00:00.18 | Katty | i believe it is a candy. |
00:01.12 | eppigy | IT IS MY WORD |
00:01.16 | eppigy | AND THEY CANNOT HAVE IT |
00:01.21 | Katty | yes, dear. of course. |
00:01.55 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
00:02.15 | [8none1] | eppigy: Sounds like you have a good legal case |
00:02.31 | Katty | volunteers to be lawyer. |
00:03.06 | eppigy | yes you may have something |
00:03.27 | jaytee | wow, I remember Kazaa. for a P2P music swapping site that thing was like a hooker with every STD known to man |
00:03.43 | eppigy | throwback |
00:03.48 | Katty | eww. |
00:03.54 | raden | LMAO |
00:04.00 | raden | i hate cisco ASA :( |
00:04.01 | [8none1] | As a friend of mine says, digital herpes |
00:04.05 | eppigy | raden: why? |
00:04.10 | Katty | the Hiv. |
00:04.17 | jaytee | ever see Ice Pirates? |
00:04.23 | [8none1] | no |
00:04.38 | raden | epigy in freaking ASDM i cannot figure out how to route internal lan to outide :( |
00:04.45 | eppigy | son |
00:04.54 | eppigy | to start close the asdm immediately |
00:05.01 | raden | okk |
00:05.03 | jaytee | really bad scifi movie with Robert Urich and the girl from Little House on the Prairie. They had Space Herpes in it. |
00:05.04 | eppigy | google asa command reference |
00:05.16 | eppigy | then |
00:05.20 | raden | i had da books in front of me |
00:05.21 | Katty | checks to see if that's on netflix |
00:05.22 | eppigy | what is your topology like? |
00:05.44 | eppigy | are there any other routers, or do they already have default routes to the asa? |
00:05.51 | Katty | woo! it's on netflix |
00:05.55 | Katty | queues it up for later tonight |
00:06.16 | jaytee | Katty, it's a crap movie |
00:06.24 | [8none1] | wow, Robert Urich . . . girl . . Little . . Herpes |
00:06.43 | Katty | that's okay. |
00:06.50 | Katty | i need something to put me sleep anyway. |
00:06.52 | raden | epigy , i have a DSL modem bridged on 0/0 a wifi bridged on 0/1 << thoose are to 2 diffrent ISP's |
00:07.02 | jaytee | not exactly Dread Pirate Roberts |
00:07.05 | raden | i have a procurve 1800 linked to ASA |
00:07.21 | eppigy | raden: you would want a default route going out the outside interface |
00:07.25 | Katty | jaytee: i watched a movie about a giant shark and octopus one night. |
00:07.36 | raden | epigy 0/1 is up at the moment |
00:07.40 | raden | outside2 |
00:07.41 | [8none1] | jaytee: You have 6 fingers on your right hand |
00:07.51 | Katty | jaytee: and a cheesy subtitled chinese movie about a chick who lost an arm and had a machine gun graphed on it. |
00:08.12 | eppigy | route <outisde int name> 0.0.0.0 0.0.0.0 <next hop ip> 1 |
00:08.57 | eppigy | as in the isp router |
00:09.02 | eppigy | for next hop ip |
00:10.24 | raden | dont i use my wan IP ? |
00:10.54 | eppigy | also if you get both isp interfaces up you can create a policy map to route certain traffic over each connection |
00:11.08 | eppigy | raden: well your outisde interface should have your wan ip |
00:11.17 | eppigy | from dhcp |
00:11.52 | raden | we have static IP's |
00:11.57 | eppigy | right |
00:12.09 | eppigy | it should still via dhcp |
00:12.19 | eppigy | I mean is this consumer internet? |
00:13.02 | raden | our wifi our static completly diffrent they authenticate via MAC |
00:13.10 | raden | business grade on both |
00:13.14 | eppigy | raden: what is the ip address and mask of your outside interface? |
00:13.16 | raden | DSL pppoe |
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00:13.28 | raden | wifi |
00:13.36 | raden | 64.108.141.237 |
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00:13.41 | raden | 255.255.255.128 |
00:13.54 | eppigy | that is a lot of ip's |
00:14.02 | raden | what u mean ? |
00:14.10 | raden | we have 1 |
00:14.13 | eppigy | that is 127 ip's |
00:14.16 | eppigy | in that subnet |
00:14.20 | eppigy | 126 |
00:14.21 | eppigy | I mean |
00:14.23 | raden | yeah |
00:14.33 | raden | spread between 2 towns |
00:14.46 | raden | DSL is |
00:14.50 | eppigy | what is the gateway ip for that? |
00:15.00 | eppigy | 64.108.141.129? |
00:15.05 | raden | gate 64.108.141.128 |
00:15.15 | eppigy | 128 is the subnet number |
00:15.30 | raden | thats what they gave eme |
00:15.37 | raden | sorry keyboar messed up |
00:15.56 | raden | DSL |
00:16.04 | raden | 69.179.99.17 |
00:16.17 | raden | 255.255.255.255 << which does not make sense |
00:16.26 | raden | 69.29.188.6 |
00:16.42 | eppigy | lets do this |
00:16.52 | eppigy | instead of making everyone in the channel want to kill use |
00:16.53 | eppigy | us |
00:17.07 | eppigy | pastbin the output of show int |
00:17.09 | CareBear\ | that's not a useful subnet mask |
00:17.11 | eppigy | and show route |
00:17.50 | jaytee | dave's my buddy, I'd never want to kill him |
00:18.04 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@70.31.3.81) |
00:18.21 | darkdrgn2k3 | hey guys, whats the major dif between 1.6 and 1.4? is there a big differnce |
00:18.52 | russellb | ~asterisk16 |
00:18.53 | infobot | new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
00:19.47 | darkdrgn2k3 | tips hat |
00:19.49 | darkdrgn2k3 | thanx |
00:20.09 | CareBear\ | I didn't get a lot of response to my observation of what might be a bug in 1.6.2.0-rc1: http://pastie.org/626741 the A has been chopped off the Allow header |
00:20.56 | Katty | and i'd tip my HAT |
00:20.57 | russellb | also, diffstat between 1.4 and trunk (dev tree for upcoming 1.6.3): 1020 files changed, 477432 insertions(+), 113817 deletions(-) |
00:21.02 | Katty | imagine THAT |
00:21.05 | russellb | In short, a ton has changed :-) |
00:23.47 | CareBear\ | Katty : was that a response to me? :) I was hoping for something more technical. Maybe even confirmation that it's a known issue. |
00:23.56 | CareBear\ | or even |
00:23.58 | CareBear\ | ~tracker |
00:24.28 | CareBear\ | ? |
00:25.00 | russellb | CareBear\: that's pretty bizarre. |
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00:25.16 | russellb | rc2 was just released, try updating to that before reporting anything |
00:25.21 | russellb | but the issue doesn't sound familiar .. |
00:25.29 | CareBear\ | russellb : could it be a freak thing with sip debugging? |
00:25.34 | russellb | it's possible |
00:25.39 | russellb | does it happen every time? |
00:25.42 | russellb | or did you see it just once? |
00:25.50 | CareBear\ | sadly not every time, no |
00:25.57 | russellb | more than once? |
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00:26.19 | CareBear\ | I've only noticed it once, the first time was very recently |
00:26.31 | CareBear\ | let me try to reproduce |
00:26.35 | russellb | Okay. |
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00:27.22 | darkdrgn2k3 | any one know of good place to get config for a spa3102? |
00:27.28 | darkdrgn2k3 | ?? spa3102 |
00:28.18 | mrbnet | I have installed asterisk on a debian system. It appears to work but every day or two I need to restart asterisk so calls will come in. |
00:28.28 | mrbnet | Any suggestions? |
00:28.52 | jaytee | RHEL 5.2 FTW! |
00:29.17 | russellb | mrbnet: something is locking up. Try installing the latest version, and if you still have a problem, join #asterisk-bugs for additional help. |
00:29.27 | russellb | latest version from asterisk.org, that is. |
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00:29.33 | [8none1] | mrbnet: I had a similar issue on a debian system with a packaged install. |
00:29.43 | manxpower | ~answers |
00:29.44 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
00:30.00 | [8none1] | It had something to do with recording ODBC CDR data and would lock channels |
00:30.14 | [8none1] | Just install from source and it should work better |
00:30.25 | manxpower | The only GOOD Asterisk package is an UNINSTALLED Asterisk package |
00:31.25 | mrbnet | 8none1: I had so much trouble with the packaged version that I installed from source. |
00:31.44 | mrbnet | 8none1: What version are you running? I am on 1.4.26 |
00:31.59 | [8none1] | I'm using 1.6 right now |
00:32.09 | [8none1] | May not be the same issue since you compiled from source |
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00:37.07 | [8none1] | Katty: I just stumbled upon this from your comment about the move you saw : http://craphound.com/images/bearsharktopus-30363-1253244193-27.jpg |
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00:37.07 | darkdrgn2k3 | welcome back |
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00:37.20 | Katty | http://www.imdb.com/title/tt1350498/ <- Movie. |
00:38.22 | Katty | From the movie: http://www.dreadcentral.com/img/reviews/megashark1b.jpg |
00:39.22 | [8none1] | I saw that in the trailer :D |
00:39.35 | [8none1] | funny |
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00:39.54 | Katty | Megaladon is a very fascinating creature. |
00:40.39 | Katty | http://upload.wikimedia.org/wikipedia/commons/thumb/0/07/Megalodon_scale1.png/800px-Megalodon_scale1.png <- for size reference. |
00:41.30 | Katty | it oculd eat a Blue Whale in a couple chomps. |
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00:44.16 | Katty | and the Plesiosaur isn't extinct either. |
00:44.25 | Katty | One was found off the coast of Japan in 1977, rotting. |
00:45.48 | Katty | http://www.discoverynews.us/DISCOVERY%20MUSEUM/CreaturesFromTheDeep/CreaturesIMAGES/Plesiosaur_4_large.jpg <- Plesiosaur |
00:46.23 | coppice | that report from japan was bogus |
00:46.54 | kevinh90 | howdy |
00:46.55 | Katty | link? |
00:47.37 | coppice | try looking up anything about plesiosaur and japan. the sensationalist ones says its a sea monster. most say it was a basking shark |
00:47.59 | Katty | googles. |
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00:58.12 | eppigy | NEIN |
00:58.33 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
00:59.42 | leifmadsen | HI! |
00:59.56 | carrar | HI!! |
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01:02.25 | shmaltz | hi!!!!!!!!!!! |
01:02.58 | [8none1] | |-| ! |
01:04.21 | shmaltz | how come I can see calls being made to sip/5371 and I don't have a sip/5371? |
01:09.58 | *** join/#asterisk jpcansa (n=jpbenavi@201.200.55.138) |
01:10.36 | shmaltz | ~anyone |
01:10.37 | infobot | rumour has it, anyone is Instead of looking for mentors from specific projects here, try the project's IRC channel, likely on this server as well. |
01:10.49 | shmaltz | ~sleep |
01:10.50 | infobot | well, sleep is overrated, and a poor substitute for caffeine. |
01:11.53 | jpcansa | does anybody has an example of code to store a number dialed byt a called on an IVR and then dial it thru a different gateway?? |
01:13.04 | shmaltz | jpcansa, example |
01:14.22 | jpcansa | shmaltz, someone calls to my IVR from pstn and dials a number, i want * to dial that number thru VoIP gateway |
01:14.42 | shmaltz | jpcansa, then just use app_dial |
01:14.49 | shmaltz | something along these lines: |
01:14.56 | shmaltz | caller is dumped in s,1 |
01:15.39 | shmaltz | s,n,Read(NUMTODIAL|plsenternum) |
01:15.40 | shmaltz | s,n,Dial(Sip/Gatway/${NUMTODIAL}) |
01:16.44 | jpcansa | good, let me try that |
01:17.36 | jpcansa | shmaltz: "plsenternum" is a recording? |
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01:18.00 | shmaltz | jpcansa, if you record it yes |
01:18.22 | shmaltz | jpcansa, please check docs for exact args for each app |
01:18.44 | jpcansa | yeah i got it |
01:18.55 | jpcansa | thanks for your help |
01:25.36 | shmaltz | anyone here as bored as I am? |
01:26.15 | shmaltz | ~bored |
01:26.16 | infobot | La ... lalalala ... beer! |
01:26.22 | shmaltz | ~beer |
01:26.23 | infobot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
01:26.33 | shmaltz | ~reset |
01:26.34 | infobot | well, reset is not influencing the message about the kernel on hyper terminal |
01:26.44 | shmaltz | ~sex |
01:26.44 | infobot | [~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep |
01:27.40 | shmaltz | ~prenant |
01:27.45 | shmaltz | ~nyc |
01:27.46 | infobot | methinks nyc is where the secret organization l.a.u.r.a. is headquartered (i'm on to you tzu!) |
01:27.57 | shmaltz | ~ny |
01:27.58 | infobot | ny is, like, a place where they make the best pizza, the best hot dogs, and the nicest hookers |
01:28.00 | shmaltz | ~nj |
01:28.01 | infobot | hmm... nj is home to the sopranos. Fogedaboudit! |
01:28.19 | shmaltz | ~wa |
01:28.20 | infobot | wha? what in $DIETY's name are you talking about? wtf? |
01:28.28 | shmaltz | ~uk |
01:28.29 | infobot | well, uk is a place where they don't know english to well |
01:28.36 | shmaltz | ~paris |
01:28.37 | infobot | [paris] Capital of France, or Paris AZ, MI, NE, OR, WA. |
01:28.46 | shmaltz | ~AZ |
01:28.47 | infobot | [az] Azerbaijan |
01:28.55 | shmaltz | ~obama |
01:28.56 | infobot | well, obama is the 44th and current president of the United States of America |
01:29.03 | shmaltz | ~clinton |
01:29.04 | infobot | hmm... clinton is the best president we've had since george bush, dammit! or the only president we've had since george bush, dammit! or the Pants Dropper in Chief or the guy blowing away small countries or the president whose staff continues to classify cryptographic software as munitions or a shitty human being or stupid or in control of nuclear weapons or a psychopath, or clit-ton |
01:29.07 | manxpower | There's nothing wrong with playing with your bot as long as you do it in private and wash your hands after. |
01:29.10 | L2Logic | ~pastebin |
01:29.10 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
01:29.28 | manxpower | you can use /msg to talk to infobot too |
01:29.50 | shmaltz | oh so some ppl are alive |
01:30.01 | shmaltz | yes but it's way more fun to talk in public |
01:30.31 | shmaltz | ~rush limbo |
01:30.32 | infobot | ACTION sees limbo is dawdling and runs over and pushes limbo out the door |
01:30.51 | shmaltz | ~lol |
01:30.51 | infobot | lol is probably stands for Laughs Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
01:31.03 | shmaltz | ~biden |
01:31.23 | shmaltz | ~iraq |
01:31.24 | infobot | [iraq] a country |
01:31.24 | L2Logic | check this out... it's probably do what you're wanting to do to specificy a specific CID for a DISA application (and record the call) http://pastebin.com/d6ea6194 |
01:31.41 | L2Logic | it'll vs it's |
01:32.23 | L2Logic | we've had a president since Reagan ? |
01:33.00 | shmaltz | L2Logic, :) |
01:33.11 | shmaltz | well we don't currently have one that is sure |
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01:35.23 | L2Logic | i don't think Arnold S. gets as much airtime as Mr. O.... even with terminator reruns |
01:36.39 | L2Logic | i remember when I was a kid.. and we had three channels.. "the president is on, every channel".. now it's, "the president's on.. all the time..." |
01:37.31 | L2Logic | next thing we hear... "We interrupt this program to share with you that Mr. Obama will not be making an appearance on the news today" |
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01:41.31 | shmaltz | :P |
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01:47.35 | raden | night Katty |
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03:16.09 | p3nguin | Here's a registration debug from a problematic softphone: http://pastebin.ca/1575494 Does everything look okay/normal as far as the registration goes? |
03:17.44 | [TK]D-Fender | p3nguin: Yes |
03:18.44 | p3nguin | If it registered correctly, I don't understand why it doesn't make calls. |
03:18.50 | p3nguin | It receives them. |
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03:23.24 | dssman | Hey all, have a quick question... if I followme an exten is there any way to make it blind to the call creator... IE only play back a ringing tone rather then "please hold while we try to locate the person you are calling [moh]" |
03:24.21 | [TK]D-Fender | p3nguin: Registeration has jak-all to do with placing a call |
03:30.32 | p3nguin | Does it not show that SIP packets are able to reach the server from the client? |
03:31.23 | [TK]D-Fender | p3nguin: Looks lie so far. |
03:31.39 | [TK]D-Fender | p3nguin: of course it could be using split info between registrar and call servers |
03:31.48 | [TK]D-Fender | p3nguin: amongst a ton of other possibilities. |
03:32.16 | p3nguin | I'm so puzzled about it. |
03:32.21 | [TK]D-Fender | p3nguin: I don't know twinkle specifically |
03:32.33 | [TK]D-Fender | p3nguin: try another soft-phone in the meanwhile |
03:32.56 | p3nguin | Tried ekiga, too. It's in the same situation. Can register and receive calls, can't make calls. |
03:34.16 | [TK]D-Fender | p3nguin: You get NO packets on call attempts |
03:34.18 | [TK]D-Fender | ? |
03:35.03 | p3nguin | correct |
03:36.04 | p3nguin | When I showed you guys earlier what debug there was when he tried making a call, the info was from qualify=yes. I turned that off, and no debug info came out when he tried making a call. |
03:36.36 | parker | hy, i'm newer with asterisk and i've have one problem, i'm using asterisk 1.4 with dahdi 2.2 and one interface E1 (INTELBRAS), i have one queue named RECEPCAO, when a call come througth E1 link the call stay mute and in 5 seconds drop it, but if a dial to the same queue by internal side, everything is OK, can anyone help me? |
03:38.38 | [TK]D-Fender | p3nguin: this tells me that yrou * side probably isn't properly forwarded |
03:39.25 | p3nguin | The particular Linux computer that these two softphones are on has a direct connection to the internet, and it acts as a gateway for one Windows computer. It's using iptables/masquerade to create NAT for the other computer. Running zoiper on the Windows computer which connects through the Linux box works correctly. Zoiper makes and receives calls. |
03:40.26 | p3nguin | That makes me think that * is correctly communicating with that remote network. |
03:40.40 | [TK]D-Fender | p3nguin: I thinking your * side is bad. not the client |
03:40.57 | [TK]D-Fender | p3nguin: And what are you doing with 2 softphones on 1 PC? |
03:41.07 | [TK]D-Fender | p3nguin: that is in itself a disaster waiting to happen |
03:41.12 | p3nguin | Trying to solve this probably, obviously. |
03:41.14 | [TK]D-Fender | p3nguin: as they fight for SIP portm etc |
03:41.24 | [TK]D-Fender | p3nguin: You are polluting your test |
03:41.52 | p3nguin | You suggested another SIP phone. Why would you suggest that if it pollutes the test? |
03:42.23 | p3nguin | We obviously have no reason to run twinkly and ekiga at the same time. That would be bad. And useless. |
03:42.27 | [TK]D-Fender | p3nguin: Hold on.. no simultaneous? |
03:42.32 | p3nguin | no no |
03:43.00 | p3nguin | Only for testing. DIdn't even run zoiper on the other machine until we went troubleshooting. |
03:43.06 | p3nguin | never two at once |
03:44.29 | parker | hy, i'm newer with asterisk and i've have one problem, i'm using asterisk 1.4 with dahdi 2.2 and one interface E1 (INTELBRAS), i have one queue named RECEPCAO, when a call come througth E1 link the call stay mute and in 5 seconds drop it, but if a dial to the same queue by internal side, everything is OK, can anyone help me? |
03:44.34 | p3nguin | I also have another Windows client on a completely different network behind NAT which runs zoiper successfully. Is it possible that zoiper is more "forgiving" about some screwed up setting that I might have? |
03:46.12 | p3nguin | And could a debug of a good call using zoiper help to diagnose anything? I don't recall if I provided that debug earlier or not. I just know there was no debug when trying to call out using neither twinkle nor ekiga. |
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03:46.37 | [TK]D-Fender | p3nguin: what do the softphoens tell you when you try to dial? |
03:47.38 | p3nguin | The status on the window shows that it is trying to place a call, but then about 30 seconds later it says that it timed out. |
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03:48.03 | *** mode/#asterisk [+o Cresl1n_] by ChanServ |
03:49.18 | [TK]D-Fender | p3nguin: Still screams networking error |
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03:49.40 | brunner | asterisk is working well with 200 callers =D |
03:50.22 | p3nguin | I'll check anything you want checked if you have an idea where to look or what to look for. |
03:51.27 | p3nguin | I wonder if it's possible to run zoiper through wine. That could be an interesting test. |
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04:00.56 | kevinh90 | hi |
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04:47.46 | CareBear\ | Where does asterisk get the username that it puts into From: when sending an INVITE ? |
04:49.39 | [TK]D-Fender | CareBear\: what do you see in there? |
04:49.44 | CareBear\ | asterisk |
04:51.01 | CareBear\ | am using asterisk as client |
04:52.48 | [TK]D-Fender | CareBear\: pastebin a complete call attempt with SIP debug enabled along with your peer config masing only passwords |
04:52.53 | [TK]D-Fender | ~pb |
04:53.03 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
04:58.16 | CareBear\ | http://pastie.org/private/sjoq4a3eszwotwv2ja5g |
04:59.28 | CareBear\ | is it a bad idea to use the same hostname both on client and on server - although the client registers with the server and is reachable through there with the hostname? |
05:00.17 | *** part/#asterisk mumtazah (n=mumtazah@142.78.48.60.wmu01-home.tm.net.my) |
05:00.50 | [TK]D-Fender | CareBear\: You are showing the call being received. I want to see the call being SENT |
05:01.21 | CareBear\ | from .9 is what is SENT |
05:01.26 | CareBear\ | the server is .7 |
05:02.25 | [TK]D-Fender | CareBear\: You aer showing me debug from the receiving server. I want to see the sender's configs & debug |
05:03.31 | CareBear\ | the debug from the sender looks just the same though.. |
05:03.46 | CareBear\ | (as for what goes onto the wire) |
05:07.24 | CareBear\ | http://pastie.org/private/u3jkoyie9ahvicseqyhtq has it, along with the extension I dial |
05:08.00 | CareBear\ | a new call-id, but otherwise same |
05:12.49 | [TK]D-Fender | CareBear\: exten => tablet,1,Dial(SIP/tablet@stuge.se,15) |
05:13.11 | [TK]D-Fender | CareBear\: You are directly dialing the host. This means you are not using your peer, and that means there is no auth attached |
05:13.11 | CareBear\ | yes? |
05:13.16 | [TK]D-Fender | CareBear\: And also no identity |
05:13.24 | [TK]D-Fender | CareBear\: hence the "asterisk" |
05:13.49 | [TK]D-Fender | CareBear\: Dial(SIP/apeernamewithusernameetc/numbertodial) |
05:14.07 | CareBear\ | but that just moves the problem one step, doesn't it? |
05:14.20 | [TK]D-Fender | CareBear\: like Dial(SIP/peter/tablet) |
05:14.35 | [TK]D-Fender | CareBear\: No, it should not show "asterisk" |
05:15.06 | [TK]D-Fender | CareBear\: Of course it'd also be good to see the callerid of the originating call (before you call from that server to the other |
05:15.20 | Carlos_Tico | TK now i am getting this ...[Sep 23 00:14:52] NOTICE[159]: chan_sip.c:14035 handle_request_invite: Call from '6002' to extension '4549142' rejected because extension not found. |
05:15.58 | Carlos_Tico | 4549142 is the number i want to dial out from the sp3k |
05:16.00 | [TK]D-Fender | Carlos_Tico: And it means just what it says |
05:16.21 | CareBear\ | D-Fender : asterisk is the originating call, I set "Peter Stuge" in oss.conf |
05:16.33 | CareBear\ | okey, let's see. |
05:16.42 | [TK]D-Fender | Carlos_Tico: this is * refusing the call you are placing. this has nothing to do with calling OUT. * is refusing the call IN |
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05:20.30 | CareBear\ | .9 sends peter@ to .7 |
05:20.36 | CareBear\ | .7 says Found peer 'peter' for 'peter' from 213.88.146.9:5060 |
05:21.01 | CareBear\ | followed by 'Failed to authenticate device "Peter Stuge" <sip:peter@...' |
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05:25.49 | CareBear\ | D-Fender : I get the idea. I have to dial out through where I am known as peter@stuge.se before From will actually show that. |
05:26.41 | CareBear\ | Makes good sense for a PBX, but maybe slightly less for a softphone, which is what I'm using asterisk on my laptop as. :) |
05:27.09 | [TK]D-Fender | CareBear\: add these to your peer : fromuser=peter , sendrpid=yes , trustrpid= yes |
05:27.20 | [TK]D-Fender | CareBear\: add those last 2 lines to the other side |
05:28.00 | CareBear\ | still get the forbidden |
05:28.18 | CareBear\ | the server has type=friend for this registration btw |
05:28.33 | CareBear\ | would that be bad? |
05:31.19 | CareBear\ | this looks incomplete: Remote-Party-ID: "Peter Stuge" <sip:@stuge.se>;privacy=off;screen=no |
05:31.47 | [TK]D-Fender | CareBear\: put "username=peter for the other side |
05:32.37 | CareBear\ | no difference - because this header is coming from my laptop |
05:32.41 | CareBear\ | maybe? |
05:33.03 | CareBear\ | (there I already have defaultuser=) |
05:33.10 | CareBear\ | =peter |
05:33.37 | [TK]D-Fender | bad parm... don't use it |
05:33.46 | [TK]D-Fender | ok, all I have time for for tonight... |
05:33.52 | [TK]D-Fender | keep at it.. you';re almost there |
05:34.00 | CareBear\ | thanks for the help so far. :) |
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07:29.41 | ThoMe | hello |
07:29.47 | ThoMe | good morning. |
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08:01.12 | ThoMe | tzafrir_laptop: hello. |
08:01.21 | tzafrir_laptop | ho |
08:01.25 | ThoMe | tzafrir_laptop: can you help me with dahdi? |
08:01.33 | tzafrir_laptop | hopefully :-) |
08:01.38 | ThoMe | i would like use chanspy (i need dahdi for this?) |
08:02.12 | tzafrir_laptop | I don't think so |
08:02.13 | ThoMe | i have this problem, when i use the function ",w" (whipser) i can only hearing but not speak. |
08:02.18 | ThoMe | now I have found this: https://issues.asterisk.org/view.php?id=15660 |
08:02.35 | ThoMe | can you help me with get this branches? |
08:02.44 | ThoMe | i havent found a doku |
08:04.28 | ThoMe | oh ok http://www.digium.com/en/docs/asterisk_handbook/downloading_compiling.html |
08:04.35 | ThoMe | svn co http://svn.digium.com/svn/asterisk/trunk asterisk <<is this ok tzafrir_laptop ? |
08:06.16 | tzafrir_laptop | BTW: (unrelated) any comments on http://docs.tzafrir.org.il/dahdi-linux/#_live_install ? |
08:07.29 | tzafrir_laptop | ThoMe, it's fixed in 1.4 , so by now latest 1.4 tarball should include it |
08:08.06 | tzafrir_laptop | That handbook page needs some updating |
08:08.16 | ThoMe | hm. have 1.4.26.2 |
08:08.17 | ThoMe | hmm. |
08:08.19 | tzafrir_laptop | Won't actually work as-is |
08:11.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
08:13.03 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-71-143.link.net.pk) |
08:23.54 | ThoMe | tzafrir_laptop: hmm. |
08:23.57 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
08:23.57 | ThoMe | problem fixed |
08:24.02 | ThoMe | not in 1.4.26.2 :_( |
08:24.10 | ThoMe | now i have |
08:24.10 | ThoMe | Asterisk SVN-branch-1.4-r219816 built by root @ asterisk01 on a i686 running Linux on 2009-09-23 08:16:46 UTC |
08:24.45 | tzafrir_laptop | it was commited on late August |
08:27.28 | ThoMe | hmm. and version 1.4.26.2 ? |
08:29.51 | ThoMe | tzafrir_laptop: is it posible in a macro when hangup this includet: |
08:29.52 | ThoMe | exten => h,1,NoOp(Coaching Start) |
08:29.52 | ThoMe | exten => h,n,Set(DB(coach-eins/${nebenstelle})=nein) |
08:29.53 | ThoMe | ? |
08:29.56 | ThoMe | a h-extention? |
08:31.37 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
08:35.17 | cjk | hi, i have not voice on local channels, except if i do something like playpack(beep) which answers the calls before dialing another party. any idea why? this problem only exists if all call legs are on SIP. If there is some ZAP in the middle it works perfectly |
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08:53.42 | ThoMe | emmm |
08:53.57 | ThoMe | is it posible ${SIP_HEADER(TO)} <<here extract only digits |
08:53.58 | ThoMe | ? |
08:54.40 | jgoo | Hej people - I was having an issue with reload commands not working, so following a google find, i did asterisk -vvvvvvvvc - it loads instantly - however reload command sticks at Reloading module 'chan_sip.so' (...) Parsing... manager*.confs |
08:55.25 | kaldemar | ThoMe: use func CUT |
08:55.45 | ThoMe | kaldemar: hm. |
08:55.56 | jgoo | http://pastebin.me/25a3f01cca2f883313c8f80cb458dd3a |
08:57.58 | ThoMe | kaldemar: emm |
08:58.00 | ThoMe | kaldemar: I use |
08:58.01 | ThoMe | <PROTECTED> |
08:58.01 | ThoMe | <PROTECTED> |
08:58.03 | ThoMe | <PROTECTED> |
08:58.11 | ThoMe | is it posible in one step extract the number? |
08:59.57 | cjk | hi, i have a call coming over ZAP to asterisk1 then it goes over SIP to asterisk2 and then it goes back over SIP to asterisk1 where I execute the echo application. if i do not answer the call on asterisk2, before going back to asterisk1 i have no voice at all. any idea? |
09:00.44 | kaldemar | ThoMe: combine those lines |
09:01.36 | kaldemar | or maybe you can't, since CUT wants a variable name |
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09:05.19 | ThoMe | kaldemar: ah, works fine |
09:05.19 | ThoMe | <PROTECTED> |
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10:02.26 | ThoMe | hello? |
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10:06.13 | so_solid_moo | Hi; could anyone tell me how long it's supposed to take an Astribank to load its firmware? |
10:06.59 | tzafrir_laptop | A few seconds |
10:07.06 | so_solid_moo | oh dear :S |
10:07.27 | tzafrir_laptop | what's the output of dahdi_hardware #? |
10:07.55 | so_solid_moo | ah; I don't have dahdi installed - using zaptel |
10:08.32 | so_solid_moo | it's seeing a "e4e4:1160 Astribank-modular no-firmware" and the xpp_fxloader script is writing the firmware to the right device |
10:17.52 | tzafrir_laptop | so_solid_moo, do you have fxload installed? |
10:18.08 | so_solid_moo | tzafrir_laptop: yup |
10:18.42 | tzafrir_laptop | http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_lsusb_test |
10:18.54 | tzafrir_laptop | you installed it after you connected the astribank? |
10:19.22 | so_solid_moo | no, it was installed from the start |
10:19.35 | so_solid_moo | the usb id is as above - e4e4:1160 |
10:19.52 | so_solid_moo | I have fpga_load available too |
10:19.57 | tzafrir_laptop | What system is it? What linux distro? |
10:20.06 | tzafrir_laptop | fpga_load is not relevant to that |
10:20.17 | so_solid_moo | Oh, sorry - it's debian lenny |
10:20.40 | tzafrir_laptop | debian lenny packages? |
10:20.45 | so_solid_moo | yeah |
10:20.55 | tzafrir_laptop | (zaptel 1.4.12 ?) |
10:21.01 | tzafrir_laptop | two issues: |
10:21.03 | so_solid_moo | 1.4.11 even I think |
10:21.21 | tzafrir_laptop | 1. that version does not support those newer astribanks |
10:21.56 | tzafrir_laptop | 2. those packages don't include the firmware . the firmware is not DFSG and thus belongs in non-free (in a separate source package) |
10:22.21 | tzafrir_laptop | try the packages from: |
10:22.33 | ThoMe | tzafrir_laptop: hello. |
10:22.52 | tzafrir_laptop | ThoMe, hi |
10:23.13 | tzafrir_laptop | so_solid_moo, hmm.. I see I didn't yet add them to my main "unofficial" source |
10:23.19 | so_solid_moo | :) |
10:23.21 | ThoMe | tzafrir_laptop: one question: when i use the dial app and i have a call with a friend (connected) is it posible when i press example "8" then hangup by my friend and run a macro X ? |
10:23.28 | so_solid_moo | tzafrir_laptop: I'd figured out 2., but not 1. - my fault, sorry |
10:25.01 | tzafrir_laptop | try packages (zaptel , zaptel-firmware) from: deb http://updates.xorcom.com/pkg-voip/repo-i386-lenny/ unstable main |
10:28.04 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
10:28.56 | ThoMe | tzafrir_laptop: is it posible press a digit when i have a call |
10:29.00 | *** join/#asterisk Moz (n=me@81.179.238.144) |
10:29.16 | ThoMe | and when i press example the digit 8 then run a macro |
10:29.17 | ThoMe | ? |
10:29.56 | tzafrir_laptop | ThoMe, look into feature codes (feature.conf) |
10:30.17 | so_solid_moo | tzafrir_laptop: thanks, sadly no difference - I just get lots of '....' on the console :S |
10:31.16 | ThoMe | tzafrir_laptop: yes but is it posible to define a own code with a macro? |
10:31.26 | tzafrir_laptop | dpkg -l zaptel | grep ^i |
10:31.32 | tzafrir_laptop | sorry |
10:31.36 | tzafrir_laptop | dpkg -l zaptel\* | grep ^i |
10:32.24 | tzafrir_laptop | actually: |
10:32.32 | tzafrir_laptop | dpkg -l zaptel zaptel-firmware | grep ^i |
10:33.05 | ThoMe | how i can reload the features.conf? |
10:33.05 | so_solid_moo | 1:1.4.12.9.svn.r4649-0.7280 and 1:1.4.12.9.svn.r4649~dfsg-0.7280 |
10:33.29 | kaldemar | ThoMe: features reload |
10:33.40 | ThoMe | No such command 'features reload' (type 'help features reload' for other possible commands) |
10:33.45 | ThoMe | No such command 'feature reload' (type 'help feature reload' for other possible commands) |
10:34.42 | ThoMe | ah |
10:34.43 | ThoMe | Dynamic Feature Default Current |
10:34.43 | ThoMe | --------------- ------- ------- |
10:34.43 | ThoMe | testfeature no def #9 |
10:36.03 | ThoMe | have |
10:36.04 | ThoMe | testfeature => #9,self,Playback,tt-monkeys |
10:36.13 | ThoMe | but but no playback tt-monkes |
10:36.14 | ThoMe | hmm |
10:40.27 | ThoMe | Set(DYNAMIC_FEATURES=testfeature) |
10:40.27 | ThoMe | testfeature => #9,callee,Playback,tt-monkeys |
10:40.30 | ThoMe | is it correct? |
10:40.39 | ThoMe | aeh testfeature => #9,self,Playback,tt-monkeys |
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10:57.35 | ThoMe | how i can hangup the calling party but not my line |
10:58.03 | ThoMe | I would like jump to X (example meetmeroom) when i hangup the callinguser |
10:59.56 | nextime | is there a way to get the current asterisk version ( 1.4 or 1.6 ) from the manager interface? |
11:00.37 | troffasky | core show version |
11:00.51 | troffasky | oh, manager interface |
11:00.53 | troffasky | nm |
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11:02.20 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
11:02.57 | nextime | or from the dialplan, so from the manager i can use EXEC or GETVAR |
11:04.26 | ThoMe | emm |
11:05.59 | ThoMe | emm, have a problem: i would like when i call with user X and I press the "1" hangup and jump to a meetme-room |
11:06.04 | ThoMe | i try: http://paste.keks.be/2052/txt |
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11:07.38 | ThoMe | but when press "1" jump to my macro and hangup() then i cant then jump to my room |
11:08.12 | ThoMe | any ideas? |
11:08.30 | ThoMe | I would like only hangup the callee not my line |
11:08.35 | *** join/#asterisk m0bius (n=mobius@mailexchange.realize.gr) |
11:13.16 | kaldemar | nextime: manager command CoreSettings will show the version |
11:14.06 | kaldemar | nextime: func VERSION will return asterisk version in dialplan |
11:15.38 | ThoMe | when i use dial() and go to my cli |
11:15.40 | ThoMe | asterisk01*CLI> soft hangup SIP/ |
11:15.40 | ThoMe | SIP/gw_7_asterisk01-08885300 SIP/53002-08824e18 |
11:15.49 | ThoMe | how i can get this SIP/gw_7_asterisk01-08885300 ? |
11:16.01 | ThoMe | $CHANNEL is SIP/53002-08824e18 |
11:16.04 | ThoMe | AND SIP/gw_7_asterisk01-08885300 ? |
11:17.10 | kaldemar | get where? |
11:18.02 | ThoMe | where, yes |
11:18.17 | nextime | kaldemar : thanks |
11:18.53 | kaldemar | ThoMe: where are you trying to get it? |
11:19.10 | ThoMe | kaldemar: hm, i would like hangup only the callee channel |
11:19.10 | nextime | kaldemar : are they present also on 1.4? |
11:19.19 | ThoMe | is it posible? |
11:19.42 | kaldemar | ThoMe: core show application SoftHangup |
11:20.09 | ThoMe | kaldemar: jep, but when i Use SoftHangup how i can hangup the callee?? |
11:21.15 | kaldemar | ThoMe: SoftHangup takes the channel as a parameter, as you should understand from the application doc |
11:22.26 | kaldemar | get the channel name by some other means, for example using core show channels concise |
11:22.39 | ThoMe | kaldemar: yes but kaldemar i would like hangup the callee |
11:22.50 | ThoMe | when i use: core show channels |
11:22.53 | ThoMe | SIP/gw_7_asterisk01- (None) Up AppDial((Outgoing Line)) |
11:22.54 | ThoMe | SIP/53022-08824e18 s@macro-dial-gateway Up Dial(SIP/461@gw_7_asterisk01|1 |
11:23.02 | ThoMe | how i can get theis SIP/gw_7_asterisk01-.... value? |
11:24.28 | kaldemar | core show channels concise |
11:25.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
11:25.49 | ThoMe | kaldemar: jep, but how i can get this in my dialplan? |
11:26.32 | kaldemar | ThoMe: with a shell script for example |
11:27.14 | kaldemar | nextime: blasted, bot seem to be in 1.6 only |
11:27.18 | kaldemar | both |
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11:31.25 | nextime | kaldemar : ok, so if i don't find the manager command, i can suppose is <= 1.4 |
11:31.27 | nextime | right, thanks |
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11:32.54 | renzoe | guys. im trying to compile dahdi but my kernel is newer that what dahdi is looking for |
11:32.57 | kaldemar | ThoMe: something like Set(callee=${SHELL(asterisk -rx 'core show channels concise' | grep ^${CHANNEL} | awk -F\! '{ print $7 }')}) |
11:33.01 | renzoe | dahdi says "You do not appear to have the sources for the 2.6.18-92.el5 kernel installed." |
11:33.08 | kaldemar | ThoMe: might be a better way to do it |
11:34.05 | tzafrir_laptop | renzoe, you need linux-devel 2.6.18-92.el5 |
11:34.08 | renzoe | but i got: kernel-devel-2.6.18-164.el5 |
11:34.19 | renzoe | hmm ok will try to yum it |
11:34.43 | ThoMe | kaldemar: but when i have more as one call, but which one? |
11:34.44 | renzoe | are you sure its linux-devel? |
11:34.49 | renzoe | im usinf centos 5.2 |
11:34.59 | ThoMe | kaldemar: example |
11:35.00 | ThoMe | asterisk01:/etc/asterisk# asterisk -rx 'core show channels concise' | grep ^${CHANNEL} | awk -F\! '{ print $7 }' |
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11:35.04 | ThoMe | (Outgoing Line) |
11:35.06 | ThoMe | SIP/442@gw_7_asterisk01|180 |
11:35.09 | ThoMe | (Outgoing Line) |
11:35.11 | ThoMe | SIP/456@gw_7_asterisk01|180 |
11:35.19 | renzoe | there's no packaged named linux-devel |
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11:37.10 | renzoe | tzafrir_laptop, any more ideas? |
11:37.18 | kaldemar | ThoMe: that's what the grep is for. take the line that you need to get. |
11:38.09 | renzoe | i cannot move on to asterisk until i install dahdi |
11:39.19 | tzafrir_laptop | renzoe, the package you need is http://vault.centos.org/5.2/os/i386/CentOS/kernel-devel-2.6.18-92.el5.i686.rpm |
11:40.08 | renzoe | but i already have a newer kernel-devel |
11:40.31 | kaldemar | do you have a newer kernel in use also? |
11:40.41 | renzoe | if i need to downgrade, how will i do that? |
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11:41.13 | renzoe | i have this one installed: kernel-devel-2.6.18-164.el5 |
11:41.51 | renzoe | i am using 64 bit version |
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11:41.57 | renzoe | of centos 5.2 |
11:42.52 | renzoe | if i dont install this. will zaptel still work? |
11:43.13 | kaldemar | you already got a link to the right package |
11:44.05 | kaldemar | dahdi is the new name for zaptel. |
11:44.19 | renzoe | so i dont need to install dahdi? |
11:44.42 | renzoe | i am trying to follow this tutorial: http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation |
11:45.05 | kaldemar | zaptel doesn't work with 1.6. you need dahdi for it. |
11:45.30 | renzoe | its just dahdi is getting my way |
11:45.38 | kaldemar | if you want to use telephony hardware and a dahdi clock, then you need it. if not, then don't install it. |
11:45.54 | renzoe | so how can i install dahdi on the newer kernel? |
11:46.11 | kaldemar | dahdi clock is needed for example in meetme conferences and iax2 trunking. |
11:46.47 | kaldemar | does uname -r match the header package version you have? |
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11:46.54 | renzoe | lemme check |
11:47.12 | renzoe | no |
11:47.23 | renzoe | uname -r says its 92.e15 |
11:47.30 | renzoe | how come |
11:47.46 | renzoe | should i uninstall the kernel devel? |
11:47.52 | renzoe | then try to reinstall it? |
11:47.53 | kaldemar | then your header package version is wrong. install the package tzafrir linked to you and proceed with the compiling |
11:48.16 | renzoe | but can i uninstall it first? |
11:48.31 | kaldemar | sure |
11:49.37 | renzoe | thanks kaldemar |
11:49.50 | ThoMe | ah, kaldemar this is the variable ${DIALEDPEERNAME} |
11:50.34 | kaldemar | ThoMe: is it the whole channel name or just the peer name? |
11:51.05 | ThoMe | the value is: SIP/gw_7_asterisk01-0882dd40 |
11:52.05 | ThoMe | but when i use SoftHangup(${DIALEDPEERNAME}|a); |
11:52.08 | ThoMe | the callee is not hanging up |
11:52.12 | ThoMe | hmm :-( |
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11:58.10 | renzoe | sigh |
11:58.22 | renzoe | i have been installing i386 version |
11:58.27 | renzoe | i have 64bit |
12:00.43 | renzoe | finally.. thanks kaldemar and tzafrir_laptop |
12:00.59 | renzoe | its doing make all now |
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12:22.49 | Dovid | anyone know what would cause this ? |
12:22.49 | Dovid | http://pastebin.com/m2672c754 |
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12:27.31 | scalex000 | good morning |
12:27.52 | cjk | hi, is there a way to send RTP before the call is answered? |
12:28.20 | cjk | i mean to the called party |
12:32.00 | *** join/#asterisk denon (i=denon@sassinak.net) |
12:32.00 | *** mode/#asterisk [+o denon] by ChanServ |
12:32.56 | m0bius | cjk: yes |
12:33.19 | cjk | m0bius, not to the calling party, but to the destination |
12:33.21 | m0bius | cjk: you might need to set Progress() before doing Playback but make sure you set Playback to play even on un-answered channels |
12:34.12 | m0bius | hum, but how will the other party hear the RTP audio if he hasn't pickup the phone? |
12:34.15 | Naikrovek | wonder if you could send a secondary data stream to another voip user. ooh |
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12:34.35 | cjk | m0bius, its not important if he doesnt hear it |
12:34.46 | cjk | but i need to bypass nat problem in one special scenario |
12:34.47 | Naikrovek | some sort of collaborative notetaking via RTP during a call... |
12:35.29 | m0bius | hum |
12:35.48 | cjk | the problem is this |
12:35.49 | Naikrovek | or even just for file transfers, for things like business cards and the like |
12:36.00 | m0bius | if the problem is nat, why don't you enable qualify to force an open connection to the router? |
12:36.04 | cjk | call comes from asterisk1 to asterisk2 and asterisk2 sends it back to asterisk1 |
12:36.12 | cjk | between ast1 and ast2 is a NAT |
12:36.24 | cjk | if i do not answer on ast2 to create some RTP, then i will not have voice |
12:36.31 | cjk | this scenario happens for call redirects |
12:37.52 | m0bius | hmmm |
12:37.58 | m0bius | that doesn't sound logical |
12:38.04 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
12:38.17 | m0bius | if nat is your problem you should focus on solving the nat |
12:38.24 | cjk | its not the NAT |
12:38.31 | cjk | i tested with all nat configuraitons |
12:38.42 | cjk | its the fact that the one that answeres the call is outside the nat |
12:39.11 | cjk | if i answer the call before redirecting wiht playback(beep) then it works |
12:39.19 | cjk | so its not 100% a nat issue |
12:39.48 | troffasky | I would take some packet captures if I were you |
12:41.34 | m0bius | I've tested multiple setups where asterisk has been in or out of nat but I never had issues |
12:41.41 | cjk | i didn the problem sounds no easy to understand, difficult to explain and more difficult to solve |
12:41.53 | m0bius | so I guess taking a capture could shed some light |
12:42.17 | troffasky | cjk, maybe a diagram would help? |
12:42.23 | cjk | yes |
12:42.25 | troffasky | picture being worth 1000 words and all that |
12:42.25 | cjk | i create one |
12:45.47 | *** part/#asterisk so_solid_moo (n=nmoo@fsf/member/alexhudson) |
12:46.10 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
12:46.32 | jaytee | wow! my old AOL account back in the 90's was mobius812 |
12:48.06 | m0bius | really? |
12:48.31 | m0bius | I've been using since the 90s as well :) |
12:48.35 | cjk | here is a quick schema: http://www.wanter-trail.org/natredirect.pdf |
12:48.57 | cjk | i showed 4 scenarios out of which 3 works, just to show that i do not have a general nat problem |
12:49.19 | m0bius | do you have reinvite on on asterisk 2? |
12:49.24 | cjk | no of course not |
12:49.52 | cjk | the problem is that in the last scenario no RTP is comming from the inside of the nat |
12:50.13 | cjk | and as such, it doesnt accept RTP from the outside, and so the conversation never takes place |
12:52.18 | kaldemar | cjk: what application are you using for the "redirect"? a Dial? |
12:52.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:52.26 | cjk | kaldemar, yes |
12:52.35 | cjk | just normal dialplan logic |
12:53.03 | kaldemar | you could use app transfer to send a SIP 302 to the originating asterisk |
12:53.05 | *** part/#asterisk ice_croft (n=nolan@mail.kubkurort.ru) |
12:53.28 | cjk | kaldemar, not an option, it should look like a normal call |
12:53.49 | kaldemar | so you want the signaling to go through both servers? |
12:53.57 | cjk | yes |
12:54.18 | cjk | kaldemar, most of the time the problem does not exist because incoming calls come from isdn or pri on asterisk2 so there is no problem |
12:54.23 | kaldemar | then disable re-invites or answer the leg. |
12:54.38 | cjk | kaldemar, reinvites are disabled |
12:54.44 | cjk | answereing has billing issues |
12:55.30 | cjk | is there a solution to do something like playback beep on answer? |
12:55.41 | m0bius | you can play an audio on answer |
12:55.47 | m0bius | on the calling party |
12:55.53 | m0bius | and the called party |
12:56.04 | m0bius | but I am not sure that would solve your problem |
12:56.50 | *** join/#asterisk jsmith (n=jsmith@asterisk/training-and-documentation-guru/jsmith) |
12:56.50 | *** mode/#asterisk [+o jsmith] by ChanServ |
12:56.56 | cjk | if asterisk2 plays it i think it will, let me check |
12:56.59 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
12:57.09 | *** topic/#asterisk by jsmith -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev, AstriCon Dis |
12:57.11 | ThoMe | hmm, i try: |
12:57.17 | ThoMe | [Sep 23 14:55:33] WARNING[27056]: pbx.c:2437 __ast_pbx_run: Channel 'SIP/53002-b6701a08' sent into invalid extension 's' in context 'from-internal-users', but no invalid handler |
12:57.26 | ThoMe | <PROTECTED> |
12:57.29 | ThoMe | is it wrong? |
12:57.51 | *** topic/#asterisk by jsmith -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs AstriCon Discount code: ac09digi |
12:58.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
12:58.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
12:58.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
12:59.36 | *** join/#asterisk voipmonk (n=voipmonk@69.172.93.45) |
13:00.11 | *** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net) |
13:00.18 | Gumug | howdy |
13:00.42 | Gumug | i need some advice |
13:00.43 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
13:02.02 | Naikrovek | ask your question |
13:02.04 | jaytee | jsmith, hi!!! |
13:02.22 | Gumug | alrighty |
13:02.40 | Katty | mew. |
13:02.47 | *** join/#asterisk engrxyz (n=zcvzxcvx@host81-143-50-89.in-addr.btopenworld.com) |
13:02.55 | jaytee | mornin Katty |
13:03.02 | Katty | hi |
13:03.03 | jsmith | jaytee: Long time, no chat... what's up? |
13:03.38 | jaytee | jsmith, not too much. getting ready to move my installation to 1.6.0.15 |
13:03.56 | jaytee | jsmith, how's things in your neck of the woods? |
13:04.49 | Gumug | i am starting work this week at a new business. We have 9 offices located in missouri and arkansas. We have 16 phone lines. Each line costs $60ish a piece to AT&T. I want to see if i can lower the costs of communication, while also implementing ACD, voice prompts, etc. |
13:05.10 | jaytee | sends jsmith a mango smoothee using app_virtual_beverage.so |
13:05.13 | Gumug | all of our business is over the phone |
13:05.13 | jsmith | Gumug: Then you're in the right place :-) |
13:05.15 | *** join/#asterisk lanning (n=lanning@212.183.136.194) |
13:05.19 | Katty | Gumug: Missouri, huh? |
13:05.22 | jsmith | jaytee: Thanks! |
13:05.24 | Gumug | yep |
13:05.37 | jsmith | jaytee: Busy here (as always) with the Asterisk training... and I'm getting geared up for AstriCon |
13:05.45 | Katty | small world. |
13:05.52 | Gumug | southwest missouri infact |
13:05.56 | jaytee | jsmith, you in Huntsville or teaching a class back east? |
13:06.09 | Katty | Gumug: i'm over in cape. |
13:06.12 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
13:06.13 | jsmith | jaytee: I'm at home until Friday, and then I fly to San Diego |
13:06.16 | Gumug | right on |
13:06.34 | jaytee | jsmith, I'm so jealous!! San Diego is a great city |
13:06.35 | Gumug | i've been looking at these hosted pbx's |
13:06.51 | Gumug | some cost a ton of money each month, for the amount of talk time we are going to have |
13:06.57 | [TK]D-Fender | Gumug: Is that 16 lines spread around? |
13:07.05 | Gumug | yes |
13:07.17 | Gumug | some offices have 2 |
13:07.19 | Gumug | some have 3 |
13:07.21 | [TK]D-Fender | Gumug: AKA each site having a line or two |
13:07.25 | Gumug | yes |
13:07.44 | [TK]D-Fender | Gumug: S2 or 3? at 16 for 9 offices, sounds like some have NONE :) |
13:08.19 | Gumug | you know |
13:08.24 | Gumug | i think we have a lot more |
13:08.29 | Gumug | some have 2 actual phones |
13:08.33 | Gumug | some have three phones |
13:08.34 | Gumug | BUT |
13:08.39 | Gumug | they are 4 line phones |
13:08.49 | Gumug | so we may have a lot more phone lines then i think |
13:08.58 | Gumug | if they are able to keep 3 or 4 calls going at once |
13:09.22 | Gumug | all i know is right now its expensive with the costs of having each line per store |
13:09.36 | Katty | store? |
13:09.39 | Katty | is this a retail chain? |
13:09.46 | Gumug | insurance broker |
13:09.51 | Katty | bummer :< |
13:09.56 | Gumug | people call in, or walk in for quotes/policies |
13:09.58 | Katty | i was excited about going shopping there for a minute. |
13:10.02 | Naikrovek | bummer? ah |
13:10.02 | Gumug | lol |
13:10.14 | Gumug | sell you some SR22 insurance |
13:10.35 | [TK]D-Fender | Gumug: Each site have broadband? |
13:10.41 | Gumug | yes |
13:10.53 | Katty | i've not been in any major accidents. |
13:11.02 | [TK]D-Fender | Gumug: then you can consolidate to a central server |
13:11.03 | Naikrovek | this is shaping up almost to be a case study for asterisk deployment |
13:11.04 | Gumug | i'm glad |
13:11.30 | Katty | any chance your locations are connected through a VPN? |
13:11.42 | Naikrovek | oh that would make it even easier |
13:12.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:12.47 | Gumug | i have no idea |
13:12.55 | Gumug | i doubt it |
13:13.00 | Gumug | i have an IT background |
13:13.10 | Gumug | i know i could figure it out if VPN were necessary |
13:13.17 | Naikrovek | that's something else you'd want to implement probably |
13:13.27 | Katty | it has lots of benefits. |
13:13.32 | Gumug | such as? |
13:13.49 | Katty | phone systems can act quirky with routing when you're going through a firewall. |
13:13.58 | Gumug | ok |
13:13.59 | Gumug | duly noted |
13:14.04 | Katty | plus, being in IT, you know all the benefits of having a VPN connection |
13:14.14 | Katty | security, data access, etc |
13:14.15 | Gumug | security |
13:14.16 | Gumug | etc |
13:14.27 | Gumug | afk sec |
13:14.27 | Katty | not to mention you can use VNC to help people |
13:14.34 | Katty | rather than /driving/ there |
13:14.49 | Naikrovek | it'll make your asterisk deployment much easier. secure email exchange between offices, secure file transfer between offices, centralized file server(s) and email server, so on and so forth, etc, etc, |
13:15.04 | Katty | dude, i can't function this morning |
13:15.11 | Naikrovek | yeah you can dude |
13:15.54 | Katty | i feel like i'm going to fall asleep on my keyboard i'm so tired. |
13:16.42 | Naikrovek | do what i do: "ugh i don't feel good, I need to go to the bathroom" then go take a nap |
13:17.05 | Naikrovek | that's what i used to do anyway, before i got diagnosed with narcolepsy |
13:17.24 | *** part/#asterisk jsmith (n=jsmith@asterisk/training-and-documentation-guru/jsmith) |
13:17.56 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
13:17.59 | Katty | i am NOT going into the bathroom here. it is /gross/ |
13:18.23 | Naikrovek | even better "i don't feel good, i'm going home to use the bathroom" nap |
13:18.23 | *** join/#asterisk troffasky (n=r00t@92-234-126-57.cable.ubr08.gate.blueyonder.co.uk) |
13:18.44 | Naikrovek | 30 mins is pure dreamy wonderment |
13:18.46 | Gumug | right now email is handled by google apps |
13:18.54 | Gumug | as well as docs |
13:18.57 | Katty | 30minutes isn't gonna do anything for me. |
13:19.04 | Naikrovek | Gumug: ah |
13:19.07 | Katty | need another 4 hours |
13:19.56 | Gumug | right now, each store has its own phone number, which is advertised in the local yellow pages, and radio spots |
13:19.58 | scalex000 | Hello |
13:20.18 | Gumug | we are thinking about going to 1 or 2 numbers for all the stores though |
13:20.22 | scalex000 | I need to interconnect BCm and asterisk using h323 or sip any ideas? |
13:20.39 | [TK]D-Fender | Katty: Do't need a VPN for VNC... |
13:21.14 | Naikrovek | well if they already have a decentralized infrastructure then a vpn probably won't do much for them |
13:21.26 | *** join/#asterisk spck (n=spck@unioncab.com) |
13:22.22 | Gumug | i'm looking at these hosed pbx's, but the cost of minutes is out of this world |
13:22.29 | Naikrovek | don't do hosten |
13:22.31 | Naikrovek | hosted |
13:22.36 | [TK]D-Fender | scalex000: Idea : throw the BCM into a dumpster and BURN IT |
13:22.47 | Gumug | Naikrovek: really? |
13:23.36 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
13:23.50 | Naikrovek | you can set up your own server in the biggest or most centralized office, and put phones in each of the other offices, and consolidate those 16 lines into a single box and probably save some money. i pay $32 per "line" per month using a voip provider |
13:24.10 | Naikrovek | unlimited incoming, unlimited free calling to US & canada as well |
13:25.22 | Gumug | save money over time right? |
13:25.36 | Naikrovek | yes, definitely |
13:25.38 | Gumug | i priced it out and some were going to cost over $500 a month |
13:25.49 | Gumug | because we are on the phone over 2hrs a day per store |
13:26.11 | Gumug | 2000 minute plans will be eroded in a day |
13:26.11 | Gumug | or 2 |
13:26.11 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
13:26.14 | Naikrovek | i have asterisk where i work. one office is in illinois, the other is in India. one asterisk server |
13:26.25 | Naikrovek | 60-80 or so extensions |
13:26.32 | scalex000 | Tk: if I was the owner I will do it but its not my. |
13:26.59 | Katty | scalex000: i think your best bet in this situation would be to hire a consultant. |
13:27.08 | [TK]D-Fender | Gumug: Where are those 2 hours talking to? |
13:27.25 | Gumug | so, would you recommend that we consolidate our #'s to 1 phone #, or would it still be possible to have numbers rerouted |
13:27.31 | troffasky | if it's site2site and you've already got internet in each site... |
13:27.44 | Gumug | [TK]D-Fender: the vast majority of our business is incoming calls |
13:27.51 | Naikrovek | Gumug: each phone could have its own phone number, or its own extension number on a single phone number, or both |
13:28.05 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
13:28.14 | Gumug | fascinating |
13:28.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
13:28.43 | Naikrovek | Gumug: you're about to enter the rabbit hole buddy :) |
13:28.44 | Gumug | people call in to get quotes on insurance, change policies, purchase policies, etc |
13:28.56 | Gumug | Naikrovek: the thing is, my role is really HR |
13:29.06 | Orbixx | How can I record a call that takes place after a client has queued? |
13:29.09 | Naikrovek | i'm overusing this word, but asterisk is pure telephonic wonderment |
13:29.15 | Gumug | lol |
13:29.26 | Gumug | is it set it and forget it easy? |
13:29.33 | Orbixx | it's more like |
13:29.35 | Gumug | i don't have the time to be a sys admin |
13:29.55 | Orbixx | set it, set it again, it's still not perfect, set it again, set it some more, then forget it for a long time |
13:30.46 | Gumug | yes, like anything |
13:30.52 | Orbixx | quite |
13:31.26 | creativx | i did a looot of setting |
13:31.29 | creativx | i even set too much |
13:31.31 | creativx | then i removed some setting |
13:31.38 | creativx | then i forgot about it and its been what.. 2 years |
13:32.01 | Gumug | do you recommend colocating an asterisk server |
13:32.01 | Katty | falls asleep on desk |
13:32.20 | Gumug | to a more secure area with regulated power, clean connection |
13:33.07 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
13:34.19 | troffasky | if you don't already have a site with decent connectivity then colo may be a good idea |
13:34.55 | Gumug | ya |
13:35.02 | Gumug | see, if the phones go down |
13:35.06 | Gumug | we are screwed |
13:36.04 | creativx | you dont have.. email? |
13:36.07 | creativx | ;) |
13:36.11 | creativx | thats what you have failover for |
13:36.14 | Gumug | lol |
13:36.14 | creativx | and redundancy |
13:36.45 | Katty | we use mobile phones for emergency events |
13:36.52 | Katty | we just have the telco forward stuff |
13:37.13 | Katty | someone please invent a drink with caffeine and melatonin |
13:37.31 | creativx | so melanotan and caffeine |
13:37.41 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
13:38.05 | Katty | melatonin effects the pineal gland. |
13:38.17 | Katty | well. not exactly. |
13:38.19 | Naikrovek | caffetonin |
13:38.32 | Katty | usualy the pineal gland secrets melatonin |
13:38.36 | creativx | Katty: do you know what melanotan is though :) |
13:38.49 | Katty | it's a bloody awesome hormone |
13:38.55 | *** join/#asterisk denon (i=denon@sassinak.net) |
13:38.55 | *** mode/#asterisk [+o denon] by ChanServ |
13:38.58 | creativx | hehe |
13:39.02 | creativx | imagine drinking it |
13:39.02 | creativx | :) |
13:39.02 | Naikrovek | Gumug: you could put the asterisk server on a UPS, so it could stay up through a couple hour power outage. |
13:39.25 | [TK]D-Fender | Naikrovek: Hours for a server? More like a generator |
13:39.27 | creativx | Naikrovek: ever heard the term mission critical |
13:39.30 | creativx | and yes |
13:39.33 | Katty | http://blogs.babycenter.com/momformation/files/2008/08/melatonin_tablet.jpg <- i guess i could crush it up and put it in my soda. |
13:39.33 | creativx | that would be one big ass ups |
13:39.49 | Naikrovek | though in that case you're likely to need to use a PoE switch, which you would then also put on the UPS to power the phones in that office |
13:39.59 | Naikrovek | well $3k can get you a natural gas powered generator |
13:40.15 | Naikrovek | one would only need battery for the 5 seconds it takes for the generator to auto-start |
13:40.18 | creativx | and what about the switching |
13:40.19 | troffasky | so, er, yeah, just get it colod |
13:40.24 | creativx | in other words yes |
13:40.25 | creativx | colo |
13:40.25 | Naikrovek | creativx: included |
13:40.27 | creativx | or |
13:40.30 | creativx | build yourself a server room |
13:40.40 | Katty | yeah don't put it in a closet :< |
13:40.44 | Katty | or the bathroom. |
13:40.44 | creativx | from scratch.. with cooling.. batteries.. and redundancy and access control and and and.. |
13:40.47 | Katty | which i /have/ seen. |
13:40.56 | Naikrovek | yeah i've seen that also |
13:40.59 | creativx | i once called an asterisk number to flush a toilet on a webcam |
13:41.00 | Katty | and MELATONIN!? |
13:41.05 | creativx | curiosity |
13:41.06 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
13:41.11 | Naikrovek | vent holes covered by saran wrap to protect them from splashing |
13:41.35 | scalex000 | this warning what its means "channel.c:3606 ast_channel_make_compatible_helper: No path to translate from H323/test-11(256) to SIP/246-b7812d48(4)" |
13:42.52 | Katty | it means it doesn't have enough melatonin |
13:44.13 | *** join/#asterisk Methose (n=Methose@38.101.237.250) |
13:44.21 | creativx | hehehe |
13:44.29 | creativx | melatonintan |
13:45.24 | [TK]D-Fender | scalex000: Means "pick codecs that * can translate. You allowed either G.729 or G.723 without licenses |
13:46.11 | beek | mornin' [TK]D-Fender |
13:46.22 | beek | waves to Katty |
13:47.50 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
13:47.53 | phix | waves to Katty and beek! |
13:48.47 | *** join/#asterisk Yuda-israel1984 (n=yo1984@62.219.144.190) |
13:49.04 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
13:49.22 | *** join/#asterisk denon (i=denon@sassinak.net) |
13:49.22 | *** mode/#asterisk [+o denon] by ChanServ |
13:49.23 | [TK]D-Fender | beek: Mornin' |
13:49.37 | Yuda-israel1984 | hi guys so this is my first time on here and i have been looking for ways to learn asteirks scripting and im finally doing it although im having trouble with something and i was wondering if anyone can help me |
13:49.40 | Katty | hugs beek |
13:49.42 | Katty | phix: hello. |
13:50.02 | Yuda-israel1984 | is anyone familiar with MyPhoneCompany ? |
13:50.02 | phix | :D |
13:50.08 | phix | nn |
13:50.58 | Yuda-israel1984 | anyone familiar then with sip messages? |
13:51.13 | Katty | yawns. |
13:51.17 | Katty | okay, gotta wake up for real. |
13:51.45 | Yuda-israel1984 | katty u a scripter? |
13:51.52 | Katty | Yuda-israel1984: no. |
13:52.23 | Katty | Yuda-israel1984: just about everyone in here is a volunteer. if no one has an answer, i would suggest being patient and asking again in a little bit. otherwise, if its high priority you might consider finding a consultant to assist you. |
13:53.10 | creativx | sound advice from Katty there |
13:53.12 | Yuda-israel1984 | im new here so i dont really know how this IRC works last time i used it was a good 10 years ago at least when i was a child |
13:54.00 | Yuda-israel1984 | i will ask later i appreciate it |
13:54.01 | Katty | kk |
13:55.28 | kaldemar | Yuda-israel1984: the best way to get an answer is to ask a specific question |
13:55.57 | *** join/#asterisk errr (n=errr@fedora/errr) |
13:55.57 | Katty | specific question: which has more caffeine? a cup of coffee or a cup of mt dew. |
13:56.03 | Katty | hi errr |
13:56.11 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
13:56.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:56.21 | Katty | hugs putnopvut |
13:56.34 | Naikrovek | ~book |
13:56.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
13:56.35 | putnopvut | oh snap |
13:56.38 | Naikrovek | Yuda-israel1984: ^^^^^^^ |
13:56.43 | putnopvut | hugs Katty back |
13:56.57 | *** join/#asterisk l2trace99 (n=jr@75.112.140.2) |
13:57.32 | *** join/#asterisk errr (n=errr@fedora/errr) |
13:58.44 | Katty | a cup of coffee (not instant) contains 90-150mg of caffeine. mew dt has 36mg per cup. |
13:58.52 | Katty | mew dt? |
13:59.03 | Katty | dang. i really AM that tired. |
13:59.04 | troffasky | is anybody running SIP over OpenVPN terminated on their * server? |
13:59.11 | Katty | troffasky: i'm not. |
13:59.20 | Naikrovek | troffasky: i'm sure someone is |
13:59.23 | Naikrovek | but i'm not |
14:00.12 | Katty | red bull only has 80mg per 8oz |
14:00.49 | Katty | that's insane. |
14:01.03 | Katty | woulda figured redbull has more caffeine than coffee. |
14:01.27 | creativx | redbull has lots of sugar |
14:01.57 | Methose | hense the crash after 10 mins |
14:02.01 | Katty | sugar doesn't do much for me. |
14:02.41 | Katty | sugar is horribly bad for you. |
14:02.55 | Katty | so is this diet mt dew i'm drinking. |
14:03.01 | Katty | it will make me hungry in about 15 minutes. |
14:03.43 | troffasky | I wonder if 20ms RTP packets are fattening |
14:03.54 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net) |
14:04.15 | Katty | troffasky: rtp packets are not fattening. |
14:04.18 | Katty | hi carlos. |
14:04.40 | creativx | Katty: diet is bad |
14:04.45 | creativx | cause of the artificial sugar |
14:04.49 | Katty | yes. |
14:04.51 | Katty | it's horribly bad. |
14:04.52 | creativx | it induces hunger |
14:04.56 | Katty | yes, it does. |
14:04.59 | creativx | plus you probably get cancer but thats not so important |
14:05.03 | creativx | :p |
14:05.09 | creativx | aspartam |
14:05.18 | troffasky | if you live in the west, chances are you'll die of cancer before anything else |
14:05.19 | Katty | it also contains phenylalanine |
14:05.41 | leifmadsen | I just don't drink pop |
14:05.56 | Naikrovek | i'm more or less addicted to caffiene free diet coke. /shame |
14:05.57 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
14:06.12 | leifmadsen | I used to drink way too much of it -- I had to quit the addiction :) |
14:06.16 | troffasky | could you explain the point of caffeine free coke? |
14:06.18 | leifmadsen | I just drink green tea now |
14:06.22 | Naikrovek | troffasky: no idea |
14:06.23 | troffasky | do you just want brown teeth? |
14:06.29 | Katty | it is in a can. |
14:06.32 | Naikrovek | tastes good? |
14:06.32 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
14:06.32 | Katty | easy accessible. |
14:06.35 | Naikrovek | plastic bottle |
14:06.41 | Katty | does not taste of funky city water. |
14:07.46 | Naikrovek | i used to drink 24 cans of pepsi (not diet) per day |
14:07.55 | Naikrovek | that's why i drink diet now. |
14:08.04 | Naikrovek | i also have narcolepsy, so caffiene is a no-no |
14:08.11 | Naikrovek | that's why i drink caffiene free |
14:08.15 | creativx | ahha |
14:08.18 | creativx | narcolepsy sounds fun |
14:08.25 | Naikrovek | yeah i about died while driving a few times |
14:08.36 | Chainsaw | creativx: It's fun until it happens to you. |
14:08.37 | l2trace99 | any one have issues with all calls within a queue dropping ? |
14:08.39 | Katty | i'd suggest not doing that anymore. |
14:08.45 | Naikrovek | wake up donig 75mph in the grass between lanes on a 4 lane highway |
14:08.50 | Naikrovek | that's a sobering experience |
14:09.04 | Katty | ah, sobering experiences. |
14:09.07 | Katty | i've had a few of those. |
14:11.55 | Katty | did i mention my dad's on facebook? |
14:11.59 | Katty | this is weird. |
14:12.21 | creativx | ouch Naikrovek |
14:12.42 | troffasky | istr you mentioning it yesterday Katty |
14:12.51 | troffasky | it wasn't particularly interesting then either |
14:13.26 | Katty | well perk up buttercup, or life is gonna be awfully boring for you! |
14:13.53 | Katty | we don't do cranky negativism here! |
14:13.57 | creativx | hehe |
14:14.00 | creativx | buttercup |
14:15.48 | Katty | my parents were such a cute couple. |
14:15.57 | Katty | http://farm4.static.flickr.com/3158/3023014977_5f88fd7035.jpg <- mom. |
14:16.09 | Katty | http://farm4.static.flickr.com/3066/3023839552_7be80cd1b5.jpg <- dad. |
14:16.16 | creativx | im not even sure why i clicked on a link with a picture of your mom |
14:16.26 | creativx | give me a link and i will click it |
14:16.34 | Naikrovek | creativx: lol same here |
14:17.20 | Katty | maybe it gives you some indication of how i look. |
14:17.20 | Katty | maybe you're just bored. |
14:17.23 | Katty | who knows! it's something intersting to look at. |
14:17.24 | Naikrovek | you posted a pic of yourself once |
14:17.27 | Naikrovek | with your dog or something |
14:17.39 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
14:17.39 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
14:17.53 | Katty | hi Deeewayne (= |
14:18.11 | Deeewayne | Katty, good morning :-) |
14:18.35 | creativx | i should find a really retarded picture link and see how many clicks it |
14:18.36 | Katty | creativx: you can't look at that picture and not smile. |
14:19.31 | spck | anyone got a recipe for answering machine detection, i.e. what delays they used? |
14:20.17 | Naikrovek | spck: i always thought detection was listening for how long a person talks when they answered |
14:21.11 | SuPrSluG | works ok if you can live with 85% accuracy |
14:22.30 | spck | doing callouts for a taxi company, not mission critical |
14:23.24 | spck | naikrovek: the amd() application has a bunch of options for delay timings, i was wondering if anyone had a set that they had a good experience with |
14:23.36 | Naikrovek | ah |
14:23.38 | Naikrovek | i haven't |
14:25.13 | SuPrSluG | these worked ok for me http://pastebin.com/m264227d2 |
14:28.58 | spck | thanks for the pasty |
14:29.20 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
14:29.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
14:29.43 | Katty | hi tony |
14:29.58 | anthm | hi |
14:30.16 | Katty | how're you dear |
14:31.50 | guax | im fine |
14:32.11 | Katty | excellent. |
14:32.22 | crazybyte | Hello to all! How can I modify a Digium Wildcard TE110P T1/E1 card sync source settings? I tried in /etc/zaptel.conf but doesn't seem to work. Thank you! |
14:32.30 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
14:32.55 | *** join/#asterisk MWE (n=michel@nl06sr01.targetmedia.nl) |
14:33.02 | MWE | hi there |
14:33.47 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
14:33.47 | MWE | hopefully some could me out... |
14:33.47 | [TK]D-Fender | CraIt does. You also have to stop * and re-init zaptel for that change to take effect |
14:33.47 | MWE | help* |
14:33.48 | mbrevda | does dahdi fax detect require answer? |
14:33.56 | [TK]D-Fender | crazybyte: And I don't see your before & after configs |
14:34.07 | [TK]D-Fender | mbrOf course it does |
14:34.16 | [TK]D-Fender | mbrevda: Of course it does |
14:34.22 | [TK]D-Fender | Gah... Autocomplete fail day |
14:34.33 | crazybyte | [TK]D-Fender, one sec (sorry I forgot about them) |
14:34.37 | Katty | [TK]D-Fender: you've been doing that a lot lately. |
14:34.42 | Katty | [TK]D-Fender: not just today. |
14:34.48 | mbrevda | [TK]D-Fender: hey - how've u been? thanks! |
14:34.53 | Katty | [TK]D-Fender: but we still love you. |
14:35.02 | mbrevda | how about the new sip fax detect? |
14:35.35 | mbrevda | couldnt find much documentation of using it |
14:35.52 | MWE | i put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put this in |
14:35.52 | MWE | <PROTECTED> |
14:36.18 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
14:36.36 | MWE | is that with originate? |
14:36.44 | *** join/#asterisk lordmortis (n=lordmort@203.59.207.20) |
14:38.47 | *** join/#asterisk Gumug (n=Gumug@98.98.181.70) |
14:38.52 | Gumug | so... |
14:38.57 | Gumug | looking at switchvox |
14:38.59 | Gumug | pretty expensive |
14:39.05 | Gumug | anyone ever used it? |
14:39.21 | spck | overpriced and i believe they have a yearly subscription requirement |
14:39.37 | crazybyte | [TK]D-Fender, here is my zaptel.conf http://pastie.org/627460 and my zapata.conf http://pastie.org/627463 |
14:39.40 | troffasky | have you considered asterisk instead? ;-) |
14:40.02 | Gumug | well, i liked the aspect of it being tied into salesforce, et all |
14:40.18 | CareBear\ | D-Fender : I was obviously missing auth= for my [peer] |
14:40.56 | crazybyte | [TK]D-Fender, also I have the following error (in asterisk console) chan_zap.c:2498 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
14:41.08 | spck | personally i didn't see any upside to their system other then it being turnkey |
14:41.36 | Gumug | sure, i guess thats the nice thing, that it's turn key |
14:41.49 | spck | but even then that's a downside |
14:41.56 | Katty | i'd like to know who invented the phrase Turn Key |
14:42.08 | spck | you have to buy the silver support contract to continue using it |
14:42.10 | Gumug | he suggested i got a SMB AA60 Appliance for each office, which would be 9 installs |
14:42.24 | spck | depends on your budget i guess |
14:42.29 | Gumug | $3,890.00 |
14:42.30 | Gumug | each |
14:42.37 | troffasky | boggles |
14:42.46 | Gumug | would asterisk be less? |
14:42.49 | spck | i spent $10k on two dell poweredge 2950 |
14:42.56 | Katty | Gumug: i'm sure someone in here would be happy to get a quote together for you. |
14:42.58 | spck | (total overkill for my install) |
14:43.06 | Naikrovek | ???? compiled and installed asterisk 1.6.1 - where did 'core show translation' go |
14:43.16 | spck | asterisk is free |
14:43.18 | Gumug | Katty: oh i'm sure |
14:43.23 | Gumug | so, he suggested i peer the systems |
14:43.23 | spck | no strings attached other then the license |
14:43.31 | Katty | i think we spent about 2k on our server. |
14:43.36 | Katty | probably another 1.5k on the pri card. |
14:43.38 | Gumug | so that if one goes down, than, the system doesn't go down as a whole |
14:43.40 | Katty | and around 100 per phone |
14:43.50 | mbrevda | grandstreams?! |
14:43.55 | Katty | polcyom 330s |
14:44.00 | mbrevda | oh, right |
14:44.03 | mbrevda | cool |
14:44.04 | Naikrovek | polycom ftw |
14:44.06 | Gumug | so, can you peer astrisk like that? |
14:44.07 | Katty | mhmm |
14:44.09 | mbrevda | +1 |
14:44.09 | spck | gumug: you can do all that with asterisk |
14:44.14 | Gumug | ok |
14:44.16 | spck | you just have to setup it yourself |
14:44.26 | Gumug | is it rocket surgery? |
14:44.27 | spck | switchvox is basically a supported preinstalled asterisk server |
14:44.36 | Katty | Gumug: it can feel like it at times. |
14:44.39 | spck | lol, no but it can be frustrating |
14:44.40 | Gumug | lol |
14:44.40 | Naikrovek | just gotta set up trunks between the systems. Gumug nope, it's easy to trunk the systems |
14:44.50 | Katty | Gumug: there are other asterisk based solutions. like Fonality |
14:44.51 | CareBear\ | Gumug : I started asterisk for the first time about a month ago |
14:44.53 | Naikrovek | and we're here to help if you need it |
14:45.15 | Katty | i believe Digium will also help, on a pay per case basis. |
14:45.17 | spck | i also have to say polycom's rock |
14:45.23 | Katty | much like microsoft. |
14:45.30 | Katty | in case you get hung up in your underwear. |
14:45.33 | Gumug | right on |
14:45.41 | CareBear\ | Gumug : I've spent maybe 20 hours on it efficiently and now I have set up a small dial plan for my incoming number, my friends can register with me, and just last night I finally got internal calls between extensions going. |
14:45.45 | spck | you can also put bounty's on features you want |
14:45.57 | Gumug | sweet |
14:46.00 | Katty | Gumug: the only drawback with Asterisk, that i've found... |
14:46.11 | russellb | is that it's too awesome? |
14:46.13 | Katty | Gumug: is if something /breaks/, there isnt' an 800 numbe ryou can call for support. |
14:46.17 | russellb | yeah, sorry about that |
14:46.24 | Katty | Gumug: there's not a Support Contract or anything like that involved. |
14:46.35 | Katty | Gumug: like you would with a Hosted Solution. |
14:46.36 | russellb | You can purchase a support contract from Digium, actually. :-) |
14:46.41 | Naikrovek | well there are consultants who lurk in here that |
14:46.46 | Gumug | so get used to being in the fetal position crying? |
14:46.49 | spck | i would say the real problem with asterisk is that it doesn't have any one specific purpose and thus has been pulled in several directions |
14:46.56 | Katty | Gumug: no. you just pay for a support contract. |
14:47.07 | Katty | Gumug: it's not Included(tm) |
14:47.12 | Gumug | yes |
14:47.19 | CareBear\ | Gumug : Nono, support is available, from the makers of asterisk themselves (Digium) as well as from independant contractors. |
14:47.26 | Gumug | well, the problem i have with hosted solutions is, they charge a fortune |
14:47.30 | Katty | russellb: that's good to know. you have fixed pricing? link? |
14:47.33 | troffasky | spck, s/problem/feature imho :-) |
14:47.33 | Naikrovek | Gumug: it's a learning curve, but fixes are never far away, if that makes sense. fixes in asterisk are very quick, with few steps, but it's hard to knwo what to do if you're new |
14:47.42 | CareBear\ | Gumug : The software you get for no cost, support is sold separately. :) |
14:47.43 | Gumug | sure |
14:47.52 | Gumug | understood |
14:47.59 | Naikrovek | Gumug: but once you know what to do, it's a pain free setup really |
14:48.03 | Naikrovek | more or less |
14:48.04 | CareBear\ | Gumug : I think you can get all the support suppliers to work on your own infrastructure. |
14:48.08 | russellb | Katty: http://www.digium.com/en/supportcenter/asterisk.php |
14:48.09 | CareBear\ | Gumug : Most if not all |
14:48.19 | spck | give yourself plenty of time to plan, design, and test |
14:48.44 | leifmadsen | and then double it |
14:48.45 | leifmadsen | :) |
14:48.52 | Naikrovek | heh |
14:49.01 | Katty | russellb: that pricing is very competative. Samsung is doing 140 per case right now--available only to certified techs. |
14:49.02 | Naikrovek | ???? compiled and installed asterisk 1.6.1 - where did 'core show translation' go |
14:49.06 | leifmadsen | most people under-estimate how long it can take to build and test a complex solution |
14:49.10 | MWE | i put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put him in |
14:49.10 | MWE | the meetme room with a pincode instead of connect him directly with the caller |
14:49.10 | [TK]D-Fender | crazybyte: Brand new card? |
14:49.23 | crazybyte | [TK]D-Fender, yes, I think so |
14:49.36 | Gumug | leifmadsen: exactly |
14:49.38 | CareBear\ | D-Fender : I didn't get a chance to say thanks last night, so thanks for your help! :) |
14:49.39 | [TK]D-Fender | crazybyte: You probably forgot to set the jumper to E1 like you're supposed to |
14:49.47 | crazybyte | I see |
14:49.54 | [TK]D-Fender | CareBear\: did you get to finish it successfully? |
14:49.55 | russellb | Katty: spread the word ;-) |
14:50.08 | CareBear\ | D-Fender : Yes, I was missing auth= for my peer, only had credentials in the register=> |
14:50.35 | [TK]D-Fender | CareBear\: Excellent. You should not have cross-site CallerID as well |
14:50.52 | crazybyte | well it's not here. it's a remote machine so i suppose they forgot to do it. any other possible reasons (that you can think of) for such issues? |
14:51.02 | Gumug | i'm more and more inclined to just do it myself |
14:51.17 | Gumug | i don't want to get hosed on hardware, if i could do it myself for 1/4 the cost |
14:51.24 | CareBear\ | Gumug : If you have a bit of time to spare and don't have to rush it, then it's a great learning experience. |
14:51.29 | Gumug | yes |
14:51.35 | Gumug | i will be invaluable then as well |
14:51.39 | CareBear\ | ;) |
14:51.50 | p3nguin | gumug: If you know how to put a computer together, you can certainly build your own asterisk server. |
14:52.03 | Gumug | been doing that for 15 years |
14:52.13 | p3nguin | Then you're already ahead in the game. |
14:52.14 | troffasky | so burn a trixbox ISO and crack on with it |
14:52.21 | Naikrovek | yeah |
14:52.26 | Naikrovek | i was going to suggest asterisknow |
14:52.28 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
14:53.20 | [TK]D-Fender | [10:51]<p3nguin>gumug: If you know how to put a computer together, you can certainly build your own asterisk server. <- build yes, configure... that's another matter |
14:53.30 | Gumug | yes |
14:53.43 | [TK]D-Fender | SILLY RABBIT TRIXBOX IS FOR KIDDIES! |
14:53.49 | Naikrovek | well yes |
14:53.51 | SuPrSluG | yeah talk him into a GUI he'll get a lot of help here then |
14:53.59 | p3nguin | hahaha |
14:54.07 | Naikrovek | but it's the best place to start if he wants to get going quickly |
14:54.10 | [TK]D-Fender | reaches for his ClueBat (tm) |
14:54.20 | l2trace99 | does anyone know if asterisk performance shows correctly in top ? |
14:54.28 | SuPrSluG | er. no |
14:54.33 | Naikrovek | l2trace99: why wouldn't it |
14:54.34 | p3nguin | I would rather he didn't have a GUI and learned how things work from the CLI first. |
14:54.36 | lirakis | its the place to start if you want to double the amount of work you have to do to learn anything |
14:54.42 | *** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk) |
14:54.44 | SuPrSluG | the best place is was and will be |
14:54.48 | SuPrSluG | ~book |
14:54.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
14:54.50 | Katty | takes cluebat away from [TK]D-Fender |
14:55.17 | p3nguin | Once you learn how to do it without a GUI, then you can learn how to use the GUI if you think it would make you more productive. |
14:55.20 | fuxu2 | fender: I always find a LART stick works better |
14:55.22 | l2trace99 | is there a threading issue that top incorrectly reads ? |
14:55.25 | [TK]D-Fender | locks down a death-grip on his ClueBat |
14:55.35 | [TK]D-Fender | Katty: MY PREEEEEECCCCCIIIOOOUUUSSS!!!!!!!!! |
14:55.38 | p3nguin | l2trace99: try htop |
14:55.50 | Katty | [TK]D-Fender: play nice!! |
14:55.54 | Katty | gives cluebat back to [TK]D-Fender |
14:56.05 | Naikrovek | you guys always think gui = stupid. freepbx shows WHAT can be done, then you get interested in how to do it via conf files. that how it worked for me, anyway |
14:56.24 | [TK]D-Fender | Naikrovek: No... Trixbox is a particularly flaming piece of shit. |
14:56.35 | Naikrovek | yes, well i'm talking of freepbx |
14:56.35 | [TK]D-Fender | Naikrovek: With f-ing sprinkles on top |
14:56.46 | mbrevda | anyone know where I can find a tiny weeny bit of documentation on sip fax detection in asterisk 1.6.2 |
14:56.49 | *** join/#asterisk Juggie (n=Juggie@99.224.81.113) |
14:57.00 | p3nguin | Ordinarily, if someone learns by way of GUI, they rely on said GUI and never bother to learn to do it without said GUI. |
14:57.04 | *** join/#asterisk tripps (n=sean@76.31.197.242) |
14:57.12 | [TK]D-Fender | Naikrovek: FreePBX instaled yourself on * compiled yourself off standard sources is fine as long as you can live within the constraints of the GUI |
14:57.30 | [TK]D-Fender | You learn jack-shit from a GUI. |
14:57.34 | mbrevda | thinks: unless they become a dev of said gui, in which case they need to learn TWO systems |
14:57.47 | Naikrovek | i learned a lot from freepbx |
14:57.51 | p3nguin | That's why I say learn on the CLI and then move to the GUI if you think it will make you more productive. |
14:58.01 | [TK]D-Fender | Every idiot I've seen who has tried took 10 times as long to UNLEARN garbage than it would for a newb to learn from scratch |
14:58.22 | Naikrovek | you guys are so polarized on the gui thing, there's no way you can be correct |
14:58.35 | Naikrovek | it's just not black and white |
14:58.36 | [TK]D-Fender | Naikrovek: Or it really is that evident :p |
14:58.52 | mbrevda | Naikrovek: +1 |
14:58.58 | mbrevda | +1, +1, +1! |
14:59.06 | Naikrovek | [TK]D-Fender: that's unlikely, i KNOW in my case, the gui taught me a lot |
14:59.07 | jaytee | "Less filling!!!" |
14:59.16 | jaytee | "Tastes great!!!" |
14:59.17 | Katty | :< |
14:59.30 | p3nguin | I've never seen a Linux SysAdmin that relied heavily on any GUI. It's always the Windows fanboys who can't handle the command line. |
14:59.46 | [TK]D-Fender | Naikrovek: before learning *, or after? |
14:59.53 | mbrevda | rubs his hand gleefuly waiting to take a big warm bite out of a delecious flame war |
14:59.57 | [TK]D-Fender | takes jaytee's beer away |
14:59.58 | Naikrovek | and it's always linux cmdline extremists who call windows users "fanboys" |
15:00.07 | Gumug | if i can setup the system for a lot less than a turnkey solution, i'll look like i walk on water |
15:00.09 | mbrevda | lol |
15:00.10 | Katty | gives jaytee an amberbock |
15:00.16 | Katty | jaytee: don't drink crap! |
15:00.18 | Naikrovek | [TK]D-Fender: in tandem |
15:00.23 | jaytee | some people like GUI based voip systems while most of us don't. Some people like drinking Miller Genuine Draft while the rest of us don't care for the taste of recycled urine. |
15:00.34 | [TK]D-Fender | Naikrovek: I'm talking post. tandem/post can be OK. |
15:00.37 | Naikrovek | [TK]D-Fender: as i saw things in freepbx, I was curious how to do those thigns without freepbx |
15:01.02 | Katty | Naikrovek: i did that too (= |
15:01.03 | [TK]D-Fender | Naikrovek: Anyone who lets the perverted terminology from GUI's sit in their head are on a LONG road back... |
15:01.08 | Katty | Naikrovek: that's how i found asterisk-stat |
15:01.17 | Katty | tho not technically part of 'freepbx' |
15:01.27 | p3nguin | gumug: Cost of hardware + cost of man hours. Asterisk is free, so you only have to spend for hardware and time. |
15:01.37 | Gumug | p3nguin: yes |
15:01.38 | [TK]D-Fender | Naikrovek: And yes, bright people starting earlier on can do fine. the problem is that people are dumb, and GUI's tend to make people lazy.... |
15:01.40 | Naikrovek | [TK]D-Fender: I agree, but to say that using a GUI automatically makes you stupid is incorrect. the converse, to say that being stupid makes you use a gui, that's usually correct in my experience |
15:01.57 | Gumug | hardware i don't think, if i build it all myself |
15:02.06 | Gumug | will cost nearly as much as purchasing a turnkey |
15:02.08 | Naikrovek | but don't get caught in the notion that a mouse dumbs you down |
15:02.16 | Naikrovek | Gumug: much less |
15:02.33 | Gumug | do i have to buy a special telelphone like card |
15:02.40 | Naikrovek | Gumug: my * server cost maybe $1500, and could probably handle 500 simultaneous calls |
15:02.45 | [TK]D-Fender | Naikrovek: directly no, but it is a common byproduct to expect to be able to jsut click your way through everything and whine when there is no "go" button |
15:02.46 | Gumug | whoa |
15:02.57 | p3nguin | gumug: It depends on how you plan to configure things. |
15:03.02 | MWE | i put someone in the meetme room with the flag b (run a background script) that script calls somebody with dial and had to set that user in the same meetmeroom. But instead of putting the callee in the meetme room there will be a bridge between the caller and the callee. When the callee hangsup the script is trying to put him again in the meetme room.. IS there a way that the background script is doing the follwing: call the callee and put him in |
15:03.02 | MWE | the meetme room with a pincode instead of connect him directly with the caller |
15:03.06 | Naikrovek | Gumug: if you want to plug into physical phone lines, yes, if you want to buy voip service from smoeone, no special hardware is required |
15:03.14 | Gumug | VOIP |
15:03.33 | Katty | what's infobot's trigger for voip telcos? |
15:03.37 | [TK]D-Fender | Gumug: You only need special cards to interface with physical lines yourself |
15:03.42 | [TK]D-Fender | ~itsplist-us |
15:03.43 | infobot | somebody said itsplist-us was Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
15:03.54 | [TK]D-Fender | ~itsplist-ca |
15:03.55 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
15:03.56 | Gumug | ok |
15:03.56 | p3nguin | gumug: I run strictly SIP phones with a SIP trunk. All I need is a regular computer with adequate resources. |
15:04.10 | Katty | Gumug: we have a pri and bandwidth.com channels. |
15:04.19 | Naikrovek | [TK]D-Fender: yes, fools like guis. but guis don't make fools. |
15:04.29 | p3nguin | sometimes |
15:04.49 | Katty | There are some situations where a GUI can be useful. |
15:05.02 | Katty | For example, Katty is on vacation. and Regular people need to make changes or alterations. |
15:05.06 | [TK]D-Fender | Naikrovek: No, but give people TV's and watch them slowly devolve into couch-potatos :) |
15:05.28 | Katty | And Katty refuses to answer the phone while on the beach at Fiji. |
15:05.32 | Katty | Excellent reason to have a GUI. |
15:05.34 | [TK]D-Fender | Katty is talking about Katty's self in the third-person again... |
15:05.39 | Katty | She does that a lot. |
15:05.40 | [TK]D-Fender | suggests upping the meds |
15:05.50 | Katty | threats [TK]D-Fender with the bat. |
15:05.53 | TJNII | smirks as Naikrovek's comments about fools and GUIs after he so firmly defended his using a GUI |
15:05.55 | Katty | s/threats/threatens/ |
15:06.24 | [TK]D-Fender | TJNII : Its the user, not the tool. |
15:06.26 | Naikrovek | TJNII: allow me to show you the half dozen putty windows I have open |
15:06.31 | [TK]D-Fender | Unless the user IS a tool :p |
15:06.35 | Naikrovek | thank you |
15:06.35 | jaytee | oh no! they're gonna do a reboot of Highlander |
15:06.36 | [TK]D-Fender | Which happens a lot |
15:06.38 | Naikrovek | that's what i'm trying to say |
15:06.48 | TJNII | Naikrovek: I just find it amusing, that's all. No need to get defensive. :P |
15:07.13 | [TK]D-Fender | Naikrovek: You are currently on "exception" status with me.... can't be said of most GUI users... |
15:07.16 | Naikrovek | i prefer command line for most things, but sometimes, for a quick thing, gui IS faster |
15:07.20 | mbrevda | Naikrovek: so use PuTTY COnection mamanger |
15:07.22 | Gumug | $30 is a lot less than our $60 we pay |
15:07.35 | Gumug | VoIP compared to analog |
15:07.38 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
15:08.57 | Katty | Gumug: you can also usually get a 'break' with volume orders. |
15:09.04 | Katty | Gumug: porting numbers is pretty easy too |
15:09.44 | Katty | eppigy: i saw this and thought of you. http://imgur.com/ePnzw.jpg |
15:11.26 | SuPrSluG | mbrevda: give poderosa a whirl |
15:11.33 | p3nguin | port rinds? |
15:12.08 | SuPrSluG | is that the liitle left overs at the switch |
15:15.20 | scalex000 | TK: I advance I little ha ha ha |
15:18.27 | *** join/#asterisk m0bius (n=mobius@mailexchange.realize.gr) |
15:18.33 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
15:21.52 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
15:22.00 | Katty | why is it so quiet? :< |
15:22.03 | Katty | pokes people. |
15:22.09 | p3nguin | ack |
15:22.16 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
15:22.44 | p3nguin | wcselby: I just learned another one of those words I can't type right. |
15:22.50 | wcselby | what was updated in the topic? the astricon discount code? |
15:22.55 | wcselby | p3nguin - what's that? |
15:23.48 | p3nguin | wcselby: "port rinds" ... I apparently can't type "pork" correctly. |
15:23.55 | wcselby | haha |
15:24.11 | p3nguin | I never knew about that one before. |
15:24.29 | wcselby | lol yeah |
15:24.47 | [TK]D-Fender | port rinds... alcoholic pig? |
15:25.01 | wcselby | i was thinking along the same lines [TK]D-Fender |
15:25.07 | *** join/#asterisk volker- (n=volker@h1311547.stratoserver.net) |
15:25.15 | wcselby | pork rinds made that have been marinated in an overly sweet wine |
15:25.20 | volker- | hi |
15:25.25 | [TK]D-Fender | I'd rather have a bottle in front of me than a frontal lobotomy... |
15:25.27 | wcselby | howdy volker- |
15:26.00 | volker- | I have questions about the SIP Protokoll. And I read the corresponding rfc |
15:26.06 | p3nguin | It's just one of those things, I guess. |
15:26.25 | *** join/#asterisk moy (n=moy@74.12.131.104) |
15:26.29 | volker- | I dont know any other irc channels about this topic so I drop my questions here |
15:26.56 | wcselby | ~ask |
15:26.57 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
15:26.58 | volker- | 1. does authentification needs encryption? |
15:27.12 | volker- | 2. how secure is MD5-sess and token? |
15:27.48 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
15:28.28 | Katty | http://farm1.static.flickr.com/88/247766036_fdeb942446_b.jpg <- check out THAT wiring job. |
15:29.17 | p3nguin | You did a marvelous job with that wiring. |
15:29.29 | Katty | :> |
15:29.42 | Katty | i love the picture sub-reddit. |
15:30.22 | wcselby | volker- - I'm not sure, but I don't think SIP needs encryption for auth |
15:30.42 | wcselby | volker- - at least not as far as asterisk is concerned |
15:31.48 | wcselby | [TK]D-Fender will probably correct me if I'm wrong |
15:32.25 | volker- | wcselby: if it needs encryption it should have a "algorithm=none" in the rfc. but it doesn't. I cant dig the information out of the RFC :( |
15:32.39 | Katty | http://imgur.com/JjQ2E.jpg :> |
15:32.41 | volker- | wcselby: errr, if it needs NO encryption :) |
15:32.48 | Katty | "MY FISH" |
15:33.06 | wcselby | Katty - I was thinking someone went otter fishing with really big bait |
15:33.57 | volker- | i'm not totaly sure how secure md5 is in connection with sip, can someone tell me? |
15:35.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
15:35.43 | Katty | does anyone here like Lentils? |
15:37.22 | coppice | I've bean eating them for years |
15:37.29 | Katty | how do you fix them? |
15:37.40 | [TK]D-Fender | coppice: I can't stand much more of this PUNishment.... |
15:37.46 | coppice | I don't eat the broken ones |
15:37.49 | *** join/#asterisk errr (n=errr@fedora/errr) |
15:38.00 | Katty | facepalm |
15:40.32 | SuPrSluG | Katty: kinda like risotto, use chicken broth and cook em adding as much as needed. |
15:41.31 | Katty | let me rephrase my question--I know how to cook them. |
15:41.34 | SuPrSluG | Katty: or make soup out of em |
15:41.43 | Katty | what is your prefered recipe that uses them. |
15:42.35 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
15:42.38 | volker- | thanks for your help |
15:42.39 | volker- | bye :) |
15:44.47 | SuPrSluG | a side with lamb |
15:45.01 | CareBear\ | is off - later people! thanks for the help so far! |
15:45.03 | *** part/#asterisk CareBear\ (i=peter@stuge.se) |
15:45.22 | Katty | mkay. |
15:45.29 | Katty | not sure where i can find lamb around here. |
15:45.36 | Katty | it's not a very commonly carried item. |
15:46.30 | [TK]D-Fender | Katty: dal makhani |
15:46.56 | Naikrovek | mmmm lamb chops |
15:47.29 | wcselby | Naikrovek - you said you have polycom ip 6000's, yes? |
15:47.30 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
15:47.43 | Naikrovek | i compiled and installed * 1.6.1 from svn a bit earlier - where did 'core show translation' go? no workie |
15:47.46 | Naikrovek | wcselby: yes |
15:47.58 | Katty | [TK]D-Fender: is this like... |
15:48.01 | Katty | [TK]D-Fender: refried beans? |
15:48.13 | wcselby | can I see how you've got your sip.conf setup for it, and maybe get a copy of your chone configs? |
15:48.17 | [TK]D-Fender | Katty: Nope |
15:48.17 | Katty | it looks good, from google images. |
15:48.20 | wcselby | i need to compare what's going wrong here on my end |
15:48.29 | [TK]D-Fender | Katty: <3 Indian |
15:48.33 | Naikrovek | wcselby: what's the problem |
15:48.46 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
15:48.57 | e4 | Apparently overnight our Grandstream phones stopped registering. It looks like they are sending sip packets to the asterisk server but asterisk logs aren't showing any activity and a tcpdump doesn't reveal any return traffic. I'm not quite sure where to look next, any ideas? |
15:49.09 | Naikrovek | mm grandstream |
15:49.14 | Naikrovek | there's a steaming pile of phone |
15:49.18 | wcselby | Naikrovek - the phone registers, you can call it, it can call out, for about three to five minutes. then it loses registration with the * server and no more workie |
15:49.20 | *** join/#asterisk aaronr (n=arussell@cpc3-stkn13-2-0-cust249.11-2.cable.virginmedia.com) |
15:49.37 | Naikrovek | e4: check any firewalls or anything, sometimes they can be blocked by adaptive software |
15:49.54 | e4 | Naivkrovek: No firewall on this interface. |
15:50.03 | Naikrovek | wcselby: loses registration?! during a call? wow |
15:50.09 | wcselby | yeah |
15:50.11 | Naikrovek | e4: are your phones and your asterisk server on the same subnet |
15:50.16 | Naikrovek | wcselby: same question |
15:50.21 | e4 | Naikrovek: Yes |
15:50.22 | *** part/#asterisk aaronr (n=arussell@cpc3-stkn13-2-0-cust249.11-2.cable.virginmedia.com) |
15:50.34 | Katty | [TK]D-Fender: traditionally, are those Lima beans or Kidney beans? |
15:50.37 | e4 | I do know enough about networking to ensure that's not the case ;) |
15:50.42 | Naikrovek | e4: okay cool |
15:50.57 | wcselby | Naikrovek - yes. I had a sip debug yesterday but can't find it. I'll make a new one today. |
15:50.58 | Naikrovek | e4: how many phones |
15:51.12 | Naikrovek | wcselby: pm me your email and i'll email you my config files |
15:51.13 | e4 | Naikrovek: Just 3, we've got a small setup. |
15:51.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
15:51.32 | [TK]D-Fender | Katty: "Other" |
15:52.56 | wcselby | e4 - you've of course rebooted the phones, done a reload on asterisk, etc? |
15:53.51 | e4 | wcselby: Yep. |
15:56.05 | e4 | The thing that I'm stuck on is that Asterisk isn't showing me *anything* in the way of logs, be it rejected connections, failed registry attempts, anything. |
15:56.12 | Katty | ponders lunch |
15:56.47 | leifmadsen | goes to get lunch! |
15:56.49 | [TK]D-Fender | e4: Global SIP debug @ * CLI shows nothing when you reboot a phone? |
15:56.52 | Gumug | $11k for fonality solution.. |
15:56.56 | Naikrovek | one thing that would be nice on polycom phones. if you could determine the model of phone from the web interface |
15:57.09 | p3nguin | wcselby: Sounds as if you're experiencing something equally as annoying as a problem I've been trying to solve. I've got this one computer which uses a softphone... both twinkle and ekiga can register to * and can receive calls, but neither of the softphones can make calls from that computer. Zoiper on a Windows box on that same remote network works perfectly, though. |
15:57.10 | Katty | leifmadsen: :< |
15:57.13 | Gumug | is $245 per phone a lot? |
15:57.14 | Katty | leifmadsen: TAKE ME WIF YOU |
15:57.15 | wcselby | e5 - have you enabled sip debug on one of the xtensions? |
15:57.27 | Katty | Gumug: depends on the type of phone. desk phone--yeah that's a bit high. |
15:57.32 | Gumug | desk phone |
15:57.40 | Katty | Gumug: polycom 330s are around 130ish |
15:57.48 | Katty | Gumug: which is your basic Desk Phone |
15:57.49 | Gumug | those are 2 line phones |
15:57.52 | p3nguin | less if you don't want a power supply with it. |
15:57.58 | Gumug | what about 3 line phones |
15:58.02 | Gumug | or 4 |
15:58.04 | Katty | lines are irrelative. |
15:58.07 | [TK]D-Fender | Gumug: You don't need |
15:58.09 | Katty | just put your phone in a group. |
15:58.15 | Katty | and assign as many lines to the group as you want. |
15:58.16 | Gumug | ok |
15:58.21 | Gumug | sounds good |
15:58.22 | Katty | it just picks up the next free one |
15:58.32 | Gumug | does asterisk have a HUD? |
15:58.33 | Naikrovek | none of that "pick up line two" BS |
15:58.35 | Gumug | web hud? |
15:58.42 | Naikrovek | Gumug: freepbx does :) |
15:58.43 | Katty | there are some 3rd party HUDs. |
15:58.48 | Gumug | i need a HUD |
15:58.53 | Katty | HUDlite, Isymphony (I'm a big fan) and FOP. |
15:58.56 | Katty | FOP isn't so Hot. |
15:59.03 | Katty | aka, Flash Operator Panel. |
15:59.03 | Naikrovek | yeah fop sucks |
15:59.18 | Naikrovek | Gumug: these are what i use: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-321 |
15:59.21 | Katty | it does an Okay job if you're just looking at stuff. it's a bit quirky. not much interaction. |
15:59.22 | e4 | <[TK]D-Fender>: Correct :/ |
15:59.30 | Katty | Isymphony works with the manager. Lets you transfer calls around via drag and drop. |
15:59.36 | Gumug | nice |
15:59.43 | Katty | We've been using it for about a year or two now. I really like it. |
15:59.48 | Katty | It's Pretty(tm) |
15:59.55 | [TK]D-Fender | e4: Dump your firwall, prove that * has bound the SIP port, etc |
15:59.58 | Katty | also has support for a jabber server, which lets you IM. |
16:00.13 | Katty | you can have a 'free' version for 5 clients, limited features. 20 bucks a client for the pro version. i think it's well worth it. |
16:00.19 | Naikrovek | [TK]D-Fender: he says he doesn't have a firewall on that interface |
16:00.22 | Katty | does call recording, call barging with little buttons. |
16:00.23 | Naikrovek | but i dunno... |
16:00.29 | Naikrovek | iptables is a monster and i hate it |
16:00.30 | [TK]D-Fender | Naikrovek: says != proof |
16:00.37 | Naikrovek | [TK]D-Fender: agreed |
16:00.55 | [TK]D-Fender | Naikrovek: and iptables has never failed me |
16:00.58 | Naikrovek | Gumug: those phones are two line phones, with up to two calls per line |
16:01.08 | Gumug | ya |
16:01.15 | Katty | i wouldn't use a 330 for a receiptionist. |
16:01.29 | Gumug | sometimes we get slammed |
16:01.31 | Katty | their display isn't very big. it's Annoying to transfer calls around. |
16:01.36 | Naikrovek | [TK]D-Fender: it's a good firewall, i've just not been able to make it do what i want. all my machines are unfirewalled and i do security on the border, but not sure if that's a good idea or not |
16:01.36 | Gumug | but that shouldn't be a problem i guess |
16:01.39 | Gumug | if 4 people call |
16:01.51 | Katty | if i recall correctly it displays the first caller. |
16:01.55 | Naikrovek | Katty: yeah the 330 is not a receptionist phone if you have a busy receptionist |
16:01.57 | [TK]D-Fender | Naikrovek: Common argument says "not". |
16:01.57 | Katty | then you have to scroll down to highlight calls after that. |
16:02.08 | Katty | it's just annoying. |
16:02.19 | Naikrovek | [TK]D-Fender: awesome. i still hate iptables though |
16:02.21 | Katty | our receiptionist has something in the 500 line i think. |
16:02.30 | Katty | an older 501 i think |
16:02.42 | Naikrovek | the 601 has some nice expansion panels. |
16:02.43 | Katty | 450 might have a bigger display. can't recall. |
16:02.50 | p3nguin | What if iptables had a GUI? :) |
16:02.54 | Katty | yeah, the 600 line has side cars. |
16:02.55 | [TK]D-Fender | Naikrovek: You're quite welcome to. I know most networking people love pf a heck of a lot more, but I never got around to learning the BSD's |
16:02.57 | Naikrovek | i still wouldn't lke it |
16:03.10 | Katty | we have no need for side cars here...we use isymphony. |
16:03.23 | wcselby | if I've got a border firewall I dump iptables first chance I get |
16:03.25 | Naikrovek | iptables makes sense but i don't know what i'm doing with it yet i guess |
16:03.31 | *** join/#asterisk diatonic1 (n=chillman@mail.clearwater-research.com) |
16:03.32 | Katty | most of the time she just clicks and drags a call to an extension and rarely touches the phone |
16:03.33 | [TK]D-Fender | Polycom IP60X = discontinued |
16:03.35 | [TK]D-Fender | Avoid |
16:03.43 | *** join/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
16:04.01 | p3nguin | naikrovek: Set default policies to DROP. Add rules to allow what you want. Done, enjoy. |
16:04.06 | Katty | some people like snomphones. |
16:04.10 | Katty | but they feel cheap. |
16:04.20 | Katty | got one upstairs in the bedroom at my house. just don't like it. |
16:04.34 | _trine | how can I change the smtp port from 25 to something else in asterisk is there a .conf file where you can change it? |
16:04.57 | p3nguin | * runs an smtpd? |
16:05.05 | coppice | Katty: they are big on ugly, but the seem more solidly built than most |
16:05.06 | Naikrovek | _trine: why would you change it, and why would you have it running on a network port |
16:05.07 | Katty | _trine: my guess is that would be your email configs. |
16:05.20 | diatonic1 | Hey all... I'm wanting to modify sending caller ID when I call a certain area code (in the US). Can anyone point me somewhere that might have the info on how to modify the dial plan? I'm running Trixbox/FreePBX |
16:05.32 | Katty | coppice: it might be that i'm used to a IP501, and this is a snom 190(i think) |
16:05.35 | coppice | Katty: they actually fix reported bugs, too, which is almost unique in this industry |
16:05.37 | wcselby | okay, I've unplugged a sip phone, then turned on SIP debugging for that extension, with the intent of grabbing a SIP debug from boot-up to when my problem occurs. However, before I plug in the phone, the cli is showing me tons of SIP traffic to the IP address of the phone that's not plugged in.... |
16:05.38 | Naikrovek | _trine: but to change it would depend on the mailer that is running |
16:05.44 | _trine | I want to send the voice messages to my email address but I donr use the standard port |
16:05.53 | Naikrovek | wcselby: ip conflict? |
16:05.57 | e4 | <[TK]D-Fender>: http://pastie.org/private/dsbebmbdaa2p5fynsazeq |
16:06.18 | Katty | _trine: asterisk uses a smtp server to send the messages. it's not an smtp server in and of itself. |
16:06.21 | _trine | it's using the default asterisk mailer |
16:06.26 | wcselby | Naikrovek - shouldn't be. I'm investigating |
16:06.36 | *** join/#asterisk carrar (i=tim@osburn.com) |
16:06.46 | Katty | _trine: you need to figure out what it's using and then edit that conf file. |
16:06.59 | _trine | Katty: thanks |
16:07.11 | [TK]D-Fender | e4: FLUSH THEM |
16:07.36 | wcselby | _trine - asterisk doesn't have a default mailer. It's using sendmail if you haven't setup anything else on the box. |
16:07.43 | _trine | I have asterisk running on a pendrive which in turn boots my router |
16:07.48 | e4 | <[TK]D-Fender>: Yes, done about a dozen times just in case, heh. |
16:07.57 | _trine | I have not set sendmail up |
16:08.07 | wcselby | _trine - so go read up on making changes to that. * does not have a built in smtpd |
16:08.09 | p3nguin | MTAs talk to each other on port 25. You sure you want to change the port? |
16:08.18 | wcselby | _trine - sendmail is setup by default on most linux distro's.... |
16:08.22 | [TK]D-Fender | _trine: * doesn't have a mailer. It jsut calls a standard sendmail script. mod it however you want |
16:08.24 | l2trace99 | mmmmm voip mtas |
16:08.35 | *** join/#asterisk Tim_Toady (n=moi@adsl290-154.kln.forthnet.gr) |
16:08.52 | _trine | OK it looks like I will have to install sendmail on my router |
16:08.58 | p3nguin | Why? |
16:09.23 | diatonic1 | _trine You should have an MTA on the * box - probably sendmail or postfix |
16:09.27 | p3nguin | Routers should, well, route. They usually don't need to be a mail relay. |
16:09.41 | wcselby | _trine - I'm not sure you fully understand what you're doing.... |
16:09.57 | p3nguin | I think you can be positive on that, wcselby. |
16:10.04 | diatonic1 | _trine see of there is a sendmail.cf file on your * box - if so, you've probably got sendmail |
16:10.06 | Gumug | iSymphony the best one? |
16:10.08 | _trine | wcselby: I not sure I'm actually here so don't worry |
16:10.10 | _trine | :P |
16:10.22 | wcselby | _trine - check the voicemail.conf file and see what file is being called to send emails |
16:10.31 | p3nguin | _trine: Just tell us the problem you're experiencing and what you THINK you want to do to solve it. |
16:10.46 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
16:11.18 | Katty | Gumug: there's no such thing as The Best |
16:11.28 | Gumug | there needs to be |
16:11.31 | Katty | Gumug: i'd highly recommend giving them all a try, and finding out what your people like best. |
16:11.35 | Gumug | would make my decision easier |
16:11.36 | _trine | I want to send the voicemail received to me email address but I forward port 25 on my router for other reasons |
16:11.43 | *** join/#asterisk carrar (i=tim@osburn.com) |
16:11.46 | _trine | /s/me/my |
16:11.53 | Katty | Gumug: one size does not fit all. |
16:12.10 | _trine | so for example I want to send the email to port 4444 |
16:12.17 | Gumug | see |
16:12.18 | p3nguin | _trine: Port forwarding is for inbound traffic to bypass the firewall. If your * box is behind that firewall, you don't need to forward anything. |
16:12.25 | Gumug | fonality is going to charge $2000 for a HUD |
16:12.41 | Katty | they will also support it. |
16:12.44 | _trine | p3nguin: I do |
16:12.44 | Katty | configure it. and install it. |
16:12.50 | Gumug | true |
16:13.03 | p3nguin | _trine: Your * box does not have an smtpd, so it cannot receive emails on port 25. |
16:13.07 | Katty | isymphony is around 20 bucks a seat license. |
16:13.17 | Katty | they won't install it for you. |
16:13.18 | Gumug | ya |
16:13.22 | Katty | but they do offer support. |
16:13.23 | e4 | [TK]D-Fender: That was it apparently, is that a fairly common thing? |
16:13.23 | Gumug | i know |
16:13.24 | Naikrovek | Katty: one size fits none usually |
16:13.29 | coppice | There are better massage chairs than the iSymphony |
16:13.32 | _trine | I don't want to receive any emails on my router |
16:13.32 | Gumug | alright i'm out of hear |
16:13.36 | Naikrovek | Gumug: laters |
16:13.39 | Gumug | thanks for the help guys |
16:13.39 | p3nguin | _trine: So unless your * box is outside of the firewall and your mail server is inside the firewall, this is a non-issue. |
16:13.39 | Katty | Gumug: byebye |
16:13.41 | Gumug | i'll be back later |
16:13.44 | Gumug | i appreciate you all |
16:13.46 | _trine | I just want to send them out on a port other than 25 |
16:13.54 | Gumug | ciao |
16:14.09 | [TK]D-Fender | e4: People putting STATE BASED rules for **UDP** and running into crazyness? It happens. I cname this one guy I know... |
16:14.19 | _trine | p3nguin: by box is not really a box it's a router |
16:14.29 | _trine | with asterisk running inside it |
16:14.34 | Naikrovek | coppice: what are some others |
16:14.44 | p3nguin | _trine: That's irrelevant at this point. |
16:15.01 | _trine | p3nguin: thanks for your help |
16:15.19 | p3nguin | _trine: * will send emails out via the local mailer daemon. Port forwarding is not needed for that to happen. |
16:15.44 | _trine | p3nguin: you obviously are not understanding my needs |
16:15.46 | p3nguin | Port forwarding is not required for outbound traffic. |
16:15.50 | e4 | [TK]D-Fender: Is an iptables accept all state-based? |
16:16.24 | Naikrovek | _trine: incoming ports and outgoing ports are different. mail doesn't LEAVE a machine via port 25, it ARRIVES at a machine on port 25 |
16:16.28 | p3nguin | If I'm not understanding it, it is because you haven't adequately described the problem. I'm trying to guide you into the right path, but you're looking down the road where you currently are. |
16:16.30 | [TK]D-Fender | e4: ACCEPT all -- anywhere anywhere state RELATED,ESTABLISHED <-------- BIG PRINT |
16:16.32 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
16:16.39 | Katty | _trine: pretty much everyone in here is a volunteer. if you feel your Needs are not being taken care of you might wait awhile and ask again later. If it's urgent you might consider finding a consultant to help you. |
16:16.46 | [TK]D-Fender | e4: accept all.. WITH BIG ADVERTISED CONDITIONS |
16:16.49 | p3nguin | Actually, mail does leave on port 25. But port forwarding has nothing to do with that. |
16:16.51 | _trine | Naikrovek: I inderstand that well lets just say 110 then |
16:16.54 | wcselby | _trine - when you say you're running a router with asterisk inside of it, what do you mean. what is your router? |
16:16.57 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
16:17.14 | *** join/#asterisk mumtazah (n=mumtazah@60.53.139.144) |
16:17.28 | Naikrovek | wcselby: some dd-wrt firmwares have asterisk in them |
16:17.35 | _trine | I have installed asterisk inside my router |
16:17.35 | _trine | it runs inside my router |
16:17.39 | e4 | [TK]D-Fender: Ah, interesting. I'll dig into that to fix it, thanks. |
16:17.43 | wcselby | what is your router _trine? |
16:17.48 | *** join/#asterisk carrar (i=tim@osburn.com) |
16:17.57 | wcselby | _trine - is it a linux box? a cisco device? what? |
16:17.58 | [TK]D-Fender | Naikrovek: See? FW FAIL :p |
16:18.06 | Naikrovek | [TK]D-Fender: yeah |
16:18.07 | _trine | it is a full asterisk and it's not ddwrt it's kamikaze |
16:18.14 | Naikrovek | _trine: ah cool |
16:18.16 | _trine | it's a WRT160NL |
16:18.18 | troffasky | wcselby, asterisk on a cisco router? what are you on? |
16:18.22 | wcselby | _trine what is kamikaze? never heard of that router brand? |
16:18.33 | troffasky | kamikaze is an openwrt release |
16:18.33 | Naikrovek | troffasky: it runs well provided you don't need to transcode anything |
16:18.35 | Katty | i think that's a linksys |
16:18.35 | wcselby | troffasky - I know you can't do that. I wanted to hear what he was trying to do |
16:18.35 | _trine | all running of an 8G pendrive |
16:18.39 | p3nguin | Sounds like a firmware for a Linksys router. |
16:18.49 | troffasky | like karmic is an ubuntu release |
16:19.04 | Naikrovek | i said that earlier. you can get open source firmwares for linksys routers that have asterisk in them |
16:19.13 | Naikrovek | which means, your router runs linux |
16:19.21 | wcselby | Naikrovek - thank you. |
16:19.26 | _trine | from my understanding asterisk has a default mailer ,, is that correct |
16:19.30 | Naikrovek | so he's got an internal mail server he wants to receive mail on |
16:19.34 | wcselby | Naikrovek - which means he's probably got sendmail installed on the router and doesn't know it. |
16:19.41 | Naikrovek | and he doesn't want the router to get in the way of that |
16:19.41 | wcselby | _trine - NO THAT IS NOT CORRECT |
16:19.42 | Katty | _trine: no. it does not have a Default Mailer built in. |
16:19.51 | wcselby | _trine - we've told you this multiple times. |
16:19.51 | Katty | wcselby: please be polite. |
16:20.02 | p3nguin | Still makes no difference on port forwarding. There is no internal LAN where the port needs to be forwarded from the outside. |
16:20.07 | Chainsaw | takes wcselby's megaphone away |
16:20.07 | Katty | wcselby: you never know what sort of day a person is having. |
16:20.08 | wcselby | _trine - asterisk uses whatever the default mailer is on your linux distro you're using |
16:20.13 | _trine | wcselby: keep your knickers on you'll wet yourself shouting like that |
16:20.20 | [TK]D-Fender | _trine: This has been explained multiple times. Lets see if this one sinks in.... |
16:20.24 | Naikrovek | _trine: asterisk does not have a mailer of its own. it uses the operating system mailer |
16:20.37 | Naikrovek | _trine: if you're running asterisk on a router, it probably does not have a mailer |
16:20.38 | Chainsaw | Providing a sendmail binary should suffice. |
16:20.44 | wcselby | laughs |
16:20.44 | p3nguin | If the system has no mailer, configure one. |
16:20.53 | [TK]D-Fender | _trine: * calls a SENDMAIL BINARY. It does not COME with a mailer. If you don't have a script to process that request and send the mail then you have NOTHING |
16:21.00 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
16:21.00 | wcselby | _trine - go to your voicemail.conf file, see what it's using to send mail, then go configure that mailer |
16:21.13 | _trine | I'm looking now |
16:21.15 | Katty | k |
16:21.16 | p3nguin | And stop THINKING about port forwarding. |
16:21.25 | wcselby | Katty - i've said it multiple time, I figured maybe caps would get his attention, since he hasn't paid any attention yet |
16:21.28 | Katty | p3nguin: THERE IS A NO YELLING POLICY. |
16:21.39 | p3nguin | katty: OKAY! |
16:21.42 | Naikrovek | i still don't get why he wants to send mail from a different port than 25. |
16:21.43 | Katty | p3nguin: KTHX |
16:21.53 | wcselby | Naikrovek - because he's already doing something on port 25. |
16:21.56 | Katty | Naikrovek: understanding is not required. |
16:22.05 | Katty | Naikrovek: we have to assume he has a logicial reason for this. |
16:22.06 | p3nguin | doesn't matter if he's using 25 or not. |
16:22.08 | wcselby | Naikrovek - some ISP's don't allow outbound mail on port 25 from residential connections |
16:22.24 | [TK]D-Fender | p3nguin: I seem to have imprinted on you ;) |
16:22.32 | _trine | I'm reasonably sure it said somewhere if you don't want to use the default mailer then un comment this and use send mail |
16:22.39 | Naikrovek | but tcp origination ports have nothing to do with destination ports. when my mail server sends mail it could do it from port 3456 or 55555, it's the destination port that matters |
16:22.44 | p3nguin | wcselby: They do, but they force you to run it through their relays. |
16:22.48 | l2trace99 | i put my mta on 2525 |
16:22.50 | Katty | please play nice while i'm gone. |
16:22.53 | l2trace99 | but no one emails me |
16:22.54 | Katty | goes to lunch. |
16:23.06 | [TK]D-Fender | I think I'm a clone now... there always seems to be two of me around! |
16:23.08 | p3nguin | No one CAN email you when your MTA is on the wrong port. |
16:23.12 | wcselby | _trine - what you're reasonably sure about doesn't matter. read the conf files and find out for yoruself what you're using |
16:23.23 | Naikrovek | such anger for a humpday |
16:23.32 | l2trace99 | oh I thought no one loved me |
16:23.35 | p3nguin | naikrovek: MTAs send out on 25 and receive on 25. |
16:23.38 | _trine | wcselby: are you a felckled face little kid |
16:23.39 | wcselby | channels [TK]D-Fender |
16:23.40 | l2trace99 | i must be really popular |
16:23.42 | [TK]D-Fender | wcselby: No reason * would be configured for anything except the generic sendmail compatible binary |
16:23.45 | _trine | quiet |
16:23.50 | l2trace99 | and I missed it |
16:23.58 | [TK]D-Fender | wcselby: And whatever is actually going on behind the scenes is completely irrelevant |
16:24.25 | wcselby | am I freckled face little kid for trying to help you out? |
16:24.32 | [TK]D-Fender | 12:23]<p3nguin>naikrovek: MTAs send out on 25 and receive on 25. <- close.. they sed TO 25 and receive ON 25 :) |
16:24.33 | wcselby | okay, sure.... |
16:24.35 | [TK]D-Fender | send* |
16:24.44 | Naikrovek | [TK]D-Fender: thank you |
16:24.54 | p3nguin | [tk]d-fender: That's what I said. |
16:25.10 | Naikrovek | if you do a tcpdump while a mail is being sent, it will almost definitely not COME FROM port 25 |
16:25.27 | carrar | correct |
16:26.08 | [TK]D-Fender | p3nguin: you said send ON 25. that is the source port, which is pretty much assuredly NOT 25. |
16:26.08 | wcselby | bleh, I've got a SIP debug to filter |
16:26.08 | [TK]D-Fender | p3nguin: separate your SRC vs DST ports :) |
16:26.08 | _trine | this is what I have read in the voicemail.conf :- |
16:26.08 | _trine | ; You can override the default program to send e-mail if you wish, too |
16:26.08 | _trine | ; |
16:26.08 | _trine | ;mailcmd=/usr/sbin/sendmail -t |
16:26.08 | _trine | ; |
16:26.08 | wcselby | have fun storming the castle boys |
16:26.09 | Naikrovek | [TK]D-Fender: i said the same thing but he didn't listen to me |
16:26.09 | [TK]D-Fender | gathers the pitchforks & villagers |
16:26.13 | _trine | that's what seems to have been the confusing bit |
16:26.14 | Naikrovek | gets a wheelbarrow |
16:26.24 | Naikrovek | _trine: your mail will not come from port 25 anyway |
16:26.31 | troffasky | _trine, install whatever package has a sendmail in |
16:26.41 | carrar | it comes from the INNERTUBES |
16:26.47 | [TK]D-Fender | _trine: * is calling sendmail. Go get a clue on how to run a mail script. |
16:26.55 | diatonic1 | _trinemodify the sendmail.cf to use a smarthost on whatever port you need, then rebuild the sendmail.mc file |
16:26.59 | [TK]D-Fender | carrar: I love running those down snowy hills! |
16:27.11 | diatonic1 | _trine: modify the sendmail.cf to use a smarthost on whatever port you need, then rebuild the sendmail.mc file |
16:27.12 | troffasky | diatonic1, openwrt will not have a sendmail.cf in |
16:27.19 | carrar | no snow here yet |
16:27.19 | _trine | But you can understand how it gets confusing when this is written in the .conf file for voicemail |
16:27.31 | troffasky | but there will be a package with a sendmail in, which is what _trine needs |
16:27.41 | Naikrovek | troffasky: not for a router there won't be |
16:27.47 | troffasky | yes there will |
16:27.56 | diatonic1 | Ahh, missed the part about it being a linksys router :) |
16:27.57 | _trine | yes there will |
16:27.57 | Naikrovek | he'll have to install new firmware with sendmail in it |
16:28.12 | _trine | no I don't need to |
16:28.19 | troffasky | Naikrovek, openwrt has a package manager like every other linux distro |
16:28.30 | troffasky | so you don't need to reinstall the whole thing when you want to add new software |
16:28.32 | _trine | Naikrovek: the wonders of kamikaze are yet to be revealed to you :P |
16:28.55 | Naikrovek | eh |
16:29.06 | Naikrovek | last time i tried openwrt or ddwrt they maxed the cpu of my router |
16:29.12 | Naikrovek | so i removed them with extreme prejudice |
16:29.19 | _trine | on this router I have 8 gigs of space for packages |
16:29.35 | _trine | lol |
16:29.41 | Naikrovek | _trine: flash drive on USB? |
16:29.47 | _trine | usb |
16:29.59 | _trine | it is running as pivot-root |
16:30.17 | troffasky | _trine, opkg install mini-sendmail |
16:30.43 | _trine | yes thanks troffasky I will |
16:31.05 | _trine | ; You can override the default program to send e-mail if you wish, too |
16:31.15 | _trine | it was that one line that through me |
16:31.28 | _trine | in voicemail.conf |
16:35.01 | Naikrovek | _trine: did you have to solder in a USB port or does that router have one |
16:35.27 | _trine | Naikrovek: no it's all done for you they are great routers for the money |
16:35.50 | Naikrovek | _trine: I just got a 310N it has no USB |
16:36.03 | _trine | Naikrovek: if you need any help getting it going I will be glad to trade info with you as at the moment I know little about asterisk |
16:36.19 | Naikrovek | _trine: k |
16:36.22 | _trine | that's the WRT160NL |
16:36.30 | _trine | I can help with |
16:36.36 | Naikrovek | ah i read about that one. the L means linux |
16:36.52 | _trine | you can install kamikaze on them |
16:37.06 | Naikrovek | i should have gotten that one, but best buy doesn't carry it locally and my budget was limited to the gift card i had in my hand |
16:37.07 | _trine | at the moment though wifi is not working well |
16:37.15 | _trine | but i don't use wifi on it |
16:37.37 | _trine | it runs asterisk very well as a single end user |
16:37.56 | _trine | I also have freeswitch installed on it |
16:39.25 | _trine | Naikrovek: the good thing about them is in my view is that they have 8 flash 32 ram a 400MHz processor and you can boot into a usb drive and use it as root |
16:39.42 | Naikrovek | wonders why his asterisk compiled from source does not have ... ooh make menuconfig to turn that on i bet |
16:39.46 | Naikrovek | _trine: nice |
16:39.53 | _trine | so you are never stuck for space |
16:40.39 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
16:41.14 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
16:41.31 | _trine | Naikrovek: you need to update the packages with :- ./scripts/feeds update -a, ./scripts/feeds install -a |
16:41.35 | *** join/#asterisk ingenius (n=alektro@host172.190-231-93.telecom.net.ar) |
16:41.49 | _trine | then you get all the other packages |
16:42.18 | _trine | then make menuconfig |
16:43.28 | Naikrovek | feeds? |
16:43.33 | *** join/#asterisk Gumug (n=Gumug@adsl-75-39-194-126.dsl.spfdmo.sbcglobal.net) |
16:43.39 | Gumug | anyone used PBX in a flash? |
16:43.53 | Naikrovek | Gumug: yeah it's not so great from what i understand |
16:44.00 | Gumug | ok |
16:44.05 | Gumug | what do you recommend |
16:44.34 | Naikrovek | well if you want to use a distro, asterisknow. but i would recommend that you install asterisk on a linux system you like rather than that, even |
16:44.49 | Gumug | i am indifferent |
16:44.52 | Gumug | i literally |
16:45.01 | Gumug | haven't run linux since the kernel 2.0.36 |
16:45.11 | Gumug | that was the last kernel i ever compiled |
16:45.35 | Naikrovek | well if this is something you want to get into, if you think you may be interested in this telephony thing, i would compile it from source and do it all manually |
16:45.47 | Naikrovek | it'll be a much simpler configuration when you're done |
16:46.30 | Gumug | nod |
16:46.44 | Gumug | what is the difference between asterisk and fastswitch? |
16:46.50 | Naikrovek | freeswitch |
16:47.07 | Gumug | sorry yes |
16:47.13 | Naikrovek | not sure, never used freeswitch. I believe the consensus is that it's pretty awesome, but doesn't have all the features of asterisk |
16:47.24 | Naikrovek | i've never used it |
16:48.11 | Gumug | right on |
16:48.13 | Gumug | afk few |
16:50.17 | saint_ | is there a physical tool beside wireshark that can be used to analyze SIP traces ? |
16:50.51 | voipmonk | fastswitch, nice name |
16:51.40 | dustybin | no work tomorrow |
16:51.41 | dustybin | :D |
16:56.21 | diatonic1 | Can anyone tell me how to modify the dial plan based on destination area code? (* newbie) Rinning Trixbox CE 2.4 |
16:56.23 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
16:56.34 | Naikrovek | diatonic1: ask in #trixbox |
16:56.54 | Naikrovek | i'm in there, i'll help you. |
16:57.10 | Naikrovek | but lets do it in there to keep this channel clear of argument |
16:59.38 | [TK]D-Fender | puts away his rusty nail embedded ClueBat (tm) |
17:05.42 | scalex000 | TK: do you have document about H323 |
17:05.43 | SuPrSluG | saint: you could use ngrep |
17:08.37 | [TK]D-Fender | scalex000: http://en.wikipedia.org/wiki/H.323 |
17:09.58 | scalex000 | TK: thanks, and document of h323.conf |
17:10.40 | [TK]D-Fender | scalex000: Go read the sample configs |
17:10.44 | tzafrir_laptop | h323.conf.sample ? |
17:11.10 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
17:13.09 | *** join/#asterisk dandate2 (n=gtejkgjk@112.202.55.116) |
17:13.39 | dandate2 | http://pastebin.ca/1576719 ext 218 is broken and cannot receive calls from queue or dial out, attemps to reach her ext-ext result in busy signal. any ideaS? |
17:15.06 | [TK]D-Fender | dandate2: Show a failed call. |
17:15.09 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
17:15.35 | dandate2 | crap man we have 10 callers waiting in queue getting told their hold position... i'll try |
17:15.45 | dandate2 | with ringall... |
17:16.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:16.07 | [TK]D-Fender | dandate2: Maybe you should should show the QUEUE STATUS while you're at it. |
17:16.52 | jgoo | hey guys - SPA400's -- I have 3 of them - with 2 the config works, when i setup a third as a trunk, it doesn't load the configuration - why?!! is there some limit? |
17:17.04 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
17:17.09 | superbeef | I have 2 asterisk boxes linked by a T1 cross over to their sangoma cards...... In a very generic sense, what would I need to do in order to have 1 box ring hte other via the T1s |
17:18.00 | [TK]D-Fender | SuPrSluG: Dial(DAHDI/G1/1234567890) |
17:18.15 | [TK]D-Fender | superbeef: Dial(DAHDI/G1/1234567890) |
17:18.33 | [TK]D-Fender | superbeef: Assuming you're running DAHDI and grouped your channels as group=1 |
17:18.48 | [TK]D-Fender | superbeef: And taht was actually rather specific... |
17:19.09 | superbeef | [TK]D-Fender: Killer... Yeah i'm using dahdi, and i think everthign is group1..... much simpler than I expected |
17:19.10 | [TK]D-Fender | superbeef: hard to believe you have 2 probably rather expensive cards and don't know how to make a basic dial command for them... |
17:19.25 | jgoo | [TK]D-Fender: why would you think that? |
17:19.46 | [TK]D-Fender | jgoo: who spends thousands without researching the basics? |
17:19.47 | jgoo | hard to believe that the more expensive a device is, the easier it is to use, or you are expected to already know about it |
17:19.58 | superbeef | [TK]D-Fender: haha.... I have this mental block in my head that thigns are different because I'm not tied to a telco on the other side |
17:20.07 | [TK]D-Fender | jgoo: Its called getting a clue about what you are getting yourself into.. |
17:20.20 | jgoo | [TK]D-Fender: you have a point there - but sometimes knowing it can do something, and being able to make a path of actions to configure it are two different things |
17:20.25 | [TK]D-Fender | superbeef: And out of curiousity... why are you linking 2 *'s via T1? |
17:20.34 | superbeef | [TK]D-Fender: performance |
17:20.36 | superbeef | [TK]D-Fender: just kdiding |
17:20.54 | [TK]D-Fender | superbeef: Migration testing I could understand... |
17:20.56 | superbeef | [TK]D-Fender: I have a lot of these guys in production, but I wanted to build a real test environment |
17:21.11 | [TK]D-Fender | SuperbTest envirnment.. sure... why not.. |
17:21.17 | dandate2 | http://pastebin.ca/1576732 |
17:21.22 | jgoo | superbeef: you want to simulate incoming calls into one of the boxes? |
17:21.31 | dandate2 | http://pastebin.ca/1576738 |
17:21.42 | dandate2 | queue 292 fails over to queue 290 |
17:21.57 | [TK]D-Fender | dandate2: Where is your peer dump? SIP debug? Queue status dump? |
17:22.12 | superbeef | jgoo: yep... i want to test them to as real world as i can get... I got burned last week deploying a box with 2 T1's cuz I couldnt test before hand |
17:22.17 | dandate2 | i never used that stuff before honestly |
17:22.31 | dandate2 | i'm just gettin stuck on a problem in a very practical setup heh |
17:22.48 | [TK]D-Fender | dandate2: BS.. you've been run through this before... "sip show peer 218" |
17:23.05 | dandate2 | i think it is caused by agent 218 being a member of both queues and tried to be rang by both at once |
17:23.52 | dandate2 | http://pastebin.ca/1576743 |
17:24.06 | [TK]D-Fender | dandate2: For a person who isn't looking, now claiming not to know how to, you sure are "thinking" a lot... |
17:24.34 | [TK]D-Fender | dandate2: New call with SIP debug enabled |
17:25.46 | dandate2 | hmm my cli says that sip debug is deprecated |
17:25.58 | Naikrovek | it also says what to use instead |
17:26.13 | Naikrovek | core sip debug <ip address> if i recall |
17:26.34 | wcselby | sip set debug peer <peer> or sip set debug ip <ipaddy> |
17:26.41 | dandate2 | got it |
17:26.42 | Naikrovek | thanks |
17:26.42 | jgoo | [TK]D-Fender: have you heard or read anything about asterisk only supporting two SPA400's as trunks? not that they are officially suppoted, but damn that 9000 is only useful as a DHCP server when doing linux installs |
17:27.09 | Gumug | how powerful a machine do i need to handle 20+ calls? |
17:27.17 | Naikrovek | Gumug: not too powerful at all |
17:27.39 | [TK]D-Fender | jgoo: PARDON? |
17:27.40 | Naikrovek | there's a voip-info page on performance somewhere |
17:27.40 | Naikrovek | let me see if i can dig it up |
17:27.40 | jgoo | I just cannot understand why my setup works... until i add a third trunk - then it doesn't even show the PBX config in freepbx =( (I know you despise that!) |
17:27.43 | Naikrovek | Gumug: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
17:27.43 | Gumug | cool, i can find it |
17:27.46 | [TK]D-Fender | jgoo: "supporting as trunks"... |
17:28.00 | jgoo | as in - using the spa400 as a trunk |
17:28.04 | [TK]D-Fender | jgoo: wrong channel <- |
17:28.19 | jgoo | having 4 channels on each trunk, with spa400s |
17:28.25 | jgoo | [TK]D-Fender: how is that? |
17:28.43 | [TK]D-Fender | 13:27]<jgoo>I just cannot understand why my setup works... until i add a third trunk - then it doesn't even show the PBX config in freepbx =( (I know you despise that!) <- because we don't care what freePBX shows |
17:28.59 | [TK]D-Fender | jgoo: Got something real to show us? |
17:29.09 | jgoo | I don't think this is a freepbx bug, I wonder if anyone is using SPA400's in this capacity, they are damn useful |
17:29.14 | [TK]D-Fender | Gumug: Any P4 will do |
17:29.20 | Katty | Everyone's playing nice still, Right? |
17:29.21 | jgoo | [TK]D-Fender: that is the problem, no error messages, nothing in /asterisk/full |
17:29.24 | Gumug | alrighty |
17:29.28 | jgoo | and nothing on over 900 -vvvvvvv |
17:29.38 | [TK]D-Fender | jgoo: No problem? Then what are you going on about? |
17:29.55 | jgoo | [TK]D-Fender: where did I say no problem? |
17:30.03 | jgoo | there is no error message |
17:30.08 | Katty | sighs. |
17:30.11 | dandate2 | http://pastebin.ca/1576747 http://pastebin.ca/1576750 about as much as i could humanly grab without finding the putty log |
17:30.12 | [TK]D-Fender | jgoo: Where is the failed call to look at? |
17:30.36 | jgoo | the problem is, the configure isn't loading - and I am trying to find where it might dump the errors - or something is going wrong after I add a third trunk, and I cannot get it to reload after that... |
17:30.42 | jgoo | [TK]D-Fender: all calls work |
17:30.53 | [TK]D-Fender | dandate2: That is QUALIFY packets, not a CALL |
17:31.08 | Gumug | my fear is i won't have enough bandwidth at the asterisk server |
17:31.09 | jgoo | [TK]D-Fender: asterisk won't load config after adding a third trunk - ergo, I cannot test a call, because calls don't work, because config isn't loading |
17:31.14 | Gumug | what should i do about that? |
17:31.26 | Naikrovek | Gumug: you'll have enough. what kind of bandwidth do the offices have |
17:31.29 | Katty | Gumug: well. you could get more bandwidth. |
17:31.30 | [TK]D-Fender | Gumug: Zantac and/or vallium |
17:31.39 | jgoo | [TK]D-Fender: I am actually pretty good at debugging (or bad, depending on who I am talking to and what about) - if I know where the error messages are, or if google does |
17:31.44 | Katty | Gumug: Also, i know bandwidth.com estimates how much bandwidth you will need per phone call. |
17:31.49 | Gumug | ok |
17:31.52 | Katty | Gumug: i don't know what it is off the top of my head anymore. |
17:31.54 | Gumug | we have DSL |
17:31.58 | Gumug | maybe 6meg plan |
17:31.59 | [TK]D-Fender | jgoo: You don't seem to be able to show a call that has "failed". |
17:32.00 | Naikrovek | Gumug: plenty |
17:32.06 | Katty | Gumug: obviously a lot of it depends on codecs. |
17:32.11 | [TK]D-Fender | jgoo: * CLI has everything we care about. Log files are meaningless. |
17:32.12 | Gumug | yes |
17:32.17 | jgoo | [TK]D-Fender: you don't seem to be able to read where I just said the calls aren't failing |
17:32.20 | Katty | Gumug: the nicer the codec, the bigger the stream. |
17:32.25 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
17:32.33 | jgoo | [TK]D-Fender: the problem is, after I put up a third trunk, devices won't resgiter |
17:32.33 | Naikrovek | Gumug: yes but 6mbit is enough for about 40 G711 calls. about 320 G729 calls. |
17:32.36 | jgoo | *register |
17:32.39 | [TK]D-Fender | Gumug: dsl is fine for your satelite locations. your main will need something substantially larger |
17:32.55 | [TK]D-Fender | jgoo: If the device won't register.. thats its problem... |
17:32.55 | Gumug | that's what i figured [TK]D-Fender |
17:32.57 | Katty | Gumug: usually the VoIP providers will have some sort of formula for you to follow, if you want to ask them. Just remember you will need the same speeds both up and down. |
17:33.03 | [TK]D-Fender | jgoo: Isn't anything on *'s side |
17:33.07 | jgoo | [TK]D-Fender: I was quite clear -- anyway, no worries, if you don't want to help, but please don't say I wasn't clear, I appreciate your help for myself and others in here |
17:33.30 | [TK]D-Fender | jgoo: Seems clearer now... SPA-side config (potential) issues... |
17:33.39 | [TK]D-Fender | jgoo: Plenty of reasons we don't recommend that model. |
17:33.42 | Katty | Gumug: if your Smaller Offices only have a handful of lines, I would say DSL will be perfectly fine. |
17:34.01 | jgoo | [TK]D-Fender: ok, I did try /sbin/service asterisk stop \ asterisk -vvvvvc - it loads fine - then at one point I was getting 'previous reload hasn't finished' but i googled that to a few dead ends and other problems |
17:34.05 | Katty | Gumug: well, depending on your upload speed. |
17:34.05 | Gumug | ya they only have up to 2 lines |
17:34.09 | Katty | Gumug: you can't forget about that. |
17:34.14 | Gumug | oh, i won't |
17:34.18 | [TK]D-Fender | jgoo: if the dev doesn't reg, thats not *'s fault |
17:34.20 | jgoo | [TK]D-Fender: is there another similar device to that? |
17:34.45 | [TK]D-Fender | jgoo: Tons. Audiocodes / Mediatrix start at 4-port and up |
17:34.58 | jgoo | [TK]D-Fender: they do reg, right before I configure another trunk, then all 50 devices stop registering, because I think the config is getting blocked while loading perhaps... I am not sure... no errors as I've said |
17:35.00 | Gumug | wish i could just have the asterisk server CALL my analog phones |
17:35.17 | Katty | Well, asterisk can call analog phones...there's a card for that. |
17:35.17 | jgoo | Audiocodes / Mediatrix run on 12v? |
17:35.25 | [TK]D-Fender | Gumug: ATA <- |
17:35.30 | dandate2 | ok complete from the putty log http://pastebin.ca/1576759 |
17:35.38 | [TK]D-Fender | jgoo: dunno what alternative power options they support |
17:35.40 | scalex000 | TK: I need a command on h323 to connect both pbx |
17:35.41 | Gumug | hmmmm |
17:35.42 | [TK]D-Fender | ~ata |
17:35.43 | infobot | from memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
17:35.57 | [TK]D-Fender | Gumug: Linksys SPA-2102 for example |
17:36.15 | Gumug | some ata's dont' work with others |
17:36.22 | Gumug | one company said it would not work if i wanted to use an ATA |
17:36.34 | [TK]D-Fender | Gumug: "it"? what is "it"? |
17:36.45 | Gumug | the linksys spa |
17:36.46 | [TK]D-Fender | Gumug: And who are "they"? |
17:36.50 | Gumug | finality |
17:36.58 | [TK]D-Fender | Gumug: BULLSHIT |
17:37.00 | Gumug | i wanted to know if i could keep my analogs |
17:37.03 | Katty | [TK]D-Fender: if you don't be nice, i'm going to staple a smiley face to your forehead. |
17:37.03 | bmoraca | fonality are a bunch of idiots |
17:37.11 | Gumug | they wanted to sell me 16 phones for $245 a peice |
17:37.16 | Katty | Gumug: well of course they do. |
17:37.20 | Gumug | lol |
17:37.22 | Katty | Gumug: they want to make money. |
17:37.26 | [TK]D-Fender | Gumug: Fonality are picky bastards about what they want to support regardless of what works |
17:37.26 | Gumug | of course |
17:37.33 | Gumug | i see |
17:37.34 | Katty | Gumug: most likely they will not be able to Support it, if you do it. |
17:37.36 | bmoraca | Gumug: what kind of phones do you currently have? |
17:37.37 | sfire | is trixbox pretty stable?? I just had a bad experience with asterisknow and I need to change it and get the phone system back working |
17:37.40 | *** join/#asterisk momelod (n=smelo@99.231.22.104) |
17:37.44 | momelod | greetings channel |
17:37.47 | Gumug | 4 line analogs |
17:37.56 | momelod | does anyone know of a command-line sip client? |
17:37.56 | Gumug | i don't know the type |
17:37.59 | bmoraca | Gumug: are they really analog or are the digital? |
17:38.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
17:38.02 | Gumug | i'm not t an office |
17:38.14 | Gumug | bmoraca: good question, i don't know |
17:38.16 | Gumug | i'll find out |
17:38.17 | [TK]D-Fender | Gumug: Call them |
17:38.24 | bmoraca | that's a necessary bit of info... |
17:38.36 | Gumug | [TK]D-Fender: will do, i'm currently in a web demo |
17:38.36 | Katty | places a bet they're toshiba phones. |
17:38.42 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
17:38.44 | superbeef | momelod: for testing, or are you actually want to do voice with it |
17:38.49 | hesco | !paste |
17:39.00 | voipmonk | toshiba |
17:39.02 | hesco | ~paste |
17:39.03 | infobot | hmm... paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
17:39.03 | voipmonk | good lord |
17:39.05 | *** join/#asterisk grEvenX (n=even@193.90.141.183) |
17:39.28 | voipmonk | katty - have u ever configured the toshiba pbx (cant remember the model name ) to do sip and have it work? |
17:39.30 | superbeef | ~taco |
17:39.31 | infobot | TACO TACO TACO! |
17:39.35 | Katty | yes. |
17:39.36 | bmoraca | if they're digital, you may be able to get a Portico TVA, which is a media gateway that emulates for certain digital sets...if they really are analog, any analog media gateway (mediatrix, etc) should work. |
17:39.42 | Katty | StrataNet. |
17:39.46 | Katty | and QSIG |
17:40.05 | Katty | not sure if it was actually sip tho ^_- |
17:40.07 | [TK]D-Fender | Note : Potico = F-ing expensive and only rarely even worth considering |
17:40.11 | momelod | superbeef: well im setting up a paging system and thought i could use the company music server. So i want to install a cli sip client on the music server and set it to auto answer.. then when people talk it just plays over the overhead speakers |
17:40.43 | voipmonk | Stratanet |
17:40.56 | bmoraca | [TK]D-Fender: no doubt...but the fact remains that if your building is wired for digital phones and you have hundreds of them already, they are per-port cheaper than replacing the phones and reterminating the cables (provided it wasn't wired CAT3) |
17:41.02 | Katty | you know, Stratanet...multiple boxes. |
17:41.05 | Gumug | bmoraca: thank you |
17:41.07 | Katty | multiple locations. |
17:41.25 | Katty | i know they were voip phones. |
17:41.29 | wcselby | blast it all |
17:41.32 | Katty | but...yeah. idk if it was sip or not |
17:41.43 | wcselby | i just finally got my sip debug cleaned up for my polycom 6000, and it's over 5000 lines on pastebin |
17:42.28 | wcselby | i don't even want to read through all of that |
17:42.58 | [TK]D-Fender | bmoraca: $125 / port to maintain old crap... hard sell. |
17:43.26 | bmoraca | if it's the difference between getting a job and not getting it, i'd sell a company a box of crap and tell them it's a "line conditioner" |
17:43.37 | [TK]D-Fender | bmoraca: You can get cheaper phones with passthrough ports, but yes there are cases where its worth it.... jsut rather rare |
17:43.37 | Katty | where is dave today. |
17:43.40 | hesco | Morning all: Last night the phone was working fine. When I got up this morning I was unable to make any outbound phone calls. Any ideas why? *CLI> output (verbose 3) and iax2 show (relevant) peer are posted at: http://bin.cakephp.org/view/1686740662 |
17:43.49 | [TK]D-Fender | bmoraca: "shit" is a condition ;) |
17:44.41 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
17:45.18 | Katty | grooves. |
17:45.59 | *** join/#asterisk errotan (n=errotan@62.201.122.123) |
17:47.19 | bmoraca | granted, i'm not a big fan of using media gateways and stuff with Asterisk, but sometimes it's a necessity. |
17:48.37 | bmoraca | in 1.6, does dbsecret work for SIP peers? |
17:49.03 | *** join/#asterisk casnik (n=Nick@fw1-e0-2.dth.xiocom.net) |
17:49.23 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
17:49.54 | ehsjoar | hesco: ping the IAX provider. It seems you have a problem reaching the IAX provider. So it could be a problem on your network or the IAX provider having issues |
17:51.19 | superbeef | momelod: so your music server would be a client to your asterisk box? |
17:52.21 | hesco | ehsjoar, I finally did a `stop now`, followed by an `asterisk -vvvgc` and it restored services. |
17:52.33 | ehsjoar | hesco: Okay, so it is working now? |
17:52.40 | hesco | can't imagine what might have changed to create that issue. |
17:52.58 | hesco | yes, at least I was able to reach my cell's voicemail |
17:53.02 | momelod | superbeef: exactly |
17:53.32 | superbeef | momelod: how will the SIP client interract with your music server |
17:53.45 | superbeef | momelod: the Music server is just hooked up to some sort of PA system right? |
17:53.53 | ehsjoar | hesco: Seems like for some reason * was not able to reach the IAX provider (or thought so). Strange that a restart would fix it. Perhaps there is a NAT problem somehow |
17:55.42 | *** join/#asterisk tlarsen (n=tlarsen@71.207.223.160) |
17:56.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
17:56.41 | Katty | ehsjoar: let us rejoice and have ice cream that your issue is fixed! |
17:57.23 | ehsjoar | Katty: It was not my issue, but hesco's. Sounds good with the ice cream though:-) |
17:57.24 | Katty | ehsjoar: should it happen again, i'm sure the provider can watch the packets that are going back and forth. |
17:57.57 | Katty | that's the spirit! |
17:58.51 | ehsjoar | Katty, hesco: Yeah, I am thinking it could be a firewall / NAT problem hesco is facing. The firewall perhaps keep the port open for a while but closes it if there is no traffic. Pure speculation on my side though |
17:59.06 | momelod | superbeef: actually the music server is mpd and i control it using the mpc client |
17:59.38 | momelod | actually i wrote an app for my cisco7960 handsets where my users can control the overhead speakers using the service button on the phone. |
18:00.13 | *** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net) |
18:00.21 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
18:00.27 | superbeef | momelod: see if pjsip does what you need |
18:01.58 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
18:07.51 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
18:08.30 | *** join/#asterisk Greek-Boy (n=greek@41.188.154.137) |
18:10.47 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
18:15.21 | bmoraca | i don't think it's possible for Valcom's website to be any more worthless than it is |
18:20.00 | *** join/#asterisk Gumug (n=Gumug@nmd.sbx09566.joplimo.wayport.net) |
18:20.22 | *** join/#asterisk ingenius (n=alektro@81-152-114-200.fibertel.com.ar) |
18:24.47 | [TK]D-Fender | bmoraca: there is very little in life that's so bad that it can't get worse. |
18:24.48 | bmoraca | DUNDi does not have a lot of informational elements to it in the CLI, does it? one would think that it would be useful to be able to output the entire DUNDi route table, but apparently one cannot |
18:25.12 | [TK]D-Fender | bmoraca: Just when you think you've hit rock bottom, that's why caterpillar makes back-hoes ;) |
18:25.18 | bmoraca | lol |
18:25.53 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
18:27.57 | kaldemar | bmoraca: it's in the database, iirc |
18:29.02 | Katty | i have a weird dns issue i can't wrap my brain around. |
18:29.26 | Katty | it makes my voicemail attachments spaz out when sending to user@domain |
18:29.32 | Katty | said domain is in the same building. on the same lan. |
18:29.37 | Katty | but it doesn't like domain.com |
18:29.40 | *** join/#asterisk _brent_ (n=_brent_@orem.jiveip.net) |
18:30.04 | Katty | we can't open domain.com in a browser either. |
18:30.04 | Katty | which might be normal. idk. don't know a lot about dns. |
18:30.34 | Katty | we own several domains which all point back to domain.com, so i just made them go to another alias. works fine. |
18:30.39 | Katty | figure that one out ^_- |
18:31.41 | Katty | exim4's mainlog is talking about MX records. |
18:32.06 | Katty | doublechecks dns server for mx record for email server |
18:32.54 | bmoraca | kaldemar: which database? it's not in astdb |
18:34.59 | kaldemar | bmoraca: astdb is what i meant. i recall checking the cache out somehow. |
18:35.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:37.32 | dustybin | [TK]D-Fender: its now fully complete: http://www.thinkdebian.org/archives/828 |
18:38.05 | dustybin | takes cover from the ClueBat |
18:40.13 | [TK]D-Fender | dustybin: [general] of IAX2 should point to a dead-end context. Phones should always explicitly refer to their context, and you should NEVER make a context named [default] |
18:40.31 | dustybin | makes note |
18:40.37 | Naikrovek | dustybin: you make that? |
18:40.44 | [TK]D-Fender | dustybin: exten => _09XX,1,Dial(IAX2/USERID@voiptalk/${EXTEN}) <-- not abstracted |
18:41.16 | [TK]D-Fender | dustybin: In your IVR, at the end of the menu audio it will HANGUP. |
18:41.29 | [TK]D-Fender | dustybin: And an invalid entry will also hangup on them |
18:41.29 | dustybin | eeek |
18:41.55 | dustybin | [FAILS] |
18:42.04 | [TK]D-Fender | dustybin: exten => 1,1,VoiceMail(1000) <- yous hould also always explicitly include the VM context to use, and you might want to consider playing a PROMPT |
18:42.15 | dustybin | double [FAILS] |
18:42.24 | Naikrovek | dustybin: i followed the compiling instructions earlier, they worked great; but you missed one little thing. this is not a fail, btw |
18:43.11 | dustybin | Naikrovek: what did i miss? |
18:43.55 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
18:43.56 | Naikrovek | dustybin: guide is awesome. i had to do a "apt-get install libncurses5-dev" when i installed build-essentials and subversion and the kernel headers. |
18:44.30 | Naikrovek | you could also change "aptitude install linux-headers-2.6.X-X-common |
18:44.30 | Naikrovek | " to "aptitude install linux-headers-`uname -r`" (those are backticks, not single quotes) |
18:45.26 | dustybin | right ok!! |
18:45.38 | dustybin | thanks!! |
18:45.56 | Naikrovek | effing awesome guide tho, i found it earlier via google i think and bookmarked it not knowing it was you :) |
18:46.12 | dustybin | its on google already? |
18:46.23 | Naikrovek | i dunno, but it's been in my browser all damn day |
18:46.30 | dustybin | well , you have to thank everybody in this channel, thats where i got most of the info from |
18:46.36 | Naikrovek | and i don't think i clicked a link in here |
18:47.09 | dustybin | im just saving the headache of finding bits and bobs from all over the place |
18:47.10 | Naikrovek | yeah it's on google |
18:47.10 | dustybin | ok |
18:47.14 | Naikrovek | second page whe i search for "debian asterisk howto" |
18:47.24 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:c42:5117:b2c9:8871) |
18:47.25 | cusco | hi |
18:47.30 | cusco | I got a problem with dahdi |
18:47.44 | cusco | <PROTECTED> |
18:47.51 | dustybin | cusco: have you intalled mhomi ? |
18:47.51 | cusco | Unloading DAHDI hardware modules: |
18:48.10 | cusco | no what is that? |
18:48.15 | dustybin | :P |
18:48.21 | cusco | ERROR: Removing 'wcte12xp': Device or resource busy |
18:48.22 | dustybin | stops trolling |
18:49.10 | Naikrovek | dustybin: would have been better if you'd said "have you been touching mohmi?" |
18:49.13 | Qwell | cusco: Did you stop Asterisk first? |
18:49.20 | cusco | yes |
18:49.47 | tzafrir_laptop | cusco, it hangs for a while or fails immediately? |
18:49.52 | SuPrSluG | lsmod |grep dahdi to see what's using it |
18:49.57 | tzafrir_laptop | anything ugly in /var/log/messages ? |
18:51.02 | Naikrovek | mmm, snack pack. cures heartburn nicely for me |
18:51.24 | Orbixx | How can I record a call that takes place after a client has queued? |
18:51.41 | *** join/#asterisk voipmonk (n=voipmonk@69.172.93.45) |
18:51.41 | Naikrovek | Orbixx: just always record all calls |
18:51.47 | Naikrovek | ... i think |
18:51.57 | Orbixx | How can that be done? :P |
18:51.58 | Naikrovek | don't think you can turn recording on mid call |
18:52.00 | Naikrovek | but i could be wrong |
18:53.02 | Naikrovek | one sec, Orbixx |
18:53.13 | dustybin | Naikrovek: i have fixed the install parts, now for the tricky bits |
18:53.22 | Naikrovek | cool beans |
18:53.34 | [TK]D-Fender | Orbixx: "core show application monitor" |
18:53.47 | dustybin | 1 [general] of IAX2 should point to a dead-end context. |
18:53.51 | kaldemar | Naikrovek: yes you can |
18:53.58 | Naikrovek | [TK]D-Fender: can he turn that on when the call changes contexts |
18:54.16 | Naikrovek | like when the call enters a context, and that context has recording set to always, will that start recording? |
18:54.17 | [TK]D-Fender | Naikrovek: ... calls... change... CONTEXTS? |
18:54.22 | Naikrovek | i don't know |
18:54.44 | Naikrovek | you pass a call to a queue, doesn't that queue have a context? |
18:54.46 | kaldemar | Dial options wWxX and featuremaps in features.conf |
18:55.20 | *** join/#asterisk TimToady_ (n=moi@adsl99-59.kln.forthnet.gr) |
18:55.28 | kaldemar | a context doesn't have anything, an extension may have. |
18:55.44 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
18:55.58 | [TK]D-Fender | Naikrovek: you = very confused |
18:56.10 | Naikrovek | [TK]D-Fender: on a few things, yes |
18:56.16 | Naikrovek | i'm no dial plan master |
18:56.28 | [TK]D-Fender | Naikrovek: unfortunate since that's 95% of * :p |
18:56.37 | Naikrovek | well i'm working on it |
18:56.43 | Naikrovek | got a lot of S going down at work |
18:57.24 | Orbixx | After a call ends, can Asterisk email the recording similarly to that of a voicemail? |
18:58.11 | [TK]D-Fender | Orbixx: go read up on Asterisk Standard Extensions |
18:59.08 | Orbixx | [TK]D-Fender: Right, action stuff on the hangup extension. |
18:59.18 | jaytee | every time I call my girlfriend from my phone in the [normal] context the call works fine but when I use a feature key to change to the [porn] context I get disconnected. She swears she isn't hanging up on me. sip debug shows nothing unusual |
18:59.23 | Orbixx | I'm just asking if it's something that Asterisk can do relatively automatically much like voicemail.conf |
19:00.02 | [TK]D-Fender | Orbixx: "core show application monitor" <- |
19:00.41 | Orbixx | I don't see anything relevant to the question I'm asking. |
19:01.12 | IBC_jkenney | I have new questions today |
19:01.29 | IBC_jkenney | I want to be able to have in the cdr records better detailed such as who hungup first |
19:02.12 | IBC_jkenney | where the call went IE it came in on DID XXXX went to main menu s 1 then they pressed 1 then it went to the sales queue thenf rom the queue it went to agent blah |
19:02.19 | IBC_jkenney | is this hard already there |
19:02.41 | [TK]D-Fender | Orbixx: Read it AGAIN |
19:03.02 | Orbixx | [TK]D-Fender: I've read it three times, I'm obviously missing something. |
19:03.09 | _brent_ | IBC_jkenney: the second part is possible if you keep track of the traversal yourself (assign it to a variable) and then put it in the CDR with cdr_custom |
19:03.18 | [TK]D-Fender | Orbixx: "m" |
19:03.35 | _brent_ | knowing who hung up might be tougher. i'd love to hear it if someone knows the answer to that part. |
19:04.10 | Orbixx | [TK]D-Fender: That just mixes the two recordings. I'm asking if Asterisk provides capability to automatically mail the recording to a mailbox much like voicemail does. |
19:04.14 | *** join/#asterisk Methose (n=Methose@38.101.237.250) |
19:04.15 | [TK]D-Fender | Orbixx: Read it AGAIN |
19:05.37 | Orbixx | I see nothing relevant to my question. |
19:05.37 | [TK]D-Fender | IBC_jkenney: the queue log already tells you who hung up |
19:06.14 | [TK]D-Fender | Orbixx: "If the variable MONITOR_EXEC is set, the application referenced in it will be executed instead of soxmix and the raw leg files will NOT be deleted automatically." |
19:06.57 | Orbixx | Right, so you're implying that Asterisk doesn't do it automatically like voicemail and I need to invoke something like "mail". |
19:07.09 | [TK]D-Fender | Orbixx: WOW. |
19:07.35 | [TK]D-Fender | Orbixx: the mere fact VM has this functionaliy semi-built-in alone is rather generous |
19:07.50 | [TK]D-Fender | Orbixx: * is up to you to configure. |
19:08.26 | Orbixx | The mere fact that Asterisk exists is generous, but I don't see why I'm being unreasonable if Voicemail does something almost exactly the same. |
19:08.32 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
19:08.39 | Orbixx | I agree, but I was asking a simple question. |
19:08.49 | Orbixx | I wasn't saying it was outrageous that it doesn't have such capability. |
19:09.04 | Orbixx | I just wanted to *know* for sure. |
19:09.22 | [TK]D-Fender | Orbixx: call recording doesn't have a config file to define which calls to record, where to send, and how to manage. |
19:09.40 | Orbixx | Thank you. |
19:09.43 | Orbixx | That's all I wanted to know. |
19:10.14 | Orbixx | For somebody who is ultimately very helpful, it's very difficult to pry the answers out of you sometimes. |
19:10.21 | Orbixx | ;) |
19:10.48 | [TK]D-Fender | Orbixx: I handed it to you multiple times. you simply could see the only option on that page which clealy let you do something on end of recording. |
19:11.00 | [TK]D-Fender | Orbixx: Don't blame me for your lack of creativity :) |
19:11.15 | jaytee | some people want their poop to come out gift wrapped it seems |
19:11.27 | [TK]D-Fender | Orbixx: You walked into this expecting a "yes" instead of looking at what the pieces already do |
19:11.29 | Orbixx | I already realised it was there. |
19:11.34 | Orbixx | And I already knew I could do it manually. |
19:11.49 | Orbixx | I just wanted to know if Asterisk could do what I could do manually - automatically, much like Voicemail. |
19:12.18 | |Rain| | so. Mr. A (100) calls Mr. B (200), they chitchat for a bit, then Mr. B (200) performs an attended transfer of Mr. A (100) to Mr. C (300). CDRs are logged for 100->200 and 200->300, but is there any way to determine that 100 talked to 300 from CDRs? |
19:12.24 | Orbixx | I don't think I'm being unreasonable? |
19:12.29 | [TK]D-Fender | Orbixx: Nope, for all the reasons that are evident by the existance and content of voicemail.conf |
19:12.43 | jaytee | |Rain|, what is that? An SAT question? |
19:13.01 | [TK]D-Fender | |nope |
19:13.08 | [TK]D-Fender | |Rain|: nope |
19:13.08 | |Rain| | I hope not, because I fail otherwise |
19:13.12 | |Rain| | [TK]D-Fender: boo |
19:13.19 | leifmadsen | there probably is with CEL |
19:13.26 | leifmadsen | but not with CDRs afaict |
19:13.52 | dustybin | [TK]D-Fender: why should this be: 1 [general] of IAX2 should point to a dead-end context. should i just make up anything and stick it in the [general] like, context=cluebat ? |
19:13.56 | jaytee | CEL? typo? for AEL? |
19:13.58 | *** join/#asterisk el_critter (n=critter@200.8.97.41) |
19:14.04 | leifmadsen | jaytee: no, not a typo |
19:14.11 | jaytee | hmmm |
19:14.17 | leifmadsen | Channel Event Logging |
19:14.17 | jaytee | goes to Google CEL |
19:14.19 | leifmadsen | ~cel |
19:14.19 | infobot | somebody said cel was Channel Event Logging |
19:14.19 | |Rain| | apparently Channel Event Logging |
19:14.20 | [TK]D-Fender | dustybin: Sure |
19:14.28 | dustybin | [TK]D-Fender: is this for security? |
19:14.33 | [TK]D-Fender | dustybin: Yes |
19:14.36 | dustybin | aye ok :D |
19:14.45 | el_critter | Anyone having problems with asterisk hanging the call but dahdi keeping the PSTN line open? |
19:14.55 | [TK]D-Fender | dustybin: So taht un-authed calls don't get unrestricted access to call out... like they do NOW |
19:15.49 | dustybin | right ok |
19:16.02 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
19:16.20 | nullable_type | How can i play a music file in the background while doing a DIAL to connect two calls |
19:16.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:16.37 | leifmadsen | Dial() with the m flag |
19:16.38 | [TK]D-Fender | nullable_type: "core show application dial" |
19:17.00 | nullable_type | thanks guys |
19:17.08 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
19:17.13 | |Rain| | [TK]D-Fender: thanks |
19:17.18 | *** part/#asterisk |Rain| (i=rain@ev.il.net) |
19:18.35 | dustybin | 2 Phones should always explicitly refer to their context, and you should NEVER make a context named [default] |
19:18.54 | dustybin | whats wrong with [default] ? im pretty sure [default] was used in the book |
19:19.16 | nullable_type | RE:DIAL, so the m option does it repeat the music file, I do not want that...... Also when i want the music to stop when the first leg can hear ring tone.... |
19:19.19 | nullable_type | Is it possible |
19:20.32 | nullable_type | may be I should use the A(x) option for DIAL? |
19:20.44 | [TK]D-Fender | dustybin: Yes, and books can have mistakes as well... |
19:21.09 | [TK]D-Fender | nullable_type: that you cannot do. |
19:21.34 | [TK]D-Fender | nullable_type: there is no pre-ring/post-ring split |
19:21.57 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
19:22.03 | dustybin | aye ok! |
19:22.08 | nullable_type | D-Fender, But does the m option loops the file or just play once |
19:22.15 | nullable_type | I want to avoid loop |
19:24.07 | wcselby | here's a new one |
19:24.17 | [TK]D-Fender | nullable_type: make it one long recording. |
19:24.19 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
19:25.03 | geneticx | Hi you all, we pay for 3 sip trunks how can I configure asterisk so that when those 3 trunks are being used to give a busy signal instead of rejecting the call..? |
19:25.29 | [TK]D-Fender | geneticx: congestion / busy <- |
19:25.34 | [TK]D-Fender | geneticx: AFTER answering |
19:27.07 | wcselby | here's a new one.....when I ping my polycom 6000 from my asterisk server from the time it boots up, it will stay registered to asterisk (and calls will work etc) for as long as I continue to ping the phone. Once I stop pinging the phone, the phone stops talking to the asterisk server, and then I can no longer ping it. |
19:27.35 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
19:28.03 | Naikrovek | wcselby: i'm starting to think switch now |
19:28.21 | jaytee | really? I'm thinking Arby's |
19:28.25 | Naikrovek | network switch problem maybe |
19:28.30 | Naikrovek | mmm arbys |
19:28.35 | Naikrovek | beef-n-cheddar ftw |
19:28.41 | wcselby | Naikrovek - every other phone we have, polycom IP 601's, cisco 7960's and 7961's, and softphones, all work over the same network, same port, etc |
19:29.03 | wcselby | but that doesn't mean it isn't a switch |
19:29.04 | jaytee | there's just something about their genetically modified vat grown roast beef that I love |
19:29.12 | wcselby | brb, helping someone in vi |
19:29.27 | dustybin | [TK]D-Fender: I have fixed it now, im pretty sure there are still errors in it, I will need to test the whole thing on my setup |
19:30.03 | Naikrovek | dustybin: you're lucky to get [TK]D-Fender's opinion on that. once he gives it his stamp of approval it'll be a really, really good, solid article |
19:30.07 | geneticx | [TK]D-Fender: Ok, so let me see if I got this correct, after my Answer () statement to have congestion /busy ? |
19:30.50 | dustybin | I will put a freenode credit at the end of the article, however, do you want to invite in extra people in this channel? |
19:31.38 | nullable_type | Hey guys where do i add an audio file to music on hold classes so i can use it in DIAL with m option? musiconhold.conf doesn't seem to have a [class] section |
19:33.00 | wcselby | nullable_type - you create your own [class] section |
19:33.22 | wcselby | so if you want a class named "companyAhold" you would create [companyAhold] and then list the specifics for that class |
19:33.47 | wcselby | but if you just want to add to your default directory |
19:34.00 | wcselby | look for the "directory" directive in your musiconhold.conf file |
19:34.11 | wcselby | on mine, it's /var/lib/asterisk/moh |
19:34.28 | nullable_type | oh cool thank you |
19:35.30 | bmoraca | mmm...kentucky fried rat... |
19:35.55 | geneticx | [TK]D-Fender: how would I type it? because congestion /busy sounds a bit too generic |
19:37.09 | bmoraca | geneticx: http://www.lmgtfy.com/?q=asterisk%20congestion |
19:38.45 | [TK]D-Fender | geneticx: Congestion() |
19:39.19 | geneticx | [TK]D-Fender: Got it, thank you. |
19:40.29 | Orbixx | Monitor() keeps exiting with non-zero, but it's been able to record before and I can't see why the channel would already be recorded. |
19:40.34 | superbeef | what's the trick to getting aterisk to compile/run with dahdi support... I built and installed dahdi, adn lib pri, and i have no DAHDI commands in the asterisk cli |
19:41.07 | [TK]D-Fender | superbeef: Did you recompile * from scratch afterwards? |
19:41.21 | superbeef | [TK]D-Fender: i feel like i have 4 times.. going to try once again |
19:41.25 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
19:41.45 | [TK]D-Fender | superbeef: Trash your source folder, and re-extract and in menuselect pay close attention |
19:42.00 | superbeef | there's a menuselect? |
19:42.11 | superbeef | i just do ./configre --with-curl then make; make install |
19:42.56 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
19:42.59 | [TK]D-Fender | superbeef: s/make/make menuselect/ |
19:43.41 | citywok | is there a way to do something like RemoveQueueMember(*) or All Queues |
19:44.20 | citywok | Pause & Unpause queue member affects all queues, but i'd like a removequeue members for all queues type function |
19:44.51 | bmoraca | how many queues do you have? |
19:44.54 | superbeef | [TK]D-Fender: it saw dahdi dependcies from menuselect.... anyway... recompiling |
19:45.10 | bmoraca | you could simply add a priority for each queue |
19:45.12 | superbeef | <PROTECTED> |
19:45.12 | superbeef | <PROTECTED> |
19:45.16 | superbeef | if that doesnt work i'm giving up |
19:45.32 | citywok | bmoraca: currently in testing 2. planning for ~100 |
19:45.50 | bmoraca | why could you possibly need that many queues? |
19:46.11 | [TK]D-Fender | bmoraca: Call center company |
19:46.16 | citywok | call center running multiple projects |
19:46.46 | citywok | but, a priority for each is a viable solution if there isn't a pre-existing function for it |
19:46.50 | [TK]D-Fender | citywok: I see a lot of value in what you'd like to do and see RQM's lack of functionaliy on this side rather disconcerting |
19:47.08 | [TK]D-Fender | citywok: it WOULD be something simple to have created... |
19:47.15 | citywok | the configs are autogenerated so i could do it simply |
19:47.27 | citywok | RQM? what does that stand for? |
19:47.33 | [TK]D-Fender | citywok: For now you'd have to make a somewhat more complex script to do this for you. |
19:47.39 | [TK]D-Fender | citywok: RemoveQueueMember |
19:47.44 | bmoraca | well, if that's the case, i suspect that you're using pins to authenticate your users...just expand that into an agi application to look and see which queues that user belongs in and use some for-next loops to take care of it |
19:47.48 | citywok | oh, har har wow i cant believe i missed that |
19:48.55 | citywok | bmoraca: yea, queue show members, all queues member of remove. or set a DB variable and add all the queues you are a member of to that |
19:49.03 | citywok | or like your first suggestion, just loop through all of them and remove you manually |
19:49.24 | citywok | [TK]D-Fender: do you think that's something i should post as a feature request on the digium site? |
19:49.32 | citywok | i've never posted a feature request, just a few bug reports |
19:49.49 | bmoraca | citywok: yes, but the second solution (agi) is more flexible...for instance, not all members may be members of all queues |
19:50.14 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
19:50.17 | citywok | bmoraca: absolutely, they would likely be a member of 2 or 3 queues max, so it would be inefficient to do 100 commands extra |
19:50.42 | *** join/#asterisk MindTheGap (n=caio@200.251.73.10) |
19:51.22 | bmoraca | citywok: it would also be inefficient to create hundreds of different login combinations based on the pins. you'll need a database backend (for external administration at the very least) and AGI would be the easiest way to interface with that for the purposes of logging in and authenticating users |
19:51.31 | *** join/#asterisk lanning (n=lanning@212.183.134.130) |
19:51.40 | [TK]D-Fender | citywok: Yes, this is worthy of a major mod |
19:52.14 | [TK]D-Fender | citywok: You should be able to remove from all/any queue any device or membername |
19:52.24 | citywok | bmoraca: the login stuff is already all handled. queues are auto-generated, i need to write a query to check for permission to log in to the queue but that's easy enough |
19:52.42 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
19:53.26 | bmoraca | i don't envy that project :P |
19:53.50 | *** join/#asterisk lanning (n=lanning@212.183.134.130) |
20:00.07 | IBC_jkenney | anyone know a good way to convert gsm to mp3 i want to convert out recordings to mp3 |
20:00.40 | wcselby | will lame input gsm? |
20:00.59 | IBC_jkenney | I belive it will |
20:01.25 | citywok | bmoraca: it's an interesting project, but we've already been using asterisk for well over a year running all of our calls (PSTN to SIP conversion on our non sip pbx) |
20:01.40 | citywok | so running all the projects out of asterisk natively isn't a HUGE step |
20:02.29 | bmoraca | gotcha |
20:02.36 | *** join/#asterisk kc8pxy (n=gecko@75-145-57-201-utah.hfc.comcastbusiness.net) |
20:03.11 | citywok | it's definitely fun. we're wrapping our pbx around our call center interfaces so that PM's can build their own phone / project queues. |
20:03.29 | jgoo | [TK]D-Fender: so - just an update - I've got it to a reproducible test case, from a clean install and setup |
20:03.30 | citywok | instead of a couple day turnaround for a project they can do it themselves in 10 minutes |
20:04.25 | jgoo | [TK]D-Fender: I have three trunks configured, none of the 50 extensions register. I disable the third trunk, not delete it, just disable, restart (powering up the 50 extensions after the restart) and all register |
20:04.27 | kc8pxy | I'm setting up an asterisk server to handle our offices calls, and i'm just getting it configged... what are the minimum config files i need for it to come up for sip channels? |
20:04.44 | jgoo | as soon as I enable the third trunk, all extensions drop... how.. |
20:04.49 | kc8pxy | config of the files, i can do, but i forget which are the minimum? |
20:05.38 | bmoraca | jgoo: that might be a question better directed to #freepbx |
20:05.38 | *** join/#asterisk l2trace99 (n=asd@75.112.140.2) |
20:06.01 | jgoo | perhaps bmoraca , maybe you are right |
20:06.03 | Katty | ATTENTION. |
20:06.06 | Katty | IT IS HUG TIME. |
20:06.08 | Katty | hugs bmoraca |
20:06.12 | jgoo | Katty: please go away |
20:06.15 | wcselby | hugs Katty |
20:06.18 | bmoraca | ouch |
20:06.20 | Katty | hugs wcselby |
20:06.26 | IBC_jkenney | hugs katty |
20:06.30 | jgoo | So much fail |
20:06.32 | wcselby | hugs the Clue Bat [tm] |
20:06.33 | Katty | we'll have none of your negativism in here, Sir! |
20:06.36 | Katty | hugs jgoo |
20:06.41 | Katty | hugs IBC_jkenney |
20:06.46 | jgoo | I think that is tantamount to rape |
20:06.47 | IBC_jkenney | again again |
20:06.48 | jgoo | and I charge for rape |
20:06.59 | Katty | you go right ahead and file a complaint. |
20:07.11 | l2trace99 | how much ? |
20:07.11 | jgoo | due to popular demand, second rapes are half price |
20:07.17 | IBC_jkenney | you can't rape the willing |
20:07.19 | IBC_jkenney | me me me |
20:07.21 | IBC_jkenney | again again |
20:07.23 | IBC_jkenney | ;) |
20:07.40 | jgoo | I know, that is why I can never score a hatrick, I usually have to take out a restraining order against the victim |
20:07.45 | jgoo | they get all clingy |
20:08.06 | *** join/#asterisk MindTheGap (n=caio@200.251.73.10) |
20:08.06 | Katty | nothing says i love you like a restraining order. |
20:08.10 | Katty | or so [TK]D-Fender says. |
20:08.11 | IBC_jkenney | I didn't know farm animals could file restraining orders |
20:08.20 | Katty | IBC_jkenney: okay, that's enough out of you. |
20:08.32 | Katty | no more negative bad thoughts! |
20:08.46 | l2trace99 | when talking about bench marking would calls be equal to channels ? |
20:08.48 | wcselby | was that a double negative Katty? |
20:08.49 | IBC_jkenney | i thought it was funny |
20:08.50 | bmoraca | farm animals...lol |
20:09.05 | Katty | IBC_jkenney: that's beside the point! :P |
20:09.28 | kc8pxy | Katty: did you intend to be redundant, or do you believe ther is such a thing as a "negative good thought" ?? |
20:09.36 | IBC_jkenney | in an attempt to make friends send jgoo a pass to the petting zoo ;) |
20:09.39 | IBC_jkenney | ok i'm done |
20:09.40 | Katty | i forgot a comma, actually. |
20:09.49 | Katty | it was supposed to be two adjetives. |
20:09.52 | IBC_jkenney | i have it out of my system now |
20:09.54 | Katty | i can't spell. |
20:10.05 | IBC_jkenney | thats ok i can't read |
20:10.06 | jgoo | kc8pxy: Katty hasn't read my thesis on patheticisms |
20:10.07 | IBC_jkenney | don't judge me |
20:10.09 | IBC_jkenney | clear |
20:10.23 | Katty | jgoo: you are correct. |
20:10.24 | kc8pxy | is there any asterisk going on here? |
20:10.25 | [TK]D-Fender | Katty: :) |
20:10.31 | Katty | kc8pxy: yes, sometimes. |
20:10.35 | IBC_jkenney | ok i need to convert gsm to mp3 a whole directory i don't want to change the file names |
20:10.38 | IBC_jkenney | just the format |
20:10.38 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
20:10.45 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
20:10.47 | IBC_jkenney | can I do it all in one klump |
20:10.48 | IBC_jkenney | ? |
20:11.02 | Katty | sox might do it |
20:11.03 | wcselby | if you write a batch file to script the process, I don't see why not |
20:11.06 | citywok | [TK]D-Fender: filed as a feature request 15947 |
20:11.21 | [TK]D-Fender | kc8pxy: I don't not never not use double negatives neither! |
20:11.39 | Katty | oh god. |
20:11.42 | Katty | that made my brain hurt. |
20:12.57 | kc8pxy | [TK]D-Fender: she didn't use a double negative. double negatives contradict. redndant is an excessive reaffirmation. :) |
20:13.13 | jgoo | [TK]D-Fender: you are bordering on being a misogynist now (if Katty is female) I say if, but the way you are all pandering to the utter fail in here is reprehensible |
20:14.03 | [TK]D-Fender | kc8pxy: yes, well there was this linguistics class in which the professor explained that unlike most languages, in Russian a double negative remains a negative, yet in no language is a double-positive a negative. |
20:14.06 | kc8pxy | jgoo: " pandering to the utter fail" ?? |
20:14.21 | [TK]D-Fender | kc8pxy: upon hearing this an attendee exclaims "Yeah right...." |
20:14.25 | jgoo | shit, we are using gender specific pronouns. yes, I am so glad you can read and quote kc8pxy |
20:14.35 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
20:14.44 | jgoo | It is embarrassing to see someone behave like that |
20:15.05 | murdock_ut | is there a way through dialplan to tell asterisk to hangup a zap/dahdi channel? |
20:15.13 | wcselby | jgoo - behaving like what? |
20:15.14 | [TK]D-Fender | jgoo: Whats wrong with accurate gender-specific pronouns? |
20:15.24 | wcselby | murdock_ut - Hangup() ? |
20:15.27 | Katty | murdock_ut: Hangup() usually |
20:15.41 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
20:15.47 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
20:15.48 | Katty | why are we having this conversation? it seems overly dramatic. |
20:15.58 | Katty | let's not have it anymore. |
20:16.09 | murdock_ut | I have a zap channel that is constantly getting stuck and I have to keep doing a soft hangup on it. |
20:16.18 | murdock_ut | It interfaces with a paging system. |
20:16.24 | [TK]D-Fender | murdock_ut: Figuring you want to trash a call for priority outbound (like 911) - System(asterisk -rx "soft hangup Zap/1-1) |
20:16.30 | [TK]D-Fender | murdock_ut: Figuring you want to trash a call for priority outbound (like 911) - System(asterisk -rx "soft hangup Zap/1-1") |
20:17.13 | murdock_ut | [TK]D-Fender: Same basic idea. |
20:17.21 | murdock_ut | I'll give that a go. |
20:18.19 | kaldemar | there's a SoftHangup application that you can use in the dialplan, no need to put it through System |
20:18.33 | jgoo | Katty: when people pander to silly little outburts, you may think it is modern and tolerant, actually it is a hate crime, it is because they think you are stupid, therefore they tolerate such ludicrous outburts. If I said, I would get kicked. Fair enough. You say it, you don't you feel good, but what it means is, they think that is what you do. Sad. |
20:18.39 | jgoo | Just a lesson for you all. |
20:19.02 | bmoraca | lol |
20:19.16 | murdock_ut | kaldemar: so there is. |
20:19.18 | murdock_ut | kaldemar: I |
20:19.23 | el_critter | murdock_ut: can you explain a little bit your problem? |
20:19.36 | murdock_ut | kaldemar: I'll look at that. |
20:19.44 | Katty | jgoo: that's nice dear. my silly little outbursts will continue. |
20:19.54 | Katty | jgoo: but feel free to put me on ignore. |
20:20.50 | [TK]D-Fender | kaldemar: Depends on which version |
20:21.02 | murdock_ut | el_critter: I have a fxs port that connections to a port on a Rauland-Borg paging system at a school. What is happening is that either asterisk or the paging system, I'm not sure which is not releasing the line after the user hangs up the phone. |
20:21.45 | el_critter | murdock_ut: what asterisk/dahdi version? |
20:22.05 | kaldemar | [TK]D-Fender: it was in 1.2 already |
20:22.47 | [TK]D-Fender | kaldemar: Ok, will look into for future reference |
20:22.49 | IBC_jkenney | <===== jgoo pissing in your cereal bowl <with love> |
20:23.00 | kaldemar | [TK]D-Fender: in 1.0 even |
20:23.16 | bmoraca | uhg, i'm never using GeoTrust for SSL certificates again |
20:23.27 | Katty | IBC_jkenney: what did we say about being so negative! |
20:23.36 | IBC_jkenney | sniff sniff |
20:23.39 | IBC_jkenney | i'm sorry |
20:23.39 | murdock_ut | kaldemar: It is an older version. I haven't got around to upgrading it. It is 1.2.22 |
20:23.45 | Katty | hugs IBC_jkenney |
20:23.53 | [TK]D-Fender | kaldemar: Wow... |
20:24.04 | [TK]D-Fender | kaldemar: Yup, goes way back... never knew |
20:24.06 | IBC_jkenney | now that is done |
20:24.20 | IBC_jkenney | lets focus on my problems beside the mental defects |
20:25.01 | IBC_jkenney | do we know of any good "free" Cheap queue monitoring software |
20:25.01 | kaldemar | [TK]D-Fender: even better, it was introduced in 0.4.0 :D |
20:25.01 | IBC_jkenney | to pull stats on agents and all that jazz |
20:25.14 | kaldemar | murdock_ut: you have the application |
20:25.27 | murdock_ut | kaldemar: ?? |
20:25.28 | kc8pxy | i have a fresh install of 1.6.2.0. what are the minimum set of config files i need to get sip channels working? i THINK i only need sip.conf and extensions.conf, and IIRC i needed modules.conf.. is that it, or do i need more? |
20:26.17 | kaldemar | murdock_ut: 1.2.22 has app SoftHangup, so no worries there. |
20:26.17 | [TK]D-Fender | kc8pxy: asterisk.conf ... rtp.conf, probably a bunch more... |
20:26.18 | murdock_ut | kaldemar: Ya. I saw that. I'm going to give that a try. Hopefully it will fix the problem until I can upgrade. |
20:28.26 | [TK]D-Fender | checkout time, BBL |
20:29.05 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
20:29.56 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
20:32.30 | tzafrir_laptop | murdock_ut, if the FXO releases the line, the FXS must notice it. The other way around is not guranteed |
20:33.40 | tzafrir_laptop | Asterisk's chan_dahdi supports disconnect notification through "kewl-start", that is power denial. But not through polarity reversal |
20:36.35 | murdock_ut | tzafrir_laptop: Ok, so the I probably have it backwards when I said I had a fxs port. It is an fxo port connecting to a fxs because the paging system is providing the dialtone. |
20:37.17 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@208.124.232.58) |
20:37.33 | darkdrgn2k3 | ok im stuck |
20:37.34 | darkdrgn2k3 | http://pastebin.ca/1577019 |
20:37.43 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.19) |
20:37.43 | darkdrgn2k3 | why am i getting "the number you have dialed is not in service" |
20:38.07 | tzafrir_laptop | murdock_ut, does it provide any sort of disconnect notification? |
20:38.17 | tzafrir_laptop | If not, do you use busydetect? |
20:39.45 | murdock_ut | It looks like I do not have that set in my zapata.conf |
20:40.09 | murdock_ut | tzafrir_laptop: Maybe I'll try that. |
20:40.35 | darkdrgn2k3 | sorry LOL |
20:40.41 | darkdrgn2k3 | so any idea what i screwed up this time? |
20:43.02 | darkdrgn2k3 | I see the call come through but then i get the "number not in service" message |
20:43.21 | Katty | blacklist it? |
20:44.46 | darkdrgn2k3 | wierd.. if i put the inbounce route to an IVR it works.. but not to an extension |
20:45.41 | Katty | any extension? or just a particular one. |
20:46.13 | darkdrgn2k3 | ok im stuck |
20:46.38 | darkdrgn2k3 | oops |
20:46.40 | darkdrgn2k3 | aany one |
20:48.58 | *** join/#asterisk maour_ (n=gnu@unaffiliated/maour) |
20:50.14 | *** join/#asterisk el_critter (n=critter@200.8.97.41) |
20:51.45 | citywok | Another issue with queues i'm running in to is if there is one member in a queue, and they are paused, a caller does not fall out fo the queue with leavewhenempty=yes |
20:54.05 | *** join/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
20:54.05 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
20:56.39 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
20:59.24 | *** join/#asterisk Jymm (i=jim@unaffiliated/jymmm) |
21:00.31 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
21:00.35 | Jymm | Just curious... Is it possible to setup asterisk to be a fax server without a modem/ata device? Just using SIP credentials |
21:00.57 | Katty | if you figure out how, let me know. |
21:01.06 | darkdrgn2k3 | me too |
21:01.21 | Jymm | Well, I can *always* whistle the tones if that helps |
21:01.26 | darkdrgn2k3 | lol |
21:01.27 | Jymm | *almost* |
21:01.31 | darkdrgn2k3 | apperntly you can also dial 666 :) |
21:01.49 | Jymm | why would I want to call myself? |
21:01.57 | darkdrgn2k3 | 666->system fax |
21:02.05 | darkdrgn2k3 | at leastr on some guis.. |
21:02.07 | Katty | not on my server! |
21:02.16 | Katty | 666 goes to the monkies. |
21:02.20 | darkdrgn2k3 | -> on some guis |
21:02.22 | Katty | or possibly weasels. |
21:02.25 | Jymm | darkdrgn2k3: I guess you were serious, but I have nfc |
21:02.40 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
21:03.00 | darkdrgn2k3 | on freepbx feature 666 is "Dial System FAX " |
21:03.05 | darkdrgn2k3 | and you get the high pitch whine of a fax |
21:03.10 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
21:03.36 | Jymm | ah |
21:03.39 | wcselby | Jymm - using iaxmodem and hylafax, it might work |
21:03.39 | Katty | does fax detect off an IVR work? |
21:03.50 | Katty | you might be able to point a 'sip trunk' to an IVR which has fax detect. |
21:03.53 | Yuda-israel1984 | hi guys im a newbie here i have a question for those that understand , i am reading the book the future of telephony 2nd edition and im starting to learn scripting and i added a sip provider and i get a weird message when i call the number it says chan_sip.c:14035 handle_request_invite: Call from '7183057535' to extension 's' rejected because extension not found. and in my script of the context i have it as 7187057535,1,answer() |
21:03.55 | Katty | for incoming. |
21:03.58 | wcselby | there's also something called spandsp but I don't know what that is, it's just on the fax page of the wiki |
21:04.12 | Katty | i've been looking forever for a pysical fax machine which will take a 'sip trunk' |
21:04.43 | wcselby | Yuda-israel1984 - you have different numbers listed "7183057535 != 7187057535" |
21:04.44 | *** join/#asterisk thansen (n=thansen@12.152.165.169) |
21:05.15 | Jymm | wcselby: you think it'll be fine for fax only? |
21:05.26 | Yuda-israel1984 | wcselby i meant it says 7183057535 its the same number |
21:05.33 | wcselby | Jymm - i haven't tested, but apparently some people have. |
21:05.51 | Katty | has never had much luck with asterisk and faxing. |
21:06.03 | wcselby | Jymm - http://www.voip-info.org/wiki/view/Asterisk+fax |
21:06.13 | wcselby | i've got faxing working, but it's using a t1 modem and pri's coming |
21:06.17 | wcselby | so it's not really over SIP |
21:06.26 | Jymm | wcselby: ah, cool. Wonder if I any of the OpenWRT devices would work with it. |
21:07.04 | wcselby | Yuda-israel1984 - check the context on the sip.conf definiton |
21:07.04 | Katty | Yuda-israel1984: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
21:07.40 | wcselby | his sip.conf context= is different from whereever he put the 7183057535,1,answer() statement |
21:07.47 | Yuda-israel1984 | i did but i have a feeling that the sip provider isnt passing the correct info |
21:08.06 | wcselby | Yuda-israel1984 - paste your sip.conf file and your extensions.conf file to pastebin |
21:08.11 | wcselby | so we can look at the whole thing |
21:08.17 | Yuda-israel1984 | ok one sec |
21:08.27 | wcselby | that error message for me as always meant the sip.conf context definition is wrong |
21:08.31 | *** part/#asterisk nullable_type (n=nullable@hq.verbx.net) |
21:08.42 | Yuda-israel1984 | [incoming_calls] |
21:08.42 | Yuda-israel1984 | exten=>7183057535,1,answer() |
21:08.42 | Yuda-israel1984 | exten=>7183057535,n,Dial(SIP/MPC/17187150001) |
21:08.59 | wcselby | Yuda-israel1984 - please use pastebin |
21:09.00 | wcselby | ~pb |
21:09.01 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
21:09.08 | Yuda-israel1984 | whats that |
21:11.47 | wcselby | Yuda-israel1984 - in your sip.conf file, what's the context= statement for your sip trunk? |
21:12.11 | Jymm | Katty: slow to load... http://www.faxsuperstore.com/ricoh-4420nf.html |
21:12.13 | wcselby | actualluy, just got to pastebin.com, paste in your sip.conf file and your extensions.conf file, and then click "send" |
21:12.22 | Yuda-israel1984 | http://pastebin.com/m2fedccc3 |
21:12.27 | wcselby | there we go |
21:12.46 | Yuda-israel1984 | as i said im a newbie im learning and i thank you all for helping me learn |
21:13.33 | wcselby | pastebin the cli output from a failed call |
21:13.40 | tzafrir_laptop | Yuda-israel1984, are you sure a call comes into that extension? |
21:13.53 | tzafrir_laptop | try adding the following extensions: |
21:13.54 | Yuda-israel1984 | thats what im not sure about |
21:14.13 | Yuda-israel1984 | when i look at the sip file i get something weird i will add it in pastebin |
21:14.35 | tzafrir_laptop | exten => _X.,1,NoOp(In Fallback for ${EXTEN}) |
21:15.11 | tzafrir_laptop | exten => s,1,NoOp(In default extension s) |
21:15.30 | tzafrir_laptop | now increase verbosity: core set verbose 3 |
21:15.46 | tzafrir_laptop | reload dialplan: dialplan reload |
21:15.56 | wcselby | alrighty, I'm out |
21:15.57 | tzafrir_laptop | and see what happens in an incoming call |
21:17.21 | Yuda-israel1984 | herre i am updating it |
21:17.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
21:18.19 | Yuda-israel1984 | http://pastebin.com/m7f40c4a9 |
21:18.42 | Yuda-israel1984 | ok i will do that tzafrir |
21:20.27 | Yuda-israel1984 | == Auto fallthrough, channel 'SIP/7183057535-0825eee0' status is 'UNKNOWN' |
21:20.35 | Yuda-israel1984 | thats the added part |
21:21.40 | *** join/#asterisk jlnt (n=jlnt@adsl-99-57-151-117.dsl.rcsntx.sbcglobal.net) |
21:26.39 | Yuda-israel1984 | Executing [s@incoming_calls:1] NoOp("SIP/7183057535-0825eee0", "In default extension s") in new stack |
21:26.43 | Yuda-israel1984 | sorry i missed that |
21:28.19 | *** join/#asterisk voipmonk (n=voipmonk@69.172.93.45) |
21:28.59 | darkdrgn2k3 | ?? asterisk16 |
21:31.55 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
21:33.39 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
21:34.18 | ebroad | when used for a peer, is * supposed to use auth=user#md5secret@domain for register? |
21:36.58 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
21:37.35 | Yuda-israel1984 | tzafrir it seems that the extension goes straight to S |
21:37.40 | Yuda-israel1984 | not to its DID |
21:37.44 | Yuda-israel1984 | why would that be? |
21:43.16 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
21:51.16 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
21:54.18 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
21:59.45 | *** join/#asterisk denon (i=denon@sassinak.net) |
21:59.54 | *** mode/#asterisk [+o denon] by ChanServ |
22:00.20 | *** join/#asterisk josefig (n=JoseFig@189.129.147.130) |
22:00.28 | *** join/#asterisk savageone (n=savageon@static-66-212-194-134.cpe.metrocast.net) |
22:00.35 | savageone | does anyone know how to monitor asterisk with snmp? |
22:01.01 | Qwell | res_snmp? |
22:01.51 | savageone | or net snmp |
22:01.56 | savageone | that's what others are saying they use |
22:02.04 | savageone | my monitoring software wants the oid |
22:02.09 | savageone | i don't know what that is hehe |
22:02.19 | savageone | or rather I know what an oid is but not what to use for asterisk |
22:02.34 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
22:02.37 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
22:03.52 | josefig | i have a doubt, i need to send 24V how can I do it with asterisk ? |
22:04.18 | Qwell | josefig: what? try explaining what you need any why.. |
22:04.27 | Qwell | and why* |
22:04.37 | josefig | okay |
22:05.46 | Naikrovek | you want to power the phone without a power adapter? |
22:05.52 | josefig | i have some extensions in my office and i need to open an electric front door with an extension for example 201 for the front door but it opens with 24V how can I do it with asterisk ? |
22:06.01 | Naikrovek | ah |
22:06.12 | el_critter | Hi, I have this extension: exten => _9X.,1,Dial(${SNTV}/${EXTEN:1},20). SNTV is PSTN. When I call a number that matches that pattern the call drops with this error: -- Nobody picked up in 20000 ms. |
22:06.39 | Naikrovek | josefig: you will need a door controller; talk to a local alarm company |
22:06.49 | _brent_ | josefig: http://cyberdata.net/products/voip/digitalanalog/intercom/index.html |
22:07.20 | _brent_ | josefig: i had a demo unit here a few months back. it's pretty cool. |
22:08.45 | josefig | but how can I send the 24V ? via USB but how with asterisk ? |
22:08.57 | *** join/#asterisk knarfly (n=vlad@c-66-176-177-82.hsd1.fl.comcast.net) |
22:09.04 | Naikrovek | asterisk doesn't send the 24V, it controls a device that sends the proper voltage |
22:09.29 | _brent_ | the device at the link i send has a dry relay switch, this connects the 24v, but it doesn't supply it |
22:10.19 | josefig | Naikrovek, yes but how can I do that? because i can get the 5V from USB and get higher to 24V |
22:10.24 | knarfly | help, I'm setting up my pbx to use bandwidth.com's SIP trunking. Unless I open the firewall wide open I can't receive calls and on some calls the callee cannot hear me. I thought SIP uses 5060-5082 and 8000-20000. I have those open but unless I open all ports I keep getting only partial performance |
22:10.54 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
22:10.58 | ebroad | josefig, check out http://www.phidgets.com/ |
22:11.09 | _brent_ | josefig: read the product info on that intercom. you send DTMF to it, it trips the switch, and you have to have 24v on one side of the switch and your door on the other |
22:11.11 | Naikrovek | josefig: you need a door controller that listens to asterisk, as _brent_ linked earlier |
22:11.22 | knarfly | 8-) |
22:11.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
22:11.28 | josefig | okay |
22:11.33 | josefig | lemme chk, thx |
22:11.45 | *** part/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
22:11.50 | FlaPer87 | hey guys, Is it possible to use raw extension with ast_writefile ? |
22:12.30 | ebroad | its a usb relay |
22:12.44 | ebroad | you might be able to trigger it via a System() call |
22:17.28 | Jymm | Just wire the output of a ATA devce direct... 100Volts should pop that door right open! |
22:18.54 | Jymm | That was a joke.... don't try it at home |
22:22.10 | ebroad | when used for a peer, is * supposed to use auth=user#md5secret@domain for register? |
22:23.08 | *** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
22:23.50 | Jymm | _brent_: That's pretty cool, ugly, but pretty cool =) |
22:26.16 | _brent_ | yeah, it seemed to work pretty well, too. it didn't do central provisioning, so it was a no-go for us, but it was pretty cool. |
22:26.37 | _brent_ | cyberdata says they're working on central provisioning |
22:26.47 | Jymm | _brent_: "cemtral provisioning" ? |
22:27.12 | _brent_ | most decent VoIP phones/ATAs will grab their configs via http |
22:27.26 | Naikrovek | ring voltage is 48V |
22:27.31 | Jymm | oh, dhcp |
22:27.54 | _brent_ | not quite the same as dhcp, but dhcp can push the http URL that the phone should download for its configs |
22:28.03 | Jymm | Naikrovek: 48V on hook, 88v off hook |
22:28.10 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
22:28.30 | Naikrovek | off-hook voltage while talking is 88V? |
22:28.39 | Naikrovek | pretty sure that's 12V |
22:29.01 | Jymm | no, I think it drops belw 48v. ON hook is 48. Ring is 88+volts |
22:29.25 | tzafrir_laptop | less than 12V |
22:29.42 | tzafrir_laptop | closer to 5V |
22:29.57 | diatonic1 | Hey Naikrovek - have you got a sec to take a look at what I posted in #trixbox and see if you have any ideas? |
22:30.10 | Naikrovek | ring is 48V on the phone system I took apart, because I still have the ring relays |
22:30.59 | Naikrovek | though i'm reading that they can indeed differ across the US |
22:31.26 | Naikrovek | the old bell telephone i'm looking at has a 48V ringer too |
22:31.31 | Jymm | http://www.epanorama.net/circuits/telephone_ringer.html |
22:32.36 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
22:32.54 | knarfly | and in case anyone else needs to be forewarned, one of my grandstream phones freaked out after only minimal use and those bums refused to replace it. |
22:33.09 | knarfly | no more gs phones for me |
22:33.21 | Jymm | either way, should energize the solenoid at least once =) |
22:33.27 | _brent_ | knarfly: not surprising. gs isn't exactly known for making a high quality phone |
22:34.00 | knarfly | nor taking care of you once you've dropped $100 on one of their phones which stops working |
22:34.14 | [TK]D-Fender | ~gs |
22:34.14 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
22:34.20 | [TK]D-Fender | ~grandstream |
22:34.21 | infobot | i heard grandstream is the Yugo of VoIP hardware. Run. Run away now.. Though therealcircut says that they're not that bad |
22:34.28 | [TK]D-Fender | knarfly: SAY IT AIN'T SO! |
22:34.31 | _brent_ | awesome :-) |
22:34.42 | _brent_ | if the infobot says it's true... |
22:36.25 | knarfly | yep, they have seen the last of my wallet |
22:36.25 | Naikrovek | polycom :) |
22:37.34 | Qwell | knarfly: who pays $100 for a gs? |
22:37.48 | Naikrovek | no joke |
22:37.54 | Naikrovek | well they do have higher models |
22:38.17 | [TK]D-Fender | DeluxeCrap (tm) |
22:39.46 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net) |
22:42.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
22:47.11 | nextime | mobile voip push system based on asterisk server side and pjsip client side is working! |
22:54.33 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-189-232.dsl.stlsmo.sbcglobal.net) |
22:54.36 | LemensTS | hey all |
22:55.03 | Naikrovek | yo |
22:55.27 | citywok | lol, we have a few grandstreams and i hate them, garbage |
22:55.43 | citywok | the aastra's i've played with have a Phenomenal speakerphone, and great XML interface |
22:56.27 | LemensTS | i launch a php script from the dial plan, and when it is done it hangs up and doesnt do the rest of the dial plan after it http://pastebin.com/m7b5eac93 |
22:56.41 | LemensTS | i thought it would go back to the next step in the dialplan, i tried agi and deadagi |
22:56.58 | Naikrovek | polycom polycom polycom. they should pay me for the amount of times i say that |
22:57.15 | citywok | i played with a cheaper polycom and wasn't all that impressed. though, it was a cheaper one to be fair. |
22:57.37 | Qwell | Naikrovek: sorry, only one of us are allowed to get paid by Polycom when we say Polycom. |
22:57.40 | Qwell | I've said too much. |
22:57.42 | citywok | i was shocked that it had a horrible speakerphone. i figured any polycom would be able to do that |
22:58.08 | LemensTS | that was with agi debug set on |
22:59.21 | diatonic1 | I love my Polycoms |
22:59.22 | [TK]D-Fender | Naikrovek: Take a number :p |
22:59.26 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@bas2-toronto44-1176437585.dsl.bell.ca) |
22:59.38 | Naikrovek | yeah |
22:59.55 | darkdrgn2k3 | ok im stumped |
23:00.00 | darkdrgn2k3 | running asterisk 1.6 |
23:00.01 | darkdrgn2k3 | http://pastebin.ca/1577184 |
23:00.11 | darkdrgn2k3 | calls keep getting "number you dailed is not in service" |
23:00.23 | darkdrgn2k3 | and frmo time to time a call will get through.... but the next dial is not in service |
23:00.37 | darkdrgn2k3 | this is only when i send the call to a extension (any extension) or voicemail |
23:00.38 | [TK]D-Fender | darkdrgn2k3: #freepbx <---- go back to MagicGUILand... |
23:00.42 | darkdrgn2k3 | if i send the call to ivr it works |
23:00.48 | darkdrgn2k3 | i been sitting there... |
23:00.54 | darkdrgn2k3 | but no one seems to have a clue..... |
23:00.58 | Naikrovek | one thing grandstream phones do have over polycom is iax2 support |
23:01.01 | [TK]D-Fender | darkdrgn2k3: This is not 2nd Level FreePBX support |
23:01.06 | darkdrgn2k3 | hopeing you cli ppl could give me an idea whats wrong |
23:01.29 | [TK]D-Fender | Naikrovek: Except Nobody cares about IAX2, and the audio has a bit of a history of instability... |
23:01.43 | nextime | is back to asterisk 1.4 after an hard fight with 1.6 issues in lates few days |
23:02.01 | Naikrovek | my iax2 trunks work great. |
23:02.05 | darkdrgn2k3 | what does " -- <SIP/pritrunkdomain-08feaf80>AGI Script recordingcheck completed, returning -1" mean |
23:02.41 | knarfly | ok the tech from bandwidth.com just called and said to open all udp ports from 1024 thru 65535....is this really a secure way to run an asterisk server? |
23:02.46 | Naikrovek | darkdrgn2k3: depends entirely on what php's return code of -1 means |
23:03.01 | [TK]D-Fender | knarfly: no functional difference |
23:03.04 | Naikrovek | knarfly: you just open them from particular hosts, and only udp |
23:03.12 | Yuda-israel1984 | anyone use MPC here?? |
23:03.15 | nextime | knarfly : why ALL > 1024 ports? |
23:03.38 | Qwell | nextime: because they are clueless. |
23:03.39 | Naikrovek | nextime: that's what he's asking |
23:03.43 | knarfly | i thought asterisk used 8000-20000 but with that it don't work with bandwidth.com |
23:04.01 | Qwell | Asterisk uses what you set in rtp.conf. |
23:04.21 | nextime | KrisWillis : just open the ports 5060 and the ports between rtpstart and stop you can read on rtp.conf |
23:04.42 | knarfly | rtpstart=10000 |
23:04.42 | knarfly | rtpend=20000 |
23:05.09 | citywok | knarfly: did bandwidth.com not give you a single IP to communicate with? |
23:05.10 | nextime | knarfly : and you can off course also change it to a less large set of ports |
23:05.21 | knarfly | that's my rtp.conf but for some reason I'm seeing log entries in my security file that says the DID provider is coming in on 64608 |
23:05.31 | Qwell | knarfly: That is their source port. |
23:05.37 | Qwell | (or yours) |
23:05.52 | Naikrovek | "coming in on" would indicate the port he's meant to listen on |
23:06.48 | knarfly | it appears than Bandwidth.com uses a bunch of different servers because the traffic comes in from various IP addresses and various IP ports |
23:07.06 | kc8pxy | i'm trying to listen to a voicepack i found, and i can't get any of my players to play gsm. any recommendations? |
23:07.17 | [TK]D-Fender | knarfly: Just do it. Doesn't make any real difference |
23:07.21 | Qwell | kc8pxy: get a player that supports gsm. |
23:07.22 | nextime | kc8pxy : use sox |
23:07.33 | nextime | and convert it to anything you can read |
23:08.06 | LemensTS | hey TKD-Fender i launch a php script from the dial plan, and when it is done it hangs up and doesnt do the rest of the dial plan after it http://pastebin.com/m7b5eac93 |
23:08.08 | darkdrgn2k3 | hmm no matter what i try to modify the script it keeps saying returned -1 |
23:08.19 | knarfly | yes but I still don't like the ideal of opening up the whole range of ports. |
23:09.30 | nextime | must find the time to open a couple of tickets on mantis |
23:10.47 | knarfly | I also don't like the way that once I start chatting on this board, there are a couple of attempts logged at unauthorized users trying to get at my asterisk server |
23:11.09 | citywok | why are you connecting to this board from your asterisk server? |
23:11.17 | nextime | rotfl |
23:11.33 | knarfly | hey 189.144.12.47 no habla espanol |
23:11.41 | nextime | chatting on irc from a server ( expecially a production one ) isn't a good idea |
23:11.54 | citywok | and some services generally check to make sure that you aren't running an open web proxy, and a few other things to make sure you aren't a compromised host |
23:12.08 | knarfly | no I'm simply watching the log file on my router |
23:12.32 | citywok | yea, that's not all that shocking |
23:13.44 | nextime | knarfly : you can even ask an irc oper to give you an cloack host |
23:13.57 | Naikrovek | don't you have to donate to get that on freenode? |
23:14.03 | nextime | so, no one except irc operators and admins can see your ip |
23:14.06 | Qwell | not an anon host, no |
23:14.23 | nextime | Naikrovek : no, if you donate you get a different one (supporter) |
23:14.39 | nextime | if you just ask to have one you have one like mine, "unaffiliated" |
23:14.39 | Qwell | You all should donate to FreeNode though, of course. |
23:14.42 | Qwell | :p |
23:14.47 | nextime | i agree :) |
23:15.14 | Jymm | made his check out to lilo |
23:15.26 | knarfly | to be honest I'd rather not hang around with a bunch of looser who don't have anything better to do that to try and crack into someone's server they worked very hard to get going. |
23:15.40 | Jymm | lol |
23:16.09 | Qwell | Jymm: I sincerely hope you aren't trolling. |
23:16.41 | carrar | wait you have a pbx at 66.176.177.82? |
23:16.44 | Yuda-israel1984 | sorry to write this again but i have a question if on a sip message where it says To: if nothing is written then it wont go to any extension now would it |
23:16.47 | carrar | I should try |
23:17.05 | carrar | fires up nmap |
23:17.44 | carrar | doh |
23:17.45 | carrar | he left |
23:18.33 | [TK]D-Fender | Yuda-israel1984: It could. If you made one it could match. |
23:19.20 | Yuda-israel1984 | TK can u please explain |
23:19.20 | kc8pxy | nextime: is it simple to tell it to convert everything in a folder, and rename it file.ogg where it was file.gsm before? |
23:20.01 | [TK]D-Fender | Yuda-israel1984: Go make an exten it can match |
23:21.27 | Yuda-israel1984 | but only the provider passes on the TO: message |
23:21.28 | Yuda-israel1984 | no? |
23:21.35 | *** join/#asterisk ESCulapio__ (n=ESCulapi@barcelord.com) |
23:22.00 | [TK]D-Fender | Yuda-israel1984: Go look at an actual call attempt and you tell ME |
23:22.08 | nextime | kc8pxy : find /path/to/your/folder -name '*.gsm' -type f -exec sox {} {}.ogg ; rename "s/\.gsm//" {}.ogg \; |
23:22.18 | nextime | something like that should work |
23:22.18 | Yuda-israel1984 | i did i get back an error |
23:22.30 | Yuda-israel1984 | of it not having the S extension |
23:22.52 | Yuda-israel1984 | meaning i am not reffering to S i want to know if i can make it pass something else |
23:24.12 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
23:25.00 | [TK]D-Fender | Yuda-israel1984: It its complaining about not finding "s" thats because you didn't tell them what exten to dial... so Asterisk told them for you |
23:25.26 | [TK]D-Fender | Yuda-israel1984: Hint.... its in your REGISTER |
23:26.31 | kc8pxy | nextime: looks like it worked. missing arg to -exec |
23:26.35 | Yuda-israel1984 | u mean i MUST add in the register line to add the /DID ???? |
23:26.50 | kc8pxy | other than that, it tried :) |
23:26.54 | Yuda-israel1984 | but i want it that it should pass me the DID being called not that i have to add each one |
23:28.10 | *** join/#asterisk jcape (n=jcape@adsl-99-37-112-194.dsl.chcgil.sbcglobal.net) |
23:28.19 | nextime | Yuda-israel1984 : something like _X. extension and get the ${EXTEN} variable? |
23:28.59 | [TK]D-Fender | Yuda-israel1984: They probably are passing you the DID... in a HEADER you have to strip off |
23:29.16 | [TK]D-Fender | Yuda-israel1984: "core show function SIP_HEADER" |
23:30.33 | Yuda-israel1984 | reading that now in the book |
23:31.43 | [TK]D-Fender | jam time, BBL |
23:32.54 | Yuda-israel1984 | thanks have a good night yall |
23:38.48 | jgoo | ok, si por |
23:38.52 | jgoo | fucking keyboard |
23:38.55 | jgoo | ok sip ports :-) |
23:39.12 | jgoo | I have 8 PAP2Ts, and they were working fnie |
23:39.48 | jgoo | now I tried introducing a third SPA400 into the mix, and it seemed to initiate a series of unfortunate events - fixing one problem exposed another - now it seems that it was luck making this work |
23:40.04 | LemensTS | do i need asterisk-addons to use mysql in the dialplan? |
23:40.06 | jgoo | and it really wants all this on separate ports - what is the requirement here? |
23:40.20 | jgoo | does anyone use this stuff? |
23:40.49 | Qwell | LemensTS: yes |
23:41.17 | jgoo | Do I need to have each port on the linksys on a different port, as it is, 5060 / 5061 - and in the extension definition, until now, the port was always 5060, and IT WORKED - now it seems it works if I set it to 5060 / 5061, depending on which port I assign the extension |
23:41.18 | carrar | I thought this is a drinking channel |
23:41.29 | jgoo | I like a fine port |
23:43.35 | jgoo | back to the issue - can people just say what the hell they do use? nobody seems to use anything, or even use asterisk in this channel half the time - what is it that the cool kids are using on asterisk? |
23:43.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
23:44.03 | bmoraca | jgoo: tcp/ip networking dictates that only one application can be active on one port at a time. so, yes, each "line" needs to register on a different port. |
23:44.09 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) |
23:44.29 | jgoo | bmoraca: please be more clear - every extension needs a seperate port? |
23:44.39 | jgoo | "line" isn't clear, and can mean almst anything - - |
23:44.51 | jgoo | so, if I have 50 extensions, i need 50 ports? |
23:46.06 | jgoo | I would love to see the documentation that states this, because I've never seen any documentation talk about more than one extension ever - and how come I've had 5-6 simultaneous calls going just during my own testing using 8 PAP2Ts? is that not runnnig more than one app on a port (also, that doesn't mean it will bind to that port on the server) |
23:46.25 | jgoo | just like firefox doesn't bind its ports to 80 no the client - it is just open to 80 on the server |
23:47.46 | jgoo | bmoraca: tl;dr - what is the way to setup 16 extensions across 8 PAP2Ts |
23:47.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
23:48.08 | *** join/#asterisk voipmonk (n=voipmonk@69.172.93.45) |
23:49.28 | jgoo | anyone? ports? anyone using more than one extension on a pbx? why is this seemingly important, yet no, until it stops magically working, and isn't really documented in the docs I've read? |
23:49.56 | carrar | Why would anyone need more then 1 extension on a pbx!! |
23:50.00 | jgoo | bmoraca: have you ever setup more than one extension? |
23:50.06 | jgoo | carrar: my thoughts exactly |
23:50.20 | jgoo | a pbx is just a fancy way to plug a phone into a wall |
23:50.39 | jgoo | one day they will make pbxes that will fit in your phone, and bye bye computer! imagien. |
23:50.42 | carrar | You can convert a soundcard to be a PBX |
23:50.42 | jgoo | damn keybaord |
23:50.43 | ehsjoar | jgoo: Sorry to jump in so late here. All client will always initiate all SIP on 5060, unless you tell them otherwise. so 1 port |
23:51.19 | jgoo | so I can run all my PAP2Ts on the same port? |
23:51.34 | LemensTS | hmmm i did this http://pastebin.com/m48dab0d4 to install asterisk addons, and i rebooted server, do a help mysql at cli and get cmd unknown |
23:51.42 | jgoo | out of nowhere I am having problems with this config - without changing anything - just restoring from backups... |
23:51.43 | el_critter | Hi, when Dial sets a call and the other side answers, does it sets a state variable somewhere? |
23:51.50 | *** join/#asterisk wr| (n=wr@p54BE7949.dip.t-dialin.net) |
23:51.55 | ehsjoar | jgoo: Yes, if they do SIP they should all use 5060 |
23:52.28 | hardwire | el_critter: what are you doing that could query the state? |
23:52.48 | jgoo | ehsjoar: why are they preconfigured to use 5060 and 5061 for line 1 / line 2 ? |
23:53.20 | jgoo | ehsjoar: and if I use line 2, the extension I use, should that port match the line 2 port? until now I didn't and it worked. But now I changed it to match *AND IT STILL WORKS* what the hell? |
23:53.26 | LemensTS | nevermind i guess thats how it is supposed to be in 1.6 |
23:53.28 | ehsjoar | jgoo: Do not know. I have installed * for 50 users. All with SIP phones configured to use port 5060 |
23:53.33 | jgoo | maths would be a lot easier if we just had to approximate our numbers |
23:53.34 | geneticx | can someone shed some light, I'm trying to set up a phone to dial automatically after 10 digits have been entered, is that something you program in asterisk or the phone itself? |
23:54.09 | jgoo | ehsjoar: me too... but it looks like it may be the problem... or at least n3glv said so |
23:54.15 | el_critter | hardwire: When one of my SIP extensions call a PSTN line (Digium TDM400P), it seems like sometimes the call is terminated (sometimes by the SIP extension, usually by the destinantion number) and asterisk or dahdi keeps the line open forever. I think maybe is a bug. |
23:54.32 | ehsjoar | jgoo: What configuration parameter on the PAP2T are you talking about |
23:54.35 | carrar | geneticx, digitmap on the phone |
23:54.38 | *** join/#asterisk MaliutaLap (n=biteme@bne.lentz.com.au) |
23:54.46 | jgoo | ehsjoar: I have 50 extensions, 16 from PAP2Ts that will be strewn around - just 3 of these extensions have issues - - - - - since I restored a valid backup... ffs |
23:55.13 | hardwire | el_critter: doh. |
23:55.15 | ehsjoar | jgoo: when you say issues, don't they register correctly with * or what issues are you seeing |
23:55.27 | jgoo | ehsjoar: SIP Port |
23:55.31 | geneticx | carrar: could it be under another name with a linksys SPA941 because I don't see anything |
23:55.41 | ehsjoar | jgoo: That can safely always be 5060 |
23:55.47 | ehsjoar | jgoo: Hold on |
23:55.50 | el_critter | hardwire: no idea? |
23:56.19 | jgoo | ehsjoar: but on the PAP2T with two ports, the second is configed to 5061 - no docs say to configure the other side like this, and it seems to work either way |
23:56.43 | geneticx | carrar: Ok i found it, thanks |
23:56.43 | ehsjoar | jgoo: Unless there is a problem with PAP2T having 2 analogue lines and for whatever reason can't use one SIP port. There is no problem for Asterisk for sure |
23:57.17 | hardwire | el_critter: Dial jumps to other extensions based on the status of a call. |
23:57.22 | jgoo | ehsjoar: ... is having two analogue lines a problem? |
23:57.30 | ehsjoar | jgoo: No |
23:57.51 | ehsjoar | jgoo: If you didn't specifically tell asterisk to listen to 5061 it will only listen on 5060 |
23:57.52 | hardwire | el_critter: sorry.. sounds like you hit a bug |
23:57.57 | jgoo | I mean, for the PAP2T - since it does put them on different ports =/ |
23:58.06 | ehsjoar | jgoo: That is in sip.conf |
23:58.18 | el_critter | hardwire: sometimes I put a timeout on Dial, the other side answers the call and then Dial hangs because of timeout. I think maybe Dial is not setting correctly some state |
23:58.22 | ehsjoar | jgoo: Perhaps PAP2T is just using that as a backup port or something |
23:58.35 | ehsjoar | jgoo: if you change it to just 5061 it will probably not work |
23:58.47 | hardwire | el_critter: turn on debug |
23:58.57 | hardwire | look for "remote side answered call" like events in your debug log |
23:59.03 | hardwire | like.. turn debug on in logger.conf |
23:59.13 | el_critter | ok |
23:59.18 | el_critter | brb then |
23:59.19 | hardwire | if you don't see any.. then dahdi is not understanding the call was answered |
23:59.23 | ehsjoar | jgoo: I am assuming that this parameter is for what port the sip server should be contacted on and not what port PAP2T is listening on |
23:59.31 | jgoo | ehsjoar: 'listen' is a loaded term for sockets - who is listening when asterisk connects the client? anyway - it has been working with 8 lines on 5061, and all 16 extensions on 5060, and all lines working - how do you explain that? |