| 00:01.33 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 00:05.20 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 00:11.29 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au) |
| 00:11.45 | Hyphenex | Anybody know if/where I can find a free version of chan_skype? |
| 00:16.02 | *** join/#asterisk SirThomas_Home (n=SirThoma@209.169.199.174) |
| 00:17.09 | SirThomas_Home | anyone here crazy/interesting enough to run Cisco phones using chan_sccp? I'm looking for firmware files. :-/ |
| 00:17.40 | Strom_C | SirThomas_Home: if you need firmware for your phones, contact your reseller |
| 00:17.51 | SirThomas_Home | yeah... that's what I thought. |
| 00:21.06 | hesco | I'm getting very close to having hylafax<-->iaxmodem<-->asterisk all working together. |
| 00:22.12 | hesco | Can someone who has done this before please advise with a sample dialplan for handling incoming and outgoing faxes? |
| 00:24.04 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 00:24.55 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:24.58 | DrAk0 | hesco, which distribution and hylafax version are you using? |
| 00:27.30 | hesco | I'm on debian, with hylafax from the debian distro, 4.3.1-7 |
| 00:28.57 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 00:31.41 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 00:40.20 | Katty | so much sleepy |
| 00:54.15 | _ShrikE | yawns |
| 01:10.06 | Olobola | is CHANNEL STATUS deprecated? |
| 01:18.37 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.214) |
| 01:19.00 | Olobola | does anyone know how I can tell the 'status' of a SIP user through realtime? I would like to know if a user is logged in, in a call etc. |
| 01:19.20 | BeeBuu | AMI |
| 01:23.56 | *** join/#asterisk SteveTotaro (n=root@96.234.217.26) |
| 01:24.03 | *** join/#asterisk EvilDeshi (n=Skunk@75-135-93-93.dhcp.mdsn.wi.charter.com) |
| 01:26.34 | SteveTotaro | hi everyone! |
| 01:28.54 | _ShrikE | hello SteveTotaro |
| 01:29.14 | SteveTotaro | slow night :) |
| 01:29.42 | *** join/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 01:29.42 | _ShrikE | very... been kinda slow all day. |
| 01:30.14 | *** part/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 01:30.18 | SteveTotaro | i have also noticed the users list has slowed significantly |
| 01:30.29 | SteveTotaro | not just today but in general |
| 01:30.47 | SteveTotaro | maybe because of better docs |
| 01:31.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:31.27 | _ShrikE | I have also noticed that there are more helpful folks in the channel lately as well |
| 01:32.47 | SteveTotaro | well if you have a minute maybe you can help me out |
| 01:33.06 | SteveTotaro | i bought three URIs http://www.dmkeng.com/Products.htm |
| 01:33.41 | Katty | i'm distracted by tv |
| 01:33.42 | SteveTotaro | now i am told that i must install asterisk from an EVB ISO |
| 01:34.17 | *** join/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 01:34.18 | SteveTotaro | they have a special version of app_rpt but i cannot just do an svn co |
| 01:34.30 | _ShrikE | that kinda sucks |
| 01:34.42 | SteveTotaro | yeah, and makes no sense |
| 01:35.19 | SteveTotaro | i am sure there must be a way around it but i am not familiar enough with svn i guess |
| 01:36.57 | SteveTotaro | wget -N http://xelatec.com/asterisk/svn_rpt_update this is the "update script" that "does not touch anything but app_rpt" |
| 01:39.48 | SteveTotaro | http://www.xelatec.com/asterisk/svn_rpt_update_a |
| 01:39.51 | *** join/#asterisk r0d3nt (i=nobody@pinky.ratman.org) |
| 01:39.59 | _ShrikE | Looking at that script now |
| 01:40.16 | rkeene | SteveTotaro, Why type of radio does this interface with ? |
| 01:40.37 | SteveTotaro | why? or what? |
| 01:40.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:40.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:41.00 | SteveTotaro | I am connecting them to ic 4000 repeaters |
| 01:41.04 | rkeene | The URIs mentioned above... the FAQ is amusing :-P |
| 01:41.33 | SteveTotaro | it is awesome tech |
| 01:42.25 | SteveTotaro | on two meter you can give phone access at a great distance |
| 01:46.50 | [gnubie] | hello all.. |
| 01:47.20 | [gnubie] | i installed a sip hard phone inside my home lan |
| 01:47.43 | _ShrikE | SteveTotaro: this will checkout the asterisk parts |
| 01:47.47 | _ShrikE | svn checkout http://xelatec.com/svn/app_rpt/asterisk/trunk /usr/src/svnwork/asterisk --username aguest --password riverside |
| 01:48.17 | *** join/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 01:48.29 | *** part/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 01:48.37 | _ShrikE | Thsi should checkout the zaptel parts |
| 01:48.39 | _ShrikE | svn checkout http://xelatec.com/svn/app_rpt/zaptel/trunk /usr/src/svnwork/zaptel --username aguest --password riverside |
| 01:48.55 | [gnubie] | calling from a pots network to extension 102 (the sip hard phone), both phones can hear each other but calling from 102 to a phone at the pots network, the callee cannot hear anything from 102 |
| 01:50.22 | [gnubie] | any idea why am i getting this problem? |
| 01:50.28 | *** join/#asterisk implicit (n=implicit@200.12.227.181) |
| 01:53.24 | russellb | _ShrikE: is the stuff in asterisk and zaptel itself not up to date? |
| 01:53.26 | russellb | and if not, why not? |
| 01:53.33 | Katty | oh |
| 01:53.39 | Katty | its russellb |
| 01:53.48 | russellb | Katty: hi2u! |
| 01:53.54 | Katty | (= |
| 01:54.11 | _ShrikE | russellb: the URI adapter SteveTotaro is using requires a special app_prt and some other things. |
| 01:54.23 | _ShrikE | errr.. app_rpt |
| 01:54.34 | russellb | oic.. |
| 01:54.56 | _ShrikE | not sure why |
| 01:56.01 | russellb | k! |
| 01:56.43 | Hyphenex | I'm trying to set up chan_skype, but I can't select /dev/dsp (does not appear in skype) even after having ran makedsp |
| 01:57.24 | russellb | that module is gross |
| 01:57.53 | russellb | and they removed copyright notices in the asterisk part that gave credit to the people who wrote the code they copied and based theirs on :( |
| 01:58.05 | _ShrikE | boooo |
| 01:58.14 | russellb | boo, indeed |
| 01:58.45 | russellb | in every community, there are good and bad citizens. |
| 01:58.51 | russellb | they are examples of the latter. |
| 01:59.10 | Katty | are you a good witch or a bad witch? |
| 01:59.21 | russellb | i don't think i'm a witch at all |
| 01:59.30 | file | moo |
| 01:59.40 | Katty | what about your dog? |
| 01:59.43 | russellb | wibble wobbles |
| 02:00.03 | file | swizzle swozzles |
| 02:00.06 | Hyphenex | russellb: What do you suggest instead of chan_skype then? |
| 02:00.12 | russellb | shrugs |
| 02:00.20 | LiNeTuX | Anyone using certificates (x509) to authenticate sip clients w/asterisk? |
| 02:00.21 | russellb | not using skype? :) |
| 02:00.47 | russellb | if it works for you, then fine ... do what you have to do |
| 02:00.54 | russellb | i just felt like speaking out a bit :) |
| 02:01.06 | russellb | LiNeTuX: Asterisk does not support that |
| 02:01.08 | Hyphenex | russellb: it doesn't work for me, that's the problem :P |
| 02:01.18 | russellb | Hyphenex: like i said, the way it works is pretty gross. |
| 02:01.23 | russellb | "works" |
| 02:01.29 | Katty | hmm. |
| 02:01.33 | LiNeTuX | russellb: I've read spatterings on voip-info.org - but can't find anything concrete. |
| 02:01.46 | LiNeTuX | Not many phone support certs either (yet) |
| 02:02.22 | russellb | Asterisk 1.6 has experimental TLS support for the signalling ... |
| 02:02.23 | Katty | yawns |
| 02:02.31 | Katty | russellb: and tapi? |
| 02:02.32 | CCFL_Man2 | Strom_C: i got a WE 51AL candle stick on ebay |
| 02:02.34 | russellb | we're working on stabilizing that ... and SRTP coming soon ... |
| 02:02.37 | russellb | Katty: no.. |
| 02:02.40 | Hyphenex | russellb: is there any other thing you know that *might* work then? |
| 02:02.40 | Katty | russellb: :< |
| 02:02.49 | Katty | russellb: give me tapi support. |
| 02:02.49 | russellb | Hyphenex: not for skype ... |
| 02:02.52 | Katty | russellb: i give you a cookie |
| 02:02.54 | russellb | Katty: mmmmmmmm nothx |
| 02:02.55 | LiNeTuX | russellb: yeah, but that's for communication... I just care about auth. right now |
| 02:03.01 | Katty | russellb: kay |
| 02:03.05 | russellb | LiNeTuX: right .. |
| 02:03.12 | Katty | russellb: make Qwell do it |
| 02:03.24 | Katty | russellb: i give you a cookie |
| 02:03.33 | russellb | heh |
| 02:03.37 | russellb | i'll put it on ... "the list" |
| 02:03.43 | LiNeTuX | query: without a VPN, what's the best way to secure SIP client authentication? |
| 02:03.52 | Katty | russellb: :< |
| 02:04.06 | CCFL_Man2 | LiNeTuX: i've always wondered that myself |
| 02:04.08 | Katty | russellb: i won't live long enough to be on a waiting list. |
| 02:04.23 | Katty | russellb: i'll put M&Ms in cookie too :> |
| 02:04.24 | russellb | that's quite sad. |
| 02:04.33 | LiNeTuX | CCFL_Man2: I mean, you can always put in crazy passwords in the auth, you can limit certain ext's to certain IP's... |
| 02:04.44 | LiNeTuX | but that's not really good security. |
| 02:04.56 | russellb | get a VPN? |
| 02:04.57 | CCFL_Man2 | LiNeTuX: even using voip on public wifi |
| 02:05.14 | LiNeTuX | CCFL_Man2: Well, that's just asking for trouble on top of crazy :) |
| 02:05.22 | russellb | authentication is silly anyway ... we should get rid of it |
| 02:05.32 | LiNeTuX | heh |
| 02:05.32 | Katty | russellb: butbutbut |
| 02:05.33 | Katty | russellb: but |
| 02:05.50 | Katty | russellb: but :< |
| 02:06.04 | CCFL_Man2 | LiNeTuX: i know, but the only solution is a vpn |
| 02:06.20 | LiNeTuX | CCFL_Man2: Until zphone support for Asterisk comes out |
| 02:06.23 | SteveTotaro | www.vosky.com |
| 02:06.24 | LiNeTuX | zphone rocks |
| 02:06.53 | Katty | i've been using zoiper of late. |
| 02:07.04 | CCFL_Man2 | LiNeTuX: why not do sip over an ssl tunnel? |
| 02:07.21 | LiNeTuX | CCFL_Man2: Because I don't have control over the end-points |
| 02:07.32 | CCFL_Man2 | or atleast, authentication over an ssl tunnel |
| 02:07.37 | russellb | yeah, zphone is cool ... but AFAIK, focuses on end to end security |
| 02:07.44 | russellb | there was a great quote on our -dev list in regards to that ... |
| 02:07.57 | russellb | supporting it is going to be tough, because "Asterisk is designed as a man in the middle attack" |
| 02:08.11 | LiNeTuX | I'd *love* to auth over a tunnel, but that brings me back to using x509 certs |
| 02:08.18 | LiNeTuX | heh |
| 02:08.27 | russellb | that's what PBX is, really, heh |
| 02:08.33 | LiNeTuX | true |
| 02:09.07 | LiNeTuX | well, Asterisk can still be MiM, but the transmission will just be noise |
| 02:09.17 | russellb | heh |
| 02:09.18 | CCFL_Man2 | russellb: you have any western electric candlestick phones? |
| 02:09.21 | SteveTotaro | _ShrikE: thanks for the URI svn string |
| 02:09.24 | russellb | CCFL_Man2: don't think so |
| 02:09.46 | CCFL_Man2 | russellb: you don't collect western electric phones? |
| 02:09.50 | russellb | why, want to give me one? :-p |
| 02:09.56 | russellb | can't say i do ... |
| 02:10.04 | russellb | i hate phones, actually |
| 02:10.04 | CCFL_Man2 | ahh |
| 02:10.08 | CCFL_Man2 | lol |
| 02:10.18 | denon | collects western UNION money orders .. |
| 02:10.23 | Katty | grins |
| 02:10.23 | denon | feel free to send me one |
| 02:10.29 | Katty | pats denon |
| 02:10.46 | CCFL_Man2 | hah |
| 02:10.53 | denon | russellb: that's weird, isnt it? most of us who love routing voice traffic actually hate phones |
| 02:11.01 | russellb | yup. |
| 02:11.08 | denon | I like calling people I want to talk to .. |
| 02:11.08 | russellb | it's extremely fun technology to work on |
| 02:11.11 | denon | but I hate ringing phones |
| 02:11.14 | russellb | lots of interesting problems to solve ... |
| 02:11.19 | denon | usually means someone's wanting me to do something |
| 02:11.21 | russellb | but damnit, that ringing phone better not be for me ... |
| 02:11.28 | denon | it always is |
| 02:11.54 | russellb | i usually ignore my office phone when it rings :-p |
| 02:12.01 | russellb | "meh, they can email me ..." |
| 02:12.25 | CCFL_Man2 | calls russellb |
| 02:12.36 | russellb | you don't know my number! |
| 02:12.46 | CCFL_Man2 | i don't |
| 02:12.50 | russellb | it's 7 |
| 02:12.53 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 02:12.56 | denon | russellb: yeah we do, it's in digium dial by name |
| 02:12.58 | rkeene | 1-900-RUSS-ELL ? |
| 02:13.11 | russellb | rkeene: how'd you guess? |
| 02:13.19 | russellb | denon: like i said, i don't answer that one :) |
| 02:13.21 | rkeene | I looked it up online. |
| 02:13.24 | denon | hehe |
| 02:13.38 | CCFL_Man2 | dials sip:7@russellb.tk |
| 02:13.56 | denon | bbl |
| 02:14.28 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 02:15.35 | rkeene | 0x24B88A172 is my phone number :-P |
| 02:15.39 | russellb | ooh |
| 02:16.37 | SteveTotaro | i got a sweet one 1.888.777.1888 |
| 02:17.06 | SteveTotaro | goes straight to comedian mail, no damn ringing |
| 02:17.12 | Olobola | how I can tell the 'status' of a SIP user through realtime? I would like to know if a user is logged in, in a call etc. |
| 02:19.03 | russellb | Olobola: it doesn't really work with realtime :( |
| 02:19.08 | russellb | unless you enable realtime caching ... |
| 02:19.17 | russellb | in which case its like ... pseudo-realtime... or something |
| 02:19.29 | SteveTotaro | realtime is funky |
| 02:19.36 | russellb | yes, it is |
| 02:19.47 | Olobola | that sucks |
| 02:19.49 | russellb | it was a nice hack ... |
| 02:19.55 | russellb | but now we need to go back and do it for real :) |
| 02:20.05 | SteveTotaro | glad to hear it |
| 02:20.21 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.214) |
| 02:20.29 | russellb | so yeah, it's sillyness is well understood at this point |
| 02:20.42 | russellb | s/it's/its/ |
| 02:20.51 | hesco | I'm getting very close to having hylafax<-->iaxmodem<-->asterisk all working together. |
| 02:20.55 | hesco | Can someone who has done this before please advise with a sample dialplan for handling incoming and outgoing faxes? |
| 02:21.04 | CCFL_Man2 | do i want sidetone service or anti sidetone service? |
| 02:21.35 | SteveTotaro | i can give you a link of a guy who reverse engineered what i did and claimed it as his own for hylafax |
| 02:22.34 | SteveTotaro | one second, let me find it |
| 02:22.58 | Olobola | what is up with CHANNEL STATUS? I can't seem to run it and I see no mention that it's deprecated. |
| 02:23.16 | russellb | i've never heard of "CHANNEL STATUS" |
| 02:24.05 | Olobola | http://www.voip-info.org/wiki/view/channel+status |
| 02:24.39 | SteveTotaro | http://blog.evaristesys.com/ |
| 02:24.56 | SteveTotaro | you can replace every I for SteveTotaro |
| 02:25.04 | SteveTotaro | "I" |
| 02:25.46 | hesco | thanks. I'll take a look at that, then. |
| 02:29.19 | hesco | SteveTotaro: Thanks. This looks like it just may be what I'm looking for. Also looks like it will take a while to digest fully. Thank you so much for the lead! |
| 02:29.46 | SteveTotaro | alex really documented the heck out of my work |
| 02:30.04 | SteveTotaro | but i suck at documentation, just would have liked a little credit ;) |
| 02:37.05 | jblack | how could I not have a single typo in a 3 page document? |
| 02:37.12 | jblack | My spellchecker must be broken. |
| 02:38.23 | LiNeTuX | +92 font? |
| 02:38.33 | jblack | Nice try! Nope. :) |
| 02:43.20 | delparnel | anyone here use a Polycom 650? |
| 02:48.26 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:48.55 | obnauticus | can anyone here explain this: http://pastebin.ca/952377 ? |
| 02:50.16 | SteveTotaro | did you try a new dongle |
| 02:50.17 | SteveTotaro | ; |
| 02:50.22 | SteveTotaro | ? |
| 02:50.54 | obnauticus | No. |
| 02:53.16 | SteveTotaro | maybe you are getting rf interference |
| 02:53.22 | SteveTotaro | or the dongle is bad |
| 02:56.40 | obnauticus | well it happens all the time. |
| 02:56.51 | obnauticus | and it's been happening since i've gotten it |
| 02:57.05 | obnauticus | and it can read data off the phone, and i've googled and seen other people with problems with my preticumar phone |
| 02:57.22 | obnauticus | and the error repeats in the same exact spot almost every time... usually |
| 02:57.36 | obnauticus | it connected one time... which i haven't seen before in the 4 months i've owned this dongle. |
| 02:58.02 | obnauticus | It always disconnects at AT+CIND=? though. |
| 03:14.59 | *** part/#asterisk infinity3 (i=brendon@saleen.netcal.com) |
| 03:16.58 | SteveTotaro | my razr v3 works perfectly |
| 03:19.58 | obnauticus | well SteveTotaro i have a question |
| 03:20.20 | SteveTotaro | fire away |
| 03:20.52 | obnauticus | I search for a device with my phone, and i pair with it with the set pin 00000, it works, then afterwards every time my asterisk box tries to connect to my phone, it sends a pair request... I enter the same pin and it fails. |
| 03:21.54 | SteveTotaro | not sure |
| 03:22.15 | drmessano | I have the same problem |
| 03:22.26 | SteveTotaro | what phone? |
| 03:22.33 | obnauticus | SPH-A640 |
| 03:22.39 | obnauticus | Samsung. |
| 03:22.50 | drmessano | SPH-A940 |
| 03:23.00 | obnauticus | oo |
| 03:23.04 | obnauticus | im starting to see a pattern. |
| 03:23.20 | drmessano | and also |
| 03:23.27 | drmessano | When you do get them paired |
| 03:23.39 | SteveTotaro | as i understand it, not all phones work well, but i can confirm the razr v3 is solid |
| 03:23.46 | drmessano | The samsung earpiece and mic do not work when Asterisk takes the call |
| 03:25.58 | *** join/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 03:26.04 | [gnubie] | i installed a sip hard phone inside my home lan.. calling from a pots network to extension 102 (the sip hard phone), both phones can hear each other but calling from 102 to a phone at the pots network, the callee cannot hear anything from 102. any idea why am i getting this problem? |
| 03:26.55 | TJNII | Where is the server in relation to the phone? |
| 03:27.24 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.22.37) |
| 03:28.09 | _ShrikE | nite ladies and gents |
| 03:29.08 | [gnubie] | TJNII: the asterisk box is the main gateway also of the extension 102 sip hard phone |
| 03:29.34 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:30.28 | *** part/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au) |
| 03:30.31 | obnauticus | drmessano, SteveTotaro: http://pastebin.ca/952408 |
| 03:30.50 | TJNII | [gnubie]: Have you tried an echo test? Is that OK? |
| 03:31.07 | [gnubie] | TJNII: no problem with it |
| 03:31.27 | TJNII | How are you connecting to the PSTN? |
| 03:32.59 | [gnubie] | the main problem is that when my sip hard phone located inside my home lan and the asterisk is my home pbx/gateway/router calls a pots phone through Zap/4, the callee cannot hear sip hard phone's voice but the caller hears the pots telephone's voice |
| 03:33.11 | SteveTotaro | obnauticus: that is over my head, sorry |
| 03:33.17 | obnauticus | Nuts |
| 03:33.36 | drmessano | Im not reading debug, screw that |
| 03:33.42 | drmessano | I closed it as fast as I opened it |
| 03:33.46 | [gnubie] | but if the caller is the pots telephone and the callee is the sip hard phone, both of them can hear its voice |
| 03:33.49 | obnauticus | lol |
| 03:33.49 | drmessano | Its too weekend for that |
| 03:33.56 | TJNII | [gnubie]: Gow are you connected to the PSTN? |
| 03:33.59 | obnauticus | it's friday |
| 03:34.05 | TJNII | <PROTECTED> |
| 03:34.05 | drmessano | week....end |
| 03:34.37 | [gnubie] | TJNII: Zap/4 pots line |
| 03:34.43 | obnauticus | week...end = sunday |
| 03:34.53 | obnauticus | businessweek...end |
| 03:35.00 | SteveTotaro | world will end in 2012 there is no such thing as weekends |
| 03:35.11 | SteveTotaro | even if is Easter |
| 03:35.38 | obnauticus | HEY |
| 03:35.43 | obnauticus | work harder not smarter. |
| 03:36.04 | SteveTotaro | work harder and smarter |
| 03:36.10 | Nugget | slack. |
| 03:36.12 | obnauticus | slack. |
| 03:36.14 | TJNII | [gnubie]: So incoming calls are fine, and clls from the sip phone to an echo test are fine? |
| 03:36.16 | SteveTotaro | and get a heart attack |
| 03:36.19 | *** join/#asterisk xfatkidx (n=chris@d149-67-218-192.col.wideopenwest.com) |
| 03:36.23 | xfatkidx | hey all |
| 03:36.31 | SteveTotaro | hello |
| 03:36.32 | [gnubie] | TJNII: yes |
| 03:36.41 | obnauticus | i bet on that day the world ends (some time in december, 2012) there will just be some naked dude running around with no pants rickrolling people. |
| 03:36.45 | TJNII | Weird |
| 03:37.23 | xfatkidx | hey; ive been looking for a good support forum for asterisk. im home for a couple weeks from surgery and thought id try something new...but im having a major error with my install |
| 03:37.25 | SteveTotaro | rickrolling? maybe i can learn a new word |
| 03:37.42 | [gnubie] | TJNII: that's why.. asterisk cli at verbose 3 doesn't give me hint for this problem |
| 03:37.43 | SteveTotaro | ~book |
| 03:37.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:37.55 | [TK]D-Fender | ~rickroll |
| 03:37.56 | jbot | hmm... rickroll is http://www.internetisseriousbusiness.com, or http://www.xkcd.com/396/ |
| 03:38.19 | xfatkidx | anyone a guru with the asteriskNOW install? Im getting a problem when the kern is mounting |
| 03:38.41 | TJNII | [gnubie]: Sip phone and the * server are on the same subnet, correct? No nat? |
| 03:38.42 | SteveTotaro | damn you, i have a core2duo and flash doesn't work |
| 03:38.52 | [gnubie] | TJNII: yes, no nat |
| 03:39.31 | TJNII | [gnubie]: Then I don't know. It's probably not a sip problem as the echo test works. I don't know about Zaptel stuff, so I can't help you debug that. |
| 03:39.44 | TJNII | [gnubie]: Odd that incoming calls work, though.... |
| 03:40.07 | xfatkidx | i get the error udevstart exited abnormally with value 0! and then kernel panic : not sync |
| 03:40.17 | [gnubie] | TJNII: thanks for trying to help. :) |
| 03:40.39 | TJNII | xfatkidx: You using freePBX, AstriskNow....? |
| 03:40.52 | xfatkidx | yeah |
| 03:41.04 | SteveTotaro | ~freepbx |
| 03:41.05 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:41.16 | xfatkidx | ahhhh, thanks so much |
| 03:41.23 | TJNII | Then you need to go to #freepbx or #astriskNOW. You're not having an astrisk problem (yet). |
| 03:41.45 | [gnubie] | this is the asterisk cli output i got when my sip hard phone calls a mobile phone through the Zap/4 pots trunk => http://www.privatepaste.com/e2VYHi3UMd |
| 03:41.56 | SteveTotaro | so is druid opensource good or not |
| 03:42.48 | SteveTotaro | i saw that all the files in /etc/asterisk are chmod 777 which is scary |
| 03:44.14 | SteveTotaro | and when is the 3com/digium product going to hit the market? |
| 03:45.27 | SteveTotaro | i don't really like fonality |
| 03:46.14 | [TK]D-Fender | SteveTotaro, What are you looking for in a system? |
| 03:46.31 | SteveTotaro | the best free gui |
| 03:46.42 | obnauticus | facepalm. |
| 03:47.05 | SteveTotaro | i have a customer with three locations and he is set on fonality |
| 03:47.42 | SteveTotaro | damn google |
| 03:48.27 | SteveTotaro | i would like to see the 3Com/Asterisk system |
| 03:48.41 | [gnubie] | kindly check this site => http://www.privatepaste.com/download/25SN1U1FvU |
| 03:49.27 | [gnubie] | that contains the tcpdump output of my sip hard phone calling a mobile phone through the Zap/4 pots trunk of the asterisk box i have at home.. |
| 03:49.28 | SteveTotaro | i am a 3com certified network telephony engineer |
| 03:50.12 | SteveTotaro | 3com has alot of backing in it's PBX products and they work great |
| 03:50.48 | Olobola | how can I tell if a sip client is logged in or in a call etc.. ? |
| 03:50.49 | rkeene | Their network switches suck :-P |
| 03:51.09 | rkeene | Olobola, sip show peers and sip show channels |
| 03:51.11 | SteveTotaro | i agree |
| 03:51.22 | SteveTotaro | i am a ccna too |
| 03:51.27 | obnauticus | me three |
| 03:51.59 | rkeene | Same here |
| 03:52.17 | Olobola | rkeene: can I query a specific client through agi? |
| 03:52.19 | SteveTotaro | i have an awesome opportunity to have a company pay for ccie but i have to do it on my own time |
| 03:52.52 | [TK]D-Fender | Olobola, AMI <- |
| 03:53.02 | rkeene | Olobola, No idea |
| 03:53.30 | [gnubie] | anyone have an idea on how to solve my problem? or, have you encountered this kind of problem? |
| 03:53.34 | Olobola | thanks, thanks |
| 03:54.48 | DoDaT69 | I was trying to install a PRI today with a sangoma a102d, but for some reason the d channel wouldnt sync.. |
| 03:55.24 | SteveTotaro | pri intense debug span X |
| 03:55.26 | DoDaT69 | I only saw 2 options in the zapata config file for D channel signalling, tried both |
| 03:55.47 | SteveTotaro | where are you? |
| 03:55.52 | DoDaT69 | atlanta |
| 03:55.58 | DoDaT69 | switch was NI1 |
| 03:56.06 | DoDaT69 | I was using pri_cpe |
| 03:56.12 | SteveTotaro | national? |
| 03:56.14 | DoDaT69 | and I did a dchan=24 |
| 03:56.14 | DoDaT69 | yea |
| 03:56.25 | DoDaT69 | I also tried the hdlc=24 |
| 03:56.36 | [TK]D-Fender | DoDaT69, are you in a postition to work on this NOW? |
| 03:56.42 | SteveTotaro | do a pri intense debug |
| 03:56.46 | DoDaT69 | I have access to the box.. |
| 03:57.11 | DoDaT69 | k |
| 03:57.12 | DoDaT69 | 1 sec |
| 03:57.21 | [TK]D-Fender | DoDaT69, pastebin your wanpipe1.conf, zaptel.conf, zapata.conf, "cat /proc/interrupts", and "wanrouter status" |
| 03:57.24 | [TK]D-Fender | ~pb |
| 03:57.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:57.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 03:57.31 | drmessano | I just farted |
| 03:57.39 | DoDaT69 | holds his nose |
| 03:58.09 | SteveTotaro | nice, silent but deadly, wet, loud, details please, farts are always funny |
| 03:59.27 | SteveTotaro | the director of IT used to come in my office, not say a word and walk out |
| 03:59.34 | drmessano | HA |
| 03:59.45 | SteveTotaro | a few seconds later the stank hit |
| 03:59.54 | obnauticus | can I use application playback to playback a wav? |
| 04:00.02 | drmessano | IT farts are funny.. like.. "HA HA, I just rebooted your DB server without telling you" funny |
| 04:00.25 | DoDaT69 | http://www.pastebin.ca/952423 |
| 04:00.29 | SteveTotaro | that was at shire pharm, the makers of adderall and many other drugs |
| 04:00.31 | DoDaT69 | I actually have 2 pri's |
| 04:00.32 | DoDaT69 | the a102d |
| 04:00.53 | DoDaT69 | I was stumped-- I even unplugged their equipment and went straigth to the smart jack |
| 04:01.29 | DoDaT69 | we are going to give it another try with the provider on monday-- tried to call sangoma today, it was a canadian holiday ;) |
| 04:01.35 | rkeene | Maybe the second PRI's D-channel is for controlling both PRIs ? |
| 04:01.48 | DoDaT69 | I asked-- they said 24 and 48 |
| 04:01.59 | DoDaT69 | so its actually 2 straight pri's |
| 04:02.02 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:02.05 | DoDaT69 | I told em I watned to switch to NI2 also |
| 04:02.13 | DoDaT69 | that way can control the CID |
| 04:02.50 | DoDaT69 | I set one PRI with Dchan signalling |
| 04:02.56 | SteveTotaro | the best is controlling ani |
| 04:03.00 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 04:03.18 | DoDaT69 | the other with the hdlc -- that was the best I could do to rule out problem on my end |
| 04:03.39 | SteveTotaro | if you are using NFAS then you can have one d chan for multiple PRIS |
| 04:03.49 | [TK]D-Fender | DoDaT69, TDMV_DCHAN = 24 - change to - TDMV_DCHAN = 0 |
| 04:04.06 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 04:04.16 | DoDaT69 | I had changed that back and forth while troubleshooting, both settings were same-- |
| 04:04.20 | DoDaT69 | I will cahnge back |
| 04:04.21 | [TK]D-Fender | DoDaT69, for EACH |
| 04:04.52 | [TK]D-Fender | DoDaT69, What are you plugging into this card? |
| 04:05.04 | DoDaT69 | pri |
| 04:05.10 | SteveTotaro | always check your cables |
| 04:05.16 | DoDaT69 | yea- those are good |
| 04:05.16 | obnauticus | with ffmpeg how do i convert a pcm to a gsm |
| 04:05.24 | [TK]D-Fender | DoDaT69, PRI direct from telco? |
| 04:05.25 | SteveTotaro | sox |
| 04:05.25 | DoDaT69 | I used their cables they had plugged into their stuff |
| 04:05.27 | DoDaT69 | yes |
| 04:05.32 | [TK]D-Fender | DoDaT69, then span=1,0,0,esf,b8zs should be span=1,1,0,esf,b8zs |
| 04:05.38 | [TK]D-Fender | DoDaT69, then span=2,0,0,esf,b8zs should be span=2,2,0,esf,b8zs |
| 04:05.53 | [TK]D-Fender | DoDaT69, hardhdlc=48 <- nope |
| 04:06.01 | [TK]D-Fender | DoDaT69, dchan=48 <- yup |
| 04:06.27 | DoDaT69 | Okay- the hdlc was generated by the wanpipe "wizard" |
| 04:06.45 | DoDaT69 | actually, whole file was, I manually added the dchan in there.. |
| 04:06.47 | SteveTotaro | wancfg |
| 04:06.52 | DoDaT69 | yea- |
| 04:07.01 | DoDaT69 | never had any issues with it before really- |
| 04:07.15 | DoDaT69 | what is the span components designate? |
| 04:07.22 | [TK]D-Fender | DoDaT69, Ok, so change as advised. Stop * & wanrouter. Restart in order, and test it out. |
| 04:07.45 | SteveTotaro | yeah, the only problem i have had using ./Setup install is with ami |
| 04:07.57 | [TK]D-Fender | DoDaT69, "span=" defines the T1 timing (probably a big part of your problem), LBO, etc. |
| 04:08.15 | DoDaT69 | pri intense debug is not valid command |
| 04:08.24 | SteveTotaro | not sure lbo has much effect on anything |
| 04:08.24 | DoDaT69 | yea-- thats dead on |
| 04:08.40 | [TK]D-Fender | DoDaT69, "span=[port],[timingsource],[lbo],[framing],[coding] |
| 04:08.43 | DoDaT69 | the carrier was there, however the dchannel wouldnt sync, so couldnt go to b channel |
| 04:08.48 | [TK]D-Fender | SteveTotaro, not in his case |
| 04:08.48 | DoDaT69 | ahhh |
| 04:09.03 | DoDaT69 | so what does the 0 or 1 designate in the timing source? |
| 04:09.09 | DoDaT69 | master/slave? |
| 04:09.19 | rkeene | 0 means you are the master source (should be used with "cpe_net") |
| 04:09.22 | [TK]D-Fender | DoDaT69, Your zapata told * IT was to provide timing, which usually ends up desync-ing with the other side |
| 04:09.26 | DoDaT69 | ah- shit |
| 04:09.36 | DoDaT69 | yea, that was my problem then.. I bet it will work now.. |
| 04:09.39 | rkeene | 1 means it's the primary master source, 2 means secondary, 3 tertiary |
| 04:09.40 | [TK]D-Fender | DoDaT69, 0=act as timer, 1=use this as primary, 2=secondary, etc |
| 04:10.02 | DoDaT69 | Hmm |
| 04:10.08 | SteveTotaro | cpe_net is good for channel banks or connecting legacy systems |
| 04:10.31 | DoDaT69 | cpe_net? |
| 04:10.38 | DoDaT69 | for signalling? |
| 04:10.46 | SteveTotaro | not in your case |
| 04:10.47 | rkeene | I'm using my PRI card as a proxy right now :-P PRI goes in 1 port, and PRI goes out to my modem bank |
| 04:11.15 | rkeene | DoDaT69, That's if you are using "0" for the timing source correctly, you will also want to use "cpe_net" (probably) -- you aren't |
| 04:11.21 | DoDaT69 | I was getting RAI alarm on one of em |
| 04:11.32 | DoDaT69 | yea, pri_cpe |
| 04:11.42 | DoDaT69 | thats the one that just made sense to me.. |
| 04:11.46 | DoDaT69 | is that correct tho? |
| 04:11.53 | SteveTotaro | yes |
| 04:12.05 | SteveTotaro | you are the cpe if taking from the telco |
| 04:12.20 | DoDaT69 | yea-- exactly what I was thinking there. |
| 04:12.28 | rkeene | Err, yeah, "pri_net" not "cpe_net" :-( (Neither for you, "pri_cpe") |
| 04:12.29 | SteveTotaro | but i had the reverse once on a FUBAR DMS100 |
| 04:12.44 | obnauticus | drmessano |
| 04:12.46 | obnauticus | what to sing with me? |
| 04:13.09 | DoDaT69 | (rkeene): I knew what you meant ;) |
| 04:13.36 | SteveTotaro | with usad (qwest) on a dms100 i had to set my box as the pri_net |
| 04:14.10 | DoDaT69 | should I enable tdm hw dtmf? |
| 04:14.30 | SteveTotaro | you should get your channels up first |
| 04:14.30 | *** part/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 04:14.40 | DoDaT69 | Okay- so only if having dtmf issues? |
| 04:14.54 | SteveTotaro | don't fix it if it not broken |
| 04:15.02 | DoDaT69 | you aint lyin there! |
| 04:15.04 | DoDaT69 | Mar 22 00:10:52 pbx kernel: wanpipe2: RAI alarm is ON |
| 04:15.20 | DoDaT69 | is it normal for that to go on, then off during the init sequence? |
| 04:16.18 | SteveTotaro | http://www.trixbox.org/forums/vendor-moderated-forums/sangoma/unable-dial-out-uk-e1-sangoma-a101d-trixbox-2-2 |
| 04:16.30 | SteveTotaro | maybe helpful, i am not in a reading mood |
| 04:16.38 | drmessano | lol |
| 04:16.44 | DoDaT69 | yea- found that today |
| 04:16.49 | [TK]D-Fender | DoDaT69, Don't forget to restart zaptel, etc |
| 04:18.06 | DoDaT69 | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 04:18.23 | SteveTotaro | good night everyone, happy Easter if you are into it |
| 04:18.44 | DoDaT69 | (SteveTotaro): happy easter to you as well! Thank you for your assistance, I really appreciate it! |
| 04:19.49 | SteveTotaro | the best part about what you are doing is seeing all the channels come up |
| 04:20.06 | DoDaT69 | ;) |
| 04:20.33 | DoDaT69 | this IS the first PRI I am turning up.. |
| 04:20.41 | DoDaT69 | sold a few of the cards/systems-- never turned it up myself |
| 04:21.20 | DoDaT69 | I usually use the a200d-- it is more budget friendly with the T1's I sell |
| 04:21.31 | DoDaT69 | smaller companies too |
| 04:22.10 | SteveTotaro | i can post what works for my card if that helps |
| 04:22.22 | DoDaT69 | you in the US? |
| 04:22.49 | rkeene | I am... I can also post my config (which is very similar to yours -- though I have 4 PRIs, and 1 is acting as a master) |
| 04:22.52 | SteveTotaro | DC/Balto |
| 04:23.46 | SteveTotaro | ok, rkeene will steer you in the right direction, i am bushed |
| 04:23.53 | rkeene | SteveTotaro, I had to go up there after Hurricane Katrina for about a month -- I didn't really care for it. |
| 04:24.20 | SteveTotaro | i plan on moving to north carolina |
| 04:24.24 | rkeene | Cool |
| 04:24.48 | SteveTotaro | but the one thing about DC is it is recession/depression proof for the most part |
| 04:25.01 | SteveTotaro | all those tax dollars |
| 04:25.03 | DoDaT69 | thank you fellas |
| 04:26.20 | rkeene | Haven't noticed anything like that here... but I live in the south (and as the song goes -- "... but we were so poor that we couldnt tell...") |
| 04:26.39 | DoDaT69 | (rkeene): where in the south? |
| 04:27.17 | rkeene | DoDaT69, Mississippi |
| 04:27.21 | DoDaT69 | oh okay- |
| 04:27.30 | DoDaT69 | thats the farthest west I ever dun been |
| 04:27.43 | DoDaT69 | boluxi |
| 04:27.51 | rkeene | I've been slightly farther west... I lived in Texas for a while |
| 04:28.01 | DoDaT69 | I am gonna travel one day |
| 04:28.08 | DoDaT69 | right now I am stuck to the ATL like stink on poo |
| 04:30.50 | rkeene | http://www.rkeene.org/tmp/zaptel.conf and http://www.rkeene.org/tmp/zapata.conf |
| 04:31.58 | DoDaT69 | ty |
| 04:32.07 | rkeene | No problem |
| 04:33.28 | DoDaT69 | Hmm |
| 04:33.36 | DoDaT69 | so you are using national for switchtype? |
| 04:33.51 | DoDaT69 | whats the big diff between that and ni1? |
| 04:33.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:36.23 | DoDaT69 | nevermind-- |
| 04:36.33 | DoDaT69 | I need sleep |
| 04:37.08 | DoDaT69 | I had to pull up some carpet at the jobsite earlier tonight-- used a peice of lcd styrofoam packaging for a pillow |
| 04:37.56 | rkeene | NI1 = "National ISDN-1", apparently an older version of "National" |
| 04:38.07 | DoDaT69 | right-- thats what we are using- |
| 04:38.08 | rkeene | ("National" = "National ISDN-2") |
| 04:38.19 | DoDaT69 | you are able to control your outbound cid with your's, right? |
| 04:38.50 | rkeene | "The main difference between National ISDN-1 and ISDN-2 is parameter downloading via components (a component being a sub-element of the Extended Facility information element). These components are used to communicate parameter information between ISDN user equipment, such as an ISDN telephone, and the ISDN switch." -- http://www.protocols.com/pbook/isdn.htm |
| 04:39.20 | DoDaT69 | my understanding is ni1 transmits number only |
| 04:39.29 | rkeene | Right |
| 04:39.29 | DoDaT69 | ni2 transmits number and name |
| 04:39.42 | rkeene | I believe that is correct, based on the information I've read |
| 04:39.52 | DoDaT69 | so, my last worry with this setup |
| 04:40.08 | DoDaT69 | is there any benifit to defining this out into groups, such as you have? |
| 04:40.21 | DoDaT69 | as long as I set my DID's for the extensions, it shouldnt matter, right? |
| 04:40.44 | DoDaT69 | and that I do not override the outbound cid on the outbound route-- |
| 04:41.07 | rkeene | I have them grouped for convience... I have 2 PRIs for the PSTN, 1 PRI for the DSN, and 1 outbound PRI for the modem bank |
| 04:41.23 | DoDaT69 | wow-- what the hell do you do? |
| 04:41.24 | DoDaT69 | Hehe |
| 04:41.29 | rkeene | So for the two PSTN PRIs, I can just refer to one group |
| 04:41.46 | *** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 04:41.56 | rkeene | It's just for our small internal phone network |
| 04:42.14 | DoDaT69 | modem bank? |
| 04:42.22 | DoDaT69 | dial up access still |
| 04:42.24 | rkeene | Yep |
| 04:42.37 | DoDaT69 | dang |
| 04:43.04 | rkeene | Some people can't get high speed internet where they need to communicate with us |
| 04:43.08 | DoDaT69 | ah |
| 04:43.14 | DoDaT69 | makes sense |
| 04:43.21 | DoDaT69 | so how are you routing the modems |
| 04:43.40 | DoDaT69 | is that actually dead ending in that machine? or do you have other hardware taking care of it? |
| 04:44.04 | rkeene | Another piece of hardware takes care of it -- it takes a PRI input and handles it |
| 04:44.29 | DoDaT69 | cool- I used to love modems in my day |
| 04:44.31 | rkeene | So calls for it come in from the PSTN or the DSN and then get routed to the outbound PRI, which is the inbound PRI for the modem pool |
| 04:44.38 | DoDaT69 | ran a bbs before this inet thing came out |
| 04:45.01 | rkeene | (dialplan just says: Dial(${DIALUPTRUNK}/1234) or something similar) |
| 04:45.02 | DoDaT69 | swift-- |
| 04:45.04 | DoDaT69 | right |
| 04:46.05 | rkeene | I ran a BBS for a very short time (using "Wildcat BBS", or something similar)... :-P |
| 04:46.20 | DoDaT69 | ye-- I liked WC! |
| 04:46.28 | DoDaT69 | I was RyBBS and PCB |
| 04:46.40 | DoDaT69 | back in the good ole os/2 warp or dos/desqview days |
| 04:46.52 | DoDaT69 | good times |
| 04:47.01 | rkeene | I still have a copy of OS/2 Warp 3.0 |
| 04:47.14 | rkeene | (On floppy disk) |
| 04:47.44 | drmessano | DOS 5.0 is the only real OS |
| 04:47.56 | obnauticus | wow |
| 04:48.01 | obnauticus | i didn't know this fucking os was fake. |
| 04:48.03 | rkeene | I have a copy of AT&T DOS 3.3 |
| 04:48.11 | obnauticus | lol. |
| 04:48.36 | drmessano | I have a copy of MS-DOS 3.3 |
| 04:49.10 | DoDaT69 | wow-- thats old |
| 04:49.18 | obnauticus | i have drmessano's mom. |
| 04:49.19 | DoDaT69 | I found my os2 warp the other day |
| 04:49.20 | rkeene | AT&T DOS is just a rebranded MS DOS. |
| 04:49.22 | obnauticus | that's even older. |
| 04:49.27 | DoDaT69 | tried to install it on a newer machine for kicks- it failed |
| 04:49.42 | DoDaT69 | that was a tough cookie when you didnt have a mouse- |
| 04:49.50 | DoDaT69 | back then it was 60$$ just for a ps2 mouse! |
| 04:49.54 | drmessano | PC Dos 1.2 |
| 04:50.00 | obnauticus | continues to play on his viola. |
| 04:50.01 | DoDaT69 | keep in mind, I was only 12 years old.. |
| 04:50.18 | DoDaT69 | that was a lot of yards to mow |
| 04:50.23 | rkeene | I ran OS/2 Warp 3.0 on my system for a while (486DX2/50, with 4MB of RAM) |
| 04:50.42 | DoDaT69 | that system ws the heat when it came out |
| 04:50.56 | DoDaT69 | I had a dx50 (never remmebered it having the /2) |
| 04:51.12 | DoDaT69 | when I oc'd it and lightening went through it-- got me a dx/4 100 |
| 04:51.17 | rkeene | But when Windows 95 came out, I switched to that for software support... and shortly thereafter to Linux (Slackware -- still running it to this day) |
| 04:51.34 | DoDaT69 | wow-- we have a good bit in common there |
| 04:51.40 | drmessano | My 386 33MHZ with 4MB and Windows for Workgroups 3.11 was my favorite system |
| 04:51.41 | DoDaT69 | I did get off slack tho |
| 04:51.46 | drmessano | Most fun I ever had |
| 04:51.49 | DoDaT69 | Hahaha |
| 04:51.57 | DoDaT69 | wow look at the time! |
| 04:51.58 | DoDaT69 | Just Kidding |
| 04:52.25 | rkeene | I first ran Windows on my 286 (Pacard Bell), Windows 3.1 though (3.11 required a 386) |
| 04:52.50 | *** join/#asterisk talntid (n=t@71-221-207-130.spkn.qwest.net) |
| 04:53.07 | drmessano | Windows 3.0 wasn't much fun on a 286 |
| 04:54.06 | rkeene | Yeah, I got rid of it and replaced it with a simpler windowing system ("Toybox 2" or something like that) |
| 04:54.36 | drmessano | 3.0 was slick.. only version you could run in 3 modes |
| 04:56.12 | rkeene | I found a copy of "Toybox II" -- still alive on the internet today! *fires up DOSBox* |
| 04:56.19 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-105-66.bltmmd.east.verizon.net) |
| 05:04.01 | DoDaT69 | again-- thanks for your help-- |
| 05:04.05 | DoDaT69 | I am going to sleep |
| 05:04.15 | DoDaT69 | Good Night |
| 05:04.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:06.03 | rkeene | Later |
| 05:06.57 | drmessano | Hmm |
| 05:15.45 | *** join/#asterisk hohum (n=dcorbe@99-204-84-225.area1.spcsdns.net) |
| 05:20.46 | jameswf-home | drmessano: hmmmm odd you have a real job now your never around |
| 05:22.29 | drmessano | lol |
| 05:22.34 | drmessano | Not during the day anymore |
| 05:22.48 | drmessano | Evenings I have been so wiped out I have been napping |
| 05:41.34 | *** join/#asterisk CCFL_Man2 (i=e8a178a5@pool-71-241-100-226.scr.east.verizon.net) |
| 05:44.36 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 05:46.33 | *** join/#asterisk talntid (n=t@71-221-207-130.spkn.qwest.net) |
| 05:46.34 | *** join/#asterisk TheSov (i=TheSov@dsl092-128-161.chi1.dsl.speakeasy.net) |
| 05:50.14 | TheSov | can anyone help me with some IVR issues? |
| 06:01.58 | obnauticus | drmessano would oyu liek to join me in a song. |
| 06:02.35 | obnauticus | We're no strangers to love. |
| 06:02.40 | obnauticus | You know the rules, and so do I! |
| 06:02.49 | obnauticus | A full commitment's what im... thinkin' of. |
| 06:02.52 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 06:02.55 | obnauticus | You wouldn't get this from any other guy. |
| 06:03.06 | obnauticus | Aiiiiiii just wanna tell you how im feelin' |
| 06:03.12 | obnauticus | Gotta make you... understand... |
| 06:03.15 | obnauticus | points at drmessano. |
| 06:06.03 | obnauticus | frowns. |
| 06:06.52 | rkeene | obnauticus, If you dial "7425" (RICK) from a phone on my network... that song plays. |
| 06:07.04 | obnauticus | my musiconhold is rick. |
| 06:07.13 | jameswf-home | ~rickroll |
| 06:07.14 | jbot | well, rickroll is http://www.internetisseriousbusiness.com, or http://www.xkcd.com/396/ |
| 06:07.21 | jameswf-home | ~fleamarket |
| 06:07.21 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 06:08.05 | obnauticus | lol. |
| 06:10.50 | drmessano | Never gonna give you up |
| 06:10.55 | obnauticus | NO |
| 06:10.58 | obnauticus | you started in the wrong place. |
| 06:11.02 | obnauticus | weait |
| 06:11.03 | obnauticus | no carry on |
| 06:11.05 | drmessano | I wasn't starting |
| 06:11.12 | drmessano | I was proclaiming my love for you |
| 06:11.17 | obnauticus | k |
| 06:11.19 | obnauticus | do it then. |
| 06:11.36 | obnauticus | Never gonna let you down.. |
| 06:11.39 | drmessano | I just did.. it doesnt mean as much when you ask for it |
| 06:11.48 | obnauticus | ... |
| 06:12.40 | obnauticus | You gonna sing or not drmessano? |
| 06:13.04 | drmessano | No, not gonna sing.. This is not #song |
| 06:13.15 | obnauticus | :| |
| 06:13.24 | obnauticus | I guess you only love me outside of #asterisk |
| 06:17.11 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
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| 06:58.16 | Olobola | so from AMI, how can get the status of a specific sip user? I can't seem to find the command. |
| 07:02.38 | jameswf-home | sip show peer <peer> sip show ..... |
| 07:04.18 | *** join/#asterisk marc\cba (n=l@cpc3-whit2-0-0-cust53.cdif.cable.ntl.com) |
| 07:07.13 | *** join/#asterisk st3r30 (n=slobos@201-212-150-29.cab.prima.net.ar) |
| 07:07.35 | st3r30 | hello, does any one here speaks spanish? |
| 07:08.51 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 07:12.01 | tzafrir_home | st3r30, some at #asterisk-es do. (yeah, a very small population) |
| 07:12.27 | st3r30 | thanks tzafrir_home |
| 07:23.18 | *** join/#asterisk hohum_ (n=dcorbe@70.6.162.128) |
| 07:32.05 | st3r30 | tzafrir_home: Im newbie installing asterisk I followed some tutorials and succesfully installed asterisk |
| 07:33.06 | st3r30 | the point is that I want to make it work with astercrm, and setting up astercrm I have to set asterisk`s user an password, |
| 07:33.41 | st3r30 | and I've never set this information during the asterisk setup, can you tellme where should I set it ? |
| 07:33.56 | st3r30 | in manager.conf ? |
| 07:37.25 | tzafrir_home | st3r30, hmm... I think that they "communicate" through mysql |
| 07:37.37 | tzafrir_home | BTW: http://astercrm.org/astercrm_documents/installation |
| 07:37.42 | tzafrir_home | recommends a chmod 777 |
| 07:37.44 | *** join/#asterisk steliosk (n=Stelios@athedsl-4401580.home.otenet.gr) |
| 07:38.14 | st3r30 | I mean this |
| 07:38.15 | st3r30 | [asterisk] |
| 07:38.15 | st3r30 | server = 127.0.0.1 |
| 07:38.15 | st3r30 | port = 5038 |
| 07:38.15 | st3r30 | username = |
| 07:38.16 | st3r30 | secret = |
| 07:38.29 | tzafrir_home | chmod 777 generally means someone didn't know about chown, or thought it complicates your system too much |
| 07:38.41 | tzafrir_home | and chose to leave you with a broken system instead |
| 07:39.25 | st3r30 | ups |
| 07:39.41 | tzafrir_home | st3r30, right, it seems you need to create a username in manager.conf |
| 07:40.14 | tzafrir_home | BTW: please don't paste here anything longer than 3 lines. Use soemthing like pastebin.ca |
| 07:40.52 | st3r30 | sorry about that |
| 07:40.58 | st3r30 | the point is that I created like [admin] secret = testing , restarted asterisk and it doesn't work , |
| 07:41.48 | tzafrir_home | There's no such thing as "does not work". Please describe what you do see and not what you don't see. |
| 07:43.03 | st3r30 | mmm, It's a little difficult for me to fully explain my self in english but i'll try it |
| 07:47.10 | st3r30 | I'm suppoused to set de daemon and set it up un /opt/asterisk/scripts ... I do it, and when I test it , I get this response, Mysql Authentication accepted , Asterisk authentication failed, so my first think is that asteris's user and secret are not properly set |
| 07:59.01 | tzafrir_home | st3r30, /opt/asterisk ? on what platform did you install asterisk? |
| 07:59.55 | st3r30 | debian |
| 08:04.54 | adorah | Hi everyone Hi Tzafrir |
| 08:23.39 | tzafrir_home | steliosk, so how did you get to use /opt/asterisk? ah, the astercrm install. hmmm |
| 08:24.26 | tzafrir_home | steliosk, oops, sorry, meant st3r30 |
| 08:25.07 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 09:56.47 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 09:57.02 | madduck | i am trying to route calls to germany via sipgate and all other calls via another provider |
| 09:57.06 | madduck | exten => _00049X.,1,Dial(SIP/${EXTEN:5}@sipgate) |
| 09:57.08 | madduck | exten => _0X.,1,Dial(SIP/${EXTEN:1}@ifi) |
| 09:57.13 | madduck | the second one works, |
| 09:58.53 | madduck | but when i dial a 00049... number, i can't make a connection |
| 10:00.08 | madduck | i get |
| 10:00.09 | madduck | SIP/2.0 100 Giving a try |
| 10:00.10 | madduck | and then |
| 10:00.14 | madduck | SIP/2.0 183 Session Progress |
| 10:00.25 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 10:00.25 | madduck | and then |
| 10:00.26 | madduck | SIP/2.0 603 Declined |
| 10:04.04 | *** join/#asterisk KaiK (n=KaiK@dslb-088-076-060-202.pools.arcor-ip.net) |
| 10:07.31 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 10:15.22 | madduck | ha! it helps to charge up the account. :) |
| 10:18.48 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 10:31.47 | *** join/#asterisk felipex (n=asdasd@88-149-172-71.static.ngi.it) |
| 10:32.03 | felipex | hi at all |
| 10:32.29 | felipex | how can i detect fax with asterisk 1.4 and sip ? |
| 10:33.40 | felipex | i read about nv_faxdetect but the newmantel site doesn't work |
| 10:42.52 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 11:34.30 | Thazza | Hey all. |
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| 12:27.38 | smace | my asterisk server is crashing :( |
| 12:28.09 | smace | has no idea why that is crashing. |
| 12:35.18 | *** join/#asterisk simNIX (n=user@82-204-21-111.dsl.bbeyond.nl) |
| 12:35.23 | simNIX | greetings |
| 12:35.42 | smace | Good Day |
| 12:39.03 | *** join/#asterisk awormus (n=wormus@adsl-074-173-206-249.sip.bct.bellsouth.net) |
| 12:44.23 | Vodak | wonderful day in the land of Cleveland |
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| 13:34.18 | Datax | Hi all |
| 13:34.29 | Datax | I have a problem with a SIP provider I'm using |
| 13:34.37 | Datax | I have 2 sip providers |
| 13:34.55 | Datax | one of them works fine for outgoing and incoming but the other one is "strange" |
| 13:35.20 | Datax | when I call a land line I use for tests through it I don't get any sound |
| 13:35.25 | Datax | nothing either way |
| 13:35.29 | Datax | yet the phone rings |
| 13:35.54 | Datax | I have noticed a difference in the CLI between the 2 providers |
| 13:36.09 | Datax | when I dial using the one that that doesn't work I get this extra line : |
| 13:36.17 | Datax | -- Packet2Packet bridging SIP/100-081fb280 and SIP/Supinfo-081f9ac8 |
| 13:36.34 | Datax | what is packet2packet bridging ? |
| 13:38.04 | Datax | anyone ? :) |
| 13:38.04 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 13:58.24 | rkeene | Packet2Packet bridging is SIP/RTP bridging between two compatible clients |
| 14:02.37 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:06.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:10.07 | eric2 | Datax: maybe you have nat issues? |
| 14:10.23 | eric2 | where the phone rings but you cannot hear the person speaking? |
| 14:10.39 | Datax | I am indeed behind nat |
| 14:10.53 | Datax | but why does one of the providers work without a problem and not the other ? |
| 14:10.53 | eric2 | does the server have a public ip? |
| 14:10.56 | Datax | no |
| 14:11.12 | eric2 | I'd say that could be a problem |
| 14:11.18 | Datax | I agree |
| 14:11.27 | eric2 | sip and having a server behind nat is a headache |
| 14:11.33 | eric2 | unless you use iax |
| 14:11.36 | Datax | but I don't understand why I don't have the same problem with the other SIP provider |
| 14:11.40 | eric2 | and iax I found was unreliable |
| 14:11.44 | Datax | no I'm using SIP with both providers |
| 14:12.46 | Datax | I know that the provider that works uses Openser |
| 14:12.56 | Datax | don't know about the other one though |
| 14:13.14 | eric2 | I don't think that should matter |
| 14:14.16 | Datax | anyone have a brilliant idea then ? ;-) |
| 14:25.32 | rkeene | You could configure your NAT to not translate packets with SPT/DPT 5060 ? |
| 14:27.45 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:32.07 | jameswf-home | lmao I just got allison to do the minimall rap with celestrial |
| 14:34.28 | jameswf-home | *Cepstral |
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| 14:36.36 | *** join/#asterisk adjohn (n=adjohn@i121-115-187-159.s05.a013.ap.plala.or.jp) |
| 14:38.16 | coppice | sounds like someone who starts cepstral with an 's' sound :-) |
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| 14:53.30 | rkeene | Allison ? |
| 14:53.56 | *** join/#asterisk S4L4D1N (n=Mak@adsl196-157-152-217-196.adsl196-13.iam.net.ma) |
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| 14:57.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:57.53 | coppice | well, you don't think those voice prompts are Mark Spencer talking, do you? |
| 14:58.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:06.08 | jameswf-home | keeps allison in a box made by dell :)) |
| 15:08.40 | coppice | you mean *sold* by Dell |
| 15:09.56 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:10.28 | jameswf-home | hate when I forget to leave before switching to the vpn .... bahh |
| 15:11.46 | polerin | this is why ssh+irssi+screen == win |
| 15:12.29 | jameswf-home | I use to do an ssh tunnel with vnc... vpn is easier. |
| 15:12.49 | polerin | i'm not using vnc |
| 15:13.13 | polerin | irssi is the irc client, I just attach to it using screen |
| 15:13.39 | jameswf-home | is not using vpn for irc... doing actual work :) |
| 15:13.45 | polerin | :D |
| 15:13.57 | polerin | yeah but you were complaining about forgetting to leave |
| 15:14.08 | polerin | so that was an answer directed at that issue |
| 15:14.10 | polerin | hehe |
| 15:14.45 | jameswf-home | in linux the vpn replaces the route which messes with live apps such as irc |
| 15:15.11 | polerin | I run the irc client on the server, not locally |
| 15:15.57 | *** join/#asterisk dijungal (n=kdaniel@cpe-65-24-202-182.insight.res.rr.com) |
| 15:16.10 | dijungal | hi anyone here every provision the GXP2020 phones |
| 15:16.14 | jameswf-home | thinks linux hates him for putting iez on it |
| 15:16.19 | dijungal | i've been trying to do that for about 3 days now... no luck |
| 15:16.23 | jameswf-home | s/iez/ie7 |
| 15:16.28 | polerin | rather, I run it inside of screen on the server, then connect to the server using ssh, and attach to that screen. that way it doesn't matter if I get disco'ed. |
| 15:16.35 | polerin | and why would you do somethign like that seriously ;) |
| 15:16.49 | jameswf-home | ~developers |
| 15:16.50 | jbot | from memory, developers is http://www.youtube.com/watch?v=KMU0tzLwhbE |
| 15:16.50 | polerin | oooh look a telephony question |
| 15:17.11 | polerin | jameswf-home: heheh yeah I understand. unfortunately. |
| 15:17.13 | jameswf-home | thinks grandsuck is a bad idea |
| 15:17.33 | polerin | but I'm a gamer so I have a windows box around anyhow |
| 15:17.37 | dijungal | i don't like grandstreams... but my client refuses to get polycoms... :S |
| 15:17.45 | dijungal | so i have to provision his grandstreams... |
| 15:17.50 | dijungal | and i've been at this waay too long |
| 15:18.15 | jameswf-home | I find a nice balance at aastre |
| 15:18.24 | dijungal | i've tried configuration generators etc.., i'm using HTTP provisioning... the phone pics up the config but does nothing |
| 15:18.25 | jameswf-home | *asstra |
| 15:18.28 | jameswf-home | bah |
| 15:19.08 | coppice | asstra has such faith in VoIP they just bought a traditional PBX business |
| 15:19.16 | dijungal | and then ever so often the web interface freezes... so i would have to restart the phone a few times.. :s |
| 15:19.43 | jameswf-home | I wrote a rick roll module for freepbx that allows you to point people at rick astley or at the fleamarket guy... but cant distribute for legal reasons (totaly sucks) |
| 15:21.37 | jameswf-home | curse the riaa |
| 15:21.54 | dijungal | anyone? |
| 15:22.03 | jameswf-home | ~grandstream |
| 15:22.04 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 15:22.13 | coppice | isn't it amazing that someone would try to protect Rick Astley recordings? :-\ |
| 15:22.37 | dijungal | ? |
| 15:22.38 | jameswf-home | I dont think i would get slammed for the minimall rap |
| 15:26.55 | _ShrikE | are you talkin bout flea market? |
| 15:28.52 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-25-220.w90-49.abo.wanadoo.fr) |
| 15:29.18 | Rico29 | hi |
| 15:29.38 | Rico29 | anybody here who knows OPAL libs ? |
| 15:31.29 | jameswf-home | heh heh |
| 15:31.32 | jameswf-home | :) |
| 15:31.38 | jameswf-home | ~fleamarket |
| 15:31.39 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 15:32.39 | Rico29 | wow |
| 15:32.51 | Rico29 | wtf |
| 15:38.52 | Qwell | jameswf-home: http://www.youtube.com/watch?v=j8oTynm8mAg |
| 15:40.57 | jameswf-home | I saw that... I am fond of http://www.youtube.com/watch?v=R6P8qssKrSw |
| 15:41.35 | jameswf-home | brb going to kill the vpn |
| 15:42.33 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
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| 16:12.47 | ZPertee | Why is Linksys ATA such a pain to configure? Can't even get it to register with Asterisk |
| 16:14.18 | _ShrikE | ZPertee: What model? Most linksys/sipura atas are dirt simple |
| 16:14.38 | *** join/#asterisk RoyK (n=roy@ip-103-52-149-91.dialup.ice.no) |
| 16:16.02 | ZPertee | _ShrikE, SPA8000 |
| 16:16.12 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 16:16.53 | jameswf-home | has heard nothing but praise for linksys phones... |
| 16:17.17 | ZPertee | _ShrikE, It won't register one particular user for some reason but it is configured as all of the rest. |
| 16:17.37 | ZPertee | ~linksys |
| 16:17.37 | jbot | i heard linksys is a tool of satan |
| 16:17.46 | ZPertee | bout sums it up |
| 16:18.10 | [TK]D-Fender | ZPertee, You are clearly thinking too hard. You have like 4 fields to fill in to get those thigns working... |
| 16:19.02 | jameswf-home | but [TK]D-Fender there are 50,000 options surely cant leave em empty or default what would the squirrels say |
| 16:19.36 | [TK]D-Fender | jameswf-home, Squirrels serve my mail.... |
| 16:19.46 | GlobeTrotter | <PROTECTED> |
| 16:19.49 | [TK]D-Fender | jameswf-home, And yeah, you DO leave the other 50,000 optiosn default :) |
| 16:20.23 | jameswf-home | GlobeTrotter: network gnomes |
| 16:20.40 | Olobola | how do I prevent a sip client from timing out? |
| 16:21.44 | jameswf-home | GlobeTrotter: redphone seems to be popular most HA asterisk systems require a ton of work to set up |
| 16:22.00 | jameswf-home | ~redfone |
| 16:22.05 | jameswf-home | ~redphone |
| 16:22.10 | jameswf-home | ~bah |
| 16:22.11 | jbot | methinks bah is everyone's other favourite word (see heh) |
| 16:23.03 | jameswf-home | http://www.red-fone.com/ |
| 16:24.13 | [TK]D-Fender | Olobola, if they have issues, what makes you think you can control them all? |
| 16:28.00 | *** join/#asterisk gitguy (n=git@67-207-141-250.slicehost.net) |
| 16:28.06 | gitguy | hi where can i find jobs with asterisk |
| 16:29.23 | jameswf-home | has an asterisk job... |
| 16:29.40 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 16:29.43 | jameswf-home | gitguy: what state |
| 16:29.58 | gitguy | i'm looking for remote |
| 16:30.05 | gitguy | online |
| 16:30.36 | jameswf-home | good luck with that... |
| 16:31.36 | gitguy | yeah i need to move out from this country :/ |
| 16:31.45 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 16:32.28 | polerin | :P |
| 16:32.50 | Olobola | [TK]D-Fender: I'm trying to issue calls based on the status of a sip client using AMI. My sip client seems to timeout after a period of time. I would like my client (exten) to appear registered whenever it's connected. |
| 16:33.04 | jameswf-home | heh http://thecontaminated.com/geeky-computer-station/ |
| 16:33.44 | [TK]D-Fender | Olobola, What do you think "timeout" implies? That its can't SEE it any more and is thus UNCONNECTED |
| 16:36.37 | Olobola | [TK]D-Fender: is my sip client causing this, or is this a default setting somewhere in asterisk? |
| 16:37.01 | [TK]D-Fender | Olobola, its the fact that its not answering. And that any number of things between the two |
| 16:37.16 | [TK]D-Fender | Olobola, The reason is not as relevant as the fact itself. |
| 16:40.44 | Olobola | [TK]D-Fender: ok. I can place a call at any time though, so I guess it's reregistering right before I place a call. So this doesn't mean at some point either asterisk or exten is unregistering? |
| 16:41.14 | [TK]D-Fender | Olobola, No, you don't need to be registered to place a call at all.... |
| 16:41.25 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:41.33 | [TK]D-Fender | ~sipregister |
| 16:41.34 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 16:43.15 | [TK]D-Fender | BRB, server maintenenance calls.... |
| 16:43.29 | [TK]D-Fender | re-ask me stuff when I'm back in 5 min or so. |
| 16:44.29 | *** join/#asterisk Chris-NB (n=chris@213162066139.public.t-mobile.at) |
| 16:47.16 | GlobeTrotter | ok,,, thanks alot guys |
| 16:57.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:59.13 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 16:59.22 | ZPertee | how do i delete .swp files? |
| 17:01.37 | ZPertee | nm |
| 17:04.42 | *** join/#asterisk samoshit (n=msauce@ool-18be2518.dyn.optonline.net) |
| 17:05.09 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 17:05.14 | [TK]D-Fender | \o/ |
| 17:05.16 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:06.33 | samoshit | hey guys, i just setup a trixbox for myself to use. can anyone recommend a free inbound voip service to use for testing purposes, until i get it set up the way i like it to connect to my real line ? |
| 17:07.12 | ZPertee | [TK]D-Fender, I want asterisk to pickup Zap/8, flash() it, SendDTMF(70), playback(answer_the_phone), and then hangup. however it just dials and then sits it won't flash the line how do I get it to keep going down the dialplan? |
| 17:07.26 | _ShrikE | samoshit: IPKALL |
| 17:07.34 | ZPertee | samoshit, why not just use a softphone such as x-lite? |
| 17:07.58 | samoshit | thanks shrike |
| 17:08.20 | samoshit | ZPertee i dunno i'm trying to keep it as realistic as possible |
| 17:08.26 | samoshit | for when i actually start using it |
| 17:08.40 | samoshit | i need to set up extensions and dial plans for a small office so.. |
| 17:09.24 | ZPertee | samoshit, no problem just sometimes X-Lite is an easier and quicker way to test as compared to setting something up with IPKALL |
| 17:09.47 | samoshit | is IPKALL as easy as signing up and entering a few lines in iax.conf ? |
| 17:10.06 | samoshit | i see what you're saying, maybe i'll use the softphone |
| 17:10.48 | ZPertee | samoshit, last time I check you have to setup a number with FreeWorldDialup, setup a number with IPKALL (which forwards to FWD), and then modify configs |
| 17:11.21 | ZPertee | samoshit, FWD isn't acutal telephone number just a sip account |
| 17:14.49 | samoshit | i see |
| 17:15.08 | [TK]D-Fender | ZPertee, Whats the point of picking up then flashing? |
| 17:15.25 | samoshit | and with your method, i have the trixbox act as it has just been dialed when it receives a call from the softphone |
| 17:15.57 | ZPertee | samoshit, that's what I would do...totally up to you though :) |
| 17:16.04 | samoshit | yeah i think i like that better |
| 17:16.12 | samoshit | now, what softphone ? |
| 17:16.18 | samoshit | freeware, lightweight, no bullshit |
| 17:16.23 | samoshit | for winxp |
| 17:16.30 | ZPertee | samoshit, X-Lite |
| 17:16.36 | samoshit | oh right, you said that :P |
| 17:16.59 | ZPertee | http://www.counterpath.com/x-lite.html |
| 17:17.34 | polerin | just go to see for herself how much better a hardphone is than soft phones :P |
| 17:18.45 | ZPertee | [TK]D-Fender, Zap/8 is connected to my old Avaya PBX, and the loudspeaker is connected to the Avaya. The idea is to pickup the line to get to avaya and then flash the channel and then say whatever I want to |
| 17:19.05 | ZPertee | [TK]D-Fender, how else would you do it? |
| 17:19.28 | [TK]D-Fender | ZPertee, I fail to see the point of grabbing a line then flashing. Is this something you'd do with an analog phone? |
| 17:20.02 | ZPertee | [TK]D-Fender, currently we pickup an analog phone and then page someone. however would be nice to get it automated |
| 17:20.06 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 17:20.23 | [TK]D-Fender | ZPertee, If you don't flash NOW, why do you need * to? |
| 17:21.02 | ZPertee | [TK]D-Fender, on our system we push the intercom button on the phone however a flash is the equivalent of the intercom button |
| 17:22.19 | [TK]D-Fender | ZPertee, So you "pickup", "flash", "dial", "wait","speak your message"? |
| 17:22.43 | ZPertee | [TK]D-Fender, yep |
| 17:22.50 | [TK]D-Fender | ZPertee, Where "flash" is the functional equivalent of your "intercom" button? |
| 17:24.15 | ZPertee | [TK]D-Fender, I am integrating Asterisk with Avaya PBX and there is special phones connected to the system which don't have a flash button but an intercom button. asterisk is connected as an extension on the Avaya and the Flash() command is the functional equivalent of your "intercom" button |
| 17:24.50 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:25.49 | [TK]D-Fender | ZPertee, so if you did this on an analog phone it works? |
| 17:26.06 | hesco | I'm still working on making an asterisk<->iaxmodem<->hylafax connection. iax2 show peer ttyIAX0 seems to show that and the iaxmodem are communicating. And I have been able to make hylafax's sendfax make the cli console show me activity. But I'm not sure how to pass th outbound fax dialplan a phone number to attempt to send to. Can anyone here please advise? |
| 17:27.11 | ZPertee | [TK]D-Fender, yeah I plugged a regular analog phone in and did the "pickup", "flash", "dial", "wait","speak your message" sequence |
| 17:27.32 | hesco | Apparently my irc client interprets an asterisk to send in bold what is between those punctuation. |
| 17:29.02 | mmlj4 | *mine too* # or not |
| 17:29.05 | [TK]D-Fender | ZPertee, Tricky to get this order right.... |
| 17:29.06 | hesco | On the bash console where I started the iaxmodem instance, I'm seeing this: Incoming call connected fax, (null), (null). |
| 17:30.00 | hesco | I'd guess that one of those nulls refers to the outbound fax's destination number and the other to the file to be faxed. How would I feed those items from the Dial application? |
| 17:32.32 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 17:32.49 | jblack | hesco: I'll paste something for you. |
| 17:33.00 | hesco | jblack: thanks! |
| 17:33.25 | hesco | I'm looking forward to seeing some clues, I appreciate your time. |
| 17:33.43 | jblack | hesco: http://pastebin.ca/index.php |
| 17:33.50 | hesco | looking now |
| 17:34.09 | jblack | I'm using IPKall. They come in over iax, asking for extension 998. |
| 17:34.30 | *** join/#asterisk UserReg_CL (n=COB@200.113.99.156) |
| 17:34.38 | hesco | jblack: your url gave me the index page. You got a link to your paste, please? |
| 17:34.51 | jblack | http://pastebin.ca/952936 |
| 17:34.52 | jblack | sorry |
| 17:35.37 | UserReg_CL | hi... for better communication is better low value MTU ? |
| 17:35.48 | hesco | thanks for that. a search on your handle turned up only very old pastes |
| 17:36.21 | jblack | I'm surprised my handle shows at all. I usually use anonymous |
| 17:36.54 | hesco | a vanity search can be a scary thing. |
| 17:37.20 | hesco | This looks like how you handle inbound faxes. |
| 17:37.33 | hesco | Any clues about routing outbound faxes, please? |
| 17:38.47 | jblack | i wasn't able to get outbound faxes working |
| 17:39.42 | jblack | I misunderstood. I thought you had trouble with inbound faxes. The problems I had with outbound faxes mostly centered around cups |
| 17:40.26 | jameswf-home | real men dont wear cups |
| 17:40.30 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-161-187.red.bezeqint.net) |
| 17:40.48 | jblack | Men that don't wear cups are rarely men for long. |
| 17:41.37 | hesco | any experience with hylafax, then? |
| 17:41.56 | hesco | Does this dialplan work for inbound using an iaxmodem? |
| 17:42.10 | jblack | Yes, it does. |
| 17:42.36 | hesco | and what is IPKALL? |
| 17:42.49 | jblack | IPKALL is a service that provides free washington state numbers. |
| 17:43.07 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:43.33 | ZPertee | [TK]D-Fender, sorry my irc went nuts! did you have any suggestions whie I was gone? |
| 17:45.13 | [TK]D-Fender | ZPertee, I'd say use a dynamic feature with features.conf that will flash the other side. So you dial the Zap, then press say *1, Asterisk will flash the line based on your features.conf and you can go from there. |
| 17:47.27 | *** join/#asterisk ManxPower (n=manxpowe@151.sub-75-201-255.myvzw.com) |
| 17:51.06 | ZPertee | [TK]D-Fender, so if I set something like that up in features.conf then if I did Dial(ZAP/8/*1) would it connect and auto flash? |
| 17:52.15 | *** join/#asterisk KaiK (n=KaiK@dslb-084-063-102-003.pools.arcor-ip.net) |
| 17:53.45 | [TK]D-Fender | ZPertee, No. |
| 17:53.56 | [TK]D-Fender | ZPertee, No auto way I can think of to sanely do this. |
| 17:54.25 | [TK]D-Fender | ZPertee, you can't tel dial itself to flash. You MIGHT be able to use M to call Flash... but I'm grey on this one |
| 17:54.33 | *** join/#asterisk ejbvanc (n=ejbvanc@c-24-22-57-253.hsd1.wa.comcast.net) |
| 17:56.09 | ManxPower | for some reason I thought Dial had a special flag like "w" to do a flash, but maybe I just used M() |
| 17:58.56 | KaiK | hello everyone |
| 17:58.59 | *** join/#asterisk jonathanpoon (n=poonj@c-76-105-5-201.hsd1.ca.comcast.net) |
| 17:59.13 | KaiK | I have one problem using MP3Player() in extension.conf |
| 17:59.33 | KaiK | it works fine with a normal mp3 file but crashes after 2 seconds using a mp3-stream |
| 18:00.06 | KaiK | it seems to be a problem with id3tags and sync, but I dont know how to handle it and didnt find any information about that |
| 18:00.44 | KaiK | does someone of you know about the problem? |
| 18:01.42 | KaiK | here is some info about it: http://www.voip-info.org/wiki/view/Asterisk+cmd+MP3Player but i dont knwo how to supress "stderr output " |
| 18:03.00 | jblack | hesco: ping |
| 18:04.30 | ZPertee | [TK]D-Fender, what do you mean by "use M to call Flash". what is M |
| 18:06.20 | [TK]D-Fender | ZPertee, a Dial option |
| 18:07.15 | ZPertee | [TK]D-Fender, oh ok executes macro |
| 18:07.53 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 18:11.01 | ejbvanc | hello everyone, i have two TDM800P cards that have FXS modules, and then i have a single digium t1 card, when I use Asterisk 1.4, the DTMF tones get mangled, but if I use Asterisk 1.2, the DTMF tones pass through to the other device connected to the TDM800P, any thoughts? |
| 18:14.36 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:15.25 | *** join/#asterisk jdg (n=jdg@203.185.181.138) |
| 18:16.14 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:18.41 | KaiK | does someone of you know about mp3player()? |
| 18:19.07 | *** join/#asterisk talntid (n=swarm@66.208.251.174) |
| 18:20.41 | Olobola | [TK]D-Fender: how can I tell (using AMI) if a sip client is connected, in a call etc? |
| 18:22.35 | *** join/#asterisk exvito (n=exvito@89-180-203-113.net.novis.pt) |
| 18:22.37 | dijungal | aaah cracked the Grandstream provisioning |
| 18:22.43 | Olobola | sipshowpeer |
| 18:23.00 | dijungal | the default template file was set to NAT Traversal = No |
| 18:23.13 | dijungal | dunno why!!!!.. sounds retarted to me |
| 18:25.25 | JunK-Y | using a grandstream!!! .. sounds retarded to me :) |
| 18:26.08 | CCFL_Man2 | ewww, grandsteam |
| 18:40.07 | UnixDog | [Mar 22 11:39:16] WARNING[1219]: chan_sip.c:3675 sip_write: Asked to transmit frame type 2, while native formats is 0x4 (ulaw)(4) read/write = 0x2 (gsm)(2)/0x2 (gsm)(2) |
| 18:42.43 | *** join/#asterisk TheSov (i=TheSov@dsl092-128-161.chi1.dsl.speakeasy.net) |
| 18:44.39 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 18:46.23 | TheSov | can anyone help me with an IVR, i went the voip-info site and the directions they give dont really work correctly i get the error "no application 'DigitTimeout' for extension X |
| 18:46.36 | ManxPower | dijungal: almost nobody uses templates with Asterisk |
| 18:47.14 | ManxPower | TheSov: The wiki is frequently wrong and almost always outdated. You need to read channelvariables.txt and upgrade.txt in your Asterisk source. |
| 18:47.29 | TheSov | thank you ManxPower |
| 18:48.01 | ZPertee | is there any reason why I can't put a splitter in an fxo port? |
| 18:48.32 | [TK]D-Fender | ZPertee, You can't plug 2 LINES into each other... |
| 18:48.38 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:48.45 | [TK]D-Fender | ZPertee, When one rings, it fries the other |
| 18:48.49 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:48.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:49.07 | [TK]D-Fender | ZPertee, Or just as bad, both ring and your card jsut fries |
| 18:49.34 | ManxPower | ZPertee: you could put a splitter for one port to go to asterisk and the other to a phone, but Asterisk will still try to dial out that port even if it's in use. |
| 18:49.58 | TheSov | can you recommend a upto-date guide for setting up an IVR? |
| 18:50.32 | ZPertee | ~thebook |
| 18:50.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:51.48 | TheSov | thanks ZPertee |
| 18:52.06 | ZPertee | TheSov, np |
| 18:55.53 | TheSov | oh man i wish i had this book a week ago... woulda saved me, well, a week |
| 18:58.27 | ZPertee | TheSov, yeah I taught myself asterisk through this book and I continue to go back to it lots |
| 18:59.08 | jblack | I keep my * book next to my desk in the same way a religous nut may keep a bible on their kitchen desk |
| 18:59.32 | TheSov | lol well i got finished right now what it took me 2 days trying to figure out, now im getting a "spawn extenstion" error but ill figure that out too with this book |
| 19:00.51 | ZPertee | TheSov, if you can't figure it out put it in a pastebin and I'll take a look |
| 19:01.08 | ZPertee | jbot, tell TheSov about pastebin |
| 19:02.04 | TheSov | ok |
| 19:02.04 | TheSov | ty |
| 19:04.25 | TheSov | ok i see the problem, when i dial the extenion its listening to the first DTMF tone and going with that, how would i make it listen to all tones until im done |
| 19:05.46 | ZPertee | TheSov, if I understand you right I would use the WaitExten() application |
| 19:05.54 | TheSov | its in their |
| 19:06.04 | TheSov | lemme pastebin it |
| 19:08.29 | TheSov | http://pastebin.com/d7487b275 |
| 19:08.55 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 19:10.38 | ZPertee | TheSov, so what happens when you call in? Does it play the al-welcome file? |
| 19:10.57 | TheSov | yes |
| 19:11.10 | TheSov | when i type 401 at the keypad |
| 19:11.20 | TheSov | it says in the console that extenion 4 is not valid |
| 19:13.06 | ZPertee | TheSov, possibly give a little more time than 10 seconds, and make sure that you are indeed pushing 4-0-1 |
| 19:13.10 | jblack | hmm. callwithus is out of 800 dids |
| 19:13.20 | TheSov | ok |
| 19:13.24 | TheSov | brb |
| 19:14.41 | ZPertee | TheSov also do you type reload in the asterisk cli after you make changes? |
| 19:15.00 | TheSov | Invalid extension '4' in context 'default', yes i did |
| 19:15.21 | TheSov | infact i went so far as to even "stop gracefully" and start it again |
| 19:15.52 | ManxPower | sounds like a context= issue to me |
| 19:16.24 | TheSov | hmm, well i only have 1 context and that is default |
| 19:16.40 | ZPertee | ManxPower, but why would it even do the background application? |
| 19:17.22 | TheSov | im gonna try making the extention 4 and see what happend |
| 19:19.12 | TheSov | that worked, sort of... now it says unable to create channel of that type SIP (no route to destination) |
| 19:20.54 | TheSov | ok its working now |
| 19:21.01 | TheSov | the sip phone was off |
| 19:21.22 | TheSov | so is their any way to make it listen to more than 1 dtmf tone? |
| 19:21.45 | *** join/#asterisk burt75 (n=hatrista@189.153.220.20) |
| 19:22.16 | burt75 | hello guys |
| 19:22.22 | *** join/#asterisk RobH (n=RobH@36-159.96-97.tampabay.res.rr.com) |
| 19:22.34 | TheSov | hello |
| 19:23.07 | burt75 | excuse me any alternative to trixbox with a "existing centos" script? |
| 19:23.08 | hesco | Shouldn't my call plan make some allowance for routing my iaxmodem calls through the diamondcard account which makes the connection to pots for me on my outbound voice calls? |
| 19:23.40 | hesco | that is, my outbound voice use diamondcard to reach the teleco network. |
| 19:23.57 | hesco | If I'm sending outbound fax to teleco connected recipients, |
| 19:24.28 | hesco | should I not need some reference to the diamondcard carrier in order to make that connection? |
| 19:24.37 | ZPertee | TheSov, to be real honest I have never had that problem. seems like asterisk isn't getting all of the digits or something weird |
| 19:24.45 | hesco | And if so, what does that look like in my dialplan? |
| 19:25.11 | TheSov | does it matter that im testing with a cell phone? |
| 19:25.46 | ZPertee | I wouldn't think so but I would try a softphone for testing |
| 19:26.09 | ZPertee | you have asterisk setup with a sip telephone number or what/ |
| 19:26.19 | TheSov | i have a sip/sip system here |
| 19:26.23 | TheSov | no pots lines |
| 19:28.27 | UnixDog | ok I have a weird issue. I can call my cousin on exten and it transcode fine polycom > ulaw > gsm > his softphone but when he calls me we get this |
| 19:28.45 | UnixDog | Mar 22 11:39:16] WARNING[1219]: chan_sip.c:3675 sip_write: Asked to transmit frame type 2, while native formats is 0x4 (ulaw)(4) read/write = 0x2 (gsm)(2)/0x2 (gsm)(2) |
| 19:28.54 | ZPertee | I personally would test with a softphone first and then use my pots later |
| 19:29.02 | UnixDog | it seems not to want to transcode |
| 19:29.35 | ZPertee | TheSov, sorry but I just don't know quite how to help you |
| 19:31.08 | ZPertee | TheSov, also instead of X for your extension I would use s so that it auto starts |
| 19:33.19 | ZPertee | TheSov, http://pastebin.ca/953061 |
| 19:34.01 | UnixDog | any idea |
| 19:34.10 | UnixDog | and the call is going sip to iax |
| 19:34.21 | UnixDog | he is on a iax softphone |
| 19:34.28 | UnixDog | I am on a polycom 550 |
| 19:36.00 | TheSov | ZPertee i have the actual incoming phone# their |
| 19:36.09 | TheSov | so i just replaced that with X |
| 19:38.13 | TheSov | is IAX better than sip? |
| 19:39.14 | riddlebox | if you have SLA setup, can you have the blf for say line 1 ring when a call comes in on line 1? I cant seem to get it to work that way? |
| 19:40.19 | ZPertee | TheSov, your dialplan looks fine then. I have never put my actual number in there for a POTS line but it is probably fine. your dialplan looks good. In my experience as long as you configue your fxo port is configured with fxs_ks signalling, pointed to the right context, and using extension s |
| 19:40.23 | [TK]D-Fender | riddlebox, 1st, * doesn't do SLA, second, its just presence, there is no "message" associated so NO, you can't. |
| 19:41.10 | riddlebox | [TK]D-Fender, what do you mean * doesnt do SLA? I have sla.conf setup and have it working on calls out, but not calls in |
| 19:41.39 | TheSov | im using sip to sip zpertee |
| 19:41.48 | TheSov | have no pots lines |
| 19:42.47 | riddlebox | [TK]D-Fender, btw I finally got a system installed(other than the one at my house :p ) |
| 19:42.48 | [TK]D-Fender | riddlebox, that is not SLA. That is a sad hack trying to fake being SLA |
| 19:43.59 | ZPertee | I thought you were using your cell phone...aparently I was day dreaming...sorry |
| 19:44.01 | riddlebox | [TK]D-Fender, well I am talking to the customer and trying to get them to not use SLA anymore, I told them to try it for a week and see if they would rather just press a 9 to dial out |
| 19:44.39 | [TK]D-Fender | riddlebox, Who needs 9 to dial out? NONE of my setups do. |
| 19:44.44 | ZPertee | i give up |
| 19:44.58 | riddlebox | [TK]D-Fender, true |
| 19:45.51 | [TK]D-Fender | ZPertee, Yes he's using his cell-phone to call in for testing. YOU were the one assuming he was calling in on a Zap channel. |
| 19:46.18 | riddlebox | but what happens if you dial, say 1800xxxxxxx, and you have an extension 180 |
| 19:47.19 | ZPertee | [TK]D-Fender, got it...finally...guess I should be sleeping instead of ircing |
| 19:47.22 | [TK]D-Fender | ZPertee, and he said : <TheSov> i have a sip/sip system here |
| 19:47.32 | [TK]D-Fender | ZPertee, <TheSov> no pots lines |
| 19:47.46 | [TK]D-Fender | ZPertee, Possibly :) |
| 19:47.52 | ZPertee | I GET THE IDEA SORRY |
| 19:48.26 | [TK]D-Fender | TheSov, If things aren't working, pastebin your dialplan, your sip.conf including your "register" line to your itsp masking only passwords. |
| 19:50.26 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 19:50.56 | madduck | so we have asterisk with a few voip handsets here and i can't tell those to call sip addresses. |
| 19:51.19 | madduck | do i have to make extensions for them with asterisk or is there some commonly accepted way to dial sip addresses with numeric keypads? |
| 19:51.25 | [TK]D-Fender | madduck, Who dials URI's from phones directly? |
| 19:52.20 | madduck | e.g. i would like to call a friend who just gave me her sip address and i'd prefer not to do this while sitting in front of a screen and screaming into some microphone. |
| 19:52.59 | [TK]D-Fender | madduck, Set up * to dial her SIP address then. |
| 19:53.19 | madduck | in extensions.conf? |
| 19:53.22 | [TK]D-Fender | madduck, * does not let you jsut enter URI's targeting it. |
| 19:53.25 | [TK]D-Fender | madduck, yes |
| 19:53.43 | madduck | what if i had four friends with sip addresses? |
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| 19:54.00 | [TK]D-Fender | madduck, Dial(SIP/jane@herproxywithfulldomain-etc) |
| 19:54.13 | [TK]D-Fender | madduck, then you'd add 4 extens for them |
| 19:54.18 | riddlebox | madduck, you can setup "extensions" which when you call that extension it calls the sip uri |
| 19:54.27 | madduck | right, i know that. |
| 19:54.38 | madduck | i was hoping there was some way to do so without having to change asterisk every time |
| 19:54.47 | madduck | a new contact comes in |
| 19:56.28 | madduck | hm. |
| 19:56.29 | madduck | Got SIP response 484 "Address Incomplete" back from 202.78.240.48 |
| 19:57.12 | riddlebox | madduck, what kind of VoIP handset is it? |
| 19:57.22 | madduck | siemens c450ip |
| 19:57.33 | madduck | with an uip address and logged into my local asterisk |
| 19:57.36 | riddlebox | and that cannot do a direct IP call? |
| 19:57.40 | madduck | s/uip/ip/ |
| 19:57.53 | madduck | wow. :) |
| 19:58.11 | madduck | s/o/a/ |
| 19:58.15 | madduck | rofl |
| 19:58.29 | madduck | riddlebox: i can tell it about one provider only |
| 19:58.43 | madduck | i cannot seem to "dial" characters |
| 19:59.18 | [TK]D-Fender | madduck, You are not following. * is not a SIP proxy. Youc an't dial a URI TOWARDS it. |
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| 19:59.35 | [TK]D-Fender | madduck, You make a NORMAL numbered extension which will acuse * to dial your URI outwards. |
| 20:00.07 | [TK]D-Fender | madduck, And if you don't want to change your dialplan to add URI's then you're going to have to make some really funky scripts to originate calls. |
| 20:01.06 | madduck | [TK]D-Fender: well, all i did was "exten => *1,1,Dial(SIP/7218@voip..." |
| 20:01.18 | [TK]D-Fender | madduck, thats fine. |
| 20:01.26 | madduck | yeah, but that does not work... |
| 20:01.55 | [TK]D-Fender | madduck, pastebin is your friend.... |
| 20:02.12 | madduck | you mean the debug output? |
| 20:03.57 | madduck | [TK]D-Fender: http://rafb.net/p/UQCmoG53.txt |
| 20:04.44 | madduck | well, a softphone also tells me "484 Address Incomplete" |
| 20:05.08 | [TK]D-Fender | madduck, They don't like "SIP/7218@voip...." that you're dialing. |
| 20:05.17 | ZPertee | TheSov, not everyone here is as severely lacking on sleep as I am :-) so if you need more help just ask the rest |
| 20:05.54 | madduck | [TK]D-Fender: okay, i'll bother them. thanks! |
| 20:06.01 | [TK]D-Fender | madduck, np |
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| 20:39.49 | metabsd | hi |
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| 21:52.08 | TheSov | ok so ive tried for an hour and am still unsuccessfull at getting the waitexten to register more than 1 dtmf singal |
| 21:52.12 | TheSov | whats goin on |
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| 22:11.09 | ZPertee | how do I get asterisk to dial a channel and then playback an audio file? I can get it to dial, the call is answered, but no playback of audio file. it hangs after the dial |
| 22:13.25 | jameswf-home | sounds like a bad sound file |
| 22:17.08 | delparnel | When my extension is unavailable, the call gets forwarded to my cell phone. Is there a way that I can differentiate between calls coming from the asterisk, and someone calling my cell number directly? I know you can add CID Name prefixes, but my cell phone only picks up CID number, not the name. |
| 22:18.01 | ZPertee | jameswf-home, so this http://pastebin.ca/953232 should work? |
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| 22:55.22 | riddlebox | [TK]D-Fender, you around still? |
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| 23:50.19 | TheSov | can anyone tell me why WaitExten(x) is only picking up the first DTMF tone that i dial? |
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| 23:50.46 | UnixDog | (X>) |
| 23:50.54 | UnixDog | dont know |
| 23:51.24 | TheSov | whoa is me |