| 00:00.08 | slav3_kitten | i'm too far out for DSL, and comcast lied to my face before we put an offer in on the property. so i have a wireless isp |
| 00:00.27 | p3nguin | dodge, get out of the way, duck, hide from |
| 00:00.31 | *** join/#asterisk Kyril (~Kyril@fedora/Kyril) |
| 00:00.43 | slav3_kitten | who also lacks backup equipment at their cell site so the 1/2 second power loss today dropped my connection like a sack of potatoes |
| 00:00.49 | p3nguin | You use VoIP with a WIRELESS ISP? |
| 00:01.03 | p3nguin | Everyone knows those two things don't mix well. |
| 00:01.23 | slav3_kitten | hey now... it's all i got |
| 00:01.34 | gusto | eh |
| 00:01.34 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 00:01.38 | carrar | You need to sell and mvoe |
| 00:01.40 | gusto | so he has wireless? |
| 00:01.44 | gusto | what kind of wireless? |
| 00:01.51 | dijib | p3nguin: but can i be done? |
| 00:01.58 | dijib | it |
| 00:02.23 | slav3_kitten | it's a ubiquiti "modem" so 2.4ghz wifi to their cell site |
| 00:02.25 | p3nguin | VoIP can work over wireless networks, but it is much less reliable and often has less quality. |
| 00:03.13 | slav3_kitten | if they a) put it into bridge mode. or b) gave me a static IP it'd work just fine in theory |
| 00:03.14 | gusto | w8 w8 |
| 00:03.31 | *** join/#asterisk P-NuT (~P-NuT_@5ad3493e.bb.sky.com) |
| 00:03.31 | gusto | so it's basically wlan, right? 2,4 ghz standard ieee802.11g, right? |
| 00:03.35 | p3nguin | How will a static IP address change anything? |
| 00:03.52 | gusto | slav3_kitten: no |
| 00:03.59 | carrar | Everyone knows static IP's go farther |
| 00:04.12 | p3nguin | Oh, right. Almost forgot about that. :/ |
| 00:04.57 | *** join/#asterisk wonderworld (~w@77.181.44.222) |
| 00:04.58 | slav3_kitten | gusto, because currently it's in a router mode and natting me. if they put it into bridge (may refuse an insist on selling me a static ip, gotten different things from different people when i've called) |
| 00:05.25 | slav3_kitten | it'll end up in bridge mode if i have static ip for sure, and should be able to be put in bridge mode with dynamic ip but they well might refuse to. |
| 00:05.27 | p3nguin | Bridging it shouldn't be hard to get, as long as the device is capable of it. |
| 00:05.56 | *** part/#asterisk P-NuT (~P-NuT_@5ad3493e.bb.sky.com) |
| 00:06.17 | slav3_kitten | they are, i've used that model device as an AP in the past. it's just locked down 100% so only they can make changes |
| 00:06.23 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 00:07.49 | slav3_kitten | it basically boils down to being at their whim p3nguin and they may make me get a static IP to put it into bridge mode. |
| 00:08.44 | p3nguin | So the static IP isn't really related, except that what your real goal is is actually a side effect of getting the static address. |
| 00:09.05 | slav3_kitten | yea |
| 00:09.49 | slav3_kitten | i really hate these bastards and as soon as any real isp is able to give me service... |
| 00:13.09 | *** join/#asterisk navaismo (~navaismo@189.191.12.116) |
| 00:13.39 | gusto | but somehow we are missing the point |
| 00:13.51 | gusto | when someone is too far for copper, there should be a fibreoptic alternative |
| 00:14.02 | gusto | because optical fiber has less resistence |
| 00:19.27 | slav3_kitten | gusto, we have a hard time getting fiber to people who live in cities. also optical fibers are ass expensive compared to coax or copper |
| 00:19.54 | *** join/#asterisk infinity1 (~brendon@216.218.216.173) |
| 00:20.19 | infinity1 | is it supposed to take an hour to download a bootrom on a polycom or is something wrong? |
| 00:20.31 | gusto | well |
| 00:20.36 | gusto | but i have no idea why |
| 00:20.49 | gusto | because theoretically optical fibres should not be that expensive |
| 00:22.53 | slav3_kitten | gusto, super strict quality control issues i'd guess. |
| 00:24.12 | gusto | tss |
| 00:24.28 | gusto | slav3_kitten: where areyou from? |
| 00:25.20 | slav3_kitten | united states |
| 00:27.03 | gusto | well |
| 00:27.04 | gusto | so |
| 00:27.16 | gusto | what kind of strict quality control are you talking about? |
| 00:27.27 | *** join/#asterisk jblack (~jblack@230.sub-70-192-128.myvzw.com) |
| 00:27.38 | slav3_kitten | when making optical fibers |
| 00:28.17 | slav3_kitten | your materials have to be perfect, can't abuse it when packing it, etc. |
| 00:30.09 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 00:32.53 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.250) |
| 00:33.55 | gusto | well |
| 00:34.14 | gusto | the industry is able to do CPU's and shit ... so what would be a problem with optical shit |
| 00:39.18 | slav3_kitten | different industries |
| 00:39.29 | slav3_kitten | optical is either glass or plastics industry |
| 00:39.54 | slav3_kitten | my experience in the plastics industry makes me wonder how anything gets done. but i worked for a shit company |
| 00:40.11 | gusto | yes |
| 00:40.39 | gusto | glass is the one |
| 00:40.50 | gusto | because with plastics you can not do like 100 km or such |
| 00:40.58 | gusto | that goes only with glass |
| 00:41.27 | gusto | i am sometimes really sick and tired of this |
| 00:41.45 | slav3_kitten | ultra low dispersion glass as well. like the kind you get in high end camera lenses and those things are ass expensive. |
| 00:41.47 | gusto | you can buy 100m cat5 for like 20 eur lately |
| 00:41.48 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 00:42.06 | gusto | and that is copper |
| 00:42.18 | gusto | and copper is expensive and hard to put out of the ground |
| 00:42.47 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.250) |
| 00:43.15 | gusto | so compared to what you need for energy and such for copper, glass manufacturing is relatively easy |
| 00:44.49 | gusto | so it's always something |
| 00:44.57 | gusto | however, wireless also can not go far |
| 00:45.12 | slav3_kitten | i wish fiber was cheaper as well, then i'd just run a nice run out to the shed next to the power conduit when we run it instead of wifi bridge |
| 00:45.39 | slav3_kitten | well wireless can go quite far provided you have LoS |
| 00:45.46 | gusto | with 2,4 ghz you can do a few kilometers, but that should not be that expensive to do with fibre optics ... but tearing up the streets may be also a problem |
| 00:46.06 | slav3_kitten | well there is 900mhz, 24ghz, etc |
| 00:46.17 | gusto | forget about that |
| 00:46.23 | gusto | Line of Sight, you are right |
| 00:46.29 | gusto | frequences ... will not do more |
| 00:46.44 | slav3_kitten | if you go low band like 30mhz you can go intercontinental distances due to skywave propagation |
| 00:46.54 | gusto | bullshit |
| 00:46.57 | gusto | that's not reliable |
| 00:47.16 | gusto | that's like pissing in the wind |
| 00:47.37 | slav3_kitten | yet ham radio guys do it all the time when the season is right an the band opens up |
| 00:47.53 | gusto | :-D |
| 00:48.00 | slav3_kitten | but generally for reliable you need los |
| 00:48.08 | gusto | well, ham radio guys ... these woodoo catchers :-D |
| 00:48.17 | gusto | that's what i said |
| 00:48.51 | carrar | HAM RADIO SAVES LIVES!!! |
| 00:49.18 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:49.29 | slav3_kitten | carrar, there is a reason i keep up on my license |
| 00:49.58 | slav3_kitten | and have like 30 vhf radios a couple uhf, and 1 hf :D |
| 00:50.09 | kikohnl | there use to be wireless telegraph systems from Honolulu to San Francisco back in 1912 |
| 00:50.18 | carrar | Might be the same reason I'm a extra |
| 00:50.26 | slav3_kitten | fist bumps carrar |
| 00:50.34 | slav3_kitten | i'm only a general sadly |
| 00:50.46 | carrar | It's so simple to upgrade |
| 00:51.01 | gusto | yes, but honolulu and san francisco is not that big of a shot |
| 00:51.04 | carrar | SUPER EASY |
| 00:51.04 | slav3_kitten | it is if i had the time to study for it |
| 00:51.12 | carrar | You don't really need time |
| 00:51.19 | carrar | just use the online thing |
| 00:51.19 | kikohnl | 2350 miles |
| 00:51.25 | gusto | when you are running quite high power and big antennas and low frequencies of some kHz |
| 00:51.25 | slav3_kitten | just memorize the exam pool? :P |
| 00:51.30 | carrar | sure |
| 00:51.38 | p3nguin | 10 meters has been pretty good during the past week. |
| 00:51.40 | carrar | If it works |
| 00:51.50 | slav3_kitten | eh not my style |
| 00:52.09 | carrar | hamradioonline.com |
| 00:52.30 | carrar | As long as you know how and were to get the information |
| 00:52.41 | carrar | no reason you have to memorize the whole world |
| 00:53.01 | carrar | I mean still learn eletronics |
| 00:53.03 | carrar | i did |
| 00:53.31 | slav3_kitten | i remember when i had time to work on projects. i have a 1/3 completed laser tag system sitting around |
| 00:54.06 | carrar | I had to get my extra, my father is |
| 00:54.08 | carrar | heh |
| 00:54.35 | slav3_kitten | oic how it is :D |
| 00:54.46 | carrar | ham radio comes and goes in phases for me |
| 00:54.46 | slav3_kitten | i have to constantly repair pinball games with my dad |
| 00:55.03 | gusto | hm |
| 00:55.09 | gusto | so ... |
| 00:55.12 | slav3_kitten | ever go to the huntsville hamfest? |
| 00:55.27 | carrar | no |
| 00:55.30 | slav3_kitten | dayton? |
| 00:55.34 | carrar | just the local ones here |
| 00:55.41 | carrar | we'll at home (WA) |
| 00:55.54 | carrar | In Japan at the moment |
| 00:55.54 | slav3_kitten | ah |
| 00:55.57 | carrar | got a jp call |
| 00:56.01 | slav3_kitten | what's in japan? |
| 00:56.09 | carrar | Japanese!! |
| 00:56.29 | slav3_kitten | lol i mean work or love |
| 00:56.29 | slav3_kitten | or holiday |
| 00:56.29 | carrar | workcation |
| 00:56.29 | gusto | we need to do something about this |
| 00:56.35 | slav3_kitten | ah |
| 00:56.40 | carrar | wife is Jpanese |
| 00:56.42 | gusto | there is too much wlan out there |
| 00:56.47 | slav3_kitten | congrats :D |
| 00:56.48 | carrar | staying at relatives |
| 00:56.59 | carrar | we do this every year |
| 00:57.06 | slav3_kitten | gusto, 24ghz wlan? |
| 00:57.15 | p3nguin | I haven't been to a hamfest in over 10 years. |
| 00:57.25 | gusto | slav3_kitten: yes, as well |
| 00:57.54 | gusto | you know, we have such problems in slovakia as well, and here in germany namely in nuremberg, here it's kind of a different problem |
| 00:58.14 | slav3_kitten | same problem i'm having with a shit wisp? |
| 00:58.18 | gusto | in slovakia, there are a lot of so-called "providers" who re-sell their DSL connection over wlan |
| 00:59.07 | gusto | and in germany here, you have a lot of flat inhabitants with wlan routers set to 100% of power output and then you have a problem connection with your own, because the 2.4 ghz band is full |
| 00:59.13 | carrar | Enable WMM on your wifi |
| 00:59.15 | slav3_kitten | ah yea that's what i get the feeling i am dealing with |
| 00:59.31 | gusto | and that's only because some ppl do not know how to set it to less, etc. how much they need |
| 00:59.50 | slav3_kitten | i think i have my access points set to 0dbm |
| 01:00.15 | slav3_kitten | seems to work pretty good inside the house |
| 01:00.16 | gusto | and of course, there are more modern wlan routers comming in, that adjust their output according to the environment, but that's taking time until the old ones will be replaced |
| 01:00.25 | gusto | i solved it with 5 ghz, so do not worry about me |
| 01:00.37 | slav3_kitten | no 900mhz? |
| 01:01.01 | carrar | Dig a ditch at night and lay fiber in it |
| 01:01.11 | gusto | slav3_kitten: we are in europe! |
| 01:01.16 | gusto | slav3_kitten: there is no 900 mhz |
| 01:01.29 | slav3_kitten | 900mhz ceases to be a frequency? :P |
| 01:01.33 | gusto | carrar: i like your idea |
| 01:01.55 | gusto | slav3_kitten: we have only GSM at 900 mhz heere |
| 01:02.04 | gusto | slav3_kitten: you can not do wlan or such in 900 mhz |
| 01:02.20 | slav3_kitten | oh yea gsm who still uses that :P |
| 01:02.28 | gusto | that would be a hard violation of regulations |
| 01:03.00 | slav3_kitten | i wish our fcc actually enforced more things here |
| 01:03.17 | slav3_kitten | oh well |
| 01:03.26 | gusto | well, the EU wanted to put UTMS on 900 mhz, but they are ignoring it ... theoretically it would be already legal to do it, but the mobile operators do not do it |
| 01:03.40 | gusto | they do LTE at 800 mhz instead |
| 01:03.46 | slav3_kitten | ah |
| 01:03.54 | gusto | and i will get a heart attack soon when this goes on |
| 01:04.46 | slav3_kitten | pace maker? |
| 01:04.56 | gusto | because 800 mhz should be left for dvb-t until channel 69 like it was all time, but no, that crazy ppl needed to open up another band for LTE |
| 01:05.14 | gusto | no |
| 01:05.26 | gusto | heart attack because where will this end? |
| 01:05.40 | slav3_kitten | same place it always end |
| 01:05.44 | slav3_kitten | s |
| 01:05.54 | gusto | what comes next? 700 mhz? 600 mhz? maybe the 400 mhz for OFDM-FLASH already? |
| 01:05.57 | slav3_kitten | with someone getting screwed but the masses not caring |
| 01:06.30 | gusto | someitimes i do not know what these regulators are paid for |
| 01:06.31 | slav3_kitten | so long as they get cat videos on their phones or something else to placate them they are good |
| 01:06.35 | gusto | but i like the czechs |
| 01:06.45 | gusto | they opened up 1800 mhz for LTE |
| 01:06.51 | slav3_kitten | i've never been outside of the US |
| 01:06.52 | gusto | that's like GSM's E band |
| 01:06.56 | gusto | so nothing new |
| 01:06.58 | gusto | and that works |
| 01:07.15 | slav3_kitten | unless you consider canada out of the country but i lived on a border of it so it does not count |
| 01:07.24 | gusto | ah |
| 01:07.27 | gusto | canada is nothing |
| 01:07.37 | gusto | they are even more fsckd up than the US |
| 01:07.45 | slav3_kitten | even to canadians :D |
| 01:07.46 | gusto | they have like a lot higher telephone bills |
| 01:08.05 | gusto | i do not understand it |
| 01:08.25 | gusto | the canadians have space and they could easily invest and connect through greenland |
| 01:08.43 | gusto | even cubans have fibre optic cable from venezuela now |
| 01:09.50 | gusto | sometimes i wish to just take all this technology away from the ppl and roll it out again and that time right |
| 01:11.26 | gusto | and it's not only telephones and internet. for example now austrian army bought some new eurofighters and they are so fucked up that not even the name for austria is written right on then |
| 01:11.30 | gusto | them |
| 01:12.24 | gusto | now the austrians are considering sending them back to EADS |
| 01:14.21 | gusto | so some companies here are too dumb to produce an ordinary fighter jet, but the americans are no better, they un-engineered their 787 dreamliner at boeing as well, that's also a shame |
| 01:15.12 | slav3_kitten | are are good at making weapons of war however |
| 01:15.33 | gusto | no |
| 01:15.58 | slav3_kitten | how do you figure? |
| 01:16.53 | gusto | because there is not one american weapon that would be somehow successful at the market ... maybe ratheon tomahawk would be a good one, but that's it then |
| 01:17.06 | slav3_kitten | i didn't say they were fantastic, we just excel at producing them in large numbers |
| 01:17.12 | gusto | yes |
| 01:17.27 | gusto | but that for the US army, which has the biggest military budget |
| 01:17.39 | slav3_kitten | and our M16 is a very successful small arm |
| 01:17.48 | gusto | so they do not need to produce good things because they do not have any of competition |
| 01:17.53 | SeRi | who fucking cares. we kick ass. bottom line. |
| 01:18.04 | gusto | hmm, not really |
| 01:18.16 | gusto | who was kicked in the ass by the US lately? |
| 01:18.25 | SeRi | Than come and take it |
| 01:18.30 | slav3_kitten | not quite AK level of success, but we were more discerning of whom we sold tooling for |
| 01:18.32 | slav3_kitten | well to sold it to |
| 01:18.36 | gusto | considering it :-D |
| 01:18.58 | slav3_kitten | gusto, afghanistan, bin laden, iraq, sadam |
| 01:19.02 | gusto | well |
| 01:19.22 | gusto | afghanistan is not a success ... that's like a shame that there are like no results after such a long time |
| 01:19.24 | SeRi | your dad, your uncle, etc... |
| 01:19.36 | gusto | bin laden ... did he ever exist? |
| 01:19.48 | slav3_kitten | gusto, afghanistan kicked the ass of russia, we are still getting shit done |
| 01:19.54 | gusto | sadam was killed by his own ppl in the end |
| 01:20.07 | SeRi | what a moron you are. |
| 01:20.09 | gusto | and iraq ... what is going on there? nothing ... like the dog died |
| 01:20.32 | gusto | slav3_kitten: maybe russians knew when it's enough |
| 01:20.34 | slav3_kitten | who me or gusto? |
| 01:20.58 | gusto | slav3_kitten: do not care |
| 01:21.00 | SeRi | slav3_kitten: are you insecure? |
| 01:21.06 | SeRi | who is not making sense here? |
| 01:21.21 | slav3_kitten | gusto, they knew that our SAM missiles we were supplying the afghani people with were dropping their planes an choppers like flies |
| 01:21.22 | gusto | american army hunting ghotst |
| 01:21.24 | gusto | ghosts |
| 01:21.42 | slav3_kitten | SeRi, i am. but i was going more for a joke lol |
| 01:21.51 | SeRi | lol |
| 01:21.54 | slav3_kitten | which failed epic style |
| 01:21.59 | SeRi | rofl |
| 01:22.11 | slav3_kitten | no need to pander to me now |
| 01:22.40 | gusto | slav3_kitten: so why is US army still there? |
| 01:22.42 | SeRi | ha |
| 01:22.57 | SeRi | because they are looking for you |
| 01:23.08 | gusto | well, they are looking at the wrong place |
| 01:23.26 | slav3_kitten | we are actually supposed to be pulling out in 2013 |
| 01:23.40 | slav3_kitten | but it'll likely take a year to fully withdraw |
| 01:23.45 | gusto | i do not fear an army whose ppl are sending p0rn mails working for the CIA then :-D |
| 01:23.52 | gusto | that's like no discipline at all |
| 01:24.14 | slav3_kitten | everyone looks at porn |
| 01:24.17 | gusto | yes |
| 01:24.35 | wonderworld | no institution killed more people than the US Army in the last 10 years. if thats "kicking ass" you are right |
| 01:24.38 | wonderworld | i think it sucks |
| 01:24.44 | gusto | but of course ... the american army is not the only one in afghanistan, for the record |
| 01:27.25 | gusto | there are more armies i make fun of ... but my favourite is the german bundeswehr ... they do not have ppl to free some ships taken over by pirates ... they need the netherland's army for that as well :-D i should make a top 10 of funniest armies ... and i am sure the american one would be among them as well :-D |
| 01:27.47 | SeRi | wtf has this channel has turned in to? |
| 01:27.57 | gusto | no, we are only talking |
| 01:28.07 | wonderworld | like #politics on EfNet |
| 01:28.23 | SeRi | no you are not. You are starting a flame war and others are following |
| 01:28.23 | gusto | went from VoIP over Wlan to Wlan alternatives to US army ;-) |
| 01:28.29 | slav3_kitten | is g729 really cpu heavy? |
| 01:28.37 | gusto | no |
| 01:29.01 | slav3_kitten | hmmm wonder why when switching to 729 call quality decreased i've read it's supposed to be better |
| 01:29.17 | gusto | because g729 is not a good codec |
| 01:29.42 | gusto | g722 is a good one |
| 01:29.47 | slav3_kitten | well i've got ulaw or 729. ulaw is uncompressed |
| 01:30.12 | gusto | i prefer g722 and g711 |
| 01:30.42 | gusto | and of course alaw over ulaw |
| 01:31.03 | wonderworld | gusto: why? |
| 01:31.10 | gusto | because it works |
| 01:31.17 | SeRi | He s probably in EU/UK |
| 01:31.20 | wonderworld | ulaw doesn't ? |
| 01:31.35 | gusto | ulaw is OK too, but why would you use it? |
| 01:31.46 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 01:31.54 | slav3_kitten | ulaw is US, alaw is UK/EU from what i understand |
| 01:32.01 | gusto | 64 kbps is better than 56 kbps |
| 01:32.03 | SeRi | wonderworld: alaw is EU/UK/ETC ulaw is NA |
| 01:32.15 | gusto | yes, that too |
| 01:32.36 | gusto | but generally you can use both, because the most VoIP providers are supporting both |
| 01:32.50 | SeRi | indeed |
| 01:33.23 | gusto | so for VoIP i would prefer the one with better quality |
| 01:33.47 | wonderworld | i'd say it depends on the circumstances |
| 01:33.59 | gusto | and that ulaw 56 kbps is because of historical reasons from what i understand |
| 01:34.32 | gusto | the americans have some differnt telephone network, so you need ulaw codec only when you are pulling it into the PTSN |
| 01:34.56 | gusto | and the same is true for EU when you are pulling to the PSTN you would need alaw |
| 01:35.08 | gusto | right? |
| 01:35.18 | slav3_kitten | what's the point of voip if it does not terminate to a PTSN |
| 01:35.18 | SeRi | yes. |
| 01:35.29 | gusto | hmm |
| 01:35.31 | wonderworld | ulaw / alaw is nice if you need to care about CPU transcoding cost |
| 01:35.32 | gusto | most of them do not |
| 01:35.36 | SeRi | !237 |
| 01:35.45 | SeRi | wrong window |
| 01:35.51 | gusto | you have a lot of VoIP lately .. and of course also mobile operators |
| 01:36.01 | gusto | and there it goes to GSM anyway |
| 01:36.15 | slav3_kitten | gusto, so much is still pstn however |
| 01:36.25 | gusto | yes it is |
| 01:36.42 | SeRi | dijib: you back? |
| 01:36.43 | gusto | but is becoming less important lately |
| 01:37.12 | gusto | for example here in germany i am among the few ones who still has a analogue telephone |
| 01:37.24 | gusto | but i am not using it, because it's too expensive |
| 01:38.28 | gusto | calling from it ... however i could make it work so that someone can call me on it, but i would need an FXO? on that (is that FXS or FXO??) |
| 01:38.28 | slav3_kitten | that's why i had our line disabled thus negating my need for fxo card an vic nm in the router |
| 01:38.53 | gusto | yes |
| 01:39.07 | slav3_kitten | fxo connects to a pstn, fxs hooks to phones |
| 01:39.15 | gusto | so FXO |
| 01:39.21 | gusto | i have only 2xFXS |
| 01:39.36 | slav3_kitten | i think i have that the right way... |
| 01:39.54 | slav3_kitten | cisco router with fxs cards? |
| 01:40.05 | gusto | but there is a linksys/cisco ATA adapter with 1x FXS and 1x FXO for 50 eur around, but i have no money and who sould call me anyway? i mean .. the VoIP number works as well, so ... ?? |
| 01:40.47 | *** join/#asterisk droemel (~droemel@p4FCAD33E.dip.t-dialin.net) |
| 01:40.48 | gusto | i would not get any benefit of that |
| 01:41.03 | slav3_kitten | 911 service |
| 01:41.15 | slav3_kitten | or the eu counterpart |
| 01:41.20 | wonderworld | you get that with voip in germany as well |
| 01:41.24 | gusto | yes ... calling the police? no way |
| 01:41.27 | wonderworld | if you have a german voip provider |
| 01:41.45 | slav3_kitten | fair chance that your voip provider could be fail while pstn works |
| 01:41.48 | gusto | i have enough trouble i do not need some amateurs to get in the way |
| 01:42.18 | wonderworld | they are obligated to offer emergency services by law |
| 01:42.26 | gusto | when the VoIP provider fails, like he did 2 days ago ... i do not call |
| 01:42.41 | gusto | i can survive without a telephone for some time i am not a girl |
| 01:43.03 | slav3_kitten | wonderworld, you and when your network connection is down due to disaster but pstn still works? how does that help you |
| 01:43.22 | wonderworld | it doesn't but i use my cell 9% of the time anyway |
| 01:43.27 | wonderworld | 99% even |
| 01:43.38 | gusto | emergency service ... i mean ... going to some german hospital and get some new kind of a virus that only exists in hospitals, no .. thanks |
| 01:43.57 | slav3_kitten | first thing that goes to hell in a disaster, cell network... |
| 01:44.12 | wonderworld | ok, i am doomed :) |
| 01:44.26 | gusto | yes |
| 01:44.33 | gusto | but not only in a desaster |
| 01:44.54 | gusto | in slovakia they measured the availability of emergency services on cell networks |
| 01:44.59 | wonderworld | but i can't rememember, that the cell network here went don't longer than an hour... |
| 01:45.03 | slav3_kitten | disaster it's likely to fail like voip, cell, pstn. |
| 01:45.04 | gusto | no provider succeeded |
| 01:45.05 | wonderworld | not many desasters in germany |
| 01:45.07 | gusto | all failed |
| 01:45.18 | slav3_kitten | which is where ham radio comes into its own |
| 01:45.25 | gusto | and that was not even during a desaster, that was like in a good weather |
| 01:46.22 | gusto | in germany, they better not try, because when they would, there would have to be some consequences and then all mobile network operators would scream that germany is not enterprise-friendly and no one wants that to happen |
| 01:46.47 | gusto | of course, we are back at ham radio |
| 01:47.02 | slav3_kitten | not my fault |
| 01:47.25 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 01:47.29 | gusto | we are not living in a people friendly environment |
| 01:48.11 | gusto | today it was on the news that some guy here in germany called the police and they came and booked him ... |
| 01:48.35 | gusto | ... so i am not calling anyone |
| 01:49.12 | gusto | still wonders why there is no asterisk11 version on freebsd-ports |
| 01:51.00 | gusto | someone here using h323 regularly? |
| 01:51.11 | slav3_kitten | wonders why gusto can't just compile it for bsd |
| 01:52.48 | gusto | of course i can compile it |
| 01:53.28 | wonderworld | is too lazy for freebsd |
| 01:53.32 | gusto | i am just wondering, because freebsd-ports have always like the new software packets in, so maybe they just do not care or something ... i do not know the maintainers of that, so i can not tell |
| 01:54.21 | gusto | on my freebsd box a 1.8.18 so the current 1.8 version is running and it was johnny on the spot there, as soon as the version came out |
| 01:54.51 | gusto | or about that same time ... not exactly the same day maybe, but soon after, i do not know |
| 01:55.01 | gusto | asterisk has a new website |
| 01:57.07 | SeRi | ok I just migrated my Arc instance with asterisk to ESXi. |
| 01:57.10 | SeRi | All ok so far |
| 01:57.26 | gusto | oh |
| 01:57.29 | gusto | ESXi |
| 01:58.03 | *** join/#asterisk coppice (~chatzilla@m121-202-21-36.smartone.com) |
| 01:58.20 | SeRi | I meant to say Arch* |
| 01:59.36 | gusto | hm, an arch - fan |
| 01:59.37 | gusto | :-D |
| 02:00.20 | gusto | however, i just took a look at the ESXi wikipedia page and there they write that ESXi only runs on x86(_64) |
| 02:00.42 | SeRi | gusto: yeap. Host is 64 bit and guest is 32 bit |
| 02:00.53 | gusto | so somehow i do not see the benefit of running ESXi instead of i have no idea ... VirtualBox or ... even KVM |
| 02:01.04 | *** join/#asterisk elico (~Thunderbi@79.181.215.110) |
| 02:01.33 | SeRi | not an arch fan nor a fan of any os |
| 02:01.46 | SeRi | os of choice is slackware |
| 02:02.04 | wonderworld | server and desktop? |
| 02:02.20 | SeRi | desktop |
| 02:02.34 | SeRi | server is arch at home and RHEL5 and 6 at work |
| 02:03.06 | gusto | yes, slackware is cool, you have one + point :-D |
| 02:03.07 | SeRi | and solaris and all kinds of other unix flavors |
| 02:03.09 | slav3_kitten | debian or slack for me |
| 02:03.18 | wonderworld | i've been doing debian only for servers since '98 or so |
| 02:03.32 | wonderworld | desktop doesn't really matter i guess |
| 02:03.43 | gusto | there are two things i like about slackware |
| 02:03.49 | wonderworld | i am on xubuntu, but could be on any other OS |
| 02:04.04 | gusto | 1.) easy and simple RC 2.) all utilities on it after install |
| 02:05.05 | SeRi | easy and simple? I wouldnt say so. you must never run slack before. |
| 02:05.16 | slav3_kitten | i really need to get an SD card for my pi so i can do raspberry pi bpx |
| 02:05.20 | slav3_kitten | pbx even |
| 02:05.25 | SeRi | easy and simple for those of us who are advance |
| 02:05.46 | SeRi | put dijib or slav3_kitten behind slack and they will cray until they are milked by their mom |
| 02:05.46 | wonderworld | i ordered a pi and never got it. ripoff |
| 02:06.03 | SeRi | wonderworld: that sucks man |
| 02:06.09 | slav3_kitten | SeRi, what? |
| 02:06.11 | SeRi | they are now shipping with 512MB of RAM |
| 02:06.15 | SeRi | I have the 256 v2 |
| 02:06.28 | slav3_kitten | i have a v1 |
| 02:06.30 | SeRi | slav3_kitten: a joke... relax |
| 02:06.32 | wonderworld | i really want one. i think it could replace my homeserver easily |
| 02:06.38 | SeRi | It can |
| 02:06.40 | slav3_kitten | SeRi, but i use slack |
| 02:06.49 | wonderworld | which is a EEE-pc at the moment |
| 02:06.50 | slav3_kitten | i even said so earlier |
| 02:06.57 | SeRi | It replace my gallery, nagios, ssh server |
| 02:07.02 | SeRi | wonderworld: ^^ |
| 02:07.33 | SeRi | slav3_kitten: dont lie. is ok. you dont have to lie to fit in. |
| 02:07.42 | SeRi | lol joking |
| 02:07.57 | slav3_kitten | lol thanks to you i just spewed coke out of my nose |
| 02:08.08 | slav3_kitten | brb i need a damp towel |
| 02:08.40 | wonderworld | no asterisk question in 1 hour.... never saw that in this channel |
| 02:08.52 | wonderworld | world is ending |
| 02:09.14 | gusto | but that could be also a good sign, or? |
| 02:09.32 | gusto | i mean ... no question, so everything runs OK, sofar |
| 02:10.14 | gusto | however, it's often not about asterisk and more about VoIP in general here |
| 02:11.21 | wonderworld | for me there is always something new to learn about * |
| 02:11.36 | wonderworld | i could read all day and still learn new things all the time |
| 02:11.54 | wonderworld | it's incredible how flexible it is |
| 02:12.10 | gusto | of course there is ... like for all software ... i for example learned yesterday, that there is sockstat command on freebsd |
| 02:12.43 | gusto | i was looking for some alternative to netstat, i wanted someting more linux like so i searched the manpages with apropos and i found it |
| 02:13.52 | gusto | i would not call asterisk much flexible ... i mean, of course you can make it small, but not until you compile it yourself, because with only taking modules out you will not reach a lot |
| 02:14.29 | gusto | so i could imagine it to be even more flexible, but that can be in the new versions, like asterisk11 i haven't tried it yet |
| 02:15.02 | gusto | maybe i should do it now :-D |
| 02:16.01 | wonderworld | i wasn't talking about making it small. it's flexible in functionality. you can script the hell out of it. it has flexible interfaces to attach nearly every database or software to it. |
| 02:16.12 | wonderworld | i think there is not much in telephony, you can't do with * |
| 02:16.43 | gusto | yes |
| 02:17.07 | gusto | asterisk does a great job of covering a lot |
| 02:17.36 | wonderworld | 3am. time to sleep..bye |
| 02:18.01 | gusto | i have 3 am here too, but that does not mean that i ll go to sleep |
| 02:27.01 | SeRi | p3nguin: You got a few secs? Looking for some help and advice |
| 02:29.16 | dijib | no |
| 02:29.31 | dijib | yeh digium res fax compatible with 11? |
| 02:29.34 | dijib | compile? |
| 02:31.56 | gusto | hm, fax is also on my todo list to try with asterisk |
| 02:36.07 | slav3_kitten | i've thought about doing the sms thing but i'm not entirely sure how that'd function andi have no real need |
| 02:38.36 | gusto | well, SMS is not really useful |
| 02:38.55 | slav3_kitten | i could see it being a nifty lil "page" type thing |
| 02:39.04 | slav3_kitten | but that's all i can honestly see it used for |
| 02:39.14 | gusto | no |
| 02:39.24 | gusto | the problem is that providers do not support it |
| 02:39.46 | gusto | and the most telephone PSTN providers as well, they do not support sending SMS in and out |
| 02:40.16 | gusto | so that would be only considerable with some Huawei like UTMS dongles |
| 02:46.18 | gusto | http://areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 someone using this one? |
| 02:46.23 | gusto | asterisk-stat |
| 02:46.36 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:49.27 | SeRi | Linux oscuridad 3.2.29 #1 SMP Mon Sep 17 13:38:05 CDT 2012 x86_64 Intel(R) Core(TM) i5-2500K CPU @ 3.30GHz GenuineIntel GNU/Linux |
| 02:49.42 | gusto | what's that? slackware? |
| 02:49.55 | SeRi | Yes. |
| 02:50.25 | SeRi | as much as I love my RAT 9 Wireless the battery life sucks |
| 02:50.38 | p3nguin | seri: What do you need? |
| 02:51.52 | gusto | that's a pitty not to see more ppl with native IPv6 addresses instead of this 2001:470: ones from HENET |
| 02:51.58 | SeRi | p3nguin: ESXi network help. |
| 02:52.08 | p3nguin | Ask your question. |
| 02:52.28 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 02:52.59 | SeRi | vmnet0 is dedicated to management and can not be used with other guest correct? |
| 02:53.26 | SeRi | vmnet1 on vswitch1 can be used by several guest correct? |
| 02:53.38 | p3nguin | Hmm. My IPv6 address doesn't seem to start with that. |
| 02:53.48 | p3nguin | Oh, right, it's not from HE.net |
| 02:54.14 | Maliuta | p3nguin: not every has native v6? |
| 02:54.33 | SeRi | your thoughts p3nguin ? |
| 02:54.48 | Maliuta | dual stack all the way, static networks of both 4 and 6. /29 and /56 |
| 02:55.01 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 02:56.35 | *** join/#asterisk mintos (~mvaliyav@114.143.164.176) |
| 02:56.57 | *** join/#asterisk ChrisInSydney (~Administr@202-129-83-200.perm.iinet.net.au) |
| 02:57.13 | ChrisInSydney | g'day all |
| 02:57.21 | SeRi | hola |
| 02:57.28 | ChrisInSydney | 'sup ? |
| 02:57.46 | p3nguin | seri: I'm not totally sure what vmnet0 is, but I have my LAN and management network both on vSwitch0, and all the VMs are attached to vSwitch0 since it is the LAN switch. |
| 02:58.02 | slav3_kitten | what's a good key combination to block caller id. i was thinking *#number |
| 02:58.27 | ChrisInSydney | here its 1831 on the PSTN network |
| 02:58.36 | p3nguin | I think I use *34 to add the last caller to the blacklist. |
| 02:58.40 | ChrisInSydney | 1831+FNN |
| 02:59.53 | ChrisInSydney | I would probably go for *46 = *HN = Hide Number |
| 03:00.02 | slav3_kitten | oh that's a great idea! |
| 03:00.03 | SeRi | p3nguin: I was reading on some site that the management interface can not be shared. I had the same thoughts as you where they all can exist in one place but I was wrong.... As soon as I fired up the VM it nocked the vspehere off line |
| 03:00.10 | SeRi | p3nguin: can you uri? |
| 03:00.14 | ChrisInSydney | *461 hide number on |
| 03:00.18 | ChrisInSydney | *460 Hide number off |
| 03:00.24 | p3nguin | As well, I have *69 to hear the last number that called and it offers a blacklisting option. |
| 03:00.38 | ChrisInSydney | *467 = *HNS = Hide Number Status = pays back whether its on or off |
| 03:01.00 | slav3_kitten | ChrisInSydney, way beyond my skill level at this point |
| 03:01.21 | navaismo | coreyf1513, one of my issues with queuemetrics was manager permission but still seeing "Agent Status cannot be determinated" still searching to fix it |
| 03:01.40 | slav3_kitten | only complicated dialplan thing i have is the wakeup call thing and i had to piece it together from the net before i actually fully understood what was going on then modified it more |
| 03:01.49 | p3nguin | seri: Let me look to see if I can figure out what vmnet0 actually is. I don't see it on the network page. |
| 03:01.49 | ChrisInSydney | slav3_kitten: Sets an astDB (or mysql) thingy you can check against |
| 03:02.15 | slav3_kitten | ChrisInSydney, i figured. just have no db experience yet |
| 03:02.39 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 03:02.49 | ChrisInSydney | Its pretty simple using astdb |
| 03:03.31 | SeRi | p3nguin: I just named my interface like that is a nic |
| 03:03.36 | SeRi | inside a vswitch |
| 03:03.39 | ChrisInSydney | Set(DB(key/name)="something") |
| 03:03.44 | slav3_kitten | ChrisInSydney, not quite gotten to that point at this moment. but i'm working my way through the book :) |
| 03:03.51 | SeRi | p3nguin: vmnic0 |
| 03:03.59 | ChrisInSydney | Then to read back ${DB(key/name)} |
| 03:04.41 | ChrisInSydney | slav3_kitten: I have to go back through the book too. All this stuff I have forgotten and new stuff I'm not too sure about |
| 03:05.13 | ChrisInSydney | Also moving all of our astdb stuff to MySQL so we can use a web app to set features for the users |
| 03:05.56 | ChrisInSydney | anyway, back to work. Just got back from lunch at the pub. I won $10.20 on a $6 bet on the greyhounds !! :D |
| 03:06.13 | SeRi | well my pi had to come off line. I need ports |
| 03:06.17 | ChrisInSydney | $4.20 profit. Thats a coffee and about 12 minutes parking |
| 03:06.24 | SeRi | shit I should of bought that damn switch |
| 03:06.38 | p3nguin | should HAVE |
| 03:06.44 | SeRi | lol |
| 03:06.46 | slav3_kitten | what i really need to figure out though is how to have an incoming sip dial extensions while allowing specific caller ID numbers to do outbound. oh and have my extension for my room call my cell if i don't pick up, then if i don't answer cell transfer to asterisk voice mail |
| 03:06.46 | p3nguin | there is no "should of" |
| 03:06.54 | p3nguin | does not exist in any language. |
| 03:07.57 | ChrisInSydney | explain the outbound some more. Not too clear on what you need |
| 03:08.28 | ChrisInSydney | he should of used "should have" |
| 03:08.42 | ChrisInSydney | p3nguin ;-) |
| 03:09.09 | p3nguin | should HAVE |
| 03:09.13 | p3nguin | there is no "should of" |
| 03:09.34 | slav3_kitten | my english is fail sometimes |
| 03:10.07 | slav3_kitten | like ChrisInSydney if i want to call in to the house on my cell, then dial out to a UK number via it |
| 03:10.30 | slav3_kitten | thus avoid the insane international pricing of my cell carrier |
| 03:10.40 | ChrisInSydney | ahh |
| 03:10.42 | ChrisInSydney | simple |
| 03:10.44 | ChrisInSydney | ish |
| 03:11.29 | slav3_kitten | i'm still unsure of the whole contexts thing so yea. i was all proud of myself that i got it dialing out even though it's not correctly working |
| 03:11.42 | ChrisInSydney | GotoIf($[CALLERID(num)=yournumber]?ivrcontext,ivrexten,1) |
| 03:11.43 | slav3_kitten | but i am going to drive to the ISP an see if i can hammer it out |
| 03:12.35 | slav3_kitten | and i've triple checked that i have nat service helpers turned off on my router |
| 03:12.40 | slav3_kitten | so it's all on my ISP at the moment |
| 03:12.41 | ChrisInSydney | put a pin in just to make sure |
| 03:13.10 | SeRi | p3nguin: something is really fucked up... lol as soon as the vm comes up it nocks the whole thing off line |
| 03:13.16 | ChrisInSydney | check caller ID, then go through to a pin |
| 03:13.16 | slav3_kitten | i've not yet got an ivr or mail boxes setup |
| 03:13.34 | ChrisInSydney | must get get back to work |
| 03:14.19 | ChrisInSydney | should of done that 10 minutes ago |
| 03:14.27 | ChrisInSydney | p3nguin: :p |
| 03:14.44 | slav3_kitten | project tomorrow is get outbound to the UK working tomorrow via isp sorted out. then i plan on setting up the voicemail boxess |
| 03:16.23 | ChrisInSydney | wonders what p3nguin would say if i "aksed" him a question |
| 03:16.27 | ChrisInSydney | C yas all |
| 03:16.39 | p3nguin | If you ask, I would answer. |
| 03:18.43 | ChrisInSydney | p3nguin, I'm sorry, I was trying to illicit a response on my use of bad english, such as "should of" "aks" "youse" etc. No matter. |
| 03:19.46 | ChrisInSydney | have fun all |
| 03:20.12 | p3nguin | Certain things, like "should of" just simply do not make any sense. Aks was determined to be a typo for ask, so I overlooked it. |
| 03:21.38 | ChrisInSydney | thats ok, its late..for some |
| 03:22.29 | ChrisInSydney | back to work for me. Cheers :-) |
| 03:22.37 | navaismo | coreyf1513, aaand its done, damm it manager permission and bad agent setup, all the f@%&# day thanks anyway |
| 03:33.23 | SeRi | p3nguin: not a go |
| 03:33.27 | SeRi | the thing just crashes. |
| 03:36.39 | ChrisInSydney | got a MySQL / ODBC / SP performance question. Just throwing this out there. |
| 03:37.19 | ChrisInSydney | Can anybody advise me if there is any potential problems with calling Stored Procedures from func_odbc.conf |
| 03:37.53 | ChrisInSydney | In theory, you can do it, but are there any "gotchas" that people haev found ?? |
| 03:38.48 | p3nguin | seri: I don't know what to say. You see how I have my vnet set up. |
| 03:39.03 | p3nguin | The management interface and the LAN are on the same vswitch. |
| 03:39.35 | SeRi | p3nguin: I dont think is that. the whole system crashes. |
| 03:39.40 | SeRi | something else is having issues |
| 03:40.09 | SeRi | yea I dont think is that |
| 03:40.39 | dijib | what are you guys working on? |
| 03:46.47 | slav3_kitten | dijib, a girl i'm interested in |
| 03:47.03 | SeRi | dijib: ESXi |
| 03:47.04 | slav3_kitten | she's not fancying the tech talk |
| 03:55.49 | dijib | still that VM server eh |
| 03:56.00 | dijib | im so not a famale |
| 03:56.01 | dijib | fe |
| 03:56.18 | dijib | i dont even know what im talking about slav3_kitten |
| 03:56.33 | dijib | i need to upgrade my digium fax but i fail |
| 03:57.09 | slav3_kitten | dijib, i'm talking with a female about electronics |
| 04:00.28 | dijib | who is that? |
| 04:00.52 | dijib | Loaded res_fax_digium.so => (Digium G.711 and T.38 FAX Techno logies (optimized for core2_64)) |
| 04:01.02 | *** join/#asterisk corretico (~luis@190.211.93.11) |
| 04:06.43 | dijib | what can i do with webmin.tgz in http://downloads.asterisk.org/pub/telephony/asterisk/webmin/ |
| 04:06.56 | *** join/#asterisk fritz09 (~Adium@2002:542e:5694:e472:713d:520:69b2:8704) |
| 04:10.32 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:13.01 | dijib | i think i fixed the fax issue finally |
| 04:13.08 | dijib | not sure about all the emailng it does |
| 04:17.06 | slav3_kitten | dijib, a chick on another network |
| 04:19.32 | p3nguin | moo |
| 04:20.12 | dijib | shoot that cow! |
| 04:20.33 | dijib | p3nguin: can i spam u in a pm? |
| 04:20.45 | p3nguin | I doubt it. |
| 04:20.53 | dijib | [Nov 13 23:18:58] == Spawn extension (fax-in, fax, 8) exited non-zero on 'SIP/voipms-00000076' |
| 04:20.57 | dijib | [Nov 13 23:18:58] -- Executing [h@fax-in:1] NoOp("SIP/voipms-00000076", "") in new stack |
| 04:21.00 | dijib | [Nov 13 23:18:58] -- Executing [h@fax-in:2] System("SIP/voipms-00000076", "sleep 10") in new stack |
| 04:21.03 | dijib | [Nov 13 23:19:09] -- Executing [h@fax-in:3] Goto("SIP/voipms-00000076", "h-SUCCESS,1") in new stack |
| 04:21.07 | dijib | [Nov 13 23:19:09] -- Goto (fax-in,h-SUCCESS,1) |
| 04:21.09 | dijib | [Nov 13 23:19:09] -- Executing [h-SUCCESS@fax-in:1] NoOp("SIP/voipms-00000076", "") in new stack |
| 04:21.12 | dijib | [Nov 13 23:19:09] -- Executing [h-SUCCESS@fax-in:2] System("SIP/voipms-00000076", "/usr/bin/tiff2pdf -o /var/spool/asterisk/fax/20121113-8009806858-swissarms-1352866462.134.pdf /var/spool/asterisk/fax/20121113-8009806858-swissarms-1352866462.134.tif") in new stack |
| 04:21.16 | dijib | [Nov 13 23:19:09] -- Executing [h-SUCCESS@fax-in:3] System("SIP/voipms-00000076", "/bin/echo "Please see attachment."|/usr/bin/mutt -a /var/spool/asterisk/fax/20121113-8009806858-swissarms-1352866462.134.pdf -s "New fax from 8009806858" -- ") in new stack |
| 04:21.20 | dijib | [Nov 13 23:19:09] -- Executing [h-SUCCESS@fax-in:4] Verbose("SIP/voipms-00000076", "Email Result = APPERROR") in new stack |
| 04:21.23 | dijib | [Nov 13 23:19:09] Email Result = APPERROR |
| 04:21.26 | dijib | [Nov 13 23:19:09] == MixMonitor close filestream (mixed) |
| 04:21.28 | dijib | [Nov 13 23:19:09] == End MixMonitor Recording SIP/voipms-00000076 |
| 04:21.31 | dijib | lol |
| 04:21.33 | dijib | i think you were right |
| 04:21.36 | dijib | made a mess of that sorry guys |
| 04:21.36 | p3nguin | lrn2fuckingpastebin |
| 04:21.38 | dijib | lol |
| 04:21.41 | dijib | god damn copy pasta all over the place |
| 04:21.43 | dijib | you guys are going to give me the banhammer arn't you |
| 04:22.26 | dijib | http://pastebin.com/vBZC2w0R |
| 04:22.51 | p3nguin | So what kind of females like to cyber about electronics? |
| 04:23.52 | dijib | electronic ones |
| 04:24.23 | slav3_kitten | p3nguin, EE ones |
| 04:24.38 | dijib | it doesnt have the email address |
| 04:25.45 | slav3_kitten | she's a bit homely but super smart |
| 04:28.39 | p3nguin | I guess you should have put in your email address. |
| 04:28.51 | p3nguin | Either in the dial plan or in the astdb. |
| 04:28.55 | p3nguin | What is EE? |
| 04:29.12 | dijib | ${DB(email/fax-manager) ts this |
| 04:29.25 | dijib | apparently in upgrading asterisk it doesnt save your databases |
| 04:29.40 | dijib | wait yes it does |
| 04:29.45 | dijib | maybe it was the machine to machine |
| 04:29.51 | p3nguin | database put email fax-manager dijib@suckit.xxx |
| 04:30.03 | dijib | thats your line. what would the database looklike? |
| 04:30.11 | dijib | ty |
| 04:38.39 | AliRezaTaleghani | weinerk: did u check it => http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 04:54.53 | slav3_kitten | if i want to strip the first 3 characters of an exten it's ${EXTEN:3} right? |
| 04:55.10 | *** join/#asterisk radic (~radic@dslb-178-007-129-012.pools.arcor-ip.net) |
| 04:59.20 | ectospasm | ~thebook |
| 04:59.20 | 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 |
| 05:00.50 | ectospasm | slav3_kitten: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.6.3 |
| 05:01.27 | slav3_kitten | ectospasm, i have the book. it is how you do it, i was just quickly working on something an figured i'd double check in here |
| 05:03.12 | ectospasm | well, then, yes, that's how you do it. |
| 05:03.25 | ectospasm | ...but I wanted to use a stronger authority than my own word |
| 05:04.11 | ectospasm | ...and that syntax is available for all channel variables, not just ${EXTEN} |
| 05:04.29 | slav3_kitten | at least the dial plan works without editing it to fix anything |
| 05:05.34 | ectospasm | that's always good |
| 05:05.53 | slav3_kitten | mhmm |
| 05:21.22 | dijib | hmm ndeed |
| 05:28.26 | *** join/#asterisk Dibbler (~Dibbler@109.157.36.151) |
| 05:29.19 | dijib | bloody postfix nightmare |
| 05:29.52 | *** part/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 05:32.49 | SeRi | dijib: postfix is simple |
| 05:32.57 | SeRi | many guides out there |
| 05:32.59 | dijib | yeh well mines broken |
| 05:33.01 | SeRi | google |
| 05:33.12 | SeRi | It was working |
| 05:33.17 | dijib | fix is 95% working its just not sending the email properly |
| 05:33.18 | SeRi | what did you do? |
| 05:33.29 | dijib | i didnt upgrade anything |
| 05:33.34 | dijib | what is its service name? |
| 05:33.47 | SeRi | postfix |
| 05:33.54 | SeRi | service postfix stop |
| 05:33.57 | SeRi | service postfix start |
| 05:34.00 | SeRi | restart |
| 05:34.03 | SeRi | reload |
| 05:34.04 | dijib | ) |
| 05:34.05 | dijib | [root@swissarms mail]# chkconfig | grep postfix |
| 05:34.05 | dijib | [root@swissarms mail]# |
| 05:34.07 | drmessano | init 0 |
| 05:34.14 | SeRi | lol |
| 05:34.31 | dijib | riiight i didnt remove anything. |
| 05:34.39 | SeRi | dijib: the fix is simple. |
| 05:34.47 | dijib | yum install postfix |
| 05:35.20 | dijib | fixed |
| 05:35.21 | SeRi | dd if=/dev/zero of=/dev/sda |
| 05:35.39 | SeRi | That command will take your pain away any day |
| 05:36.13 | dijib | testing |
| 05:36.15 | drmessano | Question: What kind of bear is best? |
| 05:36.20 | dijib | lol |
| 05:36.20 | drmessano | Answer: Black Bear |
| 05:36.27 | dijib | brown |
| 05:36.34 | drmessano | False |
| 05:36.45 | dijib | that command wouldnt do it SeRi i have that raid mirror |
| 05:37.29 | SeRi | o than simple |
| 05:37.47 | SeRi | dd if=/dev/zero of=/dev/md0 |
| 05:37.57 | SeRi | you said lavm? |
| 05:38.05 | SeRi | /s/lavm/lvm/ |
| 05:38.13 | SeRi | fuck fuck fuck |
| 05:38.15 | SeRi | time to sleep |
| 05:38.15 | dijib | lol |
| 05:38.23 | dijib | gnite SeRi |
| 05:38.29 | drmessano | You fake broke his raid |
| 05:38.41 | dijib | testing postfix faxcing now |
| 05:38.48 | SeRi | I cant I need my dd to finish so my pain can go away |
| 05:39.01 | dijib | lol |
| 05:39.38 | dijib | ok here we go, hurry up fax |
| 05:39.42 | dijib | faxrecieve |
| 05:39.55 | dijib | so the f option in recievefax |
| 05:40.05 | dijib | for the t.98 faxgateway. can use that? |
| 05:40.15 | drmessano | wtf is t.98 |
| 05:40.21 | dijib | sorry t.38 |
| 05:40.36 | SeRi | t.69 works better |
| 05:40.37 | dijib | 56k x2? |
| 05:40.47 | dijib | its negotiating at v21 |
| 05:41.32 | drmessano | What app provides receivefax? |
| 05:41.47 | drmessano | oh nm |
| 05:42.21 | drmessano | Asterisk 11 has little elves that handle the faxing |
| 05:42.32 | SeRi | drmessano: wrong. |
| 05:42.38 | SeRi | They are gnome |
| 05:42.41 | SeRi | get it right |
| 05:42.59 | drmessano | I thought we replaced them with unitarians |
| 05:43.22 | SeRi | MhhhMhhh Yea I think I remember the memo. |
| 05:43.42 | dijib | still postfix aint working |
| 05:43.52 | SeRi | your face is not working |
| 05:43.53 | drmessano | That's one hell of a sentence |
| 05:44.02 | SeRi | LMAO |
| 05:44.20 | drmessano | That's a support ticket if I ever heard one |
| 05:44.32 | SeRi | lol |
| 05:44.39 | drmessano | Title: still postfix aint working |
| 05:44.56 | drmessano | Description of issue: Broken is postfix |
| 05:44.59 | SeRi | rofl |
| 05:45.10 | dijib | think its my sudo -u command |
| 05:45.11 | drmessano | Details: Mail dont work |
| 05:45.25 | dijib | english dont working |
| 05:46.06 | SeRi | dijib: yum remove postfix;yum install postfix-for-n00bs |
| 05:46.10 | drmessano | I think its your distro |
| 05:46.16 | drmessano | CentOS sucks |
| 05:46.22 | drmessano | BAM |
| 05:46.40 | SeRi | drmessano: what is better in your world? |
| 05:47.12 | dijib | damn im running out of free faxes per day on the freesites |
| 05:47.43 | drmessano | Debian/Ubuntu, depending on application |
| 05:47.51 | drmessano | I hate RPMs |
| 05:47.53 | drmessano | I hate YUM |
| 05:47.59 | SeRi | I hate Ubuntu |
| 05:48.02 | drmessano | I hate the redhat kernel |
| 05:48.35 | SeRi | I hate canonical |
| 05:48.40 | dijib | ok testing again |
| 05:48.43 | dijib | this crap is lame |
| 05:49.24 | SeRi | drmessano: do you use sudo? |
| 05:49.54 | drmessano | sudo su |
| 05:49.55 | drmessano | lol |
| 05:50.33 | SeRi | I hate you |
| 05:50.38 | SeRi | LOL |
| 05:50.46 | SeRi | http://www.cyborggaming.com/prod/rat9.htm |
| 05:51.01 | SeRi | my new mouse ^^ |
| 05:51.34 | SeRi | works like a charm in Slackware |
| 05:51.51 | SeRi | small tweak in xorg.conf |
| 05:52.15 | SeRi | duces bitches. I am out. dd fixed my life! |
| 05:52.27 | SeRi | lol no more whisky for me |
| 05:52.29 | drmessano | lol |
| 05:53.12 | dijib | if only |
| 05:53.49 | drmessano | I wish I was drinking |
| 06:00.58 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 06:11.46 | dijib | me3 |
| 06:12.01 | dijib | would make things seem like their going better then they always do |
| 06:13.32 | dijib | this is my maillog |
| 06:13.33 | dijib | <PROTECTED> |
| 06:14.51 | *** join/#asterisk fritz09 (~Adium@pop1-1677.catv.wtnet.de) |
| 06:15.07 | dijib | kernel update time. |
| 06:35.52 | *** join/#asterisk gerhard7 (~gerhard7@82.168.115.40) |
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| 07:05.14 | kaldemar | dijib: the timeout part is what you need to fix. how is your MTA configured to use smtp.gmail.com? |
| 07:26.06 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
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| 07:50.08 | *** join/#asterisk bandroid1 (~bandroidx@205.185.117.116) |
| 07:50.29 | bandroid1 | can i use asterisk to make outgoing calls from my sip phone to my google voice number? |
| 07:50.39 | bandroid1 | i was thinking i could setup jabber on asterisk and sip to my phone |
| 07:50.44 | bandroid1 | and link them together |
| 07:51.09 | bandroid1 | i recall seeing jabber as a supported protocol for asterisk |
| 07:51.50 | bandroid1 | basically i use google voice to make calls from my pc with google chat and hate having to use a headset and want to use my sip phone instead, but there is no direct way to do it |
| 07:55.42 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 08:00.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 08:02.18 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:03.43 | mirela666 | Is there any way to hangup Originated call? |
| 08:04.18 | kaldemar | bandroid1: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 08:05.13 | ChannelZ | channel request hangup |
| 08:08.10 | *** join/#asterisk zamba (marius@flage.org) |
| 08:08.52 | *** join/#asterisk zamba (marius@80.89.47.76) |
| 08:22.32 | mirela666 | but I can't know originated channel name, at least i think I don't |
| 08:23.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 08:27.36 | *** join/#asterisk surferboy (~surferboy@41.177.76.231) |
| 08:27.47 | surferboy | hey guys |
| 08:27.53 | surferboy | I'm wondering if someone can help me |
| 08:28.05 | surferboy | I'm a total asterisk beginner |
| 08:28.26 | surferboy | anyone in? |
| 08:28.50 | *** join/#asterisk TimeRider (~steve@188.220.34.144) |
| 08:29.30 | ChannelZ | mirela666: core show channels |
| 08:29.40 | ChannelZ | ~book |
| 08:29.40 | 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 |
| 08:30.25 | surferboy | I'm looking for a way to redirect unanswered calls |
| 08:30.59 | surferboy | ChannelZ, haha |
| 08:31.14 | ChannelZ | Dial() with a timeout, and then Dial() again |
| 08:31.27 | surferboy | where? how? |
| 08:31.51 | ChannelZ | in your dialplan |
| 08:31.53 | surferboy | it's incomming calls to an ext and if they are not answered then it must go to another ext |
| 08:32.13 | surferboy | can this be done with elastix? |
| 08:32.30 | ChannelZ | I have no idea, I don't do GUIs |
| 08:32.35 | WIMPy | surferboy: You have to ask that in #elastix |
| 08:32.59 | surferboy | ChannelZ, ok without the GUI how do I do that then? |
| 08:33.44 | ChannelZ | as I said, you dial with a timeout, and then dial again (or Goto if you like) |
| 08:33.58 | ChannelZ | When the first dial times out, the dialplan continues. Do whatever you like. |
| 08:34.14 | ChannelZ | Same as how you'd send the call to voicemail, etc. |
| 08:34.29 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 08:35.17 | surferboy | ChannelZ, umm... did I mention that I'm really new to Asterisk? |
| 08:35.30 | surferboy | which file do I edit and what do I put into it? |
| 08:35.51 | ChannelZ | extensions.conf |
| 08:35.54 | ChannelZ | but really you should read |
| 08:35.55 | ChannelZ | ~book |
| 08:35.55 | 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 |
| 08:36.19 | ChannelZ | because with no basis of how anything works it's hard to explain. You must take a bit of initiative |
| 08:36.32 | WIMPy | surferboy: Don't edit the config files if you otherwise use a 3rd party configuration tool to do it. It most probably won't like you doing so. |
| 08:36.48 | ChannelZ | brb I gotta reboot my server |
| 08:38.29 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 08:39.47 | mirela666 | ChannelZ: I have my call that has Originated call that I am spying on, and all I want to do is to end that originated call on my Hangup, but I have no info what is the channel name of originated call |
| 08:43.43 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
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| 08:54.12 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:54.37 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
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| 09:05.07 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 09:15.03 | mirela666 | is there a way to retrieve channel name of the originated call, i can use app Originate or Originate through AMI... |
| 09:18.03 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-tlvqkqjjdzzkhbwj) |
| 09:18.39 | *** join/#asterisk Vince-0 (~vincent@105-236-96-84.access.mtnbusiness.co.za) |
| 09:19.53 | ChannelZ | I'd guess AMI generates an event that says but I dunno, I don't use AMI so much |
| 09:20.11 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 09:21.19 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 09:21.42 | WIMPy | I haven;t used it for a ling time, but if it doesn't tell you directly, you will at least see the event where the cannel is created. |
| 09:29.16 | x2s | Hi. Is there a way to disable the pubsub support of the jabber module? I keep running into this bug https://issues.asterisk.org/jira/browse/ASTERISK-18078 |
| 09:29.30 | Rumbles | If I update /etc/asterisk/rtp.conf to have more ports open for RTP connections do I need to reload anything in asterisk for the changes to take effect? |
| 09:30.32 | ectospasm | hmmm... I don't know of any channel driver which will force a re-read of rtp.conf |
| 09:33.08 | ChannelZ | res_rtp_asterisk at least |
| 09:40.28 | WIMPy | module reload res_rtp_asterisk perhaps. But that's just a guess. |
| 09:46.54 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 09:50.00 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:51.16 | fredericve | hi, I have a sip trunk that is capable of g711 and g729. I have a phone that is capable of g711 and g729, but prefers g729. Is there any way to tell asterisk to keep that in mind when negotiating codecs with the trunk? |
| 09:51.40 | *** join/#asterisk fritz09 (~Adium@pop1-1677.catv.wtnet.de) |
| 09:52.27 | ectospasm | fredericve: when listing codecs with "allow", order matters (first listed gets priority) |
| 09:55.59 | fredericve | ectospasm: thanks, but let me elaborate |
| 09:56.10 | fredericve | I would like to prevent asterisk from transcoding |
| 09:57.27 | ectospasm | fredericve: as long as you're not recording the G.729 channels on the Asterisk side, it shouldn't do transcoding (unless a channel is bridged which isn't configured or can't speak G.729) |
| 09:57.32 | fredericve | I have e.g. a phone which prefers g711 and a phone which prefers g729. Is there a way that asterisk can negociate the preferred codec of the phone with the trunk? |
| 09:59.04 | ectospasm | fredericve: no. In the G.711 case you'll do transcoding, since the trunk will prefer G.729 |
| 09:59.31 | ectospasm | ...unless you set two different "trunks" to the same provider, but that gets messy |
| 09:59.50 | ectospasm | ...is there any way the phone preferring G.711 can speak G.729? |
| 09:59.57 | fredericve | ectospasm: indeed. I'm trying to avoid messy :-) |
| 10:00.21 | fredericve | it can speak g729, but prefers g711 |
| 10:00.42 | ectospasm | if you tell Asterisk that it can only speak G.729, you won't do any transcoding |
| 10:01.06 | ectospasm | that means "allow=g729" in sip.conf for that phone. |
| 10:02.01 | ectospasm | the idea that a phone prefers any codec shouldn't force you to use that codec with that phone. |
| 10:03.04 | fredericve | I agree, but g729 is lower call quality, and the customer only wants to use g729 where there's limited bandwidth |
| 10:04.17 | ectospasm | then you're stuck transcoding |
| 10:04.31 | ectospasm | no real way I see around that |
| 10:06.48 | fredericve | we could patch asterisk |
| 10:07.08 | fredericve | we did it for asterisk 1.2 a long time ago |
| 10:09.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:12.26 | *** join/#asterisk Kako_ (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 10:15.19 | ectospasm | good luck with that |
| 10:19.16 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 10:23.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:23.52 | Kako | Hi, I receive every time a huge local_lostpackets value and a 0 on the rtt. In http://www.voip-info.org/wiki/view/Asterisk+RTCP i read that I should "force-enabled jitter buffer or by making sure transcoding is done". Transcoding makes no sence, both side uses alaw.. To enable jitter buffer also doesn't make sence, because it is a CAPI<->SIP connection. So far I know jitter is only for... |
| 10:23.53 | Kako | ...SIP/SIP. So what must I change to get valid values for local_lostpackets and rtt? |
| 10:24.44 | Chainsaw | Kako: Improve connectivity between the Asterisk server and the SIP client. |
| 10:24.48 | ectospasm | Kako: you've got to remember, that both halves are distinct in Asterisk. |
| 10:25.08 | ectospasm | So the CAPI "leg" is separate from the SIP leg. |
| 10:26.03 | Kako | Chainsaw: Both asterisk servers are connected with a crosschannel. How should I improve the connectivity? |
| 10:28.00 | Chainsaw | Kako: The statistics can only be for the SIP side. Work on that. |
| 10:28.01 | Kako | Chainsaw: ISDN/CAPI->Asterisk->SIP---->crosscable--->SIP->Asterisk |
| 10:28.26 | Chainsaw | Kako: Check duplex on both ethernet ports. Replace the cross cable if it isn't CAT5E or up. |
| 10:29.19 | WIMPy | It could be a re-paketization issue. I've seen them is some channeltype combinations. |
| 10:29.36 | WIMPy | Bu I haven't found out what the cause is so far. |
| 10:30.07 | Kako | Chainsaw: The fakt is, that the audio quality is absolutly lost-free and when sending 668 packets it is not possible to loose 2600. So this seems for me not a real problem but a stat problem. |
| 10:30.33 | WIMPy | So what does it sound like? |
| 10:31.50 | Kako | WIMPy: It sound for me that the given result for local_lostpackets have nothing to to with the reality. And I'm searching for the correct value :-) |
| 10:32.05 | WIMPy | The audio. |
| 10:32.24 | WIMPy | Do you have any issues at all? |
| 10:33.25 | Kako | WIMPy: The audio has CD-quality. So actually I've not have an audio issue, but in case a have one I would like to be shure that the stats I log make sence. |
| 10:33.44 | *** part/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 10:35.02 | WIMPy | Ok, it might indeed be due to packetization. I don't know how chan_capi does it, but it might push audio in as fast as possible resulting in Asteris receiving a lot more audio frames than it will |
| 10:35.07 | WIMPy | send out. |
| 10:35.21 | WIMPy | Maybe the stats don;t like that. |
| 10:36.07 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 10:37.03 | Kako | In other words, ther is no way to get a correct stat result? |
| 10:38.00 | WIMPy | I have NFI. Try to stay around and hope someone has internal knowledge of how the stats are calculated. |
| 10:38.39 | Kako | thanks, I'll be here:-) |
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| 11:12.20 | bombev | hi, is there any way to use my Skype Incoming calls to go thru my PBX server? because Skype for asterisk (digium) is no longer available |
| 11:19.49 | bulkorok | not with asterisk... |
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| 11:31.20 | kjs | are there repos for centos 6? |
| 11:36.26 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:37.10 | Kako | kjs: yes, take a look at atrpms |
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| 11:48.10 | kjs | Kako: digum rpms? |
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| 11:53.16 | Kako | kjs: http://packages.atrpms.net/dist/el6/asterisk/ |
| 11:54.02 | Kako | kjs: this rpms are probably not created by digum, in case this was the question... |
| 11:56.09 | Kako | kjs: in case you are searchin for 1.4 (asterisk14), 1.6 (asterisk16) |
| 11:57.43 | Kako | Is there a "howto" page to migrate from 1.6 to 1.8,10 or 11? |
| 11:59.20 | iulhk | using asterisk-10, using realtime asterisk, right now i have just one table for sip registration sippeers, can i add multiple table for sip registration like sip_peers1, sip_peers2, sip_peers3, etc by using one asterisk server ? |
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| 12:05.46 | kjs | Kako: 1.8 i need |
| 12:06.07 | kjs | epel has 1.8 rpms but no freepbx |
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| 12:15.24 | Kako | kjs: on the link you will find asterisk-1.8.3.2-97.el6.x86_64.rpm (but also the i386 version), addons and so on. To use atrpms with yum take a look at http://atrpms.net/documentation/install/ |
| 12:19.02 | kjs | Kako: thank, still on the hunt for a sane freepbx rpm though. |
| 12:19.28 | Kako | kjs: sorry, I've never used freepbx. |
| 12:19.41 | kjs | Kako: yeah it's not my thing either ;) |
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| 12:41.54 | iulhk | can i add multiple databases in one asterisk server ? |
| 12:46.29 | bombev | bulkorok do you think that Skype for PBX will work http://www.mhspot.com/sts/siptosis.html |
| 12:48.39 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-042-022.pools.arcor-ip.net) |
| 12:49.44 | ghost75 | is it possible to have monitor cmd record both in and out to a single file? |
| 12:50.03 | kaldemar | ghost75: MixMonitor is for that. |
| 12:50.30 | ghost75 | i am recording spammers :> |
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| 12:54.34 | bulkorok | bombev: I don't know that... I know that freeswitch has a skype module that works via a skype client |
| 12:56.00 | bulkorok | bombev: for businesses skybe offers "skype connect" |
| 12:56.17 | bulkorok | skybe!? where comes the b from!? |
| 12:58.45 | bombev | bulkorok I know that skype offer skype connect but it is kind of expensive one channel cost 5 EURO monthly |
| 12:59.28 | bombev | and if you have for example 5 sip profiles each sip profile need at least one channel |
| 13:01.08 | bulkorok | this is only one point why skype is really big s*it |
| 13:02.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 13:10.03 | ectospasm | Skype Connect is really the only way to connect Asterisk to Skype now... Skype For Asterisk is almost completely dead (and won't be resuscitated) |
| 13:11.23 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:15.54 | ghost75 | i am recording the call with monitor or mixmonitor and in h context the wav gets mailed - problem is the file is corrupted |
| 13:15.55 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 13:16.02 | ghost75 | anybody wants to look? |
| 13:16.17 | Greenlight | Are you calling stopmixmonitor? |
| 13:16.24 | ectospasm | corrupted...how? |
| 13:16.42 | ghost75 | the mailed file is smaller than the file on disk |
| 13:16.51 | ghost75 | if i listen to file on disk, its ok |
| 13:17.01 | Greenlight | Yea - you're not calling StopMixMonitor by the sounds of it |
| 13:17.07 | ectospasm | ghost75: then (mix)monitor is a red herring |
| 13:17.15 | ectospasm | ...it's a problem with voicemail to email. |
| 13:17.16 | ghost75 | ok then i try with stop |
| 13:17.30 | Greenlight | If he's calling the send, while the file is still being written, that would cause this |
| 13:17.39 | ectospasm | true enough |
| 13:18.01 | p3nguin | You don't need to StopMixMonitor for it to work |
| 13:18.15 | p3nguin | By the time the call is in the h extension, it has already stopped. |
| 13:18.22 | ghost75 | hmm |
| 13:18.29 | ectospasm | Greenlight: but that doesn't explain why he can listen to it on disk and it's fine... |
| 13:18.40 | p3nguin | I also mail a monitor file, so I know this for sure. |
| 13:18.50 | Greenlight | By the time he listens to it, it's been rwitten fully |
| 13:19.10 | Greenlight | p3nguin: I had an issue where file was corrupted because I tried to use before calling StopMixMonitor |
| 13:19.13 | ghost75 | i will just try it |
| 13:19.37 | p3nguin | Maybe you just need to wait .2 seconds before attaching it. |
| 13:19.39 | Greenlight | Indeed - it's worth a shot. If still not working, then as ectospasm said it'll be the voicemail to email part |
| 13:19.54 | ghost75 | wait time i already did add |
| 13:20.45 | ghost75 | its just "Stopmixmonitor" right? |
| 13:21.07 | Greenlight | Yup |
| 13:21.13 | ghost75 | its still broken |
| 13:21.16 | Greenlight | Oh well |
| 13:21.22 | Greenlight | You called that before the sending yea? |
| 13:21.30 | ghost75 | before hangup |
| 13:21.43 | p3nguin | Try the Wait(0.2) before running the mail app like I suggested. |
| 13:21.45 | ghost75 | in agi script 5sec wait time before mail |
| 13:21.46 | Greenlight | You see if get executed in your CLI ? |
| 13:21.54 | p3nguin | oh |
| 13:21.58 | p3nguin | 5 seconds is a while. |
| 13:22.13 | ghost75 | hmm then maybe its postfix |
| 13:23.02 | ghost75 | or because of mutt |
| 13:23.28 | p3nguin | I use mutt to attach the monitor file and mail it to me. |
| 13:23.50 | ghost75 | or its amavis |
| 13:24.04 | p3nguin | Amavis shouldn't mangle the file. |
| 13:24.49 | ghost75 | if i open in hexeditor i see some chars are different |
| 13:25.30 | p3nguin | Are you using amavis on the asterisk box, and it is between asterisk sending the mail and the stmp that is relaying it? Or is amavis on the receiving smtp? |
| 13:25.56 | ghost75 | postfix is passing smtp over amavis |
| 13:26.06 | p3nguin | before it goes outbound? |
| 13:26.16 | ghost75 | yes |
| 13:27.25 | ghost75 | is "df" something special? |
| 13:27.34 | p3nguin | the command? |
| 13:27.45 | ghost75 | its missing in corrupt file |
| 13:28.01 | ghost75 | and @f |
| 13:29.50 | ghost75 | and there is a repeating of a char sequence missing on the end |
| 13:29.50 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 13:30.37 | p3nguin | Can you troubleshoot this, avoiding amavisd, using a different relay, etc.? |
| 13:31.31 | ghost75 | i would need to change Master.cf and remove amavis from there |
| 13:31.55 | p3nguin | It will be temporary. You should try it for troubleshooting purposes. |
| 13:32.09 | p3nguin | Determine if it makes a difference, then revert your changes. |
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| 13:33.07 | ghost75 | ok |
| 13:35.18 | ghost75 | nope .. still kaboom |
| 13:36.49 | ghost75 | and mail is intern, so no relay |
| 13:39.32 | ghost75 | http://pastebin.com/uWxsbF7W <- from asterisk |
| 13:42.20 | *** join/#asterisk ozone89 (c0a7987e@gateway/web/freenode/ip.192.167.152.126) |
| 13:42.28 | p3nguin | I don't see a problem there. |
| 13:42.59 | ghost75 | i had to call mutt with env -i , otherwise was not working |
| 13:44.42 | ozone89 | hello |
| 13:45.27 | bombev | ectospasm so there is no other way to use Skype with my PBX except Skype connect |
| 13:45.29 | p3nguin | That shouldn't affect anything pertaining to file corruption. |
| 13:46.36 | ghost75 | you also send mail from asterisk? |
| 13:47.37 | p3nguin | (0718.40) <p3nguin> I also mail a monitor file, so I know this for sure. |
| 13:48.56 | p3nguin | I use mutt to mail it and ssmtp (previously used msmtp) to get the mail off the box. |
| 13:49.27 | ozone89 | i have a pluscom spgl-11, can i use it to do a voicemail box? |
| 13:50.34 | ghost75 | could be also dovecot i guess |
| 13:50.46 | p3nguin | On the receiving end? |
| 13:52.27 | p3nguin | If you have a gmail account, send your file to your gmail address to eliminate your receiving mail server. |
| 13:53.40 | [TK]D-Fender | ozone89, That device talks to the Skype app directly and lets you plug a phone into it. Asterisk has no means of directly talking to that device (no drivers), and the device itself is pretty much just a dumb audio interface. So I'm not sure where you're going to get any use out of it from... |
| 13:53.55 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 13:54.25 | ozone89 | [TK]D-Fender: okay, thanks anyway, thought it could be used for voicemail :) |
| 13:54.38 | [TK]D-Fender | ozone89, How? |
| 13:56.18 | [TK]D-Fender | ozone89, If you had it plugged into a Windows PC with the full setup it was built for, you COULD get another analog interface that Asterisk CAN use and put it there.... which would work (assuming your device does full signalling properly). It'd be kinda crappy though doing all those extra A>D>A>D conversions |
| 13:56.55 | ozone89 | [TK]D-Fender: didn't know it was skype only |
| 13:57.09 | ozone89 | tought it was just a voice modem |
| 13:57.29 | ozone89 | guess i'll have to buy a card, uh? |
| 13:58.47 | [TK]D-Fender | ozone89, It was made specifically to talk to the Skype app. It says so all over it. If skype isn't involved then what were you thinking of doing with it? |
| 13:59.15 | [TK]D-Fender | ozone89, because where does "voicemail" come in with that thing without some other implicit service attached? |
| 13:59.53 | ozone89 | [TK]D-Fender: that thing has barely the brand and model on it, so i didn't know, you said me, now i know, i said thanks anyway, what else? |
| 14:00.16 | *** join/#asterisk serafie (~erin@nat/digium/x-srjsayuiwiruvimn) |
| 14:00.18 | ozone89 | geez |
| 14:01.30 | *** join/#asterisk fritz09 (~Adium@pop1-1677.catv.wtnet.de) |
| 14:01.54 | [TK]D-Fender | ozone89, You were asking about voicemail and didn't seem to know about the Skype association (I only Googled it after you asked in here). So I'm don't understand what you were thinking with regards to it's involvement with voicemail if that skype association didn't exist. |
| 14:01.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:01.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 14:01.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:02.19 | [TK]D-Fender | ozone89, So what precisely do you want to accomplish? |
| 14:02.21 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:02.31 | ozone89 | I-DIDN'T-KNOW-IT-WAS-SKYPE-ONLY |
| 14:04.54 | [TK]D-Fender | ozone89, That's fine. I just asked what you wanted to accomplish |
| 14:05.54 | ozone89 | i wanted to make a voicemail box, and since that thing has a phone in, line in and usb, i thought it was an ADC for phone |
| 14:06.37 | [TK]D-Fender | So plug your HOME line in and answering machine on that. |
| 14:06.40 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bhfttwippfequgpb) |
| 14:06.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:06.48 | ozone89 | exactly |
| 14:06.54 | asteriskATmarmuD | hi guys. this is my scenario: someone is put into a conference (meetme). then I want to dial a number. the ringing should be heard by the one in the conference. and finally both parties should be speaking to each other in this conference. could someone please give me a hint on the required steps to do that? |
| 14:07.07 | [TK]D-Fender | ozone89, Linksys SPA-3102 <------ |
| 14:08.35 | ozone89 | ok... coulkd it work with skype too? (don't thinks so, just asking for confirm) |
| 14:09.14 | *** join/#asterisk aross42 (~aross@99.224.40.85) |
| 14:10.40 | [TK]D-Fender | ozone89, No. |
| 14:10.43 | ozone89 | ok |
| 14:10.53 | [TK]D-Fender | ozone89, It speaks SIP. |
| 14:11.03 | ozone89 | ok |
| 14:11.37 | ghost75 | p3nguin mutt needs only smtp_pass smtp_url from in .muttrc ? |
| 14:12.20 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 14:13.17 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:17.29 | p3nguin | I only have set copy=no and set record=/dev/null in my .muttrc |
| 14:17.58 | ghost75 | need to auth ... |
| 14:20.04 | ozone89 | [TK]D-Fender: thanks, i'll have a look |
| 14:20.13 | ozone89 | i'll be going now, see ya |
| 14:20.42 | ghost75 | i have it also in gmail corrupted |
| 14:20.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:20.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:21.06 | *** join/#asterisk bchia (~Adium@nat/digium/x-gdauvgqjbqdhueue) |
| 14:21.55 | p3nguin | So then you know where the problem area is. Now you just have to trim down those components until you find the faulty one. |
| 14:23.51 | ghost75 | could be only that file is still not written or mutt |
| 14:24.16 | p3nguin | It sounds like you only have postfix left to troubleshoot. You took amavisd out of the equation, you took your imap/pop3 server out of the equation... |
| 14:24.43 | ghost75 | i took out complete mail server |
| 14:24.51 | p3nguin | Can you use mutt to send to a mail box local to the box, rather than send it to an smtp? |
| 14:25.08 | ghost75 | i tried mutt to gmail and mutt to local mailbox |
| 14:25.21 | p3nguin | with no smtpd involved at all? |
| 14:25.34 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:25.45 | ghost75 | yes only mutt via smtp |
| 14:25.52 | p3nguin | I'm not sure if mutt can do that, but if it can, it's would be worth trying. |
| 14:26.11 | ghost75 | mutt logs in to gmail, no problem |
| 14:26.14 | *** join/#asterisk ChrisInSydney (~Administr@202-129-83-200.perm.iinet.net.au) |
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| 14:28.13 | p3nguin | I guess even for local delivery, it still wants a local smtpd to do the work. |
| 14:28.50 | ghost75 | if i already have tested mutt straight to gmail i dont need further testing on mailserver |
| 14:28.52 | p3nguin | Can mutt simply copy an email to a local directory under its own user account? That would eliminate smtp totally. |
| 14:28.59 | p3nguin | Oh, okay. |
| 14:29.16 | p3nguin | So you're thinking mutt is the problem? |
| 14:29.29 | ghost75 | i will try higher wait |
| 14:29.35 | p3nguin | mutt 1.5.20 |
| 14:29.42 | p3nguin | That's the version I use successfully. |
| 14:29.56 | p3nguin | I don't have any wait time at all. |
| 14:30.40 | p3nguin | The call comes in, starts up MixMonitor(), when the call dies, it jumps to extension h where some basic logic runs System() with echo|mutt in it. |
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| 14:31.47 | p3nguin | Can you test with mail rather than mutt? |
| 14:31.54 | p3nguin | mail or nail |
| 14:32.09 | ghost75 | mail has no attachment feature |
| 14:32.13 | p3nguin | Ah. |
| 14:32.17 | p3nguin | That would be a problem. :/ |
| 14:32.26 | ghost75 | can you give me your mutt cmd line |
| 14:32.51 | ghost75 | not sure it if is 1.5.20 or 1.5.21 |
| 14:33.49 | Katty | hi lads |
| 14:33.59 | ghost75 | Version: 1.5.20-9+squeeze1 |
| 14:34.53 | Katty | infobot: crittercam |
| 14:34.53 | infobot | i guess crittercam is The Birdie Breakfast Buffet!!! http://www.ustream.tv/channel-popup/birdie-breakfast-buffet |
| 14:34.59 | Katty | birdies are out in full force this morning |
| 14:35.03 | p3nguin | same => n,System(/bin/echo "See attachment."|/usr/bin/mutt -a ${MIXMONITOR_FILENAME} -s "New recording" -- ${email-addr}); |
| 14:35.33 | tonikasch | bye |
| 14:35.56 | ghost75 | yes same way |
| 14:36.07 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:36.55 | ghost75 | -d 65535 is debug |
| 14:37.59 | p3nguin | Only if your mutt was compiled with debug. |
| 14:42.30 | Katty | i'll compile your mutt in a minute |
| 14:42.49 | Katty | leifmadsen: 9 months of compiling is a long time!!! |
| 14:42.58 | leifmadsen | o.O |
| 14:43.10 | leifmadsen | oh... heh |
| 14:43.16 | Katty | leifmadsen: i was gonna say...you'll get it tmw |
| 14:43.24 | Katty | have another espresso. |
| 14:43.37 | leifmadsen | Katty: only a single thread and can't distribute the process |
| 14:43.55 | Katty | still worth it. |
| 14:43.59 | Katty | to some. (not me) |
| 14:44.06 | leifmadsen | Katty: this morning I told my wife if our kid has a peanut allergy I'm returning him |
| 14:44.22 | leifmadsen | kind of loves peanut butter and such |
| 14:44.26 | Katty | i hope she swatted you with a cereal box. |
| 14:44.36 | ghost75 | now this is cool, i updated mutt and it doesnt work anymore at all |
| 14:44.45 | jmetro | try having an allergy to grass. makes going outside fun. |
| 14:46.25 | leifmadsen | Katty: ya she wasn't the most impressed :) |
| 14:46.34 | leifmadsen | said she was going to tell on me to the kid! |
| 14:46.44 | leifmadsen | likes to talk to the baby and say, "kick harder" |
| 14:48.45 | dwayne | coppice, is this valid for spandsp? ./configure --prefix=/usr --enable-doc --enable-tests |
| 14:48.59 | dwayne | tests are not building |
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| 15:02.17 | dwayne | coppice, nm... I had some missing dependencies |
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| 15:17.04 | x2s | Hi. Is it possible that JabberSend can't send announce messages? (e.g. messages to example.com/announce/online) |
| 15:17.08 | danfromuk | Hi, I know its strictly not the correct channel for this, however, what's an easy to use asterisk-based pbx like freepbx? Are there any better than freepbx? A client wants to have their own installation and be able to manage the pbx themselves without using our hosted pbx services. |
| 15:17.53 | ghost75 | (15:35:14) p3nguin: same => n,System(/bin/echo "See attachment."|/usr/bin/mutt -a ${MIXMONITOR_FILENAME} -s "New recording" -- ${email-addr}); <- you are not calling mutt with env -i |
| 15:18.03 | p3nguin | That's right. |
| 15:18.35 | ghost75 | when i skip it, mutt doesnt work |
| 15:18.43 | p3nguin | Define "doesn't work." |
| 15:18.55 | ghost75 | i think its using env from root |
| 15:19.51 | p3nguin | In a previous version of Asterisk, I had to use sudo to invoke mutt. What version are you using? |
| 15:20.00 | ghost75 | 1.6.2 |
| 15:21.48 | p3nguin | /usr/bin/sudo -u asterisk /usr/bin/mutt ... |
| 15:21.57 | p3nguin | Configure sudoers so that asterisk can run mutt. |
| 15:22.00 | [TK]D-Fender | danfromuk, Any all-in-one ISO could do. I recommend the FreePBX supplied ISO's so far. |
| 15:22.34 | ghost75 | will try but want to avoid sudoers if possible |
| 15:22.38 | p3nguin | asterisk ALL=(ALL) NOPASSWD: /usr/bin/mutt |
| 15:23.23 | danfromuk | [TK]D-Fender: thanks. i wasn't sure if there were any others. I'll give it a test. |
| 15:23.44 | p3nguin | I'm interested to know what will be the result if you run it via sudo. |
| 15:25.25 | [TK]D-Fender | danfromuk, There are many. AsteriskNOW, Elastix, PIAF, etc |
| 15:25.56 | danfromuk | [TK]D-Fender: any idea which one is the most user friendly? |
| 15:31.48 | *** join/#asterisk RZero (~RZero@85.118.159.250) |
| 15:32.22 | navaismo | danfromuk, all are based on freepbx |
| 15:38.23 | RZero | Hi all, I have a sendfax question, we are working on the sending a fax during a Dahdi call once the user goes through an IVR and selects the options they want, then presses the ready button on their fax machine, I have it trying to send the fax but it times out, I have faxdection set to both in chan_dahdi.conf |
| 15:38.42 | coppice | dwayne: that's right |
| 15:38.43 | RZero | is there some thing Im missing out ? |
| 15:39.42 | [TK]D-Fender | danfromuk, I've already given my recommendation for that |
| 15:41.03 | danfromuk | [TK]D-Fender: thanks. wasnt sure if it was an overall recommendation or one based on useability. |
| 15:41.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:41.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:41.49 | [TK]D-Fender | danfromuk, It is the "purest" implementation to date, and is fairly modern and stable. |
| 15:45.25 | ghost75 | (16:23:55) p3nguin: I'm interested to know what will be the result if you run it via sudo. <- will try it later |
| 15:45.28 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:45.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:46.26 | RZero | this is the output from fax debug + agi debug http://pastebin.com/TG3M0udk |
| 15:48.54 | [TK]D-Fender | <PROTECTED> |
| 15:49.18 | [TK]D-Fender | I would be looking at your DAHDI debug if I were you..... |
| 15:50.25 | RZero | ok cheers TK, not sure why it tries it again though |
| 15:51.20 | *** join/#asterisk coppice (~chatzilla@123203240103.ctinets.com) |
| 15:58.01 | Rumbles | Hi guys, I've got an asterisk 1.8 server with dahdi inbound lines, the call recordings made with mixmonitor are coming through out of sync, I've seen a few posts in issues.digium.com but I'm thinking it's less likely a bug, more likely a config issue, or something up with our Dahdi lines coming in... Anyone know of any good resources which might help me, or any tips? |
| 16:04.44 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 16:10.22 | *** join/#asterisk mtbf (~ewilded@static.124.15.9.5.clients.your-server.de) |
| 16:10.42 | mtbf | <PROTECTED> |
| 16:10.45 | mtbf | <PROTECTED> |
| 16:11.06 | mtbf | it out. But I found the description would fit to 2012-04-27 19:26 +0000 [r364277] Matthew Jordan <mjordan@digium.com> (http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8-current) this issue, however 1.8.13 was released at 2012-06... |
| 16:12.44 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 16:16.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:16.58 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:22.51 | *** join/#asterisk TimeRider (~steve@88-104-147-190.dynamic.dsl.as9105.com) |
| 16:24.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:24.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:27.36 | mjordan | mtbf: that was a very, very subtle problem |
| 16:27.44 | mjordan | mtbf: are you running on a 32-bit machine? |
| 16:33.04 | mtbf | mjordan: nope, it's 64-bit one. |
| 16:36.11 | navaismo | Rumbles, what timing source are you using? |
| 16:39.51 | mjordan | mtbf: not the same problem then |
| 16:39.58 | mjordan | mtbf: that particular problem only affected 32-bit systems. |
| 16:40.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:40.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:41.02 | mjordan | mtbf: it was due to an overflow of a 32-bit long when performing calculations that involved numbers larger than 2^32 (which occurred with certain time difference calculations). On 64-bit systems, a long is typically 64 bits => most likely not the cause of your problem |
| 16:46.22 | Rumbles | navaismo, if I run dahdi_tool it advises internally clocked, is there a way to find out anything further about the timing source? |
| 16:46.55 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 16:47.00 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
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| 17:03.10 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:03.11 | gusto | hi folks |
| 17:03.11 | gusto | what's new? |
| 17:10.03 | *** join/#asterisk JoeyJoeJo (~brian@pool-173-72-191-174.clppva.fios.verizon.net) |
| 17:14.42 | ChannelZ | chicken butt |
| 17:14.47 | ChannelZ | oh no wait, that's not right. |
| 17:15.26 | JoeyJoeJo | I'm looking to move my company away from using skype. What is a good skype-like asterisk client for Windows and OSX that supports voice? |
| 17:16.30 | *** join/#asterisk fritz09 (~Adium@pop1-1677.catv.wtnet.de) |
| 17:16.47 | Chainsaw | JoeyJoeJo: Spark & Wildfire? |
| 17:17.03 | Chainsaw | JoeyJoeJo: http://www.igniterealtime.org/projects/spark/index.jsp |
| 17:17.34 | JoeyJoeJo | Thanks |
| 17:18.44 | fritz09 | hi |
| 17:22.45 | gusto | supports voice? |
| 17:22.57 | gusto | i mean voip apps have to support voice, right? |
| 17:23.45 | gusto | btw. i read on some websites that microsoft had put in a bug in skype using that someone could take over others accounts |
| 17:24.11 | gusto | so that may also be a reason for some finally move away from that *** |
| 17:24.51 | gusto | JoeyJoeJo: do you know that the new versions of asterisk support also the skype codec? |
| 17:25.14 | JoeyJoeJo | gusto: No I didn't |
| 17:25.25 | gusto | SILK or what is it's name |
| 17:25.27 | JoeyJoeJo | So does that mean I could use skype as an asterisk client? |
| 17:26.12 | gusto | you can do VoIP using asterisk and that low bandwidth codec from skype enabeling you to have such low traffic with asterisk as you would have using skype |
| 17:26.31 | gusto | but of course VoIP overhead is bigger, but apart from that |
| 17:26.49 | gusto | JoeyJoeJo: no, not yet |
| 17:27.21 | gusto | JoeyJoeJo: for that, they would also need to implement the skype protocol, so that they can communicate to each other, but there are some projets trying that as well! |
| 17:27.28 | *** join/#asterisk mokmeister (~mokmeiste@109.78.53.118) |
| 17:27.37 | gusto | but they are not far, they are still only trying to get it working |
| 17:27.56 | JoeyJoeJo | gusto: That's ok, the main reason I want to move away from skype is because of its security holes. Mainly the one that came out today about password resets |
| 17:28.30 | gusto | yes, but you have to be careful about asterisk as well |
| 17:28.36 | [TK]D-Fender | Skype is the bastard protocol of the VoIP world. |
| 17:28.47 | [TK]D-Fender | Proprietary and hostile. |
| 17:28.55 | [TK]D-Fender | Get a basic SIP phone. dozens out there. |
| 17:29.00 | JoeyJoeJo | gusto: I know, all software has bugs. But I'm in control of my own server and I can lock it down pretty well |
| 17:29.02 | [TK]D-Fender | ~softphone |
| 17:29.02 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 17:29.03 | gusto | it's not that asterisk is the "security" holy grail or something, but you can do TLS encrypted SIP and such |
| 17:29.17 | [TK]D-Fender | ~zoiper |
| 17:29.17 | infobot | [~zoiper] Zoiper (formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, Mac OS X, and Linux that can be found at http://www.zoiper.com |
| 17:29.18 | Chainsaw | gusto: Anyone ever gotten that working? |
| 17:29.22 | [TK]D-Fender | ~bria |
| 17:29.22 | infobot | [~bria] Bria is a NON-free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com . This includes licensed audio & video codecs and is full-featured. |
| 17:29.29 | Chainsaw | gusto: (It needs a patch that hasn't been applied upstream...) |
| 17:30.04 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 17:30.16 | gusto | Chainsaw: really? i have not tried it, but in the book its documented as being quite easy |
| 17:30.23 | gusto | SRTP is a bigger problem |
| 17:30.56 | Chainsaw | gusto: The SSL reader doesn't block when it should. Data will be lost. |
| 17:31.16 | gusto | Chainsaw: explain! |
| 17:31.19 | Chainsaw | gusto: https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 17:31.36 | JoeyJoeJo | ~xlite |
| 17:31.36 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, Mac OS X, and Linux that can be downloaded from http://www.counterpath.com |
| 17:32.06 | Chainsaw | gusto: The "Filip Jenicek" version is advisable for use. It is least invasive. |
| 17:32.43 | gusto | well |
| 17:32.53 | gusto | i ll have to try that out, let's see if i can reproduce it |
| 17:33.02 | gusto | but not today, maybe tomorro |
| 17:33.04 | gusto | tomorrow |
| 17:33.50 | gusto | theoretically there should not be any data loss because it's using TCP for TLS/SSL, but this seems to be a bug at the application layer |
| 17:33.57 | JoeyJoeJo | ~ekiga |
| 17:33.57 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, Mac OS X, and Linux (requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 17:34.18 | gusto | JoeyJoeJo: yes, i would suggers Ekiga or Linphone to you |
| 17:34.52 | JoeyJoeJo | Ekiga looks awesome |
| 17:34.59 | gusto | JoeyJoeJo: and Linphone enables you to see more, to look inside of it, what happens and adjust things, with Ekiga you would need to be advanced user even to get it to disable STUN |
| 17:35.14 | gusto | JoeyJoeJo: yes, it looks awesome, but is not that reliable |
| 17:35.16 | Chainsaw | But STUN is awesome. Why on earth would you want to turn it off? |
| 17:35.28 | gusto | when you have a local asterisk |
| 17:35.41 | JoeyJoeJo | What is stun? |
| 17:35.43 | JoeyJoeJo | ~stun |
| 17:35.43 | infobot | i heard stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs, or a client side method to cater to crappy sip servers, or a phaser setting |
| 17:35.49 | file | STUNing! |
| 17:35.58 | gusto | for example my asterisk runs on the router so in a lan you have to disable it so that he does not come in with the public ip |
| 17:36.50 | gusto | and with linphone it's quite easy, i can adjust everything on linphone and i have also a debug console on that one too |
| 17:37.36 | gusto | btw. starting this month i have over 50 DAB+ radios in nuremberg ... so FM is obsolete here! |
| 17:37.38 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 17:38.17 | Nivex | and thoughts on SFLphone? |
| 17:38.18 | Chainsaw | gusto: Now try it in a motor vehicle at highway speeds. |
| 17:38.46 | gusto | Chainsaw: should not be a problem ... that 200 km/h problem you only have with DVB-T |
| 17:38.58 | gusto | DAB/DAB+ is not affected |
| 17:39.38 | Chainsaw | gusto: Last time I saw a DAB radio in a vehicle it was prone to drop-outs in places where FM coped. |
| 17:39.53 | Chainsaw | gusto: Then again, perhaps you have your transmitters turned up higher in Germany then they are in the UK. |
| 17:39.56 | gusto | some DAB radios are bad, i know |
| 17:40.47 | gusto | you need to be more careful about DAB also because it uses higher frequencies and so on |
| 17:41.12 | gusto | but that problem you have with DVB-T too |
| 17:41.40 | gusto | a better quality reciever and antennas with less noise and cables with better shielding solve that problems |
| 17:42.40 | navaismo | Rumbles, try in the asterisk cli with: module show like timer |
| 17:43.11 | gusto | from 2009-2011 slovakia was upgrading from analogue to digital tv ... we had the same problem in slovakia. i had to change the coax to rg-6 and and put in a new low noise amplificator to the antenna and then it worked, with the old parts --- no way! |
| 17:43.20 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 17:43.30 | gusto | dropped the signal constantly, the same like you are saying |
| 17:44.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 17:46.01 | gusto | Chainsaw: and of course ... DAB+ with AAC+ HQ LTP is also more resilient to signal drops ... when the signal drops you do not get these bad tones you would get on MP2 |
| 17:46.43 | Chainsaw | gusto: Yes, give me white noise any day. That was bad. |
| 17:46.51 | gusto | so |
| 17:47.16 | gusto | and on AAC+ LTP you get the rythm going on but only miss some details as long as the signal is gone |
| 17:47.23 | gusto | so that is quite OK |
| 17:47.46 | gusto | the beat goes on, you only miss the details as long as the drop takes place |
| 17:49.54 | Chainsaw | gusto: Full bass-line analysis. That is impressive. |
| 17:50.40 | gusto | and as far as i understand there is also a problem with transformators as baluns (from balanced to unbalanced) which cause timing problems and thus bit errors on digital |
| 17:51.46 | gusto | you will see more of that problems comming in as soon as UK migrtes to DVB-T2 ... or did they already? |
| 17:52.31 | Chainsaw | Not sure. I have satellite television, not "freeview". |
| 17:52.55 | gusto | you satellite-fan! |
| 17:53.13 | gusto | did you hear that the russians lost control over their satellites? |
| 17:54.22 | gusto | btw what degree are you tuned in? |
| 17:54.38 | gusto | 28°E or 23°E, or where is it? |
| 17:56.01 | Chainsaw | gusto: "Astra 2" if that helps. |
| 17:56.08 | gusto | w8 |
| 17:56.18 | gusto | i ll have to look it up |
| 17:56.53 | gusto | http://de.kingofsat.net/pos-28.2E.php |
| 17:56.54 | *** join/#asterisk TimeRider (~steve@88-104-147-190.dynamic.dsl.as9105.com) |
| 17:56.55 | gusto | this one? |
| 17:58.07 | gusto | i have astra 1 & 3 then |
| 17:58.32 | Chainsaw | gusto: Yes, it's Sky. |
| 17:58.43 | gusto | hates sky |
| 17:58.59 | gusto | i am considering astra 2 because of the free tv bbc |
| 18:00.29 | gusto | however, its quite surprising that someone in britain is using satellite tv, because one would expect that the most ppl have either DVB-T(2) or IPTV |
| 18:02.47 | gusto | i mean ... britain ... that's an island, right? ... no need for satellite broadcasting |
| 18:03.31 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 18:03.46 | gusto | i wonder when Opus codec will come in to asterisk/VoIP world |
| 18:04.01 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 18:07.33 | gusto | ah, its already in maybe as CELT |
| 18:09.04 | Katty | sways |
| 18:09.16 | gusto | hm |
| 18:09.29 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 18:09.52 | Katty | omnomnom beef stew |
| 18:09.58 | JoeyJoeJo | Is there a good tutorial on how to get asterisk to authenticate against my Active Directory server? Google hasn't been helpful so far |
| 18:10.22 | gusto | oh |
| 18:10.23 | Katty | have you tried wacking it with a hammer. |
| 18:11.18 | gusto | asterisk authenticante against AD? |
| 18:11.26 | JoeyJoeJo | Yeah |
| 18:11.48 | gusto | i would suggest trying to use the AD's kerberos as authentification server for asterisk |
| 18:11.57 | gusto | maybe there is something for asterisk and kerberos |
| 18:12.24 | gusto | http://en.wikipedia.org/wiki/LDAP_Account_Manager |
| 18:12.49 | gusto | ah no, that is different software |
| 18:13.47 | gusto | http://freeradius.1045715.n5.nabble.com/SIP-RADIUS-Kerberos-authentication-td2758910.html |
| 18:13.54 | navaismo | Anyone knows a sip client for windows phone 7.5? |
| 18:13.58 | gusto | JoeyJoeJo: your idea something like that? |
| 18:15.06 | JoeyJoeJo | I wasn't planning on using radius |
| 18:15.11 | gusto | however |
| 18:15.21 | gusto | that would be kerberos then |
| 18:15.49 | JoeyJoeJo | So I need to enable kerberos in my windows server? |
| 18:15.56 | gusto | heh |
| 18:16.05 | gusto | AD already uses kerberos as far as i know |
| 18:16.12 | JoeyJoeJo | I'm not familiar with kerberos at all |
| 18:16.21 | gusto | but there seems to be no way how to interconnect that to asterisk |
| 18:17.08 | JoeyJoeJo | AD supports LDAP. I was hoping to do it that way |
| 18:17.43 | gusto | basically that should be something like a database and there are ways how to put the user list in a database, but there is some software missing that makes a connection between a database of users for asterisk and the active directory thinfs |
| 18:17.57 | gusto | however, the AD |
| 18:18.20 | gusto | but do you have some LDAP support for asterisk? |
| 18:18.50 | gusto | ah we do have |
| 18:19.06 | gusto | ~book 434 |
| 18:19.24 | gusto | let's look |
| 18:19.36 | JoeyJoeJo | I saw stuff for getting OpenLDAP + asterisk working so I hoped that AD would work too |
| 18:20.33 | gusto | yes, i just found that in the book |
| 18:21.20 | gusto | looks like an ugly hack but i have no idea if that would wokr |
| 18:21.21 | gusto | work |
| 18:21.50 | Katty | my intel server is taking off! duck a cover!! |
| 18:22.10 | Katty | s/a/and/ |
| 18:22.21 | gusto | i do not see any way how the clients are authentitating to asterisk here, it's maybe only like ... say openldap server if the user xxx is autheniticated and the openldap says: yes/no |
| 18:23.03 | gusto | why is that taking off? what is on it? |
| 18:23.04 | *** join/#asterisk Snivets (~Snivets@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 18:25.48 | n3hxs | LOL Katty, point it to the West. |
| 18:26.17 | Snivets | I have a question about a snom 760! |
| 18:26.31 | Snivets | specifically its blf setup |
| 18:27.16 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:27.19 | Snivets | when I set up the blf field to point to the correct identity (i've checked the subscriptions and it seems to be properly registered) the lights on the keys don't light up |
| 18:30.13 | mbrit | hi guys, i have a SIP client that only supports h263-2000 for video, but when i make a call the INVITE's SDP sent by asterisk contains only h263-1996 and h263-1998, where should i look to modify that? |
| 18:30.50 | gusto | what is that? |
| 18:31.09 | gusto | h263 - <year> revision? |
| 18:31.10 | mbrit | codecs sent in the SDP |
| 18:33.13 | gusto | of course, but i never seen something like h263-2000 or such |
| 18:33.29 | mbrit | (sorry i'm not too familiar with asterisk yet) but in sip.conf there is an allow=h263 and allow=h263p |
| 18:33.37 | gusto | yes, of course |
| 18:33.47 | gusto | but i do not think that would be an issue of asterisk |
| 18:34.04 | gusto | because asterisk only forwards the packets and h263 should be supported in general |
| 18:34.27 | gusto | so either the problem is something different or it's an issue between the SIP clients |
| 18:34.54 | mbrit | yup but it seems to ignore RTP packets that aren't h263 (96) or h263p (98) |
| 18:34.56 | gusto | mbrit: have you enabled video in SIP general config? |
| 18:35.19 | mbrit | yes, for example using xlite there is no problem because it works using those codecs |
| 18:36.00 | gusto | well, i would need to do some videoconferences with my soft toy first to test it :-) |
| 18:36.17 | mbrit | cool hehe |
| 18:36.29 | gusto | yes, soft toys are the best testing buddys |
| 18:36.59 | gusto | and women are the worst, because they do not have patience |
| 18:37.21 | gusto | and you need patience with asterisk, because not everything works at first shot |
| 18:37.46 | mbrit | lol, and yeah, i've seen that |
| 18:37.58 | gusto | you see |
| 18:38.31 | gusto | that's like my mother, when something is not working she calls me and says: "do make it work! do make it work!" |
| 18:39.29 | mbrit | yeah, just because you are meant to know how to fix it even without knowing what is wrong lol |
| 18:39.43 | jmetro | ...i asked a guy to ship me his harddrive once, and he sent me the entire computer in a giant box. |
| 18:40.00 | mbrit | lol |
| 18:40.29 | gusto | yes, he did not make half things done |
| 18:41.13 | ageis | Let's say my SIP provider/VoicePulse goes down, shouldn't I still be able to call locally registered/connected devices? |
| 18:43.04 | jmetro | i believe yes, as long as your server is still hooked up. my personal testing box has no sip provider and i can make calls between phones. |
| 18:43.32 | jmetro | [and coincidentally, make calls to other phones that are reachable outside of the PSTN like URI dials] |
| 18:46.47 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:50.41 | ghost75 | http://www.heise.de/autos/artikel/Bundestag-missachtet-eigene-Umweltauflagen-1749551.html |
| 18:50.53 | ghost75 | ups wrong channel |
| 18:52.44 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-eixbrcqvpivcnxnu) |
| 18:52.46 | shido6 | iru |
| 18:56.05 | ghost75 | p3nguin: same with sudo |
| 18:57.07 | *** join/#asterisk ruben231 (~OpenDial@112.198.78.99) |
| 18:58.09 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:58.13 | ruben231 | hi guys whast the use of adding variable on asterisk dialplan like this ---> 12127775678wwww3 ------------> does it mean waiting time then send dtmf number 3 _ when dialing..? |
| 18:58.50 | JoeyJoeJo | I'm getting "No matching peer found" when I try to connect to my asterisk server. That means I'm missing something from sip.conf, right? |
| 18:59.31 | ghost75 | when you try to connect? |
| 18:59.32 | ruben231 | how do i insert wwww3 on a number i dilaing like US number 2127775678 and would look like this on output ---> 2127775678wwww3 |
| 18:59.40 | ruben231 | any idea guys |
| 18:59.57 | JoeyJoeJo | ghost75: yes |
| 19:00.17 | ghost75 | you mean connect by using your phone ? |
| 19:00.30 | JoeyJoeJo | I'm using Ekiga |
| 19:01.04 | p3nguin | ruben231: That isn't a variable. |
| 19:01.15 | ghost75 | maybe you need assign phones to trunk |
| 19:01.22 | ruben231 | p3nguin: what is it..? |
| 19:01.56 | p3nguin | In a technology that supports it, the w in the phone number is .5 seconds wait time. |
| 19:02.25 | p3nguin | wwww = 2s. |
| 19:02.38 | ghost75 | you have some strange numbers |
| 19:03.20 | p3nguin | joeyjoejo: No matching peer found means that the phone is sending user credentials not matching any peer that is configured in sip.conf. |
| 19:03.26 | ruben231 | <PROTECTED> |
| 19:03.47 | p3nguin | That is not valid. |
| 19:04.28 | p3nguin | Well, technically it would be valid if you are dialing the four Ws from your phone keypad, but that's not your intention. |
| 19:05.15 | p3nguin | joeyjoejo: Did you create a peer for that phone to use? |
| 19:05.39 | JoeyJoeJo | p3nguin: No, I'm trying to use my ldap server to authenticate |
| 19:05.51 | JoeyJoeJo | I guess I just have something messed up with my ldap config |
| 19:09.32 | [TK]D-Fender | ruben231, "w"'s are things you put in the Dial APPLICATION to add a delay. This is only really applicable to ANALOG channels. Also to the SendDTMF app |
| 19:10.55 | p3nguin | joeyjoejo: I don't know how ldap adds peers to the sip configuration, but you're somehow missing either the username, password, or both. No matching peer means exactly that. |
| 19:12.37 | dijib | when im creating a certificae i have to create it as the user im going to use it from? |
| 19:13.52 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:15.01 | dijib | i just found the best domain gofaxyourself.com |
| 19:15.38 | p3nguin | You do not have to create a certificate under the user account on which you will use it. |
| 19:15.52 | p3nguin | You can create the cert anywhere and use it anywhere else. |
| 19:16.47 | dijib | so a certifate ive created and coped into my root directory is still usable by the asterisk account although unreadable becuase its in root home dir? |
| 19:17.04 | dijib | would that make sense |
| 19:17.27 | p3nguin | Correct. |
| 19:18.04 | p3nguin | The application trying to use the cert must be able to access the cert. |
| 19:19.09 | dijib | and i would have inherited permssion issues if cert is in root home dir ~ |
| 19:19.33 | p3nguin | Not inherited, just simple permission issues. |
| 19:19.48 | p3nguin | Nothing but root should be able to see what is in root's home. |
| 19:20.13 | dijib | they should have been inherited from the parent permissions of /root |
| 19:20.24 | dijib | thats how windblows works |
| 19:20.36 | dijib | recursive permissions |
| 19:23.40 | ruben231 | <PROTECTED> |
| 19:23.51 | ruben231 | <PROTECTED> |
| 19:24.09 | [TK]D-Fender | ruben231, How to do what? There is no time delay on an EXTEN pattern |
| 19:24.18 | [TK]D-Fender | ruben231, Your description of what you want to do is very ppor |
| 19:24.26 | [TK]D-Fender | poor* |
| 19:25.02 | p3nguin | dijib: If the cert files are marked as readable but the directory is not, the cert files are not readable. |
| 19:30.23 | ageis | is this proper syntax in Asterisk 1.6 Dial(SIP/${EXTEN}) |
| 19:30.57 | [TK]D-Fender | ageis, Depends what EXTEN you are on and if you have a peer that will match what was actually dialed. |
| 19:31.05 | [TK]D-Fender | ageis, it COULD be valid. |
| 19:31.36 | ageis | Ya |
| 19:34.18 | p3nguin | Could be, but probably shouldn't be if you use best practices. |
| 19:34.18 | ageis | what does the plus sign in front of the variable do? |
| 19:34.31 | p3nguin | What plus sign, what variable? |
| 19:35.01 | ageis | this symbol here + |
| 19:35.02 | ageis | any variable |
| 19:35.03 | [TK]D-Fender | .... |
| 19:35.07 | [TK]D-Fender | ageis, show us the full SAMPLE |
| 19:35.10 | p3nguin | It means + |
| 19:35.23 | p3nguin | But we need context of use to know what it will actually DO. |
| 19:35.30 | ageis | how is it interpreted by asterisk sir |
| 19:35.33 | [TK]D-Fender | ... |
| 19:35.33 | ageis | okay then |
| 19:35.34 | p3nguin | + |
| 19:35.39 | p3nguin | It's a plus. |
| 19:35.46 | ageis | how about this |
| 19:35.46 | ageis | +${EXTEN:3} |
| 19:35.52 | ageis | plus and three please |
| 19:35.55 | [TK]D-Fender | ageis, depends where it is in the line. Location matters, app matters. + isn't a magical single thing |
| 19:35.58 | p3nguin | It means + in that context. |
| 19:36.04 | [TK]D-Fender | ageis, Show us the EXACT usage. |
| 19:36.10 | ageis | there's no + on a dial pad |
| 19:36.10 | [TK]D-Fender | ageis, On it's own that means nothiner |
| 19:36.14 | p3nguin | If EXTEN:3 is 123, then it means +123 |
| 19:36.18 | p3nguin | [tk]d-fender: its |
| 19:36.21 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-faxyhafqbadeisuz) |
| 19:36.24 | [TK]D-Fender | p3nguin, Yeah yeah... |
| 19:36.34 | p3nguin | It is even less typing to do it right. |
| 19:36.40 | ageis | let's put it in a Dial command (SIP/+{$EXTEN}) |
| 19:36.44 | ageis | is that enough context? |
| 19:36.51 | p3nguin | Asked and answered. |
| 19:36.51 | [TK]D-Fender | p3nguin, bad aim... I know the proper usage. |
| 19:37.05 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 19:37.11 | [TK]D-Fender | ageis, then it will dial a SIP peer entry that literally starts with a + sign |
| 19:37.17 | shido6 | wakes up |
| 19:37.21 | shido6 | hi |
| 19:37.22 | ageis | [TK]D-Fender: cool, thanks |
| 19:37.23 | [TK]D-Fender | ageis, so you'd better have [+XYZ] in your sip.conf |
| 19:37.31 | p3nguin | I usually don't see those. |
| 19:37.44 | [TK]D-Fender | I don't think I've ever seen it either. |
| 19:37.49 | p3nguin | XYZ yes, +XYZ no |
| 19:38.39 | [TK]D-Fender | <ageis> let's put it in a Dial command (SIP/+{$EXTEN}) <-- and your variable reference is wrong there |
| 19:38.51 | ageis | correct |
| 19:38.52 | ageis | Typo |
| 19:39.08 | leifmadsen | in some cases the + is necessary |
| 19:39.17 | leifmadsen | I've had providers that required it (for whatever reason) |
| 19:39.54 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 19:39.57 | enoch | hi |
| 19:40.09 | enoch | how to hide caller id in chan_datacard? |
| 19:40.14 | *** join/#asterisk navaismo (~navaismo@189.191.12.116) |
| 19:40.14 | p3nguin | In the example of SIP/provider/+${EXTEN} , yeah. |
| 19:40.33 | ageis | Set(CALLERID(name)= something else |
| 19:40.40 | p3nguin | But in SIP/+${EXTEN} , I don't see too many people creating peers on-box that start with plus. |
| 19:40.55 | ageis | p3nguin: im still having this |
| 19:40.57 | [TK]D-Fender | nope... |
| 19:41.24 | ageis | 'extension not found' problem. As I said yesterday I want to be able to disable my SIP provider and only make calls between registered peers/devices. |
| 19:41.35 | [TK]D-Fender | ageis, PASTEBIN <- |
| 19:41.36 | [TK]D-Fender | ~pb |
| 19:41.36 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:41.43 | p3nguin | You're still not using the correct context in the peer making the call. |
| 19:41.45 | ageis | [TK]D-Fender gladly, if you'll look at my conf |
| 19:42.14 | p3nguin | In the peer entry, you need to define the CONTEXT where the extensions actually exist. |
| 19:42.14 | [TK]D-Fender | ageis, You are trying to have us validate your dialplan 1 stupid character at a time. PASTEBIN the entire call attempt and ALL of the associated dialplan, INCLUDING the context header. |
| 19:42.49 | ageis | [TK]D-Fender, sir I heard you the first time, I am preparing it |
| 19:42.54 | [TK]D-Fender | \o/ |
| 19:44.10 | *** join/#asterisk tomwish (~Tom@71-223-193-61.phnx.qwest.net) |
| 19:46.02 | *** join/#asterisk kchehab (~ck@74.114.101.188) |
| 19:46.11 | p3nguin | I guess I wasn't clear enough the first two times I told him the problem. |
| 19:46.11 | kchehab | hi |
| 19:48.33 | kchehab | can i match the incoming caller ID in a diaplan for an example exten => _X.,n,GotoIf($["${CALLERID(num)}" ==_212. |
| 19:48.59 | [TK]D-Fender | kchehab, that is an incomplete line |
| 19:49.08 | [TK]D-Fender | kchehab, Show a COMPLETE sample if you want an opinion on it |
| 19:49.28 | p3nguin | Can you match incoming caller ID in dial plan? Yes. |
| 19:49.58 | kchehab | [TK]D-Fender to use regex in matching the callerid(num) |
| 19:51.04 | [TK]D-Fender | kchehab, "core show function REGEX" |
| 19:51.21 | kchehab | [TK]D-Fender it functional using a static caller ID as exten => _X.,n,GotoIf($["${CALLERID(num)}" !="999999" but is there a new function to do that using regex as exten => _X.,n,GotoIf($["${CALLERID(num)}" != _999. |
| 19:51.32 | [TK]D-Fender | kchehab, And no, there is no "pattern" for expressions. Give the section on expressions in the book another read |
| 19:51.40 | enoch | a question: is it possible to send sms using asterisk and masking the telephone number? |
| 19:52.00 | Snivets | I fixed the snom issue, it was a subscription expiry issue as far as I can tell |
| 19:52.28 | Snivets | I'm used to the yealinks and how they are always illuminated in one way or another, that is also throwing me off a bit |
| 19:52.43 | p3nguin | kchehab: A valid callerID number will never ever match "_999." |
| 19:53.20 | kaldemar | kchehab: "${CALLERID(num):0:3}" != "999" |
| 19:53.24 | ageis | p3nguin: I understood your explanation, but I have made certain that the peers for the devices I'm using are included in the context. That's why I wanted you to look at my config. http://pastebin.com/BQMSVBpD |
| 19:53.31 | p3nguin | If you just want to match callerid numbers starting with 999, you can use part of the variable or regex. |
| 19:53.33 | ageis | [TK]D-Fender: sip.conf, extensions.conf and log. http://pastebin.com/BQMSVBpD%20http://pastebin.com/pcbRnk1L%20http://pastebin.com/EkFFc6WM |
| 19:53.36 | ageis | erm |
| 19:53.42 | kaldemar | kchehab: no regex needed for checking the first 3 characters. |
| 19:53.44 | tomwish | I am having an issue with confbridge on asterisk10. I have a generic user,bridge,menu set up and try to pass var to it using the set(CONFBDRIGE(user,etc)=var. When the conf starts it always reverts back to what is in the confbridge.conf file. here is a snippet of the code. exten => s,n,Set(CONFBRIDGE(user,template)=100) |
| 19:53.45 | tomwish | exten => s,n,Set(CONFBRIDGE(user,music_on_hold_class)=(${ODBC_MUSIC()}) |
| 19:53.45 | tomwish | exten => s,n,Set(CONFBRIDGE(user,marked)=no) |
| 19:53.45 | tomwish | exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no) |
| 19:53.45 | tomwish | exten => s,n,Set(CONFBRIDGE(user,music_on_hold_when_empty)=yes) |
| 19:53.45 | ageis | http://pastebin.com/BQMSVBpD http://pastebin.com/pcbRnk1L http://pastebin.com/EkFFc6WM |
| 19:53.45 | tomwish | exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes) |
| 19:53.46 | tomwish | exten => s,n,Set(CONFBRIDGE(user,music_on_hold_class)=(${ODBC_MUSIC()}) |
| 19:53.46 | tomwish | exten => s,n,Set(CONFBRIDGE(bridge,template)=100) |
| 19:53.46 | tomwish | exten => s,n,Set(CONFBRIDGE(bridge,sound_only_one)=welcome) |
| 19:53.47 | tomwish | exten => s,n,ConfBridge(100 -w1mM ,, ,,100) |
| 19:54.00 | p3nguin | punches a floodfucker in the face |
| 19:54.04 | [TK]D-Fender | tomaw, PASTEBIN. do NOT flood in here |
| 19:54.06 | [TK]D-Fender | ~pb |
| 19:54.07 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:54.15 | tomwish | ok. sorry |
| 19:54.40 | ageis | [TK]D-Fender: consider scenario of kevin-mobile(1337) trying to call kelly(89). take care to ignore commented out |
| 19:55.23 | p3nguin | ageis: You do not "include peers in a context." You assign a context to a peer. In the context, you create extensions that you want the peer to have access to. |
| 19:55.45 | ageis | p3nguin: Terminology. I assigned a context to the peers, and did all of that. |
| 19:55.59 | kchehab | in simple words how that can be done since i want to block a specific dialplan from pstn numbers using regex ,anyone can pastebin how to do that |
| 19:56.00 | [TK]D-Fender | ageis, "sip set debug on" <- place the call again |
| 19:56.01 | ageis | by include i meant the literal include command |
| 19:56.07 | ageis | [TK]D-Fender k |
| 19:56.14 | [TK]D-Fender | ageis, We need to see what peer it's matching and what context it's really looking in. |
| 19:57.07 | tomwish | let me try this again. i am trying to pass vars to confbridge. http://pastebin.com/MWBCbXJz I can see the return from the odbc lookup and it is correct. confbridge ignoes the vars and uses the default settings. The one I really care about is the music class. How do I get confbrdge to take the cars? |
| 19:57.11 | kchehab | or a link can guide me for that |
| 19:57.23 | ageis | [TK]D-Fender: http://pastebin.com/DtxXVyWp |
| 19:57.50 | p3nguin | kevin-mobile has a context of outgoing. outgoing incldues menu, menu contains extension 89. |
| 19:58.00 | kaldemar | kchehab: i just told you how to do what you asked. did you mean something else? |
| 19:58.01 | ageis | Yah |
| 19:58.07 | ageis | "Looking for s in banned" is this a problem? |
| 19:58.30 | [TK]D-Fender | Now we need to prove that this file you showed us is even INCLUDED properly. |
| 19:58.50 | ageis | ok |
| 19:59.01 | [TK]D-Fender | ageis, "dialplan show" |
| 19:59.25 | ageis | lol, I think this is a permissions problem. |
| 19:59.31 | ageis | my conf files were owned by root not asterisk |
| 19:59.50 | [TK]D-Fender | UNgood |
| 19:59.59 | [TK]D-Fender | fix that up, reload * |
| 20:00.03 | [TK]D-Fender | and check it |
| 20:00.42 | kchehab | kaldemar "${CALLERID(num):0:3}" != "999" this will match any number begin with 999 ??? |
| 20:01.18 | p3nguin | != means not matching |
| 20:01.18 | ageis | [TK]D-Fender: ok i did, but the dialplan still doesnt match what's in extensions.conf. all demo stuff. any idea why its not loading? |
| 20:01.29 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 20:01.56 | [TK]D-Fender | ageis, "core show settings", and "ls -la" your config folder |
| 20:03.21 | kaldemar | kchehab: the :0 means no cut 0 characters from the beginning of CALLERID(num) and the :3 limits the length to 3. so ${CALLERID(num):0:3} gives you 3 first characters of the caller id number. |
| 20:03.47 | ageis | [TK]D-Fender: hrm, looks good. does asterisk require a specific chmod? |
| 20:04.02 | p3nguin | Doesn't look good to us. |
| 20:04.22 | ageis | [TK]D-Fender: http://pastebin.com/G0N5wrNV |
| 20:04.46 | ageis | everything is -rwxrwxr-x 1 asterisk asterisk in /etc/asterisk, including parent folder |
| 20:04.59 | kaldemar | kchehab: use them as you wish. exten => _X./_999. would match calls with caller id number starting with 999 and with length of 4 or above. |
| 20:05.02 | kchehab | kaldemar please correct my dialplan exten => _X.,n,,GotoIf($[ ${CALLERID(num):0:3} != "999"]?conn,1:fail,1) |
| 20:05.16 | *** join/#asterisk mokmeister (~mokmeiste@109.78.34.118) |
| 20:05.33 | [TK]D-Fender | ageis, please dumpt the folder, etc |
| 20:05.35 | p3nguin | ageis: ps -C asterisk u |
| 20:05.56 | ageis | asterisk 7323 0.9 1.8 48168 18640 ? Ssl 15:03 0:01 /usr/sbin/asterisk -p -U asterisk |
| 20:05.57 | ageis | sure |
| 20:05.57 | kaldemar | kchehab: $[ ${ -> $[${ |
| 20:06.08 | p3nguin | Running as asterisk. Good. |
| 20:06.28 | ageis | [TK]D-Fender http://pastebin.com/nLAQRFPt |
| 20:06.38 | ageis | I'm glad we know what the problem is at least |
| 20:06.42 | kaldemar | kchehab: actually, $[ ${CALLERID(num):0:3} != "999"] -> $["${CALLERID(num):0:3}" != "999"] |
| 20:07.09 | dijib | p3nguin: whats going on here: http://pastebay.net/1155822 |
| 20:07.27 | ageis | I think I need -rw-r----- what's that, 600? |
| 20:07.35 | p3nguin | 640 |
| 20:07.38 | ageis | k |
| 20:08.05 | p3nguin | For the subdirectories, you're going to want 750. |
| 20:08.53 | ageis | orig and manager.d |
| 20:09.17 | [TK]D-Fender | pastebin the entire extensions.conf and "dialplan show" |
| 20:09.21 | [TK]D-Fender | ageis, ^ |
| 20:09.28 | kchehab | kaldemar you mean exten => _X.,n, Gotoif($[ ${CALLERID(num):0:3} != "999"] -> $["${CALLERID(num):0:3}" != "999"] ?conn,1:fail,1) sorry since i didnt get your point |
| 20:10.04 | ageis | when I load up It parses asterisk.conf and extconfig but nothing else |
| 20:10.05 | kaldemar | kchehab: replace the first one with the second... |
| 20:10.07 | ageis | [TK]D-Fender http://pastebin.com/AXwiKT8M |
| 20:10.46 | ageis | Here are my new perms still not working http://pastebin.com/UEM3VTAz |
| 20:11.06 | kchehab | kaldemar you mean exten => _X.,n, Gotoif($[ ${CALLERID(num):0:3} != "999"] != "999"] ?conn,1:fail,1) sorry since i didnt get your point |
| 20:11.14 | p3nguin | dijib: Lines 4 and 5 tell you a major problem that you need to correct. |
| 20:11.18 | kchehab | kaldemar you mean exten => _X.,n, Gotoif($[ ${CALLERID(num):0:3} != "999"] != "999"] ?conn,1:fail,1) like this bro |
| 20:11.52 | ageis | [TK]D-Fender I can tell its loading sip.conf cause the peers are in, but its not loading the dialplan |
| 20:11.53 | [TK]D-Fender | kchehab, stop putting things OUTSIDE your expression $[] |
| 20:12.05 | [TK]D-Fender | ageis, try "dialplan reload" from CLI |
| 20:12.27 | ageis | [TK]D-Fender: no such command |
| 20:12.28 | kaldemar | kchehab: no. exten => _X.,n,Gotoif($["${CALLERID(num):0:3}" != "999"]?conn,1:fail,1) |
| 20:12.34 | [TK]D-Fender | kchehab, and quotes are LITERAL, you have to have them on BOTH sides of your operator. |
| 20:12.42 | p3nguin | pbx_config didn't load, I guess. |
| 20:12.47 | kchehab | kaldemar [TK]D-Fender thanks alot |
| 20:12.51 | p3nguin | ageis: ^ |
| 20:12.53 | [TK]D-Fender | ageis, I'd check your modules folder permissions too... |
| 20:13.17 | [TK]D-Fender | ageis, Module directory: /usr/lib/asterisk/modules |
| 20:13.38 | ageis | k |
| 20:14.14 | ageis | yeah |
| 20:14.16 | [TK]D-Fender | ageis, Heck .... check them all. |
| 20:14.23 | ageis | I rsync'ed all my asterisk files as root lol. So there are 4 folder I have to check |
| 20:14.39 | p3nguin | The modules directory should have normal permissions with root ownership. chown root:root and chmod 755 |
| 20:14.44 | ageis | ok |
| 20:15.11 | ageis | how about /usr/share/asterisk |
| 20:15.25 | p3nguin | The modules should be similar. Root can own them and everyone can read them. |
| 20:15.35 | [TK]D-Fender | ageis, http://www.quickmeme.com/meme/3rs5og/ |
| 20:15.49 | ageis | :P |
| 20:15.51 | p3nguin | I don't have any /usr/share/asterisk. |
| 20:15.53 | ageis | thanks sirs |
| 20:16.51 | dijib | STARTTLS=client, relay=[127.0.0.1], version=TLSv1/SSLv3, verify=FAIL, cipher= <--THIS |
| 20:17.02 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 20:17.28 | tomwish | I am trying to play a specific moh class for confbridge when there is only 1 caller. Here is the way I start confbridge. exten => s,n,ConfBridge(100) -wM[eagleIVR1]) I am guessing that I am not setting the M flag correctly. Could somene please point me in the right direction? |
| 20:17.38 | dijib | and this : msgid=<20121114201559.GA23057@swissarms.eastlink.ca |
| 20:17.46 | dijib | im not swissarms.eastlink.ca |
| 20:18.21 | [TK]D-Fender | tomaw, what is that junk doing OUTSIDE of your closing )? |
| 20:18.44 | [TK]D-Fender | exten => s,n,ConfBridge(100) -wM[eagleIVR1]) <------- (100) |
| 20:19.01 | p3nguin | You know... 100) is the name of his conf. :/ |
| 20:19.20 | p3nguin | FIALURE <---- |
| 20:19.29 | dijib | ConfBridge(${EXTEN}) |
| 20:19.34 | [TK]D-Fender | p3nguin, http://xkcd.com/327/ |
| 20:19.45 | tomwish | I have tried it as confbridge(100 -wM[class}) and that didn't work either |
| 20:19.51 | p3nguin | Only if he wants to use the exten as the conf number. |
| 20:19.56 | p3nguin | What is the -w thing? |
| 20:20.10 | tomwish | wait for leader or marked user |
| 20:20.22 | [TK]D-Fender | tomwish, And where do you get the impression you just shove a SPACE in there and then other random chars? |
| 20:20.34 | p3nguin | [tk]d-fender: My wife has that comic printed on a t-shirt. |
| 20:20.40 | [TK]D-Fender | tomaw, App parameters are separated by COMMAS |
| 20:20.54 | [TK]D-Fender | p3nguin, entirely print-worthy. |
| 20:21.10 | tomwish | ok. let me try that. tahks for the help |
| 20:25.04 | dijib | man i suck |
| 20:25.09 | dijib | i cant fix postfix |
| 20:25.23 | dijib | fix that problem then the next |
| 20:27.06 | p3nguin | What are you using postfix for? |
| 20:27.10 | dijib | http://pastebay.net/1155826 |
| 20:27.15 | dijib | relay |
| 20:27.26 | dijib | mutt -> postfix for the fax emailing |
| 20:27.53 | dijib | no errors in asterisk cli, come out as h-SUCCESS |
| 20:29.40 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-dxazknqoimhrkvif) |
| 20:30.51 | ageis | [TK]D-Fender: Hrm, I checked all my permissions, I even purged and reinstalled. It won't load my damn extensions.conf, any ideas? |
| 20:31.22 | citywok | dijib: does your ISP block port25 outbound? |
| 20:31.26 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 20:31.38 | ageis | gives execute permissions on /etc/asterisk |
| 20:31.55 | [TK]D-Fender | ageis, pastebin ls dumps for all applicable folder, and "core show settings" and "reload" |
| 20:32.00 | [TK]D-Fender | ageis, 1 big pastebin |
| 20:32.04 | ageis | sure thing |
| 20:32.20 | citywok | ageis: is there an #include statement for a file that doesn't actually exist? that can cause extensions.conf to not load |
| 20:33.14 | ageis | citywok: ah yes |
| 20:33.19 | tomwish | D-fender, I changed it to this exten => s,n,ConfBridge((100)M[eagleIVR1]) and still doesn't work. Could you explain to me what is wrong with that? it looks ok from what I can google. |
| 20:33.27 | dijib | citywok: not sure |
| 20:33.38 | [TK]D-Fender | tomaw, exten => s,n,ConfBridge((100) <- NO DOUBLE BRACKETS |
| 20:33.41 | dijib | how does asterisk voicemail emailing work, those work |
| 20:33.48 | [TK]D-Fender | tomwish, exten => s,n,ConfBridge((100) <- NO DOUBLE BRACKETS |
| 20:33.58 | [TK]D-Fender | tomwish, Stop adding extra chars everywhere |
| 20:34.13 | [TK]D-Fender | tomwish, Where do you keep coming up with this invented syntax? |
| 20:34.30 | citywok | ageis: 'touch' it -- a blank file will work fine |
| 20:34.39 | [TK]D-Fender | M[eagleIVR1] <-- these SQUARE ones are certainly wrong. |
| 20:34.40 | tomwish | exten => s,n,ConfBridge(100,M[eagleIVR1]) is what you are saying? I am kinda slow today |
| 20:35.32 | ageis | [TK]D-Fender p3nguin citywok It's fixed, thank you everyone |
| 20:35.45 | [TK]D-Fender | ageis, Include did indeed kill it? |
| 20:36.33 | ageis | [TK]D-Fender I think so though I cant be certain. I did have an include to a nonexistent file. It didn't fix it right away after the first restart/reload, but I think it may have after a reload in CLI |
| 20:36.37 | [TK]D-Fender | citywok, It didn't used to... I vaguely recall something about them deciding to actually cause it to die. Bad move IMO |
| 20:37.03 | citywok | [TK]D-Fender: i've been bitten by that exact problem before, it's a thing. extensions ael will load, but not conf, if you have a non-existent #include file |
| 20:37.11 | ageis | yep |
| 20:37.16 | [TK]D-Fender | citywok, that's recent. |
| 20:37.29 | citywok | i probably hit it a year ago |
| 20:37.37 | citywok | in 1.8 |
| 20:37.41 | citywok | not sure about 1.4/1.6 |
| 20:38.08 | citywok | it doesn't spew any errors or helpful information, it just doesn't load the extensions.conf file -- it's incredibly frustrating |
| 20:39.18 | ageis | i'm running 1.6 |
| 20:39.21 | citywok | 1.6.x.x? |
| 20:39.43 | citywok | dijib: can you telnet to the IP:25 in the pastebin? |
| 20:40.32 | ageis | 1.6.2.7-1ubuntu1.2 |
| 20:40.32 | citywok | my guess is you can't |
| 20:41.00 | dijib | this? 74.125.136.27 |
| 20:41.03 | citywok | good lord that is old -- 1.6.2.24 is the final there |
| 20:41.19 | dijib | alt4.gmail-smtp-in.l.google.com |
| 20:41.20 | *** join/#asterisk fritz09 (~Adium@2002:542e:5694:e472:302c:eb91:4a57:72ac) |
| 20:41.27 | citywok | dijib: yes -- i can telnet to port 25 on that -- 220 mx.google.com ESMTP c2si2723133eeo.131 |
| 20:41.46 | dijib | testing... |
| 20:41.51 | dijib | y |
| 20:42.11 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 20:42.44 | citywok | that works, but postfix can't connect to it? now that IS odd. |
| 20:43.28 | tomaw | steals [TK]D-Fender's keyboard |
| 20:43.40 | dijib | telnet 74.125.136.27 25 does not seem to be working |
| 20:44.11 | [TK]D-Fender | tomaw, Yeah sorry, auto-complete in new IRC client loves foricbly showing the lowest alphabetical match without prompting... |
| 20:44.23 | [TK]D-Fender | forcibly* |
| 20:44.32 | tomaw | use irssi. it bases completions on channel activity :) |
| 20:45.01 | citywok | dijib: then your ISP is potentially blocking outbound port 25. or google is ignoring you. |
| 20:45.34 | citywok | dijib: try 66.196.118.34 (yahoo) |
| 20:45.38 | dijib | telnet: connect to address 74.125.136.27: Connection timed out |
| 20:46.01 | p3nguin | After that, try your ISP's relay address. |
| 20:46.05 | dijib | looks like the same thing |
| 20:46.48 | citywok | dijib: your ISP is blocking outbound port 25 (probably for computers with viruses spewing spam) |
| 20:47.06 | citywok | like p3nguin said, try using your ISP's mail server to relay the outbound mail through postfix |
| 20:47.08 | dijib | telnet: connect to address 66.196.118.34: Connection timed out |
| 20:47.13 | dijib | see that |
| 20:47.18 | [TK]D-Fender | tomaw, That alone is pretty much reason enough for me to switch.... |
| 20:47.43 | citywok | dijib: /etc/postfix/main.cf -- relayhost = IP.OF.ISP.RELAY |
| 20:47.56 | dijib | ty citywok |
| 20:48.10 | citywok | np |
| 20:48.53 | p3nguin | (1427.06) <p3nguin> What are you using postfix for? |
| 20:48.54 | p3nguin | (1427.26) <dijib> mutt -> postfix for the fax emailing |
| 20:49.02 | p3nguin | ^^^^^^ This is unncessary work. |
| 20:49.47 | p3nguin | You can use an smtpd that requires much less configuration. |
| 20:50.05 | p3nguin | I personally use ssmtp and used to use msmtp. |
| 20:50.19 | citywok | postfix with the default config works just fine for delivering email |
| 20:51.12 | dijib | for you but im still new |
| 20:51.55 | ghost75 | how long it takes to setup fax ? |
| 20:51.58 | citywok | dijib: yours doesn't work because your ISP is blocking you :p -- the mail daemon makes no difference in your case. it won't work either way. |
| 20:52.05 | citywok | (unless you relay through your ISP mail servers) |
| 20:53.30 | [TK]D-Fender | ghost75, Depends on you. |
| 20:53.44 | [TK]D-Fender | ghost75, Minutes. Hours. Days.... |
| 20:53.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:54.34 | ghost75 | how many config files are involved |
| 20:55.13 | ghost75 | not sure if its worth the effort :) |
| 20:55.28 | dijib | it will when i put in the local mal server |
| 20:57.03 | [TK]D-Fender | ghost75, generally no config files. Depends on HOW you intend to deal with faxing. |
| 20:57.30 | [TK]D-Fender | ghost75, Details matter so you should probably be very specific about waht direct and by what means you expect to accomplish your ends. |
| 20:57.54 | ghost75 | only send out fax over t.38 |
| 20:59.16 | [TK]D-Fender | ghost75, what is on the sending end? |
| 20:59.36 | ghost75 | the receiver of fax? |
| 21:00.03 | ghost75 | a big unknown |
| 21:01.17 | *** join/#asterisk bchia (~Adium@nat/digium/x-euvbcpyvpebzysvu) |
| 21:02.30 | [TK]D-Fender | ghost75, Sender = YOU. |
| 21:02.52 | [TK]D-Fender | ghost75, Are you plugging a PHYSICAL fax machine in? Exactly where is this "fax" starting from? |
| 21:03.06 | ghost75 | from pc |
| 21:04.32 | [TK]D-Fender | awaits a complete qualified answer. |
| 21:04.58 | ghost75 | havent looked yet for software :) but i think win7 has fax client |
| 21:05.39 | dijib | so all i need in main.cf is the relayhost set? |
| 21:05.53 | dijib | i dont need to authenticate against the smtp server |
| 21:06.00 | dijib | open relay |
| 21:06.09 | ghost75 | other option would be to hook up hardware fax on my pap2t |
| 21:08.14 | slav3_kitten | ok well i still am natted... but i have a static ip now |
| 21:09.03 | slav3_kitten | instead of a 192.168.100.x it's a 10. address space that is a 200. something externally :| |
| 21:09.16 | slav3_kitten | and there is still port mangling going on.... |
| 21:09.35 | slav3_kitten | but! i have the ability to make 2 way calling to the UK now |
| 21:09.47 | slav3_kitten | i /think/ this is an improvement |
| 21:09.54 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:09.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:20.07 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 21:21.14 | p3nguin | The static address did not make two-way audio possible. |
| 21:21.29 | p3nguin | And I thought you said a condition of that static address was bridging. |
| 21:22.57 | slav3_kitten | p3nguin, it is |
| 21:23.14 | slav3_kitten | i'm in bridge mode to a static ip, and still behind a nat |
| 21:23.18 | p3nguin | So you got bridged and still got NAT? |
| 21:23.22 | slav3_kitten | little less than pleased about this |
| 21:23.34 | slav3_kitten | yep... |
| 21:23.59 | p3nguin | If your device has both external and internal addresses and is providing nat, then it is not bridged. |
| 21:24.55 | p3nguin | I probably haven't seen the whole picture yet. |
| 21:25.21 | slav3_kitten | yea that's what i tried telling the tech support guy they had me on the phone with whom informed me bridging is only offered for static IP customers (10 bucks a month) and when asked why i don't have my public ip on my router wan port i was told "well it's all very complicated" |
| 21:26.17 | p3nguin | It's not THAT complicated. He just doesn't know what he's talking about. |
| 21:27.05 | p3nguin | If your device is now in bridged mode, and your first node (your router) has the IP address on it, then the ISP has one large NAT that all the customers are within. |
| 21:27.29 | p3nguin | Which means they were giving you double NAT before. |
| 21:27.54 | slav3_kitten | yea. so i'm paying 120 a year for a static ip to have my wisp "modem" put in bridging mode that gives me a natted ip... yep that's basically what it was before p3nguin with my nat on top of that so triple natted |
| 21:28.21 | p3nguin | What they should be doing is giving the public routed addresses to the devices and having the devices provide NAT. Or bridge the devices and put the public routed address on the first node. |
| 21:28.50 | p3nguin | You shouldn't have been using NAT in addition. |
| 21:28.59 | n3hxs | slav3_kitten, if this is true::: it's a 10. address space that is a 200. something externally it is still NAT |
| 21:29.34 | slav3_kitten | n3hxs, yea we've established i am still natted |
| 21:29.36 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 21:29.53 | n3hxs | terrible |
| 21:30.09 | n3hxs | who is the provider? |
| 21:30.16 | ghost75 | ipv4 :< |
| 21:30.23 | slav3_kitten | n3hxs, a shitty wisp out in rural tn |
| 21:30.39 | n3hxs | do you get moonshine with that? |
| 21:30.44 | n3hxs | sorry.... |
| 21:31.05 | slav3_kitten | they don't even have any form of battery backup at their cell site. power drops out for literally 1/2 a second and i'm without service for a couple mins while all their shit restarts |
| 21:31.27 | ghost75 | lol |
| 21:32.38 | n3hxs | wow |
| 21:35.33 | slav3_kitten | p3nguin, when i signed up they told me that the modem provides 10 IPs period, so it got hooked to my router to nat it for my internal network an to allow things to be setup how i wanted (3 vlans, some static routing, access lists an the like). the i was told i can't be put in bridge ode without a static ip @ 10/month. so today i got them to give me my static ip which basically means they stripped off one layer of the nat/firewa |
| 21:35.33 | slav3_kitten | ll crap and are still natting me supposedly without "firewall or alg" |
| 21:36.12 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 21:36.35 | p3nguin | I guess some improvement is better than no improvement. |
| 21:37.30 | slav3_kitten | yea i guess, but i doubt the lack of alg since it still appears as if it is being done as i send out registration on 5060 and flowroute says it's coming from 1134 |
| 21:38.46 | slav3_kitten | but i get two way calling to the uk an us |
| 21:38.48 | p3nguin | Coming from a different port shouldn't be a problem. Applications often use a source port differing from the destination port. |
| 21:39.08 | p3nguin | Actually, they almost always do. |
| 21:39.49 | slav3_kitten | yea you're right |
| 21:40.11 | dijib | im so about to effin quit this |
| 21:40.17 | dijib | i still cant get it working |
| 21:40.22 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 21:40.22 | dijib | so does it go like this |
| 21:40.27 | slav3_kitten | dijib, did you try a bigger hammer? |
| 21:40.32 | dijib | mutt -> postfix -> sendmail |
| 21:40.40 | dijib | get me RDX that will fix it |
| 21:40.54 | dijib | Nov 14 16:39:16 swissarms sm-msp-queue[27383]: qAELMZd5026761: to=solowtel@gmail.com, ctladdr=asterisk (500/500), delay=00:16:41, xdelay=00:00:00, mailer=relay, pri=174542, relay=[127.0.0.1], dsn=4.0.0, stat=Deferred: Connection refused by [127.0.0.1] |
| 21:40.59 | slav3_kitten | let me make some calls to the RCMP |
| 21:41.47 | dijib | pffft |
| 21:41.56 | p3nguin | postfix OR sendmail. Not both. |
| 21:42.00 | dijib | BATF |
| 21:42.06 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 21:42.18 | dijib | not both... either one that will work. i dont care which one |
| 21:42.43 | dym | Hey - is there a way to somehow implement the playing of a soundfile to a queuemember, before a call is passed to him? As in the memberextension rings, member picks up, file is played and then call is connected? |
| 21:42.58 | dym | (using Queue App, obviously) |
| 21:47.02 | dijib | Nov 14 16:46:24 swissarms sendmail[27867]: qAELkOOR027867: to=solowtel@gmail.com, ctladdr=asterisk (500/500), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=84533, relay=[127.0.0.1] [127.0.0.1], dsn=4.0.0, stat=Deferred: Connection refused by [127.0.0.1] |
| 21:47.21 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 21:47.38 | dijib | ok postfix. relayhost= mail.mailserver.com set |
| 21:49.55 | dym | Found it - "announce" in queues.conf is my friend :) |
| 21:52.12 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 22:00.22 | dijib | i beg of you: http://pastebay.net/1155838 |
| 22:01.53 | dym | sounds more like a postfix issue |
| 22:01.55 | dym | #postfix |
| 22:03.04 | dijib | yeh i guess eh |
| 22:03.44 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 22:11.19 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 22:17.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:19.47 | dijib | who wants to send me a fax? |
| 22:28.55 | AkkerKid | anyone if you can set a polycom phone to boot from a different tftp server without touching the physical phone? the web interface doesn't help |
| 22:29.11 | AkkerKid | can a put a file into the tftp server directing the phones to a new tftp server? |
| 22:34.25 | *** join/#asterisk fritz09 (~Adium@2002:542e:5694:e472:b86c:2d28:7319:ec41) |
| 22:34.29 | Chainsaw | AkkerKid: It should listen to DHCP options. |
| 22:34.45 | Chainsaw | AkkerKid: You could shut and then reopen the switch port it is on to powercycle it. |
| 22:35.05 | Chainsaw | AkkerKid: (I appreciate that you have no SIP session to send a notify over) |
| 22:35.25 | ruben231 | <PROTECTED> |
| 22:35.46 | p3nguin | Possibly. |
| 22:36.00 | dijib | he needs to eat his raw fish |
| 22:36.31 | ruben231 | <PROTECTED> |
| 22:36.36 | dijib | postfix is working now, fax emails are going through... now i need a Email to Fax Gateway |
| 22:36.39 | dijib | anybody |
| 22:36.59 | p3nguin | ruben231: I still don't know what you're trying to accomplish. |
| 22:38.16 | AkkerKid | Chainsaw: if the polycoms are set manually to look for the tftp server by IP... they wouldn't even bother querying DHCP for it, right? |
| 22:39.21 | AkkerKid | basically my phone server is moving to a new ip address and i need to update off-site ip phones available only via VPN and idiot users. |
| 22:39.48 | AkkerKid | I'm hoping there's a way to change the manually set tftp server address via VPN |
| 22:40.14 | Chainsaw | AkkerKid: Manual provisioning would take precedence, yes. |
| 22:40.32 | jmetro | should have done a dhcp option on the firewall |
| 22:40.52 | AkkerKid | i didn't have any luck with DHCP on the polycoms |
| 22:41.01 | AkkerKid | the linksys' worked that way though... |
| 22:41.08 | jmetro | option 66 works fine for me |
| 22:41.31 | AkkerKid | is there any way to override the manual tftp option remotely? |
| 22:41.44 | AkkerKid | jmetro: what firmware revision are you running? |
| 22:41.51 | AkkerKid | I'm still 3.0-3.2 |
| 22:43.28 | jmetro | no idea. factory grabs it fine. |
| 22:43.44 | AkkerKid | hmmm... |
| 22:44.19 | AkkerKid | maybe just asking them to reset to factory would be better than trying to have the remote users manually input a new address... |
| 22:44.25 | jmetro | probably. |
| 22:44.26 | AkkerKid | i'll give it a whirl |
| 22:44.34 | jmetro | you can remote into the phones IP and factory them |
| 22:44.41 | AkkerKid | oh yeah? |
| 22:44.46 | AkkerKid | i didn't see that in the web ui |
| 22:45.05 | jmetro | i knwo one of the brands cant factory from web.. either aastra or polycom |
| 22:45.30 | AkkerKid | is there some kind of ssh or telnet option i'm missing out on? |
| 22:46.26 | AkkerKid | i CAN factory my aastra 6757i ct from web ui |
| 22:46.50 | AkkerKid | well i'm out for the night |
| 22:46.57 | AkkerKid | thanks all! |
| 22:48.16 | jmetro | there is a button shortcut on polycoms to factory - learn it and ask the users. just make sure to set your dhcp option first or they wont go anywhere. |
| 22:48.37 | ruben231 | p3nguin: actually im doing survey call that numbers im calling are IVR and by standard press 3, i wanted to automate it on my dialplan that wait time and send dtmf 3 if possible |
| 22:49.24 | p3nguin | Use a macro. |
| 22:49.51 | p3nguin | What will you be doing after you send a 3? |
| 22:50.56 | jmetro | youre calling IVR's or are you calling users who then dial 3 to your IVR |
| 22:51.06 | p3nguin | Or by what means will you be conducting the survey after calling? |
| 22:53.28 | p3nguin | If you'll just show me the whole picture, this terrible problem you're having with having to send a 3 is probably not a big deal at all. |
| 22:54.12 | p3nguin | I can think of more than one method used for calling a given number and then sending a 3, followed by audio messages and IVR of my own. |
| 22:55.10 | p3nguin | But your lack of attentiveness in the matter is causing me to quickly lose interest. |
| 22:56.19 | ruben231 | the procedure is calling IVR numbers which have like 90 percent dtmf selection to be 3 only.. |
| 22:56.51 | ruben231 | and during the call i need somehow like a pause time (wwww) so before sending dtmf 3 |
| 22:57.14 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 22:57.54 | p3nguin | We've already established THAT part. |
| 22:57.55 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 22:58.00 | p3nguin | Move on to what I just asked you. |
| 22:58.28 | ruben231 | after sending 3 then calls will be connected thats all |
| 22:58.40 | ruben231 | after proceed to next call again |
| 23:00.29 | ruben231 | oops during sending of dtmf 3 i have a voice record message already that would play auto then after will hangup and proceed to next call again with an IVR number selecting option 3 again and play audio, thats the cycle |
| 23:01.18 | p3nguin | Now tell me HOW you are doing that. |
| 23:02.33 | ruben231 | im using a prebuilt system who have settings already for the calls, problem only is the pause time and the auto sending of dtmf3 |
| 23:02.35 | ageis | possible reasons for peer going to unavailable and " Everyone is busy/congested at this time" when they are shown as reachable/OK? |
| 23:04.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bhfttwippfequgpb) |
| 23:04.26 | p3nguin | Since you can't tell me by what means you are continuing with your survey after connecting the call, I'm only going to tell you to either use a macro within your Dial() or use SendDTMF() after the call is connected. The rest is up to you. |
| 23:05.59 | dijib | anybdy have a good guide to do the email to fax gateway |
| 23:06.37 | dijib | asterfax? |
| 23:08.54 | [TK]D-Fender | ageis: that message means nothing by itself. show us the complete call |
| 23:12.30 | ageis | [TK]D-Fender: kevin-mobile tries to call sean http://pastebin.com/2ukMSvhX |
| 23:12.42 | dijib | must be asterfax |
| 23:12.44 | ageis | (extension nums are correct) |
| 23:13.34 | [TK]D-Fender | ageis: -- Executing [91@outgoing:1] Dial("SIP/kevin-mobile-00000000", "SIP/91") in new stack |
| 23:13.52 | [TK]D-Fender | ageis: testserv*CLI> sip show peers <---- this is NO peer "91" to call |
| 23:13.56 | [TK]D-Fender | ageis: that is not a PEER NAME |
| 23:14.30 | [TK]D-Fender | ageis: Dial(SIP/thinamehadbetterbeinsip.confinsquarebrackets) |
| 23:14.47 | ageis | That's not the issue |
| 23:14.51 | ageis | That's why I just told you extension nums are correct |
| 23:14.58 | ageis | extension 91 dials sean |
| 23:15.01 | ageis | who is a sip peer |
| 23:15.04 | ageis | and is reachable |
| 23:15.04 | p3nguin | SIP/91 is not an extension number. |
| 23:15.14 | ageis | Ok |
| 23:15.17 | ageis | So how should I change my Dial() command |
| 23:15.18 | [TK]D-Fender | [2012-11-14 18:11:43] WARNING[8811]: chan_sip.c:5450 create_addr: No such host: 91 <- NOT correct |
| 23:15.23 | [TK]D-Fender | there is no [91] in sip.conf. |
| 23:15.24 | p3nguin | YOu want to call SIP/sean. |
| 23:15.30 | [TK]D-Fender | put a PEER NAME you actually HAVE in there |
| 23:15.39 | ageis | p3nguin: tried and it says "extension not found" |
| 23:15.46 | p3nguin | Of course. |
| 23:15.47 | [TK]D-Fender | you can't just reference some magical mnumber. You used a TEXT NAME |
| 23:15.51 | [TK]D-Fender | that is what you ahve to DIAL |
| 23:15.52 | p3nguin | sean is not an extension. |
| 23:15.58 | ageis | lol |
| 23:16.04 | dijib | guys is there an option? |
| 23:16.09 | ageis | You've seen my dialplan and sip.conf, can you tell me what im doing wrong? |
| 23:16.17 | p3nguin | We just did. |
| 23:16.23 | ageis | No you didn't |
| 23:16.23 | [TK]D-Fender | [18:15]p3nguinYOu want to call SIP/sean. <--- |
| 23:16.26 | ageis | is it my Dial command? |
| 23:16.40 | slav3_kitten | decides he's going to come back when IQs return to normal |
| 23:16.45 | ageis | [TK]D-Fender, I tried, and exten is not found... Do you mean to tell me I should have exten => sean in my dialplan? |
| 23:16.54 | p3nguin | ~devicenames |
| 23:16.54 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 23:17.13 | [TK]D-Fender | aeUnderstand there is a difference beter the NUMBEr you dial and the SIP ENTRY a Dial command may just HAPPEN to call |
| 23:17.19 | [TK]D-Fender | ageis: ^ |
| 23:17.26 | p3nguin | If Sean is reached by extension 91, and extension 91 dials SIP/sean, you'll be good if [sean] exists. Which it does. |
| 23:17.29 | [TK]D-Fender | I dial 91 on my phone and JOHN gets called |
| 23:17.46 | ageis | p3nguin: my Dial command is SIP/${EXTEN} is that incorrect? |
| 23:17.49 | p3nguin | No |
| 23:18.01 | p3nguin | err... |
| 23:18.02 | p3nguin | yes it is incorrect |
| 23:18.04 | ageis | Ok |
| 23:18.06 | [TK]D-Fender | ageis: You you haven't [[] entires in sip.conf with NUMBER in them? NO |
| 23:18.08 | ageis | So what should it be |
| 23:18.11 | p3nguin | damn double negatives |
| 23:18.15 | [TK]D-Fender | SIP/SEAN <------ |
| 23:18.16 | ageis | [TK]D-Fender I understand the difference |
| 23:18.30 | p3nguin | Extension 91 needs to Dial() the device the person uses. |
| 23:18.37 | p3nguin | In Sean's case, it is SIP/sean |
| 23:18.38 | ageis | The problem is the dial command in my outgoing context dials ${EXTEN} |
| 23:18.47 | p3nguin | Fix it. |
| 23:18.48 | ageis | Ok, waht's the asterisk command to abstract that in a variable? |
| 23:18.55 | ageis | er asterisk variable |
| 23:18.56 | [TK]D-Fender | NONE |
| 23:19.04 | p3nguin | I build it out in the astDB. |
| 23:19.05 | [TK]D-Fender | There is no variable. |
| 23:19.05 | ageis | p3nguin: Ok, extension 91 already does that |
| 23:19.16 | ageis | This works on a live server. Same config. |
| 23:19.19 | p3nguin | You said extension 91 dials SIP/${EXTEN} |
| 23:19.26 | p3nguin | WHich is copletely different. |
| 23:19.32 | ageis | p3nguin: No no |
| 23:19.40 | ageis | p3nguin: extension 91 in my incoming context dials SIP/sean. |
| 23:19.48 | ghost75 | http://ghostadmin.dyndns.org:8081/geoiptest.php <- what do you see here |
| 23:20.02 | p3nguin | You should be using more appropriate context names to make things less confusing. |
| 23:20.19 | p3nguin | For phones, an appropriate context name might be, surprise, "phones". |
| 23:20.26 | p3nguin | Not outgoing, not incoming. |
| 23:20.55 | [TK]D-Fender | Looking for 91 in outgoing (domain 10.0.1.21) |
| 23:20.57 | slav3_kitten | ag4ve, i use localsets to denote my phones |
| 23:21.04 | [TK]D-Fender | <PROTECTED> |
| 23:21.05 | slav3_kitten | err ageis *** |
| 23:21.07 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-qrwfxutypvetrbtg) |
| 23:21.08 | ageis | [TK]D-Fender you looked at my config on pastebin right. It has an incoming and an outgoing context |
| 23:21.10 | slav3_kitten | fucking tab fail |
| 23:21.30 | p3nguin | While you can use whatever name you want, you aslo need to be capable of discerning how the dialplan works. And with names being non-indicative, it's much harder. |
| 23:21.57 | [TK]D-Fender | ageisp3nguin: my Dial command is SIP/${EXTEN} is that incorrect? <- NONE of your sip.conf entries are NUMERIC. Yes this is WRONG. you think something can be variable based on the number. Your entires are NOT numeric at all. |
| 23:22.09 | *** join/#asterisk lantizia (~lantizia@cpc6-stok15-2-0-cust8.1-4.cable.virginmedia.com) |
| 23:22.35 | ageis | Right |
| 23:22.36 | [TK]D-Fender | Forget about "patterns", because that's just not happening. |
| 23:22.52 | p3nguin | I would never call my context for my phone something like "incoming." |
| 23:23.00 | [TK]D-Fender | You are going to have to make 1 for each because that number has so abstract association for you to a TEXT SIP peer name you intend to dial. |
| 23:23.02 | slav3_kitten | p3nguin, correct me if i'm wrong but you'd want to use like [inbound] for calls from your did to your server, [outbound] from calls from your server to a voip provider, and [localsets] for your local phones at your site? |
| 23:23.03 | p3nguin | Incoming would indicate incoming from some outside entity. |
| 23:23.08 | [TK]D-Fender | so you are going to have make a PILE of extens for this |
| 23:23.15 | ageis | Someone else set this up, I just became the admin. |
| 23:23.21 | [TK]D-Fender | ~book |
| 23:23.21 | 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 |
| 23:23.25 | p3nguin | slav3_kitten: Those names are indicative, and thereby receive approval in most cases. |
| 23:23.25 | [TK]D-Fender | ageis: ^^ |
| 23:23.31 | [TK]D-Fender | ageis: time to learn your basics |
| 23:23.36 | ageis | All the extens are set up, as I said this works on a live server. I suspect problem is with Dial() command in outgoing context. |
| 23:23.43 | ageis | Is there an asterisk variable like ${SIPID} |
| 23:23.44 | [TK]D-Fender | ageis: Chapter 5 isn't just your friend ... it's the LAW |
| 23:23.50 | ageis | Roger that |
| 23:23.53 | slav3_kitten | ageis, i fully support the book |
| 23:23.54 | [TK]D-Fender | ageis: There is is you define it. |
| 23:24.03 | p3nguin | ageis: Like I said, build it out in the astDB, then you can make it as variable as you want. |
| 23:24.19 | [TK]D-Fender | p3nguin: Lets not complicate this... |
| 23:24.20 | slav3_kitten | god i have a fucking headache. brb |
| 23:24.25 | p3nguin | Oops. |
| 23:24.47 | p3nguin | You can also make GLOBALS if you wish. |
| 23:25.07 | p3nguin | SEAN=SIP/sean |
| 23:25.18 | p3nguin | Dial(${SEAN},30) |
| 23:25.33 | [TK]D-Fender | Doesn't really simplify much. |
| 23:25.50 | p3nguin | It's actually more work that way, but it's what he seems to HAVE TO HAVE. |
| 23:26.08 | p3nguin | must have |
| 23:26.36 | ageis | Ok, I think I have a whole extra context that I just need to get rid of. |
| 23:26.48 | ghost75 | wohoo new debian has asterisk 1.8 |
| 23:27.02 | p3nguin | I don't see what the problem is with naming the contexts according to their uses and creating extensions that dial real peers. |
| 23:27.34 | [TK]D-Fender | p3gHe inherited an * he does not yet understand. That's why. |
| 23:27.39 | [TK]D-Fender | p3nguin: He inherited an * he does not yet understand. That's why. |
| 23:27.53 | [TK]D-Fender | p3nguin: Another 2-3 years and this'll all be sorted out! |
| 23:28.01 | [TK]D-Fender | (j/k) |
| 23:28.02 | p3nguin | I can't wait. |
| 23:28.20 | [TK]D-Fender | Neither can I... |
| 23:28.23 | [TK]D-Fender | heads out. |
| 23:28.32 | ageis | I'm looking for a Dial command for this 'outgoing' context (WHICH USED TO BE VOICEPULSE GATEWAY) that's flexible and will pass the variable SIP id. I think it would be better if all calls only went to one context |
| 23:28.42 | ageis | the one in which the extens are defined |
| 23:28.50 | ageis | later, thanks. |
| 23:29.10 | *** join/#asterisk Kako_ (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 23:29.24 | p3nguin | Don't use Dial if you are replacing the outgoing context with internal extensions. |
| 23:30.17 | ageis | ok, what then? |
| 23:30.19 | p3nguin | You have two choices: change the contexts of the phones' peers to use AN APPROPRIATE CONTEXT NAME where the extensions to reach people are actually located, or include that existing context that I already mentioned in that outgoing context. |
| 23:30.48 | p3nguin | Again, I tell you, "phones" is an appropriate context name for phones' extensions to be located. |
| 23:31.37 | ageis | p3nguin: yeah, the one where the extensions are is included in the outgoing context |
| 23:31.38 | ageis | very good |
| 23:31.48 | ageis | and I know you mean include => |
| 23:33.02 | p3nguin | It does not make sense for your extensions for your phones to reside in a context called "outgoing". |
| 23:33.52 | ageis | p3nguin: They never were, and that's not what I'm proposing |
| 23:34.45 | p3nguin | When I configure dial plan for local phones, I create a context named "phones" and assign it to all the local phones. |
| 23:34.47 | *** join/#asterisk moy (~moy@unvlon55-1176057127.sdsl.bell.ca) |
| 23:35.21 | p3nguin | In that "phones" context, I include any context, in a strategic order, that I want my phones to be able to access by dial. |
| 23:35.45 | ageis | that makes sense |
| 23:35.51 | p3nguin | So my phones context contains other contexts such as internals, outgoing, voicemail. |
| 23:36.12 | p3nguin | The "internals" context contains all of the extensions that are used to Dial() all of the local phones. |
| 23:36.40 | p3nguin | The "outgoing" context contains all of the extensions used to dial out of the box, typically to the ITSP to send the calls to the PSTN. |
| 23:36.56 | ghost75 | how was that called where you could see what changed from version x to y in asterisk |
| 23:37.08 | p3nguin | The voicemail context contains the extensions for the voice mail main menu as well as everyone's direct voice mail numbers. |
| 23:37.17 | p3nguin | ghost75: changelog? |
| 23:37.33 | p3nguin | You can find that on the web site. |
| 23:38.20 | ghost75 | i think was not called changelog |
| 23:39.00 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:39.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:39.05 | p3nguin | This design is an example of proper dial plan context hierarchy. |
| 23:39.55 | p3nguin | The incoming context does have the ability to reach extensions in the internals context, but the internals context does NOT have any way to reach anything other than those phones, especially not a route to the outgoing context. |
| 23:40.02 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 23:40.33 | ageis | p3nguin: but technically for what im doing, internal testing, Couldn't I have everything in one context? |
| 23:41.20 | ghost75 | ah upgrade it was called |
| 23:41.59 | slav3_kitten | ageis, technically you can do whatever you want. however if you want people to help you fix it you're going to run into issues if it's some uggly ass lump of code that needs much fuss to figure it out |
| 23:42.02 | p3nguin | Why would you want to set up things, even for testing, with extremely poor practice? |
| 23:42.41 | p3nguin | I don't buy into the bullshit about "just doing it this way for now and fix it later." |
| 23:42.52 | p3nguin | Learn to do it right. Do it right. |
| 23:43.03 | p3nguin | Even. For. Testing. |
| 23:43.10 | *** part/#asterisk ruben231 (~OpenDial@112.198.78.99) |
| 23:43.17 | p3nguin | If you learn to do it wrong, you'll continue to do it wrong. |
| 23:43.23 | ghost75 | why only have 1 context? |
| 23:43.34 | p3nguin | He prefers bad practice? |
| 23:44.03 | p3nguin | The real reason is because he doesn't know any better. That's why I'm trying to teach him a better way. |
| 23:44.10 | ghost75 | i love having loads of contexts and small ones |
| 23:44.39 | p3nguin | Having good context hierarchy is imperative. |
| 23:45.16 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-jrbvhayscmbbkafh) |
| 23:49.20 | slav3_kitten | p3nguin, i think coding in C/C++ in years has caught up to me. i keep using ; at the end of lines and // for comments purely out of habit |
| 23:49.54 | p3nguin | My dial plan has a ; at the end of every single line. |
| 23:50.05 | slav3_kitten | ; is comment no? |
| 23:50.23 | p3nguin | It is used for commenting, yes. |
| 23:50.39 | slav3_kitten | you comment every line? |
| 23:50.42 | p3nguin | It's like "this marks the end of the important stuff." |
| 23:50.54 | p3nguin | Just like you mentioned for your C stuff. |
| 23:51.17 | slav3_kitten | fair point |
| 23:51.32 | slav3_kitten | never really thought of it in that light however |
| 23:52.07 | p3nguin | AEL needs a ; at the end of each line. |
| 23:52.27 | p3nguin | I just happen to use one on the other format, too. |
| 23:53.41 | slav3_kitten | googling AEL... it looks just like C |
| 23:54.40 | slav3_kitten | like freakishly c like |
| 23:56.40 | p3nguin | ~ael |
| 23:56.41 | infobot | AEL is the Asterisk Extension Language -- a dial plan language with "C like" syntax. |
| 23:56.47 | p3nguin | heh |
| 23:57.51 | slav3_kitten | you an your fancy bot :P |
| 23:58.11 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 23:58.39 | ghost75 | ael is for super fancy dial plan? |
| 23:59.23 | p3nguin | Not really, since it is actually less flexible than the INI style of extensions.conf. |