| 00:01.54 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 00:02.37 | WasPhantom | eh? |
| 00:03.44 | _Sam-- | sorry...damn nick completion. |
| 00:03.55 | _Sam-- | i typed: sp: flakey |
| 00:03.58 | _Sam-- | and that was what came |
| 00:04.13 | _Sam-- | but you may be flakey too |
| 00:05.34 | WasPhantom | excessively so |
| 00:05.40 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 00:08.47 | [av]bani | \o/ |
| 00:09.04 | [av]bani | i wonder if gs will get bent out of shape over my gaps replacement |
| 00:14.18 | rene- | well reboot underway will this thing blow |
| 00:14.38 | Mavvie | *CLI> pri show spans |
| 00:14.39 | Mavvie | Segmentation fault |
| 00:14.39 | Mavvie | oops |
| 00:15.17 | rene- | and it did blow up, damn, |
| 00:15.22 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 00:18.45 | JunK-Y | Mavvie: which * version? |
| 00:18.59 | Mavvie | JunK-Y: mine. just adding a new command. |
| 00:19.23 | JunK-Y | and whats that command exactly? |
| 00:19.32 | Mavvie | it's nothing to worry about. |
| 00:19.43 | Mavvie | yet :-P |
| 00:21.19 | JunK-Y | gonna give me redbull? nice command then! |
| 00:21.19 | JunK-Y | :) |
| 00:21.40 | rene- | well it is alive again, im recompiling zaptel |
| 00:21.47 | rene- | hi Junky, how ist going |
| 00:21.56 | rene- | s/ist/is it |
| 00:21.57 | JunK-Y | hey rene- ! not much |
| 00:22.15 | JunK-Y | u? |
| 00:22.27 | rene- | well same as always |
| 00:23.38 | rene- | i am trying to work with signate asterisk distro, i had a hard time getting zaptel to work but its here |
| 00:32.06 | [av]bani | hmm voipsupply sales reps are difficult |
| 00:32.37 | DarthClue | [av]bani, what you trying to get from them? |
| 00:32.46 | [av]bani | a response |
| 00:33.04 | [av]bani | "hey guys anyone home? i want to buy stuff from you" |
| 00:33.05 | [av]bani | (crickets) |
| 00:33.15 | DarthClue | [av]bani, what are you trying to buy? |
| 00:33.38 | [av]bani | cisco 7970g, and zoom 5801 |
| 00:33.55 | Qwell[] | [av]bani: email cory directly, for the 7970 |
| 00:33.59 | [av]bani | i did |
| 00:34.01 | [av]bani | no response |
| 00:34.01 | Qwell[] | oh |
| 00:34.06 | [av]bani | hence crickets |
| 00:34.22 | wunderkin | when? today is a holiday |
| 00:34.26 | [av]bani | friday |
| 00:34.56 | [av]bani | i see they posted stuff on ebay since then, so someone has been active in at least putting stuff up for sale the last couple days |
| 00:38.22 | FLeiXiuS | Is it possible to share an extension amongst phones? Perhaps thake that same line and share it between the rest of the phones? |
| 00:38.47 | FLeiXiuS | s/thake/take |
| 00:38.47 | *** join/#asterisk jmcc (n=jcorgan@64-142-68-61.dsl.static.sonic.net) |
| 00:39.53 | jmcc | i'm new to writing agi scripts -- trying out python, commands to stdout work, but nothing sent to stderr gets to the console, yes, i'm calling sys.stderr.flush(), any ideas? |
| 00:40.24 | jmcc | the agi-test.agi sample script has the same problem, no console output but everything else works |
| 00:40.30 | glm2k | jmcc: some people use screen to get around that |
| 00:41.15 | glm2k | it's documented somewhere in the AGI pages on voip-infor.org |
| 00:41.16 | jmcc | you mean i can't see agi script output via stderr when running asterisk -r from command line? |
| 00:42.23 | glm2k | someone correct me but i read it on the site just this week. so it's still likely an issue |
| 00:42.26 | jmcc | i've read the wiki, i didn't see anything about it |
| 00:42.35 | jmcc | 1.2.1, btw |
| 00:42.39 | glm2k | sec...might take me a bit to find it |
| 00:42.46 | jmcc | tnx |
| 00:44.07 | glm2k | http://www.voip-info.org/wiki-Asterisk+AGI under Notes CLI output |
| 00:44.23 | glm2k | i just enable logging and multitail the logs |
| 00:44.36 | jmcc | hmmm, guess i should read more carefully :) |
| 00:44.39 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:44.45 | jmcc | will check it out |
| 00:45.01 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:45.05 | glm2k | screen output can grow because of formatting |
| 00:45.30 | glm2k | so the logs are better; logrotate and all that |
| 00:47.45 | jmcc | ok, that's exactly the issue i'm seeing, fortunately i can see everything on tty9 but I'll have to figure out the screen method, or see what shows up in the logs and set up tail for that like you said |
| 00:47.50 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 00:48.10 | mitcheloc | hello |
| 00:48.55 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 00:51.05 | *** join/#asterisk TuckerAdel (n=TuckerAd@58.160.211.158) |
| 00:54.11 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 00:55.55 | FuriousGeorge | im somewhat new to this whole svn thing, but how come i copy/pasted the command from digium's webpage and it grabbed 1.2.0 |
| 00:56.04 | FuriousGeorge | SVN-branch-1.2-r10487 to be exact |
| 00:56.27 | Nugget | what makes you think that's 1.2.0? |
| 00:56.29 | Qwell[] | because...r10487 is the latest in the 1.2 branch? |
| 00:56.50 | Qwell[] | Did you want trunk or something? |
| 00:57.42 | TuckerAdel | Hi.. when a call comes into * it comes into s and then transfers to a context for internal calls where it rings 4 other phones at the same time.... when I look at the logs it says any incoming call from the pstn number and the conneted numer is s where if im right it should say the number that picked up the call ? How should I change that in the dial plan? |
| 01:02.23 | mitcheloc | tucker, are you looking at the cdr logs? |
| 01:02.31 | TuckerAdel | yes |
| 01:03.23 | mitcheloc | hmm well my logs show which phone answered it |
| 01:03.33 | mitcheloc | tail /var/log/asterisk/cdr-csv/Master.csv, right? |
| 01:03.44 | *** join/#asterisk evilbuny (n=evilbunn@203-206-246-8.dyn.iinet.net.au) |
| 01:03.58 | TuckerAdel | <PROTECTED> |
| 01:04.08 | TuckerAdel | <PROTECTED> |
| 01:04.19 | TuckerAdel | <PROTECTED> |
| 01:05.27 | mitcheloc | well that stuff aside, does that tail command show you the info you need, i.e. is it all being properly logged to the csv file? |
| 01:06.00 | TuckerAdel | <PROTECTED> |
| 01:06.22 | mitcheloc | thats not what i asked you |
| 01:08.15 | TuckerAdel | <PROTECTED> |
| 01:08.38 | Qwell[] | looks to me like SIP/54 answered |
| 01:09.03 | TuckerAdel | <PROTECTED> |
| 01:09.27 | Qwell[] | sounds like your program is broken... |
| 01:09.42 | Qwell[] | obviously pulling the wrong field |
| 01:09.57 | TuckerAdel | <PROTECTED> |
| 01:10.20 | mitcheloc | tucker, i have used the mysql cdr logs many times and all of the data is there |
| 01:10.24 | Qwell[] | hate to break it to you...but the same thing goes into both places |
| 01:10.34 | Qwell[] | You're pulling from the wrong one |
| 01:10.56 | TuckerAdel | <PROTECTED> |
| 01:11.40 | mitcheloc | tcuker, do you see the "s" in that line you posted...? thats the destination field (i'm guessing) |
| 01:11.53 | TuckerAdel | <PROTECTED> |
| 01:11.58 | Qwell[] | You don't want dest |
| 01:12.02 | Qwell[] | you want the destination channel |
| 01:12.08 | mitcheloc | the field you want is probably called dstchannel |
| 01:12.15 | mitcheloc | tucker, http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 01:15.50 | TuckerAdel | <PROTECTED> |
| 01:16.43 | robin_z | wow ... at last, new GXP2000 software to fix that screen bug ... |
| 01:16.57 | justinu | does it work? |
| 01:17.08 | robin_z | oh, wait .. I was dreaming |
| 01:17.13 | justinu | lol |
| 01:17.17 | robin_z | AGAIN! |
| 01:18.31 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.29.212.telnor.net) |
| 01:20.30 | FuriousGeorge | i use gentoo so i never really learned how to compile stuff correctly :) if i set my make.conf to use mmx does that mean that compiles i make by hand will use mmx or do i have to specify that? |
| 01:20.37 | FuriousGeorge | or edit some other file |
| 01:23.11 | FuriousGeorge | nm, thought there would be more gentoo users in here |
| 01:23.38 | *** part/#asterisk TuckerAdel (n=TuckerAd@58.160.211.158) |
| 01:27.35 | Qwell[] | FuriousGeorge: make.conf is only for emerge |
| 01:31.10 | FuriousGeorge | Qwell[]: yeah, so im laerning |
| 01:31.13 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 01:31.28 | FuriousGeorge | just noticed distcc wasnt happening with asterisk so i can assume ive never built it with mmx support after all |
| 01:32.09 | FuriousGeorge | i started with gentoo and in the beginning it was really helpful in the "learning linux" phase but i never leared to compile right on my own :( |
| 01:32.25 | FuriousGeorge | well, properly, anyway |
| 01:32.32 | Qwell[] | "make" |
| 01:32.35 | Qwell[] | "make install" |
| 01:32.38 | Qwell[] | done and done.. |
| 01:34.54 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 01:35.50 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 01:38.56 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 01:39.01 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 01:39.33 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 01:41.07 | tengulre11 | Morning Erveryone!! |
| 01:42.07 | *** join/#asterisk Samoied (n=Samoied@200.193.110.146) |
| 01:42.23 | robin_z | OK, so I have a sip peer that allows me to set my outgoing identity, so the callerid is whatever I want it to be on ougoing calls ... |
| 01:42.44 | robin_z | some users want to rpesent the main switchboard number .. other want to present the DID number ... |
| 01:42.55 | robin_z | clues? |
| 01:43.40 | robin_z | I guess I need to manipulate the peer as part of the call setup to give the CID I want ... |
| 01:47.00 | *** join/#asterisk welles (n=welles@222.90.155.238) |
| 01:47.08 | *** join/#asterisk wellng (n=welles@222.90.155.238) |
| 01:47.15 | *** part/#asterisk wellng (n=welles@222.90.155.238) |
| 01:48.05 | websae | is there an asterisk fedora core package? |
| 01:48.07 | websae | anyone know? |
| 01:48.48 | robin_z | is the Fedora project still active? |
| 01:49.46 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 01:49.50 | FuriousGeorge | so im looking at the makefile, why is cpu=k6 under "pentium & via optimizations"? |
| 01:50.09 | robin_z | mmm .. perversity? |
| 01:50.50 | FuriousGeorge | is there a good guide to the makefile somewhere (for when the comments just arent enough) i cant find it in voipinfo |
| 01:51.00 | lunaphyte | what are park and pickup functions with regard to chan_sccp? |
| 01:55.08 | *** join/#asterisk rezzen_ (n=pkn@CPE00e081103ccb-CM0013718c3bee.cpe.net.cable.rogers.com) |
| 01:55.21 | FuriousGeorge | i cant figure out if an athlon xp owner (32bit) should uncomment PROC=K6 |
| 01:55.36 | FuriousGeorge | anyone wanna chime in? |
| 01:56.00 | robin_z | right .. so I guess I need to modify the "fromuser" of the SIP peer on a per call/user basis .. some want the switchboard number to g out, some want their DID number |
| 01:56.10 | robin_z | hmm .. now, how to do that |
| 01:57.02 | rezzen_ | n00b trying to get asterisk working as sip proxy (no zaptel or other digium h/w yet) on CentOS 4.2. asterisk -r results in "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" |
| 01:57.03 | robin_z | passing the number will be easy enough, I guess I just set it as a channel variable for each user |
| 01:57.03 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:57.03 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 01:57.19 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 01:57.28 | rezzen_ | not a modules.conf prob--at least no indication of that in messages log. |
| 01:57.37 | robin_z | rezzen_: try asterisk -c ... |
| 01:57.45 | rezzen_ | that gives me cli... |
| 01:57.49 | robin_z | -r only works if one is running ... |
| 01:57.56 | rezzen_ | aaahhhh.... |
| 01:57.59 | robin_z | so ... just plain asterisk |
| 01:58.07 | robin_z | then asterisk -r to connect to it |
| 01:58.12 | *** join/#asterisk Cool_One (n=bclinton@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 01:58.20 | robin_z | or ... better still |
| 01:58.37 | robin_z | get the CLI, enter "stop now" to stop it ... |
| 01:58.52 | robin_z | then safe_asterisk |
| 01:58.54 | rezzen_ | wow, that gap in understanding was just a lil crack. |
| 01:59.08 | robin_z | safe_astersik auto restarts it in the event of a application crash |
| 01:59.50 | rezzen_ | beaut. what service ctrl mechanism is safe_asterisk using? |
| 01:59.58 | robin_z | feck knows |
| 02:00.03 | rezzen_ | i guess its own, by the looks of things. |
| 02:00.19 | robin_z | yeah, its a perl script as I remember? |
| 02:00.24 | robin_z | or just bash? |
| 02:00.48 | Cool_One | does anyone have time to answer a questin for a newbie??? |
| 02:00.50 | rezzen_ | bash. robin_z: thanks. |
| 02:01.06 | FuriousGeorge | grrr, see now it compiles it says march=i686. when gentoo compiles stuff it says march=athlonxp, should i be worried? |
| 02:01.12 | rezzen_ | just ask yer danged question, n00b. |
| 02:01.14 | robin_z | Cool_One: domnt ask to ask, just ask |
| 02:01.53 | robin_z | FuriousGeorge: yes. if you are using gentoo, you shold be worried. |
| 02:01.59 | robin_z | its a sign of madness you know |
| 02:02.20 | Cool_One | ok... I have asterisk@home installed and running, 2 copper phone lines, 3 grandstream gxp2000 phones... system works great. I can call in and out with not problems.... But, I can not put someone on hold and pick up the second line. |
| 02:02.41 | rezzen_ | FuriousGeorge: robin_z is on the mark about that... |
| 02:02.46 | Qwell | Cool_One: sounds like a limitation of the phone. |
| 02:02.50 | robin_z | nah |
| 02:03.10 | robin_z | the gxp 2000 has a working hold button and send s the right sip message |
| 02:03.18 | Cool_One | Grandstream says I need to setup sip acocunts for each line I wan to use on my phone |
| 02:03.30 | robin_z | yes |
| 02:03.37 | Cool_One | but that would mean that my phone would have 2 extensions... |
| 02:03.43 | robin_z | no |
| 02:03.50 | Qwell | Like I said...sounds like a limitation of the phone. |
| 02:03.51 | Cool_One | so how would I transfer calls |
| 02:04.00 | robin_z | with the transfer button? |
| 02:04.34 | robin_z | extensions are NOT the same as outgoing lines |
| 02:04.50 | Cool_One | ok... transfer button... but if I had 2 sip acocunts on each phone what extension # would I use |
| 02:05.00 | Cool_One | I know |
| 02:05.17 | robin_z | the extension number of the phoen you want to trxfr to of course ... |
| 02:05.42 | Cool_One | I am really confussed |
| 02:06.01 | robin_z | sigh ... |
| 02:06.04 | robin_z | loo, |
| 02:06.05 | Cool_One | so would I setup additional sip accounts on the server for each line that I really want to use |
| 02:06.18 | robin_z | I wold set up: |
| 02:06.26 | Cool_One | and on the phone |
| 02:06.29 | robin_z | account for phone 1 = 1001 |
| 02:06.31 | Qwell | get a phone that supports multiple calls on a single account |
| 02:06.36 | robin_z | accoutn for phone 2 = 1002 |
| 02:06.48 | robin_z | these are then your "internal" numbers |
| 02:07.06 | Cool_One | ok, I understand that robin_z |
| 02:07.31 | Cool_One | so my phone does not support multiple calls on a single account? |
| 02:07.41 | robin_z | no idea |
| 02:07.44 | Qwell | Cool_One: You just said Grandstream claims it didn't |
| 02:07.59 | Cool_One | what phones do? any suggestions |
| 02:08.25 | robin_z | Id set up additional accounts for say 1011 ncoming line 1 on phoen 1, 1012, incoming line 2 on phoen 2 |
| 02:08.37 | Qwell | Cool_One: any that aren't complete crap |
| 02:08.41 | Qwell | ie; non-grandstream |
| 02:08.41 | *** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com) |
| 02:08.43 | robin_z | and then have incoming line 1 ring 1011 and 1012 at the same time, or whatever |
| 02:08.56 | robin_z | thats one way to do it ... |
| 02:09.03 | Qwell | Cool_One: ask [av]bani or _Sam--. They seem to love GS phones for some silly reason |
| 02:09.18 | robin_z | later GXP2000 firmware does asterisk blf on the buttons on the right ... |
| 02:09.36 | robin_z | bedtime!! |
| 02:09.42 | robin_z | byeeeeeeeeeeeee |
| 02:10.52 | xorsysd | Can anyone help me with a zaptel compile problem? |
| 02:12.11 | rezzen_ | thanks again robin_z |
| 02:12.44 | rezzen_ | xorsysd. maybe. maybe not. ask your question and find out. and don't ask to ask next time. |
| 02:13.02 | xorsysd | ok |
| 02:13.21 | xorsysd | The modules won't build, it gets to stage 2 and says: |
| 02:14.50 | xorsysd | Makefile.modpost:38: .config: No such file or directory |
| 02:15.13 | xorsysd | on a 2.6.15 kernel using `make linux26` |
| 02:16.51 | *** part/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:17.57 | tengulre11 | hi,all |
| 02:18.12 | rezzen_ | xorsysd: you may need to specify arch and/or other env variables. |
| 02:18.15 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:18.16 | tengulre11 | who have a IVR system contain source code? |
| 02:18.31 | tengulre11 | Windows Platform, |
| 02:18.49 | rezzen_ | xorsysd: which, i unfortunately can't help with |
| 02:19.13 | rezzen_ | have you run 'make' w/out any targets first? |
| 02:19.28 | xorsysd | yep |
| 02:19.30 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 02:19.32 | xorsysd | same error |
| 02:19.56 | xorsysd | It said something about kernel config file, so I'm going to try some things |
| 02:20.02 | rezzen_ | ensure are your requirements are met (may some missing libs, or kernel headers/source) |
| 02:20.23 | rezzen_ | s/may/maybe/ |
| 02:20.39 | rezzen_ | jbot: SNAP |
| 02:20.41 | jbot | sfsnap is probably http://sfsnap.babylonia.flatirons.org/bzflag/, or http://bzflag.org/cvs/ |
| 02:21.48 | xorsysd | lolz, my kernel's .config was not present in /usr/src/linux :p |
| 02:22.28 | lunaphyte | are chan_sccp questions off topic here? |
| 02:26.45 | *** join/#asterisk techie (i=gus@antibala.com) |
| 02:28.48 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 02:28.52 | xorsysd | Got it compiled, but I'm getting, unknown symbol in module |
| 02:30.09 | Ariel_ | lunaphyte, no but there are not that many people using it. just ask the question and you just might get a reply |
| 02:33.18 | *** join/#asterisk epablo (n=epablo@200.31.138.202) |
| 02:33.24 | epablo | Hi people |
| 02:33.40 | Ariel_ | hello epablo |
| 02:33.41 | firestrm | greetings.. |
| 02:34.46 | epablo | I got a TDM400P REV I up and running. I setup an IVR on it, but I hangup and it keeps the line open until the IVR and ultimatally the vm ends |
| 02:35.10 | Mavvie | beautiful! |
| 02:35.18 | Mavvie | finally will get normal load-balancing on my two PRIs |
| 02:35.26 | epablo | Any ideas on how to make it work better.. I'm in Venezuela and already tested using ls signaling instead of ks |
| 02:35.45 | xorsysd | *** Warning: "rtc_unregister" [/usr/src/zaptel-1.2.4/ztdummy.ko] undefined! |
| 02:35.46 | xorsysd | *** Warning: "rtc_control" [/usr/src/zaptel-1.2.4/ztdummy.ko] undefined! |
| 02:35.46 | xorsysd | *** Warning: "rtc_register" [/usr/src/zaptel-1.2.4/ztdummy.ko] undefined! |
| 02:35.46 | xorsysd | *** Warning: "crc_ccitt_table" [/usr/src/zaptel-1.2.4/zaptel.ko] undefined! |
| 02:36.00 | Qwell | xorsysd: You need CRC_CCITT in the kernel |
| 02:36.09 | firestrm | Mavvie , sounds cool.. ;) |
| 02:36.20 | xorsysd | Which is? |
| 02:36.44 | Qwell | CRC_CCITT |
| 02:37.52 | lunaphyte | i have an old 12sp+ that would register with skinny.so but couldn't complete a call, so i though i'd try sccp.so. now it won't register - asterisk says "Rejecting device: not found" and i'm not quite sure if my sccp.conf file is correct. |
| 02:38.12 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 02:39.11 | *** part/#asterisk epablo (n=epablo@200.31.138.202) |
| 02:39.32 | xorsysd | What section in the config would that be found in? |
| 02:39.34 | Ariel_ | epablo, there is also ground start look at the /usr/src/asterisk/configs/zaptel.conf.sample for the settings on it. |
| 02:42.04 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 02:45.40 | xorsysd | Is CRC_CCITT a patch or a config option? |
| 02:45.47 | Qwell | config option |
| 02:45.52 | xorsysd | What section? |
| 02:46.00 | Qwell | grep your .config |
| 02:46.41 | xorsysd | # CONFIG_CRC_CCITT is not set |
| 02:48.01 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 02:48.31 | xorsysd | Thankyou |
| 02:49.27 | xorsysd | Second night in a row for re-compiling a kernel yay |
| 02:49.33 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:54.43 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
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| 03:07.35 | mwgbc | I'm new to *. I have successfully set up outbound and have been using it for a little while. I just recently set up DID with Broadvoice. I am registered with them and can make outgoing calls but when I dial my Broadvoice number I just get a busy. Any ideas? |
| 03:08.33 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 03:09.21 | mwgbc | Wow! is it dead in here tonight |
| 03:10.30 | FuriousGeorge | im really confusd as to why asterisk is being built with march=i686 |
| 03:10.57 | FuriousGeorge | i set PROC=athlon-xp, and gentoo builds everything with march=athlon-xp, so whats the deal with that |
| 03:11.32 | xorsysd | Gentoo is really behind on stable version for Asterisk |
| 03:12.09 | FuriousGeorge | xorsysd: yeah, im building it myself, and this is my first formal introduction to Makefiles. Gentoo has spoiled me |
| 03:12.20 | *** join/#asterisk mwgbc (i=mwgbc@c-67-188-17-12.hsd1.ca.comcast.net) |
| 03:12.35 | FuriousGeorge | so im a little confused as to why PROC=i686 is commented and PROC=athlon-xp is not, yet it uses march=i686 |
| 03:13.11 | FuriousGeorge | when 1.2 came out and gentoo still had 1.0.9 masked i figured it was probably a good idea to go outside of portage |
| 03:13.45 | FuriousGeorge | mwgbc: you got a user set up for broadvoice in sip.conf? |
| 03:13.50 | FuriousGeorge | i know its well documented on the wiki |
| 03:14.53 | mwgbc | FuriousGeorge: I set up a friend in sip for broadvoice pointing to the context I wanted incoming to go to. |
| 03:14.59 | trixter | does anyone have any leads on a did provider that does 619/858 (san diego) that offers more than 2 concurrent channels (voxbone wants $20/mo for each additional channel :( |
| 03:15.23 | FuriousGeorge | what does sip show registry say |
| 03:15.30 | FuriousGeorge | "sip show registry" in cli |
| 03:16.20 | FuriousGeorge | anyone know how to get asterisk to compile with march=athlon-xp without manually setting every instance in the makefile? it seems to use i686 no matter how many comments i stick in front of that line |
| 03:16.55 | mwgbc | FuriousGeorge: sip.broadvoice.com:5060 <phone#>@s 3584 Registered |
| 03:18.12 | FuriousGeorge | you sure incoming context is set up right? |
| 03:18.29 | FuriousGeorge | exten => s,1,dial(your user) |
| 03:18.50 | FuriousGeorge | and see if the cli spits anything at you when you dial that number |
| 03:19.13 | mwgbc | I set it to Answer() then Playback() for testing purposes |
| 03:19.30 | FuriousGeorge | cli? |
| 03:19.45 | mwgbc | one second... |
| 03:20.46 | mwgbc | FuriousGeorge: Verbose set to 10 CLI was silent |
| 03:21.26 | FuriousGeorge | hmmmm |
| 03:21.41 | [av]bani | \o> |
| 03:21.42 | [av]bani | <o/ |
| 03:27.11 | FuriousGeorge | mwgbc: maybe you should turn on sip debug... someone else will have to help with that, as it may as well be greek to me |
| 03:27.30 | FuriousGeorge | [av]bani: i echoed your sentiment re: snom 360 on voip-info |
| 03:28.26 | FuriousGeorge | and i decided no way to get parking w/ leds working right w/o the api, which is beyond my scopew |
| 03:28.31 | FuriousGeorge | *scope* |
| 03:29.34 | mwgbc | FuriousGeorge: Thanks for your help. I even tried just a simple s,1,dial(<agent>) and got the same result. |
| 03:30.10 | FuriousGeorge | ive heard that people's success w/ broadvoice + * is spotty, not to discourage you, im sure its something simple |
| 03:30.16 | FuriousGeorge | find another guy using bv maybe |
| 03:30.51 | FuriousGeorge | so no one can tell me why * is using march=686 when building, despite the comment in front of that line |
| 03:31.35 | Nivex | iCEBrkr: the result of my dementia: http://trilug.org/~kjotte/tarp/ |
| 03:32.33 | *** join/#asterisk gongoputch (n=gongoput@c-68-82-194-31.hsd1.de.comcast.net) |
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| 03:49.51 | [av]bani | FuriousGeorge: i'm sure you will learn |
| 03:49.51 | [av]bani | FuriousGeorge: agi and stuff |
| 03:50.43 | *** join/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 03:51.06 | tmccrary | Is "405 Method Not Allowed" related to a codec issue? If not, does anyone know what that is? |
| 03:51.48 | FuriousGeorge | [av]bani: yeah, i got this far, right |
| 03:52.04 | FuriousGeorge | [av]bani: you know why asterisk makefile seems to pick a dumb march setting? |
| 03:55.32 | FuriousGeorge | i dunno wtf is wrong with people, i just asked that question in asterisk-dev, no one is talking there either, no answer. you'd think they'd know, why would they just ignore me? i always try to help people when im on here |
| 03:56.58 | [av]bani | FuriousGeorge: afaik it picks no march setting at all... |
| 03:57.17 | FuriousGeorge | oh but it does, and it always seems to be -i686 |
| 03:57.25 | [av]bani | FuriousGeorge: sounds like something your distro is doing |
| 03:57.36 | FuriousGeorge | but im just setting it to what i have come to believe is the correct one |
| 03:57.41 | [av]bani | if it were always picking -i686 then my x86_64 compile wouldnt work |
| 03:57.52 | FuriousGeorge | doubtful, gentoo controls all that stuff in make.conf. im in the makefile b/c im building * by hand |
| 03:58.02 | [av]bani | ahh gentoo.. that explains everything |
| 03:58.14 | FuriousGeorge | for some reason options=-m64 is uncommented in the makefile, so is proc=k8 |
| 03:58.18 | FuriousGeorge | and mine built |
| 03:58.43 | *** join/#asterisk tuxinator_linux (n=tuxinato@m190e36d0.tmodns.net) |
| 03:58.47 | FuriousGeorge | actually it confuses matters more. gentoo sets march setting in make.conf, which is where i have it set to athlon-xp |
| 04:01.50 | DarthClue | FuriousGeorge, it is possible that either noone is available to provide the answer or noone knows. I doubt you are being ignored. |
| 04:02.19 | FuriousGeorge | i dunno, i usually dont take it personally |
| 04:02.25 | FuriousGeorge | long day today |
| 04:02.32 | niZon | hmm |
| 04:02.37 | niZon | anyone here use realtime queues? |
| 04:02.52 | FuriousGeorge | had to deal with my provider today too, ugh |
| 04:03.00 | *** join/#asterisk Eggplant (i=No@64.233.107.198) |
| 04:03.17 | DarthClue | FuriousGeorge, might be time to just put it up for the night. Some days just aren't worth the trouble of getting up for. |
| 04:03.38 | FuriousGeorge | well i ahve been at it for 13 freakin hours |
| 04:03.56 | FuriousGeorge | lemme just finish remaking asterisk with new make files, see if i get calls of panic tomorrow |
| 04:04.32 | DarthClue | well, if it had just been 13 hours, I would say give it another couple, but since it's been 13 freakin hours, I would say that's about 12 too many. |
| 04:04.48 | FuriousGeorge | alls i wanna know, after all, if its a bad idea to set the damn thing manually when it insists on using i686, so i guess ill find out soon enough |
| 04:05.01 | FuriousGeorge | my day, not the makefile |
| 04:05.11 | FuriousGeorge | this has been the last hour |
| 04:05.44 | DarthClue | you using an amd64 machine? if so, i686 config shouldn't be an issue from what I can remember of some of the testing I did. |
| 04:06.43 | FuriousGeorge | barton cpu in both of them |
| 04:06.53 | FuriousGeorge | amd32 |
| 04:07.26 | FuriousGeorge | one of em is a pentium4, set that one manually to pentium4 didnt wait to see what it chose |
| 04:07.41 | mwgbc | I'm having problems with DTMF. Inbound DID is answered and sent to extension that plays BackGround() waiting for DTMF. BackGround() plays message but does not recognize DTMF tones. It does work however when it makes an outgoing call and transferes to the same extension and priority. |
| 04:08.06 | FuriousGeorge | codec? |
| 04:08.25 | mwgbc | ula |
| 04:08.27 | mwgbc | ulaw |
| 04:08.30 | FuriousGeorge | mwgbc: how did you get it working by the way? |
| 04:08.47 | FuriousGeorge | u behind nat? |
| 04:09.38 | mwgbc | FuriousGeorge: I ran sip debug and found I put a context name where it should have been an extension name. |
| 04:09.44 | tmccrary | Is "405 Method Not Allowed" related to a codec issue? If not, does anyone know what that is? |
| 04:09.46 | FuriousGeorge | ahaaaa |
| 04:10.08 | FuriousGeorge | mwgbc: sounds like ur having nat problems now |
| 04:10.12 | FuriousGeorge | is asterisk behind nat? |
| 04:10.38 | mwgbc | FuriousGeorge: I have a router on my cablemodem that does DNS for me. (my sipura box) Asterisk is on a dedicated server in Florida |
| 04:10.40 | FuriousGeorge | tmccrary: i think i ususally get "Not Allowed Here" in CLI |
| 04:10.43 | niZon | anyone use realtime queues in 1.2.4? |
| 04:10.53 | niZon | they don't seem to work without reloading |
| 04:11.49 | DarthClue | tmccrary, either codec or sip registration...when are you getting it? |
| 04:11.50 | FuriousGeorge | have you verified that you can even hear yourself speaking when you call your did? |
| 04:12.16 | mwgbc | FuriousGeorge: yes, transfered to sip phone and it works fine. |
| 04:12.56 | FuriousGeorge | then its not what i thought it was |
| 04:13.05 | FuriousGeorge | check your dtmf settings |
| 04:13.27 | FuriousGeorge | its gotta be one of those. my * is so dtmf sensitive that it pickes up tones if people are on speaker too close to me |
| 04:14.09 | FuriousGeorge | does cli react when you hit a button? |
| 04:15.43 | mwgbc | FuriousGeorge: I just ran another test. I can hear myself fine, but when pushing DTMF on calling phone it comes across audibly as barely more than a click. |
| 04:16.23 | *** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net) |
| 04:17.30 | tmccrary | THanks for the help everyone. I am getting the 405 Method not allowed here when one phone attempts to call another phone through a SIP trunk (another asterisk pbx) |
| 04:18.10 | *** join/#asterisk websae (i=websae@CPE-24-167-204-30.wi.res.rr.com) |
| 04:18.10 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 04:18.33 | mwgbc | FuriousGeorge: No, CLI does not register any DTMF activity (verbose 10) when I hit DTMF. Broadvoice said my settings for DTMF should be Inband. Is that correct? |
| 04:18.48 | websae | does anyone know how to correct this issue that i have when compiling asterisk.../usr/bin/ld: cannot find -lidn?? |
| 04:18.58 | websae | appreciate any suggestions :) |
| 04:19.00 | websae | or help |
| 04:19.25 | asterisk99 | How does one get asterisk to run after boot-up of ubuntu? |
| 04:19.30 | websae | and i am running fedora core |
| 04:21.16 | tmccrary | asterisk: I believe you edit /etc/default/asterisk |
| 04:21.34 | tmccrary | You need to change a value to YES like: RUNONSTARTUP="YES" |
| 04:21.51 | tmccrary | I forget the exact name of the variable. It's just like debians setup (no suprise there) |
| 04:22.28 | asterisk99 | tmccrary: hmmmmm no file /etc.default/asterisk |
| 04:23.01 | tmccrary | did you install apt-get install asterisk? |
| 04:23.08 | tmccrary | or did you manually install it? |
| 04:23.13 | websae | manually |
| 04:23.15 | tmccrary | ah |
| 04:23.18 | asterisk99 | tmccrary: manually |
| 04:23.25 | tmccrary | then you need to create a script and symlink it to a runlevel |
| 04:23.28 | websae | any suggestions for the /usr/bin/ld: cannot find -lidn problem? |
| 04:23.38 | websae | when compiling |
| 04:23.41 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:23.42 | websae | fedora core |
| 04:24.42 | FuriousGeorge | mwgbc: inband is for gsm, but try it |
| 04:25.19 | DarthClue | websae: try installing libidn-devel |
| 04:25.34 | websae | is there a yum package? |
| 04:25.48 | websae | i tried yum libidn-devel |
| 04:25.52 | websae | and obviously that didn't work |
| 04:26.07 | DarthClue | try up2date libidn-devel |
| 04:26.17 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
| 04:26.28 | mwgbc | FuriousGeorge: That is my current setting (inband) DTMF tones sound more like clicks audibly and CLI doesn't notice them |
| 04:27.11 | mwgbc | FuriousGeorge: I guess I can try "auto" for the DTMF mode and see if it works better. |
| 04:27.13 | websae | hrm |
| 04:27.24 | websae | i did up2date libidn-devel |
| 04:27.27 | websae | what is that suppose to do? |
| 04:29.23 | DarthClue | should install the rpm package...did it? |
| 04:29.29 | niZon | well thats no fun, no realtime users here :\ |
| 04:31.08 | DarthClue | that is correct. This conversation actually took place 2 hours ago and is simply being replayed for your convenience. |
| 04:33.23 | mwgbc | FuriousGeorge: Well, that worked. I changed DTMFmode to auto. The tones started coming in more clearly and CLI started picking them up and it correctly worked with BackGround() :) Yeah! |
| 04:34.46 | *** part/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 04:35.30 | *** join/#asterisk voip470 (n=A_mail@71.244.98.91) |
| 04:36.12 | Nugget | Mooâ„¢ |
| 04:38.42 | websae | it did not install the pacakge |
| 04:42.36 | FuriousGeorge | im glad he got it working :) |
| 04:42.38 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 04:42.55 | *** join/#asterisk welles (n=welles@222.90.155.238) |
| 04:43.27 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 04:44.29 | mzo | does anyone want to help me figure out how the hell i broke my FWD configuration? :) |
| 04:45.33 | DarthClue | websae: type in rpm -qa | grep libidn and check to see if it shows libidn and libidn-devel |
| 04:45.45 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:45.48 | *** join/#asterisk PaulHuynh (n=paulhuyn@c-68-44-237-105.hsd1.de.comcast.net) |
| 04:46.03 | PaulHuynh | help can someone help me with sip debug? |
| 04:46.26 | PaulHuynh | i have 3 spa will not register or was register and now it stop |
| 04:46.33 | PaulHuynh | it just wierd |
| 04:49.48 | *** join/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
| 04:50.15 | PaulHuynh | hello |
| 04:50.18 | PaulHuynh | anyone here? |
| 04:50.23 | pcm | no1 |
| 04:50.46 | pcm | 331 - 2 dead sessions |
| 04:50.48 | DarthClue | anyone already left, but someone might still be here, although i doubt he'll be of much help. |
| 04:51.23 | JunK-Y | hey DarthClue, long time no see u |
| 04:52.10 | DarthClue | i've been trapped in the desert on a horse with no name. |
| 04:52.44 | FuriousGeorge | ~spa300 |
| 04:52.55 | FuriousGeorge | that a sip phone? |
| 04:53.04 | FuriousGeorge | sipura? |
| 04:53.45 | PaulHuynh | np |
| 04:53.50 | PaulHuynh | it really a pap2 |
| 04:53.58 | PaulHuynh | but same as spa2000 |
| 04:53.58 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 04:54.06 | DarthClue | next time we go to diamonds, you need to remember your passport |
| 04:54.13 | PaulHuynh | it keep said offline or can't connect to server |
| 04:57.11 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
| 04:58.33 | livinded | i'm looking at buying either an ip phone or an ata, what would be better for a home user? |
| 04:58.59 | DarthClue | livinded, that depends on what you intend to do with it. |
| 04:59.21 | livinded | DarthClue, what do you mean? |
| 04:59.37 | PaulHuynh | any one? |
| 04:59.43 | PaulHuynh | any idea |
| 04:59.46 | PaulHuynh | hello |
| 05:00.47 | PaulHuynh | anyone here? |
| 05:01.12 | DarthClue | livinded, it depends on what you want to do with the phone. if you just want to use your existing phone for voip then an ata, if you want the features that come with an ip phone then get the ip phone. |
| 05:01.48 | DarthClue | PaulHuynh, I don't think anyone here is able to help right now. If I knew, I would answer ya. |
| 05:02.01 | livinded | what features do ip phones have that pstn don't and could you recomend a good ip phone, i'ml ooking at the sipura spa 3000 if i go with the ata |
| 05:02.43 | PaulHuynh | use polycom ip 501 |
| 05:02.50 | PaulHuynh | they are rock soid |
| 05:02.52 | DarthClue | I'm partial to polycoms, cisco is the high-end. go to google and dig around, you'll soon see that there are tons of features that most home users won't ever touch, but it is your choice. |
| 05:03.14 | livinded | i've heard good thigns about cisco phoens but are they really worth the money? |
| 05:03.20 | PaulHuynh | also don't forget linksys spa941 |
| 05:03.45 | PaulHuynh | yes they are feel + quality are second to none |
| 05:03.55 | livinded | the thing i like about the spa 3000 is that it has a pstn port also so i can route certain calls through my pstn line |
| 05:03.57 | PaulHuynh | we have both polycom + cisco |
| 05:04.05 | PaulHuynh | yup |
| 05:04.19 | PaulHuynh | then spa3000 is your bets choice |
| 05:04.24 | PaulHuynh | best |
| 05:05.14 | livinded | are the linksys the same as the cisco ip phones or are the linksys the consumer versions? |
| 05:06.34 | PaulHuynh | consumer version |
| 05:06.43 | PaulHuynh | but they are good for the price |
| 05:07.12 | PaulHuynh | anyone have any idea how to look for password in the sip registration? |
| 05:07.55 | rt | actually, the spa-3000 doesn't seem to pick up. |
| 05:08.47 | PaulHuynh | what i mean is i rebuilt my asterisk and i have 3 people that is not in the office and i forgot their sip password how can i look it up using a sip debug or something like that to recreate the corrcet pass so i don;t have to reconfig the whole things |
| 05:09.56 | shido6 | sip.conf is your friend |
| 05:10.13 | PaulHuynh | Registration from '121 ' failed for '58.186.22.52' - Username/auth name mismatch |
| 05:10.25 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:10.45 | PaulHuynh | ok i'm confuse i nolonger have a old copy of sip.conf |
| 05:10.59 | livinded | i think i'll end up going with the ip phone, otherwise i'd only get 1 of my pstn lines connected to asterisk |
| 05:11.10 | PaulHuynh | but i do know the ext. the device is keep try to connect |
| 05:20.08 | gongoputch | anyone running asterisk in a freebsd jail() ? |
| 05:23.20 | mzo | bleh i blew up asterisk. |
| 05:23.22 | mzo | time to rm -rf :p |
| 05:24.48 | jebba | gongoputch, i am running it in a vserver, which is similar |
| 05:25.49 | gongoputch | I am trying to figure out if hardware access is possible from the jail |
| 05:28.26 | Abydos313 | any probation officers in here to answer that question? heh |
| 05:29.09 | X-Rob | I think you need to smuggle it in up your arse. |
| 05:29.18 | DarthClue | i'm pretty sure that if you try to access the hardware from within the jail you are sentenced to solitaire confinement |
| 05:29.19 | Abydos313 | heh |
| 05:29.21 | X-Rob | 17" LCD's are going to be interesting to watch you insert. |
| 05:29.35 | Abydos313 | ouch |
| 05:29.59 | DarthClue | just a 17"? if you're gonna go thru the pain, might as well go for the 21" |
| 05:29.59 | gongoputch | groan ... |
| 05:30.07 | mzo | bleh, my aah upgrade failed on zaptel. yay =( |
| 05:30.18 | DarthClue | gongoputch, try it and find out, obviously we don't have that answer. |
| 05:31.41 | mzo | Removing zaptel module: ERROR: Module zaptel does not exist in /proc/modules |
| 05:31.41 | mzo | <PROTECTED> |
| 05:34.17 | FuriousGeorge | mzo: w/ rmmod? |
| 05:34.28 | FuriousGeorge | what happens when you lsmod? |
| 05:34.29 | mzo | yeah, i just ran setup again and it looks fine again :P |
| 05:35.29 | FuriousGeorge | one time i ran asterisk for a bit after upgrading w/o reloading new modules and when i remembered too i couldnt insert it into kernel, had to reboot after about 30 min of rebuilding asterisk and manually pointing it to the module it couldnt find |
| 05:35.37 | FuriousGeorge | rmembered to* |
| 05:36.09 | X-Rob | this one time, at band camp... |
| 05:36.16 | mzo | hahaha, oops. :) |
| 05:36.20 | mzo | now if i can get FWD to work |
| 05:36.27 | FuriousGeorge | lol |
| 05:36.32 | mzo | i'm sure there's at least five or six more rm -rf's in my future |
| 05:36.36 | FuriousGeorge | i didnt know you played the clarinett |
| 05:37.27 | FuriousGeorge | X-Rob: btw, you were right i couldnt think of a way to get my meetme parking with leds working seamlessly enough |
| 05:38.06 | FuriousGeorge | the best solution i found would have one button parking and one button answering.... |
| 05:38.22 | mzo | does anyone have a recommendation for a VOIP provider i can use to make calls to finland? Pay is okay, im sure it's cheaper than calling cards ;) |
| 05:39.09 | FuriousGeorge | finland? i have no idea, i would google fin voip providers |
| 05:39.25 | mzo | i can't read swedish =( |
| 05:39.36 | FuriousGeorge | i thought they speak fin in finland? |
| 05:39.40 | mzo | all the links come up in swedish or some other language i can't read |
| 05:39.42 | mzo | it could be? |
| 05:39.50 | FuriousGeorge | like closer to cyrillic |
| 05:39.52 | FuriousGeorge | language |
| 05:40.08 | Qwell | call Linus |
| 05:40.18 | FuriousGeorge | hes not taking my calls anymore |
| 05:41.16 | FuriousGeorge | ever since i failed to salute his kernel |
| 05:41.32 | FuriousGeorge | not funny |
| 05:41.35 | FuriousGeorge | :| |
| 05:43.03 | X-Rob | <PROTECTED> |
| 05:43.06 | X-Rob | FuriousGeorge, ^^^^ |
| 05:43.10 | X-Rob | Application Hints |
| 05:43.17 | FuriousGeorge | stop liein' |
| 05:43.17 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 05:43.19 | Guest^DJ | hi |
| 05:43.43 | X-Rob | Posted to asterisk-users subject 'call parking "hint"' |
| 05:47.28 | FuriousGeorge | X-Rob: can you link me? im having trouble finding it |
| 05:47.37 | X-Rob | http://bugs.digium.com/view.php?id=5779 |
| 05:48.23 | FuriousGeorge | oh yeah, i know about this guys branch, but its based on trunk, and i need it for production |
| 05:48.48 | FuriousGeorge | remember you said "its probably easy to back port" to which i replied "lol" and though (for you, maybe) :) |
| 05:48.59 | *** join/#asterisk clive- (n=pirch@dsl-145-31-53.telkomadsl.co.za) |
| 05:49.09 | FuriousGeorge | thought* |
| 05:52.26 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 05:53.57 | Guest^DJ | how to i generate a fake tone for 5 secs ? currently i am using exten => _01.,1,Dial(SIP/alpha/${EXTEN}|45|rt) |
| 05:54.08 | Qwell | define "fake tone" |
| 05:54.46 | Guest^DJ | as in the moment asterisk is tying to connect, instead of silent, * generates some kind of ring tone |
| 05:54.58 | Qwell | Why do you need that? |
| 05:55.28 | Guest^DJ | the silent is too long, approx 15 sec to connect. user would think is not connecting |
| 05:55.35 | Guest^DJ | and hang up |
| 05:55.57 | Guest^DJ | so by generating * tone, user would think that is connecting |
| 05:56.32 | Guest^DJ | * ring tone would ring for 10-15 secs and return to normal pstn tone |
| 05:56.55 | Guest^DJ | that way, user would know if the called number is busy/ringing |
| 05:57.47 | [av]bani | playtones() |
| 05:58.05 | Guest^DJ | ok, i try |
| 05:58.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:58.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:59.54 | Guest^DJ | [av]bani: using s,1,playtones(10), after 10 sec * would go to s,2,dial(). still there would be another 15 sec wait to connect |
| 05:59.57 | Guest^DJ | am i right? |
| 06:00.42 | [av]bani | s,1,playtones(ring) |
| 06:00.50 | [av]bani | s,n,dial() |
| 06:00.59 | [av]bani | or whatever you want to do |
| 06:01.26 | [av]bani | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones |
| 06:11.00 | Guest^DJ | it just keep ringing even the called number is busy |
| 06:11.37 | [av]bani | you need to make it stop playtones when it gets busy |
| 06:11.49 | Guest^DJ | did that |
| 06:11.52 | [av]bani | s,1,playtones(ring) |
| 06:11.54 | [av]bani | s,n,dial() |
| 06:11.58 | [av]bani | s,n,stopplaytones() |
| 06:12.04 | [av]bani | or something like that |
| 06:17.42 | *** join/#asterisk tletourneau (n=tom_remo@12-219-187-158.client.mchsi.com) |
| 06:17.44 | FuriousGeorge | mog mentioned in passing something about a registration manager in asterisk (i assume he meant in the code) |
| 06:17.52 | FuriousGeorge | to better handle my .dynu adresses |
| 06:19.00 | FuriousGeorge | anyone know anything about that? |
| 06:19.03 | FuriousGeorge | Qwell: ? |
| 06:19.09 | FuriousGeorge | im looking in your direction |
| 06:19.46 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:21.31 | tletourneau | What channel would a person go to to find help on setting * up with an multi-channel sip to pots device? |
| 06:21.59 | justinu | this is it, dude |
| 06:22.11 | tletourneau | :) |
| 06:23.24 | Qwell | These are not the droids you're looking for |
| 06:24.12 | tletourneau | I have a Vegastream Vega50, I got it to dial out over pots ok but I can't figure out how to get it to talk to * on an incoming call. |
| 06:24.50 | FuriousGeorge | you got a context in your dialplan for incoming calls tletourneau |
| 06:25.16 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:26.50 | tletourneau | FuriousGeorge I have a few, I use 2 sip providers and an aix provider. I'm just not sure what to put in for this type of device. |
| 06:27.15 | FuriousGeorge | it connects to a sip server on one end and plain old telephones on the other? |
| 06:29.12 | tletourneau | Yes, it has 8 FXO ports and one ethernet port. I've tried a dialing plan in it that tells it to send any pots traffic to my * server. I'm running A@H 2.5. |
| 06:30.09 | justinu | tletourneau: can you see the vega sending you invites on inbound call? |
| 06:30.22 | justinu | tletourneau: give up on AMP |
| 06:30.28 | justinu | no one here will help you with that |
| 06:30.29 | justinu | ~amp |
| 06:30.31 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 06:30.50 | tletourneau | OK, I didn't know, sorry. |
| 06:30.57 | justinu | it's ok |
| 06:31.05 | justinu | we can help you with asterisk stuff still |
| 06:32.07 | tletourneau | OK, let me check my logs for the invite. |
| 06:32.56 | *** part/#asterisk tletourneau (n=tom_remo@12-219-187-158.client.mchsi.com) |
| 06:34.37 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 06:37.47 | *** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz) |
| 06:41.41 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:45.29 | *** join/#asterisk welles (n=welles@222.90.170.148) |
| 06:46.36 | *** join/#asterisk Mw3 (n=mw3@national.t-error.hu) |
| 06:51.40 | remiss | good morning! |
| 06:52.20 | *** join/#asterisk Falle (n=Falstaf@falle.se) |
| 06:53.58 | Guest^DJ | hi, is there a command like Playtones(ring),15 ie. play ring for 15 seconds |
| 06:54.53 | remiss | PLaytones(ring); Wait(15); StopPlaytones(); |
| 06:54.58 | *** join/#asterisk Falle (n=falle@falle.se) |
| 06:55.42 | Guest^DJ | tried that, but that is not for my application |
| 06:56.30 | remiss | well... i can't help you if you don't tell me why that is not for your application... |
| 06:57.01 | Qwell | Guest^DJ: You don't need to play ringing...the endpoint does that on it's own |
| 06:57.03 | Qwell | its |
| 06:57.45 | Guest^DJ | remiss: i get about 10 sec of silence when connecting to pstn, instead of 10 sec silence, i want * to play generate a tone to caller |
| 06:58.08 | Qwell | Guest^DJ: Is it not playing it on its own? Show me your Dial() line |
| 06:59.10 | remiss | Guest^DJ: how about a "please wait while i try to connect you" :p |
| 06:59.23 | Guest^DJ | exten => _00.,1,Dial(SIP/alpha/${EXTEN}|45|rt) |
| 06:59.29 | Qwell | Guest^DJ: take out the damn r |
| 06:59.44 | Guest^DJ | what does the r do ? |
| 06:59.49 | Qwell | it fucks with ringing |
| 06:59.56 | Guest^DJ | lol |
| 07:00.02 | Qwell | Where did you get that example? |
| 07:00.07 | Qwell | I'm gonna go smack somebody |
| 07:00.14 | Guest^DJ | been googling |
| 07:00.22 | Qwell | show me the link you found it on... |
| 07:00.26 | FLeiXiuS | Is it possible to login 1 line with several phones? |
| 07:00.33 | Guest^DJ | hangon |
| 07:00.36 | Qwell | FLeiXiuS: sure, but it won't do much good |
| 07:00.43 | Qwell | only the last one to register will get calls |
| 07:00.55 | FLeiXiuS | Qwell: Arg, thats the problem I'm having. |
| 07:01.06 | Qwell | FLeiXiuS: intended behavior |
| 07:01.08 | Guest^DJ | Qwell: http://www.marko.net/asterisk/archives/0301/0836.html |
| 07:01.15 | FLeiXiuS | Qwell: My main reason is to have a call come through and be forwarded throughout all of the phones. |
| 07:01.26 | Qwell | FLeiXiuS: Dial(SIP/1&SIP/2&SIP/3) |
| 07:01.32 | Guest^DJ | exactly problem as mine |
| 07:01.41 | DarthClue | FLeiXiuS, ...yeah, like that...a dial group |
| 07:01.57 | FLeiXiuS | Qwell: Grr, they need to define these better in the wiki's ;-) |
| 07:02.02 | remiss | Guest^DJ: anyhow... you really shut put in a playback(pls-hold-while-try); befor dialing |
| 07:02.02 | FLeiXiuS | thanks :-) |
| 07:02.33 | remiss | s/shut/should/ |
| 07:02.47 | DarthClue | it's been so long since i've worked with this stuff...is the wiki any better or is it still a convoluted mess? |
| 07:02.47 | Guest^DJ | remiss: did tried, user dislike |
| 07:02.47 | remiss | ooohh.. fancy.. :) |
| 07:03.13 | remiss | damn.. someone broke my english today.. |
| 07:03.57 | remiss | Guest^DJ: users.. don't pay much attention to them.. |
| 07:04.19 | Qwell | DarthClue: still a convoluted mess, generally |
| 07:04.40 | Guest^DJ | remiss: they felt very ignoring |
| 07:05.10 | Guest^DJ | Qwell: after removing the r, no 'fake' tone from * |
| 07:06.10 | [hC] | sup guys |
| 07:13.19 | remiss | any music to recommend? |
| 07:16.45 | DarthClue | XM Satellite radio, but I've not yet had a chance to pipe into asterisk |
| 07:17.35 | mzo | does anyone have a clue why fwd is saying registration rejected for the last week? |
| 07:25.23 | *** join/#asterisk astar` (n=astar@ANantes-154-1-35-154.w81-53.abo.wanadoo.fr) |
| 07:25.36 | [av]bani | w00t |
| 07:27.24 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 07:27.50 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:29.41 | tzafrir | hi |
| 07:30.55 | mzo | is anyone familiar with FWD and why it'd be not working? |
| 07:34.01 | FLeiXiuS | How does asterisk map extensions to names on the speed dials? |
| 07:41.44 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:44.20 | macaco | Hey guys, morning! Has anyone been using the svn asterisk? there's a huge memory leak when establishing zap/sip channels, 132 bytes to be precise, corresponding to the pool string allocation..... anyone noticed this? |
| 07:45.22 | *** join/#asterisk pengyong (n=lala@210.21.33.60) |
| 07:47.33 | Juggie | macaco, does it exist in 1.2 |
| 07:49.17 | *** join/#asterisk anandbabu (i=ab@69-12-132-138.dsl.static.sonic.net) |
| 07:49.23 | *** join/#asterisk GBLRA (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:49.57 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:50.35 | anandbabu | when i am alone in the conference mode, i get strange noises. when some one joins, it becomes clear. has anyone eperienced this problem? |
| 07:52.41 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:55.37 | tzafrir | macaco, you said this on the -dev channel yesterday. Any reply? |
| 07:56.07 | tzafrir | macaco, could you be more specific? how exactly did you spot this leak? |
| 08:05.41 | tzafrir | anandbabu, is Meetme used with the flag M (music on hold when alone)? |
| 08:06.01 | tzafrir | does musiconhold generally work? |
| 08:08.19 | anandbabu | tzafrir, i am new to asterisk. i havent used any M flag |
| 08:08.53 | anandbabu | tzafrir, yes i have used M flag |
| 08:09.00 | anandbabu | tzafrir, exten => 1100,1, Wait(1) |
| 08:09.01 | anandbabu | exten => 1100,2,MeetMe(|MD) ; NOTE: If you add the option 'e', * will choose room #\ for you. Change the 'd' option to 'D' if you want to have a pin number for the conf\erence. |
| 08:09.54 | FLeiXiuS | How would I define a message to the users when they entered an extension not in the dial plan. |
| 08:12.16 | tzafrir | anandbabu, so you need to fix your musiconhold. |
| 08:12.19 | anandbabu | tzafrir, i removed M flag and now its silent. Thanks for the help |
| 08:12.23 | tzafrir | I'm away right now |
| 08:17.59 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-61.claranet.co.uk) |
| 08:20.47 | *** join/#asterisk lorinc (n=ang@caracas-1192.adsl.interware.hu) |
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| 08:37.46 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 08:39.07 | razu | is it normal that snom doesn't give a damn about the conf file which i've setted up in some location ? |
| 08:42.48 | razu | the problem is, that the snom reads it, but doesn't upgrade it's configuration ... :( |
| 08:42.59 | razu | anyone have any idea why is it like that ? |
| 08:50.22 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 08:50.54 | FuriousGeorge | how come ur not using the web interface? |
| 08:55.17 | *** join/#asterisk astar` (n=astar@ANantes-154-1-9-85.w81-53.abo.wanadoo.fr) |
| 08:56.07 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:57.57 | *** join/#asterisk saftsack (n=saftsack@p54A7E210.dip.t-dialin.net) |
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| 09:00.30 | [av]bani | \o> |
| 09:00.33 | [av]bani | <o/ |
| 09:01.41 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
| 09:05.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:06.23 | razu | FuriousGeorge : I need to make mass deployment to work ... I'm tired of configuring these phones over www ... |
| 09:07.04 | potsboy | hi all, is it possible to route based on DNID on a tdm card? |
| 09:07.33 | potsboy | from pstn |
| 09:07.49 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:07.51 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 09:09.46 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 09:10.06 | RoyK | morning, morons |
| 09:11.42 | ChrisUK | timmeh! |
| 09:16.08 | *** join/#asterisk WasPhantom (n=neil@203.86.197.11) |
| 09:17.50 | X-Gen | RoyK: its "morning freaks" ... morons is sooooo '70s |
| 09:18.01 | cypromis | lol |
| 09:18.10 | ChrisUK | living a lie timmeh |
| 09:18.15 | ChrisUK | ;-P |
| 09:19.48 | *** join/#asterisk saftsack (n=saftsack@p54A7E210.dip.t-dialin.net) [NETSPLIT VICTIM] |
| 09:22.01 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 09:22.52 | FuriousGeorge | is there an unofficial "preferred" scripting language to use w/ the asterisk api |
| 09:22.53 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 09:24.22 | *** join/#asterisk CKGC (i=CKGC@202.8.86.162) |
| 09:25.19 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 09:26.22 | CKGC | one question on iax |
| 09:27.29 | tzafrir | CKGC, ask away |
| 09:33.21 | *** join/#asterisk shido6 (n=ghicu6@d221-68-216.commercial.cgocable.net) |
| 09:34.47 | CKGC | tzafrir: I noticed that the "user" parameter in iax.conf is never used in the "register =>" part |
| 09:35.15 | CKGC | tzafrir: so what is it for? |
| 09:36.23 | tzafrir | CKGC, IIRC you can skip the register statement |
| 09:37.20 | tzafrir | BTW: I have the O'rreily book in front of me. It is a poor reference to iax.conf |
| 09:37.35 | tzafrir | The wiki is better |
| 09:37.40 | enemy^x | could anyone please explain to me how nationalprefix in zapata.conf works? |
| 09:39.23 | tzafrir | CKGC, I meant, that you can tell asterisk to use the data from the peer section for registration as well |
| 09:39.37 | tzafrir | I just don't remember how |
| 09:43.58 | *** join/#asterisk evilbuny (n=evilbunn@203-206-246-8.dyn.iinet.net.au) |
| 09:44.03 | saftsack | howto connect a big company with * to the telephone net? just with an E1 card? |
| 09:45.09 | Delvar | and some sticky back plastic |
| 09:45.48 | saftsack | back plastic? ^^ |
| 09:49.26 | kippi | how can I get asterisk to insert the area code if I only put in 6 numbers |
| 09:50.45 | evilbuny | kippi: exten => _XXXXXX,1,Dial(SIP/123${EXTEN}) |
| 09:50.45 | kmilitzer | Hi everyone. I need some help with dialplan magic. |
| 09:51.13 | kippi | evilbuny: do I need to tell it what the area code is or will BT do that for me? |
| 09:52.20 | remiss | 123 is the acode |
| 09:53.37 | kmilitzer | I have the following scenario: I send the call into a context where I decide for all extensions (_X.) with an AGI what to do with the call. If the AGI returns that the numbers does not exists, I send it into an context with only an i extension. So far everything is OK. But now I want to hangup and get a HANGUPREASON=1, meaning that the numbers does not exists, which then would make the SIP-Channel to generate a 404 Not Known message ... but somehow |
| 09:53.55 | kmilitzer | Any ideas how to do that? |
| 09:55.00 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 10:05.12 | *** join/#asterisk point (n=point@80.80.102.2) |
| 10:07.34 | *** join/#asterisk astar` (n=astar@ANantes-154-1-30-246.w81-53.abo.wanadoo.fr) |
| 10:08.42 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 10:12.02 | nextime | is it better to use DUNDILookup application or DUNDILOOKUP function? |
| 10:12.49 | anandbabu | do |
| 10:12.49 | anandbabu | <PROTECTED> |
| 10:12.50 | anandbabu | <PROTECTED> |
| 10:12.54 | anandbabu | <PROTECTED> |
| 10:12.58 | anandbabu | <PROTECTED> |
| 10:13.02 | anandbabu | <PROTECTED> |
| 10:13.06 | anandbabu | <PROTECTED> |
| 10:13.10 | anandbabu | echo "Using: $DIR" |
| 10:13.14 | anandbabu | done |
| 10:13.18 | anandbabu | echo |
| 10:13.31 | ChrisUK | >;) |
| 10:13.46 | nextime | anandbabu : continue isn't needed, mkdir is better with -p option :) |
| 10:24.38 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:28.51 | *** join/#asterisk KeX_WorX (n=chris@ng1.kurtkrenn.com) |
| 10:28.53 | KeX_WorX | hi |
| 10:29.08 | KeX_WorX | can i kill a sip channel which is a 'zombie' ? |
| 10:29.29 | KeX_WorX | the user completed the call, but the channel is still open |
| 10:29.36 | KeX_WorX | how do i get rid of this channel? |
| 10:29.46 | kmilitzer | soft hangup SIP/<whatever> |
| 10:30.00 | KeX_WorX | kmilitzer, thanks |
| 10:33.36 | KeX_WorX | and can i 'kick' a sip user out? |
| 10:35.39 | point | telco-64 connector - did anyone see that ? |
| 10:37.07 | kmilitzer | KeX_WorX: Sorry, don't know if that is possible ... try the sip comamnds on the cli |
| 10:37.21 | clive- | soft hangup should do it |
| 10:37.37 | KeX_WorX | kmilitzer, haven't found a cmd which would do that : / |
| 10:38.13 | KeX_WorX | i've a phone which registered nr 140, but the phone 'crashed' i re registered the phone under 141, but the phone is still under 140 reachabel |
| 10:38.34 | KeX_WorX | clive-, sonft hangup is only to close a channel? or? |
| 10:38.56 | clive- | just for a channel afaik |
| 10:39.11 | KeX_WorX | clive-, think so too |
| 10:39.20 | KeX_WorX | but to kick a user out? |
| 10:39.58 | clive- | well it will hang him up,....to deregister him, you need to modify your sip.conf and reload |
| 10:42.02 | KeX_WorX | clive-, thanks |
| 10:49.38 | remiss | what to do? |
| 10:56.45 | *** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
| 10:56.54 | pycsusz | Hi Everybody! |
| 10:58.12 | pycsusz | I have got a question again, how can I log into more conference room, with one extension? |
| 10:58.32 | pycsusz | If somebody can help me, then please do it! |
| 10:59.32 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 11:04.42 | *** join/#asterisk lele (n=lele@rivendell.ipv6.windmill.it) |
| 11:08.24 | X-Gen | :( |
| 11:09.02 | pycsusz | X-Gen You haven't got idea? |
| 11:09.41 | X-Gen | hey...i was checking my IRC connectivity there, i keep pinging out :P |
| 11:10.18 | pycsusz | X-Gen ohhh, I c |
| 11:10.39 | X-Gen | ooh nice, i'm getting pinged |
| 11:11.48 | kippi | exten => _9XXXXXX,1,Dial(Zap/g1/${EXTEN:1}) will that allow them to dial 9841571 and it will work out that its a local call |
| 11:11.50 | kippi | ? |
| 11:13.29 | pycsusz | no, it will just allow to dial 841571 |
| 11:13.58 | kippi | but BT will try and route that local? |
| 11:14.08 | pycsusz | yes |
| 11:14.13 | kippi | brill :) |
| 11:14.32 | pycsusz | it depend on your BT settings |
| 11:14.55 | X-Gen | someone should write a dialplan simulator thingum, that would answere plenty of questions |
| 11:15.18 | pycsusz | :) |
| 11:16.59 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 11:17.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:21.24 | RoyK | http://www.news.com.au/story/0,10117,18207565-13762,00.html |
| 11:22.15 | *** join/#asterisk bkw_ (n=bkw_@mc93bfa48.tmodns.net) |
| 11:23.14 | bkw_ | O M G |
| 11:23.16 | *** join/#asterisk jorgito (n=jorge@82.113.32.241) |
| 11:23.17 | jorgito | hi |
| 11:23.39 | X-Gen | TSA ? |
| 11:23.53 | bkw_ | morons with metal detectors at the airports |
| 11:23.58 | evilbuny | RoyK: ROFL |
| 11:24.17 | X-Gen | i would rather die than goto a .au hospital :P |
| 11:24.17 | *** join/#asterisk jdf (n=jon@m210e36d0.tmodns.net) |
| 11:24.17 | evilbuny | bkw_: it's not molesting if they're allowed to by law |
| 11:24.31 | jdf | heh |
| 11:24.36 | bkw_ | evilbuny, haha |
| 11:24.38 | jdf | bkw was jealous |
| 11:24.42 | jorgito | trying some instant messaging, and have idea if there is some messager (icq,msn,yahoo whatsever) which have some connector, for example i would like to inform by icq somebody that has voicemail etc. any ideas ? |
| 11:24.44 | evilbuny | X-Gen: erm? |
| 11:24.45 | bkw_ | well it was jdf that got molested |
| 11:24.50 | bkw_ | we are headed to LA :P |
| 11:24.53 | jdf | you wanted it |
| 11:24.58 | jdf | admit it |
| 11:24.58 | evilbuny | bkw_: I pity you |
| 11:25.00 | bkw_ | ewww no |
| 11:25.05 | evilbuny | LA = sux0r |
| 11:25.07 | bkw_ | pitty? |
| 11:25.09 | bkw_ | why? |
| 11:25.11 | bkw_ | no /me likes it |
| 11:25.19 | jdf | he was all polite as he grabbed my crotch |
| 11:25.23 | bkw_ | its better than oklahoma when you're only going to be there for 36 hours |
| 11:25.26 | evilbuny | bkw_: it's such a hole |
| 11:25.36 | evilbuny | I was there last week |
| 11:25.41 | bkw_ | jdf, the TSA dude |
| 11:25.46 | evilbuny | and the air was putride |
| 11:26.44 | jorgito | trying some instant messaging, and have idea if there is some messager (icq,msn,yahoo whatsever) which have some connector, for example i would like to inform by icq somebody that has voicemail etc. any ideas ? |
| 11:26.48 | benjk | the nice thing about LA is that it is pretty close to San Diego |
| 11:27.08 | jdf | LA is closer to god's blindspot |
| 11:27.09 | bkw_ | if you say so |
| 11:27.15 | evilbuny | benjk: and the mexican border/cheap booze :) |
| 11:27.43 | bkw_ | ok all these folks are sitting in the floor waiting to board... we have 30 more min... they act like the plane is going to run off and leave them if they don't hurry onto the plane |
| 11:27.51 | evilbuny | lol |
| 11:27.55 | bkw_ | dorks |
| 11:28.03 | evilbuny | bkw_: it's more a case of getting space for bags |
| 11:28.08 | evilbuny | before everyone else gets it |
| 11:28.13 | jdf | i think the guy across from bkw wants his cack |
| 11:28.14 | benjk | I was thinking about the nice restaurants in SD's Gaslamp quarter, but ok |
| 11:28.17 | bkw_ | they shouldnt' bring so fucking much |
| 11:28.19 | evilbuny | especially if you have a couple of carry ons :) |
| 11:28.26 | bkw_ | jdf, ewwww |
| 11:29.05 | bkw_ | the TSA won't allow you past with more than one plus a personal |
| 11:29.11 | bkw_ | they turn your ass back to check that shit |
| 11:29.24 | evilbuny | bkw_: sucked in :) |
| 11:29.41 | evilbuny | and SFo |
| 11:29.48 | *** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com) |
| 11:29.52 | benjk | you guys need a high speed train system |
| 11:29.53 | evilbuny | and a bunch of other airports in between ohio and Australia |
| 11:30.06 | xorsysd | Anyone here use BroadVoice? |
| 11:34.17 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 11:39.38 | *** join/#asterisk Inkubot (n=inkubot@200.74.183.55) |
| 11:40.38 | xorsysd | I'm getting a 404 while trying to register, that means it can't connect? |
| 11:41.43 | Sebb | i have a problem with asterisk and 1&1 voip here in germany: when i do an outgoing call via 1&1, when the called party answers, i get a "488 not acceptable here" as error, and hear voice for a 1/10 second or so.. how can i find out which codecs their sip-server supports? or how can i fix it? ;) |
| 11:42.18 | jorgito | how do you send sms from asterisk ? |
| 11:42.27 | *** join/#asterisk zoneout (n=cjrm@81-178-22-214.dsl.pipex.com) |
| 11:43.26 | zoneout | Hey guys, I'm using asterisk in the UK with the Digium TDM400. My FXO/FXS setup works fine except that the remote end is VERY quiet, can I amp it up easily? |
| 11:46.17 | xorsysd | What does it mean when you get a 404 on sip register?? |
| 11:49.06 | Inkubot | i've got a litle problem.. |
| 11:49.29 | Inkubot | phoneA -> router -> internet -> router -> asterisk -> phoneB |
| 11:49.40 | Inkubot | when phoneA calls phoneB everything works fine.. |
| 11:49.57 | Inkubot | the problem is when phoneB calls phoneA.. it is one way audio |
| 11:49.58 | zoneout | Does anyone know how to turn up the volume on FXO/FXS? |
| 11:50.20 | Inkubot | only phoneB can hear phoneA |
| 11:50.20 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 11:50.25 | mutilator | put asterisk and phoneA as a DMZ on their respecitve routers |
| 11:50.27 | Inkubot | do you know where is the problem ? in what side ? |
| 11:52.50 | kaldemar | zoneout: take a look at rxgain and txgain in zapata.conf |
| 11:53.51 | Inkubot | mutilator ok.. |
| 11:54.24 | mutilator | and enable nat=yes in the sip config |
| 11:54.33 | Inkubot | yes.. it is enable |
| 11:54.36 | mutilator | k |
| 11:54.45 | Inkubot | for all the sip clients.. and for [general] too |
| 11:55.25 | *** join/#asterisk evilbuny (n=evilbunn@203-206-246-8.dyn.iinet.net.au) |
| 12:00.31 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 12:02.05 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 12:02.38 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:06.19 | zoneout | kaldemar: cheers :) |
| 12:09.25 | pif | tzafrir ? |
| 12:18.24 | *** join/#asterisk appelza (n=moo@dsl-145-222-254.telkomadsl.co.za) |
| 12:18.48 | appelza | hey, could someone perhaps tell me if there is a type of logfile that shows you when an extention is in use or not? |
| 12:19.05 | appelza | I want to write a system that can show users which extentions are busy or free... |
| 12:23.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:24.20 | austinnichols10 | appelza: check out http://www.voip-info.org/wiki-Asterisk+manager+API |
| 12:24.39 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 12:25.28 | burtonez | hello, anyone has working setup with two wifi AP + asterisk + SIP wifi phones ? how about hand over roam ? |
| 12:26.15 | clive- | appelza howzit, I dunno the answer, but when you try ring that extension, it will show busy |
| 12:27.53 | appelza | tnx |
| 12:28.13 | clive- | appelza are you an asterisk developer? |
| 12:30.16 | *** part/#asterisk point (n=point@80.80.102.2) |
| 12:31.15 | *** join/#asterisk stone (n=stone@debian/developer/stone) |
| 12:32.21 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 12:32.42 | stone | I'm having trouble register my softphone (x-lite/kphone/ekiga) to my asterisk. Getting Request timeout on Registration. |
| 12:33.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:36.11 | appelza | whats asterisk's interface say when you try connect with the softphone? |
| 12:36.21 | stone | my conf: http://pastebin.com/564888 |
| 12:36.30 | appelza | clive- : no I'm not |
| 12:36.52 | appelza | Just want to make a nice interface that will show users when they can transfer a call or not based on the extention usage |
| 12:43.45 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 12:44.28 | jhiver | hi all |
| 12:46.08 | *** join/#asterisk diego_br (n=diego@200.208.241.178) |
| 12:49.48 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
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| 13:00.55 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 13:01.34 | _Paulo_ | Hi people... |
| 13:03.21 | _Paulo_ | Last night I was able to fax using an old fashioned modem connected to a TDM400P instead of app_txfax. |
| 13:04.16 | _Paulo_ | I think I still have a high error rate, but at least the transmition was completed. |
| 13:05.30 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:05.30 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 13:08.40 | FlyboySR22 | Good Morning Everyone |
| 13:12.12 | ChrisUK | Morning :P |
| 13:12.20 | ChrisUK | although its 13:17 here ;) |
| 13:12.21 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 13:13.05 | appelza | afternoon ;) |
| 13:13.28 | appelza | Come on...I cant be the only person who wants to be able to see when an extention is busy or not!?! :/ |
| 13:15.08 | stoffell | appelza, can't you see that? |
| 13:15.15 | _Paulo_ | appelza, just run "asterisk -r -x show channels" |
| 13:15.37 | _Paulo_ | appelza, if you want something simple... |
| 13:16.30 | zoneout | Is there an easy way in asterisk to detect when a Dial() has been answered? * seems to skip onto the next line in the dialplan before the lines answered. |
| 13:16.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:18.19 | kippi | is there away to lock down an extension so that you have to enter a pin to enable out going calls |
| 13:18.36 | appelza | thnx ill try |
| 13:19.07 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 13:20.20 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:21.06 | IronHelix | yeah kippi thats easy |
| 13:21.28 | IronHelix | make an exten that can only dial DISA and put a password on it |
| 13:21.34 | IronHelix | then DISA it to a context that can dial out |
| 13:21.48 | kippi | cool |
| 13:21.49 | IronHelix | DISA is an application that when you authenticate it will give you a dialtone for another context |
| 13:21.56 | kippi | I'll look that up |
| 13:22.00 | IronHelix | see also the cmd Authenticate() (just asks for a password) |
| 13:22.07 | zoneout | kippi: Not that I know of, but you could probably use AstDB and some custom dialplan stuff to do it |
| 13:22.22 | IronHelix | so you could do exten (pattern),1,Authenticate(code) then exten (pattern),2,Dial(${EXTEN}) |
| 13:23.46 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:24.32 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:24.59 | *** join/#asterisk fugitivo (n=ajf@201.255.176.40) |
| 13:26.18 | zoneout | cool, I never knew that was there... |
| 13:26.41 | IronHelix | also |
| 13:26.44 | zoneout | does anyone know how to detect a Dial() has been answered, and delay the dialplan until it has? |
| 13:26.45 | stone | Trying to register kphone to asterisk but are getting timeouts, I see that there are communication between the client and (*) (looke like the softphone is trying to auth until timeout is reached) |
| 13:26.55 | IronHelix | VMAuthenticate() will authenticate based on voicemail passwords |
| 13:26.58 | IronHelix | might be more useful |
| 13:27.03 | zoneout | On a Zap line... |
| 13:27.35 | IronHelix | thats HARD |
| 13:27.54 | zoneout | IronHelix: it is? |
| 13:27.56 | IronHelix | only way to tell (talking theoretically) is to listen for lack of ringing and IIRC zap channel doesnt do that |
| 13:28.06 | IronHelix | unlike voip channel which signals when the call picks up |
| 13:28.16 | IronHelix | analog channel does not tell you |
| 13:28.27 | zoneout | IronHelix: I heard there was an extension to do it... |
| 13:28.54 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:29.16 | IronHelix | Current local time is Tuesday Feb 21 08:29.16 AM -0500 GMT |
| 13:36.44 | *** join/#asterisk virterm (n=virterm@shiva.kanatek.com) |
| 13:41.42 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:46.37 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 13:51.04 | *** join/#asterisk Modcuts (n=info@proporta.gotadsl.co.uk) |
| 13:52.33 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 13:53.16 | *** join/#asterisk coppice (n=chatzill@242.192.17.210.dyn.pacific.net.hk) |
| 13:54.11 | _Paulo_ | coppice, Last night I was able to fax using an old external modem connected to a TDM400P instead of app_txfax. |
| 13:54.46 | _Paulo_ | coppice, lots of errors, 1 pages takes about 3 minutes. |
| 13:54.49 | coppice | wonderful. now repeat it :-) |
| 13:55.44 | fugitivo | lol |
| 13:55.58 | mutilator | O_o |
| 13:56.00 | fugitivo | _Paulo_: i have no problems faxing with a tdm400p |
| 13:56.02 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 13:56.15 | mutilator | i don't have many problems faxing |
| 13:56.20 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 13:56.30 | mutilator | even people on crappy wireless links seem to do it more than 50% of the time |
| 13:57.02 | _Paulo_ | mutilator, I not using VoIP, just PSTN |
| 13:57.16 | mutilator | yea me too |
| 13:57.19 | mutilator | kinda |
| 13:57.21 | fugitivo | _Paulo_: do you have noise on your line? |
| 13:57.26 | mutilator | well i guess not |
| 13:57.35 | mutilator | i have a t1 sent out over wireless |
| 13:57.57 | mutilator | channelized and all |
| 13:58.01 | coppice | fugitivo: that's one of those statements that has people soolishly rushing out to buy stuff :-) |
| 13:58.15 | mutilator | yea |
| 13:58.17 | _Paulo_ | fugitivo, for voice the lines are wonderful |
| 13:58.33 | fugitivo | coppice: what statement? |
| 13:58.36 | mutilator | and "software echo cancel has worked for me in 1000's of installations" is also one of the statements |
| 13:58.39 | fugitivo | coppice: noise? |
| 13:58.57 | coppice | <fugitivo>_Paulo_: i have no problems faxing with a tdm400p |
| 13:59.12 | fugitivo | coppice: it's true :) |
| 13:59.31 | coppice | maybe, but don't expect such success to be reproducible |
| 13:59.35 | _Paulo_ | coppice, with app_txfax I get "Phase 3 error" |
| 13:59.42 | fugitivo | coppice: why not? |
| 14:00.04 | coppice | have you seen how much trouble people have with fax and TDM400 cards? |
| 14:00.16 | fugitivo | _Paulo_: does echo cancellation goes to OFF when faxing? |
| 14:00.32 | _Paulo_ | I have echo cancelation off by default. |
| 14:00.54 | fugitivo | coppice: i had problems, but it was line's fault, reducing the noise solved the problem |
| 14:01.00 | coppice | _Paulo_: great sounding voice is not always much of a measure of line quality |
| 14:01.13 | _Paulo_ | sure... |
| 14:01.32 | coppice | fugitivo: if you don't get timing slips, then you are lucky |
| 14:01.56 | _Paulo_ | I have distinct results with different numbers and hardwares. |
| 14:02.19 | coppice | I used to have a voice line at home. that always sounded great. however, in wet weather a modem would not stay connected for more than a minute or two :-) |
| 14:02.22 | _Paulo_ | The worst are faxservers. |
| 14:02.24 | fugitivo | coppice: i have no problems with the x100p neither (crappy one) |
| 14:02.45 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-251.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:02.48 | coppice | the x100p should be OK for FAX. there is nothing crappy about it at all |
| 14:02.51 | _Paulo_ | I mean, the worst are PC/Modem based fax servers. |
| 14:03.12 | tzafrir | coppice, I built libunicall today. There were several warnings emiited. E.g: in one case an init function does not have return. Are you aware of those? |
| 14:03.21 | tzafrir | anyway, got to go |
| 14:03.44 | coppice | which version? |
| 14:04.41 | tzafrir | 0.0.3 pre8 |
| 14:04.44 | caio1982 | tzafrir: my unicall packages? |
| 14:04.52 | caio1982 | tzafrir: didnt get any error here |
| 14:05.07 | tzafrir | not error. but some warnings |
| 14:05.28 | tzafrir | on debian sarge |
| 14:06.02 | caio1982 | tzafrir: so you'll sync my unicall packages? :) |
| 14:06.31 | *** join/#asterisk cyonics (i=cyon@tx-71-52-77-114.dhcp.sprint-hsd.net) |
| 14:07.29 | fugitivo | coppice: what's the problem with the tdm400 for faxing? technically speaking |
| 14:07.40 | *** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
| 14:07.41 | coppice | frame slips |
| 14:08.05 | coppice | could be a driver problem, as they seem to come and go with different revisions |
| 14:08.33 | telenieko | Hi ppl. I'm trying to setup an E1 link on my asterisk box, but I have a nice RED alarm... where on the box can I see what is causing that RED alarm? (tried pri show span, pri intense debug span, zttooll...) but no clue on what's wrong ;( |
| 14:08.53 | fugitivo | ok, i'm using 1.0.8 on that box |
| 14:08.55 | coppice | tzafrir: i don't get any warnings building the current libunicall, and I don't think I have been getting any warnings since the early days |
| 14:09.11 | _Paulo_ | coppice, I would like to investigate the causes of my faxing problems. Where should I begin to look? |
| 14:09.29 | _Paulo_ | coppice, Do you heave some advice? |
| 14:09.45 | fugitivo | _Paulo_: if this helps, i use 1.0.8 in the box for faxing |
| 14:10.17 | _Paulo_ | fugitivo, asterisk 1.0.8 ? |
| 14:10.30 | fugitivo | yes |
| 14:10.33 | fugitivo | and zaptel |
| 14:11.04 | _Paulo_ | fugitivo, I'm using asterisk 1.2.4, unicall and spandsp snapshots from 2006-02-05 |
| 14:11.30 | fugitivo | _Paulo_: i don't have a box with tdm400 and 1.2.4 to test |
| 14:11.54 | fugitivo | _Paulo_: i know that the box with the tdm400 is using 1.0.8, maybe you should try that version and check if it works |
| 14:11.58 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:12.11 | fugitivo | if it works, maybe coppice is right and the problem is in the drivers |
| 14:12.22 | asteriskmonkey | is there a function in asterisk to nominalize rx and tx gains on the fly? |
| 14:13.34 | _Paulo_ | fugitivo, TDM400 is somewhat working |
| 14:13.41 | coppice | i did the early development of the modems in spandsp using a fax machine and a tdm400 for testing. it used to work very well. now with the same computer and tdm400 card things don't work at all. the OS as well as * have changed multiple times in that period |
| 14:14.01 | fugitivo | _Paulo_: i'm talking about faxing with tdm400 |
| 14:14.30 | iCEBrkr | yo yo yo |
| 14:14.40 | fugitivo | hey |
| 14:14.49 | _Paulo_ | fugitivo, Is, I able to fax |
| 14:15.04 | fugitivo | _Paulo_: what's the problem then? :) |
| 14:15.24 | _Paulo_ | fugitivo, the * server is at a colocation facilit |
| 14:15.35 | _Paulo_ | they charge by cable. |
| 14:15.43 | _Paulo_ | so i want to fax using E1 |
| 14:15.52 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 14:15.55 | _Paulo_ | instead of analog lines. |
| 14:16.08 | fugitivo | damn, i though that all this conversation was because a problem with tdm400 and fax, lol |
| 14:16.24 | macaco | I discovered the memory leak in asterisk (svn-head) in the following way: run asterisk, connect a sip channel to meetme conference, disconnect, shutdown asterisk... leaks 132 bytes of memory for each allocated channel, and this corresponds to the pool allocated in the function ast_string_pool_init called by ast_string_field_init..... valgrind traces this much: |
| 14:16.25 | macaco | ==18969== 952 bytes in 7 blocks are definitely lost in loss record 78 of 104 |
| 14:16.25 | macaco | ==18969== at 0x4A1BD7D: calloc (vg_replace_malloc.c:279) |
| 14:16.25 | macaco | ==18969== by 0x47AC82: __ast_string_field_init (utils.h:285) |
| 14:16.26 | asteriskmonkey | spandsp for the win :d |
| 14:16.26 | macaco | ==18969== by 0x41C0A3: ast_channel_alloc (channel.c:609) |
| 14:16.28 | macaco | ==18969== by 0x69FF866: ??? (chan_sip.c:2860) |
| 14:16.30 | macaco | ==18969== by 0x6A28E2B: ??? (chan_sip.c:10543) |
| 14:16.32 | macaco | ==18969== by 0x6A29AA0: ??? (chan_sip.c:11176) |
| 14:16.34 | macaco | ==18969== by 0x6A2B4BB: ??? (chan_sip.c:11309) |
| 14:16.36 | macaco | ==18969== by 0x41148E: ast_io_wait (io.c:285) |
| 14:16.38 | macaco | ==18969== by 0x6A23F8A: ??? (chan_sip.c:11457) |
| 14:16.40 | macaco | ==18969== by 0x4C28B1B: start_thread (in /lib/libpthread-2.3.5.so) |
| 14:16.42 | macaco | ==18969== by 0x5341051: clone (in /lib/libc-2.3.5.so) |
| 14:16.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.56 | fugitivo | macaco: #asterisk-dev && pastebin |
| 14:17.48 | asteriskmonkey | is there a function in asterisk to nominalize rx and tx gains on the fly? |
| 14:18.40 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 14:19.00 | [TK]D-Fender | macaco : Please do NOT spam like that again : use Pastebin |
| 14:19.01 | [TK]D-Fender | ~pb |
| 14:19.03 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:19.53 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 14:20.10 | pjz | Why am I getting ZT_SPANCONFIG failed on span 1: Invalid argument (22) ? |
| 14:20.16 | pjz | I don't have a span 1 defined in my zaptel.conf |
| 14:20.37 | macaco | fffah, ok... sorry ;) |
| 14:22.36 | asteriskmonkey | is there any know issues with asterisk and AMD? |
| 14:22.44 | *** join/#asterisk shog (n=rene@62.8.240.129) |
| 14:25.07 | shog | hello, i am looking for a free web based GUI for Asterisk. It should have an administrator interface to configure the server and also a user interface so the users can configure their phones. |
| 14:25.27 | shog | any recommendations? |
| 14:25.34 | fugitivo | anything else? |
| 14:26.28 | shog | graphical output of the server status would be nice |
| 14:27.02 | ChrisUK | your not going to find it free >;) |
| 14:27.28 | coppice | asteriskmonkey: yep. the issue is it works better than with intel CPUs |
| 14:28.01 | fugitivo | coppice: is that true? |
| 14:28.04 | shog | ChrisUK, why not? |
| 14:28.18 | ChrisUK | well im running it on 2 Dual Core AMD Opterons and it works perfect :/ |
| 14:28.24 | *** join/#asterisk lunaphyte_ (n=lunaphye@c-71-193-101-146.hsd1.mi.comcast.net) |
| 14:28.25 | coppice | its true for me, especially when the AMD machine runs 64 bit Linux |
| 14:28.36 | fugitivo | shog: you have 2 options for free, you use a crappy web interface like amp, or you develop your own |
| 14:28.46 | coppice | an X2 running spandsp blows away a xeon |
| 14:28.56 | fugitivo | coppice: hmm, i like opteron servers, but some people said it's not good for asterisk |
| 14:29.12 | coppice | i really don't understand why its so much faster |
| 14:29.15 | ChrisUK | well its fine for me got a weeks uptime so far with no issues |
| 14:29.41 | fugitivo | damn, then i'm going to start using opteron servers and not xeon |
| 14:29.58 | coppice | i've been amused recently to hear people in really huge conservative companies talk about xeons for the simple 2 cpu boxes, and opterons for anything serious :-) |
| 14:30.14 | shog | fugitivo, how come there are no decent free GUIs? |
| 14:30.30 | fugitivo | shog: make one |
| 14:30.46 | ChrisUK | people don't need them that often thats why |
| 14:31.03 | fugitivo | coppice: that's true, a lot of companies doesn't like to hear amd for servers |
| 14:31.28 | fugitivo | cool, since last upgrade my x100p is detecting hangup :) |
| 14:31.57 | *** join/#asterisk Abbas (n=Abbas@203.81.220.90) |
| 14:32.03 | coppice | fugitivo: i think you misread what I wrote |
| 14:32.11 | shog | is there a good user interface, then? |
| 14:33.00 | pjz | the CLI |
| 14:33.02 | fugitivo | coppice: oops, sorry |
| 14:33.28 | wasim | we'd even help fund it ... |
| 14:33.31 | fugitivo | coppice: here amd is not the choice for big companies |
| 14:33.32 | coppice | wasim: what is required? |
| 14:33.42 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 14:33.45 | wasim | coppice: 240 g729 channels on a dual opteron |
| 14:34.09 | [TK]D-Fender | shog : What do you expect out of a GUI and have your tried AMP yet? |
| 14:34.38 | coppice | fugitivo: several fortune 500 companies I've heard about recently have now decided xeons are only for the mickey mouse jobs, where they use Dell boxes. For 4 and 8 way they are using opteron |
| 14:35.24 | [TK]D-Fender | coppice : Actually "Mickey Mouse" (Disney/Pixar) use AMD IIRC :) |
| 14:35.26 | shog | [TK]D-Fender, i want to configure and administer the server. and i want a user interface so the regular users can configure their phones |
| 14:35.50 | fugitivo | coppice: that's good, i'd worked with opterons and i think they're great |
| 14:35.54 | *** join/#asterisk kreilmeier (n=kreilmei@h195202155002.fix.cm.kabsi.at) |
| 14:35.56 | [TK]D-Fender | shog : Since when do USERS configure their phones? And which models do you have in mind and what would the GUI do for them? |
| 14:36.08 | fugitivo | coppice: i used them as a server for diskless terminals |
| 14:36.26 | fugitivo | coppice: diskless terminals with kde+sound+all desktop capabilities |
| 14:36.45 | clive- | wasim how about PLC with g729 codec ? |
| 14:37.02 | shog | [TK]D-Fender, maybe i should try AMP first and see if it suits me. |
| 14:37.15 | appelza | Could anyone tell me why : asterisk -r -x "show channels" > /tmp/aslog works... |
| 14:37.16 | appelza | but |
| 14:37.26 | appelza | $st = `asterisk -r -x "show channels" > /tmp/aslog`; |
| 14:37.33 | appelza | doesnt work, I end up with a blank file :/ |
| 14:37.43 | wasim | fugitivo: we use icewm for diskless, kde is too fat |
| 14:38.18 | wasim | clive-: i'm waiting for sangoma to release g729 on their quad-e1 dsp |
| 14:38.41 | *** join/#asterisk newl (n=newlook@203-59-100-129.dyn.iinet.net.au) |
| 14:38.51 | fugitivo | wasim: well, the performance of those diskless terminals was awesome with kde, openoffice and all that crap, it was a dual opteron with 4gb ram and 30 diskless terminals (i couldn |
| 14:38.57 | fugitivo | (i couldn't try more) |
| 14:39.43 | *** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net) |
| 14:40.08 | fugitivo | (gentoo amd64) |
| 14:40.09 | [TK]D-Fender | wasim : which product of theirs transcodes direct to g729? |
| 14:40.16 | wasim | fugitivo: the opteron has nothing to do with the performance of the diskless, other than the nfs bit |
| 14:40.39 | fugitivo | wasim: are you kidding me? all the applications are run on the server |
| 14:40.59 | wasim | fugitivo: then we saw that kde transferred about 80 MB, while icewm was under 6, so when you boot 100 call center agents all at the same time, it helps |
| 14:41.01 | fugitivo | no local applications |
| 14:41.12 | wasim | fugitivo: then its not diskless, they are dumb-terminals |
| 14:41.18 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 14:41.30 | fugitivo | i though it was the same |
| 14:41.46 | wasim | fugitivo: diskless is when you boot off the network, but use local processing |
| 14:41.58 | fugitivo | that depends |
| 14:42.06 | fugitivo | for sound you need local processing |
| 14:42.06 | tracinet | hello all - quick question - just noticed a change from older ver. of asterisk - when a sip device is in a queue and has it call-forwarded - the device does not ring as part of queue - with v.1.2x it sends the call to the CFWD - is there a way to revert to old behavior? |
| 14:42.06 | wasim | fugitivo: dumbterm is when you just transport the display and all the processing happens at the server |
| 14:42.23 | fugitivo | wasim: you can have a mix of local and remote |
| 14:42.38 | coppice | dumbterm is G.W.Bush emulation |
| 14:42.45 | wasim | fugitivo: indeed you can |
| 14:42.49 | fugitivo | wasim: what's the name of that? :) |
| 14:42.58 | wasim | fugitivo: diskless |
| 14:43.08 | coppice | pointless? |
| 14:43.14 | fugitivo | then, it was diskless and not dumb :) |
| 14:43.16 | _Paulo_ | appelza, try $st = `asterisk -r -x "show channels"` and parsing $st after instead of reading from the file. |
| 14:43.23 | *** join/#asterisk Atreta (n=root@20150189118.user.veloxzone.com.br) |
| 14:43.29 | Atreta | Hi all |
| 14:43.53 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
| 14:44.12 | fugitivo | coppice: i don't think it's pointless |
| 14:44.14 | Atreta | can you guys give me a hand with asterisk cdr? i need it to show the billsec too in amp |
| 14:44.28 | Atreta | people in #amportal told me to come here |
| 14:44.36 | fugitivo | ~amp |
| 14:44.38 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
| 14:45.21 | Atreta | i go to ARI to get the call details but it only shows me the duration, i need the billsec too |
| 14:45.43 | potsboy | i think that module is areski not amp.. its part of ast@home |
| 14:46.03 | Atreta | yes |
| 14:46.25 | fugitivo | Atreta: anything related web interface has nothing to do with asterisk, it's just an interface |
| 14:46.27 | Atreta | jbot i was already there |
| 14:46.42 | potsboy | youll have to do a SQL select and edit the php yourself as it does not have that functionality |
| 14:46.44 | Atreta | fugitivo are you brazilian? |
| 14:46.47 | fugitivo | Atreta: try to modify the application |
| 14:46.52 | fugitivo | no |
| 14:46.58 | Atreta | hmmmmm ok |
| 14:47.01 | Atreta | thanks guys |
| 14:47.11 | fugitivo | it should be easy to add a field to the select |
| 14:47.33 | *** join/#asterisk hhoffman (n=hhoffman@n1-33-115.dhcp.drexel.edu) |
| 14:47.35 | Atreta | i'm a total noob at mysql and php |
| 14:47.36 | Atreta | :) |
| 14:47.49 | Atreta | i'm brazilian too |
| 14:48.01 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 14:48.04 | iCEBrkr | werd |
| 14:48.10 | astar` | someone has a idea on my asterisk on inbound pstn call the quality is bad but not with incoming voip calls |
| 14:48.15 | hhoffman | hi, is it possible to fax over a DID provided Teliax with rx_fax and tx_fax? |
| 14:48.16 | _Paulo_ | Atreta, lucky guy! |
| 14:48.24 | Atreta | hehe |
| 14:48.28 | astar` | and it's not a pstn prob and hardware problem |
| 14:52.55 | caio1982 | coppice: is there any known bug related to unicall/mfcr2 about 100% of cpu consumption? after 10minutes the load goes from 40% of cpu used to 100% and stays there (using g729 even forced transcoding sip2sip works fine) |
| 14:53.19 | caio1982 | (30 channels being tested for converting to cas/r2) |
| 14:53.45 | *** join/#asterisk sl16 (n=blah@tv.neterra.net) |
| 14:53.57 | coppice | there used to be issues when something feeds packets greater than 20ms to it, but I thought those problems had been fixed. I haven't received complaints for a long time |
| 14:54.09 | sl16 | hello |
| 14:54.18 | sl16 | <PROTECTED> |
| 14:54.28 | sl16 | i have 2 licenses for g729, sip.conf is ok ... i have no idea .. |
| 14:54.36 | sl16 | suggestions ? |
| 14:54.56 | caio1982 | coppice: i'll check libraries/protocols version in the buggy machine, thanks steve |
| 14:56.26 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 14:56.33 | jhiver | Hey list |
| 14:56.39 | jhiver | I don't understand something |
| 14:56.50 | jhiver | I do a NoOp(${CALLERID}) |
| 14:56.56 | jhiver | and it gives me two values? |
| 14:57.04 | jhiver | NoOp("SIP/mediant.ykoz.net-08129800", "692828070") |
| 14:57.06 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 14:57.35 | jhiver | ?? |
| 14:58.06 | _Paulo_ | in /etc/zaptel.conf, I using "span=1,1,0,cas,hdb3" |
| 14:58.07 | [TK]D-Fender | jhiver : No, thats just 1 value |
| 14:58.24 | _Paulo_ | should I use "span=1,0,0,cas,hdb3"instead? |
| 14:58.39 | [TK]D-Fender | jhiver : It always shows you which CHANNEL is initiating the Application |
| 14:58.50 | jhiver | aaah ok I understand |
| 14:58.52 | jhiver | cheers |
| 14:59.12 | jhiver | oh it seems that callerid works then :) |
| 14:59.15 | jhiver | cool |
| 14:59.32 | jhiver | now I can go code that callback agi script |
| 15:00.19 | [TK]D-Fender | :) |
| 15:00.46 | jhiver | I'm quite impressed with this audiocodes box I've bought... the stuff actually works |
| 15:01.02 | jhiver | g.729, caller id, echo cancel... phew |
| 15:01.04 | [TK]D-Fender | jhiver : What kind of gear? |
| 15:01.11 | jhiver | audiocodes mediant 2000 |
| 15:01.21 | jhiver | SIP <-> PRI 8 E1 gateway |
| 15:01.24 | coppice | audiocodes have been selling stuff for quite a while. they should have sorted it out by now |
| 15:01.48 | [TK]D-Fender | jhiver : WOW.. big league gear |
| 15:01.55 | jhiver | it's nice kit |
| 15:02.12 | jhiver | I did consider a bunch of servers with Sangoma / Digium... |
| 15:02.19 | jhiver | but I went for Audiocodes instead |
| 15:02.24 | [TK]D-Fender | jhiver : I've heard the config & docs are somewhat shitty, but that it does WORK. Does this describe your experience? |
| 15:02.37 | jhiver | it's 1U high only, has dual power supply, dual ethernet nics... |
| 15:02.39 | jhiver | yes it is |
| 15:02.45 | jhiver | I got a support contract too |
| 15:02.46 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
| 15:02.51 | paulhuynh | goodmorning |
| 15:02.57 | jhiver | so setup was easy, i.e. "make it work please" |
| 15:02.58 | jhiver | :) |
| 15:02.59 | paulhuynh | good morning everyone |
| 15:03.02 | jhiver | hi |
| 15:03.07 | paulhuynh | i have a few question |
| 15:03.24 | jhiver | I'm impressed with audiocodes voice quality too |
| 15:03.29 | paulhuynh | was i broke my * and built a new one yesterday |
| 15:03.46 | jhiver | Now if I could find a way to improve the quality of these damn * prompts... |
| 15:03.48 | paulhuynh | but i don't know all the deviced password |
| 15:04.03 | paulhuynh | the one that is not from within my office |
| 15:04.22 | paulhuynh | they all try to regsiter but got a user/pas wrong |
| 15:04.31 | jhiver | paulhuyn, are these password sent in clear or with md5? |
| 15:04.49 | paulhuynh | so how can i use sip debug to finger out the password and recreate the ext. |
| 15:05.02 | paulhuynh | well i'm not sure |
| 15:05.06 | paulhuynh | i think it md5 |
| 15:05.12 | kippi | has anyone got a good how to install the diguim IAXy |
| 15:05.13 | paulhuynh | becuase i can read it at all |
| 15:05.27 | jhiver | if it's md5 you're screwed, if it's sent clear text it's possible |
| 15:05.39 | paulhuynh | oh ok |
| 15:05.46 | paulhuynh | that and i have another question |
| 15:06.01 | paulhuynh | is there a conflict problem with pap2 with asterisk? |
| 15:06.07 | jhiver | ? |
| 15:06.13 | jhiver | I got the linksys to work with * |
| 15:06.23 | jhiver | after some faffing about |
| 15:06.25 | paulhuynh | i got two of them on the field and now they are not willing to register |
| 15:06.34 | jhiver | are you using stun? |
| 15:06.42 | paulhuynh | they was for a moment and now they stop |
| 15:06.44 | jhiver | and nat=no? |
| 15:06.52 | paulhuynh | nat=yes |
| 15:06.55 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 15:06.58 | paulhuynh | i have stun |
| 15:07.02 | paulhuynh | i have no stun |
| 15:07.12 | _Paulo_ | coppice, any advice about fixing my faxing? |
| 15:07.15 | jhiver | ah well... that must be why |
| 15:07.26 | jhiver | try it with STUN it usually improve things |
| 15:07.32 | paulhuynh | use g711u and slow down the speed |
| 15:07.44 | paulhuynh | for registering? |
| 15:08.11 | jhiver | yep and you need to "ping" the atas regularily to keep them registered |
| 15:08.20 | jhiver | I think there is an option... is it keepalive? |
| 15:08.24 | coppice | _Paulo_ its much easier fixing receive problems, since I can get logs files from people for study :-) |
| 15:08.29 | jhiver | set that both on the device and in your sip.conf |
| 15:09.03 | jhiver | otherwise the UDP ports of the NAT device will be closed if there is no traffic |
| 15:09.31 | jhiver | also make sure to force re-registration quite frequently (i.e. every hour or so) to handle changing ip addresses |
| 15:09.35 | _Paulo_ | coppice, Receiving works very well for me. |
| 15:09.50 | jhiver | or do like me and get a fritz!fonbox and forget about nat issues :) |
| 15:09.52 | CoffeeIV_ | I have a fax machine on an internal extension (via an ATA). In my dialplan, I have a System() command that notifies me of the fax, which is after the RxFax() command. Faxes from outside work fine, but faxes from the internal ext hangup and stop dialplan processing after the fax is sent but before the System() command is run. Any way to make it keep processing, then hang up ? |
| 15:10.38 | coppice | _Paulo_ there is something I suspect as causing timing issues for some people. maybe I should cook up a test version which changes that, and see how it goes for you |
| 15:10.56 | _Paulo_ | coppice, how can I gather useful information? |
| 15:11.29 | paulhuynh | but right now it said can't connect to server |
| 15:11.34 | paulhuynh | or off hook |
| 15:11.37 | paulhuynh | status |
| 15:11.57 | IronHelix | coffeeIV- but the same cmd on extension H |
| 15:12.01 | IronHelix | might get run twice then |
| 15:12.05 | IronHelix | *tho |
| 15:12.09 | coppice | that's the problem. the really useful information isn't available in your box. its in the other box. if you have two boxes to use rxfax and txfax between you might be able to gather something useful |
| 15:12.41 | CoffeeIV_ | IronHelix: I'll try it -- I can put something in the script that detects if it's being run twice on the same file -- thanks |
| 15:12.45 | _Paulo_ | I will work a setup like that. |
| 15:13.04 | IronHelix | it depends on the system command |
| 15:13.33 | IronHelix | if the system command is a script or something you could make it first check to see if its already executing... |
| 15:13.38 | paulhuynh | what command do i type is asterisk cli to get it to show sip debug for register |
| 15:13.46 | IronHelix | sip show registry |
| 15:13.49 | IronHelix | shows outbound registrations |
| 15:13.52 | IronHelix | sip show peers |
| 15:14.03 | IronHelix | might be what you want |
| 15:14.05 | _Paulo_ | coppice, I have 2 E1s from distinct telcos. |
| 15:14.25 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 15:14.37 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:14.49 | CoffeeIV_ | IronHelix: yeah, it "files" the .tif in a certain place, I can have it check for a lock file or something |
| 15:15.17 | _Paulo_ | coppice, should I put one board into another server, or can i use the same box? |
| 15:15.24 | coppice | fine. then you edit t30.c, uncomment the first line, recompile and install. when you fax you will get audio files in /tmp. get the problem to happen, and send me the audio files from both ends |
| 15:15.48 | eKo1 | I'm trying to hangup this channel using 'soft hangup' but it won't hang up. Is there anything else I can do to destroy this channel short of restarting Asterisk? |
| 15:15.58 | IronHelix | what channel is it? |
| 15:16.00 | coppice | try what's easiest first of all. if that doesn't shed light, we can try something else |
| 15:16.12 | IronHelix | remember if its a sip channel, you have to put in the full channel ie soft hangup SIP/1234-xxxx |
| 15:16.22 | _Paulo_ | coppice, You rulez. |
| 15:17.24 | mocker | Sweet. |
| 15:17.35 | mocker | Just signed up for an asterisk boot camp. |
| 15:17.57 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 15:17.58 | IronHelix | you will be assimilated, resistance is fut.... i mean cool! |
| 15:18.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:18.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:18.07 | mocker | IronHelix: :) |
| 15:18.11 | Nivex | The resistance will not be televised. |
| 15:18.12 | eKo1 | IronHelix: It is a SIP channel and I did that already. |
| 15:18.33 | Nivex | You can, hoever, join the concall :) |
| 15:18.33 | IronHelix | lol |
| 15:18.51 | IronHelix | eko- try soft hangupping the other end of the channel? or is it on a script? |
| 15:19.16 | eKo1 | There is no other end, that is the funny part. |
| 15:19.25 | IronHelix | what does shwo channels give you |
| 15:19.46 | IronHelix | *show channels |
| 15:19.47 | eKo1 | SIP/pstn-d01a (pstn-in 1 ) Up Bridged Call SIP/300-6706 |
| 15:20.12 | IronHelix | do: soft hangup SIP/pstn-d01a then soft hangup SIP/300-6706 |
| 15:20.16 | IronHelix | or did you do both of those |
| 15:20.36 | eKo1 | soft hangup SIP/pstn-d01a gives me |
| 15:20.37 | eKo1 | Requested Hangup on channel 'SIP/pstn-d01a' |
| 15:20.40 | eKo1 | and |
| 15:21.06 | eKo1 | soft hangup SIP/300-6706 |
| 15:21.07 | eKo1 | SIP/300-6706 is not a known channel |
| 15:21.21 | IronHelix | that is odd |
| 15:21.27 | eKo1 | no kidding |
| 15:21.31 | paulhuynh | does any one know where i can get some info to setup a pap2 from linksys to my asterisk@home |
| 15:21.31 | IronHelix | sip show peer pstn ? |
| 15:21.35 | paulhuynh | stun server? |
| 15:22.09 | eKo1 | I think I'll just have to restart Asterisk |
| 15:22.19 | IronHelix | does sip show peer pstn give you anything? |
| 15:22.19 | *** join/#asterisk littleall (n=littleba@cm140.epsilon174.maxonline.com.sg) |
| 15:22.20 | exonic | do you still have to run 'make linux26' for zaptel-1.2.4 on kernel 2.6 ? |
| 15:22.25 | eKo1 | IronHelix: sure |
| 15:22.27 | IronHelix | no you dont, make works fine |
| 15:22.40 | exonic | IronHelix, good deal. |
| 15:22.56 | littleall | hello, i am looking for good website about audio knowledge. who can recommend one? |
| 15:22.59 | IronHelix | it also seems to self configure udev on centos |
| 15:23.21 | IronHelix | i dunno, try restart when convenient so it waits until there are no other channels |
| 15:23.47 | IronHelix | bbl |
| 15:24.13 | remiss | littleall: audio knowledge? |
| 15:24.45 | littleall | remiss, yes, i want to understand more about audio encoding/decoding etc.... |
| 15:24.58 | remiss | en.wikipedia.org might be a good place to start |
| 15:25.01 | exonic | Does TDMoE actually work? They keep it in the asterisk.org/features but last I knew it sucked |
| 15:25.21 | *** join/#asterisk bkw_ (n=bkw_@m9436fa48.tmodns.net) |
| 15:25.25 | remiss | if you are thinking about differences between ulaw/alaw/gsm/g7** etc |
| 15:25.28 | *** join/#asterisk junglicious (n=jungle@206-225-86-167.dedicated.abac.net) |
| 15:25.56 | littleall | remiss, yes. |
| 15:26.22 | eKo1 | audio knowlegde? |
| 15:26.46 | eKo1 | i think you should consult the chapter on sound on any physics book |
| 15:26.50 | asteriskmonkey | Ok anyone here got a PRI and a digium card? |
| 15:26.58 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:27.00 | paulhuynh | HELP please |
| 15:27.01 | eKo1 | many a folk |
| 15:27.03 | paulhuynh | what does this mean |
| 15:27.04 | asteriskmonkey | ive been fighting with echo on the damn thing for 8months now |
| 15:27.05 | paulhuynh | http://pastebin.ca/42659 |
| 15:27.19 | eKo1 | asteriskmonkey: which digium card? |
| 15:27.36 | asteriskmonkey | eko1: i have a 406 and 110 |
| 15:27.48 | remiss | paulhuynh: no password? |
| 15:27.53 | eKo1 | and this is happening on both? |
| 15:28.03 | asteriskmonkey | eko1: yes |
| 15:28.09 | paulhuynh | well but i have type in the password into the pap2 |
| 15:28.11 | paulhuynh | correctly |
| 15:28.15 | paulhuynh | wha happpen |
| 15:28.31 | asteriskmonkey | eko1: so basically the uberly expensive te406 helped my issue 0 |
| 15:28.50 | eKo1 | try disconnecting one and see if that changes anything |
| 15:30.09 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 15:31.12 | asteriskmonkey | eko1: there not both connected |
| 15:31.20 | asteriskmonkey | eko1: only 1 in the machine at a time |
| 15:31.43 | asteriskmonkey | its inconsistant echo.. sometimes no echo then it creeps in 10mins later |
| 15:33.23 | paulhuynh | http://pastebin.ca/42660 |
| 15:33.28 | paulhuynh | ok no what? |
| 15:33.38 | paulhuynh | now what i shoudl do from here? |
| 15:33.54 | asteriskmonkey | the only thing i can think it is the rx/tx gain levels are incorrect, but ive tried changing them all over the place.. and watching the lione volume.. i guess only thing left to do is get my 1004hz sound test from the telco :P |
| 15:33.58 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 15:34.10 | *** join/#asterisk jdf (n=jon@m610e36d0.tmodns.net) |
| 15:34.28 | asteriskmonkey | then if that dosnt work thow my digium cards at the nearest geak and get an audiocodes gateway :D |
| 15:35.49 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:36.18 | brodiem | I have a question about the Queue command. This may be a bug, but if I do a "exten => 123,1,Queue(SomeQueue||||)", then I get put into the queue for only about a couple seconds (I hear the MoH), and then that queue "fails" if there are no agents available to take the call, and it goes to the next option in the dial plan. HOWEVER, if I use "exten => 123,1,Queue(SomeQueue)", then the call stays in the queue and I continue to hear MoH unti |
| 15:36.55 | brodiem | Using the (SomeQueue||||) is how AMP creates the dial plan, which is why I'm asking, since I want the caller to remain in the queue until someone is available |
| 15:37.58 | brodiem | so basically it goes to whatever you set the "fail over destination" in AMP for that queue right away (and timeout is set to 0 = unlimited) |
| 15:39.05 | brodiem | am I talking to myself? :) |
| 15:39.28 | *** join/#asterisk littlejohn (n=little@ppp-62-11-216-77.dialup.tiscali.it) |
| 15:39.44 | *** join/#asterisk diego_br (n=diego@200.208.241.178) |
| 15:40.03 | exonic | How would one go about proving to your telco that they are overriding your outgoing caller id? I have a pri debug on my spam and it explicitly shows the caller id being set. |
| 15:40.20 | exonic | but the receiving end gets the wrong # |
| 15:40.25 | [TK]D-Fender | brodiem : I believe you skip over the timeout value thus CAUSING a timeout. What you leave out the |||| it assumes NULL, not 0 and therefor does NOT time out. |
| 15:40.56 | brodiem | [TK]D-Fender, no it does the same when using Queue(SomeQueue|||0) |
| 15:41.20 | brodiem | AMP creates it as Queue(SomeQueue|t|||0), which also does this |
| 15:41.22 | GerbilWrk | anyone had experience loading Asterisk on a router, like the linksys wrt54g |
| 15:41.54 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 15:42.07 | asteriskmonkey | GerbilWrk: yes |
| 15:42.15 | asteriskmonkey | GerbilWrk: go to openwrt.org has instructions there |
| 15:42.28 | *** join/#asterisk DannyF (n=dannyf@dsl-cust-83-172-72-126.kringdata.net) |
| 15:43.28 | paulhuynh | http://pastebin.ca/42662 |
| 15:43.44 | docelm0 | YIPPIE! |
| 15:45.34 | brodiem | [TK]D-Fender, hmm just tried setting it to like 1200 ( Queue(SomeQueue|t|||1200) ) and it seems to work fine... wonder why setting 0 for unlimited doesn't work |
| 15:46.03 | [TK]D-Fender | 0 does not equal NULL. I found that out the hard way... this should be changed... |
| 15:46.15 | asteriskmonkey | what type of hammer would you use to smash a te-serious card to bits? |
| 15:46.21 | jarrod | what in the SIP header carries caller-id information |
| 15:46.24 | Nugget | nothing should equal NULL, not even NULL. |
| 15:47.40 | brodiem | [TK]D-Fender, and AMP only allows you to set a 20 minutes max, I guess I'll just change AMP amp so that it writes an insanely high number if you choose "0 minutes" from the drop down... hard to believe that such a limitation would be in the code for so long |
| 15:47.43 | *** join/#asterisk Abbas (n=Abbas@203.81.220.90) |
| 15:48.14 | [TK]D-Fender | AMP = suck |
| 15:48.39 | brodiem | [TK]D-Fender, is there anything better out there? I don't care so much about having a GUI but management does =/ |
| 15:48.45 | asteriskmonkey | AMP=SUCKS SO HARD IT SUCKS A BASKET BALL THORUGH A STRAW.. |
| 15:48.53 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:49.28 | eKo1 | ouch |
| 15:49.32 | eKo1 | you don't want that |
| 15:49.37 | [TK]D-Fender | brodiem : Best GUI I've seen = www.scopserv.com. Tell them Andrew sent you :) |
| 15:49.39 | *** part/#asterisk junglicious (n=jungle@206-225-86-167.dedicated.abac.net) |
| 15:49.49 | iDunno | ~amp |
| 15:49.51 | jbot | from memory, amp is NOT supported here! people using it should join #amportal |
| 15:50.10 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-144.prod-infinitum.com.mx) |
| 15:50.22 | salviadud | need help with a sipura... |
| 15:50.37 | salviadud | anybody own a spa-841? |
| 15:50.48 | brodiem | [TK]D-Fender, its hard to justify with management why they would need to pay for a GUI when there is a free one available =/ |
| 15:51.00 | asteriskmonkey | true |
| 15:51.12 | asteriskmonkey | defeats open source concep |
| 15:51.33 | asteriskmonkey | although i know joel has put huge work into scopserv |
| 15:51.35 | [TK]D-Fender | brodiem : Not when you ask their opinion on "you get what you pay for" :) |
| 15:52.01 | asteriskmonkey | its totally a good idea for a newb to get scopserv |
| 15:52.03 | asteriskmonkey | :D |
| 15:52.13 | brodiem | [TK]D-Fender, lol what's the price for a license? |
| 15:52.22 | [TK]D-Fender | asteriskmonkey : Yeah well I push Joel pretty hard... I'm the reason for all sorts of those improvements as I've been helping him debug everything :) THEM I might work for.... |
| 15:52.34 | [TK]D-Fender | brodiem : Not sure, but not too much... phone them up... |
| 15:53.00 | salviadud | so, whats the name of the GUI that does NOT cost money? |
| 15:53.10 | asteriskmonkey | the would be a m p |
| 15:53.17 | salviadud | amp? |
| 15:53.19 | salviadud | or a m p |
| 15:53.23 | [TK]D-Fender | brodiem : the ONLY reason I got a GUI here was because MGMT wanted something they though they could manage even though they have no hopes of learning enough to use it themselves, so really *I'm* stuck with it :) |
| 15:53.58 | salviadud | thats funny man, a bunch of suites that don't know what SIP stands for want to manage a PBX |
| 15:54.14 | brodiem | [TK]D-Fender same deal here, we're migrating from an existing analog PBX and they want it to be simple to manage like the existing one |
| 15:54.34 | brodiem | [TK]D-Fender so you're using ScopServ? |
| 15:54.57 | [TK]D-Fender | brodiem : My tip (guerilla warefare style) : Get AMP, let them transition and the DITCH it once you highlight the limitations.... |
| 15:55.01 | [TK]D-Fender | brodiem : Indeed. |
| 15:55.20 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:55.29 | *** join/#asterisk |Vulture| (n=Vulture@82.115.205.68.cfl.res.rr.com) |
| 15:55.36 | asteriskmonkey | my tactic.. get aah@home for ease of install on a new system .. then remake your own conf files and update src |
| 15:55.48 | brodiem | well I'm at least AMP writes the includes for adding your own custom rules so that I can at least make my own edits and not have AMP overwrite them when re-writing configs, but it's just a PITA working around AMPs limitations |
| 15:56.29 | [TK]D-Fender | brodiem : ScopServ DESTROYS AMP. Utterly. It's the best SHIT you can find for * these days :) |
| 15:56.52 | brodiem | cool..ever use PBXware? wondering how it matches with that |
| 15:58.00 | *** join/#asterisk vmedrano (n=vmedrano@196.32.128.206) |
| 16:00.01 | paulhuynh | scopServ |
| 16:00.12 | paulhuynh | anyone here use it? |
| 16:01.01 | paulhuynh | i really like switchvox demo so far |
| 16:01.24 | paulhuynh | PBXware + scopeserv cost alot of money for a small bus. |
| 16:02.14 | Cool_One | 3 |
| 16:02.30 | salviadud | why does everyone need a GUI? |
| 16:02.42 | [TK]D-Fender | paulhuynh : I guess it just depends on what your idea of expensive is, especially considering the price/functionality of digital ""key" systems... |
| 16:02.44 | cpm | because we are stupid |
| 16:02.57 | salviadud | just use the freakin' CLI with a framebuffer |
| 16:03.00 | [TK]D-Fender | salviadud : To make "suits" happy... |
| 16:03.20 | salviadud | fender's right... |
| 16:03.21 | [TK]D-Fender | cpm : Memory leak :) |
| 16:03.25 | salviadud | those damn suites... |
| 16:03.26 | cpm | heh |
| 16:03.41 | paulhuynh | yes you are right plus most GUI come as a complete system and they are over price |
| 16:04.33 | [TK]D-Fender | paulhuynh : How much? compare that to consulting fees to have someone setup and MAINTAIN * for you..... |
| 16:04.56 | paulhuynh | so far voiceone seem to be very promise |
| 16:05.07 | asteriskmonkey | man all i gotta say to all you whiny buggers out there.. you have to pay money to make money there is not rags to riches scheme in that data/telco market |
| 16:05.24 | asteriskmonkey | you wannt buy cheap phones.. youll enjoy agony of the problems they come with |
| 16:05.37 | *** join/#asterisk Sloboda (n=slob@194.42.196.254) |
| 16:05.42 | asteriskmonkey | you want to skimp on the server .. when you call volumes get up youll pay for that too |
| 16:05.52 | salviadud | the monkey is riiiiight |
| 16:06.00 | salviadud | yep, i agree |
| 16:06.25 | cpm | everyone uses their cellphones here in the US anyway, they don't need pbxes any more. |
| 16:06.27 | salviadud | if you're lazy in the linux community, you will pay for it |
| 16:06.30 | Sloboda | Hi! I am looking for softphone that supports usb-phone. Could anyone advise something? |
| 16:06.43 | salviadud | cpa buddy, I'm not in the US |
| 16:06.55 | cpm | good! |
| 16:07.02 | *** join/#asterisk santiago (n=santiago@63.245.86.179) |
| 16:07.03 | salviadud | haha |
| 16:07.07 | cpm | Where? |
| 16:07.13 | salviadud | good ol' mexico |
| 16:07.34 | cpm | Ahh, then you are suffering from cellphonitis also probably. |
| 16:07.46 | salviadud | kinda... |
| 16:07.50 | salviadud | i'll tell you this |
| 16:07.55 | salviadud | we need some damn laws here |
| 16:07.56 | Sebb | i have a problem: when i do an outgoing call, it sometimes fails.. every time the ser on the other side connects me to one specific provider.. a sip log is at http://rafb.net/paste/results/HukoVI24.html - why does asterisk destroy the call in line 189..? |
| 16:08.02 | salviadud | for example |
| 16:08.11 | salviadud | i can get a cellphone that "connects" to the internet |
| 16:08.13 | salviadud | but |
| 16:08.18 | salviadud | i can only get ports 8080 |
| 16:08.19 | paulhuynh | so what do you recommend to use PBXware. switchvox, or scopescerv |
| 16:08.22 | salviadud | and 25 |
| 16:08.26 | cpm | So do we, actually, we need exactly the opposite. We need to repeal all the crap that was passed to give the cellphone companies a free pass. |
| 16:08.39 | salviadud | i can't get the "whole" internet |
| 16:08.57 | *** join/#asterisk SpooForBrains (n=wolf@82-36-140-168.cable.ubr02.perr.blueyonder.co.uk) |
| 16:09.20 | SpooForBrains | Hey all. Could anyone help we with getting NAT traversal working for SIP? |
| 16:09.41 | cpm | ! |
| 16:09.53 | SpooForBrains | ? |
| 16:09.54 | salviadud | what's cool about asterisk, is i can call my house from my cellphone, then call my cousin in california |
| 16:10.00 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:10.11 | salviadud | its waaaay cheaper than long distance. |
| 16:10.42 | cpm | cool. |
| 16:10.43 | Sloboda | SpooForBrains, I tried iptables sip modules. They are at alpha state, really, and reboot the router during sip-registration. |
| 16:10.44 | salviadud | and in mexico, we like cheap |
| 16:10.57 | cpm | everyone likes cheap. |
| 16:11.07 | cpm | esp those who can afford not to be cheap. |
| 16:11.10 | salviadud | yeah, but, do you like tacos? |
| 16:11.29 | SpooForBrains | Sloboda: so there's no way for me to get my SIP phone working with NAT? |
| 16:11.58 | SpooForBrains | Situation is, I have a SIP phone, the office has an asterisk server, I want to get my phone talking to their server, and I have NAT at home |
| 16:12.00 | salviadud | tacos and asterisk maaaaan, thats the mexican-american dream for me |
| 16:12.12 | cpm | depends. many many years ago, in a previous life, a few hours south of mexico city, I stopped into a canteena, they had some tacos up for grabs, I grabbed. Just about did me in on tacos forever. |
| 16:12.22 | Sloboda | SpooForBrains, I'm newbie at VoIP. Well, you may use sip-proxy, * for example ;-) |
| 16:12.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:12.26 | [Airwolf] | salviadud, don't forget the coke |
| 16:12.27 | salviadud | oh my god, you got some balls there |
| 16:12.38 | salviadud | i would never get tacos in a cantina |
| 16:12.42 | cpm | I was much younger then. |
| 16:13.36 | salviadud | the coke is from colombia... not mexico |
| 16:14.06 | cpm | SpooForBrains, http://www.voip-info.org/wiki-STUN |
| 16:14.12 | salviadud | well... i'm not sure. i don't sell it or buy it so... |
| 16:14.15 | [TK]D-Fender | SpooForBrains : describe your network chain (ex : IP Phone ---> Private LAN ----> * ---> NAT Router) for your scenario |
| 16:16.12 | SpooForBrains | ( IP phone ) --- (switch) --- ( SuSEfirewall2 box ) --- interntet --- (unknown) --- (asterisk box) |
| 16:16.16 | *** join/#asterisk SomePBXUser (n=neil@96.Red-80-38-99.staticIP.rima-tde.net) |
| 16:16.27 | SomePBXUser | Hi! |
| 16:16.28 | SpooForBrains | SuSEfirewall2 box = NAT router |
| 16:16.36 | jarrod | is there a way to have SER modify the Caller-ID ? |
| 16:16.39 | SomePBXUser | Am I in the right place for newbie questions? |
| 16:16.51 | *** join/#asterisk adminguru (n=atze@fw0.prosem.net) |
| 16:17.14 | [TK]D-Fender | SpooForBrains : your SIP client definition should include : "QUALIFY=YES", "NAT=YES", and there is more if the * is behind NAT as well.. |
| 16:17.31 | [TK]D-Fender | SomePBXUser : Sure, ask away |
| 16:17.46 | SomePBXUser | :-) Thanks! |
| 16:17.53 | *** join/#asterisk chkbsd (n=chkbsd@p54AA43C0.dip0.t-ipconnect.de) |
| 16:18.01 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 16:18.04 | chkbsd | hi |
| 16:18.16 | SomePBXUser | I'm a little confused with what I actually need to set up a simple PBX (regarding hardware) |
| 16:18.18 | chkbsd | is chan_capi included in the 1.2.4 release or in the next? |
| 16:18.47 | SomePBXUser | I don't know if I need any special cards or not, since I just wan't to connect one ISDN line to the PBX to take incoming calls... |
| 16:18.51 | [TK]D-Fender | SomePBXUser : Describe the wiring and lines you have now. |
| 16:18.57 | SomePBXUser | Ok |
| 16:19.06 | SomePBXUser | ISDN goes to PBX, all the phones are IP |
| 16:19.19 | [TK]D-Fender | You have an IP telephony system already? |
| 16:19.25 | SomePBXUser | Nope :( |
| 16:19.28 | SomePBXUser | I'm trying though |
| 16:19.40 | [TK]D-Fender | ok, how many lines / phones will you need? |
| 16:19.50 | SomePBXUser | 1 ISDN line |
| 16:20.28 | SomePBXUser | Sorry (phone call) |
| 16:20.42 | SomePBXUser | Ok, it would be an internal network of IP phones |
| 16:20.59 | SomePBXUser | The idea is to be able to make outgoing calls over the ISDN line |
| 16:21.09 | *** join/#asterisk fiber0pti (n=John@c-68-35-13-238.hsd1.nm.comcast.net) |
| 16:21.14 | chkbsd | thats right ive asterisk with isdn |
| 16:21.33 | chkbsd | but in the i cant compile the chan_capi this time with 1.2.3 |
| 16:21.41 | [TK]D-Fender | SomePBXUser : Wait.. you're going to run your PBX on 1 line? For how many ext's? |
| 16:21.54 | SomePBXUser | hehe, 4 |
| 16:21.55 | chkbsd | 1.2.4 |
| 16:21.56 | SpooForBrains | Do I *require* STUN to traverse NAT? |
| 16:22.04 | SpooForBrains | I can't just forward ports? |
| 16:22.15 | SpooForBrains | (I'm sorry, I have my stupid head on) |
| 16:22.40 | [TK]D-Fender | SpooForBrains : no... you can forward ports in many cases. |
| 16:22.50 | *** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net) |
| 16:22.57 | [TK]D-Fender | SpooForBrains : can you confirm if your * server is behind NAT as well? |
| 16:23.26 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
| 16:23.33 | *** join/#asterisk coppice (n=chatzill@99.193.17.210.dyn.pacific.net.hk) |
| 16:23.54 | SpooForBrains | Heh, I was trying to figure out what * meant ... told you I had my stupid head on! |
| 16:24.32 | cpm | SpooForBrains, http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 16:24.43 | cpm | <PROTECTED> |
| 16:24.54 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 16:25.07 | Tagor | Hi, I am trying to setup my network for Asterisk on linux. But I am not really familar with subnets and broadcast. Can someone tell me what subnets and broadcast ip's I should use? |
| 16:25.12 | Tagor | This is an image of the network: http://www.mailfreeonline.com/uploader/8D812ECD.JPG |
| 16:25.13 | SpooForBrains | No, * server is not behind NAT |
| 16:26.30 | SomePBXUser | Tagor: Depends how many networks you have... a simple one to use is 192.168.1.0/24 (broadcast 192.168.1.255) this would let you use 254 IP addresses |
| 16:26.51 | Tagor | Just one network, SomePBXUser |
| 16:26.56 | Tagor | With about 5 computers |
| 16:27.27 | SomePBXUser | Ok, using the same IP scheme, your net is .0 and the broadcast IP is .255 |
| 16:28.33 | SpooForBrains | I don't see a situation on that page for me, which is * server public, me and my Grandstream SIP phone behind NAT |
| 16:30.51 | SomePBXUser | D-Fender, I'll refrase a little. Let's imagine I use * for managing 10 IP Phones from different locations. The idea is that outgoing calls can be made from any phone through *. |
| 16:31.12 | SomePBXUser | So basically I am unsure of what hardware I need, or if a simple BRI ISDN card will do the trick |
| 16:32.14 | *** join/#asterisk dezent (n=dezent@unixgeek.biz) |
| 16:32.44 | eKo1 | You either need an FXO card or a call terminator that will make the calls for you. |
| 16:32.44 | asteriskmonkey | outgoing calls are setup via contexts |
| 16:32.57 | brodiem | damn |
| 16:32.58 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.222) |
| 16:33.01 | Kernel_core | hi all |
| 16:33.02 | brodiem | sorry accidental pings |
| 16:33.31 | Kernel_core | anybody successfuly configured H323 on asterisk ?! |
| 16:34.46 | SpooForBrains | So, if the * server is public, and I'm behind NAT with me SIP phone, all I should need to do is forward the correct ports, yes? |
| 16:34.50 | Modcuts | is it possible to turn off the vm-intro.gsm on voicemail because if i remove it the vm brakes? |
| 16:36.20 | exonic | Modcuts, copy silence over it? |
| 16:36.41 | exonic | Modcuts, submit a patch =) |
| 16:36.52 | [TK]D-Fender | SomePBXUser : Well if you're looking for ISDN take a look at the AVM Fritz cards. They seemt o be the most popular ones these days (last I heard) |
| 16:37.08 | mikefoo | [TK]D-Fender: sup sup |
| 16:37.22 | SomePBXUser | Ok, so TDM01B would be ok? |
| 16:37.36 | [TK]D-Fender | mikefoo : ntm atm iykwim |
| 16:37.45 | [TK]D-Fender | SomePBXUser : not for ISDN. |
| 16:37.55 | SomePBXUser | Argh... |
| 16:38.00 | [TK]D-Fender | SomePBXUser : Digium TDM cards are for purely analog channels |
| 16:38.11 | [TK]D-Fender | SomePBXUser : Look at the AVM cards |
| 16:38.11 | SomePBXUser | Ok, thought so... |
| 16:38.52 | SomePBXUser | Yup, I've been looking at Fritz BRI Card from AVM. |
| 16:39.00 | mikefoo | [TK]D-Fender: what? lol |
| 16:39.09 | Modcuts | yeah but thats a quick fix and all other acccounts loose the message |
| 16:39.15 | SomePBXUser | Would I need any other hardware though? |
| 16:39.28 | mzo | yay, asterisk runs. yay |
| 16:39.59 | [TK]D-Fender | SomePBXUser : You'd need the PC to install * on (should be semi-decent, but nothing terribly special for the size you destribed), and the phones you intend to use. |
| 16:40.38 | SomePBXUser | Okidoki... that's about it really :-) |
| 16:40.46 | [TK]D-Fender | mikefoo : Not Too Much At The Moment I You Know What I Mean.... |
| 16:40.46 | SomePBXUser | Much appreciated! |
| 16:40.50 | [TK]D-Fender | SomePBXUser : np. |
| 16:40.59 | GerbilWrk | anyone have any recommendations to fix this. I tried installing four FXO cards in addition to the 4 port T1 card, when i did that, the T1 would not come up. I took them out, and the T1 still wouldn't come up. I rebooted several times, and eventually got the T1 up by turning up the second one also, which isn't plugged in |
| 16:41.10 | GerbilWrk | now I get this error though throughout the day, Feb 21 10:37:19 NOTICE[1567]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 16:41.13 | mzo | use the force. |
| 16:42.27 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:42.59 | dpolitech | stupid question |
| 16:43.11 | dpolitech | does anyone have a channel bank to recommend for use with *? |
| 16:43.14 | dpolitech | FXS |
| 16:43.30 | SpooForBrains | RTP would be the audio protocol, right? |
| 16:43.38 | *** join/#asterisk SupZ (n=icechat5@200-161-148-83.dsl.telesp.net.br) |
| 16:44.41 | kippi | how comes I am getting this error? *CLI> Feb 21 16:43:20 NOTICE[4161]: chan_iax2.c:3918 register_verify: No registration for peer '1001' (from 10.6.10.149) |
| 16:44.59 | kippi | do I have to add somthing in extensions.conf? |
| 16:45.21 | salviadud | i think that refers to the qualify option in iax.conf |
| 16:45.29 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 16:46.14 | salviadud | i don't think it has to do with extensions |
| 16:46.22 | Sebb | no one who speaks sip and can help me? ;) |
| 16:46.26 | *** part/#asterisk SomePBXUser (n=neil@96.Red-80-38-99.staticIP.rima-tde.net) |
| 16:46.55 | Sebb | last try - when i do an outgoing call, it sometimes fails.. every time the ser on the other side connects me to one specific provider.. a sip log is at http://rafb.net/paste/results/HukoVI24.html - why does asterisk destroy the call in line 189..? |
| 16:47.46 | kippi | i have put the conf in iax.conf |
| 16:48.20 | salviadud | well buddy, iax.conf is very important |
| 16:48.53 | salviadud | im guessing |
| 16:49.01 | salviadud | you are trying to reach 1001 |
| 16:49.06 | kippi | yeah |
| 16:49.18 | kippi | thats what I want the extension to be |
| 16:49.27 | salviadud | the extension is not the problem |
| 16:49.33 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 16:49.38 | salviadud | you don't have it registered |
| 16:49.48 | salviadud | and you don't have to |
| 16:49.55 | salviadud | you can just add a line that says qualify=nw |
| 16:50.01 | salviadud | qualify=no |
| 16:50.54 | salviadud | and, one question, 1001 is a sip phone? |
| 16:50.58 | salviadud | what is it? |
| 16:51.03 | kippi | do I need to add that to iax.conf? |
| 16:51.15 | salviadud | well... go to pastebin.ca |
| 16:51.21 | salviadud | and paste your iax.conf |
| 16:51.29 | salviadud | just take out the passwords |
| 16:51.29 | iDunno | ~pb |
| 16:51.31 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:51.32 | salviadud | or any personal stuff |
| 16:51.35 | [TK]D-Fender | dpolitech : Rhino seems to work OK... |
| 16:51.49 | dpolitech | yes |
| 16:51.52 | dpolitech | though none on ebay ;) |
| 16:52.13 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:52.23 | kippi | http://pastebin.ca/42670 |
| 16:52.47 | dpolitech | I guess that's what I'll get |
| 16:52.55 | dpolitech | have not seen anything negative about them, so... |
| 16:53.15 | salviadud | yeah, try that |
| 16:53.21 | salviadud | qualify=now |
| 16:53.24 | salviadud | i mean |
| 16:53.25 | salviadud | no |
| 16:53.26 | salviadud | not now |
| 16:54.25 | kippi | still the same |
| 16:54.34 | salviadud | did you reload? |
| 16:54.38 | kippi | yeah |
| 16:54.49 | salviadud | i guess we're both learning here aren't we... |
| 16:55.07 | salviadud | you might need to get better documented on iax |
| 16:55.11 | *** join/#asterisk gbodemantv (n=gbodeman@mail.televerde.com) |
| 16:55.13 | salviadud | i have FWD via iax |
| 16:55.22 | salviadud | let me see... |
| 16:56.10 | mzo | i've been trying fwd for iax for two days but it still says refused, i wonder if it really does take 24 hours for turning iax on to work |
| 16:56.20 | salviadud | how about |
| 16:56.20 | SibRw0rk | any get faxing to work with Asterisk? |
| 16:56.28 | salviadud | qualify=yes |
| 16:56.35 | jaiger | SibRw0rk, yes |
| 16:57.16 | SibRw0rk | jaiger: any special setup needed? |
| 16:57.36 | jaiger | not for me |
| 16:57.40 | kippi | nope |
| 16:58.08 | salviadud | damn... |
| 16:58.17 | SibRw0rk | jaiger: did you configure the ATA device just like you would anything else in extensions.conf? |
| 16:58.24 | salviadud | and the other * box you are calling |
| 16:58.39 | jaiger | SibRw0rk, ATA? |
| 16:58.47 | salviadud | how does it recognize you? |
| 16:58.58 | SibRw0rk | jaiger: yeah how you got the fax to connected to your ip network - u use a ATA device |
| 16:58.59 | jaiger | I have a channel bank and a T card |
| 16:59.03 | SibRw0rk | oh ok |
| 16:59.25 | SibRw0rk | so u just plug the regular fax into that, and gave it an extension that you've already supplied in your extensions.conf? |
| 16:59.44 | jaiger | but I have done fax through sipur ATAs. I made sure to force the codec to ulaw for that channel |
| 17:00.05 | SibRw0rk | jaiger: you hear about the T.38 patch ? |
| 17:00.10 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:00.15 | stoffell | any idea why there is no echo can board for a single span E1/T1 ? |
| 17:00.20 | jaiger | it works but I wouldn't recommend it for high quantity faxing |
| 17:00.42 | jaiger | stoffell, I have a Tellabs 2572 that does a single span |
| 17:00.55 | SibRw0rk | jaiger: how would you do high quality faxing - and don't say email2fax |
| 17:01.16 | stoffell | jaiger, i see, find it weird that the digium cards don't offer on-board echo can for the single span cards |
| 17:01.34 | *** part/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 17:02.04 | jaiger | SibRw0rk, never tried t38 but with the current state of technology I wouldn't recommend fax over IP for high quantity fax |
| 17:02.23 | jaiger | stoffell, probably not cost effective |
| 17:02.29 | SibRw0rk | jaiger: ok - thanks |
| 17:03.06 | jaiger | SibRw0rk, depends on your requirements but I might recommend HylaFAX to handle the IP network side of things |
| 17:03.30 | SibRw0rk | HylaFAX? |
| 17:03.45 | jaiger | SibRw0rk, yes a fax server for UNIX/Linux |
| 17:03.47 | kippi | anyone from digium around? |
| 17:03.59 | Modcuts | i did just wipe the vm-intro.gsm to get around her talking after the personal message, but thats a bit of bug having that on all vm. |
| 17:04.08 | SibRw0rk | jaiger: ok thanks |
| 17:04.24 | SibRw0rk | jaiger: it works well with asterisk? |
| 17:04.27 | jaiger | but last I checked it doesn't work with fax machines on the FXS side of the connection |
| 17:04.40 | jaiger | SibRw0rk, it's a stand-alone fax server for linux |
| 17:04.44 | SibRw0rk | oh |
| 17:04.44 | SibRw0rk | ok |
| 17:04.57 | SibRw0rk | i'll have to do some research into it |
| 17:04.58 | SibRw0rk | thanks man |
| 17:05.20 | jaiger | maybe iaxmodem too but I haven't tried that either. iaxmodem integrates hylafax and asterisk |
| 17:05.58 | jaiger | my own fax machines are on TDM connections and don't go over a VOIP link. that's intentional |
| 17:06.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:07.21 | *** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net) |
| 17:08.53 | _Paulo_ | coppice, I've done what you told me. Got 6 files, my original image and the received one. |
| 17:09.36 | coppice | OK. either post them somewhere and send me an e-mail pointing to them, or just e-mail the whole thing |
| 17:10.09 | _Paulo_ | coppice, thanks! |
| 17:15.40 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 17:15.45 | kippi | is there a run down of the voicemail options? |
| 17:16.22 | coppice | i thought the whole voice mail system was pretty run down :-) |
| 17:16.30 | cpm | Ouch! |
| 17:16.40 | brif8 | Can the Cisco 7920 Wireless IP Phone use either SIP and or IAX2 ? or has the asterisk SCCP support been upgraded over the last 8 months, that Asterisk can deal with SCCP directly ? |
| 17:17.59 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 17:18.12 | KriS83 | Hi |
| 17:19.32 | KriS83 | a really short and probably simple question... is it possible to use 1 sip account for multiple locations/multiple sip phones? SipUser A Logs on from IP 1.2.3.4 and 4.3.2.1. If Sip A is called, both phones ring? |
| 17:20.01 | iCEBrkr | KriS83: Won't work |
| 17:20.12 | exonic | KriS83, you want call queues |
| 17:20.15 | iCEBrkr | KriS83: But you can Dial(SIP/1&SIP/2) to make them both ring |
| 17:20.43 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 17:21.14 | KriS83 | hmm |
| 17:21.58 | KriS83 | Just out of interesst, how would sipgate support this? (obviously not using Asterisk right? :)) |
| 17:23.05 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
| 17:23.08 | docelm0 | KriS83, sipgate is asterisk |
| 17:23.34 | KriS83 | k, they support multiple logins... |
| 17:23.40 | KriS83 | how do they solve this? |
| 17:24.05 | katakefalos | could aomeone tell me if its safe to deploy an asterisk box with real time based on mysql for a production machine? |
| 17:24.06 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
| 17:24.41 | brif8 | KriS83: using the the two dial patterns like iCEBrkr said: Dial(SIP/1&SIP/2&.........) |
| 17:25.00 | Seldon1975 | hey |
| 17:25.01 | katakefalos | i have it as a stage machine but i have many issues with different devices (registration problems etc.) |
| 17:25.07 | [TK]D-Fender | kippi : Read the WIKI page, its pretty much print-and-distribute ready... |
| 17:25.21 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 17:25.44 | *** join/#asterisk zekeonfir3 (n=zekeonfi@jtzemp2.fttp.xmission.com) |
| 17:26.31 | katakefalos | is anyine in here that uses asterisk for authentication exlusiv with mysql or any other database? |
| 17:26.35 | KriS83 | brif8, doesn't make sence... cos If I logon twice with User A and Password 1234 how would there be a second extension? I would always dial SIP/A or not? |
| 17:28.01 | [TK]D-Fender | KriS83 : You CAN'T login as the same user in 2 different place. |
| 17:28.15 | *** join/#asterisk tletourneau (n=tom_remo@12-219-187-158.client.mchsi.com) |
| 17:28.32 | [TK]D-Fender | KriS83 : You need to make 2 SEPERAT accounts that will both be rung when the extension you want associated with them is dialed. |
| 17:29.35 | brif8 | KriS83: on the CLI run sip show peers, after each time you log in and you will see that it will register with the latest IP address |
| 17:29.57 | tletourneau | Hello everyone, is setting allowguest=yes in sip.conf the same as allowing non-proxy invites? |
| 17:30.13 | mzo | any fwd experts around? :) |
| 17:30.33 | *** join/#asterisk zekeonfir3 (n=zekeonfi@jtzemp2.fttp.xmission.com) |
| 17:30.54 | KriS83 | ok, I understand that.. but then docelm0 must be wrong or not? |
| 17:31.07 | KriS83 | Cos as I said sipgate does support this |
| 17:31.51 | gaupe | probably because they are running SER |
| 17:32.00 | [TK]D-Fender | KriS83 : Sipgate is not * based, right? Because what you're described is SIP-B "shared" line appearances. |
| 17:32.38 | KriS83 | ok... thats all I wanted to know :) |
| 17:32.52 | [TK]D-Fender | tletourneau : that allows un-auth'd calls to be accepted against the context specified in [general]. |
| 17:33.04 | [TK]D-Fender | And * does not support SIP-B yet. |
| 17:33.27 | [TK]D-Fender | (summer 2006 in * 1.4 is the current plan) |
| 17:33.41 | salviadud | what is sip-b? |
| 17:33.44 | KriS83 | That would be nice |
| 17:33.59 | KriS83 | [TK]D-Fender, thank you for the info |
| 17:34.07 | docelm0 | wrong about what? |
| 17:34.14 | salviadud | i believe we are using sip-a right now |
| 17:34.17 | KriS83 | docelm0, sipgate using * |
| 17:34.18 | salviadud | sip-b is better? |
| 17:34.32 | tletourneau | Thanks D-Fender, I'm trying to figure out how to get an FXO device to talk to my * box. |
| 17:35.12 | *** join/#asterisk ToTo (n=ToTo@host2-161.pool870.interbusiness.it) |
| 17:35.43 | [TK]D-Fender | tletourneau : Which? |
| 17:36.15 | tletourneau | A Vegastream Vega 50 with 8 FXO ports. |
| 17:37.25 | tletourneau | There doesn't seem to be alot of information out there about integrating these two devices. |
| 17:38.12 | tletourneau | I have the outbound working, I just can't figure out how to get the inbound working. |
| 17:38.26 | mzo | is this the wrong place to ask if someone wants to peer with me, so i can call finland? I can trade NJ to them? :P |
| 17:38.30 | coppice | _Paulo_ This TIFF goes wrong exactly as the picture starts. are you sure the problem is not with the PDF to TIFF conversion? |
| 17:38.32 | *** join/#asterisk IRC_User (n=anonymou@198.60.73.230) |
| 17:39.15 | docelm0 | KriS83, initially yes.. I know they were.. As of today good chance but who knows. |
| 17:39.20 | IRC_User | does anyone have experience with sccp in asterisk? I need some help setting up my sccp.conf for a 12SP |
| 17:39.36 | brif8 | Can the Cisco 7920 Wireless IP Phone use either SIP and or IAX2 ? or has the asterisk SCCP support been upgraded over the last 8 months, that Asterisk can deal with SCCP directly ? |
| 17:40.47 | puzzled | brif8: cisco does not do iax2. for sccp see chan-sccp.berlios.de |
| 17:41.03 | [TK]D-Fender | tletourneau : One would think that you should have it register to * just lieka phone |
| 17:42.01 | *** join/#asterisk saftsack (n=saftsack@p54A7EBE7.dip.t-dialin.net) |
| 17:42.02 | saftsack | hi |
| 17:42.08 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 17:42.22 | saftsack | is it possble to jump automatically in any extension without using immediate=yes? |
| 17:42.41 | Egonis | When I start asterisk, it quits out and notes a series of errors relating to permissions in /dev/zap in /var/log/asterisk/messages -- what do I do to change the perms? |
| 17:42.41 | IRC_User | looking for help with sccp and cisco 12SP |
| 17:43.02 | tletourneau | D-Fender : So just set it up as an extension in *? |
| 17:43.37 | mzo | what's latest version? 1.2.4? How'd i gte back down to 1.2.1? |
| 17:43.38 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 17:43.50 | nextime | Egonis : just do a "chown -R asterisk.asterisk /dev/zap" |
| 17:43.58 | salviadud | why would you want to go back down? |
| 17:44.02 | mzo | i don't know. |
| 17:44.09 | mzo | i must have messed up someting during an upgrade |
| 17:44.14 | nextime | ( assuming that your * run as user and group "asterisk" ) |
| 17:44.15 | *** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 17:44.15 | IRC_User | I use 1.2.4 and it works beautifully |
| 17:44.27 | salviadud | yep, me too |
| 17:44.35 | salviadud | 1.2.4 is nice |
| 17:44.53 | salviadud | is anyone here into social engineering |
| 17:44.55 | *** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 17:44.59 | salviadud | i know this company based in nevada... |
| 17:45.00 | IRC_User | yes |
| 17:45.08 | IRC_User | yes to social engineering |
| 17:45.09 | salviadud | i have the password that the technicians use |
| 17:45.26 | IRC_User | and they don't know you have it |
| 17:45.41 | salviadud | i used to work for a operator company based in mexico that gives outsourcing to this company |
| 17:45.47 | salviadud | of course they don't |
| 17:46.03 | salviadud | i can call them on their toll free number via FWD |
| 17:46.11 | [TK]D-Fender | tletourneau : Not certain on that model, but others like the SPA-3000 do. |
| 17:46.13 | salviadud | i want them to change the password |
| 17:46.20 | salviadud | which one should be fitting? |
| 17:46.33 | mzo | haha, i'm a dumbass |
| 17:46.44 | trixter | http://www.trxtel.com/index.php?page=Tollfree_Termination if you do enough tollfree traffic not only can you call free but you can make money off it too :P |
| 17:46.45 | salviadud | dumbass is a good password... |
| 17:46.46 | IRC_User | They should regularly, but likely that password has been un-changed since its birth |
| 17:46.47 | mzo | i must have used the tar from 1.2.1 from AAH when i upgraded and i downgraded from 1.2.3 to 1.2.1 |
| 17:46.47 | trixter | better than FWD for that reason |
| 17:46.49 | *** join/#asterisk martha (n=martha@NaTr-net1.ser.netvision.net.il) |
| 17:47.07 | salviadud | interesting... |
| 17:47.17 | tletourneau | D-Fender : I'll give that a try, thanks. |
| 17:47.23 | salviadud | think i can set it up via IAX? |
| 17:47.25 | *** part/#asterisk IRC_User (n=anonymou@198.60.73.230) |
| 17:47.30 | salviadud | thats how i have fwd setup right now... |
| 17:47.47 | trixter | the trxtel thing? iax directions are on that page |
| 17:47.59 | salviadud | right on... |
| 17:48.02 | trixter | dont even have to register with anyone |
| 17:48.10 | mzo | so if i use trx i can call toll free numbers free? :P |
| 17:48.41 | mzo | useful :) |
| 17:48.48 | trixter | you can potentially call tollfrees for a negative amount with trx :) |
| 17:48.58 | tletourneau | D-Fender : Are there reserved extension numbers in *? |
| 17:49.19 | trixter | tletourneau: no |
| 17:49.32 | tletourneau | Thanks. |
| 17:50.01 | salviadud | wow. im gonna set it up right now |
| 17:50.03 | trixter | some apps like voicemail will look for extension 0 and if you defined it clal that user.. but its not reserved for anything |
| 17:50.17 | salviadud | you see |
| 17:50.20 | salviadud | i am from mexico!!! |
| 17:50.23 | salviadud | muahahaha! |
| 17:50.23 | saftsack | trixter, do you know if its possible to jump automatically while releasing the handset of a normal analog telephone in any extension without using immediate=yes? |
| 17:50.37 | salviadud | those "toll free numbers" |
| 17:50.40 | salviadud | are not free over here... |
| 17:50.43 | [TK]D-Fender | tletourneau : not really. |
| 17:50.49 | salviadud | thanx to asterisk... now they are |
| 17:50.52 | salviadud | viva la revolucion! |
| 17:51.12 | [TK]D-Fender | saftsack : No, thats exactly the POINT of "immediate=yes" |
| 17:51.42 | saftsack | because my block dialing doesnt work with this one :( |
| 17:52.34 | saftsack | is there any reason why? |
| 17:53.51 | tletourneau | So if I set the context to from-pstn the system should treat it like any other incoming call, right? |
| 17:54.09 | saftsack | right |
| 17:54.19 | saftsack | [TK]D-Fender, do you know block dialing? |
| 17:55.16 | znoG | if I want 2 extensions to forward to each other on BUSY or NO ANSWER, but avoid endless loops of forwarding to each other... what's a good way of going about it? |
| 17:55.37 | [TK]D-Fender | saftsack : term doesn't ring a bell with me... |
| 17:55.58 | saftsack | this is dialing and THEN releasing the handset. it is an isdn feature. |
| 17:56.09 | salviadud | hey, this thing is all right |
| 17:56.23 | salviadud | i had to create a really crazy setup for toll free numbers with FWD |
| 17:56.25 | [TK]D-Fender | znog : if you are using a STDEXTEN style macro, pass it an incrementing counter for the # of hops and assign a cut-off |
| 17:56.41 | salviadud | this i like... |
| 17:56.44 | *** part/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:56.44 | [TK]D-Fender | saftsack : Sorry... thats what ISDN is DIGITAL. |
| 17:56.53 | [TK]D-Fender | Analog = dumber than dirt |
| 17:56.59 | trixter | isnt it much cleaner? |
| 17:57.18 | *** join/#asterisk exstatica (i=exstatic@redline.mednor.net) |
| 17:58.10 | saftsack | yes but my isdn telephone is handled in zap two that was it why i wrote analog |
| 17:58.17 | saftsack | not two ... too |
| 17:58.44 | *** join/#asterisk Assid (n=assid@203.115.64.13) |
| 17:58.54 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:00.52 | saftsack | [TK]D-Fender, do you have any ideas? |
| 18:02.18 | Assid | hey [TK]D-Fender: in the 501.. how do you specify a ip specifically? |
| 18:02.39 | *** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com) |
| 18:03.23 | GoRK | anyone know if there is an accessory part or something for a dell poweredge 2850 that gives me a standard 4 pin power connector for a TDM400P/TDM2400P card? |
| 18:03.58 | znoG | [TK]D-Fender: yeah not sure how i'd do it.. |
| 18:04.02 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 18:04.07 | trixter | you mean like the 5v and 12v power connectors hard drives use? |
| 18:04.13 | GoRK | trixter: yes |
| 18:04.22 | znoG | [TK]D-Fender: ie. how to check if, in the chain of call forwards, a number has already been dialed. |
| 18:05.02 | GoRK | trixter: on the dell the drives are all SCA scsi and the motherboard is connected to the PSU with a big modular connector.. there aren't really any standard power cables in there that i can find |
| 18:06.01 | tronix | GoRK: not sure, but you might have easier searches if you also include 'molex' (the brand name of that connector type) |
| 18:06.01 | mzo | trixter, how can you call them toll-frees for a negative amount? :P |
| 18:06.17 | GoRK | tronix: ah yes good idea |
| 18:06.37 | trixter | you get paid if you do volume |
| 18:07.32 | mzo | haha, so one call a day for me isn't volume ;) |
| 18:08.02 | trixter | in that case its just free |
| 18:09.21 | [TK]D-Fender | Assid : Give it a fixed IP? Either in the web interface (not suggested), or on the phone directly at boot (better), or by your DHCP server (best). |
| 18:10.20 | Assid | nah. dont wanna do from dhcp server.. wanna do it on boot.. |
| 18:10.27 | Assid | cant find what setting to edit tho |
| 18:10.45 | AndyCap | GoRK: http://delltalk.us.dell.com/supportforums/board/message?board.id=pes_cdrom&message.id=792 do you have the connector at the molex adapter for the 2800? (5th post from top) |
| 18:11.51 | Assid | nat.ip ? |
| 18:12.19 | *** join/#asterisk ToTo (n=ToTo@host2-161.pool870.interbusiness.it) |
| 18:13.09 | AndyCap | GoRK: or if you're kinky the pinout is on page 3 |
| 18:13.09 | [TK]D-Fender | Assid : You'd have to do it from the LCD on startup |
| 18:13.31 | [TK]D-Fender | Assid : its not in a provisioning file. |
| 18:13.32 | Assid | LCD ? cant provision? |
| 18:13.34 | Assid | damn |
| 18:13.36 | [TK]D-Fender | not for IP. |
| 18:14.03 | GoRK | andycap: awesome thanks |
| 18:14.06 | Assid | is it better to have a fixed ip for the phones? |
| 18:14.24 | Assid | coz apparently all of a sudden i see some issues of zombies and stuff |
| 18:14.33 | AndyCap | GoRK: dunno if the psu in a rackmount has the connector though. |
| 18:14.42 | GoRK | it has it |
| 18:14.43 | Assid | sometimes.. 1 way audio |
| 18:14.54 | *** join/#asterisk E0x (n=moya@pri-133-b32.codetel.net.do) |
| 18:14.55 | E0x | hello |
| 18:15.06 | mzo | asterisk at home must have not updated to 1.2.4 |
| 18:15.08 | GoRK | andycaP: or rather it has something similar.. i will call dell probably |
| 18:15.11 | mzo | even though this is 2.5 |
| 18:15.29 | [TK]D-Fender | Assid : Why bother with fixed IP? |
| 18:15.49 | kippi | any iaxy experts around? |
| 18:16.49 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 18:16.58 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 18:16.59 | Assid | these guys are compling of 1 way audio.. and sometimes.. i see zombies |
| 18:17.59 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 18:18.39 | shido6 | kippi, whats wrong? |
| 18:18.42 | bcnl | so am I able to do the following |
| 18:19.16 | bcnl | Set(STUB(sub)=foo) |
| 18:19.25 | bcnl | Set(STUB(sub2)=bar) |
| 18:19.26 | bcnl | ? |
| 18:20.02 | kippi | shido6: I am getting this error Feb 21 18:19:10 NOTICE[4161]: chan_iax2.c:3918 register_verify: No registration for peer '1001' (from 10.6.10.149) |
| 18:22.16 | *** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net) |
| 18:22.24 | trixter | Set(STUB(toe)=owwww) |
| 18:23.06 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 18:23.21 | asteriskmonkey | kippi: do you have your iax.conf for that devic set right for that device? |
| 18:23.30 | asteriskmonkey | how did you provision it? |
| 18:23.35 | asteriskmonkey | are you getting a blue light? |
| 18:23.51 | kippi | asteriskmonkey: no blue light |
| 18:24.00 | asteriskmonkey | then its not connected or provision right |
| 18:24.31 | kippi | so its the config on the iaxy box ? |
| 18:24.49 | asteriskmonkey | most likley :D |
| 18:24.49 | asteriskmonkey | how did you provision it? |
| 18:24.53 | shido6 | kippi, so whats in iax.conf? |
| 18:24.58 | shido6 | and how did you provision the iaxy? |
| 18:25.02 | shido6 | pastebin.ca kippi |
| 18:25.02 | *** join/#asterisk simulated (i=user@adsl-070-155-044-220.sip.bct.bellsouth.net) |
| 18:25.05 | kippi | ok |
| 18:25.29 | *** join/#asterisk stoffell (n=stoffell@d51A5826C.access.telenet.be) |
| 18:26.44 | *** part/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 18:26.51 | kippi | http://pastebin.ca/42678 |
| 18:26.58 | E0x | i have some question , exist a jabber server ( and client server ) that i can connect together with asterisk server and using jabber for registre and list the users and using the client ( with some form of voip support ) can make voice chat betewen users via asterisk server |
| 18:27.14 | gbodemantv | hey all |
| 18:27.20 | gbodemantv | quick question |
| 18:27.36 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 18:27.38 | Katty | hi lads. |
| 18:27.49 | gbodemantv | I want to convert a mp3 to gsm but it seems to sound awful when I convert |
| 18:27.52 | gbodemantv | what do you all use |
| 18:29.32 | shido6 | iax2 reload, kippi ? |
| 18:30.16 | kippi | No such command 'iax2 reload' (type 'help' for help) |
| 18:30.51 | dpolitech | you can just do a regular 'reload' |
| 18:31.00 | dpolitech | it will re-read iax.conf |
| 18:31.25 | shido6 | err... you cant iax2 reload? |
| 18:31.41 | shido6 | show modules and look for IAX2 |
| 18:32.42 | kippi | its not there |
| 18:33.21 | znoG | is G723 a licensed codec? |
| 18:33.30 | [TK]D-Fender | znoG: yup |
| 18:34.09 | shido6 | time to find chan_iax2.so and load it then |
| 18:34.16 | shido6 | you are looking for Inter Asterisk eXchange (Ver 2) , rather |
| 18:34.19 | Dr-Linux | [TK]D-Fender: hi |
| 18:35.08 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:35.27 | jpablo | hey people, anyone knows a problem in snom 360 where the phone just boots to the message: ready...serial 2.0 ? |
| 18:37.05 | kippi | shido6: so i need to install chan_iax2 ? |
| 18:37.17 | lazzarello | jpablo, no, but I got a batch of 11 snom 360s and ended up returning 4 of them for stupid bugs like that. |
| 18:38.00 | znoG | ... and has anyone tried: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ?? |
| 18:38.12 | znoG | (intel's g723/g729 code converted to Asterisk) |
| 18:38.40 | austinnichols101 | gbodemantv: use sox |
| 18:38.50 | dpolitech | make sure you read the bottom of that page |
| 18:38.52 | dpolitech | the legal stuff |
| 18:39.53 | asteriskmonkey | is there a newer patch for altering zap rx tx on the fly? |
| 18:40.04 | znoG | dpolitech: yep, i'm not looking at making any products with that code, and no patents exist in my country |
| 18:40.13 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:40.15 | znoG | dpolitech: for G729 and G723 |
| 18:40.25 | dpolitech | cool |
| 18:40.28 | dpolitech | I've tried it |
| 18:40.30 | dpolitech | worked for me |
| 18:40.35 | dpolitech | this was awhile ago though |
| 18:40.37 | dpolitech | pre *1.2 |
| 18:40.55 | znoG | ah, i see |
| 18:41.05 | znoG | so if my sipura device supports G729, it should work with this codec? |
| 18:41.07 | Juggie | it still works |
| 18:41.11 | asteriskmonkey | looking for a 1.2x veriosn of zap set tx or zap set rx command |
| 18:41.11 | dpolitech | yes |
| 18:41.14 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 18:41.16 | Juggie | though i would recommend purchasing it legit |
| 18:41.19 | Juggie | even if its not supported |
| 18:41.21 | znoG | worth a try over GSM |
| 18:41.25 | Juggie | as it supports asterisk development |
| 18:41.29 | dpolitech | right |
| 18:41.35 | dpolitech | buy from digium |
| 18:41.36 | znoG | true, true |
| 18:41.38 | Juggie | er, even if its not illegal i mean. |
| 18:41.39 | dpolitech | and you get support as well |
| 18:41.44 | dpolitech | pretty decent support too |
| 18:42.13 | dpolitech | though I've haven't even needed support with digi's g729 implementation |
| 18:43.06 | saftsack | <PROTECTED> |
| 18:43.08 | saftsack | what is that? |
| 18:44.57 | *** join/#asterisk ursuspacificus (n=ursuspac@wsip-24-249-27-197.ri.ri.cox.net) |
| 18:46.25 | kippi | where can i get chan_iax2 from? |
| 18:46.48 | shido6 | it comes with the source, Luke |
| 18:47.12 | chkbsd | is capi/isdn in the 1.2.4 release or in the next? |
| 18:47.40 | gbodemantv | tried to use GX Transcoder to convert but does not seem to convert to gsm |
| 18:47.50 | gbodemantv | even though it says it does |
| 18:47.51 | znoG | dpolitech: you still need to register with Intel to use the open source stuff I pasted before, eh? |
| 18:48.09 | gbodemantv | anybody have any ideas |
| 18:51.44 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 18:52.13 | znoG | ok i don't qualify for a non-commercial license :) |
| 18:53.31 | kippi | I am now getting this error when trying to load with iax2 |
| 18:53.32 | kippi | [chan_iax2.so]Feb 21 18:52:23 WARNING[5599]: loader.c:258 ast_load_resource: /u sr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature |
| 18:53.32 | kippi | Feb 21 18:52:23 WARNING[5599]: loader.c:391 load_modules: Loading module chan_ia x2.so failed! |
| 18:55.31 | remiss | cool! |
| 18:58.12 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 19:00.58 | wasim | anybody have a dual opteron 246 handy? need to run a couple of g729 tests |
| 19:01.07 | shido6 | heh |
| 19:01.15 | mzo | how do you get a non-commercial license? |
| 19:01.35 | salviadud | could someone please direct me to a good zaptel.conf example? |
| 19:01.52 | wasim | salviadud: make samples |
| 19:02.01 | shido6 | locate zaptel.conf.sample , salviadud |
| 19:02.06 | wasim | err, the one in the src |
| 19:02.12 | shido6 | ahh - careful with that make samples :) |
| 19:02.17 | salviadud | i meant a good one... |
| 19:02.25 | shido6 | that does what, salviadud ? |
| 19:02.52 | salviadud | i know what make samples does. im looking for something more specific |
| 19:02.59 | salviadud | for one of thos generic intel cards |
| 19:03.13 | salviadud | tp100 i think |
| 19:03.22 | jbalcomb | Lots of problems with the GXP-2000 blank screen/no ip/locking up using firmware 1.0.2.8. Anyone have any fixes? |
| 19:03.28 | Seldon1975 | Q: what's red and invisible? |
| 19:03.30 | kippi | anyone got any ideas why iax is failing to load? |
| 19:03.34 | Seldon1975 | A: no tomatoes |
| 19:03.55 | stoffell | jbalcomb, wait untill next firmware release.. depends on what hardware revision you have.. |
| 19:04.19 | stoffell | jbalcomb, we have 7 phones with 0 problems, 1 phone: blank screen/lockup/.. so, be patient.. |
| 19:04.46 | mzo | hah, i'm still trying to figure out how i downgraded my asterisk |
| 19:04.49 | mzo | thta took major skillz |
| 19:05.52 | *** join/#asterisk lalito (n=erg@201.102.4.195) |
| 19:07.41 | [Airwolf] | I just setup a new Asterisk server with a config I use everytime, but the problem is I now get these messages when trying to dial with my softphone |
| 19:07.42 | salviadud | well, did you install it like a package? |
| 19:07.44 | [Airwolf] | Feb 21 21:05:46 NOTICE[2242]: chan_iax2.c:6782 socket_read: Rejected connect attempt from 83.98.242.242, who was trying to reach '781@' |
| 19:07.50 | salviadud | or just make install? |
| 19:07.51 | jbalcomb | stoffell: ok. We have about 100 of the GXP-2000's but I'm not sure how many actually have the problem. |
| 19:07.51 | [Airwolf] | And probably I'm overlooking somthing |
| 19:07.56 | [Airwolf] | But I have no idea |
| 19:08.18 | jbalcomb | stoffell: How do you know the hardware revision? I saw some posts regarding the MAC address. Is that connected? |
| 19:08.21 | stoffell | jbalcomb, check the voip wiki regeularly for updates on the gxp-2000 page.. |
| 19:08.41 | stoffell | jbalcomb, yes, the mac addr gives an idea... |
| 19:08.45 | kippi | [Airwolf] how did you get iax working? |
| 19:08.46 | jbalcomb | stoffell: yeah, I have it bookmarked on my 'Links' toolbar. |
| 19:09.05 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:09.13 | generalhan | whats up everyone ! |
| 19:09.15 | [Airwolf] | kippi, I just made the config file. What do you mean exactly ? |
| 19:09.26 | jbalcomb | stoffell: ok. thanks. I wonder if there is a range of hardware versions that have the problem? |
| 19:09.40 | kippi | i am trying to get it to work, but if i add it to the modules I get error |
| 19:09.58 | [Airwolf] | What error do you get ? |
| 19:10.00 | stoffell | jbalcomb, i'm afraid the first bunch are affected, .. |
| 19:10.47 | generalhan | i have a quick question ... i just switched my VoIP lines out for a PRI T-1. Now a lot of my Cisco 7960s have a MEAN echo. and its a good 1-2 second delay, if i read off a phone number to some one i can say 3 digits, then i hear myself say those digits back to me in the phone. any ideas why this is happening or what i might be able to do to stop this ? |
| 19:10.48 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 19:10.52 | jbalcomb | stoffell: ok, I'll check my macs against those listed on the wiki. thanks. |
| 19:10.56 | GoRK | BTW if anyone was following my saga I found the way to get a digium card with a molex connector working in a dell poweredge 2850 .. there is a part from dell that gives you a regular molex: H2188 it is similar to the part described in the link AndyCap posted |
| 19:11.13 | GoRK | you will most likely need an extension cable as well |
| 19:11.28 | jbalcomb | stoffell: Seems 00.0B.82.03 vs. 00.0B.82.04 |
| 19:11.50 | robin_sz | oh great! at last some GCP2000 firmare to fix the broken veriosn I installed two weeks ago! |
| 19:11.57 | robin_sz | GXP |
| 19:12.57 | robin_sz | and they are giving away free money at the local bank. |
| 19:13.07 | stoffell | hehe |
| 19:13.24 | stoffell | jbalcomb, 000b8204 doesn't seem to have the prob here |
| 19:14.10 | stoffell | but.. 000b8203 does.. |
| 19:14.46 | *** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca) |
| 19:14.58 | dlublink | Hello |
| 19:15.14 | *** join/#asterisk lorinc (n=ang@caracas-1617.adsl.interware.hu) |
| 19:15.34 | jbalcomb | stoffell: That is what I'm seeing here as well. Now I just need to decide whether we realistically drop the version number just for phones that have the old MAC. :/ |
| 19:16.01 | *** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca) |
| 19:16.02 | dlublink | oop |
| 19:16.04 | stoffell | drop? i'm afraid there's no downgrade |
| 19:16.04 | dlublink | oops |
| 19:16.05 | dlublink | wrong button |
| 19:16.42 | robin_sz | stoffell: grandstream by any chance? |
| 19:16.52 | kippi | [Airwolf] can I pm you? |
| 19:17.16 | stoffell | robin_sz, hehe ;) |
| 19:17.38 | robin_sz | stoffell: I REALLY wish they would release SOMETHING to make this phone useable again :( |
| 19:17.45 | robin_sz | ANYTHING |
| 19:18.04 | jbalcomb | stoffell: ah, its all 1.0.2.X. I see that now. bummer. Guess I just need to make sure the old revision phones are physically close to the heldesk dept. =) |
| 19:18.05 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 19:18.10 | frenzy | hello all |
| 19:18.15 | stoffell | yeah robin_sz, it's a real pain, not good of them :( |
| 19:18.30 | [Airwolf] | kippi, now you can |
| 19:18.33 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:18.48 | jbalcomb | robin_sz: ditto. rediculous to not be able to rollback the version. |
| 19:18.51 | frenzy | does anyone have an idea of an open source callback application/billing for asterisk ? |
| 19:18.57 | robin_sz | stoffell: im thinking of buying some Snoms to replace the broken ones as its gone on too long now ... |
| 19:19.05 | [Airwolf] | Does anyone has seen this error before: Feb 21 21:05:46 NOTICE[2242]: chan_iax2.c:6782 socket_read: Rejected connect attempt from 83.98.242.242, who was trying to reach '781@' |
| 19:19.05 | [Airwolf] | <PROTECTED> |
| 19:19.11 | GoRK | polycoms rule! except for those things about them that kinda suck |
| 19:19.29 | stoffell | robin_sz, a "thomson st2030" looks very good, but it doesn't have any working features :( |
| 19:19.30 | robin_sz | Snom 360s are OK |
| 19:19.49 | salviadud | i get this when i run ztcfg |
| 19:19.52 | salviadud | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 19:19.54 | GoRK | 601's have poe and the rest have adaptor cables -- the poe kits purchased with the phone are about the same price as the normal phones |
| 19:19.54 | robin_sz | i have ~25 of those and 25 320s |
| 19:20.06 | salviadud | what does it mean?.... |
| 19:20.20 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 19:20.24 | jbalcomb | robin_sz: we are purchasing SNOMs as replacements as well. Mostly to deal with call quality really but the trobule with new firmware increased our urgency. |
| 19:20.38 | robin_sz | salviadud: it means that the device at address 6 youasked for, doesnt seem to exist |
| 19:20.50 | salviadud | lies!!! |
| 19:20.51 | stoffell | snom is 4x the price.. polycom is not 'too' expensive it seems. |
| 19:20.54 | salviadud | haha |
| 19:20.57 | salviadud | well, not exactly |
| 19:21.06 | salviadud | damn... this zaptel is giving me headache |
| 19:21.22 | salviadud | could a zaptel.conf file be as small as 3 lines? |
| 19:21.37 | stoffell | i think it could salviadud, what 'card' are you using? |
| 19:21.51 | salviadud | its a tp100 |
| 19:21.52 | robin_sz | jbalcomb: the GXP2000 has not been a success for us, we suffer from the "on hook" problem and now the firmware ... |
| 19:21.54 | salviadud | intel generic |
| 19:22.13 | robin_sz | salviadud: by a digium card you tightwad |
| 19:22.17 | robin_sz | buy |
| 19:22.30 | salviadud | me mexican, digium not nearby friend amigo man |
| 19:22.57 | salviadud | although, i could order it online... |
| 19:22.58 | jaiger | robin, buying the "digium" version of the card won't solve a mis-configuration |
| 19:22.59 | stoffell | tp100, sorry, don't know that |
| 19:23.10 | robin_sz | jaiger: true |
| 19:23.18 | salviadud | well, I have 2 cards |
| 19:23.20 | salviadud | both tp100 |
| 19:23.23 | salviadud | could that be the problem? |
| 19:23.25 | mzo | wish they made digium xp100s still :P |
| 19:23.36 | jaiger | the "digium card" is in fact an intel card with a heatsink glued on the chip |
| 19:23.58 | mzo | shh, don't ruin my dream :P |
| 19:24.08 | squinky86 | I followed the information on the wiki, but I can't seem to call out through net2phone with asterisk. I keep getting "Forbidden - wrong password on authentication for INVITE..." though I was calling through them just fine yesterday. Any pointers for things to look at? |
| 19:24.11 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 19:24.15 | salviadud | haha, glued |
| 19:24.18 | tletourneau | Anyone know anything about Vegastream dialing plans? I've googled it and haven't found much. |
| 19:24.22 | salviadud | thats creative.. |
| 19:24.30 | mzo | im trying to find a dialing plan for finland. :P |
| 19:24.47 | salviadud | have you seen conan o brian? |
| 19:24.55 | salviadud | he looks just like the president of finland |
| 19:25.01 | salviadud | hilarious |
| 19:25.37 | mzo | i know. |
| 19:25.48 | mzo | i make a lot of calls to finlandia. I'm going broke :P |
| 19:26.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:26.39 | Egonis | finlandia, the beverage? |
| 19:26.43 | mzo | the country ;) |
| 19:26.52 | mzo | although i do make calls while drunk on the beverage |
| 19:26.57 | Egonis | ;) |
| 19:26.59 | mzo | this is how i ended up going from 1.2.4 to 1.2.1 |
| 19:27.09 | mzo | and somehow my dialplan only dials a 900 porn number now |
| 19:27.19 | Egonis | sweet! |
| 19:27.25 | mzo | yeah, i wish i knew how i did it |
| 19:27.42 | Egonis | I should do that, to confuse the hell out of some cube monkeys |
| 19:28.22 | *** join/#asterisk XnoN (n=xnon@200.8.30.11) |
| 19:28.29 | salviadud | hey, i got my tp100 configured |
| 19:28.38 | *** join/#asterisk jontow (i=jontow@hijacked.us) |
| 19:28.39 | salviadud | did you guys know you need 2 modules to load it |
| 19:28.41 | XnoN | have asterisk.org a channel in spanish please? |
| 19:28.57 | salviadud | i can speak spanish like a gringo |
| 19:28.59 | XnoN | i dont know english so much! |
| 19:29.10 | salviadud | amigo, como esta? |
| 19:29.16 | stoffell | XnoN, your english looks okay |
| 19:29.16 | salviadud | me help a usted |
| 19:29.16 | XnoN | muy bien gracias y tu? |
| 19:29.18 | XnoN | jejeje |
| 19:29.34 | salviadud | como van the freaking calls man |
| 19:29.40 | XnoN | yes but isnt so good, I know just a little bit |
| 19:29.54 | salviadud | mira guey, si quieres hablar en español, apenas en privado |
| 19:30.08 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:30.14 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 19:30.20 | salviadud | we speak english, only, exlusiveley, lots of rednecks using asterisk ya know |
| 19:30.38 | XnoN | i need to know if with asterisk is posible create a sms server to for text service only! |
| 19:30.52 | mzo | bleh i gotta ieinstall off the cdd again, the insstall.gz only goes to 1.2.1 so it's a downgrade |
| 19:30.54 | stoffell | XnoN, yes it is, app_sms |
| 19:31.02 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 19:31.33 | *** join/#asterisk Fiskfan (n=erikdj2@h35n3fls32o286.telia.com) |
| 19:31.48 | greendisease | why do i keep getting "ouch, part reset, quickly restoring reality (0)"? |
| 19:31.50 | RoyK | morning, morons |
| 19:31.57 | [av]bani | o_o |
| 19:32.02 | remiss | o_O |
| 19:32.42 | RoyK | :D |
| 19:32.54 | hhoffman | that's mormons |
| 19:32.57 | remiss | i want to do something fun |
| 19:33.43 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 19:33.48 | badboyz | is it possible to lengthen the amount of time that a person is allowed to leave a voicemail message? |
| 19:33.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:34.00 | jpablo | hey people, a snom360 just burned on me. I have to buy another multi line ip phone, recomendations ? |
| 19:34.14 | RoyK | badboyz: SMOP |
| 19:34.43 | remiss | neutral milk hotel? |
| 19:34.50 | badboyz | RoyK: in the dialplan? |
| 19:35.04 | RoyK | nope |
| 19:35.07 | badboyz | where @ ? |
| 19:35.12 | mikefoo | badboyz: hey sup.. |
| 19:35.12 | RoyK | RTFS |
| 19:35.47 | badboyz | RoyK: if you idle on irc, to shout acronyms for assistance, then i think you are more well served leaving this channel |
| 19:35.49 | jaiger | when is the weekly dev conf call held? |
| 19:36.16 | badboyz | mikefoo: hey there :) get any new accounts setup? |
| 19:36.21 | remiss | i think it was a funny acronym |
| 19:36.24 | RoyK | badboyz: those acronyms mean something |
| 19:36.37 | RoyK | ~rtfs |
| 19:36.38 | jbot | rumour has it, rtfs is probably read the f*cking source... |
| 19:36.39 | badboyz | i know exactly what they mean |
| 19:36.55 | frenzy | what callback systems are available for asterisk ? |
| 19:37.15 | mikefoo | badboyz: still developing the frontend :) |
| 19:37.24 | RoyK | badboyz: but extending the amount of time a given person has for a voicemail message can't be done unless you just set it globally |
| 19:37.28 | *** part/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 19:37.39 | mikefoo | badboyz: recommend anyone for dids? |
| 19:37.42 | RoyK | ~smop |
| 19:37.43 | jbot | it has been said that smop is just a simple matter of programming or encrypted floor cleaning tool |
| 19:38.06 | badboyz | mikefoo: a few people in here resell for a small fee |
| 19:38.18 | *** join/#asterisk littlejohn (n=little@host191-75.pool871.interbusiness.it) |
| 19:38.21 | badboyz | ~stfu |
| 19:38.22 | jbot | it has been said that stfu is Shut the F*** Up!, or http://www.linuks.mine.nu/stfu-noob.jpg |
| 19:38.25 | badboyz | know that one royk ? |
| 19:39.03 | RoyK | badboyz: did you mean extending the time for a certain user? callerid? |
| 19:39.38 | *** join/#asterisk pengyong (n=lala@218.19.188.13) |
| 19:39.48 | badboyz | globally is fine, and its in the vm_general.inc |
| 19:43.22 | [Airwolf] | Has anyone ever seen this when calling other iax clients: chan_iax2.c: Rejected connect attempt from 83.98.242.242, who was trying to reach '781@' |
| 19:43.36 | [Airwolf] | I'm getting this on a new installation, but I don't know why |
| 19:43.50 | *** join/#asterisk j4m3s (n=debbie@gateway.digium.com) |
| 19:44.52 | *** join/#asterisk shidan (i=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 19:49.39 | _Paulo_ | How can I retrieve the dialed extension when I'm at the "i"nvalid extension? |
| 19:50.53 | Nivex | _Paulo_: ${EXTEN} should still work at that point |
| 19:53.25 | iCEBrkr | Nivex: I don't think so.. ${EXTEN} will be 'i' |
| 19:54.18 | *** join/#asterisk backblue (n=moo@87.196.0.177) |
| 19:54.27 | znoG | _Paulo_: you could do a Set(ext=${EXTEN}) when within a dialed extension, then look at that value from the i extension |
| 19:54.30 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 19:55.42 | exonic | That information is in the docs |
| 19:55.51 | exonic | http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 19:55.54 | _Paulo_ | Nivex, I got: -- Executing Set("Zap/66-1", "DIALEDNUMBER=i") in new stack |
| 19:55.56 | exonic | _Paulo_, check that URL |
| 19:57.00 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 19:57.09 | _Paulo_ | exonic, thanks, its the ${INVALID_EXTEN} |
| 19:58.20 | XnoN | alguien presente habla español? |
| 19:58.27 | *** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com) |
| 20:00.30 | Falle | are there logs from this channel on the web somewhere? |
| 20:00.47 | kippi | anyone got a how to for iax2? |
| 20:04.30 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 20:05.29 | hensema | hi |
| 20:05.37 | hensema | I'm trying to get MoH working |
| 20:05.49 | hensema | but I'm just getting noise, no music :/ |
| 20:05.59 | hensema | sounds a bit like a propeller |
| 20:06.05 | hensema | trtrtrtrtrtrtrtrtrtrtrtr |
| 20:06.20 | *** part/#asterisk E0x (n=moya@pri-133-b32.codetel.net.do) |
| 20:06.26 | hensema | any clue on what could be wrong here? |
| 20:06.32 | eKo1 | stop playing thrash metal as your MoH |
| 20:06.36 | hensema | haha |
| 20:06.51 | Nivex | hensema: check your frame size. I had to change from 30 to 20 ms on my ATA to get that to stop |
| 20:07.36 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 20:07.40 | iCEBrkr | hensema: If you're testing with your cellphone and it's GSM, it's gonna do that :P |
| 20:07.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:11.11 | hensema | right, same trouble on sip -> asterisk |
| 20:12.33 | *** join/#asterisk dragonheart (n=dragonhe@gentoo/developer/dragonheart) |
| 20:12.40 | *** join/#asterisk bkw_ (i=Jon@38.112.144.14) |
| 20:12.44 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 20:12.50 | GerbilWrk | anyone know what routers besides the wrt54g can run asterisk on them? |
| 20:13.01 | hensema | Nivex: how do you change the frame size on SIP? |
| 20:13.22 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:13.27 | mog_work | any router that runs linux GerbilWrk |
| 20:13.48 | GerbilWrk | ok, know of any other routers that run linux? |
| 20:14.18 | stoffell | GerbilWrk, check openwrt.org, there are some examples in the hardware list |
| 20:15.53 | Nivex | hensema: I did it on my ATA. I don't know how to change it in Asterisk. |
| 20:16.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:16.41 | SibRw0rk | hensema: ATA |
| 20:16.53 | SibRw0rk | is a device that hooks up regular phones to ethernet for voip |
| 20:17.17 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 20:17.22 | hensema | ah |
| 20:17.37 | hensema | well, that wouldn't be the problem then ;-) |
| 20:21.37 | SibRw0rk | hensema: what's the problem? |
| 20:22.40 | Nivex | hensema: is this happening on a Zap channel? |
| 20:22.51 | hensema | no, on SIP |
| 20:22.56 | hensema | I haven't tested ZAP yet |
| 20:22.57 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 20:23.06 | Nivex | hensema: what are you using as a SIP agent? |
| 20:23.11 | hensema | but, my collegue says on his SIP phone the music sounds fine |
| 20:23.16 | hensema | kphone |
| 20:24.10 | Nivex | hensema: under Audio preferences, what is Size of Payload set to? |
| 20:24.25 | SibRw0rk | hensema: what's the trouble with MOH? |
| 20:25.26 | *** join/#asterisk loick (n=loick@APuteaux-151-1-87-72.w86-205.abo.wanadoo.fr) |
| 20:26.10 | hensema | SibRw0rk: MoH sounds like some propeller plane is taking off |
| 20:26.46 | SibRw0rk | when you call into it, or when someone is put on hold? |
| 20:27.04 | hensema | when I call into it, running the MusicOnHold application |
| 20:27.26 | SibRw0rk | b/c i know my MOH doesn't sound right when i call into it, but when someone calls me and i put them on hold it sounds fine |
| 20:27.36 | SibRw0rk | hensema: what format is your music? |
| 20:27.41 | SibRw0rk | mp3? wav? raw? |
| 20:27.51 | hensema | just the standard mp3s shipped with asterisk |
| 20:27.58 | SibRw0rk | hensema: try converting them to raw |
| 20:28.19 | SibRw0rk | there's a walkthrough somewhere on http://www.voip-info.org/wiki-Asterisk |
| 20:28.37 | hensema | yeah, I saw it |
| 20:28.43 | hensema | I'll try, thanks |
| 20:28.45 | SibRw0rk | welceom |
| 20:28.48 | SibRw0rk | welcome |
| 20:30.14 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:30.50 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:33.52 | *** join/#asterisk lalito (n=erg@201.102.4.195) |
| 20:34.06 | simulated | dancing to* |
| 20:38.54 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 20:39.06 | _Thor | hello everyone |
| 20:39.15 | Abydos313 | hello |
| 20:39.24 | Abydos313 | just got here myself |
| 20:39.37 | Meaty | :o |
| 20:40.28 | _Thor | question: what's the dial with codec licenses g723, do I have to buy them from digium??, but they don't sell them |
| 20:40.49 | Abydos313 | i thought you only buy the 729 lic |
| 20:40.59 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 20:41.06 | _Thor | right, but I don't have g723 capabilities in my box |
| 20:41.43 | _Thor | do you know how to enable g723?? |
| 20:41.59 | Goral | ~docs |
| 20:42.00 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:42.13 | jaiger | isn't g723 the same as ulaw/alaw? |
| 20:42.21 | jaiger | ~g723 |
| 20:42.25 | Abydos313 | http://www.voip-info.org/wiki-Asterisk+codecs |
| 20:42.32 | _Thor | is it? |
| 20:42.48 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 20:42.52 | afrosheen | hey hey hey |
| 20:43.01 | jaiger | nope, 711 is |
| 20:43.13 | afrosheen | has anyone had problems with sip trunk echo on 1.24? |
| 20:43.58 | Skid | I know the person who wrote g729 |
| 20:44.06 | _Thor | who did? |
| 20:44.06 | Skid | well, I met him at a party |
| 20:44.18 | Abydos313 | Skid have him give out some home lic's for us :)) |
| 20:44.33 | Skid | he was a bit of a pretentious person tbh |
| 20:44.38 | Skid | tried to sell me a phone number for 5,000 GBP |
| 20:44.51 | _Thor | you could have had free licenses for a long time man! |
| 20:44.56 | Abydos313 | Skid how much is that in real money? |
| 20:44.57 | Abydos313 | haha |
| 20:45.00 | Skid | USD? |
| 20:45.01 | Skid | 10k ? |
| 20:45.06 | Abydos313 | i was kidding |
| 20:45.09 | Skid | oh ;P |
| 20:46.24 | _Thor | do you know that what is transfered in a codec are actually formulas, that enable the reconstruction of a given sentence in the exact same pitch and tone of voice as they were originated? |
| 20:46.52 | afrosheen | yeah british pounds are extremely valuable |
| 20:47.19 | afrosheen | no idea why |
| 20:47.44 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 20:48.20 | Assid | is there a known issue for polycom 501 with audio volume just dropping and people not being able to hear? |
| 20:48.39 | Abydos313 | i'd like to tryout the 729 for my home use :)) anyone wanna tell me how i can do that w/o buy a lic? |
| 20:48.52 | remiss | Assid: sure it's not the packets dropping instead? |
| 20:49.48 | Assid | how do i check for dropped packets for sip |
| 20:49.57 | jaiger | _Thor, you're not making sense |
| 20:50.00 | *** join/#asterisk acehunky (n=chat_jok@59.184.26.186) |
| 20:50.01 | afrosheen | Assid: I've seen phones with the coiled cable to the handset not being plugged in 100% and weirdness resulting from that..more than once |
| 20:50.44 | acehunky | can any one help me with astbill channel ? |
| 20:51.01 | *** join/#asterisk bkw_ (n=bkw_@38.112.144.14) |
| 20:51.03 | remiss | Assid: only one side can hear the other after a minute or so? |
| 20:51.09 | _Paulo_ | Can I use Authenticate with an external database |
| 20:51.30 | eKo1 | _Paulo_: hack it and send us the patch |
| 20:51.47 | Assid | remiss: nah.. sometimes they cant hear.. and then all of a sudden they can |
| 20:52.09 | Assid | also.. im geting JIT - 4 / DEL 61 |
| 20:52.14 | remiss | Assid: okay.. dunno.. possible nat-problem... |
| 20:52.19 | _Paulo_ | eKo1, if nobody have done it, I will do... |
| 20:52.24 | Assid | in LOCAL in iax2 show netstats |
| 20:52.33 | Assid | IAX2/voipjet-4 1000 4 61 0 0 0 1 0 0 0 0 0 0 0 0 |
| 20:53.06 | *** part/#asterisk shidan (i=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 20:53.13 | _Paulo_ | eKo1, I was asking because the least thing I want is reinvent the whell |
| 20:53.21 | Assid | remiss: is there a way to check sip packet loss |
| 20:53.59 | _Paulo_ | s/whell/wheel/ |
| 20:54.01 | remiss | Assid: not really.. it's udp.. just make sure all packets are properly forwarded where they should go.. don't trust routers that "support" the sip-protocol for instance.. |
| 20:54.41 | remiss | e.g. forward all tcp/udp-ports to asterisk or make sure asterisk has a public ip |
| 20:54.44 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 20:54.55 | acehunky | any astbill developer out here ? |
| 20:55.27 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 20:55.31 | remiss | any suggestions on how i should tell a person that i'm bored? |
| 20:55.37 | *** join/#asterisk redondos (n=redondos@190.48.53.26) |
| 20:55.43 | Skid | sod off you boring git? :p |
| 20:55.58 | Assid | remiss: nah.. i have it on DMZ |
| 20:56.09 | Assid | so.. i dont use any sip protocol stuff |
| 20:56.17 | remiss | uh? |
| 20:56.22 | remiss | i thought you said it was sip.. |
| 20:56.45 | Assid | DMZ = demilitarized zone.. so its available |
| 20:56.51 | Assid | SIP = transport agent |
| 20:57.05 | redondos | What's your opinion about asterisk@home? It will be my first venture with asterisk and I'm wondering if that'll really make things simpler, or more complicated. I was thinking of using debian, or something debian-based instead. |
| 20:57.07 | Assid | local phones use sip.. carrier to providers is on iax |
| 20:57.22 | remiss | oh |
| 20:57.34 | remiss | normally it's the other way around :S |
| 20:57.39 | Assid | err.. nope |
| 20:57.40 | remiss | or sip on both |
| 20:57.43 | Assid | sip phones |
| 20:57.49 | Assid | hence local clients = sip |
| 20:57.59 | afrosheen | redondos: avoid debian, you'll run into trouble with zaptel and other things |
| 20:58.02 | Assid | people dont really use iax hard phones |
| 20:58.07 | Assid | mostly sip |
| 20:58.21 | jaiger | afrosheen, in what way? I use debian on all my * servers |
| 20:58.27 | redondos | afrosheen: What's your recommendation? Nothing debian-based? |
| 20:58.29 | remiss | i was thinking about softphones.. *missing ata for now* |
| 20:59.10 | _Paulo_ | redondos, its easy to instal kernel modules in debian |
| 20:59.25 | _Paulo_ | I think its easier than other distros. |
| 20:59.29 | *** join/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl) |
| 20:59.29 | redondos | afrosheen: How about Gentoo? That's what I use it for several machines and I'm very comfortable using it, but are the ebuilds for asterisk and related products ready for production? |
| 20:59.38 | Egonis | Although I have ALSA working, I get the message: snd_pcm_open failed: no such device in /var/log/asterisk/messages |
| 20:59.42 | Assid | remiss: most softphones are sip baseed |
| 20:59.47 | redondos | My problem is not distro-based, instead, I would like to know what distribution has asterisk packages more nicely packaged. |
| 21:00.12 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 21:00.13 | Assid | afrosheen: hes saying thats not hte problem |
| 21:00.26 | Egonis | redondos: Gentoo has decent packages, but most ppl compile from source manually |
| 21:00.58 | Egonis | redondos: I use Gentoo, and with the exception of Zaptel, asterisk builds nicely |
| 21:01.00 | jaiger | Redonos, the first recommendation is don't ask any distro-related questions |
| 21:01.23 | redondos | Egonis: Alright. So what do you think about asterisk@home? Will that help me or cripple me? |
| 21:01.34 | redondos | jaiger: What do you mean? Channel rules? |
| 21:02.04 | jaiger | redondos, no. just that you'll start a Holy War type discussion with no benefit |
| 21:02.29 | redondos | Oh, I know that. But I was wondering what distro was 'readier' for Asterisk :) |
| 21:02.30 | *** part/#asterisk XnoN (n=xnon@200.8.30.11) |
| 21:02.45 | afrosheen | we've had great success with Centos 4.x, I'll leave it at that |
| 21:02.48 | _Paulo_ | redondos, centos |
| 21:02.51 | afrosheen | however, we build from source |
| 21:03.26 | Egonis | redondos: No idea |
| 21:03.27 | redondos | I see. |
| 21:03.34 | Egonis | Can anyone help me with my alsa issue? |
| 21:03.53 | generalhan | Whats up everyone... i have a quick question ... i just switched my VoIP lines out for a PRI T-1. Now a lot of my Cisco 7960s have a MEAN echo. and its a good 1-2 second delay, if i read off a phone number to some one i can say 3 digits, then i hear myself say those digits back to me in the phone. any ideas why this is happening or what i might be able to do to stop this ? |
| 21:04.11 | *** join/#asterisk [Outcast] (n=bill@222-152-213-192.jetstream.xtra.co.nz) |
| 21:04.16 | redondos | Asterisk@Home is a CentOS distro, AFAIK. I will give it a try. Though, I am more used to admining with debian tools. |
| 21:04.52 | [Outcast] | redondos: yum is great, it works more wajig |
| 21:05.10 | [Outcast] | <PROTECTED> |
| 21:05.17 | redondos | Could be, but I'm just not used to it. I guess I will have to learn it, eventually. |
| 21:06.04 | afrosheen | there's not much to learn..you issue yum install blabla and the package is installed with dependencies..if you're going with Centos/Redhat you may want Dag's repository added as well |
| 21:06.17 | [Outcast] | hey has come where rxfax will compress the image. so the fax looks squished? |
| 21:06.32 | redondos | afrosheen: Thanks for the recommendation. |
| 21:06.40 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 21:07.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:07.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:08.06 | afrosheen | np |
| 21:08.43 | redondos | Now for an implementation question: Is it possible with Asterisk to have a different menu played back if the called ID shows a specified number? |
| 21:08.51 | shido6 | yep |
| 21:08.52 | redondos | s/called/caller/ |
| 21:09.02 | redondos | Awesome. |
| 21:09.08 | redondos | hey! lovely bot :)( |
| 21:09.23 | redondos | Does anyone know what the name of the script it uses for that is? And what sort of bot it is, as well. |
| 21:10.25 | *** join/#asterisk lalito (n=erg@201.102.4.195) |
| 21:11.16 | [TK]D-Fender | redondos : No special scripts its just some GotoIf's in your dialplan |
| 21:11.49 | afrosheen | and *@home makes it easy to route stuff like that I believe |
| 21:15.10 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 21:15.27 | redondos | Oh, really? I will definitely give *@home a try, then. |
| 21:16.19 | redondos | afrosheen: But where do you specify such things, is it in Asterisk configuration files or I'd have to write some sort of script to achieve it? |
| 21:16.36 | [TK]D-Fender | redondos : extensions.conf |
| 21:18.00 | Tamarisk | Hi all simple question for you I hope you can answer for me? |
| 21:18.07 | afrosheen | redondos: in AMP, you set an incoming route for it |
| 21:18.08 | *** join/#asterisk Zodiacal (i=1232321@bdsl.66.14.242.199.gte.net) |
| 21:19.17 | redondos | Nice. Now I'm probably thinking impossible here, but can Asterisk build a menu on-the-fly? As in, can I tell asterisk hey, if this number calls, present him a menu with these options: #1, #6 and #7. Just an example. |
| 21:19.18 | Zodiacal | anyone know how to remove the 5 second timeout when dialing numbers? i would rather have it wait until i have entered an ext or a full phone number, is that posible? |
| 21:19.34 | afrosheen | with the incoming route tables you can route to extensions, ring groups, queues, alternate IVRs, etc. |
| 21:19.54 | afrosheen | no you have to create your own IVRs and they're static |
| 21:20.39 | [TK]D-Fender | redondos : thats not "on the fly", thats just "under these circumstances do this instead". |
| 21:20.46 | redondos | I see. |
| 21:20.47 | *** join/#asterisk IRC_User (n=anonymou@198.60.73.230) |
| 21:20.54 | [TK]D-Fender | redondos : At which point that'd be "yes" |
| 21:21.23 | redondos | [TK]D-Fender: Yes, well, but it would be sort of on the fly because I'd like to have a database of customers, and for each number, if they call again, the menu should present them their last choice first. I don't know if I'm being clear. |
| 21:21.41 | redondos | I will get into this when I start using asterisk, I'm buying the hardware this week. |
| 21:21.47 | Katty | [TK]D-Fender: any clue how long i'm going to have this dull pain? :< |
| 21:22.07 | _Paulo_ | while at the database subject, astdb can use an external database engine? |
| 21:22.27 | redondos | Yeah, I was going to get into that eventually. |
| 21:22.38 | [TK]D-Fender | redondos : All doable without an external database (you could use *'s internal one for that quite likely) |
| 21:22.57 | Zodiacal | any ideas? |
| 21:23.00 | [TK]D-Fender | Katty: You could offset it with a SHARP pain, but I don't think you want that.... |
| 21:23.02 | afrosheen | redondos: rather than outsmart the customers, just put a menu choice in your primary default IVR that says 'first time callers hit one, everyone else hit 2' |
| 21:23.08 | [TK]D-Fender | Katty: Mew. (belated) |
| 21:23.47 | Tamarisk | In the sip.conf when defining softphones in sip.conf, I start with : - [adrian] ¬ type=friend ¬ secret=xxxx ¬ qualify=yes ¬ nat=no ¬ host=the ip of the softphone computer or the ip of asterisk? |
| 21:23.50 | afrosheen | redondos: btw what hardware are you buying for it |
| 21:24.06 | [TK]D-Fender | Katty: Went out last night on a pseudo-date and came back to find the GF evacuated our ebdoom, dumped a bunch of stuff from the basement back up into it and has moved down there. Nice thing to come home to at 2:30am |
| 21:24.11 | znoG | could be neat though, if you were using * for tech support calls, to say to a customer: "if you are calling about tech issue <last issue opened>, please press 1, otherwise press 2" |
| 21:24.29 | Katty | [TK]D-Fender: no clue then, i take it? |
| 21:24.35 | afrosheen | yeah when my ebdooms get evacuated, I call the cops |
| 21:24.44 | [TK]D-Fender | Katty: Gone through the obvious drug choices already? |
| 21:24.53 | [TK]D-Fender | bedroom* |
| 21:24.56 | Katty | [TK]D-Fender: huh? |
| 21:25.05 | Katty | [TK]D-Fender: i'm on day 5 after surgery |
| 21:25.07 | [TK]D-Fender | Katty: for your dull pain. |
| 21:25.16 | Katty | [TK]D-Fender: 600mg of ibeuprofen |
| 21:25.34 | IRC_User | anyone familiar with sccp.conf for cisco 12SP phones? |
| 21:25.56 | [TK]D-Fender | If that doesn't doo it you may have over-adapted to it. Try switching between Acetomenophen and aspin, whichever you don't have issues with. |
| 21:26.11 | [TK]D-Fender | Asperin. |
| 21:26.24 | Katty | the pain is tollerable without painkillers |
| 21:26.27 | [TK]D-Fender | ack... I feel like shit... my insides are all scrambled today |
| 21:26.35 | Katty | but...painful |
| 21:26.41 | Katty | and i don't really want to tolerate painful :/ |
| 21:27.02 | [TK]D-Fender | Katty: Switch drugs then. |
| 21:27.26 | [TK]D-Fender | ok, I've got to get outta here... later all... |
| 21:27.29 | znoG | alcohol? |
| 21:27.32 | znoG | alcohol is good |
| 21:27.35 | [Outcast] | hey has come where rxfax will compress the image. so the fax looks squished? |
| 21:28.00 | [Outcast] | s/come/come across/ |
| 21:28.00 | redondos | afrosheen: It isn't my business, really, so the decisions were made by the guy who's putting the money. We're getting a clone. This one, to be precise: http://xrl.us/j5n2 |
| 21:28.58 | redondos | afrosheen: Plus a Sempron 2800 64bit, 512mb. How many simultaneous connections might that be able to handle if not using SIP but only playing menus and relaying to destination numbers? |
| 21:29.14 | badboyz | is it possible to take all calls that come in through a certain DID, and shift that call over to another server? |
| 21:29.27 | badboyz | i dont want to shift it based on extenion, i want a shift based on DID |
| 21:30.19 | redondos | znoG: That is almost exactly what I'd like to achieve with asterisk. And from what you've been telling me it seems doable... :) |
| 21:30.28 | FuriousGeorge | is there an "unofficial prefered" scripting language to use with the api |
| 21:30.30 | redondos | Or.. not? |
| 21:30.34 | znoG | redondos: sure, anything is doable with AGI :) |
| 21:30.42 | znoG | redondos: (that's the first thing that comes to mind anyway) |
| 21:31.02 | iCEBrkr | FuriousGeorge: C |
| 21:31.07 | redondos | AGI? I'm very new here :) |
| 21:31.18 | znoG | redondos: although if you keep it in the * database, it would be a lot faster than querying MySQL every time someone calls |
| 21:31.19 | FuriousGeorge | iCEBrkr: i didnt think C was a scripting language |
| 21:31.24 | kpettit | anybody wityh some Sangoma experience? trying to install some A200 fx cards |
| 21:31.31 | znoG | redondos: yes, AGI. www.voip-info.org << read :) |
| 21:31.33 | iCEBrkr | FuriousGeorge: hehe |
| 21:31.46 | kpettit | I got wanpipe installed, zaptel re-compiled. Trying to figure out how to configure things from there |
| 21:31.52 | redondos | znoG: Right. Well, as long as the engine is fast enough to handle a decently-sized databse :) |
| 21:31.53 | Netgeeks | AGI is evil |
| 21:32.02 | iCEBrkr | kpettit: wancfg |
| 21:32.09 | znoG | i use AGI for every call on this system, works OK for me. Although only 25 odd extensions |
| 21:32.11 | FuriousGeorge | iCEBrkr: could you lmit your responses to scripting languages :) and in the form of a question, please |
| 21:32.16 | kpettit | iCEBrkr, been using that, not working so well |
| 21:32.17 | znoG | and I don't Dial directly from the AGI, i exit and let Asterisk dial |
| 21:32.36 | iCEBrkr | FuriousGeorge: It really all depends on what you're trying to do and the amount of calls |
| 21:32.56 | iCEBrkr | znoG: AGI() is fine for something small like that. |
| 21:33.07 | kpettit | iCEBrkr, when I do the autodetect it shows "AFT-A200-SH Slot=4....." |
| 21:33.18 | iCEBrkr | ok |
| 21:33.25 | kpettit | and then it pukes |
| 21:33.28 | _Paulo_ | I want to patch app_authenticate.c so if the parameter starts with "!", app_authenticate will run an arbitrary shell comand. Is this a good idea? |
| 21:33.29 | iCEBrkr | pukes? |
| 21:33.36 | FuriousGeorge | iCEBrkr: one could use bash, no? |
| 21:33.38 | kpettit | Failed to get Card Type from selected card line! |
| 21:33.45 | iCEBrkr | kpettit: eep |
| 21:33.48 | kpettit | Failed to get Card Type from selected card line!! line: AFT-A200-SH SLOT=4 BUS=2 IRQ=11 CPU=A PORT=PRI |
| 21:33.55 | kpettit | that's the whole line. not sure why it failes |
| 21:33.57 | [av]bani | man i wish people would stop buying $80 phones and whining that they dont perform like $400 phones |
| 21:34.33 | iCEBrkr | FuriousGeorge: I'm just saying. On a large system, AGI() + PHP + MySQL is slow |
| 21:34.45 | iCEBrkr | FuriousGeorge: It really all depends on what you wanna do |
| 21:34.54 | redondos | Do you think I will I be ok if I pretend to have about 30 simultaneous calls max on a sempron 2.8 with 512 MiB of memory? |
| 21:35.34 | Mavvie | heh. olle is funny. |
| 21:35.35 | Netgeeks | AGI is evil isn't a fact it's just my own personal opinion I try to faust on all who listen..... I personally don't like it because of the resource cost involved in the use of AGI |
| 21:35.49 | FuriousGeorge | iCEBrkr: i wanna assign parking spots to meetme's control the device state, and wright a script that determines who can join, when to terminiate the room, etc |
| 21:36.23 | znoG | Netgeeks: yup, but there's also a lot to gain from it. I can't see any other way to achieve real complex dialplan stuff without using AGI |
| 21:36.39 | iCEBrkr | FuriousGeorge: I'd seriously try to keep a lot of that in extensions.conf |
| 21:36.43 | Netgeeks | actually you'd be amazed what you can do with dialplan |
| 21:36.47 | IRC_User | or use php |
| 21:37.03 | iCEBrkr | What Netgeeks said |
| 21:37.09 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:37.17 | FuriousGeorge | iCEBrkr: i looked at it, no way to get it to work right w/o assigning different sequences to join, leave, and kill the room |
| 21:37.42 | FuriousGeorge | kinda defeats the purpose of the pretty blinkey lights with the buttons |
| 21:37.58 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 21:38.38 | iCEBrkr | FuriousGeorge: Then there's the whole thought process of 'Do I really need this feature' 'do other PBXs support such features?' etc. |
| 21:38.41 | znoG | Netgeeks: using AEL or just standard dialplan? |
| 21:38.49 | FuriousGeorge | no and yes |
| 21:39.10 | Netgeeks | ael is more crippled than regular dialplan, it doesn't support all the stuff you can do in basic dialplan mode |
| 21:39.32 | iCEBrkr | I understand that Asterisk is pretty 'powerful', but think about other PBX's and what they do. You're pretty much locked into what they give you. Asterisk is a software PBX, it's meant for routing and handling calls.. Not doing 10000 cartwheels. :) |
| 21:39.37 | FuriousGeorge | its more of a project for fun of mine, and other bx have status lights with parking. barge support, and reverse transfers |
| 21:39.45 | Netgeeks | ael just looks more like a programming language to those folks who have the misfortune of only being exposed to the most recent programming languages |
| 21:41.36 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 21:42.03 | afrosheen | redondos: I suppose that hardware will work, but we've had much, much better luck with hardware we bought from dell than a server we built |
| 21:42.22 | afrosheen | for whatever reason, asterisk just wasn't happy on our custom server |
| 21:43.10 | redondos | Too bad. I hope it doesn't work like that for me. Where I live, getting non-built by yourself machines is a little out of fashion. |
| 21:43.17 | FuriousGeorge | iCEBrkr: i gotta disagree. the thrings i described above are features not cartwheels |
| 21:43.21 | redondos | See, it's too expensive to get something with a real warranty :) |
| 21:43.55 | FuriousGeorge | i would use parking spots as easy as meetmes and lose the barge for the simplicity, if i could get devstate to work with parking |
| 21:43.55 | afrosheen | redondos: well since this is a business here, we quit jacking around and got a dell 2u 2850 server |
| 21:44.09 | redondos | afrosheen: nice :) |
| 21:44.17 | FuriousGeorge | and i know there is a branch but its for head (trunk whatever) and i know theres an easy way to do this |
| 21:44.21 | afrosheen | they're really not too expensive if you get a decent account manager |
| 21:44.33 | *** join/#asterisk cpwp (n=kvirc@bacchus.hognaston.com) |
| 21:44.39 | afrosheen | we liked it so well we bought 2 more :) |
| 21:44.45 | redondos | hehe |
| 21:45.13 | cpm | I've got some dell built boxes, and some 'local dude' boxes, the local dude boxes are far superior in all aspects. and the ram slots don't melt off. |
| 21:45.16 | FuriousGeorge | how does that work anyway, there's the xmpp branch and the multiparking branch, but are they both necesarrily gonna make it into 1.4? |
| 21:45.19 | iCEBrkr | FuriousGeorge: Maybe you're going about it the wrong way? |
| 21:45.27 | iCEBrkr | FuriousGeorge: That's pretty common around here :) |
| 21:45.42 | FuriousGeorge | iCEBrkr: what is common? device states for parking spots? |
| 21:45.43 | mikefoo | afrosheen: what type of shop are you running? |
| 21:45.57 | afrosheen | mikefoo: a prototyping company for surface mount stuff |
| 21:46.00 | FuriousGeorge | thats what im shooting for. those snazzy snoms, you know |
| 21:46.13 | cpm | cool |
| 21:46.16 | afrosheen | have about 45 extensions now, a sip trunk to commpartners and a troublesome tdm400 card for backups |
| 21:46.49 | afrosheen | I think I need to call commpartners today about our weird sip echo |
| 21:46.53 | redondos | Is there no *@home for 64 bit? |
| 21:47.02 | redondos | Does asterisk run well in 64-bit mode at all? |
| 21:48.23 | znoG | daaaaamn it takes a lot to transcode any codec to speex |
| 21:48.36 | znoG | like 160ms |
| 21:48.39 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
| 21:49.13 | FuriousGeorge | iCEBrkr: if i could park directly to a spot, and monitor the chanisavail on a parking spot that solves my problem, but i cant |
| 21:49.13 | afrosheen | ouch |
| 21:49.51 | *** join/#asterisk redondos (n=redondos@190.48.53.178) |
| 21:50.01 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:50.22 | redondos | Forgive me, I'm having some connection problems. Did anyone answer my question about *@home being available for x86_64? And if Asterisk runs well at all in 64bit mode. |
| 21:52.19 | afrosheen | don't touch 64 bit anything, trust me |
| 21:52.41 | *** join/#asterisk |omni| (i=rob@net98.limelyte.net) |
| 21:52.52 | afrosheen | the chips are great but there is little to no advantage to running 64bit windows or linux right now, and alot of the software is missing, buggy, or both |
| 21:52.55 | [av]bani | x86_64 is fine |
| 21:53.04 | redondos | Heh, I've got my laptop with Gentoo in 64bit, it's pretty much ok, other than not having Flash I can do whatever I want. |
| 21:53.08 | [av]bani | i didnt have to tweak ztdummy at all |
| 21:53.15 | afrosheen | well I guess it's changed since I tried it last summer |
| 21:53.27 | [av]bani | well sure, asterisk went from 1.0 to 1.2 also |
| 21:53.32 | [av]bani | so yeah, a lot has changed |
| 21:53.39 | redondos | But it is, after all, my first experience with Asterisk and with VOIP in general, so I guess 32bit will make things easier for me. What do you think? |
| 21:53.44 | afrosheen | that has nothing to do with the state of 64 bit computing in general |
| 21:53.48 | rt | when I typed make linux26 to build it, it doesn't set USE_RTC because its guarded by an #ifdef __i386__ or something. |
| 21:53.57 | [av]bani | x86_64 has been the same since last summer; works fine for me! |
| 21:54.07 | russellb | rt: that has been changed to include x86_64 as well |
| 21:54.09 | [av]bani | and i do cross platform developmet (win32/osx/linux) on it no problem |
| 21:54.09 | afrosheen | [av]bani: what os are you running |
| 21:54.16 | [av]bani | afrosheen: fedora, and ubuntu |
| 21:54.21 | [av]bani | both work fine |
| 21:54.28 | afrosheen | wow you have ubuntu 64 running 'fine'? |
| 21:54.28 | rt | russellb: well, i stand corrected then. :-) |
| 21:54.30 | hypa7ia | anyone know if FWD's iax is being sketchy at the moment? |
| 21:54.39 | rt | getting mpg123 to compile 64 bit seemed... difficult though. |
| 21:54.47 | [av]bani | yes, in fact i built an embedded x86_64 compactflash asterisk box around ubuntu |
| 21:54.51 | russellb | hypa7ia: !!! |
| 21:54.54 | afrosheen | hmm |
| 21:54.56 | [av]bani | which i am running now |
| 21:55.02 | [av]bani | i am compiling oh323 on it as we speak |
| 21:55.17 | afrosheen | I guess things are maturing then, but again, I don't think it's at the same level as 32 bit is currently |
| 21:55.19 | [av]bani | the box i am irc'ing from is x86_64 fedora |
| 21:55.30 | [av]bani | eh? 32 bit is 32 bit on x86_64. it runs fine |
| 21:55.39 | *** join/#asterisk davidcsi (n=dvillasm@20.Red-83-32-54.dynamicIP.rima-tde.net) |
| 21:55.40 | [av]bani | it runs 32 bit libs natively. |
| 21:55.51 | [av]bani | i run 32 bit opengl games on it |
| 21:55.52 | [av]bani | no problem |
| 21:55.56 | [av]bani | zero nada zilch |
| 21:56.07 | FuriousGeorge | is it just me, or when asterisk gets an unavail cid from the telco does it display asterisk@ip |
| 21:56.17 | [av]bani | yes, thats what * does |
| 21:56.21 | [av]bani | it fills in the blanks :< |
| 21:56.23 | hypa7ia | russellb!! |
| 21:56.24 | afrosheen | haha |
| 21:56.34 | hypa7ia | how goes dude |
| 21:56.35 | [av]bani | you can change that |
| 21:56.39 | redondos | I still haven't been able to make up my mind. *@home or not? It is only available for 32 bit architectures, while I do have an x86_64-enabled CPU. |
| 21:57.10 | [av]bani | *@home is poo |
| 21:57.30 | [av]bani | unless youre a complete linux noob, youre better off installing a real distro and doing * yourself |
| 21:57.47 | redondos | Ok, that's the sort of answer I was expecting to get. |
| 21:57.49 | redondos | Thanks |
| 21:57.58 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-119-77.dsl.irvnca.pacbell.net) |
| 21:58.00 | mxmasster | hi all |
| 21:58.12 | redondos | hello. |
| 21:58.29 | [av]bani | if you are experienced in linux, you'll outgrow *@h in about 5 minutes, and wish you hadnt' wasted your time with it |
| 21:59.07 | redondos | Very useful information you're giving me. I was wondering the same thing: if *@home would tie my hands too tight. |
| 21:59.13 | afrosheen | [av]bani: on the other hand, if you're an * noob, you may find it helpful to build a working system before you want to get your hands dirty |
| 21:59.19 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:59.48 | redondos | I am a 100% asterisk/voip newbie, but am pretty much experienced with linux administration and some developing. |
| 21:59.49 | afrosheen | that way as a newbie you can see what gets written to config files when you make changes and see how everything interconnects |
| 21:59.49 | [av]bani | afrosheen: i didnt really see how *@h helped me build a "working system" any quicker than make install did |
| 21:59.58 | redondos | The 'ease of use' that @home provides, does it come from AMP? |
| 22:00.03 | redondos | Because that is.. just installable. |
| 22:00.17 | afrosheen | [av]bani: so all your * configs are irrelevant? |
| 22:00.21 | [av]bani | afrosheen: since the tools *@h installs are highly scripted and pretty convoluted even for experienced *'ers |
| 22:00.32 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.184) |
| 22:00.33 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 22:00.35 | [av]bani | afrosheen: so you dont really learn anything from reading them, than you do from the example confs |
| 22:00.53 | afrosheen | well I'm not defending *@home since I've never used it, but I'm imagining it's similar to a regular * plus AMP install |
| 22:00.59 | [av]bani | redondos: pretty much |
| 22:01.15 | mxmasster | i've heard "follow me" systems where the user is prompted to press "1" to leave a message, or "2" to find the person. If they press 2 then it asks for the name, and then calls through the list and gives the receipient the option to answer |
| 22:01.19 | [av]bani | afrosheen: and as far as guis go, amp is pretty limited. noobs wont really learn * via using it. |
| 22:01.28 | mxmasster | were can i find an example configuration for something like this? |
| 22:01.36 | [av]bani | afrosheen: amp does everything for you and hides it in complex pre written scripts, which are no good for * noobs |
| 22:01.43 | redondos | Beautiful. I've made up my mind. I understand what you say, afrosheen, but that is just not my style. Even though I am new to asterisk, I like to be able to understand how things really work. |
| 22:01.58 | [av]bani | afrosheen: * noobs are better off tinkering with the default example conf's from asterisk source install |
| 22:02.04 | afrosheen | redondos: that's fine, just buy some strong coffee and make friends with the wiki |
| 22:02.08 | [av]bani | and tbh, * is not hard |
| 22:02.22 | [av]bani | its mainly learning the voip and * terminology, which amp hides from you |
| 22:02.26 | |omni| | mxmasster: : assuming that one person may "own" each of the extensions and could be sitting at any of them, just ring them all at once..the one that answers gets the call |
| 22:02.32 | |omni| | I do that with my cell / business |
| 22:02.45 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-60-205.dialin.kamp-dsl.de) |
| 22:02.52 | mxmasster | omni: doing that now - looking for something a little more sophisticated |
| 22:03.03 | afrosheen | oh, redondos, there's a good O'Reilly book on the net about *, and it's free to download |
| 22:03.05 | davidcsi | has anyone had the problem with PRI ISDN of not getting the connect message? only sometimes.... |
| 22:03.20 | |omni| | so the system has a list of extensions and loops through one by one? |
| 22:03.40 | freat | hi... when documentation talks about "the asterisk database"... what are they referring to? For example, agi command of DATABASE PUT |
| 22:03.43 | redondos | afrosheen: heh, I guess I'll do that. You and [av]bani, you've helped a lot. Thanks for aiding my decision. |
| 22:03.47 | afrosheen | http://voipspeak.net/images/stories/orielly/AsteriskTFOT.zip |
| 22:03.52 | afrosheen | grab that and read up |
| 22:03.52 | redondos | afrosheen: Ah, nice. Thank you for the URL. |
| 22:04.02 | redondos | I will eat it ASAP. |
| 22:04.32 | redondos | Well, as soon as I build the computer. |
| 22:04.46 | *** join/#asterisk tletourneau (n=tom_remo@12-219-187-158.client.mchsi.com) |
| 22:05.07 | [av]bani | if your boss says "i have ot have a pbx in 5 minutes" you could use *@home for that |
| 22:05.19 | [av]bani | if youve never touched a voip phone or linux before |
| 22:05.38 | redondos | heh, ok. |
| 22:05.44 | [av]bani | for anything serious, dont use it |
| 22:05.54 | [av]bani | not yet anyway.. maybe they will fix it up later into something nice |
| 22:06.01 | [av]bani | amp needs a lot of work too |
| 22:06.07 | redondos | I've never administered a PBX, though. But I guess the documentation will suffice. |
| 22:06.09 | afrosheen | yeah amp is getting major changes |
| 22:06.28 | afrosheen | I think courtnage is too busy with his business to put in the hours right now |
| 22:06.31 | redondos | Can you still use AMP for monitoring? |
| 22:06.40 | mxmasster | |omni|: yes that is what it is doing now |
| 22:06.45 | mxmasster | i want the caller to record the name |
| 22:06.48 | afrosheen | yeah you can use flash operator panel to monitor, it's part of amp |
| 22:06.51 | redondos | Because some non-technical people involved in the busyness would like a graphical interface to see if the system is performing well. |
| 22:06.56 | redondos | k, cool. |
| 22:06.57 | mxmasster | and let the recipient decide to receive the call |
| 22:07.05 | redondos | s/busyness/business. |
| 22:07.11 | afrosheen | our secretary likes FOP |
| 22:07.15 | mikefoo | performing well? |
| 22:07.44 | afrosheen | yeah she just uses it to see who's on the phone etc. and what people's extensions are |
| 22:08.00 | *** join/#asterisk _deg_ (n=deg@200-233-51-145.corp.ajato.com.br) |
| 22:08.03 | redondos | mikefoo: Ok, I meant monitoring the amount of concurrent calls and see some logs. Performance, I can give them that with other tools. |
| 22:08.12 | mikefoo | ahh like a "front desk" program |
| 22:08.40 | redondos | FOP: http://www.asternic.org/demo.html |
| 22:08.55 | redondos | Neat. |
| 22:09.23 | afrosheen | for reporting..amp does that too, has graphical CDR stuff, our CFO loves that |
| 22:09.47 | redondos | Awesome. |
| 22:12.19 | badboyz | anyone get the feature pickupexten to work? |
| 22:12.25 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-251.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:13.05 | redondos | How old is Asterisk? |
| 22:13.14 | trixter | started originally about 1999 |
| 22:13.29 | redondos | Thanks. |
| 22:14.55 | freat | I would like to kick off an agi script when an agent answers a call in a queue. Any pointers on how to do this? |
| 22:16.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:17.20 | _Sam-- | im trying to register one asterisk box to another asterisk box via IAX....but the box that is registering get "rejected" when i check the iax2 show registry...and no logs on the other box...what should i check first? |
| 22:18.40 | freat | on the other box try 'iax2 debug on' |
| 22:18.55 | freat | or just |
| 22:19.00 | freat | iax2 debug, sorry |
| 22:19.11 | freat | then to turn off debugging, use 'iax2 no debug' |
| 22:21.15 | mikefoo | can anyone recommend a toll free did provider? |
| 22:21.47 | *** join/#asterisk ircritesh (n=riteshir@natint3.juniper.net) |
| 22:22.09 | ircritesh | hello... |
| 22:22.09 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 22:22.37 | ircritesh | Does anyone know how to skip out of the asterisk realtime queues to go to a voicemail??? |
| 22:23.16 | ircritesh | anyone out there??? |
| 22:23.54 | glm2k | mikefoo: try sixtel. |
| 22:23.57 | *** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0) |
| 22:24.02 | ircritesh | sixtel?? |
| 22:24.05 | glm2k | mikefoo: alias iax.cc |
| 22:24.19 | ircritesh | let me check |
| 22:24.25 | generalhan | hey guys, can anyone tell me what this means :: Feb 21 15:23:50 NOTICE[21321]: chan_sip.c:3753 copy_header: No field 'From' present to copy |
| 22:24.44 | *** join/#asterisk themikester60 (n=mikey@209-83-240-50-static.dsl.oplink.net) |
| 22:25.19 | themikester60 | has anyone noticed a lack of sensitivity with asterisk when it comes to detecting touch tones? |
| 22:25.34 | *** join/#asterisk Gir19 (i=Gir@67.189.110.174) |
| 22:25.40 | glm2k | ircritesh: that was for mikefoo. he asked for a toll free provider :) |
| 22:25.47 | ircritesh | got it ..sorry |
| 22:25.56 | afrosheen | themikester60: yeah, it's especially bad on zaptel |
| 22:26.01 | ircritesh | i wanted to know if there is a way to skip out of the asterisk realtime queues into a voicemail |
| 22:26.17 | ircritesh | dont' want the callers to keep waiting if they don't mind leaving a vmail |
| 22:26.26 | themikester60 | afrosheen, is there anyway to alter the sensitivity level? |
| 22:26.54 | ircritesh | such as press 5 to leave a vmail or wait for next available rep |
| 22:27.00 | afrosheen | themikester60: with zap, you change the rx gain..with other stuff I don't know |
| 22:27.22 | afrosheen | themikester60: what are you using for incoming calls |
| 22:27.58 | Gir19 | Ok, anyone have a link to some resources for setting up multiple digium cards, such as a TE11XP and a TD20B, I have searched voip-info and asteriskguru and cannot find anything about configuring multiple cards on a system. |
| 22:28.05 | themikester60 | both iax and zap |
| 22:28.12 | themikester60 | iax takes preference over zap though |
| 22:28.23 | afrosheen | themikester60: and you're having problems with dtmf over iax? |
| 22:28.40 | afrosheen | or zap specifically |
| 22:29.33 | *** join/#asterisk Maxx4life (n=Maxx4lif@71-35-210-12.slkc.qwest.net) |
| 22:29.45 | themikester60 | afrosheen, well I'm not exactly sure, I wasn't the one who noticed the issues at the time and its possible the person who reported the issue could have been using a zap channel, all I know is that they had issues with touch tones and the setup of the server is to use as many iax channels as possible until they run out and then default to zap |
| 22:31.10 | afrosheen | themikester60: I'd suspect it's zap's fault, since the iax trunk should have adequate volume coming from the trunk provider |
| 22:31.41 | jbalcomb | how to i let asterisk know that the third line in my PRI card is actually going out to a PBX? |
| 22:31.51 | afrosheen | themikester60: try playing with the rxgain= settings in /etc/asterisk/zap*.conf for your zap channels, go up or down by 4 and keep trying to get it to fail |
| 22:32.15 | afrosheen | themikester60: you can also use ztmonitor on each zap channel and watch the incoming volume levels live |
| 22:32.34 | afrosheen | you want them at about halfway across the graph for optimal dtmf detection |
| 22:32.53 | *** part/#asterisk ircritesh (n=riteshir@natint3.juniper.net) |
| 22:34.04 | themikester60 | afrosheen, thanks alot I'll give that a shot |
| 22:37.11 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 22:38.46 | Gir19 | Anyone have a link or info for setting up multiple digium cards, such as a TE11XP and a TD20B, I have searched voip-info and asteriskguru and cannot find anything about configuring multiple cards on a system. |
| 22:39.10 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 22:39.27 | I-MOD | Gir19: order of modprobe/insmod matters |
| 22:40.00 | I-MOD | ideally, you want to modprobe the t1/e1 card first and then the analog cards |
| 22:40.11 | I-MOD | it makes setup easier |
| 22:40.48 | I-MOD | then in zaptel.conf, do your spans and everything associated with those, and then do the analog cards below that |
| 22:41.23 | Gir19 | ok, so just setit up in the order of the cards initiations. |
| 22:41.30 | I-MOD | correct |
| 22:41.50 | Gir19 | ok, that is what I assumed, but wanted to make sure. |
| 22:43.12 | Gir19 | cause I need to get faxing to work and using a ATA just doesn't cut it, so I am hoping just putting in actual pots lines via the td20b card, that there will be fewer line errors due to network bursts. |
| 22:46.05 | Gir19 | I also have another issue that isn't as important, but my clients are still annoyed from it. Basically if they use there phone internally everything sounds fine, but if they make a call to or from POTs there is poping on the line only on their side, I have removed all chances of it being an IRQ issue, but have no clue what else it could be due to. |
| 22:46.28 | *** part/#asterisk davidcsi (n=dvillasm@20.Red-83-32-54.dynamicIP.rima-tde.net) |
| 22:48.01 | I-MOD | Gir19: dunno about that |
| 22:48.13 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 22:48.22 | freat | Gir19: any other processes on the box... maybe software raid? I had this happen with soft-raid on Linux |
| 22:48.42 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 22:49.15 | freat | Gir19: maybe test listening to MOH from the asterisk box from one of the extensions. This will ensure that Asterisk is in the loop and see if you are still getting popping. |
| 22:49.37 | freat | Gir19: could be that the SIP phones are reinviting and bypassing asterisk... just a thought |
| 22:49.58 | Gir19 | I have reinvite disabled |
| 22:50.36 | *** part/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 22:51.21 | *** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0) |
| 22:51.28 | russellb | Gir19: make sure dma is on for the hard drive ... |
| 22:51.41 | Gir19 | As for the raid it is hardware raid 1. |
| 22:51.49 | freat | Gir19: no X windows right? |
| 22:52.02 | freat | Gir19: or framebuffer |
| 22:52.22 | Gir19 | I do not know about the frame buffers, but there is no x windows. |
| 22:53.04 | freat | http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html |
| 22:53.07 | Gir19 | I also have not checked dma settings yet. that should give me a few things to check on then. thanks for the input. |
| 22:53.08 | }btorch{ | hello ... I have setup my * box to talk to a siemens officePro using a second PRI ... but asterisk keeps giving me some NOTICE: chan_zap.c: pri_channel PRI got event HDLC abort ... |
| 22:53.27 | }btorch{ | is that normal ? |
| 22:53.49 | Mavvie | }btorch{: which span? |
| 22:53.53 | }btorch{ | span 1 |
| 22:53.58 | }btorch{ | I only have one span |
| 22:54.00 | Mavvie | I got that one span 3 . |
| 22:54.04 | }btorch{ | TE110P |
| 22:54.11 | Mavvie | with the result that all my quad pris don't use span 3 |
| 22:54.33 | }btorch{ | what does that mean ? |
| 22:55.05 | afrosheen | Gir19: what do you have for outbound calls to POTS on your server |
| 22:55.09 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe203.t-net.net.ve) |
| 22:55.32 | }btorch{ | my * works almost fine using the second PRI but not it I setup the as a T1 analog with E&M ... |
| 22:56.11 | Gir19 | I am using a TE110P to a T1 with the first 12 channels set to fxs_ks and the last 12 setup for nethdlc. |
| 22:56.33 | kippi | how i am going to be able to install iax? |
| 22:56.55 | }btorch{ | Gir19: are you connecting your TE110P to a PBX ? |
| 22:57.04 | kippi | I can't find any information anywhere to help me out! |
| 22:57.19 | Gir19 | but I just got some more info that the server seems to be having frame loss to and from the T1 haven't tracked it down yet, but will start looking into that as well. |
| 22:57.26 | epablo | Hi guys.. anyone with 5 min to help me configure a tormeta2 with asterisk |
| 22:57.54 | afrosheen | Gir19: I was going to mention echo cancellation on zap channels but it probably doesn't apply to you |
| 22:58.05 | Gir19 | The TE110P is going directly to the phone company, they say it is not a pbx. |
| 22:58.36 | epablo | I'm getting a: :No functioning zap hardware found in /proc/zaptel, loading ztdummy. when loading the tor2 module |
| 22:59.11 | I-MOD | Gir19: pri? e&m? e&m wink? feat d? feat b? |
| 22:59.21 | I-MOD | what kind of t1 is it? |
| 22:59.53 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 23:00.09 | }btorch{ | Gir19: are you getting a red alarm ? |
| 23:00.32 | Gir19 | according to the phone company it is 12 fso channels and the other 12 are hdlc, but I am sure they have been missinforming me, they also said it is not using e&M or pri. |
| 23:01.26 | Gir19 | }btorch{ no it is green and I can use the card without a problem, there is just poping on the pots lines. |
| 23:01.39 | }btorch{ | I wish there were a tool to find out what signalling the T1 from PBXs or CO use |
| 23:02.21 | }btorch{ | Gir19: and you have zaptel.conf setup as e&m=1-12 ? |
| 23:03.28 | epablo | ZT_SPANCONFIG failed on span 1: Invalid argument (22) ?? Any ideas |
| 23:04.11 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:05.05 | Gir19 | }btorch{ no it is fxsks=1-12 and nethdlc=13-24 |
| 23:05.26 | I-MOD | use zttool to check the alarm state |
| 23:05.50 | I-MOD | as i'm pretty sure that fx??s is used only for analog lines |
| 23:06.12 | Gir19 | zttool does not say there are any line errors. |
| 23:06.43 | I-MOD | try setting it up for e&m and if that doesnt work, e&m wink |
| 23:06.44 | }btorch{ | I tried to setup my TE110P to use e&m wink with my Siemens T1 which was setup to use e&m winkstart and pretend to be a 24 trunk ... but I could never get the red alarm off |
| 23:07.10 | }btorch{ | it sucked |
| 23:07.26 | I-MOD | did you try featb and featd? |
| 23:07.37 | }btorch{ | who me ? |
| 23:07.39 | I-MOD | yes |
| 23:08.19 | Gir19 | well, I need to goto a meeting, I'll be back later after I try some of this stuff and see if any of it helps. Thanks again. |
| 23:08.23 | I-MOD | later |
| 23:08.28 | badboyz | is there a more technical name to call it, when you have 2 asterisk servers connected? |
| 23:08.37 | *** part/#asterisk Gir19 (i=Gir@67.189.110.174) |
| 23:08.44 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 23:09.04 | I-MOD | if you need a flashy technical term, say that they're trunked together |
| 23:09.11 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 23:09.11 | }btorch{ | is featd/b an option on /etc/zaptel.conf ? |
| 23:09.25 | }btorch{ | i don't see it |
| 23:09.57 | I-MOD | they're in zapata.conf iirc |
| 23:11.56 | }btorch{ | no I just tried setting up zaptel.conf with e&m and zapata.conf with e&m_w |
| 23:12.25 | }btorch{ | but why does it matter to configure the zapata.conf file if the red alarm is on ? |
| 23:12.58 | }btorch{ | don't I need to have the card on green before I can run * and have it working with zap ? |
| 23:17.00 | [Outcast] | can anyone how rxfax effects cpu utilization ? |
| 23:17.06 | [Outcast] | it is cpu heavy? |
| 23:17.19 | [Outcast] | s/it is/is it/ |
| 23:18.42 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 23:20.54 | epablo | Has anyone used a TE400P card? |
| 23:25.17 | websae | is that t1 card? |
| 23:25.28 | epablo | YEs a 4xt1/e1 card |
| 23:25.46 | websae | i have heard a lot of people like that card |
| 23:25.51 | websae | do you have one? |
| 23:26.27 | epablo | Yes.. I have it a new server.. but I can't seam to make it work.. |
| 23:26.39 | epablo | zaptel doesn't see it |
| 23:26.47 | websae | hrm |
| 23:28.24 | epablo | I'm installing the tor2 module.. but it doesn't let me do anymore :S |
| 23:28.41 | websae | hrm |
| 23:28.46 | *** join/#asterisk dlublink (n=dlublink@modemcable114.38-201-24.mc.videotron.ca) |
| 23:28.48 | dlublink | Hey |
| 23:28.50 | websae | I have no experience with that card, I wish i could help you |
| 23:28.58 | epablo | Hi |
| 23:29.03 | dlublink | Is there a app in the dialplan I can use to output a variable? |
| 23:29.08 | dlublink | to the console |
| 23:29.09 | epablo | websae: Thanks anyway |
| 23:30.22 | epablo | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+NoOp |
| 23:30.28 | dlublink | thanks |
| 23:30.29 | epablo | That should do the trick |
| 23:30.56 | dlublink | Variables always take the format ${CALLERID}? |
| 23:31.04 | dlublink | ${VAR_NAME} |
| 23:31.05 | websae | i just setup a new asterisk box---and i can't even see my sip phone trying to connect to it, anyone have any ideas? |
| 23:31.16 | dlublink | any nat involved? |
| 23:31.26 | websae | yes |
| 23:31.30 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 23:31.35 | websae | but i can't even see the phone hitting the server |
| 23:31.49 | dlublink | ok |
| 23:31.52 | epablo | sip debug |
| 23:31.55 | dlublink | and have you set nat=yes in sip.conf |
| 23:31.58 | websae | absolutely nothing |
| 23:32.02 | websae | yep i have nat=yes |
| 23:32.11 | dlublink | and on the sip device |
| 23:32.17 | websae | this is quite odd |
| 23:32.23 | websae | yep, it's enabled on both |
| 23:32.50 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 23:33.14 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 23:33.25 | epablo | Well guys.. got a go.. take care |
| 23:33.28 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe203.t-net.net.ve) |
| 23:33.34 | websae | any other suggestions |
| 23:33.35 | websae | ? |
| 23:35.03 | dlublink | with regards to enumlookup, anyone know what the best macro to use is? I have tried several and they give me errors about the set command |
| 23:38.56 | websae | anyone have any suggestions why I can't see my sip phone trying to register? |
| 23:39.00 | websae | i have verbose set to 10000000000 |
| 23:39.03 | websae | also have sip debug on |
| 23:40.50 | websae | i see no packets hitting the server at all from the phone |
| 23:41.14 | russellb | 100000000000? I think there are some messages a few levels above that |
| 23:42.01 | websae | what should i set it to |
| 23:42.03 | websae | set verbose |
| 23:42.08 | websae | to what? |
| 23:43.05 | russellb | i was joking |
| 23:43.05 | websae | lol |
| 23:43.05 | websae | i know |
| 23:43.05 | generalhan | hahhaha |
| 23:43.05 | websae | haha |
| 23:43.05 | russellb | we don't have anything that goes over 4 or 5 |
| 23:43.05 | websae | normally i have it like 3 |
| 23:43.05 | websae | but just for kicks and giggles i wanted to see what i am missing out on |
| 23:43.05 | websae | i am getting nothing at all from the phone |
| 23:43.09 | *** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0) |
| 23:43.45 | russellb | ok, looks like 7 is the highest value that would matter :) |
| 23:43.50 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 23:44.23 | websae | i just compiled asterisk on this new server |
| 23:44.31 | websae | and i am seeing no hits to it from my phone |
| 23:44.39 | websae | my phone registers with my other servers just fine |
| 23:44.53 | *** join/#asterisk atil (i=hugo@212-41-80-186.adsl.solnet.ch) |
| 23:44.57 | atil | hi, anybody can help me with chan_capi and NT-mode? |
| 23:45.46 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
| 23:46.03 | atil | I can see the D-channel traces in my syslog but chan_capi does not handle the call :( |
| 23:52.24 | *** join/#asterisk MoutaPT (i=MoutaPT@85.139.183.206) |
| 23:52.48 | robin_sz | meep? |
| 23:52.58 | robin_sz | so ... |
| 23:53.05 | MoutaPT | Hello, does any one has used AsterFAX with Ast@home? |
| 23:53.13 | robin_sz | the final (hah!) problem of this particular install ... |
| 23:53.56 | robin_sz | my SIP peer uses fromuser=<number> to set the outgoing CID ... |
| 23:54.19 | robin_sz | for most users on the ysytem they are happy to let the default (the main switchboard number) go through ... |
| 23:54.30 | robin_sz | a few would like to present their onw DID number |
| 23:54.46 | robin_sz | how to do that in the dial plan? huh? |
| 23:55.05 | robin_sz | im thinking get the peer, set the fromuser |
| 23:55.11 | generalhan | robin_sz: i wanted that same thing for my office. in sip.conf just put an entry in for those phones "callerid=<whatever their DID is>" |
| 23:55.14 | robin_sz | have the peer dial |
| 23:55.27 | robin_sz | ohh, right |
| 23:55.31 | generalhan | then when they make a call the CID will show up as that number |
| 23:55.43 | robin_sz | coo, that simple huh? |
| 23:55.46 | generalhan | that simple @! |
| 23:55.47 | afrosheen | yeah it's fun to play with sometimes |
| 23:55.51 | generalhan | yea |
| 23:56.07 | afrosheen | call your friends with wacky cid text and numbers |
| 23:56.09 | generalhan | my @home i set me CID as 666-666-6666 "MASTER" so that i can freak people out when i call them ! LOL |
| 23:56.17 | robin_sz | heh |
| 23:56.50 | afrosheen | it can be more useful than that, we used to have a landline at home, cancelled it but the credit cards I have are tied to that old phone number |
| 23:57.07 | afrosheen | when you get a new card you call the bank to activate and it reads the cid, so I set it to my old home phone # and it works |
| 23:57.33 | robin_sz | heh |
| 23:57.55 | afrosheen | also if you're calling someone and want them to call you back at a different number (your cellphone) before you call them you just set your CID to your cell |
| 23:57.58 | afrosheen | very handy |
| 23:58.04 | robin_sz | right |
| 23:58.13 | robin_sz | tres hand |
| 23:58.16 | robin_sz | y |
| 23:58.33 | generalhan | indeed |
| 23:59.43 | kippi | please can someone help me with iax |
| 23:59.55 | Winkie | what IAX help do you need? |
