| 00:00.21 | ibob63 | Thanks BillinOffice |
| 00:00.26 | [TK]D-Fender | [av]bani : And you're the one who has to set them all up, huh? |
| 00:00.33 | [av]bani | yep, all my toys |
| 00:01.06 | *** part/#asterisk ibob63 (n=hp@bb-87-82-24-7.ukonline.co.uk) |
| 00:01.39 | [av]bani | you know, its funny people will talk about how great the polycom sound is.. but you can only do so much with ulaw |
| 00:01.49 | [av]bani | people talk like OMG ITS CD QUALITY |
| 00:02.06 | [TK]D-Fender | [av]bani : there also general crackle in the handset, jitter, etc. |
| 00:02.15 | [TK]D-Fender | believe me... there can be a difference |
| 00:02.30 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 00:02.31 | BillinOffice | fender <- what's better? |
| 00:02.38 | [av]bani | well, the gxp2000 doesnt have jitter or crackle... though the speakerphone is too quiet |
| 00:02.42 | [av]bani | but it doesnt sound 'shit' |
| 00:03.06 | BillinOffice | I've got both on my desk, but the Polycom pleases me more... |
| 00:03.13 | BillinOffice | maybe it's just my ear but... |
| 00:03.13 | [TK]D-Fender | there is also acoustic feedback and other factors |
| 00:03.24 | BillinOffice | I haven't used the pcom's much yet. |
| 00:03.25 | [av]bani | .13 fixed the feedback |
| 00:03.37 | [av]bani | which was only on the speakerphone |
| 00:03.42 | BillinOffice | I think I'm running .9 on the GXP |
| 00:03.55 | [av]bani | not that i'd use speakerphone on it anyway... volume is too low |
| 00:03.57 | _Sam-- | they say there is a new firmware coming |
| 00:04.01 | [av]bani | thats the only real negative about it |
| 00:04.03 | BillinOffice | on the pcom? |
| 00:04.14 | [av]bani | like i keep saying, its amazing amount of phone for $80 |
| 00:04.18 | *** join/#asterisk robin_z (n=yeah@host-212-18-247-190.static.mailbox.co.uk) |
| 00:04.18 | _Sam-- | on the gxp..new firmware http://www.voip-info.org/wiki/view/GXP-2000 |
| 00:04.24 | robin_z | morning girls! |
| 00:04.44 | [TK]D-Fender | [av]bani : For $80 I'd say yes. For a company to USE though, I'd say they are worth more than GXP's |
| 00:04.53 | robin_z | _Sam--: unfortunately even with new firmware, it will still be crap |
| 00:05.00 | BillinOffice | is the .13 FW generally avail? |
| 00:05.02 | [av]bani | depends on your company, i can totally see a boiler room operation being very pleased with piles of these |
| 00:05.14 | [av]bani | :) |
| 00:05.22 | robin_z | [av]bani: yes, at least they will burn OK |
| 00:05.37 | [TK]D-Fender | robin_z : however they will have the LATEST crap! |
| 00:05.44 | robin_z | the GXP-200 sucketh bigtime |
| 00:05.45 | _Sam-- | i have 15 people who do incomng phone sales all day them on fine |
| 00:05.50 | [av]bani | they're cheap, definitely have cheap feel to it, but not cardboard |
| 00:05.55 | [av]bani | its 'ok' |
| 00:06.02 | BillinOffice | yes - it feels cheap |
| 00:06.03 | robin_z | well, mine is shite |
| 00:06.10 | BillinOffice | the handset needs some lead |
| 00:06.17 | [av]bani | yea, its light |
| 00:06.18 | robin_z | I consider not hanging up when the rx is put on hook a fault |
| 00:06.31 | robin_z | weight will not help .. desing fault. |
| 00:06.32 | _Sam-- | it is what it is...you cant compare a hyundai to a ferari...but that doesnt meant he hyundai isnt good for waht it is |
| 00:06.43 | _Sam-- | thats all i say about the phone |
| 00:06.46 | [av]bani | its more like a honda civic |
| 00:06.52 | robin_z | well, sure, but if the wheels fall off when you get in, its not actually useful as a car |
| 00:06.52 | [TK]D-Fender | BillinOffice : same problems the SPA-841 had.. hence the reason for an SPA-941 |
| 00:07.08 | [av]bani | robin_z: wheels are fine here! |
| 00:07.14 | BillinOffice | I've heard nice things about the 941. Haven't tried either. |
| 00:07.28 | [av]bani | its no cisco, but its no cheapy chinese POS either, its somewhere in between :) |
| 00:07.31 | BillinOffice | should I upgrade to .13? |
| 00:07.40 | _Sam-- | my tires roll, they could probably use an alignment, but they dont bother me that much |
| 00:07.42 | [av]bani | its marginally above shit, marginally below decent |
| 00:07.43 | robin_z | my GXP crashes randomly ... the hook switch fits into the "ear" recess of the handset, so does not actually hang up .. never can, never will |
| 00:08.05 | BillinOffice | mine has crashed too. pretty lame. |
| 00:08.07 | [TK]D-Fender | The GXP is more like a riced-up Ford Escort :D The SPA-941 is a Civic, Polycom IP 50x = Corolla, IP 60x = Camry. |
| 00:08.14 | robin_z | nah |
| 00:08.21 | [av]bani | no crashes here... |
| 00:08.22 | _Sam-- | when you say crash, you mean its locked up? |
| 00:08.23 | robin_z | ford escorts where a solid reliable workhorse |
| 00:08.27 | _Sam-- | mine has locked up. |
| 00:08.33 | _Sam-- | had to power cycle it (them) before |
| 00:08.35 | [av]bani | only on firmware upgrade |
| 00:08.35 | [TK]D-Fender | robin : reliable shit is still shit :) |
| 00:08.40 | BillinOffice | Yeah - locked up - need power cycle - mid conversation |
| 00:08.50 | _Sam-- | never seen mid conversation myself |
| 00:08.53 | [av]bani | what firmware? we using .13 here |
| 00:08.55 | robin_z | anyway .. I just bought 25 Snom 360s and 320s |
| 00:09.04 | [av]bani | BillinOffice: you using poe? |
| 00:09.06 | robin_z | oh, on the subject of shit ... |
| 00:09.07 | _Sam-- | we are .13 here, finding some bugs from what my guys are telling me |
| 00:09.18 | BillinOffice | no poe - wall wart |
| 00:09.23 | robin_z | avoid the Zyxel WiFi phone .. prestiges 2000? |
| 00:09.28 | [av]bani | BillinOffice: .13 ? |
| 00:09.31 | robin_z | you might as well use a turd |
| 00:09.32 | BillinOffice | .9 |
| 00:09.37 | [av]bani | tried .13 ? |
| 00:09.44 | [av]bani | .9 wasnt usable for me, too many bugs |
| 00:09.51 | _Sam-- | i have one of those utstarcomm cheap wifi phones |
| 00:09.52 | [av]bani | speakerphone echo |
| 00:09.52 | BillinOffice | no - don't have .13 --- been sitting out... |
| 00:10.03 | BillinOffice | waiting to hear others' war stories on .13... |
| 00:10.04 | [av]bani | its fine here, been using .13 for the past 2 weeks |
| 00:10.05 | [av]bani | no lockups |
| 00:10.08 | robin_z | my zyxel is none-working with * |
| 00:10.11 | BillinOffice | there were reported issues on asterisk-users |
| 00:10.17 | robin_z | and cant upgrade over ftp |
| 00:10.24 | robin_z | or web |
| 00:10.33 | robin_z | so .. chocholate fireguard |
| 00:10.35 | BillinOffice | speaker echo on my GXP is atrocious |
| 00:10.46 | [av]bani | yes, its .9 bug. no AEC on speakerphone. fixed in .12 |
| 00:10.59 | BillinOffice | I have the zyxel prestige 2000 |
| 00:11.11 | _Sam-- | my sales guys have reported, and ive seen it myself...where they go back to a call that was on hold on .13 and they go on hook...but the call isnt on the earpiece, they have to hang up again and pick it up again |
| 00:11.11 | BillinOffice | I like the sound, but the user interface sucks |
| 00:11.16 | [av]bani | robin_z: 24 snom 360s :() from where? |
| 00:11.50 | [av]bani | _Sam--: local hold on the gxp, or parked on * ? |
| 00:11.58 | _Sam-- | yes, local host on gxp |
| 00:11.59 | _Sam-- | hold |
| 00:12.01 | _Sam-- | sorry |
| 00:12.09 | [av]bani | i can imagine that, local stuff on gxp seems not so good |
| 00:12.17 | [av]bani | local conf for example, a bit noisy |
| 00:12.18 | *** join/#asterisk _deg_ (n=deg@201.22.27.49.adsl.gvt.net.br) |
| 00:12.24 | [av]bani | i dont think the mixing is very high quality |
| 00:12.28 | robin_z | can I tell you lot why I just LOVE the 'net? |
| 00:12.36 | BillinOffice | yeah? |
| 00:12.41 | *** join/#asterisk exstatica (i=exstatic@redline.mednor.net) |
| 00:12.50 | [av]bani | ive been planning on making conf's go to * instead |
| 00:13.27 | BillinOffice | gotta go.. |
| 00:13.28 | BillinOffice | bye |
| 00:13.32 | robin_z | well .. here I am in my factory in UK, configuring a remote * box in Geneva, feeding my industrial laser with metal and talking to you lot half way around the world ... |
| 00:13.33 | _Sam-- | cya |
| 00:13.52 | robin_z | thats pretty fscking weird when you think about it ... |
| 00:13.58 | [av]bani | only thing i havent figured out on the gxp2000 yet is MOH |
| 00:14.11 | robin_z | [av]bani; let * do moh |
| 00:14.24 | _Sam-- | i dont have any problems with it |
| 00:14.32 | [av]bani | i put em on hold and ... silence |
| 00:14.34 | robin_z | moh is an exchange thing, not a phone thing ... |
| 00:14.35 | _Sam-- | i mean, * plays the music on hold, and it works fine fos the gxp at my place |
| 00:14.43 | robin_z | good. |
| 00:14.48 | [av]bani | robin_z: the gxp supposedly does its own moh |
| 00:14.53 | robin_z | yeah, bad plan |
| 00:14.59 | _Sam-- | mine use * |
| 00:15.04 | _Sam-- | never seen a setting for anything in the phone |
| 00:15.09 | _Sam-- | to do it from the phone |
| 00:15.11 | robin_z | theres a URL I think ... |
| 00:15.14 | robin_z | daft idea |
| 00:15.14 | [av]bani | neither have i, so im trying to figure out why |
| 00:15.22 | [av]bani | why it wont use * moh |
| 00:15.31 | robin_z | anyway ... fsck that :) |
| 00:15.36 | _Sam-- | you sure its just those phones? |
| 00:15.38 | robin_z | so .. back to agi .... |
| 00:15.46 | robin_z | wtf does not my agi work huh? |
| 00:15.47 | [av]bani | some phones let you point at a moh url |
| 00:15.56 | _Sam-- | robin: what do you feed into your laser? i have some parts that i need a cnc to make |
| 00:16.15 | robin_z | _Sam--: anything up to 10mm plate steel |
| 00:16.24 | _Sam-- | could it do billet aluminum? |
| 00:16.30 | robin_z | up to 3mm |
| 00:16.47 | robin_z | I have cnc mill too |
| 00:16.54 | robin_z | and cnc press |
| 00:16.59 | robin_z | and powder coat |
| 00:17.04 | robin_z | and ... etc |
| 00:17.05 | _Sam-- | i am in the motorcycle parts and accessories biz |
| 00:17.09 | robin_z | ok |
| 00:17.10 | _Sam-- | i have a specific part i need |
| 00:17.14 | robin_z | you in UK? |
| 00:17.17 | _Sam-- | nope |
| 00:17.21 | robin_z | dang :) |
| 00:17.25 | [TK]D-Fender | I've found that in life there's very little you can't do with really REALLY BIG LASERS. Social-political problems? ZAP! Hamburger a little cold? ZAP! |
| 00:17.30 | _Sam-- | you have all this stuff for yourself? |
| 00:17.41 | robin_z | _Sam-- yeah |
| 00:17.49 | _Sam-- | do you know anything about sportbikes? |
| 00:17.57 | robin_z | [TK]D-Fender: oh I only have teeny little laser .. 2kw output or so ... |
| 00:18.32 | robin_z | _Sam--: I raced RGV250s for a few years, have a GSXR600 and run one of the UK largest motorcycle mailing lists ... so .. yeah, a bit ;) |
| 00:18.49 | _Sam-- | hmm check me out... www.kneedraggers.com (shameless self promotion) |
| 00:18.56 | _Sam-- | we are one of the largest in the US |
| 00:19.06 | _Sam-- | but i think you could help me out maybe |
| 00:19.08 | robin_z | _Sam--: google "ixion" |
| 00:19.38 | _Sam-- | i race some r6's myself :) |
| 00:20.03 | robin_z | _Sam--: I have lots of US contacts in CNC, so can probably sort you out if not myself |
| 00:20.11 | robin_z | wait, sheet change. |
| 00:21.41 | robin_z | back |
| 00:21.44 | _Sam-- | nice |
| 00:21.56 | robin_z | running 100 sheets through ... |
| 00:21.59 | _Sam-- | this is a really big idea...but its so simple its like post it notes. |
| 00:22.04 | _Sam-- | bigger than frame sliders. |
| 00:22.13 | _Sam-- | what year is your gixxer? |
| 00:22.21 | robin_z | 99 |
| 00:22.31 | _Sam-- | you need to step into the fuel injected era :) |
| 00:22.36 | robin_z | nah ... |
| 00:22.47 | robin_z | its way fast enough for the road |
| 00:22.52 | robin_z | and I dont race anymoe |
| 00:23.01 | *** join/#asterisk someunixguy (n=chatzill@static-69-95-184-175.har.choiceone.net) |
| 00:23.01 | Err | it's retarded that motorcycles aren't *all* fuel-injected these days |
| 00:23.05 | _Sam-- | i mostly just do track now |
| 00:23.10 | _Sam-- | Err: most are |
| 00:23.11 | Err | it *has* to be cheaper to build fuel-injection systems than to build carbs |
| 00:23.15 | robin_z | I mostly just do work now :) |
| 00:23.28 | _Sam-- | luckily i get to do both at the same time :) |
| 00:23.34 | [TK]D-Fender | Hey : how do I check to see if ZTDUMMY is loaded right? |
| 00:23.35 | robin_z | I have a '76 ducati too ... |
| 00:23.45 | Err | not most - many of the "race" bikes are, but I doubt 50% of all motorcycle models are |
| 00:23.52 | *** join/#asterisk fdask (i=fdask@CPE0013d479c929-CM0011e6edd218.cpe.net.cable.rogers.com) |
| 00:23.54 | fdask | hi |
| 00:23.57 | *** join/#asterisk ke4qqq (n=dad@GV-DYN-130.globalvision.net) |
| 00:24.06 | Err | (perhaps 50% of total sales - I don't know - but no bike I've ever been interested in was FI) |
| 00:24.09 | _Sam-- | i think "most" bikes above 7000 dolalrs USD |
| 00:24.10 | robin_z | Err: for 90% of riders its irrelevant, as they cant actually ride |
| 00:24.39 | _Sam-- | most of the japanese bikes of 600cc or greater are all FI these days |
| 00:24.41 | Err | robin_z: it's not about speed - it's about fuel economy, efficiency, and ease of maintenance |
| 00:24.58 | _Sam-- | and i dont know enough about harleys to make any guess |
| 00:25.01 | Err | (to me, anyway) |
| 00:25.11 | Err | few harleys are FI - only the really expensive ones, or as an option |
| 00:25.21 | Err | (not that I'm interested in a Harley, either ;-) |
| 00:25.29 | robin_z | Err: crap. its about image. thats what sells. economy? from a sports 600? dont make me laugh ... and even with carbs no one does maint |
| 00:25.55 | Err | well, I do, and I care ;-) |
| 00:25.58 | pifiu | can anyone tell me what this means? |
| 00:25.58 | pifiu | http://pastebin.ca/38284 |
| 00:26.12 | robin_z | people buy FI becase its 4bhp on last years model .. |
| 00:26.26 | robin_z | pifiu: its a url link to a pastebin item |
| 00:26.32 | _Sam-- | lol |
| 00:26.36 | pifiu | lmao |
| 00:26.45 | pifiu | the error to that link on the pastebin |
| 00:26.52 | robin_z | oh .. THAT |
| 00:26.57 | pifiu | =P |
| 00:27.22 | robin_z | you are using voipgate.com? |
| 00:27.47 | *** topic/#asterisk by twisted[asteria] -> Asterisk 1.2.2 has been released! -//- http://www.asterisk.org/ || happy birthday Cresl1n! |
| 00:27.56 | [TK]D-Fender | pifiu : mean your IAX peer is not reachable (no IP, bad route, etc) |
| 00:28.05 | robin_z | anwyay .. it means it cant open the outgoing channel |
| 00:28.08 | robin_z | as TK said |
| 00:28.21 | robin_z | fscking voipgate had no reachabel DNS today. |
| 00:28.23 | robin_z | winkers/ |
| 00:29.09 | robin_z | OK, so AGI scripts ... sigh ... |
| 00:29.28 | pifiu | no not using voipgate |
| 00:29.37 | robin_z | * says its trying to run it .. and the path is right |
| 00:29.40 | pifiu | hmmm interesting wtf though i can IAX fine |
| 00:29.48 | robin_z | its chmod 755 |
| 00:29.55 | robin_z | but it doesnt get run ... |
| 00:30.07 | robin_z | cleus? |
| 00:30.20 | file | twisted[asteria]: are you taking him out? |
| 00:30.20 | robin_z | can I run an AGI from the console? |
| 00:30.25 | file | twisted[asteria]: for the best night of his life?!? |
| 00:30.27 | *** join/#asterisk BillinOffice (n=bill@dsl092-234-029.phl1.dsl.speakeasy.net) |
| 00:30.41 | robin_z | wait .. sheet change |
| 00:31.19 | [TK]D-Fender | twisted[asteria] : How do I test to see if ZTDUMMY is loaded properly? I'm trying to get a MEETME up and running, and I've more than double checked my conf#, and PINS. they are fine, but its refusin. I'm getting dev/zap/pseudo errors |
| 00:32.15 | fdask | lsmod? |
| 00:32.18 | robin_z | back |
| 00:33.01 | [TK]D-Fender | fdask : I see zaptel, but not ztdummy |
| 00:33.03 | robin_z | TK modprobe ztdummy? |
| 00:33.16 | [TK]D-Fender | says it can't find it |
| 00:33.22 | fdask | modprobe ztdummy? |
| 00:33.23 | robin_z | opsy |
| 00:33.30 | fdask | is your kernel 2.6? |
| 00:33.34 | [TK]D-Fender | 2.4.31 |
| 00:33.38 | robin_z | OK, so you built it?? |
| 00:33.39 | [TK]D-Fender | Slackware stock |
| 00:33.41 | fdask | otherwise you'll have to make some changes in the zaptel Makefile so it compiles ztdummy |
| 00:33.52 | robin_z | by defualt its not built |
| 00:33.59 | robin_z | you need a timing source too |
| 00:34.03 | [TK]D-Fender | fdask : So it doesn't compile it in by default? |
| 00:34.07 | [TK]D-Fender | ahhh |
| 00:34.07 | fdask | not on 2.4 |
| 00:34.16 | fdask | but doesn't the 2.4 use some usb chipset or something to get the timing info? |
| 00:34.16 | robin_z | what USB do you have? |
| 00:34.22 | [TK]D-Fender | Never needed to rely on it till now... |
| 00:34.25 | robin_z | usb uhci |
| 00:34.30 | [TK]D-Fender | rob, I have OCHI and UCHI |
| 00:34.33 | [TK]D-Fender | I'm good for it |
| 00:34.40 | robin_z | both? |
| 00:34.40 | [TK]D-Fender | just didn't know it didn't compile by default |
| 00:34.57 | [TK]D-Fender | oops, just ochi |
| 00:34.58 | fdask | you wont need ztdummy on 2.4 if your using uchi, do you? |
| 00:35.04 | robin_z | fuxxored |
| 00:35.06 | *** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net) |
| 00:35.09 | [av]bani | OHCI and UHCI |
| 00:35.25 | robin_z | AFAIK its UHCI or no play |
| 00:35.28 | [TK]D-Fender | <PROTECTED> |
| 00:35.31 | [TK]D-Fender | :/ |
| 00:35.44 | robin_z | by a zap card :) |
| 00:35.45 | [TK]D-Fender | Awww bugger |
| 00:35.51 | robin_z | or upgrade to 2.6 |
| 00:36.00 | [TK]D-Fender | robin_z : I DITCHED my tdm22b |
| 00:36.11 | robin_z | uses the pseudo RT thing on 2.6 |
| 00:36.16 | [av]bani | is ztdummy even used in 2.6 ? |
| 00:36.23 | [TK]D-Fender | I really didn't want to screw with my kernel... |
| 00:36.33 | fdask | i had to load ztdummy to get rid of some warning messages in my logs |
| 00:36.56 | [av]bani | <PROTECTED> |
| 00:36.59 | pifiu | hey ok here is another one, which i think is the reason why i am having problems? |
| 00:36.59 | [av]bani | :/ |
| 00:37.00 | pifiu | http://pastebin.ca/38285 |
| 00:37.28 | robin_z | pifiu: umm because something is wrong? |
| 00:38.04 | pifiu | ok thanks robin |
| 00:38.45 | robin_z | im not being very helpful am I? |
| 00:38.51 | pifiu | no |
| 00:39.03 | fdask | anyone used a voicetronix openline card? |
| 00:39.23 | robin_z | I was almost helpful to TK to make up for it though |
| 00:39.37 | fdask | ive got an openline4 card hooked up, everythings fine save for a problem with the volume |
| 00:39.46 | fdask | can't get a decent volume level out of the card :\ |
| 00:39.59 | robin_z | sheet change ... |
| 00:41.04 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 00:42.07 | robin_z | so AGI ... |
| 00:42.26 | [TK]D-Fender | robin_z : Yeah, I seem beset upon by this little annoyance. |
| 00:42.53 | [TK]D-Fender | robin_z : I've never had to mess with my kernel before and I don't want to fsck up this box. its my home gateway and media server |
| 00:42.54 | robin_z | ztdummy? |
| 00:43.03 | robin_z | yeah .. its PITA |
| 00:43.15 | [TK]D-Fender | robin_z : Yes. in order to get it working it lokos like I'm going to need to go to 2.6 |
| 00:43.24 | robin_z | we actually chose our server to make sure it had UHCI so we could run meetme |
| 00:43.54 | *** join/#asterisk Soul (n=Soul@87-196-11-251.net.novis.pt) |
| 00:43.55 | robin_z | then we got a zap card and the problem solved itself |
| 00:44.53 | *** join/#asterisk NeonLevel (n=NeonLeve@dsl-201-129-171-113.prod-infinitum.com.mx) |
| 00:45.38 | *** join/#asterisk usam (n=alx@203.156.48.73) |
| 00:46.18 | robin_z | so ... |
| 00:46.21 | robin_z | Launched AGI Script /var/lib/asterisk/agi-bin/fax.pl // Run sfftobmp and mail it. |
| 00:46.21 | robin_z | <PROTECTED> |
| 00:46.36 | robin_z | and yes, it does exist and is chmod 755 |
| 00:46.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:46.43 | robin_z | but it never gets run :( |
| 00:46.49 | twisted[asteria] | um |
| 00:46.57 | twisted[asteria] | / comments are not valid in the dialplan |
| 00:47.04 | robin_z | ahh. |
| 00:47.13 | twisted[asteria] | use ; comments instead |
| 00:47.14 | fdask | :) |
| 00:47.20 | robin_z | fscking silly me copying and pasting from the readme |
| 00:47.29 | robin_z | silly README in chan_capi |
| 00:47.38 | robin_z | thanks twisted[asteria] |
| 00:47.41 | twisted[asteria] | np |
| 00:47.54 | twisted[asteria] | occasionally i glance over the channel and become useful for about .5 sec |
| 00:48.09 | NeonLevel | hi everybody, i'm using * as a sip server and i have two linksys pap2-na registered to this *, both linksys are on diferent networks behind linux nat's, the call goes ok so far, what i want to do is that the "media" goes straigth from one linksys to the other WITHOUT passing it through *, is this possible? thanks in advance! |
| 00:48.13 | twisted[asteria] | then my eyes glaze over and my phone rings |
| 00:49.31 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:52.59 | *** join/#asterisk edobefe (n=shigueta@69.65.149.166) |
| 00:53.59 | edobefe | hi, can an ip phone get access to more than one line with asterisk? i mean can it get calls from several POTS lines or just from one? |
| 01:01.38 | [TK]D-Fender | edobefe : depends on the phone. And don't necessarily think of phones having "lines" per se so much as the possibility of multiple simultaneous calls |
| 01:02.36 | *** join/#asterisk benjk (n=benjamin@nat.bolo.net) |
| 01:04.54 | benjk | ~seen zoa |
| 01:04.59 | jbot | zoa is currently on #asterisk (3d 16h 49m 52s). Has said a total of 229 messages. Is idling for 10h 38m 19s, last said: 'tzanger, i think most things can be solved with very cheap changes'. |
| 01:05.50 | NeonLevel | hi everybody, i'm using * as a sip server and i have two linksys pap2-na registered to this *, both linksys are on diferent networks behind linux nat's, the call goes ok so far, what i want to do is that the "media" goes straigth from one linksys to the other WITHOUT passing it through *, is this possible? thanks in advance |
| 01:06.14 | *** join/#asterisk troy (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 01:06.29 | [av]bani | NeonLevel: you want to look at 'canreinvite' |
| 01:06.39 | edobefe | [TK]D-Fender: but multiple calls means more than one at the same time, but how about multiple lines routed not at the same time to a single phone? |
| 01:06.47 | [av]bani | though if theyre behind nat, it probably wont work |
| 01:06.59 | NeonLevel | thanks [av]bani |
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| 01:13.40 | [TK]D-Fender | edobefe : Lines go into *, calls go out to phones. think of it that way |
| 01:14.05 | [TK]D-Fender | edobefe : You can make a call go from anywhere to anywhere. Its all up to your dial-pland and phone setup |
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| 01:14.13 | edobefe | [TK]D-Fender: hmm, ok, so any line can reach any phone |
| 01:15.13 | [TK]D-Fender | edobefe : All depending on your setup. |
| 01:15.36 | edobefe | [TK]D-Fender: ok, understood |
| 01:15.50 | [TK]D-Fender | You can have it so 1 line gets an IVR which does (whatever), another line rings a specific phone (or more), a third just dials your Cell phone, ec. |
| 01:16.30 | *** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 01:17.23 | [TK]D-Fender | edobefe : My home setup for instance pick up the CID of the caller, then just rings all of my phones at once then leads to VM if not answered. Basically it feels like an answering mahcine to callers. however I have access to a full PRI on top of my 1 analog line so I have up to 24 calls/in/out possible at a time. |
| 01:18.59 | Mag1KaL | Hm, I got festival/asterisk intergration to work on my box but it sounds horrible, and I don't mean the quality of the tts itself. |
| 01:19.24 | fdask | what do you mean |
| 01:20.24 | Mag1KaL | It sounds kind of distorted... the compression must be doing something weird... |
| 01:20.58 | joat | squeak/too fast? |
| 01:22.00 | Defraz | Here is something weird, I just upgraded from 1.0 astrisk to 1.2 and I have a pri into the system, when the call came in on the 1.0 asterisk it just said the number was disconnected or not in service when people called a not routed number(DID) |
| 01:22.02 | *** join/#asterisk Beastie- (n=nick@66-215-122-202.dhcp.atsc.ca.charter.com) |
| 01:22.11 | Defraz | well on 1.2 it rings the first extention on my system. |
| 01:22.47 | Defraz | I don't rmember setting anything up to do that on 1.0 am I missing something on 1.2? |
| 01:23.01 | Mag1KaL | Uhh... it works fine now... |
| 01:23.33 | Mag1KaL | Last night it sounded like crap but now after I rebooted my machine it's fine. |
| 01:27.03 | Beastie- | does anyone know much about the soft phone idefisk |
| 01:27.28 | [TK]D-Fender | Defraz : that sounds perfectly normal. Basically * is not answering the call so the telco says "not valid" |
| 01:27.43 | [TK]D-Fender | Defraz : You should account for them somehow... |
| 01:28.18 | Defraz | Well that is what I wnated but now it calls the first extention. |
| 01:28.24 | Defraz | I rather it just do what it did on 1.0 |
| 01:28.58 | Defraz | It seemed like it did that out of the box where in 1.2 do I have to set something? |
| 01:29.31 | *** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net) |
| 01:30.52 | dudes | Defraz - post your extensions/zapata.conf on pastebin |
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| 01:32.25 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-212-180-41.mia.bellsouth.net) |
| 01:33.39 | Defraz | okay what is that url again? |
| 01:33.49 | Defraz | of which distro? |
| 01:33.57 | Defraz | I mean version of asterisk? |
| 01:33.59 | [TK]D-Fender | ~pb |
| 01:34.00 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 01:34.09 | Defraz | the 1.0 or 1.2 |
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| 01:35.45 | *** join/#asterisk j_vianna (n=joaovian@207.190.206.161) |
| 01:36.18 | j_vianna | Hi guys! I need to buy some DID in USA ? |
| 01:36.38 | j_vianna | Can you guys sugest a good place to buy those ? |
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| 01:41.03 | Math` | j_vianna: how many DIDs do you need |
| 01:41.56 | Zodiacal | anyone know where this file is off hand? dialparties.agi |
| 01:42.00 | Zodiacal | path |
| 01:42.13 | Zodiacal | im new to linux obvously :P |
| 01:42.16 | Zodiacal | how do i search for files? |
| 01:42.48 | [TK]D-Fender | Zodiacal : Thats also an AMP question... something else that's not really supported here... |
| 01:42.54 | [TK]D-Fender | #amportal |
| 01:43.01 | Zodiacal | oic, thanks! |
| 01:43.01 | [TK]D-Fender | and |
| 01:43.03 | [TK]D-Fender | #linx |
| 01:43.07 | Zodiacal | k |
| 01:43.09 | [TK]D-Fender | #linux |
| 01:43.39 | j_vianna | Math: I need about 46 DID in USA and four 800 # |
| 01:44.53 | Math` | j_vianna: can I pm you? |
| 01:44.57 | someunixguy | I've been lurking for a while but since the subject of AMP came up.... just wondering what the general consensus is on it. Anyone at the intermediate/advanced level using it? |
| 01:45.07 | fdask | what is AMP? |
| 01:45.16 | someunixguy | AMP=Asterisk Management Portal |
| 01:45.16 | Math` | fdask: its called asterisk management portal |
| 01:45.23 | j_vianna | Math: sure. |
| 01:45.30 | someunixguy | (Web config w/template files, etc.) |
| 01:46.55 | [TK]D-Fender | someunixguy : AMP like all other * GUI's is EVILhttp://www.junghanns.net/downloads/bristuff-0.2.0-RC8q.tar.gz. The only time its at all validated is in large installs where you need some sort of central management that isn't a PITA and aren't linus-friendly. |
| 01:47.13 | [TK]D-Fender | Skip that url that pasted in there. |
| 01:47.15 | [TK]D-Fender | heh |
| 01:47.23 | someunixguy | lol |
| 01:47.36 | someunixguy | I can't imagine using it in a large install. |
| 01:48.45 | someunixguy | I haven't seen Thirdane's product but imagine it's similar. Too bad there isn't something good at assisting with the management of a clean config file. |
| 01:49.10 | [TK]D-Fender | someunixguy : Its called "vi" :) |
| 01:49.20 | file[laptop] | or emacs |
| 01:49.24 | justinu | bah |
| 01:49.24 | Math` | thats the best interface I've ever seen |
| 01:49.27 | justinu | emacs... what a joke |
| 01:49.27 | [TK]D-Fender | someunixguy : Goes by other accepted names too ;) |
| 01:49.27 | Math` | (vim) |
| 01:49.34 | [TK]D-Fender | I use MC, so :P |
| 01:49.37 | file[laptop] | let's not get into this... |
| 01:49.40 | Math` | lol |
| 01:49.41 | justinu | no, lets |
| 01:49.52 | file[laptop] | [TK]D-Fender: I like you, so you're allowed to use it! |
| 01:49.55 | someunixguy | I'm ashamed to say I'm a pico/nano man |
| 01:50.09 | file[laptop] | if it works for you - use it. |
| 01:50.13 | justinu | fair enough |
| 01:50.25 | someunixguy | True enough, whatever works should be the motto. |
| 01:50.51 | Err | indeed - although file[laptop] is right :-P |
| 01:51.11 | someunixguy | I've run in to two AMP installs I've been asked to clean up in the last two weeks... |
| 01:51.21 | justinu | how do you disable the screen jumping in emacs? |
| 01:51.31 | justinu | i can't use it like that |
| 01:51.33 | Err | I don't even know what "screen jumping" means |
| 01:51.47 | justinu | it means that the screen doesn't scroll one line at a time, like every other editor on the planet |
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| 01:51.56 | someunixguy | I think I've developed a brain lesion from all those circular includes they've got going on ... |
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| 01:52.09 | Err | it's not a pager, it's a text editor |
| 01:52.13 | Dr-Linux | justinu :@ |
| 01:52.15 | Err | that said, I'm sure it's adjustable - everything is |
| 01:52.30 | [TK]D-Fender | someunixguy : So how many times DOES it take you till you learn? ;) |
| 01:52.32 | justinu | well, no one I know can tell me how to change it |
| 01:52.50 | robbyt | hey guys, so what advice do you have on sugarcrm/asterisk intergration |
| 01:53.16 | justinu | besides, your argument makes no sense, because emacs acts like a pager, not a text editor |
| 01:53.17 | robbyt | without doing a lot of fancy database magic |
| 01:53.26 | robbyt | :) |
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| 01:53.44 | Err | justinu: a minute with google yielded scroll-step |
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| 01:53.52 | someunixguy | [TK]D-Fender: I'm sure by next week I'll be unable to string a sentence together and will be able to say nothing but "AMP AMP AMP ..." |
| 01:53.53 | Err | (setq scroll-step 1) turns it off |
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| 01:54.06 | justinu | err gets a cookie |
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| 01:54.27 | someunixguy | Seriously though, AMP client #1 is going back to square one and I'm building them a real config tomorrow. |
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| 01:55.05 | Newbie___ | hi all, anyone has experience installing h323 in Asterisk Version 1.2.2 |
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| 01:56.54 | j_vianna | Math`: ??? |
| 01:57.05 | [TK]D-Fender | someunixguy : So how many times DOES it take you till you learn? ;) |
| 01:57.09 | Math` | j_vianna: did you got my pms? |
| 01:57.18 | [TK]D-Fender | I helped un-AMP a guy once..... |
| 01:57.24 | Err | j_vianna: that's your problem, not his - you need to register your nick |
| 01:57.33 | someunixguy | un-AMP... lol |
| 01:57.35 | robbyt | hey guys, i'm sort of a noob- why is amp so bad?? |
| 01:57.38 | [TK]D-Fender | And convince several others that they are better off amintaining their IQ's |
| 01:57.49 | Math` | ah your nick isnt reg'd |
| 01:57.53 | justinu | amp is bad because it makes the dial plan seem very complex |
| 01:58.06 | someunixguy | I'm sure AMP has its place but it's murder if you're not intimate with it. |
| 01:58.10 | [TK]D-Fender | robbyt : * gives you control, AMP takes it away and ticks us off when people using it come here asking for help on it. |
| 01:58.22 | Math` | j_vianna: try /msg nickserv register [put your password here] |
| 01:58.27 | [TK]D-Fender | And if you're intimate with it welll... we won't go there... |
| 01:58.33 | someunixguy | haha |
| 01:58.35 | Dr-Linux | ticks :S |
| 01:58.48 | [TK]D-Fender | You don't debug AMP... you just wait till the complaints mount an a new version comes out.... |
| 01:58.51 | robbyt | hah |
| 01:59.15 | Err | most UI frontends to programs are like that |
| 01:59.16 | robbyt | yeah, i have noticed a few bugs with amp |
| 01:59.16 | [TK]D-Fender | Real dialplans are pretty easy, I see little use for it. |
| 02:00.03 | robbyt | well, thing is |
| 02:00.10 | robbyt | it gives clients easy access |
| 02:00.13 | robbyt | to modify things |
| 02:00.21 | robbyt | and it gives noobs like me a great starting point |
| 02:00.49 | alephcom_ | Not another gui war, please? |
| 02:00.55 | robbyt | that's the hardest thing about asterisk w/o amp, where to even start |
| 02:01.49 | someunixguy | True enough. When I cut my teeth on it voip-info.org wasn't around. |
| 02:02.05 | someunixguy | I had to troll the mailing list for a month before I had a clue. |
| 02:02.21 | Beastie- | does anyone know what might cause this error? pbx.c: Timeout, but no rule 't' in context 'outgoing' |
| 02:02.31 | drumkilla | If you pay attention to the coverage Asterisk gets around the net, it's obvious that AMP and Asterisk@Home get Asterisk into the hands of many that never would have touched it otherwise |
| 02:02.58 | robbyt | well, that's argueably a good thing and a bad thing, drumkilla |
| 02:03.18 | someunixguy | Not to start a flame war, but we have a lot of people getting a bad idea about Asterisk as a result of inept installations... (echo cancellation, etc.) |
| 02:03.41 | robbyt | it does lower the bar of entry quite a bit |
| 02:03.51 | drumkilla | indeed |
| 02:03.53 | robbyt | but if you have idiots install your cisco call manager, it's going to drop calls |
| 02:03.57 | robbyt | i've seen it happen |
| 02:04.14 | robbyt | if you have idiots clean your office, they're going to steal your leftovers out of the fridge |
| 02:04.17 | robbyt | i've seen that happen too |
| 02:04.17 | someunixguy | Those idiots are on the phone with cisco who fixes their problem, so call manager doesn't get a bad rep |
| 02:04.52 | robbyt | true enough |
| 02:04.58 | someunixguy | There are too many arguments in either direction so it comes out even I suppose |
| 02:05.09 | robbyt | well, the diffrence is |
| 02:05.10 | Err | heh, if crisco fixes your call manager box over the phone, you're paying them enough that you ought to have a competent admin to egin with :-) |
| 02:05.22 | Err | s/egin/begin/ |
| 02:05.24 | robbyt | that asterisk = freedom |
| 02:05.27 | robbyt | :D :D |
| 02:05.30 | someunixguy | Now if Cisco could fix that *!*#@* sip image... |
| 02:05.32 | Math` | crisco? :P |
| 02:06.10 | someunixguy | we've been testing SIP loads for weeks with them and they keep getting worse |
| 02:06.10 | robbyt | i'm sure if you payed mark enough, he'd come over to your house and write your dialplan for you |
| 02:06.30 | robbyt | v7 cisco phones? |
| 02:06.34 | someunixguy | Yeah |
| 02:06.35 | robbyt | v7 sip |
| 02:06.37 | justinu | what's wrong with the sip software? |
| 02:06.45 | someunixguy | 7.5 has a lot of problems |
| 02:06.58 | someunixguy | do 10-12 blind transfers and the phone dies |
| 02:07.01 | robbyt | i have 7.4 on my test phones, otherwise i have all polycom 501s |
| 02:07.04 | Err | justinu: did you see my solution to your emacs question before I was disconnected? |
| 02:07.12 | someunixguy | 7.4 is fine in medium load |
| 02:07.16 | robbyt | 35x polycoms, 3 ciscos |
| 02:07.17 | justinu | err: yep, you deserve a cookie! |
| 02:07.33 | someunixguy | we're on 7.9.21 and it's crap |
| 02:07.34 | Dr-Linux | 7.4 works fine for me |
| 02:07.37 | robbyt | so how does load effect the phones? |
| 02:07.53 | someunixguy | well, memory leaks get worse the more call volume you have |
| 02:08.02 | someunixguy | the blind transfer problem, for example, builds |
| 02:08.06 | robbyt | in the phones themselves? |
| 02:08.10 | someunixguy | yeah |
| 02:08.11 | robbyt | ok |
| 02:08.13 | someunixguy | the phone's UI |
| 02:08.14 | shmaltz | is there anyway that I can see in the CLI the global var? |
| 02:08.36 | robbyt | unixguy: so they eventually lockup then eh? |
| 02:08.51 | someunixguy | no, actually they get freakin weird |
| 02:09.04 | someunixguy | all sorts of odd lines and buttons vanish |
| 02:09.05 | justinu | did you put them on the reboot program then? |
| 02:09.19 | robbyt | huh- so tell me |
| 02:09.20 | someunixguy | nah, 6.5 seems fine |
| 02:09.24 | robbyt | i was thinking of using the chan_skinny |
| 02:09.27 | someunixguy | 7.4 is ok too |
| 02:09.38 | robbyt | or chan_sccp |
| 02:09.45 | robbyt | what are your thoughts on that? |
| 02:09.58 | someunixguy | I haven't tried it myself, but I understand the call features are a little limited |
| 02:10.25 | someunixguy | it's a reverse engineering situation, so i think they have enough stuff in there to allow it to work |
| 02:10.31 | someunixguy | but not much else |
| 02:10.33 | robbyt | (nod) |
| 02:11.00 | someunixguy | we run about 150 79xxs with sip give or take a few |
| 02:11.28 | Mag1KaL | I'm trying to execute a agi perl script but every time I do asterisk is telling me No such file or directory :| |
| 02:11.29 | someunixguy | we've only had a few issues with 7.5 with a couple operators |
| 02:11.47 | robbyt | so you can't downgrade eh? |
| 02:11.49 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 02:11.54 | someunixguy | Mag1KaL: make sure your file is in /var/lib/asterisk/agi-bin |
| 02:12.10 | someunixguy | We've taken those few phones to 6.5 |
| 02:12.25 | someunixguy | they're find on that load but they don't like the NTP feed so no clock |
| 02:12.41 | robbyt | hah |
| 02:12.44 | someunixguy | (too bad for those users... can't have it all right!) |
| 02:13.02 | robbyt | "do you want a clock, or a phone that works?" |
| 02:13.08 | robbyt | clock! |
| 02:13.11 | someunixguy | lol |
| 02:13.15 | someunixguy | you got that right |
| 02:13.24 | someunixguy | "How do I know when I missed a call?!?!?!?!" |
| 02:13.32 | robbyt | hah |
| 02:13.37 | Err | heh, I get irritated by all the friggin' clocks in my office - I'm glad my phone doesn't have one |
| 02:13.43 | robbyt | hey so what distro are you running? |
| 02:13.54 | Mag1KaL | someunixguy, yes it is... it actually says it has launched the script, then says it can't find it. |
| 02:13.54 | someunixguy | fedora 3 on most of our boxes |
| 02:14.16 | robbyt | aha |
| 02:14.32 | robbyt | i should give fedora a shot again, i got so pissed at fc2 |
| 02:14.45 | robbyt | jumped ship and switched to gentoo |
| 02:14.46 | someunixguy | Mag1KaL: Try running the script outside asterisk (i.e. 'perl filename.agi') and see if it runs w/o error. Might be a bad shebang |
| 02:14.55 | someunixguy | fc2 was a problem |
| 02:15.03 | robbyt | been running gentoo everywhere really, it's quite sadistic |
| 02:15.10 | someunixguy | I like fc4's implementation of yum |
| 02:15.20 | robbyt | or, should i say masochistic |
| 02:15.25 | someunixguy | very nice group install stuff |
| 02:15.29 | someunixguy | lol |
| 02:15.30 | Err | yum just keeps getting better - if only the FC releases would last longer |
| 02:15.40 | someunixguy | yeah, it's been moving too fast |
| 02:15.41 | Err | well, and be more stable |
| 02:15.42 | robbyt | yeah that's what i worry about |
| 02:15.52 | robbyt | how about asterisk in debian? |
| 02:15.53 | someunixguy | that's why we're still on 3 for production.... |
| 02:15.57 | Err | I gave up on fedora because I had to upgrade too often, and sometimes upgrades actually busted things |
| 02:15.57 | robbyt | what's the word on that? |
| 02:16.05 | someunixguy | my AMP customer is on debian |
| 02:16.09 | someunixguy | seems ok |
| 02:16.13 | Err | robbyt: ubuntu has asterisk 1.2.1 in Universe |
| 02:16.18 | someunixguy | I miss the 'service' command tho |
| 02:16.19 | robbyt | ooo |
| 02:16.24 | Err | I haven't configured it yet, but it seems to be complete |
| 02:16.30 | robbyt | does debian/ubuntu have rc-update? |
| 02:16.43 | robbyt | i think gentoo swiped a lot of their init scripts from debian |
| 02:16.54 | fdask | what |
| 02:16.57 | Err | I don't know what an rc-update is |
| 02:16.58 | Math` | I've no rc-update in debin |
| 02:17.00 | Math` | debian* |
| 02:17.06 | robbyt | ok, |
| 02:17.07 | fdask | dont they both use system v style init scripts |
| 02:17.11 | Err | debian uses /etc/init.d scripts |
| 02:17.16 | someunixguy | I installed Asterisk on Solaris this weekend for the first time, that was a nightmare |
| 02:17.25 | robbyt | rc-update is super slick in gentoo, there are /etc/init.d/ scripts |
| 02:17.27 | Math` | * on slowlaris? uh |
| 02:17.37 | someunixguy | yeah, not my idea |
| 02:17.38 | Err | everything on solaris is a nightmare to install |
| 02:17.41 | robbyt | lets say "rc-update add asterisk default" will add asterisk to runlevel default |
| 02:17.50 | robbyt | rc-update show |
| 02:17.53 | Err | if it weren't so stable when you were done, solaris would be easy to write off :-) |
| 02:17.54 | robbyt | shows everything |
| 02:17.56 | robbyt | good stuff :D |
| 02:18.01 | Err | oh, so it's like chkconfig |
| 02:18.02 | someunixguy | if you want to pay me $$$/hr to install Asterisk I'll put it on a toaster |
| 02:18.24 | someunixguy | Solaris was a bitch tho |
| 02:18.31 | robbyt | haha |
| 02:18.34 | *** join/#asterisk svenl_ (n=sven@AStrasbourg-251-1-14-121.w82-126.abo.wanadoo.fr) |
| 02:18.39 | robbyt | is there even zaptel in solaris? |
| 02:18.43 | someunixguy | Asterisk ended up in /usr/opt/asterisk/etc |
| 02:18.43 | Math` | someunixguy: then you have to put an extension to turn off the toaster, or a ringback extension to ring when your toasts are ready |
| 02:18.49 | Err | I generally write software on solaris, if I want to port it to other systems, because if it'll build there it'll build anywhere |
| 02:18.54 | someunixguy | no zaptel |
| 02:19.12 | someunixguy | and the 'install' program doesn't work |
| 02:19.18 | someunixguy | so you need to hack your makefile |
| 02:19.23 | robbyt | ugh |
| 02:19.29 | someunixguy | check that, makefile(s) |
| 02:19.43 | robbyt | so why solaris? |
| 02:19.47 | robbyt | nothing on tv? |
| 02:19.48 | robbyt | :D |
| 02:19.58 | someunixguy | Client wanted solaris, who am I to persuade? |
| 02:20.21 | Math` | I would have told the client it'd be more performant on linux |
| 02:20.25 | someunixguy | seriously though, they've got some wacky solaris app they're doing agi integration with |
| 02:20.27 | Math` | and way less hassle |
| 02:20.43 | someunixguy | the vp of engineering has a hard-on for solaris |
| 02:20.52 | someunixguy | he loves us now |
| 02:21.06 | robbyt | he must be >50 years old |
| 02:21.15 | someunixguy | yeah, and like 400 lbs |
| 02:21.19 | robbyt | sweet |
| 02:21.30 | someunixguy | it's an odd setup |
| 02:21.45 | someunixguy | he's one of those "i like it because i know it" guys |
| 02:22.07 | someunixguy | but, having just finished bashing AMP, who the heck am I to talk |
| 02:22.13 | robbyt | i tried solaris 10, booted it up, and thought "ok now what?" |
| 02:22.37 | someunixguy | it's sun hardware and it couldn't detect the nic |
| 02:22.38 | robbyt | then i reformatted a week later when i couldn't figure out how to get php5 working... |
| 02:22.43 | someunixguy | i had to load a supplement cd |
| 02:23.06 | someunixguy | none of the path vars are set, so I had to hack around that to run anything |
| 02:23.10 | someunixguy | it was PAINFUL |
| 02:23.16 | *** join/#asterisk tris_ (i=tristan@camel.ethereal.net) |
| 02:23.25 | robbyt | yeah i've done work with irix at my old job |
| 02:23.32 | someunixguy | twelve years of linux spoils a guy |
| 02:23.35 | robbyt | it was the same sort of weirdness though- |
| 02:23.41 | someunixguy | same story w/aix |
| 02:23.57 | someunixguy | "Oh, you want a COMPILER?" |
| 02:24.02 | Err | someunixguy: if you ever work on solaris again, filesystem(5) is your friend |
| 02:24.03 | robbyt | hahah |
| 02:24.14 | robbyt | what's that err? |
| 02:24.19 | someunixguy | I'm working on it tomorrow actuallt |
| 02:24.23 | robbyt | man 5 filesystem ? |
| 02:24.25 | Err | it's the manpage that tells where everything is |
| 02:24.41 | someunixguy | they realized today the default solaris partitioning scheme (99% /home, 1% everything else) is no good |
| 02:24.44 | Err | man -s5, but yes (sysv man, not bsd man - until you put /usr/ucb first in your path, of course ;-) |
| 02:24.57 | someunixguy | really? |
| 02:25.00 | robbyt | hahah! |
| 02:25.05 | someunixguy | i'll have to check that out actually |
| 02:25.20 | someunixguy | I'm going to whack /home and run a 'growfs' on the / partition |
| 02:25.27 | Err | solaris has the best documentation of any software product I've ever seen - but you *have* to use it to do anything at all |
| 02:25.30 | someunixguy | hopefully that'll fix it up |
| 02:25.36 | Err | symbolic link the crap out of the disk |
| 02:25.41 | someunixguy | haha |
| 02:25.48 | someunixguy | that'll be fun to maintain |
| 02:25.53 | Err | that's what I do to machines that were installed via solaris's I'm-uber-retarded partitioning tool |
| 02:26.25 | someunixguy | i used to do that but after six months who can remember anything |
| 02:26.30 | Err | well, local disks aren't *supposed* to hold anything - that's for network disks - which is why they partition the way they do |
| 02:26.52 | someunixguy | yeah, true enough |
| 02:27.00 | robbyt | weird! |
| 02:27.00 | Err | if you do things The Sun Way everything works just fine - but you have to learn it first, which is hard; there's a reason Sun-certified sysadmins make the money they do |
| 02:27.06 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 02:27.13 | robbyt | so all the directories should exist off a network share? |
| 02:27.13 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 02:27.29 | someunixguy | nfs it all, lol |
| 02:27.44 | Err | sun's plan is that most machines have little local storage, and the local storage that is there will be used as cache or for speed-sensitive data |
| 02:27.51 | robbyt | so then, which computer hosts the nfs shares?? lol |
| 02:28.02 | someunixguy | well, your $100k san naturally |
| 02:28.04 | Err | robbyt: a disk storage array - it's not really a computer :-) |
| 02:28.04 | robbyt | it's like an nfs circle like amp is an include circle |
| 02:28.15 | someunixguy | haha |
| 02:28.30 | robbyt | ahh, see that's getting into insane realms that i've only read about... |
| 02:28.33 | someunixguy | seriously, it's not a bad architecture |
| 02:28.57 | shmaltz | in vi how do I go to the end of the file? |
| 02:29.03 | someunixguy | and despite the painful environment it feels pretty solid |
| 02:29.03 | srt | G |
| 02:29.55 | robbyt | ok cya guys- |
| 02:30.15 | someunixguy | So, Err, since you seem to know your Solaris... When I'm doing this partitioning nonsense tomorrow, should I be able to growfs without killing the box? |
| 02:30.22 | Err | heh, I have no idea |
| 02:30.24 | someunixguy | cya robby |
| 02:30.25 | Err | I've never used it |
| 02:30.28 | someunixguy | bah |
| 02:30.39 | Err | sorry :-) |
| 02:30.42 | someunixguy | nothing like getting paid $$$/hr to fumble |
| 02:30.58 | Err | you might ask in #opensolaris - those kids know their stuff |
| 02:31.02 | robbyt | setup a vmware machine tonight ;) |
| 02:31.09 | *** join/#asterisk pengyong (n=lala@218.93.159.101) |
| 02:31.17 | someunixguy | Guess what I have open in my other IRC tab |
| 02:31.19 | someunixguy | lol |
| 02:31.38 | someunixguy | It's a pretty quiet channel though |
| 02:31.49 | Err | heh, I should re-join over there - I learned most of what I know about solaris from there, even though I've worked on solaris boxes for years |
| 02:32.18 | Err | if you ask, somebody will know - there are several solaris developers in there, and several more sun-employed troubleshooters |
| 02:32.38 | someunixguy | I haven't used IRC in... umm... 9 years or something like that but #opensolaris saved my ass on saturday |
| 02:33.48 | someunixguy | speaking of vmware, they want me to set up a vmware server tomorrow |
| 02:34.03 | someunixguy | they've got half their infrastructure running on vmware, it's sick |
| 02:37.22 | *** join/#asterisk lahaine (n=lahaine@210.64.119-80.rev.gaoland.net) |
| 02:37.26 | Newbie___ | anyone do consultancy service here? |
| 02:37.40 | someunixguy | sure, what are you looking for help with? |
| 02:38.21 | Newbie___ | h323, did it before on EL3. but now it wont work on EL4 |
| 02:38.45 | someunixguy | never worked with 323 on asterisk, sorry |
| 02:38.53 | Newbie___ | ok |
| 02:39.04 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:39.15 | Newbie___ | was pretty easy on EL3 |
| 02:39.35 | someunixguy | what are you communicating with? |
| 02:39.45 | Newbie___ | voicemaster |
| 02:39.50 | someunixguy | ah |
| 02:40.14 | littleball | hello, i have a context , at the beginning of this context, i switched to realtime, basically the real time is only used to set channel variable. So, i would like to switch back to normal dialplan. |
| 02:40.16 | littleball | how to? |
| 02:41.00 | someunixguy | I'm calling it a night. It was nice to commisserate. |
| 02:41.07 | someunixguy | cya guys |
| 02:41.11 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:42.52 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 02:43.20 | *** join/#asterisk iq (n=iq@71-214-1-78.omah.qwest.net) |
| 02:52.31 | shmaltz | is there anything wrong with this: |
| 02:52.33 | shmaltz | exten => test,5,Noop(${CUT(${ARG3},-,1)} is f1) |
| 02:52.35 | shmaltz | ${ARG3} is set to: 2-SIP/8541663-ZAP/G1 |
| 02:53.02 | *** join/#asterisk annonimous (n=annonimo@dsl-201-133-94-199.prod-infinitum.com.mx) |
| 02:53.05 | annonimous | hiya |
| 02:53.13 | shmaltz | hi |
| 02:53.22 | annonimous | hello shmaltz |
| 02:53.25 | annonimous | how are you? |
| 02:53.31 | shmaltz | fine, and you? |
| 02:53.33 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 02:53.41 | annonimous | fine fine thanks |
| 02:53.43 | annonimous | whats up? |
| 02:54.12 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 02:56.42 | *** join/#asterisk mgoh (n=goh@60.49.6.190) |
| 02:57.14 | mgoh | why we need to use channel bank? |
| 02:57.45 | *** join/#asterisk fugitivo (n=ajf@201.255.176.51) |
| 02:58.54 | *** join/#asterisk |omni| (n=rob@net98.limelyte.net) |
| 02:59.07 | |omni| | anyone using chan_sccp with cisco 79xx phones? |
| 02:59.15 | |omni| | trying to figure out a simple conference call |
| 02:59.34 | |omni| | I can transfer between two outbound, but can't seem to join the two |
| 03:00.13 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 03:04.04 | mgoh | why buddy can tell me what is the purpose of channel bank? |
| 03:04.43 | eieiyo | anybody familiar with app_rpt? |
| 03:05.56 | *** join/#asterisk jef_ (i=fischer@p54846B6B.dip.t-dialin.net) |
| 03:06.14 | tzanger | mgoh: a channel bank aggregates (usually 24) analogue phone circuits into a single digital one (T1) |
| 03:06.36 | pauldy | ? |
| 03:07.02 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 03:07.04 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:08.56 | mgoh | tzanger:I'm using 2 PRI directly to digium card from PSTN, still need channel bank? |
| 03:09.14 | tzanger | mgoh: are you planning on connecting regular everyday phones or phone lines in to asterisk? |
| 03:09.35 | *** join/#asterisk usam (n=alx@203.156.48.73) |
| 03:09.36 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 03:10.49 | mgoh | I plan to use 32 softphone and 7 ip phone and 1 ATA for FAx |
| 03:10.49 | *** join/#asterisk exism (n=jon@66.77.78.228) |
| 03:11.23 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 03:11.40 | mog_home | morning people |
| 03:11.53 | drumkilla | ? |
| 03:11.53 | tuxinator_linux | Evening mog_home |
| 03:12.02 | exism | i'm new to asterisk (i'm working on a system that's already configured), but i have managed to setup SIP with a provider and can dial out to landline and can receive calls. however, i don't hear anything on either side of the phone. where should i be looking to troubleshoot this? |
| 03:12.02 | mog_home | hows it going tuxinator_linux |
| 03:12.06 | mgoh | tzanger: I plan to use 32 softphone and 7 ip phone and 1 ATA for FAx |
| 03:12.17 | tzanger | mgoh: well then no, you don't need a channel bank |
| 03:12.22 | tzanger | morning mog_home |
| 03:12.30 | tuxinator_linux | mog_home: Doing fine |
| 03:12.41 | mog_home | grand |
| 03:12.46 | mog_home | hey tzanger hows it hanging |
| 03:13.08 | file[laptop] | !!!!!! |
| 03:16.06 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 03:17.27 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 03:18.12 | *** join/#asterisk brookshire[home] (n=matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 03:18.30 | mog_home | good stuff |
| 03:20.02 | tzanger | mog_home: not bad watching my country go down the shitter |
| 03:21.11 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:22.14 | brookshire[home] | tzanger: why is that? |
| 03:22.18 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:23.04 | *** join/#asterisk FastJack_ (i=fastjack@p5091FDC7.dip.t-dialin.net) |
| 03:23.09 | mog_home | what you say tzanger ? |
| 03:23.20 | mog_home | you could move to qwellstania with me |
| 03:23.24 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 03:23.32 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:23.58 | tzanger | brookshire[home]: cbc.ca will tell you |
| 03:24.01 | tzanger | mog_home: where? |
| 03:24.06 | tzanger | qwellstania? |
| 03:24.15 | mog_home | yes |
| 03:24.19 | mog_home | the land of qwell |
| 03:24.26 | tzanger | ahh |
| 03:24.39 | tzanger | I'd think he'd call it Qwellington |
| 03:24.41 | mog_home | whats wrong with the old usa |
| 03:25.13 | mgoh | tzanger:If we got 32 line centrex we can use channel bank to convert to T1 interface so tht I just need buy 1 port T1 digium card. I'm right? |
| 03:25.50 | tzanger | mgoh: T1s have 24 channels, not 32. E1s have 32 channels but I don't know of any E1 channel banks |
| 03:26.56 | Newbie___ | hi all, anyone has experience installing h323 in Asterisk Version 1.2.2 |
| 03:28.56 | file[laptop] | meep |
| 03:29.52 | *** join/#asterisk Cresl1n (n=matt@m495e36d0.tmodns.net) |
| 03:30.14 | Cresl1n | twisted[mobile]: !!!! |
| 03:30.16 | brookshire[home] | HAPPY BIRTHDAY! |
| 03:30.17 | Cresl1n | you rock man! |
| 03:30.22 | Cresl1n | thanks!!! |
| 03:30.23 | Cresl1n | :-) |
| 03:30.53 | mog_home | happy birthday cresl1n |
| 03:31.04 | Cresl1n | thanks mog_home |
| 03:31.11 | Cresl1n | you guys are the best of friends |
| 03:32.13 | mog_home | yup |
| 03:32.17 | mog_home | we try |
| 03:33.41 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:40.38 | Trazzz | how much bw does each sip user take? |
| 03:40.49 | Cresl1n | Trazz: ???? |
| 03:41.01 | Cresl1n | Trazz: there are so many answers to your question |
| 03:41.03 | Trazzz | bandwidth |
| 03:41.16 | Trazzz | normal call |
| 03:41.26 | Cresl1n | Trazz: that depends on a lot of factors |
| 03:41.40 | Cresl1n | but primarily depends on what codec you are using |
| 03:42.04 | Trazzz | gsm |
| 03:42.16 | Trazzz | or g711 |
| 03:42.19 | Trazzz | or g711a |
| 03:42.19 | Cresl1n | ~gsm |
| 03:42.22 | jbot | i guess gsm is a codec, operating at approx 13kbps up/down. |
| 03:42.39 | Cresl1n | I think g711 ends up being around 80-85 w/ rtp headers |
| 03:42.40 | Trazzz | ok and the others? |
| 03:42.43 | Trazzz | wow |
| 03:42.46 | Trazzz | what a hog |
| 03:43.00 | Cresl1n | Trazz: you can easily google all that |
| 03:43.06 | Trazzz | ok |
| 03:43.11 | Trazzz | 8k or 64k i thought |
| 03:44.07 | tzanger | Trazzz: yes, but that is 64kbps of AUDIO ONLY |
| 03:44.16 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-253.msy.bellsouth.net) |
| 03:44.22 | tzanger | you need to realize that that audio payload is split up into packets, and each packet has a header |
| 03:44.24 | Cresl1n | tzanger!!!! |
| 03:44.41 | tzanger | and after that, Trazzz, you need to realize that you don't take 1 second of audio and put it in 1 packet, you take 20ms of audio and put it in one packet |
| 03:45.05 | tzanger | so after it's all said and done, you end up with about 80kbps of bandwidth for a ulaw (g711) conversation |
| 03:45.13 | tzanger | (80kbps in and 80kbps out, of course) |
| 03:45.16 | tzanger | Cresl1n: hey hey hey |
| 03:45.23 | Trazzz | nice |
| 03:45.30 | Trazzz | that will kill a line |
| 03:46.03 | tzanger | yep |
| 03:46.12 | Trazzz | i take it cisco dont support gsm.. |
| 03:46.26 | [av]bani | they should, everyone does |
| 03:46.35 | tzanger | Trazzz: don't think so. g729/ulaw is generally what the commercial devices support |
| 03:46.36 | justinu | lots of hardphones don't |
| 03:46.48 | [av]bani | thats weird, since gsm is prett yuniversal |
| 03:46.50 | Trazzz | ya figured as much.. nice |
| 03:46.54 | Err | gsm isn't very efficient, though |
| 03:47.02 | tzanger | [av]bani: no, not everyone does. I can't think of any "big iron" telecom kit that includes gsm |
| 03:47.02 | justinu | sounds kinda ugly |
| 03:47.07 | tzanger | Err: huh? |
| 03:47.12 | tzanger | Err: define efficient |
| 03:47.12 | [av]bani | Err: it's plenty efficient. 4:1 compression or thereabouts |
| 03:47.16 | Cresl1n | Err: gsm is nice ;-) |
| 03:47.18 | tzanger | gsm is pretty damn good |
| 03:47.35 | Trazzz | lets all call cisco and demand gsm support |
| 03:47.39 | Trazzz | using voip of course |
| 03:47.48 | Trazzz | DoS cisco sip gateway now |
| 03:48.05 | *** part/#asterisk Cresl1n (n=matt@m495e36d0.tmodns.net) |
| 03:48.07 | Err | efficient compared to the non-free compressions they *do* support |
| 03:48.22 | [av]bani | gsm is also very low cpu |
| 03:48.31 | fdask | anyone here use a voicetronix card? |
| 03:49.02 | [av]bani | ulaw->gsm is 3ms on my pc, ulaw->g729 is 18ms |
| 03:49.11 | [av]bani | ulaw->speex is 40 :() |
| 03:49.16 | tzanger | that's nothing |
| 03:49.23 | justinu | yeah, speex is the heaviest |
| 03:49.26 | tzanger | I was experimenting with (really) light hardware |
| 03:49.31 | [av]bani | 40ms is 2 g711 frames |
| 03:49.39 | tzanger | P90 (no MMX), ulaw->ilbc was 967ms |
| 03:49.45 | Err | ouch |
| 03:49.46 | [av]bani | lol! |
| 03:50.00 | tzanger | the funny part was that I *was* able to have a conversation with it but it was... tedious |
| 03:50.15 | justinu | 10-4 good buddy |
| 03:50.33 | exism | anyone know why when calling landline through SIP i can hear fine on the landline phone but the voip phone hears nothing? (i'm not sure what is the relevant information i should provide) |
| 03:50.45 | tzanger | exism: sounds like a NAT issue |
| 03:52.09 | mgoh | tzanger:thanks |
| 03:52.49 | tzanger | mgoh: no problem at all |
| 03:54.32 | [av]bani | http://news.yahoo.com/s/ap/20060123/ap_on_hi_te/botnet_hacker |
| 03:54.51 | exism | hmm, there is no NAT going on here |
| 03:55.36 | *** join/#asterisk dissolutions (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 03:57.03 | [av]bani | its either nat or firewall |
| 03:57.14 | [av]bani | 1-way is almost always nat |
| 03:57.32 | [av]bani | theres likely nat, you just dont know it :) |
| 03:57.49 | exism | the server is on a direct connection |
| 03:58.15 | exism | i'm setting nat=never |
| 03:58.20 | exism | in case it's trying to use nat somewhere |
| 03:58.40 | [av]bani | 1-way means your return packets cant get to the voip device, either nat or firewall is blocking it |
| 03:58.46 | *** join/#asterisk S-flyp (n=cashmone@203.82.38.26) |
| 03:58.55 | exism | hmm |
| 03:59.35 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 03:59.50 | eieiyo | ~app_rpt |
| 04:00.02 | [av]bani | you could always try realigning the plasma injectors |
| 04:00.48 | tzanger | [av]bani: you sound like sivana |
| 04:01.06 | exism | on the console it says: -- Attempting native bridge of SIP/206-6758 and SIP/smf-peer.sip.o1.com-533a |
| 04:01.09 | *** join/#asterisk bmg505 (n=leon@dsl-146-5-174.telkomadsl.co.za) |
| 04:01.14 | exism | should it say it has made the native bridge? |
| 04:01.25 | tzanger | exism: depends |
| 04:02.18 | exism | so i shouldn't be alarmed if that's the last message i receive |
| 04:02.25 | tzanger | no |
| 04:02.27 | tzanger | not at all |
| 04:02.38 | [av]bani | exism: might also want to canreinvite=no |
| 04:02.58 | [av]bani | on both sip peers |
| 04:03.39 | exism | the other peer isn't defined on my side |
| 04:04.05 | exism | but i will try that |
| 04:05.13 | exism | i'm 99% sure there is no firewall action going on |
| 04:05.21 | exism | cause this a production server connected to a 100mbit line |
| 04:09.49 | tzanger | exism: that means jack shit |
| 04:10.09 | tzanger | the far side could be a little phone behind 6 NAT boxes on a GPRS connection over bluetooth |
| 04:10.21 | NewSole | hmmm.... looks like liberals have been liberated....... |
| 04:10.33 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 04:10.33 | exism | it's calling to a major voip provider though |
| 04:10.38 | exism | and then to a standard pots phone |
| 04:10.46 | NewSole | lol |
| 04:10.46 | exism | so i'd imagine it would have to be my side |
| 04:10.54 | tzanger | NewSole: yeah... as my friend said we better start brushing up on our American National Anthem |
| 04:11.10 | NewSole | yup.... |
| 04:11.19 | tzanger | I am impressed though |
| 04:11.31 | tzanger | the greens got 7% of teh votes in my riding (20 stations left to count) |
| 04:11.45 | tzanger | which for an area that has been strongly conservative since the 50s... I think is pretty good |
| 04:12.08 | NewSole | yes... but liberals needed a good but kicking to put them back in place... they were too conceeded |
| 04:12.08 | tzanger | the old fogies in this town are starting to die or be put into retirement homes without televisions by the young'uns |
| 04:12.18 | tzanger | NewSole: agreed, but the PC arent' any better |
| 04:12.43 | NewSole | no they are not... personaly I would have voted for other..... |
| 04:12.48 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 04:13.16 | tzanger | I don't believe in voting to punish someone else. I vote for who I feel best represents me which, at this point, was nobody |
| 04:13.30 | NewSole | yup... me too |
| 04:13.46 | tzanger | but the greens were "safe" in that they coudl not end up weilding any power, but voting green would get more money into their warchest for next time, and let's hope that next time they are a little better prepared |
| 04:14.38 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 04:14.48 | tzanger | I feel we need fresh blood and new ideas in ottawa... PC and libs are more or less identical, NDP would bury us ... let's get some new voices in there and see what they can do |
| 04:15.00 | *** join/#asterisk dijit0 (n=eric@69.106.51.242) |
| 04:15.01 | warthawg | anyone using a Zultys WIP2 wireless phone? |
| 04:15.10 | NewSole | I am just glad that liberals are no longer in power and the "con's" are on a short leash |
| 04:15.22 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:15.25 | tzanger | yeah I suppose it's about as good as we could hope for |
| 04:15.33 | tzanger | although I really do believe the PC should be footing the ENTIRE bill for the election |
| 04:16.21 | NewSole | personaly... the winner should pay.... not the tax payers |
| 04:16.26 | tzanger | you don't take down a government unless you are positive you have the majority of the nation behind you. tying the price of the election to the saber-rattling part(ies) ensures that it will fucking hurt if they overestimate their support |
| 04:16.33 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 04:17.04 | tzanger | if you think you have majority support then prove it by winning a majority election. If you don't win, you get pecker-slapped |
| 04:17.16 | NewSole | lol |
| 04:17.25 | NewSole | out comes the lub dud |
| 04:17.38 | tzanger | learn to work as a team instead of bitching and moaning |
| 04:18.04 | NewSole | yup... too much bitch slapping this election |
| 04:18.21 | tzanger | I'm not quite sure how to tie in that the minority leader needs to also function as part of the team but the whole "we dun like you, we're gonna take you down and cost everyone a fortune" is nonsense |
| 04:18.39 | tzanger | I really, *really* disliked the PC for that |
| 04:18.43 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:18.46 | tzanger | and what did we end up with ? the exact same fucking thing |
| 04:19.10 | tzanger | so what now, the libs will take the government down and we'll do this fucking circle jerk again? |
| 04:19.35 | NewSole | o ya.... we will be doing this again in 2 years |
| 04:19.55 | *** part/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 04:20.00 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
| 04:20.24 | BoRiS | Anyone know how to fix the Cisco 9740 (after upgrading to 7.4 and 7.5) firmware that during the upgrade, it shows "Protocol Application Invalid". |
| 04:22.12 | dissolutions | NewSole; whose in the lead? |
| 04:22.35 | BoRiS | Conservative....Yikes |
| 04:22.38 | tzanger | dissolutions: PC. 121 to lib 104 ndp 31 and one green |
| 04:22.43 | dissolutions | =O |
| 04:22.55 | tzanger | that's exactly what I was hoping for (the green) -- I want to see thm at the next federal debates |
| 04:23.03 | Trazzz | :-) |
| 04:23.18 | tzanger | if it HAS to be 4 parties, get the bloc out of there |
| 04:23.34 | tzanger | until they want to be a REAL federal party then stay the fuck out of the federal side of it |
| 04:24.25 | tzanger | time to sleep |
| 04:24.27 | tzanger | 'night |
| 04:24.56 | dissolutions | lol @ the diversity between urban vs rural |
| 04:29.04 | syle | whats all this political bullshit lol |
| 04:29.13 | syle | go smoke another joint :) |
| 04:31.39 | BoRiS | you ARE the smoke. :-p |
| 04:31.41 | rajiv | anyone using sellvoip or sipphone for origination? |
| 04:32.36 | syle | if you have lots of cash just use the best, level3 etc |
| 04:32.50 | rajiv | i need just 1 number |
| 04:32.51 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au) |
| 04:34.03 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 04:35.30 | rajiv | is level3 going to talk to me for just 1 line ? |
| 04:35.33 | syle | theres so many places just pick 1 |
| 04:36.03 | syle | no they don;t give a shit about you unless you do about 1 million minutes a month |
| 04:37.33 | syle | pisses me off but whatever, thats how the rich get richer right |
| 04:38.29 | *** join/#asterisk ast_freak (n=ast_frea@68-112-134-195.dhcp.stcl.mn.charter.com) |
| 04:38.51 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 04:39.54 | syle | hheehe i don;t really care, i just write software for voip providers like you to make money, working on a good billing system right now, 4 months into it, let me know if you need that |
| 04:41.10 | syle | all c based, no agi or database trigger overhead, all embedded into asterisk , complete speed |
| 04:45.53 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:46.47 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 04:48.05 | BoRiS | Aren't we all? |
| 04:51.01 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 04:53.51 | mgoh | how can we implement SIP refer method in ur extension? |
| 04:54.21 | file[laptop] | BoRiS BoRiS BoRiS |
| 04:56.49 | BoRiS | file!!!!!!!!!! |
| 04:56.57 | BoRiS | Wassssssssup? |
| 04:58.33 | file[laptop] | not much, what about you? |
| 04:58.54 | syle | hey file |
| 04:59.05 | file[laptop] | hi |
| 05:00.32 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:03.37 | *** join/#asterisk sudhir492 (n=sudhir@pool-71-114-70-81.washdc.dsl-w.verizon.net) |
| 05:03.40 | sudhir492 | hi all |
| 05:03.59 | tainted- | do i still have to compile mpg123 for asterisk 1.2.2 or is it integrated now? |
| 05:04.51 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:05.51 | sudhir492 | my voip provider has given me the information to program a softphone. Is it possible to feed that to asterisk and make calls using those credentials |
| 05:06.45 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:08.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-84-110-23-64.red.bezeqint.net) |
| 05:08.22 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:09.09 | *** join/#asterisk kimosabe (n=kimosabe@201.133.216.51) |
| 05:09.26 | kimosabe | can some one help me get my x lite going i need 2 enable g726 on it |
| 05:09.40 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:10.42 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:11.30 | *** join/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 05:11.54 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:12.04 | *** part/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 05:14.58 | dudes | kimosabe - sip.conf disallow=all allow=g726 |
| 05:15.03 | dissolutions | :*( Martin just stepped down :*( |
| 05:16.16 | rob0 | Martin at Digium? Stepped down from what? From Digium? |
| 05:16.33 | Himeko | not liek he had a choice |
| 05:16.44 | BoRiS | Wish liberals one. |
| 05:16.50 | BoRiS | err won |
| 05:17.00 | mog_home | martin? |
| 05:17.02 | Himeko | i wish they all died |
| 05:17.28 | dudes | that's harse |
| 05:17.33 | rob0 | oh is this some political thing? |
| 05:17.40 | mog_home | what? |
| 05:17.57 | rob0 | Canadian election? |
| 05:18.06 | dissolutions | yar |
| 05:18.10 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:18.20 | wunderkin | canada still exists? |
| 05:18.29 | BoRiS | does the US still exist? |
| 05:19.01 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 05:19.06 | rob0 | haha you mention Martin in here and I think Martin at Digium :) |
| 05:19.22 | mog_home | hasnt been a martin at digium in a while |
| 05:20.03 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:20.06 | rob0 | oh hmmmm ... it was 2 or so years ago, he helped me with my initial setup. |
| 05:22.03 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:32.21 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:33.02 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 05:35.58 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 05:36.14 | mog_home | oh woot |
| 05:36.16 | mog_home | it works |
| 05:36.45 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 05:38.25 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 05:38.47 | mgoh | can we use channel bank to convert T1 to analog phone? |
| 05:39.04 | mgoh | or what call FXS port |
| 05:39.22 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:39.26 | Corydon76-home | Not only can you, but there really isn't any other purpose for a channel bank |
| 05:39.40 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 05:39.46 | Corydon76-home | That is exactly what a channel bank does |
| 05:40.04 | justinu | mgoh: where in the world are you from? |
| 05:40.33 | *** join/#asterisk pengyong (n=lala@222.188.133.19) |
| 05:40.40 | Corydon76-home | justinu: from the land of knights and knaves |
| 05:40.57 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:40.58 | justinu | camelot? |
| 05:41.04 | copantl | can i reinstall asterisk-addons without affect my configuration of asterisk? |
| 05:41.14 | Corydon76-home | Uh, no. A Martin Gardner logical construct. |
| 05:41.30 | mgoh | singapuro |
| 05:41.33 | Corydon76-home | copantl: yes |
| 05:42.01 | Corydon76-home | justinu: knights always tell the truth, knaves always lie |
| 05:42.04 | copantl | which one of all file into etc/asterisk is gointo be affect? |
| 05:43.00 | benjk | Cory, there is at least one other use for some channel banks ... |
| 05:43.03 | Corydon76-home | You come to a fork in the road, one leads to the knaves village, the other to the knights village. You are allowed to ask exactly one question of the person standing at the fork, but you do not know whether the person is a knight or a knave. What question do you ask? |
| 05:43.15 | benjk | if they are heavy enough, you can use them as a doorstop ;) |
| 05:43.32 | SwK | anyone have a NANPA OCN -> NPA-NXX cross ref table? (complete) |
| 05:43.57 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 05:43.58 | Corydon76-home | SwK: it's proprietary data |
| 05:44.12 | SwK | not really |
| 05:44.13 | Corydon76-home | SwK: and it changes monthly |
| 05:44.24 | Corydon76-home | Telcordia charges a fee for that data |
| 05:44.25 | copantl | if i reinstall addons can be the database in /var/log/cdr-vs affected? |
| 05:44.36 | SwK | telcordia charges a fee for their tables |
| 05:44.59 | SwK | just like bellsouth charges a fee for their CNAM -> DID tables... |
| 05:45.38 | Corydon76-home | copantl: ever heard of backups? |
| 05:46.15 | copantl | of course,but that not answer my question? |
| 05:46.29 | justinu | telcordia sucks |
| 05:46.38 | copantl | it can be affected? |
| 05:46.42 | justinu | how can that not be freely avaialble? |
| 05:47.20 | Corydon76-home | justinu: that's their entire business model |
| 05:47.28 | justinu | that's weird |
| 05:47.36 | justinu | seems pretty shaky |
| 05:48.01 | SwK | justinu: the data is available from other sources, telcordia just seems to be the only one with the latest most complete dataset |
| 05:48.07 | SwK | and they charge out the ass for it |
| 05:48.08 | justinu | right |
| 05:48.15 | justinu | they're authoritative somehow |
| 05:48.16 | Corydon76-home | justinu: ever since the breakup of ATT in 1984... |
| 05:48.39 | SwK | its not really propritary data... its just tightly controlled |
| 05:48.45 | justinu | they used to be called bellcore |
| 05:48.52 | justinu | the keepers of the standards |
| 05:49.10 | sudhir492 | my voip provider has given me the information to program a softphone. Is it possible to feed that to asterisk and make calls using those credentials |
| 05:49.17 | justinu | sudhir: yes |
| 05:50.03 | sudhir492 | justinu: how should I configure? |
| 05:50.13 | justinu | like if you want the NI-2 spec, you have to pay telcordia |
| 05:50.19 | justinu | ~docs |
| 05:50.21 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:51.25 | SwK | the spec itself isnt really proprietary, its the documentation dscribing it |
| 05:51.49 | justinu | i don't get that |
| 05:51.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:51.54 | justinu | seems like a fine point |
| 05:52.19 | justinu | it should be in a public library or something |
| 05:52.27 | SwK | its like getting the v.22 specs |
| 05:52.50 | justinu | i'm impressed the FAA finally made IFR approach plates available for download online, free. |
| 05:52.58 | sudhir492 | justinu: I tried |
| 05:53.03 | sudhir492 | [braintel] |
| 05:53.04 | sudhir492 | type=peer |
| 05:53.04 | sudhir492 | auth=23456:1234@brain.net.pk |
| 05:53.04 | sudhir492 | username=2105264 |
| 05:53.04 | sudhir492 | secret=1234 |
| 05:53.04 | sudhir492 | fromdomain=brain.net.pk |
| 05:53.06 | sudhir492 | host=203.128.7.14 |
| 05:53.08 | sudhir492 | port=8891 |
| 05:53.10 | sudhir492 | allow=all |
| 05:53.12 | sudhir492 | dtmf=rfc2833 |
| 05:53.22 | justinu | sudhir492: i'm not going to help unless you pay |
| 05:53.25 | justinu | sorry |
| 05:53.27 | SwK | v.22 itself is a specification for communications, the spec, can be implemented by pretty much anyone... the problem is figuring it out, getting the docs for it tho cost a few bucks cause the ITU holds the (C) on the documentation |
| 05:53.37 | sudhir492 | hmm |
| 05:53.39 | sudhir492 | how much |
| 05:53.52 | SwK | 100USD seems fail |
| 05:54.00 | justinu | works for me |
| 05:54.01 | SwK | err fair |
| 05:54.12 | sudhir492 | not to me :-( |
| 05:54.33 | justinu | you can figure it out, it's not that tough |
| 05:54.39 | mgoh | have any method or ready make software done a call flow where it make a call when the call is connected it directly will transfer to another extension. |
| 06:00.14 | *** join/#asterisk rene- (i=rene@201.102.18.74) |
| 06:03.12 | tronix | justinu: where? (IFR apch plates) |
| 06:04.10 | justinu | http://www.naco.faa.gov/digital_tpp.asp?ver=0601&eff=01-19-2006&end=02-16-2006 |
| 06:04.34 | tronix | whoo hoo!!!!!! |
| 06:04.37 | tronix | very nice. thanks!! |
| 06:04.47 | justinu | np :) |
| 06:06.04 | justinu | unfortunately the enroute charts aren't online yet |
| 06:07.04 | tronix | it's a good start, at least. |
| 06:07.12 | justinu | true dat |
| 06:10.21 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 06:17.15 | [av]bani | anyone want to buy an h323 phone |
| 06:24.03 | *** join/#asterisk postel (n=jk@asami.zangetsu.jp) |
| 06:34.51 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 06:39.25 | *** join/#asterisk postel (n=jk@asami.zangetsu.jp) |
| 06:39.41 | *** part/#asterisk rene- (i=rene@201.102.18.74) |
| 06:41.29 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 06:45.06 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-226-83.claranet.co.uk) |
| 06:45.33 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 06:49.23 | *** join/#asterisk lalo (n=erg@201.137.152.226) |
| 06:56.07 | exstatica | anyone seen a problem liek ERROR: Module zaptel is in use by ztdummy |
| 06:56.16 | exstatica | i'm trying ot get my digium card installed |
| 06:56.32 | NirS | yes |
| 06:56.51 | exstatica | how do you fix it? |
| 06:56.59 | NirS | this means that you have a ztdummy module loaded, which is blocking you from performing a restart to the zaptel service |
| 06:57.12 | NirS | issue the following: 'rmmod ztdummy' |
| 06:57.18 | NirS | then perform a restart to zaptel |
| 06:58.24 | exstatica | No functioning zap hardware found in /proc/zaptel, loading ztdummy |
| 06:58.42 | NirS | that means that you don't have a zaptel card in the machine |
| 06:58.44 | NirS | is that correct ? |
| 06:58.47 | exstatica | i do |
| 06:59.01 | NirS | ok, in that case it means you didn't follow the instructions of installation |
| 06:59.09 | NirS | what distribution are you using ? |
| 06:59.21 | exstatica | centos |
| 06:59.27 | NirS | 4.2 ? |
| 06:59.33 | *** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net) |
| 06:59.36 | exstatica | yes |
| 06:59.46 | NirS | did you compile by running 'make linux26' ? |
| 07:00.26 | NirS | you need to compile using 'make linux26' then 'make install' |
| 07:00.45 | NirS | then follow the instructions in the README.udev file, as CentOS 4.2 uses udev |
| 07:01.05 | exstatica | hmmm i assumed astrisk@home took care of that |
| 07:01.21 | NirS | oh, it's an A@H |
| 07:01.25 | exstatica | yeah |
| 07:01.26 | NirS | well, I can't help you there |
| 07:01.33 | NirS | I'm not that familiar with it |
| 07:01.36 | FuriousGeorge | anything wrong with this: exten => _91NXXNXXXXXX,1,Dial(${POTSOUT}/ww${EXTEN:1},60,,T) |
| 07:01.45 | exstatica | still the same zaptel driver |
| 07:01.54 | FuriousGeorge | specifically that T option at the end |
| 07:01.58 | joaovianna | Hi guys, I have a problem here. I have SER as regiter for my ATA, but when I forward a call to my * box it is rejected unless I have he peer registred in my * box. Any clue to solve this problem ? |
| 07:02.12 | NirS | Furious, you have 1 too many commas before the T |
| 07:02.23 | FuriousGeorge | joaovianna: add a peer to sip.conf and start asterisk |
| 07:02.30 | NirS | the command is: Dial(Channel/Extension,timeout,options) |
| 07:02.52 | FuriousGeorge | NirS: no i dont, thats not timeout thats hash transfer |
| 07:03.03 | FuriousGeorge | there is no timeout |
| 07:03.13 | NirS | 60 is the timeout |
| 07:03.27 | FuriousGeorge | fine, but i dont specify one there which is why there are two commas |
| 07:03.38 | FuriousGeorge | T is for allow calling party to transfer |
| 07:04.16 | joaovianna | FuriosGeorge: Thanks, but I want to avoid have to add in sip.conf... |
| 07:04.35 | FuriousGeorge | joaovianna: but you just said you needed to add a peer to asterisk |
| 07:04.43 | *** join/#asterisk dasuberd1vid (n=dasuberd@gateway.digium.com) |
| 07:04.51 | FuriousGeorge | a call to my * box it is rejected unless I have he peer registred in my * box. Any clue to solve this problem ?u said: |
| 07:05.37 | NirS | in that case, remove the 60 and one of the commas, and it will work |
| 07:06.19 | joaovianna | FuriosGeorge: Thanks: I want my * as a pstn gateway. In my case I want to * making calls without a "peer" entry in sip.conf. It is possible ? |
| 07:06.44 | FuriousGeorge | NirS: what 60? |
| 07:06.52 | FuriousGeorge | theres none to remove |
| 07:07.26 | FuriousGeorge | and if i take out a comma and leave the T itll complain of now timeout specified in that context when i call. |
| 07:07.48 | FuriousGeorge | joaovianna: im not sure what you want. my understanding is that you are sending a sip call to * by way of SER, right? |
| 07:08.11 | mog_home | man ejabberd rocks FuriousGeorge |
| 07:08.11 | FuriousGeorge | and that call is being rejected because you need a peer entry for SER in *, right? |
| 07:08.19 | FuriousGeorge | lol |
| 07:08.31 | FuriousGeorge | ~FuriousGeorge |
| 07:08.36 | jbot | from memory, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 07:09.13 | joaovianna | FuriousGeorge: Yes, I'm using SER as REGISTER and then forwarding the call to *. |
| 07:09.45 | FuriousGeorge | joaovianna: i never used SER but i assume its a sip call you are sending to asterisk, right? |
| 07:10.25 | joaovianna | FuriousGeorge: Yes. |
| 07:10.59 | FuriousGeorge | mog_home: im talking about hash xfers again. if i want to allow calling party to transfer the call, will this do it: exten => _91NXXNXXXXXX,1,Dial(${POTSOUT}/ww${EXTEN:1},60,,T) (dont i need two commas there to escape the timeout * is looking for) |
| 07:11.06 | FuriousGeorge | i know this worked in 1.0.9 |
| 07:11.11 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:11.13 | mog_home | yes |
| 07:11.16 | mog_home | or || |
| 07:11.23 | mog_home | no one comma |
| 07:11.36 | mog_home | its dial(device/exten|timeout|opts |
| 07:11.37 | mog_home | ( |
| 07:12.17 | joaovianna | FuriousGeorge: Question... How authenticate the call authorizing access to PSTN without a peer entry in sip.conf for each already registred user in my other box (SER) ? |
| 07:12.17 | FuriousGeorge | joaovianna: then, I guess, to answer your question: yes, it is possible to make calls with * without adding a peer to sip.conf, but, only if they arent sip |
| 07:12.49 | FuriousGeorge | joaovianna: you call SER and have it expressly route the call for you? |
| 07:13.46 | *** join/#asterisk postel (n=jk@asami.zangetsu.jp) |
| 07:13.49 | FuriousGeorge | SER is the "gateway" to the PSTN, right? |
| 07:14.01 | joaovianna | FuriousGeorge: Yes. The user is authenticate and registred in SER and the calls (PSTN) are send to * as sip. |
| 07:14.04 | *** join/#asterisk squinky86 (n=ASGjon@64.89.118.139) |
| 07:14.17 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 07:14.17 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 07:15.00 | joaovianna | FuriousGeorge: Is hard to understand that I have to create a sip entry for each customer I have in ser... |
| 07:15.13 | FuriousGeorge | joaovianna: gimme a real world example of what you are trying to do |
| 07:15.57 | FuriousGeorge | so customer logs onto ser and when he makes a call it goes to asterisk |
| 07:16.20 | FuriousGeorge | ? |
| 07:16.50 | joaovianna | FuriousGeorge: Thanks... I have an ATA-186 registering in a free sip server (SER). When my customer call 1XXX-XXXXXXX I sent it to * for PSTN termination... |
| 07:17.02 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 07:18.02 | joaovianna | . |
| 07:18.20 | FuriousGeorge | joaovianna: you're in big trouble, not only are you gonna have to add an entry to sip.conf but youre gonna have to make a dialplan in asterisk |
| 07:18.33 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:18.36 | FuriousGeorge | (extensions.conf) |
| 07:19.05 | joaovianna | FuriousGeorge: Well, my dialplan is create on demand by AGI. |
| 07:19.16 | FuriousGeorge | oh no :) |
| 07:19.23 | FuriousGeorge | i have to sneeze |
| 07:19.27 | FuriousGeorge | ahhh ahhh |
| 07:19.30 | FuriousGeorge | ~docs |
| 07:19.34 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 07:19.34 | FuriousGeorge | !!! |
| 07:20.41 | FuriousGeorge | i know nothing about *@h or any of those automatic asterisk administrator apps, afaik *@h has a freenode channel doesnt it? |
| 07:21.01 | joaovianna | jbot: Thanks... I will check... I found "createautopeer=yes" but it put my * very vulnerable... |
| 07:21.01 | jbot | pas de quoi, joaovianna |
| 07:21.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:21.20 | FuriousGeorge | jbot: parez-vous francaise? |
| 07:21.25 | FuriousGeorge | *parlez :) |
| 07:21.47 | joaovianna | Jbot: Merci. |
| 07:21.49 | jbot | merci is, like, thanks in french |
| 07:21.59 | FuriousGeorge | joaovianna: how come your customers cant log into asterisk |
| 07:22.20 | FuriousGeorge | from their ATA? |
| 07:23.11 | FuriousGeorge | it sounds like you are using SER when you dont have to? |
| 07:23.13 | joaovianna | FuriousGeorge: I don't want my * be the SIP REGISTER. I read SER can handle more users efficiently. |
| 07:23.18 | FuriousGeorge | i could be wrong |
| 07:23.44 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 07:23.51 | ThaZZa_Work | Hey all. |
| 07:24.42 | FuriousGeorge | joaovianna: i cant comment on weather or not that's true, but if asterisk is going to be making the calls anyway, doesnt it just add more latency between asterisk and SER. how many users do you have? cant you just add more asterisk boxes? if you have that many sip customers i'd think you'd want to keep it as streamlined as possible |
| 07:25.05 | FuriousGeorge | i could be wrong |
| 07:25.21 | ThaZZa_Work | Can anyone tell me if there is an mp3 player built into asterisk 1.2.2 ? |
| 07:25.33 | FuriousGeorge | make mpg123 in source dir |
| 07:26.14 | joaovianna | FuriosGeorge: Something like http://www.voip-info.org/wiki-Asterisk+at+large |
| 07:26.16 | ThaZZa_Work | Thank you.. i knew i forgot something this morning at 3am. lol. |
| 07:26.29 | ThaZZa_Work | FuriousGeorge: Sorry that was thank you to you. :D |
| 07:27.17 | FuriousGeorge | ThaZZa_Work: np, wish they were all that easy :) |
| 07:28.31 | joaovianna | FuriosGeorge: Just a comment: I want my * only making calls to PSTN. I still have a lot of calls beetwen my customers using SER. (SIP <--> SIP) |
| 07:28.32 | ptiggerdine | mpg123 is heavily outdated and known to have security issues. |
| 07:28.45 | FuriousGeorge | joaovianna: like i said: i could be wrong :) |
| 07:29.02 | ThaZZa_Work | FuriousGeorge: I am still not getting musiconhold. the CLI just shows it starts and then stops the very next line. any ideas? |
| 07:29.17 | *** join/#asterisk pengyong (n=lala@222.188.133.19) |
| 07:29.18 | FuriousGeorge | ptiggerdine: isnt there another one that works with * |
| 07:29.53 | FuriousGeorge | ThaZZa_Work: check the moh.conf file where your moh should be, i think it should work out of the box |
| 07:29.58 | FuriousGeorge | you did do a make install right? |
| 07:30.58 | ThaZZa_Work | FuriousGeorge: Do i need to recompile asterisk after i make mpg123. cause asterisk was already compiled! |
| 07:31.07 | FuriousGeorge | nm, i dont know that you need to make install mpg123 |
| 07:31.33 | FuriousGeorge | ThaZZa_Work: i dont /think/ so but i could be wrong. are there mp3s in that dir specified in moh.conf |
| 07:32.01 | ThaZZa_Work | FuriousGeorge: but you said to make mpg123. lol.. Yes there are files. there. default moh. |
| 07:32.04 | joaovianna | ThaZZa_Work: What error message you have in your console ? |
| 07:32.08 | *** join/#asterisk EriSan (n=erisan@151.8.109.105) |
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| 07:32.57 | ThaZZa_Work | joaovianna: Upon reload.. none. |
| 07:33.10 | ThaZZa_Work | joaovianna: Following is the CLI 2 lines. |
| 07:33.30 | FuriousGeorge | ThaZZa_Work: for whatever reason i thought the make mpg123 might also install, but i use gentoo so i confuse things like that sometimes |
| 07:33.37 | ThaZZa_Work | -- Executing MusicOnHold("IAX2/207-4", "default") in new stack |
| 07:33.37 | ThaZZa_Work | <PROTECTED> |
| 07:33.37 | ThaZZa_Work | <PROTECTED> |
| 07:34.04 | FuriousGeorge | ThaZZa_Work: did you try restarting * or reload moh |
| 07:34.15 | FuriousGeorge | (guessing |
| 07:34.37 | FuriousGeorge | also check top, see if mpg123 is running |
| 07:34.38 | ThaZZa_Work | FuriousGeorge: Yep both.. The call stays connected. and doesn't hangup. so it must be doing something. |
| 07:35.37 | ThaZZa_Work | FuriousGeorge: there i think is the problem.. did a PS -A.. there is no mpg123. |
| 07:35.49 | FuriousGeorge | try running it from shell |
| 07:36.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:36.43 | ThaZZa_Work | FuriousGeorge: Doesn't exist.. :-( |
| 07:36.50 | FuriousGeorge | hmmm |
| 07:37.20 | FuriousGeorge | im at a loss, moh always just worked with every version of * ive ever installed. worked so well i thought my stupid sip client was installing some sort of terrible moh |
| 07:37.31 | FuriousGeorge | no offense to whoever wrote that piece |
| 07:37.34 | *** join/#asterisk cyber (n=kani@220.247.249.247) |
| 07:37.38 | FuriousGeorge | hope it wasnt you mog_home |
| 07:37.39 | ThaZZa_Work | FuriousGeorge: i think i just found out.. after you do make mpg123 in asterisk src. it created a mpg123 path. i think i need to make in that path. :-) |
| 07:38.14 | FuriousGeorge | ThaZZa_Work: that makes sense, you can make install in there. i knew there was something wierd about it |
| 07:38.36 | ThaZZa_Work | FuriousGeorge: make install in that path works. mpg123 exists now. :-) |
| 07:38.51 | FuriousGeorge | and moh? |
| 07:38.55 | joaovianna | ThaZZa_Work: Try ps -A|grep mpg123 |
| 07:39.17 | ThaZZa_Work | FuriousGeorge: Fixed. |
| 07:39.22 | FuriousGeorge | good job |
| 07:39.30 | ThaZZa_Work | joaovianna: Fixed. It is working. :-) |
| 07:39.48 | ThaZZa_Work | joaovianna & FuriousGeorge: Think i just needed a few more heads attached to my shoulders. :-) |
| 07:39.55 | FuriousGeorge | it was all you |
| 07:40.15 | ThaZZa_Work | FuriousGeorge: Yet you both helped to bounce ideas off. :D |
| 07:40.35 | FuriousGeorge | im bouncy like that |
| 07:40.39 | FuriousGeorge | :) |
| 07:41.09 | joaovianna | :) |
| 07:41.11 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-224-179.dsl.irvnca.pacbell.net) |
| 07:42.39 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:43.28 | ThaZZa_Work | what does this mean? Unable to find a codec translation path from g729 to slin |
| 07:43.32 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:44.46 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 07:46.33 | ThaZZa_Work | Its cool. i worked it out, yet again. lol |
| 07:47.08 | ThaZZa_Work | Home time. then to have more fun. see ya all later. :-D |
| 07:48.13 | FuriousGeorge | mog_home: remember you told me something about having fixed your flux capacitor for that xmpp.patch :). is that not on the web yet cuz i got a similar error |
| 07:48.38 | mog_home | did you check out from my svn? |
| 07:49.23 | FuriousGeorge | mog_home: cvs??!! i use gentoo, we're not allowed. j/k |
| 07:49.42 | JunK-Y | yo, weather in SF is great! |
| 07:52.40 | mog_home | svn.... |
| 07:52.43 | mog_home | not vcs |
| 07:53.32 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:53.40 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:53.49 | elephantMan | You have a lot of chance to be in SF |
| 07:54.09 | elephantMan | Weather in Paris sucks |
| 07:57.08 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
| 08:00.22 | stefanomasini | hi, can I register all 4 lines of my GXP-2000 with the same SIP account? |
| 08:08.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:08.45 | *** join/#asterisk rene- (i=rene@201.135.231.238) |
| 08:09.39 | [av]bani | what do you mean? |
| 08:09.56 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
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| 08:23.05 | Newbie___ | anyone does h323 installation consultancy service here |
| 08:25.21 | *** join/#asterisk graabein (n=gunnar@bkkb-gw.bitcon.no) |
| 08:26.12 | *** join/#asterisk lorinc (n=ang@caracas-3816.adsl.interware.hu) |
| 08:26.19 | trixter | how much does it pay? |
| 08:27.20 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 08:28.03 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:29.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@82.166.242.248) |
| 08:29.20 | Newbie___ | i pretty much can get asterisk running except for h323. keep getting error when compiling pwlib |
| 08:30.47 | Newbie___ | and yes, i read the README and use the exact method |
| 08:31.37 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:32.30 | fdask | whats the error you get |
| 08:32.37 | fdask | what os are you compiling on |
| 08:33.43 | MrChimpy | <PROTECTED> |
| 08:34.07 | MrChimpy | but the figure isn't being reflected in ${CALLERIDNUM} |
| 08:34.18 | MrChimpy | what's up there? this in inbound from an E1. |
| 08:34.57 | Newbie___ | fdask: running on whitebox linux EL 4 on dell SC 430 |
| 08:36.14 | fdask | whitebox linux? |
| 08:36.22 | Newbie___ | http://pastebin.ca/38311 |
| 08:36.39 | Newbie___ | is the same as Red hat 4 |
| 08:37.14 | Newbie___ | or is it FC4 hehe |
| 08:38.15 | *** join/#asterisk mko-025 (n=korpim@p54989F9B.dip0.t-ipconnect.de) |
| 08:38.21 | fdask | thats a weird error |
| 08:38.24 | *** join/#asterisk Tili (i=Tili@203.101.160.158) |
| 08:38.44 | Newbie___ | tell me about it, been tying to compile for the last 24 hrs |
| 08:39.10 | fdask | can you just skip building this asnparser tool |
| 08:39.14 | fdask | or do you need it |
| 08:40.03 | L|NUX | Newbie___ : they fork from RH |
| 08:40.14 | L|NUX | fdask : its for ya |
| 08:40.15 | Newbie___ | fdask: i have no idea what it does but what ever is from the README |
| 08:40.26 | Newbie___ | L|NUX: yeah |
| 08:40.37 | *** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net) |
| 08:40.45 | SERGEUS|W | i have small problem with "-r" flag |
| 08:40.51 | SERGEUS|W | /usr/sbin/asterisk -vvvvr |
| 08:40.51 | SERGEUS|W | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 08:40.53 | SERGEUS|W | but |
| 08:41.09 | L|NUX | ? |
| 08:41.17 | SERGEUS|W | it is exist |
| 08:41.18 | L|NUX | SERGEUS|W : make sure asterisk is running :) |
| 08:41.18 | SERGEUS|W | srwxr-xr-x 1 root root 0 Jan 23 13:18 /var/run/asterisk/asterisk.ctl |
| 08:41.23 | L|NUX | ps -ef | grep asterisk |
| 08:41.32 | L|NUX | SERGEUS|W : ps -ef | grep asterisk |
| 08:41.34 | Mag1KaL | How exactly is the 's' extension used? I mean what actually triggers it if the user hasn't even called anything yet? |
| 08:41.40 | SERGEUS|W | yes it's up and runing |
| 08:42.14 | SERGEUS|W | 29036 ? Ss 0:48 /usr/sbin/asterisk |
| 08:42.18 | L|NUX | hum |
| 08:42.42 | L|NUX | its working fine for me |
| 08:42.51 | L|NUX | try to restart asterisk server |
| 08:43.10 | SERGEUS|W | already tryed :) |
| 08:43.19 | Newbie___ | damn, previously did compile on whitebox EL 3 and working great |
| 08:43.27 | SERGEUS|W | "-r" refuses to work... |
| 08:43.41 | SERGEUS|W | i have no idea why |
| 08:43.44 | L|NUX | which version |
| 08:43.52 | SERGEUS|W | mine? |
| 08:44.06 | SERGEUS|W | SVN-trunk-r8447M |
| 08:44.21 | L|NUX | humm |
| 08:44.29 | Newbie___ | fdask ?? |
| 08:44.57 | SERGEUS|W | i also tryed to update it.. no result |
| 08:45.27 | *** join/#asterisk psyco-obiwan (n=cschnee@2001:4060:4419:b1:0:0:0:2) |
| 08:46.42 | psyco-obiwan | hi, i know asterisk can easily work pretty fine on a 166MHz with 128MB Ram, however does anybody know the min. requirements for *@home ?? |
| 08:46.43 | fdask | Newbie___: if you had it working on EL 3, why not go back to that |
| 08:46.47 | SERGEUS|W | any ideas? :) |
| 08:47.11 | DarkFlibble | psyco-obiwan, 166 is a litle low for asterisk... |
| 08:48.00 | DarkFlibble | according to to wiki while asterisk has been run sucessfully on a 133 we recommend a 500Mhz machine or higher |
| 08:48.15 | psyco-obiwan | DarkFlibble: i am running my external trunk gateway on a 200MHz with 96M...conferences with more than 6 users worked pretty fine without sparking the cpu at all (all codecs were translated between users..) |
| 08:48.22 | Newbie___ | fdask: EL3 is already up and running for few months but is on a clone pc. bought a new dell sc 430 but is a sata drive, which only EL 4 supports it |
| 08:49.16 | DarkFlibble | psyco-obiwan, wow... can you do show translations and paste it in a paste bin for curiosity sake? |
| 08:50.00 | psyco-obiwan | i just wanted to prepare a machine for a friend (one outgoing, one internal, conf, vm) and found that all the goodies in *@home in webinterface just yield a 404... |
| 08:50.09 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-97-200.telkomadsl.co.za) |
| 08:50.09 | psyco-obiwan | DarkFlibble: while the conf is running you mean ? |
| 08:50.25 | X-Gen | ja freaks |
| 08:50.33 | DarkFlibble | no... just anytime... it benchmarks the machine |
| 08:52.27 | bennyben | hi, I've got problems with Flash() that don't do exactly the same thing than the 'R' key on my phone, anybody can help me ? |
| 08:53.01 | bennyben | I've made a ztmonitor to listen the line to compare the 2 one, and I haven't the same signal |
| 08:53.57 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:54.16 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 08:54.16 | psyco-obiwan | DarkFlibble: http://pastebin.com/520297 |
| 08:54.57 | DarkFlibble | k...thnx |
| 08:55.00 | L|NUX | SERGEUS|W : there ? |
| 08:55.05 | SERGEUS|W | yep |
| 08:55.06 | L|NUX | SERGEUS|W : FYI, http://forums.digium.com/viewtopic.php?t=3705 |
| 08:55.27 | rene- | hi what is the default password for ftp in polycom phones? |
| 08:56.13 | DarkFlibble | psyco-obiwan, you said 6 channels with "all codecs were translated between users.." what codecs were they? mostly the high bandwidth codecs? |
| 08:56.26 | *** join/#asterisk CleanerX (n=nix@vpnwww01.rz.uni-karlsruhe.de) |
| 08:58.04 | SERGEUS|W | L|NUX, yeah, guy have the same problem |
| 08:58.15 | psyco-obiwan | ulaw and alaw mostly i guess but cant say exactly, i was just trying to get the thing breathing with all my sip phones i had (my gf thought im getting crazy talking into 6 phones concurrently ;-)) |
| 08:59.08 | DarkFlibble | psyco-obiwan, based on what you pasted, if you only run the ulaw/alaw/adpcm/slin you will manage to get a few channels from it... |
| 08:59.46 | DarkFlibble | although one change on the basic install I would make is change the prompts format to one of those from gsm... |
| 09:00.23 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:00.24 | L|NUX | humm |
| 09:00.36 | psyco-obiwan | do you say, that i wouldn't get much higher than six with this machine ? |
| 09:00.41 | DarkFlibble | since decoding/encoding gsm is cpu intensive... and you don't have a cpu with cycles to spare |
| 09:01.32 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:01.34 | psyco-obiwan | as a rule of dumb, can i say, the lower the bw of a codec the higher the cpu cost ? |
| 09:01.34 | DarkFlibble | psyco-obiwan, depends what they are doing... if they are all in ivr listening to gsm prompts you would be unlikely to get more than 1... |
| 09:01.46 | DarkFlibble | when in calls tho... it could possibly support 10 |
| 09:01.59 | DarkFlibble | psyco-obiwan, look at the tables you produced... |
| 09:02.01 | psyco-obiwan | cool to get some figures ... thx |
| 09:02.05 | psyco-obiwan | yep |
| 09:02.15 | DarkFlibble | the lower the figure the better |
| 09:02.45 | psyco-obiwan | is there a certain limit from which you can guess the number of concurrent calls translated with a certain codec ? |
| 09:03.18 | trixter | afaik there are only guesses and not real numbers |
| 09:03.28 | trixter | and it depends more than translation as to what the real load is |
| 09:03.37 | psyco-obiwan | yeah, sure but better an educated guess than nothin ;-) |
| 09:03.52 | X-Gen | someone should make a pci card that can offload that processing ;) |
| 09:04.03 | trixter | call duration, especially if you call AGIs matters, high volume low duration calls will cause more load (due to fork and exec of the agi) than lower volume higher call duration calls |
| 09:04.07 | *** join/#asterisk blop (i=blop@openbeer.be) |
| 09:04.13 | psyco-obiwan | after what i saw i was thinking i could handle dozens of calls on that hardware, didn't know i was so close crossing the limit |
| 09:04.23 | DarkFlibble | psyco-obiwan, you can attempt to guess based on the figure in the column... <guess> If you are using 20ms samples and it takes 18ms to translate between one format and another (from the table) then you will only be able to support one channel... </guess> |
| 09:04.24 | trixter | 30-50MHz per channel plus codec translation is a guess |
| 09:04.39 | trixter | depending on exactly what you are translating from/to matters as well |
| 09:04.47 | trixter | asterisk -rx show translation |
| 09:04.52 | psyco-obiwan | tanks DarkFlibble thats something i can make estimates with |
| 09:04.54 | robin_z | morning ... |
| 09:04.58 | trixter | that will let you know what your box will do for a given codec translation |
| 09:05.13 | robin_z | anyone happen to know any of the staff/owners/operators of voipgate.com ??? |
| 09:05.14 | DarkFlibble | trixter, http://pastebin.com/520297 <-- he already did it... |
| 09:05.19 | psyco-obiwan | trixter: DarkFlibble just helped me show the translation to pastebin |
| 09:05.21 | psyco-obiwan | lol |
| 09:05.33 | trixter | ahh I missed that, oh well I have to get up in 3 hours to drive to etel to give my presentation so I should sleep |
| 09:05.43 | DarkFlibble | trixter, good luck |
| 09:05.56 | trixter | thanks I will need it for the bridge crossing at 9am |
| 09:05.59 | robin_z | voipgate.com-- # wankers. name servers down, all on the same network, adjavcent ips |
| 09:06.10 | trixter | <-- hates SF traffic, although its not as bad as NJ |
| 09:06.34 | trixter | out of SF, LA, San Diego, Sacramento, NYC, Boston, NJ was by far the worst |
| 09:06.37 | DarkFlibble | robin_z, network admins can't design fault tolerent networks for shit these days |
| 09:06.51 | robin_z | so ... remind me .. a good euro sip/iax termination service is???? |
| 09:07.11 | DarkFlibble | robin_z, me in a week... |
| 09:07.20 | robin_z | DarkFlibble: you have to be particularly clueless to put all your NS's next to each other |
| 09:07.37 | psyco-obiwan | DarkFlibble: am I right that if i have an ISDN trunk comin in/out and i am translating them to ulaw/alaw that is not going to produce a great load then, right ? |
| 09:08.09 | psyco-obiwan | there was onetime a forum for exchanging sec. NS services to each other...but I cant find it anymore |
| 09:08.16 | DarkFlibble | psyco-obiwan, ISDN is natively very close to ulaw (it is ulaw I think) so it should be low load... |
| 09:08.28 | trixter | if you start with g.711 and push it over isdn you dont need that beefy of a machine |
| 09:08.32 | trixter | people do that on pIII 1000MHz |
| 09:08.45 | DarkFlibble | I use dyndns's custom dns because I know my name servers suck... |
| 09:08.49 | trixter | a few of those doing 1.5M minutes per month |
| 09:09.26 | DarkFlibble | also they have a nice web interface |
| 09:09.40 | trixter | isdn is ulaw some places alaw others |
| 09:09.42 | robin_z | I just run 2 ns on seperate class Cs in the UK and another in Lausanne .ch for safety |
| 09:09.47 | trixter | it depends largely if you are north american or not |
| 09:10.01 | DarkFlibble | ulaw in US, alaw in europe... i think |
| 09:10.05 | trixter | and g.711 doesnt really compress/decompress its quantization which has some impact but not that much |
| 09:10.21 | trixter | not compared to some other codecs |
| 09:10.34 | trixter | although asterisk treats everything as slinear internally iirc |
| 09:10.37 | robin_z | so, anyway, offically, voipgate suck. BIG TIME. 2 days now, no ns can be reached |
| 09:10.48 | trixter | everything converts to slin and from slin to something else when doing a translation |
| 09:11.07 | psyco-obiwan | trixter, are you saying that 711 should be the default to go instead of alaw/ulaw which ive seen most use if bw doesnt matter... |
| 09:11.17 | trixter | as such if there is any translation it is effectively converting twice, but slin isnt really anything other than raw, its a little different but not much |
| 09:11.30 | trixter | g711 *is* a/ulaw |
| 09:11.39 | trixter | depending on which subtype |
| 09:12.01 | psyco-obiwan | arghh...hate those buzzwords and abbreviations... |
| 09:12.16 | psyco-obiwan | its called different all over the place... |
| 09:12.24 | trixter | its not a buzzword, its the proper name of the codec :) |
| 09:12.25 | psyco-obiwan | no excuse :-) |
| 09:13.08 | DarkFlibble | robin_z, okay to pm you? |
| 09:13.14 | trixter | anyway I gotta goto sleep cause I got to get up early ... gah less than 3 hours sleep for a 4+ hour drive then 11 hours after that until my presentation, tomorrow is going to suck |
| 09:13.27 | psyco-obiwan | gl trixter and thx for the fine help! |
| 09:13.46 | trixter | rgabjs |
| 09:13.50 | trixter | \er thanks |
| 09:14.03 | psyco-obiwan | shift-left-1 ;-) |
| 09:15.13 | DarkFlibble | psyco-obiwan, you can use refurbs if cost is an issue... |
| 09:15.36 | DarkFlibble | but to be honest dell 1u poweredge servers are dirt cheap now |
| 09:15.55 | psyco-obiwan | dont say that....last time i said that i ended up setting an alpha 2100/4 275.... LOL |
| 09:15.56 | DarkFlibble | running my dev box on a 2600xp |
| 09:16.21 | psyco-obiwan | seriously i like the new Sun machines the X[42]x00 |
| 09:16.48 | DarkFlibble | for my own machines, I'll stick with proliants... |
| 09:16.55 | psyco-obiwan | installed a few of them and with debian64 they rock.. |
| 09:17.04 | DarkFlibble | got a nice quad xeon for free the other week... |
| 09:17.09 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 09:17.21 | DarkFlibble | 4x1Ghz |
| 09:18.14 | DarkFlibble | turn it on and need to go to a different room... it is a jet engine inside... |
| 09:18.45 | DarkFlibble | 4 fans in a 2 by 2 configuration... fault tolerant... |
| 09:18.47 | psyco-obiwan | i know that from the sun machines...you feel like putting a weight onto them so that they don't hover away...lol |
| 09:19.07 | stefanomasini | hi, can I register all 4 lines of my GXP-2000 with the same SIP account? |
| 09:19.08 | MrChimpy | very little chance of anything from sun ever floating away :) |
| 09:19.16 | stefanomasini | I'm getting weird behaviours... |
| 09:19.24 | psyco-obiwan | the 4100 (1U) has a battery of 6 fans which can be hot replaced...it looks really like a jet engine |
| 09:19.25 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 09:19.27 | NoRemorse | hi all |
| 09:19.29 | DarkFlibble | my firewall is bad enough... standard antec case... |
| 09:19.39 | DarkFlibble | stefanomasini, depends on a few things |
| 09:19.55 | stefanomasini | DarkFlibble, mhm... such as? |
| 09:19.55 | psyco-obiwan | stefanomasini: i think that depends on your sip provider, asterisk can afaik |
| 09:20.03 | psyco-obiwan | cant! |
| 09:20.04 | psyco-obiwan | sorry |
| 09:20.05 | stefanomasini | asterisk it is |
| 09:20.11 | DarkFlibble | capabilities of the phone... |
| 09:20.21 | stefanomasini | gxp-2000 |
| 09:20.23 | MrChimpy | seen an IBM Bladecenter? they have ENORMOUS fans. two per 14 blade chassis IIRC. they look like washing machine components. |
| 09:20.25 | DarkFlibble | capabilities of the provider |
| 09:20.31 | stefanomasini | asterisk |
| 09:20.37 | DarkFlibble | MrChimpy, yeah... |
| 09:20.45 | psyco-obiwan | i would say you cant with asterisk and Grandstream... |
| 09:20.53 | DarkFlibble | Unlikely to get given blades for free tho... :( |
| 09:20.55 | MrChimpy | and you could hear ours start up from the room *next* to the comms room :) |
| 09:21.04 | stefanomasini | psyco-obiwan: mhm... and you might be right |
| 09:21.27 | MrChimpy | df: i suspect 28 CPUs across 14 machines may be overkill for hobbyists |
| 09:21.35 | DarkFlibble | stefanomasini, I use an SPA-841 with shared line apperence |
| 09:21.48 | DarkFlibble | MrChimpy, wanna bet? |
| 09:21.50 | DarkFlibble | :P |
| 09:21.52 | MrChimpy | :) |
| 09:21.53 | psyco-obiwan | stefanomasini: i believe asterisk doesn't support multiple subscriptions for one sip account |
| 09:22.01 | Math` | it doesnt |
| 09:22.12 | stefanomasini | the thing is that I want to have all the 4 lines activated (i.e. usable to call out), but would like to return busy tone to callers if they try to call the phone and at least one line is being used... |
| 09:22.15 | DarkFlibble | SER should do tho... |
| 09:22.40 | Math` | stefanomasini: there are functions to limit the numbers of concurrent calls... |
| 09:22.44 | psyco-obiwan | stefanomasini: i guess you can do that with a bit of extensions.conf magic |
| 09:22.47 | DarkFlibble | stefanomasini, then register them as seperate lines and use a call group |
| 09:22.55 | Math` | I believe they are SetGroup() and CheckGroup() |
| 09:23.03 | psyco-obiwan | i have a gpx2000 for my home office... |
| 09:23.19 | *** join/#asterisk mgoh (n=goh@60.49.6.190) |
| 09:23.22 | MrChimpy | it's 9.30AM. I've been in work since 4.30AM. still ages until hometime! |
| 09:23.47 | NoRemorse | does anyone know how to call a default SetAccount() from the begining of a context rather than at the start of every dial plan branch>? |
| 09:23.54 | DarkFlibble | hmmm... got 15 incoming DIDs on my *home* asterisk box |
| 09:24.02 | stefanomasini | mhmm... call group. I'll check it out. thanks. But unfortunately there seems to be a bug with groups that screws up my cdr logging... |
| 09:24.18 | Math` | ah its 4:30 am here |
| 09:24.28 | DarkFlibble | 9:24am here |
| 09:24.37 | MrChimpy | df: uk? |
| 09:24.43 | DarkFlibble | yup... Leicester |
| 09:24.48 | MrChimpy | ah. london |
| 09:24.50 | psyco-obiwan | 10:24 here...and -6 degrees C *brrrr* |
| 09:25.01 | MrChimpy | m'grandfolks are in leicester :) |
| 09:25.12 | DarkFlibble | cool... |
| 09:25.20 | DarkFlibble | whereabouts? |
| 09:25.20 | Math` | -2 here and its usually -20 at that period of the year heh |
| 09:25.27 | MrChimpy | countesthorpe |
| 09:25.37 | DarkFlibble | not that far away... |
| 09:25.45 | *** join/#asterisk Tili (i=Tili@203.101.161.248) |
| 09:26.09 | MrChimpy | I still don't get why i can see : |
| 09:26.15 | MrChimpy | Accepting call from '2073096600' to '500' on channel 0/1, span 1 |
| 09:26.19 | DarkFlibble | maybe we should arrange a UK meetup for the peeps in the channel sometime |
| 09:27.08 | MrChimpy | yet in the dialplan ${CALLERNUMID} isn't filled in |
| 09:27.23 | DarkFlibble | MrChimpy, what version of asterisk you using? |
| 09:27.26 | MrChimpy | we could do with an astricon here ;) |
| 09:27.49 | *** join/#asterisk Assid (n=assid@203.115.64.10) |
| 09:28.00 | DarkFlibble | an astricon *outside* london would be nice... hotels are sooo expensive there |
| 09:28.04 | MrChimpy | 1.2.1 |
| 09:28.47 | DarkFlibble | MrChimpy, NoOP(${CALLERID(number)}) |
| 09:28.57 | DarkFlibble | thats what I have for callerid on 1.2.2 |
| 09:29.14 | DarkFlibble | ${CALLERID} was depreciated... |
| 09:29.58 | DarkFlibble | not sure thats 100% right tho... |
| 09:30.04 | Math` | it was |
| 09:30.10 | DarkFlibble | since I was a little tired at the time |
| 09:30.24 | Math` | but ${CALLERID(number)} is the new way |
| 09:32.01 | MrChimpy | are there up to date docs on this? |
| 09:32.08 | lahaine | hi |
| 09:32.22 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 09:32.29 | DarkFlibble | yes.. but they are linked in very few places |
| 09:32.51 | DarkFlibble | http://www.voip-info.org/wiki/view/Functions |
| 09:32.53 | MrChimpy | could you point me at them, or let me know how to get the DNID in the new system? |
| 09:32.53 | Math` | MrChimpy: show function CALLERID (on cli) |
| 09:32.59 | Math` | MrChimpy: or show functions to get a list |
| 09:33.01 | MrChimpy | cool. ta math |
| 09:33.19 | Math` | same as show applications :) |
| 09:34.14 | *** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 09:35.42 | DarkFlibble | got a meeting with a university later today... with regards to coding some voip products with their phd students.... what new applications/asterisk addons would you like to see? |
| 09:35.58 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:36.04 | MrChimpy | speech recognition :) |
| 09:36.15 | DarkFlibble | already got that in sphinx... |
| 09:36.22 | MrChimpy | is it any good? |
| 09:36.28 | DarkFlibble | never used it... |
| 09:36.45 | Math` | DarkFlibble: engineer a new royalty-free voice compression codec :) |
| 09:36.45 | DarkFlibble | but there are two types of recognition... |
| 09:37.04 | DarkFlibble | Math`, whats wrong with the dozens that exist already? |
| 09:37.26 | Math` | there are royalty-free codecs that can do 6-8kbps per channel? |
| 09:37.33 | DarkFlibble | lpc10 |
| 09:37.37 | DarkFlibble | :P |
| 09:37.45 | Math` | that has a decent sound? |
| 09:37.53 | DarkFlibble | you didn't say that |
| 09:37.54 | Math` | (I didnt try lpc10 til nobody supprots it) |
| 09:37.55 | DarkFlibble | :P |
| 09:38.05 | Math` | s/supprots/supports/ |
| 09:38.22 | DarkFlibble | any asterisk specific codec wont be supported that well... |
| 09:38.41 | DarkFlibble | since many providers only use the subset that is supported by all their equipment... |
| 09:38.55 | Math` | a codec is essentially an algorithm... if its open and great... its gonna be widespread |
| 09:38.56 | DarkFlibble | ie.. ulaw/gsm or ulaw/g729 |
| 09:39.27 | DarkFlibble | Math`, gsm and ilbc are open and neither are *that* widespread |
| 09:40.22 | Math` | ok |
| 09:40.42 | DarkFlibble | ilbc is specified in an rfc... |
| 09:40.53 | Math` | ok I shut up :P |
| 09:41.07 | DarkFlibble | any other suggestions? |
| 09:41.12 | NoRemorse | hey if anyone is interested, you can set accountcode in sip.conf speicificaly for each user, dont have to use SetAccount in the dialplans |
| 09:41.36 | Math` | of course you can |
| 09:41.48 | NoRemorse | yeah well you didnt say that before when ia asked lol |
| 09:42.12 | Math` | accountcode is for CDRs |
| 09:42.17 | NoRemorse | yep. |
| 09:42.28 | Math` | I didnt see the question sorry |
| 09:42.46 | Math` | (working at the same time and its 4:45 am so... tired a bit :P) |
| 09:43.02 | *** join/#asterisk _deg_ (n=deg@201.22.27.49.adsl.gvt.net.br) |
| 09:43.06 | NoRemorse | the first time I did my dialplan I mistook account code for a service tag, ie MOBILE, IDD etc. just redid it to set it as the customers account code . useing dcontext to determine what service was called now. |
| 09:43.14 | NoRemorse | yeah no prob hehe wasn't having a go at you |
| 09:43.19 | NoRemorse | question was: |
| 09:43.22 | NoRemorse | does anyone know how to call a default SetAccount() from the begining of a context rather than at the start of every dial plan branch>? |
| 09:43.25 | DarkFlibble | I'll suggest proper asterisk load balancing... |
| 09:43.29 | Math` | ah |
| 09:44.20 | DarkFlibble | so phones can log on to any box they wish and recieve calls there from any server... |
| 09:45.00 | Math` | that's basically data replication |
| 09:45.08 | Math` | and synchronization |
| 09:45.20 | DarkFlibble | Math`, yes... but asterisk is unable to do it from what I've seen... |
| 09:45.50 | Math` | uhm didnt try it but I think it can be done |
| 09:46.17 | Math` | regexten/regcontext will add the Dial() command automatically in the dialplan, you'd need to share that dialplan using DUNDi over the servers |
| 09:46.35 | Math` | and... use OpenSER or SER to load balance incoming sip calls to different servers |
| 09:47.02 | Math` | the local extensions gets routed thru dundi, the rest is sync'd (maybe even on net storage) |
| 09:47.31 | *** join/#asterisk j0 (n=dan@bb58-185-10-236.singnet.com.sg) |
| 09:47.51 | DarkFlibble | I'll look into it later today... |
| 09:48.06 | Math` | ok |
| 09:49.27 | MrChimpy | my final aim is running many IVR chat services across many servers - they should all perform identically and conferences should be distributed between servers |
| 09:49.58 | MrChimpy | not sure if DUNDi would help much there |
| 09:50.26 | Math` | DUNDi helps locating where is the phone registered, and thats almost all |
| 09:50.49 | MrChimpy | yeah. i could use it to find the current server for the conference, I guess |
| 09:50.55 | Math` | yeah |
| 09:51.02 | MrChimpy | probably best off doing it some custom way |
| 09:53.36 | *** join/#asterisk mrtwister (n=mrtwiste@cable-10-68.cgates.lt) |
| 09:53.43 | Math` | yeah probably |
| 09:54.20 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:54.35 | *** join/#asterisk TrisTan (n=yellowgi@85.102.157.233) |
| 09:54.44 | pif | hi, how is callerid suppression typically managed in the dialplan? |
| 09:54.58 | pif | a * prefix to the number? |
| 09:55.10 | pif | what is the current practice? |
| 09:58.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:58.59 | puzzled | morning |
| 10:01.33 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:03.59 | *** join/#asterisk ajav (n=ajav@58.69.204.92) |
| 10:05.00 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 10:05.22 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 10:05.31 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 10:07.04 | *** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 10:07.11 | *** join/#asterisk heath__ (n=heath__@12-215-33-205.client.mchsi.com) |
| 10:08.18 | heath__ | i've been working on this clustering framework for asterisk and i've been calling it "astcluster" for the last few days until about 5 minutes ago when i realized how horrible it sounds |
| 10:09.02 | Mavvie | with regarding to Dundi peering, is there somebody in .au I could peer with? |
| 10:10.00 | heath__ | i bet you don't want to peer into my astcluster !!! ahahahahaha |
| 10:10.14 | Mavvie | you lost. |
| 10:10.46 | dudes | heath__ - is just being a smartass dick wad |
| 10:12.11 | |vinsik| | hehe |
| 10:13.11 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:13.23 | heath__ | i just want someone else to say "astcluster" |
| 10:15.19 | h3x | asscluster. |
| 10:15.33 | heath__ | awesome |
| 10:15.34 | h3x | er. astcluster |
| 10:15.36 | *** join/#asterisk Bger (n=Bager@83.222.160.88) |
| 10:15.43 | dudes | I was wondering when you were going to hop on |
| 10:16.06 | h3x | now what did i do or not do |
| 10:16.22 | Bger | hi! |
| 10:18.38 | Bger | i'd like to ask whether asterisk can read the phone info from a panasonic PBX (so called "digital super-hybrid system" ones) |
| 10:19.00 | dudes | They'd have complete dialog! |
| 10:19.14 | h3x | read what info |
| 10:19.18 | Bger | sorry if i'm asking dumb questions, i'm really not into this stuff at all |
| 10:19.30 | Bger | h3x abouth the phone calls |
| 10:19.59 | dudes | I'm not sure, I just felt like being a smartass :) <---- haha, looks like Jennifer Lopez from Southpark |
| 10:20.04 | Bger | also, is there something that is not a complete PBX (like asterisk), that would do the same job |
| 10:20.39 | h3x | do... what job |
| 10:20.49 | *** part/#asterisk ajav (n=ajav@58.69.204.92) |
| 10:20.57 | h3x | what are you trying to do |
| 10:21.00 | Bger | h3x to read the phone calls ... from where ... to .. etc ... |
| 10:21.05 | Bger | phone billing :) |
| 10:21.14 | h3x | theres a serial port on your phone system for that |
| 10:21.20 | Bger | yep |
| 10:21.26 | Bger | there is such thing |
| 10:21.33 | h3x | thats got nothing to do with asterisk |
| 10:21.51 | Bger | sorry then ... |
| 10:22.01 | mgoh | Can T1 interface card connected to phone? or T1 interface card is just a card that can connect to PSTN? |
| 10:22.06 | Bger | do you know a program that will do this job |
| 10:22.09 | h3x | you hook it up and write a program to decypher the stuff from that port |
| 10:22.16 | Bger | ah |
| 10:22.49 | h3x | haha i told you guys my t1 phone idea was a good one |
| 10:22.51 | Bger | h3x so there are no such programs already written?... |
| 10:22.57 | h3x | maybe |
| 10:23.04 | h3x | search freshmeat for one |
| 10:23.13 | Bger | i tried ... but. .. |
| 10:23.59 | h3x | youd be better off switching your whole phone system to asterisk |
| 10:24.23 | h3x | just because its difficult to prevent people to dial certain calls on ksu's |
| 10:24.34 | mgoh | Digital Interface Cards like T1 can it connect to digital phone? |
| 10:25.18 | h3x | no |
| 10:25.38 | h3x | t1s are digital circuits that carry 24 lines |
| 10:25.40 | Bger | h3x it's not about preventing the people, it's more like there are 3 little companies using one PBX and they want to divide their bills |
| 10:25.42 | h3x | do you really need 24 lines on one phone |
| 10:26.04 | h3x | bger use account codes with your telco |
| 10:26.08 | h3x | verified account codes |
| 10:26.36 | Bger | hm |
| 10:26.49 | mgoh | no I think use channel bank to seperate it to 24 lines |
| 10:27.01 | h3x | oh you mean that |
| 10:27.09 | h3x | of course you can use a channelbank to get you 24 lines |
| 10:27.17 | h3x | asterisk can pretend to be the telco to your channelbank |
| 10:27.24 | *** join/#asterisk astr (n=ts@59.93.71.9) |
| 10:27.39 | *** part/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 10:27.46 | astr | does asterisk support vad? |
| 10:27.59 | mgoh | h3x: every single line now is consider digital or analog? |
| 10:28.06 | h3x | as in voip vad/cng? |
| 10:28.20 | h3x | mgoh... analog |
| 10:28.51 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:28.55 | astr | hsx: we meet again. yes vad as in voip/cng |
| 10:29.37 | mgoh | h3x: have any method that we can connect to digital phone that user in normal digital- PBX |
| 10:29.53 | h3x | what, isdn phones ? |
| 10:30.00 | h3x | just use voip phones |
| 10:30.17 | h3x | Oh you mean like proprietary digital phones |
| 10:30.25 | mgoh | h3x: yap |
| 10:30.28 | h3x | yes |
| 10:31.00 | h3x | www.citel.com |
| 10:31.08 | h3x | its really expensive tho |
| 10:31.12 | mgoh | h3x: really we can connect to proprietary digital phones. what hardware we need to use? |
| 10:31.12 | h3x | you may as well just buy new phones |
| 10:31.30 | astr | h3x: really, I did not know that asterisk was capable of supporting vad/cng? I searched on google. no results |
| 10:31.44 | h3x | astr: it isnt |
| 10:31.53 | NoRemorse | I have accountcode= set in sip.conf for a peering asterisk server, but it is not being sent to the cdr record, any ideas please? |
| 10:32.03 | mgoh | h3x: but proprietary digital phones provide alot of feature. or u think using feature IP Phone is better? |
| 10:32.29 | h3x | The features are determined by the citel box in this case |
| 10:32.46 | h3x | im sure its fine but we're talking over $150 an extension whereas you coul djust buy nice voip phones |
| 10:33.19 | h3x | the citel converts a amphenol connector full of 24 digital phones into voip on ethernet |
| 10:33.22 | mgoh | h3x:you are right. |
| 10:33.28 | h3x | it replaces your KSU |
| 10:33.48 | mgoh | h3x: seen like using ip phone is better and more reasonable. |
| 10:34.01 | h3x | now if you had like say 500-1000 extensions and didnt wanna rewire for ethernet i could understand using that citel box |
| 10:34.59 | h3x | snom's are cheaper now |
| 10:36.38 | mgoh | h3x: but I scare with the network traffic if all are IP base and direct to all the desk. |
| 10:36.42 | h3x | depending on what youa re doing, snom 320/360, polycom whatever, and cisco/linksys 941s are nice |
| 10:37.07 | h3x | its hardly any bandwidth if you use something like g.729 |
| 10:37.57 | astr | h3x: I went to your website and was looking for someone to talk to last friday and used your contact us form. Nobody got back to us still |
| 10:38.15 | h3x | what was your email |
| 10:38.32 | h3x | i found out my jackass developer rebooted that box and the mailserver didnt restart correctly |
| 10:39.17 | astr | :) |
| 10:39.31 | astr | h3x: anyway to contact you directly. im? |
| 10:39.43 | h3x | yeah |
| 10:40.00 | dudes | is your mail server up now? |
| 10:40.01 | h3x | so when i started up the mail server about 30 new inquries came in |
| 10:40.10 | dudes | ah |
| 10:40.14 | h3x | this is just the mail server on the webserver box |
| 10:40.16 | mgoh | h3x:any way to monitor and calc how many channel my network able to support. sometime even using single channel voip my converstation still delay or lost packet. |
| 10:40.17 | h3x | not my main email |
| 10:40.21 | h3x | you know i would have noticed that :P |
| 10:40.25 | NoRemorse | does accountcode=blah get put in cdr's if host does not =dynamic? |
| 10:41.09 | astr | h3x: do you suggest to use your contact us form again? |
| 10:41.28 | h3x | astr: i am sure i have it, but you gotta im me your email so i can find it |
| 10:41.34 | h3x | http://www.asteriskguru.com/bandwidth_calculator.php |
| 10:43.08 | astr | h3x: this room does not not allow ims. I guess we will just wait for the response and follow up |
| 10:43.34 | h3x | domain name maybe? |
| 10:44.09 | mgoh | h3x: but I dun know recently my network already use how many bandwidth? |
| 10:45.39 | h3x | mgoh let me guess |
| 10:45.48 | h3x | you have a shitty ass linksys/netgear/airlink firewall |
| 10:46.12 | dudes | I bet he has a 64kbps up |
| 10:46.18 | h3x | hahaha |
| 10:46.20 | mgoh | sure |
| 10:46.35 | h3x | both? heh |
| 10:46.45 | mgoh | 2mb actually but is broadband then speed is not consisten |
| 10:47.37 | h3x | $10 says its your firewall not your ethernet or connection speed |
| 10:47.40 | dudes | hmm, you could easily get 24 ULAW sip channels unless your provider sucks ass |
| 10:48.14 | mgoh | how about internal use? how many channel that I can support? |
| 10:48.34 | h3x | mgoh lets see, dudes has a box at my colo that does 600 calls |
| 10:48.37 | h3x | heh |
| 10:48.38 | dudes | that's a dumb question unless you're using a 10m/bit hub |
| 10:49.10 | h3x | that aint funny, i had that problem yesterday |
| 10:49.15 | dudes | I've gotten more than 500 ULAW calls going over a linksys 100m/bit switch internally NP |
| 10:49.17 | mgoh | dudes: why sometime I still fell delay for calling outside like fwd. |
| 10:49.18 | h3x | customer had a 10 meg hub |
| 10:49.36 | dudes | haha |
| 10:49.49 | dudes | I haven't used a 10m/bit hub since like 98 |
| 10:50.03 | h3x | dude your firewall sucks |
| 10:50.06 | mgoh | dudes: really my internal network is suck. I dunno why extension call asterisk echo test still not good. |
| 10:50.14 | h3x | voip is udp |
| 10:50.21 | h3x | those crappy routers dont do a good job with udp and nat |
| 10:50.25 | dudes | I got a *nix firewall (my linksys did though) |
| 10:50.36 | mgoh | h3x:100mb |
| 10:51.01 | h3x | something like a linksys wrt54g type router is probably ok |
| 10:51.12 | h3x | but most of these routers are underpowered to forward packets fast enough |
| 10:51.34 | mgoh | ic that mean I need to purchase a good switch. |
| 10:51.35 | heath__ | which is piss poor imo since that's they're job |
| 10:52.31 | mgoh | any heavy duty switch recommented? |
| 10:52.34 | h3x | unix firewall is the way to go |
| 10:52.34 | h3x | heh |
| 10:52.36 | dudes | just buy a cheap ass PII or Celery off of ebay and setup a *nix router /w decent qos rules |
| 10:53.21 | *** join/#asterisk JooZoo (n=chatzill@82-203-171-162.dsl.gohome.fi) |
| 10:53.44 | h3x | i still cant bring myself to buy amd stuff |
| 10:53.46 | mgoh | dudes, I not network or linux expert. |
| 10:53.52 | h3x | they dont have ddr2 ram support yet |
| 10:53.58 | h3x | ddr costs like 1.5x+ as much |
| 10:54.16 | dudes | AMD rocks! |
| 10:54.37 | NoRemorse | can anyone suggest any reason why accountcode= works for storing an accountcode in cdr's for a sip client user, but not for a sip asterisk peer? |
| 10:54.50 | *** join/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 10:54.58 | *** part/#asterisk _vic (n=riccardo@gw-fi.esaote.com) |
| 10:55.04 | mgoh | I heard that manage switch will help. do you guy know what brand is good to use? |
| 10:55.32 | h3x | i import some chinese shit for switches |
| 10:55.35 | h3x | they are cheap! |
| 10:55.36 | dudes | You could get a linksys switch and it'd work fine |
| 10:57.43 | dudes | Are you sure the Opteron's aren't doing DDR2? I thought they were |
| 10:57.54 | NoRemorse | can anyone suggest any reason why accountcode= works for storing an accountcode in cdr's for a sip client user, but not for a sip asterisk peer? |
| 11:00.56 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 11:02.16 | DarkFlibble | h3x, a wrt54g can do 30mbit between the lan and wan interfaces... its almost at the top of its class for home 'routers' tho |
| 11:02.34 | DarkFlibble | most max out at 10-20mbit |
| 11:03.39 | DarkFlibble | its a design tradeoff... since how many people have 20mbit broadband? (A: a lot of londoners now) |
| 11:04.10 | DarkFlibble | anyway... gotta run for this meeting... |
| 11:05.36 | DarkFlibble | for internal switching I use a gigabit switch and gigabit cards where possible, since the latency is generally 7-9 times lower.... |
| 11:05.45 | DarkFlibble | but thats just me... |
| 11:08.27 | DarkFlibble | http://forumz.tomshardware.com/network/Recommendation-router-100-MBps-internet-ftopict20292.html <-- wrt54g clocked at 27mbit... |
| 11:12.01 | *** part/#asterisk JooZoo (n=chatzill@82-203-171-162.dsl.gohome.fi) |
| 11:16.40 | *** join/#asterisk grexk (n=Server@210.213.177.217) |
| 11:18.17 | mgoh | all thanks gtg |
| 11:18.52 | grexk | hello all Im just new to asterisk. Can someone guide me with AGI. |
| 11:19.21 | grexk | why do I need to use php-cli in programming with php? |
| 11:19.35 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 11:20.11 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 11:23.17 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 11:24.31 | *** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 11:24.43 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 11:27.06 | *** join/#asterisk ermes (n=ermes@217.220.121.62) |
| 11:27.11 | ermes | ciao iddha |
| 11:28.03 | *** join/#asterisk saftsack (n=oliver@p54A7F3CB.dip.t-dialin.net) |
| 11:28.07 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 11:28.14 | saftsack | hi |
| 11:30.53 | grexk | hello |
| 11:31.37 | *** join/#asterisk fulgas (n=fulgas@209.8.233.241) |
| 11:31.46 | saftsack | grexk, do you know iaxmodem? |
| 11:33.32 | *** part/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:34.54 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 11:39.08 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 11:39.12 | NoRemorse | hi all |
| 11:39.32 | NoRemorse | is there any way to pass auth info in the DIAl command for SIP similar to IAX2? |
| 11:45.05 | grexk | sorry Im just new to asterisk |
| 11:45.45 | *** join/#asterisk bbrdrgz (n=alex@p54B03E6C.dip0.t-ipconnect.de) |
| 11:45.59 | *** part/#asterisk Bger (n=Bager@83.222.160.88) |
| 11:46.30 | NoRemorse | is there any way to pass auth info in the DIAl command for SIP similar to IAX2? |
| 11:49.58 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 11:52.02 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 11:54.04 | *** join/#asterisk nibbler_ (n=nibbler@194.150.168.18) |
| 11:57.18 | *** join/#asterisk ThaZZa_ (n=thazza@229.9.233.220.exetel.com.au) |
| 12:01.23 | grexk | GTg GOT to work with asTeriSK |
| 12:01.27 | *** part/#asterisk grexk (n=Server@210.213.177.217) |
| 12:05.39 | *** part/#asterisk rene- (i=rene@201.135.231.238) |
| 12:09.47 | *** join/#asterisk LordScinawa (i=MattiaDo@host173-174.pool8256.interbusiness.it) |
| 12:14.01 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 12:14.38 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 12:20.22 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:21.24 | *** join/#asterisk [bofh42] (n=bofh42@p54828CA6.dip0.t-ipconnect.de) |
| 12:21.41 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:23.23 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:25.53 | |vinsik| | does anybody have problems with phones connecting from behind NAT ? |
| 12:26.22 | Err | I'm sure lots of people do :-) |
| 12:26.33 | |vinsik| | how to resolve them? :D |
| 12:26.46 | |vinsik| | asterisk says -- Called phone1 |
| 12:27.00 | Err | port forwards and hackery - same as all other NAT traversals |
| 12:27.00 | |vinsik| | but phone is not ringing |
| 12:27.09 | |vinsik| | i used stun |
| 12:27.16 | |vinsik| | it makes it even worse |
| 12:27.22 | |vinsik| | qualify=yes |
| 12:27.41 | |vinsik| | then asterisk announces that phone is UNREACHABLE |
| 12:27.45 | |vinsik| | all ports are open |
| 12:28.01 | |vinsik| | i get unreachable 1-30 sek |
| 12:28.04 | |vinsik| | then it's online again |
| 12:29.04 | ermes | i can t use my tty when asterisk start at boot |
| 12:29.05 | |vinsik| | oh and happy birthday cresl1n :) |
| 12:29.42 | ermes | i can t use either other tty |
| 12:29.47 | |vinsik| | err: is there more to it than qualify, nat transversal and stun? |
| 12:29.55 | |vinsik| | err: maybe im missing something out. |
| 12:30.05 | |vinsik| | err: or is it suppose to work in such of manner? |
| 12:30.50 | Err | |vinsik|: I don't know - I don't do NAT traversal |
| 12:30.56 | Err | NATs are ugly |
| 12:31.22 | |vinsik| | ok. but what if the the same problem occures in localnetwork? |
| 12:31.32 | |vinsik| | is it because of some port settings? |
| 12:31.53 | Err | if the same problem occurs on non-NAT'd links, then you have some issues that need to be resolved |
| 12:32.12 | Err | do you have a firewall on the asterisk server, or on any of the clients, or in between the server and the clients? |
| 12:32.40 | |vinsik| | yes with configs udp 5060 and rtp 10000-12000, also configured in rtp.conf |
| 12:32.57 | |vinsik| | but it is only on the external eth |
| 12:33.03 | |vinsik| | internal has everything open |
| 12:33.08 | |vinsik| | and it sould work |
| 12:33.10 | Err | so you're *sure* that there is no firewall internally? |
| 12:33.18 | |vinsik| | yes |
| 12:33.39 | |vinsik| | well... yast seems to think so. and i checked iptables .. looks fine to me |
| 12:33.43 | Err | I'd debug it the same way I debug all network stuff - capture packets on both ends, and see what isn't getting to the other end |
| 12:34.02 | Err | (or see which end isn't responding to certain packets) |
| 12:34.07 | |vinsik| | well the server sends packets to the phone.. but it doesent reply |
| 12:34.17 | Err | "it" being the phone? |
| 12:34.21 | |vinsik| | yes |
| 12:34.26 | |vinsik| | zyxel |
| 12:34.28 | Err | sounds like your phone is misconfigured |
| 12:34.32 | |vinsik| | hmm.. |
| 12:34.40 | |vinsik| | can stun confuse it? |
| 12:34.44 | tzanger | |vinsik|: or not plugged in |
| 12:34.47 | tzanger | |vinsik|: of course it can |
| 12:34.52 | tzanger | simplify your setup |
| 12:34.53 | |vinsik| | tzanger: oh, its plugged |
| 12:34.53 | *** join/#asterisk saftsack (n=oliver@p54A7F3CB.dip.t-dialin.net) |
| 12:34.58 | tzanger | THEN start adding crap in :-) |
| 12:35.10 | *** join/#asterisk gvag11 (n=gvag11@ipa146.3.tellas.gr) |
| 12:35.10 | ermes | Err, how does asterisk start at boot ? |
| 12:35.15 | |vinsik| | tzanger: the point is that the phone has to work outside the localnetwork as well |
| 12:35.18 | ermes | there is nothing in init.d |
| 12:35.26 | tzanger | |vinsik|: well you have some work cut out for you then |
| 12:35.30 | gvag11 | hi all |
| 12:35.31 | |vinsik| | ermes: make your own init.d script |
| 12:35.42 | newl | or copy one of the ones provided.. |
| 12:35.43 | |vinsik| | tzanger: u got that right ;) |
| 12:35.52 | saftsack | some of you knows iaxmodem? |
| 12:36.09 | tzanger | |vinsik|: you need the ability for the phone to detect whether it's local or not, or to use something simpler like split-horizion dns to have the phone "see" different IPs for its proxy depending on where it is |
| 12:36.09 | |vinsik| | ermes: what distro? |
| 12:36.10 | gvag11 | does anybody knows if i can have a sort of error log from spandsp ? |
| 12:36.17 | ermes | centos |
| 12:36.17 | tzanger | saftsack: redder86 in #openpbx knows it |
| 12:36.22 | tzanger | not sure if he's up yet or not |
| 12:36.23 | ermes | rhel |
| 12:36.34 | saftsack | tzanger, thanks :) |
| 12:36.44 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 12:36.55 | saftsack | he isnt up yet but i can wait |
| 12:37.13 | |vinsik| | tzanger: ok. another question. |
| 12:37.29 | |vinsik| | tzanger: can nat=yes setting mess up the phone if its in localnetwork |
| 12:37.48 | tzanger | not sure -- I don't know the sip stack well enough to comment on that |
| 12:37.52 | *** join/#asterisk bizvn (n=nghiatha@222.252.48.198) |
| 12:38.01 | |vinsik| | anyone? can nat=yes setting mess up the phone if its in localnetwork |
| 12:38.07 | bizvn | hello all |
| 12:38.31 | ermes | Err, Found !! rc.local |
| 12:38.44 | bizvn | are there some sales men of digium here? I need some helps. |
| 12:38.59 | wasim | bizvn: you need an account number to send funds to? |
| 12:39.44 | |vinsik| | hehe |
| 12:39.52 | bizvn | ah, i don't know. I just played with asterisk for few months. And now i want to help asteirk to become popular in my country. |
| 12:40.12 | bizvn | so i really want to meet some sales man |
| 12:40.39 | Err | ermes: uh, you probably *don't* want to use /etc/rc.local |
| 12:40.52 | Err | use the sysv-style init script that comes with asterisk, and put it in /etc/init.d |
| 12:40.59 | gvag11 | does anybody knows if i can have a sort of error log from spandsp ? |
| 12:41.57 | |vinsik| | tzanger: ICMP xxx.dsl.maxinetti.fi udp port netinfo-local unreachable, |
| 12:42.00 | |vinsik| | uhh |
| 12:42.03 | *** join/#asterisk macanico (n=meca@host153.200-117-129.telecom.net.ar) |
| 12:42.34 | |vinsik| | ICMP xxx.dsl.maxinetti.fi udp port netinfo-local unreachable, <= anyone knows whats the deal with this? |
| 12:42.44 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 12:43.07 | macanico | |visi had the same problem |
| 12:43.25 | macanico | is forwarded that por on the other host? |
| 12:43.36 | |vinsik| | macanico: ? |
| 12:43.37 | bizvn | sorry, i will contact to sales of digium via email. Thks and have a good time. |
| 12:43.43 | ermes | err, why not rc.local ? |
| 12:43.59 | |vinsik| | macanico: its just a linux box that has ip_forwarding on |
| 12:44.29 | |vinsik| | CCEPT all -- anywhere anywhere |
| 12:44.31 | *** join/#asterisk pengyong (n=lala@218.93.159.101) |
| 12:44.44 | macanico | |vinsik|is behind a nat? |
| 12:44.49 | |vinsik| | in nat MASQUERADE all -- anywhere anywhere |
| 12:44.54 | |vinsik| | jes |
| 12:45.10 | |vinsik| | macanico: my phone i behind that linux box's nat |
| 12:45.17 | macanico | |vinsik| that mesasge you have when the por is closed |
| 12:45.21 | |vinsik| | macanico: how did you resolve it? |
| 12:45.27 | |vinsik| | ahh. |
| 12:45.38 | |vinsik| | so the server does not keep it alive |
| 12:45.47 | |vinsik| | or the phone? |
| 12:45.53 | macanico | |vinsik| how is? |
| 12:46.07 | macanico | adsl -- linux box -- phone? |
| 12:46.14 | macanico | somethin like this? |
| 12:46.35 | |vinsik| | oh, got it |
| 12:46.44 | |vinsik| | macanico: the server is not accepting on this port |
| 12:46.49 | macanico | yes |
| 12:46.51 | |vinsik| | macanico: no port forwarding |
| 12:46.53 | |vinsik| | :/ |
| 12:47.08 | |vinsik| | isnt nat suppose to transverse ports? |
| 12:47.15 | macanico | no |
| 12:47.32 | macanico | you have to forward to thath host |
| 12:48.12 | macanico | what kind of conection do you have? |
| 12:48.15 | macanico | adsl? |
| 12:49.19 | macanico | explain me hoy is you arch |
| 12:49.25 | |vinsik| | yes |
| 12:49.30 | macanico | and i will help you |
| 12:49.35 | Err | ermes: because rc.local removes the ability to stop/start/reload asterisk while the system is running, without doing it by hand |
| 12:49.38 | |vinsik| | i have a linux box (as router) .. |
| 12:49.50 | |vinsik| | it gets its ip from adsl modem |
| 12:49.51 | macanico | yea |
| 12:49.55 | macanico | ok |
| 12:49.57 | Err | ermes: if you use /etc/init.d, then you can run /etc/init.d/asterisk [stop|start|restart|reload] |
| 12:50.10 | |vinsik| | and has a dhcp server on another eth |
| 12:50.30 | macanico | ok |
| 12:50.41 | |vinsik| | now the asterisk server is another server in the world :) |
| 12:50.45 | macanico | is a sip phone? |
| 12:50.53 | |vinsik| | yes.. grandstream |
| 12:50.57 | macanico | ok |
| 12:51.13 | macanico | you hace to forward the por udp number |
| 12:51.19 | |vinsik| | so the ip phone gets ip from router box.. 192.168.1.148 for example |
| 12:51.23 | macanico | ok |
| 12:51.23 | |vinsik| | ok |
| 12:51.28 | macanico | you use iptables? |
| 12:51.29 | |vinsik| | how do i do that? |
| 12:51.32 | |vinsik| | yes |
| 12:51.34 | macanico | ok |
| 12:51.45 | macanico | i show you the rule |
| 12:51.45 | |vinsik| | something with postrouting .. |
| 12:51.47 | |vinsik| | ok |
| 12:51.48 | |vinsik| | than |
| 12:51.49 | |vinsik| | thanx |
| 12:52.01 | macanico | let me do it |
| 12:52.27 | |vinsik| | i know how to use iptables :) do you have the rule? |
| 12:52.59 | macanico | yep |
| 12:53.09 | macanico | con you do mi a fabor?' |
| 12:53.17 | tzanger | |vinsik|: ok, what's that mean to me? |
| 12:53.17 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 12:53.29 | macanico | the packets go to port upd 800? |
| 12:53.38 | macanico | 8000 |
| 12:54.00 | |vinsik| | 10000 -> |
| 12:54.04 | |vinsik| | rtp |
| 12:54.32 | |vinsik| | well its the same what port |
| 12:54.33 | |vinsik| | ;) |
| 12:54.43 | |vinsik| | ill configure the phones again if have to. |
| 12:55.03 | macanico | ok |
| 12:55.09 | Err | RTP rides over UDP |
| 12:55.20 | macanico | you have to do double nat |
| 12:55.24 | |vinsik| | err: yes, i know.. |
| 12:55.24 | Err | (technically it can go over TCP, but it'd be dumb to do that) |
| 12:55.29 | macanico | wait a sec |
| 12:55.38 | |vinsik| | macanico: double nat? |
| 12:55.56 | Err | adding another NAT is *never* the Right Solution |
| 12:56.14 | |vinsik| | yeah i dont think double nat is good |
| 12:56.16 | macanico | wait |
| 12:56.21 | |vinsik| | one nat is already too much :D |
| 12:56.27 | macanico | jaja |
| 12:56.46 | NewSole | to NAT or not ot NAT |
| 12:57.30 | |vinsik| | iptables -t nat -A prerouting_rule -i $WAN -p udp --dport 10000:20000 -j ACCEPT |
| 12:57.30 | |vinsik| | iptables -A input_rule -i $WAN -p udp --dport 10000:20000 -j ACCEPT |
| 12:57.34 | |vinsik| | is it something like this? |
| 12:57.47 | macanico | no |
| 12:58.07 | macanico | you have to change the dest address |
| 12:58.20 | |vinsik| | oh.. so i have to force it to the phones ip? |
| 12:58.24 | macanico | wahts your phone ip? |
| 12:58.29 | macanico | yea |
| 12:58.43 | macanico | iptables -A PREROUTING -t nat -p tcp -d 192.168.2.1 --dport 666 -j DNAT |
| 12:58.43 | macanico | --to 192.168.2.254:80 |
| 12:58.43 | macanico | iptables -t nat -A POSTROUTING -p tcp -d 192.168.2.254 --dport 80 -j SNAT |
| 12:58.43 | macanico | --to 192.168.2.1 |
| 12:58.43 | macanico | iptables -A PREROUTING -t nat -p tcp -s 192.168.2.254 --sport 80 -j DNAT |
| 12:58.44 | macanico | --to 200.81.15.68 |
| 12:58.44 | |vinsik| | iptables -t nat -A prerouting_rule -i $WAN -j DNAT --to 192.168.1.2 |
| 12:58.44 | |vinsik| | iptables -A forwarding_rule -i $WAN -d 192.168.1.2 -j ACCEPT |
| 12:58.45 | macanico | iptables -t nat -A POSTROUTING -p tcp -s 192.168.2.254 --sport 80 -j SNAT |
| 12:58.55 | tzanger | |vinsik|: no I don't think that will work... you're simply accepting the packets into the prerouting |
| 12:59.02 | tzanger | NAT and SIP aren't really great friends |
| 12:59.16 | macanico | --to 192.168.2.1:666 |
| 12:59.22 | |vinsik| | tzanger: i noticed. :( |
| 12:59.24 | macanico | upts |
| 12:59.24 | macanico | sorry |
| 12:59.29 | macanico | iptables -A PREROUTING -t nat -p udp --dport 8000 -j DNAT |
| 12:59.39 | macanico | --to 192.168.2.254 |
| 12:59.44 | tzanger | macanico: it's not that simple. You should have SER on the natting firewall if at all possible |
| 13:00.08 | SERGEUS|W | i'm trying to catch call from voxbone, i have a packet from it, but my asterisk doesn't make any actions on it, can anybody suggest me something? probably there are some common mistakes? |
| 13:00.32 | |vinsik| | macanico: thanx.. ill try this. |
| 13:00.39 | |vinsik| | tzanger: ser sounds better |
| 13:00.50 | macanico | tzayou have to forward the port 5060 |
| 13:00.51 | macanico | too |
| 13:01.17 | macanico | tzanger why if you use a direct call |
| 13:02.07 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 13:02.15 | ckruetze | Hi |
| 13:02.16 | tzanger | macanico: I'm not going to argue the finer points of SIP and NAT. If you can get it to work, great. If not, don't argue about it. |
| 13:02.49 | macanico | ii have it working fine |
| 13:02.54 | macanico | just an asterisk doin sip proxy |
| 13:02.59 | macanico | on softphones on the lan |
| 13:03.16 | macanico | and a call from to any internet |
| 13:04.02 | |vinsik| | macanico: the problem is that i take my phone everywhere with me. and they dont have a nice prerouting done for me :D |
| 13:04.26 | Err | oh, you'll *never* get your phone set up to work behind any random NAT |
| 13:04.34 | macanico | thats will be a problem :P |
| 13:04.40 | |vinsik| | :P |
| 13:04.41 | wasim | unless you have an IAX phone |
| 13:04.43 | Err | SIP doesn't work that way, which is why it's a dumb protocol |
| 13:05.18 | |vinsik| | :( |
| 13:05.25 | Err | (well, really NATs are the dumb part, but protocols that open other ports are bad in general anyway - firewalls bust them too) |
| 13:05.42 | macanico | eri only forwarded udp port range 10000-20000 to my asterisk |
| 13:05.48 | macanico | and port 5060 |
| 13:05.53 | |vinsik| | macanico: same here |
| 13:05.57 | |vinsik| | macanico: on the server |
| 13:06.18 | |vinsik| | macanico: but the phone side is the problem |
| 13:06.19 | Err | that's a *huge* swath of ports |
| 13:06.21 | macanico | and i get it working |
| 13:06.26 | macanico | yea |
| 13:06.31 | Err | (to reserve just in case SIP needs them) |
| 13:06.38 | macanico | but it works |
| 13:06.54 | |vinsik| | well what do u think is good for 300 clients? |
| 13:06.58 | macanico | source udp ports always are in that range for the calls |
| 13:07.01 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:07.05 | |vinsik| | if asterisk will even work with such an ammount |
| 13:07.12 | |vinsik| | 240 was it maximum? |
| 13:07.21 | macanico | i made som packet snifing and choose to forward that range |
| 13:07.48 | |vinsik| | macanico: rtp.conf can make it smaller.. |
| 13:07.51 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 13:07.57 | *** part/#asterisk synthetiq (n=roger@64.201.13.50) |
| 13:08.12 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 13:08.22 | |vinsik| | but its nice to know that it has nothing to do with my asterisk settings.. |
| 13:08.31 | |vinsik| | "#%%#"¤ ghm NAT and SIP |
| 13:08.38 | macanico | but the client is who chose the source port |
| 13:08.51 | |vinsik| | macanico: it asks the range from server imo |
| 13:08.58 | |vinsik| | macanico: when connecting to 5060 |
| 13:09.01 | [TK]D-Fender | I never had problems with NAT & SIP.... just need to know what to set for it... |
| 13:09.09 | Err | each endpoint chooses its port number - it *has* to |
| 13:09.17 | macanico | i can show youu some packets |
| 13:09.27 | Err | it's not possible for one endpoint to choose both port numbers - the remote port number might be in use |
| 13:10.32 | macanico | yes |
| 13:10.36 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:11.11 | macanico | 10:09:59.485925 IP 200.81.16.246.5060 > 192.168.2.1.5060: SIP, length: 384 |
| 13:11.11 | macanico | 10:09:59.558430 IP 200.81.16.246.5060 > 192.168.2.1.5060: SIP, length: 718 |
| 13:11.22 | macanico | 0:09:58.789688 IP 200.81.16.246.8002 > 192.168.2.1.15622: UDP, length 45 |
| 13:11.22 | macanico | 10:09:58.790117 IP 192.168.2.1.18708 > 200.81.16.246.8000: UDP, length 45 |
| 13:11.22 | macanico | 10:09:58.790227 IP 200.81.16.246.8002 > 192.168.2.1.15622: UDP, length 45 |
| 13:11.22 | macanico | 10:09:58.790461 IP 192.168.2.1.18708 > 200.81.16.246.8000: UDP, length 45 |
| 13:11.54 | kll | how do I increase the timeout asterisk waits when I'm using overlapdial? |
| 13:11.59 | macanico | always the source ports is etween 10000-20000 |
| 13:12.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:13.00 | |vinsik| | why then rtp.conf ? |
| 13:13.32 | macanico | well i have to tell you |
| 13:13.34 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:13.35 | macanico | that im just a newbie |
| 13:13.43 | |vinsik| | me 2 |
| 13:13.56 | macanico | i ahve been playing whit asterisk for a few days |
| 13:14.18 | macanico | but i get it working behind a nat |
| 13:14.27 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:16.10 | Ahrimanes | ~pastebin |
| 13:16.12 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:16.17 | *** join/#asterisk orion (n=orion@72.16.146.78) |
| 13:16.19 | macanico | |vinsik| you ahve and asterisk box and you whant registher your phone whit sip and make calls in or out? |
| 13:16.26 | *** join/#asterisk anti\ (n=a@200.69.233.113) |
| 13:16.36 | macanico | hi anti\ |
| 13:16.36 | macanico | como andas? |
| 13:16.49 | anti\ | isa |
| 13:16.56 | anti\ | hello world |
| 13:17.31 | Ahrimanes | http://pastebin.com/520517 <- why does this result in the default moh class being used when putting people on hold, after answering the queue? |
| 13:18.06 | *** join/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl) |
| 13:18.22 | |vinsik| | macanico: well it works for 5-30 minutes for me. ;) |
| 13:18.37 | |vinsik| | macanico: then the phones are unavailible.. |
| 13:18.50 | |vinsik| | macanico: and from time to time they are availible again |
| 13:18.50 | macanico | after that its hang up? |
| 13:19.09 | macanico | all this after do the forward? |
| 13:19.29 | |vinsik| | macanico: yesterday |
| 13:19.40 | |vinsik| | macanico: ill test your setting today |
| 13:20.23 | macanico | good |
| 13:20.39 | macanico | some snifing whill help a lot |
| 13:21.45 | anti\ | se |
| 13:21.48 | anti\ | eso quiero |
| 13:21.48 | anti\ | jaja |
| 13:21.55 | anti\ | escuchas llamadas |
| 13:22.08 | [TK]D-Fender | macanico : If you're behind NAT you need to set either EXTERNIP or EXTERNHOST, and LOCALNET in sip.conf or you won't get anywhere. |
| 13:23.40 | |vinsik| | d-fender: reaaally? |
| 13:23.50 | [TK]D-Fender | yes |
| 13:24.08 | |vinsik| | d-fender: i have a server thats for example 81.114.114.114 |
| 13:24.16 | [TK]D-Fender | unless you completely forge the daylights out of external packet with some sort of proxy or something |
| 13:24.22 | |vinsik| | d-fender: and phone is behind nat with some crappy ip |
| 13:24.30 | |vinsik| | d-fender: do i need the settings for the mainserver? |
| 13:24.37 | |vinsik| | d-fender: for the 81.114.x.x |
| 13:24.42 | macanico | asterisk as proxy maybe? |
| 13:24.50 | [TK]D-Fender | Depends if the SERVER is behind NAT or if the CLIENT is |
| 13:24.58 | |vinsik| | the client is |
| 13:25.04 | |vinsik| | server is NOT |
| 13:25.06 | anti\ | um |
| 13:25.08 | anti\ | sorry |
| 13:25.17 | anti\ | can sniff de audio call's ? |
| 13:25.31 | macanico | softophone -- nat-- asterisk -- lan --softphones |
| 13:25.36 | [TK]D-Fender | For the server behind NAT use the settings I mentioned,. For clients you should set "nat=yes" and "quality=yes" for that entry |
| 13:25.47 | [TK]D-Fender | you may need to to tell the device that its behind NAT as well |
| 13:26.38 | [TK]D-Fender | type : "QUALIFY=yes" |
| 13:26.41 | [TK]D-Fender | typo* |
| 13:27.19 | macanico | the sip.conf for the lan softphones is something like this |
| 13:27.49 | macanico | and i specify the external ip of the asterisk whit a dynamic dns |
| 13:28.39 | [TK]D-Fender | macanico : Also make sure all of your internal devices are set to "canreinvite=no" |
| 13:28.40 | Err | anti\: of course you can, if you have access to the network that's transporting them |
| 13:28.51 | *** join/#asterisk hickins (n=dtg19@213.186.161.29) |
| 13:29.16 | anti\ | access to asterik server ? |
| 13:29.20 | hickins | greetings all |
| 13:29.23 | macanico | no |
| 13:29.30 | macanico | to the trafic :P |
| 13:29.33 | macanico | jaja |
| 13:29.36 | anti\ | se we |
| 13:29.47 | anti\ | pero conque sniffias |
| 13:29.48 | anti\ | el audio |
| 13:29.49 | *** join/#asterisk Bambr (n=Bambr@213-35-236-195-dsl.end.estpak.ee) |
| 13:29.49 | anti\ | ? |
| 13:29.50 | Err | I'm sure you can record calls from the server as well |
| 13:29.58 | anti\ | aa! |
| 13:30.01 | anti\ | okok |
| 13:30.02 | macanico | aps |
| 13:30.03 | macanico | debe haber algun progie que te ensambla todo |
| 13:30.08 | anti\ | ya ta |
| 13:30.09 | anti\ | jejje |
| 13:30.17 | hickins | any can help me detect hang-ups withing perl AGI script? does setmycallback help? |
| 13:30.17 | anti\ | thank Err! |
| 13:31.30 | hickins | i need to do something when the caller hangs up |
| 13:32.04 | hickins | can anyone help please? |
| 13:33.26 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 13:35.07 | *** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
| 13:35.41 | iCEBrkr | yo yo yo |
| 13:36.46 | macanico | hickins maybe som var? |
| 13:36.48 | macanico | http://www.voip-info.org/wiki/view/DIALSTATUS |
| 13:36.54 | hickins | ICEBrkr, do you think there is ice in here to break? |
| 13:37.10 | hickins | thanks macanico |
| 13:37.57 | macanico | hickins i dont know |
| 13:37.58 | kll | how do I increase the timeout asterisk waits when I'm using overlapdial? |
| 13:37.59 | macanico | just maybe an idea |
| 13:38.26 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 13:38.36 | Lathos42 | Good Morning |
| 13:39.00 | hickins | thants fine macanico, thanks anyway I will look into it and maybe give you some feedback |
| 13:39.14 | anti\ | macanico como la flasheas con ingles |
| 13:39.16 | anti\ | :p |
| 13:39.17 | iCEBrkr | Inserting a space after commas separating the parameters will result in unexpected results. |
| 13:39.21 | iCEBrkr | e.g. |
| 13:39.23 | iCEBrkr | <PROTECTED> |
| 13:39.26 | iCEBrkr | will look for an extension " 1", i.e. with a preceding space character. |
| 13:39.27 | iCEBrkr | Someone hasn't heard of 'trim' |
| 13:39.34 | iCEBrkr | hickins: oh, u so funneeeee |
| 13:40.04 | Err | iCEBrkr: so don't do that :-) |
| 13:40.21 | iCEBrkr | Err: >>I<< Understand that, but a lot of other people don't. |
| 13:40.38 | hickins | thanks icebrkr, I only tried to cheer up little bit |
| 13:40.50 | anti\ | macanico! |
| 13:40.51 | iCEBrkr | Err: Besides, when you write something that the masses are going to be using, you should take something like white space and the like into consideration |
| 13:40.51 | anti\ | macanico! |
| 13:41.04 | macanico | si? |
| 13:41.05 | brockj49464 | With the SPA-2100 connected to * what do I need to look at to fix echo problems on the local LAN? |
| 13:41.13 | anti\ | en el gran hermano |
| 13:41.14 | Err | only if it isn't documented that the configuration language cannot handle whitespace |
| 13:41.15 | anti\ | esta 2 tipos |
| 13:41.18 | anti\ | con una notebook |
| 13:41.20 | anti\ | hablando |
| 13:41.21 | anti\ | ajajaja |
| 13:41.23 | iCEBrkr | brockj49464: I'd look into your tx/rxgain settings |
| 13:41.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:41.32 | brockj49464 | on the spa? |
| 13:41.37 | macanico | jajaa |
| 13:41.41 | anti\ | saco screen |
| 13:41.42 | anti\ | para |
| 13:41.42 | iCEBrkr | brockj49464: I believe it's in the zapata.conf file |
| 13:41.52 | macanico | a ver |
| 13:41.55 | macanico | da para conectarse anti? |
| 13:41.55 | macanico | dale |
| 13:41.56 | iCEBrkr | brockj49464: actually.. Rewind a second |
| 13:42.00 | anti\ | se |
| 13:42.01 | anti\ | dale |
| 13:42.02 | anti\ | coencta |
| 13:42.18 | anti\ | concha |
| 13:42.22 | anti\ | no se entrecorta |
| 13:42.24 | anti\ | el audio |
| 13:42.25 | iCEBrkr | Err: I dunno man, having a function white-space sensitive is pretty dumb and actually quite uncommon |
| 13:42.33 | iCEBrkr | brockj49464: Describe your problem a bit more. |
| 13:43.15 | Err | iCEBrkr: I wouldn't design a language to be white-space dependent, but plenty are |
| 13:43.25 | Err | as long as a syntax is documented, it doesn't matter *what* it does |
| 13:43.32 | iCEBrkr | Err: *cough* Pyton *cough* :) |
| 13:43.38 | iCEBrkr | err Python |
| 13:43.49 | iCEBrkr | Err: I guess so, but lets use our heads here.. Do the right thing.. |
| 13:44.59 | iCEBrkr | Most geeks/programmers are used to using function(param1, param2, ...) Someone needs to keep with the 'standards' I'm pretty sure other asterisk apps/functions don't give a rats ass about spaces in between it's parameters |
| 13:45.02 | brockj49464 | Ok I have the two lines on the SPA-2100 configured to two different ext on * which are all hooked up via 10baseT switched. When calling from SPA-2100 line 1 to SPA-2100 line 2 there is echo (also happens when I connect to external Zap/Exteral SIP) so I am thininking it is either on the client side or the * box itself. |
| 13:45.12 | Err | like I said, I wouldn't have designed it that way - but if it's defined, it doesn't really matter |
| 13:45.29 | Err | people who can't learn a language because whitespace is important probably aren't going to be able to learn it if whitespace *weren't* important |
| 13:45.38 | brockj49464 | brb |
| 13:45.45 | iCEBrkr | brockj49464: Odd, You typically get echo when you traverse PSTN/POTS lines from VoIP.. |
| 13:45.57 | iCEBrkr | Err: haha |
| 13:46.14 | iCEBrkr | Err: Now, granted, it was nice of the person to document that issue :) |
| 13:47.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:48.08 | iCEBrkr | Hrrrm, how the heck can you figure out how many lines/channels are in use? There doesn't seem to be a way to do this yet. |
| 13:48.12 | hickins | macanico, for the records, someone just told me about DeadAGI()! it might work I'll try it |
| 13:48.35 | iCEBrkr | hickins: What's not working? |
| 13:48.44 | *** join/#asterisk Abbas (n=Abbas@203.81.202.185) |
| 13:51.57 | *** join/#asterisk tRSS (n=tRSS@202.174.142.2) |
| 13:54.15 | *** join/#asterisk caedes (n=apardo@62.97.121.95) |
| 13:54.51 | brockj49464 | So where should I look for a problem. Is there a way to put the SPA-2100 in a loopback and check that out? I have reset it and restarted the * box just incase that was the problem. Could I turn the gain on the SPA down? |
| 13:55.03 | Err | iCEBrkr: "show channels" doesn't give what you want? |
| 13:55.05 | macanico | hickins ok it will be nice if it wors :P |
| 13:55.07 | iCEBrkr | brockj49464: you could. |
| 13:55.33 | iCEBrkr | Err: I need a count of available channels. Even better functionality would be a count of both used and unused channels. |
| 13:56.06 | iCEBrkr | brockj49464: You're getting echo between the two extensions on the same SPA?? |
| 13:56.37 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:57.05 | Err | so can't you use show channels and some post-processing? |
| 13:57.24 | brockj49464 | Yea, but I believe the path is to the * and back. |
| 13:58.02 | iCEBrkr | Err: The count isn't accurate? No, that's not the word |
| 13:58.07 | iCEBrkr | 2 active channels |
| 13:58.07 | iCEBrkr | 1 active call |
| 13:58.19 | iCEBrkr | Err: I made a single call. I have 72 channels available. |
| 13:58.28 | iCEBrkr | So I guess 'active call' is what I'm looking for |
| 13:58.51 | iCEBrkr | brockj49464: Yea, it most likely is.. But you shouldn't be getting echo like that. That's kinda strange |
| 13:58.57 | Err | so the "channels" it's referring to are individual data streams (one-way), in this case - so the call count is more likely useful |
| 13:59.09 | iCEBrkr | Right |
| 13:59.12 | iCEBrkr | Hrrm. |
| 13:59.23 | iCEBrkr | I might be able to hackup that part of the code to just return what I'm looking for |
| 13:59.48 | *** part/#asterisk anti\ (n=a@200.69.233.113) |
| 14:01.19 | iCEBrkr | Hrr, I wonder if I could rip this part of the code out and make a function out of it? |
| 14:01.51 | iCEBrkr | Unfortunately my C is hella-rusty |
| 14:02.32 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 14:02.38 | Err | I'd be more likely to write a script to parse the current output into what you want :-) |
| 14:02.49 | *** join/#asterisk morrece (n=moreece@196.46.142.23) |
| 14:03.02 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 14:03.03 | morrece | good afternoon all |
| 14:03.04 | iCEBrkr | Err: I'm thinking that's gonna be the route I take.... Unfortunately. |
| 14:03.20 | morrece | I have a question regarding my SIP softphone and my ASTERISK PABX |
| 14:03.45 | iCEBrkr | Err: I could just write a script to connect to the manage port and issue Command: show channels and parse from there. |
| 14:04.01 | Err | yeah, that's what I mean |
| 14:04.30 | Err | just asterisk -rx "show channels" | whatever |
| 14:05.21 | saftsack | is sourceforge.net down? |
| 14:05.32 | *** join/#asterisk leto3 (n=l@car75-1-81-57-13-34.fbx.proxad.net) |
| 14:05.41 | morrece | within my sip.conf I have added 3 extensions within my extension.conf file ... however when attempting to dial an extension I get a timeout |
| 14:06.04 | morrece | I see on my asterisk box port 5060 wich is suppose to be my bindaddr port for SIP is not running |
| 14:06.06 | morrece | why? |
| 14:06.22 | *** join/#asterisk zamsler_ (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 14:06.31 | [TK]D-Fender | morrece : Pastebin your sip.conf & extensions.conf first |
| 14:06.32 | [TK]D-Fender | ~pb |
| 14:06.33 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:06.34 | iCEBrkr | morrece: You restart or do a sip reload? |
| 14:06.53 | morrece | EXTENSION.CONF |
| 14:06.53 | morrece | [internal] |
| 14:06.53 | morrece | exten => 100,1,Dial(SIP/malcolm) |
| 14:06.53 | morrece | exten => 200,1,Dial(SIP/clinton) |
| 14:06.53 | morrece | exten => 999,1,Echo() |
| 14:06.54 | *** join/#asterisk littleball (n=littleba@cm169.epsilon169.maxonline.com.sg) |
| 14:06.57 | iCEBrkr | Nooooooooooooooo |
| 14:07.12 | morrece | iesh sorry |
| 14:07.22 | littleball | hello, i am trying to add PRI routing in my system. what is the best practice? anyone can give me some indication about this? |
| 14:07.30 | morrece | yeah I did a sip reload |
| 14:07.34 | morrece | still to no avail |
| 14:07.48 | Err | why wouldn't you name your SIP devices? |
| 14:07.50 | [TK]D-Fender | PASTEBIN! |
| 14:07.56 | iCEBrkr | Err: Cuz they're extensions... |
| 14:08.00 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 14:08.01 | *** join/#asterisk foucaulo (n=fouca@modemcable186.159-82-70.mc.videotron.ca) |
| 14:08.05 | macanico | iCEBrkr to identify more nicely? |
| 14:08.11 | Err | it's self-documenting |
| 14:08.12 | iCEBrkr | Err: And if people shuffle desks, you don't have to move their phones. |
| 14:08.26 | [TK]D-Fender | morrece : Please pastebin your entire sip.conf & extensions.conf |
| 14:08.31 | iCEBrkr | Or even if someone leaves the company and someone gets that phone.. No renaming. |
| 14:08.32 | Err | if you *do* move their phones, though, they keep their old extension - which is a Good Thing |
| 14:08.36 | morrece | ok my is simply a test setup to get it funcition on the LAN at the office first |
| 14:08.50 | Err | egrep is your friend, for renaming extensions :-P |
| 14:08.55 | iCEBrkr | lol |
| 14:09.14 | morrece | <going to pastebin.com> |
| 14:09.20 | iCEBrkr | Err: Us nerds can do that, but I'm not trying to babysit this thing... |
| 14:09.25 | fourcheeze | anyone using snom360s with * 1.2 and Destination keys? |
| 14:09.36 | fourcheeze | their operation seems to be different in 1.2 |
| 14:09.38 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 14:09.48 | fourcheeze | if a destination is not registered then the light comes on |
| 14:09.57 | fourcheeze | is it possible to change that behaviour? |
| 14:10.05 | macanico | iCEBrkr maybe it depends on the magnitudo of th pbx |
| 14:10.09 | iCEBrkr | Err: See that's the problem with us computer dorks. We can do just about anything with all the tools we have at our finger tips. Other people aren't so smart/lucky. So you really need to provide a easy solution. |
| 14:10.17 | Err | iCEBrkr: clueless people probably shouldn't administer phone switches |
| 14:10.25 | iCEBrkr | Err: Wrong attitude to have man.. |
| 14:10.29 | Err | no, it's not |
| 14:10.33 | iCEBrkr | Err: Sure it is. |
| 14:10.42 | iCEBrkr | You're thinking like a ubersmart linux person. |
| 14:10.43 | morrece | http://pastebin.com/520566 |
| 14:10.56 | iCEBrkr | Err: I used to think that way too. |
| 14:10.58 | Err | no, I'm thinking like someone who understands that you can't bumble into having a properly-configured setup |
| 14:11.18 | Err | ...this goes for IP networks, telephone systems, and just about everything else |
| 14:11.26 | foucaulo | Can somebody help me?(not able to start asterisk with spandsp installed) |
| 14:11.34 | Err | you wouldn't say that building a vehicle should be cookie-cutter - why is a telephone system any different? |
| 14:11.45 | Err | foucaulo: are you running debian? |
| 14:11.49 | iCEBrkr | Err: As it sits right now.. Asterisk would be hard to put into a company without any sort of user interface cuz of all the text files. People don't want to learn how to edit text files. PErsonally, >>I<< don't have a problem with it as it's 'normal' |
| 14:11.56 | *** join/#asterisk razu (n=razu@193.40.101.34) |
| 14:11.58 | [TK]D-Fender | macanico : Have you reset your SIP endpoints? |
| 14:12.10 | iCEBrkr | Err: Yea, but when I turn on my car, I put the key in and I turn it.. I don't have to know anything else. |
| 14:12.12 | Err | iCEBrkr: that's why there are consultants - most companies don't manage their PBXs in-house, either |
| 14:12.23 | iCEBrkr | Err: Who wants to work that hard?? |
| 14:12.33 | macanico | [TK]D-Fender what do you mean? why? |
| 14:12.36 | iCEBrkr | Err: I'm not making a service call to add/delete someones extension. |
| 14:12.47 | Err | iCEBrkr: that's common for traditional PBXs, as well |
| 14:12.55 | morrece | so what do u guys think about my problem |
| 14:12.58 | morrece | http://pastebin.com/520566 |
| 14:13.02 | foucaulo | Err: I'm using Suse 9.0 |
| 14:13.06 | [TK]D-Fender | macanico : just wondering if there is a setting they didn't pick up.. do a "sip show peers" in * CLI and see if it lists them as registered. |
| 14:13.14 | iCEBrkr | Err: I dunno about that. I've always been able to login to the PBX and add/edit extensions through an interface. It's super easy. |
| 14:13.30 | macanico | yep |
| 14:13.31 | iCEBrkr | Err: Changing a huntgroup or call routing, sure.. service call. |
| 14:14.16 | kaldemar | morrece: have you tried putting an actual ip as the bind address? |
| 14:14.45 | morrece | yes I have tried using my actual network ip of 10.0.18.200 |
| 14:14.58 | macanico | macanico/macanico (Unspecified) D 0 Unmonitored |
| 14:15.00 | littleball | hello, who has experience of routing different calls to different E1 lines? I need advice on PRI routing. Because different E1 lines are connected to different providers. I want to optimize the voice quality |
| 14:15.02 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 14:15.04 | morrece | surely my port 5060 should be open on my asterisk box right?? |
| 14:15.23 | macanico | [TK]D-Fender and if i start the softphone i get |
| 14:15.48 | macanico | morrece yep |
| 14:15.48 | [TK]D-Fender | macanico : that line looks WRONG for "sip show peers". to a "restart gracefully" |
| 14:15.53 | Err | foucaulo: the problem is that the spandsp source is calling gethostbyname, instead of ast_gethostbyname - you'll have to patch it and recompile |
| 14:16.16 | morrece | going to get my details quickly again a past them in pastebin -- 1 sec |
| 14:16.20 | macanico | macanico/macanico 200.81.16.246 D 5060 Unmonitored |
| 14:16.31 | [TK]D-Fender | macanico : Sorry, my bad.. wrong person |
| 14:16.36 | macanico | alway get that kind of output |
| 14:17.03 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 14:17.04 | [TK]D-Fender | morrece : Pastebin your "sip show peers" |
| 14:17.04 | macanico | [TK]D-Fender i was confused so i go on :P |
| 14:18.22 | [TK]D-Fender | macanico : You're still trying to work out NAT issues right? |
| 14:18.54 | *** join/#asterisk Tili (i=Tili@203.101.169.186) |
| 14:19.08 | *** join/#asterisk krustyclown (n=Dewi_sla@202.153.246.57) |
| 14:19.30 | krustyclown | hi all |
| 14:19.33 | morrece | ok well check this out while I get my sip show peers |
| 14:19.48 | morrece | http://pastebin.com/520574 |
| 14:21.13 | morrece | http://pastebin.com/520576 has all the details at the bottom includes my sip show peers |
| 14:21.35 | macanico | [TK]D-Fender no i get it to work |
| 14:22.30 | *** join/#asterisk lahaine (n=lahaine@71.68.119-80.rev.gaoland.net) |
| 14:23.48 | morrece | there is a problem with one of the IAX modules I think but that shouldnt effect me right, seeing as I am using SI? |
| 14:24.44 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 14:25.00 | johnnyb | What is the impact of specifying fxs_ks on a loopstart line? |
| 14:25.07 | morrece | so what do u guys think???? I'm a totally useless or what? |
| 14:26.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:26.09 | *** join/#asterisk apardo_ (n=apardo@62.97.121.95) |
| 14:27.03 | foucaulo | Err: ok but when you talking about spandsp source, can you just be more specific? I searched in but didn't find anything (|grep gethostbyname)... |
| 14:27.05 | [TK]D-Fender | morrece : They aren't registered. do "set verbose 10" and restart the SIP devices |
| 14:27.34 | morrece | set verbose 10 in sip.conf? |
| 14:28.20 | morrece | set verbose=10 or setverbose=10? |
| 14:29.31 | macanico | cli? |
| 14:30.17 | morrece | sorry |
| 14:30.18 | morrece | my bad |
| 14:30.21 | morrece | done |
| 14:30.48 | robin_z | quick question .. how do I change the music on hold from sound of sea/water/toilet to somethng nicer? |
| 14:31.05 | morrece | hmmmm still no port 5060 on my asterisk box though |
| 14:31.27 | morrece | thank u kindly for the help, I must return later I have a meeting to attend to thanks all again. C u soon |
| 14:31.56 | littleball | hello, how to do prefix matching in dialplan? |
| 14:32.20 | macanico | robin_z the moh |
| 14:32.41 | Err | foucaulo: I don't know, to be honest - I'm having the same problem on an ubuntu box, but I just found the solution this morning and haven't looked into it... |
| 14:34.07 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 14:34.26 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:35.15 | johnnyb | I'm trying to get asterisk running w/ multiple TDM cards, and I'm getting an error like this: |
| 14:35.27 | johnnyb | wait_for_sysfs[1795]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap5' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to <linux-hotplug-devel@lists.sourceforge.net> |
| 14:35.58 | johnnyb | I imagine that's because it doesn't expect to be getting the extra card, but how do I fix it? |
| 14:36.10 | iCEBrkr | johnnyb: Doesn't look like it has anything to do with the card. |
| 14:36.17 | iCEBrkr | johnnyb: I suspect it's a udev issue |
| 14:36.36 | MattB2 | i ahve a PRI question - our PRI interface expects to be told whether the number i'm sending to it is local or long-distance. any ideas how i go about doing that? |
| 14:36.53 | foucaulo | Err: ok, not a problem, I'm still searching, anybody had a solution? |
| 14:37.05 | iCEBrkr | MattB2: What'chu talk'n about? |
| 14:37.43 | robin_z | quick question .. how do I change the music on hold from sound of sea/water/toilet to somethng nicer? |
| 14:39.21 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:39.36 | MattB2 | iCEBrkr: area code where i am is 423. an example "local" number has areacode 706 |
| 14:39.55 | iCEBrkr | MattB2: I'm still kinda confused as to what you're trying to d. |
| 14:39.56 | *** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net) |
| 14:40.14 | MattB2 | if i just send a local number to the pri (ie 7 digits) without the 706 code, i hear ringing forever, but it never rings the other end |
| 14:40.16 | iCEBrkr | MattB2: It sounds like you're looking for some dialplan logic depending on the area you're trying to call. |
| 14:40.25 | *** join/#asterisk Defraz (i=t0tal@72.165.56.43) |
| 14:40.37 | iCEBrkr | MattB2: You can do something like that with some pattern matching. |
| 14:40.38 | Mag1KaL | Why isn't my 's' extension working... it's supposed to be called when no other exten matches the called extension right? |
| 14:40.48 | iCEBrkr | Mag1KaL: Nope |
| 14:40.52 | warthawg | i have 3 ip phones connected as extensions 101, 102, and 103. 101 works fine, it's a bt-101. 102 and 103 are zultys phones, the zip2x2 and the wip2. i can call out on both, but when i try to call in to them, asterisk tells me they are busy. any idea whether my problem is in my asterisk config or in the phone configs? |
| 14:41.21 | MattB2 | iCEBrkr: the thing is, a 7 digit number may need 706 or 423 area code. on our old analog lines you didn't need to enter the area code |
| 14:41.23 | iCEBrkr | warthawg: if the BT101 works, then it's probably your phone config, not asterisk :P |
| 14:41.26 | Mag1KaL | So how is it used then, |
| 14:41.27 | MattB2 | but our pri won't accept just a 7 digit number |
| 14:41.43 | warthawg | iCEBrkr, thanks, just thought there might be some secret thing i dunno about |
| 14:41.49 | iCEBrkr | Mag1KaL: 's' is the 'start' of your context. |
| 14:42.03 | Err | foucaulo: it's apparently in dtmftotext.c |
| 14:42.09 | iCEBrkr | MattB2: You're gonna have to get clever on your dialplan or enforce 10 digit dial. |
| 14:42.17 | De_Mon | iCEBrkr i thought s was '1' |
| 14:42.23 | iCEBrkr | De_Mon: Nope |
| 14:42.26 | robin_z | sigh .. I think its playing "calm-river" ... but I cant grep that in any of the conf files |
| 14:42.45 | iCEBrkr | robin_z: Music on Hold? |
| 14:42.52 | Mag1KaL | iCEBrkr, then what's the point? How does an application on 's' get called then? |
| 14:42.53 | robin_z | iCEBrkr yeah |
| 14:43.03 | De_Mon | robin_z take them out of your music directory |
| 14:43.06 | iCEBrkr | Mag1KaL: When you land in that context 's' is called. |
| 14:43.09 | robin_z | oh, OK. |
| 14:43.13 | MattB2 | iCEBrkr: ok, i was hoping there was something clever tyou could do with the PRI |
| 14:43.14 | iCEBrkr | robin_z: Yea, they're randomly picked to play |
| 14:43.18 | robin_z | ahh |
| 14:43.19 | iCEBrkr | MattB2: Nope |
| 14:43.22 | MattB2 | how do i know what area code to add to the 7 digit numbers thou? |
| 14:43.30 | iCEBrkr | MattB2: Just do a bunch of pattern matching in your dial plan. |
| 14:43.34 | Err | use your default area code, right? |
| 14:43.39 | Mag1KaL | iCEBrkr, but nothing is happening when I enter the context. |
| 14:43.53 | robin_z | iCEBrkr; thnaks ... I presume world-mix is just a pop beat thing? |
| 14:44.12 | saftsack | are there some nice hylafax gui clients? |
| 14:44.14 | iCEBrkr | Mag1KaL: Example? |
| 14:44.21 | MattB2 | err: that's fine for those numbers that use the default code |
| 14:44.28 | *** join/#asterisk coppice (n=chatzill@193.197.17.210.dyn.pacific.net.hk) |
| 14:44.31 | MattB2 | but round here 706 mis also a "local" number |
| 14:44.39 | MattB2 | so ppl are used to dialling the 7 digits without the 706 or 423 |
| 14:44.43 | MattB2 | so i gotta figure out which to add to the front :S |
| 14:44.48 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 14:45.48 | Err | MattB2: so you need to find out what prefixes are local with what area codes |
| 14:45.56 | Err | that's so stupid that the phone system was ever allowed to do that |
| 14:46.13 | Mag1KaL | iCEBrkr, Ok, let's say I have the s extension playing a sound and that's the only thing in the context. Now what does the user have to do to make the sound play? |
| 14:46.19 | Err | (I fortunately live in an area where 10-digit dialing is required to cross area codes, even when they're local numbers) |
| 14:46.20 | MattB2 | yup ;) |
| 14:46.22 | littleball | hello, i want to choose a specific channels based on the countrycode of the called number, how to do prefix matching in the dial plan? who can help? |
| 14:46.23 | MattB2 | ok thanks for the guidance folks |
| 14:46.25 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 14:47.20 | littleball | eg., for 44 number, i want to choose Zap/r1 but for 49 i want to choose zap/r2 |
| 14:48.45 | *** join/#asterisk Galel (n=galel@63.245.93.138) |
| 14:49.05 | foucaulo | Err: don't have file dtmftotext.c or app_dtmftotext.c in my setup... |
| 14:49.31 | Katty | ello. |
| 14:49.35 | iCEBrkr | Mag1KaL: Well, depends. |
| 14:52.06 | iCEBrkr | Mag1KaL: If you're dialing an extension, it's not going to land in 's'. If you have a Goto() statement you can land in a context and then 's' will fire |
| 14:52.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:53.16 | Mag1KaL | Ok, thanks. |
| 14:53.18 | *** join/#asterisk qazwsx (n=qazwsx@201.11.108.54) |
| 14:53.29 | synthetiq | anyone here use yate |
| 14:53.31 | synthetiq | ? |
| 14:53.57 | fourcheeze | synthetiq: a bit |
| 14:54.14 | synthetiq | opions on it vs asterisk? |
| 14:54.19 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:54.21 | synthetiq | opinions |
| 14:55.26 | Mag1KaL | asterisk has a cooler name. |
| 14:55.29 | fourcheeze | synthetiq: fewer features but more easily extensible, more lightweight, possibly less resource hungry, geared to voip technology |
| 14:55.41 | *** join/#asterisk fugitivo (n=ajf@201.255.176.51) |
| 14:55.47 | synthetiq | they are claiming 500 concurrent users and ability to laod balance amongst cloen dmachines |
| 14:55.51 | fourcheeze | synthetiq: I'm thinking of using it for h323 stuff |
| 14:55.55 | fugitivo | hello |
| 14:56.04 | fourcheeze | yes, I understand that's possible |
| 14:56.29 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:56.35 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
| 14:57.04 | synthetiq | but * cannot, * ftl =[ |
| 14:57.31 | fourcheeze | synthetiq: yate doesn't have such a large amount of developers |
| 14:57.48 | fourcheeze | asterisk is more of a standard solution |
| 14:57.58 | fugitivo | like windows |
| 14:58.15 | fourcheeze | I didn't say that |
| 14:58.30 | saftsack | how can i send a testfax with hylafaxß |
| 14:58.31 | mut | damn |
| 14:58.31 | fourcheeze | I'm considering deploying a mixed yate/asterisk solution as I think they both have strengths |
| 14:58.32 | *** join/#asterisk apardo_ (n=apardo@62.97.121.93) |
| 14:58.36 | *** join/#asterisk tRSS (n=tRSS@202.174.142.2) |
| 14:58.38 | mut | svn checkout is fscking me |
| 14:58.42 | mut | setgroup is totally gone |
| 14:58.46 | mut | dbget is totally gone |
| 14:59.17 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 15:00.32 | j0n | is it possible for a user to dial an extension while they are waiting in a queue? |
| 15:01.20 | viperdude | jontow: yes looking at defining a context for the queue which gives a mini break out menu |
| 15:01.55 | Katty | mew. |
| 15:01.56 | j0n | ahh.. I just found that. Thanks! |
| 15:01.57 | iCEBrkr | mut: Maybe you should read the Changelog once in awhile |
| 15:02.06 | mut | heh |
| 15:02.07 | mut | screw that |
| 15:02.11 | mut | it changes too much |
| 15:02.21 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:02.24 | mut | i just throw my old crap in the new crap and see what dies |
| 15:02.25 | Katty | mew? |
| 15:02.26 | iCEBrkr | mut: DBGet has been deprecated since 1.2.x yo. |
| 15:02.27 | mut | then read later |
| 15:02.32 | mut | yea |
| 15:02.35 | mut | my last install was.. |
| 15:03.01 | iCEBrkr | mut: so svn checkout isn't fsking you.. you're fscking you. :D |
| 15:03.02 | mut | CVS HEAD 2005-11-10 18:43:55 UTC |
| 15:03.14 | mut | same difference ;) |
| 15:03.19 | iCEBrkr | Not really |
| 15:03.39 | mut | stop yelling at me! |
| 15:03.43 | iCEBrkr | lol |
| 15:03.55 | Katty | :< |
| 15:04.55 | tzanger | sivana: you wouldn't give yours up |
| 15:04.58 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 15:04.59 | sivana | :P |
| 15:05.12 | Katty | this calls for cookies. |
| 15:05.23 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:06.23 | *** join/#asterisk relativeambition (n=relative@216-53-216-002.corpserv.mpinet.com) |
| 15:06.52 | relativeambition | Does anyone know if asterisk supports the brooktrout tr1034 t1 card? |
| 15:07.19 | sivana | only the binford 9000 |
| 15:07.32 | [TK]D-Fender | More Power!!! urgh urgh urgh!!! |
| 15:07.58 | coppice | relativeambition: it does not |
| 15:08.01 | iCEBrkr | Katty cookie: FPB=38vf9ohto11tcgll Allow? (Y/N/Always/neVer) V |
| 15:08.38 | relativeambition | Does anyone know if asterisk supports the brooktrout tr1114 t1 card? |
| 15:08.43 | tzanger | you don't want katty's cookies? |
| 15:09.14 | iCEBrkr | tzanger: They'd have poison in them. |
| 15:11.09 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 15:11.17 | Katty | tzanger: yes, because i am an evil evil person. |
| 15:11.29 | iDunno | yes, yes you are. |
| 15:11.32 | iCEBrkr | Katty: You have tits, right? |
| 15:11.36 | iCEBrkr | Katty: Thus.. You're evil. |
| 15:11.43 | Err | foucaulo: your best bet, then, is to look at the makefile and find out what files are used to build app_dtmftotext.so and grep them |
| 15:12.27 | Ariel_ | Morning |
| 15:12.42 | [TK]D-Fender | Katty: Mew. (belated) |
| 15:12.49 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:12.53 | Katty | [TK]D-Fender: :> |
| 15:12.58 | Ariel_ | Katty, I would hug but I think I am catching something.... argh |
| 15:13.34 | Katty | Ariel_: and i can obviously catch it over irc. |
| 15:13.56 | Katty | right? ;) |
| 15:14.08 | Ariel_ | Katty, ok just a habit.. |
| 15:14.38 | tzanger | Katty: mmmmm... evil cookies |
| 15:14.44 | Lloydie-t | <PROTECTED> |
| 15:14.52 | Ariel_ | someone giving cookies... |
| 15:15.04 | Err | Lloydie-t: don't use SIP clients :-) |
| 15:15.13 | Ariel_ | Lloydie-t, put your box with an outside address and use iptables |
| 15:15.38 | Lloydie-t | Yeah I know but most decent hard phones only support this |
| 15:16.12 | Err | Ariel_: if it's not *your* NAT it doesn't... |
| 15:16.17 | Err | (for example, roaming phones) |
| 15:16.25 | Ariel_ | Lloydie-t, I use polycom's and they don't have any issues with nat. Also most of my networks I put monowalls setups and they work great. |
| 15:17.04 | Ariel_ | Err, most setups like linksys are not a problem sipura, Polycoms and even Cisco work just fine over nat. |
| 15:17.58 | Lloydie-t | OK would * have to be on a public IP address? |
| 15:18.13 | Err | Lloydie-t: no, the server can be behind a NAT as long as you forward the ports correctly |
| 15:18.47 | Ariel_ | It would be better if it was on the public IP address. |
| 15:19.11 | Err | that is true |
| 15:19.16 | Ariel_ | Linux is able to work just like a firewall you just need to configure the iptables |
| 15:19.18 | Err | well, it'd be easier to configure, anyway |
| 15:19.35 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 15:19.53 | Lloydie-t | I'm confused. I was looking at voip.org and it rattles on about various problems regarding registration and rtp |
| 15:19.59 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 15:20.00 | Err | I wouldn't necessarily recommend that a server like asterisk run on the same machine as a firewall, though |
| 15:20.04 | Ariel_ | EriSan, why would you want to add another layer to the nat issue |
| 15:20.20 | Ariel_ | sorry EriSan it's for Err |
| 15:20.22 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-133.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:20.50 | Err | Ariel_: I wouldn't *want* to - but I would prefer not to have any services running on my firewall, since it has several networks connected to it - and I'd prefer for a hole in asterisk not to make them all vulnerable |
| 15:20.52 | Ariel_ | Err, There is no reason not to be able to use the asterisk box as a firewall. |
| 15:21.02 | Err | I just gave you one |
| 15:21.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 15:21.27 | *** join/#asterisk zaf (n=tfournet@cdm-68-228-9-79.laft.cox-internet.com) |
| 15:21.28 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:21.54 | Ariel_ | Err, many setups I do are this way. If you are putting more servers behind it that is a different story. But I would still use a monowall and forward an IP addres directly to the asterisk box and still use iptables on it. |
| 15:22.01 | Err | performance is another consideration, although with a NAT there's probably a slow link involved |
| 15:22.10 | Err | Ariel_: I know that many setups are run that way - but I don't htink it's a good idea |
| 15:22.13 | Err | lots of people do dumb things |
| 15:22.18 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 15:22.30 | klasstek | Good morning. |
| 15:22.34 | klasstek | In chan_agent agentmonitoroutgoing would it be possible to masquerade chan into an agent channel if it finds the agent to monitor? |
| 15:22.37 | Err | NAT typically implies that there aren't multiple IP addresses involved |
| 15:23.33 | Ariel_ | Err, yes correct. But if your setting up servers like asterisk and have others via the internet connect to it. have it's own IP address and setup iptables on that box. |
| 15:23.54 | Ariel_ | meeting time. brb |
| 15:24.01 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 15:24.22 | Err | sure, that's ideal - I never argued that |
| 15:24.27 | Err | that wasn't the question, though |
| 15:25.03 | Katty | Err: i love how people don't answer questions directly in here. |
| 15:25.50 | saftsack | Jan 24 16:24:53 NOTICE[31234]: chan_iax2.c:7243 socket_read: Rejected connect attempt from 127.0.0.1, requested/capability 0x8/0x4c incompatible with our capability 0xff03 |
| 15:25.57 | saftsack | what does this mean? |
| 15:26.43 | Err | I'm guessing that it means that you can't negotiate a capability set (compression codecs, maybe?) |
| 15:27.14 | saftsack | i am sending a fax over iaxmodem with the alaw codec |
| 15:27.28 | hans | in queues.conf it says you can have a member Zap/1, for example. In that case, when you don't use an agent, there's no agent login/logout right? what about timeout? |
| 15:28.16 | *** join/#asterisk techy (n=tecky@66.9.96.115) |
| 15:28.25 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 15:30.06 | Err | saftsack: is alaw supported on both ends? |
| 15:30.20 | saftsack | i added alaw to iax |
| 15:30.29 | saftsack | a second ill test now |
| 15:31.11 | saftsack | Err, now it seems to work :) |
| 15:31.39 | Lloydie-t | Re: NAT. If you have a 200 sip clients all behind various NATs and an * box servering these should not pose too many problems |
| 15:33.23 | |vinsik| | lloydie-t: ? |
| 15:33.43 | Err | I suspect that a single SIP phone behind the NAT I have at my apartment would never be able to get through |
| 15:33.48 | *** join/#asterisk mko-025 (n=korpim@p5498968C.dip0.t-ipconnect.de) |
| 15:34.09 | |vinsik| | actually it will :) |
| 15:35.05 | |vinsik| | is lloydie-t a forum bot ? |
| 15:35.20 | Lloydie-t | No Im not |
| 15:35.28 | |vinsik| | who did you reply for? |
| 15:35.39 | Err | well, it'll get out, but it won't *work* |
| 15:35.50 | |vinsik| | because i was sniffing for an answer to questions like that :) |
| 15:36.21 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:36.31 | |vinsik| | im having some problems with NAT users all connecting to one * |
| 15:37.28 | Lloydie-t | I am just trying to put together a plan for using SIP phones without having to resort to using STUN |
| 15:37.39 | |vinsik| | nice |
| 15:38.04 | |vinsik| | lloydie-t: really could use some help with this... im starting too loose my nerves with this project |
| 15:38.04 | Lloydie-t | Possibly using some sort of sip proxy |
| 15:38.39 | Lloydie-t | Ehh |
| 15:38.57 | Lloydie-t | I think we are on different things |
| 15:39.27 | *** part/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
| 15:39.38 | Err | you'll either need phones that can do that uPnP garbage and NAT boxes that do too (or some other method of tunneling through the NAT), or NATs with rules specifically to make the phones work, or SIP proxies at every NAT (as far as I know) |
| 15:41.50 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:41.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:42.38 | zoa | hey ho antony |
| 15:43.53 | anthm | hi |
| 15:48.07 | *** part/#asterisk VJ (n=vijay@203.122.28.98) |
| 15:48.37 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:49.02 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 15:49.18 | *** join/#asterisk infernall (n=wess@d221-71-12.commercial.cgocable.net) |
| 15:49.53 | hans | <PROTECTED> |
| 15:49.54 | hans | <PROTECTED> |
| 15:50.06 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 15:50.16 | hans | I suppose I have to log in? how does that work? |
| 15:50.31 | hans | and should I set up an agent or is that sufficient? |
| 15:50.32 | *** join/#asterisk brockj49464_ (n=brockj49@22.105.dhcp.hope.edu) |
| 15:51.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:52.35 | *** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 15:56.16 | [TK]D-Fender | hans : is that an external agent? (forced outbound IAX2 phone)? |
| 15:56.45 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 15:57.19 | hans | yes, I'm trying to call IAX2/fugal.net/hans, which is external to this asterisk server. That's just for me to test, though, in the end it will be SIP phones in the office, e.g. SIP/foobar |
| 15:57.23 | __chris | fs - just found this - http://www.voip-info.org/wiki-Cisco+POE - typical they dont use 802.3af! |
| 15:57.48 | hans | i'd just as soon they were just members, rather than agents |
| 15:57.52 | hans | if that's possible. |
| 15:58.00 | [TK]D-Fender | hans : I think that that kind of login will not work. * can't monitor the extensions status since it isn't a registered phone |
| 15:58.19 | [TK]D-Fender | __chris : Yup! Another reason to go Polycom ;) |
| 15:58.32 | johnnyb | Is there something special you have to do to get asterisk to start with two TDM cards? |
| 15:58.50 | hans | but if I had member => SIP/foobar and that phone is registered, would it work then? without login? |
| 15:59.37 | johnnyb | I've got two 4-port TDM cards, and asterisk will not start if I enable both of them in zapata.conf. If I go past channel => 4 it kill asterisk on startup. |
| 16:01.05 | [TK]D-Fender | hans : Correct. you are using direct TECH as agents instead of call-back logins. You could use it the way you are if you mixed in AGENT => style logins for that account and did a manual dial. |
| 16:01.32 | [TK]D-Fender | johnnyb : Do you see both in cat /proc/interrupts? |
| 16:01.43 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:01.51 | hans | [TK]D-Fender: cool, thanks |
| 16:02.03 | [TK]D-Fender | cpm : I was waiting for them too, but I've seen the procing and quality. For business users I wouldn't suggest them. |
| 16:02.15 | *** join/#asterisk LeNiN (i=djon@a1-d109.vologda.ru) |
| 16:02.19 | LeNiN | oro |
| 16:02.22 | LeNiN | ogo |
| 16:02.25 | LeNiN | to ecTb |
| 16:02.50 | LeNiN | who is alive this ? |
| 16:03.01 | [TK]D-Fender | pricing* |
| 16:03.05 | *** join/#asterisk Dream[atwork] (n=Dave@88-107-27-6.dynamic.dsl.as9105.com) |
| 16:03.12 | [TK]D-Fender | cpm : NOT good... for what you get... |
| 16:03.15 | cpm | [TK]D-Fender, You've seen the phones? The pricing is pretty agreeable, esp vs the Polycom, quality wise, I've not seen an injected molded plastic ethernet device with a lcd screen and buttons that was much better than any other. |
| 16:03.46 | cpm | I must admit I do like the polycoms I have, but they are spendy. |
| 16:04.22 | [TK]D-Fender | cpm : IP 501 = $170, SPA-941= $150 but has no PoE, 2nd eth port, lower quality speakerphone, inferior display, etc.... |
| 16:04.39 | [TK]D-Fender | cpm : For a cheap home person maybe, but not business |
| 16:04.41 | johnnyb | [TK]D-Fender: Yes, they are both in /proc/interrupts. One is at IRQ 3 and the other is on IRQ 10. |
| 16:04.55 | johnnyb | In /dev/zap I get numbers 1-8. |
| 16:05.03 | [TK]D-Fender | johnnyb : pastebin your zaptel.conf and zapata.conf |
| 16:05.05 | [TK]D-Fender | ~pb |
| 16:05.07 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:05.32 | *** part/#asterisk LeNiN (i=djon@a1-d109.vologda.ru) |
| 16:05.33 | [TK]D-Fender | cpm : I run an all-Polycom shop here, and an all-Sipura one at home w/ 941 |
| 16:06.01 | cpm | On a side note, I got a call from regional Microsoft corporate rep, who wanted to show off his Microsoft PBX system. I suppose I should call him back and see what he's offering. |
| 16:06.07 | Mag1KaL | Has anyone made an Asterisk game yet? ;) |
| 16:06.13 | [TK]D-Fender | cpm : M$ .... lol |
| 16:06.33 | cpm | I can't find any refs to a M$ pbx anywhere, I'm interested in seeing what it is. |
| 16:06.35 | Mag1KaL | MS has a PBX system? Since when? |
| 16:06.49 | fenlander | LCS 2005? |
| 16:07.11 | cpm | I guess I will call him back this week and get him to come in. |
| 16:08.27 | cpm | [TK]D-Fender, you are getting much better pricing on Polycom than I am. |
| 16:09.22 | rajiv|work | [TK]D-Fender: you really like the polycom 501 over the spa-941 ? i'mlooking at getting 6-8 phones in the next month for a small office |
| 16:10.16 | cpm | I've got only 1 501, it's a nice phone, but it is spendy. I paid $240 for it. |
| 16:11.06 | *** join/#asterisk Hmmhesays (n=Neg@72.24.227.83) |
| 16:11.43 | brad_mssw | i'd hold out for the spa-942 if you like the spa-941 |
| 16:11.44 | wunderkin | i think you can get a 601 for 250 |
| 16:11.47 | brad_mssw | due out early next month |
| 16:12.14 | brad_mssw | (at least in an office environment, where you usually only have 1 ethernet port, having a built-in 2pt switch is nice, which is what the 942 adds) |
| 16:12.35 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 16:12.39 | *** join/#asterisk Buck1tH3d (n=Buck1tH3@216.90.111.10) |
| 16:15.26 | *** join/#asterisk Zach^^ (i=chaos@dialup-4.224.213.107.Dial1.Cincinnati1.Level3.net) |
| 16:16.14 | Zach^^ | i have a dedicated server and i was to install asterisk or asterisk at home on it... it is running fedora core2 w/ cpanel right now... will this cause any problems? |
| 16:16.15 | cpm | the 942 is also a PoE device, like the 600/01 |
| 16:16.26 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com) |
| 16:17.22 | *** join/#asterisk ivanfm[wrk] (n=ivanfm@cumulus.saisp.br) |
| 16:17.31 | [TK]D-Fender | rajiv : Yeah for office use I'd really suggest it.. |
| 16:18.01 | [TK]D-Fender | cpm : Look at the price point on it though. The control over calls/linekey and registrations is definately inferior on the SPA's |
| 16:18.30 | [TK]D-Fender | cpm : I had high hopes for it, and though it is an improvement, it isn't in the same class yes, and therefor not worth the price they charge. |
| 16:18.53 | [TK]D-Fender | cpm : And for the love of God, don't compare to a 601! 601 absolutely kills it.... |
| 16:18.58 | Zand3r | I anyone using vim on Windows for Rails and have a nice vimrc file they wouldn't mind making public? I am on windows and jsut can;t find an editor or ide I like so am going back to vim but would like to make it a little friendlier. |
| 16:19.22 | Zand3r | Please ignore - stupidly typed in to wrong window ! |
| 16:19.49 | mut | what are the prequisites for mpg123? |
| 16:19.55 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 16:20.20 | mut | http://pastebin.ca/38370 |
| 16:21.31 | [TK]D-Fender | And SPA's don't really support presence yet. |
| 16:22.08 | Zach^^ | anyone using asterisk at home on FC2? |
| 16:22.34 | [TK]D-Fender | A@H is build on CentOS..... |
| 16:22.35 | Ariel_ | Zach^^, It comes with CentOS |
| 16:23.31 | Zach^^ | Ariel_ okay well i have a FC2 server that is running a webhosting and i want to install asterisk for the company and have sip phones all over... is that posible? |
| 16:23.59 | pifiu-laptop | hows everyone doing today? |
| 16:24.04 | Ariel_ | Zach^^, yes but not asterisk at home unless you edit the install files |
| 16:24.05 | hans | does the new moh files work with 8khz mono WAV files? |
| 16:24.18 | Ariel_ | Zach^^, you can setup most things there via the amp setup |
| 16:24.25 | Zach^^ | Ariel_ what would you use as a gui then? |
| 16:25.18 | Ariel_ | Zach^^, I use AMP which is part of the asterisk at home setup |
| 16:25.36 | Ariel_ | asterisk at home is a complete OS, and other programs pre-configured |
| 16:25.44 | junbug | Ariel_: sup... its Inv_Arp |
| 16:26.13 | [TK]D-Fender | Zach^^ : How big a company? |
| 16:26.14 | Ariel_ | Zach^^, http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 16:26.18 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 16:26.25 | Ariel_ | junbug, wow what a name change |
| 16:26.38 | Zach^^ | [TK]D-Fender small 3 lines |
| 16:26.40 | Ariel_ | junbug, I am doing well hope you are as well |
| 16:26.40 | junbug | Ariel_: just use it at werk |
| 16:26.52 | [TK]D-Fender | Zach^^ : how many phones? |
| 16:27.07 | Zach^^ | 4 |
| 16:27.14 | junbug | Ariel_: so far... got a job , unix support in doral |
| 16:27.20 | MattB2 | hi all... the example at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record - after s,6 it drops out on mine with "Auto fallthrough, channel 'SIP/630-1685' status is 'UNKNOWN'" - how do i make it wait for someone to enter a number? |
| 16:27.22 | [TK]D-Fender | Zach^^ : AMP really isn't worth it for such a small setup..... |
| 16:27.26 | Ariel_ | junbug, nice |
| 16:27.39 | Zach^^ | [TK]D-Fender somthing better? |
| 16:27.53 | [TK]D-Fender | MattB2 : Set "autofallthrough=no" in [general] in extensions.conf |
| 16:27.57 | junbug | yea i need to be there |
| 16:28.03 | [TK]D-Fender | Zach^^ : just run * normally w/o any gui |
| 16:28.05 | Hmmhesays | are there any country codes that start with 1? |
| 16:28.15 | [TK]D-Fender | Ariel_ : Oh.. you mean broke :) |
| 16:28.24 | Ariel_ | [TK]D-Fender, yes I do |
| 16:28.25 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 16:28.44 | MattB2 | [TK]D-Fender - aha, thanks. |
| 16:28.47 | jbalcomb | yippy! a new week of fun with Asterisk! |
| 16:28.50 | Corydon-w | Hmmhesays: there's only one |
| 16:28.53 | junbug | Ariel_: mostly asterisk? |
| 16:29.23 | Corydon-w | Hmmhesays: each country code prefix is, in its own right, unique |
| 16:29.33 | Ariel_ | junbug, no I am working with many different setups. I am helping a reseller with there support of some voip products as well. |
| 16:30.01 | [TK]D-Fender | jbalcomb : So... got any hair left? ;) |
| 16:30.50 | Corydon-w | Hmmhesays: no country has a country code which is a prefix of another country's country code |
| 16:33.31 | johnnyb | [TK]D-Fender: While working on the pastebin, I figured it out. In /etc/zaptel.conf I had "fxsks=1-4". Thanks for your time. |
| 16:33.44 | [TK]D-Fender | :) Suspected as much... |
| 16:34.16 | MattB2 | t all |
| 16:34.20 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 16:34.52 | brookshire | mattb! |
| 16:34.56 | brookshire | oh wait.. he left |
| 16:34.57 | brookshire | :( |
| 16:34.58 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 16:35.39 | jozsab1 | Hy all. Is this channel for asterisk (server) questions ? |
| 16:35.48 | brookshire | yes |
| 16:36.00 | brookshire | you can really ask anything |
| 16:36.13 | jozsab1 | where can i get answers on level3 testcases ? (channel) |
| 16:36.14 | brookshire | people are more likely to answer other questions ;) |
| 16:36.26 | jozsab1 | :) |
| 16:36.33 | brookshire | you need test cases? |
| 16:36.36 | brookshire | or just on level3? |
| 16:37.05 | jozsab1 | i'm stuck with testcase number 2.4.5 |
| 16:37.19 | jozsab1 | i realy do not understand what they are asking me |
| 16:37.35 | jozsab1 | The call is answered by SIP end, then a second call leg is setup and the media is bridged for a Three-Way Conference Call. It is hung up from the PSTN end. |
| 16:37.57 | jozsab1 | How do i do this ? |
| 16:38.49 | *** join/#asterisk apardo_ (n=apardo@62.97.121.95) |
| 16:39.44 | jozsab1 | Thanks , just don't rush me with to many answers :) |
| 16:41.32 | *** join/#asterisk crich1999 (n=crich@p54BF87DD.dip0.t-ipconnect.de) |
| 16:42.07 | brookshire | so it doesn't have to use level3 basically |
| 16:42.14 | brookshire | there should be ton of stuff on this |
| 16:42.21 | brookshire | research www.voip-info.org |
| 16:42.38 | junbug | Ariel_: is it better to colo my own box, or have someone from a pricing standpoint |
| 16:42.51 | junbug | err have someone host it |
| 16:43.05 | jozsab1 | just let others feel the pain |
| 16:43.10 | jozsab1 | :) |
| 16:43.35 | brookshire | jozsabl: http://www.digium.com/index.php?menu=case_studies |
| 16:43.36 | jozsab1 | thanks for the adress but already knew it |
| 16:43.44 | Ariel_ | junbug, http://serverpronto.com/ is a location I have setup a few boxes |
| 16:44.24 | junbug | Ariel_: ahh lemme check it out |
| 16:44.36 | Ariel_ | There here in the Miami area |
| 16:44.58 | junbug | Ariel_: do have your own company/website etc... may have business for you |
| 16:45.15 | Ariel_ | junbug, no I don't. (Too poor for that) |
| 16:45.19 | iCEBrkr | w00t! http://asteriskpbx.meetup.com |
| 16:45.30 | Ariel_ | just work of refurals |
| 16:45.52 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 16:46.09 | ceeto | Anyone here use email2fax? I'm trying to figure out how to tell it to use zap/g1 instead of zap/g0 ... |
| 16:47.54 | [TK]D-Fender | Ariel_ : You are apparently having issues with grammar as well, as "issue's" should not be possessive. :) |
| 16:48.25 | junbug | Ariel_: oh so serverpronto=colopronto |
| 16:48.37 | junbug | man they got some good rates |
| 16:48.55 | ceeto | Maybe I can approach it this way, what does TRUNK=Zap/g1 signify in my [globals] in extensions.conf? Can I just change it to Zap/g0? |
| 16:49.11 | tuxinator_linux | Ariel_: http://www.angryflower.com/bobsqu.gif |
| 16:49.16 | cpm | my issue is itching really badly. |
| 16:49.41 | jbroome | they've got creams for that |
| 16:49.51 | cpm | jbroome, thanx |
| 16:50.50 | DarkFlibble | ib |
| 16:51.56 | [TK]D-Fender | ceeto : Thats just a global variable. Do you even use it anywhere> |
| 16:51.58 | *** join/#asterisk sm7xab (n=sm7xab@h229n2c1o1095.bredband.skanova.com) |
| 16:52.48 | rajiv|work | [TK]D-Fender: does the 501 include a wall wart ? |
| 16:52.55 | rajiv|work | no PoE here |
| 16:52.58 | sm7xab | Hi! I'm trying to find out why my * server won't register with my provider. Anyone here who has a cool tip or two regarding this? Haven't managed to find anything in the docs. *=1.2.1 |
| 16:53.08 | *** join/#asterisk tehdely[ETEL] (n=delysiid@home.teambarry.org) |
| 16:53.08 | [TK]D-Fender | rajiv : I believe so. It needs a special adapter cable for PoE. |
| 16:53.14 | *** join/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net) |
| 16:53.20 | tehdely[ETEL] | morning |
| 16:53.34 | ceeto | I don't see it anywhere else in my extensions.conf ... |
| 16:53.53 | [TK]D-Fender | ceeto : then that line is useless |
| 16:53.55 | jbalcomb | [TK]D-Fender haha.. yeah, no combover yet. The 'other' phone guy disputed my interest in the Sangoma card as well as the polycoms. They want more information about my proposal. :/ |
| 16:54.24 | jbalcomb | [TK]D-Fender you wanna right a sweet blog I can add to my proposal? |
| 16:54.29 | jbalcomb | s/right/write |
| 16:54.52 | *** part/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net) |
| 16:55.01 | [TK]D-Fender | jbalcomb : Well print up both full spec sheets, take the community optinions, and add on that I use the one I suggested personally and its been flawless to date. |
| 16:55.29 | ceeto | Yea, I'm using it. |
| 16:55.30 | [TK]D-Fender | jbalcomb : Writing takes too long :) I'd do a quick phone survey though! |
| 16:56.35 | jbalcomb | [TK]D-Fender sounds reasonable. our rxgain at -4.5 and txgain at -16 seems good for us. |
| 16:57.16 | jbalcomb | [TK]D-Fender atleast thats where people can use the headsets and both sides can still hear the conversation. oh, and navigating IVRs works there too. |
| 16:57.28 | [TK]D-Fender | jbalcomb : Mine at 0.0 and 0.0 work just great! |
| 16:57.36 | jbalcomb | [TK]D-Fender no reports on whether EC has improved yet though |
| 16:57.47 | [TK]D-Fender | :O |
| 16:58.17 | jbalcomb | [TK]D-Fender yeah, i need to look into the PRI config, the PRI card config, the Asterisk compile time options, driver versions, and firmware versions. |
| 16:59.56 | [TK]D-Fender | jbalcomb : I take it your played with the AEC, and other Zaptel s/w EC's, as well as echotraining already? |
| 17:00.18 | *** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:03.15 | *** part/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 17:03.43 | jbalcomb | [TK]D-Fender AEC? other zaptel s/w EC's? just echotraining=yes |
| 17:05.53 | bkw__ | OH rrrrrrrrreally |
| 17:06.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:10.22 | *** join/#asterisk SwK (n=SwK@64.89.118.139) |
| 17:11.54 | Modcuts | I have registered the g729 codec but my trunk doesn't seem to be using it? is it phone dependent or trunk conf dependent? |
| 17:12.38 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 17:13.30 | stack_ | Is it work the ~$200 to get echo cancellation on the TDM2400P? |
| 17:13.42 | *** part/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 17:14.23 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 17:14.28 | *** join/#asterisk astoria (n=tom@user-7e5a43.user.msu.edu) |
| 17:14.34 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 17:14.39 | SpaceBass | Morning' |
| 17:16.24 | SpaceBass | I recently upgraded my AAH box and re-installed and have 4 major problems I'm trying to iron out... am hoping I can get some insight on any of them today |
| 17:17.25 | SpaceBass | 1) BAD echo on zaptel lines 2) Zaptel calls "appear" to jump contexts when I use dring 3) Faxes are recieved but are blank and finally 4) broadvoice has no audio on some of my devices (think its a bridge issue) |
| 17:17.59 | *** join/#asterisk seelen (n=_seele@200.124.172.72) |
| 17:18.18 | SpaceBass | For my zaptel echo I've tried playing with rx and tx gains and it doesnt seem to make a difference... even if the call starts with no echo, its gets progressivly worse |
| 17:18.50 | seelen | Hi, i need to figure out a good dialplan that prevents a single exten, to acces PSTN, but without having to change the rest of the extensions of context. how to do that? |
| 17:19.20 | SpaceBass | seelen, put that exten in its own context? |
| 17:21.17 | *** join/#asterisk Canadien (n=Canadien@Fd26c.f.strato-dslnet.de) |
| 17:21.40 | *** part/#asterisk Canadien (n=Canadien@Fd26c.f.strato-dslnet.de) |
| 17:22.01 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfni1.dialup.mindspring.com) |
| 17:22.10 | *** join/#asterisk roulduke_ (i=mkdhwmfn@p508D0C99.dip0.t-ipconnect.de) |
| 17:22.14 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfni1.dialup.mindspring.com) |
| 17:23.41 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 17:23.51 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 17:25.01 | seelen | SpaceBass, but that way, i wont be able to connect it to the rest of the exten to that single one |
| 17:25.19 | Hmmhesays | let me guess you are using AAH? |
| 17:25.42 | SpaceBass | Hmmhesays, which one of us? I addmitted to it already :) |
| 17:25.48 | Hmmhesays | seelen |
| 17:26.00 | Hmmhesays | SpaceBass: nothing wrong with AAH if you understand how it works |
| 17:26.07 | SpaceBass | I'm fine with AAH for my HOME install... i dont think its the root of any of my issues |
| 17:26.10 | seelen | Hmmhesays, why??.. no im asking for a dialplan |
| 17:26.13 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 17:26.15 | SpaceBass | Then again, I edit a lot of the confgs manually too |
| 17:26.19 | darwin_35 | Goodmoning |
| 17:26.25 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 17:26.29 | darwin_35 | having a iax issue on 1.2.2 |
| 17:26.33 | *** join/#asterisk RoyKa (n=roy@10.80-203-106.nextgentel.com) |
| 17:26.44 | SpaceBass | I had iax trunking issues in 1.2,2 |
| 17:26.48 | Hmmhesays | seelen: set a variable to 0 and check for that before you allow calls out the pstn |
| 17:26.56 | SpaceBass | Hmmhesays, how are you btw...been a while |
| 17:27.22 | Hmmhesays | SpaceBass: surviving up in the frozen tundra, just bought a kickass new guitar amp. |
| 17:27.23 | darwin_35 | if a user on our box originates a call to another iax box the phones ring but you dont hear the rining on the line. but if you use sip to sip its fine . |
| 17:27.24 | Hmmhesays | yourself? |
| 17:27.25 | seelen | Hmmhesays, great, what variable? |
| 17:27.32 | SpaceBass | Hmmhesays, what'd you get? |
| 17:27.50 | Hmmhesays | seelen: any one you want, or you can do it based on callerid |
| 17:28.05 | SpaceBass | Hmmhesays, I've been playing my bass a lot more these days...got some new earphones for music (Shure ec4) and noticed they are GREAT for practice and garrage band |
| 17:28.07 | Hmmhesays | seelen: assuming you are trying to keep one of your sip or iax2 peers from dialing out the pstn? |
| 17:28.21 | darwin_35 | any one here having iax to iax rining problems ? |
| 17:28.35 | darwin_35 | be it not passing the rining on the line ? |
| 17:28.41 | SpaceBass | darwin_35, I had that exact issue, hear around here that it was a known bug |
| 17:28.42 | Hmmhesays | Spacebass: i picked up a peavey XXL 100watt Head, set it on my peavey 4x12 cab |
| 17:28.53 | SpaceBass | nice! |
| 17:29.01 | Hmmhesays | Shure ec4's huh? how much those run you? |
| 17:29.03 | darwin_35 | ok |
| 17:29.19 | seelen | Hmmhesays, yes it a sipm exten i have outside the city.. i dont want it to be used as a phone line to calls to my city.. just for internal communication |
| 17:29.19 | darwin_35 | grrr |
| 17:29.24 | Hmmhesays | darwin_35, all the time, use the r flag |
| 17:29.28 | darwin_35 | well this is a major isues |
| 17:29.32 | SpaceBass | Hmmhesays, it wasn't pretty... $275 |
| 17:29.38 | darwin_35 | its not working |
| 17:29.49 | Hmmhesays | what are you calling from darwin_35 |
| 17:29.53 | SpaceBass | darwin_35, by all means, don't take my word for that... I am not 100% sure at all |
| 17:29.57 | Hmmhesays | SpaceBass: ouch |
| 17:30.01 | darwin_35 | from our box to a client box |
| 17:30.08 | SpaceBass | but I had that problem b/t 2 * boxes |
| 17:30.31 | Hmmhesays | using chan_telepath or what? |
| 17:31.12 | darwin_35 | we update to 1.2.2 over the weekend on all our servers |
| 17:31.42 | Hmmhesays | SpaceBass: I wish I could afford shure for my monitors, but nady wireless with e2's are more in my price range |
| 17:31.55 | darwin_35 | and iax stopped passing the ring tone on the line but the phones ring and this is only iax to iax calls that start out on the servers to client boxes |
| 17:32.19 | Hmmhesays | what are you using to initiate the call darwin_35 |
| 17:32.34 | darwin_35 | we pass the call from opur box to the client but the person on the line hears nothing |
| 17:32.57 | darwin_35 | calls from the pstn adn sip calls |
| 17:32.57 | SpaceBass | Hmmhesays, long story... bascially my company dragged their feet paying my corporate credit card... it went into hold... had to cough up $300 to get it out of hold so I could travel... then the corporate payment went though and I ended up with basically a personal credit... so I treated myself |
| 17:33.14 | darwin_35 | and other iax calls from client boxes |
| 17:33.22 | DarkFlibble | does anyone know of any sip/voip filtering firewalls that can filter out potential attacks to a voip network for a company? Just got a query from a client |
| 17:33.23 | Hmmhesays | SpaceBass: nice, so you play in any bands? |
| 17:33.44 | Hmmhesays | DarkFlibble: what potential attacks might those be? |
| 17:34.03 | DarkFlibble | malformed sip connections, spoofing etc... |
| 17:34.04 | jbalcomb | DarkFlibble I would think foundry or cisco would have something like |
| 17:34.24 | Hmmhesays | iptables, don't accept anything from unknown ip addresses |
| 17:34.28 | darwin_35 | I dont find a bug in the bug tracker |
| 17:34.32 | DarkFlibble | jbalcomb, I'll look into it...thnx |
| 17:34.35 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 17:34.46 | SpaceBass | Hmmhesays, used to... let it slide big time, getting my chops back now and trying to talk a few freidns into doing the same |
| 17:34.56 | darwin_35 | but when we add the r on the end of our dial line it still does not pass the rining |
| 17:35.05 | SpaceBass | but I spend so much time troubleshooting my stupid * box, who has time? |
| 17:35.06 | SpaceBass | :) |
| 17:35.13 | jbalcomb | SpaceBass agreed |
| 17:35.29 | Hmmhesays | yeah SpaceBass: i just did the same, we're building the PA right now, got a warehouse to practice in |
| 17:35.37 | darwin_35 | boss breathing down my neck |
| 17:35.44 | Hmmhesays | darwin_35 did you reload your extensions after you set the r flag? |
| 17:35.50 | darwin_35 | yes |
| 17:35.58 | *** join/#asterisk Assid (n=assid@203.115.64.10) |
| 17:36.00 | Assid | heya |
| 17:36.02 | file | iax2 debug and see what's getting passed back |
| 17:36.21 | darwin_35 | it says its passing the ring but you dont hear it |
| 17:36.27 | SpaceBass | I had some downtime today so I'm getting around to chaning all my root and domain admin passwords and trying to fix my * problems |
| 17:37.16 | darwin_35 | but the phones on the other end ring |
| 17:37.16 | Assid | umm.. questions on priority.. suppose my last priority was 5.. but i want something to ALWAYS happen last.. can i just set it to 99 and expect it to be parsed? |
| 17:37.16 | file | as an indication? |
| 17:37.16 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 17:37.16 | Hmmhesays | We have 2 dual 15' peavey's pushed by a behringer 2400w power amp |
| 17:37.16 | Assid | or is it always n+101? |
| 17:37.16 | file | because if it's as an indication, then it's up to the remote side to generate the ringing or signal it... if it's inband, then it's sent as audio |
| 17:37.16 | *** join/#asterisk crich1999 (n=crich@p54BFC05F.dip0.t-ipconnect.de) |
| 17:37.18 | SpaceBass | I have my wifi sip phones on a seperate wifi subnet with DMZ pinholes back into my lan for sip...works great for internal and PSTN calls, but when I use my BV account i get no audio |
| 17:37.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:37.20 | Hmmhesays | Assid: define "last" |
| 17:37.26 | *** join/#asterisk mog_work (i=root@user-24-236-84-48.knology.net) |
| 17:37.33 | jbalcomb | Is the /etc/zaptel.conf the only place I need to look for 'PRI card' configuration issues? |
| 17:37.36 | SpaceBass | I have BV trunk and device both set to canreinvite=no but it still seems to attempt a bridge |
| 17:37.40 | Assid | Hmmhesays: like for example.. the goto for dialstatus |
| 17:37.47 | Hmmhesays | Assid: if you are talking about hangup, use extension h |
| 17:37.53 | Assid | nah |
| 17:38.00 | darwin_35 | file rechecking |
| 17:38.01 | SpaceBass | Hmmhesays, wow...nice PA! |
| 17:38.02 | Assid | like a goto for dialstatus |
| 17:38.06 | SpaceBass | Hmmhesays, what kind of music? |
| 17:38.17 | darwin_35 | file this is only on iax |
| 17:38.18 | jbalcomb | jazz |
| 17:38.22 | Hmmhesays | SpaceBass: its getting there, Rock, punk, country/rock |
| 17:38.29 | jbalcomb | damn |
| 17:38.30 | Assid | i want it to happen.. on its own.. but i dont wanna use 'n' |
| 17:38.31 | *** part/#asterisk mog_work (i=root@user-24-236-84-48.knology.net) |
| 17:38.54 | Hmmhesays | Assid: are you talking after cmd dial? |
| 17:38.57 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 17:39.02 | jbalcomb | damn |
| 17:39.03 | jbalcomb | Is the /etc/zaptel.conf the only place I need to look for 'PRI card' configuration issues? |
| 17:39.09 | Assid | yeah.. like a followme feature.. |
| 17:39.24 | Assid | dial dial dial.. whatever.. and so forth.. |
| 17:39.31 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 17:39.33 | Hmmhesays | simultaneous? or after each other |
| 17:39.36 | Assid | but in the end.. if it doesnt work.. i want it to go to dialstatus |
| 17:39.47 | Assid | after |
| 17:39.48 | Assid | but.. |
| 17:39.51 | Assid | i want it last.. |
| 17:40.02 | Assid | im just asking if i can jump priority numbers |
| 17:40.12 | file | darwin_35: I give up. |
| 17:40.18 | Hmmhesays | Assid: why wouldn't you be able to? |
| 17:40.32 | Assid | like suppose the last priority on an extension is 7 .. can i have an entry at 95 or 99 and expect it to work? |
| 17:40.49 | darwin_35 | file trying to get to the server to test but I am blocked at the min |
| 17:41.02 | Hmmhesays | Assid: if you use a goto |
| 17:41.06 | Assid | nah |
| 17:41.08 | Assid | without goto |
| 17:41.30 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 17:41.41 | Hmmhesays | Assid: you can set dial to jump to n+101 on failure |
| 17:42.12 | Assid | yes.. but suppose i dont wanna use 'n' .. just 99.. so even if its at 6 as the last priority |
| 17:42.16 | Assid | can it jump to 99 |
| 17:42.20 | [TK]D-Fender | Priority jumping = DEAD. |
| 17:42.26 | SpaceBass | my other major problem... I have 2 zaptel lines (1 and 2) 2 has dring for my fax line... they are both in their own contexts but when I enable dring calls into zap1 come in from zap2 |
| 17:42.31 | Hmmhesays | [TK]D-Fender: theres still a flag for it in dial isn't there? |
| 17:42.37 | SpaceBass | So basically I haven't had a fax for months |
| 17:43.03 | [TK]D-Fender | Hmmhesays : For people who feel its too much work to fix their dialplans to work the PROPER way |
| 17:43.04 | Hmmhesays | Assid: is there some specific reason you don't want to use goto? |
| 17:43.09 | Assid | realtime |
| 17:43.17 | [TK]D-Fender | but deprication claims ALL..... in time... |
| 17:43.23 | Assid | liie i dont have 6,7,8----98 |
| 17:43.35 | Hmmhesays | Assid: use goto |
| 17:43.50 | Katty | Hmmhesays: you're mister popular today. |
| 17:44.00 | SpaceBass | this is my zapata-auto.conf http://pastebin.ca/38387 if anyone sees why calls would "jump" contexts I'd be very appericative |
| 17:44.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:44.11 | Hmmhesays | goto(${EXTEN},98) |
| 17:44.24 | Hmmhesays | Katty: hey, i guess so |
| 17:44.25 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:44.52 | Hmmhesays | I like how voipjet returns "no answer" if their service is on the fritz, that does loverly things for my dialplan |
| 17:45.31 | [TK]D-Fender | Hmmhesays : Goto(98) would do the same... |
| 17:45.35 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 17:45.37 | Assid | voipjet been having problems recenrtly |
| 17:45.43 | Assid | use the west server |
| 17:45.59 | [TK]D-Fender | Hmmhesays : I do that for diaplans where I like to leave breathing room. |
| 17:46.08 | Hmmhesays | [TK]D-Fender: indeed |
| 17:48.28 | Hmmhesays | darwin_35: trying answering the call in the originating box before you send it |
| 17:49.18 | darwin_35 | we tried that and it screws up the billing in our cdr |
| 17:49.27 | Hmmhesays | reset cdr before you dial |
| 17:49.53 | darwin_35 | this is a realtime box I cant just reset it everytime |
| 17:50.04 | darwin_35 | but I think I found some info on it |
| 17:50.11 | Hmmhesays | you are using realtime for config? |
| 17:50.36 | jbalcomb | [TK]D-Fender can we PM for a minute? |
| 17:50.39 | darwin_35 | our whole setup here is realtime |
| 17:51.03 | Hmmhesays | so why can't use the reset cdr command? |
| 17:51.04 | darwin_35 | but Ithink I found a note on the rining not being passed on iax |
| 17:51.17 | darwin_35 | my boss says not to |
| 17:51.28 | Hmmhesays | tell your boss where to stick it |
| 17:51.29 | darwin_35 | just doing as told |
| 17:51.32 | Hmmhesays | :D |
| 17:51.40 | darwin_35 | no I need this job |
| 17:52.04 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:52.11 | brad_mssw | Hmmhesays: he's with teliax ... |
| 17:52.30 | [TK]D-Fender | jbalcomb : sure |
| 17:52.35 | darwin_35 | yes yes iam |
| 17:52.36 | *** join/#asterisk dijit0 (n=eric@69.106.49.200) |
| 17:52.46 | jbalcomb | you start it please, i dont know how. :/ |
| 17:54.03 | Hmmhesays | where did you apply for that job darwin_35? |
| 17:54.41 | *** part/#asterisk qazwsx (n=qazwsx@201.11.108.54) |
| 17:55.48 | darwin_35 | on our website |
| 17:56.06 | darwin_35 | send a resume to daldworth@teliax.com |
| 17:56.17 | darwin_35 | but you cant have my job |
| 17:56.30 | file | unless you're of more value. |
| 17:56.33 | darwin_35 | but you could be a phone and ticket jockey |
| 17:56.40 | _Sam-- | lol |
| 17:56.44 | SERGEUS | SIP behaviour is very strange |
| 17:56.48 | twisted[asteria] | hah |
| 17:57.02 | SERGEUS | it brakes all calls after 2nd ring |
| 17:57.07 | file | which wouldn't take... er nevermind |
| 17:57.08 | Hmmhesays | darwin_35: i bet I probably could |
| 17:57.09 | _Sam-- | darwin_35: youhave a boss? |
| 17:57.15 | twisted[asteria] | SERGEUS, that sounds like a device |
| 17:57.20 | darwin_35 | yes |
| 17:57.29 | SERGEUS | when i enabled "sip debug" - it started work normaly |
| 17:57.31 | twisted[asteria] | SERGEUS, or a 10 second timer on dial |
| 17:57.35 | SERGEUS | weired |
| 17:57.38 | twisted[asteria] | oh |
| 17:57.40 | darwin_35 | I am the noc/phone/ticket/what ever support right now |
| 17:57.55 | SERGEUS | twisted wait a second |
| 17:58.10 | darwin_35 | the only big issue I am still miffed about is the mysql/asterisk mem leak |
| 17:58.21 | darwin_35 | in realtime |
| 17:58.22 | _Sam-- | i thought you were a sales guy / owner guy, you work hard that hard for someone else? |
| 17:58.41 | file | so track down the leak and fix it |
| 17:58.45 | darwin_35 | no thats David |
| 17:58.50 | dijit0 | if anyone can help, what do i need to set up on my router and asterisk to allow idefisk to connect to asterisk OVER the internet?? cause it registers fine when i set the address to the local network, but i want to be able to connect through my public IP |
| 17:58.51 | _Sam-- | i thought you were david, my fault |
| 17:58.54 | Hmmhesays | teliax website doesn't have much info about using asterisk as an origination point |
| 17:58.55 | SpaceBass | can someone take a look at my zapta.conf and see if they see an issue: http://pastebin.ca/38389 |
| 17:59.00 | Hmmhesays | which I find odd |
| 17:59.03 | darwin_35 | he adn a few others own I am buying into it slowly |
| 17:59.13 | _Sam-- | Hmmhesays: you dont need much...just two lines in extensions.conf and iax/sip.conf |
| 17:59.18 | SpaceBass | basically it works fine until I get a call on zap2, then all calls on zap1 appear as if from zap2 |
| 17:59.26 | Hmmhesays | _Sam-- i know |
| 18:00.35 | *** join/#asterisk ]Louise[ (n=S_E_L_i_@85.102.157.233) |
| 18:00.38 | Hmmhesays | does teliax have any voipjet style service, where they're just the termination provider? |
| 18:00.40 | SERGEUS | twisted[asteria], if you interested: http://pastebin.ca/38390 |
| 18:01.09 | _Sam-- | Hmmhesays: im sure based on the number of minutes you are using that guy david from teliax would make you a termination deal |
| 18:01.16 | _Sam-- | its all about the minutes |
| 18:01.35 | Hmmhesays | not many minutes, just looking for a good backup provider |
| 18:01.52 | _Sam-- | i use iax.cc as a backup to teliax (www.iax.cc) |
| 18:01.54 | _Sam-- | cheap termination |
| 18:01.57 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.197) |
| 18:02.06 | AgiNamu | Anyone know how to make asterisk send a SETUP ACKNOWLEDGE ? |
| 18:02.17 | SERGEUS | twisted[asteria], description: ii called myself via voxbone, phone start ringing, after second ring i've heard a "busy" sound and phone stoped ringing |
| 18:02.20 | Hmmhesays | as we know voipjet goes up and down more than a couple horny teenagers on prom night |
| 18:02.28 | AgiNamu | lol |
| 18:02.35 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:02.40 | SERGEUS | i'll try to reproduce this behaviour... |
| 18:03.08 | brad_mssw | Hmmhesays: yeah, you don't have to get a phone number with teliax ... |
| 18:03.27 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-212-180-41.mia.bellsouth.net) |
| 18:03.29 | Hmmhesays | looking for a per minute charge, not buying a block of minutes |
| 18:03.36 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-212-180-41.mia.bellsouth.net) |
| 18:03.36 | _Sam-- | you could probably also use broadvoice if all you need is termination |
| 18:03.43 | brad_mssw | Hmmhesays: i've got experience with sixtel (iax.cc), teliax, and junction networks ... teliax's latency is too high, sixtel isn't too reliable ... junction networks is pricy |
| 18:03.51 | twisted[asteria] | SERGEUS, what is the output of 'show version' on the CLI? |
| 18:03.54 | SpaceBass | arrruugg broadvoice is a bad word around my house right now |
| 18:04.01 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 18:04.21 | _Sam-- | brad_mssw: that is probably not accurate info, since its only accurate for YOUR routes and backbone provider. |
| 18:04.28 | SERGEUS | twisted[asteria], Asterisk SVN-trunk-r8447M |
| 18:04.30 | brad_mssw | _Sam--: that is correct |
| 18:04.33 | _Sam-- | my routes to teliax are perfect, iax.cc has never not one not worked... |
| 18:04.34 | Hmmhesays | brad_mssw: yeah you do get what you pay for |
| 18:05.33 | brad_mssw | Hmmhesays: yep, junction seems good, just pricy ... using them for all outbound termination right now |
| 18:05.54 | brad_mssw | Hmmhesays: teliax is lagging at 70-80ms unless you're on cogent |
| 18:06.13 | justinu | hey brad, i did actually get a response from junction the other day |
| 18:06.17 | brad_mssw | sixtel, usually pretty good, but loses sync |
| 18:06.27 | Hmmhesays | 70-80ms isn't bad |
| 18:06.29 | darwin_35 | send us a traceroute of that pls and what network your on .. |
| 18:06.37 | brad_mssw | Hmmhesays: not fax capable at that latency |
| 18:06.41 | darwin_35 | support@teliax.com |
| 18:06.49 | brad_mssw | darwin_35: been there, done that ... multiple times |
| 18:06.54 | Netgeeks | Hey areski, you here right now? |
| 18:06.55 | Hmmhesays | Good thing I don't fax over ip |
| 18:07.06 | _Sam-- | there has been packet loss at teliax's connection to rockynet |
| 18:07.10 | _Sam-- | 1-3% yesterday |
| 18:07.12 | brad_mssw | Hmmhesays: yeah, it's not bad for voice ... though it does drop out worse from time to time |
| 18:07.28 | darwin_35 | I have forwarded them all thus far to rockynet for further routing repair |
| 18:07.36 | twisted[asteria] | SERGEUS, are you sure it works fine with sip debug enabled? |
| 18:08.05 | brad_mssw | darwin_35: been a couple of weeks since my first complaint ... routes were fine until 3 weeks ago |
| 18:08.09 | SpaceBass | Anyone know why * would attempt a bridge when canreinvite=no is set at the trunk and device? |
| 18:08.32 | Math` | SpaceBass: a native bridge and reinviting is two different things |
| 18:08.38 | brad_mssw | _Sam--: yeah, exactly, latency and packetloss, neither of which are good for voip :/ |
| 18:08.53 | SERGEUS | yes, however i can't reproduce it now - so probably it was my fault, but i can PM a full log to you - it started to work after i called "sip debug ip ...." |
| 18:09.00 | areski | Netgeeks, Hi, yes a part of me is here |
| 18:09.04 | AgiNamu | Does anyoen know if there's an easy way to substitute your own generic bridge? |
| 18:09.12 | SpaceBass | Math`, maybe I'm misunderstanding the two then... I'm having problms with no audio on my sip trunks for devices with no internet access |
| 18:09.15 | areski | Netgeeks, how u doing ? |
| 18:09.33 | brad_mssw | anyone have anything to say about nufone ? |
| 18:09.38 | _Sam-- | brad_mssw: if you ping/traceroute/mtr/whatever to teliax are you seeing the packet loss also at the last hop? (may need to let it run for a few) |
| 18:09.43 | SpaceBass | IE phones on a subnet that route back to my * box... so when it attempts the bridge (at least thats the point on the CLI) the call connects but I have no audio |
| 18:10.02 | Math` | SpaceBass: usually asterisks breaks down the RTP stream into frames to be processed, a native bridge is when it just proxies the RTP stream |
| 18:10.09 | tronix | brad_mssw: i haven't config'd my account with nufone yet -- still working on basic * setup but they're pretty quick on tech support and inexpensive. no idea on call quality yet |
| 18:10.11 | SpaceBass | brad_mssw, I've played with nufone... seems ok |
| 18:10.45 | tronix | they did provision my 800 toll-free DID on the spot, tho |
| 18:10.58 | SpaceBass | Math`, so in my case I want a native bridge...* talkes to the sip trunk and my device talks to * is that correct? |
| 18:11.01 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
| 18:11.08 | brad_mssw | _Sam--: not getting packet loss right now it seems |
| 18:11.15 | brad_mssw | _Sam--: just high latency |
| 18:11.27 | _Sam-- | im seeing it from different hosts/nets |
| 18:11.32 | Math` | SpaceBass: if you have the same codec, the native bridge is automatic |
| 18:11.46 | brad_mssw | _Sam--: what utility are you using to measure packetloss ? |
| 18:11.53 | _Sam-- | im using mtr |
| 18:12.13 | SpaceBass | Math`, if could be a codec issue... but I never had this problem before... just appeared when I re-installed AAH |
| 18:12.20 | SpaceBass | setup is virtually the same |
| 18:12.26 | brad_mssw | _Sam--: let me install that real fast and take a look |
| 18:12.33 | Math` | SpaceBass: whats the problem? no audio on one side? |
| 18:12.39 | _Sam-- | http://www.bitwizard.nl/mtr/ (dont know if that is a current URL) |
| 18:13.06 | brad_mssw | _Sam--: already installed man :) |
| 18:13.17 | _Sam-- | decent tool |
| 18:13.20 | brad_mssw | _Sam--: no loss yet ... only up to 35 attempts |
| 18:13.29 | SpaceBass | Math`, no audio on both sides when using my wifi sip phones...which are on a segregated wifi subnet with DMZ pin holes to my * box (calls to other sip devices and over pstn work fine) |
| 18:13.32 | _Sam-- | im at 1.9% for 402 packets |
| 18:13.44 | _Sam-- | are these your last two hops? |
| 18:13.44 | _Sam-- | 12. f-5-0-0-cd2.rockynet.com 1.5% 402 63.1 65.0 57.5 292.5 24.6 |
| 18:13.44 | _Sam-- | 13. voip-co2.teliax.com 1.7% 402 62.1 62.8 60.1 83.6 2.1 |
| 18:13.50 | Math` | SpaceBass: try setting nat=yes in the SIP config for those phones |
| 18:13.59 | robin_z | ok, something is very wrong with my music on hold .. it sounds very broken ... like digital crap |
| 18:14.02 | SpaceBass | Math`, it is set to yes |
| 18:14.03 | robin_z | clues? |
| 18:14.06 | brad_mssw | _Sam--: yep, same last 2 |
| 18:14.16 | rajiv|work | anyone use sellvoip.net, gizmo, or sipphone for origination ? |
| 18:14.16 | Math` | robin_z: get the good mpg123 version |
| 18:14.23 | SpaceBass | Math`, take that back... was not set to yes for that device... let me try that |
| 18:14.26 | robin_z | ahh. good being? |
| 18:14.30 | _Sam-- | restarted the mtr...its 0% now for 50 |
| 18:14.37 | twisted[asteria] | SERGEUS, hmm... strange. sounds more like a coincidence, i just read through chan_sip.c to make sure that we don't have any code blocked out by debugging |
| 18:14.43 | rajiv|work | _Sam--: press J in mtr to see jitter |
| 18:14.46 | twisted[asteria] | anywho |
| 18:14.48 | brad_mssw | _Sam--: wait, got packet loss |
| 18:14.48 | twisted[asteria] | lunch time |
| 18:14.57 | brad_mssw | _Sam--: 0.8% |
| 18:14.59 | _Sam-- | rajiv: THANKS. |
| 18:15.04 | brad_mssw | _Sam--: last 2 hops too |
| 18:15.05 | _Sam-- | never knew the J |
| 18:15.05 | SERGEUS | twisted[asteria], sorry for a false alarm :) |
| 18:15.20 | *** join/#asterisk ToTo (n=ToTo@host225-87.pool8256.interbusiness.it) |
| 18:15.21 | *** join/#asterisk Lurr_ (n=pr0ph3t@adsl-212-180-41.mia.bellsouth.net) |
| 18:15.21 | brad_mssw | _Sam--: err, bad, got a few other hops with some loss too |
| 18:15.23 | twisted[asteria] | SERGEUS, it's okay, better to be safe than sorry |
| 18:15.25 | *** part/#asterisk Lurr_ (n=pr0ph3t@adsl-212-180-41.mia.bellsouth.net) |
| 18:15.36 | rajiv|work | _Sam--: it's not the same jitter that you see in * iax channels but still useful |
| 18:15.43 | SpaceBass | Math`, thanks for the tip, but nat=yes had no effect |
| 18:16.15 | brad_mssw | _Sam--: wow, mtr is pretty neat, showing the different routes it takes on multiple attempts |
| 18:16.32 | robin_z | Math`: im on debian runn mpg321 ... is this the problem? |
| 18:16.40 | Math` | it is |
| 18:16.48 | Math` | apt-get remove mpg321 |
| 18:16.50 | darwin_35 | t |
| 18:16.53 | robin_z | sigh .... is there an apt-gettable thing that will work? |
| 18:16.53 | _Sam-- | mpg123 |
| 18:16.56 | Math` | then: make mpg123 into *'s source tree |
| 18:17.07 | _Sam-- | apt-get mpg123 |
| 18:17.09 | Math` | no |
| 18:17.18 | _Sam-- | i apt-got my mpg123 on debian |
| 18:17.23 | Math` | apt-get'ing mpg123 installs mpg321 ;) |
| 18:17.26 | robin_z | _Sam-- no, you didnt |
| 18:17.35 | _Sam-- | mpg123 - MPEG layer 1/2/3 audio player |
| 18:17.36 | _Sam-- | bs |
| 18:17.41 | _Sam-- | its right there in my apt-cache search |
| 18:17.46 | _Sam-- | mpg123 - MPEG layer 1/2/3 audio player |
| 18:17.46 | _Sam-- | mpg123-esd - MPEG layer 1/2/3 audio player with Esound support |
| 18:17.46 | _Sam-- | mpg123-nas - MPEG layer 1/2/3 audio player with NAS support |
| 18:17.46 | _Sam-- | mpg123-oss-3dnow - MPEG layer 1/2/3 audio player for 3DNow! machines |
| 18:17.46 | _Sam-- | mpg123-oss-i486 - MPEG layer 1/2/3 audio player for i486 machines |
| 18:17.47 | _Sam-- | mpg123-el - a front-end program to mpg123 audio player on Emacsen |
| 18:17.49 | Math` | you probably have a 3rd party source giving you that |
| 18:17.53 | _Sam-- | of course i do |
| 18:17.57 | robin_z | OK, where your .deb source for that? |
| 18:18.00 | _Sam-- | i have a zillion sources |
| 18:18.00 | malverian[work] | Is there a builtin function in Asterisk for converting a string date to unix time? |
| 18:18.14 | Math` | it takes less time to compile it than to update the source list |
| 18:18.29 | robin_z | first you have to find it ... |
| 18:18.31 | Math` | robin_z: "make mpg123" in asterisk's directory... |
| 18:18.37 | robin_z | thats it? |
| 18:18.41 | Math` | its gonna download it and compile it |
| 18:18.42 | Math` | yeah |
| 18:18.44 | darwin_35 | dont use mpg123 use madplayer |
| 18:19.00 | *** join/#asterisk DShepherd (n=DShepher@port0002-abm-adsl.cwjamaica.com) |
| 18:19.04 | DShepherd | hey |
| 18:19.37 | _Sam-- | thanks for the tip... i never knew make mpg123 would download and install it |
| 18:20.18 | DShepherd | my jug is doing something on asterisk can I use the logo on my poster? |
| 18:20.25 | DShepherd | jog =lug |
| 18:20.28 | DShepherd | lug* |
| 18:20.32 | SpaceBass | Math`, it looks like both the device and broadvoice (sip trunk) use the same codec, but still no audio with nat=yes |
| 18:20.56 | tzanger | DShepherd: generally that isn't an issue if it's noncommercial, but contact digium, they own the trademark |
| 18:21.14 | DShepherd | tzanger, ok thanks |
| 18:21.16 | *** join/#asterisk gongoputch (n=gongoput@pcp01486721pcs.limstn01.de.comcast.net) |
| 18:21.37 | justinu | SpaceBass: are you getting rtp packets from broadvoice? |
| 18:22.04 | SpaceBass | on my router I have udp 5060 and tcp 10,000-20,000 forwarded b/t my wifi sip phone and * box |
| 18:22.19 | SpaceBass | on my edge router I have the same ports forwared to the * box from the outside |
| 18:22.37 | SpaceBass | justinu, I know I;m getting UDP to the * from BV b/c my decices on the same lan as my * box work fine |
| 18:23.01 | justinu | spacebass: use rtp debug, or ethereal to verify |
| 18:23.28 | justinu | also look at the sip SDP, and make sure the IPs look sane... i can take a look at the pasts if you need help interpreting it |
| 18:23.38 | SpaceBass | justinu, not too comfortable with etheeral yet, didnt know about RTP debug... trying it now |
| 18:23.56 | *** join/#asterisk ukh (n=ukh@ibook-wifi.svansen.se) |
| 18:24.22 | Hmmhesays | SpaceBass: yes that is grand |
| 18:24.50 | SpaceBass | ok, enabled RTP debug and made a call, didn't see anything debug related on the CLI |
| 18:25.00 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 18:25.11 | Hmmhesays | what problem are you having? |
| 18:25.39 | justinu | spacebass: ok, that's a problem. |
| 18:25.56 | justinu | spacebass: turn on sip debug, and paste the invite/200ok messages to pastebin |
| 18:26.07 | SpaceBass | no audio at all on my wifi sip devices when using my sip trunk - they are on a seperate wireless subnet with dmz pin holes for 5060 and rtp back to my * box....pstn and calls to other devices work fine |
| 18:26.36 | darwin_35 | unagi |
| 18:26.40 | *** part/#asterisk DShepherd (n=DShepher@port0002-abm-adsl.cwjamaica.com) |
| 18:26.42 | darwin_35 | sushi |
| 18:26.42 | justinu | it's probably a stupid nat issue |
| 18:26.48 | darwin_35 | nat |
| 18:26.49 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 18:27.11 | darwin_35 | nat = small bug that infest plants and networks |
| 18:27.24 | iDunno | that's a *g*nat. |
| 18:27.58 | [TK]D-Fender | the "g" is silent like the "p" in swimming :D |
| 18:28.00 | SpaceBass | http://pastebin.ca/38397 |
| 18:28.20 | SpaceBass | suspect it could be a nat issue... stated when I replaced my asterisk box....same general topography worked previously |
| 18:29.00 | *** join/#asterisk benjk (n=benjamin@70.103.248.130) |
| 18:29.26 | Hmmhesays | i should cook a steak tonight |
| 18:29.37 | Hmmhesays | yes yes I should |
| 18:29.39 | SpaceBass | me cooked a flat iron steak last night....quite good |
| 18:29.47 | Hmmhesays | or maybe a hobo meal |
| 18:29.55 | Hmmhesays | all wrapped up in tinfoil n shit |
| 18:30.21 | stack_ | Is it work the ~$200 to get echo cancellation on the TDM2400P? |
| 18:30.22 | justinu | SpaceBass: invite looks ok, but I didn't get the 200 OK ack from your ast box |
| 18:30.34 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 18:30.41 | SpaceBass | justinu, let me try again... maybe didnt copy it all |
| 18:31.16 | *** join/#asterisk meriad (i=mreith@unaffiliated/meriad) |
| 18:31.21 | meriad | VOIP BABY! |
| 18:31.33 | meriad | ]Louise[- where do i get free pr0n vids again? |
| 18:31.49 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-133.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 18:31.58 | *** join/#asterisk FastJack (i=fastjack@p5091FDC7.dip.t-dialin.net) |
| 18:32.00 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-133.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 18:32.03 | *** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee) |
| 18:32.25 | SpaceBass | justinu, not entirely sure about the sip messages so I'm not sure if I'm capturing the right ones: http://pastebin.ca/38399 |
| 18:33.19 | justinu | k, looking |
| 18:33.28 | meriad | hey, first time asterisks user just wodnering if there are any detailed guides explaining VOIP, and what not, and imsetting iot up on my box.. Debia.. etc.. |
| 18:33.38 | fugitivo | ~docs |
| 18:33.42 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:34.04 | SpaceBass | fyi 10.1.0.x is my lan subnet and 10.1.1.x is my wifi subnet 71..... is my public IP |
| 18:34.58 | *** part/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 18:35.02 | SpaceBass | justinu, interesting... I'm noticing something... it says : INVITE sip:5002@10.1.0.77:5060 SIP/2.0 |
| 18:35.15 | SpaceBass | but my device is 10.1.1.77 IE the 3rd octet is wrong |
| 18:35.17 | justinu | SpaceBass: ok, take a look at line 596 |
| 18:35.28 | justinu | here's the real problem |
| 18:35.38 | Zodiacal | anyone know if its posible for a cisco 7960 phone to show if someone has there phone in use? not nessiarly which phone line they are one, but just if an extention is off hook? |
| 18:35.47 | justinu | that whole message is your ast box telling broadvoice that you answered the inbound call |
| 18:36.03 | justinu | check line 618 |
| 18:36.11 | *** join/#asterisk loick (n=loick@APuteaux-151-1-13-40.w82-120.abo.wanadoo.fr) |
| 18:36.16 | justinu | that's your ast box telling broadvoice to send the RTP to IP 10.1.0.40 |
| 18:36.20 | justinu | which broadvoice obviously can't do |
| 18:36.23 | SpaceBass | the c=in ip4 ? |
| 18:36.31 | justinu | check in your sip.conf and make sure externip=<your external ip> |
| 18:36.52 | SpaceBass | justinu, I am fairly sure I do NOT have that set... let me do that |
| 18:37.10 | justinu | also make sure localnet=10.0.0.0/255.0.0.0 |
| 18:37.15 | SpaceBass | justinu, can I add that in the settings for the trunk? |
| 18:37.34 | [TK]D-Fender | SpaceBass : Verify your netmask <---- |
| 18:37.36 | justinu | no, it's global |
| 18:38.08 | SpaceBass | I'm using class C internally... 255.255.255.0 |
| 18:38.20 | robin_z | in a recent survey of men about what they liked best in a blow-job ... 8% said the physical sensation, 5% said the feeling of domination .. and ... |
| 18:38.29 | justinu | doesn't matter, because ast needs to consider all 10.0.0.0 local to it |
| 18:38.30 | robin_z | 77% said "25 minutes of peace and quiet" |
| 18:38.53 | robin_z | ;) |
| 18:38.53 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 18:40.10 | [TK]D-Fender | robin_z : about the %77 - 95% lied about their duration, and the other 5% can chalk it up to incompetance :D |
| 18:40.28 | Hmmhesays | 25 minute bj? sweet geebus |
| 18:40.33 | Hmmhesays | i would have long gotten bored |
| 18:40.38 | Hmmhesays | started watching tv or something |
| 18:40.58 | }btorch{ | I'm trying to setup iax channels on my iax.conf file and on my sip.conf I always used [<phone number>] ... now I'm setting up as [<first name>] , how can I assign an extension to that number? |
| 18:42.16 | SpaceBass | justinu, tried those lines in sip_nat.conf ... didnt work...no adding to sip.conf in the [general] context |
| 18:42.21 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 18:42.28 | }btorch{ | I want someone to be able to call say 1555 and then the macro that I have created will grab that {$EXTEN} and dial it |
| 18:42.32 | justinu | SpaceBass: eh? |
| 18:43.15 | SpaceBass | justinu, added those lines in sip.conf (externip and localnet) and it didn't fix it |
| 18:43.18 | SpaceBass | checking the debug now |
| 18:43.35 | justinu | ok, ast shouldn't be replying to broadvoice with your internal IPs |
| 18:43.41 | justinu | try restarting ast completely |
| 18:43.55 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:43.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:44.04 | SpaceBass | restart gracefully |
| 18:44.09 | SpaceBass | or do I need a reboot? |
| 18:44.13 | justinu | no reboot |
| 18:45.02 | SpaceBass | what was the message header you looked at to determine what IP my ast box was sending to BV? |
| 18:45.15 | justinu | it's in the 200ok |
| 18:45.26 | justinu | c=IN IP4 |
| 18:46.02 | *** join/#asterisk Zach^^ (i=chaos@dialup-4.224.213.107.Dial1.Cincinnati1.Level3.net) |
| 18:46.03 | Zach^^ | i am trying to install amp and i get this error http://pastebin.com/520964 |
| 18:46.10 | fugitivo | ~amp |
| 18:46.11 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 18:46.51 | Zach^^ | fugitivo there is noone in amp alive |
| 18:47.03 | fugitivo | not our problem |
| 18:47.25 | justinu | lol |
| 18:47.38 | SpaceBass | at least the script doesnt read: people using it should go to hell" |
| 18:48.02 | Math` | then go to #hell |
| 18:48.04 | Math` | :P |
| 18:48.08 | fugitivo | hehe |
| 18:48.28 | *** join/#asterisk Coccyx (n=clint@typhoon.org) |
| 18:48.29 | *** join/#asterisk Qwell[laptop] (n=chatzill@70.103.248.130) |
| 18:48.35 | SpaceBass | lol |
| 18:49.34 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 18:50.03 | [TK]D-Fender | SpaceBass : No.. the trip to hell is bundled with the software :) |
| 18:50.30 | SpaceBass | LOL |
| 18:50.56 | SpaceBass | ARRRUUUGGGG this is still broken |
| 18:52.15 | justinu | SpaceBass: paste the sip debug again |
| 18:52.34 | rajiv|work | justinu: i think you mean "pastebin ... again" |
| 18:52.45 | justinu | he knows what I mean :P |
| 18:52.58 | SpaceBass | the latest: http://pastebin.com/520983 |
| 18:53.17 | SpaceBass | oh you want me to paste it in the channel...ok...here goes |
| 18:53.17 | SpaceBass | :) |
| 18:53.25 | justinu | ok, well, there's certainly an improvement here |
| 18:53.30 | justinu | you're telling BV the right IP now |
| 18:53.55 | SpaceBass | I hit reply on on a entire business unit email the other day... responded to like 2k people... first and last time I make that mistake... I hate when people don;t send those as a bcc |
| 18:54.07 | justinu | lol |
| 18:54.13 | tzanger | SpaceBass: don't reply all :-) |
| 18:54.18 | justinu | SpaceBass: try RTP debug now |
| 18:54.21 | justinu | see what happens |
| 18:54.29 | SpaceBass | line 195 |
| 18:54.33 | SpaceBass | shows local ip |
| 18:54.41 | denon | SpaceBass: it's because of people like you, that we limit recipient counts to 20 on outbound SMTPs |
| 18:54.44 | denon | :) |
| 18:54.46 | justinu | yeah, that's an INVITE to your internal phone |
| 18:54.53 | denon | I mean, really -- that's what list servers are for |
| 18:55.11 | cpm | [TK]D-Fender, I gotta question for ya, err, If I brought dial-up in to my asterisk box, can asterisk handle routing (using ani data) data calls to my NxT1 dialup servers? So I could do TDMoE ? is this reasonable? |
| 18:55.20 | denon | SpaceBass: we expect nothing less from an MCSE <G> |
| 18:55.30 | justinu | MCSE? lol |
| 18:55.32 | justinu | sorry |
| 18:55.45 | [TK]D-Fender | cpm : uhhh.... not a clue. |
| 18:55.46 | justinu | i ouldn't admit that in this channel |
| 18:55.49 | justinu | wouldn't |
| 18:56.00 | cpm | thanks. |
| 18:56.28 | SpaceBass | lol |
| 18:56.29 | cyburdine | anyone know how to turn echo cancelation ON on a sip channel? is that even possible? |
| 18:56.29 | denon | s/in this channel/anywhre but on a resume to a clueless management type/ |
| 18:56.31 | cpm | It's a wierd question. Rather than having T1s for voip and T1s for in-dial data calls, could munge' |
| 18:56.41 | fugitivo | cyburdine: NO |
| 18:56.42 | [TK]D-Fender | cyburdine : Nope |
| 18:56.50 | cpm | em all together that way. if it works |
| 18:56.50 | fugitivo | cyburdine: you shouldn't have echo on a sip channel |
| 18:56.54 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:57.07 | SpaceBass | it was a worthless cert and has nothing to do with my current job... but it makes my friends think i know what I am talking about |
| 18:57.11 | [TK]D-Fender | echo is supposed to be compensated for at the PSTN end. |
| 18:57.13 | justinu | cpm: theoretically you should be able to do what you want to do |
| 18:57.20 | justinu | SpaceBass: what's happening with rtp debug now? |
| 18:57.50 | SpaceBass | justinu, call not going through at all...not related to RTP debug...not sure what is happening |
| 18:57.52 | cyburdine | hmm |
| 18:58.00 | jpablo | snom phones are ugly, but when you use then they are actually nice. |
| 18:58.01 | cpm | justinu, thanks |
| 18:58.03 | justinu | SpaceBass: tell you what. run this command as root: tcpdump -s0 -w trace.cap |
| 18:58.16 | justinu | SpaceBass: make your call, then send me trace.cap |
| 18:58.23 | justinu | email in pm |
| 18:59.03 | steve___ | are they any solid cli softphones out there? |
| 19:00.33 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 19:00.36 | drumkilla | steve___: if you download iaxclient, they have testcall or testclient or something |
| 19:00.37 | *** join/#asterisk SYS64738 (n=giaco@host230-254.pool81123.interbusiness.it) |
| 19:00.39 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 19:01.01 | }btorch{ | how can I compile the cdr_pgsql module after I compiled * ? |
| 19:01.25 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 19:02.32 | Math` | }btorch{: not only you can but you should |
| 19:02.47 | *** join/#asterisk salmandr (n=salmandr@dyn-112-51.uwnet.wisc.edu) |
| 19:02.57 | steve___ | drumkilla i am using iaxcomm and outbound audio has less than one sec lag. It is very usable, but annoying. |
| 19:03.32 | *** join/#asterisk NewSole (n=dave@MTL-HSE-ppp175292.qc.sympatico.ca) |
| 19:04.10 | zoa | steve, did you try idefisk ? |
| 19:04.11 | meriad | So like soft phones are PC phones and hard phones are not :> |
| 19:04.17 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:04.21 | SpaceBass | i quite like iaxcomm on os x |
| 19:04.21 | zoa | what is the roundtrip time to your server ? |
| 19:04.29 | zoa | SpaceBass: did you try idefisk on mac ? |
| 19:04.29 | zoa | :) |
| 19:04.38 | gongoputch | anyone know if these phones will work with asterisk http://www.bsdmall.com/bsdmall/gxenipte.html ? |
| 19:04.57 | SpaceBass | zoa, not yet... but it looks slick |
| 19:05.02 | zoa | gongoputch: yes it works |
| 19:05.06 | }btorch{ | Math`: I guess it did not compiled when I originally compiled asterisk... I just went back to the src location and tried to do a make cdr_pgsql.so but I got an ASTEISK_GPL_KEY error and others |
| 19:05.11 | gongoputch | cool. |
| 19:05.23 | Qwell[laptop] | but it's not a very good phone... |
| 19:05.25 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 19:05.34 | gongoputch | is FreeBSD a common choice for asterisk? |
| 19:05.41 | gongoputch | Qwell[laptop]: why? |
| 19:05.49 | Qwell[laptop] | gongoputch: it just isn't great |
| 19:05.58 | Qwell[laptop] | There is a reason it's < $80 |
| 19:06.01 | gongoputch | Qwell[laptop]: sound quality? |
| 19:06.08 | Qwell[laptop] | overall |
| 19:06.32 | gongoputch | I am looking to get my feet wet, and the price point seemed good |
| 19:06.52 | gongoputch | but, if it is broken, that won't help me learn. |
| 19:06.56 | *** join/#asterisk jkitchen (n=kitchen@ca-yorbalnd-cuda2-c1a-157.anhmca.adelphia.net) |
| 19:07.02 | jkitchen | howdy folks |
| 19:07.02 | brad_mssw | gongoputch: echo issues |
| 19:07.03 | Qwell[laptop] | Take a look at the SPA-941...I hear those are good |
| 19:07.06 | Qwell[laptop] | fairly cheap too |
| 19:07.17 | gongoputch | ah, I will google |
| 19:07.31 | brad_mssw | gongoputch: either go for a 941 linksys, or just go with a sipura ata and a regular phone |
| 19:07.34 | gongoputch | echo can be a real bugger |
| 19:07.45 | pifiu-laptop | hey qwell wasup |
| 19:08.07 | gongoputch | like that http://www.bsdmall.com/bsdmall/linksysspa941.html ? |
| 19:08.13 | Qwell[laptop] | yes |
| 19:08.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:08.20 | brad_mssw | gongoputch: yes |
| 19:08.21 | *** join/#asterisk EriSan (n=erisan@81-174-25-141.f5.ngi.it) |
| 19:08.27 | FuriousGeorge | i got two issues with asterisk no one has been able to figure out. first of all, #-transfer isnt working, despite everything being set up right |
| 19:08.39 | [TK]D-Fender | I would vote against the 941 these days if you don't get it a fair bit cheaper than $150.... |
| 19:08.41 | *** join/#asterisk NSGN (n=brandonb@cpe-66-69-197-25.austin.res.rr.com) |
| 19:08.44 | NSGN | hello all |
| 19:08.49 | Qwell[laptop] | [TK]D-Fender: Are they that much? |
| 19:08.54 | Qwell[laptop] | I was thinking $120 or so |
| 19:09.15 | jkitchen | are there any issues with meetme and IAX? I have an IAX feed from junction networks and for some reason when I call in via IAX I can't speak on the conference, but my sip clients on the local network can |
| 19:09.19 | NSGN | so what is the state of free services as far as calls to and from the internet and the PSTN? |
| 19:09.27 | [TK]D-Fender | Qwell[laptop] : $150 pretty much everywhere I've seen. If you've got a link, I;m interested. |
| 19:09.32 | jkitchen | with the same configuration files and asterisk 1.0 it all works, but I need the recording features of 1.2 |
| 19:09.36 | Qwell[laptop] | nope, never looked...just assumed |
| 19:09.40 | gongoputch | $140 on that link I posted |
| 19:09.42 | NSGN | i've heard some people talk about free local pstn gateways and such |
| 19:09.51 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 19:10.03 | [TK]D-Fender | With Polycom IP 501 @ $170 I'd sooner go with that. |
| 19:10.17 | gongoputch | [TK]D-Fender: why? |
| 19:10.18 | NSGN | was i mistaken in what they were talking about or are there just not very many? |
| 19:10.40 | justinu | we've had this discussion many many times on this channel |
| 19:10.46 | justinu | fender should put together a page on the wiki |
| 19:10.53 | FuriousGeorge | the other issue is that ive got 5 srevers all logged into eachother via iax. theyre dynamic so i use a dynamic ip to dns service, and it works great except one box that cant log into another for more than a few minutes a day |
| 19:10.57 | justinu | why the polycom501 kicks ass |
| 19:11.22 | gongoputch | I am sorry for adding to the noise. |
| 19:11.23 | FuriousGeorge | justinu: does it have snazzy led's like the snom? |
| 19:11.29 | brad_mssw | jkitchen: timer issue? meetme required zap or ztdummy if i remember correctly somewhere |
| 19:11.40 | *** join/#asterisk hans (n=fugalh@dhcp25.cs.nmsu.edu) |
| 19:11.41 | justinu | FuriousGeorge: no, not the 501 |
| 19:11.50 | NSGN | ....anyone? |
| 19:12.29 | Netgeeks | snazzy leds? is that an engineering term? |
| 19:12.49 | [TK]D-Fender | gongoputch : IP 501 has better speakerphone, contact list capabilities, PoE optional, 2 eth ports, 10/100, slightly better feel, better speakerphone. better use of screen. Thats to start :) |
| 19:13.13 | gongoputch | 2 ethernet ports? why? |
| 19:13.14 | FuriousGeorge | so no one knows why my #-xfers dont work? before i upgraded to 1.2 thye worked so well that if someone in the room was dialing on speakerphone and hit # it would ask me to xfer |
| 19:13.23 | justinu | for stations with only one lan drop |
| 19:13.26 | gongoputch | (I am looking at pix now) |
| 19:13.26 | [TK]D-Fender | gongoputch : For plugging in-line with a PC if need be. |
| 19:13.35 | gongoputch | neat. |
| 19:13.47 | gongoputch | didn't know you could do that |
| 19:13.52 | [TK]D-Fender | gongoputch : Save you having to run another path to your switch |
| 19:14.13 | gongoputch | how do they manage that? |
| 19:14.26 | [TK]D-Fender | Linksys had a chance with the SPA-941, but with Atacomm's agressive Polycom pricing, I just can't find a place for it much these days.... |
| 19:14.26 | gongoputch | mini-switch in the device? |
| 19:14.44 | [TK]D-Fender | gongoputch : yup. Its got 2 eth ports side-by-side |
| 19:14.53 | znoG | the main problem with Linksys is their incompetent staff that are supposed to do technical support |
| 19:15.01 | justinu | yeah, it's got a vlan capable switch in the phone |
| 19:15.04 | [av]bani | [TK]D-Fender: the aastra too ... |
| 19:15.12 | [TK]D-Fender | gongoputch : I have 4 daisy-chained together in customer service right now. |
| 19:15.14 | gongoputch | I am getting kinda pricey for a side poject ... |
| 19:15.17 | [av]bani | just about everything beats the 941 now ... |
| 19:15.23 | justinu | aastra, polycom, and gxp2000 do it |
| 19:15.27 | justinu | cisco too |
| 19:15.31 | justinu | anything else? |
| 19:15.48 | [av]bani | [TK]D-Fender: linksys better get their shit together, 942 just 'aint quite it' |
| 19:15.58 | [TK]D-Fender | [av]bani : True, but not at the 941's price-point. The IP 501 is so close as to challenge it and in most cases win. |
| 19:16.06 | gongoputch | would a pair of the spa941s do for a VOIP lab ? |
| 19:16.16 | robin_z | polycom is OK, but still a bit tacky |
| 19:16.21 | [av]bani | [TK]D-Fender: aastra, polycom, and snom all come in at nearly the same price point and kill the 941 |
| 19:16.30 | [TK]D-Fender | 942 = waste <- devalidates the SPA line for me for anything but home (922/921 would be suggestable perhaps) |
| 19:16.43 | robin_z | Snom 360s are 8nice* |
| 19:17.13 | gongoputch | but those seem ~ $200 |
| 19:17.23 | robin_z | and some ... |
| 19:17.27 | robin_z | but, they are NICE |
| 19:17.40 | gongoputch | I just wanna do some 'proof of concepts' |
| 19:17.50 | robin_z | grandstream then |
| 19:17.54 | robin_z | cheap but shit. |
| 19:17.59 | [av]bani | [TK]D-Fender: 942 is 'too little too late' |
| 19:18.03 | gongoputch | and work up to 'working prototypes' |
| 19:18.12 | robin_z | eh? |
| 19:18.16 | [av]bani | they coulda made it do somuch more, in firmware. they chose not to. |
| 19:18.23 | robin_z | whats wrong withthe standard development model ... |
| 19:18.30 | robin_z | if it compiles, ship it? |
| 19:18.41 | cyburdine | sorry for the delay in response.. but back to echo cancellation might there be a way to simulate it? |
| 19:18.43 | cyburdine | <PROTECTED> |
| 19:18.43 | gongoputch | robin_z: would those be a better choice than the 941 for me |
| 19:19.13 | [TK]D-Fender | <[av]bani> [TK]D-Fender: 942 is 'too little too late' <- agreed. |
| 19:19.22 | robin_z | gongoputch: the GXP2000 is cheap and tatty for "real" use, but is the cheapest going pretty much. fine for just calling and stuff |
| 19:19.22 | *** join/#asterisk baltaruiz (n=baltarui@201.145.93.243) |
| 19:19.35 | jbalcomb | [TK]D-Fender the tar is on its way |
| 19:19.36 | cyburdine | we can probably write something on our own... but it would be awesome to be able to call somthing up in asterisk |
| 19:19.59 | robin_z | I fear * now needs a Skype channel like NOW |
| 19:20.11 | [TK]D-Fender | gongoputch : Serious question : after this "proof of concept" what would you final implementation be like? How many phones? What would the people use them for ? (everyone need speakerphone?) |
| 19:20.17 | [TK]D-Fender | jbalcomb : k |
| 19:20.20 | NSGN | maybe i should condnse my question. is it my imagination, or is it possible to find a free gateway to make calls from the internet to the PSTN? |
| 19:20.31 | robin_z | NSGN: no. |
| 19:20.33 | zoa | robin_z: give me 25.000 euro and i will make you one :p |
| 19:20.38 | jbalcomb | gongoputch I have 120 GXP-2000. I can't say its the phone just yet but we have lots of trouble with echo, dropped calls, etc. |
| 19:20.40 | [av]bani | [TK]D-Fender: and now they've killed the spa-3000 also. seem to be hellbent on alienating end users. |
| 19:20.41 | baltaruiz | its your imagination |
| 19:20.59 | [TK]D-Fender | [av]bani : What exactly did they do to the 3000? |
| 19:21.17 | gongoputch | <PROTECTED> |
| 19:21.17 | robin_z | gxp2000s suck harder than a cheerleader on spring break |
| 19:21.21 | [TK]D-Fender | jbalcomb : Dropped calls would likely be a T1 synch problem on your side. |
| 19:21.21 | NSGN | robin_z: thank you. there seem to be a lot that work the other way around though |
| 19:21.25 | [av]bani | [TK]D-Fender: they stopped selling them to end users |
| 19:21.45 | [av]bani | [TK]D-Fender: if you recall, i ranted about that yesterday |
| 19:21.55 | [TK]D-Fender | [av]bani : Didn't see the notice on voipsupply or atacomm yet... |
| 19:22.03 | gongoputch | <PROTECTED> |
| 19:22.25 | jbalcomb | [TK]D-Fender T1 how to diagnose a synch problem? do I need info from the telco? what config options apply? |
| 19:22.56 | [av]bani | red box |
| 19:24.21 | jbalcomb | is there a replacement/alternative for the SPA-3000? (besides the GS HandyTone) |
| 19:24.22 | [TK]D-Fender | jbalcomb : You need to check for frame slips or incremental errors on the line. Call your telco and have them monitor it for a while. |
| 19:24.35 | [TK]D-Fender | jbalcomb : My TE405P's did that like NUTS. |
| 19:24.42 | gongoputch | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44252262144.htm <- that a good phone? |
| 19:25.00 | robin_z | sigh .. integrated emergency routing .. what a daft idea. dangerous too. |
| 19:25.05 | [TK]D-Fender | gongoputch : What kind of call volume do you expect? |
| 19:25.06 | baltaruiz | whois asterisk |
| 19:25.13 | [av]bani | jbalcomb: handytone is it |
| 19:25.16 | jbalcomb | [TK]D-Fender I'm on it. |
| 19:25.18 | SpaceBass | justinu is a life saver... thanks for spending that much time troublshooting my issue! |
| 19:25.43 | robin_z | rather than trying to route emergency calls to pstn, its just safer to drop them |
| 19:25.44 | gongoputch | [TK]D-Fender: SOHO, like a dozen a day @ each office. |
| 19:25.56 | [TK]D-Fender | gongoputch : Polycoms are all very solid phone. |
| 19:26.00 | [av]bani | frame slips are almost always a clock sync problem |
| 19:26.28 | gongoputch | [TK]D-Fender: I think i will get a pair of these. |
| 19:26.31 | stack_ | Is it work the ~$200 to get echo cancellation on the TDM2400P? |
| 19:26.32 | robin_z | if you drop the emergency calls and some one dies, well, no one will ever know, and you wont be sued |
| 19:26.42 | [TK]D-Fender | gongoputch : if you are cheap you can get them ATA's @ $70 each and they can plug a regular phone in. Or from there I'd say Polycom. If they don't need speakerphone IP 301, otherwise IP 501. |
| 19:26.51 | robin_z | if you route them and screw up, someone MIGHT know ... and then you get sued. |
| 19:27.09 | jbalcomb | Regarding the Polycom: Paul Chase's view (our other phone guy) http://pastebin.com/521038 |
| 19:27.29 | jbalcomb | I'd be happy to take rebutles to my next meeting |
| 19:27.36 | gongoputch | [TK]D-Fender: thanks for the advise. |
| 19:27.42 | [TK]D-Fender | jbalcomb : That attachment doesn't look like a TAR.... |
| 19:27.56 | jbalcomb | [av]bani: we have sucky fax situation using HandyTones |
| 19:28.10 | pifiu-laptop | pastebin down for anyone? |
| 19:28.34 | cpm | works fine here |
| 19:28.45 | darwin_35 | ok if I had off a dial plan to one of you how much o move it to mysql |
| 19:28.49 | robin_z | <aol>me too</aol> |
| 19:28.50 | darwin_35 | and realtiime calls |
| 19:28.57 | [av]bani | [TK]D-Fender: one common complaint ive heard about polycom is that polycom buried regular use stuff deep in menus |
| 19:29.10 | [av]bani | [TK]D-Fender: like the complaint on that page about transfer |
| 19:29.12 | jbalcomb | [TK]D-Fender hrmm.. that is odd. i definitely did the tar, pulled it over with WinSCP and emailed it. |
| 19:29.13 | justinu | jbalcomb: that's gotta be old |
| 19:29.24 | jbalcomb | [TK]D-Fender I'll have a double check |
| 19:29.40 | gongoputch | [TK]D-Fender: could asterisk conference a bunch of these? |
| 19:29.49 | justinu | my polycom has a transfer button |
| 19:29.51 | *** join/#asterisk newmedian (n=np@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 19:29.58 | jbalcomb | justinu I am not sure. I suggested we purchase 3 polycom 501s for testing and that was his response. |
| 19:30.01 | pb__ | jbalcomb: we have sucky reliability using handytones. our ht488 and at least one of the ht386s seems to just seize up from time to time and need power cycling. haven't dared try faxing yet. |
| 19:30.23 | justinu | jbalcomb: the 300 is an older phone, and probably an older sip image |
| 19:30.27 | [av]bani | jbalcomb: what problem with handytones? fax seems to be an issue with _everything_ :) |
| 19:30.45 | Hmmhesays | it seems odd to me that sixtel gives you sip and iax2 config |
| 19:30.53 | jbalcomb | pb__ i dont know the models on our but i know we have trouble. ive been recommended to the SPA-2002s |
| 19:31.00 | Hmmhesays | so they use sip for incoming numbers? |
| 19:31.12 | pb__ | jbalcomb: yah, I'm just about to order a couple of Sipuras to play with. |
| 19:31.30 | jbalcomb | [av]bani i don't have details yet. the other phone guy handles the faxing still. i just hear about it everyday. |
| 19:31.31 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 19:31.37 | [TK]D-Fender | jbalcomb : http://pastebin.com/521045 |
| 19:31.40 | [av]bani | jbalcomb: he's talking about the 300, maybe the 501 has a transfer button |
| 19:31.47 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 19:31.53 | franck | Hi all |
| 19:32.40 | justinu | i loaned my polycom 601 to someone |
| 19:32.46 | justinu | i'm stucking using my aastra 480i |
| 19:33.03 | franck | Sometimes I cannot register my sip client. I get a 401 unauthorised but if I insist it goes, what could be the reasons? |
| 19:33.13 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 19:33.36 | [TK]D-Fender | SPA-2002 is a great value for an ATA |
| 19:33.38 | jbalcomb | [TK]D-Fender merci. i'll be addressing his statements Thursday. |
| 19:33.56 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 19:34.04 | [TK]D-Fender | justinu: I set up an "extra" IP 600 from here at home yesterday and its monitoring my SPA-3000 for "line in use" :) |
| 19:34.12 | [TK]D-Fender | jbalcomb : you bet |
| 19:34.19 | jpablo | hi, i upgraded my snom 360 to firmware 5.0 and i have SIP Disabled! according to the display, and it is asking my for a licence :S WTF? |
| 19:34.22 | DrWho17 | Is there any way built in for asterisk to detect modem calls (I've got some people making dialup calls through VoIP, and they aren't supposed to) |
| 19:34.27 | jpablo | the snom page says nothing |
| 19:34.33 | justinu | fender: interesting |
| 19:34.34 | DrWho17 | I need to kill the call if it's a modem call |
| 19:34.38 | klictel | hi all |
| 19:34.43 | pifiu-laptop | you can move dialplans to MySQL? |
| 19:34.47 | [TK]D-Fender | jbalcomb : Fix that tar, and tar.gz it please... |
| 19:34.48 | *** part/#asterisk newmedian (n=np@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 19:35.14 | [TK]D-Fender | justinu : For the PSTN port obviously. Kind neat... |
| 19:35.24 | *** join/#asterisk Defraz (n=t0tal@67.158.135.29) |
| 19:35.42 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-153-54-57.mia.bellsouth.net) |
| 19:35.48 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-153-54-57.mia.bellsouth.net) |
| 19:36.11 | *** join/#asterisk areski (n=areski@73.Red-83-60-89.dynamicIP.rima-tde.net) |
| 19:36.38 | [TK]D-Fender | jbalcomb : It looks like mashed up concatenated uncompressed text <- |
| 19:36.47 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-153-54-57.mia.bellsouth.net) |
| 19:36.52 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-153-54-57.mia.bellsouth.net) |
| 19:37.15 | franck | Sometimes I cannot register my sip client. I get a 401 unauthorised but if I insist it goes, what could be the reasons? |
| 19:37.23 | jbalcomb | strange. if i do a tar -tvf it looks fine. it definitely aint gzipped though. |
| 19:37.28 | pb__ | jpablo: did you read the "upgrading to 5.0" notes in the snom wiki? I think they talk about that. |
| 19:37.38 | jbalcomb | [TK]D-Fender strange. if i do a tar -tvf it looks fine. it definitely aint gzipped though. |
| 19:38.04 | [TK]D-Fender | jbalcomb : tar-zcvf jbalcomb.tar.gz /etc/asterisk/ |
| 19:38.31 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:38.41 | jbalcomb | [TK]D-Fender you got it |
| 19:38.56 | MstlyHrmls | jbalcomb: re your pastebin - 1),6),7) & 8) seem like configuration issues to me |
| 19:39.41 | *** join/#asterisk canada2 (n=info@s142-179-166-27.ab.hsia.telus.net) |
| 19:39.48 | jbalcomb | MstlyHrmls agreed. i'm trying to avoid too much politicing but i think he might just be jerking my chain for ego. |
| 19:40.05 | *** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 19:40.23 | MstlyHrmls | jbalcomb: 4) & 9) doesn't seem right, but I don't have a 300 handy to test |
| 19:41.07 | klictel | does anyone knows if there is a new iaxyprov for 1.2.2? |
| 19:41.11 | DarkFlibble | are there any wifi sip phones (or iax) that support WPA? |
| 19:41.19 | *** join/#asterisk Peste (i=Peste@195.230.162.134) |
| 19:41.19 | MstlyHrmls | jbalcomb: egos? they *never* factor in ;-) |
| 19:41.27 | jbalcomb | [TK]D-Fender sent |
| 19:41.32 | [TK]D-Fender | MstlyHrmls : I want at least 1 IP 301 here to test with... just so I know what's what. |
| 19:41.32 | Peste | hello everyone :) |
| 19:41.45 | franck | DarkFlibble: WPA? |
| 19:41.56 | Peste | can somebody help me ? |
| 19:41.59 | DarkFlibble | Wireless protected access... sucessor to WEP |
| 19:42.00 | jbalcomb | MstlyHrmls yeah, I wish. hard enough to keep my own in check. trying to work with someone elses feels like play 'operation' |
| 19:42.03 | hans | what's the name of the standard or whatever it is that says what's a valid phone number for PSTN? |
| 19:42.16 | DarkFlibble | hans, NAPTA? |
| 19:42.34 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 19:42.34 | MstlyHrmls | jbalcomb [TK]D-Fender: yeah the 30x is the one model I don't have handy :-7 |
| 19:42.35 | jbalcomb | hans i think theres more than one |
| 19:42.48 | jbalcomb | MstlyHrmls how do you feel about the 501? |
| 19:43.06 | hans | for coming up with dialplans |
| 19:43.10 | *** join/#asterisk kshumard (n=kshumard@gateway.digium.com) |
| 19:43.15 | }btorch{ | jobsanyone here using firefly with asterisk ? |
| 19:43.23 | [av]bani | firefly was a great series |
| 19:43.31 | MstlyHrmls | jbalcomb: I like it. I prefer 60x, but... |
| 19:43.36 | Mark_Halverson | is there anyway to validate callerid? meaning if callerid <> NXXNXXXXXX then callerid="" |
| 19:43.38 | DarkFlibble | nanp even |
| 19:43.45 | }btorch{ | agree but I'm talking about the iax softphone |
| 19:44.17 | }btorch{ | I have just installed it but it keeps asking me to register a phone with the firefly network !!! |
| 19:45.02 | jbalcomb | DarkFlibble isnt there one for local/LD numbers and another for international? |
| 19:45.14 | [TK]D-Fender | MstlyHrmls : You'd BETTER prefer the 60x :D |
| 19:45.14 | Peste | can somebody help me with " |
| 19:45.19 | Peste | can somebody help me with "app_dial.c:805 dial_exec: Unable to create channel of type 'Zap'" |
| 19:45.39 | DarkFlibble | jbalcomb, international varies on a country by country basis... |
| 19:45.44 | jbalcomb | Peste there should be another line connected with that |
| 19:45.47 | RoyK | Peste: that can be anything |
| 19:45.51 | canada2 | <PROTECTED> |
| 19:46.00 | MstlyHrmls | [TK]D-Fender: :-D |
| 19:46.03 | jbalcomb | Blame Canada!! |
| 19:46.07 | Peste | hmm.. mom |
| 19:46.11 | [TK]D-Fender | .... |
| 19:46.27 | jbalcomb | canada2 Your anus is bleeding. |
| 19:46.37 | file | canada2: you will go away, you will not pass go, you will not collect $200 |
| 19:46.53 | DarkFlibble | for example +3536237126 is a valid irish number... 10 digits... +441162222222 is a valid uk number... so the best you can do is compare against *known* national plans |
| 19:47.05 | jbalcomb | [TK]D-Fender Could you please trying to control your Canadians? |
| 19:47.12 | fugitivo | canada2: did you try ebay? |
| 19:47.13 | justinu | lol |
| 19:47.34 | [TK]D-Fender | jbalcomb : ... Virus Scan Resultbad file Unknown virus scanner failure Virus Found |
| 19:47.34 | [TK]D-Fender | Note: There is no cure available for the virus on the file jbalcomb.tar.gz |
| 19:47.37 | justinu | fender: tes canadiens sont fou! |
| 19:48.07 | jbalcomb | [TK]D-Fender werd. |
| 19:48.09 | [TK]D-Fender | justinu : De quoi tu parles, tabarnac?! |
| 19:48.19 | Peste | <PROTECTED> |
| 19:48.19 | Peste | Jan 24 21:48:22 NOTICE[8145]: app_dial.c:805 dial_exec: Unable to create channel of type 'Zap' |
| 19:48.19 | Peste | <PROTECTED> |
| 19:48.19 | Peste | <PROTECTED> |
| 19:48.19 | Peste | <PROTECTED> |
| 19:48.24 | [TK]D-Fender | jbalcomb : rename the file and resend |
| 19:48.27 | justinu | q'est que c'est tabarnac? |
| 19:48.36 | jbalcomb | [TK]D-Fender |
| 19:48.38 | DarkFlibble | Peste, use a pestebin |
| 19:48.39 | [TK]D-Fender | Peste : All free lines are busy! |
| 19:48.47 | znoG | is there any app in Asterisk to "take" somebody else's call? (ie. phone is ringing for somebody and you are at another desk and you want to take their call) |
| 19:48.55 | *** join/#asterisk thosa (n=thosa@p54878033.dip0.t-ipconnect.de) |
| 19:48.56 | Peste | but i dont use anything |
| 19:48.57 | Math` | Peste: what about.... installing zaptel^ |
| 19:49.05 | jbalcomb | [TK]D-Fender done |
| 19:49.05 | Peste | i did |
| 19:49.10 | Math` | znoG: its called call pickup |
| 19:49.16 | *** part/#asterisk thosa (n=thosa@p54878033.dip0.t-ipconnect.de) |
| 19:49.19 | znoG | Math`: that's probably it.. it can be done? |
| 19:49.26 | [TK]D-Fender | justinu : french for "tabarnacle". remember that the French are typically more religious and their swearing is often related to the church. |
| 19:49.38 | [TK]D-Fender | YAY!!! 3 cheers for K-lining! |
| 19:49.40 | justinu | lol |
| 19:49.41 | znoG | Math`: looks like it can |
| 19:49.42 | Math` | ol |
| 19:49.53 | Math` | [TK]D-Fender: we ofter spell it "tabarnak" tho :P |
| 19:50.09 | Peste | so i installed zaptel and it's now loaded? |
| 19:50.30 | justinu | je parle de tes canadiens, comme canada2 |
| 19:50.32 | Hmmhesays | hmm i need to fix my dp for hunting through service providers |
| 19:51.45 | Hmmhesays | i should probably be using macro's |
| 19:51.48 | Peste | Math`: what can i do? or what did i wrong |
| 19:51.53 | *** join/#asterisk yiddoX (n=yiddoX@host-84-9-43-72.bulldogdsl.com) |
| 19:52.18 | [TK]D-Fender | Math` : Whatever :) |
| 19:52.26 | yiddoX | can anyone assist me in getting inbound calling working via a cisco gateway? |
| 19:52.35 | justinu | for money |
| 19:52.38 | justinu | someone will help |
| 19:52.59 | [TK]D-Fender | jbalcomb : OMG, it just failed again. Try ZIPing the tar.gz |
| 19:53.04 | *** join/#asterisk heB_z0rL (n=heB_z0rL@p5492F193.dip.t-dialin.net) |
| 19:53.15 | DarkFlibble | rar will pass most virus scanners |
| 19:53.38 | DarkFlibble | failing that... base64/uuencode the file... |
| 19:55.34 | hans | [default] |
| 19:55.34 | hans | mode=files |
| 19:55.34 | hans | directory=/usr/share/asterisk/moh |
| 19:55.35 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 19:55.49 | hans | and if I stick a 8khz 1-channel WAV in that dir, should that work? |
| 19:55.56 | hans | (1.2.1) |
| 19:56.55 | *** part/#asterisk heB_z0rL (n=heB_z0rL@p5492F193.dip.t-dialin.net) |
| 19:57.44 | [TK]D-Fender | hans : Typically MOH uses MP3 files.... |
| 19:58.10 | Math` | hans: as long as you configure musiconhold.conf accordigly |
| 19:58.17 | jkitchen | brad_mssw: nah, i have zaptel/ztdummy modules in the kernel. plus meetme wouldn't work at all if ztdummy wasn't loaded |
| 19:58.30 | hans | that's my musiconhold.conf snippet right there. |
| 19:58.45 | Hmmhesays | wow oh wow, that guy was numb in the head |
| 19:58.48 | hans | moh files show shows what I'd expect (the wav file is there) |
| 19:58.54 | Hmmhesays | "pri connector" LOL |
| 19:59.13 | hans | when I'm put on hold it says it's playing music on hold, and no errors |
| 19:59.15 | hans | but I hear nothing. |
| 20:00.58 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:01.03 | *** join/#asterisk Koenvi (n=koenvi@d54C2956E.access.telenet.be) |
| 20:01.06 | Hmmhesays | i love it when people just start spewing terms out |
| 20:01.43 | [TK]D-Fender | Hmmhesays : Yeah.. I mean really... what are the odds their flux capacitor is causing random phone disconnects?! |
| 20:02.48 | nroej | good evening everybody |
| 20:03.33 | Hmmhesays | [TK]D-Fender: this guy was looking for "internal translations for his pri connector" |
| 20:03.48 | Hmmhesays | no language barrier eithe |
| 20:03.50 | DarkFlibble | [TK]D-Fender, you need to put the phone on the floor...its the electrons struggling to go uphill... |
| 20:03.50 | Hmmhesays | *either |
| 20:03.54 | *** join/#asterisk HeyEveryBody (n=Aces1Up@ip70-189-157-31.lv.lv.cox.net) |
| 20:03.58 | DarkFlibble | :P |
| 20:04.28 | *** join/#asterisk XIN01OZ (n=askme@pcp03218165pcs.hlcrs201.al.comcast.net) |
| 20:05.30 | Peste | so.. can anybody help me to solve this "Unable to create channel of type 'Zap'"? zaptel and libpri are installed |
| 20:06.21 | DarkFlibble | is the chan_zap loaded? |
| 20:06.49 | DarkFlibble | have you configured the conf file? |
| 20:06.57 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 20:07.12 | Peste | yes |
| 20:07.14 | Koenvi | you need to configure zaptel.conf and zapata.conf |
| 20:07.39 | Hmmhesays | i don't think i've ever used macro in a production box, has anyone else? |
| 20:07.58 | Peste | zapata.conf: |
| 20:08.08 | Koenvi | Peste, what kind of hardware are you using |
| 20:08.09 | Peste | [channels] |
| 20:08.09 | Peste | language=en |
| 20:08.09 | Peste | context=internal |
| 20:08.09 | Peste | switchtype = euroisdn |
| 20:08.09 | Peste | pridialplan=UNKNOWN |
| 20:08.10 | Peste | signalling=pri_cpe |
| 20:08.23 | Peste | iechocancel=yes |
| 20:08.23 | Peste | callerid=asreceived |
| 20:08.23 | Peste | echocancelwhenbridged=no |
| 20:08.23 | Peste | echotraining=800 |
| 20:08.23 | Peste | group=0 |
| 20:08.24 | DarkFlibble | Peste, use a pastebin |
| 20:08.24 | Peste | channel=>1-15,17-31 |
| 20:08.25 | [TK]D-Fender | Peste : Use pastebin gaddamnit! |
| 20:08.28 | [TK]D-Fender | ~pb |
| 20:08.29 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:08.31 | Peste | pastebin? |
| 20:08.36 | justinu | grrr |
| 20:08.45 | DarkFlibble | second time I told you |
| 20:09.13 | Peste | k |
| 20:09.30 | Peste | i have a DIGIUM TE110P card |
| 20:09.44 | Peste | connected with a cross e1 cable |
| 20:10.12 | Peste | to a nms card (dont no type) |
| 20:10.18 | [TK]D-Fender | jbalcomb : Awaiting the next attempt.... At worse send it to my other address |
| 20:10.48 | Koenvi | I have a TE110P running at the office... but can't get to the config from here |
| 20:11.25 | justinu | peste: is the zaptel kernel module loaded? |
| 20:11.30 | Koenvi | from what I remember zapata looks ok |
| 20:11.54 | Peste | i loaded with modprobe zaptel and then modprobe wcte11xp |
| 20:11.59 | Peste | no errors |
| 20:12.21 | Peste | but i dont know how to show the loaded modules |
| 20:12.27 | Koenvi | what does "zap show channels" give in the CLI |
| 20:12.38 | Koenvi | lsmod |
| 20:13.20 | Peste | everything ok with lsmod |
| 20:14.12 | Peste | see pb |
| 20:14.30 | Koenvi | and there are some pri commands in CLI |
| 20:14.59 | HeyEveryBody | does anyone have any experience in setting up something like a 40-line autodialer system? with asterisk? |
| 20:15.10 | znoG | sounds like an AutoSpammer |
| 20:15.12 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 20:15.17 | kink0 | hello |
| 20:15.36 | Peste | pri show span 1: no Pri running on span 1 :( |
| 20:15.44 | kink0 | someone can give me what signalling are ussing to connect PRI E1 ? |
| 20:15.52 | HeyEveryBody | znog yeh a client of mine wants it to call a database of numbers and send an automated message. |
| 20:16.14 | kink0 | I am trying to connect Digium TE405 to Stargate 2N PRI E1 |
| 20:16.25 | Koenvi | Peste, did you define the spans? |
| 20:16.29 | nroej | humm can someone here try to call my ekiga sip uri? wanna see if my *s are configured well |
| 20:16.38 | Koenvi | I think it's in zapata.conf, but not sure |
| 20:17.27 | justinu | spans are configured with zaptel.conf |
| 20:17.48 | *** join/#asterisk backblue (n=moo@87-196-46-161.net.novis.pt) |
| 20:17.57 | Peste | no i didnt |
| 20:18.55 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 20:19.19 | Peste | ok |
| 20:20.09 | kink0 | if I load zaptel + wct4xxp, is sussposed I must get ACTIVE Pri in the other extreme , even if I do not start Asterisk ? |
| 20:20.31 | justinu | eh? |
| 20:20.41 | justinu | your PRI won't come up unless asterisk is running, i think |
| 20:20.53 | Koenvi | I think the D-channel is handled by ast |
| 20:20.55 | justinu | your T1/E1 layer will come up |
| 20:21.02 | justinu | but not the D |
| 20:21.16 | kink0 | well , was to isolate zaptel.conf and zapata.conf |
| 20:21.42 | kink0 | I get always Layer1: DEACT on the other end |
| 20:21.42 | Peste | so what? |
| 20:21.49 | Katty | le Nugget |
| 20:22.06 | justinu | kink0: layer 1 generally refers to the T1/E1 layer |
| 20:22.30 | kink0 | yes, but I think must be ACTIVE all time |
| 20:23.05 | justinu | yeah |
| 20:23.13 | justinu | first of all, is it e1, or t1? |
| 20:23.14 | kink0 | and I am not sure about signalling , because I have not used pri-cpe due to a asterisk error if I use as document says for a TE405 |
| 20:23.19 | kink0 | is E1 |
| 20:23.36 | justinu | k |
| 20:23.43 | justinu | make sure the span= line in zaptel.conf is correct |
| 20:23.46 | justinu | then run ztcfg -vvv |
| 20:23.52 | kink0 | I set signalling to fxsk |
| 20:23.57 | justinu | that's wrong |
| 20:24.04 | kink0 | yes, ztcfg -vvvvvvvvv goes fine, no errors |
| 20:24.13 | justinu | signalling should be bchan=1-15 |
| 20:24.16 | justinu | dchan=16 |
| 20:24.17 | kink0 | what wrong ? signalling ? |
| 20:24.22 | justinu | bchan=17-30 |
| 20:24.40 | kink0 | hmmm , ok, if I set signalling bchan in zaptel.conf, what signaling to use in zapata.conf |
| 20:24.41 | kink0 | ? |
| 20:24.52 | justinu | switchtype=euroisdn |
| 20:24.53 | justinu | probably |
| 20:24.57 | kink0 | I did use bchan, but... asterisk crashes with error while load chan_zap |
| 20:25.08 | justinu | yeah, that's some silly problem with chan_zap |
| 20:25.08 | kink0 | let me try again so... |
| 20:25.14 | justinu | it won't work if the signaling doesn't match |
| 20:25.21 | kink0 | ahh is normal that crashes ?? |
| 20:25.27 | justinu | yeah, i've seen that |
| 20:25.46 | kink0 | I had bchan(zaptel) and euro(zapata) |
| 20:25.53 | kink0 | ok, I will try one more time just now |
| 20:26.01 | justinu | pastebin your files |
| 20:26.04 | justinu | lets see if they're ok |
| 20:27.05 | kink0 | Changing signalling on channel 1 from FXO Kewlstart to Clear channel ... |
| 20:27.10 | justinu | that's better |
| 20:28.11 | kink0 | ok, now at zapata.conf ... I have switchtype=euroisdn |
| 20:28.17 | kink0 | but what signalling ? |
| 20:28.27 | kink0 | pri-cpe appears don't work for me. |
| 20:28.30 | justinu | pri_net probably |
| 20:28.37 | kink0 | ok, will try pri_net |
| 20:28.42 | justinu | since your other card probably expects to be CPE side |
| 20:29.00 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:29.16 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:29.19 | kink0 | chan_zap.c:6859 mkintf: Signalling requested on channel 1 is FXO Kewlstart but line is in PRI Signalling signalling |
| 20:29.22 | kink0 | and crashes |
| 20:29.36 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 20:29.37 | justinu | k, lets see zapata.conf |
| 20:29.41 | kink0 | but... now I set from FXOKew to bchan ... |
| 20:29.49 | *** join/#asterisk Doda (n=doda@81-235-161-106-no21.tbcn.telia.com) |
| 20:29.55 | justinu | pastebin zaptel.conf and zapata.conf |
| 20:31.01 | kink0 | ok... now I need to remember how use pastebin !! |
| 20:31.09 | justinu | it's pretty simple |
| 20:31.19 | kink0 | yes, but I used rarely :) |
| 20:31.27 | justinu | it's just cut and paste |
| 20:32.21 | *** join/#asterisk mog_work (i=root@user-24-236-84-48.knology.net) |
| 20:32.50 | kink0 | http://pastebin.com/521162 |
| 20:33.22 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:34.04 | kink0 | http://pastebin.com/521163 |
| 20:34.04 | Peste | how can i create a span for e1 and in which config |
| 20:34.25 | kink0 | justinu, hehehe, yes, was simple... there are my two config files. |
| 20:34.31 | kink0 | zapata.conf and zaptel.conf |
| 20:34.53 | justinu | kink0: i can't remember if E1 allows you to use channels 1-30, or 1-31 |
| 20:35.12 | justinu | if you only put bchan=17-30 in zaptel.conf, you might need to change it to 17-31 |
| 20:35.23 | justinu | if there's ANY mismatch in zaptel.conf and zapata.conf, chan_zap likes to just crash |
| 20:35.41 | kink0 | yes, that is happening , crashing |
| 20:36.02 | tzafrir_home | Juggie, in E1 the dchannel is 16 |
| 20:36.10 | tzafrir_home | 1-16, 17-31 |
| 20:36.26 | tzafrir_home | justinu, that is |
| 20:36.27 | justinu | that's what he's got |
| 20:36.37 | justinu | 1-15 |
| 20:37.01 | justinu | kink0: uncomment the other groups in zapata.conf |
| 20:37.03 | kink0 | <PROTECTED> |
| 20:37.26 | justinu | also, in each group, specify the signalling |
| 20:37.28 | kink0 | but I have a signalling set ... probably is expecting other signalling ? |
| 20:37.34 | justinu | group=1 |
| 20:37.38 | justinu | signalling=pri_net |
| 20:37.39 | kink0 | ok, will set in each group now. |
| 20:37.58 | justinu | it has to be just right, or else it doesn't work |
| 20:39.11 | kink0 | chan_zap.c:10546 setup_zap: Unknown signalling method 'pri_net' |
| 20:39.11 | kink0 | Jan 24 21:40:57 ERROR[2946]: chan_zap.c:10171 setup_zap: Signalling must be specified before any channels are. |
| 20:39.22 | justinu | maybe it's prinet, or pri-net |
| 20:39.24 | justinu | i can't remember |
| 20:39.31 | kink0 | this is what happens if I set for every one group |
| 20:39.43 | kink0 | ok, I will search how is named signalling variable... |
| 20:39.56 | Doda | I'm am working with a solid DTMF caller id solution for Swedish networks and the Digium TDM400p card. I have some questions where to find some documentations about processes and register setups for the TDM400p PCI card. Can anyboudy help me? |
| 20:39.59 | justinu | pri_net is right |
| 20:40.14 | justinu | kink0: http://www.digium.com/asterisk_handbook/zapata.conf.pdf |
| 20:40.30 | kink0 | yes , pri_net sintax is ok |
| 20:46.37 | *** join/#asterisk Joe_ (n=jdunn@yayformarmots.com) |
| 20:46.38 | Doda | I have a working solution for DTMF caller id signalling for FXS connected phone. But a Asterisk server warns me about that the process running the DTMF signalling is stopped for to long time (DTMF signalling caller id is called from the function zt_call()). It would be nice if somebody with expert knowledge about asterisk could guide me for a "chick" solution without risking inteference with some outer subsystems in the Asterisk software? |
| 20:47.28 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:49.41 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 20:52.59 | *** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
| 20:53.04 | buzzyd | hi all |
| 20:53.38 | buzzyd | does anyone know of a simple restart script in the event that asterisk crashes it starts it up again without my input |
| 20:53.49 | fugitivo | safe_asterisk ? |
| 20:54.03 | buzzyd | I seem to be having an issue with meetme crashing asterisk |
| 20:54.09 | *** join/#asterisk M|kee (n=ol@outbound.infosysinc.com) |
| 20:54.13 | buzzyd | safe_asterisk for me? |
| 20:54.24 | fugitivo | yes |
| 20:54.45 | buzzyd | ah cheers :) |
| 20:55.27 | M|kee | the Asterisk Forums are down, so I'll ask here. Can Cisco IOS talk to Asterisk? |
| 20:58.26 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 20:58.27 | }btorch{ | how can I setup my voicemail password for the first time ? |
| 20:59.11 | }btorch{ | I added an entry on the voicemail.conf file without a password and when I try to call the voiceman cmd I cant login |
| 20:59.40 | h3x | M|kee: like a couple of teenage girls |
| 20:59.55 | jpablo | M|kee, yes |
| 21:00.31 | AndyCap | M|kee: what kind of meaningful conversation did you expect to take place between the twe? |
| 21:01.19 | jpablo | M|kee: I have a cisco 2821 working as a PRI gateway for asterisk. |
| 21:02.54 | M|kee | I have Cisco 2600's and 3600s with FXS cards |
| 21:02.59 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de) |
| 21:03.14 | M|kee | I wanted to dial through the Cisco's and have them tell Asterisk to dial out via FXO |
| 21:03.46 | M|kee | Cisco using a dialpeer and Asterisk using what ever it uses |
| 21:04.09 | *** join/#asterisk TrickyR (n=MT@157.246.8.43) |
| 21:04.42 | *** join/#asterisk rend (n=rend@cpe-67-10-81-6.houston.res.rr.com) |
| 21:05.04 | rend | whats a quick way to get started with asterisk? i have a voip sip provider and a linux box here. |
| 21:05.51 | justinu | ~thebook |
| 21:05.52 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 21:06.36 | *** join/#asterisk Assid (n=assid@59.183.57.186) |
| 21:07.45 | M|kee | AndyCap or jpablo any ideas? |
| 21:07.49 | hans | am I the only one that notices x-lite in osx crashes when I'm testing conferences in asterisk? |
| 21:07.51 | *** part/#asterisk Koenvi (n=koenvi@d54C2956E.access.telenet.be) |
| 21:08.13 | Assid | just curious.. will mon-mon work in gotoiftime .. although we should only be using 'mon' once |
| 21:08.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:08.56 | Hmmhesays | anyone else in here using sixtel right now? |
| 21:09.11 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:09.22 | brad_mssw | Hmmhesays: I've got a sixtel acct |
| 21:10.03 | jbalcomb | [TK]D-Fender the zipped, gzipped, tar is on its way |
| 21:10.06 | Hmmhesays | brad_mssw, why do they give you sip peer settings |
| 21:10.17 | Hmmhesays | and they call it "inbound" and there is no sip register line |
| 21:10.20 | jbalcomb | [TK]D-Fender next I attach each one as plain text |
| 21:11.06 | *** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com) |
| 21:11.34 | brad_mssw | Hmmhesays: eh? where are you looking at ... on their control.sixtel.net, if you go to help, you have options between iax config and sip config |
| 21:11.40 | brad_mssw | Hmmhesays: don't see any peer settings there |
| 21:11.45 | brad_mssw | Hmmhesays: just friend |
| 21:12.34 | Hmmhesays | i'm setting this up for a client, the email they sent has both |
| 21:12.51 | Hmmhesays | i'm guessing I can kick the sip part |
| 21:13.11 | brad_mssw | yeah, i'm using only iax to them ... |
| 21:14.00 | rend | when trying to compile asterisk, i get: configure: error: termcap support not found |
| 21:16.30 | [TK]D-Fender | jbalcomb : hold onto that though, just shove them on FTP or something |
| 21:16.45 | [TK]D-Fender | jbalcomb : Or make me an account on your server so I can pick them up. |
| 21:18.20 | [TK]D-Fender | jbalcomb : the last one bombed too... FFS |
| 21:18.50 | [TK]D-Fender | Hotmail is pissing me off today... |
| 21:18.58 | *** join/#asterisk arcy (n=arcanum@ppp54-adsl-118.ath.forthnet.gr) |
| 21:20.06 | kpettit | with AMI what's the easiest way to get the full active channel from a SIP extension? |
| 21:20.23 | kpettit | For example I have SIP/200 and I need to get SIP/200-xxxxx |
| 21:20.28 | jbalcomb | [TK]D-Fender thats crazy. lemme send them from my hotmail |
| 21:20.29 | *** join/#asterisk pengyong (n=lala@218.93.159.101) |
| 21:21.23 | kpettit | I'm basically trying to get a list of active calls so I can do transfering through AMI |
| 21:22.01 | [TK]D-Fender | jbalcomb : Sure. Whatever. Its getting ridiculous.... |
| 21:22.41 | kpettit | but to do that I need the full Channel ID. Right now I'm parsing through the entire output of "Action: Status" but that's kind of painfull |
| 21:22.53 | *** join/#asterisk zikos (n=zxx@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 21:23.06 | cpm | Okay, my first asterisk server ever, I love it, it's joyful, I've had other folks asterisk servers to play with, but this the first (my own) asterisk box. Set it up saturday, all was joyful, got a number from voicepulse, configured, all is well. Now, since yesterday, I can no longer connect any outbound dialing, Session looks like this http://pastebin.com/521247 |
| 21:23.08 | jbalcomb | [TK]D-Fender yeah, maybe microsoft is bad at email. |
| 21:23.18 | kpettit | lo cpm |
| 21:23.22 | [TK]D-Fender | jbalcomb : first time in ages I've had any problems... |
| 21:23.27 | cpm | hey kpettit, how you man? |
| 21:23.50 | kpettit | busy, been programing AMI stuff all day. It's pretty fun |
| 21:24.37 | cpm | sounds like fun, YOu still doing the same gig? Or have you found that funner gig yet? |
| 21:24.42 | *** part/#asterisk zikos (n=zxx@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 21:24.51 | jbalcomb | [TK]D-Fender ive never seen it like this either. maybe our email is being handled by Asterisk now or something. |
| 21:24.55 | jbalcomb | HAHAHAH! |
| 21:25.08 | kpettit | all asterisk all the time |
| 21:25.13 | cpm | nice |
| 21:25.29 | jbalcomb | [TK]D-Fender Hotmail wont even let me attach the file because of the virus check. |
| 21:25.53 | kpettit | cpm you using the right context in extensions.conf? |
| 21:25.54 | rend | does asterisk help anyone get laid? |
| 21:26.00 | [TK]D-Fender | jbalcomb : Then I'll have to grab it when I get home then. |
| 21:26.07 | cpm | I was on saturday :) Lemme look again. |
| 21:26.23 | [TK]D-Fender | read : Laid off perhaps... |
| 21:26.36 | rend | maybe there is a dating service using asterisk |
| 21:26.48 | jbalcomb | I was thinking more like laid out.. |
| 21:26.59 | jbalcomb | by his mom |
| 21:27.21 | rend | milf? |
| 21:27.47 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 21:28.07 | *** part/#asterisk ukh (n=ukh@ibook-wifi.svansen.se) |
| 21:28.28 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 21:28.45 | }btorch{ | damn how can I call someone if they don't have a number but a name between the [] |
| 21:29.23 | }btorch{ | I can't figure this out ... is there a way to assign a number to a channel ? |
| 21:29.30 | SplasPood | Is there currently a solution to allow multiple separate customers on the same asterisk box to park calls at will, and yet prevent Customer A from picking up Customer B's parked calls...? |
| 21:29.49 | kpettit | SplasPood, you have them in different call groups |
| 21:30.14 | SplasPood | oh? Lemme read up |
| 21:30.34 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 21:30.37 | [TK]D-Fender | ok, heading home, bbiab |
| 21:30.37 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:30.53 | Seldon1975 | hey all; my PCom501s have 3 speed dial hard-keys on the left of the display; the top one of these is assigned to my Extension number - anyone know how to make this button useful (ie: pint to an actual speed-dial entry)? |
| 21:30.55 | SplasPood | kpettit: Oh nice... And this worked /w 1.0 stable? |
| 21:31.06 | Seldon1975 | ciao D-Fender |
| 21:31.14 | kpettit | Have no idea, never used * that old |
| 21:31.33 | SplasPood | kpettit: yea I'd generally try to avoid it as well :) |
| 21:32.10 | kpettit | 1.0.9 is what i started using * |
| 21:32.25 | pifiu-laptop | what does "call rejected by xxx.xxx.xxx.xxx no authority found" mean? |
| 21:32.40 | justinu | iax authentication issue |
| 21:33.37 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 21:33.55 | pifiu-laptop | meaning? |
| 21:34.00 | SplasPood | kpettit: well whatever version was released as the first "proper" 1.0 stable |
| 21:34.12 | SplasPood | kpettit: i'm already in the process of testing tho, so don't worry :) |
| 21:34.17 | warthawg | i still got them 'asterisk sez zultys phones are busy when they're not' blues |
| 21:34.20 | warthawg | <sigh> |
| 21:35.36 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 21:36.00 | s34n | chan_sip.c:3414 process_sdp: Insufficient information for SDP (m = '', c = '') |
| 21:37.11 | pifiu-laptop | wow i hate iax2 or i just suck at this shit |
| 21:37.13 | s34n | does it make sense that an INVITE might come from a different IP than RTP data? |
| 21:37.25 | file | yes, but the INVITE still has to contain SDP :) |
| 21:37.31 | file | with the info in it. |
| 21:38.00 | SplasPood | kpettit: hrm.. you define pickupgroup and callgroup = to say... 2 in sip.conf, then any users with the same pickup/call group can pickup those calls, but others cannot? |
| 21:38.29 | kpettit | yes. I don't have a config in front of me but I think that's how it works |
| 21:38.41 | kpettit | Been a couple months sense I did one of those |
| 21:38.49 | Seldon1975 | anyone here using Polycom501's? |
| 21:39.01 | kpettit | Seldon1975, what you need? |
| 21:39.07 | SplasPood | heh, well either this installation is old enough that it lacks the feature, or I'm not doin something right |
| 21:39.10 | Seldon1975 | the first speed-dial button is assigned to my extension |
| 21:39.25 | Seldon1975 | kpettit: in other words, if I press it, it dials my own extension |
| 21:39.33 | Seldon1975 | kpettit: its not useful |
| 21:39.37 | kpettit | haha |
| 21:39.41 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 21:39.45 | SpaceBass | hey again folks |
| 21:39.45 | Seldon1975 | kpettit: can I make this map to an SD entry? |
| 21:40.11 | s34n | file: the debug shows both an m and a c |
| 21:40.16 | Seldon1975 | kpettit: like the other two speed dial hard keys |
| 21:40.20 | kpettit | Seldon1975, are you talking the Extension buttons on the top left of the phone? |
| 21:40.20 | s34n | file: so why is * complaining? |
| 21:40.29 | SpaceBass | I'm not getting my fax and voicemail emails... is there a way to specify smtp host, etc? |
| 21:40.30 | Seldon1975 | kpettit: yeah |
| 21:40.36 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 21:40.37 | Seldon1975 | kpettit: the three blue keys |
| 21:40.46 | file | s34n: because it may not be exactly right, and the parser might be freaking out? |
| 21:40.50 | Seldon1975 | kpettit: the other two display speed-dial entries |
| 21:40.51 | kpettit | Those are not speed dials, those show your extension |
| 21:41.03 | kpettit | I usually configure them like I would different lines on a phone |
| 21:41.15 | Seldon1975 | kpettit: hmmm |
| 21:41.17 | s34n | file: m=audio 32776 RTP/AVP 0 |
| 21:41.26 | kpettit | Your supposed to be able to pick up two calls per button |
| 21:41.30 | kpettit | so 6 in total |
| 21:41.43 | Tall-guy | Gents, I'm just messing with $DIALSTATUS in my extensions.conf, and I've seen two different ways of doing it...trying to find which is the preferred (right) way.. |
| 21:41.45 | s34n | file:c=IN IP4 my.rtp.ip.address |
| 21:41.50 | Tall-guy | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+goto This way? |
| 21:41.56 | Seldon1975 | kpettit: ever since I added SDs to the phones directory, the first two entries have been assigned to the lower two buttons |
| 21:42.10 | Seldon1975 | kpettit: my users would like to utilise all 3 in this way |
| 21:42.15 | Tall-guy | or this way? http://www.asteriskguru.com/tutorials/voicemail.html |
| 21:42.32 | Seldon1975 | kpettit: do you know if that's possible? |
| 21:42.41 | kpettit | I haven't tried anything like that |
| 21:42.44 | kpettit | not sure |
| 21:42.47 | file | s34n: pastebin a sip debug |
| 21:42.51 | Seldon1975 | kpettit: ok thanks anyway |
| 21:43.05 | kpettit | I usually just use them as extension buttons for placing and reciving calls from different extensions |
| 21:43.16 | Seldon1975 | kpettit: how do you configure them? |
| 21:43.18 | rend | what is a good program to help configure asterisk? i dont want to have to setup a web app... |
| 21:43.25 | Seldon1975 | in the mac-phone.cfg? |
| 21:43.26 | kpettit | I use FTP |
| 21:43.34 | kpettit | yeah |
| 21:43.36 | SpaceBass | anyone know how I can have * use my local SMTP server to deliver VM and faxes rather than its own? |
| 21:43.39 | Seldon1975 | ok |
| 21:43.51 | Seldon1975 | where in the Admin guide should I look for that configuration? |
| 21:44.03 | Tall-guy | spacebass: I do that. |
| 21:44.24 | SpaceBass | Tall-guy, where do you configure the SMTP host? |
| 21:44.44 | kpettit | Seldon1975, don't have a admin guide to look at. When I download the different sip and bootrom firmware it has exmaple configs that i use |
| 21:44.46 | tzafrir_home | rend, vi? |
| 21:44.52 | Tall-guy | space: you need SOME sort of smtp on the asterisk box..I think the default is EXIM or something like that.....and all you do is tell it to send all mail thru your "real" smtp server |
| 21:44.53 | cpm | exit |
| 21:44.54 | Seldon1975 | kpettit: ok thanks anyway |
| 21:44.56 | cpm | heh |
| 21:45.15 | Tall-guy | space: it's not an asterisk setting, its a linux setting |
| 21:45.22 | SpaceBass | Tall-guy, ahhh |
| 21:45.28 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 21:45.29 | tzafrir_home | Actually, asterisk expects a /usr/sbin/sendmail |
| 21:45.33 | gongoputch | how much CPU power /memory do you need to drive a couple of IP phones and 2 lines with asterisk? |
| 21:45.42 | Tall-guy | space: or more accurately, the smtp server program on your server. |
| 21:45.47 | SpaceBass | Tall-guy, basically I'm not getting most of my VM and Faxes coming through and I assume its a routing issue and they are going out and back in |
| 21:45.47 | tzafrir_home | Which can be a real MTA, like sendmail, postfix or exim |
| 21:45.51 | Tall-guy | tzafrir: sure, aint they all compliant though? |
| 21:45.56 | Tall-guy | (yeah, what you said) |
| 21:46.01 | SpaceBass | not sure whats included in AAH, assume sendmail |
| 21:46.04 | tzafrir_home | But it can also be nullmailer or similar |
| 21:46.09 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 21:46.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:46.41 | SpaceBass | yeah, sendmail |
| 21:46.52 | tzafrir_home | Tall-guy, nullmailer et al. will have a problem if they have failed to send the message on first shot. |
| 21:46.57 | mogorman | vrey littlte gongoputch |
| 21:46.58 | Tall-guy | ah, tru. |
| 21:47.03 | tzafrir_home | sendmail? yuck. Use postfix |
| 21:47.17 | SpaceBass | don't feel like swapping it out if I don't have to.... |
| 21:47.25 | SpaceBass | tried postfix and it was pretty complex for me |
| 21:47.39 | brookshire | how is postfix more complex than sendmail? |
| 21:47.41 | gongoputch | mogorman: so I would be safe with a 450 mhz P-II and 384 MB ? |
| 21:47.45 | tzafrir_home | SpaceBass, basically there is very little to change in the default config |
| 21:48.00 | SpaceBass | never had this problem until the latest release oh AAH.... it might be a DNS issue... |
| 21:48.02 | mogorman | yeah |
| 21:48.13 | gongoputch | cool, thnx |
| 21:48.17 | mogorman | only problem you might hit is pci compatibility |
| 21:48.27 | mogorman | tdm card needs 2.2 pci bus |
| 21:48.28 | mogorman | or higher |
| 21:48.28 | darwin_35 | any one on 1.2.2 having sip header issues ? |
| 21:48.34 | gongoputch | is there a webmin module for asterisk? |
| 21:48.39 | mogorman | yeah |
| 21:48.42 | mogorman | but its no good |
| 21:48.48 | gongoputch | very cool |
| 21:48.52 | darwin_35 | ? |
| 21:48.55 | gongoputch | oh, not so much then |
| 21:49.03 | tzafrir_home | There is one useless and obsolete module |
| 21:49.27 | gongoputch | I messed with * a while back, and I am not afraid of CLI and conf files, but it was crazy. |
| 21:49.49 | tzafrir_home | gongoputch, well, script it a bit |
| 21:50.05 | tzafrir_home | gongoputch, also, with 1.2 you have the nice #exec config directive |
| 21:50.11 | mogorman | its not hat bad |
| 21:50.22 | gongoputch | I'll give another whirl |
| 21:50.50 | *** join/#asterisk linlin2 (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 21:51.03 | Katty | file: snob. |
| 21:51.16 | file | Katty: meep? |
| 21:51.19 | Katty | file: you didn't bug me last week. |
| 21:51.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:51.28 | SpaceBass | darwin_35, like what kind of issues? |
| 21:51.39 | file | Katty: awww :( |
| 21:51.55 | SpaceBass | ok, nslookup on the * box is showing the correct internet MX record for my SMTP host... rules out DNS |
| 21:51.59 | darwin_35 | File do you know of any current issues with sip headers on 1.2.2 |
| 21:52.11 | file | darwin_35: can you be any more vague? |
| 21:52.16 | darwin_35 | it seems a client is getting bad header packets |
| 21:52.27 | file | aka samples of the offending packets are good |
| 21:52.51 | *** join/#asterisk intensedr (n=scolson@209.172.11.52) |
| 21:52.54 | darwin_35 | getting one now |
| 21:52.59 | Hmmhesays | lol |
| 21:53.03 | Hmmhesays | i need a beer |
| 21:53.04 | *** join/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 21:53.04 | Hmmhesays | now |
| 21:53.10 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:53.17 | gongoputch | is FreeBSD a common choice of OS for an * installation? |
| 21:53.28 | darwin_35 | we use it and love it |
| 21:53.30 | rend | what is a good program to help configure asterisk (gui)? i dont want to have to setup a web app... |
| 21:53.35 | Hmmhesays | more common than windows i bet |
| 21:53.59 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:54.13 | hanchi | does anyone know the website for the apherion (spelling??) project, is supposed to be a NEBA 5 class asterisk project for E911 and carrier class |
| 21:54.19 | tzafrir_home | rend, what kind of GUI? |
| 21:54.33 | Katty | file: you still love me, right? |
| 21:54.40 | file | Katty: of course! |
| 21:54.42 | Katty | k |
| 21:54.50 | tzafrir_home | hanchi, look for #openpbx |
| 21:54.50 | tzanger | Katty: I got your new year's card today |
| 21:54.59 | tzanger | at least I think that's what it is -- I haven't opened it yet |
| 21:55.06 | Katty | tzanger: wow, that's a little late |
| 21:55.09 | Hmmhesays | its anthrax |
| 21:55.14 | tzafrir_home | hanchi, a different project, a different motivation(?), but somewhat similar mindshare |
| 21:55.20 | Katty | Hmmhesays: lies. |
| 21:55.32 | Hmmhesays | I could use a new project, anyone got some work for me? |
| 21:55.32 | Hmmhesays | lol |
| 21:55.39 | file | Hmmhesays: make me money. |
| 21:55.41 | file | there's your poject! |
| 21:55.44 | *** join/#asterisk Qwell[laptop] (n=chatzill@70.103.248.130) |
| 21:55.44 | file | er project |
| 21:56.05 | Hmmhesays | it seems the tap is a little dry lately for small quick jobs for bar money |
| 21:57.15 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 21:57.30 | *** join/#asterisk Muckl (n=yo@p54BEDA24.dip.t-dialin.net) |
| 21:57.42 | Hmmhesays | maybe cause I help to many people out for free, lol |
| 21:58.00 | SpaceBass | ok, regarding my mail issue...using my local DNS as my server but I cannot resolve local hosts, only internet |
| 21:58.13 | rend | tzafrir_home: linux. |
| 21:58.29 | Cybertoy | ok .. I have a weird one here... I'm using the latest and greatest asterisk I just checked out of svn ... |
| 21:58.31 | Zodiacal | is this site down for anyone else? http://forums.digium.com/ |
| 21:58.44 | Cybertoy | I'm using 2 GotoIfTime statements in a row ... and nothing happens after that. |
| 21:58.55 | Cybertoy | when I comment out the second statement it goes on ... |
| 21:59.19 | Cybertoy | does that make sense? |
| 21:59.26 | Cybertoy | zodiacal, it's not working for me either, the site that is. |
| 21:59.45 | Zodiacal | :/ |
| 22:00.44 | Cybertoy | exten => 102,1,GotoIfTime(23:00-07:00,mon-fri,*,*?local,998,1) |
| 22:00.44 | Cybertoy | ;exten => 102,n,GotoIfTime(23:00-09:00,sat-sun,*,*?local,998,1) |
| 22:00.44 | Cybertoy | exten => 102,n,NoOp(============jumping into Macro==============) |
| 22:00.44 | Cybertoy | exten => 102,n,Macro(stdext,102,SIP/phone1&SIP/phone2&SIP/101) |
| 22:00.44 | Cybertoy | If I uncomment that second line it never gets to the NoOp ... |
| 22:00.48 | Qwell[laptop] | !pb |
| 22:00.52 | Qwell[laptop] | ~pb |
| 22:00.54 | Qwell[laptop] | stupid laptop |
| 22:02.01 | }btorch{ | does iax2 work with text messages ? |
| 22:02.13 | pifiu-laptop | im fed up wtih iax2 rigt now |
| 22:02.15 | pifiu-laptop | wow lol |
| 22:02.20 | pifiu-laptop | i fucking suck at this shit |
| 22:02.40 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 22:03.04 | Hmmhesays | tell us how you really feel pifiu-laptop |
| 22:04.30 | darwin_35 | http://pastebin.ca/38437 file here |
| 22:04.48 | darwin_35 | he is saying the toute line is wrong |
| 22:05.09 | darwin_35 | that it should point to the internal ip it shoul dbe going to not the proxy |
| 22:05.11 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 22:05.34 | twisted[asteria] | can I not use functions from within NoOp()? |
| 22:05.36 | file | that's not corrupted for one thing, and the route stuff is already known about |
| 22:06.03 | twisted[asteria] | ie, exten => s,n(seed),NoOp(Seeding Cycle ${DB(MY/CYCLE)=1}) |
| 22:06.09 | twisted[asteria] | it's not working.. |
| 22:06.15 | hanchi | is anyone aware of * being used for E911 at a PSAP |
| 22:06.16 | darwin_35 | where is the bug know I dont find a report in bugs.digium.com |
| 22:06.28 | file | I don't memorize bug numbers. |
| 22:06.39 | tzanger | file: uh, why not? you lazy fucker |
| 22:06.44 | twisted[asteria] | anyone? |
| 22:06.46 | *** join/#asterisk srodgers (n=mzone2k1@63.111.4.170) |
| 22:06.48 | file | makes my head hurt :( |
| 22:06.51 | file | twisted[asteria]: Should work... |
| 22:06.56 | twisted[asteria] | file, that's what I thought |
| 22:07.07 | file | twisted[asteria]: except I didn't think you could set variables like that... |
| 22:07.08 | twisted[asteria] | yet, it doesn't.... i'm trying to figure out if i'm going insane or not |
| 22:07.11 | tzanger | twisted[asteria]: I don't think you can do that, you're trying to do an assignment without a Set() |
| 22:07.18 | twisted[asteria] | it's not setting a variable, it's the DB function |
| 22:07.27 | *** part/#asterisk arcy (n=arcanum@ppp54-adsl-118.ath.forthnet.gr) |
| 22:07.36 | Qwell[laptop] | Don't you still need to use Set? |
| 22:07.40 | Qwell[laptop] | or, in this case, SET |
| 22:07.41 | darwin_35 | is ti listed as routing or how ? |
| 22:07.48 | twisted[asteria] | Qwell[laptop], not according to the synopsis :) |
| 22:07.49 | darwin_35 | I dont find it under sip |
| 22:08.02 | Qwell[laptop] | You actually read those things? ;] |
| 22:08.14 | twisted[asteria] | Qwell[laptop], yes, when it's something i need a refresher on |
| 22:08.19 | *** join/#asterisk Krill (n=majestic@210-84-11-13.dyn.iinet.net.au) |
| 22:08.26 | file | twisted[mobile]: try using set first... |
| 22:08.58 | twisted[asteria] | file, k, although if it's required to use set, it should be noted in the 'show function' crap. |
| 22:09.05 | twisted[asteria] | i'd change it but i haven't yet got svn installed on this mac |
| 22:09.08 | darwin_35 | I need to see the bug to pass it on to the customer and my boss |
| 22:09.24 | darwin_35 | not finding anything to do with packet routing |
| 22:09.45 | *** join/#asterisk hans (n=fugalh@128.123.45.209) |
| 22:10.55 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 22:10.56 | twisted[asteria] | file, that doesn't work either. |
| 22:10.58 | file | darwin_35: it's not packet routing |
| 22:11.06 | twisted[asteria] | omg this is so broken |
| 22:11.07 | darwin_35 | what is it then |
| 22:11.46 | file | darwin_35: http://bugs.digium.com/view.php?id=6284 http://bugs.digium.com/view.php?id=6240 |
| 22:11.56 | file | note oej's response |
| 22:12.02 | file | and note that all I did was type in route for the search :P |
| 22:12.16 | *** join/#asterisk TokyoJimu (n=jimmy@198.51.175.64) |
| 22:15.03 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 22:17.56 | darwin_35 | is 1.2.0 still in svn |
| 22:18.04 | Qwell[laptop] | darwin_35: yes |
| 22:18.16 | Qwell[laptop] | but, why? |
| 22:18.31 | darwin_35 | this client with the issue has 3415 numbers threw us |
| 22:18.49 | darwin_35 | and this bug is causing major broken calls o |
| 22:19.14 | darwin_35 | and they are on 1.2.0 |
| 22:19.20 | Qwell[laptop] | upgrade |
| 22:19.24 | darwin_35 | so I now have to build a box to match |
| 22:19.48 | darwin_35 | they cant just up grade |
| 22:19.55 | Qwell[laptop] | why? |
| 22:20.14 | tainted_ | yea why |
| 22:20.39 | tainted_ | say it darwin_35 |
| 22:20.41 | darwin_35 | to many clients on all the time . and my boss said just build a box with 1.2.0 and shift thier account to it |
| 22:20.50 | Qwell[laptop] | umm |
| 22:21.02 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:21.02 | Qwell[laptop] | so... |
| 22:21.04 | tainted_ | your boss is dumb |
| 22:21.07 | Qwell[laptop] | let me get this straight |
| 22:21.07 | intensedr | Anyone know why if I get a Call thru my Asterisk box and I answer it too quickly it drops the call? |
| 22:21.11 | Qwell[laptop] | You're building a NEW box |
| 22:21.17 | Qwell[laptop] | but you can't install a new version...because... |
| 22:21.26 | tainted_ | intensedr what tech |
| 22:21.28 | Qwell[laptop] | there are too many clients on the NEW box? |
| 22:21.55 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:22.08 | intensedr | Im not sure was wondering if its a known issue? |
| 22:22.18 | SpaceBass | anyone with asterisk@home 2.x mind sending me the /etc/mail/sendmail.cf file? |
| 22:22.37 | darwin_35 | the client has a box on thier site |
| 22:23.07 | darwin_35 | that is 1.2.0 we upgraded this wekend with new servers and asterisk 1.2.2 |
| 22:23.12 | tainted_ | intensedr SIP? IAX2? TDM? |
| 22:23.13 | darwin_35 | and this issue started |
| 22:23.16 | intensedr | SIP |
| 22:23.25 | tainted_ | what provider |
| 22:23.30 | darwin_35 | and they are getting a lot of broken calls |
| 22:23.32 | tainted_ | soft client? |
| 22:23.33 | tainted_ | ata? |
| 22:23.36 | tainted_ | ip phone? |
| 22:23.41 | intensedr | ip phone |
| 22:23.45 | tainted_ | darwin_35 waht do u mean broken calls |
| 22:23.51 | tainted_ | which ip phone |
| 22:24.01 | darwin_35 | calls dropping off |
| 22:24.02 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 22:24.10 | tainted_ | complete disconnect or audio cut out |
| 22:24.19 | intensedr | Like a Gargling noise |
| 22:24.23 | darwin_35 | audio cut out and dropoff |
| 22:24.26 | intensedr | and audio cut out |
| 22:24.50 | tainted_ | darwin_35 are clients behind NAT/local to asterisk server? |
| 22:24.54 | tainted_ | intensedr which ip phone |
| 22:25.14 | tainted_ | intensedr what codecs are you using |
| 22:25.20 | tainted_ | intensedr are u transcoding? |
| 22:25.25 | darwin_35 | I dont know what thier clienst use . but we send over 3 thound numbers to this client and they then connect to thier clients |
| 22:25.30 | tainted_ | intensedr what version asterisk |
| 22:25.43 | darwin_35 | we found the bug in digum |
| 22:25.48 | darwin_35 | file pointed it out |
| 22:25.53 | tainted_ | which bug? |
| 22:26.00 | darwin_35 | 6240 |
| 22:26.45 | tainted_ | hmm |
| 22:26.47 | tainted_ | interesting |
| 22:27.04 | *** join/#asterisk oceanlan (n=irc@cpe-69-133-109-130.woh.res.rr.com) |
| 22:27.16 | syle | if you assign callerid in realtime sip table or by just having the field at all disable people's ability to send their own callerid? |
| 22:28.31 | darwin_35 | yes tainted thats the issue and a ugly one |
| 22:29.39 | tainted_ | interesting |
| 22:29.48 | tainted_ | might be responsible for my dropped calls as well |
| 22:31.39 | }btorch{ | anyone here uses jabber with * ? |
| 22:32.35 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:32.39 | intensedr | Tainted they are Sipura Phones |
| 22:32.46 | s34n | file: sorry. I'm working on it. |
| 22:32.49 | darwin_35 | i hope they fix it soon |
| 22:33.58 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 22:34.16 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 22:38.21 | SpaceBass | anyone with asterisk@home 2.x mind sending me the /etc/mail/sendmail.cf file? |
| 22:39.03 | shmaltz | SpaceBass, what you trying to config? |
| 22:39.15 | justinu | SpaceBass: a word of advice... give up on a@h |
| 22:39.18 | Qwell[laptop] | sendmail would be my guess |
| 22:39.22 | shmaltz | SpaceBass, have you tried #sendmail |
| 22:39.33 | shmaltz | Qwell, thanks :P |
| 22:39.34 | SpaceBass | shmaltz, trying to get my .cf file back...trashed mine and m4 appears to be broken in this distro |
| 22:39.59 | *** join/#asterisk JimmyGulp (n=james@ns0.esagroup.co.uk) |
| 22:40.07 | SpaceBass | as for A@H...most of the time it works well for my purposes... i do write some dialplans manually but its just so nice for home use |
| 22:40.15 | justinu | k |
| 22:40.38 | fdask | hey guys, i'm trying to use a Dial() function in my dialplan, but i'm stuck for what I should put for channel |
| 22:40.58 | fdask | i dont think i'd use zap, because ive got a voicetronix card |
| 22:41.00 | SpaceBass | fdask, did you try "show application dial" in the CLI? |
| 22:41.12 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 22:41.37 | fdask | no, looking at that now tho |
| 22:41.40 | SpaceBass | justinu, but the next box i build, even for home use will be straight ast not AAH |
| 22:41.47 | fdask | still not sure what i'd put for Technology |
| 22:42.04 | fdask | like i need to tell it to use my card, but im not sure how i do that |
| 22:43.18 | SpaceBass | i dont know anything about voicetronix cards.... sorry |
| 22:44.23 | fdask | hrm |
| 22:44.55 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net) |
| 22:45.03 | Tall-guy | http://www.voip-info.org/wiki-Asterisk+channels |
| 22:45.11 | Tall-guy | Has a pointer to voicetronix cards....need VPB channel |
| 22:45.37 | Tall-guy | and vpb.conf |
| 22:46.09 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 22:47.09 | fdask | checking that out. thanks tall-guy |
| 22:47.09 | *** join/#asterisk sac|h0p|werk (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
| 22:47.16 | Tall-guy | fdask: google is your friend |
| 22:47.23 | *** join/#asterisk bweschke (i=bweschke@129.sub-70-197-76.myvzw.com) |
| 22:47.54 | eKo1 | is anybody having trouble downloading asterisk from svn? |
| 22:47.57 | eKo1 | i keep getting 400 Bad Request (http://svn.digium.com) |
| 22:48.38 | Qwell[laptop] | eKo1: behind a proxy? |
| 22:48.50 | eKo1 | kinda |
| 22:48.53 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 22:49.14 | eKo1 | not really |
| 22:49.17 | Qwell[laptop] | yeah...proxies don't usually like svn |
| 22:49.24 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 22:49.28 | brookshire | you can try cvs |
| 22:50.28 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 22:50.37 | eKo1 | rats, i wanted to use svn |
| 22:51.06 | kpettit | I'm doing some AMI stuff. When trying to do a transfer to a external phone number is that still the "Exten:" |
| 22:51.18 | kpettit | or do I use something differnet. |
| 22:52.39 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 22:55.59 | SYS64738 | I am tryng to configure asterisk to connect to FWD, I receive calls, but I cannot do them |
| 22:56.14 | SYS64738 | where could I search for errors ? |
| 22:57.03 | *** join/#asterisk scon (n=scon@dslb-084-057-005-052.pools.arcor-ip.net) |
| 22:57.43 | kpettit | SYS64738, type "asterisk -vvvr" and try looking to see what happens |
| 22:58.02 | SYS64738 | kpettit, thanks |
| 22:59.57 | SYS64738 | I see only the registered message |
| 22:59.58 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 23:00.00 | *** join/#asterisk jmcc (n=jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 23:00.03 | SYS64738 | nothing else |
| 23:00.55 | *** join/#asterisk [1]EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 23:01.08 | *** join/#asterisk hans (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:01.17 | Grubs | Does anyone have format_mp3 working for MusicOnHold under 1.2.2? |
| 23:01.28 | jmcc | is it possible to send MWI messages to SIP phones which aren't on the same * server? that is, i want to have voicemail on a different box from where the SIP phones are registered. How to make MWI work? |
| 23:01.39 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 23:03.35 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 23:06.26 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 23:06.48 | *** join/#asterisk scon_ (n=scon@dslb-084-057-014-201.pools.arcor-ip.net) |
| 23:07.08 | [TK]D-Fender | jbalcomb : you there? |
| 23:10.36 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
| 23:10.58 | gongoputch | is there a good softphone for FreeBSD? |
| 23:12.44 | jmcc | how do people implement centralized voicemail on an * box but with SIP phones in different locations--how does MWI in that case? |
| 23:12.58 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 23:13.12 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 23:14.19 | *** join/#asterisk chrisb1 (n=chrisb@66.111.33.40) |
| 23:14.31 | chrisb1 | hiyas |
| 23:14.40 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-133.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:15.08 | gongoputch | I am reading some ratings sites, and it seems that the Polycom SoundPoint IP 301 is getting edged out by the Snom 320 ... is that the consensus here? |
| 23:16.31 | *** part/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 23:18.32 | darwin_35 | they are killing me |
| 23:18.59 | Grubs | I dont think screaming will get you anywhere. |
| 23:19.01 | file[laptop] | okay, you're using free software... and you want Digium to fix the bugs... for free... |
| 23:19.07 | file[laptop] | maybe you should learn how to fix 'em |
| 23:19.38 | *** join/#asterisk knut (n=hans@p54866F34.dip.t-dialin.net) |
| 23:19.42 | darwin_35 | but this issue of the route has to be fixed . its killing us |
| 23:19.47 | knut | hi there |
| 23:20.11 | file[laptop] | okay, so? |
| 23:20.35 | darwin_35 | I crawl and beg and plead |
| 23:20.41 | darwin_35 | for it to be fixed |
| 23:20.52 | xachen | Asterisk is real buggy |
| 23:20.55 | xachen | get over it :p |
| 23:21.39 | chrisb1 | could anyone help me and point me in the right direction as to where i can find detailed information on DIALPLAN construction options (specifically the context area spoken of in: http://www.digium.com/handbook-draft.pdf) which doesnt seem to elaborate on what options can go into this section? :-) |
| 23:22.59 | chrisb1 | basically trying to figure out what: |30|tTL(123456:60000:30000) at the end of my dialplan call means :-) |
| 23:23.29 | Grubs | chrisb1: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 23:23.44 | *** join/#asterisk _vic (n=riccardo@81.174.56.78) |
| 23:23.49 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 23:24.08 | chrisb1 | ahhah tyvm grubs |
| 23:24.49 | Grubs | :) |
| 23:28.02 | _vic | hi there. is a configuration with a cisco_827-4v (adsl+4 pots), an fxo, asterisk and some voip provider known to be working? i'm able to call out echo and time service of isp but when receiving a call is only modo-directional (out -> in = ok, in -> out = no sound) :-( |
| 23:28.31 | alephcom_ | Lol, now it's not nigerians getting rid of millions. It's american soldiers getting rid of saddams money. Gotta love spam |
| 23:30.26 | [av]bani | funny, i thought it was russian oil companies getting rid of embezzled money |
| 23:30.51 | jkitchen | I thought it was a stranded cosmonaut promising millions when he gets home if you help him out |
| 23:31.14 | [av]bani | last week it was deposed african dictators looking for assistance in shipping money out of the country |
| 23:31.28 | jkitchen | I haven't gotten a phishing email in ages ;| |
| 23:31.54 | jkitchen | well, nigerian phishing at least |
| 23:32.02 | jkitchen | I get lots of emails from banks |
| 23:32.12 | [av]bani | paypal is really insecure, i keep getting mails telling me to enter my credit card |
| 23:32.19 | [av]bani | about 2-3 per day |
| 23:32.20 | jkitchen | and ebay and paypal |
| 23:33.33 | alephcom_ | jkitchen: If you want them sendme your email address and I'll forward mine to you. :-) |
| 23:33.48 | jkitchen | alephcom_: nah, that's ok ;) |
| 23:34.48 | *** join/#asterisk Cazper_ (n=cazper@c5100A229.sdsl.catch.no) |
| 23:35.11 | *** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net) |
| 23:35.38 | chrisb1 | is anyone familiar at all with the behaviour of the L() variable in dialstrings (call time limits and warnings) ? |
| 23:35.58 | joaovianna | chrisb1: Yes... |
| 23:35.59 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 23:36.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:36.36 | chrisb1 | i was just wondering if the Y/Z parameters (warn @ Yms, repeat @ Zms) works correctly if X is less than them? |
| 23:36.46 | chrisb1 | or is it best to calculate the Y/Z parameters based on X? |
| 23:37.24 | chrisb1 | behaviour atm seems to be that if X < Y|Z, warning is immediate, followed by another warning straight away - and then nothing |
| 23:38.04 | chrisb1 | sorry for noob questions - i just got thrown into this project yesterday and have until friday to get it working :P |
| 23:38.48 | joaovianna | Well, I'm using for a calling card application. Just test it... |
| 23:40.47 | _vic | thanks a lot. |
| 23:40.51 | *** part/#asterisk _vic (n=riccardo@81.174.56.78) |
| 23:41.21 | knut | where can i find detailed documentation about the voicemail thing ? is the software beween voicemail() a script which could be manipulated? |
| 23:43.27 | *** join/#asterisk Cazper (n=cazper@c5100A229.sdsl.catch.no) |
| 23:43.37 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:46.10 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 23:48.04 | knut | hum anyway thank you ;) |
| 23:48.30 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 23:50.20 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 23:50.37 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 23:50.57 | Mavvie | I'm looking for somebody who is connected to the DUNDI network. |
| 23:51.18 | Grubs | voicemail is fairly configurable I think. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 23:58.57 | *** part/#asterisk zaf (n=tfournet@cdm-68-228-9-79.laft.cox-internet.com) |
