| 00:00.01 | Psykick | DrunkenHME: ActiveX control |
| 00:00.10 | *** join/#asterisk Camisa (n=Camisa@c-67-176-161-7.hsd1.in.comcast.net) |
| 00:00.37 | DrukenHME | ~amp |
| 00:00.40 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 00:00.41 | *** join/#asterisk FastJack (i=fastjack@p5091FD7F.dip.t-dialin.net) |
| 00:00.43 | Psykick | DrunkenHME: http://203.170.71.26/iax-webTeleFon2/start.asp |
| 00:00.47 | *** part/#asterisk [av]bani (n=[av]bani@washuu.anime.net) |
| 00:01.16 | mog_work | lol |
| 00:01.24 | mog_work | that is awesome |
| 00:01.25 | infinity1 | Psykick: what software are you using for im/presence? |
| 00:02.09 | mog_work | should use astjab... |
| 00:02.21 | Psykick | crappy firefly |
| 00:02.36 | Psykick | I like this iax web phone better |
| 00:02.49 | Flauto | psykick, what is iaxwebfone for? |
| 00:03.01 | Flauto | what does it do? |
| 00:03.33 | Psykick | once you've got it setup you can place calls using click2dial using your own DB of phone numbers |
| 00:03.33 | *** join/#asterisk exstatica (i=exstatic@haw-207-182-243-123.vel.net) |
| 00:03.57 | Psykick | YOU of course need to do the programming to get the numbers from the DB |
| 00:04.07 | Psykick | but it takes care of all the telephony side of things for you |
| 00:04.27 | Psykick | including allocating which device to use for mic + speakers |
| 00:04.28 | Flauto | nice, but it is too deep for me |
| 00:04.43 | Psykick | I'm using it in a PHP app I've put together |
| 00:04.58 | Psykick | nice simple click2dial company phone book |
| 00:06.01 | kippi1 | ye |
| 00:06.15 | *** join/#asterisk anon-troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 00:07.54 | *** join/#asterisk [Adamo] (n=adamo@32.59.2.18) |
| 00:08.57 | *** part/#asterisk [Adamo] (n=adamo@32.59.2.18) |
| 00:10.41 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 00:10.55 | *** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 00:12.05 | *** join/#asterisk DovidB (n=sentback@ool-44c05dde.dyn.optonline.net) |
| 00:17.30 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 00:19.21 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 00:19.27 | ObsidianX | hey folks, in the manual it describes EAGI as being able to send the incoming audio on "file descriptor three"... how would i go about accessing that |
| 00:23.37 | *** join/#asterisk cnet2 (n=nada@200.122.157.91) |
| 00:25.05 | cnet2 | hi, has anyone had problems installing the TDM2400p? I have xorcom installation (debian + asterisk 1.0.9) and the digium wildcard is not recognized, and show this on lspci 'Ethernet controller: Unknown device d161:2400 (rev 11)' |
| 00:26.00 | cnet2 | could it be irq conflicts? |
| 00:30.43 | ObsidianX | what kernel do you have |
| 00:33.41 | Darwin35 | we regret to inform you but the asterisk project just had its plug pulled. tiiiime to grab what you can and run for it...... |
| 00:33.41 | ruud_org | lspci unknown device just means the lspci program doesn't have the manufacturer id->name mapping for this particular card, doesn't mean anything about whether or not the kernel itself has appropriate drivers |
| 00:33.58 | ruud_org | (i.e. you can have a working card with unknown device shown up in lspci and vice versa) |
| 00:37.20 | Darwin35 | O wit they just repluged asterisk in . all is ok |
| 00:38.55 | *** join/#asterisk Jestre (n=ack@dargo.trilug.org) |
| 00:39.07 | Zodiacal | anyone know if i need a fxo module for each line? so for one 4 port TDM400P card i would need 4 x100m pxo modules? |
| 00:39.21 | Zodiacal | pxo = fxo |
| 00:39.26 | mog_work | yes |
| 00:39.32 | mog_work | for each analog line you want |
| 00:39.43 | mog_work | you need an fxo to connect to pstn |
| 00:39.56 | Zodiacal | and if im using sip phones i woudn't need fxs right? |
| 00:40.19 | Zodiacal | fxs modules that is |
| 00:40.23 | Zodiacal | i think i get it |
| 00:40.26 | Zodiacal | mog_work Thank You! |
| 00:40.35 | mog_work | yeah you dont need it |
| 00:41.00 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 00:41.11 | alephcom_ | Hello everyone. |
| 00:42.03 | mog_work | hello |
| 00:42.26 | xheliox | Howdy |
| 00:44.22 | *** join/#asterisk darkskiez (n=darkskie@bb-195-172-53-125.ukonline.co.uk) |
| 00:52.08 | Darwin35 | excuse me watch where your walking all these cables on the floor lead to something |
| 00:52.33 | justinu | Darwin35: wtf? |
| 00:52.40 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 00:53.36 | Darwin35 | wtf =wheres the fruit |
| 00:53.58 | justinu | are you like some semi-intelligent bot or something? :P |
| 00:54.34 | Ciber | Darwin35 hi! |
| 00:54.43 | Ciber | maybe not |
| 00:56.22 | Darwin35 | no |
| 00:56.26 | Darwin35 | just having fun |
| 00:56.41 | Darwin35 | letting the brain got semi mush and back again |
| 00:56.47 | justinu | lol |
| 00:56.54 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 00:56.58 | *** join/#asterisk Jick (n=Jick@209-83-240-53-static.dsl.oplink.net) |
| 00:57.33 | warthawg | can anyone point me towards a sample [inbound] that just rings the phone and lets me answer it? |
| 00:57.34 | Jick | in my dialplan, how can I configure the timeout (t) extension to repeat the last message, rather than hang up on the caller? |
| 00:57.36 | justinu | darwin35: you ever said that to someone? |
| 00:57.42 | justinu | [16:52] Darwin35: excuse me watch where your walking all these cables on the floor lead to something |
| 00:58.22 | Darwin35 | eveeryday when I work in the noc |
| 00:58.41 | Darkhalf | A noc with cables on the floor? Awesome-o. |
| 00:58.46 | warthawg | does the noc have raised flooring? |
| 00:59.10 | justinu | reminds me of the phillipines telcom office at one wilshire |
| 00:59.16 | justinu | fucking cables everywhere! |
| 00:59.31 | Darkhalf | It's been a while since I've helped myself to cat5 spaghetti. |
| 00:59.31 | warthawg | in manilla? |
| 00:59.35 | Darwin35 | yes but they run cabless across the floor to cross connect sometimes |
| 01:03.34 | wunderkin | wart, heh thats in LA |
| 01:04.09 | wunderkin | almost got a house out there justin, we could have been neighbors :P |
| 01:04.19 | Jick | when I set my extensions.conf up, I temporarily set the timeout section to play just a gsm and hangup, but now it's time I change it to something helpful. How can I set it to repeat the last thing the user heard? Surely this is a popular function of the timeout extension... |
| 01:04.22 | wunderkin | and qwell.. and a few others im sure ;D |
| 01:06.07 | Darwin35 | man 6pm and feel like 9 |
| 01:06.26 | justinu | qwell lives about as far away from me as you can be, and still say you live in LA |
| 01:06.30 | justinu | like 70 miles or something |
| 01:07.09 | Nugget | why would anyone want to be able to say that they live in LA? :) |
| 01:07.20 | justinu | i don't think anyone wants to |
| 01:07.26 | justinu | it's just where I live |
| 01:07.54 | Darkhalf | Go a hundred miles south. |
| 01:08.06 | Darkhalf | ...And a bit east. |
| 01:08.15 | *** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) |
| 01:09.07 | justinu | temecula? |
| 01:09.14 | [av]bani | hm, gxp2000 isnt as crappy as i thought it would be |
| 01:09.30 | hardwire | its not supposed to be crappy |
| 01:09.38 | Darkhalf | justinu: Even more south. But you got the right idea! |
| 01:09.40 | [av]bani | for $80 one doesnt expect much |
| 01:09.52 | Ciber | that's good to hear |
| 01:09.59 | Ciber | i just ordered 12 of those lol |
| 01:10.06 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
| 01:10.20 | justinu | escondido |
| 01:10.21 | [av]bani | speakerphone has serious echo, some volume issues with handset |
| 01:10.22 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 01:10.25 | [av]bani | functionality wise seems ok |
| 01:10.36 | _DAW-LAPTOP | hello |
| 01:10.40 | justinu | apparently the latest firmware solves that handset volume issue |
| 01:10.47 | Darkhalf | justinu: I'm in the SD area, but the idea I was trying to convey was "away from LA". |
| 01:11.10 | [av]bani | too bad the buttons on the side are only speed dial, not usable otherwise |
| 01:11.41 | Ciber | now if only i could stop asterisk from thinking my fxs channel is not an fxo |
| 01:11.45 | [av]bani | it also doesnt pick up timezone from dhcp |
| 01:12.20 | justinu | well, i agree, that this city sucks |
| 01:12.37 | justinu | i'm always getting it thrown in my face too, which doesn't help the situation :P |
| 01:13.31 | Ciber | i'm seriously thinking of throwing this damn server out the window |
| 01:15.02 | *** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 01:16.01 | Ciber | yay finally fixed |
| 01:16.04 | Ciber | stupid thing |
| 01:16.13 | justinu | now throw it out the window |
| 01:16.17 | Ciber | lol |
| 01:16.24 | Ciber | was close to |
| 01:16.31 | Ciber | was driving me nuts all day |
| 01:17.21 | Ciber | zap identifier set as g0 was ringing my other extension even though it was a fxs not a fxo |
| 01:17.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 01:17.45 | Ciber | changed it to 4 and it's now sending the damn calls to the fxo channel |
| 01:18.17 | silentfury | i realize this is probably the wrong channel, but I'm looking for help with sipX.. |
| 01:18.23 | Ciber | beyond me why it's sending them to channel 1 when it's in group 1 not 0 |
| 01:19.42 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 01:20.03 | Ciber | grrr now it's saying the person at ext 201 is unavailable |
| 01:20.14 | Ciber | i can make calls from it just fine |
| 01:22.00 | Ciber | ahh there we go fixed |
| 01:24.57 | *** join/#asterisk svenl (n=sven@AStrasbourg-251-1-52-28.w82-126.abo.wanadoo.fr) |
| 01:25.28 | *** join/#asterisk silentfury_ (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 01:29.20 | *** part/#asterisk silentfury_ (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 01:32.47 | litage | are there softphones that use g729? |
| 01:34.44 | justinu | eyebeam |
| 01:35.09 | Jick | does anyone know how I can set zaptel and wctdm to be modprobed on startup and ztcfg and asterisk to be launched on startup in CentOS v4.2? |
| 01:35.30 | mog_work | type make config in both source trees |
| 01:36.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 01:37.19 | svenl | mmm, i just built the zaptel drivers on powerpc, with 2.6.15, and got : insmod: error inserting 'wctdm.ko': -1 Invalid module format |
| 01:37.29 | svenl | anyone seen that already ? |
| 01:45.47 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:46.55 | *** part/#asterisk Jestre (n=ack@dargo.trilug.org) |
| 01:58.54 | *** join/#asterisk scolsuckz (n=scolsuck@202.58.252.15) |
| 01:59.18 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:01.43 | *** join/#asterisk Pegger (n=peg@pool-68-163-192-243.bos.east.verizon.net) |
| 02:05.24 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 02:09.33 | Jick | hmmm |
| 02:09.55 | *** join/#asterisk Mavantix (n=mavantix@69-168-33-232.chvlva.adelphia.net) |
| 02:11.07 | Jick | okay. i got zaptel and asterisk to stick service scripts in /etc/init.d by running make config and i can now add those scripts to the runlevels so they start on boot, but what about the wctdm kernel module? how do i configure that to start at boot? |
| 02:14.13 | Mavantix | anyone know why my outbound SIP line doesn't generate a "ringing noise" when it's calling out? It's dead silent, even though it does place the call. (Broadvoice.com) |
| 02:14.34 | Mavantix | Is there a dialplan command or something for this? |
| 02:15.19 | *** join/#asterisk svenl_ (n=sven@AStrasbourg-251-1-48-86.w82-126.abo.wanadoo.fr) |
| 02:15.31 | Mavantix | or maybe more specifically, should I be using the Ringing() cmd? |
| 02:15.39 | Mavantix | ...in my dialout plan. |
| 02:16.06 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 02:17.39 | *** join/#asterisk zu (n=zu@38-pool1.ras14.floca.alerondial.net) |
| 02:17.43 | zu | hy all |
| 02:20.21 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 02:21.24 | Mavantix | damn it, dial(,,r) is my answer. Thanks for listening to me ramble ;) |
| 02:22.24 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
| 02:23.57 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:24.11 | litage | when testing codecs and trying to achieve "high quality" calls, what can be configured that affects call quality? |
| 02:24.30 | justinu | usually nothing |
| 02:24.34 | justinu | possible the rate |
| 02:24.35 | *** join/#asterisk kart_179 (n=kart@201.3.87.81) |
| 02:26.05 | litage | then how do you control the amount of jitter, echo cancellation, etc? |
| 02:26.18 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:26.40 | zu | hardware echo cancellation :) |
| 02:26.50 | justinu | eyebeam has a omode you can run it in that will drop packets and add jitter |
| 02:26.58 | justinu | asterisk does too, on iax |
| 02:28.41 | litage | if you're only using sip and want to determine [a] which codec(s) work best for your situation, and [b] what "settings" for those codecs are best, what can you do? |
| 02:29.12 | justinu | if you were really serious, you'd buy some kind of RTP test generation software/hardware |
| 02:29.18 | justinu | from somebody like ixia, i guess |
| 02:29.21 | zu | if you need to determine what codec you need to use you need to start with bandwidth requirements vs channels |
| 02:29.34 | justinu | but i'm not sure what you mean by "settings" |
| 02:30.15 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 02:31.20 | litage | justinu: i'm not sure what i mean by "settings" either. i was under the impression that when you select a codec to use, you can configure them slightly so that they're better tailored for your needs |
| 02:31.27 | litage | is that incorrect? |
| 02:31.49 | justinu | as far as I know, the codecs have no real settings you can tweak |
| 02:32.07 | justinu | there may be things like jitter buffers you can play with, but the codecs themselves expect a steady stream of packets |
| 02:32.21 | zu | thats what qos is for |
| 02:32.42 | justinu | you can artificially induce things like jitter and packet loss to see how certain codecs sound in various network conditions |
| 02:32.42 | zu | choosing the correct codec is just doing the bandwith/quality calc |
| 02:32.48 | litage | ah i see |
| 02:33.14 | zu | do you want 30kbits a sec tx/rx or 300 |
| 02:33.22 | litage | within asterisk, what can be "configured" to modify the amount of jitter, echo, and whatever else occurs? |
| 02:33.35 | *** join/#asterisk awptix (n=jonathan@vzadsl-pppoe-nyny-66-97-3-13.tellurian.net) |
| 02:33.36 | zu | or in the case of sip g729 53 kbits a sec tx/rc |
| 02:33.39 | zu | rc/rx |
| 02:33.47 | awptix | Hey guys, anyone know of a simple Asterisk Web GUI? |
| 02:33.59 | awptix | just something to be able to check voicemail, etc. with |
| 02:34.11 | mrdigital | does anyone have pics of there * Setup? |
| 02:34.19 | mrdigital | awptix: yes |
| 02:34.22 | mrdigital | Voiceone |
| 02:34.23 | [av]bani | sip is no fun with nat |
| 02:34.26 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 02:34.28 | test34 | anyone using packet cable ? |
| 02:34.41 | _Vile | awptix: http://www.voip-info.org/wiki-Asterisk+GUI for alternatives |
| 02:35.02 | awptix | mrdigital: is that free? |
| 02:35.06 | mrdigital | voiceone yes |
| 02:35.12 | Nugget | http://slacker.com/photos/computers/SlackerNOC <-- pics. the asterisk server is in there somewhere. |
| 02:35.41 | mrdigital | nugget: your systems? |
| 02:35.45 | Nugget | yeah |
| 02:35.50 | mrdigital | ever heard of a desk? :) |
| 02:35.54 | Nugget | *shrug* |
| 02:35.59 | Nugget | I go in there about once every three months |
| 02:36.27 | *** join/#asterisk NoVaZuR (n=novazur@ALamentin-104-1-35-252.w81-248.abo.wanadoo.fr) |
| 02:36.44 | mrdigital | what room is it? |
| 02:36.49 | Nugget | my server room |
| 02:36.49 | NoVaZuR | hi ! |
| 02:36.49 | mrdigital | Server room? |
| 02:36.50 | awptix | hmm... slacker.com = nice domain :) |
| 02:36.54 | mrdigital | what kinds of boxes |
| 02:37.39 | Nugget | from left to right: quad xeon 450 linux, athlon something, linux, celeron openbsd, dual opteron freebsd, athlon freebsd. |
| 02:37.57 | NoVaZuR | is it possible with a IAX softphone, to call a SIP number like foo@domain ? |
| 02:37.58 | mrdigital | what you use em for? |
| 02:38.00 | Nugget | and a 6502 running atari dos. :) |
| 02:38.08 | Nugget | just stuff |
| 02:38.23 | Nugget | http://slacker.com/~nugget/stuff/SlackerNOC.pdf |
| 02:38.37 | Nugget | ouch |
| 02:38.43 | file[desk] | tastes like chicken |
| 02:39.03 | mrdigital | nugget: can i pm> |
| 02:39.11 | Nugget | I don't know. has it worked for you in the past? |
| 02:39.23 | mrdigital | ?? |
| 02:39.25 | _Vile | see #irc for help |
| 02:39.32 | mrdigital | no i mean can i pm you |
| 02:39.34 | mrdigital | lol |
| 02:39.38 | Nugget | I have no idea. try and see. |
| 02:39.43 | mrdigital | lol nm |
| 02:39.53 | litage | in terms of call/voice quality, what is configurable within asterisk? |
| 02:40.07 | Nugget | I dunno what irc network it is where people are expected to ask permission to pm first. I've never been on it, that's for sure. |
| 02:41.04 | loko | file[desk] you get a ticket? |
| 02:41.09 | Nugget | I need to take another photo. there's a few more machines in there now. |
| 02:42.05 | file[desk] | loko: I'll talk on vr-oasis in a sec |
| 02:42.19 | loko | k |
| 02:42.39 | awptix | Nugget: some people tend to think that it's not polite to PM without permission |
| 02:43.01 | awptix | and in bad netizen practice. |
| 02:43.31 | justinu | i think it's impolite for people i don't know to call me, but it doesn't stop them |
| 02:44.21 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-a2403a9a2fa4b146) |
| 02:44.51 | *** part/#asterisk NoVaZuR (n=novazur@ALamentin-104-1-35-252.w81-248.abo.wanadoo.fr) |
| 02:45.05 | Nugget | yeah, I just don't know why people think there. |
| 02:45.11 | Nugget | where is the place that's training people to think that? |
| 02:45.21 | hhoffman | hi, I'm hooking up a payphone to a asterisk box and am wondering the best way to capture the tone generated when a coin is dropped. any ideas? |
| 02:45.30 | justinu | maybe it was qwell yelling at people for pm'ing him |
| 02:45.37 | Qwell | likely |
| 02:45.40 | Qwell | What'd I miss? |
| 02:45.57 | Nugget | heh |
| 02:46.10 | *** join/#asterisk Brixius (n=Brixius@c-69-180-132-70.hsd1.mn.comcast.net) |
| 02:49.28 | Qwell | here's the way I see it |
| 02:49.36 | Qwell | If you msg me, 2 things happen |
| 02:49.45 | Qwell | 1) I get distracted, and can't help other people at the same time. |
| 02:49.56 | Qwell | 2) Nobody benefits from my help, except the selfish ass who msg'd me |
| 02:50.04 | zu | Qwell: you need a Qwell queueing system |
| 02:50.17 | Qwell | zu: I have one |
| 02:50.21 | Qwell | cash gets priority |
| 02:50.28 | zu | lol |
| 02:50.38 | zu | Thats why I have a paypal account :) |
| 02:50.51 | justinu | qwell: i'm on your side, i think you have a right to yell at people for msg'ing you |
| 02:51.40 | DrukenHME | who's got toronto calling? |
| 02:53.36 | litage | what settings/configuration in asterisk can be used to control call/voice quality? |
| 02:53.37 | justinu | what do you need? |
| 02:53.47 | justinu | litage: you can chose the codec, that's it. |
| 02:54.05 | litage | justinu: you can't do anything else within asterisk? |
| 02:54.16 | litage | [to modify call quality] |
| 02:54.21 | justinu | no |
| 02:54.30 | wunderkin | /msg justinu can u plz help me install asterisk@home k plz thx |
| 02:54.38 | justinu | lol |
| 02:55.04 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 02:55.06 | litage | justinu: what about the frame or sampling rate? |
| 02:55.12 | justinu | that's handled by the codec |
| 02:55.29 | Qwell | well... |
| 02:55.30 | litage | ah |
| 02:55.38 | Qwell | rtp packetization |
| 02:55.47 | Qwell | there is something on the tracker, to implement that |
| 02:55.55 | litage | Qwell: tracker? |
| 02:56.01 | Qwell | bugs.dgium.com |
| 02:56.06 | Qwell | digium* |
| 02:56.15 | justinu | ah, if you want asterisk to act as a TDM gateway, that might be handy |
| 02:56.34 | justinu | but if you're acting as a soft switch or simply rtp proxy, that's hadnled by the terminating devices |
| 02:56.39 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 02:58.04 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:58.10 | justinu | oh, and it'd be useful if asterisk is being used as a media server |
| 02:58.32 | hhoffman | can anyone tell me how asterisk acts when a dmtf tone is sent over the fxo card after asterisk has answered the line? |
| 02:58.59 | hhoffman | er, dtmf |
| 03:01.13 | hhoffman | ah, very good :-) |
| 03:01.47 | *** join/#asterisk diego_br (n=diego@200-231-134-59-mns.cpe.vivax.com.br) |
| 03:02.00 | DrukenHME | hhoffman: the fxo should just create the tone... |
| 03:02.12 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 03:02.46 | drumkilla | well, it recognizes it? |
| 03:02.50 | drumkilla | i don't quite understand the question :) |
| 03:03.20 | drumkilla | and then the DTMF is passed on to whatever is currently handling the channel ... |
| 03:03.41 | Qwell | drumkilla: asterisk is! :p |
| 03:03.41 | hhoffman | DrunkeHME: I have a setup that is Payphone->CO->fxo->asterisk ... When I drop a coin in the payphone a tone is generated, I'd like asterisk to do something when that happens |
| 03:03.48 | awptix | Nugget: it's kind of like someone calling you on the phone, uninvited, trying to pester you. |
| 03:03.52 | drumkilla | it may be some application, or the channel may be bridged (in a call with another channel), so the dtmf event would just go across the bridge ... |
| 03:04.07 | DrukenHME | oh... |
| 03:04.36 | hhoffman | drumkilla: can I write an extension to do something with the tone? |
| 03:04.47 | DrukenHME | you need to get together with uhm,... shit.. who was that... Drray i think... |
| 03:05.22 | drumkilla | hhoffman: yes |
| 03:05.32 | DrukenHME | well, the tones the phone makes, are for the co... |
| 03:05.35 | justinu | some guy was talking about interfaces for bell coin control |
| 03:05.43 | *** join/#asterisk jef_ (i=fischer@p54846968.dip.t-dialin.net) |
| 03:05.52 | DrukenHME | it tells the telco that you paid for the call... and they can connect you.. |
| 03:06.05 | hhoffman | drumkilla: ah! ok... so, if I'm at the console I'll presumably see what the tone translates to? |
| 03:06.09 | DrukenHME | it's the OLD way of doing it |
| 03:06.18 | justinu | it's the cool way of doing it :P |
| 03:06.26 | DrukenHME | i don't belive it's dtmf |
| 03:06.32 | drumkilla | wait, this is an arbitrary tone? |
| 03:06.33 | justinu | right, it's not |
| 03:06.47 | drumkilla | ok, then that will require code to be written ... |
| 03:06.47 | justinu | it's the bell coin control stuff real payphones use |
| 03:07.04 | justinu | i bet spandsp/libsupertone can do it |
| 03:07.07 | hhoffman | coin is 1700Hz+2200Hz |
| 03:08.01 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 03:08.38 | DrukenHME | hhoffman: uhmm... i think you will find, diffrent tones.... |
| 03:08.52 | DrukenHME | one for .05, .10 and .25 |
| 03:08.53 | DrukenHME | :) |
| 03:08.58 | *** part/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
| 03:09.32 | hhoffman | DrunkenHME: sorry... I should have clarified, this is for a quarter... that's all we are going to accept |
| 03:09.47 | hhoffman | this is for a research project, not a real life production setup :-) |
| 03:10.14 | Qwell | high rollers! |
| 03:10.36 | DrukenHME | was just pointing that out :) |
| 03:10.44 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 03:10.56 | Qwell | I can only afford to test with nickels. :( |
| 03:10.59 | DrukenHME | my payment does it's own control |
| 03:11.42 | hhoffman | hehe :-) |
| 03:14.32 | DrukenHME | has anyone actually gotten privacy manager to work? |
| 03:19.28 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 03:25.31 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 03:26.06 | Brixius | anyone gotten cdr_odbc to work? |
| 03:26.18 | DrukenHME | it works |
| 03:26.28 | DrukenHME | just don't ask me how... :) hehehehe been too damn long |
| 03:26.42 | Corydon76-home | Yeah, works fine |
| 03:27.12 | Corydon76-home | Anyone gotten func_odbc to work? |
| 03:27.17 | Brixius | arrrg, pain in the ass then.... well for me, although it would probably help if I knew how to test that unixODBC was installed properly... |
| 03:27.39 | Corydon76-home | Brixius: isql db user pass |
| 03:28.00 | Corydon76-home | db, of course, is the DSN that you configured in odbc.ini |
| 03:28.15 | Corydon76-home | You DID configure odbcinst.ini and odbc.ini, right? |
| 03:28.19 | awptix | hmm |
| 03:28.22 | awptix | did it die in here? |
| 03:28.36 | clyrrad | anyone here interested in setting up fax to email, willing to pay please PM me |
| 03:30.03 | Brixius | ya I setup odbcinst.ini and odbc.ini |
| 03:30.12 | litage | what can be done on a server (not necessarily within asterisk) to control call/voice quality? |
| 03:31.24 | Brixius | I've installed myodbc and tested that I can connect to the database with mysql client from the * box and query it, so security is correct. |
| 03:31.34 | wunderkin | anyone here interested in violating lame patents.. hah.. |
| 03:32.00 | justinu | fax to email is patented? |
| 03:32.15 | wunderkin | ya according to bkw hes gettin sued for it |
| 03:32.32 | wunderkin | lame lame lame |
| 03:32.48 | justinu | by jfax? |
| 03:33.03 | wunderkin | shrug, maybe |
| 03:33.12 | DrukenHME | wuts func_odbc? |
| 03:35.17 | clyrrad | Can anyone here setup fax to email on an Asterisk server? |
| 03:36.34 | Brixius | Corydon76-home: I get '[ISQL]ERROR: Could not SQLConnect' when I try to connect using isql |
| 03:36.46 | Qwell | Corydon76-home: I have. ;) |
| 03:36.56 | Corydon76-home | Well, try adding -vv before the dsn |
| 03:37.07 | Corydon76-home | and it'll probably tell you why |
| 03:38.52 | sivana | lol... wow... now they're opening private mail in the name of security |
| 03:39.02 | sivana | amazing... nice privacy |
| 03:40.14 | sivana | lmao |
| 03:40.25 | hhoffman | huh? |
| 03:40.29 | sivana | "Customs and Border Protection is charged with making sure that terrorists and terrorists' weapons don't enter the country," |
| 03:40.37 | sivana | kinda hard to put them in a letter and mail it |
| 03:40.59 | sivana | that's funny |
| 03:41.05 | hhoffman | url? |
| 03:41.08 | sivana | the excuses they use to violate people's privacy |
| 03:41.14 | sivana | http://www.cnn.com/2006/US/01/09/terrorism.mail.reut/index.html |
| 03:41.34 | hhoffman | danke |
| 03:42.29 | sivana | I wish people cared more about government and keep them accountable |
| 03:43.23 | hhoffman | sigh, me too |
| 03:44.00 | Brixius | I'm guessing that will be a topic of discussion at the Infragard meeting I'm supposed to goto tomorrow. |
| 03:44.38 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:44.53 | sivana | I know there's a balance that needs to be struck.. but geez |
| 03:45.15 | hhoffman | Brixius: not to mention: http://www.schneier.com/blog/archives/2006/01/anonymous_inter.html |
| 03:45.25 | *** join/#asterisk ctooley (n=SugarGue@jc1-111.moment.net) |
| 03:46.03 | Brixius | I think they need to be kept accountable too. And the police should be setting an example, not running stop signs and lights to get to lunch as I've seen them do. |
| 03:46.30 | sivana | yup |
| 03:47.00 | *** part/#asterisk ctooley (n=SugarGue@jc1-111.moment.net) |
| 03:47.41 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 03:47.55 | shmaltz | Brixius, how do you know that what you seen was for lunch? |
| 03:48.07 | sivana | heh |
| 03:48.24 | Brixius | Because I was going to the same place and saw them at the counter ordering. |
| 03:48.38 | justinu | how dare you question legal authority in a time of war! |
| 03:48.44 | sivana | lol |
| 03:48.48 | sivana | time of war |
| 03:48.51 | sivana | that's funny |
| 03:49.12 | awptix | "This is Adam Charlie Four, we're in pursuit of a double cheeseburger and a 20oz coke. Over." |
| 03:49.17 | justinu | you're either with us, or you're against us |
| 03:49.17 | shmaltz | Brixius, the call got cancelled on the way :P |
| 03:49.29 | shmaltz | lol |
| 03:49.40 | sivana | justinu: you're funny |
| 03:49.41 | Brixius | If it's official, have the lights on and be running code... |
| 03:50.03 | justinu | sivana: the world changed on 9/11 |
| 03:50.11 | justinu | :P |
| 03:50.24 | sivana | maybe... but most of it is just Bush |
| 03:50.38 | shmaltz | sivana, most of what is just bush? |
| 03:50.38 | justinu | i'm just using the rhetoric |
| 03:50.50 | justinu | being a good little mouthpiece |
| 03:50.56 | shmaltz | sivana, you not even American, so stfu |
| 03:51.02 | sivana | lol |
| 03:51.15 | justinu | yeah, your opinion is not welcome |
| 03:51.18 | sivana | shmaltz: have you heard of the lumber disputes with Canada and US? |
| 03:51.34 | Brixius | Ok with isql -vv dsn un pw I get '[unixODBC][Driver Manager]Data source name not found, and no default driver specified' |
| 03:51.37 | shmaltz | sivana, I just want to see you ruthless canadaians react if the falls are destroyed |
| 03:51.38 | sivana | I bet you say no |
| 03:51.47 | justinu | lol |
| 03:51.48 | shmaltz | sivana, no, whats that? |
| 03:51.51 | Brixius | I have it in /etc/odbc.ini |
| 03:52.24 | justinu | need any wood? |
| 03:52.27 | sivana | softwood lumber dispute... you don't know because it's not on your news... |
| 03:52.35 | sivana | only the war on terrorism is on your news |
| 03:52.38 | justinu | "need any wood?" - george w. bush |
| 03:52.46 | sivana | heh |
| 03:52.49 | shmaltz | Brixius, what do you have in odbc.ini? the 20 oz coke and burgers? |
| 03:53.00 | Nugget | the nuggets! |
| 03:53.04 | shmaltz | sivana, any urls? |
| 03:53.16 | Brixius | I've put export ODBCINI=/etc/odbc.ini in the asterisk startup script and have it in my exports list on my shell prompt. |
| 03:53.30 | sivana | shmaltz: http://www.google.ca/search?q=canada+softwood+lumber+dispute&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official |
| 03:53.31 | Brixius | sorry, this is back to the cdr_odbc issue I was having... |
| 03:53.55 | justinu | hey, stay on topic |
| 03:54.01 | justinu | the candians and americans were fighting |
| 03:54.04 | Brixius | Corydon76-home: Ok with isql -vv dsn un pw I get '[unixODBC][Driver Manager]Data source name not found, and no default driver specified' |
| 03:54.06 | justinu | lets see who wins |
| 03:54.07 | sivana | lol |
| 03:54.13 | Brixius | Corydon76-home: I've put export ODBCINI=/etc/odbc.ini in the asterisk startup script and have it in my exports list on my shell prompt. |
| 03:54.26 | shmaltz | sivana, it rings a bell, it went down though I think |
| 03:54.29 | sivana | the nafta tribunal already ruled in favor of Canada and the US is ignoring it |
| 03:54.40 | Brixius | sorry, took a hard left there... |
| 03:54.44 | Brixius | haqha |
| 03:54.53 | justinu | sivana: oh, we tend to ignore rulings that aren't in favor of us |
| 03:54.57 | justinu | deal with it |
| 03:55.00 | sivana | you think :) |
| 03:55.11 | sivana | anyhow |
| 03:55.21 | justinu | oceania, tis of thee! |
| 03:55.46 | justinu | http://en.wikipedia.org/wiki/Oceania_%28fiction%29 |
| 03:55.51 | shmaltz | this explains it right |
| 03:55.52 | Corydon76-home | Brixius: Use pastebin to paste the contents of your odbcinst.ini and odbc.ini files |
| 03:55.52 | shmaltz | http://www.cbc.ca/news/background/softwood_lumber/ |
| 03:55.54 | Corydon76-home | ~pb |
| 03:55.56 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 03:56.13 | shmaltz | sivana, nafta is only there to serve the bigger one, not the smaller one, dont kid yourself |
| 03:56.28 | sivana | I know how the US works |
| 03:56.34 | justinu | good |
| 03:57.22 | *** join/#asterisk licued (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 03:57.29 | justinu | http://en.wikipedia.org/wiki/Ingsoc |
| 03:58.32 | shmaltz | sivana, it has nothing to do with the US, it has always been that way and will always stay that way, comitees and orgenizations are only there to serv the one in power, just consider this quote: |
| 03:58.33 | shmaltz | http://www.quotecha.com/quotes/quotation_16578.html |
| 03:59.14 | justinu | except that the USA is a federal republic :P |
| 03:59.20 | Brixius | Corydon76-hom: http://pastebin.ca/36324 |
| 03:59.37 | *** join/#asterisk DrJones1 (n=mrjones1@ip68-105-251-187.lu.dl.cox.net) |
| 03:59.46 | shmaltz | You do not lead by hitting people over the head - that's assault, not leadership. |
| 03:59.48 | shmaltz | -Dwight D. Eisenhower |
| 04:00.04 | shmaltz | justinu, but it's still a democracy |
| 04:00.05 | DrJones1 | if i buy a polycomm 301 from voipsupply |
| 04:00.10 | DrJones1 | do i have to buy a seperate ac adapter? |
| 04:00.35 | justinu | a federal republic with a long history of democractic ideals, perhaps |
| 04:00.57 | Corydon76-home | Brixius: so you typed: isql MySQL-asterisk user pass |
| 04:01.00 | Corydon76-home | right? |
| 04:01.09 | shmaltz | DrJones1, no, it's included |
| 04:01.17 | Corydon76-home | CaSe-SeNsItIvE |
| 04:01.20 | shmaltz | Cisco is the only one selling those seperate |
| 04:01.26 | Brixius | Corydon76: yep |
| 04:01.38 | justinu | it's really an oligharchy |
| 04:01.50 | justinu | of corporations and special interest groups |
| 04:01.52 | Corydon76-home | Is there actually a /usr/lib/libmyodbc3.so ? |
| 04:02.05 | Brixius | Corydon76: I'm checking that now |
| 04:02.35 | DrJones1 | can anyone recommend the best headset on voipsupply under 200$? |
| 04:02.53 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:03.16 | shmaltz | DrJones1, with or without speakers? |
| 04:03.21 | shmaltz | you want multi line? |
| 04:03.25 | shmaltz | single line? |
| 04:03.27 | DrJones1 | no, just single line |
| 04:03.32 | DrJones1 | without speakers, i suppose |
| 04:03.37 | DrJones1 | the speakerphone will do just fine |
| 04:03.38 | shmaltz | try the Uniden UIP200 |
| 04:03.48 | *** join/#asterisk Shakh (i=Shakhruz@83.221.171.232) |
| 04:03.57 | DrJones1 | that sip phone? |
| 04:03.58 | Qwell | multiline headset with speakers? |
| 04:03.59 | shmaltz | the only drawback with it is that it doesn't suppport auto answer |
| 04:04.02 | DrJones1 | i was just looking for a headset |
| 04:04.06 | shmaltz | yes it's sip phone |
| 04:04.19 | DrJones1 | i just want something to plug into my rj-12 plug on my cisco 7960 |
| 04:04.19 | shmaltz | oh sorry |
| 04:04.21 | DrJones1 | :) |
| 04:04.25 | DrJones1 | but something nice |
| 04:04.28 | DrJones1 | something from voip |
| 04:04.30 | DrJones1 | supply |
| 04:04.35 | Qwell | why voipsupply? |
| 04:04.38 | shmaltz | I thought you wanted a phone |
| 04:04.48 | DrJones1 | cause im already ordering polycomm 301 from them |
| 04:04.51 | DrJones1 | and they did me right in the past |
| 04:04.51 | justinu | buy a plantronics, they're nice enough |
| 04:04.54 | shmaltz | there are better places then voipsupply for headsets |
| 04:05.08 | DrJones1 | will i save more than 15$ on a 130$ headset? |
| 04:05.50 | shmaltz | anybody know of any softphones with the the USB hardware that will do the transcoding on the hardware, and not on the host CPU? |
| 04:08.53 | *** join/#asterisk _-_ (n=nabudoco@red-corp-201.143.59.44.telnor.net) |
| 04:08.54 | Brixius | Corydon76: yep the library is there. |
| 04:10.25 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:11.28 | sivana | hrm.. do you convert mp3 to ulaw (ul) ? |
| 04:14.17 | shmaltz | <PROTECTED> |
| 04:14.57 | Brixius | Corydon76: per the message it seems like it can't find the odbc.ini file for whatever reason. |
| 04:15.04 | jbroome | shmaltz: received |
| 04:15.07 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 04:18.47 | Qwell | mogorman: Corydon76-home said no |
| 04:19.03 | mogorman | ? |
| 04:19.09 | mogorman | about the bug |
| 04:19.12 | Qwell | yeah |
| 04:19.20 | shmaltz | My grandmother started walking five miles a day when she was sixty. She's ninety-seven now, and we don't know where the hell she is. |
| 04:19.22 | shmaltz | Ellen DeGeneres |
| 04:19.37 | *** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com) |
| 04:20.09 | mogorman | heh |
| 04:20.10 | NDT | blah blah |
| 04:20.21 | NDT | How is everyone on this fine and wonderful evening? |
| 04:20.58 | Nugget | loopy. |
| 04:21.02 | NDT | hehe |
| 04:21.07 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:21.08 | Qwell | shmaltz: She's walked around the globe 17 times |
| 04:21.18 | shmaltz | I don't know what to think: |
| 04:21.19 | shmaltz | http://www.breitbart.com/news/2006/01/09/D8F1I0K03.html |
| 04:21.59 | shmaltz | Qwell, not 17 |
| 04:22.19 | shmaltz | 5*365*37=67525 |
| 04:22.22 | NDT | heh...well thats a different news tio read there |
| 04:22.33 | NDT | err to |
| 04:24.17 | Qwell | shmaltz: yes, now divide that by the radius of each |
| 04:24.35 | Qwell | (which is ~3963 miles) |
| 04:24.39 | Qwell | of Earth* |
| 04:24.47 | shmaltz | Qwell, but earth is 24000 miles |
| 04:25.23 | justinu | circumferences of the equater |
| 04:25.25 | justinu | equator |
| 04:25.36 | Qwell | heh, why was I doing radius? |
| 04:25.36 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
| 04:26.04 | NDT | maybe you were doing the financial thing again...and radius accounting? |
| 04:26.18 | NDT | ;) |
| 04:26.25 | Qwell | :p |
| 04:26.35 | shmaltz | circumferences of the equater = 24 900.9261 miles |
| 04:26.48 | dorphalsig | ok, after a roundtrip ... I still have the D-Channel problem |
| 04:26.49 | Qwell | okay, so almost 3 times :p |
| 04:27.01 | dorphalsig | Jan 9 18:57:56 WARNING[24335] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 04:27.07 | shmaltz | dorphalsig, where round trip around the world? |
| 04:27.16 | dorphalsig | shmaltz .. almost |
| 04:27.16 | shmaltz | dorphalsig, what dchannel problems? |
| 04:27.19 | justinu | dorphalsig: there is no dchannel on a MFCR2 circuit :P |
| 04:27.24 | NDT | dorphalsig: heh...mine just started that too |
| 04:27.32 | shmaltz | that means that the D channel is not up |
| 04:27.40 | dorphalsig | justinu ... yeah it seems I dont have an MFCR2 |
| 04:27.42 | litage | does a softphone/ip phone/ata do jitter buffering, or does asterisk do it? |
| 04:27.46 | shmaltz | dorphalsig, have you verified with your provider that its up? |
| 04:27.49 | justinu | dorphalsig: lol |
| 04:27.56 | dorphalsig | shmaltz -> yes |
| 04:28.02 | dorphalsig | justinu -> dont laugh! |
| 04:28.19 | justinu | dorphalsig: i'm laughing because I wasted my time with you on the MFCR2 stuff :P |
| 04:28.23 | dorphalsig | justinu -> The standard damn protocol here in south america is R2 |
| 04:28.41 | shmaltz | dorphalsig, then configs are wrong, is it up and you get this message from time to time? or is it not up at all? what does pri show span n show? |
| 04:28.44 | dorphalsig | justinu -> And the telco has no idea what is selling =S |
| 04:28.52 | justinu | so what is it? |
| 04:29.08 | dorphalsig | justinu --> it works with PRI but I get the Dchan error |
| 04:29.23 | justinu | after a while? |
| 04:29.30 | dorphalsig | shmaltz -> it works for a while then falls down, I restart zaptel and everything is nice again for a while |
| 04:29.47 | shmaltz | dorphalsig, what motherboard do yo have? |
| 04:29.51 | NDT | mine just goes through and first says no D channel available then says using 24 anyway...then says the D channel is up lol...and then works anyway |
| 04:30.02 | shmaltz | also, I'm assuming its connected to telco and not to PBX |
| 04:30.06 | justinu | i'd suspect a problem with the E1 layer, or possibly the zaptel hardware |
| 04:30.31 | justinu | i guess there's no error counters in digium's software/hardware, so you can't easily look at the E1 performance |
| 04:31.14 | dorphalsig | I dunno |
| 04:31.17 | NDT | What other quadspan cards work well that are cheaper then digium? heh |
| 04:31.42 | NDT | That 405 was $1260 we bought |
| 04:31.44 | justinu | sangoma |
| 04:31.46 | *** join/#asterisk [hC] (n=lisa@209.200.137.24) |
| 04:31.47 | dorphalsig | NDT --> There are some spanish guys that cloned Digium's cards ... Dunno about performance/support but they are half price |
| 04:31.48 | justinu | i dunno about price |
| 04:31.49 | mogorman | sangoma is digium only competitor really |
| 04:31.51 | justinu | but they work |
| 04:31.53 | mogorman | varrion |
| 04:32.12 | dorphalsig | hey mogorman! |
| 04:32.12 | mogorman | sangoma and digium are about same price |
| 04:32.14 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:32.16 | mogorman | hey dorphalsig |
| 04:32.17 | [hC] | oh.. thats it.. mog is fired for saying the s word. |
| 04:32.18 | [hC] | :P |
| 04:32.26 | shmaltz | I knew it |
| 04:32.33 | mogorman | mog is above the law |
| 04:32.36 | dorphalsig | shmaltz -> what? |
| 04:32.37 | mogorman | wtf whats with my nick |
| 04:32.48 | shmaltz | that I'm disconnected and talking to dead air |
| 04:32.59 | Nugget | mog mog bo-bog, banana-fana fo fog. |
| 04:33.09 | mog_home | if i had a nickle... |
| 04:33.22 | shmaltz | anyhow, I didn't get any answers if you answered about the motherboard |
| 04:33.27 | file[laptop] | a Canadian nickle :P |
| 04:33.32 | mog_home | oh lame |
| 04:33.37 | file[laptop] | muahahaha |
| 04:33.45 | mog_home | with the exchange im like out 3 cents |
| 04:33.53 | mog_home | not to mention gas |
| 04:33.58 | justinu | lol |
| 04:34.03 | [hC] | haha |
| 04:34.07 | Nugget | hah |
| 04:34.10 | file[laptop] | isn't it sad? |
| 04:34.15 | NDT | Yeah what is it with Sangoma and Digium anyway? These two companies hate each other or something? |
| 04:34.18 | shmaltz | dorphalsig, what motherboard you using? |
| 04:34.23 | mog_home | bad blood ndt |
| 04:34.25 | justinu | i don't think so |
| 04:34.28 | mog_home | its long story |
| 04:34.32 | justinu | oh, i guess they do :P |
| 04:34.39 | NDT | ahh |
| 04:34.51 | justinu | sangoma is french? |
| 04:34.56 | mog_home | canadian |
| 04:35.00 | justinu | or is it just a french guy that answers the ir tech support line |
| 04:35.14 | mog_home | where are they file |
| 04:35.20 | file[laptop] | Ontario |
| 04:35.33 | justinu | ontario is a big place |
| 04:35.39 | file[laptop] | what, you want the exact place? |
| 04:35.47 | justinu | yes, gps coordinates |
| 04:35.49 | JunK-Y | missisauga, no? |
| 04:35.51 | file[laptop] | picky people |
| 04:35.52 | file[laptop] | Markham |
| 04:35.58 | NDT | We want a satellite picture of the exact building... |
| 04:35.59 | [hC] | mmm.. ham. |
| 04:36.07 | JunK-Y | NDT: google earth! |
| 04:36.09 | Qwell | JunK-Y: You on googletalk? |
| 04:36.12 | NDT | hehe have it |
| 04:36.16 | file[laptop] | I'm hungry for pizza, but it's like 12:36AM... |
| 04:36.31 | NDT | I was looking down at my house in it yesterday |
| 04:36.32 | justinu | if you don't have pizza delivery at 12:30am, i pity you |
| 04:36.35 | JunK-Y | file: then take froot loops! |
| 04:36.36 | justinu | and the place you live |
| 04:36.39 | file[laptop] | oh |
| 04:36.43 | file[laptop] | I have fruit loop bars |
| 04:37.11 | file[laptop] | ooh and a muffin |
| 04:38.06 | shmaltz | Yeah I'm thirty-six, but on the show I'm thirty-two. Nobody wants to watch a thirty-six year old woman, so they decided to make me thirty-two. Much more appealing somehow. |
| 04:38.08 | shmaltz | Ellen DeGeneres |
| 04:38.08 | Qwell | JunK-Y: are my messages not going through? I'm getting yours.. |
| 04:38.29 | mog_home | qwell! |
| 04:38.44 | Qwell | mog_home: howdy |
| 04:38.52 | file[laptop] | Qwellish Qwell! |
| 04:39.35 | shmaltz | A fanatic is one who can't change his mind and won't change the subject. |
| 04:39.37 | shmaltz | Winston Churchill |
| 04:39.43 | JunK-Y | Qwell: nothing here... :( |
| 04:39.48 | [hC] | NDT: http://maps.google.com/maps?f=q&hl=en&q=50+McIntosh+Drive,+Markham+Ontario,&btnG=Search&ll=43.855481,-79.357767&spn=0.01408,0.036521 |
| 04:39.48 | [hC] | There. |
| 04:39.51 | JunK-Y | feel free to msg me. |
| 04:40.06 | *** join/#asterisk dtev001 (n=mikeh@cpe-24-168-18-30.si.res.rr.com) |
| 04:40.16 | [hC] | I would have been able to produce that alot faster if i werent on dialup right now :( |
| 04:40.37 | dtev001 | hey there.. evening all. anyone out there familiar with the odbc module for asterisk... I was wondering if the new 1.2 or 1.2.2 versions support calling mysql 5.0 stored procedures from the MYSQL() function in asterisk |
| 04:40.59 | justinu | right near an airport |
| 04:41.01 | justinu | great, i can fly there |
| 04:43.20 | _Sam-- | i think, but do not know for sure...if you are trying to interact with mysql aside from CDR/Realtime stuff...it might be easiest to use php to interface with mysql and have asterisk call the php from agi or something |
| 04:43.35 | *** join/#asterisk jahani (n=k@adsl-112-43-192-81.adsl.iam.net.ma) |
| 04:44.08 | dtev001 | ok cause i need to update a web app from the ivr, and the php guys only want me to access via the stored procedures.. |
| 04:45.06 | _Sam-- | i dont understand but ok! |
| 04:45.15 | _Sam-- | within the ivr menu you could call php at any step |
| 04:45.21 | _Sam-- | and have it do wahtever you want to sql |
| 04:45.36 | NDT | [hC]: hehe...was afk a sec...good work |
| 04:45.37 | _Sam-- | at least in my mind you could |
| 04:45.41 | dtev001 | yeah... was trying to keep it in extension_custom.conf to keep them from messing w/my stuff :) |
| 04:46.18 | _Sam-- | you could |
| 04:46.23 | _Sam-- | well maybe |
| 04:46.37 | _Sam-- | just include it |
| 04:46.49 | _Sam-- | kapil> _Sam--: with newer kernels you can use "tmpfs" instead of a ramdisk. if you |
| 04:46.49 | _Sam-- | <PROTECTED> |
| 04:46.49 | _Sam-- | <PROTECTED> |
| 04:46.53 | _Sam-- | errrr sorry |
| 04:46.55 | _Sam-- | wrong paste |
| 04:47.14 | *** join/#asterisk Dibbler__ (n=Dibbler@snaddy.plus.com) |
| 04:47.19 | dtev001 | my concern was does the mysql add-on use a driver that will choke on calling a stored proc... cause stored proc's are a mysql 5.x thing |
| 04:48.49 | Qwell | dtev001: Can you write a view that calls the SP? |
| 04:49.31 | dtev001 | i think so.. i have to check with them, i havent played with the bleeding edge mysql 5.x yet |
| 04:49.48 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) |
| 04:50.18 | *** join/#asterisk brookshire[home] (i=matt@69.247.184.46) |
| 04:51.22 | *** join/#asterisk dudes_ (n=dudes@12-215-33-205.client.mchsi.com) |
| 04:51.46 | _Sam-- | how long has asterisk been able to talk to mysql directly with MYSQL()? |
| 04:51.54 | _Sam-- | forever? |
| 04:51.55 | Qwell | _Sam--: a while |
| 04:51.59 | _Sam-- | sorry |
| 04:52.04 | _Sam-- | i always used php to do what i needed |
| 04:52.06 | dtev001 | and it works well/fast |
| 04:52.09 | mog_home | yeah |
| 04:52.11 | _Sam-- | yeah im seeing that now |
| 04:52.13 | Qwell | I like func_odbc, heh |
| 04:52.36 | dtev001 | maybe i will use perl and call it from exten this way they cant fudge with it |
| 04:54.44 | _Sam-- | give me some real world examples of using mysql queries in a dialplan aside from doing comparisons on caller id? |
| 04:56.17 | dtev001 | sure.. looking up a users account balance.. |
| 04:56.31 | _Sam-- | i see |
| 04:56.57 | dtev001 | or recharging a calling card account |
| 04:57.21 | litage | does a softphone/ip phone/ata do jitter buffering, or does asterisk do it? |
| 04:57.24 | dtev001 | i actually did one to read back a users stock portfolio |
| 04:57.26 | _Sam-- | you do inserts into sql from asterisk? |
| 04:57.44 | _Sam-- | like to recharge a calling card? |
| 04:58.25 | _Sam-- | i guess the backend stuff that charges the cardholder and manages the account must write to the sql |
| 04:58.37 | dtev001 | havent figured that out yet... sent the info to a perl script and did it from there |
| 04:58.38 | _Sam-- | the asterisk is just a front end for entering digits |
| 04:58.38 | *** join/#asterisk florz_ (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 04:58.44 | *** part/#asterisk loud (n=ariel@cypher.punk.net) |
| 04:58.51 | dtev001 | yep |
| 04:59.09 | _Sam-- | interesting stuff |
| 05:00.07 | trixter | _Sam-- give me some real world examples of using mysql queries in a dialplan aside from doing comparisons on caller id? |
| 05:00.11 | trixter | do you really want that? |
| 05:00.19 | trixter | gotta inystall mysql from addons but its easily possible |
| 05:00.44 | _Sam-- | i do that now |
| 05:01.04 | dtev001 | trixter -> you can do all sorts of stuff.. screen pop's for caller id's, account info queries, etc. |
| 05:01.05 | _Sam-- | we modify caller id via php if the incoming caller id is in our database of customer info |
| 05:01.34 | dtev001 | we use mysql() to look up do not call list as well |
| 05:01.53 | trixter | in ael2 format MYSQL(Connect connid localhost username password database); MYSQL(Query resultid ${connid} select * from mytable); MYSQL(Fetch fetchid ${resultid} col1 col2); ... |
| 05:01.58 | *** join/#asterisk ecronin (n=root@widget.gizmolabs.org) |
| 05:02.13 | dtev001 | prevent accidental calls to dnc people from sales people extensions. unless they are in our sugarcrm, then it lets em go.. |
| 05:02.20 | trixter | you asked from the dialplan didnt know you were refering to an agi or some external thing ... |
| 05:02.26 | _Sam-- | trixter: i just didnt know any real practical applications of needing to use sql queries in a dialplan..because i never had to..i was looking for examples of what people do with them |
| 05:02.52 | trixter | MYSQL(Query resultid ${connid} SELECT pattern\,route from routes where \'${phonenumber}\' like concat(pattern\,\'%\') and route != \'\' order by length(pattern) desc); |
| 05:03.17 | trixter | there is my simple but practical example, I have about 467,000 routes in a database mapping country codes, prefixes, etc to specific providers and do LCR |
| 05:03.19 | trixter | :) |
| 05:03.25 | ecronin | anyone here familiar with the zaptel driver? |
| 05:03.35 | trixter | ecronin: no what is that? |
| 05:03.43 | ecronin | like with its internals |
| 05:04.01 | dtev001 | i have 3.6 million people in the NJ do not call list alone :) |
| 05:04.15 | dtev001 | trixter & sam... here is an example |
| 05:04.16 | trixter | how many are wireless? |
| 05:04.21 | dtev001 | exten => s,7,MYSQL(Query resultid ${testdb} SELECT\ `phone`\ FROM\ `cc_card`\ WHERE\ `username`=\"${Secret}") |
| 05:04.21 | dtev001 | exten => s,8,Noop(result1 ${resultid}) |
| 05:04.21 | dtev001 | exten => s,9,GotoIf($[empty${resultid} = empty]?30:10) |
| 05:04.21 | dtev001 | exten => s,10,MYSQL(Fetch foundrow ${resultid} phone) |
| 05:04.21 | dtev001 | exten => s,11,Noop(phone ${phone}) |
| 05:04.22 | dtev001 | exten => s,12,GotoIf($["${foundrow}" = "1"]?100:30 |
| 05:04.24 | dtev001 | exten => s,30,Playback(wrong-try-again-smarty) |
| 05:04.26 | dtev001 | exten => s,31,Goto(testdb,s,3) |
| 05:04.28 | dtev001 | exten => s,100,AGI(cepstral.pl|You Are Current Balance is ${credit} dollars) |
| 05:04.31 | JunK-Y | dtev001: |
| 05:04.33 | JunK-Y | ~pb |
| 05:04.34 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 05:04.40 | trixter | wireless is illegal to call regardless of DnC status per 47 CFR 64.1200 and 16 CFR 310 |
| 05:04.58 | _Sam-- | nice i should be copying some of these down :) |
| 05:05.10 | dtev001 | doesnt stop people... i get calls on my cell |
| 05:05.25 | _Sam-- | that gives me some ideas for my dotcom biz... |
| 05:05.42 | _Sam-- | i could provide some more automated tools via tel |
| 05:06.00 | trixter | its illegal to do that with an automatic dialer unless you have prior written consent effectively |
| 05:06.00 | dtev001 | sam: yeah its a WOW feature for customers :) |
| 05:06.18 | ecronin | I'd like to decode a non-standard DTMF tone, and I think from reading the driver that this is done in hardware on fxs... anyone know if this is the case? |
| 05:06.37 | dtev001 | trixter: there are exceptions, political surveys & collection calls are just 2 |
| 05:06.44 | *** join/#asterisk nvrs (i=RUR@65.93.97.70) |
| 05:06.48 | trixter | but regardless, the national DnC is $15,400/year despite the fact that anything the government creates aside from national security things is public domain and they cant sell ... sigh |
| 05:07.00 | trixter | dtev001: not for mobiles just for the DnC |
| 05:07.04 | dtev001 | i know we paid it |
| 05:07.18 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:07.23 | trixter | I got it exempt becuase one of my customers didnt qualiufy to honor it but they wanted to :) |
| 05:07.29 | trixter | but its illegal for me to share it, go figure |
| 05:07.29 | dtev001 | trixter: i understand the fee.. someone has to pay those govt employees to maintain the list LOL |
| 05:07.40 | dtev001 | trixter: yep me too.. |
| 05:07.54 | dtev001 | someone should sue under FIA (freedom of information act) |
| 05:08.20 | trixter | the problem is that its a combo fcc and ftc thing, and the fcc is basicalyl an unconstitutional agency anyway |
| 05:08.51 | trixter | seperation of powers doctrine, yet the fcc has enforcement (executive) court (judicial) and rule making (legislative) departments |
| 05:09.11 | dtev001 | trixter: yeah thats why i havent heard of anyone getting fined and actually paying it yet. |
| 05:09.28 | trixter | then there is the interference with states rights against the 10th amendment and body of the constitution itself ... basically they can only do something if it involves interstate and foreign commerce and a local telephone call doesnt generally |
| 05:10.01 | dtev001 | here is a question... is there a way to stream live audio to your music on hold ? |
| 05:10.08 | trixter | so technically the FCC doesnt have the power to pass any rules over that, nor does congress have any power to pass laws enabling the fcc to do that, but as long as people accept things like this they will continue |
| 05:10.15 | trixter | dtev001: yes |
| 05:10.17 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:10.23 | dtev001 | how ? |
| 05:11.15 | trixter | you can specify a url for mpg123 instead of files, which is generally the easiest |
| 05:11.20 | dtev001 | cool, now i just need to see if bloomberg or cnbc stream to a supported format :) |
| 05:11.34 | trixter | that would be illegal |
| 05:11.40 | trixter | unless you buy a license of course |
| 05:11.49 | dtev001 | we subscribe to it :) |
| 05:11.59 | trixter | subscribe with redistribution rights? |
| 05:12.16 | dtev001 | we have it working on a legacy phone system... it was one of the make or breaks for replacing with asterisk |
| 05:12.22 | trixter | when you use music or any content for MoH you are effectively redistributing |
| 05:12.28 | trixter | why you cant just rip a CD and stick it in there |
| 05:12.38 | dtev001 | cause cnbc is live.. |
| 05:12.42 | mog_home | thats why we give some away |
| 05:12.48 | mog_home | because it can get tricky |
| 05:13.22 | trixter | yeah there are a couple of royalty free sites out there, there are people that create their own content and distro it royalty free, BMG for a small-medium pbx would be like $200/year for its titles iirc |
| 05:13.37 | dtev001 | seems the brokers think that customers like listening to it while they are on hold |
| 05:13.46 | trixter | but anything roytaly free is perfectly fine ... radio stations generally arent because they want you to pay to rebroadcast :/ |
| 05:14.08 | trixter | MIT got an exemption for some of what they were doing becuase htey did have a license and tied it into their existing license ... although there was a stink about em oding that |
| 05:14.22 | dtev001 | trixter: yeah, we have some kind of arrangement already cause its been working for a while and I remember seeing a bill |
| 05:14.52 | *** join/#asterisk Mipsalawishus (n=Mipsalaw@ip70-191-191-239.fv.dl.cox.net) |
| 05:14.53 | trixter | for streaming media here is the really stupid part.. an average radio station pays like $0.02 per song they play to the label. if that same radio station streams their broadcast they are expected to pay $0.05/song in addition!@#$!@$#! that is messed up |
| 05:15.30 | dtev001 | another question... has anyone used videophones with asterisk... any recommendations |
| 05:15.36 | trixter | that may have changed, I know that many radio stations are starting to go back on and stream so aparently some lciensing settlements were reached it wasnt that long ago when most radio stations totally dropped their streams becuase of that |
| 05:16.16 | mog_home | yes there is innomedia |
| 05:16.23 | mog_home | and now grandstream dtev001 |
| 05:16.52 | dtev001 | mog_home: anything special to do on asterisk to make it work ? |
| 05:17.06 | mog_home | videosupport=yes |
| 05:17.16 | mog_home | in sip.conf |
| 05:17.43 | dtev001 | wow that's easy... |
| 05:17.49 | mog_home | yeah |
| 05:17.52 | dtev001 | any idea what ports it uses ? |
| 05:17.55 | mog_home | but we cant transcode video |
| 05:17.57 | mog_home | it uses rtp |
| 05:17.59 | mog_home | for sip |
| 05:18.21 | dtev001 | ok so my std firewall config will work.. |
| 05:18.26 | mog_home | and we support h.263 and soon h.264 |
| 05:18.36 | dtev001 | do both the phones have to be the same type (since we cant transcode) |
| 05:18.44 | mog_home | yes |
| 05:20.17 | dtev001 | where do i configure moh again ? |
| 05:20.58 | dtev001 | trixter: sorry i should say where do i configure moh to stream from a source (want to test) |
| 05:21.00 | mog_home | musiconhold.conf |
| 05:22.08 | dtev001 | just change the mp3:/var/lib/asterisk/mohmp3 |
| 05:22.17 | dtev001 | to mp3: websource ? |
| 05:24.16 | trixter | yeah why not |
| 05:24.47 | trixter | mp3:http://sex.net/streams/moaning or whatever should work cause its passed to mpg123 |
| 05:24.59 | trixter | some people have reported they need to specify an empty directory as well or it doesnt work |
| 05:25.11 | dtev001 | lol |
| 05:25.14 | trixter | well a directory with 1 0 byte file .. I never had to but some people have said they did |
| 05:26.07 | *** join/#asterisk Hmmmm (n=Hmmmm@221.135.51.19) |
| 05:26.30 | dtev001 | i will try it tommorrow |
| 05:28.01 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:31.32 | mog_home | hmm gnite |
| 05:36.01 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 05:36.02 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 05:48.18 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 05:52.26 | sivana | how would I convert this to the new CDR() function? exten => s,1,GotoIf($[$ACCOUNTCODE != ""],s,gotac) |
| 05:54.04 | Corydon76-home | That doesn't work anyway |
| 05:54.19 | sivana | is the CDR(ACCOUNTCODE) still a variable |
| 05:54.26 | sivana | ${ACCOUNTCODE}? |
| 05:54.31 | Corydon76-home | Essentially |
| 05:54.31 | sivana | what do you mean |
| 05:54.38 | Corydon76-home | ${CDR(accountcode)} |
| 05:55.04 | sivana | is it case sensitive? |
| 05:55.12 | Corydon76-home | The CDR is |
| 05:55.32 | sivana | ok, so all the names are lower case |
| 05:55.33 | Corydon76-home | All functions are all uppercase |
| 05:55.46 | sivana | ya |
| 05:55.51 | sivana | and the variable? |
| 05:55.51 | Corydon76-home | Actually, I think the keywords are all insensitive |
| 05:55.56 | sivana | ok |
| 05:56.10 | sivana | exten => s,1,GotoIf($[${CDR(accountcode)} != ""],s,gotac) |
| 05:56.29 | sivana | so this should work? |
| 05:58.05 | Corydon76-home | I'd use the other syntax, but yeah |
| 05:58.23 | Corydon76-home | $[foo${FOO} = foo] |
| 05:58.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:58.43 | trixter | it takes a pretty insensitive coder to disregard the 5th bit like that |
| 05:58.53 | sivana | what other syntax? |
| 05:59.09 | Corydon76-home | Prefixing both with a constant |
| 05:59.13 | Corydon76-home | as in above |
| 05:59.54 | sivana | trying to default the accountcode if it's blank |
| 06:00.35 | sivana | and convert it all to the new formats |
| 06:00.41 | sivana | http://pastebin.ca/36329 |
| 06:01.17 | Qwell | $[${LEN(${FOO})} == 0] |
| 06:01.35 | Qwell | or, 1 =? |
| 06:01.48 | Qwell | -eq, heh |
| 06:01.50 | Corydon76-home | just 1 |
| 06:01.58 | sivana | $[${LEN(${ACCOUNTCODE})} == 0] ? |
| 06:02.09 | Corydon76-home | Just a single = |
| 06:02.15 | Qwell | GotoIf($[${LEN(${FOO})} = 0]) |
| 06:02.23 | sivana | so to check if there's something I can > 0 ? |
| 06:02.29 | Qwell | != 0 |
| 06:02.33 | sivana | ah :) |
| 06:02.35 | Corydon76-home | either/or |
| 06:02.54 | sivana | so ${ACCOUNTCODE} is still valie |
| 06:02.56 | Qwell | can ${LEN()} return -1? |
| 06:02.56 | sivana | valid |
| 06:02.59 | Qwell | sivana: sure |
| 06:03.09 | Corydon76-home | Qwell: no |
| 06:03.14 | sivana | but to set it, then I do CDR(accountcode) = blah |
| 06:03.36 | Corydon76-home | You don't have enough memory for LEN to return negative |
| 06:03.38 | Qwell | use CDR() for both |
| 06:03.51 | Corydon76-home | sivana: yes |
| 06:04.35 | sivana | Jan 10 00:57:55 WARNING[1439]: pbx.c:1098 pbx_retrieve_variable: ${ACCOUNTCODE} is deprecated. Please use ${CDR(accountcode)} instead. |
| 06:04.59 | Qwell | yeah |
| 06:05.24 | Corydon76-home | Uh, is that your production machine? |
| 06:05.25 | sivana | GotoIf($[${LEN(${CDR(accountcode)}})} !=0) |
| 06:05.28 | sivana | ya |
| 06:05.35 | Qwell | god that's ugl |
| 06:05.35 | Qwell | y |
| 06:05.38 | Corydon76-home | Why are you using trunk for production? |
| 06:05.45 | sivana | cuz I like living on the edge :) |
| 06:05.49 | Corydon76-home | You should be using 1.2 |
| 06:05.58 | sivana | ya, probably |
| 06:05.59 | Qwell | 1.2 doesn't have func_odbc. ;) |
| 06:06.05 | Corydon76-home | We're about to break trunk. Hard. |
| 06:06.15 | sivana | ok.. I won't update it again then :) |
| 06:06.22 | Corydon76-home | No, but func_odbc will compile in 1.2 just fine |
| 06:06.29 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 06:06.40 | Qwell | I'll use 1.2 when I actually move to production, maybe |
| 06:06.47 | Qwell | I watch the commits, so I'm good |
| 06:06.51 | sivana | I'm actually provisioning a new box... so I'm trying to bring my confs up to snuff |
| 06:07.11 | Qwell | (and I dev at work, so...) |
| 06:07.16 | Corydon76-home | Yeah, those deprecated warnings are only in trunk, not 1.2 |
| 06:07.21 | sivana | ya |
| 06:07.36 | Corydon76-home | They'll continue to be there in 1.4 |
| 06:07.46 | sivana | so this would correct then? GotoIf($[${LEN(${CDR(accountcode)}})} !=0) |
| 06:07.47 | Qwell | then, whammo |
| 06:07.49 | Corydon76-home | and removed for 1.6 |
| 06:07.54 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 06:08.04 | Qwell | sivana: no |
| 06:08.13 | Qwell | got an extra } |
| 06:08.14 | sivana | missing closing ] |
| 06:08.24 | Qwell | and that |
| 06:08.30 | Qwell | remove the second } |
| 06:08.40 | Qwell | and put the ] after the 0 |
| 06:09.00 | sivana | so this would correct then? GotoIf($[${LEN(${CDR(accountcode)})} !=0]) |
| 06:09.06 | Qwell | should be good |
| 06:09.12 | sivana | that is ugly :) |
| 06:09.21 | Corydon76-home | Except that you don't have a target for your GotoIf |
| 06:09.24 | Qwell | ~striplastdigit |
| 06:09.26 | jbot | rumour has it, striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
| 06:09.29 | Qwell | THAT is ugly |
| 06:09.31 | *** join/#asterisk watchy (i=watchy@adsl-69-152-41-250.dsl.ltrkar.swbell.net) |
| 06:09.35 | watchy | anyone seen " |
| 06:09.43 | sivana | exten => s,1,GotoIf($[${LEN(${CDR(accountcode)})} !=0],s,gotac) |
| 06:09.43 | Qwell | watchy: No, but I've seen ' |
| 06:09.47 | watchy | anyone seen "Protocol Application Invalid" on a 7960? |
| 06:09.58 | DrJones1 | i have |
| 06:09.59 | watchy | the phone was working fine like 2months ago |
| 06:10.01 | Corydon76-home | sivana: change the first comma to a question mark |
| 06:10.03 | DrJones1 | you have to steup the firmware |
| 06:10.04 | DrJones1 | for some reason |
| 06:10.06 | DrJones1 | no idea why |
| 06:10.07 | watchy | i just turned it on tonight to do some shit |
| 06:10.11 | watchy | and it gives that |
| 06:10.19 | watchy | DrJones1. got a website on what to do? |
| 06:10.34 | sivana | Corydon76-home: darn.. all my macros had a comma |
| 06:10.36 | sivana | GotoIf($[${LEN(${CDR(accountcode)})} !=0]?s,gotac) |
| 06:10.42 | DrJones1 | watchy not really |
| 06:10.50 | DrJones1 | just get lots of cisco firmwares |
| 06:10.54 | DrJones1 | its in the cisco sdatabase |
| 06:10.56 | DrJones1 | as a known problem |
| 06:11.07 | watchy | ah |
| 06:11.28 | watchy | whys this thing gotta do this shit. it was working fine |
| 06:12.16 | watchy | i cant even get into the config on the damn phone |
| 06:12.40 | sivana | so I don't use SET() anymore for it? |
| 06:12.49 | sivana | ${CDR(accountcode)}=${ARG2} |
| 06:13.11 | Corydon76-home | Yeah, you use Set |
| 06:13.13 | sivana | ok |
| 06:13.14 | Corydon76-home | Just not SetVar |
| 06:13.15 | Qwell | Set(${CDR(accountcode)}=${ARG2} |
| 06:13.17 | Qwell | ) |
| 06:13.31 | sivana | ya |
| 06:13.33 | sivana | ok |
| 06:16.00 | sivana | http://pastebin.ca/36330 |
| 06:16.24 | sivana | if accountcode is blank, it checks arg 2 to see if there is a value |
| 06:16.56 | Qwell | missing a } on the second LEN |
| 06:17.22 | *** join/#asterisk salmandr (n=salmandr@66-188-101-214.dhcp.mdsn.wi.charter.com) |
| 06:17.34 | sivana | right after the ) |
| 06:17.42 | Qwell | yeah |
| 06:17.50 | watchy | anyone able to point me where to get a 7960g sip image? |
| 06:18.05 | sivana | watchy: I think you need to get it from Cisco? |
| 06:18.23 | watchy | i dunno |
| 06:18.26 | watchy | i just need it |
| 06:18.36 | watchy | can i console into this phone using a cisco cable? |
| 06:18.43 | sivana | ya, but I think you need to buy it from Cisco |
| 06:18.53 | sivana | cisco firmware isn't free |
| 06:18.57 | watchy | well i got sip image like 7.4 |
| 06:19.07 | watchy | but i need teh image to reflash this fone |
| 06:19.09 | watchy | if its a bad flash |
| 06:19.26 | sivana | you should be able to get it from your TAC account |
| 06:19.35 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
| 06:19.37 | watchy | i rarely deal with cisco stuff |
| 06:19.39 | *** part/#asterisk salmandr (n=salmandr@66-188-101-214.dhcp.mdsn.wi.charter.com) |
| 06:19.41 | watchy | so i sure aint got a tac acct |
| 06:20.07 | sivana | not sure what the cost is for a maint contract, but it's not expensive for voip firmware |
| 06:20.23 | sivana | then you can download it as much as you want |
| 06:20.32 | *** join/#asterisk Navman (n=icechat5@62.108.206.82) |
| 06:20.40 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 06:20.43 | watchy | yea |
| 06:20.49 | watchy | i'll just google till i find i t on some ftp |
| 06:22.26 | harry8 | has anybody had any problems loading ztdummy on FC1? |
| 06:22.42 | harry8 | when I load that module, I don't get the voice prompts |
| 06:23.00 | harry8 | but when i remove it, it works fine, I don't have this problem in FC4 |
| 06:23.21 | harry8 | the reason i need ztdummy is for meetme() |
| 06:24.21 | sivana | thanks Qwell / Corydon76-home for the help |
| 06:27.00 | kuku5 | what is the best free windows soft phone ? |
| 06:27.18 | sivana | I like xlite.. not sure if it's freely available anymore |
| 06:29.11 | sivana | if not, let me know, I have it |
| 06:29.21 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 06:29.25 | *** join/#asterisk CpuID (n=nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 06:29.49 | kuku5 | ok |
| 06:30.09 | *** part/#asterisk fduplex (i=irc@CPE0080c6efe7fc-CM0012254495cc.cpe.net.cable.rogers.com) |
| 06:30.14 | watchy | i think i fixed the phone sigmounte_ |
| 06:30.16 | watchy | sivana |
| 06:30.20 | watchy | i did a factory reset |
| 06:30.24 | CpuID | hey ppls, anyone ever come across * picking up calls on zaptel channels...but not actually picking them up? according to the * console its picking up and going through the IVR menu, yet all i hear on the handset calling from the other end is it still ringing |
| 06:30.38 | CpuID | first time its ever come up here, seems quite disturbing |
| 06:31.27 | watchy | hmm ok maybe that didnt fix it |
| 06:31.32 | sivana | :) |
| 06:31.43 | watchy | i need a dhcp server |
| 06:34.34 | harry8 | are there any alternatives to ztdummy for a pure voip setup |
| 06:35.31 | sivana | harry8: for timing? |
| 06:35.51 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:36.01 | Qwell | ztdummy |
| 06:36.09 | Qwell | That *is* the alternative. |
| 06:36.14 | sivana | doesn't 2.6 use USB or something |
| 06:36.22 | Qwell | rtc |
| 06:36.23 | sivana | ya |
| 06:36.40 | sivana | I think for 2.4 you're stuck with ztdummy |
| 06:36.59 | sivana | oh.. there's an astrtc hack somewhere... |
| 06:37.00 | *** part/#asterisk Navman (n=icechat5@62.108.206.82) |
| 06:37.01 | Qwell | still need ztdummy with 2.6 |
| 06:37.05 | sivana | something like that |
| 06:37.35 | harry8 | on fc1 |
| 06:37.50 | harry8 | when i use ztdummy, i get no sound from the voicemail or conference |
| 06:38.07 | harry8 | of fc4 it works but the audio is choppy on SMP - works on non SMP |
| 06:38.13 | Qwell | voicemail doesn't use a timer |
| 06:38.25 | harry8 | i get no streaming |
| 06:38.32 | Qwell | Then it's unrelated.. |
| 06:38.33 | sivana | o ic... on 2.4, ztdummy uses USB |
| 06:39.43 | harry8 | <PROTECTED> |
| 06:39.43 | harry8 | <PROTECTED> |
| 06:39.49 | sivana | http://www.voip-info.org/wiki/view/Asterisk+timer |
| 06:39.49 | harry8 | but then there is no sound |
| 06:39.56 | sivana | zaprtc |
| 06:39.59 | harry8 | when I unload the ztdummy module |
| 06:40.01 | harry8 | it works |
| 06:40.09 | harry8 | rmmod ztdummy |
| 06:41.38 | harry8 | hmm |
| 06:41.45 | harry8 | http://lists.digium.com/pipermail/asterisk-dev/2004-September/006022.html |
| 06:42.01 | harry8 | interesting, i guess asterisk really needs to run on hardware not just virtual machines heheh |
| 06:42.35 | harry8 | maybe that's why I'm getting choppy audio, since vmware the usb interface is emulated |
| 06:42.52 | sivana | hehe |
| 06:43.26 | harry8 | oh well live and learn :p |
| 06:45.16 | lilo | there'll be a small server restart in a few moments; affected users, about 1,300 |
| 06:45.29 | Qwell | NOO!!!! |
| 06:45.39 | Qwell | okay, go ahead |
| 06:45.56 | Qwell | hand |
| 06:45.59 | sivana | heh |
| 06:46.00 | Qwell | not hang...dear lord |
| 06:46.15 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 06:48.02 | watchy | hrm |
| 06:48.14 | watchy | imn going home |
| 06:48.22 | watchy | ill fix this cisco phone tommorow or try |
| 06:48.54 | watchy | this protocol application invalid sucks |
| 06:49.48 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-96.claranet.co.uk) |
| 06:50.46 | Hmmmm | im new to asterisk, can someone suggest a doc to help me get started with setting up an fc4 asterisk server? |
| 06:50.54 | sivana | got I hate infomercials... they're so lame |
| 06:51.02 | sivana | ~docs |
| 06:51.04 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:51.17 | sivana | start here: http://www.voip-info.org/wiki-Asterisk |
| 06:52.06 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.50.97) |
| 06:54.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:54.19 | *** part/#asterisk CANO-1982 (i=alejandr@201.255.50.97) |
| 06:54.20 | *** join/#asterisk EriSan (n=erisan@151.8.109.110) |
| 06:55.03 | *** join/#asterisk CANO-1982 (n=alejandr@201.255.50.97) |
| 06:55.30 | *** join/#asterisk NghtLght (n=ircproxy@202.179.3.171) |
| 06:58.11 | harry8 | is asterisk planning on supporting shared lines? |
| 06:58.37 | drumkilla | ooh, Asterisk has AI? |
| 07:00.53 | sivana | ? |
| 07:02.32 | SkramX | Asterisk is your daddy |
| 07:03.48 | sivana | who's your daddy |
| 07:04.17 | *** join/#asterisk tainted_ (n=identd@ppp-71-134-157-119.dsl.irvnca.pacbell.net) |
| 07:05.00 | *** join/#asterisk [gfe]tHermO (n=thermo@p54AB4C17.dip0.t-ipconnect.de) |
| 07:05.06 | tainted_ | which companies to toll-free origination? |
| 07:05.10 | sivana | we do |
| 07:05.13 | sivana | us/ca |
| 07:05.27 | SkramX | sivana: uh, no ads in #asterisk, I thought |
| 07:05.29 | tainted_ | sivana what price? |
| 07:05.32 | sivana | hehe |
| 07:05.48 | sivana | he asked a question :) |
| 07:05.51 | sivana | and we do :) |
| 07:06.01 | sivana | so does Broadvoice, Nufone, Teliax, etc... |
| 07:06.01 | SkramX | sivana: yalls website isnt even up |
| 07:06.02 | SkramX | whatever |
| 07:06.09 | sivana | what website? :p |
| 07:06.42 | tainted_ | i'm looking for around 2-3/min |
| 07:06.45 | tainted_ | USD |
| 07:06.49 | sivana | try Nufone |
| 07:06.55 | sivana | that's only US origin |
| 07:07.11 | tainted_ | u guys don't do us only? |
| 07:07.24 | sivana | we can, but the rate's the same |
| 07:07.34 | tainted_ | what is that in USD |
| 07:08.21 | sivana | about .427 |
| 07:08.41 | sivana | if you only need US origin.. I'd use Nufone |
| 07:09.39 | sivana | sorry.. 0.0437 |
| 07:09.43 | sivana | ack 427 |
| 07:10.27 | sivana | Teliax has it at Incoming Toll-free .029/min |
| 07:10.42 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
| 07:10.42 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
| 07:10.42 | *** join/#asterisk xbmodder (i=nobody@unaffiliated/xbmodder) |
| 07:10.42 | *** join/#asterisk sac|h0p (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) [NETSPLIT VICTIM] |
| 07:10.42 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM] |
| 07:10.42 | sivana | Nufone has it a 0.02 |
| 07:11.43 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 07:14.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:15.33 | *** join/#asterisk xylo (n=as@c-67-182-205-227.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 07:15.33 | *** join/#asterisk TaSo (i=licucude@68.199.132.160) [NETSPLIT VICTIM] |
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| 07:15.33 | *** join/#asterisk ChkDigit (n=ChkDigit@static65-87-226-124.regina.accesscomm.ca) [NETSPLIT VICTIM] |
| 07:15.33 | *** join/#asterisk xkev (i=kevin@199.104.120.26) [NETSPLIT VICTIM] |
| 07:15.34 | *** join/#asterisk denon (i=denon@sassinak.net) [NETSPLIT VICTIM] |
| 07:15.34 | *** join/#asterisk dcoulson (n=dcoulson@wilbur.geekcolony.net) [NETSPLIT VICTIM] |
| 07:15.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:15.34 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) [NETSPLIT VICTIM] |
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| 07:15.35 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 07:15.36 | *** join/#asterisk malverian[work] (n=pawalls@209.208.68.61) [NETSPLIT VICTIM] |
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| 07:18.31 | *** join/#asterisk linlin (n=linlin@c-67-184-231-233.hsd1.il.comcast.net) |
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| 07:19.47 | *** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org) [NETSPLIT VICTIM] |
| 07:20.01 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
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| 07:22.20 | *** join/#asterisk sancho|laptop (n=user@adsl-68-120-224-179.dsl.irvnca.pacbell.net) |
| 07:24.00 | Pegger | wow |
| 07:24.17 | Pegger | hello |
| 07:24.19 | CpuID | yep server reboots |
| 07:24.58 | Pegger | hey do most of the voip componies like nufone and other just buy a bunch of t1 lines |
| 07:25.24 | Pegger | and then the diffrent customers just share the diffrent lines |
| 07:25.35 | Pegger | or is there somethign that I am missing |
| 07:26.14 | sivana | well... they probably have a limited number of trunk lines.. probably in groups of PRIs or DS3s |
| 07:26.34 | Pegger | sivana, well yes they are limited by the numebr of lines that they have |
| 07:26.40 | sivana | right |
| 07:26.46 | sivana | Nufone I think has DS3s |
| 07:26.55 | Pegger | what about the international stuff |
| 07:27.05 | sivana | when you close a channel, it's available for someone else |
| 07:27.06 | Pegger | how many lines is a ds3 3t1? |
| 07:27.12 | sivana | not sure.. it's a shit load |
| 07:27.31 | Pegger | sivana i get the chanel closes and then somewone else can then use it |
| 07:27.34 | *** join/#asterisk watchy- (n=watchy@h124.184.255.206.cable.cmdn.cablelynx.com) |
| 07:27.35 | sivana | I believe they route their international to their trunk provider |
| 07:27.44 | sivana | Nufone is all TDM |
| 07:27.46 | watchy- | man this damn protocol error is pissing me off |
| 07:27.49 | sivana | they don't have any VoIP routes |
| 07:27.50 | Pegger | sivana, who is their trunk provider? |
| 07:28.15 | sivana | I'm not sure... they may use a couple.. |
| 07:28.36 | sivana | but it would be someone like Level 3 or Global Crossing or their called |
| 07:28.37 | Pegger | sivana, what are some of them? |
| 07:28.50 | Pegger | oah ok |
| 07:28.57 | sivana | Qwest |
| 07:29.07 | sivana | they buy millions of minutes and then resell them |
| 07:29.15 | Pegger | wow that must be a complicated dialplan |
| 07:29.27 | sivana | probably not |
| 07:29.41 | sivana | if you have 3 carriers.. then you only need to do a failover |
| 07:30.14 | Pegger | well you have one context for all of the us and those calls go out on the ds3 |
| 07:30.29 | Pegger | and then anything else goes out on the trunk right |
| 07:30.40 | sivana | ds3 = trunk |
| 07:30.51 | watchy- | anyone know how to get console access to a 7960g |
| 07:31.13 | Pegger | sivana oha ok well you dont think they have any pstn lines? |
| 07:31.21 | sivana | that's it |
| 07:31.28 | sivana | ds3 = trunk = pstn access |
| 07:31.34 | Pegger | watchy can i pm you |
| 07:31.44 | watchy- | yea |
| 07:31.47 | watchy- | go for it homie |
| 07:31.57 | Pegger | sivana, i get some of the terms confused they are so many |
| 07:32.04 | sivana | ds3 = A carrier of 45 Mbps bandwidth. One DS3 channel can carry 28 DS1 channels. |
| 07:32.07 | watchy- | i wonder why this phone would work fine for like months then just not work |
| 07:32.20 | sivana | = 28 T-1 channels |
| 07:32.45 | sivana | = 672 voice circuits |
| 07:33.02 | watchy- | thats alot |
| 07:33.26 | Pegger | wow that is a lot of phone lines |
| 07:33.45 | sivana | ~ds3 |
| 07:33.47 | jbot | somebody said ds3 was 23 T1 channels, or 672 individual B channels. |
| 07:33.58 | Pegger | so how big a server woudl you need to be able to handle that much traffic?? |
| 07:34.09 | sivana | not sure :) |
| 07:34.11 | dudes | is a DS3 28 T1's |
| 07:34.24 | sivana | I think so... I think jbot is wrong |
| 07:34.28 | sivana | or google is :) |
| 07:34.29 | dudes | I think so too |
| 07:35.17 | dudes | 672/24 = 28 |
| 07:35.21 | sivana | ya |
| 07:35.48 | sivana | jbot, ds3 is 28 T1 channels, or 672 individual B channels. |
| 07:35.49 | jbot | ...but ds3 is already something else... |
| 07:35.56 | Pegger | sivana, so how redicilous a machine would you need to be able to handle 672 channels |
| 07:36.06 | sivana | probably more than one :) |
| 07:36.22 | sivana | jbot, ds3 is 28 T1 channels, or 672 individual B channels. |
| 07:36.24 | jbot | ...but ds3 is already something else... |
| 07:36.29 | Pegger | sivana, well is a dS3 supsos to be one cable? |
| 07:36.32 | sivana | ~ds3 |
| 07:36.34 | jbot | from memory, ds3 is 23 T1 channels, or 672 individual B channels. |
| 07:36.46 | sivana | Pegger: it's probably delivered over fiber |
| 07:37.13 | sivana | http://en.wikipedia.org/wiki/DS3 |
| 07:37.17 | Pegger | sivana, then how would you convert it to something that a digum card can understand |
| 07:38.09 | wunderkin | jbot, no a ds3 is 28 T1 channels, or 672 individual B channels. |
| 07:38.10 | jbot | okay, wunderkin |
| 07:38.13 | sivana | I'm not sure.. I don't work with them.. I'm still at DS1 :) |
| 07:38.17 | wunderkin | ~ds3 |
| 07:38.19 | jbot | extra, extra, read all about it, ds3 is 28 T1 channels, or 672 individual B channels. |
| 07:38.24 | sivana | aah.. that's what I forgot |
| 07:38.26 | sivana | thanks |
| 07:39.08 | Pegger | so if it came in on fiber how would you make it so a digum card would be able to understand it ? |
| 07:39.34 | wunderkin | demux it to t1s, you are crazy to put a whole ds3 in one box anyway |
| 07:39.42 | sivana | it might be a BNC cable... and I'm sure you'll need another piece of equipment between your box and the card |
| 07:40.43 | watchy- | i hate cisco |
| 07:40.59 | sivana | hehe |
| 07:41.11 | watchy- | i hope they go bankrupt |
| 07:41.56 | sivana | can't find a sip image? |
| 07:42.02 | sivana | it's only like $60 or something |
| 07:42.08 | sivana | less than $100 I'm sure |
| 07:42.32 | watchy- | i need it right now |
| 07:43.42 | *** join/#asterisk MatsK (n=mk@195.58.126.150) |
| 07:43.49 | sivana | it's a violation to give it out.. and most people respect their agreement with Cisco |
| 07:44.00 | sivana | especially for the money they pay :) |
| 07:44.10 | *** join/#asterisk Fantasy (i=cooL__gi@server.ivinskis.kursenai.lm.lt) |
| 07:44.27 | watchy- | well i got 2 7960s |
| 07:44.35 | watchy- | ones broke cause cisco is a bunch of cock suckers |
| 07:44.38 | sivana | just buy 1 maintenance agreement |
| 07:44.38 | watchy- | i just wanna fix it |
| 07:44.54 | watchy- | i dont mind buying shit |
| 07:44.57 | Qwell | sell it for cheap |
| 07:44.58 | *** join/#asterisk Hakan (i=Miss-tUR@server.ivinskis.kursenai.lm.lt) |
| 07:45.00 | watchy- | i just want firmware now to fix it |
| 07:45.08 | watchy- | i got stuff i gotta get done tommorow |
| 07:45.13 | Qwell | I'll give you $100 for the broken 7960 |
| 07:45.25 | sivana | hehe |
| 07:46.01 | sivana | then you can take that and buy the TAC for the other one :) |
| 07:46.06 | watchy- | haha |
| 07:46.11 | watchy- | im rich i dont need $ |
| 07:46.14 | watchy- | i just need firmware |
| 07:46.20 | Qwell | Then give me the broken 7960, and quit bitching |
| 07:46.24 | sivana | hehe |
| 07:46.27 | watchy- | and i really doubt cisco will send me firmware in 5minutes |
| 07:46.49 | sivana | well, you could call first thing tomorrow and ask for a rush |
| 07:46.59 | Qwell | and in 3 weeks, you'll have a working phone |
| 07:47.02 | sivana | I personally doubt you'll find free firmware tonight |
| 07:47.17 | watchy- | id paypal a bitch $20 for it haha |
| 07:47.26 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 07:47.28 | watchy- | jesus christ i just want my phone to work |
| 07:47.31 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 07:48.04 | watchy- | it broke for no reason |
| 07:48.07 | sivana | He might help, but I doubt it |
| 07:48.23 | watchy- | yea i doubt it to since he dont exist |
| 07:48.34 | Qwell | So what, Christmas is a scam? |
| 07:48.39 | Qwell | great...just great |
| 07:48.42 | sivana | hehe |
| 07:48.43 | Qwell | What's next, Santa is fake? |
| 07:48.51 | Pegger | really |
| 07:48.52 | sivana | and the tooth fairy... sorry |
| 07:49.12 | watchy- | yea christmas sucks |
| 07:51.08 | sivana | alright.. I'm going to bed.. it's 2:50am and I need to be up in 3 hrs.. hehe |
| 07:51.41 | watchy- | im going to kill myselfd because cisco sucks |
| 07:53.57 | jyukes | yo who has cheapest DIDs right now? XO, GlobalCrossing, Level3? |
| 07:59.07 | *** join/#asterisk bkw__ (n=bkw_@70.142.36.51) |
| 08:01.39 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 08:02.30 | Pegger | jyukes, how much do did ushilly cost from the big guys? |
| 08:02.39 | Pegger | like xo and level3 |
| 08:03.48 | watchy- | how the hell do i register to get sip software? |
| 08:04.00 | Qwell | watchy-: get a support contract |
| 08:04.23 | watchy- | where |
| 08:04.33 | Qwell | dunno, try like cdw |
| 08:04.41 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 08:05.16 | watchy- | god |
| 08:05.21 | watchy- | i wish i had a gun id shoot myself |
| 08:13.36 | *** join/#asterisk postel (n=jk@asami.zangetsu.jp) |
| 08:18.59 | Gordo | <PROTECTED> |
| 08:19.40 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 08:20.17 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 08:20.37 | Pegger | Gordo, i woudl not think so |
| 08:22.23 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 08:24.27 | Gordo | Pegger: do you have anything you can chekc it against..? |
| 08:26.48 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:29.40 | *** join/#asterisk bsdfreak (n=alex@daemon.bsdgeek.com) |
| 08:29.42 | bsdfreak | hi |
| 08:30.07 | Pegger | so sip set up presently |
| 08:30.16 | bsdfreak | anyone know of a good program for windows that allows someone to use bluetooth to detect the presence of their cell phone and when it's not detected have asterisk (through the manager or otherwise) forward calls to that cell? |
| 08:30.22 | Pegger | i am trying to get it working at home behind my firewall |
| 08:30.43 | bsdfreak | Asterisk Desktop Manager claimed to have this functionality, but it looks like the bluetooth plugin was removed |
| 08:32.11 | Pegger | that would be neat to have for my linux box, somethign that jsut runs in the background |
| 08:32.19 | bsdfreak | heh |
| 08:32.25 | *** join/#asterisk agh (n=agh@84.241.40.106) |
| 08:32.29 | bsdfreak | well ADM used to do that for linux, too, evidently |
| 08:32.52 | agh | hi gays |
| 08:33.00 | hugo-v6 | gd morning |
| 08:33.05 | agh | i have probelem can any bodey help my |
| 08:33.12 | agh | ? |
| 08:34.12 | *** join/#asterisk startu_net (n=startu@145.116.0.37) |
| 08:34.16 | startu_net | http://purl.oclc.org/NET/NewYear-2005-text |
| 08:34.37 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 08:34.46 | agh | i want config cisco with asterisk any body nows |
| 08:34.52 | agh | ? |
| 08:35.46 | hugo-v6 | is it possible to adjust the sound-gains for sip-phones on asterisk? |
| 08:36.12 | agh | yes you can |
| 08:36.25 | *** part/#asterisk startu_net (n=startu@145.116.0.37) |
| 08:36.39 | agh | you must set codec for you probelem hugo-v6 |
| 08:37.25 | Mimmus | is anyone in a good mood to suggest me some good low-cost IP phone? |
| 08:38.00 | Mimmus | I tried a few of ATCom AT-320 but results are below expectations |
| 08:38.06 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 08:38.43 | agh | any body nows how config cisco fxo with asterisk ? |
| 08:39.43 | agh | heloo |
| 08:39.45 | agh | ? |
| 08:39.57 | agh | any body see my massege ? |
| 08:41.31 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:42.01 | {zombie} | agh: I see it |
| 08:42.04 | hugo-v6 | agh: what do u mean? i set disallow=all \n allow=ulaw |
| 08:43.19 | {zombie} | agh: but you probably need to be a little more clear about exactly what you are trying to do. |
| 08:43.44 | {zombie} | have you seen this page? http://www.voip-info.org/wiki-Asterisk+cisco+FXO |
| 08:44.41 | agh | zombie: i see this link and set up but incomming csll from the psdn not route |
| 08:47.13 | agh | hugo-v6: see this link http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 08:49.51 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 08:52.36 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 08:54.52 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:54.58 | hugo-v6 | agh: i looked at this link already a few times, but looked again and cant find a option concerning my problem. may you bump my nose in it? |
| 08:55.08 | Qwell | My mission: Get 5-6 people in an IRC channel to switch to IAX/SIP from Skype... |
| 08:55.12 | Qwell | Think it'll be difficult? :D |
| 08:55.30 | zoa | yes it won't work |
| 08:56.27 | Ikarus | Hrm, my phone number just disappaered |
| 08:56.58 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
| 08:57.07 | agh | hugo-v6:you must ajust the your phone codec |
| 08:58.16 | hugo-v6 | agh: any suggestions? i use ulaw |
| 08:59.05 | hugo-v6 | and we call via asterisk into pstn (via isdn) |
| 08:59.48 | potsboy | hey all, idea's on what cause this: WARNING[4467]: chan_zap.c:2282 pri_find_dchan: No D-channels available! Using Primary channel 95 as D-channel anyway! ? |
| 09:00.29 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:01.19 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:01.34 | Mimmus | is grandstream budgetone 102 a good phone? |
| 09:01.55 | Qwell | no |
| 09:01.57 | h3x | no |
| 09:02.04 | Qwell | nothing with the word "grandstream" is "good" |
| 09:02.08 | Qwell | not in the slightest |
| 09:02.22 | h3x | the atas probably arent as bad as the phones but they still sucks |
| 09:02.25 | h3x | suck too |
| 09:02.47 | Mimmus | uao! |
| 09:03.07 | Mimmus | suggestion for another low-cost phone? |
| 09:03.41 | agh | any body can set cisco for incomming call to cisco ? |
| 09:03.51 | agh | to asterisk |
| 09:03.59 | agh | any body can set cisco for incomming call to asterisk pbx |
| 09:04.13 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 09:04.20 | agh | any body can set cisco for incomming call to asterisk pbx |
| 09:04.21 | Qwell | agh: try asking again. I don't think everybody heard you the first 8 times |
| 09:05.02 | Mimmus | I cannot propose an IP phone at 100 Euro ($) when an analog phone costs 20! |
| 09:05.14 | Qwell | Mimmus: then don't.. |
| 09:05.28 | Qwell | telephony isn't cheap |
| 09:05.30 | agh | qwell:whats you recommnad ? |
| 09:05.35 | h3x | analog phones cost more coz of the adapter you need to use |
| 09:05.36 | h3x | sort of |
| 09:05.39 | Qwell | agh: I recommend you stop asking every 30 seconds |
| 09:06.01 | hugo-v6 | Mimmus: the sipura phone is well too and doesnt cost that much. also got allnet a new phone (based on sipura) |
| 09:06.32 | Mimmus | sipura... I will look... |
| 09:06.52 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 09:06.52 | h3x | sipura is now cisco/linksys |
| 09:07.02 | h3x | linksys 941 is a good phone |
| 09:07.19 | hugo-v6 | aeh s/allnet/linksys/ damn |
| 09:07.24 | hugo-v6 | i meant linksys |
| 09:07.45 | hugo-v6 | but i still dont get it in .de |
| 09:07.55 | h3x | use snom in .de |
| 09:07.55 | h3x | heh |
| 09:08.02 | agh | qwell: stop man |
| 09:08.04 | hugo-v6 | i _do_ use snom ;) |
| 09:08.08 | hugo-v6 | the 190's |
| 09:08.25 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 09:08.26 | Qwell | Stop what? |
| 09:08.26 | opus_ | yo |
| 09:08.28 | Mimmus | snom is too expensive, 170Euro |
| 09:08.42 | Qwell | Mimmus: telephony ain't cheap...you need to understand that |
| 09:08.44 | nfi|ermes | anyone knows what this can mean ? http://pastebin.com/498970 |
| 09:08.44 | Mimmus | thus sipura/cisco/linklsys is the same thing |
| 09:08.46 | opus_ | how did asterisk 1.2.1 change music on hold .conf file in regards to classes? |
| 09:08.47 | Qwell | Go price an Avaya or Nortel |
| 09:08.54 | hugo-v6 | Mimmus: not in ek ;) and my last customer got them for about 120+taxes ;) |
| 09:09.17 | hugo-v6 | does nortel now support sip? |
| 09:09.24 | Qwell | got me |
| 09:09.39 | Qwell | but their POS PBX will cost you tens of thousands for just a few lines |
| 09:10.13 | Mimmus | Qwell: ok, but an headphone+microphone works well and costs 2-3E! |
| 09:10.27 | Qwell | Then do that |
| 09:10.31 | Mimmus | hugo-v6: what's ek? |
| 09:10.46 | Qwell | but, softphones cannot be passed as a telephony solution |
| 09:10.58 | hugo-v6 | Mimmus: sorry was .de and means purchase price |
| 09:12.15 | hugo-v6 | damn its sad that there is no tx/rx gain setting for sip |
| 09:12.15 | Qwell | hugo-v6: there don't need to be any |
| 09:12.15 | Qwell | just raise/lower your volume |
| 09:12.16 | Mimmus | Qwell: of course. But I need tofind a good compromise cost/quality (as usual) |
| 09:12.31 | Qwell | Mimmus: quality stops at the door when you use crap hardware (grandstream) or softphones |
| 09:12.51 | Mimmus | hugo-v6: I know that in germany there are good prices. I will look at germen online reseller |
| 09:13.02 | hugo-v6 | Qwell: u mean lower mic volume? |
| 09:13.04 | nfi|ermes | anyone knows what this can mean ? http://pastebin.com/498970 |
| 09:13.05 | Qwell | hugo-v6: yes |
| 09:13.24 | hugo-v6 | Mimmus: damn me i still dont sell online :/ |
| 09:13.25 | opus_ | spend more now, or spend even more later :) |
| 09:13.32 | Mimmus | Qwell: in fact, my Atcom sucks but my 1,5Euro headphones rocks! |
| 09:13.40 | hugo-v6 | Qwell: hmmm ill try that. thanks for the hint |
| 09:14.03 | Qwell | hugo-v6: If you're getting echo on SIP, there is a big problem |
| 09:15.41 | hugo-v6 | Qwell: i got an echo on all (snom) phones: the caller on the sip-phone hears himself realy loud. the called doesnt hear it but normal sound. |
| 09:16.02 | hugo-v6 | s/sound/talk/ |
| 09:16.05 | Qwell | wasn't there a firmware issue with the snoms, that caused echo? |
| 09:16.14 | Qwell | might've been another phone |
| 09:16.26 | hugo-v6 | wohoo nice bot :> |
| 09:16.38 | *** join/#asterisk Simon- (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
| 09:16.43 | RoyK | ~seen wasim |
| 09:16.46 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 1d 5h 1m 27s ago, saying: 'hehe'. |
| 09:17.23 | hugo-v6 | Qwell: the next thing ill do now is calling snom (beside that in rel. notes of one firmware they told they got echo cancellation now) |
| 09:19.28 | Qwell | bed time |
| 09:19.37 | hugo-v6 | sleep well Qwell |
| 09:19.41 | RoyK | where? |
| 09:20.01 | hugo-v6 | here its 10h20 am |
| 09:20.27 | Mimmus | hugo-v6: eh eh, in Italy too |
| 09:20.50 | hugo-v6 | Mimmus: since we got same timezone ;) |
| 09:21.26 | hugo-v6 | so... i have to move my poor ass now to $customer. |
| 09:21.29 | hugo-v6 | l8r ppl |
| 09:32.34 | *** join/#asterisk [gfe]tHermO (n=[gfe]tHe@193.174.26.59) |
| 09:32.42 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:33.40 | Mimmus | and what's about polycom? |
| 09:34.37 | *** join/#asterisk lehel (n=ddd@82.79.20.17) |
| 09:36.09 | Ikarus | hrm, getting the message zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff)., anyone got an idea what setting other then signalling I would have to change (tried both bri_cpe and bri_cpe_ptmp) |
| 09:36.21 | oej | Qwell: MOrning! |
| 09:37.07 | kippi | hey |
| 09:38.11 | opus_ | www.asteriskdocs.org died already? |
| 09:38.35 | zoa | no it didnt afaik |
| 09:38.36 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 09:38.53 | oej | zoa: Happy new Year! |
| 09:38.58 | zoa | probably just some problems with the server |
| 09:38.58 | kippi | is there away to push text to my handsets? are there any bits of software that let you do this? I have Grandstream GXP-2000 |
| 09:39.00 | zoa | hey olle |
| 09:39.09 | zoa | chestito nova godina |
| 09:39.19 | oej | zoa: Gott Nytt Ar! |
| 09:40.11 | zoa | that sounds like a disease!!! |
| 09:40.34 | opus_ | musiconhold is suppose to play a mp3 infinitely right? -- so if it hangs up then there probably is a bug, eh? |
| 09:41.02 | zoa | sounds like it yes |
| 09:42.10 | Mimmus | where can I look for problems when my conversations drop sometime for 0.5-1.0 sec? |
| 09:42.24 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:42.50 | opus_ | mimmus zaptel timmer and asterisk doesn't work that well anyway |
| 09:43.04 | opus_ | well:) if you use the asynchronous meetme patch:) |
| 09:43.44 | opus_ | http://lists.digium.com/pipermail/asterisk-users/2003-August/018908.html |
| 09:44.18 | opus_ | I wonder why that patch never made it into trunk? I tried patching that to 1.2.1 but ran into problems |
| 09:45.00 | Mimmus | opus_: any help? zttool gives me always 100% |
| 09:45.21 | opus_ | Mimmus iax, sip, what are you doing when you get the drop? |
| 09:46.15 | Mimmus | opus_: sip at both ends or sip + PRI |
| 09:46.51 | Mimmus | opus_: it is only a brief drop but is not comfortable |
| 09:46.57 | opus_ | mimmus, if it happeneds on pure sip then it is network problems. try checking the tos bit with 'tcpdump -i eth0 udp -vvvv' |
| 09:47.15 | opus_ | mimmus check to make sure all your hardware gear is in full-duplex mode |
| 09:47.30 | opus_ | mimmus make sure its 100% 802.1p compatible.. |
| 09:47.35 | *** join/#asterisk fulgas (n=fulgas@s3.http-tunnel.com) |
| 09:47.36 | Mimmus | opus_: interesting... is tos bit a phone setting? |
| 09:48.17 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 09:48.50 | Mimmus | opus_: I have TOS=0 on my phone |
| 09:48.57 | opus_ | thats your problem |
| 09:49.02 | opus_ | you want tos=0x18 atleast |
| 09:49.03 | fulgas | hey |
| 09:49.29 | Mimmus | opus_: but if I haven't QoS on my network do I need to set it anyway? |
| 09:49.42 | opus_ | nope, there is no point. |
| 09:50.48 | opus_ | unless you know how to send packets back into time |
| 09:50.56 | opus_ | :) |
| 09:51.03 | opus_ | oej is working on a patch for it i bet |
| 09:51.59 | Mimmus | opus_: ok, I understand... but I think that my network is 'good' (100 Mbps full-switched, no cacaded switches, one-port for device, etc) |
| 09:54.25 | opus_ | just look up the model number and 802.1p in google |
| 09:54.53 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 09:55.05 | opus_ | anyone know a really good lame/sox script that can 'fix' my mp3 files? |
| 09:58.29 | saftsack | hi |
| 09:59.39 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 10:02.00 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:04.37 | nfi|ermes | anyone knows what this can mean ? http://pastebin.com/498970 |
| 10:05.42 | opus_ | how do I create a RIFF file? |
| 10:05.45 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 10:07.28 | RoyK | wtf is a riff file? |
| 10:07.37 | MrChimpy | rasterised iff |
| 10:07.46 | MrChimpy | or is it audio |
| 10:07.47 | MrChimpy | i forget |
| 10:07.55 | iDunno | audio, isn't it |
| 10:07.55 | opus_ | i'm going to RIFF somebodies balls out |
| 10:08.04 | RoyK | a jazz riff |
| 10:08.05 | opus_ | yup |
| 10:08.05 | MrChimpy | my memory of them is back in the distant early 90s |
| 10:08.06 | iDunno | though it might be a Rasterised TIFF |
| 10:08.11 | iDunno | ;) |
| 10:08.14 | MrChimpy | yeah, it's audio :) |
| 10:09.28 | Mimmus | opus_: I'm looking at 802.1p... it doesn't seem too difficult to implement.. |
| 10:11.37 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:12.52 | opus_ | musiconhold doesn't like anything i'm giving it |
| 10:13.47 | nfi|ermes | RoyK |
| 10:14.47 | *** join/#asterisk vivekjj (i=1076@203.199.110.93) |
| 10:15.23 | RoyK | nfi|ermes |
| 10:15.46 | vivekjj | hello everybody |
| 10:15.57 | nfi|ermes | can you help me to understand this debug ? http://pastebin.com/498970 |
| 10:16.23 | nfi|ermes | what the hell is |
| 10:16.27 | RoyK | nfi |
| 10:16.32 | nfi|ermes | yes |
| 10:16.33 | RoyK | a generic nfi |
| 10:16.51 | nfi|ermes | :| |
| 10:16.59 | vivekjj | can anyone tell me how do I get sip calls comming from other servers |
| 10:17.02 | RoyK | or perhaps a wtf |
| 10:17.26 | nfi|ermes | nfi ? wtf ? |
| 10:17.31 | RoyK | ~nfi |
| 10:17.33 | jbot | hmm... nfi is No Fucking Idea |
| 10:17.33 | RoyK | ~wtf |
| 10:17.40 | vivekjj | do incomming calls comming from other sip servers land in sip.conf or iax.conf |
| 10:17.48 | RoyK | ~wtf is wtf |
| 10:17.53 | RoyK | ~wtf wtf |
| 10:18.04 | RoyK | ~wtf nfi |
| 10:18.15 | RoyK | ~lart nfi|ermes |
| 10:18.29 | vivekjj | scardinal: do incomming calls comming from other sip servers land in sip.conf or iax.conf |
| 10:18.47 | RoyK | vivekjj: all calls 'land' in extensions.conf |
| 10:19.12 | RoyK | (sip|iax|*).conf just define _where_ in extensions.conf they are handled |
| 10:19.40 | vivekjj | RoyK: how do i tell my asterisk that the calls are comming from there, and how does it come to know where to land up in extensions.conf |
| 10:19.47 | iDunno | incoming calls end up in extensions.conf :) |
| 10:19.58 | iDunno | because you've told it what context it's in, surely :P |
| 10:20.02 | vivekjj | ok |
| 10:20.08 | vivekjj | i will try it, thanks |
| 10:20.17 | iDunno | :) |
| 10:20.23 | vivekjj | thanks a lot |
| 10:20.27 | *** part/#asterisk vivekjj (i=1076@203.199.110.93) |
| 10:26.51 | *** join/#asterisk saftsack (n=oliver@p54A7E253.dip.t-dialin.net) |
| 10:39.31 | saftsack | oh well .... iaxcomm uses spandsp :( |
| 10:41.15 | Mimmus | saftsack: what's iaxcomm? |
| 10:42.21 | zoa | what does it use spandsp for ? |
| 10:42.36 | saftsack | for converting tiffs in dsps |
| 10:42.39 | saftsack | and on the other way |
| 10:43.00 | zoa | but iaxcomm is a softphone |
| 10:43.40 | saftsack | oh i meant iaxmodem |
| 10:43.42 | saftsack | sry |
| 10:43.45 | zoa | ah :) |
| 10:49.06 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 10:52.30 | *** join/#asterisk DannyF (n=dannyf@c-25bbe455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 10:57.41 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 11:00.39 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:02.21 | Mimmus | is there any support for sip-over-tcp in Asterisk 1.2.x? |
| 11:02.28 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:02.39 | saftsack | is sip not always over tcp? |
| 11:02.43 | saftsack | or is it over udp? |
| 11:02.57 | Mimmus | saftsack: 'normally' is over udp |
| 11:03.00 | zoa | sip is over tcp |
| 11:03.03 | RoyK | sip can run over both |
| 11:03.05 | RoyK | but asterisk only supports sip over udp |
| 11:03.05 | zoa | rtp is over udp |
| 11:03.06 | saftsack | ok |
| 11:03.09 | RoyK | zoa: and udp... |
| 11:03.09 | zoa | euh no |
| 11:03.19 | zoa | so both can be both actually |
| 11:03.19 | RoyK | zoa: asterisk only uses sip over udp |
| 11:03.25 | zoa | yes correct |
| 11:03.38 | zoa | im sleeping |
| 11:04.01 | zoa | maybe because there are 5 people asking me things while im writing here |
| 11:04.05 | RoyK | zoa: btw, it looks like the sip jb is pretty stable... |
| 11:04.11 | zoa | royk, told ya so :p |
| 11:04.23 | zoa | we did a few million calls through it |
| 11:04.32 | RoyK | zoa: so now the only problem is..... how stable is asterisk 1.2..... |
| 11:04.38 | oej | Is this in rtp or in channel? |
| 11:04.40 | zoa | less stable then the jitter buffer |
| 11:04.43 | zoa | this is in channel.c |
| 11:04.44 | zoa | im off |
| 11:04.45 | zoa | food |
| 11:04.49 | zoa | will be back in 1 hour |
| 11:04.58 | RoyK | last time we tried upgrading, the server crashed after six hours |
| 11:05.11 | *** join/#asterisk krstone (n=krstone@eden-out.rutgers.edu) |
| 11:06.53 | kippi | is there away that someone could dial 6690 and it would route accross to an external number? exten => 6690,1,Dial(Zap/g1/(01010016690)) ? |
| 11:08.22 | saftsack | what are country codes and howto find my country code? |
| 11:09.25 | *** join/#asterisk EvilRick (n=bob@196-28-86-129.wdsl.co.za) |
| 11:09.50 | EvilRick | anyne managed to gook up a UTstarcom 1000 wifi phoen to asterisk |
| 11:09.58 | EvilRick | mines giving me problems |
| 11:11.21 | Ikarus | With BRIstuff I am getting zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat=0xff) Is it possible that this is simply cable length + bad termination ? |
| 11:16.22 | nfi|ermes | i can t find the fix to this bug : http://bugs.digium.com/view.php?id=131 |
| 11:17.08 | nfi|ermes | markester writes to have submitted this fixes, but i can t see where |
| 11:17.41 | saftsack | _Sam--, hi are you here? |
| 11:20.12 | oej | nfiermes: That was a long time ago, in 2003 |
| 11:21.06 | saftsack | someone here has hylafax? |
| 11:21.08 | nfi|ermes | but the problem is still there, in cvs version, download3ed few time ago |
| 11:21.30 | oej | nfiermes: File a bug report then. A lot has happened since then. |
| 11:22.00 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:30.08 | trixter | saftsack what are country codes and howto find my country code? |
| 11:30.21 | trixter | astbill.com has a listing largely started by me, but no longer really maintained by me |
| 11:30.25 | saftsack | i think i have 49 |
| 11:30.39 | trixter | there is about 500k entries in the database that astbill has |
| 11:30.42 | trixter | and its not even complete |
| 11:30.45 | saftsack | is the country code the number, which is dialed before the number if i want to dial in my own country? |
| 11:31.15 | trixter | country code refers to the number that represents a given country.. 44 is the UK 49 germany 1 is 17 countries (that is a mess) |
| 11:31.26 | trixter | how you set up dialing depends on whether or not you need to dial it |
| 11:31.28 | saftsack | ok so i did it right |
| 11:31.58 | *** join/#asterisk apardo (n=apardo@75.Red-81-44-215.dynamicIP.rima-tde.net) |
| 11:34.52 | saftsack | trixter, are you experienced with hylafax? |
| 11:35.06 | trixter | sometimes |
| 11:35.22 | nfi|ermes | oej |
| 11:35.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:35.53 | saftsack | i have hylafax here with 2 modems. one is connected to my tdm card and the other on my fax. howto show hylafax which modem is for dialing out and in and which one is for my fax device? |
| 11:36.19 | trixter | why do you have faxes in between your fax and tdm card? |
| 11:36.30 | puzzled | morning |
| 11:36.32 | nfi|ermes | can this adapter : http://www.legend.co.uk/hardware/voip_phone_adapter-386.php be considered a sip proxy ? |
| 11:36.35 | trixter | you can lose potentially some functionality between your fax and the remote fax |
| 11:36.50 | saftsack | trixter, what do you mean? |
| 11:37.02 | saftsack | i have a fax here connected with a modem to hylafax |
| 11:37.10 | trixter | by having the modems in between the features that are supported would ultimately be that of the modems |
| 11:37.20 | trixter | which may not be as robust as the actual faxes involved |
| 11:37.23 | saftsack | and the other modem is connected to asterisk which leads it to isdn |
| 11:37.28 | trixter | and most modems appear to have buggy firmware when it comes to faxing |
| 11:37.42 | saftsack | that are fax modems |
| 11:37.51 | trixter | and you just plugged the phone jack of your fax machine into a modem? |
| 11:38.01 | saftsack | yes |
| 11:38.04 | trixter | um |
| 11:38.07 | trixter | ~fxo |
| 11:38.16 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 11:38.16 | trixter | ~fxs |
| 11:38.23 | jbot | somebody said fxs was foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 11:38.23 | saftsack | did that before but i hate spandsp |
| 11:38.25 | saftsack | because i want to archive all faxes |
| 11:38.30 | trixter | read what jbot said |
| 11:38.46 | trixter | you plugged two fxo devices together they can only speak to an fxs device ... that wont work |
| 11:38.53 | trixter | what generates dialtone? the fax nor the modem will |
| 11:39.08 | trixter | what reads dtmf? neither generally will |
| 11:39.10 | saftsack | i know what a fxs is and i connected my fax directly to it first and it worked |
| 11:39.18 | trixter | you need to plug the fax into an fxs device |
| 11:39.24 | saftsack | hmm sure? |
| 11:39.35 | saftsack | i think so too |
| 11:39.37 | trixter | your fax was able to signal to your modem that it was dialing and your modem was able to answer? |
| 11:39.54 | saftsack | dunno didnt try it |
| 11:39.55 | trixter | those are pretty nnifty modems |
| 11:40.01 | trixter | you just said you did |
| 11:40.06 | trixter | saftsack i know what a fxs is and i connected my fax directly to it first and it worked |
| 11:40.13 | saftsack | yes that worked |
| 11:40.15 | trixter | ahh |
| 11:40.22 | trixter | well how will two fxo devices talk to each other? |
| 11:40.32 | saftsack | doesnt work |
| 11:40.32 | trixter | a modem and a fx machine are both fxo devices |
| 11:40.37 | trixter | exactly |
| 11:40.40 | saftsack | i know what you mean and i got it |
| 11:40.45 | trixter | so you cant do it that way no matter what you do with hylafx |
| 11:40.51 | saftsack | but now is my question are there any fxs devices for hylafax? |
| 11:42.27 | saftsack | trixter, ? |
| 11:42.43 | trixter | ahh.. afaik no hylafax is designed to talk to modems |
| 11:43.00 | trixter | you could fake it a little but it would take spandsp |
| 11:43.01 | saftsack | talk to modems? |
| 11:43.16 | saftsack | so hylafax cant send faxes to a fax device? |
| 11:43.16 | trixter | even iaxmodem uses spandsp - and that provides a suitable interface to hylafax |
| 11:43.28 | trixter | how would it directly interface with the fax device? |
| 11:44.23 | saftsack | but spandsp isnt stable |
| 11:44.44 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 11:45.22 | saftsack | or? |
| 11:45.42 | trixter | why isnt it? |
| 11:45.44 | saftsack | arent there any longer tested stable solutions for connecting a fax digitally to the computer? |
| 11:46.15 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 11:46.16 | saftsack | trixter, because many people said, that it doesnt run good on their asterisk |
| 11:46.32 | trixter | how were they trying to run faxes? |
| 11:46.39 | trixter | a lot of people try to run it via voip and blame spandsp |
| 11:46.50 | saftsack | humm ... |
| 11:46.56 | trixter | others dont have enough cpu to properly run it and blame the software |
| 11:47.06 | trixter | many people have used spandsp for a lot of faxes and generally they work |
| 11:47.14 | trixter | its not 100% but its really high |
| 11:47.25 | trixter | most modems arent 100% either infact a lot of modems are worse than spandsp |
| 11:48.08 | Ikarus | wish there was some more info instead of source diving |
| 11:48.10 | Ikarus | ah well |
| 11:48.17 | saftsack | but would you youse spandsp for productive working? |
| 11:49.33 | Ikarus | saftsack: with the reliability of VoIP in not dropping anything, no |
| 11:49.41 | saftsack | i havent no voip |
| 11:49.46 | saftsack | havent voip |
| 11:49.52 | saftsack | just isdn |
| 11:50.04 | Ikarus | saftsack: then use hylafax + ISDN4Linux |
| 11:50.33 | saftsack | yes thought so too, but the only free ntba is used by my asterisk |
| 11:50.41 | saftsack | so it has to go over asterisk |
| 11:51.04 | trixter | spandsp does not reuqire voip |
| 11:51.25 | trixter | it never has required voip, infact documentation talks about how that is not a good thing |
| 11:51.28 | saftsack | yes i know and it works here without hylafax but i need hylafax for sending faxes from computers |
| 11:51.40 | trixter | instead a fxs/fxo line or digital circuit (bri, pri, etc) should beu sed |
| 11:51.54 | trixter | you can use iaxmodem to expose a modem interface to hylafax |
| 11:52.11 | saftsack | yes i same idea here |
| 11:52.11 | trixter | iaxmodem connects to asterisk via an iax2 channel, localhost is prefered, that way you can still use hylafax |
| 11:52.18 | trixter | iaxmodem uses spandsp for its dsp work |
| 11:52.24 | trixter | http://sf.net/projects/iaxmodem |
| 11:52.32 | trixter | it even has a hylafax modem definition ready to go |
| 11:52.40 | saftsack | yes i know iaxmodem and i dismissed it before |
| 11:52.44 | saftsack | but now ill give it a try |
| 11:52.59 | trixter | if you want to use hylafax you need to be able to use a modem |
| 11:53.15 | saftsack | ? |
| 11:53.37 | trixter | if you need ot use a modem you *either* have to do a bunch of switching internally to get the real fax machine to talk to your modem (ie fax->fxs->fxo->modem->hylafax) or use something like spandsp |
| 11:53.41 | saftsack | doesnt iaxmodem simulates a modem for hylafax? |
| 11:53.49 | trixter | yes |
| 11:54.00 | trixter | and for what you want it appears to be the better solution becuase you want hylafax |
| 11:54.06 | *** part/#asterisk krstone (n=krstone@eden-out.rutgers.edu) |
| 11:54.21 | saftsack | is app_txfax.so a good thing? |
| 11:55.29 | saftsack | trixter, so how can i send a fax into the telephone net? i doesnt think, that app_txfax works with a hfc card, or? |
| 11:55.50 | Ikarus | saftsack: it would work |
| 11:56.04 | saftsack | would you recommend it? |
| 11:56.07 | Ikarus | considering that a fax is just a modem transmission (which can be transmitted easily over ISDN) |
| 11:56.46 | Ikarus | saftsack: I would suggest bypassing asterisk |
| 11:57.33 | Ikarus | saftsack: using ISDN4linux you can handle different inbound numbers with different applications and outbound is no issue at all |
| 11:57.44 | saftsack | ok |
| 11:57.59 | saftsack | but there is a problem that is always there |
| 11:58.09 | saftsack | howto connect the hardware fax to hylafax? |
| 11:58.21 | *** join/#asterisk [gfe]tHermO (n=[gfe]tHe@193.174.26.59) |
| 11:58.40 | Ikarus | saftsack: err ? |
| 11:58.43 | sulex | what happened to www.asteriskdocs.org? I get connection refused, do you guys have a mirror URL or another source for docs? cheers |
| 11:58.47 | Ikarus | what do you want to do EXACTLY |
| 11:58.50 | *** join/#asterisk Navman (n=icechat5@62.108.206.82) |
| 11:59.09 | saftsack | send faxes from my hardware fax into the world and the computer here should save the faxes for archiving it |
| 11:59.20 | saftsack | the computers should be able too to do this |
| 11:59.24 | Ikarus | May I say, GAH |
| 11:59.30 | saftsack | GAH? |
| 11:59.31 | Ikarus | That is one hell of a hack |
| 11:59.40 | saftsack | hell of a hack? what? |
| 11:59.40 | Ikarus | As in, not a pretty solution |
| 12:00.20 | saftsack | why? is there a better solution for archiving faxes? |
| 12:00.50 | Ikarus | I would suggest using a scanner and a computer to fax and archive, instead of a hardware fax |
| 12:01.01 | Ikarus | because it reduces the number of modem connections that have to go right |
| 12:01.21 | saftsack | and howto dial then? |
| 12:01.43 | Ikarus | saftsack: write a small app for the computer to run to provide a dial thingy |
| 12:01.46 | Ikarus | pretty basic stuff |
| 12:02.00 | *** part/#asterisk Navman (n=icechat5@62.108.206.82) |
| 12:02.29 | saftsack | yes but i want to use a fax because it runs easier |
| 12:03.08 | zoa | sulex what are you looking for ? |
| 12:03.08 | saftsack | Ikarus, do you know howto configure txfax? |
| 12:03.44 | Ikarus | saftsack: yes, but aslong as you don't know what the issues are with first trying to receive and then resend a fax, it would not be a wise choice to use that |
| 12:04.08 | saftsack | why? this would be the way i want to do it |
| 12:05.20 | Ikarus | saftsack: yes, and you are not a paying customer, so I would like you to make it easy for yourself and figure out for yourself what the problems are with faxing |
| 12:05.39 | Ikarus | (hint, Asterisk is more of a PROBLEM then a solution most likely) |
| 12:05.49 | saftsack | paying customer? |
| 12:06.19 | Ikarus | saftsack: yes, as in, I am helping you for free |
| 12:06.39 | zoa | http://www.asteriskguru.com/tutorials/spandsp.html |
| 12:06.42 | zoa | -> for the faxing |
| 12:06.50 | Ikarus | zoa: spandsp is unreliable |
| 12:07.07 | saftsack | yes thats nice :), but i have to use a hardwarefax :( |
| 12:07.18 | Ikarus | saftsack: which means I am not inclined to make everything as you say, as you would probably need me to maintain it |
| 12:07.50 | saftsack | what did i maintain? |
| 12:07.53 | Ikarus | It is possible to do using asterisk and spandsp, you would receive the fax on an FXO and send it out again on a FXS using spandsp in both cases |
| 12:08.12 | zoa | then use zaptel to send it to a hardware fax |
| 12:08.19 | saftsack | yes i did configure it already this way |
| 12:08.20 | sulex | zoa, I'm new to asterisk, I need to create an IVR that releases PIN codes for accessing websites. I want to know if asterisk is what I'm looking for or not. The point is that I don't like to make dumb questions and I would to get through all the doc first. That url is shown in the doc page of the asterisk.org website. Wondering why it refuses connections |
| 12:08.29 | zoa | dont use spandsp for that |
| 12:08.35 | zoa | just accept it on the fxs and send it to the fxo |
| 12:08.42 | saftsack | but i dont know if it runs stable |
| 12:08.46 | zoa | sulex: yes |
| 12:08.48 | zoa | it can do it |
| 12:09.01 | zoa | are you looking for the ebook ? |
| 12:09.04 | saftsack | and i dont know howto tell txfax that it should youse the misdn channel for send faxes |
| 12:09.29 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 12:09.45 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:10.26 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
| 12:10.48 | sulex | zoa: last question and than I start immediatly to read. If I have a list of PINs, within a text file or another source like a mysqldb, can asterisk dinamically create the IVR prompt on its own or dow I have to handle the text2speech translation by some other software? |
| 12:11.02 | sulex | (apologies for my english) |
| 12:11.04 | zoa | you will need some text2speech translation |
| 12:11.14 | zoa | but that could be embedded in asterisk |
| 12:11.22 | zoa | like sphynx |
| 12:11.34 | zoa | meaning asterisk cannot do it out of the box, but you can make it do it |
| 12:13.06 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:13.07 | sulex | zoa: got it, thank you very much for your help. Hint for the chan: put the asteriskguru.com URL in the topic so nobody will ask the question I made about doc ;9 |
| 12:13.11 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 12:13.14 | sulex | heheh, thankyou again ;) |
| 12:15.07 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 12:15.40 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 12:15.56 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) |
| 12:16.55 | saftsack | exten => s,2,Dial(misdn/1/${NUMMER}) |
| 12:16.55 | saftsack | exten => s,3,txfax(${FAXFILE}|caller) |
| 12:17.04 | saftsack | this doesnt work, but dont know why |
| 12:22.45 | saftsack | how can i hangup and spawn another extension? Hangup and then Goto doesnt work :( |
| 12:22.54 | fulgas | got a T405p and it gives me this error kernel: Unassigning channel 0/16!...anyone knows what's the issue? |
| 12:29.35 | tzanger | saftsack: that's because the dialplan doesn't move off the "dial" until AFTER the hangup |
| 12:29.47 | saftsack | yes but how to resolve that? |
| 12:29.51 | saftsack | sole |
| 12:30.09 | tzanger | you need to use the M option or a callfile and dump the call into a context with txfax |
| 12:30.42 | saftsack | what does the M option do? |
| 12:30.56 | tzanger | you need to think outside the box a little :-) " |
| 12:31.05 | tzanger | er "How do I connect two parts of the dialplan together" |
| 12:31.14 | *** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu) |
| 12:31.39 | saftsack | tzanger, german? |
| 12:32.02 | tzanger | I am of german descent but I don't speak or read it very well |
| 12:32.13 | saftsack | :) |
| 12:32.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:32.46 | saftsack | tzanger, so do howto to this now :) ? |
| 12:33.04 | saftsack | the caller is a fax device |
| 12:33.20 | tzanger | have you read up on the M option yet? |
| 12:33.23 | saftsack | and after rxfax it should hangup the fax and then txfax should send it into the world |
| 12:33.36 | caio1982 | seems that Hakan and Fantasy are bots for porn spam/flood |
| 12:33.41 | saftsack | i dont think, that i need the M option |
| 12:33.54 | tzanger | well you can also use a callfile but I doubt you want to do that |
| 12:34.11 | saftsack | ok so ill read what the m option does do |
| 12:34.23 | tzanger | I didn't say m, I said M |
| 12:35.50 | saftsack | is M the option for soxmix? |
| 12:35.55 | tzanger | ? |
| 12:35.58 | tzanger | did I fuck that up? |
| 12:36.24 | caio1982 | lilo: maybe they can be klined or at least banned? |
| 12:37.06 | tzanger | bah |
| 12:37.09 | tzanger | M won't do what you want |
| 12:37.10 | tzanger | try G |
| 12:37.22 | tzanger | I've never used G though (mind you I've never used M either, which is why I thought it'd work) :-) |
| 12:38.19 | tzanger | M executes a macro, but the docs say it executes it BEFORE the answer. G transfers both to a given part of the dialplan upon connect |
| 12:39.03 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
| 12:42.07 | *** join/#asterisk denny (n=denny@pdpc/supporter/professional/denny) |
| 12:43.57 | *** part/#asterisk denny (n=denny@pdpc/supporter/professional/denny) |
| 12:51.14 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:51.20 | saftsack | tzanger, where to append such options? |
| 12:51.48 | saftsack | because there arent brackets in hangup for giving it options |
| 12:51.51 | DrukenHME | tzanger: doesn't g allow continue after hangup ? |
| 12:52.03 | saftsack | yes and i want exactly that |
| 12:52.11 | saftsack | Druken, do you know how it works? |
| 12:52.36 | DrukenHME | it's a dial option |
| 12:53.37 | saftsack | yes but hangup isnt a dial |
| 12:53.45 | DrukenHME | exactly |
| 12:54.13 | saftsack | so how2 do this now? |
| 12:54.22 | *** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net) |
| 12:54.25 | DrukenHME | don't know what ya want to do... |
| 12:55.01 | *** join/#asterisk cj-rm (n=cjrm@81-178-22-214.dsl.pipex.com) |
| 12:55.11 | saftsack | i want to hangup and then goto another context |
| 12:55.31 | cj-rm | I'm looking for some chimes and jingles to play to callers through asterisk. Are there any freely available? |
| 12:56.13 | *** join/#asterisk agh (n=agh@84.241.40.106) |
| 12:56.15 | cj-rm | preferably quite short, certainly no longer than 5 seconds |
| 12:56.30 | agh | hi |
| 12:56.33 | DrukenHME | saftsack: well, i do belive hangup is just that, hangup and terminate.... |
| 12:56.36 | *** join/#asterisk drray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 12:56.42 | saftsack | Druken, hmm ok |
| 12:56.49 | DrukenHME | now if you make the caller hangup, then you can continue |
| 12:57.13 | agh | any body nows about asterisk cisco incoming psdn seting ? |
| 12:57.15 | saftsack | the caller is a fax and the fax is received from rxfax in my extension |
| 12:57.34 | saftsack | and now i want to send the fax with txfax to the net but i want to hangup the fax first |
| 12:57.59 | DrukenHME | oh.. |
| 12:58.03 | saftsack | ? |
| 12:59.07 | DrukenHME | doesn't spandsp do that all on the fly? |
| 12:59.20 | saftsack | i doesnt think so |
| 12:59.22 | *** join/#asterisk chimpers (n=MrChimpy@smtp-gw.amplefuture.com) |
| 12:59.36 | agh | how to route fxo psdn to asterisk |
| 12:59.46 | agh | ? |
| 13:04.21 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:05.15 | *** join/#asterisk Miss-tURk[off] (n=PLease@62.220.217.65) |
| 13:05.16 | *** join/#asterisk [S][I][N][E][M] (n=AceofBas@62.220.217.65) |
| 13:06.29 | drray | stupid crontab |
| 13:06.45 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 13:07.16 | saftsack | can i dial with asterisk as asterisk as a telephone? for example for just play a voice? |
| 13:08.25 | [gfe]tHermO | yes, you can. |
| 13:08.29 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 13:08.49 | saftsack | yes :) with a call file or? |
| 13:08.58 | [gfe]tHermO | just type 'dial <extension>@<context>' in the CLI |
| 13:09.10 | saftsack | and in the config? |
| 13:09.39 | [gfe]tHermO | u want to hear the file @your speaker or want asterisk to connect somewhere and play the file? |
| 13:09.57 | agh | how to route fxo incomin to asterisk ? |
| 13:10.32 | saftsack | i doesnt want to do anything like this. i want to send a fax but i thought that its easier to explain if i send a sound |
| 13:10.42 | saftsack | in general i just want to create a call |
| 13:10.48 | [gfe]tHermO | ah ok.... |
| 13:10.54 | saftsack | but without the telephones which are connected |
| 13:11.05 | benjk | drop a call file into /var/spool/asterisk/outgoing |
| 13:11.14 | [TK]D-Fender | agh : You haven't described you situation. What MODEL are you using? What kind of "FXO"? Details would help.... |
| 13:11.31 | saftsack | benjk, yes thought so too ;) |
| 13:11.32 | [gfe]tHermO | i guess the call file would be the best choice then... |
| 13:11.33 | agh | cisco 3660 |
| 13:12.37 | saftsack | what channel should i set in the callfile? |
| 13:12.37 | drray | with the two fxs card? |
| 13:12.37 | drray | agh? |
| 13:12.37 | agh | yes |
| 13:12.57 | saftsack | for sending a fax? |
| 13:12.57 | agh | no voice call only |
| 13:14.30 | tzanger | DrukenHME: no that's not what he wants, as far as I can tell. He wants to call a fax machine and transmit a fax. |
| 13:14.34 | tzanger | you can't use 'g' to do that |
| 13:14.50 | tzanger | saftsack: have you read the asterisk handbook? It seems obvious that this is not the case |
| 13:15.41 | saftsack | tzafrir_laptop, im reading something over callfiles |
| 13:15.50 | saftsack | tzanger, but i dont know what you mean |
| 13:15.52 | tzanger | callfiles aren't quite how you want to do this, I don't think |
| 13:15.59 | tzanger | what's wrong with |
| 13:16.00 | tzanger | [dofax] |
| 13:16.06 | tzanger | exten => s,1,TxFax(...) |
| 13:16.10 | tzanger | exten => s,2,Hangup |
| 13:16.11 | tzanger | and then |
| 13:16.22 | kippi | on the grandstream is there away to send text to the phones? |
| 13:16.27 | tzanger | exten => 1234,1,Dial(...,,G[dofax,s,1]) |
| 13:16.30 | tzanger | <PROTECTED> |
| 13:17.00 | tzanger | or rather G[dofax^s^1] |
| 13:17.12 | saftsack | yes ok i didnt know that i can call a context in a dialstrin |
| 13:17.15 | saftsack | thanks :) |
| 13:17.28 | tzanger | saftsack: if you read the help for Dial() you did know this :-) |
| 13:17.40 | saftsack | humm, thats right ;) |
| 13:17.43 | tzanger | <PROTECTED> |
| 13:17.44 | tzanger | <PROTECTED> |
| 13:17.44 | tzanger | <PROTECTED> |
| 13:18.11 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:19.38 | agh | i have cisco 3660 with 4 port fxo and i want route incomming call to asterisk |
| 13:21.27 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 13:21.39 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
| 13:24.32 | drray | agh - via sip? |
| 13:24.40 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 13:24.48 | Mimmus | tzanger: is there a way to trap return code from TxFax? |
| 13:25.17 | tzanger | Mimmus: no. |
| 13:25.30 | tzanger | if it sets a dialplan variable then you can use that |
| 13:26.01 | kippi | has anyone used the 4620SW phones with SIP? |
| 13:26.20 | Mimmus | tzanger: I know an application (AsterFax) that is able to trap all error conditions via Manager API. How is this done? |
| 13:26.30 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
| 13:26.37 | tzanger | Mimmus: you'd have to look at the app, i am not familliar with itat all |
| 13:27.42 | Mimmus | tzanger: ok, thanks. But is there not a way to trap erroro conditions from TxFax (destination BUSY/NOANSWER/etc)? |
| 13:28.16 | tzanger | ? TxFax does not handle BUSY/NOANSWEr/etc.. that's the job of Dial |
| 13:28.49 | Mimmus | tzanger: uh? thus is incorrect to use directly TxFax? |
| 13:29.04 | tzanger | Mimmus: how do you get TxFax to dial? |
| 13:30.12 | Mimmus | tzanger: ah, I use a call file pointing to a context starting with TxFax |
| 13:30.53 | tzanger | Mimmus: yes, but a callfile has two legs. One to do the connecting, and one to dictate what's done with the connected channel. txfax is in the latter. the former has the Dial, and you handle ${DIALSTATUS} in there |
| 13:31.29 | Mimmus | tzanger: can you give me some other suggestion? I'm struggling since many months! |
| 13:31.34 | tzanger | in other words, the context txfax is in does not get executed until the Dial() says the line is up and answered. |
| 13:31.42 | tzanger | Mimmus: what exactly are you trying to do? |
| 13:31.58 | Mimmus | tzanger: sending reliably a fax... |
| 13:32.33 | tzanger | over what |
| 13:32.37 | saftsack | Jan 10 14:07:06 WARNING[9458]: app_dial.c:1143 dial_exec_full: Invalid timeout specified: 'G[faxoutgoing]' |
| 13:32.38 | Mimmus | tzanger: PRI |
| 13:32.38 | saftsack | :( |
| 13:33.04 | tzanger | saftsack: please, take 30 seconds and examine your Dial() statement. The error is obvious |
| 13:33.05 | saftsack | you wrote [] brackets in your example but the readme says () |
| 13:34.11 | saftsack | why should i examine my dialstring? the variable NUMMER exists |
| 13:34.36 | Mimmus | tzanger: I'd like to trap all error conditions (max attempts/NOANSWER/BUSY/it is not a fax/etc) |
| 13:34.59 | tzanger | you will not get "it is not a fax" as I said you cannot get the return code of txfax |
| 13:35.01 | saftsack | oh |
| 13:35.09 | saftsack | <PROTECTED> |
| 13:35.20 | Mimmus | tzanger: ok but NOANSWER/BUSY/FAILED? |
| 13:35.28 | tzanger | Mimmus: you do that in the context that has the Dial() |
| 13:35.28 | saftsack | exten => _9.,6,Dial(misdn/1/${NUMMER},G[faxoutgoing^s^1]) |
| 13:35.34 | tzanger | saftsack: still not right |
| 13:35.43 | saftsack | round brackets? |
| 13:35.48 | tzanger | look at the help for Dial(), paying special attention to its parameters |
| 13:35.58 | Mimmus | tzanger: using a call file? |
| 13:35.59 | tzanger | Dial can take 3 parameters |
| 13:36.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:36.08 | tzanger | make sure you have them |
| 13:36.17 | tzanger | Mimmus: *sigh* |
| 13:36.22 | tzanger | exten => s,1,Dial() |
| 13:36.31 | tzanger | exten => s,2,Goto(s-${DIALSTATUS}) |
| 13:36.43 | tzanger | exten s-BUSY,1,NoOp(Line Was Busy) |
| 13:36.44 | *** join/#asterisk netzwerkgoettin (n=spillerm@p54A56D89.dip.t-dialin.net) |
| 13:36.45 | saftsack | tzanger, do you mean the timeout as the third option? |
| 13:36.46 | Mimmus | tzanger: I'm very sorry :( |
| 13:36.50 | bjohnson | just send him to superdial |
| 13:36.50 | tzanger | exten s-BUSY,2,Hangup |
| 13:36.56 | netzwerkgoettin | hi there |
| 13:37.06 | tzanger | exten s-CONGESTION,1,NoOp(congestion) |
| 13:37.12 | tzanger | exten s-CONGESTION,2,Hangup |
| 13:37.13 | tzanger | etc. |
| 13:37.33 | Mimmus | tzanger: and does call file points to s,1 ? |
| 13:38.18 | saftsack | exten => _9.,6,Dial(misdn/1/${NUMMER},,G[faxoutgoing^s^1]) |
| 13:39.16 | tzanger | saftsack: that looks a lot better |
| 13:39.21 | saftsack | :) |
| 13:39.35 | tzanger | Mimmus: if that's what you need, yes. :-) |
| 13:39.43 | saftsack | maybe next time ill look better at the things if i think that its a tipeerror |
| 13:40.02 | tzanger | saftsack: you just need to slow down and really look at what you're asking Asteriskt o do |
| 13:40.17 | saftsack | yes that would maybe good |
| 13:40.59 | tzanger | I get the same way, which is why I'm trying to get you to stop following in those particular footsteps... They create a bad path :-) |
| 13:41.13 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 13:41.17 | rkioko | hi guys |
| 13:41.26 | }btorch{ | has anyone here setup asterisk with FWD ? |
| 13:41.27 | rkioko | hows sangoma compared to digium cards |
| 13:41.29 | saftsack | but now i have another serious error. the telephone at the callered people rings very short and then all is stop |
| 13:42.08 | netzwerkgoettin | saftsack: but it *does* ring - that's more than i can say ;) |
| 13:42.11 | *** join/#asterisk BriSch (n=BriSch@barthe.GFAI.de) |
| 13:42.23 | saftsack | do you mean timeout? |
| 13:42.38 | tzanger | rkioko: about the same |
| 13:42.52 | [TK]D-Fender | rkioko : Well.... compare the specs. PCI interoperability, H/W echo can specs, IRQ sharing capabilities, platform dependence, etc. What do you think? |
| 13:42.56 | tzanger | saftsack: is something hanging up? |
| 13:43.16 | tzanger | [TK]D-Fender: the sangoma cards are not all that much better at sharing IRQs. |
| 13:43.17 | saftsack | no, thats the problem but i will retest with set verbose 7 now |
| 13:43.37 | tzanger | the nextgen sangoma cards sound like they will have *killer* echo cancellation though |
| 13:43.43 | netzwerkgoettin | no i can't call out at all |
| 13:43.53 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 13:44.00 | rkioko | SS7 |
| 13:44.22 | rkioko | is my main issue |
| 13:44.44 | Mimmus | tzanger: thanks again, I will try... |
| 13:44.48 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 13:44.48 | *** join/#asterisk RaYmAn-B1 (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 13:44.52 | rkioko | want to peer with someone and they require the gateway to support SS7 |
| 13:45.00 | [TK]D-Fender | tzanger : Dunno about that. None of the other listed incompatibilities (E1000, a number of Intel MB chipsets (7205 and others)), and then there is the raw portability (PCI voltage & platform) |
| 13:45.56 | [TK]D-Fender | rkioko : Digium cards do't support SS&, but Sangoma's do. Can anyone confirm on this? |
| 13:46.01 | tzanger | [TK]D-Fender: the TE405/406/410/411 and A104 use the exact same xilinx FPGA. Digium decided to not put voltage translators on the part and made a 3.3v and 5v version. Sangoma chose the wiser option IMO. |
| 13:46.25 | tzanger | [TK]D-Fender: IIRC the PCI incompatibilities on the Digium stuff has been VASTLY improved in the last 6 months |
| 13:46.48 | rkioko | will asterisk support SS7 on sangoma cards fully |
| 13:47.19 | *** join/#asterisk RoyKa (n=roy@ti200720a080-0856.bb.online.no) |
| 13:47.20 | saftsack | tzanger, maybe the fax does answer it but i dont know |
| 13:47.28 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:47.40 | saftsack | no it does a hungup |
| 13:47.45 | saftsack | <PROTECTED> |
| 13:47.46 | }btorch{ | does the extension at the end of a register => needs to be active ? |
| 13:47.54 | saftsack | thats the device where the hardware fax is connected |
| 13:47.58 | }btorch{ | a real sip [extension] ? |
| 13:48.25 | [TK]D-Fender | rkioko : Read this and follow some links : http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup |
| 13:48.42 | [TK]D-Fender | rkioko : Which seems to say Digium cards can do SS7. |
| 13:49.02 | rkioko | yes, ive gone through it |
| 13:49.36 | [TK]D-Fender | And have you tried it? |
| 13:49.47 | rkioko | not yet |
| 13:50.08 | rkioko | still seeking info on the pros and cons of either |
| 13:52.07 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:52.21 | [TK]D-Fender | http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+faq |
| 13:52.40 | netzwerkgoettin | can anyone please help me? am not able to call out via sip :( |
| 13:52.46 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:52.47 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:53.01 | [TK]D-Fender | netzwerkgoettin : Please describe the problem. |
| 13:53.44 | saftsack | <PROTECTED> |
| 13:53.44 | saftsack | <PROTECTED> |
| 13:53.44 | saftsack | <PROTECTED> |
| 13:54.10 | netzwerkgoettin | have a pc with a hfc-s card, NTBA and ISDN phone connected |
| 13:54.29 | netzwerkgoettin | phone rings on incoming calls, can phone, no problems here |
| 13:55.09 | saftsack | tzanger, i think i need a callfile or something like this |
| 13:55.16 | netzwerkgoettin | but asterisk acts my dialed digits as hangup |
| 13:55.40 | Katty | cold. |
| 13:56.05 | Zeeek | wet |
| 13:56.07 | saftsack | tzanger, after rxfaxing the while i dial a number where the fax has to send but the faxdevice hangs up then i think |
| 13:56.38 | netzwerkgoettin | i can configure my mailbox via #9xxx, too; all the keypad numbers with a leading # work |
| 13:57.15 | Katty | [TK]D-Fender: mew? |
| 13:59.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 13:59.43 | shmaltz | gm e1 |
| 13:59.45 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 14:00.11 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:00.16 | Katty | :> |
| 14:00.24 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:01.04 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 14:01.21 | [TK]D-Fender | Katty: mew. |
| 14:01.45 | saftsack | is there any option for the dialstring that asterisk ignores that the device on the dialers site hung up already? |
| 14:02.50 | Katty | [TK]D-Fender: your statement did not parse |
| 14:04.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:05.44 | saftsack | [TK]D-Fender, do you know howto continue in an extension if the dialer hung up? |
| 14:06.24 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 14:08.16 | Katty | saftsack: just keep going with the next thing |
| 14:08.25 | saftsack | ok |
| 14:08.37 | Katty | sometimes i use congestion |
| 14:08.39 | LostFrog | If I have 6 FXO lines I need to connect to *. Should I spend $500-600 for a Adit 600 CB (with the advantage of it being outside the PC and therefore less susceptible to crashing *), or a $837 for a TDM2402B (With the advantage of less problems with CallerID, Disconnect Supervision and Echo)? |
| 14:08.39 | [TK]D-Fender | saftsack : Never tried. |
| 14:09.09 | Katty | LostFrog: whichever you value as more important. |
| 14:09.22 | [TK]D-Fender | LostFrog : Have you considered a lower port density setup? |
| 14:09.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:09.36 | LostFrog | [TK]D-Fender: like two TDM400s? |
| 14:09.44 | [TK]D-Fender | You could use 2 TDM400's, or 2 A102's |
| 14:10.40 | [TK]D-Fender | Err.. A200 |
| 14:11.06 | LostFrog | From past experience, I hate TDM400s. |
| 14:11.21 | netzwerkgoettin | no one an idea? |
| 14:11.45 | Katty | netzwerkgoettin: all four digits, starting with 9 |
| 14:11.49 | Druken | i'm with you LostFrog |
| 14:11.58 | Druken | TDM400's SUCK BALLZ! |
| 14:12.06 | Katty | Druken: chill. |
| 14:12.15 | [TK]D-Fender | LostFrog : Did you factor in the T1 card? |
| 14:12.15 | CANO-1982 | I have a problem loading the ztdummy module with my 2.6 kernel |
| 14:12.20 | Druken | morning Katty |
| 14:12.21 | Zeeek | I have two TDM400s. They work as designed I think |
| 14:12.28 | darkskiez | Im sick and tired of my tdm400 modules going dead until the module is reloaded |
| 14:12.38 | [TK]D-Fender | LostFrog : A200 looks pretty decent and cheaper for your setup : http://store.myphonecall.co.uk/store/shopdisplayproducts.asp?id=83 |
| 14:12.42 | Zeeek | what version of * ? |
| 14:12.55 | CANO-1982 | everything in the compilation process went fine, but I just can?t load the module |
| 14:12.56 | netzwerkgoettin | katty : i mean #9 as prefix for voicemail, followed by mas |
| 14:12.59 | netzwerkgoettin | msn sorry |
| 14:13.09 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:13.15 | Katty | netzwerkgoettin: never tried. |
| 14:13.18 | CANO-1982 | with asterisk 1.2 |
| 14:13.30 | netzwerkgoettin | i put a mp3 in an folder an can hear it by my phone using #100 |
| 14:13.38 | netzwerkgoettin | but i cannot dial out |
| 14:14.33 | Druken | well, who needs to dial out anyways |
| 14:15.17 | netzwerkgoettin | sometimes, i even want to make a call myself ;) |
| 14:15.26 | LostFrog | I guess I need to find a US source for that, [TK]D-Fender. |
| 14:15.48 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 14:15.59 | LostFrog | netzwerkgoettin: 1-900? |
| 14:16.03 | shmaltz | http://www.nobodyhere.com/toren.hier |
| 14:16.18 | netzwerkgoettin | LostFrog: what do you mean? |
| 14:16.23 | Zeeek | netzwerkgoettin why can't you dial out? By what channel? |
| 14:16.29 | LostFrog | netzwerkgoettin: nm |
| 14:16.36 | CANO-1982 | I have a problem loading the ztdummy module with my 2.6 kernel. Everything in the compilation process went fine, but I just can't load the module. I'm using Asterisk 1.2 |
| 14:16.47 | Druken | LostFrog: you need 6 FXO ? |
| 14:16.54 | [TK]D-Fender | LostFrog : I'm still looking for North America myself... Its the only place I've found a price for it at all and it seems to be on par with TDM400. |
| 14:17.00 | LostFrog | Druken: at the moment. |
| 14:17.20 | netzwerkgoettin | debug mode says accepting overlap voice call from '91' to '<unspecified>' on channel 0/2, span 1 |
| 14:17.25 | Druken | LostFrog: possibly more in the future? |
| 14:17.27 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:17.33 | [TK]D-Fender | LostFrog : Again : did you factor in the T1 card for your ADIT setup or do you have an extra port for it already? |
| 14:17.34 | Dandre | hello all, |
| 14:17.58 | Zeeek | netzwerkgoettin what hardware are you trying to to talk to? |
| 14:18.05 | netzwerkgoettin | Zeeek: asterisk acts my dialed digits as a hangup |
| 14:18.30 | Zeeek | and you put your config in pastebin ? Igf not now's the time |
| 14:18.31 | LostFrog | [TK]D-Fender: I already have a T1 |
| 14:18.36 | LostFrog | [TK]D-Fender: I already have a T1 Digium card. |
| 14:18.55 | netzwerkgoettin | Zeeek: try to call a sipgate number over sipgate |
| 14:18.57 | Druken | i'd go with the channelbank option |
| 14:19.09 | Dandre | I have 2 asterisk boxes and I would like to pickup a call to one extension of box 1 by one extension of box2. Is it doable? |
| 14:19.33 | [TK]D-Fender | Dandre : Sure |
| 14:19.46 | LostFrog | Druken: That's what Im thinking, just making sure that I'm not making a strategic mistake. |
| 14:20.00 | *** join/#asterisk shmaltz_ (n=mybox@mail.dmaven.com) |
| 14:20.09 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:20.13 | saftsack | [TK]D-Fender, how can i generate a call with asterisk without a dialfile? |
| 14:20.14 | netzwerkgoettin | pastebin? |
| 14:20.15 | Zeeek | netzwerkgoettin put your extensions (the part with the dial command) in the pastebin |
| 14:20.20 | Zeeek | ~pastebin |
| 14:20.23 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:20.23 | Druken | the channel bank allows you to expand if needed, and from my experince the PRI cards are more stable then the TDM |
| 14:20.35 | Dandre | I have put pickupgroup and callgroup on both sides but that doesn't seem to work |
| 14:20.49 | [TK]D-Fender | saftsack : No experience with faxing or call-files. |
| 14:21.08 | saftsack | [TK]D-Fender, ok, thanks |
| 14:21.21 | netzwerkgoettin | thx a lot, will try... |
| 14:21.25 | Zeeek | netzwerkgoettin figure out how to use pastebin, I'll be back in 30min :) |
| 14:21.34 | netzwerkgoettin | ;) |
| 14:22.58 | *** join/#asterisk in-side (n=lowgitek@es-217-129-27-34.netvisao.pt) |
| 14:23.01 | in-side | Hi there |
| 14:23.11 | Dandre | [TK]D-Fender: I have put pickupgroup and callgroup on both sides but that doesn't seem to work. Is there anything else to do? |
| 14:23.13 | in-side | does anybody here uses ser or asterisk with radius ? |
| 14:23.21 | in-side | when I mean ser I mean openser |
| 14:24.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:24.42 | [TK]D-Fender | Dandre : ActuallyI think I'd misread your question. I'm not sure... |
| 14:25.26 | saftsack | <PROTECTED> |
| 14:25.30 | saftsack | but why :( |
| 14:25.32 | Dandre | :-( |
| 14:25.34 | saftsack | theres no reasons given |
| 14:26.13 | [TK]D-Fender | LostFrog : A102 averages out to about $600 USD for 6 FXO. Mind you I'm sure its cheaper on this side of the Atlantic, but would ahve to find it for sale somewhere. |
| 14:26.25 | in-side | saftsack: turn on debug |
| 14:26.31 | in-side | verbose debug |
| 14:26.40 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:26.48 | [TK]D-Fender | LostFrog : Might be that the ADIT is a better choice for you. |
| 14:26.53 | saftsack | *CLI> set verbose debug |
| 14:26.53 | saftsack | Verbosity is now OFF |
| 14:26.54 | saftsack | Oo |
| 14:27.07 | *** join/#asterisk ozwald (n=spiri@flcemir-244-249.fusionbroadband.net) |
| 14:27.17 | in-side | ya but turn it on |
| 14:27.54 | saftsack | ok |
| 14:28.04 | Katty | i need an austarlian sponser. |
| 14:28.07 | LostFrog | Thanks all. |
| 14:28.18 | Katty | australian, too. |
| 14:28.45 | saftsack | how? |
| 14:29.16 | saftsack | *CLI> set verbose debug |
| 14:29.19 | saftsack | is that right? |
| 14:29.28 | ozwald | I recently setup a asterisk box.. and Ive been having fun tinkering around.. Im having trouble with some things tho.. Im not using a sip phone to dial out.. I want to use my did to dial in.. then I went it to put me on a inner extension so I can dial out.. I have been tinkering but I have had no luck yet... anyone have any ideas? |
| 14:29.32 | iCEBrkr | WHAT A WEEK! |
| 14:29.36 | iCEBrkr | oh. wait. it's only tuesday. :-/ |
| 14:30.25 | in-side | logger.conf |
| 14:30.26 | in-side | full => notice,warning,error,debug,verbose |
| 14:30.32 | in-side | and set debug to full |
| 14:31.00 | in-side | iCEBrkr: :s |
| 14:31.32 | in-side | my stupid openser ser just refuse to send any package for the radius server what hell |
| 14:32.14 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 14:32.19 | *** part/#asterisk [gfe]tHermO (n=[gfe]tHe@193.174.26.59) |
| 14:32.43 | saftsack | no debug output given to me :( |
| 14:32.53 | saftsack | but i think its a real hungup |
| 14:33.21 | ozwald | can anyone give me ideas or possible a direction I should look in? |
| 14:33.23 | saftsack | my asterisk gets a fax with rxfax from my hardwarefax. and then ill send it with txfax. howto generate an asterisk call now? |
| 14:33.46 | in-side | sorry can't help you much I don't use zap channels in asterisk |
| 14:34.09 | in-side | saftsack: are you using it for FoiP? |
| 14:34.16 | ozwald | in-side you know about voip n stuff/ |
| 14:34.16 | in-side | if so... good luck |
| 14:34.29 | in-side | ozwald: what is voip n stuff ? |
| 14:34.31 | saftsack | in-side, no for FoISDN |
| 14:34.32 | shmaltz_ | http://talibandating.cjb.net/ |
| 14:35.12 | ozwald | ermm like connecting it to a pstn gateway.. dialing out from your box.. and dialing into it.. |
| 14:35.26 | in-side | ozwald: try ..why ? |
| 14:35.48 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 14:36.15 | ozwald | erm.. cause Im trying to figure out the extensions.. like making it so I can dial into my box.. switch to a inside extension and then use my box to dial out |
| 14:36.36 | in-side | ozwald: there are a plenty of tutorials to follow man |
| 14:36.44 | in-side | better is to stick with one |
| 14:36.49 | ozwald | kk |
| 14:37.03 | in-side | and you have asterisk book |
| 14:37.12 | in-side | that is a good resource and it is free |
| 14:37.23 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:37.37 | in-side | search by Asterisk The Future of Telephony |
| 14:40.30 | *** join/#asterisk svenna_ (n=svenna@p548D00B6.dip0.t-ipconnect.de) |
| 14:40.38 | saftsack | aaaaahhhh nothing is working here |
| 14:41.03 | saftsack | after rxfax all gets crap :( no dialing works |
| 14:41.16 | saftsack | rxfax ended well and then :/ |
| 14:42.34 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 14:42.57 | *** join/#asterisk tengulre (n=tengulre@219.144.202.165) |
| 14:43.36 | saftsack | <PROTECTED> |
| 14:43.50 | saftsack | is this an errormsg? |
| 14:44.06 | DrDeke | I don't believe so. |
| 14:44.09 | DrDeke | Well, i mean it depends :) |
| 14:44.17 | DrDeke | My calls always do that when I hang them up. |
| 14:44.22 | saftsack | this message comes everytime when i hang up |
| 14:44.26 | DrDeke | :) |
| 14:44.26 | saftsack | yes, thats true |
| 14:44.52 | DrDeke | i guess it does make one wonder what would make it exit non-non-zero :) |
| 14:44.57 | saftsack | :) |
| 14:45.07 | saftsack | are you experienced with macros? |
| 14:45.16 | DrDeke | Nope, have never used them in * |
| 14:45.29 | saftsack | ok, did you ever use callfiles? |
| 14:45.43 | DrDeke | I've played around with them but only in a very simple way |
| 14:46.08 | saftsack | ok, so you dont know howto start them out of the extensions.conf? |
| 14:46.18 | DrDeke | Hmm |
| 14:46.24 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:48.02 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:48.15 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 14:48.15 | DrDeke | The only way I would know to do that would be to make a System() call from the dialplan to make a copy of a callfile and move it into the Asterisk outgoing spool directory. |
| 14:48.32 | saftsack | hmm ok |
| 14:48.47 | DrDeke | I don't know if that's a good way to do it though, and I've never tried it. |
| 14:49.25 | saftsack | i think its a hard way :( |
| 14:50.38 | saftsack | because after sending the fax my faxdevice hangs up, what is right |
| 14:50.57 | saftsack | but then i want to do a call for sending the fax into the net |
| 14:51.02 | saftsack | with isdn |
| 14:51.13 | tzanger | saftsack: still on this? |
| 14:51.26 | saftsack | tzanger, yes |
| 14:51.28 | tzanger | what is the full "plan" ... |
| 14:51.35 | tzanger | perhaps I'm just misunderstanding something |
| 14:51.46 | saftsack | tzanger, ok ill comment it and give it to you |
| 14:51.50 | DrDeke | Ahh, well, I don't know anything about how to send a fax with Asterisk :( |
| 14:51.52 | saftsack | 2 seconds ... |
| 14:52.09 | tzanger | file[desk]: around? |
| 14:52.15 | tzanger | file[laptop]: around? |
| 14:53.12 | Mimmus | DrDeke: eh eh, I'm not alone! tzanger will be furious |
| 14:53.30 | *** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu) |
| 14:53.31 | DrDeke | lol; I must have missed something (?) :) |
| 14:54.01 | Mimmus | DrDeke: step-by-step instructions by tzanger was not enough for me! |
| 14:54.20 | DrDeke | ohhhh about fax? |
| 14:54.20 | DrDeke | yeah |
| 14:54.32 | saftsack | http://pastebin.com/499279 |
| 14:55.40 | *** join/#asterisk jovan (n=giovanni@host211-204.pool8541.interbusiness.it) |
| 14:55.54 | Mimmus | DrDeke: yes, I'm currently still unable to trap all error conditions sending a fax |
| 14:55.58 | jovan | hi |
| 14:56.18 | DrDeke | I have never even bothered with Asterisk+fax; I don't need it and it sounds like it would be irritating :) |
| 14:56.39 | DrDeke | (On the other hand, I don't NEED chan_bluetooth, and it is definitely irritating since I can't compile it, but I keep trying anyway... ;)) |
| 14:57.00 | saftsack | tzanger, do you have an idea? |
| 14:57.26 | }btorch{ | is there a FWD test number that we can try to see if it works ? |
| 14:57.39 | *** join/#asterisk zyke (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 14:57.52 | saftsack | do you mean me? |
| 14:58.07 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 14:58.15 | tzanger | saftsack: good |
| 14:58.30 | tzanger | saftsack: now in [raus] do something a little different |
| 14:58.48 | saftsack | what? :) |
| 14:58.52 | tzanger | exten _9.,1,Goto(faxrelay,${EXTEN},1) |
| 14:58.59 | tzanger | put your fax relaying in a different context |
| 14:59.08 | tzanger | and now move what you have for _9. into there |
| 14:59.14 | tzanger | and in [faxrelay] add this |
| 14:59.44 | tzanger | exten => h,1,Goto(faxout,s,1) |
| 14:59.46 | tzanger | and in your [faxout] |
| 14:59.57 | saftsack | ok thanks :) |
| 15:00.02 | tzanger | exten => s,1,Dial(misdn/1/${NUMMER},,G[faxoutgoing^s^1]) |
| 15:00.10 | tzanger | basically the logic is as follows |
| 15:00.17 | *** join/#asterisk Dovid (i=dovi5988@250.sub-70-192-84.myvzw.com) |
| 15:00.18 | saftsack | so just exten => _9.,1,Goto(faxrelay,${EXTEN},1) in my raus extension for fax? |
| 15:00.29 | tzanger | dialing 9+anything jumps to receive the fax |
| 15:00.43 | tzanger | you need to jump to a different context because you will be using the special 'h' extension to postprocess the fax |
| 15:00.54 | saftsack | ok :) |
| 15:00.55 | tzanger | the h extension is where the dialplan goes on hangup |
| 15:01.00 | saftsack | so now testing |
| 15:01.07 | tzanger | you need to jump to another context because you don't want to run it again when txfax hangs up |
| 15:01.13 | saftsack | tzanger, i searched for h for three days :) |
| 15:01.15 | tzanger | and faxout just sends the fax |
| 15:01.16 | DrDeke | No, the "H" extension is where proprietary PBXes go when they die because they are evil. |
| 15:01.19 | DrDeke | (bow, bow, bow) |
| 15:01.21 | DrDeke | :) |
| 15:01.27 | saftsack | youre a god :) |
| 15:01.31 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 15:01.39 | tzanger | no, I just have more experience with asterisk |
| 15:01.53 | saftsack | yes thats true ;) |
| 15:01.54 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 15:04.16 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:04.49 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 15:06.14 | mut | is the only way to get an extention to show in the manager api to use an actaul extension for it? |
| 15:06.32 | mut | cause when calls are placed via a macro it shows s as the destination |
| 15:06.37 | mut | but if i make some generic |
| 15:06.51 | mut | XXXXXXXXXX,1,dial(blah) |
| 15:06.55 | mut | it will show |
| 15:10.43 | Mimmus | it seems that my * has some problem with (rare) automatic responders: it doesn't detect answer or hangup |
| 15:10.59 | Mimmus | PSTN line is a E1 PRI (Italy). Any idea? |
| 15:11.12 | *** join/#asterisk gugaiz (n=gugaiz@host197.200.61.156.ifxnw.com.ar) |
| 15:11.31 | saftsack | 10538 root 15 0 32152 6964 4008 S 99.9 1.4 1:29.92 asterisk asterisk runs loop ^^ |
| 15:11.40 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 15:13.12 | gugaiz | hi, I need integrate asterisk with raidus server, what are your recomendation? |
| 15:13.21 | *** join/#asterisk aNaSTaCia_geBeri (n=History@85.108.150.190) |
| 15:13.30 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:13.56 | gugaiz | or something that tell me, when call start and when call end |
| 15:14.34 | harryvv | nice, vancouversun.com has had a major crash. no web site. |
| 15:15.34 | *** join/#asterisk jyukes (n=jameshot@138.89.253.56) |
| 15:15.46 | saftsack | tzanger, hmm the h line doesnt work :( |
| 15:15.50 | saftsack | exten => h,1,NoOp("Hallo") |
| 15:15.54 | RoyKa | harryvv: works for me (tm) |
| 15:16.03 | saftsack | i see nothing like this on my console after the hangup |
| 15:16.10 | *** join/#asterisk jahani (n=k@adsl-54-34-192-81.adsl.iam.net.ma) |
| 15:16.22 | jahani | hi |
| 15:16.27 | harryvv | Roy? I even typed in the dns and nothing comes up. |
| 15:16.28 | saftsack | hi |
| 15:16.30 | jahani | possible to start asterisk on 2 ports? |
| 15:16.35 | jahani | 5060 and other |
| 15:16.51 | *** join/#asterisk jyukes_ (n=jameshot@138.89.253.56) |
| 15:17.16 | harryvv | okay, its firefox |
| 15:17.34 | harryvv | having a issue with not bring up that site. |
| 15:18.07 | saftsack | tzanger, i telled crap. there was a hallo :) |
| 15:18.10 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 15:18.11 | saftsack | <PROTECTED> |
| 15:21.30 | saftsack | CC_RELEASE_COMPLETE|CONFIRM [TE] port:1 |
| 15:21.33 | saftsack | what does that mean? |
| 15:21.35 | netzwerkgoettin | can anyone explain me the message "starting zap/2-1 at outgoing,<number>,1 failed so falling back to exten 's' |
| 15:21.54 | saftsack | es trifft sonst nichts zu |
| 15:21.59 | saftsack | ohh english ^^ |
| 15:22.00 | saftsack | sry |
| 15:22.11 | jsharp | You're missing anything that matches <number> in your outgoing context? |
| 15:22.40 | netzwerkgoettin | thats crazy... |
| 15:23.05 | jsharp | That's me. |
| 15:23.26 | netzwerkgoettin | (what will a woman get who cannot phone for several days ;) ) |
| 15:24.12 | netzwerkgoettin | but why should i write the outgoing number into my extensions.conf? |
| 15:24.33 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:24.57 | harryvv | mabey a number change? |
| 15:25.01 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:25.08 | harryvv | netzwerkgoettin a number of reasons. |
| 15:25.16 | }btorch{ | anyone can help me with a fwd registration using asterisk ? |
| 15:25.17 | netzwerkgoettin | *sigh* |
| 15:25.24 | jsharp | What do you have in your outgoing context in extensions.conf |
| 15:25.35 | }btorch{ | I keep getting a failed to authenticate on register to fw.pulver.com |
| 15:25.49 | }btorch{ | timout |
| 15:26.17 | netzwerkgoettin | only three columns |
| 15:26.47 | netzwerkgoettin | the first sets the caller id, the second sets caller id name |
| 15:26.51 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 15:27.27 | jsharp | Show me just one of the lines? |
| 15:27.45 | netzwerkgoettin | 3.: _x./<my_sip_number>,3,Dial(SIP/${EXTEN}) |
| 15:28.02 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 15:29.28 | jsharp | Are you sure you're setting your right CallerID on the call coming into Asterisk? Try it without the /<my_sip_number>. |
| 15:29.42 | nettie | Hi guys, I'm running the latest stable branch and having a problem with MP3Player.. when I try to play an mp3 file I get this error: Jan 10 16:25:30 NOTICE[18113]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0. The system doesnt have any soundcard installed. Anyone know what could be wrong please? Thanx in advance. |
| 15:30.12 | *** join/#asterisk pengyong (n=lala@218.93.102.142) |
| 15:30.15 | harryvv | why in the world are you playing mp3 player on a asterisk only system? |
| 15:30.30 | Katty | harryvv: to annoy you. |
| 15:30.35 | nettie | ehehe |
| 15:30.37 | Katty | harryvv: why in the world do /you/ care? |
| 15:30.38 | nettie | well |
| 15:31.08 | harryvv | katty, because its been stated that asterisk is to only run by its self..thats what makes it most reliable and stable. |
| 15:31.10 | mog_work | do you have hw? |
| 15:31.21 | Katty | harryvv: why don't you stop complaining. |
| 15:31.27 | mog_work | haryvv stop trolling |
| 15:31.31 | Katty | harryvv: help him if you want, otherwise, why don't you keep your stupid comments to yourself. |
| 15:31.34 | jsharp | Uhh. Isn't he running app_mp3player? Like, playing an MP3 to a channel? |
| 15:31.41 | mog_work | yes |
| 15:31.45 | mog_work | it is |
| 15:32.12 | gugaiz | quit |
| 15:32.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:33.19 | nettie | I wanted to play a few audio streams for a long time with different codecs to see how the system performs. I dont have many clients to test the box, so I thought it was a pretty interesting test to "load" it. |
| 15:33.49 | mog_work | do you have hw nettie? |
| 15:33.51 | harryvv | katty, not complaining but you need to listen more about the requirments to run asterisk. |
| 15:33.55 | mog_work | or ztdummy |
| 15:34.00 | nettie | it's a VPS |
| 15:34.02 | Katty | harryvv: do i? |
| 15:34.10 | Katty | harryvv: do you know how many time i hear people like you? |
| 15:34.12 | Katty | harryvv: they don't help. |
| 15:34.14 | jsharp | Uhoh. Rumble in the bronx. |
| 15:34.19 | Katty | harryvv: all they do is say lolzthatSUCKS |
| 15:34.26 | Katty | harryvv: i know plenty, dear. |
| 15:34.31 | Katty | harryvv: it's your attitude that's annoying me. |
| 15:34.40 | nettie | c'mon guys |
| 15:34.46 | nettie | I didnt want to start a war |
| 15:34.54 | Katty | nettie: it's not you that started this. |
| 15:35.10 | nettie | please .. Katty I get this but at least I'm the cause |
| 15:35.22 | Katty | nettie: people have been doing this for ages. people come in asking for help, and instead of getting help, they just get a smack in the face. i'm sick of it. |
| 15:35.33 | harryvv | katty, there is no attitude in this conversation. Its just a known fact to run asterisk as a telephony server you dont run anything else on it. |
| 15:36.09 | Katty | harryvv: is that so eh? |
| 15:36.14 | Katty | harryvv: a /known/ fact? |
| 15:36.15 | nettie | harryvv: well.. MP3Player is a feature of asterisk |
| 15:36.26 | nettie | it's not MY MP3PLAYER software |
| 15:36.33 | Katty | harryvv: i'll just say whatever, and move on. |
| 15:36.39 | [TK]D-Fender | harryvv : I run my internet gateway, samba, X + KDE for playback on my HDTV, and more on mine..... |
| 15:36.40 | Katty | harryvv: think what you like. |
| 15:36.49 | nettie | maybe I wasnt very deeep in the detail |
| 15:36.54 | [TK]D-Fender | OH and X-10 stuff. |
| 15:36.59 | nettie | I'm sorry for that |
| 15:37.33 | nettie | but asterisk has a module to play them.. so I suppose to be in the right place. |
| 15:39.14 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 15:39.29 | cj-rm | I have an asterisk context setup to Answer, wait 2 seconds, and then call playback to play a message. I then copy a call file into the spool directory to dial out on one of our Zap FXO ports and connect to the context-extension. Unfortunately Asterisk doesn't appear to wait for the dialing of the Zap channel to finish before starting playback. Is there anyway to do this, besides having a large wait time? |
| 15:39.58 | nettie | anyway as stated before I really would like to play such mp3 streams from asterisk to see how the varius codec performs on the system which is a VPS with limited resources.. not having gazillions of SIP clients Ithink the approach to continously play audio stream could do the trick. |
| 15:41.11 | nettie | Of course most of time asterisk will only handle signalling but when it wil record messages on the voicemailboxes and do some IVR function the laod will definitely go up. |
| 15:42.32 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
| 15:42.32 | cj-rm | Does anyone have any ideas on how to make asterisk wait until a call is answered before running the rest of the extension? |
| 15:42.58 | wunderkin | cj-rm, sounds like an analog card |
| 15:43.21 | cj-rm | wunderkin: it is an analog card |
| 15:43.35 | cj-rm | wunderkin: any ideas on how to fix it? |
| 15:43.51 | cj-rm | wunderkin: it's a Digium TDM400P |
| 15:45.01 | nettie | mog_work: sorry I didnt see ur messages :) unfortunately no, and ztdummy is not compiled into the kernel and considering the nature of the system I can't add it myself.. the only thing I can do it request it and see if the VPS provide will compile it in the kernel. |
| 15:45.04 | mog_work | you can try callprogress |
| 15:45.07 | mog_work | but it doesnt always work |
| 15:45.16 | wunderkin | i think there is an undocumented (at least in show apps) option c to dial, i think that the called person has to press # or something to acknowledge |
| 15:45.22 | mog_work | if im not mistaken you need a timing source for any mp3 playback |
| 15:45.57 | wunderkin | otherwise you can use the queue and acknowledgement from that, or maybe the findme |
| 15:46.11 | wunderkin | or use a dial macro |
| 15:46.16 | [TK]D-Fender | nettie : have you considered converting your MP3's to a more native format? |
| 15:46.31 | mog_work | yeah that will probably work tk |
| 15:46.40 | mog_work | but for moh you will need timing source anyways |
| 15:46.45 | nettie | thanx for the idea TK |
| 15:47.04 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:47.10 | [TK]D-Fender | nettie : Its a freebie :) Path of least resistance and all.... |
| 15:47.12 | mog_work | yeah for just listening |
| 15:47.20 | cj-rm | wunderkin: I'm not using a call to dial though, I'm using a call file with Channel: Zap/3/9......... (where the ...'s are the relevant number). Can I still pass in options like with Dial? |
| 15:47.30 | nettie | I didnt :p I just rished to figure out if there was some trick to do it :) eheh |
| 15:47.50 | wunderkin | cj-rm, you'll have to use a local chan to dial then |
| 15:49.18 | wunderkin | why isn't dial option c documented in show apps? was it deprecated because of the dial macro? there *is* a dial option c right? i thought ive heard something like that mentioned before on the list at least |
| 15:49.45 | cj-rm | wunderkin: What are the show apps? |
| 15:51.43 | wunderkin | show appliction blah in the cli |
| 15:51.44 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 15:52.27 | cj-rm | wunderkin: ahh with ya :) |
| 15:53.32 | cj-rm | wunderkin: Should I just create a new unbound SIP channel to do this then? |
| 15:54.38 | wunderkin | wha? sip would work correctly but if you have to use an analog card then use a local channel, search for local channel in the wiki if you arent sure about it |
| 15:54.52 | cj-rm | wunderkin: ok, cool :) |
| 15:55.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:55.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:55.54 | cj-rm | wunderkin: Splendid! Thats exactly what I was wanting... |
| 15:57.06 | watchy- | anyone ever seen the error "protocol application invalid" on a 7960g |
| 16:00.46 | netzwerkgoettin | why get *every* number a ss-noservice-error? i don't understand :( |
| 16:01.17 | *** join/#asterisk Weezey (n=ohno@206.186.52.84) |
| 16:01.26 | Weezey | how can I make an AGI timeout? |
| 16:02.07 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 16:02.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 16:03.21 | netzwerkgoettin | IT WORKS |
| 16:03.29 | netzwerkgoettin | i don't understand but it works!!! |
| 16:03.47 | *** join/#asterisk lo_tech (n=lo_tech@209.36.181.24) |
| 16:04.45 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 16:05.35 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 16:07.24 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 16:09.05 | }btorch{ | are the RTP ports udp or tcp ? |
| 16:09.10 | }btorch{ | udp I assume ? |
| 16:09.13 | lo_tech | udp |
| 16:09.23 | }btorch{ | and so is the 5060 ? |
| 16:10.17 | Katty | if you're going to open port 5060 for rtp stuff... |
| 16:10.22 | Katty | you need to open all the rtp ports too |
| 16:10.32 | Katty | only iax can work over a single port |
| 16:10.45 | Katty | and that's udp 4569 |
| 16:11.09 | }btorch{ | I'm openned the ports on my pix but I used tcp instead of udp |
| 16:11.26 | }btorch{ | trying to get asterisk to register to fwd |
| 16:12.52 | anthm | you can pick a certian range in rtp.conf if you want |
| 16:13.05 | anthm | and only map that range |
| 16:13.18 | watchy- | i think my damn 7960s firmware just magically broke |
| 16:14.06 | watchy- | anyone know how to get a 7960g sip image so i can try to fix this phone |
| 16:15.19 | Weezey | didn't you give me the 79XX firmware? |
| 16:15.37 | watchy- | nope |
| 16:15.46 | watchy- | i wish i had it i think my phones flash is bad |
| 16:16.12 | watchy- | i havent powered it on in about 2months and when i did now i get "protocol application invalid" |
| 16:17.49 | *** join/#asterisk javar (n=javar@69.79.51.8) |
| 16:18.34 | *** join/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net) |
| 16:20.23 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 16:20.45 | lo_tech | could be a number of things... sloppy tftp hosting, general flash problems (image authen failed, nonexistent image, tftp errors, etc.) |
| 16:21.01 | watchy- | lo_tech: well it worked 2months ago |
| 16:21.17 | watchy- | soon as i pluged it lastnight it didnt work |
| 16:21.54 | *** part/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net) |
| 16:22.03 | *** join/#asterisk iPBX (n=owned@68-169-204-147.agstme.adelphia.net) |
| 16:23.11 | watchy- | i am guessing the firmware went bad in the flash |
| 16:23.17 | watchy- | but i dont have firmware to relash it |
| 16:23.23 | lo_tech | very rare |
| 16:23.51 | watchy- | its strange i cant even access the config in the phonee |
| 16:25.26 | watchy- | i can see it getting files off my tftp server. as soon as it gets it SIPMAC.cnf file |
| 16:25.37 | watchy- | it goes to protocol application invalid |
| 16:28.29 | watchy- | 2006/01/10 16:28:13 UTC [3664/4080]: Read request for ./SIP000ED7485582.cnf; mode=octet, from 192.168.10.3:50978 |
| 16:28.29 | watchy- | 2006/01/10 16:28:13 UTC [3664/4080]: Read request for ./SIP000ED7485582.cnf completed successfully. 3362 bytes sent to the client 192.168.10.3 |
| 16:30.54 | *** join/#asterisk voipjjs (n=voipjjs@d29-182.rt-bras.wnvl.centurytel.net) |
| 16:32.22 | watchy- | i wonder why shit has to break for no reason |
| 16:33.18 | *** join/#asterisk zaptel (n=just@nat1.inalambrica.net) |
| 16:34.00 | voipjjs | Good morning. I need to set up AAH tech support center, any takers? |
| 16:35.20 | [TK]D-Fender | As in a tech support center USING AAH, or a tech support center SUPPORTING AAH? |
| 16:35.39 | Qwell | [TK]D-Fender: You *KNOW* it's the former. :) |
| 16:35.46 | file[desk] | either way just give up now |
| 16:36.02 | [TK]D-Fender | Qwell : People are crazy... never under/overestimate them. |
| 16:37.43 | voipjjs | Supporting AAH, I have been chartered to set up a AAH support center. The company has a good source of used computers. An end user will be able to purchase a p3/p4, 512 meg, 20 gig HD, TDM4XX for around $670. THis will include 1 hour of tech support |
| 16:38.29 | file[desk] | you WILL go insane |
| 16:38.32 | file[desk] | you realize this? |
| 16:38.46 | Qwell | will go...already are...no big difference |
| 16:39.12 | [TK]D-Fender | Qwell : Sure there is.... its a question of how many people they take down with them :D |
| 16:40.13 | file[desk] | and, dependency hell strikes again |
| 16:40.26 | xachen | AAH *Shudders* |
| 16:40.29 | pif | hi, has anyone used an isdn phone with chan_capi's NT mode yet? |
| 16:42.06 | voipjjs | My client is already setting up the VOIP network, 800 number and database call tracking via the web, etc. The AAh ISO will be installed and working on the systems before being sent to customers. |
| 16:42.46 | xachen | thats just evil |
| 16:42.51 | xachen | they will all be bringing them back |
| 16:42.58 | xachen | andy ou'll need lots of support ttechs |
| 16:43.21 | file[desk] | I won't even help people with AAH if they pay me |
| 16:43.52 | xachen | its justa pain in the ass |
| 16:43.55 | xachen | doesn't even work good |
| 16:44.00 | xachen | I set it up for a client once |
| 16:44.08 | xachen | and decided afterwards just to manually config it |
| 16:44.11 | brockj49464 | Anybody know of a "softmodem" for XP? |
| 16:44.12 | xachen | asterisk rather |
| 16:44.33 | jsharp | Does AAH use the realtime database stuff or just a fancy config file writer? |
| 16:44.43 | voipjjs | Rate of pay will be $50 an hour, in 15 minute blocks, per call |
| 16:44.44 | Qwell | it uses it's own BS database |
| 16:44.56 | xachen | $50? |
| 16:44.59 | xachen | You'd be paying me $75 :) |
| 16:45.03 | xachen | if not + |
| 16:45.20 | file[desk] | voipjjs: all I can say is be prepared for pain |
| 16:45.26 | *** join/#asterisk Astar (n=astar@ANantes-154-1-9-165.w81-53.abo.wanadoo.fr) |
| 16:45.52 | jsharp | I had someone want me to write some external configuration scripts for AAH that didn't interfere with AAH's regular configurations. |
| 16:47.11 | *** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com) |
| 16:47.22 | [TK]D-Fender | file[desk] : But pain = $. |
| 16:47.35 | *** join/#asterisk bmg505 (n=leon@c1-131-12.rndf.isadsl.co.za) |
| 16:47.39 | [TK]D-Fender | the long it takes to fix a problem the more money they make. |
| 16:48.01 | Mimmus | is there anyone who can help me with a difficlut problem on a E1 PRI line? |
| 16:48.12 | [TK]D-Fender | for $50/h I'd learn and support A@H :) |
| 16:48.36 | eKo1 | Mimmus: what problem? |
| 16:49.47 | Mimmus | eKo1: Asterisk doesn't detect answer for some (rare) numbers |
| 16:49.58 | Mimmus | eKo1: especially automatic responders |
| 16:50.19 | Mimmus | eKo1: it rings indefinitely |
| 16:50.25 | *** join/#asterisk Los415 (n=los415@c-24-126-63-65.hsd1.ca.comcast.net) |
| 16:52.51 | *** join/#asterisk Ti-dan (n=eee@207.107.208.137) |
| 16:52.58 | eKo1 | are you using callprogress = yes in zapata.conf? |
| 16:53.09 | Mimmus | eKo1: no, it is a PRI/E1 line |
| 16:53.17 | voipjjs | If anybody is interested in the tech position please send an email to sales@asteriskmall.com |
| 16:54.06 | *** join/#asterisk Speeder (n=psilva@217.129.166.236) |
| 16:54.21 | eKo1 | Mimmus: give me a scenario where it rings indefinitely |
| 16:54.27 | jsharp | Maybe the far end isn't sending a correct answer supervision? |
| 16:55.23 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 16:55.34 | *** join/#asterisk AAAAA (n=sub@207.107.208.137) |
| 16:56.01 | AAAAA | Hello everybody |
| 16:56.05 | AAAAA | Someone here ?? |
| 16:56.21 | Mimmus | eKo1: an italian telecom toll free number 803789 |
| 16:56.42 | Mimmus | jsharp: surely but how can I detect this? |
| 16:57.11 | AAAAA | where can we have dids with unlimited incoming and multiple channels ?? |
| 16:57.29 | *** join/#asterisk manolo (n=manolo@200.124.172.72) |
| 16:57.43 | kippi | Hi |
| 16:57.43 | jsharp | You can watch the PRI debug logs and see if they tell you anything. |
| 16:57.52 | manolo | Hey how do i make my contact directory?? |
| 16:57.55 | kippi | I have just got this error when making asterisk |
| 16:57.56 | kippi | collect2: ld returned 1 exit status |
| 16:57.56 | kippi | make[1]: *** [app_curl.so] Error 1 |
| 16:57.56 | kippi | make[1]: Leaving directory `/usr/src/asterisk/apps' |
| 16:57.56 | kippi | make: *** [subdirs] Error 1 |
| 16:57.57 | *** join/#asterisk r0d3nt_m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 16:58.10 | kippi | anyideas what is going on? |
| 16:58.30 | jsharp | Need more lines above that to find out where the linker is esploding. |
| 16:58.43 | Mimmus | jsharp: of course but or I'm not able to see anything useful or I'm not in position to do it |
| 16:58.50 | jsharp | Oh. |
| 16:58.58 | AAAAA | are you doing your make install in /usr/src/asterisk/apps or /usr/src/asterisk/ directory ?? |
| 16:59.08 | manolo | Please i need this urgent! how do i make my directory in Asterisk??? |
| 16:59.19 | kippi | usr/src/asterisk/ directory |
| 16:59.27 | jsharp | You may have to talk with your PRI provider then, see if they can do any debugging for you. |
| 16:59.52 | manolo | kippi, in CLI? |
| 17:00.03 | kippi | yeah |
| 17:00.13 | Mimmus | jsharp: aargh, with italian telco this is almost impossible |
| 17:00.17 | *** part/#asterisk Ti-dan (n=eee@207.107.208.137) |
| 17:00.42 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 17:00.52 | jsharp | kippi: Are there lines above the "collect2: ld returned 1 exit status"? Those are the lines that we'd need to see. |
| 17:01.58 | *** join/#asterisk Simon- (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
| 17:02.39 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.197) |
| 17:02.56 | kippi | ok |
| 17:02.56 | AgiNamu | G'day |
| 17:03.00 | manolo | kippi, there is no such directory in CLI.. why is that?? |
| 17:03.02 | kippi | trying it again |
| 17:03.30 | AgiNamu | Any switching experts around? |
| 17:03.45 | AgiNamu | I'm willing to pay $$$ for a few answers. |
| 17:03.45 | AAAAA | "Dropping voice to exceptionally long queue" what can be the trouble here ?? |
| 17:03.47 | Nugget | I switched to macintosh in 2002! ask me anything! ;) |
| 17:03.51 | AgiNamu | lol |
| 17:03.52 | kippi | jsharp: there is just loads and loads of compiler info, can't see collect2 |
| 17:03.58 | AgiNamu | Specifically, Q.931 and 5ESS :) |
| 17:04.26 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 17:06.58 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj0t.dialup.mindspring.com) |
| 17:07.14 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj0t.dialup.mindspring.com) |
| 17:07.19 | watchy | anyone got 7960g firmware i can get now so i dont have to wait like 2 weeks to get a $8 cisco service agreement? |
| 17:08.15 | eKo1 | AgiNamu: are these ss7 questions? |
| 17:08.50 | *** part/#asterisk voipjjs (n=voipjjs@d29-182.rt-bras.wnvl.centurytel.net) |
| 17:09.23 | AAAAA | "Dropping voice to exceptionally long queue" |
| 17:09.44 | [TK]D-Fender | AAAAA : Sounds like massive packet-loss |
| 17:10.21 | saftsack | tzanger, hi are you here? |
| 17:10.24 | watchy | i cant believe ciscos gonna make me wait days for $8 |
| 17:10.57 | AAAAA | well, how can this happen in a lan ?? |
| 17:11.26 | AAAAA | [TK]D-Fender : well, how can this happen in a lan ?? |
| 17:11.56 | [TK]D-Fender | AAAAA : What codecs, how many simultaneous calls, and what server specs? |
| 17:11.56 | AgiNamu | eKo1, no its about why the 5ESS do in-band indications. |
| 17:12.42 | Mimmus | does anyone knows Sangoma card configuration? |
| 17:12.51 | kippi | can anyone help me out? |
| 17:12.57 | [TK]D-Fender | Mimmus : What do you need to know? |
| 17:13.16 | Mimmus | suggested (!) values for some parameters of wanpipe.conf file |
| 17:13.20 | AAAAA | [TK]D-Fender : This is asterisk 1.2.1 . We are using ulaw and something like 10/15 simulanuous calls... this is the same environement and was working with asterisk 1.0.7 |
| 17:13.32 | [TK]D-Fender | Mimmus : like? |
| 17:14.28 | Mimmus | framing, clock mode |
| 17:14.29 | Mimmus | ok, framing=HDB3 in Italy |
| 17:14.29 | Mimmus | Framing? CRC or not? |
| 17:14.31 | Mimmus | Clock? Normal or Master? |
| 17:14.31 | [TK]D-Fender | Mimmus : Sorry, don't know EU standards :/ |
| 17:14.34 | AAAAA | this looks like something new with asterisk 1.2.1 |
| 17:14.46 | [TK]D-Fender | Mimmus : Typicall clocking is "normal" (from telco). |
| 17:14.46 | jsharp | Clock should probably be normal. CRC would depend on your provider. |
| 17:15.21 | watchy | i just bought my cisco smartnet agreement for my firmware for $12 from cdw |
| 17:15.32 | watchy | can someone now send me the firmware so i don't have to wait 2 weeks |
| 17:15.38 | Mimmus | jsharp: OK, CRC seems to work with my telco, it doesn't work with a legacy Alactel PBX |
| 17:16.27 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 17:17.05 | pfn | yes, e.g. context.xml for tomcat |
| 17:17.12 | pfn | or ibm-web-bnd.xmi for websphere |
| 17:17.14 | *** join/#asterisk AlexCTI (n=alex@64.221.229.212.ptr.us.xo.net) |
| 17:17.15 | pfn | or whatever for weblogic |
| 17:17.18 | pfn | or resin, etc. |
| 17:17.20 | pfn | oops |
| 17:17.25 | pfn | damn mis-tab |
| 17:17.38 | eKo1 | hehee |
| 17:20.12 | *** join/#asterisk roulduke_ (i=5c72jrfr@p508D1778.dip0.t-ipconnect.de) |
| 17:20.27 | *** join/#asterisk rastacouette (n=astar@ANantes-154-1-73-220.w86-199.abo.wanadoo.fr) |
| 17:20.35 | Mimmus | are rxgain/txgain in zapata.conf % values or absolute values? |
| 17:20.44 | *** join/#asterisk AAAAA (n=sub@207.107.208.137) |
| 17:20.45 | denon | decibels |
| 17:20.50 | AAAAA | ought my cisco smartnet agreement for my firmware for $12 from cdw |
| 17:20.50 | AAAAA | <watchy> can someone now send me the firmware so i don't have to wait 2 weeks |
| 17:20.51 | AAAAA | <Mimmus> jsharp: OK, |
| 17:20.52 | AAAAA | ought my cisco smartnet agreement for my firmware for $12 from cdw |
| 17:20.52 | AAAAA | <watchy> can someone now send me the firmware so i don't have to wait 2 weeks |
| 17:20.52 | AAAAA | <Mimmus> jsharp: OK, |
| 17:20.58 | AAAAA | sorry guys |
| 17:21.19 | Mimmus | denon: some hint for these? |
| 17:21.25 | denon | Mimmus: 0.0 |
| 17:21.30 | denon | and fix your line problems |
| 17:21.43 | Mimmus | denon: Sangome support suggested me to 'increase' rxgain. How can I do? |
| 17:21.52 | AAAAA | "Dropping voice to exceptionally long queue" on IAX2 seems to be new in asterisk 1.2.1 |
| 17:22.01 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 17:22.16 | Mimmus | denon: are they meaningful for PRI channels? |
| 17:22.19 | denon | if "Sangome" has a reason for you to increase rxgain, then they should tell you what it is |
| 17:22.37 | denon | most installs do not need it, and if they do, its usually due to some other problem that should be fixed instead |
| 17:22.44 | Mimmus | denon: Sangome (eh eh) suggested only this |
| 17:23.16 | *** join/#asterisk brif8 (n=The_Bear@lazyjtrainingcenter.com) |
| 17:23.52 | brif8 | hi all, is chanspy support in the CVS-Head 05/20/05 03:33:04 version and if not how can it be enabled? |
| 17:24.00 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 17:24.01 | *** join/#asterisk thomastim (n=anonymou@ntserver01.thomastonschools.org) |
| 17:24.13 | thomastim | hullo! |
| 17:24.23 | AlexCTI | Hi, Anyone familiar with Xlite softphone? I can make outbound calls and the phone dial, but when the remote pick uo the phone get silence and the softphone too |
| 17:25.56 | *** join/#asterisk hnupik (n=hnupik@chello082119119139.chello.sk) |
| 17:26.55 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:27.43 | hnupik | hello, i'am doing a sip-h323 signalalling conversion, i'am thinking about using asterisk with asterisk-oh323 module for this purpose. Will then the whole call traffic (with datagrams containing the voice) have to go thru the asterisk server, or only the signaling? (i'am in a analysis phase so i'am thinking about this issue) |
| 17:28.09 | saftsack | exten => h,2,Goto(faxout,s,1) |
| 17:28.09 | saftsack | [faxout] |
| 17:28.09 | saftsack | exten => s,1,Dial(Zap/1/*) |
| 17:28.09 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 17:28.21 | saftsack | this doesnt work the call is destroyed immediately, but why? |
| 17:29.47 | watchy | i wish i had a nuclear weapon so i could visit cisco |
| 17:31.06 | watchy | how does anyone get anything done if ciscos gonna take 2 weeks to let me get a sip image |
| 17:31.40 | [TK]D-Fender | watchy : You sign up 2 weeks before you get your hands on your equipment :) |
| 17:31.48 | Seldon1975 | On my Polycom 501/601s, when I go Menu > Features > Contact Directory > Add and add an entry I get a mesage 'Busy! Please Try Again!' on the phone's LCD and it fails to add the contact |
| 17:31.56 | Seldon1975 | has anyone experienced this? |
| 17:31.58 | *** join/#asterisk Samoied (n=Samoied@201.24.73.74) |
| 17:32.01 | [TK]D-Fender | And you wouldn't want to visit Cisco with a nuke... you'd want it delivered :D |
| 17:32.07 | watchy | [TK]D-Fender: my equip was working fine |
| 17:32.07 | endre | lol |
| 17:32.19 | *** join/#asterisk thomastim (n=anonymou@ntserver01.thomastonschools.org) |
| 17:32.23 | rkioko | what gsm channel banks are recommended to work with digium /sangoma cards |
| 17:32.25 | watchy | i plug it in and i get some shit about application invalid |
| 17:32.33 | file[desk] | woot trouble ticket |
| 17:32.42 | thomastim | hey, who was the x-lite guy? |
| 17:32.51 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
| 17:32.57 | AlexCTI | me |
| 17:33.06 | AlexCTI | x-lite guy |
| 17:33.08 | thomastim | oh, sorry about that, machine has been flaking out lately |
| 17:33.19 | thomastim | i use x-lite as a SIP phone with asterisk |
| 17:33.28 | thomastim | did someone answer your questions already? |
| 17:33.31 | AlexCTI | that's what i'm try to do |
| 17:33.36 | thomastim | ok |
| 17:33.41 | thomastim | maybe i can help |
| 17:33.43 | AlexCTI | no jet, |
| 17:33.55 | AlexCTI | the issue that i have is i get silence after the call is cnnected |
| 17:34.09 | thomastim | you're talking to someone else? |
| 17:34.20 | thomastim | are you going through a NAT/firewall? |
| 17:34.29 | AlexCTI | no |
| 17:34.44 | thomastim | is the other party connected through one? |
| 17:34.47 | AlexCTI | a router |
| 17:35.04 | thomastim | can you check your voicemail and so forth? |
| 17:36.04 | rastacouette | i ve a problem with my disa configuration someone know ? |
| 17:37.30 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 17:37.43 | thomastim | Alex, still there? |
| 17:37.59 | thomastim | ...on another note |
| 17:38.14 | thomastim | is anyone here familiar with asterisk's internal functions? |
| 17:38.18 | thomastim | specifically in file.c |
| 17:38.25 | thomastim | is there a reference somewhere? |
| 17:40.31 | }btorch{ | has anyone gotten * to connect to FWd over sip ? |
| 17:41.24 | drumkilla | thomastim: http://www.asterisk.org/doxygen/ |
| 17:41.41 | [TK]D-Fender | }btorch{ : I have, whats the problem? |
| 17:41.56 | saftsack | [TK]D-Fender, hi do you have experiences with the h option? |
| 17:42.00 | hnupik | has anyone experience with h323 and asterisk? does the oph323 support teleconferencing? |
| 17:42.13 | thomastim | D'OH! thanks drumkilla |
| 17:42.14 | [TK]D-Fender | saftsack : not really, and you asked me earlier. |
| 17:42.19 | drumkilla | thomastim: no problem |
| 17:42.20 | saftsack | because it doesnt work here, that the h option dials |
| 17:42.32 | saftsack | ok i thought your answer was for the callfile |
| 17:42.37 | brif8 | hi all, is chanspy support in the CVS-Head 05/20/05 03:33:04 version and if not how can it be enabled? |
| 17:42.57 | drumkilla | you seriously need to update |
| 17:43.02 | drumkilla | very, very badly |
| 17:43.24 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 17:44.59 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:45.06 | [TK]D-Fender | saftsack : For fax, callfile, AND "h" :) |
| 17:45.15 | brif8 | yeah I agree, but it's a production server and I'm trying to convience management that it it way past due |
| 17:45.24 | saftsack | ^^ |
| 17:45.25 | _Sam-- | sackman: you faxing yet? |
| 17:45.45 | drumkilla | brif8: wanting chanspy is a good reason to update to 1.2 |
| 17:46.03 | brif8 | 1.2 the latest stable right ? |
| 17:46.11 | blitzrage | latest release , yes. |
| 17:46.28 | brif8 | cool ok thanks I'll use that info. thanks guys |
| 17:46.33 | blitzrage | we don't call it "stable" anymore :D |
| 17:46.53 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 17:46.55 | brif8 | sure I haven't come across an unstable * anyway |
| 17:47.35 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 17:48.14 | }btorch{ | [TK]D-Fender: I keep getting a failed to authenticate and timeout NOTICe |
| 17:48.53 | }btorch{ | [TK]D-Fender: I have made several changes to my sip and extension files but nothing I have tried worked so far |
| 17:49.08 | *** join/#asterisk jdv79 (n=jdv79@ool-4573b9df.dyn.optonline.net) |
| 17:49.24 | jdv79 | hello |
| 17:49.25 | }btorch{ | [TK]D-Fender: I can connect to the FWD using xten fine |
| 17:50.34 | MstlyHrmls | Seldon1975: are you using TFTP & 1.6.3 or earlier? |
| 17:50.59 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-109.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:51.02 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 17:51.04 | jdv79 | random question - does asterisk support silence suppression? |
| 17:51.32 | [TK]D-Fender | }btorch{ : Pastebin your sip.conf |
| 17:51.38 | justinu | random answer - no |
| 17:51.43 | _Sam-- | maybe cheapest and most effective way to make sure if there is a problem with the CF....just put another CF on hdb of same thing ....make grub on hdb which loads hda as root...and hda also has grub...if hda fails, you just pick different grub option to boot hdb.... |
| 17:51.44 | _Sam-- | er |
| 17:51.55 | JunK-Y | jdv79: after a record ? yes u can remove silence. |
| 17:52.27 | jdv79 | JunK-Y, i don't follow you |
| 17:52.32 | }btorch{ | [TK]D-Fender: I'm actually doing so now |
| 17:52.56 | jdv79 | during a call is what i mean |
| 17:53.12 | JunK-Y | when u record, after u finished talking, u can remove silence (blank). |
| 17:53.19 | JunK-Y | then i misunderstood the question :) |
| 17:53.26 | JunK-Y | then its no. |
| 17:53.27 | jdv79 | Juggie, thanks...:) |
| 17:54.07 | *** part/#asterisk zaptel (n=just@nat1.inalambrica.net) |
| 17:54.26 | *** join/#asterisk kiwnix (n=egarcia@58.red-82-158-154.user.auna.net) |
| 17:55.51 | }btorch{ | [TK]D-Fender: http://pastebin.com/499532 |
| 17:57.22 | *** join/#asterisk jyukes (n=jameshot@138.89.253.56) |
| 17:57.22 | *** join/#asterisk RoyK (n=roy@ti211310a080-2622.bb.online.no) |
| 17:59.27 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 18:00.25 | *** join/#asterisk heath__ (n=heath__@12-215-33-205.client.mchsi.com) |
| 18:01.11 | [TK]D-Fender | }btorch{ : lookint at it |
| 18:01.13 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 18:01.24 | rastacouette | some one knows how we can cancel a call forwarded by disa ? before the correspondant answer |
| 18:01.49 | }btorch{ | [TK]D-Fender: thanks |
| 18:03.21 | trixter | }btorch{ [TK]D-Fender: I keep getting a failed to authenticate and timeout NOTICe |
| 18:03.25 | trixter | for every provider or only one? |
| 18:03.50 | trixter | telepacket for example has for about a month now timed out a bunch on me, it appears to be their database server and not their actual system ... |
| 18:03.57 | [TK]D-Fender | }btorch{ : Your localnet is 255.255.252.0? |
| 18:04.03 | *** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu) |
| 18:04.04 | [TK]D-Fender | for subnet mask. |
| 18:04.18 | }btorch{ | [TK]D-Fender: yeah |
| 18:09.10 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:09.10 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
| 18:09.17 | thomastim | people sell routers that don't accept a netmask? |
| 18:09.52 | }btorch{ | [TK]D-Fender: yeah the numbers are right I have doubled checked it .. now I increased the register timout to see what happens |
| 18:10.14 | [TK]D-Fender | }btorch{ : Itsits not one of the 2, then I'm not sure what to say... |
| 18:10.16 | Nugget | I use an airport extreme which won't (easily) let you do NAT unless DHCP is also enabled. You have to configure it by hand with SNMP if you want to do NAT without also DHCP. |
| 18:10.30 | Nugget | it has no problem with whatever netmask I give it, though |
| 18:10.32 | file[desk] | Nugget!!! |
| 18:10.47 | }btorch{ | [TK]D-Fender: what is strange is that on the * box I tried to use xten with the same config and it worked no NAT problem or authentication |
| 18:10.54 | file[desk] | Nugget: what flavor sauce do you recommend today? |
| 18:11.06 | Nugget | BBQ |
| 18:11.22 | }btorch{ | * CLI sip show registry shows State= Auth.sent |
| 18:11.29 | zyke | is it possible to have 2 different IP addresses under host option in sip and iax.conf files? |
| 18:12.21 | *** join/#asterisk jyukes_ (n=jameshot@138.89.253.56) |
| 18:13.25 | fenlander | hi, does anyone know anything about the changes to Asterisk to support the new SPA-9000? |
| 18:13.35 | }btorch{ | [TK]D-Fender: are you behind a NAT ? |
| 18:13.52 | *** join/#asterisk BriSch (n=BriSch@dslb-084-059-114-223.pools.arcor-ip.net) |
| 18:14.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:15.40 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 18:16.01 | }btorch{ | [TK]D-Fender: hey now I got a new warnning saying probably a DNS error for registration to 7272727@fwd.pulver.com |
| 18:16.29 | [TK]D-Fender | }btorch{ : at work I am, at home, no |
| 18:16.58 | *** join/#asterisk dash (n=washort@adsl-147-100-148.bhm.bellsouth.net) |
| 18:17.34 | dash | Hi. Someone want to help me understand some elements of asterisk's behaviour with SIP? |
| 18:18.03 | nettie | hey guys anyon know how to pass the language to festival? Using text2wave afaik doesnt support language option.. uhmm any idea please? |
| 18:18.11 | dash | I have a proxy registering with asterisk; when calls get routed to asterisk by the proxy, asterisk sends a 407 |
| 18:18.48 | dash | my proxy resends the INVITE with proxy auth info, and asterisk says "Ignoring this request" and replies with a 488 |
| 18:19.09 | dash | and I have /no/ clue why. any ideas how to coax more information out of it? |
| 18:19.13 | Dandan | anyone using spa 1001 with a fax machine? |
| 18:19.15 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 18:19.18 | Dandan | i need some help... :/ |
| 18:19.25 | Dandan | it gives me unknown codec 100 |
| 18:19.26 | justinu | dash: enable full logging in logger.conf |
| 18:19.45 | *** join/#asterisk tugalone (n=tugalone@host-24-225-212-25.patmedia.net) |
| 18:19.48 | fndude | Are there any other options for this timing signal besides ztdummy and ztrtc? I am getting slooow echo filled playback from MOH.... |
| 18:19.49 | *** join/#asterisk antoni_ (n=antoni@69.79.72.66) |
| 18:19.57 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 18:19.59 | justinu | dash: but usually, 488 is a codec mismatch |
| 18:20.37 | dash | justinu: it's not a codec mismatch: if I have my useragent register directly with asterisk (instead of through my proxy), it works |
| 18:21.23 | *** join/#asterisk Ti-dan (n=eee@207.107.208.137) |
| 18:21.45 | antoni_ | hello everybody, I want to start with asterisk, what distro is recommended? |
| 18:21.54 | hnupik | does anyone have a clue what this error could be? chan_oh323.c:3385 setup_h323_connection: Call 'ip$192.168.1.2:3577/30789-b1312c4d' invalid direction request |
| 18:22.08 | justinu | dash: the other thing that'll cause 488 is if "Content-Type" is not "application/sdp" |
| 18:22.25 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 18:22.37 | dash | justinu: Hmm |
| 18:22.44 | dash | no, that seems to be right, also |
| 18:22.45 | justinu | i just looked over the code for you, and that's it |
| 18:22.53 | justinu | no other way to get 488 as a reply |
| 18:23.02 | justinu | pastebin your sip debug, lets take a look |
| 18:23.13 | dash | justinu: Huh, bizarre. Where's this at? i oughta take a look at it too |
| 18:23.25 | justinu | channels/chan_sip.c |
| 18:23.28 | *** join/#asterisk razu_ (n=razu@195.222.10.105) |
| 18:23.41 | dash | well yeah obviously. which lines :) |
| 18:23.51 | justinu | just search for 488 |
| 18:23.58 | justinu | it's kinda "all over the place" :P |
| 18:24.06 | dash | so i've noticed |
| 18:24.07 | *** join/#asterisk fulgas (n=fulgas@209.8.233.242) |
| 18:24.41 | justinu | turn on full logging, and pastebin some of the output of the full log |
| 18:24.41 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 18:25.49 | dash | justinu: http://rafb.net/paste/results/issCPB56.html |
| 18:26.27 | justinu | that's just the sip debug... there's a way to get more verbose output |
| 18:26.33 | justinu | take a look at logger.conf |
| 18:26.37 | dash | all righty |
| 18:26.44 | Katty | hi. |
| 18:26.49 | *** join/#asterisk PMantis_C (n=sswitzer@66.251.89.34) |
| 18:27.25 | dash | ah, debug |
| 18:29.24 | justinu | looks like for whatever reason process_sdp() isn't happy |
| 18:29.25 | PMantis_C | My wife is complaining about call quality, dropped calls, audio turning to one-way during a good call, etc. Best way to diagnose problems? Ethereal? |
| 18:29.30 | justinu | we'll probably see why in the debug output |
| 18:29.50 | justinu | ethereal can help you out |
| 18:29.55 | dogtanian | <PROTECTED> |
| 18:30.00 | justinu | it has some RTP analysis stuff in it |
| 18:30.01 | *** join/#asterisk FastJack (i=fastjack@p5091EADD.dip.t-dialin.net) |
| 18:30.10 | dash | justinu: yeah, but there's no RTP yet |
| 18:30.28 | *** part/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 18:30.29 | dash | justinu: well if nothing else I can crank up gdb |
| 18:30.33 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 18:30.34 | PMantis_C | dogtanian: Dunno... it's cable, so shared in neighborhood |
| 18:30.42 | dogtanian | hmm |
| 18:30.51 | dogtanian | i'd guess at net conjection |
| 18:30.52 | PMantis_C | justinu: Anything in particular I should be watching? |
| 18:30.55 | *** join/#asterisk [Mr_X] (i=mrx@85.206.141.95) |
| 18:31.14 | justinu | run a capture, load it into ethereal, and use the RTP Stream analysis |
| 18:31.18 | dogtanian | maybe see what happens if you make calls at stupid'o'clock in teh morning |
| 18:31.31 | PMantis_C | justinu: run a capture with what? |
| 18:31.35 | dash | justinu: oh heh, thought you were talking to me, never mind :) |
| 18:31.47 | justinu | dash: i bet this is a problem with speex for some reason |
| 18:31.48 | dash | PMantis_C: ethereal does capturing too |
| 18:32.01 | justinu | dash: try disallowing it |
| 18:32.08 | PMantis_C | dash: No X on the Asterisk box. :) |
| 18:32.14 | dash | PMantis_C: so use tethereal |
| 18:32.20 | dash | or remote X ;-) |
| 18:32.24 | justinu | PMantis_C: use tetheral or tcpdumop |
| 18:32.26 | justinu | tcpdump |
| 18:32.31 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 18:32.37 | dash | justinu: OK, but I doubt it -- i enabled it in desperation when this started :) |
| 18:32.39 | PMantis_C | tcpdump I know about... |
| 18:32.51 | justinu | dash: try disabling all codecs except g711u |
| 18:32.56 | dash | ethereal will read dumps in pretty much any format |
| 18:32.57 | dash | justinu: OK. |
| 18:32.58 | }btorch{ | has anyone gotten * to connect to FWd over sip ? |
| 18:33.09 | dash | }btorch{: yeah, it's not hard |
| 18:33.14 | }btorch{ | opps sorry fo that |
| 18:33.22 | PMantis_C | }btorch{: yeah.. example in extensions.conf |
| 18:33.23 | *** part/#asterisk [Mr_X] (i=mrx@85.206.141.95) |
| 18:33.26 | masonf | any ideas why my reload command doesn't print anything |
| 18:33.45 | dash | masonf: console verbosity is set too low? |
| 18:33.50 | thomastim | g711? |
| 18:33.57 | masonf | should it? I just upgraded? no lots of v's |
| 18:33.58 | }btorch{ | well my doesn't I'm running sip debug now to see if I can figure out this mystery |
| 18:34.01 | thomastim | i get horrible stream quality on that |
| 18:34.16 | PMantis_C | justinu: So, with tcpdump, should I watch communications with a specific port, IP, etc ? |
| 18:34.17 | masonf | Verbosity is at least 7 |
| 18:34.17 | masonf | voip*CLI> reload |
| 18:34.19 | thomastim | all choppy with lots of dropout |
| 18:34.27 | dash | thomastim: sure, but it's easy to support :) |
| 18:34.34 | *** join/#asterisk Weezey (n=ohno@206.186.52.84) |
| 18:34.35 | *** join/#asterisk leopardus (n=leopardu@217.22.180.105) |
| 18:34.39 | dash | justinu: Same results with just pcmu |
| 18:34.41 | Weezey | anyone in the SPA9000 webinar? |
| 18:34.45 | justinu | pmantis: use tcpdump to write an output file |
| 18:34.59 | justinu | pmantis: tcpdump -s0 -w voip.cap |
| 18:35.04 | dash | PMantis_C: probably just all udp packets |
| 18:35.10 | justinu | dash: ok, did you enable the debug output? let's look in /var/log/asterisk/full |
| 18:35.42 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@63.150.226.34) |
| 18:35.51 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
| 18:35.51 | dash | justinu: i'm not seeing any etra stuff in full |
| 18:35.55 | dash | x |
| 18:35.58 | PMantis_C | Guess I could do that for a day, then ask my wife to write down times when things happen. |
| 18:36.00 | dash | justinu: hmmm |
| 18:36.15 | PMantis_C | Perhaps I should combine this with a ping response log |
| 18:36.25 | thomastim | i have an asterisk internal functions question about playing streams. anyone familiar with the internal API? |
| 18:36.29 | thomastim | question* |
| 18:36.35 | dash | justinu: logger.conf says: full => notice,warning,error,debug,verbose |
| 18:37.07 | dash | justinu: but i'm not seeing anything |
| 18:37.15 | justinu | yeah... did you do a "logger reload"? |
| 18:37.29 | dash | justinu: lemme try that |
| 18:40.52 | thomastim | ok |
| 18:41.20 | }btorch{ | can some one pastebin a sip and extension config that works with FWD for me behind nat |
| 18:41.21 | thomastim | let me try this from another angle: does anyone here support an *PBX in a locale where you aren't using english |
| 18:41.25 | thomastim | ? |
| 18:43.15 | Nugget | maybe you should just ask your real question. |
| 18:43.53 | masonf | is anyone running freebsd from ports? |
| 18:44.01 | }btorch{ | is this correct ? To: <sip:fwd.pulver.com> |
| 18:44.07 | Ti-dan | hello, with asterisk 1.2.1, how can I make my moh directory played randomly, it just plays sequentially (using mpg123) |
| 18:44.13 | masonf | is anyone running asterisk from freebsd ports? |
| 18:44.21 | dash | }btorch{: hmm, needs a localpart |
| 18:44.21 | *** join/#asterisk SludgeMa_ (n=SludgeMe@dns1.cybergeardevices.com) |
| 18:44.57 | thomastim | Nugget: are you familiar with file streaming functions in asterisk? |
| 18:44.57 | }btorch{ | that's what I got with a sip debug on CLI though |
| 18:45.03 | Nugget | maybe you should just ask your real question. |
| 18:45.10 | Nugget | instead of trying to prequalify all of us first. |
| 18:45.23 | Nugget | if someone knows, they'll help. |
| 18:45.30 | Nugget | if not, it'll be in the logs for posterity |
| 18:45.36 | dash | justinu: Nope, no good: no debug output after the second INVITE |
| 18:46.16 | masonf | Reload will not print anything even with verbosity set above 7 |
| 18:46.17 | justinu | gdb for you, i guess then |
| 18:46.35 | dash | justinu: Probably. Going to try comparing this to a client I know works first |
| 18:46.39 | PMantis_C | justinu: tried... tcpdump -s0 udp no data from inside the packet. Would this be necessary? |
| 18:46.47 | masonf | this isn't that bad but niether does sip debug ip <hostname> |
| 18:46.51 | dash | justinu: but thank you for your help, a sanity check on this stuff is always appreciated. :) |
| 18:46.56 | justinu | np |
| 18:47.03 | justinu | dash: let us know what you find out |
| 18:47.20 | justinu | pmantis: "no data from inside the packet"? |
| 18:47.33 | dash | justinu: OK. |
| 18:48.36 | PMantis_C | justinu: It tells me: TIMESTAMP IP ADDR:port > ADDR:port, UDP, length: blah |
| 18:48.54 | justinu | pmantis: tcpdump output sucks, tell tcpdump to save it as a file with -w |
| 18:49.04 | PMantis_C | justinu: I assumed that to diagnose, someone would need to see what the packet contains |
| 18:49.06 | justinu | then trnasfer that file to your workstation, and load it in ethereal |
| 18:49.16 | obiwanmikenolte | Ti-Dan: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 18:49.27 | PMantis_C | justinu: Ok, I'll try that.. didn't know it would differ from command line to -w |
| 18:50.46 | Ti-dan | obiwanmikenolte : thank you |
| 18:51.07 | PMantis_C | justinu: Wow! What a difference in output. Thank you. |
| 18:51.24 | justinu | yeah, ethereal rules |
| 18:52.01 | PMantis_C | justinu: Hmmm |
| 18:52.12 | mrdigital | ....this global thing is getting annoying |
| 18:52.16 | tzanger | *WOW* this is fucked up |
| 18:52.22 | fulgas | anyone knwos why a t405p end gives this error "Unassigning channel 3/6! " after ztcg -vvv |
| 18:52.30 | fulgas | *2nd |
| 18:53.41 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 18:53.41 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 18:53.41 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 18:54.14 | *** part/#asterisk marv[work] (n=timr@64.89.118.139) |
| 18:54.22 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 19:00.40 | justinu | PMantis_C: do the capture while wife is on the phone |
| 19:01.02 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 19:01.11 | *** join/#asterisk ToTo (n=ToTo@host14-134.pool872.interbusiness.it) |
| 19:01.52 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 19:01.56 | *** join/#asterisk darkskiez (n=darkskie@bb-195-172-53-125.ukonline.co.uk) |
| 19:05.17 | *** join/#asterisk _DAW (n=bob@adsl-156-90-112.msy.bellsouth.net) |
| 19:10.38 | *** join/#asterisk Tili (i=Tili@202-133-65-45-dialup.sat.net.pk) |
| 19:10.41 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 19:11.18 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 19:12.26 | zyke | is it possible to have 2 different IP addresses under host option in sip and iax.conf files? |
| 19:13.09 | bsdfreak | anyone know of a good program for windows that allows someone to use bluetooth to detect the presence of their cell phone and when it's not detected have asterisk (through the manager or otherwise) forward calls to that cell? |
| 19:14.00 | *** join/#asterisk oej (n=oej@h4n1fls301o1036.telia.com) |
| 19:15.08 | *** join/#asterisk saftsack (n=oliver@p54A7CB3D.dip.t-dialin.net) |
| 19:15.10 | saftsack | hi |
| 19:15.25 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 19:15.32 | cnet2 | what modules should i load to make the tdm2400 work? |
| 19:15.35 | saftsack | tzanger: are you here? :) |
| 19:15.49 | saftsack | cnet2: beronet.com and then in documentations |
| 19:15.53 | saftsack | it is described there |
| 19:16.03 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 19:16.16 | cnet2 | saftsack thanks |
| 19:17.07 | rastacouette | some one knows how we can cancel a call forwarded by disa ? before the correspondant answer |
| 19:18.57 | tzanger | saftsack: yes |
| 19:18.59 | tzanger | kind of though |
| 19:19.13 | saftsack | ok because the h extension doesnt work :( |
| 19:19.37 | saftsack | exten => h,1,Wait(8) throws me out after just 2 seconds |
| 19:19.59 | cnet2 | saftsack: the tdm2400p is relatively new card, it wasn't on the docs.. :S |
| 19:20.09 | saftsack | oh sry |
| 19:20.17 | saftsack | what is new? the p or what? |
| 19:20.21 | cnet2 | anyone know about installation instrctions for the 2300 ? |
| 19:20.24 | cnet2 | 2400 |
| 19:20.27 | saftsack | tzanger: do you have an idea? |
| 19:20.31 | cnet2 | the new is 2400 |
| 19:20.34 | watchy | man |
| 19:20.40 | watchy | i just bought sip firmware off ebay |
| 19:20.42 | saftsack | and what was the old one big one card= |
| 19:20.47 | watchy | beats waiting 2 weeks for cisco |
| 19:21.21 | tzanger | saftsack: what card? |
| 19:21.29 | cnet2 | there's cards for T1, and cards for 4 ports TDM4XX, but the TDM24 has 24 analog ports |
| 19:21.42 | Zodiacal | oh wow do i really save that much? look at the "You Save" its a negative: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-35768056832.htm |
| 19:21.44 | saftsack | tzanger: the fax is connected over a tdm400 |
| 19:21.50 | saftsack | with zaptel |
| 19:22.17 | tzanger | and what's not waiting? |
| 19:22.45 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 19:22.54 | saftsack | you said me to use the h extension and it didnt work so i thought to test, if h works at all |
| 19:23.07 | saftsack | so i did a simple exten => h,1,Wait(8) |
| 19:23.33 | saftsack | then it starts to wait but after 1 second zap2-2 hangs up and kill the action |
| 19:23.51 | *** part/#asterisk SludgeMa_ (n=SludgeMe@dns1.cybergeardevices.com) |
| 19:25.17 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 19:25.18 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 19:25.38 | *** join/#asterisk seele_ (n=seele@200.124.172.72) |
| 19:26.23 | rayvd | Howdy doody. |
| 19:26.45 | saftsack | tzanger: do you have an idea? |
| 19:27.27 | *** join/#asterisk jahani2 (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 19:27.56 | seele_ | Hi, i need help with this: my ring groups aren't working, i put some extens to a rign group, and they all get busy tone |
| 19:29.02 | [TK]D-Fender | Zodiacal : That seems to be the NORMAL price actually.. maybe their list is wrong... |
| 19:29.26 | seele_ | <PROTECTED> |
| 19:29.55 | *** part/#asterisk Ti-dan (n=eee@207.107.208.137) |
| 19:30.51 | *** join/#asterisk Defraz (i=t0tal@72.24.26.215) |
| 19:32.26 | *** join/#asterisk opensa_ivan (n=istepani@201.250.12.15) |
| 19:33.11 | watchy | these new mac macbooks can run windows? |
| 19:34.33 | jsharp | Aieeeee! |
| 19:34.37 | Ivan_Stepaniuk | hi there |
| 19:35.16 | cnet2 | i'm installing zaptel, i loaded the wcfxo module. (the onlyone that did load, w/o errors), and with the ztcfg -v i get errors, when i change the zaptel.conf file. ( i have a tdm2401, that's one FXO module with 4 fxos) |
| 19:36.14 | cnet2 | my lspci show a unknown Ethernet card, but i have been told, that doesn't matter |
| 19:36.20 | Ivan_Stepaniuk | what is the error ztcfg gets? |
| 19:36.39 | cnet2 | Ivan_Stepaniuk: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 19:36.59 | jsharp | Your zaptel.conf file is wrong. |
| 19:37.05 | jsharp | You need to use FXS signalling on FXO cards. |
| 19:37.16 | cnet2 | i've put fxsks=1-4 |
| 19:37.16 | jsharp | "Oh. Nevermind then. |
| 19:37.59 | cnet2 | , i still think is something with the zaptel recognizing the card.. is there a way for me to check that? |
| 19:38.00 | jsharp | Look in your dmesg output |
| 19:38.13 | jsharp | and cat /proc/sys/zaptel (I think it is) |
| 19:39.01 | lo_tech | zttool 4tw! |
| 19:39.27 | *** join/#asterisk imagine (n=imagine@p54ACF20E.dip.t-dialin.net) |
| 19:39.32 | cnet2 | the zttool only shows the ZTDUMMY/1 |
| 19:39.45 | Ivan_Stepaniuk | oops |
| 19:39.47 | Ivan_Stepaniuk | no idea |
| 19:40.01 | lo_tech | bad sign...lsmod says? |
| 19:40.10 | jsharp | Its not seeing your FXO modules, then. |
| 19:40.14 | [TK]D-Fender | cnet2 : That should be the WCTDM module you should be loading.... |
| 19:40.22 | cnet2 | it shows zaptel wcfxo ztdummy |
| 19:40.25 | imagine | good evening asterisk-users ! |
| 19:40.27 | [TK]D-Fender | not WCFXO, thats only for X100's |
| 19:40.35 | *** join/#asterisk RoyK (n=roy@ti211310a080-2622.bb.online.no) |
| 19:41.33 | Ivan_Stepaniuk | i have a question related to x100p, maybe someone know; its possible to make work more than one "Generic Clone Board" on the same box? |
| 19:41.39 | cnet2 | i see..., when i try to load the wctdm i get this 'wctdm: disagrees about version of symbol zt_receive' (unknown symbol in module) |
| 19:42.06 | jsharp | Sounds like you've got mismatched modules. |
| 19:42.06 | Ivan_Stepaniuk | do # modinfo wctdm |
| 19:42.24 | Ivan_Stepaniuk | and see if all modules it depends on are loaded |
| 19:42.41 | cnet2 | depends on zaptel only |
| 19:42.47 | cnet2 | and zaptel is loaded :S |
| 19:42.51 | Ivan_Stepaniuk | :/ |
| 19:43.30 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 19:43.36 | cnet2 | there's a module called (crc_ccitt) that depends on zaptel, could that be the new card? |
| 19:43.36 | a1fa | anybody used that voice changer? |
| 19:43.40 | cnet2 | new card module i mean |
| 19:44.38 | a1fa | what are toll numbers in US? |
| 19:44.39 | a1fa | 900? |
| 19:45.04 | jsharp | 900 977 |
| 19:45.07 | justinu | 800,888,877,866 |
| 19:45.14 | lo_tech | any non-local number can incur a toll |
| 19:45.14 | a1fa | those are toll free |
| 19:45.18 | justinu | oh, toll free |
| 19:45.23 | a1fa | lo_tech : i have free longdistance |
| 19:45.31 | file[desk] | toll-free numbers can technically incur charges when dialed international too :D |
| 19:46.25 | a1fa | :p |
| 19:46.30 | a1fa | anyway |
| 19:46.38 | a1fa | i am about to apply that patch for voice changer |
| 19:46.49 | a1fa | anybody had fun with that? |
| 19:47.26 | Ivan_Stepaniuk | its possible to make work more than one of those x100p clone (ambient 3200) on the same box? zttool shows me three spans |
| 19:47.50 | Ivan_Stepaniuk | (i have 3 of these crapmodems in) |
| 19:47.58 | [TK]D-Fender | justinu : those #'s you provided were TOLL-FREE. |
| 19:49.41 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 19:49.46 | a1fa | can asterisk generate CNG? |
| 19:51.29 | justinu | no |
| 19:51.39 | justinu | fender: and all this time, i had no idea :P |
| 19:51.50 | Ivan_Stepaniuk | a1fa: i think no |
| 19:51.59 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 19:52.15 | Ivan_Stepaniuk | " Implement CNG, VAD and DTX - MustDie" is in http://www.voip-info.org/wiki/view/Asterisk+Wishlist |
| 19:52.22 | Ivan_Stepaniuk | as a whishlit item |
| 19:52.30 | Ivan_Stepaniuk | so i supose its not implemented |
| 19:52.59 | Ivan_Stepaniuk | there was a thread on the mailing list about cng a time ago |
| 19:53.31 | *** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu) |
| 19:53.42 | a1fa | yeah |
| 19:53.44 | a1fa | it kind of bugs people |
| 19:53.46 | a1fa | when you stop talking |
| 19:53.49 | a1fa | and it stops transmiting |
| 19:53.55 | a1fa | and they think you hung-up on them |
| 19:54.05 | a1fa | and then they are like" HELLO ARE YOU STILL THERE" |
| 19:54.23 | Ivan_Stepaniuk | yes, i dont know why its called "confort" noise |
| 19:54.35 | *** join/#asterisk Kernel_core (n=I@29.230.dial-up.xter.net) |
| 19:54.46 | a1fa | it should be called |
| 19:54.55 | a1fa | I-AM-STILL-HERE-NOIS |
| 19:54.56 | Ivan_Stepaniuk | my english is terrible, but there must be another word for that |
| 19:55.02 | a1fa | I-AM-STILL-HERE-NOISE |
| 19:55.08 | Ivan_Stepaniuk | yeee :) |
| 19:55.14 | a1fa | where r u from |
| 19:55.24 | dash | Ivan_Stepaniuk: thinking your NAT broke your voip /again/ is uncomfortable |
| 19:55.33 | Kernel_core | Ivan_Stepaniuk: Parooski Panimayte ? ;) |
| 19:55.42 | a1fa | Argentina |
| 19:55.43 | a1fa | <PROTECTED> |
| 19:56.05 | Ivan_Stepaniuk | me too |
| 19:56.20 | Ivan_Stepaniuk | its a small world =P |
| 19:56.45 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 19:58.05 | a1fa | check that |
| 19:58.33 | Ivan_Stepaniuk | che Buenos Aires? |
| 19:58.35 | *** join/#asterisk FastJack (i=fastjack@p5091EADD.dip.t-dialin.net) |
| 19:58.36 | dash | justinu: Interestingly, the latest release of asterisk does not send a 488, but it does say "Ignoring this INVITE request". |
| 19:59.00 | *** join/#asterisk MeneMMateo (n=MeneMMat@mateo.xs4all.nl) |
| 20:00.18 | dash | ... |
| 20:00.18 | dash | Oh. |
| 20:00.49 | rastacouette | some one knows how we can cancel a call forwarded by disa ? before the correspondant answer |
| 20:01.13 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 20:01.58 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 20:04.05 | *** join/#asterisk Vahan (i=user@phoenix.arminco.com) |
| 20:04.54 | *** join/#asterisk Speeder (n=psilva@217.129.166.236) |
| 20:05.10 | *** part/#asterisk Samoied (n=Samoied@201.24.73.74) |
| 20:05.23 | Vahan | greetings, anyone has a full list of spa-3000/2100 xml variables or example .xml ? I'm looking for the PSTN / Caller ID variable names. |
| 20:05.35 | Vahan | something like http://www.sipura.com/support/spa841faq/sample-841.xml |
| 20:06.11 | Speeder | hi, I'm having a problem setting my callerid. I'm Using Set(Callerid(num_extension)=name|num) and doesn't work.. Can u help? |
| 20:07.50 | eKo1 | Speeder: first of all this ---> Set(Callerid(num_extension)=name|num) will never work |
| 20:08.24 | *** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net) |
| 20:08.50 | brad_mssw | Set(CALLERID(name)=Myname) |
| 20:09.00 | brad_mssw | Set(CALLERID(num)=123456) |
| 20:09.01 | Vahan | Set(CALLERID(number)=value) |
| 20:09.05 | Vahan | Set(CALLERID(name)=123 |
| 20:11.32 | Vahan | btw, spa-941 rocks |
| 20:11.33 | *** join/#asterisk thomastim (n=anonymou@ntserver01.thomastonschools.org) |
| 20:11.33 | Vahan | handsfree echo can problems are 95% gone |
| 20:12.00 | Ivan_Stepaniuk | "3 channels configured." :) |
| 20:12.51 | *** join/#asterisk santiago (n=santiago@208.195.215.97) |
| 20:13.35 | *** join/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 20:14.09 | *** join/#asterisk Katty_ (n=angela@64.82.232.54) |
| 20:14.28 | rastacouette | anyone has already configured a disa on asterisk ? |
| 20:14.46 | saftsack | rastacouette: what is disa? |
| 20:14.55 | Hmmhesays | anyone ever had one install that makes you want to quit your job? |
| 20:15.00 | watchy | yes |
| 20:15.14 | watchy | i have things that make me want to slit my throat in front of customers |
| 20:15.25 | Ivan_Stepaniuk | A feature on PBX's which allows an external caller access to the PBX and all of its features such as tie and watts lines by dialing in on the DISA trunk. |
| 20:15.30 | watchy | and thats a good day |
| 20:15.41 | rastacouette | it is a remote acces on your asterisk to place a call of inside your asterisk to outside |
| 20:15.44 | Ivan_Stepaniuk | (thanks google) |
| 20:16.03 | watchy | i think i'm finnaly getting 7960g firmware |
| 20:16.10 | watchy | my friend has a cco acct with lots of access |
| 20:16.22 | watchy | i already paid cisco bitchs $12 |
| 20:16.27 | watchy | but it will take 2 weeks to setup |
| 20:16.41 | twisted[asteria] | hahhaha |
| 20:16.48 | twisted[asteria] | no it won't |
| 20:16.57 | dash | saftsack: "direct inward system access" |
| 20:16.59 | PMantis_C | ...and on tonight's news, a telephony consultant kills himself in front of customers. Experts ask if VoIP is worth the casualties. |
| 20:17.17 | PMantis_C | watchy: I couldnt' resist. :) |
| 20:17.18 | Ivan_Stepaniuk | rastacouette, the way asterisk dial plan is configured, doing a disa is not doing so much |
| 20:17.20 | twisted[asteria] | Hmmhesays, to answer your question - every other day. |
| 20:17.34 | dash | PMantis_C: someone ought to sponsor some psychiatrists to investigate the effects of VoIP on mental health |
| 20:17.43 | PMantis_C | lol |
| 20:17.54 | *** join/#asterisk CleanerX (n=nix@p54A3BF68.dip0.t-ipconnect.de) |
| 20:17.57 | twisted[asteria] | dash, that'd be hard, considering a lot of us are already crazy |
| 20:18.01 | Astar | Ivan_Stepaniuk my problem is that when i call my asterisk & after i go outside with disa i cant cancel the call |
| 20:18.11 | saftsack | dash: oha ^^ |
| 20:18.11 | thomastim | compared to our existing PBX, Asterisk makes me want to quit my job and do Asterisk installs |
| 20:18.13 | Astar | it eternaly rings |
| 20:18.20 | PMantis_C | dash: Or the effects on relationships... my wife is upset about calls dropping. I'm runngin tcpdump now to see if it catches anything. |
| 20:18.32 | Ivan_Stepaniuk | you mean, when you hang up the call still active? |
| 20:18.38 | Astar | yes |
| 20:18.41 | twisted[asteria] | PMantis_C, solution: tell her not to talk so much :P |
| 20:18.57 | dash | twisted[asteria]: well, mostly I am thinking of the day debugging this SIP stack I am writing pushed me far enough to change my nick to "z9hG4bK" |
| 20:19.13 | PMantis_C | twisted[asteria]: LOL, that'll go over well... Might I suggest, "Keeping a wife 101" ? :-) |
| 20:19.21 | Hmmhesays | i'm working with audiocodes right now, i swear this guy has to be a semi retarded half brother of the president or something |
| 20:19.22 | twisted[asteria] | PMantis_C, hah |
| 20:19.23 | tzanger | hmm |
| 20:19.23 | watchy | praise the lord |
| 20:19.29 | Hmmhesays | no farking way he got this job on his own |
| 20:19.31 | watchy | i have cisco 7960g firmware |
| 20:19.33 | watchy | praise! |
| 20:19.34 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:19.47 | tzanger | I need nonintrusive "alarm" noises... I've been told to implement "start of break/end of break" noises for the company. ugh. |
| 20:19.49 | twisted[asteria] | Hmmhesays, you should talk to SwK |
| 20:20.02 | *** join/#asterisk SexyKen (n=ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 20:20.06 | Hmmhesays | he familiar with audiocodes? |
| 20:20.06 | dash | justinu: So it looks like what's happening is that my code doesn't increment the cseq when it resends the INVITE with auth |
| 20:20.10 | twisted[asteria] | Hmmhesays, *nods* |
| 20:20.13 | dash | justinu: asterisk has a cow and ignores it |
| 20:20.15 | SexyKen | Hey guys -- How can I use a standard cordless phone with my Asterisk server? |
| 20:20.16 | justinu | dash: ahhhh |
| 20:20.18 | Hmmhesays | specifically the vxml engine? |
| 20:20.25 | twisted[asteria] | oooh okay, dunno about that |
| 20:20.27 | twisted[asteria] | lemme ask |
| 20:20.34 | dash | justinu: so i fixed that and now it's mismatching the cseq on the acks :) |
| 20:20.35 | PMantis_C | SexyKen: Get an FXS card / or FSX device |
| 20:20.36 | tzanger | SexyKen: with an FXS port |
| 20:20.43 | justinu | dash: heh |
| 20:20.52 | Vahan | SexyKen: buy a FXS adapter such as Sipura SPA-1001 or SPA-3000 |
| 20:20.54 | SexyKen | Well did I mention that the asterisk server is remotely located and is pure VOIP? |
| 20:20.58 | Hmmhesays | that would be a life saver, i'm going on vacation in 2 days and considering not coming back because of this |
| 20:21.05 | PMantis_C | SexyKen: Sorry, both abbreviations should be: FXS |
| 20:21.07 | dash | SexyKen: what vahan said, then |
| 20:21.20 | twisted[asteria] | Hmmhesays, sucks... apparently not. |
| 20:21.25 | PMantis_C | SexyKen: Or, IAXy, or TDM card, etc |
| 20:21.43 | file[desk] | Hmmhesays: !!!!!!!! |
| 20:21.50 | *** join/#asterisk subzero (n=subzero@201.238.69.14) |
| 20:21.53 | Hmmhesays | file[desk] how are you? |
| 20:21.59 | SexyKen | If the Asterisk server is remote and it's a pure VOIP solution I will need to use FXS adaptors? |
| 20:22.00 | dash | file[desk]: does he work for you? ;) |
| 20:22.08 | Hmmhesays | is anyone familiar with vxml in here? |
| 20:22.12 | subzero | hi all |
| 20:22.14 | file[desk] | Hmmhesays: great, you? |
| 20:22.33 | dash | SexyKen: get an iaxy or a sipura, get it to talk iax or sip to your asterisk server |
| 20:22.46 | Hmmhesays | file[desk] horrid |
| 20:22.52 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 20:22.54 | file[desk] | Hmmhesays: whyfor horrid? |
| 20:22.57 | subzero | looking for some help on setting up a pbx system with asterisk |
| 20:23.14 | tzanger | subzero: which PBX |
| 20:23.16 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 20:23.26 | SexyKen | Okay that's pretty cool, bestbuy wouldn't carry this stuff, would they? |
| 20:23.59 | Vahan | SexyKen: www.voipsupply.com has them at the moment. |
| 20:24.10 | Hmmhesays | file[desk] install gone horribly wrong |
| 20:24.16 | subzero | tzanger, I found an example of a 8x16 PBX and looking at having a go at it |
| 20:24.22 | PMantis_C | I have IAXy's in stock as well |
| 20:24.23 | Hmmhesays | and it came back to me cause no one is doing their freaking job |
| 20:24.27 | tzanger | doesn't tell me anything |
| 20:24.33 | tzanger | and that's likely a KSU anyway, not a pBX :-) |
| 20:26.23 | saftsack | tzanger: so do you have an idea? |
| 20:26.24 | subzero | tzanger, one sec |
| 20:26.42 | tzanger | saftsack: no I missed the explanation |
| 20:27.09 | justinu | anyone know why the startup script included with asterisk would spawn 9 instances of the asterisk process? |
| 20:27.17 | saftsack | ok. asterisk doesnt seems to do what i say ^^ |
| 20:27.20 | NDT | We have numbers pointed to our toll free...carrier sends the 10 digit dialed string...say...565455656...So the person dialed 565455656...I could read this in gnugk...but doing it like this... exten => 565455656,n,Dial(Zap/g1/5164585789) asterisk seems to only be seeing the toll free everytime |
| 20:27.33 | saftsack | it does h,1,something but it stops like all other things here :( |
| 20:28.38 | NDT | DNID seems to always show as toll free number rather then number dialed |
| 20:28.43 | PMantis_C | NDT: Do you have a line like this?: exten => 565455656,1,something here ? |
| 20:29.00 | subzero | tzanger, www.vlug.org/vlug/meetings/presentations/VLUG-Telephony.pdf that's the site... |
| 20:29.03 | file[desk] | RDNIS |
| 20:29.13 | NDT | exten => 565455656,1,Dial(Zap/g1/5164585789) <---tried it like that... |
| 20:29.24 | file[desk] | but it's only applicable on PRIs and such |
| 20:29.27 | *** part/#asterisk MeneMMateo (n=MeneMMat@mateo.xs4all.nl) |
| 20:29.36 | justinu | why is the redhat asterisk init script spawning 9 copies of the asterisk process? |
| 20:29.38 | NDT | It keeps reading 565455656 as the tollfree |
| 20:30.03 | subzero | tzanger, page 14 |
| 20:30.10 | NDT | yet in gnugk it works |
| 20:30.17 | PMantis_C | NDT: Explain "reading as the tollfree" |
| 20:31.03 | Speeder | thank's brad_mssw it's working |
| 20:31.08 | }btorch{ | is this still valid on the latest * ? Dial(SIP/${EXTEN:1}@fwd-outgoing ? |
| 20:31.18 | }btorch{ | finally got the FWD to wrok |
| 20:32.02 | NDT | PMantis_C: ok...if I dialed 565 number...gnugk would show me callerid and the number that was dialed...asterisk I thought was supposed to read that DNID as default in a context like this: exten => 5654556565,1,Dial(Zap/g1/5164585789) Yet...it is seeing 5654556565 as the 866 number instead |
| 20:32.07 | tzanger | saftsack: is your hangup dialplan trying to do anything that may require a channel to be up? |
| 20:32.37 | tzanger | subzero: ok |
| 20:33.12 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 20:33.14 | saftsack | tzanger: it just do wait(8) for testing issues atm |
| 20:33.16 | justinu | wow, thanks for the help |
| 20:33.28 | justinu | you hang around, help out tons of people's problems... when it's your turn... you get nothing |
| 20:33.36 | justinu | excellent |
| 20:33.55 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 20:34.10 | Hmmhesays | ok this pile of hell is not even trying to grab the vxml script |
| 20:34.10 | NDT | justinu: Wish I could tell ya... 8) |
| 20:34.22 | twisted[asteria] | lol |
| 20:34.27 | thomastim | justinu: perhaps it's because there are child processes that handle each connection |
| 20:34.40 | saftsack | tzanger: so it should imho work |
| 20:34.49 | jbalcomb | Is there anyway to get Asterisk to log individual state changes on a call? |
| 20:34.51 | *** join/#asterisk heymikeeh (n=me@72.29.237.163) |
| 20:34.54 | thomastim | justinu: like how Apache starts a minimum number of daemons to listen to requests |
| 20:35.00 | justinu | thomastim: then why don't they appear when I start it manually? |
| 20:35.15 | thomastim | justinu: because it takes resources to start more while under load |
| 20:35.30 | justinu | uhh |
| 20:35.32 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 20:35.41 | thomastim | justinu: i have no idea, because i just started it and about a dozen spawned |
| 20:35.44 | tzanger | saftsack: ok and? |
| 20:35.45 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 20:36.06 | justinu | this is really my first time messing with redhat init scripts |
| 20:36.09 | saftsack | after 2 seconds of waiting it breaks |
| 20:36.12 | justinu | and i'm already pisssed |
| 20:36.13 | dash | justinu: It just keeps getting better. Now Asterisk is sending 503s when I send an ACK |
| 20:36.20 | Hmmhesays | what exactly does relaxdtmf do in sip.conf? |
| 20:36.26 | thomastim | justinu: i'm not using a script. i'm just calling the binary |
| 20:36.29 | NDT | justinu: like freebsd 4.9 shows like 20 postgresql processes for me...shows all teh child processes in the same damn list drives me nuts |
| 20:36.33 | saftsack | andd the channel hangs up |
| 20:36.56 | justinu | hmm |
| 20:37.01 | [TK]D-Fender | Hmmhesays : lowers the detection threshold when you are using INBAND for DTMF. useful for shitty connections. |
| 20:37.04 | justinu | obviously, i don't "get it" |
| 20:37.12 | subzero | tzanger: what hardware would I need for that kind of setup?? |
| 20:37.31 | tzanger | is there no function that returns a timestamp anymore?? |
| 20:37.33 | *** join/#asterisk heymikeeh (n=me@72.29.237.163) |
| 20:37.45 | [TK]D-Fender | ${TIMESTAMP} ? |
| 20:37.53 | NDT | justinu: Think it all revolves around NPTL support somehow in redhat |
| 20:37.58 | thomastim | justinu: is there a problem with it, or is it the fact that it's cluttering up your process list? |
| 20:38.06 | tzanger | ahh strftime |
| 20:38.20 | thomastim | i'm using slackware and i get the same MCF of processes in the list |
| 20:38.26 | tzanger | with fantastic help, ugh |
| 20:38.31 | justinu | i dunno if there's a problem yet... but the fact that I see 9 different processes listening on UDP 5060 bugs me |
| 20:38.39 | tzanger | [TK]D-Fender: hmm that might be it |
| 20:38.41 | tzanger | subzero: do this |
| 20:38.48 | thomastim | justinu: one for each connection. make sense? |
| 20:38.50 | tzanger | exten => s,1,NoOp(time is ${TIMESTAMP}) |
| 20:38.53 | tzanger | exten => s,1,Wait(8) |
| 20:38.58 | tzanger | exten => s,3,NoOp(time is ${TIMESTAMP}) |
| 20:39.01 | NDT | justinu: Install webmin or something for a better visual on how those processes are |
| 20:39.04 | tzanger | (s,1, s,2 and s,3 of course) |
| 20:39.12 | PMantis_C | Anyone know what "BATTERY!" "NO BATTERY!" "RING!" "NO RING!" in my kernel log means? |
| 20:39.22 | *** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu) |
| 20:39.22 | tzanger | saftsack: of course the channel is hung up ... you won't get to 'h' until it is hung up |
| 20:39.48 | NDT | justinu: I love looking at the list in webmin rather then console heh |
| 20:40.02 | justinu | thomastim: i understand how httpd works, but I dind't think asterisk forked itself like that. |
| 20:40.19 | justinu | and I can't understand why when I start it from the shell, it only spawns one process |
| 20:40.22 | thomastim | justinu: well i didn't notice until you mentioned it. :> |
| 20:40.31 | justinu | but when I start it from the script, it spawns 9 |
| 20:40.34 | thomastim | oh, i see |
| 20:40.52 | thomastim | you're just doing "asterisk" from the command line? |
| 20:40.56 | justinu | yeah |
| 20:40.57 | thomastim | and it only spawns one? |
| 20:40.59 | dash | justinu: sure those aren't threads? linux assigns pid numbers to threads |
| 20:41.00 | NDT | justinu: it's funny...in fedora core 4 I see what you are saying...but in centos I don't heh |
| 20:41.00 | justinu | right |
| 20:41.10 | justinu | i'm running centos |
| 20:41.13 | dash | 'cause it's always done that for me |
| 20:41.22 | justinu | hmm |
| 20:41.23 | NDT | weird |
| 20:41.29 | watchy | what needs to be put in isc bind to specify tftp server? |
| 20:41.35 | justinu | i guess they're threads, but on another box which is actually doing stuff right now, there's only one process |
| 20:41.54 | subzero | tzanger, do what?? |
| 20:42.01 | tzanger | er not subzero, saftsack |
| 20:42.37 | PMantis_C | On Gentoo, ps -e | grep asterisk shows 22 processes. Some MOH |
| 20:42.50 | justinu | ok, i guess i won't let it bother me |
| 20:43.06 | justinu | thx guys |
| 20:43.15 | Hmmhesays | these freaking phones are sending dtmf that is slighty off in frequency |
| 20:43.31 | *** join/#asterisk q2ZvR6jR (n=mk@57.80-203-77.nextgentel.com) |
| 20:43.31 | Hmmhesays | in a 50/50 ms cadence |
| 20:46.32 | NDT | justinu: Heh look at the fedora core 4 one: http://pastebin.com/499774 |
| 20:46.36 | *** join/#asterisk forhans (n=afarhan_@59.93.67.77) |
| 20:46.36 | NDT | centos I see one lol |
| 20:46.46 | justinu | weird stuff |
| 20:46.58 | docelm0 | I use CentOS 4.2 on 20 Machines running asterisk |
| 20:47.04 | docelm0 | its my flavor of choice.. |
| 20:47.05 | justinu | yeah, i'm on centos 4.2 |
| 20:47.06 | thomastim | maybe it's an option for ps that's spec-ced in the shell profile? |
| 20:47.08 | forhans | hello all, where do the developers hang out? |
| 20:47.08 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 20:47.10 | justinu | so far so good |
| 20:47.21 | justinu | thomastim: same user, same shell |
| 20:47.35 | docelm0 | uDEV through me tho.. Didnt know it used uDEV when I was setting up my TDM card.. |
| 20:47.56 | watchy | what needs to be put in isc dhcp to specify tftp server? |
| 20:48.14 | subzero | tzanger: I have a AMD 2000MHz System to start with, the phones are not a prob. is there any other hardware I need to get??? |
| 20:48.36 | Nugget | buy a new macbook pro. :) |
| 20:48.38 | tzanger | do you plan on interfacing to PSTN? |
| 20:48.57 | tzanger | watchy: as DHCP or BOOTP |
| 20:48.58 | tzanger | ? |
| 20:49.10 | watchy | dhcp. I just wanna tell my phone a TFTP server |
| 20:50.38 | subzero | tzanger: yes I have five lines from the telephone company already. |
| 20:51.13 | tzanger | subzero: well you'll need some way to hook that up (FXO ports) -- TDM404 will do 4 of 'em. |
| 20:52.14 | *** part/#asterisk forhans (n=afarhan_@59.93.67.77) |
| 20:53.42 | subzero | tzanger: thank alot |
| 20:53.56 | tzanger | watchy: I don't see it in dhcpd.conf, you may just have to give it an option # based on the RFC |
| 20:54.08 | watchy | i found it |
| 20:54.17 | watchy | <PROTECTED> |
| 20:54.19 | }btorch{ | what do you call the last four digits on a phone ? DID ? |
| 20:54.24 | watchy | how god damn simple |
| 20:55.47 | tzanger | }btorch{: no, exchange |
| 20:55.47 | thomastim | watch it be a deprecated option now lol |
| 20:55.50 | tzanger | er no not exchange |
| 20:55.52 | tzanger | just number |
| 20:55.56 | NDT | DID is direct inward dialing heh |
| 20:56.05 | NDT | my problem atm |
| 20:56.06 | NDT | LOL |
| 20:56.15 | }btorch{ | which is what the whole number (DID) ? |
| 20:56.57 | NDT | area code + prefix + suffix isnt it? |
| 20:57.22 | malverian[work] | [TK]D-Fender, I ended up using the idea you mentioned, it worked well :) |
| 20:57.30 | malverian[work] | [TK]D-Fender, I probably should make a wiki entry for this :-P |
| 20:57.33 | tzanger | area code + prefix + exchange, maybe that is what it's called |
| 20:57.45 | tzanger | I thought exchagne was the 3 digits before the number htough |
| 20:57.47 | malverian[work] | [TK]D-Fender, I really should have done it as an AGI though ;) |
| 20:57.58 | tzanger | i.e. 5192915112 519 = area code, 291 = exchange, 5112 = number |
| 20:58.39 | justinu | last 4 digits is called the extension :) |
| 20:59.05 | NDT | justinu: Good work! you win! |
| 20:59.08 | NDT | 8) |
| 20:59.13 | justinu | what's my prize? |
| 20:59.21 | justinu | gimme gimme |
| 20:59.22 | tzanger | swift kick in the ass? |
| 20:59.28 | NDT | I dunno...maybe you will get to answer more of my dumb questions? 8) |
| 20:59.49 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 21:00.00 | justinu | probably |
| 21:00.46 | malverian[work] | [TK]D-Fender, Check out the function I made to check for schedule holidays :-P http://pastebin.ca/36393 |
| 21:01.05 | malverian[work] | Sexy ;) |
| 21:01.16 | asterboy | Just received another Polycom IP 500 phone...there are some interesting differences between other IP 500s. |
| 21:02.03 | antoni_ | What Softphone can I do with asterisk, for test |
| 21:02.25 | thomastim | x-lite |
| 21:02.32 | asterboy | A battery compartment, no jumper pins, no rubber insert top left corner and expansion slot at left. |
| 21:02.45 | thomastim | antoni_: http://www.counterpath.com/index.php?menu=download |
| 21:02.59 | justinu | asterboy: that sounds like a 601 |
| 21:03.03 | antoni_ | thomastim, tnx |
| 21:03.17 | *** join/#asterisk Falle (i=falstaf@213.141.80.88) |
| 21:03.17 | asterboy | Polycom must have been making reservations in the form factor to accomodate future options. |
| 21:03.35 | asterboy | could be a 601 mold. |
| 21:03.43 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 21:03.44 | Vahan | anyone by any chance got an example .xml for provisioning sipura 2100 or 3000? |
| 21:05.00 | watchy | YES MY PHONE IS FIXED |
| 21:05.26 | watchy | whos ready for some hot man on man love now |
| 21:06.04 | asterboy | well the only difference between a man and a woman is 1 molecule. |
| 21:06.04 | *** join/#asterisk sese1 (n=sese@host81-159-79-120.range81-159.btcentralplus.com) |
| 21:06.10 | watchy | wel |
| 21:06.13 | watchy | lets do it then |
| 21:06.20 | iDunno | let's fall in love? |
| 21:06.31 | asterboy | keep it in the family? |
| 21:06.33 | watchy | i'm down |
| 21:06.38 | watchy | are you pretty |
| 21:06.42 | asterboy | I'm up |
| 21:06.50 | watchy | huggles? |
| 21:06.57 | iDunno | no, definately boggles. |
| 21:06.58 | rob0 | boys will be boys |
| 21:07.01 | iDunno | confusedly. |
| 21:07.19 | watchy | i still dunno what was up with my phone |
| 21:07.32 | watchy | but i got the 7960g firmware and it fixed it |
| 21:08.08 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:08.32 | antoni_ | where I can find the basic configuration for asterisk |
| 21:08.49 | antoni_ | Im testing with the windows version, but I will move to linux |
| 21:08.59 | watchy | should i run 7.4 or 7.5 |
| 21:09.03 | watchy | on my 7960? |
| 21:09.07 | *** part/#asterisk dash (n=washort@adsl-147-100-148.bhm.bellsouth.net) |
| 21:10.33 | *** join/#asterisk SenorAmor (n=me@ns1.accu-com.com) |
| 21:10.35 | antoni_ | thomastim, can you give me a tutorial for starting up.... and basic configuration |
| 21:10.40 | thomastim | antoni_: i started with "phase one" of this guide http://www.wlug.org.nz/AsteriskSampleSetup |
| 21:10.48 | thomastim | antoni_: you read my mind |
| 21:10.51 | thomastim | ha ha |
| 21:10.56 | SenorAmor | Hello all. I'm having a driver issue. Is it ok to ask questions in here or should I email someone instead? |
| 21:11.06 | antoni_ | thomastim, jeje, thank you hehe |
| 21:11.07 | thomastim | antoni_: it's just a simple SIP phone set-up |
| 21:11.49 | antoni_ | thomastim, I use x-lite before, but Im installing asterisk, what do I do in asterisk? |
| 21:12.11 | antoni_ | thomastim, Im lookin on the sample setup, I got it |
| 21:12.38 | thomastim | antoni_: is this a complete new asterisk install? if so, the guide will help you set up two extensions so you can test it. |
| 21:12.41 | thomastim | ok |
| 21:13.47 | antoni_ | thomastim, ok tnx |
| 21:13.52 | thomastim | :D |
| 21:14.16 | watchy | praise allah |
| 21:14.22 | NDT | Hmmm how would you write in the dialplan if ${RDNIS} = 6092911315 then dial such and such? So it would work like this? exten => 6092911315,1,Dial(Zap/g1/8003825630) |
| 21:14.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:15.02 | NDT | hehe I pointed the extension to a sex line to test it ROFL |
| 21:15.26 | watchy | haha |
| 21:15.29 | watchy | did you get laid |
| 21:15.38 | NDT | nah...she dumped me 8( |
| 21:15.45 | watchy | sux |
| 21:15.49 | watchy | ill date you |
| 21:15.52 | NDT | lol |
| 21:15.55 | asterboy | there is always masturbation |
| 21:16.11 | watchy | or animals |
| 21:16.19 | asterboy | of furniture |
| 21:16.24 | NDT | no no...no barnyard babes |
| 21:16.31 | watchy | rape is just another word for guranteed sex |
| 21:16.34 | lesouvage | asterboy: you mean with a set of callfiles and a vibrating celphone or something like that? |
| 21:16.43 | asterboy | It puts the lotion on its skin...or it gets the hose again! |
| 21:16.56 | watchy | haha |
| 21:17.01 | NDT | I thought rape was just another name for 20yrs? |
| 21:17.03 | NDT | lol |
| 21:17.17 | }btorch{ | how come when dialing sending calls to the FWD through * sometimes I keep getting "that option is invalid" if I retry then it works |
| 21:17.31 | watchy | haha |
| 21:18.00 | asterboy | what sick God would put sex organs next to waste outlets? |
| 21:18.05 | NDT | lol |
| 21:18.33 | Ivan_Stepaniuk | lol |
| 21:18.38 | justinu | "you said rape twice" |
| 21:18.42 | justinu | "i like rape" |
| 21:18.45 | asterboy | so much for "intelligent design". |
| 21:19.12 | asterboy | I've always told women if they want to stop a rape...just shit yourself. |
| 21:19.21 | NDT | HAHAHA |
| 21:19.22 | watchy | haha |
| 21:19.26 | PMantis_C | ROFL |
| 21:19.31 | justinu | i dunno if that would be a garaunteed stop |
| 21:19.35 | justinu | some sick fucks out there |
| 21:19.35 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:20.04 | watchy | yep |
| 21:20.41 | eKo1 | and what if you just went |
| 21:21.10 | [av]bani | if this is intelligent design, then god is one sick motherfucker |
| 21:21.32 | asterboy | that has to be the conclusion. |
| 21:22.57 | NDT | heh can you use IF statements in a dialplan? |
| 21:23.18 | asterboy | surely there is a better deisgn...albeit nature is taking the most efficient method of developing organs with multi purpose...I just don't like it at all. |
| 21:23.57 | asterboy | and for cleanliness sakes...girls should wipe from front to back! |
| 21:24.58 | NDT | asterboy: Admit it...theres a lot of em so hot out there you don't care how they wipe LOL |
| 21:25.16 | justinu | you know eyebeam doesn't support early media? what a piece of shit! |
| 21:25.25 | asterboy | :P |
| 21:25.43 | asterboy | I guess I could always breath through my mouth. |
| 21:25.53 | asterboy | out of site out of mind. |
| 21:25.55 | NDT | lol... |
| 21:26.10 | justinu | you eventually learn to breath thru your ears |
| 21:26.16 | asterboy | until of course you hit a chunk...say of some undigested corn covers. |
| 21:26.27 | NDT | ack... |
| 21:27.11 | eKo1 | hahaha |
| 21:27.21 | asterboy | its all about communication. |
| 21:27.24 | eKo1 | talk about off topic |
| 21:27.37 | eKo1 | yes, and * is about communication |
| 21:27.55 | NDT | * Now talking in #asterisksickbastards |
| 21:28.01 | NDT | ;) |
| 21:28.25 | eKo1 | asstersick |
| 21:28.35 | NDT | * Now talking in #asteriskcorncovers |
| 21:28.39 | NDT | thats better |
| 21:29.05 | asterboy | lol |
| 21:31.33 | [TK]D-Fender | NDT : Yes you can use IF statements in the dialplan (GotoIF) |
| 21:32.41 | *** part/#asterisk PMantis_C (n=sswitzer@66.251.89.34) |
| 21:34.50 | NDT | [TK]D-Fender: ahh thanks 8) |
| 21:36.20 | saftsack | GotoIF rocks |
| 21:36.38 | *** join/#asterisk jyukes (n=jameshot@pcp04135114pcs.maysld01.nj.comcast.net) |
| 21:39.41 | *** part/#asterisk Vahan (i=user@phoenix.arminco.com) |
| 21:41.47 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 21:44.39 | *** join/#asterisk QuAd|Haudrauf (n=hau@port-212-202-54-134.dynamic.qsc.de) |
| 21:46.33 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 21:47.28 | justinu | turns out the early media problem with eyebeam was solved, if anyone cares |
| 21:49.45 | NDT | OK...I seem to have the syntax wrong here: exten => 5551212121,1,GotoIf($["${RDNIS}" = "4665664525"])DIAL(Zap/g1/4568798555) |
| 21:50.10 | tzanger | yeah you really have that wrong :-) |
| 21:50.14 | NDT | lol |
| 21:50.18 | tzanger | show application gotoif |
| 21:52.03 | NDT | ahh..duh see it now... |
| 21:53.01 | sshadow | hi all, is it possible with * realtime extensions to use the goto ? for example: in context default i have goto(ivr|999|1) and i have a record for the extension 999 in the ivr context. I'm ending up with the following err: Channel 'SIP/realtime-6a2c' sent into invalid extension '999' in context 'ivr', but no invalid handler. What am i missing? |
| 21:54.10 | Katty | what's a program to unrar something? |
| 21:54.51 | QuAd|Haudrauf | unrar |
| 21:54.55 | QuAd|Haudrauf | arf :/ |
| 21:55.09 | QuAd|Haudrauf | for windows try winrar for example |
| 21:55.13 | Katty | unrar is not in my apt-cache search. |
| 21:55.17 | thomastim | Katty: http://rarlabs.com/ |
| 21:55.19 | Katty | and i'm lazy. |
| 21:55.24 | thomastim | it's not open source |
| 21:55.26 | QuAd|Haudrauf | then try rar in debian apt-get install |
| 21:55.40 | QuAd|Haudrauf | later on cli, use: rar x myrarfile.rar |
| 21:55.46 | thomastim | Katty: http://rarlabs.com/rar/rarlinux-3.5.1.tar.gz |
| 21:55.47 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 21:56.20 | sshadow | katty: http://www.filzip.com/ |
| 21:56.22 | Katty | i see. |
| 21:56.32 | Katty | tzanger: do you know of any open source unrarers? |
| 21:56.41 | QuAd|Haudrauf | on my gentoo, i found rar and unrar |
| 21:56.50 | thomastim | Katty: no, it's patented or something |
| 21:56.55 | tzanger | Katty: I read that as underwear |
| 21:56.58 | QuAd|Haudrauf | so, since gentoo compiles all those stuff, rar must be opensourced |
| 21:57.01 | tzanger | Katty: but no, there is just rarlabs unrar |
| 21:57.04 | thomastim | Katty: you'll have to d/l it and manually install it as root |
| 21:57.07 | tzanger | but rar is stupid anyway just use gzip/bzip2 |
| 21:57.22 | justinu | rar is for the l337hax0rs |
| 21:57.33 | tzanger | I never understood it |
| 21:57.34 | Katty | tzanger: that's not going to happen. |
| 21:57.39 | Katty | tzanger: it wasn't my file to start with. |
| 21:57.40 | tzanger | let's rar up a zip file with an iso in it |
| 21:57.48 | Katty | tzanger: but those types of comments aren't what i'm looking for anyway |
| 21:57.48 | justinu | exactly |
| 21:58.01 | QuAd|Haudrauf | okay, rar is not opensourced :( |
| 21:58.06 | QuAd|Haudrauf | just looked after it |
| 21:58.13 | QuAd|Haudrauf | stupid RAR license |
| 21:58.33 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:58.38 | Katty | i'll just move it to the windows box and use winrar on it. |
| 21:58.49 | tzanger | Katty: why not just download the linux rar app? |
| 21:58.51 | Katty | gotta burn it anyway |
| 21:58.52 | NDT | Katty: http://www.rarsoft.com/download.htm |
| 21:59.01 | Katty | tzanger: because this machine doesn't have a burner anyway |
| 21:59.04 | Katty | tzanger: and the other one does |
| 21:59.08 | tzanger | hah |
| 21:59.13 | Katty | tzanger: AND the other one also has winrar |
| 21:59.18 | Katty | tzanger: can you say l a z y |
| 21:59.30 | tzanger | Katty: ell ay zed why? |
| 21:59.38 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:59.43 | Katty | NDT: thank you for your input, however the problem has already been addressed (= |
| 22:00.02 | thomastim | Katty i'll just move it to the windows box and use winrar on it. LOL |
| 22:00.10 | thomastim | there's one way to do it |
| 22:00.18 | *** join/#asterisk kpettit_ (n=keith@user-0cet2n7.cable.mindspring.com) |
| 22:00.24 | NDT | hehe |
| 22:00.37 | kpettit_ | is there anyway to change the default timeout for sip registration? |
| 22:01.01 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:01.02 | kpettit_ | maxexpirey dsoent' seem to do the trick |
| 22:01.03 | Katty | tzanger: i think the question is why not. |
| 22:01.11 | Katty | tzanger: that's the whole point of boxes. |
| 22:01.14 | zyke | is it possible to have 2 different IP addresses under host option in sip and iax.conf files? |
| 22:01.51 | DaveCanoe | OK... is this ever so possibly the forum to ask a relatively complicated agi question? |
| 22:02.08 | thomastim | zyke: the 'dynamic' property isn't acceptable? |
| 22:02.16 | Sedorox | twisted[asteria]: delayed.. but I got the same spam again |
| 22:02.45 | DaveCanoe | My question is: after I EXEC a DIAL command, is there a way that I can regain control of the call (to dial a different number) ? |
| 22:03.00 | Sedorox | [17:01] <aNaSTaCia_geBeri> <link removed> Free Porno Videos |
| 22:03.02 | NDT | tzanger: Hehe...when you said lets rar up a zip file with an iso in it...reminded me of something I downloaded once...split with ACE, then zip, then rar'd LOL |
| 22:03.02 | Sedorox | yay for join spam |
| 22:03.12 | Katty | :< |
| 22:03.29 | zyke | thomastim: i have someone who will be sending calls from 2 different boxes and I was wondering if there was a way to create just one entry for his account |
| 22:03.33 | DaveCanoe | I've tried this with 1.0.9 and python as the AGI. Python happily sends the command, but asterisk doesn't appear to see it until after the call completes. |
| 22:03.51 | justinu | postgres errors rock: Reason: ERROR: permission denied for relation cdr |
| 22:04.00 | DaveCanoe | A related question is that when a call is ended for L() reasons, why can't I send it somewhere else. |
| 22:04.10 | justinu | permission denied for relation... ouch |
| 22:04.19 | *** join/#asterisk ManxPower (i=ewieling@128.sub-70-197-201.myvzw.com) |
| 22:04.46 | tzanger | blah the OSS version of Qt/Win is for Mingw |
| 22:04.49 | tzanger | makes sense I guess |
| 22:04.56 | tzanger | I'll use MSVC though I think |
| 22:05.01 | NDT | justinu: Shhhhh....your gonna start a DB debate again lol |
| 22:05.11 | justinu | ming was the emporer in the movie flash gorden |
| 22:05.14 | justinu | gordon |
| 22:05.16 | justinu | he ruled |
| 22:05.29 | rayvd | gordon from sesame street? |
| 22:07.17 | ManxPower | "It's not that I don't care. Well, actually I don't care, but I get paid to fix problems." |
| 22:07.18 | justinu | NDT: everyone here knows postgres is the only true open source RDBMS :P |
| 22:07.21 | thomastim | zyke: as far as the documentation for sip.conf tells me, you can only specify one IP address. However, you can specify host=dynamic also with a defaultip=x.x.x.x option |
| 22:07.36 | NDT | Now back on the subject of corn covers..err..I mean asterisk...ahh nm sidetracked |
| 22:07.39 | NDT | lol |
| 22:08.17 | *** join/#asterisk [}btorch] (n=kvirc@208.63.19.172) |
| 22:08.24 | zyke | thomastim: i think i saw somewhere you can specify an IP range but i'm not sure if that applied to sip.conf |
| 22:08.38 | DaveCanoe | nobody? |
| 22:08.49 | QuAd|Haudrauf | hm, if i were to ask for a nice asterisk webgui and you only had one word to say, which one would it be? :) |
| 22:09.11 | NDT | justinu: You did it...your trying to start stuff in here..get the natives restless... |
| 22:09.15 | NDT | ;) |
| 22:09.34 | justinu | NDT: stirring the shit is always fun |
| 22:09.39 | tzanger | QuAd|Haudrauf: RollYerOwn |
| 22:09.54 | DaveCanoe | (and then solve the halting problem) |
| 22:10.02 | QuAd|Haudrauf | :/ |
| 22:10.29 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
| 22:11.02 | NDT | justinu: We still use postgres...but for what I am doing now using MySQL...What the heck was that database I saw advertised in a linux magazine with a cheetah? cache or something...what the hell is that one? |
| 22:11.48 | justinu | cache has been around for a while |
| 22:12.34 | NDT | I just recalled that one after remembering magazine advertisement from like a year ago |
| 22:12.36 | *** join/#asterisk Daouid (n=daouid@zux221-174-190.adsl.green.ch) |
| 22:12.53 | NDT | if I recall though it was rather pricey |
| 22:12.59 | justinu | yeah, very |
| 22:13.05 | thomastim | zyke: I tried to specify a range, e.g. 10.2.0.0/16, but it immediately turns me over to the unavailable message. my guess is that it can only be the keyword 'dynamic', and actual resolvable host/domain pair, or an IP address |
| 22:15.58 | thomastim | an* |
| 22:16.29 | justinu | thomastim: unfortunately |
| 22:16.45 | *** join/#asterisk santiago (n=santiago@208.195.215.176) |
| 22:16.50 | thomastim | you should have said something before :p |
| 22:17.11 | justinu | i was too busy bitching at you guys for not helping figure out why the damn init script spawns so many processes |
| 22:17.22 | thomastim | it's magic |
| 22:17.25 | M-I-A- | How do I set certain channels on my TDM to not Answer() only have them dial out? |
| 22:17.30 | thomastim | i'm going with the multi-thread explanation |
| 22:17.44 | *** join/#asterisk backblue (n=moo@87-196-43-130.net.novis.pt) |
| 22:17.58 | eKo1 | M-I-A-: change their context to something that has a Hangup |
| 22:18.01 | justinu | thomastim: i think you're right... and I also thing is has something to do with the fact that init is what spawns the processes when I uise the init script |
| 22:18.22 | justinu | ppid is 1 |
| 22:18.28 | thomastim | justinu: init scripts are usually pretty straightforward. what's it say inside? |
| 22:18.36 | justinu | well, it's all this redhat crap |
| 22:18.37 | M-I-A- | eKo1 will that pickup the line then hang it up? |
| 22:18.38 | justinu | to make it pretty |
| 22:18.49 | thomastim | well what's under start: |
| 22:18.55 | eKo1 | M-I-A-: nope |
| 22:19.04 | M-I-A- | eKo1 ok cool thanks!! |
| 22:19.08 | Daouid | hi, anybody could please help me resolving this one: Asterisk + HFC-s in NT mode + mISDN gives me clicks on the line with or without an NTBA (resistor for termination) PLZ Help thx a lot |
| 22:20.33 | *** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 22:20.34 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
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| 22:23.43 | zyke | thomastim: i guess i will need to create 2 entries in the sip.conf for the 2 IPs |
| 22:23.55 | justinu | thomastim: http://pastebin.com/499914 |
| 22:24.50 | justinu | i know enough about shell scripts to see it's not looping or anything, so it must be threads |
| 22:25.17 | trixter | whee looks like I will be getting a donation of the libisup ss7 stack for the sacaug.org :) |
| 22:25.37 | justinu | i still need sip-t |
| 22:28.28 | *** part/#asterisk Ivan_Stepaniuk (n=istepani@201.250.12.15) |
| 22:31.42 | *** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz) |
| 22:32.34 | Psykick | hey guys is there a way for an announcement to be played when someone calls in notifying them that we cannot take their call and after a given timeout places them in a queue? |
| 22:32.39 | *** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
| 22:32.40 | kippi1 | hi |
| 22:33.06 | kippi1 | I have just installed asterisk, when I ring the number I get a busy tone, anyideas? using a TE110p |
| 22:33.28 | Ahrimanes | Psykick: normal dialplan.. call times out, instead of voicemail insert Playback(accouncement) and the Queue() |
| 22:35.36 | kippi1 | think i know why |
| 22:37.40 | justinu | kippi1: if you seriously want people to help you, you'll need to provide a lot more information than that. |
| 22:38.03 | Ahrimanes | hehe, that was way too much work it would seem |
| 22:38.07 | justinu | of course |
| 22:39.36 | eKo1 | Psykick: |
| 22:39.39 | eKo1 | yes |
| 22:42.17 | *** join/#asterisk ToTo (n=ToTo@host14-134.pool872.interbusiness.it) |
| 22:42.19 | Psykick | err .. yah? |
| 22:42.28 | Daouid | anybody knows an answer to my previously posted problem PLZ ?? |
| 22:43.14 | Zodiacal | anyone know what cisco sip phone License's for? do i need them if i use asterisk? |
| 22:43.18 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 22:43.26 | Zodiacal | are for.. even |
| 22:43.55 | Zodiacal | is it a license for using their sip firmware or somthing? |
| 22:43.58 | wunderkin | yes |
| 22:44.01 | Daouid | zodiacal: if u need to dl the firmware yes |
| 22:44.17 | Daouid | zodiacal: it includes access to cco |
| 22:44.17 | Zodiacal | geez they charge for firmware.. |
| 22:44.20 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:44.30 | Daouid | yes |
| 22:44.55 | Zodiacal | its probably better to get them with the phone? or if i get a cisco account then i can download many versions for my phone? |
| 22:45.05 | Zodiacal | or is it like once per download i have to pay? |
| 22:45.13 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:45.16 | *** join/#asterisk Kokey (n=Kokey@dsl-201-129-167-42.prod-infinitum.com.mx) |
| 22:45.20 | Zodiacal | and i can't use one download on many phones right? |
| 22:45.44 | Daouid | zodiacal: not sure but it should be a 1 year contract |
| 22:46.14 | Zodiacal | do all sip phone companies do this? |
| 22:46.19 | Zodiacal | or just cisco |
| 22:46.30 | Daouid | you can install the firmware on any phone, but the license legally authorizes you to use the phone |
| 22:46.37 | Daouid | for which you paid for... |
| 22:46.43 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 22:46.47 | Daouid | no, other companies offer the firmware |
| 22:47.22 | Daouid | i think avaya, budgetone, snom (elmeg) etc... do have free firmwares for sip |
| 22:47.45 | Zodiacal | but will those work on cisco phones? |
| 22:47.50 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:48.05 | justinu | hah no |
| 22:48.11 | Daouid | no, I think I was not clear; the companies issue firmwares for their own phones only |
| 22:48.28 | ManxPower | Zodiacal, Cisco is the only company I'm aware of that makes you pay for the firmware. |
| 22:48.30 | Zodiacal | i had to take a chance and ask |
| 22:48.42 | ManxPower | This is one of the reasons we did not use Cisco. |
| 22:49.00 | ManxPower | Cisco phones also do not come with a power supply. That's the other reason we didn't go with them |
| 22:49.08 | DaveCanoe | cisco makes you pay for firmware for most of their products. It's a company wide thing. |
| 22:49.16 | Zodiacal | is the firmware serialized so it can ditect mutiple phones with the same firmware? |
| 22:49.21 | Zodiacal | just curious |
| 22:49.23 | ManxPower | Zodiacal, no. |
| 22:49.25 | Daouid | but they offer power over ethernet (varying standard following model) |
| 22:49.28 | DaveCanoe | My question is: after I EXEC a DIAL command, is there a way that I can regain control of the call (to dial a different number) ? |
| 22:50.06 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:50.07 | DaveCanoe | cisco doesn't _enforce_ it's licencing. But anyone who would even consider not being up-to-date in their licencing isn't a target cisco client. Rather arrogant, really. |
| 22:50.34 | Daouid | DaveCanoe : yes, I think can use the dundi command |
| 22:50.36 | Zodiacal | wunderkin , daouid, manxpower Thank You! i might not go with cisco then.. |
| 22:50.46 | ManxPower | Zodiacal, we use polycom |
| 22:50.57 | Daouid | Cisco is for fortune 500 :) |
| 22:51.39 | dudes | The school here uses Cisco |
| 22:51.47 | Daouid | I have heard that avaya is very good but it's rather expensive... |
| 22:52.31 | DaveCanoe | "dundi" ... the onlyi dundi command is "dundlookup" ... which doesn't seem to dial at all. |
| 22:53.49 | Daouid | my mistake : is it DISA : http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA |
| 22:54.23 | ManxPower | DaveCanoe, the best I've done is to not dial in an AGI |
| 22:55.28 | DaveCanoe | DISA is for letting an outside caller dial. |
| 22:55.54 | DaveCanoe | manx: OK. so I use my AGI to set variables and then use the asterisk extension language to dial? |
| 22:55.59 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 22:56.10 | DaveCanoe | My last test of that didn't work. |
| 22:56.19 | DaveCanoe | testing again. |
| 22:56.56 | antoni_ | DISA feature (Direct Inwards Services Access). |
| 22:57.16 | Daouid | as I told him... I think it's usable inside |
| 22:57.36 | Daouid | just like any command, you put it in a dialplan |
| 22:57.45 | Daouid | sorry for my english |
| 22:58.24 | Daouid | I hope I remember well enough but I might be wrong :) |
| 22:58.38 | DaveCanoe | Yeah... so I put two back to back dial commands. First one with an L(20000) and second with no limit. the end of the first dial command destroys the call ---- ie: the 2nd dial doesn run. |
| 22:59.12 | Seldon1975 | Hi guys - little help with my dialplan? http://pastebin.com/499984 when the office is closed calls are routed here; I want callers to be able to press 1 to leave a voicemail message, but if they are silent I want to hang up on them. At the moment it seem that when the timeout occurs while listening for the '1' digit, execution goes back to the first priority of this context |
| 22:59.14 | ruud_org | even if you specify the g option? |
| 22:59.18 | Daouid | what do you want to do exactly ? |
| 23:00.05 | DaveCanoe | retested with 'g' option. same failure. |
| 23:01.02 | DaveCanoe | Basically, I want the ability to redirect a call. The user prepays. When their account is low (in the background) I charge their credit card. If this fails, I want to redirect the call to an operator. |
| 23:02.15 | DaveCanoe | ... the problem is that I can't see an outside of asterisk (where the knowledge of credit cards lie) way of redirecting a call once it's in progress. the L() option "almost" works --- it even lets you play an audio file, but in the end it fails because you can't redirect the call after the L() audio plays. |
| 23:03.40 | badboyz | isnt there an option, like M that executes a macro |
| 23:04.20 | badboyz | M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers) |
| 23:04.26 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
| 23:05.07 | DaveCanoe | I want to do things when the call ends, however. |
| 23:05.56 | *** part/#asterisk QuAd|Haudrauf (n=hau@port-212-202-54-134.dynamic.qsc.de) |
| 23:09.37 | Daouid | Anybody got some isdn knowledge so?? |
| 23:09.57 | DaveCanoe | ok... that has a really odd result. |
| 23:11.28 | DaveCanoe | first I execute Dial(iax2/blah/blah/L(20000)gM(tryit)). Macro-tryit executes a Wait(10) followed by another Dial. |
| 23:11.59 | DaveCanoe | If leg one is caller, the first dial generates leg2, the destination. Then the macro's dial generates leg3 attached to leg2 (leg1 dies). |
| 23:12.38 | DaveCanoe | So the M() almost does it, but the problem with M() is that it reverses the roles of everyone before it does it's work. |
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| 23:14.29 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 23:14.42 | jpablo | Hi, anyone can recommend a GSM gateway? |
| 23:15.00 | jpablo | the 2N VoiceBlue looks nice, but i can't find it anywhere |
| 23:19.12 | Daouid | Does the word isdn scare people around here ?? |
| 23:20.03 | Nugget | anyone want to buy an idsl router? :) |
| 23:20.15 | iDunno | Daouid: damned well terrifies me :) |
| 23:20.23 | justinu | how many g729 transcodes do people think a xeon 3.0 can handle? |
| 23:20.27 | Daouid | lol |
| 23:20.39 | Daouid | PLZ ;) |
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