| 00:03.40 | Hymie | I just specify extra -vvvvv in the init script |
| 00:05.49 | CaT[tm] | hassle is that prevents asterisk from backgrounding and I'd say it has a nasty affect on the zombie population of the server. |
| 00:06.01 | CaT[tm] | the latter is a theory I'm working on though |
| 00:06.24 | *** part/#asterisk exvito (n=exvito@89.181.10.10) |
| 00:13.21 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-30.ph.ph.cox.net) |
| 00:17.53 | vutamhoan | I had problem with atxfer - the transferee only ring in 5s then hangup :( - If I use blind everything ok |
| 00:18.25 | vutamhoan | both of them use the same dialplan for transfer |
| 00:18.35 | *** join/#asterisk blitzrage[E61i] (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:18.35 | *** mode/#asterisk [+o blitzrage[E61i]] by ChanServ |
| 00:19.21 | vutamhoan | can anyone help me, thanks |
| 00:26.04 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:26.56 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-72f8609e26378947) |
| 00:28.06 | *** join/#asterisk elixer (n=seanbrig@c-69-251-152-9.hsd1.md.comcast.net) |
| 00:36.10 | *** join/#asterisk sopo2k4 (n=jam@host86-153-44-68.range86-153.btcentralplus.com) |
| 00:36.26 | sopo2k4 | hey, does anyone have experience with the Asterisk Manager? |
| 00:36.56 | sopo2k4 | for some reason using the Originate function, its parson "s" as the CALLED Number :s |
| 00:37.02 | sopo2k4 | parsing* |
| 00:37.50 | citats | sopo2k4: what exactly are you sending over? |
| 00:38.04 | sopo2k4 | -- Executing [failed@outgoingWestern:1] Dial("OutgoingSpoolFailed", "IAX2/voicepulse/01144failed") in new stack |
| 00:38.34 | *** join/#asterisk gpoppo (n=gimpop@cluckhouse.demon.co.uk) |
| 00:39.29 | citats | sopo2k4: can you stick what your sending over via the manager interface in a pastebin? |
| 00:39.30 | sopo2k4 | citats |
| 00:39.32 | sopo2k4 | http://pastebin.com/m58e5fdd4 |
| 00:39.36 | gpoppo | good evening! |
| 00:40.01 | sopo2k4 | tryna implement it into my vb application, :P can login n stuff |
| 00:41.05 | citats | sopo2k4: are you trying to dial that number under exten out the IAX2/voicepulse? |
| 00:41.19 | gpoppo | i want to try out a module - chan_cellphone, but i can only find it in source format. i haven't yet compiled asterisk, since the binary packaged in fedora seems to work fine.. |
| 00:41.34 | gpoppo | do i need to get asterisk source and compile in order to try that module? |
| 00:42.01 | sopo2k4 | yes |
| 00:42.04 | sopo2k4 | i am. |
| 00:42.26 | gpoppo | or can i just compile the module - using the source from here (?), and then drop that into the relevant directory, etc? |
| 00:42.45 | gpoppo | the source for chan_cellphone, i would get from here: http://bugs.digium.com/view.php?id=8919 |
| 00:42.46 | citats | sopo2k4: you need to use something like Channel: IAX2/voicepulse/number |
| 00:42.49 | sopo2k4 | thats what ive got in outgoingWestern context inside extensions.conf |
| 00:43.01 | sopo2k4 | ic |
| 00:43.03 | sopo2k4 | let me try |
| 00:43.05 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583527.dsl.bell.ca) |
| 00:43.17 | citats | sopo2k4: wait, so your trying to bridge 2 outgoing calls? |
| 00:43.36 | sopo2k4 | basically |
| 00:43.47 | sopo2k4 | dial 1 number, wait for answer then dial another and connect them together. |
| 00:43.56 | sopo2k4 | but i cant even get it to dial 1st number yet lol |
| 00:44.33 | citats | sopo2k4: the first number you want to call needs to be passed in your channel line |
| 00:44.53 | sopo2k4 | never mind got it ringing |
| 00:44.53 | sopo2k4 | :P |
| 00:45.25 | sopo2k4 | cheers |
| 00:45.41 | sopo2k4 | needed to put the number after IAX2/voicepulse/ :P |
| 00:45.58 | sopo2k4 | would of thought it would parse what u put in Exten tho. |
| 00:46.11 | citats | sopo2k4: heh yeah gotta supply that info, otherwise it will go to default |
| 00:46.33 | citats | sopo2k4: the stuff in exten/context/priority or app/data only gets used after the call to Channel is placed |
| 00:46.55 | sopo2k4 | ic..... |
| 00:47.05 | sopo2k4 | so it would bridge the calls? |
| 00:47.21 | sopo2k4 | im not following, cos my asterisk rung me twice lol |
| 00:47.31 | sopo2k4 | when i initated only 1 Originate commands |
| 00:47.59 | citats | sopo2k4: your basically setting it up to dial whatever is channel, then after that bridge it with whatever is at exten/priority/context you passed |
| 00:48.25 | sopo2k4 | ahh ic |
| 00:48.25 | DarkRift | Is the spa3102 a good starting point if I wanted to setup a simple test with asterisk ? |
| 00:48.47 | citats | sopo2k4: did you want it to do something besides bridge the 2 channels? |
| 00:48.53 | DarkRift | I mean, if I wanted to try it out for the first time |
| 00:49.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:49.19 | sopo2k4 | nah, out of curiosity, would what i have set as Callerid: be shown to both numbers its calling? |
| 00:50.20 | citats | sopo2k4: the callerid: var will be used for the channel:, and i would think it would just be copied to the other channel. if not you could set it in the dialplan |
| 00:50.57 | sopo2k4 | there was one other thing |
| 00:51.38 | sopo2k4 | i wanted to use IAX2/voicepulse for the first call (to the person who initated the command), and use IAX2/voipjet for the number its going to dial then bridge to the channel |
| 00:51.43 | sopo2k4 | any idea's on that one? |
| 00:52.20 | citats | sopo2k4: so leave the IAX2/voicepulse in your Channel: and then set the exten under the outgoingwestern context to use IAX2/voipjet |
| 00:52.48 | sopo2k4 | ok mate, cheers for that :P |
| 00:53.13 | citats | wife just got home, might be time for dinner. take care |
| 00:53.23 | sopo2k4 | you too |
| 00:55.25 | dlynes | So what's going to be new in 2.6? |
| 00:55.45 | dlynes | Is the biggest difference going to be the new sip stack? Or will that be getting put into 2.6? |
| 00:55.51 | dlynes | erm 1.6, I mean? |
| 00:56.46 | dlynes | CaT[tm]: you mean anything less serious than error? |
| 00:57.02 | CaT[tm] | well... not debug or verbose levels :) |
| 00:57.21 | dlynes | CaT[tm]: got a pastebin of your logger.conf file? |
| 00:57.33 | dlynes | CaT[tm]: what verbosity level do you have it set to? |
| 00:57.33 | CaT[tm] | one sec. |
| 00:57.37 | dlynes | CaT[tm]: are you using 1.2 or 1.4? |
| 00:58.03 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 00:58.29 | CaT[tm] | A.2 |
| 00:58.41 | CaT[tm] | yeah we went and splurged. |
| 00:59.19 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 00:59.47 | CaT[tm] | btw. this is the line: full => notice,warning,error,debug,verbose |
| 01:00.45 | dlynes | CaT[tm]: A.2? you mean 1.2? |
| 01:01.13 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:01.15 | CaT[tm] | this is the config file, sans comments: http://pgsql.privatepaste.com/ae1hA5DV4E |
| 01:01.24 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 01:01.42 | dlynes | CaT[tm]: in your asterisk cli, type 'set verbose 100' |
| 01:02.00 | dlynes | CaT[tm]: then type 'logger restart' |
| 01:02.13 | dlynes | CaT[tm]: then try doing something...see if you're getting higher than 'error' level logging |
| 01:03.16 | CaT[tm] | just saw some verbose lines float past. |
| 01:03.38 | CaT[tm] | hrm. how do I set this in a config file? |
| 01:03.40 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:03.43 | dlynes | there ya go then |
| 01:04.28 | dlynes | CaT[tm]: in your system startup script, wait for asterisk to start...after it's started do an "asterisk -rx 'set verbose 100'" |
| 01:05.06 | x86 | why dont yall use inittab to start asterisk like the pros do it? :P |
| 01:05.06 | CaT[tm] | dlynes: thanks. is it safe to say that once the process forks, the above can be done? |
| 01:05.06 | dlynes | CaT[tm]: but, normally if everything's functioning well, you shouldn't need that high of a verbosity level |
| 01:05.17 | dlynes | CaT[tm]: correct |
| 01:05.36 | CaT[tm] | my asterisk monkey is learning and so would like to see as much info as possible if things go boom. |
| 01:05.49 | dlynes | CaT[tm]: so, normally if I need that high of a verbose level, I type 'asterisk -r', and do the set verbose 100 manually |
| 01:06.16 | CaT[tm] | the box isn't doing much and has the disk space, so... |
| 01:07.43 | elixer | CaT[tm]: do you use safe_asterisk? |
| 01:10.14 | elixer | CaT[tm]: if you do, you can change this line: ASTARGS="${ASTARGS} -vvvvg" |
| 01:10.18 | elixer | include more 'v's |
| 01:14.33 | sopo2k4 | Im looking to use the Asterisk Manager to initate calls, however id like to give each manager.conf user a specific amount of calls allowed to be initated before contacting for more. |
| 01:14.35 | *** join/#asterisk elixer (n=seanbrig@c-69-251-152-9.hsd1.md.comcast.net) |
| 01:14.40 | sopo2k4 | any idea how i could read/set a static variable |
| 01:14.54 | elixer | static? |
| 01:15.08 | sopo2k4 | Im looking to use the Asterisk Manager to initate calls, however id like to give each manager.conf user a specific amount of calls allowed to be initated before contacting for more. |
| 01:15.15 | sopo2k4 | before ... |
| 01:16.03 | elixer | i don't understand. |
| 01:16.24 | sopo2k4 | ok hmmm..... |
| 01:16.39 | sopo2k4 | you know if someone wanted to sell calling cards using their asterisk systems. |
| 01:16.50 | elixer | yes |
| 01:17.10 | elixer | look at astdb |
| 01:17.13 | *** join/#asterisk gneill794 (n=gneill79@bob.neillnet.com) |
| 01:17.13 | sopo2k4 | how would they go about doing that but instead of being charged per minute, they have a certain amount of calls. |
| 01:17.36 | sopo2k4 | i need to know how to use the asterisk manager to read / set a variable that is stored for each user. |
| 01:17.53 | elixer | astdb |
| 01:17.57 | sopo2k4 | ok let me read up |
| 01:18.00 | elixer | database put, database get, etc. |
| 01:18.56 | CaT[tm] | elixer: that would cause a lack of forkage. |
| 01:19.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:19.15 | elixer | how's that? |
| 01:19.21 | CaT[tm] | -v implies -f |
| 01:19.49 | CaT[tm] | -d also implies -f |
| 01:20.01 | CaT[tm] | (just fyi :) |
| 01:20.49 | elixer | weird. ok. |
| 01:21.09 | sopo2k4 | elixer, using the asterisk manager, id use Action: DBget(varname=family/key) right? |
| 01:21.16 | elixer | sopo2k4: yup |
| 01:21.22 | sopo2k4 | ty |
| 01:22.23 | CaT[tm] | dlynes: btw. thanks for the assist. |
| 01:22.34 | dlynes | CaT[tm]: np |
| 01:22.51 | Corydon76-home | sopo2k4: you'd do what? |
| 01:23.14 | sopo2k4 | <sopo2k4> elixer, using the asterisk manager, id use Action: DBget(varname=family/key) right? |
| 01:23.23 | dlynes | Corydon76-home: he'd do you |
| 01:23.23 | elixer | CaT[tm]: where in the asterisk 1.2 code does -v imply -f? |
| 01:23.30 | *** part/#asterisk gneill794 (n=gneill79@bob.neillnet.com) |
| 01:23.32 | elixer | CaT[tm]: trying to find it |
| 01:23.48 | elixer | in main() somewhere? |
| 01:23.59 | CaT[tm] | elixer: manpage plus cause and affect. :) |
| 01:24.12 | elixer | CaT[tm]: nm, found it |
| 01:25.10 | Corydon76-home | sopo2k4: Action: GetVar\r\nChannel: foo\r\nVariable: DB(family/key)\r\n\r\n |
| 01:25.18 | clyrrad | This is so strange, I am trying to get ztdummy and zaptel installed so I can use Page() - I am able to compile zaptel and I get no error, but when I modprobe zaptel I get FATAL: Module zaptel not found. Can anyone help me get this going, or point me in the right direction? |
| 01:26.04 | JT | you don't modprobe zaptel. |
| 01:26.06 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-72f8609e26378947) |
| 01:26.09 | JT | but it should be there |
| 01:26.14 | Corydon76-home | clyrrad: most likely, your kernel source specified a kernel other than what's installed |
| 01:26.44 | clyrrad | Corydon76-home: I checked my ln -s for /usr/src/linux and its pointed to the proper place |
| 01:26.49 | Corydon76-home | clyrrad: it's a favorite act of many distros to add the word "custom" onto the end of your kernel version, so you'll install to the wrong directory |
| 01:27.03 | Corydon76-home | clyrrad: check /lib/modules |
| 01:27.07 | clyrrad | but in /lib/modules I see many different kernel versions there - coudl that be the issue? |
| 01:27.14 | JT | clyrrad: that's not the location of compiled modules |
| 01:27.27 | JT | it could be |
| 01:27.31 | JT | if your symlinks are bad |
| 01:27.46 | clyrrad | my symlinks to the kernel headers are good..... |
| 01:27.53 | clyrrad | im just not sure about /lib/modules |
| 01:28.12 | Corydon76-home | clyrrad: as I said, it's not about the source directory, per se. The version string has to match EXACTLY |
| 01:28.22 | JT | what about kernel source? |
| 01:28.43 | clyrrad | Corydon76-home: Yep, if I uname -r there is no EXACT match in /lib/modules - how do I correct this? |
| 01:29.00 | Corydon76-home | clyrrad: compare: uname -a to: find /lib/modules -name zaptel.ko |
| 01:29.36 | Corydon76-home | clyrrad: probably the easiest for you is to custom compile your own kernel |
| 01:30.34 | clyrrad | . /lib/modules/2.6.9-55.0.2.ELsmp/extra/zaptel.ko .........vs................6.9-55.ELsmp |
| 01:30.34 | Corydon76-home | If you know what you're doing, you can reverse what the distro packagers did to crap out your source |
| 01:30.41 | elixer | ok, i don't get it. safe_asterisk out of the box doesn't fork? |
| 01:30.50 | Corydon76-home | elixer: correct |
| 01:30.59 | Corydon76-home | elixer: it's just a shell script |
| 01:31.02 | elixer | i know |
| 01:31.04 | clyrrad | Corydon76-home: ok, so I need to re-compile my kernel? |
| 01:31.05 | elixer | i'm looking at it now |
| 01:31.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9d742b9002034fbc) |
| 01:31.21 | Corydon76-home | clyrrad: correct |
| 01:31.24 | elixer | so you shouldn't use it or you should set TTY=[nada] |
| 01:31.25 | elixer | ? |
| 01:31.43 | Corydon76-home | elixer: no, you should use it and set the tty to TTY9 |
| 01:31.48 | clyrrad | Corydon76-home: hrm.... was hoping to avoide that -alright..... thanks |
| 01:32.02 | elixer | interesting. |
| 01:32.25 | Corydon76-home | That lets you view the console on Ctrl-Alt-F10 |
| 01:32.30 | elixer | right |
| 01:32.46 | elixer | why would i want asterisk to fork? |
| 01:33.16 | Corydon76-home | Some people prefer standalone daemon mode |
| 01:33.26 | elixer | ah |
| 01:33.50 | *** join/#asterisk iBuMp (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 01:35.47 | elixer | are there any performance benefits or implications one way or the other? |
| 01:35.52 | elixer | i'm sorry to ask dumb questions |
| 01:36.04 | elixer | but i'm just now realizing that my server is set up this way |
| 01:36.04 | elixer | :) |
| 01:39.54 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:44.59 | lmadsen | clyrrad: I bet you haven't rebooted since the last time yum or whatever package manager did a system update and installed a new kernel |
| 01:45.59 | sopo2k4 | anyone know what would cause the Asterisk CLI to show this every 10 seconds |
| 01:46.00 | sopo2k4 | <PROTECTED> |
| 01:46.00 | sopo2k4 | <PROTECTED> |
| 01:46.01 | sopo2k4 | ? |
| 01:46.17 | clyrrad | lmadsen: that would be a good guess :p - this thing has not been rebooted in ages |
| 01:46.31 | lmadsen | clyrrad: I would almost bet money on that being your problem :) |
| 01:46.33 | JT | you have some cron job running or a management interface or something |
| 01:46.48 | clyrrad | lmadsen: hehe it pains me to reboot it - but that is a very good point you raise |
| 01:47.27 | lmadsen | yep -- I've run into that before. I'm five nines confident that'll fix THAT problem -- I make no guarantee to the creation of new ones :) |
| 01:48.04 | clyrrad | alright we are after hours now anyway.........lets reboot this bad boy |
| 01:50.37 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:50.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:52.19 | x86 | sopo2k4: most likely you have some sort of monitoring that connects periodically to check on things... mrtg channel usage monitoring perhaps? |
| 01:53.05 | clyrrad | lmadsen: alright reboot done - shall I recompile the zaptel again? |
| 01:53.15 | lmadsen | no I don't think so :) |
| 01:53.21 | lmadsen | (I'm being sarcastic) |
| 01:53.35 | clyrrad | lol |
| 01:54.28 | clyrrad | i still dont ahve a /lib/modules that matches my uname -r |
| 01:54.32 | clyrrad | :( |
| 01:55.13 | clyrrad | ha! no errors now :D |
| 01:56.02 | lmadsen | ;) |
| 01:56.03 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:56.09 | clyrrad | lmadsen: zaptel started up now :D |
| 01:56.16 | lmadsen | perfec |
| 01:56.19 | lmadsen | +t |
| 01:56.19 | clyrrad | woot! |
| 01:56.26 | lmadsen | and you didn't even have to build a new kernel :) |
| 01:56.37 | clyrrad | indeed! I would not have thought to do a reboot either |
| 01:56.56 | clyrrad | so used to never rebooting, I had an uptme of 243 days before tonight |
| 01:57.19 | lmadsen | heh... uptimes are overrated |
| 01:57.41 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 01:57.48 | lmadsen | that just means you have 243 days of possible kernel issues :) |
| 01:57.50 | AJaymn | Anyone have Asterisk running on Fedora Core 6? |
| 01:57.53 | clyrrad | yea - that was a good call on your part - many thanks :) |
| 01:58.32 | lmadsen | clyrrad: ;) |
| 02:00.22 | *** join/#asterisk remmo (n=junk@203.32.47.253) |
| 02:03.08 | gpoppo | i am running it on fc5 AJaymn, if that answers in any way? |
| 02:03.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:05.03 | carrar | http://ln.ooz.net/35047 |
| 02:06.29 | *** join/#asterisk vutamhoan (n=hoavq@125.235.97.221) |
| 02:08.22 | AJaymn | lol sorry :P |
| 02:09.12 | AJaymn | gpoppo stupid question.. how do i install the Development Tools? :P my datacenter only installs the base system.. and im use to the old Slackware setup wizard thing |
| 02:09.59 | gpoppo | development tools? hmm.. i'm a newbie myself.. but at a guess, there must be a corresponding asterisk-dev in rpm package format |
| 02:10.19 | gpoppo | #yum search asterisk and see what you find, perhaps |
| 02:11.00 | AJaymn | how did you install asterisk on c5? was there a howto? |
| 02:11.03 | gpoppo | you could compile asterisk from source, that's what i'm just doing |
| 02:11.22 | AJaymn | thats what i was doing too but it needs development tools |
| 02:11.23 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 02:11.32 | JT | doesn't slackware use pkg |
| 02:11.38 | gpoppo | i just installed asterisk from rpm.. |
| 02:11.50 | gpoppo | yum install asterisk.. that should handle all dependencies |
| 02:11.58 | JT | gpoppo: no. |
| 02:12.07 | JT | gpoppo: did you actually read what AJaymn's scenario was? |
| 02:12.16 | JT | slackware does not use yum. |
| 02:12.22 | AJaymn | oh no no im using Core 6 |
| 02:12.24 | JT | rpm based distros use yum. |
| 02:12.32 | JT | you mean Fedora |
| 02:12.38 | AJaymn | sorry im just use to slackwares old setup |
| 02:12.45 | AJaymn | yes the server is running Fedora core 6 |
| 02:12.54 | gpoppo | did YOU actually read what his scenario was, JT?! |
| 02:12.58 | JT | i see |
| 02:13.07 | AJaymn | lol dont fight boys :) |
| 02:13.16 | JT | gpoppo: zomg?@!?!?! |
| 02:13.20 | gpoppo | what sort of development tools? |
| 02:13.26 | gpoppo | gmake, gcc, etc? |
| 02:13.26 | JT | make and gcc i asusme |
| 02:13.28 | JT | assume |
| 02:13.34 | AJaymn | http://fedoraapp.blogspot.com/ |
| 02:13.39 | AJaymn | thats what i was looking to follow |
| 02:13.46 | AJaymn | but if there is an easier way... |
| 02:14.01 | gpoppo | is that your blog? |
| 02:14.06 | gpoppo | sorry, no. not. |
| 02:14.28 | gpoppo | yeah, just issue a #yum install asterisk |
| 02:14.45 | gpoppo | you can't go far wrong. it shouldn't bork anything |
| 02:14.48 | AJaymn | last time i did a yum install my system wouldnt reboot :P |
| 02:14.56 | gpoppo | hehe! know the feeling! |
| 02:15.12 | gpoppo | debian is no better either! |
| 02:15.17 | JT | AJaymn: using yum to install the development tools is a good idea, but not to install asterisk |
| 02:15.35 | JT | eh, the debian package management system is far better |
| 02:15.46 | gpoppo | AJaymn, I leave you in JT's capable hands! |
| 02:15.55 | AJaymn | hah thanks for the help |
| 02:16.12 | AJaymn | JT do u know what ver yum will install? |
| 02:16.28 | JT | nope |
| 02:16.46 | gpoppo | # yum search asterisk will tell you |
| 02:17.14 | JT | are we talking about development tools or the asterisk package? |
| 02:17.43 | AJaymn | sorry.. asterisk |
| 02:18.00 | gpoppo | on FC5, asterisk.i386 1:1.4.6-41.fc5 is installed from redhat RPM repository |
| 02:18.01 | JT | it's not advisable to install asterisk from packages |
| 02:18.26 | gpoppo | why not, out of curiosity? |
| 02:18.48 | JT | because usually they have stupid defaults, strange patches, or lack of patches, and old versions |
| 02:19.11 | gpoppo | ahh.. hmm.. i've got that same problem with the tarball though... |
| 02:19.30 | JT | weird, getting the latest tarball doesn't seem to get an old version |
| 02:19.42 | JT | or have really silly defaults, or any patches |
| 02:19.49 | AJaymn | ok my past asterisk boxes have been trixboxes... whats the difference from 1.4 and the 1.2 vers? is 1.4 bleeding edge stuff? or... |
| 02:19.57 | gpoppo | anyway, i'll keep my nose out, since i don't really know what i'm talking about. |
| 02:20.11 | JT | AJaymn: 1.2 is meant to be more stable, but you may aswell start on 1.4 |
| 02:20.20 | JT | as 1.2 will be going into maintenance soon |
| 02:20.46 | AJaymn | id hate to set somthing up and have to update in 2 days :P |
| 02:21.06 | JT | haha avoid asterisk then ;) they seem to be releasing a new version every 2 weeks |
| 02:21.17 | AJaymn | well u know what i mean :P |
| 02:22.23 | AJaymn | ok JT since you seem to be so diverse in Asterisk ;) I have a box that keeps going into Read Only.. is that a HD issue? or is Asterisk taking over my world? |
| 02:22.48 | JT | when it boots, or while it's already up? |
| 02:22.56 | AJaymn | after running.. |
| 02:23.05 | AJaymn | somtimes goes days without going into RO.. |
| 02:23.08 | JT | sounds like the hard drive is about to die |
| 02:23.08 | AJaymn | sometimes hours |
| 02:23.31 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 02:25.51 | gpoppo | in a root terminal you could try tail'ing /var/log/messages, to see if you get any drive errors reported |
| 02:25.56 | gpoppo | i.e.: |
| 02:26.01 | gpoppo | tail -f /var/log/messages |
| 02:26.21 | gpoppo | 'grep' the old syslog too, that might shed some light |
| 02:26.46 | RypPn | is it sata or pata? |
| 02:27.08 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-5487adc83781dab6) |
| 02:29.43 | [TK]D-Fender | JT : 1.2 IS already in maintenance only mode... security fixes only. Time to jump onto the 1.4 band-wagon now. |
| 02:29.53 | AJaymn | Warning: Unknown: open(/tmp/sess_5e4c3d4d19fbb775c9f9933e9c0e68f2, O_RDWR) failed: Read-only file system (30) in Unknown on line 0 |
| 02:30.21 | JT | [TK]D-Fender: there was recent maintenance, it seems ;) |
| 02:30.30 | JT | AJaymn: buy a new hard drive |
| 02:30.35 | AJaymn | lol |
| 02:30.55 | AJaymn | trying to setup a new box.. but was having issues installing asterisk :P |
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| 02:47.10 | CoaxD | oh my god, my nufone 800 number works |
| 02:47.16 | CoaxD | after weeks of downtime. it works |
| 02:47.25 | CoaxD | yay for really awesome unstable voip telcos |
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| 02:54.15 | J4k3 | a lot of cable, all the switches, etc. it was all going to crap |
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| 03:02.05 | [TK]D-Fender | J4k3, Wso what you're saying is... that Vitelity was the straw that broke the camels back and forced yout o change EVERYTHING huh? :) |
| 03:06.55 | *** join/#asterisk prizark_ (i=prizark_@bb220-255-230-20.singnet.com.sg) |
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| 03:11.45 | AJaymn | JT you still around? |
| 03:12.26 | JT | yes |
| 03:12.38 | AJaymn | yum install asterisk |
| 03:12.46 | Juggie | wrong window! |
| 03:12.51 | AJaymn | comes back... Nothing to do here |
| 03:13.05 | JT | AJaymn: i said not to use yum to install asterisk, but anyway |
| 03:13.12 | AJaymn | oh lol |
| 03:13.21 | JT | compile from source |
| 03:13.24 | AJaymn | geer :P |
| 03:13.59 | AJaymn | http://fedoraapp.blogspot.com/ here these good directions? |
| 03:14.36 | [TK]D-Fender | AJaymn, COMPILE <- |
| 03:15.27 | AJaymn | sorry i guess i dont understand the difference between the directions on the blog, and compliling.. |
| 03:16.03 | [TK]D-Fender | AJaymn, www.asterisk.org <- Go read the install instructions. |
| 03:18.02 | Juggie | AJaymn, that blog is pointing to old versions too |
| 03:18.47 | AJaymn | well i seen that. didnt think it would take much to change the #s to correct ones |
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| 03:22.22 | flenders | AJaymn: I had problems using packages once, installed the same version from source and problems went away... |
| 03:24.26 | AJaymn | <PROTECTED> |
| 03:26.25 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:27.29 | flenders | AJaymn: /join #fedora |
| 03:28.57 | *** join/#asterisk pacneil (n=pacneil@68.15.17.81) |
| 03:29.41 | pacneil | anybody know if there's a channel for astlinux? |
| 03:30.06 | pacneil | nm |
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| 03:35.46 | lmadsen | pacneil: ;) |
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| 03:37.02 | lmadsen | AJaymn: watch for TFoT2 in the next couple of weeks. You'll like the package matrix in Chapter 3 so you know what to install starting from a minimal install |
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| 03:42.55 | [TK]D-Fender | AJaymn, Just download the source, follow the instructions, and when (if) it whines at you about something missing, then ask about it. |
| 03:43.06 | [TK]D-Fender | AJaymn, just get off your ass and get started! |
| 03:43.27 | [TK]D-Fender | AJaymn, All this talk, talk, talk, when you should be ACTING.. sheesh..... |
| 03:44.30 | AJaymn | lol |
| 03:44.33 | AJaymn | Im trying! |
| 03:47.21 | AJaymn | http://www.asterisk.org/support/install |
| 03:47.26 | AJaymn | these directions SUCK! :P |
| 03:48.09 | JT | AJaymn: untar the tarball. |
| 03:48.21 | JT | in the root of the tarball is a file called README or INSTALL, read it. |
| 03:48.44 | AJaymn | lol |
| 03:50.20 | JT | standard practice for compiling a piece of software |
| 03:50.32 | AJaymn | that its this easy for you guys |
| 03:52.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:57.52 | Sweeper | [TK]D-Fender: hey, polycom fanboi, how can I disable the keypad on a polycom? basically, I want one phone to dial one number when it's picked up, and be basically locked out otherwise |
| 04:04.06 | lmadsen | Sweeper: setup the dialplan so that's how it works |
| 04:04.10 | lmadsen | (on the phone) |
| 04:07.49 | [TK]D-Fender | Sweeper, not sure if a "||" in your digitmap might do it (captured null) |
| 04:17.39 | [TK]D-Fender | AJaymn, its called README because you're expected to ignore it :) |
| 04:18.12 | [TK]D-Fender | AJaymn, our kind here respond wel to giant-flashing-neon-sign type filenames in tarballs :) |
| 04:22.35 | AJaymn | :o |
| 04:23.42 | *** part/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 04:32.13 | AJaymn | ok... |
| 04:32.16 | AJaymn | You do not appear to have the sources for the 2.6.20-1.2944.fc6 kernel installed. |
| 04:37.49 | [TK]D-Fender | AJaymn, "yum install kernel-headers" |
| 04:39.37 | [TK]D-Fender | AJaymn, and "kernel-devel*" |
| 04:39.48 | [TK]D-Fender | AJaymn, and "kernel-utils" |
| 04:40.37 | AJaymn | ok this scares me.. this is were i screw up last time and my box wouldnt reboot after a yum update |
| 04:41.15 | flenders | AJaymn: you're not changing the kernel... you're just installing the headers for your running kernel |
| 04:42.36 | AJaymn | yum install kernel-headers want to install a different ver then what kernel im running.. |
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| 04:46.17 | AJaymn | flenders: will this still work with wrong versions? |
| 04:46.32 | AJaymn | since when i try to compile it sees i dont have the ver devl |
| 04:48.29 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
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| 04:51.42 | AJaymn | [TK]D-Fender there someplace i can get my current ver kernel devel? |
| 04:52.02 | TheNewAndy | uname -r? |
| 04:52.04 | [TK]D-Fender | AJaymn, just pic the base, it'll pic the version automatically |
| 04:52.10 | [TK]D-Fender | that too |
| 04:52.31 | AJaymn | 2.6.20-1.2944.fc6 |
| 04:52.41 | AJaymn | This is what it wants to download: |
| 04:52.43 | AJaymn | <PROTECTED> |
| 04:54.01 | AJaymn | will that ver work? or will compiling complain its not the current kernel? |
| 04:56.02 | fujin | you'll have to recompile stuff that was built against the old one |
| 04:56.08 | fujin | zaptel, other kernel modules |
| 04:56.22 | AJaymn | http://koji.fedoraproject.org/koji/buildinfo?buildID=2623 |
| 04:56.24 | AJaymn | found it! |
| 04:56.29 | AJaymn | but what rpm will i need? |
| 04:56.39 | AJaymn | oh?! |
| 04:56.42 | fujin | what are you trying to do? |
| 04:56.45 | AJaymn | what did i say? |
| 04:56.49 | fujin | you said the r word |
| 04:56.54 | AJaymn | oh heh |
| 04:57.38 | AJaymn | im trying to compile zaptel at the moment |
| 04:57.44 | fujin | ah |
| 04:57.45 | AJaymn | i get You do not appear to have the sources for the 2.6.20-1.2944.fc6 kernel installed. |
| 04:57.54 | fujin | I'm not sure how to resolve that, sorry |
| 04:58.00 | fujin | fedora is a silly distro |
| 04:58.12 | fujin | You probably have to do something with yum, to install the -dev package for kernel |
| 04:58.16 | fujin | or the srpm or whatever. |
| 04:58.34 | AJaymn | yum wants to install a new version then the kernel i have running |
| 04:58.46 | fujin | that's probably advisable |
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| 05:09.11 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 05:09.15 | rue_mohr | hello |
| 05:09.18 | rue_mohr | have a problem |
| 05:09.20 | fujin | Hi there |
| 05:09.22 | fujin | Really? |
| 05:09.30 | rue_mohr | extensions.conf |
| 05:09.44 | rue_mohr | I created a loop |
| 05:09.54 | fujin | Great! |
| 05:09.58 | fujin | like, a bad loop |
| 05:10.01 | fujin | or a good loop? |
| 05:10.07 | fujin | mmmm. loop. |
| 05:10.13 | rue_mohr | mmm an undesired loop |
| 05:10.23 | fujin | ah. |
| 05:10.23 | rue_mohr | where is pastebin? |
| 05:10.28 | fujin | rafb.net/paste/ |
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| 05:11.07 | fujin | yuck, I'd hate to think what it's like to create loops in extension.conf |
| 05:11.31 | Zipper_32 | Just out of curiosity, what are some of the main benefits I could get from using the AEL? |
| 05:11.46 | Zipper_32 | I've been using large extensions.conf files so far. |
| 05:12.05 | rue_mohr | http://rafb.net/p/N1zQtS42.html |
| 05:12.19 | rue_mohr | its just a piece, but you can see the loop |
| 05:12.37 | rue_mohr | how do I get it to ...clear? the previously dialed digits? |
| 05:13.14 | rue_mohr | to explain, I dial 401, and I hear the dog bark repeatedly |
| 05:13.57 | rue_mohr | anyone? |
| 05:14.48 | Zipper_32 | I'm assuming that dest_dog has the line "exten => s,n,Goto(${Origin},s,1)" at the end of it? |
| 05:14.57 | rue_mohr | its just like dest_duck |
| 05:16.30 | rue_mohr | so it stores the dialed digits, and falls back through the path automatically |
| 05:16.37 | rue_mohr | must be |
| 05:16.58 | Zipper_32 | It tries to go to extension 's' in house, but there is no such extension. |
| 05:17.08 | rue_mohr | there is |
| 05:17.23 | rue_mohr | er, there isn't |
| 05:17.35 | rue_mohr | there is |
| 05:17.47 | Zipper_32 | Am I blind? |
| 05:17.55 | rue_mohr | I'd paste the whole thing if It fit in 1 screen |
| 05:18.27 | Zipper_32 | What does the 's' extension do in 'house'? |
| 05:18.43 | rue_mohr | 1 |
| 05:18.44 | rue_mohr | 2 |
| 05:18.44 | rue_mohr | 3 |
| 05:18.44 | rue_mohr | 4 |
| 05:18.44 | rue_mohr | 5 |
| 05:18.44 | rue_mohr | 6 |
| 05:18.46 | rue_mohr | 7 |
| 05:18.48 | rue_mohr | 8 |
| 05:18.50 | rue_mohr | 9 |
| 05:18.54 | rue_mohr | 10 |
| 05:18.56 | rue_mohr | 11 |
| 05:18.58 | rue_mohr | 12 |
| 05:19.00 | rue_mohr | 13 |
| 05:19.02 | rue_mohr | 14 |
| 05:19.04 | rue_mohr | 15 |
| 05:19.06 | rue_mohr | 16 |
| 05:19.08 | rue_mohr | |
| 05:19.08 | CaT[tm] | oooo. it's just like sesame st. |
| 05:19.10 | rue_mohr | [house] |
| 05:19.12 | rue_mohr | Origin=house |
| 05:19.14 | rue_mohr | include => parkedcalls |
| 05:19.16 | rue_mohr | <PROTECTED> |
| 05:19.18 | rue_mohr | exten => s,1,Answer ; Answer the line |
| 05:19.20 | rue_mohr | exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds |
| 05:19.24 | rue_mohr | exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds |
| 05:19.26 | CaT[tm] | why people don't use paste sites is beyond me. |
| 05:19.26 | rue_mohr | exten => s,n,BackGround(h_inside) ; Play some instructions |
| 05:19.28 | rue_mohr | <PROTECTED> |
| 05:19.29 | AJaymn | STOP!!!!!!!!!!!!!! |
| 05:19.30 | rue_mohr | include => numbers |
| 05:19.32 | rue_mohr | <PROTECTED> |
| 05:19.34 | rue_mohr | ;Outside line |
| 05:19.36 | rue_mohr | exten => _9,1,Dial,Zap/7 |
| 05:19.38 | rue_mohr | <PROTECTED> |
| 05:19.40 | rue_mohr | exten => t,1,Play(demo-thanks) |
| 05:19.42 | rue_mohr | exten => t,n,Hangup |
| 05:19.44 | rue_mohr | damnit |
| 05:19.46 | rue_mohr | http://rafb.net/p/5QhZwh42.html |
| 05:19.48 | rue_mohr | sigh... |
| 05:19.49 | MaliutaWrk | www.pastebin.com is apprently too hard for somepeople to figure out |
| 05:19.50 | rue_mohr | sorry all |
| 05:19.54 | rue_mohr | I did! |
| 05:19.56 | rue_mohr | I missed the url before I pasted it |
| 05:20.02 | rue_mohr | mmm |
| 05:20.04 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 05:20.06 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 05:20.14 | rue_mohr | is it still sending? |
| 05:20.27 | rue_mohr | I missed th url and it pasted the last thing I had |
| 05:20.37 | rue_mohr | http://rafb.net/p/5QhZwh42.html |
| 05:20.42 | rue_mohr | is what you were supposed to get |
| 05:21.52 | rue_mohr | will anyone still talk to me? |
| 05:23.39 | rue_mohr | hmm I dont know if i been silenced or not |
| 05:29.57 | rue_mohr | drat, just like me to break it |
| 05:32.43 | flenders | can I have 2 sip users defined on sip.conf with different codecs? |
| 05:32.47 | flenders | for example [pennytel-g711] and [pennytel-g729] with the same username and passwords, and on the dialplan I would just do SIP/${EXTEN}@pennytel-g729 or g711? |
| 05:33.01 | flenders | I would already be registered with pennytel anyway on the register line |
| 05:33.56 | *** join/#asterisk rpm (n=russell@66.183.28.233) |
| 05:35.14 | fujin | you'd probably want to do it with two usernames |
| 05:35.28 | fujin | and then have two contexts for dialing them |
| 05:35.56 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:36.18 | rue_mohr | fujin, .. you wouldn't happent to be able to hear me would you? |
| 05:37.01 | fujin | nope, sorry, I can't hear you. |
| 05:37.09 | rue_mohr | heh, thanks |
| 05:37.14 | jarod14 | hi guys |
| 05:37.23 | fujin | rue_mohr: you |
| 05:37.27 | fujin | you're doing it wrogn! |
| 05:37.29 | rue_mohr | been banned for less than that |
| 05:37.43 | rue_mohr | http://rafb.net/p/Ji8I6f23.html |
| 05:38.07 | rue_mohr | please advise |
| 05:38.24 | rue_mohr | what is the right way |
| 05:39.51 | jarod14 | rue_mohr, there is a mistake on line 114 |
| 05:40.05 | fujin | right, I'm out |
| 05:40.07 | fujin | seeyas. |
| 05:41.17 | rue_mohr | heh, so there is |
| 05:41.28 | rue_mohr | thats not causeing the loop though |
| 05:42.38 | jarod14 | I was not yet connected, when you explained your problem. I just quickly read your pastebin and saw line 114 |
| 05:42.59 | rue_mohr | oh, from house, follow it for 401 |
| 05:43.01 | rue_mohr | it loops |
| 05:45.02 | rue_mohr | transfers dont work, but I wont go there now |
| 05:45.37 | jarod14 | is the syntax dial followed by a ',' correct ? |
| 05:46.04 | rue_mohr | the problem is that it loops though |
| 05:46.12 | rue_mohr | it dials the extenstions ok |
| 05:46.37 | jarod14 | have you tried since correcting line 114 ? |
| 05:46.42 | rue_mohr | but it stores the digits, so after it gets done with duck quack and goes back to house, it falls through agsin without anyone dialing anything |
| 05:46.48 | rue_mohr | ok |
| 05:46.58 | rue_mohr | it still loops |
| 05:47.19 | rue_mohr | it loops because the digits are never consumed |
| 05:47.35 | rue_mohr | when it gets back to house, its still seeing that 401 was dialed |
| 05:47.54 | rue_mohr | I see and understand the problem, I just dont understand how to fix it |
| 05:49.00 | jarod14 | ion numbers section ,why did you use pattern like _401 and not simply 401 ? |
| 05:49.19 | rue_mohr | thats what your supposed to do isn't it? |
| 05:49.27 | jarod14 | yes |
| 05:49.39 | rue_mohr | erm |
| 05:49.42 | rue_mohr | so, its right? |
| 05:49.59 | jarod14 | I think pattern are wrong in your case |
| 05:50.16 | rue_mohr | hmm, whats the _ do? |
| 05:50.18 | jarod14 | pattern are good when you don't know the whole extension |
| 05:50.32 | rue_mohr | oh |
| 05:50.41 | rue_mohr | if I wanted previously dialed digits |
| 05:50.58 | rue_mohr | so you think I drop the _ and I'll be ok... lets try |
| 05:51.05 | jarod14 | fkor instance if exten 4010 to 4019 can de dialed and go to same thing, use pattern _401X |
| 05:52.00 | jarod14 | I meant 4010 through 4019, sorry for my english ;o( |
| 05:52.08 | rue_mohr | thats ok |
| 05:52.26 | rue_mohr | it still loops |
| 05:52.39 | jarod14 | so it still sucks ! |
| 05:53.44 | *** part/#asterisk dseeb_ (n=dcb@CPE-124-179-234-118.vic.bigpond.net.au) |
| 05:54.32 | rue_mohr | well, atleast you CAN dial the dog from inside now |
| 05:55.24 | rue_mohr | a tiny bit of progress on an otherwise completely pointless day |
| 05:55.35 | rue_mohr | :) |
| 05:59.17 | jarod14 | don't know how to clear/reset the extension memorized |
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| 06:05.14 | rue_mohr | :) |
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| 06:11.19 | katsmeow-afk | rue said his flood was an accident |
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| 06:16.45 | tengulre | hi,all |
| 06:17.23 | JT | what are these people coming out of the woodwork to speak up for rue_mohr? :P |
| 06:20.33 | *** join/#asterisk IguanaNed (n=IguanaNe@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 06:29.13 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:32.30 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 06:33.44 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 06:33.45 | RsaMan | hello |
| 06:33.59 | RsaMan | does anyone know how to recover a password for asterisk gui |
| 06:34.14 | creadurex | morning JT :) |
| 06:34.28 | JT | hi |
| 06:36.49 | tzafrir | RsaMan, view or edit /etc/asterisk/manager.conf or /etc/asterisk/users.conf , I guess |
| 06:37.12 | RsaMan | got it |
| 06:37.13 | RsaMan | thanks |
| 06:44.17 | RsaMan | how does one remove a asteriskgui |
| 06:44.49 | tzafrir | webenable=no |
| 06:44.59 | tzafrir | That preactically disables it. |
| 06:45.10 | tzafrir | Then remove the javascript files |
| 06:45.21 | tzafrir | webenable=no; in manages.conf |
| 06:45.38 | tzafrir | and disable http.conf |
| 06:46.09 | tzafrir | As for users.conf - this is something you may like or like independently |
| 06:46.17 | creadurex | JT: i was wondering if you could perhaps answer a q about isdn/gt and analog lines.. im wondering how our old ass ericsson pbx can "emulate" analog lines over our gt's? |
| 06:46.39 | tzafrir | like or dislike independently, I mean |
| 06:46.55 | JT | gt? |
| 06:48.27 | creadurex | JT: thats what my phone guy called em |
| 06:48.27 | creadurex | heh |
| 06:48.31 | creadurex | im all-ip, this is new to me |
| 06:48.52 | JT | ground start, gs? |
| 06:49.18 | creadurex | he said each box had 2 lines |
| 06:49.21 | creadurex | we have 4 of em |
| 06:49.40 | *** join/#asterisk twistedolive (n=twistedo@67.133.226.68) |
| 06:51.08 | JT | yeah not really sure what you're talking about |
| 06:51.17 | creadurex | hehe ok, then we're two |
| 06:51.17 | creadurex | :) |
| 06:51.21 | *** join/#asterisk Error_X (n=konkyl@43.81-167-237.customer.lyse.net) |
| 06:51.26 | JT | isdn2/bri? |
| 06:51.29 | citats | each box having 2 lines could be a bri |
| 06:51.35 | WilliamK | morning JT |
| 06:51.50 | citats | not sure where the gt comes from though |
| 06:52.02 | creadurex | probably the norwegian notation |
| 06:53.05 | twistedolive | Hi... Does anyone know if zaptel-1.4.4 and libpri-1.4.1 can be installed on Mac OS X 10.4? I got errors during installation... |
| 06:53.40 | JT | hello WilliamK |
| 06:53.44 | JT | twistedolive: no |
| 06:53.57 | creadurex | JT: could probably mean ground start when i think of it |
| 06:54.15 | JT | creadurex: is it digital? |
| 06:54.22 | creadurex | how can i tell JT? |
| 06:54.53 | tzafrir | twistedolive, I know that there was an old port attempt of Zaptel for OSX, but I haven't heard from it for a long time |
| 06:55.09 | tzafrir | twistedolive, zaptel in Digium is for Linux alone, AFAIK |
| 06:55.12 | citats | googling for isdn gt does produce some hits, though its not completely obvious what it is |
| 06:55.27 | tzafrir | libpri should work for other platforms. No idea about OSX |
| 06:56.18 | tzafrir | gs? some kind of analog signalling, AFAIK |
| 06:56.35 | tzafrir | groundstart, not ghostscript or grandstream |
| 06:56.35 | creadurex | citats: "Telenor ISDN GT has a capacity of two 64kbit b channels and one 16kbit d channel" |
| 06:57.07 | citats | creadurex: ahhh sounds is the same thing as an ISDN BRI here in the US |
| 06:58.02 | citats | creadurex: you have boxes that plug into these lines and then break out 2 phone lines or these lines plug directly into your existing pbx? |
| 06:58.05 | creadurex | damn norwegians having to translate everything :) |
| 06:59.36 | creadurex | i have 4 boxes that the lines go into, then its spaghetti hell out of them, either 1 or two lines into the pbx from each bri box |
| 07:01.15 | citats | creadurex: so you've got something like 7-8 phone lines? |
| 07:02.23 | creadurex | according to the phone guy we can have max 8 simultaneous calls |
| 07:02.45 | JT | creadurex: 4 BRIs |
| 07:02.49 | citats | creadurex: so it sounds like your ericsson pbx has 8 FXO ports on it |
| 07:02.52 | JT | also known as ISDN2 |
| 07:03.08 | JT | citats: doubt it, that would be a stupid setup |
| 07:03.14 | JT | nevertheless, a possible setup |
| 07:03.17 | creadurex | hehe |
| 07:03.36 | citats | JT: would be stupid, but a spaghetti hell from 4 boxes into a pbx sounds like that |
| 07:03.39 | creadurex | indeed |
| 07:03.53 | JT | creadurex: the ports connected to the pbx, are they marked with anything? |
| 07:04.30 | creadurex | i tried getting an overview back there heh, but it seems one of the bri's is connected in serial with the other |
| 07:04.49 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:04.50 | JT | i doubt it |
| 07:04.53 | creadurex | i would assume each bri should have 1 isdn line in, then 2 phone lines out into the ericsson (the stupid setup) |
| 07:05.01 | creadurex | let me go take a look :) |
| 07:05.34 | JT | i smart setup would be an isdn line in (U-line) and an RJ-45 S/T-bus connection to the pabx |
| 07:05.40 | JT | s/i/a/ |
| 07:06.00 | JT | also, 8P8C, not RJ-45, to be correct :) |
| 07:07.20 | creadurex | not marked with anything |
| 07:07.39 | JT | well what connector goes to the pabx? |
| 07:07.49 | creadurex | there is a wall socket marked with u=<num> and l=<num> |
| 07:08.09 | creadurex | which goes to a box marked NT1 |
| 07:08.24 | JT | i don't care about the line to the telco |
| 07:08.32 | JT | i care about the line from the NT1 to the pabx |
| 07:08.46 | JT | the line to the telco will be single paid |
| 07:08.48 | JT | pair |
| 07:09.35 | creadurex | yes thats the one i was tracing now, the isdn line from nt1 -> pabx is connected to one of the boards inside the pabx with 2 pairs |
| 07:10.04 | JT | it's punched down, or modular socket? |
| 07:10.58 | creadurex | modular socket it seems |
| 07:12.05 | JT | creadurex: same type as for ethernet? |
| 07:12.49 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 07:12.50 | creadurex | no, it looks like a generic 4 lead connector |
| 07:13.19 | asterisknerds | <PROTECTED> |
| 07:14.59 | JT | creadurex: weird, is there a spare RJ-45 connector or two on the NT1? |
| 07:15.13 | JT | and is only a single connector used from NT1 to pabx? |
| 07:16.21 | creadurex | let me check |
| 07:16.33 | *** join/#asterisk trustinfo-tb (n=trust@AStrasbourg-156-1-80-175.w86-204.abo.wanadoo.fr) |
| 07:17.22 | trustinfo-tb | hello |
| 07:17.49 | trustinfo-tb | i've a probleme witch module chan_misdn.so in asterisk 1.4 |
| 07:18.42 | creadurex | hehe!! ok.. good idea to sit right next to the window with my hands inside the pabx when lightning strikes outside |
| 07:19.41 | JT | sure |
| 07:20.32 | creadurex | JT: the nt1 has 3 rj45 connectors, one marked line and two marked isdn. only one is in use |
| 07:20.53 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 07:20.55 | JT | and what is it marked, pray tell? :P |
| 07:21.11 | trustinfo-tb | when i compile asterisk to have the misdn module all other module are compiled but not misdn |
| 07:21.19 | trustinfo-tb | have you any idea??* |
| 07:21.37 | *** part/#asterisk twistedolive (n=twistedo@67.133.226.68) |
| 07:22.53 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 07:24.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:26.21 | creadurex | JT: who needs marked cables? it is not marked any thing.. in either end :p |
| 07:26.40 | JT | creadurex: you just told me all ports are marked, and the cable is plugged into a port. |
| 07:26.46 | JT | what is the marking? |
| 07:27.49 | creadurex | then let me rephrase to avoid confusion :) |
| 07:30.23 | creadurex | the only things marked are the bris and the nt1 |
| 07:30.35 | creadurex | none of the cables or insides of the pabx is marked with anything |
| 07:30.46 | JT | yeah, who cares about that stuff |
| 07:31.02 | JT | all i care about is the same of the port on the NT1 that the cable to the pabx is marked |
| 07:31.06 | JT | is it really that hard? |
| 07:31.44 | JT | s/same/name/ |
| 07:33.27 | JT | earth to creadurex |
| 07:35.07 | RsaMan | hello guys |
| 07:35.27 | *** join/#asterisk mightnare (n=mike@202.164.181.222) |
| 07:35.33 | RsaMan | is it only the extensions.conf file that makes rules for answering inbound calls ? |
| 07:35.49 | RsaMan | my asterisk keeps answering the land line |
| 07:36.11 | mightnare | soft phones usually have a hold button, is there a way to simulate this on the dialplan? |
| 07:36.31 | JT | RsaMan: what did you expesct it to do? |
| 07:36.43 | RsaMan | JT: I want it to do nothing |
| 07:37.06 | JT | well that's strange, but is possible in theory |
| 07:37.06 | RsaMan | i made the mistake of configuring my asterisk server after make samples |
| 07:37.14 | creadurex | lol JT, it shouldnt be this hard no |
| 07:37.41 | RsaMan | i have an spa400 conected to my landline and that is connected to my asterisk server |
| 07:37.52 | RsaMan | but the spa400 keeps answering |
| 07:38.15 | JT | creadurex: kernel jessup, Did you observe the cable to be plugged into the port marked ISDN?!" |
| 07:38.23 | creadurex | the name of the port on the nt1 where the cable to the pabx is marked: hold your breath.......: "isdn" |
| 07:38.41 | JT | RsaMan: no experience with the spa400, but sounds like what it is designed to do |
| 07:38.47 | creadurex | YES jt! 10 points to me |
| 07:38.48 | creadurex | or you |
| 07:38.59 | JT | creadurex: congrats, it took you all that time to confirm what i already knew :P |
| 07:39.03 | creadurex | the cable is indeed safely plugged into the port marked "ISDN". hehe |
| 07:39.05 | mightnare | i was thinking it could be done using features and call MusicOnHold on the channel, but how do i bridge the channels back? |
| 07:39.11 | JT | you use BRI direct to the pabx, zomg |
| 07:39.18 | creadurex | JT: i did take some calls inbetween thank you :p |
| 07:39.48 | creadurex | kazing |
| 07:39.53 | *** join/#asterisk oej (n=olle@81-224-166-188-o1036.telia.com) |
| 07:39.54 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:39.55 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 07:40.27 | JT | a BRI has 2 * 64kbit/s B channels and 1 16kbit/s D channel |
| 07:40.49 | JT | the S-bus uses 4 wires, as it is differential serial transmission |
| 07:40.54 | JT | a tx pair and an rx pair |
| 07:41.51 | RsaMan | can someone tell me which are the essential files ? http://pastebin.ca/656599, they were all installed when i used make samples |
| 07:41.54 | *** part/#asterisk Error_X (n=konkyl@43.81-167-237.customer.lyse.net) |
| 07:41.54 | *** join/#asterisk Error_X (n=konkyl@43.81-167-237.customer.lyse.net) |
| 07:41.59 | RsaMan | i just want to use sip at this moment |
| 07:42.14 | RsaMan | when i remove all except extensions.conf an sip.conf |
| 07:42.18 | RsaMan | it no longer works |
| 07:43.06 | RsaMan | i know its a bad question |
| 07:43.17 | kaldemar | removing configuration files is not the right way to not use modules. |
| 07:43.42 | RsaMan | should i just clear the contents? |
| 07:43.50 | JT | RsaMan: then don't plug in the phone line, duh |
| 07:43.58 | RsaMan | i am not sure what to do, which ones are need for sip |
| 07:44.00 | kaldemar | put back your configs and disable the modules you don't want to use in modules.conf. |
| 07:44.09 | RsaMan | JT : :) i need to make outbound calls over the line |
| 07:44.15 | creadurex | i wnder if our UPS is gonna come into play today |
| 07:44.18 | creadurex | damn lightning storm |
| 07:44.21 | JT | RsaMan: news flash, extensions.conf bares no relationship to when your SPA-400 decides to answer the line |
| 07:44.22 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 07:44.33 | RsaMan | JT : kk |
| 07:44.49 | JT | asterisk does not answer calls on an ATA. |
| 07:45.06 | JT | the rule of thumb is the device that terminates a connection answers it |
| 07:45.17 | RsaMan | kk |
| 07:45.22 | RsaMan | i will look into it |
| 07:45.26 | JT | kkkkk |
| 07:45.48 | RsaMan | as for my sip question ? will sip.conf and extensions.conf do the trick ? |
| 07:46.11 | JT | you will need to configure them |
| 07:46.14 | RsaMan | yes |
| 07:46.39 | RsaMan | but when i delete all files except them , my sip no longer works |
| 07:46.48 | RsaMan | all .conf files |
| 07:46.52 | JT | then don't randomly delete files |
| 07:46.57 | RsaMan | is there another dependant file? |
| 07:47.01 | JT | perhaps you should look into why it fails |
| 07:47.05 | JT | read the cli output |
| 07:47.09 | RsaMan | i want to know which ones are essential |
| 07:47.11 | RsaMan | kk |
| 07:47.21 | JT | what's wrong with "ok", seriously |
| 07:47.28 | kaldemar | RsaMan: you asterisk does not work if you delete files like that. take a look at my previous comment. you have to decide for yourself what modules are essential. |
| 07:47.36 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 07:52.04 | Error_X | why is the sound so fuzzy? |
| 07:53.02 | JT | Error_X: certainly giving us a lot of information to work on there |
| 07:53.36 | RsaMan | lol |
| 07:53.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:53.40 | RsaMan | worse than me |
| 07:55.00 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 07:55.01 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:55.23 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 07:55.53 | asterisknerds | <PROTECTED> |
| 07:56.24 | kaldemar | just keeps getting better. |
| 07:57.30 | *** join/#asterisk nighty^ (n=nighty@ed39.AFL49.vectant.ne.jp) |
| 07:59.12 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 07:59.25 | jmls | morning. I've gotten a PRI error here |
| 07:59.30 | jmls | NOTICE[32422]: chan_zap.c:8466 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 |
| 07:59.39 | jmls | is this a "my issue" or a "my telco" issue ? |
| 08:00.23 | jmls | followed by |
| 08:00.25 | jmls | [Aug 14 08:58:36] NOTICE[32422]: chan_zap.c:8466 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 |
| 08:01.21 | *** join/#asterisk saftsack (n=oliver@p54A7D152.dip.t-dialin.net) |
| 08:07.25 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:07.35 | JT | jmls: is it a pri? |
| 08:09.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:12.02 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262) |
| 08:14.04 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4663747eade0bbeb) |
| 08:19.44 | Error_X | JT: When Im trying to call the voicemail or a voicemenu, its choppy aaaaand slooooooow |
| 08:20.19 | Error_X | I've tried both iax and sip |
| 08:28.56 | Uatec | Error_X, where are you dialing from? what's the route between your sip/iax device and your asterisk box? |
| 08:30.07 | flenders | Error_X: you're not on a virtual machine, are you? |
| 08:30.12 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 08:31.06 | *** join/#asterisk saftsack (n=oliver@p54A7E969.dip.t-dialin.net) |
| 08:31.24 | Uatec | i get choppy and sloooow and laggy when i call from my sip client on my mobile across gprs |
| 08:40.27 | jeremy_g | are following different |
| 08:40.28 | jeremy_g | exten=1234,1,Hangup(); |
| 08:40.33 | jeremy_g | exten=>1234,1,Hangup(); |
| 08:40.36 | jeremy_g | exten= 1234,1,Hangup(); |
| 08:45.17 | RsaMan | hi |
| 08:45.23 | RsaMan | is anyone here using an spa400 |
| 08:48.54 | RsaMan | when i call the someone via the spa400 using a sip client, the person on the other end cannot here me |
| 08:49.01 | RsaMan | but i can hear them |
| 08:51.19 | *** join/#asterisk saftsack (n=oliver@p54A7CF20.dip.t-dialin.net) |
| 08:53.54 | RsaMan | does anyone know what dtmfmode= ? |
| 08:53.54 | RsaMan | is |
| 08:53.56 | jeremy_g | RsaMan:u have a great sense of hearing, it seems |
| 08:54.02 | jeremy_g | :D |
| 08:54.15 | jeremy_g | dtmfmode shud be set to info when talking to girls |
| 08:54.22 | jeremy_g | for real men, it should be set to inband |
| 08:54.38 | jeremy_g | if you are unsure about the sex, set it to rfc.... |
| 08:55.15 | RsaMan | ok |
| 08:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:55.43 | jeremy_g | RsaMan: :D |
| 08:56.05 | asterisknerds | <PROTECTED> |
| 09:01.56 | RsaMan | :( |
| 09:02.48 | RsaMan | this might be a really stupid question, but here goes |
| 09:02.58 | RsaMan | can a sip client call an iax client ? |
| 09:03.07 | *** join/#asterisk suvir (n=chatzill@ppp-124.120.237.220.revip2.asianet.co.th) |
| 09:03.08 | RsaMan | and the other way |
| 09:05.59 | creadurex | if asterisk stands between them |
| 09:08.02 | RsaMan | where do i start with setting up asterisk for an iax server |
| 09:08.06 | RsaMan | is it similar |
| 09:08.08 | RsaMan | to sip |
| 09:08.12 | RsaMan | setting up sip ? |
| 09:10.18 | tengulre | how to cancel IAX2 channel echo? |
| 09:15.17 | *** join/#asterisk saftsack (n=oliver@p54A7FAD7.dip.t-dialin.net) |
| 09:18.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:19.07 | jeremy_g | RsaMan:do you note the difference when you change the dtmfmode to match the voice sex |
| 09:20.30 | RsaMan | :( |
| 09:20.52 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 09:23.08 | RsaMan | i want to set up asterisk as an IAX server, much like i have set it up as a SIP server |
| 09:23.10 | RsaMan | but most references i find want to connect to an existing iax server |
| 09:23.12 | RsaMan | .. |
| 09:23.18 | RsaMan | do i have the wrong end of the stick ? |
| 09:26.43 | jeremy_g | RsaMan:no, you probably have the whole stick up urz |
| 09:27.05 | jeremy_g | RsaMan:listen, |
| 09:27.26 | *** join/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-100-71.w90-24.abo.wanadoo.fr) |
| 09:27.37 | jeremy_g | 1. two sip entities got to be using same dtmf mode in order to talk |
| 09:28.02 | jeremy_g | e.g. if one is inband the other should be inband as well. inband refers to raw audio |
| 09:28.24 | jeremy_g | info means the sip info format that usually some enterprise grade pbxes use by default. |
| 09:28.40 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.16.69.revip2.asianet.co.th) |
| 09:28.40 | jeremy_g | 2. To create the iax accounts, you need to see the iax.conf |
| 09:28.46 | jeremy_g | and read ~thebook |
| 09:28.50 | jeremy_g | ~thebook |
| 09:28.58 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:29.15 | HaMYaI | Hi, how do we interrupt cmd MusicOnHold? |
| 09:29.48 | RsaMan | thanks |
| 09:30.04 | LoneShadow | Anyone used a bluetooth headset with asterisk ? |
| 09:30.11 | HaMYaI | apart from hanging up |
| 09:30.29 | tengulre | hwo to cancel the iax echo??? |
| 09:30.33 | tengulre | anybody help !! |
| 09:31.08 | LoneShadow | tengulre: are you using any ATA ? |
| 09:32.49 | LoneShadow | tengulre: check http://www.voip-info.org/wiki/view/Asterisk+echo+avoidance |
| 09:33.04 | LoneShadow | im off to bed |
| 09:33.43 | jeremy_g | HaMYaI:use it intelligently |
| 09:35.03 | HaMYaI | jeremy_g: ok, thanks for a great answer |
| 09:35.20 | HaMYaI | jeremy_g: that solved my problem |
| 09:36.19 | jeremy_g | tengulre:read page 73 for thebook |
| 09:36.40 | jeremy_g | HaMYaI: it all owes to the greatness in the question itself. |
| 09:36.57 | jeremy_g | from |
| 09:39.53 | Uatec | Is there a simpler voicemail app? |
| 09:40.03 | Uatec | there are so many options in the default one that i don't need |
| 09:40.09 | Uatec | and some of my users aren't too bright |
| 09:42.50 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 09:43.24 | bertrand^ | Uatec, you mean to read your voicemails? |
| 09:43.40 | bertrand^ | Uatec, i send them by email, some of mine aren't too bright too |
| 09:43.52 | Uatec | yeah, i do that too |
| 09:44.07 | Uatec | but i would like a simpler voicemail interface |
| 09:44.12 | Uatec | i'm going to have to write one, aren't i |
| 09:44.13 | Uatec | *sigh* |
| 09:44.17 | bertrand^ | like a web-based? |
| 09:44.49 | Uatec | no |
| 09:45.16 | Uatec | just Previous, Next, Delete, record messages |
| 09:55.12 | jmls | JT: yes |
| 09:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 09:55.40 | Uatec | i'm really annoyed |
| 09:55.58 | Uatec | i can establish a sip connection to my ISPs asterisk server |
| 09:56.02 | Uatec | from mine |
| 09:56.03 | jmls | "we notice that there seemed to be a fault between 6:30 and 9:00. However, since we have been monitoring the line there have been no further problems" |
| 09:56.06 | Uatec | but not from mine, to my other |
| 09:56.13 | Uatec | my other asterisk box denies access |
| 09:56.15 | Uatec | :'( |
| 10:09.09 | RsaMan | chan_iax2.c:6988 socket_process: Rejected connect attempt from 192.168.0.22, who was trying to reach '101@' |
| 10:09.18 | RsaMan | does this mean it is not picking up a context? |
| 10:09.23 | RsaMan | 101@ ????/ |
| 10:16.32 | *** join/#asterisk Tili (n=tili@181.17.221.87.dynamic.jazztel.es) |
| 10:24.45 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 10:26.07 | Uatec | that's what i'm getting |
| 10:26.27 | Uatec | what's your Dial() look like on the other box? |
| 10:27.19 | creadurex | oh goodie, the polycom 430 has arrived |
| 10:29.22 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 10:32.08 | *** join/#asterisk KDan (n=KDan@wakiki.gotadsl.co.uk) |
| 10:37.19 | waptaxi | hi! Has anyone an experience in connection Panasonic TDA 200 with Asterisk using E1? Especially interested in making 2 B-Channels Transfer work. |
| 10:37.52 | JT | 2BCT, ambitious |
| 10:39.23 | *** part/#asterisk HaMYaI (n=LAMER@ppp-58.8.16.69.revip2.asianet.co.th) |
| 10:40.09 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 10:42.07 | waptaxi | I'm using libpri from SVN, and there are some code regarding to 2BCT, but not working with TDA200 |
| 10:49.38 | *** join/#asterisk mtryfoss (n=root@85.19.26.76) |
| 10:51.02 | waptaxi | I found this thread http://bugs.digium.com/view.php?id=7778, this is exactly what I need, but seems to work only with Avaya or maybe there's some special settings at Panasoniic side? |
| 10:51.50 | mtryfoss | I experience random disconnects of calls, with debug message "didn't get a frame from channel". The networks is fine, and the servers is way over-dimensioned. Any tips? |
| 10:53.34 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-107-182-30.ny325.east.verizon.net) |
| 10:55.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:55.58 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 10:56.28 | asterisknerds | <PROTECTED> |
| 10:57.10 | jeremy_g | my asterisk realtime wont just read read the extensions. :( |
| 10:57.32 | jeremy_g | all my sip devices are happily registered with * but extensions wont get dialled. :( |
| 10:57.36 | jeremy_g | um running ara |
| 11:08.43 | *** join/#asterisk _Krieger_ (n=warsword@91.102.176.47) |
| 11:11.57 | *** join/#asterisk zerohalo (n=zeroHalo@h-74-2-90-66.cmbrmaor.covad.net) |
| 11:13.22 | Tili | i wonder where I can get FibreNetwork MAP of whole world to select best countries for putting data/network sites |
| 11:14.28 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 11:20.27 | Uatec | dunno |
| 11:20.34 | Uatec | but if you find out i would be itnerested to see |
| 11:20.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 11:27.34 | RsaMan | hi attemping to get a hfc-s to work with asterisk |
| 11:27.44 | RsaMan | when i compile the zaphfc drivers ,,,, |
| 11:28.01 | RsaMan | i get the following error Link /usr/src/linux-2.6 to your kernel sources first! |
| 11:28.20 | RsaMan | i have run ln -s linux-2.6.6 linux-2.6 |
| 11:28.29 | RsaMan | in my /usr/src folder |
| 11:28.34 | RsaMan | but it still gives me this error |
| 11:39.13 | tzafrir | RsaMan, no, |
| 11:39.22 | tzafrir | install proper kernel sources |
| 11:39.32 | tzafrir | make sure that the following is valid: |
| 11:39.46 | tzafrir | /lib/modules/`uname -r`/build/.config |
| 11:39.52 | tzafrir | Does that file exist? |
| 11:40.07 | tzafrir | 'build' is usually a symlink to the right place |
| 11:40.27 | tzafrir | either if you build your own if one from your distro |
| 11:40.38 | tzafrir | which distro is it? |
| 11:41.12 | tzafrir | RsaMan, check http://updates.xorcom.com/astribank/bristuff/INSTALL.html |
| 11:41.27 | tzafrir | or: http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html |
| 11:41.39 | tzafrir | if you prefer the bristuff 0.4.0 |
| 11:42.05 | RsaMan | i am using fedora core |
| 11:42.34 | tzafrir | I suspect that the ./prereq.sh script will work there. But I'm not sure |
| 11:42.49 | RsaMan | kk |
| 11:42.52 | RsaMan | i will check it out |
| 11:42.52 | RsaMan | thanks |
| 11:43.25 | tzafrir | in any case, './prereq.sh test' should give you some ideas on what to do |
| 11:43.33 | RsaMan | where is that script located ? |
| 11:43.38 | tzafrir | Please report bugs |
| 11:43.42 | tzafrir | in the tarball |
| 11:44.00 | tzafrir | You can find the content of the tarball under bristuff-current/ |
| 11:44.15 | tzafrir | (INSTALL.html is actually a symlink into it) |
| 11:48.59 | RsaMan | am i following the correct route to configure a hsf-c card to work with asterisk |
| 11:49.01 | RsaMan | ? |
| 11:50.51 | tzafrir | RsaMan, you can also try our live CD from http://updates.xorcom.com/iso/live-1.0.2.iso for sample configuration. That is still 1.2, though |
| 11:51.08 | tzafrir | One other possible route is chan_misdn, which I don't know well enough |
| 11:51.17 | tzafrir | BTW: what version of Fedora do you use? |
| 11:51.22 | RsaMan | core 6 |
| 11:51.36 | tzafrir | kernel 2.6.6?? |
| 11:51.42 | *** join/#asterisk lirakis (n=eric@69.24.142.1) |
| 11:52.25 | RsaMan | 2.6.21-1.3194.fc7 |
| 11:52.25 | engrxyz | hi, how to do a sip trace in *? |
| 11:53.18 | kaldemar | engrxyz: sip debug [ip|peer] in cli |
| 11:55.02 | engrxyz | thx got it. kaldemar, u familiar with sofia-sip? |
| 11:55.37 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 11:55.56 | kaldemar | engrxyz: no, sorry. |
| 11:56.07 | Uatec | is that a person? |
| 11:56.17 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:56.18 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:56.22 | JT | is what a person? |
| 11:56.23 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:58.33 | JT | Uatec: ? |
| 11:58.55 | Uatec | sofia-sip |
| 11:59.42 | HarryR | nah, it's nokias open-source sip stack |
| 11:59.51 | HarryR | very nice to work with aparently |
| 11:59.58 | JT | freeswitch uses sofia-sip |
| 12:00.04 | Uatec | oh cool |
| 12:00.17 | HarryR | yah, before they were using exosip, which is nearly as bad as asterisk's sip stack :\ |
| 12:00.30 | Uatec | lol |
| 12:00.45 | Uatec | can you easily change the sip stack asterisk uses?? |
| 12:00.52 | HarryR | no |
| 12:01.29 | Uatec | oh |
| 12:01.31 | Uatec | LAME |
| 12:01.46 | *** join/#asterisk oej (n=olle@static-195.84.115.62.addr.tdcsong.se) |
| 12:08.59 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 12:09.03 | RsaMan | i think i made a huge mistake |
| 12:09.43 | RsaMan | i ran the incorrect bristuff script |
| 12:09.44 | RsaMan | for 1.2 |
| 12:09.47 | RsaMan | and not 1.4 |
| 12:09.48 | RsaMan | :( |
| 12:09.51 | RsaMan | oh dear |
| 12:12.14 | RsaMan | how bad is that ? |
| 12:12.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:12.35 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.47 | *** join/#asterisk yarrix (n=yarro@88.235.32.7) |
| 12:12.50 | yarrix | hello |
| 12:13.14 | RsaMan | hi |
| 12:13.48 | engrxyz | anyone experience here with sofia-sip |
| 12:13.51 | yarrix | first time here, can I just shoot away with a technical question? |
| 12:14.01 | [TK]D-Fender | ~ask |
| 12:14.02 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:14.48 | engrxyz | registering a sip UA in * is easier compared to nokia's sofia-sip |
| 12:15.38 | Mavvie | I have some weird SIP re-invite problem with a Cisco call manager. |
| 12:16.12 | yarrix | i'm located in istanbul, turkey, and am having hangup problems with a digium TDM400 card with 2 fxo, 2 fxs modules. I found countr indications for Turkey, and recompiled zaptel with them (zonedata.c), but still when a PSTN caller hangs up, asterisk doesn't recognize the hangup. |
| 12:16.20 | *** join/#asterisk hot_wheelz (i=hotwheel@124-168-132-199.dyn.iinet.net.au) |
| 12:16.21 | [TK]D-Fender | lol just learned taht gntoo is an actual breed of penguin.... and heavily munched upon by LEOPARD seals (Mac > Gentoo) heh |
| 12:16.54 | tzafrir | yarrix, there are several ways of detecting hangups |
| 12:17.02 | [TK]D-Fender | yarrix: Ask your telco to provide CDS (Call Disconnect Supervision) |
| 12:17.27 | *** part/#asterisk hot_wheelz (i=hotwheel@124-168-132-199.dyn.iinet.net.au) |
| 12:17.29 | RsaMan | does anyone here have a zaphfc card? |
| 12:17.36 | tzafrir | detecting the busy tone is generally the one you use as a last resort. And the tone you set in the tonezones is actually not used there |
| 12:17.46 | yarrix | tzafrir: i've tried the busydetect, and polarity settings. Turk Telekom, our telco, doesnt provide the supervised service. |
| 12:18.03 | RsaMan | or ever worked with asterisk and an hfc card |
| 12:18.04 | tzafrir | yarrix, tried ks? |
| 12:18.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:21.39 | tzafrir | yarrix, you should play with busypattern in zapata.conf maybe. But first make sure you have busydetect=yes |
| 12:22.58 | tzafrir | [TK]D-Fender, now you should have known that gentoo is simply a a fully GUI-configurable, two-pane X file manager: http://packages.debian.org/gentoo |
| 12:24.22 | *** join/#asterisk adker (n=chatzill@74-33-216-174.br1.glv.ny.frontiernet.net) |
| 12:24.33 | [TK]D-Fender | tzafrir : From what I've seen, with a Leopard Seal, a Gentoo only gets ONE pain before the end ;) |
| 12:24.36 | yarrix | tzafrir: yes, running ks now |
| 12:26.26 | yarrix | tzafrir: the sound I hear, as evidenced by 3-4 minute long voicemails messages, is three short beeps, followed by 1 long beep, and this pattern repeats. where can get more info about busypattern? this pattern is different from a normal busy, btw. |
| 12:27.23 | tzafrir | it's the (frequencies of the) tones of the busy tone |
| 12:29.26 | yarrix | ok, but that doesn't change the normal busy detection, i presume? |
| 12:29.41 | Uatec | hey, does anyone know where i can get good listings of UK dial tones? specifically the outgoing ring tone? |
| 12:29.45 | Uatec | i've found a few sites |
| 12:29.48 | Uatec | but ethey're crap |
| 12:29.49 | yarrix | you're right the frequency of the tone sounds about the same as the busy tone. |
| 12:30.24 | yarrix | tzafrir: i did the stuff from here: http://www.voip-info.org/wiki/view/Asterisk+indications+Turkey |
| 12:30.58 | yarrix | recompiled zaptel, reloaded, and got the same results. |
| 12:31.43 | yarrix | i talked to the digium vendor here where we bought the tdm400 card from, and they said that there is nothing that can be done except writing your own software to detect the busy pattern |
| 12:32.34 | yarrix | turns out they are selling their 'own' asterisk pbx for about 3000 usd, which 'works' on the pSTN here, and it too uses the same card. but they won't share the code, they just sell the box. |
| 12:34.37 | Mavvie | In a SIP packet, CSeq should only increase shouldn't it? |
| 12:34.37 | HarryR | Mavvie, yes |
| 12:34.37 | yarrix | and I guess that is why no-one is using asterisk/trixbox, as it is available from the regular places, instead these 'vendors' are forcing people to buy their solution instead, which is actually almost identical. i think what they're doing is a kind of theft anyway.. |
| 12:35.04 | Mavvie | HarryR: so if I see number 102, and then 103, I shouldn't see any 102's anymore? |
| 12:35.06 | yarrix | i'm talking about my locality of course. |
| 12:35.22 | HarryR | Mavvie, never again, until you start a new sip transaction |
| 12:36.03 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 12:36.46 | engrxyz | HarryR: have u tested sofia-sip with sip uas? |
| 12:36.57 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:37.08 | puzzled | hi |
| 12:37.08 | HarryR | i've tested it with a snom 200 and the voiptalk softphone, both work fine |
| 12:37.47 | engrxyz | Harry: i got issues with sofia-sip when i try to connect spa941 in FS |
| 12:38.06 | engrxyz | it won't damn register but this same phone is seamless with * |
| 12:42.53 | HarryR | it doesn't register with asterisk either? |
| 12:43.45 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 12:45.02 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 12:45.41 | engrxyz | harryR: the spa941 sip ua register right away with * sip but with FS sofia-sip, i have no success yet |
| 12:46.07 | HarryR | uh, perhaps because freeswitch doesn't do authentication? |
| 12:46.12 | HarryR | atleast not through sip |
| 12:48.03 | engrxyz | HarryR: sofia-sip is a library in FS that handles SIP UA registration and other related SIP tasks |
| 12:48.52 | engrxyz | enabling the debug option for sofia-sip in FS will tell us that it won't accept registration |
| 12:48.58 | engrxyz | from a sip ua client |
| 12:52.41 | *** join/#asterisk YoYo (n=chatzill@pool-72-66-178-250.ronkva.east.verizon.net) |
| 12:53.09 | Mavvie | APPLICATION ERROR #1303 |
| 12:53.09 | Mavvie | Invalid value for field |
| 12:53.12 | Mavvie | yes, which field? |
| 12:54.57 | Mavvie | http://bugs.digium.com/view.php?id=10449 <- interesting SIP behaviour. |
| 12:55.10 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 12:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 12:56.03 | asterisknerds | <PROTECTED> |
| 12:56.27 | *** join/#asterisk Yoe (n=wouter@samba.grep.be) |
| 12:57.58 | Yoe | Hi! newbie here -- do I need to do something special to explain to asterisk that a specific device is an ATA, or does SIP in principle allow any device to be an ATA? |
| 12:59.24 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 12:59.28 | *** join/#asterisk egypcio (n=egypcio@postfix.tradein.com.br) |
| 12:59.39 | JT | an ATA is an ATA, what's the question? |
| 12:59.59 | lirakis | Yoe: .. yeahh .. uhh... a sip endpoint.. is a sip endpoint no matter what you call it |
| 13:00.43 | Yoe | ... so to make it work with asterisk, I'd need to configure an account for it and then somehow pass a phone number to it from extensions.conf? |
| 13:00.56 | Yoe | or is that totally off? |
| 13:01.05 | Yoe | (device in question is a Linksys SPA3102, if that matters) |
| 13:01.39 | JT | sure, pretty much |
| 13:02.06 | Yoe | right |
| 13:02.47 | [TK]D-Fender | Yoe: And ATA speaks SIP to *, thats all it cares about. The fact that it uses an analog phone (or line) behind it vs being a SIP hardphone is irrelevant |
| 13:02.50 | RypPn | Yoe: http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+5#531LinksysSipuraSPA3000FXOFXSDevice |
| 13:03.14 | Yoe | ah, interesting |
| 13:03.18 | [TK]D-Fender | Yoe: And forget this "pass a phone number to it bit". |
| 13:03.20 | JT | ~thebook |
| 13:03.22 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:03.23 | Yoe | I'd been looking on that wiki, but somehow must've missed that. |
| 13:03.36 | JT | ignore all the asterisk@home bits ;) |
| 13:03.43 | Yoe | JT: yeah, I've just found that, too :) |
| 13:07.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:07.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:10.24 | waptaxi | I've noticed that when we have ringinuse=yes in queues.conf and only one member in the queue, if this member in use, other caller can't reach him. But if we have more members, he will get second call.. |
| 13:11.33 | waptaxi | is it a bug or a feature? |
| 13:12.58 | waptaxi | tested with asterisk 1.4.10.1 |
| 13:13.09 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:15.29 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:19.01 | *** join/#asterisk Corydon76-dig (i=six@pdpc/supporter/sustaining/Corydon76-home) |
| 13:19.01 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:19.19 | lmadsen | Corydon76-dig: !!! |
| 13:20.36 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
| 13:23.13 | corruptor | hi all. Does anyone use ooh323c here? i have one easy question... |
| 13:23.19 | *** join/#asterisk sopo2k4 (n=jam@host86-153-44-68.range86-153.btcentralplus.com) |
| 13:23.31 | sopo2k4 | hey does anyone know how to decrement a variable in asterisk? |
| 13:24.06 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 13:24.23 | corruptor | Set(var = $[${var} - 1]) |
| 13:24.41 | sopo2k4 | cheers |
| 13:24.58 | sopo2k4 | is there a variable for the current user logged into the asterisk manager who initated a originate? |
| 13:26.51 | corruptor | i haven't heard about such variable. I think there isn't but i'm not sure. |
| 13:27.13 | Tako-san | Anyone available to help troubleshoot a PRI line activation? Particularly if you have experience with Telus. |
| 13:27.35 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:28.09 | sopo2k4 | ah |
| 13:29.11 | Tako-san | Is there a valid "unknown" option for switchtype in zapata.conf? |
| 13:30.35 | [TK]D-Fender | Tako-san: No. You shuold know what you ordered. phone them up if you're not sure |
| 13:30.54 | [TK]D-Fender | corruptor: white-space = BAD |
| 13:31.20 | *** join/#asterisk doughecka (n=doug@unaffiliated/doughecka) |
| 13:31.53 | lmadsen | well... not all whitespace is bad... you can use it in expressions to make it clear |
| 13:32.06 | Tako-san | [TK]D-Fender: I know what I ordered. The problem is the Telco wants me to use switchtype uknown (they done want national) but I am not sure "unknown" is a valid option. In fact I am pretty sure it is not a valid option as asterisk keeps crashing. |
| 13:32.08 | lmadsen | Set(var=$[${var} - 1]) would be the preferred |
| 13:32.33 | lmadsen | Tako-san: who the heck said to use switchtype unknown? That makes so little sense to me... |
| 13:32.37 | DarylVOIP | Hey all. Does anyone know of a way for me to run an AGI without it answering the channel? I'm trying to do a db lookup in a php agi to figure out if the caller's ANI is in my database (for callback) and play a progress tone based on success/failure, and then drop the channel without every answering it. |
| 13:33.08 | Tako-san | lmadsen: I am talking to the Telco people in charge of the cutover and that is their recommendation. From 2 different techs so far. |
| 13:33.13 | corruptor | ok my mistake. I like using whitespaces in c :). |
| 13:33.35 | *** join/#asterisk saftsack (n=oliver@p54A7FDC6.dip.t-dialin.net) |
| 13:33.55 | *** join/#asterisk hohum_ (n=dcorbe@gate.globecommsystems.com) |
| 13:34.34 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 13:34.52 | DarylVOIP | Crap - knocked my network cable out. |
| 13:35.20 | corruptor | Tako-san: there are not so many different optons for pri line in zapata.conf, i think you just need to try them all |
| 13:35.56 | Tako-san | corruptor: Nod |
| 13:36.41 | [TK]D-Fender | DarylVOIP: have you tried jsut running it without an answer? |
| 13:36.44 | Tako-san | corruptor: They finally decided to go with 5ess |
| 13:36.52 | DarylVOIP | Yes, I have - and it answers. |
| 13:37.14 | [TK]D-Fender | DarylVOIP: can you pastebin CLI @ verbose 10 & AGI debug. |
| 13:37.24 | DarylVOIP | Sure. |
| 13:39.14 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:41.41 | *** join/#asterisk prizark_ (i=prizark_@bb121-6-133-90.singnet.com.sg) |
| 13:42.25 | x86 | morning |
| 13:43.23 | DarylVOIP | Ugh.....I'll have that up in a bit. I need to roll back some changes that I was trying out. |
| 13:47.42 | *** join/#asterisk Ebola (n=Ebola@host86-139-49-76.range86-139.btcentralplus.com) |
| 13:51.06 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 13:51.57 | *** join/#asterisk saftsack (n=oliver@p54A7D091.dip.t-dialin.net) |
| 13:53.28 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:00.17 | Tako-san | Ok, I have changed switchtype to 5ess in zapata.conf but when the Telco does a line trace they say we are still using national. Can someone have a quick look at my zapata and tell me if there is something glaringly wrong in there? |
| 14:00.18 | Tako-san | http://pastebin.ca/656907 |
| 14:01.50 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 14:04.33 | [TK]D-Fender | Tako-san: Have you complete stopped * redone "ztcfg -vvvv" and restarted *? |
| 14:05.02 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:05.03 | Tako-san | i didnt do ztcfg |
| 14:06.05 | *** join/#asterisk h4890 (n=Administ@fw.pakom.net) |
| 14:06.17 | h4890 | Hello. |
| 14:06.31 | Tako-san | [TK]D-Fender: But otherwise yes |
| 14:06.40 | DarylVOIP | [TK[D-Fender: OK....got that straightened out. Here's a verbose 10 and AGI debug. |
| 14:06.41 | DarylVOIP | http://pastebin.com/d1b063cfb |
| 14:06.55 | h4890 | Has anyone here ever seen the following error: Apr 25 02:53:32 ERROR[28690]: asterisk.c:1946 main: server not verified: no authorization file, exiting. code: '1' ? |
| 14:07.16 | clyrrad | [TK]D-Fender: I got the paging working last night - wanted to say thanks again for your help! Zaptel and Ztdummy are installed and working :) |
| 14:07.56 | [TK]D-Fender | DarylVOIP: Your AGI is explicitly answering the channel |
| 14:08.05 | [TK]D-Fender | clyrrad: Good to hear |
| 14:08.42 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 14:08.43 | Tako-san | [TK]D-Fender: ztcfg -vvvv all looks good. |
| 14:08.44 | Tako-san | http://pastebin.ca/656918 |
| 14:09.12 | [TK]D-Fender | Tako-san: Now restart * and do a PRI status dump |
| 14:09.16 | h4890 | In case, no one has seen that before, do you know any faq or documentation that I might check (I did already search on asterisk.org but did not find anything)? |
| 14:09.19 | clyrrad | [TK]D-Fender: you were correct about the Page() application too, as soon as zaptel was installed that application was available, its really neat got it working with SPA-94x phones, now I just need to find out about if its possible to connect a Loud Speaker to the paging functionality |
| 14:09.25 | h4890 | (Apr 25 02:53:32 ERROR[28690]: asterisk.c:1946 main: server not verified: no authorization file, exiting. code: '1') |
| 14:09.43 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 14:09.46 | casix | hellow |
| 14:10.00 | [TK]D-Fender | clyrrad: Thre are several amps that you can run off an FXS port, etc as well as SIP based devices |
| 14:10.00 | h4890 | Hello. |
| 14:10.31 | casix | it is possible to define a sql query with a function like func_odbc but not using odbc driver, using mysql directly? |
| 14:10.32 | [TK]D-Fender | clyrrad: There is also chan_oss you can use off your sound card to a straight amp |
| 14:10.38 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 14:10.39 | clyrrad | [TK]D-Fender: yea, the problem here will be that its hosted PBX, only devices will be the endpoints at the location.... |
| 14:10.48 | *** join/#asterisk Cheetah (n=cheetah2@main-gw.bense.de) |
| 14:10.49 | Cheetah | heya :D |
| 14:10.57 | [TK]D-Fender | casix: yes, MSYQL direct from asterisk-addons |
| 14:11.02 | Tako-san | [TK]D-Fender: Showing as Lucent 5E http://pastebin.ca/656921 |
| 14:11.25 | [TK]D-Fender | Tako-san: Status: Provisioned, Up, Active <--- looks good to me |
| 14:11.33 | clyrrad | [TK]D-Fender: ah - there are SIP based devices - good to know |
| 14:11.42 | Tako-san | [TK]D-Fender: Looks good to me too. But no outbound calls are working. Inbound works just fine. |
| 14:11.48 | [TK]D-Fender | clyrrad: So FXS friven it is. |
| 14:12.21 | Cheetah | i have a mISDN (FritzCard) and a digium PRI card. I'd like to route outgoing calls over the mISDN card and if no channels are available anymore on the mISDN card, route the call via the Digium card. |
| 14:12.23 | [TK]D-Fender | Tako-san: That means you just need to fix your dialplan |
| 14:12.38 | Tako-san | [TK]D-Fender: Ok. Will look into that then. Thanks. |
| 14:12.45 | Cheetah | is there an easy way to do this? is s-CHANUNAVAIL sufficient? |
| 14:13.23 | [TK]D-Fender | clyrrad: http://www.vikingelectronics.com/products/ |
| 14:13.24 | JT | Cheetah: the next priority in the dialplan is sufficient |
| 14:13.37 | casix | [TK]D-Fender: but where can I definde the function? to use odbc i'm using func_odbc.conf with ... read=SELECT pstn FROM callersid WHERE idRemoto='${ARG1}'... |
| 14:13.42 | [TK]D-Fender | Cheetah: that exten does not imply anything. |
| 14:14.42 | [TK]D-Fender | Cheetah: you assume too much from sample macro's. Check DIALSTATUS from your dial attempt. "CHANUNAVAIL" and "CONGESTION" should be the 2 you're looking for. |
| 14:14.44 | clyrrad | [TK]D-Fender: thanks :) |
| 14:14.53 | [TK]D-Fender | JT : not really... |
| 14:14.59 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:15.09 | Cheetah | JT, so if I do something like .X_,1,Dial(ZAP/g1/....) and .X_,2,Dial(mISDN.....)? |
| 14:15.20 | [TK]D-Fender | JT : on ISDN he'll have call progress and "NOANSWER" is NOT a reaswon to use another tech ;) |
| 14:15.22 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.254) |
| 14:16.02 | h4890 | Has anyone seen anything like this before? ERROR[28690]: asterisk.c:1946 main: server not verified: no authorization file, exiting. code: '1' ? |
| 14:16.04 | JT | [TK]D-Fender: it will help ensure the call gets through :D |
| 14:16.05 | [TK]D-Fender | Cheetah: What is Zap/g1 using? |
| 14:16.23 | Cheetah | what happens if the call goes through, does the next priority gets executed as well? |
| 14:16.38 | Cheetah | Zap/g1 is using the Digium card PRI |
| 14:16.40 | [TK]D-Fender | Cheetah: Not uless you pass it the "g" option. |
| 14:16.44 | Cheetah | hmm |
| 14:16.45 | Cheetah | hang on :D |
| 14:16.55 | [TK]D-Fender | Cheetah: You'll want to check for the 2 status' I just gave you. |
| 14:17.04 | [TK]D-Fender | Cheetah: not just dial them back to back |
| 14:17.11 | Cheetah | yeah |
| 14:17.13 | Cheetah | i figured that |
| 14:17.22 | Cheetah | how do you call that behaviour? trunking? |
| 14:17.32 | *** join/#asterisk saftsack (n=oliver@p54A7CD58.dip.t-dialin.net) |
| 14:17.58 | [TK]D-Fender | Cheetah: No, its called DIALPLAN. |
| 14:18.10 | Cheetah | uh yeah, but the technique |
| 14:18.30 | [TK]D-Fender | Cheetah: there is no name for random (or less random) actions you feel like doing in your dialplan. Its just "stuff" |
| 14:18.41 | Cheetah | umkay :D |
| 14:18.51 | [TK]D-Fender | Cheetah: No buzz-word, techno-babble, etc. |
| 14:18.56 | Cheetah | Dial 2.0 |
| 14:20.50 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:21.26 | HarryR | interactive telephone agent handling environment |
| 14:22.07 | clyrrad | [TK]D-Fender: check this one out http://www.cyberdata.net/products/voip/voip-speaker.html right at par with what I am after |
| 14:23.03 | [TK]D-Fender | clyrrad: Think I saw that one listed at VoipSupply. might be an option, but this one looks more like for office rooms (wher you'd have phones anyways). Depends on your deployment. |
| 14:23.13 | *** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket) |
| 14:23.25 | [TK]D-Fender | clyrrad: You'd have to compare the variety of equipment based on the environment. |
| 14:23.34 | HarryR | voip ceiling speakers? |
| 14:23.35 | HarryR | wtf |
| 14:23.44 | clyrrad | [TK]D-Fender: yea its going to be used to page back in a warehouse |
| 14:23.57 | [TK]D-Fender | clyrrad: Oh hell no! |
| 14:24.04 | clyrrad | [TK]D-Fender: trick will be getting it to work with Asterisk |
| 14:24.09 | clyrrad | hahaha yup |
| 14:24.20 | [TK]D-Fender | clyrrad: That thing CAN'T possibly put out enough poewr for that kind of noise & distance |
| 14:24.32 | [TK]D-Fender | clyrrad: Its friggen PoE powered! |
| 14:24.47 | clyrrad | [TK]D-Fender: ah - yea that I know.... but the "idea" of the hardware is what I am after |
| 14:24.56 | [TK]D-Fender | clyrrad: Go back to viking and get a real FXS amp and horn system... |
| 14:25.11 | [TK]D-Fender | clyrrad: yes, "BAD" ideas are ideas too ;) |
| 14:25.22 | clyrrad | [TK]D-Fender: Yea I am still looking at that page too - how do I interface that horn with Asterisk? |
| 14:25.47 | clyrrad | [TK]D-Fender: remember its hosted PBX, so no sound card etc available at the location where the paging horn will be |
| 14:25.50 | [TK]D-Fender | clyrrad: Via and FXS port (slap an ATA with it) |
| 14:25.58 | [TK]D-Fender | an* |
| 14:26.01 | clyrrad | [TK]D-Fender: gotcha |
| 14:26.26 | [TK]D-Fender | ouch, voipsupply = doa |
| 14:26.50 | clyrrad | [TK]D-Fender: where would you use the PoE speakers, in an office? |
| 14:26.56 | clyrrad | quite office I should say....... |
| 14:27.22 | [TK]D-Fender | clyrrad: yes, places like entryways, cafeterias, etc. |
| 14:27.27 | [TK]D-Fender | halls... |
| 14:27.43 | clyrrad | Places where this is not warehouse noice in otherwords.... |
| 14:27.52 | clyrrad | noise* |
| 14:28.01 | [TK]D-Fender | clyrrad: Think about how loud a PoE SIP hardphone would be on speakerphone. Thats your max. Definately NOT sutable for warehouse. |
| 14:28.20 | clyrrad | [TK]D-Fender: indeed - definatlly would not do the job |
| 14:28.21 | [TK]D-Fender | any resonable office area would be jsut fine. |
| 14:28.57 | clyrrad | [TK]D-Fender: would be nice to find an ATA + AMP + Horn all in one unit kinda thing |
| 14:29.13 | clyrrad | Asterisk compatible ofcourse :D |
| 14:30.38 | clyrrad | [TK]D-Fender: hrm, intreesting thing about the ATA idea is...... I am not sure that it can support the Page option, from what I read that only works on the IP Phones, have you ever got a paging system to work as mentioned with an ATA? |
| 14:31.36 | *** join/#asterisk CVirus (n=GoD@62.135.96.152) |
| 14:33.53 | h4890 | Just to let you know... |
| 14:34.19 | h4890 | Hmm. |
| 14:34.20 | h4890 | Ok. |
| 14:34.21 | h4890 | =) |
| 14:34.22 | h4890 | Bye. |
| 14:34.31 | h4890 | And thank you for the help. =) |
| 14:34.32 | h4890 | Bye. |
| 14:34.33 | *** part/#asterisk h4890 (n=Administ@fw.pakom.net) |
| 14:34.52 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 14:35.09 | JT | some sort of mental illness there for sure |
| 14:37.08 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:37.17 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-46eafc2796f88826) |
| 14:38.17 | _Krieger_ | why dialplan is duplicated in extensions.ael and extensions.conf? is it safe, for example, to delete .ael? |
| 14:38.39 | JT | yes |
| 14:38.49 | JT | if you aren't using it |
| 14:39.05 | _Krieger_ | where to look for which file is read by *? |
| 14:40.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:40.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:41.29 | *** join/#asterisk techie (n=techie@adsl-68-127-122-88.dsl.frsn02.pacbell.net) |
| 14:42.46 | [TK]D-Fender | clyrrad: the amp you plug the ATA it will do the answering (thats its job). You don't have to tell the ATA to answer. |
| 14:43.48 | [TK]D-Fender | _Krieger_: I believe that * might have a shit-fit about not finding the config file. You can avoid this altogether by simply adding "noload => pbx_ael.so" to your modules.conf |
| 14:43.57 | *** join/#asterisk shareenergy (i=shareene@62.169.103.201.rev.optimus.pt) |
| 14:44.24 | shareenergy | anyone knows why my iaxmodem can't connect to hylafax? |
| 14:45.16 | [TK]D-Fender | shareenergy: Normally we are psychic on Tuesdays, but its changed to MONDAYS now. |
| 14:45.46 | clyrrad | [TK]D-Fender: alright, so key will be to find an amp that works with Asterisk, and add the necessary SIP Headers like I had to do for the SPA-94x devices |
| 14:46.02 | shareenergy | lolol sorry |
| 14:46.03 | shareenergy | i mean |
| 14:46.05 | [TK]D-Fender | clyrrad: No. |
| 14:46.11 | clyrrad | :( |
| 14:46.14 | [TK]D-Fender | clyrrad: You've missed the point |
| 14:46.23 | shareenergy | i have everything is normal iaxmodem registers |
| 14:46.40 | shareenergy | but when it comes to link to hylafax it gives the error |
| 14:46.45 | clyrrad | [TK]D-Fender: ok im all eyes :-/ |
| 14:46.54 | shareenergy | <PROTECTED> |
| 14:47.13 | [TK]D-Fender | clyrrad: They runn off an FXS port. They are ANALOG and can be used with ANY system. When the line its attached to RINGS, *IT* answers and you're "live". Any stupid ATA will do, and there is NO head or anything special needed to eb passed to it. |
| 14:47.55 | clyrrad | [TK]D-Fender: ohhhhhhhhhh LOL - got it, so its just acting like a "person who picked up the extension" |
| 14:48.11 | clyrrad | [TK]D-Fender: ok that makes sense |
| 14:48.16 | [TK]D-Fender | clyrrad: Yes, and it WILL always answer, thats its JOB |
| 14:48.30 | clyrrad | [TK]D-Fender: got it, so then this should be an easy install in this case |
| 14:48.31 | [TK]D-Fender | clyrrad: I think we've acheived comprehension now :) |
| 14:48.37 | [TK]D-Fender | clyrrad: Yes, quite. |
| 14:48.39 | clyrrad | [TK]D-Fender: indeed we have :D |
| 14:49.12 | jeremy_g | what does SIP 404 not found signify? |
| 14:49.14 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:49.22 | jeremy_g | does that mean the extension dialled does not actually exist |
| 14:49.33 | clyrrad | jeremy_g: its like 404 not found as far as I understand |
| 14:50.00 | *** join/#asterisk d3wayne (n=deeewayn@c-71-228-186-75.hsd1.al.comcast.net) |
| 14:50.00 | jeremy_g | clyrrad: and whats that? |
| 14:50.12 | clyrrad | jeremy_g: Like it could not reach the end point / phone |
| 14:50.15 | jeremy_g | or does it mean the sip user being dialled does not exist |
| 14:50.31 | jeremy_g | clyrrad:doesnt give information on the exact reason why? |
| 14:51.01 | clyrrad | jeremy_g: have you used sip debug for more info? |
| 14:51.03 | *** join/#asterisk fjean5 (n=fjean5@atelka.info) |
| 14:51.24 | fjean5 | hello guys , how are you |
| 14:51.31 | jeremy_g | clyrrad:yeah, it seems that my extensions.conf table is not being read at all from the realtime db |
| 14:51.56 | jeremy_g | i wonder why? |
| 14:52.09 | [TK]D-Fender | jeremy_g: PASTEBIN your extensions.conf and the CLI output of your failed attempt at verbose 10 & SIP debug enabled |
| 14:52.14 | fjean5 | anybody knows a good iax2 provider to terminate calls with callerid, in Canada for a call-center |
| 14:52.47 | jeremy_g | [TK]D-Fender:ok,here i go |
| 14:53.12 | [TK]D-Fender | fjean5: www.unlimitel.ca |
| 14:53.39 | fjean5 | [tk]d-fender, thanks but they dont do call-centers |
| 14:53.54 | [TK]D-Fender | fjean5: How do you figure? |
| 14:53.55 | clyrrad | Yup unlimitel is great - I have lots of DID's with them, but they do not offer just termination |
| 14:54.17 | clyrrad | You need to buy DID's each DID comes with 4 channels |
| 14:54.19 | jeremy_g | [TK]D-Fender:but wait, i am using asterisk realtime. so i dont have a conf rather a small extensions table |
| 14:54.33 | [TK]D-Fender | fjean5: if you jsut want to terminate to Canada for outbound, there are plenty of US ITSP's with great rates |
| 14:54.48 | [TK]D-Fender | jeremy_g: you NEED extensions.conf for real-time. |
| 14:55.01 | jeremy_g | [TK]D-Fender:its static realtime |
| 14:55.18 | fjean5 | [tk]d-fender: i can imagine, but i am looking for a reference, as i dont want choppy calls |
| 14:55.20 | [TK]D-Fender | jeremy_g: Last I recall you have to have a minimal extensions.conf with the SWITCH directive. |
| 14:55.33 | jeremy_g | [TK]D-Fender:really |
| 14:55.42 | [TK]D-Fender | fjean5: telix comes better recommended around here |
| 14:55.47 | jeremy_g | [TK]D-Fender:its implausible. |
| 14:55.56 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:56.01 | fjean5 | [TK]D-Fender: ok |
| 14:56.12 | jeremy_g | [TK]D-Fender:you only need switch if you are using asterisk realtime that dynamically loads extensions without executing a reload on cli. |
| 14:56.41 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.18.47) |
| 14:56.54 | [TK]D-Fender | jeremy_g: If you say so. Pastebin whatever backup you have then. |
| 14:57.37 | jeremy_g | [TK]D-Fender:i have the same setup (static realtime) working with a different configuration on another server still without any extensions.conf. |
| 14:57.59 | [TK]D-Fender | jeremy_g: Ok, less tell, more show... |
| 14:58.35 | clyrrad | [TK]D-Fender: what is the URL for telix? I just Google found nothing useful |
| 14:59.07 | fjean5 | clyrrad: teliax.com |
| 14:59.20 | [TK]D-Fender | Teliax. Typo |
| 14:59.59 | jeremy_g | [TK]D-Fender:with this configuration, the static ara works. http://www.pastebin.ca/656977 |
| 15:00.01 | clyrrad | ah that expalins it |
| 15:00.57 | *** join/#asterisk Merlin83b2 (n=Merlin83@office.34sp.com) |
| 15:00.58 | jeremy_g | [TK]D-Fender:with this it doesnt work, http://www.pastebin.ca/656980 |
| 15:01.46 | *** join/#asterisk saftsack (n=oliver@84.167.220.157) |
| 15:02.28 | jeremy_g | [TK]D-Fender:both are extensions.conf, this extensions.conf is passed an an argument to a script that puts it into the db. |
| 15:03.22 | jeremy_g | [TK]D-Fender:when i say it works, it means that asterisk loads the data in extensions.conf and my sip phones can dial those extensions |
| 15:03.53 | jeremy_g | [TK]D-Fender:when i say it doesnt work, it means that if an extension is dialled, asterisk would return SIP 404 |
| 15:04.01 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 15:04.14 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:05.12 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 15:07.59 | jeremy_g | [TK]D-Fender:before sending 404 it also sends a 484 address incompelte |
| 15:08.34 | *** part/#asterisk fjean5 (n=fjean5@atelka.info) |
| 15:09.24 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 15:09.24 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 15:12.19 | [TK]D-Fender | jeremy_g: I asked for CLI output..... |
| 15:12.24 | *** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:15.11 | phearless | arg.. I can't make a Transfer via the asterisk manager |
| 15:15.31 | phearless | Action: Redirect is broken ? |
| 15:15.54 | jeremy_g | [TK]D-Fender: http://www.pastebin.ca/657002 |
| 15:16.22 | [TK]D-Fender | jeremy_g: not what I asked for.... geting COLDER...... |
| 15:16.30 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:20.55 | *** join/#asterisk ZeroPing (n=none@adsl-155-210-137.owb.bellsouth.net) |
| 15:21.11 | jeremy_g | [TK]D-Fender: http://www.pastebin.ca/657012 <-- sip debug |
| 15:21.54 | jeremy_g | [TK]D-Fender:damn paste bin is limited |
| 15:23.04 | jeremy_g | [TK]D-Fender: ok check this http://www.pastebin.ca/657016 <-- sip debug |
| 15:23.08 | jeremy_g | uploaded it |
| 15:25.10 | [TK]D-Fender | jeremy_g: Well in the bad pastbin you gave me ( http://www.pastebin.ca/657002 )you didn't have the exten for : Looking for rixin2 in incoming-sip (domain 192.168.0.2) |
| 15:27.10 | jeremy_g | [TK]D-Fender:yeah ignore that, cuz its not important to entertain calls incoming from the operator |
| 15:28.05 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 15:28.07 | jeremy_g | sorry |
| 15:29.44 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 15:31.17 | *** join/#asterisk Tako-san (n=Tako-san@154.5.212.245) |
| 15:31.18 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 15:31.56 | x86 | ugh |
| 15:32.03 | x86 | anyone good with compiling sangoma drivers? |
| 15:32.09 | Uatec | bummer, i just broke my pen by putting it in my desk fan |
| 15:32.19 | Uatec | i didnt' notice till i heard a loud noise and ink went everywhere |
| 15:32.27 | Tako-san | x86: What's the problem? |
| 15:32.28 | x86 | having a big problem compiling them against zaptel 1.2.18 and linux kernel 2.6.22.2 |
| 15:32.42 | tzafrir | x86, ztdummy? |
| 15:32.52 | x86 | tzafrir: what about it? |
| 15:33.08 | tzafrir | is that where you get the error? |
| 15:33.14 | coppice | x86: a versions thing? sangoma 2.3.4 won't build against recent kernels. you need to use 3.1.x |
| 15:33.17 | x86 | error: 'struct sk_buff' has no member named 'mac' |
| 15:33.27 | x86 | coppice: this is wanpipe 3.1.3 |
| 15:33.54 | x86 | tzafrir: sdladrv_src.o |
| 15:34.01 | coppice | I use 3.1.3 OK with 2.6.20. Haven't tried a newer kernel |
| 15:34.11 | x86 | hmm, perhaps i'll downgrade |
| 15:34.13 | tzafrir | sorry. I leave it to Sangoma to deal with their drivers... |
| 15:40.56 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.2) |
| 15:42.28 | phearless | anybody can help me with Action: Redirect ? |
| 15:42.43 | phearless | when the handset that do the xfer hang up, the call is cut |
| 15:44.20 | LoneShadow | Anyone used a bluetooth headset with asterisk ? |
| 15:45.07 | wothinn | Hook your bluetooth headset in to a softphone or get a Plantronics Voyager 500A and plug that in to your hardphone. |
| 15:48.31 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 15:49.18 | LoneShadow | wothinn: I want to be able to dial some extension by pressing a button. And maybe launch a speech recognition IVR |
| 15:50.22 | LoneShadow | so no softphone or hardphone |
| 15:51.53 | x86 | no good with a vanilla 2.6.22.2 from kernel.org (was using 2.6.22 from gentoo before) |
| 15:53.59 | [TK]D-Fender | LoneShadow: ..... "would you like fries with that sir?" |
| 15:54.05 | phearless | arg |
| 15:54.07 | LoneShadow | :P |
| 15:54.19 | LoneShadow | I got the speech thingie working |
| 15:54.46 | LoneShadow | trying to see if bluetooth headset would be the final topping :D |
| 15:55.49 | LoneShadow | [TK]D-Fender: so with a bluetooth dongle + headset, what can be achieved ? |
| 15:56.11 | [TK]D-Fender | LoneShadow: Not much. This isn't Star Trek you know. |
| 15:56.28 | LoneShadow | hmm |
| 15:58.26 | *** join/#asterisk ManxPower (n=manxpowe@032-447-153.area7.spcsdns.net) |
| 15:58.41 | LoneShadow | I wonder how those new voice bluetooth headsets work, SE 662, says it can dial phone numbers for certain phones |
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| 16:02.44 | x86 | yes, it is a problem with >= kernel 2.6.22 (at least() |
| 16:02.55 | x86 | 2.6.20 solved my wanpipe 3.1.3 compilation issues |
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| 16:37.00 | shareenergy | does the asterisk needs to answer the line before it passes to iaxmodem and then hylafax? |
| 16:41.07 | rrittenhouse | The company I work for wants to look into Asterisk for their PBX. Would I be smart to look into the Business edition (they just got a quote for another PBX and it was 20K) |
| 16:41.21 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:43.42 | Qwell[] | rrittenhouse: BE provides support and such, so if you want that, it might be a good idea |
| 16:44.03 | wunderkin | cough cough cough |
| 16:44.09 | rrittenhouse | Is the interface any different? |
| 16:44.10 | Qwell[] | (disclaimer, I work for Digium) |
| 16:44.17 | russellb | the interface is the same |
| 16:44.18 | Qwell[] | rrittenhouse: no |
| 16:44.25 | russellb | (disclaimer, i also work for digium) |
| 16:44.39 | russellb | and you also get priority in getting bugs fixed by the digium engineers |
| 16:44.53 | denon | bugs? there are bugs in BE? |
| 16:45.00 | russellb | it's software |
| 16:45.02 | putnopvut | denon: lol |
| 16:45.21 | denon | russellb: so you're saying there are no bugs in hardware? |
| 16:45.44 | russellb | hardware is perfect |
| 16:45.55 | denon | *cough*x100p |
| 16:45.58 | russellb | everyone just blames the drivers |
| 16:46.20 | denon | rrittenhouse: you'd probably be happier with BE if you're not feeling extremely ambitious to dive into and maintain asterisk |
| 16:46.30 | rrittenhouse | but i am ;) |
| 16:46.37 | denon | maintain as in long term |
| 16:46.45 | denon | not just set it up 'cause it's fun |
| 16:46.47 | rrittenhouse | yeah I want it to be a part of the IT department |
| 16:46.57 | tzafrir | x100p? it's not a bug, it's a feature |
| 16:47.08 | denon | and when there's an outtage, BE gives you someone to help |
| 16:47.38 | rrittenhouse | yeah |
| 16:47.39 | [TK]D-Fender | rrittenhouse: Lets start from the beginning : Starte your NEEDS. How many phones, what kinda, and how many lines? |
| 16:47.39 | [TK]D-Fender | state* |
| 16:47.39 | rrittenhouse | good point |
| 16:47.48 | rrittenhouse | 7 external lines |
| 16:47.56 | rrittenhouse | analog (thats what we have now anyhow) |
| 16:48.05 | russellb | TDM800P :) |
| 16:48.54 | [TK]D-Fender | rrittenhouse: Have you checked to see what you can get a partial PRI for in your area? |
| 16:49.08 | rrittenhouse | Not yet I was just told that I could look into this as an option just before lunch ;) |
| 16:49.28 | rrittenhouse | I want to see if i can get their current setup and costs |
| 16:49.28 | [TK]D-Fender | rrittenhouse: Ok, do so. Next : how many phones? |
| 16:50.08 | rrittenhouse | no more than 60 |
| 16:50.23 | Qwell[] | rrittenhouse: are your phones analog right now? |
| 16:50.30 | rrittenhouse | They are currently comdial Impact phones |
| 16:50.33 | rrittenhouse | digital |
| 16:50.34 | [TK]D-Fender | rrittenhouse: basic ext's? |
| 16:50.37 | Qwell[] | yuck |
| 16:50.38 | rrittenhouse | yeah |
| 16:50.53 | rrittenhouse | Im trying to determine if we want to use softphones or hardware phones |
| 16:51.06 | rrittenhouse | I know the personal preference is going to change from person to person though |
| 16:51.13 | Qwell[] | rrittenhouse: IMO, softphones will make you very unhappy. |
| 16:51.20 | rrittenhouse | ah alright |
| 16:51.25 | russellb | agreed ... |
| 16:51.37 | [TK]D-Fender | rrittenhouse: give me a specific # of phones. |
| 16:51.38 | Qwell[] | You can get a halfway decent SIP phone for around $120 |
| 16:51.42 | Sweeper | yea. call centers are really the only place you can really get away with softphones |
| 16:51.43 | coppice | all phones make me unhappy. |
| 16:51.48 | Sweeper | Qwell[]: moar liek $87 |
| 16:51.51 | russellb | coppice: lol .. |
| 16:51.51 | Qwell[] | jbot: coppice++ |
| 16:51.55 | Sweeper | polycom 320 to the resque~ |
| 16:52.01 | Qwell[] | Sweeper: the 320 is that low? |
| 16:52.08 | Sweeper | Qwell[]: sans brick, yep |
| 16:52.11 | Qwell[] | ahh |
| 16:52.17 | Qwell[] | yeah, and the brick is like $30 I imagine |
| 16:52.20 | Sweeper | $20 |
| 16:52.28 | denon | I'm kind of fund of the linksys 900-phones too |
| 16:52.28 | Sweeper | and you can get non-official ones for $10 |
| 16:52.28 | Qwell[] | plus shipping :p |
| 16:52.31 | rrittenhouse | 60 Phones |
| 16:52.32 | denon | (the old sipuras) |
| 16:52.33 | [TK]D-Fender | screw the brick, POE <--- |
| 16:52.36 | rrittenhouse | that gives us room to grow |
| 16:52.53 | Sweeper | yea, POE is nice~ |
| 16:53.15 | Sweeper | I guess you can get a couple 48 port POE switches for < $600? |
| 16:53.31 | Sweeper | or a 48 and a 24, I guess |
| 16:53.36 | rrittenhouse | I didnt consider POE |
| 16:53.38 | rrittenhouse | hmm |
| 16:53.46 | rrittenhouse | we do have to rewire anyhow |
| 16:53.47 | denon | always consider POE :) |
| 16:53.49 | *** join/#asterisk sakic (n=sakic@adsl-227-157-12.clt.bellsouth.net) |
| 16:53.56 | [TK]D-Fender | rrittenhouse: Polycom IP 320 = 60 * 87.50 = $5250 ( http://www.telephonydepot.com/product_p/105-058-320.htm ) |
| 16:54.08 | Qwell[] | [TK]D-Fender: the 320 is the one with the switch port, right? |
| 16:54.40 | Qwell[] | and...yeah... [TK]D-Fender, you need to get telephonydepot to start giving you commission :P |
| 16:54.47 | Zipper_32 | 330 is the one with the switchport |
| 16:55.14 | [TK]D-Fender | rrittenhouse: Sangoma A200d (8 FXO) = $819.50 ( http://www.telephonydepot.com/product_p/105-052-a200brme.htm ) |
| 16:56.28 | [TK]D-Fender | rrittenhouse: 4 x D-Link DES-1228P (24 port PoE Switches) = 4 * 414.66 = $1658.64 ( http://www.antonline.com/antonline.php?op=inventory&st=DES-1228P ) |
| 16:56.39 | [TK]D-Fender | rrittenhouse: There's your whole project. (minus basic server) |
| 16:56.59 | rrittenhouse | wow :P |
| 16:58.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:58.11 | Qwell[] | or, of course, the TDM800P, rather than the Sangoma http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=1TDM808BF-01 :D |
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| 16:58.45 | Sweeper | [TK]D-Fender: why 96 ports for 60 phones? :P |
| 16:59.06 | Sweeper | well, I guess one if he wants redundancy... |
| 16:59.13 | [TK]D-Fender | Sweeper: because I suck at math :) |
| 16:59.18 | Sweeper | :D |
| 16:59.19 | rrittenhouse | lol |
| 16:59.21 | [TK]D-Fender | rrittenhouse: Knock of one of those switches! |
| 16:59.25 | rrittenhouse | ;) |
| 16:59.25 | rrittenhouse | k |
| 16:59.37 | Sweeper | damn tho, those switches seem really pricey |
| 16:59.46 | Qwell[] | pricey? for a 24 port PoE switch? |
| 17:00.07 | Sweeper | well, they're d-link, so they're probably reasonable for PoE |
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| 17:00.14 | Sweeper | it just seems they should be cheaper :P |
| 17:00.27 | Sweeper | I mean, you can get a regular 24-porter for about $100 |
| 17:00.49 | [TK]D-Fender | Sweeper: PoE Costs, and has come down a lot |
| 17:01.03 | Sweeper | apparently :D |
| 17:01.18 | Sweeper | granted, I also think wrt-54's should cost $20 |
| 17:02.49 | [TK]D-Fender | Qwell[]: Where does one go to get thier warranty provided HPEC ? |
| 17:02.56 | Qwell[] | [TK]D-Fender: call up sales |
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| 17:04.36 | rrittenhouse | So dont get the Sangoma and get the TDM800P |
| 17:04.38 | rrittenhouse | right? |
| 17:04.59 | Qwell[] | rrittenhouse: well, it's up to you of course.. people have differing opinions on the matter |
| 17:05.08 | Qwell[] | and obviously, working for Digium, mine is maybe a bit biased |
| 17:05.11 | rrittenhouse | hehe |
| 17:05.16 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 17:05.40 | Qwell[] | though, Sangoma has never contributed anything to Asterisk or Zaptel, so...yeah ;) |
| 17:05.58 | Qwell[] | (I didn't like them well before I started working for Digium) |
| 17:07.11 | Sweeper | well, I can't vouch for the analog stuff |
| 17:07.41 | Sweeper | but for t1's and such, at least you can use sangoma with more that 10% of motherboards, hdd controllers, and usb devices :P |
| 17:08.01 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 17:08.10 | Sweeper | wooohooo 1k irq's per second \o\ |
| 17:08.17 | *** join/#asterisk ivrc (n=chatzill@74.228.54.150) |
| 17:08.30 | hrmphh | would you guys get linksys (sipura) spa942 or refurbished cisco 7940g? |
| 17:08.33 | hrmphh | both phones are the same price |
| 17:08.40 | Qwell[] | hrmphh: Polycom 320 |
| 17:09.06 | hrmphh | poe? |
| 17:09.09 | Sweeper | yes |
| 17:09.13 | hrmphh | hmm |
| 17:09.16 | hrmphh | those are better? |
| 17:09.18 | Sweeper | yes |
| 17:09.19 | hrmphh | what makes you say so? |
| 17:09.24 | hrmphh | i see theyre considerably cheaper |
| 17:09.29 | Sweeper | I have a cisco, and I have a polycom |
| 17:09.37 | Sweeper | the cisco is on the closet floor |
| 17:09.39 | hrmphh | heh |
| 17:09.40 | vinafera | I am really confused be looking at various hardware choices in front of me. Based on the product description the Redfone Fonebridge2 looks like a very good product at a very good price and it appears to me that it gives failover by default. Experience has taught me that things are not what they seem |
| 17:09.44 | Sweeper | I take the polycom to bed with me |
| 17:09.45 | hrmphh | ive used 7940s in the past |
| 17:09.48 | hrmphh | and have had no problems |
| 17:09.48 | Qwell[] | Sweeper: send it here, I'll give it a good home. |
| 17:10.16 | hrmphh | oh hmm doesnt have switch |
| 17:10.18 | hrmphh | id need a 330 |
| 17:10.30 | hrmphh | will it split voice/data in to separate vlans? |
| 17:10.55 | vinafera | does anyone have experience with that product vs. having redundant servers with TE410P's in each one? |
| 17:11.01 | Sweeper | it has vlan support, if that's what you mean |
| 17:11.09 | Sweeper | the switch port is just a switch port |
| 17:11.18 | Sweeper | but the phone will be on whatever vlan you put it on |
| 17:11.37 | hrmphh | right, i just want to make sure the traffic will be on separate vlans so i can do end-to-end qos |
| 17:11.46 | Sweeper | yea, no problem there |
| 17:11.50 | hrmphh | k |
| 17:11.56 | hrmphh | so id need the 330 |
| 17:11.59 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 17:12.06 | hrmphh | which is $110 vs. the $125-30 of the linksys or cisco |
| 17:12.10 | hrmphh | but youre saying its better quality |
| 17:12.13 | Sweeper | that it is |
| 17:12.15 | hrmphh | any features i lose? |
| 17:12.28 | Sweeper | probably some softkeys |
| 17:12.47 | hrmphh | blind xfer is a hard key i take it? |
| 17:12.55 | Qwell[] | hrmphh: You don't get to deal with Cisco when things go wrong. |
| 17:12.58 | hrmphh | these things can take config via tftp? |
| 17:12.59 | hrmphh | true |
| 17:12.59 | Qwell[] | If you want to call that a feature |
| 17:13.02 | Sweeper | hrmphh: yep |
| 17:13.02 | [TK]D-Fender | hrmphh: No, you GAIN features. |
| 17:13.03 | hrmphh | TAC _used_ to be good |
| 17:13.06 | hrmphh | like 10 years ago |
| 17:13.08 | Sweeper | and http, or https, or ftp |
| 17:13.15 | hrmphh | tk; what do i gain? |
| 17:13.19 | Sweeper | microbrowser! |
| 17:13.26 | [TK]D-Fender | hrmphh: Cisco/Linksys does not really support presence, and Polycom's call handling is superior. |
| 17:13.28 | Sweeper | custom ringtones! |
| 17:13.39 | hrmphh | presence? |
| 17:13.41 | [TK]D-Fender | hrmphh: join/split is a must for me. |
| 17:13.48 | hrmphh | dunno what that is |
| 17:13.55 | [TK]D-Fender | hrmphh: Presence = blf |
| 17:13.57 | [TK]D-Fender | ~blf |
| 17:13.58 | jbot | from memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 17:14.16 | hrmphh | oh yeah |
| 17:14.42 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 17:14.48 | hrmphh | so you can snoop on your coworkers? heh |
| 17:14.56 | hrmphh | it tells you if your speed dial destinations are on the phone? |
| 17:15.15 | Sweeper | on a 330, that's pretty much irrelevant :P |
| 17:15.24 | hrmphh | why is that |
| 17:15.39 | Sweeper | only 1 speed dial possible :P |
| 17:15.43 | hrmphh | lol |
| 17:16.00 | Sweeper | well, speed dial with BLF, anyways |
| 17:16.06 | rrittenhouse | [TK]D-Fender, Is this about the right price? http://www.voiplink.com/Digium_TDM800P_p/digium-tdm800p.htm |
| 17:16.13 | rrittenhouse | if i were to go with the tdm800p |
| 17:16.13 | hrmphh | bah to digium |
| 17:16.16 | hrmphh | that hardware = garbage |
| 17:16.19 | hrmphh | i just ordered sangoma |
| 17:16.22 | rrittenhouse | ah |
| 17:16.26 | rrittenhouse | what didnt work with it? |
| 17:16.29 | rrittenhouse | or what problems did you have |
| 17:16.30 | hrmphh | umm, calls? |
| 17:16.36 | hrmphh | intermittent static, etc. |
| 17:16.42 | rrittenhouse | oh ok |
| 17:16.44 | hrmphh | on a handful of different mobos |
| 17:16.47 | hrmphh | trust me, get a sangoma |
| 17:16.50 | rrittenhouse | k |
| 17:16.51 | hrmphh | theyre years ahead |
| 17:17.12 | rrittenhouse | Just trying to research for my boss |
| 17:17.34 | vinafera | It is my guess that since all I see on the fonebridge2 is press releases and sales info they are either new or crappy or both. |
| 17:17.38 | ivrc | have an issue with outbound dialing on a TDM400P card (POTS) - calls are delivered to the destination, but * keeps ringing, doesn't detect the answer |
| 17:17.55 | ivrc | suggestions would be appreciated! |
| 17:18.06 | rrittenhouse | were a media company with FM and TV transmitters on site too so im trying to take all of that into consideration |
| 17:18.08 | hrmphh | so shared call/bridged line appearance works on the soundpoints w/asterisk? |
| 17:18.11 | rrittenhouse | two different buildings |
| 17:21.50 | hrmphh | ok so what DONT you like about the polycom 320/330? |
| 17:22.09 | rrittenhouse | yeah thats what I wanna know ;) Who heres used it? |
| 17:22.17 | rrittenhouse | Everybody here are used to Comdial Impact phones |
| 17:22.35 | xheliox | Holy smokes. I just applied the patch that uses the highres timer for ztdummy, and the zttest results are remarkable. It's so so so much better. |
| 17:25.42 | hrmphh | heh |
| 17:25.58 | [TK]D-Fender | hrmphh: Polycom = solid |
| 17:27.25 | hrmphh | yeh but if you had to pick ONE thing |
| 17:27.27 | russellb | xheliox: nice! |
| 17:27.27 | hrmphh | what dont you like |
| 17:27.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:27.56 | russellb | xheliox: note your results in the bug report please |
| 17:28.01 | russellb | xheliox: that will get the patch applied faster |
| 17:29.07 | xheliox | russellb: I was just doing that... now I want to figure out how to enable the high res timer on my Centos kernels. ;) Because that's bad ass. |
| 17:29.47 | [TK]D-Fender | hrmphh: nothing about them I don't like.... well... personally they support a power brick and I guess I technically would ratehr pay a little more on EVERY phone just to a;ways have it. |
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| 17:32.19 | hrmphh | yeah i guess im fucked if poe switch dies |
| 17:32.21 | hrmphh | single point of failure |
| 17:32.27 | hrmphh | should prob grab a few bricks |
| 17:32.52 | [TK]D-Fender | hrmphh: Keep an extra around, but PoE seriosly simplifies your wiring, not having to have a brick at each desk. |
| 17:33.11 | [TK]D-Fender | hrmphh: How many phones are you looking at buying? |
| 17:33.13 | Qwell[] | if the PoE switch dies... |
| 17:33.18 | Qwell[] | then your switch is dead too |
| 17:33.29 | Qwell[] | no amount of power backup stuff is going to fix that |
| 17:33.38 | [TK]D-Fender | hrmphh: And if your non PoE switch dies Everything dies just the same. |
| 17:33.51 | Qwell[] | we learned this the hard way last night |
| 17:34.02 | Qwell[] | all of our PCs are on UPSs, networking stuff is too, etc, etc, etc |
| 17:34.09 | [TK]D-Fender | hrmphh: PoE Allows simple power backup... its value varies on implementation |
| 17:34.10 | hrmphh | umm |
| 17:34.11 | hrmphh | qwell |
| 17:34.12 | Qwell[] | but...the power went out...for...several city blocks :p |
| 17:34.14 | hrmphh | we have other switches |
| 17:34.16 | hrmphh | that was the point |
| 17:34.19 | hrmphh | just not backup poe |
| 17:34.30 | hrmphh | qwell; buy a generator |
| 17:34.40 | Qwell[] | isn't gonna help us keep the internet link up |
| 17:34.48 | rrittenhouse | I was wondering if the POE is something I wanted too... not sure |
| 17:34.53 | Qwell[] | everything was still up, except that.. which is completely out of our control |
| 17:34.54 | Sweeper | GSM modem is the way to go Qwell[] |
| 17:34.57 | Sweeper | :D |
| 17:35.04 | Qwell[] | Sweeper: and if the cell tower dies too? |
| 17:35.06 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 17:35.12 | Sweeper | Qwell[]: bigger antennas |
| 17:35.15 | Qwell[] | heh |
| 17:35.25 | hrmphh | shrug |
| 17:35.26 | Qwell[] | pedal crank, right? |
| 17:35.28 | Sweeper | I could also sell you a nice SCPC link |
| 17:35.28 | hrmphh | we have towerstream |
| 17:35.28 | hrmphh | for backup |
| 17:35.29 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:35.57 | Sweeper | only $1500 a month for 500k/500k :D |
| 17:36.00 | hrmphh | T1, ISDN BRI, POTS, VZW, TowerStream |
| 17:36.03 | hrmphh | heh |
| 17:36.08 | hrmphh | 512k towerstream is $225/mo |
| 17:36.09 | hrmphh | not bad |
| 17:36.26 | hrmphh | full 1.5Mbps is only $380/mo |
| 17:36.40 | Sweeper | that is pretty decent |
| 17:36.46 | Sweeper | pretty stable? |
| 17:37.35 | hrmphh | yes |
| 17:37.37 | hrmphh | very |
| 17:37.46 | hrmphh | you can get 1.5-3burstable for like $450/mo too |
| 17:37.53 | hrmphh | question on the polycoms |
| 17:37.56 | hrmphh | can you do blind transfer? |
| 17:38.23 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:42.24 | hmmhesays | I'm bringing sexy back |
| 17:43.18 | [TK]D-Fender | hrmphh: Yes, blind and attended. You can also be on a 3-way call, and hang up leaving the other 2 connected |
| 17:46.37 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 17:46.57 | shay|work | hello, I'm trying to make a call to the PSTN via a Zap channel, and asterisk debugging shows this: |
| 17:46.58 | shay|work | Aug 14 23:51:35 NOTICE[4949]: app_dial.c:1097 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 17:47.50 | *** join/#asterisk anthm][ (n=anthm@66.175.194.26) |
| 17:48.06 | *** join/#asterisk AndyGraybeal (n=andy@casanueva.wifi.frognet.net) |
| 17:48.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:49.15 | AndyGraybeal | hmm.. another quick question... can i hook asterisk up to a calendar program that my workgroup would use.. something like OpenGroupware or eGroupware, and when my office is closed on the calendar, can i have the phone system use our "closed for now" message? |
| 17:49.57 | AndyGraybeal | so none of us would be able to forget to set the phone sysetm to answer the calls after one ring, and then let the caller know we're closed for the day |
| 17:50.15 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 17:50.24 | Mercestes | Not to be a troll but why is ABE still on 1.2? |
| 17:50.49 | hmmhesays | ABE? |
| 17:51.03 | Mercestes | Asterisk Business Edition. |
| 17:51.36 | puzzled | dunno. maybe still too many bugs in 1.4? |
| 17:55.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@79.179.135.2) |
| 17:57.08 | hmmhesays | holy crap brian setzer is awesome |
| 17:58.51 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-2f0951b79ebfae6d) |
| 17:59.44 | [TK]D-Fender | AndyGraybeal: Sure |
| 18:00.58 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:02.37 | sopo2k4 | anyone know whats wrong with this |
| 18:02.38 | sopo2k4 | http://pastebin.com/m402d339d |
| 18:03.43 | *** part/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 18:06.50 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 18:07.58 | sopo2k4 | im using the asterisk manager to use the command Originate, and i am parsing the variable user=managerusername but for some reason the script above isnt doing any of the extensions after the Dial. |
| 18:09.00 | Mercestes | well, in under 5 seconds, I would say, never use _. |
| 18:09.27 | Mercestes | second, none of that crap under dial is ever going to get called. |
| 18:10.11 | sopo2k4 | whys that? :s |
| 18:11.04 | Mercestes | well, for one....once you Dial, Asterisk stays on dial until you hang up. |
| 18:11.31 | Mercestes | second, once you hang up, asterisk isn't going to continue processing yoru dialplan. It will go to exten => h, but it will not continue down it's current pattern match |
| 18:11.51 | Olobola | 's' extension is not working for some reason in my default context. Calls are rejected: '1234567890@default' does not exist. |
| 18:11.58 | Mercestes | which......btw, is why you should never use _., because that matches exten => h, and s, and fax, and i, and t, and o. |
| 18:12.46 | Mercestes | Olobola, s != 123456789. Try _x. instead. or _xxxxxxxxx or whatever. |
| 18:13.09 | Mercestes | Olobola, s is for macros and gotos, not pattern matching. |
| 18:13.12 | *** join/#asterisk souzha (n=IceChat7@static-72-72-83-224.bstnma.east.verizon.net) |
| 18:13.36 | *** join/#asterisk vinafera (n=pourritu@mail.cshorecomputing.com) |
| 18:13.49 | sopo2k4 | so if i want whats under Dial to be processed when the call is finished |
| 18:13.59 | sopo2k4 | id use exten => h, |
| 18:14.00 | sopo2k4 | ? |
| 18:14.04 | Mercestes | sopo2k4, very good. |
| 18:14.15 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 18:14.31 | whywontitwork | anyone know where one can find detailed information about asterisk and faxing using Spansp? |
| 18:14.46 | sopo2k4 | cheers |
| 18:14.50 | sopo2k4 | also, one more thing |
| 18:14.54 | sopo2k4 | DBGet(foo=family/key) - would work in 1.4.9? |
| 18:15.14 | Mercestes | I honestly don't know. I haven't developed my DB-fu yet. |
| 18:15.19 | sopo2k4 | ok |
| 18:15.37 | Mercestes | whywontitwork, google FoIP, asterfax, fax detection, asterisk fax, etc. |
| 18:16.17 | Mercestes | whywontitwork, there is a wiki page on the subject listing various methods for accomplishing FoIP, including asterisk+iaxmodem |
| 18:16.17 | souzha | Hey I just bought 5x 7940s for my asterisks system, and 3 of the phones, when I plug in the power then subsequently plugin in the ethernet into the 10/100 SW port, the phone shuts down, anyone else heard of this? |
| 18:16.31 | Mercestes | souzha, where did you buy them from? |
| 18:16.39 | souzha | where else...ebay :) |
| 18:16.49 | souzha | I'm assuming they were POE |
| 18:16.55 | Mercestes | chances are your screwed. |
| 18:17.05 | Mercestes | s/your/you're/ |
| 18:17.08 | Qwell[] | s/POE/DoA/ |
| 18:17.27 | Mercestes | One: You have no warranty or support. |
| 18:17.31 | souzha | really, so there is no jumper setting or anything like that for these phones |
| 18:17.39 | Qwell[] | souzha: no, it's all "automagic" |
| 18:17.47 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 18:17.49 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 18:17.51 | Qwell[] | If your switch is trying to provide power or something, it may be freaking the phone out |
| 18:17.51 | Mercestes | two: You have to have *specific* firmware software in your tftp server to setup those phones which you can only get if you buy the phones from Cisco, there is no guarantee those phones even speak to each other. |
| 18:17.58 | whywontitwork | check your ip addresses souzha |
| 18:18.12 | souzha | well I got one of them up fine |
| 18:18.22 | souzha | granted it was a pain in the ass, went through 5 different firmwares |
| 18:18.23 | Mercestes | souzha, ...oh, that must mean the other 3 are fine too. |
| 18:18.31 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 18:18.45 | Qwell[] | souzha: send them here, I'll give them all a good home |
| 18:18.49 | souzha | haha |
| 18:19.07 | souzha | the thing is in the shipped version, the 10/100 PC port works fine |
| 18:19.07 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 18:19.21 | souzha | then I upgrade to SIP |
| 18:19.31 | souzha | and SIP doesn't recognize the port |
| 18:19.41 | Olobola | Mercestes: thanks. The lumenvox 'pizza' demo is looking for extensions s though. |
| 18:19.44 | Qwell[] | sure sounds like Cisco |
| 18:22.02 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:22.09 | souzha | So I'm feeling your recommendation would be to keep resetting the phones and praying |
| 18:23.01 | Qwell[] | well, they do seem to take a random code path on boot |
| 18:23.03 | Mercestes | Actually, my recommendation was to remove yourself from the gene-pool by electrocuting yourself with those 60v power adapters. |
| 18:23.04 | Qwell[] | one of them may work :p |
| 18:23.26 | souzha | thank god they're only 48v |
| 18:23.33 | Mercestes | nothing personal..just...it would help us escalate the average intelligence of the planet by removing all those dumb enough to order Cisco phones off of Ebay |
| 18:23.34 | Yourname` | Hello. Using SpanDSP and Asterisk, I was wondering how can we effectively send out faxes? (All documents on the web seem to be talking about the rcving capabilities.. not sending.) |
| 18:23.59 | souzha | well I did this instead of paying 7 grand for fonality |
| 18:24.20 | Mercestes | ~cheapskate |
| 18:24.24 | Mercestes | ~cheap |
| 18:24.25 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 18:24.30 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 18:24.40 | Mercestes | ~phones |
| 18:24.41 | jbot | rumour has it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 18:26.15 | J4k3 | heh, I love how everybody talks shit on grandstream |
| 18:26.23 | Qwell[] | for good reason |
| 18:26.31 | J4k3 | grandstream sells the best $32 ethernet phone money can buy! |
| 18:26.33 | robl^ | cuz they suck (and not in a good way) |
| 18:26.53 | [TK]D-Fender | souzha: 7940 = Cisco PoE, not 802.3af |
| 18:26.58 | Qwell[] | J4k3: I'll take a $3 analog phone over a grandstream |
| 18:27.02 | [TK]D-Fender | souzha: overpriced & trouble |
| 18:27.20 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 18:27.36 | J4k3 | Qwell[]: eh... considering how badly ATAs work at best, a $3 shit phone attached to one sounds like a total fucking nightmare. |
| 18:27.56 | Mercestes | woah, fbomb. |
| 18:28.00 | J4k3 | now, given the right garage sale... $3 might get you a chocolate brown IBM desk phone in perfect condition too ;) |
| 18:28.21 | J4k3 | chocolate brown IBM desk phone >>>>>>> grandsuck |
| 18:28.23 | robl^ | I have a grandstream phone.. I use it when my 5 yr old cousin wants to talk to Elmo. |
| 18:28.28 | Mercestes | J4k3, That's like saying that ........Enron sold the best $32 dollar penny stocks money could buy |
| 18:28.38 | Qwell[] | Mercestes: nice |
| 18:29.04 | J4k3 | well, the most ironic part about that list is the absolute love for polycom, and polycom can't even be bothered to throw a flash chip in their phones. |
| 18:29.37 | J4k3 | I picked up one of these grandsuck phones on my desk, took it to my friends house, plugged it into his DSL router and had an extension in another town with absolutely zero re-provisioning. |
| 18:29.42 | Mercestes | Who cares? It works. |
| 18:29.55 | J4k3 | it only works if you carry a server with you. |
| 18:30.05 | Mercestes | Uh, bs.... |
| 18:30.07 | robl^ | wrong. |
| 18:30.24 | robl^ | Polycom doesn't NEED a server |
| 18:30.43 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:30.44 | robl^ | it always checks for a server for updates, but boots from flash if it doesn't find one |
| 18:31.08 | robl^ | it just reuses the last provisioning |
| 18:31.25 | [TK]D-Fender | yup |
| 18:32.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:32.49 | robl^ | grandstream is the only phone that needs to be strapped down to a desk so it doesn't fly away if the user sneezes |
| 18:33.00 | [hC] | Why would asterisk return CONGESTION after a dial attempt came back "circuit-busy" (because the number was busy) |
| 18:33.18 | [hC] | When people dial busy numbers it always comes back as congestion instead of busy.. |
| 18:33.22 | J4k3 | robl^: well, I can buy metal locally a lot cheaper than I can ship dead weight in a box via ups or fedex. |
| 18:33.35 | [TK]D-Fender | [hC]: I've seena lot of Ni1 setups that do that. Its normal. |
| 18:33.53 | J4k3 | robl^: and you'll find electronic devices that oddly weigh a lot usually have metal manufacturered into them that has absolutely no technical value. |
| 18:34.38 | Mercestes | J4k3, so you prefer to add your own metal, I guess is what yoru saying? |
| 18:34.38 | Mercestes | because its' cheaper? |
| 18:34.38 | [TK]D-Fender | robl^: Actually the Linksys ones are too lingth, as is the Aastra 5i series (the hand set has NO weight I swear) |
| 18:34.38 | [hC] | [TK]D-Fender: so, my ideal way of handling congestion is to retry on another peer, where as busy i want to stop right away and play busy... From what I described, theres no proper way to handle this, unless i just play Busy if all peers come back congested |
| 18:34.38 | Mercestes | J4k3, I have something for you to read. |
| 18:34.38 | Mercestes | ~cheap |
| 18:34.39 | jbot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 18:34.39 | J4k3 | Mercestes: I'd rather have a nice dual 4" ringer inside. |
| 18:34.48 | J4k3 | heh |
| 18:34.59 | [TK]D-Fender | [hC]: that IS what you have to do. |
| 18:35.04 | [hC] | [TK]D-Fender: 10-4. |
| 18:35.13 | J4k3 | you mean don't buy grandstream phones and crank up an old P3-700 you found in random pieces in the closet? |
| 18:35.34 | J4k3 | dammit! I got cought doing it wrong! |
| 18:35.54 | [TK]D-Fender | J4k3: No.... that PC from the closet is already yours and clearly worth much MORE than the phones ;) |
| 18:36.04 | [TK]D-Fender | J4k3: So failure only costs a little time :)\ |
| 18:36.12 | robl^ | oops. sorry. burst of emotion |
| 18:36.31 | J4k3 | robl^: you're paying an ilec for a pri? I'm sorry dude :P |
| 18:36.32 | J4k3 | hehe |
| 18:37.01 | robl^ | J4k3: nope! My employer is paying the ILEC. |
| 18:37.22 | robl^ | I was not / am not the decision maker |
| 18:37.39 | J4k3 | well, L3's sales department sucks. |
| 18:38.00 | denon | no, its really all of L3 that sucks |
| 18:38.42 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 18:39.10 | rrittenhouse | what happened to the xlite softphone? I cant seem to dl it for linux anymore |
| 18:39.35 | robl^ | I have 2 PRIs here that seem to have a hatred of dialing one specific country code -- of course its a country where we have about 100 clients. AT&T can't seem to figure it out after about 3-4 months, 6 Level 3 support tickets, about 90 calls to support |
| 18:39.50 | rrittenhouse | ekiga wont play my audio for a call (idk whats up with that) |
| 18:40.33 | J4k3 | robl^: eh... find a voip provider in that country and buy some IP-based minutes? (you think AT&T is doing any better these days? You're fooling yourself!) |
| 18:41.16 | hmmhesays | counterpath.com ? |
| 18:41.41 | rrittenhouse | i was looking on the site and it kept going back to windows and macosx editions |
| 18:41.42 | robl^ | J4k3: already ahead of you. we are doing VoIP migration firmwide in a few months. but until then, I am not a happy camper. |
| 18:42.10 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 18:42.58 | Olobola | trying to get the lumenvox pizza demo running: sent into invalid extension 's' in context 'pizza'. Here is pizza: exten => s,1,Answer |
| 18:44.54 | hmmhesays | port your number somewhere else |
| 18:45.01 | J4k3 | hmmhesays: no porting here. |
| 18:45.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:45.10 | hmmhesays | where is here? |
| 18:45.15 | J4k3 | hmmhesays: Grapeland, TX |
| 18:45.17 | J4k3 | 936-687 |
| 18:45.20 | hmmhesays | is that legal? |
| 18:45.23 | J4k3 | yes. |
| 18:45.53 | J4k3 | I can't port my vzw numbers either (which are just 100-blocks split out another ILEC exchange, 936-546) |
| 18:46.14 | J4k3 | and the worst joke out of all that is the 911 coverage :P |
| 18:46.22 | robl^ | FCC mandates porting for all US numbers, I thought |
| 18:46.29 | J4k3 | the local dispatch office is lucky to get caller ID |
| 18:46.43 | rrittenhouse | So whats a good linux softphone (running ubuntu) |
| 18:46.44 | J4k3 | robl^: for larger telcos you're right... |
| 18:46.57 | J4k3 | robl^: like everything the FCC has always done, theres exceptions. |
| 18:47.02 | robl^ | and Verizon isn't large? |
| 18:47.16 | J4k3 | Verizon doesn't own my number, Windstream owns the numbers. |
| 18:48.30 | ivrc | Have an issue dialing out from * on a TDM400P. The call is delivered to the PSTN, the far side rings, is picked up, but * does not detect the answer. Inbound calls work fine. Suggestions will be greatly appreciated! |
| 18:48.47 | J4k3 | Windstream is one of those craptacular rural telcos that operate under a few zillion exceptions. |
| 18:49.06 | J4k3 | I mean hell, they only got ISDN BRI here in '01. |
| 18:51.31 | Yourname` | So, no experts on faxing capabilities? :( |
| 18:51.41 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 18:53.14 | *** join/#asterisk alejo_g (n=Cumplea_@200.125.83.232) |
| 18:54.10 | [TK]D-Fender | Yourname`: Hylafax + analog line |
| 18:54.29 | [TK]D-Fender | Yourname`: With a seperate modem/fax board that it supports. |
| 18:54.41 | Yourname` | [TK]D-Fender: SIP/VoIP/Asterisk/SpanDSP? |
| 18:54.50 | [TK]D-Fender | Yourname`: No. |
| 18:54.52 | robl^ | why is it everyone wants to run fax lines through a PBX? |
| 18:55.03 | [TK]D-Fender | robl^: people are cheap |
| 18:55.09 | [TK]D-Fender | s/cheap/stupid/ |
| 18:55.27 | robl^ | yeah. but the money you save is lost in the time and effort it takes to make it almost sorta work |
| 18:55.28 | [TK]D-Fender | s/stupid/cheap AND stupid/ |
| 18:55.35 | *** join/#asterisk Itiliti (n=rneubaue@74.7.36.189) |
| 18:56.05 | Itiliti | \ |
| 18:56.45 | Itiliti | I am running Asterisk 1.4.6 and I am getting a weird error when I am trying to do a directed call pickup. |
| 18:56.57 | Itiliti | I get this error: app_directed_pickup.c: No target channel found |
| 18:57.16 | Itiliti | I have all the extentions in the same pickupgroup |
| 18:59.00 | Itiliti | Do I have to add the SIP trunks I am using into the same group as well..... |
| 18:59.01 | Itiliti | ? |
| 18:59.19 | Yourname` | [TK]D-Fender: Hylafax + analog line + Asterisk? |
| 18:59.40 | [TK]D-Fender | Yourname`: What part of "leave it as far away from *" are you not getting? |
| 18:59.44 | [TK]D-Fender | :) |
| 19:00.14 | Yourname` | robl^: Help me figure out a way to send more than 30+ faxes per minute. Without having to spend $20,000. |
| 19:00.49 | Yourname` | [TK]D-Fender: lol, challenges are a good thing though! SpanDSP is a complete fax package.. and not being able to use it to send faxes is frustrating. :( |
| 19:02.54 | [TK]D-Fender | Yourname`: Feel free to suffer as much as your masochism permits. Just leave us out of it. I've smiles like yours dripping off the faces' of others just like the end-scenes from "Indianna Jones & the Raiders of the Lost Ark" |
| 19:03.24 | Yourname` | hahaha |
| 19:03.32 | alejo_g | hello im from argentina |
| 19:03.37 | Yourname` | You should be the headmaster of some school, you'd be the best! |
| 19:03.39 | Yourname` | lol |
| 19:03.41 | hmmhesays | hello from argentina |
| 19:03.45 | alejo_g | i made some changes to AGENT function to manage de devstate |
| 19:03.56 | alejo_g | for the case when the agent is making an outbound call and we dont want to recive calls from asterisk acd |
| 19:04.03 | alejo_g | and i want to know which is de right place to publish them, somebody can tell me? |
| 19:04.44 | [TK]D-Fender | ok, GTG, back significantly later.... |
| 19:05.07 | alejo_g | sorry by my poor english :) |
| 19:05.20 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 19:05.32 | elixer | howdy |
| 19:05.42 | elixer | we just got the digium TE220 PCIe card |
| 19:06.09 | elixer | based on the docs, should we just pull off all the jumpers to get t1 mode? |
| 19:06.30 | elixer | its hard to tell what is going on in the photo in the docs |
| 19:08.50 | russellb | elixer: there should be a single jumper for T1/E1 mode |
| 19:08.57 | russellb | not connecting the jumper means T1 mode |
| 19:09.04 | russellb | which is probably how it is shipped by default |
| 19:09.14 | elixer | excellent |
| 19:09.16 | elixer | thank you sir |
| 19:09.48 | russellb | you're welcome |
| 19:10.00 | russellb | thanks for supporting digium :) |
| 19:17.02 | *** join/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 19:17.09 | whywontitwork | tk is that new book out yet? |
| 19:17.25 | *** part/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 19:18.11 | ivrc | <PROTECTED> |
| 19:23.06 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 19:23.08 | elixer | russellb: you're welcome. now make my sangoma A400 card work with the TE220 ;-) |
| 19:24.27 | Olobola | I just need to place automated reminder calls through a pots line. Which card would be best? |
| 19:27.18 | russellb | pfft .. |
| 19:27.26 | russellb | :-p |
| 19:27.38 | elixer | seriously |
| 19:27.45 | elixer | i can't seem to get the two to work together |
| 19:28.24 | russellb | don't look at me :) |
| 19:28.37 | russellb | Olobola: how many pots lines? |
| 19:28.52 | elixer | from the user manual, it says to ask for additional help in #asterisk ;-) |
| 19:28.56 | denon | russellb: you know, digium could probably make pretty good money writing commercial drives for sagnoma cards |
| 19:29.07 | denon | I hear more than my fair share of sagnoma whining in here |
| 19:29.11 | denon | er sangoma |
| 19:29.27 | denon | drivers ... sheesh. |
| 19:29.38 | russellb | denon: don't hold your breath :) |
| 19:29.42 | Olobola | russellb: just one for now. I might need more someday. |
| 19:29.47 | denon | could clear more cash on the driver than the card :) |
| 19:30.10 | russellb | Olobola: TDM400P for 4 ports TDM800P for 8, depending on how much you think you'll want to expand (disclaimer, I work for Digium) |
| 19:31.11 | Olobola | russellb: ok, thanks. I'll be sure to tell'm russelb sent me. |
| 19:31.23 | russellb | heh, alright :) |
| 19:33.15 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 19:33.23 | AJaymn | Anyone using Fedora Core 6 with Asterisk? |
| 19:33.44 | *** join/#asterisk mxmasster (n=mxmasste@207.171.12.109) |
| 19:33.48 | mxmasster | hi all |
| 19:34.03 | *** join/#asterisk sacitec (n=tobi@189.129.221.82) |
| 19:34.07 | sacitec | hi |
| 19:34.07 | karleeto | russellb: i've wanted to talk to someone who works for digium for quite some time! |
| 19:34.14 | mxmasster | we have a _perfectly_ working asterisk phone system setup and there is a feature request that is driving me crazy |
| 19:34.15 | rrittenhouse | so has anybody else noticed that counterpath's xlite softphone has disappeared? |
| 19:34.22 | rrittenhouse | or is it just me? ;) |
| 19:34.28 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 19:34.31 | mxmasster | specifically the executives want shared line apearances with their assitants |
| 19:34.37 | sacitec | does anybody has troubles with zap volume level ? i'm using sangoma A200 2 fxo ports |
| 19:34.38 | AJaymn | its still there.. but GSM isnt in there codex list anymore!!! |
| 19:34.39 | karleeto | russellb: i was having some echo problems driving me crazy for a while.. i finally got the HPEC from digium and it solved the problem WONDERFULLY! |
| 19:34.53 | russellb | karleeto: that's great! |
| 19:35.00 | karleeto | russellb: have you had primarily positive feedback for HPEC? |
| 19:35.04 | russellb | yes |
| 19:35.08 | denon | ... |
| 19:35.08 | mxmasster | what we are seeing is when the line is configured on the assitant's phone it rings, but if either of them put the call on hold the other person cannot pick it up |
| 19:35.12 | denon | not from everyone :) |
| 19:35.14 | russellb | for the most part |
| 19:35.21 | russellb | except for a few things that were broken at first :) |
| 19:35.35 | denon | well .. last I checked.. |
| 19:35.52 | sacitec | does anybody has troubles with zap volume level ? i'm using sangoma A200 2 fxo ports |
| 19:35.55 | russellb | heh, i'm not in charge of that product |
| 19:35.58 | russellb | i'm not in the know. |
| 19:36.01 | robl^ | mxmasster: you have to be using 1.4.x for SLA -- and even then its still complicated and doesn't scale very well |
| 19:36.01 | karleeto | russellb: well, its good to know you'll be around in here next time i have some digium questions.. we've got 4 VOIP setups and counting, all with digium cards |
| 19:36.14 | karleeto | russellb: and we like them so far, so i suspect we'll keep using them |
| 19:36.16 | *** join/#asterisk guillote_GNU (n=bancaria@host128.201-253-17.telecom.net.ar) |
| 19:36.16 | russellb | karleeto: cool. there are a good number of digium people in here |
| 19:36.43 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 19:36.43 | karleeto | russellb: you'd be surprised at how much negativity i got from people when i was trying to solve that echo problem! |
| 19:37.01 | denon | echo problems bring out the worst in people |
| 19:37.04 | russellb | Corydon76, Cresl1n, jcolp, lmadsen (kind of :-p), Qwell[], codefreeze ...... |
| 19:37.14 | Qwell[] | what? |
| 19:37.17 | russellb | nothing :) |
| 19:37.21 | robl^ | echo bad!!!! |
| 19:37.25 | russellb | naming off digium people in the channel |
| 19:37.36 | Qwell[] | You missed a bunch :p |
| 19:38.06 | russellb | hence the "....." |
| 19:39.29 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 19:40.51 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 19:41.27 | bkruse | echo does not bring out the worst in me, but then again, i do not have to try to debug it :/ |
| 19:43.58 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 19:43.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:44.19 | *** join/#asterisk davidm870 (n=davidm87@66.150.12.150) |
| 19:47.49 | *** join/#asterisk ivanfm_ (n=ivanfm@c906b486.virtua.com.br) |
| 19:47.52 | *** part/#asterisk mxmasster (n=mxmasste@207.171.12.109) |
| 19:48.35 | Itiliti | Anyone know why I would be getting this error: app_directed_pickup.c: No target channel found |
| 19:53.20 | elixer | so i just installed a TE220, and this is what `lspci` is showing |
| 19:53.22 | elixer | 10:08.0 Communication controller: Digium, Inc. Unknown device 0220 (rev 02) |
| 19:53.33 | elixer | is that correct? the "Unknown device" bit? |
| 19:54.51 | elixer | (yes it is. sorry.) |
| 20:05.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:08.15 | *** join/#asterisk Ebola (n=Ebola@host86-139-49-76.range86-139.btcentralplus.com) |
| 20:09.09 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 20:10.18 | magic_hat | Hey everyone, I have a dialplan that works for domestic calls... exten => _1XXXXXXXXXX,2,DIAL(SIP/teliax/${EXTEN}). All of the sudden I need to make an emergency international call. What's the best way to modify that so it accepts 011 followed by a variable-length string of additional numbers? |
| 20:15.12 | magic_hat | hello? |
| 20:15.36 | russellb | _011.,... |
| 20:16.28 | russellb | '.' matches one or more of anything |
| 20:16.28 | lmadsen | russellb: kind of eh? :) |
| 20:16.48 | russellb | lmadsen: you contractors :-p |
| 20:17.01 | lmadsen | russellb: I didn't see you put 'kind of' after jcolp :D |
| 20:17.03 | russellb | basically, i was trolling. |
| 20:17.07 | lmadsen | troll!!! |
| 20:17.08 | russellb | i know, i just realized that |
| 20:17.11 | lmadsen | lol |
| 20:17.15 | russellb | so i got called out. |
| 20:17.16 | lmadsen | it's ok... I don't hate you... |
| 20:17.17 | lmadsen | that much |
| 20:17.23 | russellb | right. |
| 20:17.28 | jcolp | russellb: that's sweet that you didn't think of me as one until after <3 |
| 20:17.29 | lmadsen | hehehe |
| 20:17.33 | russellb | i was wondering if/when you'd notice :-p |
| 20:21.30 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-405f7c4f871c6288) |
| 20:21.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:22.17 | *** join/#asterisk CVirus (n=GoD@82.201.222.217) |
| 20:23.28 | elixer | so what should startup first? zaptel or wanrouter? |
| 20:23.29 | VOiCi | hey, is there any decent way of removing a module from the tdm400 |
| 20:23.36 | VOiCi | without destroying it.. |
| 20:24.12 | Qwell[] | VOiCi: just pull it off slowly and straight... |
| 20:25.24 | Qwell[] | (easier said than done, right?) |
| 20:25.44 | VOiCi | haha |
| 20:25.45 | VOiCi | true |
| 20:25.47 | VOiCi | but im done |
| 20:29.38 | sevard | Qwell[]: do you know of a method/application/dialplan like chanspy with whisper mode w/out using * 1.4? |
| 20:31.28 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 20:31.49 | sevard | Or anyone for that matter? |
| 20:37.15 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 20:39.53 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-189-7-234.lsanca.fios.verizon.net) |
| 20:40.00 | dahunter3 | Has anyone used Voip street? |
| 20:40.56 | De_Mon | sevard no, 1.4 is the first time anything like whisper mode has existed in * |
| 20:45.31 | *** join/#asterisk jkrueger (n=jkrueger@ip66-104-41-250.z41-104-66.customer.algx.net) |
| 20:46.03 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.169.196) |
| 20:46.41 | sevard | De_Mon: and even whisper mode you can only talk to one person at a time, right? you can't speak to the bridged channel |
| 20:46.49 | sevard | unless you pipe all channels through a meetme channe |
| 20:46.50 | sevard | l |
| 20:47.30 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.79.22) |
| 20:47.49 | AJaymn | Timeout error occurred trying to start MySQL Daemon. |
| 20:47.49 | AJaymn | Starting MySQL: [FAILED] |
| 20:47.54 | AJaymn | why am i getting this?! :( |
| 20:48.28 | bkruse | AJaymn: #mysql, but look in /var/log/messages and syslog and the mysql folder to see why it died |
| 20:49.08 | AJaymn | cant create PID ? |
| 20:49.27 | bkruse | AJaymn: are you running as root? |
| 20:49.33 | bkruse | rm /var/run/mysqld.pid or whatever ti is |
| 20:49.34 | bkruse | it is * |
| 20:49.46 | bkruse | killall -9 mysql, its probably already running, or a zombie process of it anyways |
| 20:50.05 | AJaymn | no process running |
| 20:50.58 | AJaymn | bkruse no mysqld.pid in run dir. only an empty myslqd dir. |
| 20:51.14 | bkruse | ps aux | grep mysql |
| 20:51.40 | De_Mon | sevard correct whisper only talks to one channel. I believe there is another application that joins a channel to a bridged call... don't ask me what it is tho |
| 20:52.18 | AJaymn | bkruse root |
| 20:52.43 | sevard | De_Mon: would it b ExtendChan? |
| 20:52.47 | sevard | be* |
| 20:52.55 | Dan0maN_Work | can anyone recommend a decent channel bank to use with * and a sangoma card? |
| 20:53.53 | *** part/#asterisk Yoe (n=wouter@samba.grep.be) |
| 20:53.55 | De_Mon | sevard could be, I just noticed it in passing |
| 20:55.15 | sevard | De_Mon: I still haven't heard any recommendations for 1.4 for business purposes, have you had any experience? |
| 20:58.03 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:00.22 | iCEBrkr | Dude. Nickserv is broken |
| 21:01.23 | ZaVoid | lol |
| 21:01.31 | fujin | how so |
| 21:04.14 | iCEBrkr | Firstly, I had a zombie IRC session somewhere/somehow so I couldn't ghost and now i can't ident |
| 21:04.26 | iCEBrkr | I wonder if somehow I have it ignored? |
| 21:04.27 | iCEBrkr | lol |
| 21:04.34 | iCEBrkr | lol |
| 21:04.37 | iCEBrkr | I'm ignoring all notices |
| 21:04.38 | iCEBrkr | pfft |
| 21:04.46 | [hC] | iCEBrkr: noob. get it together. |
| 21:04.50 | iCEBrkr | haha |
| 21:04.57 | iCEBrkr | [hC]: I forgot how to EYE ARE CEE |
| 21:05.20 | *** join/#asterisk lee_is_me (n=chatzill@12-201-103-91.client.mchsi.com) |
| 21:05.20 | [hC] | its easy, just log on to AOL keyword MIRC |
| 21:05.20 | iCEBrkr | [hC]: I'm an ASP monkey now days.. so you know how it goes.. Gotta dumb it down to fit in this shop |
| 21:05.33 | iCEBrkr | think inside the box |
| 21:05.47 | *** join/#asterisk CVirus (n=GoD@82.201.222.217) |
| 21:06.25 | lee_is_me | I'm having a tough time resolving an issue with calls that come in over a zap line where the caller id information is not present...polycom 301's will not pick up the call. Anyone seen this? |
| 21:07.50 | iCEBrkr | oh man |
| 21:07.53 | *** join/#asterisk killfill (n=killfill@201.238.233.3) |
| 21:08.47 | lee_is_me | only "reject" button works... which is bothersome for the client since there is a large group ring in place. The have to go around to each phone and hit "reject" |
| 21:13.27 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 21:14.22 | jstew | Hi, can anyone recommend an honest vendor of asterisk rackmount systems? |
| 21:17.06 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 21:18.31 | [T]ank | is there any sort of command from the AGI or dialplan that will allow me from a php script or similar to do a warm transfer? you know... where instead of just a transfer you first have a 3 way call and then the person who is transferring the call then drops off. |
| 21:18.44 | *** join/#asterisk Mercestes (n=Merceste@71.41.157.70) |
| 21:19.09 | Mercestes | If I wanted to play a recording before connecting an outbound call, (such as, "this call may be recorded") how would I do that? |
| 21:19.37 | [T]ank | Mercestes: try the A option in the dial command |
| 21:19.45 | Mercestes | googling. |
| 21:19.49 | [T]ank | i use that to dial a number and have it play a message. |
| 21:20.09 | hrmphh | http://www.opinionjournal.com/editorial/feature.html?id=110009552 |
| 21:20.18 | Mercestes | Your awesome, thanks |
| 21:20.28 | [T]ank | its something like... Dial(Zap/g1/${EXTEN}||A(wav file goes here)) |
| 21:21.15 | *** join/#asterisk guillote_GNU (n=bancaria@host128.201-253-17.telecom.net.ar) |
| 21:22.19 | *** join/#asterisk ToyMan (n=Stuart@74-38-159-246.dr01.mdtw.ny.frontiernet.net) |
| 21:23.09 | Dan0maN_Work | can anyone recommend a decent channel bank to use with * and a sangoma card? |
| 21:23.51 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 21:25.14 | Olobola | so is it possible to attach to a database from within a dial plan? Would I need to use AGI? I was using agi/php to connect to sql server before. |
| 21:25.20 | *** part/#asterisk crimethinker (n=ircuser@legacy.diamond.org) |
| 21:25.44 | fetcher | Is there a simple way to adjust music-on-hold volume, without replacing the .MP3 files? |
| 21:29.30 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 21:33.23 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 21:33.30 | whywontitwork | me again |
| 21:33.57 | whywontitwork | music on hold ??? how does one play music on hold? |
| 21:34.52 | sevard | FOURTY-TWO?? |
| 21:35.10 | sevard | whywontitwork: configure the conf file, reload moh, and put somebody on hold. |
| 21:35.56 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 21:36.47 | De_Mon | sevard what do you mean? it works but if your implimentation isn't broke don't fix it |
| 21:36.51 | *** join/#asterisk icel (n=icel@63.78.162.46) |
| 21:37.16 | De_Mon | sevard I upgraded to 1.4 recently because of the whisper mode and func_odbc goodness (yea func_odbc is backported but thats besides the point) |
| 21:37.18 | whywontitwork | link for the manual please |
| 21:37.58 | sevard | De_Mon: I use 1.2 for business installs, i've never even poked at 1.4 because I hear horror stories about it crashing and things. But, a lot of customers have been asking for features that are only available in 1.4 |
| 21:38.05 | lirakis | sigh |
| 21:38.09 | icel | anyone have experience with HA asterisk solutions? I have a couple of generic questions... |
| 21:38.10 | lirakis | informix blows :p |
| 21:38.30 | lirakis | icel: ask |
| 21:38.46 | icel | cool. IT seems like a good idea to use openser? Is that true or false? |
| 21:39.06 | lirakis | icel: thats incredibly generic.. and thus difficult to give a legitimate answer to |
| 21:39.16 | lirakis | icel: most HA situations use a load balancing prox |
| 21:39.18 | icel | i want to configure 3 servers to load balance and be okay if one goes down |
| 21:39.19 | lirakis | *Y |
| 21:39.29 | lirakis | .. and openser can act as a load balancer |
| 21:39.52 | De_Mon | sevard only problems I've had are with realtime startup option, and that was a problem in 1.2 too |
| 21:40.03 | [T]ank | anyone here used the 'atxfer' feature? |
| 21:40.05 | sevard | De_Mon: What about stability? |
| 21:40.20 | sevard | De_Mon: do you have any large installs with stability issues? |
| 21:40.23 | icel | lirakis: so what in general should i do to set that kind of situation? Do I need a mysql database for voicemail and stuff? |
| 21:40.24 | lirakis | icel: you can configure openser to act as a way to pool servers and have sip clients register to 1 but distribute requiests to your pool |
| 21:40.33 | lirakis | icel: ... |
| 21:40.42 | De_Mon | sevard none yet |
| 21:40.49 | icel | lirakis: ok |
| 21:40.49 | De_Mon | no large installs of 1.4 that is |
| 21:41.38 | icel | lirakis: i don't need real specifics, just general info |
| 21:42.58 | lirakis | icel: ... think about it... |
| 21:43.33 | icel | Well I could load balance with dns or a distribute with openser |
| 21:43.34 | s34n | if I leave the secret out of the sip.conf entry, does that allow authentication without a password, or does it prevent authentication period? |
| 21:43.54 | icel | but I am confused how multiple servers share information, such as voicemail and sip accounts |
| 21:44.09 | lirakis | icel: dns will not load balance... or properly pool servers.. round robin dns will still send calls to a failed server |
| 21:44.11 | ber123 | what do you guys use to look at bandwidth stats from console on linux |
| 21:44.24 | ber123 | i have 3 G729 calls up and am showing 192kbps which seems too high |
| 21:44.29 | ber123 | i was expecting 30kbps per call |
| 21:44.40 | ber123 | (via bwmon is where the stats are coming from) |
| 21:45.00 | lirakis | icel: sip proxy... .. .. its a single registration point for your clients... then only the proxy registers with the server |
| 21:45.11 | whywontitwork | does the mp3 files in your /var/lib/asterisk/mohmp3/ folder do they need to be a special format? |
| 21:45.11 | lirakis | icel: voicemail... http://www.voip-info.org/wiki-Asterisk+voicemail+database |
| 21:45.37 | icel | lirakis: jackpot, thanks |
| 21:45.40 | lirakis | icel: go read about sip proxies and what they do.. |
| 21:46.12 | icel | lirakis: will do. thx |
| 21:46.33 | s34n | I see my phones sending register requests into the * server, but the * server doesn't respond. |
| 21:46.55 | Dan0maN_Work | how would you make sip proxies redundant? |
| 21:47.19 | s34n | shouldn't I see resgister requests at the console with enough verbosity? |
| 21:48.53 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 21:49.03 | *** join/#asterisk RRC (n=Juanito-@201.132.142.100) |
| 21:50.32 | *** join/#asterisk travesty (n=user@ool-43509aa5.dyn.optonline.net) |
| 21:52.10 | travesty | i'm having trouble setting up a PRI. I have two * boxes, one with ast 1.2 and a sangoma card, the other with ast 1.4 a digium card and they both seem configured correctly with no errors... but the red lights are still throbbing. Can anyone point me in the right direction of some troubleshooting steps? |
| 21:53.14 | ber123 | did you read the docs at voip-info for PRI |
| 21:53.29 | travesty | yep, and the sangoma docs |
| 21:53.49 | ber123 | ok if you think you have everything right i would create a loopback to the card |
| 21:53.51 | travesty | they just aren't talking to each other and i have no idea why, or how to figure out why |
| 21:53.57 | ber123 | oh you have one plugged into the other |
| 21:54.10 | ber123 | do you have the cabling crossed over? |
| 21:54.12 | travesty | i'm trying to get the two boxes to talk to each other |
| 21:54.24 | travesty | i used the t1 cable that came with my sangoma card |
| 21:54.28 | travesty | so it should be right |
| 21:54.30 | ber123 | no |
| 21:54.35 | ber123 | i dont believe so |
| 21:54.41 | ber123 | i think most cables would be straight thru |
| 21:54.46 | travesty | let me look at the wires |
| 21:54.49 | ber123 | you need to make sure pairs 1,2 and 4,5 are reversed on the other side |
| 21:54.56 | ber123 | otherwise you are trying to connect TX to TX |
| 21:55.02 | ber123 | which will give you the red alarm |
| 21:55.19 | travesty | hmmm it does appear to be straight thru |
| 21:55.25 | ber123 | theres issue #1 :) |
| 21:55.28 | travesty | is there a reference page for making a cable? |
| 21:55.33 | ber123 | its pretty easy |
| 21:55.43 | ber123 | take the pairs which are 1,2 |
| 21:55.46 | ber123 | they are color coded |
| 21:55.51 | ber123 | and move them to position 4,5 |
| 21:56.00 | ber123 | and take the 4,5 pairs and move them to 1,2 |
| 21:56.04 | travesty | i don't want to break this cable, i want to make a new one |
| 21:56.09 | ber123 | thats fine |
| 21:56.15 | ber123 | just make sure to do that on your new one |
| 21:56.28 | sevard | travesty: just use your teeth |
| 21:56.35 | travesty | so i just make a straight thru cat5e, no matter what color order, and reverse those pairs? |
| 21:56.37 | travesty | sevard: lol |
| 21:56.55 | ber123 | well you need to make sure that 1,2 and 4,5 are twisted |
| 21:57.28 | travesty | so 1,2 -> 5,4 and 4,5 -> 2,1 ? |
| 21:57.31 | ber123 | so i would take for example blue/white blue and orange/white orange in 1,2 4,5 respectively and cross them over as orange/white orange and blue/white blue in 1,2 and 4,5 respectively |
| 21:57.57 | ber123 | no, 1,2 goes to 4,5 and 4,5 goes to 1,2 same order |
| 21:58.07 | ber123 | i've never tried reversing them in the pair order |
| 21:58.16 | ber123 | might still work but my way will definitely work |
| 21:59.25 | travesty | ok! |
| 21:59.34 | travesty | i'll give it a try, thank you so much for your help ber123 |
| 21:59.46 | travesty | <3 |
| 21:59.56 | ber123 | no problem |
| 22:00.19 | *** part/#asterisk travesty (n=user@ool-43509aa5.dyn.optonline.net) |
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| 22:20.04 | Mrtaz | hey all, i have a problem with my ZAP channel, i put in a new card, all was working fine, then I ran fxotune and restarted the server, now asterisk does not appear to have any zap commands in the CLI interface |
| 22:22.15 | tzafrir_laptop | so chan_zap failed to load |
| 22:22.27 | tzafrir_laptop | anything in the logs about chan_zap? |
| 22:22.42 | tzafrir_laptop | /var/log/asterisk/messages |
| 22:23.00 | Mrtaz | checkin |
| 22:25.28 | Mrtaz | says reload was unsucessful |
| 22:26.01 | Mrtaz | hmm...interesting item...Echo training time must be within the range of 10 to 4000 ms at line 10 |
| 22:26.09 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 22:26.40 | whywontitwork | anyone please how do i get a incoming call from zapata.conf to extensions.conf not using the s handler need to use _X.,1,Answer() because i need to read the dtmf digits comming down |
| 22:32.14 | carrar | Something like exten => _X.,1,Set(THEGOODS=${EXTEN}) |
| 22:33.01 | VOiCi | anyone use trixbox and could help me out like 3-4 minutes? |
| 22:33.10 | JT | this isn't a trixbox channel |
| 22:33.11 | lirakis | VOiCi: #trixbox |
| 22:33.12 | Qwell[] | ~trixbox |
| 22:33.21 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 22:33.49 | VOiCi | oh sorry |
| 22:33.49 | VOiCi | my mistake |
| 22:33.49 | VOiCi | thanks :) bot. |
| 22:33.51 | *** join/#asterisk Fulk (i=Fulk@i-83-67-58-126.freedom2surf.net) |
| 22:34.05 | carrar | oh read the dtmf, nm |
| 22:35.12 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:38.19 | *** join/#asterisk ShaunD (i=Shaun@cblmdm72-240-16-15.buckeyecom.net) |
| 22:39.07 | ShaunD | has anyone ever heard of using asterisk for a telephone dating service? |
| 22:40.30 | clyrrad | ShaunD: why not |
| 22:40.45 | Fulk | yeah, can't think of a better platform for helping geeks get laid |
| 22:40.56 | clyrrad | lol |
| 22:41.35 | ShaunD | I was thinking there might be a project out there but I'm not finding anything |
| 22:41.56 | clyrrad | could always make your own |
| 22:41.59 | JT | <PROTECTED> |
| 22:42.01 | Fulk | yeah |
| 22:42.32 | ShaunD | I was hoping it wouldn't come to that ;-/ |
| 22:42.38 | carrar | ShaunD, How about OpenMoko? |
| 22:42.50 | carrar | Chics love things that can vibrate |
| 22:42.56 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:42.57 | guillote_GNU | ShaunD, doing such service shouldnt be that imposible |
| 22:43.00 | neverblue | what is TDM? |
| 22:43.02 | carrar | although, I'm not sure it vibrates |
| 22:43.05 | neverblue | or what does it stand for? |
| 22:43.11 | guillote_GNU | besides there is a lot of documentation |
| 22:43.30 | Fulk | time Division Multiplexing? |
| 22:43.36 | neverblue | ah |
| 22:44.01 | JT | ShaunD: you can always pay a consultant |
| 22:44.02 | clyrrad | neverblue: http://en.wikipedia.org/wiki/Time-division_multiplexing |
| 22:44.24 | Fulk | ShaunD, yeah - get some cheap Eastern Europeans or Indians to do it for you on elancer or similar |
| 22:44.34 | neverblue | thanks clyrrad |
| 22:44.43 | clyrrad | np |
| 22:45.45 | ShaunD | anyone ever use the asterisk perl module? |
| 22:46.35 | *** join/#asterisk bjohnson (n=bjohnson@dsl-67-55-22-51.acanac.net) |
| 22:46.42 | Fulk | nope |
| 22:47.59 | citats | ShaunD: i've used it a few times before |
| 22:48.44 | tzafrir_laptop | citats, any idea if anybody is ever going to get it into CPAN? |
| 22:49.09 | citats | tzafrir_home: it is in CPAN and has been since .09 |
| 22:49.17 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:49.40 | tzafrir_laptop | hmmm...hmmm |
| 22:50.03 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-3268fcbeae12431f) |
| 22:50.14 | tzafrir_laptop | Even better, I guess |
| 22:51.30 | RRC | Anyone here on AsteriskNOW ? |
| 22:53.09 | ShaunD | think I should start a sourceforge project while I'm doing this? |
| 22:53.33 | ShaunD | or will people be assholes because I'm going to use perl? |
| 22:53.48 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 22:53.51 | JT | better than using php |
| 22:54.42 | clyrrad | ShaunD: I dont get it - why would people care if you use perl? |
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| 22:55.58 | Fulk | nowt wrong with PHP |
| 22:56.30 | CaT[tm] | oooh. I feel a language war a-brewin :) |
| 22:56.47 | JT | Fulk: what is "nowt"? |
| 22:56.57 | clyrrad | meh - they all have their place, and ppl gots preferences |
| 22:57.11 | CaT[tm] | jt: neat cattle according to me dictionary :) |
| 22:57.15 | clyrrad | language wars are dumb IMO |
| 22:57.17 | JT | hah |
| 22:57.18 | ShaunD | I'm not sure really, I've just caught a lot of shit for using it, people seem to few it as a sysadmins language now and not so much for making applications |
| 22:57.36 | CaT[tm] | shaund: they be silly. way silly. |
| 22:57.53 | ShaunD | I do everything in perl |
| 22:58.04 | clyrrad | ShaunD: then do it in PHP and please the world :p |
| 22:58.09 | JT | ShaunD: there are some modern languages that may be slightly easier if you ever plan to possibly let someone else see your code :) |
| 22:58.31 | ShaunD | like what? |
| 22:58.42 | tzafrir_laptop | JT, you speak as someone who doesn't really know perl |
| 22:58.47 | ShaunD | I thought perl was about as high level as it got |
| 22:58.56 | JT | tzafrir_laptop: sorry? |
| 22:59.08 | JT | ShaunD: hell no, there's plenty more high level languages |
| 22:59.23 | CaT[tm] | jt: high or higher? |
| 22:59.27 | ShaunD | I know there are more, but what do you mean by easier to understand? |
| 22:59.29 | JT | both |
| 22:59.43 | JT | ShaunD: perl gives you 100 ways to skin a cat |
| 23:00.00 | ShaunD | as it should be |
| 23:00.01 | CaT[tm] | its only high if the cat can be skinned in one way :) |
| 23:00.08 | JT | 90% of those ways are obsfucated and ugly to read, and unfortunately a lot of people who program in perl use these methods |
| 23:00.09 | Fulk | JT, it's Northern British for nothing |
| 23:00.26 | JT | Fulk: sounds like chav slang |
| 23:01.20 | CaT[tm] | shaund: just do it in the language that you feel will give you the best result. |
| 23:01.25 | Fulk | What do you call a chav in a filing cabinet? |
| 23:01.33 | Fulk | sorted |
| 23:01.40 | clyrrad | One of the nice things about PHP is that no matter how good or how bad the coder is - you can pretty much follow the syntax becase its the same |
| 23:01.43 | JT | haha |
| 23:01.53 | ShaunD | heh |
| 23:01.54 | Fulk | what do you call a chav in a locked box? |
| 23:01.57 | Fulk | SAFE |
| 23:02.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:02.49 | Fulk | clyrrad, have you tried any Ruby programming? |
| 23:02.50 | JT | php was never that nice a web page generation system, it should never be considered out side of web pages :P |
| 23:03.05 | clyrrad | Fulk: negative not yet |
| 23:03.13 | Fulk | I hate it |
| 23:03.20 | Fulk | trying to understand other peoples Ruby code is hell |
| 23:03.46 | clyrrad | Fulk: this is why I like PHP, Java and C++, the code all looks the same |
| 23:04.05 | Fulk | I find it very easy to decipher PHP programs |
| 23:04.13 | clyrrad | yup |
| 23:04.18 | Fulk | and Java too, lots of syntactic sugar |
| 23:04.36 | clyrrad | part of the reason its so popular i guess |
| 23:04.36 | Fulk | and Ruby code is rarely commented :-( |
| 23:04.54 | ShaunD | I I've had a lot of people tell me to get into ruby recently |
| 23:05.10 | *** join/#asterisk red9012 (n=marc3234@206-248-160-30.dsl.teksavvy.com) |
| 23:05.11 | red9012 | hi |
| 23:05.15 | ShaunD | werd |
| 23:05.22 | Fulk | ShaunD, I'm getting into Ruby on Rails |
| 23:05.44 | Fulk | I'm missing the availability of a good IDE that Java and PHP have |
| 23:05.47 | red9012 | whats the best term for the 'ringing sound' heard when you dial a number, and wait till you get connected? |
| 23:05.50 | clyrrad | I see no point to learn Ruby yet :p |
| 23:06.07 | clyrrad | red9012: ringing? |
| 23:06.21 | Fulk | clyrrad, Ruby on Rails is very quick at developing CRUD web applications |
| 23:06.23 | JT | red9012: ringing indication |
| 23:06.31 | red9012 | is it 'ringing' or 'ring tone' ? |
| 23:06.56 | clyrrad | red9012: Ring Tone is what YOU hear when somone dials YOU |
| 23:07.01 | SplasPood | Fulk: heh. RoR is good for creating CRUD |
| 23:07.10 | JT | ringing indication |
| 23:07.24 | Fulk | too many application frameworks |
| 23:07.31 | SplasPood | Fulk: If you want things less CRUDy you need another framework ;) |
| 23:07.35 | clyrrad | Kinda like .NOT |
| 23:07.40 | clyrrad | whooops i mean .NET hehehe |
| 23:07.41 | JT | also known as ringback tone |
| 23:07.55 | clyrrad | really I meant .NET hehe |
| 23:07.56 | Fulk | I've been out of the web dev loop for too long |
| 23:07.57 | SplasPood | clyrrad: actually, I hear pretty good things from people RE C#/.Net |
| 23:08.45 | Fulk | my other deciding factor is employability |
| 23:08.48 | denon | c# is actually a pretty nice language to work in .. |
| 23:08.54 | denon | though you won't hear any 'nix diehards say that |
| 23:09.05 | Fulk | using c#/.net or Java is far more employable than Ruby |
| 23:09.14 | clyrrad | agreed |
| 23:09.24 | clyrrad | PHP too |
| 23:09.59 | Fulk | for web dev, yeah |
| 23:10.15 | Fulk | but c# and Java is transferable to other projects |
| 23:10.32 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:12.03 | clyrrad | Well Java has its place, its nice that it can interact in real time between client and server - where as PHP cant |
| 23:13.00 | JT | clyrrad: what do you mean? |
| 23:13.20 | clyrrad | JT: well PHP renders server side and dump HTML code to the client... |
| 23:13.30 | Fulk | and so does Java |
| 23:13.33 | clyrrad | Java can actually interact - for example Java Games |
| 23:13.38 | JT | clyrrad: err |
| 23:13.43 | JT | you're confusing the issue |
| 23:13.45 | Fulk | Java Applets are different from Java server pages |
| 23:13.55 | JT | server side java and client side java are completely seperate things |
| 23:14.07 | JT | and you can interact with other languages |
| 23:14.11 | JT | heard of AJAX? :) |
| 23:14.25 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 23:14.25 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:14.40 | Fulk | but also illustrating my point how Java is tranferable to other projects, i.e Applets, J2EE, Console, mobiles |
| 23:15.27 | clyrrad | JT: yup AJAX is cool! |
| 23:15.35 | JT | and COMET |
| 23:16.01 | Fulk | what's comet? |
| 23:16.08 | clyrrad | have not used / heard of COMET |
| 23:16.20 | Fulk | nope |
| 23:16.36 | JT | it's a similar principle to AJAX |
| 23:16.39 | JT | except |
| 23:16.52 | JT | instead of the client sending a request and getting a reply |
| 23:16.59 | JT | the server pushes data out asynchronously |
| 23:17.03 | JT | in an event driven model |
| 23:17.09 | Fulk | hmm, interesting |
| 23:17.10 | JT | with no java or flash |
| 23:17.14 | clyrrad | so just like a desktop app |
| 23:17.20 | Fulk | I guess that's how Yahoo Finance works |
| 23:17.22 | JT | in web pages |
| 23:17.26 | Fulk | with the dynamically updated stock tickers |
| 23:17.38 | JT | Fulk: well they could also be AJAX |
| 23:17.42 | Fulk | on a timer, yeah |
| 23:17.45 | JT | or java |
| 23:18.03 | Fulk | it's not a Java applet, or flash |
| 23:18.08 | Fulk | I checked the source |
| 23:18.12 | JT | heh |
| 23:18.46 | JT | http://en.wikipedia.org/wiki/Comet_(programming) |
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| 23:19.08 | Fulk | way ahead of you |
| 23:19.14 | Fulk | the ajax/comet irc client is neat |
| 23:19.30 | Fulk | it's basically push ajax |
| 23:19.42 | JT | which is awesome |
| 23:19.48 | JT | as pull sucks |
| 23:19.54 | Fulk | well |
| 23:20.12 | Fulk | what applications can comet fulfill that ajax can't with a timer? |
| 23:20.17 | JT | for anything where the server needs to send you data upon the occurance of an event |
| 23:20.20 | JT | err |
| 23:20.32 | JT | timers to check if something has changed are a waste of bandwidth |
| 23:20.35 | JT | and have a high latency |
| 23:21.14 | JT | comet allows for more efficiency and nearer to realtime reaction speeds |
| 23:21.16 | clyrrad | yup - events are a better choice - since a timer will fire off even if nothing has changed.... makes sense |
| 23:21.20 | Vorondil | Something like meebo would be a good application of comet, then? |
| 23:23.40 | clyrrad | jT: how do you code for COMMET - what language do you do it in? Or is it you can have PHP push it etc... |
| 23:24.17 | Fulk | surely it will require a web server that supports it |
| 23:24.42 | lirakis | later |
| 23:24.44 | *** part/#asterisk lirakis (n=eric@69.24.142.1) |
| 23:24.46 | clyrrad | Fulk: yea im imagine some web dameon that interprets it...... but im curious about the coding / syntax etc |
| 23:27.08 | JT | clyrrad: the language is immaterial |
| 23:27.17 | JT | clyrrad: but having toolkits available does help |
| 23:27.27 | JT | and knowing xml and javascript will help |
| 23:28.51 | clyrrad | jT: assume I know XML and JavaScript, which I do... what else you need to know to make an APP? |
| 23:29.17 | clyrrad | jT: i mean something must push this code down to the client like PHP ? |
| 23:30.07 | JT | yeah, you'll need some sort of serverside language, i'm sure all the major ones can do it |
| 23:30.46 | JT | well, not quite sure how well developed the libraries are yet |
| 23:33.11 | whywontitwork | hi there need some help please? |
| 23:33.53 | tzanger | hmm I learned something about stainless steel today |
| 23:34.03 | tzanger | it's trivial to solder to it if you remove the oxide layer that makes stainless steel famous |
| 23:34.34 | JT | what grade of stainless steel and what solder? :) |
| 23:34.38 | whywontitwork | zaptelhow do i route a call from zapata to extension.conf not using the s handler? |
| 23:35.54 | tzanger | JT: regular old tin/lead solder, although the rohs shit will work too |
| 23:35.59 | tzanger | and what grade of stainless? who knows |
| 23:36.02 | *** join/#asterisk sacitec (n=tobi@189.129.221.82) |
| 23:36.08 | *** part/#asterisk sacitec (n=tobi@189.129.221.82) |
| 23:36.53 | JT | tzanger: good luck soldering to 316L(VM) stainless |
| 23:37.06 | JT | most stainless steels are more open to brazing or TIG welding |
| 23:37.08 | tzanger | I don't think that's what this is |
| 23:37.14 | tzanger | this is an electrical connection |
| 23:37.36 | JT | i've never seen stainless electric wires |
| 23:37.42 | JT | steel isn't a good conductor |
| 23:38.24 | tzanger | it is for what I want |
| 23:38.34 | tzanger | just a low-current ground connection to earth |
| 23:38.45 | tzanger | it's also going through a spring made of god-knows-what |
| 23:38.53 | tzanger | but the stainless content may make this impossible for that purpose |
| 23:38.54 | JT | where does the stainless steel wire go? |
| 23:39.00 | tzanger | since the oxide layer will insulate it |
| 23:39.06 | tzanger | but it will turn into a capacitor then |
| 23:39.14 | tzanger | and I'll have an excellent ac ground for static transients |
| 23:39.17 | tzanger | testing is required :-) |
| 23:39.23 | tzanger | it's not stainless steel wire |
| 23:39.39 | tzanger | it's a ground connection (which goes to a spring whcih contacts the DIN rail which is earthed) |
| 23:39.52 | JT | which bit is stainless? |
| 23:40.12 | tzanger | the little metal stamped piece which is inserted and soldered ot the PC board and which the spring rests against |
| 23:40.29 | JT | i doubt it's stainless |
| 23:40.33 | tzanger | it is |
| 23:40.50 | tzanger | they were originally giving me regular tinplate but the steel was melting too badly with their laser |
| 23:40.56 | tzanger | so they called and asked if they could use stainless instead |
| 23:41.03 | JT | i see |
| 23:41.18 | tzanger | I wanted tin plate but I guess they can't do it with the laser |
| 23:41.26 | tzanger | we're looking at getting a die made to punch these out instead |
| 23:41.31 | tzanger | then we can use cheaper metal |
| 23:41.38 | elixer | i'm really not impressed with sangoma right now. everytime i try to shutdown i get a kernel panic. |
| 23:41.42 | elixer | and i am ready to cry. |
| 23:41.50 | elixer | heh |
| 23:41.52 | Fulk | elixer, I had that |
| 23:41.52 | tzanger | elixer: don't shut down your PBX; how do you intend to take calls? |
| 23:42.02 | Fulk | cna't remember how I fixed it - I think I changed the init.d ordering |
| 23:42.22 | elixer | Fulk: i've tried every possible combination. zaptel first, wanrouter first, etc. |
| 23:42.24 | elixer | no love |
| 23:42.30 | elixer | tzanger: you make a good poinnt. |
| 23:42.38 | tzanger | that's why I'm here :-) |
| 23:42.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:42.58 | lesouvage | Whywonitwork: context=context of tour choice in zapata and context of your choice in extensions.conf. What is wrong with s? (s stands for start what it actually is) |
| 23:43.00 | elixer | any chance you can configure linux to reboot after a panic? |
| 23:43.07 | tzanger | I'm known for my insightful comments |
| 23:43.12 | elixer | instead of just sitting there with a not-so-bright look on its face? |
| 23:43.17 | tzanger | oh wait no, they said inciteful... |
| 23:43.27 | elixer | incestual |
| 23:43.30 | tzanger | elixer: yes |
| 23:43.34 | tzanger | /proc/sys/kernel/something |
| 23:43.39 | tzanger | reboot_on_panic or something like that |
| 23:44.11 | elixer | that would be appropriately named |
| 23:44.53 | elixer | Fulk: i have a Sangoma A400 and a Digium TE220 |
| 23:45.09 | elixer | Fulk: when i am just running vanilla zaptel and the digium card, all is well |
| 23:45.39 | elixer | Fulk: bring in the wanrouter b.s. and if you breath on the machine the wrong way you get a panic |
| 23:45.44 | elixer | i'm going insane |
| 23:45.56 | elixer | or at least moderately annoyed |
| 23:46.27 | JT | doesn't sound like a common problem to me |
| 23:46.36 | JT | then again you have a digium AND a sangoma card |
| 23:47.42 | elixer | well if digium had a PCIe FXS card we wouldn't be in this predicament |
| 23:47.48 | elixer | so obviously its their fault |
| 23:48.17 | JT | not really |
| 23:48.24 | JT | sangoma has PCI-e PRI cards |
| 23:48.44 | elixer | true |
| 23:48.46 | elixer | damn you |
| 23:48.49 | elixer | heh |
| 23:50.47 | elixer | i guess i could try and get support from Sangoma |
| 23:50.58 | elixer | all i need is a null modem cable and a soldering iron |
| 23:51.13 | elixer | soldering iron is only needed to shove into my own eye |
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