| 00:14.24 | *** join/#asterisk MaliutaLap (~biteme@202.124.73.77) |
| 00:20.59 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 00:21.17 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 00:28.55 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 00:30.45 | Ceeriael | Im having problems with insecure=invite,port |
| 00:31.33 | Ceeriael | it dosent seem to work... as i understand, it should "ignore" the first INVITE when beeing called on a trunk... but my asterisk tries to Authenticate instead of just picking up |
| 00:33.43 | p3nguin | What's the issue? |
| 00:34.58 | Ceeriael | i try to call my trunk (my PSTN number) |
| 00:35.10 | Ceeriael | so i get the INVITE SIP paket. |
| 00:35.24 | Ceeriael | so far so good... but my Asterisk now wants to Authenticate instead of just picking up. |
| 00:35.44 | Ceeriael | ive used the "insecure=invite,port" ... but it dosent seem to do as its supposted to do |
| 00:36.44 | p3nguin | I don't know if the order of the options matters, but it is typically written as port,invite. |
| 00:36.55 | Ceeriael | 2 sec, ill try |
| 00:37.02 | p3nguin | It's certainly worth changing to see if it makes a difference. |
| 00:37.28 | Ceeriael | yes |
| 00:37.33 | Ceeriael | 2 sec |
| 00:39.05 | Ceeriael | nope p3nguin didnt go |
| 00:39.13 | Ceeriael | ill pastebin my output... |
| 00:39.36 | p3nguin | I'd rather see the entire sip peer definition. |
| 00:39.42 | Ceeriael | sure. |
| 00:39.51 | Ceeriael | outgoing calls works.... |
| 00:42.45 | Ceeriael | http://pastebin.com/amT7NBS1 |
| 00:42.47 | Ceeriael | p3nguin ^ |
| 00:45.36 | p3nguin | ceeriael: I'd rather see you delete everything you've just shown be and replace it with this: http://pastebin.com/4g2TjHVz |
| 00:45.57 | p3nguin | Register statements go before the peers. And you don't need both a peer and a user... just a peer is enough. |
| 00:46.17 | p3nguin | If peer doesn't do the job, use type=friend instead. |
| 00:46.46 | Ceeriael | ill try and paste it. |
| 00:46.52 | Ceeriael | so this is just the PeerDetails ? |
| 00:47.05 | p3nguin | I have no clue what that even means. |
| 00:47.08 | Ceeriael | =) |
| 00:47.37 | p3nguin | If you're using something other than just Asterisk, you're asking parts of your questions in the wrong place. |
| 00:48.48 | Ceeriael | its trixbox, but ive narrowed the problem down to the Invite problem (according to the manual that is... :() |
| 00:48.56 | Ceeriael | ill try and simplify the config. |
| 00:49.00 | Ceeriael | as you requested. |
| 00:49.30 | p3nguin | Oh, you need a context, too. |
| 00:49.55 | p3nguin | Let me just show you a peer entry from my system. |
| 00:50.08 | Ceeriael | wont it default to context=incomming ? |
| 00:50.33 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 00:51.05 | p3nguin | http://pastebin.com/tFVrstkt |
| 00:52.59 | Ceeriael | i dont se any "INSECURE" setting ? |
| 00:53.12 | Ceeriael | do you authenticate when you get a call ? |
| 00:53.47 | p3nguin | Here's another example: http://pastebin.com/jhkBsTiN |
| 00:53.56 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 00:54.24 | p3nguin | All three of those entries accept calls inbound to Asterisk without issues. |
| 00:54.52 | Ceeriael | hmm. |
| 00:54.58 | Ceeriael | seems simple enough ;) |
| 00:55.18 | *** join/#asterisk wesphillips (~wphillips@adsl-75-53-137-6.dsl.hstntx.sbcglobal.net) |
| 00:55.57 | p3nguin | I guess I could debug it and see what happens. |
| 00:56.42 | Ceeriael | i just tried your first sugestion. |
| 00:56.48 | Ceeriael | it wont allow me to call out |
| 01:01.07 | p3nguin | When I take a call inbound from IPComms, I see the invite, then the call just goes. |
| 01:01.28 | Ceeriael | i think ive tracked down to a Bug: https://issues.asterisk.org/view.php?id=13644 |
| 01:01.44 | Ceeriael | so ill try the newest release of asterisk |
| 01:02.24 | Ceeriael | im atm installing in a virtual enviroment. |
| 01:02.49 | Ceeriael | thx for your efford :) though it was a far more devius problem :P |
| 01:03.33 | Ceeriael | hmmm im hungry... whats the time @ yout p3nguin ? |
| 01:03.43 | p3nguin | 1903 |
| 01:04.02 | Ceeriael | its 0203 here ;) |
| 01:04.06 | Ceeriael | im getting hungry again :P |
| 01:04.09 | p3nguin | It's past my supper time, but I haven't eaten my supper yet. |
| 01:04.16 | Ceeriael | its past my bedtime. |
| 01:04.18 | Ceeriael | <PROTECTED> |
| 01:04.27 | p3nguin | I should get busy cooking soon. |
| 01:04.33 | Ceeriael | =) |
| 01:04.36 | Ceeriael | thx for your time |
| 01:04.49 | p3nguin | Hope you get it worked out soon. |
| 01:08.01 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 01:08.21 | Ceeriael | me 2 =) |
| 01:08.44 | Ceeriael | i ought to go to bed soon... but ive just "buffed up" with coke and chocolate =) |
| 01:09.07 | *** join/#asterisk MaliutaLap (~biteme@202.124.73.159) |
| 01:10.08 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 01:16.59 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:30.15 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 01:32.36 | Ceeriael | p3nguin found the problem =) |
| 01:33.08 | Ceeriael | p3nguin you have to define "insecure=port,invite" in both Outgoing and Incomming section before it works. (somewhat of a bug...) |
| 01:33.27 | Ceeriael | http://forums.whirlpool.net.au/archive/1133054 |
| 01:33.27 | Ceeriael | under "The Solution" |
| 01:33.45 | Ceeriael | well im off. c ya |
| 01:34.48 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 01:48.31 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:58.17 | *** part/#asterisk wesphillips (~wphillips@adsl-75-53-137-6.dsl.hstntx.sbcglobal.net) |
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| 02:33.58 | phix | hi gang |
| 02:36.36 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 02:36.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:47.31 | *** join/#asterisk Krampus (krampus@ozzle.org) |
| 02:47.39 | Krampus | I'm having trouble getting inbound calls to "complete" in Trixbox. If I call in, the phone on the default route will ring, but when picked up, the recipient can't hear anything. On the calling side, it sounds like it's still ringing. Outbound seems to work fine. |
| 02:47.49 | Krampus | I've got a firewall, but i'm forwarding ports 5060-5080 and 10000-20000 to it (and a pcap dump looks like traffic is flowing on the correct ports when i call in) |
| 02:47.59 | Krampus | Anyone have any ideas as to something "easy" I may have missed? |
| 02:50.42 | pabelanger | Krampus: Your best to get support from #trixbox |
| 02:52.35 | Krampus | pabelanger: yeah, although i think this may be a problem with asterisk itself (or rather how i've configured it) |
| 02:52.58 | Krampus | pabelanger: or if the line is hard and fast among you guys, i'll go away. :P |
| 02:54.45 | pabelanger | we don't keep up with trixbox changes, so you'd get better response from them. It would also take extra time to parse through their configuration files. |
| 03:11.27 | Krampus | fair enough. :) |
| 03:11.30 | Krampus | thanks anyway though. |
| 03:11.34 | *** part/#asterisk Krampus (krampus@ozzle.org) |
| 03:18.24 | *** join/#asterisk Alagar (~avaithiya@193.220.61.145) |
| 03:27.04 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 03:33.32 | joobie | guys im having this problem with g729 - http://pastebin.com/f92f7zTE |
| 03:33.56 | joobie | line 36?looks interesting - not sure why it cant identify it |
| 03:34.07 | joobie | i did buy 10 g729 licenses and installed them ages ago.. was working fine all this time |
| 03:34.10 | joobie | any ideas? |
| 03:35.06 | p3nguin | Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 03:35.26 | joobie | ya |
| 03:36.08 | joobie | but i presumed it was not finding a codec there because above on line 36 it doesnt identify media type G.729 |
| 03:36.35 | joobie | no idea why it's not recognizing it all of a sudden.. just ran 'show g729' and i got "0/0 encoders/decoders of 20 licensed channels are currently in use" |
| 03:36.44 | joobie | so not maxing out g729 licenses.. |
| 03:38.04 | p3nguin | Do you have any control over that peer? |
| 03:38.18 | joobie | only as a client |
| 03:38.34 | joobie | it's a 3rd party peer we subscribe to |
| 03:38.36 | p3nguin | I don't know what that means. |
| 03:38.51 | p3nguin | If you have control over that peer, fix it. |
| 03:39.04 | p3nguin | If you don't, contact someone who does and ask them to fix it. |
| 03:39.29 | joobie | my phone -> asterisk -> sends out via peer -> goes to another phone i have control over |
| 03:39.40 | joobie | the "sends out via peer" is a 3rd party which we dont have access to - we just use them to route calls |
| 03:39.44 | joobie | but i have access to all other devices |
| 03:39.53 | p3nguin | That peer has no codecs being offered. |
| 03:40.12 | joobie | would that be my aterisk peer config? or the peer itself |
| 03:40.37 | p3nguin | Your system is offering damn near every codec imaginable. |
| 03:40.47 | joobie | heh |
| 03:41.05 | p3nguin | (which is poor configuration practice) |
| 03:41.46 | joobie | i havent looked into forcing a codec on an inbound call.. only on our friends/peers in sip.conf |
| 03:44.43 | p3nguin | Choose a codec that you want a certain peer to use every time. For example, ulaw. disallow=all allow=ulaw |
| 03:47.24 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 03:47.54 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 03:48.15 | Micc | anyone here know much about a2billing? |
| 03:49.53 | p3nguin | Of course that codec that you choose has to be supported on the peer system. In your particular case, that peer doesn't support ANY codecs, so no matter what you offer, the answer is still nothing. |
| 03:53.50 | joobie | p3nguin, im on the phoenw ith the peer now |
| 03:53.55 | joobie | they say they support ulaw and g729 |
| 03:54.12 | p3nguin | Let them know what you see in the SIP debug. |
| 03:54.26 | p3nguin | peer - audio=0x0 (nothing)/video=0x0 (nothing) |
| 04:07.25 | Holos | ugh.. for a DHADI group... which is sequential, descendind? g0 or G0? |
| 04:09.21 | joobie | thanks p3nguin |
| 04:09.25 | joobie | i basically got a lacky |
| 04:09.27 | joobie | he didnt kno squat |
| 04:09.43 | joobie | but they have a few outages at the moment that are not related to this according to the website - but i'll try later |
| 04:12.44 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 04:13.04 | p3nguin | Does that mean they are going to fix it? |
| 04:19.22 | ectospasm | Holos: G0 is descending, I think |
| 04:22.45 | ectospasm | Holos: IIRC, if you have g0 on a T1 PRI, Asterisk will start dialing on channel 1, but if you have G0, it will start with 23 and work its way down |
| 04:25.29 | joobie | p3nguin, no |
| 04:25.47 | p3nguin | That's not very nice of them. |
| 04:25.48 | joobie | the guy i spoke to was trying to explain to me how they support ulaw and g729, so it must be our configuration that's not using those codecs |
| 04:25.55 | joobie | he's a wank |
| 04:26.11 | joobie | we have a backup peer im routing via.. just going to wait for their outages to resolve and see hwo it goes |
| 04:26.13 | joobie | could be related |
| 04:26.49 | joobie | that "Found unknown media description format G.729 for ID 18" sounds interesitng though |
| 04:27.03 | joobie | it's kinda like there is some mention of g729 in the call, but it's not in a format recognized by asterisk |
| 04:27.13 | joobie | which could explain why it's coming back with no supoprted codecs |
| 04:27.23 | joobie | .. just a stab in the dark |
| 04:27.55 | p3nguin | If they support both ulaw and g729, if g729 was totally jacked up, that would leave ulaw... but that's not even in their negotiation. |
| 04:40.20 | *** join/#asterisk OrNix (~ornix@l49-131-248.cn.ru) |
| 04:45.04 | joobie | yer |
| 04:45.17 | joobie | i duno wat is going on.. used to work fine, but today their accounting system is playing up |
| 04:45.30 | joobie | just odd that both issues happened at the same time - so discounting it to be related |
| 04:50.36 | p3nguin | I think someone broke something. Good luck getting anyone to ever admit that, though. |
| 04:56.28 | jblack | checks to see how long is left on the g729 patent |
| 04:57.54 | jblack | 2016 supposedly. |
| 05:10.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 05:50.27 | ectospasm | joobie: be sure they haven't switched to G.729 Annex B (assuming you're using the Digium codec, that only supports G.729 Annex A) |
| 06:12.08 | *** join/#asterisk tyman_ (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
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| 06:20.40 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 06:26.14 | drmessano | 5 more years on G729, yay |
| 06:26.35 | drmessano | After Google buys Digium, they can buy the G729 patent and free it |
| 06:32.50 | *** join/#asterisk lea123 (~chatzilla@59.164.99.67) |
| 06:38.42 | *** join/#asterisk KaiTheWolf (~KaiTheWol@S0106002354f2a6cc.ed.shawcable.net) |
| 06:39.03 | KaiTheWolf | hey has anyone else had this error? chan_ooh323.h:53: fatal error: asterisk/rtp.h: No such file or directory |
| 06:39.14 | *** part/#asterisk lea123 (~chatzilla@59.164.99.67) |
| 06:40.13 | ectospasm | KaiTheWolf: sounds like you don't have the Asterisk headers installed |
| 06:40.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 06:40.46 | ectospasm | KaiTheWolf: are you using a binary distro of Asterisk (e.g. AsteriskNOW, Trixbox, Elastix, etc.)? |
| 06:41.08 | KaiTheWolf | Fresh compile. but totally new to 1.8. I'm used to 1.2 and the age. on a debian system |
| 06:41.55 | KaiTheWolf | using the tarballs from the site, nothing installed prior. brand new server. |
| 06:43.04 | ectospasm | hmmm, I dunno. I don't usually deal with h323 |
| 06:44.17 | KaiTheWolf | Is it something i can do without? ive never directly used it in my previous installs, not sure if its a behind the scenes thing |
| 06:44.24 | KaiTheWolf | its part of addons. |
| 06:45.48 | ectospasm | yeah, if you don't need to use chan_ooh323, you can ignore it, and set modules.conf to noload chan_ooh323.so |
| 06:46.13 | ectospasm | I'm not even sure if ooh323 is compatible with 1.8.0 |
| 06:47.53 | joako | Does anyone know what is the cause or how to resolve constant kernel: FXO PCI Master abort messages when DAHDI modules are loaded? |
| 06:48.46 | ectospasm | joako: what card? |
| 06:49.50 | KaiTheWolf | @ectospasm this is in a compile, how do i stop it from trying to compile ooh323? just delete the file? |
| 06:50.27 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 06:50.28 | ectospasm | KaiTheWolf: run menuselect and take chan_ooh323 out of the equation |
| 06:50.50 | joako | ectospasm: Digium X100P. I've found many postings about this error in the past week, but honestly no solutions |
| 06:51.20 | joako | I also have some clone card that's detected as Generic X101P that gives the same problems... |
| 06:51.49 | ectospasm | X100P is no longer a supported card (it hasn't been supported since I started working here). |
| 06:52.15 | ectospasm | joako: yeah, Digium doesn't maintain the X100P drivers anymore, so they're not guaranteed to work |
| 06:52.43 | KaiTheWolf | @ectospasm .. ahaha i feel like an idiot. EVERYTHING in addons, is now just in the 1.8 installer or scripts. half what i see in menuselect for addons, i saw in scripts, and the other half in menuselect for asterisk itself. seems theres no use for addons anymore |
| 06:52.55 | ectospasm | ...they're included in the DAHDI distribution for posterity, but good luck in getting those cards working |
| 06:53.07 | ectospasm | KaiTheWolf: yeah, the addons have been merged into the main tree |
| 06:53.28 | KaiTheWolf | @ectospasm That solves my problem XD |
| 06:54.53 | KaiTheWolf | @ectospasm thanks man |
| 06:55.00 | ectospasm | no problem |
| 06:55.32 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 06:59.36 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
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| 07:26.08 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:30.43 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 07:31.14 | nicola_pav | hello. how can i modify the called number in extensions.conf? |
| 07:32.10 | wdoekes2 | the called number is what you put in the Dial() app |
| 07:34.00 | nicola_pav | how can i add 00 for example? |
| 07:34.15 | nicola_pav | i want to put rules in a context |
| 07:34.23 | nicola_pav | in exnetnsions.conf |
| 07:34.28 | wdoekes2 | _X.,1,Dial(TECH/00${EXTEN}@device) |
| 07:35.01 | wdoekes2 | when you call 123, _X. matches and Dial(TECH/00123@device) is called |
| 07:35.21 | wdoekes2 | (oh, that should be TECH/device/00${EXTEN} for non-sip) |
| 07:35.54 | nicola_pav | TECH? |
| 07:38.26 | nicola_pav | wdoekes2: what if i want to cut a digit? |
| 07:38.47 | wdoekes2 | technology, see 'core show application dial' |
| 07:38.59 | nicola_pav | mind my ignorance |
| 07:39.11 | nicola_pav | what if i want to cut a digit? |
| 07:39.13 | wdoekes2 | substring handling is done like so: ${EXTEN:1:4} => start at the 2nd char and take 4 |
| 07:39.27 | wdoekes2 | 123456789 => 2345 |
| 07:39.44 | nicola_pav | counting starts from 0? |
| 07:39.49 | wdoekes2 | of course |
| 07:40.01 | nicola_pav | ok, thanks |
| 07:42.45 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
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| 07:49.36 | tzafrir | Holos, astribank_hexload is part of dahdi (tools). Maybe you didn't have libusb-devel installed? |
| 07:50.05 | *** join/#asterisk jegorvilnius (~jegor@194.126.252.241) |
| 07:50.30 | jegorvilnius | Hello I have a problem with fax detection in Asterisk 1.8 can anybody help me ? |
| 07:51.49 | ectospasm | jegorvilnius: what fax software are you using? |
| 07:52.27 | jegorvilnius | I am sending fax from Microsoft fax server to the pstn line |
| 07:53.37 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 07:54.27 | ectospasm | nope, not familiar with that |
| 07:55.18 | jegorvilnius | slaps madduck around a bit with a large trout |
| 07:56.20 | jegorvilnius | <PROTECTED> |
| 07:57.04 | ectospasm | jegorvilnius: what are you using for fax detection in Asterisk? |
| 07:57.23 | ectospasm | I'm only familiar with Fax for Asterisk |
| 07:57.59 | jegorvilnius | in chan_dahdi.conf i put the faxdetect=incoming |
| 07:58.17 | ectospasm | then what happens? |
| 07:58.25 | jegorvilnius | Executing [s@incoming:1] Answer("DAHDI/1-1", "") in new stack |
| 07:58.25 | jegorvilnius | <PROTECTED> |
| 07:58.25 | jegorvilnius | <PROTECTED> |
| 07:58.25 | jegorvilnius | <PROTECTED> |
| 07:58.25 | jegorvilnius | <PROTECTED> |
| 07:58.26 | jegorvilnius | <PROTECTED> |
| 07:58.29 | ectospasm | don't flood |
| 07:58.52 | ectospasm | is there a fax extenion in [incoming]? |
| 07:58.56 | jegorvilnius | yes |
| 07:59.07 | ectospasm | pastebin that context |
| 07:59.27 | jegorvilnius | [incoming] |
| 07:59.28 | jegorvilnius | exten => s,1,Answer() |
| 07:59.28 | jegorvilnius | exten => s,n,Wait(5) |
| 07:59.28 | jegorvilnius | exten => s,n,Dial(SIP/5555) |
| 07:59.28 | jegorvilnius | exten => s,n,Hangup |
| 07:59.28 | jegorvilnius | exten => fax,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%M%S)}-${CALLERIDNUM}) |
| 07:59.28 | jegorvilnius | exten => fax,n,ReceiveFAX(${FAXFILE}.tif) |
| 07:59.50 | ectospasm | like I said, don't flood. |
| 07:59.53 | ectospasm | ~pb |
| 07:59.53 | 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. |
| 08:00.08 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:00.39 | jegorvilnius | sorry I'm new to the irc |
| 08:02.36 | ectospasm | it's detecting the fax correctly as far as I can tell |
| 08:03.11 | jegorvilnius | Yes I know but i don't understang why it can't jump to fax exten |
| 08:03.15 | ectospasm | ...just when it jumps to fax,1, it's like everything in the fax extension doesn't exist |
| 08:03.50 | ectospasm | it's never executing the Set() or the ReceiveFax() |
| 08:04.49 | ectospasm | what does "dialplan show incoming" show in the Asterisk CLI? |
| 08:07.08 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:07.56 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:08.22 | madduck | thanks, i love it. |
| 08:08.37 | jegorvilnius | http://pastebin.com/fq1EpvGq |
| 08:11.50 | ectospasm | jegorvilnius: "core set verbose 10" "core set debug 10", and pastebin the entire CLI output of a failed fax detection attempt |
| 08:14.01 | jegorvilnius | I had already verbose and debug 10 and now it is showing the same output |
| 08:16.13 | jegorvilnius | <PROTECTED> |
| 08:18.33 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:20.37 | ectospasm | hmmm. As far as I can tell, something is hanging up the channel before it gets to the fax exten. You may want to try removing the explicit Answer() |
| 08:23.36 | *** join/#asterisk Alagar (~avaithiya@193.220.61.145) |
| 08:27.12 | jegorvilnius | without Answer() it is not detecting fax |
| 08:27.21 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:28.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 08:29.41 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:33.27 | ectospasm | jegorvilnius: it's something in your dialplan then, I'd have to see the entire context from extensions.conf in pastebin |
| 08:35.44 | *** join/#asterisk sourcode (~code@ppp-58-11-76-47.revip2.asianet.co.th) |
| 08:36.00 | jegorvilnius | http://pastebin.com/xrBWg9Ff here it is. I have copied all extensions.conf it it is not big |
| 08:36.26 | ChannelZ | What if you WaitExten(5) instead of Wait(5)? |
| 08:37.07 | ChannelZ | I don't know if fax detection unblocks the Wait |
| 08:37.50 | ectospasm | there is an error in your dialplan, but it shouldn't be affecting this: You've got two '1' priorities in the 's' extension... |
| 08:37.53 | ChannelZ | and actualyl what you just pasted is slightly messed up |
| 08:38.05 | ChannelZ | what ecto said |
| 08:38.59 | ectospasm | it shouldn't be affecting this because the dialplan execution should short circuit to fax |
| 08:39.02 | jegorvilnius | yes i know about priorities i have tried different configuration |
| 08:39.29 | jegorvilnius | it is not affecting because asterisk verbose is showing the same |
| 08:39.50 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 08:40.09 | ectospasm | jegorvilnius: what log level are you recording in the console? Check /etc/asterisk/logger.conf |
| 08:40.22 | ectospasm | it may be we're missing some of the CLI output. |
| 08:45.23 | jegorvilnius | yestarday i have made full log full => notice,warning,error,debug,verbose,dtmf,fax |
| 08:45.43 | jegorvilnius | I will show What is going in than log |
| 08:47.56 | jegorvilnius | http://pastebin.com/e8S6w06H |
| 08:49.42 | ChannelZ | How are you testing this? |
| 08:50.02 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 08:51.04 | jegorvilnius | I am sending fax to pstn line which is connected via X100p wildcard (Dahdi chan 1) with Asterisk |
| 08:51.29 | ChannelZ | https://issues.asterisk.org/view.php?id=18299 |
| 08:51.31 | jegorvilnius | and i was trying to call to that line and asterisk is dialing |
| 08:52.41 | ChannelZ | Looks like a bug |
| 08:52.59 | ectospasm | jegorvilnius: I asked that the console line is, not full |
| 08:53.13 | ectospasm | s/that/what/ |
| 09:00.17 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 09:01.02 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 09:14.52 | marl_scot | mornig, i am having big problems trying to get dtmf working over SIP->*->LCR->ISDN, any time i hit a key on my SIP phones i just get a click on the other end of the ISDN, this is stopping any access to IVR systems etc. anyone able to help? |
| 09:16.22 | WIMPy | Did you set the option to generate dtmf in your Dial? |
| 09:17.50 | marl_scot | hi WIMPy , u mean the Dial(LCR/Ext/${EXTEN:1}/:s) |
| 09:18.14 | marl_scot | from what i could find on LCR the 's' option should put DTMF inband |
| 09:18.19 | WIMPy | What's that : doing there? But yes, the s. |
| 09:18.23 | marl_scot | or am i reading it wrong? |
| 09:18.57 | marl_scot | the colon was in the example i followed, have now removed it :) |
| 09:19.20 | marl_scot | could that have casued the problem? |
| 09:19.22 | WIMPy | Does it make a change? |
| 09:19.30 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:19.51 | WIMPy | Don't know what it makes of things it doesn't know. |
| 09:21.32 | marl_scot | lol, that looks like it might be it, will get get someone to try from a hardware sip phone (was testing via softphone ) |
| 09:23.47 | marl_scot | :( seamed to work from my softphone but not from the hard sip phones :( |
| 09:24.08 | marl_scot | they are yealink T26 phones |
| 09:24.51 | marl_scot | also fromt he softpohone i seem to still get the very short attempt at dtmf then the normal tone after it |
| 09:24.55 | WIMPy | Does it work from the phone to Asterisk? e.g. with voicemailmain()? |
| 09:24.56 | jegorvilnius | can anybody can tell me how can I patch asterisk with patch which is in https://issues.asterisk.org/view.php?id=18230 |
| 09:25.26 | marl_scot | havnt got voicemail setup, but phone-phone internally works without any probs |
| 09:26.08 | WIMPy | Either set up some test extension or look into the debug log. |
| 09:26.43 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 09:27.41 | WIMPy | Actually I think you only get that level of detail when you start asterisk -c. |
| 09:28.32 | *** join/#asterisk Hatrix76 (~Hatrix76@164.Red-80-36-218.staticIP.rima-tde.net) |
| 09:28.44 | Hatrix76 | Hello, I would like to know if some of you have expierence with patton 4960, we have 3 PRI connections and in total about 7k - 10k calls per day, we need something stable and I read good things about the patton devices, but anybode using them in production here? |
| 09:29.51 | marl_scot | if i setup voicemailmain, i take it we need to know if it reconises the dtmf tones? |
| 09:29.55 | *** join/#asterisk capitan (~A@ppp-71-140-67-195.dsl.irvnca.pacbell.net) |
| 09:30.24 | WIMPy | Hatrix76: Depends on what you want to do. But due to the fact that every protocol conversion will be lossy and make debugging harder, I prefer PCI cards. |
| 09:30.51 | WIMPy | marl_scot: You would find out if it works that far, yes. |
| 09:31.10 | marl_scot | ok, sorks from my softphone, will try one of the yealinks |
| 09:32.25 | Hatrix76 | WIMPy what protocol conversion will be done (lossy or not) when using a Patton 4960 device for voice communication instead of an internal PCI Card? |
| 09:32.48 | WIMPy | Hatrix76: From ISDN to SIP. |
| 09:33.28 | marl_scot | hi WIMPy , yup voicemailmain works from the yealinks as well |
| 09:34.20 | WIMPy | marl_scot: Hmm. Should give the sam results with lcr as well, then. Are they in the same context? Using the same Dial()? |
| 09:35.52 | marl_scot | same context, one is useing : exten => 298,1,VoicemailMain() : the other is using : exten => _9.,n,Dial(LCR/Ext/${EXTEN:1}/s) |
| 09:36.39 | WIMPy | Snow! |
| 09:36.46 | Hatrix76 | WIMPy yes, but my question is, where do you lossy loose something? I mean, if the conversion from ISDN->SIP is done on the patton, SIP is used to the PBX. The internal phones are SIP as well, so, there is no conversion from the internal SIP Phones to the Patton, as the PBX does no conversion ... so why is it different? |
| 09:38.27 | WIMPy | Hatrix76: Shouldn't make a difference for your phones then. But the usual problem when converting to SIP is that when a call doesn't reach it's destination, you usually don't know what happened. |
| 09:38.48 | WIMPy | I always find that disturbing. |
| 09:39.32 | Hatrix76 | WIMPy but I guess in this case you will for sure have logs on a device like the patton, won't you? |
| 09:39.51 | WIMPy | I think some devices ass Q.850 causes to SIP, but I think the location is always lost. |
| 09:40.26 | WIMPy | Proobably, but that doesn't help the user on his phone. |
| 09:40.27 | WIMPy | ^^ add |
| 09:41.29 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 09:41.38 | marl_scot | lol, thanks WIMPy , that was my bad, tehre, was testing the dtmf when dialing INTO the box from ouside, and of course that was on a differant context to dialing OUT, it is now working, thank you VERY VERY much :) |
| 09:41.51 | Hatrix76 | WIMPy, well, you are confusing me. Do you have experience with hardware SIP Gateways? Because what I am looking for is some recommendations about the stability of patton devices. I guess what you are referring to are possible network problems additional devices ... |
| 09:41.58 | WIMPy | Great. |
| 09:42.45 | marl_scot | now all i got to do is get my BLF working :( but i think that may be a problem with * 1.4 |
| 09:42.56 | WIMPy | No, I heven't used SIP gateways, unless you count Asterisk iteself. |
| 15:12.03 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:12.03 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:13.26 | jbroome | jaytee: that doesn't bode well for your voip action |
| 15:15.17 | *** join/#asterisk notax (~notax@128-145-16-190.fibertel.com.ar) |
| 15:15.22 | notax | hello |
| 15:20.06 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 15:20.11 | raden | Naikrovek, you around bro ? |
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| 15:28.24 | jbroome | doh |
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| 15:36.52 | asterisk-learner | any expert about chanspy |
| 15:36.53 | asterisk-learner | ? |
| 15:37.42 | frigidzephyr | just ask your question |
| 15:37.59 | frigidzephyr | there are plenty of asterisk experts here |
| 15:38.38 | Kobaz | axeterisk |
| 15:39.40 | jaytee | yep, axeterisk, it's also a deodorant, not just a voip system |
| 15:40.11 | russellb | heh, a deodorant that attracts woment apparently |
| 15:40.51 | Kobaz | and it cleans your bawls |
| 15:40.53 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 15:41.05 | jbroome | and lets blind people know you're a douchebag |
| 15:41.08 | asterisk-learner | ok, here is my simple setup |
| 15:41.23 | Kobaz | it's probably not simple... but go on |
| 15:41.29 | asterisk-learner | i have 3 SIP phones ( 2 hardphones and 1 softphone (zoiper)) |
| 15:41.46 | asterisk-learner | phone 1 calls phone 2 and a call is established |
| 15:42.02 | asterisk-learner | zoiper calls extension of phone1 and is spying + whispering to it |
| 15:42.13 | asterisk-learner | thru a very simple context |
| 15:42.26 | asterisk-learner | answer() then chanspy(tech / extension) |
| 15:42.37 | asterisk-learner | ok till here ? |
| 15:44.09 | p3nguin | What's the problem? |
| 15:44.51 | asterisk-learner | till here, all is fine : zoiper can listen and whisper to phone 1 |
| 15:44.59 | asterisk-learner | however, if phone 1 puts phone 2 on hold then zoiper can neither listen nor whisper to phone 1 |
| 15:45.06 | asterisk-learner | and pressing * on zoiper dialplad will not make any diffrence |
| 15:45.18 | asterisk-learner | isn't * in chanspy, used to switched between channels ? |
| 15:45.48 | Kobaz | hmm, i've never tried spying on a channel on hold |
| 15:46.01 | Kobaz | you'll probably need to spy on the phone 2 side of the call |
| 15:46.30 | *** join/#asterisk Holos (~chad@ip-70-38-12-204.static.privatedns.com) |
| 15:48.04 | asterisk-learner | but this scenario is more like |
| 15:48.05 | frigidzephyr | the situation makes sense to me. if phone 1 presses hold, the microphone and speaker are not active. what would you want to spy on it that case? silence? |
| 15:48.27 | asterisk-learner | phone 1 is the phone of an agent, and phone 2 the one of a user and zoiper plays the role of the supervisor |
| 15:48.46 | asterisk-learner | so it make sense to whisper to the agent and not the end user |
| 15:49.05 | Kobaz | oh |
| 15:49.08 | asterisk-learner | so to you |
| 15:49.12 | asterisk-learner | this is a design issue |
| 15:49.17 | asterisk-learner | and not a mis-configuration ? |
| 15:49.35 | Kobaz | well... if the agent puts the caller on hold... then you have a channel on hold... there is no audio path to whisper anymore to the phone |
| 15:49.37 | p3nguin | It sounds normal to me. |
| 15:49.45 | Kobaz | yeah, that's proper behavior |
| 15:49.46 | frigidzephyr | right |
| 15:50.57 | asterisk-learner | mmmmmmmmm |
| 15:51.13 | asterisk-learner | can someone explain to me what is the role of the DTMF * button in chanspy |
| 15:51.14 | asterisk-learner | ? |
| 15:51.37 | asterisk-learner | isnt't it to switch channels ? |
| 15:51.39 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:51.49 | WIMPy | Err, just to make that clear: You want to whisper to the party that put the call on hold? |
| 15:52.20 | Kobaz | asterisk-learner: * switches channels that match the given pattern, yes |
| 15:52.41 | asterisk-learner | WIMPy : true |
| 15:52.53 | asterisk-learner | Kobaz : ok but what if no pattern was provided ? |
| 15:52.56 | Kobaz | asterisk-learner: you can't |
| 15:53.28 | WIMPy | Well, that's definitely not possible. |
| 15:53.34 | asterisk-learner | sorry where is the pattern provided ? |
| 15:53.39 | Kobaz | asterisk-learner: to chanspy |
| 15:53.52 | WIMPy | Someone who put a call on hold is simply not there any more. |
| 15:54.02 | Kobaz | ChanSpy([chanprefix][,options]) it's right there in the help |
| 15:54.07 | asterisk-learner | ah ok ok |
| 15:54.10 | asterisk-learner | :-$ |
| 15:54.11 | WIMPy | He *might* be on another channel. |
| 15:54.11 | Kobaz | hitting * will cycle all channels that match the prefix |
| 15:54.45 | asterisk-learner | can someone provide a solution to my scenario... should i create another channel or something ? |
| 15:54.55 | Kobaz | call the agent |
| 15:55.12 | asterisk-learner | manually thru Dial ? |
| 15:55.13 | WIMPy | There is no possible solution when the call is on hold. |
| 15:55.28 | WIMPy | Other than placing a new call. |
| 15:55.35 | Kobaz | you wont get any audio through to the agent by spying, the phone is essentially "turned off" during hold |
| 15:56.31 | asterisk-learner | mmmmm |
| 15:56.55 | Kobaz | ring the agent if you want to tell him/her something |
| 15:57.05 | Kobaz | Dial() |
| 15:57.12 | asterisk-learner | ok |
| 16:02.33 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:02.35 | wcselby | o/ |
| 16:04.41 | *** part/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 16:11.22 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
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| 16:24.36 | manoot | if i want to disable cdr_sqlite i just need to noload => cdr_sqlite3_custom.so in modules.conf right? |
| 16:24.41 | manoot | there are no other modules I need to disable? |
| 16:25.21 | manoot | in 1.8 |
| 16:29.38 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:31.05 | JunK-Y | module show and disable whatever you want to |
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| 16:32.54 | p3nguin | Is codec_dahdi.so required to transcode? |
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| 16:45.24 | *** join/#asterisk msetim (~setim@187.112.159.216) |
| 16:45.28 | msetim | hello guys |
| 16:46.17 | msetim | I'm testing 1.4 EXITWITHTIMEOUT, however I'm getting some strange event: QueueCallerAbandon instead of "QueueCallerExitWithTimeout" |
| 16:47.05 | msetim | In fact I don't know if that event exists, but its not correct to say "Abandon" when the call was dropped by asterisk because it reached the queue timeout |
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| 17:09.24 | nfi|ermes | hi all |
| 17:09.32 | nfi|ermes | very strange thing |
| 17:09.37 | nfi|ermes | <PROTECTED> |
| 17:09.53 | nfi|ermes | but /usr/lib/asterisk/modules/chan_dadi.so is there |
| 17:11.13 | Kobaz | nfi|ermes: something else isn't found |
| 17:11.53 | nfi|ermes | but what else ? |
| 17:12.03 | pabelanger | nickaugust: it is chan_dahdi.so |
| 17:12.06 | pabelanger | you have a typo |
| 17:13.12 | p3nguin | <nfi|ermes> but /usr/lib/asterisk/modules/chan_dadi.so is there <-- Doubtful. Check again. |
| 17:14.18 | Kobaz | oh. heh... yeah the typo |
| 17:14.31 | Kobaz | it really is that simple |
| 17:14.53 | Kobaz | pabelanger: sometimes with really fubar'd setups.. the module in question is found and is fine, but that module needs another one, which isn't found |
| 17:15.04 | nfi|ermes | ok |
| 17:15.09 | nfi|ermes | other error now |
| 17:15.48 | pabelanger | Kobaz: Ya, modules dependencies are tricky. A patch was merged into trunk a few months ago to help with that :) |
| 17:15.58 | pabelanger | nfi|ermes: pb them |
| 17:16.41 | nfi|ermes | ok perfect |
| 17:16.52 | nfi|ermes | i'll do |
| 17:17.14 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 17:19.32 | WIMPy | Is 4ESS PRI only? |
| 17:20.11 | pabelanger | WIMPy: Yes, ISDN |
| 17:20.42 | WIMPy | The question was if it doesn't exist for BRI. |
| 17:22.42 | pabelanger | WIMPy: I _think_ 5ESS will work on both BRI / PRI. 4ESS is PRI |
| 17:23.54 | WIMPy | Yes, I found a reference for 5ESS on BRI, bit not for 4ESS. So 4ESS is not available on BRI? |
| 17:24.40 | WIMPy | s/bit/but/ |
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| 17:41.27 | *** join/#asterisk LorenzBarth (~Miranda@62.216.165.71) |
| 17:41.30 | LorenzBarth | hi |
| 17:42.13 | LorenzBarth | i try to activate my FAX licence. But there is no tab availibleto do so in the digium register website |
| 17:43.03 | russellb | LorenzBarth: you can use the console register utility. Otherwise, please contact support - http://www.digium.com/en/supportcenter |
| 17:43.38 | LorenzBarth | i wanted to register to submit a case online |
| 17:43.46 | LorenzBarth | i already registered the licence and can use ist |
| 17:44.16 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:45.20 | russellb | infobot: digiumsuppport is <reply> With the purchase of a Digium product, you are entitled to installation support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcetner |
| 17:45.20 | infobot | okay, russellb |
| 17:45.40 | russellb | infobot: forget digiumsupport |
| 17:45.40 | infobot | russellb: i didn't have anything called 'digiumsupport' to forget |
| 17:45.41 | jaytee | ~botsnack |
| 17:45.41 | infobot | jaytee: thanks |
| 17:45.51 | russellb | infobot: digiumsuppport is <reply> With the purchase of a Digium product, you are entitled to support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcetner |
| 17:45.51 | infobot | ...but digiumsuppport is already something else... |
| 17:46.01 | russellb | infobot: forget digiumsupport |
| 17:46.01 | infobot | russellb: i didn't have anything called 'digiumsupport' to forget |
| 17:46.04 | russellb | infobot: digiumsuppport is <reply> With the purchase of a Digium product, you are entitled to support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcetner |
| 17:46.04 | infobot | ...but digiumsuppport is already something else... |
| 17:46.07 | russellb | i hate you infobot |
| 17:46.10 | LorenzBarth | hehe |
| 17:46.17 | Qwell | pp != ppp |
| 17:46.27 | Qwell | ~digiumsupport |
| 17:46.34 | Qwell | ~digiumsuppport |
| 17:46.35 | infobot | With the purchase of a Digium product, you are entitled to installation support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcetner |
| 17:46.38 | russellb | infobot: forget digiumsuppport |
| 17:46.38 | infobot | russellb: i forgot digiumsuppport |
| 17:46.44 | russellb | infobot: digiumsupport is <reply> With the purchase of a Digium product, you are entitled to support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcetner |
| 17:46.44 | infobot | russellb: okay |
| 17:46.54 | LorenzBarth | the link is wrong cetner forwards to error |
| 17:47.02 | russellb | ha |
| 17:47.06 | russellb | infobot: forget digiumsupport |
| 17:47.06 | infobot | i forgot digiumsupport, russellb |
| 17:47.06 | LorenzBarth | but center ist working :-) |
| 17:47.13 | russellb | infobot: digiumsupport is <reply> With the purchase of a Digium product, you are entitled to support. For more information on contacting Digium Support, please see http://www.digium.com/en/supportcenter |
| 17:47.13 | infobot | russellb: okay |
| 17:47.42 | LorenzBarth | but if i clck there on "submit a case" and login it tels me: |
| 17:47.46 | Qwell | infobpt: no, digiumsupport is <reply> For help: /msg russellb <Your question here> |
| 17:47.53 | LorenzBarth | "Don't see the product you purchased in the list? Register or activate your product first." |
| 17:48.03 | LorenzBarth | but how can i register my software license? |
| 17:48.07 | russellb | LorenzBarth: I guess you'll have to call |
| 17:48.08 | russellb | i don't know |
| 17:48.24 | LorenzBarth | ok. i'll use another registred product as base |
| 17:48.46 | Qwell | or you could register/activate the one that isn't in the list, like it suggested. |
| 17:49.06 | russellb | Qwell: which brings us full circle to the first question |
| 17:49.13 | LorenzBarth | but theres no tab for a software license |
| 17:49.21 | russellb | ... which was that FAX isn't listed for registration |
| 17:50.34 | espiceland | cetner? |
| 17:50.42 | espiceland | Haha. |
| 17:50.45 | russellb | espiceland: shush you |
| 17:51.06 | espiceland | suppport cetner. |
| 17:51.19 | russellb | sounds like a cause worth donating to |
| 17:51.36 | *** join/#asterisk notax (~notax@128-145-16-190.fibertel.com.ar) |
| 18:03.52 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 18:03.57 | *** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 18:04.18 | t_j | anyone know what the format of the timestamp is inh polycom log files? |
| 18:05.11 | Katty | peeks in |
| 18:05.13 | Katty | ^_^ |
| 18:05.29 | thehar | >.> |
| 18:05.30 | thehar | <.< |
| 18:05.41 | Katty | let's hug! |
| 18:05.50 | thehar | hugs Katty |
| 18:05.53 | Katty | hugs thehar |
| 18:06.25 | Katty | what's for lunch |
| 18:06.35 | Qwell | had tacos |
| 18:06.46 | Katty | tacos are yummy |
| 18:06.48 | thehar | parsnip tacos? |
| 18:07.13 | Katty | what on earth is that |
| 18:07.17 | Katty | jeezy creezy |
| 18:07.20 | Katty | what on earth is that |
| 18:07.27 | thehar | lol |
| 18:07.31 | thehar | it's a joke |
| 18:07.45 | Katty | must've gone over my head |
| 18:07.52 | wcselby | i think i'm going to have pizza buffet |
| 18:07.57 | WIMPy | Don't make jokes about food. |
| 18:07.58 | wcselby | o/ Katty |
| 18:08.09 | Katty | hugs wcselby |
| 18:08.11 | WIMPy | And now I'm hungry again, BTW. |
| 18:08.13 | Katty | wcselby: promise me you'll eat some veggies. |
| 18:08.20 | Katty | wcselby: a plate of salad |
| 18:09.13 | wcselby | Katty- how about some veggies on one of the slices of pizza? |
| 18:09.16 | wcselby | does that count? |
| 18:09.20 | Katty | no |
| 18:09.24 | wcselby | :/ |
| 18:09.28 | Katty | that doesn't have much nutritional value |
| 18:09.33 | Katty | and i want you to be happy, healthy, wcselby |
| 18:09.39 | wcselby | then I can make no promise, the salad at this place is rather....poor |
| 18:10.05 | wcselby | i am going to start exercising again tonight |
| 18:10.19 | wcselby | I've got a potential contract coming up in march next year that I'd like to get into shape for |
| 18:10.31 | wcselby | since I'll be sitting around a bunch of military types all day |
| 18:10.44 | Katty | then what better way to start than having a few veggies at lunch |
| 18:10.52 | Katty | in place of a couple slices of pizza |
| 18:11.00 | Katty | you won't feel so lethargic and tired later tonight |
| 18:11.03 | wcselby | heh |
| 18:11.10 | wcselby | this is true, that's the one really bad thing about pizza buffet |
| 18:11.19 | wcselby | the whole fighting off sleep for the rest of the afternoon |
| 18:12.07 | t_j | [TK]D-Fender: thanks |
| 18:12.22 | Katty | wcselby: it's not too late to change your mind! |
| 18:12.30 | Katty | wcselby: you could always get the salad buffet at ruby tuesdays! |
| 18:12.34 | wcselby | heh |
| 18:12.38 | wcselby | i'm meeting a client there :) |
| 18:12.44 | Katty | k |
| 18:13.01 | wcselby | but thanks for the moral support :) |
| 18:13.33 | Katty | you can do it! |
| 18:13.42 | Katty | even if it's just a handful of cherry tomatoes (= |
| 18:13.54 | Katty | or some cucumber slices |
| 18:14.07 | Kobaz | what's a word for 'other party' |
| 18:14.21 | Kobaz | like, you have an agent, and a caller... in a queue system |
| 18:14.27 | Katty | that would be husband. |
| 18:14.28 | Kobaz | caller calls in, agent picks up... |
| 18:14.33 | Kobaz | but then what if the agent calls out |
| 18:14.51 | Kobaz | the agent is now technically the caller.. but then that's confusing.. it should always be called the agent |
| 18:14.59 | wcselby | third party? |
| 18:15.10 | Kobaz | it's not a third. there's two people on the line |
| 18:15.24 | wcselby | so you're talking outbound queues? |
| 18:15.32 | Kobaz | not necessarily |
| 18:15.32 | Katty | recipient? |
| 18:15.40 | wcselby | callee |
| 18:15.47 | Kobaz | an inbound agent happens to need to make an outbound call |
| 18:15.59 | Kobaz | but i have one field to describe the 'other end' which is on_call_with |
| 18:16.07 | Kobaz | but i was trying to think of a shorter word |
| 18:16.12 | Kobaz | well, one word |
| 18:16.16 | wcselby | i guess let's step back and look at the big picture -what are you needing a name for? |
| 18:16.30 | *** join/#asterisk _zoom_ (~user@196.1.251.124) |
| 18:16.31 | Kobaz | i guess i could call it 'other', but then it's not very descriptive |
| 18:16.50 | Katty | aobc |
| 18:16.54 | Katty | agent's outbound call |
| 18:16.54 | _zoom_ | hi, what was the variable or the function tell the call progress? |
| 18:16.55 | Katty | aoc! |
| 18:16.56 | wcselby | my big client calls them callsbacks |
| 18:16.58 | wcselby | callbacks* |
| 18:17.03 | Kobaz | big picture: 13:15 <Kobaz> an inbound agent happens to need to make an outbound call |
| 18:17.08 | wcselby | since that's the only reason that the inbound agent should be making outbound calls |
| 18:17.21 | WIMPy | Kobaz: Respondent? |
| 18:17.25 | Kobaz | mmm |
| 18:17.28 | Katty | your_mom |
| 18:17.29 | Katty | ^_^ |
| 18:17.31 | wcselby | lol |
| 18:17.32 | Kobaz | but it's one field. that describes both |
| 18:17.50 | Kobaz | agent_chan, on_call_with_chan |
| 18:18.05 | *** join/#asterisk SparFux (~raoul@e182027182.adsl.alicedsl.de) |
| 18:18.06 | WIMPy | Kobaz: "remote"? |
| 18:18.17 | Kobaz | mmm, maybe |
| 18:18.22 | Kobaz | yeah that sounds good |
| 18:18.23 | SparFux | Hi all. Is there a version controlled repo for libpri somewhere? |
| 18:18.31 | Kobaz | SparFux: svn |
| 18:19.09 | SparFux | Kobaz: ah, thx. Found it in http://www.asterisk.org/developer/resources/svn |
| 18:20.19 | leifmadsen | Kobaz: o/ |
| 18:20.25 | leifmadsen | practices his juggling |
| 18:20.28 | Kobaz | hehe |
| 18:20.44 | Kobaz | you should be able to do 5 by next astricon |
| 18:20.46 | Kobaz | that's your task |
| 18:20.51 | Qwell | only 5? pfft. |
| 18:20.55 | leifmadsen | lol I'll be happy if I can do 3 consistently :) |
| 18:21.06 | leifmadsen | 5 iterations of 3? maybe. |
| 18:21.14 | Qwell | I can totally juggle 8 chainsaws |
| 18:21.26 | Kobaz | heh |
| 18:21.33 | leifmadsen | that'd be sick :) |
| 18:21.37 | leifmadsen | I'd pay to see that |
| 18:21.58 | drmessano | I can juggle 8 girlfriends |
| 18:22.03 | leifmadsen | heyo! |
| 18:22.10 | SparFux | Aha, I think libpri Changelog is somehow created from svn log output? Ok. |
| 18:22.20 | Qwell | emails drmessanos girlfriend |
| 18:22.21 | Kobaz | that's one more than the days in the week. how do you manage the schedules |
| 18:22.26 | leifmadsen | SparFux: yes they all are |
| 18:22.38 | drmessano | Google Voice makes it easy to have a bunch of Girlfriends. I can it Google Ho-Phone |
| 18:22.42 | drmessano | call it* |
| 18:23.12 | Kobaz | do you have a girl for every night of the week, and then a backup girl if one is busy? |
| 18:23.20 | drmessano | Google Voice + Asterisk = makes it look like I care, but not too much |
| 18:23.25 | leifmadsen | no he has 2 girls for 4 nights of the week |
| 18:23.46 | Kobaz | ah yes... parallelize |
| 18:24.00 | drmessano | Kobaz, weekdays are for girlfriends, weekends are for girlfriend interviews |
| 18:24.47 | drmessano | So 2 x 4 days a week, then 3 days for "interviews" to see who makes it to the next week, and what new chick(s) may come onboard |
| 18:25.10 | Kobaz | kind of like survivor |
| 18:25.12 | Katty | drmessano: i'd like to see you juggle 8 girlfriends |
| 18:25.28 | wcselby | honestly, who'd want to? |
| 18:25.29 | wcselby | :P |
| 18:25.32 | drmessano | I hate Sundays though, when I get to send the prerecorded Alison "I'm sorry, you didn't make it to the next week. Goodbye" message |
| 18:25.33 | Katty | exactly. |
| 18:25.37 | p3nguin | without a net |
| 18:26.15 | drmessano | Katty, you told me that watching me with 8 girls was your fantasy. Shuuush |
| 18:26.31 | Katty | eww. |
| 18:26.37 | Katty | do you know how many germs would be in that room!? |
| 18:27.02 | p3nguin | Does Purell come in a 55 gallon drum? |
| 18:27.06 | drmessano | Katty, as long as they are D/D free, there's Lysol for the rest |
| 18:27.23 | Katty | p3nguin: i hope it comes with a complimentary set of face masks. |
| 18:27.27 | Katty | p3nguin: drmessano is gonna need it |
| 18:27.32 | drmessano | p3nguin, Purell doesn't work well in a garden sprayer, just FYI. Don't ask. |
| 18:27.33 | wcselby | hahaha |
| 18:27.55 | wcselby | time for some lunch, I think. |
| 18:28.01 | Katty | wcselby: VEGETABLES! |
| 18:28.05 | Katty | wcselby: IMMUNE SYSTEM |
| 18:28.11 | drmessano | p3nguin, also, never ask a girl if you can spray her down with a garden sprayer. |
| 18:28.32 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 18:28.40 | p3nguin | I can see it being offensive if you've got Purell in it. |
| 18:28.49 | Katty | yeah i was gonna say |
| 18:28.52 | Katty | is it like...chocolate? |
| 18:29.07 | Katty | high pressure water? ^_^ |
| 18:29.17 | Katty | ahem. NEXT. |
| 18:29.47 | WIMPy | Already seen Jackass 3D? |
| 18:29.57 | Kobaz | drmessano: this is from experience? |
| 18:31.00 | SparFux | Digium is listed as sponsor, but isn't Digium the company all behind the asterisk + dahdi-linux + dahdi-tools + libpri ? |
| 18:31.08 | Kobaz | Katty: as long as noone is sick... i don't think there would be germ issues |
| 18:31.29 | drmessano | Kobaz, there is a reason I keep a tarp in my bedroom |
| 18:31.50 | Kobaz | taco wrap? |
| 18:31.54 | drmessano | "Asterisk Installations" can be messy |
| 18:32.00 | drmessano | ^^^ ha, on topic |
| 18:32.24 | leifmadsen | SparFux: yes |
| 18:33.30 | russellb | a very strong sponsor of the project, indeed :-) |
| 18:33.50 | SparFux | Ok. |
| 18:34.02 | SparFux | Like the only sponsor :-) |
| 18:34.17 | SparFux | Or like being the project itself? |
| 18:34.41 | russellb | depends on how you define sponsor ... plenty of other companies contribute to asterisk |
| 18:34.45 | p3nguin | "If I have one more piece of vomit pie, I'm going to pumpkin." |
| 18:34.51 | SparFux | russellb: ah, I see. |
| 18:36.58 | *** join/#asterisk tyman (~tyler@173-14-203-40-fresno.hfc.comcastbusiness.net) |
| 18:40.06 | *** join/#asterisk Lantizia (~Lantizia@188-221-11-131.zone12.bethere.co.uk) |
| 18:40.10 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 18:41.51 | *** join/#asterisk jsjc (~jsjc@213.96.102.13) |
| 18:43.09 | drmessano | Digium actually employs child labor and undocumented workers to code Asterisk |
| 18:43.14 | drmessano | The "Dev Team" is a front |
| 18:43.21 | Qwell | confirm |
| 18:43.41 | Qwell | but it's okay, because they're better coders |
| 18:43.53 | Katty | not all countries have child labor laws. |
| 18:44.01 | Katty | ;P |
| 18:44.12 | russellb | that's why we hire so many devs in Canada |
| 18:44.12 | Katty | child labor laws are awesome. |
| 18:44.16 | drmessano | Fun fact: When ever you ask one of the "Dev Team" a question, and they answer "I dont know", like on "You can't do that on television", they get green slimed |
| 18:44.17 | leifmadsen | oh snap |
| 18:44.42 | russellb | (the child labor part) |
| 18:44.50 | russellb | leifmadsen is basically an 11 year old |
| 18:44.56 | leifmadsen | I was 8 when I started |
| 18:44.57 | Katty | but a very stylish one. |
| 18:44.59 | russellb | we pay him in legos |
| 18:45.04 | leifmadsen | legos are awesome! |
| 18:45.30 | Katty | i gave star wars legos as a birthday present this past saturday |
| 18:45.36 | Katty | they were a hit |
| 18:45.40 | drmessano | Ah, which is also part of the vast Canadian Asterisk Conspiracy as well. You Cant Do That On Television was a Canadian TV show, most of the "devs" are from Canada, and they also serve Back Bacon in the cafeteria |
| 18:45.48 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:46.14 | _Corey_ | Excellent Zutroy... work hard and each day you'll get a shiny penny! |
| 18:46.19 | russellb | throws a bowling ball at drmessano |
| 18:46.43 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:47.03 | drmessano | If you can find the "Lost Asterisk Sounds" for Canada, all of them are word/phrase + eh |
| 18:47.19 | drmessano | They have been well hidden over the years, but they do exist |
| 18:48.06 | drmessano | Mark Spencer has spent tens of thousands of his own money covering up the Asterisk/Canada paper trail, but he's not fooling anyone.. There's back bacon grease on HIS hands too! |
| 18:48.29 | leifmadsen | mmmm back bacon |
| 18:52.58 | jaytee | what's up with sourceforge? |
| 18:54.02 | jbroome | other than the usual slow POSery? |
| 18:54.25 | jaytee | like as in it's down |
| 18:55.37 | *** join/#asterisk Lantizia (~Lantizia@188-221-11-131.zone12.bethere.co.uk) |
| 18:56.15 | beek | jaytee: It's up here. |
| 18:59.17 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 19:01.40 | jaytee | weird! when I click on a link from the phpmyadmin site it gives me a sourceforge page that looks like it's down but when I go there direct it's up. |
| 19:01.50 | *** part/#asterisk _zoom_ (~user@196.1.251.124) |
| 19:15.32 | *** join/#asterisk TheDavidFactor (~David@c-68-34-116-180.hsd1.md.comcast.net) |
| 19:16.54 | *** join/#asterisk a_bug (~abhinav@122.163.21.48) |
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| 19:27.20 | Katty | sways. |
| 19:28.18 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 19:33.18 | *** join/#asterisk Tigerplug292 (~Tigerplug@089-101-012222.ntlworld.ie) |
| 19:34.07 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 19:39.39 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 19:46.26 | carrar | looks at Katty sway |
| 19:48.03 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 19:48.15 | neurosys | How could I place a call on hold and continue with the dialplan? |
| 19:49.25 | leifmadsen | Transfer() and MusicOnHold() |
| 19:49.28 | russellb | you might have to be a bit more specific |
| 19:49.32 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:49.33 | Katty | sways with carrar |
| 19:49.39 | leifmadsen | you'll have to be very clever :) |
| 19:49.42 | russellb | leifmadsen: that's evil |
| 19:49.46 | leifmadsen | russellb: sure is :) |
| 19:49.55 | russellb | leifmadsen: how would you "take it off hold" ? |
| 19:49.57 | leifmadsen | a Local channel may be involved somewhere, hard to say |
| 19:50.02 | leifmadsen | russellb: Bridge() |
| 19:50.11 | leifmadsen | I've actually done this before :) |
| 19:50.14 | russellb | Transfer() is going to end the first dialplan isn't it? |
| 19:50.44 | leifmadsen | oh right, I remember now |
| 19:51.06 | leifmadsen | I put it on MusicOnHold() but didn't continue in the dialplan -- it was a multi-lot parking lot that I built before that existed |
| 19:51.08 | leifmadsen | early 1.4 |
| 19:51.52 | russellb | ah, transfer to MOH, then Bridge() to pick it back up later |
| 19:51.57 | leifmadsen | ya |
| 19:52.03 | russellb | yeah, continuing in the dialplan is the tricky part |
| 19:52.09 | neurosys | right.. :( |
| 19:52.15 | leifmadsen | now whether that works or not I'm not sure, you could end up with a zombie channel or something |
| 19:53.02 | leifmadsen | usually once you start playing audio to a channel and it gets answer you lower the number of options you have because the dialplan will tend to block |
| 19:53.02 | russellb | it really depends on what you want to do ... I think I'd need more information |
| 19:53.06 | leifmadsen | aye |
| 19:53.12 | leifmadsen | a call flow would be useful |
| 19:53.18 | russellb | I mean ... you could do StartMusicOnHold(), which will play MOH while you go off and do other stuff |
| 19:53.25 | russellb | but I don't know what we're trying to accomplish |
| 19:53.25 | leifmadsen | russellb: o/ |
| 19:53.31 | leifmadsen | russellb: we're baking cookies |
| 19:53.34 | russellb | !!! |
| 19:53.34 | Katty | perks up |
| 19:53.36 | Katty | cookies?! |
| 19:53.36 | russellb | i am jealous. |
| 19:53.37 | neurosys | Ok What im trying to do is setup a way to test roll over lines. |
| 19:53.52 | WindBack | I 'm having a randomly problem. Sometimes my calls to the PSTN (trough ISDN/PRI) goes down. Looking into pri debuging I see Asterisk sending the DISCONNET message. However, enabling more dubugging in the PBX I see the following: " Got a FRAME_CONTROL (8) frame on channel DAHDI/34-1 " More details: http://www.pastebin.ca/2000163 |
| 19:53.53 | neurosys | I used a While Loop and set a variable via Read() to get the number of lines, |
| 19:53.56 | leifmadsen | waits for the punchline |
| 19:54.07 | WindBack | Please, if anybody know about this problem helpme |
| 19:54.13 | Katty | directs leifmadsen to the back, with a cup. |
| 19:54.26 | Katty | leifmadsen: don't even think about making it electric, sir. |
| 19:54.35 | leifmadsen | Katty: I promise nothing |
| 19:55.12 | Katty | electric koolaid and knuckle sammiches |
| 19:55.19 | neurosys | leifmadsen: while each line is called and answered, it should redial the MAIN number (grabbed from the CID) and dial again, having the next line in the roll over to ring. and so on |
| 19:55.26 | neurosys | Ill pastebin. |
| 19:57.54 | neurosys | leifmadsen: this is where Im at... -> http://pastebin.ca/2000172 |
| 19:59.36 | leifmadsen | neurosys: oh! |
| 19:59.54 | leifmadsen | You could do an Originate() first possibly |
| 20:00.11 | leifmadsen | might have to trigger it via a Local channel or something so it can run independently of the MusicOnHold() |
| 20:00.31 | leifmadsen | something like that |
| 20:06.04 | *** part/#asterisk SparFux (~raoul@e182027182.adsl.alicedsl.de) |
| 20:09.58 | i_heart_asterisk | we've noticed that voicemails sometimes take a long time to get to INBOX, what could be causing that ? |
| 20:10.16 | *** join/#asterisk Benwa (~benwa-ktm@unaffiliated/benwa) |
| 20:10.31 | Benwa | hi, a good softphone for windows ? |
| 20:10.34 | leifmadsen | a long time to get to the inbox? |
| 20:10.43 | leifmadsen | Benwa: zoiper, x-lit |
| 20:10.45 | leifmadsen | x-lite |
| 20:10.51 | thehar | blink |
| 20:10.55 | i_heart_asterisk | Benwa: ekiga |
| 20:11.07 | leifmadsen | I think he said good :) |
| 20:11.14 | Benwa | :) |
| 20:11.16 | thehar | lol |
| 20:11.20 | thehar | zoiper = best |
| 20:11.23 | thehar | blink is pretty good |
| 20:11.25 | leifmadsen | my choice as well |
| 20:11.25 | thehar | x-lite is standard |
| 20:11.28 | i_heart_asterisk | oh right...ekiga is only good for freeswitch...CELT codec! |
| 20:11.28 | leifmadsen | I haven't tried blink |
| 20:11.36 | Benwa | ok i'll try all of them ;) |
| 20:11.38 | leifmadsen | x-lite usually won't get you fired :) |
| 20:11.43 | thehar | ^_^ |
| 20:11.55 | i_heart_asterisk | can dial sip uri w/ ekiga |
| 20:12.03 | thehar | same with blink |
| 20:12.11 | i_heart_asterisk | cool |
| 20:12.15 | leifmadsen | and I think any of them :) |
| 20:12.21 | i_heart_asterisk | not x-lite |
| 20:12.28 | thehar | leifmadsen: no WAY! |
| 20:12.31 | thehar | they talk SIP? |
| 20:12.33 | thehar | *boggle* |
| 20:12.34 | i_heart_asterisk | i havent used blink, looks cool though |
| 20:12.35 | leifmadsen | barely |
| 20:12.43 | thehar | with crutches |
| 20:13.29 | i_heart_asterisk | leifmadsen: someone leaves a voicemail, the person checks the voicemail box, it says no new messages, hours later, the message shows up in the INBOX |
| 20:13.43 | leifmadsen | i_heart_asterisk: what backend storage? |
| 20:14.06 | leifmadsen | I bet the file really exists -- just the MWI isn't getting sent until the phone re-registers hours later |
| 20:14.17 | leifmadsen | i_heart_asterisk: what version of asterisk? |
| 20:15.36 | i_heart_asterisk | Asterisk 1.6.1.20, messages are stored on NFS, this didnt happen w/ MWI on phone, the person called in on phone to check voicemail box and the message wasnt there, although they were sure a message was left |
| 20:15.50 | leifmadsen | NFS? yuck |
| 20:16.00 | leifmadsen | sounds like a problem with the network or NFS or something |
| 20:16.01 | Benwa | thanks to all |
| 20:16.05 | *** part/#asterisk Benwa (~benwa-ktm@unaffiliated/benwa) |
| 20:16.07 | leifmadsen | it won't be an asterisk problem likely |
| 20:16.30 | i_heart_asterisk | we have a clustered setup, NFS is necessary unless you know of any alternative |
| 20:16.30 | *** join/#asterisk dobah (~dobah@cpe-98-150-177-51.hawaii.res.rr.com) |
| 20:16.48 | leifmadsen | I use ODBC voicemail storage at some locations, and IMAP at others |
| 20:16.52 | *** join/#asterisk AsManyAs (~darynm@196-210-130-97.dynamic.isadsl.co.za) |
| 20:17.15 | thehar | you could use hdfs filesystem but that is overkill for just voicemail |
| 20:17.15 | Holos | i_heart_asterisk: Active/Passive? I run rsync --delete between the two boxes on my clusters.. |
| 20:17.53 | Holos | i_heart_asterisk: then again, we only had a few vm's per hour, and they were e-mailed / archived that way.. |
| 20:17.58 | i_heart_asterisk | Holos: no we have 1 NFS, but we are running multiple asterisk boxes w/ opensips proxy, they all need to access NFS |
| 20:18.14 | *** join/#asterisk gnarf (~gnarf@router.ether8.net) |
| 20:18.17 | AsManyAs | hi there i have a problem with a new install using asterisk 1.6 wherew hiles on a cal l the call cuts but comes back after a few seconds, this is both on a sip trunk and an fxo trunk |
| 20:18.22 | jsjc | anybode knows where I could start looking for someone that with a reasonable fee will configure a simple asterisk? (4DAHDI chans, 2 sip users) |
| 20:18.30 | Holos | i_heart_asterisk: Ah.. makes sense for that way. |
| 20:18.44 | jbroome | jsjc: your location on the planet would probably help |
| 20:18.53 | i_heart_asterisk | leifmadsen: voicemail ODBC stores VMs in a DB as blob ? |
| 20:19.04 | leifmadsen | i_heart_asterisk: yes |
| 20:19.26 | i_heart_asterisk | how is that benefitial ? |
| 20:19.32 | jsjc | jbroome I think it is not neccesary right? I am located in Spain but fisically it is all set it is just configuration (asterisk it is installed and ready to be messed up with) |
| 20:19.35 | wcselby | Katty - you should be happy, I had a salad with my pizza |
| 20:19.39 | gnarf | hey guys, I have 2 asterisk servers. the "main" server handles the voicemail, but the "remote" server handles the SIP peer... They talk to each other over IAX. is there any way I can make the remote server give message waiting indicators for voicemail boxes on the main server? |
| 20:19.57 | i_heart_asterisk | *beneficial |
| 20:20.37 | leifmadsen | you could also use something like DRBD for a replicated filesystem, or unison for syncing between locations... or just have a dedicated voicemail server ;) |
| 20:20.46 | _Corey_ | hmmm... I had a customer complain about a voicemail delay very much along the lines of what i_heart_asterisk described. no NFS or anything, just native storage with some 1.6.x build. we wrote it off to user smoking hard drugs but maybe there's something to it.. |
| 20:21.33 | leifmadsen | _Corey_: nah I bet it was the blinky lamp not coming up for them |
| 20:21.43 | leifmadsen | or hard drugs |
| 20:23.09 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 20:23.39 | _Corey_ | leifmadsen: nah, that's the thing... we were cleaning up a busted deployment where the previous vendor couldn't get MWI to work at all--seriously--and the users were obsessively checking their messages. |
| 20:24.05 | leifmadsen | makes no sense to me, but ok :) |
| 20:24.14 | gnarf | hrm |
| 20:24.16 | i_heart_asterisk | we've had a feature request to be able to delete/mark/move VM messages using http links...i know this is not possible with regular asterisk voicemail, could it work with voicemail ODBC ? |
| 20:24.23 | _Corey_ | unless it happens with some regularity there's not much to say... |
| 20:24.29 | gnarf | strange, seems the topic is pretty close to mine ;) |
| 20:24.49 | gnarf | I was thinking about just rsyncing the voicemail directory over |
| 20:24.54 | gnarf | for MWI |
| 20:25.00 | i_heart_asterisk | glusterFS |
| 20:26.14 | leifmadsen | gnarf: with ODBC voicemail it should be nearly trivial -- that's what we did a few years ago |
| 20:26.28 | gnarf | i_heart_asterisk: its not that hard to parse the directory structure of the /var/lib/voicemail, you could probably write a few php scripts to do it |
| 20:26.32 | leifmadsen | PHP + ODBC = magic |
| 20:27.38 | *** part/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 20:28.08 | Juggie | if you store your voicemail in the database |
| 20:28.11 | Juggie | which i'd recomend |
| 20:28.13 | Juggie | its easy |
| 20:28.14 | gnarf | well, ODBC seems like a potential solution, but it also seems like it might be more configuration/setup than rsyncing every 5 minutes |
| 20:30.06 | Juggie | nahhh not that bad |
| 20:30.22 | leifmadsen | ya especially since there's documentation for how to do it :) |
| 20:30.32 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 20:31.19 | Juggie | odbc w/ vm data stored in the db is nice |
| 20:31.25 | Juggie | makes for the best setup. |
| 20:45.25 | i_heart_asterisk | does asterisk install ready w/ ODBC VM support? |
| 20:46.09 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 20:47.15 | tvc123 | quick question where is the best place to look for hardware reccomendations I'm looking to build a server to support 30+ conference call users |
| 20:47.37 | tvc123 | I've googled it but most of the answers seem to be quite outdated |
| 20:48.38 | AsManyAs | <PROTECTED> |
| 20:49.04 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 20:49.11 | jbroome | tvc123: grab a dell |
| 20:50.24 | *** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au) |
| 20:50.24 | tvc123 | jbroome: anything to avoid with the dells? any hardware that hasn't worked well? |
| 20:50.30 | Holos | tvc123: 30 channels isn't a lot of load.. just about anything will support that. Acer Revo's work nicely.. |
| 20:52.24 | CaneToad | in what directory does Playback() look for audio files? I thought it would have been ${astvarlibdir}/sounds, but no |
| 20:52.43 | brainiac | How would I set up a Barge-In feature that allows talking to only the extension being barged-in on and NOT the person that the ext is talking to? |
| 20:53.04 | brainiac | I'm running Asterisk 1.4.31 |
| 20:53.26 | tvc123 | Holos: how does the load factor in when they are all in the same conference call? |
| 20:53.32 | jbroome | tvc123: no, it's linux they're fine on dells |
| 20:53.57 | jbroome | i'm running on a 2650 right now |
| 21:05.28 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 21:05.44 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 21:05.51 | *** join/#asterisk wesphillips (~wphillips@adsl-75-53-137-6.dsl.hstntx.sbcglobal.net) |
| 21:06.14 | tvc123 | jbroome: what is your call volume like? |
| 21:06.16 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 21:06.27 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 21:06.47 | *** join/#asterisk guilhermebr (~Guilherme@187.54.218.50) |
| 21:09.00 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 21:10.50 | nny | trying to get Ring answer to work ona Polycom. If I already have a <alertInfo voIpProt.SIP.alertInfo.1.value= set in my sip.cfg do I have to change it to <alertInfo voIpProt.SIP.alertInfo.2.value= (.2) for the second alert response? Using http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 21:11.31 | nny | in other words, I already have a <alertInfo voIpProt.SIP.alertInfo.1.value= defined in my sip.cfg, to add another do I need to change the second one to 2? |
| 21:12.58 | thehar | you need to probably add a sip header |
| 21:13.07 | thehar | SIPAddHeader() |
| 21:16.06 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 21:16.20 | nny | thehar: i had more, just needed to test that. it worked changing the .1 to a .2, just wasn't sure how the XML classes worked |
| 21:16.39 | nny | thehar: had that part in my dialplan, just had duplicate XML classes :\ |
| 21:16.44 | thehar | ah |
| 21:16.56 | thehar | oh if you want to add another |
| 21:16.58 | thehar | yes you change it to 2 |
| 21:21.43 | p3nguin | canetoad: If it's not in sounds, it could be in a locale subdir such as sounds/en. |
| 21:22.29 | p3nguin | canetoad: Also, what makes you think it isn't /var/lib/asterisk/sounds? |
| 21:25.02 | jaytee | yay! another successful install of Asterisk 1.6.2.x from source |
| 21:26.43 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 21:26.46 | dimm | hello |
| 21:26.56 | dimm | silence in phone |
| 21:27.11 | thehar | pick it up |
| 21:27.14 | dimm | what start point to find trouble-place? |
| 21:27.14 | thehar | maybe it will change |
| 21:28.06 | dimm | silence, but phone-program showig that dialing (time count is run) |
| 21:28.09 | wcselby | brainiac- check the w and W options of extenspy and chanspy. Not sure if they're one-sided whisper or not. |
| 21:28.21 | wcselby | dimm, what phone-program are you using? |
| 21:28.35 | wcselby | do you have speakers / a headset hooked up to the phone program? |
| 21:28.37 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 21:29.36 | joobie | hey guys.. im playing with indications.conf and changing the busy = tones.. i then do module reload res_indications, it seems to load up properly but the busy tone i hear doesnt change when i dial out from a sip phone to a busy line? |
| 21:29.39 | joobie | any ideas?? |
| 21:29.40 | dimm | wcselby: yes, i have speaker, and mic, aprogram is something starting with X letter in windows, and Twinkle in linux (opensuse 11.3) |
| 21:30.00 | dimm | wcselby: silence is present on some nimbers |
| 21:30.10 | joobie | in console i can see "Got SIP response 486 "Busy here" back from blah" - so pretty sure it's triggering the busyu |
| 21:30.22 | dimm | wcselby: another time i can dial on the same numbers and silence going off |
| 21:30.24 | wcselby | in Xlite, do you get any sound at all? like a dialtone or dtmf tones when you dial? |
| 21:30.45 | dimm | wcselby: yes, i can hear dialtone |
| 21:30.49 | dimm | Xlite yes |
| 21:31.08 | dimm | asterisk, freepbx |
| 21:31.16 | wcselby | dimm- okay, then show us a sip debug of a call without audio and also the CLI trace of the same call |
| 21:31.21 | wcselby | ~freepbx |
| 21:31.21 | 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 |
| 21:32.23 | dimm | wcselby: i can't do this right now, see you later |
| 21:32.30 | wcselby | k |
| 21:33.13 | brainiac | wcselby: thanks... I'll give that a try... that seems to be the way to do it. |
| 21:36.12 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 21:39.26 | dimm | wcselby: how i can show sip debug of a call without audio ? |
| 21:39.46 | dimm | wcselby: is it about /var/log/... ? |
| 21:40.06 | p3nguin | You see it the same way you see a sip debug of a call WITH audio. |
| 21:40.40 | dimm | wcselby: CLI trace i understand it is a output of asterisk -g -r |
| 21:41.47 | wcselby | .... |
| 21:41.59 | wcselby | correct |
| 21:42.10 | wcselby | I need you to copy and paste the output of the CLI during a call that produces no audio |
| 21:42.31 | wcselby | also, in the CLI, you need to type "sip set debug peer XXXX" where XXXX is the name of the peer you are having trouble with |
| 21:42.37 | p3nguin | "sip set debug" or "sip set debug on" is the most common way to get a SIP debug. |
| 21:43.09 | *** join/#asterisk b0gatyr (~b0gatyr@unaffiliated/b0gatyr) |
| 21:43.14 | *** part/#asterisk b0gatyr (~b0gatyr@unaffiliated/b0gatyr) |
| 21:43.21 | *** join/#asterisk b0gatyr (~b0gatyr@unaffiliated/b0gatyr) |
| 21:44.03 | dimm | ok, next working day i do this for start test toruble |
| 21:44.30 | thehar | ;] |
| 21:44.33 | thehar | oops! |
| 21:44.58 | wcselby | we'll have none of that unsolicited winking in this channel, thehar |
| 21:45.09 | p3nguin | It usually does not take an entire day to collect sip debug. |
| 21:45.22 | thehar | hey! |
| 21:45.52 | p3nguin | Bah, my pills are not helping my back pain and I'm getting grouchy. |
| 21:46.15 | thehar | the key isn't to take more |
| 21:48.25 | p3nguin | I would take more, but then I'd run out before I'm due to run out and I'll be completely without any. |
| 21:49.13 | p3nguin | For that reason, I take meds as prescribed. |
| 21:51.21 | wcselby | p3nguin- what kind of back pain and what kind of pills do you take? |
| 21:53.46 | p3nguin | wcselby: I've got degenerative joint disease and end-plate spurring at L5-S1 along with a bulging disc and neural foraminal stenosis. I'm currently taking Percocet, but it's not helping as much as I would like. |
| 21:54.09 | wcselby | wow |
| 21:55.06 | wcselby | i've got scoliosis and I've had a bulding disc before - it's not fun. usually able to work through it all now a days (I wasn't when I had the disc), but nothing like what you've got |
| 21:55.15 | wcselby | bulging* |
| 21:56.10 | wcselby | does alieve / motrin help at all, along wtih the percocet? |
| 21:56.22 | p3nguin | I've had issues with it for years, and the pain has progressively gotten to the point where I finally went to see a doctor about pain management for it. |
| 21:57.05 | p3nguin | I'm not supposed to take any additional NSAIDs because of liver and kidney issues. |
| 21:57.29 | p3nguin | I was recently hospitalized for acute renal failure. |
| 21:57.36 | wcselby | that's what the doc's told me to take along with whatever pain meds I had during the bulging disc. I think it was vicodin |
| 21:57.37 | wcselby | ouch |
| 21:57.42 | wcselby | well, yeah, not an option for you then |
| 21:58.20 | p3nguin | My body laughs at Vicodin. It does nothing for my back. |
| 21:58.51 | wcselby | i can imagine |
| 21:59.48 | wcselby | most of the stronger pain pills would tear up my stomach, but I think a part of that was that the docs kept trying to give me the stuff the drug reps had just given them. for instance, I never had darvocet or percocet, but some of the others I did |
| 22:00.14 | wcselby | my wife could really only take darvocet - everything else made her sick and throw up |
| 22:00.20 | wcselby | but her issues were different, of course |
| 22:00.55 | p3nguin | In additional to the pain in the low back, the sacral pain is a bother when sitting or when getting up from sitting... I had been taking tramadol trying to find something to stop the pain, and it actually did help with the pain in my tail bone area, but still not helping the back pain. |
| 22:01.37 | p3nguin | I would take two tramadol 50mgs and a Tylenol. |
| 22:01.50 | wcselby | man, sorry to hear that |
| 22:02.07 | p3nguin | We've tried the class III drugs, now we're on to class II drugs. |
| 22:02.14 | wcselby | heh |
| 22:02.16 | wcselby | yeah sounds like it |
| 22:02.45 | p3nguin | If the Percocet doesn't do it, I think the next step is Dilaudid. |
| 22:03.07 | p3nguin | They gave me Dilaudid by IV when I was in the hospital -- good stuff, it is. |
| 22:03.16 | wcselby | heh |
| 22:03.24 | p3nguin | REALLY good stuff. ;) |
| 22:03.31 | wcselby | yeah |
| 22:03.33 | nny | anyone have any creative solutions for a CDR database using mysql w/asterisk that is in excess of 1 million records? Pretty decent iron running it, but mysql only uses one core per query, and it takes quite a bit. |
| 22:03.34 | wcselby | narcotic though |
| 22:03.49 | nny | The only things we have come up with is compartmentalization of the database and maybe some indexing etc |
| 22:03.51 | p3nguin | It's a c II just like the Percocet is. |
| 22:04.09 | wcselby | yeah, these are drugs that are way above and beyond anything I've taken |
| 22:04.39 | wcselby | i don't ever want to end up hooked on any of them etiher, so I've always shied away from taking too much or more than prescribed, etc |
| 22:04.48 | _Corey_ | checks Tivo to make sure "Intervention" is queued up. |
| 22:06.49 | p3nguin | Actually, these are 5/325, so I could increase the dosage before switching to another drug. |
| 22:07.04 | p3nguin | They make oxycodone in 5, 7.5, and 10. |
| 22:07.21 | p3nguin | (I think 2.5, too, but that's moot) |
| 22:07.53 | *** join/#asterisk BlackMaria (~Freischut@modemcable094.155-178-173.mc.videotron.ca) |
| 22:09.22 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 22:15.32 | jaytee | later all |
| 22:15.35 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:16.09 | wcselby | time to head out |
| 22:16.11 | wcselby | good luck p3nguin |
| 22:20.21 | tyman | will this configuration ( http://pastebin.com/aAu12YvQ ) make the 'messages' button of polycom 650 take the user directly to the voicemail pwd prompt on user's respective vmail box? |
| 22:21.08 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 22:21.11 | tyman | Can't easily test this as I'm remote and don't want to have client stuck in a "try again please" loop |
| 22:30.18 | *** join/#asterisk gnarf (~gnarf@c-98-213-203-220.hsd1.il.comcast.net) |
| 22:30.31 | gnarf | has anyone setup cisco 7975 with asterisk? |
| 22:31.29 | *** join/#asterisk simplydrew (~simplydre@c-76-22-213-70.hsd1.ky.comcast.net) |
| 22:33.19 | Nugget | I've got plenty of 45s and 65s. Should be the same as a 75 |
| 22:37.32 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 22:39.49 | gnarf | i was just curious if they were as much of a pain as the 7960 :) |
| 22:40.35 | _Corey_ | 7960s are nice and easy compared to their newer stuff |
| 22:41.24 | _Corey_ | I've found the newer 7961s and 7941s to be a little more glitchy |
| 22:42.04 | _Corey_ | We haven't bought/sold any cisco phones in 4+ years |
| 22:42.15 | JerJer | i like the SPA942 |
| 22:42.31 | gnarf | the linksys phone? :) |
| 22:42.32 | _Corey_ | The Sipura derived stuff is totally different |
| 22:42.59 | _Corey_ | Cisco rebranded them, now they're SPA5XX |
| 22:44.15 | _Corey_ | Cisco was actually at Astricon in 09 pitching them |
| 22:45.31 | gnarf | hrm |
| 22:45.37 | gnarf | this phone isn't getting DHCP |
| 22:45.38 | gnarf | grr |
| 22:45.52 | gnarf | DHCP is being offered, but it doesnt seem to accept |
| 22:46.34 | _Corey_ | They can take a while |
| 22:47.15 | frigidzephyr | i second the SPA942 |
| 22:47.16 | _Corey_ | From what I've seen, they try to hang onto whatever address they last got for several dhcp requests until they give up |
| 22:49.54 | gnarf | my boss really likes the feel/weight of the 7960/7970 series phones, i tend to agree |
| 22:50.06 | gnarf | sadly, the software on them is BS |
| 22:50.15 | gnarf | it still doesn't want to DHCP |
| 22:51.27 | _Corey_ | Yeah, my first choice for offices is still Polycom |
| 22:51.43 | _Corey_ | they're a solid phone with good SIP software |
| 22:52.03 | tyman | Can anyone help me with a rather urgent polycom issue with users not getting able to get their vmail from the messages button? I have my client standing around the phones waiting for a fix to this issue. Config's are here ( http://pastebin.com/aAu12YvQ ) |
| 22:52.43 | russellb | how much do we get if we fix it for your client? :-) |
| 22:53.34 | _Corey_ | tyman: it's a simple fix for the button |
| 22:53.58 | _Corey_ | tyman: add msg.mwi.1.callBack="555" where 555 is your vm ext |
| 22:54.18 | tyman | _Corey_: isn't that in my config? |
| 22:54.27 | tyman | 8500 is the vmail ext |
| 22:54.38 | _Corey_ | i'm guessing because it's in a duplicate section |
| 22:54.43 | Qwell | the msg.mwi stuffs goes in <mwi>, not <msg> |
| 22:55.01 | Qwell | at least your first block there |
| 22:55.15 | Qwell | the bypassInstantMessage is right, I believe |
| 22:55.28 | tyman | this is UCS 3.3.1 |
| 22:55.30 | _Corey_ | yeah, i think it should all be in the second part |
| 22:55.51 | tyman | these are separated into reg-basic.cfg and reg-advanced.cfg |
| 22:56.14 | tyman | the reg-adv is the top snippet and the reg-basic is the bottom snippet |
| 22:56.53 | Qwell | http://pastebin.com/WJv5ynQi |
| 22:57.23 | Qwell | err |
| 22:57.31 | Qwell | drop the first msg.mwi on line 6 |
| 23:01.01 | tyman | Qwell: the UCS 3.3.1 cfg templates have no top-level mwi key |
| 23:01.32 | tyman | the only mwi attributes are from the snippets from my original pastie |
| 23:02.31 | tyman | top level key is <msg> with mwi attributes |
| 23:05.13 | tyman | Should callbackMode by set to "subscribe" instead of "contact"? The polycom docs aren't clear to me on that topic |
| 23:06.17 | Qwell | http://pastebin.com/6FBdYQgK |
| 23:06.21 | Qwell | that should be the minimum required |
| 23:07.29 | Qwell | you'll probably want to use different open/close tags for <msg> |
| 23:07.35 | Qwell | you get the idea. |
| 23:10.41 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 23:20.35 | *** join/#asterisk Pio (~sean@207.181.14.102) |
| 23:21.26 | Pio | when my ATA gets a call with no caller ID, asterisk seems to be setting the caller ID to the name of the sip peer it comes in on.. i'd like to change that to 'private caller' or something, is callerid= in sip.conf the appropriate way to change that? |
| 23:22.07 | Pio | the sip.conf documentation says callerid is used when nothing else is available, but then it says "Defaults to asterisk".. and im not getting "asterisk", im getting the name of my sip peer |
| 23:22.56 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 23:25.36 | neurosys | leifmadsen: Finished and working! :D |
| 23:25.45 | neurosys | leifmadsen: All about Originate :) |
| 23:25.45 | leifmadsen | neurosys: nice! |
| 23:25.49 | leifmadsen | heck ya :) |
| 23:25.56 | leifmadsen | glad we got you pointed in the right direction |
| 23:26.02 | neurosys | leifmadsen: Would you like the script via PB? |
| 23:26.15 | leifmadsen | nah that's ok |
| 23:26.17 | leifmadsen | thanks tho! |
| 23:26.25 | neurosys | Cool :) |
| 23:26.40 | neurosys | Thank you! :) |
| 23:33.11 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 23:34.22 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 23:34.28 | *** join/#asterisk guilhermebr (~Guilherme@187.54.218.50) |
| 23:37.29 | *** join/#asterisk docfxit_ (4b4f060a@gateway/web/freenode/ip.75.79.6.10) |
| 23:39.32 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 23:39.39 | docfxit_ | How can I add IP addresses that are allowed to us asterisk? |
| 23:40.30 | p3nguin | Did you consider your firewall? |
| 23:40.47 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 23:42.57 | docfxit_ | I'm wondering if locking it down with the firewall would be the best way to go. I need to allow all the phones and for me to VNC into Ubuntu. |
| 23:46.23 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 23:46.33 | diemos | herroz |
| 23:48.02 | frigidzephyr | hai |
| 23:56.07 | bmoraca_work | docfxit_: localnet in sip.conf's general context, as well as permit and deny at the peer level. alternatively, iptables could be used to directly restrict your sip port |