| 00:05.12 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:06.27 | *** join/#asterisk smash- (~i2@c-98-232-245-30.hsd1.or.comcast.net) |
| 00:06.44 | *** join/#asterisk janeNarak (~jane@jane.lru.ac.th) |
| 00:09.17 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:10.19 | ManxPower | plutopathz, download the source code from downloads.digium.com |
| 00:11.45 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:11.56 | plutopathz | ManxPower, thanks man! i got it to work using just "./configure-gv.x" instead of "./configure-gv". i guess the ".x" does something else or is the package itself |
| 00:12.28 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 00:12.50 | doolittlework | hi there is the SHELL Command still used? |
| 00:12.54 | tzafrir | ./configure-gv ? What is that? |
| 00:13.05 | tzafrir | doolittlework, SHELL is a function |
| 00:14.58 | doolittlework | tzafrir: can i use it to run a system command and pass the result into a variable to be used in dialplan? |
| 00:15.41 | tzafrir | Set(foo=${SYSTEM(echo hi)}) |
| 00:15.49 | tzafrir | Or something along those lines |
| 00:16.05 | tzafrir | s/SYSTEM/SHELL/ |
| 00:21.08 | doolittlework | exten => _X.,n,Set(foo=${SYSTEM(date)}) |
| 00:22.01 | doolittlework | this is not working if i call a NoOp(the date is ${foo} do you know what may cause this not to work? |
| 00:23.08 | tzafrir | doolittlework, use SHELL . SYSTEM was a typo |
| 00:23.28 | tzafrir | core show function SHELL |
| 00:23.37 | doolittlework | i cant seem to be setting variables in my dialplan |
| 00:25.06 | doolittlework | to set a variable is Set(one=1) and then NoOp(${one}) should give me 1 but i am getting nothing |
| 00:25.19 | doolittlework | am i right or am i missing the plot? |
| 00:27.13 | [TK]D-Fender | doolittlework: I think you are completely failing to SHOW US |
| 00:28.27 | doolittlework | i figured it out i forgot to put a , so it was not getting to my noop |
| 00:28.51 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 00:28.57 | doolittlework | this is what i want to accomplish if you guys have some time to steer me in the right direction |
| 00:29.45 | doolittlework | when making an outbound call i get the date using epoch store it to a variable start time |
| 00:30.16 | doolittlework | the when the call ends i am using the option g to continue on with dialplan |
| 00:30.58 | [TK]D-Fender | ${EPOCH} <------ |
| 00:31.10 | doolittlework | then i get the date again and pass it to a second variable |
| 00:31.37 | doolittlework | [TK]D-Fender: thats the one please dont mind spelling im typing shorthand |
| 00:31.47 | tzafrir | doolittlework, please show the relevant part of the dialplan, and the CLI trace from running it |
| 00:33.03 | [TK]D-Fender | doolittlework: So far you only need 1 set |
| 00:33.10 | [TK]D-Fender | doolittlework: And no external commands |
| 00:33.26 | doolittlework | once i have both the times i use a script to compare the two and give me duration in HH:MM:SS |
| 00:33.34 | [TK]D-Fender | doolittlework: congratulations on your "over-complication of the day" award however :) |
| 00:33.38 | doolittlework | ok will paste now |
| 00:34.01 | [TK]D-Fender | doolittlework: Holy shit * already has all the date/time functions for this |
| 00:34.04 | *** join/#asterisk bmg505 (~leon@196-209-7-27-rndf-esr-3.dynamic.isadsl.co.za) |
| 00:34.11 | [TK]D-Fender | doolittlework: Whats with all the "external" crap? |
| 00:37.25 | doolittlework | http://pastebin.com/qcBZRa62 |
| 00:38.49 | doolittlework | i need to get the bill sec in format hh:mm:ss to dump to a com port for a custum install customer |
| 00:39.52 | doolittlework | their management system bills the customer automaticly using the cdr of their legacy pbx |
| 00:40.03 | doolittlework | i want to remove the legasy |
| 00:40.58 | doolittlework | but need to get this working first, in the long run, and when i know what the hell i am doing i want the linux and asterisk to contrl the whole process |
| 00:42.29 | [TK]D-Fender | doolittlework: "core show function SHELL" |
| 00:42.55 | doolittlework | busy reading up on it thanks for the direction |
| 00:45.37 | doolittlework | No function by that name registered. |
| 00:46.07 | doolittlework | do i have recompile asterisk to get this install or is it a addon module? |
| 00:46.08 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:51.32 | [TK]D-Fender | doolittlework: What version are you using? |
| 00:52.00 | doolittlework | Asterisk 1.4.24 |
| 00:52.11 | doolittlework | home test box |
| 00:52.32 | [TK]D-Fender | doolittlework: ANCIENT |
| 00:52.38 | doolittlework | do i need to up to 1.6 to use SHELL? |
| 00:52.40 | doolittlework | lol |
| 00:52.43 | [TK]D-Fender | doolittlework: Wake up and smell the toast burning... |
| 00:52.45 | doolittlework | like me |
| 00:54.34 | *** join/#asterisk wgoulevitch (~wgoulevit@ppp118-208-96-181.lns20.bne4.internode.on.net) |
| 00:56.22 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 00:58.46 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 00:58.52 | *** join/#asterisk bmg505 (~leon@196-209-7-27-rndf-esr-3.dynamic.isadsl.co.za) |
| 01:06.04 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:11.34 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 01:24.00 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 01:45.48 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:47.07 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:48.28 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:48.40 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:50.07 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:51.42 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:54.50 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:56.32 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 01:58.31 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 02:00.33 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 02:02.40 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 02:04.55 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 02:05.32 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 02:06.46 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 02:06.52 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 02:14.28 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 02:24.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 02:26.35 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:42.49 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 02:43.18 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 02:43.29 | *** join/#asterisk QaDeS (~mklaus@p4FC71CD2.dip0.t-ipconnect.de) |
| 03:06.59 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 03:07.45 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:17.26 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 03:17.38 | trelane | is there a guide to using AMI via telnet or similar? |
| 03:18.57 | trelane | telnet'ing to the port, sending an "Action: login" with valid credentials and I'm not getting _ANY_ response |
| 03:20.37 | trelane | http://pastebin.com/KVCmcbQE is a pastebin of the telnet session, and of rasterisk -vvvr |
| 03:25.50 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:25.53 | [TK]D-Fender | trelyou need 2 LFs |
| 03:26.00 | [TK]D-Fender | trelane: you need 2 LFs |
| 03:27.05 | trelane | 2LFs? |
| 03:27.18 | [TK]D-Fender | trelane: enter twice |
| 03:27.26 | trelane | ooh line feed |
| 03:27.43 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 03:28.00 | ManxPower | the difference between reading and reading carefully. |
| 03:28.11 | *** join/#asterisk italorossi (~italoross@187.111.235.214) |
| 03:28.45 | trelane | ManxPower, yeah the guide I used had a blank line |
| 03:28.55 | trelane | didn't actually indicate it was an additional LF |
| 03:29.21 | [TK]D-Fender | trelane: Didn't have to say. It SHOWED you. You just didn't look |
| 03:29.53 | ManxPower | trelane, the guide came from the docs/ directory of the Asterisk source? |
| 03:30.47 | trelane | mangala, no |
| 03:30.49 | trelane | ManxPower, no |
| 03:32.19 | *** join/#asterisk jezier (~jezier@nat.zgierz.net.pl) |
| 03:33.10 | trelane | ok, with the extra LF, and a deleted errant space in the username I'm logged in |
| 03:38.43 | *** join/#asterisk ideaman (~ihaveapla@74-81-241-158.static.sdyl005.digis.net) |
| 03:39.21 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-178-117.mia.bellsouth.net) |
| 03:51.34 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 03:54.52 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 03:55.10 | *** join/#asterisk _Eagle_ (~nick@wasteland.net) |
| 03:57.26 | _Eagle_ | when using dahdi_dummy (no hardware card), what are the expected (and acceptable) results for dahdi_test? i've read online that it should be 99.975 or higher, I'm getting averages of 99.94 to 99.96 with lowest values of 99.5 or 99.6... but someone told me I'm fine? |
| 03:58.31 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:04.51 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 04:05.56 | _Eagle_ | i'm trying to figure out if i have a bad server or not, so if anyone knows what numbers are acceptable, i'd appreciate the help |
| 04:06.10 | *** join/#asterisk BrendanMcc (3ce5fc51@gateway/web/freenode/ip.60.229.252.81) |
| 04:06.46 | *** join/#asterisk QaDeS_ (~mklaus@p4FC71913.dip0.t-ipconnect.de) |
| 04:08.11 | BrendanMcc | hey guys, new to all this stuff. got my asterisk box running - can make internal phone calls on the same site no problems. I have a SNOM 320 handset at home that has registered with the remote asterisk box, and i can make calls... the people on the site where asterisk is located can hear the phones ringing. I just cant seem to get any audio across the line externally... anything im missing? |
| 04:08.43 | BrendanMcc | p.s. i havent even tried to bother with trunks / inbound outbound routes etc until i get extensions working first |
| 04:09.03 | _Eagle_ | Brendan: probably a firewall issue |
| 04:09.41 | BrendanMcc | ok... any idea on what ports i need to open and on which side of each network? |
| 04:09.59 | _Eagle_ | Brendan: SIP signalling comes across on port 5060 usually, but then the actual RTP audio stream flows using UDP between 2 semi-random ports higher up |
| 04:10.22 | _Eagle_ | so the fact that the phone is ringing is probably because the SIP signal is getting there |
| 04:10.36 | _Eagle_ | but the RTP streams are probably blocked in one or both directions |
| 04:10.43 | BrendanMcc | ah ok... i notice RTP settings on the handset itself... maybe i should play with that first? |
| 04:10.47 | *** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com) |
| 04:11.00 | *** join/#asterisk Ta^3 (~tacvbo@189.146.182.11) |
| 04:11.11 | _Eagle_ | i don't know the default range of ports for asterisk for RTP... do a quick google search.... |
| 04:12.32 | _Eagle_ | is your server on a public ip or behind NAT? |
| 04:13.20 | pabelanger-lap | ~sipnat |
| 04:13.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:13.34 | pabelanger-lap | BrendanMcc: ^^^ |
| 04:14.48 | _Eagle_ | still hoping for an answer on my dahdi_test results question |
| 04:14.51 | BrendanMcc | can u take a look at this (phone web gui RTP Settings Tab) http://imagebin.org/101941 |
| 04:16.07 | _Eagle_ | Brendan: most ip phones you can leave most of the stuff default... just set the sip proxy, username, password, and if necessary, turn nat and keepalive on... the problem you have is probably a firewall issue, not inside the phone gui |
| 04:17.50 | _Eagle_ | Brendan: unless your asterisk server is behind NAT, then there are more problems to deal with |
| 04:18.10 | _Eagle_ | Brendan: if your server is on a public ip, then your phones can be behind NAT and it should still work fine |
| 04:18.21 | _Eagle_ | but only if you fix your firewall issues |
| 04:18.40 | _Eagle_ | signalling with no audio is 99% of the time a firewall issue in my experience |
| 04:22.25 | _Eagle_ | last call... can anyone help with my dahdi_test numbers? i gotta get outta here soon |
| 04:23.27 | BrendanMcc | one sec |
| 04:23.33 | BrendanMcc | thanks for your fast replies etc |
| 04:23.36 | BrendanMcc | um |
| 04:24.01 | BrendanMcc | the Asterisk box is behind a cable internet connection. this cable internet connection is also a router which runs NAT |
| 04:24.14 | _Eagle_ | everyone else seems to be asleep, busy, lagged to hell, or just plain ignoring us |
| 04:24.21 | BrendanMcc | lol |
| 04:24.35 | ChannelZ | we're all out getting laid |
| 04:24.42 | BrendanMcc | lmoa |
| 04:24.43 | ChannelZ | what do you mean, help with dahdi_test numbers |
| 04:24.57 | _Eagle_ | uhh.. right... we're all computer geeks... that doesn't happen :-) |
| 04:25.07 | BrendanMcc | haha |
| 04:26.18 | _Eagle_ | channelz: what's acceptable with dahdi_dummy (no hardware) for dahdi_test results? I read online it should be at least 99.975, but i'm getting averages of 99.94 to 99.96 with lowest numbers of 99.5x and someone told me i should be fine, but i'm worried about the low numbers |
| 04:26.32 | *** join/#asterisk residentgrey (~residentg@pool-71-191-49-141.washdc.fios.verizon.net) |
| 04:26.37 | _Eagle_ | all my other servers have digium cards in them and get 99.997+ |
| 04:26.54 | ChannelZ | I wouldn't consider 99.94 "low" |
| 04:27.01 | ChannelZ | Does the thing work? |
| 04:27.47 | BrendanMcc | yer, Asterisk box behind a NAT router.. not sure what i need to do here exactly... not even sure where to find these sip.conf files etc. i know my way around linux CLI as a beginner... i know how to VI to edit files and the like... just some simple steps to get this NAT issue resolved would be appreciated... that doco looks a bit like french to me, as i dont understand all the terms |
| 04:27.50 | _Eagle_ | brandan: if the asterisk box behind the cable connection is on a public ip, then that's great... but if it's on a private IP (i.e. behind NAT) then you're going to have to do some weird things to get your phones to talk to it across the internet |
| 04:28.04 | ChannelZ | it could just mean the clock on your machine is not so great |
| 04:28.28 | _Eagle_ | channelz: haven't tested asterisk yet, was just checking dahdi_test results first |
| 04:29.12 | _Eagle_ | channelz: i've had really bad luck in the past with Supermicro servers with bad timing chips... bad clock chips.... and although this is a different model/year, i wanted to test the timing before i go any further |
| 04:29.21 | BrendanMcc | _Eagle_: what sort of weird things :p it has a private IP on the local network, so its behind NAT |
| 04:29.30 | pabelanger-lap | ~sipnat |
| 04:29.30 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:29.40 | pabelanger-lap | BrendanMcc: read this ^^^ |
| 04:29.45 | BrendanMcc | cool |
| 04:30.09 | _Eagle_ | channelz: my other supermicro systems had really bad clocks and asterisk wouldn't work at all in some cases... so i'm worried that a 99.94 number means this is yet another example of a poor clock on a supermicro system |
| 04:30.48 | _Eagle_ | i need this to be reliable, for a production system... and i need to buy at least 10 more of these... so i want to make sure they're reliable BEFORE i buy 10 of them |
| 04:30.52 | ChannelZ | well again who cares, unless you're doing tons of MeetMe and it totally doesn't work right... |
| 04:31.36 | *** join/#asterisk d00gster (~dt@94.96.12.59) |
| 04:31.53 | BrendanMcc | pabelanger-lap: the aoc thing looks like what im looking for... i can SSH to the Centos Asterisk box... where would I find sip.conf and rtp.conf? |
| 04:32.30 | _Eagle_ | channelz: the last supermicro box i tried with a bad clock, i couldn't get music on hold to work... and audio was choppy with background/playback |
| 04:32.47 | devdvd | supermicro freaking sucks |
| 04:33.05 | devdvd | quite possibly the worst servers on the planet and that list includes dell |
| 04:33.32 | _Eagle_ | devdvd: what do you recommend for a good 1u server? preferably something not huge/long because space is tight |
| 04:34.01 | devdvd | depends on how much your looking to spend |
| 04:34.27 | _Eagle_ | as little as possible without getting a total piece of crap |
| 04:34.32 | pabelanger-lap | BrendanMcc: /etc/asterisk |
| 04:34.38 | _Eagle_ | i.e. under $1000 if possible |
| 04:34.51 | BrendanMcc | pabelanger-lap: ok ta |
| 04:37.39 | _Eagle_ | devdvd: what would you say are the best 3 servers, in order? |
| 04:37.40 | devdvd | _Eagle_: if your looking to go real cheap and still get something half decent eventhough im not a fan of dell (just bad history with their compatability with non-dell stuff) id say look at the PE R410 or if you dont mind going over just a little the PE R510 is a slightly better server |
| 04:38.13 | devdvd | or you can go with the PE 2970 if you wanna go with an AMD proc instead of intel |
| 04:38.57 | _Eagle_ | i've always hated dell... too big, too long, too heavy, requires rails.... last time i tried dell, it wouldn't fit in the cabinet and i was told to use a hacksaw to cut off the end to "make it fit" |
| 04:39.48 | _Eagle_ | at this point, i wouldn't even take a dell server if it was free |
| 04:39.55 | devdvd | none of them will be knock your socks off wow but having a team of gerbils do your data processing is better than a super micro |
| 04:40.21 | devdvd | really? ive never had a problem fitting a dell server ina standard 4 post rack |
| 04:41.11 | _Eagle_ | do you say that (about supermicro) from personal experience? i've always heard people talk very highly of them, but my experience has been poor... they LOOK nice, they LOOK well built and solid and well laid out inside... but they always seem to be poor performers for me |
| 04:41.20 | devdvd | personal experience |
| 04:42.10 | _Eagle_ | well, this was a poweredge 2550 and i think i tried a 1650... they were horrible... never used them -- sold them to a friend for pennies on the dollar |
| 04:42.29 | devdvd | what size is your rack? |
| 04:43.20 | _Eagle_ | dunno... might have measured it back then (years ago) but don't remember... whatever the standard they used at this data center was... they're all the same there. |
| 04:43.31 | devdvd | (for all the ladies in the room that was not a trick question. i was refering to his server rack) |
| 04:43.39 | _Eagle_ | but i also remember the dell required at least 2 people to lift, and probably a third to put it into the cabinet |
| 04:45.11 | devdvd | yea the older dell servers were beasts, even the 1950 was pretty unwieldy but i managed to get it racked myself |
| 04:45.12 | _Eagle_ | 90% of installations and maintenance i have to do by myself, and i have bad knees and a bad back, so i can't use servers i can't lift |
| 04:45.26 | devdvd | well |
| 04:45.32 | devdvd | most servers are gonna be fairly heavy |
| 04:46.00 | _Eagle_ | the supermicro's were always very small and light.... 15 inches deep.... this latest one with the numbers quoted above is only 10 inches deep |
| 04:46.48 | _Eagle_ | i don't need dual xeon with 4 hard drives and all that extra junk... just a decent small 1U |
| 04:47.27 | devdvd | either that or start looking into .5U |
| 04:48.05 | _Eagle_ | i could probably go 20 inches deep... maybe 22 at most.... beyond that just won't work for me i think |
| 04:48.33 | _Eagle_ | i've never seen a .5U server... who makes those? |
| 04:49.05 | devdvd | just google it |
| 04:49.29 | devdvd | i dont know much about them except they exist |
| 04:50.30 | _Eagle_ | any other brands besides dell? |
| 04:50.34 | _Eagle_ | that you recommend? |
| 04:50.48 | devdvd | probably not within your price range |
| 04:50.55 | devdvd | can look at the hp stuff |
| 04:51.03 | devdvd | maybe the entry level ibm/lenovo servers |
| 04:51.44 | _Eagle_ | my price range might have to go up to accomodate my needs |
| 04:51.59 | devdvd | i did find a dell server that might fit your needs and your price if you decided that as a last resort. the R210 is 15.5in deep. |
| 04:52.24 | _Eagle_ | i mean, i'm not having luck with the supermicro... so maybe my budget needs to increase |
| 04:52.57 | _Eagle_ | devdvd: interesting... i'll check out the R210... thanks... but i bet that needs rails, eh? |
| 04:53.40 | devdvd | yea |
| 04:54.21 | devdvd | personally though, i love the rapid rails |
| 04:55.04 | _Eagle_ | i gotta run... thanks for the advice |
| 04:55.26 | devdvd | the price isn't bad, they are really good quality (they easily supported the 1950 which is about 30-40lbs at full extension |
| 04:55.40 | devdvd | ok |
| 04:55.40 | devdvd | later |
| 04:56.25 | _Eagle_ | the cabinets we use have #12 screw holes.... not those square holes like dell uses |
| 04:57.21 | _Eagle_ | anyway, thanks... seeya |
| 04:59.10 | devdvd | lol, dell dont use the square holes for the rapid rails |
| 05:04.14 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 05:07.10 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 05:10.35 | *** join/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 05:18.16 | *** part/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 05:38.10 | *** join/#asterisk hariom (~earth@122.170.32.244) |
| 05:38.28 | hariom | Is there any robust Python library for Asterisk AGI and AMI? |
| 05:45.11 | *** part/#asterisk BrendanMcc (3ce5fc51@gateway/web/freenode/ip.60.229.252.81) |
| 05:53.38 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 06:07.05 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 06:11.14 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:48.19 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 06:48.35 | *** join/#asterisk lukehasnoname (~lukehasno@cpe-66-69-74-19.satx.res.rr.com) |
| 06:49.48 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 07:03.43 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 07:06.12 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 07:12.51 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:13.30 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 07:39.14 | *** join/#asterisk Godfather_ (~aaaa@193.153.129.150) |
| 07:39.44 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-239.vic.bigpond.net.au) |
| 07:41.13 | joobie | hey guys.. i have an IVR for callers coming in on the ISDN.. wanted to get the optimum quality for the voice recording that is played.. the ISDN uses alaw, so thinking I need to make this recording alaw to avoid any realtime conversion when it's playing.. if this is the case, are there various alaw qualities and what is the best that I can use and be supported still? |
| 07:46.08 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 07:53.50 | pabelanger-lap | joobie: alaw recording already ship with Asterisk. http://downloads.asterisk.org/pub/telephony/sounds/releases/ |
| 07:54.32 | joobie | ya.. though what is the optimal alaw encoding that can be usedD? |
| 07:54.46 | joobie | like for wav for example., i dont think u can do stereo.. has to be mono |
| 07:54.50 | joobie | and 8000hz |
| 07:54.51 | joobie | etc etc |
| 07:59.33 | tzafrir | joobie, that said, the overhead for wav/slin -> alaw conversion is really small |
| 08:00.54 | tzafrir | pabelanger-lap, speaking of sound files: I've adapted the packages to use a single repackaged tarball |
| 08:00.57 | pabelanger-lap | joobie: no point going over 8kHz, since ISDN doesn't support it |
| 08:01.43 | pabelanger-lap | tzafrir: Ya, I seen that. I was going to try it later today. How did you get around the timestamp issue for the .orig.tar.gz? |
| 08:01.54 | tzafrir | I didn't |
| 08:02.30 | tzafrir | There are other repackaged packages (asterisk was repackaged until lately. dahdi-linux still is) |
| 08:02.46 | tzafrir | I can't think of a decent way around this |
| 08:03.21 | pabelanger-lap | tzafrir: How do the debian maintainers not run into conflicts when uploading the .orig.tar.gz to buildd? |
| 08:03.42 | tzafrir | You make sure you only upload it once. |
| 08:03.58 | tzafrir | A second upload should not upload the orig tarball anyway |
| 08:05.42 | pabelanger-lap | tzafrir: true, but if I was to then create a new package with a local .orig.tar.gz (debuild -S) and upload it, the checksum in .dsc will mismatch with the previous uploaded one. I guess the maintainers download the first .orig.tar.gz via apt-get? |
| 08:06.15 | tzafrir | yes |
| 08:07.23 | tzafrir | That's also why svn-buildpackage likes to have tarballs in a separate directory - to make it simple to keep the tarballs |
| 08:14.47 | pabelanger-lap | tzafrir: Ya, I'll have to play with it some more. I was also looking to the upstream folder option but not played with it. |
| 08:15.50 | tzafrir | what do you mean by "folder option"? |
| 08:31.50 | joobie | tzafrir, yaa.. just thinking though, this IVR is going to be perm and only played down the ISDN, so figured may as well eliminate the conversion i/o |
| 08:32.15 | joobie | pabelanger, what about stereo.. all my wav's thus far have been mono |
| 08:32.18 | joobie | also bitrate |
| 08:33.58 | coppice | joobie: do you have a stereo ISDN to use stereo recordings with? |
| 08:34.08 | joobie | i dont think so |
| 08:34.11 | joobie | i thought mine was mono |
| 08:34.24 | coppice | exactly |
| 08:34.29 | joobie | is all ISDN mono? |
| 08:34.47 | coppice | ever seen a stereo telephone? |
| 08:34.57 | joobie | headset |
| 08:35.01 | joobie | two speakers |
| 08:35.27 | joobie | wasnt sure if that was just mono or stereo |
| 08:37.14 | coppice | they don't even give you good quality mono. they certainly don't use data to provide a second channel for stereo |
| 08:39.32 | *** join/#asterisk BANSAL (~bansal@117.199.125.53) |
| 08:50.21 | *** part/#asterisk janeNarak (~jane@jane.lru.ac.th) |
| 08:54.42 | joobie | cool cool thanks |
| 08:55.53 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 08:59.24 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:13.00 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 09:13.04 | [sr] | gud morning! |
| 09:15.29 | *** join/#asterisk ming_zym (~ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 09:15.39 | ChannelZ | oh hai |
| 09:16.41 | [sr] | :) |
| 09:19.59 | *** join/#asterisk yuracagran (~yuracagra@74.115.160.188) |
| 09:25.48 | *** join/#asterisk mloe (~whocrz@217.79.80.213) |
| 09:26.15 | mloe | hi, guys, can you give me a VoIP provider who can give Caller ID Spoofing? |
| 09:26.31 | mloe | I want to connect via SIP protocol :) |
| 09:26.59 | mloe | I saw this "spoofcard dot com", but I'm not sure - is this support SIP |
| 09:28.08 | ChannelZ | vitelity seems to |
| 09:29.10 | mloe | :)) |
| 09:29.12 | mloe | thank you! |
| 09:29.18 | mloe | bye bye |
| 09:30.16 | joobie | ergh.. anyone got tips in converting an audio stream to something compatible with asterisk? I hvae it at the moment in a Stereo WAVE 44100khz 32bit.. it sounds great.. as soon as i convert to 8000khz 16bit mono, it sounds like crap.. even to the point where it sounds like it's chopping off the "s" in the word "Thanks" which is being spoken.. |
| 09:30.26 | joobie | im using adobe soundsbooth |
| 09:30.32 | ChannelZ | of course, phones sound like crap as well. |
| 09:32.43 | joobie | it's going to sound worse nod |
| 09:32.45 | ChannelZ | If you're recording prompts you're probably better off doing so at 16/8kHz rather than downsampling from such a high rate |
| 09:32.49 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 09:32.52 | joobie | but wondeirng if there's some tips in converting to keep it as good as possible |
| 09:33.09 | joobie | ahh damn |
| 09:33.11 | iscario | hello, i would like to know if it is possible to launch an AGI script from the dialplan in background (my script writes in the asterisk native db and has a wait function , so i can't wait to go to the next extend until it finished) |
| 09:33.15 | joobie | already recorded at such a high rate |
| 09:34.36 | ChannelZ | the difference isn't likely to be massive |
| 09:35.03 | joobie | ahh |
| 09:35.21 | ChannelZ | but recording voice prompts at 32-bit is somewhat pointless |
| 09:35.21 | joobie | my main concern is that the word "thanks" becomes "thank" when i drop the quality down |
| 09:35.33 | joobie | yer |
| 09:35.34 | joobie | true |
| 09:36.20 | *** join/#asterisk miltux (~miltux@62.1.135.192.dsl.dyn.forthnet.gr) |
| 09:47.42 | iscario | please, where can i find piece of information avt asynchronous agi ? |
| 09:48.26 | joobie | iscario, wat do u mean |
| 09:49.12 | iscario | joobie: i read somewhere (?) that it existed an asynchronous mode for agi script |
| 09:49.36 | iscario | joobie: i don't know what it is, but maybe it can help me to solve my problem |
| 10:04.31 | iscario | no idea ? |
| 10:08.29 | [sr] | WIMPy: working great, i just let freepbx manage all configuration files without install the asterisk one's :) and i have all 3 things working good, LCR+asterisk+freepbx, kool! |
| 10:08.47 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:14.08 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 10:15.30 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 10:18.31 | *** join/#asterisk sourcode (~code@ppp-58-8-99-166.revip2.asianet.co.th) |
| 10:19.50 | joobie | iscario, why do you want to use asynchronous agi? |
| 10:19.57 | joobie | iscario, not exactly sure why you need a feature like this.. |
| 10:20.21 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:20.33 | joobie | agi afaik just calls an external script.. there's no reason you cant make your agi script asyncronous? but interested to know what you want to do this |
| 10:20.44 | iscario | joobie: maybe^^ i will paste what i wrote here few minutes ago |
| 10:20.57 | iscario | joobie: hello, i would like to know if it is possible to launch an AGI script from the dialplan in background (my script writes in the asterisk native db and has a wait function , so i can't wait to go to the next extend until it finished) |
| 10:20.57 | joobie | iscario, wats the end goal? |
| 10:21.29 | joobie | ahh |
| 10:25.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:25.59 | iscario | joobie: no idea how i can do that ? |
| 10:26.43 | joobie | sorry man eating dinner |
| 10:26.45 | joobie | im not sure |
| 10:26.48 | joobie | http://lists.digium.com/pipermail/asterisk-users/2010-January/243074.html |
| 10:26.51 | joobie | just reading around though |
| 10:26.58 | joobie | taht's a good start |
| 10:27.00 | joobie | brb dinner |
| 10:28.14 | iscario | thank you and good dinner then :) joobie |
| 10:33.52 | *** join/#asterisk Krolik13 (~Krolik13@5ac3137a.bb.sky.com) |
| 10:35.10 | Krolik13 | hi. what Web-Management application do you recommend to use to display call statistics from CDR database, including links to the mp3 files - recorded calls with Monitor application of asterisk ? |
| 10:36.12 | *** join/#asterisk iscario (~quassel@241.244.71-86.rev.gaoland.net) |
| 11:00.31 | *** join/#asterisk Z_God (~julius@wlan234015.mobiel.utwente.nl) |
| 11:03.52 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 11:18.59 | *** join/#asterisk my007ms (~my007ms@196.219.63.12) |
| 11:19.35 | my007ms | any example in sending sms over FXO channel ? |
| 11:22.32 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:29.43 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 11:31.31 | emora | Does anyone have comments on the Sangoma U100 Usb-fxo? I'm thinking of trying these for a simple, small office setup. |
| 11:39.47 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 11:41.10 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 11:54.02 | *** join/#asterisk BANSAL (~bansal@117.199.118.248) |
| 11:54.51 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 12:04.09 | *** part/#asterisk kk7av (~rob@pdpc/supporter/professional/rob-p) |
| 12:05.37 | *** join/#asterisk cusco (~trilili@213.141.21.122) |
| 12:05.41 | cusco | hi |
| 12:05.57 | cusco | once the call goes into h => extension, can I still use ${ANSWEREDTIME} variable? |
| 12:16.00 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 12:17.39 | cusco | can't I NoOp(${RDNIS}) ? |
| 12:17.42 | cusco | it comes empty |
| 12:23.47 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 12:24.44 | hariom | Hi, How to enable app_jack. What libraries do I need to select it in make menuselect? |
| 12:33.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:34.29 | *** join/#asterisk henkoegema (~henkoegem@78-21-68-204.access.telenet.be) |
| 12:36.12 | henkoegema | What is the advantage of a sip proxy (f.e. Kamailio) in front of Asterisk? |
| 12:41.41 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 12:47.47 | hariom | has anybody used app_jack? |
| 12:49.06 | hariom | henkoegema: sip proxy can handle a large load then Asterisk. |
| 12:49.31 | tzafrir | hariom, not me, but why do you ask? |
| 12:50.19 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 12:50.43 | hariom | Because I want to use and I am not able to figure out what all libraries it requires |
| 12:50.56 | tzafrir | libresample, or whatever it is called |
| 12:52.15 | *** join/#asterisk Pazzo (~ugelt@sadsl-246059.rol.raiffeisen.net) |
| 12:56.16 | *** join/#asterisk sourcode (~code@ppp-58-8-99-166.revip2.asianet.co.th) |
| 12:56.30 | hariom | tzafrir: I already have libsamplerate (and dev) but its not enabled in make menuselect |
| 12:56.52 | henkoegema | hariom: in which case should I use a sip proxy together with Asterisk ? |
| 12:56.58 | tzafrir | libsamplerate? libresample? |
| 12:57.17 | hariom | libsamplerate. There is no libresample. Only resample |
| 12:57.45 | hariom | When you want to handle VOIP calls and want to do SIP call handling with SIP Proxy |
| 12:58.26 | tzafrir | I wonder if it would be possible to use libsamplerate instead of libresample |
| 12:58.52 | hariom | henkoegema: http://www.voip-info.org/wiki/view/Asterisk+SIP+not-proxy |
| 12:59.03 | tzafrir | duh. I confused the two |
| 12:59.29 | henkoegema | hariom:thanks |
| 13:00.31 | hariom | tzafrir: when I search in synaptic, I get libsamplerate and resample as well. libsamplerate is for modifying sample rate of a sound file |
| 13:01.34 | WIMPy | I see libresample in audacity. |
| 13:02.58 | tzafrir | hariom, app_jack seems to be built here with libresample-dev |
| 13:04.58 | hariom | there is no such package in ubuntu hardy |
| 13:05.09 | hariom | I can't find in apt-get or synaptic |
| 13:09.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:13.34 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 13:17.35 | *** join/#asterisk fofware (~fabian@190.7.25.160) |
| 13:21.41 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 13:34.28 | x-demon | anyone using skype asterisk channel from digium? |
| 13:34.34 | x-demon | what about voice quality? |
| 13:36.36 | *** join/#asterisk ming_zym (~ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:37.55 | *** join/#asterisk fofware (~fabian@190.7.25.160) |
| 13:38.21 | tzafrir | hariom, sorry: libresample1-dev |
| 13:38.53 | tzafrir | Though I suspect it was not there when hardy was released. |
| 13:39.12 | tzafrir | Should be trivial to backport, I suspect. It's a simple library package |
| 13:42.12 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 13:42.55 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:45.26 | *** join/#asterisk fofware (~fabian@190.7.25.160) |
| 13:48.30 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:57.11 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 14:03.09 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 14:06.00 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-178-117.mia.bellsouth.net) |
| 14:07.16 | *** join/#asterisk ikaRus1 (~none@80.179.36.48.static.012.net.il) |
| 14:07.24 | ikaRus1 | hi |
| 14:08.18 | ikaRus1 | doesnt 'database get AMPUSER/XXX device' supposed to return all devices used by XXX user? |
| 14:09.20 | ikaRus1 | .m vk4akp hi de 4X4XX ;-) |
| 14:11.37 | *** join/#asterisk QaDeS_ (~mklaus@p4FC71913.dip0.t-ipconnect.de) |
| 14:12.07 | [TK]D-Fender | ikaRus1: Wrong channel... #freepbx <--------- |
| 14:29.01 | ikaRus1 | yep |
| 14:29.06 | ikaRus1 | i noticed |
| 14:29.13 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 14:30.26 | *** part/#asterisk ikaRus1 (~none@80.179.36.48.static.012.net.il) |
| 14:38.30 | *** join/#asterisk af_ (~getsmart@78.134.22.42) |
| 14:43.21 | *** join/#asterisk ming_zym (~ming_zym@115.171.198.247) |
| 14:53.51 | *** join/#asterisk PuraVida (~PuraVida@S0106001e52f5dd45.vc.shawcable.net) |
| 14:55.57 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 15:00.00 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 15:11.46 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:23.39 | *** join/#asterisk nalbaxxx (~nalbax@78.250.251.243) |
| 15:24.03 | nalbaxxx | hi |
| 15:24.16 | *** join/#asterisk digiv (~mlhess@141.214.234.28) |
| 15:26.26 | tzafrir | nalbaxxx, hi |
| 15:26.55 | tzafrir | ~ask |
| 15:26.56 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:58.36 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:58.36 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.33 (2010/06/18), *-Addons 1.6.1.4, 1.6.0.6 (2010/06/08), 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.2 (2010/06/08) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bug |
| 15:59.25 | *** join/#asterisk m_tadeu (~quassel@89.181.28.68) |
| 15:59.46 | pabelanger-lap | pokes infobot |
| 15:59.50 | coppice | the LPC10 in * is kinda broken and sounds awful. the one in FreeSwitch works right, and isn't too bad |
| 16:01.02 | my007ms | so what is max codec do he utilization in bandwidth ? |
| 16:01.02 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 16:01.14 | my007ms | s/he/the :) |
| 16:01.22 | p3nguin | so close! |
| 16:01.39 | p3nguin | s/so/You were/ |
| 16:01.44 | my007ms | <PROTECTED> |
| 16:13.55 | tzafrir | coppice, "the one in FreeSwitch" is the one from SpanDSP? |
| 16:14.23 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 16:19.25 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 16:19.44 | Dovid | hi. what causes this ? Exceptionally long voice queue length queuing to Local/XXXXXXX@to-outbound-c886,1 |
| 16:20.40 | my007ms | does codec bandwidth utilization different from freeswitch then asterisk ? |
| 16:20.48 | *** join/#asterisk exelnet (~exelnet@i59F7E018.versanet.de) |
| 16:21.15 | *** join/#asterisk tengulre (~tengulre@221.182.46.7) |
| 16:22.58 | Dovid | a codec is a codec |
| 16:31.08 | *** join/#asterisk Pazzo (~ugelt@sadsl-246059.rol.raiffeisen.net) |
| 16:31.20 | *** join/#asterisk freeedrich| (eeePC@hansaserver.de) |
| 16:32.43 | *** join/#asterisk lukehasnoname (~lukehasno@cpe-66-69-74-19.satx.res.rr.com) |
| 16:33.06 | coppice | tzafrir: yes |
| 16:33.26 | coppice | tzafrir: assuming you mean LPC10 |
| 16:33.37 | tzafrir | yes, that's what i meant |
| 16:35.19 | *** join/#asterisk Krolik13 (~Krolik13@5ac3137a.bb.sky.com) |
| 16:35.23 | Krolik13 | hi. what Web-Management application do you recommend to use to display call statistics from CDR database, including links to the mp3 files - recorded calls with Monitor application of asterisk ? |
| 16:40.38 | dlynes | Krolik13: for the voicemail, you've got vmail.cgi in the extras folder of your asterisk distribution |
| 16:41.17 | dlynes | Krolik13: it'll show the voicemails in whatever format you configured to record them in, from your voicemail.conf file |
| 16:45.22 | *** join/#asterisk QaDeS (~mklaus@p4FC71913.dip0.t-ipconnect.de) |
| 16:45.54 | *** join/#asterisk MasterO (~MasterO@proxy5.med-web.com) |
| 16:47.46 | Krolik13 | dlynes> i'm just recording all calls that passes thourh my asterisk pbx. So I wanna have an web app that shows me the call statistics with links to the recordered mp3 of the conversation. It is not about voicemails. |
| 16:47.55 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 16:47.59 | dlynes | Krolik13: ah |
| 16:48.52 | dlynes | Krolik13: you'd have to write that yourself, because you would had to have written the dialplan yourself...there's no standard recording directory or filename format for the recordings, or standard for how to record them in general (other than the application to do it) |
| 16:49.20 | dlynes | Krolik13: unless of course you're using freepbx or something, and then you're better off asking your question in #freepbx |
| 16:50.21 | *** join/#asterisk digiv (~mlhess@141.214.234.28) |
| 16:52.50 | *** join/#asterisk azrtyo (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 16:52.56 | azrtyo | hi there |
| 16:53.07 | azrtyo | i can't equivalent of this one |
| 16:53.13 | Krolik13 | dlynes> it is not problem of recordint the call. I just need the web-app that will display the statistics from the CDR beside the recorded call. I'm just looking what web app to try for it |
| 16:53.30 | azrtyo | wget http://downloads.digium.com/pub/telephony/sounds/asterisk-moh-freeplay-ulaw.tar.gz |
| 16:53.42 | dlynes | Krolik13: yes, but you want to be able to play back the recording, right? |
| 16:54.01 | dlynes | Krolik13: it has to know where the actual recording is, and there's no standard implemented for where that recording will be |
| 16:54.10 | dlynes | Krolik13: or even in what format it'll be |
| 16:54.28 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 16:54.36 | dlynes | Krolik13: it could be multiple files, it could be all one file, it could be mp3, it could be wav, gsm, who knows? |
| 16:54.38 | Krolik13 | dlynes> I see... what web-app do you recommand, that i'll be able to recode to include this functionality? I |
| 16:55.15 | Krolik13 | I need an web-app, that will has a lot of ways (filters, etc) to display the call details |
| 16:55.49 | *** join/#asterisk fenrus (fenrus@oklart.com) |
| 16:56.01 | azrtyo | ???? |
| 16:56.06 | azrtyo | anyone there ? |
| 16:56.10 | fenrus | yes |
| 16:56.23 | azrtyo | any equivalent of this one plz : wget http://downloads.digium.com/pub/telephony/sounds/asterisk-moh-freeplay-ulaw.tar.gz |
| 16:56.34 | fenrus | download it |
| 16:56.35 | azrtyo | i can't find this package |
| 16:56.55 | azrtyo | not Not Found |
| 16:58.36 | drmessano | freeplay MOH no longer exists |
| 16:58.40 | dlynes | Krolik13: try http://freshmeat.net/search?q=asterisk+cdr&submit=Search, and modify any of those to suit your needs...just tell it to filter out recordings, instead of calls |
| 16:58.58 | azrtyo | so which one replace that one ? |
| 16:59.01 | drmessano | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-moh-opsound-ulaw-current.tar.gz |
| 16:59.16 | azrtyo | ok thanks drmessano fenrus |
| 17:10.38 | azrtyo | i try to compile sox-14.0.0 at the end i got this error |
| 17:10.53 | azrtyo | make[1]: Leaving directory `/usr/src/sox-14.3.1/src' |
| 17:10.54 | azrtyo | make: *** [all-recursive] Error 1 |
| 17:11.02 | azrtyo | what i have to do exactly ? |
| 17:12.35 | jblack | that's not the error. The real error is before that. Regardless, this is the wrong place. |
| 17:14.08 | x-demon | asterisk gui (2.0 |
| 17:14.27 | x-demon | asterisk gui (2.0 branch) throws me error that i does not have needed privileges |
| 17:15.21 | x-demon | The GUI does not have necessary privileges. Please check the manager permissions for the user ! |
| 17:15.24 | x-demon | that |
| 17:15.40 | x-demon | but i have privileges for it, as said in make checkconfig... |
| 17:16.22 | tzafrir | what user is asterisk running as? |
| 17:16.49 | *** join/#asterisk mloe (~whocrz@217.79.80.213) |
| 17:16.52 | mloe | Hello, guys, I saw one article "Asterisk auto-dial out" about auto-dialing central but it explains how to make one-by-one .call files |
| 17:17.33 | mloe | is there any way to make 1 call file with the rules and then call all the numbers for numbers.txt with the call file's rules? |
| 17:17.36 | mloe | or something.. |
| 17:18.05 | mloe | *from numbers.txt |
| 17:18.13 | x-demon | tzafrir, asterisk |
| 17:18.33 | x-demon | gui has been build as root... |
| 17:18.34 | tzafrir | x-demon, does it have write permissions to /etc/asterisk and files there? |
| 17:18.45 | x-demon | tzafrir, yes |
| 17:19.07 | x-demon | oh wait... |
| 17:20.50 | x-demon | well, looks like it loading now... |
| 17:21.03 | x-demon | is that only first time it reload page ~20 times? |
| 17:22.18 | x-demon | well it stopped at parsing config files |
| 17:22.24 | x-demon | can tabulations be a problem? |
| 17:25.57 | x-demon | it actually parses all files |
| 17:26.00 | x-demon | i can see in logs |
| 17:26.13 | x-demon | but then stops at "Parsing config files" message |
| 17:30.34 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:39.02 | pabelanger-lap | mloe: possible yes, using local channels, but I doubt this would work in your case. Since you are looking to autodial. |
| 17:39.27 | nalbaxxx | i try to change asterisk directory record, i have try to use MixMonitor ( exten => _0[1-7]XX,n,MixMonitor(/var/www/taranis/media/conference-${CALLERID(num)}.wav), but that doesnt, someone have an idea ? ( sorry for my english ) |
| 17:41.09 | nalbaxxx | +work |
| 17:41.50 | pabelanger-lap | nalbaxxx: core show application MixMonitor |
| 17:42.43 | pabelanger-lap | nalbaxxx: ${MIXMONITOR_FILENAME} |
| 17:44.53 | nalbaxxx | thx pabelanger |
| 17:56.17 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
| 18:04.36 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 18:10.27 | mloe | pabelanger-lap so there is no way? |
| 18:10.33 | mloe | I've created the VBS script |
| 18:10.45 | mloe | it's very easy to make .call files from .txt file automatically |
| 18:11.11 | mloe | for example: 700 numbers for 3 seconds :) |
| 18:11.42 | mloe | but I needed to import the database from .txt file directly from asterisk, not windows script for making the .call files.. |
| 18:14.37 | pabelanger-lap | mloe: For each call you plan to make, you need to tell asterisk to dial. So in your case, a 1 to 1 relation for your .call files. How you generate them is up to you |
| 18:19.48 | x-demon | so, is it possible to fix asterisk gui? |
| 18:20.13 | mloe | :) |
| 18:20.16 | pabelanger-lap | x-demon: what is broken? |
| 18:20.33 | x-demon | pabelanger-lap, it stops on message "Parsing config files" |
| 18:20.37 | mloe | so, I can't make 1 setting for all numbers.. |
| 18:20.57 | pabelanger-lap | mloe: no |
| 18:21.06 | mloe | okay, thanks |
| 18:21.15 | mloe | can I run asterisk on live-cd linux? |
| 18:21.27 | mloe | I mean USB stick |
| 18:21.29 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@icm218138-orange.orange.sk) |
| 18:21.49 | pabelanger-lap | mloe: yes |
| 18:22.01 | mloe | asterisknow will be awesome but it will be hard to put on usb stick maybe.. |
| 18:22.26 | mloe | actually, for only dialing machine I don't need asterisknow.. |
| 18:22.35 | mloe | cool, I'll try now :) |
| 18:22.41 | mloe | thanks again |
| 18:23.12 | x-demon | is there any web gui that works directly with config files? |
| 18:23.30 | x-demon | btw, i need only peers/registers monitor and CDR reports |
| 18:23.32 | mloe | AsteriskGUI? |
| 18:23.54 | x-demon | mloe, not working for me :( |
| 18:24.06 | mloe | :| |
| 18:24.10 | x-demon | my guess is that because i have #include directives in configs |
| 18:24.13 | x-demon | in sip.conf |
| 18:25.19 | pabelanger-lap | x-demon: most likely. AsteriskGUI is currently unmaintained ATM |
| 18:25.27 | x-demon | too bad :( |
| 18:25.36 | x-demon | asterisk lacks good gui |
| 18:25.45 | x-demon | of course i can use configs... |
| 18:26.42 | pabelanger-lap | x-demon: your welcome to submit patches to AsteriskGUI; even become the maintainer |
| 18:26.58 | x-demon | pabelanger-lap, i'm not a developer, btw |
| 18:27.19 | x-demon | i know how it works, but not more |
| 18:27.25 | x-demon | i can code only in sh :D |
| 18:31.22 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:31.31 | exelnet | hi, is it possible to redirect an incoming sip call to another server? server1 owns the domain but server2 got asterisk running. so i want all calls on server1@domain to get redirected to server2. |
| 18:32.30 | *** join/#asterisk d4rkstar (~bruno@93-41-29-219.ip79.fastwebnet.it) |
| 18:34.54 | exelnet | ah ok this might help me: http://www.google.de/search?hl=de&q=sip+redirect+server&aq=f&aqi=&aql=&oq=&gs_rfai= |
| 18:39.54 | *** join/#asterisk AlHafoudh__ (~AlHafoudh@78.141.102.211) |
| 18:51.53 | dlynes | Does asterisk have an ability to specify a voicemail template from one user to the next? |
| 18:52.55 | dlynes | Or is 'emailbody=' and 'emailsubject=' global only parameters? |
| 18:53.12 | Mhaddog | been unable to access the web manager... |
| 18:57.45 | dlynes | I was kinda hoping they'd changed to be available on a per-user basis, but I guess not...thanks anyways |
| 19:11.10 | *** join/#asterisk bio-tty (~c@35.99.189.109.customer.cdi.no) |
| 19:12.24 | lukehasnoname | Does anyone know where an updated "Practical Asterisk for 1.4 AND 1.6" is online? The site advertised in the book is only 1.4 and isn't synced with the book at all |
| 19:18.12 | *** join/#asterisk lukehasnoname (~lukehasno@cpe-66-69-74-19.satx.res.rr.com) |
| 19:29.48 | *** part/#asterisk PuraVida (~PuraVida@S0106001e52f5dd45.vc.shawcable.net) |
| 19:32.04 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 19:32.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:49.36 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 19:52.48 | *** join/#asterisk nextgens (~florent@freenet/developer/nextgens) |
| 19:53.08 | nextgens | hi |
| 19:54.22 | nextgens | I'm looking for a SIP handset (DECT or WIFI, possibly the former) which has a decent support of SIP extensions and that I could connect to an asterisk server |
| 19:54.37 | nextgens | does anyone have a recommendation for a specific model? |
| 19:54.45 | nextgens | I'm in the UK |
| 19:54.55 | nextgens | oh, and obviously, cheap is better |
| 19:55.28 | Chainsaw | nextgens: Siemens C450 & friends. |
| 19:55.31 | WIMPy | Whatever you do, go for DECT. |
| 19:56.23 | nextgens | Chainsaw> these have their own ipbx, don't they? |
| 19:56.29 | WIMPy | Oh and I'd also avoid DECT bases with SIP, better go ISDN or POTS. |
| 19:56.37 | nextgens | I think dad has one |
| 19:56.50 | Chainsaw | nextgens: They speak SIP, so they'll deal with whatever PBX you prefer. Even Asterisk. |
| 19:58.49 | nextgens | Chainsaw> but does the handset talk sip or is it the base? |
| 19:59.10 | nextgens | if I have more than a handset, can I call one or another separately? |
| 19:59.28 | Chainsaw | nextgens: The base, yes. |
| 19:59.37 | Chainsaw | nextgens: Because you want the handset on DECT, not WiFi. Trust me on this. |
| 20:00.23 | nextgens | well, if only for battery life, yes |
| 20:00.35 | nextgens | but ultimately I'd like the handset to talk sip, not the base |
| 20:00.50 | nextgens | is any vendor doing SIP-over-DECT or something? |
| 20:00.56 | nextgens | or IP-over-DECT ? |
| 20:01.00 | Chainsaw | nextgens: Not that I'm aware of, no. |
| 20:01.06 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 20:01.17 | Chainsaw | nextgens: But DECT base stations will co-exist quite happily. Just have pairs. |
| 20:01.43 | WIMPy | You could run around with an old laptop wit a PCMCIA-III DECT card and use that as handset. |
| 20:02.13 | Chainsaw | nextgens: Or be defiant, buy an overpriced SIP-over-WiFi handset and then cry me a river when the battery life sucks and the intermittent latency near the edge of the coverage gives all your callers robot voices. |
| 20:03.47 | nextgens | nah, at this stage I'm convinced that I don't want WiFi |
| 20:04.06 | nextgens | unless it can do both |
| 20:04.24 | WIMPy | Good choice. |
| 20:04.29 | WIMPy | Not the handset. |
| 20:07.13 | Chainsaw | nextgens: Individually addressable extensions will require a DECT base per handset, at least on the Siemens chagall platform. |
| 20:07.46 | Chainsaw | nextgens: Unfortunately I don't know of other DECT SIP-capable platforms. Cisco opted for SIP-over-WiFi, as did Polycom. |
| 20:08.39 | WIMPy | The Simes things support 5 sip accounts, but I found the one I tried rather buggy. |
| 20:09.00 | Chainsaw | WIMPy: Sure, but you can't assign specific accounts to specific handsets. |
| 20:09.19 | WIMPy | On the one I tried, you could. |
| 20:09.27 | nextgens | from what I read the S450IP is the one I should get |
| 20:09.34 | nextgens | but then it's not available anymore |
| 20:09.40 | p3nguin | ~a580 |
| 20:09.41 | infobot | [~A580] The Siemens Gigaset A580 IP with ECO DECT technology is multi-line so you are free to register up to 6 handsets for 6 SIP accounts from different providers and make up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call. See http://gigaset.com/hq/en/product/GIGASETA580IP.html for details. Cost is about $70USD for one handset with base. Extra A58H handsets are around $40USD each. |
| 20:09.48 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 20:09.57 | [sr] | hi WIMPy |
| 20:09.58 | nextgens | so there is A580 IP and C475 IP now |
| 20:10.03 | WIMPy | Hmm. Maybe it was 6, not 5. |
| 20:10.15 | WIMPy | Hi [sr] |
| 20:10.32 | [sr] | WIMPy: lcr+*+freepbx working :) |
| 20:10.54 | fenrus | lol, i thaught you were talking about AS5800 and C7 over IP |
| 20:11.04 | fenrus | *goes back to sleep* |
| 20:11.48 | WIMPy | LOL |
| 20:12.07 | Chainsaw | It must be a 500-series thing, because I have a C475 here and I see no way to assign SIP accounts to handsets. |
| 20:13.13 | nextgens | okay, so the consensus seems to be for the a580 |
| 20:13.16 | nextgens | banco. |
| 20:13.22 | WIMPy | That's still on my to-do-list, to find out if old NAS could be used as hardware interfaces. |
| 20:31.18 | nextgens | thanks for your time :) |
| 20:31.19 | *** part/#asterisk nextgens (~florent@freenet/developer/nextgens) |
| 20:41.53 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:55.23 | *** join/#asterisk ELSEGO (~Jc@unaffiliated/elsego) |
| 20:55.28 | *** part/#asterisk ELSEGO (~Jc@unaffiliated/elsego) |
| 21:00.17 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:13.17 | *** join/#asterisk DelphiWorld (~Delphi@41.200.20.56) |
| 21:13.19 | DelphiWorld | hi all |
| 21:13.29 | DelphiWorld | where i enter stun server in grandstream gxp phones? |
| 21:16.50 | *** part/#asterisk DelphiWorld (~Delphi@41.200.20.56) |
| 21:39.30 | *** join/#asterisk ShKoDrAnI (~ardit24@cpe-95-107-197-219.wifi.tring.al) |
| 21:39.42 | *** part/#asterisk ShKoDrAnI (~ardit24@cpe-95-107-197-219.wifi.tring.al) |
| 21:51.53 | *** join/#asterisk grinder13 (~grinder@cpc2-sgyl2-0-0-cust1286.sgyl.cable.virginmedia.com) |
| 21:57.18 | carrar | Happy Fathers Days to those of us who are Fathers! |
| 21:58.07 | [sr] | only now? |
| 21:58.09 | jblack | even the disowned ones? |
| 21:58.21 | [sr] | it was about a month ago here |
| 21:58.30 | jblack | It's today in the US |
| 21:58.35 | [sr] | ah |
| 22:01.42 | carrar | yeah |
| 22:03.13 | [sr] | i see |
| 22:03.13 | [sr] | nice |
| 22:04.09 | [sr] | going to sleep |
| 22:04.10 | [sr] | see ya |
| 22:07.44 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 22:09.14 | E-bola | I use setvar in my sip.conf to specify which outgoing number to use for each sip account. I would like to set another variable for each local sip device as well, that defines which logical group function it is a part of (reception, sales etc) But since i need it for incoming calls im not quite sure i can do that. So i guess my question is if there is way to query the var for a given sip account? |
| 22:14.34 | [TK]D-Fender | E-bola: Don't believe you can poll that |
| 22:15.54 | E-bola | [TK]D-Fender: Have'nt found a way yet, so your prolly right |
| 22:16.02 | E-bola | I started making globals vars instaid |
| 22:16.12 | E-bola | but its a bit tiresome, wish i could find some smart way of handling this |
| 22:17.53 | E-bola | i think ile just make it all manual, its only 25 phones afterall |
| 22:21.02 | [TK]D-Fender | AstDB <- |
| 22:22.19 | E-bola | I could use that as well, but as far as i know its supposed to be called from dialplan or outside scripts |
| 22:22.27 | E-bola | which doesnt make it optimal for what i'd like |
| 22:22.37 | E-bola | I need to query properties of a sip account defined in sip.conf |
| 22:23.09 | E-bola | if i have to set the variables somewhere else, it defeats the purpose of what im trying to do. which is to only have to keep track of the property 1 place |
| 22:25.39 | *** join/#asterisk m_tadeu (~quassel@89-180-200-187.net.novis.pt) |
| 22:30.35 | [TK]D-Fender | E-bola: imperfect world |
| 22:31.33 | E-bola | indeed, like for example like how the snom 320's doesnt seem to want to auto update to new firmware automatically anymore :( |
| 22:32.37 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 22:43.02 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 22:57.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:58.22 | Trixboxer | Hi, Im trying to install chan_gtalk in asterisk 1.4, but it is not available in "make menuselect" |
| 22:59.09 | Trixboxer | Is there any way to install chan_gtalk in1.4 ? |
| 23:01.44 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:02.57 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:06.05 | [TK]D-Fender | Trixboxer: do you SEE it in menuselect? |
| 23:06.37 | Trixboxer | yes but with "XXX".. Im using 1.4.9 |
| 23:11.49 | Trixboxer | [TK]D-Fender, Am I doing it wrong ? |
| 23:14.34 | [TK]D-Fender | Trixboxer: you are clearly missing a pre-requisite. LOOK at the screen.... |
| 23:15.38 | Trixboxer | It "Depends on: iksemel(E), res_jabber(M)" |
| 23:15.50 | Trixboxer | I have iksemel |
| 23:16.25 | Trixboxer | Figuring out how to install res_jabber .. |
| 23:17.08 | p3nguin | Is it really that difficult? |
| 23:18.26 | *** join/#asterisk Middleman (~Dystopia@dsl-173-206-241-120.tor.primus.ca) |
| 23:18.33 | p3nguin | I should check into the gtalk module. I already use jabber, so it shouldn't be too much work. |
| 23:18.52 | Middleman | hey anyone have an isue with apache, after installing asterisknow 1.7 with 1.4 gui |
| 23:19.23 | Middleman | i ran the service cmd to start it, its not evn there |
| 23:19.37 | Middleman | www was loading b4 reboot |
| 23:21.40 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 23:22.48 | [TK]D-Fender | wonders what "1.4 GUI" even means |
| 23:24.23 | path | :] |
| 23:24.44 | p3nguin | asterisk-gui 1.4? |
| 23:29.07 | *** join/#asterisk MiserySoft (~Lee@109.180.106.90) |
| 23:30.04 | [TK]D-Fender | p3nguin: Never possessed such a version # |
| 23:30.19 | [TK]D-Fender | p3nguin: and it doesn't follow *'s #'s |
| 23:30.50 | p3nguin | It was merely conjecture. |
| 23:31.50 | p3nguin | Wow, a new configuration menu... |
| 23:31.51 | p3nguin | When did we get this? |
| 23:33.20 | [TK]D-Fender | p3nguin: My guess.. AsteriskNOW 1.7 |
| 23:33.45 | *** part/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 23:34.04 | p3nguin | I'm referring to the "make menuselect" menu in 1.4.33. |
| 23:34.04 | Trixboxer | p3nguin, I have install iksemel from yum and also from source, even gnutls is there, I have done "make clean" but still "make menuselect" says XXX for chan_gtalk |
| 23:34.21 | Trixboxer | but somehow res_jabber is not showing up.. |
| 23:34.37 | p3nguin | [*] chan_gtalk |
| 23:34.56 | p3nguin | It's there for me, so I'm not sure what might be wrong with yours. |
| 23:35.08 | Trixboxer | XXX 4. chan_gtalk |
| 23:35.31 | Trixboxer | are you using same 1.4 ? |
| 23:35.33 | p3nguin | You're on the channel drivers section? |
| 23:35.39 | Trixboxer | sorry 1.4.9 |
| 23:35.41 | Trixboxer | yes |
| 23:35.49 | p3nguin | <p3nguin> I'm referring to the "make menuselect" menu in 1.4.33. <--- 1.4.33 |
| 23:35.58 | p3nguin | I don't do antiques. |
| 23:36.21 | WIMPy | p3nguin: You installed newt because of dahdi_tool? |
| 23:36.33 | p3nguin | I don't know what newt is. |
| 23:36.54 | Trixboxer | trying on that |
| 23:36.54 | WIMPy | Some alternative for ncurses. |
| 23:36.58 | p3nguin | Actually, I remember seeing that name before, but I'm actually familiar with it. |
| 23:37.09 | WIMPy | If you got it, you get another menu. |
| 23:37.21 | p3nguin | I probably have it installed. I'll check in a minute. |
| 23:38.18 | *** join/#asterisk bjhaid (~IceChat7@41.206.15.1) |
| 23:39.06 | Trixboxer | Hey, I got [*] chan_gtalk in 1.4.33 |
| 23:39.25 | p3nguin | nod |
| 23:39.57 | Trixboxer | thanks for "1.4.33" |
| 23:40.58 | p3nguin | Hmm, I've actually had libnewt installed since Oct 10, 2009. |
| 23:41.24 | p3nguin | and I've upgraded at least three versions of asterisk since then. |
| 23:42.11 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 23:42.39 | p3nguin | Ah well. It's not all that important, I suppose. |
| 23:44.14 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 23:44.23 | bjhaid | i am a beginner, i would like to know if my RJ 11 is a FXS or FXO port? |
| 23:45.01 | p3nguin | Where is the RJ-11? |
| 23:45.07 | p3nguin | In a phone, in a wall? |
| 23:45.26 | p3nguin | Does it connect to the phone company? Do you plug a phone into it? |
| 23:46.10 | bjhaid | the RJ 11 on my notebook |
| 23:46.20 | p3nguin | a modem jack? |
| 23:46.49 | bjhaid | yes, wondering if it could be a FXS or FXO port too |
| 23:47.25 | p3nguin | Technically, it might be an FXO. There's not much chance of turning it into an FXS. |
| 23:48.02 | bjhaid | okay, so i can use it as an FXO port, how would i know if it is a FXO port? |
| 23:49.34 | p3nguin | Does it connect to a provider's jack to connect you to the phone lines so you can dial out? |
| 23:50.32 | p3nguin | ~fxo |
| 23:50.33 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 23:50.43 | p3nguin | ~fxs |
| 23:50.44 | infobot | [fxs] foreign exchange station - type of port you need to connect an analog device (phone, fax machine) to a pbx. This is the type of port found in your wall jack. |
| 23:50.49 | pabelanger-lap | bjhaid: not going to work without a specific channel driver |
| 23:50.55 | bjhaid | i have not tried connecting it to a provider's jack yet, just wanting to know if i could connect it and if it would work |
| 23:51.34 | bjhaid | pabelanger-lap: how abt the dahdi drivers would they not work for it? |
| 23:51.35 | p3nguin | This seems futile, so I'm done for now. |
| 23:52.22 | pabelanger-lap | bjhaid: no, not going to work. |
| 23:52.39 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 23:52.45 | pabelanger-lap | bjhaid: modem != fxs/fxo\ |
| 23:53.02 | bjhaid | pabelanger-lap: thanks |