| 00:05.51 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 00:10.14 | *** join/#asterisk jo3sm1th (n=email@c83-250-80-70.bredband.comhem.se) |
| 00:10.31 | jo3sm1th | Can anyone send me an invite to use Google Voice? |
| 00:11.18 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 00:11.57 | *** join/#asterisk sahafeez_ (n=sahafeez@commodity-noc-48.sc09.org) |
| 00:14.09 | Blackvel | Unable to join queue 'queue_isdn_ntba' |
| 00:14.11 | Blackvel | why? |
| 00:14.53 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp138-119.adsl.forthnet.gr) |
| 00:15.31 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:17.28 | Blackvel | can i not join queue with empty members? :) |
| 00:18.18 | jaytee | depends on what options for the queue you configured in queues.conf |
| 00:20.22 | Blackvel | right |
| 00:20.35 | Blackvel | without member => is returns immediately |
| 00:22.23 | Blackvel | JOINEMPTY will be QUEUESTATUS |
| 00:25.02 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 00:34.23 | *** join/#asterisk josema2 (n=me@122samana71.codetel.net.do) |
| 00:40.04 | *** join/#asterisk Sier (n=sier@unaffiliated/sier) |
| 00:41.27 | jblack | wazza |
| 00:43.32 | jaytee | just got back from seeing 2012 |
| 00:43.39 | prgmrchris | any good? |
| 00:44.09 | jaytee | special effects were excellent although alot of it really stretches the suspension of disbelief |
| 00:44.59 | jblack | jaytee me too! |
| 00:45.18 | jblack | yeah. total destruction porn |
| 00:45.35 | jblack | just like the youtube video. :) |
| 00:47.12 | jaytee | the cleavage content or bare breasticles content could have been higher |
| 00:49.59 | snadge | thats right.. if the worlds ending in 2012.. i want to see maximum tits |
| 00:50.19 | jblack | yeah. less save-the-dog , more gratuitious titties |
| 00:50.57 | jblack | what did you think of all the product placement in the first 1/4? |
| 00:51.15 | jaytee | I tend to ignore it all |
| 00:51.33 | jblack | even that much? how? |
| 00:51.49 | jaytee | I have a great content filter :-) |
| 00:51.51 | hardwire | really? now i have to see it |
| 00:52.07 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 00:52.08 | jblack | i must have seen sony 30 times in 5 minutes |
| 00:52.21 | jaytee | yeah, Viao laptops |
| 00:52.33 | jblack | and tvs |
| 00:52.40 | snadge | so sony is responsible for the end of the world.. i see ;) |
| 00:52.57 | jblack | funny how all the tvs were ntsc. wtf? |
| 00:53.00 | jaytee | Woody Harrelson was great playing himself as some crazy guy with a different name |
| 00:53.14 | jblack | snadge them and kellogs, yeah |
| 00:53.33 | hardwire | I KNEW IT |
| 00:54.02 | jblack | yeah. "just be yourself, man" |
| 00:54.12 | PMantis | Hi, I'm compiling dahdi-linux inside a chroot, and I'm getting "Relocatable linking with relocations from format elf64-x86-64..." messages. Tried: KVERS="2.6.24-25-generic" make TARGET=elf32-i386 Ideas? |
| 00:56.22 | PMantis | Used debootstrap with i386 arch hardy. My workstation is actually 9.10 64 bit. Must be a way to avoid the conflict... |
| 00:56.51 | jblack | woody rocks. ;) |
| 00:57.20 | PMantis | Oh, forgot to say Ubuntu, but I bet that can be assumed from the name and version I referenced. |
| 00:58.07 | jaytee | PMantis, never tried compiling in a chroot, especially from 64 bit to 32 bit. |
| 00:58.13 | jblack | hardy? you're kidding, right? |
| 00:58.14 | jaytee | ~wglat |
| 00:58.37 | jaytee | ~wglwat |
| 00:58.37 | infobot | wglwat is, like, well, good luck with all that |
| 00:58.40 | jblack | after hardy ther has been intrepid, jaunty and karmic |
| 00:59.03 | PMantis | hardy is still the latest LTS release |
| 00:59.07 | jaytee | and coming soon Lesbian Llama |
| 00:59.15 | PMantis | ROFL |
| 00:59.27 | jblack | does hardy even getting security updates? |
| 00:59.42 | jaytee | yes it does and it still gets backports too |
| 00:59.50 | PMantis | Yeah, for 5 years on the server... only 18 months old |
| 01:00.03 | jaytee | 3 years on the desktop |
| 01:00.32 | jaytee | but why are you using a chroot? and going from 64 to 32? |
| 01:01.05 | jblack | he wants to buuld it on his system, and copy it across |
| 01:01.27 | jaytee | ah |
| 01:01.46 | PMantis | I'm building a separate installation for a 32 bit environment, but my desktop it 64bit. Suppose I could build a VM with hardy on it and scp the chroot there, but there must be an easier way. |
| 01:01.59 | jaytee | sounds like a recipe for FAIL with lots of gooey toppings |
| 01:03.00 | PMantis | What's funny is, I did it on my laptop, and that's running 64 bit jaunty. |
| 01:03.37 | prgmrchris | rm -rf ubuntu |
| 01:04.15 | PMantis | smacks prgmrchris with a trout |
| 01:04.17 | PMantis | :) |
| 01:04.21 | jaytee | lol |
| 01:04.33 | jaytee | real men run RHEL or CentOS |
| 01:04.43 | PMantis | bah |
| 01:04.58 | prgmrchris | real men go bare metal |
| 01:05.12 | PMantis | LFS? |
| 01:05.38 | prgmrchris | na |
| 01:07.18 | PMantis | tries it again on his laptop after copying things over from the desktop. |
| 01:07.25 | jblack | pmantus you shouldnt bother chrooting |
| 01:08.29 | jblack | gcc and co can cross compile just fine if they are told to do so. surely *s build system can pass the appropriate march flags down the pipe |
| 01:09.01 | PMantis | jblack: I was trying to avoid the long story, but the chroot is actually a live CD environment that I'm working on. |
| 01:09.35 | jblack | you want to put * on a livecd? |
| 01:09.39 | PMantis | Makes for a nice easy * demo to clients on their hardware. |
| 01:10.02 | jblack | not a bad idea |
| 01:10.09 | PMantis | ...and something to challenge my abilities. |
| 01:10.49 | jblack | your chaqllenge is already hard enough without cross compiling |
| 01:10.54 | jaytee | sounds like the challenge part has reached the point of negative returns |
| 01:11.23 | jblack | i'd downshift to a 32 bit system for developing this |
| 01:12.16 | jaytee | 3 years ago, Chinese calendar year of the cow .....Mad Cow disease. 2 years ago, Chinese calendar year of the bird .....Avian flu. This year, Chinese calendar year of the pig .... swine flu. Next year, is the year of the cock .........Anyone worried? |
| 01:12.36 | PMantis | I had it all working, but then I added the chroot to a svn repository, and a kernel upgrade inside the chroot caused all kinds of hell with depmod.d/.svn and the environment is toast. So, I'm rebuilding a different way from the ground up: Only have the scripts that make the chroot in SVN... not the chroot itself. |
| 01:12.49 | jblack | jaytee: we're fucked? |
| 01:13.25 | jaytee | according to the Magic 8 Ball "outlook does not look good!" |
| 01:14.01 | jblack | after how bad 09 was for me, i'm scared. honestly |
| 01:15.00 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 01:15.21 | jblack | no katty here? |
| 01:15.25 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:24f5:8fd6:7f2d:73d0) |
| 01:16.22 | jaytee | probably napping, she stuffed herself on turkey earlier |
| 01:16.44 | *** join/#asterisk odi_cha (n=odi@85.155.238.217.dyn.user.ono.com) |
| 01:17.00 | PMantis | notes the distinct difference in compiling between his dual core 2.0 GHz laptop, and his quad core 2.8 GHz desktop. |
| 01:17.11 | jblack | ahhh |
| 01:17.32 | [TK]D-Fender | bbiab |
| 01:17.39 | *** join/#asterisk acxty (n=acxty@190.6.194.152) |
| 01:17.51 | PMantis | wants distcc!! |
| 01:18.25 | jblack | now that i think about it, has the dr been around much lately? |
| 01:19.40 | jblack | PMantis: its your nerves.... |
| 01:20.21 | jblack | jaytee: does sending you energy in space wars do you any good? |
| 01:20.55 | *** join/#asterisk odi_cha (n=odi@85.155.238.217.dyn.user.ono.com) |
| 01:22.21 | acxty | Hi guys, I am having lots of troubles making sugarcrm connect to asterisk. I already configure manager.conf and save the information on the system configuration on sugarcrm. I see the phone icon next to the contact phone number but when I try to make a call it popup the window and says Error Connecting to asterisk server. Already test the connection by telnet and works fine. Any ideas what may be wrong? |
| 01:23.18 | *** join/#asterisk odi_cha (i=559beed9@gateway/web/freenode/x-sasikmkclemzexwn) |
| 01:23.47 | jblack | acxty: try watching a call attempt in the asterisk console with debug and verbose at 9 |
| 01:24.01 | jblack | acxty: my bet is failed auth |
| 01:24.58 | jblack | acxty: btw, has sugar fixed the permissions problems for new users? |
| 01:25.55 | acxty | jblack, already try doing that, but it doesn't appear anything on CLI |
| 01:26.10 | *** join/#asterisk joat (n=joat@70.174.79.200) |
| 01:26.14 | jblack | then its sugar |
| 01:26.29 | acxty | I test the connectio by telnet with the exact username and password and works fine |
| 01:28.51 | *** join/#asterisk rhoyerboat (i=1000@c-98-245-149-46.hsd1.co.comcast.net) |
| 01:29.37 | jblack | just more proof that its sugar |
| 01:30.02 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 01:30.11 | acxty | jeje, I had been looking all over but haven't found anything yet |
| 01:30.11 | jblack | hi |
| 01:30.26 | Katty | comes bearing gifts of turkey and pumpkin pie |
| 01:30.43 | jblack | heh |
| 01:31.02 | Katty | how was your day dear |
| 01:31.03 | acxty | Hi Katty, remember me? I was asking for a callcenter solution and you told me about sugarcrm |
| 01:31.14 | Katty | not really, but that's okay. |
| 01:31.19 | Katty | i have the memory of a ferret. |
| 01:32.13 | jblack | not bad saw 2012 |
| 01:32.29 | voipmonk | did u like it , jblack ? |
| 01:32.40 | Katty | is it any good? |
| 01:32.45 | jblack | disaster porn. loved it. ;) |
| 01:32.50 | voipmonk | me, too |
| 01:32.53 | Katty | hmm |
| 01:32.53 | acxty | ok, I install sugarcrm and install the asterisk integration patch. I am getting the phone icon for each contact phone number, but when I click on it says Error Connecting to asterisk server. Already check and recheck the configuration on manager.conf and still cannot make the phone call |
| 01:33.05 | Katty | i think me and ryan will go see it tonight |
| 01:33.07 | Katty | now that you mention it |
| 01:33.10 | Katty | checks movie showtimes |
| 01:33.14 | voipmonk | must see on a large screen and enough sound so u can barely hear yourself laugh |
| 01:33.24 | jblack | its worth the ticket |
| 01:33.46 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 01:34.00 | voipmonk | I thought they would do more with the auto show cars tho |
| 01:34.11 | jblack | acxty: until you can get sugar to at least attempt to connect, theres nothing we can do |
| 01:34.21 | voipmonk | got the feeling Sony and bentley may have paid a pretty penny for the product placement |
| 01:34.39 | jblack | maybe sugar's logs can get you help in their channel |
| 01:35.50 | jblack | voipmonk: feeling? heh. they dont put up walls of kellogs withoout a big bribe. |
| 01:36.57 | voipmonk | I liked the geographic placement of the getaway boats |
| 01:37.05 | jblack | yeah, its a theatre show. |
| 01:37.47 | Katty | hmm. starts in 30 min |
| 01:37.50 | jblack | deletes a spoiler |
| 01:37.53 | Katty | guess we outta get going (= |
| 01:37.55 | Katty | poofs |
| 01:37.55 | jblack | go! |
| 01:38.12 | Sier | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 01:38.20 | Sier | do I need to reboot the machine to fix this? |
| 01:38.33 | jblack | sier ps aux |grep asterisk |
| 01:38.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:39.12 | Sier | there are like 10 processes running |
| 01:39.26 | Sier | including /bin/sh /usr/sbin/safe_asterisk and other = /usr/sbin/asterisk -f -U etc.. |
| 01:39.41 | voipmonk | killall -9 safe_asterisk asterisk |
| 01:40.05 | Sier | worked.. wonderful |
| 01:40.30 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:41.47 | jblack | something serious is going on though. getting multiple process like that is quite unusual |
| 01:44.13 | Sier | I installed Freepbx.. |
| 01:44.25 | Sier | but it's not working properly.. |
| 01:46.00 | PMantis | Sier: That's probably best addressed in #freepbx |
| 01:46.09 | [TK]D-Fender | ~freepbx |
| 01:46.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:46.54 | Sier | thanks.. |
| 01:54.39 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 01:59.51 | Trupsalms | please help i want to start from the beganning of mysql database and freepbx |
| 01:59.59 | Trupsalms | i keep getting errors |
| 02:00.08 | voipmonk | drop the db and start over |
| 02:00.57 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 02:02.01 | Trupsalms | need instruction on dropping the db |
| 02:03.35 | jblack | wrong channel. they do freepbx support in #freepbx |
| 02:13.23 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 02:17.41 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 02:25.41 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 02:25.59 | jblack | the channel is quiet. a little too quiet |
| 02:26.13 | russellb | ~panic |
| 02:26.14 | infobot | somebody please DO something! |
| 02:27.26 | jblack | dances the electric slide |
| 02:27.38 | jblack | join in! you know ya wanna |
| 02:29.14 | KavanS | farts really loud |
| 02:31.41 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 02:31.51 | snadge | a really loud fart would probably make a good voicemail message |
| 02:36.27 | jblack | <PROTECTED> |
| 02:37.40 | jplank | anyone have a wav for a ringback?I can't seem to find one anywhere |
| 02:39.54 | snadge | does anyone here run asterisk on top of xen? |
| 02:40.47 | snadge | asterisk now used to have downloadable xen images.. but they're no longer there.. i made my own xen image by rsyncing an asterisknow install across, and installing the xen kernel |
| 02:43.40 | acxty | sugarcrm people never answered |
| 02:45.26 | *** join/#asterisk jmacz (n=jmacz@190.25.115.67) |
| 02:47.58 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 02:55.23 | *** join/#asterisk jmacz (n=jmacz@190.25.115.67) |
| 02:55.49 | jblack | acxty: this is still the wrong place. sorry |
| 02:56.58 | acxty | jblack, ;) |
| 03:04.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:06.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:15.38 | *** join/#asterisk jmacz (n=jmacz@190.25.115.67) |
| 03:22.35 | dlynes | acxty: you're looking for an asterisk plugin for sugarcrm? |
| 03:23.40 | dlynes | acxty: nvm....just scrolled back...forget I asked |
| 03:24.39 | prgmrchris | snadge: i do |
| 03:42.53 | snadge | if i use 64bit opensuse 11.1 as the dom0.. asterisk can't register its trunk, just uses up all memory in a loop repeatedly trying to register unsuccessfully |
| 03:43.55 | snadge | the same install on virtualbox (hardware emulation) is able to register fine |
| 03:46.25 | snadge | prgmrchris: what do you use for your dom0? |
| 03:57.37 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 04:12.21 | *** part/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 04:13.42 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 04:23.37 | prgmrchris | snadge: elastix |
| 04:34.59 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 04:42.20 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 04:51.23 | *** join/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) |
| 04:53.14 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 04:54.35 | *** join/#asterisk prah (n=karl_m_j@cpe-075-177-109-229.triad.res.rr.com) |
| 04:55.13 | prah | i was curious if someone would help me with free pbx |
| 04:57.56 | *** part/#asterisk prah (n=karl_m_j@cpe-075-177-109-229.triad.res.rr.com) |
| 05:00.18 | fofware | Hello, I'm running Asterisk14 in OpenWrt and when I try to use Set(CALLERID...., I get error msg, Function CALLERID not registered, anyone know in what module can be this function? |
| 05:01.13 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 05:07.45 | ChannelZ | there is a func_callerid.so |
| 05:09.39 | Katty | returns from movie |
| 05:09.59 | fofware | ChannelZ: thanks I need find in what package it's |
| 05:10.13 | ChannelZ | I'm not sure whose responsibility it is to load it |
| 05:10.42 | [TK]D-Fender | I'd love to know who packages a core module from * separately like that |
| 05:11.45 | hardwire | heh |
| 05:11.47 | fofware | [TK]D-Fender: hello :o) |
| 05:12.21 | Katty | jblack: i liked the movie |
| 05:13.51 | fofware | maybe in same package of VoiceMail |
| 05:15.05 | [TK]D-Fender | VoiceMail has its own package too? Why not just package EVERY module of * separately while you're at it... |
| 05:15.40 | [TK]D-Fender | wonders what retarded packaging setup is being referred to here... |
| 05:15.44 | ChannelZ | fofware: when you say "package" are you actually talking about a distro package? |
| 05:16.24 | ChannelZ | ok then |
| 05:17.11 | ChannelZ | I guess he must, openwrt == embedded.. so, tight on memory to just have everything bundled in I guess |
| 05:19.22 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 05:20.51 | jblack | Katty: Cool |
| 05:21.18 | jblack | I don't wanna blow you off, but I'm in the middle of something atm |
| 05:38.31 | *** join/#asterisk Tech_Travis (n=tech_tra@cpe-76-87-9-130.socal.res.rr.com) |
| 05:39.09 | Tech_Travis | Can the voicemail pin be encypted like the secret password in sip.conf? |
| 05:45.21 | [TK]D-Fender | Tech_Travis: no |
| 06:00.29 | Tech_Travis | D-Fender: okay, thanks. |
| 06:02.46 | *** join/#asterisk lordmortis (n=lordmort@124.169.192.175) |
| 06:07.12 | lordmortis | anyone can tell me about requirecalltoken and older asterisk clients? |
| 06:07.57 | lordmortis | (i'm getting an error saying that the client doesn't support calltoken but i can't see how to turn it off, have tried requirecalltoken=no in the context for that user and it's not working?) |
| 06:15.55 | snadge | elastix doesnt mention xen anywhere? |
| 06:16.05 | snadge | it just looks like its based on centos.. similar to asterisknow |
| 06:19.32 | *** join/#asterisk cmendes0101 (n=none@173.58.40.28) |
| 06:20.35 | *** join/#asterisk mumtazah (n=mumtazah@60.51.18.40) |
| 06:22.44 | *** join/#asterisk cmendes0101 (n=none@173.58.40.28) |
| 06:49.57 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 07:04.57 | Sier | I'm trying to install asterisk-addons 1.6* , and I'm getting this error: http://pastebin.com/m72398650 |
| 07:07.40 | ChannelZ | hmm what are those funky characters? |
| 07:07.57 | *** join/#asterisk chendy (n=chatzill@116.30.194.113) |
| 07:08.30 | Sier | ChannelZ very good question.. I'm trying to install an older version now |
| 07:11.22 | Sier | with an older version now I'm getting errors related to mysql |
| 07:11.36 | ChannelZ | progress! |
| 07:11.52 | Sier | http://pastebin.com/d412b0157 |
| 07:14.26 | ChannelZ | do you have the mysql development stuff installed? |
| 07:14.29 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 07:16.00 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 07:16.08 | b14ck | sup |
| 07:17.01 | ChannelZ | mah weiner |
| 07:17.13 | Sier | ChannelZ no |
| 07:17.20 | Sier | I'm trying to figure out which packages to install |
| 07:17.31 | b14ck | lol |
| 07:17.50 | Sier | http://pastebin.com/d691237eb |
| 07:21.14 | *** part/#asterisk mumtazah (n=mumtazah@60.51.18.40) |
| 07:21.49 | ChannelZ | probably none of those.. |
| 07:22.10 | ChannelZ | oh no wait I missed it the first time |
| 07:22.11 | ChannelZ | libmysqlclient15-dev |
| 07:24.57 | *** join/#asterisk cmendes0101 (n=none@173.58.40.28) |
| 07:27.27 | Sier | 1) Using res_config_mysql at the same time as res_config_odbc can create system instability on some systems. Please load only one or the other. |
| 07:27.49 | Sier | Just saw this note on the README @ asterisk-addons .. don't know what it means yet, but maybe this could be the solutions |
| 07:27.53 | Sier | solution* |
| 07:31.21 | ChannelZ | it means don't use two database drivers at once |
| 07:37.35 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 07:42.26 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 07:42.36 | DND | guys, what software can run ULAW files? |
| 07:43.12 | Sier | I'm getting mad @ asterisk-addons :( |
| 07:43.41 | DND | why? |
| 07:44.02 | Sier | won't install |
| 07:44.11 | DND | are you compiling it? |
| 07:44.23 | DND | or doing it via yum |
| 07:44.59 | Corydon76-dig | DND: do you mean read ulaw files? |
| 07:45.53 | Sier | compiling it |
| 07:45.57 | Sier | latest problem: http://pastebin.com/d2ce07ae7 |
| 07:46.33 | Corydon76-dig | Sier: run ./configure |
| 07:46.52 | Sier | I did, it ran clean |
| 07:47.20 | Corydon76-dig | On the same machine on which you're compiling? |
| 07:47.22 | Sier | http://pastebin.com/d299d9d8e = ./configure |
| 07:47.34 | Sier | yes, machine is IA64.. not sure if this has anything to do with it |
| 07:47.56 | DND | Corydon76-dig, well at least playing ulaw files |
| 07:48.19 | Corydon76-dig | DND: sox |
| 07:48.28 | DND | sorry for windows |
| 07:48.39 | Corydon76-dig | I don't use windows |
| 07:48.54 | DND | well its great that sox is cross platform :D |
| 07:48.59 | DND | just saw it |
| 07:49.12 | *** join/#asterisk chendy_ (n=chatzill@116.30.194.113) |
| 07:49.36 | Corydon76-dig | Sier: install libmysql-dev |
| 07:54.36 | Sier | still nothing.. |
| 07:54.40 | Sier | I'm going to install version 1.4* |
| 07:54.58 | *** join/#asterisk mattwj2002 (n=matt@Wikisource/Mattwj2002) |
| 07:54.59 | ChannelZ | what specifically from addons are you wanting? |
| 07:55.37 | Sier | I'm following the FreePBX guide(http://www.corenetworks.com.au/wiki/doku.php?id=install_voip_server), which includes addons |
| 07:55.42 | mattwj2002 | hi guys |
| 07:55.49 | Sier | but instead of installing 1.4, I was trying 1.6 |
| 07:56.06 | mattwj2002 | what is a good chatroom to talk to about Google Voice? |
| 07:56.31 | mchou | mattwj2002: righet here :) |
| 07:56.38 | mchou | right* |
| 07:56.47 | mattwj2002 | sweet |
| 07:56.49 | mattwj2002 | :D |
| 07:56.58 | mattwj2002 | I am very impressed |
| 07:57.15 | mattwj2002 | especially with the voice to text |
| 07:57.18 | mchou | what part specifically? |
| 07:57.24 | mchou | nm |
| 07:57.38 | mattwj2002 | :) |
| 07:57.46 | mattwj2002 | have you tried it mchou? |
| 07:57.49 | mchou | visual voicemail is great |
| 07:57.54 | mattwj2002 | price isn't bad too |
| 07:58.03 | mchou | I've been with GV for 2 yrs |
| 07:58.11 | mattwj2002 | nice |
| 07:58.16 | mchou | back when it was grand central |
| 07:58.37 | mattwj2002 | neat |
| 07:58.49 | mattwj2002 | yeah I have 5 phone numbers :-s |
| 07:59.05 | mchou | no that they bought gizmo5 I hope they whip gizmo5 into shape |
| 07:59.15 | mchou | now* |
| 07:59.20 | mattwj2002 | me too |
| 07:59.37 | mattwj2002 | I think they are going after Skype |
| 07:59.43 | mchou | gizmo5 has been suckage for a very long time |
| 08:00.57 | mchou | just wish gv allowed direct sip connection |
| 08:01.39 | mattwj2002 | yeah that would be nice |
| 08:01.41 | mchou | but I guess they're worrkied about fraud and abuse |
| 08:02.02 | mattwj2002 | good point |
| 08:02.38 | mchou | imagine telemarketrs calling you up on you GV # |
| 08:02.40 | mattwj2002 | with gizmo can't you hookup sip some how? |
| 08:02.49 | mchou | that would be total suckage |
| 08:02.53 | mattwj2002 | o.O |
| 08:03.04 | mattwj2002 | indeed |
| 08:03.07 | mchou | I dont want to go thru gizmo |
| 08:03.21 | mattwj2002 | I don't blame you |
| 08:03.24 | mchou | unreliable, high latency, and buggy as all hell |
| 08:04.07 | mattwj2002 | I am sure Google will clean some of that up |
| 08:04.08 | mattwj2002 | :) |
| 08:04.10 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 08:04.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 08:04.19 | mchou | not to mention no one can keep up with their TOS that changes every month |
| 08:05.20 | mattwj2002 | yeah it does suck |
| 08:06.01 | mattwj2002 | but Google Voice isn't meant to be like level3 |
| 08:06.10 | mattwj2002 | at least not right now |
| 08:06.19 | mchou | no, definitely not |
| 08:06.53 | mattwj2002 | for a basic consumer product it works well |
| 08:06.54 | mattwj2002 | :) |
| 08:07.28 | mchou | mattwj2002: why do you need 5 gv #s? |
| 08:07.47 | mchou | the whole point is to considate into one # |
| 08:07.48 | mattwj2002 | oh I don't have 5 gv #s |
| 08:07.55 | mchou | consolidate* |
| 08:07.57 | mattwj2002 | I have 5 telephones |
| 08:08.00 | mattwj2002 | oops |
| 08:08.05 | mattwj2002 | I have 5 phone numbers! |
| 08:08.12 | mattwj2002 | I have one google number |
| 08:08.13 | mattwj2002 | :) |
| 08:08.15 | mchou | ok |
| 08:08.22 | mchou | that makes more sense |
| 08:08.23 | mattwj2002 | 3 are voip |
| 08:08.55 | mchou | which voip providers you using? |
| 08:09.05 | mattwj2002 | hehe it is kind of funny.....Google attacks me when it rings |
| 08:09.12 | mattwj2002 | nothing fancy |
| 08:09.18 | mattwj2002 | just Skype and Vonage |
| 08:09.28 | mattwj2002 | went the lazy routes |
| 08:09.28 | mchou | vontage |
| 08:09.34 | mattwj2002 | yeah :( |
| 08:09.44 | mattwj2002 | more like bonage |
| 08:09.45 | mattwj2002 | :( |
| 08:09.52 | mchou | that too |
| 08:10.01 | mattwj2002 | have you ever used ooma? |
| 08:10.07 | mchou | how much you paying for vonage? |
| 08:10.14 | mattwj2002 | $33 per month |
| 08:10.25 | mattwj2002 | that is with tax and all |
| 08:10.28 | mchou | I used ooma demo at Frys. Wasnt impressed |
| 08:10.34 | mattwj2002 | really? |
| 08:10.36 | mattwj2002 | why? |
| 08:10.38 | mchou | noisy as all get out |
| 08:10.45 | Sier | why pay if you can get it for free? |
| 08:11.14 | mchou | yeah, plus I havent figured out ooma's business plan |
| 08:11.29 | mchou | to me it looks just like sunrocket |
| 08:11.41 | mchou | pay up now, fold later |
| 08:11.42 | mattwj2002 | Sunrocket? |
| 08:11.54 | mchou | yeah, defunct |
| 08:11.58 | mattwj2002 | bummer |
| 08:12.14 | mattwj2002 | I actually like Skype.... |
| 08:12.26 | mchou | Skype used to suck too |
| 08:12.29 | mattwj2002 | I have done Asterisk....I just get tired of maintaining it |
| 08:12.32 | mchou | no cid |
| 08:12.38 | mchou | back in the day |
| 08:12.47 | mchou | dropped calls |
| 08:13.11 | mchou | I remember the days when skype couldnt even process credit cards |
| 08:13.19 | mattwj2002 | wow |
| 08:13.20 | mchou | that was SUPER lame |
| 08:13.22 | mattwj2002 | that is crazy |
| 08:13.34 | mattwj2002 | yeah skype still drops calls |
| 08:13.57 | mchou | that occured the same time when they had 1yr call usa unlimited for like $35 |
| 08:14.01 | mattwj2002 | but it works fine for talking to my dad |
| 08:14.07 | Sier | can you setup auto-attendants with skype? |
| 08:14.07 | mchou | for the whole year |
| 08:14.13 | mattwj2002 | nice |
| 08:14.39 | mchou | Sier: if you use asterisk skype, yeah :) |
| 08:14.42 | mattwj2002 | they have it for $30 per year if you get unlimited outgoing for about the same price |
| 08:14.55 | mattwj2002 | is that out? |
| 08:14.56 | mattwj2002 | :D |
| 08:15.05 | mchou | that is available |
| 08:15.09 | mchou | right now |
| 08:15.21 | mattwj2002 | :O |
| 08:15.24 | mchou | you have to pay for the module |
| 08:15.25 | mattwj2002 | how much? |
| 08:15.42 | mchou | I think it's around $60/yr or so |
| 08:16.01 | mchou | dont recall, really |
| 08:16.13 | mattwj2002 | :D |
| 08:16.22 | mattwj2002 | puts it on his Christmas list |
| 08:16.28 | mattwj2002 | seriously I might |
| 08:16.37 | mchou | I dunno |
| 08:17.02 | mchou | that asterisk sype module has all sorts of crazy limitations |
| 08:17.12 | mchou | read the TOS very carefully |
| 08:17.19 | mattwj2002 | $66 |
| 08:17.22 | mchou | skype* |
| 08:17.57 | mattwj2002 | for a single channel though |
| 08:17.58 | mattwj2002 | :( |
| 08:18.06 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 08:19.00 | mattwj2002 | mchou what else can you tell me about it? |
| 08:19.09 | mchou | about what? |
| 08:19.12 | mattwj2002 | do you need to install Skype on that server? |
| 08:19.16 | mattwj2002 | skype for Asterisk |
| 08:19.17 | mchou | nope |
| 08:19.21 | mchou | stand alone |
| 08:19.31 | mchou | exccept for asterisk, of course |
| 08:20.06 | mchou | you install the skype module on the asterisk server |
| 08:20.20 | mattwj2002 | hehe |
| 08:20.23 | mchou | and off you go |
| 08:20.27 | mattwj2002 | :D |
| 08:21.09 | mattwj2002 | well I have two DIDs through Skype..... |
| 08:21.21 | mattwj2002 | how does that work if you have a single license? |
| 08:21.25 | mchou | even though asterisk is a bitch to maintain I still prefer sip |
| 08:21.49 | mattwj2002 | definitely |
| 08:21.50 | mchou | mattwj2002: beats me |
| 08:21.59 | mattwj2002 | sip or better yet iax |
| 08:22.11 | mchou | I havent really looked into it since I'm still leary of skype |
| 08:22.11 | mattwj2002 | I actually prefer iax over sip |
| 08:22.20 | mchou | iax is buggy |
| 08:22.36 | mattwj2002 | well here is the thing.... |
| 08:23.02 | mattwj2002 | I lay down $66 bucks and I have an Asterisk server with unlimited incoming and outgoing calls |
| 08:23.10 | mattwj2002 | :D |
| 08:23.34 | mchou | I lay down $0 with google voice I have the same thing :) |
| 08:23.43 | mattwj2002 | hehe |
| 08:23.48 | mattwj2002 | gizmo5? |
| 08:23.56 | mchou | nah |
| 08:24.11 | mattwj2002 | how then? |
| 08:24.13 | mchou | any voip provider that does inbound for free |
| 08:24.21 | mattwj2002 | oh yeah |
| 08:24.28 | mattwj2002 | :) |
| 08:25.04 | mchou | I've been pretty impressed with sipgate so far in US |
| 08:25.36 | mchou | 15ms reply from my asterisk |
| 08:25.51 | mchou | even my router isn't that close :) |
| 08:26.07 | mattwj2002 | O_o |
| 08:26.11 | mattwj2002 | 15 ms? |
| 08:26.17 | mattwj2002 | dang! |
| 08:26.18 | mchou | yup |
| 08:26.21 | mchou | speedy |
| 08:26.25 | mattwj2002 | do you live in California? |
| 08:26.30 | mchou | yup |
| 08:26.39 | mchou | tracerout is like 8 hops |
| 08:26.42 | mchou | iirc |
| 08:26.51 | mchou | phenomenal |
| 08:27.13 | mattwj2002 | nice |
| 08:27.15 | mattwj2002 | :) |
| 08:28.07 | mchou | it's just too bad sipgate only has numbers in timbuktu right now |
| 08:28.24 | mchou | rural areas that are not even on the map |
| 08:28.44 | mattwj2002 | :) |
| 08:28.55 | mchou | umm, free numbers |
| 08:29.17 | mchou | I'm sure with paid you get better choices |
| 08:30.43 | mattwj2002 | nice |
| 08:30.47 | mattwj2002 | free aye? |
| 08:30.59 | mchou | 1 free inbound # |
| 08:31.30 | mattwj2002 | nice |
| 08:31.31 | mattwj2002 | :D |
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| 08:32.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 08:33.09 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 08:33.09 | *** mode/#asterisk [+o angler] by ChanServ |
| 08:33.44 | *** join/#asterisk oej (n=olle@87.96.134.125) |
| 08:33.50 | Corydon76-dig | I don't quite understand this, but apparently outbound to Canada on my provider is $0.10/minute, yet incoming from Canada to my toll-free number is $0.06/minute. |
| 08:34.07 | mchou | Corydon76-dig: hah |
| 08:34.11 | Corydon76-dig | You'd think toll-free would always be more expensive... |
| 08:34.46 | mchou | not if they charge more up front for your toll free DID :) |
| 08:34.55 | Corydon76-dig | $5/mo |
| 08:35.04 | mchou | wow |
| 08:35.10 | mchou | that's pretty good |
| 08:35.20 | Corydon76-dig | binfone.com |
| 08:35.21 | mchou | Corydon76-dig: whic provider is this? |
| 08:35.26 | mchou | nm :) |
| 08:35.47 | Corydon76-dig | delivery over IAX, too |
| 08:35.59 | mchou | yeah, iax is overrated |
| 08:36.06 | mchou | buggy |
| 08:36.07 | Corydon76-dig | or SIP, your choice |
| 08:36.37 | Corydon76-dig | Eh, there's far more bugs on SIP... |
| 08:36.48 | mchou | you'd think so |
| 08:36.59 | Corydon76-dig | I'm a developer. I know so. |
| 08:36.59 | mchou | but iax connects are terrible |
| 08:37.38 | Corydon76-dig | In my experience IAX connects work just fine |
| 08:37.42 | mchou | iax gets all wonky |
| 08:37.52 | mchou | sip at least works |
| 08:38.02 | Corydon76-dig | Quantity 'wonky' |
| 08:38.19 | mattwj2002 | well I need to go to bed |
| 08:38.22 | mattwj2002 | thanks everyone |
| 08:38.24 | mchou | wonky mean connections get dropped mysteriously |
| 08:38.28 | mattwj2002 | thanks mchou! |
| 08:38.36 | mattwj2002 | good night all! |
| 08:38.49 | Corydon76-dig | mchou: it's not mysterious at all. Same things happen on SIP. |
| 08:39.04 | *** part/#asterisk mattwj2002 (n=matt@Wikisource/Mattwj2002) |
| 08:39.10 | mchou | I rarely have dropped connection on sip |
| 08:39.20 | Corydon76-dig | Generally due to non-response on a critical packet |
| 08:39.38 | mchou | iax you can count on dropped connections like clockwork |
| 08:39.47 | Corydon76-dig | I submit that you're comparing different providers on IAX versus SIP |
| 08:39.57 | mchou | nope |
| 08:40.00 | Corydon76-dig | FUD |
| 08:40.01 | mchou | same providers |
| 08:40.08 | Corydon76-dig | Which providers? |
| 08:40.28 | mchou | it's possible they implemented stuff incorrectly on their end |
| 08:40.33 | mchou | but I doubt it |
| 08:40.44 | Corydon76-dig | GIGO |
| 08:41.00 | *** join/#asterisk mextlan (n=mextlan@212.179.28.1) |
| 08:41.03 | Corydon76-dig | You doubt that crappy software engineers exist? |
| 08:41.04 | mchou | considering the same stuff happened with multiple providers |
| 08:41.11 | *** part/#asterisk icyValk77 (n=icyValk7@host81-155-31-214.range81-155.btcentralplus.com) |
| 08:41.12 | Corydon76-dig | Which providers? |
| 08:41.34 | Corydon76-dig | C'mon, now, if you're going to say that IAX sucks, you need to say which providers suck with IAX... |
| 08:41.41 | mchou | Corydon76-dig: why? |
| 08:41.45 | Corydon76-dig | because I've had ZERO problems using IAX |
| 08:41.52 | mchou | Corydon76-dig: these are reputable providers |
| 08:42.08 | Corydon76-dig | mchou: apparently not, if they're dropping calls |
| 08:42.22 | mchou | dude, everyone in this channel recommends sip over iax |
| 08:42.43 | Corydon76-dig | Dude, I'm in this channel (an op, even), and I don't |
| 08:42.49 | mchou | they arent dropping calls |
| 08:43.00 | mchou | they are dropping connections |
| 08:43.02 | Corydon76-dig | You're contradicting yourself now |
| 08:43.44 | Corydon76-dig | Just admit that you have no real basis for claiming that IAX sucks |
| 08:43.45 | mchou | no, apparent you're to dense to understand the difference between connection and call |
| 08:44.00 | mchou | I'd be lucky even to get a call |
| 08:44.17 | *** mode/#asterisk [+b %mchou!*@*] by Corydon76-dig |
| 08:44.25 | ChannelZ | maybe you don't know what you're doing |
| 08:44.41 | Corydon76-dig | No trolling |
| 08:45.13 | ChannelZ | Not trolling. This conversation is cyclical with him |
| 08:45.44 | ChannelZ | I too wonder why if IAX works so poorly there isn't rioting on the net about it. This is the first I'm hearing of it.. |
| 08:46.18 | Corydon76-dig | I'm sure some people wish it would just go away and make their lives simpler |
| 08:46.38 | Corydon76-dig | Kind of like how the ITU wishes SIP would go away and everybody would just use H.323 |
| 08:46.51 | ChannelZ | heh I thought IAX did make life simpler :) |
| 08:47.40 | Corydon76-dig | I'm going to bed. |
| 08:47.55 | ChannelZ | should |
| 08:49.55 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 08:50.05 | fofware | hello, What I doing wrong? I have Asterisk1.6 running and others asterisk1.4 conneted with iax, each one have different context but all incoming call go to context of first one |
| 08:53.54 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:54.48 | ChannelZ | hard to say without seeing configs |
| 09:00.28 | *** join/#asterisk Malkor (n=marco@hlle-d9ba5fe6.pool.mediaWays.net) |
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| 09:08.44 | TJNII | Pong, to whoever pinged me |
| 09:08.48 | TJNII | Hours ago.... |
| 09:14.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
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| 09:56.11 | elliot98 | anybody home? |
| 10:02.31 | TJNII | Ask a question and see if you get an answer/ |
| 10:03.37 | kaldemar | well, that was a question. the answer is yes. |
| 10:04.39 | TJNII | kaldemar: You know what I meant, Mr. literal. |
| 10:05.42 | kaldemar | TJNII: of course i do. :) i wasn't picking on you, but him. |
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| 10:30.18 | Meaw | hi, anyone using sangoma here? it was working fine until the teleco company here changed something now when i do: wanrouter summary it says "Disconnected" |
| 10:30.43 | Meaw | I checked the E1 cable, it is working fine |
| 10:31.05 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 10:36.43 | *** join/#asterisk lozarythmic (n=loz@80-45-84-233.static.dsl.as9105.com) |
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| 10:47.17 | elliot98 | hi y'all! |
| 11:06.34 | *** join/#asterisk DND (n=arabia@80.227.221.34) |
| 11:07.03 | DND | guys, any software that can play ulaw in windows? |
| 11:10.21 | *** join/#asterisk Tako-san (n=jmkiffia@p1216-ipad308osakakita.osaka.ocn.ne.jp) |
| 11:13.38 | DND | ok let me change that. which takes a lot of resources? ongoing call + recording(always) set to ulaw format OR ongoing call + recording(always) set to wav format? |
| 11:14.47 | TJNII | DND: Depends on the call format |
| 11:16.51 | DND | the call format is g711 |
| 11:17.04 | DND | because we are using x-lite |
| 11:20.36 | tzafrir_laptop | here's something to test: make Asterisk dial (in DTMFs) a number such as 818181<something> |
| 11:20.58 | tzafrir_laptop | If busydetect is enabled and busycount=7 , this will cause a hangup |
| 11:20.59 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.236.196) |
| 11:23.43 | bn-7bc | +7, |
| 11:23.50 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
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| 11:31.54 | Meaw | hi, i have two sangoma card.. was working fine, but the teleco changed something now when i plug the E1 cable it says "Disconnected" |
| 11:32.00 | Meaw | i tested the cable, it works fine |
| 11:32.03 | Meaw | any thoughts? |
| 11:32.43 | DND | did you asked the telco if they also changed something on their settings? |
| 11:33.11 | Meaw | they asked me to disable the crc in quintum, I did in quintum and it works fine |
| 11:33.23 | Meaw | but i have no idea what to do in sangoma settings to get it working |
| 11:38.57 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 12:06.51 | *** join/#asterisk lordmortis (n=lordmort@124-169-65-230.dyn.iinet.net.au) |
| 12:22.21 | TJNII | God, this song is terrible. About a little crippled girl named Mary Christmas living in an orphanage. I think it needs to go in MoH. |
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| 13:20.43 | dlynes | DND: I believe audacity can play ulaw in windows (http://audacity.sf.net/) |
| 13:24.32 | *** join/#asterisk Faithful (n=Faithful@115.147.205.174) |
| 13:24.39 | dlynes | Meaw: was the quintum showing disconnected in its status as well? |
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| 13:53.47 | *** join/#asterisk voipmonk (n=voipmonk@69.172.97.20) |
| 13:56.31 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 14:05.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:06.14 | *** join/#asterisk tlir (n=tlir@bzq-179-128-126.static.bezeqint.net) |
| 14:10.23 | *** join/#asterisk dD0T (n=dD0T@unaffiliated/dd0t) |
| 14:14.45 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 14:18.08 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 14:26.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:32.38 | *** join/#asterisk doubletoker (n=lol@adsl-218-255-35.jax.bellsouth.net) |
| 14:32.43 | doubletoker | hello |
| 14:32.52 | doubletoker | quick question if anyone is here |
| 14:33.34 | doubletoker | I'm just learning asterisk, and I'm wondering when someone hangs up the phone |
| 14:34.02 | doubletoker | can your dialplan run clean up scripts when that happens? |
| 14:37.57 | russellb | doubletoker: Yes. Look at the 'h' extension |
| 14:38.02 | russellb | That is exactly what it is for. |
| 14:38.32 | russellb | exten => h,1,NoOp(OMG ${CHANNEL} hung up!} |
| 14:38.47 | acxty | Hi guys, Is there a way to listen someone else conversation. I am trying to configure a mini callcenter. What I am trying to so is that a supervisor can hear what the agent is doing |
| 14:39.03 | russellb | acxty: *CLI> core show application ChanSpy |
| 14:39.11 | acxty | thanks |
| 14:40.59 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 14:42.37 | *** part/#asterisk doubletoker (n=lol@adsl-218-255-35.jax.bellsouth.net) |
| 14:44.32 | *** join/#asterisk KingDavidNYC (n=Chris123@pool-96-224-162-219.nycmny.east.verizon.net) |
| 14:45.12 | *** join/#asterisk Tim_Toady (n=moi@adsl113-172.kln.forthnet.gr) |
| 14:45.37 | KingDavidNYC | hello everybody!! |
| 14:48.13 | KingDavidNYC | anybody here? |
| 14:48.16 | voipmonk | yes |
| 14:48.19 | voipmonk | whats up? |
| 14:48.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:49.10 | KingDavidNYC | hi! |
| 14:49.16 | voipmonk | hello! |
| 14:49.21 | KingDavidNYC | please help, I ha ve a weird problem |
| 14:49.31 | voipmonk | whats the problem? |
| 14:49.41 | voipmonk | and make sure you use pastebin.ca for debug information |
| 14:49.50 | KingDavidNYC | I can't send a freaking invite! |
| 14:49.57 | voipmonk | dang |
| 14:49.59 | voipmonk | you're cursed |
| 14:50.21 | voipmonk | someone said.... screw that guy.... lets get him before he can send an invite.... cuz no one will come anyway! |
| 14:50.48 | KingDavidNYC | it is on a linksys ATA, and I have to send it as a sip trunk... |
| 14:51.08 | KingDavidNYC | for some reason, the outgoing number disappears from the invite header |
| 14:51.29 | Pan3D | "It's like throwing a party where..." |
| 14:52.11 | KingDavidNYC | I want to force a "sip:outgoingnumber@ataip:ataport" to the ATA |
| 14:52.26 | KingDavidNYC | anybody's done that before? |
| 14:52.28 | voipmonk | brb |
| 14:52.31 | voipmonk | rebooting |
| 14:53.42 | [TK]D-Fender | KingDavidNYC: Show us your configs and what you;re doing |
| 14:53.51 | KingDavidNYC | sure |
| 14:56.06 | KingDavidNYC | here is the pastebin of the extension as siptrunk |
| 14:56.08 | KingDavidNYC | http://pastebin.com/m67bd4419 |
| 14:56.36 | KingDavidNYC | linksys 3102, I want to treat the pstn port as a sip trunk |
| 14:56.46 | *** join/#asterisk voipmonk (n=voipmonk@69.172.97.20) |
| 14:58.36 | KingDavidNYC | that way, I can send exten => xxx,3,Dial(SIP/phonenumber@test3102) |
| 14:58.36 | KingDavidNYC | the idea is that I want the pstn port to receive the invite, as if it was a sip trunk |
| 14:59.06 | KingDavidNYC | this, supposedly, will cause the ATA to dial-out the number on its own |
| 15:00.08 | KingDavidNYC | I do receive the invite..but in the format To: sip:X.X.X.X@ip:port |
| 15:00.23 | KingDavidNYC | but the outgoing number dissappears :( |
| 15:01.45 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:01.59 | voipmonk | what ata are you using? |
| 15:02.05 | [TK]D-Fender | KingDavidNYC: pastebin the call attempt with SIP DEBUG enabled |
| 15:03.06 | KingDavidNYC | ok, but I can only do a cut and paste of the last part of the cli, as I dont know how to route the output to a log file |
| 15:04.00 | [TK]D-Fender | KingDavidNYC: todays magic words are "scrollback buffer" |
| 15:04.10 | voipmonk | hides his ">" |
| 15:04.16 | Pan3D | sits down on his mat and waits for a story |
| 15:04.40 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:07.15 | *** join/#asterisk Zambezi (i=Zulu@80.67.9.2) |
| 15:07.23 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 15:07.44 | voipmonk | C'mon KingDavidNYC :) did you fix your scrollback? You may be able to set it to unlimited or at least 2000 |
| 15:08.05 | KingDavidNYC | tk-fender: I do, but I would prefer to capture the entire thing |
| 15:08.10 | KingDavidNYC | here is the pastebin |
| 15:08.12 | KingDavidNYC | http://pastebin.com/m7a89e749 |
| 15:08.50 | KingDavidNYC | key is this line: 106.***********->>>>>>>>>>>>>HERE->>>>>>>>To: <sip:96.224.162.219:5061> |
| 15:09.03 | KingDavidNYC | where absent is the outgoing number |
| 15:09.05 | [TK]D-Fender | KingDavidNYC: Now try again with VERBOSE 10 |
| 15:12.01 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 15:12.17 | KingDavidNYC | ok, here it is: http://pastebin.com/m342fe5e8 |
| 15:13.22 | KingDavidNYC | I have syslog monitor open on the ATA, and I just dont see anything reaching the ATA |
| 15:13.31 | [TK]D-Fender | KingDavidNYC: .... try again |
| 15:13.37 | KingDavidNYC | ok |
| 15:16.17 | KingDavidNYC | here is another bin: http://pastebin.com/m4b35ef68 |
| 15:19.09 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 15:19.25 | KingDavidNYC | I also did a wireshark trace, same thing, the ATA does not receive the packets |
| 15:19.50 | *** join/#asterisk bn-7bc (i=bjarne@mac.lan.noare-1.holmedal.net) |
| 15:20.19 | KingDavidNYC | ALso, I wish I could just sent a fake invite, but every tool I have tried forces a pre-registration of the line |
| 15:26.11 | [TK]D-Fender | kindOK, we've got a problem here... you aren't showing me the ENTIRE call. I don't see your Dial() being issued) |
| 15:26.41 | [TK]D-Fender | KingDavidNYC: ^ |
| 15:27.17 | KingDavidNYC | you mean, seeing that on the debug? |
| 15:27.22 | [TK]D-Fender | KingDavidNYC: Yes |
| 15:27.51 | *** join/#asterisk bbt (n=sam@180.189.139.92) |
| 15:27.53 | KingDavidNYC | I issue the dial on the dialplan---> exten => +17868377115,3,Dial(SIP/12128665414@test3102) |
| 15:28.11 | KingDavidNYC | I call into that +1786 which is a DID |
| 15:28.48 | KingDavidNYC | but on the last debug I sent you, that was the entire debug |
| 15:30.15 | voipmonk | 12128665414 isnt listed in either of those pastebin posts |
| 15:30.15 | [TK]D-Fender | KingDavidNYC: Where is the VEBOSE I asked for? |
| 15:30.26 | [TK]D-Fender | KingDavidNYC: We don't see your dialplan being executed |
| 15:30.37 | KingDavidNYC | I set it to 10 |
| 15:31.32 | *** join/#asterisk jmacz (n=jmacz@190.25.115.67) |
| 15:31.49 | [TK]D-Fender | KingDavidNYC: there is no verbose in there |
| 15:32.04 | [TK]D-Fender | KingDavidNYC: So either your selection is bad or your ability to follow instructions |
| 15:32.17 | KingDavidNYC | by the way, only the dialing is supposed to happen, but the ATA will never be able to terminate the call |
| 15:32.40 | KingDavidNYC | do you want to see the dialplan executing part? |
| 15:32.53 | voipmonk | yes. |
| 15:32.56 | [TK]D-Fender | KingDavidNYC: I want to see the ENTIRE call. |
| 15:32.57 | KingDavidNYC | ok |
| 15:33.21 | KingDavidNYC | I am going to set verbose to 99 |
| 15:33.39 | voipmonk | what would really be helpful is if u can include that in the sip debug. |
| 15:33.56 | KingDavidNYC | how do I increase scrollback? |
| 15:33.56 | voipmonk | from just before you make the call to after you hangup |
| 15:34.23 | voipmonk | you will need to look in the preferences of your terminal program |
| 15:34.31 | voipmonk | what are you using , btw? |
| 15:35.38 | KingDavidNYC | putty.exe |
| 15:36.50 | voipmonk | KingDavidNYC: http://www.softpedia.com/screenshots/PuTTY-Tray_4.png see where it says scrollback |
| 15:37.00 | voipmonk | sorry , lines of scrollback |
| 15:37.20 | KingDavidNYC | no idea |
| 15:38.16 | KingDavidNYC | 200 lines!, cool, I am gonna increase it |
| 15:42.13 | KingDavidNYC | ok guys, I got the entire debug, here: http://pastebin.com/m177dd40c |
| 15:42.18 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) |
| 15:42.25 | KingDavidNYC | verbose=50 |
| 15:42.27 | DigitalFlux | Guys |
| 15:42.38 | DigitalFlux | i need to create a UNIX user account on my asterisk box |
| 15:42.52 | DigitalFlux | where if any one SSHed into this machine with that account |
| 15:43.05 | DigitalFlux | he will get an asterisk CLI |
| 15:43.24 | DigitalFlux | and if he quited the CLI he gets logged out |
| 15:43.28 | DigitalFlux | any ideas on how to do that ? |
| 15:47.31 | voipmonk | DigitalFlux: sure - you will need to make friends with the shell resource file |
| 15:47.54 | DigitalFlux | voipmonk: What is the shell resource file, /etc/shells ? |
| 15:49.31 | voipmonk | no DigitalFlux |
| 15:49.43 | voipmonk | if u create a user bumbum |
| 15:50.10 | voipmonk | you will find it in /home/bumbum/.bashrc <--------- if you use bash for example |
| 15:50.20 | voipmonk | thats where you want to work your magic |
| 15:52.11 | DigitalFlux | OK, i know the bashrc file, how can i use it for that purpose ? |
| 15:52.20 | DigitalFlux | what should i input in it ? |
| 15:52.36 | voipmonk | first you'll need to think about what you want to do |
| 15:52.45 | voipmonk | and how you would do it if you were to login yourself and manually do it |
| 15:52.55 | voipmonk | then add that into the file and test... good luck! |
| 15:53.25 | voipmonk | turns the SpoonFeed 3.0 down a notch... |
| 15:56.14 | *** join/#asterisk Malkor (n=marco@hlle-d9ba5fe6.pool.mediaWays.net) |
| 15:57.12 | DigitalFlux | OK here is another question about Asterisk Queues |
| 15:57.18 | DigitalFlux | using any strategy |
| 15:57.29 | DigitalFlux | consider round robin in this case |
| 15:57.49 | DigitalFlux | with 3 extensions in the queue |
| 15:57.52 | voipmonk | waits for the question |
| 15:58.10 | DigitalFlux | consider the first extension rang but no one answerered |
| 15:58.16 | DigitalFlux | and so the 2nd and the 3rd |
| 15:58.32 | DigitalFlux | the queue behaviour is to ring on the 1st one again |
| 15:58.59 | voipmonk | if there are only 3 members then sure |
| 15:59.06 | voipmonk | toundrobin is what you're looking for |
| 15:59.18 | DigitalFlux | what i need to do is don't ring again on the extension that no one answerered at |
| 15:59.24 | elliot98 | what do you think of this site: http://asteriskuptospeed.linuxinnovations.com ? |
| 15:59.52 | voipmonk | doesnt resolve |
| 16:00.11 | DigitalFlux | Can i configure the Queue Strategies "logic" in asterisk ? |
| 16:00.39 | elliot98 | oish...take out the subdomain |
| 16:00.47 | elliot98 | only http://linuxinnovations.com |
| 16:02.25 | [TK]D-Fender | DigitalFlux: vi app_queue.c |
| 16:03.30 | DigitalFlux | [TK]D-Fender: OK, any other way to do that function i need to do without going into C code ? :) |
| 16:03.59 | KingDavidNYC | tk-fender: tk, did you see my last bin with the entire call? |
| 16:04.05 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:05.04 | elliot98 | I'd like to know how it can further help people |
| 16:06.27 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:50e5:8c49:4941:ab15) |
| 16:08.36 | [TK]D-Fender | KingDavidNYC: line 343 = * isn't even trying to call it |
| 16:10.59 | KingDavidNYC | tk-fender: in that case, the question is: how do I send the invite to the ATA? |
| 16:11.56 | KingDavidNYC | [TK]-FENDER: 1) I have to treat the ATA port as if it was a sip port |
| 16:11.58 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-70.twcny.res.rr.com) |
| 16:12.42 | KingDavidNYC | [TK]-Fender: ... that's the only way the port is going to receive an outgoing number that it can then dial forward |
| 16:12.44 | [TK]D-Fender | KingDavidNYC: sip show peer test3102 |
| 16:13.13 | [TK]D-Fender | [11:11]<KingDavidNYC>[TK]-FENDER: 1) I have to treat the ATA port as if it was a sip port <- WTF? |
| 16:13.32 | KingDavidNYC | tk-fender: yes |
| 16:14.23 | KingDavidNYC | tk-fender: the entire thing is that I have to send a fake invite to the ATA |
| 16:14.48 | KingDavidNYC | tk-fender: this "fake" invite must have an outgoing number |
| 16:15.45 | KingDavidNYC | tk-fender: this will cause the ata to dial that number out |
| 16:16.13 | [TK]D-Fender | KingDavidNYC: WTF is a FAKE invite? You'r DIALING it. this is a REAL call. |
| 16:16.21 | [TK]D-Fender | KingDavidNYC: What is this nonsense you're going on about? |
| 16:16.22 | drmessano | lol |
| 16:17.10 | drmessano | InfiniteImprobability() |
| 16:17.13 | KingDavidNYC | tk-fender: It is not a real call, the ATA has no way of terminating the call, nor to accept registrations |
| 16:17.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.28 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 16:17.45 | drmessano | I register the SPA3102 to asterisk |
| 16:17.55 | drmessano | and send it calls all day long |
| 16:18.00 | drmessano | Never a problem |
| 16:18.03 | KingDavidNYC | tk-fender: the reason I am doing this is because if the ATA receives an invite, it will pickup the pstn line and will dial the dtmf digits |
| 16:18.17 | KingDavidNYC | tk-fender: that' is all I want to accomplish |
| 16:18.36 | drmessano | Register the PSTN port to Asterisk |
| 16:18.47 | drmessano | Then send it a call as you would any device |
| 16:18.51 | drmessano | peer/number |
| 16:18.54 | drmessano | Works fine |
| 16:20.24 | drmessano | You're WAY overthinking this |
| 16:20.50 | *** join/#asterisk reglisse (i=500d51f8@gateway/web/freenode/x-jdctgobnrcfueqxa) |
| 16:20.56 | reglisse | Hi everyine |
| 16:21.20 | elliot98 | does anyone know when 1.6.2 is due out? |
| 16:21.22 | reglisse | i'm looking for some help concerning Asterisk httpstatus issues ... |
| 16:21.42 | drmessano | elliot98: When its ready |
| 16:21.46 | reglisse | can someone help me please ? |
| 16:22.04 | KingDavidNYC | drmessano: mmmmm... it didn't work for me, but ok guys, I am gonna try it now ... I really been trying this in certain methods |
| 16:22.18 | drmessano | KingDavidNYC: been working for me for 2 years |
| 16:23.01 | drmessano | reglisse: Not really.. you haven't actually asked a question yet |
| 16:23.03 | KingDavidNYC | drmessano: I work with sipura 3000s like for 2 years |
| 16:23.08 | elliot98 | good answer |
| 16:23.33 | drmessano | KingDavidNYC: Ok, so you should have this done already |
| 16:23.47 | reglisse | drmessano: true , ok .... i've installed the asterisk server and configured manager.conf |
| 16:24.06 | reglisse | and httpd.conf correctly (checked with make checkconfig) |
| 16:24.48 | reglisse | the thing is when I run the command 'asterisk -rx "http show status"' .... the server is desabled and there is no prefix :( |
| 16:24.50 | KingDavidNYC | drmessano: actually, I want to make it as a dialer, where asterisk sends the invite, the ata dials the pstn line, and the the user picksup an analog handset connecte to the same line and continues the conversation |
| 16:25.03 | drmessano | reglisse: httpd.conf? |
| 16:25.16 | KingDavidNYC | drmessano: ok, I am gonna try it the obvious now :) |
| 16:25.20 | drmessano | reglisse: As in, Apache's config? |
| 16:25.25 | KingDavidNYC | be back |
| 16:25.34 | KingDavidNYC | \ |
| 16:25.41 | KingDavidNYC | sorry |
| 16:25.50 | drmessano | KingDavidNYC: Thats not gonna work |
| 16:26.09 | reglisse | drmessano: what do I have to do wirth Apache's config ? |
| 16:26.34 | drmessano | reglisse: NOTHING.. I asked YOU which httpd.conf you are talking about |
| 16:27.01 | reglisse | ... |
| 16:27.06 | reglisse | the asterisk one |
| 16:27.17 | reglisse | <PROTECTED> |
| 16:27.36 | reglisse | Here is what i get :: HTTP Server Status: Prefix: Server Disabled |
| 16:27.50 | drmessano | http.conf |
| 16:27.54 | drmessano | NOT httpd.conf |
| 16:28.41 | reglisse | yeah sorry |
| 16:28.49 | reglisse | that's the file, http.conf |
| 16:29.03 | drmessano | .... |
| 16:29.12 | drmessano | Pastebin the entire file |
| 16:30.08 | drmessano | KingDavidNYC: So somehow you want a phone connected to the PSTN line to get a ring generated by the SPA3102, and for it to answer and continue a call from Asterisk? |
| 16:30.14 | drmessano | KingDavidNYC: No, doesnt work that way |
| 16:32.19 | [TK]D-Fender | KingDavidNYC: I tond you about this before... ORIGINATE a call to the FXS port and when it answers, have it Dial the FXO. |
| 16:33.01 | [TK]D-Fender | drmessano: He's trying to make the most ass-backwards dialout system ever. Including running analog phones in PARALLEL with teh SPA FXO <- |
| 16:33.25 | reglisse | drmessano: here it is : http://pastebin.com/m34a950af ==> manager.conf, http.conf, and the result of the command |
| 16:33.27 | drmessano | [TK]D-Fender: Yeah, the last part of that is what got me |
| 16:35.10 | *** join/#asterisk tlir (n=tlir@77.126.35.195) |
| 16:36.42 | drmessano | reglisse: Have you reloaded or restarted asterisk since adding this config? |
| 16:37.12 | reglisse | drmessano: yes, several times .... |
| 16:38.29 | reglisse | drmessano: I just reloaded it (/rt |
| 16:38.42 | reglisse | and restarted it .... always the same |
| 16:39.42 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.21) |
| 16:41.07 | reglisse | drmessano: any guess ... |
| 16:42.05 | drmessano | Which version of Asterisk? |
| 16:42.45 | reglisse | Connected to Asterisk 1.6.2.0~rc2-0ubuntu1 |
| 16:43.02 | drmessano | hmmm |
| 16:43.17 | reglisse | drmessano: I've installed from the Karmic Ubuntu repos |
| 16:46.02 | drmessano | reglisse: Something is missing then |
| 16:46.23 | drmessano | reglisse: I just created http.conf, never had it set up before, and it's working |
| 16:46.33 | reglisse | :( |
| 16:46.59 | reglisse | drmessano: can you pastebin me your files ? |
| 16:47.14 | drmessano | Dude |
| 16:47.21 | drmessano | I used YOUR http.conf |
| 16:47.34 | drmessano | Apparently that package is missing something |
| 16:48.05 | reglisse | ^^ |
| 16:48.16 | reglisse | Sorry i'm a bit asleep |
| 16:48.52 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 16:49.15 | reglisse | drmessano: thanks for your help |
| 16:51.16 | *** join/#asterisk Iamnacho (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 16:52.33 | tlir | asterisk is sending SIP OPTIONS messages to extensions. is this behavior possible to configure so it wouldn't send these? |
| 16:53.01 | [TK]D-Fender | tlir: Its sending it because you set "qualify=yes" for them |
| 16:53.40 | tlir | [TK]D-Fender: thanks for that clarification. I'll look it up |
| 16:54.22 | *** mode/#asterisk [-b %mchou!*@*] by Corydon76-dig |
| 16:57.37 | *** join/#asterisk psilikon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 17:01.31 | KingDavidNYC | tk-fender: I'm back.. I can't do it as you say... where I call the voip port and then forward to the pstn... I must initiate the call on the pstn port |
| 17:02.04 | KingDavidNYC | tk-fender: the reason is that we dont want asterisk to pay for the call, we want the carrier on the pstn line to pay |
| 17:02.32 | KingDavidNYC | tk-fender: in this case, asterisk just sends the invite and then gets out of the call |
| 17:03.23 | drmessano | KingDavidNYC: You have this all wrong |
| 17:03.23 | KingDavidNYC | tk-fender: here is what I found in my research..."You setup the pstn line sip port ip address as an outgoing trunk. You send a sip invite to the pstn line ip address : port number with the digits to be dialed at that address. On the PSTN Line tab you setup Answer Call Without Reg: Yes, One Stage Dialing: Yes. The SPA3102 should receive the incoming sip invite and dial the number on the FXO port" |
| 17:03.35 | Katty | you know. |
| 17:03.53 | Katty | scrambled eggs, which were laid yesterday, are amazing. |
| 17:04.11 | voipmonk | somehow I read , you laid them |
| 17:04.15 | voipmonk | sorry |
| 17:04.16 | voipmonk | :) |
| 17:04.17 | drmessano | KingDavidNYC: Why does asterisk need to get out of the call path? |
| 17:04.21 | voipmonk | i prefer it my way tho |
| 17:04.22 | Katty | and that probably means having eggs that were laid on the same day... |
| 17:04.28 | KingDavidNYC | tk-fender: " |
| 17:04.28 | KingDavidNYC | If you have an analog handset connected to the pstn line itself you should be able to pickup the handset and listen to what is going on the pstn line." |
| 17:04.31 | Katty | well. talk about good breakfast (= |
| 17:05.24 | *** join/#asterisk manxpower (n=ewieling@16.sub-70-223-82.myvzw.com) |
| 17:05.50 | Katty | i also made fried potatoes. |
| 17:05.55 | Katty | with onion and mushrooms. |
| 17:06.07 | Katty | i think i might die from pure happiness. |
| 17:07.05 | KingDavidNYC | drmessano: * has to get out of the call because we dont want * to pay for the call... on a pstn line, if you pick up the line and dial a number, and then someone else picksup the same line, and you hangup, the call continues with the other person and is terminated by the carrier on the pstn line |
| 17:07.23 | KingDavidNYC | Katty: do you like to eat? |
| 17:07.28 | Katty | i like to cook |
| 17:08.04 | drmessano | KingDavidNYC: Asterisk doesnt pay for anything.. |
| 17:08.05 | Katty | it's mah hobby |
| 17:08.14 | mchou | haha |
| 17:08.35 | drmessano | KingDavidNYC: Asterisk is a switch, and jsut because it bridges a call between the FXS and FXO ports on the SPA-3102 doesnt mean it's dialing out your ITSP |
| 17:08.37 | manxpower | KingDavidNYC: Do your analog lines have this feature? (it's usually called Conference Drop and Transfer, not Three-Way calling). |
| 17:08.41 | drmessano | ~book |
| 17:08.41 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:08.51 | manxpower | Also just because you drop out of the call doesn't mean that you are not going to be billed for the call. |
| 17:08.52 | drmessano | Matter of fact, Asterisk doesnt do anything you dont tell it to do |
| 17:09.45 | Katty | KingDavidNYC: http://42ndrecipestreet.blogspot.com/ |
| 17:10.21 | KingDavidNYC | All being said, can somebody please help me do the following: "You setup the pstn line sip port ip address as an outgoing trunk. You send a sip invite to the pstn line ip address : port number with the digits to be dialed at that address. On the PSTN Line tab you setup Answer Call Without Reg: Yes, One Stage Dialing: Yes. The SPA3102 should receive the incoming sip invite and dial the number on the FXO port." |
| 17:11.01 | drmessano | KingDavidNYC: We're trying to.. thats BASIC SPA3102 FXO dialing |
| 17:11.06 | manxpower | KingDavidNYC: I managed to do that once. I followed the confusing docs on the internet. |
| 17:11.31 | voipmonk | chuckles |
| 17:11.35 | manxpower | But that has nothing to do with what you asked earlier. |
| 17:11.48 | KingDavidNYC | it is all the same |
| 17:12.00 | KingDavidNYC | I know it all sounds basic, but there is a tricky part |
| 17:12.05 | manxpower | voipmonk: Just about the only thing I don't like about SPAs is the confusing way they do their FXO port. |
| 17:12.31 | voipmonk | manxpower: I called KingDavidNYC and he refused my help, good luck |
| 17:12.42 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 17:12.47 | KingDavidNYC | the trikcy part is to start the call on the pstn |
| 17:12.52 | drmessano | manxpower: He wants to originate a call via the PSTN port on a SPA-3102 then have the call bridged to the FXS device on the SPA-3102.. thats one dial command.. and he refuses to listen |
| 17:13.27 | KingDavidNYC | voipmonk: I thank you hugely for your offer of help, but I hope you understand that I dont think this warrants me giving you access to my router |
| 17:13.30 | manxpower | drmessano: *shrug* Then what he is trying to do has nothing to do with Asterisk. |
| 17:13.52 | drmessano | manxpower: Yes it does, he wants Asterisk to originate the call |
| 17:14.06 | KingDavidNYC | voipmonk: besides anything else, I dont think I will learn |
| 17:14.11 | manxpower | drmessano: you just said he wants to originate the call on the FXO. |
| 17:14.28 | drmessano | manxpower: Yeah, DIAL from ASTERISK |
| 17:14.33 | KingDavidNYC | I want asterisk to send the INVITE, that will originate the call |
| 17:14.45 | manxpower | that would be originating the call from Asterisk |
| 17:14.59 | voipmonk | KingDavidNYC: empty your cup |
| 17:15.05 | manxpower | KingDavidNYC: Why do you want to do this crazy thing? |
| 17:15.49 | KingDavidNYC | sure, I will tell you... We sell a service for building magment |
| 17:15.51 | drmessano | KingDavidNYC: There is nothing tricky here.. this is a simple DIAL from Asterisk |
| 17:16.06 | KingDavidNYC | this app is to be installed in a front desk of a residential bldng |
| 17:16.34 | drmessano | DIAL(SIP/SPA3102PEER/Number) |
| 17:16.36 | KingDavidNYC | ...and uses the phone line on the desk of the porter |
| 17:17.07 | manxpower | KingDavidNYC: you have some kind of modem on the computer the app is runnig on? |
| 17:17.59 | KingDavidNYC | manxpwer: correct, this is done now with a modem, but I have managed to convince mangmt that this can be done a lot better with a linksys 3102 |
| 17:18.10 | KingDavidNYC | no drivers, no activex. etc |
| 17:18.35 | manxpower | If not then the "uses the phone line on the desk of the porter" is about as useful of a description as "magical fairies push magical phone buttons" |
| 17:18.55 | Katty | http://42ndrecipestreet.blogspot.com/2009/11/dads-fried-potatoes.html <- |
| 17:18.58 | voipmonk | chuckles |
| 17:19.09 | manxpower | So you have an application and it magically makes phone calls. OK. |
| 17:19.46 | manxpower | Personally I'd have the application connect to Asterisk via the Manager port and issue a Manager Originate command. |
| 17:19.51 | KingDavidNYC | what it will do is: porter pick the number of the apt it wants to call to, clicks on it, then asterisk sends an invite to the ATA, which will pickup the pstn line and dial the digits, just like the modem does now |
| 17:19.58 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 17:20.35 | manxpower | KingDavidNYC: Other than your very vague info on exactly how the application talks to Asterisk, this si done every day by millions of Asterisk users. |
| 17:20.41 | drmessano | voipmonk: The refusal is what I would expect.. He's convince you make a hamburger by some complicated trickery of baking two small loaves of bread with a hamburger patty in the middle |
| 17:20.44 | KingDavidNYC | manxpower: I could do that, but the problem is that we dont want asterisk to ring to the extension first |
| 17:21.03 | drmessano | So you dial the PSTN line first, then bridge the call |
| 17:21.15 | manxpower | KingDavidNYC: I never said anything about "ringing an extension". I said "dialout the fxo port" |
| 17:21.16 | drmessano | Again, basic |
| 17:21.24 | voipmonk | well... I did have an ulterior motive..... I wanted to use goat cheese inside the hamburger |
| 17:21.48 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 17:22.04 | manxpower | So you application asks Asterisk to dial out via SIP. (asterisk doesn't know what the port type is, that' all in the SPA) |
| 17:22.05 | KingDavidNYC | drmessano: I dial the pstn line first, and then dont do anything else.. the porter picks up the handset and continues the conversation... all call progress is made by parsing the syslog debug of the ATA |
| 17:22.19 | voipmonk | christ |
| 17:22.30 | drmessano | KingDavidNYC: Yes, again, this is BASIC SIMPLE ASTERISK |
| 17:23.31 | KingDavidNYC | manxpower: when you use the AMI to originate a call, the AMI first calls your extension, then you have to answer it, and then connects you with the outgoing number |
| 17:23.41 | manxpower | KingDavidNYC: YOU DO NOT HAVE TO DO IT THAT WAY |
| 17:23.50 | KingDavidNYC | ? |
| 17:23.52 | drmessano | You dial out the FXO then you dial the extensions.. There is no trickery.. and nothing fancy on the SPA-3102.. Matter of fact, it's 4 changes to the FACTORY SPA3102 config to make it work |
| 17:24.37 | psilikon | Can you register multiple ATAs and/or ATA telephone interfaces to one SIP account so they will ring simultaneously with an incoming call? |
| 17:25.02 | mchou | psilikon: that depends on the sip provider |
| 17:25.09 | drmessano | psilikon: You can, but thats not the best way to do it.. use multiple SIP accounts and Dial them all |
| 17:25.21 | drmessano | mchou: I think he means in Asterisk |
| 17:25.33 | KingDavidNYC | ok guys, truth is, I have not worked on those ATAS in years.. Perhaps I just need to dedicate a couple of hours to refresingh my knowledge opf its basic functionality |
| 17:25.36 | manxpower | KingDavidNYC: Make the 2nd leg of the Originate go to an extension that does a Wait(9999) |
| 17:25.44 | mchou | psilikon: if the provider uses (only) asterisk then no |
| 17:25.57 | drmessano | KingDavidNYC: This isnt an ATA problem.. The ATA config is no different |
| 17:26.16 | mchou | psilikon: if they use sip proxies then chances are yes |
| 17:26.24 | drmessano | mchou: WTF??? |
| 17:26.32 | mchou | hmm?? |
| 17:26.44 | mchou | wtf on what? |
| 17:26.50 | manxpower | asterisk does not allow you to register more than one device to the same SIP account. |
| 17:26.56 | KingDavidNYC | maxpower: but who answers the first leg? |
| 17:27.04 | psilikon | I am trying to have only one number per store and three phones (through ATAs). Callcentric tells me there are 3 incoming channels by default. How can I make each phone ring with an incoming call? |
| 17:27.14 | drmessano | He wants MULTIPLE phones to ring when a call comes in.. Youre going off on this whole multiple SIP registrations to the ITSP again.. not everyone wants to do this crazy shit you came up with |
| 17:27.16 | manxpower | KingDavidNYC: Depends on how you define the first leg. the non-PSTN leg can be answered by the dialplan. |
| 17:27.40 | drmessano | psilikon: You register all 3 to asterisk as different SIP devices and ring all 3 when a call comes in |
| 17:27.45 | mchou | drmessano: yes, but that's what he asked |
| 17:27.51 | drmessano | mchou: No its not |
| 17:28.12 | manxpower | psilikon: why not have one one incoming channel from the ITSP then handle the rest in standard Asterisk Dial(SIP/123&SIP.346&SIP/457) |
| 17:28.16 | mchou | drmessano: sure it is |
| 17:28.30 | drmessano | mchou: No, it is not.. reread.. |
| 17:28.39 | mchou | multiple concurrent sip registrations with same credentials |
| 17:28.52 | mchou | all endpoints ring |
| 17:28.59 | manxpower | mchou: he asked only because he doesn't know the correct question to ask. |
| 17:29.06 | drmessano | mchou: This is #ASTERISK not #ITSP, he didn't say he wanted them registered TO THE PROVIDER |
| 17:29.09 | mchou | first one get piked up, rest stop ringing |
| 17:29.15 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 17:29.19 | mchou | picked* |
| 17:29.22 | KingDavidNYC | manxpower: non-pstn answered by the dialplan, ok, and then with the dialplan I send the call to the pstn, and that will cause to ATA to put the pstn line off-hook and dial the numbers...mmm...could work |
| 17:29.29 | psilikon | drmessano, I was trying to avoid building a PBX for each store. Is this something an ITSP can typically handle. This question isn't so much asterisk related as it is general VoIP... I didn't know where else to ask this one... |
| 17:29.44 | drmessano | psilikon: Asterisk would handle each one |
| 17:29.58 | manxpower | psilikon: NO! You have 1 PBX, phones from threee stores register to the one Asterisk PBX |
| 17:29.58 | mchou | psilikon: google "sip parallel forking" |
| 17:30.05 | drmessano | .... |
| 17:30.15 | mchou | psilikon: it's mentioned in one of the SIP rfcs |
| 17:30.18 | manxpower | psilikon: ITSPs do not typically do what you want to do. |
| 17:30.19 | drmessano | Thats the WRONG way to do it |
| 17:30.34 | KingDavidNYC | maxpower: which is by the way, what TK-FENDER has been telling me all along :) |
| 17:30.44 | mchou | a lot of ITSPs support parallel forking |
| 17:30.52 | manxpower | KingDavidNYC: Why have you not been listening to him? |
| 17:30.52 | drmessano | Encouraging him to do so is stupid.. |
| 17:30.54 | mchou | voip.ms does |
| 17:31.08 | mchou | what's so stupid about it? |
| 17:31.19 | mchou | it's in the rfc, for christ sake |
| 17:31.38 | KingDavidNYC | manxpower: excuse me, but you kind of like were more detailed |
| 17:31.42 | drmessano | mchou: Just because it WORKS doesnt mean it's the best implementation for what he wants to do.. |
| 17:31.52 | mchou | Schulzinne didnt think it was stupid |
| 17:31.58 | manxpower | KingDavidNYC: [TK]D-Fender just expects you to read more docs. |
| 17:32.00 | psilikon | Hmmm.. so it might just be easier to get a hosted asterisk box and have it do all the work. |
| 17:32.08 | drmessano | psilikon: yes, indeed |
| 17:32.22 | KingDavidNYC | I am gonna try it, I will be around |
| 17:32.45 | manxpower | KingDavidNYC: I used to do a couple of hundred calls a week doing something similar |
| 17:33.17 | manxpower | But instead of using Originate, used .call files and instead of the non-PSTN leg going to a Wait(999), it went to VoicemailMain |
| 17:33.22 | drmessano | KingDavidNYC: Thats how all of your mass-dialers work in Asterisk.. and that was explained to you 3 different ways |
| 17:33.42 | KingDavidNYC | manxpower: I still fail to see how can the pstn dialing occur if no one has lifted the handset, but ok guys, I am gonna test it as you say |
| 17:34.02 | drmessano | ^^^^^ manxpower: Thats been the back and forth all morning |
| 17:34.57 | KingDavidNYC | maxpower: I wrote my own AMI handler, that I can do easily |
| 17:34.57 | drmessano | Asterisk originating a call doesnt need a handset lifted or an extension to be involved in the call until YOU tell it to |
| 17:35.05 | manxpower | KingDavidNYC: in Asterisk all calls have two legs. In your case you have a PSTN leg and the "other leg" The other leg can go anywhere you want it to. |
| 17:35.44 | KingDavidNYC | thank you guys, you are very kind |
| 17:35.54 | manxpower | KingDavidNYC: you need to read the Asterisk book |
| 17:36.08 | voipmonk | 4 times |
| 17:36.21 | psilikon | anyone suggest a place to host an * box in the US? |
| 17:36.23 | KingDavidNYC | :) |
| 17:36.40 | drmessano | psilikon: Not on the shelf in my home office.. |
| 17:36.41 | voipmonk | how many simultaneous calls, psilikon ? |
| 17:37.00 | psilikon | voipmonk, 12 - 15 |
| 17:37.34 | voipmonk | any plans to do more? ;) |
| 17:37.45 | manxpower | KingDavidNYC: NEVER think that an "extension" is a physical device. It's mot. |
| 17:37.48 | *** join/#asterisk hjorth (n=hjorth@linux3.kill.dk) |
| 17:37.48 | manxpower | And it's not too. |
| 17:38.02 | psilikon | voipmonk, maybe |
| 17:38.47 | manxpower | And if you keep thinking that an extension is a physical device then you'll constantly be designing things wrongly in your PBX |
| 17:39.03 | voipmonk | is it an image like vmeare xen or openvz? |
| 17:39.10 | voipmonk | vmware, rather |
| 17:39.45 | drmessano | An extension isn't even a device.. |
| 17:40.11 | KingDavidNYC | manxpower: in a few minutes I will have it |
| 17:40.51 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:41.11 | psilikon | Is connecting multiple phone in parallel to one "pstn" port on a SIP known to work well. Do they typically have the current t support a few? |
| 17:42.00 | *** join/#asterisk Alagar (n=Administ@122.164.33.151) |
| 17:42.13 | manxpower | psilikon: you can't connect multiple phones to a SIP connection. |
| 17:42.46 | manxpower | Heck, you can't even connect one analog phone to SIP. |
| 17:43.03 | manxpower | You connect analog phones to analog ports, not SIP "things" |
| 17:43.06 | drmessano | manxpower: Not even with a good staple gun? |
| 17:43.34 | outtolunc | hot glue ftw |
| 17:43.43 | psilikon | manxpower, isn't that what the cable company does when they bring a voip modem to your house. Don't they just connect the analog port to your NID and star out to your phones? |
| 17:43.45 | drmessano | Whats that line about trying to nail grape jelly to a wall? |
| 17:43.57 | p3nguin | If you have an ATA that supports enough REN on the analog port, you could connect a whole bunch of phones to it. |
| 17:44.04 | manxpower | psilikon: yes, but that has nothing to do with SIP or anything else. |
| 17:44.07 | drmessano | With an ATA |
| 17:44.38 | drmessano | manxpowers point was the lack of a entity to make the connection.. You cant plug an analog phone into SIP |
| 17:44.51 | drmessano | You cant plug an IP phone into SIP |
| 17:44.52 | manxpower | They just plug multiple phones into an analog port. The device with the analog ports could be running SIP, MGCP, or my great aunt Lulu's beef stew. |
| 17:45.00 | psilikon | So my phrasing might be off but you know what I am asking right? Connect multiple phones to one ATA. |
| 17:45.06 | drmessano | Yes you can |
| 17:45.16 | manxpower | psilikon: phrasing is important. |
| 17:45.44 | drmessano | You can do all sorts of crazy shit like have an extension ring an IP phone and an ATA with a cordless on it |
| 17:45.46 | manxpower | If you can't express what you want then you don't actually know what you want. |
| 17:46.46 | drmessano | or you can do like mchou and register 20 phones to one SIP account with your provider and have a giant SIP wirenut |
| 17:46.55 | manxpower | a SINGLE wrong letter in a Dial line can fully open up your PBX to hackers. |
| 17:47.24 | mchou | dude, a itsp that uses a decent sip proxy is no wirenut |
| 17:47.30 | mchou | drmessano: ^^^ |
| 17:48.17 | mchou | drmessano: it's just the smart thing to do |
| 17:48.34 | drmessano | mchou: You are SOOOOO hung up on this being about PROTOCOL and STABILITY.. I am talking about using an ITSP like a fucking PSTN line in your house with 20 phones connected to a pair of copper is STUPID |
| 17:48.48 | psilikon | Ok so I connect three phones to one ATA. Call comes in, rings all phones and is answered by employee number 1. Since my provider gives me 3 incoming channels; can 2 more calls come in and ring all on-hook phones to be answered by employees 2 and 3? |
| 17:48.49 | mchou | what?? |
| 17:49.00 | mchou | drmessano: it's not on a pair of copper |
| 17:49.08 | drmessano | mchou: Christ |
| 17:49.09 | Katty | does anyone have any rabbit recipes? |
| 17:49.19 | mchou | it's in many different geographic locations |
| 17:49.21 | manxpower | psilikon: are you still talking about 3 phones connected to one port on an ATA? |
| 17:49.28 | psilikon | manxpower, yes |
| 17:49.57 | drmessano | mchou: let me make this easier for you to understand.. Registering 20 phones to ONE SIP account is ****LIKE***** putting 20 phones on a pair of copper.. Whats the point? |
| 17:50.08 | drmessano | Sure, but its the SAME ACCOUNT |
| 17:50.09 | *** join/#asterisk mattwj2002 (n=matt@Wikisource/Mattwj2002) |
| 17:50.10 | mchou | no it's not |
| 17:50.12 | manxpower | psilikon: then you are a complete idiot and don't even understanding what "three phones bridged to one ATA port" even means. |
| 17:50.12 | drmessano | Same number, |
| 17:50.23 | drmessano | Limited to x number of concurrent calls |
| 17:50.27 | mattwj2002 | hi guys |
| 17:50.28 | mattwj2002 | :) |
| 17:52.28 | manxpower | three phones connected to the same port on an ATA is exactly like three phones connected to one PSTN line. |
| 17:52.28 | outtolunc | that was good timing <G> |
| 17:52.28 | mchou | drmessano: the point is simpler administration |
| 17:52.28 | drmessano | Its like putting your entire company on a single SKYPE account |
| 17:52.39 | mchou | no need to deal with a billion different sign-ons |
| 17:52.39 | drmessano | Its like putting your entire company on a single SKYPE account |
| 17:52.39 | drmessano | Whats the point? |
| 17:52.39 | mattwj2002 | are you guys talking about skype asterisk plugin? |
| 17:52.39 | drmessano | No |
| 17:52.39 | mattwj2002 | oh |
| 17:52.39 | p3nguin | katty: You can cook it just like pan-fried chicken. |
| 17:52.39 | psilikon | manxpower, what about the 3 incoming channels bit. |
| 17:52.39 | manxpower | psilikon: then you need three analog ports |
| 17:52.39 | mchou | drmessano: no, it's allowing your employees to have more than one endpoint w/o dealing with a zillion credentials |
| 17:52.47 | Katty | p3nguin: i know how to cook it. i want a recipe. |
| 17:52.47 | Katty | p3nguin: do you have one? |
| 17:53.10 | drmessano | mchou: You're failing to see how this is creating ONE extension for those 20 users and no different than running copper all over your office |
| 17:53.29 | mchou | yeah, except there's no copper involved |
| 17:53.31 | drmessano | mchou: Have you ever set up a PBX before? |
| 17:53.33 | manxpower | drmessano: you'll never convince mchou any differently. |
| 17:53.46 | manxpower | mchou: Obviously knows how SIP works, not how Asterisk does SIP. |
| 17:53.54 | mchou | lol |
| 17:54.11 | drmessano | manxpower: the problem is he doesnt know how PEOPLE work.. or how a business works |
| 17:54.11 | mchou | manxpower: that's my point, actually |
| 17:54.16 | drmessano | manxpower: He sounds like he's set up for home and thats it |
| 17:54.29 | manxpower | Asterisk does not allow multiple devices to register to the same account. It's as simple as that. All the discussion in the world is not going to make Asterisk work any differently from that. |
| 17:54.33 | mchou | just becuse asterisk has limitations with sip doesnt mean rest of world has to suffer |
| 17:54.40 | mchou | manxpower: ^^^ |
| 17:54.45 | psilikon | manxpower, I want the 3 phones to behave as if they were connected to the PSTN but I want to be able to use the other two channels to receive calls even if one is in use. |
| 17:54.49 | manxpower | Personally I like the way Asterisk does this. |
| 17:54.54 | drmessano | manxpower: Complete lack of understand of the needs of a business beyond 3 employees.. because thats as far as you could really go with a system like that |
| 17:55.19 | manxpower | psilikon: how would you do that if you WERE only using PSTN? |
| 17:55.23 | hardwire | psilikon: sounds like you want a hunt group from your telco. |
| 17:55.26 | hardwire | and no pbx |
| 17:55.30 | mchou | manxpower: Schulzrinnes begs to differ with you, and I'd him him more credibility than you |
| 17:55.31 | p3nguin | katty: When I said that you can cook it like chicken, I didn't mean the actual cooking part, I meant use your favorite chicken recipe. |
| 17:55.42 | mchou | s/him/give |
| 17:56.03 | drmessano | mchou: Asterisks limitation is a good thing, IMO.. It keeps people from doing something stupid like your SIP wirenut and actually creating a real phone system |
| 17:56.42 | mchou | I fail to see how having more credentials reduces workload for admin |
| 17:56.45 | psilikon | manxpower, I want it to be like a PSTN connection where all phones ring for an incoming BUT not like a PSTN in regards to being able to receive more calls when one phone is in use. |
| 17:56.52 | hjorth | anybody got any secret stuff laying around that i cant find on google, with asterisk implemantation against cisco media gateway (Or just someone who has done this)? I am having quite a hazzle with T.38... |
| 17:57.13 | mchou | drmessano: both for ITSP or end user |
| 17:57.21 | manxpower | psilikon: I'm not interested in helping you further. |
| 17:57.29 | drmessano | mchou: I fail to see how using one SIP account on a provider for 20 office phones is anything other than running copper all over the place.. Its no different, and exactly why you have a PBX |
| 17:57.29 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 17:57.29 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM] |
| 17:57.29 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 17:57.50 | drmessano | mchou: Nobody said it made admin easier, so stop trolling |
| 17:57.55 | hardwire | mchou: a variable per registration (realdevice=00500x82) and a 8-10 line macro can fix that btw. |
| 17:58.24 | *** join/#asterisk cnu (i=cnu@113.80-203-44.nextgentel.com) |
| 17:58.26 | mattwj2002 | with skype for asterisk is their a demo version available by chance? |
| 17:58.27 | hardwire | or even using astdb to determine the realdevice.. or updating extensions.conf with that info as well. |
| 17:58.32 | mattwj2002 | 30 days or anytime |
| 17:58.45 | drmessano | mattwj2002: You asked this the other day, no |
| 17:58.45 | manxpower | mattwj2002: nobody here uses Skype for Asterisk |
| 17:58.46 | mchou | hardwire: dude, 10 line macro? you must be joking. how about 0 line macro with a real sip proxy? |
| 17:58.58 | psilikon | manxpower, thats cool, thanks for the help tho |
| 17:59.15 | hardwire | mchou: sounds like you have a solution that works for *YOU* |
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| 17:59.55 | mattwj2002 | manxpower so it is not well supported? |
| 17:59.57 | drmessano | hardwire: 20 extensions using the SAME SIP credentials isn't a PBX.. it's a copper line with 20 walmart phones attached.. he refuses to see that |
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| 18:00.06 | hardwire | drmessano: I know |
| 18:00.07 | drmessano | mattwj2002: I use it |
| 18:00.12 | drmessano | mattwj2002: It not hard to set up |
| 18:00.23 | hardwire | PBX = predictable results in most cases |
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| 18:00.51 | mattwj2002 | drmessano how is the sound quality? |
| 18:00.51 | drmessano | mattwj2002: As I told you, you dont need a demo.. you either need it and spend the money or your dont.. Same as why there is no demo for G729 |
| 18:00.51 | hardwire | -> coffee hut where the cute people live. |
| 18:00.51 | drmessano | mattwj2002: Same as anything else |
| 18:00.52 | manxpower | mattwj2002: as far as I can tell it's just that nobody cares about Skype for Asterisk |
| 18:00.55 | mchou | so far no one has said _why_ concurrent registrations (or the old fashion idea of multiple extensions) is a bad idea |
| 18:01.04 | drmessano | mchou: Actually, I have |
| 18:01.07 | psilikon | manxpower, I know exactly how I would handle this with an * box but was wondering if an ITSP could handle that for me. |
| 18:01.21 | mchou | where I have given several reasons why it's actually a good idea |
| 18:01.26 | manxpower | psilikon: contact an ITSP and ask them. |
| 18:01.27 | carrar | I want more candy |
| 18:01.35 | mattwj2002 | okay |
| 18:01.54 | hardwire | mchou: there's a lot of reasons including scalability, licensing problems, redundant framework, etc.. |
| 18:01.55 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:02.12 | psilikon | manxpower, i'll have to do that tomorrow. Thanks for the help man! |
| 18:02.14 | carrar | man |
| 18:02.17 | mchou | drmessano: then tell me why the (old) idea of having multiple extension for one individual person is so piss poor |
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| 18:02.23 | carrar | everyone split out |
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| 18:02.38 | carrar | doh |
| 18:02.41 | p3nguin | psilikon: I can't see how you'll ever get more than one call at a time using a single ATA and multiple phones on the analog port. |
| 18:02.41 | mchou | hardwire: bullcrap |
| 18:02.47 | drmessano | mchou: 1 out of 1000 offices would find having all their phones tied to one SIP registration, which means one VM box, no device <> device calling, etc, useful.. Everyone else on the planet would not |
| 18:02.56 | mchou | hardwire: you're telling sip proxies arent scalable? |
| 18:03.01 | carrar | YAHTZEE |
| 18:03.03 | hardwire | mchou: feel free to tune in. |
| 18:03.05 | mchou | hardwire: stop your bs |
| 18:03.26 | drmessano | mchou: Go into a workplace other than McDonalds and spend 10 minutes watching phone use.. No one wants a party line |
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| 18:03.45 | voipmonk | wow |
| 18:03.45 | hardwire | is he even talking about a party line? |
| 18:03.47 | mchou | dude, parallel forking is not a party line |
| 18:03.51 | p3nguin | psilikon: Now, on the other hand, you can get one call on each ATA at a time, and you can set up a sub account on your ITSP for each ATA if you want. |
| 18:04.03 | hardwire | I thought he was talking about outbound groups |
| 18:04.08 | carrar | has arrived for his Sunday entertainment |
| 18:04.11 | psilikon | p3nguin, yeah I am coming to that realization. I'm am trying to get off the PSTN to save some money... then I will have to spend all the saving implementing a pbx based solution to do what I want. |
| 18:04.24 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
| 18:04.27 | mchou | psilikon: PRECISELY |
| 18:04.34 | drmessano | hardwire: He's talking about registering 20 phones to ONE SIP ACCOUNT at say voip.ms |
| 18:04.35 | mchou | psilikon: admin headaches |
| 18:04.37 | mchou | avoid |
| 18:04.38 | drmessano | ONE NUMBER |
| 18:04.43 | carrar | haha |
| 18:04.46 | carrar | DO IT |
| 18:04.49 | drmessano | ONE account |
| 18:04.52 | hardwire | drmessano: sure.. I see whhhyyy he wants to do it. |
| 18:04.58 | mchou | I never said 20 |
| 18:05.03 | p3nguin | psilikon: I've got more tied up in my single ATA than any other single device in my entire system. |
| 18:05.04 | carrar | 21 |
| 18:05.09 | mchou | maybe 3 or 4 per individual |
| 18:05.14 | drmessano | heh |
| 18:05.17 | mchou | drmessano: ^^^ |
| 18:05.21 | carrar | I made my ATA's out of sound cards |
| 18:05.37 | drmessano | So now it's multiple devices per multiple accounts |
| 18:05.39 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 18:05.42 | carrar | and soldering wire |
| 18:05.44 | manxpower | How will you bill for this extra service of allowing more than device? |
| 18:05.47 | drmessano | I hear a PBX can do that |
| 18:06.12 | hardwire | mchou: no offense but there is a certain abstraction level you're omitting. |
| 18:06.24 | mchou | drmessano: there is clearly a valid use case for muliple extensions per individual |
| 18:06.25 | drmessano | Do they get billed for calling each other |
| 18:06.30 | drmessano | ? |
| 18:06.39 | carrar | A PBXATA |
| 18:06.46 | manxpower | mchou: you mean multiple devices, not multiple extensions, don't you? |
| 18:06.46 | hardwire | the PBX is responsible for the dialplan no matter if it's "special" with the channel driver. |
| 18:07.10 | hardwire | the sip proxies may allow you to handle it.. but they are doing so as the dialplan provider understanding their own sip channel. |
| 18:07.18 | mchou | manxpower: "extensions" in the old sense (like multiple phone in your house) |
| 18:07.20 | hardwire | it's the same thing.. just implemented differently. |
| 18:07.44 | psilikon | I *want* to go the PBX route, that would be fun as hell but I don't know if the franchise owner wants the ancillary costs nor If I'll have the time with my full-time job and part-time classes :( |
| 18:07.47 | manxpower | mchou: do unlike anything asterisk people would be calling an extension |
| 18:07.47 | drmessano | [13:08] <mchou> manxpower: "extensions" in the old sense (like multiple phone in your house) <-- this is why home users need to get Skype and stay away from #Asterisk |
| 18:07.59 | drmessano | Lets talk about PBX's |
| 18:08.02 | mchou | manxpower: endpoints or devices if you prefer |
| 18:08.17 | drmessano | Not how to maximize your Skype experience with multiple phones from Wal Mart |
| 18:08.19 | hardwire | mchou: for what it's worth.. only one registration needs to be made, multiple devices can call through the proxy using those credentials if it's configured correctly for that. |
| 18:08.22 | hardwire | which is by default. |
| 18:08.31 | mchou | drmessano: no, except tall the extensions arent in your house |
| 18:08.35 | hardwire | the device making the registration just needs to point the inbound to the correct phones. |
| 18:08.37 | *** join/#asterisk tlir (n=tlir@77.126.35.195) |
| 18:08.41 | mchou | s/tall/all |
| 18:08.56 | drmessano | mchou: No, they wouldnt be in my house.. I actually have a PBX at home and skip this mickey mouse crap |
| 18:09.05 | mchou | hahah |
| 18:09.06 | hardwire | mchou: have you tried not registering any of the phones and seeing if they can make outbound calls? |
| 18:09.24 | mchou | if you used a real itsp you wouldnt even bother with PBS |
| 18:09.26 | drmessano | Why not just use Yahoo Messenger |
| 18:09.27 | mchou | PBX |
| 18:09.30 | manxpower | of course the phones can make outbound calls if they are not registered. |
| 18:09.32 | mchou | that's the point |
| 18:09.43 | manxpower | Everyone should know that. |
| 18:09.45 | hardwire | mchou: I thought you had a lot of respectable input in this channel.. weird.. |
| 18:09.50 | hardwire | goes off for his coffee. |
| 18:09.58 | mchou | or specifically asterisk, because of it's limitation |
| 18:10.05 | p3nguin | psilikon: The savings may not be seen instantly, but think about the long-term. I haven't spent more than $2 in phone calls this month. That's a $33 savings over having a traditional telco. |
| 18:10.15 | drmessano | mchou: Really, so a real ITSP is gonna allow me to pick up my phone and dial 3 digits to ring the phone at work, 5 of my friends houses, my parents house, etc? Because they're all on my PBX here |
| 18:10.28 | carrar | I register all my SIP Phones to AT&T |
| 18:10.30 | drmessano | mchou: PLEASE TELL ME HOW |
| 18:10.39 | drmessano | mchou: I am ready to believe! |
| 18:10.41 | carrar | heh |
| 18:10.49 | mchou | drmessano: yeah, that's called free in network calling :) |
| 18:11.08 | drmessano | mchou: 3 digit extensions? |
| 18:11.13 | drmessano | mchou: Sweet, how? |
| 18:11.16 | p3nguin | hahaha |
| 18:11.28 | carrar | 3 digits!! |
| 18:11.30 | carrar | Thats crazy |
| 18:11.41 | drmessano | 7 digit extensions for the win |
| 18:11.54 | carrar | IMPOSSIBLE!!! |
| 18:12.07 | mchou | nothing impossible about it |
| 18:12.15 | drmessano | So how do I do it? |
| 18:12.19 | drmessano | Give me a provider |
| 18:12.21 | KingDavidNYC | manxpower: man, I fail to see how I can join the voip line to the pstn line during the originate call |
| 18:12.23 | mchou | that's what aliases in sip proxies are for |
| 18:12.24 | *** join/#asterisk dymaxion (n=dymaxion@202.82.134.130) |
| 18:12.26 | carrar | I force everyone to dial 20 digits |
| 18:12.40 | p3nguin | 20? |
| 18:12.46 | carrar | sometimes 30 |
| 18:12.47 | p3nguin | That's all? |
| 18:12.52 | carrar | no |
| 18:12.53 | p3nguin | That's more like it. |
| 18:12.55 | drmessano | mchou: I want an example of one I can set up TODAY for that |
| 18:13.07 | drmessano | mchou: I will dump asterisk and switch this afternoon |
| 18:13.14 | mchou | drmessano: I know for sure ekiga.net will let you do it. (5 digits though) |
| 18:13.28 | drmessano | No, 5 digits are for suckfaces.. I want THREE |
| 18:13.38 | carrar | 2 |
| 18:13.39 | drmessano | I think I said that 3 or 11 times |
| 18:13.39 | carrar | 1 |
| 18:13.47 | hardwire | drmessano: chances are he is registering 3-4 times to an outbound provider and changing his inbound endpoint each time without knowing it. |
| 18:14.01 | p3nguin | drmessano: If you use VoIP.ms and subaccounts, you can do it, but it won't give you free calling between them. |
| 18:14.04 | drmessano | hardwire: Nonsense, mchou would know better |
| 18:14.06 | hardwire | which is totally legit if you don't mind getting calls to completely random phones. |
| 18:14.10 | mchou | hardwire: you have zip idea what you're talking about |
| 18:14.15 | hardwire | mchou: YOUR MOM! |
| 18:14.19 | mchou | hardwire: look up the SIP RFC |
| 18:14.34 | drmessano | hardwire: mchou would never make a baseless troll like the one that got him banned last night |
| 18:14.38 | hardwire | mchou: which part? |
| 18:14.42 | psilikon | p3nguin, I hear ya. I was just looking at hosted asterisk plans, they want some $$$ monthly. If I host my own my concern is then getting the bandwith to handle 15-20 incoming calls |
| 18:14.45 | KingDavidNYC | maxpower: I use the AMI originate, then the dialplan answers the the first leg on the voip line, but then I dont know how to transfer the leg to the pstn line, so that the outgoing leg is dialed on the pstn line |
| 18:14.52 | mchou | hardwire: "forking" |
| 18:15.35 | carrar | psilikon, you still need bandwidth for 20 calls out of the same building |
| 18:15.40 | hardwire | mchou: got a TOC index number? |
| 18:15.47 | carrar | regardless where the PBX is |
| 18:15.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:16.19 | hardwire | regardless of what the SIP RFC says.. its not a "requirement" |
| 18:16.27 | p3nguin | psilikon: You're going to need something with moderately high speed at every location. |
| 18:16.49 | psilikon | carrar, 4 separate buildings. |
| 18:16.53 | carrar | ah |
| 18:16.57 | carrar | use LASERS |
| 18:16.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.03 | mchou | hardwire: lol |
| 18:17.10 | mchou | hardwire: nice dodge |
| 18:17.18 | KingDavidNYC | drmessano? |
| 18:17.31 | mchou | http://www.faqs.org/rfcs/rfc3428.html, section 6. Proxy Processing |
| 18:17.32 | hardwire | mchou: I'm not dodging anything |
| 18:17.43 | psilikon | I have business class RR at each location. I am thinking I need colo space OR another cable line at my office to host this pbx and need atleast 2mpbs up |
| 18:17.48 | hardwire | it's just plain not a requirement to do parallel forking |
| 18:17.59 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 18:18.17 | carrar | hardwire, you could use g.729 |
| 18:18.17 | hardwire | I also don't think it's a good idea for scalability reasons, billing, dialplan prediction... |
| 18:18.18 | psilikon | 2mbps up |
| 18:18.20 | p3nguin | psilikon: If you already have cable internet at each place, you're ready to set up * at your main office. |
| 18:18.25 | carrar | don't need 2megs for that |
| 18:18.32 | carrar | err psilikon I mean |
| 18:18.52 | outtolunc | KingDavidNYC: think in reverse, have the originate call the pstn line, if connects, dial the context,exten,priority in the dialplan (and answer) |
| 18:18.56 | psilikon | 85k per incoming SIP right? |
| 18:18.56 | mchou | hardwire: yeah, why didnt YOU write the SIP RFC? lol |
| 18:19.07 | carrar | psilikon, at G.711 |
| 18:19.10 | mchou | hardwire: scalability my ass |
| 18:19.10 | hardwire | mchou: this has nothing to do with SIP |
| 18:19.25 | p3nguin | psilikon: Change to GSM and the requirement should decrease. |
| 18:19.34 | mchou | hardwire: tell me again how a sip proxy is less scalable than asterisk |
| 18:19.46 | HaMF | Hi, can anyone tell me what the "# termtype: te" in the zaptel.conf means? (genzaptelconf on a TE110P card created this comment) |
| 18:20.02 | hardwire | mchou: I never said that? |
| 18:20.17 | mchou | hardwire: [10:18:18] <hardwire> I also don't think it's a good idea for scalability reasons, billing, dialplan prediction... |
| 18:20.20 | carrar | psilikon, 20 calls at g.729 is about 624kbps |
| 18:20.24 | mchou | wtf??? |
| 18:20.29 | mchou | hardwire: ^^^ |
| 18:20.40 | hardwire | I also don't think it's a good idea *FOR ASTERISK TO IMPLEMENT* for scalability reasons, billing, dialplan prediction... |
| 18:20.42 | KingDavidNYC | outotlunch: if the originate calls the pstn line first, then the pstn line will ring, I dont wnat that |
| 18:20.48 | carrar | psilikon, http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 18:20.53 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 18:21.06 | mchou | hardwire: especially considering ITSPs do all of the stuff you mentioned |
| 18:21.14 | outtolunc | then just have the context,exten,priority do an actual Dial(yadda,,,,,) |
| 18:21.25 | outtolunc | yadda=pstn group |
| 18:21.33 | psilikon | carrar, ooooh thanks. from Cisco even ;) |
| 18:21.35 | drmessano | outtolunc: This is the 5th time we explained this |
| 18:21.43 | carrar | heh |
| 18:21.48 | mchou | hardwire: ok, so you're saying asterisk has limitations. what else is new? |
| 18:21.50 | KingDavidNYC | outtolunch: ok |
| 18:22.01 | hardwire | drmessano: catch me up later... I don't really need this sort of functionality but some of my clients might and if he can show us the light I suppose it's worth looking into. |
| 18:22.39 | drmessano | hardwire: Do what? |
| 18:22.49 | mchou | hardwire: now you're catching on :) |
| 18:23.11 | drmessano | hardwire: I think this is a horrible idea, and I eschew and chew it |
| 18:23.12 | p3nguin | psilikon: What's your current rate on your cable, 5 down / 1 up? |
| 18:23.15 | hardwire | mchou: actually.. I'm just trying to be non-objective.. give it a try. |
| 18:23.19 | hardwire | bbl |
| 18:24.22 | *** join/#asterisk QaDeS (n=mklaus@p4FC72C8C.dip0.t-ipconnect.de) |
| 18:24.25 | drmessano | mchou giving credibility to the others point of view? he told one of the developers they didnt know the difference between a connection and a call |
| 18:24.44 | drmessano | So apparently he knows more than them too.. The rest of us don't have a chance :) |
| 18:25.03 | mchou | drmessano: It was clear in the context |
| 18:25.16 | *** join/#asterisk Nivex (i=ac1f0304@user-0c2h3cu.cable.mindspring.com) |
| 18:25.25 | mchou | selective quoting wont prove anything |
| 18:25.57 | carrar | Y*A*W*N Next topic!! |
| 18:26.01 | voipmonk | ... |
| 18:26.16 | carrar | BBQ for lunch? |
| 18:27.07 | psilikon | p3nguin, 10 down 1 up @ $55/month |
| 18:27.20 | carrar | plenty |
| 18:27.23 | carrar | for voice |
| 18:27.53 | p3nguin | psilikon: 1 Mbps outbound should be plenty for all the calls you can possibly make in a small office. |
| 18:27.56 | carrar | Though running your voice on the same with data might cause some issues |
| 18:28.05 | voipmonk | heheh |
| 18:28.12 | p3nguin | Depending on how much data usage you have |
| 18:28.19 | voipmonk | you sound like bill gates, several years ago |
| 18:28.27 | p3nguin | ;) |
| 18:28.44 | drmessano | mchou: It really doesn't matter, just builds a behavioral pattern.. I think carrar is a moron, but if I can't prove he's wrong about something, and he can prove I am the idiot, I am probably the dense one when I continue to argue |
| 18:28.59 | drmessano | No offense, carrar.. It's just business |
| 18:29.07 | carrar | WTF#$&)(*$&!@# |
| 18:29.09 | drmessano | Don't take it personally |
| 18:29.13 | carrar | YOU"RE GOING DOWN MR!! |
| 18:29.18 | drmessano | ha |
| 18:29.44 | carrar | Been on IRC since 89, not the first time someone has called me a name |
| 18:29.47 | psilikon | p3nguin, duh... oh yeah my incoming calls are on my downlink. My outgoing which will be very little is using my uplink 1mpbs crap... so 10mbps IS plenty :) |
| 18:30.50 | p3nguin | psilikon: Now the bottleneck could come on the upstream side of your main office if that's where you put your * box. |
| 18:30.51 | carrar | after having several servers on EFNET, nothing effects me |
| 18:30.58 | carrar | it's all good! |
| 18:31.02 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 18:31.19 | drmessano | carrar: I have been on IRC since they called it "emacs: network edition".. so yeah, not new to me, either |
| 18:31.31 | p3nguin | psilikon: But at this point, I would try it and see how your usage looks. |
| 18:32.01 | carrar | h4X0r |
| 18:32.30 | drmessano | The other thing about using Asterisk is that it's a big funnel.. You can run a bunch of G722 phones on it and use GSM to your provider and your internets all look cool |
| 18:32.43 | carrar | I make money with Asterisk, call me a moron all you want :) |
| 18:32.52 | drmessano | :( |
| 18:32.53 | carrar | heh |
| 18:33.23 | drmessano | That reminds me.. I need to get an SPA3102 ready for the new office |
| 18:33.24 | drmessano | heh |
| 18:33.30 | mchou | and their lies the rub |
| 18:33.40 | drmessano | "there" |
| 18:33.43 | mchou | it's all about the $$ |
| 18:33.46 | carrar | drmessano, use soundcards!! |
| 18:34.03 | carrar | where is that guy anyways |
| 18:34.36 | drmessano | carrar: Going back to my old job.. need to get my SPA3102 back on the wall in the telco closet connected to the copper for my office line |
| 18:34.59 | drmessano | Google voice has nothing on an SPA3102 regged back to my home PBX |
| 18:35.10 | drmessano | I get teh calls everyware |
| 18:35.11 | Nivex | ~phones |
| 18:35.12 | infobot | 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. |
| 18:35.51 | voipmonk | google voice gives me buzzing |
| 18:35.58 | carrar | get on the wave! |
| 18:36.07 | drmessano | carrar: Do you have invites? |
| 18:36.20 | p3nguin | Google voice doesn't let me do oubound calling from asterisk anymore. |
| 18:36.30 | dlynes | eh? aastra 480i is ranked better than aastra 5i series? and cisco's ranked even lower? |
| 18:36.34 | drmessano | p3nguin: give it a few weeks |
| 18:36.58 | p3nguin | Even using Gizmo, I can't get calls to go out. |
| 18:36.58 | outtolunc | has 1 invite left |
| 18:37.06 | mchou | p3nguin: still works here |
| 18:37.12 | drmessano | p3nguin: Google bought Gizmo5.. Expect them to fix up the interconnect and make that the SIP registrar leg of GV |
| 18:37.18 | carrar | drmessano, no, I don't use it |
| 18:38.04 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 18:39.03 | drmessano | Damn, they suspended new user signups |
| 18:39.49 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:40.13 | voipmonk | I don't like my conversations indexed and searched by google |
| 18:40.35 | drmessano | Except they dont? |
| 18:40.59 | carrar | There was a funny googl wave logo pic someone sent |
| 18:41.03 | carrar | trying to find it |
| 18:42.00 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 18:42.07 | joako | why can't verbosity be set above 2147483647? |
| 18:42.25 | carrar | that induces time warps |
| 18:42.59 | drmessano | http://en.wikipedia.org/wiki/2147483647 |
| 18:43.17 | drmessano | http://en.wikipedia.org/wiki/December_12,_2012 |
| 18:43.23 | drmessano | Coincidence? I think not |
| 18:43.25 | Nivex | what is the difference between the polycom IP 320 and 321 ? |
| 18:43.32 | drmessano | Nivex: 1 |
| 18:43.48 | p3nguin | happiness |
| 18:44.10 | carrar | haha |
| 18:44.11 | carrar | here it is |
| 18:44.12 | carrar | http://i.imgur.com/zVn91.jpg |
| 18:44.12 | psilikon | p3nguin, I need to see what type of cpu I need to do that too |
| 18:44.29 | p3nguin | psilikon: Got any old computers lying around? |
| 18:45.01 | psilikon | p3nguin, I've got one p4 2Ghz with 1 gig ddr and one p4 2.8 with 2 gig ddr2. |
| 18:45.02 | Nivex | drmessano: surely that 1 means something or there wouldn't be different models |
| 18:45.53 | p3nguin | psilikon: I would expect either system to be fine for you. I run a PIII 933MHz with 512M non-ddr. :) |
| 18:46.31 | p3nguin | psilikon: If you're worried, go with the faster system. |
| 18:46.35 | psilikon | p3nguin, i've got a p3 733 with 512mb sdram but it only handles 2 calls. How many are you trying to handle? |
| 18:46.50 | p3nguin | psilikon: Mine handles far more than two. |
| 18:47.02 | Nivex | found it: more internal memory |
| 18:47.11 | mchou | psilikon: P4 is gonna cost you more in electricity than you gonna save with going voip |
| 18:47.46 | mchou | an average P4 system draws 145W from the wall |
| 18:48.19 | psilikon | mchou, I thought I read that the p3 was in fact less power hungry than the p4 but my boss swears it is the other way around. |
| 18:48.19 | mchou | go with P3 or lower |
| 18:48.44 | mchou | measure it and prove your boss wrong |
| 18:49.06 | drmessano | Here we go |
| 18:49.15 | mchou | lol |
| 18:49.26 | drmessano | mchou: Not everyone is a single line home user |
| 18:49.53 | drmessano | mchou: We dont care that your eMachine didn't make your calls free-er |
| 18:50.07 | mchou | drmessano: and your point is? P3 wont support more than 1 line? |
| 18:50.19 | drmessano | mchou: No, your theories don't support more than 1 line |
| 18:50.54 | drmessano | mchou: You don't scale well |
| 18:51.25 | mchou | that's cause I dont have P4 in my brain |
| 18:51.44 | mchou | you know, hyperthreading and all that jazz |
| 18:52.26 | hardwire | sheesh its 1F out |
| 18:52.30 | drmessano | mchou: if I have 12 analog lines and I switch to voIP, you're telling me the cost of the power for the PC offsets my savings? |
| 18:52.50 | hardwire | only if it's a p3 |
| 18:52.55 | mchou | drmessano: heh |
| 18:52.57 | drmessano | hardwire: Damn |
| 18:53.08 | mchou | hardwire: or lower :) |
| 18:53.13 | hardwire | I use via c3 |
| 18:53.26 | hardwire | but not because I'm cheap |
| 18:53.27 | mchou | hardwire: there you go |
| 18:53.40 | drmessano | I have all my phones registered to one Skype account |
| 18:53.47 | hardwire | because I'm a self-righteous asshole. |
| 18:53.54 | drmessano | Sure, I have to IM my Dad so I can use the phone.. So what |
| 18:54.14 | carrar | heh |
| 18:54.25 | joako | drmessano: So If I run a 64-bit system I can set verbose higher? |
| 18:54.29 | drmessano | I only have to remember one username and password, and I saved 11 cents last month |
| 18:54.48 | hardwire | joako: like at a post 32 bit level? |
| 18:55.02 | drmessano | joako: It's open source.. sure you could.. |
| 18:55.07 | elliot98 | the changelog says the dial application to tell the ip phones not to count a call as missed if timeout |
| 18:55.11 | elliot98 | what option is this? |
| 18:55.21 | hardwire | you want verbose to be 2**48? |
| 18:55.35 | mchou | no mersenne prime |
| 18:55.51 | mchou | 2^48-1 |
| 18:56.12 | p3nguin | Verbose level has the same max on my 64-bit system as it does on a 32-bit system. |
| 18:56.20 | mattwj2002 | :( |
| 18:56.21 | hardwire | mchou: I did a lot of thinking on the drive to the hotties hot stop coffee shack.. |
| 18:56.37 | hardwire | and I decided a lot of things that aren't worth saying until somebody else other than you cares about it. |
| 18:56.48 | hardwire | I hope you're not offended.. but I think that's the right approach. |
| 18:56.48 | mchou | hardwire: you bought a hottie home? |
| 18:56.53 | mattwj2002 | Skype for Asterisk doesn't work with unlimited plans |
| 18:56.54 | mattwj2002 | :( |
| 18:56.59 | hardwire | mchou: no.. but they are really cute at the coffee shack |
| 18:57.48 | mchou | hardwire: I'm not offended. Just realize ITSP use sip proxies for many reasons (including forking) |
| 18:58.14 | hardwire | Many reasons indeed, there is always justification for one thing or the other. |
| 18:58.24 | mchou | hardwire: which is to say plenty of people have given it more thought than you and I have and IMPLEMENTED it |
| 18:58.41 | hardwire | I've put around an hour of thought into it. |
| 18:59.04 | hardwire | I dunno how much you've thought about it.. but you certainly argued about it. |
| 18:59.05 | mchou | In fact I think you'll be hard pressed to find ITSPs that have not implemented sip proxies |
| 18:59.41 | mchou | like I said, no skin off my back |
| 18:59.45 | hardwire | mchou: yup, I find they do it mostly for security, logging, and pre-processing. |
| 19:00.01 | hardwire | one of my itsp uses openser coupled with sippy (which they use for logging sip traces) |
| 19:00.19 | hardwire | I may end up using it to handle inbound account tagging using sip headers |
| 19:00.48 | mchou | hardwire: yup, openser supports forking |
| 19:00.53 | hardwire | but each time I think about putting a sip proxy on the edge I find I'm loosing things I'm familiar with. |
| 19:01.11 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 19:01.15 | hardwire | like, our SBC will no doubt be asterisk since I need a multi-media gateway more than just SIP |
| 19:01.22 | mchou | and that's why you hang out here on #asterisk |
| 19:01.24 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 19:01.48 | hardwire | But I may end up using opensips for registration once I get large enough to need that sort of thing. |
| 19:01.56 | mchou | water finds it's own level |
| 19:02.05 | joako | I can't get Asterisk 1.6.1.9 GUI to work. I don't get anything from manager debug.... |
| 19:03.10 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 19:03.27 | hardwire | mchou: but I realized how many dialplan decisions would take place in the sip driver when you mentioned multiple registrations.. and how so many functions would need fixups to implement it. |
| 19:03.47 | mchou | what? |
| 19:04.08 | hardwire | Well, take channel availability |
| 19:04.09 | mchou | it has nothing to do with your dialplan |
| 19:04.13 | hardwire | and channel state |
| 19:04.21 | hardwire | and fair distribution |
| 19:04.30 | hardwire | and other things that business pbx's need above and beyond wholesale providers |
| 19:04.39 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 19:04.54 | hardwire | not to mention the problems with when each sip registration goes to a sip endpoint that automatically answers |
| 19:05.13 | hardwire | whoever has the shortest network path will always win |
| 19:05.40 | hardwire | I can see why multiple registrations per user was left out until it had to be dealt with. |
| 19:05.48 | hardwire | so.. does it need to be dealt with? |
| 19:05.58 | mchou | I didnt say it wasnt complex |
| 19:06.12 | mchou | for devs |
| 19:06.27 | hardwire | I've had my share of complains in this channel aimed at the devs |
| 19:06.41 | mchou | but for admins and users there's definitely valid use cases |
| 19:06.41 | hardwire | and my answer was simple.. and I expect one day to give the same answer on my own open projects |
| 19:06.48 | hardwire | Tits or get the fuck out. |
| 19:07.35 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 19:07.38 | mchou | considering all the open source sip prxies that I know of support forking, I'd say you're mistaken |
| 19:07.46 | mchou | proxies* |
| 19:07.50 | hardwire | sip proxies are for sip, rabbit. |
| 19:07.58 | hardwire | asterisk pbx is for muuuuuuuch more. |
| 19:08.09 | hardwire | and it's goals aren't the same |
| 19:08.14 | p3nguin | I have an IAX2 trunk, so how does a SIP proxy help me? |
| 19:08.36 | mchou | I didn't imply asterisk and sip supported the same goals |
| 19:08.37 | hardwire | I have a hard time understanding why you aren't registering once to your ITSP and distributing the inbound/outbound calls using your own very cheap and RTP free open sip proxy? |
| 19:08.41 | p3nguin | And furthermore, I know the difference between "it's" and "its." |
| 19:08.53 | mchou | hardwire: nor are they necessarily mutually exclusive |
| 19:08.54 | hardwire | unless it's so you don't need an extra machine running. |
| 19:09.05 | hardwire | p3nguin: we had this discussion last week didn't we? |
| 19:09.12 | p3nguin | perhaps |
| 19:09.31 | hardwire | I have a hard time with my '.. sue me. |
| 19:09.41 | p3nguin | sues |
| 19:09.49 | hardwire | removes his ' key. |
| 19:09.56 | hardwire | mails it to p3nguin as payment |
| 19:10.12 | p3nguin | Don't feel bad -- you aren't the only one who is having trouble with that key. |
| 19:10.34 | hardwire | p3nguin: Don't feel too special there bronco. :) |
| 19:12.03 | mchou | hardwire: why would I even want or need to do that? |
| 19:12.23 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 19:12.32 | hardwire | because you are responsible for distribution of your calls above all else. At least in the context of this channel. |
| 19:13.11 | joako | Anyone knows how I can debug the asterisk GUI? |
| 19:13.23 | hardwire | if your itsp supports parallel invite, multiple dial, or other PBX related features then it's a definate bonus to use them. |
| 19:13.35 | hardwire | But I don't see where your problem is if you already have what you want. |
| 19:14.08 | mchou | hardwire: If the admin can be offloaded "upstream" I'd much rather do that. since upstream there is a PBX already |
| 19:14.34 | hardwire | would you excuse me while I headdesk? |
| 19:14.42 | hardwire | *HEADDESK* |
| 19:15.01 | hardwire | moving on.. I need to find some django peeps to run things off of.. bbl |
| 19:15.03 | p3nguin | lol |
| 19:15.37 | p3nguin | I can't believe this debate has lasted for this long. |
| 19:15.52 | ChannelZ | darn, and I just woke up. |
| 19:16.15 | hardwire | It's not a Sunday without some sort of *bating. |
| 19:17.06 | mchou | p3nguin: considering you didnt even know what the concept of parallel forking was until I asked you about it I'd say you are a bit unqualified to judge |
| 19:17.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:18.00 | hardwire | It doesn't take qualification to be bored. |
| 19:18.28 | mchou | right, the province of small minds |
| 19:19.40 | hardwire | what a prick... |
| 19:20.47 | mchou | hardwire: I wasnt referring to you |
| 19:20.55 | hardwire | I can't quite figure out if you were insulted that p3nguin had input at all or that p3nguin mentioned how long something associated with you had gone on making the judgement that it may have been too long. |
| 19:21.05 | hardwire | mchou: that doesn't make you less of a prick. |
| 19:21.42 | mchou | hardwire: speak for you self man |
| 19:21.48 | mchou | yourself* |
| 19:22.16 | ChannelZ | He's not, I think you're a prick too |
| 19:22.38 | hardwire | I think he may have been saying "Speak for yourself" as in "tell your own story" as in "don't judge people" |
| 19:22.42 | mchou | you came back from the coffee shop and restarted this jazz |
| 19:22.46 | hardwire | a few lines after telling p3nguin he has a small mind. |
| 19:22.48 | hardwire | anyhoot. |
| 19:23.00 | ChannelZ | :) |
| 19:23.50 | hardwire | mchou: I wanted to figure out if you were on to something.. and in my own mind I was comparing sip to zap as far as zap groups were concerned and trying to rationalize the reasons why multiple-registrations weren't popular enough to implement.. thats why I brought it back up. |
| 19:23.51 | drmessano | I don't think he's a prick.. I think he's a narrowminded prick that has no better understanding of the inner workings of a typical business than a 5 yr old with Dad's briefcase |
| 19:24.26 | drmessano | See, I came to your defense |
| 19:24.41 | drmessano | Credit plz |
| 19:24.48 | hardwire | drmessano++ |
| 19:24.53 | ChannelZ | snaps |
| 19:24.57 | hardwire | I'm gonna go hug something cute. |
| 19:24.58 | hardwire | bbl. |
| 19:25.16 | *** part/#asterisk mattwj2002 (n=matt@Wikisource/Mattwj2002) |
| 19:27.31 | carrar | http://www.youtube.com/watch?v=O2rGTXHvPCQ |
| 19:27.32 | carrar | haha |
| 19:28.34 | ChannelZ | yeah I saw that |
| 19:28.37 | ChannelZ | Quick, take a picture of the screen before it's ALL GONE! |
| 19:29.43 | drmessano | HAHAHA |
| 19:29.58 | drmessano | Set an alarm.. notify? |
| 19:30.08 | ChannelZ | I know this is where _I_ come if I don't want to be overheard. |
| 19:30.18 | carrar | heh |
| 19:30.44 | carrar | t0u(|-| my 5y573m 4|\|d i7 wi11 347 u y0u|\|g |
| 19:30.51 | drmessano | <ChannelZ> I know this is where _I_ come if I don't want to be overheard. <-- I just googled that and pulled it from the infobot logs |
| 19:31.08 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 19:31.11 | drmessano | Ok, not really |
| 19:31.11 | ChannelZ | hehe |
| 19:31.17 | drmessano | But give it 24 hours |
| 19:31.19 | carrar | I'm bouncing through a 101 IP's |
| 19:31.19 | drmessano | lol |
| 19:31.32 | drmessano | ZOMG ---> 341.17.25.11 |
| 19:31.36 | drmessano | I GOT TEH IP |
| 19:31.48 | carrar | heh |
| 19:31.48 | Xetrov` | ya ok. |
| 19:31.49 | ChannelZ | the 555 of the internet |
| 19:33.15 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 19:39.55 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 19:43.10 | p3nguin | mchou: So you think that your asking me about parallel forking instantly made me go from not knowing about it to knowing about it? Are you really that stupid? What I told you want that Asterisk does not accept multiple registrations on SIP. I don't recall there being any more to it than that. |
| 19:54.53 | hardwire | I was wondering when that was going to happen. |
| 19:56.17 | p3nguin | Yeah, I can't sit around 100% of the time waiting on mchou to make ridiculous assertions. |
| 19:57.09 | hardwire | Anybody have a 4 port sangoma T1 card they want to lease/rent? |
| 19:57.11 | hardwire | heh |
| 19:57.31 | p3nguin | As opposed to sell/give away? |
| 19:57.43 | hardwire | I can't afford sell, I'll gladly take gives. |
| 19:57.51 | hardwire | thanks p3nguin ! |
| 19:58.20 | p3nguin | Wouldn't renting for a long period end up costing more than just buying one? |
| 19:58.29 | hardwire | I need it for a week. |
| 19:58.38 | p3nguin | oh, then I understand your position. |
| 19:59.05 | hardwire | this is giving more credence to my listigi project.. where I want people to list all things and making them available to the "re" market for free. |
| 19:59.07 | p3nguin | For a week, they should just let you borrow it for free. |
| 19:59.29 | hardwire | they being who exactly? I would expect some sort of insurance for the card. |
| 19:59.42 | psilikon | thanks for the help today guys! I'm out. |
| 20:00.02 | p3nguin | "They" being the person (singular they) or group who would have ownership of the card. |
| 20:00.19 | hardwire | p3nguin: do you live in consumer U.S.A? |
| 20:00.21 | hardwire | :) |
| 20:01.00 | p3nguin | Yeah, and I would loan hardware to a trusted individual with legitimate needs. |
| 20:01.10 | hardwire | p3nguin: you're rare. |
| 20:01.15 | p3nguin | Unfortunately for you, I do not have said card. |
| 20:01.19 | hardwire | I would too.. but my boss would never allow it. |
| 20:05.56 | mchou | p3nguin: dont even make me post the log of that conversation |
| 20:07.21 | mchou | p3nguin: it's very clear from that earlier conversation you knew weren't even aware of the concept of parallel forking |
| 20:07.47 | mchou | s/knew// |
| 20:08.03 | hardwire | thanks bot. |
| 20:08.10 | hardwire | so helpful/redundant/spammy. |
| 20:09.21 | p3nguin | mchou: Please post the entire thing. |
| 20:09.34 | mchou | alright |
| 20:09.37 | TJNII | Please don't. |
| 20:09.38 | mchou | here goes |
| 20:10.05 | ChannelZ | utilitizes the /ignore command and makes life happy |
| 20:11.57 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 20:12.39 | *** join/#asterisk acxty (n=acxty@190.6.194.152) |
| 20:12.56 | p3nguin | channelz: That's a powerfull ignore you have; I guess it prevented me from getting the message from him, too. |
| 20:13.23 | ChannelZ | My gift to you :) |
| 20:14.18 | drmessano | Distributed ignore |
| 20:15.04 | p3nguin | That's the next best thing to being able to punch someone in the face over IRC. |
| 20:15.24 | ChannelZ | /bitchslap |
| 20:15.30 | ChannelZ | that's what we need |
| 20:15.43 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:16.20 | p3nguin | Ah, a new protocol: BSoIRC |
| 20:17.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:23.41 | hardwire | dial-a-frap |
| 20:23.43 | hardwire | dial-a-fwap |
| 20:24.14 | Amorsen | Multiple registrations would be very helpful in Asterisk |
| 20:24.29 | Amorsen | It's the one thing I sometimes think of putting a sip proxy up for |
| 20:25.09 | hardwire | praise be the tweet |
| 20:26.21 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 20:27.18 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 20:28.49 | hardwire | Amorsen: it would fit some corner cases pretty well, but is a hefty overhaul for sure. |
| 20:29.06 | hardwire | I think that's what I took away from earlier.. not sure if anybody else did. |
| 20:29.25 | p3nguin | Only mchou is really interested in configuring things that way, anyway. |
| 20:29.50 | hardwire | well.. interest would be good to know |
| 20:30.29 | p3nguin | The majority wants to be able to communicate between devices, which seems to require a different SIP peer/user for each device. |
| 20:31.13 | hardwire | in this channel I think that's a safe assumption. |
| 20:31.15 | Amorsen | It's just handy for users to have both a wireless phone, a mobile phone, and a fixed phone |
| 20:31.41 | Amorsen | You can reasonably easily do that for normal calls, but then they want those 3 phones in a queue |
| 20:31.41 | p3nguin | I have all of those things. |
| 20:31.54 | hardwire | Amorsen: defining USER200=SIP/200&SIP/202-wire3less&Zap/32 in the dialplan seems to help with that |
| 20:31.58 | Amorsen | Where all of them stop ringing if one of them is busy |
| 20:32.06 | hardwire | Dial(${USER200}) |
| 20:32.06 | Amorsen | hardwire: Yes, until you hit queues |
| 20:32.29 | hardwire | Amorsen: local channels ftw? |
| 20:32.30 | Amorsen | Then you do Local/whatever in queues |
| 20:32.34 | hardwire | dynamic queue members ftw? |
| 20:32.36 | Amorsen | And then Asterisk dies |
| 20:32.43 | hardwire | Amorsen: not in my case |
| 20:32.48 | hardwire | Asterisk lives.. |
| 20:33.09 | hardwire | Amorsen: whats your local channel destination look like? I bet I can help it not die. |
| 20:33.23 | Amorsen | I can give you the core dumps |
| 20:33.31 | hardwire | I wouldn't know what the hell to do with those. |
| 20:33.37 | Amorsen | Well there you go then |
| 20:33.53 | p3nguin | If it's dumping core, sounds like you have a more serious problem. |
| 20:34.11 | p3nguin | I've installed some really stupid Dial() rules and never caused * to crash. |
| 20:34.38 | Amorsen | Well, it only happens at relatively high call volumes |
| 20:35.10 | Amorsen | And we aren't talking Dial(), we're talking Queue with Local as members |
| 20:35.12 | p3nguin | Is it possible that you are exhausting the available memory? |
| 20:35.21 | Amorsen | Nope |
| 20:35.28 | hardwire | Amorsen: which will eventually hit a Dial() |
| 20:35.34 | Amorsen | True hardwire |
| 20:35.44 | hardwire | But I see what you're saying. |
| 20:35.51 | Amorsen | Apparently Local in a queue is a bit of a worst case for Asterisk locking |
| 20:36.13 | Amorsen | It also has other disadvantages |
| 20:36.13 | p3nguin | What would be a valid example of that? |
| 20:36.14 | hardwire | I'm having a hell of a time with a quad port PRI card and nobody has been able to help much with it.. I think Digium Support is gonna have to kick in. |
| 20:36.19 | p3nguin | I'm wanting to test it. |
| 20:36.55 | Amorsen | It doesn't die every time, I haven't found the precise scenario yet |
| 20:37.09 | p3nguin | So it does work sometimes? |
| 20:37.26 | Amorsen | Yes, 99% and a bit |
| 20:37.30 | p3nguin | How are you implementing it? I want to test it. |
| 20:37.34 | Amorsen | And I believe that the problem doesn't affect 1.4.x |
| 20:37.42 | p3nguin | (which I am using) |
| 20:37.53 | p3nguin | I still want to test it. |
| 20:38.26 | p3nguin | In queues.conf, member => what? |
| 20:38.36 | Amorsen | Just add Local/something@somewhere to a Queue, then make something@somewhere Dial a few phones |
| 20:39.14 | Amorsen | In my case another bug means that /n is necessary after somewhere, but whether that affects the problem is also unknown |
| 20:40.04 | hjorth | Amorsen: Does the queue members go "Invalid"? |
| 20:40.19 | Amorsen | hjorth: No, I've been through that bug |
| 20:40.29 | hjorth | :o) |
| 20:41.05 | mchou | p3nguin: http://ibot.rikers.org/%23asterisk/20091017.html.gz around 6:35:36 |
| 20:41.37 | Amorsen | Although the fix doesn't appear to be in 1.6.0.18rc3 |
| 20:41.41 | Amorsen | Odd |
| 20:41.46 | mchou | p3nguin: that's the whole thing for everyone to see |
| 20:43.24 | Amorsen | But anyway, life would be a lot easier if Asterisk would just accept multiple registrations |
| 20:43.34 | mchou | p3nguin: 06:39.41 p3nguin It certainly wouldn't make all the devices ring when the DID is called. |
| 20:43.45 | hjorth | Amorsen: I have fooled around with 1.6.0.18rc3 today... Didnt even managed to get a call trough it.. Downgrading to 1.6.0.17 fixed it.. So its must be a bit buggy.. |
| 20:44.05 | mchou | p3nguin: right....That's how familiar you are with the concept of parallel forking |
| 20:44.16 | TJNII | 6 hours? Really? You two have been bickering for over 6 hours? |
| 20:44.30 | p3nguin | mchou: Since * doesn't accept multiple registrations, calling the DID wouldn't make two devices ring. |
| 20:44.41 | mchou | Amorsen: before you were here we all went through a big debate about it |
| 20:45.12 | mchou | p3nguin: stop dodging, the context is THERE for everyone to see |
| 20:45.42 | mchou | Amorsen: let's just say some folks refuse to accept the concept on this channel |
| 20:46.10 | TJNII | follows ChannelZ's precident and /ignores the problem away. |
| 20:46.15 | mchou | Amorsen: they even have a hard time acknowledging its existence in SIP RFC |
| 20:47.47 | p3nguin | I don't see how asterisk's (in)ability is defined by the RFC. |
| 20:47.54 | mchou | Amorsen: after I point folks to chapter and verse they still keep on arguing how it's got limited usefulness |
| 20:48.39 | p3nguin | If I register one phone as 1000 and the turn around and register another phone as 1000, when someone calls 1000, do you think both devices will ring (assuming the correct extens are in place)? |
| 20:48.39 | Amorsen | Well, most Asterisk-competitors can do it, but then most Asterisk-competitors don't consider themselves B2BUA's but something else |
| 20:48.54 | mchou | Amorsen: right. and that's cool |
| 20:49.23 | mchou | Amorsen: but a lot of folks on this channel get all defensive |
| 20:50.05 | p3nguin | If they will, I want someone to let me know, 'cause I was misinformed. |
| 20:50.16 | mchou | Amorsen: like someone is stealing their lunch at the mere mention of parallel forking |
| 20:51.17 | drmessano | I remember not more than a month ago when mchou didn't even know this existed, and now he acts like anyone who doesn't know what it is or doesn't agree it's the best thing ever is moron |
| 20:51.40 | mchou | drmessano: I have long know that this capability existed |
| 20:51.45 | mchou | known |
| 20:51.55 | mchou | I just didnt know it was in the RFC |
| 20:52.12 | p3nguin | can't care less if it's in the RFC |
| 20:53.54 | hjorth | Any of you guys ever implemented Asterisk <-> cisco media gateway? |
| 20:57.21 | hjorth | I am having some difficulties with T.38 |
| 20:58.13 | jblack | Let's have a big group hug |
| 20:58.34 | hardwire | could use a group massage |
| 20:58.34 | hardwire | back hurts. |
| 20:59.11 | Amorsen | More like group therapy, after T.38... |
| 20:59.23 | jblack | I don't even have to look back and see the context. It's always mchou versus someoneone else. |
| 20:59.44 | *** join/#asterisk sahafeez (n=sahafeez@commodity-noc-48.sc09.org) |
| 21:00.36 | p3nguin | Heh, yeah. Always trying to force misinterpretation on someone. |
| 21:01.38 | jblack | p3nguin: Remember when I used to be like that? |
| 21:01.59 | p3nguin | It wasn't the same, but yes. |
| 21:02.28 | jblack | What? I was usually right, he's usualyl wrong? =) |
| 21:03.16 | jblack | I'm sorry mchou. I don't mean to pick a fight with you. You know a lot, and you're trying to share it with others. |
| 21:03.28 | hardwire | indeed.. |
| 21:03.35 | p3nguin | Your methods of doing so just suck. Hard. |
| 21:04.13 | p3nguin | Maybe it's a language barrier. |
| 21:04.19 | jblack | Maybe he's german. |
| 21:04.26 | hardwire | is Prickish a language? |
| 21:04.44 | jblack | Nope. |
| 21:05.25 | mchou | right, unlike you guys, I generally avoid going ad hominem on folks |
| 21:05.32 | mchou | just stick to facts |
| 21:05.40 | p3nguin | amorsen: For what it's worth, I do not like the behavior of Local/123@context in queues. |
| 21:05.40 | hardwire | My buddy at the HP linux testing lab is dealing with overseas development types and is constantly amused at the language/ethical/social barriers. |
| 21:05.53 | jblack | ?!? |
| 21:06.09 | jblack | Maybe your irc client is lagged. I already apologized and complemented you. |
| 21:06.31 | mchou | jblack: no, I was referring to certain name-calling |
| 21:06.34 | ChannelZ | yawns |
| 21:06.46 | jblack | Ok. |
| 21:06.50 | hardwire | covers ChannelZs mouth |
| 21:07.00 | Amorsen | p3nguin: Neither do I, it's tricky to get reasonable behaviour out of |
| 21:07.10 | jblack | Congratulations, channel! You now officially graduate High School! Please leave your drama at the door, and be happy! |
| 21:07.13 | Amorsen | But it's certainly better than not having any way of doing it at all |
| 21:07.21 | ChannelZ | Sorry, that _was_ rude. I am getting over a cold. |
| 21:07.24 | p3nguin | amorsen: The only way it would be acceptable for me is to create a context just for queues. |
| 21:08.03 | hardwire | Amorsen: Local chans as opposed to adding each device to the members list. |
| 21:08.09 | p3nguin | amorsen: And if I was going to do that, I might as well leave them in queues.conf. |
| 21:08.32 | Amorsen | hardwire: Yes, because doing that makes the two others ring even when you're busy on the third one |
| 21:08.35 | p3nguin | I use member => SIP/123 in my queues.conf. |
| 21:08.45 | hardwire | Amorsen: oh.. I see what you're up too. |
| 21:09.08 | hardwire | Amorsen: that should really be tied to the priority system somehow, you know. |
| 21:09.20 | p3nguin | SIP/123,2 ? |
| 21:09.21 | Amorsen | Possibly, but it would be quite complicated |
| 21:09.30 | hardwire | Amorsen: I use Local with Group |
| 21:09.34 | hardwire | It works really well. |
| 21:09.37 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 21:09.37 | hardwire | even on 1.2.x |
| 21:09.40 | p3nguin | I can see a use for doing it this way, but I do not like it. |
| 21:09.47 | Amorsen | If you could add SIP/123&SIP/124 as a queue member, that would work too |
| 21:10.25 | drmessano | [16:06] <mchou> right, unlike you guys, I generally avoid going ad hominem on folks <-- Laughable, since you got banned last night for insulting a dev |
| 21:10.32 | drmessano | Just sayin |
| 21:10.33 | hardwire | Amorsen: Agreed, they are putting the complication of specific configuration in your hands, for sure. |
| 21:10.47 | jblack | Hmm. I wonder how pickup groups could fit into this |
| 21:11.21 | ChannelZ | grrph. I wish I could re-order the function keys on these SPA phones |
| 21:11.25 | mchou | drmessano: the guy was intentionally being obtuse, and I called him out for being dense. nothing wrong with that |
| 21:11.39 | jblack | mchou: cmon, try and be nice too, ok? |
| 21:11.46 | jblack | please? |
| 21:11.51 | p3nguin | If you got banned, someone must have thought there was something wrong with it. |
| 21:12.05 | mchou | oh right |
| 21:12.10 | drmessano | mchou: Actually he asked you for facts and you went "ad hominem" on him.. |
| 21:12.12 | jblack | p3nguin: You too? |
| 21:12.19 | drmessano | Should I pastebin the log? |
| 21:12.20 | p3nguin | Me? Nice? |
| 21:12.28 | mchou | sure, go right ahead |
| 21:12.30 | jblack | Guys, All you, please. Because I asked nicely! Grow up! |
| 21:12.33 | mchou | drmessano: ^^^ |
| 21:12.37 | hardwire | ugh |
| 21:12.46 | Amorsen | brb phone |
| 21:12.52 | hardwire | brb life |
| 21:12.54 | mchou | let's see who went of the handle on that one |
| 21:13.02 | mchou | s/of/off |
| 21:13.21 | p3nguin | I love how you rarely terminate your expressions correctly. |
| 21:13.33 | p3nguin | Makes me think you've never used sed a day in your life. |
| 21:13.46 | ChannelZ | is reminded of the un-PC but funny 'fighting on the internet' poster |
| 21:13.52 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 21:13.54 | hardwire | ok.. stop.. collaborate and listen.. everybody needs to understand this is a growth op.. :) |
| 21:14.11 | jblack | p3nguin: In all fairness, many editors don't require that the expression be closed. |
| 21:14.20 | jblack | In particular, vim. |
| 21:14.22 | p3nguin | Sounds lazy. |
| 21:14.31 | *** join/#asterisk TravisB (n=TravisB@ool-457e2525.dyn.optonline.net) |
| 21:14.35 | TravisB | drama? |
| 21:14.40 | p3nguin | yes, please. |
| 21:14.46 | jblack | Isn't that a goal of system administration? To be lazy? |
| 21:14.56 | TJNII | wishes this was a message board so he could post the "butthurt report form." |
| 21:14.56 | TravisB | Sounds about right to me. |
| 21:15.00 | ChannelZ | hells yeah |
| 21:15.11 | hardwire | hah |
| 21:15.14 | TravisB | when I was a system admin I tried having the computer do work for me, instead of the other way around. |
| 21:15.35 | jblack | seems to have struck an intrinsic truth |
| 21:15.59 | drmessano | http://www.pastebin.ca/1672605 |
| 21:16.03 | hardwire | s/lazy/super rad/g |
| 21:17.24 | mchou | drmessano: why dont you post the 10 lines or so after the ban |
| 21:17.33 | jblack | Oh, you got banned? |
| 21:17.40 | drmessano | Gladly |
| 21:17.55 | mchou | jblack: check drmessano pastebin |
| 21:18.17 | jblack | I'd say you insulted the wrong person, mchou. |
| 21:18.42 | mchou | jblack: he was being intentionall obtuse |
| 21:18.52 | jblack | setting asside the ban issue, I think iax2 is a pretty good protocol. |
| 21:19.10 | drmessano | http://www.pastebin.ca/1672608 <-- Updated, but not sure what the point is |
| 21:19.22 | jblack | mchou: You're not going to convince anyone of anything by calling your 'opponent' words like "dense" and "obtuse". |
| 21:19.23 | drmessano | He wasn't |
| 21:19.35 | drmessano | He asked you for facts |
| 21:20.03 | mchou | and I gave him the facts |
| 21:20.16 | jblack | And if you understand that, then you need to decide what your goal is.. Either you're trying to educate someone, or you're trying to stir up trouble. |
| 21:20.49 | jblack | mchou: Very few people are going to hear your point if you're insulting them at the same time. Putting people on the defensive doesn't make them receptive. It does the opposite. |
| 21:21.00 | ChannelZ | rings the winning bell |
| 21:21.02 | mchou | jblack: that convo hardly rises to a bannable offense |
| 21:21.57 | jblack | We can discuss that if you want, but I'd say banning someone for an insult isn't unreasonable. I don't know if it's right or wrong, but it's certainly subject to reason, and thusly reasonable. |
| 21:22.23 | drmessano | mchou: you failed to support your claim with facts.. You didnt list specific providers or cite specific instances.. When you were asked for specifics, you went ever further off the deep end claiming everyone in here recommends SIP over IAX.. Well, that's not true, do NOT speak for me, please, and again, you cited no specifics |
| 21:22.33 | drmessano | Broad generalizations and insults are not making a case |
| 21:22.35 | hardwire | svn log -v http://svn.digium.com/svn/asterisk/trunk/channels/chan_iax2.c | grep tilghman | wc |
| 21:22.53 | jblack | It's ok to disagree, and it's ok to state it, and it's fantastic if you support it. Really! |
| 21:23.04 | jblack | I spend most of my days disagreeing with people. =) |
| 21:23.32 | mchou | drmessano: I fail to see why I needed to provide names of the providers. Why, he was going to go to those places and fix it himself? |
| 21:23.32 | jblack | I try to not classify them as stupid just for disagreeing with me. I could be wrong, after all. =) |
| 21:24.04 | jblack | mchou: if you make a statement, shouldn't you be willing to support it? |
| 21:24.14 | hardwire | I need chips and salsa |
| 21:24.17 | hardwire | and kittens |
| 21:24.23 | drmessano | mchou: He asked you for some shred of evidence or some specifics.. one of those could have been a provider and some specific circumstances.. |
| 21:24.25 | ChannelZ | just had nachos, sans cats |
| 21:24.45 | mchou | jblack: the point was, there was nothing he could do about it |
| 21:24.56 | drmessano | mchou: You couldn't even state "I used provider X and my connection dropped every 10 mins" .. Just "IAX sucks, I KNOW IT, K" |
| 21:24.59 | jblack | mchou: Certainly you'd agree that someone that is an op in this channel isn't dense.. especially about asterisk. |
| 21:25.09 | jblack | Can we agree on that point? |
| 21:25.14 | mchou | jblack: instead he accused me of ulterior motives, as the lines following the ban show |
| 21:25.35 | drmessano | mchou: Sure, because you were trolling.. Obviously you werent here to provide facts |
| 21:25.43 | jblack | If I promise to follow you in your thought process later, will you follow in my thought process now? |
| 21:25.51 | mchou | like I had an axe to grind against iax or something |
| 21:25.59 | drmessano | mchou: You also lied about everyone recommending SIP over IAX, to support some claim you couldn't back up |
| 21:26.22 | jblack | drmessano: Oh, not you too. "lied" ? |
| 21:26.33 | mchou | baloney. That's no line |
| 21:26.33 | *** join/#asterisk n3tz666 (n=n3tz666@host123-29-dynamic.13-79-r.retail.telecomitalia.it) |
| 21:26.36 | mchou | lie* |
| 21:26.37 | jblack | can't you help me settle thigns down, rather than stir them up? |
| 21:26.38 | n3tz666 | Hi all |
| 21:26.39 | drmessano | jblack: Made up? |
| 21:26.46 | drmessano | jblack: Fabricated? |
| 21:27.01 | jblack | How about "wrongly claimed" ? |
| 21:27.03 | ChannelZ | hint: the more you talk to him, the more he talks back |
| 21:27.24 | jblack | What am I doing.... |
| 21:27.25 | mchou | every time someone comes into this channel complaining about iax connectivity, the first response is "go sip" |
| 21:27.36 | jblack | I love you all. Especially you, mchou! |
| 21:27.44 | ChannelZ | lets fap! |
| 21:28.01 | jblack | mchou: I've seen seen that frequently. |
| 21:28.29 | *** part/#asterisk n3tz666 (n=n3tz666@host123-29-dynamic.13-79-r.retail.telecomitalia.it) |
| 21:28.38 | jblack | It drives me crazy, because I think IAX2 is fantastic. I think people got bit hard by IAX2 a few years ago, and old wounds heal slow. |
| 21:29.15 | jblack | but I never know. Maybe they know something that I don't... though each time I've approached someone, the core of it seems to be an old bias that just sticks around. |
| 21:29.26 | Amorsen | Well in the beginning people got hit hard by SIP and went IAX2, then the reverse, and now they're probably about equal |
| 21:29.58 | drmessano | I think mchou is taking the recommendation that IAX2 is NOT better than SIP, use SIP because everyone else is, as "IAX2 sucks and everyone says so".. No, I am not gonna tell someone to switch from SIP to IAX2, there is no need.. But if they want to implement IAX2, it works great. |
| 21:30.18 | jblack | too many people fail to keep in mind that circumstances change, and what was right a year ago, may not still be right today. |
| 21:30.43 | carrar | GO SIP |
| 21:30.44 | jblack | I've seen seen reputable people in this channel say that IAX2 sucks. |
| 21:30.52 | carrar | yip yip ahray |
| 21:31.09 | *** join/#asterisk dajhorn (n=chatzill@user-0cet72i.cable.mindspring.com) |
| 21:31.24 | jblack | That's not everybody, of course... Then again, it only takes one person to scream fire in a movie theatre to cause a panic... |
| 21:31.28 | carrar | X.25! |
| 21:31.52 | drmessano | jblack: Sure it is.. Because now even you admit everyone says IAX2 sucks, or so mchou will claim |
| 21:31.55 | jblack | Here's the sad thing. A couple of the providers I use have dropped iax. |
| 21:32.24 | jblack | Oh, don't admit everyoen says iax2 sucks. I just said myself that I think that it's fantastic, and I'm a subset of everyone. |
| 21:32.43 | jblack | So right there, we're at "not everyone". |
| 21:33.02 | ChannelZ | of course this is a microcosm, a support channel.. people generally come in here with their problems, not to give status reports about how everything is working perfectly. |
| 21:33.11 | p3nguin | I use IAX2 trunks. |
| 21:33.23 | p3nguin | I feel like it's a preference. |
| 21:33.26 | dlynes | ChannelZ: Everything is working properly. |
| 21:33.34 | jblack | I love iax2. I'm really sad that it seems to me that providers are dropping support for it. |
| 21:33.40 | TJNII | I used to, back before my friends shut off their * servers. Worked great. |
| 21:33.56 | jblack | I think sip was designed before nat and firewalling became so popular, and it shows. |
| 21:34.07 | ChannelZ | dlynes: Whew! There's one! :) |
| 21:34.16 | Amorsen | It wasn't really though |
| 21:34.26 | drmessano | Google doesnt use IAX |
| 21:34.30 | Amorsen | They just decided to ignore the problem |
| 21:34.31 | drmessano | Google is the standard |
| 21:34.36 | drmessano | Therefore, IAX sucks |
| 21:34.42 | carrar | haha |
| 21:34.43 | drmessano | and Google said so |
| 21:34.53 | ChannelZ | I use IAX between work and home, though it's not heavily utilized. But it sure as hell was easier to setup through firewalls on both sides than SIP |
| 21:34.57 | jblack | what did he say? I had him ignored. |
| 21:35.01 | Amorsen | Not that IAX has a much easier time of it, IAX reinvites over NAT must have the same issues |
| 21:35.45 | jblack | Amorsen: Sure, though I'm not sure why iax2 would get the blame for that, since it's the closing of one channel and the opening of a new one |
| 21:36.17 | jblack | thinks harder and sorts asees amorsen's point |
| 21:36.18 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:36.27 | jblack | sort of sees, that is |
| 21:36.30 | dlynes | We have IAX2 going between the CPE and the CO, and then SIP from endpoints to CPE, and CO to upstream...the odd customer wants IP centrex endpoints, so they're on SIP, but we prefer to have an asterisk box for the CPE so that we can do IAX2 |
| 21:36.33 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 21:37.07 | dlynes | The only problem I've ever had with IAX2 (that I can remember) is incompatibilities between certain versions of asterisk |
| 21:37.27 | Amorsen | If you want audio to get directly between devices, NAT will kill you. I think SIP with STUN/ICE is probably closest to a "solution" for that right now |
| 21:37.54 | *** join/#asterisk Meaw (i=dino@188.161.187.177) |
| 21:38.14 | *** join/#asterisk Sier (n=sier@unaffiliated/sier) |
| 21:38.27 | Amorsen | We just avoid NAT, which sidesteps the issue, but not everyone is so fortunate |
| 21:38.31 | jblack | The only time I've seen a problem with iax2 is when there's serious problems on a machine.. I've found that IAX2 doesn't tolerate time going backwards, but time shouldn't go backwards in the first place... |
| 21:39.26 | jblack | sip seems more tolerant, as there's no call trunking going on. |
| 21:39.34 | dlynes | jblack: you mean like one machine has a different date than the other, or one machine in particular's clock is running in reverse? |
| 21:39.52 | jblack | dlynes: I mean a machine's clock jumping backwards. |
| 21:40.15 | dlynes | jblack: ah...so most of the time it's going forwards, but every once in a while, it takes a jump backwards? |
| 21:40.40 | jblack | Yeah. It was a xen bug back in .. well pardon the pun.. but back in the past... |
| 21:41.29 | jblack | It's long, long since been fixed, but there were a couple versions where the wall clock could stop, go backwards, jump forwards. |
| 21:41.56 | dlynes | ah...I guess i'll never run into that problem then...don't plan on running asterisk in a VPS or VPE any time soon |
| 21:41.56 | ChannelZ | like ntp |
| 21:42.00 | jblack | Truely timing hell, and there were many things affected far worse than IAX2... for example, NFS would have conniption fits. |
| 21:42.19 | TJNII | Poorly configured ntp, that is. |
| 21:42.36 | jblack | TJNII: Pardon? |
| 21:43.08 | dlynes | TJNII: I have problems with ntp all the time even on devices I can only configure a server, and nothing else about ntp |
| 21:43.15 | jblack | Oh, I see. |
| 21:43.19 | TJNII | jblack: Response to ChannelZ. If NTP is configured properly it will not just reset the clock, but slow it down. |
| 21:43.47 | TJNII | Or at least that is my understanding from reading about the ocnfigs.... |
| 21:43.51 | jblack | No ChannelZ, it wasn't ntp, which suicided itself when it saw basic physics violations. :) |
| 21:44.41 | jblack | It was way up in kernelspace. Dom0 wasn't controlling the hardware for the domu's correctly, and so from there.... |
| 21:45.11 | jblack | hardware clocks dont' typically change backwards 100s of thousands of seconds ..... |
| 21:45.21 | ChannelZ | I meant in general for clocks to go backwards |
| 21:45.28 | ChannelZ | or jump backwards anyways |
| 21:45.42 | jblack | Ahh. THere's more than one clock in a system, believe it or not. |
| 21:45.55 | Sier | Not sure if this is the right palce to ask.. All my wireless clients are being dropped when I receive phone calls.. My phone is a 5.8Ghz phone, so I don't know what's going on.. Using PAP2T-NA |
| 21:45.59 | Amorsen | Yes, and they all suck |
| 21:46.05 | Amorsen | At least on the PC architecture |
| 21:46.13 | jblack | there's the system clock, the hardware clock... the kernel might still have a counter or three within, still... |
| 21:46.23 | carrar | haha |
| 21:46.24 | Amorsen | There's more than one hardware counter |
| 21:46.28 | jblack | well, imagine the disaster when the hardware clock starts jumping backwards by large amounts. |
| 21:46.35 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:46.38 | jblack | That just don't happen! |
| 21:47.20 | jblack | except on certain old versions of xen, it did. Which drove NFS and iax2 insane. |
| 21:47.31 | jblack | among other things, of course. |
| 21:48.17 | jblack | is there someone I can trust that will be here in 30 minutes? |
| 21:48.22 | jblack | I need a big favor |
| 21:48.23 | carrar | Sier, what mode of wifi are you using what what model of cordless phone? |
| 21:48.27 | Amorsen | Read the kernel timekeeping code and weep |
| 21:48.30 | carrar | Sounds like overlapping frequencies |
| 21:48.32 | jblack | that will be really easy |
| 21:48.43 | jblack | p3nguin: ping |
| 21:48.45 | jblack | drmessano: ping |
| 21:48.51 | ChannelZ | Does it require having all your fingers? |
| 21:49.00 | jblack | No, just 1. |
| 21:49.12 | drmessano | ? |
| 21:49.15 | TJNII | Allright, but no gay stuff, okay? |
| 21:49.29 | ChannelZ | yeah, sounds supicious |
| 21:49.37 | ChannelZ | err suspicious even |
| 21:49.39 | Sier | carrar Wifi G Channel 10, Cordless phone = Uniden TRU9585 (5.8Ghz Digital) |
| 21:49.59 | Sier | I coulnd't find a place to "set up" the channels on the Uniden phone... |
| 21:50.26 | TJNII | Sier: Unplug the phone, leave the pap2 connected. See what happens. Disconnect the pap2, take the coordless off-hook. See what happens. |
| 21:51.57 | ChannelZ | Is the wifi 802.11n? I thought the other standards were down in the 2Ghz area |
| 21:52.24 | Amorsen | 5.8GHz is the US equivalent of DECT |
| 21:52.28 | Amorsen | Not Wifi at all |
| 21:52.57 | drmessano | I thought DECT was the US Equivalent of DECT? |
| 21:53.11 | Sier | Only one question, do I have to forward any ports to the pap2? |
| 21:53.20 | drmessano | 5.8GHZ phones are not DECT in the US |
| 21:53.49 | Amorsen | True, I'm wrong. Sorry |
| 21:54.19 | carrar | Sier, you sure that phone is 5ghz only? |
| 21:54.21 | Amorsen | But either way we aren't talking wifi |
| 21:54.38 | Sier | carrar it says on the base 5.8Ghz ||| Digital |
| 21:54.46 | carrar | and positive you are G |
| 21:54.51 | carrar | What wifi device? |
| 21:54.57 | Sier | yes, my router is capable of N, but I chose G only |
| 21:55.04 | Sier | it's a netgear WNR1000 |
| 21:55.10 | carrar | N works in 5ghz btw |
| 21:55.21 | p3nguin | jblack: yeah? |
| 21:55.25 | Sier | I chose, Up to 54Mbps, Channel 06 |
| 21:55.32 | Sier | up to 54Mbps = G, correct? |
| 21:55.58 | carrar | thats like saying a corvette can do up to 54 |
| 21:56.07 | jblack | p3nguin: I got it. Thanks. |
| 21:56.09 | Sier | The device doesn't have a option to choose B/G/N, etc.. It only has: Up to 54Mbps, up to 65Mbps and Up to 150Mpbs |
| 21:56.18 | Amorsen | Sier: the phone isn't Wifi at all, so if you're trying to pair it with a Wifi access point you'll be disappointed |
| 21:56.19 | hardwire | an idea why asterisk would cause the context switches per 10 minutes to go from 10k to 495k |
| 21:56.22 | p3nguin | jblack: alright |
| 21:56.22 | hardwire | with no load |
| 21:56.38 | hardwire | and eventually one million :) |
| 21:56.42 | ChannelZ | Amorsen: the issue is that the phone is interfering with the wifi, not that he's trying to connect them (I thought?) |
| 21:56.57 | Sier | ChannelZ correct. |
| 21:57.25 | Sier | Up to 54 Mbps - Legacy Mode with maximum speed of up to 54 Mbps for b/g networks. (Router description for option "Up to 54Mbps") |
| 21:57.25 | p3nguin | drmessano: I have a 5.8 GHz cordless phone (in the US). What is it if it's not DECT? |
| 21:57.50 | Katty | peeks head in |
| 21:58.09 | Amorsen | Sier, just avoid channels over 140 or so and you should be fine |
| 21:58.20 | Katty | ALLO |
| 21:58.22 | TJNII | waves to Katty |
| 21:58.30 | carrar | Sier, get a polycom Spectralink Wifi phone and be done with it |
| 21:58.32 | Katty | waves at TJNII |
| 21:58.40 | Sier | I can choose from channels 01 to 11 , which should I choose? |
| 21:58.50 | Amorsen | Sier: That's 2.5GHz channels |
| 21:58.57 | Amorsen | 5.8GHz won't mess with those |
| 21:59.08 | Katty | p3nguin: hmm. that's good to know. i never thought about replacing chicken with rabbit, but rabbit is all white meat. |
| 21:59.19 | TJNII | Katty: I'm eating a breakfast I decided to cook, only cost me one fingerprint! |
| 21:59.30 | Katty | TJNII: what did you make for breakfast? |
| 21:59.39 | TJNII | Katty: Cornbread. |
| 21:59.48 | TJNII | Well within my skill set. |
| 21:59.48 | jblack | hmmm. cornbread... |
| 21:59.56 | Katty | hmm. cornbread. haven't made that in awhile. |
| 22:00.06 | Katty | i like it with green chilies or chopped jalpeno in it |
| 22:00.07 | p3nguin | katty: I would also use chicken recipes on squirrel, but I doubt that you want to eat squirrel, if I know you. ;) |
| 22:00.10 | jblack | I love hush puppies. I think I miss those the most. |
| 22:00.23 | Katty | p3nguin: no, i highly doubt i'll be eating squirrel. |
| 22:00.33 | Katty | p3nguin: but i'm not opposed to eating it, if i were hungry enough. |
| 22:00.47 | drmessano | p3nguin: DSSS or FHSS |
| 22:00.49 | ChannelZ | I should get a varmint rifle so I can shoot the squirrels off my roof that wake me up |
| 22:01.00 | jblack | ChannelZ: Then et 'em? |
| 22:01.12 | carrar | Sier, it's possible your cordless phone has really bad spurious harmonics |
| 22:01.19 | ChannelZ | mebbe. Never had it |
| 22:01.43 | Katty | http://hphotos-snc3.fbcdn.net/hs005.snc3/11236_102777489741957_100000291201041_75300_2136153_n.jpg <- dad's rabbits that will be ready in a couple more weeks |
| 22:02.03 | Amorsen | carrar: Then it really shouldn't have passed approval. But who knows. |
| 22:02.07 | ChannelZ | very cute |
| 22:02.11 | carrar | Sier or your wifi is seeing the cordless phone on 5Ghz (wifi N mode) and trying talk to it, creating issues for your G clients |
| 22:02.15 | carrar | possibly |
| 22:02.29 | carrar | hard ot say |
| 22:02.48 | p3nguin | drmessano: I have no idea. I just know it says 5.8 GHz Extended Range cordless phone on it. |
| 22:02.51 | Katty | new zealands are BIG rabbits. meant for eating. |
| 22:02.58 | Amorsen | N is probably running on 2.5GHz in this case, since the router isn't offering 802.11a |
| 22:03.00 | drmessano | p3nguin: I was answering you, not asking |
| 22:03.09 | p3nguin | drmessano: oh, sorry. |
| 22:03.13 | jblack | Katty: "meant for eating". That's precious. =) |
| 22:03.34 | carrar | Sier, buy a Cisco 1231G and be done with it :) |
| 22:03.41 | Katty | http://www.normelstud.co.uk/resources/07.jpg <- full grown new zealand. |
| 22:03.45 | jblack | You, poor bunny, exist to fill my tummy |
| 22:03.48 | Katty | of course you'd never eat a fullgrown bunny. |
| 22:03.48 | p3nguin | drmessano: I also know that there are phones that explicitly state things like DECT 6.0 on them. I don't know a lot about the cordless technologies, though. |
| 22:03.55 | Sier | carrar send me the money :) |
| 22:04.00 | jblack | wow, look at that thing. |
| 22:04.07 | carrar | $45 on ebay |
| 22:04.10 | carrar | You can afford it |
| 22:04.12 | drmessano | p3nguin: DECT is a standard and is implemented on the 6.0GHZ band in the US. |
| 22:04.28 | Katty | jblack: the parents are spoiled rotten, and won't be eaten obviously (= |
| 22:04.44 | Katty | jblack: they smaller rabbits, when they hit fryer stage, sell for about 5 bucks each |
| 22:04.47 | jblack | not if they make fat kids like that! |
| 22:05.04 | jblack | I'd keep rabbits if I didn't live in the city. |
| 22:05.09 | jblack | Practically free food. |
| 22:05.25 | Katty | yeah, but they will taste pretty wild unless you feel them corn. |
| 22:05.33 | Katty | not that corn is expensive. |
| 22:05.39 | jblack | I don't mind gamey meat. |
| 22:05.57 | Katty | raise rabbits in the house |
| 22:06.09 | jblack | I already stink the joint up enough on my own. |
| 22:06.12 | Katty | hehe , k |
| 22:06.24 | Katty | oh! i must show you my dad's chickens |
| 22:06.35 | jblack | You, my dear katty, are an enigma |
| 22:07.07 | TJNII | "An enigma wrapped in a riddle warpped in .... secret sauce." |
| 22:07.10 | Katty | http://farm3.static.flickr.com/2703/4105745219_0543b20558_b.jpg <- you can see the rabbit hutches in the background. |
| 22:07.10 | *** join/#asterisk Meaw (n=dino@188.161.187.177) |
| 22:07.26 | Katty | jblack: why do you say that |
| 22:07.47 | jblack | shrugs |
| 22:07.57 | Katty | http://farm3.static.flickr.com/2795/4105744293_e67de3f5f6_b.jpg <- hens. |
| 22:08.05 | carrar | farm girl Kattty? |
| 22:08.17 | Sier | carrar worked fine!! |
| 22:08.18 | Katty | carrar: i was raised on...an almost farm? |
| 22:08.32 | Sier | I like chicken :) |
| 22:08.33 | Katty | carrar: nothing was killed. |
| 22:08.34 | carrar | almost awesome!! |
| 22:08.40 | Katty | carrar: it was more of a petting zoo farm. |
| 22:08.46 | carrar | heh |
| 22:08.53 | carrar | best kind |
| 22:08.59 | Katty | carrar: but my parents had chickens, and rabbits, and goats. |
| 22:09.03 | carrar | Killing a friendly cow is tramatizing |
| 22:09.14 | Katty | they didn't have room for a cow. |
| 22:09.28 | Katty | but i wouldn't mind a jersey |
| 22:09.37 | Amorsen | Anything without a name is fair game for eating... |
| 22:09.38 | Meaw | Do you have a zebra? |
| 22:09.45 | Katty | no. |
| 22:09.49 | Katty | i have 5 ferrets and a german shepherd. |
| 22:09.56 | Katty | can't have livestock within city limits |
| 22:09.57 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
| 22:10.10 | carrar | My parents raise some cows for meat, became friends with it then ate it, and kept it's skin |
| 22:10.20 | Meaw | you can paint a zebra like the one we have here. |
| 22:10.23 | Sier | do you also give support to FOP here? |
| 22:10.25 | Meaw | paint a donkey |
| 22:10.27 | carrar | I have this huge cow hide in my cloeset |
| 22:10.35 | Katty | Sier: a lot of us work with FoP here. |
| 22:10.44 | Katty | Sier: but it's not Commercial Support |
| 22:11.22 | Sier | mine doesn't work ... I see the "Asterisk Flash Operator Panel", -- No Timeout, and 3 icons on the left, but that's it.. everything else is blank.. |
| 22:11.24 | Sier | what can I do? |
| 22:12.08 | Katty | well you can pastebin some stuff and see if anyone has the time to help you |
| 22:12.20 | Katty | another option is to find a consultant, or contact the makers of FoP for assistance |
| 22:12.48 | Katty | as a personally suggestion, i would go with Isymphony instead of FoP |
| 22:13.16 | Katty | i like it a /lot/ better. you can register for 5 licenses for free...and they have a team of people which have always been very helpful when i've ran into issues |
| 22:13.35 | Katty | you can also pay for a full product license and get some extra features. |
| 22:13.41 | Katty | infobot: isymphony? |
| 22:13.42 | infobot | rumour has it, isymphony is http://www.i9technologies.com/ |
| 22:13.46 | Katty | ^- link |
| 22:20.36 | dan__t | hi. |
| 22:20.37 | Sier | thanks |
| 22:21.43 | dan__t | So when it comes to AGI... Where should one generally draw the line between having a dialplan mostly in Asterisk, or mostly in AGI, et etc. What are some good rules of thumb? |
| 22:24.23 | jblack | hmmm. When you're starting to get into a lot of variable maninpulation and looping, time to start considering a switch to agi. |
| 22:25.36 | jblack | Also, if you're doing something external to asterisk, it may be time to start pulling some of it out into a script with better support. |
| 22:27.46 | Katty | i was going to go make some english muffins, but i appear to be out of yeast :< |
| 22:30.31 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 22:36.30 | Godfather_ | hi, how can i dial trought my spa3102? now i'm been able to recieve calls from pstn, but i've no idea how to call trought it |
| 22:36.56 | Godfather_ | here is my pastebin where i set up the spa http://pastebin.com/m5e3a1b8c |
| 22:37.54 | manxpower | If anyone knows about Windows login problems, please /msg me. |
| 22:38.06 | Godfather_ | i want for example, all 971XXXXXX dials via the spa3102 |
| 22:38.59 | manxpower | Godfather_: Dial(SIP/971XXXXXX@sipconentryforspafxo) |
| 22:39.14 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:39.31 | Sier | "Could not reload the FOP operator panel server using the bounce_op.sh script. Configuration changes may not be reflected in the panel display." |
| 22:39.35 | Godfather_ | manxpower, what's sipconentryforspafxo ? |
| 22:39.41 | manxpower | Sier: Try the correct channel. |
| 22:39.48 | manxpower | Godfather_: the entry in sip.conf for that port on that device. |
| 22:40.38 | Godfather_ | manxpower, i posted the entry of sip.conf, i've no idea what port i should put there |
| 22:41.14 | Sier | manxpower they're all sleeping over there :) |
| 22:41.32 | manxpower | Godfather_: I see exactly one entry there. You need to have BOTH ports register, IIRC. |
| 22:41.47 | manxpower | Your SPA has two ports right? |
| 22:42.03 | Godfather_ | fxs and fxo yes |
| 22:42.06 | manxpower | Sier: Welcome to the world of FreePBX. |
| 22:42.19 | manxpower | Godfather_: and so each port should register to Asterisk as a different sip user, right? |
| 22:42.20 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 22:42.26 | Godfather_ | i've registered both manxpower , i just posted one |
| 22:42.40 | manxpower | Godfather_: we can't help you if you don't provide complete information |
| 22:43.14 | Godfather_ | manxpower, its necesary the registry of the fxs port? |
| 22:43.18 | Godfather_ | i've no idea |
| 22:43.47 | p3nguin | I use my SPA-3102 as a "splitter" so that I can use a regular phone to make selective calling out either PSTN or Asterisk, depending on what is dialed. The 3102's dialplan is what makes the decision whether to send it out the FXO or Ethernet port. |
| 22:44.03 | Godfather_ | manxpower, http://pastebin.com/m6a4a131d |
| 22:44.07 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 22:44.33 | Godfather_ | p3nguin, so i have to configure it |
| 22:44.40 | Godfather_ | by the webbrowser no? |
| 22:45.06 | p3nguin | yes |
| 22:45.30 | p3nguin | By default, most of the calls you make will go out over SIP (if you configured the SIP account on it). |
| 22:46.18 | Godfather_ | p3nguin, i've configured 2 account, 102 for the FXS and 105 for the fxo (pstn line) |
| 22:46.37 | Godfather_ | p3nguin, do you have any web about this topic? |
| 22:51.54 | TravisB | well |
| 22:51.58 | TravisB | that wasn't so smart. |
| 22:52.04 | TravisB | Good thing I use znc. |
| 22:52.12 | TravisB | restarted dbus... not thinking it'd shut EVERYTHING down. |
| 22:52.49 | p3nguin | godfather_: Are you connecting your 3102 to a wall jack for PSTN or to Asterisk? |
| 22:53.40 | Godfather_ | p3nguin, my 3102 is connected to the wall jack (fxo port) and also to asterisk (rj45) |
| 22:53.53 | drmessano | Thats ok, route -f on a windows server is bad.. Just so you know |
| 22:53.56 | p3nguin | godfather_: Then you only need one SIP account on Asterisk. |
| 22:54.06 | drmessano | How bad? Couldnt tell you.. i cant connect |
| 22:54.11 | Godfather_ | p3nguin, i did that, that works |
| 22:54.17 | Godfather_ | i can recieve calls from pstn |
| 22:54.26 | Godfather_ | in my softphones over sip |
| 22:54.34 | Godfather_ | but no idea how to call over the pstn |
| 22:55.02 | p3nguin | godfather_: Pick up the phone, dial something. What happens? |
| 22:56.04 | Godfather_ | mmm |
| 22:56.21 | Godfather_ | p3nguin, i registered the pstline account like the other users accounts |
| 22:56.28 | Godfather_ | nothing |
| 22:56.41 | Godfather_ | http://pastebin.com/m6a4a131d |
| 22:56.43 | p3nguin | godfather_: Do you have the PSTN jack on the 3102 going to Asterisk? |
| 22:56.49 | Godfather_ | the [105] |
| 22:56.56 | Godfather_ | account is the pstline account |
| 22:57.32 | Godfather_ | p3nguin, my spa3102 is connected to the wall jack(fxo), to a phone(fxs) and to my router with rj45 |
| 22:57.35 | p3nguin | godfather_: Then what makes you think you need to configure the pstn line account with your SIP information? |
| 22:58.06 | p3nguin | Some reasonable forethought is required. |
| 22:58.20 | Godfather_ | p3nguin, to been able to get the calls from pstn? |
| 22:58.37 | p3nguin | godfather_: Does your phone company require you to register to them? |
| 22:58.47 | Godfather_ | no.. |
| 22:59.04 | Godfather_ | p3nguin, wait, i believe you dont understand me |
| 23:00.02 | Godfather_ | well, to recieve calls from the pstn and getting those dials to a voipphone of your network |
| 23:00.17 | Godfather_ | you have to register a account in the asterisk for the pstnline no? |
| 23:01.45 | *** part/#asterisk Malkor (n=marco@hlle-d9ba5fe6.pool.mediaWays.net) |
| 23:02.05 | Godfather_ | this is what i have not, in the pstn-line tab, in proxy and registration i put the proxy (ip of asterisk), user id (105) and password(test) |
| 23:02.15 | Godfather_ | *this is what i have now |
| 23:02.43 | Godfather_ | and in the dialplan inside the spa3102 in the dialplan i have (S0<:s>) |
| 23:03.27 | p3nguin | Seems like you should have left the default dialplan intact and configured your SIP acount on line 1. |
| 23:03.31 | Godfather_ | then, in extensions.conf in the default context, i can get the dials from pstn, exten => s,1,Dial(..) |
| 23:04.05 | Godfather_ | p3nguin, i registered in asterisk the line 1 too |
| 23:04.13 | Godfather_ | with another extension |
| 23:04.33 | p3nguin | I'm going to ask once more and see if you can get it this time. Are you connecting your PSTN jack to Asterisk, or are you using SIP over Ethernet? |
| 23:05.08 | p3nguin | I keep asking you, hoping that you'll understand the reasoning behind the question. |
| 23:05.34 | prgmrchris | :) |
| 23:05.46 | Godfather_ | language problem maybe, sorry |
| 23:06.07 | Godfather_ | the 2ond one i think |
| 23:06.24 | Godfather_ | wall jack -> fxo spa3102 |
| 23:06.40 | Godfather_ | fxs jack -> analog phone |
| 23:06.54 | Godfather_ | rj45 3102 -> rj45 router |
| 23:07.41 | p3nguin | right. |
| 23:07.48 | p3nguin | Now think about that for a while. |
| 23:08.22 | Godfather_ | p3nguin, while(true)? .. |
| 23:08.24 | Godfather_ | lol |
| 23:09.42 | Godfather_ | No idea what you are trying to say to me, its simply, i want a voipphone to call to the pstn, and the spa3102 its supposed to do that |
| 23:10.14 | Godfather_ | in the config of my spa, i can see ("VoIP-To-PSTN Gateway Setup ") |
| 23:10.27 | Godfather_ | and inside this, "VoIP-To-PSTN Gateway Enable" |
| 23:10.59 | Godfather_ | VoIP Caller Auth Method:"none", but no idea to tell in extensions.conf how to call trought it |
| 23:11.05 | Godfather_ | sorry for the bold |
| 23:11.06 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 23:11.08 | p3nguin | Do you want people to be able to use the device to make landline calls into the device and then route them over Asterisk? |
| 23:11.54 | p3nguin | And make SIP calls to the device and route them out the PSTN? |
| 23:12.32 | Godfather_ | i want people (the users that are registered in my sip.conf) to make calls to the pstnline |
| 23:13.13 | Godfather_ | for example, i want to dial +3466666666(with my voipphone,softphone..) and go it thorught the pstnlike |
| 23:13.22 | *** join/#asterisk _matt (n=matt@2001:470:9002:1:20b:cdff:fe04:843a) |
| 23:14.23 | Godfather_ | i mean, without the spa the sip.conf users can't call to +346666666, i want those user are been able to call with the spa3102 |
| 23:17.54 | Godfather_ | i think i have to create a user for outgoing calls and other for incoming calls |
| 23:18.03 | Godfather_ | or somethink like that |
| 23:18.18 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:19.30 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 23:21.45 | dan__t | yo jblack, sorry about that - thanks for answering. |
| 23:22.00 | dan__t | I had to run off for a bit. |
| 23:22.06 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 23:24.17 | dan__t | I'd be using AGI to pull external data and things like that. With any luck I hoped to store said data as variables in Asterisk |
| 23:25.07 | *** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-17-198-no168.tbcn.telia.com) |
| 23:27.58 | *** join/#asterisk Whtsup (n=sssi@WimaxUser3715-136.wateen.net) |
| 23:28.18 | Whtsup | my pdd is 15 sec can i low my pdd rate |
| 23:28.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:29.12 | Whtsup | anyone alive |
| 23:29.46 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 23:30.17 | hardwire | Only us trolls. |
| 23:31.05 | Whtsup | can i low my post dial delay |
| 23:31.12 | Whtsup | current is 15 sec |
| 23:34.06 | hardwire | whats your real name Whtsup? |
| 23:35.05 | hardwire | doesn't matter.. you talk just like a friend of mine does. |
| 23:35.13 | hardwire | Whtsup: calls going out over sip/iax/zap? |
| 23:35.21 | Whtsup | zap |
| 23:35.27 | Whtsup | sorry dahdi :) |
| 23:35.57 | hardwire | no big deal |
| 23:36.06 | hardwire | is it PRI or POTS? |
| 23:36.06 | Whtsup | how can i low |
| 23:36.12 | Whtsup | post |
| 23:36.17 | Whtsup | fxo card |
| 23:36.20 | Whtsup | 8 channel |
| 23:36.37 | hardwire | plug a telephone into the POTS line and dial.. then measure the delay. |
| 23:36.57 | hardwire | see if that's the problem first. |
| 23:37.00 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1242514871.dsl.bell.ca) |
| 23:37.14 | darkdrgn2k | hey , how do i register a Cisco 7940 (sip firmware) with asterisk? |
| 23:37.18 | darkdrgn2k | (i cant receive incomming calls) |
| 23:37.20 | Whtsup | i have check |
| 23:37.23 | Whtsup | 6 sec |
| 23:37.44 | hardwire | Whtsup: 6 * 2 = 12 + fudge of 2 seconds = 15 |
| 23:37.48 | hardwire | are you dialing in over pots as well? |
| 23:38.01 | hardwire | or are you using a sip phone that dials quickly? |
| 23:38.58 | hardwire | I had a client that was dealing with that.. we switched them to PRI to reduce dial time since they didn't anticipate dialing twice. |
| 23:39.09 | hardwire | once for the inbound, once for the outbound |
| 23:39.35 | Whtsup | well i m doing termination |
| 23:39.41 | hardwire | can you describe your setup? |
| 23:39.51 | hardwire | the prettier the picture the better. |
| 23:40.01 | Whtsup | simple is that |
| 23:40.17 | Whtsup | when i dial from my sip phone to my pstn pot it takes 15 sec |
| 23:40.36 | Whtsup | when i dial directly from my pstn line it take 5 to 6 sec |
| 23:40.43 | hardwire | can you describe your setup a little more? |
| 23:41.01 | Whtsup | simple is that |
| 23:41.18 | Whtsup | my setup is on local network |
| 23:41.51 | hardwire | unless I know how you've configured things and routed the call.. I can't help much. |
| 23:41.52 | Whtsup | when i dial from softphone pdd is 15 sec |
| 23:42.07 | hardwire | I can suggest you run originate from the asterisk cli and dial out directly from asterisk and measure the time that takes. |
| 23:42.24 | Whtsup | yes |
| 23:42.28 | hardwire | 'originate Zap/g1/localnumber extension 600@demo' |
| 23:42.31 | Whtsup | 15 sec |
| 23:43.03 | hardwire | yes what? |
| 23:43.18 | Whtsup | wait |
| 23:43.22 | Whtsup | i will show u my dial plan |
| 23:44.02 | Whtsup | exten => _4492X.,1,NoCDR() |
| 23:44.03 | Whtsup | exten => _4492X.,2,Wait(1) |
| 23:44.03 | Whtsup | exten => _4492X.,3,Dial(Dahdi/r0/0${EXTEN:4},60,) |
| 23:44.08 | Whtsup | this is my dial plan |
| 23:44.23 | *** join/#asterisk acxty (n=acxty@190.6.194.152) |
| 23:44.29 | hardwire | how is your dahdi channel set up? |
| 23:44.46 | Whtsup | setup means ? |
| 23:44.51 | hardwire | configured. |
| 23:44.54 | Katty | >.< |
| 23:45.31 | Whtsup | i m using elastic |
| 23:45.36 | Whtsup | elastix |
| 23:45.40 | hardwire | that doesn't really answer my question |
| 23:45.41 | Whtsup | default installation |
| 23:45.50 | Whtsup | i didnt got ur question |
| 23:45.52 | Whtsup | :) |
| 23:47.03 | Whtsup | signalling=fxs_ks |
| 23:48.53 | Whtsup | anything shud i change in my dialplan |
| 23:49.02 | Whtsup | hardwire can u suggest anything |
| 23:49.13 | *** join/#asterisk Sier (n=sier@unaffiliated/sier) |
| 23:49.24 | hardwire | honestly.. no.. I can't because I don't know damn near enough about your setup. |
| 23:49.38 | hardwire | I offered up the 'originate' test.. didn't get feedback |
| 23:49.56 | hardwire | did you bother testing how quickly your sip phone connects to asterisk by having it call a local application? |
| 23:50.17 | hardwire | what do the phone logs say about when the dial happens and when the connect happens |
| 23:50.42 | Whtsup | as soon i can click dial |
| 23:50.48 | Whtsup | call goes to asterisk |
| 23:51.35 | Whtsup | i have check there is no delay between my softphone and asterisk |
| 23:51.36 | hardwire | now check the logs |
| 23:53.39 | Amorsen | It just sounds like DAHDI waits a long time for a dial tone. Such is the nature of analog lines |
| 23:54.13 | Whtsup | hm |
| 23:54.13 | hardwire | Amorsen: It's either that or the DTMF is tooooo long. |
| 23:54.27 | Amorsen | There is that. Put an analog phone in parallel on the line and listen in |
| 23:54.39 | hardwire | Whtsup: I'm wondering if its 6sec + dial time and dial time is 6 - 7 seconds. |
| 23:54.54 | hardwire | Amorsen: or use the monitor tool |
| 23:55.15 | Amorsen | That's so fancy |
| 23:55.18 | hardwire | heh |
| 23:55.22 | hardwire | or he could run monitor |
| 23:55.27 | hardwire | err.. Monitor() |
| 23:55.33 | hardwire | instead of the dahdi monitor |
| 23:55.42 | hardwire | just make sure it records before channel bridging |
| 23:56.16 | hardwire | Whtsup: plug in a phone.. start a timer.. dial the number at a reasonable rate and wait till you get a connect then shut the timer off. |
| 23:56.22 | Whtsup | hmm |
| 23:56.35 | Whtsup | ok |
| 23:58.35 | *** join/#asterisk Circlefusion (n=circlefu@96-28-115-69.dhcp.insightbb.com) |
| 23:59.18 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
