| 00:00.54 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 00:01.21 | Nivex | I tried emailing voicepulse but that probably went into the abyss. I'll try calling them tomorrow if I don't get anywhere futzing with it tonight. |
| 00:02.35 | Nivex | was just hoping for "Oh yeah, I had the same problem and they gave me this password: ******" |
| 00:04.15 | WIMPy | Maybe it's calculated from the serial no? |
| 00:04.18 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 00:04.25 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:09.41 | slav3_kitten | Nivex, tried http://www.question-defense.com/2009/07/26/how-to-factory-reset-a-sipura-spa-2000-to-default-settings |
| 00:12.45 | Nivex | slav3_kitten: that is pretty much the same procedure BenC[UK]'s post outlined. I do not get a "Press 1" or "Press 12" to confirm. I am prompted for a password. |
| 00:13.33 | slav3_kitten | it's a different reset code |
| 00:14.09 | slav3_kitten | thus i thought it may succeed |
| 00:14.22 | Nivex | slav3_kitten: right. 877778 is a user level reset. 73738 is a full factory. the 877778 works but I still can't get into the admin level menus |
| 00:14.49 | Nivex | (for reference 877778 is USRRST and 73738 is RESET) |
| 00:14.51 | slav3_kitten | it seems like you'd want a full factory reset instead of a user level |
| 00:15.12 | Nivex | slav3_kitten: that is correct, but full factory prompts me for a password that I do not have. |
| 00:15.34 | slav3_kitten | i assume you've tried the typical ones? |
| 00:15.42 | Nivex | aye |
| 00:16.08 | Nivex | does know how to use Google |
| 00:16.20 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 00:16.57 | slav3_kitten | never said you didn't, different eyes have different views of the same problem |
| 00:19.12 | Nivex | I even came across a procedure for flashing a stock firmware onto it, but it took advantage of a security gap that was later patched |
| 00:19.33 | slav3_kitten | oh apparently cisco now owns them |
| 00:19.43 | Nivex | so basically I need a tiny bit of non-publicly available knowledge, which I was hoping someone in here might have through experience |
| 00:20.10 | slav3_kitten | nah but since you have to deal with cisco i'll refrain from laughing at you |
| 00:20.34 | slav3_kitten | < just spent days fighting cisco products |
| 00:21.49 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:23.27 | WIMPy | Have you searced for a serial port? |
| 00:25.29 | *** join/#asterisk bmg505 (~leon@196-209-120-6.dynamic.isadsl.co.za) |
| 00:30.55 | Nivex | WIMPy: only thing I've found looking at the board is a two pin unsoldered pad. likely an unused jumper of some kind |
| 00:31.06 | Nivex | could be reset. could also be self-destruct :) |
| 00:32.02 | WIMPy | Nothing with at least 3 pads? |
| 00:41.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:41.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:50.54 | paulc | Nivex: Could you sniff the network traffic after it boots up - maybe see if it hits a provisioning server to refresh its config.. (then spoof that locally maybe?) |
| 00:51.58 | *** join/#asterisk bmg505 (~leon@196-209-120-6.dynamic.isadsl.co.za) |
| 00:57.25 | Nivex | paulc: yep, did a packet capture on it already. it's not trying to provision. just SIP register as "cancelled@uaswundef.voicepulse.com" |
| 01:02.59 | p3nguin | moo |
| 01:03.06 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 01:03.24 | n8ideas | can anyone give me some hints on Asterisk ODBC configuration |
| 01:03.40 | n8ideas | it seems like we have frequent issues with blocking on long database lookups / writes |
| 01:03.47 | n8ideas | enough to cause peers to go unreachable |
| 01:03.54 | n8ideas | or the SIP server to stop responding to options messages |
| 01:04.11 | n8ideas | this is just with a few hundred peers and a local DB server |
| 01:06.26 | jpsharp | lag on the database server? |
| 01:09.16 | paulc | Nivex: Hmm.. interesting register line ;-) Cancelled account got pushed via provisioning, then it gave up trying to reprovision ever again? |
| 01:09.28 | Nivex | paulc: apparently |
| 01:10.31 | n8ideas | there are issues with lag on the DB server |
| 01:10.35 | n8ideas | or there ostensibly could be |
| 01:10.56 | n8ideas | which wouldn't be a huge issue, except that it blocks literally ALL SIP activity on the box |
| 01:11.18 | jpsharp | Yes it does, unfortunately. |
| 01:12.10 | n8ideas | that's an extremely poor design |
| 01:13.08 | n8ideas | are there steps I can take to mediate this? |
| 01:16.16 | jpsharp | make sure you have rtcachefriends=yes in sip.conf |
| 01:16.40 | n8ideas | yeah.. that's all good, and I do |
| 01:16.42 | n8ideas | but I qualify peers |
| 01:16.52 | n8ideas | and they will miss qualification intervals on DB reads that take a long time |
| 01:17.29 | n8ideas | I have literally seen options messages on the LAN go from 20 ms to 2000 ms |
| 01:17.35 | n8ideas | and under any load, it now happens multiple times / day |
| 01:19.40 | jpsharp | No suggestion, then, other than kicking your DBA/DB Server in the pants. |
| 01:19.51 | *** join/#asterisk dgeary2 (~david@49.176.65.82) |
| 01:19.51 | n8ideas | Yeah... I wish I had that option |
| 01:20.02 | n8ideas | it seems like it's unavoidable to have DB latency at some points |
| 01:20.17 | n8ideas | but you can't have a delayed CDR write take down hundreds of peers |
| 01:21.08 | jpsharp | unless you can some how replicate your SIP peers table to an on-server database. |
| 01:21.14 | n8ideas | we do that already |
| 01:21.28 | n8ideas | but that doesn't eliminate the possibility of a blocking call doing the same thing |
| 01:21.33 | jpsharp | True. |
| 01:21.49 | n8ideas | has anyone messed with the connection pooling? |
| 01:21.52 | n8ideas | or tried to scale test it? |
| 01:22.10 | n8ideas | I have no idea what those options would do, or what the advantage/disadvantage may be |
| 01:22.15 | n8ideas | very little in the way of documentation on this either |
| 01:22.25 | jpsharp | I did a little bit when I wrote the mysql CDR module, but that was years ago. |
| 01:22.39 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 01:22.48 | jpsharp | And a lot of people have had their hands in that code since then :) |
| 01:22.48 | n8ideas | like should share_connection be on or off? |
| 01:22.54 | n8ideas | haha.. I can imagine |
| 01:23.06 | n8ideas | and would a limit => of something higher work better |
| 01:23.10 | n8ideas | would that avoid blocking calls? |
| 01:23.35 | jpsharp | Up until the point where you start running out of sockets or filedescriptors on the server. |
| 01:24.12 | n8ideas | or would the multi-connection model eliminate the possibility? |
| 01:24.19 | n8ideas | but I can control against that, to some extent |
| 01:24.23 | n8ideas | we can make mysql prune inactive connections |
| 01:24.34 | jpsharp | Honestly, I don't know. I'm not much of a database tuner. |
| 01:24.40 | n8ideas | and mysql scales to a few dozen concurrent writes easily, which is more than what I need |
| 01:24.47 | n8ideas | yeah... I don't know either, which is what's driving me crazy |
| 01:25.11 | n8ideas | it's not the easiest thing in the world to test either |
| 01:30.03 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 01:30.18 | *** join/#asterisk dgeary2 (~david@49.176.65.82) |
| 01:46.45 | gusto | heh |
| 01:46.55 | gusto | asterisk 11 is out i ve read about it |
| 01:46.57 | gusto | hmm |
| 01:46.58 | gusto | interesting |
| 01:48.10 | p3nguin | Doesn't mean you need to switch to it tonight. |
| 01:48.25 | WIMPy | gusto: There's already the 2nd version. |
| 01:48.43 | p3nguin | First bug fix was released today, I guess. |
| 01:49.15 | p3nguin | I'll probably start using it in a year. |
| 01:50.07 | *** join/#asterisk droemel (~droemel@p4FCACF6A.dip.t-dialin.net) |
| 01:55.58 | nobodyshome | p3nguin: hows it going man? |
| 01:56.09 | p3nguin | meh |
| 01:56.49 | ddijibb | just meh eh |
| 01:56.53 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:56.56 | ddijibb | i was in a conf with Seri yesteray |
| 01:57.02 | ddijibb | catchin up |
| 01:57.02 | p3nguin | That's the long and short of it, yes. |
| 01:57.06 | ddijibb | he says you seem to be busy |
| 01:57.20 | ddijibb | me2 im still in the same banana boat |
| 01:57.53 | ddijibb | did you ever get a pcie mobo for that card? |
| 01:58.05 | SeRi | p3nguin: how are you? |
| 01:58.18 | p3nguin | I sure didn't. I have the card in my desk drawer, waiting for something to try it in. |
| 01:58.22 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-taslwcepupfdqwhx) |
| 01:58.27 | ddijibb | speak of el chupacabra |
| 01:58.29 | SeRi | ddijibb: waz up |
| 01:58.49 | ddijibb | someone stole my tag cuz im a tard and didnt reg it |
| 01:59.07 | ddijibb | and i have been listening to the policia scanner all day through my conf's MOH |
| 01:59.25 | p3nguin | It was you. |
| 01:59.54 | ddijibb | p3nguin: you know my domain right? |
| 02:00.15 | ddijibb | uri is 2336@ |
| 02:01.02 | p3nguin | I'm not sure, but I know that you registered the nick 1 year, 12 weeks, 6 days, 23 hours, and 45 minutes ago. |
| 02:01.50 | Nivex | paulc: ah ha! It does try to provision after being up for awhile. It's hitting sipura-profiles.voicepulse.com over HTTPS, so I can't see the GET string thoguh |
| 02:02.58 | ddijibb | p3nguin: jump in |
| 02:05.45 | SeRi | p3nguin: is been a while. |
| 02:06.08 | SeRi | there you go :) |
| 02:06.10 | dijib | im back |
| 02:06.12 | dijib | from the dead |
| 02:06.13 | p3nguin | There you went. |
| 02:06.15 | SeRi | ghosted |
| 02:06.16 | dijib | BRAINS!! |
| 02:07.01 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:07.02 | p3nguin | Don't be afraid of ghosts. |
| 02:07.17 | SeRi | lol! |
| 02:07.20 | dijib | y u affraid of conferences |
| 02:07.22 | dijib | ??? |
| 02:07.52 | carrar | conferences are a great time to pull out the laptop and do something else!! |
| 02:08.31 | carrar | or pull out your iphone and tweet about about what you had for lunch |
| 02:08.40 | carrar | -about |
| 02:08.57 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-nfdwmaqfadwrtvby) |
| 02:09.01 | p3nguin | I don't use an iPhone and I don't use twitter, so I'd have to do something else. |
| 02:09.11 | SeRi | +1 |
| 02:09.34 | SeRi | p3nguin: you saw what happen to CC? |
| 02:09.40 | dijib | p3nguin: when why are you not comming to the URI i sent you |
| 02:09.45 | dijib | google plus is just not enough |
| 02:09.50 | dijib | and your never on son |
| 02:10.01 | p3nguin | I only know they took a hit from the storm. |
| 02:10.15 | p3nguin | My never on son? |
| 02:10.21 | p3nguin | I don't have a never on son. |
| 02:10.36 | p3nguin | As a matter of fact, I don't have a son at all. |
| 02:10.38 | SeRi | They also got DDoS for 3 weeks straight... |
| 02:10.43 | dijib | i PM'ed you the uri p3nguin |
| 02:10.43 | SeRi | p3nguin: lol |
| 02:11.01 | p3nguin | Oh. I don't really mind of they got attacked. I don't like them and I don't use them. |
| 02:11.13 | p3nguin | s/of/if/ |
| 02:11.28 | SeRi | p3nguin: yea I just thought it was funny to know that they are hosted out of an office in lower Manhatten |
| 02:11.39 | p3nguin | They suck. |
| 02:11.49 | SeRi | LOL |
| 02:11.52 | SeRi | Indeed |
| 02:12.15 | p3nguin | Did you take advantage of $10 ports and move your CC DIDs to VoIP.ms? |
| 02:12.30 | SeRi | Yes i did |
| 02:12.34 | SeRi | still on the process |
| 02:12.47 | SeRi | I only had one |
| 02:12.50 | p3nguin | oh |
| 02:12.53 | SeRi | and only used it for my bridge |
| 02:13.27 | SeRi | so the most traffic it ever saw was like twic this year |
| 02:13.42 | SeRi | s/twic/twice/ |
| 02:13.57 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:14.00 | SeRi | jelapanos |
| 02:17.22 | dijib | your working much p? |
| 02:18.06 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 02:20.58 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 02:27.42 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 02:36.33 | gusto | WIMPy: the second version of what? |
| 02:36.38 | gusto | WIMPy: when do you sleep? |
| 02:36.44 | WIMPy | gusto: Asterisk 11 |
| 02:36.56 | WIMPy | When I feel the need to do so. |
| 02:37.18 | WIMPy | And BTW: I'm sleeping quite a lot. |
| 02:37.54 | gusto | cool |
| 02:38.00 | gusto | hi p3nguin |
| 02:38.06 | gusto | p3nguin: how are you? |
| 02:39.47 | dijib | p3nguin is affraid of facebook |
| 02:40.05 | p3nguin | I'm better now that I got my sushi. |
| 02:40.09 | gusto | did quit facebook some weeks ago |
| 02:41.06 | dijib | SUPER SISHI! |
| 02:41.13 | dijib | su |
| 02:41.32 | SeRi | shishi as an chichis? |
| 02:41.35 | p3nguin | sishisu? |
| 02:45.28 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 02:46.28 | WIMPy | wonders what to break tonight... |
| 02:46.51 | SeRi | http://doc.astlinux.org/userdoc:board_jetway_nf9hql-525 |
| 02:46.56 | gusto | WIMPy: asterisk 11 |
| 02:47.18 | WIMPy | gusto: I've been using that for serveral months already. |
| 02:49.02 | *** join/#asterisk jsjc (~Adium@189.Red-2-136-116.dynamicIP.rima-tde.net) |
| 02:53.37 | SeRi | p3nguin! |
| 02:56.18 | SeRi | p3nguin: you around? You still have your Atom system? |
| 02:56.47 | p3nguin | I've never had an Atom system. |
| 02:56.51 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:57.22 | gusto | I have some Intel Atoms |
| 02:57.25 | gusto | but not here |
| 02:57.49 | SeRi | p3nguin: for what ever reason I keep thinking you had one. |
| 02:57.53 | gusto | ah, no ... here is one ... the Acer Aspire One AOA150, but that's a 32bit one |
| 02:58.19 | gusto | Atom N279 |
| 02:58.23 | gusto | N270 |
| 02:58.31 | SeRi | I just got me one of this: http://doc.astlinux.org/userdoc:board_jetway_nf9hql-525 |
| 02:58.42 | SeRi | I hope it gets here by Friday :D |
| 02:58.46 | gusto | but i have also an atom 230, atom 330 and 525 |
| 02:58.54 | gusto | all intel mini-itx boards |
| 02:58.55 | WIMPy | Gusto: Do your card readers work? |
| 02:59.01 | gusto | WIMPy: no |
| 02:59.12 | gusto | WIMPy: ehm ... yes, when you boot that shit with cards inside |
| 02:59.18 | WIMPy | Hmm |
| 02:59.20 | gusto | however |
| 02:59.22 | gusto | i do not use them |
| 02:59.34 | dijib | p3nguin: you busy or what? |
| 03:00.18 | WIMPy | gusto: Oh, tht one is easy: Enable PCI hotplugging in your kernel. But mine haven't worked at all for quite some time. I think it might have broken evene befor 3.0. |
| 03:00.28 | WIMPy | even |
| 03:00.42 | WIMPy | Or rather move the e one word on :-) |
| 03:00.43 | p3nguin | Yeah, I'm trying to eat my sushi. |
| 03:01.11 | SeRi | nom nom nom |
| 03:01.20 | gusto | WIMPy: i am using CentOS6 with 2.6.32 on that |
| 03:02.20 | dijib | having trouble with your chop sticks? |
| 03:02.41 | gusto | WIMPy: does your wlan card work? mine did not since i bought it, so i replaced it first with a broadcom and now i have a intel inside, that works great and i have also 5 GHz HT40+ on that |
| 03:03.00 | gusto | WIMPy: i replaced the wlan cards in all my laptops with intel cards |
| 03:03.49 | WIMPy | gusto: Yes. Used to work without any issues. But with older kernels I had to rmmod the driver before sending it to sleep and than re-load it again afterwards. |
| 03:04.02 | WIMPy | definitely prefers Atheros |
| 03:04.37 | gusto | i burnt two of them |
| 03:04.41 | p3nguin | I'm not using chopsticks for California roll. |
| 03:04.43 | gusto | NEVER EVER atheros for me! |
| 03:05.16 | gusto | prefers broadcom ... then intel and then ralink |
| 03:05.19 | SeRi | p3nguin: lol |
| 03:05.44 | gusto | but i have two atheros ones in my wndr3800 ... that's an atheros AP |
| 03:10.29 | gusto | hm? |
| 03:14.37 | p3nguin | https://lh5.googleusercontent.com/-3dmsYG9KuGI/UJiAAL2EU-I/AAAAAAAAAEo/h1IxP47zItw/s507/IMAG0091.jpg |
| 03:16.42 | WIMPy | also had fish, but I definiteley prefer it cooked. |
| 03:17.11 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 03:17.16 | p3nguin | I don't eat raw fish. |
| 03:17.19 | *** join/#asterisk radic (~radic@dslb-178-007-128-094.pools.arcor-ip.net) |
| 03:17.58 | p3nguin | My wife will eat raw salmon or tuna, but I prefer my seafood to be cooked. |
| 03:18.23 | dijib | your just a vegan in denial |
| 03:18.30 | gusto | p3nguin: that fish looks like shit |
| 03:18.33 | WIMPy | That's not real sushi then. |
| 03:18.34 | dijib | looks good though |
| 03:18.37 | gusto | i do not understand how someone can eat that |
| 03:18.44 | dijib | sushimi |
| 03:18.53 | gusto | some time i send you picture of what i eat |
| 03:18.54 | dijib | i love sushi |
| 03:18.56 | p3nguin | I don't see any fish. |
| 03:18.58 | gusto | heee |
| 03:19.21 | dijib | if it smells like salmon keep on jammin', if it smells like trout, get the hell out |
| 03:19.28 | p3nguin | It's a friggin' California roll (slightly modified for my liking). |
| 03:19.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 03:19.59 | p3nguin | Cucumber gives me indigestion, so that's left out. |
| 03:21.41 | dijib | p3nguin: are you out to dinner or something with the wife or do you have some free geek time? |
| 03:22.01 | p3nguin | I'm at home. |
| 03:22.20 | gusto | so |
| 03:22.35 | gusto | p3nguin home alone |
| 03:22.36 | gusto | :-D |
| 03:22.49 | p3nguin | I wish. |
| 03:22.56 | p3nguin | I could get some sleep if I were home alone. |
| 03:23.02 | SeRi | lol |
| 03:23.10 | SeRi | I know how that goes |
| 03:23.10 | gusto | why not? |
| 03:26.27 | p3nguin | I don't understand the question. |
| 03:27.54 | gusto | why are you not home alone/you can not sleep, ha? |
| 03:28.22 | p3nguin | If I were home alone, I could sleep. |
| 03:28.52 | p3nguin | With people making noise and bugging me, I will have to stay awake for another couple hours. |
| 03:29.20 | SeRi | kids.... wife..... |
| 03:29.21 | dijib | well then join our conference |
| 03:29.25 | SeRi | you name it |
| 03:29.25 | dijib | dogs |
| 03:29.59 | p3nguin | I don't remember the SIP URI. |
| 03:30.10 | dijib | i sent it to u |
| 03:30.10 | gusto | you have a conference? cool |
| 03:30.18 | dijib | 2663@dijib.sytes.net |
| 03:30.35 | bchia | Anyone know where do download/how to update selinux module for asterisk? (I've got 1.7.1) and can't connect to remote cli with selinux enforcing http://pastebin.com/AfTj4fEX |
| 03:30.48 | SeRi | disable selinux |
| 03:30.52 | dijib | bchia: DISABLE |
| 03:30.54 | SeRi | selinux is satan |
| 03:31.09 | WIMPy | guesses that almost everyone in here has at least *a* conference. |
| 03:31.09 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 03:31.12 | bchia | lol - I was told today disabling selinux is evil |
| 03:31.15 | dijib | chconfig selinux off |
| 03:31.33 | gusto | be gobe satan! |
| 03:31.34 | SeRi | thats not going to work |
| 03:31.35 | SeRi | lol |
| 03:31.39 | gusto | no |
| 03:31.47 | bchia | https://twitter.com/russellbryant/status/265575500796272640 |
| 03:31.48 | gusto | thats in /etc/selinux/config |
| 03:31.56 | gusto | and then set ruleset to disabled |
| 03:31.57 | SeRi | gusto: you win |
| 03:32.12 | dijib | gusto jump in that sip uri |
| 03:32.24 | dijib | u too p3nguin |
| 03:32.57 | gusto | dijib: eh, does not work for me yet |
| 03:33.19 | gusto | seems i have to put in an extension |
| 03:33.25 | bchia | If there's a new version of the policy I'd much rather grab that and leave selinux enforcing |
| 03:33.55 | SeRi | dijib: is hosting a bchia You have to offer your first born blod and one eye |
| 03:34.13 | SeRi | and it will appear fuck |
| 03:34.16 | SeRi | typos |
| 03:34.32 | bchia | that is how I feel sometimes while troubleshooting! |
| 03:35.06 | WIMPy | Yes, I like it when trouble shooting gets you int more trouble. |
| 03:35.14 | WIMPy | s/int/in to/ |
| 03:35.16 | SeRi | WIMPy: LMAO |
| 03:35.59 | WIMPy | That's why I hate myself for having started with Asterisk. |
| 03:51.40 | p3nguin | seri: http://xkcd.com/386/ |
| 03:53.41 | SeRi | fucking awesome |
| 03:54.54 | gusto | somehow does the URI 2663@dijib.sytes.net not work |
| 03:55.15 | p3nguin | It works for the rest of us. |
| 03:55.20 | gusto | asterisk says to me that he is locally bridging it with my phone and then it hangs up |
| 03:55.48 | gusto | i checked the SIP debug, there is everything OK |
| 03:56.01 | p3nguin | How are you making the call? |
| 03:56.04 | gusto | so the other side hangs up on me |
| 03:56.18 | gusto | i put in an extension and then called that |
| 03:56.58 | p3nguin | Here is an example to make the call from the asterisk CLI: |
| 03:57.01 | p3nguin | channel originate SIP/2663@dijib.sytes.net extension 216@phones |
| 03:57.01 | SeRi | gusto: http://www.youtube.com/watch?v=QbzQRft7zGA |
| 03:57.01 | gusto | Dial(SIP/dijib.sytes.net/2663) |
| 03:57.20 | p3nguin | If your phone is extension 216 in context 'phones' that will work for you. |
| 03:57.43 | gusto | i ll try that |
| 03:57.45 | p3nguin | SIP/dijib.sytes.net/2663 needs to be SIP/2663@dijib.sytes.net |
| 03:57.50 | SeRi | SIP/2663@emintech.sytes.net |
| 03:57.53 | p3nguin | But you can just originate it quicker. |
| 03:58.01 | SeRi | p3nguin: beat me to it |
| 03:58.03 | SeRi | :) |
| 03:58.10 | gusto | i tried SIP/2663@dijib.sytes.net as well, that does the same |
| 03:59.38 | gusto | p3nguin: i do not have the 'originate' function in channel |
| 04:00.24 | WIMPy | gusto: Then you should have upgraded some time in the last years. |
| 04:02.14 | dijib | http://pastebin.com/M13gaiXb |
| 04:02.59 | p3nguin | gusto: He has a broken config. |
| 04:03.59 | gusto | p3nguin: who? |
| 04:04.10 | p3nguin | gusto: Are you using alaw? |
| 04:04.37 | dijib | [Nov 5 22:59:51] WARNING[7969]: chan_sip.c:6621 sip_write: Asked to transmit frame type ulaw, while native formats is (alaw) read/write = ulaw/ulaw |
| 04:05.01 | p3nguin | Someone is using alaw and the other is using ulaw. |
| 04:05.03 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 04:05.14 | gusto | p3nguin: of course |
| 04:05.29 | gusto | no, that IP that's me |
| 04:06.08 | gusto | ok, so it was about the codecs |
| 04:06.13 | gusto | i already thaught that |
| 04:06.26 | SeRi | he is allowing alaw in a sec |
| 04:06.41 | dijib | gusto: try again |
| 04:06.46 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:06.49 | dijib | we think your using alaw |
| 04:07.34 | p3nguin | If dijib will add alaw, save the change, and sip reload, it should be okay. |
| 04:08.34 | gusto | well |
| 04:08.51 | gusto | i have ulaw and alaw inside, so i have both, no problem at my side |
| 04:09.08 | gusto | <PROTECTED> |
| 04:09.08 | gusto | <PROTECTED> |
| 04:09.14 | WIMPy | That looks more like a missing translation path to me. |
| 04:09.28 | p3nguin | How does one get a translation path? |
| 04:09.35 | gusto | yes, that could be the case |
| 04:10.01 | gusto | codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 |
| 04:10.06 | gusto | no, that one is in as well |
| 04:10.33 | gusto | Module Description Use Count |
| 04:10.36 | gusto | codec_ulaw.so mu-Law Coder/Decoder 0 |
| 04:10.39 | gusto | codec_gsm.so GSM Coder/Decoder 0 |
| 04:10.42 | gusto | codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 |
| 04:10.45 | gusto | codec_alaw.so A-law Coder/Decoder 0 |
| 04:10.48 | gusto | codec_g722.so ITU G.722-64kbps G722 Transcoder 0 |
| 04:10.50 | WIMPy | p3nguin: core show translation path <codec> |
| 04:10.50 | gusto | codec_g726.so ITU G.726-32kbps G726 Transcoder 0 |
| 04:10.54 | gusto | 6 modules loaded |
| 04:11.05 | WIMPy | gusto: Don't flood the channel. |
| 04:12.31 | gusto | http://pastebin.com/mxxYMfEf |
| 04:12.35 | gusto | here you are |
| 04:14.47 | gusto | [Nov 5 22:59:51] WARNING[7969]: channel.c:5251 set_format: Unable to find a codec translation path from (alaw) to (ulaw) |
| 04:14.51 | gusto | is on his side |
| 04:14.52 | gusto | not on mine |
| 04:15.15 | gusto | so we were searching at the wrong end :-D |
| 04:15.25 | SeRi | mc gusto try again |
| 04:15.27 | ChannelZ | It would be funny if it were actually zlaw |
| 04:15.29 | p3nguin | I'm glad I don't have those problems. |
| 04:15.29 | dijib | gusto: retry one more time ive added a bunch of sound files |
| 04:15.48 | gusto | however, as long as my configuration does permit both, also alaw and ulaw, he should pick the other one if the first does not work for both |
| 04:16.10 | WIMPy | Looks like we're back to the fishy bits. |
| 04:16.17 | gusto | well |
| 04:16.17 | WIMPy | gives it a try... |
| 04:16.19 | gusto | it works |
| 04:16.23 | p3nguin | http://pastebin.com/z2yHwLjw |
| 04:18.01 | gusto | but you have still an asymetric line :-D |
| 04:18.17 | SeRi | *assymetric* |
| 04:18.22 | gusto | i bet you can hear my sound but i can not hear yours |
| 04:18.31 | SeRi | gusto we cant hear you |
| 04:18.54 | ChannelZ | all I hear are farting noises. |
| 04:18.56 | SeRi | thinks WIMPy needs to jump in |
| 04:19.10 | gusto | http://pastebin.com/16ZfRhDQ |
| 04:19.29 | gusto | SeRi: that's because i was not talking :-D |
| 04:19.50 | SeRi | lol |
| 04:20.05 | gusto | but my data was sent out |
| 04:20.15 | gusto | but nothing came in |
| 04:20.37 | gusto | normaly when you make a call you have in RTP sending and recieving packets, there it was only sending |
| 04:20.39 | SeRi | can you hear us? |
| 04:21.04 | gusto | like i said, i was only sending according to the log and of course i did not hear anything |
| 04:21.25 | gusto | but i did hear that sound file |
| 04:21.39 | gusto | that ... there are three participants in conference or sth like that |
| 04:21.44 | gusto | so i hear that |
| 04:22.15 | p3nguin | channel request hangup ... |
| 04:22.24 | gusto | yes, i hung up |
| 04:23.34 | WIMPy | can listen |
| 04:24.19 | WIMPy | But the audio is a little choppy. |
| 04:25.34 | dijib | its because im recompiling asterisk |
| 04:28.12 | slav3_kitten | ok anyone remember the company that is selling licenses to upgrade sip firmware on cisco phones? |
| 04:28.30 | slav3_kitten | think it was 20 bucks a model number |
| 04:29.38 | SeRi | slav3_kitten: piratebay.xxx |
| 04:30.34 | slav3_kitten | uh no |
| 04:30.38 | SeRi | LOL |
| 04:30.59 | slav3_kitten | they were selling smartnet contracts of some form |
| 04:36.36 | dijib | gusto: ONE MORE TRY, i think ive fixed it |
| 04:36.53 | gusto | dijib: oh |
| 04:37.05 | dijib | now i thin i was missing codecs |
| 04:38.21 | slav3_kitten | dijib, that'd do it |
| 04:38.40 | *** join/#asterisk Brokedcomputer (~ce@184.71.199.202) |
| 04:38.50 | dijib | gusto: are you going to try? |
| 04:39.08 | SeRi | slav3_kitten: join 2663@dijib.sytes.net |
| 04:39.28 | slav3_kitten | SeRi, how? |
| 04:39.59 | SeRi | slav3_kitten: SIP/2663@dijib.sytes.net |
| 04:40.38 | slav3_kitten | SeRi, don't i need a public ip and ports forwarded to do that? |
| 04:40.50 | p3nguin | hmm? |
| 04:40.51 | dijib | no its called a sip uri |
| 04:40.57 | p3nguin | It's a URI. |
| 04:40.57 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 04:40.59 | SeRi | slav3_kitten: no. |
| 04:40.59 | dijib | enter it into softphones |
| 04:41.01 | p3nguin | Just call it. |
| 04:41.06 | dijib | or mae ne extension in asterisk |
| 04:41.14 | p3nguin | or originate it from the CLI. |
| 04:42.07 | slav3_kitten | hmm |
| 04:42.23 | gusto | i am going to sleep a little |
| 04:42.41 | gusto | maybe someday i will set up a conference server, i need it anyway |
| 04:42.56 | gusto | first of all because of local phones here |
| 04:43.17 | gusto | so that more ppl can take the phone and talk INTO the running call(conversation) |
| 04:43.44 | p3nguin | slav3_kitten: Hi. How's it going? |
| 04:43.45 | gusto | i wanted to do it long time ago but somehow i did not come to it |
| 04:43.59 | slav3_kitten | p3nguin, it's going? |
| 04:44.00 | gusto | and that dijib bridge seems to be the old one |
| 04:44.16 | gusto | i would like to try that chan_bridge thing |
| 04:44.52 | ChannelZ | OT: Anyone running Mac OS 10.8.1 who can send me a kernel module from it? |
| 04:44.55 | p3nguin | He's using ConfBridge. |
| 04:45.05 | gusto | well |
| 04:45.25 | gusto | i thaught it's the old one where you need that DAHDI drivers for |
| 04:45.30 | gusto | you know what i mean |
| 04:45.35 | gusto | that old conferencing module |
| 04:45.38 | ChannelZ | MeetMe is the old one |
| 04:45.41 | jpsharp | That's MeetMe. |
| 04:45.41 | gusto | yes |
| 04:45.51 | ChannelZ | ConfBridge is the new hotness |
| 04:45.53 | p3nguin | I have guns. I have ammo. |
| 04:46.00 | WIMPy | gusto: And what kind of difference does it make to you when you just join a conference? |
| 04:46.05 | p3nguin | Lots of ammo, I'd like to add. |
| 04:46.06 | dijib | ssh-copy-id -p 222 -i '~/.ssh/id_rsa.pub' root@ |
| 04:46.08 | ChannelZ | So do I but what I really need is a kernel module |
| 04:46.30 | jpsharp | ConfBridge rocks. |
| 04:46.38 | *** join/#asterisk |ance|ot (~macbook@186.81.74.189) |
| 04:46.40 | SeRi | ConfBridge is a hotness |
| 04:46.46 | ChannelZ | With its cock out |
| 04:46.59 | SeRi | dijib: likes that |
| 04:47.24 | jpsharp | I dunno about cock out. I don't swing that way. |
| 04:48.01 | ChannelZ | Just a wiggle then? |
| 04:48.24 | WIMPy | LMFAO? |
| 04:48.57 | ChannelZ | Penis jokes. ha HA! |
| 04:49.17 | SeRi | p3nguin: https://danieldefense.com/rifles/mid-length/daniel-defense-m4-carbine-v5.html |
| 04:49.46 | slav3_kitten | ugh |
| 04:50.02 | slav3_kitten | not daniel defense |
| 04:50.04 | SeRi | ChannelZ: ROFL |
| 04:50.14 | SeRi | slav3_kitten: why? |
| 04:51.30 | slav3_kitten | SeRi, over priced. you could get a stripped lower an build something just as nice if not nicer with ease |
| 04:54.41 | *** join/#asterisk unicron (~unicron@the.wrong.domain.name) |
| 04:59.55 | dijib | . |
| 05:00.07 | SeRi | slav3_kitten: Not really. It is the cheapest on the line. compare to others... |
| 05:00.27 | SeRi | look at lwrc |
| 05:00.32 | SeRi | they are way high |
| 05:00.41 | slav3_kitten | lwrc is the high end of high end |
| 05:00.49 | slav3_kitten | also their piston uppers are fucking amazing |
| 05:01.21 | p3nguin | I wouldn't mind going to a piston system. |
| 05:01.26 | SeRi | so which one is it? is it to much or to little? If you can afford a lwrc than you get a DD. bottom line |
| 05:01.36 | dijib | guns :'( |
| 05:01.50 | p3nguin | I never did get around to cleaning my gas port and tube on my direct impingement rifle. |
| 05:01.55 | slav3_kitten | if i were going to build an M4 style ar i'd build a proper M4, it'd be 14" upper, SBR tax stamp, etc |
| 05:02.20 | SeRi | Thats some cash right there. Though an SBR would be nice |
| 05:02.24 | slav3_kitten | p3nguin, under normal operation you should not need to really clean the gas port or tube |
| 05:02.24 | WIMPy | is out |
| 05:02.35 | p3nguin | I've never cleaned it in 10 years. |
| 05:02.37 | ChannelZ | I have like 5 dirty guns in the safe (including the AR) I really need to get to |
| 05:02.55 | slav3_kitten | ChannelZ, my only dirty gun is my 20ga |
| 05:03.08 | ChannelZ | But enough about me, lets talk about zombies |
| 05:03.15 | p3nguin | It wasn't shooting good the last time I went out with it, and I never got around to cleaning it to see if that was the issue. |
| 05:03.15 | slav3_kitten | i'm trying to see how many rounds before it starts fucking up. so far 1k rounds |
| 05:04.00 | slav3_kitten | what was it not doing properly p3nguin? |
| 05:04.10 | slav3_kitten | FTE FTF etc? |
| 05:04.35 | p3nguin | If I remember correctly, it was ejecting the fired case, but it was not picking up a new round. |
| 05:05.10 | slav3_kitten | weak or strong ejection, and what mags were you using? |
| 05:05.10 | p3nguin | Almost like it wasn't throwing the bolt back far enough to pick up a new one. |
| 05:05.44 | p3nguin | I tried a few different mags. |
| 05:06.31 | p3nguin | If I would manually charge after firing, there was no problem picking up the new round and going fully into battery. |
| 05:06.51 | p3nguin | Basically it was a single-shot rifle with auto eject. :/ |
| 05:06.59 | ChannelZ | sounds like weak gas |
| 05:07.09 | slav3_kitten | lol, did you lube it prior to your range outing? |
| 05:07.12 | ChannelZ | <insert taco bell joke> |
| 05:07.27 | SeRi | lol |
| 05:07.35 | p3nguin | I sprayed it with some RemLube. |
| 05:07.46 | p3nguin | (not rim lube) |
| 05:08.39 | slav3_kitten | could be excessive drag on the bolt carrier slowing it down, could be an issue with your buffer, if they were aluminum GI mags the feed lips can get bashed around |
| 05:08.40 | SeRi | ok g/n guys |
| 05:08.43 | unicron | i want a AA-12 |
| 05:09.27 | p3nguin | I was using various steel mags, including my trusty old Colt. |
| 05:09.32 | slav3_kitten | unicron, are you american, own a fire company that sells to law enforcement, etc? |
| 05:09.51 | slav3_kitten | firearm* |
| 05:09.59 | unicron | i'm american and no |
| 05:10.20 | slav3_kitten | because the AA-12 is open bolt an thus a machine gun and subject to the gun control act of 1968 |
| 05:10.44 | slav3_kitten | which basically says you can buy a MG so long as you pay the tax stamp an it was manufactured prior to 1968 |
| 05:11.00 | slav3_kitten | AA-12 is way post 68 |
| 05:11.12 | unicron | but it's soooo sweet |
| 05:11.40 | slav3_kitten | lol |
| 05:11.54 | slav3_kitten | know what's way cooler? winchester 97T |
| 05:12.05 | SeRi | slav3_kitten: That probably applies to your state. I can get a stamp, llc/trust,class 3 lic and I can own any auto fire arm |
| 05:12.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-speytopbsceryvpd) |
| 05:12.42 | slav3_kitten | SeRi, the GCA of 1968 is a federal law |
| 05:13.23 | unicron | i'm a proud virginian |
| 05:13.28 | unicron | http://washingtonexaminer.com/virginia-tops-nation-in-machine-gun-ownership-maryland-is-5th/article/2511444 |
| 05:13.43 | slav3_kitten | that's why the DIAS's have been drying up an old UZI's go for 10k in full auto. they are all pre 1968, you need to be a special exception to the GCA such as an active mil/le supplier with demonstration weapons |
| 05:14.57 | SeRi | you must have it wrong. I been looking in to buying witha Class 3 and i can walk out with a brand new Auto. |
| 05:15.39 | SeRi | something is not right... I need to research morew |
| 05:15.46 | slav3_kitten | SeRi, i've been looking into it heavily. you can walk out with a auto so long as the proper parts are pre 68 |
| 05:16.17 | SeRi | HU? |
| 05:16.20 | slav3_kitten | like i can go buy a auto AR with a pre 68 drop in auto sear. everything else is new but the part that makes it auto is pre68 |
| 05:16.33 | SeRi | design? |
| 05:16.38 | SeRi | is that what you mean? |
| 05:16.48 | slav3_kitten | huh? |
| 05:17.11 | p3nguin | What we have here is a failure to communicate. |
| 05:17.37 | slav3_kitten | http://en.wikipedia.org/wiki/Firearm_Owners_Protection_Act |
| 05:18.27 | p3nguin | I doubt the store down the road with a shop on the back which manufactures the parts is going to sell me a firearm with parts from the 1960s. |
| 05:18.53 | slav3_kitten | p3nguin, then you sir would be wrong. |
| 05:19.06 | slav3_kitten | this is for fully automatic firearms |
| 05:20.08 | p3nguin | So you think the parts are from the 1960s. |
| 05:20.42 | slav3_kitten | http://appliedballisticssystems.com/Class_III_Info.html |
| 05:21.14 | slav3_kitten | p3nguin, it depends on the firearm. you can make a legal fully automatic AR-15 using a pre 1968 drop in auto sear |
| 05:21.19 | slav3_kitten | that'd be the only 60's vintage part |
| 05:21.25 | slav3_kitten | and they can be weld repaired |
| 05:21.34 | p3nguin | But I'm talking about new parts. |
| 05:21.51 | p3nguin | And seri was talking about having the proper C3 license. |
| 05:22.04 | SeRi | ^^ |
| 05:22.14 | slav3_kitten | in a nutshell if you want a machinegun legally as a private citizen. you are going to get a gun with pre 1968 parts |
| 05:23.03 | slav3_kitten | if you start a business that sells things like the AA-12 to the LE/Mil you have a special exemption for having possession of post 1968 machine guns |
| 05:24.02 | SeRi | I give up. |
| 05:24.06 | SeRi | :) |
| 05:25.39 | slav3_kitten | SeRi, ok here's how it works. you want to own a machine gun legally it will be manufactured and registered prior to 1968. |
| 05:26.07 | SeRi | slav3_kitten: I trust you :) |
| 05:26.23 | unicron | http://www.calguns.net/calgunforum/archive/index.php/t-582987.html |
| 05:26.27 | p3nguin | If he gets his class 3 federal license, why should he care about the whole 1968 crap? |
| 05:26.29 | unicron | woohoo! |
| 05:26.35 | unicron | but way out of my price range :( |
| 05:26.49 | slav3_kitten | if you want a dealer sample.. then, you need a gun that existed as a dealer sample prior to 1986, and a proper FFL, and a letter from a LE agency that syas they'd like a demonstration. |
| 05:28.05 | p3nguin | I repeat: |
| 05:28.06 | p3nguin | If he gets his class 3 federal license, why should he care about the whole 1968 crap? |
| 05:29.15 | slav3_kitten | if you want any other new MG. You'd need the full manufacturer's license, and then could get one as a sample. also you should care about the 1968 crap because the ATF cares greatly about it. |
| 05:29.45 | p3nguin | If he was born in 1968, would that help? |
| 05:30.54 | slav3_kitten | no, the gun needs to be born 68 or previous... |
| 05:31.10 | SeRi | lmao. ok I am out fo realz. g/n |
| 05:31.18 | slav3_kitten | night SeRi |
| 05:31.25 | SeRi | g/n |
| 05:34.06 | unicron | btw your wiki link says the thing you are talking about was a 1986 amendment to a 1968 act, so you're a couple decades off anyway |
| 05:35.53 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:35.59 | slav3_kitten | unicron, fact o the matter is unless you're a manufacturer of machineguns you're not going to get a post 68 machinegun |
| 05:36.57 | drmessano | What year does my Asterisk version need to be from if I want to spray unknowing victims with calls from my automatic wardialer? |
| 05:37.00 | unicron | then this guy shouldn't be posting his illegal actions on youtube http://www.youtube.com/watch?v=WOoUVeyaY_8 |
| 05:37.12 | jpsharp | drmessano: Doesn't matter, you just have to mill your own IAX trigger sear. |
| 05:37.59 | slav3_kitten | for the AA-12 to be on the civilian market you'd need to convert it from fully automatic open bolt to closed bolt semi automatic. also that guy can make that video so long as a representitive of the company that manufactures the gun is there |
| 05:38.04 | drmessano | jpsharp, is that IAX2 or IAX? As you know, IAX was banned in the Asterisk 1.2 release act of 2004 |
| 05:38.36 | unicron | he was facilitating sales of those guns to civilians for $17k in july and apparently has not been arrested |
| 05:38.53 | jpsharp | Only in nations that ratified the ITU anti-terrorism-drug treaty. |
| 05:39.13 | drmessano | jpsharp, you must compile Asterisk with closed bolt switches to enable IAX and then only may you use it in the presence of Mark Spencer |
| 05:39.30 | drmessano | Unless of course he his holding a post-1968 automatic weapon |
| 05:39.38 | drmessano | In which case, you're both going to prison |
| 05:39.43 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:39.51 | jpsharp | I choose to go to a women's prison. |
| 05:39.58 | slav3_kitten | unicron, do you know anyone who got one for 17k? |
| 05:40.10 | WIMPy | jpsharp: Think what you ask for. |
| 05:40.24 | unicron | no, i just saw a forum post referencing some tweets |
| 05:41.01 | drmessano | Locked in a prison full of women. That would be hot for like 14 seconds, until you realize youre LOCKED IN A PRISON FULL OF WOMEN. |
| 05:41.17 | slav3_kitten | unicron, i can tweet anything i please an post any bs in a forum i want. doesn't mean it happened |
| 05:41.43 | drmessano | I think you're all full of crap for referencing anything on the internet, period |
| 05:41.54 | drmessano | Everyone knows that the internet is fake, mostly |
| 05:42.13 | jpsharp | Yeah, that russian bride I ordered...man, what a disappointment. |
| 05:42.17 | slav3_kitten | the auto assault 12 was designed in the 70's and the aa-12 was 2k5 so no way in hell he's selling them to civilians legally |
| 05:42.30 | slav3_kitten | it's just not happening |
| 05:43.09 | unicron | again, your year is wrong, it's 86 not 68 |
| 05:43.59 | drmessano | jpsharp, I couldn't afford a russian bride, so I got a chinese bride. Two limbs fell of the first week, and I couldn't get the seller to respond to my requests for a return. |
| 05:44.58 | drmessano | This is precisely why I am voting for Mitt Romney tomorrow. He will put a stop to that sort of shenanigans |
| 05:45.01 | jpsharp | And after you had sex with her, you wanted it again an hour later? |
| 05:45.51 | drmessano | jpsharp, The sex was terrible. She could never get my orders right |
| 05:46.11 | jpsharp | snorks soda. |
| 05:46.42 | drmessano | "Why the hell is there a dog in here... oh, damnit.. Not WITH a DOGGIE" |
| 05:46.57 | unicron | in 2008 i wrote in smedley butler. this year i shall write in bunnista, the fictional communist rabbit |
| 05:47.47 | slav3_kitten | my fault unicron i sometimes get dyslexic with numbers like that |
| 05:47.53 | slav3_kitten | to be transferrable for private ownership, it must have been made before May 19, 1986 |
| 05:48.08 | slav3_kitten | so he could be selling auto assault 12's legally for 17k |
| 05:48.09 | unicron | it's an easy mistake slav3, since it's amending a 68 law |
| 05:48.11 | drmessano | The election is rigged anyway. Expect lots of fraud, a recount, a tie, and Dec 18th for the Supreme Court to decide Romney is the winner. |
| 05:48.50 | slav3_kitten | unicron, easy mistake that i made like twice after you corrected me. thus i feel like a dumbass |
| 05:49.38 | unicron | the main thing voting accomplishes is to grant an undeserved sheen of legitimacy upon the status quo. if nobody voted, nobody could claim to have a mandate |
| 05:49.40 | jpsharp | drmessano: And then 3 days later, the Mayan calendar explodes. |
| 05:50.10 | drmessano | I'm not surprised the law was amended in 1986. After Commando, Die Hard, and the first Lethal Weapon, you just couldn't trust anyone with that kind of firepower |
| 05:50.39 | drmessano | You couldn't just have Nakatomi plaza taken again.. |
| 05:50.54 | drmessano | Nevermind the loss of Agent Johnson and Special Agent Johnson (no relation) |
| 05:51.18 | jpsharp | So when the world ends on Dec 21, is it happening at 0000GMT? Or 0000 in whatever timezone would be appropriate for the Mayan capital city? |
| 05:51.58 | drmessano | jpsharp, depends on the TZ setting on your machine |
| 05:52.03 | WIMPy | 00:00 local time to avoid DoS. |
| 05:52.25 | jpsharp | Americas/New_York, so I guess I have to wait till New York gets it again. |
| 05:53.09 | drmessano | Maybe it's whatever TZ the Flying Spaghetti Monster has his Windows 98 box set to..? |
| 05:53.26 | jpsharp | sends powerful mindvibes to the chucklhead who decided that Americas/New_York was needed instead of EST5EDT. |
| 05:53.57 | jpsharp | May the fleas of a thousand and one camels infest your pubes. |
| 05:55.02 | drmessano | You can still add that to your autoexec.bat file on any Linux box |
| 05:55.07 | drmessano | Not that it does anything |
| 05:55.46 | jpsharp | Probably the same nozzle who coined MiB and KiB. |
| 05:58.08 | drmessano | My hobby: Referring to GB as GHZ and KB as KHZ |
| 05:58.37 | drmessano | 4GHZ of RAM is awesome |
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| 06:15.01 | [TK]D-Fender | C'mon baby ghz so good.... |
| 06:15.43 | slav3_kitten | holy crap that sucked |
| 06:15.46 | slav3_kitten | pants |
| 06:16.17 | WIMPy | have you been ciscod again? |
| 06:17.16 | slav3_kitten | decided to do a shit load of moving late at night |
| 06:17.40 | jpsharp | I used to do that too piss off my downstairs neighbor. |
| 06:17.43 | WIMPy | Moving what? |
| 06:17.52 | slav3_kitten | my body |
| 06:18.00 | slav3_kitten | i have nerve problems |
| 06:18.27 | slav3_kitten | i also loath pain medication |
| 06:23.25 | carrar | drmessano, and gigabyte cycle ram is even better |
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| 06:47.15 | slav3_kitten | huh |
| 06:55.38 | ChannelZ | meh |
| 06:59.01 | jpsharp | hem |
| 06:59.47 | WIMPy | hme |
| 07:02.06 | slav3_kitten | wonder why one of my connects is showing as a short on my lan roamer |
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| 07:27.38 | bulkorok | hi |
| 07:31.30 | jpsharp | hi |
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| 07:31.51 | Sheeplet | lo all |
| 07:34.56 | jpsharp | hi |
| 07:43.48 | AliRezaTaleghani | I am looking for a way to log the channels hangup cause... |
| 07:44.22 | AliRezaTaleghani | I mean after the Calls has been droped! i need to see the side with has been hanged the call up |
| 07:44.24 | AliRezaTaleghani | :-/ |
| 07:44.49 | AliRezaTaleghani | NoOp(Hangupcause: ${HANGUPCAUSE}) |
| 07:44.50 | bulkorok | check cdr_custom.conf |
| 07:45.12 | AliRezaTaleghani | but this is nut enoght for me... cos i see a respawn on channel and nothing more going on... |
| 07:45.47 | bulkorok | at first you can make sip debug to check how sends the BYE |
| 07:45.53 | bulkorok | who |
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| 07:47.06 | AliRezaTaleghani | bulkorok: I need this to be always in my logs |
| 07:47.11 | AliRezaTaleghani | not just for one call |
| 07:47.15 | bulkorok | mmh |
| 07:48.10 | AliRezaTaleghani | how can i for the call to go untill last priority in dialplan, even after hangup? |
| 07:49.28 | bulkorok | the h-extension will be issued after hangup... so you can use Log(NOTICE.${VARIABLE_YOU_WANT}) to /var/log/asterisk/messages |
| 07:49.45 | bulkorok | exten => h,1,Log(...) |
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| 07:59.36 | AliRezaTaleghani | bulkorok: tnx, I will check it r-now |
| 07:59.38 | AliRezaTaleghani | :-) |
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| 10:12.13 | lantizia | Hey is a value like 'jitterbuffers=4' in sip.conf even valid anymore? all the other settings seem to be jbSOMETHING |
| 10:12.59 | lantizia | only reference I've found to the 'jitterbuffers' value (plural) is in relation to zapata |
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| 10:24.51 | lantizia | i'm going to assume it's bollocks and leave it out - can't find any official docs about it other than in the old zapata conf |
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| 10:46.27 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-052-093.pools.arcor-ip.net) |
| 10:46.45 | ghost75 | is it not possible to call a macro from h context? |
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| 10:48.45 | bulkorok | macro is deprecated in 11... use gosub |
| 10:48.56 | Kako | Hi, is there a way to change the accontcode of a channel using AMI? |
| 10:49.20 | ghost75 | i have 1.6 |
| 10:49.52 | *** join/#asterisk Rumbles (~stock@77.107.183.144) |
| 10:51.18 | ghost75 | first line in macro: exten => s,1,NoOp(Executing macro-throttleoff) |
| 10:51.37 | ghost75 | maybe s not works when already in h ? |
| 10:52.13 | kaldemar | macros were deprecated in 1.6.0 already. |
| 10:54.33 | kaldemar | and yes, macros can be executed in the hangup extension. |
| 10:55.53 | ghost75 | so why it doesnt work in h context |
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| 10:58.22 | kaldemar | ghost75: it does. |
| 10:58.37 | kaldemar | ghost75: show what you have. |
| 10:59.07 | kaldemar | one can't possibly tell you why your dialplan does not do what you expect it to if you don't show it. |
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| 11:02.04 | ghost75 | kaldemar: http://pastebin.com/cFWG3Kwe |
| 11:02.32 | kaldemar | and CLI output of a call? |
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| 11:03.23 | ghost75 | i was look at was only saying executing h but nothing else |
| 11:05.47 | ghost75 | http://pastebin.com/aEkSt11V |
| 11:06.12 | ghost75 | however the other macro gets executed |
| 11:06.18 | kaldemar | that does not match your dialplan. |
| 11:06.31 | ghost75 | why |
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| 11:06.53 | kaldemar | the context is different, the macro is not executed... |
| 11:07.28 | ghost75 | dont assume i post my phone number here |
| 11:07.40 | kaldemar | i don't give a shit about your phone number. |
| 11:07.55 | ghost75 | i put those xxxx |
| 11:08.05 | kaldemar | but what your CLI output shows does not include executing a macro. |
| 11:08.23 | ghost75 | yes |
| 11:08.23 | kaldemar | it is the output of "exten => h,1,Hangup" |
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| 11:10.56 | kaldemar | if you have a Hangup, the dialplan execution stops. show the real dialplan. |
| 11:12.09 | ghost75 | this is all in h: http://pastebin.com/jAfeUtgy |
| 11:13.29 | kaldemar | "-- Executing [h@phones9xxxx:1] Hangup("SIP/10-00000000", "")" <-- that clearly says context "phones9xxxx", extension "h", priority "1" and application "Hangup" |
| 11:13.45 | kaldemar | you are showing two different things. |
| 11:15.38 | ghost75 | http://pastebin.com/X2gsyZn1 |
| 11:15.46 | ghost75 | nothing more than that |
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| 11:17.26 | kaldemar | ghost75: include => macro-DND is to blame. don't include macros like that. |
| 11:17.44 | ghost75 | why it should call this? |
| 11:18.17 | kaldemar | when you are in [phones998780] and a hangup occurs, a hangup extension in that context is searched for. since you have included other context that have hangup extensions in them, the first included one matches and is executed. |
| 11:19.06 | kaldemar | and you even include the context to itself which makes no sense. |
| 11:19.16 | kaldemar | your hangup extension is in the wrong plave. |
| 11:19.18 | kaldemar | place |
| 11:19.55 | ghost75 | what you mean with include to itself |
| 11:21.30 | kaldemar | forget about it. misread hastily. |
| 11:22.49 | ghost75 | ok thx i changed the one line to s,n instead h,n and it works |
| 11:24.34 | ghost75 | and this can be problem if i need mulitple contexts with h |
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| 11:26.20 | kaldemar | it is a problem. |
| 11:26.32 | ghost75 | so better avoid context for h ? |
| 11:26.46 | kaldemar | "context for h" does not compute. |
| 11:27.59 | kaldemar | don't put hangup extensions in contexts that you include from others, unless you know exactly what you're doing. |
| 11:30.43 | ghost75 | in context spamtrap i use h just to send mail after call is hangup |
| 11:33.14 | kaldemar | and that will work as long as you use a goto to go to the context. when you use Goto, the so called current context will change. upon a hangup, the place to look for a hangup extension changes along with that. |
| 11:33.59 | ghost75 | ok |
| 11:34.24 | ghost75 | have to remember this h include thing |
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| 11:41.39 | scgm11 | hi, is there any working libss7 for asterisk 10? |
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| 11:51.42 | scgm11 | |
| 11:51.43 | scgm11 | hi, is there any working libss7 for asterisk 10? |
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| 12:23.56 | scgm11 | 0? |
| 12:24.03 | scgm11 | is there any working libss7 for asterisk 10? |
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| 12:28.03 | bulkorok | it's not the time |
| 12:28.28 | bulkorok | right now here are at least 2-5 people reading... |
| 12:28.52 | scgm11 | ok, maybe later |
| 12:28.57 | bulkorok | 4-5 hours... |
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| 13:21.22 | Kako | Hi, is there a way to change the accountcode of a already existing channel using AMI? |
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| 13:27.13 | BenC[UK] | <PROTECTED> |
| 13:27.15 | BenC[UK] | oops |
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| 13:28.17 | WIMPy | Kako: SetVar should do it. |
| 13:30.18 | Kako | WIMPy: Variable: CDR(AccountCode) or Variable: AccountCode or... What is the right syntax? I've tried to find out byusing Getvar, but the result ist empty. |
| 13:31.17 | WIMPy | You need to use the CDR function and you need to supply the channel. |
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| 13:32.01 | Kako | WIMPy: Variable: CDR(AccountCode) worked, thanks. |
| 13:33.31 | Kako | Another question. I've parked a call with AMI by using Action: Park. How can I UnPark it? Must I combine Originate+Bridge? This sounds unecessary complicated... |
| 13:34.10 | WIMPy | Unpark it from/to where? |
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| 13:34.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:35.48 | Kako | I've the Channel name of the parked call and whant to redirect it to an extension. Normaly the same extension as i set as timeou-channel. |
| 13:36.42 | WIMPy | One that doesn't have an open channel at that point? |
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| 13:38.09 | Kako | yes |
| 13:38.27 | WIMPy | I guess you know the answer then. |
| 13:39.09 | Kako | :-( in other words Originate and Bridge? Would it work with Transfer? |
| 13:39.46 | WIMPy | What do you want to transfer? |
| 13:40.27 | Kako | the parked chanell to an new extension that hast no channel at that point |
| 13:40.38 | Kako | (Redirect) |
| 13:41.11 | Rumbles | I know this isn't the place to ask (sorry about that) but I have asked in the #hylafax channel a few times and mailed the mailing list and I'm not getting anywhere, I was wondering if anyone here knew much about hylafax+iaxmodem+asterisk - I have got it working successfully on one server, and not on another, can anyone help at all? |
| 13:41.18 | WIMPy | That would be a piece of dialplan then. |
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| 13:41.49 | WIMPy | And yes that would work if you don't use parking but just send the calls to an extension doing MusicOnHold instead. |
| 13:42.44 | WIMPy | Thinking about it. I guess it shouldn;t matter if it's parked. |
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| 13:44.23 | WIMPy | Are you trying to make a PBX out of Asterisk? |
| 13:44.29 | Kako | hm, i was houping to find an easier way... :-( |
| 13:45.31 | Kako | BTW, I can now set the AccountCode with Setvar so that it is visible with CoreShowChannels, but it doesn't get to the CDR after the call is finished. Any idea why? |
| 13:46.04 | Kako | I set it on both channels, but nothing in the CDR |
| 13:46.07 | [TK]D-Fender | because you should be setting the function directly |
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| 13:47.21 | Kako | [TK]D-Fender: In other words "Action: Setvar + Variable: CDR(AccountCode)" is not the right way? How can I set it drectly? |
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| 13:52.08 | kaldemar | Kako: try accountcode instead of AccountCode. |
| 13:53.13 | Rumbles | If anyone can offer any help with my hylafax issue, the details can be found here: http://goo.gl/qlHg3 I appreciate any assistance :) |
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| 13:54.06 | kaldemar | Kako: disregard that, the fields are not case sensitive. |
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| 13:55.20 | WIMPy | Kako: Where are you writing the CDRs? Does the field exist at all? |
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| 13:56.26 | Kako | kaldemar: thanks, I've tried it anyway... with the predicted result :-) |
| 13:57.16 | Kako | WIMPy: yes, the field exist. When creation a call with originate the field "accountcode" is set as expected. The CDR ist saved in a mysql-db. |
| 13:58.11 | WIMPy | Maybe somethign that resets it again later? |
| 14:00.12 | Kako | hm... could be. That sound like a problem because I'm using "Gemeinschaft 3.1" and the created rules are not easy to follow and very complex. |
| 14:00.17 | *** join/#asterisk serafie (~erin@nat/digium/x-mccqrjrrcbdlaztw) |
| 14:01.00 | Kako | (the default rules of GS31) |
| 14:01.01 | WIMPy | That's what all the GUIs have in common and makes it almost impossible to support such installations. |
| 14:02.37 | *** join/#asterisk bchia (~Adium@nat/digium/x-whchirqnlvfhaxqe) |
| 14:03.06 | WIMPy | However, I'd still love to see a comparison of them. |
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| 14:05.29 | Kako | Well, I use it, but since I started to use asterisk (2-3 month ago) I've already found a couple of bugs in gs31 when trying to find out because things dont work as expected... |
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| 14:06.04 | Kako | The sourcecode is... well, it is like it is :-) |
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| 14:51.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:58.33 | Rumbles | Anyone here know much about Asterisk+hylafax+iaxmodem ? I managed to set it up on a server with SIP trunks, but can't get it to work on a server using ISDN lines |
| 15:01.42 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 15:03.19 | WIMPy | Rumbles: You should tell us as exactely as possible what your issue is to get help. |
| 15:05.45 | Rumbles | okay WIMPy, sorry, I didn't want to keep asking over and over :) Basically, when faxes come in on my server that doesn't work the iax.raw audio file I generate in my tests is silent |
| 15:05.54 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216.191.106.165) |
| 15:06.41 | Rumbles | I have put some more details in the mailing list entry here http://goo.gl/qlHg3 |
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| 15:10.07 | WIMPy | I can't access that information. |
| 15:10.43 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:10.43 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:11.11 | Rumbles | how about this |
| 15:11.11 | Rumbles | https://groups.google.com/d/topic/fa.hylafax/AxYAYJ0dcMY/discussion |
| 15:11.43 | WIMPy | Nope |
| 15:12.17 | Nivex | well that was awesome. I called VoicePulse tech support about my adapter and they gave me the unlock password. |
| 15:12.25 | mirela666 | What could be the cause of not good quality of .wav file recorded with MixMonitor??? Maybe my version of sox on the system? |
| 15:13.00 | Rumbles | you can't access that WIMPy ? |
| 15:13.35 | WIMPy | Rumbles: Nope. It tells me I need to enable JavaScript. |
| 15:14.00 | *** join/#asterisk masked (~forexampl@hpavc/masked) |
| 15:14.03 | masked | hi |
| 15:14.28 | Rumbles | right okay, 1 sec, I think it'll be elsewhere :) |
| 15:14.40 | masked | does anyone have any code snippets for configuring the confbridge module in asterisk gui? |
| 15:14.46 | slav3_kitten | in voicemail.conf do you have to supply an email address? |
| 15:15.45 | [TK]D-Fender | masked, AsteriskGUI is dead and hasn't been developed for 2 years. I highly recommend you plan a migration from it. |
| 15:16.13 | masked | really? |
| 15:16.25 | masked | version 2.0 hasn't from svn hasn't been developed in 2 years? |
| 15:17.03 | Rumbles | WIMPy, can you load this: http://comments.gmane.org/gmane.comp.telephony.fax.hylafax.user/36595 |
| 15:17.04 | [TK]D-Fender | masked, the tinyiest of mods only as a few corporate customers have required. Go to #asterisk-gui and double their population.... |
| 15:17.26 | masked | what other stand-alone gui's do i have to choose from? |
| 15:17.32 | masked | im running asterisk10 on freebsd |
| 15:17.51 | [TK]D-Fender | #freepbx\ |
| 15:17.54 | jmetro | make your own |
| 15:17.57 | WIMPy | Rumbles: Yes |
| 15:18.03 | masked | i've already had to hack it up a bit to get it to work with freebsd, i might as well just write a new html/js for confbridge |
| 15:18.05 | Rumbles | okay cool :) |
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| 15:18.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:22.03 | Rumbles | any chance you have any idea what I need to look at WIMPy ? |
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| 15:33.02 | Rumbles | watches the tumble weeds |
| 15:38.26 | WIMPy | Rumbles: Don;t you get any sound in either direction? |
| 15:42.04 | Rumbles | you do in one direction |
| 15:42.15 | Rumbles | just the iax.raw that is silent after the pickup sound |
| 15:42.53 | Rumbles | which I think is the audio file that is generated by iaxmodem, so iaxmodem is hearing nothing (I think) |
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| 15:45.36 | WIMPy | Thou would sound like an issue elsewhere. |
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| 15:51.26 | WIMPy | Have you tried to call it from a phone? Did you hear anything? |
| 15:51.32 | Rumbles | I believe it's an issue with either Hylafax or Iaxmodem, I'm leaning more towards Iaxmodem right now and trawling through their mailing lists for answers... where would you think the issue is WIMPy ? And do you think rebooting the server could help? Or, do you think it could be some missing kernal modules I could load? |
| 15:51.49 | Rumbles | If I ring with a phone it just rings at my end |
| 15:52.07 | Rumbles | I changed the destination to go to the main call center and it worked fine, 2 way transmission |
| 15:52.36 | WIMPy | So it doesn't even answer? |
| 15:52.55 | WIMPy | That would certainly explain why it doesn't get any audio. |
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| 15:53.24 | WIMPy | What does the *CLI tell you at that point? |
| 15:53.50 | Rumbles | when I say a very busy phone server, I mean it, it's so fast I can't even watch the CLI :/ |
| 15:54.01 | Rumbles | I'd have to log it then trawl through |
| 15:54.16 | WIMPy | grep |
| 15:55.01 | Rumbles | If I ring it off our office fax machine, from the speaker at this end you can hear it pick up |
| 15:55.48 | WIMPy | And do you hear the modem? |
| 15:56.11 | Rumbles | wait, I just tried ringing it and it make a high pitched squeek, then fax noise |
| 15:56.22 | Rumbles | no ringing, sorry |
| 15:56.36 | WIMPy | Not to bad, I guess. |
| 15:56.40 | WIMPy | too |
| 15:56.55 | Rumbles | well, that means that it's at least behaving as a fax machine :/ |
| 15:57.41 | Rumbles | I'm just going to compare the audio with my working server |
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| 16:03.37 | SuPrSluG | hello all, can a queue penalty go above 9? |
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| 16:17.46 | mathi | hi |
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| 16:18.04 | mathi | since I upgraded to Asterisk 11, I don't see any logs anymore in the console |
| 16:18.15 | mathi | though verbose is set to 1, and all my logs use 1 as level |
| 16:18.30 | jmetro | core set verbose 200000 |
| 16:18.42 | Nivex | it's over 9000! |
| 16:20.52 | mathi | jmetro, now I see logs, but I see that Verbose function doesn't output anything |
| 16:23.31 | mathi | please help ;( |
| 16:23.52 | WIMPy | Hmm. I also had the impression yesterday that teher still something going on with verbose levels. |
| 16:24.54 | Rumbles | WIMPy, just changed it from ISDN to SIP line, worked first time D: D: D: |
| 16:25.37 | jmetro | mathi: verbsoe should definitely output stuff. try core set debug 100000 |
| 16:25.37 | WIMPy | Rumbles: So it's your interface that isn't working? |
| 16:26.53 | Rumbles | the ISDN lines work fine for voice calls.... so not sure tbh |
| 16:27.33 | WIMPy | Rumbles: Is it about receiving or about sending faxes? |
| 16:27.58 | Rumbles | receiving, sorry :) |
| 16:28.54 | WIMPy | Ok, Don;t see what should go wrong in that direction. |
| 16:29.27 | Rumbles | mmmmmm, nor do I :/ |
| 16:31.30 | WIMPy | I just tried and it looks like there are only two sates. verbose 0 disables verbose and any other number will enable verbose 0 and 1. Higher numbers don't work. |
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| 16:32.10 | [TK]D-Fender | mathi, SHOW us |
| 16:34.13 | WIMPy | [TK]D-Fender: Try http://wimpy.yeti.dk/pastebin |
| 16:35.58 | [TK]D-Fender | hrm |
| 16:36.54 | SeRi | another day at work |
| 16:36.57 | SeRi | ugh |
| 16:37.18 | bulkorok | work for today finished... |
| 16:37.21 | bulkorok | yeah :-) |
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| 16:37.42 | WIMPy | Looks like the daemons verbose level sets the maximum level for the remote console. |
| 16:38.16 | WIMPy | i.e. looks like you should configure verbose=99 in your asterisk.conf in order to be able to use it on the remote console. |
| 16:39.00 | mjordan | WIMPy: are you using aliases at all? |
| 16:39.12 | WIMPy | no |
| 16:39.35 | mjordan | kk. To me, that's a bug. If the CLI reports back "remote console verbosity set to 9", and it isn't, well... yeah. |
| 16:40.12 | mjordan | You should be able from a remote console set the verbosity for the remote console to the desired level, and not have that impact the core verbosity (which was the point of the feature that went into 11 in the first place) |
| 16:40.30 | WIMPy | I guess, the remote consoles filter is set to 9, but the server only sends as much as it was started with. |
| 16:40.57 | WIMPy | Yes, I think so. |
| 16:41.10 | mjordan | it shouldn't. If the feature is working properly, as I understand the feature, the two consoles (core / remote) should be treated independently |
| 16:41.46 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 16:41.48 | jacekowski | hi people |
| 16:41.48 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 16:41.51 | Rico29 | hi all |
| 16:42.03 | mathi | [TK]D-Fender, sorry was on the phone, what can I show you? the code or the logs ? |
| 16:42.06 | mathi | it's a simple verbose |
| 16:42.07 | Rico29 | I have a little question about say.conf |
| 16:42.23 | mjordan | mathi: read above. WIMPy and I were just discussing it with a sample CLI snippet he provided. |
| 16:42.26 | jacekowski | i've got question about echo cancelation, is it required on ISDN? |
| 16:42.27 | Rico29 | I'm trying to make my voicemail system announcing a phone number |
| 16:42.35 | [TK]D-Fender | mathi, WIMPy seems to have backed it up already and this is very odd indeed |
| 16:44.09 | WIMPy | I changed my verbose=1 to verbose=9 in asterisk.conf and after a core restart now, it works as expected. |
| 16:44.18 | WIMPy | 99 even. |
| 16:45.01 | mathi | WIMPy, so you encounter the same problem? I'm confused... |
| 16:45.13 | WIMPy | yes. I did. |
| 16:45.36 | Rico29 | but when I use "saynumber(0123456789)", it tells me 123 millions etc... and I want it to say "zero one twenty three..." |
| 16:45.36 | mathi | WIMPy, so what is the solution for this? |
| 16:46.04 | WIMPy | Rico29: Use SayDigits |
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| 16:46.16 | WIMPy | mathi: 6 lines up |
| 16:46.40 | WIMPy | Err. |
| 16:46.48 | mjordan | mathi: its a bug in Asterisk 11. A feature was introduced that lets you control the verbosity at the console granularity, i.e., you can have a verbosity set on the core that differs from a remote console attached. There is a problem with it, please file an issue (or WIMPy can) and we'll get it in queue to get fixed. As WIMPy pointed out, you can work around it by setting the core verbosity to be greater than the verbosity you'd like on |
| 16:46.54 | WIMPy | Rico29: You want a mixed version of digits and numbers? |
| 16:46.59 | Rico29 | WIMPy> with saydigits, it says "zero one two three four ..." |
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| 16:47.26 | [TK]D-Fender | Rico29, SayNumber is correct. You want a very custom format and should parse it out yourself |
| 16:47.32 | Rico29 | WIMPy> what I want is what I wrote in say.conf : _pho[n]e:0[1-9]XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2} |
| 16:48.04 | Rico29 | but doesn't seem to work as expected |
| 16:48.08 | Rico29 | did I miss something ? |
| 16:48.28 | WIMPy | mjordan, mathi: I will do it. |
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| 16:50.50 | mathi | WIMPy, thank you bro! |
| 16:50.52 | mathi | works |
| 16:51.04 | Rico29 | is there a way to reload say.conf file ? |
| 16:52.06 | WIMPy | ASTERISK-20654 |
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| 16:54.41 | Rico29 | ok found it, app_playback.so |
| 16:55.19 | Rico29 | but it doesn't solve my problem |
| 16:55.25 | Rico29 | can I have help please ? /D |
| 16:55.27 | Rico29 | :D |
| 16:56.25 | SeRi | sends help to Rico29 |
| 16:56.43 | Rico29 | needs more help |
| 16:58.11 | Rumbles | slaps aberrios for not checking the Vegastream earlier |
| 16:58.38 | aberrios | =D |
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| 17:00.14 | mathi | where can I have the meaning of ${HANGUPCAUSE} ? e.g. what does the number returned stand for? |
| 17:00.18 | zeno | hi all |
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| 17:01.35 | WIMPy | mathi: Q.850 or google for "isdn cause codes" |
| 17:01.50 | BorjaGVO | Hi everyone...anyone can tell me how should I use MusicOnHold()? I'm doing this in order to try and the only thing I hear is silence: http://pastebin.com/AeEyYMqt |
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| 17:02.38 | Rico29 | BorjaGVO> waht does CLI show ? |
| 17:02.56 | scgm11 | hi, is there any libss7 library compatible with asterisk 10? |
| 17:03.50 | coreyf1513 | WIMPy: it's not Q.931? |
| 17:03.56 | BorjaGVO | Rico29: This: http://pastebin.com/gLywYEUw |
| 17:04.32 | Rico29 | can you add timestampŝ |
| 17:04.33 | WIMPy | coreyf1513: Q.850 is only about cause codes. |
| 17:05.06 | WIMPy | BorjaGVO: Do you have any music available? |
| 17:05.44 | BorjaGVO | WIMPy: yes... |
| 17:05.55 | BorjaGVO | :/ |
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| 17:06.15 | scgm11 | none using ss7? |
| 17:07.37 | WIMPy | scgm11: Not too many, I guess. But then I would understand better if you asked for a compatible channel, not a lib. |
| 17:08.18 | scgm11 | ok, I though the libss7 of digium was the standard way |
| 17:08.50 | scgm11 | is there any other? digium has dropped support? |
| 17:10.21 | WIMPy | I only knew the danish one. |
| 17:10.40 | WIMPy | But for Asterisk you need a channel. And that channel might need a lib. |
| 17:11.20 | scgm11 | I have used libss7 the digium one with ast 1.8 |
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| 17:12.36 | scgm11 | my question is if there is support for 10 |
| 17:12.42 | scgm11 | the channel driver is dahdi |
| 17:12.50 | scgm11 | works like libpri |
| 17:13.39 | Qwell | scgm11: What makes you think it wouldn't work with Asterisk 10? |
| 17:13.46 | [TK]D-Fender | Why wouldn't it work? |
| 17:13.51 | scgm11 | because it doesn't compile |
| 17:14.03 | Qwell | What doesn't compile? |
| 17:14.04 | scgm11 | against the 10 sources |
| 17:14.09 | scgm11 | the libss7 |
| 17:14.30 | Qwell | libss7 doesn't care what version of Asterisk you have installed. |
| 17:15.02 | scgm11 | mmm maybe is a dependency then |
| 17:15.08 | scgm11 | let me check again |
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| 17:16.57 | scgm11 | qwell: check this |
| 17:16.58 | scgm11 | http://pastebin.com/PbATC98D |
| 17:17.42 | WIMPy | Check your compiler flags. |
| 17:17.52 | WIMPy | That's not a real issue. |
| 17:18.33 | WIMPy | That is, get rid of the -Werror. |
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| 17:20.21 | Rico29 | ok still need help |
| 17:20.34 | Rico29 | I've tried many things, but nothing works |
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| 17:22.44 | Rico29 | still have error : pbx.c:10179 pbx_builtin_saynumber: We were unable to say the number 067697...., is it too large? |
| 17:23.18 | scgm11 | how can I change those compilers flags? I always use the same I just made make |
| 17:23.31 | Rico29 | ./configure --help |
| 17:24.23 | WIMPy | Or edit CFLAGS in the makefile. |
| 17:24.51 | Rico29 | WIMPy> no idea for my saynumber problem ? ;) |
| 17:25.36 | Rico29 | or [TK]D-Fender |
| 17:25.41 | Rico29 | ? |
| 17:25.42 | WIMPy | Rico29: I haven't tried to (re)configure that so far. |
| 17:25.47 | Rico29 | ok |
| 17:26.25 | WIMPy | Oh, wow. Lots of changes to chan_misdn. Some outdated things really refuse to die. |
| 17:26.26 | scgm11 | this has no configure the make file is the digium default |
| 17:26.30 | scgm11 | i will check |
| 17:26.31 | scgm11 | the makefile |
| 17:26.55 | Rico29 | there's no configure script anymore ? |
| 17:27.08 | scgm11 | seems no |
| 17:27.14 | Rico29 | ok |
| 17:27.15 | scgm11 | at least I can't find it |
| 17:27.50 | scgm11 | I will get rid of -Werror flag |
| 17:27.52 | scgm11 | ok?? |
| 17:28.15 | [TK]D-Fender | Rico29, You are showing things in bits and pieces. and only rarely at all |
| 17:28.28 | Rico29 | ok, will pastebin it |
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| 17:29.47 | Rico29 | ... tomorrow |
| 17:29.54 | Rico29 | time to go home (in france) |
| 17:30.02 | Rico29 | good afternoon all ! |
| 17:30.26 | [TK]D-Fender | ... |
| 17:32.35 | beardy | Always interesting how they expect people helping them for free on their free time if they go home from a paying job in the middle of it eh. |
| 17:33.52 | file | bananas. |
| 17:34.31 | beardy | Cream pudding. |
| 17:35.07 | file | omg... banana pudding... |
| 17:35.22 | theRawr | yum |
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| 18:25.24 | paulc | Nivex: You aruond? |
| 18:25.26 | paulc | or around even |
| 18:28.58 | Nivex | paulc: yo |
| 18:30.29 | JunK-Y | paulc ! |
| 18:36.36 | navaismo | So for the libpri-1.4.13 dahdi needs to be installed first? If I recall libpri was before dahdi. |
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| 18:43.38 | glaz | Hi, how can I configure a Queue() to be able to receive DTMF to send the caller to another extension? i.e.: Press 9 to leave voicemail while in Queue(). |
| 18:44.18 | [TK]D-Fender | glaz, context= <------ |
| 18:44.27 | [TK]D-Fender | documented in the sample config..... |
| 18:45.06 | glaz | context= in the section of my queue in queue.conf |
| 18:45.12 | glaz | right? |
| 18:45.22 | [TK]D-Fender | yes |
| 18:45.27 | navaismo | ~itsp-us |
| 18:45.37 | glaz | ok let me test it. |
| 18:45.40 | navaismo | ~itsp |
| 18:45.40 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:45.51 | navaismo | ~itsplist-us |
| 18:45.51 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:49.30 | glaz | [TK]D-Fender: it works, thanks, also, is it possible to play a Background() message and put the caller in the Queue() right away and not wait for Background() to finish? |
| 18:50.19 | [TK]D-Fender | glaz, No |
| 18:51.02 | glaz | So, I should configure my queue greeting message in queues.conf ? |
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| 18:52.09 | [TK]D-Fender | There isn't really a "greeting". Ther is a periodic announcement, or whatever you play before. But it is not a "greeting"/. |
| 18:52.28 | glaz | ok |
| 18:52.42 | glaz | I guess it can't be done then |
| 18:54.02 | jblack | you might be able to pull a trick by using musiconhold |
| 18:54.56 | glaz | true |
| 18:58.21 | *** join/#asterisk Shane-S (~chatzilla@c-68-46-130-94.hsd1.nj.comcast.net) |
| 18:59.58 | Shane-S | I have some total newbie questions that most sites are either dumbing down for home VOIP services, or they assume I know. First is say I have internet access, do I need a voip service/provider to make/receive calls or would something like Google Voice work? |
| 19:03.48 | jmetro | with just an internet connection you can do URI dialing and other internet dialing, or do something like make your own little phone network across the country. |
| 19:06.57 | glaz | also, I'm trying to play an audio stream for my musiconhold, it says it started but I can't hear anything |
| 19:07.13 | glaz | the stream is http://5283.live.streamtheworld.com:3690/CHMPFMAAC_SC |
| 19:07.20 | glaz | it plays fine in winamp |
| 19:07.42 | Qwell | glaz: Does it play in mpg123? |
| 19:07.56 | glaz | how can I tell, I am remotely connected to the server |
| 19:08.02 | glaz | I see a bunch of yyyyyyyyy |
| 19:08.13 | Qwell | So then clearly not. |
| 19:09.14 | glaz | anyone running linux who can test it ? |
| 19:09.24 | [TK]D-Fender | Shane-S, If you want to call the real world, you need some sort of service provider. |
| 19:12.14 | Shane-S | [TK]D-Fender: If I want to just test the system out, having only internet connectivity what would I be looking for or could I use. I am just wanting to learn and play, but I keep seeing DID, TRUNK LINES, etc., but not one is explaing what they do, search for POTS lines I find out analog and needing FXS? cards or something...very confusing to when the ideal is just pick up and dial in, pick... |
| 19:12.16 | Shane-S | ...up and dial in. |
| 19:12.36 | Qwell | Shane-S: POTS = analog |
| 19:12.41 | [TK]D-Fender | Shane-S, You don't need ANY of these just to learn * |
| 19:13.15 | Shane-S | [TK]D-Fender: How would I emulate the calling inside / calling outside process? |
| 19:13.25 | [TK]D-Fender | Shane-S, You can use whichever you like at whatever point you feel like, but they are not required for basic learning. Feel free to jsut set up a softphone or two and learn how all the other bits work |
| 19:13.44 | *** join/#asterisk zeno (~zeno@88.103.102.96) |
| 19:13.58 | [TK]D-Fender | Shane-S, A call is a call is a call. You could say "well this is where I WOULD be calling outside... but I'll just do something so I know my call was processed right before going out. |
| 19:15.29 | glaz | can I show you my musiconhold.conf config ? |
| 19:16.04 | Qwell | glaz: get your stream app to actually play the stream first |
| 19:16.05 | [TK]D-Fender | ~did |
| 19:16.05 | infobot | somebody said did was Direct Inward Dialing, or just a phone number |
| 19:16.07 | [TK]D-Fender | ^ |
| 19:16.18 | *** join/#asterisk chris-NB (~chris@home.fuerstaller.com) |
| 19:16.27 | WIMPy | ... which would actually be non-did. |
| 19:16.51 | glaz | Qwell: there's no way I can remotely play the sound right ? |
| 19:16.51 | [TK]D-Fender | Shane-S, A DID points to some provider and they pass the call to you. If you have a boring copper line from the telco, that's one way. I fyou have an ITSP that is another. Maybe you have a Digital phone circuit like ISDN, etc. |
| 19:16.52 | Shane-S | [TK]D-Fender: Thank you! |
| 19:16.56 | [TK]D-Fender | ~itsp |
| 19:16.57 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:17.39 | [TK]D-Fender | Shane-S, Asterisk processes calls. Doesn't really matter from where, or if you plan on taking that call and bridign it outwards. You can lear all of the processes around it jsut fine with jsut a softphone |
| 19:17.40 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 19:18.13 | Shane-S | [TK]D-Fender: I want to learn because we have a POTS system with 4 copper lines. I would like to kill our bills by using something like Comcast VOIP for $99 per line versus the $350+ per line we are charged in fees |
| 19:18.57 | [TK]D-Fender | Shane-S, How much actual talk time do you use, and to where? What direction are the calls? (inbound vs outbound) |
| 19:19.33 | *** join/#asterisk bchia (~Adium@nat/digium/x-wmufgrrxdjcccmzj) |
| 19:19.58 | Shane-S | [TK]D-Fender: Never broke it down, I just see the $$ we spend, 5K a year in Verizon bills |
| 19:19.58 | glaz | I actualy get this error, [Nov 6 14:10:06] DEBUG[21965]: res_musiconhold.c:1131 moh_class_destructor: mpg123 pid 22018 and child died after 50176 bytes read |
| 19:20.11 | [TK]D-Fender | Shane-S, Sometimes you may need to support up to X calls at a time, but maybe oly very rarely, so you don't need them as "dedicated". Different providers offer different packages with difference in their cost effectiveness depending on your specific needs. |
| 19:20.23 | Shane-S | [TK]D-Fender: I would say it is 60/40 in/out |
| 19:20.54 | [TK]D-Fender | Shane-S, First I'd calculate actual talk time. |
| 19:21.48 | glaz | Qwell: it plays |
| 19:22.54 | Shane-S | [TK]D-Fender: okay I will have to request the phone bills from out accounts payable |
| 19:22.59 | Shane-S | our* |
| 19:23.06 | [TK]D-Fender | Shane-S, just one sample to keep in mind : http://www.phonebooth.com/business-phone-plans |
| 19:23.25 | [TK]D-Fender | Shane-S, You should know where you are now to better plan where you're going. |
| 19:23.48 | jmetro | i like how the say unlimited auto attendants |
| 19:23.51 | Shane-S | I was just figuring on 4 dedicated system line that could be utilized from 8am - 4pm in near 100% capacity |
| 19:24.24 | [TK]D-Fender | Shane-S, Ok, you are in the demographic that tends to make "unlimited" offering a profitable choice. |
| 19:25.35 | glaz | Qwell: I think I'm getting further now, I get tons of these though; monmp3thread: Read 39 bytes of audio while expecting 1600 |
| 19:27.14 | Shane-S | [TK]D-Fender: That confuses me more...I don't have "users" I have 4 lines that any of 50 extensions can utilize if free..that is why VOIP is confusing me...I would never pay per user, we aren't a call center. Just a small business who receive calls when needed, and make them when needed. |
| 19:27.41 | paulc | Nivex: Back at my desk, delayed reply - sorry.. I was going to say "how about a wildcard/404 handled page on a local (spoofed) provisioning HTTP server, that returns a blank/empty (unlocked) profile to the device?" |
| 19:27.54 | paulc | JunK-Y: Salut! Ca va?! Long time no speak mate! |
| 19:28.28 | Nivex | paulc: I was thinking of trying that but it was very much locked in to HTTPS so I figured the cert verification would fail. |
| 19:28.29 | JunK-Y | yeah, lets talk tonight, im in rush now |
| 19:28.43 | Nivex | paulc: moot point now. They gave me the unlock code :) |
| 19:28.53 | paulc | Haha awesome - problem solved then :) |
| 19:29.07 | paulc | JunK-Y: Sure - tout a l'heure |
| 19:30.07 | Shane-S | [TK]D-Fender: Looks like I will just have to let my boss call in some phone companies and just pay $$$ cause this is all beyond my understanding. Thank you for the help, at least now I know I would be underwater tackling this. |
| 19:30.22 | Qwell | ~book |
| 19:30.22 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:30.28 | Qwell | This is all very simple stuff. |
| 19:34.44 | [TK]D-Fender | Shane-S, That one might be a whole hosted service |
| 19:35.00 | [TK]D-Fender | Shane-S, Forget that for now, but there are "per line" concept pricing offerings out there. |
| 19:35.21 | [TK]D-Fender | Shane-S, As in "per dedicated channel" |
| 19:35.56 | [TK]D-Fender | Shane-S, It's nowhere near as bad as you feel it might be |
| 19:36.27 | *** join/#asterisk jblack (~jblack@170.sub-70-192-131.myvzw.com) |
| 19:36.48 | Shane-S | Qwell / [TK]D-Fender : That is where this is looking like dark muddy water for me is the real connectivity. Asterisk Now was amazingly easy to get software phone working at home...so i thought I could help...but seems so confusing. |
| 19:37.33 | [TK]D-Fender | Shane-S, Well you're learning about more tha just the dumb phone lines you have now... so naturally there is more. |
| 19:38.19 | Shane-S | Anyone used phybridge? Another issue we have is re-wiring for Cat5/PoE VOIP |
| 19:39.04 | Shane-S | Phybridge I found on PhoneLady.com apparently you can use Cat3 with it and an adapter... |
| 19:39.24 | *** join/#asterisk breezy (~breezy@74.115.41.6) |
| 19:41.02 | Shane-S | Of course that tosses in a new wrench for me learning Asterisk...is how that thing interfaces, I am hoping it is just plug and play, ie LAN port to Asterisk Box or Switch, and then Phybridge makes Cat3 just like Cat5....soo much new stuff to learn... |
| 19:41.19 | Qwell | That sounds like a terrible idea. |
| 19:41.20 | WIMPy | Shane-S: the easiest way to start is to keep your existing phone system and just replace you lines with VOIP accounts. |
| 19:42.27 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 19:42.39 | WIMPy | And if you've got 4 wires of cat3 that's good enough for ethernet. |
| 19:42.47 | [TK]D-Fender | Looks like so horribly misrepresented ATA/gatewaty |
| 19:42.55 | WIMPy | Just don't try fast ethernet. |
| 19:43.04 | Nivex | mmm 10 megabit |
| 19:43.13 | WIMPy | Looks like a DSLAM like solution including power feeding to me. |
| 19:43.34 | [TK]D-Fender | Shane-S, What is that mosntrosity for? |
| 19:43.35 | Qwell | Whatever it is, I'm sure it's stupidly expensive. |
| 19:43.51 | WIMPy | It is. |
| 19:44.17 | WIMPy | Don't you have networked PC where you want to put phones? |
| 19:44.27 | [TK]D-Fender | Shane-S, Big tip since you're just starting out : don't bring ideas of equipement you THINK you need. State what you have and you want it to do and we can better advise on what will acoomplish the task for you. |
| 19:44.29 | Shane-S | one second...on a call lol |
| 19:45.22 | WIMPy | We should have support for Up0 phones. |
| 19:47.58 | Shane-S | [TK]D-Fender: We currently have a 12 year old Vodavi Digital Phone system using 4 incoming/outgoing POTS lines, with 43 extensions. I am hoping to be able to provide an phone system replacement for that. |
| 19:48.47 | *** join/#asterisk briguy (~BriGuy@74.115.41.6) |
| 19:48.56 | [TK]D-Fender | Shane-S, Seems that's a dead-end. |
| 19:49.19 | [TK]D-Fender | Shane-S, as far as recyclability is concerned |
| 19:49.31 | [TK]D-Fender | Shane-S, Its a rip&replace. **OR** |
| 19:49.49 | [TK]D-Fender | Shane-S, You could just replace the lines going in the front |
| 19:50.14 | [TK]D-Fender | Shane-S, And use your existing system as-is and it simply doesn't know that the lines you plugged in aren't what you used to have. |
| 19:50.23 | Shane-S | [TK]D-Fender: The wiring is what I was "hoping" to keep in place as that has been quoted as 7-8K at $125-175 per drop. |
| 19:50.46 | WIMPy | Don't you have networked PC where you want to put phones? |
| 19:50.46 | [TK]D-Fender | Shane-S, You don't have ethernet in the appropriate spots? |
| 19:51.09 | Shane-S | No, all wireless now. |
| 19:51.36 | Qwell | 43 desktops, on wireless? |
| 19:51.54 | Shane-S | We have ethernet, but it is goes to old closets, looking to move everything to an MDF or and IDF, wiring currently goes to a celing mounted cabinet, the Cat3 is in its own room hence I want to use that. |
| 19:52.03 | WIMPy | Sounds like a good idea to shove in some cables anyway. |
| 19:52.09 | Qwell | You can't put a switch in the cabinet? |
| 19:52.09 | [TK]D-Fender | Shane-S, Ok, if you also want to replace your existing system then you'll still need all new hardware. Three are some 2-wire digital converters out there for older handsets (not yours), or you could go analog for those |
| 19:52.12 | Shane-S | no 250 laptop/netbooks on wireless :P |
| 19:52.19 | Shane-S | School |
| 19:52.53 | Qwell | You're very seriously overthinking this. |
| 19:54.35 | [TK]D-Fender | I personally like the idea of investing in infrastructure. wiring the drops would be a good thing |
| 19:55.28 | *** join/#asterisk briguy (~BriGuy@74.115.41.6) |
| 19:56.49 | Shane-S | [TK]D-Fender / Qwell : Yeah, I agree, I actually want pink or purple colored Cat5, and I want dedicated switches ...actually just a 48 port PoE if they cables will reach |
| 19:57.11 | [TK]D-Fender | Shane-S, 5e/6. 5 = dead |
| 19:57.26 | [TK]D-Fender | Actually ... just 6. Why screw yourself so much shorter down the line... |
| 19:57.34 | [TK]D-Fender | labour is the worst part anyway |
| 19:58.18 | file | moar infrastructure! |
| 20:00.29 | Shane-S | [TK]D-Fender: okay, that (aside from handsets) was my largest cost on my estimating spreadsheet, so I think the consensus is I should just do it. I do feel the same, just my estimates are nearing 20K for a phone system supported by me, while a corporate offer came in at 30K (licensing aside each year) |
| 20:01.32 | Shane-S | I just wanted to present significant savings that would be reliable and supported (meaning not a one man band, they are "F"ed if I leave deal). |
| 20:01.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-xcgluorztharbzky) |
| 20:02.08 | *** join/#asterisk cheasee (~cheasee@vie-91-186-159-054.dsl.sil.at) |
| 20:02.11 | cheasee | hi |
| 20:02.28 | Shane-S | I will have to see if I can find other companies to quote Cat 6 runs lower, because then I can "justify" it as ANY system would use that |
| 20:02.40 | [TK]D-Fender | Shane-S, What is the other offer comprised of? |
| 20:03.34 | Shane-S | No idea, they got the quote and said we can't afford that in the budget, and I starting thinking of Asterisk Now. |
| 20:03.57 | Shane-S | I am waiting on a Shoretel and Avaya Quote I requested, so that I have the details |
| 20:04.26 | [TK]D-Fender | Plenty of ex-pats from each of those here... |
| 20:04.50 | jmetro | ew shore-tel.... ewww avaya |
| 20:05.05 | [TK]D-Fender | All sharing similar opinions... |
| 20:06.04 | *** join/#asterisk The_Hatta (ident@165-154-121-72.ispnetbilling.com) |
| 20:06.52 | Shane-S | Sad part is we have VERY basic needs, dial in with a prompt menu, extensions, voicemail, internal calling (multi-line), and we used to have call out restrictions until cell phones |
| 20:07.16 | jmetro | it took me about an hour to set that up the first time I installed asterisk on a virtualbox |
| 20:08.06 | Shane-S | No on hold music, no voicemail to email, no dial in to dial out (for hiding personal numbers), no "find me" re-routing/forwarding, and we rarely even move phones/extensions |
| 20:09.21 | Shane-S | The biggest feature we use/must have is 911 button direct to 911 center and certain phones that must light up a 911 button and display the extension that dialed it. |
| 20:10.59 | Shane-S | jmetro: that is why I thought I could help save $$, but the infrastructure upgrade is half the cost, then the phones, the system seems to be the cheapest part. |
| 20:12.16 | Shane-S | I just found PBX in a Flash blog post about using an Atom based barebones, and having a hardware server for under $300 |
| 20:12.49 | [TK]D-Fender | Shane-S, Your needs are petty by scale. My watch could do this ... and it's ANALOG |
| 20:15.04 | jmetro | if you already have internet in place, its just a matter of getting more bandwidth or better internet to support voip |
| 20:15.39 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 20:16.04 | Shane-S | I have the bandwidth, I have Comcast at 5Mb and FiOS at 15Mbs (probably higher in burst, but that is the max I have realistically seen from the lines) |
| 20:17.07 | *** join/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 20:17.17 | Shane-S | I also have them using saturation load balancing. Sadly the slower Comcast is the primary, because Verizon stole my fiber pair once and we were down for a week...until a tech finally check the fiber and it was dark. |
| 20:17.56 | TheKernel[work] | I'm trying to setup Call Parking, but when I transfer a call to 700 it tries to send it out the Outbound trunk, how do I configure the dial plan to send the calls to the application? |
| 20:19.32 | [TK]D-Fender | TheKernel[work], use a better pattern for your trunk |
| 20:19.45 | [TK]D-Fender | TheKernel[work], And pay attention to your INCLUDE order in the dialplan |
| 20:19.51 | Shane-S | alright guys thank you for the help/info/tips |
| 20:20.22 | [TK]D-Fender | Shane-S, You can delay your harware choice BTW |
| 20:20.40 | [TK]D-Fender | Shane-S, All you need is about $100 worth of ATA's |
| 20:20.49 | Shane-S | [TK]D-Fender: ? |
| 20:20.49 | [TK]D-Fender | Shane-S, And that will let you get off your analog lines. |
| 20:21.12 | [TK]D-Fender | Shane-S, And you can plan out your central hardware upgrade while saving on line costs. |
| 20:21.13 | Shane-S | What would those be could you link me to them? |
| 20:21.25 | [TK]D-Fender | Shane-S, Linksys PAP2T-NA |
| 20:21.28 | [TK]D-Fender | ^ |
| 20:21.48 | Shane-S | hmm...I might hang in here to ask you more.. |
| 20:22.30 | [TK]D-Fender | Shane-S, Remember you have several motivations in play. LIne costs are line costs and a very real consideration. This is also the EASIEST to change. |
| 20:23.14 | Shane-S | [TK]D-Fender: Now see your suggestion...now makes me angry with our phones guys....because I asked them... |
| 20:23.21 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 20:23.41 | [TK]D-Fender | Shane-S, It ain't Raw-Cat Sigh Hence. |
| 20:23.49 | Shane-S | [TK]D-Fender: If we rented modems from say Comcast that take VOIP and make it so any standard phone can use it, would the same phone system we have work |
| 20:24.06 | [TK]D-Fender | Shane-S, Unplug 4 wires from telco. Plug 2 into each PAP2. Send calls over ITSP. The end. |
| 20:25.21 | Shane-S | oh I see what you are saying, pay for VOIP using POTS, I thought these were to use VOIP and send it into my system as an Analog signal |
| 20:25.45 | [TK]D-Fender | it IS |
| 20:25.57 | [TK]D-Fender | use your system AS-IS and just send the calls over VoIP |
| 20:26.03 | [TK]D-Fender | save on line costs NOW |
| 20:26.26 | [TK]D-Fender | and while you're banking savings it sure becomes easier to seel the "upgrade" idea to the boss... |
| 20:26.28 | Qwell | [TK]D-Fender: err, PAP2? |
| 20:26.35 | [TK]D-Fender | Qwell, It'll do the job :) |
| 20:26.53 | Qwell | I suppose you're right. |
| 20:26.55 | [TK]D-Fender | Fine, splurge anther few $$ for some slightly nicer ATA's! |
| 20:27.10 | Qwell | I was thinking about it backwards. That's just evil. |
| 20:27.28 | Shane-S | hmm aren't these the exact saem things Vonage or Comcast rent you? |
| 20:27.35 | [TK]D-Fender | Pretty much |
| 20:28.00 | *** join/#asterisk Jinxed-_ (93b128b1@gateway/web/freenode/ip.147.177.40.177) |
| 20:28.11 | [TK]D-Fender | Vonage first used the Sipura SPA-2000 ATA which was bought out by Linksys |
| 20:28.17 | Shane-S | I am tempted to try it with a number to see.....but the local installer said I need a special card to use that method |
| 20:28.29 | [TK]D-Fender | Shane-S, You don't |
| 20:28.41 | [TK]D-Fender | Not for REPLACING your lines. |
| 20:28.51 | [TK]D-Fender | to INTERFACE with them however.. that's another piece of hardware |
| 20:28.53 | Shane-S | you are really making we want to go punch them in the balls |
| 20:28.57 | [TK]D-Fender | Direction matters. |
| 20:29.09 | Jinxed-_ | Any ideas on what to look for in the situation where you have one way audio which is very choppy, but audio the other way is perfectly clear |
| 20:29.12 | [TK]D-Fender | GOOD. That means you're starting to understand... |
| 20:29.33 | Jinxed-_ | in the same call |
| 20:29.35 | [TK]D-Fender | Jinxed-_, difference in BW. |
| 20:29.49 | Jinxed-_ | [TK]D-Fender: difference is bandwidth within the same call? |
| 20:29.51 | [TK]D-Fender | Jinxed-_, Got a small upstream? Might be choking. |
| 20:30.20 | Shane-S | [TK]D-Fender: I don't know the internals...so assuming i do that, and I have dial tone, and I plug them into the same spot the POTS line used to plug in, theoretically my Vodavi card know no difference? |
| 20:30.35 | [TK]D-Fender | yup |
| 20:30.52 | [TK]D-Fender | Your Vodavi is jsut a dumb phone when you get down to it |
| 20:32.37 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 20:32.47 | Shane-S | That is what I thought...I felt like a moron when they told me I can't switch without a card, when I sat there going...but I can plug a standard home phone into that jack in the phone system and get tone, and i can use VOIP router wire and get tone, how the hell does it know there is a difference |
| 20:33.09 | [TK]D-Fender | Answer : There isn't any. |
| 20:33.28 | jmetro | dialtone in a voip system is conveniently generated by the server, because otherwise the crisp, clear tone of pure signal makes you think the line is dead. Same reason why servers have a built in static noice generator for the silent parts of conversations. |
| 20:34.34 | Shane-S | Like the time they told me the voicemail system was dead...before the $4K card....and I took the power supply appart as it wasn't power up, and a fuse in the AT PS blew...it was soldered on..so I ask for a new one and the read me the riot act, and swore to me I broke it. |
| 20:35.49 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 20:35.50 | [TK]D-Fender | Shane-S, Ah well, a VM car is another matter... they tend to have a little exrta signalling around them, etc |
| 20:36.01 | [TK]D-Fender | And yes, telcos love blaming (and billing) you for everything |
| 20:36.18 | Jinxed-_ | [TK]D-Fender: can you monitor it somehow to see if that is the issue in asterisk? |
| 20:36.29 | [TK]D-Fender | So if You're finding the idea of needing special hardware for this a distasteful as the rest of us ... by all means continue your migration plans... |
| 20:36.46 | [TK]D-Fender | Jinxed-_, It's your connection. Check at your border. |
| 20:37.02 | [TK]D-Fender | Jinxed-_, Comapare the in vs out |
| 20:37.54 | Jinxed-_ | I think I have ennough bandwidth just worried about delay |
| 20:38.26 | Shane-S | [TK]D-Fender: okay will do thanks...now I want to try and get another coaxial line in here and steal my dads Comcast modem/voip router and see if i can use it here (same county). Then try adding that line to the card...hopefully the number isn't hard wired into the system |
| 20:39.52 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 20:40.06 | Jinxed-_ | [TK]D-Fender: is there anything I can configure on asterisk to make asterisk more tolerant of lossy networks with delays of around 400ms |
| 20:40.28 | Jinxed-_ | I'm not worried about delay but choppyness is awful |
| 20:41.12 | [TK]D-Fender | Shane-S, Number should be tied to the modem. |
| 20:41.23 | *** join/#asterisk Micc (~Mic@199.38.209.68) |
| 20:41.44 | [TK]D-Fender | Jinxed-_, If you've already got jitter buffers enabled... then you're way beyond spec |
| 20:42.42 | Jinxed-_ | [TK]D-Fender: How do I make sure my jitter buffers are enabled and maxed out |
| 20:42.45 | *** join/#asterisk jsjc (~Adium@189.Red-2-136-116.dynamicIP.rima-tde.net) |
| 20:42.55 | jmetro | jitter buffers add delay |
| 20:43.29 | Jinxed-_ | [TK]D-Fender: 400 ms is round trip (ping) |
| 20:44.23 | Jinxed-_ | alright i will do some googling |
| 20:44.26 | Jinxed-_ | thx for the help |
| 20:45.44 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:48.35 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
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| 21:15.33 | ghost75 | (21:42:07) [TK]D-Fender: Jinxed-_, If you've already got jitter buffers enabled.. are there any recommended values? |
| 21:24.31 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:24.31 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:24.40 | ghost75 | jbenable=yes + jbforce=yes in sip.conf should tell Asterisk to dejitter incoming audio from SIP channels no matter what the receiving leg can handle. |
| 21:24.51 | ghost75 | so jitterbuffer helps only for incoming audio? |
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| 21:54.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:56.30 | *** join/#asterisk Conker (~Conker@static-207-54-111-107.ptr.terago.net) |
| 21:57.20 | Conker | Hello all! I'm trying to setup asterisk and for whatever reason is doesn't seem to be listening on any udp or tcp ports, im stumped |
| 21:59.47 | [TK]D-Fender | Show us |
| 22:00.14 | Conker | Show you? |
| 22:00.32 | [TK]D-Fender | that Asterisk is running but no ports bound |
| 22:03.42 | Conker | http://pastebin.com/5NaSeTiw |
| 22:05.35 | [TK]D-Fender | And the part about proving Asterisk is actually running? |
| 22:06.08 | pabelanger | Conker, *CLI> sip show settings |
| 22:06.11 | pabelanger | pb your results |
| 22:06.15 | Conker | line 10. |
| 22:06.51 | ghost75 | netstat -na | grep 5060 |
| 22:07.50 | Conker | No such command 'sip show settings' |
| 22:08.01 | Conker | ghost75, nothing |
| 22:08.19 | pabelanger | Conker, then you are no loading chan_sip |
| 22:08.30 | pabelanger | *CL> module load chan_sip.so |
| 22:09.23 | ghost75 | can bindport be changed to a different setting than "port" ? |
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| 22:14.23 | [TK]D-Fender | ghost75: Like? |
| 22:14.40 | ghost75 | like bindport=8060 port=5060 |
| 22:15.16 | [TK]D-Fender | doesn't make sense |
| 22:15.22 | [TK]D-Fender | * binds to a single port |
| 22:16.19 | ghost75 | listening is bindport and peer is port |
| 22:17.07 | [TK]D-Fender | port = DEST port for outcalls |
| 22:17.54 | ghost75 | my isp uses 5060 but i dont want to be reachable on 5060 |
| 22:18.23 | Conker | pabelanger: your correct, wasn't loaded, so now it is listening |
| 22:20.15 | [TK]D-Fender | Conker: then bindport is it |
| 22:21.04 | [TK]D-Fender | ghost75: rather |
| 22:22.01 | ghost75 | i tried bindport 8060 but i could not call out |
| 22:24.12 | Conker | [TK]D-Fender, bindport? |
| 22:24.36 | [TK]D-Fender | ghost75: that is something ELSE you should be showing us. |
| 22:24.55 | ghost75 | not now |
| 22:24.56 | [TK]D-Fender | Conker: Bad aim, don't worry about it |
| 22:25.27 | ghost75 | i think nat is needed when bindport!=port |
| 22:25.57 | Conker | [TK]D-Fender, ok |
| 22:26.19 | [TK]D-Fender | ghost75: There is no association between the two |
| 22:27.39 | ghost75 | i have also no port forwarding on router |
| 22:28.00 | [TK]D-Fender | THAT is a tragic failure... |
| 22:28.03 | [TK]D-Fender | for inbound anyway |
| 22:28.14 | ghost75 | but till now it worked |
| 22:30.06 | [TK]D-Fender | It might if the re-reg frequency is high enough otherwise the port would close behind it |
| 22:33.37 | ghost75 | thats maybe why incoming port needs to be same as outgoing |
| 22:33.51 | [TK]D-Fender | No. |
| 22:33.58 | ghost75 | hmm |
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| 22:57.37 | *** join/#asterisk keyy1 (~keyy@host5.global.tygrys.net) |
| 22:58.54 | keyy1 | hi there, iv got a problem with recieving sms from dect phone, everything goes ok, message is sent to asterisk from phone, but phone keeps saying that message was not send |
| 23:01.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:08.43 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.10.0 (2012/11/06), 1.8.18.0 (2012/11/06), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:11.33 | plundra | Eller tre nya kistor med besvikelser i röda rummet? :) |
| 23:11.39 | plundra | Oops, wrong channel. |
| 23:16.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gmyqsqfadhhxpeaz) |
| 23:18.18 | Nugget | A Moose once bit my sister |
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| 23:24.44 | SeRi | p3nguin: whats going on. |
| 23:24.53 | SeRi | dijib: you in? |
| 23:26.41 | ghost75 | what? |
| 23:40.45 | SeRi | ping ghost75 |
| 23:40.49 | SeRi | lol |
| 23:41.14 | *** join/#asterisk sktyau (~nick@monitor.ic.ipera.net.au) |
| 23:41.40 | cheasee | how is it possible to add a script output for Set(CALLERID(name)=$scriptoutput) what is executed before via System() ? |
| 23:42.10 | cheasee | i couldnt find a possibility to add a script to set() ? |
| 23:45.14 | sktyau | How are you passing the results of your shell script / whatever back into Asterisk? |
| 23:45.25 | cheasee | thats the question :D |
| 23:46.47 | cheasee | my script accesses my phonebook to check what name is calling, now i would like to set that name as callerid |
| 23:47.20 | cheasee | i can return the name in any way but i dont know how to tell asterisk about that? |
| 23:49.38 | sktyau | If you are running 1.6+ look at http://www.voip-info.org/wiki/view/Asterisk+func+shell - but unless your phonebook is a text file somewhere and you are simply grep'ing it (Yuck) you might benefit from AGI + [favouritelanguage] . System wasn't really designed to do what you are trying to do. |
| 23:49.59 | cheasee | phonebook is a google acc ;) |
| 23:50.13 | *** part/#asterisk ghost75 (~trechber@dslb-088-064-052-093.pools.arcor-ip.net) |
| 23:50.51 | *** join/#asterisk navaismo (~navaismo@189.241.62.150) |
| 23:51.08 | sktyau | Hi All, is anyone here familiar with (my perceived) issues regarding peer selection where there are multiple peers with the same host. I can find bucket loads of posts / "bugs" about it but not as much as a dirty hack that explains a way around the problem. |
| 23:51.17 | cheasee | ah, sktyau thanks! seems that link is exactly what im looking for! :) |
| 23:52.05 | sktyau | cheasee, Do Google provide an API for that? If so then you would def. be better off trying to do this with AGI than directly with system IMHO. |
| 23:52.53 | cheasee | sure google has an api for that.. https://developers.google.com/google-apps/contacts/v3/ |
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| 23:57.59 | lvlinux | Anybody here familiar with extensions.lua and/or extensions.ael? |
| 23:58.34 | Kobaz | what about them |
| 23:58.48 | lvlinux | All I really want to know is are they necessary? Can I delete them as long as I have extensions.conf? My Astlinux install has all three files. |
| 23:59.26 | lvlinux | when I do dialplan reload, it seems to get everything in all three of them. |
| 23:59.47 | Kobaz | if you don't need the configuration in those files then you can delete them |
