00:00.04 | Baylink | carrar: you do realise that that answer's worse than useless, right? :-) |
00:00.07 | *** join/#asterisk NateHB (n=noone@static-108-0-194-65.lsanca.dsl-w.verizon.net) |
00:02.25 | NateHB | Any aastra gurus around here, ive got a few 57is, I've got them all plugged into cisco router, all those ports are tagged native VLAN1000, some reason, 1 aastra phone with VLAN enabled, LANID set to 1000 works fine gets its ip from DHCP, the rest all cant get an IL address, i can plug my laptop into the same port and its gets a correct address, any ideas? |
00:04.18 | Gugge | dont enable vlan on the phones |
00:04.42 | NateHB | the only phone that actually works has VLAN enabled |
00:04.48 | NateHB | thats what gets me |
00:04.52 | paulc | [Jan 16 15:35:29] NOTICE[19663]: utils.c:1074 ast_wait_for_output: Timed out trying to write |
00:04.53 | Gugge | and your notebook doesnt ... but it works |
00:05.07 | paulc | uh, ignore that.. bloody right mouse click |
00:06.37 | NateHB | maybe i need to restart the switch |
00:06.53 | NateHB | fucking POS cisco catylst |
00:07.05 | NateHB | i meant frelling |
00:07.18 | NateHB | sorry, i relize theres probably alot of kids around here |
00:07.46 | *** join/#asterisk Akiraaa (n=Akiraaaa@79.112.34.79) |
00:08.48 | voipmonk | its not the cat's fault |
00:10.56 | NateHB | the cat? |
00:11.21 | NateHB | voipmonk, you think it might be the cabling? |
00:11.33 | paulc | <mx:Button x="46" y="187" label="Refresh" click="phpPlayerData.send()"/> |
00:11.49 | paulc | curses the mouse (as opposed to the cat) |
00:11.54 | *** join/#asterisk EwanPMcLean (n=ewanmcle@89.241.235.97) |
00:11.56 | paulc | sorry for the spam/copy'n'paste junk |
00:13.53 | Gugge | NateHB: i think i would look at the operator before looking at the cables or the cat ... when a notebook works |
00:14.08 | *** join/#asterisk Akiraaa (n=Akiraaaa@79.112.33.170) |
00:22.53 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
00:37.10 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
00:37.34 | drmessano | Sooooo |
00:37.39 | paulc | So? |
00:38.18 | drmessano | So I try to download some firmware and provisioning apps from Cisco's site |
00:38.54 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
00:38.59 | paulc | And how's that going for you? |
00:39.54 | drmessano | and can't figure out what to do with the files I actually downloaded.. Each of the ZIP and EXE archives opens in 7-ZIP and contains .text .rdata and .data files |
00:40.05 | drmessano | I'm guessing there's some backend issue on the site.. |
00:40.32 | paulc | ..or are those the firmware files you're meant to push to the router somehow? |
00:40.38 | drmessano | Nope |
00:40.38 | paulc | <-- not a Cisco expert, just guessing |
00:41.11 | drmessano | There should be .bin files in the ZIP archives and the EXE's should actually execute in windows, and they don't |
00:53.06 | *** join/#asterisk voipmonk (n=shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
00:59.43 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
01:07.19 | *** join/#asterisk hardwire (n=spencers@69-161-30-106.static.acsalaska.net) |
01:07.20 | hardwire | meh |
01:08.32 | ChannelZ | feh |
01:09.02 | *** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) |
01:16.25 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
01:23.23 | *** part/#asterisk Baylink (n=jra@65.34.94.81) |
01:32.24 | *** part/#asterisk EwanPMcLean (n=ewanmcle@89.241.235.97) |
01:34.59 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
01:44.13 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
01:49.28 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
02:01.17 | *** join/#asterisk aidinb (n=Aidin@adsl-71-159-228-75.dsl.sndg02.sbcglobal.net) |
02:03.07 | *** join/#asterisk Diffen2 (n=diffen2@c-737de555.042-17-73746f11.cust.bredbandsbolaget.se) |
02:03.09 | p3nguin | drmessano: What were you trying to download and install? |
02:04.52 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
02:06.40 | Diffen2 | evning. i have a really stupid question. when i call from my asterisk to my pstn gw i dont have any headers at all if i look at the call in wireshark. strange thing is that the call gets through so it works. have i missed out anything in the sip.conf? |
02:13.16 | drmessano | p3nguin: I figured it out |
02:19.15 | *** join/#asterisk infobot (i=ibot@rikers.org) |
02:19.15 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #aster |
02:24.15 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
02:24.24 | kam187 | hi |
02:24.37 | kam187 | does asterisk support vad for detecting answer? |
02:25.10 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) |
02:29.23 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
02:29.54 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
02:31.32 | *** join/#asterisk joobie (n=joobz@CPE-143-238-230-89.vic.bigpond.net.au) |
02:38.40 | *** join/#asterisk chendy (n=chatzill@204.152.211.137) |
02:39.03 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
02:44.28 | *** join/#asterisk pawz (n=pawz@ppp118-208-61-62.lns20.bne1.internode.on.net) |
02:48.20 | *** join/#asterisk jblack (n=jblack@71.181.248.16) |
02:50.32 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
02:57.44 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
03:04.28 | *** join/#asterisk Katty (n=User@adsl-70-253-161-187.dsl.stlsmo.swbell.net) |
03:04.32 | Katty | hi. |
03:05.21 | ChannelZ | ahoyhoy |
03:12.51 | *** join/#asterisk mog (n=mog@c-71-228-185-24.hsd1.al.comcast.net) |
03:12.51 | *** mode/#asterisk [+o mog] by ChanServ |
03:14.34 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
03:19.41 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
03:28.02 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
03:28.13 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
03:33.42 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
03:34.09 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
04:06.03 | *** join/#asterisk dkirker-openmobl (n=dkirker@openmobl/ceo/dkirker) |
04:07.46 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
04:17.37 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
04:21.06 | Katty | i'm am irritated at our company. |
04:21.20 | Katty | they are talking about relationships between white and black people to be 'crossbreeding' |
04:21.25 | Katty | i find it harsh, and untasteful. |
04:21.33 | Katty | cranky. |
04:25.15 | *** join/#asterisk etnos (i=etnos@c-75-74-66-161.hsd1.fl.comcast.net) |
04:39.32 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
04:51.41 | *** join/#asterisk geneticx (n=etnos@adsl-2-215-86.mia.bellsouth.net) |
04:58.48 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
05:07.24 | kam187 | hmm how do i stop progress from a sip channel being passed back |
05:07.31 | kam187 | i want to control it in the extensions.conf |
05:11.29 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
05:25.18 | Katty | eppigy: ping |
05:26.13 | *** join/#asterisk manxpower (n=ewieling@216.186.151.147) |
05:26.22 | manxpower | any music geeks around |
05:26.27 | ChannelZ | Isn't crossbreeding between species? |
05:26.28 | *** join/#asterisk linuxviewer (n=example@ip72-222-249-220.ph.ph.cox.net) |
05:26.36 | linuxviewer | anyone using Cisco 7940 phone with Asterisk? |
05:26.45 | ChannelZ | I am a geek that likes music |
05:27.44 | manxpower | What does F with a subscript 4 mean? |
05:28.54 | kam187 | hmm |
05:35.49 | ChannelZ | in what context? |
05:36.37 | manxpower | As in the note F subscript 4 is 349.23 Hz |
05:36.50 | manxpower | F in the 4th Octave? |
05:37.18 | *** join/#asterisk pawz (n=pawz@ppp118-208-61-62.lns20.bne1.internode.on.net) |
05:37.31 | ChannelZ | well yeah then it'd be the F above 'middle C' |
05:37.49 | ChannelZ | I don't think the superscript has particular significance |
05:38.11 | ChannelZ | which is why I asked as maybe I had no idea what you were talking about |
05:38.17 | kam187 | hmm how do i overide the progress between two calls? |
05:38.23 | ChannelZ | hang up |
05:38.51 | kam187 | :-/ |
05:38.55 | manxpower | A USA dialtone is more or less F above middle C (350) + A above middle C (440) |
05:39.15 | kam187 | i dont want the answer on one side to go through to the other, i want to control it with answer() |
05:39.34 | ChannelZ | huh? |
05:39.47 | manxpower | kam187: Until you answer the call, all progress is generated by the local SIP phone. |
05:40.00 | kam187 | i have a h323 call come in, and in the extensions it dials a sip provider out |
05:40.06 | kam187 | but that providers answers right away |
05:40.36 | manxpower | kam187: not much you can do about that other than get the provider to not answer or get a different provider. |
05:40.43 | kam187 | even while its ringing, so i want to do some kind of ring detection using background detec or something |
05:40.53 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
05:40.56 | manxpower | kam187: I wish you the BEST of luck. |
05:41.18 | kam187 | well i could modify sip.c to not pass the progress pretty easily but i'd rather not |
05:43.11 | kam187 | ok so u dont know of any way to overide progress between the two legs? |
05:43.14 | ChannelZ | then how do you decide when the other end picks up? |
05:44.03 | kam187 | by monitoring the audio either using dsp.c, speech() or backgrounddetect() |
05:44.29 | ChannelZ | and what is the point of doing this? |
05:44.57 | kam187 | call progress detection using vad |
05:46.16 | ChannelZ | ...and what is the point of doing this? |
05:46.32 | kam187 | fo billing purposes |
05:46.38 | kam187 | most voip routers support this |
05:46.39 | kam187 | google it |
05:47.18 | kam187 | for example chan_dahdi has line detection built in |
05:49.04 | manxpower | Yes, and as it says in the config file it's experimental and buggy. |
05:49.11 | ChannelZ | manxpower: so are you wanting to play dialtones on the piano or.. :) |
05:49.47 | manxpower | ChannelZ: I'm trying to come up with a dialtone different from the default, but still is still pleasing to the hear. |
05:50.08 | kam187 | manxpower: yeah thats ok i'm just saying i'm not asking for some wierd wako bizare thing that no one else on the earth would ever want |
05:50.10 | manxpower | So I tried CE, didn't like it, trying FC now. |
05:50.59 | manxpower | kam187: I don't know what part of the world you are from but any carrier that answered all calls is a POS carrier. there is NO reason for a carrier to do this. |
05:51.20 | manxpower | And it, in fact, illegal in the USA, AFIK. |
05:51.36 | ChannelZ | kam187: I didn't say you were, just wondering what the end game was |
05:51.37 | kam187 | yeah true, but u have the same problem when u connect to analog phone lines |
05:51.54 | manxpower | kam187: Your carrier is using analog lines? |
05:51.56 | kam187 | which is probably why only the analog channel drivers support it |
05:52.05 | kam187 | tbh i dont really care about this carrier |
05:52.26 | kam187 | i'm more interested to get the functionality working on askterisk so i know its an option |
05:52.31 | manxpower | Well if by "support" you mean "randomly disconnect calls when enabled" then yes it is "supported" on DAHDI/Zaptel |
05:53.10 | manxpower | kam187: get out some books on DSP programming and let us know when you have a patch. 8-) If it was easy to do someone would have done it already. |
05:53.13 | kam187 | lol is it that bad? |
05:53.36 | kam187 | i know how DSPs work :P |
05:54.03 | manxpower | kam187: there is anecdotal evidence it likes to think higher pitched voices (women) or loud voices (men) are "BUSY" tones. |
05:54.32 | kam187 | ok |
05:54.37 | manxpower | It's a great feature to enable when a husband and wife are arguing. |
05:54.42 | kam187 | haha |
05:54.46 | ChannelZ | HIII...YOOOUUU...HAAAAVVE...REEEAACHED...BOOOOB |
05:55.35 | ChannelZ | (speaks monotone) |
06:00.54 | ChannelZ | Hmm. Not far from where I work, there is one single block that conatins 4 bail-bonds places, 2 criminal attorneys, a behavioral health center, and a post office. |
06:01.13 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
06:10.57 | *** join/#asterisk pawz (n=pawz@ppp118-208-61-62.lns20.bne1.internode.on.net) |
06:18.46 | Katty | hugs ChannelZ |
06:25.13 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
06:32.23 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
07:14.04 | *** join/#asterisk lordmortis (n=lordmort@124-169-172-70.dyn.iinet.net.au) |
07:21.21 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
07:41.49 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
07:41.54 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
07:52.08 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
08:08.33 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
08:49.01 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
08:57.08 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
09:12.41 | tzafrir | kam187, with SIP, as with any decent VoIP protocol (and with ISDN, in DAHDI) you get proper out-of-band signalling as to the state of the call |
09:12.51 | tzafrir | No need to guess it by listening to the line |
09:13.40 | tzafrir | Can you be more specific about where calls don't properly disconnect with SIP? |
09:17.05 | ChannelZ | I don't think his issue is disconnect; he wants to know when the remote end actually picks up the phone for accurate billing |
09:20.44 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
09:22.45 | tzafrir | This should also be reported. Your call only starts at that point |
09:24.59 | ChannelZ | well doesn't it depend on the channel (and perhaps the carrier if you're using an ITSP)? |
09:25.45 | ChannelZ | Like on analog lines DAHDI reports the channel as "ANSWERED" as soon as it's done dialing |
09:27.34 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
09:28.00 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
09:38.06 | *** join/#asterisk pawz (n=pawz@210.56.81.113) |
09:39.20 | *** join/#asterisk djabbour (n=djabbour@ool-457be7e8.dyn.optonline.net) |
09:39.51 | djabbour | Are there any good metrics on how much transfer (in GB say), asterisk traffic uses on average per call volume? |
09:49.09 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
09:49.18 | *** join/#asterisk smooth_penguin (n=smoove@59.95.23.61) |
09:50.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
09:57.01 | *** join/#asterisk porter (n=terdon@unaffiliated/porter/x-000001) |
10:14.33 | *** part/#asterisk _bugz_ (n=bugz@adsl-99-129-212-35.dsl.lsan03.sbcglobal.net) |
10:24.42 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.38.122) |
10:45.00 | *** join/#asterisk dennisG (n=dennisG@84.30.136.208) |
10:47.22 | *** join/#asterisk pawz (n=pawz@210.56.91.157) |
10:50.43 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
11:00.39 | *** join/#asterisk mrchrisadams_ (n=mrchrisa@82-45-160-216.cable.ubr05.hari.blueyonder.co.uk) |
11:10.15 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
11:23.43 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
11:27.16 | benngard | now i am lost... |
11:27.28 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
11:27.37 | benngard | from cli: channel originate SIP/0317998985 extension 0317998975@inputinterior.se <- works |
11:28.01 | benngard | from ami: |
11:28.01 | benngard | Action: Originate\r\nChannel: SIP/0317998985\r\nExtension: 0317998975\r\nContext: inputinterior.se\r\nPriority: 1\r\nTimeout: 15000\r\nCallerID: Magnus Benngard<0317998975\r\n\r\n> |
11:28.18 | benngard | == Starting SIP/0317998985-0000003b at inputinterior.se,,1 failed so falling back to exten 's' |
11:28.59 | benngard | i cant se the difference, and why ",," |
11:29.39 | benngard | end the end is ofc: <0317998975>\r\n\r\n |
11:35.11 | tzafrir | benngard, obviously the 'extension' did not get through |
11:35.31 | benngard | sure but i cant se what i am doing wrong |
11:36.08 | benngard | exten => 0317998975,1,ExecIf($[${DB_EXISTS(CFIM/0317998975)}]?Goto(${DB(CFIM/0317998975)},1) exists in context inputinterior.se |
11:36.10 | tzafrir | use 'Exten' |
11:36.16 | tzafrir | not 'Extension' |
11:36.21 | benngard | lets try |
11:38.24 | benngard | that worked, thx |
11:39.51 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
11:56.55 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |
12:34.22 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
12:39.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
12:41.53 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.143.83.dsl.dyn.forthnet.gr) |
12:50.23 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
12:50.44 | k-man | are the siemens cordless voip phones any good? |
12:50.52 | k-man | like this one: http://www.ryda.com.au/Siemens-C470IP-Voip-Cordless-Phone-p/c470ip.htm |
12:52.18 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
13:16.28 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
13:19.35 | *** join/#asterisk coppice (n=chatzill@238.168.17.210.dyn.pacific.net.hk) |
13:26.10 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.142.208.dsl.dyn.forthnet.gr) |
13:28.56 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
13:46.21 | benngard | i have some: http://gigaset.com/shc/0,1935,se_sv_0_152411_rArNrNrNrN,00.html really good sound and very easy to configure with * |
13:57.17 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
14:02.57 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
14:08.12 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
14:09.01 | benngard | if i know an extension and wanna know the the name of the channel connected to that extension, how do i do that (in an easy way)? |
14:21.48 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
14:43.25 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
14:53.29 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
14:53.33 | sun28 | moin \o/ |
14:54.08 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
14:54.26 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
14:54.28 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
15:08.01 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
15:10.28 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
15:13.27 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.239.104.dsl.dyn.forthnet.gr) |
15:19.31 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
15:25.02 | *** join/#asterisk katoen (i=037eb61f@xs8.xs4all.nl) |
15:25.48 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
15:30.18 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) |
15:40.22 | *** join/#asterisk Alagar (n=Administ@122.164.37.250) |
15:44.25 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
15:46.15 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
15:49.34 | *** join/#asterisk scunizi (n=scunizi@ip72-197-240-157.sd.sd.cox.net) |
16:04.23 | *** join/#asterisk Heretic (n=fallen@dsl-246-123-89.telkomadsl.co.za) |
16:04.53 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
16:13.29 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
16:13.32 | _abc_ | hello |
16:14.04 | _abc_ | is there a way to search the ekiga.net phonebook ? they claim to support ldap but i can't seem to go through ?! |
16:14.21 | _abc_ | of course i have an ekiga account |
16:16.19 | *** join/#asterisk atha (n=atha@unaffiliated/athayde) |
16:17.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
16:17.39 | scunizi | I just loaded asterisk and gastman.. running gastman asks for a host name.. the only one I have is a free ekiga account.. but with that it won't connect.. any hints to get past the main screen? |
16:17.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
16:20.46 | jaytee | you need to put in the hostname of your asterisk server for gastman and gastman is soooooo old and not supported |
16:21.12 | scunizi | jaytee: ah.. is there an alternative? |
16:22.03 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
16:23.01 | jaytee | scunizi, what are you looking for? a graphical management interface for asterisk? |
16:23.07 | scunizi | jaytee: yes. |
16:23.45 | scunizi | jaytee: and a tutorial for cli management ... most of what I've found is pretty old |
16:23.56 | jaytee | then you probably want to look at installing AsteriskNOW 1.5 or whatever the latest version is and choosing the Freepbx interface. |
16:24.51 | jaytee | but anything with Freepbx on it is going to limit your flexibility in making Asterisk do things "outside the box" |
16:25.36 | scunizi | jaytee: understandable.. |
16:26.19 | jaytee | for CLI management just type help for a list of commands and then if you want more info on a particular CLI command type help "command name" without the quotes |
16:26.39 | jaytee | for example help sip will show you all the CLI commands related to sip |
16:27.15 | jaytee | and while the last version of the book only covers up to version 1.4 it's still mostly relevant |
16:27.20 | jaytee | ~book |
16:27.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
16:27.34 | jaytee | ~botsnack |
16:27.34 | infobot | jaytee: aw, gee |
16:28.04 | scunizi | jaytee: that's great.. thanks for the tips and link. |
16:28.11 | jaytee | yw |
16:30.36 | scunizi | jaytee: you running asterisk on WinXX? just curious because of the reference on cli "help <command>" |
16:32.29 | jaytee | scunizi, WinXX? I run Asterisk on RHEL 5.2 and CentOS 5.2. While there is a "port" of Asterisk for Windows, it's garbage and should be avoided at all costs. |
16:35.20 | jaytee | scunizi, I can manage my asterisk servers from a Windows workstation by running Putty (Windows version) to connect via ssh to a secure terminal session on the asterisk server and then I run asterisk -vvvvvvr to connect a remote asterisk console. Asterisk runs as a service on all my asterisk servers. |
16:36.06 | *** join/#asterisk klochan (n=klochan@95-27-105-153.broadband.corbina.ru) |
16:38.04 | scunizi | jaytee: ok.. I was curious because "help <command>" on my linux box doesn't do anything.. |
16:39.02 | jaytee | scunizi, help is not a linux command, it's an asterisk command typed in the asterisk CLI, not the linux command prompt. |
16:42.37 | scunizi | jaytee: ah! you can see how new I am to this.. how do I get to the asterisk cli? |
16:43.47 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
16:43.53 | jaytee | scunizi, type asterisk -vvvvvvr from a linux command prompt |
16:44.36 | jaytee | if you get an error message about asterisk.ctl not being found then asterisk IS NOT running as a service and you'd have to type asterisk -vvvvvvvc instead |
16:45.19 | jaytee | scunizi, what linux distro are you running? |
16:45.43 | *** join/#asterisk phdpeabody (n=peabody@77-21-14-170-dynip.superkabel.de) |
16:45.48 | *** join/#asterisk cesar_CR (n=cesar@201.199.168.170) |
16:46.09 | scunizi | jaytee: you might laugh but ubuntu 8.04 with a VM running 9.10 kubuntu.. I installed asterisk in the vm |
16:47.25 | jaytee | scunizi, that shouldn't matter unless you need TDM hardware to connect to the PSTN in which case a VM will not be the right solution. |
16:47.32 | phdpeabody | OK, I was reading the wiki but I've got a retarded question.. if I install an asterisk server, what do I need to do to make/receive calls? |
16:47.51 | jaytee | phdpeabody, read the book |
16:47.54 | jaytee | ~book |
16:47.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
16:49.37 | scunizi | jaytee: this is mostly for experimentation.. I sell an IP-PBX that has a linux kernel but that's where it stops resembling anything else.. good system but got my curiosity up about asterisk.. I'd also like to implement it at home for my other business. With a sip trunk and muliple channel capability.. |
16:49.59 | phdpeabody | ok I'll read the book if you can yes/no two questions.. 1. Do I need to have some special provider access to make/receive international calls 2. Is there a low-cost module plug that I can connect a standard phone to VOIP? |
16:51.34 | jaytee | phdpeabody, you'd need an account with an ITSP to provide a sip "trunk" to make and receive calls outside your own network. many offer international plans. |
16:52.01 | jaytee | ~itsplist-eu |
16:52.16 | jaytee | ~itsplist |
16:52.42 | jaytee | hmmm, can't remember how to get the ITSP list for Europe |
16:53.07 | jaytee | ~itsplist-germany |
16:53.10 | phdpeabody | heh I don't suppose it's on the website? |
16:53.29 | jaytee | voip-info.org? might be, not sure |
16:53.48 | phdpeabody | I was thinking asterisk.org but I'll check that one too |
16:53.49 | jaytee | or you can google ITSP and look for a provider in germany |
16:54.12 | hardwire | gives phdpeabody a treat |
16:54.43 | jaytee | and for an analog phone you'll need an ATA adapter like a Linksys SPA2102 or PAP2T-NA |
16:56.48 | phdpeabody | I think the linksys is discontinued, but cool thanks |
16:57.21 | jaytee | no, the SPA2102 is still sold in the US, your market might be different |
16:58.41 | Kobaz | hmm |
16:58.52 | Kobaz | what would cause echo on an ip to ip call |
16:59.29 | Kobaz | i have a sip phone, going to asterisk, and then an iax trunk to another asterisk box which sends the call to another sip phone |
16:59.38 | jaytee | mic volume set too high? is one of the sets on speakerphone? |
16:59.49 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
17:00.19 | Kobaz | headset |
17:01.14 | Kobaz | it's like asterisk is echoing back whatever audio it gets back to the caller |
17:01.33 | scunizi | jaytee: trying to start with asterisk vvvvvvr or c results in errors for a non-existant asterisk.conf and extconfig.conf .. as well as permission issues.. should asterisk be started as root? |
17:01.44 | *** part/#asterisk phdpeabody (n=peabody@77-21-14-170-dynip.superkabel.de) |
17:02.42 | jaytee | scunizi, yes it should be started as root and there are configuration steps you have to do. now would be a good time to download and start reading the book. |
17:03.27 | scunizi | jaytee: did.. do the download.. and yes.. I'll read.. I figured these basic questions would/should be answered in there.. |
17:04.16 | jaytee | the book also contains a section for running asterisk as non-root but it's best to start with the simple things first. You'll also need to edit some conf files and there's a section of the book that gets you up and running fairly quickly. |
17:04.46 | Kobaz | ph |
17:04.47 | Kobaz | oh |
17:04.47 | Kobaz | hmm |
17:04.50 | Kobaz | i think it's the soft phone |
17:05.03 | scunizi | cool.. thanks. I'll be afk for a while.. atleast until after the chargers game :) |
17:05.13 | Kobaz | the sip phone on the other end of the asterisk iax link is a sip soft phone (twinkle) |
17:05.19 | Kobaz | it's creating echo |
17:05.53 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
17:05.56 | Kobaz | it's really annoying |
17:06.09 | manxpower | Kobaz: get better heassets |
17:06.16 | manxpower | and headsets too |
17:06.18 | Kobaz | it's the soft phone |
17:06.30 | manxpower | that is very unlikely |
17:06.46 | Kobaz | well, as unlikely as it is... that's what it is |
17:06.54 | manxpower | Sucks to be you. |
17:06.56 | Kobaz | i don't have a mic on the pc with the soft phone |
17:07.10 | Kobaz | if i mute all the channels in the mixer, the echo goes away |
17:07.44 | jaytee | try turning the mic volume in the mixer down or if this is ALSA try lowering your MUX setting |
17:07.52 | Kobaz | mic volume is zero |
17:07.54 | manxpower | Kobaz: Does the issue go away when you switch to a different softphone? |
17:08.10 | Kobaz | yeah, zoiper works no problem |
17:08.31 | manxpower | Odd. |
17:08.37 | p3nguin | Zoiper has echo cancellation... does the other phone you're using? |
17:08.52 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
17:09.36 | Kobaz | not sure |
17:10.26 | Kobaz | it's not checked |
17:10.29 | Kobaz | heh |
17:10.31 | Kobaz | hmm |
17:10.41 | Kobaz | i didn't think you needed echo cancellation with ip to ip calls |
17:10.49 | *** join/#asterisk cesar_CR (n=cesar@201.199.168.170) |
17:11.36 | Kobaz | i turned on echo cancellation in twinkle and now the echo is much less, still a very low volume echo |
17:12.55 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
17:14.00 | manxpower | echo cancelation on IP phones cancel the speaker/mic/speakerphone echo, not far end analog loop echo. |
17:15.05 | manxpower | Polycom phones are the same way. You can't cancel the far end analog loop echo in the IP phone because the latencies are far too high, it has to be done at the PSTN/VoIP conversion point. |
17:15.35 | manxpower | But with a headset you should not get much echo. |
17:16.23 | Kobaz | yeah the polycom handles it fine |
17:17.40 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
17:17.47 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
17:18.04 | Kobaz | since there's no moh since it never got the request |
17:18.07 | Kobaz | er, wrong window |
17:20.35 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
17:28.25 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
17:28.53 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
17:41.59 | *** join/#asterisk NateHB (n=noone@static-71-116-246-221.lsanca.dsl-w.verizon.net) |
17:43.11 | NateHB | hey guys, my asterisk server is multihomed, is there any way to set one program to use a certain outside gateway, and let asterisk use its interface? |
17:43.51 | NateHB | I've got openfire installed on the same box as asterisk, and I want to make sure openfire doesnt use the same bandwith as asterisk |
17:46.48 | manxpower | NateHB: What interface is used is up to the OS. |
17:49.09 | Kobaz | NateHB: #networking |
17:50.15 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
18:02.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
18:08.22 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
18:10.24 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
18:14.40 | *** join/#asterisk Nephfl (n=none@rrcs-67-78-149-118.se.biz.rr.com) |
18:16.34 | Nephfl | hello, I have always used polycom ip phones, i now want to connect some phones to a hosted asterisk pbx from behind a nat firewall and dynamic ip, so I imagine a sip phone with stun would do the job, what do yuo guys recommend? |
18:18.31 | Nephfl | is this thing on? |
18:19.27 | Nephfl | is anyone here? |
18:26.30 | ChannelZ | nope |
18:30.57 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
18:31.26 | p3nguin | Does a NoOp() stop a PlayTones(), or do I need to explicitly use StopPlayTones() or some other sound? |
18:32.32 | p3nguin | nephfl: Asterisk can handle NAT pretty well most of the time, so there shouldn't be any problem. Just follow the guide for setting up NAT with SIP. |
18:32.38 | p3nguin | ~sipnat |
18:32.38 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
18:36.04 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:c5f:79bd:a077:acde) |
18:36.05 | cusco | hi |
18:36.57 | cusco | Im trying to make a php script connect into asterisk's manager with fsockopen. Has anybody done that? |
18:38.44 | cusco | http://paste.debian.net/56949/ |
18:38.48 | ChannelZ | yes but not lately |
18:39.13 | ChannelZ | you need \r\n for one |
18:39.18 | cusco | well im not being able to read anything |
18:39.23 | cusco | ouch |
18:39.28 | cusco | :/ ok |
18:40.05 | cusco | not enough |
18:40.14 | cusco | I can echo the first gets($socket) |
18:40.34 | cusco | if I try to echo it a second time (right down the bottom) |
18:40.47 | cusco | the script hangs, probably because it has nothing to echo/read |
18:41.03 | cusco | (I added the \r) |
18:41.22 | cusco | it only echos "Asterisk Call Manager/1.1" |
18:41.48 | *** part/#asterisk dgilmore (n=dgilmore@fedora/dgilmore) |
18:45.02 | ChannelZ | it might not be parsing the line breaks right, try using fread instead |
18:45.05 | Nephfl | any suggestions for wholesale sip terminaton providers without volume requirements and no channel limits? |
18:46.01 | cusco | with fread I need to specify a number of bytes. I can do so but it errors out anyway |
18:46.18 | ChannelZ | re-paste what you have so far |
18:46.31 | cusco | actually it hangs |
18:46.32 | cusco | ok |
18:47.20 | cusco | http://paste.debian.net/56951/ |
18:48.14 | cusco | it echoes the first fgets() but hangs on the fread() |
18:49.29 | cusco | ow!! |
18:49.42 | cusco | I replaced the '\r\n' with "\r\n" |
18:49.45 | cusco | it works! |
18:49.52 | cusco | Response: Success |
18:49.53 | cusco | Message: Authentication accepted |
18:50.00 | cusco | dang |
18:50.44 | ChannelZ | yah was just going to say that, \n and other escapes aren't evaluated in single quotes |
18:51.19 | hluesea | greg |
18:53.20 | cusco | ok thanks |
19:02.51 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
19:05.34 | cusco | and don't need \r |
19:05.35 | ChannelZ | Nephfl: Flowroute? |
19:11.01 | dlynes | Nephfl, if you're in Canada, Navigata is the best I've used |
19:11.40 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
19:11.42 | p3nguin | VoIP.ms also has a presense in Canada, so they could be another option. |
19:13.25 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
19:15.10 | *** join/#asterisk Mango (n=Mango@d154-20-97-118.bchsia.telus.net) |
19:15.41 | Mango | Hello. Can someone please remind me what pbx_config.so does? |
19:16.08 | Kobaz | Mango: loads your dialpla |
19:16.09 | Kobaz | n |
19:16.09 | benngard | i wrote a small app that (by a simple html/php/ami) could transfer a call that i have answered, but i had i hard time to find the channel my extension was connected to, did use "Action: CoreShowChannels" and looked for my extension and "BridgedChannel", that worked but was not fun at all to parse, is there a better way to do it? |
19:16.22 | Mango | Kobaz: Thanks |
19:16.43 | p3nguin | Is _doVoiceMail something specific to OderlyQ, or to Asterisk in general? |
19:17.41 | *** join/#asterisk Katty (n=User@adsl-70-253-161-187.dsl.stlsmo.swbell.net) |
19:17.47 | Katty | :>>> |
19:17.54 | Katty | 4 new breeds of bird are in the yard this morning! |
19:18.13 | Kobaz | it's raining! |
19:18.15 | Kobaz | RAIN! |
19:18.17 | Kobaz | in january |
19:18.18 | Mango | My wife would like to know which ones :) |
19:18.19 | p3nguin | Your milkshake brings all the birds to the yard. |
19:18.21 | Kobaz | all my snow is going away |
19:19.18 | benngard | i i wish my snow could go away |
19:19.31 | benngard | just a brown mess outside |
19:19.53 | Kobaz | it's gotta snow some more before the end of winter |
19:20.05 | Katty | a white breasted nuthatch, a red bellied woodpecker, a tufted titmouse, and a american goldfinch |
19:20.06 | Kobaz | i still have a season ski pass, and i haven't gone enough for it to pay for itself yet |
19:20.22 | Mango | Katty: Yay =) |
19:20.28 | Katty | :> |
19:20.35 | Mango | is married to an ornithologist |
19:20.48 | dlynes | bird lover |
19:21.06 | benngard | i am swedish, but i do (roughly) understand what kind of bird it is |
19:21.16 | Katty | Mango: you should share my crittercam with her. |
19:21.30 | dlynes | mice have tits? |
19:21.35 | Kobaz | of course |
19:21.49 | Katty | they are mammals. they do have live young. |
19:21.55 | dlynes | why are their tits tufted, though? |
19:22.00 | Katty | what? |
19:22.18 | dlynes | tufted mice tits |
19:22.21 | dlynes | erm |
19:22.24 | dlynes | tufted titmouse |
19:22.30 | dlynes | snickers. |
19:22.31 | Katty | dlynes: http://upload.wikimedia.org/wikipedia/commons/d/d7/Tufted_titmouse_perching_2006-11-23.jpg |
19:22.48 | Kobaz | that's not a mouse |
19:22.58 | dlynes | no kidding |
19:23.01 | dlynes | ripped off |
19:23.01 | Katty | no, it's a titmouse. |
19:23.14 | dlynes | I thought i was going to get to see some titties |
19:23.17 | Mango | Kobaz: Wow! You must be an ornithologist too =) |
19:23.52 | Kobaz | totally |
19:23.55 | Mango | Katty: What country do you live in? |
19:24.05 | dlynes | not to mention that breast on those nuts |
19:24.12 | dlynes | and that woody pecker |
19:24.16 | Katty | Mango: Central USA |
19:24.21 | dlynes | Katty's got a dirty mind |
19:24.32 | ChannelZ | BORING - if you type 'tits' into google it doesn't auto-suggest anything |
19:24.46 | benngard | censur! |
19:25.15 | dlynes | but i guess his belly's red from getting poked with a woody pecker all the time |
19:25.33 | *** join/#asterisk titter` (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
19:25.46 | dlynes | and titter`'s here just in time |
19:25.51 | benngard | ChannelZ: type breast into google and se what u get as first ;) |
19:26.20 | dlynes | benngard, scarless breast enlargements? |
19:26.28 | dlynes | erm implants, that is? |
19:26.39 | titter` | titter means to giggle, no clue whatcha talking about |
19:27.02 | titter` | http://pastebin.com/m25010808 -- this is flooding my console, and my pri lines are down |
19:27.05 | dlynes | titter`, oh...we were just talking about mice tits |
19:27.07 | titter` | anyone know what the deal is with this |
19:27.21 | ChannelZ | Wikipedia. |
19:27.29 | ChannelZ | I'm sick of Wikipedia |
19:27.32 | Kobaz | titter`: have you tried reloading dahdi? |
19:27.35 | dlynes | ChannelZ, why's that? |
19:27.44 | Kobaz | titter`: anything good in your system log? |
19:27.46 | titter` | I am going to, but I am curious what would cause this error |
19:28.00 | dlynes | a filled up scheduler |
19:28.00 | ChannelZ | I don't know. |
19:28.14 | dlynes | don't ask me what the scheduler is, though |
19:28.14 | *** join/#asterisk DelphiWorld (n=Miranda@41.104.103.211) |
19:28.16 | DelphiWorld | hi |
19:28.22 | titter` | It just happened my boss called me freaking ... they fired the old asterisk guy after his less than stellar setup lead to a box getting rooted and a very expensive six digit hack with intl calls to cuba |
19:28.23 | DelphiWorld | any iristel customer here? |
19:28.35 | ChannelZ | (although 'breast' on wikipedia has some wikiporn) |
19:28.37 | dlynes | titter`, beauty |
19:28.50 | titter` | dahdi_tool isn't even installed -.- |
19:28.54 | titter` | brb. |
19:29.05 | dlynes | DelphiWorld, iristel? |
19:29.27 | Kobaz | ooo wikiporn |
19:30.08 | DelphiWorld | dlynes: http://www.iristel.ca ;) |
19:30.36 | DelphiWorld | dlynes: dyrect lines, give me one! ;) |
19:30.51 | dlynes | Kobaz, it even has some pictures of that oddity that's become popular recently with the mcdonald's diet.....man titties |
19:31.18 | dlynes | DelphiWorld, wow...Canadian...never heard of them, though |
19:31.21 | Kobaz | heh |
19:32.00 | DelphiWorld | dlynes: register with it;) |
19:32.06 | dlynes | ah...based out of markham, on |
19:32.28 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
19:32.40 | dlynes | DelphiWorld, they any good, or do they suck? |
19:32.51 | DelphiWorld | dlynes: very good;) |
19:32.53 | titter` | installed dahdi-tools and dahdi_tool didn't install -.- |
19:33.05 | dlynes | coverage areas across canada, including bc |
19:33.13 | Kobaz | http://www.urbandictionary.com/define.php?term=McGurgles |
19:33.20 | dlynes | I wonder why I never heard of them in BC, and I'm in the telecom industry there |
19:33.43 | ChannelZ | Do you eventually get the McRuns? |
19:34.01 | Kobaz | i would think so... after eating the McMountain of food |
19:34.46 | Mango | I've never heard of them either...in BC as well. |
19:35.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:35.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
19:36.02 | Mango | Looks expensive |
19:36.08 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
19:36.28 | DelphiWorld | Mango: google for algeria connect |
19:36.38 | *** join/#asterisk darkrift_lp (n=dark@65.92.170.72) |
19:37.12 | DelphiWorld | iristel is providing algeria did |
19:37.30 | DelphiWorld | brb |
19:37.46 | darkrift_lp | someone have an idea if it's a good idea to have a TDM card and an asterisk installation through a virtual machine ? Is it reliable for the communication between the card and the asterisk serv, or there's actually some issue with that ? |
19:38.20 | Kobaz | darkrift_lp: your first challange is getting the tdm card actually working in a virtual machine |
19:38.43 | Kobaz | darkrift_lp: and your second challange is making sure your virtual machine gets very frequent and accurately timed execution timeslices |
19:39.09 | darkrift_lp | yeah, that'S the problem |
19:39.27 | Kobaz | asterisk is very time-sensitive, for example there's certain things that must be sent out every 20ms |
19:39.43 | Kobaz | just buy a $299 quad core dell server |
19:40.44 | darkrift_lp | well if I need to buy a new one I'll do that, I just didn't want to have another server in the server room because we wanted to have limited number of servers ... (not enough space and cost saving) |
19:42.42 | *** join/#asterisk wepy (n=wepy@ip70-179-126-55.dc.dc.cox.net) |
19:42.44 | wepy | hello |
19:43.19 | titter` | hmm, dahdi_tool doesn't install when I compile dahdi-tools |
19:43.45 | wepy | what is the service called where I get a phone number that's forwarded to my asterisk box? |
19:43.47 | darkrift_lp | thanks for the info Kobaz |
19:43.55 | wepy | (but I don't lease the phone line, i only handle IP) |
19:44.01 | Kobaz | welcome |
19:44.18 | wepy | heh |
19:44.43 | Kobaz | hmm |
19:45.44 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
19:47.18 | wepy | Is there a kind of service where I get an incoming number and the ability to dial out to traditional phones? |
19:47.28 | drmessano^ | wepy: That would be an ITSP |
19:47.46 | Kobaz | something very strange is going on with originate and musiconhold over iax |
19:47.50 | p3nguin | wepy: a DID, maybe? |
19:47.51 | drmessano^ | wepy: Theres no "forwarding" here, this is a native transport, no more or less than analog or PRI |
19:48.00 | drmessano^ | ~itsp |
19:48.00 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
19:48.00 | wepy | ah thanks |
19:48.27 | wepy | do most ITSP providers handle hunt-modes and voicemail? |
19:48.35 | Kobaz | no |
19:48.38 | drmessano^ | no |
19:48.41 | wepy | or could i just have all calls routed to my asterisk system? |
19:48.59 | wepy | hm |
19:49.09 | Kobaz | that's generally how it is, yes |
19:49.33 | wepy | actually i want to have a single number, where, if I'm at work, it rings my work phone, but if i'm elsewhere, it dials my cell phone |
19:49.46 | wepy | but it seems like i'd need to least traditional phone service for that ;\ |
19:49.54 | drmessano^ | Why? |
19:49.54 | wepy | s/least/lease |
19:50.00 | drmessano^ | Didnt we just go over this |
19:50.21 | darkrift_lp | ~itsplist-ca |
19:50.21 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
19:50.37 | wepy | if I used ITSP, it's call -> ITSP -> my asterisk -> ITSP -> cell phone|work |
19:50.39 | p3nguin | If you have Asterisk, you can use FollowMe or even basic dialplan with a sequence of Dial() commands. |
19:50.57 | wepy | hm |
19:51.26 | wepy | would latency be an issue if it all had to pass through the ITSP twice and my asterisk box once? |
19:51.58 | drmessano^ | And if you have Ma Bell, it's call > price inflated pass through ma bells switch > Phone or cell |
19:52.00 | p3nguin | Not really. Network latency is the only problem if you pass in and out of networks. |
19:52.29 | drmessano^ | $20 a month vs $75 a month is a no brainer |
19:52.31 | wepy | was hoping to save $ ;) |
19:52.37 | wepy | hah |
19:52.54 | drmessano^ | How are you gonna save money using Analog? |
19:53.20 | wepy | can't.. |
19:53.33 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
19:53.53 | dennis00 | so, where's Obelix? |
19:54.18 | wepy | how do you get service from bell? |
19:54.21 | wepy | or the like |
19:54.53 | drmessano^ | Same way you have for the last 100 years.. call them |
19:55.20 | drmessano^ | They probably have a hotline/priority queue just for new analog signups |
19:55.28 | wepy | so the phone company would perform ITSP services? |
19:55.33 | drmessano^ | No |
19:55.37 | wepy | i thought they were all analog ;) |
19:55.49 | wepy | oh |
19:56.43 | drmessano^ | The last thing you want is any sort of VoIP services from your traditional phone company |
19:57.37 | drmessano^ | Their single biggest expense is maintaining all that horrid 100+ year old infrastructure, and you get to share in that with all the saps still using analog lins |
19:57.39 | drmessano^ | lines* |
19:58.03 | drmessano^ | Find a nice ITSP and forget about that hole in your wall |
19:58.11 | wepy | heh |
19:58.56 | wepy | how do you know which ITSP's are good? |
19:59.10 | wepy | and do they usually let you test latency? |
19:59.24 | drmessano^ | Google is your friend.. try <name of provider>+sucks or <name of provider>+problem |
19:59.30 | wepy | haha |
19:59.33 | drmessano^ | Test how? |
19:59.42 | drmessano^ | Just ping their gateway |
19:59.49 | wepy | not sure, actually so much of it might have to do with my home ISP |
20:00.14 | wepy | drmessano^: so do you think I even need my own asterisk system? |
20:00.19 | drmessano^ | if you see one that looks good, try it.. if it doesnt work, dump em |
20:00.34 | drmessano^ | wepy: Only you can answer that |
20:00.36 | Kobaz | hmmmmm |
20:00.39 | Kobaz | pokes [TK]D-Fender |
20:01.26 | dlynes | drmessano^, actually...not quite true |
20:01.33 | dlynes | drmessano^, we save money by going analogue |
20:01.58 | drmessano^ | How so? |
20:02.12 | dlynes | We provide voip to the demarc room |
20:02.23 | dlynes | And then provide analog from there to the individual units |
20:02.41 | drmessano^ | What does that have to do with getting an analog line from Ma bell? |
20:02.46 | dlynes | Way cheaper to use existing cat 3 infrastructure than to run all new cat 5 cabling |
20:02.56 | wepy | hm |
20:02.58 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
20:02.58 | dlynes | you said analog...you didn't say it had to be from ma bell |
20:03.10 | drmessano^ | The entire conversation was about PROVIDERS |
20:03.15 | drmessano^ | Now HOW YOU WIRE THE INSIDE |
20:03.19 | drmessano^ | Not* |
20:03.19 | dlynes | we're a provider :) |
20:03.32 | dlynes | and we provide analog to the customer |
20:03.33 | wepy | has anyone heard of a good ITSP for the northeast USA? |
20:03.41 | p3nguin | sure |
20:03.49 | dlynes | they could care less whether it's voip before it gets to them, and for that matter, they don't even know |
20:04.03 | p3nguin | VoIP.ms and Flowroute are both fine and they serve all of USA. |
20:04.10 | wepy | thanks :D |
20:04.12 | drmessano^ | dlynes: and obviously you dont use ANALOG from ma bell, so what I said _IS_ entirely true |
20:04.29 | drmessano^ | dlynes: great, still out of context |
20:04.32 | dlynes | nah...but we've used analog from telus and bell telephone |
20:04.47 | dlynes | and they can still be cost effective...depends on your needs |
20:04.57 | drmessano^ | great, and I used to have an orange bicycle |
20:05.03 | dlynes | lol |
20:05.06 | dlynes | but seriously |
20:05.16 | dlynes | i wouldn't provide a business all voip lines |
20:05.26 | drmessano^ | He's one person wanting a line |
20:05.28 | dlynes | I'd use voip for their overflow and long distance |
20:05.54 | ChannelZ | 'cept flowroute seems not to be taking on new customers |
20:05.55 | dlynes | doesn't take much to get you riled up, does it? =) |
20:06.03 | drmessano^ | Maybe we should sell him on the merits of PRI too |
20:06.20 | dlynes | ChannelZ, vitelity's decent |
20:06.35 | dlynes | ChannelZ, I'm in Canada, and they're still not bad from here (they're in Denver) |
20:06.49 | ChannelZ | hmm thanks, I'm in Denver :) |
20:06.50 | wepy | can you use ipsec with any of the ITSPs? |
20:06.54 | ChannelZ | I knew of teliax |
20:07.03 | drmessano^ | dlynes: Well, when some people decide to jump in so they can make some point entirely out of the context of the conversation so they can look smart, it's annoying. Just sayin |
20:07.12 | dlynes | lol |
20:07.17 | p3nguin | If you're going to use Vitelity, you might as well go VoIP.ms since they seem to be giving lower rates and they resell Vitelity. |
20:07.41 | Kobaz | p3nguin: you can get lower rates if you go direct with vitelity |
20:07.55 | drmessano^ | Flowroute isnt taking on new customers? |
20:08.05 | dlynes | p3nguin, i'm getting pretty damned cheap rates from vitelity...can't see how you can get much lower than I already have |
20:08.08 | p3nguin | What do you mean by "go direct" with them? You pay, they give you service. |
20:08.17 | Kobaz | 0.8 cents a minute incoming |
20:08.21 | ChannelZ | drmessano^: well not through the website anyway |
20:08.27 | Kobaz | 1.5 cents with voip.ms, isn't it? |
20:08.31 | drmessano^ | Oh nice |
20:08.48 | drmessano^ | That doesnt sound good |
20:08.49 | titter` | [Jan 17 02:33:28] NOTICE[31198] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
20:08.56 | Kobaz | go direct instead of through a reseller |
20:09.04 | titter` | After that, the error went crazy |
20:09.11 | titter` | every 3 seconds |
20:09.15 | dlynes | titter`, is your d channel down? |
20:09.24 | titter` | it's up now |
20:09.25 | Kobaz | titter`: you don't have a rhino card by any chance? |
20:09.28 | titter` | no |
20:09.31 | Kobaz | that's good |
20:09.51 | titter` | it's a TE122 |
20:10.04 | dlynes | titter`, your d chan is up now, and it's still giving you those errors? |
20:10.12 | titter` | no error now |
20:10.16 | titter` | stoped asterisk |
20:10.19 | titter` | and restarted dahdi |
20:10.21 | titter` | and all is well |
20:10.24 | dlynes | titter`, ah |
20:10.31 | titter` | but want to know what caused it |
20:10.34 | dlynes | titter`, but given time, it happens again? |
20:10.42 | titter` | first time i've seen this lol |
20:10.42 | dlynes | titter`, usually caused by your d chan going down |
20:11.04 | titter` | it causes our pri to ring busy obviously, and my CIO flipped out |
20:11.08 | dlynes | titter`, but i've never had it spam me |
20:11.24 | titter` | Last time something like this happened, the server was hacked |
20:11.30 | dlynes | titter`, i've only ever had it happen sporadically, and only maximum three times in a row |
20:11.36 | drmessano^ | Voip.ms' termination is cheaper than Vitelity, as is origination for this area |
20:11.39 | titter` | That is when they let go the old PBX guy |
20:12.25 | titter` | dlynes: thanks. still trying to figure out what the hell that scheduler error is |
20:12.44 | titter` | dylnes: it started with the d chan going down, then it spammed that scheduler error over and over |
20:12.50 | dlynes | titter`, anyways...d chan going down is usually a problem on the CO end |
20:13.11 | dlynes | titter`, but if you're getting a scheduler error at the same time, I would think it's probably an error on your end |
20:13.18 | wepy | what's d chan? |
20:13.19 | dlynes | titter`, might even be a configuration error |
20:13.42 | *** join/#asterisk DelphiWorld (n=Miranda@41.104.103.211) |
20:13.43 | dlynes | wepy, it's where your error correction, did information, ... all comes in on for PRIs and BRIs |
20:13.47 | DelphiWorld | dlynes: i am back |
20:13.54 | dlynes | DelphiWorld, congratulations |
20:14.03 | DelphiWorld | dlynes: for what? lol |
20:14.08 | dlynes | shurg |
20:14.13 | dlynes | Just being facetious :) |
20:14.13 | titter` | dlynes: i'll tear down dahdi, and reinstall it ... it looks like this was converted from zaptel config files |
20:14.41 | DelphiWorld | dlynes: do you will by around tomorow? |
20:14.44 | titter` | dlynes: any suggestions for this ... any locations that dahdi lives besides /etc/dahdi |
20:14.52 | dlynes | DelphiWorld, possibly |
20:15.04 | Kobaz | man |
20:15.08 | dlynes | titter`, no idea...last time I worked with a PRI was when dahdi didn't exist |
20:15.10 | Kobaz | these asterisk bugs are getting weirder and weirder |
20:15.24 | dlynes | Kobaz, aren't they already weird enough? |
20:15.31 | drmessano^ | Kobaz: Aren't all bugs weird? |
20:15.34 | Kobaz | no, this is really weird |
20:15.40 | Kobaz | some bugs are straightforward |
20:15.53 | *** join/#asterisk klochan (n=klochan@78-106-111-127.broadband.corbina.ru) |
20:16.13 | dlynes | Kobaz, it's complaining about some problem with windows on a linux box? |
20:16.15 | *** join/#asterisk darkrift_lp (n=dark@65.92.170.72) |
20:16.35 | Kobaz | heh |
20:16.35 | Kobaz | what? |
20:16.45 | wepy | why isn't more of the world using sip@blah for dialing? |
20:16.46 | drmessano^ | I would think the general concept of "It shouldnt be doing that" would be classified as weird |
20:17.00 | wepy | instead of analog phone nubmers.. |
20:17.03 | dlynes | Kobaz, you said the bugs were weird...so i guess they're not that weird, after all :) |
20:17.06 | Kobaz | heh |
20:17.11 | drmessano^ | wepy: Because the old get old, and the young get stronger.. it may take a week, and it may take longer |
20:17.24 | drmessano^ | They got the guns, but we got the numbers |
20:17.31 | dlynes | wepy, because that would just be completely obtuse? |
20:17.36 | wepy | is there a way for vonage customers or skype users to dial sip@ "numbers"? |
20:18.00 | DelphiWorld | wepy: see skype for sip |
20:18.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
20:18.18 | Kobaz | all i've been doing all week is submitting bug reports |
20:18.19 | drmessano^ | Better yet, see Skype for Asterisk |
20:18.41 | Kobaz | https://issues.asterisk.org/view.php?id=16627 |
20:19.53 | wepy | maybe more companies using voip should announce their sip@ numbers |
20:20.02 | carrar | or maybenot |
20:20.06 | wepy | what do you call sip@ numbers? that saying feels old/dumb |
20:20.21 | wepy | sip addresses? |
20:21.26 | drmessano^ | SIP URI dialing |
20:21.32 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |
20:23.18 | Kobaz | what i find, is everything is all fine and good if you're doing simple stuff in asterisk |
20:23.31 | Kobaz | or stuff that you've done before, and it all works fine |
20:23.34 | titter` | well on a sidenote, I successfully got my asterisk to trunk to shoretel ... shoretel is pretty misinformed on asterisk, and the kb they provided was wrong |
20:23.39 | Kobaz | as soon as i try something new... i run into a million bugs |
20:23.52 | Kobaz | drmessano^: does that happen to you? |
20:24.06 | wepy | thanks.. |
20:24.08 | carrar | titter, yeah it would be even better it Shortel phones would work with Asterisk |
20:24.13 | carrar | it=if |
20:24.20 | Kobaz | every new project I start... or even any time i add a new feature to an existing project... i wind up submitting like 5-10 asterisk bugs |
20:24.32 | carrar | Shortel makes nice phones |
20:24.35 | wepy | maybe just 1 more question: I noticed that some ITSPs have DIDs in some countries, and others do not.. |
20:25.01 | dlynes | Kobaz, try doing blf and/or sla...you'll hit a whole raft of bugs there, too |
20:25.02 | wepy | is there a way to acquire a DID from one ITSP, then have incoming calls on that DID forwarded to an ITSP/asterisk box near home? |
20:25.11 | wepy | like combining DIDs from multiple ITSPs |
20:25.12 | titter` | Shoretel is nice, hardware is nice, some of the features are nice ... but the cost ... bleh. License per extension, per voicemail box, per sip trunk, on and on |
20:25.13 | dlynes | Kobaz, blf and sla were so broken for us, we had to abandon the project |
20:25.21 | Kobaz | yeah |
20:25.26 | Kobaz | i've had lots of problems with blf |
20:25.44 | dlynes | Kobaz, but it wasn't just asterisk that was broken |
20:25.44 | drmessano^ | Kobaz: I find a bug here and there, and sometimes something seldom used will get broken as hell, but overall I wouldnt say that anything complex is usually broken.. I will say that in the past there were far more issues with regression, but that's gotten much better over time |
20:25.57 | dlynes | Kobaz, our phone's firmware was broken w.r.t. blf, too |
20:25.59 | titter` | carrar: do you have a asterisk -> shoretel setup right now? |
20:26.05 | carrar | no |
20:26.10 | carrar | You doing it over SIP? |
20:26.18 | dennis00 | Can I set a callerid with noop? |
20:26.18 | dlynes | Kobaz, and the company moves like snails on bugs...and half the time don't admit it's an issue on their end |
20:26.21 | titter` | yep |
20:26.25 | Kobaz | heh |
20:26.40 | Kobaz | i need to hire a full time c coder whose job it is to fix asterisk bugs |
20:26.42 | carrar | titter, should be easy, I hear Shortel doesn't support authentication |
20:26.55 | *** join/#asterisk Deiz (n=swh@unaffiliated/deiz) |
20:26.58 | dlynes | Kobaz, digium's way better about fixing bugs than any hardphone manufacturers are about fixing firmware bugs |
20:27.05 | titter` | I setup a * server next to the shoretel switches so no nat is involed ... then use sip from asterisk to shoretel on the lan ... and iax to asterisk over the wan for the rest of my asterisk boxes |
20:27.08 | *** join/#asterisk DelphiWorld (n=Miranda@41.104.101.60) |
20:27.10 | Kobaz | polycom does a good job fixing firmware bugs |
20:27.18 | drmessano^ | Digium is better about fixing bugs than Microsoft |
20:27.22 | Corydon76-dig | dlynes: Hush, we need all the help we can get |
20:27.27 | Kobaz | who wants to move to central pa, and fix asterisk bugs for me |
20:27.28 | dlynes | Kobaz, aastra's good about fixing bugs that they admit exist |
20:27.38 | dlynes | Kobaz, but a lot of the time, they don't admit the problems exist |
20:27.41 | p3nguin | dennis00: No, but you can set Caller ID with CALLERID(num)= |
20:27.47 | dlynes | Corydon76-dig, ? |
20:28.10 | Corydon76-dig | would love it if 100 companies each hired coders to do nothing but work on and post patches for bugs in Asterisk |
20:28.17 | Kobaz | Corydon76-dig: i'm going to do that |
20:28.24 | Kobaz | Corydon76-dig: i'll be one of those companies |
20:28.36 | dlynes | Corydon76-dig, yeah..it would be nice...but at the same time, digium's still doing a good job with what they've got |
20:28.39 | drmessano^ | Seriously, lets count the number of bugs in Windows that show up as updates on patch Tuesday that go back to >>> earliest supported version vs those of Asterisk.. |
20:28.43 | dennis00 | p3nguin: in sip.conf or extensions.conf? not like this? exten => _X.,1,NoOp,${CALLERIDNAME} |
20:28.43 | Corydon76-dig | Kobaz: sweet, the community appreciates your assistance |
20:29.09 | Kobaz | Corydon76-dig: i appreciate the good access to the developers that's available.. it helps things move along nicely |
20:29.25 | Kobaz | but watch out... i'm gonna have you guys on speed dial soon |
20:29.29 | p3nguin | dennis00: In extensions.conf, Set(CALLERID(num)=12345) |
20:29.33 | Corydon76-dig | For 5 years, I worked for a reseller, fixing bugs and contributing features to Asterisk |
20:29.44 | wepy | i worked at a place that used shoretel |
20:29.50 | dlynes | Corydon76-dig, did you interpret what i said as digium bashing, or something? It wasn't, in case you misunderstood |
20:29.52 | p3nguin | dennis00: NoOp() does NOTHING. |
20:30.01 | wepy | it was alright, but we had lots of problems with cisco gear internally |
20:30.03 | Kobaz | p3nguin: sure it doesn't |
20:30.18 | Kobaz | p3nguin: NoOp prints stuff to the console output if you're in verbose mode |
20:30.19 | dlynes | p3nguin, well...not quite nothing |
20:30.19 | wepy | much less reliable than pots |
20:30.22 | titter` | Corydon76-dig: have you ever seen an error like this before http://pastebin.com/m25010808 |
20:30.30 | dennis00 | p3nguin: thank you. |
20:30.32 | dlynes | p3nguin, it allows you to insert meaningful status and/or error messages in your logs |
20:30.33 | Corydon76-dig | dlynes: I hear you, but I want to encourage, not discourage, outside companies to help work on Asterisk |
20:30.42 | dlynes | Corydon76-dig, ah |
20:30.44 | p3nguin | <PROTECTED> |
20:30.45 | p3nguin | [Synopsis] |
20:30.45 | p3nguin | Do Nothing |
20:30.45 | p3nguin | [Description] |
20:30.45 | p3nguin | <PROTECTED> |
20:30.48 | dlynes | Corydon76-dig, didn't realize i was discouraging them |
20:30.50 | drmessano^ | I found a random Avaya phone at work once. I sprayed the edge of the trash can with Lysol after disposing of it because the phone accidentally hit the rim on the way in. |
20:30.52 | Kobaz | p3nguin: that's a lie |
20:30.59 | p3nguin | kobaz: Then file a bug on it. |
20:31.10 | Kobaz | i think that's my job now |
20:31.15 | Kobaz | all i do is file asterisk bugs |
20:31.22 | Corydon76-dig | titter`: I'm just here momentarily. I have laundry to do today |
20:31.28 | p3nguin | dlynes: You should be using Verbose() for that. |
20:31.37 | Corydon76-dig | Big, massive piles of laundry |
20:31.42 | *** part/#asterisk DelphiWorld (n=Miranda@41.104.101.60) |
20:31.43 | titter` | Corydon76-dig: no hurry, it happened after this [Jan 17 02:33:28] NOTICE[31198] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
20:31.57 | titter` | It spammed over a million lines in my log file lol |
20:32.26 | dlynes | p3nguin, verbose is a dialplan command? |
20:32.32 | p3nguin | correct |
20:32.32 | Kobaz | application |
20:32.54 | p3nguin | Verbose(1,some stuff you want to see) |
20:33.11 | tzafrir | titter`, some lower-level error. e.g. bad line |
20:33.22 | dlynes | p3nguin, i.e. Verbose(Call coming in for the DID ${ARG1} for the customer ${ARG2} from ${ARG3})? |
20:33.22 | p3nguin | or Verbose(3,some stuff you don't want to see until verbose level is 3) |
20:33.27 | Kobaz | okay so, hmm |
20:33.30 | Kobaz | what else am i going to do today |
20:33.35 | dlynes | p3nguin, oh...so even better than noop then |
20:33.37 | p3nguin | dlynes: exactly |
20:33.43 | dlynes | p3nguin, I've been using noop for years :0 |
20:33.58 | dlynes | I guess verbose is a somewhat new dialplan application? |
20:34.00 | drmessano^ | NoOp is teh win |
20:34.06 | tzafrir | Verbose(3,...) is effectively NoOp, right ? |
20:34.19 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
20:34.38 | Kobaz | dlynes: it's "new" as of 2004 |
20:34.39 | Kobaz | r3581 | citats | 2004-08-05 22:12:54 -0400 (Thu, 05 Aug 2004) | 2 lines |
20:34.39 | Kobaz | Add app_verbose.c to cvs (bug 2212) |
20:34.53 | dlynes | ah |
20:35.03 | EmleyMoor | Is there any way to get Zoiper to show whether a contact on my box is on the phone or not? |
20:35.06 | dlynes | so it probably didn't exist in asterisk when i first started using it, then |
20:35.08 | wepy | what's a SIP trunk? |
20:35.12 | EmleyMoor | (IAX2 account) |
20:35.13 | p3nguin | wepy: nothing |
20:35.16 | Kobaz | ~siptrunk |
20:35.17 | infobot | well, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
20:35.19 | dlynes | but it's still pretty old |
20:35.20 | drmessano^ | NoOp is the ultimate man application in Asterisk.. "Sure, you can come over, but don't expect me to commit to anything or care" |
20:35.33 | wepy | cool |
20:35.39 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
20:36.00 | titter | tzafrir: thanks, could you elbaorate a little more ... I have seen a forum post that Digium has said this was a bug (Oct. 09), and it was fixed with a patch to libpri |
20:36.01 | Kobaz | wepy: when people say 'sip trunking' they mean, handling a bunch of calls over sip |
20:36.11 | drmessano^ | Woman: "I love you" -- Man: "NoOp()" |
20:36.28 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
20:36.32 | dennis00 | I see noop all the time. |
20:36.36 | titter | tzafrir: http://forums.digium.com/viewtopic.php?p=136339&sid=2c8ef807d7c32dc6a3ee5b21c9cad646#p138057 |
20:36.42 | tzafrir | titter, in most cases it's caused by a lower-level error (below libpri) |
20:36.47 | dennis00 | But maybe they just meant noob. |
20:36.49 | tzafrir | what version of libpri do you have? |
20:36.58 | drmessano^ | "Danny, did you break that??!!!??!!" "Um, NoOp()" |
20:37.17 | titter | 1.4.10.1 |
20:37.18 | p3nguin | dennis00: Just because you see it often does not make it the best application for the job. |
20:37.36 | dennis00 | p3nguin: I have changed it, I want my new sim card to test it. |
20:37.52 | tzafrir | Hmm... so we now also have Shawn Bright on -users. For a moment I thought it was Sean |
20:38.03 | Kobaz | wiggity |
20:38.17 | wepy | ~iax |
20:38.18 | infobot | hmm... iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
20:38.42 | dlynes | ~iax2 |
20:38.43 | infobot | well, iax2 is http://www.voip-info.org/wiki-IAX |
20:38.43 | p3nguin | I call it "eye axe" |
20:38.47 | tzafrir | titter, that's a different error |
20:38.59 | drmessano^ | p3nguin: It's EEKS |
20:39.02 | *** join/#asterisk LemensTS (n=customgt@71.86.32.146) |
20:39.04 | wepy | what's a good SIP-capable cordless phone for a home? |
20:39.15 | wepy | maybe witha few nice features, but most importantly, bug free |
20:39.16 | EmleyMoor | p3nguin: Isn't that a Dutch soccer team? |
20:39.19 | EmleyMoor | <g> |
20:39.35 | LemensTS | [Jan 17 14:32:16] WARNING[30560]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) <--thats normal if a sip phone is offline and you try to call it right? |
20:39.36 | dlynes | wepy, try siemens...my coworker's pretty happy with his |
20:39.38 | titter | tzafrir: looks the same to me, minus the line number -- http://pastebin.com/m25010808 http://pastebin.com/d57e37e5b |
20:39.48 | p3nguin | wepy: You can use any cordless phone you want... with an ATA hooked to it. |
20:39.54 | wepy | ok thanks |
20:40.03 | Katty | yawns |
20:40.05 | wepy | ~ata |
20:40.06 | infobot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
20:40.09 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
20:40.09 | p3nguin | wepy: An ATA will turn any phone into a VoIP phone. |
20:40.10 | wepy | effing acronyms :) |
20:40.21 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
20:40.26 | wepy | i don't have a regular phone ;) |
20:40.30 | wepy | i'm all IP at home |
20:40.40 | Kobaz | eyepeeee |
20:40.48 | p3nguin | So you want a cordless with a SIP base. Okay. |
20:41.06 | wepy | sip base.. |
20:41.18 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
20:41.40 | wepy | i'd like to be able to dial sip URI's and also regular phone numbers, but all of it would go through the ITSP i choose |
20:41.56 | p3nguin | For example, my cordless doesn't have an RJ-14 on it for a regular phone cord, it only has an RJ-45 for an Ethernet cable. |
20:42.22 | EmleyMoor | Can presence hints from Asterisk work over IAX2? |
20:42.44 | wepy | yea |
20:42.44 | p3nguin | yes |
20:42.51 | wepy | actually, wifi would be nice here.. |
20:43.00 | wepy | but cordless -> rj45 would also work |
20:43.12 | p3nguin | Now you've gone from a cordless phone to a Wi-Fi phone. |
20:43.33 | p3nguin | Most people don't like Wi-Fi and SIP together. |
20:43.37 | EmleyMoor | I have just started using Zoiper Communicator and presence working would be a definite plus point |
20:43.48 | titter | tzafrir: I will try libpri 1.4.10.2 based on this bug report https://issues.asterisk.org/view.php?id=15892 |
20:43.56 | titter | tzafrir: thanks |
20:44.03 | EmleyMoor | uses his N95 as a WiFi SIP phone, when it works |
20:44.23 | p3nguin | emleymoor: Just put the hint in your dialplan. exten => 2040,hint,IAX2/2040 |
20:44.29 | wepy | wifi's not stable enough? |
20:44.57 | EmleyMoor | p3nguin: Hmmm... already there but seems only to work over SIP |
20:45.24 | p3nguin | exten => 1234,hint,SIP/1234 |
20:46.09 | p3nguin | Either Zoiper isn't working with presence correctly or you've specified the wrong Tech in the dialplan. |
20:46.46 | EmleyMoor | I have hints for the two "user" extensions that can detect when the user is using any of their phones... but Zoiper doesn't see it |
20:47.20 | EmleyMoor | (X-Lite and recent ekiga do) |
20:48.09 | p3nguin | "core show hints" |
20:48.34 | drmessano^ | wepy: get a nice ATA like a Linksys PAP2 and a $20 DECT phone from walmart |
20:48.38 | p3nguin | The second column shows the tech/channel |
20:48.50 | drmessano^ | Win, win.. When the phone tech changes, toss it and get the next $20 phone |
20:48.57 | EmleyMoor | Shown truncated due to the length |
20:49.14 | p3nguin | Does it show IAX2/abc123? |
20:49.20 | p3nguin | or SIP/ |
20:49.32 | EmleyMoor | Yes and yes |
20:49.44 | p3nguin | hmm |
20:54.51 | *** join/#asterisk wepy (n=wepy@ip70-179-126-55.dc.dc.cox.net) |
20:56.49 | wepy | is you use a DECT phone and ATA, how can you make the phone dial sip URI's? |
20:56.58 | wepy | maybe asterisk can translate some special numbers into URIs? |
20:57.05 | EmleyMoor | core show hints shows that the hint changes to InUse if my partner's Zap phone is in use - but no sign of any meaningful presence readout at all in Zoiper |
20:57.14 | [TK]D-Fender | wepy: What kind of SIP URI's are you intending on dialing? |
20:57.47 | p3nguin | emleymoor: Sounds like a Zoiper problem, then. |
20:58.20 | EmleyMoor | Is there another IAX2 softphone for Linux that can do presence? |
20:58.20 | wepy | [TK]D-Fender: not sure, but I assume some day everyone will have one :) |
20:58.43 | p3nguin | So you wanted Zap presense, not IAX2 presense. I guess I misunderstood that. |
20:59.09 | p3nguin | And I can't spell presence. |
20:59.59 | EmleyMoor | p3nguin: One hint per user covers all phones each user has |
21:00.24 | EmleyMoor | (be they IAX2, SIP, Zap) |
21:00.44 | drmessano^ | Wepy: You can set up extensions in Asterisk to dial a user@host. When the day comes that we use URIs for phones it will likely be picking a name out of an address book, much like we do on smartphones, and clicking dial. |
21:00.52 | p3nguin | If a "user" has all those techs on his phone, Asterisk should be able to show hints for each tech. |
21:00.58 | Katty | this salad is so awesome. |
21:01.18 | EmleyMoor | ... or "phones of each of tth |
21:01.22 | Katty | it has balogna, almonds, quacamole, and snyder's honey mustard nibblers. |
21:01.27 | EmleyMoor | ... or "phones of each of those techs"? |
21:01.48 | p3nguin | I don't like balogna, so I would have to toss it. |
21:01.55 | p3nguin | tosses katty's salad |
21:02.02 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
21:02.08 | EmleyMoor | It's not so much what it shows for as what it shows to. |
21:02.55 | [TK]D-Fender | wepy: Stop inventing problems that don't exist and move on to something productive |
21:03.29 | Kobaz | hmm |
21:03.43 | Kobaz | this problem with chanspy not working when the channel isn't generating audio is a real pain |
21:04.28 | p3nguin | If the channel isn't generating audio, what would you listen to during the spying? |
21:04.32 | Kobaz | whisper |
21:04.38 | EmleyMoor | Silence? |
21:04.39 | Kobaz | whisper is also affected |
21:05.13 | Kobaz | it's not silence, it's a lack of audio frames |
21:05.25 | Kobaz | you can transmit silence |
21:05.25 | Katty | p3nguin: well, i was going to put chicken on it |
21:05.30 | Kobaz | and that's okay |
21:05.38 | Kobaz | but if there is no audio data, whisper just flat out breaks |
21:05.39 | Katty | p3nguin: but i nukerwaved it for too long and it tasted funny |
21:06.03 | EmleyMoor | Kobaz: What are you trying to achieve? |
21:06.21 | Kobaz | EmleyMoor: being able to whisper into a channel that may or may not be recieving audio |
21:06.44 | EmleyMoor | Kobaz: And if it's not, how do you expect to be able to? |
21:06.56 | Kobaz | by fixing the bug in chanspy |
21:07.23 | Kobaz | i need to find where the reads are done, and make them non-blocking |
21:07.51 | Kobaz | that's the plan anyway, probably will be more involved than that |
21:11.10 | dennis00 | p3nguin: I would like my callerid spoofing to work with budggetphone, but there probably is no way? |
21:11.52 | p3nguin | dennis00: If you set the CALLERID(num) on the channel when you Dial() out... if it does not work, then there is probably no way. |
21:12.29 | dennis00 | Too bad. |
21:13.37 | p3nguin | They might be like voipbuster. |
21:14.10 | p3nguin | You have to authenticate phone numbers before you can use them as outbound CID, and then you have to select the numbers from the list in their phone app. |
21:17.19 | [TK]D-Fender | [16:11]<p3nguin>dennis00: If you set the CALLERID(num) on the channel when you Dial() out... if it does not work, then there is probably no way.No, LOTS of other things can affect this |
21:17.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
21:17.53 | p3nguin | Let him know what else he can do... I'm sure he's interested if there is another way. |
21:18.13 | [TK]D-Fender | dennis00: pastebin your sip peer, and debug from a call attempt including changing the CID |
21:19.09 | dennis00 | ok |
21:31.17 | dennis00 | http://pastebin.ca/1755165 |
21:32.01 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
21:33.45 | wepy | voip.ms has a great web site heh |
21:34.02 | wepy | they explain the services very clearly.. love it |
21:34.25 | p3nguin | I use VoIP.ms for my toll-free DID and termination. |
21:35.34 | *** join/#asterisk ruied (n=ruied@bl7-220-227.dsl.telepac.pt) |
21:36.27 | [TK]D-Fender | dennis00: in [31107142866] add "sendrpid=yes" , "trustrpid=yes" and retry |
21:40.29 | dennis00 | [TK]D-Fender: same thing. |
21:44.18 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:49.09 | dennis00 | Can anybody a provider that does not block thijs? |
21:52.58 | Kobaz | hmm |
21:53.19 | Kobaz | splicing in audio with chanspy is not always nicely handled |
21:57.43 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
21:58.39 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
21:59.42 | dennis00 | I am looking for a voip provider that does not require mobile number and accepts callerid spoofing. |
21:59.44 | *** join/#asterisk p3nguin_ (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
22:03.24 | Kobaz | someone |
22:03.26 | Kobaz | er |
22:03.35 | Kobaz | someone broke originate in 1.6.0 svn :( |
22:08.17 | *** join/#asterisk ruied (n=ruied@bl7-220-227.dsl.telepac.pt) |
22:09.43 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
22:10.28 | *** join/#asterisk jhirley (n=jhirley@adsl-145-34-13.mia.bellsouth.net) |
22:14.12 | *** join/#asterisk Matt_A (n=matt@s15365979.onlinehome-server.com) |
22:15.31 | Kobaz | https://issues.asterisk.org/view.php?id=16628 |
22:15.37 | Matt_A | Hello, is it true that Freeswitch has a better architecture than Asterisk? |
22:16.29 | Kobaz | haha |
22:17.07 | Kobaz | it depends on the definition of better |
22:19.04 | *** join/#asterisk darkrift_lp (n=dark@65.92.170.72) |
22:23.05 | *** join/#asterisk jhirley_ (n=jhirley@adsl-161-65-146.mia.bellsouth.net) |
22:26.03 | titter | Should I recompile Asterisk after upgrading libpri |
22:26.08 | Kobaz | no |
22:26.37 | Kobaz | it's a dynamic library, and the ABI is the same |
22:27.08 | titter | Kobaz: thanks |
22:31.58 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
22:34.52 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
22:37.24 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
22:37.32 | Kobaz | man |
22:37.38 | Kobaz | all this bug reporting is making me hungry |
22:39.16 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
22:42.55 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
22:43.59 | *** join/#asterisk Raden (n=Raden@71.89.121.119) |
22:44.39 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.38.122) |
22:46.44 | *** join/#asterisk ruied (n=ruied@bl7-220-227.dsl.telepac.pt) |
22:47.34 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
22:53.34 | dennis00 | Are lots of people using Voipbuster with Asterisk? |
22:57.23 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
23:09.23 | *** join/#asterisk jhirley (n=jhirley@adsl-145-3-56.mia.bellsouth.net) |
23:11.56 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
23:38.56 | wepy | how much jitter is OK on a link that's about 200ms round trip? |
23:39.01 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
23:40.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
23:42.43 | *** join/#asterisk diatonic_afk (n=diatonic@mail.clearwater-research.com) |
23:48.22 | *** join/#asterisk p3nguin (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
23:48.58 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
23:49.18 | *** part/#asterisk wepy (n=wepy@ip70-179-126-55.dc.dc.cox.net) |