| 00:08.18 | *** join/#asterisk awkfu (n=awkfu@66.162.90.57) |
| 00:08.59 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:09.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:09.28 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 00:15.51 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:18.08 | carrar | osburn.com/swineflu.jpg |
| 00:23.24 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 00:23.28 | timeshell | Got SIP response 482 "Loop Detected" back from 0.0.0.0 |
| 00:23.41 | timeshell | What causes this to happen? |
| 00:24.56 | ambush276 | hey guys need a tad bit of help |
| 00:24.57 | ambush276 | ok |
| 00:25.02 | ambush276 | with skype to sip |
| 00:25.09 | ambush276 | when i call my pbx(skype number which goes through the gateway) |
| 00:25.28 | ambush276 | itw orks fine ,, but when i hit extension 11 (outside line, like a 1800 number) the skype2sip says busy |
| 00:25.36 | ambush276 | what do i do so that it can place the call back out through skype? |
| 00:32.46 | KyleK | timeshell: asterisk thinks its calling itself for whatever reason |
| 00:33.06 | timeshell | KyleK, Yah, but it's not |
| 00:33.11 | timeshell | That's the strange part of it |
| 00:34.01 | timeshell | KyleK, Not only that, I have the same error on the two servers that register to each other and are supposed to be forwarding the extensions to each other. |
| 00:34.19 | KyleK | ambush276: which skype to sip, the one thats in beta or some 3rd party thing |
| 00:34.31 | ambush276 | 3rd party |
| 00:34.33 | ambush276 | by nch |
| 00:34.36 | ambush276 | im signed up for the beta. |
| 00:34.43 | ambush276 | but havent gotten any results back yet as to my status |
| 00:37.03 | timeshell | KyleK : Now forwarding SIP/7007-092875f0 to 'Local/5221@DID_LUSI_PBX' (thanks to SIP/LUSI_PBX-0928b670) |
| 00:37.17 | timeshell | Why is it going to "Local" |
| 00:37.31 | timeshell | That's not the address defined for the trunk |
| 00:37.53 | KyleK | ambush276: does the one you're using attach to a skype client? fire up another skype client for outgoing calls? |
| 00:38.24 | KyleK | timeshell: im actally a total noob with this, i still need to set up my * to work behind a nat |
| 00:38.37 | KyleK | ~sipnat |
| 00:38.38 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:38.38 | ambush276 | it starts up a client |
| 00:38.53 | ambush276 | but only one is running |
| 00:38.59 | KyleK | can you make it start up two? |
| 00:50.06 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:53.15 | *** join/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 00:58.54 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 00:58.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:26.47 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 01:29.13 | ambush276 | no., |
| 01:29.16 | ambush276 | i wish i could |
| 01:29.26 | ambush276 | does anyone know hwo to do a hairpin effect for this program? |
| 01:29.39 | tfrew | hairpin effect? |
| 01:32.38 | xheliox | tfrew: Yeah, I'm glad I'm not the only one who has never heard that term. |
| 01:33.02 | tfrew | i have no idea what ambush means by hairpin effect |
| 01:33.14 | *** join/#asterisk nassy (n=nassy@207-38-194-41.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:35.25 | xheliox | tfrew: Me either. |
| 01:39.51 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:48.23 | ambush276 | harpin effect. |
| 01:48.25 | ambush276 | rlly? |
| 01:48.32 | ambush276 | im not really a voip expert but that is a basic term.. |
| 01:48.35 | ambush276 | basically |
| 01:49.02 | ambush276 | a call coems in throuhg an SIP trunk, goes to PBX system, then pbx extension is on an external line, so it goes back through the same SIP trunk to call out.. |
| 01:49.08 | ambush276 | im trying to do this with Skype to sip? |
| 01:49.10 | ambush276 | anybdy? |
| 01:49.50 | tfrew | so thats what thats called? |
| 01:50.06 | tfrew | does the skype phone have an incoming did with incoming minutes? |
| 01:50.15 | ambush276 | yea |
| 01:50.18 | ambush276 | the sykpe phone worsk fine |
| 01:50.20 | ambush276 | like |
| 01:50.26 | ambush276 | incoming works fine and goes to the pbx fione |
| 01:50.32 | ambush276 | but the outgoing is the problem |
| 01:50.41 | ambush276 | ext. 11 is set to an externa line, and when it is dialed it says "busy" |
| 01:51.50 | tfrew | are you using follow me or dialing out like an extension? |
| 01:54.38 | jaytee | ambush276, Skype does not work with Asterisk without the beta software for Skype support |
| 01:54.49 | ambush276 | ... |
| 01:54.53 | ambush276 | jaytee |
| 01:54.56 | ambush276 | the bridge.. |
| 01:54.57 | ambush276 | ughgh |
| 01:55.10 | ambush276 | 'http://www.nch.com.au/skypetosip/ |
| 01:55.15 | ambush276 | for u jaytee |
| 01:55.26 | jaytee | and Asterisk doesn't like hairpins |
| 01:55.37 | tfrew | ambush276: if your skype acount has skypein, you can make a custom extension that dials the skype did |
| 01:55.53 | tfrew | although, that uses up both the sip trunk and skypes minutes |
| 01:55.56 | ambush276 | but.. |
| 01:56.00 | ambush276 | the problem is the outgoign is through skype |
| 01:56.02 | ambush276 | so it wont go anyhwere |
| 01:56.05 | ambush276 | the incoming is on skype |
| 01:56.08 | ambush276 | and the outgoing is on skype.. |
| 01:56.10 | ambush276 | unless. |
| 01:56.12 | ambush276 | ... |
| 01:56.34 | tfrew | i don't understand what you are asking |
| 01:56.43 | jaytee | interesting site. hadn't seen that before |
| 01:57.40 | [TK]D-Fender | Aht he insanity continues.... |
| 01:57.59 | jaytee | [TK]D-Fender ? |
| 01:58.39 | [TK]D-Fender | jaytee: ambush276 here has been "not getting" it for some time, and extreme hand-holding has been more that difficult |
| 01:59.11 | ambush276 | Tk hye! |
| 01:59.13 | ambush276 | ya.. |
| 01:59.15 | ambush276 | well new problem |
| 02:00.09 | jaytee | [TK]D-Fender, ah! kind of like the guy with the "488 - Loop detected" error that kept asking and asking about it but would never post any details when asked. |
| 02:00.47 | drmessano | Ok so like um |
| 02:01.03 | [TK]D-Fender | yup |
| 02:01.11 | drmessano | If I download and use this skype to sip thing, right.. and my calls don't work.. umm |
| 02:01.24 | ambush276 | no... different question |
| 02:01.28 | ambush276 | it works |
| 02:01.33 | ambush276 | it's just busy when dialing out |
| 02:01.44 | drmessano | So its not working |
| 02:01.44 | ambush276 | do i need to have 2 clients running? or is there a setup thing? |
| 02:01.51 | ambush276 | incomign works fine now |
| 02:01.56 | ambush276 | it gives a busy signal when dialing out |
| 02:01.58 | jaytee | drmessano, you probably forgot to install the flash plugin component. remember, it's SIP>FLASH>SKYPE |
| 02:02.06 | drmessano | ROFLLLLL |
| 02:02.26 | [TK]D-Fender | ambush276: you trying to use your SIP>Skype to take calls in from skype and then send them out to other contacts> |
| 02:02.31 | drmessano | ambush276: That application is a piece of shit.. |
| 02:02.40 | drmessano | ambush276: I used it, it sucked |
| 02:02.53 | drmessano | ambush276: Now go outside and get some air |
| 02:02.56 | ambush276 | what sip trunk should i use? |
| 02:03.01 | smultron | anyone here have experience with the company Fonality? i was looking to use their software/hardware, but wondering if it would be better just to hire a consultant to put together a more open Asterisk system... |
| 02:03.02 | drmessano | ambush276: Pet a bird, feed a cow |
| 02:03.17 | drmessano | smultron: Fonality sucks |
| 02:03.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:03.30 | smultron | drmessano: any supporting arguments? ;) |
| 02:03.35 | jaytee | "Engineer Tim, Never Forget!!!!" |
| 02:04.32 | drmessano | smultron: They repackage other companies applications and try to call them a "shiny turd" |
| 02:04.55 | drmessano | smultron: bad ethics.. Want supporting arguments, read the trixbox.org forums |
| 02:05.02 | jaytee | and they kick little puppies too! |
| 02:05.19 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:05.20 | [TK]D-Fender | sometimes even big puppies |
| 02:06.14 | drmessano | smultron: They gained a bad reputation in the way they experiment on their open source users, with shitty code, crappy updates that break systems, a real bad attitude, and their shady attempts to monetize their offerings |
| 02:06.29 | drmessano | They also are very slack on security |
| 02:06.55 | drmessano | Their trixbox CE and trixbox Pro systems are notoriously insecure, which you can easily google for |
| 02:06.56 | smultron | drmessano: is there a place to get a consultant who can put together systems? i don't have the time/expertise to put one together myself |
| 02:07.34 | smultron | i'm just wary of close-source software in general |
| 02:07.44 | drmessano | watches smultron get 100 private messages |
| 02:07.59 | smultron | waits.... |
| 02:08.02 | smultron | :P |
| 02:11.02 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:19.26 | tfrew | smultron: the company i work for would be more then happy to be your consultant |
| 02:19.29 | tfrew | neteasyinc.com |
| 02:19.59 | tfrew | which would then be me talking with you to sell you either a trixbox or asterisknow server |
| 02:20.14 | tfrew | (with a hardend setup) |
| 02:20.29 | tfrew | are you looking to buy a server or do you already have appropiate hardware? |
| 02:20.42 | [TK]D-Fender | likes his servers chewy & tender. |
| 02:22.26 | drmessano | Ewwww |
| 02:22.33 | drmessano | Trixbox.. bleh |
| 02:22.44 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 02:23.35 | smultron | tfrew: we'd need new hardware (IP phones & server) |
| 02:23.52 | drmessano | What exactly is a trixbox server with a "hardened setup"? |
| 02:24.05 | tfrew | all passwords changed |
| 02:24.09 | drmessano | LOl |
| 02:24.10 | tfrew | iptbales firewall |
| 02:24.11 | eppigy | TRABAJO |
| 02:24.31 | tfrew | fail2ban applied to sip and iaz |
| 02:24.33 | tfrew | iax* |
| 02:24.41 | tfrew | and other tweaks |
| 02:25.08 | tfrew | it takes several hours longer to clean up the mess, but once the server is setup properly, trixbox is pretty decent for the end user |
| 02:25.25 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 02:25.29 | eppigy | i like to touch it |
| 02:26.08 | drmessano | Several hours you could build a box from scratch using FreePBX and Asterisk, have it hardened from the get-go |
| 02:26.23 | drmessano | Wasted time polishing a turd |
| 02:26.48 | tfrew | true |
| 02:27.18 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 02:28.12 | drmessano | Thats also installing using SVN and being able to deploy security updates without praying Fonality doesnt roll out bad RPMs like they do every so fuckin often and tank a bunch of boxes |
| 02:28.13 | timeshell | Hey, anyone know how to resolve Got SIP response 482 "Loop Detected" |
| 02:28.26 | drmessano | Pretty bad when getting exploited is as risky as the update process |
| 02:28.45 | timeshell | It's calling the correct context point to a remote server here... -- Called LUSI_PBX/5221 |
| 02:29.01 | timeshell | But for some reason, I still get .... -- Got SIP response 482 "Loop Detected" back from 0.0.0.0 |
| 02:29.45 | [TK]D-Fender | timeshell : Stop talking to yourself |
| 02:30.00 | timeshell | = anyone |
| 02:30.03 | timeshell | cool |
| 02:30.55 | timeshell | Comon TK, you're the expert. Why is this happening? |
| 02:31.47 | *** join/#asterisk juanIMP (n=Juancho@201.245.254.74) |
| 02:32.19 | *** part/#asterisk nassy (n=nassy@207-38-194-41.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:33.01 | [TK]D-Fender | timeshell : Clearly you're looping shit and FFS you haven't shown a God damn thing through hours of whining. WHERE'S THE FUCKING PASTEBIN? :) |
| 02:33.44 | timeshell | TK : Well DUH |
| 02:33.49 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 02:34.03 | timeshell | TK : It so happens that I can't see how it could possibly be looping. |
| 02:34.14 | eppigy | wow dog |
| 02:34.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:34.37 | [TK]D-Fender | timeshell : Of course you can't see, you're blind as a bat and have no clue what you're doing. |
| 02:34.53 | timeshell | Aw, be nice TK |
| 02:35.09 | [TK]D-Fender | timeshell : And the reason you don't ahve configs and CLI output with SIP debug enabled for you broken calls is....? |
| 02:35.11 | timeshell | Here's your pastebin |
| 02:35.13 | timeshell | http://www.pastebin.ca/1410055 |
| 02:35.26 | drmessano | sends timeshell some hamthrax |
| 02:36.29 | [TK]D-Fender | timeshell : Try again, this time a COMPLETE call with SIP DEBUG. I've beaten you over the head for this enough in the past already. |
| 02:36.40 | [TK]D-Fender | timeshelland the CONFIS |
| 02:36.44 | [TK]D-Fender | CONFIGS* |
| 02:37.02 | eppigy | UND KIENER EIR |
| 02:37.09 | drmessano | MEIN KAMPF |
| 02:38.21 | [TK]D-Fender | -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/7007-092847f8", "0 > 0 ?1-CHANUNAVAIL,1:1-out,1") in new stack |
| 02:38.28 | [TK]D-Fender | \o/ broken dialplan too |
| 02:39.54 | timeshell | TK : No, Dialplan not broken |
| 02:39.59 | timeshell | Already checked that |
| 02:40.38 | drmessano | Ich bin ein asterisker |
| 02:40.38 | timeshell | Read the CLI output again |
| 02:40.50 | timeshell | yah, yah |
| 02:41.04 | [TK]D-Fender | timeshell : Yes that GotoIf is broken |
| 02:41.29 | [TK]D-Fender | timeshell : More specifically the expression in it |
| 02:41.47 | drmessano | (0)>(0) |
| 02:44.10 | [TK]D-Fender | (o)(o) ? |
| 02:44.37 | drmessano | (. ) ( .) |
| 02:45.31 | [TK]D-Fender | drmessano: Itty bitty..... |
| 02:47.58 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 02:48.25 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 02:48.45 | ltd | Is there any way to disable MWI NOTIFY sip messages from being sent? |
| 02:49.16 | [TK]D-Fender | ltd : Stop telling your peers to watch VM |
| 02:49.28 | [TK]D-Fender | ltd : "mailbox=" <---------- |
| 02:49.51 | ltd | [TK]D-Fender: Okay, so they only send it if mailbox= is set? |
| 02:50.03 | timeshell | http://www.pastebin.ca/1410061 |
| 02:50.10 | [TK]D-Fender | ltd : yup |
| 02:50.18 | timeshell | TK : Relevant stuff from debug there] |
| 02:51.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 02:51.04 | [TK]D-Fender | timeshell ; -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/7007-092a8f58", "SIP/LUSI_PBX/5221") in new stack |
| 02:51.23 | [TK]D-Fender | timeshell : INVITE sip:5221@142.46.193.200 SIP/2.0 |
| 02:51.24 | timeshell | Yah, and? |
| 02:51.29 | eppigy | oh boi |
| 02:51.32 | timeshell | That's correct |
| 02:51.39 | [TK]D-Fender | timeshell : <--- SIP read from UDP://127.0.0.1:5060 ---> |
| 02:51.48 | *** join/#asterisk thansen (n=thansen@c-24-10-236-153.hsd1.ut.comcast.net) |
| 02:51.51 | [TK]D-Fender | timeshell : The IP is having * talk to ITSELF |
| 02:52.02 | [TK]D-Fender | timeshell : Look at your PEER |
| 02:52.04 | drmessano | As in, like a loop? |
| 02:52.09 | timeshell | 142.46.193.200 ISN'T the server this is coming from |
| 02:52.16 | [TK]D-Fender | drmessano: Wait... what are you trying to say? |
| 02:52.26 | eppigy | EUREKA |
| 02:52.50 | timeshell | Well HOW does that HAPPEN?! |
| 02:52.51 | drmessano | [TK]D-Fender: Now i've found, that the world is round, and of course, it rains, everyday...? |
| 02:52.55 | *** join/#asterisk intralanman (n=intralan@174-157-165-12.pools.spcsdns.net) |
| 02:52.58 | timeshell | That's what I'm trying to find out guys |
| 02:53.02 | [TK]D-Fender | timeshell : Its coming in over the LOOPBACK. WTF do you think that means? |
| 02:53.09 | timeshell | I didn't tell it to talk to or set 127.0.0.1 anywhere |
| 02:53.22 | timeshell | I didn't ask what it means TK |
| 02:53.27 | timeshell | I know what it means. |
| 02:53.30 | drmessano | sings "Will it go round in circles" |
| 02:53.33 | timeshell | I don't know WHY it's happening |
| 02:54.03 | eppigy | disable your lo interface |
| 02:54.10 | [TK]D-Fender | timeshell :Where are the configs? You didn't provide a complete call either. |
| 02:54.48 | drmessano | eppigy: LOL |
| 02:54.49 | timeshell | after it goes to loop, it just tries to redirect as local extension and fails because there is no local 5221 |
| 03:14.04 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 03:16.24 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 03:18.34 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 03:18.36 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 03:18.48 | ruben23 | hi |
| 03:26.17 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 03:26.39 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:28.26 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:45.24 | timeshell | http://www.pastebin.ca/1410082 |
| 03:45.31 | timeshell | TK ^^^ |
| 03:47.48 | *** join/#asterisk spck (n=spck@75-134-20-74.dhcp.mdsn.wi.charter.com) |
| 03:47.51 | spck | lo |
| 03:48.35 | spck | anyone know if asterisk-gui works with 1.6? |
| 03:49.56 | timeshell | asterisk-gui version 2 SVN does |
| 03:50.02 | timeshell | Mostly |
| 03:50.04 | timeshell | :p |
| 03:51.16 | drmessano | timeshell: Why dont you paste your whole config so someone can actually waste some time trying to help you |
| 03:52.36 | timeshell | drmessano: Ok look. I'm trying to figure out where it's going wrong. Although there is a SIP trunk registered to the remote server, the call doesn't appear to be touching it as nothing shows up in the remote CLI |
| 03:53.20 | timeshell | SO, I'm going back to the local server trying to figure out how the hell it's being directed to the local loopback address |
| 03:53.51 | timeshell | I have 5 trunks going to other servers that work without a problem |
| 03:54.08 | timeshell | It's this one trunk that behaving stupid and it's configured EXACTLY the same as the others. |
| 03:55.14 | drmessano | timeshell: You look. No one is going to take you seriously unless you're willing to pastebin something meaningful. Specific information has been requested and repeatedly ignored by you. |
| 03:55.37 | timeshell | I'm TRYING to FIND something meaningful to PASTE |
| 03:56.14 | pdmmm | whoa |
| 03:56.25 | pdmmm | we need an intervention! |
| 03:56.51 | timeshell | Like, DUH. I'm not sitting around here picking my nose for the last 30 mins. |
| 03:57.15 | spck | i q's about dundi if you want to answer them instead |
| 03:57.20 | spck | er got* |
| 03:57.24 | [TK]D-Fender | timeshell : You should be looking at the registration info on that peer. You see the IP that * sends to and that it is counted as a local interface and simply looping back. I see no networking dump, and I never got the complete call I asked for |
| 03:57.37 | drmessano | ^^^^ |
| 03:57.53 | [TK]D-Fender | is tired of asking for the same thing repeatedly |
| 03:58.12 | [TK]D-Fender | spck: Go ahead and jsut ask, and see who answers. |
| 03:58.41 | timeshell | TK : That is the most helpful answer I've gotten so far. |
| 03:58.53 | drmessano | LOL |
| 03:58.55 | spck | i'm just looking for documentation on dundi in a call center enviroment |
| 03:59.19 | spck | if i can find the doc's i can figure it out, they just seem to be non-existent |
| 03:59.45 | drmessano | [TK]D-Fender: yes, screw your information requests.. keep the educated guesses coming |
| 03:59.51 | *** join/#asterisk CrazyTux (n=brandon@99.53.97.235) |
| 04:00.09 | spck | i'm trying to setup a redundant load balanced system with a combo of realtime mysql and dundi |
| 04:00.18 | [TK]D-Fender | should hire Terry Tate |
| 04:00.35 | pdmmm | haha |
| 04:00.38 | pdmmm | office linebacker! |
| 04:01.06 | [TK]D-Fender | pdmmm: Ain't NOTHIN' going down in MY house!! BIIIIIITTTCCCHH!!! |
| 04:01.18 | [TK]D-Fender | WOOO!!!!!!!!!!!! |
| 04:01.25 | pdmmm | fuck ya |
| 04:01.37 | pdmmm | i need to find that video |
| 04:01.46 | pdmmm | i've been killing myself on java shit all day |
| 04:01.59 | drmessano | [TK]D-Fender: I have a peer.. and the username is ______, tell me, in latin, using 13 letters or less who invented the ice cream and why I have one way audio on my peer |
| 04:02.17 | drmessano | Oh, and you have 10 seconds |
| 04:02.22 | spck | forty-two |
| 04:02.24 | drmessano | Meaningful answers only, please |
| 04:02.38 | spck | ^^ meaning of life |
| 04:02.42 | drmessano | and don't expect a pastebin |
| 04:02.50 | drmessano | ^^ the universe, and everything, as well |
| 04:03.03 | [TK]D-Fender | drmessano: Evinrude |
| 04:03.21 | [TK]D-Fender | (or at least kept it from melting....) |
| 04:03.25 | drmessano | lol |
| 04:03.40 | drmessano | You left off the part about NAT, and the fact I am using newbuntu |
| 04:03.43 | drmessano | Sorry :( |
| 04:03.49 | [TK]D-Fender | drmessano: Thats right. Outboard boat motors owe their existance to a guy who didn't want his ice-cream going soft |
| 04:04.05 | [TK]D-Fender | drmessano: And ice-cream pre-dates Emperor Nero |
| 04:04.21 | [TK]D-Fender | drmessano: Lets just say a hard date ain't happening for soft-ice-cream :) |
| 04:04.25 | drmessano | lol |
| 04:04.42 | [TK]D-Fender | is a massive repository of useless trivia |
| 04:05.04 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 04:05.17 | drmessano | spck: You do know what the last things the Dolphins said before leaving the planet, dont you? |
| 04:05.21 | tfrew | addes [TK]D-Fender to /etc/yum.repos.d/useless.repo with a priority of 100 |
| 04:05.23 | drmessano | thing* |
| 04:05.33 | spck | so long and thanks for all the fish? |
| 04:05.37 | drmessano | Very good |
| 04:05.53 | spck | DON'T PANIC! |
| 04:06.14 | drmessano | Fonality <-- Likely to be the first against the wall when the revolution comes |
| 04:06.30 | spck | i talked my boss out of buying one of those |
| 04:06.39 | drmessano | Then finding out they were indeed the first agains the wall when the revolution came |
| 04:06.42 | spck | bought two dell servers for the same price |
| 04:11.56 | spck | is there much documentation out there for dundi and asterisk? |
| 04:12.35 | drmessano | Not much |
| 04:12.50 | [TK]D-Fender | spck: DUNDI is not a HA solution, its for peering for UNLIKE extensions |
| 04:15.19 | spck | ok |
| 04:15.54 | spck | i work for a medium sized cab company, i have two servers that i would like to load balance, do you have any suggestions? |
| 04:16.48 | jdblack | set up your phones and providers to try both servers. |
| 04:17.31 | spck | won't they always choose the first server then? |
| 04:18.15 | spck | and what about scalability after two servers? |
| 04:18.51 | thehar | spck: there is an 08 astricon presentation on load balancing |
| 04:19.31 | thehar | leif did it i think? |
| 04:22.23 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 04:28.39 | [TK]D-Fender | spck: What exactly are you load balancing? And what is your idea of "medium" sized? |
| 04:29.14 | tfrew | load balancing: opensips handles registration, round robin to a group of asterisk servers, all sharing a unified database |
| 04:30.44 | spck | we have a phone staff that is taking orders for cab rides |
| 04:31.09 | spck | we have written our own dispatching software and are using asterisk-java to interface with the servers |
| 04:31.36 | spck | i guess my operation is small in comparison those i've worked at before |
| 04:31.50 | spck | at most we are going to have 35 phones on the system |
| 04:32.08 | [TK]D-Fender | spck: You might want to try answer my question with a NUMBER. Maybe several. Perhaps even a reasonable guess of those numbers. |
| 04:32.27 | spck | i thought i just did |
| 04:32.28 | [TK]D-Fender | spck: My WATCH could handle you needs.. and its ANALOG <- |
| 04:32.58 | spck | despite the sarcasm i know better |
| 04:33.09 | spck | i'm building this system to cover our needs for the next 10 years |
| 04:33.56 | [TK]D-Fender | spck: * boxes can normalloy support 2 TDM interfaces. Sangoma makes 8 port PRI cards. In E1 mode that is 4 * 30 * 2 = 240 channels. Terminated to PRI |
| 04:34.31 | spck | well i only have 2 pri lines, i know pathetic |
| 04:34.36 | [TK]D-Fender | spck: If you quadruple your system in that time a P4 from years ago is STILL plenty |
| 04:34.49 | tfrew | http://www.voipsupply.ie/products/Redfone-foneBRIDGE2-Dual-T1%7B47%7DE1-Ethernet-Bridge.html |
| 04:34.57 | tfrew | use those with a round robin setup |
| 04:34.57 | [TK]D-Fender | EWWWWWWWWWWWWWW!!!!!!!!!!!!!!!!! |
| 04:35.01 | spck | you know for preaching about answering peoples questions.... |
| 04:35.42 | tfrew | hmm |
| 04:35.51 | [TK]D-Fender | spck: Oh, I've answered the need for load-balancing and that DUNDI isn't really a normal strategy for it. |
| 04:36.24 | [TK]D-Fender | spck: And hopefully give you a sense that your requirements should not require excessive measures within any reasonable time-frame |
| 04:36.45 | [TK]D-Fender | spck: This should be "peace of mind" |
| 04:36.56 | spck | what's a typical load balancing scheme then? |
| 04:39.14 | [TK]D-Fender | spck: Typical scenarios are more ITSP and application server based with SER (or similar) as a RR hub. this completely changes deployment strategies as well with phones normally registering to SER and * only usef for back-end service |
| 04:39.31 | [TK]D-Fender | spck: And we're talking 500+ sized installs here |
| 04:40.34 | spck | RR hub? |
| 04:40.39 | spck | oh round robin |
| 04:40.40 | [TK]D-Fender | Round Robin |
| 04:40.43 | hardwire | road runner! |
| 04:40.50 | [TK]D-Fender | Rural Road? |
| 04:40.55 | hardwire | meep meep |
| 04:41.05 | hardwire | really redonkulous |
| 04:41.38 | spck | well i have to kick ass servers i need to get running by july 1st so this ought to be fun |
| 04:41.42 | spck | er two* |
| 04:42.18 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 04:44.27 | spck | can i run opensips on the same box i'm running * |
| 04:44.42 | spck | ? |
| 04:45.14 | [TK]D-Fender | spck: Yes, typically you'll run * on another port like 5070, etc |
| 04:45.39 | tfrew | or use esx |
| 04:45.45 | tfrew | esxi* |
| 05:21.43 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 05:26.30 | spck | well thanks for the point in the right direction guys |
| 05:28.36 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-240-118-231.hsd1.ky.comcast.net) |
| 05:29.13 | jeffspeff | what is meant by a "skinny session" ? as in Starting Skinny session from 85.190.0.3 and WARNING[30702]: chan_skinny.c:5621 get_input: Skinny Client sent less data than expected. Expected 4 but got 3. |
| 05:31.23 | tfrew | jeffspeff: skinny is cisco phones proprietory protocall |
| 05:31.33 | tfrew | could you possible just flash them to the sip firmware? |
| 05:31.41 | tfrew | that would make them easier to use with asterisk |
| 05:34.05 | *** join/#asterisk Kisu (n=k@ip70-179-88-179.dc.dc.cox.net) |
| 05:40.14 | drmessano | SIP is FAR less useful on Cisco phones |
| 05:40.35 | drmessano | If you can get Skinny working, you can use more of the phones features |
| 05:40.43 | tfrew | like mwi? |
| 05:40.51 | drmessano | Yeah |
| 05:40.57 | tfrew | oh |
| 05:41.07 | drmessano | BLF, etc |
| 05:41.08 | tfrew | i dunno, i spend to much time tweaking polycom xml |
| 05:41.22 | tfrew | i don't look at the other phones enough |
| 05:41.35 | tfrew | acually, i've never touched a cisco phone before |
| 05:41.45 | tfrew | so sorry jeffspeff, i lied :) |
| 05:47.47 | [TK]D-Fender | did once, and still feels dirty |
| 05:48.09 | tfrew | good job |
| 05:51.31 | jeffspeff | i'm not using cisco phones though |
| 05:51.39 | coppice | cisco phones make good paper weights |
| 05:52.12 | jeffspeff | i'm using sjphone (softphone)... and that IP address isn't one that i'm familiar with... maybe something with my provider? |
| 05:54.28 | tfrew | jeffspeff: then set your softphone to use sip |
| 05:54.35 | jeffspeff | it is |
| 05:54.40 | tfrew | or use xlite |
| 05:55.01 | jeffspeff | the ip that it says it's starting a skinny session with isn't my ip address |
| 05:55.03 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 05:55.10 | tfrew | odd |
| 05:55.15 | jeffspeff | the server 192.168.11.41 and my computer is 192.168.11.9 |
| 05:55.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@dslb-088-072-212-085.pools.arcor-ip.net) |
| 05:55.27 | jeffspeff | definately not in my network lol |
| 05:56.27 | jeffspeff | could my number provider (i think they're called a DID provider, correct me if i'm wrong, which i think i am) be using Cisco equipment that would cause that warning? |
| 05:57.08 | tfrew | your voip provider should be passing you a sip trunk |
| 05:57.17 | tfrew | unless your account is set to use skinny |
| 05:57.36 | tfrew | for example, teliax can send either sip or iax out to your device |
| 05:57.47 | jeffspeff | hmmm... i am using teliax |
| 05:58.11 | jeffspeff | do i need to contact them and specify that i want a sip trunk? |
| 05:58.47 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 06:05.08 | jeffspeff | tfrew, do i need to contact them and specify that i want a sip trunk? |
| 06:06.37 | tfrew | i would think so |
| 06:06.53 | tfrew | better yet just get an account with sombody else |
| 06:06.58 | tfrew | teliax, or callcentric |
| 06:07.09 | tfrew | callcentric has good local number unlimited plans |
| 06:14.03 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 06:23.48 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
| 06:30.27 | jeffspeff | tfrew, i am with teliax |
| 06:30.39 | tfrew | set your account to iax |
| 06:30.47 | tfrew | use den.teliax.net |
| 06:30.57 | tfrew | as your proxy |
| 06:31.13 | jeffspeff | i use sip... and den.teliax.net is my proxy |
| 06:31.51 | tfrew | i would recommend iax for teliax |
| 06:32.05 | tfrew | it is somewhat better |
| 06:38.29 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 06:42.12 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 06:43.00 | jeffspeff | tfrew, i'm not all that experience with *, but isn't sip and iax completely different? also, i'm trying to set my CID, and I'm having troubles, if i showed you my extensions.conf could you point me in the right direction? |
| 06:43.31 | tfrew | i don't really read conf files |
| 06:43.47 | tfrew | i print them out into brail and finger them |
| 06:44.14 | jeffspeff | lol |
| 06:44.15 | jeffspeff | http://pastebin.ca/1410156 |
| 06:46.41 | tfrew | i see a problem |
| 06:47.33 | jeffspeff | the "satan <666>" part was a joke that i was just playing on a co-worker/friend who then informed me that my joke was lost because it didn't show that on his CID... it just said "out of area" |
| 06:47.36 | jeffspeff | what is the problem? |
| 06:48.24 | tfrew | that was the problem |
| 06:48.31 | [TK]D-Fender | jeffspeff: maybe the fact Teliax looks at your number and REJECTS it. Tried a valid looking number? |
| 06:48.43 | jeffspeff | lol, let me try |
| 06:51.45 | jeffspeff | ok, with this set " jeffclay_cid=Pioneer Computers <2709080013> " it only shows my number, but not the name |
| 06:51.48 | tfrew | teliax will pass any cid you want |
| 06:51.53 | tfrew | but it has to be valid |
| 06:52.02 | tfrew | teliax won't pass the name |
| 06:52.06 | tfrew | with sip |
| 06:52.09 | tfrew | change to iax |
| 06:52.17 | tfrew | like i was saying... i deal with teliax alot |
| 06:53.06 | jeffspeff | so on their web interface, just change from sip to iax, and i can still use sip on my system and it will work better? |
| 06:53.17 | tfrew | no |
| 06:53.26 | tfrew | you have to setup an iax trunk |
| 06:53.34 | tfrew | and remove the sip trunk |
| 06:53.55 | tfrew | its a pain yes, but you will be happier with the way the system works over iax |
| 06:54.04 | tfrew | you can close all those ports |
| 06:54.10 | tfrew | only need one open with iax |
| 06:54.33 | tfrew | and teliax isn't called tel(iax) for no reason ;) |
| 06:55.35 | jeffspeff | what are the main pros and cons between sip and iax? i thought sip was the mainstream, feature loaded, the thing to use ... you know what i mean.. |
| 06:56.23 | tfrew | sip is the main way nowadays to move voip traffic |
| 06:56.52 | tfrew | but for asterisk to asterisk communication, iax is the way to go |
| 06:57.00 | tfrew | and teliax supports iax |
| 06:57.04 | tfrew | for that reason |
| 06:57.14 | tfrew | it solves alot of problems that sip has |
| 06:57.30 | tfrew | one major being that the media and everything is all in the same packet |
| 06:58.31 | jeffspeff | how different is it from sip for dialplan stuff? |
| 06:59.38 | jeffspeff | on the ten scale where 10 is know's quite a bit, and 0 is none at all... i'd give myself a 3 on SIP and a 0 on IAX... |
| 07:00.15 | tfrew | hey, i'm a freepbx user |
| 07:00.21 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 07:00.30 | tfrew | i don't understand the conf's very much at all :P |
| 07:01.32 | jeffspeff | i'm not trying to flame or anything, but if IAX is so much better (which i'm reading about at http://www.voip-info.org/wiki/view/IAX+versus+SIP ) then why is SIP the mainstream for voip? |
| 07:02.00 | tfrew | because iax is directly tied into asterisk pbx |
| 07:02.10 | tfrew | granted, some other softwares support the protocall |
| 07:02.20 | tfrew | but the majority of systems run over sip |
| 07:02.36 | tfrew | it's kind of the same argument of why microsoft is so big when they suck so hard |
| 07:02.47 | jeffspeff | lol, i understand that one. |
| 07:03.57 | jeffspeff | ok, i plan to have people connect to this server as an off-site pbx for soho's and have their phone register to my server directly... can IAX still do that? |
| 07:06.35 | tfrew | no |
| 07:06.49 | tfrew | you would then still have to have sip ports open on firewall/forwarded |
| 07:07.06 | tfrew | but if asterisk is behine a nat, good luck getting your phones to work behind another nat |
| 07:08.03 | jeffspeff | ok, so i would use IAX trunk to teliax, and then use sip.conf to register sip users for the "customers" ? |
| 07:08.11 | [TK]D-Fender | jeffspeff: Because IAX wasn't an IETF standard, and nobody gave a crap. IAX is not routable, was not made for ENDPOINTS, etc. Inter ASTERISK eXchange. |
| 07:08.30 | tfrew | what d-fender says |
| 07:08.51 | [TK]D-Fender | And there is no reason CID name should pass via IAX and not SIP |
| 07:09.06 | jeffspeff | oh, thanks [TK]D-Fender that actually helps me understand it a lot better |
| 07:09.48 | jeffspeff | so i could use iax trunk to teliax provider, and then have sip channels for users and it would work seemlessly correct? |
| 07:10.17 | tfrew | assuming your asterisk service can handle the amount of active calls |
| 07:10.36 | tfrew | a midgrade dualcore can handle 100-200 calls depending on codec |
| 07:10.48 | [TK]D-Fender | jeffspeff: * processes calls, whats on either end does not particularly matter |
| 07:11.39 | jeffspeff | 2x 3ghz xeons, 4gb ram ECC, 15k scsi hdd's raid 0 |
| 07:12.17 | [TK]D-Fender | RAID 0? EW |
| 07:12.34 | tfrew | change that to raid 10, 1, or 5 |
| 07:12.39 | [TK]D-Fender | jeffspeff: And What are you looking to support over this? |
| 07:12.46 | tfrew | (or 6 possibly) |
| 07:13.14 | jeffspeff | well, i only have 2x 70gb drives for it right now so i wanted as much hdd space as i could atm... i have 6 hot swap bays still open that i plan to use |
| 07:13.29 | tfrew | ok |
| 07:13.34 | tfrew | so use raid 1 with lvm |
| 07:13.45 | tfrew | add drives in blocks of 2 |
| 07:13.59 | tfrew | and just expand your /var/spool/asterisk with lvm |
| 07:14.12 | [TK]D-Fender | jeffspeff: And how much space do you really need? |
| 07:14.15 | tfrew | don't ever use raid 0 on your server |
| 07:14.29 | jeffspeff | [TK]D-Fender, i own a small local computer business that services local business like law firms, doctor offices, and other various ones, and i'm wanting to get them (and others) set up with my system |
| 07:14.31 | [TK]D-Fender | jeffspeff: What are you expecting to pump through your server? |
| 07:14.33 | tfrew | i have a years worth of call data recording all calls, and thats only 30gb |
| 07:16.36 | jeffspeff | not sure on call volume, but hoping to have somewhere around 30 or more users |
| 07:16.46 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 07:17.16 | jeffspeff | when i get ready to roll it out, i'm going with a fiber line 10mb up and 10mb down... that'll only be about $150 USD p/m |
| 07:17.35 | [TK]D-Fender | jeffspeff: .... a P3 could do this |
| 07:18.29 | jeffspeff | yeah, but that's just starting off, i have big (fingers crossed) plans for it... nobody in my area is offering anything close |
| 07:19.10 | jeffspeff | and several of my clients aren't happy with their current POTS based PBX... so i'm gonna go out on a limb and give this a shot |
| 07:20.13 | [TK]D-Fender | jeffspeff: How much you don't know about the scaling of services you're preparing to off your customers is beginning to scare me. |
| 07:21.09 | jeffspeff | [TK]D-Fender, what do you mean? you think i'm trying to "jip" my customers? |
| 07:21.55 | [TK]D-Fender | jeffspeff: No... just that you don't know this going in. Like applying at a garage as a mechanic and asking "How do I do an oil change?" |
| 07:22.07 | jeffspeff | lol |
| 07:22.30 | jeffspeff | i just thought that sip was the way to go, and so sip is what i've been doing |
| 07:23.52 | [TK]D-Fender | jeffspeff: And SIP is fine, in fact many ITSP tend to have audio issues on their IAX termination they don't on SIP. |
| 07:23.58 | jeffspeff | i know that i'm not experienced greatly for this, but i don't plan to charge like i am... one of my clients is willing to go with me as a test run to help me work out the kinks for only the costs that i have |
| 07:24.10 | [TK]D-Fender | jeffspeff: which you would think shouldn't happen, yet does |
| 07:24.47 | *** part/#asterisk tfrew|asleep (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 07:24.51 | coppice | there are far more providers supporting SIP than supporting IAX, so SIP keeps more options open. Swapping between then shouldn't be hard, though |
| 07:26.07 | jeffspeff | [TK]D-Fender, so for what i'm trying to do; should i use all sip or iax trunk with teliax and then sip with clients? |
| 07:26.47 | [TK]D-Fender | jeffspeff: I recommend SIP all around. Easier to debug, more commonly supported, and certainly more stable |
| 07:27.57 | jeffspeff | [TK]D-Fender, ok, thanks... any ideas on how to get my name and number CID to go through on teliax with sip? |
| 07:28.32 | [TK]D-Fender | jeffspeff: CALL THEM |
| 07:28.43 | jeffspeff | lol, will do |
| 07:28.47 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 07:28.48 | [TK]D-Fender | jeffspeff: Its quite doubtful you can even set your name |
| 07:28.56 | coppice | nah, too easy. look for a more arcane way |
| 07:29.28 | jeffspeff | [TK]D-Fender, something that i've been wondering... to get * set up for fax/email and set up for voicemail/email... the e-mail server doesn't have to be setup on the * server does it? i'd hope not, but where are the configs for that? |
| 07:29.32 | [TK]D-Fender | coppice: That kind of ritual sacrifice is a little expensive in this new ecomony. A gota... REALLY... |
| 07:29.35 | [TK]D-Fender | goat* |
| 07:30.20 | [TK]D-Fender | jeffspeff: * calls the standard "sendmail" script. You can sub this as you please |
| 07:30.45 | [TK]D-Fender | jeeAnd NOW you've broung "fax over VoIP" into the equation which puts you in WAY over your head |
| 07:30.49 | [TK]D-Fender | brought* |
| 07:30.55 | [TK]D-Fender | ugh... typing degrading |
| 07:31.54 | *** join/#asterisk ectospasm (i=ectospas@i.hate.your.vhosts.shellium.org) |
| 07:32.13 | jeffspeff | [TK]D-Fender, the * site is offering that free module or whatever (slept several times since i read it last) for faxing... is that not easy to implement? |
| 07:32.39 | jeffspeff | and i've been reading up on faxing, doesn't seem too involved |
| 07:33.00 | ectospasm | jeffspeff: what do you mean "implement"? As in setup? It's not hard, no |
| 07:33.21 | [TK]D-Fender | jeffspeff: T.38 <---- this is not "fax for asterisk". |
| 07:33.38 | ectospasm | T.38 is fax over voip. |
| 07:33.43 | [TK]D-Fender | coppice: Feel free to jump in(s/in/on/) at any time :) |
| 07:33.56 | ectospasm | essentially |
| 07:36.10 | jeffspeff | [TK]D-Fender, http://www.digium.com/en/products/software/faxforasterisk.php |
| 07:37.43 | coppice | <PROTECTED> |
| 07:37.49 | [TK]D-Fender | jeffspeff: You've got a lot or work and reading ahead of you.. |
| 07:38.06 | jeffspeff | i'm fully aware... |
| 07:38.06 | [TK]D-Fender | ok, I'm off, later all |
| 07:38.48 | ectospasm | coppice: I'm told it works, I haven't heard any reports otherwise |
| 07:39.27 | coppice | I've heard from a couple of people who couldn't get it to work, but I don't know who's fault that was |
| 07:40.00 | jeffspeff | what would be the best way to implement fax on a voip system? |
| 07:40.19 | ectospasm | jeffspeff: what version of Asterisk? |
| 07:40.44 | ectospasm | I'd recommend 1.6, since T.38 only works in passthrough mode on 1.4 |
| 07:40.53 | jeffspeff | 1.6.x i downloaded and compiled a few weeks ago |
| 07:40.58 | coppice | there is free fax support built into asterisk 1.6, if you build it on a system with spandsp-0.0.6pre11 installed |
| 07:41.04 | jeffspeff | 1.6.0.9 |
| 07:41.20 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 07:41.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 07:41.41 | jeffspeff | coppice, i saw that in the modules list, but didn't install it... how does that module work? |
| 07:42.06 | ectospasm | coppice: FAX and FFA obviate the need for spandsp. |
| 07:42.41 | jeffspeff | ectospasm, i'm fairly new to *... FFA and spandsp mean nothing to me atm |
| 07:42.50 | coppice | they only give one channel for free. how many will pay when free things work so well |
| 07:43.27 | ectospasm | coppice: even the author of spandsp says it's voodoo magic, he doesn't understand why it works at all. |
| 07:43.34 | jeffspeff | coppice, i'm referring to the free module built into 1.6... |
| 07:43.52 | jeffspeff | coppice, well at least now i am. :P |
| 07:43.54 | coppice | I am the author of spandsp, and I know exactly why it works |
| 07:44.28 | ectospasm | Oh, I stand corrected. I remember reading years ago that it didn't work well in all cases. |
| 07:45.07 | jeffspeff | how does the built in fax module in 1.6 work? |
| 07:45.07 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 07:45.42 | jeffspeff | do you just "email" a fax number with the fax saved as an attachment? |
| 07:46.07 | coppice | Most people who have trouble get it for one of two reasons - their machine timing is screwed up, or their asterisk can send a reliable stream. Digium have finally had to address the second issue in chan_dahdi with a fudge to get their own commercial module out the door. spandsp and attractel gain from this, too |
| 07:46.28 | ectospasm | jeffspeff: set faxdetect in sip.conf and chan_dahdi.conf, have a fax extension to receive inbound faxes, and use ReceiveFAX and SendFAX |
| 07:47.01 | ectospasm | You can write a script that makes a call file and sends the fax when you want to e-mail it. |
| 07:47.27 | coppice | Digium should have released V.34 FAX, and done something the free options can't |
| 07:48.18 | jeffspeff | what about an IP fax machine? i can see a lot of business clients not liking the extra steps required to scan the document, then open it on comptuer and try to e-mail it... they like the simplicity of a fax machine |
| 07:49.06 | ectospasm | sounds unnecessary, and a waste of paper. |
| 07:49.23 | coppice | Most high end combo FAX machine have a T.37 or T.38 option. some cheap ones, like the Brothers, have T.37 |
| 07:49.51 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-212-085.pools.arcor-ip.net) |
| 07:49.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 07:50.48 | jeffspeff | ectospasm, scanning and emailing seems unnecessary to people that are used to puting a paper in the slot and dialing the number... fax/email would be fine for me, but i can see a lot of my business clients not liking it and having a hard time with it |
| 07:51.06 | ectospasm | jeffspeff: they probably printed out the thing in the first place |
| 07:51.19 | jeffspeff | ectospasm, no doubt |
| 07:51.26 | ectospasm | Why not set up a virtual printer that prompts them to fill out the fax cover sheet, then send it without wasting paper? |
| 07:51.59 | ectospasm | no scanning at all (and putting it into the traditional fax machine is scanning it anyway) |
| 07:52.26 | jeffspeff | ectospasm, example: one of my clients is a pain management center... they often times send lots of medical records to other doctors, they'd have to scan and email a large amount of their files |
| 07:52.37 | ectospasm | Worse comes to worse, you could always hook the analog fax machine to an FXS port on an Asterisk machine, and use ReceiveFAX |
| 07:53.14 | ectospasm | jeffspeff: they've got until 2012 to start using electronic medical records (in the US, anyway), might as well start the migration now. |
| 07:53.23 | jeffspeff | lol, true |
| 07:53.55 | jeffspeff | ok, lets entertain this virtual printer idea... |
| 07:53.59 | ectospasm | yes, it's a pain in the behind, but it's gotta be done sometime, waiting just makes it more painful. |
| 07:54.10 | coppice | Medical is one of the strongest markets for FAX, and still growing |
| 07:54.21 | jeffspeff | they are in the process of going paperless... not sure how long down the road it is though |
| 07:54.58 | ectospasm | jeffspeff: I don't know the specifics of writing a virtual printer driver, but it can't be too difficult. |
| 07:55.43 | ectospasm | at least you could write a script that converts it to PDF (or TIFF) and sends it to Asterisk |
| 07:55.54 | jeffspeff | so, they would print the document to the virtual printer, which would then send the (i'm asuming .tiff) image through the * server and then it can be recieved by a regular fax machinge... correct? |
| 07:56.25 | ectospasm | at the far end, yes, assuming the Asterisk machine had a PSTN connection (should be OK with voip, too) |
| 07:56.28 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@dslb-088-072-212-085.pools.arcor-ip.net) |
| 07:58.58 | jeffspeff | ectospasm, i'm using teliax sip trunk |
| 08:00.07 | ectospasm | you may need to look at their documentation or ask them whether they can do T.38 |
| 08:00.16 | ectospasm | if they can, you're in business. |
| 08:00.44 | jeffspeff | ectospasm, and this is with the built in fax module in 1.6 correct? |
| 08:01.18 | ectospasm | built-in? FAX isn't built in, you have to install it separately, get a license, etc. |
| 08:01.27 | ectospasm | even FFA you need a license key. |
| 08:02.19 | jeffspeff | when i was in make menuslect, i saw a fax module... and i believe coppice referred to it earlier as well |
| 08:02.32 | jeffspeff | fax module or feature |
| 08:02.47 | ectospasm | not the same as Digium's offering. |
| 08:03.05 | ectospasm | ...that's probably the spandsp stuff |
| 08:03.13 | jeffspeff | yes it was |
| 08:03.13 | ectospasm | Which I (obviously) know little about. |
| 08:03.31 | jeffspeff | the one Digium's offering works quite a bit better i assume? |
| 08:03.56 | ectospasm | it has commercial backing, ymmv whether it works better. |
| 08:04.50 | ectospasm | disclaimer: I'm biased, as I work for Digium |
| 08:04.58 | jeffspeff | oh, np |
| 08:05.43 | jeffspeff | what about recieving a fax... does that just go to an e-mail address specified in the context? |
| 08:06.18 | ectospasm | not exactly |
| 08:06.38 | jeffspeff | can you elaborate? |
| 08:07.01 | ectospasm | it gets sent to the fax exten in your incoming context, where you need to ReceiveFAX(), then you can throw it wherever you need using dialplan or system scripts. |
| 08:07.18 | ectospasm | (probably both) |
| 08:08.00 | ectospasm | You can use dialplan logic and functions (FAXOPTS) to help figure out where you need to send it. |
| 08:10.25 | jeffspeff | forgive me, but i'm completely new to pbx systems faxing.. 1) do you have to have seperate numbers? for fax and voice or does the system auto distinguish between the two? 2) I assume you can route the fax based on the number it's sent to in relation to what e-mail address recieves; correct? |
| 08:10.49 | jeffspeff | *between the two on the same number |
| 08:11.37 | ectospasm | what do you mean "separate numbers?" Different DIDs? It would be helpful, each user could have their own FAX DID, or their own DID, which faxdetect determines when there's a fax there. |
| 08:11.53 | ectospasm | There's a manual for the Fax for Asterisk stuff |
| 08:12.26 | ectospasm | I'm not at work so I can't download it to check, but you can create a login on digium.com and look at it yourself. |
| 08:12.35 | coppice | where is that manual? |
| 08:12.50 | ectospasm | coppice: one sec |
| 08:14.04 | ectospasm | https://www.digium.com/en/supportcenter/documentation/viewdocs/FAX |
| 08:15.51 | ectospasm | well, it's getting late, and I need to go to bed. Catch y'all l8r... |
| 08:16.06 | jeffspeff | ectospasm, i'm wanting to setup a remote sip proxy for soho's (mostly doctors offices, law firms, and a few other varying business), if Joes Law Firm has one DID, they can use that # for both fax and voice, and * will detect when a fax comes in on SIP and send to the appropriate e-mail address; and Jans Chiropractic can do the same with a single DID. correct? |
| 08:17.01 | ectospasm | jeffspeff: if I'm not mistaken about how the fax detection works. I admit I'm no expert. |
| 08:17.42 | jeffspeff | ok, thanks for your help. I'm getting off too. it's 3:17am here |
| 08:19.37 | *** join/#asterisk af_ (n=getsmart@88-149-230-139.dynamic.ngi.it) |
| 08:21.55 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 08:25.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 08:33.59 | coppice | ectospasm: it looks like res_fax is very similar in use to the existing FAX stuff, except for the FAXOPTS stuff |
| 08:51.53 | *** join/#asterisk bionoid (i=terje@mesyah.com) |
| 09:04.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 09:06.58 | bionoid | [May 1 12:48:24] ERROR[3485] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied <-- My log is full of these, but the file exists and is 660 asterisk:asterisk - any idea what I might have done to acheive this? ;-) |
| 09:13.44 | *** part/#asterisk abchirk (n=rapunzel@cl-2502.ham-01.de.sixxs.net) |
| 09:15.29 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:35.55 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 09:51.48 | *** join/#asterisk nirs (n=nirs@dslb-088-072-212-085.pools.arcor-ip.net) |
| 09:56.21 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 09:58.17 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:18.08 | chiwawa_42 | Hi ! I have a CID support issue on an OpenVox A400P FXS port. I could fix it on a PAP2T by ajusting the ring waveform, frequency and voltage, but I can't find out where to change these settings on the zaptel configuration (or zonedata.c). Where may I change these settings ? |
| 10:19.17 | chiwawa_42 | Also, is there any comprehensive documentation about the tones settings syntax, if possible covering both zaptel and linksys ATAs ? |
| 10:42.53 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 10:46.26 | ruben23 | hi i got asterisk server on my network behnd firewall and i have a remote agent wants to login to my asterisk by softphones, how do i setup this that the remote user can login as regular SIP client on my network. |
| 10:54.55 | Superbartt | port forwarding the sip and rdp ports? |
| 10:57.10 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 11:04.54 | ruben23 | Superbartt:on my gateway/firewall box..thats all..? |
| 11:06.44 | Superbartt | hmm, could be that the other end also has to forward some ports to get it working properly |
| 11:07.05 | Superbartt | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions read this |
| 11:15.57 | ruben23 | <PROTECTED> |
| 11:16.05 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 11:22.49 | *** join/#asterisk missinglink (n=missingl@CPE-58-161-41-204.nsw.bigpond.net.au) |
| 11:26.21 | missinglink | my server crashed, does aircrack-ng keep a log if it finds a password correctly? |
| 11:26.40 | missinglink | oops |
| 11:26.43 | missinglink | sry |
| 11:39.01 | *** join/#asterisk missinglink (n=missingl@58.161.41.204) |
| 11:48.25 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 11:54.09 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:00.10 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 12:13.29 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 12:45.06 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 12:48.12 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:48.12 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 12:53.38 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 13:00.03 | *** join/#asterisk juanIMP (n=Juancho@201.245.254.74) |
| 13:04.35 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:04.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:12.18 | *** join/#asterisk rOfLz (n=unit90@119.160.20.227) |
| 13:18.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:20.06 | *** join/#asterisk outtolunc (n=me@166.128.239.4) |
| 13:22.39 | rOfLz | I am having problem when calling out call get establish but no voice on both ends ( this would not be a nat problem coz I do not have any router or firewall coz both asterisk and user are on same lan) |
| 13:23.49 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 13:24.57 | *** join/#asterisk moy (n=moy@66.78.97.66) |
| 13:27.30 | outtolunc | i needed a good morning laugh, thanks |
| 13:28.11 | *** join/#asterisk moy (n=moy@66.78.97.66) |
| 13:35.19 | *** join/#asterisk moy (n=moy@66.78.97.66) |
| 13:37.30 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 13:39.20 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:39.20 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:42.48 | rOfLz | I am having problem when calling out call get establish but no voice on both ends ( this would not be a nat problem coz I do not have any router or firewall coz both asterisk and user are on same lan) |
| 13:46.10 | nirs | r0flz, are you sure you are using the proper codecs on all sides ? |
| 13:46.25 | nirs | can you please paste into pastebin.com your configs so we can look at these ? |
| 13:49.34 | rOfLz | I have set disallow=all then allow=ulaw and alaw |
| 13:49.44 | nirs | ok, sounds about right |
| 13:49.57 | nirs | are you sure you don't have a NAT problem in there ? |
| 13:50.09 | nirs | are you using FreePBX to configure? or hand configurations ? |
| 13:51.30 | rOfLz | coz I am using trixbox machine which uses asterisk 1.4 |
| 13:51.35 | juanIMP | rOfLz: have you already definde localnet |
| 13:52.14 | rOfLz | I have tried localnet as well but condition was same |
| 13:53.13 | juanIMP | in trixbox, where did you defined it?? |
| 13:53.29 | rOfLz | nirs: I have trixbox and client on same lan there is no firewall or router within both |
| 13:53.38 | rOfLz | in sip.conf |
| 13:54.03 | juanIMP | you must do it in sip_custom.conf rOfLz |
| 13:54.06 | rOfLz | and in sip_nat.conf as well |
| 13:54.17 | rOfLz | hmm |
| 13:54.25 | juanIMP | sip.conf is generated by trix-box |
| 13:55.30 | rOfLz | so which conf file will asterisk use sip_custom.conf or sip.conf |
| 13:55.44 | juanIMP | both of them |
| 13:56.30 | rOfLz | do I need to define localnet in sip_custom.conf as well ? |
| 13:56.40 | juanIMP | in sip.conf there is an include of sip_custom.conf |
| 13:57.00 | juanIMP | yep in sip_custom.conf |
| 13:57.52 | rOfLz | juanIMP: lemme try this but do u have any other clue ? |
| 13:59.06 | juanIMP | nope :( , have you check the CLI? |
| 13:59.50 | juanIMP | full log??? everery thing seems to work properly? |
| 14:00.19 | juanIMP | the codecs in the phone or softphone are correct? |
| 14:01.36 | juanIMP | you shoult only use the codecs of sip peers in the phones. |
| 14:09.14 | *** join/#asterisk moy (n=moy@66.78.97.66) |
| 14:15.30 | juanIMP | does it works rOfLz ?? |
| 14:15.48 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 14:18.30 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 14:26.45 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:35.52 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:45.22 | *** join/#asterisk CrazyTux (n=brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 14:55.07 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 15:01.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 15:07.05 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-212-085.pools.arcor-ip.net) |
| 15:07.05 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:08.42 | *** part/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 15:15.01 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 15:16.08 | *** join/#asterisk columbo (n=PeterFal@pool-71-177-208-38.lsanca.dsl-w.verizon.net) |
| 15:19.51 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 15:23.41 | *** join/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 15:44.41 | *** part/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 15:49.48 | Kobaz | hmm |
| 15:50.07 | Kobaz | it seems that GoSub needs an exact match on the exten? |
| 15:50.47 | Kobaz | it doesn't look like it does pattern matching |
| 15:58.34 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 15:58.36 | pmhaddad-work | oh sendmail how i hate thee |
| 16:07.02 | *** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar) |
| 16:07.25 | *** join/#asterisk columbo (n=PeterFal@pool-71-177-208-38.lsanca.dsl-w.verizon.net) |
| 16:16.15 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 16:23.28 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 16:31.02 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
| 16:35.36 | *** join/#asterisk crevetor (n=crevetor@bas1-montreal07-1176436428.dsl.bell.ca) |
| 16:36.29 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 16:43.44 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:45.11 | *** join/#asterisk jetole (n=Joe@204.13.0.100) |
| 16:57.19 | chiwawa_42 | Hi ! I have a CID support issue on an OpenVox A400P FXS port. I could fix it on a PAP2T by ajusting the ring waveform, frequency and voltage, but I can't find out where to change these settings on the zaptel configuration (or zonedata.c). Where may I change these settings ? |
| 16:57.23 | chiwawa_42 | Also, is there any comprehensive documentation about the tones settings syntax, if possible covering both zaptel and linksys ATAs ? |
| 17:09.44 | *** join/#asterisk puzzled (n=foobar@5353384C.cable.casema.nl) |
| 17:11.42 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 17:18.47 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:21.45 | *** join/#asterisk jshas (n=psuka@220-241.187-72.tampabay.res.rr.com) |
| 17:23.31 | *** join/#asterisk crevetor (n=crevetor@bas1-montreal07-1176499941.dsl.bell.ca) |
| 17:24.10 | crevetor | hi |
| 17:24.19 | crevetor | I have a really weird issue |
| 17:24.47 | crevetor | I have an asterisk server directly on the internet with peers behind nat |
| 17:25.01 | crevetor | Thos peers all have nat=yes, qualify=5000 |
| 17:25.18 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 17:25.40 | crevetor | when I do a sip show peers they all appear with th right IP, status as OK (with the response time) |
| 17:26.29 | crevetor | But if I do a SIP debug I see that asterisk is sending options packet to their rfc1918 IP |
| 17:27.10 | crevetor | but also to the public IP behind which they are |
| 17:36.57 | pmhaddad-work | anyone know why i need a seperate authenticate priority for meetme in 1.6.0.9? |
| 17:37.21 | pmhaddad-work | MeetMe(400,i,1234) just will not work |
| 17:37.36 | pmhaddad-work | MeetMe(400,ip,1234) doesnt work either |
| 17:38.04 | pmhaddad-work | i have to do 400,1,Authenticate(1234) 400,n,MeetMe(400,ip) |
| 17:39.36 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-3-57.wi.res.rr.com) |
| 17:40.29 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 17:46.33 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 17:59.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:10.12 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:10.42 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 18:12.10 | *** join/#asterisk eliel (n=eliels@120-17-235-201.fibertel.com.ar) |
| 18:27.25 | *** join/#asterisk Gnewt (n=vector@207.115.69.54) |
| 18:27.52 | Gnewt | My DISA is not working... I have an inbound number from IPKall, which worked for a while... but now when I call in it seems that it doesn't recogize any DTMF tones. |
| 18:28.03 | Gnewt | I tried DISA and an IVR to ring my extensions, both unresponsive |
| 18:28.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:31.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@dslb-088-072-212-085.pools.arcor-ip.net) |
| 18:37.03 | *** join/#asterisk juanIMP (n=Juancho@201.245.254.74) |
| 18:44.32 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 18:47.23 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 18:55.01 | *** join/#asterisk jtodd (n=jtodd@dslb-088-072-212-085.pools.arcor-ip.net) |
| 18:55.01 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:57.25 | jeffspeff | anybody here have any luck sending name and number CID with Teliax SIP ? I can only send number... |
| 19:01.22 | *** join/#asterisk hi365 (n=hi365@94.159.178.171) |
| 19:01.43 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:02.01 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 19:12.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:17.24 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 19:35.01 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.118.136) |
| 19:35.36 | DelphiWorld | hello |
| 19:35.44 | DelphiWorld | please do you provide trixbox support here ? |
| 19:36.11 | drmessano | Nope |
| 19:36.41 | tzafrir_laptop | try #trixbox for starters.. |
| 19:36.51 | DelphiWorld | drmessano: ok |
| 19:37.17 | DelphiWorld | tzafrir_laptop: no responce in it |
| 19:37.58 | [TK]D-Fender | DelphiWorld: Not our problem. |
| 19:38.00 | drmessano | Their forums are the preferred |
| 19:38.07 | drmessano | trixbox.org |
| 19:39.24 | *** join/#asterisk jsidhu (n=chatzill@c-24-23-223-14.hsd1.ca.comcast.net) |
| 19:49.20 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 19:52.34 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279582078.dsl.bell.ca) |
| 19:53.48 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:05.46 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 20:06.47 | kn0x | i cant get sangoma wanpipe to compile |
| 20:06.58 | kn0x | dahdi source is installed and compiled without a hitch |
| 20:08.54 | kn0x | i get http://pastebin.ca/1410626 |
| 20:09.07 | kn0x | error: âstruct dahdi_spanâ has no member named âechocanâ |
| 20:15.05 | kn0x | [TK]D-Fender: you around? |
| 20:15.45 | [TK]D-Fender | kn0x: I'd recommend using the version their WIKI recommends. 3.3.16 |
| 20:16.22 | kn0x | i thought thats wat i d/ld |
| 20:16.23 | kn0x | sec |
| 20:16.30 | kn0x | ohh |
| 20:16.31 | kn0x | shit |
| 20:16.32 | kn0x | okay |
| 20:16.47 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:16.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:20.08 | Emrah86 | yo... when I checkout asterisk-addons 1.4 and install it, i can't find cdr_addon_mysql.so in /usr/asterisk/modules Do you have any idea how i can load this module. I've already tried to load it from modules.conf, but no succes... |
| 20:20.57 | [TK]D-Fender | Emrah86: Go prove it compiled. Then verify that you actually ran the install following that. |
| 20:24.30 | *** join/#asterisk hi365 (n=hi365@94.159.177.166) |
| 20:27.10 | Emrah86 | [TK]D-Fender: what do you mean by compiled? I just downloaded the last version from the site, make / ./configure / make install ... still the same :( |
| 20:27.37 | [TK]D-Fender | Emrah86: Go look that the .so was even built |
| 20:30.58 | Emrah86 | [TK]D-Fender: no he doesnt build that .so |
| 20:31.45 | [TK]D-Fender | Emrah86: Then you'r probably missing a dependency that would be stated rather clearly if you did make menuconfig |
| 20:32.33 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:36.42 | Emrah86 | [TK]D-Fender: you are right... i get Depends on: mysqlclient(E) |
| 20:36.57 | Emrah86 | [TK]D-Fender: thanks for the info, ive learnt something new :-) |
| 20:43.16 | Emrah86 | [TK]D-Fender: fixed, thnx again for your tip |
| 20:46.53 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 20:51.13 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 20:51.46 | kn0x | [TK]D-Fender: same error with 3.3.16 source |
| 20:52.38 | *** join/#asterisk jsidhu (n=chatzill@c-24-23-223-14.hsd1.ca.comcast.net) |
| 21:01.26 | *** join/#asterisk telecos (n=sergio@42.166.219.87.dynamic.jazztel.es) |
| 21:03.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:04.02 | KyleK | does "sip set debug" show packets going both ways or just outgoing? |
| 21:09.26 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:10.51 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:15.07 | [TK]D-Fender | KyleK: both |
| 21:15.16 | *** join/#asterisk goofy03 (n=kvirc@146.110.203-77.rev.gaoland.net) |
| 21:15.23 | goofy03 | Hi |
| 21:16.18 | goofy03 | is there nvfax moduule for asterisk 1.6.1 |
| 21:16.23 | goofy03 | ? |
| 21:18.55 | ambush276 | hey |
| 21:19.08 | ambush276 | can anyone help |
| 21:19.13 | ambush276 | is it SIP or extension problem?? |
| 21:19.15 | ambush276 | http://pastebin.ca/1410675 |
| 21:21.56 | ambush276 | nybody? |
| 21:23.37 | russellb | you have no peer in sip.conf called "sam" |
| 21:27.57 | goofy03 | chan_dahdi.c: Fax detected, but no fax extension |
| 21:28.47 | goofy03 | i have iax extension on it ? |
| 21:28.58 | [TK]D-Fender | goofy03: Means what it says. Your channel detected a fax tone, but you don't have a fax exten in the current context |
| 21:29.43 | goofy03 | but s ok in freepbx |
| 21:30.04 | [TK]D-Fender | goofy03: FreePBX is NOT supported in here |
| 21:30.08 | [TK]D-Fender | ~freepbx |
| 21:30.09 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:32.23 | goofy03 | ok |
| 21:32.31 | goofy03 | is there nvfax moduule for asterisk 1.6.1 ? |
| 21:33.18 | goofy03 | where to download the source ? |
| 21:35.14 | [TK]D-Fender | goofy03: Last I heard that module was abandoned by its authors t the start of * 1.4 |
| 21:36.24 | KyleK | gah I got * to make a sip call with the external IP, but its moving the RTP stream from the external IP to an internal IP |
| 21:38.29 | KyleK | hey would "module reload chan_sip.so" reload sip.conf? |
| 21:40.50 | _ShrikE | KyleK: sip reload |
| 21:42.57 | KyleK | thx |
| 21:44.11 | *** join/#asterisk CrazyTux (n=brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 21:45.49 | *** join/#asterisk amaache (i=amaache@196.46.252.50) |
| 21:49.52 | amaache | Hi |
| 21:50.52 | amaache | I have TDM400 -22 with 2 FXO 2 FXS but i dont have the signal tonality in the FXS |
| 21:52.39 | *** join/#asterisk InHisName (n=InHisNam@c-68-80-23-194.hsd1.pa.comcast.net) |
| 21:53.39 | InHisName | I need to capture a call with dial tones, messages etc to an audio track. How do I do that? |
| 21:53.48 | InHisName | In my asterisk. |
| 21:54.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:54.18 | InHisName | Like to start at off hook and end at on hook |
| 21:58.46 | hi365 | Qwell: ping |
| 22:03.10 | kn0x | anyone else have trouble compiling sangoma wanpipe drivers? |
| 22:13.54 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 22:14.57 | leifmadsen | InHisName: use Monitor() |
| 22:15.24 | leifmadsen | use it prior to do Dial() without the 'b' (only record when bridged) option |
| 22:15.54 | leifmadsen | you'll need to do immediate=yes and capture the dtmf though in order to trigger the dialplan on a DAHDI channel |
| 22:15.58 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 22:16.37 | *** join/#asterisk ingenius (n=alektro@OL77-237.fibertel.com.ar) |
| 22:16.59 | leifmadsen | so basically you'll have to do immediate=yes, which will then generate a tone (fake dialtone), and then you can enter the dtmf from the phone, and send the numbers along to dial() |
| 22:17.11 | leifmadsen | runs off to dinner |
| 22:24.34 | InHisName | Thanks, leifmadsen. Now to read further as soon as I find where I save my manuals. |
| 22:31.05 | *** join/#asterisk ^GoKu-^ (n=aambriz@187.134.46.21) |
| 22:31.19 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 22:33.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:36.13 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-229-16.telecom.net.ar) |
| 22:56.25 | *** join/#asterisk tfrew (n=rendwe@c-68-57-89-103.hsd1.va.comcast.net) |
| 23:05.29 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:05.57 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:07.17 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:07.26 | *** join/#asterisk hi365 (n=hi365@94.159.177.166) |
| 23:11.40 | kn0x | sangoma drivers wont compile under debian lenny |
| 23:12.19 | kn0x | make[2]: *** [/usr/src/wanpipe-3.4.1/kdrvtmp/sdla_tdmv.o] Error 1 |
| 23:12.29 | kn0x | and compilation FAILZ :( |
| 23:15.10 | tfrew | pastebin the whole ./configure and make |
| 23:20.36 | kn0x | troubled: ok |
| 23:20.40 | kn0x | err |
| 23:20.42 | kn0x | tfrew: |
| 23:21.36 | kn0x | http://pastebin.ca/1410758 |
| 23:22.13 | tfrew|afk | k |
| 23:23.38 | tfrew|afk | did you install a source deb or get the tgz from asterisk |
| 23:23.49 | kn0x | 1.6.1 lastest tarball |
| 23:24.19 | tfrew|afk | and the latest for dahdi |
| 23:24.31 | kn0x | yeup |
| 23:25.25 | kn0x | ive tried many different versions of the wanpipe driver |
| 23:25.47 | kn0x | the 3.5 3.4.1 3.3.16 3.3.16.22 |
| 23:25.53 | kn0x | all fail @ same error |
| 23:25.59 | kn0x | i dont know what im missing -_- |
| 23:28.02 | tfrew|afk | odd |
| 23:28.06 | tfrew|afk | try the latest svn |
| 23:28.30 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:31.09 | KyleK | exten => 123,n,Dial(SIP/line1,60) <-- if I have an FXO hit the 123 extension when it gets a call, does * answer the call while its dialing SIP/line1? |
| 23:31.23 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 23:31.26 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:34.39 | tfrew|afk | no |
| 23:35.20 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:40.13 | *** join/#asterisk [gquit]bombadil (n=dana@70.94.3.57) |
| 23:42.14 | *** join/#asterisk pmhaddad-lappy (n=pmhaddad@24-247-43-183.dhcp.mrqt.mi.charter.com) |
| 23:43.56 | *** join/#asterisk AJayMN (n=root@24-159-234-93.dhcp.mdsn.wi.charter.com) |
| 23:44.45 | AJayMN | I installed 1.4 over a previous 1.2 it was working, but noticed some of my custom dial stuff wasnt so reinstalled 1.2 and its loaded.. Only problem is Conference rooms are not working |
| 23:44.55 | AJayMN | when you go into one it hangs up.. i dont know what to check |
| 23:45.05 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 23:45.27 | drmessano | Zaptel |
| 23:45.48 | AJayMN | would zaptel change just by downloading 1.4 and addons and installing them? |
| 23:45.59 | drmessano | You need 1.4 Zaptel for 1.4 |
| 23:46.03 | AJayMN | usally if zaptel isnt working it would just say that is in invalid conference room |
| 23:46.15 | AJayMN | Im trying to get 1.2 working again.. |
| 23:46.20 | drmessano | I know |
| 23:46.32 | drmessano | Do you want the answers, or do you have them? |
| 23:46.35 | InHisName | leifmadsen: I tried to set it up. Got the voice message captured but not tones etc. Used monitor() and immediate=yes, but not the other. not sure what that is yet or where to put it. |
| 23:46.39 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 23:46.39 | *** mode/#asterisk [+o file] by ChanServ |
| 23:46.51 | drmessano | Check your Zaptel |
| 23:47.05 | AJayMN | dont want to be stupid but how? :P |
| 23:47.35 | drmessano | Did you install Zaptel 1.4 during your brief foray into 1.4? |
| 23:47.44 | AJayMN | no |
| 23:47.54 | AJayMN | thats why i cant figure out what happen |
| 23:48.03 | drmessano | How did you "reinstall" 1.2? |
| 23:48.07 | drmessano | Source? |
| 23:48.14 | AJayMN | yes |
| 23:48.16 | *** join/#asterisk propellerhead (n=yogurt2u@host66.190-229-16.telecom.net.ar) |
| 23:48.33 | AJayMN | removed the modules directory so as not to have 1.4 modules trying to load in 1.2 |
| 23:48.41 | drmessano | ok |
| 23:48.52 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:48.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:48.59 | drmessano | I would reinstall Zapte; |
| 23:49.01 | drmessano | I would reinstall Zaptel |
| 23:49.19 | AJayMN | k |
| 23:49.34 | AJayMN | does zaptel have a module? |
| 23:49.42 | AJayMN | i didnt delete the dir only renamed it |
| 23:51.07 | drmessano | Yes, but theres much more to it |
| 23:51.12 | drmessano | Kernel modules |
| 23:51.19 | drmessano | err.. drivers |
| 23:52.20 | AJayMN | dang :P |
| 23:53.33 | drmessano | It takes like under a minute to reinstall |
| 23:53.39 | drmessano | Whats the deal? |
| 23:54.29 | AJayMN | <PROTECTED> |
| 23:54.43 | AJayMN | last time i screwed with this.. my server locked up and had to have the datacenter restart it |
| 23:54.54 | AJayMN | its the weekend :( no one there.. $100 emergency call |
| 23:54.59 | AJayMN | thats why im worred :P |
| 23:55.30 | drmessano | Well, dunno what to tell ya |
| 23:56.11 | *** join/#asterisk kumarphilly (n=kumar@66.160.197.30) [NETSPLIT VICTIM] |
| 23:57.19 | drmessano | If Zaptel causes that much of a problem with your server, maybe you need to find out why its unstable and fix that first |
| 23:58.57 | AJayMN | its not zaptel it was when i did the kernel update |
| 23:59.03 | AJayMN | it snag'd at reboot |
| 23:59.41 | drmessano | Ok, but we're not updating the kernel |