| 00:01.13 | sbingner | rudholm, actually it's a bit flaky with 1 ;) |
| 00:02.15 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 00:02.23 | rudholm | sbingner: yeah, I'm going to have to experiment with different arrangements, but since most of my phones are vintage, CID isn't really an issue (I have two talking CID boxes, CID on my cordless, and CID on my Squeezeboxes, which is plenty) |
| 00:03.39 | sbingner | rudholm, yea... 20 seemed pretty reliable for my phone ;) |
| 00:04.19 | rudholm | sbingner: did the phone ring? |
| 00:04.28 | rudholm | sbingner: I imagine some electronic ringers could react that fast |
| 00:06.50 | sbingner | rudholm, nope it didn't |
| 00:07.16 | sbingner | I think may 'understand' that means don't really ring or something |
| 00:07.53 | sbingner | when I had larger values (like 1000), I got a full first ring then a short ring thereafter |
| 00:09.13 | sbingner | 50 even worked for no ring |
| 00:09.26 | _VoicemeUpDotCom | 8004444444|45|g |
| 00:09.34 | _VoicemeUpDotCom | dialign this.. but its not continuing after.. |
| 00:09.42 | _VoicemeUpDotCom | i think along the lines theres an hangup .. maybe |
| 00:10.52 | sbingner | _VoicemeUpDotCom, the originating channel didn't hang up did it? |
| 00:11.19 | _VoicemeUpDotCom | yes |
| 00:11.29 | _VoicemeUpDotCom | not term.. but orig did.. or me in this case |
| 00:11.37 | _VoicemeUpDotCom | any test number i can use that wont work ? |
| 00:11.42 | _VoicemeUpDotCom | i tried 9994444444 |
| 00:11.49 | _VoicemeUpDotCom | but seems cough.. its being routed |
| 00:14.04 | sbingner | 'g' -- goes on in context if the destination channel hangs up <-- would be kinda pointless to keep going if the source channel wasn't there anymore ;) |
| 00:14.37 | sbingner | try calling your cell phone and hang up on yourself *shrug* |
| 00:14.45 | *** join/#asterisk Jon335 (i=jon335@unaffiliated/jon335) |
| 00:15.03 | Jon335 | anyone here played with app_cepstral or app_swift? |
| 00:15.20 | Jon335 | neither can detect any of my voices |
| 00:16.07 | _VoicemeUpDotCom | mmm ok |
| 00:16.42 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:16.51 | nybble | back, once again i am |
| 00:17.15 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 00:20.43 | rudholm | sbingner: yeah, I'll experiment with different durations to see what works for me |
| 00:22.37 | _VoicemeUpDotCom | hmm i oculd use hangup then |
| 00:22.54 | _VoicemeUpDotCom | but i cant change the contex tim in.. and surely cant do that |
| 00:27.07 | fab5freddy | i can't get voice mails to pickup |
| 00:27.22 | fab5freddy | in extensions.conf i have |
| 00:27.37 | fab5freddy | exten => 1234,1,Dial(SIP/user,30) |
| 00:27.46 | fab5freddy | but after 30 seconds it don't stop ringing |
| 00:28.17 | fab5freddy | exten => 1234,2,VoiceMail(777@context) |
| 00:28.32 | fab5freddy | is sip.conf i have mailbox=777@context |
| 00:28.41 | polerin | meh. This is probably not an issue that people are likely to want to pick up and talk about in here, but I'm going to ask anyway :P Anyone successfully unlocked a vonage router then re-registered with vonage? The longterm aim is to get the second port registerd via SIP to my * test box so I don't have to use a soft phone |
| 00:28.55 | fab5freddy | in voicemail.conf i have [context] 777 => 0001, user, user@email.com |
| 00:29.16 | sbingner | polerin, I dont think you even have to unlock it for that |
| 00:29.36 | aptura | fab send it to pastebin.ca |
| 00:29.43 | polerin | sbingner: oh? |
| 00:29.52 | sbingner | polerin, I think they LET you use the second port |
| 00:29.57 | sbingner | polerin, call them and ask ;) |
| 00:30.12 | sbingner | or google of course |
| 00:30.14 | polerin | I don't see any user configurable stuff at all ;P |
| 00:30.23 | sbingner | what ATA? |
| 00:30.47 | polerin | RT31P2 |
| 00:31.45 | polerin | possibly a firmware update. |
| 00:31.46 | fab5freddy | aptura: http://pastebin.ca/427780 can you tell what my issue is? |
| 00:34.15 | sbingner | polerin, try to find something that will let you find the local device key-- then download your config from vonage and decrypt it, I can't tell you how because I don't remember but there are utilities to do that. Once you have that decrypted you'll know what the admin login is |
| 00:36.01 | polerin | hmm. well I'm pretty sure I can unlock it, but I'm just trying to make sure I can re-authenticate :P |
| 00:36.56 | sbingner | you'll have to either decrypt the config file to get your device ID key or decrypt it to get the username and password.... or point it back at their TFTP which would overwrite your config |
| 00:37.13 | sbingner | there is stuff talking about doing that with the PAP2 -- it may relate |
| 00:37.20 | polerin | nods |
| 00:37.40 | _VoicemeUpDotCom | polerin, |
| 00:37.49 | polerin | what it looks like for unlocking this one is spoofing the auth server, and I can probably work that out. |
| 00:37.59 | _VoicemeUpDotCom | you will need to use the vonage virtual softphone @ 9.99 anyhow |
| 00:38.04 | polerin | no |
| 00:38.11 | _VoicemeUpDotCom | else they block \ |
| 00:38.17 | polerin | i'm not looking to connect asterisk to vonage |
| 00:38.24 | _VoicemeUpDotCom | asterisk to ata |
| 00:38.26 | _VoicemeUpDotCom | same |
| 00:38.33 | _VoicemeUpDotCom | ah |
| 00:38.39 | polerin | I'm actually going to be switching to broadvoice, I'm just wanting to play |
| 00:38.40 | _VoicemeUpDotCom | no luck yet.. |
| 00:39.39 | polerin | though to be honest if/when I got the SIP for reauth of the router, I could probably just stick it in the asterisk anyway :P |
| 00:40.25 | polerin | I've got someone who loves unlocking devices, always makes him giggle insanely :P |
| 00:40.34 | polerin | and he has one of these, and isn't using vonage so doesn't have to worry about it anymore :P |
| 00:41.12 | fab5freddy | http://pastebin.ca/427780 can somebody check this out and tell me my issue, thanks |
| 00:45.21 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 00:46.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:46.54 | pfn | wtf is libtdl |
| 00:47.19 | nextime | Hi. Ther's a way to use junghanns gsm cards ( so i need to use ztgsm ) with asterisk 1.4.2? i don't want to install all bristuff cause i use mISDN for hfc and bri cards |
| 00:47.49 | pfn | configure isn't finding unixODBC because it can't find libtdl... |
| 00:48.34 | pfn | oh, it's ltdl |
| 00:48.57 | pfn | nevermind, libtool |
| 00:49.00 | pfn | stupid centos install... |
| 00:51.19 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 00:53.29 | _VoicemeUpDotCom | exten => _573204XXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 00:53.32 | _VoicemeUpDotCom | sound right ? |
| 00:53.36 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:53.44 | _VoicemeUpDotCom | oh |
| 00:54.08 | _VoicemeUpDotCom | its matching exten => _X.,1 first.. even its its after..wahts the rule agiain ? |
| 00:55.25 | nybble | night all |
| 00:55.31 | _VoicemeUpDotCom | nig |
| 00:56.23 | _VoicemeUpDotCom | ok so its puts the include AFTE the others hmm |
| 00:59.40 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:00.08 | _VoicemeUpDotCom | guess its friday night |
| 01:02.41 | fab5freddy | I am getting "Unable to re-open DSP device |
| 01:02.58 | fab5freddy | i believe is what causing my voice mail not to work |
| 01:04.13 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:04.20 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:04.21 | aptura | module not installed? |
| 01:04.36 | fab5freddy | aptura: how do i install it? |
| 01:04.38 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:06.43 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 01:08.46 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:11.51 | *** join/#asterisk nikko (n=nikko@69.85.203.178) |
| 01:17.44 | pfn | so is ael the preferred dialplan config these days? |
| 01:21.19 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 01:22.51 | *** join/#asterisk AlexCeli (n=alex@190.42.145.41) |
| 01:23.20 | AlexCeli | excuse me... |
| 01:24.01 | AlexCeli | how I can set callerid from a PHP script? I call a PHP from AGI, I wan't to set the callerid num and name |
| 01:24.29 | AlexCeli | I try with |
| 01:24.31 | AlexCeli | echo "Set(CALLERID(name)=\"".$wnombre."\")\n"; |
| 01:24.31 | AlexCeli | echo "Set(CALLERID(num)=\"".$wtelefono."\")\n"; |
| 01:24.36 | AlexCeli | without luck |
| 01:25.33 | AlexCeli | someone can help me? |
| 01:28.10 | *** join/#asterisk Jon335 (i=jon335@unaffiliated/jon335) |
| 01:32.42 | AlexCeli | please |
| 01:32.46 | AlexCeli | nobody can help me? |
| 01:33.36 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 01:34.10 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 01:34.52 | asteriskguy | has anyone here setup an asterisk cluster or realtime configuration yet? |
| 01:36.49 | _VoicemeUpDotCom | is the exten h .. an exten or prio ? |
| 01:41.05 | *** join/#asterisk cyanid3 (n=jhowe@24-180-124-56.dhcp.aldl.mi.charter.com) |
| 01:42.33 | AlexCeli | <AlexCeli> excuse me... |
| 01:42.33 | AlexCeli | <AlexCeli> how I can set callerid from a PHP script? I call a PHP from AGI, I wan't to set the callerid num and name |
| 01:42.33 | AlexCeli | <AlexCeli> I try with |
| 01:42.34 | AlexCeli | <AlexCeli> echo "Set(CALLERID(name)=\"".$wnombre."\")\n"; |
| 01:42.34 | AlexCeli | <AlexCeli> echo "Set(CALLERID(num)=\"".$wtelefono."\")\n"; |
| 01:42.35 | AlexCeli | <AlexCeli> without luck |
| 01:42.37 | AlexCeli | <AlexCeli> someone can help me? |
| 01:47.13 | _VoicemeUpDotCom | if i include exten => h,1, form another include can i have 2 ? |
| 01:47.23 | _VoicemeUpDotCom | can h be sisolated by context |
| 01:55.15 | BSD_Tech | the callerid is set fron sip or iax.conf |
| 01:55.46 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 01:59.51 | ManxPower | _VoicemeUpDotCom: in my experience lettered extensions are always lower in priority than a lettered exten in the context. |
| 02:02.43 | _VoicemeUpDotCom | kk |
| 02:03.27 | _VoicemeUpDotCom | hmm to include a fiel its include => file.conf ? |
| 02:04.56 | cyanid3 | I'm having some problms getting incoming calls to come in using the FXO channel. Things work when I type in extension 7777. Here's my config files: http://www.pastebin.ca/427867 |
| 02:05.43 | cyanid3 | This is using an analog terminal adapter. Is there something that I'm missing? |
| 02:09.20 | _VoicemeUpDotCom | ohehe well to include a file tis waht again |
| 02:09.27 | _VoicemeUpDotCom | let me ~google |
| 02:12.12 | tzafrir_laptop | anybody here tried using zapscan (the utility from the asterisk-gui, not the asterisk application) to detect hardware with a TDM400P card in which not all ports had modules? |
| 02:20.40 | *** join/#asterisk rpm (n=russell@S010600111155e117.vc.shawcable.net) |
| 02:23.08 | AlexCeli | <AlexCeli> excuse me... |
| 02:23.08 | AlexCeli | <AlexCeli> how I can set callerid from a PHP script? I call a PHP from AGI, I wan't to set the callerid num and name |
| 02:23.08 | AlexCeli | <AlexCeli> I try with |
| 02:23.08 | AlexCeli | <AlexCeli> echo "Set(CALLERID(name)=\"".$wnombre."\")\n"; |
| 02:23.08 | AlexCeli | <AlexCeli> echo "Set(CALLERID(num)=\"".$wtelefono."\")\n"; |
| 02:23.09 | AlexCeli | <AlexCeli> without luck |
| 02:23.11 | AlexCeli | <AlexCeli> someone can help me? |
| 02:24.24 | wunderkin | ~wiki |
| 02:24.28 | wunderkin | ugh |
| 02:32.01 | *** join/#asterisk Caplain (n=Shayne@c-68-60-62-95.hsd1.mi.comcast.net) |
| 02:32.47 | Caplain | i seem to have an inbound call issue with nufone |
| 02:32.57 | *** join/#asterisk viking78 (i=viking78@66-168-102-94.dhcp.jcsn.tn.charter.com) |
| 02:33.09 | Caplain | debug shows something comming on sip but it doesnt go through |
| 02:37.38 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 02:38.16 | ManxPower | AlexCeli: do not put the callerid info in quotes |
| 02:41.05 | polerin | Apr 6 21:27:16 WARNING[2663]: chan_iax2.c:7126 socket_read: Call rejected by 192.246.69.186: Unable to negotiate codec |
| 02:41.27 | polerin | making outbound iax2 call through fwd |
| 02:46.46 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
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| 02:48.12 | *** part/#asterisk nikko (n=nikko@69.85.203.178) |
| 02:48.17 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
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| 02:49.36 | polerin | mmph. bed time |
| 02:49.38 | polerin | night all |
| 02:58.14 | *** part/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 03:06.59 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
| 03:07.31 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 03:07.44 | jets | Juggie! |
| 03:07.56 | jets | Qwell! |
| 03:08.00 | jets | russelb! |
| 03:08.55 | Juggie | jets! |
| 03:09.10 | Innatech | Innatech! |
| 03:09.13 | Innatech | oh....nevermind. |
| 03:10.20 | jets | you should. I'm at home... |
| 03:10.33 | jets | that doesn't say much i kind of played hooky today |
| 03:11.52 | Innatech | I'm waiting for a manual to finish printing, so I can lug it around all weekend and pretend I'm going to read it before monday. |
| 03:13.13 | *** join/#asterisk Caplain (n=Shayne@c-68-60-62-95.hsd1.mi.comcast.net) |
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| 03:26.29 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:32.16 | *** join/#asterisk Dandan (i=dandan@ip68-9-233-149.ri.ri.cox.net) |
| 03:32.20 | Dandan | hello all :) |
| 03:32.30 | Dandan | anyone willing to sell s100u here? |
| 03:32.45 | Dandan | or borrow for a month? |
| 03:32.48 | [TK]D-Fender | EWWWW why would you want one of those? |
| 03:32.57 | Dandan | Hey D-Fender :) |
| 03:33.05 | Dandan | I am trying to interface it with my slug - nslu2 |
| 03:33.24 | Dandan | since ztdummy is not an option as a timing source |
| 03:33.36 | Dandan | Zaptel Version: 1.2.11 Echo Canceller: MG2 |
| 03:33.36 | Dandan | ztdummy: This module requires the kernel HZ setting to be 1000 ticks per second |
| 03:33.47 | Dandan | and can't be loaded :-/ |
| 03:34.04 | [TK]D-Fender | Dandan, Guess what... the S100U isnt' a valid timing source. |
| 03:34.14 | Dandan | oh sh*t... |
| 03:34.14 | [TK]D-Fender | Dandan, it doesn't generate steady IRQ's |
| 03:34.29 | Dandan | relly? |
| 03:34.32 | Dandan | Hm :( |
| 03:34.46 | [TK]D-Fender | Dandan, its a dumb USB device... don't expect miracles... if the USB port its on isn't enough you're dead meat regardless |
| 03:34.59 | Dandan | any available and reliable replacements? |
| 03:35.19 | Dandan | (voip-info is surprisingly numb about it) |
| 03:35.20 | [TK]D-Fender | Dandan, Yes, you can get all sorts of PC's to replace your slug with that will do the job just fine ;) |
| 03:35.39 | Dandan | :-P yeah, but it IS NOT THE POINT :) |
| 03:36.02 | Dandan | I would like to try, to play, to experiment |
| 03:36.24 | Dandan | don't you love 266 MHz, 32 MB of RAM and debian on it? :) |
| 03:36.29 | [TK]D-Fender | Dandan, try that street corner 3 blocks down ;) |
| 03:36.59 | Dandan | hm... |
| 03:37.07 | Dandan | good advice, thanks |
| 03:37.11 | Juggie | soo totally offtopic, go see grindhouse. |
| 03:37.13 | Juggie | its awesome :) |
| 03:37.56 | Dandan | lol :) |
| 03:38.00 | Dandan | i just saw 300 |
| 03:38.04 | Dandan | enough gore for today |
| 03:40.00 | [TK]D-Fender | 300 looks too damn much like a video game.... |
| 03:40.41 | Dandan | it is watchable... |
| 03:41.06 | [TK]D-Fender | And of course the usual villification of the antagonist and grandiose gestures of the leads. cheap trash, expensively made and with B&W flair that tries too damn hard. |
| 03:41.40 | Dandan | that is a critique that I haven't heard in a looong time |
| 03:41.47 | [TK]D-Fender | Dandan, Many things are "watchable", but I don't want to be feeling the 9$ was a waste and start searching for a spoon to pluck my eyes out with after. |
| 03:41.58 | Dandan | it is based on a comic book! what do you expect? |
| 03:42.22 | coppice | they had weird posters here for 300. they made it look like a voodoo movie |
| 03:42.32 | [TK]D-Fender | Dandan, everything I just stated :) I *MAY* watch it once my frined downloads/encodes it :) |
| 03:42.41 | Dandan | [TK]D-Fender: that's why I spend my money wisely, on 'watchable' stuff I am trying not to spend a dime if there are other means of watching the movie |
| 03:43.33 | Dandan | coppice: lol :) |
| 03:45.21 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 03:45.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:46.55 | Juggie | MOG! |
| 03:47.06 | Dandan | hm, |
| 03:48.57 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 03:49.03 | PakiPenguin | hello |
| 03:49.12 | mog | Juggie, |
| 03:49.16 | Dandan | hey :) |
| 03:49.19 | PakiPenguin | is it possible to set disa password from mysql? |
| 03:50.44 | [TK]D-Fender | PakiPenguin, Use Authenticate instead and look it up in the dialplan' |
| 03:50.53 | PakiPenguin | ah |
| 03:51.50 | Dandan | Heh, I do not know d-fender if you remember me, but since the last time I came here, trying to configure sipura - I moved on - now I am working in an environment of 24 servers and 600 users in 11 different locations - sweet asterisk 1.0.* and 0.* hideout! |
| 03:52.08 | Dandan | s/sipura/sangoma/ |
| 03:52.12 | PakiPenguin | :) |
| 03:52.44 | Dandan | jbot: lol :) RegEx rullz! |
| 03:53.11 | *** join/#asterisk bmg505 (n=leon@196.209.179.147) |
| 03:53.15 | wunderkin | ~amiadog? |
| 03:53.25 | jets | ~jets |
| 03:53.46 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 03:53.49 | JoelSolanki | Hi all |
| 03:54.02 | wunderkin | i think jbot is being retarted again |
| 03:54.02 | JoelSolanki | is there any setting for early media in asterisk sip configuration ? |
| 03:54.16 | JoelSolanki | is the canreinvite stuff ? |
| 03:54.17 | wunderkin | retarded .. or something.. yeah.. |
| 03:55.12 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 03:56.01 | JoelSolanki | ? |
| 03:56.20 | JoelSolanki | anybody can give me little idea on early media |
| 04:01.53 | Dandan | hmmm |
| 04:02.11 | Dandan | looks like you can live without ztdummy and still have iax2 and MOH... |
| 04:03.30 | [TK]D-Fender | Dandan, you never needed zaptel for MoH, and only needed it for IAX2 if you were planning on TRUNKING with it. |
| 04:03.43 | [TK]D-Fender | Dandan, So whats the deal with Sangoma? |
| 04:03.57 | Dandan | Sangoma works, as far as I remember, at the other company |
| 04:04.02 | Dandan | that I no longer work for... |
| 04:04.28 | [TK]D-Fender | Dandan, I meant more "what is your issue with them NOW"? |
| 04:04.36 | Dandan | now I work for the company that has many more asterisk installations but are all 1.0.* and I would like to start planning to upgrade... |
| 04:04.50 | Dandan | there is no issue, NOW I can install and configure with ease |
| 04:05.03 | Dandan | thanks to this channel too :) |
| 04:05.50 | [TK]D-Fender | Dandan, Ok, thought you had a question buried in there somewhere... silly me! |
| 04:05.57 | *** join/#asterisk AzianFlu (n=Yamabush@arcane/supporter/yamabushi) |
| 04:06.33 | Dandan | nah, can somebody share from the top of his/her head the main advantages of 1.4? (or major selling points?) |
| 04:06.44 | Dandan | I do not want to go through Changelog... |
| 04:06.54 | [TK]D-Fender | Dandan, what do you feel you're MISSING? |
| 04:07.49 | Dandan | hm, voicemail issues - the light comes up whtn the message is being left, there are not voicemail groups... and round robin agents - there is no first-second-third, etc... |
| 04:07.52 | Dandan | it is always RR |
| 04:08.08 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 04:08.57 | [TK]D-Fender | Dandan, I'd suggest you recode for 1.2 compliance, and migrate their first ensuring that your settings are 1.4 compliant as well and await a better reason to just ight to 1.4 |
| 04:09.22 | Dandan | Hm, that will be tough |
| 04:09.51 | Dandan | somebody created his own mysql interface, absolutely crazy manager.pl scripts and many many pitfalls that I do not want to dig into... |
| 04:09.54 | Dandan | just yet... |
| 04:10.16 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 04:10.17 | Dandan | but your avice seems valuable |
| 04:10.47 | [TK]D-Fender | jump right into* |
| 04:11.06 | Dandan | right :) |
| 04:15.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:17.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:18.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:19.27 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:19.45 | Qwell | bkruse_home: !!! |
| 04:20.12 | bkruse_home | Qwell!! |
| 04:20.23 | bkruse_home | qwell that was awesome |
| 04:20.28 | Qwell | that was |
| 04:20.33 | bkruse_home | you have a very pretty and sweet wife btw |
| 04:20.37 | Qwell | :D |
| 04:20.45 | bkruse_home | shes super cool |
| 04:21.14 | coppice | they're all sweet, until you get them home :-) |
| 04:21.21 | Qwell | haha |
| 04:21.27 | Qwell | coppice: You must be married ;) |
| 04:21.40 | Qwell | and by the sounds of it, for several years ;) |
| 04:22.02 | bkruse_home | haha |
| 04:22.20 | AzianFlu | lol |
| 04:22.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:23.08 | viking78 | I love my wife of almost 10yrs, she's much more reliable than my internet connection |
| 04:23.41 | wunderkin | must be better than drako's |
| 04:23.53 | coppice | that might say a lot about your internet connection |
| 04:24.02 | drako | wunderkin, lol |
| 04:24.14 | bkruse_home | coppice: score! |
| 04:24.22 | bkruse_home | *high five* |
| 04:24.41 | coppice | i really really love my wife of over 10 years, but she doesn't always do her best to make that easy |
| 04:24.55 | [TK]D-Fender | viking78, Got a hot pic of your internet connection to share? :) |
| 04:24.58 | AzianFlu | I love my wife of over 11 years, she makes it easy to go to work every day.... ;) |
| 04:25.03 | drako | women never make anything easy. |
| 04:25.19 | Qwell | drako: I think AzianFlu just proved that wrong :) |
| 04:25.19 | coppice | AzianFlu: nice |
| 04:25.36 | drako | lol |
| 04:25.56 | [TK]D-Fender | qwell : ok, fine.. care to share one of your wifeas well? Perhaps a side-by-side shoot? |
| 04:26.31 | [TK]D-Fender | ~[TK]D-Fender |
| 04:26.39 | jbot | it has been said that [tk]d-fender is rockin' the casbah !!! |
| 04:26.39 | Qwell | hey, I'm on Comcast - I doubt you wanna see that |
| 04:26.45 | Qwell | s/jbot/viking78/ |
| 04:26.54 | [TK]D-Fender | viking78, More like "recently ready to move on" |
| 04:27.22 | coppice | [TK]D-Fender: really dusty guitar? :-\ |
| 04:27.34 | [TK]D-Fender | viking78, I had to drop a really hot chick I got involved with too fast following a LTR breakup. Rebound issues... |
| 04:27.41 | [TK]D-Fender | coppice, funny :) |
| 04:27.46 | [TK]D-Fender | coppice, NO ;) |
| 04:28.02 | bkruse_home | Qwell: your jbot skills never fall short of amazing |
| 04:30.52 | coppice | wifey has spent the last 3 days in china. I have serious problems sleeping without snuggling up to her. thankfully she's on her way home :-) |
| 04:32.04 | bkruse_home | res_wife.so? |
| 04:32.25 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
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| 04:34.46 | Dandan | :) |
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| 05:21.11 | bkruse_home | mog: there? |
| 05:25.01 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 05:26.33 | nick125_lappy | Hmm, I've been trying to implement sipbroker with my asterisk box, however, if I try to send a call to the PSTN normally, it just seems to sit there trying to push the call to sipbroker, any ideas? |
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| 06:03.31 | nick125_lappy | Slightly off-topic, but, does anyone here know how to "force" e164.org to do the verification call? |
| 06:18.48 | *** join/#asterisk Matrix9 (i=MiniMe@s142-179-197-109.ab.hsia.telus.net) |
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| 08:08.46 | mDuff | coppice: my wife is getting back tomorrow from 2 months in Dallas. I feel your pain. |
| 08:11.00 | mDuff | coppice: I had a coworker going through there on business pick up the chihuahuas a few weeks back -- it was bad enough sleeping without her; not even having the dogs snuggling up made it that much worse. |
| 08:11.28 | mDuff | nick125_lappy: I have no experience with or knowledge of sipbroker -- but how are you deciding which calls to send out a given trunk? |
| 08:13.16 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:15.51 | *** join/#asterisk svenna_ (n=svenna@p548d3ee1.dip0.t-ipconnect.de) |
| 08:27.58 | ecam | i need desperate help! |
| 08:28.09 | ecam | i cant seem to get my SIP trunk working |
| 08:29.22 | ecam | <PROTECTED> |
| 08:31.00 | ecam | okay, after a while it registers |
| 08:31.35 | ecam | now i can make a external call through the trunk line to a asterisk extension |
| 08:31.41 | ecam | but all it does is ring |
| 08:32.02 | ecam | when i try to pick up the call, i get a beep and the call disconnects |
| 08:32.55 | ecam | when i try to call an external number through the trunk from an asterisk extension, asterisk tells me that the system is busy |
| 08:36.17 | ecam | <PROTECTED> |
| 08:36.24 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
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| 08:45.57 | tzafrir_laptop | ecam, what phones are they? pastebin some relevant information (sip.conf , cli trace, etc.) |
| 08:45.59 | tzafrir_laptop | ~pb |
| 08:46.11 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 08:46.47 | tzafrir_laptop | ecam, generally, try to convince us that you configured your system correctly ;-) |
| 08:48.42 | ecam | hmm |
| 08:49.11 | CrashHD | anyone have problems with sip adaptive jb? |
| 08:49.17 | CrashHD | pops and clicks |
| 08:49.17 | ecam | <PROTECTED> |
| 08:52.07 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 08:53.26 | ecam | here's my sip.conf |
| 08:53.28 | ecam | http://pastebin.ca/428120 |
| 08:53.55 | ecam | <PROTECTED> |
| 08:55.52 | ecam | this is what gets logged when i make a call from an external line through the trunk to asterisk |
| 08:55.54 | ecam | http://pastebin.ca/428121 |
| 08:55.55 | tzafrir_laptop | the phone registers, right? |
| 08:56.02 | ecam | the phones register |
| 08:56.09 | ecam | internal calling works fine |
| 08:56.16 | ecam | the trunk's the prob |
| 08:58.20 | tzafrir_laptop | what do you mean by "not working"? Do you get any error? |
| 08:58.34 | ecam | <PROTECTED> |
| 08:58.56 | ecam | the external phone gets disconnected |
| 08:59.00 | ecam | no ringing tone |
| 08:59.07 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 08:59.19 | ecam | and the only thing that i see change on asterisk's logs is http://pastebin.ca/428121 |
| 08:59.34 | ecam | nothing else happens, none of the internal phones ring |
| 09:03.40 | ecam | <PROTECTED> |
| 09:12.41 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
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| 09:19.08 | ecam | hello? |
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| 09:38.04 | *** join/#asterisk JanJoost (n=JanJoost@janjoost.xs4all.nl) |
| 09:38.10 | JanJoost | hi all |
| 09:38.32 | JanJoost | I've got a small question regarding asterisk, someone here willing to lend me their brain for a few minutes? |
| 09:38.56 | JanJoost | idea is: I have an ISDN connection, and a few telephone numbers associated with that connection |
| 09:39.05 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 09:39.31 | JanJoost | I can't seem to find how I can lead a specific external phone number to an internal number |
| 09:39.47 | JanJoost | is that possible at all? |
| 09:40.22 | JanJoost | (i.e.: external number 001 goes to internal number 1, external number 002 goes to internal number 2 etc) |
| 09:40.55 | florz | JanJoost: You do know what the dialplan is? |
| 09:41.06 | JanJoost | florz: it gets works |
| 09:41.08 | JanJoost | err |
| 09:41.15 | JanJoost | worse: I have a fully working setup |
| 09:41.24 | JanJoost | some background: i've got everything to work |
| 09:41.33 | JanJoost | but based around one external phone number |
| 09:41.44 | JanJoost | and now that I have hooked up my nokia e70 to my asterisk |
| 09:42.12 | JanJoost | I'd like to use one of the so called secondary numbers (a second public phone number, routed to my isdn connection) |
| 09:42.23 | JanJoost | to call through to my nokia e70 over sip |
| 09:42.29 | florz | JanJoost: Well, if you do know what the extensions.conf looks like, do the obvious =:-) |
| 09:42.31 | JanJoost | but I can't seem to find the magic :) |
| 09:42.55 | JanJoost | florz: I know how to process the external calling party number |
| 09:43.08 | JanJoost | but I can't find the magic associated with 'number that has been called' |
| 09:43.19 | florz | JanJoost: From asterisk's point of view, there is no real difference between an "internal" and an "external" number ... |
| 09:43.57 | florz | JanJoost: There is just "the number that has been dialed by the incoming channel/call". |
| 09:44.06 | JanJoost | florz: I half-and-half expected to do some magic in modem.conf, something saying: 'msn1=incomingnumber1, msn2=incomingnumber2' etc |
| 09:44.23 | JanJoost | mmm |
| 09:44.34 | florz | JanJoost: Ohh, you are using i4l? |
| 09:44.39 | JanJoost | florz: yes |
| 09:44.58 | JanJoost | oh, sorry, that probably was significant :) |
| 09:45.11 | florz | JanJoost: Hmm. Does it really on one MSN only? |
| 09:45.21 | florz | s/on/react on/ |
| 09:45.39 | JanJoost | florz: it reacts on all msns |
| 09:45.41 | ecam | <PROTECTED> |
| 09:46.04 | JanJoost | but I can't get it to do some magic based on the msn used |
| 09:46.05 | *** join/#asterisk Caplain (n=Shayne@c-68-60-62-95.hsd1.mi.comcast.net) |
| 09:46.24 | JanJoost | at least: I don't know how |
| 09:47.07 | florz | JanJoost: You do know how to "do some magic based on the number called by a SIP client"? |
| 09:47.28 | JanJoost | yup |
| 09:47.42 | JanJoost | I do have the feeling I'm missing something blatantly obvious here |
| 09:47.57 | florz | JanJoost: Well, basically I said it already =:-) |
| 09:47.58 | JanJoost | and I am going to feel really really stupid once someone enlightens me :) |
| 09:48.18 | JanJoost | heh |
| 09:48.20 | JanJoost | ok |
| 09:48.34 | JanJoost | I was thinking the wrong way around, I guess :) |
| 09:48.38 | florz | JanJoost: For asterisk, the MSN is just "the number the ISDN line dialed into your asterisk" :-) |
| 09:49.40 | JanJoost | florz: that I can handle, I expected to have to assign that to some variable |
| 09:49.44 | JanJoost | in order to use it later on |
| 09:49.51 | JanJoost | but Life Gets Better |
| 09:50.06 | JanJoost | and I shouldn't want to do complicated things, I suppose :) |
| 09:51.50 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 09:56.57 | JanJoost | ok, thanks! |
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| 10:25.13 | *** part/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 10:25.46 | ecam | anyone knows why my calls are disconnecting? |
| 10:26.23 | *** join/#asterisk friedrich| (n=friedric@e177242023.adsl.alicedsl.de) |
| 10:26.35 | ecam | i make a call from an external line, the internal phone rings |
| 10:26.50 | ecam | i pick up the phone, then the call ends |
| 10:30.11 | ecam | the log i get is this |
| 10:30.12 | ecam | http://pastebin.ca/428180 |
| 10:30.35 | ecam | 1234567 is the external number |
| 10:31.16 | ecam | <PROTECTED> |
| 10:31.30 | ecam | <PROTECTED> |
| 10:31.37 | ecam | 5555555 is the external number |
| 10:32.02 | ecam | someone please take a look and reply asap, thanks! |
| 10:44.04 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 10:58.37 | *** join/#asterisk jrenzema (n=renzema@58.252.186.81.lund.res.dyn.perspektivbredband.net) |
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| 11:31.08 | jrenzema | Hi all. I have what is probably a basic problem for the gurus here. I have two inbound sip numbers (...332 an ...336)from the same provider/server. However, regardless of which one I call, the call seems to be appearing on the ...336 number. |
| 11:31.37 | jrenzema | As I hope to handle the calls in different fashions (differnt IVR paths), this is not good. |
| 11:39.28 | tzafrir_laptop | jrenzema, any chance you explicitly set the callerid in the sip user settings? |
| 11:40.09 | ecam | <PROTECTED> |
| 11:40.36 | ecam | <PROTECTED> |
| 11:40.54 | ecam | <PROTECTED> |
| 11:41.20 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 11:41.24 | jrenzema | In sip.conf? I do not have any callerid line defined for the trunk line. |
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| 12:07.06 | polerin | ecam: are you getting any messages from the CLI, and do you have debug and verbosity set higher so you can see them? |
| 12:07.36 | polerin | sorry didn't scroll back far enough |
| 12:09.47 | *** join/#asterisk zapata (n=user@chello213047080026.4.14.vie.surfer.at) |
| 12:32.17 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 12:32.17 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. |
| 12:32.22 | *** join/#asterisk Solaris444 (n=chatzill@203-166-235-97.dyn.iinet.net.au) |
| 12:33.17 | polerin | ecam: no I meen as in in your inbound trunk just have exten=> s,1,Dial(SIP/100) |
| 12:33.51 | polerin | (or whatever the proper syntax is, I'm sleepy and a newb. Just trying to break it down somewhat..) |
| 12:34.13 | polerin | err inbound trunk extention |
| 12:34.17 | ecam | <PROTECTED> |
| 12:35.19 | ecam | <PROTECTED> |
| 12:36.08 | ecam | <PROTECTED> |
| 12:36.13 | polerin | yeah. |
| 12:36.47 | polerin | well you have context=from-internal as your trunk |
| 12:37.08 | ecam | <PROTECTED> |
| 12:37.10 | Solaris444 | hi guys. |
| 12:37.17 | ecam | <PROTECTED> |
| 12:37.20 | Solaris444 | is there a useful web interface for asterisk? |
| 12:37.22 | polerin | I'm looking at old pastebins |
| 12:37.46 | ecam | <PROTECTED> |
| 12:38.02 | polerin | repaistebin your new sip and the relevant contexts from extensions.conf |
| 12:38.04 | Solaris444 | the other thing I was wondering... asterisk seems to want to run as root. |
| 12:38.09 | ecam | <PROTECTED> |
| 12:38.11 | Solaris444 | Any way to make it run as a limited user? |
| 12:38.49 | ecam | <PROTECTED> |
| 12:39.08 | Solaris444 | ok |
| 12:39.10 | ecam | <PROTECTED> |
| 12:39.29 | Solaris444 | do you run it as root? |
| 12:39.39 | ecam | <PROTECTED> |
| 12:40.04 | Solaris444 | oh. you don't consider that bad practise? |
| 12:40.04 | ecam | <PROTECTED> |
| 12:40.08 | Solaris444 | oh right |
| 12:40.16 | ecam | <PROTECTED> |
| 12:41.01 | Solaris444 | how so? |
| 12:41.05 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 12:41.06 | Solaris444 | in a virtual machine? |
| 12:41.09 | ecam | <PROTECTED> |
| 12:41.11 | Solaris444 | ah |
| 12:41.18 | Solaris444 | right. doesn't help me. |
| 12:41.20 | Solaris444 | :/ |
| 12:41.23 | ecam | <PROTECTED> |
| 12:41.42 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 12:43.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:44.45 | Solaris444 | nah it's all good. |
| 12:47.30 | polerin | ecam: I've got to work soon, if nothing else, create a [test-inbound] context and set your inbount trunk to that, and try to have it ring one sip user directly rather than working on groups first |
| 12:47.47 | polerin | if it works then at least you have that to work from |
| 12:47.54 | ecam | <PROTECTED> |
| 12:47.56 | ecam | <PROTECTED> |
| 12:48.03 | ecam | <PROTECTED> |
| 12:48.04 | ecam | <PROTECTED> |
| 12:49.06 | ecam | <PROTECTED> |
| 12:51.09 | polerin | I didn't see it, whats the link? |
| 12:51.48 | ecam | <PROTECTED> |
| 12:51.52 | ecam | <PROTECTED> |
| 12:51.54 | ecam | <PROTECTED> |
| 12:52.05 | ecam | <PROTECTED> |
| 12:52.13 | polerin | oh |
| 12:52.15 | polerin | well |
| 12:52.18 | ecam | <PROTECTED> |
| 12:52.19 | polerin | theres part of your problem :) |
| 12:52.27 | polerin | possibly |
| 12:52.28 | polerin | :P |
| 12:52.32 | ecam | <PROTECTED> |
| 12:52.46 | ecam | <PROTECTED> |
| 12:52.56 | polerin | do you have a [from-trunk] in the extentions? |
| 12:53.12 | ecam | <PROTECTED> |
| 12:53.19 | polerin | that is your dialplan, ie, wtf you are trying to do with the incoming calls |
| 12:53.24 | polerin | what's in it? |
| 12:53.29 | ecam | <PROTECTED> |
| 12:53.32 | ecam | <PROTECTED> |
| 12:53.33 | ecam | include => from-pstn |
| 12:53.33 | ecam | <PROTECTED> |
| 12:53.33 | ecam | [from-pstn] |
| 12:53.33 | ecam | include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations |
| 12:53.34 | ecam | include => ext-did-direct ; MODIFICATOIN (PL) put before ext-did to take precedence |
| 12:53.36 | ecam | include => ext-did |
| 12:53.38 | ecam | include => from-did-direct ; MODIFICATOIN (PL) for findmefollow if enabled, should be bofore ext-local |
| 12:53.41 | ecam | exten => fax,1,Goto(ext-fax,in_fax,1) |
| 12:53.49 | ecam | i don't know what i'm doing >_< |
| 12:54.06 | polerin | heh |
| 12:54.06 | ecam | but all i want is to make incoming calls ring a group of phones |
| 12:54.22 | ecam | and the first to pick up the call gets it |
| 12:54.42 | ecam | you know, department extensions, all phones in the department ring, blah blah |
| 12:54.55 | polerin | ok, try this first, put an exten=> s,1,Dial(SIP/100) ; I think that's the right syntax |
| 12:55.02 | polerin | in your from-trunk |
| 12:55.03 | ecam | in a new context? |
| 12:55.05 | ecam | okay |
| 12:55.28 | ecam | leave everything that's already in my from-trunk intact? |
| 12:55.49 | *** join/#asterisk friedrich| (n=friedric@e177242023.adsl.alicedsl.de) |
| 12:58.08 | polerin | you might try to comment out the include you've got in there, just to isolate that one command |
| 12:58.34 | ecam | okay |
| 13:00.44 | polerin | heh, this is the blink leading the blind really :P but give it a shot. |
| 13:00.50 | ecam | okay |
| 13:00.55 | ecam | so i try calling in? |
| 13:01.21 | polerin | I assume you've saved/relaoded/etc |
| 13:01.23 | polerin | yeah |
| 13:01.39 | ecam | okay, i'm getting that announcement from my telco that my line is not in use |
| 13:02.03 | polerin | heh |
| 13:02.13 | polerin | ok I have to run to work, I'll be back on in about 30 or so |
| 13:02.23 | ecam | lol |
| 13:02.24 | ecam | well |
| 13:02.38 | ecam | http://pastebin.ca/428260 |
| 13:02.51 | ecam | look at lines 13 to 17 |
| 13:02.56 | ecam | when you have the time |
| 13:03.04 | ecam | have a nice day! |
| 13:05.00 | ecam | argh that's it, i'm going to create a new install, try from scratch |
| 13:05.17 | ecam | but i'll keep this one just in case anyone figures out the solution |
| 13:05.42 | polerin | ecam |
| 13:05.45 | polerin | before you do that |
| 13:05.56 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:05.59 | polerin | copy your extentions.conf to a backup |
| 13:06.11 | polerin | then go through chapter.. err.. 5 on the pdf book |
| 13:06.13 | polerin | it'll help |
| 13:06.20 | polerin | I think it |
| 13:06.23 | polerin | I think it's 5 |
| 13:06.27 | ecam | okay |
| 13:08.37 | ecam | oh yeah, you mean the o'reilly book right? |
| 13:08.57 | polerin | I think. It's on the site under documentation |
| 13:09.07 | polerin | the chapter dealing with the extensions.conf |
| 13:09.07 | ecam | okay |
| 13:09.14 | ecam | chapter 5: dialplan basics? |
| 13:12.24 | menfin | for those would be interested this book was translated in french |
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| 13:49.13 | ecam | oh yeah |
| 13:49.47 | ecam | can anyone tell me if its advisable to use asterisk in FC5? |
| 13:49.58 | ecam | or SuSe 10 |
| 13:50.11 | ecam | i.e. distributions with Selinux built in |
| 13:52.43 | *** part/#asterisk moranil (n=moranil@122.162.73.244) |
| 13:53.38 | polerin | ecam: not a clue |
| 13:53.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 13:53.51 | polerin | but regardless, you have to get the dialplan working first |
| 13:54.10 | polerin | or rather, if ou don't get the dialplan working, the rest of it won't either :P |
| 13:55.05 | ecam | well, i'm thinking of starting from scratch |
| 13:55.23 | ecam | so i was considering which os to use too |
| 13:55.35 | ecam | i'm getting a hang of extensions now :) |
| 13:56.29 | polerin | nods |
| 13:57.17 | polerin | I'm using debian right this second? |
| 13:57.32 | polerin | .. without the question. |
| 13:58.07 | ecam | okay... |
| 13:58.09 | ecam | debian |
| 13:58.10 | ecam | hmm |
| 13:58.22 | ecam | i have a limited selection of os to choose from |
| 13:58.34 | ecam | basically the server holding asterisk will have other functions too |
| 13:58.51 | polerin | not neccisarily the best idea, but one I'm having to share |
| 13:58.53 | ecam | as in, it's supposed to be an all-in-one server |
| 13:59.12 | ecam | yeah, dedicated systems are expensive |
| 13:59.53 | ecam | alternatively, i can run xen on it and run multiple OSes virtually |
| 14:00.00 | ecam | but that's inefficient |
| 14:00.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:01.58 | polerin | yeah, it's better to just run it on one os. Still, the issue is that * is processor intensive apparently :/ |
| 14:02.20 | *** part/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:02.28 | ecam | i'm going to keep it simple |
| 14:02.29 | ecam | lol |
| 14:02.34 | ecam | no conference calls and stuff |
| 14:02.46 | ecam | just passing the voice streams |
| 14:03.03 | polerin | well.. if your box has to do transcoding/etc .. yeah |
| 14:03.26 | ecam | nah, no conference calls and no voicemail and stuff means no transcoding |
| 14:03.34 | ecam | as long as i keep a uniform standard |
| 14:04.07 | [TK]D-Fender | transcoding isn't bad unless we're talking G.729. Us "show translation" for the relative load from one to another... |
| 14:04.34 | [TK]D-Fender | voicemail is NO issue. |
| 14:05.41 | ecam | i see |
| 14:06.00 | ecam | speaking of which |
| 14:06.20 | ecam | is there a way to make the console scroll page by page? |
| 14:06.35 | ecam | typing help prints the whole list of commands available |
| 14:07.07 | ecam | and my screen has limited screen estate |
| 14:07.14 | [TK]D-Fender | ecam, user a client that lets you scroll back |
| 14:07.37 | ecam | recommend me one, i'm not using X |
| 14:07.46 | ecam | hmm, screen... |
| 14:08.12 | [TK]D-Fender | ecam, well I usually access mine through PuTTY on a Windows PC.... |
| 14:08.39 | [TK]D-Fender | ecam, if you are only on text-mode Linux, I have no idea what to suggest |
| 14:09.13 | ecam | yeah, i'm currently using text-mode |
| 14:09.31 | ecam | i can get sshd running though, i think i shall do that |
| 14:09.35 | [TK]D-Fender | ecam, and thats all you have? on and PC there? |
| 14:10.07 | [TK]D-Fender | bleh.. can't type today, I'm coming out dyslexic... |
| 14:10.21 | [TK]D-Fender | Only one PC there? |
| 14:10.25 | ecam | lol, yeah, i'm having a little problem understanding you, sorry |
| 14:10.33 | ecam | nope, i'm just using vmware |
| 14:13.27 | ecam | hmm, i have this question |
| 14:13.35 | ecam | so i define these contexts |
| 14:13.50 | ecam | then i set up all my sip clients |
| 14:14.03 | ecam | then assign each client a context |
| 14:14.34 | ecam | so when a user tries to make a call with one of the sip clients, what happens? |
| 14:14.47 | ecam | or do contexts only apply for incoming calls? |
| 14:15.01 | [TK]D-Fender | ecam, correct |
| 14:15.16 | [TK]D-Fender | ecam, (more or less) |
| 14:16.00 | ecam | okay, so only sip clients that receives calls need a context? |
| 14:16.31 | [TK]D-Fender | ecam, no, only those that PLACE calls need a context. |
| 14:16.40 | ecam | ahh... okay |
| 14:16.49 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:17.16 | [TK]D-Fender | ecam, Though why you'd want to leave a SIP device ou there that can only ANSWER calls is beyond me :) |
| 14:17.45 | ecam | well, shouldn't a sip device work both ways? |
| 14:18.17 | ecam | i still need help figuring out how to set up asterisk |
| 14:18.36 | polerin | D-fender: call centers |
| 14:18.41 | ecam | i'm confused as to how to configure |
| 14:18.42 | [TK]D-Fender | ecam, if a phone has no context (or points to an empty one) there there is nothing that they are allowed to DIAL. |
| 14:18.58 | ecam | ohh, okay |
| 14:19.06 | [TK]D-Fender | polerin, even call centers let you call other internal people and services like VM. |
| 14:19.14 | polerin | ecam: think of contexts as rooms, and extentions as things within that room |
| 14:19.22 | polerin | [TK]D-Fender: depends on the call center :P |
| 14:19.31 | [TK]D-Fender | polerin, How would you transfer a call to someone else for better support if you couldn't? :) |
| 14:19.33 | ecam | rooms, things within rooms, got it! |
| 14:19.57 | [TK]D-Fender | polerin, I see where the "pole" comes from.... now have that thing disinfected :D |
| 14:19.59 | ecam | what about trunk lines? are they treated differently? |
| 14:20.09 | polerin | [TK]D-Fender: I've seen a sales call center where there were no calls |
| 14:20.36 | polerin | ... and what is that supposed ot meen :P |
| 14:21.06 | [TK]D-Fender | polerin, I admit that your scenario is possible, but if I can't dial 911 when the guy next to me has a coronary, I've got "issues" with the working envirnment... |
| 14:21.26 | ecam | lol, good one |
| 14:21.42 | [TK]D-Fender | I thought so.... |
| 14:22.04 | [TK]D-Fender | polerin, if this is your office, I've got a lawsuit with your name on it ;) |
| 14:22.11 | polerin | actually it's pol-erin not pole-rin |
| 14:22.13 | polerin | :P |
| 14:22.26 | polerin | and yes, I didn't like that work much :P |
| 14:22.46 | polerin | I'm in a dispatch position now, and I spend way too much time on the phone :P |
| 14:23.10 | polerin | ecam: they still have to start in a room |
| 14:23.33 | ecam | but do calls that come from the trunk be treated as incoming or outgoing calls? |
| 14:23.50 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 14:23.51 | polerin | incoming |
| 14:23.51 | [TK]D-Fender | ecam, Forget the term "trunk" and "lines". You need to seperate TECH from EXTENSIONS.. Extensions.conf sets up the rules of what NUMBERS you are allowed to dial (with fixed, and/or pattern matches). What HAPPENS when you dial a numberyour are allowed to is a COMPLETELY different matter. |
| 14:24.59 | [TK]D-Fender | ecam, I can have : exten => 500,1,Dial(SIP/500) and this will let me dial 500 from my phone... now it just HAPPENS to ring a SIP device (do not call a SIP device an EXTENSION!! this is where you can throw yourself off) |
| 14:26.12 | polerin | ecam: work through that sample dial plan in that chapter, if you haven't |
| 14:26.14 | brian | I want to have a call queue |
| 14:26.15 | [TK]D-Fender | ecam, equally I could have had : exten => 500,1,Dial(ZAP/1/911) and the cops would already be on their way.... or exten => 500,1,VoicemailMain(500@default) and you could be picking up voicemail from box 500. |
| 14:26.50 | ecam | i see... |
| 14:26.50 | brian | But the Asterisk Queue module seems to be a little much for what I want |
| 14:27.08 | ecam | so i guess i could try starting from scratch now |
| 14:27.13 | polerin | [TK]D-Fender: btw, do you have a sec to look at somthing? |
| 14:27.14 | [TK]D-Fender | brian, you haven't been here 1 minute and already you are contradicting yourself :) |
| 14:27.24 | ecam | i think i messed up my previous one too much |
| 14:27.25 | [TK]D-Fender | brian, go get some COFFEE!!!!! |
| 14:27.26 | polerin | ecam: just try from scratch on the extentions, don't bother reinstalling |
| 14:27.43 | brian | I was thinking of writing my own queue in AGI but it seems that it's kind of more complicated than I realised after I thought it through. |
| 14:27.50 | [TK]D-Fender | ecan, yes, trash your crap and start over right! :) |
| 14:27.50 | ecam | okay |
| 14:28.10 | brian | [TK]D-Fender: Well, the Queue module is for when you have multiple "agents" |
| 14:28.20 | brian | [TK]D-Fender: There is only one "agent" always |
| 14:28.20 | ecam | gee, and this is only evaluating |
| 14:28.43 | [TK]D-Fender | ecam, whats to evaluate? it all just works.... |
| 14:29.00 | brian | [TK]D-Fender: I don't want them to login, I want Asterisk to call their mobile phone and then transfer the callers to his channel. |
| 14:29.48 | [TK]D-Fender | brian, is there any "waiting around" in this case? |
| 14:30.03 | brian | [TK]D-Fender: Yes, they will be waiting in a linear queue. |
| 14:30.11 | brian | [TK]D-Fender: But I want the queue to be real simple... |
| 14:30.15 | [TK]D-Fender | brian, otherwise you are just DIALING him which is 1 line of dialplan... |
| 14:30.28 | brian | [TK]D-Fender: I only want it to call him once |
| 14:30.31 | ecam | D-Fender, i gotta get myself prepped for it, I need to set it up with minimal downtime on the server and the phone network |
| 14:30.40 | brian | [TK]D-Fender: It works like this...the application is activated... |
| 14:30.50 | brian | [TK]D-Fender: And then it calls his mobile phone |
| 14:30.55 | [TK]D-Fender | brian, Dial(Zap/g1/1234567890). Whats the problem then? |
| 14:31.16 | [TK]D-Fender | brian, just dial his damned phone! :) |
| 14:31.17 | brian | [TK]D-Fender: There is music on hold and stuff i don't want in the queue module. |
| 14:31.29 | [TK]D-Fender | I jsut told you... DIAL! not QUEUE. |
| 14:31.32 | brian | [TK]D-Fender: You don't get it, there is a queue to talk to this person. |
| 14:31.44 | brian | [TK]D-Fender: Lots of people will call and we need to manage the callers. |
| 14:31.45 | [TK]D-Fender | brian, app_dial.so <- the key to the universe. |
| 14:32.00 | brian | [TK]D-Fender: You don't get it! |
| 14:32.04 | brian | [TK]D-Fender: I need a waiting line! |
| 14:32.18 | [TK]D-Fender | brian you aren't telling me anything more that you jsut want to ring one guy. |
| 14:32.24 | brian | Yes... |
| 14:32.32 | brian | We are setting up a toll free number |
| 14:32.39 | [TK]D-Fender | brian, there are 2 major problems with this.... |
| 14:33.00 | polerin | [TK]D-Fender: he wants to ring the number once, then have the guy answering multiple incoming without re-ringing the cell |
| 14:33.09 | brian | When the application is activated, I want it to call the guy's mobile phone and place him in the channel. |
| 14:33.36 | brian | And then I want people to be in a queue to be transferred to this channel. |
| 14:33.37 | [TK]D-Fender | brian, if cell-phone VM picks up, you're toast. If you are not uing a PRI, you're toast. If the phone is off and the cell company answers with an "out of range / service / whatever", you're toast. |
| 14:34.11 | brian | [TK]D-Fender: When it calls him it will have him press some buttons to verify he actually answered. And he is the one activating the application so he'll answer his phone when it rings. |
| 14:35.01 | brian | [TK]D-Fender: They have a problem where they have people call in to his cell phone but he gets so many calls that call waiting causes him to drop calls |
| 14:35.10 | [TK]D-Fender | brian, this is a very painful thing to attempt to do.... |
| 14:35.23 | brian | [TK]D-Fender: Why is it painful? |
| 14:35.33 | [TK]D-Fender | ecam, When you say down-time... are you already running * for your site? |
| 14:35.49 | brian | polerin: do you understand what i'm talking about |
| 14:35.58 | polerin | brian: yes, though i'm a newb. |
| 14:35.59 | [TK]D-Fender | brian, way too much coding... and the guy will either get MoH o or ringing. |
| 14:36.13 | brian | [TK]D-Fender: what are you talking about |
| 14:36.32 | brian | [TK]D-Fender: I just want to know if I can use the Asterisk queue plugin |
| 14:37.54 | polerin | D: btw, could you do it like this... set up a conference, attach the cell to the the conference, then queue the incoming and one at a time add them to the cnference |
| 14:38.16 | [TK]D-Fender | brian,he will have to pick-up within the call timeout period, and the call cannot be answered by anything other than HIM, or its game-over. Understand? |
| 14:38.45 | polerin | d: well, thats on the init of the call, the other ones... well he's not wanting the guy to pick up at all |
| 14:38.46 | [TK]D-Fender | polerin, No, the cell is not WAITING in queue. |
| 14:38.52 | ecam | D-Fender, something like that, we're using some brainless voip->analogue thinngy |
| 14:39.11 | ecam | but it aint * |
| 14:39.21 | brian | What are you talking about [TK]D-Fender |
| 14:39.22 | [TK]D-Fender | ecam, so whats the downtime for? |
| 14:39.37 | brian | [TK]D-Fender: Asterisk is only calling ONE PERSON |
| 14:39.49 | ecam | i have to pull the voip line off the brainless thing to plug it into * |
| 14:39.59 | brian | [TK]D-Fender: Other people are calling into the toll free number! |
| 14:40.06 | brian | [TK]D-Fender: To talk to that ONE PERSON. |
| 14:40.07 | [TK]D-Fender | ecam, not for setup & testing you don't... not until the last second. |
| 14:40.18 | brian | [TK]D-Fender: They need to WAIT IN LINE to talk to that ONE PERSON. |
| 14:40.32 | polerin | D: your not listening. you have you agent (the cell) He doesn't hang up at all. then you have supplicants (people calling him). when you start it out you have it call the agent and connect him to the conference. Then, set up a queue to enter the conference. |
| 14:40.36 | polerin | for the supplicants |
| 14:40.58 | brian | polerin: Yes...That is what I want. |
| 14:41.00 | ecam | i wasn't too sure when initial tests showed that i couldt get any incoming calls from the trunk line working... |
| 14:41.15 | brian | polerin: What is the best way to do that? |
| 14:41.21 | [TK]D-Fender | polerin, YOU don't get it... the cell isn't going to wait in the stupid confernence to get a damn call! |
| 14:41.25 | ecam | but now that i know its my contexts's that's having problems, maybe there wont be downtime |
| 14:41.34 | brian | [TK]D-Fender: What the hell are you talking about? |
| 14:41.48 | polerin | [TK]D-Fender: why not? |
| 14:41.53 | brian | [TK]D-Fender: It's a person! |
| 14:41.58 | polerin | [TK]D-Fender: if he doesn't hang up, why is he going to get disconnected? |
| 14:42.04 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 14:42.15 | polerin | brian: <--- newb. :P |
| 14:42.37 | [TK]D-Fender | polerin, This guy gets a lot of calls. he wants the calls queued. I'm sure this guy doesn't waant to sit ON A CALL all day waiting for someone to ARRIVE. |
| 14:42.54 | brian | That's not how it works. |
| 14:42.56 | polerin | actually, yes, that's what he's asking for. |
| 14:43.03 | [TK]D-Fender | polerin, no, its NOT |
| 14:43.18 | brian | [TK]D-Fender: ummm |
| 14:43.25 | brian | [TK]D-Fender: you don't know the facts |
| 14:43.44 | polerin | ok lets look at a different way |
| 14:43.45 | brian | [TK]D-Fender: I didn't ask you to cry about problems that you think will arise. |
| 14:43.48 | [TK]D-Fender | brian, if I'm missing something its because you didn't provide them. |
| 14:43.54 | polerin | hold on a sec |
| 14:44.02 | polerin | lets take a standard queue. |
| 14:44.04 | brian | If you think your missing something you shouldn't assume and fill stuff in. |
| 14:44.13 | polerin | where an agent logs in, and takes calls |
| 14:44.23 | [TK]D-Fender | brian, and if you think these things aren't complication that will break you system and don't have to be considered go back to dreamland.... |
| 14:44.23 | polerin | these calls are fed to the agent one call at a time |
| 14:44.43 | polerin | this agent doesn't get to choose weither to answer or not |
| 14:44.47 | brian | [TK]D-Fender: If the guy hangs up the queue is killed. |
| 14:44.52 | brian | [TK]D-Fender: not everyone will get to talk to him |
| 14:44.56 | polerin | when they log in, it will auto answer, untill they log out. |
| 14:45.32 | [TK]D-Fender | polerin, no, agents do not auto-answer by any stretch of the imagination nor to they inherently get kicked out. |
| 14:45.40 | polerin | [TK]D-Fender: FSCK |
| 14:45.44 | polerin | listen to me for a second |
| 14:45.48 | polerin | I know these queues exist. |
| 14:45.53 | polerin | I'm sitting in one right now |
| 14:46.04 | brian | Ok I am going to go implement this on my own since you guys are stupid and don't get it. |
| 14:46.20 | [TK]D-Fender | polerin, Yes, and I've been in here for 4 years, am an * consultant and RUN a damned call-center on it. |
| 14:46.22 | brian | Is it really that difficult to understand? |
| 14:46.51 | polerin | [TK]D-Fender: so none of your agents Can ever have auto answer ever, because it's just not technically possible ever |
| 14:46.52 | [TK]D-Fender | brian, ltes try this from the top and then correct me, ok? |
| 14:47.08 | [TK]D-Fender | polerin, No, I'm saying its by no means a REQUIREMENT. |
| 14:47.17 | polerin | ... |
| 14:47.21 | polerin | rephrase |
| 14:47.34 | [TK]D-Fender | polerin, what if the phone I'm rining is an analog phone? How do you make that "auto-answer"? |
| 14:48.04 | polerin | by not hanging it up? |
| 14:48.15 | [TK]D-Fender | polerin, and why would someone every get kicked out if all calls were auto-answer? |
| 14:48.23 | [TK]D-Fender | brin Ok, lets start over... |
| 14:48.53 | polerin | [TK]D-Fender: define "get kicked out" do you meen a supplicant? |
| 14:49.01 | [TK]D-Fender | brian, 1. Call comes in. Enteres a queue to talk to your 1 guy who has only a cell-phone. |
| 14:49.17 | [TK]D-Fender | brianso far so good? |
| 14:50.35 | [TK]D-Fender | brian, then the queu tries to call the cell.and you want him to press something to attempt to be actually connected to the caller? |
| 14:51.03 | brian | The queue doesn't call shit. |
| 14:51.28 | brian | The guy activates the AGI application and the website announces that you can now call in. |
| 14:51.40 | polerin | D-fender: you aren't listening to the person who is asking the question. you are assuming. brian wants the cell to NEVER HANG UP. |
| 14:51.52 | polerin | so this guy can talk to person after person. |
| 14:52.02 | polerin | brian: correct? |
| 14:52.03 | brian | The guy on the cell hangs up when he wants to. |
| 14:52.10 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 14:52.11 | brian | And then it deactivates the application. |
| 14:52.11 | polerin | yeah sorry was about to rephrase |
| 14:52.39 | brian | ANd everyone on the queue is told "no longer available for calls" |
| 14:54.00 | polerin | brian: what input do you want the person on the cell to hear when someone new is put onto his line? |
| 14:54.28 | polerin | brian: do you want him to have to answer the phone, or just beep in his ear |
| 14:54.40 | garreel | BrianHV: it seems to me like using agentcalbacklogin to do queue login/logout |
| 14:55.59 | brian | I'd rather use Dial and then have him dynamically added as an agent. |
| 14:56.02 | *** join/#asterisk Chr1z28 (n=Chr1z@adsl-074-166-041-208.sip.asm.bellsouth.net) |
| 14:56.12 | [TK]D-Fender | brian, So the CELL calls in to accept the next caller for each call? |
| 14:56.20 | brian | No. |
| 14:56.25 | brian | Asterisk calls the cell. |
| 14:56.34 | brian | One time. |
| 14:57.08 | brian | The call in sessions will be probably not more than an hour long probably. |
| 14:57.44 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:57.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:57.44 | brian | s/probably not/not |
| 14:58.38 | [TK]D-Fender | brian, ok, so he's on his first call. How does it end, what happens then? |
| 14:58.51 | brian | He can press * to hang up on the person.... |
| 14:59.12 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 14:59.32 | polerin | brian: Lets see if I can phrase this correctly. Asterisk calls the cell one time. That connection stays active untill the cell is hung up. While the cell is still connected, the * opens a queue to talk to this gentleman. One at a time people from this queue are connected to the cell, without hanging up the cell. |
| 15:00.00 | polerin | when one person from the queue hangs up, the next person is connected, without having to re-call the cell |
| 15:00.10 | brian | pretty much |
| 15:00.18 | brian | it's not like it's some super complex system |
| 15:00.22 | polerin | yeah |
| 15:00.29 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 15:00.37 | brian | i just need to know how to add a agent dynamically |
| 15:00.54 | polerin | [TK]D-Fender: can you see that system? |
| 15:01.01 | brian | using AgentCallbackLogin will be deprecated |
| 15:01.18 | brian | So I need to use Dial |
| 15:01.40 | wunderkin | agentcallbacklogin is not what you want anyway, you need add/removequeuemember |
| 15:02.11 | polerin | wunderkin: the problem I'm not sure of (because i'm a newb).. when the call hangs up, will it hang up the cell? |
| 15:02.22 | brian | polerin: no |
| 15:02.30 | [TK]D-Fender | brianI think I see a way. You DO queue the calls like normal. let them sit in line. You will have a process running to see if the queue has people waiting. if so, check and see if the agent (CELL) is currently logged in. |
| 15:02.44 | brian | [TK]D-Fender: no |
| 15:02.47 | polerin | brian: i know you don't want it, i'm curious about weither it will do it regardless |
| 15:03.03 | brian | [TK]D-Fender: if he's not logged in it will give them a message that he's not available for calls and to check the website |
| 15:03.16 | brian | [TK]D-Fender: I'm using FastAGI |
| 15:03.50 | [TK]D-Fender | brian, if the agent is NOT logged, spawn a call-file (or similar to have * call the cell. The cell can then answer that call and * will prompt him to log-in. upon doing so he will call the "QueueLogin" and sit in there like an old-style permanent agent until he quits because of their being no more callers |
| 15:03.55 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 15:04.13 | polerin | [TK]D-Fender: your not getting something, He doesn't want the cell to be called at any time. |
| 15:04.14 | brian | [TK]D-Fender: no |
| 15:04.21 | polerin | [TK]D-Fender: only when this dood WANTS calls |
| 15:04.32 | brian | polerin: exactly |
| 15:04.50 | brian | The whole purpose of the system is to allow users of the website to call in when he wants users of the website to call in. |
| 15:04.53 | [TK]D-Fender | brian, fine, give him a simple option to stop the call-out. Who said that was a big deal!? |
| 15:05.01 | brian | And without giving out his cell phone number. |
| 15:05.16 | wunderkin | then you can use the queue options joinempty=no and leavewhenempty=yes |
| 15:05.17 | brian | Or whatever number we set it to call. |
| 15:05.33 | brian | I understand that wunderkin |
| 15:05.33 | polerin | [TK]D-Fender: ok let me ask it this way. Can you do this. |
| 15:05.41 | wunderkin | ok what is the question |
| 15:05.46 | brian | What i don't understand is how to add the cell phone user as an agent? |
| 15:06.08 | wunderkin | agentcallbacklogin is not what you want anyway, you need add/removequeuemember <-- |
| 15:06.18 | brian | I think it'd be best to use a MeetMe conference (it's a hack so that I can map keys to functions) |
| 15:06.30 | wunderkin | or i guess when you call out, you can have it go to an agentlogin |
| 15:06.49 | wunderkin | that would be easier i guess wouldnt it |
| 15:06.51 | brian | wunderkin: The agent doesn't need to have a password. |
| 15:07.00 | polerin | [TK]D-Fender: can you set up a conference where it is limited to two people, then connect the cell to one side of that. for the other side have a queue to enter the confernce |
| 15:07.01 | [TK]D-Fender | brian, you shouldn't add him as a direct-dial agent. he should be called and told to log in when there are callers and he wants to accept calls. |
| 15:07.17 | brian | [TK]D-Fender: what are you talking about |
| 15:07.27 | brian | [TK]D-Fender: who said ANYTHING about that? |
| 15:07.36 | polerin | brian: he's trying to let the guy hang up when the queue is empty |
| 15:07.43 | polerin | brian: but that's not part of what you specified |
| 15:08.02 | brian | [TK]D-Fender: you keep assuming that I want to do things that I don't want to do. |
| 15:08.22 | brian | [TK]D-Fender: You're assuming that I'm a huge newbie |
| 15:08.28 | brian | [TK]D-Fender: I just never worked with Queue |
| 15:08.37 | wunderkin | brian, when you originate the call, the other side of it should be agentlogin... there you go |
| 15:08.39 | brian | [TK]D-Fender: I just want to know how I can do this |
| 15:09.18 | brian | I want a MeetMe conference to be the agent though. |
| 15:09.34 | wunderkin | why do you want to use meetme |
| 15:09.50 | brian | I don't think there's any other ways to escape the call to perform dialpad functions |
| 15:09.57 | [TK]D-Fender | brian, And you are using the wrong tools for formulating your solution. Im telling you what will fulfill your GOAL. the fact you can't process this is not my problem. |
| 15:10.17 | brian | Basically I want him to be able to press like *9 to block the caller from ever calling again, etc... |
| 15:10.42 | polerin | [TK]D-Fender: you aren't even listening to what the goal is |
| 15:10.55 | [TK]D-Fender | brian Your right you don't understand queues or the agent system, or the APPS involveld and understand that a remote Cell user is the WORST kind of thing to try to implement for a variety of reasons. |
| 15:11.16 | polerin | [TK]D-Fender: I realize you do this for a living and you know more about * than I do, but you are trying to implement something different |
| 15:11.27 | brian | [TK]D-Fender: You really don't get it. |
| 15:11.47 | brian | Let's say I have a phone...it could be a cell phone or a landline or whatever. |
| 15:11.53 | brian | I go to a web UI. |
| 15:11.56 | [TK]D-Fender | polerin, No, you don't understand that you have to deal with this in a round-about way because * queues SUCK and don't have all the brains needed to do this in a way that seems NORMAL. |
| 15:11.58 | brian | And I enter in login information. |
| 15:12.13 | wunderkin | hey if this guy wants to sit on an hour long cell phone call thats ok, they only want calls when this guy is logged in.. that part is doable.. sucky.. but that is their problem |
| 15:12.14 | brian | And I activate the call in feature. |
| 15:12.27 | wunderkin | the only problem i see is doing the dtmf dispositions |
| 15:12.27 | polerin | [TK]D-Fender: you are still trying to implement something different than what he's asking for |
| 15:12.35 | brian | wunderkin: Yeah, how would I do that? |
| 15:12.49 | brian | wunderkin: Seems like the only way would be to use a MeetMe conference. |
| 15:12.54 | [TK]D-Fender | brian, * queues are very limited and you have to cheat to get the job done for what you want. If what I'm suggesting doesn't feel "normal", you would be CORRECT. thats because * was never NORMALLY deisgned to handle the shit you want to do! |
| 15:13.06 | [TK]D-Fender | brian, so you ahd BETTER learn how to "cheat" and fast. |
| 15:13.09 | brian | [TK]D-Fender: It's called a hack. |
| 15:13.17 | brian | [TK]D-Fender: I do want to cheat... |
| 15:13.24 | brian | [TK]D-Fender: I want a MeetMe conference channel to become the agent. |
| 15:13.29 | wunderkin | i think gnudialer had to mod the source to do that... im not sure how else you would atm... i dont think meetme is going to help any.. but i have not really used it so i would have to see |
| 15:14.04 | [TK]D-Fender | MeetMe will not work as part of this solution! Get it through your heads! |
| 15:14.09 | brian | yes it will |
| 15:14.13 | brian | I've used MeetMe before |
| 15:14.16 | polerin | [TK]D-Fender: wow, seriously, yell much? |
| 15:14.22 | [TK]D-Fender | brian, then you clearly have all the answers. Go do it. |
| 15:14.25 | [TK]D-Fender | ~wglwat |
| 15:14.27 | jbot | wglwat is, like, well, good luck with all that |
| 15:14.40 | brian | Just forget it. I'll figure it out. |
| 15:14.46 | brian | You don't want to get it. |
| 15:15.02 | wunderkin | i know what you are doing and i have to implement the same thing |
| 15:15.11 | wunderkin | that i have been putting off for awhile... |
| 15:15.22 | brian | wunderkin: interesting |
| 15:15.25 | [TK]D-Fender | wunderkin, A world of pain awaits... |
| 15:15.34 | [TK]D-Fender | *sigh* |
| 15:15.40 | polerin | [TK]D-Fender: let me ask a simple question |
| 15:15.40 | brian | It's not pain |
| 15:15.41 | wunderkin | hehe yes everything i do with asterisk is a hack |
| 15:16.04 | [TK]D-Fender | wunderkin, we'll ahve to swap stories later... those 430's still acting up? |
| 15:16.45 | brian | It's not that difficult to hack Asterisk to work this way. |
| 15:16.50 | wunderkin | :-) yes..! it took awhile but they actually agreed to exchange them for 501s and poe cables, like they said before... and do an advanced replacement w/o security.. wow! |
| 15:17.00 | brian | Maybe if your only using extensions.conf |
| 15:17.08 | brian | But I'm using FastAGI so I can organize the logic a lot better |
| 15:17.33 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) [NETSPLIT VICTIM] |
| 15:18.37 | wunderkin | so maybe by the end of next week they will all be in and working... i have to implement that SER on a WRT solution :P another hack :D hehe.. i was looking at milkfish for that.. unless i can figure it all out on my own.. |
| 15:19.28 | [TK]D-Fender | wunderkin, Oh god.. do you mean that redundant scenario we talked about a month or so ago? |
| 15:19.48 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:20.10 | wunderkin | yes... i wanted to verify that was the problem... guess so! the next priority was dealing with the reboots and that has been proven so we are finally over once this is finished... |
| 15:20.20 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) [NETSPLIT VICTIM] |
| 15:20.41 | brian | :( |
| 15:21.00 | brian | why is asterisk so uncustomizable |
| 15:21.45 | [TK]D-Fender | wunderkin, *shudder* |
| 15:22.06 | wunderkin | only took 6 months plus |
| 15:25.53 | wunderkin | ive seen other people that do 60 sec regs so i dont see what is the problem |
| 15:26.55 | wunderkin | maybe they dont use the phone as much |
| 15:27.25 | brian | :( |
| 15:27.31 | wunderkin | what? |
| 15:27.39 | brian | I hate asterisk |
| 15:28.21 | brian | Everyone in here confused me :( |
| 15:28.56 | *** join/#asterisk Fieldy (i=hjnt79QL@gentoo/contributor/Fieldy) |
| 15:29.08 | wunderkin | thank you for making me realize what i had to do is not going to be easy... now i have to find that gnudialer patch and see if it even applies anymore |
| 15:29.40 | brian | wunderkin: halp |
| 15:29.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:30.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:30.25 | brian | I was thinking of just writing my own queue code in AGI. |
| 15:32.55 | brian | wunderkin: what is that patch for? |
| 15:33.19 | wunderkin | to allow dtmf dispositions on chan_agent |
| 15:34.44 | mocker | Hmm, new place I'm starting has a trixbox setup. I've only done straight config files before.. |
| 15:34.52 | mocker | To convert them or not.. ;) |
| 15:34.59 | *** join/#asterisk MrWup (n=neil@80-46-38-124.static.dsl.as9105.com) |
| 15:35.13 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 15:39.04 | brian | wunderkin: I found something that may be of interest to you: http://www.voip-info.org/wiki/view/Agents+without+agent+channel |
| 15:41.30 | wunderkin | yeah im aware of that |
| 15:42.47 | brian | whats wrong w/ that |
| 15:43.23 | wunderkin | maybe you can do something with feature keys |
| 15:51.14 | polerin | is there a way to send text to a phone? |
| 15:51.34 | polerin | (for example, if I have a test extension, and I want to call from the console to it and get text feedback |
| 15:53.02 | *** join/#asterisk bkw__ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 15:53.03 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
| 15:56.21 | [TK]D-Fender | polerin, Depends on what phones |
| 16:00.40 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:03.03 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:05.05 | polerin | [TK]D-Fender: i'm not using a phone right this sec because i'm at work ;P |
| 16:05.11 | polerin | i'm sshed to my test box and have text only |
| 16:05.34 | [TK]D-Fender | <PROTECTED> |
| 16:06.13 | polerin | again, i'm learning. It would be nice to be able to get some feedback while I run through examples and play |
| 16:06.21 | polerin | without having to bother my wife |
| 16:07.02 | [TK]D-Fender | polerin, Well you asked a completely blind question without any substance. Some phones you can send "text" to, others you can't. |
| 16:07.49 | polerin | I said " Call it from the console and get text feedback" sorry if I missed the " to the console" |
| 16:08.07 | polerin | I know the softfone can't receive text, already tried it ;P |
| 16:08.17 | [TK]D-Fender | polerin, What kind of feedback are you thinking here? |
| 16:08.55 | [TK]D-Fender | polerin, AFAIK the only phones you can "push" text to right now are Cisco's |
| 16:09.10 | polerin | can you push text to the console? |
| 16:09.53 | polerin | ie dial 100@internal |
| 16:09.58 | polerin | then get text back to it |
| 16:10.08 | gambolputty | I want push text too |
| 16:10.10 | polerin | or what's the syntax so I can TIYAS? |
| 16:10.13 | [TK]D-Fender | polerin, you can do that in a few different ways I might think. Give a complete example of an interaction you want to do. |
| 16:10.17 | gambolputty | for some custom apps I have in mind |
| 16:11.01 | [TK]D-Fender | TIYAS? |
| 16:11.15 | wunderkin | sayit..backwords.. |
| 16:11.18 | polerin | try it yourself and see |
| 16:11.31 | [TK]D-Fender | polerin, Ok, I not answering these question like signing off blank cheques..... |
| 16:11.46 | wunderkin | you don't want a blank check? |
| 16:11.46 | [TK]D-Fender | polerin, Give a complete scenario and then you'll get an answer. |
| 16:11.52 | polerin | I did |
| 16:11.58 | polerin | I have a [test] context |
| 16:12.10 | polerin | I want to be able to call an extention in it and have text pushed back to me |
| 16:12.11 | [TK]D-Fender | polerin, sorry "send text" and "see text at console" don't cut it. |
| 16:12.58 | [TK]D-Fender | polerin, polerin And this new sample does not involve the "console" |
| 16:13.21 | polerin | yes it does, as the only way I can call it currently is from asterisk -rc |
| 16:13.28 | polerin | dial 100@test |
| 16:14.05 | [TK]D-Fender | polerin, well if you want to see "stuff" at console thats what "NoOp and VERBOSE are for. |
| 16:14.20 | brian | How do I dial a remote SIP user with Asterisk Dial command? |
| 16:14.24 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:14.36 | [TK]D-Fender | polerin, but these have NOTHING to do with being able to send text to a PHONE. So you are asking about apples and oranges here. |
| 16:14.39 | polerin | Dial(SIP/user@host.com) |
| 16:14.54 | wunderkin | this is not a fruit stand! |
| 16:15.03 | *** join/#asterisk docelmo (n=vircuser@c-68-45-140-42.hsd1.de.comcast.net) |
| 16:15.14 | [TK]D-Fender | wunderkin, right... nobody here but... VEGETABLES! |
| 16:15.25 | wunderkin | #veggietales |
| 16:15.25 | docelmo | Can someone tell me the application used for reading DTMF? |
| 16:15.33 | polerin | [TK]D-Fender: give me the benifit of at least having RTFM a hair. When I do a dial from the cli console, it is aping a phone correctly? |
| 16:15.39 | [TK]D-Fender | docelmo, "show application read" |
| 16:15.41 | polerin | s/correctly/correct |
| 16:15.58 | docelmo | [TK]D-Fender thanks.. I thought it was only for variables.. |
| 16:16.34 | MrWup | is there any e-mail style server for debian which would work really well with microsoft outlook on windows xp client machines? i need groupware and all that stuff |
| 16:16.35 | polerin | now, that being said, thanks for the better answer. |
| 16:16.57 | [TK]D-Fender | polerin, What phones can see what the CONSOLE can see? It has limitations that your average SIP phone doesn't. It can do things an average SIP phone can't. Extend that logic to Zap analog channels, or SCCP, etc. |
| 16:16.57 | MrWup | (im trying to set up a voicemail system, but first i need e-mail so I can send voicemail as attachments to users email accounts) |
| 16:16.59 | MrWup | could be so cool |
| 16:17.22 | polerin | i'm not saying it IS a phone, but that in that context it is aping a phone at least slightly |
| 16:17.46 | [TK]D-Fender | polerin, Your test will not work on any kind of phone I've ever seen. Only in CONSOLE. so whats the point of doing something one way in "test" that will not have ANY use in your real-world usage after? |
| 16:17.58 | polerin | because right now I can't do anything else :P |
| 16:18.08 | [TK]D-Fender | polerin, thats called WASTING TIME |
| 16:18.14 | [TK]D-Fender | polerin, Yours and now MINE. |
| 16:18.16 | polerin | it's also called Learning |
| 16:18.51 | [TK]D-Fender | polerin, well I've already given you the answer to this. |
| 16:19.10 | [TK]D-Fender | [TK]D-Fender> polerin, well if you want to see "stuff" at console thats what "NoOp and VERBOSE are for. |
| 16:19.12 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 16:19.18 | polerin | yes you have, and thank you. I was going with the hack I knew |
| 16:19.53 | [TK]D-Fender | so do exten => 100,1,Verbose(Yay, I'm dialing shit now!) |
| 16:19.59 | *** join/#asterisk Dimitripietro (n=ssdsdfa@modemcable251.229-130-66.mc.videotron.ca) |
| 16:20.29 | [TK]D-Fender | polerin, and for the only way I've ever heard of this applying to an actual PHONE : http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services |
| 16:24.25 | [TK]D-Fender | MrWup, *'s voicemail is not sufficient for your needs? |
| 16:24.40 | polerin | [TK]D-Fender: thats a perfect answer, just making sure :P |
| 16:26.14 | [TK]D-Fender | polerin, Wel you ask one thing and then want the answer formulated to something that is not at all the same and WONDER why ew're confused and feeling we're wasting our time? :) |
| 16:27.15 | polerin | [TK]D-Fender: heh, I tend to give the short answer (ie "the syntax is blah() but do this instead") and then smack the n00b about |
| 16:27.18 | polerin | different styles ;) |
| 16:27.56 | [TK]D-Fender | polerin, Yes yours would fall under the scope of "insane" :) |
| 16:28.44 | MrWup | [TK]D-Fender, well, not really. i dont want people to have to check their voicemail via their phones. i want it all linked in with e-mail. so management can be sure people are replying to their business messages, and can keep a proper log of all voicemails |
| 16:29.06 | MrWup | would be so much tidier for administration if it were linked in with messaging |
| 16:29.24 | MrWup | thing is, doing that is easy now ive worked out how to use agi and php. only problem is now finding a decent alternative to exchange server |
| 16:29.29 | polerin | well there's usually a method to my madness. Yah have to understand I'm not used to the * methodology so much as webdev/video issues |
| 16:29.32 | MrWup | (which is far too expensive and cumbersome) |
| 16:29.41 | [TK]D-Fender | MrWup, then use *'s VM -> email unctionality that it already has and set "delete=yes|attach=yes" |
| 16:29.53 | MrWup | i need some sort of groupware messaging server which will run on debian and be compatible with windows XP clients running MS OUtlook |
| 16:30.02 | [TK]D-Fender | polerin, indeed there is a madness to your method... |
| 16:30.04 | polerin | MrWup: if nothing else, there is ogo, but wow if that's not a piece anyway |
| 16:30.42 | polerin | (above is not a suggestion, it's a joke) |
| 16:32.08 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 16:32.20 | BSD_Tech | so how did the phone conf go yesterday |
| 16:32.39 | [TK]D-Fender | BSD_Tech, I practically because the host :) |
| 16:32.47 | [TK]D-Fender | became* |
| 16:32.49 | [TK]D-Fender | shfhdffd |
| 16:33.38 | BSD_Tech | again in english |
| 16:33.38 | BSD_Tech | lol |
| 16:33.38 | [TK]D-Fender | ack... tired headachy, playing guitar and getting ready to head out to martial arts.... words escape me and the fingers just start typing :) |
| 16:33.57 | BSD_Tech | ahh |
| 16:36.44 | polerin | [TK]D-Fender: which style? |
| 16:37.03 | [TK]D-Fender | polerin, on guitar, sweeping right now, martial arts : Tenshen Shodan Katori Shinto Ryu. |
| 16:37.19 | polerin | if you say tkd, i'm going to laugh because that's the first thing I thought of when I saw your nick |
| 16:37.22 | polerin | er.. aww ;) |
| 16:37.41 | polerin | karate variant? not one i'm famiar with |
| 16:37.54 | Qwell | street fighter variant |
| 16:37.59 | [TK]D-Fender | TSKSR > TKD, no competition :) |
| 16:38.09 | polerin | don't start tkd bashing. |
| 16:38.22 | polerin | sport tkd is trash, but trad isn't to be sneezed at :) |
| 16:38.32 | polerin | at least if you keep your stupid hands up |
| 16:38.55 | [TK]D-Fender | polerin, TSKRS is samurai WEAPONS arts in the kobudo family. I draw my katana and lets see how all your hand-to-hand techniques deal with STEEL :D |
| 16:39.25 | [TK]D-Fender | I never said it would be a FAIR fight, now did I? ;) |
| 16:39.36 | [TK]D-Fender | Right now I'm on katana & bo. |
| 16:39.46 | Qwell | pfft, I could take you |
| 16:39.49 | [TK]D-Fender | Naginata comes in another year +/- |
| 16:39.50 | polerin | heh |
| 16:40.24 | [TK]D-Fender | qwell, from your last description you attested to being short and a relatively small rfame. I am NEITHER, and better armed ;) |
| 16:40.29 | Qwell | what're you gonna do, parry? :P |
| 16:40.33 | [TK]D-Fender | qwell, Gun-fu! |
| 16:40.42 | polerin | lol |
| 16:42.10 | polerin | someone's watched equlibrium |
| 16:42.10 | [TK]D-Fender | I am actually practicing left-handed now and inverting the katas to through of right-handers who won't realize till the end that their lst move won't work :) |
| 16:42.31 | [TK]D-Fender | Equilibrium was a nigfty movie.... over-the-top, but fun regardless |
| 16:42.35 | polerin | yeah |
| 16:42.50 | polerin | [TK]D-Fender: I don't do formal weapons traning, but I fight against people that have |
| 16:43.28 | [TK]D-Fender | I hope to be able to learn daisho with my sensei. I've heard there have been 1-2 students in the apst who've done it. |
| 16:43.41 | [TK]D-Fender | polerin, hand to hand works on blunt, no so much on sharp.. esp LONG sharp. |
| 16:44.06 | [TK]D-Fender | polerin, I did some aikido with works against staff & knife, but long + sharp = dead :) |
| 16:44.17 | polerin | you didn't see what I said :P |
| 16:44.23 | polerin | Shortspear + shortsword |
| 16:45.20 | Qwell | any of you folks using alphanumeric iax2 users? |
| 16:45.25 | Qwell | on 1.4 |
| 16:45.33 | Qwell | or 1.2, I guess |
| 16:45.56 | [TK]D-Fender | ok, I'm off.. BBIAB |
| 16:46.16 | Qwell | I really, really, really do not believe this bug report |
| 16:48.17 | *** join/#asterisk beeb (n=b@220-253-45-208.VIC.netspace.net.au) |
| 16:49.15 | beeb | hi |
| 16:50.51 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 16:51.44 | InHisName | I rebooted my * and now my Sunrocket is giving me a 404 error on registration. |
| 16:53.30 | InHisName | polerin , got a serious asterisk question instead then ? |
| 16:54.32 | polerin | actually it was a troubleshooting question for you |
| 16:54.49 | polerin | but i'm a newb so i'm hushing :) |
| 16:56.19 | InHisName | Try me anyway as I might be able to help you if it is an easy question. Don't hush as there isn't much going on at the moment. |
| 16:57.39 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 16:58.18 | polerin | static IP? |
| 16:58.26 | InHisName | Hello, CunningPike, speak something. |
| 16:58.49 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:59.08 | polerin | InHisName: I notice your on comcast, so I was curious if it's the front box on a comcast connection |
| 16:59.09 | InHisName | I have done static a little, at least in my home lan. |
| 16:59.40 | InHisName | 1st box is cable modem, then astlinux. |
| 16:59.45 | polerin | if you are registering via that IP, it is possible for it to change. We give out long leases, but they do occasionally flip up |
| 16:59.55 | InHisName | A combo linux and asterisk box. |
| 17:00.05 | polerin | cable modem is a bridge. The mac the DHCP sees is your external NIC |
| 17:00.47 | brian | I'm having a problem... |
| 17:00.51 | polerin | if you are trying to register with the external ip, make sure it's what you are expecting :) |
| 17:01.02 | InHisName | "We" give out long leases ? who is "we" ? |
| 17:01.09 | polerin | <--- works for comcast |
| 17:01.17 | Qwell | k/ickban polerin |
| 17:01.19 | Qwell | oops :P |
| 17:01.24 | polerin | heh |
| 17:01.32 | Qwell | polerin: upgrade me :) |
| 17:01.40 | brian | I'm using Dial() with context justintvagent and extension s... |
| 17:01.42 | brian | [justintvagent] |
| 17:01.42 | brian | exten => s,1,AddQueueMember(justin) |
| 17:01.44 | polerin | hey, trust me, I'm a good person to have contact with if you are having problems/getting the run around |
| 17:01.46 | Qwell | polerin: what do you do there? |
| 17:01.53 | brian | I'm then using AddQueueMemeber as such... |
| 17:01.58 | *** join/#asterisk friedrich| (n=friedric@e177249157.adsl.alicedsl.de) |
| 17:02.01 | brian | It adds the queue member ... |
| 17:02.03 | InHisName | I was working fine, until I had modem problem. Comcast exchanged it for another. I STILL have the same IP as before, amazing isn't it? |
| 17:02.15 | brian | But it hangs up after it adds the queue member. |
| 17:02.23 | polerin | no, swapping modems won't give you a different IP |
| 17:02.26 | Qwell | polerin: go get ahold of the editors of The Consumerist - they do a piece on "confessions from techs" at ISPs and telcos and stuff |
| 17:02.34 | Qwell | would love to hear a couple from Comcast ;) |
| 17:02.35 | polerin | the modem is a bridge. The dhcp lease is granted to your NIC's interface |
| 17:02.52 | polerin | Qwell: I'm a dispatcher/advanced tech. I provide tech support to the techs in the field. Been doing that 3 years, was a field tech specialising in modems for another 3 |
| 17:03.07 | Qwell | ahh, yeah, I'm sure they'd love to hear from you then |
| 17:03.11 | polerin | heh |
| 17:03.38 | brian | When you add someone as a queue member |
| 17:03.47 | polerin | I do the best I can for everyone I touch. I'm a rarity. I also have a big head and get in lots of arguments with my management over procedure/etc, so I draw heat too :P |
| 17:03.50 | brian | Isn't it supposed to like...not hang up? |
| 17:04.16 | polerin | brian: rofl. |
| 17:04.22 | InHisName | Can you interperet the error: Apr 7 12:48:33 WARNING[1383]: chan_sip.c:9745 handle_response_register: Got 404 Not found on SIP register to service 3145630068@sunrocket.com, giving up ? polerin |
| 17:04.24 | polerin | We'll get to you shortly *click* |
| 17:05.03 | polerin | InHisName: as I said. I'm a newb when it comes to * so not really. Sounds to me like its not contacting sunrocket correctly though. |
| 17:05.11 | polerin | **shrugs** |
| 17:05.44 | polerin | almost anyone in here is better qualified to answer, it's why I was going to stifle that question :) |
| 17:05.59 | InHisName | oh |
| 17:07.25 | InHisName | 261 people ? logged into asterisk here and only 3 are reading message as far as I can tell. Must be gobs of other ar eon break, a big one! |
| 17:08.12 | polerin | <-- queen of idle |
| 17:09.30 | Qwell | polerin: so, does your being here mean Comcast is gonna start using asterisk? ;) |
| 17:09.43 | etfonhomey | Need help with setting up presence. I have an * server and two sip phones - one a Polycom 301 and the other Xlite softphone |
| 17:09.58 | polerin | Qwell: ... bwahahaha |
| 17:10.07 | polerin | **cough cough hack** |
| 17:10.33 | Qwell | I'll take that as a "we already do" |
| 17:10.43 | etfonhomey | Can anyone help with that? |
| 17:10.44 | polerin | lets put it this way. The internal webapp I designed is written in PHP. That's the only peice of OSS about it though, IIS and MSSQL. |
| 17:12.05 | *** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 17:12.14 | InHisName | brian, what is the queue member thing do ? Like when on hold for real person on an airline etc ? |
| 17:12.50 | polerin | Qwell: I'm here because I'm a friend of corydon's and I'm helping my wife set up her small buisness |
| 17:12.55 | Qwell | ahh |
| 17:13.44 | polerin | just out of curiousity, I haven't seen congestion with aguments. does it take an int as a time limit? |
| 17:13.48 | Qwell | So, when my net goes out, I should call him to get ahold of you? ;) |
| 17:13.53 | ManxPower | etfonhomey: We could, but we have already answered that question many, many times both here and on the mailinglists. |
| 17:13.53 | ManxPower | and we are tired of it. |
| 17:14.22 | polerin | do you live in nashville/surrounding areas? |
| 17:14.23 | polerin | :P |
| 17:14.32 | Qwell | polerin: Huntsville - close enough :P |
| 17:14.52 | polerin | heh, not close enough for me to dispatch, but I can look at some of your stats and give you some suggestions if you are having problems. |
| 17:15.09 | Qwell | and yeah, it takes seconds as an arg |
| 17:15.15 | Qwell | core show application congestion |
| 17:15.36 | polerin | ah, ok so you can look up args that way. awesomeness, thanks. |
| 17:15.36 | brian | ugh |
| 17:15.46 | *** join/#asterisk Rhab (n=Rhab@p3EE0D4BF.dip.t-dialin.net) |
| 17:15.54 | Rhab | hi |
| 17:16.02 | brian | How come when I originate a call through the manager that AddQueueMember it hangs up after it AddQueueMember? |
| 17:16.05 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:16.05 | InHisName | hi, Rhab |
| 17:16.12 | brian | How do I make it not hang up? |
| 17:16.37 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:16.58 | InHisName | brian what is AddQueueMember ? do. What queue is that ? |
| 17:17.06 | brian | I'm basically doing Dial() with a context of agent and in the agent starting context it does AddQueueMember(queue_ |
| 17:17.08 | wunderkin | all addquemember does is add the person as a member to the queue... |
| 17:17.19 | Rhab | anyone here experienced with fax-recieving through misdn? |
| 17:17.29 | brian | As a member...like...as a agent right wunderkin? |
| 17:17.49 | polerin | brian: btw, when you get that done, may I see your work for future reference? |
| 17:18.11 | brian | I think this is going to be like, closed source proprietary. |
| 17:18.30 | Rhab | I can revieve faxes, but the generated tif files are broken |
| 17:18.33 | Rhab | somehow |
| 17:18.37 | brian | ugh |
| 17:19.42 | polerin | brian: that's fine, a description in private if you can would be ok. I just like knowing how problems I've looked at were resolved ... if possible :) |
| 17:20.40 | wunderkin | brian, what you use depends on how you are doing things... |
| 17:21.26 | ManxPower | Brian: if your call is in the dialplan then do something else in the dialplan after the addqueue member. |
| 17:24.23 | *** join/#asterisk anthm (n=anthm@m015f36d0.tmodns.net) |
| 17:24.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:26.40 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 17:33.25 | brian | Is an Asterisk channel the same thing as an interface? |
| 17:33.37 | beeb | <PROTECTED> |
| 17:33.49 | brian | ManxPower: Such as? |
| 17:34.22 | brian | I was under the impression that I'd be logged in as an agent and added to an agent channel if I did AddQueueMember |
| 17:35.24 | brian | But apparently it's hanging up on me. |
| 17:36.03 | brian | Probably because there is no one waiting in the queue... |
| 17:36.08 | wunderkin | no |
| 17:36.21 | brian | then why? |
| 17:36.58 | polerin | Apr 7 12:32:11 WARNING[2663]: chan_iax2.c:7126 socket_read: Call rejected by 192.246.69.186: Unable to negotiate codec <--- iax2 to FWD. outbound call. |
| 17:37.04 | ber_ | I just signed up with a SIP provider but I cant seem to get outbound calls working |
| 17:37.05 | polerin | any suggestions? |
| 17:37.15 | ber_ | when i look at sip debug it says i need to authenticate my invite message |
| 17:37.17 | ber_ | but i dont know how to do that |
| 17:37.27 | brian | wunderkin: heeeelp |
| 17:37.27 | ber_ | inbound/DID calls work ok |
| 17:38.24 | polerin | brian: such as Verbose(Debug - step past Add to queue) |
| 17:38.52 | wunderkin | brian, for the second leg of the call you can't do addqueuemember... you either have to use agentlogin (which will use chan_agent and i'm not sure how you will get dtmf dispositions from that, i don't think you can use features) or something else, i am trying to figure out how to do it myself so you will have to hold onto your panties |
| 17:39.25 | brian | wunderkin: AddQueueMember does agentlogin I thought... |
| 17:39.42 | wunderkin | they are both different apps |
| 17:39.59 | brian | Dynamically adds interface to an existing queue (i.e. the interface "logs on" to the queue, as an agent does with AgentCallbackLogin). |
| 17:41.16 | brian | I think I'm going to just wrap it in a MeetMe conference... |
| 17:41.28 | brian | That's probably the only way this will work. |
| 17:41.36 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:41.56 | garreel | you need to define queues in queues.conf |
| 17:42.06 | garreel | then define agents in agents.conf |
| 17:42.20 | garreel | then assig the agents to queues in queues.conf |
| 17:42.26 | Qwell | don't use agents |
| 17:42.29 | Qwell | agents suck |
| 17:42.46 | polerin | brian's thing is a little different than the standard queue though |
| 17:42.53 | Qwell | how so? |
| 17:43.01 | polerin | brian: you should probably re-ask the full question |
| 17:43.11 | garreel | then make agents logon with agentlogin or agentcalbacklogin |
| 17:43.45 | wunderkin | no you dont want to use agentcallbacklogin |
| 17:43.49 | polerin | err.. well.. for one he doesn't want the agent's connection to disconnect when the other end hangs up, and he needs the agent to able to use dtmf to disconnect/reconnect calls |
| 17:44.16 | garreel | then agentlogin |
| 17:44.17 | Qwell | use a Local channel with features. Don't use agents |
| 17:44.28 | Qwell | agents are a very very bad idea |
| 17:44.58 | polerin | Qwell: basicly, he is going to have an agent log in on a web page, have * call the specified number, then set up a queue to connect person after person to it without disconnecting the agent (ie making them need to answer the call over and over) |
| 17:45.17 | Qwell | then you can't use AddQueueMember |
| 17:45.23 | Qwell | s/you/he/ |
| 17:45.38 | Qwell | a meetme could work, but that's hackish |
| 17:45.40 | polerin | ... jbot does regex? awesomeness |
| 17:45.49 | polerin | s/awesomeness/that is badassed |
| 17:45.50 | brian | Qwell: well |
| 17:45.55 | brian | Qwell: I think hackish is the only way? |
| 17:45.59 | polerin | :/ |
| 17:46.49 | Qwell | trouble with doing it that way, is if the person disconnects, the queue member (the meetme) will still be logged in |
| 17:47.01 | Qwell | so calls will still be sent to it |
| 17:47.05 | brian | well I'm using FastAGI and I'm monitoring events... |
| 17:47.12 | brian | from the AMI interface |
| 17:47.22 | brian | So I should be able to fix that up from within my FastAGI script. |
| 17:47.32 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177819133.dsl.bell.ca) |
| 17:50.11 | garreel | Qwell[]: why are agents a very very bad idea? |
| 17:50.44 | Qwell | because agents can cause some really ugly bugs |
| 17:51.00 | polerin | heh |
| 17:51.03 | garreel | any examples? |
| 17:51.16 | brian | Qwell: How do I use a local channel with features with Asterisk Queue? |
| 17:51.17 | Qwell | crashing in stuff that is in no way related to agents |
| 17:51.34 | Qwell | brian: in 1.4 there is a sample queues document - it explains most of that |
| 17:51.39 | Qwell | at least how to do the local channels |
| 17:51.39 | polerin | anyone have a sec to look at a basic error line and tell me where to start digging/rtfming? |
| 17:51.49 | garreel | is this related to 1.4 too? |
| 17:51.59 | Qwell | bbl |
| 17:52.00 | brian | I'm using like...1.2 |
| 17:52.26 | garreel | bbl? |
| 17:52.31 | brian | :( |
| 17:52.37 | polerin | lol |
| 17:52.52 | polerin | brian: he's at least given you a starting point |
| 17:53.00 | brian | I don't know where to begin anymore. |
| 17:53.34 | polerin | from the haze that is my newbness, i'd start with local channels, not that I know where that is |
| 17:53.42 | fab5freddy | How do I find out the IP that users from my the outside world would use to connect to my Asterisk network? |
| 17:54.01 | polerin | fab5freddy: whats your network setup? |
| 17:54.17 | fab5freddy | polerin: dynamic ip |
| 17:54.27 | polerin | not quite what I meen, but ok. |
| 17:54.36 | polerin | is your * box behind a firewall or NAT router? |
| 17:54.54 | fab5freddy | polerin: i can connect find on my lan using 192.168.2.11 but i assume this is not the same for those connecting from the outside world |
| 17:55.09 | polerin | ok, is that a static ip? |
| 17:55.33 | fab5freddy | polerin: i get internet from a dsl router |
| 17:55.39 | polerin | if not, you should probably assign it a static somewhere outside of the DHCP range |
| 17:56.00 | polerin | ok. What's the lowest IP your router gives out? 2.10? |
| 17:56.13 | fab5freddy | polerin: yes |
| 17:57.10 | polerin | ok. I'd reccomend you look into assigning your box a static Ip between 2.2 and 2.9. Are you using SIP or iax for your provider connection? |
| 17:57.22 | fab5freddy | polerin: sip |
| 17:58.01 | cspot | 5060? |
| 17:58.12 | fab5freddy | 5060 sounds right |
| 17:58.13 | brian | I can't even get this crap to Dial() and then join a Meetme. |
| 17:58.22 | polerin | ok, find what port SIP listens on, then forward that port on the router to the static IP you set on your * box. |
| 17:58.51 | fab5freddy | polerin: thanks for the help |
| 17:59.02 | polerin | then either look in your router to see what your external ip is, look at the output of register when you reload, or go to something like whatsmyip.com |
| 17:59.25 | polerin | (that's the ip you need to have people connect to) |
| 17:59.41 | *** join/#asterisk Fieldy (i=c9HzMhuP@gentoo/contributor/Fieldy) |
| 17:59.45 | polerin | btu you need to make sure the ports are forwarded, otherwise the router will just drop the packets |
| 18:00.08 | polerin | .. I think. SIP doesn't tunnel by default does it? |
| 18:00.15 | fab5freddy | polerin: so they would connect something like 70.65.2.67:5060 |
| 18:00.41 | polerin | yeah, if that's your ip. |
| 18:00.49 | polerin | btw |
| 18:00.54 | polerin | as you are on irc, just whois yourself :) |
| 18:01.12 | polerin | that hostname resolves to your external ip |
| 18:01.48 | fab5freddy | i did that but it's not a number ip |
| 18:02.11 | fab5freddy | can i use it just the same? |
| 18:02.23 | polerin | it's a host name. you can either nslookup <host name> or ping <host name> and it'll give you the numeric |
| 18:02.30 | polerin | and yes, you can just use it the same |
| 18:03.06 | cspot | also do /dns on that output to get ip |
| 18:04.09 | polerin | depends on your client, some don't have it. |
| 18:05.59 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 18:05.59 | Rhab | Hi allz |
| 18:06.24 | Rhab | I can receive faxes via asterisk->iaxmodem->hylafax |
| 18:06.37 | Rhab | but the tif files are somehow broken |
| 18:06.58 | Rhab | when i open them on a windows pc they cant be displayed |
| 18:07.17 | Rhab | any ideas what went wrong? |
| 18:08.30 | Rhab | the files are just about 30kb big |
| 18:08.38 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 18:08.58 | polerin | Rhab: can you open them on an *nix box/the gIMP? |
| 18:09.24 | polerin | (just curious, don't give any weight to my questions :P ) |
| 18:09.46 | Rhab | hm atm i dont have a possibility to check this |
| 18:09.57 | Rhab | only my server is running debian |
| 18:10.02 | Rhab | with no gui |
| 18:11.04 | fab5freddy | on nslookup i get Server: 192.168.2.1 |
| 18:11.17 | fab5freddy | Address: 192.168.2.1#53 |
| 18:11.21 | Rhab | is there any programm to check what is wrong with the images? |
| 18:11.45 | *** join/#asterisk Fieldy (i=2LR9uCQ7@gentoo/contributor/Fieldy) |
| 18:11.48 | polerin | fab5freddy: nslookup <hostname to look up> |
| 18:13.01 | *** part/#asterisk Fieldy (i=2LR9uCQ7@gentoo/contributor/Fieldy) |
| 18:13.05 | polerin | Rhab: well you can get the windows version of the gimp, and if it doesn't open I'd say something is corrupted, but again, i are newb. |
| 18:13.43 | red9012 | In using streaming music on hold. Does each music on hold class result in a new stream, or one stream can be shared among all classes? |
| 18:14.00 | Rhab | what would be a normal size for a one page fax? |
| 18:14.07 | Rhab | *file size |
| 18:15.54 | *** join/#asterisk SmaxTheFrog_ (n=M_A_X@91.64.25.7) |
| 18:16.26 | SmaxTheFrog_ | hi, what is the value for letting it ring endlessly if nobody takes the call..just empty ? |
| 18:17.56 | brian | Adding a Zap channel as an agent doesn't seem to be working. |
| 18:19.03 | garreel | BrianHV: do you mean adding a zap channel to a queue?... it should work |
| 18:20.44 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:22.31 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 18:31.03 | brian | garreel: I added a zap channel as a "agent" |
| 18:31.21 | brian | garreel: I added a zap channel as a queue member.... |
| 18:31.48 | brian | garreel: So I expect when I call and execute Queue() that it would take me to the Zap channel |
| 18:31.51 | brian | But that's not the case.... |
| 18:32.22 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-7b5136890c000103) |
| 18:32.22 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:32.58 | tzafrir_laptop | brian, please tell us what *does* happen, not what doesn't happen... |
| 18:33.07 | brian | http://rafb.net/p/LN7FOx28.html |
| 18:33.15 | brian | I have it set to queue ringing, right? |
| 18:33.21 | brian | Basically I just hear infinite ringing. |
| 18:33.28 | brian | There is the asterisk console output pasted for show queue |
| 18:34.00 | tzafrir_laptop | Zap/pseudo? |
| 18:34.05 | brian | yes |
| 18:34.17 | tzafrir_laptop | you did use real zaptel channels, right? |
| 18:34.17 | brian | That's the MeetMe conferences channel... |
| 18:34.29 | brian | tzafrir: no |
| 18:34.43 | brian | tzafrir: I'm using the hardware emulator kernel module |
| 18:35.18 | Rhab | Anyone here using hylafax?? How big is a usual fax in tiff format? |
| 18:35.26 | brian | Let me see something. |
| 18:36.55 | tzafrir_laptop | brian, hardware emulator? what do you mean? |
| 18:37.44 | Rhab | would be really helpful if some1 could just check the size of one of his faxes |
| 18:37.45 | brian | The kernel module that fakes the timer. |
| 18:37.53 | tzafrir_laptop | ztdummy? |
| 18:37.57 | brian | yes |
| 18:40.41 | brian | when I try add the SIP channel it says: |
| 18:40.47 | brian | SIP/fwd.pulver.com-08691000 (dynamic) (Invalid) has taken no calls yet |
| 18:43.32 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:44.05 | *** join/#asterisk Fieldy (i=7RDFHgxA@gentoo/contributor/Fieldy) |
| 18:48.43 | mDuff | When determining the MONITOR_FILENAME in use by a queue, are there any variables which indicate the channel of the agent who is accepting the line? |
| 18:49.03 | Rhab | Anyone here using hylafax?? How big is a usual fax in tiff format? |
| 18:49.12 | mDuff | Rhab: let me check. |
| 18:50.08 | mDuff | There's quite a bit of variance. Most are around 25K; a few are 60K; the largest is 228K. |
| 18:50.21 | Rhab | ok fine that helps |
| 18:50.44 | Rhab | i just configured everything and sent some test faxes |
| 18:51.07 | Rhab | the tif files were aroung 30kb but i couldnt open them |
| 18:51.13 | Rhab | with my windows pc |
| 18:51.24 | mDuff | Ahh. I convert 'em all to PDF before sending them to my users. |
| 18:52.11 | Rhab | i did that once but the pdf was broken and only 1K big |
| 18:52.35 | mDuff | ...and looking at it, the 228K one is several pages. |
| 18:52.50 | Rhab | u convert them with FaxDispatch? |
| 18:53.06 | Rhab | i mean u use FILETYPE=pdf? |
| 18:53.12 | mDuff | Rhab: No, FaxDispatch only looks up the destination email addresses and sets a few variables. The conversion happens... oh, yes. |
| 18:53.54 | Rhab | hm ill try to manually convert them an put them on samba share |
| 18:54.04 | Rhab | lets see if theyre ok |
| 18:54.13 | mDuff | I think I had some issues with the PDF conversion, but it was pretty obvious from the logs what went wrong, and easy to fix. |
| 18:54.20 | mDuff | Don't remember much more than that, though. |
| 18:55.15 | wunderkin | if you cant open the original file why would converting it help? |
| 18:55.19 | Rhab | wow, i got some weird output when i tried tiff2pdf |
| 18:55.27 | Rhab | hm dont know |
| 18:55.31 | mDuff | wunderkin: TIFF is a complicated format, and Windows doesn't necessarily parse it properly. |
| 18:55.41 | Rhab | ^^ |
| 18:55.43 | wunderkin | use another program? |
| 18:55.46 | mDuff | wunderkin: Well, let me restate that. TIFF has a lot of possible subformats, and they vary quite a bit. |
| 18:55.52 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 18:55.56 | wunderkin | the windows fax viewer thing works doesnt it? |
| 18:56.12 | tzafrir_laptop | Rhab, ImageMagick's programs (e.g: convert) should be able to handle multi-page tiffs and convert them to PDFs or whatever |
| 18:56.12 | Rhab | no its displaying nothing |
| 18:56.26 | mDuff | wunderkin: doing the conversion serverside means you don't need to worry about getting proper/compliant software on every single client that someone might ever want to use. |
| 18:56.42 | mDuff | wunderkin: also, doing the conversion with UNIX tools means you get useful logs, where as Windows software frequently... doesn't. |
| 18:56.43 | Rhab | imagemagick is for *nix? |
| 18:56.49 | mDuff | Rhab: it's cross-platform. |
| 18:57.08 | Rhab | ok ill check it |
| 18:57.10 | brian | I'm using the queue application. |
| 18:57.18 | tzafrir_laptop | Rhab, what distro do you se? there's probably a package of it in that distro |
| 18:57.21 | brian | How can I make the queue lead to a meetme conference? |
| 18:57.23 | mDuff | Rhab: Well, I'd start by seeing what's going wrong with tiff2pdf. |
| 18:57.37 | Rhab | first thing i want to find out is if my .tif files from the test faxes are broken |
| 18:57.54 | mDuff | Rhab: If they're not confidential, you can send them to me and I'll take a look. |
| 18:58.04 | Rhab | tzafrir_laptop i use debian 3.1 stable |
| 18:58.08 | mDuff | Rhab: I'm ccd@isgenesis.com |
| 18:58.29 | Rhab | not at all, i just copied some article |
| 18:58.40 | tzafrir_laptop | apt-get install imagemagick |
| 18:58.49 | Rhab | ok w8 |
| 18:59.07 | Rhab | ill have to relogin to access my samba share, brb |
| 18:59.12 | *** part/#asterisk Rhab (n=Rhab@p3EE0D4BF.dip.t-dialin.net) |
| 18:59.44 | mDuff | Brian: Do you want that if the queue times out, or do you want the conference to pick up all calls in the queue, or...? |
| 18:59.59 | brian | mDuff: I want the conference to pick up calls in the queue. |
| 19:00.42 | mDuff | Brian: Erm. Why are you using a queue at all, in that case? |
| 19:00.44 | brian | I tried adding the zap channel as a queue member but didn't work. |
| 19:00.49 | brian | because |
| 19:01.11 | brian | I only want two people in the conference at the same time |
| 19:01.40 | brian | There is one person sitting in the conference |
| 19:01.47 | *** join/#asterisk Rhab (n=Rhab@p3ee0d4bf.dip.t-dialin.net) |
| 19:01.51 | brian | I want there to be a queue to talk to this one person sitting in the conference |
| 19:01.53 | Rhab | ok |
| 19:02.13 | Rhab | maybe its just a stupid problem |
| 19:02.27 | Rhab | that i dont have the access rigth to the fax files |
| 19:02.29 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:02.34 | mDuff | Brian: Okay. Then why are you using a conference at all, in that case? Set the one person up as an agent; that way they have a persistent call that transitions between the other party. |
| 19:03.06 | brian | mDuff: Because if I set them up as an agent then it will hang up and then re-ring their phone for every caller. |
| 19:03.28 | Rhab | lol im just too newb |
| 19:03.30 | mDuff | Brian, no, it won't. That's if they're a member, not an agent. Just a minute. |
| 19:03.41 | mDuff | Brian, just a moment while I find some docs. |
| 19:03.42 | Rhab | i have to do some chmod i guess |
| 19:03.45 | brian | mDuff: I tried to add them as a member dynamically |
| 19:03.57 | brian | mDuff: And it just hung up after it added them as a member |
| 19:04.40 | brian | I don't care how it works, I tried everything (using MeetMe, etc) and I can't get this to work the way I want it too |
| 19:05.00 | mDuff | Brian: see AgentLogin (which is the behavior you want), and compare to AgentCallbackLogin (which is the behavior you don't want) |
| 19:05.12 | brian | Well |
| 19:05.19 | brian | The person is called by the Asterisk server |
| 19:05.43 | brian | I want the Asterisk server to call them, and then I want the Asterisk server to make them an agent. |
| 19:06.02 | brian | And then people call in I want it to queue each caller one by one to this one person! |
| 19:07.43 | mDuff | Brian, sounds reasonable, and I'm pretty sure chan_agent is the Right Way to do it. |
| 19:08.02 | Rhab | YEEEES its working! |
| 19:08.18 | Rhab | lawl |
| 19:08.24 | brian | Let's say I add them as a member... |
| 19:08.32 | *** join/#asterisk shinux__ (n=shinux@208.70.5.150) |
| 19:08.37 | brian | How do I get them on a agent channel?!? |
| 19:08.46 | brian | Or any channel... |
| 19:09.03 | brian | I don't even care what channel as long as the damn queue can queue callers to it! |
| 19:09.23 | mDuff | Brian: The agent channel is a member -- and you can even have it hardcoded as a member in queues.conf if you want. You log a person into the agent channel using AgentLogin. |
| 19:10.34 | *** join/#asterisk Caplain (n=Shayne@c-68-60-62-95.hsd1.mi.comcast.net) |
| 19:11.02 | Rhab | mDuff, do i have to change RecvFileMode to get more access? |
| 19:11.52 | Rhab | or change the faxmaster? |
| 19:13.17 | mDuff | Rhab: aren't your hooks running as the same user as faxq and hfaxd? If so, they shouldn't really be having permissions problems. That said, I personally tend to replace my faxrcvd scripts with stubs that strace the real scripts (strace -o /tmp/faxrcvd.strace -s 2048 -ff faxrcvd.real), to be able to find out exactly what's failing. |
| 19:13.33 | mDuff | erm. |
| 19:14.05 | mDuff | strace -o /tmp/faxrcvd.strace -s 2048 -ff "$(dirname "$0")/faxrcvd.real" "$@" |
| 19:14.09 | mDuff | something like that. |
| 19:14.37 | Rhab | hm i think i run everything as root |
| 19:15.05 | Rhab | but in FaxDispatch i send the mail to another user |
| 19:15.19 | mDuff | Rhab: really? Most hylafax packages I'm aware of install and use non-root users for security. |
| 19:15.29 | Rhab | uucp? |
| 19:15.55 | Rhab | mbe ur right and i dont know about it |
| 19:15.57 | mDuff | uucp, or a new one called fax or hylafax or something. Depends on what the person who packaged it for your distro ate for breakfast that morning. |
| 19:16.10 | Rhab | ok im using uucp |
| 19:16.17 | Rhab | for iaxmodem and stuff |
| 19:16.49 | mDuff | It'll be the hfaxd or faxq process that you care about in this case. Use ps or somesuch to see what they're running as. |
| 19:17.02 | brian | mDuff: That will hang up |
| 19:17.22 | brian | mDuff: And ring every time a new call comes in from the queue |
| 19:17.37 | brian | mDuff: I want the agent to stay on the line and go through the queue |
| 19:18.02 | mDuff | Brian: chan_agent stays online and plays music-on-hold between calls if you use AgentLogin instead of AgentCallbackLogin. |
| 19:18.18 | Rhab | puh,i just wanna have all the faxes mailed to a user and access them via imap from the clients |
| 19:18.27 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 19:19.30 | *** join/#asterisk Nase (n=Nase@p54bc87fa.dip0.t-ipconnect.de) |
| 19:20.03 | Rhab | ls -la says: -rw------- 1 uucp uucp 31744 2007-04-02 21:53 fax000000001.tif |
| 19:20.18 | mDuff | Rhab: that's fine, though, if your hook scripts are executing as the uucp user -- which they should be. |
| 19:20.23 | Nase | hi everybody |
| 19:21.04 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 19:21.14 | Rhab | what u mean by hook scripts? |
| 19:21.35 | Rhab | the conversion to pdf? |
| 19:21.37 | mDuff | Rhab: incoming calls are handled by hooks, typically/especially faxrcvd |
| 19:21.47 | Nase | I'm an asterisk-newb. |
| 19:21.48 | mDuff | Rhab: yes, the hooks are responsible for actually doing the PDF conversion. |
| 19:22.33 | Rhab | ok, how can i configure hylafax to grant rights for the fax file to my user |
| 19:22.40 | mDuff | Rhab: You can specify a different hook to use for incoming faxes by specifying FaxRcvdCmd in your config.ttyIAX* files, but most folks don't. |
| 19:23.04 | mDuff | Rhab: The hook script runs as the uucp user. The uucp user has read permissions. So permissions are fine. |
| 19:23.22 | Nase | can anyone tell me if it is possible to route the signal of the line-in of my sound-card into asterisk? is that something like Console/dsp? |
| 19:23.52 | Rhab | okay but when the fax is send to my user (SENTO=xam, in FaxDispatch) this user lack rights |
| 19:23.54 | mDuff | Rhab: What you need to do is figure out why else the scripts are failing, which is why I've been urging you to trace them with strace (or something simpler, if you're so inclined -- just doing a "set +x" and redirecting stderr to a file couldn't hurt) |
| 19:24.14 | mDuff | Rhab: Why would xam need to have rights to the file? uucp is the user sending the email. |
| 19:24.46 | Rhab | yeah sry, im just a little slow and not familiar with this stuff |
| 19:24.55 | *** join/#asterisk leejohn (n=itsme@58.69.34.238) |
| 19:26.59 | leejohn | hi! good day, anyone here? i have a toll free service from my telco, i have a little problem as soon as asterisk starts to answer the call they charge me although there's no one on the phone |
| 19:27.37 | wunderkin | don't use Answer then |
| 19:27.52 | *** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 19:27.57 | leejohn | if i don't use exten => xxxxx,1,Answer there's no audio |
| 19:28.11 | Rhab | mDuff the faxfiles attached to the notification email are just 1kb, but should be 30 |
| 19:28.46 | mDuff | Rhab: Right. Which is why you need to figure out what's going wrong. Which is why generating an strace and reading through it would be a Good Idea. |
| 19:29.28 | Rhab | I try hard to use this strace but im totaly newb with it |
| 19:29.34 | Rhab | give me some minutes |
| 19:29.35 | *** join/#asterisk sh4g (n=jsb@l007090.zseriespenguins.ihost.com) |
| 19:29.35 | wunderkin | leejohn, you mean no ringing and then when you pick up there is audio |
| 19:29.52 | leejohn | wunderkin: no audio at all |
| 19:30.29 | wunderkin | and when you use Answer there is audio? that does not make sense |
| 19:30.29 | *** part/#asterisk sh4g (n=jsb@l007090.zseriespenguins.ihost.com) |
| 19:30.32 | Caplain | what file format are the sounds that the Playback command uses? |
| 19:30.40 | leejohn | wunderkin: yes there's an audio |
| 19:31.13 | polerin | Caplain: I know they can use gsm, they may be able to use others as well |
| 19:31.25 | *** join/#asterisk p4r14h (n=j0sh@69.92.145.178) |
| 19:31.35 | Caplain | okay |
| 19:31.35 | mDuff | Caplain: there are a lot of formats supported; the big difference between them is size and performance. You can ask Asterisk for a list; "core show codecs" should give you one (though it may not be quite all of those) |
| 19:31.57 | Caplain | okay |
| 19:32.06 | mDuff | Caplain: "core show translation" will give you an idea of the performance differences between those codecs. |
| 19:32.48 | mDuff | Caplain: slin tends to be fastest, since it's what newer Asterisks are using internally. Also, note that not quite all of them in that list are supported for file operations -- speex, for instance, isn't -- but most are. |
| 19:33.31 | Caplain | well say i have an mp3 and i want to sue that in a playback command, what file type do i need to convert it to? |
| 19:34.00 | Caplain | oh i found a link |
| 19:34.02 | leejohn | Caplain: if you are using 1.2 then i suggest use rawplayer |
| 19:34.05 | Caplain | nm thanks anyways |
| 19:34.15 | MrWup | ok heres an idea i got: PSTN caller -> asterisk -> SIP user (SIP user is talking to PSTN caller) i want to press a button on my SIP handset (aastra 9133i) and the PSTN will immediately be transfered to someone else's voicemail |
| 19:34.18 | MrWup | can it be done? |
| 19:34.41 | Caplain | oh, another thing is when i play mp3's via moh or mp3player the sound is distorted and loud |
| 19:34.46 | Caplain | is there a way to fix that? |
| 19:35.29 | leejohn | Caplain: since i don't use mp3player i don't really know, but i do suggest rawplayer instead, it works fine for me though you should convert your mp3 into raw format |
| 19:35.49 | Caplain | okay |
| 19:36.04 | leejohn | wunderkin: i will try to use 1.4 for my issue do you think it will work? |
| 19:36.32 | wunderkin | leejohn, uh... i dunno.. i have nfi so i left it for someone else |
| 19:36.46 | leejohn | wunderkin: ok thanks anyway man |
| 19:37.15 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 19:38.07 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 19:45.33 | Nase | newb question: is there a way to "route" the soundcards line-in signal to a caller? |
| 19:45.53 | Nase | or do i need a softphone for that? |
| 19:46.37 | Nase | i want people to call in, authenticate and here what is currently spoken into the microphone. |
| 19:46.50 | Nase | is that possible? |
| 19:47.30 | Nase | oops. people should _hear_ of course ;) |
| 19:49.22 | Nase | yes or no would be ok :) |
| 19:50.00 | polerin | Nase: probably is but just out of curiosity, what is your intent? |
| 19:50.13 | polerin | oh, sorry, nm you said that. .. |
| 19:52.10 | Nase | hmm. what's the general direction to go there? the Console channel with ALSA? |
| 19:52.46 | Nase | can i Play(Console/dsp) ? or some thing like that? |
| 19:53.52 | polerin | not sure. newb :) |
| 19:53.57 | polerin | <-- newb even |
| 19:54.08 | Nase | :) |
| 19:55.23 | Nase | i've read dozens of books and wikis and faqs today - i didn't find anything |
| 19:57.02 | Nase | well, at least i've got a better overall understanding of asterisk now ... ;) |
| 19:59.58 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 20:01.08 | polerin | lol, yeah that does happen when you are RTFMing :) |
| 20:01.13 | *** join/#asterisk lachesis (n=lachesis@ASt-Lambert-151-1-31-78.w82-124.abo.wanadoo.fr) |
| 20:01.15 | *** part/#asterisk Rhab (n=Rhab@p3ee0d4bf.dip.t-dialin.net) |
| 20:02.00 | *** join/#asterisk bawb2 (n=bawb2@ppp-70-248-21-90.dsl.tpkaks.swbell.net) |
| 20:02.54 | zapp-branigan | i have one error in the gumstix |
| 20:02.58 | zapp-branigan | (does /var/run/asterisk.ctl exist?) |
| 20:03.03 | Qwell | zapp-branigan: does it? |
| 20:03.15 | zapp-branigan | the file is present |
| 20:03.27 | zapp-branigan | and show the number of pid |
| 20:03.27 | etfonhomey | Is there an extra sounds package that I'm forgetting? I can't find enter-ext-of-person.gsm in my new 1.4.2 installation. Any ideas? |
| 20:03.29 | Qwell | Does the user you're running asterisk as have permission to it? |
| 20:03.35 | zapp-branigan | root |
| 20:04.18 | zapp-branigan | 0777 |
| 20:04.20 | red9012 | In using streaming music on hold. Does each music on hold class result in a new stream, or one stream can be shared among all classes? |
| 20:04.33 | leejohn | Qwell: can i ask a question please, can i use a progress apps together with ringing apps on sip channel so that when there's no one answer the call my telco won't bill me ? |
| 20:05.05 | Qwell | No, you don't need to use those.. the device in question will provide the tones |
| 20:06.04 | leejohn | Qwell: so how can i do this, if i don't use answer apps the caller doesn't hear ringing |
| 20:06.27 | Qwell | Just Dial the user, and it'll ring for him, unless the provider (or you) is doing something silly |
| 20:06.32 | MrWup | if i wanted to have a button which when pressed would blind xfer the caller to someone's voicemail, would i need to use an application map? |
| 20:07.08 | leejohn | Qwell: if i do that's yes the sip phone is ringing but the caller doesn't hear a ring is this the intended behavior? |
| 20:07.25 | Qwell | no, if you're answering or something, they won't hear ringing |
| 20:07.57 | *** part/#asterisk Nase (n=Nase@p54bc87fa.dip0.t-ipconnect.de) |
| 20:08.43 | *** join/#asterisk stkn_ (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 20:09.05 | leejohn | Qwell: your are correct, if i don't answer the incoming channel using the answer apps the sip phone rings, but the caller can't hear that it's ringing |
| 20:09.07 | luke-jr | http://www.AllKnowingBall.com/question.php?pageid=3684 |
| 20:09.17 | luke-jr | :o |
| 20:09.20 | Qwell | leejohn: then your provider is doing something stupid |
| 20:09.38 | Qwell | that or you're doing something silly, like perhaps Dial()ing with ,r |
| 20:09.59 | leejohn | Qwell: thank you, i understand now, that they should do it, nope i don't use the option r |
| 20:10.05 | wunderkin | he also said previously that if he does not use Answer then there is no audio when he picks up the phone? really seems like a communication problem here |
| 20:10.21 | leejohn | wunderkin: yes, it's my fault |
| 20:10.46 | leejohn | wunderkin: i can tell that there's an audio if i don't use answer apps, i tried it a while ago you are correct |
| 20:13.31 | Qwell | what kind of audio? |
| 20:13.49 | Qwell | like, the actual call? |
| 20:13.53 | leejohn | Qwell: yes |
| 20:14.06 | Qwell | yeah, they're doing something stupid then... |
| 20:14.43 | leejohn | Qwell: so i will pursue this issue's to them, thanks a lot |
| 20:15.29 | lachesis | Hi, I have set up an IVR on my pbx but I have a problem with DTMF recognition. I had dtmfmode=auto in sip.conf. DTMF works well when I call with my IAX softphones, but not when I call with a regular phone (not voip). If I put dtmfmode=inband, then DTMF recognition works with the regular phone, but I am afraid it's going to be unreliable or that mobile phone user might not be able to type in dtmf... Any idea about what i |
| 20:16.21 | lachesis | I thought dtmfmode=auto meant that asterisk would switch to inband when necessary, but apparently it does not... |
| 20:18.44 | leejohn | lachesis: how about setting this to rfc2833? |
| 20:18.53 | lachesis | oh... I just notice I have dtmfmode=rfc2833 in the incoming sip trunk.... maybe that's the problem... |
| 20:19.02 | lachesis | well I tried everything. |
| 20:19.16 | lachesis | only inband works when the regular call in |
| 20:19.31 | leejohn | lachesis: you are using 1.2 or 1.4? |
| 20:19.40 | lachesis | I'm going to try to remove the dtmf line from the sip trunk. |
| 20:19.44 | lachesis | 1.2 |
| 20:20.40 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:20.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:24.18 | lachesis | nope, it's not working... I tried to change dtmfmode in the trunk setting but it didnt change anything.... |
| 20:26.12 | *** join/#asterisk jd98 (n=jdio@ip70-172-219-188.ri.ri.cox.net) |
| 20:27.03 | *** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
| 20:28.39 | lachesis | I dont get it... |
| 20:28.41 | trevarthan | Hello. I'm using asterisk 1.2.x. I need to randomize some IVR menu paths, but the Random() dialplan cmd looks like a poor choice, considering that I'll be adding new paths on a regular basis and each path needs an equal percent chance of being triggered per call. What's the best way to do this? |
| 20:29.50 | Innatech | maybe farm it out to adhearsion, if you're feeling adventurous? |
| 20:30.03 | Innatech | or some other external helper.... |
| 20:30.36 | *** join/#asterisk [hC] (n=hardcore@adsl-63-200-45-107.dsl.snfc21.pacbell.net) |
| 20:31.23 | lachesis | it works only if I put dtmfmode=inband in sip.conf. If I try to put dtmfmode=inband anywhere else (like in the trunk setting) or use the SIPDtmfMode Application, then it wont switch to inband... |
| 20:31.36 | lachesis | I guess sip.conf might have the last word... |
| 20:32.01 | trevarthan | Innatech: Are you talking to me? |
| 20:32.05 | leejohn | are you receiving a call from a sip channel or zap? |
| 20:32.05 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 20:32.18 | leejohn | lachesis: are you receiving a call from a sip channel or zap? |
| 20:32.48 | leejohn | lachesis: oic, sip trunk |
| 20:33.28 | lachesis | yes, sip well what i didnt say i that it is an outgoing call => it's a callback application. |
| 20:33.44 | lachesis | but i dont thing it matter |
| 20:34.46 | *** join/#asterisk omm (n=omm@rrcs-70-61-148-55.central.biz.rr.com) |
| 20:34.48 | leejohn | lachesis: are you aware that you can't use inband signalling if you are using a loose codec like gsm and g.729? what codec you are using? |
| 20:35.21 | lachesis | yes but well the thing is that only inband works. |
| 20:35.33 | omm | Anyone successful with queues and penalties? |
| 20:36.10 | lachesis | but I have to set dtmfmode=inband in sip.conf and then it works but I wish to keep out-of-band dtmf for phone that needs it (or support it ) |
| 20:38.32 | lachesis | What I would like is to find out why dtmfmode=auto does not work. If I understood properly, it would take outofband dtmf by default but switch to inband when necessary. |
| 20:40.14 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
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| 20:41.40 | Innatech | trevarthan: yes, I was. |
| 20:42.49 | lachesis | hmmm.... it seems the problem is with the callback. |
| 20:43.03 | lachesis | well I call in, it works. |
| 20:43.52 | trevarthan | Innatech: I wouldn't mind writing a Perl script or something, but I don't want to use a big complicated framework for this simple issue. However, I don't know how to run an external Perl script. The System() cmd doesn't allow me to return a value... |
| 20:44.07 | lachesis | softphone using sip-trunk1 call a regular phone and plays a menu = regular phone cant input dtmf. |
| 20:44.24 | lachesis | no sorry I messed up my sentance. |
| 20:44.26 | [TK]D-Fender | trevarthan, Time to make an AGI |
| 20:44.51 | trevarthan | [TK]D-Fender: yeah, I was just looking at that... |
| 20:45.52 | Innatech | trevarthan: Yup, AGI is cool. Still, this might be worth a look, if you enjoy Ruby: http://adhearsion.com/ |
| 20:46.10 | Innatech | I've only read about it thus far, but it looks attractive. |
| 20:46.39 | trevarthan | Innatech: I don't enjoy Ruby. I know Perl, PHP, C, C++, C#, Shell, Python, OCaml, and a few others. I don't know and don't want to know Ruby at this time. |
| 20:46.55 | Innatech | Well, there ya go. Easy decision! |
| 20:47.09 | omm | Is there a way to log out static agents? I want to use penalties to keep the call order static... (always starts with the first agent)... But penalties always calls the first phone even when its NO ANSWER.... Should I auto DND that user or log them out of the queue maybe pause them? |
| 20:47.53 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:49.02 | [TK]D-Fender | omm, no, thats what static means. You can PAUSE them however, |
| 20:50.22 | omm | I would like to test the PAUSE and see if that will ring the next agent |
| 20:50.57 | omm | if so then I need to figure out how to program the Polycoms soft button to toggle pause and auto pause |
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| 20:51.55 | omm | [TK]D-Fender, is there a program that can monitor paused agents? |
| 20:52.11 | omm | Call center manager wants to see when agents are away from the phone. |
| 20:52.27 | [TK]D-Fender | omm, if you're running polycom, use a microbroswer script to show their status there. |
| 20:52.47 | omm | I can do that on the 501s |
| 20:52.47 | [TK]D-Fender | omm, as for mapping a key, leave a line-key clear and make a speed-dial for it. |
| 20:52.59 | omm | ok |
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| 21:12.47 | lachesis | Ok now I understand better my problem : on outgoing call, asterisk wont hear dtmf from the receiving telephone. (it's a problem for my callback application). I found forum posts where it says the solution is to put exten => s,n,SIPDtmfMode(inband) in a context of the call but it wont work with me... |
| 21:16.01 | [TK]D-Fender | lachesis, What does your provider use? |
| 21:16.11 | omm | Is there a way to auto DND a user after no answer? |
| 21:16.46 | [TK]D-Fender | omm, yes its in queues.conf . this is not DND, this is auto-pause |
| 21:17.04 | omm | ok |
| 21:17.07 | omm | I'll look |
| 21:17.42 | lachesis | Well, I am not sure : if I have dtmfmode=auto, it works well on incoming call, I dont have problem to recognize dtmf. |
| 21:18.08 | lachesis | it's only on outgoing call that it does not work. |
| 21:18.14 | [TK]D-Fender | lachesis, Set a fixed mode and be done with it. This is not something that CHANGES between you and your provider. |
| 21:21.35 | lachesis | well ok, I put inband on in the trunk setting and it works but is not that a problem if I try to use a mobile phone ? |
| 21:21.48 | *** part/#asterisk SmaxTheFrog_ (n=M_A_X@91.64.25.7) |
| 21:22.10 | lachesis | I understood that mobile phone sent out-of-band dtmf... |
| 21:22.20 | trevarthan | [TK]D-Fender: I'm reading up on asterisk AGI. I think I'll use Perl for it's ubiquity under linux. All I need is a random number right now, so I plan to just use SET VARIABLE within the AGI script and exit. Will exiting the AGI script cause any problems? Or will control simply return to asterisk's dialplan at priority + 1? |
| 21:22.49 | [TK]D-Fender | trevarthan, the latter. you'll do just fine. |
| 21:23.01 | trevarthan | cool. thanks. |
| 21:26.46 | russellb | trevarthan: if all you need is a random number, why not just use UNIQUEID ? |
| 21:27.17 | trevarthan | russellb: I don't need a unique number. I need a random number between 1 and 10 or 1 and 5 or whatever. |
| 21:27.50 | russellb | ah, well then I think there is also a RANDOM() dialplan function |
| 21:28.11 | russellb | core show function RAND |
| 21:28.35 | trevarthan | russellb: yes, but imagine trying to use Random() that will 40 different possible branches, then adding a branch once per day. Not pretty or easy. |
| 21:29.03 | trevarthan | russellb: The RAND function is only available in 1.4+ and I need to target 1.2.x. |
| 21:29.07 | russellb | ah |
| 21:29.34 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 21:29.36 | russellb | alrighty, i'll shut up then :) |
| 21:29.55 | trevarthan | frankly, a round-robin approach would work equally well for me, but I don't see an easy way to do that either. |
| 21:30.46 | trevarthan | it's OK. AGI looks easy enough. perhaps not as efficient as I would like, but it should work. |
| 21:31.10 | russellb | well, for round robin, you could keep a number in the astdb |
| 21:31.33 | russellb | and just increment it on each call or something |
| 21:32.16 | russellb | or just use AGI and be done with it, heh |
| 21:33.50 | trevarthan | russellb: actually, the astdb is not a bad idea at all. No external deps either. I like it. |
| 21:34.44 | trevarthan | I considered using MySQL, but rejected it because I didn't want to deal with usernames and passwords and such. I like the astdb idea though. |
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| 21:36.02 | russellb | the only issue i guess is that you can't really ensure that the operation of getting the number, and then incrementing it, is "atomic" |
| 21:36.10 | russellb | so multiple calls may get the same number at the same time ... |
| 21:36.38 | russellb | in 1.4 (i think, maybe just trunk), there is a MacroExclusive application that would solve that |
| 21:37.21 | trevarthan | russellb: I can live with lack of atomicity in this application. In fact, I prefer it because I really don't trust random numbers to be true statistically random numbers in linux..... |
| 21:37.44 | russellb | heh, that's cool |
| 21:37.48 | russellb | well, i'm off ... good luck |
| 21:37.57 | trevarthan | later. thanks for the help. |
| 21:38.04 | russellb | no problem |
| 21:38.50 | Innatech | ooh, I'm shoving this scrollback into my .txt repository of doom. That's good stuff. |
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| 21:44.31 | trevarthan | how do I run functions in the asterisk CLI? |
| 21:44.51 | trevarthan | I see `show functions`, but how do I run them? |
| 21:50.20 | [TK]D-Fender | trevarthan, Don't think you can. the CLI is not a fully functioning dialplan engine... |
| 21:50.40 | trevarthan | ok. np. thanks. |
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| 21:54.33 | mmartinn | What purpose does WaitEvent serve in the Manager interface? |
| 22:00.27 | Qwell | mmartinn: To wait for an event |
| 22:00.49 | Qwell | We use it in the GUI for monitoring for updates, so we can show channels in use and such |
| 22:01.19 | Qwell | of course, with the GUI, it opens closes a rawman connection to do it |
| 22:01.21 | mmartinn | Qwell: But if you are just sitting, connected to the manager interface, it sends you events anyway |
| 22:01.32 | Qwell | not always |
| 22:01.46 | Qwell | I mean, not always - as in, you won't always be "just sitting" |
| 22:01.59 | mmartinn | Is this more for the HTTP interface to it? |
| 22:02.23 | Qwell | I think it may have existed before that, but yeah, it works well for that |
| 22:02.43 | mmartinn | I can connect to the manager interface and just not run any commands it spits out events all day long |
| 22:02.52 | Qwell | sure |
| 22:02.54 | mmartinn | What does it not deliver without WaitEvent? |
| 22:04.40 | *** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net) |
| 22:10.56 | *** join/#asterisk kn0x (n=pinochle@c-67-176-194-29.hsd1.il.comcast.net) |
| 22:11.00 | kn0x | hey |
| 22:11.14 | kn0x | do i need ztdummy for anything besides meetme |
| 22:11.41 | kn0x | ive allways installed zaptel, but i have a vps that wont take it.. |
| 22:12.08 | kn0x | wat actually uses ztdummy besides meetme? |
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| 22:14.02 | Innatech | kn0x: not sure, but I do know that there is a way to support ztdummy on VPS's. There's a provider that focuses on hosted FreePBX that has posted about it a bunch. |
| 22:15.43 | Innatech | kn0x: the first few google results for "ztdummy vps" seem useful. |
| 22:15.53 | kn0x | well its not that exactly |
| 22:16.10 | kn0x | i have another problem |
| 22:16.33 | kn0x | i have a client with a fedora core 4 box, w/o kernel sources |
| 22:17.08 | kn0x | the other problem was another client who has a virtuozoo box that wont accept kernel modules |
| 22:17.12 | kn0x | its not reallly a vps |
| 22:17.17 | Innatech | Ah. |
| 22:17.17 | kn0x | its a psuedovps |
| 22:17.23 | kn0x | but for this one |
| 22:17.38 | Innatech | The solution lylix.net uses for ztdummy is for virtuozo, IIRC. |
| 22:17.39 | kn0x | i cant get the kernel sources (its an obscure version) |
| 22:18.11 | Innatech | so far as the Fedora box goes.....I'm afraid I can't offer any advice. I solve Fedora problems with an Ubuntu LTS install CD. |
| 22:18.45 | Innatech | And I rely on plenty of help to get CentOS to do my bidding when I must. |
| 22:18.59 | kn0x | lol |
| 22:19.23 | kn0x | do you know though, if asterisk can run fine w/o ztdummy? |
| 22:19.37 | kn0x | im just using it to run an ivr and some other things |
| 22:19.51 | mmartinn | Yes, Asterisk definitely runs without ztdummy |
| 22:19.57 | Innatech | I feel like saying yes, its OK--but I'm not an authority . |
| 22:20.17 | mmartinn | I have it running in a vmware image right here, without any zt drivers |
| 22:20.46 | kn0x | cool |
| 22:21.03 | kn0x | yeah i mean, i've never had to worry about not installing zaptel |
| 22:23.23 | kn0x | ohh an INnatech, i dont think lylix will be willing to show me how to do ztdummy on the virtuozo box |
| 22:24.20 | Innatech | well....they've posted about how its done in a number of places. |
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| 22:25.47 | kn0x | ohh really |
| 22:25.49 | kn0x | hm |
| 22:26.49 | polerin | kn0x: it will run, but it needs it for some things (music on hold, and a couple others) |
| 22:27.25 | polerin | if you don't need a feature that requires it, i think it will be fine <grain of salt, still a newb here ;) > |
| 22:28.07 | Innatech | kn0x: this might be worth a gander : http://www.telephreak.org/papers/vpa/ "Virtual Private Asterisk." |
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| 22:36.48 | polerin | hmm |
| 22:36.54 | polerin | exten=> s,n,Noop(foo), lable_name |
| 22:37.12 | polerin | that's the right way to set a lable of "lable_name" right? |
| 22:37.42 | Qwell | s,n(label_name),DoStuff() |
| 22:37.51 | polerin | ah, braindead, thanks |
| 22:41.07 | kn0x | Innatech- thats the problem though |
| 22:41.23 | kn0x | this is a Virtuozo vps, with a shared kernel |
| 22:41.30 | kn0x | so i dont have accesss to load kernel modules |
| 22:41.48 | Innatech | right. Hold on, let me try and find the posts I saw referencing Virtuozo. |
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| 22:44.40 | polerin | exten => s,n(menu_entry),Wait(1) |
| 22:44.56 | polerin | still saying menu_entry is an invalid label :/ |
| 22:45.40 | Innatech | kn0w: take a look here, where jd from lylix argues with "bubba." He talks about a shared ztdummy under virtuozzo being better than a vmware solution. http://www.trixbox.org/forums/trixbox-forums/open-discussion/hosting-installing-trixbox-vps-or-vds |
| 22:45.59 | polerin | hmm... the label doesn't have to be declared before the gotoiftime() does it? |
| 22:47.46 | Innatech | kn0x , that is. not kn0w. |
| 22:47.46 | InHisName | Can anybody interperet the error: Apr 7 12:48:33 WARNING[1383]: chan_sip.c:9745 handle_response_register: Got 404 Not found on SIP register to service 3145630068@sunrocket.com, giving up ? |
| 22:49.27 | Innatech | kn0x: from another poster, same thread : In an OpenVZ VPS you have root access, but you can not load/unload kernel modules. This is not a problem, because the node administrator can grant you to access Zaptel, which runs once at the hardware node level. |
| 22:51.38 | *** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 22:53.37 | Innatech | kn0x: one more link : http://www.4psa.com/docs/voipnow/voipnow_virtuozzo_integration.html |
| 22:57.32 | kn0x | Innatech- hey thanks a lot |
| 22:57.37 | kn0x | that is interezting |
| 22:58.05 | Innatech | no prob. I thought so too. |
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| 23:47.35 | *** join/#asterisk linagee_ (n=linagee@unaffiliated/linagee) |
| 23:47.39 | linagee_ | hello! :) |
| 23:48.10 | linagee | is it possible to run an SMS gateway? |
| 23:48.36 | linagee | but probably not what you'd normally think. in reverse. to receive SMS messages for your DIDs |
| 23:48.59 | linagee | or is that some elite club that only cingular, tmobile, etc. belong to? |
| 23:49.41 | ManxPower | linagee: Actually yes. |
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| 23:50.36 | ManxPower | In other parts of the world cell carriers are more open to interconnection with large customers. It is not like that in the USA |
| 23:50.39 | linagee | ManxPower: O_o |
| 23:51.02 | linagee | ManxPower: if i put asterisk into debug mode and sent an SMS to my DID, would i get anything? |
| 23:51.12 | linagee | (and waste $0.10) |
| 23:51.32 | ManxPower | you'll waste a lot more than 10 cents trying to make it work. |
| 23:51.40 | linagee | LOL |
| 23:52.11 | `p4r14h | send the SMS messages from google SMS :D |
| 23:53.01 | linagee | ok so uhm. of course. nothing came though. :) |
| 23:53.02 | linagee | lol |
| 23:53.03 | linagee | :( |
| 23:53.14 | linagee | ManxPower: is it some elite club? should i give up trying now? heh |
| 23:53.35 | ManxPower | linagee: most of the "SMS" info you see on the web is about EUROPE and the rest of the world. |
| 23:53.38 | linagee | ManxPower: i'd imagine it just uses TCP/IP or something |
| 23:54.01 | linagee | ManxPower: like some server is listening on a port. says, "i have these DIDs here" |
| 23:54.08 | ManxPower | Hmm? No. |
| 23:54.12 | linagee | heh |
| 23:54.19 | russellb | lol ... |
| 23:54.30 | ManxPower | In the rest of the world you call the SMSC's modems and do a 1200bps data burst. |
| 23:54.31 | russellb | if only it was that easy |
| 23:54.34 | linagee | so you mean http://www.voip-info.org/wiki-Asterisk+cmd+Sms is europe stuff? |
| 23:54.41 | ManxPower | linagee: YES! |
| 23:54.46 | linagee | ah |
| 23:55.09 | blitzrage | no, ManxPower love lieing to people |
| 23:55.11 | blitzrage | I see it all the time |
| 23:55.18 | blitzrage | :) |
| 23:55.39 | ManxPower | linagee: Perhaps you missed http://www.voip-info.org/wiki/view/SMS+Service+in+the+USA which is the first link on the page http://www.voip-info.org/wiki-Asterisk+cmd+Sms |
| 23:55.45 | linagee | blitzrage: suuure you are. :) |
| 23:56.08 | linagee | oh whoops. yep. i did. |
| 23:56.16 | linagee | thx |
| 23:56.52 | linagee | ManxPower: wow. hah. the info on that page seems so vague. |
| 23:57.06 | linagee | ManxPower: i guess that's how they're able to keep charging $0.10/message. "it's magic" :) |
| 23:57.15 | ManxPower | linagee: now you begin to see how far behind the USA is compared to the rest of the world when it comes to telecom |
| 23:57.39 | linagee | ManxPower: sending out might be possible. find that carrier's gateway or web page gateway or email gateway |
| 23:57.52 | linagee | ManxPower: but send/receiving into the sms system itself (like in the "raw" i guess you'd call it..) |
| 23:57.57 | ManxPower | In most of the rest of the world incoming calls to a cell phone are free to the cell phone user. The CALLER pays a premium rate for calling a cell number. |
| 23:58.14 | linagee | ManxPower: wow. that's like a normal landline call |
| 23:58.25 | linagee | the caller has the pay burden |
| 23:58.36 | ManxPower | linagee: Um, SMS is a very specific protocol. |
| 23:58.55 | ManxPower | linagee: it is also why cell phones are more common in europe than in the usa |
| 23:58.56 | linagee | ManxPower: "very specific" = proprietary = $$$. did i get that just about right? :) |
| 23:59.03 | linagee | lol |
| 23:59.39 | ManxPower | linagee: http://smsforum.net/doc/download.php?id=smppv34 |