| 00:18.55 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 00:18.58 | X-Rob_ | ~centosbug |
| 00:19.07 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 00:36.30 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:48.00 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 00:55.54 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 01:22.27 | docelmo | WHADUP! |
| 01:32.11 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:34.02 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 01:37.07 | Asterisk_Newbie | bye |
| 01:37.16 | *** part/#asterisk Asterisk_Newbie (n=a_ti_tu_@bl7-133-238.dsl.telepac.pt) |
| 01:55.38 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:57.12 | paolob | Guys, anyone knows how can I get asterisk sending a DTMF to an existing (i.e. answered) call? I connect to the pstn via sipura spa3000 ATA. Any help is appreciated. Thank you! |
| 01:57.40 | dongs | maybe try using inband dtmf and no compression between ata and asterisk box. |
| 01:58.12 | paolob | dongs, could you explain more simply, please? |
| 01:58.34 | dongs | i'm not sure how much simpler it could be |
| 01:59.10 | dongs | set dtmf for inband mode for the sipura peer config, and disallow all codecs except ulaw |
| 01:59.50 | paolob | dongs, the fact is that when asterisk executes the dial application, it doesn't keep executing the next priorities until the calls ends, so that placing a SendDTMF application after the dial one haven't any effect |
| 02:00.20 | dongs | oh THATs waht oyu mean. |
| 02:00.29 | dongs | i thought you meant like, pressing DTMF after answering a call. |
| 02:00.32 | dongs | like physically on the phone. |
| 02:00.45 | dongs | yea i duno what to do you about that. |
| 02:00.57 | paolob | and placing a D() option in the dial application sends the DTMF before connecting the caller with called |
| 02:01.03 | dongs | right. |
| 02:01.33 | paolob | but what is a D() option executed that way? I find in unuseful! |
| 02:01.56 | dongs | I find a lot of thigns in asterisk unuseful |
| 02:02.43 | paolob | dongs, so you don't know how can I get * sending the DTMFs tones to the called person? |
| 02:03.08 | dongs | i can think of a lamehack way of doing it with meetme, but otherwise, no |
| 02:03.40 | dongs | you could bridge caller + dtmf generating extension into a meetme and let it go, but that;s so dumb I cant believe i even thought of that. |
| 02:03.47 | paolob | Meetme doesn't work in my ubuntu installation, someone told me I should recompile * |
| 02:06.25 | paolob | dongs, But thinking in making the call in two moments? 1st moment: dial the called number - then wait some seconds so that the called person answers - and then sending the 2nd part (the DTMF). Isn't a way to make this in * |
| 02:06.26 | paolob | ? |
| 02:08.31 | paolob | That would be useful for companies with some menu before presenting a human, sending the DTMF with asterisk would permit to navigate automatically in the company's menus and present directly the human to the caller |
| 02:12.18 | dongs | paolob: no |
| 02:12.32 | paolob | dongs, :-( |
| 02:13.11 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:13.24 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 02:38.15 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 02:39.59 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 02:40.39 | paolob | [TK]D-Fender, do you know a way to get * send automatically dtmf tones to a called person after the connection is established? I connect to the pstn through a sipura spa-3000? For example in order to have * navigate automatically in the menu of a company... |
| 02:43.14 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 02:43.35 | dlynes_home | paolob: Use the D() option of Dial() |
| 02:44.33 | paolob | dlynes_home, unfortunately the D() option sends the dtmf tones _before_ (why?!?) the sipura connects me with the called person |
| 02:45.26 | paolob | dlynes_home, would it be a bug? |
| 02:48.02 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 02:48.45 | mbit | hey does anyone know a fix for this? |
| 02:48.45 | mbit | e know a fix for this |
| 02:48.45 | mbit | <mbit> Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK |
| 02:48.45 | mbit | <mbit> Timestamp: 00192ms |
| 02:49.40 | file | who says that is broken? |
| 02:51.37 | mbit | it basically rings then hangs up after 2 rings |
| 02:53.45 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 03:00.11 | *** join/#asterisk Qwell[\0] (n=chatzill@unaffiliated/qwell) |
| 03:00.46 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:01.50 | *** join/#asterisk reco (n=reco@user-0cdfan9.cable.mindspring.com) |
| 03:02.22 | reco | does anybody know a good how to for asterisk on debian for a total nebe like me? |
| 03:07.18 | Qwell[\0] | ~docs |
| 03:07.25 | jbot | [docs] 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 |
| 03:07.25 | Qwell[\0] | ~wikis |
| 03:07.27 | jbot | wikis is, like, http://www.voip-info.org |
| 03:07.32 | Qwell[\0] | bbl, shit blowing up |
| 03:08.45 | russellb | ... |
| 03:08.46 | russellb | gross |
| 03:11.05 | file | so russellb |
| 03:11.41 | russellb | so |
| 03:11.43 | russellb | hi |
| 03:11.49 | file | so hi |
| 03:12.12 | russellb | sup kid |
| 03:12.26 | file | Doctor Who!!! you? |
| 03:12.31 | russellb | yes? |
| 03:12.33 | russellb | what? |
| 03:12.48 | file | potato |
| 03:12.54 | russellb | cheese ballz |
| 03:12.57 | file | or better, Chee______ |
| 03:13.08 | russellb | B U R G E R !!!_--!@#@$!!!!!!!!! |
| 03:13.17 | file | omg omg omg |
| 03:13.24 | russellb | like |
| 03:13.26 | russellb | O |
| 03:13.26 | russellb | M |
| 03:13.27 | russellb | G |
| 03:14.09 | file | HAWT |
| 03:16.52 | russellb | so file |
| 03:16.57 | file | so russellb |
| 03:17.12 | russellb | i totally thought we were talking in #asterisk-dev |
| 03:17.18 | russellb | i'm usually not this silly in #asterisk |
| 03:17.30 | file | it's a slow evening |
| 03:17.39 | *** mode/#asterisk [+o file] by russellb |
| 03:17.50 | file | now I will take over the channel |
| 03:18.03 | file | but I'm lazy |
| 03:18.09 | russellb | i bet there are other people here ... lurking |
| 03:18.13 | russellb | ... lurkers ... |
| 03:18.31 | file | yesssss |
| 03:18.33 | reco | jbot: thanx |
| 03:18.33 | jbot | reco: my pleasure |
| 03:18.48 | reco | jbot: are you real? |
| 03:19.06 | russellb | if by real you mean a bot, then yes |
| 03:19.17 | reco | nice |
| 03:20.24 | reco | russellb: what is the recommended way to install asterisk on debian. use the packages inlcuded in the debian tree or build from source? |
| 03:20.47 | russellb | eh, it's up to you ... as long as the packages are up to date it's probably fine |
| 03:21.04 | russellb | but packages do include some patches not officially supported by the asterisk dev team |
| 03:21.22 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 03:21.31 | hads|home | Did somebody say lurkers? |
| 03:21.38 | reco | russellb: 1.0.7 |
| 03:21.44 | russellb | reco: that's ancient |
| 03:22.01 | russellb | reco: you should download and install the latest version of 1.2 |
| 03:22.03 | file | once upon a time it was the latest |
| 03:22.09 | russellb | or pull it from the 1.2 branch in svn |
| 03:22.17 | russellb | 1.0.7 was a pretty solid release ... |
| 03:22.29 | file | russellb: you're biased |
| 03:22.30 | russellb | but, i mean, even the 1.0 branch is up to 1.0.11 i think |
| 03:23.31 | russellb | so? |
| 03:23.34 | reco | russellb: i cee thanx |
| 03:23.43 | russellb | I <3 1.0 |
| 03:23.45 | russellb | it's sentimental |
| 03:23.56 | reco | :) |
| 03:24.18 | *** part/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 03:24.20 | file | I bet you printed out it's source and used it to cover your walls |
| 03:24.21 | russellb | i run trunk on my boxes, so don't listen to me :) |
| 03:24.31 | russellb | no, but that's a hot idea |
| 03:24.49 | russellb | we should print books of the asterisk source |
| 03:24.54 | *** part/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
| 03:24.58 | file | yesssss |
| 03:24.58 | reco | is sombody using here zimbra with asterisk? |
| 03:25.09 | file | ugh |
| 03:25.10 | russellb | we use zimbra |
| 03:25.15 | russellb | but ... not tied into asterisk |
| 03:25.16 | *** join/#asterisk AJmn (n=mycock@70.59.126.206) |
| 03:25.31 | russellb | file: lulu.com |
| 03:25.35 | russellb | file: my brother's wife works there |
| 03:25.49 | AJmn | hey guys i know alot factors into this... but how many calls should i be able to do on a dual 500mhz machine with 512Meg ram? using mix of G711 and g729? |
| 03:26.00 | file | russellb: o rly? |
| 03:26.01 | reco | i am pretty new i think of moving from stalker communigate to zimbra, do you like it? |
| 03:26.11 | file | reco: meh it's decent, I have my gripes |
| 03:26.19 | russellb | i have gripes as well ... |
| 03:26.29 | russellb | but it's cool overall |
| 03:26.31 | file | I think we all do |
| 03:26.40 | russellb | it's just little things |
| 03:28.32 | file | little big big little |
| 03:29.05 | russellb | file: quick, convert the asterisk source into a PDF |
| 03:29.13 | russellb | i want to know how much this will cost .... |
| 03:29.24 | file | a lot! |
| 03:29.30 | file | unless we outsource to China |
| 03:30.01 | russellb | file: for a 200 page 8.5 by 11 inch book, less than $10 !!!! |
| 03:30.15 | russellb | we are so doing this with 1.4.0 |
| 03:30.39 | file | :D |
| 03:31.34 | russellb | or ... SVN-trunk-r36959 |
| 03:31.36 | russellb | whatever |
| 03:32.08 | file | russellb: why are you not partying? |
| 03:32.15 | russellb | because i have no friends? |
| 03:32.19 | file | lame |
| 03:32.22 | file | I'll be your friend! |
| 03:32.56 | russellb | it's not much more costly to do hardback |
| 03:33.08 | russellb | or ... hardcover |
| 03:33.10 | russellb | whatever. |
| 03:36.46 | *** join/#asterisk bobbercheng (n=chatzill@218.242.131.66) |
| 03:37.27 | Corydon76-home | russellb: Nashville had the 3rd largest fireworks display in the world tonight |
| 03:37.38 | russellb | nice |
| 03:38.10 | Corydon76-home | Let that be a lesson to all you larger American cities: Nashville is kicking your ass |
| 03:39.05 | file | o rly |
| 03:39.22 | *** part/#asterisk AJmn (n=mycock@70.59.126.206) |
| 03:39.44 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 03:39.54 | Corydon76-home | file: they said something like 7000 fireworks |
| 03:40.29 | file | not enough! |
| 03:49.57 | *** join/#asterisk [1][Outcast] (n=outcast@219-89-206-239.adsl.xtra.co.nz) |
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| 03:52.17 | *** part/#asterisk P-NuT (n=nut@fw.office.unitedip.net.au) |
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| 03:55.39 | *** join/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 03:57.31 | *** join/#asterisk Paavum (i=Dorphals@pcsp168-254.supercabletv.net.co) |
| 03:58.05 | Paavum | Hello. I uninstalled mysql4 and installed some mysql5 binaries |
| 03:58.34 | Paavum | however when I want to use the mysql_cdr it says libmysqlclient.so.14 not found |
| 03:59.02 | Paavum | so I created a link named libmysqlclient.so.14 that pointed to libmysqlclient.so.15 |
| 03:59.05 | Paavum | (which I have= |
| 03:59.24 | Paavum | but I am still getting the funny message "libmysqlclient.so.14 not found" |
| 04:02.15 | chandi | Hi, I've got a problem between my sipura ATA and my asterisk box that are on the same network. Sometimes when I make a call or I receive a call from the sipura it drops the outgoing RTP. I know it's between the sipura and the asterisk because it never happens when I make calls from the asterisk box (inbound or outbound). Anybody has an idea ? |
| 04:03.19 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 04:04.33 | *** join/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
| 04:04.42 | seb- | how remove echo? |
| 04:05.13 | Corydon76-home | You can't remove echo |
| 04:05.16 | seb- | my Grandstream Handytone 286 analog-2-digital claims it can but doesn't say how |
| 04:05.40 | seb- | lemmie find exact quote |
| 04:05.42 | seb- | ... |
| 04:05.50 | Corydon76-home | Echo is a law of nature. |
| 04:06.08 | seb- | Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise |
| 04:06.08 | seb- | Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) |
| 04:06.24 | seb- | Corydon-w: G.168? |
| 04:06.28 | *** join/#asterisk RoyKa (n=roy@chello080109196173.3.graz.surfer.at) |
| 04:06.41 | seb- | Corydon76-home: G.168 i think |
| 04:06.54 | Corydon76-home | Note that the channel name is #asterisk not #grandstream |
| 04:07.38 | Corydon76-home | In other words, go ask the vendor |
| 04:07.54 | seb- | maybe G.168 is a general protocol to take care of echo effects? |
| 04:10.38 | chandi | nobody has an idea about my prob ? |
| 04:13.06 | *** part/#asterisk dec (n=tom@ppp206-151.lns1.adl2.internode.on.net) |
| 04:13.14 | Corydon76-home | Apparently not |
| 04:13.32 | file | Your call can not be completed as dialed. Please check the number and try your call again later. |
| 04:13.35 | *** join/#asterisk tipizo (n=tipizo@c-24-34-63-95.hsd1.ma.comcast.net) |
| 04:13.59 | chandi | ok ;) |
| 04:14.16 | tipizo | when I type this export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 04:14.17 | chandi | what's the number I should call then ? ;) |
| 04:14.25 | file | 411 |
| 04:14.34 | file | tipizo: we do not use CVS at all anymore |
| 04:14.42 | CunningPike | ~cvs |
| 04:14.50 | jbot | [cvs] concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn. |
| 04:14.52 | tipizo | How i get the latest head |
| 04:15.07 | CunningPike | ~svn |
| 04:15.09 | jbot | methinks subversion is version control software. see http://subversion.tigris.org/ it aims to be a better CVS than CVS. |
| 04:15.20 | CunningPike | svn.digium.com |
| 04:15.26 | Corydon76-home | svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk |
| 04:21.14 | tipizo | thanks |
| 04:23.40 | *** join/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 04:26.02 | nomego | how do I configure an entry in sip.conf for an ekiga softphone? |
| 04:26.27 | tipizo | suppose that my asterisk server is behind a firewall... And I would like to use my ATA to where ever I'm to connect to asterisk server |
| 04:26.32 | tipizo | how do do that? |
| 04:27.03 | Trionnis | port forwarding |
| 04:27.21 | Trionnis | sip and rdp ports to the * ip |
| 04:27.30 | tipizo | I try it it does not work? |
| 04:27.31 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 04:27.45 | Trionnis | you sure you have the right ports? |
| 04:27.49 | Trionnis | :) |
| 04:27.52 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:28.01 | Trionnis | and explain "does not work" please |
| 04:28.11 | Trionnis | doesn't connect, no audio, etc? |
| 04:29.12 | tipizo | I'm using an SPA2000 and try to point to my asterisk server from friend house.. I got no dial tone |
| 04:29.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:29.44 | Trionnis | and have you checked the logs? |
| 04:29.48 | Trionnis | to see what's happening? |
| 04:30.16 | tipizo | no i did not |
| 04:30.25 | Trionnis | might be a good place to start :) |
| 04:30.54 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:30.58 | tipizo | OK I will try it again l |
| 04:31.01 | tipizo | thanks |
| 04:31.11 | Trionnis | you should get some indication as to the issue |
| 04:31.36 | Trionnis | if not, turn on sip debug and be ready for a crapflood (if it's even connecting) |
| 04:33.22 | tipizo | great idea... I will try that |
| 04:33.45 | Trionnis | ok :) |
| 04:33.49 | tipizo | do i need a stun server! |
| 04:33.51 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 04:33.55 | Trionnis | hmm |
| 04:33.58 | Trionnis | you shouldn't |
| 04:34.12 | Trionnis | make sure you have nat enabled in sip.conf |
| 04:34.39 | Trionnis | I'm assuming "connecting from a friend's house" means the ATA is on the back side of another nat router? |
| 04:35.03 | tipizo | Yes, that is correct! |
| 04:35.15 | Trionnis | e.g. ATA -> his router -> internet -> your router -> asterisk |
| 04:35.23 | Trionnis | want an easy solution? |
| 04:35.27 | Trionnis | get an IAXy |
| 04:35.29 | Trionnis | :) |
| 04:35.59 | Trionnis | sip + nat stinks, imho |
| 04:36.18 | tipizo | ATA --> his router --> internet internet-->my router -->asterisk |
| 04:36.35 | Trionnis | yeah |
| 04:36.45 | Trionnis | that's a bad way to try to run sip |
| 04:36.55 | Trionnis | it can be done, but I personally wouldn't prefer it |
| 04:37.03 | tipizo | What Do i need to look for in sip.conf to make sure that nat is enabled? |
| 04:37.40 | Trionnis | under your channel config for that particular device, you'll want "nat=1" |
| 04:37.59 | Trionnis | also "host=dynamic" couldn't hurt |
| 04:38.12 | tipizo | what is IAXy |
| 04:38.30 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-116-157.dsl.chcgil.sbcglobal.net) |
| 04:38.36 | Trionnis | http://www.digium.com/en/products/hardware/s101i.php |
| 04:38.45 | Trionnis | uses IAX instead of SIP |
| 04:39.06 | Trionnis | IAX = much better NAT handling, and also has jitter compensation if you want/need it |
| 04:39.15 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 04:39.44 | BugKham | anyone knows how to get mpg123 compiled for 64-bit CPU? |
| 04:41.01 | Trionnis | can't say I do mate |
| 04:41.02 | Trionnis | sorry |
| 04:41.09 | tipizo | great...how would IAXy fit in my existing confiuration... |
| 04:41.14 | Trionnis | it's an ATA |
| 04:41.19 | Trionnis | you would replace the Sipura with it |
| 04:42.30 | tipizo | What if I turn my asterisk server into router/firewall using shorewall... Would that be a good solution? |
| 04:42.56 | Trionnis | well |
| 04:42.58 | Trionnis | it could be |
| 04:43.04 | Trionnis | depends a lot on hardware |
| 04:43.16 | Trionnis | it takes a bit to crosscode channels and such |
| 04:43.21 | nomego | I should be able to connect an FXO-gateway beside my regular phone until I get it all working, right? |
| 04:43.28 | Trionnis | but yes, that might make it a bit easier |
| 04:43.48 | Trionnis | well, you'd need 2 phones, or a 2 line phone |
| 04:43.49 | Trionnis | but yes |
| 04:43.54 | BugKham | anyone knows how to get mpg123 compiled for FC5 x86_64? |
| 04:44.13 | BugKham | or knows of anyother mp3 stream player for * |
| 04:44.24 | nomego | maybe mpg321 works |
| 04:44.38 | BugKham | nomego, u reckon? |
| 04:44.44 | tipizo | don't understand why would i need two lines |
| 04:44.50 | nomego | BugKham: try it |
| 04:44.55 | Trionnis | well, you said "regular phone" |
| 04:45.04 | Trionnis | I'm assuming you're referring to a POTS line? |
| 04:45.12 | tipizo | nope...sip phone |
| 04:45.15 | Trionnis | ah |
| 04:45.23 | tipizo | that's what i'm using now |
| 04:45.24 | Trionnis | yeah, you might have some trouble with routing though |
| 04:45.24 | nomego | I said regular phone ;) |
| 04:45.37 | Trionnis | doh |
| 04:45.40 | Trionnis | my bad |
| 04:45.41 | Trionnis | haha |
| 04:45.50 | Trionnis | too much going on at once :) |
| 04:46.14 | BugKham | nomego, it was last released on 2002 I don't think it will work |
| 04:46.52 | nomego | how about me.. would an FXO-gateway work as a regular phone? |
| 04:46.58 | tipizo | All I care to do right now is cary my ata with me on the road... have it connect to my asterisk server ... get a dial tone and make phone calls |
| 04:47.14 | Trionnis | then an IAXy will be the easiest no-hassle solution for you |
| 04:47.16 | BugKham | nomego, but will give it a try |
| 04:47.39 | nomego | BugKham: you do that |
| 04:47.58 | tipizo | kinda vonage like |
| 04:48.09 | Trionnis | well |
| 04:48.13 | Trionnis | something to consider |
| 04:48.19 | tipizo | please |
| 04:48.21 | Trionnis | Vonage has their gateways on the live net |
| 04:48.31 | *** part/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:48.34 | Trionnis | NAT messes with SIP something terrible |
| 04:48.58 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:49.03 | Trionnis | I'm guessing this is on a residential connection? |
| 04:49.23 | *** join/#asterisk RalphieII (n=ralph@c-67-162-230-243.hsd1.tx.comcast.net) |
| 04:49.39 | RalphieII | anyone in tonight? for real? |
| 04:49.40 | tipizo | forget about nat... Let's connect my asterisk server directly to the net using shorewall to turn it into a router/firewall |
| 04:49.53 | Trionnis | then you should be ok so long as you open the proper ports |
| 04:50.06 | Trionnis | and add those 2 lines to sip.conf as I mentioned |
| 04:50.17 | file | also setup sip.conf, externip/externhost and localnet with your external IP information and local network information |
| 04:50.20 | tipizo | that would 5060:5082 and 10000:20000 |
| 04:50.28 | tipizo | for sip |
| 04:50.35 | RalphieII | My install of trixbox 1.1 fails, anyone have info on this? |
| 04:50.37 | Trionnis | whatever you're using |
| 04:50.51 | Trionnis | -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx' |
| 04:51.02 | Trionnis | er |
| 04:51.04 | Trionnis | points :) |
| 04:51.29 | RalphieII | ahh |
| 04:51.48 | RalphieII | anyone in tonight? for real?( |
| 04:51.58 | RalphieII | :( |
| 04:52.04 | tipizo | chief thanks so very much!!! |
| 04:53.01 | Trionnis | certainly |
| 04:55.01 | tipizo | Would you recommand using asterisk head or stable? |
| 04:55.10 | Trionnis | stable |
| 04:55.16 | luke-jr_ | depends on what you want to do |
| 04:55.23 | russellb | trunk! |
| 04:55.23 | Trionnis | true |
| 04:55.31 | russellb | jk ... |
| 04:55.32 | Trionnis | if it's basic sip, stable should be just fine for you |
| 04:55.33 | luke-jr_ | for some things, trunk is more stable than 1.2 |
| 04:55.35 | tipizo | so stable will be 1.2.9.1 |
| 04:56.10 | luke-jr_ | Trionnis: if he wants to use ugly extensions.conf stuff, maybe =p |
| 04:56.25 | luke-jr_ | russellb: hurry up with 1.4 =p |
| 04:56.32 | russellb | don't look at me ... |
| 04:56.45 | file | it's all my fault |
| 04:56.53 | Trionnis | yes, we know |
| 04:56.58 | Trionnis | ;) |
| 04:57.31 | luke-jr_ | file: is it? |
| 04:57.33 | tipizo | how do I get the stable using svn |
| 04:57.50 | luke-jr_ | tipizo: rtfw? |
| 04:57.55 | Trionnis | haha |
| 04:57.56 | Trionnis | ouch |
| 04:58.50 | tipizo | It's better to ask when you don't know.... |
| 04:59.06 | *** join/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 04:59.15 | *** part/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 04:59.23 | tipizo | you showed a while ago to do this svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk |
| 04:59.39 | tipizo | I'm assuming this is head not the stable version |
| 05:00.10 | file | there's no "stable" version, there is a 1.2 release branch though |
| 05:00.12 | file | where only bug fixes go |
| 05:00.19 | file | http://svn.digium.com/svn/asterisk/branches/1.2 |
| 05:00.37 | file | so therefore... svn co http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 05:00.41 | luke-jr_ | tipizo: HEAD refers to the latest code for any version-- past, stable, or trunk |
| 05:01.00 | luke-jr_ | file: and even some bugfixes aren't allowed :( |
| 05:01.37 | tipizo | you guys are fabulous... thanks for the education |
| 05:01.39 | file | I sense a bug tracker incident |
| 05:02.08 | luke-jr_ | file: well, nobody cares to fix AEL, deeming AEL 2 the fix-- but there's no release with AEL 2 |
| 05:02.55 | luke-jr_ | that's one of my reasons to be anxious over 1.4 =p |
| 05:02.59 | file | uh huh |
| 05:03.05 | tipizo | But I saw on asterisk.org version 1.2.9.1 is that head or stable? |
| 05:03.11 | file | 1.2.9.1 is a release |
| 05:03.21 | luke-jr_ | tipizo: HEAD of 1.2 at the time of the release |
| 05:03.23 | file | note I'm not using the term stable |
| 05:03.58 | luke-jr_ | tipizo: there's HEAD of 1.2 and HEAD of trunk. two heads. |
| 05:04.10 | Trionnis | yeah, you want stable, get a cisco |
| 05:04.19 | Trionnis | (yes, I'm kidding) |
| 05:04.23 | luke-jr_ | =p |
| 05:04.32 | hads|home | Argh! This two headed monster will kill us all! |
| 05:04.34 | tipizo | Kinda confuse a bit... 1.2 and 1.2.9.1 |
| 05:04.45 | file | Trionnis: hehe |
| 05:04.47 | luke-jr_ | tipizo: 1.2 is the branch |
| 05:04.50 | Trionnis | ^^ |
| 05:04.54 | luke-jr_ | tipizo: 1.2.9.1 is the release/tag |
| 05:05.25 | tipizo | I don't want to be ban so I will not ask anymore question... |
| 05:05.40 | file | 1.2.9.1 is a snapshot of the 1.2 branch at a specific time when 1.2.9.1 was released, it won't get updated - 1.2 will get updated and another snapshot will occur to create the next release |
| 05:05.50 | luke-jr_ | tipizo: FWIW, I think Trionnis was scared of himself being banned for a bad joke, not expecting you to get banned |
| 05:05.52 | file | you won't get banned... it takes a lot to get banned |
| 05:05.58 | Trionnis | yeah |
| 05:06.01 | Trionnis | just look at me |
| 05:06.10 | Trionnis | I'm the channel clown, and they haven't booted me yet ;) |
| 05:06.24 | luke-jr_ | they should, as a joke |
| 05:06.30 | Trionnis | I'm suprised they haven't |
| 05:06.32 | Trionnis | :) |
| 05:08.00 | luke-jr_ | file: so when can I expect 1.4? =p |
| 05:08.09 | luke-jr_ | at least an alpha or beta or something? |
| 05:09.07 | file | O wpm |
| 05:09.09 | file | gah |
| 05:09.11 | file | I won't answer that |
| 05:09.30 | Trionnis | second Tuesday of next week |
| 05:09.37 | Trionnis | not a day earlier! ;p |
| 05:09.40 | file | there's only so many and so much time |
| 05:09.43 | file | er so many people |
| 05:10.22 | luke-jr_ | file: aww :( |
| 05:10.43 | luke-jr_ | not even a 1.3 or something? =p |
| 05:10.46 | file | Trionnis: I thought there's only one Tuesday in a week :P |
| 05:10.55 | Trionnis | exactly my point |
| 05:10.57 | Trionnis | :) |
| 05:11.05 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 05:11.08 | file | next week... is possible... |
| 05:11.11 | file | we'll see |
| 05:11.16 | Trionnis | o_O |
| 05:11.20 | Trionnis | O RLY? |
| 05:11.21 | luke-jr_ | really? o.o |
| 05:11.49 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 05:12.37 | file | actually, I need to write myself a note now |
| 05:12.40 | file | you've given me an idea |
| 05:12.44 | luke-jr_ | ? |
| 05:12.45 | Trionnis | I did? |
| 05:13.10 | file | actually I have to write about 3 notes as some other stuff just flooded back into my thoughts |
| 05:13.17 | Trionnis | guess I need to leave the channel now... I can't be *productive* fer god's sake |
| 05:14.15 | file | I hope my replacement hard drive arrives ... ::looks at clock:: today ... I want to get my workstation back to working before I leave |
| 05:14.57 | file | silly moving parts |
| 05:15.11 | luke-jr_ | so it has 47 minutes + [0..23] hours to get here... =p |
| 05:15.21 | file | it's 2:15AM here :D |
| 05:15.48 | luke-jr_ | ... |
| 05:15.52 | luke-jr_ | what timezone is that anyhow? |
| 05:15.58 | Trionnis | est I'm guessing |
| 05:16.00 | luke-jr_ | middle of the ocean? |
| 05:16.06 | luke-jr_ | no, Eastern is 1:15 AM |
| 05:16.10 | Trionnis | hm |
| 05:16.26 | Trionnis | atlantic standard time? |
| 05:16.32 | file | Trionnis: bingo |
| 05:16.33 | Trionnis | maybe he lives on a boat? |
| 05:16.33 | luke-jr_ | doesn't the ocean fill the next timezone? |
| 05:16.38 | Trionnis | :> |
| 05:16.52 | luke-jr_ | file: you use satellite or smth? |
| 05:16.57 | luke-jr_ | from a boat? |
| 05:16.58 | luke-jr_ | O.o |
| 05:17.00 | file | never! |
| 05:17.07 | luke-jr_ | o.O |
| 05:17.15 | luke-jr_ | file: hey, so what are the ideas/notes? =p |
| 05:17.49 | file | uh one I can't say |
| 05:17.56 | luke-jr_ | o.o |
| 05:17.59 | file | one is whether I want to pack my Polycom phone and take it with me |
| 05:18.10 | file | and the other is... darn, I forgot it |
| 05:18.21 | luke-jr_ | what's the one you can't say? |
| 05:18.26 | luke-jr_ | before you forget it.... |
| 05:18.30 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:18.54 | file | nope! |
| 05:19.05 | luke-jr_ | aww, can't type it either? =p |
| 05:19.38 | file | my brain is preventing it! |
| 05:19.42 | luke-jr_ | or you have some fancy voice recognition IRC? =p |
| 05:19.49 | luke-jr_ | oy |
| 05:19.50 | luke-jr_ | oh* |
| 05:19.54 | luke-jr_ | we can fix that then |
| 05:20.25 | file | I really don't remember this second thing and it's really bugging me |
| 05:20.25 | file | maybe it's the second one |
| 05:20.32 | luke-jr_ | hm |
| 05:20.46 | luke-jr_ | is it to offer me a job? |
| 05:21.00 | file | I am *so* not the person for that |
| 05:21.03 | luke-jr_ | lol |
| 05:21.26 | *** join/#asterisk tlow (n=tlowe@omfg.wtf.no) |
| 05:21.41 | luke-jr_ | fireworks? |
| 05:22.10 | file | pfft |
| 05:23.13 | luke-jr_ | download a movie? |
| 05:23.22 | file | it was something to do with packing |
| 05:23.31 | luke-jr_ | oh, you moving? |
| 05:23.42 | file | no, going elsewhere for a week |
| 05:23.52 | luke-jr_ | fed up with that weird timezone, eh? |
| 05:23.55 | file | I remembered it when I went to Subway and told myself to write a note when I got back but forgot |
| 05:24.24 | luke-jr_ | maybe you need to check the weight of a suitcase or such |
| 05:24.49 | luke-jr_ | or pack your toothbrush |
| 05:24.50 | file | oh, need to remember to pack an extra set of clothes in my backpack |
| 05:25.03 | file | I have horrible luck with luggage |
| 05:25.03 | luke-jr_ | is that it? or something else? |
| 05:25.39 | file | nope, that's it! |
| 05:25.44 | luke-jr_ | great |
| 05:25.52 | luke-jr_ | so what was that first note again? =p |
| 05:25.55 | file | I still won't tell you the other thing though |
| 05:26.00 | luke-jr_ | aww ;) |
| 05:26.50 | Trionnis | I know |
| 05:27.02 | Trionnis | he's going to code a free porn auto-downloader into 1.4 |
| 05:27.06 | luke-jr_ | Trionnis: you ask, you gave him the idea |
| 05:27.10 | Trionnis | I did? |
| 05:27.15 | Trionnis | that doesn't happen |
| 05:27.19 | Trionnis | I don't have good ideas |
| 05:27.27 | Trionnis | just smartass wisecracks |
| 05:27.29 | Trionnis | :> |
| 05:27.35 | luke-jr_ | * Trionnis salivates with anticipation |
| 05:27.35 | luke-jr_ | <-- mog (i=ejabberd@68.62.237.103) has left #asterisk |
| 05:27.35 | luke-jr_ | <file> actually, I need to write myself a note now |
| 05:27.35 | luke-jr_ | <file> you've given me an idea |
| 05:28.05 | file | it's crazy as a canary! |
| 05:28.14 | luke-jr_ | so am I! |
| 05:28.14 | luke-jr_ | maybe |
| 05:28.23 | luke-jr_ | I don't know how crazy canaries are tho |
| 05:28.28 | luke-jr_ | but I am tired! |
| 05:28.36 | luke-jr_ | and probably don't make sense |
| 05:28.36 | file | keep it gay! |
| 05:28.41 | luke-jr_ | :o |
| 05:30.12 | luke-jr_ | if you delete a time zone, does that make every hour 2 and a half minutes longer? |
| 05:30.34 | file | maybe |
| 05:30.54 | luke-jr_ | so let's delete 8 time zones |
| 05:31.13 | luke-jr_ | to make a nice even 0x10 hour day |
| 05:31.31 | luke-jr_ | and each hour can be 150% as long |
| 05:31.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 05:32.20 | luke-jr_ | ok |
| 05:32.23 | luke-jr_ | I really should go to bed |
| 05:32.24 | luke-jr_ | goodnight |
| 05:32.33 | luke-jr_ | btw |
| 05:32.37 | luke-jr_ | the current hexadecimal time is |
| 05:32.42 | luke-jr_ | .3:ad |
| 05:32.48 | luke-jr_ | ttyl :) |
| 05:34.32 | Trionnis | hmm |
| 05:34.33 | Trionnis | bed |
| 05:34.36 | Trionnis | sounds like a good idea |
| 05:34.37 | Trionnis | night all |
| 05:34.42 | *** part/#asterisk Trionnis (i=lordkuri@12.206.2.116) |
| 05:35.36 | *** part/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 05:37.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:38.44 | *** join/#asterisk MISGroup (n=sales@vt-williston-cuda1c4-226.sbtnvt.adelphia.net) |
| 05:39.33 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 05:43.43 | MISGroup | Has anyone not been able to delete an extension from A@H? I created a receptionist at extension 0 and now I am unable to go into the config page to modify it. Can't delete it either....ideas? |
| 05:44.07 | drray | try the asterisk at home irc channel? |
| 05:44.40 | MISGroup | tnx |
| 05:46.05 | *** join/#asterisk MISGroup (n=sales@vt-williston-cuda1c4-226.sbtnvt.adelphia.net) |
| 05:47.57 | docelmo | ACK! A@H What a SUCK ASS application.. Geesh buy the book and learn the RIGHT way to configure asterisk.. This canned shit is for the birds |
| 05:51.26 | drray | don't be coy |
| 05:54.54 | MISGroup | Hmmmm well I was given two weeks to implement a phone solution for a non-proffit org. I was able to get them set up with a A@H system for 15 extensions. Give me another month and I might understand your narrow minded comment. |
| 05:55.30 | MISGroup | I had not dealt with * yet and this was a quick intro..... |
| 05:56.19 | russellb | i really don't like it when people so harshly put down the efforts of others that are just trying to make asterisk easier to use |
| 05:56.33 | russellb | any efforts like that that are released as *free* should be applauded. |
| 05:56.43 | drray | I agree |
| 05:57.04 | drray | I referred him to the a@h group because I don't have the first clue how to help him |
| 05:57.04 | MISGroup | *nod* |
| 05:57.14 | drray | other than making him use asterisk |
| 05:57.16 | drray | :) |
| 05:57.43 | russellb | yeah, well generally it's not supported in this channel |
| 05:57.49 | russellb | because the help for that is different |
| 05:57.57 | russellb | *not* because there is a hostile environment towards it |
| 05:58.00 | drray | I do think that asterisk@home is a solution in search of a problem |
| 05:58.02 | russellb | which many people here make it out to be |
| 05:58.57 | russellb | anyway, end rant :) |
| 05:59.14 | docelmo | A@H Has its on place and that is for the novice that wants an easy unflexable system. I have tried it and found it easier to compile from source |
| 06:00.36 | MISGroup | so clairify this a bit.... If I am using the CLI of an A@H box would you consider that Asterisk? |
| 06:02.19 | russellb | probably not, no :) |
| 06:02.52 | russellb | problems with freepbx generated configuration would probably best be addressed by those familiar with it. |
| 06:03.11 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:03.55 | MISGroup | understood. |
| 06:04.56 | docelmo | If you type asterisk -r your using asterisk in a round about way. Your still controlled by what am lets it do |
| 06:05.13 | docelmo | I have tried some custom things with amp and it just drove me nuts to get it to work |
| 06:06.38 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 06:19.39 | seb- | say if i hear my voice twice when i talk...is that what they mean by 'echo' ? |
| 06:19.51 | seb- | is it common and is there anything i can do about it? |
| 06:24.56 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 06:25.01 | Assid | heya |
| 06:25.50 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
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| 06:49.32 | docelmo | seb what hardware are you using? |
| 06:50.19 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 06:54.19 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 06:54.57 | jhiver | hi all |
| 06:55.30 | jhiver | 1) do g729 codecs work for FreeBSD |
| 06:55.50 | jhiver | 2) What is a good store with fast response to buy g729 licenses |
| 06:55.52 | jhiver | ? |
| 06:55.56 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:56.15 | *** part/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 07:03.10 | docelmo | jhiver, to my knowledge BSD isnt supported for g729 via digium |
| 07:03.39 | docelmo | You can however check digium's website, or their FTP server where the codecs are listed. |
| 07:04.16 | acehunky | docelmo i recently saw g729 compiled codec on digium ftp site |
| 07:04.24 | acehunky | for freebsd |
| 07:11.26 | jhiver | cool |
| 07:11.27 | jhiver | thanks |
| 07:14.59 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 07:15.09 | dlynes_home | docelmo: there's g729 compiled for freebsd on digium's ftp site as well as a register utility for freebsd....they're just not officially supported |
| 07:15.57 | dlynes_home | jhiver: just go to digium's web site to buy your g729 licenses; then use the g729 codec and register utility for freebsd |
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| 07:18.38 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:19.14 | jhiver | mhhh let's hope I have linux binary compat enabled on the freebsd box :) |
| 07:20.50 | jhiver | hey what should I grab for opteron processor: i368, i586, i686 ? |
| 07:22.48 | Assid | hey docelmo! |
| 07:22.50 | Assid | wassup man |
| 07:23.46 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 07:25.30 | Assid | jhiver: i think i686 |
| 07:25.36 | Assid | unless they have AMD specific |
| 07:26.42 | jhiver | ./register: Exec format error. Binary file not executable. |
| 07:26.47 | jhiver | f*ck |
| 07:26.51 | BugKham | hi, anyone using * on FC5 x86_64? |
| 07:26.57 | Qwell | jhiver: chmod? |
| 07:27.03 | jhiver | is there no bsd compatible 'register' utility ? |
| 07:27.08 | jhiver | yeah it's been chmoded |
| 07:27.29 | BugKham | hi, wanna know what you use for playing mp3 files |
| 07:27.30 | jhiver | ok found it |
| 07:27.46 | BugKham | dlynes_home, what OS are you using for *? |
| 07:28.10 | jhiver | ./register G729-273393D3 |
| 07:28.11 | jhiver | ELF interpreter /libexec/ld-elf32.so.1 not found |
| 07:28.11 | jhiver | Abort |
| 07:28.23 | jhiver | aaargh |
| 07:29.40 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:32.20 | dlynes_home | BugKham: linux |
| 07:33.10 | dlynes_home | jhiver: are you using the freebsd register utility, or the linux register utility? |
| 07:34.13 | dlynes_home | jhiver: and why do you need linux binary compatibility enabled on the bsd box? |
| 07:34.24 | dlynes_home | jhiver: i never had it enabled on mine |
| 07:36.11 | jhiver | ok |
| 07:36.18 | jhiver | I've tried both actually |
| 07:36.25 | jhiver | so i will try first the BSD one |
| 07:36.51 | jhiver | it's for freebsd 5.2, I use 6.0, but hell |
| 07:36.58 | dlynes_home | ah |
| 07:37.01 | dlynes_home | I've only used 6.0 |
| 07:37.11 | jhiver | I get this |
| 07:37.14 | jhiver | e82-103-133-162e# ./register |
| 07:37.14 | jhiver | ELF interpreter /libexec/ld-elf32.so.1 not found |
| 07:37.14 | jhiver | Abort |
| 07:37.19 | dlynes_home | I couldn't tell you if it works on 5.2 or not |
| 07:37.28 | jhiver | no I use 6.0 too |
| 07:37.32 | jhiver | I use this toos: |
| 07:37.33 | LoneShadow | anyone using ubuntu or debian here ? |
| 07:37.34 | jhiver | tool: |
| 07:37.36 | dlynes_home | Yeah...do you not have ELF support installed? |
| 07:37.41 | jhiver | ftp://ftp.digium.com/pub/asterisk/g729/unsupported/freebsd-5.2.1/ |
| 07:37.50 | jhiver | not sure, I'm not that good with freebsd :) |
| 07:37.55 | dlynes_home | one sec...i'll tell you what i grabbed |
| 07:38.23 | jhiver | dlynes_home, how do you enable that? is there a port or maybe something in /usr/src/ ? |
| 07:39.55 | dlynes_home | ftp://ftp.digium.com/pub/asterisk/g729/unsupported/freebsd-5.2.1/register |
| 07:40.01 | dlynes_home | is that what you used? |
| 07:40.21 | dlynes_home | I have no idea how to enable elf...one sec |
| 07:42.27 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:43.08 | jhiver | yeah I used that |
| 07:43.28 | jhiver | maybe it's because I have a 64 bit (AMD Opteron) architecture? |
| 07:44.01 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:49.14 | *** part/#asterisk reco (n=reco@user-0cdfan9.cable.mindspring.com) |
| 07:52.22 | dlynes_home | jhiver: and you're running 64-bit freebsd? |
| 07:53.43 | jhiver | I'm not sure, that was installed for me, I guess I can dmesg to check |
| 07:53.54 | dlynes_home | yeah...i suspect your'e running 64-bit freebsd |
| 07:54.03 | dlynes_home | and you forgot to install the 32-bit compatibility libraries |
| 07:54.28 | jhiver | right, as I said that wasn't set up by me, so... |
| 07:54.38 | jhiver | think i can fix it with /sbin/sysinstall? |
| 07:55.01 | dlynes_home | i have no idea |
| 07:55.12 | dlynes_home | i'm almost completely useless when it comes to freebsd |
| 07:55.18 | jhiver | ok :) |
| 07:55.23 | jhiver | so am I |
| 07:55.28 | dlynes_home | I was only running it for a while because we had someone in house that knew it |
| 07:55.31 | jhiver | although I find it very stable for asterisk |
| 07:55.37 | dlynes_home | but I got tired of having to rely on him for everything |
| 07:55.45 | dlynes_home | and the lack of compatibility with asterisk |
| 07:56.07 | skrusty | morning |
| 07:56.08 | dlynes_home | so i said forget it, and made slackware our standard |
| 07:56.22 | jhiver | :) |
| 08:00.02 | linlin | I cant seem to build the asterisk-addons components, I get errors when attempting to "make": http://pastebin.ca/79365 |
| 08:00.32 | mitcheloc | linlin, can't you use odbc support in asterisk? |
| 08:01.05 | linlin | i'm sorry? |
| 08:01.13 | linlin | i dont understand what you mean |
| 08:01.21 | mitcheloc | you should also keep your support requests limited to one channel, not #asterisk & #freepbx |
| 08:01.49 | mitcheloc | is the reason you want asterisk-addons for the mysql logging? |
| 08:03.04 | linlin | well, im not sure why I want it, im following the FreePBX "INSTALL" file |
| 08:04.14 | mitcheloc | okay well keep the questions in #freepbx :) no offense, but they will know more |
| 08:06.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:08.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:15.38 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:18.57 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 08:19.03 | FreezeS | hello |
| 08:19.22 | FreezeS | I've got a problem: MoH in a queue stops after 2 loops |
| 08:19.39 | FreezeS | anyone else had this problem ? |
| 08:20.38 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 08:21.57 | RoyKa | hi |
| 08:24.56 | dlynes_home | hola |
| 08:25.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:26.49 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:32.38 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 08:34.58 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 08:35.40 | AltnTab | how to make _0. and _088XXXXXXX work together ? |
| 08:36.33 | AltnTab | without putting them in different contexts, they have to be viewable by all users |
| 08:36.55 | dlynes_home | AltnTab: and how is putting them into different contexts going to prevent that? |
| 08:37.41 | AltnTab | dlynes_home, i think by specifing wich user on which context uses |
| 08:38.13 | dlynes_home | AltnTab: you can define contexts and include those contexts in other contexts though |
| 08:39.02 | dlynes_home | AltnTab: [operator] exten => _0.,1,blahblah [ld] exten => _088XXXXXXX,1,blahblah [mycontext] include => operator ; include => ld |
| 08:39.05 | AltnTab | dlynes_home, yes, i know ... but how can i use those two work together |
| 08:39.27 | AltnTab | dlynes_home, is this going to solve the problem ? |
| 08:39.32 | dlynes_home | the above example will make the operator context take precedence over the ld context |
| 08:39.49 | AltnTab | because everything goes thru _0. |
| 08:39.58 | dlynes_home | if you include [mycontext] in your user's context, or assign the mycontext to your user |
| 08:40.28 | dlynes_home | Yes, but becuase you're using the include => statements, it forces asterisk to include the extensions in the specific order that you've specified |
| 08:41.25 | RoyK[at] | methinks the ordering logic should be rewritten...... |
| 08:41.37 | dlynes_home | So, you can either include operator first, or include ld first, depending on which one you want to take precedence |
| 08:41.47 | RoyK[at] | splitting it up into several contexts just makes a mess |
| 08:41.55 | *** join/#asterisk tparcina (n=tparcina@lns01-1072.dsl.iskon.hr) |
| 08:42.03 | tparcina | good day channel |
| 08:42.10 | dlynes_home | RoyK[at]: you mean the ordering logic within the asterisk code? |
| 08:42.14 | AltnTab | dlynes_home, no, no sorry for messing thigs up |
| 08:42.18 | tparcina | hi dlynes! |
| 08:42.21 | AltnTab | they are now in one context |
| 08:42.25 | dlynes_home | heya tparcina |
| 08:43.24 | dlynes_home | RoyK[at]: nobody's stopping you from fixing the code |
| 08:43.27 | RoyK[at] | dlynes_home: yes |
| 08:43.29 | RoyK[at] | hehe |
| 08:43.33 | RoyK[at] | no, i know |
| 08:43.38 | tparcina | does anybody use voipbuster? |
| 08:44.03 | AltnTab | dlynes_home, so i have to put this ext which i prefer in default context and include the other which is with lower priority in other context ? |
| 08:44.04 | tparcina | i read on their web pages that you can make free sip calls |
| 08:44.10 | AltnTab | tparcina, i use them |
| 08:44.25 | tparcina | AltnTab: you use them with asterisk? |
| 08:44.26 | AltnTab | i have several accounts |
| 08:44.30 | AltnTab | yes |
| 08:44.36 | AltnTab | tparcina, |
| 08:44.58 | tparcina | AltnTab: are cals realy free? |
| 08:45.13 | AltnTab | to some destinations |
| 08:45.28 | AltnTab | the register thing is free and you have 1 minute free call |
| 08:45.50 | tparcina | AltnTab: yes, to somthing more than 30 destioations... it's cool! |
| 08:45.52 | AltnTab | but other prices are much cheaper than other |
| 08:46.13 | AltnTab | i mean than other VoIP operators |
| 08:46.25 | AltnTab | the quality is good |
| 08:46.25 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 08:46.28 | AltnTab | i use g726 |
| 08:46.53 | tparcina | AltnTab: so, i need to register (do i have to pay anything?) and i can call for 1 min for free. if call last longer than 1 min they will charge me? is that the way they work? |
| 08:47.30 | AltnTab | tparcina, no afrer a minute the call disconnects |
| 08:47.41 | FreezeS | does anyone know why MoH is stopped in a queue after 2 loops ? |
| 08:47.50 | AltnTab | if the charge you they have no way of taking your money or finding you |
| 08:47.56 | AltnTab | the method is prepaid |
| 08:49.37 | nounoursfre | AltnTab : You have implant the codec G726 in your asterisk ? |
| 08:49.43 | AltnTab | tparcina, the deal is of you want to talk with free destinations for more than a minute pay us 10 bucks |
| 08:50.48 | AltnTab | nounoursfre, i was using G.711 at first but i've used it with gsm gateway and there was no compatible codecs |
| 08:51.04 | AltnTab | nounoursfre, and i moved to the next free one, g.726 |
| 08:51.50 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 08:51.50 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 08:52.06 | tparcina | AltnTab: how do you make phonecalls that last more than 1 min? |
| 08:52.07 | nextime | hi, i have a problem with a random crash with latest 1.2 svn branch when i get more that 15/20 calls on a zaptel PRI ( Wildcard TE405P ). any hint on it? |
| 08:52.54 | AltnTab | tparcina, i pay 10 bucks which i have to spend on calling within 3 months |
| 08:53.01 | AltnTab | after that my money are gone |
| 08:53.33 | AltnTab | tparcina, i am not shure but 10 is the minium price |
| 08:55.27 | tparcina | AltnTab: do you know where are their servers located? are they all in USA or they have some in the rest of the world? |
| 08:57.27 | AltnTab | tparcina, i think they are somewhere in Europe, i am not shure |
| 08:57.57 | tparcina | AltnTab: can you please send me your sip.conf and extensions.conf that are related with voipbuster? |
| 09:01.19 | AltnTab | tparcina, it is very basic as connecting to any other SIP provider, first use register => user:pass@name than define [name]username,secret,codec,host=sip1.voipbuster.com |
| 09:02.42 | AltnTab | in sip.conf, all destinations have to be dialed with the soecific country prefix ex: america 001...../// |
| 09:03.08 | AltnTab | dlynes_home, i worked, tnx :) |
| 09:06.34 | *** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
| 09:07.28 | telenieko | Hi! When I originate a call from a call file (/var/spool/asterisk/outgoing) how can I set the CallerID of the call? I mean, my phone rings, I pickup, then when asterisk calls the other party there's no callerID. thx :) |
| 09:09.20 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 09:10.15 | tparcina | <AltnTab> tparcina, it is very basic as connecting to any other SIP provider, first use register => user:pass@name than define [name]username,secret,codec,host=sip1.voipbuster.com |
| 09:10.15 | tparcina | * RoyK[at] has quit IRC (Read error: 104 (Connection reset by peer)) |
| 09:10.15 | tparcina | <AltnTab> in sip.conf, all destinations have to be dialed with the soecific country prefix ex: america 001...../// |
| 09:10.40 | tparcina | AltnTab: thank you, i'll register now and try their services. |
| 09:14.27 | tparcina | AltnTab: final question, is it posible to register without downloading and installing their software? |
| 09:14.55 | hads|home | tparcina: All this info is on their site. |
| 09:23.14 | tparcina | hads|home: thank you, i have look on their site and i couldn't find that is possible to register without downloading their software. so i have asked here. it wouldno't be the first time that is possible something that they haven't put on their web pages... |
| 09:25.20 | x86 | how would i represent this UK DID in E164 format: 08458681772 |
| 09:28.40 | RoyK[at] | <PROTECTED> |
| 09:28.46 | RoyK[at] | tparcina: que? |
| 09:28.51 | *** join/#asterisk Kizmet (n=kizmet@00-nsi-ad-act.au.argon.net.au) |
| 09:29.17 | Kizmet | file, can i download the whole svn tree now :P |
| 09:30.42 | tparcina | RoyK: que? - sorry, i don't speake zulu language :)) |
| 09:32.06 | *** join/#asterisk otaku42 (n=otaku42@madwifi/developer/otaku42) |
| 09:32.09 | otaku42 | moin all |
| 09:32.15 | jhiver | right... now on my hosted server I can run digium's register tool (for g729) |
| 09:32.19 | jhiver | However I get: |
| 09:32.22 | jhiver | Unable to determine hostid. You must have at least one ethernet card |
| 09:32.27 | jhiver | and it doesn't reg... |
| 09:32.32 | jhiver | I use FreeBSD 6.0 |
| 09:33.51 | jhiver | any ideas? |
| 09:33.53 | otaku42 | question: when asterisk registers for a SIP account for, for example, 1800 seconds, it will schedule to re-register in 1785 seconds. is there a way to modify this 15 seconds difference with an option in the configuration file? |
| 09:34.30 | Kizmet | otaku42, does it really make a difference ? |
| 09:34.50 | otaku42 | Kizmet: in my case yes. it's a very special situation, but it definitely makes a difference. |
| 09:34.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:35.17 | Kizmet | hmm im sure you can set it sumwhere |
| 09:35.35 | RoyK[at] | jhiver: it probably searches for eth0, which doesn't exist on fbsd iirc |
| 09:35.47 | RoyK[at] | http://karlsbakk.net/fun/crazy_japanese_sign.jpg |
| 09:36.03 | otaku42 | Kizmet: astiersk is running in an vmware server which has a huge rtc lag. this needs to be fixed otherwise, but for now working around this by using a larger "reregsiter offset" would help. |
| 09:41.26 | Kizmet | otaku42, *cough* Xen :P |
| 09:42.41 | otaku42 | Kizmet: yes, but unfortunately i can't easily change the solution that is already in use here :( |
| 09:42.53 | Kizmet | :( |
| 09:56.27 | tparcina | can anybody tell me one free calling number in USA? - i would like to try voipbuster but i don't have anybody to call... |
| 10:10.52 | *** join/#asterisk Zerthimon (n=Zerthimo@80.74.110.153) |
| 10:11.34 | Zerthimon | greetings |
| 10:11.58 | Zerthimon | !help |
| 10:13.28 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 10:13.42 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 10:16.40 | *** part/#asterisk otaku42 (n=otaku42@madwifi/developer/otaku42) |
| 10:20.28 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 10:20.53 | *** join/#asterisk E-bola (i=bola@a80-186-38-125.elisa-laajakaista.fi) |
| 10:21.18 | E-bola | CAn anybody help me figure out how to connect my asterix server to the outside world? |
| 10:21.42 | E-bola | Im new to asterisk, and not completely sure how i enable asterisk to receive calls from normal phones, and how i can call out to normal phones |
| 10:21.56 | E-bola | I want to use a SIP provider, but im not sure how to set that up in asterisk |
| 10:22.22 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 10:27.04 | *** join/#asterisk AuPix (n=AuPix@81.171.226.204) |
| 10:28.29 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 10:28.34 | AuPix | Has anyone got asterisk/trunk to configure h323 yet? |
| 10:29.27 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 10:42.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:56.52 | tzafrir | ~help |
| 10:58.22 | BugKham | ~help |
| 11:01.10 | *** part/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 11:03.09 | Zerthimon | ~docs |
| 11:03.11 | jbot | docs is probably 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 |
| 11:06.00 | *** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net) |
| 11:08.45 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 11:19.00 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 11:19.02 | vivek | hello all |
| 11:20.03 | vivek | What's the best ata with aix2 support and fxo and fxs ports ? I would be nice if they keep updating codecs (i know its a uptopian dream but ...) spa3k seems nice except for the lack of aix2 and global ip codecs ... |
| 11:21.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:24.10 | *** part/#asterisk tparcina (n=tparcina@lns01-1072.dsl.iskon.hr) |
| 11:38.09 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 11:39.39 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 11:42.26 | Zerthimon | anyone alive here and able to help with zap ? |
| 11:42.28 | jhiver | hey guys |
| 11:42.57 | jhiver | do you know of any hardware solution that can handle SIP and / or H323 codec conversion and optionally which does echo cancellation? |
| 11:43.24 | vivek | jhiver: spa3k handles sip ... |
| 11:43.32 | vivek | and spa2k does sip too ... |
| 11:43.41 | vivek | but no h323 ... |
| 11:44.06 | jhiver | errrr |
| 11:44.12 | jhiver | spa3k is like one channel |
| 11:44.26 | jhiver | I need something with more capacity, for softswitching type requirements |
| 11:44.31 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 11:44.31 | jhiver | like minimum 30 channels |
| 11:45.52 | jhiver | Asterisk doesn't handle g723 which is kind of annoying |
| 11:46.42 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 11:47.26 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 11:48.08 | BugKham | why is mpg123 still with 1.2.9.1 package? |
| 11:48.37 | BugKham | in the wiki it said "In Asterisk 1.2.x and above you no longer need to use the mpg123 player .." |
| 11:50.27 | *** join/#asterisk acehunky (n=chat_jok@59.184.28.92) |
| 11:55.13 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:03.55 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
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| 12:09.08 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 12:10.24 | *** part/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 12:17.21 | *** join/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 12:18.44 | negativecreep | hi all |
| 12:18.45 | negativecreep | i am planning to run asterisk on openvpn. |
| 12:19.09 | negativecreep | the client will be behind a nat while the asterisk server is on a public ip. |
| 12:19.32 | negativecreep | i wnat to setup a tunnel between the asterisk server and the client and then communicate between them on private ip addresses. |
| 12:19.37 | negativecreep | is it possible? |
| 12:20.46 | *** join/#asterisk B4 (n=B4@202.69.48.245) |
| 12:21.15 | B4 | hi ... anyone alive at this time? |
| 12:21.23 | X-Gen | Nope |
| 12:21.31 | negativecreep | :) |
| 12:21.39 | B4 | two! |
| 12:21.59 | AltnTab | Anyone said, alive ?! ;) |
| 12:22.06 | B4 | hmm no |
| 12:22.09 | B4 | lol |
| 12:22.18 | AltnTab | hmodes, my mistake ... back 2 sleep |
| 12:22.33 | B4 | k who can I poke regarding problem with E1 outbound calls |
| 12:22.48 | *** join/#asterisk ACiDV (n=acidv@modemcable247.11-37-24.mc.videotron.ca) |
| 12:23.15 | negativecreep | not me.. |
| 12:23.19 | B4 | anyone? |
| 12:24.09 | ACiDV | Hi all =^) Anyone have an idea of what can be this error : ERROR[11790]: pbx.c:5913 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 12:24.41 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:24.42 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:25.20 | ACiDV | showed on CLI 10-20 lines per seconds. Using Asterisk SVN branches... |
| 12:25.25 | B4 | Vorondil ... bjohnson ... can you guys help for problem related to E1 outbound calls? |
| 12:25.55 | bjohnson | I don't have E1 |
| 12:26.02 | negativecreep | guys...does it sound plausible to run asterisk over a vpn..using a private ip for the asterisk server and a private ip for the client.. |
| 12:26.08 | negativecreep | just one client needs to connect. |
| 12:26.10 | negativecreep | over the vpn. |
| 12:26.17 | bjohnson | negativecreep: yes |
| 12:26.50 | ACiDV | except this error about Data Buffer Size, asterisk work perfectly on a server (callcenter) w/ 8 full T1 |
| 12:26.53 | bjohnson | but depending on the protocol, it doesn't have to be a full vpn. If a one port protocol, a ssh tunnel would suffice |
| 12:28.32 | negativecreep | bjohnson: thnx for the explanation...could u point me to some link which explains this in a bit more detail. |
| 12:28.40 | negativecreep | or if you can explain this ssh scenario. |
| 12:28.42 | negativecreep | i am using sip |
| 12:28.46 | negativecreep | but looking into using iax. |
| 12:28.58 | bjohnson | negativecreep: sorry no. try google |
| 12:29.11 | bjohnson | search for ssh forwarding |
| 12:29.37 | bjohnson | it's a standard ssh function, all clients should have it |
| 12:29.44 | negativecreep | bjohnson: i do understand ssh forwarding. |
| 12:29.45 | negativecreep | :) |
| 12:29.47 | bjohnson | it won't work with SIP though |
| 12:30.02 | bjohnson | should work with iax |
| 12:30.07 | negativecreep | ahan |
| 12:30.13 | bjohnson | (I haven't done it myself) |
| 12:30.18 | negativecreep | hmm.. |
| 12:30.20 | negativecreep | k |
| 12:30.31 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 12:30.34 | Bert- | hello there :) |
| 12:30.38 | negativecreep | any iax softphones out there for linux? |
| 12:30.43 | Bert- | today is big day ;) |
| 12:31.11 | B4 | I am getting Cause: INVALID_NUMBER_FORMAT (28) in PRI debug :( |
| 12:31.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:31.31 | B4 | IAX2 works great over VPNs |
| 12:31.48 | Bert- | I wondering if Asterisk is able to make difference between 3, and 301 in IVR, as 3 is a submenu, and 301 a direct extension |
| 12:32.05 | negativecreep | thnx B4..i am starting work on iax. |
| 12:32.25 | Bert- | I mean, I want to be able to press 3 for legal, or dial 301 to have agent 301 ringing |
| 12:32.45 | [TK]D-Fender | Bert- : Yes. |
| 12:32.51 | B4 | I have used IAX with SSL tunnels as well as SSH, IPSEC and PPTP ... world with everything |
| 12:32.53 | Bert- | [TK]D-Fender okay |
| 12:32.55 | Bert- | don't tell me |
| 12:33.03 | Bert- | juste want to find by myself |
| 12:33.03 | Bert- | :) |
| 12:33.11 | Bert- | but juste wanted to know if it is possible :) |
| 12:33.41 | B4 | should be possible with correct timeouts |
| 12:34.15 | [TK]D-Fender | If you have an exten for 3, and 301 in the same context, if you push the first digit, timeout rules will detemine when to accept taht as the final answer becuase the user may want to continue to dial another valid exten, namely 301. |
| 12:34.19 | ACiDV | Found the source of my 'Data Buffer Size Exceeded' message.... related to a Music On Hold class that have no file in the directory... |
| 12:34.26 | *** join/#asterisk nettie (i=[U2FsdGV@85-18-54-38.ip.fastwebnet.it) |
| 12:35.07 | B4 | nettie ... any idea on E1 outbound calls? |
| 12:35.27 | nettie | B4 I'm sorry I only use asterisk to do sip2sip |
| 12:36.27 | B4 | k thanks |
| 12:36.38 | nettie | I just have an fxo card in the box I didnt configure it yet. |
| 12:37.24 | B4 | oh ok |
| 12:37.45 | nettie | Guys, asterisk v1.4 will be based on the current "trunk", considering 1.4 is almost out I supposed that "trunk" is going to be pretty stable.. am I right? |
| 12:38.04 | B4 | yeah trunk is stable |
| 12:38.13 | nettie | B4 you're using it? |
| 12:38.57 | nettie | are you also using jitterbuffer with sip channel? |
| 12:40.26 | B4 | yes nettie I have trunk running |
| 12:47.05 | *** join/#asterisk GyrosGeier (n=richter@p54997B55.dip.t-dialin.net) |
| 12:47.07 | GyrosGeier | hi |
| 12:47.34 | B4 | hi |
| 12:47.51 | GyrosGeier | I'm rebuilding chan_misdn (as an external package) against current Asterisk, and get bitten by |
| 12:48.06 | GyrosGeier | #define pthread_mutex_t use_ast_mutex_t_instead_of_pthread_mutex_t |
| 12:48.26 | tzafrir | ~pb |
| 12:48.33 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 12:48.58 | GyrosGeier | I know for a fact that pthread_mutex_t is correct at this place because it is the ABI of the mISDN library, not Asterisk ABI. |
| 12:50.25 | nounoursfre | the chan asterisk france is #asteriskfr |
| 12:52.10 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:52.50 | GyrosGeier | is there any sane way to get rid of these #defines? It seems the only thing I can do is #undef them, but I'd like to do that cleanly |
| 12:55.19 | *** join/#asterisk variables (n=variable@69-172-36-47.atlsfl.adelphia.net) |
| 12:58.53 | *** part/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 13:00.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:02.02 | nounoursfre | do you have test asterisk and google talk ? |
| 13:03.02 | Bert- | I have a(nother) stupid idea: |
| 13:03.48 | Bert- | Is a wayto make somthing like that : dial myreal phone number + an extension, and be routed directly to the correct agent ?? |
| 13:04.42 | Bert- | a thing like that: +335000000#2002 so call 335000000 (where there is an asterisk), then asterisk see the #2002 and dial 2002 ?? |
| 13:04.58 | Bert- | it is a little piggy I reckon |
| 13:05.01 | Bert- | :) |
| 13:05.08 | Bert- | but it could be great |
| 13:06.02 | [TK]D-Fender | Bert- : That woul be like inventing a phone number which sorry, just doesn't work. |
| 13:09.34 | tzanger | hey |
| 13:09.39 | tzanger | who here has little kids at home? |
| 13:09.42 | tzanger | I have a question |
| 13:10.07 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:10.08 | tzanger | I have three kids of "little piggy" age... every one of them, upon finishing the little piggy game, will give you their other foot to do that one too |
| 13:10.36 | *** part/#asterisk B4 (n=B4@202.69.48.245) |
| 13:10.56 | *** join/#asterisk Marso (i=NA@c-71-226-191-170.hsd1.ga.comcast.net) |
| 13:11.51 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 13:12.43 | rob0 | My little ones are getting bigger now, but they still like little piggies. |
| 13:12.59 | rob0 | 9 and 8 |
| 13:13.01 | noky | i want to know how can i block the "183 Session Progress" from my SER 0.8.14... |
| 13:13.11 | noky | anybody know? |
| 13:13.31 | *** join/#asterisk s0lid (n=s0lid@202.147.29.146) |
| 13:14.08 | GyrosGeier | noky, why? |
| 13:14.16 | [TK]D-Fender | noky : Mellita #2 filters |
| 13:14.36 | tzanger | rob0: yeah my 7 year old likes it |
| 13:14.42 | tzanger | the 10yo is a little old for it :-) |
| 13:14.48 | tzanger | but the 2 and 5 year old love it |
| 13:14.51 | Ahrimanes | exten => _XXXXXXX.,10,Set(__ACODE=${SIPPEER(${CALLERIDNUM:2}:accountcode)}) <- any reason this shouldnt fetch the accountcode of the sippeer with username matching ${CALLERIDNUM:2} ? |
| 13:14.55 | kay2 | [TK]D-Fender: when I am talking to someone, how can I invite a third one to the conversation ? |
| 13:15.02 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 13:15.12 | kay2 | [TK]D-Fender: like 3 way conferancing |
| 13:15.20 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:15.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:15.22 | [TK]D-Fender | kay2 : Depends on the phone/tech |
| 13:15.29 | kay2 | [TK]D-Fender: sip |
| 13:15.30 | tzanger | once in a while with the older ones I will have the last little piggy get dragged off by the big bad wolf (and drag them across the floor) |
| 13:15.34 | *** part/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 13:15.37 | Qb3rt | is there a manner to run a stress test on an asterisk server??? with a software or i dont know? |
| 13:15.41 | rob0 | haha |
| 13:15.42 | [TK]D-Fender | kay2 : And depends on the phone... |
| 13:15.59 | kay2 | [TK]D-Fender: is it to the phone to do the mixing ? |
| 13:16.04 | kay2 | [TK]D-Fender: or to asterisk |
| 13:17.05 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:17.35 | noky | [TK]D-Fender: hi! |
| 13:17.40 | noky | what is mellita #2 ? |
| 13:18.29 | noky | GyrosGeier: because i have a bug with my gatewaysip,, and this sip message don't do me nothing to my things |
| 13:18.37 | [TK]D-Fender | kay2 : For SIP it'd be the phone |
| 13:18.55 | [TK]D-Fender | noky : Makes great coffee ;) |
| 13:19.01 | noky | xD |
| 13:19.13 | noky | i don't have good documentation from the SER =( |
| 13:19.20 | noky | please help me [TK]D-Fender |
| 13:19.44 | *** part/#asterisk nounoursfre (n=nounours@213.161.196.217) |
| 13:19.54 | [TK]D-Fender | noky : sORRY, NEVER USED IT. |
| 13:20.32 | kay2 | [TK]D-Fender: for IAX ? |
| 13:21.08 | noky | :( |
| 13:22.03 | [TK]D-Fender | kay2 : IAX is also done by the phone IIRC |
| 13:22.55 | kay2 | [TK]D-Fender: well beside ZAP, what is done by asterisk ? |
| 13:23.20 | [TK]D-Fender | kay2 : Maybe you should try asking a more meaningful and specific question. What are you trying to do? |
| 13:24.02 | [TK]D-Fender | kay2 : * just passes packet, and the only time it sits in the stream is when recording, or apps like MeetMe. But when you Dial someone its typically just flows through. |
| 13:26.05 | kay2 | [TK]D-Fender: ok, but basically, If I want to invite 4 people for a meetme without having to add something in meetme.conf |
| 13:26.07 | kay2 | how can I do ? |
| 13:26.15 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 13:27.05 | kay2 | [TK]D-Fender: isnt it possible to make a conferance and the meetme room created on the fly ? |
| 13:28.41 | BugKham | anyone still using mpg123 in 1.2.x? |
| 13:29.02 | kay2 | BugKham: what could be used instead ? |
| 13:29.33 | BugKham | kay2, for MOH that's what I mean |
| 13:30.07 | nortex | kay2, Check out app_conference |
| 13:30.23 | *** join/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 13:30.28 | [TK]D-Fender | kay2 : There are ways of making dynamic conference rooms in MeetMe but I don't know the details. Another option is having 2 sip phones on your * box bridge everyone. So SIP/A calls 1 other person as well as SIP/B. SIP/B then conferences in another call and presto, you have 4 people in "conference" all done on the phone level. |
| 13:30.42 | BugKham | kay2, I'm trying to compare the voice quality between the built in mp3 support in asterisk-addons and mpg123 |
| 13:30.47 | [TK]D-Fender | BugKham : I did for a while. |
| 13:31.05 | BugKham | [TK]D-Fender, so what did you end up with? |
| 13:31.28 | BugKham | [TK]D-Fender, the mpg123 I used was better than the built in one |
| 13:33.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:33.02 | BugKham | [TK]D-Fender, but when I tried to use the custom feature with mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 13:33.21 | BugKham | it still doesn't work that well |
| 13:33.41 | BugKham | [TK]D-Fender, any idea? |
| 13:33.48 | [TK]D-Fender | BugKham : I never really noticed the differnce personally. |
| 13:34.27 | [TK]D-Fender | BugKham : Nor did I delve into it since it never matterd that much. |
| 13:34.32 | jhiver | aaargh |
| 13:34.49 | BugKham | [TK]D-Fender, it really bothers me here =( |
| 13:35.09 | jhiver | can't get to install digium's g729 codecs, and the illegal ones neither |
| 13:35.23 | jhiver | so I have ZERO alternative :-/ |
| 13:35.44 | *** join/#asterisk walhala (n=niolou@stardust.noc.frontier.fr) |
| 13:37.10 | *** join/#asterisk nounoursfr (n=nounours@213.161.196.217) |
| 13:37.34 | [TK]D-Fender | jhiver : Whats the trouble with Digium's? |
| 13:42.30 | jhiver | [TK]D-Fender, I can't use the register tool with FreeBSD |
| 13:42.39 | jhiver | it's telling me it can't find any network interfaces |
| 13:42.47 | jhiver | and so refuses to register the codec |
| 13:43.17 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:43.35 | jbalcomb | marklar |
| 13:44.11 | m4rkl4r | marklar, marklar |
| 13:44.46 | CoaxD | jhiver: Sounds to me like you need to fix that. |
| 13:45.00 | nortex | polo |
| 13:45.11 | jbalcomb | [TK]D-Fender: Happy Belated Canada Day!! |
| 13:47.06 | [TK]D-Fender | jbalcomb : SHUP YUO! |
| 13:47.13 | [TK]D-Fender | jbalcomb : ;) |
| 13:47.19 | [TK]D-Fender | jbalcomb : How goes the war? |
| 13:47.51 | jbalcomb | [TK]D-Fender: ;) I got the damn AMI to give the username and extension based on the IP I'm pulling from ArpWatch. |
| 13:48.24 | jbalcomb | [TK]D-Fender: So no I have a Table with MAC, IP, Username, and Exten. |
| 13:48.35 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 13:48.41 | faberk64 | hi folks |
| 13:49.17 | jbalcomb | [TK]D-Fender I need to code the interface and make massive tables of the configs for the phones. That'll give me an alpha level working system. |
| 13:49.22 | faberk64 | how is possible to strip-off only centain numbers from called number? |
| 13:49.28 | Dovid | hello |
| 13:49.54 | [TK]D-Fender | jbalcomb : So your app scans and then compares against the * SIP registry? |
| 13:49.59 | faberk64 | I need to stri only 39 from ONLY the numbers beginning with it |
| 13:50.18 | Dovid | What do u mean by certain ? |
| 13:50.22 | faberk64 | if the beginning is different, do not touch it |
| 13:50.42 | faberk64 | I mean, that ONLY 39 must be stripped |
| 13:50.44 | [TK]D-Fender | faberk64 : GotoIf + Set {thevar:2} |
| 13:50.49 | jbalcomb | [TK]D-Fender: yes'm. SIPpeers to get all registered users, pulls the ObjectName field, and the does a SIPshowpeer <exten> for each one. |
| 13:51.11 | *** join/#asterisk Galeras (n=Galeras@litigaractivos1.att.net.co) |
| 13:51.14 | [TK]D-Fender | jbalcomb : Whats the goal? If its already defined in * what do you do with this newfound association? |
| 13:51.21 | Galeras | Hi! |
| 13:51.26 | faberk64 | {39:2} |
| 13:51.26 | Dovid | Meaning ? |
| 13:51.30 | faberk64 | is this? |
| 13:51.35 | Dovid | nno |
| 13:51.42 | Dovid | Can I PM ? |
| 13:51.45 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
| 13:52.06 | [TK]D-Fender | faberk64 : No. Pastebin the section of your dialplan you'd like to integrate this into. |
| 13:52.12 | [TK]D-Fender | ~pb |
| 13:52.14 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 13:52.18 | jbalcomb | [TK]D-Fender: The interface need to have accurate information so you can choose a user to change, and it uses the exten in automatically building the config for the phone. |
| 13:52.39 | Galeras | Please, tell me how can i get datetimes from queues.log. Thanks |
| 13:53.10 | [TK]D-Fender | Galeras : its the first field and is in the WIKI page for it. Its a UNIXTIME standard notation. |
| 13:53.20 | jbalcomb | Galeras: download queuemetrics and go through their queueLoader.pl |
| 13:53.38 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 13:53.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:53.57 | jbalcomb | ~UNIXTIME |
| 13:54.12 | jbalcomb | jbot: UNIXTIME |
| 13:54.41 | Galeras | Thanks to all: UnixTme |
| 13:54.42 | [TK]D-Fender | jbot : You are NOT all-knowing! |
| 13:54.44 | jbot | [TK]D-Fender: I think you lost me on that one |
| 13:54.51 | [TK]D-Fender | jbot : Obviously. |
| 13:55.01 | [TK]D-Fender | :D |
| 13:55.54 | jbalcomb | [TK]D-Fender: What is Canada Day in celebration of? |
| 13:56.23 | jbalcomb | damnit, there goes my dangling preposition again.. |
| 13:56.27 | BugKham | [TK]D-Fender, u know if musiconhold.conf will be reloaded with the "reload" from CLI? |
| 13:56.36 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 13:56.52 | BugKham | [TK]D-Fender, doesn't seem to work for me |
| 13:57.37 | *** join/#asterisk Skymarshal (n=Skymarsc@wlan-5.uni-koblenz.de) |
| 13:58.18 | [TK]D-Fender | jbalcomb : Careful or the Grammar Rangers may send the ninjas to take you out ;) |
| 13:58.36 | [TK]D-Fender | BugKham : You have to unload/reload the module IIRC. |
| 13:58.59 | Skymarshal | exten => _0903100X,1,Set(ZAEHLER = $[${ZAEHLER} + 1]|g) |
| 13:59.00 | Skymarshal | allways results in: Setting global variable 'ZAEHLER ' to ' 0' which is not ok because it should be 1. Any ideas? It is working with "-" and that makes me running wild. |
| 13:59.07 | BugKham | [TK]D-Fender, oh, ok |
| 14:00.24 | [TK]D-Fender | Skymarshal : Whats with the "|g" at the end? |
| 14:01.05 | Skymarshal | [TK]D-Fender: |g should be global (refering to the documentation) |
| 14:01.18 | [TK]D-Fender | Skymarshal : and I'd suggest removing the extra whitespace as well. |
| 14:04.08 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 14:04.24 | [TK]D-Fender | Skymarshal : Wouldn't hurt to NoOp the var followed by that Eval you re doing to see if the contexts work sanely spereately. |
| 14:05.13 | Skymarshal | [TK]D-Fender: I try that. |
| 14:07.40 | Bert- | hmm |
| 14:08.08 | Bert- | does someone has ever interconnected asterisk with a Nextone softswitch please ? |
| 14:09.50 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 14:10.26 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 14:11.23 | PerlStalker | away |
| 14:18.10 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:18.34 | *** join/#asterisk s0lid (n=s0lid@202.147.29.146) |
| 14:21.23 | *** join/#asterisk hohum (n=dcorbe@12.195.58.237) |
| 14:27.41 | *** join/#asterisk pdavid (n=chatzill@adsl-072-151-167-100.sip.mob.bellsouth.net) |
| 14:27.45 | pdavid | morning all |
| 14:28.06 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 14:31.57 | *** join/#asterisk Skarmeth (n=Skarmeth@201009012196.user.veloxzone.com.br) |
| 14:32.17 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
| 14:32.28 | Persilon | Hi |
| 14:32.59 | Persilon | I'm trying to use rxfax and txfax... Is there anyway of making an extension use tx and another rx to test send and recieve ? |
| 14:34.43 | *** join/#asterisk [koss] (i=koss@adsl-67-39-192-129.dsl.bcvloh.ameritech.net) |
| 14:34.55 | GyrosGeier | sure |
| 14:35.15 | GyrosGeier | you need a command file, as described in the txfax docu |
| 14:35.20 | [koss] | i need a Dual RJ45 PoE IP phone -- im between Polycom IP430 and Linksys 942 -- any opinions? |
| 14:35.45 | GyrosGeier | <aol>me too</aol> |
| 14:36.12 | *** join/#asterisk jcmoore (n=jcmoore@picard.ojc.nuvio.com) |
| 14:36.41 | Hmmhesays | whoa |
| 14:36.48 | [TK]D-Fender | [koss] : depends what you want out of a phone. |
| 14:37.11 | [koss] | nothing special |
| 14:37.27 | [TK]D-Fender | [koss] : Polycom supports more calls, but fewer line keys, better sound quality and pysical construction. |
| 14:37.40 | [TK]D-Fender | [koss] : IP430 would eb a fine choice for most uses. |
| 14:38.11 | [koss] | Do I need anything special (besides a PoE switch) for the IP430 to work over PoE...? |
| 14:38.28 | Persilon | GyrosGeier: I tried using Dial(SIP/[rxextension],10) and then txfax, but the xfax extension doesn't answer |
| 14:38.30 | [TK]D-Fender | [koss] : Nope. |
| 14:38.33 | GyrosGeier | [koss], judging from the first hits on google, respectively, the Linksys seems better due to sRTP support |
| 14:39.01 | GyrosGeier | Persilon, yes, because you are calling a SIP phone |
| 14:39.06 | [TK]D-Fender | GyrosGeier : Polycom supports it now as well IIRC. Aside from the fact that * DOESN'T yet... |
| 14:39.18 | GyrosGeier | Persilon, you need to call an extension |
| 14:39.21 | [koss] | hrmmm do I really need SRPT on a LAN? |
| 14:39.32 | [TK]D-Fender | [koss] : Paranoia knows no bounds. |
| 14:39.34 | Persilon | GyrosGeier: that's the caller parameter on txfax ? |
| 14:39.49 | noky | i want to know how can i block the "183 Session Progress" from my SER 0.8.14... |
| 14:40.04 | GyrosGeier | Persilon, no, your call is never dispatched via the dialplan |
| 14:40.13 | [koss] | the screen on the linksys looks nicer |
| 14:40.22 | [koss] | that's my expert analasys |
| 14:40.37 | Persilon | GyrosGeier: so how can I call the rx extension to test ? |
| 14:41.02 | [TK]D-Fender | [koss] : Iffy. On the IP 430 its a close call. (from the pics I've seen). I have every other model they put out so far. |
| 14:41.07 | GyrosGeier | Persilon, it has been quite a while since I did that the last time |
| 14:41.13 | gaupe | [koss]: look into the thomson st2030 too |
| 14:41.23 | GyrosGeier | Persilon, (I used to be Debian maintainer for that stuff) |
| 14:41.46 | Persilon | GyrosGeier: I used to use debian :P |
| 14:41.57 | GyrosGeier | [koss], the Linksys might be able to speak Skinny as well. |
| 14:42.13 | [TK]D-Fender | GyrosGeier : Not sure thats a "plus" ;) |
| 14:42.23 | [TK]D-Fender | GyrosGeier : Who wants a dumb phone? |
| 14:42.35 | [koss] | I might need a "nicer" phone too for "executives" |
| 14:42.45 | [koss] | so they can feel special -- and a receptionist phone |
| 14:42.46 | gaupe | GyrosGeier: it hasn't been , the 941/942 is based on the ATA-adapters |
| 14:43.06 | [TK]D-Fender | [koss] : Better reason to go Polycom, for unified provisioning. So 501/601 for managers/receptionists |
| 14:43.13 | *** join/#asterisk netoguy (n=skelley@64-199-141-122.ip.mcleodusa.net) |
| 14:43.24 | [TK]D-Fender | [koss] : IP 601 + attenedant modules = nice |
| 14:43.40 | GyrosGeier | [TK]D-Fender, Skinny specifies a protocol for the display IIRC |
| 14:44.01 | [koss] | ok cool, polycom it is :) |
| 14:44.16 | [koss] | here's a nother doozey -- what do y'all think of fonality ? |
| 14:44.23 | GyrosGeier | gaupe, I see; As Linksys == Cisco these days it might have been |
| 14:44.56 | gaupe | GyrosGeier: it is, these phones are sipuras - cisco is just the brand printed on |
| 14:45.07 | GyrosGeier | gaupe, (also I remember someone having a similar phone at the Asterisk booth at a fair lately and driving it via Skinny) |
| 14:45.13 | [TK]D-Fender | [koss] : Picture solitary confinment. GUI = loss of control. |
| 14:45.28 | [koss] | lol ok... |
| 14:45.42 | gaupe | GyrosGeier: the ciscos are similar ;) |
| 14:45.43 | [koss] | so just vanilla asterisk no "aftermarket" GUI? |
| 14:45.55 | netoguy | does anyone know if Wireless IP Phones work with repeaters like the Linksys WRE54G? |
| 14:46.00 | [TK]D-Fender | [koss] : You typically choose * to both save money and gain control. Going with a GUI undoes much of that gain. |
| 14:46.12 | GyrosGeier | netoguy, sure, a phone cannot see that it's a repeater |
| 14:46.19 | acehunky | hello room |
| 14:46.27 | [TK]D-Fender | [koss] : That's what I advise from most situations. Describe the setup you're looking at doing. |
| 14:46.34 | acehunky | can anyone guide me if asterisk supports video over SIP ? |
| 14:46.46 | GyrosGeier | netoguy, just keep in mind a repeater eats half of your bandwidth and doesn't do QoS usually |
| 14:46.49 | netoguy | GyrosGeie, do you know how it handles switching between multiple repeaters? |
| 14:46.54 | acehunky | I could see H.264 stuff on the wiki.. |
| 14:47.00 | netoguy | Should it work seamlessly? |
| 14:47.05 | GyrosGeier | netoguy, depends on the phone |
| 14:47.09 | GyrosGeier | netoguy, it *should* |
| 14:47.29 | [TK]D-Fender | acehunky : Yes |
| 14:47.29 | acehunky | Video over Asterisk .. any one ? |
| 14:47.36 | netoguy | GyrosGeier, do you have any experiece with any of the wifi phones? |
| 14:47.44 | [TK]D-Fender | acehunky : Done it with eyeBeam before. |
| 14:47.48 | acehunky | hi [TK]D-Fender |
| 14:47.54 | GyrosGeier | netoguy, in principle, an 802.11 station is supposed to watch all beacons with the same ESSID and switch to another AP if it gets a stronger signal |
| 14:48.14 | acehunky | [TK]D-Fender is there any document that you can point me to ? on getting this working with Eyebeam .. |
| 14:48.21 | GyrosGeier | netoguy, in practice, they switch lazily when the S/N ratio gets really bad. |
| 14:48.23 | acehunky | or probably with the new Video IP Phone from Grandstream |
| 14:48.31 | [koss] | [TK]D-Fender: I'm replacing an old telrad PBX (bought used a year ago by someone nolonger here). In any event -- We have about 35 cubes+office with 1 RJ45 drop right now. In a year we'll need about 55 or so. So I just need a relatively small phone system |
| 14:48.53 | GyrosGeier | netoguy, I don't have particular experience with phones, I just wrote an 802.11 stack once. |
| 14:49.27 | [TK]D-Fender | acehunky : Its all in the WIKI. just enable the video codecs and that should be it. |
| 14:49.33 | netoguy | GyrosGeier, thank you for the info. |
| 14:49.44 | GyrosGeier | netoguy, if you want a "professional" solution, go with proper wiring and "managed" or "lightweight" APs; I specifically recommend Aruba's stuff as the APs handle handover if the stations are too dumb |
| 14:50.22 | [TK]D-Fender | [koss] : Ok for that size sure, just do it "plain vanilla". GUI only pays off if you have morons admining it, or your setup is huge and config files become too heavy |
| 14:50.32 | GyrosGeier | (basically, the APs kick stations out that have other APs nearer) |
| 14:50.42 | acehunky | http://www.voip-info.org/wiki-Asterisk+video ? [TK]D-Fender |
| 14:51.54 | [TK]D-Fender | acehunky : yup |
| 14:52.06 | acehunky | How do i get the codecs ? sorry but i am kinda new to this area .. is there any patent with the h.264 codecs ? |
| 14:52.14 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 14:53.31 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 14:53.52 | acehunky | aaah [TK]D-Fender i just read that we dont need video codecs inside asterisk .. but it needs to be in the client .. :) |
| 14:53.55 | [TK]D-Fender | acehunky : No need, they will be used in "passthrough" mode. * will not affect them. |
| 14:55.54 | acehunky | yeah ... just read it .. i think i need to update the wiki ;) its written in those small letters in comments .. like one of those hidden charges on any product brochure:P |
| 14:56.35 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-163.rockynet.com) |
| 15:00.08 | GyrosGeier | hmm |
| 15:00.19 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-4-128.telkomadsl.co.za) |
| 15:00.24 | |oranjia| | hello world :) |
| 15:00.36 | Dovid | Hello :) |
| 15:01.02 | |oranjia| | Does anyone have Agi experience? I need a bit of help sorting out a little scriptlet |
| 15:01.21 | [TK]D-Fender | GyrosGeier : Which ATA's support PoE? I've never heard of any.... |
| 15:02.07 | *** join/#asterisk FRUMUSHELUL (n=nobody@83.218.216.199) |
| 15:02.16 | noky | i want to know how can i block the "183 Session Progress" from my SER 0.8.14... |
| 15:02.16 | FRUMUSHELUL | hello. i need some help on asterisk |
| 15:02.19 | noky | i want to know how can i block the "183 Session Progress" from my SER 0.8.14... |
| 15:02.48 | FRUMUSHELUL | how can i interconnect GnuGk with Asterisk? |
| 15:03.13 | [TK]D-Fender | noky : Perhaps you should ask in #ser .... |
| 15:03.16 | *** join/#asterisk s0lid (n=s0lid@124.106.215.82) |
| 15:03.35 | *** join/#asterisk jaike (n=a@203.115.188.120) |
| 15:03.41 | FRUMUSHELUL | Netmeeting->GnuGk->Asterisk->PSTN |
| 15:04.30 | noky | [TK]D-Fender: nobody answer in #ser =( |
| 15:04.46 | GyrosGeier | [TK]D-Fender, there are a few |
| 15:04.57 | jbalcomb | woot. got me three more Polycom IP 501s. |
| 15:05.01 | GyrosGeier | [TK]D-Fender, problem is that they are ridiculously expensive |
| 15:05.24 | GyrosGeier | [TK]D-Fender, and I already spent 250 EUR on a 16-way PoE switch |
| 15:06.02 | |oranjia| | if anyone can give me some thoughts : http://pastebin.com/740852 |
| 15:06.04 | |oranjia| | :) |
| 15:07.36 | [TK]D-Fender | noky : Well here certainly isn't the place to spam it. You'd probably have better luck ont he mailing lists. |
| 15:08.28 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:09.55 | pdavid | anyone here using trixbox by chance? |
| 15:10.22 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 15:10.29 | jbalcomb | hey pdavid, according to the title of this channel you might find trixbox help in #freepbx. |
| 15:10.50 | pdavid | jbalcomb: that is certainly true! sorry! :) |
| 15:11.00 | jbalcomb | pdavid: otherwise, i haven't heard of anyone using trixbox in here. ;) |
| 15:11.09 | pdavid | actually, how about anyone using a sipura spa3000 |
| 15:11.13 | E-bola | Anybody tried to use asterix with live communication server? |
| 15:11.26 | pdavid | i am having difficulty using vertical activation codes on a line connected to an spa3000 |
| 15:11.44 | pdavid | any *xx numbers are not getting through to my * box |
| 15:12.03 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:12.35 | [TK]D-Fender | GyrosGeier : You can get a 24 Port PoE switch for $366 USD |
| 15:13.58 | [TK]D-Fender | pdavid : probably because the ATA us isung them and not passing them on. |
| 15:14.15 | pdavid | Fender: any thoughts on getting it to work with the spa3k? |
| 15:14.19 | pdavid | the config pages are a dizzying array of options! |
| 15:14.51 | [TK]D-Fender | pdavid : there are 2 pages that refer to CLASS codes and are blatantly worded. If you can't figure them out, then something is seriously wrong... |
| 15:15.19 | pdavid | which 2 pages? |
| 15:15.21 | [TK]D-Fender | pdavid : 1 page to choose the code to associate with an INTERNAL CLASS feature, the other to ENABLE/DISABLE it. |
| 15:15.37 | [TK]D-Fender | pdavid : Go into the SPA's web admin and look. |
| 15:15.38 | jbalcomb | [TK]D-Fender: Do you have a polycom ip 501 config file? |
| 15:15.47 | [TK]D-Fender | jbalcomb : Yup, same as any other really. |
| 15:16.08 | [TK]D-Fender | jbalcomb : only diff in my config is the user/pass, and line assignment. |
| 15:16.30 | jbalcomb | [TK]D-Fender: what about the auto-answer? |
| 15:16.32 | [TK]D-Fender | jbalcomb : jbot : If you said you have "3 more", then you should already HAVE a good config to base off of. |
| 15:16.48 | [TK]D-Fender | jbalcomb : Never got it working (didn't try too hard at it though_ |
| 15:17.02 | pdavid | Fender: under Regional and Line 1? I see a list of available vertical activation codes, and a bunch of supplementary service subscription options |
| 15:17.03 | jbalcomb | [TK]D-Fender: i didnt use config files to do the first two. |
| 15:17.22 | [TK]D-Fender | :O |
| 15:17.28 | jbalcomb | [TK]D-Fender: I'm also unclear about which files do what. |
| 15:17.44 | [TK]D-Fender | jbalcomb : So you've never provisioned them before? |
| 15:17.48 | [TK]D-Fender | jbalcomb : Any of them? |
| 15:17.50 | jbalcomb | [TK]D-Fender no |
| 15:17.58 | jbalcomb | [TK]D-Fender web interface |
| 15:18.04 | [TK]D-Fender | jbalcomb : How my Polycom's do you own at this point, and which models? |
| 15:18.05 | file | [TK]D-Fender: geez one letter off in french and strawberry turns into fresh |
| 15:18.22 | jbalcomb | [TK]D-Fender i have 5 ip 501s now and 4 or 5 841/941s |
| 15:18.32 | jbalcomb | [TK]D-Fender and an spa-2002 |
| 15:18.53 | [TK]D-Fender | file : a few decades ago a guy was executed because of a misplaced comma in a telegram, so I'll let you off easy this time ;) |
| 15:19.23 | file | well I just glanced at this Subway thing and my mind translated it into "Eat Strawberry" until I reread it |
| 15:19.29 | [TK]D-Fender | jbalcomb : go grab 1.6.6. and untar it in a folder on your server and look at the sample files there. |
| 15:19.49 | [TK]D-Fender | jbalcomb : only a few settings to change for them to get up and running, and they'll auto-update to the latest firmare in the same shot. |
| 15:19.50 | jbalcomb | [TK]D-Fender okidoki |
| 15:20.33 | GyrosGeier | [TK]D-Fender, ack, plus shipping and handling; also, I need it to do VLAN and lots of other stuff. |
| 15:20.33 | jbalcomb | [TK]D-Fender i need to get all the config options figured out so i can build my DB table at some point too. cunningpike said he'd be glad to help out with the polycom configs |
| 15:20.54 | Bert- | no one here ever tried asterisk and nextone ? |
| 15:21.04 | jbalcomb | Bert- Not I. |
| 15:21.11 | [TK]D-Fender | GyrosGeier : D-Link DES-1526 does VLAN/QoS as well though I never peronally played with it... |
| 15:21.32 | GyrosGeier | [TK]D-Fender, urgh, but it's a D-Link |
| 15:21.43 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.143) |
| 15:22.22 | [TK]D-Fender | GyrosGeier : Those work great for me... |
| 15:22.27 | vader-- | hello |
| 15:22.33 | vader-- | hi defender |
| 15:22.40 | [TK]D-Fender | vader-- : y0 |
| 15:22.51 | vader-- | weekend good? |
| 15:23.12 | GyrosGeier | [TK]D-Fender, well, the lowest price I see is 422 EUR |
| 15:23.26 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 15:24.45 | vader-- | Ok i have something that they are asking for on this phone system but im not sure how to do |
| 15:24.54 | vader-- | im using the cisco 7940G phones |
| 15:25.21 | vader-- | right now we have an old avaya system where the phones can show you when someone is on the phone |
| 15:25.25 | [TK]D-Fender | vader-- : Meh |
| 15:25.38 | [TK]D-Fender | GyrosGeier : Big ripoff where you are... |
| 15:25.42 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:25.45 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 15:26.34 | [TK]D-Fender | vader-- : No idea how to show presence on those in a normal way. You could always amke a "Services" XML browser page that polls AMI for it. |
| 15:27.05 | vader-- | they want some way on the new phones for the secretaries to see if their boss is on the phone |
| 15:28.11 | vader-- | hmmm interesting |
| 15:28.12 | tzanger | AMI? |
| 15:28.23 | tzanger | asterisk management interface? |
| 15:28.42 | [TK]D-Fender | tzafrir : Asterisk Manager Interface. I wrote a script to do that for my Polycom's in addition to using Buddy Watch for presence. |
| 15:29.28 | [TK]D-Fender | tzafrir : That way it bypasses the BW limit and also includes names and other info if I so choose (like VM count, etc) |
| 15:30.07 | *** part/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 15:31.11 | rob0 | Woohoo! |
| 15:31.36 | *** join/#asterisk Arno[Slack] (n=arnaud@gra94-6-82-229-221-134.fbx.proxad.net) |
| 15:31.41 | rob0 | (too early here for beer) |
| 15:33.30 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
| 15:33.44 | Arno[Slack] | is there any Asterisk's gurus to give me their trick about how the manage load balancing with Asterisk please ? |
| 15:33.56 | Arno[Slack] | *how they |
| 15:34.01 | file | it's all smoke and mirrors |
| 15:34.53 | jbalcomb | Arno[Slack]: two servers with an IAX link? |
| 15:35.07 | Arno[Slack] | yes or SIP |
| 15:35.25 | Arno[Slack] | but IAX is ok |
| 15:35.40 | jbalcomb | Arno[Slack]: I don't know of a load balancing feature though, i think you just manually pick which lines and extensions get handled by which server |
| 15:36.09 | Arno[Slack] | hum... no : it need to be able to evolve |
| 15:36.31 | Arno[Slack] | I can use dundee instead of manually did it ;) |
| 15:36.53 | Arno[Slack] | I wondered if somebody used a true redundancy and load balancing soluton |
| 15:36.59 | jbalcomb | Arno[Slack]: IAX is /ok/? ... Inter-Asterisk eXchange protocol.. |
| 15:37.01 | Arno[Slack] | *solution |
| 15:37.29 | jbalcomb | did you look on the wiki and google yet? |
| 15:37.41 | [TK]D-Fender | Arno[Slack] : SER <- |
| 15:37.46 | Arno[Slack] | I meant : I have no restriction on that IAX or SIP I don't really care, I prefer IAX because indeed it's more accurate and powerfull |
| 15:37.51 | rob0 | mog: Give yourself a raise. |
| 15:38.35 | Arno[Slack] | [TK]D-Fender: yes I saw SER, are you using it ? |
| 15:38.49 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.146) |
| 15:39.08 | Arno[Slack] | jbalcomb: but.. of course I do not ask without searching |
| 15:39.11 | mog | ? |
| 15:39.23 | rob0 | You deserve it, you know you do. :) |
| 15:39.46 | Arno[Slack] | I wanted to know if someone have already used some solutions like that and have some tricks |
| 15:40.01 | jbalcomb | [TK]D-Fender: where to d/l the 1.6.6 firmware and configs? |
| 15:40.13 | Arno[Slack] | like : "do not use that it's a real pain in te butt" or something like that ;) |
| 15:40.45 | *** join/#asterisk MonkeyHugs (n=MonkeyHu@63.149.122.94) |
| 15:41.40 | Arno[Slack] | and HSRP ? |
| 15:44.10 | *** join/#asterisk s0lid (n=s0lid@124.107.26.34) |
| 15:46.20 | LoneShadow | asterisk is exiting with 01 code when its parsing modules.conf |
| 15:47.03 | LoneShadow | this is a fresh install, anything I can do to enable more debug prints ? |
| 15:47.14 | LoneShadow | I tried gdb asterisk, and run -vvvgc |
| 15:48.35 | rob0 | gdb is useless for configuration errors. |
| 15:48.44 | rob0 | what module is failing? |
| 15:49.03 | LoneShadow | it dosnt give any failure messages, just exits |
| 15:49.03 | rob0 | maybe the modules.conf file format is wrong? |
| 15:49.16 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:49.35 | LoneShadow | I thought that would be the problem, copied all the /etc/asterisk/* from my working machine to this |
| 15:49.45 | LoneShadow | still fails at same place |
| 15:50.11 | LoneShadow | actually, it was trying to load format_mp3.so I think |
| 15:50.19 | LoneShadow | <PROTECTED> |
| 15:50.20 | LoneShadow | Program exited with code 01. |
| 15:50.20 | brad_mssw | wow, asterisk 1.2.9.1 just totally barfed on us in production ... it was still running, but was in some sort of crazy ring loop, no one could answer calls etc, had to forcibly restart asterisk |
| 15:50.34 | brad_mssw | everything is sip here |
| 15:50.39 | MonkeyHugs | Anyone know why conference participants would be able to dial into a conference bridge, but not be able to hear each other? |
| 15:50.39 | brad_mssw | including incoming |
| 15:51.19 | brad_mssw | MonkeyHugs: timing issues maybe (got ztdummy or a real zap card?) |
| 15:51.33 | jarrod | hey, what are some home dsl routers that will support multiple sip phones behind them (polycom) and terminate to a SER box running rtpproxy/nathelper properly |
| 15:51.51 | xheliox | Is there a list of officially supported server hardware by Digium? |
| 15:51.51 | jarrod | will the newer linksys dsl/cable routers work ? |
| 15:52.29 | file | xheliox: like complete systems? |
| 15:53.00 | RoyK[at] | LoneShadow: just asterisk -gvvvvvc |
| 15:53.00 | RoyK[at] | and pastebin the output |
| 15:53.01 | RoyK[at] | ~pb |
| 15:53.02 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 15:53.31 | kay2 | Someone here uses MeetMe() ? |
| 15:53.35 | LoneShadow | RoyK[at]: I just did noload for format_mp3.so, and it loaded fine |
| 15:53.36 | xheliox | file: Like specifically the HP DL360. :) I just want to be able to tell someone 100% that the TE210P will work in it. |
| 15:53.45 | jbalcomb | [TK]D-Fender: Where do you get the polycom firmware and configs? |
| 15:53.54 | jarrod | jbalcomb: from your reseller |
| 15:54.06 | file | xheliox: http://www.digium.com/en/docs/misc/compatibility_notes.php that's all I know about |
| 15:54.09 | file | unless... |
| 15:54.19 | MonkeyHugs | Brad_mssw: there is a TE410P installed in the server, but I will look into the T1 config to make sure timing is not an issue. Thank you for your suggestion. |
| 15:54.48 | xheliox | I thought there was a list of servers Digium had "certified". |
| 15:54.57 | xheliox | Maybe it's the incompatible list I'm thinking of. :) |
| 15:54.58 | file | I know there's one for business edition |
| 15:55.25 | jarrod | i know the dell 1850 works well with all the versions of the digium 3.3v cards |
| 15:55.32 | file | you said... DL360? |
| 15:55.42 | MonkeyHugs | Kay2: is there an alternative to MeetMe()? :) |
| 15:55.50 | file | it's on the list for BE compatibility so it was tested with it... so it should be fine |
| 15:55.59 | xheliox | I'll take a gamble on if it works for business edition, it works for the community edition. |
| 15:56.05 | LoneShadow | any idea why format_mp3.so might fail ? |
| 15:56.08 | xheliox | Thanks. :) Is that business list public? |
| 15:56.16 | file | http://www.digium.com/en/docs/ABE/abe_compatibility.php |
| 15:56.17 | *** join/#asterisk OdysseyVoiceSolu (n=OdysseyV@24-48-145-188.atlsfl.adelphia.net) |
| 15:56.44 | xheliox | Gracias. :) |
| 15:56.51 | OdysseyVoiceSolu | hello |
| 15:57.05 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 15:57.09 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 15:57.16 | Qwell | murf: hey |
| 15:57.28 | OdysseyVoiceSolu | i am in need of assistance with my dial plan |
| 15:57.31 | file | woot murf |
| 15:57.46 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 15:57.46 | murf | Qwell: howdy! |
| 15:57.50 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 15:57.53 | FlatFoot | hello all |
| 15:58.04 | Qwell | murf: mind a quick msg? |
| 15:58.18 | OdysseyVoiceSolu | ? |
| 15:58.22 | FlatFoot | odd question can't find a good answer anywhere can someone help ? |
| 15:58.23 | murf | Qwell: Go for it |
| 15:58.35 | RoyK[at] | LoneShadow: bingo |
| 15:58.36 | RoyK[at] | LoneShadow: format_mp3 isn't really something you'll need anyway. prolly some old file. rebuild -addons |
| 15:58.44 | RoyK[at] | xheliox: the incompatible list is just there since the digium hardware isn't good enough :P |
| 15:58.46 | RoyK[at] | LoneShadow: rebuild -addons |
| 15:58.52 | LoneShadow | hmm |
| 15:59.12 | LoneShadow | so is format_mp3 required only to mp3 music on hold ? |
| 15:59.29 | LoneShadow | play* |
| 15:59.37 | xheliox | RoyK[at]: If I had purchased these cards, they'd be another brand. |
| 15:59.37 | OdysseyVoiceSolu | i need to send an annoucement to the person being called before they are able to talk, does anyone know how to go about that? |
| 15:59.51 | FlatFoot | need to direct outgoing dependant on my cli to a different real number that is bound to my * box |
| 15:59.59 | FlatFoot | not quite sure if that makes sense |
| 16:01.21 | file | getting 100% compatibility is hard, but we try |
| 16:03.49 | nettie | e' free? |
| 16:03.52 | nettie | whoops |
| 16:03.53 | nettie | :) |
| 16:03.58 | MonkeyHugs | Brad_mssw: Loaded ZTDummy and people are able to hear eachother now in the conference bridge. Thanks Brad! |
| 16:05.05 | *** join/#asterisk quadrata (n=spork@dynamic-64-115-24-203.isp.broadviewnet.net) |
| 16:05.09 | brad_mssw | ... ahh, always nice when teliax goes down |
| 16:05.36 | brad_mssw | glad our numbers will be ported away from them in 2 days |
| 16:05.55 | file | yeekz even their site is dead to me |
| 16:06.06 | Spy000007 | Always nice to come back from vacation to an outage... Thanks Teliax! |
| 16:06.34 | brad_mssw | it's amazing, they have how many servers, yet they're always going down |
| 16:06.57 | *** join/#asterisk watchy (n=watchy@office2.gwhsi.com) |
| 16:07.29 | watchy | i have a network issue. when i trace route an ip on my network the last hop responds twice. anyone seen this? |
| 16:07.38 | file | xheliox: according to a wildly cool Matt in support, the DL360 is one of the recommended servers :D |
| 16:07.59 | brad_mssw | i think a lot of it is their ISP's fault (rockynet), but they have servers in other locations, so they should be fault-tolerant for origination and termination ... |
| 16:08.09 | brad_mssw | wow, they're back up |
| 16:08.17 | Spy000007 | Certainly is rocky... |
| 16:08.31 | file | Spy000007: that was too easy |
| 16:08.33 | xheliox | file: Thanks. :) I generally don't have problems with Digium cards working, but when I do, it's a frickin nightmare. This can't end up being a nightmare, not this time. :) |
| 16:08.43 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 16:08.50 | Spy000007 | Probably had to reboot their Windows XP Microsoft Access database holding customer login info... |
| 16:11.19 | rob0 | "Append ANI2" an option in Asterlink's control panel, would that effect my caller ID? Could I just use SetCallerID() in my dialplan? |
| 16:11.38 | file | rob0: that sends ANI2 on incoming calls, appends it to the end - two digits |
| 16:11.49 | rob0 | SetCallerID("G W Bush" <202-xxx-xxx>) |
| 16:12.28 | rob0 | file: did I sign up for the wrong plan? I can't figure out how to get inbound SIP working. |
| 16:12.37 | rob0 | (I signed up for Extreme.) |
| 16:13.03 | file | did you read the instructions in the email? |
| 16:13.13 | file | you have to setup in the control panel how to route incoming calls if you're on extreme |
| 16:14.24 | RoyK[at] | xheliox: i don't think te410p exists in any other brand, but then, you can always get Sangomas |
| 16:14.41 | rob0 | HAHA ... the email was BLOCKED |
| 16:14.46 | RoyK[at] | does anyone know if there are any works to get native sangoma support for asterisk? |
| 16:14.49 | RoyK[at] | without zaptel? |
| 16:15.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:15.25 | rob0 | Brian replied to me, but I still don't have the welcome mail. |
| 16:19.48 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:22.16 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
| 16:22.40 | murf | file: a long paused woot to you, too! ;^) |
| 16:23.09 | kay2 | MonkeyHugs: with what version of asterisk do you use MeetMe() |
| 16:23.21 | RoyK[at] | rob0: show function CALLERID |
| 16:25.35 | Persilon | HI |
| 16:25.44 | Persilon | I need some help with rxfax and txfax |
| 16:26.28 | MonkeyHugs | Kay2: 1.2.9.1 |
| 16:26.47 | Persilon | they don't seem to give the right tones (I'm calling from a fax-modem) and it reachs absolutetime without sending or recieving anything |
| 16:27.25 | murf | Hey guys, I'm trying to build a conferencing box, need a web gui for setting up/controlling/billing. Was going to use GDS s/w, but they put it on hold. Any other options anyone knows of? |
| 16:28.13 | MonkeyHugs | Murf: there is Druid and astUNI |
| 16:28.57 | MonkeyHugs | astUNI is more robust but more involved to setup |
| 16:31.35 | murf | MonkeyHugs: where do you get astUNI? Google just failed. |
| 16:31.38 | MonkeyHugs | Murf: make the astGUIclient |
| 16:31.51 | MonkeyHugs | make that astGUIclient |
| 16:32.08 | murf | OK. and Druid? |
| 16:32.12 | MonkeyHugs | Yup |
| 16:32.33 | MonkeyHugs | I beleive the full version of Druid cost about $80 |
| 16:32.57 | MonkeyHugs | astGUIclient is free |
| 16:33.30 | CunningPike | In a similar vein, what are people using for ACD reporting? |
| 16:34.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:36.07 | kay2 | MonkeyHugs: well it's weird, when I do a MeetMe(|Md) is says conference number 0 is not valid |
| 16:36.21 | *** join/#asterisk s0lid (n=s0lid@61.28.161.132) |
| 16:36.24 | rob0 | file: please oh please oh please can I get the welcome email? I promise I won't block it. |
| 16:36.26 | kay2 | MonkeyHugs: how do you create a conferance room dynamically on the fly ? |
| 16:36.42 | file | rob0: I don't work at Asterlink anymore lol |
| 16:36.47 | rob0 | oh no!! |
| 16:36.53 | rob0 | nm then ;) |
| 16:36.54 | file | it's possible to get it resent though |
| 16:37.01 | file | someone just has to queue it up |
| 16:37.12 | *** part/#asterisk jaike (n=a@203.115.188.120) |
| 16:37.44 | rob0 | I sent an email asking them to do so. Brian replied to part of it. |
| 16:38.52 | rob0 | Where are you working now? |
| 16:39.12 | file | Digium |
| 16:39.23 | rob0 | cool! Telecommuting, I guess? |
| 16:39.28 | file | yessir |
| 16:39.42 | rob0 | congrats |
| 16:39.51 | file | thankies |
| 16:40.05 | rob0 | I might apply there too. Not sure if they need me, tho. I'm not a * expert yet. |
| 16:40.30 | MonkeyHugs | Kay2: pastebin your meetme.conf file |
| 16:41.20 | MonkeyHugs | kay2 Your syntac looks correct. Try setting your conf room to 1000 and see if you get the same result. |
| 16:45.49 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:46.15 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:46.24 | kay2 | MonkeyHugs: I don't wanna have anything in meetme.conf |
| 16:46.33 | kay2 | MonkeyHugs: I just want the room to be created dynamically |
| 16:46.57 | kay2 | MonkeyHugs: Meetme(1000|Md) does the same thing |
| 16:47.20 | kay2 | MonkeyHugs: in meetme.conf everything is commented tho |
| 16:48.57 | Dr-Linux | again, anybody familiar with SCCP? |
| 16:49.27 | [TK]D-Fender | Dr-Linux : If Qwell couldn't help you, you're d00med |
| 16:50.24 | Dr-Linux | [TK]D-Fender: i have only problem remains, my cisco phone doesn't hangup |
| 16:51.14 | [TK]D-Fender | Dr-Linux : tried changing the firmware on it? |
| 16:51.32 | Dr-Linux | [TK]D-Fender: the firmware is 3.1 the latest one |
| 16:51.48 | Dr-Linux | [TK]D-Fender: i'm asking, bcoz i think it's not phone problem .. |
| 16:52.17 | Dr-Linux | [TK]D-Fender: look here >> http://pastebin.ca/78962 |
| 16:52.39 | Dr-Linux | only phone is still connected to the asterisk server |
| 16:53.31 | [TK]D-Fender | Dr-Linux : tried pickup up the handset and hanging it up again? |
| 16:54.25 | tzanger | holy shit |
| 16:54.28 | tzanger | holy shit |
| 16:54.28 | Dr-Linux | [TK]D-Fender: cisco 7935 is a conference device, it has no handset |
| 16:54.40 | tzanger | I got this BT module working with uClinux |
| 16:57.18 | [TK]D-Fender | Dr-Linux : Yeah, well is has a way to go offhook to dial... so go figure it out. |
| 16:57.57 | *** join/#asterisk Tier_1 (n=tier@c-67-176-28-65.hsd1.co.comcast.net) |
| 16:59.17 | Dr-Linux | [TK]D-Fender: there are many things in the sccp.conf, but i can't understand all, and can't find help on google |
| 16:59.39 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 17:02.10 | Dr-Linux | [TK]D-Fender: in my sccp.conf i can see for offhook: |
| 17:02.11 | Dr-Linux | ;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none. |
| 17:02.37 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 17:03.34 | RoyK[at] | Dr-Linux: noload => chan_skinny.so |
| 17:03.35 | RoyK[at] | :) |
| 17:04.09 | Dr-Linux | RoyK[at]: that's already done. chan_skinny is not loaded |
| 17:04.34 | vivek | hello all what's the best ata for iax2 which has fxo and fxs ports and supports lots of codecs especially globalip sound and other low bandwidth codecs ... ? i have a spa3k but I don't think it will support ip sound (dosn't look like they will update their firmware ... |
| 17:04.50 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 17:05.20 | smackus | hi all |
| 17:05.26 | smackus | is there a cleaner way to do this? http://pastebin.ca/79720 |
| 17:05.54 | RoyK[at] | Dr-Linux: how do you expect to make sccp work without chan_skinny? skinny is the sccp module :) |
| 17:06.34 | Dr-Linux | RoyK[at]: you are wrong |
| 17:06.44 | RoyK[at] | ~sccp |
| 17:06.46 | jbot | somebody said sccp was Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 17:06.53 | RoyK[at] | ~skinny |
| 17:06.55 | jbot | skinny is, like, a common name for SCCP, the VoIP protocol used by many Cisco phones, or what people look like when they put computing above eating |
| 17:07.20 | RoyK[at] | ~royk |
| 17:07.21 | jbot | [royk] that viking asterisk guru, or your friend |
| 17:07.30 | vivek | iax2 is better than sip right ? |
| 17:07.31 | [TK]D-Fender | ~[TK]D-Fender |
| 17:07.33 | jbot | you are probably rockin' the casbah !!! |
| 17:07.39 | [TK]D-Fender | vivek : Depends for wht. |
| 17:08.22 | vivek | [TK]D-Fender: hmmz i need to cross firewalls ... so i suppose that means yes to AIX2 and no to SIP |
| 17:08.25 | Dr-Linux | ##linux guys didn't answer, for my question |
| 17:08.38 | Dr-Linux | how can i check my currentl runleve and runlevel history? |
| 17:08.42 | vivek | Dr-Linux: what was your question ? |
| 17:08.55 | RoyK[at] | Dr-Linux: runlevel |
| 17:09.08 | vivek | can i get sip working with just two ports that are open ? |
| 17:09.11 | vivek | 80 and 81 ? |
| 17:09.12 | Dr-Linux | vivek: runlevel history |
| 17:09.30 | RoyK[at] | Dr-Linux: runlevel |
| 17:09.35 | vivek | Dr-Linux: YOU WANT TO GO BACK UP YOUR RUNLEVEL HISTROY ? |
| 17:09.44 | RoyK[at] | vivek: on tape |
| 17:09.48 | vivek | or terminal commands ? |
| 17:10.00 | RoyK[at] | Dr-Linux: init 6 might help |
| 17:10.03 | dasenjo | Hi, why am I getting "Function TIMEOUT not registered |
| 17:10.03 | dasenjo | " error on an * 1.2.7.1? |
| 17:10.06 | vivek | sorry about the caps ... pressed the wrong keys |
| 17:10.18 | Dr-Linux | vivek: terminal command that show me my runlevel history, |
| 17:10.26 | RoyK[at] | Dr-Linux: runlevel |
| 17:10.28 | RoyK[at] | Dr-Linux: runlevel! |
| 17:10.32 | RoyK[at] | ~lart Dr-Linux |
| 17:10.39 | Dr-Linux | vivek: actually my box is getting auto reboot, |
| 17:10.47 | RoyK[at] | ROTFL |
| 17:10.51 | vivek | init 2 ;) |
| 17:10.52 | Dr-Linux | RoyK[at]: that shows only current runlevel :) |
| 17:10.56 | vivek | init 0 |
| 17:10.58 | vivek | shutdown 0 |
| 17:11.09 | Dr-Linux | vivek: i know all that |
| 17:11.12 | vivek | lol or just take a hammer and break your hdd ;) |
| 17:11.14 | Dr-Linux | but i wanna know history |
| 17:11.20 | RoyK[at] | Dr-Linux: runlevel shows current and the last one |
| 17:11.24 | vivek | it must be in some log ... |
| 17:11.43 | RoyK[at] | Dr-Linux: and init 6 reboots the machine, of course |
| 17:11.57 | Dr-Linux | yeah, i know that |
| 17:12.36 | Dr-Linux | RoyK[at]: my network guys told me that, machine is getting auto reboot, bcoz this server is changing it's runleve auto .. |
| 17:12.55 | RoyK[at] | then kill cron and at and go through the config |
| 17:12.57 | Dr-Linux | i think he is wrong, so i just wanna verify it to see runlevel history |
| 17:13.01 | RoyK[at] | the runlevel should NEVER change |
| 17:13.21 | Dr-Linux | RoyK[at]: not using cron though |
| 17:13.28 | vivek | that's strange ... |
| 17:13.53 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 17:13.56 | Dr-Linux | look here his reply: |
| 17:13.57 | Dr-Linux | Thank you for your request. |
| 17:13.57 | Dr-Linux | I can see from the logs that this server is changing runlevels automatically, |
| 17:14.25 | dasenjo | there are a few functions in my *, why? |
| 17:14.42 | Dr-Linux | hhm.. |
| 17:14.54 | Dr-Linux | i think no one every use chan_SCCP :) |
| 17:16.14 | [TK]D-Fender | Dr-Linux : People choose * to be able to use COMMODITY equipment with OPEN standards. Skinny is anything but. |
| 17:17.27 | Dr-Linux | hhm.. |
| 17:17.45 | Dr-Linux | [TK]D-Fender: would you like to have a look on my sccp.conf ? |
| 17:18.03 | dasenjo | what did I do? why don't you help .. |
| 17:18.23 | dasenjo | you don't even asnwer or kid me .. :p |
| 17:18.37 | CunningPike | dasenjo: Ask a question that makes sense :) |
| 17:19.39 | dasenjo | I need to get absolutetimeout working on my server .. but I got an error saying the TIMEOUT function is not registered |
| 17:20.02 | CunningPike | dasenjo: So, register it. Look in your modules.conf |
| 17:20.08 | dasenjo | there is no func_timeout.so or something like that .. how can I register it? |
| 17:21.02 | *** join/#asterisk negativecreep (n=xaeem@202.147.167.204) |
| 17:21.02 | dasenjo | there is no func_* on my modules.conf even |
| 17:21.09 | CunningPike | dasenjo: You can load it from the CLI using load func_timeout.so, but you should also add a load => line in your modules.conf |
| 17:22.13 | CunningPike | dasenjo: Here is some good modules.conf information: http://www.voip-info.org/wiki/view/Asterisk+Slimming |
| 17:23.18 | *** join/#asterisk my_name (n=juxhin@nathan.epi.usf.edu) |
| 17:23.28 | my_name | hello |
| 17:24.11 | negativecreep | hi guys |
| 17:24.15 | my_name | i just installed asterisk on a sun system with a configured dialogic card in it. |
| 17:24.18 | *** join/#asterisk R-MAN (n=raficmas@i-195-137-114-169.freedom2surf.net) |
| 17:24.24 | negativecreep | i am having lots of noise on iax2 calls while sip calls have next to none. |
| 17:24.39 | my_name | after trying to run it i get a few errors |
| 17:24.39 | R-MAN | hey guys |
| 17:24.56 | my_name | error number one is res_odbc.c:479 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 17:25.00 | negativecreep | any ideas? |
| 17:25.12 | [TK]D-Fender | Dr-Linux : I don't touch Skinny, and I've only USED a Cisco phone ONCE. |
| 17:25.13 | Hmmhesays | what endpoints are you using negativecreep |
| 17:25.22 | Hmmhesays | I use crisco all the time, but sip |
| 17:25.32 | [TK]D-Fender | Dr-Linux : You've really got to get that boss of yours from buying shit without going through testing first. |
| 17:25.34 | R-MAN | Has anyone delt with im sure the famous issue of getting choppy or robtic sound when you make a outbound call? |
| 17:25.52 | [TK]D-Fender | my_name : Masochist.... |
| 17:26.24 | my_name | [TK]D-Fender, i am not sure why you say that |
| 17:26.32 | my_name | please explain |
| 17:26.38 | Dr-Linux | [TK]D-Fender: it's okey, no problem, everything is very fine to me, only hangup problem. |
| 17:26.41 | negativecreep | Hmmhesays: endpoints? |
| 17:26.52 | Dr-Linux | [TK]D-Fender: maybe that's due to dialplan? :S |
| 17:26.56 | Hmmhesays | what are you using to send your calls |
| 17:27.01 | [TK]D-Fender | my_name : Picking all the hardest tech's and platforms to get running all at once. |
| 17:27.23 | [TK]D-Fender | Dr-Linux : Dialplan doesn't make a phone hang up. A phone hanging up makes it hang up. |
| 17:27.24 | negativecreep | dsl on both ends if u r talking about internet connectivity..delay is like 70ms |
| 17:27.48 | negativecreep | i am using iaxcomm to make iax calls. |
| 17:27.53 | negativecreep | asterisk 1.2.6 |
| 17:28.51 | Dr-Linux | [TK]D-Fender: this phone doesn't have it own functions, all it works from the sccp.conf |
| 17:28.59 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 17:29.27 | [TK]D-Fender | Dr-Linux : Well hanging up isn't a CONFIG option. |
| 17:29.27 | Dr-Linux | [TK]D-Fender: something wrong with line1, 2 and .... something wrong with lines stuff :S |
| 17:29.43 | popvoxdave | I need to detect a busy signal on an outbound SIP call placed through a SIP termination provider. |
| 17:29.52 | my_name | [TK]D-Fender, the platform doesn't seem to have any problem with the card. because it pics up calls with no problem. |
| 17:29.59 | popvoxdave | I am just getting a 183 back and the early media is the busy tone. |
| 17:30.21 | Dr-Linux | [TK]D-Fender: in first two tries it provides me "call end" option, but third time it doesn't :S |
| 17:30.34 | [TK]D-Fender | Dr-Linux : That = BUG |
| 17:30.35 | popvoxdave | There is no indication in the signalling that it's busy. Is this a feature that needs to be turned on in the provider's gateways or is there a decent way to detect busy audio in asterisk on a SIP channel? |
| 17:30.48 | [TK]D-Fender | Dr-Linux : change your firmare revision and pray to the God's at Cisco. |
| 17:31.04 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:31.06 | dasenjo | CunningPike, as said, there is no func_timeout.so at all |
| 17:31.13 | rob0 | Eh, Cisco? Eh, Pancho! |
| 17:31.16 | [TK]D-Fender | popvoxdave : Its supposed to com from the provider... |
| 17:31.46 | Dr-Linux | [TK]D-Fender: you think it's firmware problem, not configs? |
| 17:33.38 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 17:33.49 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 17:33.53 | *** join/#asterisk ToyMan (n=stuq@74-32-56-214.dsl1.mdl.ny.frontiernet.net) |
| 17:34.02 | The_X | Hi folks, anyone got MWI + asterisk + 79** phones working? |
| 17:34.11 | *** join/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 17:34.18 | phigwork | i'm having a problem getting this zap wildcard tdm400p working |
| 17:34.20 | Hmmhesays | not in skinny |
| 17:34.27 | Hmmhesays | in sip |
| 17:34.31 | phigwork | asterisk quits saying unable to specify channel 1, unable to open channel 1, etc |
| 17:34.31 | CunningPike | dasenjo: What was the error message again? |
| 17:34.39 | Paavum | Hello. Anybody working with Druid? |
| 17:34.50 | Hmmhesays | The_X: what protocol are you using? |
| 17:34.51 | Paavum | I cant install the damned demo thingy |
| 17:34.56 | The_X | sip |
| 17:34.58 | phigwork | no such device or address |
| 17:34.58 | dasenjo | CunningPike, function TIMEOUT not registered |
| 17:35.27 | CunningPike | dasenjo: And what's the line in your dialplan that generates the error? |
| 17:35.43 | The_X | running 1.2.7.1 |
| 17:35.44 | dasenjo | CunningPike, after "CLI> load func_timeout.so" /usr/lib/asterisk/modules/func_timeout.so: cannot open shared object file: No such file or directory |
| 17:35.51 | Hmmhesays | The_X: mailbox=mailboxnumber@context |
| 17:36.00 | [TK]D-Fender | Dr-Linux : if it works 2 times, and on the third it fails.. use your imagination.... |
| 17:36.52 | CunningPike | dasenjo: But what is the line in your dialplan that gave the original error in the first place? |
| 17:36.54 | dasenjo | CunningPike, Set("Zap/1-1", "TIMEOUT(absolute)=0") |
| 17:37.03 | Hmmhesays | The_X: do what I said or give me access and paypal me a 20 |
| 17:37.06 | The_X | I just want the freaking cisco phone light to blink |
| 17:37.08 | The_X | that's about it |
| 17:37.18 | Dr-Linux | [TK]D-Fender: that's what i'm thinking .. but can't find out.. and that's why i think there is something in sccp.conf :S |
| 17:37.18 | The_X | Hmmhesays, sure ;) |
| 17:37.22 | dasenjo | as appears in the cli |
| 17:37.25 | phigwork | the_x: which cisco phone? |
| 17:37.28 | The_X | 7960s |
| 17:37.29 | phigwork | out of curiosity |
| 17:37.55 | [TK]D-Fender | Dr-Linux : if it works twice, THEN fails, its clearly a BUG in the firmware otherwise why would it have worked at all? |
| 17:38.03 | Dr-Linux | [TK]D-Fender: i just made a call to one of my SIP user at USA and that went very good and i had an option "endcall" and that works |
| 17:38.09 | phigwork | i want to get the 7910 working |
| 17:38.17 | phigwork | but i think it's going to be a pain |
| 17:38.25 | Dr-Linux | [TK]D-Fender: hhm... yeah, maybe you are right |
| 17:38.47 | CunningPike | No maybe about it - [TK]D-Fender is always right |
| 17:38.59 | Dr-Linux | [TK]D-Fender: btw, i'm using Cisco's defulat firmware, i just loaded .xml file to make it connect to my asterisk box. |
| 17:39.56 | dasenjo | CunningPike, this is the correct way to set absolute timeout on 1.2.7.1 right? |
| 17:40.14 | [TK]D-Fender | <- not always right, just loaded with common sense (a rare trait actually), a lot of intuition, and tries hard :) |
| 17:40.27 | vivek | Dr-Linux: lol you could make a call to India and ty if you want ;) i have a sip phone and ata lol |
| 17:40.31 | CunningPike | dasenjo: Please provide the line in your dialplan, not the output from your CLI |
| 17:40.36 | vivek | and its connected ... |
| 17:40.39 | The_X | there you got |
| 17:40.42 | The_X | go |
| 17:40.44 | The_X | thanks folks |
| 17:41.12 | vivek | er can i somehow use spa 3000 to place a aix2 call ? |
| 17:41.17 | Hmmhesays | no |
| 17:41.31 | [TK]D-Fender | vivek : No. It doesn't do IAX2. |
| 17:41.33 | Dr-Linux | vivek: chal bay apna kaam kar :P |
| 17:41.53 | vivek | Dr-Linux: lol kya yaar tu bhi india mein ho ? |
| 17:42.11 | Dr-Linux | vivek: heh nahin yaar, i'm not |
| 17:42.15 | dasenjo | CunningPike, exten=s,2,Set(TIMEOUT(absolute)=0) |
| 17:42.36 | Dr-Linux | vivek: i'm from your enemy count... :P |
| 17:42.44 | vivek | Dr-Linux: lol ok |
| 17:43.09 | vivek | we don't have enemies ... atleast I don't care ... |
| 17:43.21 | Dr-Linux | vivek: i have lot of sip users and servers, that's not a problem :) |
| 17:43.26 | [TK]D-Fender | 1 well placed nuke can change that quick ;) |
| 17:43.34 | CunningPike | dasenjo: Looks good - I'm just trying to remember which module provides TIMEOUT()...... |
| 17:43.47 | *** part/#asterisk netoguy (n=skelley@64-199-141-122.ip.mcleodusa.net) |
| 17:44.03 | vivek | [TK]D-Fender: really i don't see how ... i would be dead before i know it ;) |
| 17:44.11 | Paavum | Hello. Is anybody working with Druid? |
| 17:44.44 | vivek | besides i live near an airforce base so its probably a sure target ;) haha |
| 17:45.05 | Dr-Linux | <PROTECTED> |
| 17:45.05 | Dr-Linux | <PROTECTED> |
| 17:45.31 | *** join/#asterisk pdthome (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 17:46.00 | [TK]D-Fender | CunningPike : pbx_functions.so I believe |
| 17:46.04 | trelane_ | can I pass an argument of some sort to the zap modules and try to steer IRQ's when loading the module |
| 17:46.32 | CunningPike | [TK]D-Fender: Ah - thanks |
| 17:47.03 | [TK]D-Fender | vivek : not true.... major civilian target first to demoralize the enemy, THEN major military installations. If you go for military first they have more time to react since thats where the defenses are strongest. |
| 17:47.08 | CunningPike | dasenjo: pbx_functions.so, apparently |
| 17:47.56 | vivek | [TK]D-Fender: in a nuclear war if you take out airforce bases and cities that's the end of it ... |
| 17:47.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net) |
| 17:48.37 | Hmmhesays | anyone ever configured a dialpeer in an as5300? |
| 17:49.51 | *** join/#asterisk BugKham (i=CKGLOB@125.24.3.20) |
| 17:50.13 | BugKham | anyone using madplay for moh at all? |
| 17:50.19 | rob0 | WAR!! |
| 17:50.28 | RoyK[at] | ~lart rob0 |
| 17:50.31 | rob0 | ouch |
| 17:50.38 | rob0 | that was uncalled for. |
| 17:50.46 | Hmmhesays | dude you so yelled WAR |
| 17:50.46 | generalhan | Hmmhesays: im having the same issue as The_X ... my MWI doesnt come on at all on some phones and stays on, on others ... i went through and put mailbox=XXXX@default in the sip.conf for half of my entries and still after reboot none of the cisco ones are working ?? |
| 17:50.57 | trelane_ | Hmmhesays, yeah, what is it good for? |
| 17:51.09 | dasenjo | CunningPike, thanks a lot!! |
| 17:51.15 | rob0 | Well I expected nastiness, but not LAWYERS, ugh. |
| 17:51.24 | CunningPike | dasenjo: Don't thank me, thank [TK]D-Fender |
| 17:51.42 | dasenjo | pbx_functions is the correct module |
| 17:51.54 | Hmmhesays | generalhan cisco's? |
| 17:51.57 | dasenjo | [TK]D-Fender, thank you |
| 17:52.02 | generalhan | 7960s |
| 17:52.07 | [TK]D-Fender | ywc |
| 17:52.18 | Hmmhesays | what f/w? |
| 17:52.24 | *** join/#asterisk ToyMan (n=stuq@74-32-56-214.dsl1.mdl.ny.frontiernet.net) |
| 17:52.25 | generalhan | you just helped The_X with this same issue ... ive been dealing with it for a while and changing the mailbox line doesnt help me at all |
| 17:52.31 | generalhan | 8-3-00 |
| 17:52.47 | phigwork | i'm trying to install a tdm400p card |
| 17:52.50 | phigwork | getting chan_zap.c: Unable to open channel 1: No such device or address |
| 17:52.58 | Hmmhesays | I'd have to log in and poke around |
| 17:53.01 | phigwork | what am I forgetting? |
| 17:53.21 | rob0 | phigwork: ztcfg -vv |
| 17:54.28 | rob0 | phigwork: Likely a misconfigured zaptel.conf |
| 17:54.33 | phigwork | hm |
| 17:54.44 | phigwork | it says "did you forget that FXS channels use FXO signalling" etc |
| 17:55.08 | phigwork | so does that mean they should be backward in /etc/zaptel.conf or in /etc/asterisk/confs? |
| 17:55.10 | phigwork | or both? |
| 17:56.13 | rob0 | well I have fxoks=1 and fxsks=1 where channel 1 is my FXS and 5 is my FXO |
| 17:56.43 | rob0 | (The FXS is on a TDM card with 4 slots for modules.) |
| 17:56.53 | phigwork | you have both =1? |
| 17:57.08 | rob0 | ooops no |
| 17:57.21 | rob0 | fxsks=5, sorry |
| 17:58.02 | Hmmhesays | generalhan: that is kind of odd that they are staying ON |
| 17:58.10 | rob0 | Channel 01: FXO Kewlstart (Default) (Slaves: 01)\nChannel 05: FXS Kewlstart (Default) (Slaves: 05) |
| 17:59.37 | rob0 | genzaptelconf is your friend |
| 18:00.09 | rob0 | I think [TK]D-Fender suggested it to me some days ago. [TK]D-Fender is your friend too. :) |
| 18:00.48 | Paavum | Hello. Is anybody working with Druid? |
| 18:01.21 | *** part/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 18:01.35 | [TK]D-Fender | rob0 : Nope, never suggest these "auto" tools. Anything worth doing is worth doing yourself. |
| 18:02.07 | generalhan | Hmmhesays: i know .. and that is the one that im most concerned with too |
| 18:02.27 | generalhan | Hmmhesays: this is just a guy who is leasing space from us and he is non to pleased with the light in his face |
| 18:02.47 | nettie | hey guys anyone know if installing a second asterisk server on the same box is fine? I would like to try the trunk version and slowly migrate to it. I'm woried mostly about paths or other possible issues. I dont have a spare box at the colo and virtualization is not possible. Thanx in advance. |
| 18:03.10 | generalhan | i was thinking about just downgrading his phone only back to 8-2-00 because it didnt happen with that f-w ... the only problem is that i was having issues with the phone locking up durring a transfer with that f-1 |
| 18:03.47 | walhala | is there some french people here ? |
| 18:04.00 | [TK]D-Fender | walhala : Pas de chance... |
| 18:04.16 | walhala | :) |
| 18:04.36 | CunningPike | nettie: I would say that that is a high-risk strategy, at best |
| 18:05.13 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 18:05.15 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:05.18 | Hmmhesays | well I can login and try to fix it, but I ain't doing it for free |
| 18:05.38 | nettie | CunningPike yeah.. maybe the best way is just migrate them considerign I already did some tests and keep stable src handy to replace it if problems happend |
| 18:05.48 | phigwork | chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such device |
| 18:05.49 | phigwork | whoot |
| 18:05.50 | phigwork | :) |
| 18:05.58 | Hmmhesays | in the words of anthm yesterday "i'm a busy sumnabitch" |
| 18:06.22 | generalhan | Hmmhesays: i understand ... my boss is the cheapest person you will EVER know so that wont work ... ill keep messing with it ... i was just looking for ideas |
| 18:06.39 | Hmmhesays | sip debug and watch your notify messages |
| 18:06.49 | Hmmhesays | cheap people are only cheap till customers start getting in their face |
| 18:07.09 | anthm | busy *as* a son-na-ma-bitch |
| 18:07.36 | generalhan | Hmmhesays: is there a way to do a sip debug and save the output somewhere? my CLI scrolls soo fast that i cant make anything from debugs |
| 18:07.53 | Hmmhesays | um, I use putty ssh, 4000 line buffer and logging |
| 18:08.19 | generalhan | if i put verbose to 1 and use a sip debug will that still work ? |
| 18:08.27 | phigwork | am I missing some sound drivers or something? |
| 18:08.36 | generalhan | cause without seeing all the phone calls i could prolly make something of it |
| 18:08.57 | generalhan | but we get about 30 calls every 2-3 minutes so its tough to watch all that ! |
| 18:09.24 | CunningPike | nettie: If you only have one server, you can use some clever symlinks to point /usr/sbin/asterisk at whatever version you want to run at any time - that would make switching even faster |
| 18:09.40 | CunningPike | nettie: But I would seriously caution against running trunk in production |
| 18:10.34 | Qwell[] | CunningPike: sissy |
| 18:10.35 | Qwell[] | :P |
| 18:10.44 | CunningPike | Qwell[]: :P |
| 18:11.51 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:12.02 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.236) |
| 18:12.37 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 18:13.19 | Hmmhesays | generalhahn: up sip debug a single ip |
| 18:16.06 | nettie | CunningPike as far as I can see 1.4 is almost out |
| 18:16.17 | nettie | CunningPike isnt 1.4 based on trunk |
| 18:16.18 | nettie | ? |
| 18:16.26 | Qwell[] | nettie: it is |
| 18:16.45 | nettie | CunningPike this means trunk is supposed to be pretty stable imho |
| 18:16.45 | CunningPike | nettie: Yes - and we'll be upgrading once it's a stable release - it has lots of stuff in it that we want |
| 18:17.08 | nettie | CunningPike I cannot live anymore without jitterbuffer on sip chan |
| 18:17.10 | CunningPike | nettie: It ain't neccessarily so..... |
| 18:17.50 | CunningPike | nettie: We're waiting for called party ID, Polycom ACD functions........ |
| 18:18.32 | nettie | CunningPike ok, those are cool features BUT without jitterbuffer it's not even possible make a decent call! |
| 18:19.19 | CunningPike | nettie: We take the view that we run whatever version we need to get stuff we need working - sounds like you need trunk :) |
| 18:19.41 | nettie | CunningPike yeah .. it's not that I need it to play cards :) |
| 18:19.49 | nettie | CunningPike I'm definitely in a bad position atm, |
| 18:20.09 | CunningPike | nettie: What sort of network do you have that you need jitterbuffer so bad? |
| 18:20.22 | nettie | CunningPike eheh.. I have the offices phones (polycoms) behind DSL and the asterisk server is at the colo |
| 18:20.36 | CunningPike | nettie: Ah - that'll do it |
| 18:21.04 | nettie | CunningPike if the upstream is heavely we stard having problems. |
| 18:21.27 | nettie | CunningPike conidering the upstream is only 256Kb youcan imagine what a couple of meial could do |
| 18:21.30 | *** join/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 18:21.36 | nettie | meial=emails |
| 18:22.08 | CunningPike | nettie: Yuck - if I were you, I'd be running a teeny asterisk box inside your LAN and IAX2 trunking to your colo. |
| 18:22.20 | nettie | CunningPike that's the second otion |
| 18:22.23 | nettie | option |
| 18:22.45 | chandi | Hi, I've got a strange problem. When I make a call from my sipura ATA through my asterisk on the same lan... the sipura won't receive the audio. It's strange since it works on calls that are only from the asterisk to the sip provider. |
| 18:22.46 | nettie | considering I stil have a tftp server for phones provisioning and stuff. |
| 18:23.04 | nettie | CunningPike jitterbuffer on IAX2 is great? |
| 18:23.08 | chandi | One more: when the callee's line is busy it does ring normally on the sipura |
| 18:24.13 | nettie | CunningPike do you know if 1.2 and trunk configuration files are 100% compatible? |
| 18:24.36 | CunningPike | nettie: It's not so much the jitterbuffer being great, it's that a) IAX has one b) trunking makes more efficient use of your bandwidth c) a local server prevents internal calls having to pass through your colo server d) the firewalling is easier........ |
| 18:25.32 | pdthome | nettie: iax2 is pretty much meant for what you are doing, have a small server like Cunning is saying and send outbound calls over the iax trunk |
| 18:26.19 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 18:26.28 | nettie | CunningPike yeah.. right know interna calls are going out, because of NAT I actually configured it disabling re-invites |
| 18:26.40 | pdthome | ya the firewalling is much better |
| 18:27.08 | CunningPike | nettie: Yup - I really would look at a local server - it doesn't have to do much, and can be a really small box - how many local phones have you got? |
| 18:27.36 | nettie | about 8 |
| 18:27.42 | nettie | I Alreadyhave a local server |
| 18:27.44 | nettie | we use |
| 18:27.49 | nettie | for smb and stuff |
| 18:27.54 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:27.56 | vivek | iax2 is better at nat ? so its better for a firewall blocked lan with only a couple of ports open ? |
| 18:28.28 | CunningPike | nettie: You could try a shared server - might be better with a dedicated one, though |
| 18:28.30 | nettie | I actually have an epigy pbx as well which is left from a couple of project we had eheh |
| 18:28.39 | CunningPike | vivek: Totally |
| 18:28.43 | nettie | CunningPike sure |
| 18:28.57 | file | I don't wanna run away, but I can't take it... I don't understand... if I'm not made for you then why does my heart tell me that I am |
| 18:29.02 | nettie | CunningPike I'll definitely use the one we laready have |
| 18:29.07 | pdthome | vivek: http://www.voip-info.org/wiki/view/IAX it talks about the natting issue |
| 18:29.11 | vivek | CunningPike: hmmz can i somehow make sip work in the same environ ... i have a couple of sip adapters ... |
| 18:29.42 | CunningPike | vivek: You can, but you need to open the correct ports on your firewall...... |
| 18:30.15 | vivek | CunningPike: I can't do that ... i don't control the firewall :( ... |
| 18:30.40 | CunningPike | nettie: Shouldn't be a problem with a light load - try it :D |
| 18:30.58 | nettie | CunningPike okie I'll give it a shot :) |
| 18:31.02 | CunningPike | vivek: Then you need IAX, and probably on a port that's open already. Sounds like SIP is out of the question |
| 18:31.08 | vivek | CunningPike: only 80 and 81 are open and RTP packets do flow through those ports ... (atleast skype and google talk work ...) |
| 18:31.16 | *** join/#asterisk tod (n=tod@207.54.140.109) |
| 18:31.29 | pdthome | vivek: do you have any udp ports open? |
| 18:31.36 | vivek | CunningPike: that's bad news ... with two spa 3ks in the bag .. |
| 18:32.02 | tod | hello, does anyone know if you can do a regex compare with an "if" statement in ael? |
| 18:32.06 | vivek | pdthome: any traffic other than http and https ports are blocked .. |
| 18:32.17 | CunningPike | vivek: Why can't you get the firewall configured the way you need it? |
| 18:32.28 | pdthome | did they maybe forget dns? |
| 18:32.34 | vivek | CunningPike: cos i am inside my college ... |
| 18:32.43 | pdthome | a lot of times you will find 53 udp open |
| 18:32.45 | vivek | and ican't talk to the admins .. they are all crazy ... |
| 18:33.03 | CunningPike | vivek: Sucks to be you :) |
| 18:33.16 | vivek | CunningPike: yeah it does ... |
| 18:33.27 | vivek | I am hoping iax2 will help me out ... |
| 18:33.28 | CunningPike | vivek: IAX softphone? |
| 18:33.33 | pdthome | vivek: do you nave nmap? |
| 18:33.43 | vivek | pdthome: hmmz whats nmap ? |
| 18:34.05 | vivek | CunningPike: i need to get my parents a hardphone though .. |
| 18:34.07 | Nugget | nmap is a unix utility that lets you probe remote machines. |
| 18:34.20 | vivek | they are not going to muck around with a computer .. |
| 18:34.22 | Nugget | see what ports are open, make an educated guess about what OS it's running, etc. |
| 18:34.35 | vivek | Nugget: they run linux |
| 18:34.43 | vivek | all the servers i.e. |
| 18:34.44 | pdthome | http://www.insecure.org/nmap/download.html |
| 18:34.53 | vivek | All outside access is via proxy servers ... |
| 18:35.04 | pdthome | vivek: so you are trying to connect from you to your parents? |
| 18:35.11 | vivek | pdthome: yes |
| 18:35.13 | Nugget | it's a handy network diagnostics tool that can be used to some success for malicious or harmless shenanigans. |
| 18:35.26 | CunningPike | vivek: Skype? |
| 18:35.26 | vivek | Nugget: tnx ;) |
| 18:35.34 | vivek | CunningPike: Skype works ... |
| 18:35.48 | CunningPike | vivek: So, use that then..... |
| 18:35.48 | vivek | but not feasible .. |
| 18:36.03 | jbalcomb | vivek: setup a proxy server at your parents place using port 80 or 443 |
| 18:36.04 | vivek | no computers at home ;) they simply can't use one .. |
| 18:36.22 | vivek | jbalcomb: ok i can do that on a router ... |
| 18:36.22 | pdthome | vivek: do you have any hosts outside your college? |
| 18:36.36 | jbalcomb | vivek: easy enough solutions |
| 18:36.47 | vivek | pdthome: i can arrage one at home on my wrt router ... |
| 18:36.54 | vivek | wrt54gl router i.e. |
| 18:37.09 | jbalcomb | vivek: whats all this yacking about if you can just do that? |
| 18:37.25 | pdthome | vivek: there are some skype hard phones these days |
| 18:37.57 | vivek | jbalcomb: er i need more info than just put up a proxy server .. |
| 18:38.12 | pdthome | http://www.engadget.com/2006/06/28/philips-adds-a-new-skype-phone-to-their-voip-lineup/ |
| 18:38.26 | pdthome | cordless skype phone, that should work for them |
| 18:38.37 | salviadud | skype suxxors |
| 18:38.57 | vivek | pdthome: i would like to work with what i have atm ;) which is two spa 3ks ... |
| 18:38.58 | salviadud | can't get an ata to work with skype... |
| 18:39.07 | CunningPike | salviadud: It has its uses - this happens to be one of them....... |
| 18:39.16 | vivek | getting hardphones ... for skype seems crazy .. |
| 18:39.22 | pdthome | salviadud: lets see... it's free for skype2skype, they already have the infrastructure, there are plenty of devices, yep for his purposes it sucks don't use it |
| 18:39.36 | CunningPike | vivek: You should talk to Dr-Linux - he's ramming a square peg into a round hole too :) |
| 18:39.44 | salviadud | i'm just talking about the protocol. |
| 18:39.58 | pdthome | vivek: you don't have a bastion host to speak of currently, you don't have inbound port access, and your parents can't use a computer, you are not leaving alot of options :) |
| 18:40.07 | jbalcomb | salviadud then you might clarify when making such generalizations |
| 18:40.17 | Dr-Linux | CunningPike: what's up? :) |
| 18:40.37 | salviadud | jbalcomb, i will now speak in the third person |
| 18:40.42 | *** join/#asterisk hess\n (n=hess@201.44.216.94) |
| 18:40.45 | vivek | pdthome: yeah i see that ... but how does a proxy help ? I could do something about the proxy outside my college .. |
| 18:40.47 | hess\n | hello |
| 18:40.57 | CunningPike | Dr-Linux: Not much - get your skinny phone working yet? And, before you ask, I have no clue about skinny |
| 18:41.11 | *** join/#asterisk n3^ (n=n3@63-253-43-58.ip.mcleodusa.net) |
| 18:41.16 | pdthome | well you can tunnel stuff via proxy to an external host from your location, from your parents house I would assume you won't have the same firewalling issues |
| 18:41.22 | *** join/#asterisk Samoied (n=Samoied@200.213.47.92) |
| 18:41.36 | vivek | pdthome: no firewall issues at home ... |
| 18:41.39 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 18:41.49 | salviadud | and thats about it |
| 18:41.54 | Dr-Linux | CunningPike: heh, everything works fine but only problems remains, it doesn't hangup after 2 tries. |
| 18:41.58 | pdthome | <PROTECTED> |
| 18:42.17 | vivek | pdthome: cool that could work .. maybe i can run the proxy on my ruoter ... |
| 18:42.21 | vivek | at home .. |
| 18:42.34 | pdthome | possibly, the wrt is a little tight on resources so you'll have to give it a shot |
| 18:43.04 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 18:43.09 | *** part/#asterisk tod (n=tod@207.54.140.109) |
| 18:43.14 | jbalcomb | i thought i saw an asterisk implementation that your could load on the linksys wrt? |
| 18:43.26 | vivek | jbalcomb: yes there is one ... |
| 18:43.27 | pdthome | i think there is one |
| 18:43.38 | jbalcomb | resources can't be that tight then.. |
| 18:43.55 | jbalcomb | just change the default ports that it listens on and your done.. |
| 18:44.02 | pdthome | well if he loads asterisk and a tunnel server, etc. etc etc... it could get tight quick |
| 18:44.22 | n3^ | I have an AGI script triggering Festival... anyone know the best way to interrupt it similar to the background command? |
| 18:44.27 | pdthome | with the size of his project though cpu power shouldn't be a problem |
| 18:44.39 | pdthome | only real worry is ram |
| 18:44.40 | jbalcomb | pdthome any reason port forwarding wouldn't suite as well as a tunnel server? |
| 18:44.45 | vivek | 16mb could be a problem though ... |
| 18:45.17 | pdthome | jbalcomb: he said his college is full proxy, so he has to get out somehow |
| 18:45.34 | jbalcomb | unless the college is doing content based packet filtering... |
| 18:45.45 | Hmmhesays | could be |
| 18:45.53 | vivek | hmmz why can't i just try using port 80 as my sip port and connect to an external provider ? |
| 18:45.56 | pdthome | Parents house -----> (INTERNET) <--- |Proxy only FW| <---- vivek |
| 18:46.03 | jbalcomb | pdthome: so his phone connects to port 80 on his parents router? |
| 18:46.16 | pdthome | if it's a true proxy fw that won't work |
| 18:46.18 | pdthome | sip aint http |
| 18:46.20 | n3^ | should I be using the ExternalIVR command? |
| 18:46.20 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:46.42 | pdthome | proxy tunnel software makes all the stuff going across look like valid http traffic |
| 18:46.49 | pdthome | it just depends on what they are using |
| 18:46.54 | jbalcomb | oh.. i wondered why they called them different names when i thought they were the same protocol... |
| 18:47.27 | CunningPike | vivek: If I was looking after your uni's network, I would be filtering port 80 to just http requests. It's unlikely that they will allow inbound UDP on port 80 |
| 18:47.29 | pdthome | the port 80 thing might work, but the chances that they also opened up 80 udp are pretty slim |
| 18:47.44 | jbalcomb | how is /valid/ http traffic different than any traffic destined for port 80? |
| 18:48.05 | jbalcomb | CunningPike inbound wouldn't be 80 |
| 18:48.17 | pdthome | they dont' allow inbound anything to vivek |
| 18:48.24 | pdthome | he can go outbound on 80 and 443 |
| 18:48.28 | pdthome | via a proxy |
| 18:48.29 | jbalcomb | the matter of udp being open is a much more valid concern |
| 18:48.49 | vivek | well as far as i know they just filter urls and look for some 3000 blacklisted words .. |
| 18:48.54 | jbalcomb | only way to know is to try |
| 18:49.03 | pdthome | then it's either IDP or a real proxy |
| 18:49.07 | vivek | i can't do that untill i get there ... |
| 18:49.10 | jbalcomb | vivek: if that is the case then you should be fine |
| 18:49.38 | pdthome | either way you are probably screwed without a proxy that looks like legit http/https traffic |
| 18:49.47 | pdthome | well legit https traffic is a bit of a funny statement |
| 18:50.04 | BugKham | how to uninstall zaptel drivers? |
| 18:50.20 | BugKham | "make uninstall" doesn't exist |
| 18:50.33 | vivek | pdthome: i am flumoxxed... i shuld put up a proxy and try and tunnel in ? |
| 18:50.39 | jbalcomb | pdthome yeah.. you think they are checking the packet and doing fuzzy logic on the contents to see if it matches what /valid/ html would be? |
| 18:50.40 | vivek | or tunnel out ... |
| 18:50.55 | vivek | jbalcomb: i don't think they do that ... |
| 18:51.02 | pdthome | jbalcomb: ya, GET/POST/HEAD headers, etc... |
| 18:51.17 | jbalcomb | vivek: i'd be a bit surprised if they did |
| 18:52.04 | pdthome | vivek: to test it just setup an ssh server on port 80 somewhere and see if you can connect, if you can, your good |
| 18:52.22 | pdthome | i would doubt they have 80 udp open out, that doesn't make much sense |
| 18:52.30 | pdthome | so you will have to have a tunnel of some sorts |
| 18:52.33 | pdthome | either way |
| 18:52.56 | vivek | so how does skype work its way through and google talk ? |
| 18:53.02 | pdthome | but a softphone with an ssh tunnel would be much easier |
| 18:53.12 | *** part/#asterisk BugKham (i=CKGLOB@125.24.3.20) |
| 18:53.18 | *** join/#asterisk BugKham (i=CKGLOB@125.24.3.20) |
| 18:53.42 | pdthome | skype is tcp |
| 18:53.51 | pdthome | and google talk is jabber based which is also tcp |
| 18:54.05 | pdthome | If the above is not possible, Skype versions 0.97 or later can use a HTTPS/SSL proxy. In order to do that, you have to configure the proxy address in Internet Explorer options. Then Skype will be able to use it as well. |
| 18:54.08 | Hmmhesays | so switch your sip shiat over to tcp |
| 18:54.18 | pdthome | skype automatically uses your proxy settings |
| 18:54.25 | pdthome | i would assume gtalk does the same |
| 18:54.47 | vivek | Hmmhesays: hmmz are you joking ? ;0 |
| 18:54.49 | pdthome | so in theory with skype if you can browse the internet it will work |
| 18:54.54 | jbalcomb | Hmmhesays: isn't it more customary to use IAX for external phones? |
| 18:55.08 | pdthome | jbalcomb: hehehe, that was our initial suggestion but he has sip phones already purchased |
| 18:55.14 | Hmmhesays | is there a customary? If you're using hard phones, use tcp |
| 18:55.15 | pdthome | or ATAs i believe |
| 18:55.39 | jbalcomb | Hmmhesays: there certainly is a customary. it's usually what ends up working best for the most people. |
| 18:55.46 | vivek | i think i will go ahead and get aix2 ata's ... |
| 18:55.47 | *** join/#asterisk linlin (i=linlin@c-67-184-152-231.hsd1.il.comcast.net) |
| 18:55.47 | jbalcomb | ATAs i thinks |
| 18:55.59 | Hmmhesays | SER handles tcp sip just fine |
| 18:56.00 | pdthome | vivek: try a softphone from school before you purchase |
| 18:56.03 | pdthome | see what you can make work |
| 18:56.18 | jbalcomb | vivek: buy them a PC from me and using MSN messenger with headsets |
| 18:56.26 | Hmmhesays | or drop an openvpn wrt in front of the phone |
| 18:56.29 | linlin | What are the fees associated with using asterisk for a small business environment, only 5-15 employees, only maybe 10 phones or so. Is there licencing involved? |
| 18:56.32 | pdthome | someone in here I am sure will let you connect to their server when you are ready to test. I have a couple of servers sitting on the net for testing you can use if I am around when you are testing |
| 18:56.37 | Bullseye_Network | ~flush |
| 18:56.39 | Bullseye_Network | ops |
| 18:56.41 | Bullseye_Network | sorry |
| 18:56.43 | Bullseye_Network | lol |
| 18:56.50 | Hmmhesays | linlin: 10 bucks a chan if you want g.729 thats about it |
| 18:56.53 | jbalcomb | linlin: perhaps digium.com would be a good place to go for that |
| 18:57.04 | pdthome | openvpn rcks |
| 18:57.06 | pdthome | rocks even |
| 18:57.17 | Hmmhesays | ~8ball kick in the nuts? |
| 18:57.19 | jbot | Please ask again. |
| 18:57.21 | vivek | pdatnx tnx i will check back in 10 days in that case ... ;) |
| 18:57.24 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 18:57.29 | vivek | er i mean pdthome |
| 18:58.13 | Hmmhesays | no thanks to me eh? |
| 18:58.14 | Hmmhesays | wtf |
| 18:58.22 | pdthome | vivek: np |
| 18:58.27 | vivek | Hmmhesays tnx ;) |
| 18:58.30 | pdthome | lol |
| 18:58.35 | vivek | and tnx to everyone who helped |
| 18:58.46 | Hmmhesays | i use openvpn on wrt's that go back to an asterisk box, takes care of pretty much every network situation you would come across |
| 18:59.01 | boch | guys, do you know what this msg means: WARNING[278]: channel.c:2328 set_format: Unable to find a codec translation path from g729 to slin |
| 18:59.24 | file | boch: Asterisk wants to transcode g729 to signed linear, and you have no codec installed to do it |
| 19:00.14 | boch | whats signed linear? a kind of codec? im only trying to do g729 passthrough |
| 19:00.58 | [TK]D-Fender | boch : your endpoints dont match if its trying to translate |
| 19:03.14 | vivek | Hmmhesays: are you in the black gold territory to use vpn for voip ? i hear UAE blocks all voip calls ... my college servers kinda seem like that ;) |
| 19:03.31 | Hmmhesays | black gold? |
| 19:03.39 | boch | [TK]D-Fender: is there a way to fix this without buying a g729 license ? |
| 19:03.39 | vivek | blackgold is oil ... |
| 19:03.47 | Qwell[] | boch: don't use g729? |
| 19:03.47 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 19:03.56 | Hmmhesays | vivek kind of |
| 19:04.00 | vivek | UAE is united arab emarites ... |
| 19:04.03 | vivek | Hmmhesays: ok ;) |
| 19:04.20 | boch | Qwell[]: i wish i could, but im using it only for passtrough |
| 19:04.50 | Qwell[] | boch: If you're using any apps, (background, playback, voicemail, etc) that doesn't have g729 sounds...it'll transcode |
| 19:04.53 | Qwell[] | and you need a license to transcode |
| 19:05.15 | Qwell[] | you can get g729 sounds for those though, of course |
| 19:05.59 | Juggie | or, you could use sox and generate g729 files for every sound file * has in a few minutes. |
| 19:06.04 | *** join/#asterisk davegrin (n=da5id@24-38-58-10-st.lndnnh.adelphia.net) |
| 19:06.15 | boch | just Answer, Dial and Hangup, i have a license in other ast, can i copy it? i think the file is codec_g729-gcc-pentium4.so |
| 19:09.03 | *** part/#asterisk BugKham (i=CKGLOB@125.24.3.20) |
| 19:09.22 | file | boch: is that the Intel one? |
| 19:09.53 | [TK]D-Fender | boch : Well if its passthrough you shouldn't have an issue. verify that both ends support it and that * KNOWS it. |
| 19:14.10 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.236) |
| 19:17.48 | *** join/#asterisk my_name (n=j@nathan.epi.usf.edu) |
| 19:18.13 | my_name | hello |
| 19:18.37 | my_name | i just installed asterisk and seems to be working "fine" but i have no clue as to what to do with it |
| 19:18.41 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:18.51 | quadrata | heh |
| 19:19.06 | [TK]D-Fender | my_name : ... |
| 19:19.07 | [TK]D-Fender | ~docs |
| 19:19.08 | jbot | somebody said docs was 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 |
| 19:19.10 | [TK]D-Fender | ~book |
| 19:19.11 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:20.40 | my_name | hmmm let me read through that then |
| 19:20.47 | my_name | i'll get back to you |
| 19:21.13 | [TK]D-Fender | my_name : Was your statement about how to configure it or what to USE it for? |
| 19:21.29 | my_name | i know what my general purpose was i know asterisk should accomplidsh that |
| 19:22.04 | my_name | i just don't know how to continue on i hope the book helps |
| 19:22.25 | CunningPike | Is FOP still the best 'console' solution? |
| 19:22.37 | my_name | [TK]D-Fender right configuration |
| 19:22.38 | [TK]D-Fender | my_name : Ok, well read the book, its a good one and should give you a good place to start from. |
| 19:22.51 | [TK]D-Fender | CunningPike : Depends what for. What are your goals? |
| 19:23.37 | CunningPike | [TK]D-Fender: Right now, our receptionist has a Nortel console that lets her assign incoming calls to extensions, see who's on the phone....... |
| 19:23.57 | CunningPike | Divert to voicemail etc |
| 19:24.13 | [TK]D-Fender | CunningPike : How many ext's to monitor? |
| 19:24.33 | smackus | if I use the AgentLogin command to log in, how do i get them to log out? exten => 2001,2,AgentLogin() |
| 19:24.43 | CunningPike | [TK]D-Fender: Fewer than can be accomodated on a 601 with sidecar(s) ;) |
| 19:24.49 | smackus | when I call the extension it asks for user name and pass, then says logged in. |
| 19:24.56 | smackus | if I dial it again, it says user already logged in |
| 19:25.01 | [TK]D-Fender | smackus : they log themselves out by #, or get kicked off for not answering, etc |
| 19:25.05 | CunningPike | [TK]D-Fender: But we're interested in a 'soft' solution |
| 19:25.53 | smackus | so when i dial my login extension, it asks first for user name then #, then pass then #. When do I give it #? |
| 19:25.54 | [TK]D-Fender | CunningPike : Hrm... thats what FOP is for.... You could always make your own easily enough (to see who's on the phone at least), but for "active" features that'd require some real work. |
| 19:26.15 | [TK]D-Fender | smackus : You do that on the phone that IS logged in. |
| 19:26.35 | smackus | so i dial the extension, then just press #? |
| 19:26.54 | my_name | see ya folks i'm gonna be reading for a while |
| 19:27.06 | CunningPike | [TK]D-Fender: OK - I'll go with FOP - I just wanted to check before I installed it that it hadn't been overtaken by a better one |
| 19:27.53 | [TK]D-Fender | CunningPike : not to my awareness... I am considering updating my Polycom MB one for "web" use.... |
| 19:28.09 | [TK]D-Fender | smackus : No. Do you know how AgentLogin works? |
| 19:28.25 | CunningPike | [TK]D-Fender: Well, if anyone would know, you would ;) Thanks |
| 19:30.44 | [TK]D-Fender | CunningPike : Not true.... I avoid GUI's wherever possible, don't work with DB's, and noting fancy involving AMI or Apache. |
| 19:30.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:31.19 | CunningPike | [TK]D-Fender: Ya, I hear you - this is our first foray into anything like that |
| 19:31.30 | [TK]D-Fender | I've only logged into an AMP install a handful of times remotely to look at a guys system, and I am running a GUI here as well, but have little experience in the setup and full capabilities of them all |
| 19:34.31 | rob0 | Can "host=" in a sip.conf use a wildcard, like say, "host=*.asterlink.com"? |
| 19:34.56 | quadrata | I would think not - name has to be resolvable |
| 19:35.08 | boch | [TK]D-Fender: have a minute to see this http://pastebin.ca/79847 ? |
| 19:35.24 | file | rob0: no. |
| 19:35.25 | smackus | [TK]D-Fender: i was using agentcallbacklogin, and was suggested to switch to agentlogin. |
| 19:35.41 | [TK]D-Fender | boch : clearly not compatable. Pastebin your sip.conf entries for those 2 devices. |
| 19:35.43 | smackus | it is not working the way I thought it did, so to answer your question, no I dont know how it works |
| 19:36.02 | rob0 | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+permit-deny-mask is probably what I want. |
| 19:36.03 | [TK]D-Fender | smackus : Sounds like AgentCallbackLogin didn't kick them out. |
| 19:36.37 | *** part/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 19:36.43 | smackus | well, we have no one logged in. |
| 19:36.47 | smackus | restarted the system |
| 19:36.55 | smackus | and now are trying to log them in using agentlogin. |
| 19:37.14 | vader-- | Which do you guys like better FOP or HUDLite? |
| 19:37.29 | boch | [TK]D-Fender: here it is http://pastebin.ca/79849 |
| 19:38.08 | [TK]D-Fender | boch : What are these 2 enpoints? |
| 19:38.40 | rob0 | Asterisk+sip+permit-deny-mask doesn't show CIDR addressing, do I have to use the full netmask, /255.255.255.0 instead of /24? |
| 19:39.06 | [TK]D-Fender | boch : And what is that place that you are dialing? |
| 19:39.58 | [TK]D-Fender | rob0 : Forget masks, isn't safe when dealing with DNS to simply assume that it'll always fall in a range. |
| 19:47.16 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 19:47.42 | *** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk) |
| 19:48.38 | AuPix | russellb, are you on line? |
| 19:52.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.236) |
| 19:55.44 | boch | [TK]D-Fender: the first one is a spa2100 and the second one a provider (only uses g729) |
| 19:58.35 | Dr-Linux | Sphinx Voice recognition program is working for anyone? |
| 20:00.32 | [TK]D-Fender | boch : you really shouldn't be dialing by IP and they are clearly not liking your codec choice. |
| 20:01.16 | [TK]D-Fender | boch : do a SIP debug on the call to confirm what they're offering |
| 20:02.50 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:04.25 | *** join/#asterisk adorah (n=Administ@87.69.72.228) |
| 20:09.18 | *** part/#asterisk MonkeyHugs (n=MonkeyHu@63.149.122.94) |
| 20:21.57 | *** join/#asterisk jsaunders (n=root@70.71.224.65) |
| 20:22.02 | jsaunders | Good day all. :D |
| 20:24.47 | jsaunders | Anyone had issues with a Dial() string such as "SIP/101&IAX2/201" where it will dial both channels simultaneously, and if I answer say the IAX2 channel, the softphone (SJPhone in this case) will continue ringing? |
| 20:26.14 | [TK]D-Fender | jsaunders : No. It will stop. |
| 20:26.26 | jsaunders | So it must be SJPhone. |
| 20:29.04 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:29.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.236) |
| 20:39.09 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 20:42.31 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 20:43.51 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 20:44.19 | *** join/#asterisk hohum (n=dcorbe@12.195.58.237) |
| 20:45.30 | ManxPower | ~docs |
| 20:45.35 | jbot | [docs] 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 |
| 20:45.35 | ManxPower | ~book |
| 20:45.37 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:47.55 | *** join/#asterisk dacleric (n=dacleric@p54823429.dip0.t-ipconnect.de) |
| 20:49.37 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 20:50.56 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:02.51 | *** join/#asterisk skraelings001 (n=skraelin@201.230.47.19) |
| 21:02.54 | *** join/#asterisk Shoragan (n=shoragan@datenfreihafen.org) |
| 21:03.21 | skraelings001 | hello everybody |
| 21:03.28 | *** part/#asterisk quadrata (n=spork@dynamic-64-115-24-203.isp.broadviewnet.net) |
| 21:04.15 | CunningPike | Is AstManProxy still on the go? |
| 21:04.35 | skraelings001 | i have problems with directed pickup app. i'm just able to pick sip and iax call but not zap. any idea? |
| 21:09.57 | ManxPower | skraelings001, put the console info on pastebin.ca |
| 21:16.59 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 21:17.37 | CunningPike | No-one knows if AstManProxy is still a going concern? |
| 21:18.03 | *** join/#asterisk backblue (n=moo@87-196-69-122.net.novis.pt) |
| 21:18.28 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 21:19.02 | PakiPenguin | can i increase the parking time |
| 21:20.05 | popvoxdave | yes, astmanproxy is on version 1.21 |
| 21:20.48 | Vorondil | PakiPenguin: i believe features.conf would be what you're looking for |
| 21:20.55 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 21:21.17 | PakiPenguin | i want to increase it more then 60 secs |
| 21:21.36 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
| 21:22.33 | ManxPower | CunningPike, Yes, it is. Why do you think it is not. |
| 21:22.42 | ManxPower | PakiPenguin, Yes. |
| 21:23.56 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 21:24.53 | *** join/#asterisk jwa (n=jwa@columbia80.com) |
| 21:25.08 | skraelings001 | ManxPower: sorry late, i have added portions of dial plan http://pastebin.ca/79908 |
| 21:25.16 | CunningPike | ManxPower: I don't - just making sure..... |
| 21:25.40 | popvoxdave | I put up an early version of astmanproxy 1.21 a couple of days ago which had a typo and caused a crash. |
| 21:25.50 | popvoxdave | pls be sure you get latest 1.21 rev if you are downloading fresh. |
| 21:27.09 | CunningPike | popvoxdave: Great - thanks for the headsup |
| 21:27.49 | nortex | What causes this? http://pastebin.ca/79910 |
| 21:28.33 | jwa | hello .. i've got an older version of asterisk ( SVN-branch-1.2-r25015 ) that's been segfaulting and locking up. should i download the tarball & rebuild it, or should I fetch the latest 1.2 branch ( branches/1.2 ) ? |
| 21:29.30 | nortex | It has happened 3 times today and I can't figure out why there is a deadlock |
| 21:29.50 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 21:30.35 | ManxPower | jwa, only released tarballs are expected to work. |
| 21:30.45 | blebleble | anyone ever try something like this, i want an internal fax machien (ata / sip) to send all of its faxes to our hylafax server and then have hylafax queue the items and send them out via a true modem/pstn line doable? suggestions? |
| 21:31.15 | jwa | ManxPower: that's what I would expect .. I inherited this setup & I'm trying to figure out the best way to fix it :-) |
| 21:31.29 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 21:31.29 | ManxPower | nortex, Nobody knows what causes it, Digium seems to say "it's just a warning, not an error, it's not a problem." |
| 21:31.51 | ManxPower | blebleble, FaxOverVoiceOverIP is not reliable |
| 21:32.11 | P4C0 | hello guys, I'm having problems sending email notifications (for mailboxes) is there a way to tell asterisk to use a smtp server? |
| 21:32.24 | ManxPower | P4C0, no. |
| 21:32.30 | rob0 | well ... maybe |
| 21:32.33 | ManxPower | it will use the local sendmail command. |
| 21:32.49 | jwa | I'm also getting 'app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)' -- lots of 'em .. what do they mean? |
| 21:32.52 | rob0 | nail(1) is a replacement for /bin/mail (BSD mailx) which can use SMTP. |
| 21:32.59 | ManxPower | or whatever command you tell it in voicemail.conf |
| 21:33.05 | CunningPike | P4C0: You can try using sendmail.cf to set options..... |
| 21:33.23 | ManxPower | jwa, that means it cannot send a call to that destination. |
| 21:33.27 | *** join/#asterisk knarfly (n=bwatson@12.42.132.26) |
| 21:33.34 | nortex | ManxPower, It is for me, because once it starts to deadlock all incoming and outgoing calls drop. |
| 21:33.36 | ManxPower | Does "sip show peers" show those devices as having an IP address/ |
| 21:33.41 | P4C0 | ManxPower, CunningPike thanks |
| 21:33.42 | ManxPower | nortex, I know. |
| 21:33.49 | *** join/#asterisk |dennis| (n=dennis@200.32.215.84) |
| 21:33.56 | ManxPower | Perhaps you need to report it. Try #asterisk-dev or #asterisk-bugs |
| 21:34.20 | nortex | ManxPower, I'll try |
| 21:34.25 | ManxPower | rob0, that would be useful if voicemail used /bin/mail |
| 21:35.02 | *** join/#asterisk alvariux (n=Administ@201.112.57.231) |
| 21:35.20 | alvariux | hello |
| 21:35.52 | alvariux | anybody? |
| 21:36.04 | ManxPower | alvariux, ask your question or go away |
| 21:36.16 | phigwork | Anyone using FWD through IAX? |
| 21:37.05 | *** join/#asterisk knarfly (n=bwatson@12.42.132.26) |
| 21:37.18 | alvariux | im getting some problems registering to broadvoice |
| 21:37.33 | *** part/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 21:37.34 | rob0 | "mailcmd=/bin/mail" in voicemail.conf? I don't know if that would work tho; /bin/mail (nail) might have different requirements. |
| 21:37.50 | phigwork | I'm wondering if they've disabled 411 and *1800 through their iax gateway. Anyone know? |
| 21:38.01 | rob0 | phigwork: in fact IAX was the only way I could get FWD working. SIP fails. |
| 21:38.28 | *** join/#asterisk R-MAN (n=raficmas@i-195-137-114-169.freedom2surf.net) |
| 21:38.34 | phigwork | rob0: I can register and I -think- I'm able to place calls to other FWD numbers |
| 21:38.53 | phigwork | but when I try 411 or *1800xxxxxxx it rejects the call |
| 21:39.09 | rob0 | haha, I tried to sign up at their forums yesterday ... my MX rejected the confirmation |
| 21:39.16 | alvariux | Probably a DNS error for registration |
| 21:39.38 | blebleble | Manpower: that was kind of my point of trying this, forcing faxes to go internal to a hylafax server, and then piping them out over pstn |
| 21:39.56 | rob0 | Jul 4 07:15:10 miniluv postfix/smtpd[25477]: warning: Illegal address syntax from mail.pulver.com[192.246.69.184] in MAIL command: <forum-no-reply@freeworlddialup.com "fwd user forums"> |
| 21:39.59 | phigwork | rob0: can you try ringing me over fwd? |
| 21:40.10 | rob0 | phigwork: sure |
| 21:40.19 | alvariux | im using this register => 8322010274:password@sip.broadvoice.com |
| 21:40.35 | alvariux | any idea |
| 21:40.43 | alvariux | everything seems ok |
| 21:42.27 | CunningPike | What are people using for ACD stats? |
| 21:44.07 | Nugget | Every time I see "ACD" I parse it as Apple Cinema Display and get confused. |
| 21:44.24 | CunningPike | :D |
| 21:44.28 | CunningPike | ~acd |
| 21:44.29 | jbot | it has been said that acd is All Cats Down, a Jazz term used when the musicians are passed out drunk (props to ManxPower) |
| 21:44.30 | Nugget | there aren't enough TLAs to go around. |
| 21:44.40 | P4C0 | do I need to have sendmail running as a daemon to send mails?? humm |
| 21:44.40 | *** part/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 21:45.05 | dlynes_office | I like jbot's definition better than CP's :) |
| 21:45.08 | smackus | I started using agentlogin today instead of agentcallbacklogin... now I periodically have asterisk just quit and shutdown. Has anyone else had an experience like this? |
| 21:45.12 | *** part/#asterisk alvariux (n=Administ@201.112.57.231) |
| 21:45.23 | smackus | I had this same issue with mixmonitor, it went away when I went back to monitor. |
| 21:45.34 | smackus | wondered if there is anything like that going on with agentlogin |
| 21:45.45 | CunningPike | Hey, dlynes_office - where ya bin? |
| 21:45.49 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:45.50 | dlynes_office | Just busy |
| 21:45.59 | dlynes_office | Bought a new laptop on Friday |
| 21:46.06 | dlynes_office | Great sale at Future Crap |
| 21:46.14 | CunningPike | dlynes_office: Neato - what did you get? |
| 21:46.16 | dlynes_office | dual core laptop for $1000 :) |
| 21:46.22 | CunningPike | Not bad |
| 21:46.23 | dlynes_office | with 1GB RAM to boot |
| 21:46.26 | dlynes_office | expandable to 4GB |
| 21:46.33 | CunningPike | Nice |
| 21:46.56 | dlynes_office | Yeah...i didn't want to be stuck with one that was only expandable to 2GB's, in case I needed more |
| 21:47.18 | CunningPike | Always need more :) |
| 21:47.26 | dlynes_office | It'll be mostly a development machine |
| 21:47.32 | dlynes_office | but i will use it for site administration, too |
| 21:47.49 | dlynes_office | So it has Linux 2.6.17.3 on it, and Windows XP |
| 21:48.07 | dlynes_office | And i'm putting all the network scanning tools and VOIP bandwidth tools |
| 21:48.12 | dlynes_office | on it |
| 21:48.58 | CunningPike | Neat - I'm just installing CentOS on a box to run AstManProxy so we can host FOP or HUDLite, ACD stats etc on it |
| 21:53.45 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:53.51 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:53.53 | Dr-Linux | hi all |
| 21:56.54 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 21:57.29 | Dr-Linux | rob0: why? |
| 21:57.40 | rob0 | 21:46 < dlynes_office> dual core laptop for $1000 :) |
| 21:58.16 | dlynes_office | rob0: ? |
| 21:58.25 | dlynes_office | rob0: oh...nvm |
| 21:58.36 | dlynes_office | rob0: it was a toshiba satellite |
| 21:58.45 | Dr-Linux | dlynes_office: what's up? |
| 21:58.48 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
| 21:58.59 | dlynes_office | not much |
| 21:59.05 | dlynes_office | Just busy setting up some phones |
| 21:59.06 | rob0 | Toshibas used to be real good, but I've heard their quality has gone down (I wouldn't know) |
| 21:59.10 | dlynes_office | gotta install a new system tomorrow |
| 21:59.17 | dlynes_office | rob0: some people have had bad luck with them |
| 21:59.24 | dlynes_office | rob0: but in general, they're still pretty good |
| 21:59.32 | rob0 | My first computer was a Toshiba laptop, 8086 CPU :) |
| 21:59.41 | dlynes_office | rob0: for the most part, you probably want to go with a toshiba satellite pro, or higher |
| 21:59.56 | dlynes_office | rob0: but i figured for $1000, who cares |
| 21:59.57 | rob0 | it was a real champ until /dev/kid poured Coca-Cola in it. |
| 22:00.05 | skraelings001 | ManxPower: i think i've solved, when i make a call like this 61072XX and it should dial 2XX the current extension is 61072XX and not 2XX so Pick up must be checking if the extension 2XX has been called, but it hasn't been as we see, so i should modify the current extension. |
| 22:00.08 | rob0 | haha ... mine was $1k too |
| 22:00.10 | Dr-Linux | dlynes_office: okey, when you get free then lemme know, i wanna discuss something with you |
| 22:00.24 | dlynes_office | Dr-Linux: what's she look like? |
| 22:01.15 | Dr-Linux | dlynes_office: Cisco 7935 |
| 22:02.25 | dlynes_office | oh |
| 22:02.38 | dlynes_office | thought you were talking about some hot new girl you had your eye on :((( |
| 22:02.53 | dlynes_office | thought you were gonna show us some naked pictures :(( |
| 22:03.10 | CunningPike | Of Dr-Linux?? No thanks :O |
| 22:03.12 | Dr-Linux | dlynes_office: not eye, i was all on her a day before. |
| 22:04.02 | CunningPike | Better not let Katty hear you talking like that...... |
| 22:04.04 | Dr-Linux | CunningPike? |
| 22:04.17 | dlynes_office | CunningPike: moi? |
| 22:04.28 | CunningPike | dlynes_office: ;) |
| 22:04.46 | dlynes_office | I don't know her from adam :) |
| 22:04.52 | Dr-Linux | CunningPike: katty has big ..... so i don't like with her |
| 22:04.54 | dlynes_office | I talk to her...that's about it |
| 22:05.32 | Dr-Linux | dlynes_office: she is not a hot girl, she is an old lady |
| 22:05.40 | dlynes_office | Dr-Linux: who? |
| 22:05.50 | Dr-Linux | so forgot about her and let's hug 7935 |
| 22:05.55 | Dr-Linux | dlynes_office: Katty |
| 22:05.59 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:06.07 | dlynes_office | ah...you already tried hitting on her? |
| 22:06.19 | dlynes_office | you sad, sad individual... |
| 22:06.32 | Dr-Linux | dlynes_home: No, but he tried hitting on all |
| 22:06.43 | dlynes_office | he? |
| 22:06.48 | dlynes_office | katty's a transtesticle? |
| 22:07.03 | Dr-Linux | ~dict transtesticle |
| 22:07.22 | dlynes_office | ~dict transvestite |
| 22:07.41 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 22:07.51 | dlynes_office | TripleFFFF: !!! You're just in time |
| 22:07.57 | TripleFFFF | wow why |
| 22:07.58 | TripleFFFF | <PROTECTED> |
| 22:08.00 | dlynes_office | TripleFFFF: We're talking about transtesticles |
| 22:08.04 | CunningPike | TripleFFFF: Run! Run for your life! |
| 22:08.05 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:08.10 | TripleFFFF | anyoen had dropped calls .. its the router im sure |
| 22:08.16 | CoffeeKid | has anyone ever got a polycom (or other hard phone) working over a vpn? |
| 22:08.18 | TripleFFFF | bold mode off |
| 22:08.27 | Dr-Linux | :S |
| 22:08.46 | TripleFFFF | the guy was saing its me.. when even is vonage crap drops after 300 sec |
| 22:08.47 | Dr-Linux | dlynes_office: let's fuck cisco 7935 ? |
| 22:08.49 | dlynes_office | what is it? why do so many peeps want to run sip phones over a vpn? |
| 22:08.54 | TripleFFFF | so its the router that flushes something every 300 |
| 22:09.08 | dlynes_office | TripleFFFF: set qualify=300 |
| 22:09.09 | TripleFFFF | vpn so no ./rtpcapture |
| 22:09.11 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
| 22:09.24 | CoffeeKid | its a polycom 301, we wanna do it so its secure, and we can have a connection to work, with an extension from home:) |
| 22:10.33 | CoffeeKid | my whole network is routed through the vpn... through a linux server (ip tables), i can get ftp access to the phone to update software, but can't call in or out. |
| 22:11.06 | CoffeeKid | well, routed as in, any traffic to that subnet goes through the vpn, not all traffic. |
| 22:11.16 | *** part/#asterisk jwa (n=jwa@columbia80.com) |
| 22:11.26 | Dr-Linux | dlynes_office: you are helping me only with sex :( but not asterisk |
| 22:11.48 | CoffeeKid | sex is easier than asterisk :) |
| 22:12.07 | P4C0 | CoffeeKid, agree |
| 22:12.30 | TripleFFFF | dlynes_office touhgt qualify would just make as unavail of ping ove xxx.. |
| 22:12.44 | TripleFFFF | as in qualify 2000 .. if ping > 2sec.. mark as not there |
| 22:12.53 | TripleFFFF | hence needs 2000 ms to be qualified |
| 22:13.14 | skraelings001 | c u |
| 22:13.16 | P4C0 | I found the problem... if I send a mail like sendmail -f myuser@mydomain touser@todomain it works, but not sure how asterisk is sending it... cause he add my domain to the relay part or something strange... how can I modify this? |
| 22:13.22 | CoffeeKid | anyone have any clue about my vpn deal? anyone give it try? |
| 22:13.23 | TripleFFFF | i think its something in there that just craps it out |
| 22:14.21 | CoffeeKid | P4C0: what do you mean "he add my domain to the relay part"? |
| 22:14.37 | nortex | CoffeeKid, I have Polycom 500/600's connected via a site-to-site VPN |
| 22:14.45 | Dr-Linux | anybody ever use Cisco 7935 phone? |
| 22:14.50 | CoffeeKid | nortex awesome, how'd you get it to work? |
| 22:15.00 | riddlebox | if you have an alternate number with broadvoice, do you just have to have the main number in sip.conf, or both? |
| 22:15.23 | P4C0 | CoffeeKid, not sure, but if I send something calling the command directly it works, and throw asterisk it doesnt... not sure how exactly is asterisk invoking sendmail |
| 22:15.26 | pdthome | does the wrt get any more stable with the custom firmware on it, mine reboots a few times a day with the factory software |
| 22:16.22 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
| 22:16.37 | dlynes_office | TripleFFFF: i've found the wrt54g's huge poc's...anything over 300ms makes them unusable |
| 22:16.48 | CoffeeKid | P4C0: perhaps try an AGI script? |
| 22:16.52 | dlynes_office | TripleFFFF: maybe 1000 for the wrt54g |
| 22:17.02 | dlynes_office | TripleFFFF: maybe it was 300 for the netgears |
| 22:17.03 | P4C0 | CoffeeKid, AGI script? what is that? |
| 22:17.10 | TripleFFFF | well still |
| 22:17.18 | dlynes_office | TripleFFFF: but the wireless routers definitely seem to be worse than the wired ones for that |
| 22:17.28 | TripleFFFF | something is weird since it drops exaacty 300 sec after answer |
| 22:17.28 | CoffeeKid | P4C0: an external script written in php or something, that asterisk calls when you tell it to. |
| 22:18.01 | P4C0 | CoffeeKid, naa, now it works, I just missed the -t in the asterisk call ;) thanks |
| 22:18.07 | P4C0 | I mean in the sendmail call |
| 22:18.18 | CoffeeKid | P4C0: ah, cool, glad i could help.. hehe |
| 22:18.30 | *** part/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 22:20.07 | TripleFFFF | hey .. your pos arugment doesnt hold.. |
| 22:20.08 | TripleFFFF | lol |
| 22:20.16 | TripleFFFF | since x-lite works over 1 hour no prob |
| 22:20.25 | TripleFFFF | only the pap2-t and pap2 from vonage do that |
| 22:20.30 | dlynes_office | waht do yo umean? |
| 22:20.40 | TripleFFFF | xlite i can make calls for 23940852-0947529380 seconfs |
| 22:20.43 | dlynes_office | what does pap2 have to do with whether linksys is a pos or not? |
| 22:20.56 | TripleFFFF | pap2t-na and pap2-vonage they drop after 300 sec.. |
| 22:21.16 | TripleFFFF | well.. i kind of got to the point its the router.. but if xlite works.. its the paps.. now.. why .is a good one |
| 22:21.24 | *** join/#asterisk Eecplat (n=ouarf@AStDenis-105-1-54-164.w80-8.abo.wanadoo.fr) |
| 22:21.30 | TripleFFFF | im returning 50 of tese crap pos' |
| 22:21.41 | dlynes_office | linksys routers? |
| 22:21.44 | dlynes_office | or pap2t-na? |
| 22:21.51 | TripleFFFF | pap2-t under a linksys.. |
| 22:21.57 | TripleFFFF | pap2vonafe under same linksys |
| 22:22.06 | TripleFFFF | xlite on pc under same linksys |
| 22:22.09 | dlynes_office | I'd just return the whole boatload and all the chinamen that came on it, too |
| 22:22.17 | TripleFFFF | yeha |
| 22:22.37 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 22:23.05 | *** join/#asterisk CvR (n=CvR@cw.global-player.com) |
| 22:23.55 | TripleFFFF | so yeah |
| 22:23.56 | TripleFFFF | <PROTECTED> |
| 22:23.56 | TripleFFFF | <PROTECTED> |
| 22:23.56 | TripleFFFF | <PROTECTED> |
| 22:23.57 | dlynes_office | I blame America! |
| 22:24.03 | TripleFFFF | its bush |
| 22:24.06 | TripleFFFF | i just know it |
| 22:24.24 | TripleFFFF | he just messed all pap's so we use normal lines and he can tap us |
| 22:25.06 | TripleFFFF | well.. thing is all points to linksys.. UNLESS hmmm yeah !!!!!!!!!!!!!!!!! |
| 22:25.09 | TripleFFFF | 1.2.9.1 |
| 22:25.19 | TripleFFFF | ill revert to something else brb |
| 22:29.20 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 22:33.34 | P4C0 | is there a way to customize the language of the voicebox? |
| 22:34.31 | CoffeeKid | w00t! got my hardphone to work over vpn! |
| 22:34.38 | P4C0 | I mean I can record a message but even like there's like a fixed voice telling "after the tone say your message and hangup or press the # key" or like that |
| 22:34.42 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:37.32 | TripleFFFF | voicemail(s1231231234 |
| 22:37.35 | TripleFFFF | S |
| 22:37.39 | TripleFFFF | s = silent pos |
| 22:37.51 | Dr-Linux | anybody ever use Cisco 7935 phone? |
| 22:41.24 | *** join/#asterisk morex (i=morex@host86-133-5-49.range86-133.btcentralplus.com) |
| 22:41.26 | morex | Hello all |
| 22:41.36 | morex | Anybody seen these before? |
| 22:41.46 | morex | Jul 5 20:19:46 WARNING[2917]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256) |
| 22:42.11 | morex | I'm getting them when I try to unpark a local channel connection through two sip phones... |
| 22:43.01 | CvR | Hi all! I've got some problems connecting Asterisk via a TE110p to a DMS100. Whenever I place a call pri debug shows the SETUP message being replied to with RELEASE COMPLETE (Cause 44) Similarly if I call the DID associated with the PRI I can see an incoming SETUP which * replies to with CALL PROCEEDING, but the dms100 then immediately sends a RELEASE (Cause 6). If * sends the call to a SIP phone * replies to the SETUP with CALL PROCEEDING and ALER |
| 22:43.01 | CvR | TING, but then immediately receives a RELEASE (Cause 6) from the dms100. --- I'm stuck. Does anybody have an idea what could be wrong? Does anybody have the time to look at the complete output of pri debug? |
| 22:43.15 | [TK]D-Fender | P4C0 : Yes * supports multiple languages |
| 22:43.57 | PerlStalker | Is it possible to disable voice mail forwarding in asterisk's v/m sub system? |
| 22:44.20 | [TK]D-Fender | PerlStalker : Yup. Go read the WIKI |
| 22:44.42 | CoffeeKid | I have a strange issue with music on hold... if more than 2 people are on hold, the music sound really terrible, anyone ever run into this? |
| 22:46.04 | P4C0 | [TK]D-Fender, how can I change it? for mailboxes... need to change vm-intro and auth-thankyou |
| 22:46.39 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 22:46.42 | PerlStalker | [TK]D-Fender: Ok. I must be dense today. Is the wiki just asterisk.org? |
| 22:46.58 | Agrajag- | gday. using the Echo command i hear myself about 0.5 - 0.75 seconds after i talk. this is on a local lan, ping time to the box running asterisk is about 0.140. is this normal? |
| 22:47.21 | CoffeeKid | PerlStalker: voip-info.org/asterisk-wiki ... i think |
| 22:47.35 | Agrajag- | im using u-law. using iaxcomm as the softphone |
| 22:47.54 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 22:50.35 | riddlebox | does anyone have a broadvoice number with an alternate number on that account? |
| 22:50.35 | CoffeeKid | Agrajag-: doesn't sound normal, should be realtime... but I've never used u-law |
| 22:53.01 | TripleFFFF | so .. |
| 22:53.13 | rob0 | ~wiki |
| 22:53.23 | TripleFFFF | ~wako |
| 22:53.59 | Agrajag- | CoffeeKid: it's about the same with GSM too |
| 22:54.30 | Agrajag- | i have no idea what the issue is though or how to fix it |
| 22:55.17 | *** part/#asterisk morex (i=morex@host86-133-5-49.range86-133.btcentralplus.com) |
| 22:55.20 | P4C0 | exactly where should I put the language=fr for voicebox? |
| 22:58.34 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 23:07.20 | knarfly | Anyone had any luck setting up X-Lite to connect to * from remote location? |
| 23:08.00 | Nugget | tens of people have, for sure. |
| 23:08.15 | [TK]D-Fender | P4C0 : Do that in your device setup or in the dialplan before you execute it |
| 23:08.33 | knarfly | Anyone care to help out a dummy who can't get it working...? |
| 23:08.38 | [TK]D-Fender | P4C0 : You need to download a language pack. Go get the professional ones made by June Wallack |
| 23:09.00 | [TK]D-Fender | P4C0 : it is a complete replacement for all the defaults |
| 23:09.30 | P4C0 | [TK]D-Fender, thanks, but exactly where? device setup? you mean sip.conf? |
| 23:10.17 | [TK]D-Fender | P4C0 : For SIP phones yes. You define the "default" language on an interface level first, then in your dialplan (like in an IVR). |
| 23:10.32 | P4C0 | [TK]D-Fender, thanks! |
| 23:10.59 | [TK]D-Fender | P4C0 : Whenevr * plays a sounds it looks in the /[language] folder for a native language version before falling back to the default (english) |
| 23:11.29 | Dr-Linux | question: |
| 23:11.30 | Dr-Linux | exten => _346XXXXXXXX,11,GotoIf($[ "${CHANNEL}" : "SIP"]?41) |
| 23:11.30 | Dr-Linux | exten => _346XXXXXXXX,12,GotoIf($[ "${CHANNEL}" : "195"]?21) |
| 23:12.14 | Dr-Linux | my first line works fine, but 2nd doesn't work. is there anyway i can use both string in the same line? |
| 23:12.22 | Dr-Linux | or anyway to make 2nd one work as well? |
| 23:12.52 | Dr-Linux | the second never returns true |
| 23:13.31 | *** join/#asterisk yanz7 (n=tysovka@ool-182f74a0.dyn.optonline.net) |
| 23:14.35 | yanz7 | hi everyone...i would like to ask few newbie questions about seting up trixbox |
| 23:14.56 | Dr-Linux | yanz7: #freepbx |
| 23:15.02 | yanz7 | thanks |
| 23:15.20 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:17.27 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 23:17.48 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 23:26.27 | CunningPike | Well, that bites - I was just about to install AsteriskGuru Query Stats when their web site went down |
| 23:29.46 | TripleFFFF | lol |
| 23:31.34 | *** join/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 23:35.14 | CunningPike | ~seen pop |
| 23:35.16 | jbot | pop <pop@194.88.109.42> was last seen on IRC in channel #debian, 686d 9h 4m 59s ago, saying: 'help! i've got a problem, when I execute: echo test | mail root -> then the command waits 3 seconds, and goes back on the prompt, how is that possible? i'm just mailing to the local user. when i kill named (which runs locally) it goes very fast, but i still need ... |
| 23:35.19 | CunningPike | ~seen popvoxdave |
| 23:35.20 | jbot | popvoxdave is currently on #asterisk (10h 4m 57s). Has said a total of 6 messages. Is idling for 2h 9m 30s, last said: 'pls be sure you get latest 1.21 rev if you are downloading fresh.'. |
| 23:36.08 | *** part/#asterisk knarfly (n=bwatson@12.42.132.26) |
| 23:36.47 | CunningPike | Does anyone know if AstManProxy helps with applications like HUDLite and FOP? |
| 23:37.25 | CunningPike | Is it a valid strategy to connect those applications to AstManProxy and have it make a single connection to asterisk? |
| 23:39.15 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 23:41.19 | mitcheloc | they already have their own proxies |
| 23:41.41 | CunningPike | OK - good, thanks |
| 23:42.24 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 23:42.27 | CunningPike | Just for fun try this: 'yum whatprovides php' :S |
| 23:44.50 | [TK]D-Fender | Dr-Linux : How about you NoOp that var before you go testing it.... |
| 23:45.07 | [TK]D-Fender | Dr-Linux : And maybe use the = sign instead of : |
| 23:45.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:50.33 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 23:53.51 | phigwork | Anyone seen this randomly come up in the cli or in logs? chan_sip.c: handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 23:53.57 | phigwork | dunno what that means |
| 23:54.02 | phigwork | tho it is just a warning |
| 23:58.31 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:59.57 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
