| 00:00.09 | frk2 | guess not |
| 00:02.16 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:02.40 | Malthus | real TDM (T1/PRI) or analog? |
| 00:03.34 | justinu | he's probably talking about a TDM400 or 2400 |
| 00:04.04 | Malthus | why would Digium name their cards like that? |
| 00:06.33 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:06.49 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 00:07.46 | *** join/#asterisk enzo123 (n=enzo123@5.sub-70-192-180.myvzw.com) |
| 00:08.50 | justinu | Malthus: good question |
| 00:12.17 | frk2 | yeah - TDM 4000 |
| 00:12.18 | frk2 | 400 |
| 00:12.32 | frk2 | so analog |
| 00:12.38 | frk2 | i have about 8 of these boxes |
| 00:12.44 | frk2 | disconnection happens on exactly 3 |
| 00:13.09 | frk2 | the call would just go *poof* in the middle |
| 00:13.44 | frk2 | i wonder if using fxs_ls would hlep |
| 00:13.55 | frk2 | since there is no disconnect supervision in this place |
| 00:14.22 | justinu | does it happen on all 4 lines of the 3 boxes? |
| 00:14.54 | frk2 | yes |
| 00:14.57 | frk2 | checked that too |
| 00:15.07 | frk2 | manually disabled zap channels |
| 00:15.13 | justinu | have you asked your telco to test the lines for shorts or other defects? |
| 00:15.16 | frk2 | i wonder if this could be a telco issue |
| 00:15.29 | frk2 | no i haven't. Thats a very hard thing to explain to clients :) |
| 00:15.31 | justinu | a short might cause the tdm400 to think the line disco'd and it hangs up |
| 00:15.40 | frk2 | hmmm |
| 00:15.50 | frk2 | but there's NOTHING in /var/log/messages even with debug |
| 00:15.58 | frk2 | thats exactly what i was thinking myself |
| 00:16.02 | justinu | the other thing to do would be to perhaps swap out a known good card with a potential bad card |
| 00:16.17 | justinu | see if it follows the card or the lines |
| 00:16.27 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:16.30 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:16.35 | frk2 | did that too justinu |
| 00:16.37 | frk2 | two cards |
| 00:16.45 | frk2 | on one of the locations |
| 00:17.14 | frk2 | what else- on the system level- could possibly cause this? |
| 00:17.28 | frk2 | i mean i do have samba running on this machine |
| 00:17.37 | justinu | irq conflicts |
| 00:17.45 | frk2 | so ive increased the priorioty to -19 for asterisk |
| 00:17.59 | justinu | the tdm400 is pretty touchy WRT sharing IRQs |
| 00:18.40 | frk2 | its not sharing irq's |
| 00:18.54 | frk2 | cat /proc/interrupts says it uses 21 alone |
| 00:18.55 | justinu | then i'd have to suspect a line problem |
| 00:19.05 | justinu | does audio ever pop or crack? |
| 00:19.09 | frk2 | nopes |
| 00:19.16 | frk2 | audio is CRYSTAL clear |
| 00:19.20 | frk2 | just goes- poof |
| 00:19.29 | justinu | weird |
| 00:19.52 | frk2 | the user once did complain the voice went robotic |
| 00:19.57 | frk2 | but that was a bad mpg123 process |
| 00:20.01 | frk2 | hogging CPU |
| 00:20.05 | frk2 | i wonder if Samba is the culprit |
| 00:20.09 | justinu | doubt it |
| 00:20.36 | frk2 | busydetect is also up to 8 |
| 00:20.38 | frk2 | so it cant be that |
| 00:20.55 | justinu | you said you tried removing that, right? |
| 00:21.22 | frk2 | no. i increased it to 8 |
| 00:21.31 | frk2 | if i remove it the lines get stuck |
| 00:21.42 | justinu | but do they ever disconnect? :) |
| 00:22.11 | frk2 | will never be able to check |
| 00:22.16 | justinu | you said you checked /var/log/messages? what about /var/log/asterisk/full? |
| 00:23.34 | frk2 | nothing unusual |
| 00:23.48 | *** join/#asterisk stp (i=stp1800@68-235-136-100.atlsfl.adelphia.net) |
| 00:23.51 | justinu | crank up the verbosity and the debug level |
| 00:23.59 | justinu | maybe you can get a clue asto why the line disco'd |
| 00:24.58 | docelm0 | anyone in here use plainvoip? |
| 00:24.59 | frk2 | well |
| 00:25.10 | frk2 | debug is on in the wctdm module |
| 00:26.02 | docelm0 | MSG me if you do. Also you MUST use asterisk to connect to us. I need someone to try out an API I created. Curious to know how it works.. :P |
| 00:26.14 | Qwell[] | docelm0: Give me an account :p |
| 00:26.23 | docelm0 | Go signup for one |
| 00:26.31 | Qwell[] | $$$ :P |
| 00:26.46 | frk2 | so juntinu- this is not a common problem? |
| 00:26.48 | docelm0 | Dude.. PV is cheap enough w/o me giving money away |
| 00:26.52 | Qwell[] | heh |
| 00:26.57 | frk2 | would let me to believe that its the telco |
| 00:27.03 | frk2 | you think using fxs_ls would help? |
| 00:27.08 | justinu | frk2: i dunno if I would say common, but I've heard about it |
| 00:27.09 | docelm0 | BUT for someone who IS already registered I will give em $2 credit for testing.. :) |
| 00:27.24 | justinu | using fxs_ls would rule out the "short" theory, i believe |
| 00:27.45 | frk2 | exactly |
| 00:28.09 | frk2 | more funny stuff |
| 00:28.11 | justinu | then how does asterisk deal with disco supervision? |
| 00:28.16 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 00:28.23 | frk2 | it doesnt.. just busydetect |
| 00:28.25 | Qwell[] | justinu: with really big fros |
| 00:28.35 | frk2 | there is really no disconnect super in my country anyways :) |
| 00:28.37 | frk2 | oh |
| 00:28.39 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 00:28.42 | frk2 | more interesting facts i forgot to share |
| 00:28.45 | justinu | frk2: so why use _ks? |
| 00:28.58 | frk2 | this ONLY happens during the daytime |
| 00:29.03 | justinu | sunspots!! |
| 00:29.08 | frk2 | it doesnt happen in the evening! |
| 00:29.11 | justinu | wrap your pbx in tinfoil |
| 00:29.19 | frk2 | :) |
| 00:29.24 | frk2 | im also wondering if its heat |
| 00:29.24 | justinu | seriously, maybe it's a weak PSU? |
| 00:29.29 | justinu | or something like that. |
| 00:29.38 | justinu | pbx on a UPS? |
| 00:29.39 | frk2 | theres no airconditioning at this location |
| 00:29.40 | *** join/#asterisk zwelch (n=kumquat@pdpc/supporter/sustaining/zwelch) |
| 00:29.44 | frk2 | yes its on a UPIS |
| 00:29.45 | zwelch | join #asterisk-dev |
| 00:29.45 | frk2 | UPS |
| 00:30.06 | frk2 | and a temperature of 110 deg F is not common in this land |
| 00:30.12 | jsaunders | Does anyone know the default amount of time between sip registration attempts w/ *? |
| 00:30.13 | frk2 | so i dont know if thats the problem |
| 00:30.17 | jsaunders | And perhaps how to change? |
| 00:30.32 | *** join/#asterisk IceManRISK (n=kart@201.66.46.249) |
| 00:30.33 | frk2 | jsaunders- thats a property of the sip client |
| 00:31.00 | frk2 | I guess heat would cause random bullshit |
| 00:31.06 | jsaunders | regseconds= for sip client? |
| 00:31.07 | jsaunders | k |
| 00:31.07 | frk2 | it always has in my experience :) |
| 00:31.22 | jsaunders | tnx frk2, you pwn |
| 00:31.41 | frk2 | glad i make you happy |
| 00:31.55 | justinu | what country? |
| 00:31.57 | jsaunders | Well, considering you're the only person today to answer one of my 5 or so question, yep. :) |
| 00:32.11 | frk2 | Pakistan |
| 00:32.14 | justinu | ah. |
| 00:32.19 | frk2 | it gets HOT as hell in here |
| 00:32.28 | justinu | i would think heat would cause more serious problems, like kernel panics |
| 00:32.32 | justinu | or system freezes |
| 00:32.41 | justinu | but who knows, maybe the TDM card is real picky. |
| 00:32.42 | frk2 | i guess that would be extreme heat |
| 00:32.55 | frk2 | but i have seen modems act up under heat |
| 00:32.57 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 00:32.59 | justinu | what's the story on your PBX PSU? |
| 00:33.12 | frk2 | power supply |
| 00:33.12 | frk2 | ? |
| 00:33.14 | justinu | yeah |
| 00:33.21 | justinu | is it strong, or a cheapy? |
| 00:33.30 | frk2 | hmm |
| 00:34.07 | justinu | does the TDM400 card need additional power, or does it take everything in from the PCI slot? |
| 00:34.07 | frk2 | my client is known to use cheap ass stuff |
| 00:34.15 | frk2 | it needs power |
| 00:34.27 | justinu | +/- 12VDC, right? |
| 00:34.29 | frk2 | you are right.. PSU can definitely cause issues |
| 00:34.41 | frk2 | 5v or 12v- not sure |
| 00:34.55 | justinu | some systems have voltage monitors... maybe the signal is right on the edge or something |
| 00:35.18 | frk2 | hmm |
| 00:35.23 | generalhan | hey when my systems hurting in KNOW its the heat .... cause 110 IS common here ! lol |
| 00:35.24 | frk2 | will keep that in mind |
| 00:35.33 | justinu | anyways, that's about all I can think of |
| 00:35.45 | frk2 | general where u at? |
| 00:35.48 | generalhan | AZ |
| 00:35.50 | frk2 | thanks dude |
| 00:35.54 | justinu | np |
| 00:35.57 | justinu | what part of pk? |
| 00:35.59 | frk2 | its been quite enlightening |
| 00:36.03 | doughecka | what are the symptoms? |
| 00:36.13 | generalhan | pk ? |
| 00:36.30 | frk2 | the machine is in lahore.. smack in the middle.. known for very dry heat spells |
| 00:36.35 | justinu | islamic republic of pakistan |
| 00:36.49 | frk2 | islamic my ass really |
| 00:36.52 | justinu | lol |
| 00:36.55 | generalhan | haha |
| 00:37.12 | frk2 | i get more messed up here than i ever did in the US |
| 00:38.06 | frk2 | awright- so If my telco dont have disconnect supervision.. fxs_ks is useless |
| 00:38.09 | frk2 | is that right? |
| 00:38.14 | justinu | yeah |
| 00:38.17 | justinu | P for Punjab, A for Afghania (the Afghan areas), K for Kashmir, S for Sindh and tan for Balochistan. An i was later added to the English rendition of the name to ease pronunciation. |
| 00:38.17 | frk2 | im gonna eliminate that variable then |
| 00:38.21 | justinu | ^^ how interesting! |
| 00:38.58 | frk2 | heh |
| 00:39.03 | frk2 | well |
| 00:39.08 | frk2 | its supposed to mean land of the pure |
| 00:39.42 | justinu | yeah, here's the whole quote: |
| 00:39.43 | justinu | The name "Pakistan" (IPA: /paːkɪst̪aːn/) means "Land of the Pure" in Urdu and Persian and was coined in 1933 by Choudhary Rahmat Ali, who published it in the pamphlet Now or Never[5] as an acronym of the names of the "Muslim homelands" of western India — P for Punjab, A for Afghania (the Afghan areas), K for Kashmir, S for Sindh and tan for Balochistan. An i was later added to the English rendition of the name to ease pronunc |
| 00:40.17 | justinu | http://en.wikipedia.org/wiki/Pakistan |
| 00:40.28 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 00:41.08 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 00:50.27 | *** join/#asterisk mgob (n=goldenol@65.171.196.23) |
| 00:50.35 | mgob | hi |
| 00:50.54 | mgob | anyone have working copies of auto config files for the Thomson ST2030 phone? their configuration guide is crypttttic. |
| 00:54.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:55.58 | *** part/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
| 00:57.51 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 01:00.32 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:00.42 | paolob | ciao raga! |
| 01:01.00 | paolob | C'è un amministratore che mi può cancellare Sodoma? È per un cambio di redirect |
| 01:01.33 | paolob | excuse me, I were in the wrong channel... |
| 01:01.34 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:01.41 | Snake-Eyes | hey, any one got ideas as to why I would be getting Got SIP response 400 "Bad Request" back on one Asterisk box and not another. I have compared the files (sip.conf, extensions , any one know of any where else I should look? |
| 01:01.58 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 01:02.03 | *** join/#asterisk adker (n=adker@74-33-211-200.br1.glv.ny.frontiernet.net) |
| 01:07.25 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:12.00 | *** join/#asterisk juice (n=juice@mo-71-50-26-181.dhcp.sprint-hsd.net) |
| 01:15.48 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 01:15.51 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 01:17.50 | *** join/#asterisk opus_ (n=opus@68.216.187.60) |
| 01:17.54 | opus_ | anyone here use app_amd ? |
| 01:30.26 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 01:31.34 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:32.28 | *** join/#asterisk Parvaresh (i=bartali@213.207.218.66) |
| 01:32.59 | Parvaresh | hmm |
| 01:33.11 | Parvaresh | any guide on how to config a cisco 7940 for asterisk |
| 01:33.18 | generalhan | TONS of them |
| 01:33.55 | generalhan | http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx |
| 01:34.02 | Parvaresh | how can i get the firmware |
| 01:34.04 | generalhan | http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960 |
| 01:34.08 | Parvaresh | i dun have an account on cisco |
| 01:34.11 | generalhan | you need a cisco contract |
| 01:34.21 | Parvaresh | dun have any |
| 01:34.28 | Parvaresh | no free release of it? |
| 01:34.30 | generalhan | then you cant get them |
| 01:34.31 | generalhan | nope |
| 01:34.37 | generalhan | like cisco is gonna release ANYTHING for free |
| 01:34.58 | Parvaresh | hmm |
| 01:35.01 | Parvaresh | k |
| 01:41.31 | *** join/#asterisk bluegrass (n=irc1@209-6-185-254.c3-0.wth-ubr1.sbo-wth.ma.cable.rcn.com) |
| 01:41.35 | Parvaresh | which verion of firmware is mostly recommended for asterisk |
| 01:42.04 | Qwell | Parvaresh: whatever the latest currently is |
| 01:42.14 | Parvaresh | cool |
| 01:42.20 | Parvaresh | then v8.xxx should be fine |
| 01:42.29 | Qwell | sure, try it |
| 01:42.47 | Qwell | worst case scenario, you brick it, and send it to me |
| 01:43.34 | *** part/#asterisk bluegrass (n=irc1@209-6-185-254.c3-0.wth-ubr1.sbo-wth.ma.cable.rcn.com) |
| 01:44.42 | generalhan | haha |
| 01:45.57 | *** join/#asterisk bluegrass (n=irc1@209-6-185-254.c3-0.wth-ubr1.sbo-wth.ma.cable.rcn.com) |
| 01:46.20 | generalhan | well ive had about all the work i can handle for a day ... i think its time to go home ! |
| 01:46.48 | generalhan | "have good mash-pitting" every one ! |
| 01:47.02 | *** join/#asterisk nigelr (n=nigelr@ninja.nobiscuit.com) |
| 01:47.13 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:47.22 | Tier_1 | is the asterisk svn down |
| 01:48.31 | protocoldoug | hmm when i send a Musiconhold(), my console says it starts and then stops, any ideas? |
| 01:48.32 | protocoldoug | http://pastebin.com/725830 |
| 01:49.05 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:49.31 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 01:52.04 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:54.32 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 01:56.50 | nigelr | I've got a problem with calls from cellphones coming in on a PRI and going straight to voicemail |
| 01:57.00 | nigelr | a landline calling an unregistered SIP extension with voicemail works fine - they hear the voicemail message and can leave a message |
| 01:57.08 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 01:57.16 | nigelr | a cellphone calling an unregistered SIP extension with voicemail doesn't - they get dead air. |
| 01:57.32 | nigelr | it's like callprogress isn't working properly |
| 01:58.07 | nigelr | if the sip extension is registered, ie. it rings, the cellphone works fine. |
| 01:58.24 | opus_ | nigelr what is the sip extension? a sip phone? |
| 01:59.18 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 01:59.28 | nigelr | in this case it's eyebeam running on a PC, but it's the same if you use a sip phone and leave it unplugged or whatever. |
| 01:59.38 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 01:59.59 | opus_ | you don't have qualify=yes and also your SIP registration timeout is set to something huge |
| 02:00.21 | opus_ | set qualify=30 and set your reg timeout to 20secs if it is a small network |
| 02:00.55 | opus_ | eyebeam any good? the phone right? |
| 02:01.25 | nigelr | yep, seems to work ok. I don't think it's the sip registration, the client has been turned off for more than a day now |
| 02:01.27 | sevard | that softphone takes so much system resources |
| 02:01.29 | sevard | god. |
| 02:01.45 | nigelr | will try the qualify thing though, after I look up what it does :) |
| 02:02.02 | nigelr | it's weird that it's only GSM cellphones that have the problem though |
| 02:02.53 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 02:03.46 | *** join/#asterisk camelon (n=chiardon@206.106.249.85) |
| 02:04.23 | nigelr | opus_: that only changes the sip side of things. this is related to what device the caller is calling from ie. cellphones don't work, landlines do. I'm betting the result would be the same if it was a IAX or a zap extension, |
| 02:04.30 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 02:04.32 | kernel20 | hi there |
| 02:04.56 | camelon | Night |
| 02:06.37 | kernel20 | my xlite client that runs through VPN seems to be distorted, but in his side my voice sounds so well, any ideas? is there any settings in asterisk to fine tune the voice? |
| 02:07.26 | opus_ | you can't do VOIP over VPN because you are basically encapsulating real time traffic in something that doesn't support realtime transport... |
| 02:07.29 | opus_ | i know, it really sucks :) |
| 02:07.43 | *** part/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 02:08.04 | kernel20 | in my case VOIP works over VPN |
| 02:08.44 | kernel20 | MY VOIP server is inside my internal lan |
| 02:08.44 | sevard | opus_: I just finished reading a study that said encapsulation increases the quality of UDP streams |
| 02:09.03 | kernel20 | thats why clients needs to be in VPN so that they can connect to VOIP SERVER |
| 02:09.29 | sevard | I know it makes sense that a VPN would totally destroy the integrity of the call but they said after much real world testing it is quite infact the opposit |
| 02:09.37 | opus_ | sevard : i just read a study that if you paint your car red, you have a 90% chance of winning any car race. |
| 02:09.41 | opus_ | :) |
| 02:09.55 | sevard | opus_: death2u |
| 02:10.05 | nigelr | I guess what I'm looking for is a way to generate a ring signal before going to voice mail. |
| 02:10.06 | kernel20 | sevard? |
| 02:10.36 | kernel20 | what are you implying? |
| 02:10.59 | sevard | kernel20: No idea, I read that study and tested it over my VPN and found it to be true. But if opus_ is an 'expert' perhaps you should listen to him |
| 02:11.18 | kernel20 | true? of what? |
| 02:11.22 | kernel20 | opus_? |
| 02:11.24 | opus_ | no idea, let me check it out |
| 02:11.30 | nigelr | that might make sense. Encapsulating many udp packets together may have the effect of increasing latency but reducing jitter. |
| 02:11.32 | opus_ | i would just use regular RTP over WAN ? |
| 02:11.45 | kernel20 | ok here is the situation |
| 02:12.00 | kernel20 | all client machine is connected via vpn |
| 02:12.01 | sevard | that my call quality was increased after sending UDP data over my VPN. I found SIP signaling on 5060 took longer to set up and RTP streams took a couple miliseconds longer to settle |
| 02:12.11 | sevard | but I found there wern't any blips or background badness. |
| 02:12.14 | camelon | with regard to a PRI wich be the best functional choice looking to decrese faults: let the PRI clocked internally or clocked by the telco? |
| 02:12.23 | sevard | It took a whole lot more bandwidth to encapsulate. |
| 02:12.36 | kernel20 | what the SIP? |
| 02:12.53 | sevard | My analysis may have been fubar though. Things always go wrong in scientific tests |
| 02:12.54 | opus_ | long over VPN or faster over VPN? and why ? |
| 02:13.14 | kernel20 | RTP? |
| 02:13.19 | kernel20 | i getting lost here |
| 02:13.22 | sevard | It seemed to take longer to settle and took longer for signaling, but once I was in call the call quality was great. |
| 02:13.27 | sevard | kernel20: you need thebook |
| 02:13.32 | sevard | ~thebook |
| 02:13.33 | jbot | from memory, thebook is somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 02:13.44 | brodiem | what is it that people use SER for in conjunction with asterisk? Is it mainly to provide a proxy service when dealing with multiple * machines? |
| 02:14.08 | opus_ | kernel20, VOIP won't work over VPN no matter what, is what I have experienced so far in my testing. |
| 02:14.33 | kernel20 | hmm so what should i do? |
| 02:14.38 | sevard | opus_: In my testing VoIP works like a charm over a VPN and I know several people here who use it. You just have to have the _bandwidth_ for it. |
| 02:14.46 | kernel20 | i need to have my server to have a public ip? |
| 02:14.55 | opus_ | servard what VPN do you use? |
| 02:15.16 | sevard | opus_: I tested on openvpn and hamachi |
| 02:15.19 | opus_ | kernel20, yes. external IP |
| 02:15.29 | opus_ | servard: cool |
| 02:15.31 | sevard | kernel20: what your experiencing might be a nat issue ? maybe ? |
| 02:15.43 | sevard | kernel20: I don't know |
| 02:15.53 | kernel20 | i can ping to both nodes with no problem |
| 02:16.35 | kernel20 | so u mean if i have voip on vpn, it requires a lot of bandwidth? |
| 02:16.45 | opus_ | servard: then I guess I will try it over again. theoretically if you had a strong RTP implementation and a good jitter buffer, and your VPN respected 802.1p, it could work but not even the big boys like cisco can get that right so why bother wasting time on it is my expri |
| 02:16.47 | sevard | encapsulation always requires more bandwidth |
| 02:16.51 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:16.52 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:17.21 | kernel20 | hmmm |
| 02:17.31 | protocoldoug | what would cause a MusicOnHold() to start and then stop immediately, http://pastebin.com/725830 |
| 02:17.45 | opus_ | encapsulating RTP packets is pointless when you can just send it out the WAN gateway |
| 02:18.12 | opus_ | in fact, there are some VPN protocols that use RTP!!! |
| 02:18.26 | kernel20 | i use openswan opus_ |
| 02:18.33 | kernel20 | but dunno of RTP thing |
| 02:18.47 | opus_ | because it allows you to reinvite VPN sessions, so you can roam seamlessly your session:) |
| 02:19.16 | sevard | bbiab |
| 02:19.48 | opus_ | i think its an awesome use of RTP to VPN over it, like the reverse of what you guys are talking about |
| 02:20.25 | kernel20 | hmmm so your recommendation is that i need to have my voip server have a public ip |
| 02:20.43 | kernel20 | so that RTP wont be an issue? |
| 02:21.12 | [TK]D-Fender | protocoldoug : That looks perfectly normal to me. You call the MoH app, it start, adn then stops because the channel hung up. |
| 02:21.32 | opus_ | yes |
| 02:21.32 | kernel20 | opus_? |
| 02:21.44 | kernel20 | is that yes for me? |
| 02:21.51 | opus_ | <kernel20> so that RTP wont be an issue? <--- so that VPN wont be an issue, yes. |
| 02:22.02 | kernel20 | hmmm |
| 02:22.06 | kernel20 | ok ill try it |
| 02:22.10 | kernel20 | thanks for the advise |
| 02:22.24 | opus_ | kernel20, and make sure every piece of hardware end-to-end supports 802.1p. otherthen that you should never have problems. |
| 02:22.38 | protocoldoug | [TK]D-Fender, yeah the music never plays, the hang up is a few seconds later... but this brings me to another question -- you need to use an external app to play an mp3? |
| 02:22.47 | kernel20 | opus_: i dont have any hardwares yet |
| 02:22.54 | kernel20 | all are softphone at the moment |
| 02:23.11 | kernel20 | i only installed asterisk |
| 02:23.17 | kernel20 | clients use softphones |
| 02:23.25 | protocoldoug | [TK]D-Fender, cause maybe that's the problem, i'm not trying to use an extra app, like mpg123 or whatnot |
| 02:23.32 | opus_ | kernel20, you are going to hate this. softphones are really crappy as well |
| 02:23.44 | kernel20 | huh? |
| 02:23.48 | nigelr | camelon: clocked by the telco, definitly. |
| 02:23.55 | [TK]D-Fender | protocoldoug :Not necessarily |
| 02:24.02 | kernel20 | but when i call to local lan, voice seems to be great |
| 02:24.06 | opus_ | the only softphone that has ever got it right is Skype, other then that I haven't been impressed with any other softphone |
| 02:24.21 | [TK]D-Fender | protocoldoug : You need to instal "format_mp3" which comes with the asterisk-addons pacage downloadable seperately |
| 02:24.25 | opus_ | and you can't use Skype with asterisk yet. |
| 02:24.31 | kernel20 | i know |
| 02:24.41 | protocoldoug | [TK]D-Fender, ahhh ha that i haven't done, *thumbs up* thank you |
| 02:24.49 | kernel20 | the reason i set up asterisk is for corporate use |
| 02:24.57 | kernel20 | i guess x-lite works great |
| 02:25.20 | [TK]D-Fender | protocoldoug : Glad to help |
| 02:25.49 | kernel20 | opus_: u mean to say even if I have the VOIP server a public IP still the voice from other client machine over vpn will have problems? |
| 02:26.28 | kernel20 | the one i like with asterisk is the conferencing (meetme) |
| 02:26.41 | kernel20 | it works very great, skype conferencing is a mess |
| 02:27.44 | camelon | nigelr: but if the QoS from the Telco is so bad . . . is to risky doing the timme with the card? wich could be the consequence from this?TIA |
| 02:27.55 | kernel20 | opus_? |
| 02:28.19 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:28.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:31.18 | *** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 02:31.36 | watchy | anyone got a url for a polycom presence setup? |
| 02:32.14 | [TK]D-Fender | watchy : Its all over the WIKI. |
| 02:33.12 | [TK]D-Fender | watchy : look up "presence" in sip.cfg, set to "1", reboot your phone, add some buddy's and enable buddy watch on them. Make sure they match a hint exten in your phone's context. |
| 02:35.56 | *** join/#asterisk BugKham (i=BugKham@202.8.86.162) |
| 02:36.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:36.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:36.30 | nigelr | camelon: the only important thing is that your end matches the telco. The telco isn't going to listen to timing info from you, so you must use timing info from then. |
| 02:36.35 | nigelr | from them I mean |
| 02:37.19 | nigelr | otherwise you will get poor quality calls or no calls at all. |
| 02:37.32 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 02:38.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 02:40.48 | camelon | nigelr: but now getting the timing from the telco the calls quality is less than poor! what strategy to overcome that? |
| 02:42.48 | ManxPower | camelon, your problem is not timing. |
| 02:43.21 | websae | ManxPower: do you work for a large VoIP company? |
| 02:43.34 | ManxPower | websae, no. |
| 02:43.52 | camelon | ManxPower: at this time wich are your diagnosis? |
| 02:44.23 | ManxPower | camelon, I don't have one. But if you are getting your timing from the telco and you have poor call quality then your problem is not a timing problem. |
| 02:44.42 | ManxPower | IRQ conflicts are another common cause of poor call quality |
| 02:46.01 | camelon | ManxPower: allways the IRQs are good . . .i've pasted it her without any comment from the comunity |
| 02:46.47 | ManxPower | poaste the URL again |
| 02:47.16 | Malthus | ManxPower, hey |
| 02:47.26 | ManxPower | Hello, Malthus |
| 02:47.27 | camelon | ManxPower: a minute . . . must connect |
| 02:47.46 | Malthus | ManxPower, I tried the mixed voice/data T1 |
| 02:47.48 | brodiem | I know this isn't an SER channel, but can SER as a SIP proxy direct its calls to asterisk machines as a load balancer? And detect broken network links and use a failover? |
| 02:47.52 | Malthus | it worked without a hitch :) |
| 02:48.05 | ManxPower | Malthus, cool |
| 02:48.24 | ManxPower | brodiem, You didn't search the Wiki, did you? |
| 02:48.34 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 02:49.25 | brodiem | ManxPower, well I found this key phrase, but didn't include the "automatic" part I was looking for: Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population |
| 02:50.49 | ManxPower | http://www.google.com/search?hl=en&q=site%3Alists.digium.com+ser+failover&btnG=Google+Search |
| 02:51.27 | *** join/#asterisk Quension (i=quension@66.7.99.222) |
| 02:51.58 | brodiem | ManxPower answers the question.. |
| 02:52.46 | ManxPower | It's not about what you know -- it's about knowing how to find the answers you don't know. |
| 02:55.05 | *** join/#asterisk autobus (n=autobus@80.172.14.203) |
| 02:55.09 | autobus | hi all. |
| 02:55.19 | autobus | its possible help-me.. |
| 02:55.20 | autobus | ? |
| 02:55.26 | camelon | ManxPower: http://pastebin.ca/57000 |
| 02:55.26 | Malthus | sure |
| 02:55.48 | autobus | good |
| 02:56.23 | ManxPower | camelon, do you have 4 CPUs or does the system have hyper threading? |
| 02:56.24 | camelon | ManxPower: like a zen master . . .you are in the parh |
| 02:57.11 | camelon | ManxPower: have hyper threading |
| 02:57.16 | [TK]D-Fender | ManxPower : Just pray its not an oncoming train ;) |
| 02:57.32 | ManxPower | camelon, turn off hyper threading. |
| 02:57.42 | autobus | i hav this situation: |
| 02:57.43 | autobus | exten => 100,1,Set(CALLER=${CALLERID(num)}) |
| 02:57.43 | autobus | exten => 100,2,Hangup() |
| 02:57.43 | autobus | exten => h,1,Wait(10) |
| 02:57.43 | autobus | exten => h,1,System(/var/lib/asterisk/scripts/callback.sh ${CALLERID(num)}) |
| 02:57.48 | autobus | the problem is: |
| 02:58.04 | autobus | the intruction wait not function in this case. |
| 02:58.13 | autobus | but i dont understand why |
| 02:58.24 | ManxPower | camelon, http://www.google.com/search?hl=en&q=site%3Alists.digium.com+hyperthreading+problem&btnG=Google+Search |
| 02:58.28 | ManxPower | autobus, use pastebin.ca |
| 02:58.30 | ManxPower | ~pb |
| 02:58.31 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:58.47 | watchy | tk: make a hint exten for every phone? |
| 02:58.49 | [TK]D-Fender | autobus : multiple problems. you have 2 priorities numbed "1" in "h" |
| 02:59.17 | camelon | ManxPower: in the same path like you . . .finding answers i don`t know . . . why i must turn off hyperthreading? |
| 02:59.17 | [TK]D-Fender | autobus : second "h" will never get called if YOU explicitly kill the channel with "hangup" |
| 02:59.29 | autobus | no sorry this example is nott correct. i have correct in extensions.conf |
| 02:59.35 | ManxPower | camelon, I don't know. |
| 02:59.45 | ManxPower | But to many people have reported it fixed problems..... |
| 02:59.48 | [TK]D-Fender | autobus : pastebin the sample you'd like help with then |
| 02:59.58 | autobus | hum |
| 03:00.10 | watchy | tk: make a hint exten for every phone? |
| 03:00.12 | ManxPower | autobus, PASTE the extensions.conf info, do not type it or you will waste our time. |
| 03:00.17 | [TK]D-Fender | watchy : yup |
| 03:00.37 | ManxPower | s/ to / so / |
| 03:00.37 | watchy | and that hint is just used for presence and thats it correct? |
| 03:00.38 | autobus | ManxPower sorry |
| 03:01.19 | autobus | [TK]D-Fender , the channel hangup sucessful |
| 03:01.35 | [TK]D-Fender | watchy : Correct, it tells * to watch the tech/device and associate it with the exten you provide. then any phones using the context containing the hints can track them |
| 03:01.35 | autobus | but im interesting creat one wait time. |
| 03:02.06 | [TK]D-Fender | autobus : when yuo call "hanup" you call IMMEDIATLY dies. The "h" exten will NOT get called. |
| 03:02.34 | autobus | and, what is a solution? |
| 03:02.49 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:03.01 | [TK]D-Fender | autobus : pastebin what you have now so I can have an idea what you want to do. |
| 03:03.03 | camelon | ManxPower . . the other data in the paste are OK from your experience? |
| 03:03.05 | watchy | tk: sweet |
| 03:03.45 | ManxPower | camelon, yes. |
| 03:05.26 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 03:05.43 | justinu|laptop | Tessier Ashpool, S.A. |
| 03:06.43 | camelon | ManxPower: thanks . . .what other aspects (probably causes) to look? |
| 03:08.09 | [TK]D-Fender | justinu|laptop : Were you here when we were getting a info on the new Polycom phones? |
| 03:09.23 | justinu|laptop | nope |
| 03:09.26 | watchy | anyone got the polycom 1.6.6 firmware? |
| 03:09.35 | justinu|laptop | what'd I miss? |
| 03:09.46 | watchy | id trade a taco for it |
| 03:10.36 | [TK]D-Fender | justinu : IP 430. A 301 w/ PoE builtin (no special cable), and comes with brick, speakerphone, and rumoured backlight. All in an IP 301 frame. |
| 03:10.54 | [TK]D-Fender | watchy : What model(s) do you have? |
| 03:10.57 | justinu|laptop | nice, price? |
| 03:11.12 | watchy | tk: i got like 26 501s and 4 601s |
| 03:11.16 | autobus | please see: http://pastebin.com/725905 |
| 03:11.22 | watchy | i need it for the 601 to gimme more then 7 things |
| 03:11.24 | [TK]D-Fender | watchy : Attendant modules? |
| 03:11.26 | watchy | 7 buddys |
| 03:11.40 | watchy | yea i got 2 601s with 1 attendant and 2 601s with 2 |
| 03:12.04 | autobus | the sintax its not corrrect for make one wait time? |
| 03:12.31 | websae | *syntax |
| 03:12.34 | autobus | after Hangup command |
| 03:12.44 | watchy | exten => 500,hint,SIP/peername |
| 03:12.52 | watchy | should 500 be the extension of the phone? |
| 03:13.15 | [TK]D-Fender | autobus : You don't want to do "hangup". you want to basically tell the CALLER to hangup so your scripts gets called. So you should play a recording saying "please hang up now" and keep looping it till they do hangup. THEN your "h" will get called |
| 03:13.28 | [TK]D-Fender | watchy : Good reason to want 1.6.6. |
| 03:13.29 | justinu|laptop | [TK]D-Fender: any pricing info? |
| 03:13.55 | watchy | tk: you got it layin around? |
| 03:14.23 | [TK]D-Fender | justinu : Not yet, but there isn't a lot of room between the 301 & 501 $ wise. basically I have a feeling either the line will go up in price or the 301 & 501 my budge a bit on both sides to compensate it. |
| 03:14.28 | autobus | but the objective is, not Answer the call |
| 03:14.38 | autobus | this is the callback aplication. |
| 03:14.50 | [TK]D-Fender | justinu|laptop : Basically the IP 430 would become the defacto business phone for * IMO. |
| 03:15.00 | Qwell | 430? |
| 03:15.13 | [TK]D-Fender | Qwell : New model in testing. |
| 03:15.14 | Qwell | oh. nm |
| 03:15.44 | autobus | [TK]D-Fender you understand my objective? |
| 03:15.58 | justinu|laptop | [TK]D-Fender: sounds promising |
| 03:16.15 | [TK]D-Fender | autobus : Yes, person calls that exten, and then it waits for them to disconnect, then calls them right back. |
| 03:16.22 | autobus | yes |
| 03:16.24 | autobus | right |
| 03:17.02 | autobus | but its necessary wait 5 seconds. for phone avaible to receive the call. |
| 03:17.10 | autobus | understand? |
| 03:17.13 | [TK]D-Fender | autobus : but for it to place the call AFTER the disconnect the only real wy to do it is to wait for the CALLER to hangup. There is no way for * to hang up on the call on purpose and then continue on by itself. Its jsut the way things work. |
| 03:18.01 | [TK]D-Fender | autobus : Yes I understand the delay. You just need to loop the caller with a message saying "please hang up now" and wait for the caller to ahng up. its the only way that will work. |
| 03:19.34 | autobus | its possible creat one script in php or another lanuage for wait 5 seconds |
| 03:19.46 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 03:19.56 | Jaxxan | hey guys |
| 03:20.07 | Jaxxan | ~zaptel |
| 03:20.09 | jbot | from memory, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 03:20.26 | Jaxxan | I have a problem with outbound calls thru my PRI |
| 03:20.29 | Jaxxan | it's double ringing |
| 03:20.42 | Jaxxan | Asterisk is generating Ringing as well as the MSC |
| 03:20.50 | Jaxxan | this is only for outbound calls |
| 03:20.59 | {zombie} | are you adding the 'r' flag to the Dial command? |
| 03:21.08 | Jaxxan | let me check |
| 03:21.33 | Jaxxan | exten => s,104,Dial(${TRUNKGROUP}/${ARG1}) |
| 03:21.49 | Jaxxan | where ARG1 is the phone number |
| 03:21.57 | Jaxxan | so no, i'm not |
| 03:22.54 | {zombie} | ok. I don't know if there's an option to suppress the ringing |
| 03:23.13 | Jaxxan | this started when i upgrade from zaptel-1.0.9.1 to zaptel-svn |
| 03:23.19 | Jaxxan | upgraded |
| 03:23.20 | autobus | [TK]D-Fender its possible run other scripts after hungap right? |
| 03:24.47 | [TK]D-Fender | autobus : yes. |
| 03:24.50 | Malthus | deadagi |
| 03:25.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:25.02 | nigelr | ManxPower: did you see my earlier question re. cellphone calls going straight to voice mail? |
| 03:25.33 | autobus | [TK]D-Fender what is the objective of: h in the intruction? |
| 03:26.27 | sevard | autobus |
| 03:26.32 | sevard | ~thebook |
| 03:26.33 | jbot | thebook is probably somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 03:26.35 | Jaxxan | ~overlapdial |
| 03:27.13 | [TK]D-Fender | autobus : yeah you really should download THEBOOK and read up on *'s standard extensions. |
| 03:27.29 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 03:28.01 | autobus | ok thanks |
| 03:29.39 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
| 03:30.31 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:32.29 | websae | sevard: did you email me? |
| 03:34.59 | littleball | hello, in the sip.conf file, how to register my own asterisk with remote sip provider? especially, if the remote sip server host name is different than the domian/realm name? |
| 03:36.48 | ManxPower | littleball, sip.conf.sample |
| 03:39.43 | autobus | [TK]D-Fender agi wait for digit command function and solve my solution? |
| 03:40.27 | autobus | Usage: WAIT FOR DIGIT <timeout> |
| 03:41.03 | [TK]D-Fender | autobus : I strongly suggest you follow my hint to continuosly playback the "please hangup now" recording and just wait for them to do it. |
| 03:41.45 | autobus | but its necessary the system answer the call right? |
| 03:43.44 | *** join/#asterisk papa_e (i=papa_e@ip68-4-40-21.pv.oc.cox.net) |
| 03:44.04 | papa_e | hrmm, no real-time voice recognition in * |
| 03:47.59 | [TK]D-Fender | autobus : You can't have a call hanup that you never answered. |
| 03:48.26 | [TK]D-Fender | papa_e : You sure looked hard, didn't you? Try Sphinx. |
| 03:49.59 | papa_e | tk, uh, yeah, have sphinx2 setup and it sure doesn't work well |
| 03:51.10 | [TK]D-Fender | papa_e : Yeah I hear its spotty, but can work. |
| 03:51.37 | papa_e | well, even for something simple like yes/no or 1,2,3,4, it's unreliable as hell |
| 03:52.02 | papa_e | like 70% accuracy at best, on a crystal clear link, with the caller speaking LOUD |
| 03:52.48 | papa_e | but then again, it's a complex piece of software, and i've probably missed some configuration options somewhere |
| 03:56.14 | Jaxxan | god i dont even know what i'm searching for |
| 03:59.31 | Jaxxan | can someone explain to me what overlap dialing is ? |
| 04:00.13 | Jaxxan | sending overlap digits.. wtf is that ? |
| 04:01.07 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:02.24 | Jaxxan | ahhh, there it is |
| 04:04.59 | justinu|laptop | Jaxxan: try priindication=outofband |
| 04:05.14 | Jaxxan | i just found http://bugs.digium.com/view.php?id=6690 |
| 04:06.23 | Jaxxan | that describes my problem to a T |
| 04:06.46 | Jaxxan | where does priindication=outofband go ? |
| 04:06.49 | Jaxxan | zapata.conf ? |
| 04:06.52 | justinu|laptop | yes |
| 04:07.10 | *** join/#asterisk bigmac4444 (n=mtur2848@CPE-144-131-193-158.qld.bigpond.net.au) |
| 04:07.22 | bigmac4444 | G'day all =) |
| 04:07.30 | Jaxxan | but that's just for busy/congestion right ? |
| 04:08.03 | justinu|laptop | there's an alerting in PRI also |
| 04:08.25 | justinu|laptop | If you set progressinband=never or you don't set anything and let it default, then you do in fact get inband progress (dual ring sound). If you set it to "yes" or "no" then you do NOT get the dual ring sound. |
| 04:08.29 | justinu|laptop | from the notes |
| 04:08.35 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-117-03.telkomadsl.co.za) |
| 04:09.26 | justinu|laptop | so maybe not priindication |
| 04:09.42 | justinu|laptop | maybe it's in sip.conf |
| 04:10.21 | Jaxxan | i should be able to just stick that in [general] right ? |
| 04:10.25 | justinu|laptop | i think so |
| 04:12.39 | Jaxxan | yup that worked |
| 04:12.47 | [TK]D-Fender | ok, Im done fotr the night. Later all |
| 04:13.00 | justinu|laptop | good deal |
| 04:13.25 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:13.31 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 04:14.48 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 04:17.21 | autobus | its easy creat deadeAGI for wait 2 seconds? |
| 04:17.32 | autobus | its possible help me. |
| 04:20.53 | justinu|laptop | anyone ever use the G() option to app_dial? |
| 04:21.14 | Jaxxan | badass, thanks justinu |
| 04:21.35 | justinu|laptop | np |
| 04:23.23 | Snake-Eyes | hi, any one got ideas as to why I would be getting Got SIP response 400 "Bad Request" back on one Asterisk box and not another. I have compared the files (sip.conf, extensions , any one know of any where else I should look? |
| 04:23.46 | Jaxxan | check your firewall? are you going thru nat ? |
| 04:31.45 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:32.45 | autobus | its easy creat deadeAGI for wait 2 seconds? |
| 04:32.50 | littleball | hello, does cisco AS5300 support SIP? |
| 04:32.54 | autobus | its possible help me! |
| 04:39.47 | papa_e | snake, like jaxxan mentioned, it's a network problem |
| 04:40.16 | papa_e | turn on sip debugging and you'll pinpoint it in two seconds |
| 04:40.24 | Jaxxan | autobus: i have no clue what you're asking. rethink and retype your question. |
| 04:40.35 | SplasPood | littleball: yes, but cisco's website might be a better place to ask :P |
| 04:43.25 | Jaxxan | littleball: i got 4 AS5350's as our SIP Media Gateways |
| 04:44.20 | justinu | nice |
| 04:45.43 | Jaxxan | wow |
| 04:45.49 | Jaxxan | i closed all but 1 trouble ticket today |
| 04:45.54 | Jaxxan | i should take tomorrow off |
| 04:45.55 | justinu | lucky you |
| 04:47.04 | watchy | anyone here skilled in fxotune? |
| 04:47.19 | Jaxxan | so tomorrow i get to move from wireless backhauls at my residence for internet to our OC3 Fiber... i'm excited. |
| 04:47.26 | Snake-Eyes | Jaxxan, i have replaced the box eg given it the same ip and everything, only difference is asterisk configs |
| 04:47.56 | justinu | you have fiber optic termination equipment at home? |
| 04:47.58 | Jaxxan | did you turn on sip debugging like papa_e stated ? |
| 04:48.06 | Snake-Eyes | yes |
| 04:48.11 | Jaxxan | justinu: i have a cell site in my front yard |
| 04:48.25 | justinu | is it yours? |
| 04:48.36 | Jaxxan | company i work for (= |
| 04:48.41 | justinu | interesting arrangement |
| 04:48.42 | Jaxxan | might as well be mine, i have keys |
| 04:49.22 | Snake-Eyes | Jaxxan, I have even captured the packest, they look almost the same |
| 04:49.26 | watchy | no fxotune masters here? |
| 04:49.50 | Jaxxan | Snake-Eyes: if you've ruled out network problems, then you have a problem with your configs somewhere, when does the error occur ? |
| 04:50.02 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:50.47 | shido6 | i think im getting the hang of this stupid max tnt |
| 04:51.42 | Snake-Eyes | Jaxxan, The box sends the invite (both invites look the same, maybe some minor diff) then the other side (pstn termination provider) sends back a 400 "Bad Request" with the one box |
| 04:52.07 | Snake-Eyes | its using sip |
| 04:52.41 | Snake-Eyes | I've gone through all the configs and I can't see what might cause this ... |
| 04:53.10 | Jaxxan | so you got an asterisk box trying to send a call to your upstream provider and they're saying 400 bad request ? |
| 04:53.30 | znoG | how do you know your provider is working OK? |
| 04:53.34 | papa_e | snake, maybe your provider needs to clear the sip registration |
| 04:53.35 | znoG | can you make a call using a standard SIP client? |
| 04:53.36 | Snake-Eyes | looked at extensions and sip.conf, wondering if theres some where else I should look |
| 04:53.43 | Jaxxan | and you're sure you got your sip.conf entries are correct for registration ? |
| 04:54.09 | Snake-Eyes | provider doesnt need/want registration |
| 04:54.21 | Snake-Eyes | it done by IP |
| 04:54.44 | Jaxxan | and you turned your old box off ? |
| 04:54.56 | Snake-Eyes | swapped ips |
| 04:55.08 | Snake-Eyes | when ever i put the old box back it works fine |
| 04:55.25 | Snake-Eyes | all sip calls work |
| 04:55.35 | Jaxxan | maybe arp hasn't updated ? |
| 04:55.39 | Jaxxan | i dunno |
| 04:56.02 | znoG | Snake-Eyes: your old box has the same asterisk version running? |
| 04:56.14 | littleball | hello, i am looking for pstn termination in india ocean countries. who has? like Bangladesh and and Sri Lanka |
| 04:56.17 | Snake-Eyes | this new box has frontend install on it, that has changed some configs |
| 04:56.22 | Snake-Eyes | znoG, yes |
| 04:57.17 | Jaxxan | you set bindaddr=XXX.XXX.XXX.XXX in sip.conf right ? |
| 04:57.19 | znoG | Snake-Eyes: there has to be a difference in the configs .. sip packets are that similar, huh? |
| 04:57.42 | Snake-Eyes | only thing i can think of is that this new frontend has changed some some where, but i cant find it |
| 04:58.01 | Snake-Eyes | Jaxxan, bind=0.0.0.0 for both |
| 04:58.18 | Jaxxan | try binding it to the ip address your provider expects |
| 04:58.24 | Snake-Eyes | znoG, yea, i just dont know where it is ;( |
| 04:58.32 | Snake-Eyes | Jaxxan, ok |
| 04:58.56 | Jaxxan | are you behind a NAT ? |
| 05:00.09 | Jaxxan | if you are make sure you set your externip= and localnet= entries |
| 05:00.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:00.25 | Snake-Eyes | the server isnt behind a nat |
| 05:00.45 | Snake-Eyes | bind didnt work ;( |
| 05:01.09 | justinu|laptop | Jaxxan: what area are you located in? |
| 05:01.14 | Jaxxan | american samoa |
| 05:01.15 | watchy | any fxotune masters here? |
| 05:01.36 | justinu|laptop | cool, interesting place |
| 05:01.41 | Jaxxan | small |
| 05:01.54 | Jaxxan | but i do a job that's more of a hobby and i get paid to play with technology. |
| 05:02.41 | Jaxxan | i dunno Snake-Eyes, maybe pastebin your configs |
| 05:02.52 | Jaxxan | and the errors |
| 05:03.18 | Jaxxan | i'm going home though, end of the day for me. |
| 05:03.39 | Jaxxan | talk to ya'll later |
| 05:04.29 | *** join/#asterisk mitcheloc (n=mitchelo@209.76.232.56) |
| 05:05.17 | Snake-Eyes | noo come back ;( |
| 05:05.30 | watchy | i have 8 lines hooked up on 2 tdm400s |
| 05:05.40 | watchy | the lines on the first card echo so fucking baddly |
| 05:05.44 | bigmac4444 | question please... |
| 05:05.46 | *** join/#asterisk jontow (i=jontow@hijacked.us) |
| 05:06.08 | watchy | but the 2nd card fxotune fixes the echo insanely well |
| 05:06.57 | bigmac4444 | can asterisk append numbers to the front of a dialed number before passing to a gateway? eg: I type 1234 on VoIP phone and asterisk will add 444 in front making it 4441234 then pass that whole number to the gateway? |
| 05:07.13 | watchy | yea i dunno how |
| 05:07.16 | watchy | but thats possible |
| 05:07.17 | bigmac4444 | LOL |
| 05:07.24 | bigmac4444 | thats a start =) thx |
| 05:07.44 | watchy | im kinda a newbie at stuff like that |
| 05:07.54 | bigmac4444 | that makes 2 of us |
| 05:08.18 | bigmac4444 | whos joy? |
| 05:08.51 | Snake-Eyes | mwuhaha found the file where the problem is |
| 05:09.00 | Snake-Eyes | happy = joy |
| 05:09.51 | Snake-Eyes | http://www.answers.com/joy&r=67 |
| 05:10.17 | *** join/#asterisk chendy (n=Daiyan_C@218.80.71.156) |
| 05:11.00 | watchy | im about to kill this tdm card |
| 05:11.20 | bigmac4444 | found info on append trailing digits, but not leading digits. used to be one called prefix but is obsolete now? |
| 05:12.41 | watchy | echo ratio = 0.0060 (68.1 / 11367.0) |
| 05:12.44 | watchy | is that good? |
| 05:12.48 | watchy | i would think so |
| 05:13.30 | Qwell | ~striplsd |
| 05:13.37 | Qwell | ~striplastdigit |
| 05:13.39 | jbot | striplastdigit is probably ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
| 05:14.09 | Qwell | or am I totally misunderstanding the question? |
| 05:14.24 | watchy | qwell: he wants to add more digits |
| 05:14.37 | watchy | like someone dials 8366666 he wants it to be 4448366666 |
| 05:14.50 | Qwell | umm |
| 05:15.02 | Qwell | SET(SOMEVAR=123${SOMEVAR}) |
| 05:15.09 | watchy | unless im looking at the wrong dudes question |
| 05:15.23 | watchy | qwell: you know anything about tdm cards? |
| 05:20.43 | jontow | hrm.. wtf.. my SPA2002 seems to not work :/ |
| 05:21.01 | jontow | will pickup an IP via DHCP, and I can set it to static via the IVR.. but i can't communicate with it |
| 05:21.58 | bigmac4444 | yea, append digits to the front of a dialed number coming into asterisk |
| 05:22.14 | bigmac4444 | dial 1234, asterisk adds a leading 444 |
| 05:22.28 | bigmac4444 | becoming 4441234 |
| 05:24.07 | jontow | :( grr |
| 05:24.16 | jontow | i think it may be toast, that sucks |
| 05:29.07 | bigmac4444 | Dial(Zap/g0/555${EXTEN}) |
| 05:31.10 | bigmac4444 | ok, thats easy enough, now the leading 0 needs to be dropped before the append. |
| 05:32.59 | bigmac4444 | exten => _07X.,4,Dial(SIP/555${EXTEN:0}@${PSTN_GATEWAY},60,tT) - something like that i presume, but i believe the order it works would be incorrect |
| 05:37.10 | bigmac4444 | opps, the 0 would be 1 for one leading digit |
| 05:38.25 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 05:50.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 05:52.03 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:55.43 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 06:05.54 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 06:09.54 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:11.38 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:13.10 | *** join/#asterisk buzzdee (n=buzzdee@174.56.204.212.ediscom.de) |
| 06:24.05 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 06:26.30 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 06:29.46 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 06:36.34 | *** join/#asterisk Tili (n=Tili@cm33.gamma249.maxonline.com.sg) |
| 06:38.19 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 06:38.59 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 06:44.50 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:45.00 | *** join/#asterisk lorinc (n=ang@caracas-2722.adsl.interware.hu) |
| 06:45.02 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:46.43 | [hC] | anyone here worked with sangoma a200/remora cards? |
| 06:47.33 | [hC] | i am presuming that you have to specifically ask for daughter boards and the backplane connector? they sent me two unique cards instead. |
| 06:54.26 | *** join/#asterisk copz (n=sadf@nrjsoft13.networx-bg.com) |
| 06:54.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:00.01 | *** join/#asterisk mikasaari (n=mikasa@blender/coder/mikasaari) |
| 07:02.11 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:03.55 | mikasaari | Hi. I do have Digium TE110P card working with Asterisk. I have configuration which is working nicely with Digium high end card (in different E1 line). The line is configured like bchannels: 1-15,17-30 and dchannel 16. Lower channels are working correctly but when 17-30 channels are used, there is no voice at all. Now the question is, should TE110P card support all those bchannels 1-15 and 17-30 ? (I am big noobie) |
| 07:04.23 | opus_ | yeah digium sucks, you should have bought sangoma:) |
| 07:04.48 | mikasaari | Is the reason the digium card, or could it be in the E1 line ? |
| 07:05.04 | opus_ | your D channel is fucked up, you should open a ticket with your provider |
| 07:05.12 | opus_ | ALSO you are running with Digium |
| 07:05.18 | mikasaari | :) |
| 07:05.22 | mikasaari | I opened the ticket |
| 07:05.34 | opus_ | switch to sangoma if you are running E1 |
| 07:05.45 | opus_ | sorry, i probably pissed off a bunch of fan boys |
| 07:05.47 | opus_ | fuck'em |
| 07:05.49 | mikasaari | They claim that there is configuration fault in our digium configs ( I disagree with them, same configs work in different card) |
| 07:05.58 | mikasaari | ;) |
| 07:06.01 | opus_ | your d channel is misconfigured |
| 07:06.07 | mikasaari | this is clear ! |
| 07:06.09 | mikasaari | Thanks a lot |
| 07:06.49 | oej | mikasaari: Do you have the switch on the board set for E1? |
| 07:06.57 | mikasaari | oej, I have yes |
| 07:07.13 | mikasaari | oej, And I used the jumper to enable the E1 |
| 07:07.15 | oej | mikasaari: The digium card fully supports PRI over E1, so there has to be a config error |
| 07:07.45 | oej | Check your zapata conf, that you have correct channel numbers |
| 07:07.55 | drray | and that ALL channels are configured |
| 07:08.01 | drray | even if you are not using them |
| 07:08.15 | drray | er, all spans |
| 07:08.29 | oej | drray: Why is that? |
| 07:08.48 | mikasaari | oej, I checked those and all seems to be correctly configured. All channels get up when starting the asterisk |
| 07:08.53 | oej | I just do configure all spans out of habit, but did not know it was a problem |
| 07:09.01 | opus_ | oej, i see people are into voodoo when figuring out asterisk problems. that sucks! |
| 07:09.05 | drray | I had an issue where not configuring a span boned me |
| 07:09.27 | oej | opus_: Well, roll up your sleeves and try to fix it. That's what I did. |
| 07:09.41 | opus_ | asterisk is pretty good with error messages -- apparently not that good jesus |
| 07:10.02 | drray | are you running asterisk -vvvvvvvvvvc |
| 07:10.03 | drray | ? |
| 07:10.06 | oej | Well, this could be either zaptel or asterisk |
| 07:10.20 | opus_ | drray, you should test more and read more RFCs, what you said is completely illogical. having a configured span on a dead port has NO effect on a good port:) |
| 07:10.42 | oej | What RFCs cover that? |
| 07:11.12 | drray | opus you are probably right, however, when I was having problems getting my PRI working, configuring all the spans solved it |
| 07:11.15 | opus_ | oej, it doesn't exist/. |
| 07:11.22 | opus_ | oej, but should |
| 07:11.24 | drray | or my issue went away at teh same damn time |
| 07:11.25 | mikasaari | I fear that in my case the E1 Line is misconfigured (I claim this because they have fixed the line 3 times now and first time they connected the line to wrong place, next time they misconfigured the Node and now they do not want to answer to my calls) |
| 07:11.39 | *** join/#asterisk chendy (n=Daiyan_C@218.80.61.245) |
| 07:11.40 | opus_ | drray, thats voodoo debugging |
| 07:11.56 | opus_ | drray, try the inverse and it wil still be the same. |
| 07:12.21 | drray | why would I try the inverse now that my card is working? isn't that snatching defeat from the jaws of victory? |
| 07:12.29 | oej | mikasaari: Run zttool to see the various lines |
| 07:12.41 | oej | zttool will show you if you have incoming connections on each channel of the E1 |
| 07:12.47 | opus_ | mikasaari, your D channel is probably line #16 or something misconfigured wrong. read the docs over and over again until you find the problem -- its pretty simple |
| 07:13.07 | drray | mikasarri - are you sure you have a D channel to begin with? |
| 07:13.25 | opus_ | drray -- your diagnoistics approach is flawed. update it:) |
| 07:13.50 | mikasaari | My D channel is #16. If lower bchannels are working nicely, I thought uppers should work too |
| 07:14.34 | opus_ | your upstream provider might try to "correct" the problem which could cause more problems in your testing.. open a ticket with your provider but make sure you understand the documenation 110% before wasting their tim |
| 07:14.35 | opus_ | time |
| 07:14.40 | mikasaari | Also all configurations are copied from working environment where is exactly same kind of E1 line, just bigger Digium card |
| 07:14.55 | kmilitzer | mikasaari: Do you have the problem in both directions, i.e. incoming calls as well as outgoing? |
| 07:14.57 | mikasaari | opus_, I agree with you |
| 07:16.00 | opus_ | <oej> zttool will show you if you have incoming connections on each channel of the E1 |
| 07:16.16 | opus_ | oej is the top expert here, what ever he saids goes! :) |
| 07:16.50 | opus_ | he is more then likely giving you 100% right track to resolution to your problem |
| 07:17.18 | mikasaari | kmilitzer, Hard one. When calling in I can see the span (b channel I think it is) from asterisk debug, but when calling out I think I do not see the span. I think I have to make 30 calls now :) |
| 07:17.47 | opus_ | shit, how did i spend $400 in 3 days. fucking worthless usd |
| 07:18.05 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 07:19.22 | dlynes_home | opus_: in eur? |
| 07:19.38 | opus_ | no im in USA |
| 07:19.47 | oej | opus_: I am not an zaptel expert by any means |
| 07:19.49 | dlynes_home | opus_: no...i meant where you spent $400 in 3 days |
| 07:19.54 | kmilitzer | mikasaari: Well, I am not sure how the channel allocation of E1 is working as I only use SS7, but if it the like, then you will need 30 parallel calls ;) |
| 07:20.32 | oej | kmilitzer: which ss7 driver do you use? |
| 07:20.33 | opus_ | on the west coast, on stupid shit like dinner for clients and dumb shit |
| 07:20.43 | mikasaari | kmilitzer, I fear that is just like that :/ |
| 07:20.44 | kmilitzer | oej: chan_ss7 ... works good |
| 07:20.45 | dlynes_home | opus_: ah |
| 07:20.52 | oej | kmilitzer: The sifra.dk one? |
| 07:20.54 | dlynes_home | opus_: phear Canada *blink* |
| 07:20.56 | kmilitzer | oej: Yes |
| 07:20.58 | mikasaari | kmilitzer, At least it seems so when looking the debug screen |
| 07:21.20 | opus_ | Cananda? it was 85 all day here :) |
| 07:21.24 | oej | kmilitzer: Thanks for letting me know. People ask me during trainings. |
| 07:21.29 | opus_ | Canada sorry |
| 07:21.35 | dlynes_home | opus_: it's been around that here, too |
| 07:21.45 | opus_ | nice |
| 07:21.45 | dlynes_home | opus_: i'm in Vancouver, not Toronto |
| 07:21.57 | kmilitzer | oej: If you/people at your training want more info just contact me |
| 07:22.05 | opus_ | i work with a guy in Vancouver , which is 12 hours away from where I am at |
| 07:22.32 | kmilitzer | mikasaari: Where are you located and what is you telco? Will they be helpful in debugging? |
| 07:23.19 | mikasaari | kmilitzer, I am in Finland. Operator is not willing to do too much at all |
| 07:23.35 | oej | kmilitzer: You work with development of the driver? |
| 07:23.50 | mikasaari | kmilitzer, After 3 weeks of trying to get them to do something, I think I will change the telco |
| 07:24.14 | kmilitzer | mikasaari: I know that for ss7 you can and have to do a channel allocation test before taking a line in use. Maybe there is something equal for an E1 |
| 07:24.45 | kmilitzer | oej: Mainly I am only working with it, but I submited a few small patches to sifira |
| 07:25.18 | mikasaari | kmilitzer, I reset all channels when starting up the asterisk (tested without reseting as well). 1-15 channels work correctly but still those upper ones won't :( |
| 07:26.12 | opus_ | oej, i figured out how to get sip registration to work with exosip |
| 07:26.12 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:26.14 | opus_ | it was pretty easy, you might be interested. |
| 07:26.36 | kmilitzer | mikasaari: Sorry, I was a bit unclear. The channel allocation Test is done from the switch side to make sure all bchannels carry voice ... at least in ss7 as I already said |
| 07:28.09 | mikasaari | kmilitzer, My fault (big noobie I am). From switch side, do you mean the switch before my Digium card ? |
| 07:28.36 | kmilitzer | mikasaari: switch side = side of the telco |
| 07:28.58 | mikasaari | kmilitzer, Nice nice, ok I will ask this as well from them, if I somehow get them to answer to my calls :) |
| 07:29.27 | kmilitzer | mikasaari: Maybe you should first check if something like that is implenetd in E1 definitions |
| 07:29.41 | mikasaari | kmilitzer, I will yes. |
| 07:30.17 | kmilitzer | Willl anyone of you visit Astricon Berlin? |
| 07:31.11 | opus_ | <- cluecon |
| 07:31.15 | opus_ | 2006 :) |
| 07:31.59 | kmilitzer | opus_: Where and what would that be? |
| 07:32.00 | opus_ | i'm going to HOPE, then Cluecon, then DEFCON and home |
| 07:32.21 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 07:32.25 | opus_ | www.cluecon.com |
| 07:32.46 | opus_ | www.defcon.org |
| 07:33.14 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 07:33.19 | opus_ | and |
| 07:33.24 | opus_ | www.hopenumbersix.net |
| 07:33.47 | oej | No, I won't visit any Astricons |
| 07:33.50 | kmilitzer | opus_: cluecon sounds good, but I fear my boss won't like to the idea of a US based conference :( |
| 07:33.59 | kmilitzer | oej: Why? |
| 07:34.14 | opus_ | kmilitzer, too bad because USA is about 5 to 10 years ahead of the rest of the world |
| 07:34.44 | dlynes_home | opus_: so why is it that europe always gets stuff ten years before the US? |
| 07:35.24 | dlynes_home | I'm still waiting to see a diesel vw gti, or a diesel audi a3 |
| 07:35.41 | kmilitzer | Well at least we put more channels on a PRI line and have a better quality of our TV signal ;) |
| 07:35.56 | opus_ | we will be manufacturing the algae that makes your diesel for your audi, you will see |
| 07:36.09 | opus_ | :) |
| 07:36.19 | kmilitzer | dlynes_home: Audi equiped a racing car for Le mans 24 hours race with a Diesel |
| 07:36.25 | dlynes_home | cool |
| 07:36.40 | dlynes_home | I'm a major vw nut :) |
| 07:36.58 | kmilitzer | dlynes_home: VW is far too expensive ... |
| 07:37.04 | dlynes_home | the rabbit's being reintroduced in june |
| 07:37.10 | opus_ | i am looking for a mercedes benz CDI 320 diesel right now myself:) 58mpg |
| 07:37.13 | dlynes_home | and audi isn't? |
| 07:37.22 | dlynes_home | audi way too bloody expensive |
| 07:37.45 | zoa | mercedes is for old people |
| 07:37.46 | dlynes_home | opus_: you mean a benz smart card? |
| 07:37.48 | dlynes_home | opus_: you mean a benz smart car? |
| 07:37.57 | opus_ | no, not the A class (?) |
| 07:38.06 | opus_ | the E 320 CDI diesel :) |
| 07:38.15 | dlynes_home | i have no idea wtf that is |
| 07:38.18 | opus_ | A class is _banned_ in usa |
| 07:38.18 | tparcina | hi group! hi dlynes |
| 07:38.25 | dlynes_home | the only benz cdi i know of is the smart car |
| 07:38.35 | dlynes_home | opus_: why? |
| 07:38.45 | oej | zoa: Oh, have you bought a mercedes now? |
| 07:38.45 | zoa | the A probably doesnt do 3.2 litre :p |
| 07:38.49 | zoa | hehe |
| 07:38.53 | zoa | no no |
| 07:38.58 | zoa | i drive an old bmw now |
| 07:39.03 | zoa | 10 years old |
| 07:39.05 | tparcina | big discusion about asterisk going on here :)) |
| 07:39.12 | zoa | yes |
| 07:39.26 | oej | Porting Asterisk to the car stereo |
| 07:39.28 | zoa | my onboard computer is running asterisk |
| 07:39.35 | zoa | guess somebody stole it |
| 07:39.40 | opus_ | asterisk mpg |
| 07:40.13 | kmilitzer | There are computers for the slot for the car audio ... |
| 07:40.26 | opus_ | dlynes_home, the A calls, more german engineer -- sorry for the spam . bleh http://www.mercedes-benz.com/content/mbcom/international/international_website/en/com/international_home/home/products/passengercars/a-class/aclasscoupe.html |
| 07:40.27 | kmilitzer | ... common x86 hardware ... would work for asterisk ;) |
| 07:40.33 | opus_ | class sorry |
| 07:40.49 | opus_ | hey you guys hear about people hacking into cars via bluetooth??!? |
| 07:41.06 | opus_ | apparently it is really happening, via bluetooth snarfing |
| 07:42.36 | dlynes_home | opus_: why would that make it illegal in the US? |
| 07:44.11 | opus_ | emissions regulation |
| 07:44.30 | opus_ | or something weird, no idea. |
| 07:44.59 | dlynes_home | opus_: ah...thought that was only an issue in California and Washington state? |
| 07:45.39 | opus_ | depends, Detroit has no emission laws what so ever. you can almost drive without a muffler |
| 07:46.02 | opus_ | but to import the car you need to put it through all sorts of regulatatory tests |
| 07:46.16 | opus_ | for example, here we do not have a Nissan Skyline |
| 07:46.44 | opus_ | becuase apparently Nissan decided it wasn't profitable to crash test one for the testing and put it in the market for general purchases |
| 07:47.04 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:47.24 | dlynes_home | nissan skyline's only in Asia, isn't it? |
| 07:48.50 | opus_ | europe |
| 07:49.12 | opus_ | who would want to crash one into a wall:) |
| 07:49.22 | dlynes_home | Crash Test Dummies |
| 07:49.38 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 07:50.04 | *** join/#asterisk DimitrisCreteHer (n=iraklion@olon.ath.forthnet.gr) |
| 07:52.45 | DimitrisCreteHer | hello. Can someone help make capi to work (a@h). I have an ISDN AVM B1 active card. |
| 07:55.43 | *** part/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 08:01.24 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 08:01.31 | *** join/#asterisk motu (n=motu@192.165.166.181) |
| 08:02.47 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 08:03.14 | DimitrisCreteHer | Hello. Can someone help me to make CAPI to work ? |
| 08:03.36 | dlynes_home | Asterisk: The Open Source PBX -=- http://www.asterisk.org/ || FreePBX/AMP/Asterisk@Home Users should join #FreePBX for assitance |
| 08:04.18 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) |
| 08:04.23 | dlynes_home | DimitrisCreteHer: The reason we don't usually help A@H users in here is because the configuration files are all configured kinda weird |
| 08:06.02 | DimitrisCreteHer | I don't care abour a@h configuration files. I just want my centos to have capi support |
| 08:08.07 | shido6 | i retract that statement |
| 08:08.14 | shido6 | the max tnt is kicking my ass |
| 08:10.12 | Sonderblade | is there a way to put the stuff you can write in the [general] section of extensions.conf into extensions.ael? |
| 08:14.10 | *** join/#asterisk motu (n=motu@192.165.166.143) |
| 08:15.39 | autobus | its easy creat deadeAGI for wait 2 seconds? |
| 08:15.40 | autobus | its possible help me! |
| 08:15.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:15.55 | opus_ | sure |
| 08:19.47 | *** join/#asterisk chapeaurouge (n=chapeaur@vilhost1.vision.lu) |
| 08:20.36 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:20.37 | *** join/#asterisk littlejohn (n=little@host16-75.pool8717.interbusiness.it) |
| 08:28.54 | Splat | anyone able to give me a hint on weather or not I want to "Build optimized CAPI driver without CAPI manager?" with my Eicon Diva BRI 2m adapter? |
| 08:31.10 | chapeaurouge | hi all.. how can i find out which codec was used for a particular phone call? |
| 08:31.44 | Shaun2222 | anybody know of a solution for ztdummy needing rtc? in Xen rtc wont load. |
| 08:32.37 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:32.37 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:34.24 | *** join/#asterisk cstomi (n=chatzill@22-36.adsl.etel.hu) |
| 08:36.52 | litage | when writing settings in sip.conf, is it okay to put quotes around each setting's value? eg: useragent = "Asterisk 1" |
| 08:39.42 | opus_ | litagen, no |
| 08:39.52 | opus_ | wait, maybe |
| 08:40.00 | opus_ | try experimenting.. learn gdb |
| 08:40.12 | opus_ | <Snake-Eyes:#ser> have you made the trusted file and put your details in there |
| 08:40.12 | opus_ | <DimitrisCreteHer> hello. Can someone help make capi to work (a@h). I have an ISDN AVM B1 active card. |
| 08:40.12 | opus_ | *** Signoff: psk ("Client exiting") |
| 08:40.12 | opus_ | <rkr245:#ser> Snake_Eyes:just now i enabled mediaproxy and its running fine |
| 08:40.12 | opus_ | <rkr245:#ser> so can i remove those lines from my configuration? |
| 08:40.12 | opus_ | *** tessier_ has left channel #asterisk because ("Leaving") |
| 08:40.14 | opus_ | *** tessier_ has left channel #ser because ("Leaving") |
| 08:40.16 | opus_ | *** L|NUX (n=linux@202.5.145.58) has joined channel #asterisk |
| 08:40.18 | opus_ | *** motu (n=motu@192.165.166.181) has joined channel #asterisk |
| 08:40.20 | opus_ | <Snake-Eyes:#ser> these lines: <rkr245> if (!is_uri_host_local()) if (is_from_local() || allow_trusted()) { |
| 08:44.11 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:45.37 | autobus | its easy creat deadeAGI script for wait 2 seconds? |
| 08:46.14 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 08:47.42 | Sonderblade | have anyone encountered this bug http://lists.digium.com/pipermail/asterisk-dev/2005-July/013754.html or know a workaround for it? |
| 08:50.10 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:50.20 | Faithful | Why is it when my router drops out momentarily the * has to be rebooted? |
| 08:50.41 | *** join/#asterisk cfh (n=luca@host18-109.pool82186.interbusiness.it) |
| 08:50.56 | oej | litage: Wrong channel, read topic and ask for help in #freepbx |
| 08:51.25 | *** part/#asterisk cfh (n=luca@host18-109.pool82186.interbusiness.it) |
| 08:53.49 | xbit` | rxfax, txfax supports fax relay? |
| 08:54.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:58.25 | *** part/#asterisk DimitrisCreteHer (n=iraklion@olon.ath.forthnet.gr) |
| 09:02.44 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.00 | *** join/#asterisk fulgas (n=fulgas@209.8.233.248) |
| 09:03.04 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 09:07.15 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 09:07.35 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:08.52 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 09:10.42 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 09:10.42 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 09:12.40 | sshadow | hi, is anybody using Druid? |
| 09:12.41 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:13.47 | Alex | Uh, if you have one phone registered with one asterisk, and then bring up a different asterisk but using the same user/pass, on a different box.. and register the phone with the new asterisk.. howcome the phone still receives incoming calls form the old one? |
| 09:13.52 | Alex | Is it due to the phone staying registered? |
| 09:14.10 | Zeeek | IAX or SIP? |
| 09:14.20 | Alex | SIP |
| 09:14.34 | Zeeek | there should be an unregister option somewhere |
| 09:14.47 | Zeeek | in phone config |
| 09:14.53 | Alex | Thanks Zeeek. |
| 09:15.03 | Zeeek | I should say the *may* be |
| 09:15.10 | Alex | I suppose I could just reboot the phone |
| 09:15.35 | Zeeek | if it's a sipura that'd be the best way - takes about 8 seconds |
| 09:16.31 | Zeeek | oej that is not swenglish! |
| 09:16.43 | oej | Ahhh |
| 09:16.45 | oej | Me bad |
| 09:16.56 | Zeeek | anyway I'm talking fargo |
| 09:17.04 | Zeeek | yaaaa... ok then... |
| 09:17.41 | Zeeek | the formal name may be scandahoovian, actually |
| 09:18.22 | *** join/#asterisk pbx1 (n=pbx1@58.69.229.213) |
| 09:24.40 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 09:28.25 | *** join/#asterisk sergey (n=sergey@195.151.15.22) |
| 09:29.29 | *** join/#asterisk oelewapperke (n=oelewapp@85.158.215.1) |
| 09:29.41 | oelewapperke | how do I get asterisk to NOT require authentication from a certain peer ? |
| 09:29.49 | oelewapperke | I need 2 things |
| 09:29.58 | oelewapperke | it DOES need to authenticate when calling to that peer |
| 09:30.08 | oelewapperke | but is MUST NOT require authentication back |
| 09:32.24 | RoyK | oelewapperke: insecure=very |
| 09:33.01 | RoyK | oelewapperke: that is, you need a 'user' and a 'peer' with different auth settings |
| 09:33.14 | oelewapperke | and I need the user to authenticate ? |
| 09:33.18 | oelewapperke | or the peer ? |
| 09:35.20 | oelewapperke | RoyK: ? |
| 09:36.29 | autobus | its possible, make one wait after one hungap command? |
| 09:36.35 | autobus | for make more intructions? |
| 09:37.26 | autobus | after no sorry! |
| 09:37.30 | autobus | later |
| 09:37.49 | autobus | for example: |
| 09:37.50 | autobus | exten => 100,1,Set(CALLER=${CALLERID(num)}) |
| 09:37.50 | autobus | exten => 100,2,Hangup() |
| 09:37.50 | autobus | exten => h,1,Wait(10) |
| 09:37.50 | autobus | exten => h,1,System(/var/lib/asterisk/scripts/callback.sh ${CALLERID(num)}) |
| 09:37.53 | RoyK | ~pb |
| 09:37.55 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 09:38.43 | RoyK | oelewapperke: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer |
| 09:40.53 | RoyK | autobus: the second should be exten => h,2,..... |
| 09:41.13 | autobus | yes. i have correct in extensions.conf! |
| 09:41.18 | autobus | sorry. |
| 09:41.25 | autobus | h,1 |
| 09:41.27 | autobus | and h,2 |
| 09:41.38 | autobus | but wait not function |
| 09:42.17 | sergey | Hi How make all number exept 71 as first dialed digits? |
| 09:45.15 | RoyK | autobus: ??? |
| 09:45.18 | RoyK | sergey: wot? |
| 09:45.56 | autobus | RoyK wait command no function. |
| 09:46.15 | autobus | execute the 100,2,Hangup() |
| 09:46.32 | autobus | and no wait 10 seconds for next command. |
| 09:46.38 | RoyK | autobus: pastebin the verbose output |
| 09:46.40 | RoyK | ~pastebin |
| 09:46.41 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com |
| 09:47.16 | RoyK | also, I'd rather put the pause in the callback script |
| 09:47.32 | RoyK | also, use AGI instead of System :) |
| 09:48.18 | autobus | yes! |
| 09:48.26 | autobus | correct! |
| 09:48.34 | sergey | RoyK, sorry my mistake |
| 09:49.31 | autobus | RoyK its possible you give me one example of the script? |
| 09:49.57 | RoyK | see the example scripts in the source |
| 09:54.16 | Zeeek | see RoyK |
| 09:55.11 | Zeeek | jbot sucks rocks |
| 09:58.10 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 09:59.55 | chapeaurouge | ~p2bin |
| 09:59.56 | jbot | from memory, p2bin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
| 10:00.26 | Zeeek | ~b2b |
| 10:04.03 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) |
| 10:05.44 | BugKham | how to see codec used by an active channel? |
| 10:06.04 | BugKham | from CLI, for example |
| 10:07.30 | *** join/#asterisk RaYmAn-Bx (i=rayman@cl-305.ede-01.nl.sixxs.net) |
| 10:08.45 | chapeaurouge | cat your_file | ./paste2pastebin.pl |
| 10:11.24 | Zeeek | BugKham show the channel |
| 10:12.13 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 10:12.21 | tparcina | hi group! |
| 10:18.28 | tparcina | has anybody have this problem - Maximum retries exceeded on transmission; Hanging up call 6cebaaec71cb9b5d670018b026076fdc@192.168.2.10 - no reply to our critical packet. |
| 10:21.01 | Sonderblade | i have one extensions.ael and one extensions.conf is there an easy way in asterisk to switch between the two dial plans? |
| 10:27.03 | mut | ah man |
| 10:27.07 | mut | mornin shift ugh |
| 10:27.22 | mut | i've been playing with a spider crawling on my montior for 15 min now |
| 10:27.33 | mut | chasing it with the mouse cursor |
| 10:28.13 | Zeeek | keeping busy? |
| 10:28.34 | mut | i can't remote into one of the servers, max sessions |
| 10:28.35 | mut | =\ |
| 10:28.40 | RoyK | Sonderblade: both are parsed to asterisk's dialplan |
| 10:28.52 | mut | i dun feel like walking back there and killing off sessions at the console either |
| 10:28.58 | mut | too cold |
| 10:29.14 | Sonderblade | RoyK: so what happens if you have the same contexts and extensions in both files? |
| 10:29.28 | Zeeek | where is it cold in this season? |
| 10:29.47 | RoyK | Sonderblade: probably the same as happens if you have double up in extensions.conf |
| 10:29.48 | mut | in an airconditioned server room |
| 10:29.58 | RoyK | Zeeek: norway |
| 10:29.59 | mut | and outside... |
| 10:30.01 | Zeeek | Sonderblade a reason for doing that would be interesting. What is it? |
| 10:30.20 | Zeeek | RoyK well it is only about 15C here |
| 10:30.21 | RoyK | Zeeek: 12 degrees and light rain |
| 10:30.35 | RoyK | well, not rain now. we've even seen the sun today |
| 10:30.58 | Zeeek | looked like rain today but so far none - join #weather |
| 10:31.16 | Sonderblade | Zeeek: i have a good dialplan in extensions.conf but i want to make a new one using the AEL syntax in extensions.ael |
| 10:31.19 | mut | todays high is supposed to be 15 here |
| 10:31.25 | mut | i doubt it'll reach that |
| 10:32.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:33.03 | puzzled | hi |
| 10:33.12 | Zeeek | hey |
| 10:33.13 | RoyK | hi puzzled |
| 10:35.10 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 10:35.35 | tparcina | i gota this real big problem that hang's up my calls. can sombody take a look and sugest anything? - http://pastebin.ca/57058 |
| 10:37.03 | tparcina | RoyK, i have mentioned to you yesterday that you should mouve to some not so well organized country on south of europ :)) |
| 10:37.19 | tparcina | nad you could be swimming now :) |
| 10:37.26 | tparcina | and you could be swimming now :) |
| 10:40.34 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:45.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:46.00 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 10:50.49 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 10:52.44 | Zeeek | Sonderblade use includes in extensions.conf to add and remove "conflciting" extensions |
| 10:53.48 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 10:54.19 | Sonderblade | Zeeek: is there a command for including extensions.ael in extensions.conf? |
| 10:54.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:55.09 | Zeeek | no, I don't think so. I just meant, put the doubled ones in an include and change that each time. Personally I don't use AEL. I think it will eventually evolve, maybe. Right now I don't need it |
| 10:55.20 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 10:55.21 | Tagor | Hi |
| 10:55.26 | Tagor | I've a very strange problem: |
| 10:55.41 | Tagor | Since yesterday I have problems with calls. If someone calls me or I call him I don't hear anything |
| 10:55.51 | Tagor | Though the IVR and voicemail work fine |
| 10:55.59 | Tagor | Internal echo test also works fine |
| 10:56.08 | Tagor | Any idea how this is caused? |
| 10:56.43 | Zeeek | Tagor what happened yesterday to make the behavior change? |
| 10:56.49 | *** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 10:56.50 | Tagor | Nothing |
| 10:56.57 | Tagor | I haven't changed anything |
| 10:57.29 | Tagor | I suddenly got this problem. The only thing I know is that there was a very short energy interuption |
| 10:57.32 | Zeeek | it's just a machine, it can't be sick or anything like that. What protocol, what provider, what phone, what network setup, what NAT etc etc |
| 10:57.35 | Tagor | So the server had to be restarted |
| 10:57.55 | Tagor | But I didn't change anything in the configs |
| 10:58.40 | Tagor | SIP, 12connect, Grandstream GXP 2000 also tried X-lite, phone connects to asterisk server and asterisk server connects to provider, DMZ to the server |
| 10:59.14 | Tagor | Asterisk also records the call. I noticed that all files are 44 bytes |
| 10:59.17 | Zeeek | Tagor you'll have to sip debug and see what happens |
| 11:00.02 | Tagor | The normal procedure, it says ringing then it says ok and then when I hangup it says 'normal hangup' |
| 11:00.20 | Sonderblade | Zeeek: well AFAIK, you can't mix "regular" asterisk dialplan syntax with AEL syntax |
| 11:00.29 | *** join/#asterisk chendy (n=Daiyan_C@222.67.29.25) |
| 11:01.11 | Zeeek | Sonderblade no you can't. I meant remove the conflicting ones, comment them out in the include. Nevermind, it's too hard to explain. Bottom line, it doesn't work the way you want |
| 11:01.43 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 11:05.40 | Zeeek | Tagor can you try another provider? |
| 11:05.51 | Zeeek | try FWD or sipgate free |
| 11:06.23 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 11:07.25 | Zeeek | since the echo test works, it seems to be on the incoming/outgoing end, not within you * setup |
| 11:15.00 | BugKham | Zeek, which of the show channel output indicates codec used? |
| 11:15.17 | BugKham | NativeFormat? |
| 11:16.02 | BugKham | Zeek, looks like it's different btw Zap and SIP chans also |
| 11:16.18 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 11:16.45 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 11:16.58 | Zeeek | do a sip show channels |
| 11:18.07 | Zeeek | or ZAP show channel 5 |
| 11:20.01 | BugKham | hmm, it says nothing about codecs |
| 11:20.02 | *** join/#asterisk chapeaurouge (n=chapeaur@vilhost1.vision.lu) |
| 11:20.20 | Zeeek | maybe you got a bad asterisk |
| 11:20.28 | BugKham | oh, i see it now |
| 11:20.37 | BugKham | Default law:? |
| 11:21.56 | *** join/#asterisk Tagorr (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 11:22.03 | Tagorr | Not sure whether you get this message: |
| 11:22.06 | Tagorr | Zeeek >> I found the problem which I had before. For some reason it stops transmitting sound after a specific date. I had this in February too. I just set date to 2001 and everything works fine |
| 11:22.08 | Tagorr | Anyone an idea what this stupid asterisk millenium bug is caused by? |
| 11:24.20 | Delvar | yeah in the mailing lists there was an unsign/signd integer mixup... should be fixed in 2.5.7 and trunk |
| 11:24.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:24.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:30.20 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 11:30.21 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 11:32.05 | Splat | anyone here using Eicon Diva Server cards and have a 100 number indial range? |
| 11:32.53 | Szolke | hi all. We have a configuration problem. We would like to register our client (PC) to our server (PC) with SIP. the client send the register request but it cant get back nothing. |
| 11:33.33 | *** join/#asterisk coppice (n=chatzill@199.203.17.210.dyn.pacific.net.hk) |
| 11:34.32 | Szolke | the client get this message: Jan 2 00:04:00 WARNING[186]: chan_sip.c:9760 handle_response_register: Got 404 Not found on SIP register to service sipteszt1@217.xxx.32.207, giving up |
| 11:35.59 | znoG | there's no real hunt groups in Asterisk, right? they just need to be configured in the dial plan as needed |
| 11:36.34 | coppice | why does that make them less than real? |
| 11:37.13 | znoG | it doesn't, really, in most PBXs you can say .. "add hunt group <ext>" and define the list of extensions |
| 11:37.27 | znoG | where in *, you have to set it up in the dial plan, but it works the same. |
| 11:38.36 | Malthus | looks like a hunt group, acts like a hunt group, hunts like a hunt group :) |
| 11:39.00 | znoG | oh wait.. it IS a hunt group! |
| 11:39.55 | Malthus | suppose someone wrote an * frontend that acted just like that, and you never knew about extensions.conf? |
| 11:40.03 | Malthus | what mighty matters we ponder :P |
| 11:45.04 | *** join/#asterisk miguel3239 (n=chatzill@h-68-167-124-171.cmbrmaor.covad.net) |
| 11:53.13 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:54.49 | xheliox | Stupid question: When someone dials a phone number out of the Asterisk box, and then they want to enter dtmf tones for menu selections --- Asterisk intercepts those, so they're not sent properly to the remote end... how can I isolate the dtfm so it's sent to the remote end, and not Asterisk after a call has been placed? |
| 11:55.02 | *** join/#asterisk chapeaurouge (n=chapeaur@vilhost1.vision.lu) |
| 11:56.54 | coppice | why is it called a hunting group? shouldn't it be a hunting pack? |
| 11:58.56 | Sonderblade | is there a program that can generate asterisks conf files for you? |
| 11:59.26 | RoyK | yes, vim |
| 11:59.55 | coppice | will only vim work? what about ajax? |
| 12:02.50 | chapeaurouge | anyone know how i can tell which codec a phone is using/has used? |
| 12:03.17 | RoyK | sip show channels ? |
| 12:03.51 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:05.16 | chapeaurouge | RoyK, thx. |
| 12:07.33 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 12:07.57 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 12:09.22 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:13.08 | eipi | why i can't compile asterisk-addons? I'm receiving this error: from format_mp3.c:20: /usr/include/asterisk/strings.h:264: error: syntax error before "__extension__"? |
| 12:20.03 | *** join/#asterisk aze (n=aze@ACayenne-101-1-6-66.w81-248.abo.wanadoo.fr) |
| 12:21.14 | *** join/#asterisk Tili (n=Tili@cm33.gamma249.maxonline.com.sg) |
| 12:21.24 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:23.18 | *** join/#asterisk kevinfcn (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 12:29.42 | *** join/#asterisk littlejohn (n=little@host16-75.pool8717.interbusiness.it) |
| 12:30.59 | Szolke | Can you help me to configure sip peers? |
| 12:36.09 | *** join/#asterisk jpbotelho (n=jpbotelh@201.7.108.130) |
| 12:40.53 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 12:45.44 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:45.53 | *** join/#asterisk JimmyCarter (n=flod@213083175015.sonofon.dk) |
| 12:49.15 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 12:49.33 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 12:54.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:54.44 | autobus | when i user hangup script, i think the call hungap now right? |
| 12:54.55 | autobus | i have another instruction in next lines |
| 12:55.36 | autobus | but the call only hangup when conclude the all introctions |
| 12:55.41 | autobus | why! |
| 12:55.42 | autobus | ? |
| 12:56.27 | *** join/#asterisk esculapio__ (i=elvyn@200.88.44.66) |
| 13:00.36 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 13:01.16 | Dovid | Hi all |
| 13:01.33 | Dovid | Can anyone help me with System() and ssh ? |
| 13:03.06 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 13:03.07 | Dovid | ? |
| 13:05.33 | Dovid | Anyone awake here ? |
| 13:05.40 | *** join/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 13:05.44 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:08.36 | RoyK | <PROTECTED> |
| 13:08.49 | RoyK | Dovid: use agi instead. far better |
| 13:08.49 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:09.10 | Dovid | I am bad with agi. I dont know scripting well |
| 13:09.50 | Dovid | Do u know where I can find an example that has ssh |
| 13:10.09 | Dovid | I am tryin to write a script that will ssh into a diffrent server and reboot it etc. |
| 13:11.05 | *** join/#asterisk keyhack (n=keyhack@68.236.93.225) |
| 13:11.11 | RoyK | System(ssh someserver /sbin/reboot) |
| 13:11.47 | Dovid | Where do I pur in the root and pass ? |
| 13:11.57 | shiznatix | in won |
| 13:12.08 | shiznatix | i* |
| 13:12.27 | Dovid | System(ssh root:pass@someserver.net /sbin/reboot) ? |
| 13:13.10 | RoyK | docelm0: no... you need to use signed keys - you can't pass the password like that. also, i'd say better use another user and add sudo access |
| 13:13.45 | RoyK | ssh-keygen -t dsa and copy the .ssh/id_dsa.pub file to the destination server's .ssh/authorized_keys |
| 13:13.48 | RoyK | rtfm :) |
| 13:14.03 | Dovid | lol |
| 13:14.12 | Dovid | I am still learnin linuz |
| 13:14.15 | Dovid | Linux* |
| 13:14.29 | JimmyCarter | Is it possible to make attended transfer via the Manager API? |
| 13:14.58 | RoyK | Dovid: http://defindit.com/readme_files/ssh.html |
| 13:15.06 | *** join/#asterisk coppice (n=chatzill@66.166.17.210.dyn.pacific.net.hk) |
| 13:15.12 | Dovid | thanx |
| 13:16.26 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 13:19.39 | cytrak | can someone please help me out on this ... I have tried so many things to improve my sound quality and still is crapy as hell ->http://pastebin.com/726477 |
| 13:19.52 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 13:19.59 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) |
| 13:20.18 | cytrak | I'm talking about when I call from my iax idefisk phone to a cell or a landline phone |
| 13:20.45 | Flauto | is there a way to set expiry for individul register? |
| 13:21.41 | *** join/#asterisk enerv (n=enerv@200.233.70.28) |
| 13:21.41 | cytrak | my voice volume sounds pretty low and when my iax client doesn't speak I hear scrachy noises |
| 13:21.42 | *** join/#asterisk mitka (n=mitka@62.76.244.194) |
| 13:21.55 | mitka | hi |
| 13:22.02 | Flauto | hi |
| 13:22.14 | mitka | whats up |
| 13:22.23 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.182.Dial1.SanJose1.Level3.net) |
| 13:22.30 | mitka | flauto how large is ur asterisk |
| 13:22.32 | mitka | set p |
| 13:22.49 | Flauto | it is not large |
| 13:23.00 | Katty | the only type you should type 'ur' is when you're referencing the Land of Ur. |
| 13:23.05 | Flauto | but i do have multiple sip registers |
| 13:23.43 | mitka | hmmm..multple sip registers... |
| 13:23.51 | mitka | whta is that exactly |
| 13:24.02 | mut | sux to the land of ur |
| 13:24.05 | Flauto | one of them needs expiry=3600 |
| 13:24.19 | Flauto | but one of them would not work with expiry=3600 |
| 13:25.58 | mitka | any idea what type of processor should i get to asterisk to support 200 extensions |
| 13:26.15 | Flauto | hmm.... |
| 13:26.25 | Flauto | i really don't know |
| 13:26.28 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 13:26.31 | tzanger | mitka: start out with a regular old P4 as fast as you can sanely get, with a half gig of memory and regular IDE disk |
| 13:26.42 | tzanger | see how it goes, and do your own testing. "200 extensions" says nothing |
| 13:26.44 | Flauto | the most extensions that i have are less than 10 |
| 13:27.09 | tzanger | but a regular old P4 system is a GREAT way to start. it's not a wasted investment and it's relatively cheap |
| 13:27.17 | kay2 | what is needed to leave a video voicemail ? |
| 13:27.17 | cytrak | any ideas on how to improve sound quality ? |
| 13:27.22 | tzanger | or AMD. don't bugger around with Xeons or x86-64 until you can see you need it |
| 13:27.59 | [TK]D-Fender | kay2 : Whatever time it takes YOU to code it. |
| 13:28.47 | coppice | what's a video voicemail? sign language? :-\ |
| 13:28.49 | mitka | ok...thanks for the info |
| 13:29.16 | [TK]D-Fender | coppice : Beware of BRAILLE! |
| 13:29.46 | [TK]D-Fender | TTT? Text-To-Touch? :) |
| 13:29.48 | mitka | what type of codec are you using current |
| 13:29.51 | Flauto | hi coppice, tkd good morning |
| 13:30.08 | mitka | which codec....do u all recommend |
| 13:30.19 | coppice | FLAC |
| 13:31.08 | tzanger | [TK]D-Fender: text-to-text :-) |
| 13:31.19 | bkw_ | isn't that what SMS is? |
| 13:31.39 | coppice | SMS is fiddly thumbs to text |
| 13:31.51 | tzanger | coppice: heh |
| 13:31.57 | ManxPower | cytrak, What is the nature of the bad audio? |
| 13:32.43 | mitka | anyone know whats the best codec |
| 13:32.44 | coppice | Flauto: hi |
| 13:32.51 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 13:32.56 | coppice | mitka: FLAC |
| 13:32.57 | ManxPower | cytrak, also put your /etc/zaptel.conf on pastebin |
| 13:33.26 | ManxPower | cytrak, volume is controled by rxgain and txgain |
| 13:33.32 | jaybuffet | i have a question, before i go replacing all these phones... is there a way to use our existing samsung prostar dcs system with asterisk ? |
| 13:33.42 | tzanger | jaybuffet: ahh, integration |
| 13:33.55 | Flauto | coppice, is there a way to set one of the registers to a certain expiry? |
| 13:33.56 | tzanger | jaybuffet: depends on what level you want. I am half-assed integrating a Norstar MICS with Asterisk |
| 13:34.11 | jaybuffet | tzanger: how so? |
| 13:34.35 | tzanger | jaybuffet: Asterisk - PRI - MICS |
| 13:34.52 | tzanger | the MICS can have the PRI as TIE trunks or just external trunks (I'm using the latter) |
| 13:34.53 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:35.27 | jaybuffet | tzanger: do you lose some functionality |
| 13:35.36 | ManxPower | tzanger, how much would the software upgrade be to use the PRU as TIE trunks? |
| 13:35.38 | tzanger | jaybuffet: no, but you don't get all the features of * in the MICS |
| 13:35.56 | ManxPower | s/PRU/PRI |
| 13:36.09 | tzanger | ManxPower: well I can use it as TIE right now, but I lose the ability to route outgoing calls through the PRI then (TIE = remote "internal" network IIRC) |
| 13:36.24 | ManxPower | tzanger, Ah. |
| 13:36.35 | tzanger | I have an MCDN license which will consolidate the PRI for external and remote internal calls, but it runs a proprietary signalling protocol called SL1 |
| 13:37.15 | tzanger | if I had another MICS with that license I could theoretically reverse-engineer it by putting a 2-port Digium card in the middle and DACSing the channels and just monitoring the traffic, but time is something I'm very short on |
| 13:37.57 | ManxPower | tzanger, if you DACS Asterisk can't touch the DACS'd channels |
| 13:38.14 | tzanger | ManxPower: no, but the zaptel driver can |
| 13:38.25 | tzanger | the DACSing is done just to monitor, not mangle |
| 13:38.26 | ManxPower | tzanger, if you had another PRI port on the MICS..... |
| 13:38.41 | ManxPower | tzanger, monitor in what way? |
| 13:38.43 | tzanger | once I have enough of the protocol figured out I can remove the DACS and see if I can get * to play nice and pretend it's a mICS |
| 13:38.51 | tzanger | ManxPower: yeah, but DTIs are expensive as hell |
| 13:38.53 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 13:39.05 | ManxPower | ah, to write a SL1 channel driver? |
| 13:39.14 | tzanger | ManxPower: no, just add sl1 to libpri |
| 13:39.14 | ManxPower | tzanger, you can find them cheaper on eBay |
| 13:39.20 | tzanger | ManxPower: true enough |
| 13:39.49 | ManxPower | I would have done that, but I'm sure our Nortel is not licensed for PRI or T-1 |
| 13:40.38 | Szolke | Can you help me to register two asterisk with SIP? |
| 13:41.31 | tzanger | oh yeah you need the license for the DTI too |
| 13:41.33 | tzanger | *rolls eyes* |
| 13:42.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:42.42 | *** join/#asterisk IMG-SD (n=IMG-SD@ismex.imperialgroup.ca) |
| 13:43.56 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 13:45.20 | jaybuffet | are most of you guys in the telephony industry, or where do you learn about all this stuff ? |
| 13:46.24 | tzanger | jaybuffet: a lot of us have had some kind of telephony experience, but I think most of us are just tinkerers |
| 13:46.27 | Flauto | i think some of them are pros |
| 13:46.44 | tzanger | I designed and helped run a 30k-user dialup ISP for a few years |
| 13:46.50 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.182.Dial1.SanJose1.Level3.net) |
| 13:47.02 | tzanger | and I design industrial hardware and software (not at all related to telephony) fora living |
| 13:47.19 | mitka | must be expert.... |
| 13:48.40 | tzanger | no I'm no expert, I just have some experience and I'm tenatious as hell |
| 13:49.37 | jaybuffet | yeah.. i'm getting a little overwhelmed here... sure replacing the existign phon system with an asterisk sol would probably be easy enough, but in trying to save money, maybe there is a way to use existing equipment, then you mention integration, and i'm like.. uhhh.. ok |
| 13:49.53 | tzanger | jaybuffet: the easiest thing to do is trash the existing system |
| 13:50.17 | jaybuffet | tzanger: but generally not the most cost effective |
| 13:50.18 | tzanger | however most business spend around $15-20k on their phones and to get rid of them without knowing they'll like the new one gives anyone the screaming heebie-jeebies |
| 13:50.31 | jaybuffet | tzanger: exactly |
| 13:50.32 | tzanger | jaybuffet: so generally you start out small |
| 13:50.35 | jake1932 | jaybuffet: is it a digital PBX system, or IP based? |
| 13:50.53 | jaybuffet | jake1932: digital PBX |
| 13:50.56 | tzanger | I started out by adding 4 analog trunk lines to our MICS and routing them into a TDM400 with 4 FXS ports. |
| 13:50.59 | ManxPower | I used to manage techops for a small ISP that did PRI ISDN, Dialup, and DSL. After than I did tech consulting on LAN/WAN for about 10 years. |
| 13:51.16 | tzanger | then I programmed the MICS that any long distance call was to use Trunk B lines, and any local call was to use Trunk A lines |
| 13:51.32 | brif8 | I have DNIS enabled. exten => mynumber,1,Macro(Inbound) then in [macro-inbound] I have exten => s,1,Answer ... How can I in [macro-inbound] see mynumber, (which number was called) and how I got to [macro-inbound] ? |
| 13:51.34 | tzanger | (trunk A = the 12 POTS lines from the telco, trunk B = the 4 POTS "lines" from Asterisk) |
| 13:51.34 | jake1932 | jaybuffet: yeah - could be real expensive to try to make use of existing phones |
| 13:51.38 | *** join/#asterisk amorith (n=nahirean@unaffiliated/nahirean) |
| 13:51.55 | tzanger | that was very simple integration and let me experiment with VOIP |
| 13:52.01 | ManxPower | brif8, you didn't read README.variables did you? |
| 13:52.06 | mitka | whats MICS?? |
| 13:52.37 | jaybuffet | seems to be nortel proprietary stuff |
| 13:52.41 | tzanger | no voicemail or anything |
| 13:52.57 | tzanger | but it let me experiment |
| 13:53.17 | tzanger | and I could get a DID and ring one of those 4 lines and the receptionist would pick up since as far as the MICS was concerned, it was just another line ringing |
| 13:53.48 | tzanger | then I got a channel bank and pulled all the POTS lines from the telco into * and ran all the FXS ports into the MICS |
| 13:54.04 | tzanger | so every single call went through asterisk, even if it would just bounce through the CB and go out to the telco again |
| 13:54.17 | tzanger | when we moved, we got a PRI and plugged it into asterisk |
| 13:54.27 | tzanger | and then got a PRI module for the MICS and plugged it into asterisk |
| 13:54.41 | jaybuffet | tzanger: seems simple enough :-/ |
| 13:54.42 | tzanger | (same as the channel bank solution, but cheaper and more powerful) |
| 13:54.59 | tzanger | the MICS can assign DIDs to specific extensions |
| 13:55.07 | tzanger | so I assigned our main # to the reception phone as usual |
| 13:55.21 | *** join/#asterisk Siarom (n=gurgel@sec16.secrel.com.br) |
| 13:55.31 | tzanger | but then created "fake" DIDs of 0000XXX where XXX = the extension # -- so I can ring any extension directly from Asterisk by dialing 0000+extension |
| 13:55.45 | *** part/#asterisk Siarom (n=gurgel@sec16.secrel.com.br) |
| 13:55.46 | tzanger | it's cheap integration... the MICS still thinks it's an extenral call so it's not true integration |
| 13:56.09 | tzanger | but it'll let me add a few polycom phones to Asterisk and mostly have them ork and feel like internal extensions |
| 13:56.27 | tzanger | i.e. the MICS has a dialplan that 8XX dials out the PRI. Asterisk sees a call to 8XX and calls one of hte polycoms |
| 13:56.48 | tzanger | and the polycoms dial 2XX to reach our norstar extensions... asterisk sees a 2xx and dials the PRI 00002xx |
| 13:57.06 | tzanger | still can't do voicemail or call parking or anything, but it's getting closer |
| 13:57.20 | tzanger | norstar really ties your hands behind your back because they do NOT want you doing this |
| 13:57.40 | ManxPower | tzanger, do you ever do consulting on nortel stuff? |
| 13:57.56 | tzanger | ManxPower: a little |
| 13:58.22 | ManxPower | because your MICS stuff is pretty much what we want to do, but don't know of any decent local nortel people that have any understanding of what we want. |
| 13:58.36 | coppice | merdian option 11..... er, sorry, meridian catch 22 :-) |
| 13:58.45 | ManxPower | the last consultant quoted us 15 hours to just route 30xx calls out the CO ports. |
| 13:58.50 | *** join/#asterisk Siarom (n=gurgel@sec16.secrel.com.br) |
| 13:58.54 | tzanger | my current norstar projects include screwing with an MCK telebridge I got from an anonymous source and hacking the TE405 to do optical interconnects so I can truly replace the KSU |
| 13:58.58 | tzanger | 15 HOURS? |
| 13:59.09 | tzanger | with a FastRAD it's an hours worth of work for 100 extensions |
| 13:59.11 | tzanger | (at least on PRI) |
| 13:59.34 | ManxPower | tzanger, yeah, he was going to manually program every 31xx combo |
| 13:59.51 | ManxPower | ..er... 30xx |
| 13:59.51 | lunk | even a bash loop would be faster than that |
| 14:00.17 | tzanger | still |
| 14:00.44 | tzanger | on a M7234 (I think that's the model of the phone, 24 soft buttons, 2 line display) it would take no more than 2 or 3 hours |
| 14:00.50 | ManxPower | Actually we wanted 2xxx - 7xxx to go out the CO port, 1xx to stay the way it is (1xx are current nortel extensions) |
| 14:00.51 | tzanger | it's only a hundred DNs |
| 14:01.12 | ManxPower | tzanger, no wildcards? |
| 14:01.20 | tzanger | actually you don't even do that, you use a wildcard (create a dial rule) |
| 14:01.23 | tzanger | yeah |
| 14:01.35 | tzanger | you create a call routing table if you don't already have one |
| 14:02.13 | ManxPower | tzanger, can you do it via the modem/serial port on the system? |
| 14:02.35 | tzanger | I have 9XXXXXXX go out PRI-A, 911/9911 go out PRI-A, 91XX... and 90XX... to go out the same interface... it should NOT be difficult to set up |
| 14:02.41 | tzanger | ManxPower: you need a RAD/FastRad |
| 14:02.57 | Modcuts | Any reason why if you log in via ssh and don't allow root login and switch user you can't use asterisk console? |
| 14:02.57 | ManxPower | tzanger, what is that? |
| 14:02.57 | tzanger | the serial port on the system is retarded IIRC |
| 14:03.03 | tzanger | ManxPower: little gray box like an ATA but with a serial port instead of an FXS port |
| 14:03.31 | ManxPower | tzanger, well I think there is a modem on the system currently. |
| 14:03.31 | ManxPower | don't know much else about it. |
| 14:04.29 | mitka | is it possible to use ip phones but connect it with different wire than lan cable |
| 14:04.42 | *** part/#asterisk Siarom (n=gurgel@sec16.secrel.com.br) |
| 14:04.45 | mitka | because if i use analog phones the wire are much thinner |
| 14:05.02 | ManxPower | mitka, no. If you did that then it would not be an IP phone. |
| 14:05.05 | tzanger | mitka: not really, just 802.11. :-) what kind of connection do you want? ThickNet? |
| 14:05.22 | tzanger | or token ring |
| 14:05.34 | ManxPower | tzanger, pervert |
| 14:05.57 | tzanger | ManxPower: :-) |
| 14:06.23 | mitka | is there alternative for lan cable... |
| 14:06.35 | tzanger | mitka: such as? |
| 14:07.05 | mitka | because....in my building wiring with lan cable is difficult as 100 extension are need |
| 14:07.07 | mitka | to set uyp |
| 14:07.08 | coppice | hey, there's product differentiation for you. a VoIP phone with BNC and token ring connectors instead of RJ45 |
| 14:07.09 | mitka | set |
| 14:07.19 | coppice | you could throw in an AUI too |
| 14:07.32 | zoa | haha :) |
| 14:07.34 | mitka | do u have the website |
| 14:09.40 | tzanger | mitka: well then no |
| 14:09.53 | tzanger | mitka: use regular PSTN phones and channel banks... (eww) |
| 14:10.13 | tzanger | or use something like a citel gateway for the existing phones, but that's kind of stinky too |
| 14:10.17 | mitka | yeah...guess thats the only choice...but sip phones...have many other features |
| 14:10.28 | mitka | like text message...caller id |
| 14:11.07 | tzanger | yep |
| 14:11.18 | tzanger | if you can't rewire the building then you're going ot have to look at alternatives |
| 14:11.29 | tzanger | do you have cat5 going ot ever desk already? |
| 14:11.44 | tzanger | if so, you can "cheat" as most phones have built in switches... you plug the phone into the LAN, and the computer into the phone |
| 14:11.51 | tzanger | I never really liked that though, but that's just me. I'm a bit of a purist |
| 14:12.44 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:13.07 | mercestes | yea, rebooting the phone kicks the comp off the Internet but other than that...works pretty ok. |
| 14:13.31 | mercestes | you can also buy littlke linksys 4 port switches and dump them all over the building |
| 14:13.53 | mercestes | so wherever you have 1 wire you suddenly have 3. |
| 14:13.54 | mitka | ok...good ideas |
| 14:13.55 | Quension | cat5 can serve multiple purposes if it's just an issue of how much to run |
| 14:14.08 | mitka | thx |
| 14:14.12 | mercestes | NP |
| 14:14.26 | Quension | you can take lines that aren't used for ethernet and run phone over them |
| 14:14.36 | Quension | up to 4 pairs |
| 14:15.05 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 14:15.16 | mitka | i am not going to ptovide internet or internet the only purpose is for calling |
| 14:15.17 | Quension | sometimes you can get away with two 100basetx runs on a single cat5 as well |
| 14:15.20 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 14:15.29 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:15.33 | Quension | I meant classic analog phone with the 4 pair thing, that wasn't clear |
| 14:15.52 | mitka | is there something simpler such to support only calling... |
| 14:15.53 | mitka | ok |
| 14:16.03 | mitka | but classic phones |
| 14:16.13 | mitka | doesnt support advance features |
| 14:16.18 | mitka | like |
| 14:16.21 | Quension | yeah, it doesn't directly answer your question |
| 14:16.39 | Quension | I was just mentioning it in case it proved useful |
| 14:16.46 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:16.52 | tzanger | Qwell: you mean 2 pairs |
| 14:16.55 | tzanger | but I don't recommend that |
| 14:17.39 | mitka | the only choice i have now to have analog phones with 2 pair wire |
| 14:17.40 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 14:17.49 | mitka | being lookin for alternative... |
| 14:17.49 | Quension | yeah, it's a hack and can easily have issues with long runs |
| 14:18.07 | Quension | besides confusing the hell out of anyone who has to maintain it later ;) |
| 14:18.29 | mitka | is it possible to connect ip phones with nornal analog phone line |
| 14:18.52 | mitka | the nornal copper wire |
| 14:19.18 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 14:19.39 | Quension | no, there's nothing pratical that would work for that, that I know of |
| 14:22.54 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:23.29 | mitka | thz |
| 14:23.31 | mitka | thx |
| 14:23.40 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:24.06 | tzanger | mitka: no not really. you can screw with networking over phonelines type of equipment but you will just end up disappointed |
| 14:25.09 | *** join/#asterisk John-Z (n=lotek@phrank.aus.us.siteprotect.com) |
| 14:25.24 | John-Z | Wow, this channel is popular! |
| 14:25.25 | mitka | how do i interconnect two asterisk system in 2 separtate building |
| 14:25.27 | zoa | yes |
| 14:25.30 | zoa | its all because of me |
| 14:25.35 | zoa | they are all my groupies |
| 14:25.38 | John-Z | Really.. wow. |
| 14:25.38 | zoa | especially tzanger |
| 14:25.40 | John-Z | ;) |
| 14:25.51 | mitka | i heard could be done with iax2 trunking |
| 14:25.54 | zoa | watch out for my special charms |
| 14:25.58 | John-Z | I'm in the process of setting up my first Asterisk system.. thought I would lurk a moment. |
| 14:26.01 | zoa | mitka: yes can be done |
| 14:26.08 | Zeeek | zoa you going to astricon? What city? |
| 14:26.16 | zoa | John-Z: asterisk is very popular, hence the amount of people here |
| 14:26.18 | mitka | but will iax2 work without internet |
| 14:26.18 | tzanger | no, I'm oej's groupie, sorry |
| 14:26.20 | unixgeek | mitka: yes, you will want to use iax2 to connect between asterisk boxes. |
| 14:26.24 | zoa | zeeek, i dont know yet, i would go for the people |
| 14:26.29 | zoa | but if its in several cities |
| 14:26.31 | tzanger | mitka: it's easy |
| 14:26.33 | zoa | they will all be very small |
| 14:26.36 | tzanger | I've done it dozens of times |
| 14:26.40 | *** join/#asterisk aze (n=aze@ACayenne-101-1-15-24.w80-8.abo.wanadoo.fr) |
| 14:26.46 | zoa | so i will loose a fortune and see only few people if i go |
| 14:26.48 | Zeeek | zoa certainly, but then you can mingle better :) |
| 14:26.50 | tzanger | you create a sip or iax2 connection between them and just route the calls |
| 14:26.53 | zoa | but i will try to arrange something |
| 14:26.58 | zoa | dunno yet |
| 14:26.59 | Zeeek | come to Paris |
| 14:27.01 | zoa | will need to see |
| 14:27.07 | zoa | paris or berlin seem the cheapest to me |
| 14:27.10 | Zeeek | you're not far are you? |
| 14:27.11 | zoa | as i can easily drive there |
| 14:27.13 | zoa | no |
| 14:27.16 | mitka | its within the same city |
| 14:27.18 | zoa | paris = 3 hours drive |
| 14:27.21 | mitka | sama rea |
| 14:27.22 | brif8 | ManxPower: ok I found the ${DNID} variable. If I'm using this as part of the file name to monitor/record a call how can I find what extension actually answers the call, seeing multiple extensions ring when the number is dialed ? |
| 14:27.23 | mitka | area |
| 14:27.26 | Zeeek | drive? That means burning gas! polluting the planet! |
| 14:27.32 | russellb | zoa: hey! |
| 14:27.34 | mr_horsepower | paris = 17h drive! :D |
| 14:27.47 | Zeeek | or 1244 hours from L.A. |
| 14:27.52 | mr_horsepower | :D |
| 14:27.52 | John-Z | Paris is a 10 hour plane trip from here.. :) |
| 14:27.55 | John-Z | Dont feel so bad. |
| 14:28.13 | Zeeek | paris is a one minute trip down two floors |
| 14:28.20 | John-Z | hah. |
| 14:28.33 | russellb | zoa: I hope to start focusing on the new jitterbuffer next week, with the plan to get it merged by the end of the month |
| 14:28.34 | Zeeek | quick someone give me the best RSS URI you have on anything |
| 14:28.53 | *** join/#asterisk rva (n=rafa@200.210.51.130) |
| 14:29.21 | *** join/#asterisk abatista (n=Ariel@70.46.87.154) |
| 14:29.49 | *** join/#asterisk iulius (n=iulius@adsl-145-179-107.asm.bellsouth.net) |
| 14:30.07 | zoa | russellb, cool |
| 14:30.16 | zoa | please let me know if you find such timestamp problems |
| 14:30.29 | zoa | we use it in production without issues |
| 14:30.33 | zoa | but roy has issues with it |
| 14:31.53 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 14:32.37 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:32.50 | russellb | will do ... RoyK seems to quite often have problems that nobody else ever sees :) |
| 14:32.54 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:32.55 | vooduhal | How would one call "agent logoff Agent/1234" via agi? |
| 14:33.04 | vooduhal | Exec doesn't see agent as an application. |
| 14:33.40 | jaybuffet | could someone that knows something tell me what this does (http://www.samsung.com/Products/OfficeNetwork/DigitalKeySystem/OfficeNetwork_DigitalKeySystem_iDCS500.asp) I talked to the guy that installed our existing phone system, and he said we would need that to use our existing phone system for voip... would this tie into asterisk in someway or would this be in place of asterisk.. again.. i'm confused |
| 14:34.16 | tzanger | jaybuffet: typically speaking (I have not looked at the link) any vendor's solution to "bring VOIP to the system" is proprietary and hard to use with asterisk. |
| 14:34.29 | C4T3l | true |
| 14:34.32 | jaybuffet | tzanger: thats what i am affraid of |
| 14:35.16 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 14:35.39 | tzanger | that link is for an entire PBX, not just a card to enable VOIP |
| 14:36.19 | triple-e | vmail.cgi is broken |
| 14:36.38 | jaybuffet | tzanger: ok so it replaces asterisk then... |
| 14:36.58 | C4T3l | jbot are you there? who is mercestes? |
| 14:37.01 | triple-e | it shows the vm file as being broken or unavailable |
| 14:37.24 | triple-e | when i go into the vm direcotory it shows the file as not being readable |
| 14:37.44 | triple-e | and owned by root |
| 14:37.55 | triple-e | how did i hose that up, :-) |
| 14:38.22 | tzanger | jaybuffet: it's an entire PBX |
| 14:38.25 | mercestes | chown asterisk vvoicemail file. |
| 14:38.29 | tzanger | i.e. it replaces the system you have now |
| 14:39.01 | triple-e | every time i get a voicemail its created with the same permisssions level |
| 14:39.15 | [TK]D-Fender | triple-e : So any good news following the end of that teleconference yesterday? SIP 2.0, IP 430, /other dates? |
| 14:39.38 | triple-e | just a mid june release on SIP2.0 |
| 14:39.49 | Alex | Anyone know of a SIP client for Symbian 9? |
| 14:40.13 | MikeJ[Laptop] | Alex, the sipx client may work on Symbian. |
| 14:40.14 | *** join/#asterisk XanaXa (n=m@ppp-69-219-158-119.dsl.chcgil.ameritech.net) |
| 14:40.17 | [TK]D-Fender | triple-e : Thats excellent news. Did you notice anything substantial about it? |
| 14:40.41 | [TK]D-Fender | triple-e : Like radical redeisn of LCD, etc? |
| 14:40.45 | Alex | Thanks, MikeJ[Laptop] |
| 14:40.45 | triple-e | i'll pull it once its available and test it in our lab -- the security seems interesting but something my clients aren't gonna pay to impliment |
| 14:40.49 | file | ~centosbug |
| 14:40.50 | jbot | hmm... centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 14:41.16 | triple-e | no backlit LCD -- i asked three times if there was a schedule, so did other people so they are getting alot of push back on the back lit LCD |
| 14:41.44 | XanaXa | hey guys I have a quick question, how do I allow Asterisk to register devices on multiple subnets? My asterisk server is on the 192.168.6.0 subnet and I have a phone on the 192.168.1.0 subnet. |
| 14:42.06 | [TK]D-Fender | triple-e : I meant if SIP.20 was going to change its layout, more than just adding backlight, etc. |
| 14:42.35 | [TK]D-Fender | XanaXa : does your default route on * allow packets to go both ways seamlessly? |
| 14:43.00 | brif8 | ManxPower: ok I found the ${DNID} variable. If I'm using this as part of the file name to monitor/record a call how can I find what extension actually answers the call, seeing multiple extensions ring when the number is dialed ? |
| 14:43.08 | XanaXa | ok I might have asked too soon, default GW is net set in CentOS |
| 14:43.08 | triple-e | security stuff, probibly a bunch of bug fix's for them to jump from 1.6 to 2.0 --- i heard from tech that they were fixing my bug |
| 14:43.11 | XanaXa | :) |
| 14:43.22 | [TK]D-Fender | XanaXa : if so the only quesion would be if you're using NAT for OUTSIDE connections. If thats the case then you need only add another localnet clause in SIP.CONF to say that 192.168.1.0 is *also* local. |
| 14:44.09 | [TK]D-Fender | XanaXa : Well without a default route you sure as shit aren't getting very far outside your subnet now are you? :) |
| 14:44.11 | ManxPower | brif8, I dunno. It's in the logs. |
| 14:44.19 | XanaXa | bah I feel like a damned idiot but I am glad I found this channel though |
| 14:44.21 | [TK]D-Fender | XanaXa : Or at least a static one! |
| 14:44.30 | triple-e | what should the permission settings be on the voicemail directory |
| 14:44.42 | [TK]D-Fender | XanaXa : Don't worry, all SORTS of idiots find this room! ;) |
| 14:44.54 | triple-e | <---- one of them |
| 14:45.05 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:45.13 | XanaXa | it is working fine now, I am new to Asterisk, just got 2 phones and setup a server last night and I love it |
| 14:45.37 | Sonderblade | can you make an asterisk pattern that matches the same thing that both _XXX and _XXX# matches? |
| 14:46.01 | [TK]D-Fender | Sonderblade : You need to match each seperately. |
| 14:46.01 | brif8 | ManxPower: I realize it is in the cdr log, but it would be great if the filename showed the extension who answered the call also |
| 14:46.07 | triple-e | TK: can you take a look at your box and tell me your perm's on your voicemail dir |
| 14:46.15 | Sonderblade | [TK]D-Fender: why? |
| 14:46.15 | *** join/#asterisk woodhead_ (n=woodhead@pool-72-68-92-146.nwrknj.east.verizon.net) |
| 14:46.23 | ManxPower | ${EXTEN} contains that info. |
| 14:46.34 | [TK]D-Fender | Sonderblade : thats jsut the way * works. period. |
| 14:46.36 | ManxPower | Oh! You means the DEVICE!!!!!!!! |
| 14:46.48 | ManxPower | Since an extension can have 500 devices associated with it |
| 14:47.02 | Sonderblade | * should learn regexps |
| 14:47.13 | triple-e | extension can have 500 devices associcated with it ? |
| 14:47.15 | [TK]D-Fender | Sonderblade : you should learn * :) |
| 14:47.23 | *** join/#asterisk Ahrimanes (n=michael@62.61.133.90.generic-hostname.arrownet.dk) |
| 14:48.03 | blitzrage | the IP501 screen looks so weird if you look at it from an angle... |
| 14:48.05 | ManxPower | triple-e, Why would it not? |
| 14:48.17 | blitzrage | its like it projects the pixels |
| 14:48.28 | blitzrage | but if you look at it from an angle, you can see double pixels |
| 14:48.42 | [TK]D-Fender | blitzrage : its the "double-glass" effect... |
| 14:48.43 | triple-e | i don't know how to do that -- im hoping you can point me to something that i can learn something |
| 14:48.48 | [TK]D-Fender | blitzrage : ! ! ! |
| 14:48.53 | blitzrage | [TK]D-Fender: yah -- I don't remember seeing that on my IP500 |
| 14:49.05 | blitzrage | [TK]D-Fender: god, I could go for some of that this weekend |
| 14:49.12 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:49.20 | ManxPower | exten => 122,1,Dial(Zap/1&SIP/abcd&MGCP/fred&Zap/4&Zap/5.... |
| 14:49.22 | brif8 | ManxPower: device/extension when the number is called (dial,SIP/100,SIP/200,SIP/300,,Ttr) any of the three can answer. this is only known when the call is picked by eg 200. If I put ${EXTEN} in the file name I don't know that since the call has not been answered when monitor starts |
| 14:49.40 | blitzrage | brif8: that's incorrect syntax |
| 14:49.50 | ManxPower | brif8, wrong. The call is NOT PICKED UP BY EXTENSION 200!!!!!! |
| 14:50.01 | ManxPower | The call is picked up by the device with SIP user id 200 |
| 14:50.07 | C4T3l | should use & |
| 14:50.08 | *** join/#asterisk pb__ (n=pb@82-70-217-41.dsl.in-addr.zen.co.uk) |
| 14:50.25 | ManxPower | I think brif8 is using some silly macro for his example. |
| 14:50.30 | brif8 | ManxPower: ok my mistake, ok then how do I find the device ID then of who answered the call |
| 14:50.40 | triple-e | ManxPower: I see what you were talking about, I guess i was dreaming of a multicast one to many call |
| 14:50.45 | ManxPower | brif8, now you are asking the right question. I have no idea. |
| 14:50.56 | brif8 | sorry |
| 14:51.07 | ManxPower | We use the MAC address for the SIP user ID so we never make the mistake of thinking of a device as an extension |
| 14:51.11 | ManxPower | as SO many do. |
| 14:51.45 | ManxPower | brif8, you realize that using the "r" option to dial is like wearing a sign on your back that says "I am an idiot, kick me." |
| 14:51.50 | [TK]D-Fender | ManxPower : Exten is something your dial, devices is something * can ring IN the exten. Why can't people get this? *sigh* |
| 14:52.13 | ManxPower | [TK]D-Fender, because all the damn examples out there have device ids that look like extensions. |
| 14:52.22 | [TK]D-Fender | I swear key-system PBX's make people dumber (than they already are) |
| 14:52.30 | blitzrage | yay |
| 14:52.47 | brif8 | ManxPower: explain further on the "r" option ? |
| 14:53.19 | ManxPower | brif8, "r" means overide any sound the caller should be hearing and make the caller hear ringing, even if the caller should be hearing "the number you called is dosconnected" |
| 14:53.22 | ManxPower | don't use it. |
| 14:53.34 | brif8 | got it |
| 14:53.48 | ManxPower | brif8, also your Tt will allow EITHER side transfer the call using # |
| 14:53.54 | Zeeek | ManxPower great definition - it should go in the docs |
| 14:54.08 | rva | does anyone know sipura disconnect tones for siemens pbx? |
| 14:54.20 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:54.28 | Zeeek | but that would only take 3 seconds! :) |
| 14:54.30 | ManxPower | and brif8 pays for the call HAHAHAHAHA! |
| 14:54.38 | brif8 | so drop the T and just leave t |
| 14:54.56 | ManxPower | brif8, which one you use depends on the direction of the call. |
| 14:55.08 | ManxPower | better to use the transfer feature of the phone and not use t or T at all |
| 14:55.19 | brif8 | ok will do |
| 14:55.37 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.204) |
| 14:56.37 | ManxPower | and read the docs, don't just use an example some random person posted. |
| 14:57.19 | Zeeek | is it widely known that there is an open source SIP phone extension for Firefow? |
| 14:57.42 | lunk | i'm fat, but not that wide Zeeek |
| 14:57.55 | lunk | is there? |
| 14:58.05 | ManxPower | Zeeek, No idea, other than the fact that a softphone running in a browser is too kinky even for me. |
| 14:58.15 | Zeeek | yes. |
| 14:58.24 | Zeeek | I mean yes, there is |
| 14:58.38 | Zeeek | http://www.openwengo.com |
| 14:58.56 | Zeeek | and of course there is a non-free service you can use with it - but not required |
| 14:59.10 | ManxPower | all softphones suck |
| 14:59.19 | Zeeek | Wengo is a mainstream operator in France and they chose to stay open-source |
| 14:59.38 | Zeeek | well I too am addicted to hardphones, but there is a place for the softphone |
| 14:59.52 | mut | recycle bin? |
| 14:59.55 | Zeeek | having a PC on to talk on the phone is soooo 1999 |
| 15:00.07 | mut | Zeeek: pda foo |
| 15:00.34 | coppice | ManxPower: so asterisk must suck. its just a big multi-channel softphone |
| 15:00.48 | ManxPower | coppice, no, it's a PBX |
| 15:01.04 | salviadud | yeah, a hacker's pbx |
| 15:01.15 | ManxPower | if you used Asterisk as a softphone (soundcard, microphone, speakers) then it would suck too. |
| 15:01.35 | coppice | and the polycoms, snoms and ciscos must such, cos under those cases they are no different from any other softphone |
| 15:02.17 | salviadud | if you could get a polycom to handle iax2, that would be great |
| 15:03.02 | ManxPower | coppice, they run on an embeded OS with dedicated hardware. |
| 15:03.28 | ManxPower | If you ran a softphone on an embeded OS that nobody can get into and use dedicated hardware, then that so called "softphone" would not suck. |
| 15:03.47 | brif8 | Monitor(wav|filename) has to be before the call is picked up by a device right, it can't be after |
| 15:03.50 | ManxPower | dedicated hardware that the user cannot change or upgrade, that is. |
| 15:04.10 | ManxPower | brif8, you can run it after, but it won't record anything |
| 15:04.22 | coppice | so, asterisk must suck very very badly, since it tries to be many softphones all at once |
| 15:04.35 | Zeeek | heh |
| 15:04.38 | tzanger | softphones just suck because the interface is idotic. |
| 15:04.40 | tzanger | hardphones are just softphones with a decent interface |
| 15:04.49 | *** part/#asterisk woodhead_ (n=woodhead@pool-72-68-92-146.nwrknj.east.verizon.net) |
| 15:05.00 | Zeeek | the rectal joystick isn't available except for ciscos |
| 15:05.28 | coppice | if you want a phone for a call centre I think the interface of a softphone is much the superior one, as it integrates with the rest of the workstation |
| 15:05.37 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:06.05 | triple-e | found VM error |
| 15:06.11 | triple-e | asterisk was running as root |
| 15:06.24 | ManxPower | softphones suck because they they are run on any old random piece of slightly compatable PC hardware. They also suck because they are poorly writen and poorly designed. |
| 15:06.29 | Blackthorn | Hi, could somone tell me a little bit about the music on hold feature? What type of file format is used or available and can you listen to a live stream and what type of stream if possible? |
| 15:06.38 | blitzrage | all softphones suck because they are trying to look like a phone |
| 15:07.16 | tangel | is it possible to direct IP dial a vonage number? |
| 15:07.22 | coppice | blitzrage: now that is a broadly valid point. they are interface paradym challenged |
| 15:07.24 | lunk | my softphone has a crank on the side |
| 15:07.32 | mr_horsepower | ppl, anyone have used disa to collect number from a comercial pbx? |
| 15:07.34 | unmanagedaway | who here has worked in a call center ... I have and have to say login/out of a crappy lucent phone with codes and everything else to remember blows, and I have worked in a call center with some really crappy IP software phones |
| 15:07.35 | blitzrage | make the softphone look like MSN messenger so it hides in the task bar and only pops up a small window when you need it, then click on it to answer / or reject |
| 15:07.35 | tangel | i'm pissed that when i currently call a vonage customer i'm going from my voip->pots->voip |
| 15:07.52 | unmanagedaway | so everything can have its faults |
| 15:07.53 | salviadud | yeah, i've worked at a call center |
| 15:07.53 | blitzrage | coppice: exactly |
| 15:08.05 | salviadud | the soft phone we used sucked mayor a$$ |
| 15:08.15 | blitzrage | coppice: they haven't "gotten" it yet -- softphones suck because of interface, and not because of their usage |
| 15:08.18 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 15:08.37 | salviadud | it got the job done though... |
| 15:08.57 | wasim | call centers don't really require a soft phone interface, they just log on once, and stay logged on for 8 hours without really doing much, an occasional transfer, an occasional pause/unpause |
| 15:09.15 | tangel | anyone try calling vonage customers direct IP? =D |
| 15:09.25 | blitzrage | tangel: I wouldn't expect that you could |
| 15:09.32 | salviadud | if you want to call vonage, without vonage, you can try FWD |
| 15:09.40 | salviadud | i think |
| 15:09.45 | tangel | what would fwd do? |
| 15:09.53 | salviadud | you dial a prefix, then the vonage number |
| 15:09.54 | tangel | i'm sure i could call direct to the persons IP |
| 15:10.00 | tangel | phone@IP:10000 |
| 15:10.06 | blitzrage | tangel: if you're sure, why are you asking us? |
| 15:10.09 | salviadud | well. try it out dude |
| 15:10.09 | tangel | but i was hoping i could do phone@sip.vonage.com or something |
| 15:10.21 | tangel | i don't want to have to ask people for their ip address |
| 15:10.28 | salviadud | what a crime |
| 15:10.33 | salviadud | they'd have to trust you |
| 15:10.41 | tangel | Host sip.vonage.com not found: 3(NXDOMAIN) |
| 15:11.11 | unmanagedaway | wasim... hmm it is not that easy, time was also tracked via the phone... Working on projects, meetings, lunch, hell even taking a peee was all tacked by codes entered into a phone |
| 15:11.11 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
| 15:11.22 | tangel | gay.. voip is totally gay.. that e914 or whatever thing needs to catch on |
| 15:11.50 | salviadud | haha, gay? |
| 15:11.55 | Makenshi | e164? |
| 15:11.59 | salviadud | come one man, you can't gender offend voip |
| 15:12.10 | Makenshi | gay isn't gender specific |
| 15:12.32 | jaybuffet | that was fun.. |
| 15:12.33 | salviadud | so, voip is happy then |
| 15:12.44 | tangel | hah.. yeah, that's it. |
| 15:12.45 | jaybuffet | nice meeting with the boss to get him excited about asterisk |
| 15:13.04 | tangel | oh well, the forum posts i can find say vonage doesn't allow for it |
| 15:13.08 | Makenshi | voip would work if there was one standard for signalling and one standard for rtp |
| 15:13.24 | salviadud | iax2 again |
| 15:13.28 | [TK]D-Fender | Makenshi : They're called SIP & RTP :) |
| 15:13.41 | salviadud | no need for rtp |
| 15:13.51 | Makenshi | except there's h.323, sip, skinny, iax2, and gsm, alaw, ulaw, g719, ... |
| 15:13.55 | coppice | Really Terrible Protocol |
| 15:14.11 | [TK]D-Fender | Makenshi : half of that list are CODEC's, not protocols... |
| 15:14.11 | ManxPower | SIP and RTP were written by a bunch of hippies on LSD -- or at least that's what it looks like from reading the standards. |
| 15:14.13 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:14.17 | wasim | Stopped In Progress |
| 15:14.26 | salviadud | damn those hippies |
| 15:14.38 | salviadud | they were probably using FreeBSD |
| 15:14.47 | wasim | Makenshi: don't forget good ol' mgcp |
| 15:14.49 | tangel | ok.. so i have her ip now and i can direct dial from x-lite and it works |
| 15:15.11 | tangel | how can i do a direct sip connection through asterisk? .. my cisco phone doesn't seem to support url dialing so i need to setup an extension that gets forwarded |
| 15:15.17 | coppice | I don't know if they were on LSD, but i'm bloody sure they'd never actually used a phone |
| 15:15.33 | Blackthorn | Hi, could somone tell me a little bit about the music on hold feature? What type of file format is used or available and can you listen to a live stream and what type of stream if possible?] |
| 15:15.38 | coppice | wasim: we all want so much to forget MGCP |
| 15:15.39 | Makenshi | seems there's not much talk about TRIP |
| 15:15.48 | salviadud | Blackthorn, read the book |
| 15:15.51 | Makenshi | i think TRIP is a great alternative to e164 |
| 15:15.53 | coppice | does anyone use TRIP? |
| 15:16.12 | Makenshi | we do and oregon state do |
| 15:16.18 | coppice | The Really Irrelevant Protocol |
| 15:16.20 | *** join/#asterisk bamp (n=iraklion@olon.ath.forthnet.gr) |
| 15:16.20 | Makenshi | as for the rest... dunno |
| 15:16.21 | Makenshi | lol |
| 15:16.26 | salviadud | damn, my keyboard doesn't have a tilde... |
| 15:16.36 | coppice | ~~~~~~~ |
| 15:16.41 | tangel | <PROTECTED> |
| 15:16.45 | salviadud | ~thebook |
| 15:16.46 | jbot | thebook is probably Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org |
| 15:16.46 | coppice | cut and paste at will |
| 15:16.54 | salviadud | there ya go Blackthorn |
| 15:16.55 | Makenshi | rfc3219 |
| 15:16.55 | wasim | tangel: search for "Fear and Loathing in Las Vegas" |
| 15:17.01 | tangel | heh |
| 15:17.11 | coppice | vovida have a free implementation of trip |
| 15:17.36 | Blackthorn | i've read it dosn't tell you practicly anything aobut the file formats or streaming audio |
| 15:17.39 | blitzrage | OT: http://www.youtube.com/watch?v=dMH0bHeiRNg |
| 15:17.48 | salviadud | yes it does |
| 15:17.56 | salviadud | asterisk likes mp3 without tags |
| 15:18.05 | salviadud | and with constat bitrate |
| 15:18.19 | salviadud | constant |
| 15:18.43 | blitzrage | there is no mention of TRIP in 'thebook' |
| 15:18.48 | Zeeek | is there a cellphone-readable file format for vmail? |
| 15:19.07 | Makenshi | blitzrage, http://www.rfc-editor.org/rfc/rfc3219.txt |
| 15:21.25 | kay2 | is there a way to play a voicemail with video ? |
| 15:21.27 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-132-61.dsl.chcgil.sbcglobal.net) |
| 15:21.35 | kay2 | or an option to add to Voicemailmain ? |
| 15:21.36 | blitzrage | Makenshi: haha -- I like how its trying to be BGP, OSPF, and IS-IS :) |
| 15:21.47 | blitzrage | kay2: yah -- as long as the phone supports it, it should just work |
| 15:23.01 | mr_horsepower | damm, you guys does not connect nothing with comercial pbx? |
| 15:23.16 | coppice | one of the key problems with a lot of the media related RFCs is you read them and still have no clue why anyone would cook them up |
| 15:23.21 | salviadud | mr_horsepower. what are you trying to say? |
| 15:23.53 | unmanagedwork | see my question still stands ... how can I, if at all, set the "Q.931 dialplan" on a per call basis, |
| 15:24.24 | mr_horsepower | allways i try to speak with someone over here, to learn something new, no one have made that. |
| 15:24.24 | salviadud | mr_horsepower. are you russian? |
| 15:24.24 | unmanagedwork | trying to do some twiddle with a switch ... |
| 15:24.24 | blitzrage | mr_horsepower: everything is new -- you have to learn how to do it yourself |
| 15:24.25 | mr_horsepower | no :D |
| 15:24.35 | salviadud | mr_horsepower. are you hindu? |
| 15:24.47 | mr_horsepower | blitzrage: i'm not magic, this is not my side problems only |
| 15:25.02 | mr_horsepower | changing experience it's good. |
| 15:25.09 | blitzrage | you get what you pay for :) |
| 15:25.18 | salviadud | this dude writes just like the hackers from blacklist |
| 15:25.20 | blitzrage | if you need something complex done... hire a consultant |
| 15:25.23 | [TK]D-Fender | kay2 : THERE IS NO SUCH THING AS VIDEO. Get over it! its a miracle alone that RTP can pass it between 2 compatible phones, but as it is cross compatibility is flakey enough. |
| 15:26.07 | mr_horsepower | not so much complex, the problem it's that i dont have to much experience with commercial pbx |
| 15:27.36 | mr_horsepower | some commercial pbx implement hard and obscure ways from dialing a number! :o |
| 15:27.40 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:28.26 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 15:28.51 | salviadud | mr_horsepower. you talk funny |
| 15:29.06 | salviadud | where are you from? |
| 15:30.18 | mr_horsepower | portugal dude, my english needs practise. |
| 15:30.44 | salviadud | o voce fala portugues |
| 15:30.50 | blitzrage | mr_horsepower: still sounds like you need a consultant if you don't know how to use the commercial PBX |
| 15:30.52 | salviadud | well, my portuguese needs work |
| 15:30.54 | *** join/#asterisk Ox7a69 (n=Ox7a69@83.175.220.178) |
| 15:31.06 | mr_horsepower | i sayd portuguese, not brasilian. |
| 15:31.14 | [TK]D-Fender | I'm fluent in gibberish! |
| 15:31.24 | unmanagedwork | Location: Portugal |
| 15:31.30 | unmanagedwork | based on ip |
| 15:31.50 | unmanagedwork | http://www.dnsstuff.com/tools/ptr.ch?ip=82.102.1.42 |
| 15:31.52 | salviadud | well, i guess the portuguese from brazil and portugal are not the same then |
| 15:32.00 | Nivex | "Lady, I only speak two languages: English and Bad English." |
| 15:32.07 | mr_horsepower | blitzrage: if someone here, asks something i know, i can try to help! |
| 15:32.28 | mr_horsepower | salviadud: native portuguese its from portugal. |
| 15:32.43 | mr_horsepower | salviadud: portuguese in brasil, its brasilian. |
| 15:33.24 | salviadud | mr_horsepower. are the women in brasil hotter than the women in portugal? |
| 15:33.27 | mr_horsepower | unmanagedwork: so easy to "just ask" :D |
| 15:34.00 | mr_horsepower | salviadud: it depends on the woman. |
| 15:34.24 | salviadud | mr_horsepower. i'm looking for big booty girls |
| 15:34.34 | blitzrage | my friend just got back from south america -- apparently they have the hottest women :) |
| 15:34.36 | coppice | salviadud: depends if the air con is on or not |
| 15:34.55 | mr_horsepower | coppice: :) |
| 15:34.59 | coppice | death valley has the hottest women |
| 15:35.01 | salviadud | copz. i agree, that really determines if they're hot |
| 15:35.08 | *** join/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
| 15:35.31 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 15:35.52 | salviadud | coppice. are you from cali? |
| 15:35.53 | [TK]D-Fender | coppice : Nah, NY's "spontaneous combustion" win's out. exothermic > endothermic ;) |
| 15:35.53 | jaybuffet | can someone walk me through on how to sound intelligent when i am talking to a voip provider.. what questions do i need to ask ? |
| 15:36.06 | jeremib | is there anyway to edit the voicemail options so I can say in my message "press 1 to try me at my cell phone", then it transfers the call to my cellp hone? |
| 15:36.20 | mr_horsepower | i love NY, wonder if some day will ever live there. :D |
| 15:36.20 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:36.38 | *** join/#asterisk fulgas (n=fulgas@209.8.233.239) |
| 15:36.42 | [TK]D-Fender | jeremib : use either * or 0. Read up on the exit extens for VoiceMail. |
| 15:36.56 | jeremib | will do, thanks! |
| 15:37.26 | [TK]D-Fender | jeremib : Easily done. |
| 15:38.16 | salviadud | [TK]D-Fender. does that come up on the book? I was trying to find about voicemail.conf and found very little info |
| 15:38.46 | ManxPower | salviadud, voicemail.conf.sample does not have the information you are looking for? |
| 15:38.53 | [TK]D-Fender | salviadud : I never said it was in VOICEMAIL.CONF. |
| 15:39.07 | [TK]D-Fender | READ PEOPLE! |
| 15:39.16 | salviadud | hehe |
| 15:39.23 | ManxPower | "show applications" |
| 15:39.39 | salviadud | so, i should just look into the sample file |
| 15:39.44 | [TK]D-Fender | ManxPower : I'm not sure they're ready for such awesome power... |
| 15:39.54 | [TK]D-Fender | salviadud : Samples SUCK. |
| 15:40.19 | ManxPower | salviadud, The sample config files, the "show application X" in the CLI, the docs directory, the mailing list archives, the mailing list and finally here. |
| 15:40.53 | salviadud | show applications is pretty nice... |
| 15:41.19 | salviadud | ManxPower. you the man |
| 15:41.49 | ManxPower | "show applications like voice" |
| 15:42.06 | ManxPower | and even better "help" in the CLI. |
| 15:43.32 | unmanagedwork | http://bugs.digium.com/view.php?id=3493 |
| 15:43.39 | unmanagedwork | think I found what I need... |
| 15:43.44 | Sonderblade | is there a Case application/statement? |
| 15:43.51 | jaybuffet | whats a good ethernet switch for voip in a mixed network environment... or should the voip network be physically seperate |
| 15:45.42 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 15:46.34 | ManxPower | ARGH! The new CLEC wants to hand us off a "dual T-1" as an ethernet connection, not as two DXS-1 or 1 V.35. |
| 15:46.52 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:46.58 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:46.58 | lzhang | whenever I call app AMD() in my dialplan, it hangs and seems to get stuck in an infinite loop... has anybody experienced something similar to this, or heard of a solution? |
| 15:47.30 | ManxPower | fs-1*CLI> show application amd |
| 15:47.30 | ManxPower | Your application(s) is (are) not registered |
| 15:47.30 | ManxPower | fs-1*CLI> |
| 15:47.39 | ManxPower | perhaps you should ask the author of app_amd |
| 15:47.46 | wasim | lzhang: try INTEL() |
| 15:49.05 | lzhang | wasim: is that an application? |
| 15:49.37 | jeremib | [TK]D-Fender - if i use the exit extens for VM, does * and 0 do the same thing? so I couldn't have * = call cell and 0 = operator? |
| 15:49.40 | ManxPower | lzhang, Asterisk does not come with app_amd. |
| 15:49.55 | ManxPower | jeremib, read "show application voicemail" |
| 15:49.55 | mitka | has anyonde intregated asterisk with gsm channel bank |
| 15:50.07 | coppice | AMD = A Mythical Detector |
| 15:50.07 | [TK]D-Fender | jeremib : They do whatever you tell them to. |
| 15:50.09 | file | ManxPower: yes it does... |
| 15:50.14 | jeremib | ok |
| 15:50.22 | ManxPower | file, not 1.2 |
| 15:50.34 | *** join/#asterisk ToTo (n=ToTo@host134-88.pool8256.interbusiness.it) |
| 15:50.34 | jeremib | oh i see now, thanks ManxPower |
| 15:50.40 | file | silly 1.2 |
| 15:50.48 | ManxPower | hence "show application amd" returning Your application(s) is (are) not registered |
| 15:51.10 | lzhang | ManxPower: AMD is awesome, when it's working... I have it running on one asterisk install, but I can't seem to get it going on any others |
| 15:51.31 | ManxPower | lzhang, I/m happy for you. It still does not come with 1.2 |
| 15:52.21 | lzhang | yes I am aware of that... is this channel only for apps that come with 1.2 release or something? |
| 15:53.05 | brettnem | hey anyone else notice both of the iax2 voicepulse gateways just died? |
| 15:53.40 | mitka | anyone tired gsm gateway with asterisk |
| 15:53.47 | brettnem | what is app amd? |
| 15:53.47 | jeremib | yes brettnem |
| 15:53.59 | brettnem | jeremib: so I'm not the only one? |
| 15:54.17 | jeremib | well, i just tried to call my number an i get a fast busy, not even hitting my asterisk |
| 15:54.20 | lzhang | app_amd = answering machine detection |
| 15:54.21 | jeremib | though it's still registered |
| 15:54.34 | brettnem | jeremib: ok, mine just came back up |
| 15:54.47 | brettnem | jeremib: my backup gateway |
| 15:55.30 | brettnem | still won't connect.. |
| 15:55.31 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 15:55.32 | a1fa | hey |
| 15:55.54 | a1fa | sometimes when i recieve inbound calls and user type in extension numbers |
| 15:55.58 | a1fa | sometimes it freaks out |
| 15:56.08 | a1fa | for example, i dial "552" |
| 15:56.12 | brettnem | jeremib: ok, it's back up now.. like down for 10 minutes |
| 15:56.12 | a1fa | it dials "104" |
| 15:56.16 | unmanagedwork | http://pastebin.com/726716 |
| 15:56.18 | jeremib | thanks brettnem |
| 15:56.19 | a1fa | that is insane |
| 15:56.22 | unmanagedwork | AMD and asterisk 1.2.6 |
| 15:56.47 | a1fa | chan_sip.c:2542 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 15:56.51 | a1fa | maybe this is the output |
| 15:56.53 | a1fa | i dont know |
| 15:58.06 | lzhang | unmanagedwork: yes, those are the exact instructions I followed, I have it working on one box running 1.2.7.1 but on none others (also running 1.2.7.1) |
| 15:59.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:59.54 | *** join/#asterisk blumer (n=blumer@slobber.ruffdogs.com) |
| 16:00.14 | puzzled | hi |
| 16:03.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:03.18 | puzzled | I'm trying to use exten => _*06. and in features.conf changed disconnect => *0 to *3 but for some reason dialing *06<number> doesn't work. anything I forgot/did wrong? |
| 16:04.01 | blumer | I am a complete voip n00b, but I am troubleshooting a problem for someone. I am trying to determine whether or not an asterisk installation is receiving DTMF codes. At this point, what I've been able to do is do a tcp dump of the port on which Asterisk is listening and watch the incoming traffic when I push buttons, but I have not seen a discernable difference in traffic. Their application was formerly able to detect tones, but suddenly stopped. I |
| 16:04.01 | blumer | s there an "easy" way to determine if the tone is reaching the machine? Thanks. |
| 16:04.08 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 16:04.10 | blumer | Wow, that was really long. Sorry. :\ |
| 16:06.19 | salviadud | what chan are you using? |
| 16:06.26 | salviadud | is it SIP? |
| 16:06.28 | salviadud | iax2? |
| 16:07.22 | blumer | iax2, I believe. |
| 16:07.23 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:07.42 | blumer | How can I determine for certain? |
| 16:08.06 | salviadud | try the Background application |
| 16:08.27 | salviadud | if it doesn't recognize it there, well, you might know for certain if it's detecting DTMF |
| 16:08.36 | salviadud | do a small ivr |
| 16:08.48 | salviadud | and test it |
| 16:10.09 | blumer | "Background" application ... |
| 16:10.15 | blumer | clarify, please? |
| 16:10.16 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 16:10.20 | salviadud | ok |
| 16:10.37 | salviadud | Background works like playback |
| 16:10.40 | salviadud | it plays a sound file |
| 16:10.42 | salviadud | but |
| 16:10.51 | salviadud | background waits for an extension to be dialed |
| 16:11.07 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:11.11 | TheCops | When I'm using Dial command to one of my phone, I've got that error: May 19 12:00:31 WARNING[4662]: chan_sip.c:1973 create_addr: No such host: |
| 16:11.14 | TheCops | Someone know why ? |
| 16:11.32 | brettnem | hey, is it just me or does /var/lib/asterisk/sounds/letters/s.gsm ACTUALLY say "F" |
| 16:11.46 | blumer | salviadud: is this an executable that should be on the machine? Or a command to be execute from within ... the asterisk cli? |
| 16:11.54 | brettnem | TheCops: could be your dial syntax |
| 16:12.06 | TheCops | brettnem, no, the phone is ringing |
| 16:12.09 | TheCops | but got that warning |
| 16:12.19 | salviadud | blummer, this is a dialplan application, it should be added in your extensions.conf |
| 16:12.36 | salviadud | have you read the book? |
| 16:12.40 | blumer | got it. |
| 16:12.44 | salviadud | background is basic pbx stuff man |
| 16:12.48 | *** part/#asterisk jeremib (n=netnameu@c-71-203-209-162.hsd1.tn.comcast.net) |
| 16:12.56 | brettnem | TheCops: we'd have to see the full cli output including the dial statement to know anything |
| 16:13.15 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:13.48 | blumer | salviadud: I have not--total asterisk rookie trying to rapidly collect information to solve a customer's problem pronto |
| 16:14.13 | blumer | when you say "the book", referring to the man pages, or a specific book? |
| 16:14.31 | r_evolution|afk | ~thebook |
| 16:14.32 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org |
| 16:14.39 | snitt | The Book |
| 16:14.45 | brettnem | blumer! |
| 16:14.46 | blumer | great--thank you! |
| 16:14.48 | brettnem | ~gwypf |
| 16:14.49 | jbot | i guess gwypf is Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 16:15.12 | brettnem | we got all sorts of canned bot responses |
| 16:15.17 | blumer | :) |
| 16:15.38 | blumer | excellent--thank you guys for pointing me in the right direction. |
| 16:16.12 | snitt | :)) |
| 16:16.29 | kay2 | someone knows a SIP client that does H.263 |
| 16:16.29 | kay2 | ? |
| 16:16.34 | zoa | eyebeam i think |
| 16:17.30 | kay2 | zoa: not free, xlite is free tho but dunno about h263 |
| 16:19.21 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:19.45 | *** join/#asterisk master_of_bcm (n=AMione@cust-206-40-173-219.bos-static.gis.net) |
| 16:20.09 | master_of_bcm | hi all, how do you set the password with h323 phones? |
| 16:20.44 | master_of_bcm | I have 4 different phone shere and non of them cna connect to the server, it looks like h323 phones dont have user/passwords? |
| 16:21.50 | master_of_bcm | I only ahv experieience with sip phones |
| 16:26.12 | blitzrage | kay2: minisip.org I think |
| 16:26.22 | master_of_bcm | any idea? |
| 16:27.19 | [TK]D-Fender | kay2 : eyebeam & Ekiga do H.263 |
| 16:37.32 | *** join/#asterisk lithi (n=irssi@67.71.46.240) |
| 16:38.37 | *** join/#asterisk PBXtech (n=nik@70.89.247.188) |
| 16:38.39 | *** join/#asterisk gvainfo (n=gvainfo@AGrenoble-257-1-34-62.w86-206.abo.wanadoo.fr) |
| 16:38.39 | brif8 | can you just reload queues.conf or do you have to reload |
| 16:38.44 | gvainfo | hi |
| 16:39.24 | PBXtech | anyone ever heard of a 7940 with a laptop off the PC port that gets an IP number and can ping anything EXCEPT the default gateway? |
| 16:39.27 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:39.54 | Qwell[] | PBXtech: maybe the gateway blocks icmp |
| 16:40.13 | PBXtech | nope ping if i bypass the phone |
| 16:40.30 | PBXtech | its the phone |
| 16:40.35 | PBXtech | but its all the phones |
| 16:40.58 | *** join/#asterisk Lino` (n=Lino@i577BF40A.versanet.de) |
| 16:41.06 | PBXtech | makes it so the laptop cant get to the internet |
| 16:42.21 | *** join/#asterisk obanta (n=obanta@CPE-24-27-129-176.neb.res.rr.com) |
| 16:42.37 | dlynes_home | Is there any reason why I would be getting 100K of data on the receive queue for my rtp port, that asterisk never empties? |
| 16:45.15 | *** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net) |
| 16:48.54 | *** part/#asterisk blumer (n=blumer@slobber.ruffdogs.com) |
| 16:49.21 | *** join/#asterisk tuxd00d (n=tuxinato@adsl-63-205-99-182.dsl.lsan03.pacbell.net) |
| 16:51.24 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:54.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:56.07 | CunningPike | Morning, ladies and gents |
| 16:56.58 | CunningPike | I have a question relating to qualify=yes - is it nuts to put every one of 400 Polycoms on that setting? |
| 16:57.23 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:57.49 | Qwell[] | CunningPike: a qualify packet is small compared to rtp |
| 16:57.50 | [TK]D-Fender | CunningPike : should be OK. |
| 16:58.52 | CunningPike | Great - thanks - I just wanted to make sure it wouldn't kill stuff |
| 16:59.02 | brif8 | does "reload" drop the current calls or just reloads all modules like queues.conf ? |
| 16:59.11 | CunningPike | brif8: The latter |
| 16:59.30 | brif8 | CunningPike so it will not effect calls in progress |
| 16:59.38 | CunningPike | brif8: No |
| 16:59.48 | CunningPike | brif8: I do it all the time |
| 17:00.15 | CunningPike | brif8: It might affect calls in progress if a change you make breaks something....... :) |
| 17:00.35 | brif8 | no just changing the timout value in queues.conf |
| 17:01.15 | ManxPower | reload should never drop current calls. |
| 17:01.21 | CunningPike | brif8: What you can do for that is 'reload app_queue.so' (I think) that will only reload your queues |
| 17:01.39 | *** join/#asterisk ariel_ (n=Ariel@70.46.87.154) |
| 17:04.00 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:04.03 | *** join/#asterisk abatista (n=Ariel@70.46.87.158) |
| 17:06.57 | rva | i have transfer working from features.conf... |
| 17:07.03 | rva | from ip to ip extensions |
| 17:07.09 | rva | but if i call another asterisk server... |
| 17:07.24 | rva | the transfer happens there!!! and not in my asterisk. is that normal? |
| 17:07.56 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 17:08.26 | [TK]D-Fender | rva : pastebin the way you call this other server. |
| 17:08.45 | [TK]D-Fender | ~pb |
| 17:08.53 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:09.25 | rva | [TK]D-Fender: i just dial....over a sip trunk! what should i pastebin? |
| 17:10.36 | *** join/#asterisk ivanfm (n=ivanfm@c90604d7.virtua.com.br) |
| 17:10.38 | *** join/#asterisk salmandr (n=ben@mdsnwigjbas01-pool10-a181.mdsnwigj.tds.net) |
| 17:12.01 | *** join/#asterisk obanta (n=obanta@CPE-24-27-129-176.neb.res.rr.com) |
| 17:12.12 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 17:12.28 | mroth_imm | is anyone aware of what a flood of these messages: |
| 17:12.31 | mroth_imm | May 19 13:11:12 WARNING[6591]: Avoided deadlock for '0x2aaab0647100', 10 retries! |
| 17:12.37 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:12.44 | mroth_imm | means and how they can be resolved...Asterisk seems to be running fine other than that |
| 17:13.00 | brif8 | where is the profit margin in being a VoIP termination provider |
| 17:13.14 | mroth_imm | they've been coming through since around 10am... |
| 17:15.22 | *** join/#asterisk Skarmeth (n=Skarmeth@200164212156.user.veloxzone.com.br) |
| 17:16.16 | mroth_imm | is this thing on? |
| 17:16.24 | jaybuffet | if i have sip.conf => mysql,asterisk,ast_config in my extconfig.conf file, do i still need the actual sip.conf file? |
| 17:21.12 | Corydon-w | Yes |
| 17:21.39 | mroth_imm | Corydon-w: any hints on the avoided deadlock messages? |
| 17:21.43 | Corydon-w | extconfig is only for configuring users and peers, not for general config |
| 17:22.06 | Corydon-w | mroth_imm: are you thinking I know the problem and I'm just teasing you? |
| 17:22.13 | mroth_imm | i see the value in quotes is a channel, but it is not listed by 'sip show channels'...we are only using sip channels |
| 17:22.24 | mroth_imm | no, i was thinking that you were afk |
| 17:23.13 | mroth_imm | i apologize for my lack of psychic abilities |
| 17:23.17 | Corydon-w | mroth_imm: if I knew how to fix it, it would already be in trunk |
| 17:23.31 | mroth_imm | is there a way to kill the channel? |
| 17:23.47 | Qwell[] | soft hangup |
| 17:23.47 | Corydon-w | No, only soft hangup |
| 17:23.49 | *** join/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 17:24.16 | chino[server] | can someone give me some resources for voip providers in the north american region ? |
| 17:24.25 | Qwell[] | ~wikis |
| 17:24.27 | jbot | i guess wikis is http://www.voip-info.org |
| 17:24.28 | Qwell[] | chino[server]: search there |
| 17:24.40 | mroth_imm | okay...how do i tie the value listed back to the sip channel that needs hung up? |
| 17:25.11 | Corydon-w | It's listed in 'show channels' |
| 17:25.28 | Corydon-w | soft hangup works on the general channel name |
| 17:26.06 | [TK]D-Fender | rva : Pastebin exactly what you dial. |
| 17:26.24 | [TK]D-Fender | rva : and pastebin what the receiving end does with the call. |
| 17:27.48 | chino[server] | you guys know of any that allow you to use channels as needed and pay for usage at the end of hte month ? instead of a fixed channel plan ? |
| 17:28.29 | [TK]D-Fender | chino[server] : VoicePulse Connect |
| 17:28.37 | chino[server] | thanks :] |
| 17:30.28 | mroth_imm | i see nothing resembling the value in the error messages in the output of 'show channels' |
| 17:30.38 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 17:32.12 | mroth_imm | looking at the source, i'd say the value in the error message is the memory address of the channel structure |
| 17:33.07 | *** join/#asterisk meppl (i=mephisto@meppl.net) |
| 17:33.14 | mroth_imm | "/* c is surely not null, but we don't have the lock so cannot access c->name" |
| 17:33.30 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:33.55 | meppl | where does the sangoma-cards "A104" run best? |
| 17:34.05 | dlynes_home | meppl: in a pci bus |
| 17:34.16 | meppl | and with wich OS |
| 17:34.26 | dlynes_home | meppl: Windows, Linux, Solaris |
| 17:34.32 | blitzrage | [TK]D-Fender: you have a link for that polycom firmware? Some website I remember has it available, but I forget the address |
| 17:34.34 | meppl | so, its terrible in windows |
| 17:34.50 | meppl | windows often craches |
| 17:34.53 | dlynes_home | meppl: no...it's supported on Windows, Linux and Solaris |
| 17:35.04 | blitzrage | windows doesn't crash on me |
| 17:35.25 | dlynes_home | meppl: windows only crashes often if you have buggy drivers, buggy hardware, viruses, spybots, malware, spyware, adware, .... installed |
| 17:35.30 | *** join/#asterisk ivanfm (n=ivanfm@c90604d7.virtua.com.br) |
| 17:35.38 | meppl | i only found this beta-drivers for windows ftp://ftp.sangoma.com/WINDOWS/A101_A102_A104/ |
| 17:35.39 | Qwell[] | dlynes_home: or Windows |
| 17:35.44 | blitzrage | dlynes_home: totally agreed -- keep your PC clean, and it runs remarkably well |
| 17:35.53 | meppl | i have no viruses... |
| 17:35.55 | dlynes_home | Qwell: nah...windows since windows 2000 is pretty stable |
| 17:36.05 | Qwell[] | ME came out after 2000 |
| 17:36.06 | dlynes_home | Qwell: actually since nt 4 |
| 17:36.07 | blitzrage | w2k sucks -- winxp kicks ass |
| 17:36.13 | Qwell[] | ME came out way after NT4 |
| 17:36.15 | meppl | dlynes_home, so, windowsXP crashes, if i install the hardware-abstraction-driver |
| 17:36.25 | dlynes_home | blitzrage: not quite true....windows xp home edition freaking blows |
| 17:36.27 | blitzrage | "buggy driver" |
| 17:36.30 | meppl | dlynes_home, windows2k crashes, if i install the protocoll-driver |
| 17:36.32 | blitzrage | dlynes_home: oh yah -- I agree :) |
| 17:36.41 | blitzrage | dlynes_home: I just assume we're taking XP Pro-SP2 :) |
| 17:36.47 | Qwell[] | SP2...pfft |
| 17:36.53 | dlynes_home | But yeah...XP Pro SP2 is extremely stable |
| 17:37.07 | blitzrage | I'm afraid Vista is going to be ME all over again |
| 17:37.24 | dlynes_home | meppl: are you installing the "beta" version of the driver? |
| 17:37.42 | meppl | there is no information for wich windows it was tested |
| 17:37.47 | meppl | also for linux |
| 17:37.53 | meppl | no information which kernel |
| 17:38.00 | meppl | dlynes_home, yes |
| 17:38.11 | dlynes_home | meppl: ummm...i don't know about windows, but they say it works on all versions of linux back to 1.0 |
| 17:38.13 | meppl | dlynes_home, there are no other drivers to download for windows |
| 17:39.11 | dlynes_home | ftp://ftp.sangoma.com/WINDOWS |
| 17:39.17 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 17:39.39 | meppl | dlynes_home, yo, there is the beta-driver |
| 17:39.54 | meppl | in linux im too stupid too install it, i think |
| 17:39.57 | meppl | momentaneous |
| 17:40.03 | dlynes_home | probably |
| 17:40.05 | meppl | because: WARNING: Kernel source directory /lib/modules/2.6.15-1-k7/build not found! |
| 17:40.18 | dlynes_home | meppl: that's because you don't have kernel-dev installed |
| 17:40.28 | meppl | that folder is only available for 2.4-kernel |
| 17:40.40 | meppl | in my system |
| 17:41.05 | *** join/#asterisk BugKham (n=BugKham@125.24.9.169) |
| 17:41.41 | *** join/#asterisk ToTo (n=ToTo@host134-88.pool8256.interbusiness.it) |
| 17:42.01 | dlynes_home | meppl: go to ftp.kernel.org/pub/linux/kernel/v2.6 and download the source code for 2.6.15.7 or older, do make menuconfig, edit your makefile so that it installs to a different directory than the default (so you don't overwrite your existing kernel), then make bzlilo ; make modules ; make modules_install |
| 17:42.39 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 17:42.46 | meppl | :/ |
| 17:42.50 | dlynes_home | ? |
| 17:42.57 | meppl | dlynes_home, that needs time |
| 17:43.06 | meppl | okay |
| 17:43.06 | dlynes_home | meppl: not that much time |
| 17:43.14 | meppl | thank you for help |
| 17:43.32 | dlynes_home | meppl: Takes about an hour or so to do a kernel configuration if it's your first time (less than 1/2 hour if you're comfortable with it) |
| 17:43.51 | dlynes_home | meppl: and if you've got a decently fast machine, it should take about 20 minutes to do a compile |
| 17:44.06 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 17:44.13 | Assid | hey VoicePulse: you around? |
| 17:44.32 | dlynes_home | meppl: if it's an asterisk box you're installing it for, install the bare minimum, and compile all network drivers as modules (in case you need to swap out one network card and replace it with another) |
| 17:44.34 | *** join/#asterisk CYPRESS_A (n=nate@216-230-88-10.client.cypresscom.net) |
| 17:45.07 | dlynes_home | meppl: make sure you install the crc_ccitt and rtc modules as well (in case you want to use ztdummy for timing) |
| 17:45.27 | meppl | dlynes_home, so |
| 17:45.38 | meppl | dlynes_home, /lib/modules/2.6.15-1-k7/build should be for the headers |
| 17:45.42 | dlynes_home | meppl: if you've got a limited number of pci slots, compiling in APIC and SMP support as well |
| 17:45.47 | dlynes_home | meppl: correct |
| 17:46.04 | meppl | in my system i of course i can install the headers for 2.6-kernels |
| 17:46.05 | dlynes_home | meppl: /lib/modules/2.6.15-1-k7/build/include/linux |
| 17:46.13 | meppl | but then they are in another directory |
| 17:46.36 | *** join/#asterisk geiseri (n=geiseri@dsl-207-245-69-126.cust.oldcity.dca.net) |
| 17:46.53 | meppl | i must look how to tell the sangoma-setup, that its in another directory |
| 17:47.03 | geiseri | hi, has anyone got some experiance with iaxclient? |
| 17:47.11 | dlynes_home | meppl: Well, either learn how to configure your distribution of choice properly, or learn how to compile and install a kernel from source |
| 17:47.24 | meppl | /usr/src/linux-headers-2.6.15-1-k7/ |
| 17:47.41 | meppl | pf |
| 17:47.44 | meppl | i create a link |
| 17:47.48 | Assid | geez.. where is VoicePulse ? |
| 17:47.59 | Assid | cmon guys.. pay attention to irc |
| 17:48.28 | dlynes_home | ~seen VoicePulse |
| 17:48.33 | jbot | voicepulse <n=contact@unaffiliated/voicepulse> was last seen on IRC in channel #asterisk, 1d 21h 34m 31s ago, saying: 'I am investigating the issue we are discussing in PM.'. |
| 17:48.33 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 17:48.53 | meppl | dlynes_home, okay, it works :-P |
| 17:48.58 | meppl | thank you for help |
| 17:48.59 | Assid | man.. whats what he told me yday |
| 17:49.02 | dlynes_home | no problem |
| 17:49.15 | dlynes_home | meppl: but seriously...you should learn how to compile the kernel eventually, too |
| 17:49.27 | BadPacket | Assid: why not call them - since they're the only company with actual phone support |
| 17:49.34 | dlynes_home | meppl: if you ever need new kernel features, sticking yourself to your distro is going to be painful |
| 17:49.42 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:49.51 | Assid | yeah.. am just doing that |
| 17:50.08 | *** join/#asterisk ChristianASGI (n=Christia@64.89.118.139) |
| 17:50.17 | dlynes_home | heh...i wish we could run as a company without phone support :) |
| 17:50.21 | dlynes_home | that'd rock :) |
| 17:50.34 | dlynes_home | no more bitching, whining customers on the phone anymore :) |
| 17:50.55 | meppl | dlynes_home, im able to compile the linux-kernel... |
| 17:50.57 | BadPacket | Assid: can't really expect them to provide support on irc... since we all jump all over them every time they say something :) |
| 17:50.58 | flujan | guys, i'm having problems with a MFC r2 signalling. |
| 17:51.14 | flujan | Where I can paste the error? |
| 17:51.18 | CunningPike | Our phone system must be working great - our customers haven't called us to complain for ages..... ;) |
| 17:51.18 | dlynes_home | ~pb |
| 17:51.20 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:51.39 | dlynes_home | CunningPike: 99% of the time when we get complaints |
| 17:51.46 | dlynes_home | CunningPike: it's because Telus has screwed something up |
| 17:51.52 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 17:51.58 | Assid | great.. their DTMF is now having a rpoblem |
| 17:52.01 | flujan | do you use pastebin.com? .ca? |
| 17:52.08 | Qwell[] | flujan: doesn't matter |
| 17:52.14 | ChristianASGI | what channel should I ask a question about pridialplan being set from the dialplan to flag some calls as private isdn? |
| 17:52.15 | dlynes_home | flujan: doesn't matter...just pick one |
| 17:52.20 | flujan | ok |
| 17:52.39 | ChristianASGI | what channel should I ask a question about pridialplan being set from the dialplan to flag some calls as private isdn? |
| 17:52.54 | dlynes_home | ChristianASGI: no need to repeat so often |
| 17:52.58 | Qwell[] | ChristianASGI: ask 5 more times, and you'll get an answer, I promise |
| 17:52.58 | dlynes_home | ChristianASGI: it's this channel |
| 17:53.19 | flujan | Quension, dlynes_home here it goes... http://pastebin.com/726922 |
| 17:53.23 | ChristianASGI | I thought I whispered the first one to russellb |
| 17:53.36 | flujan | please, take a look. I trying my best without success... :( |
| 17:54.08 | dlynes_home | flujan: yeah...that's out of my league...i've never been able to get faxing of any description to work properly |
| 17:54.26 | flujan | dlynes_home, thanks... anyway! |
| 17:54.29 | dlynes_home | flujan: If you're able to catch coppice on here, he's the author of spandsp |
| 17:55.14 | *** join/#asterisk Loceur (n=noneya@vsas.veedix.com) |
| 17:55.16 | flujan | dlynes_home, how can I contact coppice? |
| 17:55.24 | dlynes_home | flujan: just catch him on here |
| 17:55.39 | dlynes_home | flujan: he's on here often enough |
| 17:55.49 | dlynes_home | ~seen coppice |
| 17:55.52 | jbot | coppice is currently on #asterisk (4h 40m 46s). Has said a total of 25 messages. Is idling for 2h 5m 45s, last said: 'AMD = A Mythical Detector'. |
| 17:56.01 | MikeJ[Laptop] | coppice is chatting on #openpbx right now |
| 17:56.06 | flujan | thanks... |
| 17:56.07 | flujan | :) |
| 17:56.28 | coppice | flujan: there is nothing wrong in what you have pasted |
| 17:56.43 | flujan | coppice, hi... nice to meet you! :) |
| 17:57.06 | flujan | coppice, I'm having difficults in testing the channel. How can I test it to see if its working? |
| 17:57.09 | Assid | great |
| 17:57.16 | Assid | now voicepulse has DTMF issues |
| 17:57.30 | flujan | coppice, I'm trying to connect a legacy proprietary pbx to asterisk using a T1 crossover cable. |
| 17:57.33 | BadPacket | Assid: ? |
| 17:57.53 | Assid | try calling them |
| 17:57.58 | Assid | i cant send dtmf |
| 17:58.32 | BadPacket | trying now |
| 17:58.51 | flujan | coppice, and congratulations for the nice work. It will be amusing if it were merged in the default asterisk installation. :) |
| 17:58.54 | coppice | flujan: you haven't actually described a problem yet |
| 17:59.11 | *** join/#asterisk jeffik (n=Jeff@Maroon-103-179.ADSL.NetSurf.Net) |
| 17:59.13 | BadPacket | Assid: i just got through and hung up on them |
| 17:59.23 | *** join/#asterisk vengeance0 (n=jdspence@mail.cai-engr.com) |
| 17:59.36 | vengeance0 | Hello |
| 18:00.01 | Assid | weird |
| 18:00.04 | vengeance0 | Could anyone tell me what I should do to compile Zaptel drivers with 4KSTACKS |
| 18:00.10 | flujan | coppice, http://pastebin.com/726937 there is protocol failure |
| 18:00.27 | Qwell[] | vengeance0: recompile your kernel with 4k |
| 18:00.52 | meppl | dlynes_home, so, sangoma: i tried different computers and different windows-versions and the only available beta-drivers were really unstable and difficult to install |
| 18:00.56 | meppl | and bad documentation |
| 18:01.04 | flujan | coppice, I saw the documentation and didn't know how to test my system... sending call throught the channels and so on. :( |
| 18:01.06 | meppl | installation in linux works much better |
| 18:01.09 | vengeance0 | thanks |
| 18:01.10 | dlynes_home | meppl: they're famous for bad documentation |
| 18:01.14 | Qwell[] | meppl: Why are you in here, asking for sangoma support on windows? |
| 18:01.22 | flujan | coppice, I didn't use the testcall program because of that. :( |
| 18:01.34 | flujan | coppice, where can I find additional documentation ? |
| 18:01.59 | dlynes_home | Qwell: maybe he's hoping to get asterisk/zaptel/sangoma working on windows :) |
| 18:02.11 | Qwell[] | #astwin32 |
| 18:02.15 | chino[server] | http://connect.voicepulse.com/Default.aspx that web site is killing me don't they have a 800 number ? |
| 18:03.02 | *** join/#asterisk XanaXa (n=m@ppp-69-219-158-119.dsl.chcgil.ameritech.net) |
| 18:03.04 | dlynes_home | heh..cool |
| 18:03.13 | dlynes_home | you do a whois on them, and you don't even get a phone number :) |
| 18:03.18 | SpaceBass | WOO HOO!!! My WIP330 should be delivered today! |
| 18:03.20 | ChristianASGI | Anyone have any insight into flagging PRI calls as private from the dialplan instead or zapata so that only select calls can be marked as private? pridialplan marks all calls out the channels as private or whatever flag you use. I just need to mark some calls as private. |
| 18:03.37 | Assid | BadPacket: dont work fro me |
| 18:03.43 | Qwell[] | dlynes_home: report them...phone number in dns records are mandatory now, I believe |
| 18:03.52 | dlynes_home | ah...nvm voicepulse: 732-339-5100 |
| 18:03.58 | wunderkin | ChristianASGI, setcallingpres=yes, and for the dialplan, show application setcallingpres |
| 18:04.11 | dlynes_home | it was just one solid number that looked like some kinda strange zip code :) |
| 18:04.54 | chino[server] | so thats their phone number ? |
| 18:05.00 | dlynes_home | chino[server]: i guess so |
| 18:05.20 | wunderkin | ChristianASGI, i mean in /etc/asterisk/zapata.conf, usecallingpres=yes |
| 18:05.21 | dlynes_home | chino[server]: that's their administrator's phone number |
| 18:05.34 | coppice | flujan: its hard to say from that log. you need to add "loglevel = 255" to your unicall.conf file and try again. you will get a more detailed log with enough info to see what is going on |
| 18:05.37 | dlynes_home | chino[server]: Ketan Patel |
| 18:05.48 | ChristianASGI | thank you wunderkin. SetCallerPres doesn't appear to include a "private" flag option. |
| 18:05.54 | chino[server] | im calling it lol |
| 18:06.03 | flujan | coppice, thanks very much... I will do this now. :) |
| 18:06.06 | mroth_imm | is anyone aware of what a flood of these messages: |
| 18:06.09 | mroth_imm | May 19 13:11:12 WARNING[6591]: Avoided deadlock for '0x2aaab0647100', 10 retries! |
| 18:06.13 | mroth_imm | means and how they can be resolved...Asterisk seems to be running fine other than that |
| 18:06.15 | BadPacket | Assid: odd, what happens? |
| 18:06.22 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:06.26 | chino[server] | lol its their real phone number |
| 18:06.29 | mroth_imm | looking at the source, i'd say the value in the error message is the memory address of the channel structure |
| 18:06.32 | chino[server] | thats bullshit |
| 18:06.36 | chino[server] | they should have it on the site too |
| 18:06.44 | dlynes_home | chino[server]: why would they? |
| 18:06.48 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:06.50 | *** join/#asterisk miguel3239_ (n=chatzill@h-68-167-124-171.cmbrmaor.covad.net) |
| 18:06.57 | dlynes_home | chino[server]: they don't want people irritating them with their problems ;) |
| 18:06.59 | brettnem | mroth_imm: it means your totally f-d |
| 18:07.01 | mroth_imm | so i'm unaware of how to hangup the channel |
| 18:07.15 | Dr-Linux | i'm just going to install asterisk on quad at datacenter, what asterisk and zaptel version is recommended? |
| 18:07.16 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 18:07.18 | mroth_imm | not really, the switch is handling over 120 calls fine, other than that message |
| 18:07.27 | brettnem | mroth_imm: it will soon die |
| 18:07.27 | dlynes_home | mroth_imm: from the cli soft hangup channelname, so like soft hangup zap/1-1 |
| 18:07.48 | mroth_imm | the trouble is there is no channel name in the message, it's the address of the channel structure |
| 18:08.00 | chino[server] | dlynes_home: for sales!!!! |
| 18:08.02 | mroth_imm | i need a way to go from the address to the channel name to issue a soft hangup |
| 18:08.07 | Assid | BadPacket: okay u think its this prvider |
| 18:08.17 | Assid | anwaysy.. i'l;l call them using another provider |
| 18:08.30 | BadPacket | Assid: must be - I used my cell phone and it was fine |
| 18:08.39 | mroth_imm | the message has been persistent since 10am, so I'm not too convinced it will just go away on its own |
| 18:08.40 | dlynes_home | wtf? |
| 18:08.52 | dlynes_home | high of 75 for the weekend, but during the week it was like 85 or 90 |
| 18:09.03 | mroth_imm | or alternately, a way to tell which channel has been up for four hours |
| 18:09.29 | brettnem | mroth_imm: when I get that error. my box is on it's last legs. |
| 18:09.36 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 18:09.51 | flujan | coppice, here is the updated log: http://pastebin.com/726948, and thanks for the help and patient. I will write a how-to or something like that to help newbies like me. :) |
| 18:09.54 | mroth_imm | yeah, i've seen it lead to problems before too... |
| 18:10.12 | brettnem | "avoided deadlock" is a total crock of shit |
| 18:10.29 | dlynes_home | brettnem: ? |
| 18:10.39 | brettnem | oh. haha.. I said that out loud! |
| 18:10.45 | BadPacket | Assid: I've used sipdiscount - dtmf never works with them |
| 18:10.54 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:10.56 | brettnem | "avoided deadlock" = "I'm dieing.. please kill me |
| 18:10.59 | dlynes_home | key word: 'discount' |
| 18:11.02 | Assid | actually used to work fine |
| 18:11.08 | Assid | anyways.. this is something weird |
| 18:11.14 | Assid | incioming call on voicepulse |
| 18:11.17 | dlynes_home | brettnem: nah...avoiding deadlocks is a normal course of operations for threaded code |
| 18:11.30 | Assid | should be coming in from voicepulse context.. its coming from voicepulse outgoing context |
| 18:11.35 | Assid | how the hell is that |
| 18:11.37 | brettnem | dlynes_home: dieing isn't... |
| 18:11.38 | dlynes_home | brettnem: i don't know why they bother issuing a message from the code for that |
| 18:11.43 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 18:11.50 | BadPacket | Assid: you should fix your config before you bother them |
| 18:11.51 | dlynes_home | brettnem: yeah, but dying has nothing to do with avoiding deadlocks |
| 18:12.00 | mroth_imm | considering i've been seeing the same message from the same channel for 4.5 hours, i'd say it's not normal in this instance |
| 18:12.01 | wunderkin | dlynes_home, i think it was put into debug level |
| 18:12.06 | BadPacket | Assid: RTFE |
| 18:12.17 | Assid | RTFE ? |
| 18:12.19 | mroth_imm | it's a WARNING |
| 18:12.23 | BadPacket | read the ***cking example |
| 18:12.29 | brettnem | dlynes_home: when it happens stuff breaks.. I don't care if its a freakin warning |
| 18:12.39 | dlynes_home | mroth_imm: It's not a warning; trust me...I'm a programmer |
| 18:12.40 | BadPacket | I pasted their example config and took it from there... worked right away |
| 18:12.54 | Assid | err.. calls are coming in fine |
| 18:13.06 | Assid | just saying tis weird to show as the outgoing call instead of incoming |
| 18:13.07 | mroth_imm | wunderkin: it displays a pointer address, so I'm SOL unless I can somehow map it to a name to hang it up...any ideas? |
| 18:13.17 | stack_ | We are getting an interference/staticy/popping noise on some calls on our PRI... someone told me this may be a codec issue... would that be true? |
| 18:13.18 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 18:13.20 | mroth_imm | the channel's been up for a LONG time, so maybe I can id it that way? |
| 18:13.25 | mercestes | reload chan_zap.so? |
| 18:13.38 | BadPacket | that has nothing to do with them |
| 18:13.41 | mercestes | killall -9 asterisk? |
| 18:13.43 | Assid | i know |
| 18:13.47 | Assid | im just saying its weird |
| 18:13.49 | brettnem | rm -rf * |
| 18:13.51 | southtel | In Realtime/static, can I do the equivalent of an "#include <filename>" ? |
| 18:13.51 | BadPacket | you're weird |
| 18:13.56 | mercestes | rm -dvfr Yes |
| 18:14.01 | Assid | either hwich way.. the call is dropping out after 2 mins or so |
| 18:14.10 | Assid | only happening with calls terminating over them |
| 18:14.10 | mercestes | dont' jsut delete....delete verbosely...and force it. |
| 18:14.30 | dlynes_home | rm -rf / |
| 18:14.34 | brettnem | I just told the people at the cafeteria that their meat in their hamburgers i funky. |
| 18:14.37 | brettnem | then I bought one |
| 18:14.42 | BadPacket | meat? |
| 18:14.46 | brettnem | MEAT |
| 18:14.54 | CunningPike | stack_: What card do you have? |
| 18:14.56 | BadPacket | spamburgers |
| 18:15.09 | stack_ | CunningPike: Digium TE110p |
| 18:15.19 | brettnem | more liek cannedburgers |
| 18:15.31 | dlynes_home | stack_: are you getting any frame slippage? |
| 18:15.49 | stack_ | dlynes_home: can you define frame slippage? |
| 18:15.50 | CunningPike | stack_: What is the zttest output like? (min/max/avg) |
| 18:15.50 | unmanagedwork | http://www.bozosoft.com/mike/meat/brains-article.html |
| 18:15.57 | BadPacket | i dunno assid, sounds like it's something on your box - is it always exactly 2 minutes? |
| 18:15.59 | unmanagedwork | PorkBrain burgers |
| 18:16.09 | Assid | nah |
| 18:16.10 | Assid | random |
| 18:16.10 | brettnem | that is freakin gross |
| 18:16.12 | Assid | sometimes less |
| 18:16.13 | Assid | and no |
| 18:16.15 | dlynes_home | stack_: you'll get messages on your console about bad something or other HDLC |
| 18:16.15 | Assid | nothing with my box |
| 18:16.20 | Assid | last time they had this problem |
| 18:16.23 | Assid | i called them |
| 18:16.23 | BadPacket | I've been on hold with Dell for the last 30 minutes on a call through them |
| 18:16.23 | Assid | they fixed it |
| 18:16.28 | Assid | that was a few months ago |
| 18:16.55 | Assid | hell.. i have these guys talking for over 5000 seconds.. |
| 18:16.57 | Assid | works fine |
| 18:17.11 | Assid | i think they screwed something up during the change over to connect01/02 |
| 18:17.14 | brettnem | hey anyone know the status of Sonic using voip for their drive thru windows and sending customers to call centers in India? |
| 18:17.36 | stack_ | dlynes_home, yes, I get "PRI got event: HDLC Bad FCS (8) on Primary D-channel..." but they don't coincide with the interference |
| 18:17.38 | lzhang | brettnem: that's hilarious |
| 18:17.48 | brettnem | lzhang: I don't *think* I'm making it up |
| 18:17.48 | Assid | BadPacket: dont get me wrong.. they are a good service |
| 18:17.49 | unmanagedwork | "All the more brilliant, then, that the label includes a recipe for scrambled eggs and brains, which sound like the kind of meal that keeps bypass surgeons in business. The recipe leads off with instructions to "drain brains" -- I'll bet someone at Armour is still chuckling over that one." |
| 18:17.57 | Assid | i just wish they clean up the kinds of the new servers |
| 18:18.01 | BadPacket | Assid: all of them suck, they just suck less |
| 18:18.05 | dlynes_home | stack_: that can cause the popping noise and/or dropped calls afaik |
| 18:18.08 | lzhang | brettnem: I used to work at Sonic... man it would've been nice if we didn't have to take orders |
| 18:18.13 | stack_ | dlynes_home: the HDLC stuff pops up sporatically and doesn't seem to affect anything |
| 18:18.16 | Assid | cause now even legacy ones are messed up |
| 18:18.17 | brettnem | BadPacket: I miss the days I used to be able to drive down to Dell.. :( |
| 18:18.18 | coppice | flujan: that doesn't look like a MFC/R2 link. |
| 18:18.19 | dlynes_home | stack_: however, CunningPike would know more about it than I would |
| 18:18.30 | dlynes_home | stack_: he's got quite a bit of experience with the pri cards |
| 18:18.36 | brettnem | lzhang: makes sense if you think about it.. they don't need to talk to people in the local store. |
| 18:18.51 | dlynes_home | stack_: but that's usually indicative of a hardware problem |
| 18:18.53 | CunningPike | stack_: dlynes_home is overstating my usefulness ;) |
| 18:18.59 | dlynes_home | lol |
| 18:19.09 | stack_ | CunningPike: hehe |
| 18:19.09 | BadPacket | brettnem: i gotta tell you, their support blows |
| 18:19.09 | flujan | coppice, I called the proprietary pbx help desk and they say they use this kind of link... So I went to try it. |
| 18:19.16 | BadPacket | brettnem: dell, that is |
| 18:19.23 | Assid | BadPacket: i really find them one of the least sucky ones out there. so i guess in a way the best to use.. heck i got aruond 4-5 accounts with them |
| 18:19.28 | brettnem | BadPacket: it does since they moved it international |
| 18:19.29 | CunningPike | stack_: However, you should have zero HDLC errors - the first thing to check is your zaptel.conf - can you pb it for me? |
| 18:19.29 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
| 18:19.40 | BadPacket | brettnem: damned foreigners |
| 18:19.43 | stack_ | CunningPike, sure, just a sec |
| 18:19.44 | brettnem | BadPacket: I used to drive to dell and buy computers in a store here.. |
| 18:19.50 | flujan | coppice, how can I detect what kind of link it is? Maybe there are giving me wrong information... |
| 18:19.54 | *** join/#asterisk Splas (n=jwb@206.252.198.100) |
| 18:19.55 | brettnem | hey we're all foreigners, right? |
| 18:20.03 | BadPacket | brettnem: I was kidding |
| 18:20.10 | brettnem | I know. ;) |
| 18:20.11 | CunningPike | stack_: Let's take this to a pm session |
| 18:20.14 | dlynes_home | You're all foreigners to me |
| 18:20.21 | dlynes_home | Unless you're Canadian :) |
| 18:20.36 | brettnem | hmmm.. the ultimate irony |
| 18:20.36 | Assid | arnet canadians french? |
| 18:20.47 | Qwell[] | Assid: only the french ones |
| 18:20.48 | dlynes_home | bu shi |
| 18:20.55 | flujan | coppice, at least is it configured like one MFC/R2? |
| 18:20.56 | coppice | flujan: the signalling bit patterns from the other end do not look like the normal patterns for an R2 link. The other end seems to be responding to Asterisk setting the link to idle, but its not responding in the right way |
| 18:21.02 | dlynes_home | womenda zhongguo ren :) |
| 18:21.11 | Assid | que ? |
| 18:21.32 | *** join/#asterisk Splas (n=jwb@206.252.198.100) |
| 18:21.35 | dlynes_home | no, we're all chinese :) |
| 18:21.37 | BadPacket | verizon sucks |
| 18:21.39 | brettnem | awesome.. check it out |
| 18:21.39 | brettnem | http://www.nytimes.com/2006/04/11/technology/11fast.html?ex=1302408000&en=fba08e17788e24c9&ei=5090&partner=rssuserland&emc=rss |
| 18:22.26 | flujan | coppice, and what if the link is blocked in the end point? Does the link appear to be blocked or something like that? |
| 18:22.32 | brettnem | wow: Software tracks her productivity and speed, and every so often a red box pops up on her screen to test whether she is paying attention. She is expected to click on it within 1.75 seconds. |
| 18:23.11 | coppice | flujan: R2 supports blocking |
| 18:23.44 | BadPacket | meat |
| 18:24.10 | Assid | call centers can take your throat away mn |
| 18:24.14 | Assid | imagine talking so much |
| 18:24.26 | flujan | coppice, thanks very much for the help. I will contact them and see if something is wrong. |
| 18:24.37 | flujan | coppice, see ya... :) |
| 18:24.40 | brettnem | "If you're in L.A.... and you hear a person with a North Dakota accent taking your order, you'll know what we're up to," McDonald's Chief Executive Jim Skinner told analysts |
| 18:24.43 | flujan | thank you all guys! ;) |
| 18:24.46 | *** join/#asterisk gvainfo (n=gvainfo@AGrenoble-257-1-45-227.w86-206.abo.wanadoo.fr) |
| 18:24.58 | Qwell[] | outsourcing to ND? |
| 18:25.05 | Qwell[] | That's kinda cool...better than I expected |
| 18:25.07 | BadPacket | damned foreigners |
| 18:26.04 | brettnem | How about just put festival and sphinx behind the wheel at mc-donald and eliminate like 50,0000,000,00,,0000 jobs |
| 18:26.06 | mercestes | 1.75 seconds? OMG, the average human reactiontime is around 1.6 |
| 18:26.09 | Assid | docelm0 ? |
| 18:26.10 | coppice | if you're in bangalore and your curry order is taken by someone with a north dakota accent, you'll know they've outsourced the work |
| 18:26.19 | brettnem | mercestes: see, plenty of lag built in there |
| 18:26.21 | Qwell[] | coppice: give it time |
| 18:26.26 | Qwell[] | it'll come full circle |
| 18:26.41 | brettnem | coppice: won't you be disappointed to know your curry was cooked in north dakota as well? |
| 18:26.42 | mercestes | gah, the things teenagers will do for minimum wage. |
| 18:26.50 | Qwell[] | brettnem: nah, better... just give everybody the touchscreens that the cashiers have |
| 18:26.51 | dlynes_home | Qwell: shouldn't be long with the value of the us dollar lately :) |
| 18:27.06 | brettnem | Qwell: that's kinda like the old automat resturants |
| 18:27.07 | Qwell[] | even idiots can do that job...and many mcdonalds customers are idiots (or former employees), so... |
| 18:27.08 | mercestes | bet I could start a porn industry for 5.35 an hour in the right cities. |
| 18:27.16 | Qwell[] | No offense to the mog man, btw |
| 18:27.29 | coppice | last time I looked there were about 100,000 people in Bangalore working permanent nights in call centres. Add in hyderbad and other places. how can that many bloody phones need answering? :-\ |
| 18:27.36 | gvainfo | sorry to interrupt you, just a question for debian stable + asterisk - is there a "proven" good way aside compiling to get the zaptel pseudo device? |
| 18:27.42 | Qwell[] | and if an order gets screwed up, you now have nobody to blame but yourself |
| 18:27.51 | Qwell[] | win-win-win, imo |
| 18:27.51 | dlynes_home | coppice: and lahore (where Dr-Linux works in a call center) :) |
| 18:27.59 | Qwell[] | except for like...mcdonalds cashiers...they lose |
| 18:28.10 | BadPacket | mcdonalds sucks |
| 18:28.22 | coppice | the pakistan call centres tend to be fairly small scale. |
| 18:28.38 | brettnem | An automat was a form of a cafeteria-style restaurant in which simple foods, usually coffee, sandwiches, and other fare such as macaroni and cheese, were served to the clientele by means of coin-operated vending machines. |
| 18:28.56 | Qwell[] | brettnem: yeah... |
| 18:30.15 | coppice | dlynes_home: there are quite a few substantial call centres in china, serving chinese communities outside china |
| 18:30.30 | dlynes_home | coppice: heh...I wouldn't doubt it |
| 18:30.48 | dlynes_home | coppice: I guess you know how many chinese we have in vancouver, eh? |
| 18:30.58 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:31.04 | coppice | less than we have here :-) |
| 18:31.10 | dlynes_home | well, duh :) |
| 18:31.34 | dlynes_home | But Richmond, which is about 300K people, is probably 70% Chinese |
| 18:32.17 | coppice | and mostly cantonese speakers |
| 18:32.24 | dlynes_home | yep...mostly hong kongese |
| 18:32.38 | coppice | nei sik m'sik gong gwong dung wah? |
| 18:32.45 | BadPacket | godblessyou |
| 18:32.47 | dlynes_home | wo buzhi dao |
| 18:32.50 | jaybuffet | when i start asterisk using "/etc/init.d/asterisk start" i cannot connect to the console using "asterisk -r", but if i start it from the exec i can connect fine... why would that by... it complains about the asterick.ctl file when i cannot connect but the file exists |
| 18:32.58 | mercestes | Gazundheit |
| 18:33.32 | BadPacket | "nei sik m'sik gong gwong dung wah" = "Hungry?" |
| 18:33.33 | Flauto | if i did not understand wrongly, using moh-native, there is no way for me to control the loudness of the player? |
| 18:33.36 | coppice | se pu tung wah ping yam ho nan |
| 18:33.52 | mercestes | ~mercestes |
| 18:33.59 | jbot | i heard mercestes is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. |
| 18:33.59 | BadPacket | " se pu tung wah ping yam ho nan" = "no" |
| 18:34.00 | dlynes_home | He's saying he wants to go to Henan, to pick up some sexy ladies |
| 18:34.22 | Flauto | coppice, what is that language? |
| 18:34.36 | dlynes_home | Flauto: Cantonese, I think...or maybe Hakka, or Hokkien |
| 18:34.49 | Flauto | are you guys chinese? |
| 18:35.00 | dlynes_home | I'm Caucasian, but I do speak some Mandarin |
| 18:35.06 | mercestes | I think I'm turning japanese. I really think so. Does that count? |
| 18:35.15 | coppice | i'm one of those rare blonde blue eyed chinese |
| 18:35.20 | dlynes_home | lol |
| 18:35.28 | Flauto | really? i am chinese and i speak mandarin |
| 18:35.38 | Flauto | haha |
| 18:35.51 | Flauto | i am a real one |
| 18:36.12 | coppice | ngoh m'sik teng pu tung wah. ngoh ji sik gwong dung wah |
| 18:36.37 | Flauto | coppice, that is not real mandarin pin yin |
| 18:36.49 | coppice | its cantonese |
| 18:36.51 | dlynes_home | Flauto: It's some kinda weird cantonese romanization |
| 18:36.59 | Flauto | hehe |
| 18:37.20 | coppice | it says I can't speak mandarin, only cantonese |
| 18:37.29 | dlynes_home | I'll probably never understand cantonese...it's too gutteral |
| 18:37.44 | dlynes_home | well, and add to the fact that I know very few cantonese |
| 18:37.55 | dlynes_home | Almost everyone I know is from the mainland |
| 18:38.07 | dlynes_home | mostly near Hangzhou and Beijing |
| 18:38.24 | coppice | more than one person who came from the north to hong kong as a child told me they found it easier to master english than cantonese |
| 18:38.25 | Flauto | i am from beijing |
| 18:38.36 | dlynes_home | coppice: lol |
| 18:38.37 | coppice | i was in hangzhou just yesterday |
| 18:38.44 | jaybuffet | file /var/run/asterisk.ctl exists and its owned by the asterisk user, yet i stall cant connect to "asterisk -r"... is that because im logged in as root.. even though i added the root user to the asterisk group |
| 18:38.55 | dlynes_home | Yeah...I'd love to go to west lake one of these years |
| 18:38.58 | Flauto | i am going back to beijng in a week |
| 18:39.02 | CunningPike | jaybuffet: It means that asterisk isn't starting from init.d |
| 18:39.04 | dlynes_home | West lake is extremely beautiful |
| 18:39.30 | jaybuffet | <PROTECTED> |
| 18:39.32 | coppice | west lake is nice again now. it was a building site a couple of years ago. hotels, conference centres, and stuff |
| 18:39.47 | dlynes_home | I know a couple people that live in Hangzhou, so the chances of me going to Hangzhou next time instead of returning to Beijing is much higher |
| 18:40.03 | CunningPike | jaybuffet: It's not working then - check your init script |
| 18:40.12 | coppice | go to long tseng. its really nice there |
| 18:40.13 | CunningPike | Also your logs for messages |
| 18:40.20 | dlynes_home | coppice: yeah...probably everyone's moving to hangzhou to see the beautiful girls :0 |
| 18:40.44 | dlynes_home | coppice: long tseng? You mean Long Ceng? |
| 18:41.10 | coppice | dunno how you pin yin ize it. the place near hangzhou where the tea grows |
| 18:41.11 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 18:41.13 | Flauto | come on guys |
| 18:41.16 | Flauto | answer my question |
| 18:41.22 | dlynes_home | coppice: Oh...you mean Long Jin |
| 18:41.29 | dlynes_home | coppice: long = dragon, jin = well |
| 18:41.39 | coppice | i don't jave a chinese IME set up on this machine |
| 18:41.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:41.43 | Flauto | when i use noh-native, i have no control the loudness? |
| 18:41.51 | dlynes_home | Flauto: correct |
| 18:41.57 | jaybuffet | seems like init.d/asterisk is using different config files..... |
| 18:42.07 | CunningPike | jaybuffet: There ya go :) |
| 18:42.17 | brettnem | Flauto: I have a native MOH installation and the audio is WAY too loud |
| 18:42.18 | *** join/#asterisk azzie (n=az@azzie.net) |
| 18:42.23 | dlynes_home | coppice: you mean longjin cha...the very famous green tea, right? |
| 18:42.26 | Flauto | so, what is the point of using it then? |
| 18:42.27 | azzie | anybody uses Dash911 here ? |
| 18:42.49 | brettnem | azzie: I've looked into it, but too expensive |
| 18:42.53 | dlynes_home | Flauto: the volume's just fine for me |
| 18:43.10 | CunningPike | Flauto: The point is not having to use mpg123 |
| 18:43.12 | Flauto | hmm... |
| 18:43.17 | brettnem | I've had lots of customer complaints with the loudness |
| 18:43.17 | dlynes_home | Flauto: but I adjust it in another program before I dump it on the hard drive |
| 18:43.17 | azzie | brettnem, buck fifty per customer you mean? |
| 18:43.27 | CunningPike | Flauto: You can deal with it when creating your files in the first place |
| 18:43.30 | brettnem | azzie: yeah and monthly charges.. |
| 18:43.32 | coppice | yeah. dragon well of tea fame |
| 18:43.43 | brettnem | CunningPike: I tried that.. couldn't do anything to reduce the volume |
| 18:43.43 | dlynes_home | coppice: yeah...in mandarin, it's Long Jin |
| 18:43.53 | *** join/#asterisk dapatrick (n=dapatric@dsl253-031-098.phl1.dsl.speakeasy.net) |
| 18:43.54 | azzie | brettnem, i'm screwing around with them now... what do you use? |
| 18:44.03 | coppice | I think the road signs say Long Tseng |
| 18:44.09 | CunningPike | Will the real Flauto please stand up? |
| 18:44.11 | blitzrage | anyone here using Polycom IP501s? Just curious what your experiences with the various bootroms and SIP images are, and which versions you recommend? |
| 18:44.13 | brettnem | azzie: I'm a carrier.. so I do things the old fashioned way |
| 18:44.28 | azzie | brettnem, lucky :-) |
| 18:44.37 | jaybuffet | do you have to be logged in as the same user as asterisk is running in order to do "asterisk -r" |
| 18:44.38 | dlynes_home | coppice: maybe that's in Hokkien then |
| 18:44.38 | jaybuffet | ? |
| 18:44.38 | brettnem | blitzrage: I use them.. good phone. horrid XML files |
| 18:44.45 | CunningPike | blitzrage: We use IP501s with the latest of each - no problems - they just keep getting better |
| 18:44.59 | Assid | CunningPike: you got the latest 1.6.6 right ? |
| 18:45.01 | blitzrage | which is the current versions? |
| 18:45.03 | brettnem | I haven't tried the new sip load with the buddy list patch yet |
| 18:45.06 | blitzrage | of the bootrom and the sip.ld? |
| 18:45.11 | brettnem | I run I think 1.6.4 |
| 18:45.14 | CunningPike | Assid: Correct - it adds presence icons to the line keys |
| 18:45.21 | dlynes_home | coppice: after all, Zhejiang borders Fujian |
| 18:45.24 | Assid | im running 1.6.3 |
| 18:45.25 | blitzrage | where do you get the images? I have a new IP501 -- do I have to sign up for something? |
| 18:45.37 | brettnem | Earlier versions could do buddy lists, but could only do 7 at a time |
| 18:45.38 | coppice | most road signs in china have mandarin pin yin for the english part. I usually read the chinese, so I don't always take too much notice |
| 18:45.44 | Assid | bootrom - i thought they said only update to 3.x if you have problems |
| 18:45.46 | blitzrage | I realized I can download the files from 'freedomphones' :) |
| 18:45.48 | CunningPike | blitzrage: From your reseller - or search the list archives |
| 18:45.50 | Assid | cause you cant downgrage again |
| 18:46.07 | brettnem | blitzrage: I don't think 1.6.6 is available on freedomphones |
| 18:46.17 | blitzrage | doesn't look like it |
| 18:46.18 | Assid | yeah.. i didnt see it last time either |
| 18:46.19 | blitzrage | just 1.6.5 |
| 18:46.23 | Flauto | cunninglike, how are you doing. i am using moh-native now, but the the sound is kind of quiet. i am playing classical music on it, that might be the problem |
| 18:46.30 | Assid | yeah |
| 18:46.33 | dlynes_home | coppice: yeah...i just figured it might be hokkien rather than mandarin because Taiwanese usually use Yale romanization, rather than hanyu pinyin |
| 18:46.42 | brettnem | Flauto: I wish mine was quiet.. mine is so loud it makes people want to hang up on hold |
| 18:46.46 | CunningPike | Flauto: Yes - it might well be |
| 18:46.51 | Assid | there should be a changelog file man.. would be nice to see the difference in the versions |
| 18:47.04 | brettnem | a changelog file man? |
| 18:47.06 | Flauto | hehe |
| 18:47.15 | Assid | changelog file |
| 18:47.18 | brettnem | "Hello, I'm the changelog file man" |
| 18:47.33 | brettnem | sorry.. ;) |
| 18:47.39 | Assid | geez.. leave out a few punctuations.. and you get shot in here |
| 18:47.54 | CunningPike | 1.6.6 isn't general release, I don't think - latest SIP is 1.6.5, and latest bootrom is 3.1.3.0131 |
| 18:47.59 | dlynes_home | Flauto: Try adjusting the amplitude in a waveform manager before saving it on the hard drive |
| 18:48.00 | brettnem | oh come on.. we're family.. we all pick on each other |
| 18:48.06 | *** join/#asterisk ScubaDude (n=fritzbra@196.207.41.251) |
| 18:48.28 | ScubaDude | lo all |
| 18:48.50 | Assid | hell im using 1.6.5.0043 as per what sip.ver says |
| 18:48.57 | Flauto | yeah, i guess that the the only way then |
| 18:49.03 | coppice | well, on tuesday evening I was enjoying dragon well tea beside west lake :-) |
| 18:49.13 | Flauto | taiwanese now is using pin yin |
| 18:49.18 | *** part/#asterisk ScubaDude (n=fritzbra@196.207.41.251) |
| 18:49.20 | CunningPike | Assid: Are you having problems with it? |
| 18:49.33 | stack_ | randomly our caller id's show up as "asterisk", but most of the time it shows the correct callerid... any ideas? |
| 18:49.38 | Assid | CunningPike: nah |
| 18:49.39 | coppice | Flauto: not very often |
| 18:49.39 | Flauto | and also |
| 18:49.51 | Assid | wasnt really looking at upgrading unless its got something really worth it |
| 18:49.56 | blitzrage | stack_: sendrpid=yes and trustrpid=yes |
| 18:50.01 | CunningPike | stack_: 'asterisk' is what gets inserted for a blank CID |
| 18:50.09 | CunningPike | We used a macro to change is |
| 18:50.13 | CunningPike | s/is/it/ |
| 18:50.15 | stack_ | blitzrage: where do those go? |
| 18:50.20 | blitzrage | stack_: sip.conf |
| 18:50.29 | blitzrage | stack_: check the sip.conf.sample file for more information |
| 18:50.36 | stack_ | blitzrage: thans |
| 18:50.50 | stack_ | s/thans/thanks/ |
| 18:51.54 | dlynes_home | Flauto: heh...the taiwanese here, are still very much deadset against using pinyin |
| 18:52.02 | dlynes_home | Flauto: they all still insist on using zhuyin |
| 18:52.13 | Flauto | is there a way to set expiry for a single register under sip.conf |
| 18:52.34 | dlynes_home | Flauto: maxexpiry=nnnn, i think |
| 18:52.37 | coppice | dlynes_home: but they left, so they have no say :-) |
| 18:52.50 | Assid | dlynes_home: dont you have to set it in the phone/client ? |
| 18:52.56 | dlynes_home | Flauto: yeah...maxexpiry and defaultexpiry |
| 18:53.10 | coppice | for typing zhuyin is very popular in taiwan |
| 18:53.32 | Flauto | well, if i set up that, it is for all registers though |
| 18:53.32 | chino[server] | is 1 cent per minute allot ? |
| 18:53.34 | Assid | chino[server]: nah.. just 1 cent |
| 18:53.42 | dlynes_home | Flauto: oh yeah...i guess that's only global |
| 18:53.57 | chino[server] | what do you normally pay with a pstn link ? |
| 18:53.59 | Assid | Flauto: you can set the expiry in oyur client |
| 18:54.01 | dlynes_home | Assid: yeah, you can set it in the client, too |
| 18:54.03 | chino[server] | i thought most polaces you get unlimited calls for liek 30 a month |
| 18:54.04 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:54.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:54.08 | Flauto | no way to set only one? |
| 18:54.11 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:54.14 | dlynes_home | chino[server]: you don't...you just pay a monthly charge |
| 18:54.17 | dlynes_home | Flauto: in the client |
| 18:54.32 | xp_prg | can the Asterisk extensions function PlayBack handle ogg? |
| 18:54.33 | blitzrage | chino[server]: yah -- but it depends how many minutes a month you use. $30 will give you 3000 mins a month |
| 18:54.37 | dlynes_home | chino[server]: i think you want www.vonage.com |
| 18:54.51 | chino[server] | voiceplus voip says 11$ per month for 4 channel up/down and its 1cent per minute for outgoing |
| 18:55.10 | Assid | chino[server]: no.. 1 cent to CERTAIN locations |
| 18:55.18 | chino[server] | yes i know that |
| 18:55.18 | *** join/#asterisk mgob (n=goldenol@65.171.196.18) |
| 18:55.19 | blitzrage | usually North America |
| 18:55.24 | chino[server] | but is that a good rate ? |
| 18:55.24 | dlynes_home | chino[server]: is 1c/mi cheap for what? |
| 18:55.26 | Flauto | dlynes how? |
| 18:55.34 | dlynes_home | chino[server]: I'm getting 0.0065/mi |
| 18:55.42 | mgob | anyone have issues with pickup() only working on some extensions? or contexts? |
| 18:55.43 | Assid | you get 0.0065 ? |
| 18:55.46 | blitzrage | chino[server]: its pretty standard -- you're not going to get much cheaper unless you buy bulk minutes |
| 18:55.49 | Assid | who the hell do you use? |
| 18:55.55 | dlynes_home | Flauto: it depends on your client |
| 18:55.58 | chino[server] | so only for outgoing means if someone calls one of my 4 channels then its free ? |
| 18:56.00 | mgob | I am just doing Pickup(${EXTEN:2}) and it works for one context but not the other |
| 18:56.03 | *** join/#asterisk Enderson (n=enderson@smtp.gentoo.org) |
| 18:56.12 | dlynes_home | Assid: a canadian company |
| 18:56.19 | Flauto | what you mean by client? |
| 18:56.19 | Assid | dlynes_home: got a site? |
| 18:56.30 | dlynes_home | Flauto: your softphone or your hardphone |
| 18:56.34 | blitzrage | dlynes_home: heh -- carrier, or reseller? |
| 18:56.37 | dlynes_home | Flauto: or your ata or your sip gateway |
| 18:56.43 | Assid | dlynes_home: hell hows their quality? |
| 18:56.43 | dlynes_home | blitzrage: carrier |
| 18:56.50 | blitzrage | dlynes_home: yah, that makes sense |
| 18:57.03 | dlynes_home | blitzrage: it's wholesale pricing |
| 18:57.07 | blitzrage | dlynes_home: yep |
| 18:57.12 | blitzrage | dlynes_home: sounds like it :) |
| 18:57.28 | Flauto | the register is for a service |
| 18:57.29 | dlynes_home | blitzrage: but those are direct routes, not shit routes |
| 18:57.53 | blitzrage | dlynes_home: i.e. they have a PRI in the locations they terminate to? |
| 18:58.02 | dlynes_home | blitzrage: we have another one that's cheaper for us to terminate to than it is for us to terminate to locally |
| 18:58.05 | chino[server] | dlynes_home: you have a link ? |
| 18:58.30 | blitzrage | this doesn't sound like a consumer business guys |
| 18:58.42 | dlynes_home | blitzrage: i.e. it's cheaper per minute to terminate to them, than what it costs as a fraction of our pri |
| 18:58.53 | blitzrage | dlynes_home: interesting :) |
| 18:59.03 | dlynes_home | chino[server]: how many minutes do you push per month? |
| 18:59.06 | blitzrage | VoIP is neat -- but it's just a fad |
| 18:59.23 | Assid | dlynes_home: you got their site for me ? |
| 18:59.38 | dlynes_home | Assid: how many minutes do you push per month? |
| 18:59.39 | chino[server] | dlynes_home: i have no idea |
| 18:59.53 | chino[server] | blitzrage: its not a fad |
| 18:59.59 | Assid | around 30 bucks worth ? with voicepulse rates |
| 19:00.11 | blitzrage | chino[server]: I know -- its a joke... |
| 19:00.12 | dlynes_home | Assid: then there's no way these guys would even talk to you |
| 19:00.16 | chino[server] | oh lol |
| 19:00.22 | Assid | how many mins do these guys need? |
| 19:00.30 | dlynes_home | Assid: it's wholesale, not retail |
| 19:00.34 | Assid | damn |
| 19:00.41 | chino[server] | so whsay 1 cent per minute for outgoing calls they literally mean that if poepple call me its free ? |
| 19:00.46 | dlynes_home | if you have to ask, you're not pushing enough |
| 19:01.23 | Assid | well.. if i club everyone together.. prolly around 100 bucks worth.. i guess |
| 19:01.33 | blitzrage | chino[server]: no -- it'll be 1c/min, either direction usually |
| 19:01.40 | brettnem | where are you guys terminating calls to? |
| 19:01.41 | dlynes_home | Assid: even just one of my customers is pushing about $150/mo to the Punjab |
| 19:02.11 | r_evolution | Assid - you ever manage to get the DID you were wanting? |
| 19:02.20 | Assid | well.. found a place |
| 19:02.25 | dlynes_home | btw |
| 19:02.34 | dlynes_home | Anyone able to get me a Montreal DID? |
| 19:02.38 | blitzrage | dlynes_home: I think I can |
| 19:02.46 | Assid | you tried didx ? |
| 19:02.46 | *** join/#asterisk bartpbx (n=bartpbx@p54B01B29.dip0.t-ipconnect.de) |
| 19:02.47 | dlynes_home | I just need one, not a whole whack |
| 19:02.54 | [TK]D-Fender | dlynes_home : Unlimitel terminates here |
| 19:02.58 | blitzrage | dlynes_home: yah -- what kind of plan you need? |
| 19:03.00 | dlynes_home | rehan wallah wallah bing bang is an idiot |
| 19:03.14 | bartpbx | hello, i have a problem compiling zaptel on a x86_64 system |
| 19:03.31 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:03.35 | dlynes_home | blitzrage: no idea offhand...I just have one customer that happens to need a did there |
| 19:03.43 | Assid | bartpbx: dont use x86_64 |
| 19:03.47 | Assid | i learned that the hard way |
| 19:03.47 | blitzrage | dlynes_home: cool, let me know -- I can get DIDs there |
| 19:04.12 | bartpbx | assid: ok, why not? are there any known problems witth x86_64? |
| 19:04.16 | dlynes_home | blitzrage: let me do a brief check to see what her current calls are to montreal, and it'll probably be about 1/2 that volume |
| 19:04.31 | blitzrage | dlynes_home: sounds good |
| 19:04.33 | stack_ | blitzrage: I made those changes to sip.conf and we just had an inbound call with the following: "CallerID returned with error on channel..." |
| 19:04.45 | Assid | bartpbx: known .. not sure.. ijust had a terrible experience |
| 19:04.45 | *** join/#asterisk trelane (i=trelane@66.93.203.199) |
| 19:04.51 | Assid | i tink someone else did too |
| 19:05.01 | Assid | not sure. but it might be on google as well |
| 19:05.04 | blitzrage | stack_: need more info -- pastebin the dialplan call flow |
| 19:05.05 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:05.24 | blitzrage | with some noop()'s to show what the ${CALLERID(name)} and ${CALLERID(number)} return |
| 19:05.32 | *** part/#asterisk Enderson (n=enderson@smtp.gentoo.org) |
| 19:05.55 | dlynes_home | blitzrage: the one she needs specifically is for the 350 CO |
| 19:05.59 | bartpbx | google says nothing about a Problem |
| 19:06.56 | dlynes_home | [TK]D-Fender: what's unlimitel? |
| 19:07.03 | [TK]D-Fender | dlynes_home : www.unlimitel.ca |
| 19:07.04 | dlynes_home | [TK]D-Fender: and you're in Montreal? |
| 19:07.09 | [TK]D-Fender | dlynes_home : Indeed |
| 19:07.18 | dlynes_home | [TK]D-Fender: ah...thought you were American :0 |
| 19:07.19 | Flauto | dyynes, go to a conference room somewhere, so i can hear your mandarine |
| 19:07.30 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 19:07.31 | dlynes_home | Flauto: lol...my mandarin's not that good :) |
| 19:07.39 | dlynes_home | Flauto: it's quite weak, actually |
| 19:07.42 | stack_ | blitzrage: http://pastebin.ca/57162 |
| 19:07.43 | Flauto | hehe |
| 19:07.46 | dlynes_home | Flauto: i need to find a mandarin wife :) |
| 19:07.46 | r_evolution | TK's one of *those* guys... |
| 19:08.08 | dlynes_home | Flauto: just haven't had the time to take more courses since my last trip to China |
| 19:08.42 | r_evolution | yikes! |
| 19:08.56 | blitzrage | stack_: no idea.. that should not have affected a Zap channel at all |
| 19:08.56 | Flauto | good luck. i am mandarin, but my gf is cantonese |
| 19:09.03 | dlynes_home | Flauto: btw...from what i hear |
| 19:09.14 | dlynes_home | Flauto: when you go back to beijing, you won't recognize it...it's changed a lot |
| 19:09.39 | dlynes_home | Flauto: see? that's the trouble with living in the us |
| 19:09.49 | dlynes_home | Flauto: you end up meeting cantonese girls, not mandarin girls :) |
| 19:09.56 | bartpbx | assid, what was your hard way to learn it? |
| 19:10.05 | bartpbx | what was the problem? |
| 19:10.16 | dlynes_home | Flauto: personally, I like mandarin speaking girls a lot more |
| 19:10.20 | Assid | wrell.. something to do with the zaptel timing device |
| 19:10.32 | dlynes_home | Flauto: they seem to be a lot more feminine |
| 19:10.37 | blitzrage | dlynes_home: I terminated almost a million minutes last month -- will your friends talk to me? :) |
| 19:10.43 | Flauto | not true |
| 19:10.45 | dlynes_home | blitzrage: lol |
| 19:10.47 | Flauto | it really depends |
| 19:10.56 | dlynes_home | blitzrage: one of them only terminates about a million a month, himself |
| 19:11.03 | blitzrage | dlynes_home: 945,557.69 actually :D |
| 19:11.08 | dlynes_home | blitzrage: the other one I'm sure terminates more than that |
| 19:11.11 | stack_ | blitzrage: it seems as though everything coming in on the TDM400 is not getting caller id info..., works fine on the TE110p |
| 19:11.18 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-5-22.slkc.qwest.net) |
| 19:11.21 | Assid | i really should sum up their minutes of usage |
| 19:11.55 | dlynes_home | blitzrage: once we get going though, I hope to be terminating at least 100K/mo by the end of the year |
| 19:12.02 | rollergrrl | In regards to 5ess, 4ess, and NI2... is there one that is preferred over another? |
| 19:12.03 | Assid | oh crap |
| 19:12.06 | r_evolution | girl != katty in here? |
| 19:12.07 | Assid | i gotta wake up in 5 hrs |
| 19:12.19 | dlynes_home | rollergrrl: mi5 |
| 19:12.29 | r_evolution | watching bond? |
| 19:12.33 | rollergrrl | dlynes_home: hilarious |
| 19:12.35 | blitzrage | mi:3 |
| 19:12.50 | dlynes_home | rollergrrl: oops...thought you were just putting random digits and letters together :) |
| 19:12.58 | r_evolution | this channel will self-destruct in 10 seconds... |
| 19:13.02 | Assid | 5 hrs 2 mins |
| 19:13.06 | rollergrrl | r_evolution: why do people say that every time I come in here? |
| 19:13.27 | r_evolution | i dunno... i suppose because everyone gets accustomed to Katty being the only girl here |
| 19:13.40 | Assid | we have a girl in here? |
| 19:13.41 | dlynes_home | Nah... rollergrrl is in here regularly, too |
| 19:13.49 | dlynes_home | So is Corydon-w |
| 19:13.49 | mercestes | girls? where? |
| 19:14.02 | Assid | shes almost non existent.. barely any 'talk time' |
| 19:14.04 | dlynes_home | But Corydon-w is married |
| 19:14.05 | rollergrrl | I'm not ask talkative as katty though |
| 19:14.12 | blitzrage | dlynes_home: looks like I can only get 514 areacode for now... |
| 19:14.25 | mercestes | quick, everyone put on your "dress pocket protectors." |
| 19:14.30 | dlynes_home | blitzrage: I said 350 CO, not 350 NPA :) |
| 19:14.33 | Corydon-w | I'm a girl? |
| 19:14.41 | r_evolution | hah @ mercestes |
| 19:14.41 | rollergrrl | close to one |
| 19:14.45 | rollergrrl | hehe |
| 19:14.51 | dlynes_home | Corydon-w: Oh...thought you were cause i heard you talking about your husband or something the other night |
| 19:14.57 | rollergrrl | He's gay |
| 19:14.58 | blitzrage | dlynes_home: oh -- then I have no idea what you're asking :) |
| 19:15.01 | dlynes_home | oh |
| 19:15.06 | mercestes | it's that whole civil liberties thing. |
| 19:15.18 | chino[server] | blitzrage: they sid incoming is free |
| 19:15.25 | blitzrage | chino[server]: good for them |
| 19:15.32 | dlynes_home | blitzrage: like 1-514-350-xxxx |
| 19:15.42 | brettnem | I like that bumper sticker "I didn't need my civil liberties anyway..." |
| 19:15.52 | [TK]D-Fender | dlynes_home : Why 350 for CO? |
| 19:16.06 | dlynes_home | [TK]D-Fender: local to another CO I guess |
| 19:16.17 | blitzrage | dlynes_home: oh -- well, then you didn't give me enough info, lol |
| 19:16.21 | dlynes_home | [TK]D-Fender: the thing is, I don't know what's local there, and what isn't, and niether did she |
| 19:16.22 | blitzrage | is Montreal only one NPA? |
| 19:16.33 | [TK]D-Fender | dlynes_home : total waste of an idea. thats just "on-island". typically anything counting as down-town Mtl has the same calling range. |
| 19:16.40 | Assid | man.. i got some crazy ass power fluctuations |
| 19:16.40 | rollergrrl | Anyway... should stick my PRIs with 5ess and NI2... should I just throw a dart? |
| 19:16.46 | [TK]D-Fender | blitzrage : Ummm NO :D |
| 19:16.54 | blitzrage | [TK]D-Fender: ok, didn't think so |
| 19:17.01 | rollergrrl | Let me fix that sentence |
| 19:17.04 | Assid | undervoltage mostly.. lights become dimm and sometimes.. TOOOOOO bright |
| 19:17.08 | [TK]D-Fender | blitzrage : We're moving to forced 10-digit dialing in a month and adding AREA codes to Montreal :) |
| 19:17.09 | rollergrrl | Anyway... should I stick my PRIs with 5ess or NI2... should I just throw a dart? |
| 19:17.18 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 19:17.22 | blitzrage | [TK]D-Fender: you don't have 10-digit now? pffffft -- sooooo behind the times ;) |
| 19:17.52 | [TK]D-Fender | blitzrage : You keep importing our greatest export : Anglophones ;) |
| 19:18.20 | blitzrage | :D |
| 19:18.30 | blitzrage | [TK]D-Fender: you're not a separatist are you? *glares* |
| 19:18.34 | [TK]D-Fender | "I'm an alien. I'm a legal alien. I'm an englishman in Quebec" |
| 19:18.48 | blitzrage | [TK]D-Fender: you're not a conservative are you? *glares* |
| 19:19.03 | *** join/#asterisk adorah (n=FreePBX8@87.69.72.228) |
| 19:19.12 | Assid | docelmo ? |
| 19:19.16 | Assid | you around |
| 19:19.24 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:19.34 | [TK]D-Fender | blitzrage : Don't glare at me child! |
| 19:19.47 | mercestes | So, I heard at work the other day that mexico was talkiing about legalizing pot......that would be funny....for the first time in world history....illegal americans would be run out of mexico. |
| 19:19.54 | blitzrage | [TK]D-Fender: oh I'll *glare* -- and you'll LIKE IT |
| 19:19.56 | Assid | is there any changelog for the differences between 1.6.2 and 1.6.5 ? |
| 19:20.25 | [TK]D-Fender | Assid : Yes. |
| 19:20.47 | [TK]D-Fender | Assid : And SIP 1.6.6 is out and 2.0 coming next month. |
| 19:21.14 | Assid | [TK]D-Fender: not public yet right ? and which file contains the changes? |
| 19:21.23 | dlynes_home | [TK]D-Fender: wtf? you don't have ten digit dialing yet? I think even thunder bay has ten digit dialing |
| 19:21.25 | bartpbx | Assid: I would realy like to try it myself with x86_64. Did you had the same problem compiling zaptel? |
| 19:21.46 | bartpbx | it looks like i typo but ths can not be i think |
| 19:21.47 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:21.48 | Assid | bartpbx: i dont remember what the problem was EXACTLY.. but i remmeber something about zaptel and that causing me alot of problems |
| 19:21.59 | blitzrage | [TK]D-Fender: how do you get your files?! :D |
| 19:22.00 | [TK]D-Fender | Assid : Polycom publicly published their changelog in PDF format. |
| 19:22.11 | blitzrage | [TK]D-Fender: changelogs are where? |
| 19:22.27 | [TK]D-Fender | blitzrage : By asking for them :) |
| 19:22.32 | adorah | How the ITEF is going to tackle seriously nt raversal issues in SIP? |
| 19:22.35 | [TK]D-Fender | blitzrage : Only Polycom's site |
| 19:22.42 | blitzrage | [TK]D-Fender: but where on the polycom site? |
| 19:22.48 | [TK]D-Fender | blitzrage : Changelogs are free, SOFTWARE is not. |
| 19:22.50 | Assid | [TK]D-Fender: do you have to update the sip.cfg if you update the sip files? |
| 19:22.55 | [TK]D-Fender | blitzrage : Go look you lazy ass! |
| 19:22.59 | [TK]D-Fender | :D |
| 19:23.01 | blitzrage | [TK]D-Fender: I did! couldn't find them |
| 19:23.30 | Assid | yeah neither could i |
| 19:23.30 | r_evolution | :( |
| 19:23.30 | Assid | i think we turning blind man |
| 19:23.30 | [TK]D-Fender | Assid : Not if you're on SIP 1.6 already |
| 19:23.30 | r_evolution | not everyone is a disciple of the church of polycom TK ;) |
| 19:23.30 | dlynes_home | [TK]D-Fender: I just needed an exchange/CO that's local to 1-514-705-xxxx |
| 19:23.38 | Assid | can we get 1.6.6 off you ? ;) ? |
| 19:23.55 | mercestes | he won't share, I tried. |
| 19:24.01 | Assid | haha |
| 19:24.02 | jaybuffet | figured out why i couldnt connect if anyone cares... i didnt set the ASTVARRUNDIR var correct in the Makefile when i compiled it... once i changed it.. everything works |
| 19:24.07 | Assid | was always worth a try |
| 19:24.17 | dlynes_home | blitzrage: are you familiar with what's local to 1-514-705? |
| 19:24.32 | blitzrage | dlynes_home: sorry, no idea :( |
| 19:24.42 | dlynes_home | blitzrage: ah...didn't think so |
| 19:24.43 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.172) |
| 19:24.51 | mercestes | I basically called up some random small voip provider and screamed at them about my sip configs until they emailed them to me and apologized for not being able to find me in their database. |
| 19:25.00 | brif8 | anyone know where is the profit margin in being a VoIP Termination Provider. I mean most offer +/- 3c/min and numbers at $5.00 per month. Yet they probably pay 1c/min from CLEC. |
| 19:25.21 | bartpbx | hm. strange`.. any c / makefile specialist interessted in having a look at this problem? |
| 19:25.29 | r_evolution | mercestes... that's like what we used to do as kids with hardees |
| 19:25.38 | r_evolution | go into hardees and bitch out whoever was working the counter... and get free food |
| 19:25.39 | r_evolution | dirty. |
| 19:25.58 | bartpbx | this seams to be a Problem in zaptel-1.2.5 on x86_64 the same version is runing fine on a none x86_64 system |
| 19:26.30 | Assid | i stil dont getwhy polycom wont just open their site for free downloads |
| 19:26.30 | dlynes_home | [TK]D-Fender: is unlimitel residential only? |
| 19:26.39 | chino[server] | gsm takes up 64 kb of band width does that stand for kilo bytes or kilo bits ? |
| 19:26.46 | [TK]D-Fender | dlynes_home : what kind of usage youplanning for that DID? |
| 19:27.02 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 19:27.03 | [TK]D-Fender | dlynes_home : No, they do both. and metered / normal as well. |
| 19:27.05 | brif8 | chino[server]: bits |
| 19:27.17 | adorah | GSM shouldn't take that much bandwidth.. |
| 19:27.27 | r_evolution | little b = bits... big B = bytes :) |
| 19:27.33 | schirpich | how do you assign a extention to a queue? |
| 19:27.34 | dlynes_home | [TK]D-Fender: probably about 1000mi/mo or so |
| 19:28.01 | chino[server] | brif8: and you talk about your isp have 256k up your talking about bits or bytes ? |
| 19:28.06 | mercestes | ............ |
| 19:28.46 | r_evolution | mercestes... the abuser of small companies :( |
| 19:28.47 | Assid | unlimitel .. 4c/min ? |
| 19:28.54 | mercestes | ~mercestes |
| 19:28.56 | jbot | hmm... mercestes is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. |
| 19:28.56 | bartpbx | hm.. anyone has a x86_64 system not runing suse and could try to compile zaptel? |
| 19:29.07 | r_evolution | someone was feeling a little goth |
| 19:29.13 | [TK]D-Fender | dlynes_home : Unlimitel is for you then. Get a DID from them ($2.5/mo) and calls will be $.011/min |
| 19:29.21 | blitzrage | lol |
| 19:29.24 | blitzrage | ~blitzrage |
| 19:29.26 | jbot | well, blitzrage is a super cool fellow |
| 19:29.30 | [TK]D-Fender | dlynes_home : 5 channels mac. |
| 19:29.31 | blitzrage | neat |
| 19:29.45 | Assid | not max.. .011 /min |
| 19:29.49 | Assid | unlimitel huh |
| 19:29.49 | [TK]D-Fender | ~[TK]D-Fender |
| 19:30.00 | [TK]D-Fender | Jbot Cloaking enabled! ;) |
| 19:30.25 | adorah | r u talking about Cablevision 500 min for 19.95$/month? |
| 19:30.46 | Assid | aargh.. can a sip provisioning cause a problem if you arent updating the bootrom.. and the phone reboots in the middle? |
| 19:30.53 | Assid | im having CRAZY power problems |
| 19:31.21 | [TK]D-Fender | Assid : yes SIP updates take 2 reboots |
| 19:31.32 | [TK]D-Fender | Assid : Verify, then save. |
| 19:31.43 | Loceur | how do I pick a handset for my spanking new asterisk server? What's ya'lls favorite phones? |
| 19:31.53 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 19:32.09 | r_evolution | my favorite method is to grab a dartboard loceur... |
| 19:32.12 | brettnem | Loceur: I've used SNOM, Polycom and Cisco.. I like the Cisco the most.. but SNOM is pretty flexible.. |
| 19:32.13 | r_evolution | pin a few company logos up |
| 19:32.16 | r_evolution | and throw darts :) |
| 19:32.20 | Assid | okay.. its actually updated.. just that its rebooting with this power problems |
| 19:32.30 | Assid | really dont knwo what to do besides unplug and sit |
| 19:32.33 | r_evolution | TK will threaten your life if you get anything other than Polycom though |
| 19:32.49 | mercestes | Polycom is nice...if you can make it FREAKING WORK.......it's hell getting it to work for the first time. |
| 19:32.53 | brettnem | I'd stick with one of those three |
| 19:33.00 | brettnem | mercestes: bah, it ain't that bad |
| 19:33.03 | mercestes | but after that it's ok.......unless you ever have to reboot....that takes a week. |
| 19:33.15 | brettnem | rebooting the polycom phones is damn slow |
| 19:33.26 | mercestes | my grandma boots faster than a polycom |
| 19:33.30 | brettnem | I have two polycoms on my desk now.. they are nice phones.. no doubt |
| 19:33.44 | Assid | mercestes: actually no.. i love it. gotta thank [TK]D-Fender tho |
| 19:33.47 | r_evolution | heh... polycom's greatest feature... the web interface only takes a blazing 5 minutes to be active |
| 19:33.52 | r_evolution | after the phone boots |
| 19:34.05 | brettnem | and any small change.. BAMO reboot |
| 19:34.07 | [TK]D-Fender | mercestes : Dunno about that. Always worked right off for me.... |
| 19:34.09 | Loceur | thanks guys |
| 19:34.11 | brettnem | 5 minutes.. cup of coffee |
| 19:34.15 | blitzrage | Assid: http://www.polycom.com/resource_center/0,1454,pw-26-482-10533,FF.html |
| 19:34.26 | mercestes | Polycom boards: Bug: "It takes a full 2 minutes before the web interface of a polycom phone is accessible after a successful boot." Solution: "Make the boot last 2 minutes longer." |
| 19:34.29 | [TK]D-Fender | brettnem : But unlike Snom's you don't have to reboot them HOURLY ;) |
| 19:34.32 | brettnem | but I have hundreds of polycoms installed.. most people seem to really like them |
| 19:34.37 | brettnem | [TK]D-Fender: I don't have that problem! |
| 19:35.10 | blitzrage | Polycom hasn't updated their documents page -- latest changlog is 1.6.3 it looks like |
| 19:35.12 | mercestes | but yea, once a polycom is working and ON....it's rock solid... |
| 19:35.13 | Assid | blitzrage: which one ? |
| 19:35.17 | brettnem | SNOMs allow you to specifiy in DHCP a HTTP location in the TFTP parameter for the config fils.. that is SWEET |
| 19:35.20 | Assid | i have the admin guide |
| 19:35.25 | blitzrage | "I'm breathing so I guess I'm still alive, even thought signs seem to tell me otherwise" |
| 19:35.30 | Assid | dont see anything in there with words "changelog" |
| 19:35.45 | brettnem | I saw the polycom changelog the other day.. I'm sure of it |
| 19:35.47 | blitzrage | nevermind -- I didn't read careuflly enough |
| 19:36.12 | blitzrage | I thought you're looking for "Release Notes" |
| 19:36.22 | mercestes | Cisco's are ok too.....dealing with all the bootroms on a Cisco can be painful tho |
| 19:36.50 | [TK]D-Fender | brettnem : Can do tahtwith Polycom's as well.... |
| 19:37.03 | brettnem | [TK]D-Fender: no, with polycom you have to point to a TFTP server. |
| 19:37.14 | [TK]D-Fender | Cisco's a re great phones... SIP isn't always stable and its DEFINATELY tricky, and expensive, but YGWYPF |
| 19:37.27 | Assid | err.. where is the serial number located on the phone? |
| 19:37.30 | [TK]D-Fender | brettnem : So entirely not true :) |
| 19:37.34 | Assid | i mean below the mac addresS? |
| 19:37.37 | mercestes | Grandstreams are the absolute best!!!!! (door stoppers) |
| 19:37.42 | tzanger | mercestes: heh |
| 19:37.50 | brettnem | [TK]D-Fender: you can make polycom's boot a config off a HTTP server?? |
| 19:37.52 | tzanger | grandstreams are great residential low call volume phones |
| 19:37.52 | MstlyHrmls | brettnem: nope, with 3.x you can give it a URL to the <mac>.cfg file |
| 19:38.03 | brettnem | MstlyHrmls: Very cool.. glad to be wrong. :) |
| 19:38.03 | tzanger | brettnem: yes, with latest bootroms and firmware (I have not tried this) |
| 19:38.09 | Assid | white sticker or black one? |
| 19:38.13 | [TK]D-Fender | brettnem : FTP, TFTP, HTTP, HTTPS, jsut about anything since BootROM 3 |
| 19:38.18 | brettnem | it still takes 10 minutes to boot tho? |
| 19:38.24 | tzanger | brettnem: yes :-( |
| 19:38.24 | [TK]D-Fender | brettnem : 2 |
| 19:38.36 | r_evolution | debating w/ TK in regards to polycom != wise |
| 19:39.07 | brettnem | hmm.. what happened to the udev permissions in FC4? |
| 19:39.08 | [TK]D-Fender | r_evolution : only because I'm RIGHT :) I work with them, I use them at home, and I consult them. Experience counts. |
| 19:39.38 | brettnem | [TK]D-Fender: i use them alot too.. they are dependable. I like the phone alot.. the config file totally SUCKS |
| 19:39.41 | [TK]D-Fender | And then I'm going to cert for either distribution or support. Depending aon whats profitable for my consultancy. |
| 19:39.41 | r_evolution | actually i was going for the only because you hunt dissenters down and kill them |
| 19:40.01 | mercestes | The polycom config file is written in total klingon. |
| 19:40.01 | blitzrage | I'd like to program something that provisions and creates the .XML files... |
| 19:40.14 | [TK]D-Fender | brettnem : I'd rephrase that as "sure its not pretty", but I get what needs done done in no time and with certainty. |
| 19:40.15 | brettnem | The only real beef I have with them is the boot time.. if you make a program to automatically make the configs, it no big deal |
| 19:40.15 | blitzrage | but then I'd have to get good at programming :) |
| 19:40.23 | CunningPike | blitzrage: I have some shell scripts...... |
| 19:40.42 | [TK]D-Fender | Mine are ready to go on the server from before I even take a new one out of its box. |
| 19:40.52 | MstlyHrmls | mercestes: XML, klingon, very similar :-) |
| 19:41.00 | [TK]D-Fender | blitzrage : No, you could script it with 4 lines of SED. period |
| 19:41.09 | brettnem | [TK]D-Fender: same here.. we have a web gui that makes the polycom files.. but I still don't like it.. |
| 19:41.09 | blitzrage | pffft -- look at the XML for i-911 -- that'll make you go nuts |
| 19:41.26 | brettnem | blitzrage: I've looked at that.. nasty.. |
| 19:41.30 | [TK]D-Fender | brettnem : I always provision mine by hand however. |
| 19:41.33 | blitzrage | brettnem: fun stuff eh? :D |
| 19:41.39 | C4T3l | ive been wanting to write a sed script for a poly config do you have an ex? |
| 19:42.07 | [TK]D-Fender | C4T3l : if you know how to use SED, just make a template and search & replace! |
| 19:42.19 | [TK]D-Fender | C4T3l : and NO, I have no need for such things. |
| 19:42.21 | C4T3l | i need to learn sed |
| 19:42.32 | brettnem | just do everything in PERL |
| 19:42.39 | brettnem | I mean.. really |
| 19:42.55 | mercestes | Nah...VB.Net |
| 19:43.00 | mercestes | *ducks* |
| 19:43.14 | mercestes | *saw that one coming* |
| 19:43.17 | C4T3l | my company has a bad habit of ordering fones last minute and having to rush-config |
| 19:43.44 | [TK]D-Fender | Well any phone I get here takes me maybe 2 minutes to prep.... no biggie. |
| 19:43.48 | C4T3l | i havent been here for too long so i'm trying to make it easy on myself |
| 19:43.52 | brettnem | I've written config generators for both cisco and SNOM from sip.conf.. I don't know why there isn't something out there that does this stuff. |
| 19:44.25 | [TK]D-Fender | brettnem : For poly its harder because of XML's free-form nature and lack of line breaks. |
| 19:44.27 | brettnem | I use SETVARs and place the MAC address in sip.conf along with a LINE=1 or LINE=2.. No dummy proofing, but it works |
| 19:44.38 | brettnem | [TK]D-Fender: s/// |
| 19:45.09 | C4T3l | hmm. very intersting |
| 19:45.15 | *** join/#asterisk mozveren_1 (n=mozveren@checkphone-174-194.cnt.nerim.net) |
| 19:45.18 | mozveren_1 | hello all |
| 19:45.28 | brettnem | yeah, I put all sorts of junk in those setvars.. I put flash operator panel labels |
| 19:45.34 | mozveren_1 | i have problem with a A102 sangoma card |
| 19:45.35 | brettnem | and if they should appear on the FOP |
| 19:45.39 | mozveren_1 | doesn't |
| 19:45.42 | mozveren_1 | work |
| 19:45.48 | mozveren_1 | I have launch wancfg |
| 19:45.56 | mozveren_1 | enter my configuration |
| 19:46.08 | mozveren_1 | but after when I start wanrouter, I have some troubles |
| 19:46.15 | mozveren_1 | anyone can help m ? |
| 19:46.59 | Assid | how the hell d you register our product with polycom? |
| 19:47.13 | Assid | it says the serial number is wrong |
| 19:47.48 | [TK]D-Fender | Assid : Serial # is the MAC <- |
| 19:48.05 | [TK]D-Fender | Fish for all! |
| 19:48.05 | Assid | thats what i tried |
| 19:48.09 | r_evolution | you're done with your fish TK |
| 19:48.15 | mercestes | *hides and tries to be non-descript. |
| 19:48.59 | Assid | keeps saying "please validate and re-enter serial number again (2501,40) |
| 19:49.23 | [TK]D-Fender | Assid : PM me what you're entering |
| 19:51.19 | dlynes_home | [TK]D-Fender: 0.011/mi even if all calls are redirected to my asterisk server? |
| 19:52.15 | r_evolution | dlynes : about that termination provider? |
| 19:52.21 | dlynes_home | r_evolution: ? |
| 19:52.42 | r_evolution | i was just curious to know who you were using... you mentioned how good they were earlier |
| 19:52.45 | [TK]D-Fender | dlynes_home : Where else would they be going? |
| 19:52.59 | r_evolution | or so i thought... i wasn't paying full attention |
| 19:53.18 | dlynes_home | [TK]D-Fender: from their did, call forwarding to a cell phone in 778 NPA? |
| 19:53.36 | [TK]D-Fender | dlynes_home : AH |
| 19:53.41 | dlynes_home | r_evolution: I don't remember ever mentioning they were good |
| 19:53.48 | dlynes_home | r_evolution: i just said they were cheap |
| 19:53.49 | [TK]D-Fender | dlynes_home : that'd be double-rate. |
| 19:53.58 | [TK]D-Fender | dlynes_home : 2 channels don't forget. |
| 19:54.12 | dlynes_home | [TK]D-Fender: why do i need two channels, though? |
| 19:54.22 | *** part/#asterisk meppl (i=mephisto@meppl.net) |
| 19:54.24 | r_evolution | oh... well how is the quality then? good? fair? pathetic? |
| 19:54.36 | [TK]D-Fender | dlynes_home : One for the call to come in on, the other to dial and bridge to cell of course |
| 19:54.43 | dlynes_home | r_evolution: probably fair, unless it's a direct route |
| 19:55.31 | dlynes_home | [TK]D-Fender: so one zap channel and one sip or iax channel |
| 19:55.56 | r_evolution | worth looking into. rates? |
| 19:56.09 | dlynes_home | [TK]D-Fender: iow, it'd be pretty stupid to only have one channel |
| 19:56.26 | dlynes_home | r_evolution: are you a reseller? |
| 19:56.53 | r_evolution | nah. we provide service |
| 19:57.06 | dlynes_home | so you're a reseller then, or an interconnect |
| 19:57.11 | dlynes_home | right? |
| 19:57.39 | [TK]D-Fender | dlynes_home : Well if you want them to terminate a DID direct to Cell, no need for SIP. If you want to do the termination then just 1-way with them, and SIP/IAX, yor choice |
| 19:58.01 | [TK]D-Fender | dlynes_home : I'm an informed consumer, and consultant :) |
| 19:58.09 | dlynes_home | lol |
| 19:58.17 | r_evolution | technically... but when you say reseller i think wholesale |
| 19:58.27 | dlynes_home | and so do i |
| 19:58.41 | r_evolution | but yes we provide service to consumers... and we also have an in-house pbx that i'm in the process of getting up... |
| 19:58.49 | r_evolution | but the wiring here is... a mess to say the least |
| 19:59.00 | dlynes_home | so you provide service to businesses or something then? |
| 19:59.10 | dlynes_home | i.e. long distance service? |
| 19:59.10 | r_evolution | business and residential |
| 19:59.40 | r_evolution | well yes we also provide ld service on the pstn... |
| 19:59.43 | dlynes_home | so, iow, you're a reseller |
| 19:59.56 | r_evolution | but we also provide voip service |
| 20:00.02 | dlynes_home | i.e. you're a VAR |
| 20:00.05 | r_evolution | so yes... we resell service to consumers |
| 20:00.26 | r_evolution | why do you ask? |
| 20:00.36 | SpaceBass | Anyone with a DATA t1 have a sense of what codec to use to save bandwidth? |
| 20:00.50 | r_evolution | 729 :) |
| 20:00.52 | tzanger | SpaceBass: if all youw ant to do is save bandwidth, use a heavily compressed codec |
| 20:01.02 | tzanger | lpc10, g726, g729, gsm |
| 20:01.05 | dlynes_home | r_evolution: because I don't want my wholesalers getting pissed off at me for sending them a bunch of weenies looking for a $20/mo all you can eat plan |
| 20:01.13 | tzanger | (in increasing order of audio quality) |
| 20:01.29 | SpaceBass | tzanger, i guess it wasnt a good question...trying to determine if 1024 is enough bandwidth to reasonably do VoIP for home and still have room for data |
| 20:01.41 | [TK]D-Fender | tzanger : G726 is higher than ALL the rest... |
| 20:01.44 | SpaceBass | i know on paper it should be enough for several concurrent calls, but.... |
| 20:01.47 | r_evolution | not quite dlynes :) |
| 20:01.54 | [TK]D-Fender | tzanger : its 1/2 ULAW.. cmon... |
| 20:01.56 | tzanger | ack you're right |
| 20:02.11 | [TK]D-Fender | tzafrir : At least in the capacity * has implemented... |
| 20:03.06 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 20:14.02 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:15.11 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 20:16.44 | SpaceBass | Stupid fedex man won't bring my WIP330 phone and I need to run errands |
| 20:21.25 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 20:21.36 | MattH | Hi.. I'm running 1.2.6 can anyone tell me why *8 pickup might not be working for me? |
| 20:23.41 | [TK]D-Fender | MattH : Improper setup in features.conf, pickup group improperly defined in each tech's setup.... |
| 20:24.28 | MattH | yeah it says "nothing to pickup" |
| 20:24.56 | [TK]D-Fender | MattH : well I guess you'd better check the settings of whatever it was that you were expecting to be able to pick up. |
| 20:25.24 | MattH | so if I'm trying to pickup an extension (sip phone) what exactly needs to be in the config? that's probably what I'm missing |
| 20:25.56 | [TK]D-Fender | your target has to be identified. |
| 20:26.04 | MattH | k |
| 20:26.06 | MattH | http://www.voip-info.org/tiki-index.php?page=Asterisk+callgroups+and+pickupgroups |
| 20:26.09 | MattH | this will probably answer my question |
| 20:26.44 | [TK]D-Fender | TFM usually does ;) |
| 20:26.52 | *** join/#asterisk loonacy (n=loonacy@24-117-254-250.cpe.cableone.net) |
| 20:26.54 | mercestes | LOL |
| 20:27.05 | MattH | yup there we go.. missing my call group :) |
| 20:28.36 | MattH | AHA! there we go |
| 20:29.13 | loonacy | Hello, i'm trying to make a macro that will take a 7 digit number, add an area code, and either dial a SIP number if there's an account on that server, or dial out an IAX trunk if that extension doesn't exist.. How do i tell if an extension exists? |
| 20:29.32 | loonacy | I tried Goto() but if the extension doesn't exist it just complains there's no invalid handler. |
| 20:31.45 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:32.12 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 20:34.38 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 20:35.00 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 20:37.54 | XanaXa | hey guys can someone tell me if I need to enable custom_extensions for them to work? Or is there a way to see if they are enabled? |
| 20:38.47 | Tier_1 | are you using asterisk or freepbx |
| 20:39.01 | Tier_1 | if freepbx join #freepbx |
| 20:39.12 | XanaXa | ok thanks |
| 20:39.27 | Tier_1 | II have abounty for a agi |
| 20:39.33 | Tier_1 | any agi guys here |
| 20:41.37 | loonacy | Can anybody point me in the right direction for handling a Goto when the extension doesn't exist? I'd like to be able to continue the dialplan instead of it complain about no invalid handler. |
| 20:41.47 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:43.13 | *** part/#asterisk Loceur (n=noneya@vsas.veedix.com) |
| 20:43.16 | *** join/#asterisk Obitus (n=erik@212.17.129.15) |
| 20:43.37 | r_evolution | why not handle that GoTo in the i extension loonacy? |
| 20:43.47 | r_evolution | i.e. exten => i,1,GoTo(etcetcetc) |
| 20:45.03 | mercestes | There just isn't a syntax for "If this then do this" in extensions.conf is there? |
| 20:45.16 | Dr-Linux | /usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never defined |
| 20:45.16 | Dr-Linux | make: *** [zaptel.o] Error 1 |
| 20:45.16 | Dr-Linux | [root@ivr zaptel-1.2.5]# |
| 20:45.16 | lzhang | anybody else have been having problems compiling asterisk-addons svn trunk? |
| 20:45.23 | Dr-Linux | what could be the reason? |
| 20:45.57 | Mother | greetings |
| 20:46.07 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:46.11 | mercestes | I mean..I could use GotoIf but man that's fargin messy for 20 checks |
| 20:46.18 | loonacy | r_evolution: I have : exten => s,2,GoTo(${ARG1},${ARG2},1) <-- in s extension, I never get to i extension. |
| 20:46.30 | SuperLag | For call queues... I'm reading the docs on the voip-info.org site and I've got a question. "Fewest Completed Calls".... how is this calculated? does it reset every 24 hours? |
| 20:46.41 | Dr-Linux | i have installed Libpri packege, but having this error while installing zaptel |
| 20:46.44 | Tier_1 | any one here good with agi |
| 20:47.11 | SuperLag | It says, "fewestcalls: ring the one with the fewest completed calls from this queue", but it doesn't say when/how it calculates that value |
| 20:47.23 | SuperLag | Any ideas? |
| 20:47.27 | mercestes | what_agi? php_agi perl_agi bash_agi c_agi |
| 20:48.06 | Tier_1 | wll what ever its best done it |
| 20:48.21 | Tier_1 | I was thinking perl or php |
| 20:48.38 | mercestes | it's your choice. |
| 20:48.46 | Tier_1 | it for asterisk |
| 20:48.51 | mercestes | it all reads stdin and stdout |
| 20:49.21 | r_evolution | loonacy... you must be getting to the i extensions somewhere if * is complaining about no invalid extension handler |
| 20:49.23 | Tier_1 | nerc pvt me I will explain as to not hog the channel |
| 20:50.10 | Tier_1 | I have a exten I want to convert to a agi |
| 20:51.37 | Obitus | Anyone here tried the Nokia N80 SIP/softphone with Asterisk? |
| 20:52.32 | *** part/#asterisk chino[server] (n=daquino@e82-103-128-114s.easyspeedy.com) |
| 20:52.53 | loonacy | r_evolution: Yeah, I just figured it out.. I was trying to use i in the current context, but now i realize it was going to i in the new context i was going to. |
| 20:54.30 | r_evolution | :) |
| 20:54.41 | r_evolution | isn't it odd? sometimes just talking about a problem helps you figure out the answer |
| 20:55.20 | loonacy | Now i just have to figure out how to handle it if they dial an area code i don't have a context for. |
| 20:56.09 | CYPRESS_A | Ok folks got one that is driving me bannanas: Can the background application utilize a local variable. I got it working with a asterisk defined variable, but not a user/local defined variable. I have a pastebin at http://cpp.enisoc.com/pastebin/6963 I appreciate anyones help. |
| 20:59.56 | Tier_1 | ///merc ou there |
| 21:00.09 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:01.32 | lzhang | how do I get format_mp3 to not compile in asterisk-addons? |
| 21:02.09 | Tier_1 | edit the makefile comment it out |
| 21:02.10 | [TK]D-Fender | lzhang : Whats wrong with it compiling there? |
| 21:03.05 | r_evolution | hey loonacy |
| 21:03.17 | lzhang | I'm trying to compile latest trunk version, but it seems to choke on format_mp3 |
| 21:03.26 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 21:03.52 | lzhang | http://pastebin.com/727222 here's what it looks like |
| 21:04.53 | wunderkin | why do you even bother with format_mp3 |
| 21:05.13 | lzhang | wunderkin: I'm trying to not compile it |
| 21:05.27 | wunderkin | oh |
| 21:05.45 | *** kick/#asterisk [CYPRESS_A!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted[asteria] (DO NOT PRIVMSG FLOOD) |
| 21:05.48 | wunderkin | i've never used asterisk-addons |
| 21:05.59 | *** mode/#asterisk [-o twisted[asteria]] by twisted[asteria] |
| 21:06.04 | lzhang | I need the res_mysql stuff |
| 21:06.28 | *** join/#asterisk CYPRESS_A (n=nate@216-230-88-10.client.cypresscom.net) |
| 21:06.31 | wunderkin | /msg twisted[asteria] like; /msg twisted[asteria] o; /msg twisted[asteria] m; /msg twisted[asteria] g |
| 21:07.13 | CYPRESS_A | anyone an expert on background()? |
| 21:07.37 | lzhang | ok it looks like I'm getting strings.h problems with other stuff in addons too, commenting out the Makefile part for format_mp3 yields other errors when compiling app_saycountpl |
| 21:07.43 | wunderkin | let us put on our rocket scientist hats |
| 21:08.23 | wunderkin | CYPRESS_A, try asking your question |
| 21:09.43 | *** join/#asterisk jtoy (n=toy@cust-206-40-173-219.bos-static.gis.net) |
| 21:09.52 | *** part/#asterisk jtoy (n=toy@cust-206-40-173-219.bos-static.gis.net) |
| 21:09.55 | CYPRESS_A | Can the background application utilize a local variable. I got it working with a asterisk defined variable, but not a user/local defined variable. I have a pastebin at http://cpp.enisoc.com/pastebin/6963 |
| 21:10.54 | [TK]D-Fender | CYPRESS_A : Please note the extra spaces in the exectution of your "setglobalvar" |
| 21:11.34 | *** join/#asterisk watchy2 (n=watchy@70.238.57.237) |
| 21:11.45 | watchy2 | is there a dial command to ring every phone? |
| 21:11.52 | CYPRESS_A | exten => s,1,SetGlobalVar(CUSTOMER = CYPRESS); |
| 21:11.57 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 21:12.00 | [TK]D-Fender | watchy : Its called "Dial" |
| 21:12.05 | CYPRESS_A | around the "=" ? |
| 21:12.10 | [TK]D-Fender | CYPRESS_A : Look at what it EXECUTES as. |
| 21:12.23 | watchy2 | tk: i want to dial EVERY phone using a panic extension |
| 21:12.27 | CYPRESS_A | correct " // " |
| 21:12.31 | watchy2 | i sold a system to a .gov contractor who makes big explosives |
| 21:12.31 | r_evolution | holy crap watchy. |
| 21:12.46 | [TK]D-Fender | CYPRESS_A : remove your extra spaces |
| 21:12.46 | watchy2 | if something happens they want to beable to dial say 666 and it ring all phones |
| 21:13.08 | watchy2 | so they can get someone to answer right away |
| 21:13.08 | BadPacket | I have an OC-48 in my pants |
| 21:13.09 | [TK]D-Fender | watchy : Dial is for you! |
| 21:13.15 | lzhang | watch2: try the wiki and check out allpage.agi, it intercoms all the phones |
| 21:13.29 | watchy2 | yea i know i need dial fender |
| 21:13.39 | watchy2 | but i doubt Dial(AllPhones) is gonna work |
| 21:13.48 | CunningPike | ~seen stack_ |
| 21:13.50 | jbot | stack_ is currently on #asterisk (3h 2m 7s). Has said a total of 14 messages. Is idling for 2h 2m 39s, last said: 'blitzrage: it seems as though everything coming in on the TDM400 is not getting caller id info..., works fine on the TE110p'. |
| 21:13.53 | watchy2 | i was just curious about anymore info on it |
| 21:14.10 | watchy2 | but ill check allpage and dial on the wiki |
| 21:14.16 | lzhang | allpage is what you need |
| 21:14.17 | BadPacket | ~seen docelm0 |
| 21:14.19 | jbot | docelm0 <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 20h 47m 10s ago, saying: 'BUT for someone who IS already registered I will give em $2 credit for testing.. :)'. |
| 21:14.40 | watchy2 | lz: it just intercoms all phones so they can yell HELP WERE GONNA DIE i guess |
| 21:14.52 | watchy2 | which is what we need |
| 21:15.15 | lzhang | yeah you set up the phones to autoanswer so the phones don't even have to be picked up |
| 21:15.36 | watchy2 | but it will only auto answer on that one extension correct |
| 21:15.52 | watchy2 | if that say 666 extension is dialed |
| 21:15.52 | lzhang | it will autoanswer on every phone you set it up with |
| 21:15.55 | watchy2 | otherwise its all normal |
| 21:16.06 | lzhang | correct |
| 21:16.14 | lzhang | only for when allpage is called |
| 21:16.21 | watchy2 | thats what we want for sure |
| 21:16.33 | CYPRESS_A | D-Fender: Thanks a million, been bugging me for for a couple of days. |
| 21:16.50 | [TK]D-Fender | CYPRESS_A : NEXT!!!! (c) BKW |
| 21:16.54 | watchy2 | Dfender is awesome. |
| 21:17.00 | watchy2 | he helped me alot lastnight |
| 21:17.07 | watchy2 | lzhang: got any exp with fxotune? |
| 21:17.16 | lzhang | no idea what you're talking about :) |
| 21:17.29 | watchy2 | its to tune tdm interfaces for echo |
| 21:17.34 | lzhang | any experience compiling latest asterisk-addons? |
| 21:17.49 | *** join/#asterisk dynamicpulse (n=0a0800f9@12.208.56.217) |
| 21:17.56 | watchy2 | make don't work? |
| 21:18.03 | lzhang | nope |
| 21:18.31 | watchy2 | whats it say |
| 21:19.06 | Tier_1 | where do you set the timezone for voicemail playback so it plays the proper tiime for thier timezone |
| 21:19.35 | Tier_1 | insted of people in NY getting time statements in Mountin time |
| 21:24.51 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 21:25.19 | lzhang | watchy: http://pastebin.com/727222 |
| 21:25.24 | lzhang | watchy |
| 21:25.43 | lzhang | : also when I try to not compile format_mp3, it fails for string.h stuff on the other modules too |
| 21:26.43 | watchy2 | hmm |
| 21:26.51 | watchy2 | you try make clean and then make? |
| 21:26.54 | watchy2 | what os you using? |
| 21:27.01 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 21:27.03 | lzhang | debian sarge |
| 21:27.14 | lzhang | yeah, I've make cleaned and maked a few times already |
| 21:27.34 | SpaceBass | I have a device that wants to know SIP proxy / port and registrar address and port....shouldnt both proxy and address by my Asterisk box and port 5060? |
| 21:28.01 | Strom_C | SpaceBass: yes |
| 21:28.17 | SpaceBass | cant get the darned thing to register |
| 21:28.32 | watchy2 | you got the newest asterisk installed? |
| 21:28.38 | Strom_C | SpaceBass: is there NAT involved? |
| 21:29.01 | lzhang | watchy2: yeah, I svned and compiled trunk for asterisk, zaptel, libpri --- it works |
| 21:29.36 | watchy2 | strange |
| 21:29.47 | watchy2 | i'll try to compile it. maybe its broke |
| 21:30.12 | lzhang | watchy2: cool, thanks! |
| 21:30.21 | nextime | what's the best ( best = most stable ) ss7 support in *? |
| 21:30.55 | watchy2 | 1.2.2 lzhang? |
| 21:31.20 | lzhang | svn trunk... svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 21:31.30 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 21:31.32 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 21:33.22 | SpaceBass | Strom_C, its working now...new WIP330 phone...has a web interface but it apparently doesnt include all the settings |
| 21:33.29 | SpaceBass | had to set some stuff in the phone itself |
| 21:34.11 | Strom_C | SpaceBass: ah. |
| 21:34.39 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 21:35.18 | watchy2 | <PROTECTED> |
| 21:36.04 | lzhang | watchy2: zaptel, libpri, asterisk, asterisk-addons? |
| 21:36.16 | lzhang | that was quick |
| 21:36.17 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 21:37.12 | watchy2 | yea |
| 21:37.17 | watchy2 | i had them dloaded already |
| 21:37.25 | watchy2 | on a fast box |
| 21:37.28 | lzhang | damn must be something wrong with my box |
| 21:37.32 | *** part/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 21:37.37 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 21:42.41 | *** join/#asterisk file (n=file@216.237.114.82) |
| 21:43.44 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 21:47.03 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:47.07 | *** part/#asterisk unmanagedwork (n=unmanage@64.89.118.139) |
| 21:49.03 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:51.52 | RoyK | <PROTECTED> |
| 21:53.48 | [Airwolf] | I'm trying to create a macro, so I can give users both a SIP and IAX account. But I want to check if the channel is availible first. Now there is a ChanIsAvail function, but I don't really know how to implement it for my situation. |
| 21:54.02 | [Airwolf] | Does anyone have some experience with this ? |
| 21:54.31 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 21:54.39 | [TK]D-Fender | <PROTECTED> |
| 21:55.38 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:56.51 | [Airwolf] | [TK]D-Fender, that works. The problem is that FOP doesn't work anymore then. |
| 21:57.09 | [Airwolf] | And besides, it creates an error. And I don't like errors. :P |
| 21:57.30 | XanaXa | anyone here have a good tutorial on using mp3's for music on hold |
| 21:59.18 | *** join/#asterisk kapsel (i=kapsel@irc.thinkgeek.dk) |
| 22:00.18 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 22:00.25 | kapsel | I have just setup an Asterisk server that we got more or less pre-configged from a company, it worked for a while, but now we can't dial local extension numbers, it just hangs up and leaves a "someone called you" message on the number you call any ideas to what could have caused this? |
| 22:00.49 | dlynes_home | kapsel: have you tried rebooting your phones? |
| 22:00.51 | kapsel | we have 20 extension numbers and 20 phones with to-ip adapters, everything works "except for that" |
| 22:01.19 | kapsel | dlynes_home: i have not, no. |
| 22:01.24 | dlynes_home | kapsel: try it |
| 22:01.34 | *** join/#asterisk hayburn (i=chiaborg@concorde.hayburn.net) |
| 22:02.52 | dlynes_home | OMG!!!! |
| 22:02.58 | dlynes_home | they're coming out with a new rocky movie? |
| 22:03.01 | watchy2 | iyea |
| 22:03.02 | watchy2 | yea |
| 22:03.03 | watchy2 | they are |
| 22:03.08 | dlynes_home | That's funny |
| 22:03.15 | dlynes_home | Like there wasn't enough already? |
| 22:03.26 | *** part/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 22:03.42 | RoyK | omfg. rocky..... |
| 22:03.42 | dlynes_home | Still gonna be sly stone? |
| 22:03.47 | [TK]D-Fender | dlynes_home : old news, as is the plans for Rambo 4 |
| 22:03.53 | kapsel | dlynes_home: hmm, strange. it did not help, i think ill have to check my adapter configs again. this is one huge mess thou, check out this picture of my setup right now: http://gallery.kapsel.dk/computerting/19052006158.jpg.html?g2_imageViewsIndex=1 - its 10 adapters, one ip phone, and 20 wireless phone bases |
| 22:03.54 | RoyK | rotfl |
| 22:03.58 | [TK]D-Fender | dlynes_home : yup for both |
| 22:04.03 | dlynes_home | dood |
| 22:04.05 | dlynes_home | that's messed up |
| 22:04.32 | dlynes_home | kapsel: tried rebooting the router, too? |
| 22:04.49 | kapsel | dlynes_home: the router? |
| 22:05.03 | dlynes_home | kapsel: wtf????? |
| 22:05.13 | RoyK | [TK]D-Fender: heh. the guy is like 60 |
| 22:05.14 | dlynes_home | kapsel: Did I just step into the twilight zone? |
| 22:05.19 | *** join/#asterisk serg_b (n=serg_b@9i.ru) |
| 22:05.45 | dlynes_home | kapsel: your phone system looks like something the cat dragged in |
| 22:05.56 | kapsel | hehe, its an crappy image, but its just temp while we're testing it |
| 22:06.08 | kapsel | has to be moved somewhere else and setup with better cabling |
| 22:06.12 | dlynes_home | ah |
| 22:06.15 | RoyK | rotfl. if the stuff about rocky is right, it's not just _with_ stallone, but _by_ stallone as well :) http://en.wikipedia.org/wiki/Rocky_Balboa |
| 22:06.17 | dlynes_home | anyways...try rebooting the router |
| 22:06.25 | kapsel | but having adapters and wireless phone bases gives alot of cables, really. |
| 22:06.40 | dlynes_home | kapsel: yeah, but anything involving wireless is going to be a huge headache |
| 22:06.50 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:07.10 | kapsel | dlynes_home: nah, wireless is no problem. all the bases are configged to their phones etc. from their older setup, its no big deal at all actually. |
| 22:07.40 | dlynes_home | kapsel: yeah...i've never had a pain free wireless setup |
| 22:07.47 | dlynes_home | kapsel: there's always at least one problem |
| 22:07.50 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:07.56 | dlynes_home | kapsel: but there's always at least one windows machine, too |
| 22:09.25 | kapsel | as long everything is working next monday |
| 22:09.38 | kapsel | we're using isdn for inbound calls |
| 22:09.46 | kapsel | = huge problems right now |
| 22:09.52 | dlynes_home | fun fun fun |
| 22:10.02 | dlynes_home | i love solving other people's problems |
| 22:10.04 | dlynes_home | not |
| 22:10.13 | kapsel | im not asking you to solve my problems. |
| 22:10.27 | dlynes_home | wasn't suggesting you were |
| 22:10.34 | dlynes_home | but you get to solve someone else's problems :0 |
| 22:11.43 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 22:12.08 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 22:12.22 | Z_God | I'm getting this with asterisk 1.2: |
| 22:12.22 | Z_God | chan_modem.c:833 modem_request: This channel driver is deprecated. Please see the UPGRADE.txt file. |
| 22:12.29 | FarrisG | The docs aren't clear, does AsterFax require the use of AMP? |
| 22:12.34 | Z_God | there isn't any info about a proper update though |
| 22:12.58 | Z_God | only some stuff about mISDN which doesn't seem to support much hardware |
| 22:14.30 | *** join/#asterisk Kokey (n=jramirez@201.133.218.194) |
| 22:15.32 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:15.54 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.172) |
| 22:16.42 | tdonahue-laptop | anyone here use the isc dhcp server know what the option name is for the tftp servers that polycom looks for? |
| 22:17.55 | [TK]D-Fender | tdonahue-laptop : 66 |
| 22:17.56 | dlynes_home | tdonahue-laptop: explain what the difference is between the tftp servers that polycom looks for and the tftp servers that any other device looks for? |
| 22:19.03 | tdonahue-laptop | i don't know which option it was looking for |
| 22:19.09 | *** join/#asterisk ghost99 (n=neville@222-153-92-225.jetstream.xtra.co.nz) |
| 22:21.09 | tdonahue-laptop | there we go... tftp-server-name is equivilent to option 66 |
| 22:22.49 | *** join/#asterisk adker (n=adker@74-33-211-200.br1.glv.ny.frontiernet.net) |
| 22:25.15 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:25.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:25.45 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:25.57 | ghost99 | [tk]-Defender .. you awake :) ? |
| 22:28.15 | jart | While working with Asterisk developers, I created a patch that utilized RSA technology to encrypt VoIP conversations that could be transfered across the internet and not be detected by the National Security Agency. |
| 22:28.20 | jart | Before posting my patch on the internet, I wrote Mark Spencer a letter notifying him of my intentions to defect to the FreeSwitch project |
| 22:28.29 | jart | After the patch had been posted, Digium deployed the entire Asterisk development team to DDOS the servers distributing the patch. Digium then contacted FreeSwitch intelligence telling them that the purpose of my patch was to put a rootkit in the FreeSwitch codebase. |
| 22:28.39 | jart | With conflicts heated and the open source telephony world at the brink of full scale war, a cunning FreeSwitch developer discovered my intentions to defect and the patch was surreptitiously merged in to the FreeSwitch tree. |
| 22:29.05 | jart | and everyone lived hapily ever after |
| 22:29.26 | file | w, t, f |
| 22:30.28 | FarrisG | Are there any decent guides to safely upgrading Asterisk? I'm running a very old version (CVS-v1-0-01/15/05-19:47:01) and would would like to get current. |
| 22:30.38 | Qwell[] | FarrisG: there is an upgrade.txt, or something |
| 22:30.46 | *** join/#asterisk Kokey (n=jramirez@201.133.218.194) |
| 22:31.00 | FarrisG | Qwell[]: In the cvs repo? |
| 22:31.15 | Qwell[] | FarrisG: iirc |
| 22:31.16 | jart | file: it's a spoof of the Hunt For Red October plot |
| 22:31.53 | file | interesting |
| 22:34.39 | De_Mon | how can I determine which device created a channel? like SIP/user1 or ZAP/5 ... what two variables do I need? |
| 22:35.49 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:36.22 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:37.34 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 22:37.50 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 22:38.06 | ringhals | hey everyone |
| 22:38.40 | ringhals | I ahve a problem that I can't seem to wrap my head arround |
| 22:39.07 | ringhals | I am working on a remote call center type applicaton and am going to use an iax soft phone |
| 22:40.07 | ringhals | the problem I am having is that when I dial an extension (on another machine) I iax from one to the other and then the native bridge is accomplished |
| 22:40.49 | ringhals | this is good other than the server where the call originates drops the channel so I can no longer monitor that iax phone.. any suggestions? |
| 22:41.31 | jart | canreinvite = no |
| 22:41.39 | ringhals | I have that set |
| 22:41.55 | ringhals | I think anyway.. LOL 1 sec let me check |
| 22:41.56 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 22:41.59 | file | that's for SIP |
| 22:42.04 | jart | oh yea |
| 22:42.19 | file | you want notransfer=yes |
| 22:42.45 | jart | that's the one |
| 22:42.56 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 22:43.04 | ringhals | notransfer=yes .. cool I will give that a shot (I knew I missed something simple) |
| 22:43.52 | CunningPike | Funny how it's notransfer=yes, instead of transfer=no :D |
| 22:44.58 | *** join/#asterisk SoMeOnEnUlL (n=morris@p1774-adslbkkct1.C.csloxinfo.net) |
| 22:45.03 | file | actually in latest trunk, it's both |
| 22:45.06 | file | with transfer=no being preferred |
| 22:45.13 | russellb | file: yay |
| 22:45.18 | russellb | file: svn is alive!!! |
| 22:45.23 | file | russellb: o rly? |
| 22:45.29 | jart | file: ya rly |
| 22:45.41 | ringhals | thanks a ton guys I knew I could count on you |
| 22:45.48 | jart | ringhals: <3 |
| 22:46.05 | russellb | in the trunk, there is an even cooler iax transfer option |
| 22:46.17 | russellb | that allows you to only transfer the media, but not the signalling |
| 22:46.28 | SoMeOnEnUlL | hi, can anyone here tell me what could be the problem with asterisk when I hear the playback but can't record or can't do the echo test? |
| 22:46.46 | jart | SoMeOnEnUlL: one way audio? firewall problems? |
| 22:46.51 | SoMeOnEnUlL | yea |
| 22:46.55 | SoMeOnEnUlL | that's what i thought |
| 22:47.10 | SoMeOnEnUlL | but, i opened all the port |
| 22:47.18 | SoMeOnEnUlL | and, i use STUN server |
| 22:47.54 | jart | is troubleshooting with IAX a possibility for you? |
| 22:47.57 | SoMeOnEnUlL | so, i was thinking maybe codec problem? |
| 22:48.43 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 22:48.57 | Renacor | is there an app that can log into asterisk and grab all the used channels? |
| 22:48.59 | SoMeOnEnUlL | yea |
| 22:49.21 | Renacor | so you could do a zapbarge |
| 22:49.22 | SoMeOnEnUlL | i'll see |
| 22:51.02 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:52.38 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 22:52.42 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 22:53.15 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:01.27 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 23:01.50 | Jaxxan | hey guys |
| 23:01.56 | Dr-Linux | Is there an option where I can put someone on hold w/o having music? Or is this a global item? |
| 23:02.10 | Dr-Linux | i mean on cisco 7940 phone? |
| 23:02.16 | Jaxxan | where can i find /var/lib/asterisk/sounds/* in their original .wav format ? |
| 23:02.53 | Jaxxan | or another format other than .gsm |
| 23:03.08 | Jaxxan | yeah yeah i know i can sox, but i think that loses quality when converting back to wav |
| 23:03.14 | CunningPike | Jaxxan: Trawl the list postings - someone has published these |
| 23:03.21 | CunningPike | About a couple months ago |
| 23:03.23 | wunderkin | kristian |
| 23:03.30 | Jaxxan | list postings ? |
| 23:03.35 | CunningPike | ~list |
| 23:03.37 | jbot | one warez list being sent |
| 23:03.51 | CunningPike | jbot, jbot, jbot |
| 23:04.00 | dlynes_home | lol |
| 23:04.12 | CunningPike | Jaxxan: asterisk-users mailing list |
| 23:04.14 | dlynes_home | ~mailinglist |
| 23:04.15 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:04.18 | Jaxxan | woot |
| 23:04.24 | CunningPike | smartarse ;) |
| 23:04.54 | Jaxxan | need to remove the " , " from the http://www.asteriskguru.com/archives link |
| 23:05.11 | dlynes_home | why? |
| 23:05.19 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:05.21 | Qwell[] | because dumb clients add the comma to the link |
| 23:05.23 | Jaxxan | if ya click on the link it doesn't work (= |
| 23:05.24 | dlynes_home | the comma being there makes perfect sense in the English language |
| 23:05.47 | Qwell[] | erm |
| 23:05.51 | Qwell[] | ,or and there" |
| 23:05.54 | CunningPike | Jaxxan: Works in my client - your client may not parse it proper;y |
| 23:05.59 | Jaxxan | prolly |
| 23:06.02 | Qwell[] | CunningPike: like I said...dumb clients |
| 23:06.10 | CunningPike | Qwell[]: ;) |
| 23:06.11 | Jaxxan | its tough finding a really good irc client for my mac |
| 23:06.16 | CunningPike | Colloquy |
| 23:06.19 | Jaxxan | using Colloquy atm |
| 23:06.25 | dlynes_home | Jaxxan: Chatzilla |
| 23:06.27 | CunningPike | :S - I have no comma |
| 23:06.35 | CunningPike | With Colloquy |
| 23:06.48 | Jaxxan | maybe i should upgrade |
| 23:06.54 | Jaxxan | i got v2.0 |
| 23:07.04 | CunningPike | 2.0.1 is latest |
| 23:07.19 | *** join/#asterisk shaun222 (i=Shaun@tina.ndcservers.net) |
| 23:09.39 | Renacor | god I hate phpclasses.org |
| 23:09.59 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 23:11.45 | dlynes_home | Renacor: try cpan.perl.org instead, then |
| 23:12.15 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 23:19.30 | *** part/#asterisk hayburn (i=chiaborg@concorde.hayburn.net) |
| 23:19.43 | *** join/#asterisk xcoyote (n=farfan@dsl-200-95-78-249.prod-infinitum.com.mx) |
| 23:20.22 | *** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net) |
| 23:20.34 | xcoyote | question: which function can i use in order to get the unixtime (milliseconds since 1 jun 1970) ${timestamp} returns something different |
| 23:20.42 | }btorch{ | which ports do I have to open on the firewall for my iax2 clients ? |
| 23:25.59 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 23:31.31 | *** part/#asterisk jeffik (n=Jeff@Maroon-103-179.ADSL.NetSurf.Net) |
| 23:32.25 | kapsel | im having a strange problem. im using pap2 voip adapters to my asterisk setup, and depending on what phone i plug on, some of them wont recieve calls (but they recieve a message that someone called), and some of them recieves calls just fine. |
| 23:32.30 | kapsel | any ideas? |
| 23:35.18 | FarrisG | is there a specific method of upgrading from a 1.0 version to 1.2, or do I need to just install fresh from CVS and then manually move over my old config? |
| 23:36.30 | *** join/#asterisk XanaXa (n=m@ppp-69-219-158-119.dsl.chcgil.ameritech.net) |
| 23:37.31 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 23:39.24 | *** join/#asterisk jahani (n=k@adsl196-64-229-217-196.adsl196-16.iam.net.ma) |
| 23:39.37 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 23:42.39 | XanaXa | guys I am having a lot of trouble using extension such as *61 for weather or *62 I don't think any of my custom_extensions.conf entries are being used, is there something I am might have missed, do you have to enable these somewhere? |
| 23:45.09 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 23:48.29 | xp_prg | has anyone run Asterisk::Manager perl on windows with activestate? |
| 23:49.00 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:49.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:49.43 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 23:49.48 | *** join/#asterisk zwelch (n=kumquat@pdpc/supporter/sustaining/zwelch) |
| 23:50.29 | Jaxxan | exten => *98,1,Goto(86,1) ; as an example |
| 23:50.57 | thx2000 | Anyone feel like being a good samaratin and answering a few questions regarding a multiple site implementation |
| 23:53.25 | De_Mon | no but ask anyway someone else may answer |
| 23:58.35 | Qwell[] | feel like != going to anyways |
| 23:58.56 | Qwell[] | guess it wasn't that important though |
