| 00:00.39 | edwar64896 | ? |
| 00:00.50 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:00.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:00.59 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
| 00:00.59 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 00:00.59 | *** join/#asterisk teleweb (n=buffy@dD577512A.access.telenet.be) [NETSPLIT VICTIM] |
| 00:03.12 | [TK]D-Fender | rob0 : Tru, but handing it to them up-front is a little different than trying to hack on in later. Criminal activity is spurred primarily by opportunity. |
| 00:16.11 | *** join/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 00:16.45 | Narkov- | any ideas why the manager API command Status returns all ActionID's and not just the specified one? |
| 00:17.20 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 00:19.16 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:22.53 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.141) |
| 00:23.04 | Druken | radio shark ? |
| 00:29.19 | UForgotten | video killed the radio shark? |
| 00:34.26 | [TK]D-Fender | Put all the blame on VCR |
| 00:34.45 | edwar64896 | IMHO the manager actionID is used to simulate 'blocking' within the API. |
| 00:34.56 | edwar64896 | therefore they should be unique across commands |
| 00:35.14 | Narkov- | bah...so how would one track a call started with originate? |
| 00:35.41 | CtRiX | no |
| 00:36.09 | CtRiX | ah misteries of manager api |
| 00:36.17 | Narkov- | lol...it seems so |
| 00:37.12 | [1]BigBadHoss | hey whats the difference between a SoundPoint IP 500 and a 501? |
| 00:37.13 | edwar64896 | how do you want to track it? |
| 00:37.14 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 00:37.31 | edwar64896 | The ActionID isn't a tracking mechanism. |
| 00:37.40 | Narkov- | i just want to know call status, ringing, "active" and completed/failed..etc |
| 00:37.47 | edwar64896 | it is simply used to tell the client when the function call initiated by the AMI has 'returned |
| 00:38.07 | edwar64896 | \ |
| 00:38.47 | edwar64896 | I'm doing this in TCL using the callerid and Source/Destination ID's coming back in the manager events. |
| 00:39.35 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 00:39.39 | Narkov- | damn hey...no better solutions? :( |
| 00:39.49 | edwar64896 | nope;. |
| 00:40.02 | edwar64896 | and I agree, it's a pain in the ass, but at least it works. |
| 00:40.22 | edwar64896 | I would prefer a more effective way of tracking calls from originate too. |
| 00:40.44 | Narkov- | all it needs is a unique ID tagged to the call at originate |
| 00:41.00 | edwar64896 | It might be possible to track this stuff using channel variables. |
| 00:41.17 | edwar64896 | and UserEvents? |
| 00:42.04 | Narkov- | hrmm |
| 00:42.13 | edwar64896 | you can set a channel variable on originate |
| 00:42.32 | edwar64896 | it might be possible to then display that channel variable when the events come through that relate to that channel |
| 00:42.43 | lunaphyte | is it actually expected that you run vmail.cgi as root? this is so annoying. |
| 00:42.43 | edwar64896 | you might need to patch channel.c to do it but it should be trivial. |
| 00:45.11 | Narkov- | i might specify a variable at originate then have the dialplan update status to mysql...messy :( |
| 00:46.15 | edwar64896 | what do you want the status information for? |
| 00:46.20 | edwar64896 | if you don't mind me asking? |
| 00:46.50 | Narkov- | to originate the right number of calls for our max trunk capacity |
| 00:47.08 | Narkov- | and to schedule retries..etc |
| 00:47.17 | edwar64896 | outbound callcentre? |
| 00:47.22 | Narkov- | correct |
| 00:47.30 | edwar64896 | predictive? |
| 00:47.47 | Narkov- | nope |
| 00:47.48 | [TK]D-Fender | [1]BigBadHoss : 501 has more memory for firmware mostly. You don't want to buy a 300/500/600 over their upgraded versions 301/501/601 |
| 00:48.11 | edwar64896 | limiting the number of calls on an outbound trunk is normally done using SetGroup |
| 00:48.14 | [TK]D-Fender | [1]BigBadHoss : IP 601 adds support for the Attendant Modules taht the 600 does not support |
| 00:48.15 | Narkov- | i want max bandwidth/trunk utilization without over/under shooting it |
| 00:48.41 | edwar64896 | that should be a fixed number then |
| 00:48.53 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 00:48.56 | edwar64896 | depending on your available bandwidth and level of risk ;-) |
| 00:49.08 | Narkov- | correct...but how many calls are active Now()? |
| 00:50.25 | Narkov- | ahhh...i get you...but i still need the outcome of the call...probably better handled through dialplan actions...hrmm |
| 00:50.35 | edwar64896 | I would take a look at the SetGroup page on voip-info and see if this inspires you. |
| 00:50.42 | edwar64896 | this allows you to count up the number of calls |
| 00:50.46 | edwar64896 | going into a particular context. |
| 00:51.01 | edwar64896 | you can then limit calls based on a check of the number of calls in that context at any time |
| 00:51.51 | edwar64896 | where r u in qld? |
| 00:52.31 | Narkov- | Brisbane, Northside |
| 00:53.31 | edwar64896 | sunnier than down here in Melbourne then! |
| 00:54.58 | Narkov- | brb |
| 00:59.23 | *** join/#asterisk teknoprep (i=teknopre@unaffiliated/teknoprep) |
| 00:59.41 | teknoprep | is BLA comming into asterisk's future? |
| 01:01.20 | mog | its already there |
| 01:01.21 | UForgotten | there's lots of blah's already |
| 01:01.24 | UForgotten | lol mog |
| 01:01.26 | mog | footy? |
| 01:01.37 | UForgotten | depends on the phone I s'pose |
| 01:01.39 | orlock | nuh |
| 01:01.55 | JT | football, in this case not the soccer type of football |
| 01:02.01 | orlock | i dont understand why people get so worked up about guys in shorts running through mud after a bit of leather |
| 01:02.20 | JT | i know |
| 01:02.23 | JT | incredibly boring |
| 01:02.34 | JT | watching braindead meatheats lumber around a field |
| 01:02.40 | JT | meatheads |
| 01:02.42 | UForgotten | people pay millions of dollars to support it, too. Same for hockey. |
| 01:05.10 | orlock | JT: http://jwr.d2p.com.au/CarStuff/torrie/spoiler/ |
| 01:06.39 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:06.53 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:24.08 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:32.06 | lunaphyte | is this message: dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) : possibly just the result of a particular phone not being on? |
| 01:32.27 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 01:32.34 | hads | lunaphyte: Yup |
| 01:32.38 | lunaphyte | ok |
| 01:34.23 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:37.25 | *** join/#asterisk yatesy_ (n=yatesy@unaffiliated/yatesy) |
| 01:37.35 | *** part/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 01:39.33 | *** join/#asterisk meshuga (i=meshuga@c-71-231-141-145.hsd1.or.comcast.net) |
| 01:39.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:43.41 | *** join/#asterisk dextro (n=dextro@24-155-108-234.dyn.grandenetworks.net) |
| 01:45.02 | dextro | I have been battling with this for a while now but have gotten no where; all of my calls to 'Console/dsp' show "Already have a call on the OSS channel"; chan_oss is loaded and show channels shows no calls... any ideas? |
| 01:45.02 | edwar64896 | anyone on yet that knows the dsp.c code ? |
| 01:45.43 | edwar64896 | could something have a hold of /dev/dsp ? |
| 01:45.51 | edwar64896 | or could it be a permissions issue on /dev/dsp? |
| 01:46.08 | edwar64896 | are you running asterisk as root or asterisk? |
| 01:49.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 01:50.59 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 01:53.25 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 01:55.08 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 01:56.46 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:57.50 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 01:58.16 | *** join/#asterisk angler_ (i=angler@nat/digium/x-cf7f236b46e8a20c) |
| 01:58.51 | axscode | just want to ask if TDM22B when installed.. if all 4 LED light is always ON? or it will only be ON if you plugged a phoneline!? |
| 01:59.19 | shmaltz | axscode, always on, as long as the modules are loaded |
| 02:00.08 | axscode | thanks, whats the problem if im calling and when the remote phone picks-up but it still ringing!? |
| 02:01.04 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 02:01.04 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 02:06.04 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 02:06.28 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 02:09.55 | riddlebox | has anyone ever worked on a vodavi starplus STSe? |
| 02:14.01 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 02:14.02 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 02:15.16 | *** join/#asterisk DEader (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 02:15.19 | DEader | hello |
| 02:18.25 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 02:23.56 | axscode | how do i create dialplan.. that when zap 3 is in use.. itll use zap 4... or zap4 is in use itll use zap 3 |
| 02:23.59 | axscode | :) |
| 02:24.21 | davidded | ww |
| 02:24.34 | axscode | ? |
| 02:24.47 | davidded | sorry, got wild with my keyboard ;) |
| 02:25.28 | davidded | but you can use chanisavail if the zap ata cards do support it |
| 02:25.36 | davidded | i think they should |
| 02:25.45 | davidded | i just recently wanted to do something similar, |
| 02:26.10 | davidded | however, zap bri cards cannot detect an already present bearer on a b channel before dialing... |
| 02:26.32 | axscode | im using TDM22B do you have any experience about it.;.. |
| 02:26.59 | wunderkin | use channel groups |
| 02:27.09 | davidded | no, sorry. actually, i only have isdn and sip equipment directly attached to my * |
| 02:27.29 | axscode | wunderkin: will you guide me a lil.... |
| 02:28.05 | davidded | wunderkin: ok, thx, i should have thought about this... that's how bristuff maps the two b-channels of an isdn line |
| 02:28.13 | axscode | in zapata.conf ... ill group them ? |
| 02:28.33 | wunderkin | axscode, in /etc/asterisk/zapata.conf, use group= and in extensions.conf use Dial(Zap/g1/num) |
| 02:29.02 | axscode | g1 ----> group = 1 in zapata ? |
| 02:29.39 | wunderkin | yes |
| 02:29.53 | davidded | yes, you'll have to specify the group=1 with your other settings before you finally map the channel |
| 02:30.35 | axscode | in extension itll be automatically alias as g1 ? or do i have to alias them? |
| 02:33.13 | wunderkin | try it and see |
| 02:34.45 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:34.55 | *** join/#asterisk r0d3nt|m (i=anonymou@tinfoilhat.net) |
| 02:35.57 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:39.49 | *** part/#asterisk UForgotten (i=uf0rg010@ip24-250-224-98.ga.at.cox.net) |
| 02:42.06 | axscode | the loadzone.. do it really needs to be there? im in the philippines.. i think there is none for ph |
| 02:44.59 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 02:46.45 | *** join/#asterisk davidded (n=davidded@p54B1DBA1.dip.t-dialin.net) |
| 02:46.51 | davidded | hi back |
| 02:51.15 | davidded | wunderkin: are you experienced with isdn? |
| 03:22.58 | *** join/#asterisk sumasuma (n=sumkji@61.14.86.23) |
| 03:23.20 | sumasuma | anyway to get pcm files ? |
| 03:23.37 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:23.51 | *** join/#asterisk ajungem (n=ajungem@200.119.237.59) |
| 03:24.59 | Sargun | sumasuma, from what |
| 03:27.00 | *** join/#asterisk evilmnky (n=evilmnky@216-106-185-169.ds1-static.mia1.net.ststelecom.com) |
| 03:30.34 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:34.59 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 03:43.41 | *** join/#asterisk blue`sky (n=crazy_gu@ppp-70-225-190-144.dsl.chmpil.ameritech.net) |
| 03:46.21 | sumasuma | Sargun: I want to play to my audio hardware directly from asterisk ! |
| 03:46.46 | sumasuma | without any encoding or decoding |
| 03:52.56 | *** join/#asterisk bmg505 (n=leon@c1-234-1.rndf.isadsl.co.za) |
| 03:55.59 | L|NUX | hello every one |
| 03:56.13 | L|NUX | cam some one take a look into this http://pastebin.ca/181568 |
| 03:56.19 | L|NUX | *can* |
| 03:57.26 | *** join/#asterisk mog_home (n=mogorman@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:57.26 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 03:59.43 | [TK]D-Fender | L|NUX : pastebin "cat /proc/interrupts" |
| 04:00.36 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 04:01.33 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 04:01.52 | _-Jon-_ | Hey everyone |
| 04:02.03 | _-Jon-_ | I'm having a little problem with getting my custom music on hold to work |
| 04:02.15 | L|NUX | wait |
| 04:02.29 | _-Jon-_ | Wait? |
| 04:02.32 | sumasuma | _-Jon-_: if the problem is little i guess you can solve it by yourself |
| 04:02.51 | L|NUX | [TK]D-Fender : http://pastebin.ca/181572 |
| 04:03.11 | _-Jon-_ | I'm sure I could if I dedicated enough time to it, however it's probably something so small that I'm just not seeing it |
| 04:07.29 | [TK]D-Fender | L|NUX : Do the same for "ztcfg -vvvv" |
| 04:14.04 | *** join/#asterisk hassler (n=hassler@cpe-65-31-37-112.woh.res.rr.com) |
| 04:14.40 | hassler | hi all! anyone familiar with BLF on a Grandstream GXP2000? |
| 04:15.11 | *** join/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 04:18.31 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:19.49 | L|NUX | sorry got dc |
| 04:20.26 | _-Jon-_ | Okay I have a new music on hold problem.. mpg123 seems to blast the music, not really sure why |
| 04:20.36 | L|NUX | [TK]D-Fender : this is the output of cat /proc/interrupts http://pastebin.ca/181572 |
| 04:20.36 | _-Jon-_ | I used quietmp3 |
| 04:21.03 | benjk | L|NUX, no zaptel drivers loaded |
| 04:22.51 | L|NUX | benjk : its loaded |
| 04:22.55 | L|NUX | let me show you bro |
| 04:23.06 | L|NUX | [root@korea asterisk]# lsmod | grep zaptel |
| 04:23.06 | L|NUX | zaptel 196868 1 wanpipe |
| 04:23.06 | L|NUX | crc_ccitt 6209 1 zaptel |
| 04:23.11 | [TK]D-Fender | benjk : He's running a Sangoma card, you won't see any Digium stuff in there |
| 04:23.24 | L|NUX | [TK]D-Fender : got me right |
| 04:23.24 | L|NUX | :) |
| 04:23.45 | [TK]D-Fender | L|NUX So wheres the PB I asked for? |
| 04:24.06 | *** join/#asterisk kronos003 (n=kronos00@c-68-40-112-183.hsd1.mi.comcast.net) |
| 04:24.07 | L|NUX | http://pastebin.ca/181572 |
| 04:24.24 | benjk | he still needs zaptel and his pastebin output didn't show it, so if he doesn't show the information required then of course he won't get correct responses |
| 04:24.54 | [TK]D-Fender | L|NUX : That wasn't what I last asked for.... |
| 04:25.12 | [TK]D-Fender | benjk : You won't see it there for a function Wanpipe setup.... |
| 04:25.21 | L|NUX | [TK]D-Fender : i am sorry but i hybernate my system |
| 04:25.31 | L|NUX | and moved my system from one place to another |
| 04:25.35 | [TK]D-Fender | [00:07] <[TK]D-Fender> L|NUX : Do the same for "ztcfg -vvvv" |
| 04:25.39 | L|NUX | ok |
| 04:25.43 | L|NUX | thx |
| 04:25.44 | benjk | chan_zap doesn't talk to wanpipe devices |
| 04:25.50 | [TK]D-Fender | L|NUX : Yeah i just saw the login/out that I hadn't noticed earlier |
| 04:26.15 | L|NUX | http://pastebin.ca/181577 |
| 04:26.20 | [TK]D-Fender | benjk : Its all transparent, and /proc/interrupts won't show Zaptel, trust me, thats fine. |
| 04:26.34 | benjk | still his info is insufficient |
| 04:27.15 | [TK]D-Fender | L|NUX : And "warouter status"? |
| 04:27.19 | [TK]D-Fender | wanrouter* |
| 04:28.26 | L|NUX | http://pastebin.ca/181579 |
| 04:28.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.23 | [TK]D-Fender | L|NUX : Damn.... everything looks fine..... try modprobing zaptel and restart * with "asterisk -gvvvvvc" |
| 04:29.33 | L|NUX | okies |
| 04:29.37 | L|NUX | ok |
| 04:30.22 | L|NUX | okies now its work |
| 04:30.29 | L|NUX | but another issue |
| 04:30.29 | L|NUX | :( |
| 04:31.06 | L|NUX | http://pastebin.ca/181581 |
| 04:31.29 | L|NUX | D channel down :( |
| 04:31.38 | benjk | check your cabling |
| 04:31.53 | benjk | red alarm means the card cannot see the network |
| 04:32.17 | L|NUX | thats another Channel :) |
| 04:32.23 | L|NUX | another span |
| 04:32.29 | L|NUX | but i am asking about which is connected :( |
| 04:33.12 | benjk | ah ok |
| 04:35.28 | benjk | check your console output when asterisk starts up |
| 04:35.43 | benjk | I am sure there is something in there relating to the pri channels |
| 04:36.01 | hassler | hi all! anyone familiar with BLF? |
| 04:36.46 | benjk | you also might want to check the special device files in /dev |
| 04:37.02 | L|NUX | ok |
| 04:38.27 | benjk | BLF? Bacon Lettuce Falafel? |
| 04:39.25 | wunderkin | milf? |
| 04:39.57 | [TK]D-Fender | Mostly Inert Liposuctioned Fat? |
| 04:40.27 | hassler | busy line function |
| 04:40.40 | hassler | although bacon lettuce falafel sounds good too |
| 04:41.03 | *** join/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 04:41.18 | L|NUX | benjk : there is no file with name w1g1 and w2g1 in /dev |
| 04:41.51 | bhrobinson | can anyone tell me what I am doing wrong that I cannot get my softphone to connect to my asterisk server? I am sure it is a NAT thing, but not sure what I need to add or do |
| 04:42.54 | [TK]D-Fender | bhrobinson : pastebin your sip.conf masking only passwords and describe the full network path between * and your softphone |
| 04:42.55 | [TK]D-Fender | ~pb |
| 04:42.59 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 04:44.09 | *** join/#asterisk Rahail (i=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 04:44.44 | *** part/#asterisk hassler (n=hassler@cpe-65-31-37-112.woh.res.rr.com) |
| 04:48.33 | bhrobinson | [TK]D-Fender: ok, pastebin is at: http://pastebin.ca/181586 |
| 04:48.56 | bhrobinson | my machine is on IP 10.25.2.182 going through a sonicwall 1260 |
| 04:49.06 | bhrobinson | the pub IP is 198.211.206.194 |
| 04:49.17 | bhrobinson | the server is on a public ip |
| 04:49.22 | bhrobinson | what else do you need? |
| 04:52.00 | bhrobinson | by the way, and please dont beat me up too much... this is a trixbox... I am trying to get away from them one by one... |
| 04:55.27 | [TK]D-Fender | bhrobinson : You didn't pastebin the INCLUDED files..... |
| 04:56.06 | bhrobinson | my bad... |
| 04:56.16 | bhrobinson | give me a moment please |
| 05:01.58 | bhrobinson | [TK]D-Fender: there is nothing in any of the 3 other files |
| 05:03.02 | [TK]D-Fender | bhrobinson : Congratulations, you are missing ALL of the settings you need for * to work behind NAT |
| 05:03.25 | bhrobinson | lol |
| 05:03.29 | bhrobinson | why thank you |
| 05:05.08 | [TK]D-Fender | under [general] add : |
| 05:05.11 | [TK]D-Fender | nat=yes |
| 05:05.16 | [TK]D-Fender | canreinvite=no |
| 05:05.42 | [TK]D-Fender | localnet=10.0.0.0/255.0.0.0 |
| 05:05.56 | bhrobinson | on the server? |
| 05:05.57 | [TK]D-Fender | externip=198.211.206.194 |
| 05:06.11 | bhrobinson | the server is on a pub ip though |
| 05:06.58 | [TK]D-Fender | bhrobinson : Wait, you mean * is public and the client is NAT'd? |
| 05:07.02 | bhrobinson | yeah |
| 05:08.05 | [TK]D-Fender | bhrobinson : You should still have "canreinvite=no" in [general] and need to have "qualify=yes" and "nat=yes" in your peer setup. |
| 05:08.45 | [TK]D-Fender | Also note that you have your softphone set up as a peer only which isnt' approprite for a phone. Should be "friend" |
| 05:09.30 | bhrobinson | you mean the ethos at the bottom? |
| 05:10.41 | bhrobinson | that is where I have a friend routing his long distance through my spare trunks |
| 05:10.59 | bhrobinson | and his b2bua has to do a reinvite on that |
| 05:11.47 | [TK]D-Fender | bhrobinson : No, reinvites are completely unnecessary and inadvisable. |
| 05:12.03 | [TK]D-Fender | And forget about reinvites behind NAT |
| 05:14.21 | bhrobinson | ok, I added the settings you suggested,and still no connection |
| 05:16.12 | [TK]D-Fender | bhrobinson :show it all, and what did you do to try to put it in effect? |
| 05:18.00 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 05:18.18 | stephane_ | jour |
| 05:19.23 | bhrobinson | on the ext I had all but the qualify matching, so I added that and reloaded the system. I then set my Linksys phone to point to it. It failed, so I did a sip debug on my ip address < front of the firewall> and have saved it to http://pastebin.ca/181593 |
| 05:21.12 | [TK]D-Fender | bhrobinson : I just asked you to repaste you new final configs and tell me what you think you did to put what you were to showme into effect..... |
| 05:22.10 | bhrobinson | I would be glad to, but 90% of it is in the trix/freepbx mysql setup... so it is not in the conf files... |
| 05:22.58 | bhrobinson | but I added to the extension in the web page the fields you were talking about... |
| 05:23.35 | bhrobinson | [TK]D-Fender: I loaded 6 trixboxes thinking it would "be the easy way" and I am having to go back and swap them all.. |
| 05:23.41 | [TK]D-Fender | bhrobinson : Will you just give the info that I asked for already....... |
| 05:25.25 | bhrobinson | ok |
| 05:25.30 | bhrobinson | one moment |
| 05:26.40 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 05:27.52 | bhrobinson | [TK]D-Fender: http://pastebin.ca/181599 |
| 05:29.01 | [TK]D-Fender | bhrobinson : And where is the "nat=yes", "canreinvite=no", and "qualify=yes" entries for your peers I told you to put in? |
| 05:29.24 | bhrobinson | since this is a trix install, it is parsed from the mysql |
| 05:29.47 | [TK]D-Fender | bhrobinson : NO. Those files are all that matters. Learn this and learn it FAST. |
| 05:30.15 | [TK]D-Fender | Trixbox is blowing your damned changes away and the MySQL garbage is meaningless. |
| 05:30.41 | bhrobinson | [TK]D-Fender: I know... that is why I was trying to tell you... I loaded 6 of these, and I am migrating them one by one |
| 05:31.00 | bhrobinson | the ONLY reason I cannot do it on the clean box, is that it is onsite |
| 05:31.14 | [TK]D-Fender | Fix those peer entries (which should be "friend" anyways), reload SIP and stop letting FreePBX overwrite your changes! |
| 05:31.21 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:31.32 | bhrobinson | which peer... the ethos? that is not my phone |
| 05:31.56 | [TK]D-Fender | bhrobinson : Where is your phone in that mess? |
| 05:32.15 | bhrobinson | it is not in there... that is what I am trying to tell you... |
| 05:32.30 | [TK]D-Fender | ... |
| 05:32.54 | bhrobinson | should I make a clean sip profile for an extension and just bypass the trix settings? |
| 05:33.03 | file | sweet, total power failure |
| 05:33.04 | [TK]D-Fender | bhrobinson : So basically you have spent this time showing me NOTHING? |
| 05:33.31 | [TK]D-Fender | What trix settings? There is no entry for your phone from what you now tell me. |
| 05:33.44 | bhrobinson | yes there is |
| 05:34.03 | [TK]D-Fender | bhrobinson : You'd better point it out pretty clear then.... |
| 05:34.12 | bhrobinson | are you familiar with trix? those settings are parsed with an odbc connect and generated on the fly |
| 05:34.25 | bhrobinson | so, if you log on to the ip... 216.85.121.28 |
| 05:34.33 | bhrobinson | use the stupid web crap |
| 05:34.41 | bhrobinson | you will see the extension info |
| 05:35.14 | Kerry_G | dont ever do that |
| 05:35.22 | [TK]D-Fender | bhrobinson : PM me the login |
| 05:36.21 | bhrobinson | Kerry_G: the ip? |
| 05:38.34 | *** join/#asterisk DEader (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 05:38.36 | DEader | hello |
| 05:38.48 | DEader | i know the trunk version of asterisk support video |
| 05:39.16 | DEader | is there a video playback command i can use to play back video when a user dial an extension |
| 05:39.58 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 05:41.59 | bhrobinson | not sure who the ass is that shut down my machine.. but that is gonna cost me an hour drive |
| 05:42.38 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 05:43.01 | DEader | ? |
| 05:43.18 | Kerry_G | never ever ever post a public IP with a default password, there are 251 people in this room |
| 05:43.41 | Kerry_G | I see it happen several times a day |
| 05:46.02 | Kerry_G | bhrobinson: you will be lucky if the passwords werent changed |
| 05:46.25 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 05:53.49 | tengulre11 | what's error ? Unable to open '/dev/zap/pseudo' |
| 05:55.51 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 06:03.16 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:03.21 | *** join/#asterisk revstray (n=rev@ip-d88bdd4b.shfw.dedicatedcentral.com) |
| 06:04.04 | revstray | g'day all |
| 06:04.17 | revstray | re: /topic, yes - we will all miss lilo |
| 06:05.34 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:08.42 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-137.range81-158.btcentralplus.com) |
| 06:18.01 | *** join/#asterisk oQPa (i=Spain@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 06:26.43 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:38.03 | *** join/#asterisk sdsf (n=x@phoenix.di-net.ru) |
| 06:38.06 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:38.15 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 06:47.33 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 06:48.43 | *** part/#asterisk ajungem (n=ajungem@200.119.237.59) |
| 06:50.20 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 06:53.01 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 06:53.08 | S^P | Hi, is it possible to configure asterisk SIP on two ot more ports? |
| 06:53.41 | zouzou | how to enable musiconhold when when pressing the hold key on the phone? |
| 06:55.36 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 06:55.36 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 06:56.28 | S^P | zouzou: I think its a phone feature, and it plays "music on hold" without sending any DTMF on channel. |
| 06:58.30 | *** join/#asterisk hermuli (n=Eladamri@a88-112-252-73.elisa-laajakaista.fi) |
| 06:59.13 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 06:59.23 | kaldemar | hello. |
| 06:59.40 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 07:00.49 | tzafrir | hi dorel__ |
| 07:00.53 | kaldemar | would anyone know if there's a way to restart individual channels on a TDM400P without restarting zaptel or asterisk? |
| 07:01.19 | tzafrir | kaldemar, using a little patch to chan_zap |
| 07:01.24 | dorel__ | hey |
| 07:02.05 | tzafrir | hmmm, actually it is part of 1.4 |
| 07:02.12 | kaldemar | tzafrir: certainly, do you know if someone has already implemented such a thing? |
| 07:02.20 | tzafrir | for 1.2, look at http://bugs.digium.com/view.php?id=6955 |
| 07:02.56 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 07:03.01 | kaldemar | thanks. |
| 07:03.03 | *** join/#asterisk h3x (n=hex@24.81.2.254) |
| 07:04.00 | tengulre | Hi,all |
| 07:04.11 | kaldemar | i'm really looking for a way to reload single channels, not all of them, but maybe that can be modified.. |
| 07:05.48 | tengulre | how spell it ,"sigoma"? for asterisk. |
| 07:06.05 | tengulre | singoma? |
| 07:06.11 | tengulre | singouma? |
| 07:06.13 | tzafrir | kaldemar, what for? |
| 07:06.23 | tzafrir | sangoma? |
| 07:06.29 | tzafrir | what hardware? |
| 07:06.30 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:06.35 | tzafrir | TDM400P? |
| 07:06.38 | tengulre | oh! Yes!! |
| 07:06.45 | tengulre | tzafrir: thanks |
| 07:07.22 | tzafrir | zap_restart will indeed disconnect all calls. However consider rewriting zapata.conf and then re-issuing zap restart |
| 07:07.36 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:07.49 | kaldemar | tzafrir: i have e&m channels that tend to fall and i need to get the channels back up without running the whole switch down. |
| 07:07.53 | tzafrir | you can always destroy a specific channel with "zap destroy channel" |
| 07:08.00 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-137.range81-158.btcentralplus.com) |
| 07:08.13 | kaldemar | tzafrir: and yes, TDM400P |
| 07:08.52 | tzafrir | TDM400P? the 4 port analog card? |
| 07:08.56 | kaldemar | yes. |
| 07:10.35 | tzafrir | A more selective "channel add" would require more work. I wonder if it would be possible to run zap_setup() with a mask of channels so it will just add those. |
| 07:10.42 | tzafrir | Still tricky. |
| 07:11.27 | kaldemar | indeed. i didn't expect it to be simple. :) |
| 07:12.07 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-137.range81-158.btcentralplus.com) |
| 07:14.57 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:15.13 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-137.range81-158.btcentralplus.com) |
| 07:15.18 | zouzou | s^P: my phone doest support music on hold,is ther a way to configure it in the PBX? |
| 07:16.12 | zouzou | musiconhold() coammand let you play music on hold but i need to get it when i press the on hold key |
| 07:16.21 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:16.58 | zouzou | how can i make asterisk recognise that i press the on hold key on my phone |
| 07:16.59 | Ahrimanes | morning |
| 07:18.24 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:19.20 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-137.range81-158.btcentralplus.com) |
| 07:19.21 | S^P | zouzou: I told you HOLD key on phone, never send any DTMF on channel. so there is no way to recognize it in PBX |
| 07:21.50 | JT | zouzou: you haven't even said what type of phone it is |
| 07:25.42 | oQPa | FXS or FXO ? |
| 07:26.09 | oQPa | for a normal telephone line |
| 07:28.49 | JT | oQPa: ?? what is the question? |
| 07:29.34 | oQPa | FXS = asterisk to asterisk or FXS = asterisk to Telephone line |
| 07:30.33 | tzafrir | zouzou, it should send "flash", right? or is it a SIP phone? |
| 07:30.37 | oQPa | FXO = pc to thelepone or pc to telephoneline |
| 07:31.04 | tzafrir | oQPa, who are you asking, exactly? |
| 07:31.12 | kaldemar | oQPa: FXO to telephone line, FXS to telephone |
| 07:31.41 | oQPa | this is the exactly response thanks kaldemar |
| 07:32.10 | oQPa | im see purchae a card |
| 07:32.17 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 07:32.51 | *** part/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 07:37.21 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:38.08 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:43.39 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 07:49.22 | zouzou | Its a sip phone: XLITE |
| 07:52.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:54.03 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 07:54.16 | zouzou | does xlite send sny dtmf on the channel if i press the hold key? |
| 07:54.24 | dyn | hi |
| 07:54.48 | JT | so it's a soft phone |
| 07:54.54 | JT | why would it send dtmf |
| 07:54.59 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 07:55.01 | JT | it's not an analogue line |
| 07:56.01 | dyn | last time I talked here someone who's distributing Billion ISDN cards |
| 07:56.10 | zouzou | so there is no ay to enable a music onhold in asterisk when i prees the hold key in xlite |
| 07:56.15 | S^P | zouzou: configure musiconhold class in asterisk, it will work. |
| 07:56.16 | dyn | I'd have some questions about them |
| 07:56.19 | zouzou | cause it will never recognise it?\ |
| 07:56.26 | dyn | or towards anyone familiar with Billion BiPAC cards |
| 07:56.52 | S^P | zouzou: it do and work perfect. |
| 07:57.26 | *** join/#asterisk UlbabraB (n=UlbabraB@81.74.72.205) |
| 07:58.22 | *** join/#asterisk Sephiroth (n=Sephirot@pdpc/supporter/active/Sephiroth) |
| 07:58.48 | *** part/#asterisk Sephiroth (n=Sephirot@pdpc/supporter/active/Sephiroth) |
| 08:08.14 | *** join/#asterisk diablopico (n=diablopi@ip68-101-147-222.sd.sd.cox.net) |
| 08:08.30 | diablopico | hello,,, anyone up for helping me with IAX |
| 08:09.09 | diablopico | socket_read: Call rejected by xx.xx.xx.xx: No authority found |
| 08:09.57 | benjk | your username/password is wrong or the context/extension doesn't exist |
| 08:10.21 | *** join/#asterisk cacrus (n=cacrus@213.132.40.2) |
| 08:10.27 | diablopico | thanks,,, i get a little confused with what side needs that and what doesnt |
| 08:10.43 | benjk | many of the error messages totally suck |
| 08:10.45 | kristalino | hi. Wich free windows softphone would you recommend ? |
| 08:10.58 | benjk | they throw you off more than they are helpful |
| 08:11.01 | diablopico | sjphone works for me |
| 08:11.16 | diablopico | thanks benjk |
| 08:11.37 | cacrus | I haev just bought a FXO card X100P from ebay , installed it in the machine , seup trixbox , But I can not make any out going call , it always says all the circuits are busy , pleast try again later " anyone faced this porblem ? I haev defined outgoing rout thru the trunk zap/1 |
| 08:12.09 | benjk | you can get a better idea of what's wrong by using the debug commands in the CLI |
| 08:12.19 | benjk | iax2 debug in this case |
| 08:12.20 | cacrus | I have normal analog phone line in the LINE jack of FXO card |
| 08:12.32 | JT | btw it's probably not a real X100P. cacrus |
| 08:12.37 | cacrus | yes its noty |
| 08:12.44 | cacrus | its clone , but its detected fine , |
| 08:12.56 | cacrus | and I don't have any Irq sharing on it |
| 08:13.08 | cacrus | I just wanna test it with one analog line |
| 08:13.17 | cacrus | can anyone help |
| 08:13.19 | JT | clones are often defective btw |
| 08:13.25 | JT | and this isn't really a trixbox support channel |
| 08:13.30 | diablopico | ok ,, i get a successful register from the * i send the call from to the * i send to |
| 08:13.45 | benjk | there are no clones, btw, its all the same, only difference is a heatsink to conceal the chip |
| 08:13.45 | diablopico | the call goes to the recieving , and gets rejected . |
| 08:14.14 | cacrus | so any help ? |
| 08:14.20 | JT | and some of the chips are rejects |
| 08:14.24 | JT | < JT> and this isn't really a trixbox support channel |
| 08:14.33 | benjk | diablopico, use iax2 debug to see more datail as to why it rejects the call |
| 08:14.48 | tzafrir | cacrus, does it show on 'zap show channels' ? |
| 08:14.53 | cacrus | but I thoughht it has to do with asterisk |
| 08:14.53 | diablopico | cool,, thanks benjk |
| 08:15.02 | cacrus | let me check |
| 08:15.06 | benjk | all of the cards sold today are either second hand, or made from refurbed or reject chips |
| 08:15.18 | benjk | because the chips are no longer manufactured |
| 08:15.22 | JT | cacrus: trixbox is not just asterisk, it has a lot of things on top which make it hard to support |
| 08:15.36 | benjk | in other words those cards are a gamble, hit and miss |
| 08:15.42 | JT | the error you gave is a trixbox error message, not asterisk |
| 08:15.54 | JT | actually |
| 08:16.00 | JT | you havent event specified the error |
| 08:16.04 | JT | -t |
| 08:16.04 | rob0 | FWIW I'm having a similar problem with an FXO clone on *. I've figured my card is hosed. |
| 08:16.29 | rob0 | <-- about to buy a TDM FXO module |
| 08:17.02 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:17.02 | benjk | the only thing those PCI modem cards are good for is to get a zaptel timing source |
| 08:17.21 | cacrus | btw I am very new to this asterisk this , please let me know if I need any other card , other than FXO X100P to simply use one line |
| 08:17.32 | tzafrir | benjk, heck, they're nice to get started with Asterisk |
| 08:17.39 | tzafrir | (for a testing system) |
| 08:17.40 | cacrus | tzafrir: I don't have zap command |
| 08:17.42 | benjk | get a sipura 3000 |
| 08:17.49 | JT | cacrus: does it need to be in the asterisk box? |
| 08:17.58 | JT | if not, an external sip box is cheapest |
| 08:17.59 | tzafrir | cacrus, chan_zap.so wasn't built? |
| 08:18.01 | JT | like benjk said |
| 08:18.02 | rob0 | Mine worked for awhile. Then just started crashing mysteriously. |
| 08:18.12 | tzafrir | Have you built asterisk on your own? |
| 08:18.15 | cacrus | No |
| 08:18.38 | cacrus | I have just installed trixbox while the X100P was installed inthe box |
| 08:18.41 | tzafrir | you run this from the asterisk cli, not the linux cli, righT? |
| 08:19.03 | cacrus | yes I use web FREEPBX |
| 08:19.22 | dyn | P[ 1] stack_init: Cannot add layer 4 to this port. <<<-- what does that mean? I get this error message from chan_misdn |
| 08:19.59 | benjk | dyn, you should have used BRIstuff as I told you |
| 08:20.03 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 08:20.05 | jeremy_g | hi |
| 08:20.10 | cacrus | I can really install th pound key asterisk distribution , but is there any guide to configure sinfle X100P and one analog line setup ? |
| 08:20.31 | JT | pound key asterisk distribution? |
| 08:20.50 | dyn | benjk: hi! I'm trying to follow about ALL instructions you guys are giving me, it's a tough task :) |
| 08:20.50 | rob0 | #* |
| 08:20.52 | jeremy_g | is there a nice editor for configuring * that can make things easy (like color coding for .confs, autocomplete :P,..) |
| 08:20.57 | cacrus | thsi used to be asterisk@home |
| 08:21.21 | benjk | cacrus, you shouldn't waste people's time with that crap, get proper equipment |
| 08:21.46 | rob0 | cacrus: there are instructions at the wiki, but a dead card is hard to resurrect. |
| 08:21.59 | jeremy_g | i am really tired of seeing the black screen and vi |
| 08:22.23 | benjk | dyn, I told you that BRI is the most straightforward way to get those cards working, pretty much out of the box |
| 08:22.44 | rob0 | cacrus: try stopping asterisk, unloading all relevant modules, reloading modules and restarting asterisk. |
| 08:23.28 | jeremy_g | is there even some tags for emacs |
| 08:23.47 | cacrus | benjk : I am anewbi to asterisk , all I need to know if there is any guide to setup single FXO for single analog line , |
| 08:23.48 | rob0 | Relevant modules: "rmmod -v wcfxo zaptel" |
| 08:23.51 | dyn | benjk: last time i was in the middle of installing it when someone pointed out it's useless, badly designed and obsolete and it's not the proper way doing things :( I aint have an openion here but I'm really losing the point among all the openions :) |
| 08:24.55 | benjk | and I told you that the very same people who told you that would call you a troll and kick you off this channel if you said this about zaptel |
| 08:24.56 | benjk | YET |
| 08:25.03 | benjk | BRIstuff is zaptel |
| 08:25.30 | tzafrir | benjk, and chan_capi . and app_segfault. and a couple of other things |
| 08:25.45 | benjk | fact is that the market leader and most experienced company for HFC BRI cards is Junghanns |
| 08:26.02 | benjk | and fact is that they wrote this stuff for those cards |
| 08:26.31 | benjk | fact is, it works out of the box with the card you have |
| 08:26.41 | JT | indeed |
| 08:26.48 | JT | asterisk development is primarily US |
| 08:26.58 | JT | where BRI use is exceedingly rare |
| 08:27.02 | dyn | okay |
| 08:27.19 | JT | and the only bri card from digium is not released fully yet |
| 08:27.51 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 08:27.53 | hank | hi |
| 08:28.44 | dyn | benjk: i get the very same message with asterisk-bristuff (the debian packaged version) |
| 08:28.50 | dyn | benjk: P[ 1] stack_init: Cannot add layer 4 to this port. |
| 08:29.12 | dorel__ | how do i implement an authentication based on the user entering a 4 digits number? |
| 08:29.13 | dyn | benjk: in spite of misdn is configured, loaded and detected my card according to dmesg output |
| 08:29.16 | benjk | have you removed all your previous ISDN drivers first? |
| 08:29.18 | *** join/#asterisk Stephnie (i=Stephnie@217.160.253.61) |
| 08:29.19 | Stephnie | hi |
| 08:29.38 | dyn | benjk: you mean misdn?! |
| 08:29.41 | JT | bristuff doesn't use misdn? |
| 08:29.43 | benjk | indeed |
| 08:29.44 | Stephnie | I have downloaded a g729 codec from digium ... |
| 08:29.49 | benjk | BRIstuff uses Zaptel |
| 08:30.10 | benjk | and you shouldb't be using the packaged version either |
| 08:30.19 | Stephnie | but still getting nothing when I do show translation |
| 08:30.25 | dyn | ok |
| 08:30.28 | benjk | BRIstuff has a script that does everything for you |
| 08:30.31 | Stephnie | do I need to register with digium first? |
| 08:30.34 | benjk | including cownload |
| 08:30.39 | benjk | download |
| 08:30.40 | JT | Stephnie: did you pay? |
| 08:30.46 | Stephnie | not yet |
| 08:30.49 | benjk | all you need is the initial bootstrap |
| 08:30.53 | JT | well there you go |
| 08:30.55 | benjk | BRIstuff does the rest |
| 08:31.18 | dyn | benjk: which script is that? i only see tarballs here at junghanns.net |
| 08:31.33 | Stephnie | what is the procedure by the way ... will it work after paying digium ? |
| 08:31.35 | benjk | and yes, zaptel has its downsides, BUT for a single BRI port, its good enough |
| 08:32.02 | JT | Stephnie: it's all in the voip-info wiki |
| 08:32.06 | JT | voip-info.org |
| 08:32.18 | Stephnie | ok... |
| 08:32.56 | *** part/#asterisk Stephnie (i=Stephnie@217.160.253.61) |
| 08:34.06 | zouzou | how can i uninstall asterisk? |
| 08:35.10 | rob0 | zouzou: that's a question for your OS. |
| 08:35.22 | rob0 | read your OS documentation. |
| 08:35.51 | benjk | you can use my uninstall script |
| 08:35.53 | zouzou | on red hat linux |
| 08:36.11 | rob0 | cacrus: /msg is not polite, but yes, what you want is possible. But those FXO cards are often junk. |
| 08:36.32 | zouzou | where can i found it |
| 08:36.43 | benjk | http://www.sunrise-tel.com/downloads/AsteriskShellScripts.zip |
| 08:37.18 | benjk | its the uninstall_asterisk.sh script, it's got built in help, will archive your configs/logs etc |
| 08:38.10 | zouzou | thanks man |
| 08:38.24 | benjk | it won't delete astman and it assumes the addmailbox script is in /usr/sbin |
| 08:38.58 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:50.36 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:55.39 | *** join/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 09:00.59 | *** part/#asterisk bhrobinson (n=brobinso@mail1.nt-it.com) |
| 09:01.28 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 09:01.48 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:02.19 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.52 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 09:05.01 | angryuser | hi everybody, i am unable to compile zaptel-1.4.0-beta1 i got http://pastebin.ca/181683, any ideas? |
| 09:05.13 | tzafrir | benjk, you don't need the addmailbox script |
| 09:06.55 | *** part/#asterisk cacrus (n=cacrus@213.132.40.2) |
| 09:09.09 | tzafrir | angryuser, can you give some 5 or so lines of log above the part you pasted? |
| 09:09.23 | angryuser | ok |
| 09:09.30 | tzafrir | BTW: to make make spew untranslated messages, try tomething like: |
| 09:09.36 | tzafrir | LANG=C make |
| 09:09.55 | tzafrir | or: LC_ALL=C make |
| 09:10.16 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 09:10.17 | angryuser | http://pastebin.ca/181689 |
| 09:13.21 | benjk | tzafrir, I didn't claim otherwise, all I am saying is that this uninstall script assumes the addmailbox script to be in /usr/sbin if it is there |
| 09:14.28 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:16.16 | S^P | angryuser: translate this for us :) make[4]: *** Pas de règle pour fabriquer la cible « 1.4/zaptel-1.4.0-beta1 ». Arrêt. |
| 09:16.43 | angryuser | No rule to compile « 1.4/zaptel-1.4.0-beta1 » Stop |
| 09:18.02 | angryuser | or No rule to compile the subject « 1.4/zaptel-1.4.0-beta1 » Stop, this one is better:) |
| 09:19.28 | angryuser | i got suse 10.1, did ./configure an make, got errors on make |
| 09:19.33 | S^P | Erreur 2: there should be another error before this one |
| 09:20.03 | angryuser | pasting all log |
| 09:20.07 | S^P | can you please paste all output. |
| 09:20.10 | S^P | k |
| 09:22.24 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 09:23.57 | angryuser | http://pastebin.ca/181701 all is here |
| 09:24.50 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 09:29.08 | S^P | anygryuser: did you try 1.2 version before this? |
| 09:30.21 | angryuser | i have a problem with the last version of 1.2, but a earlier version works fine for me |
| 09:30.50 | angryuser | anyway alls this 1.4 stuff was just to see new asteriks features |
| 09:32.23 | *** join/#asterisk hastru (n=asterisk@port-83-236-235-195.static.qsc.de) |
| 09:32.33 | hastru | hello all! |
| 09:33.27 | S^P | complied fine with me. |
| 09:38.51 | tzafrir | angryuser, have you figured this? |
| 09:38.59 | tzafrir | SUBDIRS=/home/aster/Desktop/aster 1.4/zaptel-1.4.0-beta1 |
| 09:39.24 | angryuser | hmmm |
| 09:39.55 | tzafrir | try changing in the makefile: SUBDIRS="$(PWD)" instead of SUBDIRS=$(PWD) |
| 09:44.28 | angryuser | not worked, i got http://pastebin.ca/181717 |
| 09:48.23 | angryuser | ok i succefully compiled:) i renamed aster 1.4 to aster1.4:) |
| 09:52.09 | MrChimpy | baa |
| 09:55.39 | *** join/#asterisk ceb` (n=qzxcd@paranoia.coldev.org) |
| 09:55.43 | ceb` | hoi |
| 09:57.14 | *** join/#asterisk oQPa (i=name@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 09:57.29 | Dr-Linux | tzafrir: a quick question |
| 09:58.37 | tzafrir | ask |
| 09:59.07 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 09:59.32 | hwt | hi. asterisk replies 500 Internal Error when calls with SIP FROM = anoynymous@domain.com. |
| 09:59.38 | hwt | is this normal behavior? |
| 09:59.50 | Dr-Linux | As we have implemented VLANs in Network. So we are facing some issue in dial out calls we receive following error when try to call out from VLAN 2 but from VLAN 1 it’s working perfectly. We can receive all the calls on soft phone but when try to call out found the following error so please look into it and do the needful. |
| 09:59.52 | hwt | is there away to accept such calls and show them as hidden or something? |
| 10:00.19 | Dr-Linux | Subnet for VLAN 1:- 192.168.0.0/24 |
| 10:00.19 | Dr-Linux | Subnet for VLAN 2:- 192.168.150.0/24 |
| 10:00.19 | Dr-Linux | <PROTECTED> |
| 10:00.46 | *** join/#asterisk h4mm3r` (n=h4mm3r@213-140-16-186.fastres.net) |
| 10:01.01 | Dr-Linux | tzafrir: please let me know if you have any clue? |
| 10:04.37 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 10:10.25 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 10:10.25 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) [NETSPLIT VICTIM] |
| 10:10.53 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 10:14.51 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:16.16 | *** join/#asterisk Kuto (n=kuto@125.60.241.24) |
| 10:18.01 | Kuto | has anyone tested asterisk without any fx* card? |
| 10:21.18 | benjk | of course |
| 10:21.20 | benjk | analog sucks |
| 10:21.45 | benjk | serious users will use BRI or PRI cards |
| 10:21.52 | benjk | and there are plenty of them |
| 10:22.04 | hwt | or just SIP. |
| 10:22.08 | benjk | or no hardware at all |
| 10:22.19 | benjk | VoIP only |
| 10:22.50 | Kuto | yes voip only without any hardware except sip |
| 10:26.58 | *** join/#asterisk Dr^Mouse (n=noneof@user-5445f864.lns4-c13.telh.dsl.pol.co.uk) |
| 10:28.15 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:29.52 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:37.01 | *** join/#asterisk slx (n=slx@88-149-154-95.f5.ngi.it) |
| 10:39.51 | slx | hello, asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1q on debian, i'm using chan-capi 0.3.6 on AVM Fritz!, everything works fine outbound and inbound isdn calls, what happen is that sometime incoming calls are not served by asterisk as expected, basically the remote party ears the telephone ringing but on this side the Answer() command won-t start... |
| 10:40.09 | slx | any suggestion? |
| 10:40.58 | slx | this happens always using a mobile with a particular carrier to call asterisk |
| 10:42.09 | benjk | I am a little confused by your statement "using chan-capi" while you also say you use BRI-stuff |
| 10:42.20 | benjk | cause BRIstuff doesn't use CAPI |
| 10:43.23 | benjk | if you did use BRIstuff, then you could do pri intense debug and see what exactly happens when those calls come in |
| 10:43.26 | slx | benjk, that's the BRIstuffed package from debian |
| 10:43.53 | benjk | well, all I can say is that BRIstuff does not use CAPI |
| 10:43.53 | benjk | no matter where you got it from |
| 10:44.16 | benjk | BRIstuff is based on Zaptel |
| 10:44.23 | slx | so you think my problem should belong to the BRIstuffed asterisk I use? |
| 10:44.46 | benjk | I don't know, but something in your statement is not right |
| 10:44.57 | benjk | if you are using CAPI, then you are not using BRIstuff |
| 10:45.06 | benjk | if you are using BRIstuff, then you are not using CAPI |
| 10:45.20 | dyn | mutually exclusive |
| 10:45.25 | dyn | i believe you're looking for that term |
| 10:45.28 | dyn | :) |
| 10:45.31 | benjk | well, its a different set of drivers |
| 10:46.03 | benjk | BRIstuff comes with drivers that conform to the Zaptel interface |
| 10:46.22 | benjk | and it also patches Libpri so it can be used for BRI |
| 10:46.44 | slx | the fact is Zaptel enable does denies to asterisk to use chan capi? |
| 10:46.46 | benjk | that way a BRI card looks to Asterisk like a PRI interface |
| 10:47.02 | benjk | or more precisely it looks like a Zaptel PRI interface |
| 10:47.28 | benjk | you cannot have two different drivers handle the same piece of hardware |
| 10:47.52 | benjk | and if you are using CAPI drivers, then Asterisk can only talk to them via chan_capi |
| 10:48.14 | benjk | BRI stuff provides Zaptel based BRI drivers and thus Asterisk can use chan_zap to talk to them |
| 10:49.05 | slx | benjk, I'm using fcpci and chan_capi regardless the fact the asterisk i use is bristuffed... |
| 10:49.26 | slx | sip works, iax works, capi works, certain carriers calls fails... |
| 10:49.52 | benjk | well, you are not using BRIstuff then |
| 10:50.08 | benjk | even if you are using the BRIstuff applied patches |
| 10:50.17 | benjk | you don't actually make use of it |
| 10:50.25 | slx | benjk, exactly |
| 10:50.27 | benjk | hence my comment that your statement is confusing |
| 10:50.50 | benjk | whether you have BRIstuff patches applied or not doesn't really matter so much |
| 10:50.52 | slx | benjk, lol you right, but I had to report my debian package in its full version naming |
| 10:50.58 | benjk | because you are using CAPI and chan_capi |
| 10:51.55 | benjk | of course it is possible that having the BRIstuff patches applied and using chan_capi may have some side effect that causes your problem, but I don;t think so |
| 10:52.22 | benjk | anyway, with chan_capi, it seems that your debugging options are fairly limited |
| 10:52.31 | kristalino | hi. Wich free windows SIP softphone can i use ? |
| 10:52.40 | benjk | I have seen this several times |
| 10:53.30 | benjk | people tried to debug a problem that I would have easily debugged with BRIstuff using the pri intense debug command, but they couldn't seem to find any such debug as to getting the same level of information with chan_capi |
| 10:53.44 | key2 | is there any app for asterisk to test the charge load |
| 10:54.13 | benjk | and some other folks who were present who actually knew something about chan_capi (I really dont) said that there was no easy way to get that info |
| 10:55.29 | slx | benjk, what pisses me off is the fact that apparently the calls are failing are setup in the same way as the one who do work |
| 10:56.30 | slx | benjk, if i call from my mobile everything is fine, other mobile with other carrier I can ear the NT1 plus ringing I see the call on the CLI but the Answer() it's just not passed to the call |
| 10:56.46 | slx | weird |
| 10:57.12 | *** join/#asterisk l-fy (n=l-fy@yate/developer/l-fy) |
| 10:57.18 | *** part/#asterisk l-fy (n=l-fy@yate/developer/l-fy) |
| 10:57.54 | slx | benjk, http://pastebin.com/793963, they end like this |
| 10:57.58 | benjk | that's why I say, if you were actually using BRIstuff, you could see what's happening by using pri intense debug |
| 10:58.09 | benjk | but with capi it appears that your visibility is limited |
| 10:58.39 | slx | benjk, ok, got the hint, I wipe out bristuff |
| 10:58.45 | slx | thanks ;0 |
| 10:59.16 | slx | s/bristuff/capi/ |
| 11:01.11 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 11:02.44 | benjk | I am not saying you need to replace anything |
| 11:03.01 | benjk | I am only saying that I know how to use zaptel related commands to debug this kind of problme |
| 11:03.13 | benjk | but that I don't know how to use chan_capi to debug it |
| 11:03.55 | benjk | and from earlier such encounters, I recall that nobody else knew either (even if they knew chan_capi well) |
| 11:04.20 | benjk | the debugging commands are channel specific |
| 11:04.28 | slx | benjk, trying to use debug CAPI, i'm noting that the ISDN SETUP is different |
| 11:04.37 | slx | *noticing |
| 11:04.38 | benjk | so since you use chan_capi, the commands I know to use, cannot be used |
| 11:04.56 | slx | debug capi, fyi ;) |
| 11:05.33 | benjk | like I said, we have have several encounters here where the information that the debug function in chan capi didn't provide the required level of detail |
| 11:05.41 | *** join/#asterisk Tili (n=tili@202.133.65.126) |
| 11:05.51 | benjk | but as I said, I know very little about chan_capi |
| 11:08.58 | *** join/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 11:11.35 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 11:13.19 | Narkov- | how successful are people with NVFaxDetect? |
| 11:14.44 | *** join/#asterisk resync (n=resync@unaffiliated/resync) |
| 11:15.04 | axscode | hi guyz.. just want to ask.. i have a TDM22B ... [1][2][3][4] where 1-2 are fxo and 3-4 are fxs module, i put a phone on port 2 and my pstn on port 3. now i tried dialing to zap/3 going to pstn call it rings and nobody picksup even its a pizza delivery number. and zap/2 i put a phone and it rings together with zap/3 |
| 11:15.32 | *** join/#asterisk potsboy (n=potsboy@vc-196-207-32-253.3g.vodacom.co.za) |
| 11:15.34 | resync | hi, anyone knows how to get the free upgrades for Asterisk Business Edition from Digium? |
| 11:16.22 | benjk | yeah, call/mail Digium |
| 11:16.38 | darkskiez | svn up ? |
| 11:16.41 | darkskiez | :) |
| 11:16.52 | resync | mm okay, thanks benjk :) |
| 11:16.55 | axscode | anyone good with TDM22B, can anyone help me please.. |
| 11:17.46 | *** part/#asterisk resync (n=resync@unaffiliated/resync) |
| 11:19.43 | potsboy | Hey all am planning to do my dcap, any advise as to what docs etc. I should run through? |
| 11:20.27 | Dr-Linux | benjk: do you have any clue on my question? |
| 11:20.37 | Dr-Linux | looks like it's kinda source port issue |
| 11:21.05 | benjk | what question? |
| 11:21.41 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:23.08 | Dr-Linux | we have implemented VLANs in Network. So we are facing some issue in dial out calls we receive following error when try to call out from VLAN 2 but from VLAN 1 it’s working perfectly. We can receive all the calls on soft phone but when try to call out found the following error so please look into it and do the needful. |
| 11:23.16 | axscode | <PROTECTED> |
| 11:23.21 | axscode | what does it means by 3-1 ? |
| 11:23.46 | Dr-Linux | axscode: first group 3rd channel/port |
| 11:23.57 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 11:24.07 | Dr-Linux | axscode: 1 is group 3 is channel number |
| 11:24.18 | Dr-Linux | benjk: did you saw the question? |
| 11:24.34 | axscode | oh.. but im dialing using zap/3/ |
| 11:24.45 | Dr-Linux | i see |
| 11:24.46 | axscode | i mean.. im dialing using Zap/g3 |
| 11:24.57 | *** part/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 11:25.16 | potsboy | Nobody have any advise on dcap material ? |
| 11:25.31 | axscode | <PROTECTED> |
| 11:25.57 | axscode | im dialing using /g3/ i have channel 3 and channel 4 as member of my g3 |
| 11:26.15 | Dr-Linux | axscode: how many cards you have? |
| 11:26.23 | RoyK | axscode: zap/3-1 means channel one in group three |
| 11:26.58 | benjk | and I thought it meant 3-1=2 thus channel 2 |
| 11:27.03 | Dr-Linux | correct |
| 11:27.10 | axscode | o.. thanks RoyK.... |
| 11:27.24 | Dr-Linux | benjk: what you thought about my question? :) |
| 11:27.56 | benjk | I think it is unreasonable to ask here |
| 11:28.13 | benjk | you have to do a little more footwork and ask more specific questions |
| 11:28.26 | Dr-Linux | well, |
| 11:28.28 | benjk | otherwise its like "hey can somebody please do my work" |
| 11:28.42 | axscode | i have TDM22B ... [fxo][fxo][fxs][fxs] in FXO module what should it be a phone or a phoneline? |
| 11:28.47 | benjk | besides, the info is really insufficient to answer anyway |
| 11:28.59 | Dr-Linux | benjk: i'm using 2 softphone Xlite and SJphone, same extension can dialout from xlite |
| 11:29.11 | Dr-Linux | but can't dialout from SJphone |
| 11:29.21 | MRH2 | hi - i am getting an NMI error with zaptel after a certain revision - any suggestions on how to narrow it down? |
| 11:29.23 | Dr-Linux | so not sure what's the probelm |
| 11:29.38 | benjk | Dr-Linux, sip debug is your friend |
| 11:29.48 | RoyK | Dr-Linux: it's because you're not supposed to fiddle with that during ramadan! pray, goddamn you! |
| 11:29.50 | Dr-Linux | benjk: i'm just guessing it's source prot problem |
| 11:29.54 | benjk | compare the successful session output with the unsuccessful session output |
| 11:29.58 | Dr-Linux | hhm.. good idea |
| 11:30.12 | potsboy | Mhr2, disable hyrer threading in bios |
| 11:30.29 | benjk | RoyK, I think you got the wrong idea about Ramadan |
| 11:30.31 | Dr-Linux | RoyK: ohh i just done my pray and i'm on fasting |
| 11:30.45 | RoyK | Dr-Linux: I'm just having fun :) |
| 11:30.47 | benjk | Ramadan != Sabbath |
| 11:30.49 | MRH2 | yeah its looks like a zaptel prob though |
| 11:30.52 | RoyK | I know i know... |
| 11:31.04 | Dr-Linux | Sabbath? |
| 11:31.10 | Dr-Linux | what's Sabbath? |
| 11:31.38 | benjk | yeah, the Jews have this thing called Sabbath and they cannot do any work then, not even switching on or off the light in the room |
| 11:32.09 | Dr-Linux | ohh |
| 11:32.11 | Dr-Linux | who is jew? |
| 11:32.16 | Dr-Linux | you? |
| 11:32.22 | Dr-Linux | i'm not a jew |
| 11:32.25 | Dr-Linux | i'm a Muslim |
| 11:32.26 | RoyK | Dr-Linux: our 'friends' the Jews' saturday prayer |
| 11:32.32 | Dr-Linux | and proud to be |
| 11:32.34 | RoyK | Dr-Linux: http://en.wikipedia.org/wiki/Shabbat |
| 11:32.39 | RoyK | Dr-Linux: calm down :) |
| 11:33.18 | benjk | Dr-Linux, that's why I said: Ramadan != Sabbath |
| 11:33.23 | Dr-Linux | benjk: please don't attach Ramdadan and Sabbath |
| 11:33.33 | Dr-Linux | benjk: oo i'm sorry |
| 11:33.39 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:33.44 | benjk | != is very much a detaching operator |
| 11:34.00 | Dr-Linux | benjk: yes i observe that later |
| 11:34.03 | Dr-Linux | also i'm not good in C |
| 11:34.10 | benjk | ;) |
| 11:35.00 | potsboy | Join #holy-war :) |
| 11:35.08 | benjk | heh |
| 11:35.14 | JT | at c |
| 11:35.16 | benjk | I think I abstain from that one |
| 11:35.16 | JT | or life... |
| 11:35.17 | MRH2 | vpm450m - is that the newer hardware echo can? |
| 11:35.47 | benjk | voice per minute: 450 million units |
| 11:36.03 | benjk | still doesn't make sense though |
| 11:36.09 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:36.12 | jeremy_g | hi |
| 11:36.19 | jeremy_g | things have changed |
| 11:36.39 | benjk | really? |
| 11:36.46 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 11:36.49 | axscode | <PROTECTED> |
| 11:36.51 | jeremy_g | in extension.conf, there is no more 'u' unavailability thingie |
| 11:36.58 | jeremy_g | :P |
| 11:37.31 | axscode | <PROTECTED> |
| 11:37.35 | jeremy_g | is it true? do we now do the basic checkings whether caller is busy or unavailable through different ways rather than having a u or b extension |
| 11:38.03 | jeremy_g | IsChanAvai() |
| 11:39.01 | jeremy_g | axscode:whats the problem |
| 11:39.06 | jeremy_g | axscode:what do you want to do |
| 11:40.35 | axscode | jeremy_g: i have a new TDM22B, arrived last friday... as on my current configs.. when i dial going out to pstn.. the fone rings at the end.. but when its picked up it still ringging. |
| 11:41.35 | jeremy_g | i know this problem |
| 11:42.18 | jeremy_g | let me try to recall |
| 11:42.58 | jeremy_g | axscode:stop reading tutorials, read that asterisk future of telephony book, it has the answer |
| 11:43.06 | jeremy_g | just do a find in the pdf document |
| 11:43.13 | jeremy_g | there was some option u have to change |
| 11:43.22 | MRH2 | any one know what was the previous hardware digium echo can refered to as - like in the 411? |
| 11:45.56 | *** part/#asterisk Narkov- (n=Narkov@c220-237-73-248.kelvn1.qld.optusnet.com.au) |
| 11:46.26 | axscode | jeremy_g: which part im supposed to read? |
| 11:46.57 | jeremy_g | axscode:just ctrl-f dial tone |
| 11:47.05 | jeremy_g | axscode:or beep |
| 11:47.50 | jeremy_g | oops, i didnt see ur above msges buddy, i do have pdf, if u need still, ask |
| 11:48.10 | axscode | 99% sir.. thanks.. |
| 11:49.17 | dorel__ | from asterisks' console, can i run a script command, like Dial("123123131")? |
| 11:49.22 | axscode | <PROTECTED> |
| 11:49.33 | axscode | dial tone? |
| 11:50.18 | axscode | oh.. something in the zone |
| 11:51.19 | tzafrir | dorel__, if you use 1.4, use originate |
| 11:51.40 | tzafrir | Dial is something to be run from the dialplan, not from the CLI |
| 11:52.20 | tzafrir | The dialplan is basically the "script" that an asterisk channel executes |
| 11:54.47 | Mw3 | if i'd like to use zapras to access my pbx from home when there's some network outage do i need isdn at home as well (i have a pri in my pbx)? |
| 11:54.57 | Mw3 | or i can connect with an analog modem from home |
| 11:56.00 | benjk | zapras only works with ISDN |
| 11:56.21 | Mw3 | so both end has to be isdn |
| 11:56.32 | benjk | however, you can easily set up a dial-in service so you can dial in over an analog line using a modem |
| 11:57.32 | dorel__ | right |
| 11:57.36 | benjk | all you need for that is a PCI modem or a serial interface with an external modem |
| 11:57.41 | dorel__ | ok ill create my own dialplan in a script then :) |
| 11:58.13 | Mw3 | benjk: that's nice, but i only have a pri connection to the pstn |
| 11:58.42 | benjk | might still be able to do it anyway |
| 11:58.56 | benjk | not with zapras though |
| 11:59.40 | Mw3 | hm, maybe i can do something with iaxmodem |
| 11:59.50 | benjk | yeah, I was just thinking about that |
| 12:00.16 | dorel__ | what would happen if i run Dial(ZAP/g0/054311010\r\n) on a pri? or other strings... |
| 12:00.18 | Mw3 | i'll try it |
| 12:03.24 | dorel__ | or something like Dial(ZAP/g0/054123456xjh#) |
| 12:03.30 | dorel__ | i wonder what will happen |
| 12:04.29 | benjk | it will give you an error because the number is malformed |
| 12:05.28 | benjk | also, not sure if you can have a group 0 |
| 12:07.04 | dorel__ | yeah i can |
| 12:07.32 | dorel__ | but i think when i had something like Dial(ZAP/g0/0541231231$) it dialed somewhere, no idea where though :) |
| 12:07.59 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:13.05 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 12:13.24 | fourcheeze | ugh |
| 12:13.45 | fourcheeze | someone here's arranged to port a fax number to our incoming provider |
| 12:13.50 | fourcheeze | without checking first |
| 12:14.17 | fourcheeze | what's the shortest route from asterisk 1.2.7 to support incoming faxes to some degree? |
| 12:14.29 | fourcheeze | don't really care how ATM |
| 12:18.19 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 12:20.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:25.38 | *** join/#asterisk shodan (n=shodan@216.113.99.88) |
| 12:26.41 | creativx | anyone here use ip10s from swissvoice? |
| 12:27.10 | shodan | a long time ago I read some article about building your own DSL isp , it said you could get a straight copper pair connection to someone else in your neighbourhood , is that true ? is there such a service available to end users ? |
| 12:27.39 | benjk | its called a dry line |
| 12:28.04 | benjk | it usually goes through your local exchange though |
| 12:28.47 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:30.52 | shodan | is that something I can just ask bell (bell canada , I'm in quebec) to install or will they just say "stfu noob go away" ? |
| 12:31.36 | benjk | telcos usually offer dry lines, but sometimes their sales people don't know |
| 12:31.59 | benjk | so if they tell you they don't have this, you have to be persistent |
| 12:32.32 | RoyK | anyone that knows if it's possible to somehow convert an ethereal trace to an audio file or two? |
| 12:33.47 | JT | cat ethereal_trace > /dev/dsp |
| 12:33.51 | shodan | and these are the kind of lines I can just plug a pair of dsl transceivers at each end and it'll work ? |
| 12:34.09 | JT | shodan: you need 4 wire HDSL modems or similar |
| 12:34.13 | JT | i have a small pile here |
| 12:36.28 | shodan | hmm I was going to go through all the trouble of installing a tower at both locations just to get a long range wifi link going , this looks a lot less trouble ! |
| 12:36.41 | shodan | JT, how much is that worth ? |
| 12:38.35 | *** join/#asterisk _ViperNetworks (n=Nitesh@65.48.63.178) |
| 12:41.16 | [TK]D-Fender | shodan: Yes you can get Dry-Line DSL in montreal, thats what I'm running on with a Sangoma S518 ADSL PCI card |
| 12:41.34 | [TK]D-Fender | shodan: Got to www.pubnix.net cand give them a call. Great place run by Unix geeks who seriously know their stuff. |
| 12:41.41 | benjk | note that dry lines have downsides |
| 12:42.27 | benjk | first, the telephone guy may accidentally use your line and give it to somebody else because it doesn't have a dialtone |
| 12:42.48 | benjk | secondly, dry lines are more prone to corrosion because they don't carry line voltage |
| 12:42.55 | shodan | [TK]D-Fender, what about in the lost backwoods of St-Liguori ? is that the kind of service you find anywhere there are phone lines , or only where you'll find dsl ? |
| 12:43.36 | [TK]D-Fender | shodan: DSL is DSL. If you can't get it normal, you can't get it period |
| 12:43.44 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 12:44.19 | benjk | D-Fender, not if you want to run your own DSL from A to B via the exchange |
| 12:44.27 | shodan | oh , so a dry line is dsl ? |
| 12:44.33 | benjk | because the exchange needs no DSLAM then |
| 12:44.48 | benjk | no, a dry line is a straight pair of copper from A to B |
| 12:44.54 | shodan | that's the lost hole where I live => http://maps.google.com/maps?f=q&hl=en&q=st-liguori+qc+ca&ie=UTF8&z=13&om=1&iwloc=A |
| 12:45.02 | [TK]D-Fender | shodan: Dry-line DSL is just a line with DSL sync on it but no dialtone for phone service |
| 12:45.12 | *** join/#asterisk baconbuttie_uk (n=bob@host-84-9-143-249.bulldogdsl.com) |
| 12:45.30 | baconbuttie_uk | 'lo |
| 12:45.34 | benjk | but he wants a dry line between two end points |
| 12:45.36 | [TK]D-Fender | shodan: Plein-milieu de nul-part! |
| 12:45.53 | coppice | dry line DSL is a huge pain |
| 12:46.10 | *** join/#asterisk ltd (n=z@202-161-29-79.dyn.iinet.net.au) |
| 12:46.15 | sxpert-work | dry line works well here |
| 12:46.17 | [TK]D-Fender | coppice: Not for me yet. |
| 12:46.30 | shodan | ok , that's a dry line I want , not dry dsl |
| 12:46.50 | key2 | [TK]D-Fender: u speak french ? :) |
| 12:46.50 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 12:46.50 | [TK]D-Fender | coppice: Flawless since install and not a bad thing to say about my provider. |
| 12:47.05 | [TK]D-Fender | shodan: You jsut want to rent the copper and run your own service on it from A to B? |
| 12:47.19 | coppice | because the telco monkeys can't hear anything when they stick a butt phone across the line they think its not in service and keep ripping it out for reuse |
| 12:47.39 | [TK]D-Fender | key2: Yes, I'm from Montreal, QC myself |
| 12:48.21 | shodan | [TK]D-Fender, yes exactly , as an alternative to a point to point long range wifi link |
| 12:48.23 | [TK]D-Fender | coppice: Never heard of over here since the telco itself offers dry-line service. |
| 12:48.52 | [TK]D-Fender | shodan: Ok, that falls under "leased line" territory and if they'll even offer it at all its cing to cost you. how long a distance? |
| 12:48.58 | [TK]D-Fender | going* |
| 12:49.04 | coppice | they offer it here, but the monkeys keep ripping them out |
| 12:49.15 | shodan | about 700 meters |
| 12:49.19 | sxpert-work | coppice: that sucks |
| 12:49.36 | JT | shodan: does your area have stormwater drains? |
| 12:49.40 | sxpert-work | shodan: go check with the local authorities if you can install something in the sewers |
| 12:50.09 | shodan | not all the way to location B |
| 12:50.27 | [TK]D-Fender | shodan: Yeah, 700M shouldn't be a big deal, but dispatching a tech for it if the service even exists (you KNOW they don't want to put your gear on their site) they are going to make you seriously pay for it... |
| 12:50.32 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:50.38 | mut | what is unemployment funded by? |
| 12:50.45 | sxpert-work | mut: profit |
| 12:50.52 | mut | .. |
| 12:50.52 | [TK]D-Fender | mut: "The Employed" |
| 12:50.57 | mut | i mean |
| 12:50.57 | sxpert-work | sorry, that's "generated by" |
| 12:51.01 | mut | yea |
| 12:51.11 | mut | taxes? |
| 12:51.13 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:51.20 | [TK]D-Fender | indeed. |
| 12:51.35 | key2 | [TK]D-Fender: weird, I never thought u could speak french :) and specially spelling it with the dashes |
| 12:51.35 | mut | hm |
| 12:51.45 | sxpert-work | mut: depends where... in the US, it's "tough luck dude, screw you" |
| 12:51.58 | [TK]D-Fender | key2: You'll never see my type with accents however ;) |
| 12:52.07 | mut | ? |
| 12:52.13 | shodan | benjk, how much should a dry line cost ? |
| 12:52.18 | key2 | [TK]D-Fender: "plein-milieu de nul-part" <- Shithole ? |
| 12:52.36 | sxpert-work | key2: yeah |
| 12:52.37 | mut | shodan: only cost me $14/mo |
| 12:52.37 | [TK]D-Fender | key2: "middle of nowhere" |
| 12:52.53 | sxpert-work | "shithole" is a perfect translation :D |
| 12:52.54 | mut | til verizon figured out we were running sdsl over it |
| 12:53.03 | [TK]D-Fender | My dry line costs me 5$ |
| 12:53.16 | mut | yea they're cheap |
| 12:53.32 | sxpert-work | murf: lol |
| 12:53.35 | sxpert-work | mut: lol |
| 12:53.36 | sxpert-work | even |
| 12:53.51 | sxpert-work | mut: what happened then ? |
| 12:54.00 | shodan | mut is that 14/month per end points ? |
| 12:54.07 | [TK]D-Fender | For $50 CAD tax-inc I get 5000/8000 (solid) unlimited bandwidth, fixed IP, on a dry loop |
| 12:54.14 | mut | cost $35/loop |
| 12:54.22 | mut | then they told us no |
| 12:54.38 | sxpert-work | mut: they can't do that, once the contract is signed, I guess |
| 12:55.08 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:55.19 | sxpert-work | mut: how expensive would be to pull a fiber across town via the sewers ? |
| 12:55.39 | mut | no idea |
| 12:55.41 | darkskiez | dont forget the repair work |
| 12:55.44 | coppice | depends if you have your own trained rats |
| 12:56.20 | sxpert-work | coppice: muahahaha |
| 12:56.38 | shodan | not an option here , I think I can rent space on the utility pole , but that's like 1$/per pole/per month |
| 12:56.56 | coppice | BT in the UK used to use mice to pull a thin nylon thread through pipework |
| 12:57.39 | sxpert-work | neat |
| 12:57.47 | shodan | I guess a rc car would do the job too hehe |
| 12:58.00 | sxpert-work | shodan: over 700m, that's pretty cheap... |
| 12:58.11 | *** part/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 12:58.25 | sxpert-work | shodan: that's about 50 bucks a month or so. |
| 13:00.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:00.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:00.27 | [TK]D-Fender | shodan: For 700M how many poles? I think that would probably be cheaper..... |
| 13:00.58 | [TK]D-Fender | shodan: Actually..... I'm SURE it will be cheaper. |
| 13:01.01 | shodan | 50$ that's too close to the price the local WISP would charge |
| 13:01.11 | sxpert-work | hah |
| 13:01.26 | sxpert-work | but with 50$, you can pull a fiber and have gigabit across |
| 13:01.40 | sxpert-work | ask that from the local WISP :D |
| 13:01.48 | benjk | use ip over carrier pigeon then |
| 13:01.55 | sxpert-work | hahaha |
| 13:01.59 | [TK]D-Fender | benjk: Nah.... smoke signals! |
| 13:02.11 | sxpert-work | that too |
| 13:02.14 | benjk | that wont work at night though |
| 13:02.32 | benjk | low bandwidth too |
| 13:02.33 | [TK]D-Fender | mouahahahhahahhahaha *cough* |
| 13:03.32 | sxpert-work | [TK]D-Fender: cough with the smoke, heh... told you to quit smoking |
| 13:04.03 | benjk | you said Canada, right? why not just dig a trench and roll your own fiber |
| 13:04.04 | mut | maaan |
| 13:04.21 | mut | friday we had a guy from lucent our here trying to sell us one of their lucent compact switches |
| 13:04.25 | mut | the thing is badass |
| 13:04.37 | mut | $340k price tag wasn't that nice tho |
| 13:04.50 | Damin | At NOTACON last year, I saw a demonstration of TCP over Tin can and String.. was good for about 40 baud. :) |
| 13:04.52 | [TK]D-Fender | mut: What kind of "switch"? |
| 13:05.03 | coppice | a light switch |
| 13:05.31 | sxpert-work | mut: lol |
| 13:05.31 | mut | class 4/5 telco |
| 13:05.40 | Damin | coppice: Meaning that once you turn it on, your lights go out because it sucks so much power. :) |
| 13:05.45 | sxpert-work | mut: and an * box probably does exactly the same, ;D |
| 13:05.54 | sxpert-work | benjk: rotfl |
| 13:05.54 | mut | you wish |
| 13:05.58 | Damin | sxpert-work: I wouldn't count on it. ;) |
| 13:06.12 | Damin | sxpert-work: It's hard to beat Lucent at the Switch game.. they practically invented it.. |
| 13:06.15 | sxpert-work | Damin: right, the * box would probably do more |
| 13:06.41 | shodan | [09:04:21] benjk you said Canada, right? why not just dig a trench and roll your own fiber <== uh ? I don't get it |
| 13:06.42 | benjk | the asterisk box will do more deadlocking thats for sure |
| 13:06.43 | Damin | sxpert-work: No way in hell. :) |
| 13:07.05 | benjk | shodan, never mind |
| 13:07.27 | sxpert-work | Damin: the lucent thing will deadlock your credit card for a very long time |
| 13:07.48 | benjk | the idea is that CA is less densely populated than many other places |
| 13:08.08 | coppice | I've met one or two very dense canadians |
| 13:08.09 | Damin | sxpert-work: It's a matter of scale. |
| 13:08.14 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 13:08.22 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:08.26 | sxpert-work | coppice: define "dense" :D |
| 13:08.27 | shodan | oh ok |
| 13:08.29 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:08.31 | De_Mon | Hrm.. woke up this morning and all my sip peers were UNKNOWN and none of them could dial in or out |
| 13:08.32 | Damin | sxpert-work: Asterisk is not the right answer to evey question.. ;) |
| 13:08.52 | Damin | coppice: Meaning that they are so dense they won't float in water? |
| 13:08.54 | De_Mon | restarting asterisk fixed it, but it'd be nice to know it woln't happen again 8) |
| 13:09.02 | sxpert-work | Damin: it's a matter of setting the * box the proper way so that the thing scales :D |
| 13:09.08 | benjk | coppice how many lb/ft3 |
| 13:09.23 | benjk | lb/ft^3 even |
| 13:09.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:09.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:09.39 | coppice | isn't .ca metric territory? |
| 13:09.42 | Damin | sxpert-work: Actually, when you really look at the costs of doing a carrier deployment, Asterisk starts to cost more when you get over 1,000 concurrent channels.. |
| 13:09.42 | sxpert-work | the "dense" refers to the bone contents of the cranial cavity ? |
| 13:09.52 | sxpert-work | Damin: hmmm |
| 13:10.00 | benjk | how many kg/m^3 |
| 13:10.06 | coppice | i remember they have km/h road signs to confuse the americans |
| 13:10.18 | angryuser | [TK]D-Fender: hi, i need to manage asterisk from remote machine, call transfer, state of lines/sip clients, any idea how? |
| 13:10.38 | De_Mon | angryuser flash operator pannel for one |
| 13:10.44 | Damin | sxpert-work: For example, if you do an objective comparison of the cost / port of a Cisco PRI gateway versus Asterisk w/ PRI cards, the Cisco actually comes out to be cheaper when you get to 8 ports. |
| 13:10.50 | mut | hm |
| 13:11.00 | angryuser | thank you |
| 13:11.01 | sxpert-work | coppice: no, more like the americans are confusing themselves by still using stupid imperial units despite the superior metric system being in existence for more than 200 years |
| 13:11.01 | benjk | http://www.gordonengland.co.uk/conversion/density.htm |
| 13:11.16 | mut | O_o |
| 13:11.31 | benjk | yeah, but it wasn't invented by an American |
| 13:11.34 | sxpert-work | Damin: is that only hardware, or did you factor in the CCM licenses ? |
| 13:11.44 | Damin | sxpert-work: And although you can do it, you shouldn't really run more than 4 PRI in a single Asterisk box. At that point, a 5450 or an APX 8000 w/ a DS3 card looks quite attractive! ;) |
| 13:12.01 | Damin | sxpert-work: Well, no one in their right mind would run CCM for Class 5 features.. |
| 13:12.11 | sxpert-work | Damin: how about using that sangoma DS3 card :D |
| 13:12.29 | Damin | sxpert-work: Show me one.. it's vapor still.. |
| 13:12.40 | benjk | You can use a Sangoma A108 for 8 ports |
| 13:12.43 | mut | heh |
| 13:12.54 | sxpert-work | right. that runs... 5K EUR or so |
| 13:13.02 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:13.08 | mut | i can run 2688 t1's off this thing |
| 13:13.18 | benjk | 3995 USD actually |
| 13:13.19 | sxpert-work | still cheaper than whatever similar thing dialogic^wintel^w(whatever) has |
| 13:13.38 | fourcheeze | how do I renegotiate a codec during a call? |
| 13:13.49 | fourcheeze | most customers are happy with g729 from my upstream provider |
| 13:14.00 | sxpert-work | mut: how expensive would that be (linecards & shit) ? |
| 13:14.00 | fourcheeze | so I want to use that wherever possible and pass it through |
| 13:14.06 | fourcheeze | however some want g711 |
| 13:14.09 | angryuser | Flash operator panel is stable? no crashes? |
| 13:14.36 | fourcheeze | so for particular extensions I want to send back a message to the provider "resend in g711" |
| 13:14.37 | mut | lets see |
| 13:14.39 | fourcheeze | how can I do this? |
| 13:14.48 | fourcheeze | can sip handle that? |
| 13:14.50 | mut | what his lil summary shows is.. |
| 13:14.57 | mut | k nvm i was lookin at the wrong number |
| 13:15.04 | Druken | 2688 t1's? |
| 13:15.07 | mut | up to 448 ds1's |
| 13:15.11 | mut | 120 ds3's |
| 13:15.33 | mut | ~21,000+ voip calls |
| 13:15.40 | mut | shit i don't own it |
| 13:15.49 | mut | but one of these days |
| 13:15.53 | Damin | sxpert-work: Well, consider that if you want to use Asterisk, you have no valid 723 licenses, you'll have to buy 729 licenses and you won't get solid T38 (yet). Then, consider the machine and the engineering time to load it in and doing a cost comparison of using Asterisk as a TDM gateway versus a Cisco shows that the Asterisk solution costs just about the same, but you lose many features.. You gain flexibility, but you lose some real important things lik |
| 13:15.54 | benjk | angryuser, FOP is, but the Asterisk Manager interface through which it connects isn't |
| 13:15.55 | mut | it will be mine! |
| 13:16.07 | Bobcat991966 | Damin, Im with you on this one, when it comes to a large call Centre Lucent does what it does very well and me personally I would not put * in the same league but then again there is something to be said for an open archatecture. Its a trade off and * is not right for every circumstance just like Avaya is not the answer for everthing. |
| 13:16.18 | sxpert-work | Damin: why would you need G723 or 729 for ? |
| 13:16.22 | mut | $340k sticker price is bit much for our market right now |
| 13:16.47 | Damin | sxpert-work: because if you are a carrier, you want to terminate traffic from as many possible locations you can! :) |
| 13:17.20 | fourcheeze | sxpert-work: you want g729 so that you can get adequate quality with low bandwidth |
| 13:17.22 | Damin | sxpert-work: I do millions of minutes of month using 723 from Indian call centers.. Why? because the 723 license that ships w/ Windows allows them to do some pretty insane things.. |
| 13:17.31 | Druken | mut: gee... ya think ? |
| 13:17.40 | mut | :P |
| 13:17.41 | sxpert-work | Damin: hmmm. my carrier uses G711 alaw :D |
| 13:17.48 | FlatFoot | hello all any good debian peeps in today |
| 13:17.49 | Druken | i'd pay that for a house... not something to go in my house... :) |
| 13:18.17 | mut | yea.. |
| 13:18.30 | FlatFoot | exit |
| 13:18.31 | mut | it'd go in our SBC CO |
| 13:18.35 | Damin | sxpert-work: In any case, you do not do Asterisk any favors when you pitch it into a situation that it's just not designed for. It was designed as a PBX, not a Switch. :) |
| 13:18.37 | FlatFoot | ooops wrong screen |
| 13:19.09 | Damin | sxpert-work: It's not branded as "Asterisk: The Open Source Switch" ;) |
| 13:19.34 | benjk | it actually says "Asterisk, a telephony toolkit" in each file |
| 13:19.36 | Damin | sxpert-work: That being said, Asterisk for the Small Business is a completely kick-ass model, and it rocks as an application server... |
| 13:21.16 | fourcheeze | FlatFoot: got a debian problem? |
| 13:21.58 | sxpert-work | Damin: a switch is just an oversized PBX :D |
| 13:22.22 | sxpert-work | also, you can use lots of small asterisk boxes connected on a big ethernet swithc :D |
| 13:22.27 | benjk | a switch is a pbx without deadlock issues |
| 13:22.40 | fourcheeze | what is a dealock issue? |
| 13:22.40 | benjk | thus asterisk doesn't qualify |
| 13:22.57 | Damin | sxpert-work: You do not understand the cost structure or investment. If you did, you wouldn't keep harping on this. |
| 13:23.08 | benjk | its when your server decided to stop all work and wait for itself |
| 13:23.22 | Damin | fourcheeze: A situation where something in the core causes all call processing to cease. |
| 13:23.39 | Damin | fourcheeze: Like what happens when I try to run more than 230 concurrent SIP calls through an Asterisk box. |
| 13:23.54 | sxpert-work | benjk: that's just a bug. |
| 13:23.55 | Juggie | i've seen it happen w/ 1 iax call |
| 13:23.58 | Damin | fourcheeze: Deadlock means "It's dead Jim" |
| 13:24.00 | benjk | no its not |
| 13:24.03 | sxpert-work | benjk: 5ESS crashing has been seen |
| 13:24.10 | benjk | its a systemic problem |
| 13:24.17 | Damin | sxpert-work: it's a fundamental design flaw actually.. |
| 13:24.17 | sxpert-work | same goes for Alcatel E10B3s used around here |
| 13:24.20 | Bobcat991966 | I dissagree Expert-Work. A switch is just for moving calls around not nessassarily for VM, Sreenpops, Call recording (Nice Systems) integration with Work Force Managment. I have installed 500 million calls a year call centers and have never installled a swith. |
| 13:24.22 | benjk | indeed |
| 13:25.02 | Juggie | 500million calls, not bad :) |
| 13:25.26 | Damin | sxpert-work: In a true carrier environment, one call does not have the power to affect anything other than itself. :) |
| 13:26.00 | jtexter3 | I believe the difference between a switch and a PBX is just semantics. I've seen the same piece of Avaya equipment being used as a switch, and as a PBX, it's just all in how you configure it. That's how I see * |
| 13:26.02 | sxpert-work | Damin: hmmm... |
| 13:26.17 | sxpert-work | jtexter3: semantics and sticker price :D |
| 13:26.23 | Damin | sxpert-work: But again.. I love Asterisk, support it, use it, advocate it.. but in the right situations.. When you try to push it into a realm that it is just not designed for, you are going to create more problems.. |
| 13:26.25 | benjk | and no deadlocks |
| 13:26.51 | Juggie | benjk, feel free to fix them |
| 13:27.03 | benjk | we'll do |
| 13:27.13 | sxpert-work | Damin: well, tell you what. my DSL carrier uses cirpack stuff for switches. despite it priced accordingly, it fsck up pretty badly rather often |
| 13:27.21 | Damin | Juggie: Some of the issues require a significant change to the core design, much of which is on the table for 1.5 and beyond.. |
| 13:27.29 | benjk | not in a way that Digium would accept though |
| 13:27.30 | Damin | sxpert-work: Cirpack != Lucent. ;) |
| 13:27.37 | Juggie | Damin, i know i understand what causes a deadlock :) |
| 13:27.50 | sxpert-work | Damin: heh, well... |
| 13:28.01 | Damin | sxpert-work: And your DSL provider obviously sucks ass. :) |
| 13:28.24 | sxpert-work | Damin: not really. as long as you only consider the IP connection, they're the best around here :D |
| 13:28.50 | Damin | sxpert-work: Well, it's been fun, but I have to go look at a customer issue.. :) |
| 13:28.56 | sxpert-work | hah : |
| 13:29.07 | sxpert-work | Damin: have fun hacking your switches |
| 13:31.51 | benjk | Juggie, that's simple: locks cause deadlocks |
| 13:33.47 | Juggie | no kidding :P |
| 13:33.59 | sxpert-work | benjk: depends the way they're used :D |
| 13:34.00 | benjk | remove all the locks, no more deadlocks |
| 13:34.24 | sxpert-work | benjk: more crashing though, when carpet gets pulled from under threads |
| 13:34.30 | benjk | no threads |
| 13:34.50 | Juggie | no threads? |
| 13:34.50 | sxpert-work | what's the point of locks if no threads are used ?N |
| 13:35.16 | benjk | well, no pthreads to be more precise |
| 13:35.29 | sxpert-work | hmm ? |
| 13:35.54 | sxpert-work | what sort of threads are used then, if not pthreads ? |
| 13:36.04 | benjk | coroutines |
| 13:36.08 | sxpert-work | gah... |
| 13:36.10 | sxpert-work | no wonder |
| 13:36.21 | benjk | Erlang uses coroutines, scales magnificently |
| 13:36.45 | benjk | there are a bunch of C coroutine libs too |
| 13:36.59 | sxpert-work | does that make efficient use of multicores ? |
| 13:37.17 | benjk | depends on the implementation |
| 13:37.21 | sxpert-work | hmmm... |
| 13:37.45 | sxpert-work | does coroutine give anything that the pthreads don't ? |
| 13:37.54 | benjk | the von Behren paper presented at Usenix 2003 is good reading on this |
| 13:38.33 | benjk | its a little confusing at first because they use the term threads (coroutines are sometimes called user-level threads) but their implementation uses coroutines |
| 13:38.47 | sxpert-work | hmm. |
| 13:38.53 | benjk | coroutines are lightweight, about the same cost as a function call |
| 13:38.59 | benjk | and they need no locks |
| 13:39.16 | benjk | they are deterministic, you always have a defined state |
| 13:39.19 | sxpert-work | benjk: they're not really multiprocessing though |
| 13:39.26 | benjk | never any in-between states |
| 13:39.45 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 13:39.48 | benjk | like I said, depends on the implementation |
| 13:39.55 | sxpert-work | in the time of windows 3.1, this was called "cooperative multitasking" |
| 13:40.23 | benjk | Ericsson have done some good work on multi CPU systems running Erlang based software |
| 13:40.30 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:40.41 | benjk | that's a different issue though |
| 13:40.46 | coppice | Win 3.1 is *not* co-routines |
| 13:40.58 | benjk | using coroutines for different *processes* is not a good idea |
| 13:41.12 | benjk | using coroutines for different threads is |
| 13:41.39 | benjk | processes don't share data, that's a case for preemptive scheduling |
| 13:41.53 | sxpert-work | pthreads share data |
| 13:42.05 | benjk | precisely my point |
| 13:42.08 | sxpert-work | and they allow for pre-emptive scheduling |
| 13:42.18 | benjk | precisely thats the rpoblem |
| 13:42.21 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 13:42.29 | benjk | because you get non-determinism |
| 13:42.33 | sxpert-work | I fail to see where there's a problem here |
| 13:42.46 | sxpert-work | and ? |
| 13:42.52 | benjk | non-deterministic behaviour is the problem |
| 13:43.01 | sxpert-work | it's a matter of using mutexes on data properly |
| 13:43.28 | sxpert-work | non-deterministic behavior is only a problem for developpers that haven't grasped the concepts |
| 13:43.29 | benjk | well, there are quite a number of scholars who disagree with that gospel |
| 13:43.38 | benjk | read the von Behren paper |
| 13:43.57 | sxpert-work | things like ABS breaks use threads to do things and manage to handle the non-deterministic behavior properly |
| 13:44.32 | jeremy_g | what is the channel Local for? |
| 13:44.32 | coppice | do you know of a threaded ABS system? |
| 13:44.36 | Druken | uhg... monday morning business is a pain in the ass |
| 13:44.38 | sxpert-work | coppice: yeah |
| 13:44.39 | jeremy_g | whats really so special about it? |
| 13:44.41 | benjk | there is also an article which delivers mathematical proof that correctness cannot be guaranteed with preemptive scheduling and shared data models |
| 13:44.49 | sxpert-work | the bosch stuff uses thet |
| 13:44.53 | sxpert-work | that even |
| 13:45.06 | sxpert-work | they have multiple threads, with varied priorities. |
| 13:45.19 | benjk | google for Apache versus YAWS |
| 13:45.24 | coppice | car system makers are usually paranoid about async behaviour. No threads, to tasks, no interrupts is the usual methodology |
| 13:45.31 | benjk | read the von Behren paper |
| 13:45.36 | jeremy_g | is this #realtime |
| 13:45.39 | sxpert-work | the external communication thread has lower priority compared to the one that checks for the pressure applied on the break pedal |
| 13:45.46 | Damin | Well. that was easy.. "Customer: I installed a new application andwindows XP won't boot, should I restore w/ Norton GoBack?" "Me: Yep. Let me know if you need anything else. CLICK" |
| 13:45.56 | sxpert-work | Damin: lol |
| 13:46.16 | sxpert-work | jeremy_g: heh |
| 13:46.39 | jeremy_g | explain the philosophy of Local channel to me, i know how to use it? |
| 13:47.01 | pigpen | Question: is IAXmodem worth setting up now, or should I wait for Asterisk 1.4 ? |
| 13:47.25 | jeremy_g | yeah, why shud we wait for ast 1.4 |
| 13:47.26 | sxpert-work | benjk: link to said paper ? |
| 13:47.50 | pigpen | For did inbound routing of faxes via hylafax. |
| 13:48.02 | pigpen | ^^^clarification... |
| 13:48.06 | coppice | pigpen: sounds like you don't realise 1.4 and iaxmodem are completely unrelated |
| 13:48.27 | Druken | what is iaxmodem for? |
| 13:48.53 | pigpen | I do realize that....but I read that voicemail AND faxing are integrated into * ver 1.4. |
| 13:48.55 | coppice | its a class 1 fax modem build upon spandsp |
| 13:49.12 | pigpen | but thanks.....you never know who might be a newbe. |
| 13:49.18 | coppice | pigpen: where did you read that? |
| 13:49.47 | GaryH | jeremy_g: have you looked at http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels ? |
| 13:50.06 | pigpen | http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-14-unveiled.asp |
| 13:50.09 | pigpen | "Unified Messaging- integrates voicemail, email, and fax into a central mailbox where users can send, retrieve and manage all of their messages using any communication device. " |
| 13:50.18 | sxpert-work | benjk: heh, found it. to tell you the truth, we have researchers here that wrote the exact opposite in their own papers :D |
| 13:50.23 | *** join/#asterisk sif519 (n=ask@CPE00119576d845-CM00111adf8414.cpe.net.cable.rogers.com) |
| 13:50.39 | benjk | ftp://ftp.inf.puc-rio.br/pub/docs/techreports/04_15_moura.pdf |
| 13:50.44 | jeremy_g | GaryH:yeah |
| 13:50.47 | benjk | http://www.usenix.org/events/hotos03/tech/full_papers/vonbehren/vonbehren.pdf |
| 13:50.49 | pigpen | coppice, as you see...my question does have a "bit" of merit. |
| 13:50.59 | sif519 | I'm semi new to all this can someone help me with a musichold problem ? |
| 13:51.04 | coppice | you mean "T.38- allows IP FAXes to pass through the server" ? |
| 13:51.17 | pigpen | keep reading.... |
| 13:51.19 | benjk | and YAWS TOTALLY TOTALLY THRASHES APACHE here ... |
| 13:51.19 | benjk | http://www.sics.se/~joe/apachevsyaws.html |
| 13:51.28 | pigpen | 3 lines down. |
| 13:51.40 | coppice | oh, I see "Unified Messaging- integrates voicemail, email, and fax into a central mailbox where users can send, retrieve and manage all of their messages using any communication device." |
| 13:51.51 | Damin | pigpen: I believe the point is that you can use IMAP to store stuff for Voicemail if I am not mistaken. Not that it receives Faxes.. :) |
| 13:51.54 | coppice | well I guess that;s news to everyone who has played with it |
| 13:52.06 | pigpen | yes...I have downloaded the 1.4 beta2... |
| 13:52.16 | pigpen | and dug through it..but no metion of fax other than t.38 |
| 13:52.32 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:52.37 | coppice | and the T.38 is just an incomplete passthrough mode |
| 13:52.52 | pigpen | ah..ok. |
| 13:52.56 | Damin | pigpen: And now that you have downloaded it are you "integrating voicemail, email and fax into a central mailbox where you can manage all of your messages using any communication device."? |
| 13:53.26 | pigpen | well, I downloaded it...mainly to see what all I have to change to get my dialplan working.... |
| 13:53.27 | Damin | pigpen: Like I said, that is most likely IMAP support for app_voicemail. |
| 13:53.32 | coppice | well, it should be a complete one, but when it meets the real world..... :-) |
| 13:53.40 | Damin | pigpen: Which I'm not even sure is complete. :) |
| 13:53.40 | sxpert-work | benjk: I have yet to get an explanation on how this would work on multi-processor machines |
| 13:53.47 | jeremy_g | benjk:this is not apache vs yaws, shut the fuck up :P |
| 13:54.08 | pigpen | yeah...I was thinking when I read it, that it is probably imap support. |
| 13:54.15 | sxpert-work | benjk: of course, as I read it, erlang does not use the undelying OS (it's thus an OS in itself, like you could probably run DOS then) |
| 13:54.47 | benjk | actually pthreads are a bad idea on multi-processor systems when you get to a decent nnumber of cpus |
| 13:54.51 | coppice | pigpen: you can install spandsp, and add fax functions to *. there sren't any built in, though |
| 13:54.58 | benjk | http://www.computer.org/portal/site/computer/menuitem.5d61c1d591162e4b0ef1bd108bcd45f3/index.jsp?path=computer/homepage/0506&file=cover.xml&xsl=article.xsl |
| 13:55.05 | jeremy_g | mmm who said erlang? |
| 13:55.23 | coppice | Mrs Erlang? |
| 13:55.44 | Splas | hrm.. anyone port any of the app_swift/app_cepstral modules to 1.4? I'm reluctant to bother using an AGI :P |
| 13:56.08 | coppice | Erlang should have been British. Anyone that interested in queue should be British |
| 13:56.33 | jeremy_g | benjk and sxpert-work seem to be some educated dudes finally |
| 13:57.11 | jeremy_g | coppice:do you know what erlang is? |
| 13:57.23 | coppice | duh! |
| 13:58.08 | benjk | So Modula-2, Ada, Io and some other actor based languages are also operating systems then? |
| 13:58.08 | sxpert-work | coppice: are you inferring that brits are gays or something ? |
| 13:58.09 | fourcheeze | heh |
| 13:58.10 | benjk | pfffft |
| 13:58.17 | fourcheeze | nothing wrong with queueing |
| 13:58.27 | coppice | no. the British just love queuing |
| 13:59.00 | benjk | fourcheeze, only if you queue in |
| 13:59.10 | coppice | Big hunky Brits just can't wait to go to the post office and stand behind a hundred other people. its a national obsession |
| 13:59.27 | SplasPood | haha |
| 13:59.42 | SplasPood | they get in queue behind the rooo-ter |
| 14:00.17 | sxpert-work | the british should all have moved to poland |
| 14:00.31 | sxpert-work | massive queuing there 20 years ago :D |
| 14:00.33 | benjk | why poland? |
| 14:00.39 | benjk | haha |
| 14:00.53 | benjk | reminds me of a joke from Eastern Germany |
| 14:01.19 | sxpert-work | "where's east ?" "the direction the trains full of food go ... |
| 14:01.26 | benjk | somebody comes into a department store and asks "No TV sets here?" |
| 14:01.47 | jeremy_g | question: what is difference between checking the availability of a channel and going to extension (n+101) |
| 14:02.01 | benjk | and the clerk answers "No TV sets is on the fourth floor, this floor is no furniture" |
| 14:02.10 | tzanger | benjk: hahaha |
| 14:02.10 | SplasPood | heh |
| 14:02.12 | sxpert-work | jeremy_g: going to 101+n is deprecated |
| 14:03.22 | jeremy_g | then ? |
| 14:03.29 | jeremy_g | whats the alternative |
| 14:03.58 | SplasPood | most apps return status in a variable |
| 14:04.04 | SplasPood | you check that variable and then do whatever.. |
| 14:04.27 | tzanger | jeremy_g: goto foo-${DIALSTATUS} |
| 14:04.29 | tzanger | is what I use |
| 14:04.47 | coppice | I wonder if the download server for Erlang queues all the download requests? :-) |
| 14:05.02 | sxpert-work | hmmm. this coroutine stuff is rather ugly :D |
| 14:05.05 | SplasPood | coppice: Like I said.. depends on how well their roo-ter is working |
| 14:05.13 | scanna | hi all |
| 14:05.30 | jeremy_g | hi scanna |
| 14:05.35 | *** join/#asterisk cwedwards (n=chatzill@lncinsw1.cin.hdm.net) |
| 14:05.37 | sxpert-work | especially the concept of 'yield' |
| 14:05.39 | scanna | I don't understand something about callerid and cdr... |
| 14:05.41 | benjk | its actually less ugly than pthreads, or are you talking about Erlang syntax? |
| 14:05.50 | jeremy_g | tzanger:thanks |
| 14:05.59 | SplasPood | scanna: what about it |
| 14:06.14 | jeremy_g | what is it ? |
| 14:06.34 | scanna | i have 2 sip extensions, 100 (callerid=100) and 101 (callerid=101) |
| 14:06.47 | cwedwards | Is there anyone around available who can provide support for installation of my TDM400P? It does not appear to work. Whenever I try to make calls with it, despite having the trunk setup in Asterisk, it tells me that all circuits are busy |
| 14:06.49 | jeremy_g | aww cute!! |
| 14:07.00 | benjk | yield() is a very nice concept, because it allows you to decide what is and what isn't atomic |
| 14:07.02 | sif519 | what would cause Sep 25 10:06:43 NOTICE[23564]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 14:07.29 | benjk | sif519, no zaptel timing |
| 14:07.31 | scanna | if in the dialplan i change the callerid with set(CALLERID(number)=1234567) in the cdr the callerid is alwais the original (100 or 101) |
| 14:07.34 | coppice | that's all this about atomic yield? :-\ |
| 14:07.49 | sif519 | benjk : how do you fix that ? |
| 14:07.59 | scanna | why? it should be 1234567.... |
| 14:08.13 | benjk | install ztdummy or a PCI modem that goes by the name of X100P |
| 14:08.26 | sif519 | ty |
| 14:08.44 | cwedwards | It's nice that digium support lines lead to a dead busy signal, but at the cost of this thing I expect better than a more than 1 year out of date manual with incorrect information in it, and a total lack of drivers or correct download location |
| 14:08.58 | RoyK | global warming is nice. so far the mean september temperature in oslo has been 4,2°C above normal |
| 14:09.04 | sxpert-work | benjk: thing is, this doesn't work on multi-processor systems |
| 14:09.16 | sxpert-work | unless you're using a virtual machine thing |
| 14:09.21 | scanna | can someone help me? |
| 14:09.24 | jeremy_g | RoyK:and mean winters would be -25 |
| 14:09.29 | jeremy_g | :P |
| 14:09.40 | sxpert-work | you can't prevent the 2 procs from running stuff asynchronously |
| 14:09.48 | RoyK | jeremy_g: it rare gets that cold around here...... |
| 14:10.15 | pigpen | RoyK, yeah...winter here lasts about 1 day. |
| 14:10.21 | benjk | pthreads don't work well on multi-processors |
| 14:10.24 | RoyK | pigpen: where? |
| 14:10.29 | benjk | http://www.computer.org/portal/site/computer/menuitem.5d61c1d591162e4b0ef1bd108bcd45f3/index.jsp?path=computer/homepage/0506&file=cover.xml&xsl=article.xsl |
| 14:10.31 | pigpen | San Antonio, TX |
| 14:10.35 | RoyK | benjk: why shouldn't they? |
| 14:10.48 | benjk | read the paper in the URL |
| 14:11.25 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 14:11.36 | [TK]D-Fender | cwedwards: Thats a neat trick.... how exactly does Asterisk tell you "all circuits are busy"? |
| 14:12.24 | Damin | Juggie: Nah.. not much of a Hockey Fan.. :) |
| 14:12.32 | cwedwards | When I try to make the call, that's what it does. However, I know the line works because if I put a single-line phone on it, I get dialtone and can make calls |
| 14:12.41 | cwedwards | therefore, Asterisk is telling me all circuits are busy |
| 14:12.47 | DrAk0 | while im trying to connect to asterisk with ekiga i get this error http://pastie.caboo.se/14788 |
| 14:12.56 | RoyK | benjk: one day with not too much to do, i will |
| 14:13.09 | cwedwards | I'm assuming the card isn't installed correctly because the trunk and outbound routes were setup per guides from voip-info.org |
| 14:13.38 | sxpert-work | benjk: as I said, we have ppl here writing the exact oposite in their own papers |
| 14:13.50 | cwedwards | this is where Digium says to come for support, so I'm here |
| 14:14.07 | benjk | in a nutshell it says that once Intel and AMD roll out the 16, 32, 64, 128 core systems, software that appeared to work will fail |
| 14:15.39 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:15.39 | benjk | sxpert, there is a mathematical proof in the Lee paper, the only way you can debunk it is by a) find an error in the proof or b) deliver a counter proof that shows it doesn't apply to xyz cases |
| 14:16.07 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
| 14:16.31 | jtexter3 | scanna: Have you tried doing a NoOp right after with CALLERID(number) to see if it's the value you expect? |
| 14:16.44 | SplasPood | scanna: thats cause it's CALLERID(num) not number |
| 14:18.52 | *** join/#asterisk lorinc (n=ang@caracas-1643.adsl.interware.hu) |
| 14:22.02 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 14:22.33 | pingwin | i'm trying to use realtime sip. I have 2 static phone configurations in sip.conf and two in mysql. |
| 14:22.33 | scanna | also with CALLERID(num) same results |
| 14:22.43 | pingwin | sip show peers <id> show the same info profile. |
| 14:23.03 | pingwin | I can call the static peers from the realtime peers. but the realtime peers cannot call each other or be called from the static peers |
| 14:23.18 | pingwin | I have the password in the secret column. does md5 secret neet to be set? |
| 14:23.43 | pingwin | I have read that the md5secret column follows md5('<ext>:asterisk:<password>') |
| 14:23.47 | scanna | jtexter3: in the display of the called phone the callerid showed is 1234567 |
| 14:24.17 | pingwin | even still, it doesn't change what happens.. can anyone help, or point me in a direction of documentation that would help me? |
| 14:24.21 | jtexter3 | have you tried setting CALLERID(ani)? |
| 14:24.41 | *** join/#asterisk eject_ck (n=eject@rubin-gw.neocm.com) |
| 14:25.58 | scanna | jtexter3: with callerid(ani) in cdr the src is 1234567 and the callerid 100 |
| 14:26.03 | pingwin | and I can call the demo extensions on the server from the realtime extensions |
| 14:26.12 | pingwin | so it seems to be an issue of calling the realtime extensions |
| 14:26.17 | scanna | it changes only the source |
| 14:26.25 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 14:28.21 | *** join/#asterisk siffy519 (n=ask@CPE00119576d845-CM00111adf8414.cpe.net.cable.rogers.com) |
| 14:30.49 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 14:31.13 | zouzou | what the best sound formats to use with asterisk:voicemail and IVR |
| 14:31.34 | essobi | the closest one to the codec you're using the most often on the peers. |
| 14:31.45 | essobi | IE less transcoding = less cpu usage |
| 14:32.27 | zouzou | i need the best sound quality (don t care about CPU consuming0 |
| 14:32.33 | *** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co) |
| 14:33.03 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 14:34.56 | blaylock | who is this Rob (lilo) fella? |
| 14:35.03 | blaylock | creator of lilo? |
| 14:35.06 | pingwin | the guy who started freenode |
| 14:35.30 | pingwin | a car hit him while he was riding his bike |
| 14:36.07 | pingwin | can anyone help me with my realtime sip issue? |
| 14:36.32 | pingwin | if not I'll start writing all the realtime users as static sip configs. I just need to get this resolved. |
| 14:36.35 | blaylock | man |
| 14:36.42 | jtexter3 | scanna: Unfortunately, I haven't done much with changing CDR values, and then check the data in the database |
| 14:36.50 | blaylock | whats the problem? |
| 14:36.53 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 14:36.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:37.10 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-0674d480f5a1f685) |
| 14:37.48 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 14:37.57 | pingwin | i have 2 realtime sip users and 2 static sip users. I can call the static sip's and demo ext from the realtime phones. but I can't call the realtime phones at all |
| 14:38.02 | blaylock | is anyone here running multiple digium cards in one asterisk box? |
| 14:38.16 | blaylock | hmm |
| 14:38.28 | blaylock | i must be out of the loop, whats realtime sip? |
| 14:38.32 | jtexter3 | blaylock: yeah, I am |
| 14:38.44 | pingwin | the sip user config running from a database |
| 14:39.04 | blaylock | jtexter3, do they simply show up as the next available spans? |
| 14:39.21 | blaylock | jtexter3, like first 4 port card 1-4 second 5-8 |
| 14:39.56 | blaylock | pingwin, hmm sorry i havent gotten that deep into asterisk yet |
| 14:40.27 | pingwin | blaylock: tis cool, just need to figure out if it's an asterisk bug, if so I'm going to skip it and move on to writing static sip configs |
| 14:40.34 | pingwin | i don't want to , but if I have to I will |
| 14:40.57 | blaylock | pingwin, yeah i guess sometimes it has to be the hard way |
| 14:41.37 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:42.45 | jtexter3 | blaylock: that's correct. Also, there is a rotary dial on the cards. Make sure one is set to 0, and the other is set 1 |
| 14:43.12 | blaylock | jtexter3, cool thanks |
| 14:43.21 | blaylock | jtexter3, how is that dial set? |
| 14:43.42 | StyleWarz | Anybody here who could gimme a hint how to solve the following: I have Nokia E60 which is connected via WIFI/SIP. If somebody calls in, it should first try to connect over SIP, and if that channel is unavail (like with ChanIsAvail) it should connect the call over GSM to my phone |
| 14:43.42 | blaylock | jtexter3, through software or some type of dip switch |
| 14:43.53 | StyleWarz | But there is no solution with queues right? |
| 14:44.44 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 14:45.12 | *** part/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:45.44 | jtexter3 | blaylock: It's a rotary dial on the top of the card, close to the front of the card. You'll need a small phillips to set it. |
| 14:45.58 | blaylock | jtexter3, cool thanks |
| 14:47.19 | [TK]D-Fender | StyleWarz: You wouldn't use queues for that. Its about3 lines of dialplan logic. |
| 14:48.08 | StyleWarz | [TK]D-Fender: well i thought about ChanIsAvail(SIP/myuser) |
| 14:48.21 | StyleWarz | then n+101 do a dial to my GSM phone |
| 14:48.33 | StyleWarz | but how can i catch if just killed the call on my cell |
| 14:48.39 | StyleWarz | like i don't want that call |
| 14:48.45 | FlatFoot | anyone good with debian ? |
| 14:48.57 | StyleWarz | FlatFoot: That question kills itself ;) |
| 14:48.59 | FlatFoot | i'm having a blonde day need help |
| 14:49.32 | fourcheeze | FlatFoot: just ask |
| 14:49.33 | FlatFoot | using th tzconfig and the time is coming up about 8 hours out |
| 14:50.07 | FlatFoot | need to change the time can i find somewhere in there to specify a time server to use |
| 14:50.13 | FlatFoot | rather than the in built stuff ? |
| 14:50.30 | fourcheeze | ls -l /etc/localtime |
| 14:51.01 | FlatFoot | lrwxrwxrwx 1 root root 28 2006-09-25 19:19 /etc/localtime -> /usr/share/zoneinfo/Etc/GMT0 |
| 14:51.33 | FlatFoot | should be 15:51 |
| 14:52.54 | StyleWarz | [TK]D-Fender: Could u give me some good lines for this? since i really don't get how that is supposed to work |
| 14:52.58 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:53.51 | FlatFoot | fourcheeze: that was my output |
| 14:54.15 | fourcheeze | are you in gmt? |
| 14:54.17 | [TK]D-Fender | cwedwards: please pastebin the error for us to see showing that it is indeed the card at fault. |
| 14:54.19 | FlatFoot | need to change the time cos i get res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 14:54.19 | [TK]D-Fender | ~pb |
| 14:54.24 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:54.33 | FlatFoot | fourcheeze: yeah just south of london |
| 14:54.45 | fourcheeze | you probably want what I have then |
| 14:54.55 | fourcheeze | lrwxrwxrwx 1 root root 33 2006-01-12 01:23 /etc/localtime -> /usr/share/zoneinfo/Europe/London |
| 14:55.09 | fourcheeze | then try |
| 14:55.16 | FlatFoot | yeah the time now is 15:55 |
| 14:55.39 | StyleWarz | [TK]D-Fender: or is this possible with a queue too? |
| 14:55.40 | fourcheeze | cool |
| 14:56.00 | FlatFoot | aha where ? |
| 14:56.08 | [TK]D-Fender | StyleWarz: Its like 2 lines..... c'mon.... |
| 14:56.10 | fourcheeze | Farnham, Surrey |
| 14:56.16 | FlatFoot | i am in sunny (tr)Ashford Kent |
| 14:56.31 | [TK]D-Fender | StyleWarz: "show application gotoif" and look at the ${DIALSTATUS} variable |
| 14:56.45 | StyleWarz | [TK]D-Fender: ah thanks, that was the info i was looking for |
| 14:56.55 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:57.08 | benjk | about ~12.000 miles either way or so |
| 14:57.11 | FlatFoot | benjk: you must be large to get both sides at the same time |
| 14:57.37 | FlatFoot | fourcheeze: any idea about this time prob ? |
| 14:58.00 | fourcheeze | did you set your symlink? |
| 14:58.04 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 14:58.18 | FlatFoot | symlink for what ? |
| 14:58.25 | fourcheeze | <PROTECTED> |
| 14:58.48 | FlatFoot | just did a quick install downloaded the debian netinstall |
| 14:58.51 | ManxPower | Ugh. I just spend $300 on surge protectors |
| 14:59.03 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:59.06 | fourcheeze | FlatFoot: you need to set your /etc/localtime to link to where you are |
| 14:59.44 | FlatFoot | fourcheeze: might be a bit thick but i thought i did that with the tzconfig |
| 14:59.48 | fourcheeze | rm /etc/localtime |
| 14:59.49 | fourcheeze | ln -s /usr/share/zoneinfo/Europe/London /etc/localtime |
| 14:59.57 | fourcheeze | then do |
| 15:00.01 | fourcheeze | ntpdate time.uk2.net |
| 15:00.17 | fourcheeze | <PROTECTED> |
| 15:00.17 | fourcheeze | Mon Sep 25 16:00:09 BST 2006 |
| 15:01.02 | FlatFoot | that would work but i need to get ntpdate |
| 15:01.07 | DrAk0 | while im trying to connect to asterisk with ekiga i get this error http://pastie.caboo.se/14788 |
| 15:01.26 | fourcheeze | apt-get install ntpdate |
| 15:01.59 | FlatFoot | fourcheeze: sorry just done that , can't type or think today |
| 15:02.12 | FlatFoot | luverly job i now have the correct time |
| 15:02.13 | GerbilWrk | what do you need to setup to have Asterisk e-mail the voicemails? |
| 15:02.24 | FlatFoot | that netinstall realy is basic/minimal |
| 15:02.42 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 15:02.48 | benjk | GerbilWrk: Postfix or Exim |
| 15:03.00 | ManxPower | I see that today is Newbie Day |
| 15:03.05 | benjk | or Sendmail (cough) |
| 15:03.13 | ThaZZa | Any able to help me compile asterisk 1.4 on a fresh debian stable system? |
| 15:03.31 | GerbilWrk | benjk, thanks |
| 15:03.38 | FlatFoot | fourcheeze: thanks for that now all i got to do is get this damn Snom 320 to send dtmf properly |
| 15:03.42 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 15:04.10 | fourcheeze | FlatFoot: np |
| 15:04.12 | benjk | you need to edit /etc/asterisk/voicemail.conf too |
| 15:04.33 | FlatFoot | benjk: was that for me ? |
| 15:04.43 | benjk | no, for GerbilWrk |
| 15:04.58 | FlatFoot | soory got my eye on nine things |
| 15:05.22 | benjk | byonic eye, eh? |
| 15:05.37 | benjk | or is it bionic? |
| 15:05.41 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp39-37.adsl.forthnet.gr) |
| 15:05.44 | FlatFoot | yeah just an office full of planks who won't leave me alone |
| 15:06.20 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 15:06.40 | benjk | like so .. "fix it a**hole! how long is this going to take?" ??? |
| 15:07.04 | FlatFoot | yeah that an the ones who repeat themseleves 98 time |
| 15:07.20 | FlatFoot | s cos they can't remeber what you just told them 3 seconds ago |
| 15:07.40 | benjk | we had this trader who got so enraged that he through a big 19 inch CRT out of the window |
| 15:07.44 | benjk | or so he thought |
| 15:08.38 | benjk | the thing was still plugged in so the power cord caught it and it lost its momentum on the way to the window, fell down on his foot and broke his ankle |
| 15:08.45 | FlatFoot | nice |
| 15:08.54 | FlatFoot | bet that was a little painful |
| 15:08.57 | ManxPower | LOL! |
| 15:09.10 | benjk | he was like a wild animal then |
| 15:09.33 | benjk | but he made millions for the bank, so he was untouchable |
| 15:09.38 | ManxPower | benjk, to bad you did not have a video camera, you could have posted the video online |
| 15:09.53 | benjk | that was back in 1994 |
| 15:10.00 | ManxPower | benjk, I hate people like that. The only people that should be untouchable are tech people. |
| 15:10.10 | benjk | heh |
| 15:10.25 | ManxPower | benjk, 1994? I guess you could have posted an ASCII Art animation, but it would not have been as funny. |
| 15:10.39 | benjk | the story alone was funny enough |
| 15:11.03 | Qwell | alt.humor.monitors-falling-on-people |
| 15:11.33 | benjk | mind you, if the monitor has made it through the window, this was in the 19th floor of a Tokyo high rise above a very crowded road |
| 15:11.42 | benjk | s/has/had |
| 15:12.29 | benjk | if it had hit somebody below to death, he'd been serving time in a Japanese prison |
| 15:12.56 | Qwell | What, they wouldn't have found a scapegoat? |
| 15:13.24 | ManxPower | "The monitor just jumped out the window! I think it was working too hard and could not take the pressure!" |
| 15:13.28 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 15:13.35 | benjk | not if the company is a foreign bank and the guy a foreigner |
| 15:13.46 | benjk | hahaha |
| 15:14.17 | *** join/#asterisk saftsack (n=oliver@p54A7D6E4.dip.t-dialin.net) |
| 15:14.27 | [TK]D-Fender | ManxPower: "When all seems lost and you want to give up...... remember thats why the windows don't open" |
| 15:14.36 | benjk | heh |
| 15:14.41 | sxpert-work | of course, that was a coroutines using monitor </troll> |
| 15:15.03 | benjk | I think it was an analog monitor |
| 15:15.24 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 15:16.16 | *** part/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 15:18.13 | FlatFoot | anyone used these snom320's ? |
| 15:19.46 | FlatFoot | would do but i am on the ground floor |
| 15:19.53 | benjk | too bad |
| 15:19.57 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:20.05 | FlatFoot | although i could go to my transmitter site ( 13 floors) |
| 15:20.23 | benjk | that might just be high enough |
| 15:20.26 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:23.29 | *** join/#asterisk ptblank (n=MURDER1@68-169-166-65.lmdaca.adelphia.net) |
| 15:23.40 | RoyK | hm... is it possible to check gcc version from the makefile? |
| 15:24.00 | RoyK | PROC=k8 requires gcc >= 3.4 |
| 15:24.22 | benjk | gcc --version |
| 15:24.45 | *** join/#asterisk Tr4d3r (n=Tr4der@exchange.slmsistemas.net) |
| 15:25.57 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 15:26.32 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 15:26.33 | RoyK | GCC_MAJOR_VERSION=`gcc --version| head -1 | cut -d\ -f3 | cut -d. -f1` |
| 15:27.07 | benjk | make has some string handling functions built in, no need to pipe |
| 15:27.55 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-172-125.dhcp.southerncoastalcable.net) |
| 15:28.04 | RoyK | benjk: erm. how can I do it the best way then? |
| 15:28.10 | QbY_ | Anyone know the dial plan/digit map for a Polycom to send all digits to the Asterisk server? |
| 15:28.41 | *** join/#asterisk kristalino (i=kristali@gateway/tor/x-4694998219856e4c) |
| 15:28.50 | benjk | I can't recall it from memory, but I have done this to split the Darwin version into its three parts and it was surprisinly easy |
| 15:29.07 | benjk | check the manuals for make and look for string handling functions |
| 15:29.19 | ManxPower | QbY_, Uh, it sends all digits to the server by default |
| 15:29.47 | QbY_ | for some reason, it can't send ** |
| 15:29.54 | QbY_ | we have an extension 9000** and it can't dial the second * |
| 15:30.30 | benjk | check features.conf, probably the * is mapped to some feature |
| 15:30.39 | [TK]D-Fender | QbY_: : What phone? |
| 15:30.56 | QbY_ | IP 501 |
| 15:31.13 | QbY_ | this is what it had: [2-9] |
| 15:31.13 | QbY_ | <PROTECTED> |
| 15:31.51 | [TK]D-Fender | QbY_: : nvm, just read that : x.T|*.T|#.T |
| 15:32.03 | GerbilWrk | try adding |xxxx**| |
| 15:32.24 | [TK]D-Fender | QbY_: You want what I just put up. |
| 15:32.28 | benjk | #.T seems like a weird channel, nobody there |
| 15:33.11 | QbY_ | [TK]D-Fender-- So that will send everything, after the obvious pause, to the * |
| 15:34.42 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 15:35.22 | pingwin | are there any better options for text-to-speech other than festival? the voice is almost unuseable |
| 15:35.36 | QbY_ | pingwin.. Cepstral |
| 15:35.42 | pingwin | thanks Oby |
| 15:35.56 | rob0 | or ... hire more people to answer the phone ;) |
| 15:35.57 | m4rkl4r | Hi, Is there an agi scripting package that is currently recommended? In any language? |
| 15:36.38 | m4rkl4r | more specifically, what is currently the most usefull php AGI package? |
| 15:36.55 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 15:40.19 | m4rkl4r | most of the php AGI wrappers are rather old. Is the AGI interface stable enough that it doesn't matter? |
| 15:41.50 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:42.45 | *** join/#asterisk rgeek (i=rgeek@nat/digium/x-7fc1fe1d2bbbb2f3) |
| 15:43.54 | syzygyBSD | m4rkl4r: I prefer the python AGI module I use. It seems to be rather robust, but if you stick with php let me know if you find a good one |
| 15:44.11 | m4rkl4r | syzygyBSD: thanks |
| 15:44.23 | syzygyBSD | I have used a couple but ended up doing most of the work myself, I haven't found a very good php module yet, but that doesn't mean there isn't one |
| 15:44.58 | *** part/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 15:45.41 | [TK]D-Fender | QbY_: Correct |
| 15:45.54 | [TK]D-Fender | QbY_: That is the dialplan I use for all of my installations |
| 15:46.33 | m4rkl4r | syygyBSD: could you provide me a link to the exact python package you are using? |
| 15:46.52 | *** join/#asterisk rgeek (i=rgeek@nat/digium/x-7b59936e28e6ac2c) |
| 15:46.57 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:47.04 | syzygyBSD | well, not the exact one, but the one that I started with |
| 15:47.21 | m4rkl4r | sure. |
| 15:47.54 | syzygyBSD | pretty sure it is http://sourceforge.net/projects/pyst |
| 15:48.38 | m4rkl4r | thanks |
| 15:49.49 | syzygyBSD | ya, that is the one |
| 15:50.18 | syzygyBSD | looks like they have updated it quite a bit since I took it (2 years ago) so it should be a lot more helpful now |
| 15:50.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:53.57 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:59.16 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 16:02.24 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 16:09.16 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 16:12.01 | pjz | so I'm going to get either a digium TE207P or a TE212P... my motherbaord will take either.. any preference? |
| 16:12.13 | pjz | it's the 3.3 and 5v versions of the same card |
| 16:12.26 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 16:13.07 | FlatFoot | so i just svn'd to do a rebuild cos i am fed up now , and , make zaptel is full of errors . should i reload the who damn thing ?????????? |
| 16:13.17 | FlatFoot | or just hit it with a LARGE hammer |
| 16:13.39 | *** join/#asterisk Avalone (n=Avalone_@dial-010.vl-093-as1.avtlg.ru) |
| 16:14.57 | _alex_mx_ | pjz, i would go with the 5v simply because you can always plug into just about anything and get it up and running...a 3.3v card requires that you have a compatible "server class" mobo...my 2c |
| 16:17.33 | FlatFoot | i hate my connection only getting 6MB/Sec |
| 16:18.00 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 16:18.48 | adamowitz | Anyone here using (or has used) the D-Link DPH-540 WiFi phone? |
| 16:19.15 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-224e28676fb6c964) |
| 16:19.15 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:27.40 | pjz | _alex_mx_: ah, okay, thanks |
| 16:28.47 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 16:28.53 | angryuser | i have tdm400p with 4 FXO lines, 1 line is coupled with another one, is it possible to detect when i have incoming call, do not answer, queue outgoing call till the momone when incoming call stops on the same interface? |
| 16:29.08 | angryuser | *noone |
| 16:29.19 | angryuser | *noone rings |
| 16:29.59 | angryuser | i have tdm400p with 4 FXO lines, 1 line is coupled with another one, is it possible to detect when i have incoming call, do not answer, queue outgoing call till the moment when incoming call stops on the same interface? (my bad) |
| 16:31.12 | [TK]D-Fender | angryuser: You should not be coupling FXO ports...... very bad |
| 16:31.25 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:31.34 | angryuser | it is not a coupling, my stupid inet provider |
| 16:32.28 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-224e28676fb6c964) |
| 16:32.56 | [TK]D-Fender | blitzrage: ! ! ! |
| 16:33.41 | angryuser | i got like internet unlimited line from my provider(adsl) you know that with adsl you got the filter, and normally normal line is not interferred with adsl, well my provider route incoming calls from normal nit to internet line, and i can turn of this feature |
| 16:33.55 | blitzrage | [TK]D-Fender: ever see "Error is 0x10000" ? |
| 16:33.55 | angryuser | *i cant |
| 16:33.56 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 16:35.18 | *** join/#asterisk cwedwards (n=chatzill@lncinsw1.cin.hdm.net) |
| 16:35.26 | cwedwards | Is there a digium representative here? |
| 16:35.39 | [TK]D-Fender | blitzrage: I believe your <mac>.cfg file is pointing to a non-existant file. |
| 16:36.27 | Qwell | cwedwards: depends on the question |
| 16:36.42 | [TK]D-Fender | cwedwards: Several, but a while back I asked you to provide CLI output to show us up-front the error you are receiving. |
| 16:38.32 | angryuser | so Fender my lines are not coupled, let's say i dont need answer on incoming, and wait when it stops ringing before open Zap, is it possible? |
| 16:39.18 | [TK]D-Fender | angryuser: You mean you don't want * to answer incoming calls, only to place outgoing calls on that line? |
| 16:39.34 | cwedwards | I am calling already for hardware installation support |
| 16:40.23 | cwedwards | Fender: Never got that message; I got disconnected at some point and then didn't realize nickserv was keeping me out of the channel |
| 16:40.32 | cwedwards | what CLI output would you like? |
| 16:40.39 | [TK]D-Fender | cwedwards: As you wish, but if you are asking for help in here it would be appreciated if you could provide us with some of the details we'll need to help identify your problem. |
| 16:40.46 | angryuser | exactly, also i need to check before placing outgoing call is that interface rings, and if yes queue it till it stops |
| 16:40.55 | cwedwards | I would be happy to do so |
| 16:41.00 | [TK]D-Fender | cwedwards: The output of a failed call (inbound/outbound) on "verbose 10" |
| 16:41.13 | *** join/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 16:41.14 | file | [TK]D-Fender: I just want |
| 16:41.18 | *** join/#asterisk oej (n=oej@191.Red-88-9-110.dynamicIP.rima-tde.net) |
| 16:41.47 | cwedwards | hrm, not sure how I get that. asterisk -r for the cli, then make a call, it fails, how do I get verbose 10? |
| 16:41.56 | cwedwards | I'm afraid I'm relatively new to linux in general and Asterisk |
| 16:42.09 | [TK]D-Fender | angryuser: What you are looking for is extremely trick if not technically impossible. The phone can begin ringing before your card can even acknowledge it and you could in that moment pick up a line only to be grereting by that inbound caller |
| 16:42.39 | [TK]D-Fender | cwedwards: in * CLI do "set verbose 10" then capture the screen output for that failed call and put it in a pastebin |
| 16:42.41 | [TK]D-Fender | ~pb |
| 16:42.43 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 16:42.43 | angryuser | i agree, a bit tricky |
| 16:42.48 | [TK]D-Fender | I'd suhhest www.pastebin.ca |
| 16:42.52 | [TK]D-Fender | suggest* |
| 16:43.29 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 16:43.48 | [TK]D-Fender | angryuser: I'm really not sure how you'd do this without some custom circuitry. or a nasty amount of scripting... |
| 16:43.54 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:44.58 | cwedwards | is there any way to capture the text from the java ssh terminal? |
| 16:45.11 | angryuser | Fender, all coz my providers router is total **;) |
| 16:45.13 | cwedwards | I can't seem to do that. I suppose I could do a screenshot instead |
| 16:45.20 | [TK]D-Fender | cwedwards: What Java SSH interface? |
| 16:46.10 | quid246 | hehe... what would cause a TORTURE response from ${DIALSTATUS}... not that I have one, but just saw it in the docs. |
| 16:46.14 | [TK]D-Fender | angryuser: You should be able to tell your telco to automatically forward your calls to your other # using a service like *72 |
| 16:46.57 | angryuser | i also heard that my provider is using MGCP client for his inet phone, any wayt to connect directly asterisk to it? |
| 16:47.37 | cwedwards | I'm using Trixbox 1.1 so the java ssh terminal in it |
| 16:47.48 | cwedwards | and apparently I can't copy and paste text from it |
| 16:47.58 | cwedwards | so I'm trying the screenshot into a .rtf document into pastebin |
| 16:48.11 | [TK]D-Fender | cwedwards: Well if you can provide that screensho it may be of use but unfortunately : |
| 16:48.18 | [TK]D-Fender | ~trixbox |
| 16:48.19 | jbot | [trixbox] NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 16:48.20 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:48.34 | cwedwards | yes, I know, I've been there |
| 16:48.39 | [TK]D-Fender | cwedwards: It may be excessively hard to debug this because of that |
| 16:49.11 | cwedwards | ok. Well I appreciate that. So are you telling me Digium cards are not compatible with Trixbox? |
| 16:49.29 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-9729cc0be9713420) |
| 16:49.29 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 16:49.39 | quid246 | I dunno why there is so much animosity towards Trixbox here... after all, * is at the core of it. |
| 16:50.05 | quid246 | Maybe the purists feel something has been taken away rfom them by the masses. |
| 16:50.12 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:50.19 | aptura | because it gives bussiness the power to install it them selves rather then call a consultant? |
| 16:50.28 | quid246 | aptura: Rightly so. |
| 16:50.51 | cwedwards | if that's the answer, then my response is to pack the card up, ship it back and get my money refunded, and cancel the project |
| 16:50.56 | rob0 | Grandstream ATA's ... I've heard they suck? Linksys/Sipura preferred? |
| 16:51.05 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:51.25 | quid246 | cwedwards: Your Digium card will have no problems working with Trixbox, since it's just a repackaged Asterisk |
| 16:51.34 | cwedwards | After all, the only reason I bought the digium card was for installation support, and evidently you're telling me because I'm installing it on a server running the trixbox package I'm not elligable? |
| 16:51.47 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:51.47 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:51.48 | cwedwards | quid: that's what I thought. I'll deal with telephone support. |
| 16:52.02 | cwedwards | and ask my other questions in #freepbx as requested |
| 16:52.02 | rob0 | Did Digium tell you no support for trixbox? |
| 16:52.05 | cwedwards | I appreciate your time and efforts |
| 16:52.07 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 16:52.18 | cwedwards | rob0: they haven't told me no support. Fender is indicating that. |
| 16:52.56 | rob0 | I think the point is it is off topic HERE, not that Digium can't or won't provide the installation support you paid for. |
| 16:53.00 | quid246 | cwedwards: And if still doesn't work, then gt yourself a Snagoma.. or if it's not a critical app, an OpenVox card |
| 16:53.02 | [TK]D-Fender | cwedwards: I did not say that. You can contact Digium for direct support, just that it is not directly supported in here. |
| 16:53.14 | quid246 | oops Sangoma |
| 16:53.24 | [TK]D-Fender | cwedwards: PM |
| 16:53.31 | cwedwards | Fender: Ok. Thank you. Have a good day sir. |
| 16:53.37 | *** join/#asterisk juice (n=juice@mo-76-0-45-81.dhcp.embarqhsd.net) |
| 16:54.57 | angryuser | is there any wayt to register MGCP client on remote server? |
| 16:55.26 | rob0 | quid246: I can only guess from seeing other frontends to complex software, I don't have experience with trixbox et al. In making things pointy-clicky easy for non-tech administrators, such frontends often break things that should work. |
| 16:56.09 | rob0 | My expertise is with mail, Postfix in particular. You couldn't write a frontend to do what I do. |
| 16:57.15 | *** join/#asterisk gissi (n=silvio@32.104.18.240) |
| 16:57.23 | ManxPower | aptura, Telecom is complicated enough businesses will still have to hire a consultant |
| 16:57.29 | aptura | Just saw a new Avaya IPPhpone install at homedepot. Was impressed with the phones design and the fact it had a lit display. |
| 16:57.38 | aptura | ManxPower Ohh I know |
| 16:58.31 | aptura | Not only complicated also critical. Thats the plus side of this biz if you know it well then your services are needed if a clients system goes down. |
| 16:59.04 | aptura | other then aastara what other phones have a lit display? |
| 16:59.26 | [TK]D-Fender | aptura: Aastra phones are backlit, as well as the Linksys SPA-942, and the overprices Polycom IP 650 |
| 17:00.13 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 17:00.36 | aptura | Okay then that will be a option if the customer wants it. |
| 17:00.55 | [TK]D-Fender | Backlight is a nifty option, but I wouldn't pick my phone over it. |
| 17:02.17 | angryuser | is there any software whit is capable to register MGCP client on remote server? |
| 17:02.23 | angryuser | *which? |
| 17:03.25 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 17:04.22 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
| 17:04.22 | *** mode/#asterisk [+o kram] by ChanServ |
| 17:07.16 | ManxPower | angryuser, This is VoIP. ALL servers are "remote" |
| 17:08.03 | angryuser | <ManxPower> my server is local for me;) |
| 17:09.20 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:10.01 | sahafeez | question - i have a nice up and running asterisk system that i installed last year - oct or dec. i have not upgraded since then. the system works fine. any reason to upgrade? |
| 17:10.12 | zparta | yes |
| 17:10.15 | zparta | bugfixes :P |
| 17:10.25 | zparta | securityfixes and so forth |
| 17:10.39 | CtRiX | no. no reason for asterisk to upgrade. |
| 17:10.47 | CtRiX | if it works, don't touch it. |
| 17:11.03 | CtRiX | every bugresolved introduces at leastother 2 bugs... |
| 17:11.50 | sahafeez | ok. like asking an economist 2 opinions per person ;) |
| 17:12.01 | aptura | sahafeez which version |
| 17:12.06 | sahafeez | well the box is behind a nice big firewall and no access is allowed to it. |
| 17:12.12 | sahafeez | hum - how do i find out? |
| 17:13.18 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 17:13.27 | aptura | when you connect to asterisk at the cli it will show you above the cli |
| 17:13.35 | aptura | or show version |
| 17:13.52 | sahafeez | Asterisk CVS HEAD built by root@voice-gateway on a i686 running Linux on 2005-11-01 23:43:35 UTC |
| 17:13.57 | sahafeez | does not say much |
| 17:14.05 | sahafeez | show version files = a big list |
| 17:14.49 | aptura | or in /usr/src/asterisk version name if its your ownly download of asterisk. |
| 17:14.58 | revstray | anyone know if there is an update/upgrade schedule on the OpenBSD port of Asterisk? |
| 17:15.30 | sahafeez | did not think about that. what file in /usr/src/asterisk |
| 17:15.49 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 17:16.18 | sahafeez | i remember i pulled it from CVS.. |
| 17:16.25 | mog | i thought asterisk ran on openbsd |
| 17:16.46 | sahafeez | Asterisk 1.2.0-beta2 released |
| 17:17.17 | sahafeez | from the changelog |
| 17:17.39 | hmmhesays | yeah I compiled that for mipsel a coupld days ago |
| 17:19.55 | sahafeez | so any compelling reason to upgrade? |
| 17:20.33 | blitzrage | fixing something that isn't broken is always a compelling reason :) |
| 17:25.36 | nettie | Hi guys, I just noticed that my wildcard fxo takes a while to detect the hangup, anyone know if this could be tuned please? |
| 17:25.43 | angryuser | i am searching for a external 1 FXS<>pstn supporting MGCP, or directly a phone |
| 17:26.38 | fiel | nettie: probably need remote disconnection supervision turned on from your line provider |
| 17:27.11 | nettie | fiel ehehe anything simpler? |
| 17:27.12 | nettie | eheh |
| 17:27.28 | nettie | they'll never ever ever ever ever do it |
| 17:27.28 | nettie | eheh |
| 17:27.55 | fiel | nettie: nope, without that then there is nothing to really determine if the line dropped |
| 17:28.00 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 17:28.16 | Dr-Linux|work | please someone have a look >> http://pastebin.ca/182030 |
| 17:28.23 | nettie | fiel pretty sad then.. |
| 17:28.32 | Dr-Linux|work | i'm recieving a lot of stuff on CLI regarding this IP address |
| 17:28.46 | Dr-Linux|work | i can ping this ip as well, but don't know what's wrong :S |
| 17:28.59 | fiel | nettie: well, its a physical thing on the line, nothing the card can do to provide a feature that isn't on the line |
| 17:29.03 | revstray | my reason for asking about the upgrade, is the 1.4 version coming. I have no compelling reason to upgrade right now. |
| 17:29.29 | Tall-guy | dr-linux: why do u think something is wrong? |
| 17:29.37 | nettie | fiel I understand.. well I think I just have to live with that then :( thanx a lot to pointing it anywa.. |
| 17:29.57 | revstray | and in any case, I am still setting Asterisk up for the first time :) Just doing my due diligence. |
| 17:31.38 | Dr-Linux|work | Tall-guy, : bcoz i don't know what's 69.106.137.172 |
| 17:32.16 | gissi | revstray: If I were you, I would setup a new box (or even a virtual one with Xen or VMWare) and test out the new features of the 1.4, maybe you find something that worth implementing for your users... |
| 17:32.32 | Tall-guy | dr-linux: to me, it looks like a sip client is trying to connect to you from that ip address, and then the debug shows asterisk determining whether or not the request is coming from a local ip as defined in your config, or external....(and it determines its external, and thereofore substitutes your externlip in the reply) |
| 17:32.44 | *** part/#asterisk fiel (n=blitzrag@CPE0016b614c984-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 17:32.56 | revstray | gissi: alright, will do as soon as I get this one going. :) |
| 17:33.27 | revstray | this will be fully a softphone setup, so a bit of a learning curve |
| 17:34.33 | Dr-Linux|work | Tall-guy, so what you suggest what should i do? |
| 17:34.58 | Tall-guy | dr-linux: well, i don't think anything is "wrong" except that someone is hitting you who you don't know who they are.....so block 'em at the firewall |
| 17:36.14 | Dr-Linux|work | Tall-guy, he/she is hitting alot |
| 17:36.45 | Tall-guy | dr-linux: daily? or just today? |
| 17:36.56 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 17:37.00 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 17:37.38 | nettie | hey guys anyone knoe why musiconhold sounds distorted please? |
| 17:37.45 | nettie | it's very noisy |
| 17:37.54 | nettie | had it runnign perfectlyon my old box |
| 17:38.08 | Dr-Linux|work | Tall-guy, today |
| 17:38.12 | nettie | anyone know what I could check pleasE? |
| 17:38.34 | Dr-Linux|work | Tall-guy, i just debug the IP and i found the reason |
| 17:38.42 | Dr-Linux|work | justinu|laptop, you around? |
| 17:39.05 | nettie | Ithink i figured out |
| 17:41.41 | nettie | newest mpg123 versions are not working |
| 17:42.16 | syzygyBSD | there is one version in pretty much all the documentation that you are suposed to use |
| 17:42.35 | syzygyBSD | I think it is .40r... or something like that, I haven't looked at them for a long time |
| 17:42.36 | *** join/#asterisk malahoo (n=malahoo@70-33-145-223.agstme.adelphia.net) |
| 17:42.48 | [TK]D-Fender | Hey, I've got an FC3 server I need to get TFTP running on. Can someone remind me the name of the package I need to install for it? |
| 17:43.11 | syzygyBSD | oh... gimme a sec |
| 17:43.37 | CunningPike | nettie: Use native moh |
| 17:43.40 | syzygyBSD | tftp? |
| 17:43.41 | CunningPike | ~moh |
| 17:43.42 | jbot | well, moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
| 17:44.07 | syzygyBSD | [TK]D-Fender: for me on fedora core 3... 'apt-get install tftp' works |
| 17:44.16 | Tall-guy | tkd: http://rpm.pbone.net/index.php3/stat/4/idpl/1540792/com/tftp-server-0.34-0.1.fc3.rf.x86_64.rpm.html |
| 17:44.28 | syzygyBSD | but as you can see I have apt installed... a little different then you I think |
| 17:44.36 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-60-201.red.bezeqint.net) |
| 17:44.40 | [TK]D-Fender | Tall-guy: Yeah I jsut got that for my arhc, whats the service name it installs? |
| 17:44.44 | [TK]D-Fender | arch* |
| 17:45.34 | zparta | is there any SIP number i can call to check if my asterisk is working? |
| 17:45.57 | hmmhesays | so is anyone in here using asterisk2billing? |
| 17:46.16 | malahoo | Hi - I have an * system set up, and it works for the most part. But some extensions always ring as "busy" or "not available". Is this a good place to ask for help? |
| 17:46.42 | Tall-guy | tkd: tftpd from xined? |
| 17:46.43 | rob0 | malahoo: Definitely maybe. |
| 17:47.07 | faberk64 | Hi guys |
| 17:47.14 | faberk64 | I'm getting this problem: if I call a busy phone in my office from another phone inside, I got the busy signal, but If I call from outside to an office phone and it is busy, I have no tone only silence until the line go down. |
| 17:47.14 | faberk64 | If I got a phone off hook: if I call it from outside, it keeps ring free. |
| 17:47.18 | hmmhesays | zparta: that depends on what services you have it hooked up to |
| 17:47.24 | *** join/#asterisk _ViperNetworks (n=Nitesh@65.48.63.178) |
| 17:47.36 | zparta | hmmhesays: at this moment none :P |
| 17:47.49 | zparta | cant i call an ip or something? :P |
| 17:47.57 | *** join/#asterisk easement (i=easement@ip67-94-5-100.z5-94-67.customer.algx.net) |
| 17:48.00 | [TK]D-Fender | Tall-guy: I think I got it. Seems to load under xinetd |
| 17:48.15 | Tall-guy | tkd: thats what I meant by xined? but I can't type. |
| 17:48.24 | [TK]D-Fender | xinetd* |
| 17:48.30 | [TK]D-Fender | Tall-guy: Thanks for your tips. |
| 17:48.44 | trelane_ | so the WCTDM24XXP is really a DEC Tulip (compatible) ethernet adapter |
| 17:48.56 | trelane_ | cute |
| 17:49.45 | *** join/#asterisk Splas (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 17:49.45 | easement | I have been googling like crazy trying to wrap my head around things and have a basic noob question: I am looking to buy a Linksys WRT54GL and install OpenWRT and Asterisk on it. What is the good/best VoIP client to go with? |
| 17:49.48 | file | trelane_: not exactly |
| 17:50.28 | easement | All the sites mention that it won't work with skype and I got consufed with mention of googleTalk supporting SIP. |
| 17:51.00 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 17:52.40 | file | the TDM24XXP is a weapon... that is all. |
| 17:53.11 | rob0 | easement: * is both a VoIP client and server, supporting SIP and IAX2 and others. So I don't understand the question. Do you want to know about softphones? |
| 17:53.17 | malahoo | I have a bunch of Sipura 841's. I see them register (with asterisk -r then "sip debug peer xxxx"), and NOTIFY. But when I try to ring the extension, it reports "busy". |
| 17:53.27 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 17:53.31 | malahoo | I also see "All channels busy/congested" sometimes |
| 17:53.52 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 17:54.57 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 17:55.45 | easement | malahoo : I think the proper term is provider. I want to call out to a regular old telephones. |
| 17:56.26 | easement | rob0 : I think the proper term is provider. I want to call out to a regular old telephones. |
| 17:56.34 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:56.36 | malahoo | You want to buy minutes to connect to the POTS? |
| 17:56.59 | rob0 | There are origination and termination providers. Some do both. |
| 17:57.12 | malahoo | There are several ways to do that. If you'll be spending more than $2000/month, you can connect direct to e.g. GlobalCrossing. |
| 17:57.41 | malahoo | We use a service called junctionnetworks.com - simple to set up, IAX, low per-minute charges |
| 17:57.43 | easement | I'm mainly just doing it as a learning experience. |
| 17:58.01 | zparta | is there any free provider i can try my asterisk setup with? |
| 17:58.14 | Juggie | yes |
| 17:58.16 | malahoo | easement: yeah I'd say try junctionnetworks. Cheap. |
| 17:58.17 | zparta | quality doesnt matter as its only for testing :P |
| 17:58.18 | rob0 | Okay, search the Wiki for "ipkall" and "stanaphone". Both free termination providers. |
| 17:58.29 | rob0 | they work quite well. |
| 17:58.29 | Juggie | theres an iax provider which gives 25cents of calls |
| 17:58.38 | Juggie | the name eludes me now |
| 17:58.38 | rob0 | termination? Am I using the right word? |
| 17:58.44 | Juggie | but you get a 25cent credit to try it |
| 17:59.00 | rob0 | They give you a phone number which POTS can call. |
| 17:59.48 | rob0 | Voipjet looks pretty good for origination (?? outbound calling), cheap and a good introductory offer. |
| 17:59.59 | rob0 | I use Asterlink for outbound. |
| 18:00.03 | hmmhesays | rob0: i think you have them backwards |
| 18:00.14 | *** join/#asterisk xxxAGENTExxx (n=x@201.65.48.130) |
| 18:00.16 | rob0 | (never did figure out the toll-free number.) |
| 18:00.19 | hmmhesays | VoipJet is considered a termination provider in my nomenclature |
| 18:00.25 | rob0 | hmmhesays: thanks, I always forget :) |
| 18:00.40 | jbeez | whats a nomenclature |
| 18:00.53 | jbeez | im j/k :P whats up guys |
| 18:00.56 | malahoo | What's the best way to debug SIP? |
| 18:01.18 | xxxAGENTExxx | alguem aqui fala português? |
| 18:01.47 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:02.22 | malahoo | I'm seeing the * box talk to the phones, and the phones can connect to the * box (e.g. to call a POTS number), but calling one of the phones by extension always gives me a busy signal |
| 18:02.52 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:02.52 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:02.54 | easement | thaks guys. |
| 18:03.00 | rob0 | I think you generally have to pay for termination (outbound calls), but there are numerous free origination (inbound calls) providers, namely ipkall and stanaphone. |
| 18:03.14 | xxxAGENTExxx | algum brasileiro por aqui? |
| 18:03.19 | xxxAGENTExxx | preciso de ajuda para implementar o asterisk |
| 18:03.23 | SomeJ | malahoo : do you have a dialplan to call the phones? |
| 18:03.27 | rob0 | Asterlink's termination in USA is dirt cheap. |
| 18:03.32 | malahoo | I do. |
| 18:03.41 | GerbilWrk | malahoo, are your phones behind a nat, and if so, do you have reinvite enabled? |
| 18:04.03 | malahoo | SomeJ: it works for some phones, and i swear it used to work for the all of them |
| 18:05.02 | SomeJ | malahoo: the ones that do not work, are they on a local network or behind any kind of nat/firewall? |
| 18:05.21 | malahoo | GerbilWrk, The phones are NATed, the * box has a public IP. I'm not sure how to enable reinvite. |
| 18:05.38 | malahoo | SomeJ: the ones that don't work are NATed behind a PIX 501 |
| 18:05.56 | GerbilWrk | malahoo, check the sip.conf, make sure those have nat=yes, and canreinvite=no |
| 18:06.47 | malahoo | I see a canreinvite option that's supposed to be in sip.conf (here http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Checking that... |
| 18:07.01 | SomeJ | malahoo: may also want to set qualify=yes, that will help for keeping the pinhole open in the nat |
| 18:08.06 | Juggie | if you just want to try iax, try voipjet. |
| 18:08.16 | Juggie | they will give you 25cents of free service |
| 18:09.10 | malahoo | GerbilWrk: sip.conf has NAT=yes and canreinvite=no for all extensions |
| 18:09.14 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 18:09.27 | backblue | anyone uses realtime, when i do some reload all the cached information goes away again, any option to asterisk does not do that? |
| 18:09.39 | GerbilWrk | malahoo, then try doing qualify=yes |
| 18:09.56 | *** part/#asterisk easement (i=easement@ip67-94-5-100.z5-94-67.customer.algx.net) |
| 18:10.14 | malahoo | Ok - turning on qualify, will do asterisk -rx "reload", brb |
| 18:11.28 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 18:12.33 | *** join/#asterisk davidded (n=davidded@p54B1DBA1.dip.t-dialin.net) |
| 18:12.35 | davidded | rehi |
| 18:12.43 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:13.32 | dgilmore | anyone have any idea why a t1 card configured for span=1,1,0,esf,b8zs will show in dmesg as TE110P: Span configured for CAS/HDB |
| 18:13.40 | [TK]D-Fender | PIX + NAT +* = serious trouble |
| 18:14.13 | [TK]D-Fender | dgilmore: Look in your zapata.cong |
| 18:14.30 | xxxAGENTExxx | algum brasileiro? |
| 18:14.36 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 18:14.37 | xxxAGENTExxx | alguem aqui fala português? |
| 18:14.40 | xxxAGENTExxx | preciso de ajuda |
| 18:14.45 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:14.56 | *** join/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 18:15.03 | backblue | xxxAGENTExxx: n falas inglês? aqui não é um bom canal, para falar em português. |
| 18:15.34 | dgilmore | [TK]D-Fender: thats just from loading drivers zapata.conf has nothing to do with that /etc/zaptel.conf sets things and its correct |
| 18:16.25 | backblue | xxxAGENTExxx: fala aqui em inglês, eu tenho de sair. |
| 18:16.28 | dgilmore | [TK]D-Fender: http://paste.ausil.us/31 is what i have in /etc/zaptel.conf |
| 18:16.42 | xxxAGENTExxx | :( |
| 18:16.45 | xxxAGENTExxx | i dont speak english |
| 18:17.23 | [TK]D-Fender | dgilmore: You seem to configure more channels than you have spans for. |
| 18:17.50 | dgilmore | [TK]D-Fender: i have a T1 card a 4 port FXO card and a 24 port FXS card |
| 18:18.27 | eliXier | 24port crazy |
| 18:18.32 | dgilmore | [TK]D-Fender: do i need to configure spans for the non T1 cards |
| 18:18.54 | dgilmore | eliXier: i couldnt convince the bosses to go to sip handsets |
| 18:18.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:19.04 | *** join/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 18:19.09 | [TK]D-Fender | dgilmore: Terrible..... |
| 18:19.23 | [TK]D-Fender | dgilmore: That you chose to use a TD2400P for it = even worse |
| 18:19.37 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:19.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:20.16 | fjean5 | guys, how are you...would you know how to connect asterisk to Voice Master (VM) by any chance ? I keep getting 406 not acceptable |
| 18:20.26 | dgilmore | [TK]D-Fender: :) i considered a pair of 12 port sangoma cards |
| 18:20.36 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 18:20.56 | [TK]D-Fender | dgilmore: Zap analog channels as a whole is a mistake. You should have gotten a mass-gateway like Mediatrix or AudioCodes. |
| 18:21.20 | tmccrary | Has anyone here setup an audiocodes gateway? I have an 8 port FXO and getting it to work with asterisk sucks. I can dial out through it, but not in (or at least the audiocodes gateway doesn't forward the call to asterisk) |
| 18:21.31 | [TK]D-Fender | dgilmore: Snagoma for that many ports is just impractical. |
| 18:22.22 | dgilmore | [TK]D-Fender: thats why i went the digium route. and i probably didint do it the best way but its done now |
| 18:22.38 | [TK]D-Fender | dgilmore: Oh well. Where are you located? |
| 18:22.49 | dgilmore | [TK]D-Fender: Peoria Illinois |
| 18:23.24 | [TK]D-Fender | dgilmore: You likst 24 E&M channels, that would imply CAS...... |
| 18:23.38 | [TK]D-Fender | list* |
| 18:23.46 | [TK]D-Fender | dgilmore: Not running PRI? |
| 18:24.01 | dgilmore | [TK]D-Fender: i have a DVA through AT&T its not PRI i get 24 channels for voice |
| 18:24.32 | dgilmore | [TK]D-Fender: it works perfect on one system but not the other |
| 18:24.39 | [TK]D-Fender | dgilmore: Ok, well the statebent of if being CAAS would be correct. |
| 18:24.49 | [TK]D-Fender | bleh, can't type today... |
| 18:25.01 | dgilmore | [TK]D-Fender: i can never type :D |
| 18:25.20 | dgilmore | esf with b8zs is what its supposed to be |
| 18:25.50 | *** part/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 18:25.58 | CunningPike | Does anyone know of a good way of providing audio out for an overhead paging system? |
| 18:27.19 | [TK]D-Fender | dgilmore: Ok, what is the actual problem? |
| 18:27.39 | [TK]D-Fender | CunningPike: Dedicated amplifier off an FXS port. |
| 18:27.40 | dgilmore | [TK]D-Fender: dmesg is showing CAS when it should show :q |
| 18:27.57 | dgilmore | ESF/B8ZS |
| 18:28.00 | dgilmore | gahh |
| 18:28.11 | dgilmore | [TK]D-Fender: i cought your misstyping :D |
| 18:28.25 | [TK]D-Fender | dgilmore: Not sure about that... does it work? |
| 18:28.53 | dgilmore | [TK]D-Fender: the system using ESF/B8ZS works im setting up the second one now |
| 18:29.08 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 18:29.19 | CunningPike | [TK]D-Fender: We have an amplifier all set up, with a cable from one of its audio inputs terminated in an RJ-11 jack. I can't just plug that into an FXS port on something like an SPA-3000, can I? |
| 18:29.25 | MikeJ | fun toys.... http://digg.com/mods/Bluetooth_Western_Electric_500_handset_mod |
| 18:30.09 | [TK]D-Fender | CunningPike: Its a raw audio amplifier? |
| 18:30.21 | CunningPike | [TK]D-Fender: Yup |
| 18:30.23 | [TK]D-Fender | CunningPike: You'll need an FXS interface for it. |
| 18:30.44 | CunningPike | [TK]D-Fender: OK - I'm looking at the wiki page now....... |
| 18:30.46 | [TK]D-Fender | Check the WIKi under "paging fxs" and you'll get a litst of manufacturers including "Viking" |
| 18:31.21 | CunningPike | [TK]D-Fender: Any better than any others? |
| 18:31.41 | [TK]D-Fender | CunningPike: I don't have any personal experience with any of them, but I've heard Viking works well. |
| 18:31.43 | malahoo | SomeJ and GerbilWrk: I added qualify=yes to sip.conf, did a reload at the CLI, power-cycled the phone, but I still get a busy (CLI reports "SIP/2005-1614 is circuit-busy") |
| 18:31.50 | CunningPike | [TK]D-Fender: k, thanks |
| 18:32.48 | *** join/#asterisk Cyt (n=danielcy@athedsl-109694.otenet.gr) |
| 18:33.06 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.145.155) |
| 18:33.16 | dgilmore | [TK]D-Fender: thanks for your help. Ill test it after everyones left today and see if it works and is just showing me bogus info |
| 18:35.04 | [TK]D-Fender | dgilmore: But does it WORK? |
| 18:35.50 | dgilmore | [TK]D-Fender: im not sure I have to wait until 5pm before i can test it. I had notice it was showing the wrong info. so i thought id ask |
| 18:37.27 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:39.15 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.242) |
| 18:40.45 | *** join/#asterisk adamowitz (n=adamowit@ip68-9-201-27.ri.ri.cox.net) |
| 18:41.02 | adamowitz | Anyone here online with FWD? I'd like to make a test call. If so, please give me your number so I can call you. |
| 18:41.05 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-be728fda35d6cead) |
| 18:41.05 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:41.43 | adamowitz | Or, if you'd rather call me: 728125 |
| 18:41.56 | *** join/#asterisk rokerr (n=rowan@38.99.162.187) |
| 18:43.38 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:46.34 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 18:46.34 | jtexter3 | Is anyone here doing a "hosted" pbx with Asterisk, either as a provider or as a user? |
| 18:47.00 | *** join/#asterisk loomis53 (n=loomis53@precisionresearchinc.com) |
| 18:48.04 | loomis53 | is anyone here familiar w/ doing predictive dialing w/ CfMC? |
| 18:48.39 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:48.44 | jbeez | If i'm getting this..... WARNING[5204]: Unable to open pseudo channel for timing... Sound may be choppy. which file holds that setting to change? |
| 18:49.08 | Corydon-w | /dev/zap/ctl |
| 18:49.24 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:49.36 | jbeez | thats the device, but which config file holds the info to point to the correct device? |
| 18:49.47 | Corydon-w | There isn't one |
| 18:49.57 | jbeez | you just make the device? |
| 18:50.04 | tmccrary | Audiocodes MP-118 FXO is a POS |
| 18:50.09 | jtexter3 | jbeez: Did you compile and install Zaptel? |
| 18:50.18 | Corydon-w | If you don't have Zaptel hardware, then load ztdummy |
| 18:50.27 | malahoo | I guess I'll try to set up IPSec between the PIX and the * box, to get rid of NAT. That should make things work easier, right? |
| 18:50.41 | tmccrary | IPsec is horrible at NAT traversal :) |
| 18:50.57 | tmccrary | ah, but in this case it sounds like you would be okay |
| 18:51.06 | tmccrary | as the NAT box is doing the virtual routing |
| 18:51.11 | jbeez | here is all I did, apt-get install asterisk, apt-get install subversion on a fresh debian box |
| 18:51.20 | Corydon-w | I hear IPv6 will solve all of our problems wrt NAT |
| 18:51.32 | Qwell | Corydon-w: I hear it bakes cookies too |
| 18:51.44 | tmccrary | ohh, cookies |
| 18:51.46 | Corydon-w | Qwell: and brews coffee |
| 18:51.54 | jtexter3 | Yeah, if only the industry would pick up IPv6. Although, a friend and I were discussing, and it's quite possible with NAT that IPv6 may never come to be |
| 18:51.56 | mut | i heard it was espresso |
| 18:51.59 | Qwell | Corydon-w: new version picks and grinds the beans too |
| 18:52.07 | *** join/#asterisk new2voip (n=new2voip@secure.kayacorp.com) |
| 18:52.16 | syzygyBSD | sweet, I'ma hafta get me some of this IPv6 |
| 18:52.26 | jtexter3 | jbeez: I've never used apt-get to install the asterisk packages. I would try apt-get install zaptel |
| 18:52.43 | dgilmore | jtexter3: ipv6 will come in. it will get forced by asia where there is a short supply of ip's |
| 18:53.09 | Corydon-w | I actually asked our ISP if we could get IPv6 addresses, in addition to our IPv4 addresses... His response was that that was an upgrade to his core routers that he didn't want to have to do |
| 18:53.18 | dgilmore | jtexter3: though ive heard stories of isps in asia providing nat only service. i.e. you dont geta public ip from the isp |
| 18:53.37 | dgilmore | Corydon-w: ive gotten told that also |
| 18:53.53 | [TK]D-Fender | Corydon : What ELSE do your Rice Crispies say to you? :) |
| 18:54.01 | tmccrary | I asked our ISP(ATT)'s engineer about getting allocated ipv6 address space.... he said I'd have to be on Internet2 |
| 18:54.21 | jbeez | jtexter3: I'm doing that now.... how do you install your asterisks installations? |
| 18:54.24 | adamowitz | Anyone here online with FWD? I'd like to make a test call. If so, please give me your number so I can call you. |
| 18:54.26 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 18:54.28 | jtexter3 | dgilmore: That's basically what our decision came down to, that IPv6 was supposed to come about to solve the shortage of IP addresses, but ISP's have been slow to adopt because it takes upgrades across the board, and NAT essentially solves the primary issue, which is lack of IP addresses |
| 18:54.42 | tmccrary | NAT also makes thing extra fun |
| 18:54.55 | Juggie | nat isnt the problem people think it is. |
| 18:54.57 | tmccrary | Hooray for NAT! |
| 18:55.08 | Corydon-w | NAT isn't the solution that people think it is |
| 18:55.16 | new2voip | I get this asterisk error when I try and dial using sjphone. WARNING[24424]: pbx.c:2377 __ast_pbx_run: Timeout, but no rule 't' in context 'outgoing' Incoming calls work fine. Hope someone here knows what the problem is. thx |
| 18:55.17 | jtexter3 | jbeez: I'm a CentOS user, though I've considered using Debian. Currently, I download the release tarballs from the asterisk.org site, and compile by hand |
| 18:55.27 | mtaht4 | NAT is like a 70 billion dollar a year *industry* now. |
| 18:55.28 | jbeez | you can hop into an existing ipv6 implementation w/ a tunnel on openbsd |
| 18:55.29 | Juggie | agreed, it would be all great to have ipv6, but right now nat isnt the horrible problem people assume that it is. |
| 18:55.30 | Corydon-w | A good firewall can protect machines with real IPs as well as a NAT can do |
| 18:55.33 | loomis53 | So that's a no to my predictive dialer question, i take it :) |
| 18:55.37 | dgilmore | jtexter3: yeah it also breaks lots of stuff. it wont be until most of asia is ipv6 only taht businesses elsewhere demand ipv6 space it will come out |
| 18:55.38 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:55.41 | DrAk0 | while im trying to connect to asterisk with ekiga i get this error http://pastie.caboo.se/14788 |
| 18:56.20 | Corydon-w | loomis53: There is not predictive dialer in Asterisk, though there are various third party modules floating around that do that |
| 18:56.20 | dgilmore | jtexter3: i use asterisk on Fedora and CentOS. i build my own rpms |
| 18:56.30 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:56.33 | jbeez | jtexter3: do you have some sort of step by step guide to installing by hand? is there any reason you use centOS over debian? |
| 18:56.40 | Corydon-w | s/not/no/ |
| 18:56.43 | Juggie | centos is great :) |
| 18:56.52 | Corydon-w | jbot: thanks |
| 18:56.52 | jbot | Corydon-w: gern geschehen |
| 18:57.02 | jbeez | really I haven't used it, I don't know anything about it |
| 18:57.23 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 18:57.24 | jtexter3 | jbeez: I've never tried Debian to be honest. I like CentOS because it's based off Redhat Enterprise, which appears to be extremely solid. |
| 18:57.32 | dgilmore | jbeez: i use CentOS because its stable long supported and im a RHCE |
| 18:57.37 | [TK]D-Fender | jbeez: If you're already comfortable with Debian, then there is no reason to qwitch. Only thing I can say is forget about downloading * in package format. Get the FTP tarballs and compile it yourself |
| 18:57.52 | jtexter3 | CentOS is simply Redhat Enterprise, with all the logos and stuff that says Redhat stripped out |
| 18:58.04 | tmccrary | and the support |
| 18:58.18 | Juggie | who needs support :) |
| 18:58.36 | tmccrary | depends on your skill level, I suppose |
| 18:59.06 | jbeez | listen, I started with redhat 4.2 like 7 years ago, i switched to freebsd w/ its fabulous ports and packages system that downloads tar files, deps, compiles, everything for you, but I'm trying to learn asterisk now, so I've choosen a popular linux distro(debian) to run it on, and I might be going about it the wrong way, but i wanna learn something |
| 18:59.07 | jtexter3 | I agree with [TK]D-Fender, I would stick with whatever you are comfortable with, whether that's Debian, Slack, CentOS, BSD, or whatever |
| 18:59.07 | hmmhesays | vicidial, asterisk2billing both have em |
| 18:59.11 | loomis53 | jbot, any idea if it's more of a kludge or a solution that can be put into production for 140 station shop? we're shopping for dialers and someone mentioned asterisk as a possabillity, but the lack of tech expertise in the company makes me think we're better off w/ a packaged solution, despite the cost. |
| 18:59.59 | pigpen | Hi all..I am getting some goofy stuff on my pri,: |
| 19:00.00 | pigpen | !! Got S-frame while link down |
| 19:00.00 | pigpen | !! Got I-frame while link state 2 |
| 19:00.00 | pigpen | !! Got S-frame while link down |
| 19:00.01 | pigpen | <PROTECTED> |
| 19:00.10 | Juggie | please, no channel spamming |
| 19:00.16 | pigpen | !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! |
| 19:00.20 | pigpen | yeah..sorry... |
| 19:00.24 | jtexter3 | jbeez: If you are just getting started, I highly recommend the Asterisk book by O'Reilly |
| 19:00.45 | pigpen | but I am thinking that my telco was jacking with my pri... |
| 19:00.58 | jtexter3 | But basically, download the tarballs from asterisk.org. They are libpri, zaptel, asterisk, asterisk-addons, and asterisk-sounds |
| 19:01.09 | jtexter3 | Unpackage, and compile in that order |
| 19:01.41 | pigpen | I would like to confirm my suspicions... |
| 19:01.52 | jtexter3 | Now, question for someone with Debian experience, does "make config" in zaptel and asterisk work the same in Debian as it does with CentOS? |
| 19:02.05 | Juggie | yes |
| 19:02.17 | Juggie | wait, is there even a make config? |
| 19:02.23 | jbeez | I'm horrible with books :/ |
| 19:02.28 | Juggie | what are you trying to do? |
| 19:02.50 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 19:03.35 | jtexter3 | Juggie: make config in zaptel and asterisk simply put scripts in /etc/rc.d/init.d, so you can do service zaptel start, and service asterisk start |
| 19:03.44 | jtexter3 | As well as set both to start automatically |
| 19:03.44 | Juggie | ah, |
| 19:03.48 | Juggie | yes i forgot about that |
| 19:03.53 | tmccrary | god audiocodes equipment is horrible |
| 19:04.02 | Juggie | yes it *should* handle any distro |
| 19:04.08 | Juggie | thats not to say that it definitally does |
| 19:04.08 | jtexter3 | For zaptel, it also dumps a file into /etc/sysconfig so you can tell Zaptel which modules to load |
| 19:04.10 | Juggie | but it should ;) |
| 19:05.10 | [TK]D-Fender | tmccrary: A tad cryptic I'd say, bout not "horrible" |
| 19:06.32 | tmccrary | If I'm setting up an Audiocodes FXO gateway, should I treat Asterisk as a Proxy, Gateway or Registrar? |
| 19:07.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:07.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:09.10 | [TK]D-Fender | tmccrary: All 3. |
| 19:09.26 | [TK]D-Fender | tmccrary: From the sounds of things. |
| 19:09.41 | [TK]D-Fender | tmccrary: Aim for Registrar first |
| 19:09.41 | tmccrary | So thats a guess |
| 19:09.41 | tmccrary | ? |
| 19:09.55 | tmccrary | I've tried all three and each one individually |
| 19:12.01 | jbeez | jtexter3: if I download a centOS iso, how long would it take to setup an asterisk box to mess around with and learn on? Like how long is the centOS install, and the asterisk compiles, etc? |
| 19:12.18 | justinu|laptop | like an hour? |
| 19:13.29 | rokerr | jbeez: what about the rpath appliance? |
| 19:13.43 | jbeez | jbeez: what is that? |
| 19:13.45 | jbeez | err |
| 19:13.51 | jbeez | rokerr: what is that? :) |
| 19:14.15 | Nivex | http://www.rpath.org/rbuilder/project/asterisk/ |
| 19:14.22 | rokerr | ya that |
| 19:14.22 | jbeez | which version of centOS should I d/l? 4? |
| 19:14.36 | justinu|laptop | think they're up to 4.4 now |
| 19:14.42 | Nivex | Is anyone here going to Ohio LinuxFest (http://www.ohiolinux.org/) ? |
| 19:14.44 | tmccrary | its official now |
| 19:15.14 | dgilmore | tmccrary: yep it si |
| 19:15.42 | jtexter3 | jbeez: CentOS, probably talking an 1-1.5 hours by the time you install, download updates |
| 19:16.06 | jtexter3 | jbeez: Compiling asterisk, probably 5-10 minutes depending on the power of the machine |
| 19:16.48 | jbeez | is centOS available as an ISO image to d/l? |
| 19:17.13 | justinu|laptop | real men use PXElinux installs |
| 19:17.16 | dgilmore | jbeez: cd and DVD isos |
| 19:17.22 | Qwell | real men netboot |
| 19:17.23 | Nivex | jbeez: I would imagine so. CHeck their website. |
| 19:17.31 | justinu|laptop | PXE is netboot, nub |
| 19:17.33 | jtexter3 | jbeez: www.centos.org |
| 19:17.35 | justinu|laptop | :P |
| 19:17.38 | Qwell | netboot is way cooler |
| 19:17.41 | jbeez | im in one of the mirrors now, trying to find an iso |
| 19:17.54 | dgilmore | i use rarpd and tftp alot to do installs :D |
| 19:18.46 | justinu|laptop | oldsk00l sun way |
| 19:18.47 | *** join/#asterisk catlee (n=catlee@tor149-99-33-106.dedicated.rogerstelecom.net) |
| 19:18.50 | catlee | Hello |
| 19:18.52 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 19:18.57 | dgilmore | justinu|laptop: new school sun way |
| 19:19.09 | Qwell | /mode +q Idle |
| 19:19.14 | Qwell | stupid xchat |
| 19:19.18 | Idle | ;p; |
| 19:19.20 | Idle | er, lol |
| 19:19.22 | justinu|laptop | well, they been doing that for years |
| 19:19.47 | catlee | I'm thinking of buying a linksys spa-3102 to get started with voip and * at home. Are there any gotchas with this ATA? |
| 19:19.48 | dgilmore | justinu|laptop: my brand new sun hardware does it that way. so its what i use to egt linux on |
| 19:20.42 | justinu|laptop | yeah, just funny that a sparc 20 uses the same method |
| 19:20.56 | Qwell | ss5 does too |
| 19:21.11 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:21.12 | justinu|laptop | anything with openfirmware |
| 19:21.16 | [TK]D-Fender | catlee: So you want to use it to bring in your existing analog line? |
| 19:21.17 | Qwell | obp |
| 19:21.18 | dgilmore | justinu|laptop: its just suns way i think the macs work nearly teh same way also |
| 19:21.24 | justinu|laptop | yep |
| 19:21.35 | Qwell | obp rocks |
| 19:21.47 | dgilmore | Qwell: it does indeed. |
| 19:21.47 | *** join/#asterisk gbarosio (n=gbarosio@host172038.metrored.net.ar) |
| 19:22.14 | dgilmore | im going to setup my home asterisk box on my SB150 one day just to have a sparc one |
| 19:22.14 | Qwell | I like how the new ones even have a layer before that, heh |
| 19:22.14 | catlee | Both to connect to my analog line, but primarily so that I can use my analog handsets for voip |
| 19:22.19 | justinu|laptop | intel macs are using EFI firmware tho, iirc |
| 19:22.39 | dgilmore | justinu|laptop: yeah they are same as itanium |
| 19:23.08 | gbarosio | hello there, got a quickie one, I need to find out the last login attempt from a user, and I am not finding that info around, sip show user <user> doesn't shows that. Any hints? |
| 19:23.23 | dgilmore | Qwell: i dont like that the ALOM can be telneted into but no ssh |
| 19:23.23 | Nugget | telnet is eeeeeeevil! |
| 19:23.25 | catlee | if all goes well, then I'd like to get rid of my analog line all together...but that's a ways off yet I think |
| 19:23.39 | justinu|laptop | better than the sun serial LOM |
| 19:23.47 | Qwell | dgilmore: yeah, that's kinda...meh |
| 19:24.14 | *** join/#asterisk bbz_ (n=will@static-216.87.37.130.primary.net) |
| 19:24.25 | [TK]D-Fender | catlee: Its a great little unit. You should be happy with it. |
| 19:24.35 | *** join/#asterisk Chicago (n=Chicago@c-71-201-141-218.hsd1.il.comcast.net) |
| 19:24.38 | dgilmore | has anyone here put * on a sparc linux box? |
| 19:24.44 | Qwell | dgilmore: yep |
| 19:24.55 | bbz_ | anyone have experience w/ /dev/dsp and telling which channel to output audio on? mono left / mono right ? |
| 19:25.00 | catlee | [TK]D-Fender: from reading the manual, it seems as though it might be running * itself :) |
| 19:25.03 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 19:25.15 | Qwell | catlee: I don't believe it does. One of them do however. |
| 19:25.21 | Qwell | allegedly, anyhow |
| 19:25.28 | Chicago | Anybody using asterisk to voice control their computer? |
| 19:25.47 | Qwell | Chicago: like "computer, kill Flanders"? |
| 19:25.59 | tmccrary | Yes, I am using it lightning row |
| 19:26.05 | Chicago | Qwell: Yeah |
| 19:26.39 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:26.48 | Chicago | Qwell: I was thinking "computer play me a song" |
| 19:27.19 | QbY_ | Chicago.. That's like using a shotgun to kill a housefly. |
| 19:27.32 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:27.40 | [TK]D-Fender | catlee: No, nothing of the sort. Its just got a lot of little configuration bits in there. So many ways it can intelligently route calls |
| 19:27.54 | gbarosio | any idea on how to find out when did a user performed a connection against asterisk? the equal to `last` for a linux, but in the asterisk shell |
| 19:27.56 | *** join/#asterisk eivindtr (n=wingnut-@i170192.directconnect.no) |
| 19:28.18 | Chicago | QbY_: I was thinking of using asterisk to talk to the computer to queue up music... ( or more than one caller using asterisk to queue up music ) |
| 19:28.24 | gbarosio | any hint / idea, google seems flacky on that one. |
| 19:28.31 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:28.33 | eivindtr | Hey all. Can someone tell me if it's right that for queues, rrmemory and roundrobin seem to be doing the same thing? |
| 19:29.19 | *** part/#asterisk rokerr (n=rowan@38.99.162.187) |
| 19:29.33 | dgilmore | Qwell: did it work well? |
| 19:29.36 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:29.36 | Qwell | dgilmore: sure |
| 19:29.53 | Qwell | my box is a bit...umm...beefy, though |
| 19:29.59 | dgilmore | Qwell: cool i love my sparc linux so it will be fun to do |
| 19:30.03 | Qwell | excessively so |
| 19:30.15 | gbarosio | chaps, I am not an asterisk freak, and I need some guidance, just the a hint, please :) |
| 19:30.19 | dgilmore | Qwell: what is it? |
| 19:30.24 | Qwell | dgilmore: t2000 :p |
| 19:30.37 | dgilmore | Qwell: cool i have a T1000 |
| 19:30.57 | dgilmore | but ill probably put * on either my ultra60 or SB150 |
| 19:31.16 | dgilmore | Qwell: what distro are you running on it |
| 19:31.34 | Qwell | dgilmore: I had to go with Gentoo.. ubuntu was being incredibly stupid |
| 19:31.49 | dgilmore | Qwell: im running aurora on mine |
| 19:31.57 | Qwell | I run aurora on my ss5.. |
| 19:32.00 | Qwell | it's done me well |
| 19:32.20 | dgilmore | i installed ubuntu to debug some hardware issues it was nasty |
| 19:32.24 | dgilmore | :) |
| 19:32.31 | gbarosio | centos, asterisk 1.2.6, come on guys, I'll pay the beers later :} |
| 19:32.35 | blaylock | Has anyone ever seen anything like this before? !! Got reject for frame 78, retransmitting frame 78 now, updating n_r!!! |
| 19:32.44 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:33.40 | jtexter3 | gbarosio: What's up? |
| 19:33.43 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 19:33.50 | gbarosio | jtexter3, thanks |
| 19:33.57 | gbarosio | I need to find out |
| 19:34.14 | gbarosio | how to get a user timestamp for the last succesfull log |
| 19:34.18 | gbarosio | login |
| 19:34.28 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:34.51 | gbarosio | the equivalent to last, or lastlog |
| 19:35.20 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:35.28 | jtexter3 | The last login to the box, via a terminal, ssh, or something? |
| 19:35.40 | gbarosio | to asterisk |
| 19:36.09 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-106-254.red.bezeqint.net) |
| 19:36.18 | jtexter3 | hmm.....not so sure there is such a beast |
| 19:36.19 | gbarosio | got sip users, with passwd's, and I want to find out if employees that are not anymore in the company are still using the asterisk services |
| 19:36.50 | gbarosio | cause the sip.conf file has not been pruned, or not as it should. |
| 19:37.38 | jtexter3 | The only thing I could think of would be to query your CDR's, and prune the sip.conf |
| 19:37.57 | jtexter3 | sip show peers might help, by checking status |
| 19:38.17 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 19:38.20 | gbarosio | ta, the sip.conf prune is about to come, but as a sysadmin, I've been requested on that info. Obviously I am not the asterisk guy in the company :) |
| 19:38.22 | gbarosio | thanks chap |
| 19:39.57 | gbarosio | sip show peers. That actually works for a script running during a few days, cheers. |
| 19:40.09 | gbarosio | bye |
| 19:40.10 | *** part/#asterisk gbarosio (n=gbarosio@host172038.metrored.net.ar) |
| 19:40.14 | *** join/#asterisk l-kc (n=lokesh@estrela.nortenet.pt) |
| 19:42.16 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 19:47.16 | *** join/#asterisk kannan (i=1000@61.8.147.164) |
| 19:48.46 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 19:51.29 | jbeez | downloading the pound key distro cd |
| 19:52.18 | pingwin | how are the asterisk sounds created? recordings from asterisk or from a text-to-speech synthisizer? |
| 19:52.50 | Qwell | pingwin: huh? |
| 19:53.05 | *** part/#asterisk wonderfrog (n=mabrown@h69-11-204-58.69-11.unk.tds.net) |
| 19:53.11 | *** join/#asterisk Derkommissar (n=Cubanito@226.cimatelecom.com) |
| 19:53.13 | quid246 | The old fashioned way.. a human voice |
| 19:53.16 | pingwin | sorry, I'm using gentoo so these sounds were installed with asterisk |
| 19:53.19 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:53.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:53.27 | pingwin | cool, thats all I was wanting to verify. |
| 19:54.26 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 19:55.10 | Derkommissar | Sorry to ask a basic question,,,,, I have a digium board, and randomly the t1 wouls just drop active calls. I can never catch it while is happening...... anyone suffering this or have sofferd this.... is there anything i did wrong in the configuration of zaptel ? |
| 19:57.29 | *** part/#asterisk Derkommissar (n=Cubanito@226.cimatelecom.com) |
| 19:57.29 | *** join/#asterisk Derkommissar (n=Cubanito@226.cimatelecom.com) |
| 19:57.33 | Derkommissar | Hello ? |
| 20:01.28 | tmccrary | Does anyone know what kind of caller ID method ATT uses with centrex lines? |
| 20:01.35 | tmccrary | Bellcore |
| 20:03.53 | [TK]D-Fender | Derkommissar: Make sure your card is on its own IRQ and that it is successfully taking clock sync from the telco |
| 20:03.58 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 20:04.55 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 20:05.36 | *** join/#asterisk nekoyokoshima (i=akira@194.204.98.8) |
| 20:05.46 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 20:06.28 | nekoyokoshima | i need some help |
| 20:06.51 | nekoyokoshima | actually my problem isnt asterisk but a zaptel card |
| 20:07.22 | _DAW | hi mates |
| 20:07.26 | nekoyokoshima | whenever i plu in the T210P zaptel card On Hold Music and the VoiceMail systems won't playback anything |
| 20:07.35 | nekoyokoshima | *plug |
| 20:11.08 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:11.38 | *** part/#asterisk DrAk0 (n=ljd@unaffiliated/luisjose) |
| 20:15.19 | *** part/#asterisk Derkommissar (n=Cubanito@226.cimatelecom.com) |
| 20:15.21 | *** join/#asterisk joaovianna (i=joaovian@ool-4354d1a8.dyn.optonline.net) |
| 20:16.05 | luke-jr_work | What would be responsible for T1 quality being crappy compared to IP? |
| 20:16.17 | joaovianna | I'm trying to set asterisk binding for 2 different IP addresses in the same network card... I seens like I only can register in the first IP... Any clue ? |
| 20:17.17 | *** join/#asterisk moprilo (n=jjohn@201.192.35.138) |
| 20:17.25 | joaovianna | <PROTECTED> |
| 20:18.11 | moprilo | hi, i was looking for a posibly linksys router with a fxo and a fxs.. but that would do QoS. |
| 20:18.15 | moprilo | anyone know of any? |
| 20:18.28 | moprilo | ohh, and that i could install linux so i can install asterisk on it |
| 20:20.05 | [TK]D-Fender | moprilo: Sorry, no such thing |
| 20:20.51 | moprilo | i was looking to offer my clients a small embedded system, where i could install linux |
| 20:21.09 | moprilo | so that i wouldn't have to installa PC at their permises |
| 20:21.45 | [TK]D-Fender | moprilo: There is a new * dev kit appliance coming out that sounds like what you want. |
| 20:22.03 | moprilo | who's making it? |
| 20:22.04 | bkw_ | but you forget that thing ain't worth 4k |
| 20:22.05 | [TK]D-Fender | moprilo: Till then (even probably still), you're petter off with a small server |
| 20:22.17 | bkw_ | that whole embeded thing is a rip off galore |
| 20:22.40 | joaovianna | moprilo: Did you try Linksys WRT54G ? |
| 20:22.47 | [TK]D-Fender | bkw_: Never said it was WORTH it. |
| 20:22.55 | moprilo | but wrt54g doesn't have fxo-s or fxs.. |
| 20:22.57 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:23.03 | tmccrary | gateways |
| 20:23.04 | bkw_ | haha true |
| 20:23.07 | [TK]D-Fender | joaovianna: He's looking for an "all-in-one-miracle-box:. |
| 20:23.15 | moprilo | jaja :D |
| 20:23.16 | bkw_ | he needs to buy cisco then |
| 20:23.22 | tmccrary | Audiocodes are cryptic as hell as a wise man once said :) |
| 20:23.26 | [TK]D-Fender | ~glwaot |
| 20:23.40 | moprilo | well i could try a wrt with another linksys box, like one of those with fxo+fxs |
| 20:23.41 | [TK]D-Fender | ~glwat |
| 20:24.01 | joaovianna | moprilo: Use one device to have asterisk installed and then use any ATA available... You don't have fxo or fxs in a PC anyway... |
| 20:24.05 | [TK]D-Fender | moprilo: Nothing you can install Linux & * on as well. |
| 20:24.09 | *** join/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 20:24.14 | Cherebrum | Why does asterisk do this: http://pastebin.ca/182184 |
| 20:24.41 | Cherebrum | it seems to use up all of the channels on the PRI at once |
| 20:25.45 | Juggie | 8ball says, your dialplan is broken. |
| 20:26.09 | bkw_ | no 8ball says that your libpri is out of whack |
| 20:26.13 | bkw_ | skew |
| 20:26.23 | bkw_ | no no |
| 20:26.29 | bkw_ | 8ball says don't use asterisk |
| 20:26.32 | Juggie | nah, zap calling zap. |
| 20:26.39 | Juggie | something is amuck in that dialplan |
| 20:26.48 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-989f71e8ff0b8146) |
| 20:27.46 | Cherebrum | It's pretty simple.. it just says exten => _2XXX,1,Dial(ZAP/G1/${EXTEN}) |
| 20:28.01 | *** join/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 20:28.09 | ANTILOCAS | anybody can help me to update the firmware of my ATA |
| 20:28.16 | [TK]D-Fender | Cherebrum: What on the other end of that PRI? |
| 20:28.21 | Juggie | Cherebrum, yes, but what channels are in group1 |
| 20:28.29 | ANTILOCAS | i cant understand the manual |
| 20:28.56 | Cherebrum | The other end is a Nortel PBX |
| 20:29.38 | Juggie | so you are trying to do what, dial in from the nortel pbx, and then dial out to the nortel pbx? |
| 20:29.39 | *** part/#asterisk Sargun (n=xbmodder@atarack/staff/xbmodder) |
| 20:30.11 | Cherebrum | Juggie: yes.. for IVR functions |
| 20:30.13 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/xbmodder) |
| 20:30.58 | Juggie | i think you are essentially creating a loop. |
| 20:31.37 | Cherebrum | how is it looping? my dialplan has no loops in it and it doesn't run thru any dialplan between those messages |
| 20:32.33 | Cherebrum | it seems to just be trying to bridge random channels together |
| 20:33.12 | Cherebrum | should I not be using exten => _2XXX,1,Dial(ZAP/G1/${EXTEN}) |
| 20:33.14 | Cherebrum | ? |
| 20:33.23 | Juggie | does the nortel give you outbound access? |
| 20:33.32 | Juggie | eg, dial 8xxxxxxx to dial your cell or something? |
| 20:33.55 | Cherebrum | I think so. I haven't tried |
| 20:33.55 | Juggie | er, pstn access i should say |
| 20:34.01 | Juggie | try dialing your cell phone or something |
| 20:34.06 | Juggie | see if it still happens |
| 20:34.14 | Cherebrum | it doesn't happen all the time |
| 20:34.21 | Juggie | eg, so you change that _2XXX,1,Dial(ZAP/G1/${EXTEN}) to Zap/G1/yourcell |
| 20:34.40 | Juggie | or some other line outside the nortel |
| 20:34.43 | Juggie | and see what happens |
| 20:35.42 | Cherebrum | sometimes I get: http://pastebin.ca/182195 |
| 20:36.15 | Juggie | yah, try dialing an external number |
| 20:36.21 | Juggie | and see what happens please. |
| 20:36.59 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:37.54 | Cherebrum | Juggie: I can't easily do that. The box is on the other side of the US from me |
| 20:38.07 | nekoyokoshima | whenever i plug in the T210P zaptel card On Hold Music and the VoiceMail systems won't playback anything....without it everythings works out......any clue?? |
| 20:38.15 | nekoyokoshima | *TE210P |
| 20:38.32 | Juggie | then how are you generating calls into the nortel pbx? |
| 20:38.51 | *** join/#asterisk alex-lt (i=alex@myliu.braske.net) |
| 20:39.05 | Cherebrum | hold on.. I guess I can hack something up |
| 20:39.11 | alex-lt | i have a question just for statistic |
| 20:39.24 | alex-lt | how many developers are in asterisk project? |
| 20:40.13 | file | more then 1, less then a billion |
| 20:40.39 | alex-lt | but the really active ones? |
| 20:40.50 | *** join/#asterisk vazir (i=anton@82.198.21.17) |
| 20:41.01 | file | define "really active"... there's varying degrees... some people contribute regularly, some people do it as a good, some contribute once a year... |
| 20:41.15 | alex-lt | file: ummm... the situation is ... |
| 20:41.24 | Cherebrum | Juggie: It won't let me dialout over the PSTN |
| 20:42.04 | alex-lt | i have a friend, who hired _2_ developers, who say that they will code a really good voip app |
| 20:42.04 | alex-lt | the server and client side |
| 20:42.11 | Cherebrum | http://pastebin.ca/182203 |
| 20:42.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:42.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:42.30 | Qwell | alex-lt: well, it depends on what the app does/is |
| 20:42.30 | file | "really good voip app" is rather vague |
| 20:43.00 | alex-lt | Qwell: "better than asterisk" one |
| 20:43.20 | diclophis | what can be better than asterisk at voip? |
| 20:43.23 | file | it would take a lot of time and effort |
| 20:43.25 | Qwell | and they plan on having it done by 2007? |
| 20:43.29 | alex-lt | file: same i say |
| 20:43.35 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 20:43.45 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 20:43.48 | alex-lt | Qwell: lol, they are coding it already for almost a year |
| 20:43.50 | tRSS | I need some help with my contexts! |
| 20:43.55 | Qwell | ...why? |
| 20:44.21 | alex-lt | i really don't believe they could do it even by 2010 |
| 20:44.53 | alex-lt | unless the'ire copy/pasting code from opensource projects and trying to mke something from it |
| 20:45.05 | tRSS | i cleaned up my extensions.conf file to remove extra junk from it. I have a user in a context and i do an include => local-line , but the user is able to dial everywhere surprisingly!!! |
| 20:47.41 | alex-lt | i juts want to give my friend some statistick like how much people are active in devloping of asterisk, by saying 'active' i mean contributing code atleast once a week |
| 20:49.39 | catlee | Ok, stupid questions coming...Is a SIP URL like an e-mail address? i.e. <extension/username>@<hostname/ip>? |
| 20:49.48 | file | alex-lt: 20 people I can count who are regular active to some degree |
| 20:50.17 | alex-lt | good, that'll be good for a start |
| 20:50.33 | alex-lt | thanks for the info file |
| 20:50.54 | alex-lt | file: btw, what do you think about advertising in freenode? |
| 20:51.00 | alex-lt | mainly... in this channel ? |
| 20:51.07 | file | define advertising |
| 20:51.13 | Qwell | alex-lt: It's in poor taste |
| 20:51.17 | Qwell | depending |
| 20:51.28 | file | yeah, it depends... |
| 20:51.32 | alex-lt | advertising - msg'ing in private with advertisment of other software |
| 20:51.43 | alex-lt | with words like "it's better than asterisk" |
| 20:51.44 | file | ungood |
| 20:51.48 | alex-lt | i think so too |
| 20:52.13 | Qwell | alex-lt: let me guess... |
| 20:52.38 | alex-lt | file: then you should consider this: http://pastebin.ca/182212 |
| 20:53.20 | file | interesting |
| 20:54.03 | alex-lt | better than asterisk |
| 20:54.05 | alex-lt | heh |
| 20:54.15 | file | well, they started from scratch and targetted something different |
| 20:54.48 | file | so the statement is valid depending on what you are trying to accomplish |
| 20:55.09 | alex-lt | 09-25 23:51:14 <Cherebrum> Not cool! I was mearly suggesting that maybe your friend could make use of that instead of writing his own voip softswitch since they weren't happy with asterisk |
| 20:55.15 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 20:55.21 | alex-lt | Cherebrum: why can't you just tell this out here if you think so |
| 20:55.32 | *** join/#asterisk pdtwork (n=ptinsley@adsl-156-8-237.bna.bellsouth.net) |
| 20:55.39 | alex-lt | like "maybe try freeswich, maybe it will fit your needs" |
| 20:55.47 | alex-lt | why do you have to "hide" and adv in pm's ? |
| 20:55.55 | Cherebrum | because the asterisk guys and the freeswitch guys don't generally get along so as to not cause a big fuss I said so privately |
| 20:56.11 | file | all depends on how you behave |
| 20:56.18 | alex-lt | i agree with file |
| 20:56.18 | file | I wouldn't have minded it |
| 20:56.25 | pdtwork | oil and water with some of the guys though |
| 20:56.36 | alex-lt | i don't think anyone would make a big deal if you'd tell it nicely with some arguments |
| 20:57.07 | alex-lt | or just offer trying it out if asterisk doesn't feet my friends needs |
| 20:57.35 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:58.07 | *** join/#asterisk terrapen (n=cjs@166.70.183.98) |
| 20:58.10 | terrapen | howdy |
| 20:58.24 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:58.34 | terrapen | anybody know of an industrial-strength WiFi VoIP phone? |
| 20:58.43 | terrapen | like something that i could give our warehouse employees |
| 20:58.51 | Cherebrum | terrapen: The hitachi one is nice |
| 20:58.58 | MikeJ | :P |
| 20:59.00 | alex-lt | Cherebrum: oh, and PS. is it me or is the link not working? :) |
| 20:59.02 | terrapen | IPC-5000? |
| 20:59.03 | aptura | does the hitachi take a beating? |
| 20:59.11 | Cherebrum | Which link? |
| 20:59.12 | terrapen | these guys will deliver a beating |
| 20:59.12 | aptura | can it be dropped many times and work |
| 20:59.14 | catlee | And if a SIP URL is like an email address, how would you have multiple SIP clients in a residential setup using *, with a dynamic IP address? |
| 20:59.20 | terrapen | it will get dropped, it will get knocked around |
| 20:59.32 | alex-lt | Cherebrum: the one you gave me in the PM |
| 20:59.38 | terrapen | i need something seriously industrial-strength |
| 20:59.39 | Cherebrum | freeswitch.org? |
| 20:59.44 | terrapen | sounds like a good idea for a business plan |
| 20:59.45 | aptura | might want to consider a custom made silocone jacket around it or something. |
| 20:59.46 | catlee | Would you have to connect to the main * server first, and then dial an extension, or could you somehow connect to the final extension via the main * server |
| 20:59.48 | alex-lt | oh, it's switch not swich :) |
| 20:59.48 | terrapen | industrial-strength IP phones |
| 20:59.56 | [TK]D-Fender | terrapen : None. get an industrial analog phone and put it on an ATA |
| 21:00.08 | terrapen | drop-proof, spill-proof, idiot proof WiFi and wired IP phones |
| 21:00.11 | MikeJ | file, my favorite answer... depends... |
| 21:00.26 | pdtwork | is there a good way to tell the channels that are currently being recorded? |
| 21:00.32 | terrapen | d-fender, I'm not sure if an industrial cordless phone would work in our warehouse...it's rather large |
| 21:00.35 | alex-lt | oh, btw guys, i'm and idiot in asterisk, never used it, but just a question: is it possible to connect GSM cards to asterisk ? |
| 21:00.42 | terrapen | and a phone with a single base station would not work well |
| 21:00.50 | terrapen | alex, check out the wiki, there is stuff on GSM there |
| 21:00.58 | Cherebrum | alex-lt: gsm for celluar phone audio calls? |
| 21:01.09 | alex-lt | terrapen: i've just wanted to know if it's a yes or no ;) |
| 21:01.15 | alex-lt | but i understand that it's a yes |
| 21:01.15 | terrapen | well, yes |
| 21:01.19 | pdtwork | alex-lt: there are some sip gsm gateways |
| 21:01.26 | alex-lt | Cherebrum: yap |
| 21:01.40 | Cherebrum | alex-lt: suncomm makes an inexpensive GSM SIP gateway |
| 21:01.49 | pdtwork | i would look into those, then it will work with pretty much anything, direct sip phone, asterisk, whatever |
| 21:01.54 | Cherebrum | Suncomm SC-375 I think |
| 21:02.01 | terrapen | the hitachi IP-5000 looks delicate |
| 21:02.07 | terrapen | it would never last in our warehouse |
| 21:02.22 | Cherebrum | terrapen: I would just use a nice rugged cordless phone and an ATA |
| 21:02.27 | AndyCap | and a GSM card for the * box: http://www.voismart.it/index.php?option=com_content&task=view&id=101&Itemid=146 |
| 21:02.30 | Cherebrum | much cheaper |
| 21:02.31 | pdtwork | it's not too bad, more sturdy than most of the wifi phones |
| 21:02.34 | alex-lt | Cherebrum: you know, 'inexpensive' sounds differently in different countries :) |
| 21:02.52 | alex-lt | for me 500$ is reallllly expensive :) |
| 21:03.11 | terrapen | cherebrum, a traditional cordless phone would not work in our warehouse |
| 21:03.17 | terrapen | our warehouse takes up many acres |
| 21:03.19 | sx-wks | besides, the thing has interesting vulnerabilities : http://www.derkeiler.com/Mailing-Lists/Full-Disclosure/2005-11/0550.html |
| 21:03.19 | alex-lt | 500$ is my sallary lol :) |
| 21:03.25 | sx-wks | (and more, google for them :D) |
| 21:03.42 | terrapen | alex, you get what you pay for. |
| 21:03.42 | Cherebrum | terrapen: then you need something with repeaters |
| 21:03.57 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:04.12 | Cherebrum | I wouldn't put something like a wifi phone in a warehouse |
| 21:04.15 | alex-lt | terrapen: yeah i understand |
| 21:04.38 | AndyCap | hmm, voismart.it would do well to have pages in english as well |
| 21:04.44 | pdtwork | sx-wks: it's mainly default config stuff... I don't consider that a vuln. And the other is information disclosure |
| 21:08.45 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 21:08.45 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.4 beta now available!!! Start testing!!! -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx -=- http://pastebin.ca/ for showing others large amounts of text -=- We'll miss you Rob (lilo). |
| 21:09.27 | AndyCap | but what band are these phones in? 900 and 480? judging from the name |
| 21:09.35 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:09.45 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:10.09 | terrapen | yeah, we have a lot of wifi gear out there already |
| 21:10.13 | florz | "interfear" is neat =:-) |
| 21:10.14 | *** join/#asterisk type0 (n=type0@29-8-178-69.static.gci.net) |
| 21:10.20 | type0 | anyone know of a SIP fax machine? |
| 21:10.29 | terrapen | type0 heh |
| 21:10.42 | AndyCap | florz: scares the other users away? |
| 21:10.42 | terrapen | that's a good one |
| 21:10.43 | tRSS | type0: you mean FoIP |
| 21:10.55 | florz | AndyCap: hehe :-) |
| 21:11.08 | type0 | well, im assuming that if you can route a call over asterisk, there's gotta be SOME fax solution? |
| 21:11.40 | tRSS | i guess, no one is in the mood to resolve some real dialplan issues today |
| 21:11.59 | tRSS | type0: check the wiki, i have seen something about it some time back |
| 21:12.05 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 21:14.21 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:14.49 | *** part/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 21:15.37 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:16.29 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 21:16.51 | hmmhesays | ok is there any good way to tell if a call was reinvited with "show channels"? |
| 21:16.59 | catlee | ok, so maybe they were really stupid questions :) |
| 21:17.09 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:17.11 | pdtwork | there are no stupid questions |
| 21:17.20 | pdtwork | only prickish answers |
| 21:17.25 | catlee | ha! |
| 21:17.26 | quid246 | haha nice |
| 21:17.46 | pdtwork | type0: fax machines are very finicky and don't like to be put across voip |
| 21:18.07 | pdtwork | you can look into T.38 or something like IAXModem with Hylafax for faxing |
| 21:18.16 | pdtwork | but you really need a PSTN connection somewhere in the mix |
| 21:18.24 | pdtwork | PRI or normal pots |
| 21:18.58 | catlee | I'm just trying to understand what a home setup would look like with multiple SIP clients |
| 21:19.56 | *** join/#asterisk hohum (n=dcorbe@host81-148-10-194.in-addr.btopenworld.com) |
| 21:19.59 | luke-jr_work | pdtwork: sure... tell that to my fax machine |
| 21:20.10 | luke-jr_work | pdtwork: faxing over ulaw/alaw works fine |
| 21:20.14 | pdtwork | phone line/voip provider <--internet/pstn--> Asterisk <--ethernet--> SIP handset |
| 21:20.33 | pdtwork | luke-jr_work: for very specific setups yes it can work, but to suggest that across the board is very misleading |
| 21:20.48 | pdtwork | it doesn't work for many configs and it damn sure doesn't work on some of the faster protocols |
| 21:20.53 | luke-jr_work | pdtwork: it's worked for every fax I've ever had to send |
| 21:21.08 | luke-jr_work | you mean compressed codecs |
| 21:21.11 | pdtwork | luke-jr_work: good for you, your lucky. |
| 21:21.16 | luke-jr_work | you can't get faster than ulaw/alaw |
| 21:21.28 | pdtwork | no even with those codecs you can have problem |
| 21:21.39 | pdtwork | dropped or latent packets will screw up a fax transmission |
| 21:21.56 | pdtwork | fax machines are stupid enough to take silence as the delimiter between one side and the other side's turn to talk |
| 21:22.00 | pdtwork | so it screws up the handshake over time |
| 21:22.15 | tRSS | i need some help with my contexts and call restrictions. I have pastebin'ed here: http://pastebin.ca/182232 |
| 21:22.18 | AndyCap | the arguments are here: http://soft-switch.org/foip.html |
| 21:24.55 | pdtwork | MikeJ: howdy partner |
| 21:25.10 | pdtwork | no idea why I just went into western mode btw... |
| 21:25.20 | catlee | pdt: how are the individual SIP handsets addressed? Or are they only addressable from *, and not from "outside"? |
| 21:25.30 | pdtwork | I am in nashville downtown maybe thats part of the problem |
| 21:26.10 | pdtwork | catlee: the phones show up as sip devices in asterisk and you can setup a dialplan to have them dialed from the outside various ways |
| 21:26.48 | pdtwork | by routing a DID (normal phone number) or as an option on an auto attendant menu (press 2 for sally, 3 for bob...) |
| 21:27.30 | catlee | Ok, that's for an incoming PSTN call, right? |
| 21:27.58 | pdtwork | or voip, you can address extensions via a voip call as well so you could route those directly to the phones |
| 21:28.24 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:28.39 | catlee | via auto attendant, or could somebody call the handset directly? |
| 21:28.40 | pdtwork | with a PRI you have context around the inbound calls you can have DIDs with a pots line you just get ring to know the line is ringing (in some places you can get multiple rings to tell calls apart) but that is kind of a hack |
| 21:28.59 | tmccrary | Is there a way to jump to an extension based on ${calleridnum}? |
| 21:29.07 | pdtwork | if you have a unique way to identify the call 5551212 could go to phone 1 or the auto attendent |
| 21:29.46 | catlee | ok |
| 21:30.47 | pdtwork | tmccrary: there is mention of that here http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 21:31.08 | pdtwork | you do it like exten => placecallcamein/calleridtomatch,1,Playback(tt-monkeys) |
| 21:31.12 | C6Vette | tmccrary: exten => s/calleridnumber,1,Goto(whereever,s,1) |
| 21:31.53 | catlee | Could you have * register itself as more than one SIP client, so calls to 12345@provider.com, 12346@provider.com, etc. all go to *, and then * passes them off to the relevant handset, depending on what # was called? |
| 21:32.03 | pdtwork | sure |
| 21:32.22 | *** join/#asterisk MrTelephone (n=MrTeleph@c199-243-115-15.hb.picriverisp.net) |
| 21:32.37 | ManxPower | tmccrary, Goto(${CALLERIDNUM},1) |
| 21:32.47 | tmccrary | C6Vette: Thanks but this god damn stupid audiocodes gateway doesn't provide an extension when it signals asterisk, so asterisk goes to the s extension |
| 21:32.55 | tmccrary | thanks |
| 21:33.03 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 21:33.05 | MrTelephone | I'm trying to compile asterisk on Debian and asterisk will not compile chan_zap.c. The zaptel source is working fine but asterisk will still not compile with chan_zap.c |
| 21:33.13 | MrTelephone | Does anyone have any suggestions? |
| 21:33.18 | ManxPower | tmccrary, THAT is an issue. What sort of interface do you have between the audiocodes and the PSTN? |
| 21:33.34 | ManxPower | MrTelephone, I suggest you tell us what the error message is. |
| 21:33.43 | catlee | pdtwork: guess I'll just have to play around with it and see :) Thanks for the help |
| 21:33.57 | pdtwork | thats a good idea, you won't hurt anything |
| 21:34.13 | MrTelephone | there is no error message. When I run asterisk there are no zap commands on the CLI. I do notice when I compile the gcc command is missing -DZAPTEL_OPTIMIZATIONS. |
| 21:34.18 | pdtwork | and you can get some pretty cheap sip accounts that are pay as you go until you are comfortable with it and want to use it for sure |
| 21:34.32 | catlee | yeah, I've come across quite a few of those |
| 21:34.40 | *** join/#asterisk daysmen3 (n=primus@host217-44-109-13.range217-44.btcentralplus.com) |
| 21:34.43 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:34.44 | catlee | I've also see things like sipbroker.com and voxalot.com |
| 21:35.06 | catlee | Not really sure what value these add |
| 21:35.19 | tmccrary | Manx: I am using an Audiocodes 8 Port FXO gateway with pots lines |
| 21:35.42 | pdtwork | catlee: for sip only stuff in a month or so I would suggest taking a peek at freeswitch. We are currently knee deep in a new sip stack so we don't have registration in the new one yet, not a good fit for you right now. |
| 21:35.58 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:35.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:36.00 | hmmhesays | so can someone tell me the easiest way to tell if a sip call was reinvited? |
| 21:36.03 | ManxPower | MrTelephone, and you installed Zaptel from source? |
| 21:36.11 | MrTelephone | yes |
| 21:36.20 | ManxPower | tmccrary, you can't know what the dialled number is on POTS |
| 21:36.20 | tmccrary | ManxPower: Do you have experience with Audiocodes at all? |
| 21:36.39 | ManxPower | tmccrary, no. I just remember seeing lots of messages about how terroble they are to configure. |
| 21:36.41 | MrTelephone | I have 2 debian boxes. One asterisk compiles fine.. the other doesn't :( I'm trying to find out why asterisk is compiling differently |
| 21:36.54 | tmccrary | ManxPower: Yes, but I should be able to assign some kind of extension to it from the FXO -> SIP, like line 1, 2, 3, etc |
| 21:37.23 | ManxPower | tmccrary, now you get into the realm of "it's terrible to configure" |
| 21:37.23 | tmccrary | Thanks though, the caller ID thing is good enough |
| 21:37.25 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 21:37.32 | tRSS | help me guys: http://pastebin.ca/182232 |
| 21:37.36 | tmccrary | ManxPower: Oh, I've been there for the past 5 hours or so... ;) |
| 21:37.44 | catlee | pdtwork: freeswitch would be something I run on my server? |
| 21:37.48 | ManxPower | MrTelephone, what happens if you do a "modprobe zaptel" on the problem box? |
| 21:37.49 | tmccrary | But your tip about CID just about got me finished, thanks again |
| 21:38.12 | ManxPower | tmccrary, send a paypal for 1/2 of the amount of money my suggestion saved you to eric@fnords.org |
| 21:38.24 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 21:39.03 | MrTelephone | I get no output from that command but when I type in dmesg I can see that the card is recognized. tdm24xxp |
| 21:39.13 | MrTelephone | modprobe wctdm24xxp |
| 21:39.15 | ManxPower | MrTelephone, idd,. |
| 21:39.18 | ManxPower | odd too |
| 21:39.20 | pdtwork | catlee: ya, like asterisk it's another project there are others too, yate, openpbx, trixbox, etc... |
| 21:39.32 | pdtwork | pick the one that fits you best |
| 21:39.45 | tmccrary | Sadly, I don't have paypal. |
| 21:39.51 | ANTILOCAS | pdtwork is there one free for windows? |
| 21:39.53 | mog | trixbox = asterisk |
| 21:40.39 | ManxPower | tRSS the incoming call is not matching the user you think it is. |
| 21:40.47 | pdtwork | mog: true, but packaged up al perdy for those less interested in the guts |
| 21:41.06 | ManxPower | tRSS, in sip.conf [general] put context=INVALID then in each device in sip.conf section put the correct context= line |
| 21:41.08 | mog | yes, but the other two are not asterisk |
| 21:41.13 | catlee | I like guts :) |
| 21:41.13 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 21:41.32 | pdtwork | ANTILOCAS: I can't speak for the rest but freeswitch core developers test it on Windows, Linux, OS X, and occasional testing on BSD |
| 21:41.48 | pdtwork | Solaris is in the works |
| 21:42.00 | catlee | debian packages? :) |
| 21:42.21 | ManxPower | tRSS, a device will go to ONE context. If you don't specify a context then it will go to the in in [general] |
| 21:42.22 | pdtwork | freeswitch doesn't have any package management yet, debs and rpms are being discussed |
| 21:42.45 | tRSS | ManxPower: that would double my work. then i have to connect every context to every other context!? |
| 21:42.55 | ManxPower | pdt, anyone in telecom that is not interested in "the guts" then they should not be in telecom |
| 21:43.07 | MrTelephone | -DZAPTEL_OPTIMIZATIONS I don't get this when I compile on the machine that I'm having problems with |
| 21:43.09 | pdtwork | ManxPower: some people do things they don't want to do :) |
| 21:43.14 | MrTelephone | Maybe I should check the zaptel source |
| 21:43.24 | ManxPower | tRSS, You missed my point. If you do not specify a contect for a device in sip.conf it will STILL GO TO A CONTEXT |
| 21:44.01 | catlee | cool, I'll check them out |
| 21:44.08 | tRSS | I understand that point very well, ManxPower. I guess, I have confused myself, give me a moment to re-think the situation. |
| 21:44.23 | ManxPower | tRSS, contexts are hard to understand. |
| 21:44.33 | ManxPower | and even harder to get right |
| 21:44.45 | tRSS | ManxPower, I had them nailed. The whole mess started, when I restructured my extensions.conf file |
| 21:44.46 | MrTelephone | I was using the zaptel trunk svn maybe its broken now |
| 21:45.09 | MrTelephone | /usr/src/zaptel-1.2/zaptel.c:188: warning: 'fcstab' defined but not used |
| 21:45.11 | MrTelephone | is that normal |
| 21:45.21 | ManxPower | MrTelephone, Perhaps you should stick to RELEASED stuff. |
| 21:45.28 | qdk | what should i look for when i wanna use a SIP-account similar as IAX2 to transfer calls between servers? |
| 21:45.39 | MrTelephone | I was using fxotune from the trunk release because of its extra features |
| 21:45.49 | ManxPower | MrTelephone, ah. |
| 21:45.50 | tRSS | ManxPower; so basically, you mean I should have the context=whatever against every device in the sip.conf and then do include => statements in the extensions in conf to allow access to other contexts? |
| 21:46.08 | ManxPower | tRSS, Yes. That is the correct way to do it. |
| 21:46.33 | tRSS | ManxPower: but thats a lot of work, specially, if I have a huge, complicated dialplan! :( |
| 21:46.38 | ManxPower | and put context=INVALID in sip.conf [general] so when Asterisk accepts calls from random unauthenticated devices the calls don't go anywhere. |
| 21:46.40 | tRSS | I guess, I have no other option! |
| 21:46.49 | MrTelephone | Zaptel Version: SVN-branch-1.2-r1468 Echo Canceller: KB1 |
| 21:46.55 | MrTelephone | is that the kind of output you were talking about? |
| 21:47.02 | ManxPower | tRSS, It's much less work then being fired for allowing random people to connect into your box and make international calls. |
| 21:47.04 | *** join/#asterisk daysmen3 (n=primus@host217-44-109-13.range217-44.btcentralplus.com) |
| 21:47.21 | *** join/#asterisk type0 (i=type0@159-76-74-65.gci.net) |
| 21:47.24 | tRSS | ManxPower: I guess, that makes sense! :) |
| 21:47.36 | MrTelephone | Found a Wildcard TDM: Wildcard TDM2400P (24 modules) <-- after probing wctdm24xxp |
| 21:47.39 | ManxPower | tRSS, No matter WHAT you do, you will have to properly inclide => contexts in the dialplan. |
| 21:48.00 | MrTelephone | finally.. |
| 21:48.01 | ManxPower | And no matter WHAT I do, I can't seem to type today. |
| 21:48.07 | tRSS | lol |
| 21:48.31 | MrTelephone | Now its compiling properly after I switched trunk to svn.. the thing is trunk release of zaptel is working on the other machine |
| 21:48.40 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 21:48.41 | MrTelephone | svn = branches I mean |
| 21:48.52 | *** part/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 21:48.58 | tRSS | may be I should start putting include => statements everywhere in my dialplan, before its too late. |
| 21:49.05 | *** join/#asterisk netvulture (n=vulture@host72.netvulture.com) |
| 21:49.29 | ManxPower | tRSS, maybe you need to write a structured plan before you start mucking with the dialplan. |
| 21:49.42 | mercestes | I have phones who are getting their name and username getting criss-crossed around. |
| 21:49.59 | MrTelephone | Can I ask a question somewhat related to asterisk? Everytime I upgrade the kernel it reconfigures the grub bootloader but it changes the root image to /dev/hdc1 when it should be /dev/sda1. Do you know where to fix this problem? |
| 21:50.06 | mercestes | say I have phone1/phone1 and phone2/phone2 in "sip show peers" I keep ending up with phone1/phone2. Different phones, different contexts, different passwords... |
| 21:50.08 | mercestes | why is this happening? |
| 21:50.11 | ManxPower | tRSS, here is what *I* do. I have a context for each type of PSTN access (local, toll, international, 900 numbers), then each device is in one of those contexts. |
| 21:50.42 | ManxPower | mercestes, pastebin your sip.conf |
| 21:51.14 | *** part/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 21:51.26 | *** join/#asterisk Rubicon33 (n=dxm@dslb-084-058-218-232.pools.arcor-ip.net) |
| 21:51.27 | mercestes | It's asterisk realtime, so it's in mysql, do you want the appropriate rows? |
| 21:51.56 | Rubicon33 | hello to everybody |
| 21:52.49 | Rubicon33 | somebody here who can help me with misdn and nt-port `? |
| 21:54.29 | tRSS | ManxPower: I working along those lines as well |
| 21:57.04 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:57.34 | mercestes | manxpower: Here, http://pastebin.ca/182277 The appropriate two rows. |
| 21:58.31 | joaovianna | I'm trying to set asterisk binding for 2 different IP addresses in the same network card... I seens like I only can register in the first IP... Any clue ? |
| 21:59.08 | qdk | Phone <- SIP-> asterisk <- SIP -> SIP-provider... is a simple register => in sip.conf be enough for the communication between my asterisk and my provider? |
| 22:00.39 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-100-156.dsl.peoril.sbcglobal.net) |
| 22:00.43 | *** part/#asterisk Rubicon33 (n=dxm@dslb-084-058-218-232.pools.arcor-ip.net) |
| 22:00.49 | ManxPower | mercestes, I cannot help you with Realtime |
| 22:01.11 | ManxPower | qdk, register is ONLY to get INCOMING calls from your provider. |
| 22:01.48 | *** join/#asterisk Kraig (n=sebas@nat1.inalambrica.net) |
| 22:02.10 | qdk | ManxPower: Ok, so i need a type=peer in sip.conf, right? |
| 22:02.27 | [TK]D-Fender | qdk : And possible a USER as well. |
| 22:02.34 | ManxPower | qdk, peer would be for calls Asterisk -> Provider. |
| 22:02.38 | [hC] | <3 type=friend |
| 22:02.41 | [TK]D-Fender | qdk : And in many cases "friend" covers both |
| 22:02.49 | tRSS | ManxPower: now I have another problem. When I want finance people to talk to IT, I do an include => IT in the finance context, but this allows finance people to dial international as well. Have I screwed up someplace? |
| 22:02.54 | ManxPower | register only informs the provider what your IP address is, it does nothing else. |
| 22:03.07 | ManxPower | tRSS, yes, you screwed up. |
| 22:03.13 | tRSS | :) |
| 22:03.30 | qdk | ManxPower: ok, thx a lot... i gives a bit more to go with. |
| 22:03.35 | ManxPower | tRSS, why do you have IT and finance in different contexts. |
| 22:04.16 | ManxPower | tRSS, Do this. create contexts for local, long distance, and international. Have NOTHING in those contexts except for exten lines to call those types of numbers. |
| 22:04.30 | tRSS | because I want IT to be able to dial anywhere they want, but finance shouldn't be able to do so. Secondly, IT has a different IVR for voicemail/call forwarding/etc, while finanace has a simple voicemail IVR |
| 22:04.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 22:04.51 | ManxPower | example: [local-trunks] exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1} |
| 22:05.16 | ManxPower | then create a context for devices that are only allowed to call local. |
| 22:05.34 | ManxPower | example: [local-access] include => local-trunks include => extensions |
| 22:05.39 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 22:05.48 | ManxPower | where [extensions] is where all the Dial lines for extensions are. |
| 22:05.53 | jbalcomb | since when is my GXP-2000 looking for a .htaccess file on my provisioning server? any one know what should be in it? |
| 22:06.02 | tRSS | thats what I have done!!! can I include multiple contexts in sip.conf? |
| 22:06.07 | ManxPower | for devices that can call local and extensions you would have context=local-access in sip.conf |
| 22:08.44 | *** part/#asterisk MrTelephone (n=MrTeleph@c199-243-115-15.hb.picriverisp.net) |
| 22:09.37 | ManxPower | tRSS, http://pastebin.ca/182294 |
| 22:09.53 | Kraig | jbalcomb: you mean http://httpd.apache.org/docs/1.3/howto/htaccess.html ? |
| 22:10.35 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 22:10.59 | jbalcomb | Kraig: well, i'll check that out for general info but i used to fake fm.grandstram.com/gs/ to auto-provision my gxp-2000s and now I'm getting an error about accessing a .htaccess file. |
| 22:11.06 | ManxPower | tRSS, or if you MUST. http://pastebin.ca/182295 |
| 22:11.22 | tRSS | ManxPower: I think I have it nailed now. testing it now. give me a minute and I will provide you a feedback (nice imaginative secret: stupidexample!) :) |
| 22:11.24 | jbalcomb | I'm wondering if GS changed something or if I've just broken something I didn't realize in the last month or two. |
| 22:12.21 | jbalcomb | [TK]D-Fender: Have you noticed anything like this? |
| 22:12.34 | ManxPower | jbalcomb, I think I know what your problem is. |
| 22:12.44 | ManxPower | jbalcomb, you are using Grandstream, that is your problem. |
| 22:13.25 | [TK]D-Fender | jbalcomb : Sorry.. you know me well enough to know I'd never dirty my hands working with a GXP :) |
| 22:14.08 | jbalcomb | haha! I win cause I don't care! |
| 22:14.25 | [TK]D-Fender | ManxPower : I'm taking a litking to the Aastra 480i as well. Another decent choice. |
| 22:14.28 | ManxPower | jbalcomb, no HTTP client should EVER try to access .htaccess. Preventing client access to .htaccess file is a basic requirement of web security. |
| 22:14.32 | jbalcomb | but i do feel ya. it's just that they were here before me and that is what I have to work with. |
| 22:14.35 | CrashHD | aastra's I really enjoy |
| 22:14.35 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 22:14.44 | CrashHD | 480's and the 9133's |
| 22:15.10 | [TK]D-Fender | 9133 is just too low-end for my tastes. |
| 22:15.21 | CrashHD | just looks low end |
| 22:15.23 | CrashHD | sounds good |
| 22:15.42 | jbalcomb | ManxPower: hrm.. ok, maybe i jacked my apache config or something wierd when I wasn't looking. i had to make some changes to my configs to get the Polys to auto provision. |
| 22:15.42 | ManxPower | I guess I really should pick up all the tools, cables, and 30 AV modulators off the living room floor before the BF gets home. |
| 22:15.48 | [TK]D-Fender | CrashHD : I meanin terms of user interface. I'm sure the audio & physical quality are sufficient. |
| 22:16.04 | CrashHD | agreed |
| 22:16.25 | [TK]D-Fender | CrashHD : I supposed for well-roundedness I should pick myself up one of each.... |
| 22:16.33 | [TK]D-Fender | CrashHD : Like I have with Polycom alreadt |
| 22:16.34 | CrashHD | most my users don't want to mess with their phones |
| 22:16.42 | CrashHD | that is what I did |
| 22:16.57 | *** join/#asterisk saftsack (n=saftsack@p54A7D32F.dip.t-dialin.net) |
| 22:20.20 | [hC] | does anyone know of ANY brand of wireless headset + lifter that works on polycom phones? |
| 22:20.28 | [hC] | I realize the plantronics lifter is a no go. |
| 22:20.39 | aptura | CrashHD as in clients or users in your company and what do thay dont want to mess with? |
| 22:21.31 | MikeJ | my favorite headset is: http://digg.com/mods/Bluetooth_Western_Electric_500_handset_mod |
| 22:21.32 | CrashHD | most users (with in our company) as well as the various companies I deal with do not want to use any of the menu options of their phone, regardless of the phone interface, they would A) rather call tech support and have them deal with it or b) use some type of cpu interface |
| 22:21.37 | MikeJ | but it doesn't come with a lifter |
| 22:23.16 | [TK]D-Fender | [hC] : like I told you before 1 simple screw makes the Plantronics lifter work jsut fine |
| 22:23.33 | [hC] | [TK]D-Fender: do you have a picture you can show me of what youve done? |
| 22:23.46 | [hC] | [TK]D-Fender: and of course, like i mentioned before, the handset also has to be functional. |
| 22:23.52 | [hC] | [TK]D-Fender: which kinda makes things hard. |
| 22:24.33 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:25.24 | [TK]D-Fender | [hC] : Not hard. you just leave it flat on the table |
| 22:25.31 | Druken | does anyone have experince with the linksys 9000 ? |
| 22:25.50 | [hC] | [TK]D-Fender: do you have a pic? |
| 22:26.30 | [TK]D-Fender | [hC] : All you doo is lok at the top edge of the lifter bar. You'll realize it has a hollowed back. you take a 1/2" screw and screw it into the middle of that outer edge so it depresseds the hook-switch. end of story. |
| 22:26.39 | [TK]D-Fender | [hC] : I took it down. |
| 22:26.53 | *** join/#asterisk nentis (n=nentis@hotblack.opensourcery.com) |
| 22:27.03 | *** part/#asterisk rgeek (i=rgeek@nat/digium/x-7b59936e28e6ac2c) |
| 22:27.05 | [TK]D-Fender | [hC] : They're all wired now having come to realize that cordless blows when you can't tell who's calling or transfer or anything |
| 22:27.15 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:27.52 | nentis | I've sucessfully got the intercom feature to work with a Snom 320 (dial #104 sends "SIPAddHeader("Call-Info:sip:10.0.0.167\;answer-after=0")"). |
| 22:27.59 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:27.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:28.03 | nentis | I would like both sides to beep indicating an intercom session has been established. |
| 22:28.09 | [hC] | [TK]D-Fender: oh you were using the dark plastic attachable lifter bar, that comes in the box? |
| 22:28.12 | nentis | I can cause the caller to hear a beep, but not the receiver. |
| 22:28.24 | nentis | Anyway to have the Snom auto-answer, then beep? |
| 22:30.08 | [TK]D-Fender | [hC] : Sounds about right. its a flat-edge little bar thats jsut part of the lifter. the back was hollow like I said so ti was easy to wedge a small screw into. |
| 22:30.51 | type0 | make[3]: *** [/root/zaptel-1.4.0-beta1/pciradio.o] Error 1 |
| 22:30.51 | type0 | make[2]: *** [_module_/root/zaptel-1.4.0-beta1] Error 2 |
| 22:30.51 | type0 | make[2]: Leaving directory `/usr/src/kernels/2.6.9-42.0.2.EL-i686' |
| 22:30.52 | type0 | make[1]: *** [linux26] Error 2 |
| 22:30.52 | type0 | make[1]: Leaving directory `/root/zaptel-1.4.0-beta1' |
| 22:30.52 | type0 | make: *** [all] Error 2 |
| 22:30.55 | type0 | why would I be getting that? |
| 22:31.54 | ManxPower | nentis, why not answer after 1 ring and set the ring to be a beep |
| 22:31.58 | *** join/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 22:32.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:32.31 | nentis | hm. |
| 22:32.36 | nentis | I'll look into that. |
| 22:33.46 | fjean5 | guys, it probably has been covered somewhere but why the digits are not influenced the SetLanguage () ? it's working for normal prompts, but...is there a quick workaround ? |
| 22:34.18 | fjean5 | I can't overwrite my english prompts... |
| 22:34.27 | ManxPower | fjean5, Digits in voicemail or digits in saydigits? |
| 22:34.35 | fjean5 | voicemail |
| 22:34.45 | ManxPower | fjean5, you need to set the language in voicemail.conf |
| 22:35.06 | ManxPower | At least, I THINK you do. |
| 22:35.13 | fjean5 | mm, ok, let me check that |
| 22:35.20 | ManxPower | does SayDigits use the correct language if you use SetLanguage? |
| 22:35.32 | ManxPower | fjean5, I assume you are using Asterisk 1.2? |
| 22:35.36 | [TK]D-Fender | it certainly should... |
| 22:35.37 | fjean5 | yes |
| 22:35.42 | fjean5 | let me check |
| 22:35.56 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 22:37.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:37.33 | [TK]D-Fender | Hmmm..... 480i's configurable soft-keys are pretty damn nice..... |
| 22:38.02 | [TK]D-Fender | Ok, I'm going to add them to my list of recommendable phones :) |
| 22:38.11 | [TK]D-Fender | ManxPower : Ever played with one? |
| 22:38.31 | fjean5 | ManxPower: I am not seeing anything about language in voicemail.conf |
| 22:38.59 | [TK]D-Fender | fjean5 : pastebin the call attempt |
| 22:39.06 | ManxPower | [TK]D-Fender, Nope. My largest customer has 400 people, 18 offices, in 2 states and 3 full time IP people and 1 consultant. Supporting ONE brand of phone is more than enough work. |
| 22:40.11 | [TK]D-Fender | ManxPower : Ok, I can sympathize. If eve you get the oportunity it'd be a good experience for you I think. |
| 22:40.47 | ManxPower | [TK]D-Fender, we standardized on Polycom |
| 22:40.52 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 22:40.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:40.56 | ManxPower | good or bad, that is the standard. |
| 22:41.17 | [TK]D-Fender | ManxPower : I love my Polycom's too... jsut saying if you were to ever look at anything else, that I believe you'd like the 480i. |
| 22:41.50 | [TK]D-Fender | ManxPower : Warning : SIP 2.0.1. has a presence bug thats being tracked by Polycom that i phoned in. |
| 22:42.00 | ManxPower | [TK]D-Fender, if I'm ever crazy enough to do an asterisk install for a new customer, I'll look at it. |
| 22:42.17 | justinu|laptop | hey d-fender, they ever fix that sidecar buddy watch issue? |
| 22:42.18 | MstlyHrmls | [TK]D-Fender: what's the presence bug? |
| 22:42.25 | [TK]D-Fender | ManxPower : So thats jsut one, and for them you are a consultant? |
| 22:42.37 | [TK]D-Fender | MstlyHrmls : It just flat-out doesn't work :/ |
| 22:42.44 | ManxPower | [TK]D-Fender, Huh? |
| 22:42.58 | [TK]D-Fender | MstlyHrmls : i see * sending the message, but the Poly's just seemto ignore it. |
| 22:43.08 | ManxPower | I have 2 clients with Asterisk installs (1 client with 6 or so installs, 1 client with 2 installs) |
| 22:43.14 | fjean5 | [TK]D-Fender: here is the trace: http://pastebin.ca/182330 |
| 22:43.19 | [TK]D-Fender | ManxPower : Thats just one customer and you are a consultant for them? |
| 22:43.22 | MstlyHrmls | [TK]D-Fender: neat! |
| 22:43.54 | ManxPower | [TK]D-Fender, the 400 employee customer? Yes, I am a consultant for them. |
| 22:44.03 | [TK]D-Fender | fjean5 : you sure you HAVE french digits in the right place? |
| 22:44.05 | ManxPower | Their primary non-windows constultant |
| 22:44.15 | fjean5 | yes... |
| 22:44.44 | fjean5 | [tk]D-Fender: i am sure |
| 22:45.06 | [TK]D-Fender | fjean5 : do verify for yourself.... |
| 22:45.34 | fjean5 | [tk]D-fender: if i remove the digits directory from within "fr" it doesn't change anything |
| 22:45.47 | [TK]D-Fender | fjean5 : "ls /var/lib/asterisk/sounds/digits/fr" |
| 22:46.01 | fjean5 | [tk]d-fender: exact |
| 22:46.13 | [TK]D-Fender | fjean5 : Actually your path is the reverse of what I jsut wrote. |
| 22:46.30 | [TK]D-Fender | digits/fr, not fr/digits |
| 22:46.38 | fjean5 | no prob |
| 22:46.42 | [TK]D-Fender | fjean5 : What * release are you on? |
| 22:47.14 | [TK]D-Fender | you should have an FR directory under "digits", not the other way around |
| 22:47.58 | fjean5 | [tk]d-fender: 1.2.8 mmm, so it should be ...sounds/digits/fr ? |
| 22:48.31 | fjean5 | [tk]d-fender: and not ...sounds/fr/digits , is that what you are saying |
| 22:49.16 | fjean5 | [tk]d-fender: let my try that |
| 22:49.30 | [TK]D-Fender | fjean5 : that is indeed what I am saying. |
| 22:49.38 | [TK]D-Fender | m'ecoute toi-la! ;) |
| 22:50.13 | hmmhesays | [TK]D-Fender: how do you tell if a call was reinvited on the CLI |
| 22:50.14 | *** join/#asterisk neoalex (n=neoalex@user-0cdfjlp.cable.mindspring.com) |
| 22:50.14 | fjean5 | haha |
| 22:50.31 | [TK]D-Fender | fjean5 : pis? |
| 22:50.44 | neoalex | hi guys... quick question: what does DYN stand for in "sip show peers" output? |
| 22:51.11 | [TK]D-Fender | hmmhesays : I think if you see some acnowledgement to the native bridge attempt. not jsut a "connect" because * could be in between |
| 22:51.17 | fjean5 | [tk]d-fender: nope |
| 22:51.22 | fjean5 | not working.. |
| 22:51.29 | [TK]D-Fender | neoalex : Dynamic IP I think. |
| 22:52.47 | neoalex | aah yes.. right... and I put it there :)) |
| 22:52.56 | neoalex | thanks |
| 22:53.45 | neoalex | bye |
| 22:55.06 | murf | Anyone else on the face of the whole earth trying to get Asterisk to talk to METASWITCH?? |
| 22:57.11 | aptura | never heard. got a url on that murf |
| 22:57.12 | murf | Heh, heh! Doesn't look like it. I guess I'm pretty unique! A one-in-a-million type guy! Hey! |
| 22:57.29 | murf | aptura-- what url do you need? |
| 22:57.35 | Jeekay | is there some way to have an addressbook on my phones via asterisk? |
| 22:57.45 | Jeekay | that people on their hardphones could scroll through |
| 22:57.56 | aptura | I know * has issues talking to nortel switches. Loose audio after 15 min |
| 22:58.22 | ManxPower | Jeekay, no, that is TOTALLY a function of the phone |
| 22:58.32 | aptura | jeerkay depends on the phone |
| 22:58.49 | joe | anyone using polycom 301 and 501 have a solution for doing company directories which won't screw up when merging w/ local directories? ie the xml files..... |
| 22:59.30 | Jeekay | on most normal PBXs you can program a central address book |
| 22:59.49 | CunningPike | joe: We gave up - the phonexxxxxx-directory.xml files are for the phone to store its directory contents in the event of a failure, not for loading a central directory |
| 22:59.53 | murf | aptura-- trying to get 1.4 to talk to METASWITCH, which apparently has just been updated too. Trying to place a call out. the metaswitch ain't responding to authentification. |
| 22:59.54 | ManxPower | Jeekay, Asterisk is not a "normal pbx" |
| 23:00.10 | CunningPike | aptura: Our Asterisk talks to our Nortel 100% |
| 23:00.21 | aptura | Cunning really! |
| 23:00.36 | CunningPike | aptura: I'd be looking for work if it didn't |
| 23:00.40 | joe | CunningPike: :/ k, so you don't prepopulate them or anything? |
| 23:00.46 | aptura | what model ???? |
| 23:01.11 | CunningPike | joe: No - we let people make their own entries, except when we are deploying a new phone that needs a BLF key |
| 23:01.21 | aptura | Cunningpike pm me i have some interesting questions to ask about a local company |
| 23:01.22 | CunningPike | aptura: Nortel 11C |
| 23:01.32 | joe | CunningPike: blf key? |
| 23:01.40 | CunningPike | ~blf |
| 23:01.43 | jbot | [blf] Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 23:01.56 | joe | ah yes |
| 23:02.11 | joe | and the beeping thing whatever that's called |
| 23:02.23 | joe | thanks CunningPike, good to know ... |
| 23:02.25 | ManxPower | [TK]D-Fender, any suggestions for Asterisk and Poycom firmware versions for use with a Polycom Sidecar? |
| 23:02.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:02.56 | [TK]D-Fender | ManxPower : 1.6.7 |
| 23:03.04 | ManxPower | [TK]D-Fender, thanks |
| 23:03.35 | [TK]D-Fender | ManxPower : np. 1.6.6. upped support to 48 buddies, and 2.0.1. breaks presence so its an easy choice :) |
| 23:05.16 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.166.246) |
| 23:07.13 | *** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 23:09.57 | *** join/#asterisk vin2install (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 23:11.54 | vin2install | Does anyone know where I can insert a code that can pull up a webpage automatically when an incoming call comes? |
| 23:12.38 | fjean5 | on the same computer ? |
| 23:12.55 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:12.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:12.58 | vin2install | yea |
| 23:13.45 | fjean5 | I don't know but you could System (....) your browser exec ? |
| 23:14.33 | vin2install | what do you mean |
| 23:15.12 | fjean5 | exten => 4192,2,System(/bin/yourbrowser ) |
| 23:16.35 | vin2install | how would it distinguish when a phone rings then which specific computer to pull the browser up onlu |
| 23:16.38 | vin2install | only |
| 23:17.00 | fjean5 | oh, so it's not on the same computer |
| 23:17.10 | cekc | how come when I put my Digium TE110P card in my system I can't hear audio from voicemail/moh anymore? |
| 23:17.43 | vin2install | the asterisk system is seperate and everybody has bt-101 |
| 23:17.57 | vin2install | with seperate computers |
| 23:18.38 | fjean5 | vin2install: well, if your client registers an IP address onto the asterisk to let him know where it is, it can use netcat to send a msg to the PC. There a listener could get the message and bring up the browser. That is one quick solution, there are others. |
| 23:20.19 | fjean5 | vin2install: as for an example, check this page: http://www.voip-info.org/wiki/view/Asterisk+call+notification |
| 23:23.51 | vin2install | I was thinking on something like voicerd with sugarcrm login |
| 23:24.11 | *** part/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 23:25.23 | fjean5 | well there are actually several solutions for this; an app that connects to the manager can do the job as well |
| 23:25.25 | aptura | Has anyone heard of bridging a asterisk box with the Nortel MCS5200? I just got off the phone with one of our local CLEC's who competes against Telus. Thay are having difficulty bridging the Nortel MCS5200 switch to asterisk. Seems the issue deals with how long the sip channel stays open. The audio shuts down every 15 min with mostly dead air. The engineers have not found a cause of this. The company abandoned the lab testing phase becau |
| 23:27.26 | vin2install | Since Sugarcrm already has an asterisk module integrated would it be hard to tell it to do something else to like pull up the website on a certain incoming call event that its already listening to. |
| 23:30.43 | *** join/#asterisk Samoied (n=Samoied@201.21.232.88) |
| 23:30.48 | fjean5 | vin2install: couldnt say 100%, but it should.... |
| 23:31.29 | fjean5 | aptura: do you mean conversations are terminated after 15min |
| 23:31.35 | aptura | no |
| 23:31.37 | vin2install | where would I put in the code? Would it be in the agi or php script |
| 23:32.03 | fjean5 | aptura: ok, over dead air only then |
| 23:32.07 | aptura | just the audio quits. This nortel switch is a 70k channel unit costing 2 million dollars. |
| 23:33.59 | fjean5 | ok, a good tcpdump on a call would give you some hints... |
| 23:36.03 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 23:36.45 | fjean5 | vin2install: i don't know as I dont' use it.. |
| 23:37.15 | vin2install | k |
| 23:37.51 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:38.27 | redder86 | I just want everyone to understand how *stupid* this looks and how difficult it makes it to understand what purpose "blah" serves... |
| 23:38.30 | redder86 | if (write(chan->alertpipe[1], &blah, sizeof(blah)) != sizeof(blah)) |
| 23:38.58 | justinu|laptop | are you trolling? :P |
| 23:39.06 | redder86 | no sir, complaining |
| 23:39.16 | justinu|laptop | same thing around here, i guess |
| 23:39.33 | redder86 | trolling would be if I said that there's lots to complain about |
| 23:39.38 | justinu|laptop | heh |
| 23:39.48 | justinu|laptop | i'm with you man |
| 23:39.52 | justinu|laptop | blah isn't cool |
| 23:40.25 | *** part/#asterisk fjean5 (i=fjean5@modemcable012.205-203-24.mc.videotron.ca) |
| 23:40.35 | redder86 | it would be really nice for a person who can read code, like me, to accomplish it *quickly* rather than having to dig into what "blah" is for |
| 23:40.43 | justinu|laptop | they'll just tell you to fix it |
| 23:40.44 | redder86 | now, in this case it's not too difficult to figure out |
| 23:40.47 | justinu|laptop | then not accept your patch |
| 23:40.52 | redder86 | exactly |
| 23:40.55 | justinu|laptop | because you didn't sign a disclaimer |
| 23:41.02 | redder86 | prolly |
| 23:41.10 | justinu|laptop | or they didn't like the var names |
| 23:41.17 | redder86 | apparently "blah" is better |
| 23:41.21 | justinu|laptop | anyways, im done trolling now ;) |
| 23:41.44 | nentis | ManxPower, thanks for the tip. Single-ring works. (re: Intercom beep sound for both parties) |
| 23:42.32 | aptura | Who knows of a phone with a intercom button on it |
| 23:43.55 | redder86 | aptura: any phone with a programmable button would work |
| 23:44.14 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:45.23 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:45.58 | aptura | redder, this would be for ease of use no training needed. Where employee turn over if common and emplyee station changing is often. Avaya has two intercom buttons on there phone. One called right the other called left. I did not ask but I suspect its for the right side or left side of a store. |
| 23:46.44 | designdream | i want to have an intercom with polycom 501.. is it possible? |
| 23:46.58 | CunningPike | designdream: Yes - check out 'auto-answer' |
| 23:47.29 | CunningPike | aptura: You can program a button on the Polycom, and then use one of the spare labels to label it 'Intercom' |
| 23:47.39 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:47.41 | ManxPower | CunningPike, I think the correct answer is "check the damn archives, this is not the first time the question has been asked" |
| 23:47.59 | ManxPower | ~mailinglist |
| 23:48.00 | 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 |
| 23:48.04 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.160) |
| 23:48.18 | CunningPike | ManxPower: I could say that, but I like to be a) friendly and b) helpful. I was a newbie once |
| 23:48.29 | designdream | CunningPike: so do i make a sip extension for it and have all the phones subscribe to the same 'intercom' extension? |
| 23:49.45 | CunningPike | designdream: You can program a line key to dial a speed dial that dials an extension that then uses the 'auto-answer' feature of Polycoms to have them answer when dialed. Check the wiki for 'auto-answer polycom' or 'paging polycom' |
| 23:51.28 | designdream | thanks! |
| 23:51.43 | CunningPike | designdream: You're welcome |
| 23:53.55 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:54.29 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 23:55.17 | *** part/#asterisk revstray (n=rev@ip-d88bdd4b.shfw.dedicatedcentral.com) |