| 00:00.28 | Naikrovek | fogrider: you've got some disk errors, if you've a backup, restore from it. if not, use freepbx in asterisknow to create a freepbx backup, then reinstall, and restore |
| 00:01.05 | Naikrovek | fogrider: #asterisknow can help you far more than us |
| 00:01.15 | Naikrovek | people from digium lurk in there |
| 00:01.22 | Naikrovek | you'll be much better off posing your problem in there |
| 00:01.54 | fogrider | Naikrovek: i've been in there since i joined here... no response. |
| 00:02.27 | p3nguin | Free support isn't always instantaneous. |
| 00:02.27 | fogrider | good idea about the backup. but i did already install a license from digim for g729 that i don't wanna lose. |
| 00:02.29 | Naikrovek | it's slow, but patience will serve you well |
| 00:02.47 | p3nguin | You won't lose your license. |
| 00:03.01 | Naikrovek | they're host based and you'll be fine |
| 00:03.06 | Naikrovek | you can always call digium and explain |
| 00:03.10 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 00:03.27 | p3nguin | The license allows you to run X number of channels. |
| 00:03.57 | Naikrovek | each license allows you to run 1 |
| 00:04.03 | Naikrovek | unless that's changed recently |
| 00:04.19 | p3nguin | What if you have one license for 42 channels? |
| 00:04.42 | *** part/#asterisk my007ms (~nnnnnnnnn@196.219.63.12) |
| 00:04.47 | ChannelZ | you're playing semantics between 'license' and 'license file' |
| 00:11.46 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 00:12.45 | *** join/#asterisk jppialasse (~jppialass@64.235.198.36) |
| 00:12.53 | doctorray | Not to beat a dead horse, but I can't find info re: 1.6.2, my phone hints are "idle" and never changing. is this the whole call-limit thing, or is there something else I should be looking for? |
| 00:18.50 | *** join/#asterisk diegomad (mad@186.81.139.114) |
| 00:20.14 | jaytee | http://tinyurl.com/yb3x3y9 |
| 00:20.15 | jppialasse | hello |
| 00:20.39 | jppialasse | is there a way to force hangup during a conversation after 1 hour ? |
| 00:21.05 | *** join/#asterisk coppice (~chatzilla@93.176.64.202.dyn.pacific.net.hk) |
| 00:22.17 | jaytee | jppialasse, you can set Timeout(absolute) |
| 00:22.38 | doctorray | jaytee: nice |
| 00:22.39 | p3nguin | doctorray: It was solved earlier by using counteronpeers rather than the invalid limitonpeers. |
| 00:22.43 | WIMPy | doctorray: Probably yes. Take a look at the UPGRADE-1.6 file for changed keywords. |
| 00:24.26 | *** join/#asterisk bmoraca (bmoraca@66.242.162.254) |
| 00:24.35 | hardwire | sighs |
| 00:25.17 | bmoraca | echos |
| 00:25.36 | hardwire | public sip phone dials public asterisk server.. public asterisk server dials other public asterisk server.. answer by other public asterisk server.. codec fail |
| 00:25.36 | bmoraca | sitting in the datacenter waiting for a system to fail is not fun |
| 00:25.55 | carrar | bmoraca, thats why they have monitoring tools! |
| 00:26.15 | carrar | or perhaps you are the tool |
| 00:26.18 | bmoraca | carrar: i'm waiting for it to fail so i can swap out the hardware if need be...DC is 45 minutes from my house |
| 00:26.35 | bmoraca | it's already failed twice and i've swapped out two components...waiting to see if it does it again |
| 00:27.00 | carrar | Why wait |
| 00:27.08 | carrar | do it when it's not critical |
| 00:27.23 | bmoraca | carrar: because i think replacing the power regulator unit fixed the issue |
| 00:27.41 | bmoraca | and it's always critical :P |
| 00:28.01 | hardwire | I can force the codecs.. but I can't let the sip phone and the final destination hash out which to use it seems |
| 00:28.52 | bmoraca | hardwire: if all codecs are available to all devices, there shouldn't be a problem. and if the media path doesn't include the first asterisk box, then there really shouldn't be a problem |
| 00:29.02 | hardwire | alas.. there is |
| 00:29.11 | hardwire | all points support g729/g711 |
| 00:29.19 | bmoraca | weird |
| 00:29.33 | hardwire | ends up with a mixed g711/g729 session |
| 00:29.37 | hardwire | and of course.. that don't work |
| 00:29.56 | bmoraca | sounds like a priority issue |
| 00:30.02 | bmoraca | codec priority, that is |
| 00:30.08 | hardwire | I really dislike priority issues |
| 00:30.39 | bmoraca | don't think you can influence which codecs asterisk chooses, though...at least, that's what i recall from the last discussion about it |
| 00:30.41 | hardwire | esp when I want to walk a call from server to server to provider |
| 00:31.01 | hardwire | SIP_CODEC is working fine for helping me test what the heck is going on |
| 00:31.16 | hardwire | BYE! |
| 00:32.05 | jppialasse | jaytee> thanks |
| 00:33.05 | jaytee | jppialasse, you're welcome |
| 00:34.00 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:34.30 | fogrider | ok... getting some info from sip debug ...... Really destroying SIP dialog '10 bah bah@127.0.0.1' Method : REGISTER |
| 00:34.36 | fogrider | Any ideas what that means? |
| 00:36.22 | fogrider | i get that message with a different bahbah number about every 30 seconds. i receive nothing when another phone tried to connect to asterisk server |
| 00:37.53 | *** join/#asterisk rservers (~brian@unaffiliated/rservers) |
| 00:39.41 | fogrider | that is the only message i receive from sip debug set |
| 00:39.46 | hardwire | meh.. multi-hop codec negotiation of doom |
| 00:39.57 | fogrider | anywhere else i can look to clues to what is going on with this thing? |
| 00:40.36 | hardwire | turn verbose down |
| 00:41.38 | fogrider | i got it set to verbose 9.... |
| 00:42.18 | fogrider | ok, the bab bah number is actually a set of two different bah bah numbers. one after another, just continues to repeat. |
| 00:44.10 | *** join/#asterisk blaines (~blaines@120.sub-75-242-114.myvzw.com) |
| 00:44.44 | fogrider | 107945af1abb56c91e3b2f311c46e77e@127.0.0.1 and 097022666e419fc33d1124cf4e38563c@127.0.0.1 which ofcourse 127.0.0.1 is the same machine's loopback IP. but the question is what the heck is it registering for and over and over. |
| 00:45.16 | blaines | can someone recommend a place to purchase asterisk compatible phones |
| 00:45.22 | blaines | and maybe a good asterisk phone |
| 00:46.04 | p3nguin | Many people prefer to use Polycom phones with Asterisk. They get them from various places. |
| 00:47.19 | blaines | p3nguin: cool - amazon.com? |
| 00:47.29 | blaines | or cdw i guess too |
| 00:47.56 | p3nguin | telephonydepot.com |
| 00:48.18 | p3nguin | voipsupply.com |
| 00:48.22 | leifmadsen | www.e4voip.com |
| 00:48.29 | leifmadsen | I love their service, prices, and delivery times |
| 00:48.42 | p3nguin | You heard it here, first! |
| 00:48.49 | leifmadsen | :D |
| 00:52.36 | fogrider | hey, when i'm at the command line and i type sip reload, and hit enter. and i don't get no respone from the CLI, what does that mean? |
| 00:53.39 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:54.25 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 00:55.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:55.44 | fogrider | or if i do an "stop now", and then try and run asterisk and get a message "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?" and no files exist in the /var/run/asterisk folder.. what do ya do? |
| 00:57.32 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 00:59.37 | leifmadsen | fogrider: sounds like a deadlock (in reference to your first question) |
| 01:00.07 | leifmadsen | fogrider: the answer to the second question is, "you did a stop now... asterisk stopped... so you can't connect to the console" |
| 01:00.23 | leifmadsen | runs off to watch Lost. |
| 01:00.31 | fogrider | hey dude... |
| 01:01.05 | blaines | thanks for the recommendations |
| 01:01.16 | fogrider | i believe that asterisk is running under the wrong user...how do i check that? |
| 01:03.56 | jaytee | nice dining table! |
| 01:04.37 | *** join/#asterisk Magicblaze0071 (~sony@fl-67-235-198-191.dhcp.embarqhsd.net) |
| 01:06.28 | beefpastry | anybody here have any dialplan experience with freepbx (#freepbx is dead) |
| 01:07.50 | Slugs_ | fogrider, ps aux | grep asterisk |
| 01:11.07 | *** join/#asterisk Magicblaze0071 (~sony@fl-67-235-198-191.dhcp.embarqhsd.net) |
| 01:12.47 | *** join/#asterisk khratos (~khratos@190.166.134.1) |
| 01:14.27 | *** join/#asterisk jsjc (~chatzilla@115.131.205.97) |
| 01:15.21 | jsjc | i have been reading the asterisk queues tutorial from orderlyq that is very understandable. I just wonder how will be possible to mix DAHDI users with SIP? any where I coul dget some info on this? |
| 01:17.01 | Naikrovek | jsjc: endpoints are endpoints as far as i know. try it and see if it works. if not, then come in here and ask why. |
| 01:17.17 | leifmadsen | jaytee: thanks! :) |
| 01:17.54 | jsjc | Naikrovek: for example this macro agent-dial has this ext exten => dial,1,Dial(SIP/${DB(AGENT/${ARG1}/PHONE)},30,rotwh) that means that will call trough sip |
| 01:18.04 | Naikrovek | leifmadsen: you overpiad for yer phone if you bought it from that place, looks like |
| 01:18.08 | jsjc | so depending on extension I should do a goto if |
| 01:18.29 | jsjc | 1XXDAHDI 2XX SIP got to each depending on ext that will work? |
| 01:18.33 | jsjc | theroretically yes? |
| 01:18.42 | Naikrovek | exten => dial,1,Dial(SIP/whatever&whatever&whatever) |
| 01:19.02 | *** join/#asterisk CRCinAU (~CRCinAU@irc.crc.id.au) |
| 01:19.14 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 01:19.32 | Naikrovek | i don't know the DAHDI syntax for extensions but just Dial() them all at once, separated by & |
| 01:19.38 | Naikrovek | oh you don't know ahead of time if it's sip or DAHDI |
| 01:19.40 | Naikrovek | i see |
| 01:19.44 | Naikrovek | well i'm sure there's a way |
| 01:19.48 | Naikrovek | i just dunno what it is |
| 01:20.32 | jsjc | ok will check... |
| 01:21.02 | *** part/#asterisk CRCinAU (~CRCinAU@irc.crc.id.au) |
| 01:21.11 | *** join/#asterisk CRCinAU_ (~CRCinAU@irc.crc.id.au) |
| 01:21.24 | CRCinAU_ | so. |
| 01:21.40 | CRCinAU_ | asterisk + xen on a 250hz kernel timer.... |
| 01:21.41 | *** join/#asterisk blaines (~blaines@44.sub-75-242-150.myvzw.com) |
| 01:21.53 | CRCinAU_ | is it possible without thing failing really badly? :\ |
| 01:23.17 | *** part/#asterisk CRCinAU (~CRCinAU@irc.crc.id.au) |
| 01:23.27 | *** join/#asterisk CRCinAU_ (~CRCinAU@irc.crc.id.au) |
| 01:23.38 | CRCinAU_ | and why can't I do anything as my normal nick? :| |
| 01:25.23 | CRCinAU_ | I tried running 1.6.2.6 in an asterisk paravirt machine and things were - well, woeful. |
| 01:26.08 | CRCinAU_ | faxing died completely (maybe spandsp issue?), voicemail did weird things and paused at random times |
| 01:26.19 | CRCinAU_ | even if it was the only vm running at the time :| |
| 01:35.07 | *** join/#asterisk Magicblaze0071 (~sony@fl-67-235-198-191.dhcp.embarqhsd.net) |
| 01:39.46 | fogrider | slugs: i run ps aux | grep asterisk i see where root is runing /bin/sh /usr/sbin/safe asterisk -U asterisk -G asterisk and then i see asterisk is running /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 01:40.12 | fogrider | or whoever is listening ..... |
| 01:47.14 | ChannelZ | do what now? |
| 01:49.05 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:50.11 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 01:50.44 | fogrider | well, asterisk is broke. computer rebooted and now i can't get any sip connections. that's what i see when asterisk is running when i do ps aux | grep asterisk. i turn on sip set debug, and i only get these messages, Really destroying SIP dialog '107945af1abb56c91e3b2f311c46e77e@127.0.0.1' Method: REGISTER and 097022666e419fc33d1124cf4e38563c@127.0.0.1 .... over and over again. |
| 01:51.21 | *** join/#asterisk blaines (~blaines@17.sub-75-245-206.myvzw.com) |
| 01:52.10 | ChannelZ | your phones are registering to localhost? |
| 01:52.35 | fogrider | nope... don't have any phones on the localhost... that's one issue |
| 01:53.08 | fogrider | i have multiple other phones that are on same network, but after the computer got rebooted due to power,SIP doesn't work anymore. |
| 01:54.04 | ChannelZ | well I'd say you have some crazy-ass network issues occuring.. or you put a register => line in sip.conf to yourself.. |
| 01:54.24 | fogrider | i don't remember ever seeing that message before the problem happened. I'm just tring to get everything back working the way it was. |
| 01:54.54 | fogrider | everytning was working fine, power went out, computer came back on, no more workie. I didn't add any code or edit any file. |
| 01:54.55 | ChannelZ | Have you turned on SIP debug and looked at what is going on, what user agent these are coming from |
| 01:55.18 | fogrider | yes, those are the only messages that come when i turn on sip debug. |
| 01:55.36 | ChannelZ | That isn't what I wasked |
| 01:55.40 | ChannelZ | asked |
| 01:55.42 | fogrider | only two messages over and over again, they 30 secnods i get another one |
| 01:55.50 | fogrider | ok... sorry. |
| 01:55.55 | fogrider | what you want me to do? |
| 01:56.05 | ChannelZ | What user agent are they coming from? |
| 01:56.09 | ChannelZ | Pastebin some |
| 01:56.39 | fogrider | where do i see what user agents they are coming from? |
| 01:57.19 | ChannelZ | in the headers of the sip messages |
| 01:57.50 | ChannelZ | gotta go, almost time for LOST |
| 01:58.31 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 01:58.44 | fogrider | that message starting with "Really" is the only message i get |
| 02:07.24 | *** join/#asterisk blaines (~blaines@48.sub-75-245-87.myvzw.com) |
| 02:15.33 | fogrider | wow i fixed it.... that's crazy . i can't belive that is what broke it. unbelieveable. i finally tracked it down thou.... crazy crazy |
| 02:19.06 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 02:21.53 | jaytee | fogrider, what was it? |
| 02:26.36 | jsjc | is there any possibility of changing the greeting of voicemail from shell? |
| 02:28.42 | fogrider | jaytee: let's just say i learned something important about asterisk.... if your dns fails then sip fails. |
| 02:29.44 | jaytee | ah! I see |
| 02:29.44 | mutax | :] |
| 02:31.29 | fogrider | damn, i guess the power going out and internet down at same time, i believed it to be the power shutting off causing a shutdown that caused problem. but it was actually the extended time without DNS resolving ;( |
| 02:34.52 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:37.09 | *** join/#asterisk crycrow (~Victor@195.16.79.4) |
| 02:38.32 | crycrow | hi |
| 02:41.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:41.54 | *** part/#asterisk crycrow (~Victor@195.16.79.4) |
| 02:45.20 | Naikrovek | hi |
| 02:48.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
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| 03:27.57 | spenguin[work] | if I make changes to the mysql db where all the data regarding agents etc, do I need to reload asterisk? |
| 03:28.02 | spenguin[work] | or any specific module? |
| 03:32.59 | *** part/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 03:39.09 | ChannelZ | I think that's kind of the point of 'realtime config' |
| 03:39.51 | ChannelZ | though a module might be working off of stale info, like a call in progress |
| 03:42.40 | spenguin[work] | ah k ChannelZ |
| 03:51.54 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 03:58.34 | ChannelZ | ..but I don't run RT so hell if I know! |
| 03:58.56 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 04:00.16 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 04:07.02 | joobie | hey guys.. got a scenario i thought i'd bounce around to see what you guys think a good approach is here... |
| 04:07.13 | patrb | so im any reason why app_addon_sql_mysql isnt compiling (1.6.20)? I have mysql devel and asterisk both installed...no errors during compiling...it just seems to skip the app...anyone else experience this? |
| 04:08.21 | joobie | basically when a particular number rings, asterisk dials about 5 handsets around the offset at the same time.. anyone of those 5 handsets can collect the call.. the problem is after hours, 1 person is still in the office, just on a different phone.. it would be great if the person can collect the call if this is the case.. |
| 04:09.09 | patrb | joobie: you can detect the time in your dialplan and then send it to your alternate extension after x time |
| 04:09.21 | joobie | times vary |
| 04:09.28 | joobie | i think what i need is some way to notify on the phone |
| 04:09.34 | joobie | like a light flash |
| 04:09.38 | joobie | but not actually ring? |
| 04:09.58 | joobie | because the user doesnt want to hear his phone ring everytime a call comes through for the main number, only if it's specifically for his extension |
| 04:10.09 | joobie | but the user can also hear the other 5 phones ring.. so he knows when a calls coming through |
| 04:10.22 | joobie | so ideally if he can visually see the call coming through on the handset or sumthen and answer it? |
| 04:10.48 | joobie | im getting all 5 phones to ring at the moment by using Dial() with multiple sip extensions in there |
| 04:10.54 | patrb | not sure how to help...id get a firm time that youre sure everyone else is gone by |
| 04:11.00 | patrb | and tell the user to deal with it until that time |
| 04:11.23 | patrb | ...or id write a php script w/ a button for the last user out to press to switch over the dial plan |
| 04:11.41 | patrb | but good luck training your users to use that script lol |
| 04:11.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:12.45 | joobie | patrb, the issue is say all 5 users are out at lunch |
| 04:12.48 | joobie | or say they finish a bit earlier |
| 04:12.58 | joobie | if the call comes through, this user should have the option to answer |
| 04:13.08 | joobie | even during the day fo rexample.. maybe 1 is in the toilet and 4 are out.. |
| 04:13.16 | joobie | variable scenarios iwth the time so a set schedule wont work in this cae |
| 04:13.17 | joobie | case |
| 04:22.16 | luke-jr | is anyone else getting random SIP calls that try to dial some random number and don't talk when you answer? :/ |
| 04:23.11 | luke-jr | stuff like 000442073479999 |
| 04:23.59 | *** join/#asterisk Jumpie (n3rdz@ip68-98-31-152.ph.ph.cox.net) |
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| 04:57.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:58.10 | ChannelZ | no but I don't have SIP open to random people. It's probably just script-kiddies prodding your system to see if they can exploit it |
| 04:59.34 | ChannelZ | joobie: you could use Pickup() |
| 05:02.28 | joobie | ChannelZ, so have the single user dial something on his phone then make that use pickup() to grab the call? |
| 05:06.10 | ChannelZ | yeah.. it's a little wierd depending on your setup, because you're picking up an "extension" as opposed to a device.. so it depends a little on how your dialplan is setup |
| 05:06.43 | ChannelZ | But basically I have a *7XXX extension, so I can dial *7123 and it will pickup extension 123 if it is ringing. |
| 05:07.50 | ChannelZ | But since an extension is a manifestation of the dialplan, it means someone had to have dialed extension 123 to make the phone ring in the first place |
| 05:09.39 | ChannelZ | So I'm not sure what happens if for instance your dialplan only consists of an 's' extension which answers the call, yells at the caller a little, and then Dial()s all your device which ring.. I guess you'd have to Pickup(s) which might or might not be bad (say 5 calls have come in on other lines that all go through that context/exten..) |
| 05:11.02 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 05:11.39 | ChannelZ | In my case the dialplan says some things and then if the person doesn't enter an extension specifically, I do a Goto(internal,200,1) which jumps to a new context/exten which then Dial()s a default device (call it the 'secretary'). So I can then Pickup(200) from that same context to snag the call. |
| 05:30.58 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:35.36 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 05:37.14 | Defraz | Hey all I am trying to make it so when I dial 7XXX it will dial out my zaptel card channel group 1 with a # and 3 digits that follow the 7. |
| 05:37.19 | Defraz | exten => 7XXX,1,Dial(g1/#${EXTEN:3}) is what I have so far. |
| 05:38.36 | Defraz | so if I dial 7120 it should send #120 out the zap channel |
| 05:38.46 | Defraz | But it only seems to send 7120 |
| 05:39.18 | kerframil | Defraz: no, because the pattern isn't prefixed with an underscore |
| 05:39.22 | ChannelZ | well I'm surprised it's dialing anything at all because ${EXTEN:3} means 'cut off the first 3 digits' |
| 05:39.53 | ChannelZ | and there's that too |
| 05:39.53 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:40.20 | ChannelZ | so to combine you want _7XXX and ${EXTEN:1} methinks |
| 05:44.19 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 05:44.28 | Defraz | okay I got it but it isn't going out the zap |
| 05:44.36 | Defraz | I changed the 3 to a 1 |
| 05:44.48 | Defraz | so it hacks the 7 off |
| 05:45.10 | Defraz | but it is not going out the right channel I think |
| 05:45.12 | Defraz | Executing [6102@from-internal:2] Dial("SIP/9000-b782d010", "g1/#102") in new stack |
| 05:45.19 | ChannelZ | Oh you want DAHDI/g1 not just g1 |
| 05:45.19 | Defraz | hmmm |
| 05:45.39 | Defraz | I am using the older zaptel drivers |
| 05:45.43 | spenguin[work] | does anyone know about "Security check failed" error in ekiga by any chance? |
| 05:45.44 | Defraz | so it would be zap/g1 |
| 05:45.48 | ChannelZ | ok then Zap/G1 |
| 05:47.08 | Defraz | okay got it |
| 05:47.11 | Defraz | works slick. |
| 05:47.35 | Defraz | I am playing with a dumb starcall system for a school. |
| 05:47.40 | Defraz | it is obnoxious |
| 05:54.56 | spenguin[work] | nvm, stupid ekiga bug |
| 05:58.07 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
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| 06:53.14 | kamh | hi |
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| 07:03.34 | kruemeltee | hello all together :-) |
| 07:05.52 | ChannelZ | at this hour, unlikely |
| 07:06.31 | kruemeltee | I have a grandstream SIP Telephone that has two Lines for calling ... If I have this phone as an agent within a queue the second line is calling if the agent is currently talking ... how am I able to prevent * for calling this phone on the second line? |
| 07:06.59 | *** join/#asterisk fiddur (~fiddur@192.121.104.72) |
| 07:07.05 | kruemeltee | excuse me for talking so lately ... but here it's 9 o'clock in the morning ;-) |
| 07:07.27 | kruemeltee | maybe there's another one ;-) |
| 07:07.35 | kruemeltee | just like you :-) |
| 07:08.32 | ChannelZ | Usually the phone has a 'call waiting' function you can turn off so it will reject the second call |
| 07:09.10 | ChannelZ | Specifically that phone, I have no idea where to tell you to look besides generically 'in its configs' |
| 07:11.34 | kruemeltee | I have a grandstream GXP1200 ... |
| 07:11.51 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:11.52 | kruemeltee | wait a moment, I'll check the configurator of the phone |
| 07:13.15 | kruemeltee | yea ... I found the "call waiting" function ... and it's enabled ... if disabling this one the second line will not be used anymore? |
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| 07:25.09 | ChannelZ | kruemeltee: presumably |
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| 09:25.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:25.48 | refsgaard | Hi all - Peter from solutors.com here. I got a call from a customer who after upgrade to the latest Asterisk it getting random disconnect after 8 min. And its always 8min but its random on who gets disconnectet. Any one have an idea? |
| 09:26.19 | refsgaard | They are dialing into a Codian MCU - (we work with Tandberg video conference and dont have any experience in Asterisk) |
| 09:26.39 | Chainsaw | refsgaard: It would be best to have your customer run: core set debug 10 & core set verbose 10 |
| 09:26.58 | Chainsaw | refsgaard: Before/during the disconnection, Asterisk events should be logged. We'll need to know these events to help you furhter. |
| 09:27.40 | refsgaard | i will get that :) Thx |
| 09:38.09 | *** join/#asterisk joobie (~joobie@CPE-124-180-152-53.lnse4.lon.bigpond.net.au) |
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| 09:42.36 | sprite-- | Are older asterisk versions archived anywhere? |
| 09:45.43 | sprite-- | nm found it |
| 09:50.11 | hc_e | Moin. question about 'dialplan reload'. Am I correct to assume that any channel in the 'UP' state during the reload is not affected by it? |
| 09:50.26 | hc_e | I.e., is the old dialplan kept until no channels are active in it anymore? |
| 09:54.33 | Dovid | hc_e: correct |
| 09:54.49 | Dovid | aprite--: all versions of asterisk are on asterisk.org |
| 09:55.33 | Dovid | aprite--: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 09:55.54 | hc_e | Dovid: ah good. Do you know what happens if i Goto another context and from there jump back to the old one, which has been reloaded in the meantime? |
| 09:56.18 | sprite-- | Dovid: Thanks. Found it already :) I had a chef recipe and was using another download folder where older versions were being removed/ |
| 09:57.36 | tzafrir_laptop | hc_e, hmm... I think this assumption is wrong |
| 09:57.42 | tzafrir_laptop | But it is easy to test |
| 09:57.57 | tzafrir_laptop | using a dialplan with Wait(60) |
| 09:58.26 | *** join/#asterisk kartik (~koolkarti@117.199.113.80) |
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| 10:02.45 | sprite-- | ugh wtf |
| 10:02.50 | sprite-- | It's not running my custom recipe |
| 10:03.01 | sprite-- | wrong channel |
| 10:03.02 | mutax | WIMPy: you nightmare is back ;o) |
| 10:04.20 | mutax | Hi everybody. I am trying the second day now to get lcr working. Whatever I do, LCR is not picking up the line. neither direct nor with asterisk. All I get is a 'queued hangup' by lcr. When using the lcr vbox I get the same cause as in asterisk: 101 - message in wrong state or sth. |
| 10:04.31 | mutax | Has somebody any clue what happens there? |
| 10:14.02 | *** join/#asterisk bbt (~Sam@dsl-220-253-76-42.NSW.netspace.net.au) |
| 10:14.54 | bbt | just did a yum update and noticed some asterisk packages were installed. upon retarting amportal i get this error: /usr/sbin/safe_asterisk: line 145: 3288 Segmentation fault |
| 10:15.43 | Chainsaw | bbt: Asterisk is not a great fan of restarting gracefully. |
| 10:15.59 | Chainsaw | bbt: Sometimes it just explodes instead. Try stopping it more forcefully and then restarting it. |
| 10:17.48 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.59) |
| 10:17.49 | bbt | heres the full pastbin: http://paste.org/pastebin/view/17357 |
| 10:18.06 | bbt | more forcefully in what way? |
| 10:19.18 | bbt | is there a pid i should be looking for? |
| 10:19.24 | mutax | bbt: /etc/init.d asterisk stop ; ...start |
| 10:20.26 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 10:20.40 | bbt | tried that, also service asterisk start/stop and im still getting the error |
| 10:22.23 | bbt | slightly different errors: http://paste.org/pastebin/view/17358 |
| 10:23.43 | *** join/#asterisk elzid (~IceChat7@host81-143-42-174.in-addr.btopenworld.com) |
| 10:30.37 | mutax | bbt: you can try to start it manually - Is it a debian? |
| 10:30.44 | bbt | centos |
| 10:31.11 | mutax | bbt: I guess asterisk is executed i a chroot and some libraries are updated and some not |
| 10:31.18 | *** join/#asterisk maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 10:31.29 | mutax | bbt: and ine of the libs causes a segfault. you have to find out which one |
| 10:32.01 | mutax | bbt: so you should find out what ${ASTSBINDIR} is, go there, maybe do a chroot into the corresponding dir |
| 10:32.05 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.59) |
| 10:32.17 | mutax | bbt: and execute asterisk -vvv -c (as the user) |
| 10:33.06 | mutax | ah: (core dumped) -> you can also try to find the corefile |
| 10:33.10 | bbt | the last thing that gets loaded is: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) nd then it segs |
| 10:33.33 | mutax | can you find out if that was part of the update? |
| 10:33.50 | mutax | either the asterisk is too new or the lib is I guess. |
| 10:34.11 | mutax | you could also try to put the corefile into gdb |
| 10:34.24 | mutax | gdb -core THE_COREFILE /path/to/asterisk |
| 10:34.46 | mutax | but I dont think this will help much, because there are no bedugging symbols in asterisk (usually) |
| 10:35.22 | bbt | these are all the updates: http://paste.org/pastebin/view/17359 |
| 10:36.08 | bbt | where would the core file be located ? |
| 10:36.58 | mutax | bbt: I think somewhere asterisk can write to, but not sure. It can be configured. But you can try finding a file just named 'core' in te asterisk dirs |
| 10:37.46 | mutax | bbt: have you tried rebooting, I see kernel modules beeing updated there |
| 10:38.24 | mutax | one quick way to find out if it is codec_g726.so - locate the file and rename it to codec_g726.so.bak |
| 10:38.43 | bbt | ok.. i did a reboot after the update, i usually do that as a general practice |
| 10:39.24 | refsgaard | Chainsaw i got an update with the log :) Can i get you to take a look at it |
| 10:40.21 | mutax | bbt: okay, then I recommend renaming the so files. can you throw a 'ldd /path/to/asterisk' intpo pastebin? |
| 10:40.45 | Chainsaw | refsgaard: Certainly. Do you have it on pastebin somewhere? |
| 10:40.55 | mutax | just to get a geberal idea how the linking looks. |
| 10:41.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:41.15 | bbt | tried rename the codec_g726.so and now it coredumps after: app_macro.so => (Extension Macros) |
| 10:41.18 | mutax | another thing: you can try to disable the 'safe' excution of asterisk to test if its working |
| 10:41.27 | refsgaard | ChanServ hold on i will :) |
| 10:42.12 | refsgaard | Chainsaw : http://pastebin.com/iewujR2g |
| 10:43.36 | bbt | mutax, what would a sample path for the ldd command look like? |
| 10:43.47 | Chainsaw | refsgaard: It's redacted, which isn't too helpful. |
| 10:43.47 | bbt | trying to figure out what i should be looking for |
| 10:43.49 | mutax | well, find the asterisk binary |
| 10:43.54 | Chainsaw | refsgaard: But they're calling a macro called "hang up". |
| 10:44.06 | Chainsaw | refsgaard: So it seems... like they're telling Asterisk to hang up and it's doing as told. |
| 10:44.12 | mutax | ldd shows all linked libs and if they can be found |
| 10:45.08 | refsgaard | ohh i see |
| 10:45.15 | refsgaard | is there somehow we can get a better log? |
| 10:45.22 | bbt | got it: http://paste.org/pastebin/view/17361 |
| 10:46.12 | mutax | If I were you I would do three things: try to execute asterisk directly without chroot or anything (you did that already, didn't you, just 10 minuteds ago ?) |
| 10:46.12 | mutax | Then I would rename the libs that seem to segfault on loading and do an update/reinstall of all asterisk-related packages. Have you ever compiled something for asterisk on your on this installation? |
| 10:46.32 | bbt | no i didnt |
| 10:47.13 | mutax | hmm. strange. |
| 10:47.20 | bbt | no not for asterisk, but i have for other stuff in the past |
| 10:47.23 | mutax | so try renaming the .so files -.- |
| 10:48.02 | mutax | at some point it will stop booting because of them missing, but when youre lucky we find out what lib causes the segfault. |
| 10:48.10 | bbt | what would the best way to execeute asterisk directly ? |
| 10:48.16 | bbt | asterisk & |
| 10:48.17 | mutax | Or you could try the reinstall. Maybe something is broken.. |
| 10:48.17 | bbt | ? |
| 10:48.19 | mutax | no |
| 10:48.24 | mutax | asterisk -vvv -c |
| 10:48.32 | mutax | vor very very verbose with console |
| 10:48.46 | mutax | ehrml. s/v/f/ |
| 10:49.39 | mutax | so if centos is using a chroot this seems to be broken to me (old libs in there) |
| 10:49.56 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 10:50.10 | mutax | maybe there is a commend to update the chroot automagically - but I dont know the installation on centos |
| 10:53.30 | bbt | hahaha |
| 10:53.43 | bbt | i found it |
| 10:53.58 | mutax | cool - tell me ;) I am curious |
| 10:53.59 | WIMPy | mutax: Hi. Not having any more luck today? |
| 10:54.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:54.18 | mutax | WIMPy: not really. outgoing calls are working, only incoming not |
| 10:54.19 | bbt | mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules_ |
| 10:54.25 | bbt | something in there isnt right |
| 10:54.43 | WIMPy | mutax: Is Audio ok now? |
| 10:54.46 | mutax | WIMPy: lcr receives a hangup as soon as the line is signalled. I am debugging in the lcr core |
| 10:54.48 | bbt | it was renaming the modules so i figured to move the entire folder |
| 10:54.58 | mutax | WIMPy: yap audio is okay now |
| 10:55.01 | bbt | s/it/i/ |
| 10:55.17 | bbt | loaded fine |
| 10:55.23 | WIMPy | mutax: Ok, at least something. What did you do? |
| 10:55.23 | mutax | WIMPy: but audio is okay - on the slower machine -.- |
| 10:55.33 | WIMPy | Ah, ok |
| 10:55.51 | mutax | WIMPy: changes from my dual core t61 ibm 2,4 GHz machine to an old celeron 1.7 GHz single core |
| 10:56.08 | mutax | WIMPy: are you familiar with the lcr debug output? |
| 10:56.33 | WIMPy | mutax: I prefer a trace. Much easier to read. |
| 10:56.47 | *** join/#asterisk mhaddog (~mhaddog@32.Red-88-25-252.staticIP.rima-tde.net) |
| 10:57.21 | mutax | WIMPy: I can pastebin one in a second, maybe I just found sth. |
| 10:58.54 | Chainsaw | refsgaard: For a better log, your customer needs to stop trying to interpret it. |
| 10:59.07 | Chainsaw | refsgaard: Which means, they need to stop cutting out what they think is irrelevant, because it might not be. |
| 10:59.38 | Chainsaw | refsgaard: I'm happy for them to scrub phone numbers or other confidential data, but I need the full trace. |
| 10:59.44 | bbt | mutax, is there an easy way to tell which module isnt right? apart from renaming them all individually until i find the culprit? |
| 10:59.59 | WIMPy | mutax: The output of misdn_log would be great. |
| 11:00.47 | mutax | bbt: humm. ldd on each one? |
| 11:01.02 | *** join/#asterisk Akiraaa (~Akira@92.81.196.47) |
| 11:01.08 | mutax | bbt: the culprit is, only asterisk can tell you which one is not loading -.- |
| 11:01.15 | refsgaard | Chainsaw thx allot :) Will pass that on. Very nice help here btw. |
| 11:01.32 | mutax | bbt: you can try to find the files in /lib and /usr/lib and copy that versoins manually into the chroot |
| 11:01.38 | Chainsaw | refsgaard: :) |
| 11:01.48 | mutax | bbt: then you should at least have the latest version of them |
| 11:02.05 | bbt | ok |
| 11:02.06 | mutax | bbt: or you can execute a diff on each file agains the versoin in your system |
| 11:02.14 | mutax | then you find the one that was not updated |
| 11:02.26 | mutax | WIMPy: I just made a trace for you ;) |
| 11:02.44 | mutax | WIMPy: I think the isdn adaptor is sending a ISDN message that lcr is not awaiting |
| 11:05.06 | mutax | WIMPy: http://pastebin.com/v77zgL73 |
| 11:05.51 | mutax | WIMPy: I am not sure, but I found this: (output of pid added by myself): |
| 11:05.51 | mutax | (in mISDN_handler() line 1864): Received MPH_INFORMATION_IND for port 0 (Ext). l3m->pid=65535 |
| 11:06.10 | mutax | and this pid is not handled in the switch statement later in the code |
| 11:06.40 | WIMPy | mutax: could you give me the output of misdn_log instead? |
| 11:06.47 | mutax | jup |
| 11:07.09 | refsgaard | Chainsaw just got this short log...Dont seem that the customer gets tha "whole log concept", sorry about that :) but here we go |
| 11:07.11 | refsgaard | [Apr 14 06:40:35] WARNING[10396] chan_sip.c: Maximum retries exceeded on transmi |
| 11:07.13 | refsgaard | ssion 189171ee66acf73773f62b4054d5f697@10.20.124.111 for seqno 2 (Critical Respo |
| 11:07.14 | refsgaard | nse) -- See doc/sip-retransmit.txt. |
| 11:07.16 | refsgaard | [Apr 14 06:40:35] WARNING[10396] chan_sip.c: Hanging up call 189171ee66acf73773f |
| 11:07.18 | refsgaard | 62b4054d5f697@10.20.124.111 - no reply to our critical packet (see doc/sip-retra |
| 11:07.21 | refsgaard | nsmit.txt). |
| 11:07.32 | refsgaard | sorry about the paste |
| 11:07.53 | refsgaard | ohhh wait |
| 11:08.02 | refsgaard | i read about that somewhere on google |
| 11:08.05 | refsgaard | is that a NAT problem? |
| 11:08.32 | Chainsaw | refsgaard: It certainly looks like it, because it's trying to send to an internal 10-range IP address. That's RFC1918 and not routable. |
| 11:09.19 | mutax | WIMPy: http://pastebin.com/8nQZHkkZ |
| 11:09.29 | Chainsaw | refsgaard: They may want to set nat=yes on your sip.conf entry. |
| 11:10.02 | Chainsaw | refsgaard: If that does not clear the problem up entirely, they should set directmedia=no (assuming Asterisk 1.6.2) or canreinvite=no (on older versions). |
| 11:10.17 | refsgaard | thx allot :) |
| 11:10.22 | mutax | refsgaard: dunny, but maybe localnet helps too and settoing the external hostname? |
| 11:10.27 | Chainsaw | refsgaard: (They go together, both the nat & directmedia should be set) |
| 11:10.50 | refsgaard | will try that :) |
| 11:11.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.144.143) |
| 11:12.21 | *** join/#asterisk davido1 (~davido@p54B0A723.dip0.t-ipconnect.de) |
| 11:12.28 | WIMPy | mutax: Hmm. No answer at all? |
| 11:12.30 | davido1 | hello irc |
| 11:12.40 | davido1 | I have a very simple question... |
| 11:12.46 | mutax | WIMPy: no, nothing. I set lcr to send me on voicemail |
| 11:12.55 | mutax | WIMPy: and it keeps recording amtpy wav files |
| 11:13.03 | davido1 | Is it just me? Or "#exec" doesn't work in ael files in asterisk 1.6? |
| 11:13.56 | *** join/#asterisk Pazzo (~ugelt@host156-36-static.14-79-b.business.telecomitalia.it) |
| 11:13.57 | mutax | WIMPy: (in ea_message_port() line 2561): EPOINT(1) release with cause=101 location=5 (terminal '', caller id '03062005858') |
| 11:13.58 | mutax | trace : 14.04.10 13:56:25.123 EP(1): RELEASE port 0 from CH(1) cause value=101 location=5-Remote-PBX |
| 11:13.58 | mutax | (in port_disconnect_release() line 2065): EPOINT(1) current multipoint cause 0 location 0, received cause 101 location 5. |
| 11:14.04 | WIMPy | mutax: I don't see it accepting the call. It's only offered twice nothing more. |
| 11:14.15 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 11:14.18 | mutax | cause value 101 feels strange. |
| 11:14.42 | mutax | if this is the one from the table, its a 'message not compatible with state' |
| 11:15.28 | davido1 | anybody? |
| 11:15.39 | mutax | WIMPy: jup, extactly that's what happens. asterisk and lcr think they got on hook, try to play audio or sth. then lcr signales the hangup, although they are not on hook -.- |
| 11:16.00 | mutax | davido1: i am not in ael, but isn't # a comment? |
| 11:16.11 | WIMPy | mutax: I don't see anything in your latest pastebin. Only the call offered twice, no repl whatsoever. |
| 11:16.26 | mutax | davido1: ah, no. bullshit, sorry |
| 11:16.33 | WIMPy | mutax: Only 'reveived' lines. |
| 11:16.34 | davido1 | Hehe, no problem mutax |
| 11:16.46 | mutax | WIMPy: jap. hmmmmmm |
| 11:17.07 | mutax | WIMPy: the strange thing is, outgoing calls are working. |
| 11:17.07 | davido1 | Anyway, no, usually u can use "#include" and "#exec" directives in *.conf files |
| 11:17.20 | mutax | WIMPy: so the cabling should be okay |
| 11:17.40 | WIMPy | mutax: agree |
| 11:17.55 | WIMPy | mutax: Probably some strange signalling. |
| 11:18.53 | mutax | can you interpet the first packet? |
| 11:19.07 | mutax | 14.04.2010 14:07:47.803671 received  8 bytes prim = 0102 id=0000ffff |
| 11:19.14 | WIMPy | Nothing strange about that, AFAICS |
| 11:19.18 | mutax | k |
| 11:19.31 | WIMPy | Oh, that's just layer1 stuff. |
| 11:19.54 | mutax | (in mISDN_handler() line 1903): the link became active before l2establish timer expiry. |
| 11:20.16 | mutax | so maybe layer2 is not working? |
| 11:20.32 | mutax | arrgh |
| 11:20.50 | WIMPy | How to you make ooutbound calls then? |
| 11:20.51 | mutax | I had these still in the interface.cfg - I am trying again.. narrgh! layer1hold yes ; layer2hold yes |
| 11:22.23 | mutax | what is name : cause_65 ? |
| 11:22.35 | mutax | Bearer capability not implement.? |
| 11:22.43 | WIMPy | mutax: Yes, unfortunaletly that prevents lcr from using the interface, if L1 is down. But I need it with Sphairon IAD as that will otherwise deactivate L1 while in conversation :-( |
| 11:23.04 | mutax | WIMPy: okay, l2 is working -.- |
| 11:23.23 | mutax | andI have loads of log from misdn_log ;) |
| 11:23.43 | WIMPy | good. |
| 11:25.06 | mutax | WIMPy: http://pastebin.com/KsgBP1JQ |
| 11:27.40 | pentanol | hi guys |
| 11:28.43 | pentanol | I want jump to another value if there hit something odd |
| 11:29.08 | pentanol | i.e... exten => _XXX,9,Gotoif($[$TRUNKS{${NUM_TRUNK} = ${SRV_ID}]?selfpass,14) exten => selfpass,14,NoOp() |
| 11:29.17 | mutax | WIMPy: this sounds familiar, although complete other techique http://blog.krisk.org/2008_03_01_archive.html |
| 11:30.01 | pentanol | if left equals right value it should jump to number 14 of priority |
| 11:30.25 | pentanol | but I got an error sent into invalid extension 'selfpass' in context 'default', but no invalid handler |
| 11:31.14 | *** join/#asterisk chasing`Sol (~rc4@217.54.176.166) |
| 11:31.14 | WIMPy | mutax: That really looks strange. |
| 11:31.25 | mutax | WIMPy: what exactly? |
| 11:31.43 | mutax | WIMPy: Maybe I'll try to call the manufacturer -.- |
| 11:31.51 | mutax | WIMPy: how hard is it to get a fritz card running? |
| 11:31.52 | WIMPy | mutax: Status/Cause 63. Why? |
| 11:32.03 | mutax | WIMPy: its in the logs |
| 11:32.10 | mutax | i made a trace in parallel |
| 11:32.16 | WIMPy | I can't see anything special in that SETUP. |
| 11:32.48 | mutax | but the cause 65 is after the hangup |
| 11:33.04 | mutax | which seem normal, considering it wants a bchannel after hangup |
| 11:33.28 | mutax | this is the cuplrit: CH(1): MT_RELEASE_COMP INDICATION U<-N cause : location = 1 : value = 101 layer 1 : up |
| 11:33.40 | mutax | some message that lcr does not like |
| 11:33.45 | *** join/#asterisk coppice (~chatzilla@93.176.64.202.dyn.pacific.net.hk) |
| 11:33.58 | bbt | mutax, i fixed it. i compiled a module by hand ages ago and forgot about it, turns out all i had to do was remove it. yay |
| 11:34.04 | mutax | I think I'll try to get the fritz card working to see if its my hardware |
| 11:34.08 | mutax | bbt: :DD |
| 11:34.47 | mutax | WIMPy: and if i still have the nerves I'll connect them with a crosses cable and try that out |
| 11:34.49 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 11:35.37 | mutax | WIMPy: and maybe I'll write to the manufacturer of that analog to isnd adapter... its a german company. |
| 11:36.02 | mutax | WIMPy: so I could even drive there and slap them if they f.up the protocol ;o) |
| 11:36.36 | WIMPy | mutax: Fritzcard depend on if it's HFC based or not. If it's HFC it's the same. |
| 11:36.52 | mutax | WIMPy: yesa, at least I can et rid of USB ;o) |
| 11:37.12 | mutax | 01:0e.0 Network controller: AVM Audiovisuelles MKTG & Computer System GmbH A1 ISDN [Fritz] (rev 02) |
| 11:37.24 | mutax | I think thats not HFC based... |
| 11:38.21 | WIMPy | Then you will have to try capi, I guess. |
| 11:41.48 | mutax | ja |
| 11:41.50 | WIMPy | mutax: It looks to me the adaptor doesn't like the proceeding message. But I really don't see why there is a status message. |
| 11:42.40 | mutax | hmm. I'll try the fritz card. can yo point me directly to the entry in the logfile? |
| 11:42.54 | mutax | I could try to add an if-clause ignoring that message? |
| 11:43.06 | mutax | or do you mean the hardware drops the connection? |
| 11:43.42 | WIMPy | mutax: Yes, It looks like it disconnects after the recovery. |
| 11:44.13 | mutax | :[ odd |
| 11:44.21 | WIMPy | However it's also strange, that it happens twice. |
| 11:44.49 | mutax | I think the call is announced twice |
| 11:45.22 | mutax | with the first two 'RING's on the analogue line I get some data on the isdn d channel |
| 11:47.04 | WIMPy | Ooops |
| 11:47.31 | WIMPy | Bug on my side. bbiab |
| 11:48.49 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:50.01 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 11:54.52 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:02.26 | WIMPy | mutax: Yes, I see all messages twice, except for the first setup. So that looks rather strange. (again) |
| 12:03.10 | mutax | hmm.... |
| 12:03.19 | davido1 | pentanol: it looks okay, unless you don't have all the priorities for selfpass there, I don't know why it wouldn't work |
| 12:03.21 | WIMPy | mutax: The really interesting part is the message at 14:21:30.053027 with an unknown protocoll. |
| 12:03.27 | mutax | WIMPy: well... CAPI not installed, chan_capi disabled! <- my next problem ;) |
| 12:03.36 | mutax | WIMPy: ill look into that |
| 12:04.14 | *** join/#asterisk rookmoot (rookmoot@ns1.wigo.fr) |
| 12:04.20 | rookmoot | hi ppl |
| 12:04.40 | mutax | WIMPy: its in there at least once more... as you sai |
| 12:04.42 | mutax | d |
| 12:04.52 | rookmoot | i wonder if i can do video/speech with multi users (like a conference) using asterisk ? |
| 12:05.19 | mutax | rookmoot: audio is no problem |
| 12:05.31 | rookmoot | mutax, and for video ? |
| 12:05.42 | mutax | WIMPy: I think I'll make lcr ignore this one.... |
| 12:06.07 | mutax | rookmoot: you need to put the videostreams of the people tgether in one larger video? |
| 12:06.16 | WIMPy | mutax: It looks to me as if lcr tries to decode that message and fails. I can't decode it, either. |
| 12:06.16 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 12:06.29 | mutax | WIMPy: it writes it in the log |
| 12:06.56 | mutax | debug.log:(in mISDN_handler() line 1864): Received MPH_INFORMATION_IND for port 0 (Ext). l3m->pid=65535 |
| 12:07.18 | rookmoot | mutax, i want to have a videostream for each ppl who have joinded the conference |
| 12:07.33 | WIMPy | mutax: I think that just a consequence, not the error itself. |
| 12:07.33 | rookmoot | same for audio |
| 12:07.49 | mutax | rookmoot: dunno, sorry |
| 12:08.06 | mutax | WIMPy: hmm. I'll make it ignore that, then setup capi ;) |
| 12:09.47 | WIMPy | mutax: Have you found out, where it starts to process received frames? |
| 12:10.04 | davido1 | Anyway, anyone can help with ael? |
| 12:10.31 | mutax | WIMPy: well I grepped for Received MPH_INFORMATION_IND |
| 12:10.37 | mutax | its in lcr/mISDN.cpp |
| 12:10.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:11.01 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:12.33 | mutax | WIMPy: no, nothing changes. |
| 12:13.21 | WIMPy | mutax: I try to find my way. But not sure if I'm going to find the spot. |
| 12:13.44 | mutax | me not either. |
| 12:13.48 | mutax | I focus on capi now |
| 12:14.08 | mutax | if the adaptor emits broken ISDN signalling It's a bit wasted time debugging lcr that much |
| 12:14.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:14.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:15.33 | WIMPy | Looks lake a classic double trouble to me. The Adaprot sends something we don't understand, and lcr doesn's just ignore it. |
| 12:15.52 | rookmoot | leifmadsen, lol mdr |
| 12:15.58 | leifmadsen | file: When you have a channel module (lets say SIP) and you want to send a call to another channel type (lets say IAX2), is the path of the call from chan_sip -> the Asterisk core -> chan_iax2 ? |
| 12:15.59 | mutax | WIMPy: great, mISDN just made me a kernel oops |
| 12:16.38 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:16.47 | WIMPy | mutax: Maybe that's related to undecodable message as well? |
| 12:16.55 | leifmadsen | file: I'm trying to understand the relationship between channel modules, and we're picturing it as a "gateway" to the Asterisk core, which then has some sort of internal language to pass off the information...? |
| 12:17.01 | mutax | no. i tried accessing /dev/capi20 |
| 12:17.55 | WIMPy | never liked capi |
| 12:18.04 | mutax | whooa |
| 12:18.11 | mutax | segfault on ls /dev |
| 12:18.24 | mutax | proicess gets killed immiediately |
| 12:18.29 | mutax | time for a reboot of my server |
| 12:18.38 | WIMPy | Erm!?! |
| 12:18.49 | ariel_ | Morning folks |
| 12:19.10 | mutax | WIMPy: stat() on /dev/capi20 makes a kernel oops |
| 12:20.02 | WIMPy | mutax: This is getting stranger and stranger. Do you have gremlisn on your system? |
| 12:20.14 | mutax | dunno |
| 12:20.22 | mutax | but each call of 'ls' now makes a kernel oops |
| 12:20.25 | mutax | strange indeed |
| 12:20.35 | mutax | the isdn card got pudged |
| 12:20.40 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 12:20.43 | mutax | rmmod -f is evil :( |
| 12:21.04 | WIMPy | mutax: It says so in the manuall. |
| 12:21.07 | WIMPy | -l |
| 12:21.34 | mutax | :} |
| 12:22.22 | davido1 | someone help with ael? |
| 12:22.38 | leifmadsen | ~ask |
| 12:22.39 | 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. |
| 12:23.04 | tuxx- | =] |
| 12:23.25 | mutax | I am here against my will, YES ;) 30 hours of my life *wine* |
| 12:23.31 | mutax | WASTED :D |
| 12:23.42 | davido1 | Can I use the #exec directive in ael? In asterisk 1.6.2, that is |
| 12:23.59 | leifmadsen | *whine* :) |
| 12:24.03 | davido1 | Because I keep getting a Syntax Error |
| 12:24.07 | leifmadsen | *wine* is a delicious drink |
| 12:24.16 | WIMPy | mutax: I've got the impression the problem is in misdn, not lcr. |
| 12:24.21 | davido1 | whilst *whine* is rarely drunk |
| 12:24.56 | coppice | some people drown in whining |
| 12:24.56 | mutax | leifmadsen: yep, wine would be a great idea,only its 3 pm here - a bit too early. and my engrish got worse. |
| 12:25.05 | mutax | WIMPy: I think so, yes |
| 12:25.07 | leifmadsen | mutax: only 8:25am here |
| 12:25.16 | davido1 | 14.25 here ;) |
| 12:25.37 | davido1 | PPL!!! Someone has used the #exec directive in ael??? |
| 12:25.47 | davido1 | So much wine... aye aye :P |
| 12:25.50 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 12:26.05 | leifmadsen | davido1: if they had, they would have said so by now. Be patient. |
| 12:26.34 | davido1 | true... I'm extremely impatient :P |
| 12:26.37 | mutax | in vino veritas |
| 12:26.51 | mutax | davido1: google? *covers* |
| 12:28.08 | davido1 | mutax: nah, tried alr. It's too ambiguous, I know it works in asterisk 1.4, but no one appears to have my problem with 1.6. And no one says that it doesn't work, but no one says it does. I dunno |
| 12:31.52 | WIMPy | mutax: What exactely did you try to disable in lcr? |
| 12:33.06 | *** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net) |
| 12:33.27 | TheCompWiz | can someone give me a few ideas for trying to figure out what makes ZOMBIE'd calls? |
| 12:34.10 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:34.16 | mutax | WIMPy: http://pastebin.com/zPSEx4Hc |
| 12:34.20 | coppice | I think most people calling from telemarketing call centres are zombies |
| 12:34.28 | leifmadsen | brains! |
| 12:34.37 | mutax | coppice: no, asterisks :) |
| 12:34.39 | leifmadsen | Rule 1: Cardio |
| 12:35.19 | WIMPy | mutax: Did you comment out the whole outer case? |
| 12:35.44 | mutax | no, i added the case 65535 and return 0 in the pid-case |
| 12:35.50 | WIMPy | Oh, you changed the case, yes. |
| 12:36.19 | mutax | *grr* the avm website where i should get the capi drivers for the card is down -.- |
| 12:36.28 | mutax | *the world hates me* |
| 12:36.31 | WIMPy | That's what I would have tried, as well. |
| 12:37.03 | mutax | but lcr or misdn emits the hangup in either case |
| 12:37.09 | WIMPy | mutax: No, it just saves you from finding out that you need an older kernel for it :-) |
| 12:37.17 | mutax | hmpf. okay |
| 12:37.39 | mutax | WIMPy: Is the capi-chan the riht thing in asterisk? |
| 12:37.47 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:37.48 | mutax | WIMPy: it seems i am hitting old manuals again |
| 12:38.07 | WIMPy | mutax: No, it sends status, then proceeding and then it receives a RelComp |
| 12:39.02 | WIMPy | Interestingly however it also sends two alerting after the first RelComp. |
| 12:39.04 | mutax | WIMPy: Grr. i dont even get capi running |
| 12:39.31 | mutax | WIMPy: the signaling is in the d channel, right? |
| 12:39.40 | WIMPy | yes |
| 12:39.43 | mutax | WIMPy: isnt the NT supposed to echo that back? |
| 12:40.07 | WIMPy | That's the E(cho) channel. |
| 12:40.13 | mutax | hmm |
| 12:40.18 | WIMPy | That's only used for collision detection. |
| 12:40.22 | mutax | right, that was on a seperate one |
| 12:40.48 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 12:40.53 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 12:41.06 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 12:41.44 | Skeeter- | [TK]D-Fender, do you know if the new dahdi fixes the problem encounter with the sangoma A200 card, i didnt see anything within the release notes |
| 12:42.06 | [TK]D-Fender | Skeeter-: "The problem"? I'm sorry... could you be a little more vague please? |
| 12:42.26 | leifmadsen | Skeeter-: you'll either have to try, or look at the release notes. I don't think [TK]D-Fender knows everything :) |
| 12:43.09 | Skeeter- | [TK]D-Fender, The issue, where when you restart wanrouter, you had to unplug and replug the fxo lines in order to make em active |
| 12:43.29 | leifmadsen | That sounds like a problem with the wanrouter |
| 12:44.07 | Skeeter- | i know |
| 12:44.15 | [TK]D-Fender | Skeeter-: Indeed, that clearly has nothing to do with DAHDI |
| 12:44.23 | Skeeter- | but commenting the line 4177 in dadhi-base.o fix the problem |
| 12:44.23 | leifmadsen | why would the DAHDI modules fix that then? |
| 12:44.30 | leifmadsen | Skeeter-: then file an issue |
| 12:44.36 | [TK]D-Fender | Skeeter-: Did you fix my blotchy colour printer output problem yet? |
| 12:44.53 | Skeeter- | color printer?? wth are you talking about |
| 12:45.01 | leifmadsen | I think that's his point... |
| 12:45.06 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:45.30 | Skeeter- | im confuse |
| 12:45.35 | leifmadsen | Skeeter-: he's a dick. |
| 12:45.55 | Skeeter- | leifmadsen, but he is smarter then both of us together |
| 12:45.56 | [TK]D-Fender | leifmadsen: Correction : 'He has a dick" |
| 12:46.04 | leifmadsen | Skeeter-: I disagree |
| 12:46.27 | Skeeter- | Sangoma told me to hand edit dahdi-base.o AND NOT edit the wanpipe driver itself |
| 12:46.29 | leifmadsen | not sure why you would presume that |
| 12:46.38 | leifmadsen | Skeeter-: then you need to file an issue, time to move on |
| 12:46.48 | leifmadsen | Skeeter-: if there is no issue that you can point to, then it won't get fixed |
| 12:46.56 | [TK]D-Fender | leifmadsen: unchecked self-deprecation. |
| 12:46.56 | leifmadsen | when the issue is closed, then it is fixed |
| 12:47.30 | Skeeter- | well, im not sure i wanna reach asterisk and tell them their product as an issue, ill probably get ban from here |
| 12:47.39 | Skeeter- | ill keep edting the files myself |
| 12:47.43 | [TK]D-Fender | Skeeter-: I'd go with leifmadsen's recommendation. If thats the fix they offer, then submit it as such. I don't think it should take long to make it in. |
| 12:48.02 | [TK]D-Fender | Skeeter-: It won't get fixed otehrwise. Submit it. |
| 12:48.08 | leifmadsen | Skeeter-: that is not correct -- if that were true, we woudln't have a bug tracker |
| 12:48.21 | Skeeter- | ban tracker |
| 12:48.21 | Skeeter- | hahaha |
| 12:48.55 | leifmadsen | in fact, I'd say that is a stupid statement you just made |
| 12:50.54 | [TK]D-Fender | davido1: I couldn't find any reference to this #exec directive in AEl you were referring to in the source docs or the WIKI. What are you attempting to do, and please link to some reference to it if possible |
| 12:51.56 | mutax | *grr* stupid question. for a fritz a1 card via capi20 - what do i need? I enabled capi20 and capifs in the kernel, loaded the modules. capiinfo says capi is not installed, as dies asterisk. do i need firmware for the card or what am I missing? most of the howtos are referencing to kernel 2.4 and seem to be outdated. |
| 12:52.32 | Skeeter- | if i post an issue for dahdi 2.2.X, will it be considered? |
| 12:52.50 | WIMPy | Oh, the wonderful world of proprietary drivers. |
| 12:53.03 | davido1 | [TK]D-Fender: http://www.sfr-fresh.com/unix/misc/asterisk-1.6.2.6.tar.gz:a/asterisk-1.6.2.6/doc/tex/asterisk/node22.html |
| 12:53.11 | mutax | WIMPy: -.- |
| 12:53.15 | leifmadsen | Skeeter-: fine |
| 12:53.37 | davido1 | [TK]D-Fender: I'm able to use the #include directive, but not the #exec... |
| 12:53.52 | WIMPy | mutax: Tried an T1 the other day. The driver is in the standard kernel, but it looks like it's impossible to load the firmware with a recent kernel. So it's pretty pointless. |
| 12:54.04 | mutax | *.* |
| 12:54.15 | Skeeter- | registration system seems slow |
| 12:54.16 | mutax | THE WORLD HATES ME -.- |
| 12:54.35 | Skeeter- | mutax, join my #hatered |
| 12:54.57 | WIMPy | mutax: Or something borke while I upgraded the kernel, but I don't believe too much in coincidents. |
| 12:55.28 | WIMPy | Didn't bother to downgrade again, yet. |
| 12:55.44 | mutax | now the usb adaptor hangs up and the two pci cards are the same make and model and dont have a driver any more? oh my gosh |
| 12:56.14 | [TK]D-Fender | davido1: And did you add the option they specify to asterisk.conf? |
| 12:57.18 | davido1 | [TK]D-Fender: Yup, that's why I'm starting to think that it might be that it just doesn't work for *.ael files in asterisk 1.6 |
| 12:58.07 | [TK]D-Fender | davido1: You're on 1.6.2.6? |
| 12:58.33 | WIMPy | mutax: Go to the next store and buy an el cheapo card. But that won't fix you adaptor and the interoperability issues, of course. |
| 12:58.41 | mutax | pfft - avm has a php-script as a gateway to their ftp, and the ftp is up and running -.- |
| 12:59.13 | davido1 | [TK]D-Fender: Nope, 1.6.2.0 |
| 12:59.21 | mutax | WIMPy: hmm - only i dont want to keep a big pc running 24/7. thought a small board without fans but USB would be ok -.- |
| 12:59.46 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:59.50 | [TK]D-Fender | davido1: I couldn't find that doc myself in 1.6.2.5. I'd recommend upgrading now.... |
| 13:00.12 | WIMPy | mutax: Get the number ported to sipgate. |
| 13:00.27 | mutax | :D |
| 13:01.27 | davido1 | [TK]D-Fender: Thanks, good to know that it might be a version issue then. Funny thing is that it seems to work in asterisk 1.4... :s |
| 13:01.35 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 13:01.54 | [TK]D-Fender | davido1: I find it odd that I couldn't find a reference to it anywhere myself |
| 13:02.08 | mutax | yea, prop. drivers: /usr/src/fritz/src/driver.h:101: error: conflicting types for ‘driver_init’ |
| 13:03.01 | davido1 | [TK]D-Fender: I know! And in the changelog from 1.4 to 1.6 it says that that directive can be used in all of asterisk configuration files |
| 13:03.06 | davido1 | o.O |
| 13:03.24 | WIMPy | I never liked AVM, tho I have to say their recent FritxBoxes seem to have very good interoperaility. |
| 13:04.23 | Skeeter- | Can i modify an issue? |
| 13:05.01 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:06.37 | leifmadsen | Skeeter-: which issue? you can only modify one that you submitted I think |
| 13:06.51 | leifmadsen | although to modify an issue you may need to be a bug marshal |
| 13:06.55 | Skeeter- | well i cant figure out how, but its minor |
| 13:07.01 | leifmadsen | lucky for you, I am a bug marshal |
| 13:07.08 | leifmadsen | what needs to be changed on which issue? |
| 13:08.07 | Skeeter- | https://issues.asterisk.org/view.php?id=17183. The point is, the Sangoma way to fix it is with 2.2.0 and i used 2.2.1, the line they ask to hand edit is 4077 when in the 2.2.1 version is 4177 |
| 13:14.00 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 13:14.34 | Dovid | leifmadsen: There were no bug marshals around so I put in a ticket. Am I missing something here or can it be a bug ? https://issues.asterisk.org/view.php?id=17180 |
| 13:15.45 | bent_screwdriver | anyone use a program like Asterisk Desktop Assistant, formerly Snapanumber, that is a little more stable? (click-to-call) |
| 13:16.20 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:17.13 | mutax | http://opensuse.foehr-it.de/ <- drivers for AVM cards and latest kernels |
| 13:17.33 | davido1 | bent_screwdriver: have you seen asterisk desktop manager? It's not thaaaaat good (no integration with outlook) but it's okay |
| 13:18.23 | mutax | WIMPy: HA! |
| 13:18.35 | mutax | 1 fcpci running fcpci-a400-18 A1 3.11-07 0xa400 18 |
| 13:18.39 | davido1 | bent_screwdriver: http://adm.hamnett.org/node/3 |
| 13:18.57 | bent_screwdriver | davido1: thanks, i'll take a look |
| 13:19.10 | davido1 | bent_screwdriver: sure mate |
| 13:19.24 | Skeeter- | leifmadsen, thanks i got an email, that you added an note |
| 13:20.14 | WIMPy | mutax: Not that much choice there. |
| 13:20.40 | mutax | WIMPy: i unpacked the rpms manually applied the patches and compield in in debian |
| 13:20.58 | mutax | now i have to see if asterisk sees any calls in capi |
| 13:21.04 | WIMPy | hardware wiese |
| 13:21.43 | WIMPy | -e |
| 13:21.47 | mutax | WIMPy: hmm. capi debug shows data, only nobody picks up :) |
| 13:21.51 | mutax | WIMPy: jap |
| 13:23.03 | *** join/#asterisk Khratos (~khratos@190.80.226.201) |
| 13:26.30 | [TK]D-Fender | BRB |
| 13:27.39 | mutax | WIMPy: guess what... http://pastebin.com/e63DtkwK |
| 13:28.05 | bent_screwdriver | when installing aserisk via yum is there a way to also save the rpm's in case i need to rebuild down the road and the repo isn't available? |
| 13:28.25 | mutax | WIMPy: its not lcr or misdn. its the isdn adaptor -.- |
| 13:29.26 | WIMPy | mutax: That's what I meant by 'double trouble'. Both sides have some sort of misfeature and the combination makes a no-go. |
| 13:29.45 | mutax | WIMPy: this is the fritz card, not taking the call |
| 13:30.13 | *** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es) |
| 13:30.19 | WIMPy | mutax: I haven't tried capi recently, it's not based on misdn layers, is it? |
| 13:30.57 | mutax | WIMPy: outgoing calls work like a charm! |
| 13:31.09 | mutax | or at leas, originate on the cli |
| 13:31.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:31.52 | davido1 | bent_screwdriver: I thought your distro would keep the downloaded rpm's somewhere... You should check that with your distro. |
| 13:32.07 | *** join/#asterisk voipnet-tech (~russ@rrcs-24-39-187-146.nys.biz.rr.com) |
| 13:33.51 | mutax | WIMPy: yup, outgoing is prefect |
| 13:35.26 | WIMPy | mutax: Hmm. Do I miss something or is the call rejected immediately with capi? |
| 13:36.02 | mutax | i think so, yes. basically the same with the usb adaptor |
| 13:36.05 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-172-249.clienti.tiscali.it) |
| 13:36.13 | mutax | I see a cause 101 again |
| 13:37.26 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:38.07 | bent_screwdriver | davido1: CentOS - i did updatedb and locate *.rpm and it only found one, and it wasn't any yum install rpm. i will do some more research i guess... |
| 13:41.03 | bent_screwdriver | davido1: looks like you have to add this to yum.conf and then it will save them: keepcache=3D1 |
| 13:41.22 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:42.14 | davido1 | bent_screwdriver: oh, good to know. |
| 13:44.09 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 13:45.12 | mutax | <PROTECTED> |
| 13:45.12 | mutax | <PROTECTED> |
| 13:45.12 | mutax | :( |
| 13:48.19 | bent_screwdriver | davido1: actually just found that it is keepcache=1 on cent. the other settings was from a forum and not sure what distro it was for. just thought I would let you know so i didnt mess you up down the road...;) |
| 13:49.54 | davido1 | bent_screwdriver: I appreciate it. :) |
| 13:52.00 | *** join/#asterisk gelpg (~chatzilla@94-21-99-125.pool.digikabel.hu) |
| 13:53.09 | gelpg | hi |
| 13:53.16 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:53.37 | gelpg | I have to solve a problem and I'm not shure if asterisk is capable of |
| 13:54.00 | gelpg | I have an ISDN line |
| 13:54.24 | gelpg | actually 2 |
| 13:54.47 | gelpg | so when the fist 2 channel is busy |
| 13:55.23 | gelpg | can asterisk detect if it is a third incoming call and give a forward to another phone number? |
| 13:56.35 | WIMPy | gelpg: I don't think so. But you can tell your telco. Or get the two lines bundled. |
| 13:56.55 | Corydon76-dig | gelpg: Yes, use ChanIsAvail |
| 13:57.45 | WIMPy | Corydon76-dig: I don't think you will get into the dialplan at all. |
| 13:58.11 | Corydon76-dig | WIMPy: if he's got 2 ISDN lines, he's got 4 voice channels |
| 13:58.43 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:59.20 | gelpg | yes, my probles is that I cannot organize these lines into a ptp |
| 13:59.26 | gelpg | only a ptmp |
| 13:59.41 | davido1 | Corydon76-dig: Yes, so ChanIsAvail would work for outgoing calls, but wouldn't detect if there's a third incoming call on that line... I don't know if that's the problem.. Maybe I misunderstood? |
| 13:59.52 | WIMPy | gelpg: ptmp bundles are possible. |
| 13:59.55 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:59.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.01 | gelpg | and I have to keep a certain phone number |
| 14:00.07 | Corydon76-dig | gelpg: test each line in succession, total up the channels in use, and you have an answer. Just note that your own channel will appear to be busy, so you don't want to take any action until 3 lines are in use |
| 14:00.18 | gelpg | my problem is the incoming, not the outgoing |
| 14:00.28 | WIMPy | gelpg: Set a call forward on busy with your operator. |
| 14:01.11 | gelpg | the operator sais it should be done by me, who operates the phone exchange |
| 14:01.50 | WIMPy | gelpg: They won't set up forwarding for you? |
| 14:02.00 | gelpg | nope |
| 14:02.25 | gelpg | the're saying it should be done by me |
| 14:02.47 | WIMPy | gelpg: But you can set up forwarding on their switch? |
| 14:03.06 | mort_gib | gelpg: You get things confused |
| 14:03.34 | mort_gib | gelpg: You either configure BRI to use PTP or PTMP, but that decision is made by your provider! |
| 14:03.35 | gelpg | it"s possible, so please fix me |
| 14:03.46 | mort_gib | gelpg: He he, fix you!! |
| 14:03.52 | dgd_ | :) |
| 14:03.54 | gelpg | ok, let me clear the situation |
| 14:04.07 | gelpg | there is ptmp with a certain phone number |
| 14:04.08 | mort_gib | gelpg: Yes, lets clear the situation :-) |
| 14:04.16 | mort_gib | Who will fetch beer?? |
| 14:04.17 | gelpg | they cannot move this number into ptp |
| 14:04.28 | *** join/#asterisk smooth_penguin (~smoove@59.95.50.214) |
| 14:04.33 | WIMPy | gelpg: The easiest way wold be to get a phone to set up the forwarding rule. I don't know os any way to do it with Asterisk. With old i4l there was a thing called cfctrl. |
| 14:05.04 | WIMPy | s/ os / of / |
| 14:05.08 | gelpg | so i have to keep this pmp which has only 2 channel |
| 14:05.12 | mort_gib | gelpg: Your provider will be able to bundle lines, by assigning a primary number for a bundle, and several secondary numbers |
| 14:05.15 | mort_gib | if required |
| 14:05.43 | mort_gib | The choice between PTP/PTMP is not associated with bundling! |
| 14:05.48 | gelpg | if there is a third incoming, i have to give a command or something to forward the third call to a phone number |
| 14:05.58 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 14:06.14 | gelpg | mort_gib: yes, we asked them |
| 14:06.15 | mort_gib | No, you have to ask your provider to bundle the lines, or whatever they call it |
| 14:06.32 | gelpg | but they can't do it with that phone number |
| 14:06.34 | mort_gib | gelpg: And they said, bla bla, can't do that! |
| 14:06.39 | mort_gib | gelpg: not a first |
| 14:06.43 | gelpg | we can get new phone numbers |
| 14:06.46 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:07.02 | mort_gib | gelpg: You do NOT have to do anything on Asterisk for this to happen |
| 14:07.04 | gelpg | but we have to keep the old ones |
| 14:07.12 | mort_gib | !! (!..!) |
| 14:07.31 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 14:07.40 | gelpg | so it can be done by the phone company |
| 14:07.50 | mort_gib | Hey, you ask them to pour down whatever number you want, in the bundle of lines you have |
| 14:08.04 | WIMPy | gelpg: They can do round about everything, if they know how :-) |
| 14:08.05 | mort_gib | on analogue lines, it's one number per line |
| 14:08.13 | mort_gib | Not so with BRI/PRI |
| 14:08.28 | gelpg | I have a BRI |
| 14:08.39 | mort_gib | You can have 30 channels and one number, or 30 numbers on two channels |
| 14:08.56 | mort_gib | Depending on the marketing dept of the loco telco |
| 14:09.10 | mutax | WIMPy: hmm. now its working.... |
| 14:09.11 | mort_gib | Where are you in the world?? |
| 14:09.14 | gelpg | yes, theoretically it works |
| 14:09.20 | mort_gib | hungary |
| 14:09.20 | gelpg | welcome to Hungary |
| 14:09.22 | gelpg | :-d |
| 14:09.29 | mutax | WIMPy: dunno why, but incoming calls are working |
| 14:09.36 | mort_gib | Don't know about Hungary :-) |
| 14:09.42 | mutax | WIMPy: *investingating* |
| 14:09.43 | WIMPy | AFAIK there's usually a limit of 16 MSN/BRI |
| 14:09.45 | gelpg | it is in central europe |
| 14:09.49 | mort_gib | I do Asterisk in UK/Gibraltar/Spain |
| 14:09.57 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 14:09.59 | mort_gib | Then you are PTP |
| 14:10.09 | gelpg | no, ptmp |
| 14:10.16 | WIMPy | mutax: what combination works? |
| 14:10.18 | mort_gib | Ah, ok... |
| 14:10.23 | gelpg | i can ask to switch into ptp |
| 14:10.27 | mort_gib | Same S**T |
| 14:10.38 | gelpg | but they can't move this number into ptp |
| 14:10.52 | gelpg | because they need a 0-9 open range |
| 14:11.04 | gelpg | and unfortunatelly it's occupied |
| 14:11.07 | mort_gib | You can forward a call, but not if the channel is in use, only the telco can do this |
| 14:11.13 | mutax | WIMPy: fritz card |
| 14:11.27 | mutax | WIMPy: and debug=99, verbose=99 ;) |
| 14:11.29 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:11.35 | WIMPy | mort_gib: Sure ytou can, but not with Asterisk. |
| 14:11.43 | *** join/#asterisk moy (~chatzilla@74.12.133.16) |
| 14:11.48 | mort_gib | You need to INSIST that they trunk/bundle the ISDN circuits |
| 14:11.55 | mutax | maybe exten => foo,1,Ringing helped |
| 14:12.04 | gelpg | ok, thanks for the advice |
| 14:12.13 | mort_gib | WIMPy: He wants to forward the next incoming call on a busy circuit |
| 14:12.27 | mutax | WIMPy: I learned there is a timeout of 4 seconds in the capi (?) so when asterisk doesnt take the call it gets rejected... |
| 14:12.41 | mutax | WIMPy: andrining sets this timout to 120 secs |
| 14:12.45 | WIMPy | gelpg: However bundling of ptmp lines can have soem side effects, like some supplementary services no longer working. |
| 14:12.47 | mort_gib | That is DIFFICULT on BRI, unless you have "uhm" access to the telcos network |
| 14:13.31 | WIMPy | mort_gib: No, that's called CD and can even be done by better phones. |
| 14:14.35 | WIMPy | Or by using the ancient i4l. |
| 14:15.10 | mort_gib | If you have one BRI circuit, and both channels are busy, then how can you know that a third call is waiting?? |
| 14:15.35 | WIMPy | mort_gib: Just like any incomming call. |
| 14:15.50 | WIMPy | The number of calls is not limited by B-cahnnels. |
| 14:16.23 | WIMPy | You can easily have 10 simultaneous incomming calls on a single BRI. |
| 14:16.38 | WIMPy | You just can't accept all of them. |
| 14:16.39 | mutax | WIMPy: immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were received on incoming call (no destination number yet) ;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE. info like REDIRECTINGNUMBER may be lost, but this is necessary for drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE. |
| 14:16.45 | WIMPy | But you can chose which to accept. |
| 14:17.01 | mutax | WIMPy: commenting this out makes the call unanswered |
| 14:17.05 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:17.33 | WIMPy | mutax: Right. The sending complete was missing. Damn. That really should have been obvious :-( |
| 14:17.54 | mutax | well, I learnd it the hard way then... |
| 14:18.02 | WIMPy | I's definitely call that a bug. |
| 14:18.09 | WIMPy | I'd |
| 14:18.09 | mutax | question: does this help us with the usb thingy? |
| 14:18.37 | patrb | so I'm trying to figure out the reason why app_addon_sql_mysql isnt compiling (1.6.20)? I have mysql devel and asterisk both installed...no errors during compiling...it just seems to skip the app...anyone else experience this with asterisk-addons-1.6.20? |
| 14:18.57 | WIMPy | mutax: You could try a s extension. |
| 14:19.50 | mutax | WIMPy: did that. |
| 14:19.57 | WIMPy | mutax: And slap th manufacturer of that box. And KDG as well for selling it. |
| 14:20.03 | mutax | :) |
| 14:21.09 | *** join/#asterisk Faithful (~Faithful@121.91.118.214) |
| 14:21.11 | WIMPy | is always amazed how missing things never draw your attention. :-( |
| 14:21.18 | mutax | ;) |
| 14:22.06 | Katty | hi |
| 14:22.10 | Katty | my asterisk does not work at all |
| 14:22.11 | Katty | how to fix pls |
| 14:22.22 | mutax | ~ask |
| 14:22.23 | 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. |
| 14:22.27 | patrb | oh katty |
| 14:22.28 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 14:23.02 | patrb | hey katty, have you ever compiled asterisk-addons-1.6.20? |
| 14:23.12 | Katty | yesh |
| 14:23.36 | Katty | why did yours asplode |
| 14:23.38 | patrb | do you have any idea why when I compile it, it skips app_addon_sql_mysql completely...no errors |
| 14:23.46 | patrb | im about to paste it to paste bin |
| 14:23.49 | jhirley | o/ |
| 14:23.53 | Katty | hmmmmmmmm |
| 14:24.02 | Katty | my first guess would be some sort of dependency problem |
| 14:24.04 | Katty | hi jhirley |
| 14:24.06 | ba1dy | patrb: mysql libs installed? |
| 14:24.11 | patrb | mysql-devel |
| 14:24.16 | jhirley | Hello Sunshine !! |
| 14:24.23 | Katty | but i use postgresql |
| 14:24.41 | Katty | so i doubt i would have bumped into the issue regardless |
| 14:24.48 | Katty | hugs jhirley |
| 14:24.53 | patrb | ba1dy: I have mysql-devel installed..did yum search mysql | grep libs and nothing came up |
| 14:25.00 | patrb | katty: np, thanks anyway |
| 14:25.19 | ba1dy | patrb: i am not very familiar with centos |
| 14:25.24 | davido1 | see ya guys |
| 14:25.25 | *** part/#asterisk davido1 (~davido@p54B0A723.dip0.t-ipconnect.de) |
| 14:25.42 | Katty | what about something like... |
| 14:25.43 | Katty | libmysql |
| 14:25.58 | patrb | no matches :( |
| 14:26.00 | Katty | libmyodbc - the MySQL ODBC driver |
| 14:26.25 | patrb | 1 sec, let me paste my compile |
| 14:27.58 | Katty | YOUR ONE SECOND IS UP SIR |
| 14:29.03 | patrb | doh |
| 14:29.04 | patrb | http://pastebin.com/AvjgSZeV |
| 14:30.05 | patrb | I may have just figured it out... |
| 14:30.14 | patrb | yup..problem solved |
| 14:30.24 | *** join/#asterisk kash (~kash@190.196.71.206) |
| 14:30.24 | Katty | ya weren't holdin yer tongue right |
| 14:30.53 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 14:31.01 | Katty | hi Skeeter- |
| 14:31.10 | Skeeter- | https://issues.asterisk.org/view.php?id=17183#bugnotes . I cant undertstand anything he is saysing |
| 14:31.17 | Skeeter- | Katty, Morning |
| 14:31.20 | patrb | heh..so i didnt have mysql-devel installed when I intially compiled the addons....never went back and ran the configure script after installing mysql-devel |
| 14:31.30 | patrb | facepalms himself |
| 14:31.50 | Katty | ohnoes! |
| 14:32.01 | *** join/#asterisk kartik (~koolkarti@117.199.113.80) |
| 14:32.07 | Katty | Skeeter-: which he? |
| 14:32.09 | kash | Hi everybody ... just a little question .. i m looking for a good CDR Viewer .. i m looking in google , but all the CDR Viewer that i have founded got last version in 2008 , astcdrviewer , asternic cdr stats ... i just need something to see call stats .. like a call center monitor . . |
| 14:32.23 | Skeeter- | Katty, check the issue i posted |
| 14:33.19 | Katty | Skeeter-: yeah, but which person |
| 14:33.26 | Katty | Skeeter-: there's like 4 people in the conversation |
| 14:33.31 | Corydon76-dig | kash: Part of the problem is that CDRs suck for doing anything but a 1950s era telephone call |
| 14:33.38 | Skeeter- | pabelanger |
| 14:33.44 | Skeeter- | wil32 is me |
| 14:34.10 | *** join/#asterisk pkecastillo (~pirruar@190.113.141.122) |
| 14:34.17 | Katty | Skeeter-: ask leifmadsen |
| 14:34.26 | pkecastillo | Hello guys! |
| 14:34.45 | Katty | kash: we use asterisk-stat |
| 14:34.55 | Skeeter- | ill just ask to close the casse |
| 14:35.07 | Skeeter- | it will get too technical for me |
| 14:35.19 | Corydon76-dig | Skeeter-: You inlined a patch. Patches have to be uploaded as a separate file to the issue |
| 14:35.51 | Skeeter- | i didnt know that i inlined a patch |
| 14:35.58 | Skeeter- | i dont even know how patchs work |
| 14:36.38 | *** join/#asterisk kartikm (~koolkarti@117.199.113.80) |
| 14:36.41 | Corydon76-dig | Skeeter-: Any output generated with a diff command |
| 14:37.00 | Skeeter- | do i need to put that ouput into a txt file? |
| 14:37.13 | Katty | i would guess yes |
| 14:37.29 | Corydon76-dig | Yes |
| 14:38.02 | *** join/#asterisk kartik (~koolkarti@117.199.113.80) |
| 14:40.49 | dandre | hello, |
| 14:42.05 | Katty | hi |
| 14:42.06 | [TK]D-Fender | kash: Call centers (inbound) usually imply use of Queues which have their own logging, not CDR. CDR viewers have little reason to change as CDR does not change and there are only so many ways to look at them. This is not the case with Queues |
| 14:43.29 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 14:43.36 | pabelanger | Help! I found a bug! |
| 14:43.39 | dandre | is there any way to modify the text displayed on the phone while calling a party? |
| 14:43.39 | dandre | I have some shocuts managed by my asterisk dialplan and when the user dials this shortcut I would like to display the real number (eg if 1234 is dialed, the phone should display "Dialing +1 800 123456" |
| 14:43.40 | pabelanger | :) |
| 14:43.47 | [TK]D-Fender | pabelanger: SQUISH IT!!!! |
| 14:44.07 | [TK]D-Fender | dandre: Depends on the phone. |
| 14:44.30 | pabelanger | Anybody using chan_skype with Asterisk 1.6.2? I am having a difficult time getting it to actually work. IE: None of my skype users are logging. |
| 14:44.31 | dandre | I am using thomson st2030 |
| 14:44.43 | [TK]D-Fender | dandre: What does their very fine manual say? |
| 14:44.53 | pabelanger | And I don't seem to see any actually debugging from the driver. |
| 14:44.58 | dandre | nothing about this |
| 14:45.12 | [TK]D-Fender | dandre: That is likely your answer |
| 14:45.19 | WIMPy | dandre: You could try sendtext. |
| 14:45.24 | dandre | but I haven't found any solution whtever the phone used |
| 14:45.32 | [TK]D-Fender | dandre: PBX's don't tell SIP phones what to display normally. |
| 14:45.33 | dandre | ok |
| 14:46.00 | [TK]D-Fender | pabelanger: What makes this a "bug" exactly? |
| 14:46.17 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:46.21 | wcselby | o/ |
| 14:47.24 | kash | thanks .. lets go to google to learn about the queues .. |
| 14:48.09 | pabelanger | [TK]D-Fender: :) Don't mind the bug comment when I joined. Was joking around. I don't believe it to be. |
| 14:49.39 | kruemeltee | does anybody know what kind of hardware I need for about 150 telephones to handle? |
| 14:49.55 | pkecastillo | guys, when call with command "originate", and next is an application, for example (playback (tt-monkeys)). |
| 14:49.58 | pkecastillo | whats is the event or message that launches asterisk to find and launch hangup?. |
| 14:50.02 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:50.03 | wcselby | kruemeltee - what kind of telephones? |
| 14:50.05 | [TK]D-Fender | kruemeltee: Doing what? |
| 14:50.13 | kruemeltee | SIP Telephones ... all of them |
| 14:50.40 | wcselby | kruemeltee - how many concurrent calls, what the general plan (office system, call center, etc)? |
| 14:50.50 | WIMPy | kruemeltee: Sounds expensive. Not to buy but in operation. |
| 14:50.50 | [TK]D-Fender | pkecastillo: "launch hangup"? Huh? |
| 14:51.16 | WIMPy | pkecastillo: When the application exits. |
| 14:51.34 | pkecastillo | sorry, what generates the hangup... |
| 14:51.46 | [TK]D-Fender | pkecastillo: Who's hangup. There are now 2 calls. |
| 14:52.00 | [TK]D-Fender | pkecastillo: The original, and the Originated one. |
| 14:52.04 | kruemeltee | there is a call center with 8 agents ... and I think about maximum 100 similiar calls |
| 14:52.22 | wcselby | what is a similar call? you mean concurrent? |
| 14:52.29 | kruemeltee | absolutely maximum ... I think mostly about 20-40 similiar calls |
| 14:52.34 | [TK]D-Fender | kruemeltee: Coming in over what? How many simultaneous calls? How many being recorded? Conferencing? |
| 14:53.05 | wcselby | kruemeltee - you've got PRI/analog coming into the premises or you planning on doing this all over voip? |
| 14:53.09 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 14:53.19 | kruemeltee | no conferencing ... and no recordings |
| 14:53.41 | [TK]D-Fender | kruemeltee: Any modern system can do this then. |
| 14:53.59 | pkecastillo | humm... so,, I can put 2 or 3 applications per originate? |
| 14:54.12 | pkecastillo | or only one app? |
| 14:54.30 | [TK]D-Fender | pkecastillo: When you tell it an app, its a 1-shot deal. Otherwise point it to an EXTENSION |
| 14:54.44 | wcselby | gotta run to a meeting |
| 14:55.03 | wcselby | kruemeltee - from what you've said, sounds like [TK]D-Fender is right...any modern, dual/quad core system can handle this |
| 14:55.35 | wcselby | kruemeltee - the important thing with an asterisk server is processor. don't skimp on ram, but ram is less important than a good processor. |
| 14:55.38 | wcselby | anyways, i'm off |
| 14:55.39 | pkecastillo | humm, I understand!... |
| 14:56.46 | kruemeltee | I've got a Sip-Provider that let me call 20 silmutaneous calls ... after this we have 8 elmeg ICT 880 connected to asterisk ... and they want to call "intern" (means from one elmeg to another) with the help of asterisk |
| 14:57.10 | *** join/#asterisk geneticx_wrk (~jorge@host-208-88-126-198.biznesshosting.net) |
| 14:57.45 | [TK]D-Fender | kruemeltee: Ok, looks like a basic SIP>Analog FXS gateway. SHould be easy enough |
| 14:58.36 | kruemeltee | we've got an ibm e-server 345 |
| 14:58.46 | [TK]D-Fender | kruemeltee: That should do as well |
| 14:59.09 | wcselby | well, short meeting |
| 14:59.24 | wcselby | get there and find out the other person has another meeting going on |
| 14:59.28 | wcselby | i love meetings! |
| 14:59.34 | wcselby | </sarcasm> |
| 14:59.44 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.15 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 15:00.33 | *** join/#asterisk Tim_Toady (~moi@188.4.0.11.dsl.dyn.forthnet.gr) |
| 15:00.42 | TSM | i seem to be getting a lot of 'pbx.c: Data Buffer Size Exceeded!' has anyone seen this error? |
| 15:00.48 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:03.19 | p3nguin | tsm: How's the memory usage for asterisk? |
| 15:08.39 | TSM | err |
| 15:09.26 | p3nguin | noraml? high? |
| 15:10.03 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:11.08 | mdg | <3 meetings |
| 15:11.28 | *** part/#asterisk mutax (~mutax@erde.kf.TU-Berlin.DE) |
| 15:13.44 | kruemeltee | says goodbye :-) |
| 15:17.36 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:22.43 | *** join/#asterisk Z_God (~julius@wlan229147.mobiel.utwente.nl) |
| 15:27.11 | TSM | p3nguin: sos was pulled away from my desk, 2G ram, total/1723596 used/1709376 free/14220 buffers/187804 chached/867496 |
| 15:27.25 | TSM | p3nguin: i dont seem to be having any problems though |
| 15:28.17 | p3nguin | If asterisk is using 1723596, there's a problem. |
| 15:29.36 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 15:29.45 | TSM | p3nguin: the server has httpd/dhcpd/ftp/mysql/op_server all running on it too |
| 15:30.05 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 15:30.20 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:30.42 | p3nguin | Wonderful. Now how much memory is asterisk using? |
| 15:31.11 | McBoingbo | Phone is dropping calls getting "NWIF: nw_task() - Can't find associated CCB!" in the app-log for the device, any ideas? |
| 15:34.27 | idespinner | p3nguin, is PS the command you would use to find out memory usage? is VMsize the proper column? |
| 15:34.58 | p3nguin | I would probably use top or htop, but ps would also show it (provided the right options). |
| 15:36.05 | *** join/#asterisk skatz (~skatz@static-217-133-145-50.clienti.tiscali.it) |
| 15:36.21 | *** join/#asterisk Pazzo (~ugelt@host156-36-static.14-79-b.business.telecomitalia.it) |
| 15:41.50 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 15:42.37 | TSM | p3nguin: err let me check |
| 15:44.20 | p3nguin | ps l -C asterisk |
| 15:45.04 | TSM | p3nguin: 4 100 3323 3317 15 0 632848 35848 stext Sl ? 29:22 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 15:45.42 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 15:46.12 | adnc | hello, how can i find out if a channel is currently in use or free from the cli? |
| 15:46.28 | p3nguin | adnc: core show channels |
| 15:46.56 | adnc | p3nguin, if the channel is in use will it show active? |
| 15:47.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:48.54 | skatz | hi everybody, i have a problem with background(). It starts playing my welcome message but it's being truncated after some seconds. Could anyone help me? |
| 15:49.04 | adnc | p3nguin, thank you for your tipp, do you know if there is a agi codesnipped that does extract the state from the output of core show channels? |
| 15:49.10 | p3nguin | tsm: 632848k virt and 35848k res isn't really too much, but I'm not sure if there is a limit set for processes on your system. |
| 15:49.41 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:49.52 | *** join/#asterisk lazni (~lazni@118.71.103.4) |
| 15:51.10 | p3nguin | adnc: There's probably something, but I'm not a programmer. |
| 15:53.15 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 15:53.45 | TSM | p3nguin: not that im aware off, its been running since jan1st without reboot but * has been reloaded several times, if it not causing a problem i dont think i should be concerned about it too much, duno |
| 15:54.30 | p3nguin | tsm: But there IS a problem, otherwise you would not have offered up that error for someone to analyze. |
| 15:54.49 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:56.03 | p3nguin | My novice suggestion was that asterisk was using too much memory, thus causing the error that you displayed. |
| 15:57.12 | TSM | p3nguin: true, it needs to be logged and noted, but its not a priority 1, seems now after changing a setting and reloading its gone away, not sure for how long, ile have to monitor it |
| 15:57.34 | p3nguin | Check the memory usage after you've reloaded it. |
| 15:58.01 | *** join/#asterisk rare1980_ (~rare1980@115.186.4.131) |
| 15:59.27 | TSM | p3nguin: not sure now as i think i had already done the last ps after the reload, darn, it seems to only have started 2h ago until i did the reload, poop ok ile check back every few days to see if it comes up again |
| 16:00.15 | *** join/#asterisk eliyahud (~eliyahu@85.65.125.39.dynamic.barak-online.net) |
| 16:00.18 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:01.34 | wcselby | has anyone here used any professional music on hold recording companies? |
| 16:01.44 | wcselby | or can recommend one? |
| 16:01.50 | p3nguin | You could put something like "0 3 */2 * * /bin/ps l -yC asterisk" into crontab, and the output should be emailed to you. |
| 16:03.02 | *** part/#asterisk lazni (~lazni@118.71.103.4) |
| 16:04.37 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:05.10 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 16:05.35 | TSM | p3nguin: cool ile do that |
| 16:05.42 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 16:06.54 | edwin_quijada | FastAGi perl can run in windows>? |
| 16:07.10 | edwin_quijada | I need run an AGI script from windows server |
| 16:08.42 | tzafrir_laptop | edwin_quijada, I suppose windows can listen on TCP ports and such. Not sure, though |
| 16:09.58 | tzafrir_laptop | wcselby, what's so special about "music on hold"? You're looking for a professional musician? |
| 16:10.07 | edwin_quijada | tzafrir_laptop: I have read about FastAgI perl module but it says just about Linux and use other modules just 4linux |
| 16:10.56 | p3nguin | My thought about that is to get permission (or buy license) to use music that you like. You'll probably spend less money on the licensure than you would on fees to have music created for you. |
| 16:11.10 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:11.34 | wcselby | tzafrir_laptop - I'm looking for a client, that doesn't like the default asterisk hold music. they could really use a professional hold music / advertisement thing. i've had another client that has done it in the past, but I can't remember who they used. |
| 16:11.35 | tzafrir_laptop | There should be plenty of places to find such music that can be relicensed |
| 16:11.37 | wcselby | it's more than just music |
| 16:11.43 | tzafrir_laptop | e.g. http://magnatune.com/ |
| 16:11.59 | wcselby | it's the recorded message along with the music |
| 16:12.42 | tzafrir_laptop | Or see the CREDITS file in the existing moh file. |
| 16:12.43 | wcselby | right now they've got me looking at streaming an internet radio station |
| 16:12.54 | paulc | wcselby: Check out shockwavesound.com maybe? |
| 16:12.58 | p3nguin | You can hire Allison to do your messages if you use her voice in the IVR. |
| 16:13.05 | Qwell | tzafrir_laptop: are you familiar with how Linux handles FS cache of very large files? |
| 16:13.19 | wcselby | she doesn't do our IVR, we had a former employee record that |
| 16:13.20 | tzafrir_laptop | Qwell, not anything special |
| 16:13.20 | Qwell | (think 2GB files, where only a portion of the file is read at a time) |
| 16:13.24 | wcselby | sorry, the client did |
| 16:13.31 | wcselby | i'm here so much I sometimes feel like an employee |
| 16:13.33 | Qwell | would it be trying to cache the entire file? |
| 16:14.10 | tzafrir_laptop | I suppose it would be trying to cache on the block level. But I'm not really sure |
| 16:14.20 | Qwell | that would make sense |
| 16:14.47 | Qwell | tzafrir_laptop: any idea where I'd be able to look to find out? it's a bit of a specialized area |
| 16:15.39 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 16:18.24 | idespinner | Qwell, for what its worth, with other PBX's for the Music/Message on hold, we've used the 'on hold plus' device successfully |
| 16:20.53 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:21.16 | wcselby | something like www.please-hold.net I guess. looking for more options than just one. i'll keep looking |
| 16:21.58 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:26.41 | *** join/#asterisk [1]sassyn (~sassyn@bzq-82-80-242-222.cablep.bezeqint.net) |
| 16:26.44 | [1]sassyn | hi all |
| 16:26.49 | [1]sassyn | can someone please help |
| 16:27.10 | [1]sassyn | It seems that my asterisk disconnect user after 16min of call |
| 16:27.40 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 16:28.12 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:30.57 | Chainsaw | [1]sassyn: Please issue core set verbose 10 & core set debug 10. |
| 16:31.09 | [1]sassyn | Chainsaw, and then what? |
| 16:31.13 | edwin_quijada | Asterisk::FastAGi can run in Windows? |
| 16:31.13 | Chainsaw | [1]sassyn: And pastebin us a trace of when it disconnects. Do not redact it, just insert a ==trouble starts here== marker. |
| 16:31.21 | Chainsaw | [1]sassyn: And then I will tell you why. |
| 16:31.31 | Chainsaw | [1]sassyn: Along with suggestions on what to do about it. |
| 16:31.41 | [1]sassyn | Coool |
| 16:31.44 | [1]sassyn | Give me a sec |
| 16:32.24 | [1]sassyn | the only problem is that I need to make 16 min call |
| 16:33.01 | [1]sassyn | Chainsaw, can u tell me maybe where to look in the conf files? |
| 16:33.21 | Chainsaw | [1]sassyn: Not without a trace, no. I'd like to know what's wrong first. |
| 16:35.08 | [TK]D-Fender | edwin_quijada: FastAGI = sockect connection. Doesn't matter what language or platform |
| 16:36.17 | [1]sassyn | OK |
| 16:36.22 | [1]sassyn | Session is up |
| 16:36.30 | [1]sassyn | call is runing |
| 16:36.36 | [1]sassyn | let's see in 15 min |
| 16:36.38 | [1]sassyn | Chainsaw, Thank U |
| 16:42.14 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:43.52 | edwin_quijada | [TK]D-Fender: I read that just run in linux? so Can I run my scripts in windows>? |
| 16:48.15 | [TK]D-Fender | edwin_quijada: What part did you not understand? |
| 16:48.54 | [TK]D-Fender | edwin_quijada: FastAGI works over a network socket. It doe not matter what is on the other side as long as it talks on that socket |
| 16:50.43 | *** join/#asterisk toddejohnson (~toddejohn@ppp-70-226-210-28.dsl.spfdil.ameritech.net) |
| 16:55.32 | [1]sassyn | Chainsaw, Well I found out that it is disconnect only when talking to long distant call |
| 16:56.10 | Chainsaw | [1]sassyn: And how are you terminating that call? ISDN? Analog? |
| 16:56.57 | [1]sassyn | No it is a SIP TRUNK |
| 16:58.53 | Chainsaw | [1]sassyn: Okay. Then I will need the call trace that I requested a while ago. |
| 16:59.10 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 16:59.18 | [1]sassyn | Chainsaw, well it is on call to outside |
| 16:59.27 | [1]sassyn | 5:40 now for long disctant |
| 17:01.19 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:06.24 | Naikrovek | never once have i ever seen anyone misspell "distance" in that way |
| 17:07.10 | p3nguin | nugget: http://tinyurl.com/y4e9hp9 |
| 17:19.03 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 17:19.31 | Nugget | eyes p3nguin |
| 17:21.55 | [1]sassyn | Chainsaw, |
| 17:22.27 | [1]sassyn | Here is it http://pastebin.com/RX7haKtm |
| 17:24.00 | [1]sassyn | Chainsaw, any idea |
| 17:24.35 | Chainsaw | [1]sassyn: Remember what I said about not redacting the output? |
| 17:24.41 | Chainsaw | [1]sassyn: Please take it to heart. |
| 17:25.14 | [1]sassyn | Chainsaw, what do u mean I didn't understand |
| 17:25.29 | Chainsaw | [1]sassyn: You will give me the log, the whole log and nothing but the log. |
| 17:25.48 | [1]sassyn | Really destroying SIP dialog '6e0238e630fe65e01511cbc93972f306@172.16.0.17' Method: OPTIONS |
| 17:25.49 | [1]sassyn | Really destroying SIP dialog '5e28270e18d74f18055ee53f0457c3c3@172.16.0.17' Method: OPTIONS |
| 17:25.49 | [1]sassyn | Really destroying SIP dialog '11efbe16249d0cf663adda63324fe293@172.16.0.17' Method: OPTIONS |
| 17:25.49 | [1]sassyn | Really destroying SIP dialog '1a8e05a27357b5d70f788c0507566cd5@172.16.0.17' Method: OPTIONS |
| 17:25.49 | [1]sassyn | Really destroying SIP dialog '72d1f54a1bc45f641d89900f4aef0e0a@172.16.0.17' Method: OPTIONS |
| 17:25.49 | [1]sassyn | Really destroying SIP dialog '09f9ab3b6a27112a534fd4dc36a15873@172.16.0.17' Method: OPTIONS |
| 17:25.50 | [1]sassyn | Really destroying SIP dialog '176dedec3aee8204255c1b75777af435@172.16.0.17' Method: OPTIONS |
| 17:25.50 | [1]sassyn | Really destroying SIP dialog '3d1f4b145159d1122ab4e8717d12f922@172.16.0.17' Method: OPTIONS |
| 17:25.51 | [1]sassyn | Really destroying SIP dialog '42b02a4916b58ad81722a72b777d6027@172.16.0.17' Method: OPTIONS |
| 17:25.51 | [1]sassyn | Really destroying SIP dialog '407915cd239d229e53acb3d53454f04c@172.16.0.17' Method: OPTIONS |
| 17:25.53 | [1]sassyn | hoooooppppppppppppps |
| 17:25.55 | [1]sassyn | sorry |
| 17:26.16 | Chainsaw | Please don't do that. It's very bad for Fender's blood pressure. |
| 17:27.36 | [1]sassyn | I'm sorry it was by mistake |
| 17:27.40 | [1]sassyn | Sorry again |
| 17:27.50 | [1]sassyn | Any way I have alog in var/log/asterisk |
| 17:29.19 | [1]sassyn | Maybe this can help? http://pastebin.com/DBekkrEP |
| 17:31.13 | [1]sassyn | any idea? |
| 17:33.47 | Chainsaw | [1]sassyn: Still missing the beginning... |
| 17:35.08 | toddejohnson | I'm having a horrible time with incoming calls contexts. http://pastebin.com/kT3MtWjx is the sip config and it is calling from-trunk-sip-BW-SIP-A instead of from-bandwith. |
| 17:35.33 | toddejohnson | This just started after the upgrade to 1.6.2. Any ideas how to get this fixed? |
| 17:36.12 | [TK]D-Fender | toddejohnson: those are not contexts. Next that only shows part of your config and not an actual call attempt that failed |
| 17:36.24 | [TK]D-Fender | toddejohnson: enable SIP DEBUG at * CLI and pastebin a complete failed attempt |
| 17:37.21 | Naikrovek | [TK]D-Fender: you should hotkey that request, so you don't have to type it each time |
| 17:37.37 | [TK]D-Fender | Naikrovek: Nah... |
| 17:37.59 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 17:38.08 | elliot98 | slips in |
| 17:38.25 | Naikrovek | elliot98: not unnoticed. |
| 17:38.34 | toddejohnson | [TK]D-Fender: the call rings through. BW-SIP-A I think is outgoing and from-bandwith-A was supposed to be incoming i thought. It's somewhat freepbxized. |
| 17:38.51 | Naikrovek | then #freepbx is where you wanna ask |
| 17:39.24 | elliot98 | hiya all! |
| 17:39.35 | [TK]D-Fender | toddejohnson: "somewhat"? What's that supposed to mean? |
| 17:39.41 | elliot98 | anyone here have much experience with fail2ban? |
| 17:40.28 | *** join/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 17:40.28 | kazaa_lite | hi all |
| 17:40.28 | kazaa_lite | why do I get: chan_iax2.c:3979 __auto_congest: Auto-congesting call due to slow response? |
| 17:40.29 | kazaa_lite | do i need to set something like sessions per second? |
| 17:40.35 | kazaa_lite | how can i configure asterisk to handle more number of sessions per second? |
| 17:41.11 | toddejohnson | [TK]D-Fender: It is freepbx, but It was just working with asterisk 1.6.0. |
| 17:42.09 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 17:43.05 | [TK]D-Fender | toddejohnson: "Was"? If you're not on 1.60, what are you on? You need to be more clear about your circumstances. Also you haev 2 peers with the same host, etc.... seriously this looks like bad duplication, and you haven't included a failed call yet |
| 17:45.37 | toddejohnson | [TK]D-Fender: I'm on asterisk 1.6.2 now I upgraded from 1.6.0 and that is when my incoming calls stoped working properly. They still rang through but stopped removing the +1 from the did and callerid. The +1 removal is in from-bandwith. |
| 17:46.33 | [TK]D-Fender | toddejohnson: If the call is coming in then your inbound routes are bad. |
| 17:46.51 | [TK]D-Fender | toddejohnson: take this up in #freepbx |
| 17:46.55 | [TK]D-Fender | ~freepbx |
| 17:46.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:48.20 | kazaa_lite | how can i configure asterisk to handle more sessions per second? |
| 17:48.37 | Qwell | kazaa_lite: buy better hardware |
| 17:48.55 | leifmadsen | asterisk does not have any imposed limit |
| 17:49.24 | toddejohnson | [TK]D-Fender: I currently don't understand how you can have 2 sip trunks one for incoming and one for outgoing and how to specifiy which is which. I though it had something to do with type=peer. |
| 17:50.00 | toddejohnson | [TK]D-Fender: Like you said it looks duplicated. I think that is part of the problem. |
| 17:50.22 | kazaa_lite | if saying freeswitch is not crime here, then there is some equivallent to 'fsctl sps' in asterisk as well |
| 17:50.29 | kazaa_lite | i cannot recall:( |
| 17:51.55 | Qwell | what are you expecting such a command to do? |
| 17:52.27 | *** join/#asterisk [1]sassyn (~sassyn@bzq-82-80-242-222.cablep.bezeqint.net) |
| 17:52.34 | kazaa_lite | i want to increase default number of calls or sessions asterisk can handle per second |
| 17:52.44 | Qwell | <leifmadsen> asterisk does not have any imposed limit |
| 17:52.56 | [1]sassyn | Chainsaw, sorry i got disconnected I'm doing this again |
| 17:53.06 | kazaa_lite | ok |
| 17:53.52 | leifmadsen | increase your hardware capacity, or have your system run multiple asterisk processes and spread the calls out amongst them |
| 17:54.09 | leifmadsen | the latter part of my sentence is left as an exercise for the reader |
| 17:54.40 | oldhack | I am putting ASTLINUX on an Alix 2d3+b and am wondering what size CF card I should use? |
| 17:54.59 | Qwell | oldhack: whatever you think is necessary |
| 17:55.10 | Qwell | we don't know anything about your requirements |
| 17:55.33 | oldhack | just a home ast box |
| 17:56.03 | oldhack | Qwell: thx |
| 17:56.18 | [1]sassyn | Chainsaw, http://pastie.org/919742 much better now |
| 17:56.38 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.230.8.dsl.dyn.forthnet.gr) |
| 17:57.07 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 17:57.21 | Chainsaw | [1]sassyn: [Apr 14 20:55:02] NOTICE[27256] chan_sip.c: Disconnecting call 'SIP/012-SIP-Out-00000057' for lack of RTP activity in 31 seconds |
| 17:57.30 | Chainsaw | [1]sassyn: The other side has stopped sending audio. |
| 17:58.09 | [1]sassyn | So it is my problem or the SIP trunk Problem? |
| 17:58.25 | [1]sassyn | Chainsaw, What do u think? |
| 17:58.38 | [1]sassyn | Chainsaw, cause this happend every 15 min |
| 17:58.42 | [1]sassyn | all the time |
| 17:58.46 | Chainsaw | [1]sassyn: I would say it is the problem of whatever device/provider you're talking to or the router you're going through. |
| 17:59.02 | Qwell | more likely the latter |
| 17:59.07 | leifmadsen | [1]sassyn: ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity |
| 17:59.07 | leifmadsen | <PROTECTED> |
| 17:59.15 | leifmadsen | from sip.conf.sample |
| 17:59.29 | Qwell | leifmadsen: yeah but if he isn't getting RTP anymore, there's some other reason for it |
| 17:59.37 | leifmadsen | true dat |
| 17:59.43 | [TK]D-Fender | qwlMal-notified hold herhaps |
| 17:59.49 | leifmadsen | maybe he is behind nat and not responding to a message they expect |
| 17:59.56 | [TK]D-Fender | Qwell: Mal-notified hold perhaps |
| 18:00.30 | [1]sassyn | OK |
| 18:00.35 | [1]sassyn | Let me check |
| 18:00.47 | [1]sassyn | rtptimeout=30 |
| 18:00.50 | Qwell | sounds like a portmap timeout to me |
| 18:01.03 | [1]sassyn | in my conf |
| 18:01.26 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:03.05 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:04.26 | *** join/#asterisk mlr78 (~mratliff@phoenix.warren-wilson.edu) |
| 18:05.47 | mlr78 | hey guys! I've not done this in the past, but I need a way of differentiating between a call coming from the outside world and a call coming from inside |
| 18:06.01 | mlr78 | I'd like to set a double ring for outside calls and a single for inside |
| 18:06.55 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:07.16 | mlr78 | this would be calls coming into my phone, not out. when a call is made from someone's cell for example to my asterisk phone, I'd like my phone to do a double ring |
| 18:12.53 | *** part/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 18:14.11 | p3nguin | mlr78: That isn't typically something asterisk is going to handle. |
| 18:14.38 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:15.24 | wcselby | an e1 is 30 concurrent channels, right? |
| 18:15.29 | wcselby | a voice e1* |
| 18:16.32 | [TK]D-Fender | mlr78: What phone are you using? |
| 18:16.53 | mlr78 | linksys spa942 and 946 |
| 18:16.57 | [TK]D-Fender | wcselby: 31 channel in CAS (very rare to find), 30 in PRI |
| 18:17.16 | [TK]D-Fender | mlr78: Those phones support distintive ringing via setting a SIP header before yuo call it |
| 18:17.47 | TSM | [TK]D-Fender: every heard of DASS2? |
| 18:17.48 | [TK]D-Fender | mlr78: Very well documented on the WIKI. Go read up on "linksys auto-answer" there (same method fo auto-answer as for setting other ringers) |
| 18:17.58 | [TK]D-Fender | TSM: Nope... thats a new one for me |
| 18:18.01 | TheCompWiz | wcselby: depends on how you configure it. 30 b-channels max... 1 d-channel (pri) or 31 channels with e&m wink & such. |
| 18:18.11 | mlr78 | perfect thanks! |
| 18:19.10 | p3nguin | [tk]d-fender: Cisco 7940/7960 doesn't support distinctive ring on SIP, right? |
| 18:19.19 | [TK]D-Fender | p3nguin: Not sure.... |
| 18:20.29 | TSM | [TK]D-Fender: prolly does not matter as its now defunct but some people in the UK still have it as BT did not force migration only ceased new installs, its a precurser to EUROISDN E1 signaling, electricly compatable though |
| 18:23.04 | TSM | [TK]D-Fender: some systems had it supplied via CAT5 from the fiber termination point (mabey 3 because of how old the original spec was) some also had it via BNC from the fiber term point, (NTE in the UK( |
| 18:23.33 | TSM | [TK]D-Fender: anyway useless info nowerdays but never know when someone will ask the question |
| 18:24.14 | [TK]D-Fender | tmsdefunct UK standards? Not on my priority list :) |
| 18:24.19 | [TK]D-Fender | TSM: defunct UK standards? Not on my priority list :) |
| 18:27.04 | kerframil | p3nguin: apparently, you may set the Alert-info header |
| 18:27.08 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:27.40 | *** join/#asterisk websae (~IceChat7@207-118-145-17.dyn.centurytel.net) |
| 18:27.43 | kerframil | p3nguin: "If the INVITE message from the server contains an Alert-Info header, distinctive ringing is invoked. The format of the header is "Alert-info: x." The value of "x" can be any number. This header is received only by the phone and is not generated by the phone." |
| 18:28.20 | websae | anyone have billing solution on top of asterisk they want to sell or license? |
| 18:30.27 | p3nguin | kerframil: I'll see if I can make that do anything. |
| 18:30.38 | kerframil | p3nguin: I can't test it as I use sccp |
| 18:30.57 | p3nguin | I tried SCCP and it sucked. I had to go back to SIP. |
| 18:31.14 | kerframil | p3nguin: it only sucks if you use chan_skinny ;) |
| 18:31.21 | p3nguin | I'm sure SCCP is great using Call Manager, but it wasn't pleasant with Asterisk. |
| 18:31.37 | p3nguin | I used chan_skinny. What do you use? |
| 18:31.51 | kerframil | p3nguin: Federico Santulli's branch of chan-sccp |
| 18:32.59 | luke-jr | is anyone else getting random SIP calls from Chinese IPs that try to dial some random number and don't talk when you answer? :/ |
| 18:34.07 | *** part/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 18:35.05 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:37.05 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 18:37.11 | p3nguin | kerframil: "SIPAddHeader(Alert-Info: 2)" works. |
| 18:37.28 | dajhorn | luke-jr: They are looking for open DISAs. |
| 18:37.48 | kerframil | p3nguin: jolly good |
| 18:38.03 | luke-jr | dajhorn: what's the usual way to block them? :/ |
| 18:38.06 | TheCompWiz | is tempted to build a honey-pot trap for some o' those. |
| 18:38.10 | p3nguin | Now if I can just make the distinctive ring different... |
| 18:38.15 | [TK]D-Fender | luke-jr: fail2ban |
| 18:38.17 | luke-jr | TheCompWiz: RBL? :) |
| 18:38.33 | luke-jr | [TK]D-Fender: if it was failing, I wouldn't care... |
| 18:38.34 | TheCompWiz | RBL? no... just leave them in limbo & keep letting them *think* they're close. |
| 18:38.56 | dajhorn | luke-jr: I don't publish SIP URLs, so I disallowed anonymous inbound sip sessions. |
| 18:38.56 | bmoraca_work | luke-jr: don't accept SIP calls from unauthenticated peers. |
| 18:39.24 | luke-jr | what's the point of SIP if not for the anonymous calls? :P |
| 18:39.28 | [TK]D-Fender | luke-jr: No... * logs the failed attempts. Fail2ban is a log scanner to detect hack attempts and auo-blocks them via iptables, etc |
| 18:39.29 | wcselby | bmoraca_work - that's step one. step two is setup fail2ban (or something similiar) that blocks failed sip reg attempts |
| 18:39.29 | jstapleton | did the IMAP integration against Exchange 2000 per "Asterisk: The Future of Telephony". Now, when I call voicemail, I get "[Apr 14 14:18:19] DEBUG[13240]: app_voicemail.c:5225 leave_voicemail: Before find_user |
| 18:39.29 | jstapleton | [Apr 14 14:18:19] WARNING[13240]: app_voicemail.c:5227 leave_voicemail: No entry in voicemail config file for '1601' |
| 18:39.40 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.35.181) |
| 18:39.44 | luke-jr | [TK]D-Fender: there's no failure |
| 18:39.56 | [TK]D-Fender | luke-jr: Yuo mean they ARE authing and getting through? |
| 18:40.02 | luke-jr | [TK]D-Fender: they're anonymous |
| 18:40.18 | jstapleton | But the extension is there. What is the best way to troubleshoot this IMAP stuff? I am told that there are no logs because Asterisk is talking directly to Exchange server... |
| 18:40.41 | p3nguin | <PROTECTED> |
| 18:40.42 | bmoraca_work | luke-jr: send unauthed sip calls to a context that doesn't allow them to dial to the PSTN? |
| 18:40.43 | wcselby | TheCompWiz - i've been thinking of setting up something similiar (honeypot with weak sip accounts and no outbound routes), just to see what happens |
| 18:40.49 | p3nguin | Should be clear what's wrong. |
| 18:40.55 | luke-jr | bmoraca_work: they're not hitting the PSTN, they're hitting my house phones -.- |
| 18:41.22 | [TK]D-Fender | luke-jr: Any special rason you're allowing anonymous calls? |
| 18:41.28 | luke-jr | [TK]D-Fender: so people can call me? |
| 18:41.33 | bmoraca_work | i'm not sure why anyone would open unauthenticated SIP calls. sounds like a terribly stupid idea. |
| 18:41.37 | dajhorn | luke-jr: No different than any other spam. Perhaps you could use something like the geoip package and block all of China. |
| 18:41.55 | p3nguin | It's pretty hard to get calls via SIP URI if you don't allow anonymous calls. |
| 18:41.58 | *** join/#asterisk xAvia (~matt@cpe-69-204-26-134.buffalo.res.rr.com) |
| 18:42.12 | p3nguin | Configure the contexts and dialplan accordingly, and there should be no problem. |
| 18:42.15 | wcselby | luke-jr - do you not have any connection with a sip provider or something? why would you need to allow anonymous calls to receive calls? |
| 18:42.15 | [TK]D-Fender | luke-jr: Ok, if you have generic SIP like that.... I suppose you could build in your own protection by breaking down the SIP call to check for IP's / count times, etc. |
| 18:42.36 | [TK]D-Fender | wcselby: Its rare but some people do allow that. |
| 18:42.40 | luke-jr | [TK]D-Fender: they're using multiple IPs |
| 18:42.50 | [TK]D-Fender | luke-jr: A lot? |
| 18:42.51 | luke-jr | wcselby: e164 & SRV records, at least |
| 18:42.56 | kerframil | luke-jr: http://ipdeny.com/ipblocks/data/countries/cn.zone |
| 18:42.57 | TheCompWiz | you could black-hole all of china ... |
| 18:43.02 | luke-jr | [TK]D-Fender: I'm setup iptables to block two class Bs |
| 18:43.10 | p3nguin | Not everyone wants to accept calls ONLY via DID. |
| 18:43.26 | wcselby | yeah |
| 18:43.44 | [TK]D-Fender | luke-jr: I mean how many different sources have yuo tracked so far? |
| 18:43.49 | Katty | peeks in |
| 18:44.03 | luke-jr | [TK]D-Fender: at least 10 unique IPs |
| 18:45.18 | *** join/#asterisk jsmith (~jsmith@asterisk/training-and-documentation-guru/jsmith) |
| 18:45.18 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:45.18 | [TK]D-Fender | luke-jr: Well as I said you could make your own little logging method even in pure dialplan to create a ban rule. |
| 18:47.36 | *** join/#asterisk epixcore (~irc@81.217.154.57) |
| 18:47.39 | *** join/#asterisk theron (~Theron@67-133-88-65.dia.static.qwest.net) |
| 18:47.57 | epixcore | ich hab nen problem mit asterisk (saydigits) kann mir mal eben wer helfen bitte? |
| 18:50.29 | TheCompWiz | yet another reason not to allow anonymous sip. |
| 18:52.04 | [TK]D-Fender | TheCompWiz: Required for E.164 and other things. He wants it and the use is valid... just needs to be controlled a bit |
| 18:52.31 | TheCompWiz | indeed... but I still won't :D |
| 18:52.53 | TheCompWiz | if you make free calling from anywhere... people will abuse it & try to make $$$ |
| 18:53.02 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:53.16 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 18:53.51 | TheCompWiz | hell... look at email. 300k+ emails blocked since Monday. and 0 were actually from a REAL person. |
| 18:54.14 | TheCompWiz | and that's just for 1 domain. |
| 18:55.40 | epixcore | Hello, when i let my asterisk server call my mobile phone, and asterisk says some digits with SayDigit() and i hangup on my mobile phone the saydigits command still runs but i need to go forward in the dial plan, any ideas? |
| 18:56.02 | TheCompWiz | it'd be great if someone could setup a token-based system for e.164... i.e. you look it up... you get a special token valid for a period of time... which would allow you to make an anonymous sip call.... without it... you get nada. |
| 18:56.46 | TheCompWiz | if you hangup... the pbx should hangup too. |
| 18:57.29 | epixcore | >TheCompwiz: but he does it not! it executes the SayDigits() command for every digit and does not hangup |
| 18:58.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:01.51 | *** join/#asterisk lallous (~lallous@213.204.73.95) |
| 19:04.14 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 19:04.30 | [TK]D-Fender | epixcore: What are you using to call your cell? |
| 19:05.28 | epixcore | >[TK]D-Fender: i am using the SIP Protocol, the calls were startet from Call files |
| 19:05.37 | voxter | man, someone needs to patch the g729 PLC logger to report the channel that it detected packet loss on.. otherwise is somewhat of a useless message. |
| 19:05.47 | [TK]D-Fender | epixcore: Does * get a disconnect message from your carrier? |
| 19:07.06 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 19:07.25 | epixcore | >[TK]D-Fender: seems not so |
| 19:08.21 | *** join/#asterisk rare1980_ (~rare1980@115.186.3.193) |
| 19:08.45 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 19:08.57 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:09.38 | [TK]D-Fender | epixcore: then tell your carrier to fix their signalling. * is doing its job. Make them do theirs |
| 19:10.49 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 19:10.50 | Katty | hi |
| 19:11.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:12.07 | *** join/#asterisk TimeRider (~steve@82.132.139.34) |
| 19:12.10 | drfreeze | Can someone tell me the point of using option 'r' on incoming calls on a PRI? |
| 19:12.40 | drfreeze | Is this a feature that is always needed, or just if a ring is not provided |
| 19:13.06 | mlr78 | SIPAddHeader was what I needed...thanks [TK]D-Fender |
| 19:13.39 | rare1980_ | hi all from where i can find out asterisl cli logs??? |
| 19:14.01 | bmoraca_work | whatever you set in /etc/asterisk/asterisk.conf |
| 19:14.09 | drfreeze | I turned this feature off for outgoing calls because I had a case where with the 'r' the call just rang forever, but without the 'r' the automated service would answer |
| 19:14.13 | rare1980_ | thanks let me check |
| 19:17.26 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.36 | *** join/#asterisk synthetiq (~User@208.90.33.43) |
| 19:18.07 | synthetiq | anyone use sipp and know how to have the program end if its still running after x seconds via the command line? |
| 19:19.20 | [TK]D-Fender | ~pb |
| 19:19.21 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 19:19.24 | [TK]D-Fender | epixcore: ^^^^^^^^ |
| 19:27.47 | *** join/#asterisk nny (~Scott@64.203.239.83) |
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| 19:28.17 | *** part/#asterisk ebroad (~EB@72.11.213.195) |
| 19:29.30 | Naikrovek | katty |
| 19:29.36 | Naikrovek | you are afk lately |
| 19:31.08 | *** join/#asterisk kash (~kash@190.196.71.206) |
| 19:31.24 | kash | hi again ! somebody know how to convert wav to alaw ? |
| 19:31.30 | epixcore | http://pastebin.com/deb2dqHd - this is what my asterisk says ... |
| 19:33.40 | bmoraca_work | kash: a wav could be an alaw... |
| 19:33.47 | p3nguin | kash: "file convert infile.wav outfile.alaw" didn't work for you? |
| 19:34.28 | Naikrovek | sox |
| 19:34.32 | Naikrovek | oh maybe not alaw |
| 19:34.57 | kash | format_wav.c:131 check_header: Not in mono 2 |
| 19:35.11 | kash | this is the file convert message |
| 19:35.37 | p3nguin | Okay, so the wav file isn't valid to asterisk. It wants 8k mono 16-bit. |
| 19:36.27 | kash | i will try that .. thanks .. |
| 19:40.38 | kash | sox ever change all the stuff .. i m looking many places how to convert wav 44 in wav 8 ... but is not working because all is deprecated .. |
| 19:40.44 | kash | sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
| 19:40.59 | kash | -w and resample are deprecated .. |
| 19:41.04 | chazzam | try -2 instead |
| 19:41.47 | chazzam | well, I've got nothing for resample, but try -2 instead of -w |
| 19:45.50 | nny | has anyone had a user complain of "random beeping" almost like a DTMF (not from hitting any keys). Using sangoma card/ cisco SPA phones.... Have two systems, one with T1 card and one with 8 Port... both seem to be doing it |
| 19:46.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:47.53 | wcselby | nny - do you have CLI output from when someone hears the random tones? |
| 19:48.01 | wcselby | to see if an actual DTMF tone was detected by asterisk? |
| 19:48.19 | jsmith | nny: No, but a friend of mine reported it on an SPA ATA |
| 19:48.33 | wcselby | yeah i've had it on my spa |
| 19:48.35 | wcselby | erm |
| 19:48.39 | wcselby | pap2t |
| 19:49.01 | wcselby | mostly when my wife is talking |
| 19:49.20 | bmoraca_work | has anyone used a MySQL cluster as an ODBC datasource for asterisk? does it work fairly well? |
| 19:49.21 | wcselby | * actually sees a dtmf tone |
| 19:50.06 | jsmith | bmoraca_work: Sure... if you like MySQL... |
| 19:50.11 | Nugget | ew mysql |
| 19:50.30 | jsmith | bmoraca_work: If you know the current state of MySQL and it's owners, you might consider PostgreSQL instead :-p |
| 19:51.44 | bmoraca_work | does postgreSQL support synchronous replication? does it have easy-to-use GUI management tools (like phpmyadmin)? if not, then it doesn't fit the bill, regardless of the fact that Oracle now controls MySQL |
| 19:52.31 | jsmith | bmoraca_work: There are several different replication methods that can be used with PostgreSQL (Slony, Bucardo, Tungsten, just to name a few) |
| 19:52.48 | jsmith | bmoraca_work: And of course, there's phppgadmin if you must use a web gui |
| 19:52.56 | jsmith | is a command-line commando |
| 19:53.28 | bmoraca_work | i have better things to do than write SQL scripts to perform basic maintenance, so I'll take a GUI tyvm |
| 19:54.36 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:54.48 | nny | wcselby: no fraid not, trying to get it to happen and see it in CLI is proving difficult |
| 19:55.12 | nny | jsmith interesting, i am not sure where to even start looking |
| 19:56.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:57.15 | jsmith | nny: There's plenty of good information on the web.... I'd start at http://www.postgresql.org/docs/ if I were you |
| 19:59.03 | nny | think that was meant for bmoraca_work no? |
| 19:59.48 | jsmith | Oh yeah, sorry |
| 19:59.51 | Nugget | I prefer londiste for postgresql replication. can't recommend slony. |
| 20:00.05 | jsmith | Nugget: I don't like Slony either, but it's an option |
| 20:00.06 | wcselby | jsmith - you referring to this --> http://www.zacks.com/stock/news/32888/Oracle%92s+MySQL+Gets+a+Boost ? |
| 20:00.23 | wcselby | haven't touched pgsql in a long time |
| 20:00.41 | wcselby | haven't paid attention to the mysql stuff in a long time either, hadn't realized sun was bought by oracle. |
| 20:00.49 | wcselby | although I still use mysql |
| 20:00.54 | Nugget | postgresql is faster, safer, and free-er. I don't understand why people use mysql other than just dumb inertia |
| 20:00.56 | jsmith | wcselby: Naw... just the general state of affairs, where all the former core developers of MySQL have left, and all the exciting stuff isn't happening in MySQL any more |
| 20:01.18 | jsmith | (Drizzle, MariaDB, etc.) |
| 20:01.19 | wcselby | jsmith - ahh. interesting, interesting. |
| 20:01.45 | wcselby | now I'm going to have to learn all about pgsql... |
| 20:02.02 | wcselby | i almost got a mysql cert a coulple years ago for work, but in the end they decided I didn't need it. |
| 20:02.07 | jsmith | wcselby: Never hurts to learn more, especially about a really cool database :-) |
| 20:02.17 | jsmith | used to run one of the largest MySQL installations in the world |
| 20:05.07 | wcselby | cool |
| 20:13.55 | wcselby | anyone use pandora here? |
| 20:14.27 | Naikrovek | yeah |
| 20:14.36 | Naikrovek | the music site? |
| 20:14.39 | wcselby | yeah |
| 20:14.42 | Naikrovek | yeah |
| 20:14.59 | wcselby | can you only listen using their website and or their own app, or do you get like an m3u or pls file to play in your media player? |
| 20:15.12 | Naikrovek | wcselby: only through their site or their apps |
| 20:15.18 | wcselby | gotcha |
| 20:15.27 | Naikrovek | and free accounts only get 40 hrs per month i think |
| 20:15.37 | wcselby | yeah |
| 20:15.37 | Naikrovek | so be sure to pause when you leave the computer |
| 20:15.39 | wcselby | i've heard that |
| 20:16.24 | wcselby | was just trying to see if there was a way I could stream a pandora station into asterisk musiconhold |
| 20:16.53 | Naikrovek | that would probably be a violation of pandora's terms of use, but the old analog hole will serve you well here if you're bent on it |
| 20:16.56 | jsmith | Legally, no. |
| 20:17.02 | jsmith | Technically, it's not worth your time |
| 20:17.02 | wcselby | nah, not bent |
| 20:17.05 | wcselby | just exploring options |
| 20:17.12 | wcselby | picky client |
| 20:17.13 | Naikrovek | wikipedia has a huge list of free music |
| 20:17.17 | Naikrovek | well |
| 20:17.23 | wcselby | lol |
| 20:17.41 | Naikrovek | if your client has rebroadcast rights they can put anything they're licensed to rebroadcast in for hold music |
| 20:17.45 | Naikrovek | for example |
| 20:17.54 | jsmith | wcselby: Tell the client that legally, you have to spend a bunch of money with ASCAP (or your respective music licensing agency) to be able to stream commercial music like that |
| 20:17.55 | Naikrovek | if they buy muzak for phone systems, they can use that legally |
| 20:18.23 | Naikrovek | there are products that you can buy that give you rights to rebroadcast over phones. muzak is just one of them |
| 20:18.37 | wcselby | figured "hey, if I can get them to tell me some songs they actually WANT, I can create a pandora station for them". instead i get told "some easy listening music, from like the 60's to 80's, or something" |
| 20:18.50 | *** join/#asterisk moy (~chatzilla@74.12.121.97) |
| 20:19.07 | wcselby | good to know jsmith |
| 20:19.23 | Naikrovek | i think $5000/yr will get them anything and everything for phones |
| 20:19.30 | wcselby | heh |
| 20:19.38 | Naikrovek | for some people that's cheap for others, that's insanity |
| 20:19.43 | wcselby | i'm sure it would |
| 20:20.06 | wcselby | these people didn't like the idea of paying 300 for a custom music on hold from a professional music on hold service |
| 20:20.13 | Naikrovek | hah |
| 20:20.14 | wcselby | with voice messages, etc. |
| 20:20.22 | Naikrovek | love those something-for-nothing people |
| 20:20.25 | wcselby | i think maybe if I couch it the right way, they may. |
| 20:21.05 | wcselby | "well, here are your options - 300 fior custom hold music, 5000 for rebroadcast rights for commerical music, or xxx legal fees for breaking the law" |
| 20:21.16 | wcselby | lol |
| 20:21.25 | Naikrovek | yeah |
| 20:21.28 | wcselby | just looking at options at this point really |
| 20:21.36 | Naikrovek | they do NOT f**k around with piracy |
| 20:21.53 | wcselby | i've already told them I wouldn't just rip some mp3's from their cd's |
| 20:22.13 | joako | Is there a place I can buy a list of valid phone numbers? |
| 20:22.28 | Naikrovek | joako: you can lease them usually for $1/month each |
| 20:22.39 | Naikrovek | in small quantities anyway |
| 20:22.41 | wcselby | joako - there are a lot of did providers out there |
| 20:22.47 | Naikrovek | if you want 10,000 numbers it's probably cheaper per number |
| 20:23.05 | wcselby | or is he talking phone numbers so he can call them? |
| 20:23.11 | joako | wcselby, I am aware. But what I would like to do is buy a list of valid numbers do I don't call disconneted numbers |
| 20:23.13 | Naikrovek | ooh bad |
| 20:23.17 | Naikrovek | spammer |
| 20:23.19 | wcselby | heh |
| 20:23.30 | wcselby | joako - i'm not sure you'll find that kind of support in this channel |
| 20:23.33 | wcselby | joako sorry |
| 20:23.34 | Naikrovek | yeah you won't |
| 20:23.45 | Naikrovek | but no one has those lists, that's what wardiallers are for |
| 20:23.58 | Naikrovek | the list would change every 5 minutes anyway |
| 20:24.18 | Naikrovek | also |
| 20:24.21 | joako | But the problem is when I dial sequentially and filter out the DNC many numbers are not connected, so the ASR is very poor |
| 20:24.42 | Naikrovek | joako: if you're not obeying the do not call list when you do that you're going to be in dark water, legally |
| 20:24.48 | patrb | joako: no sympathy |
| 20:24.51 | Naikrovek | none at all |
| 20:25.03 | wcselby | joako - i think you misunderstood me.... |
| 20:25.37 | Naikrovek | wcselby: http://www.bmi.com/licensing/entry/534929/pdf533669_1/ |
| 20:26.09 | Naikrovek | holy crap |
| 20:26.13 | Naikrovek | that's cheap |
| 20:26.36 | Naikrovek | number of trunk lines used to provide music on hold: 10. fee: $227 |
| 20:26.37 | Naikrovek | wow |
| 20:26.59 | wcselby | where'd you get that number Naikrovek? |
| 20:27.20 | wcselby | ahh I see it in the pdf |
| 20:28.09 | wcselby | so, for BMI's music, for this client, it'd be 791 per year |
| 20:28.19 | wcselby | how much for Sony's, Time Warner's, etc....? |
| 20:28.52 | jsmith | wcselby: That's why you go to someone like ASCAP |
| 20:29.03 | Naikrovek | do you really need EVERY song |
| 20:29.05 | jsmith | wcselby: They'll license you everything under the sun |
| 20:29.30 | wcselby | Naikrovek - i don't, but if we just pipe radio music into MOH, how can we determine which label the music is from? |
| 20:29.36 | wcselby | jsmith - that's what I'm looking for |
| 20:29.48 | Naikrovek | wcselby: they let you know what you can play and what you can't |
| 20:30.03 | Naikrovek | i couldn't find any way to obtain rebroadcast licenses at ascap |
| 20:30.44 | p3nguin | Don't you just pay them and they do the rest? |
| 20:30.46 | bmoraca_work | screw it, i'll use SQL Server. postgresql doesn't have a good synchronous cluster mechanism and MySQL's is a pain in the ass |
| 20:32.28 | *** join/#asterisk elzid_ (~IceChat7@5ace4173.bb.sky.com) |
| 20:33.12 | Naikrovek | bmoraca_work: postgresql 9 will have awesome synchronous clusters |
| 20:33.29 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 20:33.30 | bmoraca_work | that certainly doesn't help me now, though |
| 20:34.32 | TheCompWiz | anyone know what causes "ZOMBIE" calls? |
| 20:35.18 | Naikrovek | bmoraca_work: but it might influence your db decision now |
| 20:35.34 | Naikrovek | easier to upgrade from postgresql 8 to 9 than from mssql to postgres |
| 20:35.47 | bmoraca_work | mssql already supports the kind of clustering i need |
| 20:36.17 | Nugget | mssql is pretty solid and transact sql is nice |
| 20:36.49 | Naikrovek | not saying it isn't |
| 20:38.21 | Corydon76-dig | TheCompWiz: Transfers |
| 20:38.30 | TheCompWiz | wasn't a transfer tho |
| 20:38.58 | TheCompWiz | incoming call from dahdi... user answers... the call immediately hangs up. |
| 20:39.12 | Corydon76-dig | That's not a zombie call |
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| 20:39.40 | TheCompWiz | ok... then why does the log show the call becoming "SIP/1020001-000002a2<ZOMBIE>" |
| 20:39.58 | Corydon76-dig | TheCompWiz: Call pickup, perhaps? |
| 20:40.16 | TheCompWiz | not sure I understand... wha? |
| 20:40.28 | Corydon76-dig | Anything which does a masquerade will cause a zombie, though they should disappear almost immediately |
| 20:41.14 | Corydon76-dig | only reason a zombie wouldn't disappear is if the channel isn't getting serviced properly following the masquerade |
| 20:41.19 | TheCompWiz | ok... hm... |
| 20:45.42 | elzid_ | guys, need help with web-meetme again - I get "ast_writefile: No such format" at point of recording even though I'm setting recording file and format variables - the thing is when I use MeetMe(,rDMpc) in my dialplan, conference records but only because of the D parameter - but I cannot use this in the setup... |
| 20:46.09 | elzid_ | this is because it allows the caller to dynamically create conference rooms |
| 20:46.32 | elzid_ | I need to have conferences, pins pre-defined on the web interface by a host/admin |
| 20:46.37 | elzid_ | any help is appreciated guys |
| 20:50.06 | *** join/#asterisk moos3 (~rgenthner@rrcs-24-39-23-74.nys.biz.rr.com) |
| 20:50.17 | moos3 | can anyone help me with a hylafax issue |
| 20:52.53 | elzid_ | any ideas where the format is read from - clearly not from the MEETME_RECORDINGFORMAT variable |
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| 20:57.57 | moos3 | can anyone help me with this error http://pastie.org/920115 ? |
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| 20:59.01 | WIMPy | moos3: Check te spelling of Dial. |
| 21:00.20 | elzid_ | some debug RE my problem guys: |
| 21:00.42 | elzid_ | http://pastebin.com/08kPyW8e |
| 21:00.49 | moos3 | WIMPy, new error http://pastie.org/920115 |
| 21:01.28 | elzid_ | http://pastebin.com/WV6ixgHP |
| 21:03.08 | elzid_ | my extensions: http://pastebin.com/EkqhP5Vt |
| 21:03.29 | WIMPy | moos3: I still read Dail and agree that the parameter syntax is not correct. |
| 21:04.16 | moos3 | WIMPy, reload teh paste |
| 21:04.28 | moos3 | it shouldn't have dail but dial |
| 21:04.36 | moos3 | it use to work |
| 21:04.55 | moos3 | i'm not sure why it stopped working or worked in the first place |
| 21:04.56 | WIMPy | It doesn't change |
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| 21:05.21 | moos3 | WIMPy, try now |
| 21:05.31 | moos3 | i have hylafax up and running |
| 21:06.00 | [TK]D-Fender | moos3: GOTO, not DIAL |
| 21:06.05 | moos3 | ok |
| 21:07.17 | kash | hi everybody .. i know that maybe is a stupid question .. but i read all the tutorial , and have something that i dont understand about the agents and queues .. is necesary a Sip account ? how the agent connects to the server ? |
| 21:07.22 | moos3 | http://pastie.org/920115 |
| 21:08.03 | [TK]D-Fender | moos3: read Goto's INSTRUCTIONS. |
| 21:08.18 | [TK]D-Fender | moos3: You don't seem to understand the parameters you have to pass. |
| 21:08.20 | moos3 | [TK]D-Fender, i have tried is with ,s,1 |
| 21:08.34 | [TK]D-Fender | moos3: What I see is wrong and clearly doesn't match the instructions. |
| 21:09.01 | [TK]D-Fender | moos3: And you aren't jumping to an "s" extension in there eather.. there IS NO "s" |
| 21:09.19 | [TK]D-Fender | moos3: You need to look at the patterns you CAN match where you are going. |
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| 21:10.32 | moos3 | k |
| 21:11.04 | wcselby | moos3 - Goto(fax-in,${EXTEN},1) |
| 21:11.20 | moos3 | k |
| 21:11.22 | moos3 | thanks |
| 21:12.44 | moos3 | now to figure out this bit 2010-04-14 17:12:08] WARNING[26572]: pbx.c:3753 __ast_pbx_run: Channel 'DAHDI/3-1' sent into invalid extension '7043' in context 'fax-in', but no invalid handler |
| 21:13.38 | moos3 | fixed |
| 21:16.32 | moos3 | is away: In search of the answers. |
| 21:18.45 | ChannelZ | ponders a $100 2.8Ghz P4 to replace is aging 600mhz P3 server |
| 21:20.18 | jblack | sounds worth it |
| 21:20.45 | jblack | unless that's just the CPU, in which case you may want to shop more |
| 21:21.19 | ChannelZ | heh no it's a whole Dell refurb, 512mb, 40gb HD which I will replace with one of many larger spares I have lying around |
| 21:27.45 | idespinner | ChannelZ, a new intel atom board with case, ram, and cpu will run around $200(1.6ghz dualcore) |
| 21:28.20 | drfreeze | Help!!! |
| 21:28.48 | drfreeze | -- have office where incoming calls cannot be heard -- just started a few minutes ago |
| 21:30.00 | ChannelZ | Can you be more vague? |
| 21:33.42 | drfreeze | ChannelZ: -- ....--..2...232..3s.sa ..2.2 -- is that better? |
| 21:34.41 | WIMPy | drfreeze: It's probably your nick. |
| 21:36.17 | drfreeze | WIMPy: I rebooted the router |
| 21:36.40 | drfreeze | rebooting the asterisk server now |
| 21:36.48 | drfreeze | I'll check the nic when it comes up |
| 21:37.22 | WIMPy | No your nick. I was joking. |
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| 21:43.26 | ChannelZ | LOL |
| 21:43.37 | nny | hmmph. Have an 8 port card, have the system setup to just test one line. When I dial in it hits my dialplan etc. When I try to dial out it just gives back congested/Unknown (0) |
| 21:44.30 | nny | ugh nm |
| 21:44.38 | ChannelZ | drfreeze: we're not clairvoyant - what are the "incoming calls"? VoIP? Analog DAHDI? Radio frequencies picked up by your fillings? What are the phones? Can the other side hear you? |
| 21:45.08 | idespinner | nny, probably a config issue maybe with you dahdi groups or dialplan |
| 21:45.08 | nny | shit I want to get calls with my fillings |
| 21:45.26 | drfreeze | ChannelZ: I've got 5 calls in the last few minutes. All of my PRI accounts are having problems |
| 21:45.35 | drfreeze | It appears to be the telco provider |
| 21:45.35 | nny | idespinner: yup, automagic dahdi_conf generated by sangoma card used group 0, we use 1 by default :\ |
| 21:45.41 | idespinner | ChannelZ, what channel driver does that? |
| 21:45.46 | idespinner | is it in trunk? |
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| 21:47.37 | ChannelZ | What, fillings? |
| 21:47.38 | idespinner | why oh why are originate commands so hard to track :/ |
| 21:47.43 | idespinner | ChannelZ, yes! |
| 21:47.44 | ChannelZ | chan_lucilleball.so |
| 21:47.58 | ruben23 | hi |
| 21:48.58 | Corydon76-dig | chan_crystalball.so |
| 21:49.34 | idespinner | save on costs by bypassing the costly PSTN network and route your calls over magic crystal balls? |
| 21:49.59 | ChannelZ | kinkt |
| 21:50.04 | ChannelZ | err kinky |
| 21:50.13 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 21:51.42 | Corydon76-dig | I route mine over shiny disco balls |
| 21:55.46 | Pan3D | dances |
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| 22:33.40 | nny | hmm |
| 22:34.51 | nny | i have a xfer to cellphone that does an "s,n,TrySystem(asterisk -rx "channel originate DAHDI/g1/${WHO:3} application Bridge ${BRIDGEPEER}")" it works great, but when the caller hangs up with the bridged channel it dumps the call back into "default" |
| 22:38.01 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:38.38 | p3nguin | What's the difference between System() and TrySystem()? |
| 22:39.12 | bmoraca_work | damnit |
| 22:39.18 | bmoraca_work | realtime only works with mysql? |
| 22:41.21 | nny | p3nguin: heh not sure, this thing was a bit of a hack to get the system to Originate a call, IIRC we had to use System because I couldn |
| 22:41.23 | Corydon76-dig | p3nguin: if the command within the System fails to execute for whatever reason, System will cause the dialplan to exit |
| 22:41.33 | nny | couldn't call the Originate and Bridge natively from the command line |
| 22:41.35 | Corydon76-dig | TrySystem will continue, no matter what |
| 22:41.52 | nny | er natively from the dialplan |
| 22:42.09 | Corydon76-dig | bmoraca_work: res_config_odbc |
| 22:42.46 | bmoraca_work | yeah, i got it...file doesn't exist without asterisk-addons, but asterisk addons doesn't list that in make menuselect...weird |
| 22:43.09 | bmoraca_work | or something |
| 22:43.33 | Corydon76-dig | For various reasons, I don't recommend mysql realtime for use in production systems |
| 22:44.04 | Corydon76-dig | Well, not without some major caveats |
| 22:45.12 | bmoraca_work | i don't have a problem using it for sip peers and moderate (non-pattern-based) dialplan |
| 22:49.17 | bmoraca_work | hrm |
| 22:49.19 | bmoraca_work | that's weird |
| 22:49.46 | bmoraca_work | tells me some of the columns don't exist in the table, but they surely do |
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| 23:26.30 | Kobaz | ~book |
| 23:26.31 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 23:30.36 | jblack | I took it apart. It can do, under my deks lamp, 3.8v @ 17mw |
| 23:30.59 | jblack | in simpler words, rejigger their circuit a bit, add a second battery, and you can hide an AVR in here. |
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| 23:32.01 | jblack | I've done (barely) 328s on 0.6ma |
| 23:32.41 | manxpower | jblack: on the wrong channel? |
| 23:33.01 | jblack | sorry. |
| 23:33.04 | Defraz | hah |
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| 23:41.06 | bmoraca_work | hrm |
| 23:42.24 | bmoraca_work | i think res_config_odbc is warning me incorrectly about collumns that don't exist, even though they do |
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