00:00.15 | Vlat- | if they're now wanting to do - we use STUN |
00:00.28 | epablo | OK |
00:01.14 | epablo | Has anyone used EAGI with perl. I need to access that FD 3 to send audio, but I don't know how |
00:01.47 | Vlat- | clyrrad: 're you here ? |
00:03.53 | *** join/#asterisk Delta34 (n=delta34o@198.87.24.253) |
00:04.09 | *** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net) |
00:05.42 | *** join/#asterisk file[laptop] (n=jcolp@142.166.94.161) |
00:05.55 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
00:06.23 | *** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net) |
00:06.27 | *** part/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net) |
00:06.53 | *** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net) |
00:07.01 | Delta34 | hi all, i been trying to answer for this but no luck, is their a way to display the called party's name to the calling party, say if i dailed 4000 it will tell me that i dialed the operator |
00:07.07 | dan__t | bah#*@!&($#!$#*@ |
00:07.23 | Delta34 | this is using cisco 7960 phones and asterisk |
00:07.38 | Delta34 | i read its a limitaion of sip |
00:08.12 | Vlat- | Delta34: CallerID is evil |
00:08.52 | Vlat- | usually it doesn't work...unless you're in US, and have got the perfecit gatway chain |
00:09.21 | Delta34 | no its for internal users on the asterisk box |
00:09.30 | syle2 | i bought one of those rhino channel banks, anyone played with them before? |
00:09.32 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
00:09.37 | Delta34 | old analog pbx has this feature |
00:09.52 | leszq | bye |
00:09.54 | *** part/#asterisk leszq (n=leszq@82.177.97.254) |
00:09.54 | cio | Just joined - What feature Delta34? |
00:10.38 | Delta34 | so if i dialed ext 4000, on my cisco 7960 phone, it will tell me i dialed the opeartor at ext 4000 |
00:10.39 | MikeJ[Laptop] | Delta34, yes, you can do that |
00:10.47 | Delta34 | how? |
00:11.01 | MikeJ[Laptop] | wait..on the dialing phone or the called phone? |
00:11.20 | Delta34 | on the dialing phone, i know callerid shows up on the called phone |
00:11.28 | *** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
00:11.30 | MikeJ[Laptop] | it works now to show the called phone whatever you want, for example if it is a que call, what que it is coming from |
00:11.35 | kingtux | hell all |
00:11.40 | kingtux | hello |
00:11.47 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
00:11.58 | MikeJ[Laptop] | on the 7960's I beleive you can send xml stuff back to the screen, I do not know if that is implemented anywhere |
00:12.46 | supaigtr | Is there a way to disable a zaptel card and use asterisk with card still intact? I'm trying to see if my audio problems are card. |
00:12.59 | Delta34 | dont think u can send xml stuff that way, the xml stuff is for directories or services features |
00:13.11 | kingtux | having trouble with ztdummy mod....I'm getting this error when I try to modprobe it. I have no digium cards just VOIP....FATAL: Error inserting ztdummy (/lib/modules/2.6.9-11.EL/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
00:13.11 | kingtux | FATAL: Error running install command for ztdummy |
00:18.14 | supaigtr | kingtux: is the module installed in the modules directory for the running kernel? |
00:19.03 | cio | On debian, I should be able to compile cvs with just my kernel-headers, right? I shouldn't have to use the entire source tree?! |
00:20.14 | *** join/#asterisk coppice (n=chatzill@48.201.17.210.dyn.pacific.net.hk) |
00:20.55 | *** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
00:20.55 | *** mode/#asterisk [+o drumkilla_laptop] by ChanServ |
00:21.30 | marc324 | how do you force * to use thr db instead of conf files |
00:22.43 | *** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
00:23.30 | kingtux | yes |
00:23.42 | kingtux | i see it there |
00:23.52 | kingtux | ztdummy.ko |
00:24.19 | *** join/#asterisk Math` (n=math@modemcable231.182-70-69.mc.videotron.ca) |
00:25.10 | supaigtr | Kernel compiled with symbols? |
00:29.09 | *** join/#asterisk simprix (n=simprix@24-231-248-225.static.aldl.mi.charter.com) |
00:29.27 | simprix | Does anyone have experience with voicepulse connect ? |
00:31.29 | brookshire[home] | i do! |
00:31.55 | file | Mattttttt |
00:33.28 | *** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey) |
00:33.51 | cio | How stable is the 1.2beta1? |
00:33.59 | brookshire[home] | more stable than 1.0.9 |
00:34.00 | brookshire[home] | :) |
00:34.24 | brookshire[home] | in some situations |
00:34.26 | cio | How about vs 1.0.7? |
00:34.41 | brookshire[home] | i think 1.0.7 has a security hole |
00:35.16 | simprix | brookshire[home]: how do you like voicepulse |
00:35.26 | brookshire[home] | voicepulse is awesome |
00:35.39 | brookshire[home] | i'm setting it up on my new computer as we speak actually |
00:35.42 | file | brookshire[home]: wazzup? |
00:35.45 | simprix | cool |
00:35.48 | brookshire[home] | but i've had them for 6 months |
00:35.49 | simprix | the service is ok |
00:35.50 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
00:36.02 | brookshire[home] | and they have a number in my area |
00:36.03 | brookshire[home] | :) |
00:36.54 | kingtux | man |
00:36.57 | *** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
00:37.49 | cio | Anyone running 1.2beta1 in production? |
00:38.40 | supaigtr | cio: I was as of today. |
00:39.02 | InfraRed | anything to report |
00:39.06 | brookshire[home] | hey file |
00:39.06 | cio | Meaning you're still running it or you change from that version to another? |
00:39.38 | supaigtr | I'm backing down but I think that'll create more problems. RIght now I have major problems with IAX2 to IAX2 that have been ongoing. |
00:40.13 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve) |
00:41.33 | shido6 | taco bell |
00:41.50 | supaigtr | How do I stop the real zaptel driver loading and force ztdummy to load instead? |
00:42.36 | simprix | brookshire[home]: do i need a normal voicepulse account also |
00:42.37 | cio | mmmmmmm... taco bell.... |
00:42.42 | *** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
00:42.56 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj0e.dialup.mindspring.com) |
00:43.05 | brookshire[home] | i don't think so |
00:44.49 | RoyK | <PROTECTED> |
00:45.46 | supaigtr | cio: Its gotten so complicated I don't know whats broken. |
00:45.53 | NetSkier | Anyone here attending Astricon? I am wondering how good it has been so far. |
00:46.03 | RoyK | running cvs head in production is for madmen and jerjer |
00:46.05 | RoyK | that is |
00:46.14 | InfraRed | |<-------------------->| this good |
00:46.18 | InfraRed | not to scale |
00:46.19 | NetSkier | lol |
00:46.37 | InfraRed | :) |
00:46.53 | RoyK | s/madman/other madman/ |
00:46.57 | brookshire[home] | netskier: #astricon |
00:47.08 | NetSkier | I have been wondering if I should drive thru rush hr traffic tomorrow to crash the party. |
00:47.12 | NetSkier | brookshire: thnx |
00:49.59 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
00:50.17 | supaigtr | Anyone seen calls to drop to on-hold music then back? |
00:51.44 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
00:52.38 | marc324 | how do you force asterisk to load from db? |
00:54.01 | brookshire[home] | as in realtime? |
00:54.07 | marc324 | yes |
00:54.21 | marc324 | i have the db setup (hopeso). |
00:54.43 | brookshire[home] | http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime |
00:54.44 | marc324 | but now as a a test, i want to load the extensions from db |
00:56.28 | brookshire[home] | basically in extensions.conf you will just do something like |
00:56.31 | brookshire[home] | switch => Realtime/mycontext@realtime_ext |
00:56.40 | blitzrage | yo! |
00:56.57 | brookshire[home] | BLITZ!!! |
00:57.04 | blitzrage | brookshire[home]: !!!!!!!!!!!!!!!!!!!!!! |
00:57.10 | blitzrage | brookshire[home]: we're missing you at astricon |
00:57.21 | HiltonT | how is Astricon? |
00:57.33 | *** join/#asterisk dos000 (n=dos000@i216-58-62-73.cybersurf.com) |
00:57.38 | brookshire[home] | http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album02&id=IMG_3334 |
00:57.41 | HiltonT | (if only I could get A@H to work, I could actually play with *!!! |
00:57.42 | brookshire[home] | BEST PIC EVER! |
00:57.48 | Vlat- | btw, by the astericon |
00:58.06 | HiltonT | rofl |
00:58.07 | brookshire[home] | HiltonT: just drop A@H and do a hardcore install ;) |
00:58.15 | Vlat- | if there's a consolidated solution, like ser+asterisk together |
00:58.30 | brookshire[home] | much less headache imho :) |
00:58.33 | HiltonT | I would, but I want to play a little first before I get my head around Linux/BSD again |
00:58.36 | Vlat- | i would be happy to receive any info about it |
00:58.37 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
00:58.44 | Vlat- | in other words - let me know :) |
00:58.53 | brookshire[home] | hilton: can you install debian? |
00:58.53 | HiltonT | been quite a while since I had Linux/BSD here to work on |
00:58.55 | cio | in extensions.conf, if I have 4 lines in a TDM400P, my TRUNK should say ZAP/g1-g4? |
00:59.06 | HiltonT | Debian - shouldn't be an issue |
00:59.16 | dos000 | HiltonT, last time i tried A@H it went smooth. I just did not like centos thz all |
00:59.27 | HiltonT | Asterisk fails to start here |
00:59.46 | brookshire[home] | install debian |
00:59.54 | HiltonT | thi the 100-clone card gets detected as does the rest of the hardware (P2B-DS, dual Cel-522, 512MB, SCSI HDD) |
00:59.55 | brookshire[home] | then follow this: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian |
01:00.00 | Vlat- | merge ser and asterisk together...and you won't need 5300/5350 anymore |
01:00.05 | dos000 | HiltonT, tell what is the error, someone can maybe help |
01:00.10 | HiltonT | brookshire; thanks, reading |
01:00.13 | cio | Or ZAP/g1-4? |
01:00.47 | HiltonT | no http loaded, AMP shows "Asterisk" as not started, php script errors when creating an extension - which error do you want me to start with? :) |
01:01.14 | dos000 | Vlat-, except for the g729 stuff ... not sure how many simultaneous g729 you can do on a P43G machine |
01:01.14 | brookshire[home] | lol |
01:01.14 | HiltonT | Xorcom Rapid - that worth a look? (since it is Debian anyway) |
01:01.18 | brookshire[home] | yeah |
01:01.20 | brookshire[home] | i like it |
01:01.32 | Vlat- | dos000: g729 isn't needed anymore i guess |
01:01.36 | cio | Anyone? |
01:01.38 | Vlat- | yes, it's a good codec |
01:01.40 | dos000 | Vlat-, why ? |
01:01.50 | Vlat- | it have...maybe 2-3 years |
01:02.07 | brookshire[home] | i agree |
01:02.16 | dos000 | Vlat-, fill me in .. i just came back few months on the voip arena |
01:02.17 | Vlat- | right now ILBC sometimes can suffer a concurency to 729 |
01:02.19 | brookshire[home] | g729 will be replaced with a codec with higher quality |
01:02.20 | brookshire[home] | :) |
01:02.28 | Vlat- | for example |
01:02.30 | brookshire[home] | like mp3 quality |
01:02.31 | HiltonT | I have Xorcom here, may try that first |
01:02.35 | dos000 | brookshire, which one ? |
01:02.38 | Vlat- | we're using 729/711 world-wide |
01:02.57 | brookshire[home] | dunno.. quess we'll find out |
01:03.02 | Vlat- | and when it's no 729 compatibility - 711 play a best |
01:03.02 | simprix | brookshire[home]: did you have to have a phone number or did they give you oner |
01:03.03 | brookshire[home] | maybe skype's |
01:03.10 | Vlat- | erm |
01:03.17 | tzafrir_laptop | HiltonT, xorcom is good :-) |
01:03.17 | Vlat- | skype must die :) |
01:03.24 | brookshire[home] | simprix: you do not have to have a phone number with voicepulse connect |
01:03.26 | HiltonT | cool, will look at that |
01:03.28 | dos000 | hell with skype |
01:03.29 | brookshire[home] | but you can buy one extra |
01:03.29 | Vlat- | they're using hacked ilbc |
01:03.39 | HiltonT | Skype, schmype |
01:03.59 | Vlat- | we had several customers, switched to skype |
01:04.11 | tzafrir_laptop | Vlat-, no, they're using another codec that is also called ilbc. Or actually: anohter variant of ilbc. A patented one |
01:04.15 | dos000 | Vlat-, but seriously how many can you do in asterisk ? |
01:04.16 | Vlat- | after some month they're went back |
01:04.22 | NetSkier | HiltonT: tzafrir wrote Xorcom. |
01:04.55 | Vlat- | tzafrir: any difference? we can use uncompressed 711 now |
01:05.02 | NetSkier | I like xorcom too. |
01:05.08 | Vlat- | and it make no difference on current dsl speeds |
01:05.24 | Vlat- | 9kbit or 64kbit... |
01:05.38 | Vlat- | is there any difference if user have 3mbit ? |
01:05.45 | tzafrir_laptop | Their variant uses much less bandwidth. But if you can use g711, don't bother |
01:05.53 | dos000 | Vlat-, ha ... i have people on dial up calling from remote third world countries over vsat ! |
01:06.03 | Vlat- | dos000: asterisk is our media backend |
01:06.10 | tzafrir_laptop | But with our current DSL, uplink is still a problem |
01:06.15 | brookshire[home] | yeah.. in that case you would need g729 |
01:06.23 | dos000 | Vlat-, any reboot horror stories ? |
01:06.30 | brookshire[home] | g729 is about the only codec that can fit on dialup |
01:06.31 | Vlat- | dos000: we don't use it for primary, coz it have a ..interesting.. implementation of sip protocol |
01:06.42 | Vlat- | as PBX it acts the best |
01:06.54 | tzafrir_laptop | NetSkier, I'm trying to add more providers to the add-trunk script. Any ideas? |
01:07.14 | Vlat- | dos000: we have ser + asterisk, and these working the best |
01:07.21 | NetSkier | Not really; I am an asterisk newbie. |
01:07.36 | Vlat- | ser can handle a 10000 connection at dual Xenon |
01:07.40 | Vlat- | Xeon |
01:07.50 | dos000 | Vlat-, but tell me did you experienced unscheduled down time at all due to ser and/or asterisk |
01:08.00 | Vlat- | asterisk can do anything with codec conversion |
01:08.09 | Vlat- | dos000: no |
01:08.29 | NetSkier | tzafrir: I just hit upon xorcom to jumpstart my debian efforts. Great way to get started quickly. |
01:08.38 | Vlat- | dos000: we have backup boxes. but, to be honest, we did not need them in the past 3 years |
01:08.57 | dos000 | Vlat-, wow ... how do you rate asterisk over sems for vmail ? |
01:09.01 | Vlat- | i don't want to tell the asterisk suck (it's great software) |
01:09.16 | Vlat- | neither (ser is great) (great software, but the config........) |
01:09.18 | HiltonT | then, I suppose tzafrir is a tad on the biased side, which isn't necessarily bad :) |
01:09.35 | Vlat- | together they're making a real-working thingie |
01:09.44 | *** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
01:09.59 | NetSkier | HiltonT: Quick, install it quickly, while he is here, so you can bug him for help. ;) |
01:10.00 | CoaxD | Okay, who's gonna /msg me an NT Server 4.0 reg key? :) |
01:10.18 | dos000 | Vlat-, what do you use for failover ? |
01:10.27 | brookshire[home] | coax: astalavista.com ? |
01:10.31 | CoaxD | My license arrives tomorrow. i need to do the install today. C'MON. hah |
01:10.45 | CoaxD | brookshire: Microsoft keeps most of these search engines pretty damn well tied down in that arena. *g* |
01:11.28 | *** join/#asterisk starman (n=inshift@host06.alica.hyatthsiagx.com) |
01:11.36 | Vlat- | dos000: just to make things clear |
01:11.42 | brookshire[home] | first off.. why nt? |
01:11.45 | dos000 | CoaxD, NT is'nt that the os from the time of lion heart ? |
01:11.46 | Vlat- | also we're using 3 5350 |
01:11.49 | brookshire[home] | you might as well use window 95 |
01:12.15 | Vlat- | and "Asterisk suck/SER rulez or vice-versa"...sorry, but it make no difference |
01:12.20 | brookshire[home] | windows N(o)T |
01:12.37 | CoaxD | dos000: Pretty much |
01:12.40 | CoaxD | That said, i still need it |
01:13.20 | dos000 | Vlat-, you mentioned you had a failover solution .. what is it based on ? |
01:13.26 | Vlat- | dos000: there're a rack, in about 5 meters from me |
01:13.53 | Vlat- | dos000: 2 Intel2.4 ghz servers as ser |
01:14.20 | Vlat- | if one doesn't respond - every transaction forwardet to the second by hardware |
01:14.26 | dos000 | Vlat-, how do you do failover |
01:14.49 | dos000 | Vlat-, my question is how does the forwarding happen |
01:14.52 | Vlat- | dos000: 2 SER machine and self-written scripts |
01:15.06 | Vlat- | dos000: it does on t_on_failure[x] |
01:15.13 | kingtux | i'm trying to make asterisk-addons and i'm gettting this error .. |
01:15.17 | kingtux | cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory |
01:15.17 | kingtux | make: *** [cdr_addon_mysql.o] Error 1 |
01:15.22 | Vlat- | ser's default feature |
01:15.44 | brookshire[home] | that's not an error.. it's a feature! |
01:16.00 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
01:16.04 | Vlat- | t_on_failure[1] { |
01:16.18 | NoRemorse | hello, how do I disbale CLI presentation to a peer please? is it just SetCallerID() ? |
01:16.19 | kingtux | well i'm looking to store cdr data...so this will screw me up right |
01:16.21 | Vlat- | <PROTECTED> |
01:16.28 | Vlat- | <PROTECTED> |
01:16.29 | Vlat- | { |
01:16.30 | dos000 | Vlat-, i dont get it. does your failover run (on hw) below sip or in sip ? |
01:16.47 | brookshire[home] | kingtux: looks like it can't find the asterisk.h file |
01:16.53 | Vlat- | dos000: all we have is a sip, nothing more or less |
01:17.13 | Vlat- | IAX is the thing we don't really like (and we can describe why) |
01:17.26 | kingtux | now i'm getting this error ... |
01:17.35 | brookshire[home] | of course.. i don't have that either |
01:17.39 | kingtux | FATAL: Error inserting ztdummy (/lib/modules/2.6.9-11.EL/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
01:17.39 | kingtux | FATAL: Error running install command for ztdummy |
01:18.18 | brookshire[home] | that seems like the kernel module is not matching up with the kernel version |
01:18.19 | Vlat- | dos000: asterisk is the best for PBX solutions. Very great |
01:18.21 | brookshire[home] | uname -a |
01:18.40 | Vlat- | dos000: but I have no idea, why the people using it at ITSP level |
01:18.59 | Vlat- | yes, it's more confortable |
01:19.09 | brookshire[home] | kingtux: you did compile asterisk first and install it before you tried to compile the addons right? |
01:19.29 | MikeJ[Laptop] | brookshire, arn't you supposed to be in CA? |
01:19.31 | Vlat- | but less configurable. so IMHO, SER is for signalling, * is for anything other |
01:19.39 | brookshire[home] | mikej: who would run digium? |
01:19.40 | brookshire[home] | :) |
01:19.44 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
01:20.08 | kingtux | yes compiled it fist |
01:20.14 | brookshire[home] | did you install? |
01:20.18 | brookshire[home] | make install |
01:20.34 | Vlat- | some time ago i was in ser devteam |
01:20.49 | Vlat- | and it make me cry, the people using asterisk for the proxy |
01:21.07 | Vlat- | becose it's several hundreds time SLOWER |
01:21.12 | kingtux | make install |
01:21.21 | Vlat- | 'comz it's for PBX solutions |
01:21.41 | brookshire[home] | i'm sure one day it will have a proxy |
01:21.47 | brookshire[home] | heh.. just not anytime soon |
01:22.14 | Vlat- | brookshire[home]: you're alredy have it |
01:22.21 | HiltonT | ok - Xorcom installing as I type... |
01:22.33 | brookshire[home] | hehe.. not a real one |
01:22.35 | brookshire[home] | ;) |
01:22.49 | Vlat- | brookshire: grep the source |
01:22.53 | supaigtr | Vlat: Any pointers on SER TNT and * vm? |
01:23.11 | Vlat- | supaigtr: vm like voicemail ? |
01:23.15 | supaigtr | Yea. |
01:23.17 | Vlat- | if so, forget about SEMS |
01:23.24 | supaigtr | SEMS? |
01:23.36 | Vlat- | SER Media System |
01:23.41 | brookshire[home] | vlat: you know.. as long as i've used asterisk.. i've never had to deal with sip.. so i don't really care, lol |
01:23.50 | Vlat- | Asterisk voicemail is JUST GREAT |
01:24.07 | brookshire[home] | mainly just iax and tdm :) |
01:24.09 | Vlat- | you just need forward() your ser to local asterisk backend |
01:24.37 | supaigtr | Vlat: I'd like use the best of * meetme, etc, best of maxtnt PSTN - SIP, and SER lot of normal users with SIP to the TNT with some voicemail. |
01:24.40 | dos000 | Vlat-, how you rate asterisk against sems ? |
01:24.48 | Vlat- | brookshire: sorry, i think i just did a mistake, thought about only sip |
01:25.26 | Vlat- | dos000: SEMS is 10%, Asterisk is 100% |
01:25.41 | Vlat- | all: SER is for routing |
01:25.50 | dos000 | Vlat-, what is missing .. just on the voicemail side of things |
01:25.51 | Vlat- | all: Asterisk is for user's pleasure :) |
01:25.55 | *** join/#asterisk Weezey (n=ohno@CPE001195cf5c03-CM0014e8267934.cpe.net.cable.rogers.com) |
01:26.03 | supaigtr | * vm and meetme seems very stable, other things like IAX trunking, SIP support seem a bit flaky at times. |
01:26.05 | NetSkier | Vlat-: I would like to know why you don't like iax2. |
01:26.08 | Vlat- | SER is just a router, nothing more |
01:26.28 | dos000 | Vlat-, i am talking about sems not ser per say |
01:26.30 | Vlat- | NetSkier: stop. we doesn't work with IAX |
01:26.42 | brookshire[home] | iax is good for asterisk to asterisk |
01:26.49 | brookshire[home] | but.. for ip phones.. |
01:26.54 | supaigtr | SER = softwitch vendors worst nightmare. Something that works that doesn't cost 6 digits. |
01:26.54 | brookshire[home] | iax is lacking for support |
01:26.54 | Vlat- | i had several (negative) thoughths about IAX |
01:27.10 | Weezey | (IAX slept with his dog) |
01:27.11 | cio | Is there a trick to getting tapi notify turned on with asterisk? I can place tapi calls, but I can't get it to notify on inbound ... |
01:27.19 | Vlat- | for example - rtp/sincro broke cause rtp/sincro broke |
01:27.20 | supaigtr | iax between boxen doesn't work so hot all the time. |
01:27.33 | Vlat- | e.g. 30ns timeout for rtp |
01:27.35 | NetSkier | Weezey: Heh. Well, I sleep with my dog too. |
01:28.03 | dos000 | supaigtr, but its funny how all things ser related are still discussed behind doors. chec the #ser for example |
01:28.08 | Vlat- | dos000: sems (f)worked at me...for 15 minutes |
01:28.21 | HiltonT | NetSkier; I think I slept with your dog, too :) |
01:28.22 | supaigtr | dos000: True. |
01:28.31 | Vlat- | dos000: then I removed the package and installed back asterisk |
01:28.32 | brookshire[home] | wow this channel is getting freaky |
01:28.37 | HiltonT | lol |
01:28.58 | Vlat- | dos000: SEMS can do nothing right now. To be honest, it's not beta. It's the pre-alpha |
01:29.05 | Vlat- | look at the CVS |
01:29.09 | dos000 | Vlat-, thanks |
01:29.31 | NetSkier | HiltonT: Well, my dog is a male, if that helps at all. ;0 |
01:29.40 | NetSkier | ;) |
01:29.48 | Vlat- | so, i'll say once more |
01:29.53 | supaigtr | Vlat: know of any publicly accessable example of ser, tnt, and asterisk? |
01:29.55 | brookshire[home] | that's hot |
01:29.56 | Vlat- | SER if for signalling |
01:30.04 | Vlat- | Asterisk is for media |
01:30.14 | Vlat- | supaigtr: www.onsip.org |
01:30.22 | dos000 | supaigtr, do you have tnt in your hand ? |
01:30.27 | Vlat- | supaigtr: completely working example |
01:30.34 | supaigtr | Yep. Bunch of them. |
01:30.41 | brookshire[home] | asterisk is also for in-house pbx :) |
01:30.42 | supaigtr | Left over from dial up switching to PON. |
01:30.48 | dos000 | supaigtr, check voip-info.org as well |
01:30.51 | Vlat- | o!!! |
01:30.57 | supaigtr | Vlat:Close I can get. |
01:31.14 | Vlat- | The CISCO Hungary motherfuckers brought a beer for me |
01:31.26 | supaigtr | I read the article and have it working with * no problems. Just turn off auth but SER is where i have problems. |
01:31.51 | Vlat- | rather bring me a $12000 for hardware-testing |
01:32.11 | dos000 | supaigtr, what kind of problems. I am about to buy tnts myself |
01:32.50 | *** join/#asterisk nnnnnn (n=killfill@pc-200-74-17-222.asturias2.pc.metropolis-inter.com) |
01:32.51 | Vlat- | supaigtr: let's examine. asterisk cand handle x connections at amount of time as router. SER handles x*500 connections at the same configuration |
01:32.53 | nnnnnn | hi.. |
01:33.02 | supaigtr | Pain to get setup. U'll want someone who know TNT. I'll be glad to help of course. It took like a week to got thru the millions of options to get it working. |
01:33.07 | Vlat- | i'm not asterisk addict, neither se |
01:33.08 | Vlat- | r |
01:33.30 | dos000 | supaigtr, did you had someone in here helping ? |
01:33.34 | supaigtr | vlat: I just want to use * for the VM and meetme. 10,000 account in SER / TNT and 1000 or so VM. |
01:33.45 | Vlat- | i just own the company in EU. and ser+asterisk is the most cost-effective solution right now |
01:33.48 | supaigtr | dos000: Nobody in here except BKW he helped. |
01:33.55 | dos000 | supaigtr, i would really like to get in touch with you if it no a problem |
01:34.05 | Vlat- | 'coz Cisco5350 is $20000 |
01:34.16 | nnnnnn | i have a abox directly connected to the internet. if i make x-lit connect to the public IP the phone doesnt recieve any sound, but, when i connect to 127.0.0.1 it works.. |
01:34.30 | supaigtr | No problem. I'd really like some extra help getting everything working with ser. The TNT is noauth so I've got that part. |
01:34.33 | nnnnnn | how can i see where the problem is?.. |
01:34.56 | dos000 | supaigtr, did you buy it used ? |
01:34.57 | supaigtr | dos000 U getting tnt for ser or *? |
01:35.08 | Vlat- | the only problem - i have to moderate myself sometimes |
01:35.11 | dos000 | supaigtr, ser+asterisk for vmail ! |
01:35.19 | supaigtr | dos000: New. They were used for dialup and fax applications. |
01:35.26 | supaigtr | We have the same goal. |
01:35.28 | Vlat- | it's asterisk channel, and i don't want to make someone cry :) |
01:36.06 | dos000 | supaigtr, mind to take this in pv ? |
01:36.12 | supaigtr | k |
01:36.36 | Vlat- | when asterisk will get the real REGISTAR, we'll change to it |
01:36.50 | Vlat- | SIP REGISTAR |
01:37.02 | Vlat- | currently it's sip1.1 i guess |
01:37.08 | supaigtr | I kinda think it should stay seperated. |
01:39.31 | Vlat- | Asterisk is the PBX solution, and it do it at the best. I just don't understand the people trying to get it to ITSP level |
01:39.52 | supaigtr | Right. |
01:40.29 | supaigtr | Theres alot to do on the PBX side before trying to replace ser. Also I have problem using * as TDM gateway. TNT hardware works much better. |
01:41.01 | Vlat- | SER is just a proxy |
01:41.09 | Vlat- | Asterisk+TDM <> SER |
01:41.12 | Vlat- | works fine |
01:41.23 | Vlat- | but rather borrow a C5300 |
01:41.32 | supaigtr | echo, interrupts are a problem. TNT = no problem |
01:41.37 | Vlat- | 5300 is onlu $11000 now |
01:42.31 | supaigtr | :) |
01:42.32 | Vlat- | so I don't really understand |
01:43.08 | Vlat- | asterisk maillist is full of replies like "Our customers aren't sastificated with our service" |
01:43.14 | *** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
01:43.33 | Vco | i belive SER is actually a "router" not a proxy |
01:43.35 | Vlat- | damned god, and who told you Asterisk is business-grade solution??? |
01:43.55 | Vlat- | Vco: you're right, i didn't choise a right expression |
01:44.15 | Vlat- | use ser as router/proxy. use Asterisk as the backing end |
01:44.17 | MikeJ[Laptop] | plenty of business use asterisk |
01:44.19 | Vlat- | or maybe SEMS |
01:44.30 | MikeJ[Laptop] | just use it right |
01:44.33 | Vco | asterisk suxours as SIP routing compared to SER :( |
01:44.37 | Vlat- | use Cisco equipment |
01:44.42 | Vlat- | maybe Vocaltek |
01:45.08 | Vlat- | but just to make it sure. SW solutions never will reach the HW |
01:45.25 | MikeJ[Laptop] | asterisk isn't a proxy, it's not meant to be... if you are doing high volume sip termination, yeah use ser.. ser and asterisk together allow you to do some good stuff |
01:45.37 | Vlat- | Btw, aster1.2 SIP is much more than buggy |
01:45.44 | Vco | i mean even simple concepts like multiple SIP registrations etc.. |
01:45.55 | Vlat- | see it before |
01:46.06 | Vlat- | Ast. is the best PBX we had found |
01:46.13 | Vlat- | even hardware ones |
01:46.25 | Vco | hell ya... |
01:46.30 | Vlat- | but like ITSP-grade solution it's nothing |
01:47.00 | Vlat- | it's the way I hate SER and I like SER |
01:47.02 | Vlat- | it's config |
01:47.06 | Vco | ya |
01:47.10 | supaigtr | I've had more than one customer revolt on * and buy a lucent or panasonic. |
01:47.14 | Vlat- | i can do virtually everything with it |
01:47.44 | brookshire[home] | supaigtr: i would love to know the outcomes of that, are they happier with that? |
01:47.47 | *** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
01:47.58 | Vlat- | i have the except customer ? ok. i'll write the codes to config and the customer's HW will work |
01:48.16 | chidex | I take it that asterisk 1.0.9 is reasonably stable? |
01:48.21 | Vlat- | in the case of Asterisk we'll need to submit CVS change and wait for a month |
01:48.27 | Vco | anyone else have problems with Asterisk <-> Asterisk SIP and DTMF? |
01:48.35 | supaigtr | brookshire: Most are happy with the pana. Not the lucent. But * has lots of problems in the business world. |
01:48.37 | Vlat- | chidex: more than stable |
01:48.49 | Vlat- | chidex: using it at primary |
01:49.30 | *** join/#asterisk forrestc{hm} (n=forrestc@206.127.77.82) |
01:49.42 | forrestc{hm} | Hello everyone. |
01:49.43 | chidex | vlat: didn't get that last bit |
01:49.51 | forrestc{hm} | Got a bizzare one tonight |
01:50.12 | Vlat- | chidex: pastle it in, please |
01:50.12 | nnnnnn | hey Vlat-, when my software sip phone connect to asterisk, but doesnt get any sound from it, i.e. the ansear mashine voice record, what does it mean?.. |
01:50.31 | clyrrad | What is the best codec to use for IAX to IAX communication? |
01:50.33 | nnnnnn | it can send sounds.. |
01:50.41 | Vco | is it behind nat? |
01:50.45 | Vlat- | nnnnnn: set your phones to use nat+stun. btw, grandstream ? |
01:50.46 | forrestc{hm} | nnnnn: probably a firewall issue, assuming you don't have an audio problem |
01:50.53 | forrestc{hm} | clyrrad: it depends |
01:50.57 | clyrrad | on what? |
01:50.59 | Vlat- | nat=yes usually helps |
01:51.03 | brookshire[home] | clyrrad: inhouse or over the internet? |
01:51.07 | clyrrad | over internet |
01:51.16 | brookshire[home] | gsm or g729 |
01:51.21 | forrestc{hm} | clyrrad: bandwidth versus quality versus packet loss. |
01:51.23 | nnnnnn | hm.. Vlat- you mean on sip.conf? |
01:51.37 | forrestc{hm} | clyrrad: I personally use ulaw for everything. |
01:51.59 | forrestc{hm} | clyrrad: no transcoding issues, better quality, no license issues, etc. |
01:52.04 | Vlat- | nnnnnn: 1) set up ip forwarding on customer routers 2) set up stun server at customer hardware to stunserver.org |
01:52.05 | clyrrad | I have g711u right now, makes a strange sound until the call is connected for some odd reason |
01:52.15 | Vlat- | and don't forget about DNS |
01:52.17 | forrestc{hm} | clyrrad: but takes more bandwidth. |
01:52.22 | HiltonT | xorcom rapid: what are the default passwords for the extensions, and how are these change? |
01:52.39 | Vlat- | if you want just to do it work at everyone |
01:52.45 | brookshire[home] | hilton: you mean for voicemail? |
01:52.47 | Vlat- | than forget about it |
01:52.57 | HiltonT | nope, to attach my SIP hardphone to it |
01:53.03 | Vlat- | it's impossible even at B2BUA |
01:53.07 | HiltonT | and then later, for voicemail :) |
01:53.28 | forrestc{hm} | vlat: I've had almost zero problems with the sipura stuff with nat=on and no stun. |
01:53.37 | clyrrad | forrestc{hm}.... Any idea why some IAX to IAX calls pass the Caller ID and other show up as Anonomous? If I call from the same number to the same source some times the CID will show up othertimes it wont..... |
01:53.53 | Vlat- | forrestc{hm}: and the local network ? :)) |
01:54.03 | brookshire[home] | clyrrad: that is a configuration problem ;) |
01:54.13 | forrestc{hm} | vlat: we're an ISP with dozens of different types of customer routers. |
01:54.18 | brookshire[home] | you can set the caller id to be whatever usually |
01:54.33 | Vlat- | forrestc{hm}: NAT user trying to call another NAT user. 192.168.0.155 trying to call 10.0.0.221 |
01:54.35 | forrestc{hm} | vlat: Asterisk is on a real IP address which helps |
01:54.44 | forrestc{hm} | vlat: oh.. That does suck. |
01:54.49 | brookshire[home] | if no caller id specified.. then anonymous |
01:54.56 | clyrrad | brookshire[home].... Its kind of hard to spot a configuration issue with this when some times it works and other times it wont under the same testing conditions. |
01:55.03 | clyrrad | Could it be CVS HEAD? |
01:55.04 | Vlat- | forrestc{hm}: we had solved it by the ser way |
01:55.23 | Vlat- | forrestc{hm}: asterisk does the rtp proxying, when it needed |
01:55.32 | Vlat- | +2 line to config |
01:55.44 | brookshire[home] | intresting.. do you control the end that is giving you an anonomous back? |
01:55.59 | clyrrad | Yes I control both |
01:56.08 | supaigtr | clyrrad: It does that. I have the same problem. Sometimes it send CID sometimes not. |
01:56.11 | Vlat- | so I have no idea |
01:56.15 | forrestc{hm} | vlat: we basically decided just to use asterisk as a B2BUA and our PSTN gateway (actually a farm of asterisk boxes). |
01:56.30 | clyrrad | supaigtr.... Are you using CVS HEAD as well? |
01:56.35 | Vlat- | forrestc{hm}: to be honest - have no idea |
01:56.56 | brookshire[home] | intresting.. i'll look into it |
01:56.59 | Vlat- | forrestc{hm}: we're using SER as proxy. It have more than flexible config to do everything |
01:57.02 | brookshire[home] | if you can reproduce it |
01:57.11 | brookshire[home] | i would post it to the bug tracker |
01:57.11 | Vlat- | forrestc{hm}: so we solve our problems with SER |
01:57.32 | Vlat- | when we need the MEDIA we forward to ser |
01:57.34 | clyrrad | brookshire[home] was that message to me? |
01:57.38 | Vlat- | for example for 404 |
01:57.40 | brookshire[home] | yes sir |
01:57.42 | supaigtr | clyrrad: Yep. |
01:57.42 | brookshire[home] | :) |
01:57.51 | brookshire[home] | vlat: we get it.. ser is great for sip :) |
01:58.15 | clyrrad | How to reproduce it is a damn good question LOL.... there is no rhyme or reason to it that I can see, sometimes it works and other times it wont |
01:58.25 | Vlat- | [default] exten 404 on asterisk say "User is not available, press 1 to leave the message" |
01:58.32 | clyrrad | supaigtr.... have you had any luck finding the EXACT condition that causes it to happen? |
01:58.34 | forrestc{hm} | clyrrad: have you looked in the Asterisk console with debugging/verbose turned up |
01:58.48 | supaigtr | clyrrad: I have a simlar intermittent problem with muting audio in one direction as well with iax |
01:59.08 | Vlat- | press 2 to call it again (+call time to my billing( |
01:59.20 | brookshire[home] | brb |
01:59.21 | Vlat- | +3 to contact the supprot team |
01:59.24 | HiltonT | Xorcom seems to have installed fine and be running fine, but NFI how to add/remove extensions, nor how to register my hardphone to ext 400 |
01:59.25 | Vlat- | (payable) |
01:59.32 | clyrrad | forrectc{hm}, yes when CID is not sent I do not see anyting different on the CLI and I am running about 10 v's when I load * |
01:59.55 | forrestc{hm} | clyrrad: have you done a "set debug 255" |
02:00.04 | clyrrad | supaigtr are you using CVS head? |
02:00.14 | Vlat- | Asterisk = PBX. The guy wrote SIP module for it it's a fat german guy |
02:00.15 | supaigtr | Yep. |
02:00.25 | Vlat- | (2 month ago it's married man) |
02:00.33 | Vlat- | so don't expect a lot |
02:00.33 | clyrrad | forectc{hm} i just did that now |
02:00.35 | NetSkier | HiltonT: try asteriskguru.com for some phone configuration hints. |
02:00.41 | *** part/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
02:01.05 | NetSkier | HiltonT: google for asteriskguru if I got the url wrong. |
02:01.11 | HiltonT | just don't know the password set by default in Xormcom Rapid, and apparently little documentation, but reading that site... |
02:01.26 | HiltonT | (I know how to config this phione fine, just not Rapid) |
02:01.34 | NetSkier | HiltonT: basicly edit /etc/asterisk/extensions.conf |
02:01.36 | forrestc{hm} | clyrrad: I'm trying to figure out how to spit a debug out of some sort with the CID Just before the dial. |
02:01.45 | HiltonT | k, editing now... |
02:01.54 | supaigtr | noop |
02:01.57 | forrestc{hm} | clyrrad: how often does this occur? |
02:01.59 | clyrrad | forrestc{hm} even with that debug set the same thing is happening, with no errors on the CLI |
02:02.05 | clyrrad | randomly |
02:02.10 | clyrrad | but all the time |
02:02.15 | clyrrad | if that makes sense* |
02:02.27 | forrestc{hm} | clyrrad: what did you say the calling source was? |
02:02.31 | clyrrad | What I mean is every time I call it will happen, but the odd time the CID will work |
02:02.47 | clyrrad | its IAX to IAX |
02:03.35 | chidex | vlat: so are you using 1.0.9 in a production environment then? |
02:03.43 | supaigtr | clyrrad: U had any audio problems? Not often but it happens. 1 -3 sec audio lost on far end IAX? |
02:03.58 | forrestc{hm} | clyrrad: an IAX to IAX trunk, or actually using an IAX ATA or phone. |
02:03.59 | Vlat- | chidex: yes |
02:04.26 | clyrrad | Yes i have Audio problems like a JITTER or a lag at the beginning of every call, takes about 3 seconds, then its gone for the remainder of the call, its like it is doing handshaking of some kind |
02:04.38 | chidex | vlat: did you apply any custom patches? |
02:04.46 | clyrrad | I have an * box, and an IAX ATA |
02:05.08 | forrestc{hm} | clyrrad: so you basically want the CallerID to work on the IAX ATA. |
02:05.10 | Vlat- | chidex: nothing |
02:05.17 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
02:05.22 | chidex | vlat: cool |
02:05.36 | forrestc{hm} | clyrrad: is the a call from * to the ATA or from the ATA to the *? |
02:05.36 | clyrrad | Yes |
02:05.40 | chidex | vlat: how big is your setup? |
02:05.54 | Vlat- | chidex: 1000+ customer |
02:05.58 | clyrrad | The call is from my cell phone to the * box which sends to the ATA |
02:06.17 | forrestc{hm} | clyrrad: how are you getting the call into the * box? |
02:06.18 | chidex | vlat: oh yeah, using ser! :) |
02:06.28 | clyrrad | from my VOIP provider |
02:06.53 | forrestc{hm} | clyrrad: so you're taking a did from a voip provider into your * box and passing onto an IAXy or similar? |
02:07.01 | clyrrad | exactly |
02:07.09 | Vlat- | chidex: it make no differnece for me. if ser is working for that setup - it's OK. if * would work - that would be OK |
02:07.27 | forrestc{hm} | clyrad: what protocol are you talking to your voip provider? IAX also? |
02:07.32 | Vlat- | there's no time for the holy wars |
02:07.33 | clyrrad | yes |
02:07.46 | clyrrad | IAX to the VOIP provider to Me, then from Me IAX to the ATA |
02:07.58 | Vlat- | fuck |
02:08.14 | forrestc{hm} | clyrad: have you done anything which indicates that you are actually receiving CID On the * box every time? |
02:08.20 | Vlat- | Uncensored Terminator II it's absoluterly different movie !!! |
02:08.43 | clyrrad | sorry didnt follow your last question |
02:09.10 | Vlat- | about IAX and ATA |
02:09.16 | NetSkier | Vlat-: I have not heard of the uncensored version. Where can I get it? |
02:09.27 | Vlat- | if you have 100+ customers you would like to do it |
02:09.34 | Vlat- | - don't do it |
02:09.42 | forrestc{hm} | clyrad: there are two places this could be broken... either the cid from your provider, or cid from * to the ATA. |
02:09.57 | Vlat- | there're several problemms with IAX |
02:10.04 | Vlat- | for example E164 |
02:10.12 | forrestc{hm} | clyrad: what I'm wondering is if you know whether or not you are getting the CID On the * box from the provider each time |
02:10.26 | clyrrad | hrm.... its a good question |
02:11.03 | forrestc{hm} | Hold on.. |
02:11.11 | Vlat- | NetSkier: i have it here. and i heared neither. but it's here!!! and the TII movie end do not stops with T1 melting |
02:11.19 | HiltonT | hhmmm, seems no simple (as in straightforward) way to edit the files in Xorcom Rapid (extensions, for example) and no docs, and no idea what the default passwords are, and no way to connect my hardphone to ext 401, for example :( |
02:11.33 | forrestc{hm} | clyrrad: try a iax2 debug on the command line. |
02:11.36 | Vlat- | NetSkier: there're a 15 minute trail by Connor's thinks |
02:11.58 | clyrrad | ok done |
02:12.02 | clyrrad | and got an annon that time |
02:12.24 | forrestc{hm} | clyrrad: did you get the anon from the iax box? |
02:12.31 | forrestc{hm} | er iax debug? |
02:12.49 | clyrrad | I called from a phone connected to the * box to my cell phone that time and got Annon on my cell CID |
02:12.50 | NetSkier | HiltonT: I booted it into single user to set the root password. Then booted into multiuser mode. |
02:13.11 | HiltonT | I'm ssh'ed into it |
02:13.21 | forrestc{hm} | clyrrad: that was backwards from what I heard you say before. |
02:13.36 | NetSkier | HiltonT: Then I used the editor nano, a super small emacs, to edit /etc/apt/sources.list to add the regular debian repositoriers. |
02:13.37 | clyrrad | yes, right now I tried it the other way around |
02:13.50 | forrestc{hm} | clyrrad: both ways work differently |
02:13.56 | forrestc{hm} | clyrad: different set of issues. |
02:14.01 | clyrrad | that what im checking right now :) |
02:14.10 | HiltonT | of the 3 distros - Astlinux, A@Home (and its latest Beta) and Xorcom Rapid, at least Rapid actually starts running Asterisk!!! |
02:14.22 | NetSkier | Then I became root, and ran apt-get'ed some packages, like emacs, and whatever else I wanted. |
02:14.27 | clyrrad | seems to work evrey time when i call from cell phone to the IAX box |
02:14.32 | HiltonT | I don't want to make it a full Debian system, I want some docs on how to drive the distro! |
02:14.36 | clyrrad | Asterisk box i mean * |
02:14.43 | NetSkier | Then, with my favorite editor, I edited /etc/asterisk/extensions.conf. |
02:14.45 | HiltonT | it has vim on it - I'm happy with vim |
02:14.49 | forrestc{hm} | clyrrad: you get cid every time from the cell phone. |
02:14.54 | forrestc{hm} | clyrrad: right? |
02:14.55 | NetSkier | great; use vim then. |
02:15.04 | NetSkier | but you need to be able to become root. |
02:15.12 | Vlat- | NetSkier: also, the guys are messing tottally (shocker, rude bite for example) with Mrs. Connor at the start. There no such thing in the uncens movie |
02:15.14 | forrestc{hm} | clyrrad: doesn't work from the phone to the cell phone. |
02:15.19 | NetSkier | to edit those files, IIRC. |
02:15.22 | HiltonT | I ssh'ed in as root :( |
02:15.32 | clyrrad | Yes, every time i call from my cell phoen to the * box CID works, but this is a SIP phone connected directly to the * box, thats the one that always gets the CID |
02:15.34 | NetSkier | ok |
02:15.43 | clyrrad | so I think your right, the problem is between * and the IAX box |
02:15.50 | clyrrad | not with the VOIP provider |
02:15.55 | Nugget | no, it's his left. |
02:16.08 | forrestc{hm} | clyrrad: does the iax box ever work? |
02:16.20 | Vlat- | about IAX... it has no native CID |
02:16.35 | Vlat- | in the latest revisions it was diverted |
02:16.36 | HiltonT | the extensions.conf in /etc/asterisk doesn't match the extensions shown in its console UI :( |
02:16.46 | Vlat- | but still to buggy imho |
02:17.09 | clyrrad | The IAX box works in the sense that i can send and receive calls, but its CID never works when I call from the IAX box to my cell phone |
02:17.35 | forrestc{hm} | clyrrad: are you sure you have callerid set right int he iax.conf? |
02:17.36 | *** join/#asterisk wolfson (i=hehe@nc-71-2-31-68.dyn.sprint-hsd.net) |
02:17.50 | clyrrad | woops that was a lie, it just worked this time |
02:18.05 | clyrrad | that was the first time I got CID off the IAX box |
02:18.17 | forrestc{hm} | clyrrad: could you let me know what happens in these cases: |
02:18.20 | clyrrad | and I have made no changes to the configs |
02:18.23 | forrestc{hm} | SIP ATA Call to cell phone |
02:18.28 | forrestc{hm} | IAX ATA Call to cell phone |
02:18.33 | forrestc{hm} | Cell phone call to SIP ATA |
02:18.38 | forrestc{hm} | Cell phoen call to IAX ATA |
02:18.57 | forrestc{hm} | SIP ATA might mean SIPphone if it's not an ATA> |
02:19.03 | clyrrad | Ok SIP to Cell I get random if the CID will work or not |
02:19.22 | clyrrad | Same with IAX ATA to Cell, its random if it will work or not |
02:19.31 | clyrrad | Cell phone to SIP ATA works EVERY time |
02:19.32 | Vlat- | XXXX <-> AABBCC |
02:19.41 | clyrrad | Cell phone to IAX works Randomly |
02:20.00 | forrestc{hm} | clyrrad: one more: SIP phone to IAX phone, |
02:20.11 | Vlat- | forget about getting the proper CID every time |
02:20.12 | forrestc{hm} | on and one more after that: iax phone to sip phone. |
02:20.37 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
02:20.40 | clyrrad | Nope it sais NO Data when i do that |
02:20.48 | Vlat- | CallerID is the unstardatesising piece of shi... the thing sometimes don't work |
02:20.51 | forrestc{hm} | vlat: I'm about ready to tell him the same thing... sounds like his provider is screwing with CID. |
02:21.02 | Vlat- | for example |
02:21.02 | NoRemorse | hello, has anyone got a h323 gateway O can test some calls to please? trying to debug my openh323 setup here |
02:21.08 | Vlat- | US has the own CID format |
02:21.22 | NoRemorse | *I |
02:21.23 | Vlat- | Russia don't evem has CID, but it's working |
02:21.44 | Vlat- | fucking Europe has something standartized, but it doesn't works |
02:21.48 | forrestc{hm} | clyrrad: what do you have in your sip.conf and iax.conf |
02:22.02 | Vlat- | Korea had found the OWN CID |
02:22.16 | clyrrad | well in SIP.conf I dont have anyting for the IAX phone or ATA if that is what you mean |
02:22.17 | Vlat- | Japan can convert from US CID |
02:22.32 | clyrrad | and in IAX.conf i have the register that registeres the DID with my provider |
02:22.35 | Vlat- | and there're only the primary countries |
02:22.46 | forrestc{hm} | clyrrad: no, I'm just wondering how/where you are setting the CID for the ATA's. |
02:22.46 | clyrrad | Vlat... sounds like we have a real mess huh |
02:22.56 | Vlat- | it's a REAL mess |
02:23.07 | clyrrad | Oh, the CID is being passed from my VOIP provider |
02:23.23 | Vlat- | i have about 10 exceptions for CID-s in my ser.cfg\ |
02:23.26 | clyrrad | so then i do this SetCallerID(<${CALLERIDNUM}> |
02:23.28 | forrestc{hm} | vlat: In all reality, the US CID system seems to work pretty well as long as everything is provisioned for it. |
02:23.29 | clyrrad | in the dial plan |
02:23.45 | forrestc{hm} | clyrrad: have you tried hardcoding $CALLERIDNUM |
02:23.47 | Vlat- | forrestc{hm}: US CID works 100% in US |
02:23.48 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
02:24.12 | clyrrad | forrestcm{hm} yes i get the exact same results some times it works and other times it does not |
02:24.18 | Vlat- | i will tell the other thing ;) |
02:24.21 | clyrrad | I thought i may be a CVS HEAD bug |
02:24.28 | Vlat- | There's the Elcatel |
02:24.41 | Vlat- | the ATC manufacture operator |
02:25.03 | Vlat- | there's the Alcatel too |
02:25.08 | forrestc{hm} | clyrrad: I suspect more likely that your provider is not honoring CID from you for some reason. |
02:25.12 | Vlat- | and there's the Ericcson |
02:25.12 | forrestc{hm} | clyrrad: which provider? |
02:25.22 | clyrrad | unlimitel |
02:25.31 | Vlat- | also there're about 5-th of Chineese ATC providers |
02:25.58 | Vlat- | and EVERY OF THEM is incompatible at CID level with each other |
02:26.12 | forrestc{hm} | clyrrad: see http://www.voip-info.org/tiki-index.php?page=Unlimitel |
02:26.18 | clyrrad | Vlat-> I see that has been frustrating you for some time huh? :P |
02:26.22 | Vlat- | personally me had to check this, so i know what about i speaking |
02:26.48 | Vlat- | clyrrad: i'm rather hardware-man than SW-man...so of course it's frustating for me :)) |
02:27.12 | clyrrad | forrestc{hm} LOL |
02:27.15 | clyrrad | Nice link |
02:27.25 | clyrrad | here I am pulling my hair out over my configuration |
02:27.39 | clyrrad | I wonder why that is a problem |
02:28.00 | Vlat- | HW-man means 3-floor ATC + Ser :)) |
02:28.03 | forrestc{hm} | clyrrad: they probably don't use PRI or SS7 or similar for their trunks, which don't support CID. |
02:28.25 | Vlat- | atc can not do a trick in this case |
02:28.34 | forrestc{hm} | I said that wierd... but you get the idea hopefully. |
02:28.52 | Vlat- | by the way |
02:29.09 | forrestc{hm} | On the ATA side, I suspect it *might* be an issue where the ATA isn't 100% callerid-friendly or there's a setting to get it to pass CID Correctly, etc. |
02:29.16 | Vlat- | CID is a little circuit in the end-customer's ATC |
02:29.41 | Vlat- | if it's inside ATC, you can do any thing |
02:30.07 | forrestc{hm} | clyrrad: might also be the CID box if it's really an ATA. |
02:30.10 | clyrrad | The ATA I am using does not seem to have any such setting its a GNET VP168I not sure if anyone is farmiliar with that one or not |
02:30.15 | Vlat- | but it'll replace the append_rpid_hf by the own nubmer |
02:30.33 | Katty | mew. |
02:30.35 | forrestc{hm} | clyrrad: not here, but like vlad indicates CID sometimes is a mess. |
02:30.53 | clyrrad | LOL, yup and i think this is a perfect example |
02:30.54 | forrestc{hm} | Hmmm.. back to my bizzare SIP problem myself. |
02:31.00 | clyrrad | thanks for helping me get to the bottom of it |
02:31.08 | clyrrad | that link was the final proof that my configs are fine |
02:31.38 | clyrrad | But there is a seperate issue with the ATA not sending and receiving caller ID |
02:31.42 | Katty | mew? |
02:32.07 | Vlat- | btw, that circuit has Atmel ROM :) $xxxx and we can reset it :))))) |
02:32.22 | Katty | let's trying rephrasing. |
02:32.24 | Katty | hi! |
02:32.31 | supaigtr | Sup. |
02:32.32 | Katty | obviously no one speaks kat around here |
02:32.54 | Vlat- | to be serious |
02:33.11 | Vlat- | the last-end GW forms the CID |
02:33.28 | Vlat- | if it don't want to do - the user receive CID you set up |
02:33.54 | Vlat- | if ANY GW at the path trying to change CID - you have nothing to do with it |
02:33.57 | forrestc{hm} | Is anyone aware of a SIP call audio distortion issue with HEAD? |
02:34.06 | forrestc{hm} | ulaw codec |
02:34.44 | clyrrad | forrest what kind of distoration is it? Like a LAG or JITTER? or just scrambled audio? |
02:34.52 | forrestc{hm} | scrambled audio. |
02:35.03 | Vlat- | forrestc{hm}: what doest the ulaw have to do with sip ? |
02:35.07 | Vlat- | -t |
02:35.10 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
02:35.15 | forrestc{hm} | calls *all* start out fine but get more and more distorted. |
02:35.24 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
02:35.36 | forrestc{hm} | vlat: you can run different codecs on sip |
02:35.40 | Brijn | Good evening all |
02:35.58 | Vlat- | forrestc{hm}: let's start it again: what equipment have you got. CISCO? Vocaltec? PC with SER ? |
02:36.00 | forrestc{hm} | vlat: i.e. ulaw (aka 711u), 729, etc. |
02:36.14 | Vlat- | forrestc{hm}: of course |
02:36.19 | forrestc{hm} | vlat: I have an asterisk box with a 4 port digium card. |
02:36.31 | forrestc{hm} | vlat: I am passing *MODEM* calls through this asterisk box with no problems. |
02:36.44 | *** join/#asterisk crash3m__ (n=crash3m@unaffiliated/crash3m) |
02:36.48 | Vlat- | forrestc{hm}: codec is codec. signalling is signalling. billing is billing. etc.... |
02:36.52 | forrestc{hm} | SIP calls to/fron the asterisk box sound fine, then slowy get worse. |
02:37.06 | Vlat- | dos000: #ser is dead |
02:37.37 | forrestc{hm} | digium card means digium T1 card aka TE405. |
02:37.45 | dos000 | Vlat-, long live to <> ? |
02:37.49 | JamesDotCom | Vlat-: so :~ |
02:37.52 | Katty | wickets. |
02:38.02 | JamesDotCom | still need to build the channel up |
02:38.08 | Vlat- | dos000: of course i can tell the * is the ....!!!! ..........!!! ..!!!! |
02:38.19 | clyrrad | forestc{hm} does the distortion happen all the time no matter how many calls the PBX is taking on? |
02:38.21 | Vlat- | but it wouldn't be the true |
02:38.25 | forrestc{hm} | clyrrad: yep. |
02:38.37 | clyrrad | so even with 1 call in and out you get this problem? |
02:38.40 | forrestc{hm} | and every call starts out fine. |
02:38.40 | dos000 | Vlat-, long live to openser ? |
02:38.43 | forrestc{hm} | clyrrad: yep. |
02:38.48 | Vlat- | forrestc{hm}: are you serious???? |
02:38.52 | clyrrad | how long is it before they go sour? |
02:38.53 | Vlat- | the MODEM ????!!!!! |
02:38.59 | forrestc{hm} | clyrrad: different. |
02:39.04 | Vlat- | dos000: never tried openser |
02:39.05 | forrestc{hm} | clyrrad.. usually a few seconds. |
02:39.13 | clyrrad | thats what I was goning to suggest too the modem or the card may be bad |
02:39.34 | forrestc{hm} | clyrrad: *VOICEMAIL* calls to the asterisk box do the same thing. |
02:39.36 | clyrrad | do you have another card you can try to rule out defective hardware? |
02:39.58 | clyrrad | always from the same phone? or with any phone? |
02:40.01 | tamp4x | what u trying to do with ser |
02:40.12 | Vlat- | modem with 10+... |
02:40.13 | forrestc{hm} | Different ATA's... and were working *fine* with an earlier version of asterisk. |
02:40.16 | Vlat- | interesting |
02:40.20 | Katty | omgwtfulolzkthxbi |
02:40.22 | tamp4x | i just set up a stateless roundrobin router with it |
02:40.29 | Vlat- | but if the proto is IAX |
02:40.32 | forrestc{hm} | I'm also passing 56K modem calls at 53K through the T1 ports. |
02:40.35 | Vlat- | it would suck well |
02:40.53 | forrestc{hm} | Got 2 T1's attached to Telco PRI's and 2 T1's attached to the modem bank... all is well. |
02:40.57 | tamp4x | i had one of my programmers write an iax2 router =] |
02:41.35 | clyrrad | hrm.... that is strange, so different ATX's and different phones, all produce the same audio problem even if the calls are not going accross your digium card? |
02:41.49 | forrestc{hm} | Heck I have one caller which has been on a modem call 2:15 between two channels on teh digium card. |
02:41.52 | forrestc{hm} | clyrrad: yep. |
02:42.04 | forrestc{hm} | clyrrad: and what's weird is it sounds codec related. |
02:42.18 | forrestc{hm} | clyrrad. I.E. I get "phonetic" distortion, not loss. |
02:42.20 | clyrrad | I would agree |
02:42.34 | clyrrad | does sound like codec to me too |
02:42.38 | *** part/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
02:42.40 | clyrrad | I wonder if you can re-install the codec's |
02:42.42 | forrestc{hm} | but I'm running 711u, which isn't really a lossy codec. |
02:42.51 | forrestc{hm} | it's either there or it's not. |
02:43.09 | clyrrad | you compiled * right? |
02:43.13 | forrestc{hm} | yes. |
02:43.13 | clyrrad | from CVS |
02:43.19 | forrestc{hm} | yes. |
02:43.20 | clyrrad | when? |
02:43.29 | forrestc{hm} | last 48 hours. |
02:43.48 | forrestc{hm} | let me get a version. |
02:43.49 | clyrrad | have you checked for any updates since then? |
02:43.57 | forrestc{hm} | somewhat.. |
02:44.04 | forrestc{hm} | looked at the CVS Commits. |
02:44.11 | clyrrad | how are you checking your cvs version? |
02:44.13 | forrestc{hm} | didn't see anything seemingly related. |
02:44.54 | FuriousGeorge | anyone ever seen fax detection not working despite the user setting it up right? is there some bug i dont know about |
02:45.06 | clyrrad | Fax over VOIP? |
02:45.25 | FuriousGeorge | clyrrad: no |
02:45.28 | FuriousGeorge | pots |
02:45.40 | forrestc{hm} | clyrad: from the file i touch when I do a cvs |
02:45.41 | clyrrad | sorry cant help u on that one |
02:45.58 | forrestc{hm} | October 10th 21:39 |
02:46.10 | forrestc{hm} | MDT |
02:46.13 | clyrrad | My CVS was from October 1st |
02:46.17 | clyrrad | take your CVS from there |
02:46.41 | clyrrad | becase we know I have that CVS, and I dont have the problem your having.... |
02:46.57 | clyrrad | so if you date back then and get your files from there and still have the same problem then it must be hardware |
02:47.47 | forrestc{hm} | I'll play with that a bit later tonight. |
02:48.18 | clyrrad | its the only other thing i can think to suggest, becase I am not sure if you can manually re-install the codecs to see if that is the problem |
02:48.54 | forrestc{hm} | Before I do that I'm going to see if I can find a tool which will let me grab the RTP packets and play them back "non-streamed" to determine if it's bad data being sent by asterisk or if it's a network issue. |
02:49.44 | forrestc{hm} | I'll write a quick chunk of perl to strip the ulaw stream out of a tethereal dump if I can't find a tool to do it automatically. |
02:50.46 | crash3m__ | I know one exists to rip some type of stream, wanting to say they called it a forensic tool heh |
02:50.48 | clyrrad | do you have alot of traffic accross your network? |
02:50.58 | *** join/#asterisk cwetter (n=cwetter@68-114-46-75.dhcp.stbr.ga.charter.com) |
02:51.27 | clyrrad | if not and this is all on your internal LAN you should have to worry about a network issue |
02:51.39 | clyrrad | should'nt* |
02:51.55 | supaigtr | forreestc: You can tell if network is bad with ipperf. |
02:52.30 | cwetter | when I run modprobe wcfxo I get 'ZT_CHANCONFIG failed on channel 4: No such device or address (6)'. What do I do? |
02:53.03 | forrestc{hm} | clyrrad: I really doubt it's a network issue. If it is it's related to the network card in machine, not the transport to the SIP Devices. |
02:53.48 | forrestc{hm} | hey this looks promising... rtpdump |
02:53.57 | supaigtr | forrestc: ipperf can tell you if UDP iax2 traffic can make it from point a to b realiably. |
02:54.59 | cwetter | Is s110m an FXO or FXS Card? |
02:55.18 | clyrrad | forrestc are you using VOIP at all with your Asterisk box? |
02:55.38 | crash3m__ | wtf else would you be doing besides VoIP with an asterisk box? |
02:55.39 | forrestc{hm} | supiagtr: what's really bizzare is that RTCP indicates *no packet loss* or other issues on the call |
02:56.00 | forrestc{hm} | clyrrad: yep. The call problems are SIP. |
02:56.14 | forrestc{hm} | crash3m: We use it as a call router around here T1 to T1. |
02:56.40 | forrestc{hm} | cupiagtr: and RTCP usually *KNOWS* what is going on. |
02:56.46 | clyrrad | But for me the Jitter or Lag whatever it 'SHOULD' be called, it goes away after 2-5 seconds like handshaking is taking place.... know what i mean? |
02:57.21 | forrestc{hm} | clyrrad: probably an issue with the jitterbufer tuning up. It might also be something to do with echo cancellation. |
02:57.49 | clyrrad | its one of the 2 havent been able to fix it yet, wanted to see if you were having the same issue or if you solved that part already |
02:58.49 | supaigtr | I have IAX2 muting in one direction every so many calls. I can't find any problems. Its like a ghost problem. |
02:58.54 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
02:59.46 | clyrrad | Dont have the Muting problem, I use IAX2, both directions sound good, it is just a bit of jumping of the call at the beginning, sounds more like a lag than anyting else |
03:00.02 | supaigtr | iax2 show netstats? |
03:00.28 | clyrrad | hrm... never tried this before |
03:01.49 | clyrrad | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
03:01.49 | clyrrad | IAX2/unlimitel-1 1000 -1 0 -1 -1 0 -1 0 0 0 0 0 0 0 0 |
03:01.53 | clyrrad | how does that look to you? |
03:02.23 | supaigtr | RTT looks bad. |
03:02.32 | clyrrad | Really? |
03:02.35 | supaigtr | Oh. Is jitterbuffer on? |
03:02.42 | supaigtr | Both sides? |
03:03.04 | supaigtr | RTT is Round Trip |
03:03.19 | clyrrad | on the IAX box all I have is jitter size and its set to 0 |
03:03.19 | supaigtr | I have like 2-6 ms |
03:03.43 | clyrrad | in IAX.CONF i have forcejitterbuffer=no under [general] |
03:03.52 | Vlat- | damned good UK production |
03:04.15 | clyrrad | supaigtr... what do you have set for jitter? |
03:04.16 | Vlat- | by the way |
03:04.26 | *** join/#asterisk docE (n=docE@dsl001-136-136.lax1.dsl.speakeasy.net) |
03:04.26 | Vlat- | if you're using the jitter |
03:04.26 | supaigtr | I've been using the new jitterbuffer which fills out the rest of those values. Seems to work. Ihave since reverted to stable astersik but I think my muting was related to zaptel card. I unloaded that module and loaded ztdummy and going to test tommorow. |
03:04.33 | Vlat- | just add +-10 to it |
03:04.49 | supaigtr | I just turned on jitterbuffer=yes. the new jb is supposed to do things for you. |
03:04.55 | Vlat- | it would be more or less real value |
03:05.07 | docE | sup sup from Astricon! |
03:05.11 | clyrrad | supaigtr in iax.conf under [general]? |
03:05.20 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
03:05.26 | *** join/#asterisk Soulz--- (n=Soulz---@host-137-132-43-194.imcb.nus.edu.sg) |
03:05.28 | docE | pictures available @ http://astri2005.netdr.biz More coming! |
03:05.30 | Soulz--- | hi all |
03:05.32 | clyrrad | Vlat are you refering to setting Jitter Size =10 in my IAX ATA? |
03:05.32 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
03:05.36 | Vlat- | docE: what's new at asteric ? |
03:05.55 | Vlat- | clyrrad: i meant to set up the adaptive jitter |
03:06.11 | Soulz--- | i am getting Set Absolute Timeout to 15 timeouts when ever a incomming call |
03:06.14 | supaigtr | clyrrad: there or in each iax section. I do it in both. |
03:06.15 | docE | Lots of more companies going to be supporting asterisk |
03:06.18 | Soulz--- | any idea how to fix that please? |
03:06.41 | clyrrad | supaigtr... you have jitterbuffer=yes in your general and each context of your iax.conf? |
03:06.58 | Vlat- | <PROTECTED> |
03:07.00 | clyrrad | Vlat, i do not have such a setting in my ATA |
03:07.20 | Vlat- | e.g. endpoint to ....austria [xxx]xxxxxxxx for example |
03:07.28 | supaigtr | Yep. |
03:07.37 | clyrrad | hrm... let me give that a shot |
03:07.42 | Vlat- | c. |
03:07.46 | supaigtr | Vlat-: I just want to connect two offices. |
03:07.58 | Vlat- | clyrrad: ATA like ATA186? |
03:08.55 | Vlat- | supaigtr: look, i really have less experinece with IAX. But with SIP it usually take 10-20 minutes |
03:09.01 | Vlat- | with H323 - 30 minutes |
03:09.19 | docE | I test for dCAP tomorrow! YAY! |
03:09.24 | Vlat- | (10mins to tell the people WHY they're shouldn't use H323) |
03:09.28 | brookshire[home] | yay! |
03:09.42 | brookshire[home] | everyone uses h323 |
03:09.44 | brookshire[home] | hehe |
03:09.50 | brookshire[home] | well... all the big boys |
03:09.54 | clyrrad | supaigtr LOL i think that fixed it :) |
03:10.01 | Vlat- | give me the man use h323 |
03:10.10 | Vlat- | i'll make Uncle Bens from him |
03:10.41 | Vlat- | or maybe Uncle Vlad |
03:10.58 | Vlat- | 3 years ago i had to wrote h323 applications |
03:11.17 | Vlat- | openh323 made a more than bad impression to us |
03:11.22 | Vlat- | with the fscking bugs |
03:12.37 | brookshire[home] | :( |
03:13.37 | brookshire[home] | katty: he's getting drunk in cali |
03:13.39 | brookshire[home] | :( |
03:14.01 | Katty | brookshire[home]: i think he's actually is disneyland |
03:14.06 | brookshire[home] | haha |
03:14.08 | Katty | brookshire[home]: are you at astricon as well? |
03:14.08 | brookshire[home] | probably so |
03:14.12 | brookshire[home] | no.. |
03:14.14 | Katty | k |
03:14.15 | brookshire[home] | i had to stay home |
03:14.24 | brookshire[home] | are you his nextel bud? ;) |
03:14.46 | Katty | oh, perhaps ;) |
03:14.53 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve) |
03:14.53 | Katty | i do beep beep on occasion |
03:14.59 | brookshire[home] | haha |
03:15.03 | docE | Katty? Female edition? nice |
03:15.04 | brookshire[home] | always while we are playing pool |
03:15.05 | epablo | Hi people.. |
03:15.18 | file | Katty! |
03:15.22 | Katty | brookshire[home]: i can't help it if i call him while he's playing pool! |
03:15.24 | epablo | On what var is the result/return code of a call get set |
03:15.30 | Katty | file: yay, file! |
03:16.07 | file | heyyyyyyyy |
03:16.09 | file | wazzup? |
03:16.31 | Katty | my muscle content |
03:16.42 | Katty | forearms ripple now :> |
03:16.46 | epablo | If I dial, lets say zap. On what var do I get my diconnect cause? |
03:17.01 | file | ooh |
03:17.47 | docE | ep its lemme check |
03:17.48 | docE | I have it |
03:17.52 | supaigtr | clyrrad: What was the fix. |
03:18.10 | docE | you know this is on the WIKI right? |
03:19.06 | epablo | let me see if I can find it, sorry for my lazyness. I think it is late and I'm low on tea |
03:19.48 | docE | http://www.voip-info.org/wiki/view/Asterisk+variables |
03:20.13 | epablo | Thanks! |
03:20.14 | docE | Get some JOLT, Mt Dew, or something |
03:20.48 | docE | I have 4 cans of Mt Dew next to me in the Code Room |
03:20.55 | Katty | i /will/ have these forearms: http://www.propstore.com/images/products/410/tombraid-angjolgundisplay2.jpg |
03:21.35 | docE | Kat do you have a picture of you now? |
03:21.40 | epablo | Those cans give me big headaches.. I think it is the Caffeine level, or the other stuff they put in them |
03:21.47 | Katty | docE: 92 of them, at last count. |
03:21.57 | docE | Just caffeen! they are good! |
03:22.06 | docE | Can I have the URL? |
03:22.09 | Katty | docE: no |
03:22.13 | docE | :( |
03:22.17 | docE | why? |
03:22.25 | docE | Im not a bad person.. Just curious.. :D |
03:22.26 | Katty | i don't know you |
03:22.37 | docE | ok who in here knows me? |
03:22.54 | *** join/#asterisk chrisdavid (n=cjones@69.90.193.151.novuscom.net) |
03:22.57 | wunderkin | sure i know you docE |
03:22.59 | docE | Twisted or Damin could vouch for me if they were in here.. but I think they are down stairs |
03:23.02 | wunderkin | /msg docE whats the url? |
03:23.20 | docE | Sorry wouldnt give it out if she gave it to me. |
03:23.30 | docE | Not in my nature to kill someones trust. |
03:24.13 | epablo | Katty: Blur out the face (TV style) and let us see you. ;) |
03:24.48 | Katty | file: zomg, lookit! http://perso.wanadoo.fr/psylo-vision/_images/wallpapers/_first/Tomb-Raider-cot%E9.jpg |
03:24.52 | Katty | file: purrty! |
03:25.42 | Katty | docE: just because twisted knows you doesn't make it All Better |
03:25.45 | file | oooooooh |
03:26.09 | docE | sigh.. |
03:26.11 | docE | women |
03:26.29 | Katty | docE: oh yes, i'm unreasonable now because i don't want to share. |
03:26.34 | Katty | docE: shame on me. |
03:26.43 | docE | hehe.. no not even like that.. |
03:26.51 | Katty | k |
03:26.54 | docE | Women just like to be secretive.. thats all.. :) |
03:27.04 | docE | My wife does the same thing to me and drives me nuts |
03:27.05 | file[laptop] | Katty is silly |
03:27.19 | Katty | file[laptop]: you'll have to braid my hair like miss croft at the next convention! |
03:27.40 | file[laptop] | oh no!!! |
03:27.49 | Katty | oh yes! |
03:27.49 | *** part/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
03:27.51 | Katty | yes you will |
03:28.03 | Katty | or i will steal your danish and OJ |
03:28.35 | file[laptop] | nooooooo :( |
03:29.03 | Katty | hair or death! little red cookbook! |
03:29.33 | file[laptop] | redrum |
03:29.38 | L|NUX | please donate www.pakistanhelp.com |
03:30.55 | file[laptop] | Katty: how was your day? |
03:31.30 | Katty | file[laptop]: medium rare. |
03:31.40 | file[laptop] | ooh |
03:31.43 | Katty | file[laptop]: with a hint of windows, actually |
03:31.51 | Katty | silly routing and remote access causing me headaches |
03:31.51 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
03:31.55 | jdv79 | Oct 12 23:31:12 WARNING[10400]: chan_zap.c:1573 zt_set_hook: zt hook failed: Device or resource busy |
03:31.58 | jdv79 | anyone? |
03:31.59 | Katty | i threatened it with a stick |
03:32.08 | Katty | and by stick i mean reconfigured it |
03:34.02 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
03:34.23 | supaigtr | If you had to get something fixed and had to bet on HEAD is broken or the zap card you aren't using is causing audio problems with IAX2 to IAX2 which one would you go with by morning. Revert to stable, or stay with HEAD but figure out how to disable the zaptel card remotely and use ztdummy. |
03:34.46 | fiber0pti | Anyone have any suggestions for some good and inexpensive voip phones? Somewhere between $75-$150 retail.. |
03:34.57 | clyrrad | CAD or USD? |
03:35.17 | supaigtr | clyrrad: What was the fix? |
03:35.30 | clyrrad | jitterbuffer = yes :) :) :) |
03:35.38 | *** join/#asterisk tim_scott (n=war@d198-166-220-253.abhsia.telus.net) |
03:35.41 | tim_scott | Hello all. |
03:35.50 | tim_scott | Is there anyone here who could spare a moment to answer a question or two? |
03:36.30 | supaigtr | clyrrad: I think my audio loss is the fact that the machine still has a zaptel TDM card but it isn't used. |
03:37.32 | supaigtr | tim_scott: Ask away. |
03:37.32 | tim_scott | Alrighty. |
03:37.32 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
03:37.54 | tim_scott | Okay. |
03:38.12 | clyrrad | did you disable all intrupts that your not using in your BIOS? I read that can screw things up |
03:38.31 | epablo | Anyone know where a file generated call returns it's disconnect cause. Just checked dialstatus and hangupcause.. with no luck |
03:38.46 | tim_scott | I doing a presentation on Asterisk on Friday at ny local college. I was going to set up an Linux machine running Asterisk there, which would communicate over IAX with a machine at work. |
03:39.15 | Soulz--- | hello all |
03:39.18 | tim_scott | The college has port 4569 blocked, so I set the bindport=80 in iax.conf, on both the machine at work and at college. |
03:39.30 | Soulz--- | when ever i dial a sip/iax2 extenion, it always goes to voicemail |
03:39.37 | Soulz--- | though the phones are working |
03:39.44 | supaigtr | I did but it still had problems. I unloaded the module but nobody here seems to know if that will make things work. I know it doesn't show up in interrupts now. The problem is intermittent and I won't know until tommorow. By then they'll be ready to rip * out of the business. |
03:39.47 | tim_scott | the work machine will not register with the machine at college, but I can still place calls from work -> college. |
03:40.04 | tim_scott | But I can't make calls from college -> work, because "work" is the machine doing the register=>'ing. |
03:40.07 | *** join/#asterisk st3v (n=st3v@netblock-66-245-213-120.dslextreme.com) |
03:40.19 | tim_scott | Anyone have any suggestions? Where is the registration traffic going? :/ |
03:40.31 | Soulz--- | can anyone give me to some pointers on where i should look? |
03:40.31 | *** join/#asterisk kb1_kanobe (n=krisbout@h24-207-96-50.cst.dccnet.com) |
03:40.42 | epablo | tim_scott: sounds like a FireWall problem |
03:40.59 | clyrrad | supaigrt.... LOL yah some people just wont get it |
03:41.11 | kb1_kanobe | evening all. |
03:41.12 | tim_scott | So where does the registration traffic go? |
03:41.28 | kb1_kanobe | anything scary in cvs-head at the moment? |
03:41.32 | tim_scott | The firewall port cannot be unblocked, it's their security policy to be a massive pain in the ass. |
03:41.41 | Katty | anyone know who's hosting the cluecon gallery? |
03:41.49 | Katty | or, more importantly, where Junky's gallery is? |
03:41.58 | Katty | cause i know who's hosting it |
03:41.59 | docE | Im hosting Astricon's.. :) |
03:42.14 | Soulz--- | when ever i dial a sip/iax2 extenion, it always goes to voicemail, anyone? |
03:42.21 | st3v | I am setting up an asterisk server, using a Rhino channel bank. Should I get the 24 port POTS FXS bank and replace one of the 4 port FXS modules with a $380 FXO module, or just keep the channel bank all FXS, and get the digium card with 4 FXO ports built in? |
03:42.49 | tim_scott | Soulz, can you be more specific? |
03:42.52 | Vco | well |
03:42.56 | Vco | do the math |
03:43.14 | Soulz--- | timscott: when ever i dial from different ip phones (internally and externally) |
03:43.23 | Soulz--- | all sip calls/iax seems to go to vm |
03:43.27 | st3v | I mean performance wise, they are almost the same price. |
03:43.43 | epablo | tim_scott: You are right can't be the FW.. both side are sure to have port 80 in/out open |
03:44.02 | Vco | whats a TDM40B go for? |
03:44.07 | epablo | tim_scott: unless it is set only for specific IP's |
03:44.15 | supaigtr | TCP UDP theres a difference http is TCP |
03:44.34 | st3v | <PROTECTED> |
03:44.44 | tim_scott | :/ |
03:44.58 | st3v | I would get the TDM04B |
03:45.48 | docE | yes and asterisk only uses UDP ports. Not tcp |
03:45.50 | tim_scott | supaigtr: I'm not following. Was that comment directed at me? :/ |
03:46.02 | Vco | more importantly, will you need the extra space for FXS later... |
03:46.13 | Soulz--- | externally meaning when external did calls a internal sip extension |
03:46.21 | Soulz--- | i get to go straight to vm |
03:46.24 | Vco | either way you're going to need a card if thats the case |
03:46.26 | Soulz--- | this is quite strange |
03:46.33 | Vco | so then it really wouldn't matter |
03:46.34 | st3v | well we only need 18 extensions now, but maybe more later |
03:46.36 | glm2k | st3v: same functionality. are you on a budget? |
03:46.58 | Vco | go channelbank... |
03:47.10 | glm2k | i would do the same as well. |
03:47.15 | glm2k | saves a pci slot too |
03:47.36 | Soulz--- | http://pastebin.ca/25359 |
03:47.45 | Vco | and who knows...maybe you'll need more that the 4 ports later... |
03:47.52 | Vco | making the TDM card a waste |
03:47.54 | supaigtr | tim_scott: Yep. UDP is likely blocked |
03:48.02 | Soulz--- | AGI Script dialparties.agi completed, returning 0 or Everyone is busy/congested at this time |
03:48.08 | Soulz--- | dunno where i should look |
03:48.12 | st3v | yeah |
03:50.22 | *** join/#asterisk bmg505 (n=leon@rndf-146-13-10.telkomadsl.co.za) |
03:50.25 | tim_scott | supaigter: Possible. I'll find out. |
03:50.27 | tim_scott | Thanks. |
03:50.49 | epablo | tim_scott: Colleges normally block UDP.. At least mine did |
03:50.49 | kb1_kanobe | is it possible to delete specific voicemail messages from specific boxes through the manager interface? |
03:51.11 | tim_scott | So the registration traffic is going over UDP? |
03:51.39 | Soulz--- | tim: do u need more info? |
03:52.35 | epablo | tim_scott: it all goes over UDP |
03:54.06 | tim_scott | UDP port what? 4569? |
03:56.40 | epablo | tim_scott: Thats the defualt.. but if you can manage the FW guy to open any of the to you.. You can set it up |
03:56.47 | Katty | iax is 4569, me thinks |
03:57.09 | tim_scott | He won't open up any of the ports.. URG. He said it's a security risk... >_< |
03:57.15 | Katty | pffft |
03:57.19 | tim_scott | Exactly. |
03:57.25 | tim_scott | So is there another option? Would it help if I pasted what I see on my screen? |
03:57.37 | tim_scott | I'm really screwed on this one, presentation is tommorow and friday. :/ |
03:57.41 | tim_scott | Anyone have any suggestions? |
03:57.57 | epablo | Can you set up a VPN? |
03:58.03 | Katty | yes, VPN will work |
03:58.10 | Katty | and make yourself part of the internal network |
03:58.21 | tim_scott | I can't do any port-forwarding stuff. I can only use port 80. |
03:58.23 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
03:58.36 | Katty | tim_scott: you don't have to if your firewall does vpn |
03:58.41 | Katty | tim_scott: and most do |
03:58.55 | tim_scott | The firewall at college? Hell, I don't know. |
03:59.00 | Katty | i'm sure it does |
03:59.05 | Katty | even the cheapies have vpn |
03:59.12 | tim_scott | I would assume. |
03:59.19 | Katty | the symantec model 100 for 10 users has vpn |
03:59.21 | tim_scott | Sadly, I don't know how to do that. |
03:59.31 | epablo | I don't think he will let you doit. but you can use openvpn on both machines.. |
03:59.31 | Katty | not going to learn any quicker (= |
03:59.34 | tim_scott | I don't have enough time to mess with VPN patches. |
03:59.40 | Katty | k |
03:59.51 | tim_scott | I have like, 8 hours before I have to start setting up. |
03:59.53 | tim_scott | :/ |
03:59.58 | Vco | oh so you're saying you have a life or something.. |
04:00.16 | Vco | ahh |
04:00.47 | tim_scott | I know I probably asked this already, but could someone explain to me in detail why simply setting bindport=80 in iax.conf on both sides of the firewall will not work? |
04:01.01 | tim_scott | I mean, I would assume it would work, but the one end can't register with the other. :/ |
04:01.12 | *** part/#asterisk chrisdavid (n=cjones@69.90.193.151.novuscom.net) |
04:01.54 | tim_scott | Pardon my ignorance, I just don't understand why it isn't working. :S |
04:02.06 | kb1_kanobe | tim_scott: sorry, just kind of coming in mid-discussion. What are you trying to do? |
04:02.14 | tim_scott | Heh >_< |
04:02.23 | supaigtr | tim_scott: They allow TCP on port 80 not UDP |
04:02.35 | tim_scott | Assuming UDP is allowed. |
04:02.52 | epablo | tim_scott: Then their is no problem |
04:02.54 | tim_scott | kb1_kanobe, I'm setting up an asterisk system at my local college that needs to communicate via IAX with an Asterisk server at work. |
04:03.03 | kb1_kanobe | server to server? |
04:03.09 | tim_scott | 4569 is blocked inbound and outbound, so I set bindport=80 in my iax.conf file. |
04:03.11 | supaigtr | It would work if udp was allowed. |
04:03.12 | tim_scott | Yes, server to server. |
04:03.23 | tim_scott | Well, UDP is allowed. |
04:03.30 | supaigtr | It should work then. |
04:03.34 | tim_scott | But the one server isn't register=>'ing with the other. |
04:03.37 | tim_scott | See, it's _not_ working. |
04:03.39 | Katty | file[laptop]: i suddenly got sleepy :< |
04:03.53 | tim_scott | What additional information do I need to provide to help someone help me ? :S |
04:03.56 | kb1_kanobe | I don't use register between my servers. |
04:04.05 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
04:04.17 | supaigtr | That has nothing to do with registering. If both * boxes are doing IAX over 80 and it worked on the orignal port it should work on 80. |
04:04.18 | kb1_kanobe | as longs as udp is allowed both ways and there is no nat, it should be simple enougg. |
04:04.28 | tim_scott | There is NAT. |
04:04.32 | Vco | or a proxy server? |
04:04.33 | kb1_kanobe | s/enougg/enough. |
04:04.36 | supaigtr | On both sides NAT? |
04:04.46 | tim_scott | Both sides. |
04:04.57 | epablo | Both need to reg |
04:04.59 | kb1_kanobe | ewwww... out of my league. Sorry. |
04:05.12 | tim_scott | Like I said, I can Dial() work -> college, but not back. |
04:05.13 | supaigtr | If you can't get a public ip on one side it won't work. You only need one side to register. Side behind nat. |
04:05.44 | kb1_kanobe | can both sides register w/ a third party server elsewhere? |
04:05.45 | tim_scott | Oh. |
04:05.54 | tim_scott | supaigtr: that sounds like it makes sense. |
04:06.06 | tim_scott | So they can't simply register with each other? |
04:06.10 | docE | kb1_kanobe yes |
04:06.32 | docE | if bedind nat No. . You would have to setup DMZ or tunnel your port to the private IP |
04:06.40 | kb1_kanobe | docE: if he were to use notransfer on the man-in-the-middile, that could work for him...? |
04:07.14 | tim_scott | AUGH |
04:07.24 | tim_scott | That's what I thought. |
04:07.38 | tim_scott | kanobe: latency would probably be killer if I did that. |
04:07.46 | Vco | ? |
04:07.59 | tim_scott | I tried setting up a proxy at work, but bouncing my calls work -> proxy -> college added too much latency. |
04:08.02 | kb1_kanobe | unless you can get access to something in the DMZ at the site, or nearby (network-wize) |
04:08.24 | tim_scott | I'll have to find out tommorow. I have a few hours setup time. *shudder* |
04:08.30 | tim_scott | Let's hope this isn't a total flop. Thanks guys. |
04:08.33 | kb1_kanobe | VPN will be the way to go... |
04:08.38 | tim_scott | At least my suspicians were confirmed. |
04:08.46 | kb1_kanobe | good luck :-/ |
04:08.50 | tim_scott | I'll see if I can plug into a DMZ, that'll probably be easier. |
04:08.55 | kb1_kanobe | optimal. |
04:09.08 | tim_scott | Goodnight. |
04:09.10 | *** part/#asterisk tim_scott (n=war@d198-166-220-253.abhsia.telus.net) |
04:10.01 | docE | the man in the middle option would not work for the nat issue |
04:10.45 | kb1_kanobe | No? Damn nat. I assumed he'd be able to register both with a 3rd party and, as long as notransfer was enabled then all rtp traffic would relay through there. |
04:15.22 | fiber0pti | Does anyone have experience with setting up multiple Sipura SPA-841 phones with asterisk? |
04:17.18 | Katty | newp. |
04:20.22 | docE | yes |
04:20.23 | docE | why? |
04:20.46 | docE | they are like setting up any other ATA or Sip phone.. You assign extensions and they are happy |
04:20.59 | Katty | and they frolic about the network |
04:21.00 | fiber0pti | cool.. so it was easy? |
04:21.08 | fiber0pti | I might be setting up 100 of them |
04:21.12 | Katty | with their happy rtp packets. |
04:21.15 | fiber0pti | I'm looking at them because of price |
04:21.50 | Katty | the only phones i've really worked with are the polycom500s |
04:22.11 | fiber0pti | Katty; I will be setting up 10 of those soon.. any suggestions? |
04:22.16 | fiber0pti | anything that stood out? |
04:22.31 | Katty | fiber0pti: if you're going to setup a .wav file for a ringtone, make sure it's ulaw 8 mono |
04:22.44 | fiber0pti | hehe.. k |
04:22.58 | docE | if you setup 100 of them might I suggest using a tftp config on the phones |
04:23.05 | docE | it will make your life MUCH easier trust me! |
04:23.05 | fiber0pti | I got 16 of them from ebay for $100 a piece |
04:23.09 | Katty | tftp-- |
04:23.11 | Katty | ftp++ |
04:23.24 | Katty | unless you have a firewall |
04:23.25 | supaigtr | ftp or https are only options for the poly |
04:23.29 | fiber0pti | docE: I've used tFTP on a few cisco 7960.. it was great |
04:23.31 | Katty | which i hope you do, since you have IP phones. |
04:23.39 | supaigtr | The thing that sucks is buddy lists and DHCP. |
04:23.41 | Katty | supaigtr: no, they do tftp as well |
04:23.52 | supaigtr | Not with the newest bootrom. |
04:23.55 | Katty | supaigtr: that's what i originally tried. and then decided it was silly. |
04:24.04 | Katty | silly like a... |
04:24.06 | Katty | ...balloon. |
04:24.09 | fiber0pti | I would actually prefer ftp over tftp just becuase of some issues I've ran into with tftp |
04:24.26 | supaigtr | :) Katty.. How did you handle DSS buttons / console? |
04:24.27 | HiltonT | hhmmm, can AMP be run with Xorcom Rapid? |
04:24.36 | docE | at anyrate fiber good luck with that.. All I can say is take a week and plan out your deployment. Cause a poorly planned deploymnet will make your life hell |
04:24.44 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
04:24.54 | Katty | brookshire[home]: yay! |
04:24.54 | fiber0pti | doce: thanks.. I will try my best |
04:25.46 | Katty | fiber0pti: and if your polycom 500s require rebooting a lot...let me know. |
04:25.53 | docE | but if you need help by all means do ask questions.. I believe the only dumb question is the one not asked.. |
04:26.01 | Katty | fiber0pti: i haven't /quite/ figured out why mine do that. like 5 times during the day. |
04:26.05 | docE | I am usually here at docE, Docelm0 or Docelmo |
04:26.12 | Katty | fiber0pti: there's this weird issue of they can hear you, but you can't hear them |
04:26.16 | fiber0pti | weird.. how long do they take to come up after rebooting? |
04:26.17 | Katty | fiber0pti: yet, it's not an rtp problem |
04:26.27 | *** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com) |
04:26.33 | Katty | fiber0pti: i'm all confuzzled and want to look at sip debug, but haven't had a chance yet |
04:27.03 | fiber0pti | KAtty: interesting.. does the reboot render them useless at that time? What if someone is on the phone? |
04:27.09 | supaigtr | Katty: is that an intermittent problem? |
04:27.11 | docE | anywho.. off for niccotine bbiaf.. |
04:27.13 | Katty | supaigtr: no |
04:27.18 | supaigtr | Hmm. |
04:27.22 | Katty | fiber0pti: hmm? |
04:27.27 | Katty | fiber0pti: they reboot and the phone is fine |
04:27.33 | supaigtr | The reboot is usually from timestamp changing on FTP. |
04:27.35 | Katty | fiber0pti: tis very odd. |
04:27.44 | Katty | we're not using ftp |
04:27.45 | fiber0pti | indeed.. I will see if mine do it too |
04:27.51 | Katty | we're MANUALLY rebooting |
04:27.59 | Katty | kthx |
04:28.05 | Katty | fiber0pti: excellent |
04:28.25 | Katty | also! this is a girly rant. |
04:28.28 | Katty | no need for solutions. |
04:28.35 | fiber0pti | hehe.. |
04:28.39 | ender | I use IP301s and IP501s. I've never had to reboot them aside from config changes. |
04:28.49 | fiber0pti | good to know, as well |
04:29.00 | supaigtr | We built a we UI to mass configure phones here. IT works ok but we have problems with * still. |
04:31.20 | JerJer[mobile] | girly rants are fun |
04:31.32 | HiltonT | girly pants are more fun |
04:31.40 | Katty | bye. |
04:31.41 | JerJer[mobile] | if they are on the floor |
04:31.49 | JerJer[mobile] | damn i scared katty off :( |
04:35.00 | Damin | Katty: Was someone talking shit to you? |
04:35.36 | Katty | mrow? |
04:36.33 | Katty | Damin: have you insaned? |
04:36.46 | Vco | i think JerJer was making some mention of wanting to wear her pants...... |
04:36.49 | Katty | Damin: why would /anyone/ do something like that? |
04:38.42 | Katty | Damin: that's what i thought. |
04:39.25 | Katty | supaigtr: sorry, you missed half the conversation. |
04:39.26 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
04:40.07 | Damin | I think I missed half the conversation too.. |
04:40.14 | Katty | exactly. |
04:40.15 | Damin | Katty: Should I smack him for you? |
04:40.32 | Katty | violence never solved anything except deadlocks. |
04:40.41 | MikeJ[Laptop] | yes.. you should smack him |
04:40.45 | MikeJ[Laptop] | :P |
04:40.56 | Katty | right. now i'm quite confused. |
04:41.02 | MikeJ[Laptop] | hey Damin... what the hell are you doing online.. go get a drink boi |
04:41.06 | Katty | because Brain Surgery Sucks. or so i hear. |
04:41.52 | Damin | MikeJ[Laptop]: I would.. but I don't know where people are drinking at... |
04:41.59 | MikeJ[Laptop] | the bar? |
04:42.00 | Damin | MikeJ[Laptop]: And I'm catching up on Email right now.. |
04:42.00 | Katty | i think i'll just stop while i'm ahead (= |
04:42.12 | *** join/#asterisk Eight (n=blake@12-227-171-175.client.mchsi.com) |
04:42.16 | Damin | Katty: Probably a good idea.. Maybe I should smack you? |
04:42.30 | Katty | Damin: not a very wise idea. |
04:42.32 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
04:42.34 | MikeJ[Laptop] | yay cygwin!!! |
04:42.42 | Damin | MikeJ[Laptop]: Huh? |
04:42.43 | MikeJ[Laptop] | soooo close now |
04:42.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
04:42.53 | MikeJ[Laptop] | almost have it all compiling |
04:42.59 | docE | YAY! |
04:43.07 | MikeJ[Laptop] | figured out all the makefile ugliness |
04:43.18 | MikeJ[Laptop] | and figured out how to do it not sooo ugly |
04:43.27 | Katty | where is twisted? |
04:43.39 | JerJer[mobile] | last i heard they were going to dinner |
04:43.42 | Damin | docE: Who are you? |
04:43.50 | MikeJ[Laptop] | and did it without all the junk that the asteriskwin32.com guy had to do to make a loader and do all kninds of weird stuff... |
04:44.21 | MikeJ[Laptop] | so I still build .so files on top of dll's so all the module loader stuff does not have to be totally re-written |
04:44.38 | Katty | g'night. |
04:44.56 | MikeJ[Laptop] | which is good |
04:45.10 | docE | Im sitting across from you dude.. |
04:45.13 | MikeJ[Laptop] | makes the makefiles a bit fun tho |
04:45.20 | MikeJ[Laptop] | not from me |
04:45.25 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
04:45.29 | MikeJ[Laptop] | cuz I'm in deroit |
04:45.43 | MikeJ[Laptop] | :P |
04:46.02 | JerJer[mobile] | detoliet |
04:46.07 | MikeJ[Laptop] | yes |
04:46.11 | MikeJ[Laptop] | is mark there ? |
04:46.16 | JerJer[mobile] | somewhere |
04:46.27 | MikeJ[Laptop] | tell that boi to get a drink... |
04:46.28 | JerJer[mobile] | Alison is too :) |
04:46.32 | Damin | MikeJ[Laptop]: Mark is totally pissed off at you dude. |
04:46.40 | MikeJ[Laptop] | at me? |
04:46.41 | MikeJ[Laptop] | why? |
04:46.43 | Damin | MikeJ[Laptop]: Yeah.. |
04:46.55 | Damin | MikeJ[Laptop]: I don't know.. |
04:47.02 | MikeJ[Laptop] | well that's silly |
04:47.17 | Damin | MikeJ[Laptop]: What did you do? |
04:47.20 | MikeJ[Laptop] | dunno |
04:47.32 | Katty | why don't you /ask/ like grownups? |
04:47.34 | MikeJ[Laptop] | been a stranger sense I got back from boston.. |
04:47.42 | MikeJ[Laptop] | ummm.. ask what? |
04:47.48 | Katty | if he's upset with you. |
04:48.01 | Katty | or i suppose you could do the girly could shoulder thing for a few years. |
04:48.08 | Katty | s/could/cold/ |
04:48.10 | MikeJ[Laptop] | well.. I would assume if somone is mad at me, that they would say somthing to me |
04:48.17 | HiltonT | I like the girly cold shoulder thing :) |
04:48.23 | Katty | right. i was going to bed. |
04:48.33 | MikeJ[Laptop] | so I am guessing that they are just joking w/ me |
04:48.36 | HiltonT | nite Katty |
04:48.42 | MikeJ[Laptop] | if not.. then that's just silly |
04:48.50 | HiltonT | MikeJ[Laptop]; 'zactly |
04:49.20 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
04:50.17 | file | tomorrow is shopping time... oh joy |
04:50.23 | MikeJ[Laptop] | for me? |
04:50.27 | file | nope! |
04:50.30 | MikeJ[Laptop] | :( |
04:50.33 | MikeJ[Laptop] | why not? |
04:51.28 | jets | FILE |
04:51.36 | file | well if you want to pay for it all, sure - then shopping time |
04:51.37 | HiltonT | because you are a dude, MikeJ[Laptop], and shopping isn't something we enjoy doing |
04:51.37 | file | oh no it's jets |
04:51.37 | jets | what are you shopping for hMMM? |
04:51.44 | file | sneakers. |
04:51.49 | jets | oh whatever have u even talked to me this whole trip HMMM? |
04:51.50 | Damin | OK.. |
04:51.52 | Damin | going to drink.. |
04:52.02 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
04:52.03 | jets | who's drinking and where? :) |
04:52.13 | docE | Probably in the bar.. |
04:52.31 | docE | Jet arent you like 18 or something? |
04:52.58 | jets | Nope i'm 22 bitches, i was at the bar lastnight with a bunch of geeks |
04:53.04 | jets | which bar |
04:53.24 | docE | the one down stairs.. :) |
04:53.31 | docE | I am in the code room |
04:53.36 | docE | I may go drink.. dunno.. |
04:54.09 | Vco | thats not a br, that the gay crack house |
04:54.35 | docE | who are you? |
04:55.05 | jets | i'm brian... fun hair |
04:55.09 | jets | gonna speak about callmanager |
04:55.14 | jets | being replaced by * |
04:55.32 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
04:55.41 | HiltonT | CallManager == Hell Expensive! |
04:55.47 | HiltonT | (well, it IS Cisco!!!) |
04:56.09 | Vco | And thats even coming from someone with the name Hilton......... |
04:56.10 | jets | come on baby light my fire! |
04:56.14 | HiltonT | lol |
04:56.19 | jets | heh |
04:56.23 | HiltonT | I look nothing like Paris (thank fsck) |
04:56.24 | jets | i'm a hilton |
04:57.21 | *** join/#asterisk bweschke (n=bweschke@m810f36d0.tmodns.net) |
04:59.01 | jets | tyler hilton was at von for a few hours in march |
04:59.15 | HiltonT | who's that? |
04:59.31 | jets | google |
05:01.30 | HiltonT | oh, yay - teenybop |
05:01.50 | HiltonT | another Britney Spears... |
05:02.54 | jets | ya |
05:02.59 | jets | fscking loaded though |
05:03.01 | jets | and likes asterisk |
05:04.03 | loud | jets |
05:04.13 | loud | next call manager will be SIP enabled. |
05:04.18 | loud | i saw it today. |
05:04.21 | MikeJ[Laptop] | g'night all |
05:04.27 | loud | cisco noticed they are losing money. |
05:04.31 | bweschke | on second thought.... now that I'm back to my room I'm pretty damn tired... not gonna come back down to the code zone. good night damin and jeremy.. catch you guys tomorrow |
05:05.12 | jets | loud: "sip enabled" doesn't 4.0 do sip already? |
05:06.15 | jets | anything fun going on at the code zone |
05:06.26 | loud | 5.0 |
05:06.35 | loud | Q1 2006 |
05:07.28 | jets | hrmmm no a deployment of 4.0 at a college were using soft phones and doing some sip calls to asterisk........ maybe not trunking, etc,etc, but endpoints like polycom hardphones, softphones and asterisk |
05:07.52 | loud | you sure they are not using cisco's sip proxy ? |
05:08.22 | jets | yep i'm sure they are using callmanager |
05:08.50 | jets | http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a00801f8e18.shtml |
05:19.15 | clyrrad | Anyone here had custom messages recorded by Alyson? |
05:21.11 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-82.cybersurf.com) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk NetSkier (n=ns@ca-redbch-cuda1-c3a-199.stmnca.adelphia.net) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
05:21.11 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
05:23.33 | kb1_kanobe | errr.... has the +101 jump behaviour been depreciated, or deactived in cvs-head? |
05:23.59 | wasim | that would not be good |
05:24.17 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:24.21 | kb1_kanobe | indeed. |
05:24.40 | kb1_kanobe | I must be missing something... more digging. |
05:26.10 | kb1_kanobe | oh, for ff.... sake. Why oh why is my shiney newly-upgraded asterisk server imploding. |
05:29.31 | *** join/#asterisk svadoe (n=svadoe@216.230.147.187) |
05:29.59 | kb1_kanobe | Hmmm... weasles appear to be eating my platters. |
05:30.51 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
05:36.43 | *** join/#asterisk jtodd (n=jtodd@host06.alica.hyatthsiagx.com) |
05:42.53 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
05:43.07 | niZon | anyone here ever done any kind of point to point link over dry copper pairs? |
05:43.16 | supaigtr | Yep. Everyday. |
05:43.30 | tessier | Wet copper pairs tend to corrode. |
05:43.41 | *** join/#asterisk Qwell (n=chatzill@pool-71-108-241-125.lsanca.dsl-w.verizon.net) |
05:43.41 | niZon | supaigtr what kind of speed can you get? |
05:43.41 | SplasPood | I prefer mine lightly salted |
05:44.01 | supaigtr | Depends on eq. Theres stuff out there to do 10 meg + |
05:44.02 | kb1_kanobe | SDSL w/old netopia 7100C were quite popular. |
05:44.22 | niZon | I'm looking to do it over about 1.5Km or so |
05:44.24 | supaigtr | Got boxes of those. Back to back required firmware. |
05:44.30 | niZon | as fast as possible |
05:44.39 | supaigtr | niZon: just one connection or multiple? |
05:44.45 | kb1_kanobe | the 7100c could do it out of the box if memory serves. |
05:44.55 | niZon | depends on what kind of speeds i can get with one connection |
05:45.06 | supaigtr | netopia had a firmware for the "CO" end |
05:45.14 | supaigtr | 1.5 is going to be SDSL |
05:45.45 | niZon | do you have links to any info on required hw? |
05:45.49 | supaigtr | You could get a coppermountain concentrator and bond to 6 meg or so. |
05:45.55 | niZon | hm |
05:46.04 | supaigtr | You'd need three pair. |
05:46.04 | kb1_kanobe | niZon: http://www.pbs.org/cringely/pulpit/pulpit20010823.html |
05:46.45 | supaigtr | net2net has some small stuff thats current tech. |
05:46.53 | niZon | ok |
05:47.24 | supaigtr | Fiber works better. Microwave ain't so bad either. |
05:47.32 | niZon | I'd like to get about 10Mbit |
05:47.44 | niZon | fiber is a nono, can't get it to the building |
05:47.47 | kb1_kanobe | hell, even 802.11 wifi (interference excepted) |
05:47.54 | supaigtr | I've got a pair of new waverider eq that does 10 mb. |
05:48.04 | niZon | wifi, maybe if we can get roof space |
05:48.30 | supaigtr | I'll love to get rid of it and it don't require much. Ethernet to a little 1ft panel. self contained. |
05:48.38 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve) |
05:48.43 | supaigtr | I replaced with a 90meg link. |
05:48.47 | kb1_kanobe | niZon: http://www.star-os.com |
05:49.01 | niZon | hm |
05:49.34 | supaigtr | I think the wr is rated at 72meg but expect 10 -30 in real world. |
05:49.50 | supaigtr | They setup in a day and work well. |
05:49.55 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
05:50.28 | niZon | they look interesting |
05:51.17 | supaigtr | NLOS but I recommend little in the way. Couple trees isn't a big deal but it won't work thru mountain of dirt and rock. |
05:51.22 | Vco | motorola canopy wireless |
05:51.24 | Vco | 900mhz |
05:51.28 | supaigtr | Sucks!!!! |
05:51.45 | supaigtr | I've got 2 of those systems in boxes. |
05:51.56 | Vco | fired them up ? |
05:52.00 | niZon | 900mhz is a tad slow.. |
05:52.27 | supaigtr | Yea for about 6 months. too much in that spectrum even in rural areas. Most pager system interfer. |
05:52.49 | supaigtr | Its hard to compete with a 120watt pager system in more areas than we have cell service. |
05:52.50 | Vco | i noticed they have new higher speed backhauls now too... |
05:53.07 | supaigtr | the waverider is 5.8 |
05:53.10 | supaigtr | iSM band |
05:53.42 | swm_ | What happened to S2SM band? |
05:53.47 | Vco | is still say this is cool |
05:53.48 | Vco | http://cgi.ebay.com/ebaymotors/Mobile-Cellular-Cell-Tower-Truck-Antennas-Ham-Radio_W0QQcmdZViewItemQQcategoryZ63739QQitemZ4581983789QQrdZ1 |
05:53.52 | supaigtr | THe only microwave backhaul that is realiable for carrier use is the 10+gig licensed bands. |
05:54.13 | Vco | just paint WARDRIVER across the sides and drive around |
05:54.56 | supaigtr | No last mile wireless works good unless its point to point and well designed. |
05:55.20 | supaigtr | Played with mesh but we named it mush networking :) |
05:55.43 | supaigtr | Fiber is sooooo much better and more realistic in meeting demands. |
05:55.58 | Vco | oooh.. they have 300mb backhauls actually.. |
05:56.07 | supaigtr | Thats not user rates. |
05:56.49 | supaigtr | A good 655mb sonet microwave system starts at about 120,000 installed and licensed. |
05:56.50 | swm_ | Anyone tried infrared transmissions with a high powered foocused infrared beam between two points? |
05:57.49 | Vco | you mean a laser pointer, morse code chart and a twitchy finger? |
05:57.57 | kb1_kanobe | never tried it, but canon used to make an atm155 system using short range lasers. |
05:58.41 | supaigtr | That went belly up. canobeam or something like that. It looked like an atm camera. |
05:58.49 | kb1_kanobe | yeah. |
05:59.06 | supaigtr | Saw one on ebay for around 8000 a few months ago. |
05:59.20 | kb1_kanobe | lol - I think I saw same. |
05:59.28 | swm_ | Well a 130 Watt Laster (not millawatt like those stupid pens) but, it can go clear half way around the world, with repeaters of course. 35.8 miles and you need another reeater or your shooting off earth into the space |
06:00.19 | kb1_kanobe | bah - piggyback on one of those reflectors they left up on the moon for earth-moon distance measurements. |
06:00.22 | kb1_kanobe | ;-) |
06:00.31 | supaigtr | Anyone have any sonet hardware? whiterock or dmx? |
06:01.04 | supaigtr | If you're will to break the rules there all sort of things to do. |
06:01.05 | tuppa | $INSERT_OBLIGATORY_AUSTIN_POWERS_LASER_JOKE_HERE |
06:02.51 | *** join/#asterisk mthem (i=merlintm@64.235.245.133) |
06:02.54 | swm_ | Why not install a 250,000,000,000 Watt Laster powered from a 680 array of satellite solar conductors with a huge 65,000 Micro Ferit capacitors in outer space and nuke things around the worls, starting with Geoge Bush |
06:03.01 | *** join/#asterisk srfrog (n=cag@209-250-4-64.convergentaz.net) |
06:03.52 | supaigtr | nite ppl |
06:03.54 | Vco | becaue that would make you some sort ofpinko commie terrorist |
06:04.18 | *** part/#asterisk srfrog (n=cag@209-250-4-64.convergentaz.net) |
06:04.38 | *** join/#asterisk gres (n=serg@62.152.85.99) |
06:04.41 | swm_ | Yeah totall self destructive too... Launches it into deep space and explodes heh... Not recoverable, no proof |
06:05.15 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
06:05.47 | NoRemorse | hi, does OH33 use a hardwired CID? (ie the "aliases" in oh323.conf) or can it pass on correct CID for each call? |
06:05.53 | NoRemorse | *OH323 |
06:06.29 | HiltonT | ok - is there anywhere I can find a doc that lets me get my head around the extensions.conf file and Asterisk dialplans? I guess that this is the thing I need to grok before I move forward |
06:06.47 | NoRemorse | extensions.conf.sample |
06:07.36 | swm_ | -BOOM-BOOM-BOOM-BOOM-BOOM Bannnnnnnggggggggggggggggggggggggggggggg |
06:07.40 | HiltonT | doesn't exist in Xorcom Rapid :( |
06:20.12 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
06:20.36 | *** part/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
06:23.08 | *** join/#asterisk dasuberdavid (n=david@dsl001-136-136.lax1.dsl.speakeasy.net) |
06:26.22 | *** join/#asterisk JerJer[mobile] (n=jj@68.123.154.34) |
06:27.14 | JerJer[mobile] | MikeJ[Laptop]: :( |
06:27.45 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
06:28.44 | FuriousGeorge | anyone ever install a doorphone? |
06:29.27 | *** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net) |
06:29.41 | brookshire[home] | glorified doorbell? |
06:29.49 | brookshire[home] | :) |
06:31.46 | HiltonT | FuriousGeorge; yeah, we have as part of a Home Security package, but not (yet) a VOIP one |
06:36.12 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
06:37.00 | HiltonT | but, if I can get my head around Asterisk, that's likely to change |
06:38.06 | *** join/#asterisk jeffik (n=Administ@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
06:38.14 | FuriousGeorge | brookshire[home]: yeah, but its for a business in the city so you dont wanna just buzz anyone in |
06:38.22 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
06:38.27 | FuriousGeorge | so its only glorified in a very practical way |
06:38.39 | HiltonT | we do this for residential installs |
06:38.48 | HiltonT | (and I mean EXPENSIVE res installs) |
06:39.08 | FuriousGeorge | HiltonT: b/c you sell proprietary hogwash :) |
06:39.26 | HiltonT | where there's a tradesman's entrance to the house which can be remotely opened and locked, as can the internal door from that room to the rest of the house |
06:39.32 | HiltonT | FuriousGeorge; yes, that we do! :) |
06:39.36 | HiltonT | and I want to change this |
06:39.54 | FuriousGeorge | HiltonT: you've come to the right place |
06:40.03 | HiltonT | that's why I'm here! |
06:40.33 | HiltonT | I just need to grok how to config * to let me have my hardphone register to it, and I'll be underway (Xorcom Rapid) |
06:40.53 | HiltonT | and then I can figure out dialplans and such funishness |
06:41.01 | FuriousGeorge | did you check voip-info |
06:41.02 | wasim | ~docs |
06:41.04 | jbot | it has been said that docs is 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 |
06:41.46 | HiltonT | reading that now (and have been for a while now) |
06:41.54 | HiltonT | thanks jbot you rock |
06:42.15 | HiltonT | handbook-draft is, well, not really useful |
06:42.29 | HiltonT | asteriskdocs, heading over... |
06:43.30 | brookshire[home] | voip-info.org |
06:43.38 | FuriousGeorge | HiltonT: how hard can it be? [username] secret=,dynamic=,allow=,regexten=,callerid=,etc. |
06:43.41 | brookshire[home] | :) |
06:43.47 | FuriousGeorge | HiltonT: voip-info.org is where its at |
06:44.06 | HiltonT | yeah, but there's sooooo much there to read thru to get to the real stuff :) |
06:44.23 | brookshire[home] | asterisk o'reilly book? |
06:44.26 | brookshire[home] | :) |
06:44.35 | FuriousGeorge | brookshire[home]: got it, havent looked at it yet |
06:44.37 | HiltonT | voip-info |
06:44.42 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
06:44.50 | jets | I MIGHT BE |
06:44.51 | jets | i might not be |
06:44.53 | jets | hmmM! |
06:44.56 | FuriousGeorge | HiltonT: just search for Xircom |
06:45.15 | FuriousGeorge | or Xorcom or whatever |
06:45.18 | Damin | http://www.speakeasy.org/~gomez/owlhouse/images/misc/CamelToads2.jpg |
06:45.18 | HiltonT | or maybe even Xorcom :) |
06:46.11 | HiltonT | camel toads - rofl |
06:46.33 | brookshire[home] | Xircom was an old modem, lol |
06:46.41 | HiltonT | yup |
06:48.19 | riksta | hahaha |
06:48.27 | mutilator | anyone know how to fix win2k only showing one processor when 2 are installed? and using HT so it should actually show 4 processors.. |
06:49.02 | brookshire[home] | umm... no |
06:49.07 | HiltonT | FuriousGeorge; sure, but where does that go... |
06:49.33 | HiltonT | mutilator; was W2K installed in SMP mode, or was the 2nd CPU added later? |
06:51.34 | FuriousGeorge | HiltonT: why, in sip.conf of course, sometime after register, if youve got one of those |
06:51.45 | HiltonT | sip.conf? |
06:51.54 | HiltonT | oh, someone earlier said extensions.conf |
06:51.57 | mutilator | not sure, i got it tho |
06:52.00 | HiltonT | NFwonder I'm lost! |
06:52.09 | mutilator | had to change my computer type to ACPI multiprocessor PC |
06:52.13 | mutilator | was Standard PC |
06:52.21 | FuriousGeorge | [general] blah blah blah register=> blah blah [user] blah blah [peer] blah blah [friend] blah blah <--- sip.conf |
06:52.29 | HiltonT | standard PC is single CPU most definitely |
06:52.37 | mutilator | yea |
06:52.40 | HiltonT | FuriousGeorge; ta, blah, ta |
06:52.48 | mutilator | found teh ms docs |
06:52.57 | mutilator | took longer than i expected googling ~3 minutes |
06:52.58 | HiltonT | and MS doesn't support the change like that to make it SMP properly |
06:53.00 | mutilator | so i asked here in the mean time |
06:53.04 | brookshire[home] | well.. extensions.conf pretty much ties everything together |
06:53.05 | FuriousGeorge | HiltonT: seriously, thats where it goes in the doc |
06:53.12 | brookshire[home] | but sip.conf setups your sip channels |
06:53.24 | HiltonT | yeah, I gathered, I "blah" a lot, too, that's all :) |
06:53.44 | FuriousGeorge | you got your general section with a register=> and section for friends peers and users |
06:54.05 | FuriousGeorge | whats the problem? if you can grasp that go fill in the values from the example |
06:54.11 | FuriousGeorge | docs |
06:54.17 | HiltonT | there's no problem |
06:54.20 | HiltonT | I said "thanks" |
06:54.27 | FuriousGeorge | oh |
06:54.29 | FuriousGeorge | your welcom |
06:54.32 | HiltonT | :) |
06:54.55 | FuriousGeorge | so you got your hardphone going? |
06:55.00 | HiltonT | not yet |
06:55.15 | HiltonT | I only installed * this morning, never played before |
06:55.37 | HiltonT | (tho, the hardphone was working with AstraTEL (an Aussie SIP supplier) earlier) |
06:56.16 | FuriousGeorge | HiltonT: you know how to change settings in the phone to point it to ur * server? |
06:56.19 | mutilator | well shit |
06:56.21 | mutilator | that wasn't good |
06:56.30 | mutilator | BSOD changing it to acpi multiproc |
06:56.32 | FuriousGeorge | mutilator: rm -rf? |
06:56.34 | HiltonT | FuriousGeorge; know how to drive the phone fine, just not * |
06:56.45 | HiltonT | mutilator; exactly why they don't support it :) |
06:56.55 | HiltonT | FuriousGeorge; fdisk works better |
06:57.08 | mutilator | yeh our other server works fine tho =\ |
06:57.25 | HiltonT | really, running Win 2K these days is asking for problems |
06:57.34 | FuriousGeorge | HiltonT: username of phone corresponds to [ ] entry after general in sip.conf |
06:57.35 | HiltonT | esp. if it EVER connects to the Internet |
06:57.58 | *** part/#asterisk Qwell (n=chatzill@pool-71-108-241-125.lsanca.dsl-w.verizon.net) |
06:58.04 | FuriousGeorge | secret is obviously password |
06:58.49 | FuriousGeorge | disallow all and allow a codec that will work with the phone |
06:59.25 | FuriousGeorge | pick a context for the phone (maybe something like admin_caller, or outgoing or something) |
06:59.37 | FuriousGeorge | that covers the essentials |
07:00.00 | HiltonT | hhmmm, me needs to do some more reading ... "context"? |
07:00.06 | HiltonT | was fine up until there! |
07:00.32 | HiltonT | and I gather [user] would be [4001] or whatever the extension number will be |
07:00.36 | FuriousGeorge | dont think to much into it. an incoming call has a context "its incoming" |
07:00.39 | brookshire[home] | http://www.asterisk.org/glossary |
07:00.41 | brookshire[home] | :) |
07:00.44 | mutilator | good thing they added the "last known good config" boot option |
07:00.53 | HiltonT | mutilator; definitely :) |
07:00.57 | FuriousGeorge | if its from your sip provider thats a context |
07:01.15 | dan__t | hi |
07:01.17 | brookshire[home] | context is like a group |
07:01.20 | dan__t | i'm back, who missed me. |
07:01.32 | brookshire[home] | [default] |
07:01.35 | HiltonT | nah, want this to be internal only right now, then expand it to go external, then use AstreSIP (for now) to terminate to a POTS number |
07:01.36 | brookshire[home] | [menu[ |
07:01.41 | brookshire[home] | [menu] |
07:01.42 | brookshire[home] | etc |
07:02.03 | FuriousGeorge | HiltonT: oh yeah i forgot, you gotta make your hardphone a friend (type=friend) if u want it to take and make calls |
07:02.06 | FuriousGeorge | thats important |
07:02.15 | HiltonT | I really need to read more! |
07:02.51 | HiltonT | now I'm totally in the dark! |
07:02.56 | FuriousGeorge | HiltonT: you can get your phone logged in but it wont make or take calls till u hook that up in the dialplan |
07:03.29 | HiltonT | I can't make it log in - that's why I need to read up on how * handles extensions |
07:03.41 | HiltonT | and half you said made sense! |
07:03.43 | HiltonT | :) |
07:03.52 | brookshire[home] | well.. first you need to get the channel working |
07:03.56 | HiltonT | half *what* you said made sense :) |
07:04.08 | FuriousGeorge | its easy, under [hardphone] in sip.conf you say context=hardphone, and under [hardphone] in extensions.conf you do a exten=> _X.,1,dial(sip/provider/${EXTEN}) |
07:04.09 | HiltonT | first I need to understand what I'm doing! |
07:04.13 | brookshire[home] | then you can hook it into extensions.conf :) |
07:04.33 | HiltonT | sure, for ppl who understand this! |
07:04.47 | FuriousGeorge | HiltonT: forget calling, lets focus on logging your phone in |
07:04.56 | HiltonT | sure |
07:05.00 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
07:05.23 | FuriousGeorge | the first section of sip.conf is general, forget that for now, the second section is for peers,users,and friends (peer+user=friend) |
07:05.48 | HiltonT | ok, but there's no "second" part in Xorcom default install |
07:05.53 | HiltonT | (listening) |
07:05.58 | FuriousGeorge | make a friend entry for your hardphone there. whats in the [] is its username what comes after the secret= on a subsequent line, is the pw |
07:06.08 | *** join/#asterisk voipguy (n=voipguy@196.200.26.42) |
07:06.10 | HiltonT | username = extension number? |
07:06.30 | FuriousGeorge | username in the phone = whats in the [ ] in sip.conf |
07:06.31 | FuriousGeorge | get it |
07:06.44 | Dr_Ray | [ray] |
07:06.45 | HiltonT | yup |
07:06.49 | Dr_Ray | extension=100 |
07:06.56 | Dr_Ray | user = ray |
07:07.05 | Dr_Ray | secret = ray |
07:07.12 | Dr_Ray | erm username |
07:07.18 | FuriousGeorge | yeah username |
07:07.52 | FuriousGeorge | and i believe its regexten=101 |
07:08.15 | FuriousGeorge | type=friend , you probably want |
07:08.25 | kb1_kanobe | is newjb in stable or only cvs-head? |
07:08.39 | HiltonT | ok, got that done |
07:08.48 | FuriousGeorge | kb1_kanobe: afaik, only 1.0.9 is stable |
07:08.54 | FuriousGeorge | 1.0.X |
07:09.09 | HiltonT | with regexten |
07:09.35 | FuriousGeorge | fire up a console |
07:09.39 | HiltonT | done |
07:09.39 | FuriousGeorge | asterisk -rvvvvvvvv |
07:09.50 | FuriousGeorge | set verbose 50 |
07:09.58 | FuriousGeorge | sip show peers |
07:10.09 | *** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net) |
07:10.18 | HiltonT | done |
07:10.37 | FuriousGeorge | and |
07:10.49 | HiltonT | shows 501 - 510, none of which is what I just configured (4001) |
07:11.02 | FuriousGeorge | oh no, u using a@h or something |
07:11.11 | HiltonT | Xorcom |
07:11.29 | voipguy | any asteriskwin32 users here? |
07:11.33 | FuriousGeorge | how come you have extensions 501 -510 |
07:11.41 | HiltonT | I don't want to get my head around both Linux (been quite some time) and * at the same time |
07:11.51 | HiltonT | NFI - I didn't config 'em |
07:12.07 | FuriousGeorge | NFI? |
07:12.14 | HiltonT | aahhhhh, I see my phone trying and failing to register (with an incorrect username and such) |
07:12.21 | HiltonT | no fscking idea |
07:12.40 | FuriousGeorge | ~pb |
07:12.43 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
07:12.46 | Dr_Ray | isn't anything above set verbose 10 useless? |
07:12.49 | FuriousGeorge | sip.conf |
07:13.14 | brookshire[home] | above 4 |
07:13.16 | brookshire[home] | i think |
07:13.18 | brookshire[home] | maybe 3 |
07:13.53 | FuriousGeorge | i heard it was 11 |
07:14.06 | FuriousGeorge | ~verbose |
07:14.08 | jbot | from memory, verbose is try running the verbose option on that command and looking at output for likely problems. |
07:14.13 | brookshire[home] | 69 is the only true level of verbose ;) |
07:14.24 | Dr_Ray | all I ever get is 68 |
07:14.25 | HiltonT | lol - guess what my password is! |
07:14.36 | brookshire[home] | secret? |
07:14.38 | FuriousGeorge | id harly call slurping and muffled moaning verbose |
07:14.51 | Dr_Ray | the you do me, and I'll owe you one bit |
07:14.53 | FuriousGeorge | hardly* |
07:15.38 | brookshire[home] | heh.. |
07:15.40 | brookshire[home] | so yeah |
07:15.49 | brookshire[home] | not being picky |
07:15.54 | brookshire[home] | but wouldn't it be byte |
07:16.02 | HiltonT | ok - I cleaned up the comments from sip.conf, and here 'tis... |
07:16.05 | HiltonT | [general] |
07:16.05 | HiltonT | context=disabled-sip-insecure-read-getting-started |
07:16.06 | HiltonT | port=5060 |
07:16.06 | HiltonT | bindaddr=0.0.0.0 |
07:16.06 | HiltonT | srvlookup=yes |
07:16.06 | HiltonT | #include "sip-reg.d/*.conf" |
07:16.07 | HiltonT | #include "sip-phones.d/*.conf" |
07:16.10 | HiltonT | [test] |
07:16.12 | HiltonT | regexten=4001 |
07:16.12 | FuriousGeorge | HiltonT: |
07:16.13 | HiltonT | username=test |
07:16.15 | FuriousGeorge | ~pb |
07:16.17 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
07:16.17 | HiltonT | secret=69 |
07:16.17 | HiltonT | type=friend |
07:16.28 | brookshire[home] | <PROTECTED> |
07:16.32 | HiltonT | sorry :) |
07:17.04 | HiltonT | http://pastebin.ca/25372 |
07:17.15 | brookshire[home] | much better ;) |
07:17.26 | HiltonT | :) now I know aboot it, I'll use it :) |
07:18.14 | brookshire[home] | so.. does it register? |
07:18.23 | HiltonT | nope |
07:18.34 | HiltonT | hang on... |
07:19.28 | FuriousGeorge | http://pastebin.ca/25373 <-- that looks good but ur commenting wrong |
07:19.51 | FuriousGeorge | u sure you got the credentials right on your phone? |
07:20.19 | FuriousGeorge | if you do put a context=something and a correspond [something] in extensions.conf |
07:21.00 | HiltonT | yup - user/authuser = "test" and password = "69" and sip proxy and domain = "192.blah" |
07:21.13 | FuriousGeorge | and what does cli say |
07:21.19 | HiltonT | Oct 13 17:21:04 NOTICE[2084]: chan_sip.c:7761 handle_request: Registration from 'test <sip:test@192.168.69.253>' failed for '192.168.69.27' |
07:21.37 | *** join/#asterisk Pikoro (n=pikoro@db.sunny-net.ne.jp) |
07:21.46 | HiltonT | what's that aboot extensions.conf? |
07:22.06 | FuriousGeorge | put a context=something and a correspond [something] in extensions.conf |
07:22.36 | Pikoro | ok, finally got my outbound calls working with sound via sip, however, i can't figure out why i can't receive incoming calls via sip.. gotta be routes but i can't find anyting in extensions.conf or sip.conf that might be affecting it |
07:22.57 | FuriousGeorge | what context is your general section of sip.conf in? |
07:23.06 | Pikoro | from-trunk |
07:23.15 | HiltonT | whereaboots in extensions.conf - me know nosing |
07:23.25 | Pikoro | ahh, not talking to me :D |
07:23.47 | HiltonT | I have no idea what context it is in |
07:23.58 | FuriousGeorge | HiltonT: extension.conf should have only a [general] [global] and [something] in it for now, one on each line |
07:24.16 | HiltonT | some macro-stdexten stuff too |
07:24.28 | HiltonT | and macro-stdmeetme |
07:24.34 | FuriousGeorge | save that one as extensions.conf.example |
07:24.39 | FuriousGeorge | and make your own |
07:24.51 | FuriousGeorge | with those three lines |
07:24.57 | HiltonT | exactly what I'm doing :) |
07:25.18 | FuriousGeorge | Pikoro: do you have a user entry for your sip provider in sip.conf |
07:25.32 | Pikoro | user entry? yes |
07:25.47 | HiltonT | ok - 3 lines with those headings |
07:25.58 | HiltonT | nowt else |
07:25.58 | FuriousGeorge | is that in the incoming from sip context you want it to be in |
07:26.11 | HiltonT | huh? |
07:26.12 | FuriousGeorge | HiltonT: reload at cli and see if phone registers |
07:26.16 | HiltonT | :q |
07:26.19 | HiltonT | :) |
07:26.50 | FuriousGeorge | if it doesnt leave secret= blank and same for phone |
07:27.12 | HiltonT | failed |
07:27.41 | HiltonT | clearing secret= |
07:28.11 | HiltonT | I still have all those extensions listed |
07:28.22 | FuriousGeorge | "reload" |
07:28.27 | HiltonT | did that |
07:28.37 | FuriousGeorge | stop asterisk and start it |
07:28.52 | HiltonT | on debian, how? |
07:29.00 | FuriousGeorge | ? |
07:29.05 | HiltonT | never used Debian before |
07:29.09 | FuriousGeorge | just do a stop gracefully at the cli |
07:29.14 | HiltonT | (only DeadRat) |
07:29.19 | FuriousGeorge | and then start it with an asterisk -cvvvvvvvvvvvvvvvvvvvv |
07:29.25 | HiltonT | k |
07:30.01 | HiltonT | No such command 'stop' |
07:30.05 | FuriousGeorge | your iax.conf could have peers in it too |
07:30.10 | FuriousGeorge | stop gracefully |
07:30.26 | HiltonT | sorry |
07:30.43 | HiltonT | ready |
07:30.48 | Dr_Ray | stop now |
07:30.49 | FuriousGeorge | ready |
07:30.54 | HiltonT | all still there :( |
07:31.04 | FuriousGeorge | check iax.conf |
07:31.11 | Dr_Ray | I don't like stop gracefully, it stops taking new calls |
07:31.29 | HiltonT | I have no calls in/out, initial config :) |
07:31.30 | FuriousGeorge | Dr_Ray: he's not quite in production yet :) |
07:31.35 | Dr_Ray | true dat |
07:31.36 | Dr_Ray | :) |
07:31.39 | HiltonT | FAR, far from it :) |
07:32.40 | HiltonT | ok - iax.conf... |
07:33.04 | FuriousGeorge | HiltonT: doesnt your Xorcom distro come with phone example configs? |
07:33.20 | FuriousGeorge | for what your supposed to put in sip.conf |
07:33.26 | HiltonT | no example configs |
07:33.31 | HiltonT | just pre-configured files |
07:33.38 | HiltonT | which are obviously not too great |
07:33.55 | FuriousGeorge | preconfigured sip.conf with example config for your phone? |
07:33.57 | HiltonT | hence why I'm flailing here |
07:34.02 | HiltonT | nope |
07:34.08 | FuriousGeorge | what phone is it? |
07:34.17 | HiltonT | netcomm v85 |
07:34.20 | HiltonT | (on eval) |
07:34.25 | HiltonT | don't like it |
07:34.50 | HiltonT | but, I can config it in its web interface without issue - can connect to a number of SIP providers |
07:35.26 | HiltonT | can connect it to my AstraSIP and also (one at a time) my FWD account |
07:35.46 | FuriousGeorge | http://www.netcomm.com.au/VoIP/#V100 |
07:35.48 | FuriousGeorge | that thing? |
07:36.14 | HiltonT | nope |
07:36.16 | HiltonT | hang on |
07:36.31 | HiltonT | the V85 a few line sup |
07:37.53 | FuriousGeorge | change your credentials on phone from test to 4001 |
07:38.06 | HiltonT | really basic web config - can't backup/reload data, can't pre-config phonebook nor speed dial entries... |
07:38.18 | HiltonT | 2 secs... |
07:39.36 | HiltonT | still no go |
07:39.54 | FuriousGeorge | hmm |
07:40.11 | *** join/#asterisk Corydon76-home (i=green@pdpc/supporter/sustaining/Corydon76-home) |
07:40.13 | FuriousGeorge | ok i got an idea, get a softphone installed on a pc on that network and see if that thing gets logged in |
07:40.17 | FuriousGeorge | try x-lite |
07:40.20 | FuriousGeorge | www.xten.com |
07:40.32 | HiltonT | I decided a while back (nice pie for lunch) to not get frustrated at this, just to get it working - MUCH better revenge :) |
07:40.40 | HiltonT | k, running now |
07:42.00 | HiltonT | k, failed as "test", trying "4001" |
07:42.16 | HiltonT | failed |
07:42.24 | HiltonT | blank secret |
07:42.28 | FuriousGeorge | so the softphone is failing too? |
07:42.30 | HiltonT | yup |
07:42.46 | HiltonT | Oct 13 17:42:31 NOTICE[4607]: chan_sip.c:7761 handle_request: Registration from 'Quark HT <sip:4001@192.168.69.253>' failed for '192.168.69.28' |
07:42.49 | mutilator | man what a pain in the ass |
07:42.55 | HiltonT | yup |
07:42.59 | mutilator | wonder if i do a full restore |
07:43.00 | FuriousGeorge | throw in a host=dynamic in there, and make sure you have a context=something that corresponds to extensions.conf |
07:43.03 | mutilator | if it'll detect it |
07:43.08 | mutilator | i just updated to the latest bios |
07:43.11 | HiltonT | full restore, or "over the top reinstall" |
07:43.28 | HiltonT | host=dynamic in where? sip.conf? |
07:43.29 | mutilator | i don't wanna overtop reinstall our main mail/web server |
07:43.37 | FuriousGeorge | yah sip.conf |
07:44.12 | HiltonT | there's no context=something |
07:44.20 | HiltonT | that under [test] as well? |
07:44.34 | FuriousGeorge | under test put a context=something |
07:44.41 | FuriousGeorge | in extensions.conf [something] |
07:44.55 | HiltonT | done |
07:45.54 | FuriousGeorge | reload try to log softphone in |
07:45.54 | HiltonT | ain the process |
07:45.54 | mutilator | grrr |
07:45.54 | FuriousGeorge | sip reload |
07:45.55 | mutilator | the promise drivers didn't wor |
07:45.55 | mutilator | k |
07:45.55 | mutilator | not detecting the raid now |
07:45.55 | mutilator | one problem after another |
07:45.55 | FuriousGeorge | promises promises |
07:45.55 | HiltonT | same error |
07:45.55 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
07:45.55 | HiltonT | I HATE PROMISE CONTROLLERS |
07:50.25 | *** join/#asterisk ^X-works (n=drttrtr@81-208-62-98.ip.fastwebnet.it) |
07:51.53 | mutilator | heh |
07:52.05 | mutilator | i shoulda stayed in bed tonite |
07:52.06 | mutilator | and said forget it |
07:52.14 | mutilator | i wanted to be outta here an hour ago |
07:53.42 | *** join/#asterisk [Outcast] (n=bill@222-152-255-158.jetstream.xtra.co.nz) |
07:54.17 | [Outcast] | greetings any bug marshalls in the room? |
08:03.09 | xming | anyone using chan_missdn with 1.2-beta1? |
08:07.04 | shido6 | heh, poor mutilator |
08:10.19 | xming | strings.h seems to be gone in the beta1, while I have it in the CVSHEAD of several months ago |
08:12.09 | xming | oops |
08:12.40 | xming | strings.h is still in beta1 but not on the latest CVS ? |
08:15.58 | *** join/#asterisk Poincare (n=jefffnod@dD5779806.access.telenet.be) |
08:20.07 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
08:20.09 | mutilator | argh! |
08:22.35 | skrusty | anyone know why in cvs head i keep getting "not a local SIP domain" |
08:22.47 | skrusty | when a device tries to register |
08:32.13 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
08:32.28 | nfi|ermes | hi all |
08:33.06 | uter | how do i disable hisax? |
08:34.14 | iDunno | rmmod hisax |
08:35.38 | uter | well, i don't want it to be started at all |
08:36.51 | wasim | then don't modprobe it |
08:38.09 | iDunno | if you're using hotplug, add it in to the blacklist |
08:40.02 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
08:40.15 | uter | on one of my debian systems it works without blacklist |
08:41.23 | uter | well, i think i'll try to find the difference between both systems |
08:44.31 | *** join/#asterisk case_ (n=case@pdpc/supporter/student/case-) |
08:44.31 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
08:44.38 | case_ | hello |
08:44.49 | case_ | i'm making some tests with Asterisk and i would like to generates some calls without any special devices, any ideas? |
08:46.18 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
08:47.03 | iDunno | just set up some sip channels and use softphones? |
08:48.02 | case_ | do you have a softphone to suggest? |
08:49.01 | iDunno | well, there's xlite for windows, or kphone or gnome-meeting for linux |
08:49.15 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
08:50.45 | case_ | iDunno: thanks a lot |
08:50.45 | e3g | how to allow only 1 IP Address to Hit my Asterisk???? IF SIP Registration is not there....?? |
08:51.32 | e3g | I meant to say...a specific IP Address? |
08:52.57 | nfi|ermes | <PROTECTED> |
08:53.29 | nfi|ermes | sometimes it works, sometimes not |
08:53.38 | nfi|ermes | i think some nat prooblem |
08:53.43 | nfi|ermes | any suggestion ? |
08:53.53 | e3g | have you tried NAT=yes ? |
08:53.57 | *** join/#asterisk tsetane (n=tse@212.4.33.75) |
08:54.10 | e3g | Is your asterisk at public IP Address? |
08:54.29 | case_ | e3g: can't you do that with your firewall ? |
08:54.57 | nfi|ermes | n |
08:55.02 | nfi|ermes | it's in my lan |
08:55.04 | e3g | case_: I havent worked on firewall stuff.....I think there is LOKKIT .... or let me know about one? |
08:55.20 | case_ | e3g: what is your OS ? |
08:55.24 | e3g | RH9 |
08:55.27 | nfi|ermes | i tried nat=1 or nat=0 |
08:55.42 | case_ | e3g: i'm suggest you to have a look on firestarter |
08:55.47 | e3g | nfi|ermes : try NAT=YES |
08:56.04 | case_ | it's a gui to controll iptables, the linux kernel embended firewall |
08:56.34 | e3g | what about if I have minimum Installation? |
08:56.47 | e3g | or FREEBSD? |
08:57.07 | *** join/#asterisk JoseHap (n=Jose@202-197.246.81.adsl.skynet.be) |
08:57.13 | JoseHap | hi |
08:57.30 | JoseHap | does anyone know poe cabling scheme of polycom phones |
08:57.30 | case_ | e3g: you can controll iptables with scripts (not that hard), for netbsd i don't know |
08:57.43 | *** join/#asterisk tsetane (n=tsetane@212.4.33.75) |
08:57.45 | JoseHap | is it possible to make it ourselves ? |
08:57.47 | case_ | s/netbsd/frebsd/ |
08:58.10 | HiltonT | POE is pins 7+8 (normally) |
08:58.20 | e3g | case_: alright....thanks |
08:58.40 | nfi|ermes | e3g, it's incredible . With nat=1 , it works after i deactivate and reactivate my lan adapter in the cleint |
08:58.40 | case_ | hum... can someone tell me if my gnomemeeting 1.2.1 can do SIP or if i need the cvs version? |
08:58.44 | case_ | e3g: you're welcome |
08:58.54 | JoseHap | because the polycom 300 doesn't work with a poe 802.2af switch and std cable |
08:59.02 | JoseHap | i read i need an adaptor |
08:59.47 | HiltonT | 300 is supposed to be af compliant, therefore a standard Cat5 should do it |
09:00.22 | JoseHap | on the datasheet is mentionned 802.3af and cisco poe, but optional ;-) |
09:00.48 | HiltonT | and you bought the option? |
09:00.54 | JoseHap | i test it on a small netgear poe switch which works great with grandstream phones |
09:01.06 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
09:01.08 | JoseHap | no, i didn't buy it |
09:01.18 | HiltonT | and you wonder why it doesn't work? |
09:01.23 | HiltonT | :) |
09:01.28 | JoseHap | but as it's an external adaptor i suppose it only changes pins layout |
09:01.35 | JoseHap | but i do not know how |
09:01.48 | HiltonT | t's pi$$poor |
09:02.38 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:04.35 | JoseHap | ok, so maybe i revert 8 and 7 pins to test |
09:04.43 | HiltonT | maybe |
09:04.52 | HiltonT | do you really like the phone? |
09:04.55 | JoseHap | maybe i kill the phone :D |
09:18.13 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
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09:18.24 | *** mode/#asterisk [+o twisted|astricon] by ChanServ |
09:20.25 | *** join/#asterisk malabar (n=mala@bkkb-gw.bitcon.no) |
09:21.23 | *** part/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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09:35.30 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
09:35.31 | kippi | hey |
09:35.48 | kippi | I need a ISDN 2 card, what would people recomend? |
09:36.50 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
09:37.12 | zobia | hello everyone. do u know how to transfer a zap channel to a meetme conference? |
09:39.12 | wasim | zobia: exten => 1,1,Meeme(blah) |
09:39.34 | wasim | zobia: where blah exists in meetme.conf and exten 1 is in a context that is accessible to the zap channel |
09:39.57 | wasim | zobia: there was an obvious typo in that command above, show application meetme for more details |
09:40.39 | zobia | okay. thanks a ot |
09:40.53 | htims | wasim: and how does it work when i want to tansfer someone i've called already to an conference? |
09:41.00 | wasim | sure, donate a little something to earthquake relief :) |
09:41.24 | wasim | htims: use manager api to transfer the call, or use # transfer, see features.conf |
09:41.57 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
09:42.06 | wasim | 40k+ dead, 50k+ injured, 2 million homeless, winter approaching, is not a pretty sight ... |
09:42.13 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
09:45.23 | RoyK | shit |
09:45.32 | RoyK | wasim: and hospitals are standing? |
09:46.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
09:46.15 | puzzled | morning |
09:46.49 | case_ | errr... i don't understand how to generate a call from a sidphone to asterisk (well, i can call asterisk from the sidphone, but how to configure Asterisk to accept this call?) |
09:47.49 | *** join/#asterisk corne (n=corne@ndn-165-157-254.telkomadsl.co.za) |
09:47.52 | JoseHap | do you mean sipphone ? |
09:48.06 | case_ | yes i mean it |
09:48.25 | RoyK | sidphone, sid, the kid that breaks toys...... |
09:48.26 | RoyK | :) |
09:49.08 | case_ | that's the revenge of the sid :) |
09:49.21 | RoyK | http://www.nextron.no/main.php3?PI=11&PNO=13167 |
09:49.23 | RoyK | lol |
09:49.41 | JoseHap | case_ i cannot understand the question about your phone |
09:50.16 | JoseHap | you want asterisk to make something with the call, like answering and playing beeps or music ? |
09:51.19 | case_ | JoseHap: i want asterisk to queue the call |
09:52.03 | case_ | i'm in a call center scheme here. there are incoming calls, some queues, and operators to answer |
09:52.15 | case_ | i don't bother about operators right know |
09:52.35 | case_ | what i want is generating incoming calls with a sipphone and queue the call (forever) |
09:55.10 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
09:55.16 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
09:56.55 | zobia | hello. any one how to transfer a call with agi? |
09:58.15 | zobia | actually i want to transfer a to a meetme in the agi. |
09:58.58 | corne | hi case_ i have just join, if i understand you correctly you want to set up queueing on you asterisk. if that is the case then i can tell you that it is very easy to setup a queue if you are using asterisk@home |
10:00.10 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
10:09.22 | RoyK | eeeeeerrrrrr |
10:09.25 | RoyK | strange |
10:09.32 | RoyK | I try to dial into my new snom 320 |
10:09.36 | RoyK | and it answers 404 |
10:09.38 | RoyK | not found |
10:09.41 | RoyK | stupid |
10:11.20 | case_ | corne: thank you very much but JoseHap is helping me in private... |
10:23.45 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
10:47.58 | gordonjcp | hello |
10:48.43 | gordonjcp | is there a good reason why asterisk wants so many rtp ports? |
10:50.37 | malabar | who says asterisk wants many rtp ports? |
10:51.07 | gordonjcp | well, in rtp.conf it's set up to use 10000-20000 |
10:51.25 | gordonjcp | ten thousand possible rtp ports seems a tad excessive |
10:51.33 | RoyK | gordonjcp: well |
10:51.52 | RoyK | gordonjcp: http://bugs.digium.com/bug_view_page.php?bug_id=3986 is a good reason to allocate a bunch |
10:53.09 | gordonjcp | well, it seems that only a couple of people are having the problem |
10:53.54 | malabar | adjust it to your needs, what is it? 2 per SIP-connection? 1 per IAX-connection? Havent't used 1.0.7... |
10:54.19 | RoyK | gordonjcp: there aren't that many people that use asterisk in large itsp setups |
10:56.31 | HiltonT | RoyK; any reason in particular? |
10:56.35 | *** join/#asterisk contrabanda (n=G@213.131.37.202) |
10:56.41 | contrabanda | hiii |
10:56.49 | contrabanda | please help me with voicemail |
10:57.14 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
10:57.22 | RoyK | HiltonT: reason for what? |
10:57.31 | Akelavlk | Hello, has anybody experience with spanDSP? |
10:57.41 | HiltonT | not being used in large ITSP setups |
10:57.42 | contrabanda | exten => 111,1, Answer |
10:57.43 | contrabanda | exten => 111,2, Dial(SIP/dito,10); |
10:57.43 | contrabanda | exten => 111,3, Playback(custom/111) |
10:57.43 | contrabanda | exten => 111,4, Voicemail, 111 |
10:57.43 | contrabanda | exten => 111,5, Hangup |
10:57.43 | contrabanda | exten => 111,103,Voicemail,b111 |
10:57.45 | contrabanda | exten=> 111, 104, Hangup |
10:57.48 | Akelavlk | My question is if spanDSP is ok or not.. |
10:57.52 | contrabanda | here ia my conf |
10:58.09 | RoyK | ~pb? |
10:58.10 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
10:58.21 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
10:58.42 | contrabanda | ok sorry |
10:58.57 | razu | contrabanda : do you have an account in voicemail.conf ? |
11:00.20 | contrabanda | yes |
11:00.28 | contrabanda | i got such error |
11:00.31 | contrabanda | Oct 13 17:53:44 WARNING[2934]: app_voicemail.c:2406 leave_voicemail: No entry in voicemail config file for ' 111' |
11:00.45 | razu | well |
11:00.58 | razu | the answer should be infront of u |
11:01.09 | contrabanda | 111 => 111, Dito Sekhniashvili, naxalovka@mail.ru |
11:01.09 | RoyK | HiltonT: stability, perhaps? |
11:01.25 | HiltonT | dunno - wondered if there was a reason, not a guess is all |
11:01.34 | HiltonT | maybe no-one's played that much :) |
11:01.45 | HiltonT | as for stability, have you found * to be unstable? |
11:01.47 | Akelavlk | What about spanDSP? |
11:02.08 | contrabanda | whats a problem with my voicemail? |
11:02.17 | HiltonT | it doesn't work |
11:02.28 | RoyK | HiltonT: yes. lots of things needed fixing. and still there are lots of more |
11:02.42 | HiltonT | aha - getting better, though |
11:02.52 | HiltonT | stable in a 30-50 user office, I gather |
11:03.01 | RoyK | asterisk is good for small systems, but never designed to run anything larger than the small office |
11:03.06 | HiltonT | I assume 1.2.0 will rock fairly hard |
11:03.20 | RoyK | HiltonT: we're running a few thousand users on a custom patched 1.0.7 |
11:03.27 | HiltonT | what's the definition of "small" tho? |
11:03.44 | RoyK | HiltonT: do not assume that. it's not really well tested, and prolly won't be when it comes out as 'stable' |
11:03.49 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
11:03.50 | HiltonT | cool - sounds like a reasonable size! |
11:04.00 | HiltonT | what was patched, roughly, to handle this? |
11:04.22 | HiltonT | (I'm *just* getting into *) |
11:04.25 | razu | contrabanda : get rid of the annoyng spaces after commas. and check again |
11:10.13 | contrabanda | ok |
11:10.40 | HiltonT | RoyK; patches? |
11:10.48 | contrabanda | i got the same error |
11:12.36 | contrabanda | is there any rule for creating voicemail mailbox? |
11:14.54 | RoyK | HiltonT: it's not patched to handle the load. it's patched to stop crashing and to stop blocking certain clients from sending INVITES and to handle mysql integration better and a few other things |
11:15.52 | HiltonT | aha - I gather you passed these patches, especially the "stop crashing" ones back to the team for inclusion? |
11:16.24 | HiltonT | MySQL integration would be useful, as would Outlook/Exchange integration (tho I gather it does this thru CAPI) |
11:18.34 | HiltonT | because less crashing is obviously nice :) |
11:18.39 | HiltonT | in any situation |
11:19.50 | RoyK | all are open |
11:29.14 | *** join/#asterisk skydiver (i=skydiver@skydiver.no) |
11:30.27 | skydiver | anyone here who has the correct setup to get a skinny/SCCP phone working with asterisk? I've got it all set up, but I can only dial out to the PSTN and to other SIP phones, but not to other skinny/SCCP and not from SIP/PSTN and inwards. guess this relates to missing extensions, but I haven't been able to find any good examples on how this is done. :-) |
11:33.06 | *** join/#asterisk aor (n=bob@202-197.246.81.adsl.skynet.be) |
11:33.15 | aor | Hi everybody ... |
11:33.22 | JoseHap | hi aor ;-) |
11:34.02 | aor | I'm (desesperately :) trying to make zaptel 1.2 working with a centos 4.1 with standard kernel 2.6.10 |
11:34.17 | aor | I follow the README for udev |
11:34.35 | aor | the devices are created, even if I get this message on the modprobe wctdm |
11:35.07 | zobia | hello can asterisk tranfer a iax2 phone to a meetroom? |
11:35.25 | mbranca | eh, you need to wait some secs after modprobing before launching ztcfg .... |
11:35.30 | mbranca | udev is not immediate |
11:35.31 | mbranca | :) |
11:35.50 | aor | well, it seems that black magic is with me :) |
11:36.04 | aor | i just rmmod and modprobe again wctdm and it is working :s |
11:36.30 | aor | mbranca : yeah i even checked that the devices where created, but I got an error with chanzap |
11:36.35 | aor | but now it is working |
11:36.44 | aor | the machine is damn to slow to my fingers ;-) |
11:39.34 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
11:42.11 | aor | ok, got the tip : I need to modprobe zaptel first, wait a few seconds, then launch wctdm and everything is ok |
11:45.10 | skydiver | anyone with some skinny/SCCP experience here? =) |
11:47.25 | *** join/#asterisk Ldr (n=Lode@194.206.157.226) |
11:51.29 | JoseHap | aor is definitely too fast ;-) |
11:52.49 | zobia | hello. anyone how to transfer a established call to a meetroom in agi. |
11:54.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
11:56.27 | kippi | hi |
11:57.53 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235) |
11:58.01 | MuppetMaster | Hello |
11:58.30 | MuppetMaster | Per this post: http://forums.digium.com/viewtopic.php?t=1892 - does anyone have, no of where to find an Asterisk server connected to an incoming line in India/Pakistan? |
12:00.15 | kippi | if I have ISDN line and I want to put my fax number on it, Do I just get a grandstream box for the fax and just route the calls over asterisk to the fax? |
12:03.09 | skydiver | anyone here who has the correct setup to get a skinny/SCCP phone working with asterisk? I've got it all set up, but I can only dial out to the PSTN and to other SIP phones, but not to other skinny/SCCP and not from SIP/PSTN and inwards. guess this relates to missing extensions, but I haven't been able to find any good examples on how this is done. :-) |
12:03.39 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235) |
12:07.09 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
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12:18.12 | kippi | I need some kinda reporting software, whats the best on out there? |
12:24.51 | RoyK | kiko69: reporting what? |
12:24.58 | RoyK | echo is a good reporting command |
12:25.28 | *** join/#asterisk chidex (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
12:26.50 | *** join/#asterisk freaek (i=goldspe@stallion.psilocy.be) |
12:27.11 | *** join/#asterisk melven (n=melven@d220-236-133-3.dsl.nsw.optusnet.com.au) |
12:27.21 | freaek | good evening all |
12:28.44 | melven | hello has anyone used a digium s100u on a fedora core 3 machine ? |
12:30.20 | freaek | sorry melven, not me, I'm using slackware :) |
12:38.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:42.13 | melven | <PROTECTED> |
12:42.26 | melven | i'm just trying to get it to work period . |
12:44.20 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
12:46.13 | *** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey) |
12:48.27 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
12:49.05 | lehel | hello |
12:49.26 | tzafrir_laptop | hi |
12:50.48 | Ariel_ | Morning all |
12:58.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
13:02.40 | uter | is there a way to tell asterisk the general language? |
13:02.56 | Katty | mew. |
13:03.05 | uter | not just for an extension or channel, but for all the things? |
13:06.15 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
13:06.16 | *** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home) |
13:06.22 | Katty | mew? |
13:06.34 | mistral | i am a little stuck on a problem to configure outbound calls |
13:06.35 | Ahrimanes | muw? |
13:06.45 | tzafrir_laptop | uter, you can speak English in any channel |
13:06.52 | mistral | how would i just perform a catch all ? and dial it on a linue ? |
13:07.20 | tzafrir_laptop | uter, but you refer to SetLanguage? |
13:07.33 | Ariel_ | So anyone know if the 1.2 version of asterisk is office out yet? |
13:07.39 | Katty | mew :< |
13:08.22 | Katty | :> |
13:08.31 | Katty | yay, someone answered me for a change |
13:08.34 | tzafrir_laptop | Ariel_, if it's not in $TOPIC, it didn't happen |
13:08.43 | tzanger | I always say good morning, Katty |
13:09.02 | mistral | so i guess nobody can hlep me then |
13:09.03 | Katty | tzanger: :>> |
13:09.07 | Ariel_ | tzafrir_laptop, yes I did not see it on any mail list either, bummer |
13:09.17 | *** join/#asterisk jontow (i=jontow@ws.woflsys.net) |
13:09.20 | tzanger | mistral: you *really* and I mean *really* need to read the handbook |
13:09.21 | tzanger | it's covered there |
13:09.23 | uter | tzafrir i don't want to use setlanguage in every extension |
13:09.27 | *** join/#asterisk IzNoGooD (n=marc@iznogood.demon.nl) |
13:09.31 | Katty | tzanger: i never read the handbook. |
13:09.38 | tzafrir_laptop | mistral, your answer is not clear |
13:09.42 | Katty | tzanger: more like browsed several pdfs. |
13:09.44 | tzanger | Katty: you're not asking questions that are answered clearly in the handbook. :-) |
13:09.49 | Katty | tzanger: and bugged the crap out of Hmmhesays ;> |
13:09.51 | uter | so i would like to specify it as a global |
13:09.55 | IzNoGooD | Is there any special I need, to get asterisk to play the demo? |
13:09.56 | Ariel_ | mistral, a catch all is like exten => _.,Dial(Blah) but it's dangerest and I would not use it. |
13:10.10 | IzNoGooD | I have a connection through sip and misdn, but both I don't hear any sound |
13:10.17 | tzafrir_laptop | uter, you can set it in sip.conf/iax.conf . But I know of no global way in the dialplan |
13:10.19 | IzNoGooD | Both connect to the asterisk demo |
13:11.07 | uter | tzafrir, i tried it in zapata.conf for incoming calls, but it didn't work :( |
13:11.12 | iDunno | hmm. |
13:11.17 | mistral | you see i have read the aterisk hand book i just dont follow how the pattern matching works on the numbers |
13:11.29 | Katty | tzanger: does jim van mag(etc) talk in here? |
13:11.37 | Katty | meg(etc) i mean |
13:11.50 | *** join/#asterisk coppice (n=chatzill@190.196.17.210.dyn.pacific.net.hk) |
13:11.58 | iDunno | Now using the zaphfc driver, and a zaphfc card, it just doesn't appear to register at all :( |
13:12.36 | Ariel_ | mistral, the files in /usr/src/asterisk/configs/extensions.conf.sample has lots of samples for you to see dialing pattern matching. |
13:13.07 | Ariel_ | Katty, it's an isdn type card used in the EU |
13:13.36 | Katty | Ariel_: isdn type card? |
13:13.44 | Katty | Ariel_: i take it it's not for analog lines. |
13:13.51 | tzanger | Katty: yes, he's JimVanM when he's here |
13:13.52 | tzanger | but it's not often |
13:13.56 | tzanger | nicfe |
13:14.03 | Katty | tzanger: k...does he show up at conferences? |
13:14.08 | tzanger | I just got a request to fax something to 750 different locations |
13:14.17 | Katty | tzanger: email it instead! |
13:14.21 | tzanger | this'll be a good test for txfax :-) |
13:14.24 | Katty | tzanger: save trees! |
13:14.32 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
13:14.36 | tzanger | Katty: a LOT of these places do not have or prefer not to use email |
13:14.51 | tzanger | Katty: I'd agree with you but I know what our customers are like :-S |
13:14.51 | Katty | if they have any sense they'll have a kyocera copier which takes the fax and -> to email |
13:14.58 | tzanger | conferences? I have no idea |
13:15.03 | Katty | WHAT |
13:15.07 | tzanger | he tends to frequent the TorAstricon meetings |
13:15.14 | Katty | what sort of backworld place to these people live in? |
13:15.20 | tzanger | Katty: if they had any sense they'd prefer email in the first place |
13:15.22 | Katty | we've only got 10 people here and /we/ have email! |
13:15.36 | tzanger | Katty: a lot of our customers are literally in the middle of nowhere |
13:15.42 | iCEBrkr | Dude, my parents use email religiously.. |
13:15.49 | Katty | of course we're the ones who setup networks too and stuff, but anyway |
13:15.51 | tzanger | iCEBrkr: I'm not saying old people can't use email |
13:15.54 | iCEBrkr | That's a scary concept considering my dad is a motorhead mechanic |
13:15.56 | tzanger | I know a lot that do |
13:15.58 | Katty | tzanger: i'm in the middle of nowhere :P |
13:16.05 | Katty | tzanger: missouri has corn, cows, and fields. that's it :P |
13:16.10 | tzanger | but I'm saying that some of our customers don't have email |
13:16.14 | Katty | sad. |
13:16.17 | tzanger | Katty: heh, I'm in rural ontario |
13:16.17 | Katty | sad sad sad. |
13:16.27 | tzanger | some of our customers don't have computers |
13:16.35 | Katty | eek! |
13:16.40 | Katty | i'd die. |
13:16.43 | tzanger | where's coppice |
13:16.48 | iCEBrkr | tzanger: But I know where you're coming from. I worked at a mortgage place and a bunch of the appraisers didn't have/use email. They expected faxs and Fed-Ex'd documents. |
13:16.49 | Katty | napping. |
13:17.09 | tzanger | I wonder if a P4/whatever can send 23 simultaneous faxes |
13:17.12 | IzNoGooD | Any ideas why I don't get any sound? |
13:17.24 | Katty | great |
13:17.27 | Katty | i get to go answer phones. |
13:17.34 | Katty | and take care of /walk ins/ |
13:17.35 | iCEBrkr | Katty: Sweet |
13:17.50 | Katty | our nextel staff is going to an All Day conference :<<<< |
13:18.02 | tzanger | fun |
13:18.05 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
13:18.36 | Katty | no, /not/ fun |
13:18.40 | Katty | people are evil |
13:18.44 | Katty | EVIL |
13:19.11 | iCEBrkr | Katty: Yea, well. Ya know, birds of a feather. :P |
13:19.26 | tzanger | iCEBrkr: exactly |
13:19.28 | Katty | big difference between birds and people. |
13:19.29 | iCEBrkr | :) |
13:19.37 | Katty | birds flock, and they get along with flocking. |
13:19.49 | Katty | they don't go visit nextel stores and bitch at the staff who work in the IT department |
13:20.01 | iCEBrkr | Katty: My phone doesn't work!!! *Whine* |
13:20.03 | Katty | because they think everyone is a Dedicated Nextel Team Member |
13:20.11 | iCEBrkr | Katty: I keep getting dropped calls!! *Whine* |
13:20.16 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
13:20.25 | iCEBrkr | You warmed-up yet? |
13:20.27 | Katty | iCEBrkr: nono, it's more like this |
13:20.42 | Katty | iCEBrkr: zomgidroppedmyphoneinthetoilet MAKE IT WORK NOW OR I"M TALKING TO YOUR MANAGER |
13:20.52 | iCEBrkr | lol |
13:20.59 | tzanger | Katty: do they hand the phone to you? |
13:21.04 | iCEBrkr | OH NOEZ!!! |
13:21.09 | Katty | tzanger: actually, no. |
13:21.14 | Katty | tzanger: i know nothing about nextel stuff. |
13:21.17 | tzanger | THANKFULLY no I'd think :-) |
13:21.21 | Katty | tzanger: except that it's amr and IDEN network. |
13:21.34 | tzanger | I have no idea what that means |
13:21.36 | Katty | tzanger: and that sms is just an email address |
13:21.46 | Katty | tzanger: nextel stuff isn't my department ;) |
13:21.59 | iCEBrkr | Speaking of SMS. |
13:22.03 | Katty | yay, SMS |
13:22.12 | Katty | if you put your number, under gaim, under uhh...aim stuff, me thinks |
13:22.15 | Katty | and talk to it |
13:22.20 | Katty | it'll send a message to a mobile |
13:22.22 | tzanger | Katty: can you get me the format of an SMS message that lights up the phone's MWI? I'm sure it's just a specific header, something like MWI: active,8095124533 or MWI: inactive |
13:22.25 | iCEBrkr | I want my SMS to be able to SMS people. I guess I could just use their providers gateway... |
13:22.48 | Katty | tzanger: no clue. all i know is that if i send an email to myphonenumber@messaging.nextel.com it's an SMS |
13:23.03 | Katty | tzanger: which is darn handy when a server hiccups (= |
13:23.11 | Katty | tzanger: or when i get a voicemail, etc. |
13:23.24 | tzanger | Katty: yeah that's pretty standard, I was hoping to light up MWI on the phone (so it shows voicemail waiting, not just an SMS waiting) |
13:23.37 | Katty | tzanger: oh, i've been working on that, too. |
13:23.53 | Katty | tzanger: i tried an email with an attachment ..with several different encodings too....never worked :< |
13:24.04 | tzanger | yeah |
13:24.04 | Katty | tzanger: apparently they use a different type of server for that. |
13:24.13 | Katty | tzanger: it's not an email server. |
13:24.16 | tzanger | there are SMS gatweays that charge $0.17/SMS that can light/delight MWI they claim |
13:24.19 | tzanger | so I know it's possible |
13:24.27 | tzanger | yeah they use their SMS gateway |
13:25.09 | iCEBrkr | tzanger: There's gotta be a 'free' SMS provider out there. :) |
13:25.19 | tzanger | nah |
13:25.21 | iCEBrkr | or one that offers like 100 SMS/mo |
13:25.24 | iCEBrkr | :( |
13:25.24 | Katty | tzanger: i'll ask. |
13:25.32 | Katty | tzanger: but our nextel staff doesn't know |
13:25.35 | tzanger | I want to do it myself I don't want to rely on an external gateway |
13:25.38 | Katty | tzanger: and this area's staff doesn't know |
13:25.44 | tzanger | I'd love to set up an email address on my domain and say to send it there so I could see |
13:25.45 | Katty | tzanger: so i'll have to pester corporate level |
13:25.49 | Katty | tzanger: and i'm sure they wouldn't tell me |
13:25.53 | tzanger | but I don't think they'll do it |
13:26.03 | iCEBrkr | tzanger: Yeah, I'd like to get my own SMS thing set up without relying on the providers gateway |
13:26.11 | Katty | nextel people are really dumb. |
13:26.11 | tzanger | Katty: yeah :-( I was bugging telus here in ontario and they just gave up and said it wasn't psosible which is a bullshit answer |
13:26.20 | Katty | and when you get to the ones who know stuff, they're too busy to talk to you |
13:26.22 | tzanger | even if I had to call an 800# TAP gateway to do it I could |
13:26.35 | iCEBrkr | TAP |
13:26.35 | iCEBrkr | lol |
13:26.37 | iCEBrkr | Old skewl |
13:26.49 | tzanger | iCEBrkr: my other idea was to just get a phone and send the message that way |
13:26.50 | Katty | skwerly |
13:27.00 | *** join/#asterisk mlynch (n=mlynch@email.gcom.com) |
13:27.16 | iCEBrkr | I wrote a TAP interface in Turbo Pascal when I was like 15? |
13:27.28 | Katty | i did some HTML when i was 13 |
13:27.30 | tzanger | iCEBrkr: :-) |
13:27.35 | iCEBrkr | So I could send pages to my PageNet pager :P |
13:27.35 | Katty | i was all bouncy and excited |
13:27.36 | coppice | TAP was a areal PITA |
13:27.38 | Katty | it sucked ;) |
13:27.42 | Katty | but i was happy! |
13:27.42 | tzanger | some of the documentation I have says that that's all it is is high speed TAP these days |
13:27.54 | mutilator | geocities w00t |
13:28.01 | tzanger | I had a pagenet pager! |
13:28.04 | Katty | pfft, geocities |
13:28.08 | Katty | i never needed geocities. |
13:28.14 | mlynch | Can anyone help out a noob? |
13:28.15 | mutilator | heh |
13:28.16 | Katty | i was on irc from like 10 |
13:28.19 | tzanger | coppice: do you forsee any issues in faxing two pages to 750 faxes with app_txfax? |
13:28.28 | iCEBrkr | tzanger: Yeah, but did you WarDial pagers with a targets phone number? |
13:28.31 | Katty | there's always someone on irc that will let you use their server heh |
13:28.31 | mutilator | so what'de ya do host it off your 28k modem |
13:28.33 | iCEBrkr | >:) |
13:28.38 | mutilator | ah |
13:28.44 | tzanger | I have two .tiff files, I need to figure out how to make one 2-page .tiff file yet |
13:28.45 | iCEBrkr | mlynch: What's up noob!! |
13:28.54 | Katty | mutilator: yay apache on 28k ;> |
13:28.57 | tzanger | iCEBrkr: Hahahahaha no no I never did that |
13:29.00 | Katty | mutilator: hey! i did that for awhile! |
13:29.04 | iCEBrkr | tzanger: Can't ImageMajick do that? |
13:29.05 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
13:29.08 | mutilator | heh |
13:29.09 | mutilator | me too |
13:29.10 | Katty | Uther_P: (= |
13:29.13 | coppice | tzanger: those who have trouble with spandsp normally have it when sending. I have yet to find out why. many people seem to send in volume OK |
13:29.19 | tzanger | iCEBrkr: perhaps I'm not sure :-) I have to make sure these tiff files are in the right format too |
13:29.22 | mutilator | my dialup stayed connected for days at a time usually |
13:29.28 | Katty | mutilator: woah. |
13:29.30 | mutilator | most i ever did was 7 day straight connection |
13:29.38 | Katty | mutilator: mine stays connected for a few hours now. |
13:29.38 | mutilator | was wicked cool |
13:29.46 | tzanger | coppice: ok, this will be an interesting test. I'm curious to see if my CPU can handle 24 9600 baud fax sessions at once |
13:29.49 | Katty | mutilator: i bet. |
13:29.56 | mutilator | ya most isp limit it now, this was back when i got it through the community college |
13:29.57 | iCEBrkr | tzanger: Yeah. Give IM a shot, there's a ton of command line tools to manipulate different image formats. I'm learning stuff almost every day |
13:29.59 | tzanger | actually I think I only have 15 B channels turned up so that might be the limiting factor |
13:30.00 | mlynch | Installed asterisk with samples on RHFC2 with 2.6.13.3 kernel. Everything works but I can't seem to get the connection to digium IAX demo server to do anything |
13:30.06 | mutilator | hope on and browse yahoo with lynx and stuff |
13:30.10 | mutilator | hop* |
13:30.19 | Katty | mmm, lynx |
13:30.20 | Katty | and pine |
13:30.22 | Katty | remember pine? |
13:30.25 | coppice | tzanger: 24 should OK on a modern machine |
13:30.25 | mutilator | yep |
13:30.27 | Katty | it was hottt. |
13:30.28 | mutilator | used pine for out email |
13:30.30 | Uther_P | pine is bad |
13:30.31 | mutilator | our* |
13:30.32 | tzanger | yeah this is a P4/something (single processor) |
13:30.43 | Katty | Uther_P: it was dreamy in the mid 80s |
13:30.45 | Uther_P | buggy buggy buggy |
13:30.52 | tzanger | P4/2.4G |
13:30.56 | mutilator | having fun with my .plan file |
13:30.57 | Uther_P | had a few code exploit butz |
13:31.01 | Katty | Uther_P: whiine whine whine ;) |
13:31.04 | Uther_P | er bugz |
13:31.05 | mutilator | put all kinda cool ascii art in it |
13:31.07 | iCEBrkr | tzanger: cat /proc/cpuinfo |
13:31.12 | tzanger | iCEBrkr: I just did |
13:31.14 | iCEBrkr | hehe |
13:31.24 | coppice | tzanger by dual 2.4GHz machine handles more than 48 in tests :-) |
13:31.37 | tzanger | :-) okay then |
13:31.41 | Uther_P | eek... try mutt... its cooler anyway in my opinion |
13:31.42 | iDunno | Katty: poor child. |
13:31.48 | Katty | iDunno: don't poor child me. |
13:32.08 | iDunno | Katty: using pine, though... only unfortunates do that ;) |
13:32.09 | Katty | iDunno: if you're going to poor child me, say it over the fact i have to use exchange |
13:32.20 | Uther_P | haha |
13:32.21 | Katty | iDunno: with outlook...and windows |
13:32.22 | *** join/#asterisk Assid (n=assid@203.115.64.57) |
13:32.27 | *** part/#asterisk IzNoGooD (n=marc@iznogood.demon.nl) |
13:32.29 | tzanger | coppice: it's curious how the 'ht' indicator bug is still there in 2.6.13 |
13:32.40 | iDunno | Katty: *shudder* I feel exceptionally sorry for you all of a sudden |
13:32.47 | Katty | iDunno: kthx. |
13:33.13 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
13:33.22 | Uther_P | the samba team emulated the window networking... I wonder why noone has tried to emulate the exchange server |
13:33.33 | Katty | who would want too? |
13:33.36 | Assid | i think there is a free exchange |
13:33.40 | Katty | for the calendar and tasks and stuff? |
13:33.45 | Assid | there is one i think |
13:33.50 | Uther_P | for efficiency |
13:33.50 | Assid | openexchange i think its called |
13:33.55 | Assid | sf/freshmeat for it |
13:33.55 | tzanger | I used that |
13:33.56 | tzanger | it's crap |
13:34.03 | tzanger | we use exchange4linux now |
13:34.08 | Katty | my next one is going to be Hula Project |
13:34.22 | Katty | mister nixon recommended it |
13:34.25 | tzanger | SuSE OpenExchange Server blew goats with amazing speed |
13:34.39 | coppice | who? Richard? |
13:34.52 | mutilator | tzanger... exchange4linux actaully works for ya? |
13:34.54 | tzanger | Exchange4Linux rocks -- 100% python, uses Postgres, stores EVERYTHING in postgres, Outlook client is in python too (but closed source) |
13:34.54 | Katty | coppice: peter |
13:34.54 | coppice | "There will be no whitewash in the open source" |
13:34.56 | tzanger | mutilator: sure does |
13:35.11 | mutilator | ya gotta pay for the windows driver thing tho right? |
13:35.20 | tzanger | I don't care about that. $50/seat is fine |
13:35.23 | mutilator | i think thats why i opted out of it |
13:35.37 | tzanger | the backend is open source |
13:35.37 | tzanger | that was the key |
13:35.37 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
13:35.37 | mutilator | ya |
13:35.37 | tzanger | I want to migrate off outlook eventually |
13:35.37 | Katty | now then! |
13:35.43 | Katty | so i had that polycom 500 problem |
13:35.47 | mutilator | it's a one time forever updates free fee right? |
13:35.54 | coppice | tzanger: do it today |
13:35.55 | nfi|ermes | in my default dialplan there is: $[foo${ARG3} = foo]?3:2 . Anyone knows what is this ? |
13:36.00 | Katty | if anyone recalls, we were having the issue of we couldn't hear them, but they could hear us. |
13:36.05 | Katty | and then you manually rebooted the phone |
13:36.09 | Katty | and it was magically all better. |
13:36.13 | Katty | until it goofed up again. |
13:36.16 | mutilator | heh well my office wanted to use it for the outlook calendaring feature |
13:36.20 | Katty | soooooo.......i watched the rtp packets |
13:36.23 | mutilator | so i just built a nifty one into the website |
13:36.23 | Assid | 30G ? |
13:36.24 | tzanger | coppice: I hope to do it tonight actually |
13:36.26 | Katty | and it's not a port thing |
13:36.33 | Assid | nice.. what the hell you storing in there |
13:36.36 | Katty | any Other suggestions of things to look at? |
13:36.46 | tzanger | not keen on tying up all the B channels during working hours |
13:36.55 | Katty | and someone better answer me :P |
13:37.06 | Katty | or i shall start poking. |
13:37.06 | tzanger | Katty: exchange4linux |
13:37.19 | Katty | tzanger: i don't think we're talking about the same problem. |
13:37.19 | tzanger | outlook doesn't know it's not talking to Exchange, which means that all the good stuff works |
13:37.35 | Katty | tzanger: nonono, the polycom problem. |
13:37.36 | tzanger | and all your data sits in an open system |
13:37.36 | Katty | tzanger: you silly sod :P |
13:37.48 | tzanger | there's something really *cool* about doing arbitrary SQL queries on your contacts, todos and even emails :-) |
13:37.51 | tzanger | Katty: oh :-) |
13:38.05 | tzanger | it's solved a few problems for us already (being able to access all our outlook data) |
13:38.18 | *** join/#asterisk Caede (n=caede@sentry.zoom.com) |
13:38.28 | tzanger | hmm I could poke this all through nufone's fax gateway too |
13:38.29 | Katty | tzanger: after roughly 8-10 calls person can no longer hear any incoming audio. manually reboot phone, everything is ok. |
13:38.36 | tzanger | Katty: hmm |
13:38.37 | Katty | tzanger: explain /why/ |
13:38.46 | Katty | tzanger: also, please note, i watched rtp packets. |
13:38.51 | tzanger | is it allocating RTP ports outside of what Asterisk is saying it can do? |
13:38.52 | Katty | tzanger: nothing seemed unusual. |
13:38.58 | Katty | tzanger: no, that's what i looked at |
13:39.04 | Katty | tzanger: it's not too high. |
13:39.09 | tzanger | Katty: have you called polycom support? |
13:39.15 | Katty | tzanger: :< |
13:39.20 | Katty | tzanger: why would i do a silly thing like that? |
13:39.40 | Katty | tzanger: you know they'll give me the run around. |
13:39.51 | tzanger | Katty: then you politely tell them you'll send the phone back |
13:39.54 | Katty | tzanger: and then it'll come down to a We Don't Know type thing. |
13:40.03 | iDunno | *YAY*! |
13:40.04 | Katty | tzanger: but it's not one phone. it'll all 10 of our phones. |
13:40.11 | iDunno | I R TEH W1NN0R! |
13:40.17 | Katty | tzanger: something tells me they're not /all/ bad |
13:40.21 | tzanger | Katty: then you say, "that's terribly saddenning. I guess I'll send my phones back and go with the Cisco solution. I thought you'd be able to help." |
13:40.23 | iDunno | (or, rather, at bloody last I've got working ISDN) |
13:40.34 | Katty | tzanger: ha, that won't work. |
13:40.36 | tzanger | Katty: do you have another phone to test? Have you done a factory reset? |
13:40.49 | Katty | tzanger: yes, i've already done a factory reset. |
13:41.02 | Assid | Katty: restart network |
13:41.10 | Katty | Assid: that's not the issue either. |
13:41.26 | Assid | Katty: try pinging the ip from the linujx box |
13:41.26 | Katty | Assid: we've had power outages where everything was reset. changed nothing |
13:41.33 | Katty | Assid: that's fine too. |
13:41.40 | Katty | Assid: these phones recieve calls perfectly fine |
13:42.05 | Assid | hrmm.. weird |
13:42.09 | Katty | tzanger: i really thought it was the rtp port issue problem thingy |
13:42.21 | Katty | tzanger: but it went up to something like 2257 and dropped back down to 2222 |
13:42.28 | Katty | tzanger: the person never rebooted their phone in between |
13:42.42 | Katty | i /really/ need to take a look at sip debug |
13:43.01 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
13:44.22 | syle | what is happening with intel and asterisk? they are talking about shipping some intel machine with asterisk business edition in nov/05 |
13:44.35 | Ahrimanes | link? |
13:44.45 | syle | http://blog.tmcnet.com/blog/tom-keating/voip/intel-and-asterisk.asp |
13:45.03 | tzanger | hmm |
13:45.18 | syle | http://www.voipplanet.com/solutions/article.php/3551016 |
13:45.39 | *** join/#asterisk KeX_WorX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
13:45.44 | KeX_WorX | hi |
13:46.06 | KeX_WorX | can some1 tell me how i tell asterisk to log infos bout queues? |
13:46.24 | KeX_WorX | this should be an queue_log file in /etc/log/asterisk or? |
13:46.29 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
13:46.44 | KeX_WorX | saw this on an asterisk srv. now i set up a new one and there isn't such a file : / |
13:46.54 | syle | idk what your logger.conf is set to log dude |
13:46.58 | mutilator | i ought to get one of them fancy asterisk certifications |
13:47.03 | Caede | Okay... frustrated-at-the-end-of-my-wits question. I have a zaptel TE110P card connected to a PBX using E&M (wink-start) signalling. We get constant clicking/popping sounds on any call traversing through the zap interface. I don't see the clicks/pops when using ztmonitor, and it's only heard by the TDM-end (at the PBX side). Does this sound like timing or something else? If I turn on... |
13:47.04 | Caede | ...extra extra debugging for zaptel, I get oodles of "T1: Lost our place, resyncing" (offset 28, although that's not in the message) |
13:47.09 | mutilator | probly start being a resume stuffer here soon |
13:47.24 | mutilator | one of the useful ones anyway |
13:47.30 | KeX_WorX | syle, what do i have to write in logger.conf ? |
13:47.54 | KeX_WorX | i've these files from both srv's one with quele_log file, one without |
13:48.01 | KeX_WorX | the logger.conf files are exactly the same |
13:48.35 | syle | enable your debug log |
13:48.36 | KeX_WorX | *grml : ) |
13:48.39 | KeX_WorX | queue_log |
13:48.39 | KeX_WorX | sry |
13:48.41 | *** join/#asterisk apardo (n=w0w0@1.Red-83-46-192.dynamicIP.rima-tde.net) |
13:49.03 | nfi|ermes | in my default dialplan there is: $[foo${ARG3} = foo] . Anyone knows what is this ? |
13:49.19 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
13:49.23 | *** join/#asterisk IzNoGooD (n=marc@iznogood.demon.nl) |
13:49.31 | nfi|ermes | does it check if the ${ARG3} is empty ? |
13:49.41 | Akelavlk | Hello. Has anybody experiences with HylaFax of spanDSP? |
13:50.00 | Uther_P | nfi|ermes: yes, |
13:51.16 | syle | i think arg3 is a call to a macro possibly |
13:51.31 | syle | so check where its calling it |
13:51.38 | nfi|ermes | thx Uther_P |
13:53.02 | Katty | yay, new server |
13:53.13 | Katty | rev drive, 2 300 gig hard drives with raid 1 |
13:53.18 | Corydon76-home | nfi|ermes: you could also do $[${LEN(${ARG3})}] |
13:53.20 | Katty | pretty little box. |
13:53.45 | Katty | too bad windows is going on it :< |
13:53.47 | Corydon76-home | but that would be the opposite |
13:53.58 | funxion | why cant I find wct2xxp anywhere in zaptel |
13:53.58 | Beirdo | awww, poor little box. |
13:54.02 | Katty | Beirdo: i know :<< |
13:54.09 | Corydon76-home | since an empty ARG3 would have length 0, evaluating to false |
13:54.13 | KeX_WorX | syle, had to uncomment queue_log => in logger.conf |
13:54.16 | Beirdo | these things happen, I guess |
13:54.19 | Beirdo | good morning, Katty |
13:54.39 | KeX_WorX | syle, but on the srv where the loging works, this line is commented and it logs? how is that? u know? |
13:56.25 | syle | did you read the file? |
13:56.31 | syle | cause it defaults to yes |
13:57.55 | IzNoGooD | What is the best channel to get help on using asterisk? |
13:58.17 | jake1932 | here is the asterisk channel |
13:58.52 | IzNoGooD | Ok, the question is, I try to use asterisk with misdn, and the connect works |
13:59.04 | IzNoGooD | and asterisk starts playing the demo-congrats |
13:59.09 | IzNoGooD | but I don't hear any sound |
13:59.31 | *** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it) |
13:59.45 | *** join/#asterisk mlynch (n=mlynch@email.gcom.com) |
14:00.13 | jake1932 | IzNoGooD: can you get audio using other devices (SIP/IAX/etc)? |
14:00.22 | IzNoGooD | no |
14:00.23 | FABRIZIOxxx | hello all .. how can i test the efficiency of the lan .. i'm using tracepath and i'm getting strange results ... is it ok .. ? |
14:00.28 | IzNoGooD | sip isn't working either |
14:00.36 | jake1932 | IzNoGooD: any error messages |
14:00.51 | mlynch | Can anyone help a noob get digium iax demo server demo working? |
14:01.06 | IzNoGooD | jake1932: none |
14:01.12 | funxion | does anyone here have a TE210p? |
14:01.20 | jake1932 | IzNoGooD: have you done "rtp debug" from the CLI? |
14:01.30 | IzNoGooD | Trying |
14:02.25 | IzNoGooD | jake1932: outputs <Sent RTP packet to 82.161.0.227:49194 (type 0, seq 60201, ts 528960, len 160)> which is my external ip |
14:02.50 | IzNoGooD | but both hosts asterisk and sip-client are on localnet 10.0.0.x |
14:02.59 | jake1932 | IzNoGooD: should be sending from the asterisk box to your softphone or hardphone ip |
14:04.01 | IzNoGooD | should I set the localip somewhere? |
14:04.07 | IzNoGooD | Where does it get my externel ip from? |
14:04.17 | jake1932 | IzNoGooD: using NAT? |
14:04.26 | IzNoGooD | Not in between |
14:04.39 | IzNoGooD | I have a seperate router |
14:04.43 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
14:04.48 | IzNoGooD | with nat |
14:04.52 | jake1932 | IzNoGooD: sip or iax? |
14:04.57 | IzNoGooD | sip |
14:05.02 | jake1932 | IzNoGooD: sip is sip.conf - iax is iax.conf |
14:05.10 | IzNoGooD | yes, sip.conf |
14:09.32 | KeX_WorX | syle, where is that what u said? that queue_log defaults to yes ? can't find it |
14:10.12 | syle | in your logger.conf dude |
14:10.20 | KeX_WorX | can't find it |
14:10.26 | KeX_WorX | asterisk 1.0.7 |
14:10.28 | syle | ; This determines whether or not we log queue events to a file (defaults to yes). |
14:10.28 | syle | ;queue_log = no |
14:10.28 | syle | ; |
14:10.54 | KeX_WorX | nop. that's not in my file |
14:11.02 | syle | oww no idea then dude, i use latest cvs myself |
14:11.11 | tzanger | ok app_txfax returns 0 or -1, how do I see that in the dialplan? |
14:11.26 | tzanger | n+101 isn't around anymore I don't think and README.variables doesn't show any kind of "application result code" |
14:11.52 | *** join/#asterisk mithro (n=tim@tagung-233-198.tagung.uni-hamburg.de) |
14:12.13 | syle | how is it not around |
14:12.18 | KeX_WorX | this is my logger.conf http://pastebin.com/392243 |
14:12.34 | KeX_WorX | use the deb package. perhaps it's in a new version |
14:13.18 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
14:13.45 | syle | is 1.0.7 latest stable? |
14:14.04 | KeX_WorX | ähm, dunno : ) |
14:14.14 | file[laptop] | no, 1.0.9 is the latest |
14:14.15 | KeX_WorX | 'latest' stabel on debian |
14:14.16 | KeX_WorX | k |
14:14.45 | syle | is realtime stuff at least supported? |
14:14.52 | file[laptop] | in stable? no |
14:15.04 | IzNoGooD | ah sip works now |
14:15.08 | KeX_WorX | looked at a logger.conf on a 1.2-beta1 asterisk. ther is the queue_log = no |
14:15.13 | syle | ie: extconfig.conf |
14:15.19 | Ariel_ | 1.0.9.2 does not support realtime |
14:15.28 | syle | iaxusers peers etc |
14:15.45 | file[laptop] | stable does not support realtime. |
14:15.51 | syle | well that is gay |
14:15.57 | file[laptop] | not really, it's bug fixes stuff |
14:15.59 | Ariel_ | no it's great that it does not support realtime |
14:16.08 | syle | depends on what your doing |
14:16.12 | syle | for me it would suck |
14:16.26 | file[laptop] | then don't use stable |
14:16.26 | Ariel_ | if you need realtime you don't use stable |
14:16.26 | syle | i don;t :) |
14:17.14 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
14:17.22 | syle | umm how can you not see advantage is realtime |
14:17.26 | syle | in |
14:17.39 | syle | you new to asterisk? |
14:18.02 | tzafrir_laptop | syle, it has some atvantages, but a whole slew of problems. |
14:18.10 | Ariel_ | syle, no I have been working with asterisk for over 3 years. I can do allot faster things with normal setup then realtime. |
14:18.30 | syle | how can you automate adding iaxusers etc from webpage without it |
14:18.32 | Ariel_ | it's not complete and has issues |
14:18.43 | tzafrir_laptop | syle, sure |
14:18.43 | Ariel_ | manager api |
14:19.04 | syle | manager api? how so? |
14:19.06 | tzafrir_laptop | syle, edit config file and reload |
14:19.19 | tzafrir_laptop | AMPortal does that |
14:19.40 | syle | i;ve played with manager api in perl, never saw a way to add realtime users |
14:19.56 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
14:20.10 | tzafrir_laptop | to add real-time users you simply add them to the table, right? |
14:20.16 | syle | right |
14:20.17 | Ariel_ | syle you submit then your perl does a write then a reload |
14:20.29 | syle | a reload on production! |
14:20.31 | syle | you nuts |
14:20.41 | Ariel_ | syle, it works great |
14:20.49 | Essobi | Anyone remember the name of that sip program you could use to hand craft packets to check for responses to servers? |
14:20.51 | syle | you must not have many users |
14:20.57 | Ariel_ | you can reload sip by it's self and also extensions by them self |
14:21.03 | tzafrir_laptop | syle, but then Asterisk has much more work to do at call time, just when it needs to be most responsive |
14:21.22 | syle | i;m to bleeding edge as it is it is scary |
14:21.24 | Ariel_ | prayer time bbl |
14:21.33 | syle | cvs head and mysql 5.x for trigger support |
14:21.53 | Essobi | ;) |
14:21.59 | Essobi | That is bleeeding edge. |
14:22.09 | Essobi | But triggers are sweeet. |
14:22.17 | syle | yes i love them! |
14:22.25 | Caede | Okay... guess my first question was too long. How 'bout this: any reasons NOT to use icc to compile Asterisk? |
14:22.29 | syle | god i don;t ever want to go back to 4.x |
14:22.34 | Essobi | We're about to migrate some stuff to 5, for triggers and real sub-selects, and views |
14:22.54 | tzafrir_laptop | syle, then why not use pgsql and have tried-and-tested triggers? |
14:23.01 | tzafrir_laptop | and sub-selects |
14:23.36 | syle | because that would take more days of learning postgres and i really would rather not commit that, + i hate fact you have to export a whole table just to back it up |
14:23.46 | mutilator | anyone know if it's possible to upgrade a win2k pc from Standard PC mode to ACPI multiproc pc mode or MPS multi proc mode w/o reinstalling win2k over top.. i tried to just update from standard to acpi last night and i had to restore windows cause it bsod.. |
14:24.04 | syle | same with innodb, but not my myisam tables |
14:24.31 | tzafrir_laptop | syle, BTW: did you read about oracle and innodb? |
14:24.34 | syle | + realtime support is native in asterisk with mysql , not postgres |
14:24.40 | syle | postgres needs ODBC |
14:24.54 | tzafrir_laptop | isn't there native postgres in head? |
14:24.56 | syle | is supported natively in asterisk with mysql |
14:25.34 | *** part/#asterisk mlynch (n=mlynch@email.gcom.com) |
14:25.37 | Essobi | some poeple are just comfortable with what they know |
14:25.42 | Essobi | I know I am. :) |
14:25.49 | syle | pretty much, i been using mysql since it first came out |
14:26.17 | Essobi | mysql does outshine pgsql when it comes to speed in large large large databases too. |
14:26.23 | tzafrir_laptop | Oracle recently bought the company that develops Innodb: http://www.oracle.com/corporate/press/2005_oct/inno.html |
14:26.35 | Essobi | Nice. |
14:26.40 | syle | really? i thought mysql shined on smaller tables |
14:26.58 | syle | read an article somewhere postgres performs better on more data tables |
14:27.07 | darkskiez | and then there was sqlite |
14:27.29 | syle | but bastards were probably doing comparison on myisam instead of innodb so maybe they were getting table locking instead of row locking |
14:27.31 | *** join/#asterisk _T3_ (n=rposada@53.228.uio.satnet.net) |
14:27.46 | tzafrir_laptop | actually, if you don't need to access the data from a different server, sqlite would probably mean much less administration headache |
14:28.20 | syle | thats scary, oracle will no longer maintain it opensource i bet |
14:29.02 | syle | i am very found of triggers |
14:29.13 | syle | spent a whole day with them over the weekend |
14:29.22 | tzafrir_laptop | syle, it's GPL. The worst they could do is stop maintaining it. But from the POV of MySQL AB the worst that could happen is they could no longer resell it as part of their non-free product |
14:29.30 | syle | abit tricky to learn to do complex shit, but its great when you know it |
14:30.26 | syle | although i guess monday wasn;t a holiday for you guys |
14:30.39 | syle | was canadian thanksgiving here |
14:31.28 | syle | anyways, i;ve never seen how to add a user to a config file over manager API |
14:32.38 | syle | but i disagree with calling sip reloads all the time for realtime stuff, i hope extconfig.conf makes it into stable soon |
14:34.47 | Ariel_ | syle, only way it makes it is if the 1.2beta1 becomes stable release. |
14:37.37 | tzanger | ok why the fuck isn't * seeing my outgoing .callfiles |
14:38.28 | tzanger | wasn't there some kind of module I had to load ot get that functionality |
14:38.39 | tzanger | /var/spool/asterisk/outgoing exists, I drop a file in and it isn't picked up |
14:38.48 | syle | you use mv right? |
14:39.34 | syle | well i;ll make my script that places calls into that directory |
14:39.36 | syle | paste |
14:39.49 | syle | cp /etc/asterisk/callfile.txt /etc/asterisk/callfile.txt.old |
14:39.49 | syle | chown asterisk.asterisk /etc/asterisk/callfile.txt |
14:39.49 | syle | mv /etc/asterisk/callfile.txt /var/spool/asterisk/outgoing/callfile1.txt |
14:39.49 | syle | cp /etc/asterisk/callfile.txt.old /etc/asterisk/callfile.txt |
14:39.55 | syle | simple shell script |
14:40.08 | iDunno | anyone using the zaphfc driver? |
14:40.13 | iDunno | and has working callerid? |
14:40.25 | tzanger | yeah |
14:40.35 | tzanger | but it's not getting picked up. usually the file is deleted within a few seconds of being put there |
14:40.51 | syle | tail /var/log/asterisk/debug |
14:40.55 | syle | after you do it |
14:41.14 | tzanger | I am tailing it |
14:41.15 | tzanger | nothing |
14:41.40 | syle | CLI show anything? |
14:42.17 | syle | idk if you have verbose on or not |
14:42.37 | Dr_Ray | is the file a valid call file? |
14:42.46 | tzanger | no cli norally would show it being processed |
14:42.53 | tzanger | it's like the part of asterisk that schedules this is not on |
14:43.05 | syle | did you try restarting asterisk? |
14:43.54 | syle | only reason i;ve ever heard of it doing that is if your timestamp on that callfile is in the future |
14:44.16 | Dr_Ray | syle - that's an awesome feature btw |
14:44.31 | tzanger | syle: yeah |
14:44.37 | syle | wakeup call stuff |
14:44.40 | syle | yeah |
14:44.41 | syle | :) |
14:44.47 | Dr_Ray | yeah, saves a cron job |
14:45.00 | *** join/#asterisk case_ (n=case@pdpc/supporter/student/case-) |
14:46.01 | tzanger | syle: hmm |
14:46.03 | tzanger | don't think so |
14:46.14 | tzanger | I mean even if the callfile's garbage asterisk usually deletes it |
14:47.26 | tzafrir_laptop | the call file needs to be readable by asterisk. If it's not immedietly deteled , it is probably not readable |
14:47.29 | syle | it just sits there? |
14:47.49 | syle | did you chown it to the asterisk users before placing it there? |
14:47.55 | tzafrir_laptop | chown/chmod it before you move |
14:47.59 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
14:48.03 | nfi|ermes | how shuold i choose wich codec to use ? |
14:48.04 | syle | well he says it does not show up in his debug log at all though |
14:48.27 | syle | debug would show you reason it failed usually, permission issue or whatever |
14:48.31 | Brijn | Quick q, where do you define on what IP's asterisk will listen for incoming registration requests.. |
14:48.31 | *** join/#asterisk darwin35 (n=darwin35@208.139.193.178) |
14:48.35 | tzafrir_laptop | nfi|ermes, sip show channels / iax2 show channels |
14:48.35 | tzanger | found it syle |
14:48.39 | tzanger | pbx_spool.so was noload'ed |
14:48.48 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
14:48.56 | syle | playing to much with modules.conf? |
14:49.01 | darwin35 | good morning my fellow asterisk tramps,whores, sluts, and all others |
14:49.09 | syle | hehe |
14:49.22 | darwin35 | have you Prosatuted your asterisk box latly |
14:49.36 | darwin35 | have you made it work its tits and ass off |
14:50.26 | darwin35 | has it made you so happy you could just burst with joy |
14:50.56 | darwin35 | no I am just happy got my first Pay check from new job |
14:51.05 | syle | kewl, what are you doing? |
14:51.05 | darwin35 | its nice to be working again |
14:51.08 | *** join/#asterisk Ocean_big (n=morey@ax113-4-82-227-179-176.fbx.proxad.net) |
14:51.12 | darwin35 | www.teliax.com |
14:51.14 | Ocean_big | Hi! |
14:51.19 | darwin35 | Support office |
14:51.39 | syle | you do asterisk stuff? |
14:52.05 | syle | or you the new phone monkey hehe |
14:52.07 | darwin35 | everything from support to loading up new servers with asterisk realtime to |
14:52.23 | darwin35 | testing |
14:52.38 | syle | kewl what database you using for realtime? |
14:52.38 | limbique | how to do pickup with asterisk? |
14:52.38 | *** join/#asterisk royth1 (n=royth1@200.121.129.178) |
14:52.44 | Ocean_big | i need help with asterisk... i 'm newbee. Someone to help me ? |
14:52.46 | darwin35 | I am going to be building 4 new boxes today if the boss gets here with them |
14:52.58 | tzanger | ok that sent WAY too fast to be successful |
14:53.01 | *** join/#asterisk rikstah (n=rick@84.93.87.216.plusnet.ptn-ag2.dyn.plus.net) |
14:53.06 | tzanger | postgres |
14:53.08 | tzanger | postgres |
14:53.08 | darwin35 | dont ask for help . just state the issue your having and wait for a responce |
14:53.11 | syle | :( |
14:53.24 | rikstah | hey, i just registered on the voip-user.org wiki and clicked the validation link i was emailed, but i still can't sign in...any ideas pls? |
14:53.30 | tzanger | what's worng with pg? It Just Works |
14:53.53 | darwin35 | how is pg clustering now days |
14:53.58 | syle | i;m not saying anything is wrong with it, i;m asking what db darwin uses for his realtime stuff |
14:54.22 | darwin35 | here we use MYSQL at the min but I want to move to PG |
14:54.44 | darwin35 | but I have not heard the update on the clustering issues they had 4 months ago |
14:55.09 | Ocean_big | my problem : asterisk dont answer any incomming call, I can only call (SIP or non-IP phone) |
14:55.14 | syle | i heard people switching from postgres to mysql for backup db servers heheh |
14:55.31 | tzafrir_laptop | Ocean_big, what do you see on the CLI? set verbose 3 |
14:55.52 | tzafrir_laptop | ~pb |
14:55.54 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
14:56.19 | *** join/#asterisk hotgrits (n=hotgrits@192.160.238.156) |
14:57.10 | Ocean_big | tzafrir_laptop, I 'm newbee.. I use Asterisk@home, i don't understand what is the CLI. |
14:57.34 | *** join/#asterisk gdh (i=foobar@bum.net) |
14:57.40 | tzafrir_laptop | Ocean_big, as root, run: asterisk -r |
14:57.46 | darwin35 | well I am going to setup a 2 pg boxes today and test clustering and 1 fbsd box |
14:57.50 | Ocean_big | ok |
14:57.53 | darwin35 | and go on from there |
14:58.01 | gdh | Quickie - is 1.2.0 final likely to be announced at Astricon. or am I better just installing beta1 on a fresh box? :) |
14:58.32 | case_ | i still can't call my asterisk server with a sipphone :( |
14:58.44 | *** join/#asterisk salmandr (n=salmandr@mdsnwinas02pool2-a226.mdsnwi.tds.net) |
14:59.00 | Ocean_big | tzafrir_laptop,it's ok.. i can see " Verbosity is at least 3 |
14:59.50 | tzanger | ok zaptel bug |
15:00.22 | tzanger | IAX2 call to a * box which dial(Zap/g1) -- if the zaptel channel is busy it sends back BUSY but IAX2 sees hangupcause 21 - rejected... not busy |
15:01.03 | tzafrir_laptop | Ocean_big, good. Now what happens when you call? |
15:01.29 | *** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net) |
15:01.54 | tzafrir_laptop | gdh, if it were to be announce, it were already be announced, I guess |
15:02.11 | tzanger | I have to explicitly say Busy() in the dialplan or I get a rejection not busy even though chan_zap is clearly saying busy |
15:02.30 | case_ | when i try to call asterisk with a linphone, it says "user cannot be found at given address". i want that anybody can call the asterisk server, what should i do? |
15:02.36 | Ocean_big | tzafrir_laptop, nothing ^^ |
15:02.38 | tzafrir_laptop | dgh, and if it is going to be released today, you really don't want to install it. Give it a day or two to sort out the most obvious problems |
15:02.39 | gdh | tzafrir: I wondered if they'd leave it until the last day, tho :) |
15:03.11 | tzafrir_laptop | If you want to experiment, you can always take current HEAD |
15:03.29 | Ocean_big | tzafrir_laptop, but when i call, all seem to be OK |
15:03.49 | tzafrir_laptop | Ocean_big, how do you know that the call attempt did get to the Asterisk machine? Is it SIP? |
15:04.07 | gdh | tzafrir: Don't suppose you've got any 1.2.x xorcommed debs? :) |
15:04.25 | tzafrir_laptop | of the beta. Uploading them now... |
15:04.52 | gdh | haha nice timing :) paste a sources.list line when you're ready :) |
15:06.20 | Ocean_big | tzafrir_laptop, i don't know.. i think the call doesn't attemp asterik machine... but why ? asterisk machine is connect on my phone line. I call it with my mobil phone |
15:06.54 | tzanger | coppice: you around? txfax seems to be dying immediately and even with option caller|debug I'm not getting any output saying why |
15:07.16 | coppice | sad, isn't it? |
15:07.16 | Ocean_big | i 'm french, there is something to customize for french phone ? :s |
15:07.55 | tzanger | coppice: ha |
15:08.09 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
15:08.33 | *** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net) |
15:08.52 | tzanger | coppice: I just called my own number with txfax and the * cli says it hung up, but I am still hearing beeping |
15:09.24 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
15:10.45 | johnm | Has anyone else had problems with * HEAD, where we redifne the sent CLI. If the CLI we recieve is less than 5, we set it to somethitng else. Basically what happens is: If the original CLI is something (anything) it works. if the original CLI is blank, then we set it.. just like we do any other time.. and it doesn't work. it simply doesn't send a CLI. |
15:11.13 | johnm | is anyone else having problems with this? |
15:11.56 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
15:14.23 | *** join/#asterisk srsergio (n=asterisk@194.red-217-216-155.user.auna.net) |
15:14.44 | *** join/#asterisk [_gordo_] (n=gordo@bl5-187-197.dsl.telepac.pt) |
15:15.09 | [_gordo_] | zt_maxpulsetime for Matra Nortel pbx ? |
15:15.17 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
15:15.38 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
15:15.46 | generalhan | whats up everyone ? |
15:16.55 | *** join/#asterisk fabsoft (n=settete@mercedes.cs.unibo.it) |
15:17.07 | fabsoft | salve a tutto |
15:17.08 | tzanger | ahh it's the local channel that is hanging up |
15:17.10 | fabsoft | i* |
15:17.11 | tzanger | once the call is bridged |
15:17.51 | tzafrir_laptop | Ocean_big, so go a bit below to the network level |
15:18.01 | generalhan | i have a question about the asternic FOP: i want to have 2 different FOPs to run so i copied all the files do a new dir and adjusted where the flash_dir is, but when i run the op_sever.pl from the new directory it still shows the old FOP, even on the new URL ? any ideas ? |
15:18.30 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
15:18.38 | rikstah | is anyone using a bluetooth headset and asterisk to talk on SIP(etc) calls ? |
15:18.46 | tzafrir_laptop | you could run a sniffer to check that SIP packets are indeed sent. I like to use tcpdump for that: tcpdump -n 'udp port 5060' |
15:18.56 | syzygyBSD | I have used it before, not right at this minute though |
15:19.12 | rikstah | syzygyBSD that to me? if so is it realiable |
15:19.32 | syzygyBSD | rikstah: have you used a bluetooth headset with a cellphone? |
15:19.42 | rikstah | syzygyBSD yeah of course |
15:19.47 | Uther_P | what is 'syzygy' BSD ? |
15:19.47 | fabsoft | i live in italy, a have got a telecom isdn TN mode network card, and also a hfc-pci chipset card, what channel i need to use hfc-pci for send/receive call ? |
15:20.07 | tzafrir_laptop | Ocean_big, you can also run 'sip debug' on asterisk. This dumps every sip packet asterisk recieves, and is very verbose . Don't try to understand them. But you'll easily see when sip packets arrive |
15:20.13 | Ocean_big | tzafrir_laptop, now, i 'm sure that the call attempt the astersik machine, because i had plugged a non-ip phone behind the card. |
15:20.21 | syzygyBSD | Uther_P: the only word in english that has 3 y's and no other vowels |
15:20.35 | Ocean_big | the Soft PHONE is OK, i can call with it |
15:20.45 | syzygyBSD | rikstah: it is just as reliable as witha cell phone I think |
15:20.55 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
15:21.04 | rikstah | syzygyBSD did you use the app_bluetooth thing? |
15:21.09 | tzafrir_laptop | I don't understand exactly what you sa. What is working and what isn't? |
15:21.23 | Ocean_big | the network is very basic only one PC and the asterisk machine |
15:21.27 | *** join/#asterisk snitt (i=snitt@snitt.info) |
15:21.34 | Ocean_big | (and a routeur) |
15:21.38 | Uther_P | syzygyBSD: ahh... a solar or lunar eclipse, for examples, are syzygy |
15:21.41 | Uther_P | 's |
15:21.43 | syzygyBSD | rikstah: no, I just had it setup like a bluetooth headset in windows using IBM's drivers |
15:21.53 | syzygyBSD | Uther_P: also any 3 objects in alignment |
15:21.56 | rikstah | ahh sorry i forgot to mention i'm on LINUX :P |
15:22.11 | syzygyBSD | rikstah: ya, I haven't done it on linux |
15:22.44 | Ocean_big | tzafrir_laptop, outgoing call are OK, and Incoming call are not OK :(.. |
15:22.50 | *** join/#asterisk fugitivo (n=ajf@201.255.102.19) |
15:22.51 | fabsoft | i live in italy, a have got a telecom isdn TN mode network card, and also a hfc-pci chipset card, what channel i need to use hfc-pci for send/receive call from pstn network ? |
15:22.55 | Uther_P | heh... like the old saying about needing to "get your ducks in a syzygy" |
15:22.56 | fugitivo | hello |
15:23.14 | Uther_P | ...or a game of 3 card monty :P |
15:23.30 | Uther_P | interesting word... it is forever burned into my vocabulary |
15:23.31 | tzafrir_laptop | Ocean_big, outgoing calls and incoming calls to where? to that certain IP phone? |
15:23.56 | darwin35 | back in while calls are taking off |
15:24.08 | [_gordo_] | how do i dial a flash hook from my softphone |
15:24.09 | darwin35 | billibg/support/wheres the boss |
15:24.16 | [_gordo_] | i can't make it work |
15:24.32 | *** part/#asterisk darwin35 (n=darwin35@208.139.193.178) |
15:26.39 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235) |
15:26.41 | MuppetMaster | Hello |
15:27.16 | MuppetMaster | Anyone know how to invoke the 'On-Demand' recording that may be configured for an extension within AMP? Per this screen on the demo: http://demo.coalescentsystems.ca/config.php?display=3 |
15:27.26 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
15:27.43 | IzNoGooD | Unhandled Message: prim 281 len -6 from addr 51400201, dinfo 0 on port: 1 |
15:27.50 | IzNoGooD | That's what I get on an misdn channel |
15:27.59 | IzNoGooD | sound isn't working |
15:28.09 | IzNoGooD | any idea's? |
15:28.23 | Ocean_big | tzafrir_laptop, outgoing and incoming from non-ip phone |
15:28.39 | tzafrir_laptop | Anybody for some asterisk 1.2 (beta1) debs? |
15:28.50 | MuppetMaster | tzafrir_laptop: ? |
15:28.53 | Ocean_big | Ip phone are OK |
15:29.04 | tzafrir_laptop | deb http://rapid.dotsrc.org/ experimental/ |
15:29.33 | *** join/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk) |
15:29.36 | lehel | i'll try it tzafrir_laptop |
15:29.44 | IzNoGooD | me 2 |
15:31.11 | tzafrir_laptop | I'll commit quite similar changes to the pkg-voip svn soon |
15:31.27 | tzafrir_laptop | I just hope everybody's happy with my version |
15:32.00 | Ocean_big | tzafrir_laptop, the CLI doesn't show anything when i call the asterisk machine with a non-ip Phone |
15:32.00 | ret28 | Does Asterisk's implementation check against From: forgery ? |
15:32.54 | Uther_P | how would a forged sip packet do any good? other than maybe annoying the crap out of someone |
15:32.59 | gdh | tzafrir: cheers :) |
15:33.42 | ret28 | Uther_P: Pretending to be from a SIP domain that it had no right to be ... or it's possible I'm misunderstanding SIP addressing |
15:34.01 | ret28 | (that is, calling somebody pretending to be somebody else) |
15:34.26 | Uther_P | but doesn't it use the uri to know how to send reponses back to the originator? |
15:34.44 | ret28 | I thought it just sent them back along the same connection |
15:34.50 | syle | anyone using SER? |
15:34.52 | mutilator | [11:33:59] <Pyro> Dark-Fx: walked up to the sdc..it went like this "yeah my arm might be broken", "we can fit you in after the weekend" |
15:34.52 | mutilator | [11:34:03] <Pyro> so i drove to hancock |
15:34.55 | _DAW | Hello |
15:34.57 | ret28 | (especiall true for a TCP SIP connection) |
15:35.12 | Uther_P | but asterisk doesn't use tcp sip |
15:35.20 | ret28 | Mmm, but it'll probably do the same for UDP |
15:35.24 | _DAW | Has anyone here used the TOUCH_MONITOR variable with one touch record? |
15:35.31 | ret28 | That is, sending it back to the originating IP address |
15:35.36 | Uther_P | I would think that the protocols would be the same regardless of its transport proto |
15:35.59 | ret28 | I suppose I ought to test this internally (say by forging a voip provider), and seeing what it tries to send to the outside world |
15:36.16 | tzanger | coppice: listening to txfax and the fax machine (xerox) talk |
15:36.49 | Uther_P | ret28: if my memory serves me, the uri is used to responses... if you hit a proxy first, it encapsulates your sip message in its own, then reinvites (if applicable) later to take itself out of the middle |
15:36.53 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.171.235) |
15:36.54 | tzanger | coppice: I can hear them try to negotiate, the fax warble is replaced quickly by normal modem sounds (noisy hiss) but then it stops and goes back to warble a few more times, hissing again, then just dropping out |
15:38.01 | Uther_P | ret28: I don't think sip cares what the udp packet header says the originating ip address was |
15:38.44 | Uther_P | ret28: but... I imagine someone could do something similar by faking themselves as a proxy |
15:38.46 | *** join/#asterisk heka (n=heka@82.114.68.123) |
15:39.08 | ret28 | Damn, this client really doesn't like me trying to make it spoof :> |
15:39.10 | Uther_P | and encapsulating its own message |
15:39.34 | Uther_P | ret28: so formulate your own messages |
15:39.51 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
15:39.59 | ret28 | At the moment, I'm thinking that'll be more effort than beating this client about, but I may turn out to be wrong :) |
15:40.28 | ret28 | I was under the impression that proxying was done by keeping the SIP signalling going via the proxy (as it'll be low bandwidth), but redirecting the RTP |
15:40.41 | ret28 | I can't think of a particularly good reason to want to offload the SIP signalling anyway |
15:41.29 | Uther_P | ret28: the media transport is established by means communicated inside the sip messages |
15:41.48 | Uther_P | the sip can work just fine without the media working at all |
15:41.49 | InfraRed | http://docs.info.apple.com/article.html?artnum=86816 |
15:42.10 | Uther_P | haha |
15:42.12 | Uther_P | who bothered |
15:42.42 | *** join/#asterisk TK9 (n=Administ@p54B28C56.dip0.t-ipconnect.de) |
15:42.56 | ret28 | "Walk to your destination by putting one foot in front of the other. /Do not walk into walls with your iMacG5/." |
15:43.43 | Uther_P | refrain from jumping, hoping or skipping while carrying your g5 as you are likely to trip on your own stupidity |
15:43.51 | InfraRed | it took 2 months of fine tuning the document too |
15:44.02 | ret28 | Is it bad politics to say something offensive about Mac users here? :) |
15:44.03 | Uther_P | didn't know quite how to word it? |
15:44.21 | Uther_P | its only safe politics to degrade microsloth |
15:44.43 | ret28 | Well, many of my best friends are Mac users ... |
15:44.50 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
15:46.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
15:46.38 | danzig | Hello * Gurus :-) Question: I have a lot of Grandstream GXP2000 hardware phones belonging to students in a dorm, each student has a SIP account (with Musimi.dk, think Vonage or something) which is a SIP peer, when the phone tries to dial out Asterisk does a Dial SIP/the-persons-trunk . Asterisk says bridging call natively, it works fine, but I give very bad error messages. If the trunk says... |
15:46.40 | danzig | ...408, the phone inside gets told 503. If teh trunk says 503, the phone inside gets 503. If the trunk does not exist, the phoen inside gets told 503. Is there a good/easy way of letting more informative error messages through? |
15:46.40 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
15:47.50 | Uther_P | the trunk says 408 to your asterisk box, and it reports 503 to the phone? |
15:47.53 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
15:48.34 | ret28 | Hrrmm, this isn't good |
15:50.00 | ret28 | I've successfully spoofed a From, and asterisk doesn't even try to lookup or contact the real server for that domain |
15:50.05 | danzig | Yep, I think so. I am doing |
15:50.06 | ret28 | (as far as tcpdump can tell) |
15:50.07 | danzig | exten => _XXXXX.,5,Dial(SIP/${account}/${EXTEN},120) |
15:50.31 | ret28 | If the SIP module gave me another variables, I could execute a System call to a check in the dialplan |
15:50.34 | ret28 | But nooo ... |
15:50.47 | *** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
15:51.11 | ret28 | *more variables |
15:52.22 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
15:52.43 | Uther_P | danzig: do the grandstreams even have a different action to take when receiving a 408? I wouldn't think so... the trunk says to forward somewhere else to asterisk, asterisk has nowhere else to forward, so it gives a resource unavailable to the phone... seems like what its doing is reasonable to me |
15:52.44 | ret28 | Is this a Major Security Hole ? |
15:52.56 | af_ | mhh, there is a webphone for asterisk, with sources available? |
15:53.12 | Uther_P | af_: several |
15:53.21 | Uther_P | you mean a soft phone? |
15:53.36 | Uther_P | ret28: did the sip responses come back to you? |
15:53.41 | af_ | no a stuff Icould phone without install nothing |
15:53.47 | af_ | not a softphone |
15:53.47 | Uther_P | hah, what? |
15:54.02 | af_ | I push a button on a web page, that's all |
15:54.13 | ret28 | Uther_P: Yeah, it all validated as a call, finished dialling, went to my voicemail |
15:54.31 | Uther_P | hrm... I've never seen one... but I bet you could write a front end for a softphone |
15:54.52 | Uther_P | ret28: what did the caller id say for who the caller was from? |
15:55.18 | Uther_P | ret28: if you wanna see the sip messages as they come in, then turn sip debug messages on for your clients ip |
15:55.33 | Uther_P | that'll tell you if your "spoof" even worked |
15:55.49 | tzafrir_laptop | ret28, how have you authenticated? |
15:56.10 | danzig | Uther: yes, it is reasonable. The GXP displays the error number on the display. The problem is my users get confused - they do not understand the diffenrence between congestion as in you have no trunk and congestion as in party B is on the phone. Is there anything I can do, easier than a big block of 'jump here on chan-unavailable', 'jump there on something-else' etc.? |
15:56.39 | Uther_P | danzig: yea... you can have your dialplan report congestion if the dial fails |
15:56.44 | ret28 | My dialplan uses SetCallerID(${CALLERID}@${SIPFROMDOMAIN}) , which gives me a callerid of "totallybogususer1234@voiptalk.org" |
15:57.03 | ret28 | callerchan=SIP/voiptalk.org-b5a00470 (from the voicemail txt file) |
15:57.08 | ret28 | So it just believed what my client fed it |
15:57.29 | ret28 | tzafrir_laptop: It hasn't, I'm supposed to be simulating a totally foreign call by a 3rd party trying to talk to my SIP address |
15:57.47 | ret28 | (I've not got a client connected and registed with my SIP at the moment, so it's dropping to voicemail) |
15:57.52 | tzafrir_laptop | ret28, AFAIK caller-ID is something that can be spoofed. You should only trust it if you have a good reason |
15:58.19 | ret28 | tzafrir_laptop: But the actual SIP From address is being spoofed successfully as well |
15:58.30 | ret28 | (note that I set the callerid for incoming SIP based on the From address) |
15:58.35 | Uther_P | yea, heh... caller id is as irrelivant as the From field in an email... you can make it look like it's comming from whoever you want |
15:58.41 | tzafrir_laptop | ret28, you authenticate the sip user, right? |
15:58.58 | ret28 | tzafrir_laptop: Erm, no. That's the point, it's for people who aren't local users, total 3rd parties trying to call me. |
15:59.02 | tzafrir_laptop | ret28, give a caller-id based on that sip user |
15:59.15 | danzig | Uther: But I always get congestion/503, no matter what goes wrong... So I do not really need more reporting of congestion/503. What I really need is congestion only if the called party is engaged, and other errors (maybe recorded messages) on other errors. Anyone already done this? |
15:59.36 | tzafrir_laptop | ret28, well, you shouldn't trust their caller-IDs. If you do: don't complain |
15:59.37 | ret28 | And did the SIP designers really make the mistake that's long been realised with SMTP? ;) |
16:00.02 | Uther_P | danzig: called party is engaged? you mean on the phone? if you mean the called party is using the phone, that messages is 486, not 503 |
16:00.04 | ret28 | tzafrir_laptop: How do I actually know that somebody calling from @voiptalk.org is a VoIPTalk user then? |
16:00.07 | ret28 | (for example) |
16:00.09 | tzafrir_laptop | ret28, there is no simple way to verify that address on-line |
16:00.12 | ret28 | This isn't a closed system |
16:00.21 | Uther_P | ret28: its not a mistake, per se... its pointless |
16:00.24 | ret28 | It's "I have a SIP address, call me from whoever your provider is" |
16:00.37 | tzafrir_laptop | ret28, otherwise I can easily concive a DOS attack on voiptalk.org . |
16:00.51 | ret28 | tzafrir_laptop: With callbacks? |
16:00.53 | tzafrir_laptop | Not to mention a dictionary attack to discover valid names |
16:00.56 | ret28 | *verification callbacks |
16:00.56 | Uther_P | ret28: what if you are bouncing of a proxy, and a firewall? then you couldn't get back to the originator to verify it |
16:01.01 | terrapen | I prefer an OS/2 attack. |
16:01.04 | Uther_P | you would jsut have to go back the way you came |
16:01.24 | Uther_P | heh, the old smurf method |
16:01.34 | ret28 | OK, what about by DNS? |
16:01.46 | tzafrir_laptop | a DOS attack leaves you with the ability to run DOS alone. That is why nobody would do a OS/2 attack |
16:01.52 | ret28 | Do a lookup for the A, and both SRVs, compare against From host |
16:01.57 | tzafrir_laptop | ret28, spofable just the same |
16:02.03 | ret28 | I'm not suggesting verifying the originating user, just the domain |
16:02.18 | Uther_P | ret28: that call could be bouncing off many different places... proxies, firewalls? your lookup wouldn't work the way you think |
16:02.29 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
16:02.37 | Uther_P | your proxy could be (and often is) a different domain the the originator |
16:02.39 | danzig | Uther: Hmmm.... Am getting 503 on _everything_ Must go check log files... Maybe my provider is causing the problem? I thought it was my fault, with the bridging, that everything ended up as a 503 inside... |
16:03.04 | ret28 | So, back to the original question ... is there any way to have trustworthy sources in SIP? |
16:03.05 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfjs5.dialup.mindspring.com) |
16:03.34 | ret28 | Uther_P: I was under the impression that users typically proxied through their VoIP provider |
16:04.25 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
16:04.35 | Uther_P | danzig: if your truck reported 408 (timeout).. then your server would try other providers if it had any more... when it exhausts them, then it reports to the caller that 503, the service is unavailable... the service the user is trying to call, which would include any and all paths to that service the sip server has |
16:04.44 | *** join/#asterisk terracecomm (n=terracec@164.216-123-230-0.interbaun.com) |
16:05.21 | Uther_P | the phone would only get a 408 if it was going to the trunk directly... as the trunk would expect that it may not be the only method out |
16:05.58 | Uther_P | danzig: some people have external proxies that bounce their calls through firewalls |
16:06.09 | Uther_P | err,, that was directed to ret28 |
16:06.25 | terracecomm | did anyone find the problem with building 1-2-0-beta1 on centos 4 |
16:06.35 | ret28 | Howcome their voip provider isn't doing that? |
16:06.37 | terracecomm | seem to be missing a dependency |
16:07.00 | redder86 | coppice: hi |
16:07.06 | ret28 | (where 'doing that' == 'providing a proxy' btw) |
16:07.44 | Uther_P | because sometimes both sides are behind nats and/or firewalls... and need someone outside to bounce off of first, to establish the natd entry |
16:07.47 | danzig | Uther: Thanks, now I am beginning to understand. I apologize for my stupidity. There is always only one path out - what do I write in extensions.conf to get Asterisk to give the internal phone the error the trunk reported, _not_ go on to the 'next stage' - which is nothing/timeout/congestion/hangup in most cases? |
16:08.33 | Uther_P | danzig: not sure...hrm. I don't know that you can through the dialplan... probably can with an agi though |
16:08.48 | Uther_P | why would the end users need to know *why |
16:08.52 | Uther_P | * they can't get out |
16:08.56 | ret28 | Uther_P: Confused ... is the implication that one's VoIP provider wouldn't have public IPs? |
16:09.14 | Uther_P | isn't getting the beeping telling them they can't get out enough? why would they need to know why? |
16:09.24 | Uther_P | ret28: welcome to MY world, ahah |
16:09.31 | ret28 | I think my 'solution' would be to just add 'Suspicious' to the callerid where lookups don't match :) |
16:09.40 | *** join/#asterisk vp7 (n=vp7@193.27.41.43) |
16:09.41 | ret28 | Uther_P: Running your own SIP domain/server behind NAT? :) |
16:09.47 | Uther_P | ret28: my VoIP provider is also my T1 provider, and they used RFC1918 addresses for their SIP and media servers |
16:09.54 | Uther_P | it was a major pain in my ass |
16:10.14 | ret28 | Don't their SRV records point to the same place as they NAT out of then? :/ |
16:10.21 | Uther_P | ret28: yea, its behind my firewall... being translated to a public ip address from there |
16:10.43 | Uther_P | ret28: they DON'T nat out... heh... the packets come to ME from internal addresses |
16:10.46 | mmlj4 | suggestions on where to buy digium hardware? |
16:10.49 | vp7 | Hello! Could anyone tell me, if it's possible to run asterisk with H.323 support on FreeBSD? I'd like to run Asterisk as office PBX and our provider can give us only H.323 |
16:10.56 | Uther_P | they can do that because they are also the T1 provider |
16:11.08 | mmlj4 | vp7: yes, of course |
16:11.24 | ret28 | Uther_P: :/ |
16:11.34 | ret28 | My sympathies |
16:11.58 | ret28 | It's still utterly mindboggling that the SIP drafters didn't think "Hmm, e-mail From: forgery ... hey, we ought not to fall into that!" |
16:12.06 | Uther_P | ret28: so it's technically a local network to them... but it was a bitch, because I had to allow the rfc1918 addresses inbound from the external interfaces... then translate them... THEN make sure that my firewall didn't try to translate or drop the packets on the way back out |
16:12.57 | danzig | Uther: A valid point... Main reason is that we have people getting frustrated because they cannot tell the difference between the 3 most common cases: |
16:12.58 | danzig | - the remote party is actually engaged |
16:12.59 | Uther_P | finally I just setup the cisco router to act as a nat FROM the T1 to the internal network as the public ip address of mydefault gateway |
16:13.00 | danzig | - the person has not even set a trunk up for themselves and has absoloutly no chance of ever getting out (I could handle this with a jump on chan-unavailable) |
16:13.01 | danzig | - our service provider is foobarred |
16:13.03 | danzig | Therefore they keep on trying, thingking that the remote party is home, talking on the phone, so now is a good time to call. |
16:13.40 | Uther_P | danzig: yea, for an admin's purpose, I can understand... but unless your users are admins... it does'nt really make any difference to them whether they know... since they can't do anything about it anyway |
16:13.56 | ret28 | I think I'll just have to live with a small number of users coming up with "(SUSS)" in the callerid on my phone :D |
16:14.15 | Uther_P | danzig: naa... if the other party was buzy the phone would get back a 486 message, not a 503 message |
16:14.46 | Uther_P | danzig: 503 == service unavailable.... 486 == Busy Here |
16:14.51 | *** join/#asterisk JerJer[mobile] (n=jj@68.123.154.34) |
16:14.53 | ret28 | I've seen people suggesting an SPF equivalent, which makes more sense than for e-mail (as there's no forwarding issues, because SIP forwarding is done via HTTP style redirects) |
16:15.37 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
16:17.29 | *** join/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
16:17.40 | *** part/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
16:17.43 | Uther_P | ret28: I think as far as reinvites from one nat to another... in the sip messages, the clients should specify to each other their external ip's (which they do), and have both send packets out at the same time.... currently one of them tries to start the media pathway to the other which is listening.... but if both of them were to just send a few packets out before the pathway is established, then both sides would have already poked a hole throught their |
16:18.19 | ret28 | Oh, STUN |
16:18.37 | Uther_P | that way the initial packets don't get dropped at the nat with no entry to forward them |
16:19.01 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
16:19.31 | danzig | Uther: Thanks. Maybe solution is that last step in dialplan should not be congestion, but a recorded mesasge. The important thing is that they can easily (by the sound) tell the difference between busy and service unavailable... |
16:19.31 | ret28 | I'm not so concerned about the actual media IP origination, just the SIP messages being authentic (which can go through a provider proxy without being too heavyweight) |
16:19.39 | lvp | hello |
16:20.20 | Uther_P | danzig: but their phone's display should say "486" on it if the call is busy.... besides... isn't it a faster busy tone for a 503 than for a 486? |
16:20.28 | danzig | Or maybe I should fiddle with indications.conf? |
16:20.37 | lvp | any idea if any hard SIP phone has support for using LDAP queries to implement a central phone book ? |
16:21.01 | lvp | (or any other way which is scalable and has search features) |
16:21.14 | danzig | Hmmm... am having trouble testing, Do not have any relaiably busy numbers to call :-( May have to make one... |
16:21.45 | Uther_P | lvp: you could probably create an agi on asterisk and map a prefix of an extension to do the lookup |
16:21.57 | danzig | Hmmm... too close for normal users, I fear. ;-) |
16:22.08 | Uther_P | danzig: call the phone number that you are dialing out of, heh |
16:22.57 | lvp | U: hmm.. |
16:23.15 | danzig | No, there are 4 lines available. Maybe I have set it up silly, but one can perfectly well call onesself - the line 2 lamp just starts flashing ;-) |
16:24.07 | tzanger | coppice: what TIFF format should I have my file to send in? |
16:26.05 | ret28 | On another less controverisal note ... if I have SRV records for my SIP domain, but no A records, how many users would that mess about? |
16:26.25 | ret28 | (as in, any common clients/PBXes lacking SRV support these days?) |
16:27.05 | Uther_P | ... clients without srv support wouldn't be able to use it |
16:27.24 | ret28 | But are there many of those? |
16:27.36 | Uther_P | *shrug* not to my knowlege |
16:27.43 | ret28 | Hmm, ok |
16:27.47 | Uther_P | most VoIP providers support srv these dayz |
16:27.55 | Uther_P | errr VoIP devices rather |
16:28.21 | *** join/#asterisk bgtroll (n=bgtroll@pirus.securax.be) |
16:28.32 | lvp | ret: quite a lot |
16:29.04 | lvp | ret: at least soft clients |
16:30.00 | ret28 | Is it common for users to make calls straight from their phones without any interaction with or proxying through the provider? |
16:30.37 | Uther_P | ret28: you mean ip to ip calls? |
16:30.43 | tzanger | http://www.brehanbrand.com/images/lj/cameltoads.jpg |
16:30.54 | ret28 | Uther_P: Mmm |
16:31.32 | Uther_P | tzanger: bwahaha |
16:31.42 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
16:32.03 | tzanger | I hope that busybody dropped dead from embarassment |
16:32.10 | Uther_P | hehe, hell yea |
16:32.17 | Uther_P | thats what he/she gets for snooping |
16:32.30 | ret28 | Hahaha, that's great |
16:32.36 | ret28 | And yes, justice! |
16:34.18 | Uther_P | "please let me know what camel toads are and how I might be able to tell if he is smoking, taking, or licking them"... well... he'd probably only be licking, pounding or rubbing them.... a much different experiance to 'smoking the toad", but just as gratifying |
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16:38.19 | ManxPower | Ugh. FedEx tracking is down |
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16:39.25 | sigterm | doh |
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16:40.08 | *** part/#asterisk TK9 (n=Administ@p54B28C56.dip0.t-ipconnect.de) |
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16:52.46 | chidex | does asterisk 1.0.9 have custom macros just like asterisk@home? |
16:53.07 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
16:53.29 | ManxPower | chidex, Asterisk@Home IS 1.0.9 |
16:54.45 | chidex | I thought asterisk@home had a little bit more done to it to help the novice |
16:56.18 | ManxPower | chidex, Asterisk@Home does a lot of stuff for the user, but the Asterisk is the same as the one you can FVT or CVS |
16:56.41 | tzafrir_laptop | chidex, asterisk@home adds custom macros on top of asterisk. You can put yours instead. |
16:57.22 | tzafrir_laptop | asterisk@home (actually AMPortal) has a very complex dialplan. A default asterisk installation is something that is much easier to grasp |
16:57.25 | SpaceBass | but the new version appears to have a bug with 2 or more zap channels in different contexts.... |
16:57.33 | *** join/#asterisk taec_ (n=phil@eventhorizon.hosting365.ie) |
16:57.45 | taec_ | Hello, I've got a TE410P card from Digium, Asterisk and a PRI ISDN line. The Line is plugged into the first interface on the card, but I'm at a loss as to how to proceed any further. Google hasn't been kind! If anyone could provide any pointers I'd really appreciate it. |
16:57.54 | SpaceBass | all in all, it can let someone get up and running quickly |
16:58.08 | ManxPower | ~docs |
16:58.09 | jbot | it has been said that docs is 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 |
16:59.19 | tzafrir_laptop | SpaceBass, the problem is how to customize it |
16:59.24 | *** join/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com) |
16:59.28 | taec_ | I've read a _lot_ of documentation, but I'm still unsure as to how to proceed with the ISDN line. Should I be setting it up with ISDN4Linux, or proceeding some other way? A small pointer in the right direction would be great. |
17:00.02 | tzafrir_laptop | taec_, not an expert, but what card do you have, and what kernel version? |
17:00.41 | *** join/#asterisk coppice (n=chatzill@190.196.17.210.dyn.pacific.net.hk) |
17:01.02 | taec_ | TE410P and 2.6 kernel |
17:01.03 | tzafrir_laptop | taec_, have you configured zaptel.conf and zapata.conf? for the PRI card? |
17:01.30 | tzafrir_laptop | ISDN4Linux is for BRI ISDN |
17:01.34 | ManxPower | taec_, ISDN4Linux is for ISDN BRI |
17:03.09 | tzafrir_laptop | taec_, it is supported by zaptel (one of the t1 drivers). I wonder if my script would have configured it slightly correctly. |
17:04.32 | taec_ | ok, thanks :) ... Yes, although I'm not sure if the values are correct. |
17:05.04 | tzafrir_laptop | taec_, zap show channels |
17:05.18 | tzafrir_laptop | if it shows nothing, the cards are not configured |
17:05.40 | tzafrir_laptop | taec_, is the module loaded? anything on /proc/zaptel/* ? |
17:06.34 | Ariel_ | ~seen shido6 |
17:06.43 | jbot | shido6 is currently on #asterisk (1h 58m 34s) |
17:07.03 | tzanger | coppice: any idea why I am not getting any debug output at all with TxFax(/path/to/file.tiff,caller,debug) ?? |
17:07.04 | Ariel_ | shido6, you around? need to talk with you? |
17:09.28 | lehel | tzanger: start asterisk with -cvvvddd |
17:10.12 | coppice | tzanger: no idea. which version are you using? |
17:10.45 | ManxPower | tzanger, Start Asterisk as "asterisk -cvvvddd" so you get STDERR to your console. |
17:10.47 | taec_ | /proc/zaptel has 1 2 3 and 4 in it ... presuming they represent the 4 interfaces on the card |
17:10.58 | taec_ | show zap channels gives a pseudo channel |
17:11.02 | ManxPower | I don't believe you'll see STDERR/STDOUT if you don't do that |
17:11.34 | ManxPower | tzanger, I have to do the same thing when I debug AGI scripts |
17:11.35 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
17:12.36 | *** join/#asterisk igori (n=Miranda@194.84.91.2) |
17:13.37 | taec_ | zap show status gives 4 alarms ... on the channels. |
17:14.29 | tzanger | coppice: just trying to get it to work here... I converted the image to exactly 1728 pixels wide and it came across, although "stretched" a little beyonda page |
17:14.32 | tzanger | just playing a bit now |
17:14.49 | ManxPower | taec_, Then there is no active lines plugged into the card |
17:15.31 | taec_ | The line that was plugged into our office PBX has been plugged out and brought straight into that card.. |
17:15.41 | tzanger | yeah it seems to be a page width thing |
17:15.44 | ManxPower | taec_, Then you have a wireing provlem |
17:16.03 | ManxPower | taec_, RED Alarm means "I don't see a line" |
17:16.22 | ManxPower | taec_, maybe you need a crossover T-1 cable rather than a straight thru T-1 cable |
17:18.29 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
17:19.38 | *** part/#asterisk bgtroll (n=bgtroll@pirus.securax.be) |
17:20.12 | tzanger | ok a normal fax is 1728 pixels wide, how many pixels "long" is a regular 8.5x11" sheet? It seems to be stretching it beyond a page length |
17:20.28 | FuriousGeorge | anyevery installed a doorphone? |
17:20.39 | *** part/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk) |
17:20.40 | *** join/#asterisk fugitivo (n=ajf@201.255.102.19) |
17:21.19 | FuriousGeorge | im looking at those analog hookups. it appears all i need is an fxo and the doorphpne |
17:21.36 | tzanger | FXS, no? are you hooking the doorphone to *? |
17:22.14 | ManxPower | ~fxofxs |
17:22.15 | jbot | from memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
17:22.24 | tzanger | ManxPower: yes I know that |
17:22.33 | ManxPower | tzanger, that was for FuriousGeorge |
17:22.41 | tzanger | heh |
17:22.47 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
17:23.10 | distortion | is there a way to send the "sip debug" output to a file? |
17:23.13 | FuriousGeorge | yeah, but i heard that a doorphone wants an fxo |
17:23.22 | queuetue | Does anyone use voipjet? Are they currently down? |
17:23.24 | ManxPower | distortion, see /etc/asterisk/logger.conf |
17:23.51 | FuriousGeorge | it doesnt want to dial anything, just ring the server |
17:24.18 | ManxPower | FuriousGeorge, It would be unusual for a doorphone to not dial |
17:24.57 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
17:26.16 | distortion | ManxPower: thank you. |
17:26.32 | FuriousGeorge | ManxPower: http://lists.digium.com/pipermail/asterisk-biz/2005-July/007071.html |
17:27.06 | FuriousGeorge | my understanding is, if they hit the "call:" button the fxo handles it like an incoming call, and thats why you use one of those |
17:28.18 | *** join/#asterisk Tili (i=Tili@202-133-67-168-dialup.sat.net.pk) |
17:28.26 | queuetue | Does anyone use voipjet? Are they currently down? |
17:28.38 | FuriousGeorge | it, the doorphone, doesnt need a dialtone or to ring. the communication always goes the same way |
17:28.46 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
17:29.02 | *** join/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
17:29.57 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
17:30.23 | taec_ | ManxPower: the line almost definitely works. It's a straight through E-1 cable which was plugged originally into our PBX and now into our TE410P card. |
17:30.34 | taec_ | ManxPower: I even went to the trouble of cable testing it there, just in case : |
17:31.17 | *** join/#asterisk morale (n=russell@secure.deadbolt.ca) |
17:31.21 | morale | has anyone had luck getting asterisk going with primus or vonage in canada? |
17:32.56 | generalhan | does anyone know how to change the background on the FOP ? which file has to be edited ? |
17:33.09 | taec_ | ManxPower: if the zaptel module is loaded and the cable is ok, would there be anything else that could be the problem? |
17:33.13 | *** join/#asterisk toddf (n=toddf@wsip-70-182-74-104.ok.ok.cox.net) |
17:34.07 | taec_ | Would there be a setting on the card I may have to configure for it to support E-1 lines? |
17:34.09 | tzanger | coppice: txfax is working as expected. I had to adjust my scanned tiff image to account for the non-square pixels |
17:34.29 | queuetue | Does anyone use voipjet? Are they currently down? |
17:35.39 | tzanger | I don't use them |
17:36.44 | jontow | woohoo |
17:36.47 | jontow | 4 SNOM 320's arrived :) |
17:37.24 | harryvv | how much per phone? |
17:37.55 | harryvv | morale, hello? |
17:38.01 | jontow | hmm, i forget.. $210? |
17:39.30 | morale | harryvv, eh? |
17:39.37 | coppice | tzanger: as expected? is that good or bad? :-) |
17:41.45 | queuetue | Who else do you use for outgoing? (VOIPjet appears to be off the air...) |
17:42.29 | tzanger | coppice: so far good, now I need to see if I can get imagemagick ot create a 2-page tiff file |
17:44.06 | morale | it looks like this inphonex.com company is the best deal.. they provide the SIP information |
17:44.20 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
17:44.26 | morale | Inphonex - SIP Open - $ - prepaid available - Canadian DIDs available $8/month .. 8$ for a basic number. |
17:44.41 | Seyr | Is there any way to set the number of rings before * answers? |
17:46.33 | tzanger | Seyr: just wait() |
17:47.23 | *** join/#asterisk scfrec (i=scfrec@scfrec.compic.ee) |
17:47.35 | Seyr | ok, thanks :-) |
17:48.22 | scfrec | trying to call from voip to PSTN via asterisk. got echo, because analog phone on second site. reading manual don't help. can anyone point to right direction? |
17:48.25 | ManxPower | taec_, there are jumpers on the board. |
17:48.25 | *** join/#asterisk l1nux (n=moi@lns-bzn-4-82-250-119-89.adsl.proxad.net) |
17:48.30 | l1nux | hi |
17:48.38 | ManxPower | However, it should set the correct mode when you use a EU span= line. |
17:49.05 | ManxPower | scfrec, Echo has to be removed at the VOIP/PSTN interface. |
17:49.43 | *** join/#asterisk brainlight (n=zeldaxxx@dsl-du-83-173-249-189.cybernet.ch) |
17:49.53 | scfrec | VOIP interface - MOSA 3704B |
17:49.57 | scfrec | at analog - can't |
17:50.02 | brainlight | heya, anybody here has an eicon diva BRI-2M working on a 2.6 kernel with asterisk? |
17:50.07 | ManxPower | scfrec, then that is the device that has to remove the echo |
17:50.10 | scfrec | if call go to GSM phone - no echo at all. only on analog |
17:50.17 | generalhan | does anyone know how to change the background on the FOP ? which file has to be edited ? |
17:50.31 | ManxPower | scfrec, correct. the cell network has to cancel out echo in the PSTN/GSM interface. |
17:50.35 | tzafrir_laptop | generalhan, background.jpg , IIRC |
17:50.50 | generalhan | i dont even see where that file is stored, or called apon |
17:50.56 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
17:50.57 | *** join/#asterisk igori (n=Miranda@194.84.91.2) |
17:51.03 | l1nux | any one get call (over nat) success with voipuser ? |
17:51.06 | generalhan | i made a background.jpg but i dont see where the old one was to write over it |
17:51.15 | tzafrir_laptop | at the same directory of thw .swf file. Check the logs of apache |
17:51.28 | scfrec | Echo Cancellation G.165/G.168 16ms - not enough ;) |
17:51.48 | ManxPower | generalhan, That issue is usually the browser caching the old image |
17:52.13 | ManxPower | scfrec, It should be plenty, since the "16ms" is the latency of the PSTN part of the call, not the VoIP part of the call. |
17:52.24 | generalhan | well if i go to the dir with the .swf file, there was never a background.jpg file there to bgin with |
17:52.33 | l1nux | Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) |
17:52.34 | l1nux | <PROTECTED> |
17:52.35 | generalhan | but it worked anyway ! |
17:52.42 | generalhan | good call guys thanks a lot ! |
17:52.58 | l1nux | what the codec need to use ? |
17:52.58 | scfrec | ManxPower: problem i hear myself becouse phone on other side transmit my voice back |
17:53.00 | ManxPower | l1nux, Um, Asterisk does not support G273.1 |
17:53.14 | l1nux | ohh.. |
17:53.24 | ManxPower | scfrec, All analog phones do that, it's just that until VoIP came along you could not hear the echo because it happened so fast. |
17:53.48 | harryvv | Here comes the voip compitition to vancouver! http://www.canada.com/vancouver/vancouversun/news/business/story.html?id=f42b3c54-c400-4bb6-9305-40c849bd85b3 |
17:54.05 | harryvv | shaw cable is now selling voip across its cable. |
17:54.08 | ManxPower | scfrec, this is covered over and over and over again in the mailing list archives and there is much information on the Wiki |
17:54.15 | ManxPower | ~mailinglist |
17:54.17 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
17:54.17 | ManxPower | ~docs |
17:54.18 | jbot | somebody said docs was 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 |
17:54.19 | scfrec | ManxPower: echo return in 0.2 seconds. |
17:54.42 | ManxPower | scfrec, you can believe me ot not. I'm still right. |
17:54.47 | scfrec | ManxPower: when i call from software phone - same problem - i hear myself |
17:55.03 | scfrec | so sound back to line on second side |
17:55.14 | ManxPower | scfrec, that is expected. |
17:55.25 | ManxPower | anywhere there is analog there will be echo. |
17:55.28 | l1nux | ManxPower, what the codec need to use with voipuser.org ? |
17:55.40 | ManxPower | l1nux, I don't know. What codecs do they support? |
17:55.45 | scfrec | ManxPower: solution? |
17:56.21 | ManxPower | scfrec, the solution is for your VoIP/PSTN gateway to cancel out the echo. |
17:56.40 | ManxPower | Since you are using a 3rd party gateway, there is NOTHING we can do to help. |
17:57.09 | scfrec | ManxPower: it's possible todo on asterisk? |
17:58.05 | ManxPower | scardinal, only if asterisk handles the PSTN/VoIP conversion. |
17:59.31 | scfrec | ManxPower: schema is nexT: phone-gateway-asterisk(my)-asterisk(provider)-dont know-phone |
18:00.20 | Ariel_ | JerJer[mobile], are you around? I have a biz question for you... 800 number service? |
18:00.29 | *** part/#asterisk Ldr (n=Lode@194.206.157.226) |
18:01.20 | InfraRed | 800 numbers |
18:01.24 | InfraRed | thanks for reminding me! |
18:01.53 | tzanger | fuck |
18:01.55 | Ariel_ | InfraRed, that a way to do it. |
18:02.11 | InfraRed | Ariel_: i have a more cunning plan :) |
18:02.16 | tzanger | I can't seem to get the return code from TxFax because the other side hangup is too fast |
18:02.28 | ManxPower | scfrec, then complain to your provider since the provider is doing the PSTN/Voip conversion |
18:02.29 | tzanger | and there's no txfax (go on in context on hangup) :-) |
18:03.04 | Ariel_ | There seems to be issues with the internet today... Some locations are down. |
18:03.07 | Ariel_ | or very slow. |
18:04.05 | harryvv | hi Ariel looks like shaw cable is invading the voip market in vancouver BC. |
18:04.31 | harryvv | Ariel, do you have a backbone web site that shows sites are down? |
18:05.12 | l1nux | ManxPower, from http://www.voipuser.org/forum_topic_330.html "g729 & gsm" ): |
18:05.36 | Ariel_ | harryvv, no |
18:05.49 | ManxPower | l1nux, H729 is a patented codec and requires a licensing fee to use |
18:05.53 | ManxPower | so I guess you must use GSM |
18:06.31 | *** part/#asterisk jcollie (n=jcollie@lt16586.campus.dmacc.edu) |
18:06.51 | l1nux | ManxPower, i dont have gsm in my ata device ): |
18:08.08 | ManxPower | l1nux, then you'll need to use ulaw or alaw from your device to Asterisk, then from Asterisk to voipuser would be GSM |
18:08.55 | l1nux | ManxPower, ohh! is possible |
18:09.12 | harryvv | Ariel, here is a snapshot of internet traffic. |
18:09.15 | harryvv | http://www.internettrafficreport.com/namerica.htm#graphs |
18:09.16 | ManxPower | l1nux, Most people use ulaw or alaw (but not both) for the local network |
18:10.17 | BrianR___ | ulaw on the lan - especially since many of the cheaper hardphones suck and fall over if you try to run too many simulataneous calls with compression :) |
18:11.08 | harryvv | Is there any current voip showsalers that can transfer my telus phone number as a did? |
18:11.20 | harryvv | whosalers that is |
18:11.29 | l1nux | ok, thanks ManxPower :) |
18:12.41 | Ariel_ | argh the wait for the zap to be realy is exten => _X.,1,Dial(Zap/g0,www/${EXTEN}) |
18:12.44 | vp7 | Hello! Could anyone tell me, if it's possible to run asterisk with H.323 support on FreeBSD? I'd like to run Asterisk as office PBX and our provider can give us only H.323 |
18:13.45 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
18:13.56 | ManxPower | Ariel_, Actually exten => _X.,1,Dial(Zap/g0/wwww${EXTEN}) |
18:14.02 | Ariel_ | vp7, asterisk can support h323 as an adon. I don't know about freebsd since I don't use it. |
18:14.17 | Ariel_ | ManxPower, thanks my head is not on right today. |
18:14.32 | ManxPower | Ariel_, and as you know it only works on analog interfaces |
18:15.09 | Ariel_ | only works on the tdm400 and the x100p not the t1 pri? or a channelized t1? |
18:15.48 | ManxPower | Ariel_, Correct |
18:15.56 | ManxPower | it might work on T-1<->Channelbank |
18:15.57 | Ariel_ | ManxPower, thanks... |
18:16.01 | vp7 | ARiel_: And how about Linux? I tried to find valid versions of openh323&pwlib but didn't find any that can be compiled successfully with asterisk. Do you know where can i donwload it? |
18:16.06 | Ariel_ | yes I know it works on a c/b |
18:16.07 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
18:16.14 | ManxPower | Oh! It WILL work on channelized T-1. |
18:16.18 | Ariel_ | vp7, yes |
18:16.26 | Ariel_ | linux works great I use CentOS |
18:16.36 | ManxPower | vp7, There are at least FOUR H323 drivers for Asterisk |
18:17.34 | ManxPower | chan_h323 (NuFone, included with Asterisk). asterisk-oh323 (3rd party download), chan_ooh323 (from asterisk-addons), and chan_woomera (openpbx.org I think) |
18:17.53 | vp7 | <PROTECTED> |
18:17.58 | *** join/#asterisk fugitivo (n=ajf@201.255.102.19) |
18:18.13 | Ariel_ | vp7, gentoo works |
18:18.22 | Ariel_ | ~docs |
18:18.24 | jbot | rumour has it, docs is 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 |
18:18.29 | l1nux | ManxPower, no sound ): |
18:18.31 | vp7 | ManxPower: Maybe i've done some wrong things, but these channels didn't wanted to compile :( |
18:18.32 | fugitivo | gentoo rocks |
18:18.33 | Ariel_ | vp7, see the wiki |
18:18.51 | Ariel_ | vp7, I don't use h323 for any of my setups. |
18:19.12 | vp7 | Ariel_: thnx, i'll try to search again. Hope this time will be more efficient :) |
18:19.35 | l1nux | is only one way audio |
18:20.20 | vp7 | Ariel_: Btw, our provider gives us H.323. Maybe the better way will be to convert it into SIP with external software? Do you know how is better? |
18:20.37 | ManxPower | l1nux, sounds like a NAT problem |
18:20.46 | brainlight | heya, anybody here has an eicon diva BRI-2M working on a 2.6 kernel with asterisk? |
18:21.11 | Ariel_ | vp7, no but there are other h323 channels that work. |
18:21.30 | l1nux | ManxPower, possible, but work fine with sipphone and others |
18:22.02 | l1nux | one way audio, is only with voipuser ): |
18:22.15 | harryvv | who has a track record of reliability and the infrastructor to keep up with demand? |
18:22.19 | Ariel_ | argh where are the people from nufone when you need them. |
18:22.39 | harryvv | and has also some kind of customer service? |
18:23.06 | *** join/#asterisk sjaak538 (n=sjaaknab@d5c53145.dsl.concepts.nl) |
18:23.50 | harryvv | ohh common somone must know from experaince? |
18:24.04 | l1nux | ManxPower, howto debug it ? |
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18:41.14 | Seyr | Im using RealTime and having problems with MWI. I have rtcachefriends=yes and I can leave a message and retrieve them, just no light on the handset for my 7960. |
18:41.52 | *** part/#asterisk reperire (n=nathan@wbs-146-168-56.telkomadsl.co.za) |
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18:50.12 | d33t | anyone around to answer a quick * question? |
18:50.32 | Uther_P | just ask... you don't need permission to ask |
18:50.41 | d33t | heh, k, thanks. |
18:51.00 | d33t | i'm trying to connect my asterisk server to sipphone and i can't get it to see incoming calls |
18:51.12 | Hmmhesays | anyone wanna buy a t1 card off me? |
18:51.17 | Hmmhesays | new |
18:51.32 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
18:51.36 | bjohnson | $5 |
18:51.36 | Uther_P | d33t: are you registering? |
18:51.37 | d33t | i can make outbound calls to other sip networks and use my "gizmo out" minutes, but nothing comes |
18:51.40 | d33t | *in |
18:51.41 | d33t | yes |
18:51.51 | d33t | it syas registered on sipphone.com |
18:52.02 | Jabroni | guys lil' question, is the pickupexten parameter introduced in the asterisk 1.0.9 release?? I added pickupexten = *9 but when a phone dials it sip returns a 484 |
18:52.04 | d33t | <PROTECTED> |
18:52.16 | Uther_P | does the context towhich it's registering in exist, and does it have an 's' extension? |
18:52.25 | asterisk99 | anyone know how to keep asterisk from chopping off 1st 1/4 second from sound files.... example "Welcome" comes out "Come" ... don't want that!!! |
18:52.43 | Seyr | asterisk99: add a Wait(1) before playing the file |
18:52.44 | *** join/#asterisk _Thor (n=Christia@rrcs-24-39-156-126.nyc.biz.rr.com) |
18:52.45 | d33t | the context exists, but i don't have an s extension |
18:52.50 | d33t | heh...... i'm such a n00b |
18:52.50 | Hmmhesays | i'd probably let it go for 4fiddy |
18:53.27 | Uther_P | d33t: actually, you probably only need an extension for the phone number that the calls would be comming in on |
18:53.27 | d33t | i thought i use the exten => SIPNUMBER,x,..... context |
18:53.34 | _Thor | anyone knows sangoma installation? |
18:53.36 | bjohnson | asterisk99: add a wait before picking up |
18:53.40 | d33t | i have this..... |
18:53.46 | asterisk99 | Seyr: You;d think that would work... but I still get COME - even with wait(2) |
18:53.47 | d33t | exten => ${SIPPHONENUM},1,NoOp(Incoming call from SipPhone Account ${SIPPHONENUM}) |
18:53.51 | d33t | and the variable is set |
18:54.03 | d33t | and of course, other lines follow |
18:54.04 | d33t | heh |
18:54.09 | bjohnson | I don't have any fiddy's |
18:54.35 | Uther_P | d33t: try hard coding the number |
18:54.40 | d33t | k |
18:54.42 | bjohnson | how about 4fathers |
18:55.08 | Uther_P | and add an 'i' extension |
18:55.14 | Seyr | asterisk99: no idea :-( |
18:55.24 | _Thor | Sangoma T1 installation, anyone knows how to configure it? |
18:55.30 | d33t | still nothing, right to sipphone vm system |
18:55.43 | d33t | i am running * in verbose mode and the call never seems to come in |
18:56.07 | Uther_P | d33t: I'm going to assume you reloaded after changing the dialplan |
18:56.12 | d33t | yes |
18:56.20 | d33t | heh, i'm not that new :) |
18:56.28 | bjohnson | and verbose is level 5 or higher |
18:56.40 | Uther_P | turn on sip debugging on your provider... see if they ever even send you an invite msg |
18:56.46 | d33t | using 6 v's |
18:56.56 | d33t | asterisk -c -vvvvvv |
18:57.44 | bjohnson | easy there fella .. we just met |
18:58.16 | Uther_P | just friends, I swear |
18:58.19 | asterisk99 | Seyr: There's a trick... answer() first... THEN Wait(1) |
18:58.19 | Uther_P | heh |
18:58.55 | d33t | Uther_P: any idea where i would turn on "sip debugging" at sipphone.com or gizmoproject.com? |
18:59.01 | d33t | i don't see anything like that |
18:59.03 | Uther_P | asterisk99: I was going to suggest that... but didn't think it would have made a diff since you put the wait on, hrm |
18:59.09 | Uther_P | d33t: in the cli |
18:59.34 | Uther_P | d33t: sip debug ip (ip of the provider) or sip debug peer (peer name) |
19:00.00 | Seyr | asterisk99: ah, I do that by default :-) |
19:00.03 | Uther_P | it auto completes... so if you just type "sip debug peer " then push tab, it'll show you the peers to choose from a list |
19:00.10 | asterisk99 | Uther_P: Astwisk is vewy twicky ;) |
19:00.50 | Seyr | Anyone know why I cannot get MWI when using Realtime with rtcachefriends=yes? |
19:00.52 | asterisk99 | anywone know why the default 's' extension does not work for PRIs? |
19:00.57 | d33t | i can't tab the peer's name |
19:00.58 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
19:01.05 | d33t | when i do sip show peers i get.... |
19:01.08 | Uther_P | d33t: type 'sip show peers' |
19:01.13 | d33t | Name/username Host Dyn Nat ACL Mask Port Status |
19:01.13 | d33t | desktop/desktop (Unspecified) D N 255.255.255.255 0 Unmonitored |
19:01.13 | d33t | cordless/cordle 192.168.1.50 D N 255.255.255.255 5060 Unmonitored |
19:01.14 | d33t | proxy01.sipphon 198.65.166.131 N 255.255.255.255 5060 Unmonitored |
19:01.15 | Uther_P | eek |
19:01.27 | Uther_P | dude, http://pastebin.ca |
19:01.54 | d33t | the desktop softphone is not running, the cordless (on a grandstream) is connected, and it seems sipphone is as well |
19:02.04 | Uther_P | d33t: sip debug peer proxy01.sipphone.com or whatever that name is |
19:02.13 | d33t | yeah, that's right |
19:02.36 | d33t | ok, debugging enabled...... try the call again? |
19:02.40 | ManxPower | <PROTECTED> |
19:03.09 | Uther_P | yea.. his problem is the proxy entry |
19:03.39 | d33t | looks like it came in |
19:03.40 | Uther_P | d33t: yea, call it... if its comming back to you, it'll dump a bunch of sip messages |
19:04.00 | d33t | it did |
19:04.22 | Jabroni | anyone uses the call pickup module ? |
19:05.31 | d33t | anything i should be looking for in there? |
19:06.17 | Uther_P | d33t: paste your sip.conf on http://pastebin.ca |
19:06.41 | d33t | give me sec, i need to strip out the comments |
19:06.47 | lehel | goodbye all |
19:07.20 | Uther_P | d33t: cat sip.conf | grep -vE '/^;/' |
19:07.59 | d33t | heh, gues that'd work well too |
19:10.27 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
19:11.44 | d33t | uther_p: sent link |
19:12.12 | *** join/#asterisk tsetane (n=tsetane@87.252.68.0) |
19:13.15 | Uther_P | d33t: register => user[:secret[:authuser]]@host[:port][/extension] is the syntax... you aren't specifying an extension, thus incomming calls are dropped into the 's' extension |
19:13.44 | d33t | ah, was never sure how that worked. like i said, n00b |
19:13.53 | d33t | so if i put my sipphone number in the extension it'll work? |
19:14.24 | *** part/#asterisk tsetane (n=tsetane@87.252.68.0) |
19:14.39 | d33t | register => 1747601xxxx:xxxx@proxy01.sipphone.com/1747601xxxx |
19:14.41 | d33t | like that? |
19:14.45 | Uther_P | d33t: yea |
19:14.57 | d33t | ok, i'll give it a go |
19:15.13 | d33t | is there a link you can send me to that explains the s, r, t, etc extensions? |
19:15.24 | Uther_P | but you can put whatever extension you want at the end there... the last option specifies what local extension calls come into from that provider |
19:15.35 | d33t | oh |
19:15.49 | d33t | so if i redo the extensions.conf to that ext it will use that |
19:15.51 | d33t | i see |
19:15.58 | Uther_P | d33t: s is default or 's'tart, t is timeout, i is invalid.... not sure about r |
19:16.18 | Uther_P | or you just put in an 's' extension |
19:16.24 | d33t | a lot of examples i read use r, i think it works like timeout but just proceeds to the next line |
19:16.38 | d33t | that would also work i guess |
19:17.04 | d33t | but i want to set it up to take calls in from FWD too |
19:17.08 | *** join/#asterisk buddah (n=djbrianc@67.110.253.129) |
19:17.19 | d33t | and handle them differnetly |
19:17.24 | buddah | does anyone know how to fix the problem with cisco ata 186's second line not ringing |
19:17.24 | buddah | ? |
19:17.28 | d33t | so i don't want to just use the default s |
19:17.36 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
19:17.38 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
19:17.54 | tzafrir_laptop | which is the current "stable" spandsp? 0.2 or 0.3? is 0.3 "fluid" or the recommended version? |
19:17.58 | Uther_P | then either specify different extensions to dump the incomming calls |
19:18.09 | d33t | yeah, 'tis my plan |
19:18.28 | d33t | awesome! |
19:18.32 | d33t | works liek a charm |
19:18.34 | d33t | thanks! |
19:18.42 | Uther_P | d33t: extensions can also be words, as long as they aren't reserved words |
19:18.46 | Uther_P | no prob |
19:18.58 | d33t | ah, that explains some of the other examples i saw |
19:19.03 | d33t | can you also use contexts? |
19:19.24 | d33t | like use /sipin and have a sipin context with an s extension? |
19:19.35 | buddah | anyone heard of pap2-na's having issues where the second line won't ring? |
19:19.37 | Uther_P | e.g. if you wanted to have FWD and SIPPHONE extensions... you end your registration with /FWD or /SIPPHONE |
19:20.17 | d33t | can i use them as contexts too then, or just as extensions in the default context? |
19:20.28 | Uther_P | no, those are extensions |
19:20.32 | d33t | ok |
19:20.51 | d33t | sweet, you taught me more than you know, heh |
19:20.52 | d33t | thanks again |
19:21.05 | Uther_P | they are dumped in the context that is specified in your sip.conf... OR you can use a provider name by creating it as a peer |
19:21.13 | Uther_P | then putting that peer in a specific context |
19:21.50 | d33t | i think that's a bit over my head |
19:21.57 | *** part/#asterisk TK9 (n=Administ@p54B28C56.dip0.t-ipconnect.de) |
19:22.03 | d33t | i'll stick with the default context |
19:22.06 | d33t | since it works :) |
19:22.16 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-209.nas28.salt-lake-city1.ut.us.da.qwest.net) |
19:22.17 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
19:22.22 | Uther_P | d33t: much like you made the sip entries for you phones... you make one for your provider, and specify a context |
19:22.58 | d33t | yeah, i've seen it donw that way before too, but i couldn't get that working either |
19:23.08 | d33t | although, now that i know a little more, i might be able to |
19:23.28 | d33t | but, is there an advantage 1 way or the other, besides he config being a little more clear? |
19:24.27 | Seyr | Anyone know why I cannot get MWI when using Realtime with rtcachefriends=yes? |
19:25.31 | Uther_P | d33t: for a simple config... no... its helpfull though, if you have many lines/providers and/or sets of extensions that must be handled a different way |
19:26.25 | Uther_P | d33t: especially if your provider will be sending calls to you where the target extension varies... then it enters the context at the extension of the number |
19:26.33 | *** part/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
19:27.39 | d33t | ok |
19:29.42 | *** join/#asterisk Tili (i=Tili@202-133-65-86-dialup.sat.net.pk) |
19:31.25 | FuriousGeorge | has anyone ever installed a doorphone? |
19:31.56 | FuriousGeorge | specifically an analog one. like the some of the ones from viking |
19:32.04 | *** join/#asterisk corne (n=corne@ndn-165-157-254.telkomadsl.co.za) |
19:34.43 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
19:35.36 | *** join/#asterisk pbd (n=plancomm@12.144.118.36) |
19:35.44 | pbd | Good afternoon, all. |
19:35.52 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
19:36.08 | jayk- | what are the advantages to using chan_sccp over SIP for the Cisco 7960 phones? should i use SIP or chan_sccp? |
19:36.22 | pbd | Jayk: Your Mileage May Vary. |
19:36.45 | pbd | I've got a 7960 on my desk right now running SCCP attached to Callmanager, and another one running SIP attached to Asterisk. |
19:36.54 | jayk- | ah. |
19:37.17 | jayk- | i have problems with the cisco phones dropping calls occasionally, getting hung up (where they won't dial) and need to be rebooted to have the problems fixed. |
19:37.19 | pbd | chan_sccp (or chan_skinny) have various strengths and weaknesses- most of which lie in the firmware revision. |
19:37.25 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
19:37.25 | jayk- | and im wondering if it is because they are SIP |
19:37.31 | pbd | Under SCCP, I've seen that problem- but never under SIP. |
19:37.43 | pbd | But I've heard other people say they've seen it too. |
19:37.50 | jayk- | i'm running SIP7.5 |
19:37.54 | pbd | So am I. |
19:37.56 | jayk- | hrm |
19:38.10 | jayk- | sometimes i might pick up phone and dial, and it won't connect, but if i try it again, it will work |
19:38.26 | pbd | That's something to check your asterisk console about. |
19:38.40 | pbd | I have seen cases where the console reports that the phone is lagged- I had to adjust for that. |
19:38.52 | jayk- | when it does that, it doesn't output anything on the console |
19:38.58 | pbd | I'm trying to track that one down- I *think* it's in relation to CDP and the switches I use. |
19:39.01 | jayk- | how did you adjust for that? |
19:39.09 | jayk- | i don't use CDP on the switches |
19:39.09 | pbd | Do you have verbose and debug turned up enough? |
19:39.16 | jayk- | i have it turned up to about 9. :) |
19:39.22 | *** join/#asterisk fordvoice (n=chrisf0r@cpe-69-133-21-43.cinci.res.rr.com) |
19:39.51 | pbd | You adjust the milliseconds of qualify in the sip.conf entry. |
19:39.54 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
19:39.55 | pbd | I find 500ms works well. |
19:40.04 | Ariel_ | Just wonder why it's so hard to get good customer service in the Voip/Telecom buisness. |
19:40.05 | jayk- | the phones are all less than 10ms |
19:40.24 | pbd | I see this even when I can ping the phones from the server at <2ms .. but Asterisk (HEAD) reports it's TOO LAGGED, at >200ms. |
19:40.43 | jayk- | huh |
19:40.43 | pbd | Some folks have reported losing calls when the TOO LAGGED message comes across.. dropouts, that sort of thing. |
19:40.46 | jayk- | so 500ms works? |
19:40.49 | Seyr | qualify isnt ping |
19:40.52 | pbd | It works better. |
19:41.05 | jayk- | qualify=500ms? |
19:41.13 | jayk- | or just 500 isnt it |
19:41.15 | pbd | No, it's not ping- there's a lot of stuff there.. I'm seeing it as the phones are basically taking too long to respond to the headers. |
19:41.24 | pbd | You don't need the 'ms' on the line |
19:41.36 | jayk- | ill try that |
19:41.57 | jayk- | so 500ms doesn't degrade from the voice quality? |
19:41.59 | pbd | Someone here gave me a pointer that CDP, or the phone's attempt to work with it, may be slowing down the phone's response. |
19:42.07 | *** join/#asterisk [_gordo_] (n=gordo@bl5-187-197.dsl.telepac.pt) |
19:42.24 | pbd | qualify doesn't have anything to do with the voice quality itself... merely how long it takes to answer the headers. |
19:42.25 | jayk- | i have tos=lowdelay |
19:42.27 | jayk- | could that be part of it? |
19:43.07 | pbd | Depends on your switch. Are you running QOS? |
19:43.10 | jayk- | nope |
19:43.19 | pbd | (you should, btw, in almost all cases) |
19:43.31 | pbd | Then tos bits won't do anything one way or another. |
19:43.32 | jayk- | i've never used it before. |
19:44.21 | jayk- | how is it configured? |
19:44.26 | pbd | QOS is defined on your routers and switches- essentially, it moves certain traffic through before other traffic. Even on a relatively uncongested lan segment, it may matter- it helps mitigate bursting. |
19:44.55 | pbd | It's configured differently on each manufacturer's network equipment. |
19:45.04 | jayk- | i have all cisco switches and routers |
19:45.15 | pbd | If you have a simple hub attached to your phones and the server, with no other devices on the hub, you wouldn't need it. |
19:45.37 | pbd | If you're running them all as part of a LAN/WAN environment, with mixed loads, etc.. you probably need QOS running. |
19:45.53 | jayk- | k |
19:45.56 | pbd | emphasis on probably- your LAN may be different. |
19:46.42 | pbd | And it's not a panacea.. it merely helps. In practice, it's a way of saying- look, switches/routers, please make sure the RTP for my phones goes through before the pr0n for the web browsers. |
19:47.09 | jayk- | k |
19:47.11 | jayk- | ill check into it |
19:47.18 | pbd | But, if there's a lot of pr0n, and only a little RTP, it might not do much for you. |
19:47.37 | pbd | Now, I'll trade someone out there that longwinded answer for any experience in setting up caller-ID in Brazil? |
19:48.15 | pbd | I'll be testing it next week, but I'm hoping someone out there has some basic experience they can share- like 'I got it to work', or 'It will never work- you're on drugs'. |
19:48.30 | *** join/#asterisk tsetane (n=tsetane@87.252.68.0) |
19:48.35 | pbd | And no, I don't have any extra to share. :) |
19:49.13 | jayk- | wow, interesting |
19:49.17 | jayk- | Oct 13 12:48:46 NOTICE[2216]: chan_sip.c:9598 handle_response_peerpoke: Peer '102' is now REACHABLE! (241ms / 500ms) |
19:49.22 | jayk- | i just turned on qualify and got that msg |
19:49.43 | jayk- | only a couple of phones did that |
19:50.24 | pbd | If you keep watching, you'll see it for all the phones eventually. |
19:50.35 | jayk- | ok |
19:50.41 | jayk- | Oct 13 12:50:16 NOTICE[2216]: chan_sip.c:9604 handle_response_peerpoke: Peer '26201' is now TOO LAGGED! (501ms / 500ms) |
19:50.43 | jayk- | got that too |
19:51.13 | *** join/#asterisk Gnurdux (n=gnurdux@69.251.241.119) |
19:51.17 | jayk- | i have callerid working here in the states. :) |
19:51.19 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
19:51.26 | jayk- | but that was pretty simple, after we converted to a PRI from a voice t1 |
19:51.31 | Gnurdux | ok.... im trying to setup asterisk to connect to fwd |
19:51.47 | pbd | Yeah, caller ID in the US is a no brainer. |
19:51.49 | Gnurdux | i installed asterisk@home on an old box with no terminal |
19:51.59 | Gnurdux | and i followed the directions to setup FWD |
19:52.07 | Gnurdux | but im not sure how to use it |
19:52.16 | obsidian-studios | if using iax2 between * boxes what should I base the bandwidth on per line using ulaw? 80k? 20k? It's more about codec than protocol right? |
19:52.33 | pbd | Jayk: if you do a sip show peers, it will show you what the lag time is currently. |
19:52.50 | pbd | obsidian: you are correct- it's more codec than protocll. |
19:53.02 | pbd | But IAX takes a little smaller bite than SIP or H323.. |
19:53.11 | pbd | we're talking about 79K vs 80K, though. |
19:53.19 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
19:53.26 | jayk- | oh yeah, look at that |
19:53.35 | jayk- | that didn't show up before (the lag time) |
19:53.50 | pbd | It may only show up if you have qualfy on. |
19:54.28 | obsidian-studios | pbd: what's realistic for licensed codecs 20k or 40k? |
19:54.38 | jayk- | i think you are right |
19:54.50 | jayk- | thanks pbd. :) |
19:54.52 | pbd | Which licensed codec? |
19:55.22 | pbd | In theory, g.729 is about 8K.. GSM around 12K. |
19:55.45 | pbd | there's a couple of good bandwidth calculators around- google for them, I don't have the urls' handy. |
19:56.15 | Gnurdux | can someone help me? |
19:56.22 | pbd | Keep in mind, there's some overhead- that's just the RTP bandwidth. I usually count around 20K for a compressed codec, and 80K for ulaw, regardless. |
19:56.23 | obsidian-studios | pbd: ok, just trying to get guestimates for now, no worries on exacts, is there a preferred licensed codec? |
19:56.39 | tzafrir_laptop | Gnurdux, if you ask your question: maybe |
19:56.52 | obsidian-studios | pbd: tyring to stick with ulaw, but I think there are bandwidth limitations at one location |
19:56.58 | pbd | Licensed, I've seen people use 729 ($10 per channel), or 723 (ridiculous cost).. or GSM (preferred in Asterisk community- it's free and pretty good). |
19:57.01 | jayk- | w |
19:57.36 | tzafrir_laptop | speex should give a good quality, but takes very much CPU |
19:57.40 | obsidian-studios | pbd: quality? pretty sure phones will run ulaw, might use different codec in phone if available to avoid transcoding |
19:57.48 | obsidian-studios | pbd: gsm is native to * right? |
19:58.01 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
19:58.06 | *** join/#asterisk darby_t (n=tom@dnx29.neoplus.adsl.tpnet.pl) |
19:58.21 | tzafrir_laptop | obsidian-studios, just as much as other codecs (ilbc, g726, etc.) |
19:58.38 | obsidian-studios | tzafrir_laptop: all modules so * does not care |
19:59.10 | pbd | I run 729, ulaw, and gsm. They all have 'native' drivers in Asterisk.. including some I haven't mentioned. To my ear, 729 and gsm sound about the same, ulaw a little better. Don't try faxing over a compressed codec, as it narrows the frequency range.. but the human ear is pretty versitile. If you have users with perfect pitch, they're going to complain at anything less than ulaw. |
19:59.54 | Gnurdux | tzafrir_laptop, the prob is that my question isnt that well-defined |
19:59.55 | pbd | a PRI runs a form of ulaw/alaw- its what you'd get on a high quality digital line, since there is no compression. |
20:00.12 | Gnurdux | i setup FWD using the direction of the asterisk@home handbook |
20:00.18 | pbd | Gnurdux: Since you have no well defined question, the best answer we can give is 'maybe'. :) |
20:00.22 | Gnurdux | and i dont know if i didnt get it setup properly |
20:00.27 | Gnurdux | or i dont know how to use it |
20:00.43 | Gnurdux | but i cant call my old FWD contacts when i tell Kphone to connect to my asterisk box |
20:01.36 | pbd | Gnurdux: Sounds like it's time to go into the asterisk console and do a little 'set verbose 4' and 'set debug 4'- and see what the messages say. |
20:01.41 | obsidian-studios | pbd: cool ty, one client is getting a PRI, but the other is a non-profit with limited budget |
20:01.56 | Gnurdux | hmm |
20:02.11 | Seyr | Anyone know why I cannot get MWI when using Realtime with rtcachefriends=yes? |
20:02.13 | Gnurdux | how do you get to the asterisk console? |
20:02.18 | tzafrir_laptop | Gnurdux, you can try to call the clock or the echo test on FWD (612, 613) |
20:02.38 | tzafrir_laptop | Gnurdux, there's also an option in the web interface to send a call to your number |
20:02.43 | Gnurdux | oh |
20:02.44 | Gnurdux | ok |
20:02.56 | pbd | Gnurdux: Get onto the linux box where you installed asterisk, and type 'asterisk -r'. It's all ugly from there. :) |
20:03.21 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
20:04.08 | sgorilla | has anyone here messed around with access grid? |
20:04.08 | Gnurdux | tzafrir_laptop, how do i tell it to call me from the web interface? |
20:04.20 | sgorilla | or have a logitech quick cam 4000 |
20:04.49 | FuriousGeorge | anyone ever installed a door phone? ive seen people say they hook to fxo which makes sense if they only have a call button, and ive heard other people say they hook to fxs, which makes sense if they can dial extensions and whatnot\ |
20:04.53 | FuriousGeorge | anyone know? |
20:05.13 | FuriousGeorge | i wouldnt wanna buy anything w/o being sure i got the right thing |
20:05.21 | pbd | Furious: Most door phones run FXS- they're essentially bat phones. |
20:05.28 | fugitivo | FuriousGeorge: you can get adapters for FXS |
20:05.29 | Gnurdux | 612 and 613 say address incomplete btw |
20:05.36 | pbd | But google for 'door phone', and see what you get. |
20:05.51 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
20:05.58 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
20:06.05 | miztic | FuriousGeorge, i'll be getting into that soon, the doorphone we have acts like a phone extention, so i expect ours to require an FXS connection |
20:06.05 | FuriousGeorge | pbd: ive been using the google search on the wiki, getting conflicting info |
20:06.33 | pbd | George: In the end, it's going to depend on the model you pick. |
20:06.41 | fugitivo | FuriousGeorge: www.2n.cz, they have nice stuff |
20:06.53 | miztic | it'll get more fun when i have to activate the door contacts with a button on the phone |
20:06.54 | pbd | I can't imagine one running on FXO, since door phones typically don't have power to them.. but anything is possible. |
20:07.00 | FuriousGeorge | miztic: ive hooked a POTS phone to an fxs before and hung it by the front door, but i need a real outside intercom for this one, weather resistant, etc |
20:07.20 | miztic | ours came with the propriatary lucent system |
20:07.24 | mcf3782 | I have a "Can Asterisk do this...." question. :) |
20:07.25 | miztic | i plan to reuse it |
20:07.26 | FuriousGeorge | pbd: thats what i figured, it would be powered and it would ring the fxo when somene hit call |
20:07.32 | fugitivo | FuriousGeorge: www.2n.cz, check Communicators section |
20:07.47 | FuriousGeorge | fugitivo: checking |
20:07.53 | Uther_P | its not uncommon for pbx systems to have functionality for door phones to buzz open the door |
20:08.14 | pbd | George: It's a lot simpler to make a door phone unpowered- needing an FXS port, as the hardware is simpler and you don't need a separate power supply. But I suppose you could make one loop start, like an alarm line, or something. |
20:08.47 | mcf3782 | I have a single POTS line from BellSouth that has 3-way calling on it. To use the 3-way feature, you flash the switch hook, get a second dial tone, dial the other number, then flash again to get all 3 parties connected. |
20:08.50 | pbd | Uther: True- but electric locks are generally higher amperage, 12v systems.. not compatible with phones. |
20:09.18 | pbd | So it's more common to have the server talk to the lock through separate wire. Added security advantage- you can't override the lock by isolating the phone. |
20:09.41 | Uther_P | pbd: yea, would be best to have another device for buzzing it open, and use the door phone function as a logic gate to it |
20:10.05 | mcf3782 | Can Asterisk put an inbound call "on hold", dial a second call, and then join them together on request when someone on the pots line does something like say dial "*2" (doesn't really matter to me what code I use). |
20:10.26 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
20:10.40 | Uther_P | mcf3782: in the dialplan,,, you would want to use the flash application.. then senddtmf, then flash |
20:11.16 | Uther_P | flash, wait(1), senddtmf(number), wait(1), flash |
20:11.24 | mcf3782 | Ahh. cool. I figured the answer was probably "yes". Just wanted to verify that before I went off and started trying to figure out how to do it. :) |
20:11.38 | Uther_P | trying to bounce calls of your work? heh |
20:12.35 | pbd | Uther: That only works if you have three way calling on your line. Simpler if you have multiple lines on your system. :) |
20:13.14 | mcf3782 | I want a way for my parents, who really don't like all my "gadgets" and stuff, to be able to just have Asterisk put them on hold, dial my cell phone, and then 3-way conference us together without them having to hang up and call my cell phone if I don't happen to be at home. |
20:13.22 | Uther_P | mcf3782: remember, in this instance there is no dial app to get blocked on for the call... do unles you expect that your provider allows for unattended conference calls, then you'll have to put in a loop in the dialplan, simulating an indefinate wait so it doesn't establish the call, then hangup on the both of you |
20:13.37 | *** part/#asterisk Gnurdux (n=gnurdux@69.251.241.119) |
20:14.23 | mcf3782 | oh. ok. good to know. Thanks for that pointer, Uther_P. |
20:14.34 | Uther_P | no problem.. i;ve done this before, heh |
20:14.38 | pbd | mcf: Then you're in fine shape. I'd personally subscribe to a low cost outbound only VoIP provider for the outbound leg.. then simply execute a DIAL application out the VoIP provider to your cell phone. |
20:15.15 | Uther_P | bah, no need for all that for just a 3 way |
20:15.17 | pbd | (it's simpler that way, and no three way calling charges). |
20:15.22 | mcf3782 | That may certainly be an option. I'm just not that far along yet. :) |
20:15.36 | pbd | I dunno about your phone company- but mine charges extra for three way calls. |
20:15.39 | *** join/#asterisk Gnurdux (n=gnurdux@69.251.241.119) |
20:15.49 | Gnurdux | i had to change my dial rules |
20:15.53 | FuriousGeorge | fugitivo: that looks like exactly what i need but no distributors in the US |
20:15.59 | Uther_P | mcf3782: btw, if it behaves anyway like it did on my provider, you might need to jackup the volume |
20:16.01 | Gnurdux | is there a way to make it forward ALL calls |
20:16.14 | Uther_P | Gnurdux: exten - |
20:16.16 | Uther_P | err |
20:16.27 | Gnurdux | in dial patterns |
20:16.29 | pbd | exten => s,1,DIAL(FWD/#) |
20:16.29 | Uther_P | exten => _X.,1,goto |
20:16.30 | mcf3782 | BS charges me $.25 each time I use the 3-way-on-demand feature. |
20:16.39 | fugitivo | FuriousGeorge: I had a meeting with them last week, i can give you a contact if you want |
20:17.30 | pbd | Ahh, BellSouth. There's a happy company.. sitting out there next to the pond, ignoring everyone. :)( |
20:17.41 | mcf3782 | pretty much |
20:18.04 | mcf3782 | Anyone who isn't a BS customer care to guess what they want per-month for CallerID? |
20:18.15 | mcf3782 | It's sad sad sad |
20:18.29 | mcf3782 | $9.95 |
20:18.52 | Gnurdux | is there a wildcard dial pattern? |
20:19.27 | pbd | Gnurdux: 's' matches everything (default), or _X. ,which matches everything at least one digit long. |
20:19.44 | Gnurdux | with the quotes or witout? |
20:19.54 | pbd | Without. |
20:20.10 | pbd | exten => s,1,app_whatever |
20:20.30 | mcf3782 | If I felt like paying that extortionist rate; then any time mom or dad called me; they'd just get routed over to the cell phone automagically without them having to do anything except just wait. But I just refuse to pay over $10/month (by the time you add taxes) for the ability to get CID data. |
20:20.54 | pbd | mcf: So port your number to Vonage or some such. |
20:21.15 | pbd | $25/month flat, all services included. |
20:21.18 | *** part/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
20:21.21 | Gnurdux | just s? |
20:21.35 | pbd | ~voip-info |
20:21.36 | jbot | i guess voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
20:21.53 | pbd | Thanks, jbot. |
20:22.13 | Gnurdux | so i should use that? |
20:22.30 | mcf3782 | can't without moving my Internet connection from BellSouth.net to something else. I have a BellSouth.net DSL Internet connection. They won't sell me DSL without me having a voice line with them. |
20:22.52 | pbd | Gnurdux: There's a million ways to write a dialplan. My preferred method is the s extension- but it's your dialplan. |
20:23.13 | Gnurdux | i just want to have 1 line that forwards EVERYTHING |
20:23.42 | pbd | mcf: So get an 'emergency use only' line from them, with DSL on it. If you push them hard enough, they may just drop your regular rates- especially if you tell them.. you know, I just got this cable modem offered in my area, and Vonage is looking fine.. |
20:23.42 | h3x0r | i wish zaptel worked on freebsd 6 |
20:23.58 | Gnurdux | theres a freebsd 6? |
20:24.03 | Gnurdux | is it beta? |
20:24.06 | h3x0r | dude |
20:24.08 | h3x0r | they are working on 7 now |
20:24.25 | Gnurdux | wait |
20:24.29 | pbd | mcf: That's the way SBC is working, in most cases. |
20:24.29 | h3x0r | 6 is on beta #5 but its pretty damn close to release |
20:24.35 | Gnurdux | freebsd 6 is stable? |
20:24.39 | Gnurdux | see i use linux |
20:24.39 | h3x0r | 7 is current for some time now |
20:24.45 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
20:24.46 | Gnurdux | dont really follow the BSD world |
20:24.59 | h3x0r | well their main web page is somewhat misleading |
20:25.05 | h3x0r | i mean yeah it says 5.whatever is stable |
20:25.16 | h3x0r | but new hardware dosent fucking work with 5 |
20:25.23 | Gnurdux | hehe |
20:25.26 | mcf3782 | pbd - tried that already. :) BS won't attach DSL to an "emergency use" class voice line. Not "can't".. "won't".. they wouldn't make as much money that way. ;) |
20:25.47 | pbd | mcf: So go cable modem, and chuck BS into the wind. :) |
20:25.48 | h3x0r | like my supermicro server's ethernet controller and sata ich7r controller |
20:25.57 | h3x0r | It aint like linux dosen't have the same problem |
20:26.08 | h3x0r | you can't time warp into the future to write device drivers for stuff that dosen't exist yet |
20:26.21 | wunderkin | mcf3782 what does changing internet providers have to do with using voip? |
20:26.23 | Gnurdux | yes i know |
20:26.31 | h3x0r | you can backport to older kernels and stick it in the stable branch |
20:26.35 | mcf3782 | I probably eventually will. The cable provider here is finally getting their act together. |
20:26.35 | Gnurdux | but i dont have supernew hardware either |
20:26.37 | h3x0r | but that defeats the purpose of the stable branch |
20:27.01 | h3x0r | and that is 10x easier to do in *bsd than linux |
20:27.04 | pbd | Hey, man- this is the telephony world. We don't run anything that hasn't been in production for 2+ years. And who needs sata for an Asterisk box anyway? |
20:27.07 | h3x0r | because theres one cvs respository for userland and kernel |
20:27.26 | h3x0r | Uhm, who uses EIDE on a Asterisk box |
20:27.30 | Katty | who wants to proof read my resume? :P |
20:27.45 | Gnurdux | who can help me? |
20:27.49 | h3x0r | SATA deals with hard drive failures better, in conjunction with software raid 1 |
20:27.55 | pbd | Katty: Having a good day, are we? |
20:27.58 | h3x0r | you unplug a drive you ran atacontrol on with sata |
20:28.01 | Katty | pbd: mrow? |
20:28.02 | h3x0r | and the system keeps running |
20:28.08 | h3x0r | you unplug a eide drive and the system dies. |
20:28.34 | h3x0r | well CF is better |
20:28.41 | h3x0r | but what about voicemail storage |
20:28.55 | h3x0r | and CDRs unless you are going to send those to another box.. but then again you STILL need a raid for that |
20:28.55 | pbd | Katty: You're putting out your resume- that's always a good sign. |
20:28.58 | h3x0r | somewhere |
20:28.59 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
20:29.03 | Katty | pbd: uh, no. |
20:29.11 | h3x0r | I am using 3ware controllers on database servers |
20:29.16 | Katty | pbd: i just want someone i know to proof read it for me |
20:29.17 | h3x0r | but that would get really expensive if i put it on everything |
20:29.28 | pbd | h3x0r: Voicemail? I use paper tape. It stands the test of time. |
20:30.06 | pbd | Katty: And you know us here? Boy, I feel honored. ;-) |
20:30.06 | mcf3782 | Wonder how many people still know what that is? :) |
20:30.16 | Katty | pbd: i don't know you |
20:30.17 | *** part/#asterisk Gnurdux (n=gnurdux@69.251.241.119) |
20:30.27 | Katty | pbd: i might let twisted or anthm proof read it for me though. |
20:30.53 | pbd | Ahem. So you asked in open forum who wants to read a resume, and now you're being picky. |
20:31.00 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
20:31.02 | Katty | well, duh |
20:31.22 | Katty | pbd: you've insaned. kthxbi |
20:31.35 | pbd | Hrm. Considering that I've been hiring people for companies for the last 10 years or so, I might have offered. But now, fergeddaboudit. |
20:32.05 | pbd | Fickle. But I'd have guessed that from your nick. |
20:32.15 | pbd | ;-) |
20:32.53 | Katty | k |
20:33.06 | Katty | twisted[asteria]: find me when you're done at that conference, kthx (= |
20:33.50 | pbd | Katty: I do have one question for you, that I started to ask you about a month ago, completely unrelated to Asterisk, but related to Linux, Samba, and 'click to print' under Windows. |
20:34.19 | pbd | I think you said you got it working. I've had my MS and Linux guys working on it for weeks now, with only minimal success. Did you say you had it working? |
20:34.58 | Katty | Hrm. Considering that I've been using linux and samba for years now, I might have offered. But now, fergeddabouit. |
20:35.20 | Katty | Fickle. But I'd have guessed that from your attitude. |
20:35.21 | pbd | Yeah, yeah. Same answer you started to give last month. Figures. :) |
20:35.32 | arp2 | get a room you two |
20:35.41 | Katty | arp2: oh but I /do/ so love mocking people. |
20:35.48 | Katty | arp2: and I shall, until they grow up. |
20:35.53 | FuriousGeorge | pbd: so i just checked a company called doorbellfon, and theirs hook up to "a spare trunk" so i assume that would be an fxo or something like that |
20:36.16 | Katty | pbd: why don't you /hire/ someone to fix that little issue of yours, hmm? |
20:36.22 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
20:36.32 | corne | lo ppl |
20:36.45 | FuriousGeorge | its seems like there are two flavors of door phone, the one button kind which wants to go to an fxo, and the kind with a full dialpad which wants an FXS |
20:37.11 | pbd | Katty: I have, thanks. Unfortunately, the best answers in the world are 'it works sometimes'. Then again, if I know someone who has gotten it to work successfully, I might even ask them. :) |
20:37.36 | ender | pbd: where is the 'click to print' dialog at? |
20:37.50 | Seyr | Anyone know why I cannot get MWI when using Realtime with rtcachefriends=yes? |
20:38.00 | pbd | Furious: Makes sense. Maybe you can get someone to ship you a loaner? |
20:38.05 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
20:38.19 | pbd | ender: Under file, run "//servername/printername" |
20:38.22 | mcf3782 | 'it works sometimes'.. isn't that pretty much a given with all things Microsoft? ;) |
20:38.35 | e3g | hi |
20:38.37 | corne | i have just tried to configure amp, but i made a mistake somewhere. amp doesn't show the status of asterisk(if a phone is ringing or call is active), but correctly indicates the extentions and trunks |
20:38.49 | pbd | The idea is that a server can have all the drivers on it, and by installing the printer off the server, the drivers get sent to you, set up, and work. |
20:39.03 | e3g | can we setup call back system with asterisk ? |
20:39.19 | ender | pbd: oh I see. We don't share printers via samba. We use cups so that Windows systems print to them via IPP |
20:39.25 | pbd | Under samba- it's not so simple. We've gotten it to send the drivers down, but they don't work in the end. Frustrating, and barely documented. Or worse- documents that say they work are either out of date or untested. |
20:40.00 | pbd | ender: We can do that too- problem is, we print in Japanese, which the CUPS drivers need additional drivers for, which aren't available or don't work well. |
20:40.06 | ender | ah |
20:40.10 | corne | can you setup callbarring with asterisk? |
20:40.27 | ender | anybody know if it is possible to do handset 'paging' w/ *? Something we used on our old phone system. |
20:41.02 | vp7 | e3g: It's possible :) |
20:41.15 | fiber0pti | ender: if the phones you are using have an auto answer functionality |
20:41.21 | e3g | vp7: then what took you so long to answer me ? ;) |
20:41.23 | pbd | ender: Also, we have some copiers with a boatload of options, which need options that aren't supported on the stock CUPS driver. |
20:41.39 | e3g | vp7 : how ???? how asterisk can do automatic dialing? |
20:41.40 | fiber0pti | ender: otherwise you'd need a separate paging system |
20:41.48 | vp7 | e3g: I'm newbie here, so was waiting for answer from GURU :) |
20:42.02 | pbd | ender: Define handset paging. |
20:42.06 | ender | pbd: er... I don't follow. Cups uses foomatic ppd files. Samba would too. how do you get around that? |
20:42.34 | e3g | vp7: when all gurus Failed ...... newbie starts from there :) |
20:42.35 | corne | i have just tried to configure amp, but i made a mistake somewhere. amp doesn't show the status of asterisk(if a phone is ringing or call is active), but correctly indicates the extentions and trunks |
20:42.43 | ender | pbd: in our old system, we could pick up the phone, hit 'page' and select handsets and then speak. All ahndsets would broadcast. |
20:42.49 | pbd | ender: If the stock ppds worked with the copier, it would print- but the options dialogs aren't there for all the other stuff. |
20:43.02 | pbd | ender: Ahh. Yes, it does support it, but in a backhanded sort of way. |
20:43.19 | ender | pbd: right, how does Samba get around that? Also, when using IPP, you install a windows side driver for the printer, and the CUPS IPP is just the URI to the printer. |
20:43.23 | e3g | vp7 : shoot!!! your suggestion |
20:43.29 | vp7 | e3g: You can create a file with calling parameters & asterisk will call where you want within several seconds. |
20:43.37 | pbd | The common method is to open up a dynamic meetme bridge, then add in auto-answer extensions from all the phones you want to page to, (make sure they're added on mute), talk, then break apart the bridge. |
20:44.01 | e3g | how asterisk can call without any event? |
20:44.12 | pbd | ender: Theoretically, you don't need the foomatic stuff- you can do the direct rpcclient calls to add the entries into the samba tdbs, so it *looks* like a real Windows print queue. |
20:44.14 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
20:44.25 | corne | e3g, does e3g stands for egg? egg in you face!! |
20:44.46 | e3g | dont you know e3g? |
20:44.55 | pbd | Once it looks like a real print queue, the driver is installed locally, and the file is printed up to the queue, which is defined by samba as pointing to a cups printer.. that's about as deep as I get with it personally. |
20:45.17 | e3g | vp7: ok I get the info in the file......now ??? how asterisk can do automatically DIAL() ? |
20:45.25 | vp7 | e3g: You receive call. Came to conclusion, that call back should be made. Create a file with required parameters. Asterisk will find this file and make a call. |
20:45.38 | corne | no, but i now super man |
20:45.57 | ender | pbd: yeah, seems like you've put your thought into it. At that point though, you have to wonder, why spend so much effort trying to emulate a windows print server, instead of just installing a windows print server. THere are lesser evil things. |
20:46.02 | vp7 | e3g: ok, let's me try to find what you have to do. wait for a bit... |
20:46.41 | e3g | vp7: after saving file.....I dont know how to let the asterisk know to dial :) ...ok I wait |
20:46.42 | pbd | ender: Basically, the problem is that I have 12 remote offices, each with a small fileserver, better suited to a linux box than to a fullblown windows server (less costly, more reliable, once developed). |
20:46.49 | pbd | For the main office, we still do run it as windows. |
20:47.09 | e3g | corne: I make MEN ...Super man, Spider man, would you like to be skunk man ??? |
20:47.32 | pbd | Although eventually, we'll axe the Win2K3 network in favor of samba PDCs and LDAP. |
20:47.36 | corne | heheeheheheheheheheheheheehhee |
20:47.40 | corne | bla bla bla |
20:47.50 | vp7 | e3g: you save file in a special directory that is scanned by asterisk for new events. so you just have to save it. i'll try to find a name of this feature & this directory |
20:48.12 | Seyr | Im using Realtime and have rtcachefriends=yes, but I do not get MWI on my 7960 when I leave voicemail. Anyone have any experience with Realtime that might have any suggestions on what to check? |
20:48.53 | pbd | Now that the channel is a little more active, I'll go for a repeat.. anyone out there handling caller-id inbound in Brazil? |
20:49.15 | *** join/#asterisk pardove (n=pardove@195.146.47.201) |
20:49.19 | e3g | vp7: I really appreciate your efforts buddy |
20:49.21 | vp7 | e3g: I've got :) Just look into http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI |
20:49.33 | ender | pbd: yeah. |
20:49.44 | corne | can you setup callbarring with asterisk? |
20:49.44 | ender | pbd: hrm, what about using those embedded 'print server' devices? |
20:49.50 | pardove | when * attempts to detect CallerID on Zap Channels |
20:49.57 | pardove | when * attempts to detect CallerID on Zap Channels? |
20:50.07 | pbd | ender: Thought about it- but most of them don't do the automatic driver downloads. At least, I haven't found one that does. |
20:50.07 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
20:50.14 | ender | pbd: ah. |
20:50.25 | vp7 | e3g: Feature is called "Asterisk auto-dial out" and docs are here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
20:50.31 | ender | pbd: w/ 12 people or so, is it really that difficult to put up a doc on how to do the driver install? |
20:50.39 | pbd | I smell a lot of Microsoft shenanigans there.. printing seems very poorly documented. |
20:50.40 | vp7 | e3g: Is this an answer to your question? |
20:51.10 | e3g | vp7: do you love your girl friend ??? :) |
20:51.17 | e3g | vp7: hey ....thankssssss!!! |
20:51.22 | pbd | Now, Katty.. does that explain to you a little better why 'just hire someone' isn't really the right answer here? |
20:51.27 | e3g | vp7: U got me there :) |
20:51.56 | pbd | Money isn't the issue- its finding someone who can actually do it, and repeat it, in our environment. |
20:51.57 | e3g | vp7: thanks man .....I appreciate that .... |
20:52.13 | vp7 | e3g: not at all :)) |
20:52.28 | pardove | when * attempts to detect CallerID on PSTN lines? |
20:53.03 | Katty | pbd: i haven't been paying attention |
20:53.05 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
20:53.22 | pbd | pardove: I'm sorry.. I think I missed the first part of your question, but perked up over caller-ID.. |
20:53.29 | Katty | pbd: but that generally happens when you have an IT department to run |
20:53.49 | Katty | pbd: i hope you find your answer. |
20:53.54 | corne | vp7, can you setup callbarring with asterisk? |
20:54.11 | pbd | Katty: You are in a good mood today. My group hasn't been bugging me- they're all waiting for the new laptops I ordered (133 of them) to arrive tomorrow. |
20:54.25 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
20:54.28 | Katty | I'm in a good mood every day. |
20:55.20 | *** join/#asterisk Emrah (n=user@adslgva0491.worldcom.ch) |
20:55.29 | pardove | pbd: it seems that * attempts to detect caller-id just after the 1st ring. but i have a case that Telco sends the caller-id before the 1st ring, so * can't detect it :-( |
20:55.37 | *** join/#asterisk Ash (i=aaron@outofband.org) |
20:55.38 | Seyr | Anyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes? |
20:55.39 | pbd | Katty: Now, am I wasting air here, or can I now apologize for (I think) somehow ticking you off over your resume, and perhaps find out if you at all know of a resource, or have gotten click to print working under samba before? |
20:55.41 | Ash | yowsa |
20:55.49 | Emrah | Hello! |
20:55.51 | Ash | hello there |
20:56.08 | pbd | pardove: ahh.. now I have to remember.. I've read on that one before. |
20:56.19 | vp7 | corne: Explain plz what does "callbarring" means. I don't know such an english word (english is not my native) |
20:56.20 | pbd | pardove: Are you in the US? |
20:56.27 | pardove | pbd:: any solution? |
20:56.30 | *** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
20:56.34 | pbd | pbd: And you're using POTS lines, not PRI, correct? |
20:56.48 | pardove | pbd: yes i'm using POTS lines |
20:56.49 | Katty | pbd: Yes, you did tick me off. |
20:56.59 | Katty | pbd: To be honest, I'm still rather ticked off about it too. |
20:57.05 | Katty | pbd: You're very presumptious. |
20:57.10 | Emrah | I've made some searches on the Internet, particularly in the Asterisks mailinglists, but I did not find any answer for my problem. I'm having a problem, perhaps with the motherboard, perhaps with the 4xdigium card... May some have have a look at this for me? |
20:57.23 | Ash | Has anybody had an issue with their TDM400P not being recognized by asterisk? |
20:57.24 | Katty | pbd: And to be frank, you're wasting your breath with me. |
20:57.26 | Emrah | I will publish the error message in pastebin |
20:57.27 | pbd | Katty: Actually, I'm not- just busy. But I'm sorry if it offended you. |
20:57.40 | Katty | pbd: let's hope it doesn't happen again, eh? (= |
20:57.45 | syzygyBSD | Ash: do you have the modules loaded? |
20:57.51 | Ash | ztcfg -vvv shows the channels as being configured just fine, and there are no errors in the modprobe |
20:58.01 | ender | 30~ IP-301s showed up today, so now I have to configure them alll. |
20:58.06 | Uther_P | *rolls eyes* |
20:58.20 | pardove | pbd: do you have any solution? |
20:58.21 | Uther_P | ever the drama major |
20:58.23 | ender | people see the phones w/ their names on it, and want them NOW. |
20:58.25 | Ash | syzygyBSD: I get this error: WARNING[4607]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device |
20:58.29 | Katty | Uther_P: naturally, i'm female! |
20:58.30 | Ash | and a few after that |
20:58.36 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
20:58.37 | pbd | pardove: You said POTS.. but are you in the US, or are we talking a non-US standard CID spec? |
20:58.37 | Katty | Uther_P: it's /normal/ for me (= |
20:58.44 | Uther_P | I know man that dont do that, heh |
20:58.45 | Ash | it's a TDM400P with two FXO modules on it |
20:58.47 | Uther_P | err many |
20:59.16 | harryvv | traditionally what cable type is T-1 or T-3 normally? |
20:59.17 | pardove | pbd: i'm not in US i have this issue in germany |
20:59.27 | Ash | harryvv: T1's are usually RJ45 |
20:59.29 | pbd | Katty: Erm, well- seems I could accuse you of also being at least one who's quick to judge- but hey. If you don't know the answer, or aren't willing to help, so be it. :) |
20:59.34 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
20:59.34 | corne | vp7,call barring is when you want to restrict an extention to only be able to dail local numbers instead of national or international numbers |
20:59.40 | harryvv | ash, rj45 is the connector. |
20:59.54 | Ash | harryvv: well yeah. I thought that's what you mean, sorry :) |
20:59.54 | harryvv | for cat5 cable |
21:00.08 | ender | harryvv: not just cat5 |
21:00.17 | Ash | all my T1's use UTP with RJ45 ends from where they come out of tthe demarc |
21:00.22 | pbd | pardove: Ok, then. Now that I'm thinking about it, I think the issue I read up on was the converse- because they didn't have a wait() int the dialplan, it was not picking up the caller id, which came in on ring two. |
21:00.26 | ender | harryvv: rj45 is a connector type. Any number of cables could be terminated into that connector type. |
21:00.37 | harryvv | I suspect its serial based cable such as those used to hook to the DTE DCE connectors on cisco routers. |
21:00.40 | harryvv | But dont know. |
21:00.58 | Ash | harryvv: you can use a plain old ethernet cable for T1, dude. |
21:01.03 | harryvv | I see |
21:01.24 | vp7 | corne: you want not to let some of your internal users to make international calls? it's easy - just set to them context that have no rules for international dialing :) |
21:01.30 | pbd | pardove: Your case is that it's coming in too early.. which is weird. I'd more suspect a protocol incompatibility. What is your PSTN interface? Digium card? |
21:01.43 | *** join/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
21:01.43 | *** mode/#asterisk [+o twisted|astricon] by ChanServ |
21:02.06 | corne | vp7, thanx |
21:02.07 | pardove | pdb: i'm using TDM400P |
21:02.57 | pardove | pdb: but a caller-id detection device catchs the caller-id on the this line |
21:03.03 | Emrah | May someone have a look at this? http://pastebin.ca/25423 |
21:03.22 | syzygyBSD | I think that is one of the cards I have in my system, also have a T100P and a TE100P, they all work fine |
21:05.06 | pbd | pardove: Right, but it's possible the digium card isn't configured to read your signalling method. |
21:05.06 | pbd | (or the tdm400 doesn't have the ability to read it) |
21:05.07 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
21:05.46 | pardove | pdb: i've also tested it with TE110P and TA750. no success! |
21:06.17 | Emrah | May someone have a look at this? http://pastebin.ca/25423 |
21:06.34 | pardove | pdb: and also tested with all cidsignallings |
21:06.35 | *** join/#asterisk [hC] (n=hardcore@dsl001-136-136.lax1.dsl.speakeasy.net) |
21:06.54 | pbd | pardove: That's indicative. Did the 110P show anything for caller ID when you debugged the span? |
21:06.59 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
21:07.33 | pbd | If the 750 wasn't able to decode it, it's possible that your telco is using a signalling method that's too unique for the cards to handle. |
21:08.03 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
21:08.24 | pardove | pdb: i didn't do that. as i know channelbank just passthroughs everything, am i right? |
21:09.25 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
21:09.56 | pbd | Pardove: Well.. maybe. Keep in mind, the 750 is going to hand you ISDN PRI, which communicates all the stuff digitally- it uses IE messages to pass things like caller-id. An analog POTS line doesn't work that way.. its more like a cheapo modem sending the signal in a frequency range that's hard to hear, between the rings. So the 750 has to catch it one way, and re-encode it to the other. |
21:09.57 | Samoied | hello all |
21:10.18 | Samoied | I have a problem with authentication i asterisk |
21:10.41 | Emrah | Anyone has an idea to help me finding where is the problem comming from? |
21:10.54 | Samoied | I want to use one userid <From> and other authid <Authorization> |
21:11.13 | denon | does anyone know if there's a changelog or other details on adtran 750 firmware updates? |
21:11.16 | Samoied | but the asterisk always use From header, not auth |
21:11.17 | denon | I thought there used to be |
21:11.19 | pbd | Emrah: It can come from all sorts of places- mostly bugs in Asterisk. |
21:11.53 | Emrah | But the card has just crashed yesterday and now it's really strange |
21:12.00 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
21:12.13 | pbd | Emrah: I see it occasionally on my h323 channels, if the gatekeeper wigs out in *just* the right way. |
21:12.15 | Emrah | Is there anyone to catch how is the connection between the card and Asterisk? |
21:12.23 | Emrah | is there any way* |
21:12.49 | Ash | So has anybody had an issue with the TDM400 channels not showing up? |
21:12.55 | Ash | log output is here: http://pastebin.com/392749 |
21:13.00 | pbd | Emrah: I'm not sure what you mean by 'catch the connection'. You can look for interrupt conflicts, etc, by doing a 'cat /proc/interrupts'.. or check your dmesg log. |
21:13.02 | Ash | this is with the kernel modules loaded |
21:13.15 | pardove | pdb: when i connect the caller-id detector to the line, it shows the caller-id just before the first ring. does it give any idea? does * or CB try to detect CID before 1st ring? if CID detector shows cid so telco CID sig. is known, isn't it? |
21:13.56 | pbd | pardove: It's known to the makers of your CID detector device- but I'll bet that a similar device from the US won't catch anything. |
21:13.59 | pardove | pdb: what is the difference between cid detector and *? |
21:14.08 | pbd | pardove: Signalling, mostly. |
21:14.53 | pbd | There's a lot of different ways to pass CID.. timing, protocol, frequency ranges, etc. And there's a few different standards. In the US, we use 'bellcore' standards.. but that's not true everywhere. |
21:14.56 | pardove | so we can add more cid sig. standards to *? is there any ref. for all known cid standards? |
21:15.12 | Emrah | Thanks a lot pbd , I will try |
21:15.27 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
21:15.56 | pbd | pardove: Big maybe there. Depends on the card drivers as to whether it will catch it at all. You'd be best off talking to Digium or Sangnoma directly on that one. |
21:16.32 | *** part/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com) |
21:16.51 | Seyr | Anyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes? |
21:17.33 | pbd | Pardove: the sipura-spa3000 I'm playing with now supports 8 different CID standards.. some of which send before 1st ring. |
21:17.53 | pbd | I'd be surprised if the Digium hardware didn't support the same standards.. but not *Very* surprised. |
21:17.54 | pardove | pbd: to me it seems that it is a generic problem to chan_zap or ... because i have the same prblem with TDM400P, TE110P TE405P and A104... |
21:18.21 | pbd | pardove: Not necessarily chan_zap- it depends on what the card passes up to it. |
21:18.24 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
21:19.50 | pardove | pbd: do you have any ref./link to CID signalling types? |
21:19.52 | pbd | pardove: Check this page out, it might help you. http://artofhacking.com/files/callerid/CLI_FAQ.HTM |
21:21.20 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
21:21.38 | pardove | pbd: thanks |
21:24.16 | *** join/#asterisk bongfrog (n=winston@dsl001-136-136.lax1.dsl.speakeasy.net) |
21:24.29 | HiltonT | love the nick :) |
21:25.38 | *** join/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com) |
21:26.06 | Trionnis | so does nufone ever answer their sales email ? |
21:26.12 | Trionnis | :) |
21:27.34 | pardove | pdb: just to make sure. does * really support these 3 cid sigs: bel202, v23, dtmf? |
21:28.09 | FuriousGeorge | so how many telephones worth of voltage do these ATA devices support? |
21:28.43 | FuriousGeorge | pbd: i went with the viking k-1700, and a c-1000 to unlock the doors from extensions |
21:30.01 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
21:30.40 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-82.cybersurf.com) |
21:31.23 | Katty | anyone want to quote me a server? |
21:31.29 | Ash | "server" |
21:31.33 | Katty | a /windows/ server |
21:31.37 | Katty | ;) |
21:31.39 | syzygyBSD | lol.. |
21:31.46 | denon | sure, 1 nice quad opteron for $25k |
21:31.51 | syzygyBSD | Katty: what is it going to be used for? |
21:31.59 | Katty | syzygyBSD: well it's windows. |
21:32.02 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
21:32.02 | Katty | syzygyBSD: so obviously a paperweight. |
21:32.05 | Trionnis | not a lot then? |
21:32.08 | syzygyBSD | how about you save your time and just burn the money now |
21:32.11 | Trionnis | dang, beat me to it |
21:32.17 | Katty | ;) |
21:32.22 | Katty | i'll take that as a no |
21:32.25 | Katty | kthxbi. |
21:32.38 | syzygyBSD | lol.. i run a couple of them, just prefer linux |
21:33.32 | FuriousGeorge | dualcore athlon64, 1 gig of ram, 4 x 250 gb sata 1,300 or so |
21:33.49 | FuriousGeorge | thats with a highend mb |
21:33.56 | ender | Katty: what specs? |
21:34.15 | Katty | ender: pretty and black. |
21:34.15 | ender | Katty: Pogo Linux makes great hardware and they can ship/support Windows too. We bought our * servers from Pogo. Very satisfied. |
21:34.22 | ender | Katty: and they're black. |
21:34.25 | denon | I wouldnt recommend an ath64 on a server .. |
21:34.28 | denon | that's what opterons are for |
21:34.29 | denon | of xeons |
21:34.32 | Katty | ender: hot. |
21:34.58 | Katty | i'm sure i'll build a xeon server. |
21:35.05 | denon | we sell a ton of em |
21:35.08 | Katty | i was just teasing about the quote. heh |
21:35.09 | ender | denon: dual-core Opterons are quite nice. |
21:35.10 | Assid | how much are opterons going for now adays |
21:35.10 | denon | way more than opteron |
21:35.17 | fifer | We have been going with Dell 1850 recmount 1u servers. Under $2k |
21:35.27 | Katty | :< |
21:35.33 | FuriousGeorge | denon: i just happened to have that quote for a network cam server |
21:35.35 | Katty | those are noisy. |
21:35.52 | syzygyBSD | get water cooling |
21:35.54 | Katty | every single 1u i see sounds like an airplane. |
21:35.54 | denon | FuriousGeorge: what kinda gear? |
21:36.11 | ender | Katty: hehe, our 1us aren't bad. But they're just P4 Prescotts w/ 2x Sata disks. |
21:36.12 | denon | Katty: not so much like a plane .. but more like a jet engine :) |
21:36.29 | FuriousGeorge | denon: snc-df70 sony cameras |
21:36.38 | Seyr | Anyone know why I cannot get MWI on a 7960 using Realtime with rtcachefriends=yes? |
21:36.40 | denon | FuriousGeorge: eth-attached? |
21:36.43 | FuriousGeorge | 1,800 with dual 17" lcd display |
21:36.50 | FuriousGeorge | denon: yeah all PoE |
21:36.57 | denon | FuriousGeorge: 1800 cams? |
21:36.57 | FuriousGeorge | no electrician required |
21:36.58 | *** join/#asterisk loick (n=loic@APuteaux-151-1-46-19.w82-124.abo.wanadoo.fr) |
21:37.00 | FuriousGeorge | no |
21:37.15 | FuriousGeorge | cams cost me 850 each, the server for 8 was what i quoted before |
21:37.20 | denon | ah |
21:37.23 | denon | what software? |
21:37.34 | FuriousGeorge | add 1400 for sony realshot |
21:37.48 | denon | been looking for good applications, open source would be nice, to handle commercial security cam installs |
21:37.52 | Assid | whats better? opteron 242 or athlon 64 3400 |
21:38.02 | denon | Assid: depends what you're doing with it |
21:38.05 | Katty | what's better? |
21:38.07 | Katty | a nice salad. |
21:38.10 | Katty | and hugs. |
21:38.20 | denon | depends how cute she is .. |
21:38.23 | denon | the one you're hugging |
21:38.24 | FuriousGeorge | denon: there are two that ive found that work with sony in mpeg4, sony software (of course) and d3data, which is actualy more $ |
21:38.30 | Katty | denon: heh. |
21:38.50 | FuriousGeorge | Katty: whats ur pretty black server gonna serv? |
21:38.56 | Assid | denon: video encoding.. gaming.. everyday shit.. |
21:38.56 | Katty | FuriousGeorge: hot chocolate, obviously. |
21:38.57 | denon | FuriousGeorge: i dont really care what transport/codec/etc they use .. usb cams would be nice, but not necessary .. and not practical anyway I guess |
21:39.06 | denon | Assid: think I'd go opteron then |
21:39.09 | Katty | FuriousGeorge: it's just gonna be a file server |
21:39.17 | denon | Assid: though the ath has some advantages .. you wont need 64 anyway |
21:39.21 | Assid | how do they differ? |
21:39.35 | FuriousGeorge | denon: i dont think theres a such thing as commercial usb camera security systems |
21:39.38 | denon | cache and instructions |
21:39.43 | denon | FuriousGeorge: nope, not really |
21:40.09 | Katty | FuriousGeorge: with 2 300 gig hds and a raid1 |
21:40.10 | denon | FuriousGeorge: sony's app any good? it comes with the cams? |
21:40.11 | FuriousGeorge | denon: mpeg4 is good b/c you can get good quality at 2kB/s which makes recording for days (i.e. commercial application) feasable on less space |
21:40.32 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
21:40.33 | generalhan | anyone know of some really good IVR software that i can use with linux ? |
21:40.42 | FuriousGeorge | Katty: how much fileserving will it do at peak hours |
21:40.44 | denon | generalhan: uh, asterisk? |
21:40.50 | Katty | FuriousGeorge: haha |
21:40.53 | shido6 | anyone know how to install Universal.pm ? |
21:40.54 | Katty | FuriousGeorge: hahahahaha |
21:40.57 | Katty | FuriousGeorge: peak? |
21:40.57 | generalhan | dennon: LOL ! lemme explain |
21:41.00 | Katty | FuriousGeorge: 10 people! |
21:41.01 | FuriousGeorge | denon: sony sw that comes with a camera just drops file on an ftp |
21:41.15 | FuriousGeorge | 10 people downloading 10 gigs or 10 megs? |
21:41.23 | Katty | more like 5 megs. |
21:41.23 | denon | FuriousGeorge: and you can easily view lots of cameras, navigate archive footages, etc right from an app? |
21:41.33 | HiltonT | then any box will do - its way small |
21:41.42 | *** part/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com) |
21:41.46 | generalhan | my boss wants, instead of people leaving messages, wants to set up a VR system that will prompt the user for their name and phone number then log those responses to a database, or even just a spreadsheet |
21:41.54 | FuriousGeorge | denon: for "easily manage multiple cameras" ur gonna want the $1400 real surveilance recording sw |
21:41.54 | HiltonT | cheap HP or Dell box, depends on what u prefer |
21:42.03 | HiltonT | or a whitebox (build it yourself) unit |
21:42.15 | denon | FuriousGeorge: from d3data? or from sony? |
21:42.22 | FuriousGeorge | denon: other cameras are compatible with cheaper sw, but they dont do PTZ outside, with only PoE for less than a grand, if at all |
21:42.34 | FuriousGeorge | denon: havent used either |
21:42.46 | HiltonT | I like the LOOK of the panasonic "HAL" camera, not seen it in action, tho |
21:42.47 | denon | FuriousGeorge: well, I mean which is the 1400 |
21:43.16 | FuriousGeorge | Katty: piii1ghz 256 mb of ram, on 100baseT network. that might be overdoing it i dunno |
21:43.19 | pardove | just to make sure. does * really have full support of these 3 cid sigs: bell202, v23, dtmf? |
21:43.42 | HiltonT | not the 256MB - that's not enough for Windows |
21:43.48 | FuriousGeorge | sony=1400 for 8 camera licence. d3data > 1700 for 10 licence (less is a 5 licence) |
21:44.05 | pardove | just to make sure. does * really have full support of these 3 cid sigs: bell202, v23, dtmf? |
21:44.10 | denon | hmm .. gets pretty steeap for only 10 |
21:44.20 | denon | -e |
21:44.44 | FuriousGeorge | denon: if ur cameras are gonna be indoor maybe you can get cheaper cameras which are compatible with cheaper software |
21:44.44 | denon | both can control robotics cams? |
21:45.00 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
21:45.10 | FuriousGeorge | denon: i believe so, but the onboard httpd of the cam probably can too |
21:45.18 | harryvv | FuriousGeorge what ccd cameras do you run? |
21:45.26 | denon | FuriousGeorge: yeah, just idiot-proofing for endusers here .. |
21:45.51 | FuriousGeorge | harryvv: i just quoted a guy for snc-df70 outdoor. i think theyre .3 MP |
21:45.58 | FuriousGeorge | remains to be seen if hes serious |
21:46.01 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
21:46.22 | denon | FuriousGeorge: so do they both give a nice mult-cam interface for remote login? |
21:46.30 | HiltonT | FuriousGeorge; seen the Panasonic cams that look like HAL (2001)? Any good? |
21:46.42 | FuriousGeorge | denon: having never used/bought/seen either sw, i cant tell you much. |
21:46.44 | harryvv | Once I buy a BTV cctv capture card going to interface my sony cctv to asterisk. if there is any motion in the backyard when im not here it will triger a script in asterisk to call me on my cell phone. |
21:46.56 | denon | FuriousGeorge: ah. . what'd you say you use? or nothing, just dump to disk? |
21:47.06 | FuriousGeorge | i can tell you ive heard sony support is terrible, and u'd think being 3rd party and so steep, support would be d3data's angle |
21:47.25 | HiltonT | I don't like Sony's non-Professional kit |
21:47.37 | harryvv | Fur, this is a clone sony brand. Runs about 100 dollars for the camera. I dont worry about it to much. |
21:47.38 | HiltonT | and their Pro gear is not in the same ballpark as what we're discussing here |
21:47.46 | HiltonT | I mean pro as in broadcast quality |
21:47.46 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
21:47.46 | denon | "scalable (i.e., a companie with locations " .. it'd be nice to buy from a company that can spell company... |
21:48.06 | FuriousGeorge | harryvv: i was asking someone about streaming the rtp from the camera to * and using libraries to make it oh323 or something |
21:48.13 | FuriousGeorge | not that i could do that |
21:48.22 | HiltonT | we use a lot of Sony Pro gear in TV and it is good. Their consumer gear is not what it used to be 7+ year back |
21:48.28 | FuriousGeorge | HiltonT: never heard of those cams, btw |
21:48.35 | HiltonT | I'll find a link |
21:48.36 | harryvv | i seee |
21:48.45 | FuriousGeorge | HiltonT: or at least they werent what i was looking for when i was looking |
21:48.48 | denon | FuriousGeorge: so do you have suggestions on a cheaper, but relatively secure internal cam? |
21:49.01 | FuriousGeorge | secure as in vandleproof? |
21:49.06 | harryvv | cam that fits inside the pc? |
21:49.06 | FuriousGeorge | vandal proof |
21:49.16 | denon | well .. in a school, but not like inner-city chicago |
21:49.16 | FuriousGeorge | lol |
21:49.24 | denon | well, in lots of schools actually .. |
21:49.25 | denon | but yeah |
21:49.25 | harryvv | denon secure cam? |
21:49.32 | FuriousGeorge | well lit area? |
21:49.38 | denon | relatively |
21:49.41 | denon | halls and such |
21:49.53 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
21:49.56 | harryvv | i think its sad to put cams in highschools. |
21:49.57 | iCEBrkr | ${UNIQUEID:1:-3} |
21:50.06 | harryvv | We never had any |
21:50.06 | FuriousGeorge | i dunno, if ur lighting is good you could use the dlink 5300 |
21:50.07 | iCEBrkr | Shouldn't that grab all but the last 3 characters? |
21:50.23 | Corydon76-home | ACK, no... Do not use a negative length |
21:50.30 | denon | harryvv: yeah .. they're gettin a grant for it .. and they're hoping it'll keep kids from pullin crap .. little bit of an intimidation factor |
21:50.38 | Corydon76-home | Negative offset is fine, though |
21:50.48 | Corydon76-home | ${UNIQUEID:-3:3} |
21:50.50 | harryvv | denon, what do you mean by pulling crap |
21:50.50 | *** join/#asterisk KranZ (n=user@216.16.193.200) |
21:50.52 | HiltonT | a couple here, http://www.smarthome.com/971302.html but not the one I was after... |
21:51.01 | FuriousGeorge | denon: what about dome cameras, arent they all drop ceiling |
21:51.03 | FuriousGeorge | s |
21:51.12 | denon | harryvv: pranks, breaking into lockers .. noogying the little guy .. whatever, I dunno |
21:51.22 | Supaplex | they have crap on a string now? what's next, wagging tails on dogs? |
21:51.25 | denon | FuriousGeorge: yeah, that could be a good option |
21:51.26 | KranZ | poop |
21:51.27 | Corydon76-home | iCEBrkr: why are you trying to break off the last 3 chars of UNIQUEID, anyway? |
21:51.39 | iCEBrkr | Corydon76-home: cuz I don't need the decimal place |
21:51.43 | KranZ | cus he's the ice breaker |
21:51.45 | harryvv | denon i can see for breaking into lockers. But what are the locks for? |
21:51.48 | Corydon76-home | iCEBrkr: that's not a decimal |
21:51.51 | FuriousGeorge | denon: isnt the sony snc-df40 an indoor dome camera with PoE |
21:51.52 | *** join/#asterisk brent21 (n=Brent21@70.88.149.221) |
21:51.59 | Corydon76-home | iCEBrkr: that's how it goes unique |
21:52.09 | denon | harryvv: Im a tech guy, I dont care what they want to use the tech for at their schools.. |
21:52.13 | iCEBrkr | exten => setup,2,SetVar(sequence_id=${UNIQUEID:1:-3}) |
21:52.17 | iCEBrkr | That's what I need to do. |
21:52.18 | denon | FuriousGeorge: no clue |
21:52.19 | Corydon76-home | They are two numbers, together they are unique |
21:52.21 | iCEBrkr | I don't need that shit... |
21:52.22 | *** part/#asterisk brent21 (n=Brent21@70.88.149.221) |
21:52.31 | Corydon76-home | Either one is NOT guaranteed unique |
21:52.38 | iCEBrkr | ah |
21:52.42 | harryvv | denon, same here. been a tech guy since age 6 :) |
21:52.48 | Corydon76-home | iCEBrkr: the first one is just the same as ${EPOCH} |
21:52.49 | FuriousGeorge | denon: if it is PoE you dont need to do electrical work, thats a big savings some times |
21:52.55 | HiltonT | oh, yeah |
21:52.59 | iCEBrkr | Corydon76-home: I was wondering why that looked familiar. |
21:53.03 | denon | FuriousGeorge: looks like a nice cam |
21:53.18 | Corydon76-home | iCEBrkr: it's EPOCH and an incrementing counter |
21:53.20 | HiltonT | even if not PoE, you can use those power injectors and break it out at the camera end |
21:53.29 | denon | FuriousGeorge: dosnt say anything about PiE .. |
21:53.32 | Corydon76-home | iCEBrkr: together, they're unique |
21:53.38 | denon | PoE |
21:53.48 | KranZ | what is a compression codec which is cpu friendly |
21:53.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
21:53.59 | iCEBrkr | Corydon76-home: agreed. I understand. |
21:54.17 | iCEBrkr | But for what I'm using it for ( for now ) I kinda need to get ride of th . |
21:54.21 | Corydon76-home | iCEBrkr: however, if you really, really want to break off everything after the period, use CUT() |
21:54.21 | iCEBrkr | err rid |
21:54.23 | denon | KranZ: ulaw |
21:54.23 | Supaplex | KranZ: a dedicated codec expansion card ;) |
21:54.43 | KranZ | denon: < 64k |
21:54.46 | Corydon76-home | i.e. ${CUT(UNIQUEID,.,1)} |
21:54.54 | KranZ | Supaplex: can u even get those working with asterisk? |
21:54.56 | denon | KranZ: well-configured speex |
21:54.59 | Corydon76-home | because the second portion is variable length |
21:55.02 | denon | FuriousGeorge: Power requirements AC 24 V 50/60 Hz, DC 12 V,POE |
21:55.14 | denon | so guess it is |
21:55.26 | Supaplex | KranZ: no idea, but if I had a free one, I'm likely to try :) |
21:55.39 | KranZ | word |
21:55.40 | denon | FuriousGeorge: cant seem to find pricing on em though |
21:55.44 | *** join/#asterisk Dougnaka (n=Doug@66.236.77.194.ptr.us.xo.net) |
21:55.49 | KranZ | denon: on what |
21:55.55 | denon | snc-df40 |
21:56.01 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-57-74.dialin.kamp-dsl.de) |
21:56.04 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
21:56.09 | Supaplex | wb lilo |
21:56.21 | denon | lilo's come to chew us out for using too much bandwidth again :) |
21:56.26 | Corydon76-home | Actually, both portions of UNIQUEID are variable length... the first portion has just already gotten to its maximum |
21:56.45 | Supaplex | la la la here's more bw down the drain. ;o) |
21:57.01 | KranZ | denon: well the price for the ceiling mount of a snc-df70n is $109 |
21:57.02 | denon | FuriousGeorge: oh. looks like ~$700 |
21:57.12 | KranZ | so its gonna be pricey |
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21:58.08 | denon | dunno if that would fly price-wise |
21:58.10 | KranZ | the df70n is around $800 |
21:58.31 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
21:58.34 | KranZ | givin the higher model number, the df40 might cost less |
21:58.37 | DanielArndt | hi, does anyone here own a German T-Com Eeumex 820 Lan ? and knows how to connect it with two ISDN (cologne) cards in NT mode, so that 4 lines can be used at onces ? |
21:59.09 | denon | probably, df40 is fixed |
21:59.52 | KranZ | denon: there's a snc-df40n for $627 |
21:59.58 | FuriousGeorge | denon: theres a sony scn-df series pdf out there, says all the models are PoE |
22:00.17 | denon | FuriousGeorge: doesnt it seem like something like this should happen for sub-400 bucks? |
22:00.30 | KranZ | http://froogle.google.com/froogle?q=snc-df40n&scoring=p&sa=N&start=50 |
22:00.31 | FuriousGeorge | also says the df40 doesnt have night vision, and is indoor |
22:00.35 | denon | some of those axis ones are fairly cheap |
22:00.47 | *** join/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net) |
22:01.13 | FuriousGeorge | denon: you gotta factor into the cost of the camera the fact that you dont need to submit plans to the city and get a licened electrician to do the wiring |
22:01.27 | FuriousGeorge | and when theres a power outage one battery backup keeps everything going |
22:01.30 | denon | FuriousGeorge: yeah .. |
22:01.32 | FuriousGeorge | while you find a gas gnerator |
22:01.39 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-209.nas28.salt-lake-city1.ut.us.da.qwest.net) |
22:01.51 | denon | FuriousGeorge: these arent real high-security areas, so they dont really care much about the important things :) |
22:02.13 | FuriousGeorge | yeah, but im tlaking about the bottom line too |
22:02.21 | denon | nod |
22:02.25 | denon | they have a staff electrician I believe .. |
22:02.32 | denon | so he'd be the guy dragging cat5 too |
22:02.34 | denon | wouldnt matter much |
22:02.38 | denon | but its still less hassle |
22:02.52 | FuriousGeorge | yeah, he can go back to changing lightbulbs faster this way :) |
22:02.56 | denon | heheh |
22:03.23 | denon | so 1700 + (10*650) would get them a 10 install setup |
22:03.32 | FuriousGeorge | plus you can tell the school: "look you not gonna sit here and watchem, just dumpem on disk and watchem if you need to |
22:03.34 | denon | plus a server |
22:03.52 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
22:03.52 | FuriousGeorge | denon: they need an ftpd to dump the files on |
22:03.56 | denon | FuriousGeorge: so with mpeg4, is it really very i/o intensive? |
22:04.07 | denon | ftpd? figured they stream right to the software. . |
22:04.40 | FuriousGeorge | denon: i think its just a matter of 13 kB/frame with jpg vs ~2kB with mpeg4 |
22:05.02 | *** join/#asterisk Damin_PDA (n=pocketir@19.sub-70-209-70.myvzw.com) |
22:05.08 | Damin_PDA | werd... |
22:05.12 | denon | FuriousGeorge: yeah, but 10-20 cams doing that .. would I need to even consider array speeds? |
22:05.29 | denon | doesnt seem like that's much traffic |
22:05.33 | FuriousGeorge | denon: the cameras are like little computers, they got firmware you log into thats free, that firmware has the ability to send ftp streams of video for free too. its your job to catch it |
22:05.59 | ender | can * play aiff files? |
22:06.02 | FuriousGeorge | denon: no, but if ur recording for security you may want redundancy. liability is expensive |
22:06.02 | denon | FuriousGeorge: but I assume an app like we were talking about before lets you easily see realtime stuff quicker |
22:06.22 | denon | FuriousGeorge: yeah, offsite mirror maybe .. compress and shuffle |
22:06.25 | GXTi | a striping array would be a plus |
22:06.30 | GXTi | or full raid |
22:06.37 | denon | 0+1 would be nice .. but .. |
22:06.38 | FuriousGeorge | denon: exactamundo, control all the cameras, bring up archived video, monitor, from one interface |
22:06.42 | denon | theyd kick and scream |
22:06.46 | GXTi | yeah |
22:06.47 | GXTi | bit pricey |
22:06.52 | GXTi | just a _bit_ ;p |
22:07.00 | denon | FuriousGeorge: so do these apps still hook ftp? or do they just do an ip stream |
22:07.10 | FuriousGeorge | both |
22:07.11 | denon | GXTi: well .. its not pricey if you want your whole array to be 40 meg .. |
22:07.21 | GXTi | denon: flash drive array ;) |
22:07.28 | FuriousGeorge | the ftp is optional, their job is to ipstream |
22:07.31 | denon | FuriousGeorge: both? watching an ftp doesnt seem very realtime |
22:07.33 | denon | ah ic |
22:07.40 | FuriousGeorge | you gotta connect something to it (video server) |
22:07.42 | GXTi | denon: you could tail the videofile on the ftp server |
22:07.51 | FuriousGeorge | to record |
22:08.01 | denon | GXTi: oh yeah, that's real platform-independant :) |
22:08.01 | FuriousGeorge | to view, the client is IE6 |
22:08.16 | FuriousGeorge | dont know about ff/mozilla support |
22:08.17 | GXTi | denon: actually its more of a problem with the player software |
22:08.24 | denon | GXTi: nod |
22:09.50 | FuriousGeorge | but of course, if ur paying 1400 for sw, it can view the cameras, too |
22:14.22 | syle2 | anyone use SER? |
22:14.33 | file[laptop] | yes no maybe so |
22:14.57 | syle2 | i been trying to figure out if it does cdr entries like cdr_mysql from mysql-addons |
22:15.13 | syle2 | how you suppose to know duration of calls |
22:15.17 | file[laptop] | it doesne't. |
22:15.27 | file[laptop] | the accounting module records all the SIP messaging for the calls |
22:15.34 | file[laptop] | then you have to use outside stuff to construct that information |
22:15.53 | syle2 | how do you get duration of call? |
22:16.01 | file[laptop] | you use outside stuff to figure it out |
22:16.10 | syle2 | ie ? |
22:16.21 | file[laptop] | I'm not going to find the software for you |
22:16.26 | file[laptop] | as I don't know it off the top of my head |
22:17.22 | syle2 | i;d rather code my own, just look for references to what variables to use |
22:17.43 | file[laptop] | variables? where? |
22:18.18 | ender | what file format should I make my voice prompts for * to use? |
22:21.06 | *** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
22:21.23 | CrazyYoss | what is a "rate center"? |
22:21.44 | *** part/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
22:21.52 | AsteriskNoob | rate center is usually a city or area in regards to billing |
22:22.11 | CrazyYoss | thank you |
22:22.18 | *** join/#asterisk bongfrog (n=winston@206.165.75.198) |
22:23.06 | AsteriskNoob | rates in boise idaho are different from rates in new york.... yeah it just defines the rate for an area |
22:23.09 | AsteriskNoob | :) yw |
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22:27.45 | *** mode/#asterisk [+o twisted] by ChanServ |
22:29.30 | KranZ | gsm |
22:33.27 | devel | word, people. i looked on the voip-info wiki for a "voicemail reference" (like an end user reference) and came up with nothing.... is there something? |
22:34.16 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
22:36.46 | Supaplex | How can I find the bug int he bug tracking system for this message? http://www.archivum.info/asterisk-dev@lists.digium.com/2005-02/msg00122.html I'm having the same issue. |
22:37.01 | Supaplex | int he/in the/ |
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22:49.15 | syzygyBSD | What? boise? |
22:50.46 | syzygyBSD | devel: is it something like http://voip-info.org/tiki-index.php?page=Asterisk+voicemail... or did you want more of a client's guide to options? |
22:52.21 | syzygyBSD | devel: something closer to http://voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
22:52.44 | devel | right, syzygyBSD, that second one is the stuff. |
22:52.59 | devel | thanks, syzygyBSD (not sure why i didn't find that....) |
22:53.39 | syzygyBSD | probably didn't look up the right voicemail command, ended up with callers being directed to leave a voicemail instead of checking theirs |
22:54.56 | devel | syzygyBSD, nah, i think i just didn't think to look at the command reference for the behaviour. it's been a long day :) |
22:57.15 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
23:02.28 | *** join/#asterisk Starmaker (n=magnus@85.8.2.169) |
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23:11.24 | *** join/#asterisk RazorJack (n=RazorJac@205.211.153.20) |
23:12.18 | RazorJack | Hey Guys, having a problem with jitter..... When someone phones in zaptel -> sip.... I hear them fine on my PAP2-NA, but they get jitter hearing me.... I have no clue where to start debugging it other than the various adjustments I have made in the zap*.conf files |
23:12.23 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
23:15.09 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
23:16.02 | drbrown | has anyone had any probs w/ asterisk T1's & gsm???? |
23:16.17 | drbrown | specifically in version 1.2beta1 |
23:16.25 | RazorJack | asterisk t1's? |
23:16.38 | RazorJack | nope |
23:16.43 | RazorJack | other than my jitter problem |
23:16.45 | drbrown | asterisk + T1 + gsm |
23:16.57 | drbrown | gsm as in voicemail, promtps and codecs |
23:17.09 | RazorJack | nope, but not using 1.2beta1 yet |
23:17.29 | drbrown | everything gsm is distorted |
23:17.41 | drbrown | only over the T1 though |
23:18.02 | drbrown | it's a te110p & a Rhino channel bank |
23:18.15 | *** join/#asterisk wcj (n=wcj@wsip-68-110-129-70.ga.at.cox.net) |
23:18.23 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net) |
23:18.28 | wcj | hello |
23:18.31 | RazorJack | hey |
23:18.39 | SpaceBass | ok... fyi.. #mtyhtv-users ... they are assholes... just had to vent |
23:18.48 | RazorJack | lol spacebass |
23:18.57 | RazorJack | I agree. |
23:18.59 | SpaceBass | every single possable topic is banned |
23:19.16 | RazorJack | I bit the bullet, and bought a wicked media center pc |
23:19.20 | wcj | will someone in here answer some basic questions about pbx's ? |
23:19.23 | Ariel_ | SpaceBass, wow nice hello |
23:19.23 | RazorJack | its awesome, will never go back to mythtv |
23:19.26 | SpaceBass | F that... my HD tivo works just fine... so what if I it doesnt integrate with my * box... not worth being in that # |
23:19.33 | SpaceBass | Ariel_: sorry :) just venting :) |
23:19.35 | RazorJack | wcj, you just joined, didnt see a question from you. |
23:19.48 | Ariel_ | SpaceBass, it's ok I understand |
23:19.51 | wcj | lol, well that was my first question |
23:20.06 | wcj | I don't know anything about pbx's or how to use them |
23:20.10 | Ariel_ | what would you like to know about pbx? |
23:20.13 | wcj | but I was thinking about setting one up |
23:20.14 | Ariel_ | ~docs |
23:20.15 | jbot | i guess docs is 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 |
23:20.21 | RazorJack | Ariel_; you a jitter fixing guru? |
23:20.22 | wcj | well hold on |
23:20.27 | wcj | just some basic questions here |
23:20.27 | Ariel_ | wcj, what do you want to do with it? |
23:20.38 | Ariel_ | RazorJack, no I am no guru |
23:20.50 | RazorJack | Ariel_; hehe, me neither.... |
23:20.50 | wcj | can I create a public voip network? |
23:21.04 | Ariel_ | wcj, great you have come to the right place. |
23:21.05 | RazorJack | wcj; FREE CALLING! WOOT! |
23:21.14 | snitt | lol |
23:21.17 | Ariel_ | RazorJack, what is the problem your having with the jitter? |
23:21.26 | wcj | lol |
23:21.32 | SpaceBass | Ariel_: i was looking for you earlier... had an amp question |
23:21.35 | RazorJack | zap -> sip.... I hear them PERFECTLY... no echo, they get jitter on the PSTN/ZAP side |
23:21.41 | wcj | so this is possible to do then? with only an internet connection? |
23:21.51 | RazorJack | not sure where to start |
23:22.14 | wcj | I think razorjack said it best, I myself am not sure where to start |
23:22.31 | RazorJack | wcj; asterisk@home or amp.voxbox.ca |
23:22.34 | RazorJack | wcj; two good starts |
23:23.08 | wcj | what is asterisk@home ? |
23:23.14 | RazorJack | 1sec, ill google it for ya |
23:23.20 | wcj | don't worry |
23:23.22 | wcj | I can |
23:23.23 | RazorJack | live linux mini cd |
23:23.29 | RazorJack | with asterisk and amp installed on it |
23:23.34 | RazorJack | http://asteriskathome.sourceforge.net/ |
23:24.12 | Ariel_ | it's a self installing asterisk setup not a live cd. It will erase your drive. |
23:24.25 | Ariel_ | but it's the fastest way to get asterisk and a great gui up and running. |
23:24.28 | RazorJack | Ariel_; sorry, ya just saw that |
23:24.39 | SpaceBass | Ariel_: how can I enable call waiting at the user level? seems to be at the device level |
23:24.41 | RazorJack | It was a live cd at one point wasnt it? |
23:24.51 | RazorJack | based on knoppix |
23:24.51 | Ariel_ | SpaceBass, *70 |
23:24.53 | RazorJack | I thought |
23:25.16 | SpaceBass | Ariel_: I did that, and in the CLI it showed the device, not the user |
23:25.17 | Ariel_ | RazorJack, there is a rapid cd which was based on Debian |
23:25.30 | RazorJack | maybe thats it, too many linux distros |
23:25.34 | Ariel_ | ahh you have the device separete from the extensions. |
23:25.41 | wcj | so basically all I have to do is install asterisk and configure it? |
23:25.54 | wcj | then purchase hardware that can interface to it? |
23:26.02 | RazorJack | wcj; installing is easy... configuring... dont call me, ill call you :) |
23:26.07 | wcj | lol |
23:26.09 | wcj | so I've heard |
23:26.12 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
23:26.15 | wcj | but I like a challenge ;) |
23:26.16 | *** join/#asterisk xyharley (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
23:26.33 | RazorJack | wcj; DONT get a clone wildcard x100, biggeste mistake I made learning asterisk |
23:27.00 | wcj | I'll keep that in mind |
23:27.00 | RazorJack | wcj; or a grandstream phone |
23:27.05 | RazorJack | wcj; they belong in the lake |
23:27.26 | wcj | well i might go for a swim if I can find some free ones down there |
23:27.31 | SpaceBass | Ariel_: well I have a device with an extension (6000) and a default user (6010)... |
23:27.35 | wcj | (long as no ameaba attack me) |
23:27.54 | wcj | anyways |
23:27.59 | RazorJack | wcj; if you wanna tweak, sure go for it, but I use my asterisk not for just testing |
23:28.06 | wcj | so I don't have to have any kind of link to a phone company? |
23:28.16 | wcj | I just install and set it up? |
23:28.17 | RazorJack | wcj; http://amp.voxbox.ca/ and http://asteriskathome.sourceforge.net/ |
23:28.23 | RazorJack | wcj; ummm you wanna call ppl right? |
23:28.28 | wcj | yes |
23:28.37 | RazorJack | wcj; skype.... |
23:28.49 | RazorJack | wcj; free world dialup |
23:29.01 | HiltonT | schype |
23:29.06 | RazorJack | wcj; vonage lite |
23:29.12 | wcj | yeah, but then I learn onthing and have nothing new to put on a resume |
23:29.12 | RazorJack | I dunno, depends on your preference |
23:29.28 | HiltonT | if only they complied with standards, maybe eBay's $2.6bn would have made sense |
23:29.41 | RazorJack | lol |
23:30.10 | HiltonT | $2.6bn - I mean, welcome back to the Tech issues a few years ago in the stockmarket! |
23:31.12 | RazorJack | Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the Universe trying to produce bigger and better idiots. So far, the Universe is winning. " |
23:31.25 | HiltonT | it always will! |
23:31.46 | RazorJack | HiltonT; you know anything about jitter? |
23:32.01 | wcj | well thank you guys for your input |
23:32.06 | RazorJack | wcj; good luck |
23:32.21 | wcj | ty |
23:32.28 | HiltonT | RazorJack; I've only started playing with * in the past 24 hours :) |
23:32.35 | RazorJack | give asterisk@home a try |
23:32.46 | RazorJack | HiltonT; I've been way longer |
23:33.10 | wcj | will do |
23:33.12 | RazorJack | HiltonT; no clue where this jitter came from, it just started a month ago after no upgrades |
23:33.50 | wcj | i'm sure i'll see you guys in here again |
23:33.50 | wcj | bye for now |
23:33.51 | SpaceBass | the latest aah seems to have more jitter for me... lot more... but I think its b/c it has more overhead and I have a slow box |
23:33.55 | Ariel_ | RazorJack, your getting a sound that is jitter? |
23:34.04 | RazorJack | Well thats the thing, how do you calculate your overhead.... |
23:34.21 | Ariel_ | RazorJack, what codec are you using how many users? |
23:34.22 | HiltonT | maybe local network bandwidth is less than it was |
23:34.44 | RazorJack | ulaw, 4 users |
23:35.06 | *** join/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net) |
23:35.08 | Ariel_ | SpaceBass, I need to change my setup sometime to device different then extensions to find do some test. But I kinda like the extension being like the devices. |
23:35.25 | Ariel_ | RazorJack, what service do you have dsl/t1 data cable? |
23:35.36 | RazorJack | I just got the PAP2-NA firmware updates from cisco today.... havent upgraded the adapter yet... but it WAS working fine |
23:35.48 | SpaceBass | Ariel_: I just did it to play around... I can easily go back to devices and extensions being the same |
23:36.00 | RazorJack | Ariel_; sip extension 200 -> pap2-na -> asterisk -> zap tdm400 -> bell |
23:40.38 | RazorJack | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P thats the card im using |
23:40.55 | RazorJack | wonder if theres a firmware update or something |
23:45.12 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net) |
23:46.10 | *** part/#asterisk Avero (n=no@dsl001-136-136.lax1.dsl.speakeasy.net) |
23:46.59 | Ariel_ | RazorJack, are you using stable or cvs head? |
23:46.59 | *** join/#asterisk ben_d (n=ben@cpe-66-66-209-96.rochester.res.rr.com) |
23:47.02 | *** part/#asterisk IzNoGooD (n=marc@iznogood.demon.nl) |
23:47.26 | RazorJack | Ariel_; stable |
23:47.42 | RazorJack | just removed couple lines from my zapata.conf |
23:47.49 | Ariel_ | you can use the zaptel drivers from cvs head which have a better echo routing. |
23:47.51 | RazorJack | context=from-pstn |
23:47.51 | RazorJack | signalling=fxs_ks |
23:47.51 | RazorJack | group=0 |
23:47.51 | RazorJack | channel=4 |
23:48.10 | RazorJack | this aint echo, its jitter, or am I missing something? |
23:48.29 | RazorJack | 1.0.9 asterisk |
23:49.14 | *** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net) |
23:50.05 | kb1_kanobe | RazorJack: if you originate the call from the opposite side does it still display that behaviour? |
23:50.11 | RazorJack | yep |
23:50.25 | kb1_kanobe | have you tried dialing into app_milliwatt() from both sides? |
23:50.39 | RazorJack | whoa, no, how do I do that? |
23:50.47 | *** join/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
23:51.06 | RazorJack | just make an extension to app_miliwatt? |
23:51.25 | Ariel_ | RazorJack, like a small buzz |
23:52.07 | kb1_kanobe | RazorJack: yes. It will generate a constant 1004hz signal. If the sip side shows the defect but the zap doesn't, so there is a problem on that path etc. |
23:52.17 | RazorJack | from the sip side, its not coming down to me, its going back jitter |
23:52.24 | *** join/#asterisk carrar (i=tim@osburn.com) |
23:52.26 | carrar | w00t! |
23:52.32 | kb1_kanobe | try zap into app_milliwatt. |
23:52.43 | RazorJack | 1s, ill do it right now |
23:54.06 | RazorJack | exten => *07,1,app_milliwatt() ? |
23:54.34 | RazorJack | im missing something :P |
23:55.00 | kb1_kanobe | try just milliwatt() |
23:55.08 | RazorJack | dont I have to load it in modules.conf ? |
23:55.22 | RazorJack | load => app_milliwatt.so ? |
23:55.25 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
23:55.43 | *** join/#asterisk wcj (n=wcj@wsip-68-110-129-70.ga.at.cox.net) |
23:56.16 | RazorJack | speaker phone on my sip extension 200 |
23:56.20 | kb1_kanobe | here we are: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Milliwatt |
23:56.21 | RazorJack | dialed *07, constant tone |
23:56.27 | RazorJack | no jitter |
23:56.40 | *** part/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
23:56.42 | *** join/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
23:56.44 | RazorJack | gonna phone in on lanline, 1 sec |
23:56.53 | *** part/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
23:57.41 | RazorJack | weird, just dialed from bell line, to my asterisk server and dialed *07, constant 1004 tone |
23:57.43 | RazorJack | no jitter |
23:58.13 | *** join/#asterisk apardo (n=w0w0@1.Red-83-46-192.dynamicIP.rima-tde.net) |
23:58.14 | kb1_kanobe | interesting. |
23:58.47 | wcj | can anyone tell me what asterisk's capabilities are with a single lan line? |
23:59.28 | fifer | Full SIP based phone server |