| 00:00.19 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 00:01.00 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:01.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:20.20 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 00:20.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b2301550d480c2d5) |
| 00:26.21 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:34.20 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 00:38.52 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 00:45.09 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 00:53.11 | hmmhesays | stacey haiduk was hot |
| 01:00.20 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 01:00.59 | Mavvie | I have tested the line and I get a fax tone although there is no dialtone before getting the fax tone. |
| 01:01.18 | Mavvie | I start hating people who know the buzzwords but don't know where to use them. |
| 01:03.02 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:06.09 | *** join/#asterisk exvito (n=exvito@89.181.50.34) |
| 01:06.10 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 01:06.10 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 01:08.35 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:09.07 | asterisknerds | <PROTECTED> |
| 01:10.28 | *** join/#asterisk CVirus (n=GoD@217.54.232.146) |
| 01:16.00 | *** join/#asterisk elixer (n=seanbrig@c-69-251-152-9.hsd1.md.comcast.net) |
| 01:20.00 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
| 01:22.02 | *** part/#asterisk Djeli (n=Djeli@ppp157-243.static.internode.on.net) |
| 01:24.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:26.11 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 01:31.27 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 01:33.23 | *** join/#asterisk Psykick (n=anon@60.234.198.147) |
| 01:33.25 | Psykick | hi guys |
| 01:33.36 | Psykick | having a problem registering the g729 codec |
| 01:34.39 | Psykick | register utility gets as far as asking me if I want to register the key (which I answer yes to) ... and then it just quits |
| 01:35.13 | Psykick | glibc version 2.4 |
| 01:36.04 | exvito | Psykick: do you have public IP conectivity on that system ? |
| 01:36.16 | Psykick | yes |
| 01:37.06 | exvito | ok... I've had trouble in the past trying to reg a non connected system... Unfortunately, by then, Digium did not provide offline means of registration... (and I believe that would still be the case) |
| 01:37.46 | Psykick | hmmm ... ok |
| 01:38.15 | Psykick | would be nice if they provided source so I could try compiling register utility on current platform |
| 01:39.02 | Voicemeup | yeah |
| 01:39.09 | Voicemeup | you need 445 out open on Firewall |
| 01:39.15 | Voicemeup | its sll comm to digium and back |
| 01:39.20 | Voicemeup | ssl i mean |
| 01:40.05 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:40.54 | *** join/#asterisk ptiggerdine (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 01:40.57 | JT | Psykick: haha, source, yeah... right. |
| 01:44.08 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 01:45.53 | exvito | ...the solution for registering should not be based on a bin utility run on the asterisk system. It would be much better if Digium provided a webpage that would supply the REG codes based on whatever info they needed (order #, NIC mac address, whatever...) -- are you listening Digium ? :) |
| 01:46.19 | Psykick | JT: I know they wouldn't ... would mean people would be able to figure out license keys |
| 01:46.41 | hmmhesays | someone is going to hell for this episode of south park |
| 01:47.09 | JT | it seems a bit pointless having G.729 and no Internet |
| 01:47.17 | JT | hmmhesays: err, why? |
| 01:48.11 | hmmhesays | cause it is so wrong |
| 01:48.12 | hmmhesays | lol |
| 01:48.16 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 01:48.21 | hmmhesays | yet hilarious |
| 01:48.23 | JT | what episode? |
| 01:48.28 | exvito | JT: one might use WAN VPN links (low bandwidth), but now public IP access for the voice system... no ? |
| 01:48.29 | hmmhesays | the satans party one |
| 01:48.38 | exvito | now=not |
| 01:48.39 | JT | oh, that one was fairly tame |
| 01:48.47 | JT | also you're implying that hell exists |
| 01:48.53 | hmmhesays | where gacy, bundy and some other serial killer are like the 3 stooges |
| 01:48.57 | JT | exvito: not very common |
| 01:50.58 | exvito | JT: I agree.. but that would depend only on IP connectivity policies, firewalling, security, etc.. |
| 01:51.25 | JT | heh |
| 01:52.30 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:06.26 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 02:09.57 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582342.dsl.bell.ca) |
| 02:12.01 | tengulre | hi,all which asterisk manager is best for beginner ,beside astclient and tribox. |
| 02:13.22 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 02:15.36 | SwK | vi or emacs |
| 02:15.56 | CVirus | Vi |
| 02:16.03 | CVirus | for me at least |
| 02:16.45 | CVirus | SwK: run vimtutor to learn vi |
| 02:16.56 | SwK | i was answering his question |
| 02:17.35 | CVirus | ah .. LOL |
| 02:17.46 | CVirus | SwK: sorry :-) |
| 02:17.55 | SwK | i use vi so much i end up having i and a all over things that are not vi like openoffice and word |
| 02:18.56 | CVirus | hehe |
| 02:20.18 | SwK | what really pisses me off is i'll be doing something in Flex (which uses Eclipse for an ide) and I'll hit end for eol and it'll go to the end of a file heh |
| 02:23.43 | tzanger | what's used for h323 these days? |
| 02:26.15 | *** part/#asterisk exvito (n=exvito@89.181.50.34) |
| 02:28.02 | Weezey | finally wireshark tells all. Okay, I updated to 1.4 and now peers that are set to nat=yes can make calls out, but when they talk directly to the asterisk box, the rtp stream tries to slam into their private IPs instead of their public IPs. |
| 02:28.11 | Weezey | any help? I have to get this fixed asap. |
| 02:29.27 | Weezey | seems to only be a problem with asterisk playback (background, voicemailmain, etc...) musiconhold plays fine. |
| 02:30.14 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 02:30.37 | *** join/#asterisk fholmes (n=fholmes@c-98-198-197-44.hsd1.tx.comcast.net) |
| 02:31.15 | *** join/#asterisk saftsack (n=saftsack@pD9E0530D.dip.t-dialin.net) |
| 02:31.40 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:35.29 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 02:35.52 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 02:36.35 | AJaymn | has someone made a script to run that will download all Asterisk dependents? |
| 02:41.41 | Corydon76-home | AJaymn: perhaps you'd be happier with package management |
| 02:42.17 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 02:42.56 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 02:43.41 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 02:44.00 | Corydon76-home | Evening, bkruse |
| 02:45.33 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-176-115.buff.east.verizon.net) |
| 02:50.33 | bkruse_home | Corydon76-home: hows it goin? |
| 02:52.46 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 02:56.01 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 02:56.55 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-0d69129a834682d9) |
| 02:57.20 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 03:03.45 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 03:05.08 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:05.35 | *** join/#asterisk CVirus (n=GoD@217.54.232.146) |
| 03:11.45 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 03:35.28 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 03:36.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:36.57 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 03:37.10 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:37.34 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:47.17 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-230-107.dsl.irvnca.pacbell.net) |
| 03:47.21 | BSD_Tech | hey guys |
| 03:47.34 | BSD_Tech | I might be wrong this is why I am checking |
| 03:48.01 | BSD_Tech | but is there a way to allow multi devices 1 sip registration |
| 03:48.11 | BSD_Tech | and have it know to ring all devices |
| 03:48.23 | BSD_Tech | or do you have to setup a ring group |
| 03:48.24 | JT | under the same sip username? no. |
| 03:48.49 | BSD_Tech | ok just confirming |
| 03:48.59 | JT | dial can call multiple phones at once |
| 03:52.06 | bkruse_home | Dial(sip/nub&sip/bsdtech&sip/bkruse |
| 03:52.15 | bkruse_home | |30)* |
| 03:52.42 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:52.42 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:53.08 | BSD_Tech | I know that |
| 03:53.25 | BSD_Tech | but I have some one wanting multi devices 1 sip registration |
| 03:53.41 | BSD_Tech | I was just confirming what I told him |
| 03:55.44 | russellb | bkruse_home: you wish :-p |
| 03:55.59 | bkruse_home | :[ |
| 03:56.02 | bkruse_home | I do. lol |
| 03:56.07 | JT | they are on drugs? |
| 03:56.08 | bkruse_home | one day maybe :D |
| 03:58.05 | russellb | JT: huh? |
| 03:58.48 | JT | the person wanting multiple devices on a single sip registration |
| 03:59.31 | russellb | ah. |
| 03:59.43 | russellb | a crapton of people want "shared extensions" |
| 03:59.49 | russellb | which is what they are gettingat ... |
| 04:00.12 | russellb | i was actually thinking of ways to implement it last night when i couldn't sleep :( |
| 04:09.17 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 04:13.55 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 04:18.31 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 04:19.10 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 04:20.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:22.41 | *** join/#asterisk MdeP (n=mdep@205-94-22-190.adsl.tie.cl) |
| 04:24.58 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.180.56) |
| 04:30.36 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:36.52 | *** join/#asterisk CVirus (n=GoD@217.54.232.146) |
| 04:40.46 | Sweeper | that should be done in the dialplan :/ |
| 04:49.26 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-230-107.dsl.irvnca.pacbell.net) |
| 04:52.44 | *** join/#asterisk mtaht4 (n=m@cpe-065-190-150-008.nc.res.rr.com) |
| 04:57.49 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 05:00.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:09.24 | *** join/#asterisk fwp (n=fwp@unaffiliated/fwp) |
| 05:11.40 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 05:16.37 | *** join/#asterisk CVirus (n=GoD@217.54.232.146) |
| 05:31.52 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-134-148.buff.east.verizon.net) |
| 05:33.40 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:35.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:41.08 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:43.18 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:45.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:50.57 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 05:51.26 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 05:51.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:02.03 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 06:09.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:15.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:29.39 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 06:30.39 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:34.57 | *** join/#asterisk saftsack (n=oliver@p54A7F857.dip.t-dialin.net) |
| 06:35.20 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 06:39.01 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:41.22 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 06:41.25 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:43.27 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:44.45 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 06:54.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:56.56 | *** join/#asterisk kv0s (n=kv0s@p4FD2265D.dip0.t-ipconnect.de) |
| 06:57.32 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:58.03 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 06:59.51 | LoneShadow | how to jump from one context to another ? |
| 07:00.49 | LoneShadow | or is that even allowed ? |
| 07:01.17 | kv0s | LoneShadow: You can "include" some other context ... |
| 07:01.24 | SwK | goto(context,exten,priority) |
| 07:01.32 | SwK | or as kv0s said include it |
| 07:02.11 | LoneShadow | I have two project gizmo accounts, for some reason even though I have separate contexts, it always chooses the same |
| 07:02.24 | LoneShadow | wanted to jump to 2nd one based on EXTEN |
| 07:04.44 | LoneShadow | SwK: the goto method, on hangup will it still remain in the 2nd context ? |
| 07:05.03 | LoneShadow | let me try it out |
| 07:05.38 | KpoH | LoneShadow: define "h" exten, it will be executed on hangup |
| 07:06.10 | LoneShadow | I have already defined h on context1 |
| 07:06.14 | LoneShadow | didnt want that to be executed |
| 07:06.43 | KpoH | can you show you diaplan in pastebin? |
| 07:06.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 07:10.58 | *** join/#asterisk humbertopt (n=ss@10001344683.0000074353.acesso.oni.pt) |
| 07:13.45 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 07:15.37 | *** join/#asterisk shtoom (n=shtoom@221-128-190-199.static.exatt.net) |
| 07:17.27 | *** join/#asterisk saftsack (n=oliver@p54A7C084.dip.t-dialin.net) |
| 07:18.09 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 07:18.53 | LoneShadow | KpoH: http://pastebin.ca/655290 |
| 07:19.16 | LoneShadow | It kind of works |
| 07:19.50 | LoneShadow | but once it jumps, it dosnt go to exten => 17471913364,1,Wait(2), instead goes to h rule with priority of 1 |
| 07:21.37 | yonahw-work | LoneShadow: what is the verbose output as the call runs? |
| 07:21.55 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:22.44 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 07:23.16 | LoneShadow | yonahw-work: http://pastebin.ca/655295 |
| 07:23.25 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 07:23.33 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:23.56 | *** join/#asterisk lbow (n=lbow@dsl-146-4-61.telkomadsl.co.za) |
| 07:24.21 | lbow | Hi! B410P card users around? |
| 07:25.00 | lbow | Anybody seen this failure - stacks of kernel messages saying "mISDN_rdata: rport queue overflow 256/256 [addr:52010201 prim:120282 dinfo:ffffffff]" |
| 07:25.25 | yonahw-work | LoneShadow: I'm not sure this will solve your problem, but try taking out the spaces in your goto |
| 07:25.31 | LoneShadow | aah |
| 07:26.01 | LoneShadow | that did the trick :D |
| 07:26.06 | LoneShadow | thanks :) |
| 07:26.07 | yonahw-work | perfect |
| 07:26.13 | yonahw-work | :) |
| 07:26.43 | LoneShadow | changed it "s" instead of the number, and it still works :) |
| 07:27.36 | LoneShadow | any of you folks played with sphinx ? |
| 07:28.24 | yonahw-work | LoneShadow: that sounds a little dangerous to me with a Callback (using the "s" extension that is) |
| 07:28.31 | *** join/#asterisk umanghc (n=Umang@ool-182fface.dyn.optonline.net) |
| 07:29.00 | LoneShadow | why so ? |
| 07:29.12 | umanghc | hey peopel |
| 07:29.40 | LoneShadow | I will be using it as "exten => 17471912345,1,Goto(custom-gizmo-us-callback,s,1)" |
| 07:29.53 | umanghc | whats the best way to get * workin with ruby on rails |
| 07:30.04 | LoneShadow | whats ruby on rails ? |
| 07:30.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:30.18 | yonahw-work | yes but if for some reason you overlook something and randomly send to the callback context whoever you sent there will be able to make calls on your dime |
| 07:30.20 | snuff-work | if u want AGI for ruby..? |
| 07:30.37 | snuff-work | check voip-info.org |
| 07:30.41 | yonahw-work | and seeing as it seems that you are only planning on using it if a specific number was dialed, why not close the door |
| 07:30.45 | SwK | someone wrote a whole agi ami dialplan etc forusing ruby for asterisk |
| 07:30.46 | snuff-work | not sure if it exists |
| 07:30.56 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 07:31.01 | LoneShadow | yonahw-work: ah ok |
| 07:31.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:31.40 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 07:31.43 | snuff-work | umanghc, http://www.voip-info.org/wiki-Asterisk+AGI |
| 07:32.11 | snuff-work | all sorts of diff AGI backends.. ruby/perl/c/php |
| 07:32.22 | LoneShadow | yonahw-work: thanks :D |
| 07:32.27 | umanghc | is RAGI the best way to go? |
| 07:32.54 | LoneShadow | I just configured flite and sphinx |
| 07:32.55 | SwK | http://adhearsion.com/ |
| 07:32.58 | SwK | for the ruby guy |
| 07:33.01 | LoneShadow | now not sure what to do with them :/ |
| 07:34.23 | umanghc | SwK: thanks |
| 07:36.00 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 07:40.44 | *** join/#asterisk harryk (n=harry@213.156.75.36) |
| 07:41.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:42.57 | *** join/#asterisk saftsack (n=oliver@p54A7D508.dip.t-dialin.net) |
| 07:45.02 | *** join/#asterisk harryk (n=harry@harry.org.ua) |
| 08:00.16 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:02.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:03.22 | tengulre | hi,all |
| 08:03.26 | tengulre | anybody here? |
| 08:05.45 | umanghc | tengulre: hey |
| 08:05.55 | KpoH | can I Dial(SIP/) like IAX with user:pass to authethicate on remote side? or only register option in sip.conf aviable? |
| 08:05.57 | tengulre | :-D |
| 08:06.01 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 08:06.09 | Uatec | \o/ |
| 08:06.29 | tengulre | I want write a softphone, anybody can give me some tips? |
| 08:06.45 | Uatec | 1) Learn a programming language |
| 08:06.53 | umanghc | asteriskNOW is just for linux? I'm trying to get Adhearsion on a Mac OS X box |
| 08:07.26 | umanghc | SwK: Does Adhearsion work on Mac |
| 08:08.00 | KpoH | tengulre: there is already nice solutions like twinkle exists, for what reason you want another softophone? |
| 08:08.21 | tengulre | kwinkle? |
| 08:08.53 | KpoH | twinkle |
| 08:09.05 | tengulre | KpoH: I using microsoft windows . ;( |
| 08:10.42 | *** join/#asterisk sacitec (n=tobi@189.129.221.82) |
| 08:10.54 | KpoH | tengulre: google for softphones, I suppose you will be surpised of all this phones :) |
| 08:12.47 | JT | Uatec: so setup your irc client properly? |
| 08:13.27 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:13.30 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:14.55 | creativx | morning JT :) |
| 08:16.09 | JT | hi |
| 08:16.28 | Uatec | I swear, i spend more time configuring software than using it. |
| 08:16.51 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:17.11 | JT | irssi requires minimal configuration to operate properly :) |
| 08:17.38 | tengulre | morning JT :) |
| 08:17.48 | creadurex | wtf |
| 08:18.08 | JT | hello |
| 08:18.19 | LoneShadow | I have configured text-to-speech and voice recognition on my asterisk. now I dont know what to do with it |
| 08:18.29 | LoneShadow | I guess it was fun setting things up :P |
| 08:20.03 | shtoom | LoneShadow : r u using lumenvox ? |
| 08:20.07 | Aurs | if I have 5 asterisk pbx'es (regpbx1-5), how can I create a single point of entry for sip registrations? (that will be "distributed" to my 5 regpbx'es) can this be done with dundi? |
| 08:20.12 | LoneShadow | shtoom: nope |
| 08:20.32 | creadurex | morning Aurs |
| 08:20.44 | Aurs | mornings creadurex |
| 08:20.46 | LoneShadow | shtoom: I am using sphinx2 |
| 08:20.52 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 08:21.21 | shtoom | LoneShadow:Oh i c is it a free solution ? |
| 08:21.27 | LoneShadow | yes |
| 08:21.33 | shtoom | cool |
| 08:22.02 | shtoom | how is performance is it working properly ? |
| 08:22.14 | LoneShadow | its not so bad |
| 08:22.23 | LoneShadow | probably decent for a personal asterisk setup |
| 08:23.03 | LoneShadow | may need to use something better for a business solution |
| 08:24.15 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.181) |
| 08:24.33 | sacitec | hi |
| 08:24.44 | sacitec | talking about bussiness solution |
| 08:24.55 | sacitec | which one do u recommend for iax ? |
| 08:25.26 | LoneShadow | err, I am just a hobbyist |
| 08:25.54 | LoneShadow | I use asterisk for my personal home phone |
| 08:26.36 | sacitec | i've been testing firefly under win |
| 08:27.13 | sacitec | it looks fancy, but it's no so good as xlite, but it has a more simple interface and a nice gui |
| 08:28.05 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 08:30.23 | LoneShadow | dont use softphones, except when I was trying to learn asterisk :D |
| 08:35.02 | Aurs | what about idefisk from asteriskguru? that softphone supports iax |
| 08:35.17 | Aurs | haven't tested it myself, though |
| 08:36.01 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:38.58 | LoneShadow | idefisk works, had tried it out once |
| 08:43.17 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-60-207.w82-120.abo.wanadoo.fr) |
| 08:46.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:48.38 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:01.54 | Uatec | hey, i've setup a new asterisk box and i don't have any of the debug data i used to get on my old one |
| 09:02.05 | Uatec | i'm dialling a number and getting "call ended" immeidately, from my phone |
| 09:02.10 | Uatec | but nothing's coming up in the CLI |
| 09:02.17 | Uatec | how can i get more info about why it's failing? |
| 09:02.40 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:03.03 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 09:03.26 | Aurs | Uatec: set a higher verbose level in cli? (set verbose 10) |
| 09:04.11 | floppp | Uatec: It's a SIP phone ? |
| 09:05.14 | deegan | Hi, we just ported a heap of phonenumbers to our current SIP provider (about 120 numbers) and instead of them routing it in on our current one trunk (we just want one number to show anyway) we got +110 SIP accounts to register. Now, if this is all i got to work with, what would be the best way to get them into the asterisk without to much hassle. do i need to make 110 new trunks or cant i just use register all of them? |
| 09:05.36 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 09:05.53 | Uatec | ahh, verbose... ty |
| 09:06.08 | Uatec | floppp, ones a sip phone, ones an iax trunk |
| 09:06.08 | JT | Aurs: OpenSER |
| 09:06.10 | Uatec | ahah: Aug 13 10:05:38 WARNING[13307]: chan_iax2.c:7140 socket_read: Call rejected by 192.168.232.157: No authority found |
| 09:07.02 | JT | deegan: a database probably |
| 09:09.15 | Aurs | JT: bah.. :P |
| 09:09.36 | JT | Aurs: you asked |
| 09:09.38 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 09:09.39 | Aurs | hehe |
| 09:09.55 | Aurs | I know, I know.. |
| 09:10.49 | tzanger | hmm |
| 09:11.00 | tzanger | bank of america says there's some irregular activity on my card |
| 09:11.09 | tzanger | I'd have to agree... I don't have a card with them :-) |
| 09:11.19 | Aurs | JT: how much do you want for a working openser.cfg? ;) |
| 09:11.26 | HarryR | `lol |
| 09:11.44 | JT | heh, i don't do openser consulting at this stage :/ |
| 09:11.48 | Aurs | hehe |
| 09:12.28 | Aurs | I guess there are thousands of companies that use openser for this purpose, but noone wants to share their config |
| 09:12.51 | HarryR | Aurs, what kinda stuff are you need with openser? |
| 09:13.15 | Uatec | I'm getting no authority found there, when i dial an iax2 call |
| 09:13.18 | Uatec | to another asterisk box |
| 09:13.30 | Uatec | how can i specify the authority that this iax user must have? |
| 09:14.06 | Aurs | HarryR: I want a single point of entry for sip registrations... but I want UAs to register to asterisk boxes. and I want client A to always register to pbx1, and client B to pbx2, etc |
| 09:14.38 | Aurs | but at the same time, I want clients to have the same config (register to same domain/ip) |
| 09:15.21 | HarryR | is there a reason specifically that you need to register with the Asterisk boxes, it'd be 100x simpler if you didn't need to |
| 09:15.27 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 09:16.04 | HarryR | We're thinking of doing something like that ourselfs, but it's just in the planning stage - not really sure if we need to do it |
| 09:16.28 | HarryR | mainly boils down to using products developed out-of-house which we need to scale up |
| 09:16.45 | Aurs | HarryR: we should put our heads together then. we have tried a few different approaches.. right now we're trying to register to openser, but cannot get polycom phones to work 100% |
| 09:17.05 | *** join/#asterisk KermitTheFragger (n=siepkes@53571672.cable.casema.nl) |
| 09:17.13 | Aurs | (and use pbx1,pbx2 etc as proxy) |
| 09:17.59 | HarryR | what doesn't work about the polycoms? |
| 09:18.05 | Aurs | inbound |
| 09:18.15 | Aurs | (nat) |
| 09:18.17 | HarryR | at all ? |
| 09:18.33 | Aurs | but if we register polycom directly on a asterisk pbx, it all works |
| 09:18.58 | HarryR | but if you register with openser & get asterisk to pass the call to extension@openser.example it doesn't work? |
| 09:19.29 | Aurs | not inbound to polycom phones, but outbound works |
| 09:19.55 | HarryR | you sure packets are even getting to the polycoms? |
| 09:20.01 | Aurs | it rings |
| 09:20.10 | HarryR | ooh right, no media |
| 09:20.12 | Aurs | but we cannot pickup |
| 09:21.04 | Aurs | when we press answer on the polycom, the ringing stops, but the display still sais "answer", so there is some kind of nat issue. haven't really tried so hard to work that one out, because we're working with other issues as well |
| 09:21.47 | HarryR | grab a quick sip trace some time and post it up on the openser mailing list :) |
| 09:24.29 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 09:24.54 | HarryR | Aurs, you need authentication on asterisk - souly for knowing who's making the call right? |
| 09:25.26 | Uatec | hey |
| 09:25.53 | Uatec | http://rafb.net/p/sHhh7187.html <-- here, i'm trying to dial 763 from a sip phone on one asterisk box, to a sip phone on another asterisk box |
| 09:26.10 | Uatec | but when I dial from the first box, i get: |
| 09:26.30 | Uatec | Aug 13 10:26:14 WARNING[13307]: chan_iax2.c:7140 socket_read: Call rejected by 192.168.232.157: No authority found |
| 09:26.34 | Uatec | on the first box |
| 09:26.57 | Uatec | and: Aug 13 10:26:36 NOTICE[22210]: chan_iax2.c:6904 socket_read: Rejected connect attempt from 192.168.232.128, who was trying to reach '763@internal' |
| 09:26.58 | Uatec | on the second |
| 09:27.05 | Uatec | why is it rejecting the call? |
| 09:27.13 | Uatec | from what i can see, it has all the authority it needs |
| 09:28.17 | Aurs | HarryR: openser has a friend in sip.conf |
| 09:28.36 | HarryR | ah fair enough |
| 09:29.27 | Aurs | so our clients register to openser.. and when they send an invite, openser routes it to a asterisk pbx |
| 09:29.46 | Aurs | to the "correct" asterisk pbx |
| 09:30.06 | HarryR | but.. you need to register to the asterisk pbx why? |
| 09:30.06 | Chris-NB | hi |
| 09:30.14 | Chris-NB | is it possible to play ealry media with asterisk? |
| 09:30.22 | tzanger | Chris-NB: yep |
| 09:30.25 | HarryR | Chris-NB, yeah, just dont Answer() |
| 09:30.27 | Chris-NB | let asterisk play early meda befor an incoming call is connected |
| 09:30.45 | Chris-NB | so just play something, then answer |
| 09:30.53 | Aurs | HarryR: don't know if we _need_ to do that.. because we'll have to rewrite most of our extensions.conf anyway |
| 09:31.05 | HarryR | yah, for example, instead of Ringing() you could play your own stuff |
| 09:31.12 | HarryR | ah k Aurs |
| 09:32.27 | Aurs | it would just make it easier if we registered directly to asterisk |
| 09:33.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:35.36 | jeremy_g | what does this imply |
| 09:35.36 | jeremy_g | Looking for s in sip (domain 192.168.0.2) |
| 09:35.36 | jeremy_g | Transmitting (no NAT) to 192.168.0.73:5060: |
| 09:35.36 | jeremy_g | SIP/2.0 484 Address Incomplete |
| 09:35.36 | jeremy_g | Via: SIP/2.0/UDP 192.168.0.73; |
| 09:38.09 | jeremy_g | why does asterisk look for an s in sip domain |
| 09:38.23 | Uatec | Why is there no documentation on how to setup an IAX trunk, or SIP trunk? |
| 09:38.32 | creadurex | there is Uatec |
| 09:38.36 | creadurex | and there is no such thing as a sip trunk |
| 09:38.59 | Uatec | Where, creadurex? |
| 09:40.06 | Chris-NB | if i call a playback() befor answer, the call is accepted by asterisk with a 200 ok |
| 09:40.07 | jeremy_g | Uatec:whats so hard with that? |
| 09:40.49 | Chris-NB | so I'ts no ealry media. the call is established and the caller is billed. but early media should not be billed? is this possible with asterisk? |
| 09:41.07 | Uatec | jeremy_g, i dont' konw |
| 09:41.09 | Uatec | it's not working |
| 09:41.15 | Uatec | all my calls are being rejected by the receiving asterisk box |
| 09:43.57 | lbow | if you want to use Playback for early media, you need the ",noanswer" option |
| 09:45.34 | Uatec | creadurex, then how do i dial a specific SIP connection to my other asterisk box? |
| 09:45.52 | *** join/#asterisk sasch (n=sasch@host117-234-static.4-79-b.business.telecomitalia.it) |
| 09:46.19 | sasch | hi all |
| 09:46.37 | sasch | i have a ubuntu-server that i try to install asterisk 1.2 |
| 09:46.52 | sasch | but when i start asterisk in shell i return this warnig |
| 09:46.53 | sasch | http://pastebin.ca/655363 |
| 09:46.57 | sasch | can help me ... |
| 09:48.17 | HarryR | you built from scratch or used a package forg it? |
| 09:48.49 | Uatec | jeremy_g, if it's so easy why isn't it workin? |
| 09:51.33 | sasch | <HarryR> before i have install asterisk 1.4 from scratch ... now i want to return to asterisk 1.2 and i have install with apt-get |
| 09:51.51 | sasch | <HarryR> excusme for my english... but i'm italian :-P |
| 09:51.56 | HarryR | sasch, make sure you remove all the modules installed by the 1.4 install |
| 09:52.05 | HarryR | sasch, then re-install 1.2 |
| 09:52.21 | HarryR | 1.2 is probably trying to load a 1.4 module and failing |
| 09:52.25 | JT | < Uatec> Why is there no documentation on how to setup an IAX trunk, or |
| 09:52.26 | JT | <PROTECTED> |
| 09:52.30 | sasch | ok |
| 09:52.31 | JT | ^ are you joking? |
| 09:54.23 | Uatec | Where? |
| 09:54.42 | JT | i think the quote was clear |
| 09:55.57 | sasch | <HarryR> i remove all in my /usr/src and in my /etc/asterisk |
| 09:56.12 | sasch | <HarryR> i remove the kernel module for zaptel |
| 09:56.27 | HarryR | nah, make sure /var/lib/asterisk/modules is cleaned between installs of more than 1 minor version different |
| 09:56.41 | Uatec | ok, no i am not joking |
| 09:56.46 | Uatec | Where is this documentation? |
| 09:56.51 | JT | ~thebook |
| 09:56.51 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:57.03 | sasch | bash: cd: /var/lib/asterisk/modules: No such file or directory |
| 09:57.34 | HarryR | uh /usr/lib/asterisk/modules |
| 09:57.40 | HarryR | it's monday, excuse me :0 |
| 10:00.14 | sasch | ok i remove all |
| 10:00.24 | sasch | now i run apt |
| 10:04.04 | *** join/#asterisk KermitTheFragger (n=siepkes@53571672.cable.casema.nl) |
| 10:08.28 | sasch | i have a tdm400p ... first and second slot i have fxo and thirth slot i have fxs |
| 10:08.52 | sasch | in wich mode i make zapatel.conf |
| 10:09.13 | Uatec | well, i'm rereading through TFOT, but there's nothing here that's helping me with my call being rejected |
| 10:11.13 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 10:12.43 | Uatec | OMFG, there's nothing bloody wrong, why doesn't it work !?!?! ARGH! |
| 10:13.15 | kombi | can someone clarify the quoting policy for me? I try to do System(echo "<?php $bluepill = 'redpill' ?>" > /some/file.txt) and it throws "No closing parenthesis found" Why might it be? |
| 10:14.36 | kombi | does * not differentiate between > and ) ? |
| 10:16.03 | kombi | System(echo "bluepill" > /red/pill.txt) works like a charm.. weired.. |
| 10:17.20 | sasch | i have one problem with zapata |
| 10:17.21 | sasch | http://pastebin.ca/655381 |
| 10:18.03 | kombi | I am a giddy goat.. |
| 10:18.32 | J4k3 | 05:18 < kombi> I am a giddy goatse.. |
| 10:18.41 | Uatec | eww |
| 10:18.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 10:19.50 | kombi | System(echo "some weired code \;") <- escape the ; because it is a comment in the dialplan |
| 10:20.04 | kombi | easy, huh? |
| 10:23.17 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 10:24.14 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 10:26.13 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 10:36.39 | krdian_ | hi |
| 10:36.44 | Uatec | so JT, do you know why i would be getting that error? |
| 10:47.03 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 10:48.42 | JT | what error? |
| 10:48.47 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 10:50.48 | Uatec | Whenever i try to dial through an iax trunk, i get: Aug 13 11:50:27 WARNING[13307]: chan_iax2.c:7140 socket_read: Call rejected by 192.168.232.157: No authority found |
| 10:50.49 | Uatec | <PROTECTED> |
| 10:50.52 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262) |
| 10:50.59 | Uatec | from the receiver |
| 10:56.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:01.22 | alin` | I put in extensions.conf this line: |
| 11:01.23 | alin` | Executing [300@200:1] SLAStation("SIP/200-099511d8", "station1") in new stack |
| 11:01.23 | alin` | <PROTECTED> |
| 11:01.23 | alin` | and, I got this error: |
| 11:01.26 | alin` | Executing [300@200:1] SLAStation("SIP/200-099511d8", "station1") in new stack |
| 11:01.28 | alin` | <PROTECTED> |
| 11:01.32 | alin` | [Aug 13 15:49:20] WARNING[4102]: app_meetme.c:770 build_conf: Unable to open pseudo channel - trying device |
| 11:01.35 | alin` | [Aug 13 15:49:20] WARNING[4102]: app_meetme.c:773 build_conf: Unable to open pseudo device |
| 11:01.38 | alin` | [Aug 13 15:49:20] WARNING[4100]: app_meetme.c:2787 admin_exec: Conference number 'SLA_line1' not found! |
| 11:01.42 | alin` | <PROTECTED> |
| 11:01.51 | alin` | what should I do in order to create a Shared Line? Could somebody tell me please? |
| 11:13.02 | Sweeper | oi, I'm wanting to detect and recieve faxes on a shared voice/fax line on a TDM400p. what's the current best bet? |
| 11:17.25 | tzanger | use the fax extension |
| 11:18.01 | Sweeper | like in spandsp? or is the fax extension builtin, and I can make a system call to hylafax to pickup? |
| 11:18.10 | JT | alin`: if you expect help, use pastebin. |
| 11:19.29 | JT | the fax extension has nothing to do with spandsp |
| 11:19.54 | Sweeper | so the latter :) |
| 11:20.12 | tzanger | asterisk will jump to the fax extension in the current context if it detects a fax tone when playing a greeting or waiting |
| 11:20.37 | tzanger | you then use it to do whatever you like... RxFax, FancyHylaThing, whatever |
| 11:22.27 | Sweeper | sexy |
| 11:25.57 | creadurex | hmm |
| 11:26.18 | creadurex | with a 9330 gn headset, which part is responsible for giving a sound on inbound calls? the softphone or the headset itself? |
| 11:26.27 | creadurex | a ringing notification that is |
| 11:27.30 | alin` | JT: ok |
| 11:28.39 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 11:32.49 | alin` | how can I list active conferences numbers |
| 11:32.51 | alin` | ? |
| 11:33.39 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 11:37.04 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:38.21 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 11:39.41 | alin` | I have in extensions.conf exten => 333,1,MeetMe(2345||9938) |
| 11:39.41 | alin` | and I obtain the answer |
| 11:39.41 | alin` | <PROTECTED> |
| 11:39.44 | alin` | <PROTECTED> |
| 11:40.10 | alin` | why? I have defined the room 2345 in meetme.conf... |
| 11:42.03 | `Sean | alin` jt already told you god damn it use pastebin stop being stupid, and use your head. |
| 11:45.19 | alin` | `Sean: this time it was no need to paste. |
| 11:45.56 | alin` | if you want to answer, ok. you are not obliget to |
| 11:47.35 | *** join/#asterisk kkn088 (n=kikoun@84.4.74.213) |
| 12:01.29 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:02.27 | alin` | how can I define a conference room for meetme? |
| 12:04.01 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:04.58 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.39 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 12:07.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:07.21 | hmmhesays | very carefully |
| 12:08.05 | hmmhesays | you probably don't have zaptel loaded |
| 12:08.27 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 12:08.57 | sakic | if someone sets up asterisk for you manually can you add a gui to it for modification purposes? |
| 12:09.41 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 12:10.47 | alin` | hmmhesays: I have loaded zaptel |
| 12:11.33 | *** join/#asterisk misk0 (n=misk0@62.48.116.68) |
| 12:11.54 | misk0 | anyone have installed trunk sip-tcptls? |
| 12:12.47 | *** join/#asterisk b0ri0 (n=b0ri0@196.219.66.14) |
| 12:12.56 | JT | sakic: not really? |
| 12:13.14 | b0ri0 | guys , am new to the whole Asterisk thing , I dont have any questions |
| 12:13.21 | b0ri0 | I am just saying Hiiii |
| 12:14.27 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:14.27 | hmmhesays | sakic: in 1.4 yes |
| 12:19.25 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 12:21.01 | *** join/#asterisk guillote_GNU (n=bancaria@host228.190-30-60.telecom.net.ar) |
| 12:21.27 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 12:22.04 | ramindia | any help on incoming call route to extension |
| 12:22.29 | ramindia | when i call DID from phone the call coming in , with Alias ID from provider |
| 12:22.45 | ramindia | how do i match that Alias and send call to extension ? |
| 12:26.21 | misk0 | ramindia: you can login into CLI and monitor incoming calls and see there number |
| 12:26.59 | misk0 | after, you can use that number in extension.conf as - exten => number,1,Dial(something) |
| 12:27.26 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 12:28.25 | ramindia | misk0: this what my ngrep show http://www.pastebin.ca/655484 |
| 12:28.31 | ramindia | any suggestions |
| 12:29.17 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:29.46 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:31.52 | jeremy_g | ramindia:whats was your ngrep cmd line syntax |
| 12:32.31 | ramindia | jeremy_g: ngrep -W byline 5060.. iam only testing my incoming call |
| 12:32.53 | jeremy_g | ramindia:oh ok |
| 12:33.01 | ramindia | iam able to see the call coming till asterisk box..with my other DID |
| 12:33.21 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 12:34.26 | alin` | how can I set * to send SIP NOTIFY MESSAGES IN PIDF_XML format, instead of DIALOG_INFO_XML format? |
| 12:34.51 | hmmhesays | you can do that? |
| 12:35.03 | alin` | hmmhesays: you are asking me? |
| 12:35.12 | *** part/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 12:35.26 | ramindia | misk0: any suggestion |
| 12:36.35 | [TK]D-Fender | alin`: You have the source code... get to work... |
| 12:36.54 | alin` | [TK]D-Fender: :) |
| 12:37.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:45.22 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 12:47.41 | jeremy_g | i just dont get it, why do i get this msg SIP/2.0 484 Address Incomplete. |
| 12:47.41 | jeremy_g | Via: SIP/2.0/UDP 192.168.0. |
| 12:48.12 | mvanbaak | because you miss the last bit of info on that ip address |
| 12:48.27 | jeremy_g | mvanbaak:nopes |
| 12:48.35 | mvanbaak | 192.168.0. is not a valid ip |
| 12:48.44 | jeremy_g | mvanbaak:thats an incompelte paste |
| 12:48.49 | mvanbaak | ah |
| 12:49.00 | mvanbaak | it looked sane with the error message :) |
| 12:49.43 | mvanbaak | what asterisk version ? |
| 12:50.03 | jeremy_g | mvanbaak:1.2.13 |
| 12:51.41 | mvanbaak | hhmm, AST_CAUSE_INVALID_NUMBER_FORMAT |
| 12:52.12 | *** join/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 12:52.28 | [TK]D-Fender | jeremy_g: pastebin your dialplan and CLI output w/ sip debug. |
| 12:53.01 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 12:53.27 | alin` | what is the difference between SLA and BLA ? |
| 12:53.35 | *** join/#asterisk vutamhoan (n=hoavq@58.187.90.91) |
| 12:53.57 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:54.11 | jeremy_g | [TK]D-Fender:ah! thats too much work, it was working perfectly with the same config |
| 12:54.19 | [TK]D-Fender | alin`: SLA = having 2 phones registered to the same "account" where BOTH ring and one can place a call on hold and the other one steals it. |
| 12:54.35 | jeremy_g | what happened to it |
| 12:54.50 | [TK]D-Fender | jeremy_g: Pastbin = too much work? I guess you were just loking for our SYMPATHY then. Try in #drphil ..... |
| 12:55.08 | [TK]D-Fender | alin`: BLF = just a pretty indicator to tell you their on the phone. |
| 12:55.15 | vutamhoan | Hi, I use atxfer but callee's ring in 5s and hangup - how can't I exten ringing time? |
| 12:55.23 | [TK]D-Fender | alin`: SLA = usable line key SHARING their identity. |
| 12:56.40 | jeremy_g | [TK]D-Fender:yeah perhaps.. |
| 12:56.47 | ramindia | how can i match wild card any incoming send to extension yyy |
| 12:57.03 | hmmhesays | um read extensions.conf ? |
| 12:57.16 | *** join/#asterisk ManxPower (n=manxpowe@015-844-731.area5.spcsdns.net) |
| 12:57.27 | [TK]D-Fender | ramindia: _. <- but this is typically rather stupid. |
| 12:58.07 | ramindia | how about any incoming match with inside digits *567* |
| 12:58.14 | hmmhesays | um read extensions.conf ? |
| 12:58.42 | mvanbaak | _.[567]. |
| 12:59.17 | mvanbaak | oh wait |
| 12:59.19 | [TK]D-Fender | mvanbaak: Ummm..... don't think so :) |
| 12:59.26 | mvanbaak | there's 567 in the number |
| 12:59.33 | mvanbaak | then loose the braces |
| 12:59.49 | ManxPower | mvanbaak: . must be the LAST char in a pattern match |
| 12:59.57 | [TK]D-Fender | mvanbaak: indeed ^^ |
| 13:00.09 | ramindia | mvanbaak: that means, anything side match with 567 will send to extension is this correct |
| 13:00.23 | mvanbaak | eh ? |
| 13:00.31 | ManxPower | ramindia: you cannot do that with Asterisk |
| 13:00.33 | [TK]D-Fender | ramindia: You need to catch something MORE global than you want and TEST it after. Go read THEBOOK, and learn how to use dialplan patterns. |
| 13:00.35 | [TK]D-Fender | ~book |
| 13:00.36 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:00.49 | mvanbaak | ah |
| 13:00.50 | ManxPower | at least you can't do it in one line match |
| 13:00.52 | mvanbaak | I do it in agi |
| 13:01.06 | [TK]D-Fender | ManxPower: Well, you CAN, but not in a way that 404's no-valid entries. |
| 13:01.10 | ramindia | ok |
| 13:01.14 | [TK]D-Fender | ManxPower: Yeah, better... |
| 13:01.54 | ManxPower | You can do anything in a dialplan, but you can't easily match <any number of any digits>567<any number of any digits> |
| 13:02.17 | *** join/#asterisk guillote_GNU (n=bancaria@host228.190-30-60.telecom.net.ar) |
| 13:02.22 | ManxPower | I can't imagine anyone needing to do so unless their dialplan design is totally screwed up |
| 13:02.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:03.12 | jeremy_g | i usually put hangup for _X. |
| 13:03.13 | [TK]D-Fender | ManxPower: * need the ability to have something like a "non-acknowledged match" property so you can test the number without respoing "trying" or "404 |
| 13:03.28 | *** join/#asterisk nighty^ (n=nighty@p3132-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 13:03.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:04.11 | *** join/#asterisk sakik (n=sakic@adsl-227-157-12.clt.bellsouth.net) |
| 13:06.02 | *** join/#asterisk ManxPower (n=manxpowe@015-822-747.area5.spcsdns.net) |
| 13:06.28 | ManxPower | I HATE USB Cell modems |
| 13:06.53 | sergee | i hate modems |
| 13:07.06 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 13:07.13 | jeremy_g | ManxPower:why |
| 13:07.18 | Uatec | I hate usb. |
| 13:07.24 | jeremy_g | Uatec: :o |
| 13:07.24 | Uatec | lol, i don't really |
| 13:07.29 | Uatec | i use it all of the time |
| 13:07.43 | jeremy_g | eat that |
| 13:07.46 | sergee | Days of hate in MacDonalds |
| 13:07.57 | jeremy_g | premier |
| 13:08.09 | Uatec | I like firewire too. |
| 13:08.09 | ManxPower | jeremy_g: If I connect it directly to the laptop it will get bumped and disconnect or break. If I put it in it's USB cradel it gets bumped and disconnected. |
| 13:08.47 | jeremy_g | ManxPower:that particular product model seem to sux |
| 13:08.53 | ManxPower | Verizon finally upgraded to EVDO in my area, about a month ago. |
| 13:09.09 | ManxPower | jeremy_g: I can't imagine any design that would be any better |
| 13:09.51 | ManxPower | I can't even remember the last time I had a PCMCIA card pop out of the slot when it was not supposed to. |
| 13:10.43 | jeremy_g | ManxPower: someone has casted a spell |
| 13:11.25 | ManxPower | someone needs the USB Cell modem shoved up their ass and that person would be a sprint sales rep. |
| 13:11.45 | jeremy_g | hehe poor guy |
| 13:12.31 | jeremy_g | you are really ruining the image of these usb modems, some of them are innocent |
| 13:12.59 | jeremy_g | like we use 3 and telia in sweden, which are terrific |
| 13:13.34 | ManxPower | jeremy_g: Can you point out a USB modem that does not have the problem if sticking out so much it gets caught on things? |
| 13:13.39 | *** join/#asterisk guillote_GNU (n=bancaria@host228.190-30-60.telecom.net.ar) |
| 13:14.01 | JT | one that's glued in |
| 13:14.06 | ManxPower | many USB devices have this problem when used with a laptop. |
| 13:14.23 | ManxPower | JT: Superglue is one of my things to try. |
| 13:14.24 | JT | s/laptop/computer/ |
| 13:14.30 | JT | usb has a dumb connector |
| 13:15.02 | ManxPower | JT: no, on a desktop the device and computer are on a desk. With a laptop you never know how/when/where the computer will be moved. |
| 13:15.03 | jeremy_g | ManxPower:i normally use them in a neat way, i never put them to such end user test of snatching the damn thing out randomly |
| 13:15.13 | mmlj4 | use a hub? |
| 13:15.19 | mmlj4 | hey ManxPower |
| 13:15.25 | JT | you never know when a peripheral will be yanked on a desk |
| 13:15.31 | ManxPower | When I got disconnected my foot bumped the USB cradel that is sitting on the bed. |
| 13:15.45 | ManxPower | which is where I and my computer are sitting. |
| 13:16.01 | mvanbaak | use bluetooth |
| 13:16.22 | ManxPower | mvanbaak: That is actually the best suggestion I've seen all morning. |
| 13:16.36 | ManxPower | not practical for me, but still the best suggestion. |
| 13:16.37 | mmlj4 | ManxPower: hey, didn't know if you knew that they redid your old place, and someone's rented it out... I passed by a month ago |
| 13:16.54 | ManxPower | mmlj4: Good for them. |
| 13:17.15 | ManxPower | My old place was not badly damaged compared to most. |
| 13:17.16 | mvanbaak | ManxPower: thank you, thank you |
| 13:17.19 | *** join/#asterisk gardo (n=gardo@125.212.12.90) |
| 13:17.30 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 13:17.43 | mvanbaak | since this week I finally have a laptop with bluetooth |
| 13:17.51 | mmlj4 | any more restlessness from the UMC natives? |
| 13:17.58 | mvanbaak | looking at bluetooth cell modem as well |
| 13:18.03 | mvanbaak | maybe I can use my phone for it |
| 13:18.11 | ManxPower | mmlj4: not that I know of. I have some IVR recordings to set up. |
| 13:18.17 | *** part/#asterisk misk0 (n=misk0@62.48.116.68) |
| 13:18.22 | mmlj4 | cool |
| 13:18.26 | ManxPower | Then they will be told to contact Hunt Brothers until I get paid. |
| 13:18.40 | ManxPower | I've not seen a check from them for at least 3 months |
| 13:18.58 | mmlj4 | also, i'm punching down the last of the wiring at ormond today... but the T won't be installed for another 2 weeks, probably :-(] |
| 13:19.30 | ManxPower | mmlj4: I'll be down starting right after laborday |
| 13:19.36 | mmlj4 | ah, ok |
| 13:19.50 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 13:19.51 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:19.54 | *** join/#asterisk didm (n=mehdi@apps2.netmsds.com) |
| 13:20.09 | mmlj4 | hey, you're in burmingscum now, is that right? |
| 13:20.59 | ManxPower | mmlj4: Steele, AL actually |
| 13:21.03 | mmlj4 | ah. |
| 13:21.21 | ManxPower | Gadsden/Rainbow City is where I do my banking, shopping, etc. |
| 13:21.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:21.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:22.12 | didm | hi guys bought licenses for g729 from digium website, and didn't receive yet the key, someone can help me please and tell me how long it takes to receive the licenses key after the purchase |
| 13:22.23 | mmlj4 | trying the woods for a change? cool |
| 13:22.35 | mmlj4 | i'd kill for a place in the middle of the woods |
| 13:22.38 | ManxPower | didm: 2 business days |
| 13:22.52 | ManxPower | mmlj4: I was never in birmingham. |
| 13:23.06 | ManxPower | It's just nobody knows where Steele, AL is. |
| 13:23.11 | didm | Thanks ManxPower, and you know if it is by email or ??? |
| 13:23.12 | mmlj4 | right "=_ |
| 13:23.16 | mmlj4 | :-) |
| 13:23.19 | ManxPower | didm: e-mail |
| 13:23.23 | ManxPower | check your spam folder |
| 13:23.31 | didm | ok thanks |
| 13:24.08 | mvanbaak | ManxPower: not even google maps ? |
| 13:24.32 | ManxPower | mvanbaak: Google maps knows where it is, but it gets the actuall location of the street address wrong. |
| 13:24.33 | didm | oh actually i did but nothing, I think you 're right bcs I place the order on saturday |
| 13:24.40 | ManxPower | it is off by a mile or two |
| 13:24.45 | mmlj4 | hey, my client sells stuff all over the world, and is sending me to the bahamas for 2+ weeks to do an install for them :-) |
| 13:24.54 | mvanbaak | mmlj4: nice ! |
| 13:25.09 | mvanbaak | mmlj4: call in sick once you're there |
| 13:25.12 | *** join/#asterisk myiagy (n=myiagy@201.64.81.78) |
| 13:25.18 | mvanbaak | bad food in plane or something |
| 13:25.22 | mvanbaak | need week to recover |
| 13:25.26 | mmlj4 | waiting for the phone to ring, supposed to be flying out this week |
| 13:26.00 | mvanbaak | check your asterisk logs, maybe you made a booboo there ;) |
| 13:26.16 | mmlj4 | i'm going to look for a condo on the beach, if possible |
| 13:26.23 | mvanbaak | happened to me once |
| 13:26.29 | mvanbaak | I was waiting and waiting for a coll |
| 13:26.38 | mvanbaak | s/coll/call/ |
| 13:26.55 | mvanbaak | after an hour I checked asterisk console and saw a lot of errors ;) |
| 13:27.01 | mmlj4 | a correctbot? smooth |
| 13:27.37 | mmlj4 | or useless if users can parse sed |
| 13:29.54 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5e9ab3be7c88c819) |
| 13:30.24 | *** part/#asterisk didm (n=mehdi@apps2.netmsds.com) |
| 13:30.42 | *** join/#asterisk didm (n=mehdi@apps2.netmsds.com) |
| 13:30.44 | *** part/#asterisk didm (n=mehdi@apps2.netmsds.com) |
| 13:30.57 | *** part/#asterisk kslater (n=kslater@24.svnf1.xdsl.nauticom.net) |
| 13:31.22 | pourriture | If I have ztmonitor running , should I be able to see an attempt to dial an extension on the RX audio level? |
| 13:31.36 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 13:31.47 | ManxPower | pourriture: yes. |
| 13:32.00 | ManxPower | actually on the tx level |
| 13:32.40 | pourriture | ManxPower: I am not ... it just sits there ... I see the recording voice on the TX side, but when I push a button, no response |
| 13:32.56 | pourriture | Does that sould like broken modem? or bad config? |
| 13:33.21 | ManxPower | pourriture: it sounds like a DTMF configuration issue. |
| 13:33.33 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 13:35.48 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 13:38.07 | *** join/#asterisk trustinfo-tb (n=trust@AStrasbourg-156-1-80-175.w86-204.abo.wanadoo.fr) |
| 13:39.50 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 13:39.55 | *** join/#asterisk trustinfo-tb (n=trust@AStrasbourg-156-1-80-175.w86-204.abo.wanadoo.fr) |
| 13:41.31 | trustinfo-tb | hello world |
| 13:42.16 | trustinfo-tb | i've a problem to install my new B410P on trixbox 2.3 with asterisk 1.4.6 |
| 13:42.22 | trustinfo-tb | can someon help me? |
| 13:43.03 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 13:45.21 | ManxPower | no. |
| 13:45.30 | ManxPower | Try the trixbox forums |
| 13:48.08 | trustinfo-tb | yes but trixbox is a web page. Asterisk the motor and the probs is with chan_misdn.so |
| 13:48.08 | *** join/#asterisk bgogol (n=bgogol@dsl017-017-116.wdc2.dsl.speakeasy.net) |
| 13:48.16 | *** join/#asterisk saftsack (n=saftsack@pD9E04B15.dip.t-dialin.net) |
| 13:48.16 | trustinfo-tb | not with trixbox |
| 13:48.43 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 13:48.43 | hmmhesays | actually trixbox uses freepbx as a gui doesn't it? |
| 13:48.46 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:48.51 | trustinfo-tb | yes |
| 13:50.46 | trustinfo-tb | when i compile asterisk to have chan_misdn.so module i have all module but no misdn |
| 13:51.25 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 13:53.07 | *** join/#asterisk zerohalo (n=zeroHalo@h-74-2-90-66.cmbrmaor.covad.net) |
| 13:54.43 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.18.47) |
| 13:55.24 | anonymouz666 | ..and a new day will dawn for those who stand long.... |
| 13:55.57 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:57.21 | flujan | hi all. :) |
| 13:58.07 | flujan | I am using 1.4.5 and sometimes All incoming calls become muted... None of the sides listen... But the call is originated. |
| 13:58.22 | flujan | today I had this problem, and it is only fixed when I restart asterisk |
| 13:58.31 | flujan | here goes the message output: |
| 13:58.32 | flujan | http://pastebin.com/d6c0b19bd |
| 13:58.58 | flujan | are someone having this kind of problem? |
| 14:00.17 | DrAk0 | why i keep getting this. |
| 14:00.18 | DrAk0 | [Aug 13 15:59:53] NOTICE[8719]: chan_sip.c:14736 handle_request_subscribe: Got SUBSCRIBE for extension 14@from-internal from 192.168.1.102, but there is no hint for that extension. |
| 14:01.31 | [TK]D-Fender | DrAk0: Because that phone is trying to check for BLF on an extension that you didn't set up a hint for. |
| 14:01.53 | DrAk0 | [TK]D-Fender, what should i do then? |
| 14:02.08 | [TK]D-Fender | DrAk0: Tell you phone to STOP, or set up the hint. |
| 14:02.10 | [TK]D-Fender | (duh) |
| 14:02.29 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 14:02.47 | DrAk0 | [TK]D-Fender, but it happens with all phones, something i can do from asterisk ? |
| 14:03.01 | asterisknerds | <PROTECTED> |
| 14:04.28 | [TK]D-Fender | DrAk0: I jsut answered your question..... |
| 14:05.54 | DrAk0 | set up the hint |
| 14:07.07 | *** join/#asterisk hohum_ (n=dcorbe@gate.globecommsystems.com) |
| 14:09.55 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:10.48 | flujan | Drako, set up hints for all the phones... :) |
| 14:10.53 | flujan | [TK]D-Fender: hi. :) |
| 14:10.59 | *** join/#asterisk hohum_ (n=dcorbe@gate.globecommsystems.com) |
| 14:11.04 | [TK]D-Fender | flujan: Good morning |
| 14:11.16 | puzzled | hi all |
| 14:11.38 | flujan | [TK]D-Fender: :) Fender, considering the problem I am having and the output I pasted, do you recommend me to updated to 1.4.10? |
| 14:11.43 | flujan | hi puzzled |
| 14:12.00 | [TK]D-Fender | flujan: Reading now |
| 14:12.06 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:12.07 | flujan | ok |
| 14:12.49 | [TK]D-Fender | flujan: debug info like taht doesn't help me much.... provide normal CLI + SIP debug for it please |
| 14:13.04 | [TK]D-Fender | flujan: And naturally I suggest you go with the latest full release. |
| 14:13.52 | flujan | [TK]D-Fender: as The problem is that this error happen sometimes during the day... I will wait it to happen again... Unfortunately, I can create this behavior of the pbx. :( |
| 14:14.21 | [TK]D-Fender | flujan: I'm not sure WHICH error, and can't tell where you wen't wrong in there. |
| 14:16.06 | flujan | [TK]D-Fender: the phones starts to be muted. Sometimes asterisk look normal on the cli ( no errors ) but no calls are processed... They are just hanged up. |
| 14:16.19 | flujan | I will try the 1.4.10 |
| 14:16.24 | [TK]D-Fender | flujan: while a call is in progress? |
| 14:16.30 | [TK]D-Fender | flujan: 1.4.10.1 <------ |
| 14:16.30 | flujan | yeap |
| 14:16.43 | flujan | ops... I missed this update. :) |
| 14:20.19 | g1powermac | Hey All |
| 14:20.30 | g1powermac | anyone recommend a really good desk wired SIP phone? |
| 14:21.30 | [TK]D-Fender | g1powermac: Any Polycom would do |
| 14:21.34 | *** join/#asterisk saftsack (n=saftsack@217.224.75.21) |
| 14:24.37 | [TK]D-Fender | g1powermac: Model I'd suggest varies based on numerous factors |
| 14:25.19 | g1powermac | yea, checking them out now |
| 14:26.16 | g1powermac | hmm, like the one with the built in ethernet switch |
| 14:26.20 | [TK]D-Fender | http://www.telephonydepot.com/Polycom_s/25.htm |
| 14:26.27 | g1powermac | less ethernet I have to run |
| 14:26.33 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:26.52 | [TK]D-Fender | g1powermac: Only the IP 320 DOESN'T have a switch. |
| 14:26.55 | g1powermac | yea |
| 14:27.01 | g1powermac | I'm looking at the Soundpoint IP 330 |
| 14:27.07 | [TK]D-Fender | g1powermac: What kind of usage willthis phone see? |
| 14:27.18 | g1powermac | just basic office use |
| 14:27.26 | [TK]D-Fender | g1powermac: Got PoE? How many phones total are you planning? |
| 14:28.03 | g1powermac | no PoE, and number of phones hasn't been determined yet at least for the remote location since we haven't finalized the lease yet :-) |
| 14:28.39 | g1powermac | I'm thinking between 5 to 6 phones, but that includes the couple of wifi sip phones |
| 14:28.44 | [TK]D-Fender | g1powermac: no estimates? |
| 14:28.57 | cpm | I wish I could actually use the built-in switch in the polycom phone |
| 14:29.12 | [TK]D-Fender | ~wifisip |
| 14:29.21 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 14:29.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 14:29.23 | g1powermac | yes, I know, you told me about them yesterday |
| 14:29.36 | [TK]D-Fender | g1powermac: ATA +normal cordless... if you know whats good for you |
| 14:29.51 | [TK]D-Fender | g1powermac: I say it so often its hard to keep track to whom :) |
| 14:30.01 | *** part/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 14:30.01 | g1powermac | I might go that way if I find the one phone is bad |
| 14:30.04 | [TK]D-Fender | cpm: And why don't you? |
| 14:30.11 | *** part/#asterisk saftsack (n=saftsack@217.224.75.21) |
| 14:30.19 | [TK]D-Fender | g1powermac: Oh yeah... you got the UTSC.... |
| 14:30.22 | cpm | [TK]D-Fender, why don't I I what? |
| 14:30.24 | [TK]D-Fender | g1powermac: *shudder* |
| 14:30.31 | [TK]D-Fender | cpm: use the switch in it? |
| 14:31.02 | cpm | [TK]D-Fender, ahh, sorry. Because the phone network and the user network are separate vlans, and I don't think the switch does vlan tagging. |
| 14:31.23 | Uatec | [TK]D-Fender, what about GPRS sip phones? |
| 14:31.39 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 14:31.47 | [TK]D-Fender | g1powermac: For basic use, IP 330 (111.95) + Power Brick (17.95) = 129.90 / desk |
| 14:31.59 | g1powermac | its a 2.4ghz one that really messes with the wifi we got |
| 14:32.06 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 14:32.10 | [TK]D-Fender | Uatec: I've heard the Nokia E61/62 is "ok" |
| 14:32.13 | Uatec | Oh dear, i've removed any requirement for a password but i'm still getting "call rejected" |
| 14:32.23 | Uatec | [TK]D-Fender, you have? i'm using an XDA Exec at the momemtn |
| 14:33.54 | [TK]D-Fender | Uatec: I'm waiting for bkruse's Seek-Rat Poject to come to fruition :) |
| 14:34.49 | Uatec | what? |
| 14:36.41 | [TK]D-Fender | Uatec: OpenMoko + MokoIAX :) |
| 14:38.18 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 14:38.23 | *** join/#asterisk suvir (n=chatzill@ppp-124.120.129.161.revip2.asianet.co.th) |
| 14:39.36 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:40.00 | Uatec | oh |
| 14:40.02 | Uatec | ACH |
| 14:40.07 | Uatec | the openmoko makes me cry |
| 14:41.24 | [TK]D-Fender | Uatec: Good way or bad? |
| 14:43.26 | *** join/#asterisk \lart (i=foobar@pool-71-168-216-181.cmdnnj.fios.verizon.net) |
| 14:44.22 | \lart | Greetings all.. Anyone have Polycom SIP 2.1.2 and bootroom 3.2.3 rev b available? My reseller has proven to be utterly useless in providing these releases - and of course, I can't DL them unless I'm a reseller. |
| 14:44.23 | *** join/#asterisk jonconley (n=joncon@12.145.191.2) |
| 14:44.31 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:46.28 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 14:46.37 | HarryR | uh, I've got 3.2.2 |
| 14:47.06 | Uatec | bad bad bad |
| 14:47.06 | \lart | HarryR, interesting, I don't even see that as an available release on polycom's site.. |
| 14:47.32 | HarryR | http://www.freedomphones.net/polycom/files/spip_ssip_bootrom_3_2_2.zip |
| 14:48.12 | \lart | they show 3.2.3 rev b, 3.1.3 rev d and 3.1.0.. |
| 14:48.20 | \lart | i'll give a look at 3.2.2 |
| 14:48.22 | \lart | thx |
| 14:49.46 | *** join/#asterisk myiagy (n=myiagy@201.64.81.78) |
| 14:50.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:51.26 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:52.13 | HarryR | \lart, we only have 3.2.2 here |
| 14:52.36 | pourriture | when working in conf files ... it is very important to make sure you spell things right :| |
| 14:53.20 | pourriture | for example ... echotraing is not appropriate |
| 14:54.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:55.12 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 14:55.17 | x86 | morning |
| 14:55.18 | whywontitwork | hi there |
| 14:55.27 | whywontitwork | need some help on call pick up? |
| 14:55.39 | x86 | whywontitwork: was that a question or a statement? |
| 14:55.40 | whywontitwork | using a 4port bri gateway |
| 14:55.40 | x86 | :P |
| 14:55.57 | whywontitwork | where does one set the pickup groups ? |
| 14:56.18 | tzafrir | whywontitwork, you'll need to be more explicit: |
| 14:56.36 | tzafrir | set up in asterisk? in the gateway? (which gateway is it?) |
| 14:56.38 | Uatec | when i type "sip show peers" i get: lucifer/s 192.168.232.128 D 5060 Unmonitored |
| 14:56.49 | Uatec | what's lucifer/s ? |
| 14:56.54 | Uatec | and no point have i put in s as a username |
| 14:57.07 | whywontitwork | and witch one do u use, callgoup= or group= and if anyone could explain the diference? please |
| 14:57.13 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 14:57.25 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 15:01.26 | Uatec | whywontitwork, in sip.conf (or iax.conf) |
| 15:01.43 | Uatec | buti don't remember the difference, i just stick everybody in the same one |
| 15:03.42 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:05.48 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:06.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:08.12 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 15:11.09 | [TK]D-Fender | Uatec: When [lucifer] registered, it listed "s" as the exten to call back on. |
| 15:13.03 | *** join/#asterisk Corydon76-lap (i=Corydon7@pdpc/supporter/sustaining/Corydon76-home) |
| 15:13.03 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 15:13.40 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:14.12 | Uatec | ahhh |
| 15:14.12 | Uatec | ok |
| 15:14.18 | Uatec | this is frustrating |
| 15:14.26 | Uatec | i'm sure i'm doing something wrong |
| 15:14.34 | Uatec | i've got the register => line in my sip.conf |
| 15:14.37 | Uatec | how do i dial out on that? |
| 15:14.44 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:14.45 | [TK]D-Fender | Uatec: You DON'T. |
| 15:14.47 | [TK]D-Fender | ~sipregister |
| 15:14.50 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 15:14.50 | Uatec | i mean... |
| 15:15.15 | Uatec | yes... |
| 15:15.18 | Uatec | i was being silly |
| 15:15.28 | Uatec | but to make a call downt he same way as the registerr line |
| 15:16.04 | Uatec | i should have something like: exten => 100,1,Dial(SIP/lucifer.uatec.net/100) |
| 15:16.36 | Uatec | yes? to dial the sip device lucifer.uatec.net with extension 100 ? |
| 15:16.51 | Uatec | where lucifer.uatec.net is another asterisk box |
| 15:18.17 | [TK]D-Fender | Uatec: You should not be dialing by host-name, you should ahve a peer set up for this with auth info, host, etc all specified |
| 15:18.36 | [TK]D-Fender | Uatec: through which you could : Dial(SIP/lucifer/12345) |
| 15:19.44 | *** join/#asterisk ta^3 (n=tacvbo@189.146.191.75) |
| 15:20.08 | Uatec | ahhhh |
| 15:20.13 | Uatec | yes, that :D |
| 15:20.17 | Uatec | i have the peer |
| 15:20.21 | Uatec | oh dear |
| 15:20.36 | Uatec | now on the peer (lucifer) i'm getting Failed to authenticate user "Spare Desk" <sip:sparedesk@192.168.232.128>;tag=as15bf0b86 |
| 15:20.49 | Uatec | sparedesk is the device that i am actually dialing from |
| 15:21.14 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 15:21.21 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 15:21.46 | Uatec | how do i make lucifer accept connections from a SIP device on a different Asterisk box? |
| 15:22.24 | Uatec | ah |
| 15:22.27 | Uatec | here we go |
| 15:22.33 | Uatec | on the originating asterisk box: |
| 15:22.34 | Uatec | Forbidden - wrong password on authentication for INVITE to '"Spare Desk" <sip:sparedesk@192.168.232.128>;tag=as15bf0b86' |
| 15:22.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:24.22 | whywontitwork | can one use the callgroups to route incoming calls? |
| 15:24.36 | whywontitwork | if yes please give sample?? please |
| 15:26.21 | x86 | [TK]D-Fender: morning :) |
| 15:28.03 | [TK]D-Fender | whywontitwork: that option has nothing to do with routing calls, that has to do with being able to pickup a call that is ringing other devices in the same group |
| 15:28.10 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:28.16 | ZaVoid | morning all |
| 15:28.25 | x86 | morning |
| 15:28.26 | [TK]D-Fender | whywontitwork: Perhaps you should reword your request from the beginning be specific about exactly what it is you want to do. |
| 15:28.34 | [TK]D-Fender | x86: Good morning |
| 15:28.35 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:29.43 | x86 | got me a nice new Sangoma A102DX this morning :P |
| 15:29.53 | *** join/#asterisk robh71_ (n=robh71@host-65-124-86-25.entouch.net) |
| 15:29.54 | ZaVoid | morning x86 and [TK]D-Fender |
| 15:30.09 | x86 | awww... how cute... baby's first PCIe telephony card ;) |
| 15:30.21 | x86 | heh |
| 15:31.15 | x86 | got a PCIe A200 also (4port FXO) |
| 15:31.24 | SplasPood | Hrm.. is there any functional diff to Dial(SIP/peer/number) vs Dial(SIP/number@peer)? |
| 15:31.26 | x86 | this ought to be interesting :P |
| 15:31.47 | x86 | SplasPood: those are the same |
| 15:31.50 | [TK]D-Fender | SplasPood: Sometimes the latter doesn't work. Also it might match against some other host entry and fail to auth, etc. |
| 15:31.59 | x86 | SplasPood: SIP/peer/number is just better sense ;) |
| 15:32.02 | SplasPood | hrm... ok |
| 15:32.08 | SplasPood | I've never used that |
| 15:32.12 | SplasPood | always done SIP/number@peer |
| 15:32.16 | SplasPood | I will now change my ways ;)" |
| 15:32.23 | x86 | good ;) |
| 15:32.33 | whywontitwork | i have two departments, SALES(SIP100&SIP101) TECNICAL(SIP102&SIP103) 2 sip users per department i need to route call to the department must first ring the first extension the the othere if both in sales are busy overflow to technical and visa versa? |
| 15:32.48 | SwK | oh jesus people... sip/user@host is just more consistant w/ how URIs are susposed to look |
| 15:32.59 | Mercestes | whywontitwork, that wasn't a question. |
| 15:33.08 | SplasPood | SwK: true |
| 15:33.13 | Mercestes | SwK: I tend to agree. |
| 15:33.18 | x86 | SwK: sure, butnot how a dialplan should be based ;) |
| 15:33.32 | Mercestes | SwK: But I'm feelin gbetter because I just had an hour long arguement with #gentoo on whether the ok button shoudl come first, or the cancel button. |
| 15:33.45 | x86 | Mercestes: hahahaha |
| 15:33.46 | SplasPood | SwK: Thats why I asked if there were differences... if there were not, I would continue doing it the way I'd been doing |
| 15:33.53 | Mercestes | x86: No, I'm being serious. =/ |
| 15:33.56 | [TK]D-Fender | whywontitwork: this is basic dialplan flow. Dial them in sequence. |
| 15:33.58 | x86 | wow |
| 15:34.02 | cellphone | the answer is clearly that the two buttons should be ordered randomly every time |
| 15:34.05 | whywontitwork | i know you can dial sip100 and then sip102 and so on , how do you overflow to the other when busy? |
| 15:34.09 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 15:34.13 | Mercestes | cellphone, That's what *I* was saying! |
| 15:34.16 | cellphone | hehe |
| 15:34.23 | Mercestes | Force users to read. =/ |
| 15:34.27 | cellphone | oh, and they should move around when you try to click them. |
| 15:34.38 | Mercestes | To deter users from using mice. |
| 15:34.39 | SwK | Mercestes, well just because someone likes to it one way or the other doesnt make doing it different from every other uri right |
| 15:34.41 | [TK]D-Fender | whywontitwork: It will flow right on through when there there is no answer for whatever reason. |
| 15:34.50 | SwK | seeing the a sip desting is just that a URI |
| 15:34.58 | SwK | nuff said i got back in the corner now |
| 15:35.00 | [TK]D-Fender | whywontitwork: If you don't want to even dial if they are on the phone, then CHECK IT FIRST using "ChanIsAvail" |
| 15:35.48 | whywontitwork | so there is no way to route an incoming call to a group whith cyclic properties?? |
| 15:36.03 | x86 | oh jesus christ |
| 15:36.10 | x86 | whywontitwork: google man... google |
| 15:36.23 | Mercestes | whywontitwork, Uh, yes, with your dialplan |
| 15:36.28 | whywontitwork | meaning s,1,Dial(group/101) |
| 15:37.19 | Mercestes | .... |
| 15:37.27 | Mercestes | group is not a dial technology, so no. |
| 15:37.31 | Mercestes | ~book |
| 15:37.32 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:37.40 | Mercestes | Read that. |
| 15:38.43 | whywontitwork | helpful arent we |
| 15:38.55 | hi365 | when i try to play a "queue-thankyou" message to a queue it plays the callers position in the queue. i dont want to playthat to the caller. how do i stop it? |
| 15:38.58 | x86 | anyone ever use an Sangoma A200 FXO/FXS card? when putting in the FXO or FXS modules, how do you know which side (red or blue) is up, and which is down? |
| 15:39.25 | hi365 | x86: you will see it only goes in one way |
| 15:39.36 | hi365 | (you can tell based on the contacts on the card) |
| 15:39.44 | [TK]D-Fender | whywontitwork: taht isn't how to dial multiple phones at once. |
| 15:39.59 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 15:40.06 | [TK]D-Fender | whywontitwork: Dial(SIP/100&SIP/200&SIP/300,30) |
| 15:40.07 | x86 | hi365: i got it to go in both ways |
| 15:40.14 | whywontitwork | google it X86 |
| 15:40.15 | [TK]D-Fender | whywontitwork: That will dial 3 phones simultaneously. |
| 15:40.31 | hi365 | x86: have a look at the contacts on the "host" cards |
| 15:40.33 | x86 | hi365: the contacts are evenly spaced, and fit either way |
| 15:40.38 | x86 | hmm ok |
| 15:40.59 | hi365 | ur right, but if you have a look youll see that only one side "touches" the host card |
| 15:41.17 | whywontitwork | i know that TK just wanted to know if there is a way to dial multiple extensions without using above sample? |
| 15:41.26 | hi365 | x86: see here: http://wiki.sangoma.com/sangoma-hardware |
| 15:41.32 | hi365 | scrool about halfway down |
| 15:41.43 | x86 | hi365: i see what you're saying... cool thanks :) |
| 15:42.57 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 15:43.59 | [TK]D-Fender | whywontitwork: No. DIAL's instructions are remakably clear. "show application dial". there is no concept for "Dial(group/whateverotherkeywordifeellikeinventing) |
| 15:44.49 | whywontitwork | thx Tk busy downloading the book, does this book cover asterisk 1.4???????????// |
| 15:45.14 | [TK]D-Fender | whywontitwork: What it covers still alrgely applies |
| 15:46.01 | [TK]D-Fender | whywontitwork: There are a number of new things (most unnecessay) that it does not clearly. the book was made for 1.2 and 1.2's syntax still apllies for the most part. There are some 1.0 entries in it though that are completely removed from 1.4 however |
| 15:46.08 | x86 | whywontitwork: multiple punctuation marks are _not_ needed |
| 15:46.10 | Corydon76-lap | The second edition (which focuses on 1.4) will be out RSN |
| 15:46.11 | [TK]D-Fender | whywontitwork: A new release is about to be made of the book. |
| 15:46.51 | whywontitwork | have they released the date on the new book? |
| 15:46.56 | x86 | Corydon76-lap: is the dCAP focused on 1.4 or 1.2? |
| 15:47.12 | Mercestes | x86: CCM |
| 15:48.06 | [TK]D-Fender | whywontitwork: Any week now. |
| 15:48.13 | whywontitwork | k thx TK |
| 15:48.23 | *** join/#asterisk Curi (n=creinero@pc-79-234-239-201.cm.vtr.net) |
| 15:48.31 | Corydon76-lap | x86, I believe the current course is focused on 1.2, although I've heard it's getting an update for 1.4 |
| 15:48.32 | [TK]D-Fender | whywontitwork: No reason not to continue reading the one thats out. |
| 15:48.46 | Curi | Hello, does anyone knows how to convert wav files to g729 format? |
| 15:49.08 | Mercestes | Curi: did you google convert wav to g729? |
| 15:49.09 | [TK]D-Fender | whywontitwork: And of course cruising through "show applications" , "show application [appname]", "show functions", 'show function [funcname]" |
| 15:49.34 | [TK]D-Fender | whywontitwork: this is the core of RTFM. If you're wondering how to use an app or function, it helps to read the INSTRUCTIONS :) |
| 15:49.42 | Curi | Mercestes: yup, and I didn't find anything |
| 15:49.50 | [TK]D-Fender | whywontitwork: This last case would have been rather evident. |
| 15:50.40 | Mercestes | Funny, my first hit was helpful |
| 15:52.24 | Mercestes | Curi: |
| 15:52.25 | Mercestes | damnti |
| 15:52.34 | Mercestes | Curi: http://www.trixbox.org/forums/trixbox-forums/help/g729-conversion-utility |
| 15:52.42 | Mercestes | Your google-fu is weak. |
| 15:53.57 | Curi | Mercestes: Oh, i guess i needed to clarify that i wanted an actual conversion software so i can perform a batch conversion, i need to convert a couple of hundreds of files |
| 15:54.10 | hi365 | when i try to play a "queue-thankyou" message to a queue it plays the callers position in the queue. i dont want to playthat to the caller. how do i stop it? |
| 15:54.19 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:54.22 | Mercestes | damnit..now I have to quit googlign Allison pics |
| 15:56.59 | jeremy_g | Mercestes: hahahaha :D |
| 15:57.03 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:57.10 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:57.28 | whywontitwork | what is the best softphone with transfer capabilities? |
| 15:58.00 | brodiem | can anyone recommend a T.38-enabled sip provider? |
| 15:58.11 | [TK]D-Fender | whywontitwork: eyebeam. |
| 15:58.12 | Mercestes | Curi: well, I give up. I accidentally found a queue statistics software piece so I'm going to play with that. |
| 15:58.24 | [TK]D-Fender | Mercestes: Perv. |
| 15:59.17 | pigpen | I am working with iaxmodem. After moving to asterisk 1.4 and the related items, iaxmodem will not attempt to register. |
| 15:59.21 | Mercestes | [TK]D-Fender, The Allison pics or the queue stats? |
| 15:59.54 | [TK]D-Fender | Mercestes: If you have to ask, you're already too far gone ;) |
| 16:00.04 | whywontitwork | Thx again TK |
| 16:00.25 | [TK]D-Fender | whywontitwork: However... |
| 16:00.26 | Mercestes | [TK]D-Fender, queue statistics are hawte |
| 16:00.27 | [TK]D-Fender | ~softphone |
| 16:00.28 | jbot | something that should be drug out into the street and shot |
| 16:00.35 | pigpen | any ideas? I have an identical setup, and did the same upgrade and iaxmodem is working fine. |
| 16:00.37 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:00.52 | *** part/#asterisk Curi (n=creinero@pc-79-234-239-201.cm.vtr.net) |
| 16:01.03 | [TK]D-Fender | lunch time... BBIOAB |
| 16:01.15 | Mercestes | wtf? What does BBIOAB mean? |
| 16:01.22 | whywontitwork | if you use the Dial command as follows Dial(SIP/100&SIP/101&SIP/103) does the first one that answer get the call? |
| 16:01.33 | De_Mon | whywontitwork exactly |
| 16:01.35 | Mercestes | Be back in ...orallly abrasive .....buttkissing? |
| 16:02.19 | De_Mon | Be Back In... Out After Breakfast |
| 16:02.49 | Mercestes | Bacon Bits Is Only After Breakfast? |
| 16:03.01 | jeremy_g | yukh bacon |
| 16:03.26 | Mercestes | Big Boys induce Oral Activities baby? |
| 16:03.32 | puzzled | whywontitwork: afaik yes |
| 16:04.05 | Mercestes | Black Birds Induce Organized Activists Bombings? |
| 16:04.21 | jeremy_g | Mercestes:a friend saw a female pig, having group sex with about 100 pigs all in a line. |
| 16:04.31 | Mercestes | jeremy_g, link? |
| 16:04.38 | *** join/#asterisk Ebola (n=Ebola@host86-139-49-76.range86-139.btcentralplus.com) |
| 16:04.41 | De_Mon | little does curi know, any commandlet can be turned into a batch converter with a little scripting |
| 16:04.48 | jeremy_g | Mercestes:he has stopped eating it :D |
| 16:04.57 | Mercestes | jeremy_g, LINK??? omg... |
| 16:05.11 | jeremy_g | the damn thing induces this effect |
| 16:05.17 | Mercestes | jeremy_g, lol |
| 16:05.26 | jeremy_g | Mercestes:i can only ask him for a footage if he made one on that trip |
| 16:05.34 | Mercestes | jeremy_g, hehehe. |
| 16:05.35 | Mercestes | Aww |
| 16:05.41 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:05.43 | Mercestes | ... |
| 16:05.49 | Mercestes | I wonder how many hits "bacon porn" google would turn up |
| 16:05.50 | jeremy_g | i have quit too |
| 16:05.59 | jeremy_g | :D |
| 16:06.02 | jeremy_g | dont try that |
| 16:06.05 | De_Mon | I shutter only thinking about it |
| 16:06.08 | Uatec | [TK]D-Fender, i'm gettin: Forbidden - wrong password on authentication for INVITE to '"Spare Desk" <sip:sparedesk@192.168.232.128>;tag=as15bf0b86' |
| 16:06.14 | Mercestes | google video search "duct tape bondage" It's funny as hell |
| 16:06.23 | Uatec | on the originating asterisk box |
| 16:06.35 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:07.07 | jeremy_g | ..searching |
| 16:07.07 | De_Mon | Uatec wrong password generally means what it says |
| 16:07.07 | Uatec | surely i don't have to authenticate each remote device individually? |
| 16:07.07 | Uatec | i know De_Mon |
| 16:07.07 | Mercestes | may not be safe for work... |
| 16:07.11 | Mercestes | <PROTECTED> |
| 16:07.12 | Uatec | but it's coming from a machine that shouldn't even be asking for the password |
| 16:07.56 | jeremy_g | Mercestes:your words and uatec 's are mixing up and sound funny as hell. .. bondage..machine asking for a wrong password..for INVITE to ''spare .. |
| 16:07.57 | jeremy_g | :D |
| 16:08.13 | Mercestes | jeremy_g, Hrm? oh, we're watching the same videos |
| 16:08.32 | Mercestes | jeremy_g, It's on google videos. It's this one chick, tickling the ever living hell out of these taped up chicks. |
| 16:08.42 | Mercestes | no sex...no nudity....just tape and raspberries. |
| 16:08.44 | [T]ank | trying to install digium TE420. I have run "make clean ; ./configure ; make menuselect" once I finish with all of that have done "make ; make install" that all seems to go fine, but then I do a make config. here is what I am getting: http://pastebin.ca/655674 |
| 16:08.53 | Mercestes | omg I couldn't stop laughing. |
| 16:08.56 | jeremy_g | Mercestes:hehehe |
| 16:08.56 | [T]ank | could anyone suggest what I may be doing wrong? |
| 16:08.56 | Mercestes | ....watching them..I mean. >.> |
| 16:09.15 | jeremy_g | Mercestes:you sure want us to take a break :) |
| 16:09.38 | Mercestes | jeremy_g, everyone needs a little BBIOAB. . |
| 16:10.27 | Uatec | the point is, my call is going: SIPDevice -> Asterisk A - > Asterisk B, then being blocked at B becuase SIPDevice has the wrong username and password |
| 16:10.37 | Uatec | why is it asking the sip device for the username and password |
| 16:10.43 | Uatec | it should be dealing with Asterisk A for that |
| 16:12.54 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 16:12.59 | Mercestes | butt banging is often around bondage. |
| 16:12.59 | *** join/#asterisk ESCulapio_ (n=elvyn@66.44.88.200.l.sta.codetel.net.do) |
| 16:13.00 | Mercestes | .... |
| 16:13.02 | Mercestes | I can't stop. |
| 16:13.03 | Mercestes | :( |
| 16:13.09 | Mercestes | Damn you, [TK]D-Fender! |
| 16:13.35 | Mercestes | bet it was a typo |
| 16:13.53 | xheliox | Anyone having transfer problems since upgrading to 1.4.10.1? (from 1.4.9)... |
| 16:14.01 | Mercestes | xheliox, cisco phones? |
| 16:14.11 | xheliox | Nope. Eyebeam.. |
| 16:14.17 | Mercestes | Then no. =/ |
| 16:14.19 | russellb | xheliox: check the bug tracker ... i think there is an open issue on that |
| 16:14.30 | Mercestes | s/no/yes/ |
| 16:14.34 | Mercestes | >.. |
| 16:14.44 | Mercestes | s/>../>.>/\ |
| 16:14.47 | Mercestes | gah |
| 16:15.57 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 16:16.12 | ESCulapio_ | quien me puede ayudar con una agi, el comando GET DATA no me funciona |
| 16:16.21 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 16:16.22 | ESCulapio_ | who can help me with one agi, commando GET DATA does not work to me |
| 16:17.53 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:19.07 | xheliox | russellb: Hmm. Not seeing one related to me.. I just discovered it seems to be when transfering a call that came into a queue. Extensions dialed directly can transfer just fine,. |
| 16:20.35 | thansen|laptop | does anyone have some suggestions for asterisk/voicexml solutions? |
| 16:21.18 | russellb | xheliox: issue 10415: "Transfers stopped working when migrating from 1.4.9 to 1.4.10" |
| 16:21.33 | xheliox | Yeah, just found it. |
| 16:21.36 | xheliox | That could be my issue... |
| 16:21.52 | xheliox | Still looking. |
| 16:22.06 | *** join/#asterisk rexile (i=elixer@65.207.74.18) |
| 16:22.46 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 16:22.50 | [TK]D-Fender | thansen|laptop: very few people care about VXML. You're best off checking the mailing lists & WIKI. There occasionally news bits about different implementations there. |
| 16:23.05 | jeremy_g | [TK]D-Fender:who are you really, are you russelb |
| 16:23.12 | Mercestes | FENDER!!!! |
| 16:23.20 | Uatec | [TK]D-Fender, do you have experience of * to * SIP connections? |
| 16:23.20 | Mercestes | What does BBIAOB mean? I have to know. |
| 16:23.21 | russellb | jeremy_g: o.O |
| 16:23.40 | thansen|laptop | [TK]D-Fender: why do people not care about it? |
| 16:23.40 | Mercestes | wait, no BBIOAB. |
| 16:23.43 | [TK]D-Fender | Mercestes: it means "its a friggen typo, DLEA WITH IT" :p |
| 16:23.50 | Mercestes | oh. |
| 16:23.54 | jeremy_g | russellb:is that some code language that i dont understand and which means yes |
| 16:23.55 | Mercestes | =/ |
| 16:23.57 | [TK]D-Fender | <recursive_sarcasm> |
| 16:24.00 | Mercestes | I came up with about 20 possibilities. |
| 16:24.14 | russellb | jeremy_g: i am certainly not the same person as [TK]D-Fender |
| 16:24.24 | Mercestes | jeremy_g, they do sleep together, tho. |
| 16:24.26 | russellb | how else could we be on IRC at the same time?! |
| 16:24.39 | jeremy_g | Mercestes:D |
| 16:24.44 | [TK]D-Fender | Mercestes: You have way to much free time (and probably a certain appendage give your recent topics) on your hands :) |
| 16:24.53 | jeremy_g | see, * runs in his family |
| 16:25.28 | russellb | or does his family run in * ? |
| 16:25.40 | Mercestes | my family just runs |
| 16:25.49 | xheliox | russellb: Dumb question, how do I easily unmerge the changes from res_features? |
| 16:25.50 | jeremy_g | yeah, that could be true as well |
| 16:26.04 | xheliox | I'm sure there's some clever svn command I don't know. :) |
| 16:26.09 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:26.13 | jeremy_g | russellb:dont answer dumb questions, its for others. |
| 16:26.22 | jeremy_g | :p |
| 16:26.23 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:26.24 | russellb | xheliox: there is a clever svn command, let me hack it up real quick ... |
| 16:26.55 | xheliox | gracias. |
| 16:27.17 | russellb | cd asterisk-1.4.10.1 ; svn merge http://svn.digium.com/svn/asterisk/tags/1.4.10.1/res/res_features.c http://svn.digium.com/svn/asterisk/tags/1.4.9/res/res_features.c . |
| 16:27.20 | russellb | see if that does it ... |
| 16:27.47 | russellb | that should generate the changes from 1.4.10.1 going back to 1.4.9 and merge them into your local copy |
| 16:28.02 | Mercestes | Wow, that is clever |
| 16:28.17 | xheliox | svn: '.' is not a working copy |
| 16:28.20 | *** join/#asterisk MercestesAlso (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:28.25 | russellb | pwned |
| 16:28.28 | russellb | ok ok ... |
| 16:28.29 | xheliox | so maybe instead of . res/res_features.c ? |
| 16:28.40 | Mercestes | :D |
| 16:28.53 | russellb | cd asterisk-1.4.10.1 ; svn diff http://svn.digium.com/svn/asterisk/tags/1.4.10.1/res/res_features.c http://svn.digium.com/svn/asterisk/tags/1.4.9/res/res_features.c | patch |
| 16:29.13 | russellb | perhaps patch -p0 ... |
| 16:29.44 | xheliox | can't find file to patch at input line 5 Perhaps you used the wrong -p or --strip option? |
| 16:29.54 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:29.58 | xheliox | would it just be easier to wget that file? and put it in the damn directory? ;) |
| 16:30.03 | russellb | no way man |
| 16:30.46 | Mercestes | xheliox, heh. I'd just mask 1.4.10 and emerge asterisk and I'd be all done and happy |
| 16:30.56 | Mercestes | Gentoo wins |
| 16:30.59 | russellb | xheliox: cd into the res dir first |
| 16:31.09 | xheliox | Mercestes: Don't even get me started. :p |
| 16:31.15 | Mercestes | lmao |
| 16:31.16 | russellb | gentoo is for n0000000bs |
| 16:31.46 | xheliox | Now that worked... thanks russellb. Now hopefully that will fix the transfer problem, I'll let you know in a jiffy |
| 16:31.57 | Mercestes | I see it this way. I want to drive a car. I want to put in a key, turn it a little, put the little indicator on "D" and stomp it and go. |
| 16:32.03 | thansen|laptop | [TK]D-Fender: so why do people not care about vxml? |
| 16:32.14 | Mercestes | I don't want to hone my pistons, rearrange my sparkplugs, and rotate my tires first everytime I want to go somewhere. |
| 16:32.50 | russellb | thansen|laptop: a better question may be why *do* people are about it :) |
| 16:32.57 | [TK]D-Fender | thansen|laptop: its another layer on top of what * does already, whats the point? |
| 16:33.43 | thansen|laptop | so what's everyone's preferred method of creating ivr-esque stuff? I'm developing in php |
| 16:33.58 | thansen|laptop | I just figured vxml would simplify the process |
| 16:34.01 | [TK]D-Fender | thansen|laptop: Have you even installed and WORKED with * yet? |
| 16:34.07 | thansen|laptop | yes |
| 16:34.23 | thansen|laptop | I have some scripts running etc |
| 16:34.28 | [TK]D-Fender | thansen|laptop: well then you should know how to build an IVR for it then. |
| 16:34.37 | thansen|laptop | agi scripts...it just seems cumbersome |
| 16:34.52 | [TK]D-Fender | thansen|laptop: extensions.conf = everything |
| 16:34.59 | *** join/#asterisk myiagy (n=myiagy@201.64.81.78) |
| 16:35.22 | thansen|laptop | extensions.conf = cumbersome and far from interactive with dbs etc |
| 16:35.59 | MrTelephone | holy shit my asterisk box has been running pretty solid these days |
| 16:36.04 | [TK]D-Fender | thansen|laptop: for the little bits that require it, thats what AGI is for. |
| 16:36.38 | thansen|laptop | [TK]D-Fender: that's the thing, *most* of my stuff is gonna be based off data in a db |
| 16:36.39 | MrTelephone | thanks guys for all your contributed support for making it happen |
| 16:37.17 | MrTelephone | thansen, cron a script to create an extensions-1.conf which is included from the original |
| 16:37.18 | [TK]D-Fender | thansen|laptop: Poor you. If the pain becomes too much remember.... thats why the windows don't open ;) |
| 16:37.48 | MrTelephone | is someone going to program includes into voicemail.conf? |
| 16:38.07 | [TK]D-Fender | MrTelephone: You already can, and have been able to for AGES |
| 16:38.14 | thansen|laptop | [TK]D-Fender: :) |
| 16:38.15 | MrTelephone | since when? |
| 16:38.30 | MrTelephone | i couldn't do it with 1.2.12 |
| 16:38.36 | MrTelephone | im using 1.2.23 or something now |
| 16:38.47 | [TK]D-Fender | MrTelephone: At least somewhere in 1.2.... you'd see this in FreePBX installs. |
| 16:38.57 | thansen|laptop | I'm just looking for the "correct" solution since I'm just starting to build this |
| 16:38.58 | [TK]D-Fender | MrTelephone: AFAIK its ALWAYS been there. |
| 16:39.12 | MrTelephone | no way there was an issue with it |
| 16:39.26 | MrTelephone | so i had to make my script check the orginal file then make the changes needed |
| 16:39.35 | MrTelephone | unless I'm on crack |
| 16:39.54 | [TK]D-Fender | MrTelephone: I would never do a drug namedd after a part of my ass...... |
| 16:40.12 | thansen|laptop | lol |
| 16:40.17 | Uatec | <PROTECTED> |
| 16:40.22 | MrTelephone | you'd have to be gay to commit suicide off the grand canyon then |
| 16:40.44 | MrTelephone | bad joke sorry |
| 16:40.54 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 16:41.05 | MrTelephone | check out msn's bad driver article today.. it made me laugh pretty hard |
| 16:42.16 | MrTelephone | You're a committed Greenie, a tireless apostle against global warming, evil corporations and any SUV. You flaunt your hippie-vegan lifestyle and fastidious demands for organic food. Your mode of transport? A decrepit, Woodstock-era VW bus that spews more pollution than a dealership full of Hummers. Drop one circle if the ashtray betrays one final hypocrisy: a pack-a-day Marlboro habit. |
| 16:42.39 | MrTelephone | anyways |
| 16:42.41 | MrTelephone | sorry for the spam |
| 16:43.24 | MrTelephone | goes on about how people have big V8 SUvs and the wife has one too to go shopping |
| 16:43.24 | HarryR | MrTelephone, VW busses are relatively economic, considering they last for so long :) |
| 16:43.24 | MrTelephone | haha |
| 16:43.24 | rexile | my marlboro habit is closer to 2 packs a day... |
| 16:43.25 | rexile | and i hate al gore |
| 16:43.26 | MrTelephone | i know this is just some guys opinion |
| 16:43.28 | rexile | so what do i win? |
| 16:43.30 | rexile | :) |
| 16:43.33 | rexile | err |
| 16:43.42 | elixer | much better |
| 16:43.46 | MrTelephone | i dunno im trying to go greener myself but its like pissing in the ocean |
| 16:44.13 | HarryR | I used to recycle a lot |
| 16:44.22 | HarryR | then I realized life sucks sometimes and stopped :\ |
| 16:44.58 | HarryR | very much somebody that doesn't give a damn what'll happen after I die ;) |
| 16:45.43 | HarryR | hopefully it'll kick in just in time to get regular 20C winters and 35C summers ;) |
| 16:46.45 | elixer | i can't convert that to fahrenheit in my head |
| 16:46.52 | elixer | i'll assume thats both very cold and very hot |
| 16:47.19 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 16:47.21 | elixer | err |
| 16:47.24 | *** join/#asterisk saftsack (n=saftsack@217.224.75.21) |
| 16:47.35 | elixer | vor something |
| 16:47.36 | *** join/#asterisk Op3r (n=Op3r@121.97.242.81) |
| 16:48.24 | HarryR | uh, 20C is my perfect temperature |
| 16:48.34 | *** join/#asterisk shareenergy (i=shareene@62.169.80.74.rev.optimus.pt) |
| 16:48.42 | HarryR | 35C is a perfect lie-in-the-garden-with-a-beer excuse |
| 16:49.04 | shareenergy | anyone know why rxfax in my case makes multiple tiff, instead of one? |
| 16:49.09 | [TK]D-Fender | elixer: 68 / 95 respectively |
| 16:49.10 | x86 | 22C is the perfect temperature |
| 16:49.23 | shareenergy | JT |
| 16:49.26 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 16:49.41 | HarryR | why 22C? |
| 16:49.56 | coppice | shareenergy: rxfax only makes one TIFF file per call. multiple files indicates multiple calls |
| 16:50.07 | x86 | HarryR: because i like 22C? :P |
| 16:50.16 | MrTelephone | haha |
| 16:50.17 | MrTelephone | yeah |
| 16:50.26 | MrTelephone | 35 dry and 20c humid or about the same |
| 16:50.33 | x86 | HarryR: why 20C? :p |
| 16:50.36 | coppice | 26C is cheaper to maintain, and is OK |
| 16:50.43 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 16:50.50 | HarryR | x86, because it's been scientificly proven to be the ideal temperature for servers |
| 16:50.54 | HarryR | and therefore, humans too |
| 16:51.02 | x86 | hahaha that must be correct |
| 16:51.05 | MrTelephone | im trying to do a cost comparison between fuel oil, propane, electric, and pellot stove |
| 16:51.07 | x86 | :p |
| 16:51.11 | shareenergy | coppice it also make one, the problem is that the file gets mixed with many things instead of a normal fax paper |
| 16:51.13 | MrTelephone | to keep my asterisk box warm |
| 16:51.14 | elixer | mmmm, 68 F |
| 16:51.21 | MrTelephone | :-/ |
| 16:51.21 | elixer | that'd be nice right now |
| 16:52.22 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:52.51 | coppice | shareenergy: I have no idea what that is supposed to mean |
| 16:56.56 | x86 | hmm |
| 16:57.10 | x86 | yall ever hear of a resturaunt called "stake and shake"? |
| 16:57.31 | x86 | i just came up with an awesome idea for a "theme resturaunt" / titty bar... |
| 16:57.35 | x86 | "shank a skank" |
| 16:57.45 | shareenergy | coppice i can't have a normal page... it cuts the image in the fax |
| 16:57.53 | tzanger | x86: heh |
| 17:00.30 | *** join/#asterisk LoneShadow (n=a@c-76-103-55-28.hsd1.ca.comcast.net) |
| 17:02.19 | xheliox | russellb: Reverting fixed the problem in res-features, I'll update the bug, just thought you should know. Thanks for your help, as always. Digium needs a developer tip jar, like a bartender. :) |
| 17:04.28 | Juggie | red bull is allways accepted. |
| 17:05.21 | xheliox | the next case of redbull is on me then :p |
| 17:07.17 | *** join/#asterisk anthm (n=anthm@adsl-68-74-96-61.dsl.milwwi.ameritech.net) |
| 17:07.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:09.42 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 17:11.20 | shareenergy | anyone can help me with fax problem on asterisk, with hfc card |
| 17:11.21 | shareenergy | ? |
| 17:13.03 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 17:13.41 | coppice | you aren't really describing your problem very well. it seems you are sending a multi-page fax to rxfax, and end up with something unexpected in the TIFF file. you haven't really explained what is wrong with the TIFF file |
| 17:16.59 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 17:18.23 | Yourname` | Hello. I have agent 7001 added in agents.conf and queues.conf, yet I get this warning after entering the password during Agent login: WARNING[3454]: chan_agent.c:1866 __login_exec: Extension '7001' is not valid for automatic login of agent '7001' ->And right then on the phone, it says please enter another extension.. what's going wrong now? |
| 17:19.39 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:22.39 | hi365 | when i try to play a "queue-thankyou" message to a queue it plays the callers position in the queue. i dont want to playthat to the caller. how do i stop it? |
| 17:22.53 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 17:23.43 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:23.44 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:30.44 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 17:34.39 | [TK]D-Fender | hi365: There is another parm in queues.conf that clearly controls that. Go read the sample again. |
| 17:34.40 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 17:34.53 | ESCulapio_ | who can help me with one agi, commando GET DATA does not work to me |
| 17:34.58 | ESCulapio_ | please |
| 17:35.02 | [TK]D-Fender | Yourname`: Check your call-back context |
| 17:35.40 | hi365 | [TK]D-Fender: it set NOT to anounce the caller position. it doesnt play the "you are caller number" file, it just syas the position number (eg. "2 please hold for the next...") |
| 17:35.55 | Yourname` | [TK]D-Fender: What do you mean call-back context? |
| 17:36.03 | [TK]D-Fender | hi365: Go verify the contents of the recording outside of the queue. |
| 17:36.37 | [TK]D-Fender | Yourname`: agents usually get callved through the dialplan. if you can't log them in under a certain extension, make sure it exists where you are telling the queue to dial it out to. |
| 17:36.46 | hi365 | [TK]D-Fender: which recording? |
| 17:37.04 | [TK]D-Fender | hi365: "queue-thankyou" obviously. |
| 17:37.32 | ESCulapio_ | help my please with appli GET DATA |
| 17:38.01 | hi365 | [TK]D-Fender: what am i testing it for? i works great. the problem is that the calleer postion is being palyed befor it, even though the option is NOT set |
| 17:38.08 | hi365 | (i.e. set not to paly) |
| 17:38.18 | [TK]D-Fender | hi365: pastebin CLI output, and your queues config |
| 17:38.26 | hi365 | k. soon |
| 17:38.27 | Yourname` | [TK]D-Fender: Under the [agents] context in agents.conf, the agents are defined there. And under the context of [testq] agents are defined there too. |
| 17:39.24 | [TK]D-Fender | Yourname`: when you log in agents that means they aren't static devices and are dialed through your DIALPLAN. Go verify that 7001 is in the context you told the login app to use |
| 17:41.12 | s34n | which timer is best to use when you don't have digium hardware? |
| 17:41.40 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 17:43.10 | [TK]D-Fender | s34n: 2.6 Kernel I would think... |
| 17:46.12 | s34n | [TK]D-Fender: using ztdummy? or replacing rtc? |
| 17:46.31 | [TK]D-Fender | s34n: ZTDUMMY *uses* rtc |
| 17:46.53 | [TK]D-Fender | s34n: it will use rtc otherwise UHCI if not available |
| 17:47.18 | s34n | [TK]D-Fender: so that is preferable to zaprtc? |
| 17:48.19 | [TK]D-Fender | s34n: Only references I've seen to zaprtc are ANCIENT |
| 17:48.31 | s34n | k. thanks. |
| 17:49.29 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 17:50.01 | [TK]D-Fender | s34n: Got a problem running standard ztdummy? |
| 17:51.26 | s34n | no. I just hadn't seen the zaprtc stuff, and thought it might be new. |
| 17:51.47 | [TK]D-Fender | s34n: Stop looking for trouble then.. you've probably got more than enough as it is :) |
| 17:51.56 | s34n | :) |
| 17:52.16 | s34n | well, that brings up my next poke... |
| 17:53.00 | s34n | is ael worth the dabble? |
| 17:53.38 | Qwell[] | s34n: of course |
| 17:54.33 | s34n | is ael becoming the new enlightened path, or is it just another idea? |
| 17:55.59 | s34n | the wiki says ael is still experimental, so I guess that answers my question |
| 17:57.23 | hi365 | [TK]D-Fender: http://pastebin.ca/655807 |
| 17:57.51 | [TK]D-Fender | s34n: Few people actual give a shit about AEL, and support is somewhat sparse. It technically just parses back to internal logic so it doesn't really offer anything new. If you have a particularly messy bit of dialplan, it might be worth it for certain bits, but on a whole I see no point to it. |
| 17:57.59 | Mercestes | s34n: I would say ael doesn't give you anything you can't do in extensions.con |
| 17:58.34 | Yourname` | [TK]D-Fender: I got it! :) |
| 17:58.36 | [TK]D-Fender | hi365: debug info looks like garbage to me. Real CLI output please... |
| 17:58.37 | elixer | codefreeze: opinions? :) |
| 17:58.43 | elixer | heh |
| 17:59.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@79.179.135.2) |
| 17:59.12 | hi365 | [TK]D-Fender: DAMN your picky! hold on |
| 17:59.35 | [TK]D-Fender | hi365: I asked for something and you feel you susbitute whatever you want for it! |
| 17:59.45 | [TK]D-Fender | substitute* |
| 17:59.47 | [TK]D-Fender | kjasdkjasldhasd |
| 18:01.59 | Yourname` | [TK]D-Fender: Why does it sometimes do a "Started music on hold" and right away "Stopped music on hold" and then say NOTICE[3607]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! -> What else did I mess up on? |
| 18:02.03 | hi365 | [TK]D-Fender: http://pastebin.ca/655813 |
| 18:02.43 | hi365 | yeh, i didnt notice that befor |
| 18:02.55 | [TK]D-Fender | ... |
| 18:03.22 | hi365 | that you wanted cli and not logfile |
| 18:03.56 | *** join/#asterisk eject_ck (i=eject_ck@galling.glancer.volia.net) |
| 18:04.20 | *** join/#asterisk saftsack (n=saftsack@217.224.75.21) |
| 18:04.39 | [TK]D-Fender | hi365: You are using the wrong options for a periodic announce... |
| 18:04.50 | hi365 | go on |
| 18:04.53 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 18:05.32 | [TK]D-Fender | How often to announce queue position and/or estimated holdtime to caller (0=off) announce-frequency = 90 |
| 18:05.47 | [TK]D-Fender | How often to make any periodic announcement (see periodic-announce) periodic-announce-frequency=60 |
| 18:06.23 | [TK]D-Fender | [root@localhost configs]# cat queues.conf.sample|grep perio |
| 18:06.24 | [TK]D-Fender | ; How often to make any periodic announcement (see periodic-announce) |
| 18:06.26 | [TK]D-Fender | ;periodic-announce-frequency=60 |
| 18:06.28 | [TK]D-Fender | ;periodic-announce = queue-periodic-announce |
| 18:06.33 | [TK]D-Fender | THAT is what you should be using |
| 18:06.37 | hi365 | [TK]D-Fender: thanks ill ahve a look |
| 18:06.50 | [TK]D-Fender | hi365: All in the sample files... READ THEM :> |
| 18:07.14 | [TK]D-Fender | hi365: Standard announce should be disabled. |
| 18:11.45 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 18:19.30 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:20.45 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 18:31.20 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 18:34.59 | *** join/#asterisk ReDNeQ (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 18:38.14 | *** join/#asterisk ygguh2 (n=concilio@ool-44c5e3c2.dyn.optonline.net) |
| 18:39.42 | ygguh2 | looking for skype help. W would like to be a ble to use our dev asterisk server to connect to skype users. Im looking for a skype channel app. Thanks. |
| 18:41.44 | wothinn | ygguh2: You and everyone else who uses Skype. It doesn't exist. If you search the web, you'll find some gateway products, but nothing as useful or fancy as a chan_skype.so, sadly. |
| 18:41.46 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:43.12 | ygguh2 | wothinn, thanks for the info. I've been searching the web for two days now with out any REAL usable results. there are two companies which utilize skypes sdk, but, you must us an x session/vnc to the server to make it work. and Thats not what we want to do. |
| 18:43.53 | *** join/#asterisk implicit_ (n=implicit@vc240149.vpn.uci.edu) |
| 18:43.56 | wothinn | Unfortunately, it's the best you're going to get. That's the fun of trying to talk to a proprietary product. |
| 18:44.15 | ygguh2 | true |
| 18:44.45 | [TK]D-Fender | ygguh2: All attempts to use Skype with * are ugly hacks. Get over it and go get a REAL soft-phone. |
| 18:44.47 | wothinn | I think there's a bounty somewhere on a chan_skype.so, but I wouldn't hold out much hope of it happening soon. It'll take a complete protocol reverse-engineering job. |
| 18:45.18 | *** join/#asterisk mtaht4 (n=m@cpe-065-190-150-008.nc.res.rr.com) |
| 18:46.19 | ygguh2 | [TK]D-Fender, we currently have about 40 linksys spa942, over 100 web bases sip softphones for our operators. Some of our customers have been asking us to connect to them over skype. Its not by choice that I ask about skype. |
| 18:51.32 | karleeto | ygguh2: i've got a few linksys spa941's, i guess they're OK phones.. but damn they don't hold a candle to our Polycom's.. of course they were half the price :) |
| 18:53.16 | ygguh2 | thats why we purchased them. while they work well, Im unable to program all four buttons, and using tftp is just horrible, it takes more time to configure the phone for tft them anything else. we had to open a few phones to install foam to cut down on the feedback when using the speaker phone. |
| 18:53.29 | shareenergy | anyone know what is the best rxgain, and txgain to receive fax on a HFC card? |
| 18:55.41 | NOT_guru | I am having some oddness when using cisco phones ( 7940 - 7960 ) in SIP mode, I get them to register to asterisk on a local subnet ( 10.0.0.X ), but can't get them to register to a "offnet" subnet ( 10.0.7.X ) suggestions? |
| 18:55.45 | [TK]D-Fender | karleeto: Where are you located? |
| 18:56.11 | [TK]D-Fender | karleeto: You can get an entry level Polycom on par with linksys for pricing in North America.... |
| 18:56.57 | Mercestes | NOT_guru, try setting the natting in the cisco phone and turn nat=off in asterisk |
| 18:57.25 | NOT_guru | will look now mercestes.. thank you brb |
| 18:59.41 | NOT_guru | I am sorry for my ignorance but what NAT settings you think I need on the phone? http://www.pastebin.ca/655933 |
| 19:01.59 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:02.55 | flujan | hi guys, i am trying to install asterisk on my mac... It is asking for fetch or wget in the make install... I installed wget and fetch using macports but asterisk doesn't find them. Anyone already have this error? |
| 19:03.43 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-7986dc20ba45039c) |
| 19:05.50 | NOT_guru | nm I seem to have corrected the issue |
| 19:05.55 | [TK]D-Fender | flujan: "unload chan_masochism.so" |
| 19:06.02 | *** join/#asterisk sysreq (n=sysreq@197.36-ppp.3menatwork.com) |
| 19:06.05 | NOT_guru | thank you so nuch for your input Mercestes |
| 19:06.20 | flujan | [TK]D-Fender: lol... I use a mac to work. :P |
| 19:06.24 | NOT_guru | it is appreciated when someone offers to lend a hand |
| 19:06.31 | flujan | but the servers run Slackware. :D |
| 19:07.18 | MrTelephone | i ordered like 4 copies of office 2007 oem and they don't even come with cds |
| 19:07.26 | MrTelephone | you need the god damn pre install kit now |
| 19:08.21 | bkruse | [TK]D-Fender: ello |
| 19:09.01 | [TK]D-Fender | bkruse: y0 |
| 19:09.53 | [TK]D-Fender | bkruse: You know... I've been thinking that with direct access to Moko's GSM you could easily install * on there for processing, and then use a local-host IAX client from there :) |
| 19:10.20 | [TK]D-Fender | bkruse: 0 transcoding (assuming GSM610compatability (maybe hardware transcoded at the client level at worst) |
| 19:10.48 | bkruse | [TK]D-Fender: There is a project, that I am currently signed up as a dev for doing asterrisk on the openmoko |
| 19:10.59 | bkruse | I, personally, do not really see why as more of a proof of concept and a cool thing to do ; |
| 19:11.00 | bkruse | ;] |
| 19:11.30 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 19:11.35 | bkruse | running asterisk and an iaxclient on top of all the other stuff on the neo? I am not sure :/ |
| 19:11.46 | *** join/#asterisk livesN[box] (n=chadkous@165.236.120.14) |
| 19:12.09 | livesN[box] | hey guys -- Is there a way to get a dynamic agent's member name to show in queue logs? |
| 19:12.45 | livesN[box] | seems like I should be able to login an agent and track them by their name rather than just what phone they are at (since agents move around sometimes between desks) |
| 19:14.03 | [TK]D-Fender | bkruse: Local VM, intelligent call handling (single client that can try IAX first, then GSM, etc), local IVR for incoming calls (privacy, etc). I can come up with TONS of great uses. |
| 19:14.26 | bkruse | [TK]D-Fender: You think the hardware could handle that? |
| 19:14.34 | bkruse | This is true, we could do a lot of cool integration with such |
| 19:14.36 | [TK]D-Fender | bkruse: The public release CPU is going to be a lot bigger... might work. |
| 19:14.50 | [TK]D-Fender | bkruse: It'd only be viable if you can eliminate transcoding. |
| 19:14.53 | bkruse | [TK]D-Fender: ohrly? Like what? |
| 19:15.14 | bkruse | What speed? cache? memory? |
| 19:15.19 | bkruse | I just wonder if itll be lagless enough to use |
| 19:15.31 | [TK]D-Fender | bkruse: 266MHz Samsung System on a Chip (SOC) <- P0 |
| 19:15.37 | bkruse | [TK]D-Fender: hmmm |
| 19:15.53 | [TK]D-Fender | bkruse: 400+MHz (can't remember the type) <- P1 |
| 19:16.03 | bkruse | [TK]D-Fender: woah, interesting... |
| 19:16.11 | bkruse | That would def do the job for one, even multiple calls |
| 19:16.41 | [TK]D-Fender | bkruse: Wouldn't need too much in terms of multiple calls... enough to handle BASIC use of GSM VS Wifi |
| 19:16.46 | Mercestes | NOT_guru, Your welcome |
| 19:16.48 | *** join/#asterisk dlynes_laptop (n=dlynes@216.113.200.191) |
| 19:16.50 | bkruse | [TK]D-Fender: right |
| 19:16.51 | [TK]D-Fender | bkruse: Great way to maximise your calling value |
| 19:17.18 | bkruse | [TK]D-Fender: This is true, you could do a lot of sweet integration with sending jabber messages, VM's, queues, tons of thigns |
| 19:17.22 | bkruse | anything asterisk has |
| 19:17.28 | [TK]D-Fender | bkruse: using * on the OM itself allows you a "single dialer" option as it can directly interface with it |
| 19:17.40 | bkruse | right |
| 19:17.46 | Qwell[] | the openmoko devs were pretty clear about asterisk |
| 19:17.57 | [TK]D-Fender | bkruse: So yeah it SEEMS wierd to want to put * on it... but the benifits really ARE cool. |
| 19:17.59 | Qwell[] | they said (in no uncertain words) that it is a dumb idea |
| 19:18.00 | bkruse | Qwell[]: .....who? and what did they say? |
| 19:18.12 | Qwell[] | bkruse: there was a mailing list thread on it |
| 19:18.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 19:18.20 | bkruse | Qwell[]: sean said it would be awesome |
| 19:18.20 | [TK]D-Fender | Qwell[]: two words "Mission Accomplished" <- |
| 19:18.28 | Qwell[] | bkruse: Sean probably "gets it" |
| 19:18.29 | Qwell[] | so yeah |
| 19:18.30 | bkruse | but I think he was thinking more proof of concept and "coolness" |
| 19:18.42 | [TK]D-Fender | Qwell[]: I've jsut come up with TONS of great reasons to do it. |
| 19:18.44 | Qwell[] | well, no, I mean...I think it'd be a great idea |
| 19:18.48 | bkruse | not so much a tool, I think the iax client alone would be sweet. |
| 19:18.55 | Qwell[] | I'm not saying don't do it... I'm just saying you might get some push back :D |
| 19:19.05 | `Sean | ? |
| 19:19.05 | [TK]D-Fender | Qwell[]: A GOOD IAx client would go a long way, but * adds that little bit extra. |
| 19:19.28 | *** join/#asterisk Strom_M (n=strom@pool-64-222-110-211.burl.east.verizon.net) |
| 19:20.15 | bkruse | Qwell[]: I am not doing that |
| 19:20.28 | Qwell[] | bkruse: I was going to :p |
| 19:20.33 | bkruse | I just think if they do it, you can configure the iax client to go to localhost, but as far as I am concerned, I am doing something else |
| 19:20.45 | bkruse | Qwell[]: you should still, no one has done anything on the project, its just sitting there |
| 19:20.46 | [TK]D-Fender | bkruse: Here it is : Faster CPU - S3C2442/400 |
| 19:21.07 | Qwell[] | I want to use those accelerometers somehow with Asterisk :p |
| 19:21.07 | bkruse | hmm |
| 19:21.50 | [TK]D-Fender | Qwell[]: (10m/s^2) * 3s = Asterisk auto-segfault? ;) |
| 19:21.57 | Qwell[] | huh? |
| 19:22.23 | [TK]D-Fender | Qwell[]: 3s drop to pavement = * HISTORY <--------- |
| 19:22.34 | Qwell[] | ahh |
| 19:27.24 | [TK]D-Fender | bkruse: Honestly if the IAX Client is really good with dialplans it would be enough to do LCR between IAX&WIFI + GSM as fallback (with some parsing) |
| 19:27.34 | [TK]D-Fender | bkruse: That alone would make a LOT of people happy. |
| 19:27.52 | [TK]D-Fender | bkruse: Could jsut trigger a dial out of the native Dialer app) |
| 19:28.22 | bkruse | based on if the configured asterisk box is availible? |
| 19:28.30 | bkruse | I would just say if its registered than pass calls through iax, if not use gsm |
| 19:28.52 | MrTelephone | i think u should develop the mgcp and sell th eproject to cisco and cash out as millionaires |
| 19:28.59 | codefreeze | elixer: As to AEL, I disagree with [TK]D-Fender and Mercestes: they overlook several advantages, that apparently mean little to them, although, I hoped on more than one occasion to convince them otherwise! |
| 19:29.01 | MrTelephone | heh |
| 19:29.23 | livesN[box] | is there a way I can log in an agent and track them by their name rather than just what phone they are at (since agents move around sometimes between desks) |
| 19:29.45 | bkruse | MrTelephone: interesting..... :] |
| 19:29.55 | Mercestes | codefreeze, that's been eating at you for awhile, hasn't it? |
| 19:29.56 | [TK]D-Fender | bkruse: Exactly |
| 19:30.05 | MrTelephone | asterisk eats call manager for breakfast |
| 19:30.29 | [TK]D-Fender | codefreeze: Of course YOU'D beg to differ :) Feel free to re-enlighten me. My opinions are not set in stone. |
| 19:30.46 | [TK]D-Fender | MrTelephone: And dies of food-poisoning by lunch ;) |
| 19:30.48 | Mercestes | mostly because he's too stoned to remember his opinions |
| 19:34.41 | codefreeze | elixer: Mercestes: [TK]D-Fender: sorry I was slow to respond, got back from lunch, sort of, and saw the blue tab. |
| 19:34.44 | *** join/#asterisk guillote_GNU (n=bancaria@host228.190-30-60.telecom.net.ar) |
| 19:35.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 19:37.37 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 19:41.16 | codefreeze | As far as recapping what I said before, I don't think I have much more to add to the mix, [TK]D-Fender; Mercestes; but you aren't being factually correct in saying that AEL adds nothing that you don't already have in extensions.conf... |
| 19:41.44 | Mercestes | codefreeze, Exactly in what instance is that not correct? |
| 19:42.57 | [TK]D-Fender | codefreeze: AEL2 is now standard, and considered non-experimental? |
| 19:43.44 | codefreeze | Mercestes: Beyond some spotty error/warnings from the extensions.conf reader, how many/much checking is done for you? AEL has many, many checks. |
| 19:44.09 | Mercestes | codefreeze, your not answering my question. |
| 19:44.29 | codefreeze | [TK]D-Fender: I think we officially lifted the 'experimental' label when we put ael2 in place. |
| 19:44.29 | [TK]D-Fender | Mercestes: unload chan_troll.so |
| 19:44.40 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:44.51 | codefreeze | Mercestes: that was just item #1. |
| 19:45.20 | [TK]D-Fender | codefreeze: Where is the best doc for it? |
| 19:45.54 | codefreeze | Item #2: spotty support? Issues usually are handled very quickly. A couple days on the average. |
| 19:46.04 | [TK]D-Fender | codefreeze: (in terms of completeness of syntax) |
| 19:46.16 | MrTelephone | I have a serious spam problem here |
| 19:46.28 | MrTelephone | I keep getting 20 emails a day from asterisk-dev, j/k |
| 19:46.29 | MrTelephone | heh |
| 19:46.36 | codefreeze | [TK]D-Fender: right now, see the AEL2 wiki on voip-info |
| 19:46.39 | MrTelephone | but there is viagra ones too from russellb or something |
| 19:47.11 | codefreeze | [TK]D-Fender: completeness of syntax? When is anything complete? |
| 19:47.13 | MrTelephone | you know who has good support, Carrier Access.. man they are good |
| 19:47.31 | MrTelephone | Digium, sangoma, carrier access, RAD, good support |
| 19:47.36 | [TK]D-Fender | codefreeze: Well as you parse backwards some docs may be better than others. |
| 19:48.35 | codefreeze | I usually update the wiki first; I think there's a copy in the doc/ dir, a little less frequently updated |
| 19:49.42 | codefreeze | Item #3: stand alone compiler: You don't have to feed your dialplan to a running asterisk to find out if you borked things. |
| 19:53.20 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 19:53.21 | codefreeze | [TK]D-Fender: I spewed about a dozen items a few days back along this line, somewhere... was it here? Or in #asterisk-dev, that are advantages to using AEL. Among the number are security fixes, insulation from changes to extension.conf format; structured programming; but I'm sure I've brought these up before. |
| 19:54.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:54.53 | [TK]D-Fender | codefreeze: security fixes? Can you elaborate a bit on this? Structured is good for complex input I admit. Has it really insulated much by way of extensions.conf changes? there've been remarkably few, and AEL itself is on a 2nd revision now... |
| 19:55.16 | [TK]D-Fender | codefreeze: For a DB driving IVR i can see the value of it though. |
| 19:55.36 | codefreeze | It'll be around a while. |
| 19:56.04 | russellb | codefreeze: you just need to write up a "why AEL?" document, drop it in the doc dir, and then link to it when you need to go through the list :) |
| 19:56.27 | lmadsen | codefreeze: you brought all this up in here about 2 days ago |
| 19:56.31 | lmadsen | just FYI :) |
| 19:56.55 | russellb | MrTelephone: you're getting viagra adds from me? that's certainly odd. |
| 19:56.56 | codefreeze | [TK]D-Fender: If you use GotoIf, you should be using AEL. Or if your extensions.conf is more than maybe 20 lines long. (trivial, in other words). |
| 19:57.05 | [TK]D-Fender | lmadsen: His wheel gets rounder every time ;) |
| 19:57.12 | *** join/#asterisk DarylVOIP (n=daryl@c-68-32-227-165.hsd1.pa.comcast.net) |
| 19:57.18 | codefreeze | lmadsen: thought so, thanks |
| 19:57.31 | lmadsen | codefreeze: hahaha... so you're saying I should be using AEL :) |
| 19:57.51 | lmadsen | I'd like to write a section on AEL in TFoT |
| 19:58.08 | russellb | lmadsen: yes, you should |
| 19:58.13 | jcolp | russellb: so that's the other business you operate! |
| 19:58.31 | russellb | your dialplans are like self inflicted pain! |
| 19:58.52 | codefreeze | [TK]D-Fender: maybe I should do the ~AEL thing in here, and flood the page each time someone (eh-hem) says it doesn't add anything. |
| 19:58.56 | lmadsen | russellb: nah... just might be painful for people with lesser dialplan fu :) |
| 19:59.07 | lmadsen | codefreeze: oh you're cheeky |
| 19:59.12 | lmadsen | I like it! :) |
| 20:00.00 | [TK]D-Fender | lmadsen>I'd like to write a section on AEL in TFoT <- funny thing to here NOW as its about to hit print ;) |
| 20:00.02 | lmadsen | russellb: only reason I didn't write something for the 2nd edition was because I ran out of time |
| 20:00.12 | lmadsen | [TK]D-Fender: I didn't mean for this edition |
| 20:00.15 | russellb | lmadsen: it's all good |
| 20:00.33 | [TK]D-Fender | lmadsen: Its absent from the one ABOUT to be released? |
| 20:00.35 | codefreeze | lmadsen: lol... I don't want to insult anybody, but sometimes I feel like I need to stick up for poor AEL |
| 20:00.40 | tzanger | haha |
| 20:01.04 | lmadsen | [TK]D-Fender: and it was in the things we wanted to write, but I ended up writing a chapter on connecting to relational databases and func_odbc instead |
| 20:01.05 | codefreeze | lmadsen: Yes, I agree, you need a big, fat chapter on AEL. |
| 20:01.28 | Yourname` | Hi codefreeze. |
| 20:01.40 | lmadsen | codefreeze: I agree. AEL would probably make my dialplans a bit easier to read |
| 20:01.40 | crimethinker | is that kind of like brainfreeze? |
| 20:01.58 | codefreeze | Yourname`: hey! how do? |
| 20:02.03 | codefreeze | crimethinker: totally |
| 20:02.17 | Yourname` | codefreeze: Good so far, you? |
| 20:02.18 | lmadsen | if you want a real advantage to AEL, it's the ability to remove about 1/3 to 1/5 of the characters you need to type: i.e. exten => _1NXXNXXXXXX,n,..... on everyline |
| 20:02.22 | DarylVOIP | Anyone know if passing a ringing channel to an AGI will always answer it? I'm trying to use an AGI to figure out if a user is callback authorized or not and now that I have the dial plan passing to the AGI bin before I dump the call I can see that its billing (on my cell that I'm calling from) |
| 20:02.40 | DarylVOIP | The goal is never to answer the channel. |
| 20:02.44 | codefreeze | Yourname`: pretty good for an old fat guy |
| 20:03.12 | codefreeze | lmadsen: Ooooo, oooo! I'll add that to the list. |
| 20:03.15 | Yourname` | codefreeze: Good haha. Remember the other day you were telling me about the "call failed to go through, reason 0" .. you said it's basically NORMAL_NETWORK_CONGESTION -> I was wondering if reason 0 = disconnect or timeout or what exactly. |
| 20:03.16 | russellb | codefreeze: fat? lol .. |
| 20:03.23 | russellb | codefreeze: and you're not old either :) |
| 20:04.03 | codefreeze | russellb: well, as far as I can tell, I'm doing my best to become both... I guess I'm failing... |
| 20:04.18 | *** join/#asterisk galeras (n=galeras@190.90.27.21) |
| 20:05.17 | russellb | dang, i'm doing pretty well at both |
| 20:05.27 | galeras | howdy |
| 20:05.46 | codefreeze | Yourname`: I think disconnect might fit in the description, but NO_ANSWER should happen for dialing timeout |
| 20:05.57 | lmadsen | codefreeze: glad I could help :) |
| 20:06.13 | Yourname` | codefreeze: Is there any documentation at all about this somewhere? :S |
| 20:06.40 | galeras | plz, give me any suggestion why ${CHANNEL(callgroup)} is empty. I have callgroup=1 in zapata.conf |
| 20:06.41 | codefreeze | russellb: ha, sounds like you might rate a failure on both accounts, like me. |
| 20:07.51 | codefreeze | Yourname`: to quote the famous saying: "Use the Source, Luke!" -- it's not the easiest to read, it's encrypted to something like ancient runes, but it does give you some answers sometimes |
| 20:08.35 | Yourname` | codefreeze: As a newb, it might be a little hard for me to track it down, lol.... as long as it's commented it should be fine. Which file do you think will have the most info? |
| 20:09.07 | galeras | is there any way to know from dialplan the span or group of the current zap channel? |
| 20:09.18 | codefreeze | Yourname`: use grep -r "The error message you see", that's how I start. |
| 20:09.28 | Yourname` | ok |
| 20:10.17 | codefreeze | Yourname`: oops, put a . or the dir where the source is at the end of that grep. |
| 20:10.47 | Yourname` | codefreeze: grep -r "call failed to go through" . -> should do it? |
| 20:10.54 | codefreeze | Yourname`: grep is your friend. I once saw a T-shirt: grep meaning life |
| 20:11.22 | *** join/#asterisk gardo (n=gardo@125.212.12.90) |
| 20:11.33 | codefreeze | Yourname`: yes, but if the caps can be off, use grep -ri instead of just -r |
| 20:11.42 | Yourname` | codefreeze: Bad news, nothing is coming up.. lol |
| 20:11.45 | Yourname` | ok |
| 20:12.22 | Yourname` | I got one file trying -ri! |
| 20:14.10 | Yourname` | Hmm, not much help. |
| 20:16.36 | galeras | nobody? |
| 20:16.43 | crimethinker | nobody? |
| 20:17.09 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 20:17.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:17.20 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 20:17.21 | [TK]D-Fender | galeras: pastebin CLI output that shows it not working, and your related configs. |
| 20:17.24 | [TK]D-Fender | ~pb |
| 20:17.24 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:18.43 | codefreeze | ~AEL |
| 20:18.44 | jbot | ael is, like, Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 20:19.38 | x86 | gah |
| 20:19.50 | x86 | i dont remember sangoma drivers being such a PITA before |
| 20:20.44 | russellb | guess you should have bought something else :-p |
| 20:20.54 | jcolp | naughty russellb |
| 20:20.59 | x86 | :p |
| 20:21.00 | russellb | sorry :-X |
| 20:21.12 | Trevor_b | x86: I always found sangoma to be much simpler then Digium, but thats just me. |
| 20:21.58 | russellb | how is not having to apply patches and install extra software not simpler? |
| 20:22.05 | x86 | Trevor_b: would you like to help me get this sangoma driver working? :P |
| 20:22.21 | x86 | russellb: the vanilla zaptel drivers compiled like a champ |
| 20:22.45 | x86 | patched zaptel and my kernel sources tree with the sangoma wanpipe patches, now my kernel wont 'make modules' anymore |
| 20:22.47 | Trevor_b | Where are you running into an issue? Compilation or configuration? |
| 20:23.04 | russellb | much simpler indeed :-p |
| 20:23.06 | Trevor_b | you had to patch the kernel too? What linux flabor you using? |
| 20:23.13 | Trevor_b | s/flabor/flavor |
| 20:23.17 | x86 | Trevor_b: gentoo |
| 20:23.27 | g1powermac | hmm, is it possible to have one sip phone to be used with multiple asterisk servers? |
| 20:23.45 | x86 | i setup an exact replica of the same box before, and it's working fine... but i forgot the process ;) |
| 20:23.47 | Trevor_b | and the setting wasnt a default kernel option then? Sorry, used to using CentOS, which we rarely need to recompile kernel. |
| 20:24.00 | x86 | Trevor_b: i dont have to recompile the kernel |
| 20:24.17 | Trevor_b | g1powermac: Using multiple registrations is simple, depending on phone manufacturer |
| 20:24.19 | x86 | Trevor_b: but i have to compile a kernel module for the wan router stuff |
| 20:24.26 | Trevor_b | oh just the module |
| 20:24.29 | x86 | yep |
| 20:24.32 | Trevor_b | whats the error? |
| 20:24.38 | g1powermac | Trevor_b, interesting |
| 20:24.39 | x86 | and after the patch, getting a bunch of bad stuff :P |
| 20:24.50 | russellb | WilliamK: 1 |
| 20:24.52 | russellb | d'oh |
| 20:24.58 | russellb | WilliamK: ignore that. |
| 20:25.44 | x86 | http://pastebin.ca/656058 |
| 20:25.53 | x86 | Trevor_b: |
| 20:26.40 | galeras | Before paste, please let me to ask if what i'm trying to do can help to fix my trouble: |
| 20:26.48 | x86 | i'm going to get a fresh kernel tree again (i saved my .config) |
| 20:26.52 | Trevor_b | x86: What zaptel you compiling against, and what kernel? Also the wanpipe source is from sangoma website, or from the CD? |
| 20:27.01 | [TK]D-Fender | ok, I'm out, bbiab |
| 20:27.15 | x86 | Trevor_b: wanpipe 3.1.0.p21 |
| 20:27.47 | x86 | Trevor_b: zaptel 1.2.18, asterisk 1.2.21.1 |
| 20:28.02 | x86 | Trevor_b: linux 2.6.22 |
| 20:28.54 | x86 | Trevor_b: you recommend i download the latest stuff off sangoma's website? |
| 20:29.00 | jcolp | 2.6.22 changed some APIs, it looks like the wanpipe you are using does not have the required changes for it |
| 20:29.05 | x86 | wanpipe 3.1.0.p21 was the latest gentoo offering |
| 20:29.09 | galeras | I have poor quality on fax pass-through in my setup: PSTN->Asterisk->Alcatel (voice calls are fine) |
| 20:29.13 | galeras | I have 2 Te210 Cards, i will try to link channels only inside same card to avoid time sync issues, could fix the fax quality? |
| 20:29.51 | x86 | jcolp: ah, ok... is there a newer wanpipe then? |
| 20:29.51 | jcolp | x86: I do not know, to answer that question I would have to go check... which you are capable of :) |
| 20:29.52 | *** join/#asterisk lirakis (n=eric@69.24.142.1) |
| 20:30.15 | lirakis | hola |
| 20:30.22 | x86 | hehe |
| 20:30.23 | galeras | Hola amigo |
| 20:30.29 | Trevor_b | x86: We downloaded the latest from Sangoma when we just did it a few weeks ago (i think the same zaptel version), but our kernels are based on CentOS 4.5, so our kernel is 2.6.9 not 2.6.22.... |
| 20:30.52 | Nivex | Cisco 7690 vs Polycom IP 501. Who wins? |
| 20:31.01 | Nivex | *7960 |
| 20:31.21 | Trevor_b | x86: either update sangoma drivers (if they have updates for your kernel) or backstep a kernel version or two (2.6.9 is known working for me). |
| 20:31.23 | *** join/#asterisk HockeyInJune (n=HockeyIn@141.157.255.106) |
| 20:32.08 | crimethinker | love your nickname, HockeyInJune |
| 20:35.08 | HockeyInJune | thnx :) |
| 20:37.11 | *** join/#asterisk prashant_jois (n=prashant@68.148.97.186) |
| 20:38.31 | prashant_jois | I'm having a problem with asterisk dying mysteriously. It dies with "Code 127" but doesn't dump its core. Does anybody know what this code is? |
| 20:39.55 | *** part/#asterisk galeras (n=galeras@190.90.27.21) |
| 20:41.14 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 20:43.16 | *** join/#asterisk pagec (n=pagec@h-74-0-107-178.nycmny83.covad.net) |
| 20:46.37 | *** join/#asterisk nentis (n=nentis@209-162-205-68.dq1mn.easystreet.com) |
| 20:46.39 | pagec | asterisk isn't producing any recorded sounds. i watch them playing in the console but nothing is coming out over the phone. anyone know what could be wrong? |
| 20:47.28 | nentis | hey voip folk. I have an Aastra 53i. Trying to find the option for setting the admin password via the tftp config. |
| 21:00.37 | nDuff | I think I've seen some tools intended to make dialplan creation (or at least IVR design) pretty much automated... but my thought on seeing those was "I don't need that GUI %#$@". Now I'm interested in handing off menu creation to one of the local suits, and such a tool would be darned handy... would anyone have a pointer? |
| 21:01.03 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:09.05 | prashant_jois | does anybody know what it means when Asterisk ends with exit status 127? It does so sporadically and doesn't do a core dump so I'm at a loss of how to trace where this is happening. |
| 21:10.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:11.23 | hmmhesays | you can tell it to do so |
| 21:12.30 | hmmhesays | http://www.voip-info.org/wiki-Asterisk+debugging |
| 21:12.46 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:16.07 | prashant_jois | hmmhesays: Asterisk is running with -g option, and it does indeed dump the core on segfaults and other problems, but just this condition does not produce a core dump. |
| 21:18.10 | *** join/#asterisk malph_work (n=chatzill@66-231-0-194.hosts.sdnet.net) |
| 21:19.08 | malph_work | if i used SCP to copy the config files from a server running asterisk 1.4 to a second server 1.4 can anyone think of a reason why blindxfer wouldn't be working? |
| 21:23.26 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:28.26 | nDuff | prashant_jois: hrm. build a debug version and run inside gdb? |
| 21:30.01 | prashant_jois | nDuff: looks like I'm going to have to do that. Thanks! |
| 21:32.23 | *** join/#asterisk Verilium (n=live@206-248-190-25.dsl.teksavvy.com) |
| 21:33.17 | Verilium | Hi there. Quick question, just to make sure, supposing I have "exten => s,10,Dial(SIP/201&SIP/202&SIP/203&SIP/204&SIP/211,20)", will that effectively make it so all 5 extensions ring at the same time? |
| 21:34.03 | wishes | i dont suppose anyone knows offhand shell script to count the items in an array do they? |
| 21:34.16 | carrar | Verilium, yes |
| 21:34.26 | Verilium | carrar: Allright, thank you. :) |
| 21:34.33 | carrar | but you might improve on that |
| 21:34.55 | Verilium | By doing a queue instead? |
| 21:35.06 | carrar | might want to make sure they are registered 1st |
| 21:35.08 | Verilium | (Reading a bit here and there as I type) |
| 21:35.14 | wishes | nm got it |
| 21:35.15 | Verilium | Hmm.. |
| 21:35.54 | carrar | what if they are on a call? |
| 21:36.40 | flenders | then the other ones ring |
| 21:37.05 | Verilium | carrar: Right now, what this extension is doing, in theory, from what I can see, is when it sees a particular callerid, it goes into ivr, then dials the 5 extensions in question. |
| 21:37.30 | carrar | like a call center? |
| 21:37.38 | carrar | then you may want to use queues like you mentioned |
| 21:37.40 | Verilium | I'm trying to do debug for someone on vacation, so I'm not even completely sure I'm understanding all of this. Ahem. :P |
| 21:38.47 | Verilium | carrar: Well, not exactly for a call center, but I suppose the idea is the same. Right now, it's just if the phone number matches the one from the phone downstairs, the 5 extensions ring, so that 'someone' can pickup, and open the door for the person downstairs who wants to come up. |
| 21:39.37 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 21:39.40 | carrar | if it's not yours don't worry about it, just use what you pasted |
| 21:40.34 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 21:40.34 | Verilium | Well, my friend is the one more knowledgeable with Asterisk, I'm more of the unix admin, but anyway.. |
| 21:41.37 | Verilium | Anyhow. Thanks for confirming. |
| 21:47.22 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:50.07 | lirakis | hrmmm |
| 21:50.24 | lirakis | i keep seeing ACL error (permit/deny) for failed sip authentications on my cli |
| 21:50.53 | lirakis | .. this is a new box.. whats wierd.. is i had on client workg.. then i changed the user/pass and i get it to work again |
| 21:51.07 | Op3r | whats the time in california now? |
| 21:51.17 | elixer | so does digium need any overpaid C# developers? just curious... |
| 21:51.18 | elixer | :) |
| 21:51.32 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 21:51.36 | Deeewayne | elixer: probably not, but good luck getting overpaid! |
| 21:52.11 | el_critter | hi |
| 21:52.30 | Deeewayne | hello |
| 21:52.32 | elixer | Deeewayne: i'm already overpaid - my goal is to continue being so |
| 21:52.33 | elixer | :) |
| 21:52.50 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 21:53.00 | elixer | just not at my current employer. |
| 21:53.20 | Deeewayne | where are you? |
| 21:53.23 | Op3r | thats good |
| 21:53.30 | Deeewayne | state...not employer ;-) |
| 21:53.31 | Op3r | Im also overpaid |
| 21:53.49 | Op3r | on a 3rd world wages though |
| 21:53.50 | Op3r | :( |
| 21:53.55 | elixer | baltimore, md |
| 21:54.08 | elixer | i could move to alabama though |
| 21:54.16 | elixer | even though its hot as hades |
| 21:54.30 | Deeewayne | especially this week |
| 21:54.57 | elixer | i've only been there once |
| 21:55.00 | elixer | birmingham |
| 21:55.30 | elixer | and when i walked outside in the morning @ 6am to smoke and immediately had every article of clothing stick to my body i decided that it wasn't my kinda state |
| 21:56.36 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 21:57.11 | *** part/#asterisk Verilium (n=live@206-248-190-25.dsl.teksavvy.com) |
| 22:01.26 | *** join/#asterisk hohum_ (n=dcorbe@gate.globecommsystems.com) |
| 22:08.27 | *** join/#asterisk mxmasster (n=mxmasste@207.171.12.109) |
| 22:08.30 | mxmasster | hi all |
| 22:08.32 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 22:09.38 | mxmasster | sipxpbx has a config server that includes support to autocreate configuration files for Polycom phones |
| 22:09.49 | mxmasster | does anything like this exist for Asterisk? |
| 22:10.09 | mxmasster | and no I'd rather not use the sipxconfig server to configure my Asterisk deployments |
| 22:14.28 | wishes | elixer: time to quit |
| 22:14.48 | wishes | smoking that is, not states :) |
| 22:15.10 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 22:15.12 | Mercestes | wishes: Seems people are taking you at your word. |
| 22:15.55 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:16.02 | wishes | yeah |
| 22:16.08 | wishes | thats not entirely a bad thing though is it? :D |
| 22:16.32 | Mercestes | Probably not. =/ |
| 22:17.13 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:17.20 | nDuff | mxmasster: not for polycom, but I've got autocreation code for a whole bunch of other phones |
| 22:17.49 | mxmasster | nDuff: thanks - i've seen some examples for Cisco, etc... but unfortunately we need Polycom |
| 22:17.58 | nDuff | mxmasster: ...and it's probably flexible enough to do its thing for polycoms as well. (runs a templating engine, supports arbitrary filters, etc) |
| 22:18.24 | nDuff | mxmasster: heh. wait a few months; we're planning on getting polycoms for any new lines we put in. |
| 22:18.41 | mxmasster | nDuff: :) thanks |
| 22:18.58 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:19.33 | [TK]D-Fender | mxmasster, I think I've heard of a perl script ofr two for that. |
| 22:19.46 | [TK]D-Fender | mxmasster, Go check the WIKI |
| 22:21.30 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:22.55 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 22:23.50 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 22:24.29 | clyrrad | has anyone here got a "paging" system to work with Linksys/Cisco hardware and Asterisk? |
| 22:25.25 | *** join/#asterisk xenon4356 (i=xenon435@pool-72-82-241-15.cmdnnj.fios.verizon.net) |
| 22:25.29 | mxmasster | [TK]D-Fender: I'm obviously not searching for the right think on the wiki - can you give me the search term? |
| 22:25.36 | xenon4356 | hello all |
| 22:26.01 | clyrrad | I want to have some sort of Group paging system, like you dial an extension, and it pages your voice to all phones in your paging group..... anyone done anything like this? |
| 22:26.53 | xenon4356 | I am looking for some help with a scripting task - I would like to be able to dial in and have the asterisk system run though a few scripts, which would in turn interact with a web site - is this even possible? |
| 22:27.58 | clyrrad | xenon4356: yes lookup AGI on Google - it will do what you need |
| 22:30.15 | [TK]D-Fender | mxmasster, You'll jsut have to look around. I do all of mine by hand. |
| 22:30.34 | [TK]D-Fender | clyrrad, "show application page" <------- |
| 22:31.28 | xenon4356 | hm ok, cool - thanks. let me give you a little more information. i'd like to make a script that will allow users to call in to reset their password to an online application. i already have the password reset forms online, they take 3 fields (last 4 of SSN, zip code, and year of birth) and if these match a database, the password is reset. I would like to automate this through the phone system. Does this sound doable? |
| 22:31.51 | [TK]D-Fender | xenon4356, Quite |
| 22:32.40 | xenon4356 | Then I guess I should say - is it easy? |
| 22:33.01 | xenon4356 | And if you can think of any keywords that might help me find what I'm looking for, much appreciated |
| 22:33.24 | clyrrad | xenon4356: "Easy" is a very relative term |
| 22:33.35 | clyrrad | xenon4356: I belive it would be easy for me, not sure about you :p |
| 22:33.50 | clyrrad | [TK]D-Fender: thanks thats what I was looking for |
| 22:34.29 | xenon4356 | Well I guess what I am trying to figure out is how asterisk would handle this - or I should say, how I would go about implementing it - is asterisk somehow able to post data to a website and then act upon the results? |
| 22:34.31 | Hymie | hey..does anyone know if there are english prompts by that june wallack? my client wants an alternate english voice, but is'nt willing to pay for all the base prompts, only the changes... |
| 22:35.15 | [TK]D-Fender | xenon4356, where would your web server (or its data source) be relative to your * server? |
| 22:36.38 | xenon4356 | same network - internal |
| 22:37.08 | [TK]D-Fender | xenon4356, then you can have your * server poll the request data from there and thats it. |
| 22:37.34 | xenon4356 | What do you mean? |
| 22:38.23 | [TK]D-Fender | xenon4356, short answer "yes", * can pull the account info from "wherever", or your other server can deposit it somewhere * can access for your purpose |
| 22:38.28 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 22:39.17 | clyrrad | xenon4356: Like I said earlier lookup AGI on Google - its what you need |
| 22:39.26 | [TK]D-Fender | clyrrad, not necessarily. |
| 22:39.43 | clyrrad | it should do exactly what he needs |
| 22:40.08 | [TK]D-Fender | clyrrad, it is a means to doing several things. Not to say a REQUIRED means. You think too specifically :) |
| 22:40.38 | clyrrad | I was refering to his question of "is it possible", and "what keywords" to lookup |
| 22:40.57 | [TK]D-Fender | clyrrad, then you clearly haven't been reading what he was asking :) |
| 22:41.00 | xenon4356 | I am looking up AGI - I'm just confused as to how to approach this - like I said the functionality already exists on a web page. if there was a way to have asterisk grab the variables i need, then post them to my page, and speak the result, that's what i'm looking for |
| 22:41.36 | [TK]D-Fender | xenon4356, You can run whatever scripting you want based on your call to verify their input and act according. |
| 22:43.11 | xenon4356 | OK -- I guess I have some reading to do. |
| 22:43.16 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:43.53 | [TK]D-Fender | xenon4356, Ok, where is the account info stored? |
| 22:44.39 | xenon4356 | its stored in Active Directory as well as a few other DBs |
| 22:45.02 | [TK]D-Fender | xenon4356, what other kinds of DB's? |
| 22:46.48 | xenon4356 | mysql, but again the functionality to reset the passwords already exists - i'm not looking to reinvent that for a whole bunch of reasons, the biggest one being security |
| 22:47.42 | [TK]D-Fender | xenon4356, well if a change to mysql is enough you can do that completely withing your * dialplan without any extra tools or excessive programming |
| 22:48.19 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 22:48.24 | xenon4356 | unfortunately it's really not - it would work for one password, but the active directory wouldn't work |
| 22:48.54 | [TK]D-Fender | xenon4356, do you know of a linux tool out there that can modify your AD accourdingly? |
| 22:49.09 | xenon4356 | I don't believe so |
| 22:49.18 | [TK]D-Fender | xenon4356, then thats your bottleneck. |
| 22:49.46 | xenon4356 | One probably exists, but again if I COULD do this by just posting this data to the web page, it would be much better and much more likely to be OK'd security wise |
| 22:50.05 | [TK]D-Fender | xenon4356, * runs on Linux and can be told to execute scripts that will do "whatever". if you can't picture a "whatever" that will sync up all of the different aspect of what you need to do then "oh well" |
| 22:50.42 | [TK]D-Fender | xenon4356, An internally generated "post" could work, and this does sound like some extra trouble |
| 22:51.12 | xenon4356 | well, what i need doesn't seem like it would be too tough, if it's possible |
| 22:52.20 | xenon4356 | if it could collect 3 variables, post that data to a web page, and respond with whether it was successful (based on the response to the posted page) that is all I need |
| 22:52.31 | [TK]D-Fender | xenon4356, its the AD stuff thats the challenge |
| 22:52.58 | Juggie | linux & active directory can talk |
| 22:53.17 | citats | [TK]D-Fender: it sounds like he doesn't need to worry about the specific database, just submit a HTTP request and whatever on the webserver handles the backend stuff |
| 22:54.00 | clyrrad | [TK]D-Fender: you mentioned Page - I dont seem to have that applicaiton on 1.2.11 even though the wiki sais it worked from v 1.2.7 - does this need to be installed seperatly? |
| 22:54.12 | xenon4356 | sorry, my dog pissed on the carpet. anyway, perhaps i'm not making myself clear. the AD stuff, the database stuff - all that is already done. the webpage works fine, provided you entered into the page the 3 variables correctly |
| 22:54.40 | xenon4356 | i would just like to be able to post these variables with data entered from a phone keypad |
| 22:56.16 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 22:56.23 | *** join/#asterisk aut (n=aut@modemcable241.90-201-24.mc.videotron.ca) |
| 22:56.38 | fujin | hi, could anyone tell me if using format_mp3 for hold music is going to have a higher impact on performance than say, playing back ulaw? |
| 22:58.19 | aut | can someone help me with codecs? i bought g729 licenses and set g729 as the only allowed codec. now i can call out, but i can't conference. when i try, i get: " No compatible codecs, not accepting this offer!" |
| 22:58.44 | fujin | and is there a way to show the allowed codecs |
| 22:58.46 | fujin | err |
| 22:58.47 | fujin | formats |
| 22:58.49 | fujin | for moh |
| 22:59.29 | aut | fujin, yes, i think there is a huge difference |
| 22:59.36 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 23:00.04 | aut | fujin: you should pre-encode your moh with the codecs you allow |
| 23:00.26 | *** part/#asterisk lirakis (n=eric@69.24.142.1) |
| 23:00.41 | fujin | I just did a load test, had 30 people in the queue with format_mp3 |
| 23:00.44 | fujin | and the box was shitting itselfs |
| 23:00.48 | fujin | load of 2.0 |
| 23:00.58 | fujin | bad quality hold music playback etc |
| 23:00.59 | wishes | encode to gsm :D |
| 23:01.09 | aut | encode to whatever the channels are already in |
| 23:01.17 | aut | so you dont have to transcode |
| 23:01.22 | fujin | yeah, gotta work out how to configure moh to accept ulaw files |
| 23:01.32 | [TK]D-Fender | clyrrad, if you don't have ti its because you don't have a Zaptel timing source as required. |
| 23:01.43 | [TK]D-Fender | clyrrad, Page uses MeetMe internally |
| 23:02.07 | aut | fujin: see: http://astrecipes.net/?n=152 |
| 23:02.22 | [TK]D-Fender | aut, what hardware are you using? |
| 23:02.26 | fujin | what should the extensions for .ulaw be? |
| 23:02.34 | [TK]D-Fender | fujin, exactly that |
| 23:02.34 | clyrrad | [TK]D-Fender: when you say "Zaptel timing source" - what exactly is that? |
| 23:02.42 | fujin | ah, cool. |
| 23:02.43 | [TK]D-Fender | clyrrad, Oh GOD..... |
| 23:02.57 | aut | [TK]D-Fender, no fancy hardware.. it's VOIP only |
| 23:03.00 | [TK]D-Fender | clyrrad, you know a TIMING source as is required for MeetMe, IAX2 trunking, etc... |
| 23:03.21 | [TK]D-Fender | aut, what devices EXACTLY are you using? |
| 23:03.28 | xenon4356 | another question while I am here - all of my calls seem to have a lot of "jitter" - i think thats the correct term - the audio is somewhat choppy. this happens both from outside to asterisk as well as from a phone on the same lan to asterisk |
| 23:03.32 | aut | just cisco 7961 |
| 23:03.35 | clyrrad | [TK]D-Fender: I have not used either of those, thats why I am asking.......sorry if my question bothers you - I am asking becase i dont know |
| 23:03.39 | xenon4356 | its not horrible by any means, but it is noticible |
| 23:03.46 | fujin | [TK]D-Fender: should a .ulaw be formatted a specific way? |
| 23:03.51 | aut | [TK]D-Fender, cisco 7961 phones. do you need to know more about the server itself? |
| 23:03.53 | fujin | bitrate, etc |
| 23:04.16 | [TK]D-Fender | aut, It could be that your PHONES don't support multiple G.729 encodings. Sipura/Linksys is notorious for this. |
| 23:04.52 | crimethinker | linksys in general is notorious for suckiness |
| 23:04.58 | crimethinker | sucking, even |
| 23:05.06 | aut | [TK]D-Fender, i tried with a cisco -> sipura/linksys earlier and had the same issues... so i started troubleshooting with two ciscos instead... hrm |
| 23:05.19 | crimethinker | with 2 fat slobbery lips |
| 23:05.48 | fujin | nevermind, got it |
| 23:05.53 | fujin | 8000bit/1chan |
| 23:05.55 | aut | [TK]D-Fender, but that would probably explain it. |
| 23:06.04 | clyrrad | anyone used the Page() application? |
| 23:06.14 | aut | [TK]D-Fender, only thing is i havent seen anything posted on the net about it... |
| 23:06.39 | [TK]D-Fender | aut, allow ulaw as secondary preference. if that works, there's your answer |
| 23:06.46 | [TK]D-Fender | clyrrad, I have |
| 23:07.33 | clyrrad | [TK]D-Fender: Yes, but you are for some reason getting annoyed by my question......... so im asking for someone who is willing to help me understand what I need to do |
| 23:07.50 | xenon4356 | anyone? |
| 23:08.06 | fujin | that tends to happen with [TK]D-Fender |
| 23:08.09 | aut | [TK]D-Fender, yes, that works :/ |
| 23:08.10 | fujin | get used to it, ask better questions |
| 23:08.25 | clyrrad | xenon4356: I had this problem when there was not enough upload bandwith |
| 23:08.54 | xenon4356 | but it happens when the phoen is on the same switch even as the asterisk box, certainly that can't be bandwidth |
| 23:10.18 | fujin | xenon4356: checked the jitter settings on your phones? what codecs? |
| 23:10.18 | clyrrad | anyone else used the Page() application or know what I need to do to get it installed / working? |
| 23:10.36 | fujin | that's a non-default application, did you buiild it yourself? |
| 23:10.50 | fujin | not registered anyways.. show application page doesn't give me any info |
| 23:10.51 | clyrrad | fujin: was that question to me? |
| 23:10.55 | [TK]D-Fender | clyrrad, Do you have a Zaptel card? |
| 23:10.59 | clyrrad | nope |
| 23:11.16 | clyrrad | I use SIP phones, pure VoIP system |
| 23:11.19 | [TK]D-Fender | clyrrad, then you you need ZTDUMMY running with Zaptel |
| 23:11.28 | xenon4356 | fujin - i have a cisco 7960....haven't messed with jitter or anything of that sort |
| 23:11.39 | [TK]D-Fender | clyrrad, Go install Zaptel & ZTDUMMY, then recompile * after starting zaptel. |
| 23:11.39 | fujin | sorry, I don't play with cisco sip phones |
| 23:11.41 | fujin | no idea about em |
| 23:11.50 | xenon4356 | it does however happen from regular phone lines coming in to the DID too |
| 23:11.53 | clyrrad | [TK]D-Fender: that will provide me the Page application? |
| 23:11.57 | [TK]D-Fender | clyrrad, then you will be able to use Page & MeetMe |
| 23:12.02 | [TK]D-Fender | clyrrad, Yes. |
| 23:12.07 | clyrrad | [TK]D-Fender: ok - thanks |
| 23:12.25 | aut | what kind of phones are "better" than the cisco 7960s? |
| 23:12.30 | [TK]D-Fender | clyrrad, I'm just shocked that after all this time you've been here you didn't know that :) |
| 23:12.37 | [TK]D-Fender | aut, Any Polycom :) |
| 23:12.38 | fujin | polycom |
| 23:12.46 | aut | maybe something that supports a low bandwidth codec in tandem |
| 23:12.49 | aut | really? |
| 23:12.53 | clyrrad | [TK]D-Fender: yea I been around a long time, but I have never had a need for this - today is the first time I was asked for it |
| 23:12.53 | fujin | I'd go with polycom or linksys |
| 23:12.56 | aut | didnt realize polycom had good voip offerings |
| 23:12.56 | fujin | from personal preference |
| 23:13.04 | fujin | polycom are probably The Best (tm) |
| 23:13.15 | JT | aut: polycom are the industry leaders in sip phones |
| 23:13.29 | aut | the soundpoints? |
| 23:13.36 | JT | yes |
| 23:13.38 | aut | they are so ugly compared to cisco ! :P |
| 23:13.57 | aut | is the voice quality comparable to the cisco? |
| 23:14.00 | [TK]D-Fender | aut, They grow on you... |
| 23:14.03 | aut | the cisco has *amazing* voice quality |
| 23:14.08 | aut | speakerphone too |
| 23:14.13 | *** join/#asterisk chrisp_83 (n=cplumley@72.18.105.112) |
| 23:14.18 | fujin | dude, they aren't ugly |
| 23:14.20 | [TK]D-Fender | aut, Polycoms MAKES Cisco's speakerphone :) |
| 23:14.21 | fujin | they are awesome looking |
| 23:14.28 | JT | aut: the audio quality is better than cisco |
| 23:14.29 | fujin | I swear I wish i hadda bought polycoms now instead of linksyss |
| 23:14.29 | aut | [TK]D-Fender, hehe, that explains it! |
| 23:14.40 | JT | as if they look ugly |
| 23:14.41 | aut | we've invested quite a bit in the cisco shit |
| 23:14.44 | aut | but im so sick of the problems |
| 23:14.46 | JT | cisco's silver plastic is ugly |
| 23:14.46 | aut | it's always something |
| 23:14.56 | chrisp_83 | Could someone please private message me regarding some help wih asterisk using VMware? |
| 23:15.50 | aut | i assume the polycoms actually support SIP rather than being a hack? :) |
| 23:15.57 | JT | yes |
| 23:16.02 | aut | that's always a plus |
| 23:16.15 | aut | im sold.. gonna order a few |
| 23:16.24 | Dan0maN_Work | heh |
| 23:16.29 | [TK]D-Fender | aut, hold your horses. |
| 23:16.31 | aut | uh oh |
| 23:16.31 | aut | hehe |
| 23:16.32 | Dan0maN_Work | i asked the same questions last week |
| 23:16.36 | [TK]D-Fender | aut, Where are you located? |
| 23:16.40 | xenon4356 | the company I work for is about to switch their whole CCME platform over to SIP...I can't wait to see this |
| 23:16.43 | aut | [TK]D-Fender, canada |
| 23:16.57 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 23:17.01 | [TK]D-Fender | aut, Ok, still worth importing from these guys : www.telephonydepot.com |
| 23:17.04 | clyrrad | [TK]D-Fender: is it ok to use the newest zaptel version with Asterisk 1.2.x? Or do I need to version match? |
| 23:17.11 | [TK]D-Fender | aut, Do you have 803.3af PoE? |
| 23:17.18 | [TK]D-Fender | clyrrad, version match |
| 23:17.39 | aut | [TK]D-Fender, yep, got the POE |
| 23:17.40 | clyrrad | [TK]D-Fender: thanks :) |
| 23:17.54 | [TK]D-Fender | clyrrad, np |
| 23:18.05 | [TK]D-Fender | aut, Ok, do you need the pass-through switched port? |
| 23:18.29 | aut | [TK]D-Fender, hrm, im using it on the cisco, but i might not absolutely have to have it |
| 23:18.47 | [TK]D-Fender | aut, If you don't IP 320 ($87.50) http://www.telephonydepot.com/product_p/105-058-320.htm |
| 23:19.05 | [TK]D-Fender | aut, If you don't IP 330 ($111.95) http://www.telephonydepot.com/product_p/105-058-330.htm |
| 23:19.07 | [TK]D-Fender | do* |
| 23:19.08 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-234-118.vic.bigpond.net.au) |
| 23:19.10 | aut | [TK]D-Fender, oh, i need more than one line too |
| 23:19.32 | Dan0maN_Work | 320/330's support 2 lines |
| 23:19.35 | [TK]D-Fender | aut, Cisco's warp your brains on the concept of "lines" |
| 23:19.36 | aut | oh, i see |
| 23:20.16 | [TK]D-Fender | aut, the 320/330 can support up to 4 CALLS simultaneously. Typically a phone is only used for a SINGLE identity but may have several CALLS going on over it at a time. |
| 23:20.32 | [TK]D-Fender | identity = registration = "line" |
| 23:20.51 | aut | yeah, i meant that i need two separate sip registrations with their own extensions... |
| 23:21.00 | [TK]D-Fender | first mental hurdle to oversome when learning about SIP & PBX's |
| 23:21.06 | aut | but i would expect to be able to conference, etc on those separate "lines" |
| 23:21.17 | [TK]D-Fender | aut, You meana single phone ould have 2 distinct identities? |
| 23:21.21 | aut | yep |
| 23:21.26 | [TK]D-Fender | aut, ...why? |
| 23:21.43 | [TK]D-Fender | aut, only valid case I've seen is where 2 people actually SHARED a phone. |
| 23:22.10 | aut | [TK]D-Fender, it's a strange setup... basically if im working for two different companies with internal phone systems, i can hook into each one separately |
| 23:22.13 | [TK]D-Fender | aut, line != call! |
| 23:22.35 | [TK]D-Fender | Ah, ok, where you're working for 2 divisions and need each reg'd sepearlety, sure |
| 23:22.41 | aut | i understand, but i have to authenticate against two separate sip servers |
| 23:22.54 | [TK]D-Fender | aut, In taht case you can have echo line key reg to a different server supporting 2 calls each. |
| 23:23.04 | [TK]D-Fender | s/echo/each/ |
| 23:23.05 | aut | cool |
| 23:23.18 | Dan0maN_Work | curious on what you guys this about this... my remote office currently has ~10-15 people in it, with plans to expand to ~25-30. when i was talking last week to a few of you about the main office and redundant servers, i should possibly get a beefy DB server, with 2 easily exchangeable * servers. should i even consider this for the remote office? or just a single beefy server? |
| 23:23.27 | [TK]D-Fender | aut : all taht on an 87.50$ phone :) |
| 23:23.33 | aut | yeah, sounds very nice |
| 23:23.41 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 23:23.46 | Zipper_32 | aut: I'm happy with my 12 IP330's and Linksys SRW224P 24Port POE Switch. They work very well together. Not very expensive either. |
| 23:23.47 | aut | while i have you on the subject, do you know of any good wireless voip phones? |
| 23:23.55 | aut | i tried the linksys... HORRIBLE RIPOFF |
| 23:23.57 | [TK]D-Fender | Dan0maN_Work, each site should have its own server. |
| 23:24.04 | [TK]D-Fender | ~wifisip |
| 23:24.05 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 23:24.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 23:24.12 | aut | hella true. |
| 23:24.42 | [TK]D-Fender | ATA + Cordless <------------- |
| 23:24.50 | honeybeebuzz | in fact Wi-Fi itself :) |
| 23:25.00 | aut | which ATA do you use? |
| 23:25.05 | aut | tried linksys there too :) |
| 23:25.21 | aut | didnt seem bad so far, but i havent tried anythign advanced with it either |
| 23:26.10 | [TK]D-Fender | Linksys wired phones are ok, but not worth it compared to Polycom (in North America). their ATA's work jsut fine at a very nice price-point. |
| 23:26.52 | Dan0maN_Work | D-Fender: you think for only 15-30 people, that i should go with 2 servers? or one beefy one with DB and * together? (i was going to have at least 1 server there running *, with offices trunked IAX) |
| 23:27.07 | [TK]D-Fender | Dan0maN_Work, not worth the DB |
| 23:27.28 | [TK]D-Fender | Dan0maN_Work, you need a server at both sites to trunk w/ IAX |
| 23:28.59 | Dan0maN_Work | D-Fender: ok. thanks for the quick info. i've gotta run |
| 23:29.59 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-124-103.pitbpa.east.verizon.net) |
| 23:34.06 | *** join/#asterisk exvito (n=exvito@89.181.10.10) |
| 23:34.16 | honeybeebuzz | ? while reading this article, http://articles.techrepublic.com.com/5100-1035_11-6123058.html I am lost about "What You Need" 2nd pointer |
| 23:34.48 | honeybeebuzz | particualrly... If you want to be able to, for example use your POTS line for your local calls and a broadband phone for your long distance, you will want to add one more FXO daughter card to your order. |
| 23:34.51 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 23:35.04 | [TK]D-Fender | honeybeebuzz, FORGET that article now. You don't have a defined need just yet. |
| 23:35.31 | [TK]D-Fender | honeybeebuzz, Do yuo have analo lines that you WANT to use with *? |
| 23:35.41 | [TK]D-Fender | honeybeebuzz, Do you have analog lines that you WANT to use with *? |
| 23:36.09 | honeybeebuzz | yes... that will stick in on FXO right... |
| 23:36.33 | [TK]D-Fender | honeybeebuzz, FXO *ports*, yes. How many? |
| 23:36.41 | honeybeebuzz | and on FXS, you connect a regualular home phone. |
| 23:37.09 | honeybeebuzz | i got one fxo and fxs |
| 23:37.17 | [TK]D-Fender | honeybeebuzz, Yes. Though I would never recommend using a PCI card for that. |
| 23:37.45 | [TK]D-Fender | honeybeebuzz, before getting too embroiled, what do yuo want to do with *? |
| 23:38.40 | honeybeebuzz | it is digium's with those two FXS and FXO daughter card |
| 23:39.10 | [TK]D-Fender | honeybeebuzz, Stop thinking about hardware right now OK, state your NEED and we'll suggest hardware accordingly. |
| 23:39.56 | *** part/#asterisk zerohalo (n=zeroHalo@h-74-2-90-66.cmbrmaor.covad.net) |
| 23:40.53 | honeybeebuzz | ok... here we go... one box with * instaled , and one analog phone line coming to house. one the same phone line, ADSL there too. |
| 23:41.17 | [TK]D-Fender | honeybeebuzz, Ok, thats what yuo HAVE. Now what do you want to DO with all of this? |
| 23:41.39 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 23:42.38 | honeybeebuzz | to have incoming call going to * box, and routed accourding to menu... 1. go to mobile number , 2. ring locally phone attached to home line 3. ring soft ext |
| 23:42.41 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:44.06 | *** part/#asterisk HockeyInJune (n=HockeyIn@141.157.255.106) |
| 23:46.49 | [TK]D-Fender | honeybeebuzz, Ok, then here's what I'd suggest : Linksys SPA-3102. Will let you take in your home line AND use 1 phone as a SIP phone. Cost +/- $75. |
| 23:48.27 | [TK]D-Fender | honeybeebuzz, http://www.telephonydepot.com/product_p/105-054-312.htm |
| 23:48.39 | [TK]D-Fender | honeybeebuzz, There, even LESS |
| 23:49.11 | honeybeebuzz | there where is * in this picture... |
| 23:52.41 | [TK]D-Fender | honeybeebuzz, the SPA-3102's FXO takes in your home line, and send the call to * via SIP ( a voip protocol). * then processes the call however your likee. From there * can call the FXS port on the SPA making al the phones attached to it ring and the rest is history |
| 23:53.10 | [TK]D-Fender | Ok, I've gotta jet for now. Thats the quick version and what I would highly recommend for you just starting off with this. |
| 23:53.42 | honeybeebuzz | keul... thanks I will draw a pic to put things first... in order. |
| 23:53.51 | [TK]D-Fender | honeybeebuzz, the SPA-3102 is a VERY versatile little unit that can be deployed in a LOT of different ways and saves you mucking with your server for a PCI card. |
| 23:54.21 | [TK]D-Fender | honeybeebuzz, the SPA-3102 would be plugged onto your LAN switch and talk to * over IP all internal to your network. |
| 23:54.27 | [TK]D-Fender | ok, outt here, later all.... |
| 23:56.21 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 23:57.02 | *** join/#asterisk d3wayne (n=deeewayn@c-71-228-186-75.hsd1.al.comcast.net) |
| 23:58.36 | *** join/#asterisk CaT[tm] (n=cat@nessie.weebeastie.net) |
| 23:59.33 | CaT[tm] | how do I get asterish to log the equivalent of -vvv etc without actually specifying it on the command line? douing full => notice,warning,error,debug,verbose in logger.conf does't seem to be doing it. |