| 00:00.54 | tzafrir_laptop | NightMonkey, the dialtone to the caller is given until someone answers. |
| 00:00.56 | tzafrir_laptop | I'm not exactly sure if with that adapter Asterisk can control when it will answer |
| 00:01.59 | NightMonkey | tzafrir_laptop: What's odd is that I do have context=incoming set up for that FXO's extension, but Asterisk isn't following that dialplan, but just giving a dial tone. |
| 00:02.15 | tzafrir_laptop | Sending it to the console is simple, though: it's a sip device for asterisk. So in sip.conf you give it the proper context name (context=target_context) |
| 00:03.02 | NightMonkey | tzafrir_laptop: I think that's what I did. Let me recheck. |
| 00:03.09 | tzafrir_laptop | NightMonkey, is the FXO port reigstered with Asteirsk? |
| 00:03.19 | NightMonkey | tzafrir_laptop: Yep. |
| 00:03.29 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 00:03.47 | tzafrir_laptop | in 'sip show users', does it show the right context name? |
| 00:05.43 | NightMonkey | tzafrir_laptop: I broke out the extension into <extension>_peer and <extenrion>_user (with type=peer and type=user, respectively). And, the <extension>_user shows up... but not the <extension>_peer... ah. |
| 00:06.00 | NightMonkey | (in sip.conf) |
| 00:06.37 | tzafrir_laptop | this is 'sip show users' . IT only shows users, not peers. |
| 00:06.56 | NightMonkey | tzafrir_laptop: Ah, duh. |
| 00:07.10 | NightMonkey | tzafrir_laptop: Thanks. ;) |
| 00:07.31 | NightMonkey | tzafrir_laptop: "sip show peers" shows the peer, too. |
| 00:07.46 | tzafrir_laptop | next thing to do: |
| 00:07.50 | tzafrir_laptop | set verbose 3 |
| 00:07.54 | tzafrir_laptop | and call in |
| 00:08.00 | tzafrir_laptop | do you see anything? |
| 00:08.13 | tzafrir_laptop | in the asterisk CLI, that is (asterisk -r) |
| 00:08.20 | NightMonkey | tzafrir_laptop: OK, one moment. |
| 00:09.25 | *** part/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 00:10.20 | NightMonkey | tzafrir_laptop: Ah, no output - I think I have a problem with the gateway - I'll troubleshoot that first. |
| 00:11.21 | tzafrir_laptop | not necessarily |
| 00:11.30 | tzafrir_laptop | no channel was created |
| 00:12.08 | NightMonkey | tzafrir_laptop: This time, the call just rang and rang, no pickup. |
| 00:12.11 | tzafrir_laptop | one option is still that it has failed to authenticate or something. Though I believe you should have seen a message about that. Not sure |
| 00:12.18 | tzafrir_laptop | try: sip debug |
| 00:12.28 | tzafrir_laptop | (warning: very verbose) |
| 00:12.37 | tzafrir_laptop | stopping that: sip no debug |
| 00:13.23 | NightMonkey | tzafrir_laptop: The device reports that it has registered, both the FXS and FXO. I'll try the debug approach. (thanks for the help, btw) |
| 00:22.22 | *** join/#asterisk niedobry (n=bbrindle@ip24-254-142-122.rn.hr.cox.net) |
| 00:24.53 | harlequin516 | I have my sip fxs device setup. It can interact locally with my asterisk server and a softphone connected to it, everything works fine. I have a problem when calling out from the fxs device calling out to the internet. Everything seems find and the calls connect, but then Asterisk says attempting native bridge and sound does not travel in either direction. |
| 00:26.15 | harlequin516 | Does anyone have a solution to make festival play like background instead of like playback |
| 00:26.17 | harlequin516 | ? |
| 00:30.51 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.195) |
| 00:48.19 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 00:48.44 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 00:51.59 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:08.23 | *** join/#asterisk crimethinker (i=ircuser@legacy.diamond.org) |
| 01:11.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:18.24 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:20.42 | *** join/#asterisk waKKu (n=wakku@unaffiliated/wakku) |
| 01:34.46 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:37.46 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 01:47.08 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:51.23 | NightMonkey | tzafrir_laptop: Ah, success! :) With help from you and http://suvault.com/joachimblog/?p=13 . |
| 01:51.56 | NightMonkey | Well, almost, but got to get my extensions.conf contexts right... But I got a call through to the console. |
| 01:54.52 | NightMonkey | And, now with an "include => default" in the pstn context, we're set. :) |
| 01:55.08 | NightMonkey | Asterisk rocks! |
| 01:55.55 | SirThomas_Home | true statement! |
| 01:58.02 | NightMonkey | Now, I've got to figure out the basics of what I did to make it work, and document the relevant parts. |
| 02:17.11 | NightMonkey | OK, where do I find a reference to the subcommands to the dial application? (e.g. Dial(<extension>,<wait>,D(1234)) |
| 02:18.18 | waKKu | maybe voip-info.org/wiki ? |
| 02:18.37 | NightMonkey | waKKu: Thanks, I'll check the wiki. |
| 02:25.57 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 02:34.46 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 02:37.06 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 02:44.30 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 02:51.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:53.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:53.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:58.18 | Sweeper | ok, I have an idea for a voip service I need to bounce off some people |
| 02:58.46 | Sweeper | hows about a hosted pbx...that the client can easily configure via xml (RESTfully) |
| 02:59.11 | Sweeper | billed per-minute |
| 02:59.44 | Sweeper | and they will also be able to initiate calls via xml requests to the system, so they can instamagically integrate voip into their web applications |
| 03:00.52 | JT | sounds alright |
| 03:01.15 | Sweeper | yay~ |
| 03:04.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:37.12 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:41.32 | *** join/#asterisk Err (n=haHA@cpe-24-165-123-44.cinci.res.rr.com) |
| 03:44.57 | Err | I realize that this is not really within the scope of asterisk, but I'm at a loss as to where I should ask... Does anyone know why I can't use the * key within the 'dialplan' for a PAP2T? |
| 03:52.14 | Err | aha! and the answer is: because I'm an idiot. :-P |
| 03:52.36 | Err | Apparently asterisk doesn't display call attempts to invalid extensions on the console with -vvv? |
| 03:55.03 | *** join/#asterisk Kaycut (n=nada@host9.200-117-210.telecom.net.ar) |
| 03:55.12 | Kaycut | hi |
| 03:55.17 | Kaycut | i have a question |
| 03:55.32 | Kaycut | when i make an asterisk server |
| 03:55.47 | Kaycut | then two clients call one to the other |
| 03:56.03 | Kaycut | the bandwith of my conection is use? |
| 03:56.18 | Kaycut | or just their bandwith |
| 03:56.41 | Kaycut | im clear? |
| 03:56.43 | Qwell | Kaycut: depends on whether reinvite is enabled |
| 03:57.06 | Qwell | sometimes reinvite breaks with NAT, so...not always |
| 03:57.13 | Kaycut | can i setup the server to use their bandwith instead my own? |
| 03:57.32 | Qwell | if they aren't behind a NAT |
| 03:57.43 | Kaycut | supose they are behind a nat |
| 03:57.55 | Qwell | that makes it a bit more difficult |
| 03:57.56 | Kaycut | there isn't another way_ |
| 03:58.01 | Kaycut | ? |
| 03:58.10 | Kaycut | can you explain? |
| 03:58.55 | *** join/#asterisk Telamon (n=telamon@blk-222-160-100.eastlink.ca) |
| 03:59.12 | *** part/#asterisk Telamon (n=telamon@blk-222-160-100.eastlink.ca) |
| 03:59.30 | Kaycut | qwell are you there? |
| 03:59.50 | Qwell | I am, but I don't think I'd be able to explain it right, right now... I'm pretty tired |
| 04:00.22 | Kaycut | anyone here can do it? |
| 04:00.36 | Kaycut | i need a light explanation |
| 04:00.41 | Kaycut | about this |
| 04:01.11 | Kaycut | i mount an asterisk server to use ip clients, not land lines, just customers with ipphones |
| 04:02.00 | Kaycut | when a call is establish, can i setup my server to use their bandwith instead my bandwith because i have just 256kb |
| 04:05.37 | Kaycut | just answer me yes or no, its all i need please |
| 04:06.08 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 04:10.24 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 04:13.33 | JT | Kaycut: the answer is: pretty much no |
| 04:14.53 | Kaycut | ok |
| 04:15.01 | Kaycut | thanks i will try |
| 04:15.19 | Kaycut | ok? |
| 04:15.27 | JT | you won't be able to make them reliably talk direct to each other |
| 04:15.45 | JT | running a service on 256kbit/s of bandwidth... what crack are you on? :P |
| 04:23.04 | Kaycut | ok |
| 04:23.05 | Kaycut | thanks |
| 04:34.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:39.51 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 04:47.32 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 04:48.03 | *** join/#asterisk littleball (n=littleba@bb220-255-152-208.singnet.com.sg) |
| 04:49.07 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 04:49.26 | littleball | hello, i am writing a simple script to monitor the asterisk |
| 04:49.31 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 04:49.50 | littleball | it is strange to me that if i run the script from Bash console, it works |
| 04:50.07 | littleball | but if i run as .sh and in background, it stucks |
| 04:51.13 | littleball | #!/bin/bash |
| 04:51.13 | littleball | while(true) |
| 04:51.13 | littleball | do |
| 04:51.13 | littleball | <PROTECTED> |
| 04:51.13 | littleball | <PROTECTED> |
| 04:51.14 | littleball | <PROTECTED> |
| 04:51.15 | littleball | <PROTECTED> |
| 04:51.18 | littleball | <PROTECTED> |
| 04:51.20 | littleball | done |
| 04:55.39 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 04:59.07 | *** join/#asterisk darius_ (n=darius@integrity.bourg.net) |
| 04:59.23 | darius_ | Where can I find documentation that covers iax Registration? |
| 05:12.29 | Qwell | The above is an example of how NOT to write a script to check if a process is running. |
| 05:23.46 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 05:24.44 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:26.33 | *** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 05:31.59 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 05:39.31 | *** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com) |
| 05:47.13 | littleball | hello |
| 05:47.34 | littleball | i run command $/usr/sbin/asterisk -r -x 'show version' , it return immediately |
| 05:47.53 | Qwell | yeah, don't do that |
| 05:47.53 | littleball | but if i put this line in a bash script, and run it in background, it never return |
| 05:47.55 | littleball | why? |
| 05:47.58 | Qwell | the script is very bad |
| 05:48.08 | littleball | Qwell, how? |
| 05:48.16 | Qwell | there are *far* better ways to check if asterisk is running. Look at safe_asterisk |
| 05:48.23 | littleball | i use this script becuase i need to monitor the status of asterisk |
| 05:48.32 | littleball | whether it is stopped or running |
| 05:53.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:53.57 | littleball | Qwell, thanks |
| 05:56.05 | littleball | Qwell thanks |
| 06:05.38 | JT | has anyone here run Asterisk on Sun Netra T1s before? |
| 06:07.27 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:11.32 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:13.30 | *** part/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 06:35.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:56.43 | dlynes | JT: I've run callweaver on them before, using Solaris 9 |
| 06:56.54 | dlynes | JT: I would imagine Asterisk wouldn't be much different |
| 07:10.22 | *** join/#asterisk rvhi0 (n=rvhi3@cpe-67-49-180-124.hawaii.res.rr.com) |
| 07:12.11 | rvhi0 | how do i register multiple phones with the same extension? |
| 07:12.47 | rvhi0 | so if the ext is called, all will ring |
| 07:16.01 | dlynes | JT: erm solaris 8 sorry |
| 07:16.11 | dlynes | JT: Here's the url to the google cached copy: http://72.14.253.104/search?q=cache:J81jbQb9-y8J:wiki.openpbx.org/tiki-index.php%3Fpage%3DEasy%2BRoute%2Bto%2BBuilding%2BOpenPBX.org%2Bon%2BSolaris+openpbx+solaris+netra&hl=en&ct=clnk&cd=1&gl=ca |
| 07:16.27 | dlynes | JT: gotta run...sleepy time |
| 07:24.43 | JT | dlynes: thanks |
| 07:24.44 | JT | nice |
| 07:24.49 | JT | not tried any other OSes? |
| 07:25.53 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 07:32.03 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 07:33.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:35.28 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 07:40.30 | *** join/#asterisk gardo (n=gardo@125.212.121.250) |
| 08:07.09 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 08:24.49 | cy303 | sup |
| 08:26.22 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 08:38.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:48.51 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 09:01.57 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
| 09:02.19 | harlequin516 | Does the Background command return immediately? Has this behavior changed? |
| 09:13.41 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 09:16.58 | *** join/#asterisk bcnx (n=root@d54C0E110.access.telenet.be) |
| 09:17.43 | bcnx | Hi all, I'd like to throw the following to you if I may: is it absolutely necessary to register SIP phones with asterisk in order to use them? |
| 09:18.18 | bcnx | I'm building a asterisk cluster with heartbeat and if I switch nodes, the phones don't work untill their next registration |
| 09:18.46 | bcnx | but it seems they don't work at all without registering, even when I set host=xxx.xxx.xxx.xxx. in sip.conf |
| 09:19.26 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 09:19.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 09:21.58 | bcnx | mmm, lot's of people, no activity, odd |
| 09:26.57 | *** join/#asterisk CryptiK (n=cryptik@adsl-75-42-64-130.dsl.scrm01.sbcglobal.net) |
| 09:27.06 | CryptiK | phew |
| 09:34.01 | pj_ | Hello world |
| 09:37.33 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 09:43.06 | J4k3 | does anyone know offhand if one can use a 'locked' wm5 wifi-capable pda phone with a sip client? |
| 09:43.18 | J4k3 | seems like a cheap&easy way to get a wireless voip handset |
| 09:48.13 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:49.29 | k31th | guys I have configured an outbound route and setup the iax trunk... i said the dialing pattern tp 9|. when i press 9 then enter the number i get "all circuits are busy now" |
| 09:49.33 | k31th | and ideas? |
| 09:51.01 | DarKnesS_WolF | k31th: paste the full dialing line.. |
| 09:51.14 | DarKnesS_WolF | and make sure that iax show registery show that ur registered with ur IAX trunk |
| 09:52.01 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 09:53.30 | k31th | DarKnesS_WolF: how can i tell if the iax trunk is registered |
| 09:54.00 | k31th | status "unmonitored" ? |
| 09:55.37 | DarKnesS_WolF | k31th: iax2 show registry |
| 09:57.44 | k31th | DarKnesS_WolF: does not look like its registered |
| 09:58.58 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 10:01.49 | k31th | DarKnesS_WolF: could this not be registering due to my box being behind a NAT ? |
| 10:04.38 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:08.05 | k31th | where can i find the log so i can troubleshoot this IAX problem? |
| 10:08.53 | DarKnesS_WolF | k31th: dude this command i sent to u will tell u if ur registered on not ! |
| 10:08.57 | DarKnesS_WolF | and eys IAX works behind NAT |
| 10:09.00 | DarKnesS_WolF | yes * |
| 10:09.19 | DarKnesS_WolF | k31th: check ur /var/log/asterisk/* logs and check logger.conf to make it using full logeing |
| 10:14.24 | k31th | CAUSE : No such context/extension |
| 10:17.25 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:20.13 | bcnx | Hi all, I'd like to throw the following to you if I may: is it absolutely necessary to register SIP phones with asterisk in order to use them? |
| 10:20.30 | bcnx | I'm building a asterisk cluster with heartbeat and if I switch nodes, the phones don't work untill their next registration |
| 10:20.45 | bcnx | but it seems they don't work at all without registering, even when I set host=xxx.xxx.xxx.xxx. in sip.conf |
| 10:23.56 | k31th | DarKnesS_WolF: http://pastebin.ca/585579 |
| 10:31.29 | DarKnesS_WolF | k31th: man |
| 10:31.33 | DarKnesS_WolF | k31th: read the freaking command |
| 10:31.37 | DarKnesS_WolF | iax2 show registry |
| 10:31.41 | DarKnesS_WolF | not iax2 show peers ! |
| 10:31.44 | k31th | DarKnesS_WolF: yeah |
| 10:31.53 | k31th | I tried that does not appear to be registered. |
| 10:31.58 | DarKnesS_WolF | and ti fix this monitor thing u can add qualify=yes in the iax.conf |
| 10:32.11 | DarKnesS_WolF | k31th: what u have in iax.conf ? |
| 10:32.18 | DarKnesS_WolF | u have the context for the peer |
| 10:32.29 | DarKnesS_WolF | and register => username:password@iax_provider ? |
| 10:32.31 | DarKnesS_WolF | or u don'g ? |
| 10:33.09 | k31th | hum this seems top put every thing in sql... |
| 10:33.14 | k31th | this is a trixbox. |
| 10:33.30 | DarKnesS_WolF | k31th: go ask in trixbox |
| 10:33.36 | DarKnesS_WolF | #trixbox |
| 10:33.42 | k31th | ok, ta |
| 10:33.57 | k31th | when i did this by hand it took two mins |
| 10:43.43 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 10:43.53 | DarKnesS_WolF | k31th: i don't use trixbox sorry |
| 10:44.02 | DarKnesS_WolF | may be u miss something with the GUI |
| 10:44.22 | DarKnesS_WolF | i think u have to redite the regisrat node from the GUI after u create it |
| 10:46.37 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 10:50.57 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 10:56.08 | k31th | DarKnesS_WolF: no idea, this is my first crack with a gui in a while. |
| 10:56.28 | k31th | tbh addint this info in the config file would be far easier. |
| 11:06.31 | *** join/#asterisk evisu (i=hIRC@bzq-88-153-46-82.red.bezeqint.net) |
| 11:07.14 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 11:09.59 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 11:10.35 | *** join/#asterisk stoffell_w (n=stoffell@fw.catsanddogs.com) |
| 11:15.07 | *** join/#asterisk stoffell (n=stoffell@d51A580BB.access.telenet.be) |
| 11:15.14 | toot | hey folks. i'm just sitting down to write a spec for a client to an asterisk manager proxy. just wondering has anyone documented something similar or? |
| 11:17.56 | toot | i notice most of the pc based apps directly talk to the manager interface - i assume this is undersirable behaviour |
| 11:18.36 | pj_ | I'm using TDMOE with a fonebridge box, and keep getting weird "Got S-frame while link down" and the D channel comes up and down all the time (I also get a lot of Q931 "RESTART" message)... Can anyone help me understand why ? |
| 11:19.22 | stoffell | toot, depends on how many talking they do :-) the manager has improved over time, so it all depends on the load it tends to get... |
| 11:20.00 | toot | are the permissions not an issue? i thought an abstraction that provided more granual access control to be appropriate? |
| 11:20.19 | toot | ie a distinct config file/db to say what users can do what eg transfer calls, barge, etc |
| 11:20.42 | stoffell | well, that depends on 'your' environment .. if you need a lot of users and separate permissions, i'd say: go the proxy way |
| 11:21.50 | toot | are there any mature ones? that allow that level of abstraction? |
| 11:22.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:22.23 | stoffell | no idea, i only know of one, being astproxy ... (if i remember correctly) |
| 11:22.29 | toot | astmanproxy seems to be the most mature but seems to have died a death |
| 11:22.44 | stoffell | oh, that's the one i was talking 'bout :p |
| 11:23.00 | toot | okay, looks like a project that should be picked up on in that case |
| 11:23.23 | toot | you know off hand what the appropriate mailing list would be to see what peoples thoughts are? :) |
| 11:24.08 | stoffell | asterisk-users? or a totally different moment in this channel, many people are US-based :) |
| 11:24.22 | toot | hehe yeah :) |
| 11:35.10 | shido6 | hey pj |
| 11:35.20 | shido6 | how close are your TDMoE devices? |
| 11:39.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:45.02 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 11:50.29 | k31th | You guys got any idea? http://www.trixbox.org/forums/trixbox-forums/help/all-circuits-are-busy-now-2 |
| 12:03.00 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:04.04 | Err | k31th: are you sure that the extension (phone) number you're sending to voiptalk is legit? That is, are you stripping off the leading 9 that you use internally before sending it off? |
| 12:04.27 | Err | (and for that matter, do they want the leading 0) |
| 12:06.59 | k31th | Err: umm in the outbound routes i have 9|. |
| 12:07.29 | k31th | If i completely removed this would it just attempt to use the trunks for all calls? |
| 12:08.58 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 12:11.37 | *** join/#asterisk jeanmimi (n=jean-mic@obs92-5-88-163-187-48.fbx.proxad.net) |
| 12:11.40 | Err | I don't understand your question |
| 12:12.03 | k31th | how do i strip off the 9 ? |
| 12:12.05 | Err | my question is, when you send the extension to voiptalk (which is the phone number that you want them to dial for you), are you forwarding on that 9? |
| 12:12.30 | k31th | I hope now |
| 12:12.34 | k31th | not * |
| 12:12.36 | Err | ...because you probably shouldn't. I don't use voiptalk, so I don't know what extension format they use, but I doubt that they need/want a prefixed 9. |
| 12:12.40 | k31th | as that would mess it up ? |
| 12:12.48 | k31th | agreed |
| 12:13.01 | k31th | err who do u use? |
| 12:13.17 | Err | I'm currently playing with voicepulse - but I don't think that matters :-) |
| 12:13.19 | jwh | you shouldn't be sending to voiptalk other than the destination number |
| 12:13.27 | jwh | +anything |
| 12:13.42 | k31th | i can find out wat its sending via the log? |
| 12:13.57 | jwh | set verbose 10 on the console if you're using it |
| 12:14.04 | jwh | then watch when you dialout |
| 12:14.45 | jeanmimi | Hi |
| 12:14.52 | Err | I'm running at verbosity=3, and I see the extension number |
| 12:15.03 | jeanmimi | I am trying to use a regexp but I am not getting any result |
| 12:15.04 | k31th | DIAL_NUMBER=01225777888 |
| 12:15.10 | k31th | seems to be ok ? |
| 12:15.22 | k31th | I think its a problem with my IAX trunks not registering |
| 12:15.34 | jeanmimi | what I am trying to get is the part between sip: and @ (which in my case is a phone number, digits only) |
| 12:15.42 | jeanmimi | so here is my regex: |
| 12:15.45 | jeanmimi | Set(called=$["${SIP_HEADER(TO)}" : "\:([0-9]\+)"]) |
| 12:16.21 | jeanmimi | and I have tried much simpler regex, even just "." but it will never match |
| 12:16.29 | jwh | k31th: perhaps |
| 12:16.40 | jeanmimi | I have checked various examples on the net and I seem to be using exactly the same syntax |
| 12:16.51 | Err | k31th: does 'iax2 show registry' show them as registered? |
| 12:17.00 | k31th | jwh: well i did iax2 show registry and they are not registered :( |
| 12:17.14 | jwh | no errors in console as to why? |
| 12:17.18 | k31th | how do i find out why ? |
| 12:17.33 | k31th | can i force them to reregister ? |
| 12:17.40 | jwh | if you're attached to console and its erroring, it'll tell you why |
| 12:17.52 | caio1982 | jeanmimi: have you tried without the parenthesis? i didnt ever know it accepts regex, butyou could ttry that |
| 12:17.53 | jwh | err, not sure |
| 12:17.55 | k31th | iax2 reload? |
| 12:18.02 | jwh | yeah probably |
| 12:18.22 | jeanmimi | caio1982: no, I will try now |
| 12:18.51 | k31th | humm not real errors. |
| 12:19.29 | Err | k31th: are you sure that your outgoing extension doesn't have to start with 44? that's what it looks like, to me, in the example extensions.conf on support.voiptalk.org site |
| 12:20.13 | k31th | Err: how would i test this |
| 12:20.37 | jeanmimi | caio1982: now I am just getting 0 as value |
| 12:21.02 | jeanmimi | caio1982: I have tried with various regex and I always 0 if not using the () |
| 12:21.34 | caio1982 | jeanmimi: no clues then, i really didnt know it could accept regex (IF it really does) |
| 12:21.36 | Err | k31th: for starters, it looks like you don't *have* to register - you can just send username:password in your Dial() call (that is, Dial(IAX2/user:password@iax5.voiptalk.org/<Extension>)) |
| 12:21.54 | k31th | ok |
| 12:22.02 | caio1982 | but the parenthesis is useless for me on it |
| 12:22.08 | caio1982 | s/is/are |
| 12:22.10 | k31th | that might explain why its not reg atm. |
| 12:22.47 | jeanmimi | caio1982: well the () are normally not useless because they allow you to take just the part you want to keep |
| 12:23.23 | Err | k31th: I don't know anything about dialing in the UK, but I *think* that you need to drop the 0 as well and replace it with a 44; so, if (for instance) you were dialing, on your phone, the number '901225777888', you'd need your extensions.conf to use Dial(IAX2/USERID:PW@iax5.voiptalk.org/44${EXTEN:2}) to strip off the leading 9 and 0 and replace them with 44 |
| 12:23.32 | k31th | WOO! |
| 12:23.39 | k31th | Err: wicked got it working |
| 12:23.49 | k31th | I had to do, this |
| 12:23.57 | jeanmimi | caio1982: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 12:24.05 | caio1982 | it's not that true for sed, the app that taught me regexs, but i just checked voip-info and dialplan regex will always return 1 or 0 to you, just to let you know if the regex matched or not... kinda stupid, imho |
| 12:24.05 | k31th | 9441225445566 |
| 12:24.12 | k31th | 44 being my country code. |
| 12:24.24 | Err | right - the rule I just told you would replace the 0 with the 44 ;-) |
| 12:24.33 | k31th | wicked ! |
| 12:24.34 | Err | (so you could still dial 90<number> and have it work internally) |
| 12:24.40 | jeanmimi | caio1982: no, I am not using the REGEXP command |
| 12:25.13 | k31th | I guess its good practice to use a 9 ? |
| 12:25.13 | jeanmimi | caio1982: using the the : operator |
| 12:25.13 | k31th | for an outgoing |
| 12:25.26 | caio1982 | jeanmimi: oh, i got it, asterisk regex then |
| 12:25.40 | Err | if you have internal extensions as well, the 9 is 'standard' - at least in the US. However, you don't need them if you don't ever direct-dial your internal extensions from themselves - it's then just One More Digit that you have to dial. |
| 12:25.44 | jeanmimi | caio1982: whatever you call it :) |
| 12:26.14 | Err | for instance, I'm using asterisk to route my home phone over VoIP, so I don't use anything like a 9 - because it'd just mean that dialing is more complicated, which would very likely confuse visitors (and my wife). |
| 12:27.06 | caio1982 | jeanmimi: the anchoring of the regex at the beginning of the line isnt important for you? you're using : instead =~ |
| 12:27.06 | k31th | Err: haha yeah true, i will be using this at work eventually and 9 seems the standard in the UK to. |
| 12:27.17 | k31th | so ill keep that but fix the 44 thing and replace it with a 0 |
| 12:27.32 | jeanmimi | caio1982: I have tried with =~ too but it was helpless |
| 12:27.50 | Err | you could (should) write a slightly more complicated dialplan that allows you to use the 44, and have it be passed directly - or dial 0, and have it be replaced with 44 |
| 12:27.56 | caio1982 | LOL, that was great... from voip-info (not edition):"THE DOCUMENTATION ABOVE FOR REGULAR EXPRESSIONS IS WOEFULLY INADEQUATE." |
| 12:28.02 | caio1982 | hehe |
| 12:28.08 | caio1982 | s/not/no |
| 12:28.35 | jeanmimi | ok then, where can I get the proper information then ?? |
| 12:28.42 | Err | at least, you should if you can actually do that in the UK - I don't know how your guys' dialing plan works on the traditional phone lines... |
| 12:29.08 | caio1982 | jeanmimi: i dont know, i'm just reading the page, not complaining :) |
| 12:29.21 | jeanmimi | :) |
| 12:29.36 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:29.37 | caio1982 | jeanmimi: have tried without escaping the parenthesis and all ? |
| 12:29.59 | jeanmimi | caio1982: am I escaping the () ?? |
| 12:30.13 | caio1982 | ops, nevermind, i'm just thinking randomly |
| 12:30.37 | caio1982 | forget it, it's too early in brazil and i'm still sleeping |
| 12:30.47 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 12:30.59 | tzafrir_laptop | hmmm... voip-info could use something like the talk page of wikipedia |
| 12:31.01 | jeanmimi | :) |
| 12:39.01 | coppice | voip-info could do with some date stamping |
| 12:41.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:42.13 | Err | jeanmimi: is it possible that the regex has to include the *entire* body? That is, that you'd have to prefix it with the equivalent of [^:]*, and suffix it with something like .*? |
| 12:42.53 | Err | (I see on the voip-info.org site a note about 123foo matching their ([0-9]+), but foo123 not - which makes me wonder if there isn't some non-standard rules to these) |
| 12:43.45 | jeanmimi | caio1982: I have figured the problem out |
| 12:43.46 | Err | it actually looks like it has to match the entire *beginning* of the expression - but the examples are sparse, so I'm shooting in the dark |
| 12:44.34 | jeanmimi | caio1982: so I guess both : and =~ make the regex anchored to the starting character |
| 12:44.51 | jeanmimi | caio1982: anyways I now have a normal regex with () and it works as expected |
| 12:45.03 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 12:45.22 | Err | man, I'm on a roll - I think I'll quit while I'm ahead |
| 12:46.38 | *** join/#asterisk saftsack (n=saftsack@pD9E0754E.dip.t-dialin.net) |
| 12:47.19 | saftsack | hi, if i handle huntgroups this way here i get messages on phones which didnt answer the calls that they missed calls. is there a way to avoid this? exten => 1,1,Dial(SIP/001&SIP/002&SIP/003) |
| 12:48.17 | k31th | If i am using IAX as inbound route to a nat can i do this via just a port fwd? |
| 12:54.00 | caio1982 | jeanmimi: really? both anchoring to ^ isn't wrong? |
| 12:54.22 | jeanmimi | only : is supposed to |
| 12:54.39 | DrukenLPY | saftsack: yeah... change the settings on the phones to not report unmissed calls |
| 12:54.57 | DrukenLPY | er, missed calls |
| 12:55.18 | saftsack | DrukenLPY: ok i thought that there is maybe a message in the sip protocoll so that the phone can detect if it is an missed call |
| 12:56.41 | DrukenLPY | nope.. the phone did ring did it not? |
| 13:00.24 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:01.35 | saftsack | it ringed |
| 13:01.40 | *** join/#asterisk hideandseek (n=jace@ip70-190-245-6.ph.ph.cox.net) |
| 13:05.39 | DrukenLPY | exactly, so it "missed" the call |
| 13:09.52 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 13:11.27 | DrukenLPY | anyone from toronto in here?? did paramount sell wonderland? i'm looking over the site... and it's not advertised as paramounts anymore.... |
| 13:22.17 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 13:25.47 | k31th | humm my inbound routing goes to voicemail all the time? |
| 13:31.29 | *** join/#asterisk guillote_GNU (n=guillote@host212.201-252-204.telecom.net.ar) |
| 13:32.09 | DrukenLPY | k31th: pastebin the dialplan, and the cli output from an incoming call |
| 13:34.29 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 13:36.10 | k31th | http://pastebin.ca/585801 DrukenLPY ^^^ |
| 13:40.43 | pj_ | Heya, I keep getting Message type: RESTART (70) on my E1, and D channel goes up and down continuously... any idea why ? |
| 13:41.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:41.26 | pj_ | (I'm on TDMOE via a fonebridge box) |
| 13:42.38 | pj_ | and a bunch of !! Got S-frame while link down |
| 13:44.28 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:47.48 | pj_ | :/ |
| 13:50.14 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:56.03 | *** join/#asterisk friedrich| (n=friedric@e177253204.adsl.alicedsl.de) |
| 14:10.39 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 14:15.18 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 14:16.42 | *** join/#asterisk lukketto (n=lukketto@host213-103-dynamic.59-82-r.retail.telecomitalia.it) |
| 14:17.28 | *** part/#asterisk lukketto (n=lukketto@host213-103-dynamic.59-82-r.retail.telecomitalia.it) |
| 14:24.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:25.34 | festr__ | hello, anyone? |
| 14:25.54 | festr__ | i need a little help with 1.4.5, i've duplicate CDR when call is NO ANSWER |
| 14:33.58 | hackeron | festr__: hmm, just tried with my 1.4.5 -- no duplicate CDR for NO ANSWER here so seems like a configuration issue -- not sure what could be causing this though. |
| 14:34.17 | festr__ | hackeron: interesting. |
| 14:34.33 | festr__ | hackeron: hackeron look at http://forums.digium.com/viewtopic.php?t=15475&highlight=cdr |
| 14:34.44 | festr__ | hackeron: these people have the same issue |
| 14:34.57 | hackeron | festr__: but I override my NO\ ANSWER to retry with a different SIP account |
| 14:35.19 | festr__ | hackeron: try to call from sip to sip and hangup during ringing |
| 14:35.47 | hackeron | festr__: here's what I do: http://rafb.net/p/i47J8q56.html |
| 14:36.23 | hackeron | festr__: well, I have sip to sip-pstn-gateway |
| 14:36.25 | hackeron | festr__: let me try |
| 14:36.30 | festr__ | hackeron: great |
| 14:37.13 | hackeron | festr__: oh, I see, yes -- if I hang up during ring, I get 2 NO-ANSWER in my CSV |
| 14:37.40 | hackeron | festr__: http://rafb.net/p/mezEVk70.html |
| 14:37.40 | festr__ | hackeron: yes. and you have in the first cdr "s" extension is it? |
| 14:37.54 | hackeron | festr__: yes, first s and then the extension |
| 14:38.01 | festr__ | so it is the same |
| 14:38.09 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 14:38.12 | hackeron | yep, file a bug report :) |
| 14:38.28 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 14:38.45 | festr__ | there are some thread in dev list and post in forums.digium. it seems no one is interested in this |
| 14:39.17 | festr__ | someone could fire the bug :) |
| 14:39.55 | hackeron | festr__: well, I guess you can always sed -i the cvs if it's a major problem, but if you think about it, if the client hung up, then the sip phone you're dialing hangs up too -- so you can still see who hangs up first |
| 14:40.08 | falz | good day. is there any setting that can be set in any conf file to enable/disable either specific NOTICE or WARNING messages when in the CLI, or just to specify the level to show? |
| 14:40.22 | falz | just upgraded to 1.4.x, and now the level of NOTICES are way high, just due to modules that it's loading |
| 14:40.36 | festr__ | hackeron: it is complete useless these two cdr |
| 14:40.42 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 14:41.05 | hackeron | falz: /etc/asterisk/logger.conf |
| 14:41.17 | festr__ | hackeron: it is a little bit complicated to filter these cdrs |
| 14:41.34 | falz | hackeron: ah hah. thanks! I see it. I suppose the other solution is to not load the modules that throw errors |
| 14:42.03 | festr__ | and generally not load modules which you do not need |
| 14:42.55 | falz | seems that the debian package autoloads all. I think I'll just unload modules and leave the console messages to the default warn/notice/err |
| 14:43.00 | falz | seems cleaner (to me) |
| 14:43.06 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 14:43.06 | hackeron | festr__: I agree having 2 lines is useless but why is it causing you that much trouble? -- if you're rading the CVS line by line you still have the unique call number so it's still easy to parse |
| 14:43.28 | festr__ | hackeron: imageine thousands cdrs |
| 14:43.33 | festr__ | hackeron: and i'm logging to mysql |
| 14:43.56 | festr__ | hackeron: this "featur" is new to > 1.4.4 |
| 14:44.36 | *** join/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 14:44.47 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 14:44.54 | hackeron | festr__: my no answer lines seem to comprise roughly 12% of all lines |
| 14:44.57 | littleball | hello, how can i know the codec used for a specific channel on CLI console |
| 14:44.59 | littleball | ? |
| 14:45.26 | hackeron | festr__: but I guess I see your point |
| 14:45.26 | festr__ | littleball: show channel name |
| 14:45.44 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 14:46.15 | littleball | thanks festr__ |
| 14:46.52 | hackeron | festr__: http://bugs.digium.com |
| 14:47.09 | littleball | festr__, does " WriteFormat: 8 |
| 14:47.09 | littleball | <PROTECTED> |
| 14:47.19 | littleball | ? |
| 14:47.33 | festr__ | littleball: show translation |
| 14:48.22 | festr__ | littleball: i would not fire bug until someone on dev confirms it IS bug |
| 14:48.49 | festr__ | s/littleball/hackeron |
| 14:49.58 | littleball | festr__, what does show translation do? |
| 14:50.17 | littleball | the description from cli console is blur on this |
| 14:50.19 | hackeron | festr__: #asterisk-dev maybe? |
| 14:50.21 | festr__ | littleball: you asked ReadFormat: 8, show translation shows you, what codec is 8 |
| 14:50.54 | festr__ | hackeron: yes, i've tryed i have to be more patient :) |
| 14:50.55 | littleball | oh, i know 8. show codec 8 |
| 14:50.58 | falz | anyone happen to know what "process_zap: Ignoring signalling" would imply? my card(s) don't support the signalling type? everytihng appears to work ok, just got this error as of 1.4.x from 1.2.x |
| 14:51.06 | festr__ | littleball: aha then nevermind :) |
| 14:53.49 | littleball | what is the difference between safe_asterisk and /usr/sbin/asterisk? |
| 14:53.59 | littleball | i found the former uses much more CPU |
| 14:54.10 | littleball | although i know it is a simple bash script |
| 14:54.12 | littleball | wrapper |
| 14:55.48 | shido6 | respawns asterisk if it dies, littleball |
| 14:55.57 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:56.06 | shido6 | and plugin ur email addy in the script so it emails you when it dies. |
| 14:56.29 | shido6 | then have that email addy send u a text for easy notification without going nagios style |
| 14:56.44 | littleball | shido6, i know. i did modify safe_asterisk because i found that it doesnot work on fedora core 5 (my old system).. But after i change, the cpu usage become higher |
| 14:58.20 | shido6 | whats your new system? :) |
| 14:59.46 | littleball | shido6, i just modify the safe_asterisk file and make it simple |
| 14:59.51 | littleball | and work |
| 15:00.12 | littleball | for this system, only 50 channel use my one cpu |
| 15:00.43 | littleball | dual core cpu, 50 channel use half of the cpu resources |
| 15:01.46 | shido6 | cool |
| 15:02.09 | shido6 | u like that dual core, eh? |
| 15:02.12 | shido6 | :) |
| 15:02.48 | littleball | i just feel strange. another system has 20 channel usage, but cpu only use 5% |
| 15:03.04 | littleball | all use 711 codec. no codec translation |
| 15:03.08 | shido6 | well |
| 15:03.13 | shido6 | there are other factors |
| 15:04.06 | littleball | how to confirm on the system that there is no codec translation happend on the first system? |
| 15:04.39 | *** join/#asterisk waKKu (n=wakku@unaffiliated/wakku) |
| 15:04.47 | shido6 | start with say.... hdparm |
| 15:04.48 | shido6 | setpci |
| 15:05.16 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 15:05.20 | shido6 | cat /proc/interrupts find out what each card is using |
| 15:05.23 | shido6 | dont share interrupts |
| 15:06.09 | shido6 | do u have allow=all setup for anything? |
| 15:06.11 | waKKu | folks.. good morning first.. - someone there can say me _with sure_ if pickupgroups works with IAX or not ? |
| 15:06.50 | shido6 | etc etc :) |
| 15:07.12 | littleball | shido6, i do not think so (allow=all) is not |
| 15:07.47 | littleball | also, interrupt is not the reason. because the first system just start to use more cpu after i use safe_asterisk to start asterisk |
| 15:08.03 | littleball | safe_asterisk is not running asterisk as deamon |
| 15:08.09 | littleball | it is using asterisk -c |
| 15:08.23 | littleball | and piping the input/output to the /dev/null |
| 15:08.40 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:09.08 | [TK]D-Fender | littleball, However you run safe_asterisk as a daemon ITSELF <- |
| 15:10.00 | littleball | basically, it cannot . but in the safe_asterisk script, it try to do. Of course, i cannot make it work |
| 15:10.18 | littleball | so, i suspect that -c use more cpu |
| 15:11.01 | littleball | #!/bin/bash |
| 15:11.02 | littleball | ulimit -c unlimited |
| 15:11.02 | littleball | run_asterisk() |
| 15:11.02 | littleball | { |
| 15:11.02 | littleball | <PROTECTED> |
| 15:11.02 | littleball | <PROTECTED> |
| 15:11.04 | littleball | <PROTECTED> |
| 15:11.06 | littleball | <PROTECTED> |
| 15:11.08 | littleball | <PROTECTED> |
| 15:11.10 | littleball | <PROTECTED> |
| 15:11.12 | littleball | } |
| 15:11.16 | littleball | run_asterisk & |
| 15:11.24 | littleball | this is the modified version of safe_asterisk |
| 15:12.05 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:16.47 | Qwell | Why are you piping input from /dev/null? O.o |
| 15:17.09 | littleball | <PROTECTED> |
| 15:17.28 | littleball | tty9 is virtual console, right? |
| 15:17.37 | littleball | have no idea what is tty9 |
| 15:18.22 | littleball | Qwell, just use null to replace normal console name |
| 15:19.38 | littleball | and i thnk maybe it is the reason it use me one cpu core resource. although it works |
| 15:27.26 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:27.48 | *** join/#asterisk jeanmichch (n=jean-mic@obs92-5-88-163-187-48.fbx.proxad.net) |
| 15:27.57 | jeanmichch | Hi |
| 15:28.49 | jeanmichch | I am registered a voip ISP and whenever I receive incoming calls, they are always sent to extension "s" |
| 15:28.59 | jeanmichch | and I dont understand why they dont get sent to the actual called number |
| 15:31.04 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:31.20 | shido6 | :) |
| 15:31.33 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:33.33 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 15:33.36 | *** join/#asterisk bbryant_ (i=brett@nat/digium/x-9f27f1b79434b28e) |
| 15:33.45 | [TK]D-Fender | jeanmichch, because of your register statement. "regerist => user:pass@host/ifyoudidn'tfillinthisslashandnumberthenitsgoingtos |
| 15:34.25 | [TK]D-Fender | register* |
| 15:34.26 | [TK]D-Fender | shdd |
| 15:34.27 | [TK]D-Fender | bleh |
| 15:34.37 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 15:34.50 | jeanmichch | you mean just the trailing slash ? |
| 15:35.12 | jeanmichch | right now i have register => user:pass@host |
| 15:35.23 | *** join/#asterisk PioneerVM2 (n=IceChat7@24-151-103-018.dhcp.nwtn.ct.charter.com) |
| 15:35.32 | jeanmichch | so you are saying that I basically need to add a / at the end ? |
| 15:35.48 | PioneerVM2 | what is the proper way to handle incoming calls on different DIDs -- do you have to use an IF statement? |
| 15:35.56 | [TK]D-Fender | jeanmichch, /12345677 |
| 15:36.07 | [TK]D-Fender | jeanmichch, * fills in /s if you don't put something |
| 15:36.18 | PioneerVM2 | right now I use exten => _XX.,1,Answer -- that takes all calls from all DIDs but i need to have separate sections depending on the DID |
| 15:36.31 | jeanmichch | the thing is that I have three phone number that when dialed end up into my asterisk |
| 15:36.41 | [TK]D-Fender | PioneerVM2, make a FIXED pattern obviously, and stop useing a cath-all like that |
| 15:36.41 | jeanmichch | so I am not sure which number should go after the slashg then |
| 15:37.04 | [TK]D-Fender | jeanmichch, Depends how your provider works. |
| 15:37.06 | PioneerVM2 | for incoming is the pattern the # that the personcalled in on? |
| 15:37.13 | PioneerVM2 | or is it the extension withinthe service |
| 15:37.16 | PioneerVM2 | within asterisk |
| 15:37.20 | [TK]D-Fender | PioneerVM2, Yes. Hence the EXTEN |
| 15:37.22 | PioneerVM2 | i thought it was the extension within asterisk |
| 15:37.30 | PioneerVM2 | yes, so how do i change based upon the incoming DID |
| 15:37.36 | [TK]D-Fender | PioneerVM2, they dialed your DID. |
| 15:37.41 | PioneerVM2 | ugh. |
| 15:37.43 | PioneerVM2 | no i have multiple DIDs |
| 15:37.55 | PioneerVM2 | i want to do something different based upon the DID they dialed to reach me |
| 15:38.10 | PioneerVM2 | i have a context [incoming] |
| 15:38.19 | [TK]D-Fender | PioneerVM2, the provider may just lump all of them together. Go test CALLERID() for dnis,dnid, etc. |
| 15:38.29 | jeanmichch | [TK]D-Fender: what exactly do you mean by "depends how your provider works" what info should I check in the sip exchanges ? |
| 15:38.41 | PioneerVM2 | ok so that is what i asked, so i need to use an if statement of some sort to check tghe caller ID string |
| 15:38.54 | PioneerVM2 | and then goto different areas of code |
| 15:39.12 | [TK]D-Fender | PioneerVM2, maybe, maybe not. you shoulod just look at the values at try. |
| 15:39.20 | PioneerVM2 | when the call first comes in what is the default extension # that is started with |
| 15:39.23 | [TK]D-Fender | jeanmichch, same test for you. |
| 15:39.27 | PioneerVM2 | before someone types anything |
| 15:39.46 | [TK]D-Fender | PioneerVM2, DEPENDS. |
| 15:41.21 | jeanmichch | at try ? |
| 15:41.32 | jeanmichch | is that somethign I'll find the sip dialogs ? |
| 15:42.18 | [TK]D-Fender | jeanmichch, PioneerVM2 : NoOp(dnis = ${callerid(dnis)}) |
| 15:42.26 | [TK]D-Fender | jeanmichch, PioneerVM2 : NoOp(dnid = ${callerid(dnid)}) |
| 15:46.31 | jeanmichch | they both show 0 |
| 15:48.57 | jeanmichch | with callerid (lower case) beyy are both =0 |
| 15:48.58 | *** join/#asterisk gardo (n=gardo@203.84.184.246) |
| 15:49.06 | jeanmichch | with CALLERID they are equal nothing |
| 15:51.05 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:57.10 | riddlebox | does anyone have a grandstream GXP2000, I have one account on it connected to my asterisk server, which works fine, but I set the second account to a friends asterisk server, but it always says not registered, I can connect a soft phone to that one though? |
| 15:58.43 | _VoiceMeUp_COM | cluecon |
| 15:59.54 | *** part/#asterisk littleball (n=littleba@bb220-255-155-254.singnet.com.sg) |
| 16:00.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:10.34 | *** join/#asterisk ToyMan (n=Stuart@74-32-11-174.dsl1.mdl.ny.frontiernet.net) |
| 16:12.38 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 16:30.20 | *** join/#asterisk nohop (n=nohup@cc501678-a.hgv1.dr.home.nl) |
| 16:30.25 | nohop | good evening, ppls |
| 16:30.33 | waKKu | afternoon ;) |
| 16:30.38 | nohop | or that :) |
| 16:30.48 | waKKu | :D |
| 16:31.23 | nohop | does anybody know of some quick-start guide or smth to asterisk ? cause i kinda grew out of the reading-for-2-weeks-before-anything-works years ago... :) |
| 16:32.11 | nohop | and is asterisk actually the best choise if i only want to use SIP ? |
| 16:32.11 | waKKu | maybe asterisk handbook can help u |
| 16:32.37 | waKKu | only sip? i would say OpenSER |
| 16:32.59 | nohop | ahh.. |
| 16:33.08 | nohop | yeah, i've seen some stuff about that... i'll take a peek into that then :) |
| 16:33.25 | nohop | maybe later i'll want to add ptsn sometime though... |
| 16:33.42 | nohop | but from what i've seen so far configuring is like... 10 times as much work as apache was 10 years ago... |
| 16:33.55 | nohop | (asterisk, that is) |
| 16:35.05 | waKKu | ehehe... |
| 16:36.44 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 16:36.59 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:37.41 | tzafrir_laptop | I'm trying to build some modules from 1.4 out of th tree (app_skel is a trivial case) and get the error: |
| 16:37.58 | tzafrir_laptop | app_skel.c:133: error: ‘AST_MODULE’ undeclared here (not in a function) |
| 16:38.20 | Qwell | tzafrir_laptop: Corydon answered basically the same question last night in #asterisk-dev |
| 16:38.30 | Qwell | <Corydon76-home> d3wayne: this does it correctly, if you want to use it for a model: http://svncommunity.digium.com/view/func_odbc/1.4/Makefile |
| 16:44.19 | ManxPower | http://www.netfunny.com/rhf/jokes/old89/sheep.832.html |
| 16:44.37 | PioneerVM2 | is there an "if" statement other than GotoIf -- i cant seem to find it |
| 16:44.50 | PioneerVM2 | i want to set a variable based upon a situation |
| 16:44.51 | ManxPower | All my users are astronomers, all the staff are engineers |
| 16:45.08 | ManxPower | PioneerVM2: then gotoif to a place that sets a variable |
| 16:45.17 | PioneerVM2 | yea i dont like that method |
| 16:45.37 | ManxPower | PioneerVM2: oh well. |
| 16:45.47 | Qwell | ${IF()} |
| 16:46.02 | PioneerVM2 | how does it work qwell |
| 16:46.10 | PioneerVM2 | the goto if seems to require a static postion # |
| 16:46.11 | ManxPower | PioneerVM2: there are docs |
| 16:46.19 | PioneerVM2 | yea i cant seem to find an IF one im looking |
| 16:46.21 | ManxPower | PioneerVM2: NO! IT can use a label |
| 16:46.29 | PioneerVM2 | how would that work |
| 16:46.57 | [TK]D-Fender | PioneerVM2, "show function IF" |
| 16:47.06 | ManxPower | I'll ssh over a 3000 ms latency connection and find an example for you |
| 16:47.24 | PioneerVM2 | oh wait i found an if page |
| 16:47.32 | PioneerVM2 | i did a search but i think "if" was too general to find what i wanted |
| 16:47.36 | PioneerVM2 | it was hitting tons of non cmd pages |
| 16:48.53 | [TK]D-Fender | PioneerVM2, "show application execif" |
| 16:49.25 | ManxPower | PioneerVM2: http://pastebin.ca/586062 |
| 16:49.38 | tzafrir_laptop | After adding -DAST_MODULE=app_skel.c I get a slightly different error: app_skel.c:133: error: ‘app_skel’ undeclared here (not in a function) |
| 16:51.20 | ManxPower | PioneerVM2: Hewre is another one: http://pastebin.ca/586064 |
| 16:53.16 | PioneerVM2 | manx thx |
| 16:54.03 | ManxPower | I will frequently set the variable I want as the priority after the gotoif and then use the gotoif to skip the setvar if needed |
| 16:54.35 | ManxPower | so the default action is to set the variable, and if the gotoif condition is true skip that setvar |
| 16:56.31 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:57.22 | [TK]D-Fender | LONG : exten => o,n,GotoIf($[${LEN(${OPER_DEST})} != 0]?check-message) |
| 16:57.41 | [TK]D-Fender | SHORT : exten => o,n,GotoIf(${LEN(${OPER_DEST})}?:check-message) |
| 16:57.54 | [TK]D-Fender | SHORT : exten => o,n,GotoIf(${LEN(${OPER_DEST})}?check-message) |
| 16:57.56 | [TK]D-Fender | rather |
| 16:58.04 | [TK]D-Fender | <- Boolean abuser |
| 16:59.16 | nohop | hey... |
| 16:59.21 | ManxPower | There's nothing wrong with abusing booleans as long as you do it in private and wash your hands after |
| 16:59.23 | nohop | f: "0031528272772" <sip:0031528272772@217.67.240.200>;tag=as26a5cc07 |
| 16:59.23 | nohop | t: <sip:nohup@sip.nohup.nl>;tag=as22f6c723 |
| 16:59.41 | nohop | how do i tell asterisk to direct that call tp sip:nohup@192.168.10.30 ? |
| 16:59.46 | nohop | s/tp/to/ |
| 17:00.02 | nohop | wow, that bot is scary :) |
| 17:00.20 | ManxPower | Dial(SIP/nohup@192.168.10.30) |
| 17:00.45 | ManxPower | of course it would be better if you had a [nohup] section of sip.conf of course |
| 17:00.45 | nohop | yea... but it's the part before that i need... cause in the examples there's only 'numbers' there :) |
| 17:01.07 | nohop | ok, well that entry would be easily added, i guess... but.. |
| 17:01.08 | ManxPower | nohop: what "part before that"? |
| 17:01.44 | [TK]D-Fender | exten => fred,1,NoOp(Yay, some schmuck dialed FRED! lolz) |
| 17:02.01 | nohop | i mean exten => somethingsomethin,Dial(nohup) :) |
| 17:02.19 | nohop | the somethingsomething part :) |
| 17:02.41 | ManxPower | rather than looking at the sip debug, perhaps you can just tell us what you want to happen. |
| 17:02.46 | nohop | or SIP/nohup... i dunno... i guess i'll have to do a couple of days on reading up on stuff after all... |
| 17:02.57 | nohop | hehe, i could |
| 17:03.06 | ManxPower | "A call comes in via SIP for extension "nohup" and I want to send that call to device fred" |
| 17:03.06 | nohop | but i'm used to ppl only answering with 'rtfm' usually |
| 17:03.10 | nohop | so i tried to avoid that :) |
| 17:03.53 | ManxPower | or "A call comes into extension 0031528272772 and I want to send it to the device nohup", for example. |
| 17:03.58 | ManxPower | So what are you trying to do. |
| 17:04.10 | nohop | i want incomming calls to sip:nohup@sip.nohup.nl to be sent to sip:nohup@192.168.10.30 |
| 17:04.28 | nohop | where sip.nohup.nl is my asterix box |
| 17:04.32 | ManxPower | I assume your asterisk box is ip.nohup.nl |
| 17:04.35 | ManxPower | ok |
| 17:04.46 | nohop | and the other one is my workstation, running a softphone thingy |
| 17:04.55 | ManxPower | exten => nohup,1,Dial(SIP/nohup@92.168.10.30) |
| 17:05.00 | ManxPower | this is not rocket science |
| 17:05.07 | nohop | ahh nope |
| 17:05.09 | nohop | that was pretty easy :) |
| 17:05.38 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 17:06.24 | *** part/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 17:06.32 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 17:06.44 | ManxPower | oops |
| 17:07.11 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:12.32 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 17:13.30 | *** join/#asterisk ccesario_ (n=ccesario@201-0-52-126.dsl.telesp.net.br) |
| 17:13.32 | nohop | hmm... |
| 17:13.46 | nohop | that did give alot more output, but nothing ringing... |
| 17:15.19 | [TK]D-Fender | ~pb |
| 17:15.20 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:15.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 17:15.25 | nohop | oh wait, i'm lying.. i think it's exacly the same output |
| 17:15.57 | *** join/#asterisk ComputerWill (n=Computer@adsl-154-223-204.ard.bellsouth.net) |
| 17:16.01 | ccesario_ | hi |
| 17:17.51 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 17:21.36 | ccesario_ | I try pass the variable to other context, but I don't give success ... http://pastebin.ca/586100 |
| 17:22.31 | ccesario_ | the variable EXTEN_X is passed to context ura-usuario, but the value is "" :/ |
| 17:22.38 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 17:23.36 | [TK]D-Fender | ccesario: you must not have any spaces in you SET call. |
| 17:24.36 | ccesario_ | [TK]D-Fender exten => _82XX,6,Set(EXTEN_X = ${EXTEN}) |
| 17:24.39 | ccesario_ | ooops |
| 17:24.52 | ccesario_ | [TK]D-Fender exten => _82XX,6,Set(EXTEN_X=${EXTEN}) this ?? |
| 17:25.46 | [TK]D-Fender | yes |
| 17:25.54 | ccesario_ | hmmmm coool |
| 17:26.00 | ccesario_ | thanks [TK]D-Fender |
| 17:26.04 | [TK]D-Fender | np |
| 17:26.15 | nohop | hmmm |
| 17:26.23 | nohop | ManxPower: is that all that i should do to make it work ? |
| 17:26.26 | nohop | cause it's not working.. |
| 17:26.31 | [TK]D-Fender | ccesario : you need to remove them from here as well : exten => _82XX,3,GoToIf($[${DIALSTATUS} = CHANUNAVAIL]?8) |
| 17:26.36 | nohop | and calling to test stuff all the time is starting to be costy now :) |
| 17:27.14 | [TK]D-Fender | nohop, PASTEBIN your output and dialplan so we can see what's wrong |
| 17:27.15 | [TK]D-Fender | ~pb |
| 17:27.16 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:27.23 | ccesario_ | [TK]D-Fender why ? |
| 17:27.39 | [TK]D-Fender | ccesarioUsually Whitspace gets counted into the comparison. |
| 17:27.47 | *** join/#asterisk gardo (n=gardo@203.84.184.246) |
| 17:27.50 | nohop | [TK]D-Fender: there's 100s of lines of debug output, not really paste-able |
| 17:27.53 | [TK]D-Fender | ccesario : exten => _82XX,3,GoToIf($["${DIALSTATUS}"="CHANUNAVAIL"]?8) <- better |
| 17:27.58 | nohop | exten => nohup,1,Dial(SIP/nohup@192.168.10.30) |
| 17:28.04 | [TK]D-Fender | nohop, ****PASTEBIN**** |
| 17:28.05 | [TK]D-Fender | ~pb |
| 17:28.06 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:28.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:28.12 | nohop | i know what paste is |
| 17:28.14 | nohop | and what a bin is |
| 17:28.15 | ccesario_ | [TK]D-Fender ohhh thanks |
| 17:28.24 | nohop | but i never heard them in combination before :) |
| 17:28.32 | [TK]D-Fender | nohop, I've seen people pastebin 300 lines of code. JUST DO IT. |
| 17:28.38 | ccesario_ | [TK]D-Fender I change all in my dial plan |
| 17:28.38 | nohop | fuck this |
| 17:28.39 | *** part/#asterisk nohop (n=nohup@cc501678-a.hgv1.dr.home.nl) |
| 17:28.43 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 17:28.43 | [TK]D-Fender | 3000* |
| 17:28.53 | [TK]D-Fender | geez |
| 17:28.57 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 17:29.36 | [TK]D-Fender | Idiot at garage : "Whats wrong with my car?!?!?!?!?!?!" |
| 17:29.46 | [TK]D-Fender | Mechanic : "Lemme look under the hood" |
| 17:29.53 | ccesario_ | hahahhahahaha |
| 17:29.55 | [TK]D-Fender | Idiot at garage : "NOOOOOOOOO!!!!!!!!!" |
| 17:30.05 | [TK]D-Fender | Idiot at garage : (Drives away) |
| 17:30.10 | ComputerWill | Idiot at garage: Use your superpowers to figure it out! |
| 17:30.35 | [TK]D-Fender | Total &*#@^#$ing retard. God that kind pisses me right the hell off. |
| 17:33.42 | ComputerWill | Does anyone here have any experience getting newer Cisco 7971g phones working? I'm actually trying to get one to work with CallManager 4.1(2), but we're not running in encrypted mode, and the phone seems to want that. I figured the Asterisk community might have a few pointers. Is this the right channel for that? |
| 17:34.09 | ComputerWill | I found this, but It's still not registering: http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960 |
| 17:41.24 | [TK]D-Fender | ComputerWill, So far you're not working with a common protocol, nor is * even involved..... |
| 17:43.07 | ComputerWill | trus |
| 17:43.13 | ComputerWill | ture |
| 17:43.21 | ComputerWill | can't type. .. true |
| 17:44.20 | ComputerWill | I just thought that similar issues might have been encountered. My phone was requesting the .tlv file, so I tried some of the regular tricks. I didn't know if the SIP loads had the same problem. |
| 17:44.46 | [TK]D-Fender | ComputerWill, * doesn't even HAVE encrption.... |
| 17:45.23 | ComputerWill | I did find a note on Cisco's site which says to issue a null certificate, but our CTLClient isn't runnning (since we didn't buy the $300 USB thingy to create TLV files). Gotta love Ci$co. |
| 17:45.52 | PioneerVM2 | any way to not have the Voice Mail system record or inform you of msgs that are hangups -- it seems to always record 1 second long msgs if the person hangs up during the announcement |
| 17:46.04 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:47.39 | [TK]D-Fender | PioneerVM2, Sounds like disconnect supervision detection delay. |
| 17:47.46 | tzafrir_laptop | hmm... regarding AST_MODULE: I forgot to make AST_MODULE a string. Now it works |
| 17:48.06 | [TK]D-Fender | PioneerVM2, And when in doubt, run an external process to do your e-mailing instead and handle this yourself. |
| 17:48.22 | PioneerVM2 | id rather not reinvent the wheel |
| 17:48.30 | PioneerVM2 | i also dont want all these empty msgs |
| 17:49.47 | PioneerVM2 | is there a way to alter that detection value? |
| 17:50.49 | [TK]D-Fender | PioneerVM2, if its analog CDS, then you'r in TFB-Land. Otherwise, get coding.... |
| 17:52.37 | PioneerVM2 | ok so there is nothing easy to play around with |
| 17:53.26 | dlynes | JT: nope |
| 17:54.17 | dlynes | JT: Both of those setups are done by me (Solaris 8 on a Netra T1, and Solaris 10 on an UltraSPARC 5 |
| 17:54.26 | dlynes | JT: |
| 17:54.27 | Sweeper | asdf |
| 17:54.32 | Sweeper | facking centos |
| 17:54.41 | Sweeper | if I was doing this on gentoo, I'd be done already |
| 17:54.48 | dlynes | JT: Qwell has apparently gotten Asterisk up and running on Solaris as well |
| 17:54.57 | dlynes | JT: on a SPARC |
| 17:55.15 | dlynes | JT: But as I understand it, a Sunfire, not a Netra |
| 17:57.05 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:04.04 | tzafrir_laptop | Sweeper, kernel-devel |
| 18:04.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:04.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:06.49 | Sweeper | tzafrir_laptop: yea, didn't work |
| 18:07.09 | tzafrir_laptop | Sweeper, uname -r; rpm -qa | grep kernel |
| 18:07.45 | Sweeper | mmm, missing kernel-xen headers |
| 18:08.09 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 18:08.32 | Sweeper | but that doesn't exit! |
| 18:08.47 | tzafrir_laptop | kernel-xen-devel |
| 18:08.54 | Sweeper | I've got that already |
| 18:09.15 | tzafrir_laptop | uname -r ? |
| 18:09.37 | Sweeper | http://pastebin.ca/586189 |
| 18:11.24 | tzafrir_laptop | either downgrade kernel-xen-devel or boot to the newer kernel |
| 18:11.35 | Sweeper | mmm, good point~ |
| 18:11.42 | Sweeper | reboot~ |
| 18:12.12 | Sweeper | ok, so this realtime thing....I want my contexts to be entirely defined in the db |
| 18:12.28 | Sweeper | none of this switch statement stuff |
| 18:14.26 | Sweeper | is that even possible? |
| 18:15.58 | Sweeper | tzafrir_laptop: yay that worked |
| 18:16.09 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 18:16.24 | [TK]D-Fender | Sweeper, For the DB stuf : GET CODING. |
| 18:16.36 | Sweeper | [TK]D-Fender: gah |
| 18:16.41 | [TK]D-Fender | Sweeper, real-time = half-assed |
| 18:16.46 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 18:16.48 | Sweeper | damnit all D: |
| 18:16.59 | Sweeper | I fork asterisk for database justice \o\ |
| 18:17.12 | Sweeper | make it entirely db-based :D |
| 18:17.20 | [TK]D-Fender | Sweeper, it SHOULD BE. |
| 18:17.33 | carrar | with postgres |
| 18:17.37 | [TK]D-Fender | Sweeper, SQLite included along with admin toos for it. |
| 18:17.41 | tzafrir_laptop | It is. Only a database of text files... |
| 18:17.55 | Sweeper | screw postgres! mysql fo lyfe |
| 18:17.56 | [TK]D-Fender | carrar, No, generic SQL with embessed license friendly "starter" |
| 18:18.19 | Sweeper | hmm |
| 18:18.26 | [TK]D-Fender | carrar, "agnostic" should be key to this. As to which embedded "starter" to sue, who cares. |
| 18:18.33 | [TK]D-Fender | use* |
| 18:18.33 | carrar | heh |
| 18:18.44 | [TK]D-Fender | go for goal, not the means. |
| 18:18.52 | Sweeper | now to find out if it would be less of a PITA to learn a new, less sopported softpbx, or learn enough C to make asterisk do what I need it to |
| 18:19.14 | carrar | I'd really love * config files to all be in a db |
| 18:19.34 | [TK]D-Fender | Sweeper, you still haven't calidated that what you want to do REQUIRES the context system ata all, or a forced record of realtime. |
| 18:19.48 | [TK]D-Fender | Sweeper, so its REWIND TIME. WTF do you WANT TO DO? :) |
| 18:19.57 | tzafrir_laptop | carrar, the table structure barely works today |
| 18:20.02 | carrar | err a live db * |
| 18:20.28 | Sweeper | I want to define contexts on the fly, without restarting the server every time it happens |
| 18:20.36 | tzafrir_laptop | There are quite a few things that simply can't be easily done with a separate model. |
| 18:20.56 | [TK]D-Fender | Sweeper, No, defining contexts on the fly is again a means to an end... what was this to accomplish for you? |
| 18:21.03 | tzafrir_laptop | Basically it means that Asterisk has no idea what the real config is, and that it could change under its nose |
| 18:21.08 | *** join/#asterisk saftsack (n=saftsack@pD9E0754E.dip.t-dialin.net) |
| 18:21.24 | [TK]D-Fender | tzafrir_laptop, you say that... as though it were a BAD thing ;) |
| 18:21.32 | *** join/#asterisk centrex (n=mythtv@c-68-62-167-203.hsd1.al.comcast.net) |
| 18:21.34 | Sweeper | oh. I'm writing a system that will allow high-level dialplan maniupulation via XML POST/GET |
| 18:21.36 | *** join/#asterisk YonahW (n=kvirc@IGLD-83-130-71-223.inter.net.il) |
| 18:21.48 | tzafrir_laptop | [TK]D-Fender, not everything can be changed without Asterisk be aware of it |
| 18:22.06 | centrex | I'm trying to find another way to provide timing to asterisk for meemte(conferences), unfortunately on this vps I have no access to the source the kernel was built with and can't compile ztdummy |
| 18:22.20 | [TK]D-Fender | Sweeper, You could pipe all your calls through an AGI if you wanted at which point it could do whatever you wanted. But that is perhaps extreme. |
| 18:22.33 | tzafrir_laptop | Sweeper, yet another one |
| 18:22.34 | [TK]D-Fender | centrex, You're DOA then. |
| 18:22.53 | Sweeper | tzafrir_laptop: eh? |
| 18:22.54 | centrex | There's no feature to pull timing from like ntp or anything? |
| 18:22.59 | [TK]D-Fender | centrex, Try "app_conference" instead. WIKI it up. |
| 18:23.01 | tzafrir_laptop | It will either end up being too generic to be useful, or some custom system like the ones we have already |
| 18:23.04 | centrex | thanks |
| 18:23.28 | Sweeper | [TK]D-Fender: might work |
| 18:23.29 | tzafrir_laptop | centrex, no access to the kernel source? what system is that? |
| 18:23.51 | Sweeper | tzafrir_laptop: eh? show me somewhere else that allows configuration via xml |
| 18:23.55 | tzafrir_laptop | you normally don't need full kernel source for that |
| 18:23.55 | [TK]D-Fender | Sweeper, Of course it would work, its a question of how complex your processing is, but in AGI at least it puts it entirely in your hands. |
| 18:23.56 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:24.00 | centrex | tzafrir, a colocated virtual private server |
| 18:24.06 | carrar | heh |
| 18:24.20 | carrar | nice choice for a * box |
| 18:24.21 | [TK]D-Fender | centrex, Don't worry... we've seen your kind before ;) |
| 18:24.36 | centrex | =( |
| 18:25.21 | tzafrir_laptop | Sweeper, have you looked at the asterisk gui? They technically do that ;-) |
| 18:25.33 | Sweeper | tzafrir_laptop: totally not the point mang |
| 18:25.42 | tzafrir_laptop | Though their database is really the asterisk config files |
| 18:25.59 | Sweeper | I'm talking about a low cost of entry service that allows easy, fast voip services in web apps |
| 18:26.33 | Sweeper | I'll write a few interface libs for some popular frameworks, and am |
| 18:26.35 | Sweeper | *bam |
| 18:26.46 | tzafrir_laptop | VoiceOne, for instance, is built around real-time asterisk config, IIRC |
| 18:27.17 | Sweeper | you just do RestVoip.connect("1234567890","1234567890"), and you have google's click ot call on your web page |
| 18:27.56 | Sweeper | it's really more of a web dev tool than anything else |
| 18:28.00 | tzafrir_laptop | Sweeper, write a proxy app that communicate with sterisk via the manager interface |
| 18:28.07 | tzafrir_laptop | Customize the dialplan a bit |
| 18:28.07 | Sweeper | tzafrir_laptop: that's part of it |
| 18:28.24 | Sweeper | but I also want to do things like IVR |
| 18:29.08 | tzafrir_laptop | IVR is just extensions.conf, and this can be used from realtime today |
| 18:29.22 | Sweeper | but I can't create extensions |
| 18:29.23 | tzafrir_laptop | (you miss some features, but this is an inherent problem of realtime) |
| 18:29.24 | Sweeper | err |
| 18:29.25 | Sweeper | contexts |
| 18:32.11 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:33.19 | *** join/#asterisk CodeBanshee (n=cfairey@194.164.236.240) |
| 18:34.18 | tzafrir_laptop | hmm... is this a limitation of realtme extensions? I have not heard of it before |
| 18:34.37 | tzafrir_laptop | not that I use realtime in any way |
| 18:35.15 | Sweeper | well |
| 18:35.22 | Sweeper | you can slap wahtever into the db |
| 18:35.38 | Sweeper | but you have to have logic in the ext.conf file to deal with new extensions |
| 18:37.36 | *** join/#asterisk guillote_GNU (n=guillote@host193.200-82-106.telecom.net.ar) |
| 18:40.25 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:44.47 | tzafrir_laptop | Sweeper, this is a Good Thing: you *can* have both extensions.conf and realtime (or even extensions.ael or whatever) |
| 18:45.18 | tzafrir_laptop | It would make sense to put macros and other "code" part in a static file. They will actually be readable |
| 18:45.35 | tzafrir_laptop | You can keep them empty and ue only the DB if that makes you happy |
| 18:45.43 | Sweeper | you're not picking up what I'm laying down, mang |
| 18:46.20 | Sweeper | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions <-- according to that, I HAVE to have stuff in extensions.conf |
| 18:47.09 | Sweeper | basically, I have to define a context and switch for each new context in the db |
| 18:47.15 | Sweeper | now, if this is wrong, yay |
| 18:49.01 | *** join/#asterisk lukketto (n=lukketto@host213-103-dynamic.59-82-r.retail.telecomitalia.it) |
| 18:49.23 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 18:58.44 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:59.28 | [TK]D-Fender | Sweeper, easy to update your extensions.conf and issue an "extensions reload" you know... |
| 19:00.13 | [TK]D-Fender | Sweeper, Can't see why generating on change is an overly difficult uor undesirable thing (hard to validate most other approaches) |
| 19:02.04 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 19:03.08 | carrar | any compeling reasons no to upgrade from 1.4.4 to 1.4.5 |
| 19:03.12 | carrar | no=not |
| 19:04.56 | Sweeper | [TK]D-Fender: uh, the entire point of using realtime is to AVOID reloads and writing to flatfiles |
| 19:05.49 | *** join/#asterisk joebob777as7 (n=C@yoda.peacefulescape.com) |
| 19:06.04 | Sweeper | besides the fact that it would be horrible once I got into clustering |
| 19:06.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:06.50 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:07.04 | joebob777as7 | can someone help me? i am trying to connect a wireless sip phone from my office to my house and i'm watching my cli it says registered sip port 52238 and then my phone tells me that it failed |
| 19:09.50 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 19:10.13 | joebob777as7 | it's an e60 nokia |
| 19:10.39 | *** join/#asterisk lovely2 (n=tylerj@fluoride.crm114.net) |
| 19:10.45 | lovely2 | hello |
| 19:11.05 | blitzrage | joebob777as7: don't think you can use a password because the phone won't respond to the 407 Proxy Auth |
| 19:11.20 | lovely2 | is there a way of creating a channel that isn't connected to a device and just rings out |
| 19:11.48 | blitzrage | lovely2: huh? |
| 19:11.59 | NOT_guru | Hello I am on asterisk 1.2.18 and I am looking for my rev number asterisk -r is not showing my rev on the connect output |
| 19:11.59 | joebob777as7 | blitzrage, so what should i change? remove my secret in the extension? |
| 19:12.07 | NOT_guru | is there another way to see your rev # |
| 19:12.28 | NOT_guru | FYI I am trying to go through the chan_unistim install faq |
| 19:12.29 | blitzrage | joebob777as7: ya, you can't authenticate using a secret, so you have to remove it and just filter on the IP address (permit/deny) |
| 19:12.43 | blitzrage | NOT_guru: cd /usr/src/asterisk ; svn info |
| 19:12.44 | lovely2 | blitzrage: what i said, eg like a sip or iax or sccp channel that just rings out, without being conntected to a channel |
| 19:13.01 | blitzrage | lovely2: everything IS a channel... |
| 19:13.06 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-00a8f49b7d914f17) |
| 19:13.10 | blitzrage | lovely2: do you mean something like a "callfile" ? |
| 19:13.11 | lovely2 | does everything provide hint data |
| 19:13.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:13.35 | *** join/#asterisk Dovid (n=Dovid@bzq-88-152-100-129.red.bezeqint.net) |
| 19:13.39 | lovely2 | i'm trying to add a new extension to a queue, so that i can light a light on a cisco phone to say people are in the queue |
| 19:13.41 | Dovid | evening ev1 |
| 19:13.44 | NOT_guru | Thank you blitzrage |
| 19:14.14 | blitzrage | lovely2: sounds almost liek you want to check out SLA |
| 19:14.35 | [TK]D-Fender | blitzrage, NO. |
| 19:14.38 | lovely2 | the sla stuff is easy, but i have no way of using it with queue |
| 19:14.53 | [TK]D-Fender | blitzrage, HE DON'T WANT NO SLA. |
| 19:14.53 | blitzrage | [TK]D-Fender: thanks for your useful input |
| 19:14.56 | lovely2 | queue provide no hint data, at least anyway of working |
| 19:14.59 | blitzrage | HE JUST WANT |
| 19:15.01 | [TK]D-Fender | blitzrage, ! ! ! |
| 19:15.03 | blitzrage | lol |
| 19:15.19 | joebob777as7 | blitzrage, i did that and it gave me the same error... it's an e60 nokia |
| 19:15.30 | [TK]D-Fender | lovely2, the only way of monitoring Queue usage is with their XML browser. |
| 19:15.51 | blitzrage | joebob777as7: no idea... you have to read the SIP trace to determine what it is doing/not doing |
| 19:16.01 | lovely2 | Fender: you have a link ? |
| 19:16.24 | lovely2 | i search for a xml service provider script, for that purpose but i couldn't find one. |
| 19:16.29 | [TK]D-Fender | joebob777as7, pastebin the failed call attempt's CPI output at verbsoe 210, SIP debug enabled, and provide the [general] section of your sip.con and the phone's entry |
| 19:16.52 | [TK]D-Fender | lovely2, Do you know at all how to enable the XML broswser on the phone? |
| 19:16.59 | lovely2 | yes, have done |
| 19:17.07 | lovely2 | i already have a bunch of services |
| 19:17.22 | lovely2 | i don't feel like writing a script to do it. |
| 19:17.32 | lovely2 | i thought somebody else would of. |
| 19:17.49 | [TK]D-Fender | lovely2, then make a web script that creates an XML page parseing out the data you want. Use AMI in your script to connect to * to gt the info or parse the output of "asterisk -rx "show queues"" and so on |
| 19:18.14 | [TK]D-Fender | lovely2, I wrote one for Polycom's XHTML MicroBrowser. |
| 19:18.28 | lovely2 | yeah, i might end up doing that. |
| 19:18.31 | [TK]D-Fender | lovely2, But don't automatically assume ther is a finished product ready to be shoved into your hands. |
| 19:18.40 | lovely2 | i like assumptions |
| 19:18.45 | lovely2 | makes my life easier |
| 19:18.57 | [TK]D-Fender | lovely2, They make an ass out of you and of "umption" ;) |
| 19:19.01 | lovely2 | like queues supporting hints |
| 19:19.19 | lovely2 | what about assumptions being the mother of all f---- ups |
| 19:19.36 | [TK]D-Fender | lovely2, Assume * will wire-transfer $1,000,00 on demand as well... if you've going to have delusions, don't be half-assed about it! |
| 19:19.48 | Sweeper | hmmmm |
| 19:19.55 | lovely2 | hahaha |
| 19:20.00 | Sweeper | looks like AGI is the way to go :/ |
| 19:20.01 | [TK]D-Fender | lovely2, INDEED |
| 19:20.06 | NOT_guru | Request wire-transfer $1,000,00 |
| 19:20.16 | Sweeper | gah, I hope ruby can run fast enough to make it work |
| 19:20.20 | NOT_guru | oops dropped a 0 |
| 19:20.26 | NOT_guru | but I would take the grand |
| 19:20.28 | NOT_guru | sorry |
| 19:20.29 | [TK]D-Fender | Sweeper, I'd sooner bet that SER is the way to dgo and just use * for termination and as an app server. |
| 19:21.03 | Sweeper | [TK]D-Fender: the apps are what I need to script |
| 19:21.31 | [TK]D-Fender | Sweeper, what kind of taks is your system doing? |
| 19:21.59 | Sweeper | ivr, voicemail, conf rooms |
| 19:22.58 | [TK]D-Fender | Sweeper, ok, well I'm sure you've gained at least a little insight here and will find something more acceptable for your means.... |
| 19:23.09 | Sweeper | snicker |
| 19:23.34 | Sweeper | it's just that I've decdied that AGI is the lesser of two evils, providing I can work out a functional paradigm |
| 19:24.12 | [TK]D-Fender | Sweeper, Plenty more evil to look into ;) |
| 19:24.44 | Sweeper | eh. yate seems like a PITA, and a poorley documented one at that |
| 19:25.03 | [TK]D-Fender | Sweeper, gone indepth with FreeSWITCH yet? |
| 19:25.42 | [TK]D-Fender | Sweeper, offers a very promising approach. I'm awaiting a "ready for public consumption" release personally |
| 19:25.50 | Sweeper | exactly |
| 19:25.55 | Sweeper | it looks ok |
| 19:26.08 | Sweeper | but it's not 'production' yet |
| 19:26.49 | harlequin516 | This seems rather undocumented to me, but I call Background in my macro, and the dialed extension goes to the context that the Macro was called from. |
| 19:33.20 | [TK]D-Fender | harlequin516, No, that is VERY documented. |
| 19:33.52 | [TK]D-Fender | harlequin516, Macro's become effectively merged with the context calling them and for the love of God STOP TRYING TO MAKE IVR'S IN MACROS PEOPLE! |
| 19:36.10 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:36.36 | *** part/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:37.02 | *** join/#asterisk colinm_ (n=col@VDSL-130-13-116-18.PHNX.QWEST.NET) |
| 19:37.41 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 19:38.27 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 19:41.22 | lovely2 | D-Fedner how did you get around the permission problems. |
| 19:41.31 | lovely2 | with your xml script |
| 19:41.39 | [TK]D-Fender | What permission problems? |
| 19:41.59 | lovely2 | i'm just using exec to call asterisk -rx \"show queues" |
| 19:42.00 | [TK]D-Fender | lovely2, I use AMI personally, and so should you. |
| 19:42.01 | lovely2 | in php |
| 19:42.04 | lovely2 | ami |
| 19:42.06 | lovely2 | what is ami ? |
| 19:42.09 | [TK]D-Fender | lovely2, And you COULD jsut runt both as root. |
| 19:42.14 | [TK]D-Fender | ~ami |
| 19:42.15 | jbot | ami is, like, the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 19:42.17 | lovely2 | ick |
| 19:42.25 | lovely2 | cool |
| 19:48.44 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:49.45 | harlequin516 | Alright I have solved my longstanding problem. apparently I have to answer the call before I can Dial, or the media path wont connect |
| 19:50.54 | Strom_M | duh |
| 19:50.55 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 19:51.24 | [TK]D-Fender | harlequin516, It'll connect... it the dial gets ANSWERED. |
| 19:52.13 | harlequin516 | I guess there is some realtionship between Dial and Answer, that I was unaware of. |
| 19:52.43 | [TK]D-Fender | harlequin516, Still barking upt he wrong tree and failing to actually describe what you are trying to accomplish.... |
| 19:53.20 | harlequin516 | [TK]D-Fender: I'm just confusing everyone because I have 12 open issues. |
| 19:53.41 | [TK]D-Fender | harlequin516, www.drphil.com |
| 19:53.44 | harlequin516 | Haha |
| 19:53.55 | harlequin516 | I'd rather commit sepuku |
| 19:54.31 | harlequin516 | So when does a call need to be answeed in a dialplan? I mean should? |
| 19:55.32 | [TK]D-Fender | harlequin516, wrong approalch. tell us exactly what you're trying to do and we can comment on HOW. |
| 19:55.32 | harlequin516 | tsuba? Is that to kill me from behind if I fail in attempt? |
| 19:56.07 | harlequin516 | I want to call out to broadvoice through asterisk from my sipura fxs. |
| 19:56.12 | [TK]D-Fender | harlequin516, Tsuba is the handguard on a sword. Before you draw you push on the tsuba to release the blade. |
| 19:56.24 | harlequin516 | oh |
| 19:56.39 | [TK]D-Fender | harlequin516, In priming it I show that I am prepared to draw. |
| 19:57.15 | harlequin516 | The calls to braodvoice werent working because at no point in my dialplan (path) was there an Answer to the call from my fxs. |
| 19:57.24 | [TK]D-Fender | <- working of 5th kyu Tenshen Shoden Katori Shinto Ryu |
| 19:58.11 | harlequin516 | So bascially I think the lesson for me is: You must Answer a Call before you Dial? |
| 19:58.21 | [TK]D-Fender | harlequin516, That does not add up.... if your dialplan immediately dials your ATA and it answers, the call gets bridged, thats it |
| 19:58.56 | harlequin516 | Yeah.. hmm You mean bridged instead of forwarded? |
| 19:58.58 | [TK]D-Fender | harlequin516, Answer is only mecessary if the time you're spending ringing your ATA exceeds the timeout for the inbound ringing. |
| 19:59.24 | [TK]D-Fender | "fordwarded" in an incorrect term. * is a B2BUA, not a proxy. |
| 19:59.27 | Strom_M | or if you need to perform the IVR menu dance |
| 19:59.41 | harlequin516 | I guess the right word maybe , bridged instead of native bridged? |
| 19:59.55 | [TK]D-Fender | harlequin516, that part doesn't factor it. |
| 20:00.19 | [TK]D-Fender | harlequin516, if a call is allowed to reinvite, it WILL. Doesnt matter if * does IVR stuff first or not. |
| 20:02.10 | *** join/#asterisk kstward9 (n=ksteward@71.174.78.77) |
| 20:02.51 | harlequin516 | [TK]D-Fender: I understand you r logic, and I agree. However my test shows otherwise. The only way that i get the voice path to connect is by Answer() before Dial(Broadvoice). |
| 20:03.17 | harlequin516 | If I do not Answer it appears to connect, but there is no voice in either direction. |
| 20:04.02 | [TK]D-Fender | harlequin516, then your SIP setup is bad. add "canreinvite=no" to [general] and all your other entries |
| 20:04.06 | harlequin516 | There must be some other implication (than timeout) to Answering a Call. |
| 20:04.32 | harlequin516 | [TK]D-Fender: But it works with Answer() ? |
| 20:04.35 | [TK]D-Fender | YES |
| 20:04.51 | harlequin516 | [TK]D-Fender: I understand the canreinvite=no, and I have done this already |
| 20:05.23 | [TK]D-Fender | Exten => _NXXNXXXXXX,1,Dial(SIP/broadvoicepeer/${EXTEN}) ; z0mg it werks!? |
| 20:06.35 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 20:09.42 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 20:11.01 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:13.10 | harlequin516 | Is it possible that a previous Answer() cmd alters the method of the bridging from a subsequent Dial() cmd? |
| 20:13.40 | Qwell | harlequin516: it's possible, but probably not, no |
| 20:15.29 | harlequin516 | OKay back to my previous question.. Is it the right behavious of calling Background() from a macro to jump to the read extension in the context from which the macro was called? |
| 20:16.37 | harlequin516 | This question may be trickier than is obvious. |
| 20:17.27 | harlequin516 | I thought that a Macro was just an easy way to gosub with parameters. I think there is more magic than that. |
| 20:17.30 | waKKu | folks.. whats softphone r u using for IAX with linux ? |
| 20:18.01 | harlequin516 | SIP softphones all suck. |
| 20:18.07 | harlequin516 | I hate SIP |
| 20:18.14 | harlequin516 | So damn complex for nothing. |
| 20:18.18 | waKKu | harlequin516: k means kde ? |
| 20:18.19 | [TK]D-Fender | harlequin516, As I said, Macros merge intot he context calling them and this is well documented |
| 20:18.39 | harlequin516 | Yeah, but its not a standard feature of KDE. |
| 20:19.04 | harlequin516 | [TK]D-Fender: Oh I must not have seen your reply. |
| 20:19.31 | harlequin516 | waKKu: I do recommend kiax. |
| 20:20.29 | waKKu | hm.. i hate kde.. but i'll try it ;) |
| 20:20.38 | waKKu | do u know anoter ? |
| 20:21.15 | harlequin516 | [TK]D-Fender: I am looking at the voip-info cmd Macro page, but I do not see where it says that. |
| 20:22.24 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 20:22.42 | harlequin516 | [TK]D-Fender: Oh I see it in Note 2. |
| 20:23.11 | harlequin516 | [TK]D-Fender: Its just not stated formally using the right words like you used. |
| 20:26.11 | YonahW | what is the best way to deactivate two modules from a tdm04B? just dont configure in zaptel.conf? |
| 20:26.43 | YonahW | dont use in zapata.conf? separate as another group? |
| 20:26.48 | harlequin516 | Gheesh: Macro cmd has a lot of quirks. |
| 20:26.56 | [TK]D-Fender | harlequin516, Its quite clear and you must not have been in an "open" mood when reading. |
| 20:27.31 | [TK]D-Fender | YonahW, Clarify what you would define as "inactive", and we'll suggest the means. |
| 20:28.32 | *** join/#asterisk zabin (n=zabin@c-68-59-30-108.hsd1.sc.comcast.net) |
| 20:28.46 | YonahW | no intention of using them nor plugging anything into them for the time being |
| 20:32.01 | rob0 | /dev/kid is asking me: what kind of monkeys are tt-monkeys ? |
| 20:34.39 | sevard | rob0: wtf |
| 20:35.47 | tzafrir_laptop | YonahW, basically both are equivalent. I would disable in zapata.conf, so /proc/zaptel sould still give a correct picture |
| 20:36.22 | [TK]D-Fender | YonahW, remove them from any achannel groupings and point them to a null context. Or jsut remove their channel ddefinition lines. |
| 20:36.46 | lovely2 | love php inline, no fuctions just one long main |
| 20:36.47 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 20:37.01 | [TK]D-Fender | rob0, CodeMokeys : http://www.jonathancoulton.com/2006/04/14/thing-a-week-29-code-monkey/ |
| 20:37.07 | rob0 | ty |
| 20:39.15 | YonahW | tzafrir, D-Fender : thanks for the advice |
| 20:40.16 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:40.54 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 20:44.50 | rob0 | haha cute :) |
| 20:47.09 | *** join/#asterisk saftsack (n=saftsack@pD9E0754E.dip.t-dialin.net) |
| 20:48.16 | lovely2 | Fender: finished :) |
| 20:55.11 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 20:55.54 | YonahW | i have a te110p and a tdm04b i run ztcfg and get an error that ZT_CHANCONFIG failond channel 29. any ideas? I pasted my configuration and ztcfg verbose at http://pastebin.ca/586437 |
| 20:56.06 | YonahW | failond=failed |
| 20:57.07 | YonahW | modules load successfully |
| 20:57.16 | [TK]D-Fender | fxsks=32-36 <- 1 channel too many. Next you need to make sure of which order your cards got initialized in. check dmesg |
| 20:57.43 | [TK]D-Fender | and you failed to specify your dchan |
| 20:58.14 | YonahW | do i need to specify the dchan? i tried it orginally specifying the dchan in between 15 and 17 |
| 20:58.42 | YonahW | clearly you are right about 36 being too many channels but why would it fail on 29? |
| 20:58.59 | [TK]D-Fender | YonahW, not sure. Are yuo getting a APRTIAL PRI? |
| 20:59.03 | [TK]D-Fender | PARTIAL* |
| 20:59.14 | [TK]D-Fender | YonahW, It'd crap out of the telco disagrees IIRC |
| 20:59.26 | YonahW | no full pri |
| 21:00.29 | [TK]D-Fender | YonahW, Ok, check the first pile of things then |
| 21:01.53 | Aces1Up | anyone here use adhearsion? |
| 21:02.41 | waKKu | [TK]D-Fender: did u had used pickupgroups ? i'm trying to use on 1.2 and get no success... |
| 21:02.48 | YonahW | D-Fender, thanks |
| 21:03.10 | [TK]D-Fender | waKKu, nope |
| 21:03.25 | [TK]D-Fender | waKKu, Although I'm quite sure they work jsut fine. |
| 21:03.49 | waKKu | [TK]D-Fender: hm.. i may do sth wrong.. |
| 21:04.03 | waKKu | or the problem is with IAX |
| 21:04.03 | [TK]D-Fender | waKKu, more than likely |
| 21:04.10 | waKKu | ;) |
| 21:04.15 | [TK]D-Fender | and no, not IAX. |
| 21:09.19 | YonahW | D-Fender: I made the necessary changes and verified that the te110p loads first, any other ideas? |
| 21:09.47 | aptura | <PROTECTED> |
| 21:12.35 | [TK]D-Fender | YonahW, Ok, this is a brand new TE110P isn't it? |
| 21:12.51 | YonahW | D-Fender, yes |
| 21:12.58 | [TK]D-Fender | YonahW, Ok, that solves it |
| 21:13.05 | YonahW | ?? |
| 21:13.25 | [TK]D-Fender | YonahW, by default its set for T1 mode ONLY (24 channels. that + your TDM04B = 28 channels and says why it dies on 29. |
| 21:13.42 | [TK]D-Fender | YonahW, Shut down your box... theres a JUMPER oyou forgot to set for E1 on it. |
| 21:13.45 | YonahW | I see but the jumper is on |
| 21:13.56 | YonahW | i checked before i put it in the box |
| 21:13.59 | [TK]D-Fender | YonahW, I'd TRIPLE check that if I were you |
| 21:14.03 | YonahW | i can shut down and check again |
| 21:14.16 | [TK]D-Fender | It screams "jumper error" all over it |
| 21:14.32 | YonahW | gonna do that right now, just because what you are saying definitely makes sense |
| 21:22.42 | YonahW | D-Fender: This is just going to shock you |
| 21:23.11 | YonahW | turns out the jumper was indeed not on properly and after fixing that ztcfg runs smoothly |
| 21:23.35 | [TK]D-Fender | YonahW, Quite welcome |
| 21:23.40 | YonahW | i could have sworn that I checked that jumper |
| 21:23.47 | YonahW | truly appreciate the assistance |
| 21:24.00 | [TK]D-Fender | YonahW, Maybe you did and it got torqued during installation... |
| 21:24.11 | [TK]D-Fender | "Shit happens" |
| 21:24.17 | YonahW | could be I installed that card first |
| 21:24.23 | YonahW | and shit definitely does happen |
| 21:24.39 | [TK]D-Fender | very minor, and not a long time to diagnose & correct. |
| 21:25.20 | YonahW | yeah if this is the entire extent of trouble I have on this setup I will be rather pleased |
| 21:25.29 | [TK]D-Fender | 30 mins + IRC delay : physical inspection time (+shutdown/restart & test) = no big deal |
| 21:27.05 | YonahW | truly not a big deal |
| 21:27.32 | YonahW | i spend more time going through rss feeds on any given day |
| 21:28.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:37.51 | Gtwy | how important is it that i run those 2 iptables commands? http://gentoo-wiki.com/HOWTO_Asterisk |
| 21:39.03 | [TK]D-Fender | Gtwy, *whatever* |
| 21:39.13 | [TK]D-Fender | ToS doesn't exist ont he internet |
| 21:40.08 | Gtwy | okay, because im getting a very open ended error message with iptables that *something* is wrong with both commands, but it doesnt say what.. and i dont know enough about iptables to understand what im typing in there to begin with |
| 21:40.12 | Gtwy | thanks |
| 21:43.38 | lovely2 | can you write an agi script that changes a extensions status ? |
| 21:54.20 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 21:56.32 | crimethinker | The USA GDP is $43,866 per person. wtf are we producing that's worth so much? don't we import everything from China? |
| 21:57.27 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:58.09 | magic_hat | hey everyone. I've seen a whole bunch of stuff re installing * on ubuntu.... anyone done this and know which approach works best? |
| 22:06.41 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 22:06.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:07.50 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 22:07.53 | *** join/#asterisk mikpel (i=mikpel@h-195-210.A183.cust.bahnhof.se) |
| 22:09.42 | Aces1Up | with asterisk 1.4 do you need a timer for the meetme function? |
| 22:10.11 | mikpel | hi is it possible to connect ericsson buisnessphones to asterisk? using a channelbank? |
| 22:11.25 | tzafrir_laptop | magic_hat, apt-get install asterisk |
| 22:11.40 | tzafrir_laptop | fastest way to get it up and running |
| 22:15.09 | *** join/#asterisk snuff-away (n=snuffy@C-59-101-151-24.mel.connect.net.au) |
| 22:16.46 | *** join/#asterisk Bryce34 (n=Bryce34@juv34-4-82-238-91-177.fbx.proxad.net) |
| 22:19.11 | Bryce34 | Hello, do you know if the RFC 4040 will be integrated in Asterisk ? |
| 22:20.50 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 22:26.12 | b1shop | anyone know how to get polycom 320 to boot from tftp? |
| 22:26.52 | __DAW | same way all the other polycoms do I imagine. |
| 22:27.11 | b1shop | i can get to the screen on the phone. i do not seem to be able to change it. |
| 22:27.23 | b1shop | and the web interface does not have an option for it |
| 22:27.25 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:29.26 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 22:30.52 | __DAW | works great through the phone for me. |
| 22:31.44 | b1shop | menu, 2, 3, 1... then? |
| 22:33.29 | __DAW | nope |
| 22:33.41 | __DAW | 2 is status |
| 22:36.11 | Bryce34 | I search a solution to using the direct-call-pickup feature with a thomson ST2030. My BLF already working find. |
| 22:36.38 | Aces1Up | what is the typical place i should look at when i dial my DID and in rings for 1 ring then goes to fast busy? |
| 22:38.07 | b1shop | err. menu, 2, 1, 3. |
| 22:38.19 | b1shop | stuck on ftp and connot seem to find a way to change to tftp |
| 22:42.35 | *** part/#asterisk zabin (n=zabin@c-68-59-30-108.hsd1.sc.comcast.net) |
| 22:44.25 | b1shop | __DAW, if you have a 320 can you help me out? |
| 22:46.08 | __DAW | have your read your manual? How are you trying to do it? |
| 22:47.35 | b1shop | all it came with is a one page start guide. i have the admin manual from the site |
| 22:50.23 | __DAW | it would be wise to do some reading up. |
| 22:51.22 | b1shop | no kidding |
| 22:51.35 | __DAW | ;) |
| 22:51.42 | b1shop | the 188 page manual contains 9 occurannence of tftp. none of which talk about changin it |
| 22:57.19 | __DAW | have you tried pressing the check key when you are on ftp and then using the left and right arrows to change? |
| 22:58.40 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 22:59.02 | b1shop | yes. |
| 22:59.37 | __DAW | then you may need to call polycom. works no probs here. |
| 23:00.16 | __DAW | and it is 3,2,1 not 2,1,3 you are looking at the status page.. |
| 23:07.07 | b1shop | menu, 3(settings), 2(advanced) and passwd, 1(Change PW). only options i have |
| 23:07.44 | __DAW | did you enter the password |
| 23:07.59 | b1shop | default of 123 i think? |
| 23:08.13 | __DAW | you need to READ.. This is all in the manual!!! |
| 23:08.33 | __DAW | dont search for keywords... READ |
| 23:12.50 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:15.54 | *** join/#asterisk cb` (n=charlie3@pool-71-103-26-157.lsanca.dsl-w.verizon.net) |
| 23:17.30 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:19.31 | *** join/#asterisk paragonc (n=paragonc@c-67-188-83-121.hsd1.ca.comcast.net) |
| 23:20.47 | paragonc | im working on a voicexml application that dials in via a toll free line - looking to know if anyone knows of a carrier who can price lower than $.015 per min |
| 23:21.15 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:27.48 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 23:28.33 | *** join/#asterisk WindBack (n=jorge@host48.190-136-109.telecom.net.ar) |
| 23:29.00 | WindBack | _VoiceMeUp_COM, hello |
| 23:29.54 | `Sean | anyone know where i can get that s/correct/correct script that jbot is runing :)? |
| 23:30.45 | WindBack | I always use xlite to connect to my asterisk server (on windows) Now in linux I'm trying to configure ekiga, but the problem that I have is that I can send DTMF |
| 23:30.46 | Strom_M | you mean a regexp parser? :) |
| 23:30.57 | WindBack | Anybody can helpme |
| 23:30.59 | WindBack | ? |
| 23:31.48 | WindBack | When I press any number the asterisk server don't detect it |
| 23:32.14 | rob0 | Press some OTHER number instead. |
| 23:32.28 | rob0 | NO NOT THAT ONE!! |
| 23:32.35 | Strom_M | *beep* |
| 23:33.14 | WindBack | When I press a number the asterisk server don't detect it |
| 23:33.27 | WindBack | rob0, |
| 23:33.51 | Strom_M | WindBack: let me guess - SIP trunking |
| 23:34.06 | Strom_M | probably either one way audio and/or wrong DTMFmode |
| 23:34.17 | rob0 | ooooooooooh nice crystal ball! |
| 23:34.26 | Strom_M | thanks |
| 23:34.32 | Strom_M | I got it on sale at Target |
| 23:34.59 | Strom_M | it works fine except it keeps telling me about the low low prices I could be paying for consumer goods |
| 23:35.28 | WindBack | Strom_M, I configure on both sides DTMF RFC2833 but it didn't found |
| 23:36.10 | WindBack | Strom_M, In ekiga RFC2833 is the only option |
| 23:37.23 | Strom_M | alright, so now we turn to NAT issues |
| 23:37.24 | joebob777as7 | anyone know why when i try to connect a sip phone from another network outside of my own i get a message saying Registered SIP 210 at <myip> port 52450 expires 3600 in my cli? |
| 23:37.35 | Strom_M | because I'm going to bet $50 you're using a NAT between ekiga and asterisk |
| 23:37.58 | Strom_M | joebob777as7: asterisk is behind a NAT, isnt it |
| 23:38.05 | joebob777as7 | yup |
| 23:38.19 | Strom_M | sigh |
| 23:38.39 | joebob777as7 | but i'm trying to connect my e60 nokia out of network... how do i get around this? |
| 23:39.02 | Strom_M | have you set externip, localnet, all the appropriate port forwarding, and so on? |
| 23:39.47 | joebob777as7 | i believe so... all i could find to set... i set up port 5060 to my box and i am using dyndns |
| 23:39.55 | WindBack | Strom_M, no betwen ekiga and asterisk there isn't nat, both are in the same LAN with similar ips |
| 23:40.13 | WindBack | (with the same network mask) |
| 23:40.34 | Strom_M | WindBack: well then perhaps Ekiga blows dead yaks. Try a different softphone. |
| 23:41.15 | joebob777as7 | Strom_M, any suggestions? |
| 23:41.20 | Strom_M | joebob777as7: do calls actually work? |
| 23:41.23 | WindBack | Strom_M, Do you know about any who work fine in gnome?? |
| 23:41.36 | Strom_M | or are you just complaining because something /looks/ wrong on teh screen? |
| 23:41.42 | joebob777as7 | Strom_M, no my phone says registration fails |
| 23:41.52 | Strom_M | WindBack: beats me; all softphones blow really |
| 23:42.10 | joebob777as7 | windback try x-lite |
| 23:42.23 | Strom_M | joebob777as7: well then you probably havent set SIP up correctly on asterisk and all NAT devices involved |
| 23:42.27 | joebob777as7 | WindBack, there are plenty of guides |
| 23:42.45 | joebob777as7 | Strom_M, ok would you mind helping me do that? :) :) |
| 23:42.47 | Strom_M | only run SIP behind a NAT if you own stock in a headache pill manufacturer |
| 23:42.55 | joebob777as7 | sorry for my extreme newbness |
| 23:42.59 | pipwerk | ~sipnat |
| 23:43.10 | jbot | i guess sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:43.10 | joebob777as7 | I own a pharmacy does that count? lol |
| 23:43.15 | WindBack | joebob777as7, xlite work fine, but sometimes it sound really bad |
| 23:43.20 | WindBack | (in linux |
| 23:43.21 | WindBack | ) |
| 23:43.21 | Strom_M | joebob777as7: pastebin.ca your configs |
| 23:43.40 | WindBack | joebob777as7, but in windows it really work fine |
| 23:44.03 | Strom_M | WindBack: well then, ekiga really does blow dead yaks |
| 23:45.11 | Strom_M | aww, i hurt his feelings |
| 23:45.45 | rob0 | Just think about the poor dead yaks. |
| 23:45.53 | Strom_M | i know :/ |
| 23:51.01 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:51.10 | Strom_M | so are you done flooding me? |
| 23:51.25 | Strom_M | because when I ask you to use pastebin, I don't mean "send me a million PMs" |
| 23:51.27 | _VoiceMeUp_COM | Jun 23 19:50:55 WARNING[14968]: format_wav.c:183 check_header: Can only handle 16bits per sample: 1 |
| 23:51.39 | _VoiceMeUp_COM | that means.. 8k stereo ? |
| 23:52.07 | Strom_M | no |
| 23:52.27 | Strom_M | _VoiceMeUp_COM: you should know how ulaw/alaw companding works |
| 23:52.34 | _VoiceMeUp_COM | hmmm |
| 23:52.49 | Strom_M | i mean, you /are/ an ITSP, right? :) |
| 23:53.00 | _VoiceMeUp_COM | ahahah |
| 23:53.03 | Strom_M | you /do/ actually /know/ what you're working with, right? |
| 23:53.20 | _VoiceMeUp_COM | instead of making fun of me you could help |
| 23:53.24 | _VoiceMeUp_COM | i didnt made the files |
| 23:53.39 | _VoiceMeUp_COM | so i dont know whate they where saved in |
| 23:53.44 | Strom_M | i'm not making fun of you; i'm trying to ascertain whether you understand telephony |
| 23:53.56 | _VoiceMeUp_COM | and wats does a wav file have to do with this ? |
| 23:53.57 | _VoiceMeUp_COM | lol |
| 23:54.02 | _VoiceMeUp_COM | anyhow nevermind i found out |
| 23:54.03 | _VoiceMeUp_COM | thanks |
| 23:54.13 | Strom_M | go read ITU-T G.711 |
| 23:54.21 | Strom_M | much to be learned |
| 23:55.39 | Strom_M | joebob777as7: ok, so looking at your pastebin, you apparently didn't set externip, or localnet, or anything else you're supposed to set in order to run asterisk as a SIP server behind NAT |
| 23:56.01 | Strom_M | joebob777as7: furthermore, it looks like you're using trixbox |
| 23:56.20 | __DAW | ~trixbox |
| 23:56.20 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 23:57.15 | joebob777as7 | great since people have modified things to make them simpler there is no support. sweet |
| 23:57.18 | *** part/#asterisk joebob777as7 (n=C@yoda.peacefulescape.com) |
| 23:57.44 | pipwerk | no, there is support, just not at #asterisk |
| 23:57.51 | pipwerk | hmmm |
| 23:57.58 | pipwerk | usefull, not :( |
| 23:59.20 | Strom_M | welcome to asterisk |
| 23:59.22 | Strom_M | brain required |