| 00:03.47 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 00:05.39 | *** join/#asterisk greenjenny (n=greenjen@pool-151-200-242-33.res.east.verizon.net) |
| 00:07.05 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 00:08.20 | greenjenny | Can anyone give me an idea of what kind of hardware I need in order to convert between SIP telephones in my office and incoming/outgoing calls on our T1? |
| 00:08.38 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-90-103.w86-206.abo.wanadoo.fr) |
| 00:08.43 | JT | an asterisk server with a PRI card can connect to a T1 |
| 00:08.45 | greenjenny | I'm getting tripped up on this whole transcoding thing |
| 00:08.51 | greenjenny | ah yeah, sorry. |
| 00:08.55 | greenjenny | I have a T1 card |
| 00:09.02 | JT | alternatively you can use a SIP PRI gateway, but they're expensive |
| 00:09.03 | JT | ah ok |
| 00:09.05 | greenjenny | But do I need any sort of DSP conversion? |
| 00:09.20 | greenjenny | or is that done in the T1 card? |
| 00:09.30 | Cyber-Dogg | so... when I install asterisk... does it automatically install the zaptel drivers? |
| 00:11.34 | UKCoder | urgh.... when I look at the messages on the wire using wireshark I see "c=IN IP4 147.135.12.250" but * debug shows it as "c=IN IP4 147.135.12.128" (128 is the host sending the SIP INVITE) |
| 00:11.50 | JT | greenjenny: you only need the card... |
| 00:12.08 | UKCoder | Is there a bug in 1.4.1 that anyone knows of around SIP INVITE/RTP connection setup?\ |
| 00:13.15 | greenjenny | JT: that sounds great. |
| 00:14.46 | greenjenny | JT: so, what? the DSP is done in the CPU? |
| 00:14.53 | greenjenny | JT: that sounds pretty load heavy |
| 00:15.30 | JT | greenjenny: there is no dsp work if you're not transcoding |
| 00:15.38 | JT | modern CPUs are powerful if you are |
| 00:15.51 | greenjenny | from SIP to T1 isn't transcoding? |
| 00:16.01 | greenjenny | I was confused then |
| 00:16.18 | greenjenny | And more than happy to be set right! :) |
| 00:18.51 | boch | i think you wont need transcoding if you use ulaw or alaw for you SIP calls |
| 00:19.06 | greenjenny | boch: hey, neat! |
| 00:19.19 | tzanger | [TK]D-Fender: around? |
| 00:19.30 | tzanger | I forget, is it possible to have parking slots seen in sidecars? |
| 00:19.44 | tzanger | i.e. have 701/702/703 as hints that show up on a polycom series of line buttons? |
| 00:21.28 | JT | greenjenny: it is not transcoding if you use the same codec |
| 00:21.47 | JT | PRIs use g.711 |
| 00:21.53 | JT | ulaw or alaw, depending on country |
| 00:22.10 | greenjenny | JT: so if I use ulaw and g.711 on SIP, then I'm gold? |
| 00:22.33 | carrar | GOLDEN! |
| 00:22.46 | JT | greenjenny: depends on the country, although transcoding between the u and a versions is almost a non CPU hit |
| 00:23.05 | greenjenny | JT: yeah, ulaw in Washington DC anyways :) |
| 00:24.17 | JT | usa uses Mu-law/"u-law" |
| 00:24.23 | greenjenny | yeah |
| 00:24.32 | greenjenny | that's how I meant |
| 00:24.45 | greenjenny | JT: you have saved me an inordinate amount of time, thank you! |
| 00:25.43 | JT | transcoding is usually a transparent operation |
| 00:25.47 | Zipper_32 | tzanger: Do you have any materials on how to get the sidecars setup with asterisk? I'm trying to get basic extensions to show up on the sidecar. |
| 00:25.54 | JT | you don't need to worry about it unless you need to specially setup the codec |
| 00:26.08 | tzanger | Zipper_32: I've not done it on my own, but generally you just enable more buddies and watch them |
| 00:26.57 | Zipper_32 | okay, |
| 00:26.59 | carrar | zipper, do you have the admin guide for the phone? |
| 00:27.39 | carrar | set it up to ftp down it's config |
| 00:27.39 | robin_sz | so .. if I put somethng like : |
| 00:27.45 | robin_sz | exten => 5102,1,Dial(SIP/home,5,t,M(xrms)) |
| 00:28.06 | robin_sz | and then create an xrms macro that calls AGI ... |
| 00:28.14 | Zipper_32 | carrar: I have the configs coming off a TFTP Server, yes. |
| 00:28.24 | Zipper_32 | I do have the admin guide as well, |
| 00:28.24 | carrar | and the guide? |
| 00:28.30 | Zipper_32 | "SoundPoint IP-SoundStation IP Administrators Guide, Version 2.0.x.pdf" |
| 00:28.30 | carrar | it's in there |
| 00:28.44 | robin_sz | hte agi should get pased the channel vars when/if the phoen is picked up, right? |
| 00:28.52 | Zipper_32 | What about Asterisk customizations? I just heard about setting up 'hints' the other day. |
| 00:29.10 | carrar | yeah hints work great |
| 00:29.13 | carrar | use 1.4.4 |
| 00:29.29 | Zipper_32 | Alright then. I'll install it tonight. |
| 00:29.31 | Zipper_32 | Thanks. |
| 00:29.59 | carrar | You will need to enable "hints" |
| 00:30.04 | carrar | on the phone |
| 00:30.10 | carrar | let me look what that option is |
| 00:30.10 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:30.17 | robin_sz | I just dont think this AGI is getting called :( |
| 00:31.21 | robin_sz | I seem to have two AGI dirs, /var/lib/asterisk/agi-bin/ and/usr/share/asterisk/agi-bin/ |
| 00:31.34 | robin_sz | how can I tell whic one is in use? |
| 00:32.20 | carrar | <PROTECTED> |
| 00:32.21 | carrar | feature feature.1.name="presence" feature.1.enabled="1" |
| 00:32.34 | Zipper_32 | carrar: Thank you! |
| 00:32.37 | carrar | in order for the bw to work |
| 00:32.51 | Zipper_32 | Thank you very much. =) |
| 00:34.16 | boch | hi, anyone knows what happens with the MYSQL connID when the party hangs while runing a MYSQL() query ? |
| 00:35.17 | carrar | robin |
| 00:35.22 | carrar | try: exten => 5102,1,Dial(SIP/home,5,tM(xrms)) |
| 00:36.10 | carrar | boch, it should close it |
| 00:37.05 | robin_sz | carrar, thanks |
| 00:43.42 | boch | carrar, i also think it sould close it, today my mysqld crashed with "too many connections" or something like that |
| 00:46.49 | carrar | Asterisk the only thing accessing your db? |
| 00:47.09 | boch | yes |
| 00:47.09 | carrar | Probably should move whatever you are doing to a AGI |
| 00:47.29 | boch | i guess |
| 00:47.40 | carrar | MYSQL command is too limited |
| 00:48.29 | boch | is there a performance difference between AEL+MYSQL or extensions + agi (binary) ? |
| 00:48.48 | carrar | I tossed mysql and went with Postgres and agi's |
| 00:49.05 | carrar | must better in my view |
| 00:49.36 | carrar | I have not used AEL w/mysql |
| 00:50.18 | carrar | I use perl for agi |
| 00:51.41 | boch | have you tried c for agi ? |
| 00:51.50 | carrar | no |
| 00:52.01 | carrar | litter harder to debug |
| 00:52.13 | carrar | as you can't really make changes on the fly |
| 00:52.33 | boch | should i notice the difference between c and perl in agi ? |
| 00:52.42 | carrar | I would think so |
| 00:53.02 | carrar | assuming your C is not waiting on anything for data |
| 00:53.45 | boch | on anything? |
| 00:54.15 | carrar | Just cause you write something in C doesn't mean it's going to be faster if what is slowing it down is a database that it is making a call too |
| 00:54.38 | boch | ahh right, clear |
| 00:54.58 | boch | it will use the same db, so its the same |
| 00:55.08 | boch | thanks carrar |
| 00:55.34 | ptiggerdine_ | compiled C should be faster than perl until the point of requestion stuff from a db.. |
| 00:55.59 | ptiggerdine_ | assuming you've coded well. |
| 00:56.24 | carrar | So no calling perl to suck data out of a database within your C program? :) |
| 00:57.24 | tzanger | hmm |
| 00:57.30 | boch | good point |
| 00:57.50 | tzanger | in the wiki, it says that the DYNAMIC_FEATURES var needs to be set in order to take advantage of the 'wW' Dial flags |
| 00:57.52 | tzanger | why is that? |
| 01:00.57 | carrar | You can put it in your globals |
| 01:02.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 01:28.10 | fall0ut | DCC SEND asdfasdflkasjdf;lkjasdf |
| 01:28.23 | fall0ut | haha |
| 01:28.37 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
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| 02:00.14 | Cyber-Dogg | I'm trying to install asterisk on freebsd |
| 02:00.29 | Cyber-Dogg | everytime I do a make install I keep getting errors on newt |
| 02:00.46 | Cyber-Dogg | it says "shared library newt.52 does not exist" |
| 02:00.56 | Cyber-Dogg | but I can't find newt .52... only .52 |
| 02:01.00 | Cyber-Dogg | err. .51 |
| 02:03.54 | *** join/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 02:05.57 | bkw__ | hrm where is mog |
| 02:06.18 | bkw__ | Cyber-Dogg, don't think much effort is put into anything but linux |
| 02:06.36 | bkw__ | Linux is the target platform for Asterisk |
| 02:08.18 | russellb | plenty of people use it on FreeBSD ... |
| 02:08.18 | Cyber-Dogg | that's what I thought russellb |
| 02:08.18 | h3x | bkww |
| 02:08.19 | h3x | w |
| 02:08.19 | russellb | you could do ... ./configure --without-newt |
| 02:08.25 | h3x | dude is there anything crazy about app_start_moh that it isnt used in app_conference |
| 02:08.44 | Cyber-Dogg | how do I do that? |
| 02:08.44 | h3x | sorry, ast_moh_start |
| 02:08.49 | crimethinker | buh? |
| 02:08.50 | Cyber-Dogg | make install ./configure --without-newt |
| 02:09.44 | russellb | no, you would run the configure script with that option before running make and make install |
| 02:09.47 | russellb | note that this only applies to 1.4. |
| 02:11.25 | bkw__ | russellb, still doesn't negate the fact that digium only targets linux |
| 02:11.38 | bkw__ | and I must admit that FreeBSD has really messed things up |
| 02:11.47 | Cyber-Dogg | russellb: I'll give it a shot that way |
| 02:11.55 | bkw__ | its hard to compile correctly on it.. things moving around and messing with you from version to version |
| 02:12.01 | Corydon76-home | The community targets a number of platforms |
| 02:12.14 | bkw__ | Corydon76-home, but the Offically supported platform is Linux |
| 02:12.30 | Cyber-Dogg | bkw__: I have to get this running on linux and BSD for a project I'm doing for school |
| 02:12.38 | Corydon76-home | Sure, for business edition |
| 02:12.39 | bkw__ | Cyber-Dogg, its possible for sure |
| 02:12.46 | h3x | hey |
| 02:12.48 | Corydon76-home | but that's a whole other ball of wax |
| 02:12.57 | bkw__ | yep |
| 02:13.11 | bkw__ | but really anything out side of linux is community supported last I seen. |
| 02:13.13 | Corydon76-home | We don't support business edition in here |
| 02:13.41 | h3x | we should get people to send us $995 for OPEN SOURCE |
| 02:14.30 | Corydon76-home | h3x: you're not paying money for the source. You're paying for level one support on a business priority |
| 02:14.38 | Cyber-Dogg | russellb: I tried that... still looks for newt |
| 02:15.53 | russellb | well i don't know why it would .. and i'm too tired to look right now. |
| 02:18.50 | *** join/#asterisk kafnir (n=kafnir@c-76-18-12-243.hsd1.fl.comcast.net) |
| 02:19.38 | kafnir | hello |
| 02:20.41 | kafnir | I am new to asterisk,so can someone help me solve this error |
| 02:20.44 | kafnir | WARNING[6111]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. |
| 02:21.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:22.32 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:24.11 | *** join/#asterisk Fieldy (i=kzjxi543@gentoo/contributor/Fieldy) |
| 02:24.17 | shmaltz | what variables are available when using mixmonitor? |
| 02:29.53 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 02:31.44 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:36.07 | shmaltz | I want to get something along these lines and it's not working: |
| 02:36.08 | shmaltz | MixMonitor(test.wav,W(4)b,/usr/bin/echo "Call From ${CALLERID(all)} to ${CDR(dst)} Recorded at ${CDR(start)} |
| 02:36.10 | shmaltz | Duration was ${CDR(billsec)} > /root/call${CDR(dst)}${CDR(end)}") |
| 02:36.11 | shmaltz | why? |
| 02:37.32 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:40.01 | h3x | wtf |
| 02:40.29 | shmaltz | h3x, yes? |
| 02:41.08 | h3x | asterisk doesn't evaluate asterisk variables in a System call does it ? |
| 02:41.19 | h3x | maybe im oldschool |
| 02:41.40 | h3x | dude just use the CDRs |
| 02:56.45 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 02:57.26 | iBuMp | good evening everyone.. Question.. On a fedora box, is it better to install asterisk from RPM or just tarball/compile it?? |
| 02:59.02 | *** part/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 03:02.23 | *** join/#asterisk luckyone (n=jordan@CPE-65-28-7-102.kc.res.rr.com) |
| 03:03.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:05.01 | luckyone | I am having trouble connecting to my running asterisk process, can anyone help me? |
| 03:05.15 | iBuMp | luckyone have yoiu tried asterisk -vvvvvcr |
| 03:05.21 | luckyone | trying with this: asterisk -rv -U asterisk -G asterisk |
| 03:05.22 | iBuMp | at the console.. |
| 03:05.49 | luckyone | I am running that as root |
| 03:06.13 | iBuMp | what os? |
| 03:06.19 | luckyone | it says that it is unable to connect to remote asterisk |
| 03:06.33 | iBuMp | you sure service is running |
| 03:06.39 | luckyone | iBuMp: yes, on ubuntu |
| 03:06.54 | *** join/#asterisk Un1x_Laptop (i=Sean@72.53.146.162) |
| 03:07.02 | luckyone | iBuMp: kubuntu actually |
| 03:07.06 | iBuMp | so you tarballed it? |
| 03:08.05 | luckyone | iBuMp: I downloaded a tarballed, untarred, configured, make && make installed it |
| 03:08.37 | iBuMp | have you stopped/started * |
| 03:08.38 | luckyone | iBuMp: it works, I have tested it called it, etc |
| 03:08.42 | iBuMp | ah |
| 03:08.57 | luckyone | iBuMp: I just can't reconnect to my terminal |
| 03:09.11 | luckyone | ps -ef | grep asterisk shows it running |
| 03:09.18 | iBuMp | hrmm i always use asterisk -vvvcr or more vvvvv for verbosity |
| 03:09.23 | luckyone | right |
| 03:09.27 | iBuMp | did you try putting the v before the r? |
| 03:09.32 | iBuMp | did that matter |
| 03:09.37 | luckyone | no, that doesn't matter |
| 03:10.22 | iBuMp | sorry i dont seem to have a clue.. |
| 03:10.39 | iBuMp | your not runnig any securities on the box are you |
| 03:11.20 | Rusty1 | luckyone: ps -aux |grep asterisk says? |
| 03:12.03 | *** join/#asterisk pepepedo (n=mavveric@OL135-98.fibertel.com.ar) |
| 03:12.08 | pepepedo | Hello |
| 03:12.25 | pepepedo | I have a problem with R2 on asterisk |
| 03:12.31 | pepepedo | may someone help me pls? |
| 03:17.54 | pepepedo | Hello! |
| 03:19.00 | pepepedo | smebody know why my R2 channel naver stop of send END OF ANI? |
| 03:19.25 | *** join/#asterisk pepepedo (n=mavveric@OL135-98.fibertel.com.ar) |
| 03:20.03 | killfill_ | hey |
| 03:20.21 | killfill_ | idefisk douns isnt goo when im talking and revicing an incomming cal (ringing) |
| 03:20.27 | killfill_ | erpp |
| 03:20.47 | killfill_ | idefisk isnt very good when im talking and revicing an incomming call (ringing) |
| 03:21.08 | JT | you didn't even fix all the errors :P |
| 03:21.21 | killfill_ | heh |
| 03:21.28 | killfill_ | .. sorry.. its the beer.. :P |
| 03:22.08 | pepepedo | can someone help me with MFC/R2 chan? |
| 03:22.09 | killfill_ | has jitter buffer something to do with it? |
| 03:22.16 | JT | pepepedo: unlikely |
| 03:22.28 | killfill_ | i wish to replace ten/sip with idefisk/iax2... |
| 03:22.36 | killfill_ | xten that is. |
| 03:23.15 | pepepedo | i dont know why |
| 03:23.23 | pepepedo | my R2 channel |
| 03:23.32 | pepepedo | never stop to send END OF ANI |
| 03:23.46 | pepepedo | MFC/R2 Chan 1: <- 5 on [2/ 40/Group I /End of ANI ] |
| 03:23.46 | pepepedo | MFC/R2 Chan 1: E off -> [2/ 40/Group I /End of ANI ] |
| 03:23.46 | pepepedo | MFC/R2 Chan 1: <- 5 off [2/ 40/Group I /End of ANI ] |
| 03:23.46 | pepepedo | MFC/R2 Chan 1: E on -> [2/ 40/Group I /End of ANI ] |
| 03:23.46 | pepepedo | MFC/R2 Chan 1: <- 5 on [2/ 40/Group I /End of ANI ] |
| 03:23.48 | JT | r2 isn't even part of standard asterisk |
| 03:23.54 | pepepedo | i know |
| 03:23.56 | JT | pepepedo: jebus christ, don't do that again! |
| 03:24.01 | pepepedo | but may be someone can help me |
| 03:24.16 | JT | maybe you shouldn't flood |
| 03:24.18 | JT | ~pb |
| 03:24.30 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:24.30 | pepepedo | sorry |
| 03:30.29 | findlay | jbot: pb is also at paste.lisp.org |
| 03:30.51 | jbot | findlay: okay |
| 03:30.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:32.39 | CrazyTux | exten => _auto-login-NXXNXXXXXX,1, whats wrong with that? why wont it match sip:auto-login-npanxxext@ |
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| 03:41.11 | boch | do you know why the callback function passed to Asterisk::AGI is not exec when user hangs ? |
| 03:41.11 | blitzrage | CrazyTux: because N and X match numbers, not letters |
| 03:41.25 | CrazyTux | blitzrage, how can I match something like tht? |
| 03:41.28 | CrazyTux | s/tht/that/ |
| 03:41.35 | iBuMp | good evening everyone.. Question.. On a fedora box, is it better to install asterisk from RPM or just tarball/compile it?? |
| 03:41.50 | blitzrage | _auto-login-.,1,NoOp() |
| 03:42.00 | blitzrage | or _auto-login!,1,NoOp() |
| 03:42.18 | *** part/#asterisk holiday42 (n=me@70-57-197-218.farg.qwest.net) |
| 03:42.41 | blitzrage | something like auto-login-4165551212 should match on that pattern you have though |
| 03:43.08 | blitzrage | if it doesn't, then you have something wrong with the pattern or the value that is being matched against the pattern |
| 03:43.31 | JT | Verbose(${EXTEN}) :) |
| 03:43.41 | *** join/#asterisk `Sean (i=Un1x@CPE000c258d147c-CM000a73a94167.cpe.net.cable.rogers.com) |
| 03:48.08 | CrazyTux | JT, Verbose, whats that do? |
| 03:48.20 | CrazyTux | JT, give information on extension supplied? |
| 03:50.36 | BSD_Tech | your all fired turn in your staplers and stick pads |
| 03:50.41 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 03:51.07 | CrazyTux | BSD_Tech, :) what part of Irvine are you in? |
| 03:51.26 | BSD_Tech | Irvine is the main hub for the DSL provider |
| 03:51.35 | CrazyTux | BSD_Tech, ah orange county rather I should say than. |
| 03:51.36 | BSD_Tech | I am in Mt Washington |
| 03:51.37 | *** join/#asterisk bmg505 (n=leon@196.209.181.175) |
| 03:51.48 | CrazyTux | BSD_Tech, Not familiar with that.... |
| 03:52.05 | BSD_Tech | Glendale/EagleRock |
| 03:52.09 | CrazyTux | ah |
| 03:52.40 | BSD_Tech | I brought my own stapler and stickypads there for I get to keep them |
| 03:52.51 | BSD_Tech | lol |
| 03:56.34 | JT | BSD_Tech: tells you what the exten variable is, which will be useful |
| 03:57.22 | BSD_Tech | ? |
| 03:57.27 | CrazyTux | BSD_Tech, was to me |
| 03:58.19 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:59.01 | JT | oops |
| 03:59.05 | JT | CrazyTux: yes to you |
| 03:59.12 | CrazyTux | JT, :) thanks. |
| 03:59.21 | JT | see what asterisk sees it as, then you might see why it isn't matching |
| 03:59.48 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 04:06.44 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 04:07.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:09.10 | blitzrage | night all |
| 04:11.37 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 04:11.42 | *** join/#asterisk BBHoss_Laptop (n=Hoss@adsl-230-0-201.hsv.bellsouth.net) |
| 04:12.56 | BBHoss_Laptop | anyone know if res_snmp can be used with 1.2 |
| 04:13.03 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 04:13.17 | iBuMp | asterisk 1.4 or 1.2? Whats better/more solid? |
| 04:15.25 | JT | 1.2 is more stable |
| 04:15.28 | BBHoss_Laptop | 1.2 probably has a bigger userbase |
| 04:15.35 | BBHoss_Laptop | 1.4 has more features |
| 04:18.25 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 04:22.27 | iBuMp | im having a crazy time with * 1.4.4 and a VPN'd company.. |
| 04:22.36 | iBuMp | cant find right codecs, bandwidth combo |
| 04:23.25 | BBHoss_Laptop | good luck with vpn period |
| 04:23.34 | iBuMp | DOH! |
| 04:23.40 | BBHoss_Laptop | too much jitter for any good |
| 04:23.43 | JT | eh, it's doable |
| 04:23.50 | JT | maybe on a dodgy vpn |
| 04:23.53 | BBHoss_Laptop | ilbc or gsm are best bets |
| 04:23.56 | iBuMp | i was wondering if I should just enable firewall ports and put DMZ into effect |
| 04:24.08 | iBuMp | i bought g729a |
| 04:24.11 | BBHoss_Laptop | what exactly are you trying to do my friend |
| 04:24.15 | iBuMp | doestn work worth a &&(*@ |
| 04:24.24 | BBHoss_Laptop | now now, patience |
| 04:24.29 | JT | BBHoss_Laptop: best not to make recommendations without checking the situation :) |
| 04:24.35 | iBuMp | just find the right codec that works with the wan and allows users to dial their voice mails |
| 04:24.46 | iBuMp | i mean i get them connected but then none of the signals are picked up |
| 04:24.56 | BBHoss_Laptop | dtmf? |
| 04:24.59 | iBuMp | and with the g729 it lasts for 20 secs then stops working |
| 04:25.04 | iBuMp | we have dtmf enabled |
| 04:25.19 | BBHoss_Laptop | i know, you mean by signals=dtmf signals? |
| 04:25.19 | iBuMp | i have installed both rpm versions and tarballed |
| 04:25.24 | iBuMp | yes sorry |
| 04:25.27 | BBHoss_Laptop | k |
| 04:25.50 | BBHoss_Laptop | can you see the digits being pressed in debug mode? |
| 04:25.51 | JT | iBuMp: what dtmf mode is set in sip.conf? |
| 04:25.58 | iBuMp | rfc |
| 04:26.30 | JT | hmm |
| 04:27.25 | iBuMp | i didnt know there was a debug mode.... O( |
| 04:27.34 | iBuMp | i thought console would show all |
| 04:27.40 | iBuMp | * -vvvvvcr |
| 04:27.46 | iBuMp | is waht i ran |
| 04:28.07 | BBHoss_Laptop | there is a dtmf debug command i think, cant remember what it is |
| 04:28.11 | BBHoss_Laptop | i can check |
| 04:28.21 | iBuMp | the g729 also makes a robotic bong sound when call is originated and the call is cutoff after 20 secs |
| 04:28.25 | *** join/#asterisk InHisName (n=Administ@c-68-80-56-212.hsd1.pa.comcast.net) |
| 04:28.34 | BBHoss_Laptop | wierd |
| 04:28.40 | BBHoss_Laptop | what type of vpn |
| 04:28.54 | iBuMp | ipsec |
| 04:29.43 | BBHoss_Laptop | i was having problems with an ipsec vpn |
| 04:29.51 | BBHoss_Laptop | i had to switch to dmz |
| 04:29.59 | BBHoss_Laptop | i believe my problem was jitter |
| 04:30.01 | BBHoss_Laptop | not sure |
| 04:30.06 | BBHoss_Laptop | never could find out |
| 04:30.11 | BBHoss_Laptop | but ill look into something |
| 04:30.12 | iBuMp | hrmm.. what if i put in firewall rules allow all traffic from IP to |
| 04:30.15 | iBuMp | i got it.. |
| 04:30.26 | JT | udp vpn i hope |
| 04:30.28 | iBuMp | i will jsut create allow all rules in each VPN for eath other |
| 04:30.52 | iBuMp | even though the vpn must allow all traffic tunneled through |
| 04:31.37 | BBHoss_Laptop | if it goes through the vpn at all, it will do the same thing |
| 04:32.48 | JT | if you're getting audio just fine, dtmf should work in rfc2833 mode |
| 04:32.55 | JT | if it doesn't, it's usually an endpoint problem |
| 04:35.50 | iBuMp | all grandstream phones |
| 04:36.12 | JT | they're a problem in a phone lookalike box ;) |
| 04:36.22 | iBuMp | heh |
| 04:36.27 | ELBunce | Any iaxclient devs up? |
| 04:37.42 | BBHoss_Laptop | lol |
| 04:39.35 | BBHoss_Laptop | ibump: do tail -n 100 /var/log/asterisk/full |
| 04:39.41 | BBHoss_Laptop | look for recieved dtmf digit |
| 04:39.52 | iBuMp | ok give me a sec.. thanks,, |
| 04:39.53 | BBHoss_Laptop | tail it after you try a call |
| 04:40.01 | BBHoss_Laptop | see if it picks the numbers up |
| 04:41.11 | BBHoss_Laptop | i believe it should show the digits |
| 04:43.26 | snuffy22 | yes it should long as in your logger.conf full has 'dtmf' |
| 04:43.38 | *** join/#asterisk clever[rev] (n=clever@fctnnbsc16w-156034215154.nb.aliant.net) |
| 04:46.59 | Corydon76-home | Heh, it's amusing how many people refer to DTMF logging without quite understanding what that does |
| 04:51.53 | clever[rev] | im guessing that logs the numbers pressed on phones?:P |
| 04:52.15 | Corydon76-home | Nope |
| 04:52.25 | clever[rev] | whats it do then? |
| 04:52.41 | Corydon76-home | It logs DTMF digits which are passed through a non-native bridge |
| 04:52.41 | BSD_Tech | it logs the tones for playback ? |
| 04:52.50 | clever[rev] | ahhh |
| 04:53.02 | BBHoss_Laptop | where are all dtmf digits then |
| 04:53.11 | Corydon76-home | We don't log them |
| 04:53.25 | clever[rev] | are there any cheap/easy to make devices for connecting a pc to a phone line |
| 04:53.34 | clever[rev] | such as going thru a sound card to get most of it done? |
| 04:53.39 | BSD_Tech | its called modem |
| 04:54.05 | clever[rev] | [28 23:45:30] <clever[rev]> i have several 'voice modems' |
| 04:54.05 | clever[rev] | [28 23:45:35] <clever[rev]> would those be of any use?:P |
| 04:54.06 | clever[rev] | [28 23:45:52] <russellb> nope |
| 04:54.06 | BBHoss_Laptop | so theres no way to see dtmf digits pressed through a sip or iax2 trunk |
| 04:54.26 | BSD_Tech | you can |
| 04:54.26 | clever[rev] | BSD_Tech: hmmm voice modems wont work then:P enless he's wrong |
| 04:54.45 | Corydon76-home | Some voice modems will work |
| 04:54.48 | clever[rev] | ive never even gotten a voice modem to work under winblows yet though and i gave up ages ago when we stoped using ialup |
| 04:55.01 | Corydon76-home | but they'll need to be full-duplex, which is not a common feature on modems |
| 04:55.12 | clever[rev] | id think all you realy need is a audio in/out like a sound card and some way to pick it up/off the hook |
| 04:55.22 | clever[rev] | and some way to mix the sounds between the 2 lines |
| 04:55.31 | *** join/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 04:55.36 | BSD_Tech | oom made a great full duplex modem |
| 04:55.42 | BSD_Tech | zoom even |
| 04:55.44 | BSD_Tech | lol |
| 04:55.59 | BSD_Tech | typing with a fractured finger is so fun |
| 04:56.01 | [TK]D-Fender | clever[rev], http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 04:56.04 | snuffy22 | i don't claim to know anything other than if it shows digits when i push then dtmf working |
| 04:56.12 | [TK]D-Fender | clever[rev], Go read, then come back. |
| 04:56.12 | clever[rev] | BSD_Tech: lol:P |
| 04:56.22 | BSD_Tech | you can us a lcd read out |
| 04:56.22 | pabs3 | does #include work in all asterisk configs? |
| 04:56.31 | BSD_Tech | yes |
| 04:56.46 | BSD_Tech | to include external config diles |
| 04:56.51 | BSD_Tech | fiels |
| 04:56.56 | BSD_Tech | files even |
| 04:57.28 | BSD_Tech | #include = filename.conf |
| 04:57.39 | Corydon76-home | s/=// |
| 04:57.48 | BSD_Tech | ? |
| 04:57.57 | Corydon76-home | #include "filename.conf" |
| 04:58.01 | Corydon76-home | No = |
| 04:58.01 | BSD_Tech | ok |
| 04:58.07 | BSD_Tech | thats tight |
| 04:58.12 | BSD_Tech | brain fart |
| 04:58.49 | BSD_Tech | smelly one to |
| 04:58.49 | Keltus | what kind of service provider am I looking for, if I want to redirect my toll free number to an asterisk box that I own? |
| 04:58.50 | Keltus | is it toll free DID? |
| 04:59.07 | BSD_Tech | port it to a sip or iax provider |
| 04:59.21 | BSD_Tech | like teliax |
| 04:59.31 | BSD_Tech | only name I can think of off my head |
| 04:59.42 | pabs3 | hmm, voicemail.conf says I shouldn't use #include in it, is that still the case? |
| 04:59.52 | Keltus | BSD_Tech: what's the service I'm looking for actually called? the technical name |
| 04:59.57 | BSD_Tech | what are you trying to do |
| 05:00.10 | Keltus | have a toll free number hit my asterisk box |
| 05:00.18 | BBHoss_Laptop | origination |
| 05:00.19 | Corydon76-home | Not if you want people to be able to change their passwords, no |
| 05:00.23 | BSD_Tech | a sip of iax2 provider |
| 05:00.45 | BSD_Tech | and you would have to port your number |
| 05:00.50 | BBHoss_Laptop | keltus:origination |
| 05:00.51 | Keltus | yeah I can do that |
| 05:00.54 | Keltus | is it just called "origination"? |
| 05:00.56 | BSD_Tech | wich might cost a little money |
| 05:01.06 | Keltus | I don't see any company selling origination |
| 05:01.10 | BSD_Tech | ? |
| 05:01.12 | BBHoss_Laptop | ha |
| 05:01.18 | BBHoss_Laptop | its a technical term |
| 05:01.25 | Keltus | then what do I look for |
| 05:01.25 | BSD_Tech | Broadvoice teliax voicepulse |
| 05:01.32 | BBHoss_Laptop | voipstreet is what i use |
| 05:01.37 | BBHoss_Laptop | they do 1800 |
| 05:01.48 | Keltus | okay, and what would be the cheapest rate? |
| 05:01.56 | Keltus | I've seen 2c/minute but that was for toll free DID |
| 05:02.04 | Keltus | and I guess I'm looking for "toll free origination" |
| 05:02.12 | BBHoss_Laptop | you have to balance quality and price |
| 05:02.40 | BBHoss_Laptop | what kind of volume do you do |
| 05:02.45 | Keltus | okay, I want good quality but good pricing |
| 05:02.51 | *** part/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 05:02.53 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 05:02.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:02.55 | Keltus | I guess everyone does.. |
| 05:03.11 | Keltus | say, 100 minutes a day |
| 05:03.19 | jql | I'll take fast, cheap, *and* good |
| 05:04.12 | BBHoss_Laptop | 2.9 c/min with voipstreet |
| 05:04.21 | BBHoss_Laptop | plus $2.95 a month |
| 05:04.27 | BBHoss_Laptop | $20 porting charge |
| 05:04.32 | Keltus | hmmm... okay |
| 05:04.36 | BBHoss_Laptop | the porting charge is a onetime fee |
| 05:04.40 | Keltus | right |
| 05:04.51 | BBHoss_Laptop | but that dosent include canada |
| 05:05.06 | BBHoss_Laptop | for canada support its 4.9 c/min |
| 05:05.15 | Keltus | nope I just need US |
| 05:05.23 | BBHoss_Laptop | k |
| 05:05.42 | BBHoss_Laptop | they've just been acquired by a larger company |
| 05:05.49 | BBHoss_Laptop | dont know if its bad or good |
| 05:05.57 | BBHoss_Laptop | thier support it good though |
| 05:06.24 | BBHoss_Laptop | if you want, you can test latency] |
| 05:06.29 | *** join/#asterisk ploieel (n=manni@Fb251.f.ppp-pool.de) |
| 05:06.32 | BBHoss_Laptop | ping chiv1.voipstreet.com |
| 05:07.00 | h3x | les.net is good for canada |
| 05:07.10 | h3x | hes using the old Group Telecom network |
| 05:07.30 | Keltus | I don't need canada at all |
| 05:07.37 | h3x | i know :P |
| 05:07.42 | BSD_Tech | night all |
| 05:07.45 | Keltus | how can I tell how the quality will be? |
| 05:07.53 | iBuMp | night.. im finishing an SVN install |
| 05:07.55 | BBHoss_Laptop | free 20 min |
| 05:07.58 | iBuMp | this is great! |
| 05:08.00 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 05:08.10 | Keltus | right now, I'm using get1800.com and it redirects to an ipkall number |
| 05:08.17 | Keltus | which then hits my asterisk server |
| 05:08.21 | Keltus | and the quality is BAD |
| 05:08.35 | Keltus | it sounds like there are dropped packets |
| 05:08.36 | BBHoss_Laptop | latency will kill you with that |
| 05:08.46 | BBHoss_Laptop | most likeyl jitter |
| 05:08.47 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 05:08.53 | Keltus | what's jitter? |
| 05:08.55 | h3x | what, you used switched toll free? |
| 05:09.10 | Keltus | huh? |
| 05:09.23 | BBHoss_Laptop | jitter is the variability of latency |
| 05:09.28 | BBHoss_Laptop | how stable your latency is |
| 05:09.29 | Keltus | oh |
| 05:09.30 | [pyro] | hi guys, i cant find any info on aastra phones & asterisk. Does anyone know of any urls to setup info? |
| 05:09.32 | Keltus | no idea |
| 05:09.40 | h3x | voip-info.org |
| 05:09.42 | Keltus | it's just a test setup for now |
| 05:09.49 | h3x | that reminds me i need to get my 57i working |
| 05:09.53 | h3x | i couldnt even get it to call asterisk at all |
| 05:09.56 | BBHoss_Laptop | nice |
| 05:09.59 | BBHoss_Laptop | hmm |
| 05:10.05 | BBHoss_Laptop | was thinking of getting one |
| 05:10.16 | h3x | ill bet ya that NAT wont work on one at all |
| 05:10.20 | BBHoss_Laptop | i bet that BS 300000 sq ft range |
| 05:10.27 | h3x | its got this proprietary Nortel NAT option |
| 05:10.51 | jql | I have an aastra, but I don't have remote provisioning setup with it |
| 05:11.05 | BBHoss_Laptop | anyone used the wireless |
| 05:11.11 | jql | it works well. |
| 05:11.32 | BBHoss_Laptop | what kind of range |
| 05:11.45 | BBHoss_Laptop | only thing that works for me right now is 900mhz |
| 05:11.49 | jql | dunno. My boss has a so-cal mansion and it works anywhere in there |
| 05:11.51 | BBHoss_Laptop | and barely at that |
| 05:11.53 | jql | heh |
| 05:11.58 | h3x | http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra+Phones |
| 05:12.08 | BBHoss_Laptop | i've tried DECT too |
| 05:12.16 | BBHoss_Laptop | same bullshit |
| 05:12.28 | BBHoss_Laptop | 900mhz gives like 20ft more range |
| 05:12.55 | BBHoss_Laptop | anybody know where i can get DECT that does SIP |
| 05:13.01 | BBHoss_Laptop | that ships to USA? |
| 05:13.06 | h3x | Linksys |
| 05:13.10 | jql | I have a bookmark for a phone that was scheduled to be released... |
| 05:13.12 | h3x | oh DECT |
| 05:13.19 | h3x | SIP would be over 802.11 |
| 05:13.24 | BBHoss_Laptop | no |
| 05:13.25 | JT | wifi wireless phones are useless |
| 05:13.28 | h3x | you mean DECT to the base |
| 05:13.28 | BBHoss_Laptop | the base does sip |
| 05:13.30 | BBHoss_Laptop | yes |
| 05:13.30 | h3x | ok |
| 05:13.33 | JT | h3x: no, not if the base station talks the sip |
| 05:13.38 | h3x | heh |
| 05:13.43 | h3x | why not use a 802.11 phone |
| 05:13.45 | BBHoss_Laptop | i think we got it now :) |
| 05:13.49 | BBHoss_Laptop | 2.4ghz |
| 05:13.51 | JT | because they're all absolute rubbish |
| 05:13.55 | BBHoss_Laptop | indeed |
| 05:14.10 | h3x | i know a whole hospital that uses them |
| 05:14.11 | h3x | and it works fine |
| 05:14.13 | BBHoss_Laptop | dect you can use repeaters, multiple bases and such |
| 05:14.15 | JT | and 802.11 is innapropriate for mobile voip |
| 05:14.17 | JT | lucky them |
| 05:14.23 | h3x | cisco APs |
| 05:14.28 | BBHoss_Laptop | so... |
| 05:14.41 | BBHoss_Laptop | wifi is for data |
| 05:14.44 | BBHoss_Laptop | not voice :) |
| 05:14.46 | h3x | so is VoIP |
| 05:14.47 | [TK]D-Fender | ~wifisip |
| 05:15.00 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 05:15.00 | h3x | IP |
| 05:15.00 | h3x | hehehe |
| 05:15.00 | [TK]D-Fender | ^^^^^^^ |
| 05:15.22 | BBHoss_Laptop | if the bot says it it must be true |
| 05:15.24 | h3x | use a damn cordless phone |
| 05:15.26 | h3x | on an ATA |
| 05:15.28 | h3x | EOF |
| 05:15.53 | BBHoss_Laptop | then it wont do handover |
| 05:16.05 | BBHoss_Laptop | i have a unique situation |
| 05:16.16 | BBHoss_Laptop | where i must have 5 different extensions |
| 05:16.24 | BBHoss_Laptop | AND have it cover the whole store |
| 05:16.24 | JT | h3x: exactly |
| 05:16.29 | JT | just don't use 802.11 |
| 05:16.32 | [pyro] | has anyone configured asterisk to work with the new Aastra 53/55/53i handsets? |
| 05:16.34 | h3x | cell phone |
| 05:16.39 | h3x | DONE! |
| 05:16.40 | JT | the jitter and packet loss kills it |
| 05:16.40 | h3x | hahahah |
| 05:16.53 | BBHoss_Laptop | seriously |
| 05:17.04 | BBHoss_Laptop | this is the one thing holding my whole rollout up |
| 05:17.09 | h3x | bluetooth it to a on-net call |
| 05:17.21 | h3x | that would be awesome |
| 05:17.40 | BBHoss_Laptop | that would be an "ugly hack" |
| 05:17.54 | h3x | [pyr: theres tons of asterisk setup info in the docs for those aastra phones but i havent had the patience to do it yet |
| 05:18.06 | jql | mmm... 55i is kinda sexy |
| 05:18.08 | h3x | what is up with my autocomplete |
| 05:18.14 | [pyro] | h3x: oh for the new handsets? |
| 05:18.17 | h3x | Yes |
| 05:18.21 | [pyro] | h3x cheers |
| 05:18.28 | h3x | And I had two guys from the VON show calling me every day |
| 05:18.30 | jql | much sexier than my 480s |
| 05:18.30 | h3x | about setting up the phone |
| 05:18.35 | h3x | from Aastra |
| 05:18.44 | h3x | but i was busy with toher crap |
| 05:18.55 | [TK]D-Fender | Aastra means well, but comes up lacking in several categories |
| 05:19.16 | h3x | like NAT support? heh |
| 05:19.34 | h3x | http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-CF2061A7/04/hs.xsl/19703.htm#dl_installation |
| 05:19.56 | [TK]D-Fender | the 5i series handsets have NO wieght and the speakerphone is inferior. Call handling doesn't hold a candle to Polycom, their new Pixel display is being driven by the same char-martrix engine as the 480i. |
| 05:20.24 | h3x | but the xml browser on it should be awesome |
| 05:20.29 | BBHoss_Laptop | the KIRK Telecom devices look good |
| 05:20.33 | BBHoss_Laptop | just cant find any |
| 05:21.21 | IOscanner | I can't seem to get asterisk to respond to 180 or 183 from carrier. I have progressinband=yes and removed r from dial string. Anyone have any other ideas what I can do? |
| 05:21.34 | jql | define respond |
| 05:21.38 | [pyro] | [TK]D-Fender: so you would recommend polycom over the new aastra 5i series? |
| 05:21.53 | jql | there is no response expected for 1xx |
| 05:22.01 | iBuMp | is using 3 digit compared to the default 4 a [problem? |
| 05:22.04 | IOscanner | well it is not passing the 180 or 183 to our callers. It closes the channel |
| 05:22.10 | [TK]D-Fender | [pyro], Yup. |
| 05:22.14 | [TK]D-Fender | Polycom > All |
| 05:22.19 | [pyro] | lol |
| 05:22.20 | IOscanner | just ignores them |
| 05:22.21 | h3x | i like the snom's |
| 05:22.23 | h3x | overall |
| 05:22.24 | jql | closing the channel is entirely the wrong thing for asterisk to be doing |
| 05:22.41 | BBHoss_Laptop | NEVER had ANY trouble out of p-coms |
| 05:22.49 | h3x | but i have an application where i need all these softkeys on the 57i |
| 05:23.05 | jql | snoms have even more softkeys. :) |
| 05:23.17 | IOscanner | The carrier sees it too. I am not sure why asterisk is not opening the RTP stream so I can hear the ring or TDM messages passed inband |
| 05:23.17 | BBHoss_Laptop | wonder if an 802.11a network would work |
| 05:23.18 | h3x | no those are BLF keys with bezels to mark |
| 05:23.28 | h3x | a soft key is next to a LCD :P |
| 05:23.39 | jql | yeah... |
| 05:23.44 | jql | the aastra gives you control of those? |
| 05:23.47 | h3x | yes |
| 05:23.54 | jql | interesting |
| 05:23.54 | h3x | and you can mess with everything from XML |
| 05:23.58 | h3x | and you can do XML push |
| 05:24.04 | h3x | its great for screen pops |
| 05:24.07 | IOscanner | Any ideas jql? |
| 05:24.13 | jql | like a cisco... but less bastard-step-childish? |
| 05:24.47 | h3x | right |
| 05:25.08 | BBHoss_Laptop | linksys does a dect skype |
| 05:25.11 | jql | IOscanner: You need logs. Lots of logs. (core) set verbose 4, set debug 4, sip debug, sip trace, etc... |
| 05:26.43 | BBHoss_Laptop | zyxel's v250 phone would work |
| 05:26.51 | BBHoss_Laptop | just cant buy them anywhere |
| 05:27.53 | Scrumps | evenin' |
| 05:28.00 | jql | I need to setup a voip-phone-laundering business out of the EU |
| 05:28.01 | pabs3 | Corydon76-home: does the same apply to sip.conf? |
| 05:28.32 | BBHoss_Laptop | jql:yeah |
| 05:29.35 | BBHoss_Laptop | anybody know taiwanese? |
| 05:32.40 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 05:35.11 | BBHoss_Laptop | http://www.netvox.com.tw/English/Html/V-108Cat.htm |
| 05:35.12 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 05:40.13 | *** join/#asterisk TomasuAway (n=moose@S0106000c765956b8.ed.shawcable.net) |
| 05:44.22 | BBHoss_Laptop | skype should be illegal |
| 05:46.15 | [TK]D-Fender | BBHoss_Laptop, And why is that? |
| 05:47.05 | BBHoss_Laptop | because of the people that use it |
| 05:48.15 | *** join/#asterisk mkl1525 (n=qwertz@pd9534421.dip0.t-ipconnect.de) |
| 05:48.16 | [TK]D-Fender | BBHoss_Laptop, ..... if I were to apply that method of thinking to my normal urge to give natural selection a "boost" this world would be a very empt place |
| 05:48.49 | BBHoss_Laptop | i know im just being an ass |
| 05:48.53 | BBHoss_Laptop | im frustrated |
| 05:49.21 | BBHoss_Laptop | there is a consumer solution, but no biz solution |
| 05:49.40 | mkl1525 | Hi, (* 1.2) have added a "Queue(all|20)" in my extensions.conf but the call stays in the queue although 20 seconds have already passed. Anything I missed to setup? |
| 05:49.49 | [TK]D-Fender | Skype is the bastard child of VoIP. Its for little kiddies. |
| 05:50.17 | [TK]D-Fender | mkl1525, yeah, a PILE of parms. timout is NOT the second parm. |
| 05:50.28 | [TK]D-Fender | mkl1525, "show application queue" |
| 05:50.30 | walhala | mk all|||20 |
| 05:50.53 | BBHoss_Laptop | damnit why cant i just go to voipsupply and buy a DECT base station, phones, and repeaters |
| 05:51.52 | JT | i thought voipsupply was evil |
| 05:52.37 | BBHoss_Laptop | anywhere |
| 05:52.39 | BBHoss_Laptop | ebay even |
| 05:53.18 | mkl1525 | [TK]D-Fender, thanks will try it |
| 05:53.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:53.38 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:54.59 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 05:55.03 | IOscanner | Okay here is a sip debug of the traffic I don't can't hear the RTP from the 183. http://paste.uni.cc/15873 |
| 05:55.26 | IOscanner | I have progressinband=yes and removed r from the outbound dial. |
| 05:56.09 | [pyro] | h3x: i greped the asterisk doc's for "aastra" and got nothing. Which docs did you say the setup info was in? |
| 05:56.15 | [TK]D-Fender | ok, bedtime.. I'm outta here. |
| 05:56.17 | [TK]D-Fender | later all |
| 05:56.28 | [pyro] | later [TK]D-Fender |
| 05:56.34 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 05:56.48 | IOscanner | line 229 I see the SIP/2.0 200 canceling |
| 05:57.13 | IOscanner | Why and how can I keep it from canceling |
| 05:57.26 | pabs3 | if a grandstream phone (GXP 2000) can't navigate a voicemail menu, what is likely to be the cause |
| 05:57.29 | pabs3 | ? |
| 05:57.39 | JT | dtmf problems |
| 05:57.50 | JT | doesn't seem unusual with grandstreams |
| 05:58.00 | *** join/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
| 05:58.03 | TomasuAway | I tried asking in #asterisknow, but have yet to recieve a respnse, so I'm wondering if its possibble to setup a LinksysPAP2 with asterisknow? I couldn't find anywhere to add anything like it. |
| 05:58.26 | JT | it's just a sip device |
| 05:58.35 | JT | the book tells you how to setup sip stuff |
| 05:58.35 | TomasuAway | right, and I didnt see anywhere to add it. |
| 05:58.58 | JT | ~thebook |
| 05:59.12 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:59.12 | JT | shrug |
| 05:59.12 | JT | not familiar with the gui |
| 05:59.13 | dudes | gui suck |
| 05:59.13 | TomasuAway | I had it setup via regular config files before a reinstall, but I wanted to try out the guis |
| 05:59.16 | IOscanner | Anyone see why I can't get asterisk to play the rtp stream from 183? |
| 05:59.37 | pabs3 | JT: so, probably an issue with the phone's configs? |
| 06:00.29 | JT | maybe |
| 06:00.39 | JT | they're not very good phones |
| 06:03.22 | IOscanner | Looks like the other end is canceling the stream. Am I reading the trace correct? |
| 06:04.05 | mightnare | is it possible for an agi to retrieve the sip response when dial ends? |
| 06:04.33 | pabs3 | fixed it by setting it to use SIP INFO for DTMF |
| 06:04.45 | mightnare | i see something like this "Got SIP response 486 "Busy Here" back from ..." on the cli, i was hoping that my agi can also retrieve this info |
| 06:04.58 | JT | pabs3: what was it on before? |
| 06:05.16 | JT | pabs3: and where did you change the setting, asterisk or the phone? |
| 06:05.49 | Siya | moin |
| 06:05.52 | pabs3 | JT: in-audio, and on the phone, in the account settings |
| 06:06.05 | mkl1525 | When going into queue * throws an error " File 20 does not exist in any format" so is there any way to get which file * wants to open? |
| 06:06.08 | JT | pabs3: what codec were you using? |
| 06:08.15 | pabs3 | JT: the phone says GSM during the call to the voicemail ext |
| 06:08.35 | JT | pabs3: you didn't seriously expect dtmf to work did you? |
| 06:08.50 | JT | inband dtmf only works with g.711 (ulaw or alaw) |
| 06:09.09 | pabs3 | had no idea, I'm new to all this stuff |
| 06:10.20 | JT | gsm is highly compressed |
| 06:10.27 | JT | designed to carry voice |
| 06:10.38 | JT | it doesn't reliably carry tones accurately |
| 06:11.38 | Siya | pabs3: dtmf is too much of a real sinus to be compressed ;) (once u=you know this you never forget) |
| 06:11.44 | Siya | -u= |
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| 06:19.18 | *** join/#asterisk poohbah431111 (n=rmillis@S010600183950760b.cg.shawcable.net) |
| 06:19.34 | poohbah431111 | Hello all. Anyone using Asterisk with Callcentric? |
| 06:20.35 | poohbah431111 | I have a problem where my inbound DID callers get their unavailable message instead of my PBX. Their FAQ says this may be due to the registration interval. How can I set this in asterisk? |
| 06:21.06 | *** part/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
| 06:21.31 | poohbah431111 | i have defaultexpirey=50 |
| 06:21.31 | poohbah431111 | maxexpirey=50 |
| 06:21.40 | IOscanner | Okay I have 183 sip session working to Asterisk. I have other asterisk boxes that are making calls to the carrier via an IAX trunk to our core Asterisk boxes. They don't seem to be passing the 183 RTP messages to the calls coming from the IAX trunk. |
| 06:21.51 | IOscanner | Has anyone else had this issue? |
| 06:21.53 | poohbah431111 | in my sip.conf General section? |
| 06:22.41 | poohbah431111 | ? |
| 06:22.45 | poohbah431111 | oops |
| 06:23.01 | poohbah431111 | Sounds like a big implimentation IO? |
| 06:24.39 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:25.32 | pabs3 | btw, has anyone started (or thought of doing so) a project like rockbox that presents a consistent set of features (& config & user interfaces) across multiple phone brands? |
| 06:25.58 | TomasuAway | If you have a voip provider that supports SIP and IAX, whats over all better to use? |
| 06:26.17 | JT | up to you really |
| 06:26.58 | TomasuAway | cons vs pros is sorta what I'm looking for. |
| 06:29.39 | IOscanner | TomasAway: http://www.voip-info.org/wiki/view/IAX+versus+SIP |
| 06:29.40 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:29.45 | TomasuAway | heh |
| 06:29.52 | TomasuAway | thanks |
| 06:30.13 | IOscanner | no problem |
| 06:31.08 | IOscanner | If you plan to bring to an upstream carrier via SIP and using asterisk I am finding SIP and IAX don't play well when dealing with inband vs or out band |
| 06:31.56 | IOscanner | I have been having problems getting TDM messages from SIP 180 and 183 messages to pass down an IAX trunk to other asterisk boxes |
| 06:32.03 | TomasuAway | I don't follow.. I have a rather simple config, VOIP Provider (sip/iax) + asterisk + SIP Phone |
| 06:33.14 | mkl1525 | and another problem: when using voicemail I can see that in tmp the file is growing but after caller quits the file is deleted in tmp but not moved to inbox. Any suggestions what the problem could be (disk isn't full and tmp + inbox dir have same permissions)? |
| 06:33.21 | IOscanner | If you plan to use any IAX trunks for IAX soft clients then use IAX. |
| 06:33.53 | Keltus | are there any providers that do asterisk hosting (but you can customize it and configure it) and also toll-free origination? |
| 06:34.23 | IOscanner | If not SIP might be an easy way. I am finding If I use SIP across the board it is fine. If I want to use IAX trunks or clients they don't play well. |
| 06:34.25 | Siya | TomasuAway: I guess it would be safe to say, try to match what you'll be using on your phones. makes life easier when troubleshooting as you'll only have to familiarise yourself with one protocol rather than two. |
| 06:34.39 | poohbah431111 | How do I set " Registration Expiration" in asterisk? |
| 06:35.04 | TomasuAway | I see that IAX makes dealing with ports to map simpler... |
| 06:35.32 | IOscanner | yep I like IAX trunks, but I think I am going to have to convert everything to SIP trunks |
| 06:35.33 | Siya | tomcorrect |
| 06:36.31 | *** join/#asterisk ardor (n=Miranda@ip70-170-92-65.lv.lv.cox.net) |
| 06:36.48 | IOscanner | If you ever plan to have asterisk boxes making calls via an Asterisk box VOIP (SIP) + Asterisk (SIP) + Asterisk (IAX) stick with SIP . |
| 06:37.32 | IOscanner | It might be more work, but documentation for issues between IAX and SIP are hard to find. |
| 06:37.50 | jql | I use sip end-to-end, which lets me use openser as my registrar |
| 06:40.04 | TomasuAway | thanks guys. I think I'll try IAX first, see if theres any issues, and switch back to sip later if needed. |
| 06:40.23 | IOscanner | no problem good luck |
| 06:42.06 | mkl1525 | [TK]D-Fender, have tried it with your example and the example from the wiki "exten => xxxxxxxx,2,Queue(all|tT|||30)" but the caller stays in the queue forever, any further hints? |
| 06:50.56 | mosty | pabs3: it's much simpler to standardise on a single make of phone in a single business |
| 06:51.46 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:53.25 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:53.33 | mosty | i wish there was an openiax (openser-like efficiency for iax) |
| 06:55.02 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 06:57.06 | mightnare | is it possible for an agi to retrieve the sip response when dial ends? |
| 06:57.08 | mightnare | i see something like this "Got SIP response 486 "Busy Here" back from ..." on the cli, i was hoping that my agi can also retrieve this info |
| 06:57.54 | drrt | hello |
| 06:58.23 | drrt | mightnare, you can try line status checking |
| 06:58.28 | TomasuAway | hmm, my provider has a check box on my DID config page labeled "Send ANI Prefix" what is an ani prefix? |
| 06:59.00 | mightnare | drrt: line status checking even if the channel is dead already? |
| 07:00.22 | drrt | mightnare, you can check it before the call |
| 07:00.45 | krdian_ | mightnare: try to use DeadAGI instead of AGI |
| 07:00.53 | drrt | mightnare, no so much difference when do u check line |
| 07:00.58 | Keltus | what's the best way to find a good voip provider? |
| 07:01.12 | Carlis4 | Anyone that knows a SIP<->PSTN provider that allows you to set outgoing CALLERID-number? |
| 07:01.17 | drrt | krdian_, he wants to to get line status 1st |
| 07:01.23 | drrt | as i got |
| 07:01.29 | mightnare | i'm already using deadagi, i just don't know which variable to retrieve the dial result code |
| 07:01.50 | Carlis4 | mightnare: Write a script that logs all varibles to a file and do a test-run. |
| 07:02.16 | Carlis4 | You might discover other useful variables. |
| 07:02.24 | drrt | Carlis4, can you post the way ? |
| 07:03.32 | Carlis4 | drrt: You mean the variable is not posted automatically to agi? You have to retrieve it in the dialplan? |
| 07:03.42 | mightnare | after "EXEC DIAL "SIP/200" for example, i'm trying to retrieve the dial return codes using "GET VARIABLE HANGUPCASE" |
| 07:03.59 | mightnare | ... but i don't get any |
| 07:04.33 | mightnare | though i get something like this: |
| 07:04.43 | mightnare | <PROTECTED> |
| 07:04.44 | mightnare | <PROTECTED> |
| 07:04.44 | mightnare | <PROTECTED> |
| 07:05.04 | mightnare | from the CLI |
| 07:05.20 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 07:05.39 | drrt | Carlis4, no. can you show the way how to retrieve all variables? |
| 07:10.38 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:13.29 | IOscanner | Any one have a link to creating SIP trunk between two asterisk boxes? I can't seem to find one that works. I get username/auth mis match when I try. |
| 07:24.34 | snuffy22 | mightnare, to retrieve the status of a dial.. i just use '${DIALSTATUS}' if the response is busy etc do somethin else |
| 07:24.53 | snuffy22 | look at voip-info dialstatus |
| 07:25.18 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:25.23 | mightnare | but DIALSTATUS only returns a selected set... |
| 07:25.50 | mightnare | i was hoping to get a more specific return code... like what shows on the CLI |
| 07:25.56 | snuffy22 | ahh.. |
| 07:26.07 | mightnare | Got SIP response 486 "Busy Here" back from 192.168.100.100 |
| 07:26.13 | mightnare | something like that :) |
| 07:26.15 | snuffy22 | mm.. k |
| 07:26.46 | mightnare | or... Got SIP response 603 "Decline" back from 192.168.100.100, if the user rejected the call... |
| 07:28.26 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 07:28.47 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:29.19 | TomasuAway | IOscanner, Siya: Id like to thank you two again :) got my phone working now :) |
| 07:30.40 | EvilDeshi | what are the default ports for asterisk for iax for my NAT> |
| 07:30.41 | EvilDeshi | ? |
| 07:31.04 | snuffy22 | ${SIPURI} maybe.. but doubtful |
| 07:32.09 | TomasuAway | EvilDeshi: http://www.voip-info.org/wiki-IAX ;) |
| 07:32.09 | mightnare | EvilDeshi: the port is defined inside iax.conf |
| 07:32.10 | EvilDeshi | thanks |
| 07:32.16 | mosty | IOscanner: sip doesn't support trunking btw, but anyway is this a 2-way link? ie calls initiated from both ends? |
| 07:37.01 | IOscanner | mosty: why do docs have things about SIP trunks? |
| 07:37.29 | IOscanner | Correct you will have one link for inbound and one for outbound |
| 07:37.46 | IOscanner | I can't seem to get my links to work |
| 07:37.54 | mosty | IOscanner: and are both peers defines as friends? |
| 07:38.04 | IOscanner | nope peers |
| 07:38.13 | IOscanner | should they be friends |
| 07:38.52 | BBHoss_Laptop | does it have to be SIP? I have IAX working |
| 07:39.14 | mosty | IOscanner: type=friend means that the peer both makes and receives calls |
| 07:39.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:39.32 | IOscanner | Ah that might do it |
| 07:39.38 | IOscanner | I have it registered |
| 07:39.53 | IOscanner | It was just showing up as unkown |
| 07:40.17 | BBHoss_Laptop | u can do qualify=yes and it will give you latency as well |
| 07:41.25 | IOscanner | mosty: Now I can't send calls down the link. |
| 07:42.12 | mosty | what is the error message? |
| 07:44.18 | IOscanner | SIP/2.0 407 Proxy Authentication Required is in the SIP debug |
| 07:44.54 | IOscanner | sorry that was to asterisk |
| 07:45.14 | mosty | i think that's normal, that means one peer is asking the other for authentication details |
| 07:46.32 | *** join/#asterisk tmcpr (n=tmcpr@85-189-92-116.btlnet.managedbroadband.co.uk) |
| 07:47.14 | IOscanner | found it X-Asterisk-HangupCause: No route to destination |
| 07:47.27 | BBHoss_Laptop | probably nat |
| 07:48.08 | IOscanner | I see SIP/2.0 401 Unauthorized on the remote server |
| 07:48.36 | BBHoss_Laptop | what type of auth? simple secret? |
| 07:49.05 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:49.07 | tengulre | hi,all |
| 07:50.06 | tengulre | How to config the smtp auth of voicemail when I using VoiceMail(uxxx) ? |
| 07:51.17 | mosty | tengulre: doesn't asterisk just use the local smtp server? |
| 07:51.29 | IOscanner | I didn't define. I thought md5 was default |
| 07:55.15 | BBHoss_Laptop | i would set secret=something |
| 07:55.35 | BBHoss_Laptop | then again im using iax2 not sip |
| 07:55.48 | BBHoss_Laptop | its more designed towards trunking |
| 07:56.09 | tengulre | mosty: how to send the voicemail to my public mail address? like xxxx@hotmail.com? |
| 07:58.54 | *** part/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 08:00.13 | *** join/#asterisk menil (n=meni@62.90.116.95) |
| 08:03.50 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:04.19 | mosty | tengulre: asterisk just uses the local smtp server. figure out how to configure that so you can send mail from the linux command line. then asterisk's voicemail->email should just work |
| 08:06.42 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 08:09.34 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 08:11.43 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:15.10 | tzafrir | actually asterisk uses the local MTA through the sendmail interfact, and not the local SMTP service |
| 08:15.41 | tzafrir | though in practice those tend to be handled by the same program |
| 08:16.20 | tzafrir | and right: if you can send mail with mail/mailx, you can probably send voicemail with asterisk |
| 08:23.23 | BBHoss_Laptop | ure isp blocking MTAs? |
| 08:23.58 | IOscanner | mostly: thanks I got it working. |
| 08:24.08 | IOscanner | I had to send calls direct to the carrier for now |
| 08:24.24 | IOscanner | Asterisk SIP trunks still don't pass progressinband |
| 08:24.39 | IOscanner | it must be a but or something that was overlooked. |
| 08:24.43 | JT | they're not trunks :) |
| 08:25.04 | IOscanner | I think I will just use openser to handle this. |
| 08:25.16 | IOscanner | JT: Sorry |
| 08:25.20 | IOscanner | :) |
| 08:26.02 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:27.25 | *** join/#asterisk Vec2 (n=Vec@dsl-241-198-60.telkomadsl.co.za) |
| 08:32.29 | *** join/#asterisk A[s]H (n=TnT@host117-192-static.53-88-b.business.telecomitalia.it) |
| 08:33.24 | A[s]H | DIGIUM have steal me!!! |
| 08:33.38 | A[s]H | I have payed codec but never receive key |
| 08:33.44 | A[s]H | not answer me on email |
| 08:33.46 | BBHoss_Laptop | haha |
| 08:33.50 | jql | digium have theif me |
| 08:33.51 | A[s]H | and not answer me at phone |
| 08:33.53 | BBHoss_Laptop | lemme go visit them |
| 08:33.55 | A[s]H | what i must do ??? |
| 08:34.03 | BBHoss_Laptop | they are 10 mi away |
| 08:34.11 | BBHoss_Laptop | want me to send in a tac force |
| 08:34.31 | A[s]H | digium fuck |
| 08:34.48 | BBHoss_Laptop | why didnt u just "try" the open source intel version |
| 08:34.59 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 08:35.05 | A[s]H | can u give me address? |
| 08:35.11 | BBHoss_Laptop | brb |
| 08:35.13 | BBHoss_Laptop | looking |
| 08:35.21 | A[s]H | try ??? and it's free? |
| 08:35.51 | BBHoss_Laptop | g729 right? |
| 08:35.56 | A[s]H | yes |
| 08:36.45 | BBHoss_Laptop | http://asterisk.hosting.lv/ |
| 08:36.56 | A[s]H | are free to use? |
| 08:37.13 | A[s]H | DISCLAIMER: You might have to pay royalty fees to the G.729/723 patent holders for using their algorithm. |
| 08:37.49 | tzafrir | A[s]H, may I suggest that you wait a few hours until there will actually be some people of Digium alive here? |
| 08:38.11 | A[s]H | i have payed my codec a week ago |
| 08:38.26 | BBHoss_Laptop | free for experimentation |
| 08:38.33 | BBHoss_Laptop | if you want to commercialize |
| 08:38.35 | BBHoss_Laptop | u must pay |
| 08:38.42 | A[s]H | yes |
| 08:38.54 | A[s]H | i have payed but i never receive my key-id |
| 08:39.29 | BBHoss_Laptop | what are you using it for |
| 08:39.37 | JT | you should probably ring digium |
| 08:39.50 | A[s]H | 2 hour that i ring it |
| 08:39.53 | A[s]H | no answer |
| 08:40.04 | BBHoss_Laptop | its 3:30am right now |
| 08:40.11 | A[s]H | only IVR |
| 08:40.11 | BBHoss_Laptop | wait till 10am |
| 08:40.22 | JT | rofl |
| 08:40.31 | JT | good idea to call when they're open |
| 08:40.35 | BBHoss_Laptop | yeah |
| 08:40.39 | BBHoss_Laptop | its still dark here |
| 08:40.45 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:40.58 | A[s]H | NOT SERIOUS COMPANY! |
| 08:41.08 | JT | A[s]H: not serious user |
| 08:41.16 | A[s]H | i have payed |
| 08:41.19 | A[s]H | a week ago |
| 08:41.20 | JT | "paid" |
| 08:41.20 | drrt | JT, agree |
| 08:41.26 | A[s]H | for a key-ID!! |
| 08:41.35 | A[s]H | excuse me form my bad english |
| 08:41.39 | BBHoss_Laptop | its supposed to be automatic i think |
| 08:41.45 | BBHoss_Laptop | u check your spam? |
| 08:41.49 | A[s]H | yes |
| 08:41.54 | A[s]H | not answer |
| 08:41.55 | BBHoss_Laptop | did you get an order confiration? |
| 08:42.08 | tzafrir | A[s]H, bashing them here will not help you |
| 08:42.43 | BBHoss_Laptop | you bring problems, we give solutions |
| 08:42.51 | BBHoss_Laptop | solution provided |
| 08:43.01 | tzafrir | there's probably some misunderstanding. There are probably a few folks in the channel that can help you sort out such misunderstandings |
| 08:43.43 | *** join/#asterisk bitl (n=cahe@stat-5-160.e-sky.ru) |
| 08:43.46 | *** join/#asterisk A[s]H (n=TnT@host117-192-static.53-88-b.business.telecomitalia.it) |
| 08:44.06 | BBHoss_Laptop | u in italy? |
| 08:44.08 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 08:44.08 | *** join/#asterisk matsk (n=mk@194.68.102.171) |
| 08:44.09 | *** join/#asterisk lupino3 (n=lupino3@217-133-45-108.b2b.tiscali.it) |
| 08:44.17 | lupino3 | hello * |
| 08:44.20 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 08:44.43 | lupino3 | is there a way to pass a variable to the called member of the queue? |
| 08:45.21 | snuffy22 | hmm i'd consider using the asterisk db |
| 08:45.36 | lupino3 | me too |
| 08:45.41 | snuffy22 | if the information isn't stored within the channel when its picked up |
| 08:45.49 | A[s]H | yes |
| 08:45.50 | *** part/#asterisk naxxtor (n=naxxtor@vm209.contextshift.co.uk) |
| 08:45.51 | lupino3 | and I thought of using the call id as a key for the variable |
| 08:45.59 | lupino3 | but when a queue member is called |
| 08:46.07 | lupino3 | asterisk issues a brand new channel |
| 08:46.20 | lupino3 | so I don't have an unique id :( |
| 08:46.42 | lupino3 | the problem is that I work with multiple queues associated to multiple IVRs |
| 08:47.05 | lupino3 | so I'd like to pass a variable from a given IVR to the member of the queue |
| 08:47.14 | lupino3 | and I don't know how to do it :( |
| 08:47.48 | snuffy22 | hmm must admit i dont use queues that often atm |
| 08:48.46 | lupino3 | thanks however :) |
| 08:48.48 | snuffy22 | out of interest what stuff is kept from when the call is placed in the queue till when its picked up |
| 08:49.12 | snuffy22 | maybe you can find somethin pseudo unique to work off |
| 08:50.00 | lupino3 | I thought of the caller id |
| 08:50.13 | lupino3 | but if two people with no caller id are both into the system... POOF |
| 08:50.42 | JT | global vars, astdb, odbc |
| 08:51.22 | lupino3 | thanks JT, but I don't have an unique ID to retrieve the info |
| 08:51.49 | JT | every call has a unique id |
| 08:51.52 | lupino3 | yes |
| 08:51.55 | *** join/#asterisk matsk (n=mk@194.68.102.171) |
| 08:52.05 | lupino3 | but as soon as the Queue app calls a member of the queue |
| 08:52.12 | lupino3 | Asterisk creates a new call |
| 08:52.30 | lupino3 | so the ID is different (I already tried to do it in that way, with ${UNIQUEID}) |
| 08:53.08 | A[s]H | i try to call it at 17pm (Italy) |
| 08:53.27 | A[s]H | if i not receive answer, i make a bomb and ship it |
| 08:53.33 | BBHoss_Laptop | uhh |
| 08:53.41 | BBHoss_Laptop | i wouldnt say that in a US chan |
| 08:53.56 | BBHoss_Laptop | CIA will be knocking down your door shortly |
| 08:54.00 | A[s]H | yes |
| 08:54.04 | A[s]H | sux cia |
| 08:54.19 | JT | A[s]H: why would anyone want to help you when you sound like a shit talking 13y/o? |
| 08:54.32 | BBHoss_Laptop | lolzorz |
| 08:54.42 | A[s]H | i look for help from a week |
| 08:54.46 | A[s]H | nobody help me |
| 08:54.47 | drrt | seems there no other way to talk :) |
| 08:55.03 | A[s]H | do u think it is correct? |
| 08:55.26 | JT | i don't think it's correct to start making threats, no |
| 08:55.34 | drrt | call to the office during their workday |
| 08:55.37 | JT | i think you should call digium during their business hours |
| 08:55.42 | A[s]H | i have called |
| 08:55.45 | A[s]H | always ivr |
| 08:55.53 | A[s]H | i leaved 1 million mess |
| 08:55.53 | JT | during THEIR business hours |
| 08:55.54 | A[s]H | :) |
| 08:55.56 | JT | not yours |
| 08:55.57 | *** join/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 08:56.17 | *** join/#asterisk c4colo (n=DJpyro@70-57-31-8.hlrn.qwest.net) |
| 08:56.21 | A[s]H | ok this evening i try |
| 08:56.40 | A[s]H | i hope somebody hear me |
| 08:56.45 | [o^o] | what's up? |
| 08:57.29 | A[s]H | Italy is Italy |
| 08:57.45 | A[s]H | other are only state |
| 08:57.46 | [o^o] | yes |
| 08:58.04 | [o^o] | hi c4colo |
| 08:58.13 | c4colo | hello |
| 08:58.19 | A[s]H | see u then |
| 08:58.20 | A[s]H | bye |
| 08:58.33 | A[s]H | tnk u BBHoss_Laptop |
| 08:58.43 | BBHoss_Laptop | btw |
| 08:58.56 | BBHoss_Laptop | i dont think italy is covered by that patent |
| 08:59.01 | BBHoss_Laptop | im no lawyer though |
| 08:59.13 | [o^o] | what's that g729? |
| 08:59.24 | BBHoss_Laptop | yezh |
| 08:59.34 | denke | [o^o] its a codec |
| 08:59.56 | denke | [o^o] with low bandwidth and high processor use |
| 09:00.22 | A[s]H | g729 it's nothing, 10$ on the air |
| 09:00.23 | [o^o] | yeah, 8khz |
| 09:00.36 | [o^o] | but can you tell me how much after sip? |
| 09:00.37 | BBHoss_Laptop | 8 kbit |
| 09:00.51 | [o^o] | 729a+sip= ? |
| 09:00.57 | BBHoss_Laptop | 12kbit? |
| 09:01.02 | [o^o] | no |
| 09:01.04 | BBHoss_Laptop | 16 |
| 09:01.07 | [o^o] | no |
| 09:01.12 | BBHoss_Laptop | 1kbit |
| 09:01.17 | florz | [o^o]: you don't wanna transport G729 in SIP |
| 09:01.24 | BBHoss_Laptop | iax? |
| 09:01.33 | jql | no? |
| 09:01.35 | c4colo | skinny? |
| 09:01.41 | c4colo | tcp? |
| 09:01.45 | [o^o] | h.323? |
| 09:01.53 | BBHoss_Laptop | yeah tcp rtp streams are teh l33t |
| 09:01.54 | c4colo | PCM |
| 09:01.55 | florz | no, cause all those text headers make for quite some overhead |
| 09:01.57 | [o^o] | BBHoss_Laptop that would be iax2 |
| 09:02.02 | jql | gsm? |
| 09:02.11 | [o^o] | gsm is a codec |
| 09:02.17 | jql | it's also a network |
| 09:02.19 | [o^o] | g729a is a codec |
| 09:02.40 | BBHoss_Laptop | its also an acronym |
| 09:02.50 | c4colo | I only use realmedia audio streams for my pbx |
| 09:02.50 | [o^o] | so are speex (yuk) and ilbc, and g723 and g726 |
| 09:02.53 | [o^o] | and g711a |
| 09:02.54 | [o^o] | etc |
| 09:02.58 | jql | Groupe Spécial Mobile |
| 09:03.06 | BBHoss_Laptop | LPC10!!!! |
| 09:03.18 | [o^o] | sip adds about 20kbps to any of them |
| 09:03.27 | BBHoss_Laptop | what about iax2 |
| 09:03.31 | [o^o] | so, g729 is around 28kbps |
| 09:03.34 | denke | 20? |
| 09:03.34 | jql | no, rtp does |
| 09:03.38 | jql | sip is irrelevant |
| 09:03.39 | florz | [o^o]: SIP really adds a lot more |
| 09:03.51 | jql | rtp is used by many protocols |
| 09:03.56 | [o^o] | so does iax2, but trunked iax2 can save most of that on each additional channel |
| 09:03.57 | c4colo | iax2 trunked is the lowest overhead for multiple streams, from what I have seen |
| 09:04.12 | c4colo | I was saying that [Airwolf] |
| 09:04.18 | c4colo | er damn tab |
| 09:04.21 | c4colo | [o^o] |
| 09:05.01 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:05.03 | [o^o] | so, how do you have a codec for voip and not have a transport? |
| 09:05.06 | [o^o] | I'm curious |
| 09:05.15 | florz | [o^o]: ? |
| 09:05.33 | BBHoss_Laptop | http://articles.techrepublic.com.com/5100-1035_11-6159446-2.html |
| 09:05.43 | [o^o] | well, everyone is down on sip and iax2, so? what's low cost (bandwidth) alternative? |
| 09:05.52 | BBHoss_Laptop | copper |
| 09:06.04 | c4colo | heh |
| 09:06.08 | jql | tdm ftw |
| 09:06.58 | c4colo | or in other words "analog for the win" |
| 09:07.01 | [o^o] | so, you run a codec over an analoge card? |
| 09:07.14 | c4colo | you can use GSM to PLC |
| 09:07.15 | BBHoss_Laptop | yeah g729 :) |
| 09:07.20 | jql | well, analog in the 64k sampling sense |
| 09:07.36 | [o^o] | or just 4 chans on a $500 card, plus the great legacy telco calling rates... |
| 09:08.02 | [o^o] | and line charges |
| 09:08.07 | [o^o] | and taxes, and more taxes |
| 09:08.19 | c4colo | hmm... use a [T/C]DMA encoder algorithm over a 64k analog line to reduce overhead ... hmm |
| 09:08.21 | [o^o] | and surcharges |
| 09:08.24 | jql | everyone likes taxes |
| 09:08.42 | c4colo | not taxes, surcharges! |
| 09:08.44 | [o^o] | let's not forget the "dialtone fee" |
| 09:08.46 | [o^o] | in usza |
| 09:08.48 | [o^o] | USA |
| 09:08.55 | BBHoss_Laptop | usazorz |
| 09:08.58 | [o^o] | I told em we don't want the dialtone |
| 09:09.00 | c4colo | hahah |
| 09:09.07 | [o^o] | just the connection |
| 09:09.15 | c4colo | hmm |
| 09:09.16 | [o^o] | they looked at me like I had 2 heads |
| 09:09.21 | c4colo | haha |
| 09:09.28 | BBHoss_Laptop | lol |
| 09:10.13 | [o^o] | so BBHoss_Laptop your local telco was out and your voip was not, today? |
| 09:10.15 | [o^o] | that's funny |
| 09:10.36 | BBHoss_Laptop | not today |
| 09:10.39 | c4colo | I thought analog was so much more reliable |
| 09:10.39 | BBHoss_Laptop | 2 days ago |
| 09:10.45 | BBHoss_Laptop | not round here |
| 09:10.50 | c4colo | apparently |
| 09:10.54 | BBHoss_Laptop | humidity i guess |
| 09:11.06 | BBHoss_Laptop | plus dumbass house builders that munch on fiber |
| 09:11.46 | jql | bulldozers are attracted by glass. it's a known fact |
| 09:12.04 | jql | the force is inversely proportional to the thickness |
| 09:12.07 | BBHoss_Laptop | ditch-witch |
| 09:12.36 | [o^o] | you should see how excited a diesel earthmover gets when it hits a gas line!!! |
| 09:12.52 | BBHoss_Laptop | lol |
| 09:12.59 | BBHoss_Laptop | they slice through those round here too |
| 09:12.59 | [o^o] | voooooooom |
| 09:13.31 | [o^o] | and if the plates are missaligned, there go's a $20,000 engine |
| 09:15.31 | BBHoss_Laptop | gnight all |
| 09:15.35 | BBHoss_Laptop | been grand |
| 09:15.45 | [o^o] | kk |
| 09:18.41 | *** part/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 09:19.47 | denke | Hello Everyone! |
| 09:20.13 | denke | can anybody help me in Realtime architecture? |
| 09:20.54 | snuffy22 | yes what about it denke |
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| 09:37.18 | bintut | why conference and not meetme or the other way around? |
| 09:38.26 | puzzled | hi |
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| 10:04.07 | J4k3 | why is it that my asterisk works oodles better when I switch to inferior (P3-700 vs AthlonXP 2400+) Intel hardware? :) |
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| 10:44.56 | Dovid | , |
| 10:49.34 | festr__ | is it possible to acces ${RTPAUDIOQOS} on both bridged channels? |
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| 11:04.58 | stoffell_h | tzafrir_laptop, if the H/W ok led flickers; is that normal? (or should it stay lit without flickering?) |
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| 11:15.06 | tzafrir_laptop | hi |
| 11:16.38 | tzafrir_laptop | it may be a sign of problem but not always. That led is generally related to the power supply |
| 11:22.10 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 11:23.01 | ghenry | Hi, if a user dials a PSTN number from a mobile, let's it ring a few times and then hangs up, is it normal for the phones to keep ringing for a couple of seconds on the other end after the hangup |
| 11:23.13 | ghenry | which the potential of them picking up a dead call? |
| 11:24.13 | ghenry | I understand that mobile hangup to normal lines, including VoIP, takes a couple of seconds to propogate. |
| 11:24.14 | Dovid | ghenry: it depends on your set up. probably what is happening is that the hangup packet takes time to go thru untill asterisk knows to hang up. |
| 11:24.49 | ghenry | yeah, some users are reporting dead calls |
| 11:25.19 | ghenry | best way to handle this? |
| 11:25.57 | Polis_ttt | wath can be wrong in my network, i got two asterisk-servers, and suddenly today, none of them was working. They were sending registry to my voip-providers, but none of the package from voip-provider got back to my servers. I killed all workstations at my network and it started again. Looks like some workstation was taking all voip-packages that was coming in to my netowrk. what shall i look for? |
| 11:26.08 | Dovid | i would do a debug on the system to see when asterisk gets the hang up from when the mobile user hangs up |
| 11:27.12 | ghenry | sure, good point |
| 11:27.43 | Dovid | Polis_ttt: maybe a router issue (of it not passing the packets to the proper location) or maybe an issue with one of the desktops. I would see which machine is doing it. turn it off and see what happens. also double check ur settings in the router. |
| 11:28.23 | Dovid | another thing u can do is do a SIP trace on the aserisk box. see if the astreisk box is sending out the right iP |
| 11:33.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:33.36 | Dovid | i know this is OT but does any one have a script that can SSH in to a box and then i can tell it what to do ? I want to create a php script that will restart certain services on a remote box thru an AGI |
| 11:35.26 | Polis_ttt | Dovid: whats the command for a sip-trace? |
| 11:36.08 | Dovid | set verbose 7 |
| 11:36.11 | Dovid | sip debug |
| 11:36.18 | Dovid | (I meant sip debug - not trace) |
| 11:36.28 | Dovid | you can also do this |
| 11:36.47 | Polis_ttt | Dovid: can't u use a perl-module for that, that you add to crontab? |
| 11:37.03 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 11:37.03 | Dovid | rom the CLI |
| 11:37.03 | Dovid | v |
| 11:37.07 | Dovid | ngrep -t -W byline -d any -w <SIP USER NAME HERE> port 5060 |
| 11:37.34 | Dovid | Poliss_ttt: never used perl b4 |
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| 11:39.21 | punani | i got asterisk on a dual p3 rig, would http://ftp.digium.com/pub/telephony/codec_g729/asterisk-1.2/x86-32/codec_g729a_v31_pentium3m.tar.gz be the correct codec? |
| 11:39.34 | punani | not sure what the 'm' suffix is for |
| 11:40.22 | Dovid | punai: i don kno much about it but for me I just installed asterisk + asterisk add ons and it worked for me |
| 11:48.03 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-46-82.red.bezeqint.net) |
| 11:49.05 | UnixManiac | hello all! I have the following problem: i have installed the g723 and g729 modules in order to make outgoing calls to my SIP provider over the internet. I have configured a new section in sip.conf named [mysip] with the right codecs then i added in extensions when anyone dials 41xxx to make the call throught mysip. So far so good everything works fine. But when i try to put the register => ... into the sip.conf and then starting asterisk |
| 11:49.15 | UnixManiac | anyone that can help me? |
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| 12:00.33 | evisu | anyone know why i would keep getting this error when using .call files: OutgoingSpoolFailed |
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| 12:03.29 | VJFROMGT | <PROTECTED> |
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| 12:09.44 | denke | Helo Everyone! Can anybody help me with realtime? |
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| 12:38.25 | Qwell | putnopvut: you're gonna have to explain your nick to me sometime.. |
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| 12:48.12 | [TK]D-Fender | Live... in Stereo! |
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| 12:54.14 | coppice | [TK]Fender: you forgot the decimal point |
| 12:55.30 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 12:55.57 | [TK]D-Fender | coppice: fine... 120.0" screen :) |
| 12:56.01 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:56.14 | coppice | that's better |
| 12:56.32 | [TK]D-Fender | coppice: Tough house today... |
| 12:58.06 | Zeeek | and that screen is just for the polycom |
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| 13:01.14 | [TK]D-Fender | Zeeek: No, but it is technically for my * server ;) |
| 13:03.50 | *** part/#asterisk DragonBall-Z (n=aahmed@202.5.145.13) |
| 13:04.06 | Zeeek | The Polycom phones have a wonder server built in. It takes 5 minutes to boot, but wow, what a show after! |
| 13:04.42 | Zeeek | Oops need to get home |
| 13:05.02 | *** join/#asterisk zodell (n=zodell@firebox3-sharon.sctelcom.net) |
| 13:06.15 | *** part/#asterisk zodell (n=zodell@firebox3-sharon.sctelcom.net) |
| 13:11.10 | *** join/#asterisk pejo_ (n=peter@1-1-5-39a.lio.sth.bostream.se) |
| 13:14.50 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 13:15.18 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 13:15.27 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:15.57 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:18.19 | *** join/#asterisk T1w (n=mv@81.95.240.42) |
| 13:18.35 | T1w | Any know what !! Got S-frame while link down means? |
| 13:19.35 | coppice | well, your link was down, and an S-frame was received |
| 13:19.47 | *** join/#asterisk BouYYY (n=BouYYY@81-86-77-70.dsl.pipex.com) |
| 13:20.09 | [TK]D-Fender | coppice: Doctor, doctor! ......it hurts when I raise my arm like this! |
| 13:20.13 | T1w | What is a S Frame?... |
| 13:20.22 | T1w | and should the D channel go down sometimes? |
| 13:20.24 | coppice | supervisory |
| 13:20.29 | [TK]D-Fender | T1w: Comes right before a T frame clearly... |
| 13:20.44 | coppice | depends on the provider. |
| 13:21.27 | punani | [TK]D-Fender: that a clan name or something? |
| 13:21.38 | T1w | Because when we call the pbx.. we get connection... then we see a D channel on span 1 up.. and we loose connecion |
| 13:21.42 | T1w | what can this be? |
| 13:21.44 | [TK]D-Fender | punani: Yup, a long time ago.... I still wear it... |
| 13:21.51 | punani | what game? :) |
| 13:22.07 | T1w | what can this be? |
| 13:22.09 | T1w | ups |
| 13:22.54 | *** join/#asterisk Ebola (n=Ebola@host86-137-4-175.range86-137.btcentralplus.com) |
| 13:23.19 | [TK]D-Fender | punani: Action: Half-Life |
| 13:24.38 | T1w | Any? |
| 13:25.29 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:25.35 | punani | never got into the action mod |
| 13:25.44 | punani | what you playing these days |
| 13:26.19 | Katty | morning |
| 13:26.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:26.45 | [TK]D-Fender | Katty: Mew. |
| 13:26.59 | T1w | Because when we call the pbx.. we get connection... then we see a D channel on span 1 up.. and we loose connecion |
| 13:27.04 | [TK]D-Fender | punani: These days an occasional bout of Diablo 2 and thats it.... |
| 13:28.09 | [TK]D-Fender | punani: I haven't been a gamer since dropping AHL for my FPS. |
| 13:28.34 | punani | gaming is a complete waste of time |
| 13:28.37 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:28.37 | punani | it's like an addiction ;/ |
| 13:30.06 | [TK]D-Fender | punani: Yeah I rememmber passing hours on it. On Gamespy (of whatever that server tracer was) that used to rank players & servers my clan hit #1 for a few weeks one winter. My best friend and I swapped the #player position between two weeks :) |
| 13:31.15 | jm|work | Red Bull gives you wind |
| 13:31.37 | Katty | don't you mean wings? |
| 13:31.43 | jm|work | oh. Yeah. |
| 13:31.55 | [TK]D-Fender | Katty: Dunno... you're looking pretty "winded" to me :) |
| 13:32.06 | Katty | pfft. |
| 13:32.22 | [TK]D-Fender | Katty: See... you're already expulsing hot air! |
| 13:33.39 | [TK]D-Fender | Katty: ... quickly ;) |
| 13:33.39 | Katty | my weekend was too short. |
| 13:33.39 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 13:33.39 | Katty | everytime i see irule i think hyrule |
| 13:33.39 | Katty | and then i think of zelda. |
| 13:33.40 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:33.43 | irule | hi katty! |
| 13:33.43 | Katty | hewwo. |
| 13:34.08 | irule | does threewaycalling=yes work with sip phones? I see it in zapata.conf |
| 13:35.04 | [TK]D-Fender | irule: that is a zapata option, not a SIP one. |
| 13:35.12 | T1w | What could be the problem... when ISDN calls just resets... |
| 13:35.17 | [TK]D-Fender | irule: and clearly "no" |
| 13:35.21 | T1w | we get a D Channel is up |
| 13:35.28 | T1w | and the connection is then lost? |
| 13:35.47 | UnixManiac | hello all! I have the following problem: i have installed the g723 and g729 modules in order to make outgoing calls to my SIP provider over the internet. I have configured a new section in sip.conf named [mysip] with the right codecs then i added in extensions when anyone dials 41xxx to make the call throught mysip. So far so good everything works fine. But when i try to put the register => ... into the sip.conf and then starting asterisk |
| 13:35.54 | UnixManiac | anyone that can help me? |
| 13:35.59 | irule | weird |
| 13:37.17 | denke | <UnixManiac> I ll try |
| 13:37.26 | [TK]D-Fender | UnixManiac: You stopped short of actually telling us the PROBLEM. |
| 13:39.07 | jkiff | UnixManiac: You were cut off at "then starting asterisk". |
| 13:39.25 | *** join/#asterisk Fieldy (i=6rK5ZAX7@gentoo/contributor/Fieldy) |
| 13:39.42 | *** join/#asterisk limbi-Q (n=limbique@194.178.123.2) |
| 13:39.54 | irule | UnixManiac into the sip.conf and then starting asteris{????????} |
| 13:40.09 | UnixManiac | asterisk i get an error:"temporarily anavailable".The same message occurs if i remove the codecs from [mysip] and try to make a call...my question is how do i make asterisk to use the same codecs when it tries to do the register thing?(I tried putting the same lines from [mysip] into [general] but it doesnt work :( ) |
| 13:40.24 | UnixManiac | well it get cut |
| 13:40.56 | UnixManiac | sorry! |
| 13:40.56 | [TK]D-Fender | UnixManiac: Registering shouldn't have anything to do with codecs...... |
| 13:41.29 | [TK]D-Fender | UnixManiac: And no evidence to the contrary as you have not succeeded in any attempt. |
| 13:41.34 | Qwell | s/shouldn't/doesn't/ |
| 13:42.19 | [TK]D-Fender | Qwell : this is "Bizarro World".... when the improbable is definate, and the impossible not entirely so ;) |
| 13:42.53 | UnixManiac | my sip provider told me that |
| 13:43.02 | irule | way to go [TK]D-Fender! |
| 13:43.04 | UnixManiac | that this error is due to lack of codecs |
| 13:43.11 | Qwell | during a register? no |
| 13:43.13 | Qwell | they're full of it |
| 13:43.20 | irule | yeah! |
| 13:43.23 | irule | huh? |
| 13:43.34 | UnixManiac | Qwell ok |
| 13:43.47 | UnixManiac | if you say so...you are the experts :) |
| 13:43.51 | UnixManiac | i will send them an email |
| 13:43.57 | UnixManiac | the thing is |
| 13:44.01 | UnixManiac | that i had the same problem |
| 13:44.18 | UnixManiac | when i hadnt add the codecs in sip.conf at [mysip] |
| 13:44.31 | UnixManiac | when i was trying to call the same message occured |
| 13:44.35 | UnixManiac | and i couldnt call |
| 13:45.16 | UnixManiac | when i add dissalow=all and allow=g723 and allow=729 then it worked |
| 13:45.41 | UnixManiac | but putting the same values in the [general] section doesnt do any good :( |
| 13:45.42 | Qwell | sure, but that's unrelated to registering |
| 13:45.48 | UnixManiac | ok |
| 13:46.14 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:46.24 | Qwell | speaking of registering... |
| 13:46.56 | Mercestes | I tried to register this weekend but it wanted my CC# and a payment plan so I canceled it out. |
| 13:47.01 | Qwell | lame |
| 13:47.05 | Mercestes | I didn't wanna have to call in 10 days and cancel it. =/ |
| 13:47.07 | Mercestes | Yea, I agree. |
| 13:47.16 | Qwell | didn't used to be like that |
| 13:47.51 | Qwell | there's a trial on the front page, which shouldn't have any of that - or didn't about a month ago |
| 13:48.00 | Mercestes | I was even ok with entering my CC#, but when it asked for a payment plan I was like, "uh, nah." |
| 13:48.07 | Mercestes | I'll check that out. |
| 13:48.09 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:48.17 | Mercestes | but I did try this weekend |
| 13:48.41 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 13:51.01 | *** join/#asterisk simonkern (n=simonker@p54aa8e8f.dip0.t-ipconnect.de) |
| 13:51.04 | simonkern | hi |
| 13:53.30 | mosty | what does Dial() do if it rings more than one extension, and one of those extensions is set to DND? |
| 13:54.06 | T1w | Because when we call the pbx.. we get connection... then we see a D channel on span 1 up.. and we loose connecion ? any |
| 13:54.43 | masked | <PROTECTED> |
| 13:54.45 | masked | oops |
| 13:54.46 | masked | ede |
| 13:56.11 | [TK]D-Fender | mosty: It stops trying to ring that device and keeps on trying the rest |
| 13:58.29 | mosty | [TK]D-Fender: that's what i thought, but Dial appears to stop, and then the timeout extension kicks in |
| 13:58.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:59.07 | [TK]D-Fender | mosty: Timeout is supposed to happen naturally. Pastebin it. |
| 14:01.48 | mosty | hrm, the dial command didn't have a timeout specified. i want it to ring indefinitely with the r option |
| 14:02.20 | [TK]D-Fender | mosty: Indefinate = bad, "r" = bad.... |
| 14:02.27 | mosty | i see an error message complaining about r being an invalid timeout, so i guess it was just chance that the DND response came back right before the timeout kicked in |
| 14:02.39 | [TK]D-Fender | mosty: PASTEBIN |
| 14:02.53 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:03.56 | mosty | i've already fixed the initial problem |
| 14:04.28 | mosty | just need to figure out how to specify the r option and indefinite ringing simultaneously (a customer request, not my choice) |
| 14:05.18 | Qwell | any real consultant would say "no" |
| 14:05.25 | Qwell | and explain why it's a bad thing |
| 14:05.54 | [TK]D-Fender | mosty: ......***PASTEBIN*** |
| 14:06.12 | Qwell | $20 says he's missing a | |
| 14:06.18 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 14:06.20 | [TK]D-Fender | Qwell : I was about to PM that ;) |
| 14:06.50 | [TK]D-Fender | Qwell : I jsut wanted the evidence to assassinate him on before mentioning it :) |
| 14:06.58 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:07.06 | mosty | well, mission accomplished |
| 14:07.14 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:07.26 | angryuser | can somebody tell me why is this happening? http://forums.digium.com/viewtopic.php?t=15938 (cdr_mysql related) |
| 14:07.36 | [TK]D-Fender | mosty: pwned |
| 14:08.22 | mosty | indeed |
| 14:09.03 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 14:09.35 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 14:09.56 | mosty | i get paid next week |
| 14:10.00 | mosty | i promise |
| 14:10.14 | *** join/#asterisk yacc (n=andreas@091-141-082-046.dyn.one.at) |
| 14:10.29 | [TK]D-Fender | "The Czech is in the mail...." |
| 14:11.07 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 14:11.14 | Daejeo1 | i am calling my friend box. "error- sip response 603 "declined" |
| 14:11.43 | [TK]D-Fender | Daejeo1: Pay off your credit card next time... |
| 14:12.07 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:12.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:12.11 | Katty | anthm: (= |
| 14:12.25 | Daejeo1 | [TK]D-Fender: why? |
| 14:12.27 | anthm | hey |
| 14:12.28 | [TK]D-Fender | Katty: I spent over 1hr on hold with them once only to get hung up on.... |
| 14:12.34 | Katty | nice. |
| 14:12.39 | Katty | let's not have horror story's like that. |
| 14:12.56 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:13.36 | Katty | Ooo, new hold musics. |
| 14:13.41 | Katty | must be in a different part of india |
| 14:17.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:17.07 | LeddyHM | maybe try a limo company then |
| 14:17.11 | LeddyHM | they got lots of drivers |
| 14:18.10 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 14:21.19 | [TK]D-Fender | Katty: Drivers for...? |
| 14:21.47 | Katty | an hp motherboard. |
| 14:22.15 | Katty | i can't even understand half of what this guy says |
| 14:22.30 | Katty | and i'm pretty framilier with lots of accents |
| 14:22.34 | Katty | hell, i can understand junky! |
| 14:23.04 | LeddyHM | which hp mobo drivers? |
| 14:23.08 | [TK]D-Fender | Katty: EEK |
| 14:23.18 | Katty | 945GCT-HM |
| 14:23.28 | Katty | it's an ESC apparently, not on their website either |
| 14:23.28 | LeddyHM | for what? |
| 14:23.37 | LeddyHM | updated bios? |
| 14:23.41 | Katty | and the chip for the realtek nic... |
| 14:23.41 | LeddyHM | os? |
| 14:23.44 | Katty | it's not on the realtek website either |
| 14:23.54 | Katty | LeddyHM: nic drivers. |
| 14:24.02 | LeddyHM | which os? |
| 14:24.03 | Katty | nothing on driverguide either |
| 14:24.05 | Katty | xp pro |
| 14:24.25 | LeddyHM | How long you been on hold? |
| 14:24.32 | Katty | 15 minutes |
| 14:24.44 | LeddyHM | You coulda installed a compatible nic by now ;) |
| 14:24.48 | Katty | yes |
| 14:24.50 | Katty | but this is for a client |
| 14:24.52 | Katty | it's not ours |
| 14:25.51 | LeddyHM | GL |
| 14:25.51 | errr | I have 3 systems (pbx1 2 and 3) I was wondering if there is a way to make it so the extensions on pbx2 so I could add one as a BLF to my phone on pbx1 my phone on pbx1 would know if the pbx2 extension was in use or not |
| 14:25.51 | LeddyHM | I dun like hpeepee |
| 14:25.52 | *** join/#asterisk r_evolution (i=r_evolut@208.6.94.10) |
| 14:26.33 | Katty | damnit, they hung up on me |
| 14:26.43 | r_evolution | isnt that just rude. |
| 14:28.27 | [TK]D-Fender | Katty: What PC model? |
| 14:28.44 | Katty | [TK]D-Fender: pavillion a6005y |
| 14:28.46 | r_evolution | TK I havent seen you in freaking forever. |
| 14:28.53 | r_evolution | TK alias... the cannuck. |
| 14:28.54 | *** join/#asterisk th3 (i=th3@gateway/tor/x-58353022030cb75f) |
| 14:30.00 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 14:30.01 | Katty | [TK]D-Fender: i've been all over the hp website. |
| 14:30.11 | Katty | [TK]D-Fender: i've got a spare nic in it now, maybe the stupid windows update thing will have something |
| 14:30.28 | btsteve | hello i need some help with odbc and realtime config for asterisk 1.4 |
| 14:30.34 | Daejeo1 | working |
| 14:30.46 | Daejeo1 | TK?:? |
| 14:30.55 | [TK]D-Fender | r_evolution: Been a while... but its because of your absence, not mine :) |
| 14:31.08 | *** join/#asterisk disa (n=disa@87.226.145.138) |
| 14:31.13 | r_evolution | this is true homie... this is true. |
| 14:31.16 | disa | hi, all |
| 14:31.21 | r_evolution | I went from only a little kinda busy |
| 14:31.24 | r_evolution | to way too freaking busy |
| 14:31.30 | r_evolution | in about a month |
| 14:31.37 | r_evolution | no more time for hanging in * :( |
| 14:31.43 | *** join/#asterisk ccesario (n=ccesario@200-158-227-195.dsl.telesp.net.br) |
| 14:33.14 | r_evolution | im just hiding from the world today as I am slightly discouraged with it... |
| 14:33.31 | r_evolution | simple question leads me to reject the possibility of buying anything from any voip reseller ever again. |
| 14:36.31 | r_evolution | You know... I bet if Katty ever went or ever does go to an Astricon... there would be tackleage of great amount. |
| 14:36.31 | [TK]D-Fender | My screen just arrived :D |
| 14:36.51 | r_evolution | because any time ive ever been in here... and she has as well... there's net-tackle. |
| 14:36.54 | r_evolution | oh the horror IRL. |
| 14:36.55 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 14:36.57 | [TK]D-Fender | r_evolution: Yeah... most people here haven't even SEEN a woman before ;) |
| 14:37.09 | r_evolution | You know |
| 14:37.10 | r_evolution | some days |
| 14:37.14 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 14:37.14 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 14:37.16 | r_evolution | I would wake up pitying them... |
| 14:37.23 | r_evolution | but some days... I would wake up envying them... |
| 14:37.27 | r_evolution | we call those days... PMS time |
| 14:37.40 | r_evolution | I call it "Best Western Take me Away!" |
| 14:39.44 | coppice | I wonder why the Best Westerns in China aren't Best Easterns :-\ |
| 14:39.55 | Mercestes | lmao |
| 14:39.57 | r_evolution | Maybe it should be inversed... |
| 14:40.03 | r_evolution | China should = Best Western |
| 14:40.07 | r_evolution | American should = Best Eastern |
| 14:40.16 | Mercestes | Then it'd be exotic |
| 14:41.46 | [TK]D-Fender | coppice: "Go west... life is peaceful there..." |
| 14:42.49 | r_evolution | That sounds like a lie. |
| 14:43.03 | r_evolution | West? West of here would be West Virginia... ever seen the movie Wrong Turn? Yeah nuf said. |
| 14:43.05 | coppice | Best Western does too |
| 14:43.49 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 14:43.59 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 14:45.49 | VJFROMGT | does anyone know how to create an extension which authenticate by caller id (i dont care about security) |
| 14:46.13 | Mercestes | VJFROMGT, Obviously not. Authenticates what?? |
| 14:46.28 | *** part/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 14:46.53 | VJFROMGT | well i have users who are using sip client that do not support userid and password |
| 14:47.28 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:47.36 | Mercestes | insecure=very |
| 14:48.05 | Mercestes | although I've never seen a sip client that doesn't support userid and password. |
| 14:48.14 | VJFROMGT | Mera softswitch |
| 14:48.15 | jm|work | does seem a bit odd |
| 14:48.30 | VJFROMGT | some quintom hardware |
| 14:48.32 | jm|work | skinny, maybe |
| 14:48.42 | Mercestes | quintum? lmao |
| 14:48.49 | VJFROMGT | they are really based on H323 platform but support sip |
| 14:48.59 | Mercestes | Authenticate by IP |
| 14:49.08 | Mercestes | actually...I never got hte damned thing hooked to my * |
| 14:49.14 | VJFROMGT | can astereisk auth by ip? |
| 14:49.30 | Mercestes | I hooked it directly to my Coppercom PBX and ran quintim to quintim for the entire Voip path |
| 14:49.40 | Mercestes | and came in analog on one side and out analog on the other side |
| 14:49.54 | Mercestes | It's an overglorified, retarded, overpriced 48 channel ATA. |
| 14:49.58 | Mercestes | trash the damn thing and get real phones. |
| 14:51.33 | r_evolution | big up to Mercestes. |
| 14:52.03 | VJFROMGT | i am dealing with providers who are reselling minutes |
| 14:52.23 | Mercestes | You are *their* customer. make them serve you on your terms. |
| 14:52.27 | btsteve | where can i find the mysql database schema fro asterisk |
| 14:52.29 | Mercestes | or find a new provider |
| 14:52.42 | Mercestes | btsteve: google asterisk rta |
| 14:53.40 | VJFROMGT | so there is no way of allowing authentication based on ip or caller id? |
| 14:53.56 | krdian_ | VJFROMGT: sip ? |
| 14:54.12 | VJFROMGT | yes |
| 14:55.06 | r_evolution | so just out of curiousity VJFROMGT... what exactly are you attempting to do? Route traffic from an originating IP to your * box? |
| 14:55.19 | krdian_ | VJFROMGT: set proper host of client and add allowguest=yes |
| 14:55.42 | r_evolution | or vice versa? |
| 14:55.59 | r_evolution | here's a concept... just set them as a peer... and define a context. |
| 14:56.05 | *** join/#asterisk bbryant (i=brett@nat/digium/x-255b97385f353f3c) |
| 14:57.49 | neverblue | anyone have alot of experience with both Ekiga and Twinkle? |
| 15:05.16 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 15:06.02 | irule | what is an exten => a,... for a macro? I dont see anything in http://www.voip-info.org/wiki-Asterisk+cmd+Dial and comes witin a macro in the samples |
| 15:06.21 | *** join/#asterisk \lart (i=foobar@pool-72-73-230-114.cmdnnj.fios.verizon.net) |
| 15:06.35 | *** part/#asterisk \lart (i=foobar@pool-72-73-230-114.cmdnnj.fios.verizon.net) |
| 15:07.32 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
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| 15:13.00 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 15:13.52 | codazoda | I have an asterisk box that doesn't seem to be detecting rings anymore. I want to make sure the zaptel drivers are still loaded. Is there a command I can run to check that? |
| 15:14.14 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 15:14.16 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 15:14.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:16.34 | neverblue | ~book irule |
| 15:16.46 | neverblue | grr |
| 15:16.46 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:17.07 | krdian_ | codazoda: zap show status |
| 15:17.26 | lee_is_me | I could really use some help if anyone is online. Dialed into customer site and when they try to dial out on zap lines, looks like it is trying to dial but then exits 0. Can anyone offer suggestion? |
| 15:19.33 | codazoda | I get "no such command" with "zap show status". I'm running 1.4.1, has it changed? |
| 15:20.03 | *** join/#asterisk thojo (n=ttr@0x5733db9d.bynxx19.adsl-dhcp.tele.dk) |
| 15:20.31 | codazoda | When I run zttool is says, "unable to open /dev/zap/ctl: no such file or directory" |
| 15:21.32 | neverblue | ~thebook |
| 15:21.44 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:21.54 | neverblue | there irule |
| 15:23.19 | codazoda | I had this all working, been doing great for several weeks, just died over the weekend. I'm thinking the zaptel drivers aren't loaded for some reason... Maybe a kernel update, although I'm sure I disabled those. |
| 15:23.22 | anonymouz666 | hi Katty |
| 15:23.40 | Katty | anonymouz666: allo (= |
| 15:25.57 | irule | whats up with the book? |
| 15:27.14 | [TK]D-Fender | irule: Go read up on Asterisk Standard Extensions. This is Dialplan 101 |
| 15:27.14 | anonymouz666 | how to make manager more verbose when submiting requests? |
| 15:29.24 | *** join/#asterisk FaUl (i=immo@shell.chaostreff-dortmund.de) |
| 15:29.25 | FaUl | hi |
| 15:29.52 | FaUl | i have a problem with sending CID via e1-port to our Provider |
| 15:30.16 | FaUl | the provider discards whatever i send and set it to our main-number |
| 15:30.23 | FaUl | any hints? |
| 15:30.26 | Qwell[] | FaUl: tell them not to do that |
| 15:30.31 | FaUl | i did |
| 15:30.36 | Qwell[] | and? |
| 15:30.44 | [TK]D-Fender | FaUl: Could be they don't permit setting it. Pastebin the CLI output of your attempt at verbose 10 and PRI debug enabled. |
| 15:30.47 | [TK]D-Fender | ~pb |
| 15:31.08 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:31.18 | anonymouz666 | FaUl: I got the same problem here. What country? |
| 15:31.19 | FaUl | i tried numbers out of our number-block - it doesn't work - they even tried to set clip no screening - did not help either |
| 15:31.31 | irule | what page is that? |
| 15:31.32 | FaUl | germany |
| 15:31.52 | codazoda | Yup, my kernel updated... Bad, very bad. |
| 15:31.53 | [TK]D-Fender | FaUl: PASTEBIN it please... |
| 15:32.38 | FaUl | yea, i'll do |
| 15:33.22 | anonymouz666 | FaUl: No matter what I do here, the telco always send to PSTN the main-number |
| 15:34.33 | *** join/#asterisk bird_of_Luck (n=melifaro@80.251.128.150) |
| 15:35.26 | FaUl | [TK]D-Fender: ah, pri debug was a nice hint anyway |
| 15:37.15 | FaUl | [TK]D-Fender: http://pastebin.ca/519145 |
| 15:37.24 | lee_is_me | Is dialing out on zap lines effected by * having internet access or access to DNS? |
| 15:37.55 | *** part/#asterisk th3 (i=th3@gateway/tor/x-58353022030cb75f) |
| 15:38.06 | [TK]D-Fender | FaUl: "SetCallerPres("OSS/dsp", "prohib")" never used.... |
| 15:38.07 | ghenry | does this suggest they agree on codecs? |
| 15:38.07 | ghenry | Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) |
| 15:38.24 | lee_is_me | I was just online with a customer whose zap lines would not dialout...looked like they were trying in CLI, but then whey would exit 0. Reboot * and now they are working correctly. They were having some problems with their DSL earlier... |
| 15:38.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:38.51 | FaUl | [TK]D-Fender: i tried without that anwyay |
| 15:38.57 | FaUl | anyway |
| 15:39.47 | [TK]D-Fender | FaUl: Try setting the Number alone, not name/both, ditch that "pres" deal as well and the ANI too. |
| 15:39.59 | FaUl | yea, that was my first trie |
| 15:40.06 | FaUl | didn't work either |
| 15:40.26 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 15:40.27 | [TK]D-Fender | ghenry: Yes, agree on ulaw |
| 15:40.42 | ghenry | thought so |
| 15:40.43 | [TK]D-Fender | FaUl: Perhaps try again and pastebin that... |
| 15:40.48 | ghenry | so that's not my audi oprob then |
| 15:40.52 | [TK]D-Fender | FaUl: maybe a different error.. |
| 15:41.15 | [TK]D-Fender | lee_is_me: No, Zap has nothing to do with anything else. |
| 15:41.43 | lee_is_me | thanks TK. Can you think of any reason for that behavior off hand? |
| 15:42.54 | [TK]D-Fender | lee_is_me: Nope |
| 15:43.08 | lee_is_me | Odd. OK thanks again. |
| 15:43.40 | FaUl | [TK]D-Fender: http://pastebin.ca/519157 |
| 15:45.23 | [TK]D-Fender | FaUl: Ok, phase 2 : pastebin your zapata.conf |
| 15:47.27 | FaUl | [TK]D-Fender: sorry, have to leave, anyway: here it is: http://pastebin.ca/519162 |
| 15:47.43 | FaUl | i'll be back in est. 1hour, feel free to write suggestions into query |
| 15:53.24 | irule | is there an asterisk users manual for the hole family? |
| 15:54.04 | irule | not the configuring user, but the teenager that will stress the system to its limits? heh |
| 15:54.08 | Mercestes | ~book |
| 15:54.14 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:54.14 | Mercestes | ~docs |
| 15:54.16 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 15:54.20 | Mercestes | ~wiki |
| 15:54.25 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:54.33 | Mercestes | ~wiki irule |
| 15:54.41 | Katty | burger king or mcdonalds? |
| 15:54.54 | Mercestes | Katty, BK, of course |
| 15:55.07 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-114-89.dsl.tulsok.swbell.net) |
| 15:55.09 | Katty | there's also a taco smell over there |
| 15:55.12 | Katty | and a subway! |
| 15:55.15 | Qwell[] | eww |
| 15:55.18 | Qwell[] | subway FTL |
| 15:55.30 | Katty | i like subway. |
| 15:55.39 | Qwell[] | I don't do subway anymore |
| 15:55.43 | irule | jbot is right Mercestes! wikipedia hates my guts! lol |
| 15:55.50 | Katty | i did subway two times a day while i was at cluecon |
| 15:55.59 | Katty | that was 3 days |
| 15:56.06 | Qwell[] | my wife worked there for a few days, and quit after her boss yelled at her for throwing away olives that had been sitting out for several days, and started becoming...unfresh |
| 15:56.10 | Katty | 'course nothin else vegan was around. |
| 15:56.33 | Qwell[] | (after he had told her to throw them out not 10 minutes before that) |
| 15:56.39 | file | Katty: this is the only place to be! I'll give you love and fantasy... you won't forget me easily |
| 15:57.10 | Katty | file: i don't think you ever went to subway with me. |
| 15:57.12 | Katty | file: sniffle. |
| 15:57.26 | Katty | all alone...in the subway |
| 15:57.43 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 15:57.58 | irule | subway rocks |
| 15:58.00 | Katty | wellll, tis lunch time |
| 15:58.02 | Katty | buhbye |
| 15:58.03 | Mercestes | Qwell[]: That is the quality of workers you get when you pay around 10k a year. |
| 16:00.23 | irule | Qwell[] your wife should have reported her ex-boss with his boss and mentioning you might call sanitation authorities in case HE is not punished for breaking the LAW and btw forcing him to apologyze at the least |
| 16:00.34 | woolbeo | So I upgraded from asterisk 1.2 to 1.4 last week and I am having a problem with a change that wasn't documented. I have a Dial command that ends up calling anohter dial command, and in 1.2 asterisk would honor the timeout on the first dial command, but 1.4 does not. Anyone else seen this, or have any ideas around this? |
| 16:00.47 | Qwell[] | irule: my wife is lazy - it was easier to just quit :P |
| 16:00.59 | file | lazy like Qwell! |
| 16:01.03 | Qwell[] | indeed |
| 16:01.22 | irule | yes that is what most people do, quite sadly |
| 16:03.02 | *** join/#asterisk yacc (n=andreas@091-141-082-119.dyn.one.at) |
| 16:03.22 | irule | I guess it would have been a cool adventure calling the media,cprotesting, doing all the BS to make sure you get back at the MF for yelling lol |
| 16:04.24 | irule | I would have started with -OK MF, EAT THOSE OLIVES OUT OF THE TRASH! |
| 16:04.28 | [TK]D-Fender | woolbeo: Pastbin your failed attempt at verbose 10 |
| 16:04.30 | [TK]D-Fender | ~pb |
| 16:04.32 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 16:05.55 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 16:12.07 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 16:12.14 | *** join/#asterisk jsolares (n=jsolares@216.106.168.71) |
| 16:12.52 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 16:17.03 | woolbeo | [TK]D-Fender, will do, but I won't be able to do it for 8 hours.. Its a live system, and it is part of the after hours dialplan logic... |
| 16:17.49 | woolbeo | [TK]D-Fender, Its not officially a failed attempt, because the call goes through, it just doesn't timeout when it should. |
| 16:19.23 | [TK]D-Fender | woolbeo: pastebin your dialplan |
| 16:19.42 | [TK]D-Fender | woolbeo: Because complaining without showing us anything is wasting our time :) |
| 16:25.03 | ghenry | If I'm dailing a SIP url on port 5605, why would the receiving end send back not on port 5060 <-- SIP read from 192.168.45.183:2289: |
| 16:25.24 | ghenry | * sends back to that port, so I get no audio: Transmitting (NAT) to 192.168.45.183:2289: |
| 16:25.32 | ghenry | The SIP call connects fine |
| 16:25.50 | ghenry | * sends it's ACKs back on this port, not the original 5605 like in the Dial |
| 16:25.57 | ghenry | so the audio never bridges |
| 16:26.05 | ghenry | any SIPAddHeader I can do? |
| 16:26.31 | cpm | [TK]D-Fender, do you do turn-key systems? |
| 16:27.26 | [TK]D-Fender | cpm: I do complete configs, but only SERVICE, not hardware |
| 16:27.35 | cpm | [TK]D-Fender, thanx |
| 16:27.40 | cpm | [TK]D-Fender, recommendations? |
| 16:27.47 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 16:27.50 | cpm | <PROTECTED> |
| 16:28.06 | [TK]D-Fender | cpm: For harware? Depends where you are |
| 16:28.20 | cpm | [TK]D-Fender, may I msg ? |
| 16:28.29 | [TK]D-Fender | cpm: Sure |
| 16:28.32 | cpm | thx |
| 16:28.43 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 16:29.29 | vAd0r | how does this compare w/ skype as far as security |
| 16:29.31 | vAd0r | encrypted calls etc |
| 16:32.01 | neuwald | hi folks. how to a2billing automatically do an sip reload after modifications ? |
| 16:32.01 | irule | anybody ever done a propper dual language system? |
| 16:32.17 | [TK]D-Fender | irule: Sure. |
| 16:35.34 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 16:38.07 | woolbeo | Ok Here is the relevant part of my dialplan, http://pastebin.ca/519273 for the case where asterisk 1.4 is not honoring the first dial timeout when Dial ends up calling another Dial, but it did in asterisk 1.2 |
| 16:38.19 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 16:40.40 | woolbeo | correction http://pastebin.ca/519283 |
| 16:40.49 | [TK]D-Fender | woolbeo: What context/exten should I start looking at? |
| 16:41.28 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 16:41.58 | froguz | one of the events for queue_log is COMPLETEAGENT(holdtime|calltime|origposition), where holdtime represents the caller's hold time. does this holdtime also include the time that caller was put on hold AFTER the call was answered by the agent? |
| 16:42.24 | woolbeo | [TK]D-Fender, Basically it happens when line 10 is executed.. |
| 16:42.58 | froguz | or just the time the caller wait for an agent to answer? |
| 16:43.35 | VJFROMGT | how can i allow all calls to be made via a certian extension, ie, regardless of what username password u try, you can make a call out |
| 16:43.46 | woolbeo | [TK]D-Fender, under 1.2 it would Dial the first op and if the first op didn't answer within the timeout in line 10, it would move on to the second op, but under 1.4 it just dials the first op, and never timesout. |
| 16:45.43 | woolbeo | [TK]D-Fender, to make a complicated dialplan simple, line 10 looks up what number to call, ends up calling out line 186 after all the look ups. |
| 16:45.51 | BSD_Tech | writing dial plan hurts the brain |
| 16:46.09 | vAd0r | how does this compare w/ skype as far as security and encryption? |
| 16:46.09 | BSD_Tech | and it seems no one wants to pool code together |
| 16:46.11 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:50.15 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-114-89.dsl.tulsok.swbell.net) |
| 16:51.13 | vAd0r | do i need to use srtp to get secure calls on asterisk |
| 16:51.14 | vAd0r | ? |
| 16:51.16 | BSD_Tech | O hell BKW is alive |
| 16:51.19 | BSD_Tech | run |
| 16:51.37 | BSD_Tech | Mr West how are you these days |
| 16:53.05 | mishehu | I didn't realize that asterisk supported srtp |
| 16:55.13 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 16:55.21 | Dr-Linux | guys wanna discuss an issue |
| 16:55.41 | Dr-Linux | i'm getting this alot, and calls get's disconnected: |
| 16:55.41 | Dr-Linux | Got SIP response 482 "Loop Detected" back from 192.168.0.106 |
| 16:55.42 | Dr-Linux | <PROTECTED> |
| 16:55.46 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 16:56.13 | elg | i'm curious, does the web UI in AsteriskNOW come a la carte, i.e. for an existing installation? |
| 16:56.17 | vAd0r | I am trying to find out if that is what i need for security and if it does |
| 16:56.41 | BSD_Tech | the gui is far from ready |
| 16:56.49 | BSD_Tech | its still under devel |
| 16:56.58 | vAd0r | does no one know anything about SRTP |
| 16:57.06 | Corydon76-work | It's ready; it's just not complete |
| 16:57.18 | Corydon76-work | but the stuff that is there works |
| 16:57.30 | BSD_Tech | its half ready it still has issues |
| 16:57.45 | BSD_Tech | the latest ver seems to have issues |
| 16:58.01 | Corydon76-work | Issues that you've reported? |
| 16:58.12 | BSD_Tech | pari and I are working on them |
| 16:58.38 | Corydon76-work | k, didn't realize you were working with Pari |
| 16:58.45 | BSD_Tech | but he is mia today |
| 16:59.25 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 17:01.51 | froguz | i reiterate my question. does 'holdtime' in queue_log events also include the time that caller was put on hold AFTER the call was answered by the agent? |
| 17:01.59 | vAd0r | can i make a sip call to googletalk? |
| 17:02.16 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:02.39 | rpm | vAd0r: im pretty sure you can with the jabber/xmpp support in asterisk. |
| 17:03.09 | Qwell[] | rpm: that wouldn't exactly be a sip call... |
| 17:03.20 | Siya | elg: #asterisk-gui |
| 17:03.48 | Corydon76-work | froguz: No, it does not. |
| 17:04.08 | rpm | Qwell[]: it'd be a sip call to the pbx :) |
| 17:04.12 | *** part/#asterisk elg (n=fugalh@216.31.27.110) |
| 17:04.21 | Qwell[] | or analog, or iax2, or various others |
| 17:04.31 | *** join/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 17:04.38 | Corydon76-work | froguz: it is the initial hold time prior to being connected to a queue member, only |
| 17:04.41 | vAd0r | can you allow a skype number to call your sip extension? |
| 17:05.26 | Siya | vAd0r: with skype-out or using a skype-sip gateway |
| 17:05.26 | Corydon76-work | vAd0r: theoretically, yes |
| 17:05.33 | Siya | don't ask here unless wearing a flame proof suit |
| 17:05.41 | Siya | google is your friend :) |
| 17:06.25 | vAd0r | im reading about it know. logically i would assume i setup some sort of trunk to skype and then an inbound route that routes my skype ext to my astrisk one |
| 17:06.29 | vAd0r | is that about right |
| 17:06.49 | sheldonh | ztcfg fails for my quad PRI Wildcard TE410P (2nd Gen), with "CAS signalling on span 5 conflicts with HDLC with FCS check on channel 109." zaptel-1.4.1 backported to debian etch. zaptel.conf here: http://rafb.net/p/eZAiZb87.html any advice? |
| 17:06.57 | Corydon76-work | vAd0r: that is how it would be done, yes |
| 17:07.02 | froguz | Corydon76-work, thank u. I think it should include the complete hold time. i wander if i can do that using the dialplan and the System app |
| 17:07.05 | vAd0r | thx |
| 17:07.21 | Dr-Linux | Corydon76-home: any advice for "Got SIP response 482 "Loop Detected" back from |
| 17:07.23 | Dr-Linux | ? |
| 17:07.26 | Corydon76-work | froguz: it most certainly should not |
| 17:08.18 | Corydon76-work | froguz: you're welcome to log that hold time separately, but initial time to answer is a critical statistic which would be skewed by adding later hold time |
| 17:08.21 | Qwell[] | you definitely want the distinction between the queue member putting the caller on hold, and the caller being on hold waiting for a queue memeber |
| 17:09.57 | froguz | Qwell[]: yes, it would be great having that statistic too |
| 17:10.10 | Dr-Linux | Qwell: any clue about my question :) |
| 17:11.13 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 17:14.01 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:16.03 | froguz | it's possible to record log entries from 'Started music on hold' and 'Stopped music on hold' events? |
| 17:16.15 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 17:16.40 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:16.55 | *** join/#asterisk saftsack (n=oliver@p54A7C5AB.dip.t-dialin.net) |
| 17:17.37 | Ryushin | I just upgraded to the latest version of asterisk, zaptel, and wanpipe. Asterisk keeps displaying this message "Primary D-Channel on span 2 up" and outgoing calls won't go out. |
| 17:18.00 | BSD_Tech | what ver of wanpipe |
| 17:18.28 | Ryushin | 3.1 |
| 17:18.29 | BSD_Tech | and wich sangoma card |
| 17:18.37 | BSD_Tech | 3.1 is beta |
| 17:18.47 | BSD_Tech | last I checked |
| 17:18.51 | Ryushin | Didn't say beta. |
| 17:18.54 | Ryushin | I'll downgrade. |
| 17:18.59 | BSD_Tech | hold on |
| 17:19.18 | vAd0r | lol i have to "BUY" chanskype for that to work |
| 17:19.22 | vAd0r | how stupid |
| 17:20.06 | Ryushin | BSD_Tech: Holding...... |
| 17:20.24 | BSD_Tech | BETA Drivers ............................. wanpipe-3.1.0.tgz (2007-05-18) |
| 17:20.42 | BSD_Tech | it says on the linux sangoma page |
| 17:20.52 | Ryushin | Oh, well, I just went to the ftp site. |
| 17:20.56 | BSD_Tech | http://wiki.sangoma.com/wanpipe-linux-drivers |
| 17:20.58 | Ryushin | I'll downgrade and see what happens. |
| 17:21.14 | BSD_Tech | always read the webpages |
| 17:21.33 | *** join/#asterisk stack_ (n=stack@198.30.100.203) |
| 17:21.42 | *** join/#asterisk binary-zero (n=Shakeel@unaffiliated/binary-zero) |
| 17:21.51 | binary-zero | guys i get error on compilling SCCP2 |
| 17:21.59 | binary-zero | chan_sccp.c:1260: error: incompatible type for argument 1 of âast_inet_ntoaâ |
| 17:22.00 | BSD_Tech | # STABLE Voice & Data Drivers .......... wanpipe-2.3.4-9.tgz (2007-05-17) |
| 17:22.06 | binary-zero | can any one give a clue |
| 17:22.19 | irule | how to you recommend I implement Meetme? what are your experiences? I tried Meetme(600|Aqd) in a separate extension but would like something more professional looking |
| 17:23.21 | Ryushin | BSD_Tech: Well, I started to panic so I came here first. |
| 17:24.41 | Dr-Linux | why we need pedantic=yes in sip.conf ? |
| 17:24.46 | BSD_Tech | never panic |
| 17:25.08 | BSD_Tech | always refer back to the web pages and wiki |
| 17:25.13 | binary-zero | guys any idea about SCCP issue |
| 17:25.35 | Qwell[] | binary-zero: no, chan_sccp is dead |
| 17:25.38 | irule | when I blind transfer people to a conference room, I get hang up, how can I change that behavior so that once I transferred anyone, I get a playtunes(dial)? thanks |
| 17:25.39 | binary-zero | realy ? |
| 17:25.41 | vAd0r | Banaskin got it working |
| 17:25.44 | sheldonh | how bizarre. this doesn't work: http://rafb.net/p/eZAiZb87.html but this does: http://rafb.net/p/B1gMEV91.html |
| 17:25.44 | vAd0r | he will be back on later |
| 17:25.50 | Qwell[] | binary-zero: the last release was more than a year ago... |
| 17:26.03 | binary-zero | Qwell[]: so what would be the best thing to use skinny protocol on asterisk ? |
| 17:26.04 | MrChimpy | is 1.4 considered stable yet? |
| 17:26.09 | Qwell[] | chan_skinny in 1.4 |
| 17:26.10 | vAd0r | he got 7970 and ip comunicator working |
| 17:26.13 | vAd0r | w/ sccp |
| 17:26.23 | binary-zero | oh ! and configuration would be the same Qwell[] |
| 17:26.23 | binary-zero | ? |
| 17:26.24 | MrChimpy | as in stable for real proper large scale production? |
| 17:26.24 | vAd0r | he is gonna send me the compiled module |
| 17:26.28 | Qwell[] | binary-zero: no |
| 17:26.42 | binary-zero | Qwell[]: any URL would be very helpfull if you can |
| 17:29.20 | *** join/#asterisk andrewc (n=andrewc@dsl254-017-249.sea1.dsl.speakeasy.net) |
| 17:31.51 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 17:31.54 | sheldonh | MrChimpy: 1.4 is probably _better_ for production than 1.2 |
| 17:32.00 | ifnotwhynot | hi there |
| 17:32.20 | ifnotwhynot | could some please help me set up ari recording interfase please |
| 17:33.12 | Mercestes | sheldonh, that isn't saying much |
| 17:34.47 | *** join/#asterisk gk1 (n=Greg@pool-72-67-72-172.lsanca.fios.verizon.net) |
| 17:34.47 | killfill_ | hey, |
| 17:34.47 | killfill_ | i have 2 member in my queue |
| 17:34.47 | killfill_ | Agent/01 and 02. |
| 17:34.47 | killfill_ | and strategy = roundrobin. |
| 17:34.49 | JT | 1.2 is still considered more stable |
| 17:34.56 | killfill_ | got 2 problems. |
| 17:35.12 | killfill_ | 1.- Calls get in into agent 02 (wish is second in the list) i wish to make it ring in order |
| 17:35.42 | killfill_ | 2.- When Agent X doesnt take the call, then its voicmail apears. i wish to make it ring the next agent... |
| 17:35.52 | killfill_ | is this normal? |
| 17:36.19 | [TK]D-Fender | killfill_: Yes its perfectly normal. You pointed it to a place in your dialplan that FALLS TO VOICEMAIL. |
| 17:36.35 | [TK]D-Fender | killfill_: You should really pay more attention to what you're doing :) |
| 17:36.41 | killfill_ | heh |
| 17:36.56 | killfill_ | [TK]D-Fender: |
| 17:36.58 | killfill_ | exten = 90,1,Queue(${EXTEN}) |
| 17:37.01 | killfill_ | thats all i have there... |
| 17:37.03 | sheldonh | Mercestes: i hear you, but it _is_ an answer to the question :) |
| 17:37.20 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 17:37.24 | [TK]D-Fender | killfill_: think about where your agents LOG IN. |
| 17:37.26 | msetim | Hi guys |
| 17:37.41 | [TK]D-Fender | killfill_: Agent/01 is NOT a SIP device... |
| 17:38.01 | msetim | I would like to know how to clean a sip channel frozen |
| 17:38.06 | killfill_ | ah.. hm.. |
| 17:38.17 | killfill_ | [TK]D-Fender: you mean i should take their voicmail off? |
| 17:38.54 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 17:38.54 | [TK]D-Fender | killfill_: I mean you should think about sending the calls to a place without VM associated. You should have enough smarts to know what to do with this... |
| 17:41.10 | killfill_ | hmm.. |
| 17:41.11 | BSD_Tech | no |
| 17:41.11 | killfill_ | [TK]D-Fender: but for a small support-center, its normall to use queues with agents, right?.. or ppl uses ringgorups |
| 17:41.11 | BSD_Tech | killfill hold on |
| 17:41.22 | BSD_Tech | the issue is how the gui does iot |
| 17:41.28 | killfill_ | hm.. |
| 17:41.28 | [TK]D-Fender | killfill_: yes its perfectly normal to use queuest for call centers. |
| 17:41.37 | BSD_Tech | it does not set the queues dial plan right |
| 17:41.51 | BSD_Tech | and its something I have been working to fix |
| 17:42.14 | killfill_ | BSD_Tech: ah ok. |
| 17:42.18 | [TK]D-Fender | killfill_: make ANOTHER set of extens to dial your agents that DOESN'T lead to VM or answering the line., NOT that macro/context you are using now. |
| 17:42.42 | [TK]D-Fender | killfill_: You using a GUI built dialplan? |
| 17:43.08 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 17:43.10 | woolbeo | [TK]D-Fender, I take it you gave up trying to read my hairy dialplan? |
| 17:43.15 | BSD_Tech | the gui is far from perfect |
| 17:43.21 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:43.24 | BSD_Tech | and still has loads of dial plan issues |
| 17:43.25 | killfill_ | [TK]D-Fender: yup using asterisk-gui.. but im watching whats below it.. |
| 17:43.28 | Daejeo1 | is it wise to buy Grandstream BudgeTone 101? I want to use with asterisk |
| 17:43.42 | killfill_ | [TK]D-Fender: http://pastebin.ca/519472 |
| 17:43.51 | BSD_Tech | killfill give me a min I will dig up the perfect dial pal for queus |
| 17:43.58 | killfill_ | BSD_Tech: ok ... :P |
| 17:44.08 | *** join/#asterisk saftsack (n=oliver@p54a7e15e.dip.t-dialin.net) |
| 17:44.18 | [TK]D-Fender | killfill_: FUGLY, and sorry... no GUI support from me... I *LIKE* my sanity kplzthxbibi |
| 17:44.32 | killfill_ | i see no voicmail there.. just the agents that are user with voicmails... |
| 17:45.12 | irule | when I blind transfer people to a conference room, I get hang up, how can I change that behavior so that once I transferred anyone, I get a playtunes(dial)? thanks |
| 17:45.16 | Daejeo1 | join asterisknow |
| 17:45.22 | killfill_ | [TK]D-Fender: well, trying to understand how to make a dialplan that fit my needs.. |
| 17:45.23 | sysreq | Daejeo1: i own one, and it works ok.. just don't expect anything from the speakerphone (pretty much unusable due to echo). |
| 17:45.26 | [TK]D-Fender | killfill_: thats because you have tunnel-vision and aren't looking at the context used to dial your agents |
| 17:45.52 | [TK]D-Fender | Daejeo1: ... |
| 17:45.54 | [TK]D-Fender | ~gs |
| 17:46.26 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 17:46.27 | [TK]D-Fender | ~phones |
| 17:46.39 | jbot | extra, extra, read all about it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream ... |
| 17:46.39 | sysreq | Daejeo1: but aside from that, and if it's for testing/home purposes.. it's decent. |
| 17:46.43 | saftsack | hi, i've bought an octasic soft echo module. do i have the same EC cancel quality which i have with a ec hardware board? |
| 17:47.14 | [TK]D-Fender | saftsack: Try and compare. |
| 17:47.15 | Dr-Linux | damn |
| 17:47.34 | killfill_ | hm.. |
| 17:47.39 | Dr-Linux | [TK]D-Fender: i'm facing 2 issues with a call and with 2 seconds call gets dropped |
| 17:47.44 | Dr-Linux | you can see here: http://phpfi.com/237954 |
| 17:47.46 | Dr-Linux | the output |
| 17:47.51 | LeddyHM | damn, vitelity.net charges flat rate even if you are dialing "toll free" numbers |
| 17:47.58 | Dr-Linux | first issue: loop detected |
| 17:48.05 | vAd0r | I have been trying to get my linksys pap2 to work w/ my asterisk. I keep trying to unlock it. i have went to the web page and changed the user password to 1234. I then relog into the router w/ user/1234 I try to run the tftp to it and then it prompts me for another login which i am assuming is the admin one. I can not get this thing unlocked. Please help. |
| 17:48.10 | Dr-Linux | 2, May 29 22:37:18 WARNING[25360]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 06f635f96267c3c01b620803661c3c22@192.168.0.106 for seqno 102 (Critical Response) |
| 17:48.17 | Daejeo1 | sysreq: do you recommend any other phone? |
| 17:48.28 | Daejeo1 | echo problem. |
| 17:48.32 | stack_ | Does anyone have any experience running a credit card terminal through an asterisk box... they tend to be picky and running through an ATA box will work 10% of the time |
| 17:48.33 | [TK]D-Fender | Executing Macro("Local/4086@users-1b16,2", "stdexten60|4086|SIP/4086") in new stack Executing Dial("Local/4086@users-1b16,2", "SIP/4086|20|t") in new stack |
| 17:49.10 | [TK]D-Fender | Dr-Linux: You are calling yourself! Stupid infinite loop. YOU = SILLY! |
| 17:50.21 | Dr-Linux | hhm.. |
| 17:50.39 | *** join/#asterisk ffad (n=fad@ool-18b957f5.dyn.optonline.net) |
| 17:51.04 | Daejeo1 | [TK]D-Fender: can you recommend any cheap phone? |
| 17:51.16 | Dr-Linux | [TK]D-Fender: that's correct but what about 2nd issue? |
| 17:51.20 | killfill_ | [TK]D-Fender: this is my dialplan related to the queue: http://pastebin.ca/519487 zapata has in its context: # |
| 17:51.23 | killfill_ | [DID_trunk_1] |
| 17:51.25 | ffad | using asteriskNOW, i've hooked up a custom sip service provider to register with. but because i'm behind a NAT it won't register. any suggestions? |
| 17:51.29 | sysreq | Daejeo1: that's the only one i have because that's pretty much the only thing i could afford (i'm a student).. but i've heard good things about the polycom 301. |
| 17:51.32 | [TK]D-Fender | Daejeo1: Follow the phone list above. |
| 17:51.52 | *** join/#asterisk Fieldy (i=mQXsbdAp@gentoo/contributor/Fieldy) |
| 17:52.00 | [TK]D-Fender | IP 301 = waste |
| 17:52.07 | [TK]D-Fender | IP 320/330 for low end now. |
| 17:53.04 | woolbeo | My only complaint about polycoms are no backlight |
| 17:54.15 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 17:54.38 | punani | ffad: /etc/asterisk/sip_nat.conf |
| 17:54.45 | [TK]D-Fender | woolbeo: I'f you're willing to shell out a bit more, there are 2 with it |
| 17:54.48 | sysreq | Daejeo1: the trick here is to basically ask for a phone model, and wait for people to say it's worthless and list better ones. |
| 17:54.49 | sysreq | :) |
| 17:55.09 | GreyFoxx | Can anyone here recommend a good asterisk manager proxy? Something I can let clients use snap to connect to, but will filter out information that doesn't related directly to the client? |
| 17:55.26 | vAd0r | any ideas? |
| 17:55.55 | GreyFoxx | I was going to use astmanproxy and edit the code to add filtering but it segfaults a lot and before I just write my own from scratch I thought I'd look to see if there is anything better out there |
| 17:56.29 | *** part/#asterisk zonkedout (n=matt@sd-2704.dedibox.fr) |
| 17:56.50 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 17:57.02 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 18:00.41 | [TK]D-Fender | vAd0r: Keep googling guides for unlocking it and say a PRAYER. Maybe next time you won't try to "cheap-out" and you'll buy one that won't try to stab you in the back. |
| 18:00.53 | [TK]D-Fender | ~ygwypf |
| 18:01.12 | jbot | well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 18:01.12 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:01.45 | Katty | jbot: i love you. |
| 18:01.55 | jbot | You love you.? |
| 18:02.04 | Katty | jbot: no, i love you. |
| 18:02.22 | Mercestes | Do you love me? |
| 18:02.32 | Katty | botly love is different. |
| 18:02.55 | Mercestes | .... |
| 18:03.01 | Mercestes | you don't...omg! |
| 18:03.01 | file | jbot: botsnack |
| 18:03.01 | jbot | file: thanks |
| 18:03.12 | Mercestes | =) |
| 18:03.13 | Katty | Mercestes: simmer down now. |
| 18:03.22 | Katty | Mercestes: like spaghetti sauce!! |
| 18:03.41 | Mercestes | yes ma'am |
| 18:05.29 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 18:05.57 | killfill_ | when i select roundrobin in queues.conf.. the irder is not gettin respected.. i.e. Agent 01 is first member and then 02. |
| 18:06.03 | killfill_ | 02 is ringging first... |
| 18:06.09 | irule | when I blind transfer people to a conference room, I get hang up, how can I change that behavior so that once I transferred anyone, I get a playtunes(dial)? thanks |
| 18:06.17 | killfill_ | is this normal?. or it should be ringed in order? |
| 18:07.16 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 18:07.42 | Katty | weeee! |
| 18:07.45 | Katty | hmm-home: oh |
| 18:07.47 | Katty | hmm-home: YOU |
| 18:07.51 | Katty | hmm-home: you are /so/ in trouble. |
| 18:07.56 | Katty | hmm-home: i dunno what for, but i'll think of something. |
| 18:08.16 | file | Katty: he is in trouble for... not being Hmmhesays, but instead being hmm-home |
| 18:08.21 | Katty | file: oh, right. |
| 18:08.26 | Katty | hmm-home: what file said. |
| 18:08.35 | killfill_ | :S |
| 18:08.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-5-67.red.bezeqint.net) |
| 18:10.28 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:10.57 | *** join/#asterisk saftsack (n=oliver@p54a7f29c.dip.t-dialin.net) |
| 18:11.08 | gk1 | anyone here using hudlite and have the callerid popup working correctly? Mine will not pop until i answer it, and it should pop before answering, yes??? |
| 18:11.15 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 18:12.41 | Katty | hudlite? |
| 18:12.43 | Katty | is that free? |
| 18:13.32 | [TK]D-Fender | Katty: yes |
| 18:13.51 | Katty | [TK]D-Fender: maybe this is just what i need to make the receiponist quit bugging me. |
| 18:14.12 | Katty | [TK]D-Fender: also, i posted a new recipe. twas yummy. bbq beef, in teh crockpot (= |
| 18:14.27 | killfill_ | [TK]D-Fender: does member's order matters in queues.conf? |
| 18:15.07 | gk1 | it is free, and requires you to set up the server on the asterisk box |
| 18:15.11 | gk1 | its nice looking |
| 18:15.17 | [TK]D-Fender | Katty: BEEF? What happened to Little Ms. Vegan?! :) |
| 18:15.18 | gk1 | and does all kinds of groovy things |
| 18:15.39 | hmm-home | heh |
| 18:15.51 | hmm-home | Beef, its whats for dinner |
| 18:15.56 | gk1 | when an inbound call comes in it is supposed to pop a little window, and also if you want pop a url in the browser, but the callerid doesnt pop until after the call is answered |
| 18:17.27 | killfill_ | anyone knows if order matter in queues.conf?.... |
| 18:17.31 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:18.14 | [TK]D-Fender | killfill_: in 1.4 roundrobin is the same as the old RRMEMORY which remember the last person rung. |
| 18:18.31 | [TK]D-Fender | killfill_: You seriously need to READ. this is all documented. |
| 18:19.19 | Katty | [TK]D-Fender: that was almost 6 months ago now |
| 18:19.47 | LeddyHM | Katty: FOP works just as well too |
| 18:19.48 | [TK]D-Fender | Katty: Welcome to ..... the top 'o' the food chain! |
| 18:20.05 | hmm-home | I don't know of any vegans that stay vegan for the rest of their life |
| 18:20.07 | Katty | LeddyHM: we use fop here. |
| 18:20.10 | hmm-home | meat is just too good |
| 18:20.21 | Katty | and bad for you. |
| 18:20.28 | Qwell[] | taste > health |
| 18:20.30 | Katty | in another couple years, i'll probably stop eating it again for awhile. |
| 18:20.47 | hmm-home | Katty: Well prepared selections aren't necessarily bad for you |
| 18:20.54 | *** join/#asterisk kodok (n=me@bb121-7-79-146.singnet.com.sg) |
| 18:21.10 | Katty | hmm-home: yeah, and i really only eat turkey and chicken. this beef thing was just for the boy's sake ;) |
| 18:21.21 | hmm-home | Katty: :D |
| 18:21.28 | Katty | tastes funny |
| 18:21.36 | hmm-home | like any tasty morsel you have to work it into a well rounded diet |
| 18:21.38 | MrChimpy | everyone loves beefs! |
| 18:21.45 | Katty | pfft. |
| 18:21.49 | Katty | less meat, more pasta! |
| 18:21.54 | hmm-home | I make a wicked beef stew |
| 18:22.05 | Vec2 | When a call comes in to asterisk I have set it to, transfer the call after X seconds to my mobile, however the CDR does not show the dialed number has my cellphone, only shows the incomming number, does anyone have a solution to this ? |
| 18:22.10 | hmm-home | apparently you have not had a good one |
| 18:22.11 | [TK]D-Fender | Katty: Pasta will make you fat a hell of a lot faster than Beef.... |
| 18:22.16 | hmm-home | stew is more veggies than meat |
| 18:22.23 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D493.access.telenet.be) |
| 18:22.24 | Katty | [TK]D-Fender: yeah... |
| 18:22.27 | Katty | [TK]D-Fender: i know (= |
| 18:22.52 | Katty | poor bessie |
| 18:22.56 | MrChimpy | beef n'pig! |
| 18:23.08 | hmm-home | this weekend kicked my ass |
| 18:23.10 | Katty | grilled cheese! |
| 18:23.18 | kodok | what does native bridge mean ? |
| 18:23.18 | Katty | chinese food! |
| 18:23.22 | MrChimpy | beef n'pig n'cheese! |
| 18:23.29 | Katty | that's balogna. |
| 18:23.37 | killfill_ | [TK]D-Fender: so there is no way to make queues start alwais from a member X?.. |
| 18:24.01 | MrChimpy | kdook: iirc it means there's no codec traaslation going on between each side of the bridge |
| 18:24.11 | Qwell[] | killfill_: use penalties |
| 18:24.55 | MrChimpy | katty: it's either geeky or tasty and a typo |
| 18:24.55 | killfill_ | Qwell[]: yup.. just reading about them.. looks good. but use something like the old roundrobin would make it simplir in my case.. :P |
| 18:25.07 | [TK]D-Fender | killfill_: don't think so anymore... |
| 18:25.16 | Katty | MrChimpy: it's part of this hudlite screenshot page. |
| 18:25.57 | *** part/#asterisk binary-zero (n=Shakeel@unaffiliated/binary-zero) |
| 18:25.58 | [TK]D-Fender | Katty: Its a super CRM program. You really need to completely stop & read up on it. |
| 18:26.02 | Dr-Linux | damn |
| 18:26.03 | [TK]D-Fender | Katty: BIG business |
| 18:26.10 | Katty | oh ah |
| 18:26.11 | Dr-Linux | any idea wht this happens? : |
| 18:26.11 | Katty | butbut |
| 18:26.12 | Dr-Linux | May 29 23:25:04 WARNING[25360]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission 14f926787256cb1d2a68e5b623f6bd5c@192.168.0.106 for seqno 102 (Critical Response) |
| 18:26.14 | Katty | i'm playing with hud lite right now |
| 18:26.21 | MrChimpy | yeah, googling sugarcrm works |
| 18:26.24 | Katty | sugarcrm just needs to wait a minute |
| 18:27.01 | *** join/#asterisk ManxPower (n=manxpowe@247.sub-70-221-16.myvzw.com) |
| 18:27.11 | gk1 | DR-Linux: I get those quite often as well |
| 18:27.28 | gk1 | Dr-Linux: its something to do with your nat settings somewhere |
| 18:27.57 | Dr-Linux | gk1: while call in queue, it doesn't repeat according to the time settings and hangs after give me this shit |
| 18:28.39 | Dr-Linux | gk1: where NAT involved? :S |
| 18:28.49 | gk1 | Dr-Linux: Doesnt matter where the call is queue or not, you get that message call is dropped. Check you nat settings, you should be getting a bunch of net retrans messages as well |
| 18:29.18 | Dr-Linux | gk1: NAT settings where?? |
| 18:29.22 | Dr-Linux | do you mean, in sip.conf? |
| 18:29.58 | ManxPower | usually NAT somwhere else, I imagine |
| 18:30.26 | gk1 | could be in sip.conf |
| 18:30.46 | gk1 | just check the settings for the phone that has the problem. or do they all have the problem? |
| 18:30.47 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 18:30.54 | ManxPower | there is only nat=yes or nat=no |
| 18:30.59 | gk1 | where is your server in relation to the phones |
| 18:31.05 | gk1 | same lan segment? |
| 18:31.07 | Dr-Linux | gk1: this problem is with all clients |
| 18:31.11 | ManxPower | unless asterisk is behind NAT of course |
| 18:31.18 | Dr-Linux | gk1: and i don't think it's NAT issue |
| 18:31.24 | gk1 | thats what i am thinking |
| 18:31.25 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 18:31.36 | gk1 | is the server and the phones on the same lan subnet? |
| 18:31.45 | ManxPower | Dr-Linux: Max retries exceeded means "the far end stopped respoiding" That is usually a NAT issue |
| 18:31.48 | Dr-Linux | gk1: my asterisk server and phones are located on same subnet local lan |
| 18:31.55 | MrTelephone | Can someone maybe explain why asterisk will not accept a call from-pstn with these lines? |
| 18:31.58 | MrTelephone | exten => s,1,Set(CALLERID(num)=${IF(${REGEX("^807[229,822,826,825,868]...." ${CALLERID(num)})}?${CALLERID(num):3}:1${CALLERI |
| 18:31.58 | MrTelephone | exten => s,2,Goto(dids,${CALLERID(dnid)},1) |
| 18:32.18 | ManxPower | MrTelephone: the CLI will tell you. |
| 18:32.35 | gk1 | Dr-Linux: what type of phones? |
| 18:32.46 | MrTelephone | the telco company says the number is out of service but the number is listed in [dids] context as a valid extension |
| 18:32.46 | ManxPower | at least the CLI will tell you what those things evaluate to. |
| 18:32.49 | Vec2 | When a call comes in to asterisk I have set it to, transfer the call after X seconds to my mobile, however the CDR does not show the dialed number has my cellphone, only shows the incomming number, does anyone have a solution to this ? |
| 18:32.50 | Dr-Linux | gk1: cisco's and softphones |
| 18:33.01 | gk1 | Dr-Linux: i see it mostly with AAstra and Poly's |
| 18:33.08 | ManxPower | MrTelephone: But your main issue is totally not understanding the function of extension "s" |
| 18:33.09 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:33.36 | Dr-Linux | gk1: hhm.. actually all these phones are being logged in to the queue via agent callback login |
| 18:33.43 | MrTelephone | Manxpower, I agree with you there |
| 18:33.53 | Dr-Linux | gk1: do you think think somewhere agents.conf invovled? |
| 18:33.54 | ManxPower | Extension "s" is only run if you have 1) immediate=yes 2) a loopstart or groundstart FXO port. |
| 18:34.01 | ManxPower | or 3) there is a Goto |
| 18:34.17 | ManxPower | Dr-Linux: max retries is a protocol and networking issue |
| 18:34.37 | Dr-Linux | ahhmmm, |
| 18:34.43 | ManxPower | MrTelephone: If Asterisk receives the dialed number then extension "s" will NEVER be run automatically |
| 18:34.51 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:35.03 | MrTelephone | ok then just use wildcards instead? |
| 18:35.12 | Dr-Linux | ManxPower: actually, i'm making call from asterisk server1 to asterisk server2 and all the clients are on lan with asterisk server2 |
| 18:35.14 | ManxPower | MrTelephone: USe whatever you want, but don't use "s" |
| 18:35.20 | MrTelephone | what is the recommended wildcard to use for any number? |
| 18:35.21 | gk1 | Dr-Linux: its a major sip issue, for some reason I see it a lot more with 1.4.x than with 1.2.x |
| 18:35.24 | MrTelephone | .*? |
| 18:35.33 | ManxPower | MrTelephone: that is not a wildcard |
| 18:35.48 | Dr-Linux | and both asterisk server are connected with each other with IAX2 , sip connection gives me same issue though |
| 18:35.48 | ManxPower | MrTelephone: are all your DIDs the same number of digits? |
| 18:35.53 | MrTelephone | yeah |
| 18:36.01 | ManxPower | MrTelephone: how many digits? |
| 18:36.05 | MrTelephone | 7 |
| 18:36.06 | Dr-Linux | gk1: i never seen 1.4 |
| 18:36.09 | MrTelephone | XXXXXXX |
| 18:36.23 | [TK]D-Fender | MrTelephone: .... |
| 18:36.28 | [TK]D-Fender | ~osmosis |
| 18:36.41 | jbot | osmosis is, like, the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:36.41 | ManxPower | you could get a number that is 1551212? |
| 18:36.41 | gk1 | Dr-Linux: do you have canreinvite set for your phones in sip.conf? |
| 18:36.41 | ManxPower | perhaps you want a more specific pattern match |
| 18:36.44 | ManxPower | like _NXXXXXX,1,whtever |
| 18:36.57 | MrTelephone | right |
| 18:37.05 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 18:37.19 | Dr-Linux | gk1: yes, that's set to no |
| 18:37.24 | Dr-Linux | canreinvite=no |
| 18:37.30 | ManxPower | As I;m sure you know _. would match non-number extensoins like o,T,t,a,etc |
| 18:37.40 | ifnotwhynot | dies anyone had any luck setting up asterisk recording interface ARI? |
| 18:37.49 | gk1 | Dr-Linux: thats werid |
| 18:38.00 | MrTelephone | the telco will only route 7 digit calls destined for my group of allocated pstn numbers so I should be save to _X. it |
| 18:38.02 | Dr-Linux | gk1: i'm sure somewhere agents.conf invovled |
| 18:38.03 | ManxPower | gk1: I would say a firewall issue on the linux boxes |
| 18:38.26 | gk1 | Firewall is possible |
| 18:38.27 | ManxPower | MrTelephone: you just told me that all DIDs are 7-digits. |
| 18:38.31 | ManxPower | STOP TRYING TO USE OVERLY BROAD WILDCARDS. |
| 18:38.34 | ManxPower | They will fuck you up. |
| 18:38.43 | gk1 | Dr-Linux: is iptables enabled on you rmachines? |
| 18:38.55 | MrTelephone | sorry my mistake |
| 18:39.10 | Dr-Linux | bcoz i'm facing this with only callback agents, my other sip clients are just fine |
| 18:39.11 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:39.17 | [TK]D-Fender | ManxPower: He didn't know what a proper pattern match was, do you seriously think he know how _. will match oshitafax?! |
| 18:39.25 | Katty | cool. |
| 18:39.28 | Katty | i love how people are nice to me. |
| 18:39.33 | Katty | they're zomg, a girl just said asterisk |
| 18:39.35 | Dr-Linux | gk1: where iptables? what port you are talking about? |
| 18:39.35 | Katty | *faint* |
| 18:39.46 | MrTelephone | ok then I'll use _229XXXX |
| 18:40.08 | gk1 | Dr-Linux: probably not the case as the rest of your sip phones are ok |
| 18:40.15 | ManxPower | MrTelephone: Much better |
| 18:40.40 | ManxPower | then the only things that will conflict with are extensions that start with 229 and you don't have any of those, right? |
| 18:40.43 | Dr-Linux | gk1: yes, |
| 18:40.46 | MrTelephone | then I can use ${EXTEN} instead of $CALLER(dnid) |
| 18:40.52 | Dr-Linux | something wrong with agents callback login |
| 18:40.53 | ManxPower | [TK]D-Fender: With a nick like "MrTelephone" you'd think he would know some basic telecom |
| 18:40.59 | MrTelephone | manxpower, no |
| 18:41.03 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 18:41.09 | ManxPower | MrTelephone: ${EXTEN} holds the currently executing extension number. |
| 18:41.38 | ManxPower | CALLERID(dnid) holds the originally dialed number. Usually they will be the same unless you use Gotos or IVRS |
| 18:41.41 | gk1 | Dr-Linux: never used agent callback so I couldnt tell you the mechanism it uses to connect, and why you get the errors you get. |
| 18:41.55 | MrTelephone | very good this should work well then |
| 18:42.18 | Dr-Linux | gk1: no problem and thanks for the discussion, i'm talk to you later to put light on more stuff. Bye for now |
| 18:42.21 | Dr-Linux | /gone |
| 18:42.31 | gk1 | goodluck!!! |
| 18:42.54 | ifnotwhynot | can anyone help me with setting up asterisk recording interfase please |
| 18:42.55 | ifnotwhynot | ? |
| 18:43.06 | Katty | [TK]D-Fender: i bribed sugarcrm into doing a webdemo with me. |
| 18:43.19 | ManxPower | ifnotwhynot: there is no such thing as the asterisk recording interface. |
| 18:43.27 | [[blah]asfd | ifnotwhynot: install asterisk then use the command monitor in the dial plan |
| 18:43.32 | vAd0r | I have been trying to get my linksys pap2 to work w/ my asterisk. I keep trying to unlock it. i have went to the web page and changed the user password to 1234. I then relog into the router w/ user/1234 I try to run the tftp to it and then it prompts me for another login which i am assuming is the admin one. I can not get this thing unlocked. Please help. |
| 18:43.32 | [[blah]asfd | all done |
| 18:43.55 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 18:44.00 | ManxPower | vAd0r: you can't unlock the Vonage PAP2s |
| 18:44.47 | ManxPower | they block admin logins, block TFTP updates, and block factory resets |
| 18:45.22 | vAd0r | what about w/ a console cable |
| 18:45.38 | Qwell[] | note to self: Do not take sudafed with water.. for some reason, water makes it desolve instantly. |
| 18:45.43 | ManxPower | vAd0r: where is the console cable port on the PAP2s? |
| 18:45.55 | Qwell[] | note to Pfizer: Do not make medicine taste bad |
| 18:45.57 | vAd0r | could i soldier one on? |
| 18:46.12 | punani | if you can find a soldier, give it a try |
| 18:46.14 | ManxPower | vAd0r: there IS no console port. |
| 18:46.35 | killfill_ | <PROTECTED> |
| 18:46.35 | killfill_ | <PROTECTED> |
| 18:46.52 | ManxPower | vAd0r: there have been rumors that you can unlock them if they have never been connnected to the internet |
| 18:46.54 | killfill_ | whats this?.. zaptel is hanging out my outgoing calls... :S |
| 18:46.59 | killfill_ | (te110p card) |
| 18:47.01 | ManxPower | killfill_: and you know what the next step is, right? |
| 18:47.36 | ManxPower | find out the values of DIALSTATUS and HANGUPCAUSE using Noops in the priority after the Dial |
| 18:47.49 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 18:48.01 | killfill_ | ah |
| 18:49.20 | btsteve | i am getting Response 1: Match Not Found when out switch places a sip call to asterisk for voicemail. does anyone have any idea where i should look to correct this? |
| 18:50.12 | neverblue | anyone have alot of experience with both Ekiga and Twinkle? |
| 18:51.42 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D493.access.telenet.be) |
| 18:51.55 | ManxPower | vAd0r: you are welcome to keep beating your head against the wall, but at be least polite enough not to waste our time when you are doing so. |
| 18:51.59 | FaUl | er |
| 18:51.59 | FaUl | re |
| 18:52.03 | Qwell[] | neverblue: You're never gonna get a response like that |
| 18:52.28 | FaUl | [TK]D-Fender: any idea that i missed? |
| 18:52.31 | ManxPower | btsteve: that would usually indicate a context or dialplan issue |
| 18:52.58 | [TK]D-Fender | FaUl: ah yes.. the "hidecallerid=yes" in zapata.conf is HIGHLY suspect, and I'd look at that localdialpl = unknown too... |
| 18:53.22 | FaUl | i played on that too, but it made no difference |
| 18:53.48 | neverblue | ill take my chances :) |
| 18:53.56 | Qwell[] | ~ask |
| 18:54.08 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:54.42 | [TK]D-Fender | FaUl: Could simply be that your telco has not actually allowed you to se it... |
| 18:55.01 | MrTelephone | hey manxpower thanks for the advice I appreciate your help |
| 18:55.25 | neverblue | so you feel my question is not specific, informative, complete or concise? |
| 18:55.35 | neverblue | or just off topic? |
| 18:55.42 | FaUl | [TK]D-Fender: they said |
| 18:55.49 | ManxPower | neverblue: not even close to being nformative, complete or concise |
| 18:55.52 | [TK]D-Fender | FaUl: they LIE ;) |
| 18:55.56 | FaUl | [TK]D-Fender: they even switched clip no screening on and it did not work either |
| 18:56.01 | neverblue | its concise and complete |
| 18:56.21 | ManxPower | It is vague and open ended. |
| 18:56.25 | Qwell[] | neverblue: it's incomplete, because there will be a followup question when somebody says "yes" |
| 18:56.29 | Qwell[] | incredibly incomplete |
| 18:56.37 | neverblue | lol |
| 18:56.39 | sheldonh | neverblue: but suppose the answer is "yes", what good does that do you. just ask the question you would ask if there _were_ people with experience of these things here |
| 18:56.41 | Qwell[] | sure, if that's ALL you want to know, is whether somebody knows them... |
| 18:56.41 | btsteve | if i am setting up a server that is going to do voicemail only for sip connections what should i configure for my contex or dialplan. i need the box to answer with out ay added delay |
| 18:56.43 | Qwell[] | then, yes, I do |
| 18:56.46 | Qwell[] | but now bbl |
| 18:56.46 | neverblue | why are we even having this discussion? |
| 18:56.52 | ManxPower | consuse and complete would be "Ekiga is sending a SIP 100 Proceeding, but asterisk is not giving a ringing sound to the Polycom 500 that is making the call" |
| 18:56.57 | neverblue | and look, its three ppl talking about it now |
| 18:56.58 | neverblue | lol |
| 18:57.04 | [TK]D-Fender | btsteve: ... |
| 18:57.06 | [TK]D-Fender | ~book |
| 18:57.16 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:57.16 | [TK]D-Fender | ~wikis |
| 18:57.19 | jbot | it has been said that wikis is http://www.voip-info.org |
| 18:57.19 | ManxPower | btsteve: nothing special. |
| 18:57.22 | MrTelephone | qwell where should I investigate if I have a problem with rtp streams being barged into for brief moments when I try and make an outgoing call? |
| 18:57.41 | ManxPower | MrTelephone: first define "barded into" |
| 18:57.42 | MrTelephone | Do you think thats a pri card issue? |
| 18:57.51 | ManxPower | ZapParge is an Asterisk application |
| 18:58.15 | MrTelephone | On odd calls outgoing you can briefly hear a short audio clip of what I am assuming is another rtp stream |
| 18:58.15 | neverblue | anyone who actually replied to me, Qwell, sheldonh or ManxPower, have experience with both Ekiga and Twinkle? |
| 18:58.16 | tzafrir_laptop | ZapBarge |
| 18:58.21 | ManxPower | Is the solution to your question is "don't run ZapBarge" on your outgoing calls. |
| 18:58.22 | Qwell[] | neverblue: yes, I do |
| 18:58.28 | sheldonh | <PROTECTED> |
| 18:58.33 | tzafrir_laptop | twinkle is nice |
| 18:58.34 | ManxPower | MrTelephone: the CLI is your friend. |
| 18:58.38 | neverblue | ok, we are using an older version of twinkle |
| 18:58.48 | MrTelephone | I'm not getting any errors or anything that I can see but I can check again |
| 18:59.00 | neverblue | so Qwell we are thinking of either updating to the latest, or moving onto Ekiga |
| 18:59.03 | MrTelephone | is it possible to log more verbosely to /var/log/asterisk/messages? it only logs warnings and notices |
| 18:59.05 | ManxPower | MrTelephone: I did not say "errors" |
| 18:59.12 | Qwell[] | neverblue: I've never used either. |
| 18:59.16 | ManxPower | MrTelephone: /etc/asterisk/logging.conf |
| 18:59.19 | Qwell[] | You simple asked if anybody had experience with them. |
| 18:59.20 | neverblue | do they perform closely? |
| 18:59.24 | Qwell[] | simply* |
| 18:59.30 | MrTelephone | Manxpower, i should look for bridged channel commands etc? |
| 18:59.51 | ManxPower | MrTelephone: I think it is time to step away from Asterisk ans read the damn book. You are wasting everyone's time with newbie questions, most of which you should find the answers to in the book. |
| 18:59.55 | neverblue | you ppl take the "fun" outta support |
| 19:00.16 | ManxPower | neverblue: have you ever been in a tech support call center? |
| 19:00.27 | ManxPower | I should say "have you ever worked in a tech support call center"? |
| 19:00.30 | neverblue | ManxPower, I really dont feel like answering you |
| 19:00.44 | neverblue | do your thing... |
| 19:00.52 | ManxPower | I'll take that as a "No." |
| 19:01.03 | ManxPower | neverblue: I help people that are not idiots. That is what I do. |
| 19:01.03 | neverblue | whine, complain, show me your perspective, which ever you choose |
| 19:01.24 | neverblue | ah, the "your perspective" response |
| 19:01.28 | ManxPower | I do actually help idiots, but not for free. |
| 19:01.28 | MrTelephone | manxpower i know you are very filmiliar with asterisk but if you came into a cable operators channel asking questions about db levels and rf noise I wouldn't treat you poorly |
| 19:01.29 | neverblue | hmm, never heard that before |
| 19:01.53 | ManxPower | MrTelephone: I'[ll bet you would if you got asked that questions 10 times a day. |
| 19:02.00 | MrTelephone | maybe |
| 19:02.57 | ManxPower | Of course "db levels" is a very broad topic. It does not take into account Slope or noise or ingress RF or any of those things. |
| 19:02.57 | MrTelephone | you overworked and should take a 10 minute breather |
| 19:02.57 | MrTelephone | :P |
| 19:03.11 | ManxPower | Sure saying "The video signal should be about -5db" (or whatever it should be) tells you almost nothing you need to know, but does answer the question. |
| 19:03.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 19:03.45 | MrTelephone | your filmiliar with cable networks, thats cool too :P |
| 19:03.51 | ManxPower | Or perhaps it would be like someone asking "How much loss will I get over 450ft of coax"? |
| 19:03.52 | neverblue | Qwell[]> neverblue: I've never used either. <-- doesnt that imply you have no experience with it then? |
| 19:03.54 | Qwell[] | MrTelephone: that certainly backfired |
| 19:04.02 | Qwell[] | neverblue: I have plenty of experience with them |
| 19:04.06 | neverblue | oh really? |
| 19:04.12 | neverblue | why the mixed responses? |
| 19:04.12 | [TK]D-Fender | "All answers are responses but not all responses are ANSWERS" <- |
| 19:04.13 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D493.access.telenet.be) |
| 19:04.20 | Qwell[] | neverblue: because they're two different questions |
| 19:04.23 | [TK]D-Fender | SEE ABOVE |
| 19:04.34 | Qwell[] | ~ask |
| 19:04.42 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:04.42 | ManxPower | MrTelephone: I know far too little about CATV. |
| 19:04.42 | MrTelephone | Qwell, i'm not offended because manxpower has a broad knowledge of other fields other than telephony |
| 19:04.46 | MrTelephone | I'm impressed |
| 19:04.49 | neverblue | u guys really need to join a support channel |
| 19:04.57 | neverblue | and see how its really done |
| 19:05.06 | ManxPower | MrTelephone: But when I was starting the micro-cableco, I did not come on IRC asking questions, I started reading. |
| 19:05.17 | Qwell[] | neverblue: we aren't being paid |
| 19:05.18 | neverblue | because this channel is the furthest from support I have ever seen |
| 19:05.24 | [TK]D-Fender | neverblue: www.drphil.com ... all the support you need! |
| 19:05.26 | neverblue | no one on irc is being paid |
| 19:05.30 | neverblue | but guess what |
| 19:05.34 | neverblue | they have support channels |
| 19:05.35 | ManxPower | Yes, Asterisk is a very complex and hard to learn system, but the asterisk book is a good place to start. |
| 19:05.41 | neverblue | and they are ACTUALLY good at it |
| 19:05.42 | MrTelephone | manxpower, yeah I do a lot of reading too but its nice to come online and get a couple quick answers |
| 19:05.43 | neverblue | lmao |
| 19:05.52 | ManxPower | MrTelephone: that is true. |
| 19:06.01 | ManxPower | but you are missing out on SO much. |
| 19:06.06 | neverblue | Fender, you have my permission to stop acting like your 5 |
| 19:06.08 | *** kick/#asterisk [neverblue!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Go to the other support channels then. You're being disruptive.) |
| 19:06.31 | jsolares | :S |
| 19:06.35 | jsolares | heh |
| 19:06.39 | *** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 19:06.44 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 19:06.47 | neverblue | lmao |
| 19:06.58 | neverblue | dish it out, but not able to take it? |
| 19:07.16 | ManxPower | MrTelephone: I run a 15 channel MATV/CATV system for a campground. |
| 19:07.52 | [TK]D-Fender | Lets just say our respect for people falls like the 1929 NYSE when we realize that some people will not read the basic stuff given to them and expect EVERYTHING to be fed to them. There is a difference between "helping you" and "telling you every little thing because you are a COMPLETE lazy ass". |
| 19:08.13 | neverblue | oh, how the truth hurts people |
| 19:08.30 | [TK]D-Fender | neverblue: You don't seem terribly fazed ;) |
| 19:08.32 | ManxPower | Light a fire for a man and you keep him warm for 1 night. Light a man on fire and keep him warm for the rest of his life. |
| 19:08.37 | neverblue | no, I dont ;) |
| 19:08.46 | [TK]D-Fender | jm|laptop: BURN HIM! ;) |
| 19:08.52 | jsolares | lol |
| 19:09.28 | neverblue | see still acting like a child.... |
| 19:09.29 | ManxPower | MrTelephone: on the otherhand, if you had old (but working) cable equipment that I can have, you'd be suprized at how much hand holding I'm willing to do. I would not even require dinner and drinks first! |
| 19:10.42 | ManxPower | neverblue: and oddly he very well may be the only person that can help you./ |
| 19:10.47 | [[blah]asfd | i am trying to figure out how to use md5 with the iax.conf. I am currently using plaintext. I see that I can do auth=md5 and that I can generate md5 check sums in linux, but I am not following how to do it all. I have been reading in the sample section of iax.conf. I am confused and help would be appreciated. |
| 19:10.51 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 19:11.15 | neverblue | thats is advantage, and he lets everyone know it :) |
| 19:11.36 | ManxPower | neverblue: and yet, most people seem to get along with him just fine. |
| 19:12.38 | [TK]D-Fender | [[blah]asfd: I believe that MD5 tells you system NOT to encode as MD5 because its already encoded. |
| 19:12.45 | neverblue | ManxPower, why are you even talking to me? |
| 19:12.57 | neverblue | are you trying to make a point? |
| 19:13.35 | [[blah]asfd | [TK]D-Fender: that is one thing that confused me. is iax sending the pass info encrypted already? |
| 19:13.37 | anonymouz666 | anyone in here know a way to debug the manager commands? |
| 19:13.48 | festr__ | anonymouz666: tethereal |
| 19:14.03 | festr__ | anonymouz666: tethereal or tshar 'port manager' -V |
| 19:14.05 | festr__ | :) |
| 19:14.19 | MrTelephone | maxpower what are you looking for in equipment? |
| 19:14.20 | anonymouz666 | hmm |
| 19:14.24 | [TK]D-Fender | neverblue: I've spent a ridiculous amount of time helping people here, including you on many occasions. If you can't take an obvious jest or two don't go telling me I'm a stiff or anything :) |
| 19:14.28 | anonymouz666 | ok |
| 19:14.50 | [TK]D-Fender | neverblue: so "live and let live", show you're making an effort and peace, ok? |
| 19:14.52 | festr__ | anonymouz666: but there should be more easy way i think |
| 19:15.26 | [TK]D-Fender | [[blah]asfd: your "secret" should ALREADY be MD5 encoded... thats so its not plaintext in your CONFIGS FILES. |
| 19:15.31 | MrTelephone | I have a bunch of dsr-4400s that lost their receiver ID and everyone says I have to ship to mexico to get them fixed |
| 19:15.32 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 19:15.37 | anonymouz666 | festr__: i don't know the easy way :) but its ok I can do it using ngrep |
| 19:15.54 | [TK]D-Fender | [[blah]asfd: in-somuch as yeah its sorta "there", but at least its not basic alpha, etc :) |
| 19:16.16 | festr__ | anonymouz666: and what about set debug verbose to higher levels? |
| 19:17.18 | irule | how can I prevent the error 'Auto fallthrough channel SIP status is NOANSWER' when dialing ffrom the samples macro-stdexten? it has an s-NOANSWER to voicemailmain, so, whats up? |
| 19:17.31 | anonymouz666 | festr__: nada |
| 19:18.05 | [TK]D-Fender | irule: Auto-fallthrough says you ran out of dialplan at a certain point, and nothing maginally makes * jump to an exten like that. |
| 19:18.30 | btsteve | do i need to define our switch as an incoming trunk so that the asterisk can act as voicemail for it? |
| 19:18.34 | MrTelephone | s-${DIALSTATUS} makes it goto that extension |
| 19:18.34 | [TK]D-Fender | irule: PASETBIN output & code any time you bring this sort of stuff up ok? Descriptions are often of little sue. |
| 19:18.38 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 19:18.43 | btsteve | it is sip only |
| 19:19.11 | [TK]D-Fender | MrTelephone: Want to ammend you answer to something complete and useful? :) Its missing a few things to be "help".... |
| 19:19.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:19.34 | MrTelephone | thats all I could think of at the time :( I will try harder |
| 19:19.48 | [TK]D-Fender | btsteve: You can set * to take un-auth'd calls if you have it locked down security-wise |
| 19:20.00 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 19:20.25 | [TK]D-Fender | MrTelephone: You forget the GOTO part at the beginning. "s-${DIALSTATUS}" alone isn't complete "answer" |
| 19:20.28 | btsteve | Sorry i am new to asterisk so i am not sure is i have it locked down. |
| 19:21.06 | [TK]D-Fender | btsteve: its your network, just think what could try talking to your * box directly. that is the level of exposure you are open to us you DON'T setup auth credentials. |
| 19:21.22 | MrTelephone | btsteve, i think he means putting asterisk on a private network, or using permit statements to only allow connections from specific IP's |
| 19:21.42 | [TK]D-Fender | btsteve: Indeed that is one aspect I am talking about. |
| 19:21.55 | [TK]D-Fender | btsteve: as MrTelephone said. |
| 19:21.57 | neverblue | *i have no comment* |
| 19:22.03 | [TK]D-Fender | MrTelephone: BETTER. |
| 19:22.04 | btsteve | it is on a firewalled connection. |
| 19:22.25 | irule | http://pastebin.ca/519779 this is it |
| 19:22.35 | [TK]D-Fender | btsteve: try setting up a peer/user setup on it first and see if you can do it the "normal" way first. |
| 19:22.38 | btsteve | and the switch is on the same firewalled connection. users will not connect directly they are e-mail their voice mail |
| 19:22.50 | neverblue | anyone have alot of experience with both Ekiga and Twinkle? |
| 19:23.25 | [TK]D-Fender | irule: I am NOT going through 300+ lines a crap to hunt down the relevent bits.... redo it please.... |
| 19:23.35 | [TK]D-Fender | 3000+ * |
| 19:24.03 | MrTelephone | maxpower likes to read- |
| 19:24.05 | MrTelephone | :P |
| 19:24.14 | irule | 5079 actually lol ...on its way... |
| 19:24.25 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:24.57 | Nugget | I gave up on chan_skinny, loaded the sip firmware on, and now it's unable to register with asterisk. |
| 19:25.12 | [TK]D-Fender | MrTelephone: Yes he likes to read.... USEFUL stuff that tells him things that will help him. That is neither ;) |
| 19:25.35 | [TK]D-Fender | Nugget: ..... Cisco is POO ;) |
| 19:25.41 | Nugget | indeed |
| 19:26.01 | MrTelephone | nugget: you got it to download the SIP firmware? do you have your tftp server with the config on it? |
| 19:26.16 | Nugget | yeah, I'm past all that. |
| 19:26.25 | Nugget | it's sending a register sip packet that asterisk doesn't like |
| 19:26.39 | MrTelephone | hmm I remember having a similar problem |
| 19:26.42 | MrTelephone | one sec |
| 19:26.47 | Nugget | asterisk is coming back with an unauthorized response |
| 19:26.54 | yannj_fr | I searching a solution to organise a find me, I mean someone can log on every phone with his sip account by calling an extension |
| 19:27.03 | yannj_fr | does any one have an idea |
| 19:27.18 | MrTelephone | there is a register = 1 or yes in the config file that you should make sure is set |
| 19:27.31 | Nugget | ah, sounds promising! |
| 19:27.49 | Nugget | <PROTECTED> |
| 19:27.52 | Nugget | I've got that. |
| 19:27.58 | Nugget | maybe "true" isn't what I need in there |
| 19:28.00 | sheldonh | does astbill work with php5? i've got conflicting answers from google |
| 19:28.48 | Nugget | hrm, no, I think that's fine. from the phone itself the configuration menu indicates "register with proxy: yes" |
| 19:29.30 | MrTelephone | I wasn't using xml configs myself :-/ |
| 19:30.05 | Nugget | http://lists.digium.com/pipermail/asterisk-users/2006-July/158364.html <-- that's exactly the problem I'm having |
| 19:31.00 | irule | http://pastebin.ca/519806 there |
| 19:31.09 | MrTelephone | yeah its definitely not the same problem I'm having nugget.. Mine was nat related |
| 19:31.18 | irule | it is only 1600 lines |
| 19:31.48 | yannj_fr | I searching a solution to organise a find me, I mean someone can log on every phone with his sip account by calling an extension, an idea? |
| 19:32.40 | MrTelephone | Nugget, can you use an older version like 7.4? |
| 19:33.01 | [TK]D-Fender | irule: Why am I seeing another 1000 lines when all I need is one bloody context from extensions.conf?!?! |
| 19:33.15 | *** join/#asterisk jeffgus (n=jeffgus@marlene.zimage.com) |
| 19:33.19 | btsteve | we have out switch set up as a peer in the sip.conf file, and it loads with out any problem. |
| 19:33.57 | irule | goold point lol |
| 19:34.05 | [TK]D-Fender | irule: we're trying to fix your dialplan errors, you should really wake up and stop pastbein 15 config files we DON'T CARE ABOUT. |
| 19:34.14 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:34.44 | MrTelephone | I wish XML didn't become so popular.. |
| 19:35.03 | Qwell[] | xml rocks |
| 19:35.13 | [TK]D-Fender | Katty: As Greek Gods go.... Hera might not be the best choices for a name... she spent a lot of her time trying to kill her relatives ;) |
| 19:35.15 | MrTelephone | html rocks too |
| 19:35.23 | yannj_fr | Does any one use snom phones with autoprovisionning |
| 19:36.02 | Katty | [TK]D-Fender: hera's exchange. i thought it was fitting. |
| 19:36.02 | irule | well I am not actually trying to annoy you, I just asumed you could press control-F to find macro-standarext |
| 19:36.02 | [TK]D-Fender | Katty: I see your point.... |
| 19:36.02 | Nugget | MrTelephone: There is no such animal for 79x1 phones. |
| 19:36.02 | irule | but I am reposting as we read |
| 19:36.03 | Nugget | according to the wiki, people are successfuly using these phones with the current firmware. |
| 19:36.07 | MrTelephone | nugget my application says POS3-07-4-00 |
| 19:36.13 | Nugget | nobody's been kind enough to post a working config, though. |
| 19:36.18 | Nugget | MrTelephone: that's for 79x0s |
| 19:36.21 | [TK]D-Fender | irule: If you think I'm going to bust my ass while you're being lazy and flooding me with crap you've got another thing coming.... |
| 19:36.38 | irule | i font have bad intentions |
| 19:36.41 | irule | dont |
| 19:36.44 | *** join/#asterisk blindluck9 (n=jeremiah@west-rock.rockriver.net) |
| 19:36.53 | [TK]D-Fender | irule: "God helps those who help themselves". I'm CONSIDERABLY less forgiving.... |
| 19:37.22 | [hC] | Anyone using agents where the system will call them and require them to hit '#' to take the call? Is this just the ackcall setting in agents.conf? |
| 19:37.39 | Katty | [TK]D-Fender: is this syntax right? exten => _xxx,1,Dial(SIP/${EXTEN}@context,20wW) |
| 19:37.49 | [TK]D-Fender | [hC]: Ackcall is only for agents logged in with "AgentLogin" |
| 19:37.55 | MrTelephone | nugget, can you see the phone register when your using sip debug <peer> |
| 19:37.58 | Nugget | yes |
| 19:37.59 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 19:38.15 | [hC] | [TK]D-Fender: Ah. Is there any way using callbacklogin to have the system call the person and have them accept the call before they are connected? |
| 19:38.16 | [TK]D-Fender | [hC]: if you want it acked, mod one of those call-screening Macro's into your dilaplan for your agents |
| 19:38.19 | Nugget | it sends a register which has no Authorization: header, then asterisk tells the phone to go pound sand. |
| 19:38.32 | [hC] | [TK]D-Fender: example? url? |
| 19:38.33 | MrTelephone | heh |
| 19:38.53 | [TK]D-Fender | [hC]: Samples on the WIKI, get Googling :) |
| 19:38.58 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 19:39.09 | [hC] | [TK]D-Fender: kay :) |
| 19:39.16 | [TK]D-Fender | Katty: missing a "," betwee 20 and Ww |
| 19:39.35 | *** join/#asterisk visba (n=dca[lapt@c-67-166-17-228.hsd1.co.comcast.net) |
| 19:39.43 | [TK]D-Fender | Katty: and "wW" means EITHER side of the call can initiate on-demand call recroding through features.conf |
| 19:40.04 | neverblue | Fender self taught, or did you take a course? |
| 19:40.09 | Katty | okies. |
| 19:40.21 | wwalker | I've been using the purchased from digium 729 codec for months and today "show g729" reports "No such command". "show modules" reports that the codec_g729a.so is loaded. |
| 19:40.26 | MrTelephone | nugget, try using insecure=very in sip.conf until you find out what config option your missing |
| 19:40.34 | Qwell[] | wwalker: You should call Digium support |
| 19:42.06 | [TK]D-Fender | neverblue: in *, self taught. "show application [whatever]", sample files, and the Wiki. Thats it. I RTFM. |
| 19:42.42 | neverblue | u seem like a self taught kinda person |
| 19:43.11 | [TK]D-Fender | neverblue: Oh and I harly ever skimmed the book even though i refer people to it. Once I know the answer is there its shameful that people can't read what they're given when I know the answer is sitting right there. |
| 19:43.21 | [TK]D-Fender | neverblue: God gave me eyes... I'm using them... |
| 19:43.34 | neverblue | the book is good, but not perfect |
| 19:43.45 | [TK]D-Fender | neverblue: Quite true. |
| 19:43.50 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 19:43.55 | neverblue | some people learn better from reading |
| 19:44.01 | neverblue | some dont |
| 19:44.08 | [TK]D-Fender | neverblue: I refer people to the correct place based on their questions. You won't see me refer someone to the book if its not in there. |
| 19:44.16 | neverblue | hehe |
| 19:44.25 | [TK]D-Fender | neverblue: And most people never get off their ass to read. this is the MAJORITY. |
| 19:44.26 | neverblue | u use the ~book too much :P |
| 19:44.37 | neverblue | u should run a stats website |
| 19:44.43 | [TK]D-Fender | neverblue: More like people don't read it ENOUGH :) |
| 19:45.04 | yannj_fr | Does any one use snom phones with autoprovisionning? |
| 19:45.07 | [TK]D-Fender | neverblue: cause & effect. |
| 19:45.09 | jkiff | Hey ya'll. How do you use the pickup feature? (i.e., pickupexten=*8 in features.conf.) If 299 calls 207, and 213 wants to pick it up, how is that done? *8 from 213 is giving me busy signal, and a "NOTICE[3760]: chan_sip.c:10675 handle_request_invite: Nothing to pick up". Is there something I need to include in my dialplan? Am I just retarded? :-P |
| 19:45.11 | neverblue | i can read a book three times, and lose well over 50% of it in a day |
| 19:45.19 | punani | wwalker: show translation ? |
| 19:45.22 | neverblue | i need to get my hands dirty |
| 19:45.28 | neverblue | and I like to break things |
| 19:45.36 | Katty | [TK]D-Fender: i should add you to my speed dial. |
| 19:45.38 | Katty | file: and you too |
| 19:45.50 | Qwell[] | Katty: well, I never |
| 19:45.50 | Katty | file: oh. i already have you in sd |
| 19:46.11 | Katty | Qwell[]: never what? |
| 19:46.13 | Katty | Qwell[]: :P |
| 19:46.21 | Qwell[] | umm |
| 19:46.25 | Qwell[] | I've never done a lot of things :p |
| 19:46.37 | jsolares | jkiff: you should have something like exten => _*8XXX,1,Pickup(${EXTEN:2}) and then you dial *8 and the extension you want to pickup, *8207 |
| 19:46.40 | Katty | s'ok to be inexperienced, Qwell[] ;) |
| 19:46.45 | file | yay Katty |
| 19:46.51 | Corydon76-work | Qwell is plenty experienced... |
| 19:46.52 | Nugget | ah, it's a NAT thing. |
| 19:47.02 | jsolares | i think there's also pickup groups so that you can pickup with just *8, i haven't gotten there yet tho |
| 19:47.07 | Nugget | I wonder if it's an rport range problem or something |
| 19:47.38 | Qwell[] | Corydon76-work: should ast_storage_*->get() really return an int? I assume it's supposed to return an fd... |
| 19:47.41 | Corydon76-work | Katty: he's an expert spoonologist |
| 19:47.57 | Katty | Corydon76-work: that's good to know. |
| 19:47.58 | *** join/#asterisk forrestv (n=forrestv@c-75-74-100-18.hsd1.fl.comcast.net) |
| 19:48.05 | Qwell[] | There is no spoon. |
| 19:48.16 | jkiff | jsolares: Hmm I see. If you have to do that, then what's the point of the line in features.conf? |
| 19:49.08 | jsolares | if they're sip, use pickupgroup=1 for all of them and try just *8 |
| 19:50.42 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 19:50.49 | samy_b1 | hey gys can some one tell me why i'm geting " 302 Moved Temporarily " msg when i try to call out ? |
| 19:50.50 | [hC] | [TK]D-Fender: so, I found an example of call screening. I presume that if you send Dial() into a macro using the M() argument, that it waits for the macro to complete before determining that you've answered? Otherwise the agentcallbacklogin will see the line answered and just go with it, i would guess? |
| 19:50.58 | Corydon76-work | Qwell[]: dunno, maybe |
| 19:51.20 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 19:51.21 | Qwell[] | I guess we need to talk about how it's supposed to work sometime then.. |
| 19:51.22 | jkiff | jsolares: Alrighty, I'll look into that. Thanks. :) |
| 19:51.24 | MrTelephone | u have to be a rocket scientist to figure out someone elses code :( |
| 19:51.32 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 19:51.40 | Qwell[] | MrTelephone: luckily, most of the people in Huntsville are just that. |
| 19:51.41 | blindluck9 | 302 moved temporarily usually comes up when someone pushed the CallFWD key on the phone and is redirecting it to another number |
| 19:51.56 | jsolares | jkiff: you'll also need callgroup |
| 19:52.02 | jsolares | i just tried it |
| 19:52.02 | MrTelephone | qwell what are you working on now? |
| 19:52.08 | MrTelephone | asterisk 1.4? |
| 19:52.11 | Qwell[] | MrTelephone: rockets, apparently |
| 19:53.02 | samy_b1 | blindluck9: |
| 19:53.05 | Corydon76-work | Qwell[]: I'll follow your lead |
| 19:53.27 | samy_b1 | that is only happaning when i try to call out true my did sip |
| 19:53.31 | Corydon76-work | Qwell[]: it's your idea, I just extended it a bit |
| 19:53.31 | Qwell[] | I added a null storage driver on the way home on Friday :P |
| 19:53.45 | Qwell[] | because, really...the others are res modules |
| 19:53.53 | Qwell[] | so if none are loaded...well...we've gotta do something |
| 19:54.04 | Corydon76-work | Qwell[]: committed? |
| 19:54.08 | Qwell[] | it's on my laptop |
| 19:54.13 | Corydon76-work | Ah |
| 19:54.41 | Katty | how do i dial an extension and tell it which context it's supposed to be in? |
| 19:54.57 | Katty | like exten => _xxx,1,Dial(SIP/${EXTEN}@context,20)? |
| 19:55.08 | Katty | it keeps telling me no such host: downstairs |
| 19:55.16 | neverblue | hehe |
| 19:55.19 | neverblue | ~book |
| 19:55.32 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:55.32 | jkiff | jsolares: Aye, I see. So I'll have to add a callgroup=1 and a pickupgroup=1 to every phone in sip.conf? |
| 19:55.32 | neverblue | mmmuuuhaha |
| 19:55.32 | neverblue | ~thebook |
| 19:55.35 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:56.14 | irule | http://pastebin.ca/519903 I reposted again, I dont ever recall modifying stdexten, and that is where I get the error message |
| 19:56.43 | jsolares | jkiff: correct, that's also if you want every phone to be able to pickup every call, http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 19:57.56 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 19:58.07 | jsolares | irule: whats with line 159, _0,1 wouldn't that reset n? hmmm |
| 19:58.34 | *** join/#asterisk pejo_ (n=peter@1-1-5-39a.lio.sth.bostream.se) |
| 19:59.12 | irule | thanks for pointing out the error |
| 19:59.58 | jkiff | jsolares: Hehe, I just found that page. I see, so I can make only some phones pickup'able, and only some phones able to pick up the pickup'able phones... if that makes sense. |
| 20:00.29 | irule | but I still get this error message Auto fallthrough, channel 'SIP/sip503-081e34f8' status is 'NOANSWER' |
| 20:01.27 | jsolares | can you see the stdexen ARG1 something ARG2 something? |
| 20:03.32 | [TK]D-Fender | irule: pastebin the full CLI output at verbose 10 of the call that generates the error |
| 20:03.33 | irule | you are right |
| 20:03.45 | irule | thanks jsolares |
| 20:04.00 | irule | I must get used to watching those little details :s |
| 20:04.44 | jsolares | i know what you mean, i finally got led blinking call pickup working on snom 320 on asterisk 1.4.4 |
| 20:06.52 | *** join/#asterisk _mihai_ (n=_mihai_@38.96.187.252) |
| 20:10.12 | *** join/#asterisk yacc (n=andreas@091-141-067-254.dyn.one.at) |
| 20:10.38 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 20:11.45 | jsolares | well time for lunch, ttyl |
| 20:16.47 | yannj_fr | Does any one use snom phones with autoprovisionning? |
| 20:17.41 | *** join/#asterisk pejo_ (n=peter@1-1-5-39a.lio.sth.bostream.se) |
| 20:19.04 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 20:19.54 | tuan_modulis | does this look normal to you guys? |
| 20:20.01 | tuan_modulis | May 29 16:17:25 VERBOSE[1421] logger.c: -- Called 5IP-UNLIMITEL5147624011 |
| 20:20.07 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 20:20.13 | *** part/#asterisk _mihai_ (n=_mihai_@38.96.187.252) |
| 20:20.26 | tuan_modulis | the number and channel are like glued together |
| 20:22.34 | *** join/#asterisk ploieel (n=manni@Fb2e0.f.ppp-pool.de) |
| 20:22.54 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 20:24.50 | groogs | tuan_modulis: no, and neither does 5IP instead of SIP |
| 20:25.22 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:25.35 | tuan_modulis | thx |
| 20:29.46 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 20:31.34 | irule | is it possible to authenticate among 5 or 10 different passwords as opposed to a single password as in Authenticate? |
| 20:32.11 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:34.40 | *** join/#asterisk sob0l (n=sobol@host-87-99-4-27.lanet.net.pl) |
| 20:35.03 | Corydon76-work | irule: if you read the documentation, you'll see that the first argument to Authenticate may be a pathname |
| 20:36.06 | Corydon76-work | However, if you want multiple authentication, I'd suggest using VMAuthenticate, instead, especially if each user has a voicemail box on the system |
| 20:38.45 | *** part/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 20:39.31 | *** join/#asterisk EricL (n=eric@74.9.83.194) |
| 20:39.56 | EricL | Is there a fix for the "no reply to our critical packet" bug? |
| 20:40.20 | EricL | I am running Asterisk 1.4.4 on Gentoo and I can't get my Cisco 7961G phones to hold a call for longer than 20seconds. |
| 20:42.34 | high-rez | Anyone else having problems with teliax at the moment? |
| 20:42.39 | tzafrir_laptop | EricL, is there a bug in the mantis (bugs.digium.com) ? |
| 20:42.50 | *** join/#asterisk OpenBSDFan (n=irc@unaffiliated/openbsdfan) |
| 20:43.32 | OpenBSDFan | greetings, sorry for sounding like a "noob", but what is "asterisk" used for? |
| 20:43.50 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:43.54 | high-rez | It's a telephony engine. |
| 20:43.56 | Corydon76-work | OpenBSDFan: see topic |
| 20:44.07 | high-rez | You can use it as a PBX, custom voice apps, as a service provider, etc etc etc. |
| 20:44.40 | EricL | tzafrir:http://bugs.digium.com/view.php?id=7433 I believe this is the same thing. |
| 20:44.45 | OpenBSDFan | I see.. so people would dial into it using an analog phone? |
| 20:44.55 | Qwell[] | OpenBSDFan: among many, many, many other things |
| 20:45.27 | OpenBSDFan | How would it be of any use though? one would need special hardware to accept more then a single "phone-in" right? :S |
| 20:45.38 | tzafrir_laptop | EricL, small hint: leave a space before the http:// , this will make it easier for automatic links detectors of IRC clients. Anyway, looking |
| 20:45.43 | Qwell[] | ~book |
| 20:45.55 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:45.55 | Qwell[] | ~wikis |
| 20:45.57 | jbot | i heard wikis is http://www.voip-info.org |
| 20:45.57 | Qwell[] | OpenBSDFan: start there |
| 20:46.15 | OpenBSDFan | I just need a simple explaination really.. |
| 20:46.20 | EricL | tzafrir_laptop: I usually do, but I forgot to load my default settings in BitchX, I just did it :) |
| 20:46.21 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:46.34 | Qwell[] | http://asterisk.org/about |
| 20:46.43 | Corydon76-work | OpenBSDFan: you're way beyond simple explanation already |
| 20:47.05 | OpenBSDFan | I've read that page, but that doesn't really explain it IMHO. |
| 20:47.47 | OpenBSDFan | Well it might, but it could be just over my head. |
| 20:47.48 | Corydon76-work | OpenBSDFan: explain openbsd |
| 20:47.48 | tzafrir_laptop | EricL, that bug was marked as fixed before asterisk 1.4.4 has been released |
| 20:48.02 | Corydon76-work | OpenBSDFan: doesn't it require special hardware? |
| 20:48.10 | EricL | tzafrir_laptop: I know, but that's the only bug I can seem to find similar to my problem. |
| 20:48.15 | OpenBSDFan | No it doesn't. |
| 20:48.24 | Qwell[] | OpenBSDFan: so what does it run on? |
| 20:48.41 | EricL | tzafrir_laptop: I am getting that error and the phone hangs up after 20 seconds. |
| 20:48.54 | OpenBSDFan | ..Any beige box with an old i386 would suffice. |
| 20:49.01 | Qwell[] | there you go then |
| 20:49.17 | Qwell[] | (and you're wrong, because openbsd will compile on...everything - including toasters. :P ) |
| 20:49.25 | Qwell[] | ((but then, so will asterisk)) |
| 20:49.44 | OpenBSDFan | :P I don't follow the armish port.. thank you. |
| 20:50.13 | Qwell[] | asterisk doesn't require any hardware at all, beyond "..Any beige box with an old i386" |
| 20:50.18 | tzafrir_laptop | EricL, sorry. I have no such phone... |
| 20:50.22 | OpenBSDFan | but really though, would "Asterisk" be used at home? if so.. how could it be at all useful without a special line? (for more then 1 user..)? |
| 20:50.39 | EricL | Any idea what would cause that though? |
| 20:50.49 | EricL | Since it doesn't happen on any of my other phones? |
| 20:50.50 | Qwell[] | OpenBSDFan: depends on what you want it to do |
| 20:51.16 | OpenBSDFan | Well, I would like to build my own SkyNet and fight evil monkeys.. :| |
| 20:51.29 | OpenBSDFan | sorry lol. |
| 20:51.45 | Qwell[] | well, that's certainly doable |
| 20:51.49 | *** part/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 20:52.04 | Qwell[] | monkeys are predictable, you could probably fight them with 20-30 lines of dialplan |
| 20:53.10 | OpenBSDFan | I'll rephrase my question... "What would be required to make it feisable for more then 1 person to dial in at a time.." via analog phones. |
| 20:53.25 | Qwell[] | dial in to what? |
| 20:53.49 | Corydon76-work | OpenBSDFan: a single port analog card.... $150 in hardware |
| 20:54.11 | OpenBSDFan | So I don't understand Asterisk, I though it was some sort of "server" for lack of a better word. |
| 20:54.26 | Qwell[] | OpenBSDFan: yes, and what do you want the "server" to do with your call once it gets it? |
| 20:55.46 | OpenBSDFan | I don't know.. sing the J-e-l-l-o theme song.. how does that matter? |
| 20:55.47 | Qwell[] | Does it need to play the J-e-l-l-o theme song to more than one user at a time? Does it need to start over each time somebody new calls in? |
| 20:56.02 | Corydon76-work | OpenBSDFan: the dialplan is a blank slate. It's like programming a firewall. There are tools, but the basic firewall doesn't pass any traffic whatsoever |
| 20:56.50 | OpenBSDFan | Qwell, sure why not.. hypethically// |
| 20:57.08 | Qwell[] | Then you need more than one port |
| 20:57.28 | Qwell[] | We can't define your system for you. You need to tell us what you want it to do, before we can tell you what you need. |
| 20:57.52 | *** join/#asterisk jsolares (n=jsolares@216.106.168.71) |
| 20:58.02 | OpenBSDFan | "port" ? multiple telephone lines? or a facility for receiving ships and transferring cargo to and from them. |
| 20:58.15 | Qwell[] | You only said phones, so no, you don't need any telephone lines at all. |
| 20:58.44 | OpenBSDFan | Wow, I'm so lost I need a map.. |
| 20:58.52 | Qwell[] | ~book |
| 20:58.59 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:58.59 | Qwell[] | ~wikis |
| 20:59.01 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 20:59.14 | OpenBSDFan | I think you're starting to be a little repetitive.. |
| 20:59.21 | punani | [21:42:27] * Joins: OpenBSDFan (n=irc@unaffiliated/openbsdfan) |
| 20:59.24 | punani | [21:43:26] * Joins: danalien (n=danalien@unaffiliated/danalien) |
| 20:59.29 | Qwell[] | well, we don't have any information to go on yet... |
| 20:59.37 | punani | where do these vhosts come from...? |
| 20:59.50 | punani | can't seem to set mode +x here |
| 20:59.50 | Qwell[] | punani: freenode - ask an ircop for an unaffiliated cloak |
| 21:00.00 | Corydon76-work | punani: they're hostmasks |
| 21:00.07 | EricL | tzafrir_laptop: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg181107.html |
| 21:00.09 | OpenBSDFan | The internet is a series of pipes, unaffiliated is the one to the left. |
| 21:00.11 | Corydon76-work | Donate to Freenode to get your own |
| 21:00.14 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 21:00.20 | Qwell[] | for unaffiliated, you don't need to donate |
| 21:00.23 | OpenBSDFan | They are free.. |
| 21:00.27 | Qwell[] | but, you can get a sponsor cloak |
| 21:00.28 | EricL | This is what I am thinking of doing since I really need these phones to work. |
| 21:01.46 | OpenBSDFan | Well, Thanks for the mind games.. this was a fun way to find help.. :D |
| 21:02.13 | Qwell[] | OpenBSDFan: it's kinda expected that people who come here actually know what asterisk...is |
| 21:02.36 | OpenBSDFan | It's best to expect the unexpected my friend. |
| 21:03.40 | OpenBSDFan | I for instance, have no clue what I'm doing.. thus I went out on a swift and eventful journey.. |
| 21:03.59 | OpenBSDFan | And still have no clue what I'm doing.. |
| 21:04.09 | Qwell[] | OpenBSDFan: when you're ready to tell us what you'd like to do - we'd be more than happy to help |
| 21:04.18 | Qwell[] | until then, what else can we do, really? |
| 21:05.01 | OpenBSDFan | A phone line.. A server... A constantly playing song.. and concurrently connected users. |
| 21:05.11 | Qwell[] | what is the phone line for? |
| 21:05.26 | OpenBSDFan | For people to "call" the server, via "telephone". |
| 21:05.36 | Qwell[] | a telephony doesn't need a line |
| 21:05.39 | Qwell[] | telephone* |
| 21:05.44 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 21:05.48 | OpenBSDFan | ..? Who's going to call it then? |
| 21:05.51 | Qwell[] | doesn't need a phone line, rather.. not like you're thinking |
| 21:05.58 | Qwell[] | a...phone.. directly connected to asterisk |
| 21:06.17 | Qwell[] | You DID say "analog phone" earlier, did you not? |
| 21:06.46 | OpenBSDFan | Well, Having everyone bring a phone to my house to connect directly into the "server" would be kinda.. unorthadox.. |
| 21:07.05 | Qwell[] | well, without knowing anything about your setup - it seemed pretty reasonable to me... |
| 21:07.08 | OpenBSDFan | Why not call it, via a normal telephone number.. |
| 21:07.20 | OpenBSDFan | I expect you to to read my mind... (sorry..) |
| 21:07.24 | Vec2 | When a call comes in to asterisk I have set it to, transfer the call after X seconds to my mobile, however the CDR does not show the dialed number has my cellphone, only shows the incomming number, does anyone have a solution to this ? |
| 21:07.28 | slmnhq | OpenBSDFan: you can receive calls over the internet.. a la voip |
| 21:07.30 | Qwell[] | because if you just want to hear the J-e-l-l-o theme song when getting out of the shower... |
| 21:07.38 | Qwell[] | then you wouldn't need a phone line. |
| 21:07.45 | Qwell[] | Please, be more specific with what you want. :) |
| 21:08.06 | *** part/#asterisk visba (n=dca[lapt@c-67-166-17-228.hsd1.co.comcast.net) |
| 21:08.06 | OpenBSDFan | This was hypothetical.. |
| 21:08.09 | slmnhq | OpenBSDFan: then all you need is an internet connection |
| 21:08.22 | OpenBSDFan | I'm not interested in viop. |
| 21:08.41 | OpenBSDFan | Just.. a normal.. basic telephone line.. heh |
| 21:08.53 | OpenBSDFan | voip* |
| 21:08.59 | Qwell[] | with a normal.. basic telephone line.., you would be limited to one caller |
| 21:09.40 | OpenBSDFan | ..Wow, That's what I asked.. before.. and was wondering "what hardware" would be required to get around that. |
| 21:09.59 | Qwell[] | no hardware that you can buy can do that. |
| 21:10.09 | Qwell[] | You'd need to get more lines, or a digital line like a T1 |
| 21:10.51 | Mercestes | OpenBSDFan, If you want one, normal, telephone line, buy a $10 walmart phone. |
| 21:10.57 | OpenBSDFan | I see... so you would require multiple phone lines... deja vu... So It's safe to say most people using "Asterisk" are using a "Digital line" and voip? |
| 21:11.05 | EricL | How do I change the maximum retries for a single SIP peer? |
| 21:11.08 | Mercestes | If you want the jello theme song...buy a $10 walmart radio |
| 21:11.11 | Qwell[] | OpenBSDFan: no, not really |
| 21:11.25 | Qwell[] | some people just want the J-e-l-l-o theme song to play when they get out of their shower |
| 21:11.32 | OpenBSDFan | Mercestes, But that couldn't handle more then one connection. |
| 21:11.39 | Qwell[] | or to be able to call the living room from the bed room |
| 21:11.43 | Mercestes | OpenBSDFan, Sure it can. Call waiting |
| 21:11.55 | Qwell[] | I don't even *have* a phone line at home, let alone a digital one |
| 21:12.06 | OpenBSDFan | concurrent** |
| 21:12.46 | Mercestes | Asterisk is a voice communications server, designed to handle voice communications. If you want to recieve one normal analog phone call then you need one FXO port card. if you want several, then you need several FXO ports. |
| 21:12.52 | OpenBSDFan | What fun would it be to have only "1" person listen to the lovely jingle at a time?? |
| 21:12.56 | Mercestes | if you want many, you need a T1 card hooked to a PRI |
| 21:13.15 | MindTheGap | can I pass $EXTEN on [globals] ? I need the following global: X => "sip/$EXTEN:1" to be used at the actual extension like exten=_155,1,Dial(${X}) |
| 21:13.17 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 21:13.20 | Mercestes | Or if you want to do it over voip, then you need a VoIP provider and an internet connection |
| 21:13.41 | OpenBSDFan | Got it.. |
| 21:13.43 | Mercestes | or you can hook it to an ISDN line if you wish.... |
| 21:13.55 | jkiff | jsolares: That callgroup/pickupgroup stuff works great. Thanks much. :) |
| 21:14.11 | jsolares | glad to help |
| 21:14.24 | Mercestes | Butyou kind of have to understand telephony technologies before you can fully explore what you can plug asterisk into. |
| 21:14.45 | Mercestes | The "limitations" on asterisk are really the limitations on how many concurrent call paths a given telephony medium provides |
| 21:14.47 | OpenBSDFan | So.. I'm guessing with "voip".. a multiple users could "call" this server and "listen" to the hypothetical jingle? |
| 21:15.06 | OpenBSDFan | They wouldn't require viop correct? |
| 21:15.11 | OpenBSDFan | voip* |
| 21:15.20 | Mercestes | With "voip" then you are only limited by the quality of your provider(s), your bandwidth, and the reliability of your connection from start to finish |
| 21:15.41 | OpenBSDFan | 16Mbit/1Mbit suffice? |
| 21:15.50 | Mercestes | no, your ITSP would intercept the # via other telephony technologies and switch it to you over VoIP |
| 21:16.13 | Mercestes | The real answer is "maybe." |
| 21:17.03 | OpenBSDFan | Thanks for providing me with the information I was seeking.. do they pay you? |
| 21:17.19 | Mercestes | I can run calls on Dialup using a WRT54GL linksys router. |
| 21:17.26 | Mercestes | I can also fail to run calls on a T1 |
| 21:17.54 | Mercestes | It depends more on your network backbone reliability,jitter, packet-loss, ping times, router speeds, router stability, etc. |
| 21:18.09 | OpenBSDFan | I see |
| 21:18.15 | Mercestes | You can have a 300ms ping for all I care as long as it *stays* at 300ms and doesn't flop around like a suffocating fish |
| 21:18.52 | OpenBSDFan | Does this mean with voip, anyone can become a small local dialup isp? via a broadband line? |
| 21:19.07 | MindTheGap | can I pass $EXTEN on [globals] ? I need the following global: X => "sip/$EXTEN:1" to be used at the actual extension like exten=_155,1,Dial(${X}) |
| 21:19.37 | Mercestes | Some people seem to think so. |
| 21:19.45 | Mercestes | anyone can become a *CRAPPY* ISP. |
| 21:19.52 | OpenBSDFan | :D |
| 21:20.14 | OpenBSDFan | I bet.. sounds like a fun get rich quick scheme.. |
| 21:20.18 | docelmo | but it takes someone special to become an extraordinarily crapy ISP.. |
| 21:21.28 | OpenBSDFan | So there are probably hundreds of these such ISP's looking to follow AOL's path? |
| 21:21.44 | docelmo | yep |
| 21:21.51 | docelmo | same with ITSP's.. |
| 21:22.21 | docelmo | They are all the same.. They put the smallest amount of thought and effort into building something and hoping no one challenges that they suck.. |
| 21:22.50 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 21:23.01 | docelmo | Well the ITSP game has learned from the ISP's and found to be a little more cautious when using an ITSP.. Hense the AOL of ITSP's Vonage |
| 21:23.29 | OpenBSDFan | ITSP meaning internet telephone provider? |
| 21:23.33 | docelmo | yes |
| 21:23.38 | Qwell[] | ~itsp |
| 21:23.42 | jbot | [itsp] Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 21:23.44 | docelmo | thanks Q |
| 21:24.19 | OpenBSDFan | So this has probably made it easy for people to become "TISP"'s then? |
| 21:24.42 | docelmo | that does this refer to? |
| 21:24.44 | docelmo | Asterisk |
| 21:24.49 | docelmo | ? |
| 21:24.51 | OpenBSDFan | voip.. |
| 21:24.52 | Mercestes | OpenBSDFan, Do you know how to become a millionaire in telecom??? |
| 21:25.29 | OpenBSDFan | Are you kidding? I always make a tough choice at the end of the month.. rent or chips. |
| 21:25.33 | docelmo | Pull a sellvoip |
| 21:25.40 | Mercestes | OpenBSDFan, Well, first you start out as a billionaire. |
| 21:25.42 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:25.50 | Qwell[] | OpenBSDFan: poker, or potato? |
| 21:26.00 | OpenBSDFan | You're funny.. |
| 21:26.07 | Mercestes | It's true. |
| 21:26.10 | OpenBSDFan | Doritos. |
| 21:26.53 | OpenBSDFan | I have been $5 short on my rent.. It's an evil habbit.. |
| 21:27.28 | OpenBSDFan | But that's for 2 bags.. it's a steal.. |
| 21:28.22 | OpenBSDFan | I invent well. |
| 21:28.25 | OpenBSDFan | invest* |
| 21:29.01 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 21:31.10 | OpenBSDFan | *whistles* awkward silence |
| 21:31.52 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:32.17 | OpenBSDFan | What's a TK and why do you defend it? |
| 21:33.02 | EricL | tzafrir_laptop:It appears as though 2 is XMIT_CRITICAL and 1 XMIT_RELIABLE |
| 21:33.49 | tzafrir_laptop | so those values have basically remained the same |
| 21:34.10 | EricL | Yep, but I think if I change them, I will probably have hanging open sip lines. |
| 21:34.18 | *** part/#asterisk OpenBSDFan (n=irc@unaffiliated/openbsdfan) |
| 21:34.34 | EricL | I would really prefer not to mess with the source. I am sure its probably some setting, but I will open up a bug report anyway. |
| 21:36.26 | *** join/#asterisk friedrich| (n=friedric@e177244249.adsl.alicedsl.de) |
| 21:37.50 | EricL | tzafrir_laptop: It only happens when I dial out. |
| 21:38.43 | EricL | tzafrir_laptop: It doesn't happen when I dial anything locally (ie 4 digits instead of the 10 for an external number). |
| 21:41.04 | EricL | tzafrir_laptop: And it doesn't happen on incoming calls, only outgoing calls. |
| 21:42.30 | *** join/#asterisk DigitalKNK (n=DigitalK@74.7.73.133) |
| 21:42.35 | DigitalKNK | um can anyone help me out and figure out why my sipconnect is not registering ? I just signed up with cbeyond and I am having a hard time trying to get it registered. |
| 21:43.31 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 21:44.19 | EricL | Is there anyone around that has any idea what the 20 second disconnect is about on calls outside of the * box? |
| 21:50.18 | *** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
| 21:50.37 | Mercestes | DigitalKNK, Cbeyond has very good technical support. |
| 21:51.01 | Netgeeks | Hey folks, anyone here fairly familiar with using the Manager interface? Specifically listening to events? |
| 22:03.42 | denke | Hello Guys, does anyone know what dose it mean: [May 29 23:50:59] WARNING[8437]: translate.c:677 __ast_register_translator: plc_samples 160 format 6 |
| 22:03.57 | denke | I get this message a few times at asterisk startup |
| 22:05.08 | denke | someone? .... anyone? :) |
| 22:06.58 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 22:07.00 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
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| 22:09.22 | *** part/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 22:09.24 | Mercestes | denke, it is safe to ignore that warning. Try googling the error |
| 22:09.29 | Mercestes | s/error/warning/ |
| 22:10.03 | denke | I have, but no useful information.... |
| 22:10.25 | denke | thank you for your help |
| 22:11.02 | gcbirzan | Hey... Not really asterisk related, but app_cepstral related (the one off http://www.voip-info.org/wiki/index.php?page=App_Cepstral, for version 1.2). I've managed to compile it, it kind of works, only it segfaults. Hm. It seems write_audio() is called by swift with the last parameter not actually initialised... Anyone have any hints, or knows the solution? (No, Google made me none the wiser) |
| 22:11.27 | denke | the only thing i do not understand, if it is ignoreable, than why level warning... |
| 22:12.18 | gcbirzan | denke: That's why it's not an error. :-) |
| 22:12.29 | denke | :) |
| 22:12.42 | denke | and why not debug, or notice? :) |
| 22:13.12 | *** join/#asterisk remmo (n=junk@203.62.147.6) |
| 22:14.00 | gcbirzan | (And since swift doesn't seem to be open source... :-) ) |
| 22:14.50 | denke | thanks, for all the help |
| 22:15.33 | neverblue | shouldnt that be s/warning/error/ ? |
| 22:17.00 | denke | I think it should be s/warning/debug/; but it sure is warning... |
| 22:24.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:28.06 | gcbirzan | Pfeh. Let me try another question. Anyone played with the two app_cepstrals? |
| 22:29.35 | *** join/#asterisk dave_mw1 (n=dave_mwi@CPE-65-27-7-82.kc.res.rr.com) |
| 22:30.14 | jsolares | i use a perl AGI to generate the .wav and then play it back |
| 22:30.14 | dave_mw1 | I'm trying to play a .gsm audio file...I'm not having any luck. On Fedora 6 here, tried XMMS, play, esd-play, mplayer, xine...nothings playing it. Any suggestions |
| 22:30.29 | jsolares | use sox to convert it to wav |
| 22:30.35 | dave_mw1 | well... |
| 22:30.46 | dave_mw1 | jsolares: I suppose I could...but I'd rather just play it directly |
| 22:30.54 | jsolares | no idea then :) |
| 22:31.29 | jsolares | i usually set up an extension on asterisk to test the .gsm's instead of playing them back *shrugs* |
| 22:32.48 | *** join/#asterisk k31th (n=keith@cartman.nzsolutions.net) |
| 22:33.01 | k31th | anyone else used this grandstream phones |
| 22:33.07 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 22:33.14 | k31th | they seem flakey to say the least? |
| 22:33.25 | Mercestes | ~gs |
| 22:33.27 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:33.30 | jsolares | they certainly do but at the price |
| 22:33.33 | Mercestes | ~phones |
| 22:33.35 | jbot | from memory, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 22:34.01 | Mercestes | The bot never lies. |
| 22:34.03 | jsolares | i have a snom320 and a polycom430 they put the grandstream to shame |
| 22:34.52 | Mercestes | ~botsnack |
| 22:34.52 | jbot | aw, gee, Mercestes |
| 22:36.15 | *** join/#asterisk DaveCanoe (n=Dave@66.96.20.43) |
| 22:37.04 | s0ck | jsolares: yeh>? |
| 22:37.04 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
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| 22:45.17 | *** join/#asterisk dijungal (n=chatzill@64.86.52.254) |
| 22:45.24 | dijungal | hello |
| 22:46.10 | dijungal | question: I would like to use asterisk to dial numbers and test for dial tone... i would just like to verify that the number actually rings... any suggestions? |
| 22:46.13 | *** part/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 22:46.14 | gcbirzan | jsolares: Hm. The other app_cestral doesn't return when a key is pressed... Nor does EXEC PLAYBACK which, apparently, is The Way(tm) to do it, heh. |
| 22:46.23 | dijungal | sorry not dial tone - ring tone |
| 22:47.22 | killfill_ | hi |
| 22:47.23 | gcbirzan | Hm, though, you probably weren't talking to me. |
| 22:47.24 | gcbirzan | :-) |
| 22:47.25 | jsolares | gcbirzan: why not GET DATA it'll return the key pressed |
| 22:47.34 | killfill_ | http://www.pastebin.ca/520413 <---- how cna i get more info about whats happening to my TE110p? |
| 22:47.45 | killfill_ | all calls are getting hang up |
| 22:48.50 | dijungal | so is there anyway to use asterisk to verify that phone numbers actually ring? |
| 22:48.54 | jsolares | gcbirzan: http://www.voip-info.org/wiki/view/get+data |
| 22:49.10 | gcbirzan | jsolares: I was looking at http://www.voip-info.org/wiki/view/stream+file actually. |
| 22:49.13 | gcbirzan | But, that could work. |
| 22:50.02 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 22:54.17 | dijungal | i guess noone knows |
| 22:57.56 | s0ck | dijungal: i remember reading a project on nerdvittles about something similar, have a scan :) |
| 22:58.44 | dijungal | nerdvittles? |
| 22:58.56 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:01.36 | hansin321 | I am using Xlite to register to *, and I am trying to get it to send calls out my FXO card. Can anyone see anything wrong with this line: |
| 23:01.39 | hansin321 | exten => _9XXXXXXX.,1,Dial(Zap/1/$EXTEN:1) |
| 23:02.01 | _charly_ | exten => _9XXXXXXX.,1,Dial(Zap/1/${EXTEN:1}) |
| 23:02.19 | hansin321 | _charly_: thanks. I'll give that a shot. |
| 23:04.59 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 23:05.20 | hansin321 | Thanks _charly_. I was using an older tutorial, but it was relevant for what I am doing. Works for me now. I am going through the learning curve stage right now. |
| 23:05.32 | kombi | anyone got 79x1 phones to download ringtones yet? |
| 23:05.38 | hansin321 | Must have been old syntax. |
| 23:05.48 | *** join/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
| 23:06.03 | kombi | ..cause I havn't.. |
| 23:06.06 | *** part/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com) |
| 23:06.44 | _charly_ | hansin321: i don't know about a syntax without {}, but at least >=1.2 has ${VARIABLE} for variables |
| 23:07.57 | hansin321 | This was an Onlamp article from 2002 or so, so it may have changed. Or an error, |
| 23:08.06 | dijungal | s0ck: can't seem to find an article on that |
| 23:08.14 | _charly_ | maybe |
| 23:08.24 | dijungal | i've been searching google too.. can't seem to find anything close to what i need |
| 23:08.40 | *** join/#asterisk cpurn (n=cpurn@eth4307.vic.adsl.internode.on.net) |
| 23:09.37 | hansin321 | I am trying also to get it to send calls the other way, from POTS/FXO to my registered SIP client. Can you see anything here? (I promise no more after this; I'll dig in a little deeper on my own): |
| 23:09.41 | hansin321 | exten => s,1,Dial(SIP/2000,30) |
| 23:11.03 | _charly_ | you have extension 2000 in your sip.conf? |
| 23:11.31 | cpurn | I have a TDM400P card, I can get incoming call going according to the 'Future of telephony', the incoming call is simply an answer() and echo(). I'm trying to test an analog phone by plugging it to the FXS port, however I am unable to get any dial tone signal? I have setup an extension 611 (exactly as the documentation shows) and try to dial 611, I get nothing, in asterisk log I can't see any activity, would someone be able to share a l |
| 23:12.33 | hansin321 | _charly_: I do. I'll did a littler deeper. I am reqistering to * on SIP client from NATed environment. I think I just have to get my hands dirty at first and see what I can come up with. |
| 23:13.25 | _charly_ | hansin321: what messages do you get at your asterisk console with verbose 5 ? |
| 23:16.54 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 23:18.38 | hansin321 | _charly_: -- Starting simple switch on 'Zap/1-1' |
| 23:18.38 | hansin321 | [May 29 17:18:45] NOTICE[20544]: chan_zap.c:6351 ss_thread: Got event 18 (Ring Begin)... |
| 23:18.42 | hansin321 | <PROTECTED> |
| 23:18.44 | hansin321 | <PROTECTED> |
| 23:18.51 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:19.18 | hansin321 | I'm going to mess with it a little later. See what I can figure... Thanks. |
| 23:19.30 | _charly_ | and extension 2000 is behind nat? there's a nat-setting for sip.conf |
| 23:19.38 | _charly_ | perhaps you should try this |
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| 23:33.14 | mcc532 | Hmmmm...I have never used IRC before, apologies in advance if I do this wrong. I am getting a 401 unauthorized error from my grandstream 486 that I need help with. |
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| 23:37.17 | *** join/#asterisk grogoreo (n=grogoreo@sheeran.plus.com) |
| 23:37.25 | grogoreo | hi |
| 23:39.11 | *** join/#asterisk mvand (n=mvand@CPE-65-28-181-127.neb.res.rr.com) |
| 23:40.26 | grogoreo | could you use Asterisk in replace of payed for services using the SIP or H.323 protocols? Or am I misinterpreting what it can be used for. Services like, after doing a quick google, sipgate or something |
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| 23:41.12 | grogoreo | I was wondering if it would be feasible to co-locate a server and use it for friends and family use or would the bandwidth be excessive? |
| 23:41.37 | s0ck | would colo costs negate the cost saving? |
| 23:42.09 | s0ck | or do you mean at a family members house with the phattest pipe :D |
| 23:42.26 | Netgeeks | Hey, any folks here very familiar with the manager interface? Specificall listening to events? |
| 23:42.55 | grogoreo | I'm planning to co-locate a server anyway and just thought it would be a cool idea to have my own service. |
| 23:43.03 | s0ck | very briefly looked at it for passing calls the handset via a custom app |
| 23:43.22 | *** join/#asterisk mvand (n=mvand@CPE-65-28-181-127.neb.res.rr.com) |
| 23:43.33 | acctor | I am trying to convert a 7940, I do not have a P003-08-6-00.sbn |
| 23:43.40 | acctor | err, that wasn't supposed to send :) |
| 23:43.52 | acctor | I only have a P003-08-6-00.sbn, not a P0S3-08-6-00.sbn |
| 23:43.52 | s0ck | grogoreo: i've been wondering the same thing myself |
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| 23:44.41 | s0ck | the missus is on the phone enough to her family :P |
| 23:45.47 | grogoreo | s0ck: though I haven't had a go at setting an asterisk server up yet, I would imagine since the the hardware requirements are so low and having only a few amount of people using it, it might work out. Would just be interesting to see how much bandwidth is used |
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| 23:48.27 | s0ck | the g729 codec appears to be the lowest bandwidth whoring one |
| 23:48.36 | s0ck | it will use a few more cpu cycles tho |
| 23:48.56 | s0ck | look at trixbox if you haven't set it up before |
| 23:49.03 | s0ck | it's a point and click adventure |
| 23:49.24 | grogoreo | cool, thanks for the info s0ck |
| 23:49.42 | s0ck | :) |
| 23:51.21 | cpurn | I have an analog phone hooked in into my FSX port on my TDM400 card, I have setup an 'internal' context and created a new extention 611 which simply answer() and echo(), however when I try to dial 611 from my analog phone, I get nothing... what am I missing? thanks |
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| 23:57.41 | zodell | I have posted this to forums.digium.com, but have struck out. Can anyone tell me what I should be looking for now......http://forums.digium.com/viewtopic.php?t=15692 |
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