| 00:00.23 | [TK]D-Fender | LeddyHM, FYI: Construction underway |
| 00:01.21 | Uatec | when my phone tries to connect, i get the following in my asterisk cli: |
| 00:01.22 | Uatec | Apr 21 00:59:55 ERROR[5578]: chan_sip.c:6563 register_verify: Peer 'recsdesk' is trying to register, but not configuredas host=dynamic |
| 00:01.22 | Uatec | Apr 21 00:59:55 NOTICE[5578]: chan_sip.c:11126 handle_request_register: Registration from '<sip:recsdesk@192.168.2.4>' failed for '192.168.232.196' - Username/auth name mismatch |
| 00:01.25 | Uatec | i don't get it |
| 00:01.37 | LeddyHM | tk: woo woo |
| 00:01.49 | LeddyHM | to both, that's just what I was looking for (verbose) |
| 00:02.34 | Uatec | which one is the IP of the phone connecting in via SIP? |
| 00:02.43 | Uatec | and which one is the IP it's trying to match against? |
| 00:05.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:07.08 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:07.28 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:07.54 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:08.41 | rad07 | Are freePBX and Asterisk-GUI related in any way and are they compatible with Asterisk 1.4. Is TriBox developend independently of Asterisk and does it depent on the core Asterisk code? Can I compile something that is intended for Asterisk 1.4 on TriBox |
| 00:09.21 | pariah | Happy 4/20 Everyone! :) |
| 00:10.28 | florz | 4/20? But it's 4/21! |
| 00:15.43 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
| 00:19.31 | [TK]D-Fender | rad07, * GUI is completely different from FreePBX, but they are both meant for the same goal : configuring * via a web interface. |
| 00:19.35 | [TK]D-Fender | rad07, and : |
| 00:19.38 | [TK]D-Fender | ~trixbox |
| 00:19.50 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/ |
| 00:20.49 | [TK]D-Fender | rad07, FreePBX is currently only designed for use with * 1.2.X, and this reflects on its usage on Trixbox. |
| 00:21.12 | [TK]D-Fender | rad07, So no, don't trink of using 1.4 modules with Trixbox/FreePBX |
| 00:23.42 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:28.26 | THX2000 | Anyone have any clue why BLF will work on my Aastra phone when the extension im monitoring is 1XX, but not when its anything else eg: 2XX,7XX... |
| 00:30.11 | [TK]D-Fender | THX2000, the only side that can screw that up is *'s. Check your hints... |
| 00:31.40 | THX2000 | The hints look right, and the weirdest part is when I set the phone to monitor a bogus extension in the 100's i |
| 00:31.40 | THX2000 | get an error, anything else doesn't error out at all |
| 00:31.51 | kink0 | ~openpbx |
| 00:31.54 | jbot | somebody said openpbx was something that started off as an asterisk fork, but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in asterisk. see http://openpbx.org/ for more info, or join #openpbx |
| 00:32.02 | [TK]D-Fender | THX2000, pastebin your extensions.conf context with hints, and SIP debug of a failed subscribe |
| 00:32.12 | THX2000 | kk |
| 00:33.06 | kink0 | I waste three days trying to discriminate voice and fax (T38) calls unsuccessfully :( |
| 00:33.14 | kink0 | all help will be welcome. |
| 00:37.28 | kink0 | ~t38 |
| 00:37.31 | jbot | it has been said that t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 00:43.09 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 00:45.34 | THX2000 | Pastebin: Query failure: Can't open file: 'pastebin.MYI'. (errno: 145) |
| 00:45.36 | THX2000 | am i just newb or what? |
| 00:47.50 | Uatec | now my snom is no longer trying to register |
| 00:47.51 | [TK]D-Fender | www.pastebin.ca |
| 00:48.06 | kink0 | <PROTECTED> |
| 00:48.27 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 00:48.42 | THX2000 | http://pastebin.ca/450689 |
| 00:48.54 | kink0 | well, good night, tomorrow the battle will continue with T38 :( |
| 00:48.57 | THX2000 | extensions.conf is just a testbed right now so really basic |
| 00:49.02 | adorah | has anyone here experienced with chan_cellphone? |
| 00:51.07 | kink0 | adorah, no, I did use bluetooth, then I connect a mobile terminal to USB sound cards ( works !! ), and finally I bougth some 2N GSN gateways |
| 00:52.09 | kink0 | bye |
| 00:52.41 | adorah | kink0: that is the point: If it works u don't need a FXO trunk and can use a 10$ BT adaptor and any used cellular with BT support |
| 00:54.39 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:54.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:58.37 | JerJer[mobile] | split license - bah |
| 00:58.48 | JerJer[mobile] | if it wasn't for Mark nobody would have Asterisk to fork |
| 00:58.52 | JerJer[mobile] | in the first place |
| 01:00.18 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:00.21 | THX2000 | [TK]D-Fender: Did i totally jack it up? |
| 01:00.40 | *** join/#asterisk friedrich| (n=friedric@e177253012.adsl.alicedsl.de) |
| 01:00.40 | LeddyHM | yes |
| 01:00.41 | LeddyHM | shhhhh |
| 01:00.53 | THX2000 | haha, thought so :P |
| 01:01.18 | LeddyHM | I don't even know what the discussion is |
| 01:01.21 | LeddyHM | :) |
| 01:02.03 | THX2000 | and even u know i jacked it up :P |
| 01:02.48 | LeddyHM | exactly |
| 01:02.54 | LeddyHM | I'm THAT good! |
| 01:02.56 | [TK]D-Fender | THX2000, Running the 1.4 devstate patch? |
| 01:03.32 | THX2000 | Probably not...just the stock 1.4.2 |
| 01:05.05 | [TK]D-Fender | THX2000, ok, well I'm not sure on that formats operation under the base... |
| 01:06.34 | THX2000 | umm...the what now? |
| 01:35.46 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 01:47.38 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 01:48.08 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 01:56.46 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 01:58.41 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 02:00.09 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 02:01.09 | *** join/#asterisk Phoe (n=blank@mail.jetaservices.com) |
| 02:02.23 | Phoe | I want to use the values stored in the CALLTRACE family in AstDB, but I dont know how to find the key associated with my call... Does that make sense? |
| 02:05.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:12.16 | *** join/#asterisk hijacked (i=D8Uz@cerebus.clandestineresearch.com) |
| 02:17.56 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:22.49 | *** join/#asterisk plasmid2 (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 02:23.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 02:29.34 | LeddyHM | Is there a way to change the callerid on forwards? |
| 02:29.57 | LeddyHM | or is that done at the phone level and just uses the global id setup in *? |
| 02:33.06 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 02:33.19 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 02:33.52 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 02:34.43 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:42.32 | *** join/#asterisk SwK[Work] (n=SwK@12-214-191-109.client.mchsi.com) |
| 02:44.31 | JerJer[mobile] | LeddyHM: how are you doing a forward? dialplan based or on the sip phone directly ? |
| 02:46.22 | *** join/#asterisk brussel_ (n=brussel@cpe-72-130-172-213.san.res.rr.com) |
| 02:48.07 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:50.10 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 02:50.36 | AJaymn | what Barge is available for SIP ? there ZAPBarge but is there SIPBarge or somthing? I cant find the doc |
| 02:51.50 | wunderkin | chanspy although it has sucked in the past for me, i have not used it for awhile |
| 02:52.22 | AJaymn | thats it :) thanks |
| 02:53.21 | JerJer[mobile] | or just sniff the RTP packets :) |
| 02:53.30 | JerJer[mobile] | ...all else fails, that is |
| 02:55.27 | AJaymn | :P |
| 02:56.09 | *** join/#asterisk _10nix_ (n=hyjnx@ool-4577788e.dyn.optonline.net) |
| 03:04.08 | _10nix_ | i have a bit of an extensions.conf question, i was wondering if anyone oculd give me a hand with |
| 03:09.15 | LeddyHM | jer: phone based |
| 03:21.57 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:22.28 | *** join/#asterisk dj-fu (i=aj@unaffiliated/dj-fu) |
| 03:24.18 | *** join/#asterisk friedrich| (n=friedric@e177251006.adsl.alicedsl.de) |
| 03:35.20 | *** join/#asterisk annonimous (n=annonimo@189.136.62.254) |
| 03:35.24 | annonimous | good night |
| 03:36.46 | annonimous | when i call with a sip trunk i can establish communication but if i hang up the call the sip trunk answer with an "all circuits are busy now..." how can i fix this? |
| 03:37.07 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:41.15 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 03:41.26 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 03:46.51 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:47.37 | *** join/#asterisk rmayorga (n=rmayorga@168.243.73.11) |
| 03:48.50 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 03:49.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 03:50.55 | *** join/#asterisk voltagex (n=voltagex@124-254-120-152-dsl.ispone.net.au) |
| 03:51.11 | voltagex | is there a sip standard for caller id or does Asterisk just do the best it can? |
| 03:52.53 | *** join/#asterisk bmg505 (n=leon@196.209.180.80) |
| 03:53.17 | *** join/#asterisk beave (n=beave@bundy.vistech.net) |
| 04:00.51 | Jameno123 | hehe |
| 04:00.51 | Jameno123 | "Ôá„Û–9#¿!!1Ikerin " |
| 04:00.51 | Jameno123 | nice information in the queue_log :) |
| 04:01.01 | Jameno123 | for the "AGENT" field |
| 04:01.33 | Jameno123 | err :) overflow i guess? |
| 04:04.31 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 04:09.43 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 04:12.19 | annonimous | when i call with a sip trunk i can establish communication but if i hang up the call the sip trunk answer with an "all circuits are busy now..." how can i fix this? |
| 04:12.41 | *** part/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 04:13.51 | *** join/#asterisk kerlnel20 (n=kerlnel2@210.238.181.187) |
| 04:13.53 | kerlnel20 | hi there |
| 04:14.01 | kerlnel20 | can i customized wengo? |
| 04:14.29 | kerlnel20 | i am planning to create an app that will connect to my asterisk |
| 04:14.37 | kerlnel20 | and make calls and conferences |
| 04:14.39 | kerlnel20 | any ideas? |
| 04:17.51 | kerlnel20 | any ideas? |
| 04:19.12 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-73.socal.res.rr.com) |
| 04:38.42 | kerlnel20 | can i customized wengo? |
| 04:38.44 | kerlnel20 | i am planning to create an app that will connect to my asterisk |
| 04:38.49 | kerlnel20 | any ideas? |
| 04:39.10 | *** join/#asterisk killfill (n=killfill@201.215.221.113) |
| 04:39.12 | killfill | hi |
| 04:39.14 | killfill | what is VPM? |
| 04:50.03 | joe_f | kerlnel20: ask better questions |
| 04:50.04 | joe_f | ha |
| 04:50.25 | *** join/#asterisk hfb (n=hfb@adsl-75-39-161-90.dsl.irvnca.sbcglobal.net) |
| 04:52.28 | kerlnel20 | is there any free dll of or ocx so that i can connect to asterisk using my app |
| 04:52.58 | *** join/#asterisk jarg (n=jarg@189.157.162.135) |
| 04:53.08 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 04:56.13 | Qwell | kerlnel20: STOP ASKING THE SAME QUESTIONS REPEATEDLY |
| 04:56.20 | Qwell | I've told you that *DOZENS* of times now. |
| 05:00.16 | *** join/#asterisk McFrosty (n=no@sc-18-24.sckans.edu) |
| 05:01.47 | kerlnel20 | you dont tell me anything Qwell |
| 05:02.04 | kerlnel20 | so please if you have idea maybe you can shed light |
| 05:02.11 | Mavvie | kerlnel20: then listen to my answer: setup a tcp session towards the manager interface. |
| 05:02.27 | Qwell | kerlnel20: Remind me how many times I've kicked and/or banned you from various channels? |
| 05:03.20 | kerlnel20 | i am just asking for any ocx or dll |
| 05:03.20 | Qwell | ask an intelligent, well crafted question, ONCE, and wait for an answer |
| 05:03.21 | kerlnel20 | so that i can use |
| 05:03.21 | JT | you are asking for silencing |
| 05:03.21 | kerlnel20 | for this little project |
| 05:03.21 | JT | i suggest you pay someone with a clue |
| 05:03.21 | kerlnel20 | there is no meaning if you get harsh |
| 05:03.22 | JT | that makes no sense |
| 05:03.36 | Mavvie | kerlnel20: setup a tcp session towards the manager interface and you can do all you want. |
| 05:03.43 | kerlnel20 | how MaartenB |
| 05:03.46 | killfill | where does zaptel installs zaptel.h? |
| 05:03.47 | kerlnel20 | how Mavvie |
| 05:03.56 | Qwell | killfill: /usr/include/zaptel/ |
| 05:04.02 | killfill | thanks |
| 05:04.02 | Mavvie | kerlnel20: bind, socket, connect, and then read/write. |
| 05:04.14 | kerlnel20 | any ocx? |
| 05:04.25 | JT | stfu about the ocx already |
| 05:04.26 | Mavvie | nah, just tcp |
| 05:04.28 | JT | sheesh |
| 05:04.36 | Qwell | Mavvie: trust me when I say this... it would be in your best interest to just stop there. :) |
| 05:11.57 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 05:15.47 | mattwj2005 | anyone around? |
| 05:15.52 | mattwj2005 | anyone want to chat? |
| 05:16.14 | Mavvie | yeah. a/s/l please |
| 05:16.33 | mattwj2005 | sure |
| 05:16.42 | mattwj2005 | 25/m/Minneapolis |
| 05:16.47 | mattwj2005 | you Mavvie? |
| 05:17.03 | Mavvie | Minneapolis? |
| 05:17.09 | Mavvie | Only two things come from Minneapolis... |
| 05:17.15 | mattwj2005 | yeah? |
| 05:17.15 | Mavvie | Bulls and gayman. |
| 05:17.30 | mattwj2005 | Shows his horns |
| 05:17.31 | Mavvie | I don't see any horns, so are you a gayman? |
| 05:17.40 | mattwj2005 | nice Family Guy joke ;) |
| 05:18.17 | mattwj2005 | I blew $40 |
| 05:18.38 | Mavvie | actually, it's from a military school movie. |
| 05:18.45 | Mavvie | long before family guy got married. |
| 05:18.59 | mattwj2005 | gambled....at supper.....had a beer....and drank some soda |
| 05:19.07 | mattwj2005 | nice |
| 05:19.34 | Mavvie | naah, Minneapolis is nice. |
| 05:19.55 | Mavvie | I once dated a girl from that state. |
| 05:20.00 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 05:20.05 | mattwj2005 | cool |
| 05:20.09 | mattwj2005 | asl? |
| 05:20.12 | Mavvie | One night she said "you may kiss me where is smells strange" |
| 05:20.19 | Mavvie | so we went to her state. |
| 05:20.30 | mattwj2005 | lol |
| 05:20.36 | mattwj2005 | nice |
| 05:21.05 | Mavvie | are you sure you want to keep talking to me? |
| 05:21.29 | mattwj2005 | it doesn't matter |
| 05:23.22 | McFrosty | Does Asterisk allow you to make free PC to landline calls? at college and just need a way to call home |
| 05:23.40 | mattwj2005 | not really |
| 05:23.53 | mattwj2005 | you have to pay for landline calls |
| 05:23.58 | mattwj2005 | I like teliax.com |
| 05:24.09 | McFrosty | yeah I see that every where...was hoping this was the cure |
| 05:24.25 | *** part/#asterisk killfill (n=killfill@201.215.221.113) |
| 05:24.34 | mattwj2005 | you can find a few services for free 800 numbers |
| 05:24.37 | mattwj2005 | that is about it |
| 05:24.50 | mattwj2005 | there are also some free conference rooms |
| 05:25.36 | McFrosty | hmmm ok will look around some more...thx :) |
| 05:26.01 | mattwj2005 | okay cool....I am a hobbist. let me know if I can help |
| 05:26.57 | McFrosty | I use windows mostly but been messing with kubuntu and I remembered hearing of asterisk so just was checking in :) |
| 05:28.22 | mattwj2005 | are you a hobbist or are you working on a cooporate solution? |
| 05:28.57 | *** join/#asterisk joe--f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 05:29.04 | mattwj2005 | here is a list of some free stuff you can do |
| 05:29.05 | mattwj2005 | http://wiki.ekiga.org/index.php/Fun_Numbers |
| 05:29.32 | joe--f | where's a good doc on installing asterisk and testing it with x-lite? |
| 05:29.46 | joe--f | http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html has driven me crazy |
| 05:30.04 | McFrosty | I guess I would be conisdered a hobbist...unemployed college student about to move to a rental house and trying to find a way to call ppl for free... :P I think it's a dead end search tho |
| 05:30.31 | JT | you can call them for free if they use voip |
| 05:30.35 | JT | but not landlines |
| 05:30.48 | McFrosty | Back home they have no internet :/ |
| 05:30.56 | mattwj2005 | oh man that sucks |
| 05:31.02 | McFrosty | Yeah |
| 05:31.10 | mattwj2005 | otherwise you could setup an ATA |
| 05:31.16 | mattwj2005 | if they had a descent broadband |
| 05:32.09 | mattwj2005 | teliax.com has numbers you family can call locally.....but it is $25 per month |
| 05:32.16 | mattwj2005 | I know what it is like being in college |
| 05:33.24 | McFrosty | I checked out there site....might consider it |
| 05:34.24 | mattwj2005 | also have you checked out www.voip-info.org ? |
| 05:35.00 | joe--f | should 'telnet <asterisk-server-ip> 5060' be .. um.. not timing out? ha |
| 05:35.20 | McFrosty | Nope but will now.. :P ... can't sleep and getting ready to start moving into the rental house so just started googling open source voip hoping it could help |
| 05:36.00 | mattwj2005 | the guys in the chat room are great |
| 05:36.05 | mattwj2005 | very helpful people :) |
| 05:37.45 | GreyFoxx | JoelSolanki: ...... actually normally I would expect you to get a connection refused since telnet uses TCP, and asterisk is listening on UDP 5060 |
| 05:37.52 | GreyFoxx | err that was for joe--f |
| 05:38.45 | mattwj2005 | McFrosty one more thing |
| 05:38.52 | mattwj2005 | www.skype.com |
| 05:39.12 | mattwj2005 | that is the cheapest voip around...$29 per year unlimited outgoing in the US and Canada |
| 05:39.34 | McFrosty | yeah, I have used skype before but just chatting for a job I used to have...never used it to call just IM |
| 05:42.21 | McFrosty | mattwj2005 have you used skype to call ppl? just curious on if there is any lag or if it's just like using a normal landline phone |
| 05:42.27 | mattwj2005 | yup |
| 05:42.35 | mattwj2005 | depends |
| 05:42.50 | mattwj2005 | sometimes the quality is good....sometimes it sucks |
| 05:43.30 | mattwj2005 | I have never had an incoming number |
| 05:44.07 | McFrosty | Okay, thx for all the suggestions :) |
| 05:44.13 | mattwj2005 | you can use Skype by the minute too |
| 05:44.43 | mattwj2005 | your welcome .... I graduated Dec 2005.....I know how it sucks to be a poor college student :) |
| 05:45.17 | McFrosty | lol yeah... i will be graduating this coming Dec |
| 05:45.44 | mattwj2005 | congrats |
| 05:45.51 | mattwj2005 | what will you degree be in? |
| 05:45.57 | McFrosty | Computer Science |
| 05:46.08 | mattwj2005 | man your a smart dude |
| 05:46.09 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 05:46.19 | McFrosty | lol |
| 05:46.28 | mattwj2005 | Computer Science is tough....I took a few Computer Science courses for fun |
| 05:47.14 | McFrosty | I personally think the CS program here is horrible.. I made the mistake of coming because I got a small football scholarship |
| 05:47.27 | McFrosty | The only math required was college algebra |
| 05:47.38 | mattwj2005 | BS? |
| 05:47.58 | mattwj2005 | Bachelor of Science? |
| 05:48.17 | McFrosty | Not totally sure :P |
| 05:48.30 | mattwj2005 | 2 year degree or 4 year? |
| 05:48.33 | McFrosty | 4 |
| 05:48.50 | mattwj2005 | dang |
| 05:48.58 | mattwj2005 | I had a degree in Telecommunications |
| 05:49.04 | mattwj2005 | *have |
| 05:49.09 | mattwj2005 | I had to take Calcus |
| 05:49.10 | mattwj2005 | :S |
| 05:49.26 | *** join/#asterisk af_ (n=getsmart@81-174-47-36.f5.ngi.it) |
| 05:49.29 | mattwj2005 | good ole C- |
| 05:49.47 | McFrosty | We have had 4 different instructors teaching CS .. new one every year I been here... Luckily the instructors got better each time and the one we have now is awesome but it's a little late for me now :/ |
| 05:49.57 | McFrosty | It's a very small school |
| 05:50.46 | McFrosty | There redoing the whole program and I think calc 1 is required now |
| 05:52.37 | mattwj2005 | my college has lost a couple of people this year |
| 05:54.34 | McFrosty | I think this college is up to 18:1 student teacher ratio |
| 05:54.46 | mattwj2005 | that is descent |
| 05:55.11 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:55.23 | mattwj2005 | speaking of college man I feel bad for Virginia tech students |
| 05:55.32 | McFrosty | Most of my CS classes have 4 of us.. we are the only "classic" CS majors |
| 05:55.43 | mattwj2005 | lol |
| 05:56.04 | McFrosty | a lot of digital arts CS majors tho |
| 05:56.28 | mattwj2005 | any good looking calls? |
| 05:56.33 | mattwj2005 | *gals? |
| 05:57.39 | McFrosty | Yeah and most I would guess have some money .... again a reason why i wish i didn't come here... 22-24k/year i think ... state school in Texas was 13k I think |
| 05:58.23 | mattwj2005 | ouch! |
| 05:58.43 | McFrosty | http://www.sckans.edu/digital/ |
| 05:58.52 | McFrosty | computer science part of the schools web site |
| 06:00.00 | mattwj2005 | I got an Associate Degree and a Bachlor's Degree |
| 06:00.09 | mattwj2005 | I owe able 25k to 30k |
| 06:00.14 | mattwj2005 | *about |
| 06:00.55 | McFrosty | lol, I think I cover maybe about half of the 22k in grants and scholarships...rest is all loans....it's gonna hurt when I get out |
| 06:01.59 | mattwj2005 | well wait until you get a 2 week paycheck with 4 digitals for dollar signs |
| 06:02.19 | mattwj2005 | $####.## in other words |
| 06:02.28 | McFrosty | That will be nice :) |
| 06:04.01 | mattwj2005 | college is expensive |
| 06:04.14 | mattwj2005 | but it pays in the long run |
| 06:04.24 | McFrosty | I agree to that |
| 06:04.32 | mattwj2005 | lets say you owe $40,000 |
| 06:04.43 | mattwj2005 | people spend that on car |
| 06:04.49 | McFrosty | lol |
| 06:04.52 | McFrosty | yeah that they do |
| 06:04.59 | mattwj2005 | in 5 to 10 years....you have a nice hunk of metal |
| 06:05.00 | mattwj2005 | :) |
| 06:06.53 | McFrosty | I assume you run linux? |
| 06:07.15 | mattwj2005 | depends what computer your referring to |
| 06:07.21 | McFrosty | ahhh |
| 06:07.23 | mattwj2005 | I have a multiple os network |
| 06:07.27 | mattwj2005 | at home |
| 06:07.30 | McFrosty | nice |
| 06:07.39 | McFrosty | I have my PC and my laptop |
| 06:08.11 | mattwj2005 | if you have a nice speedy computer...you could always try AsteriskNOW using vmware |
| 06:08.48 | McFrosty | I really don't know what Asterisk does |
| 06:09.04 | McFrosty | I read on the site about PBX or something but I never heard of it until tongiht |
| 06:09.10 | McFrosty | tonight* |
| 06:09.13 | mattwj2005 | okay |
| 06:09.52 | mattwj2005 | basically you take phones -> asterisk server -> internet -> wherever |
| 06:11.09 | mattwj2005 | wherever could be to a service that provides phone service aka lan line |
| 06:11.24 | mattwj2005 | it could be to friends with other asterisk servers |
| 06:11.34 | mattwj2005 | or some of those fun services I showed you |
| 06:12.11 | JT | lan line? |
| 06:12.33 | mattwj2005 | the pstn |
| 06:12.51 | joe--f | how can i completely/cleanly remove asterisk? 'apt-get remove asterisk', isnt doing it completely |
| 06:12.51 | JT | lanD? |
| 06:13.57 | JT | uninstalling usually never removes everything |
| 06:14.08 | mattwj2005 | yeah it sucks |
| 06:14.16 | mattwj2005 | you can always reinstall |
| 06:14.56 | JT | mattwj2005: you meant land line? |
| 06:15.13 | mattwj2005 | you know your mom's house, your dad |
| 06:15.20 | mattwj2005 | dad's house, cell phone, etc |
| 06:15.36 | JT | so the PSTN |
| 06:15.42 | JT | but not lan line :) |
| 06:15.43 | mattwj2005 | yeah |
| 06:15.51 | mattwj2005 | sorry I had a beer tonight :S |
| 06:15.55 | JT | hah |
| 06:16.03 | joe--f | mattwj2005: i'm trying to to 'apt-get install asterisk', but im getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 06:16.32 | mattwj2005 | what happens when you do apt-get install asterisk |
| 06:16.37 | mattwj2005 | have you tried restarting your computer? |
| 06:16.41 | joe--f | yeah |
| 06:17.00 | joe--f | 'apt-get install asterisk' goes fine, but at the very end i get that error ^^ |
| 06:17.38 | joe--f | i get that also when doing 'asterisk -r'.. i had it working earlier, but got impatient and thought i screwed the config so i wanted to restart the install.. bad idea now.. :( |
| 06:18.07 | joe--f | i'll try to do a full autoremove and restart and install again.. sorry ha |
| 06:18.10 | mattwj2005 | try restarting |
| 06:18.15 | joe--f | ya... :( |
| 06:18.30 | mattwj2005 | what does your /etc/apt/sources.list look like? |
| 06:20.47 | joe--f | mattwj2005: first line: deb http://de.archive.ubuntu.com/ubuntu/ edgy main restricted |
| 06:20.58 | joe--f | restarted and it's still crapped |
| 06:22.01 | mattwj2005 | do you have everything uncommented for in the default /etc/apt/source.list ? |
| 06:22.07 | mattwj2005 | do a apt-get update |
| 06:22.16 | mattwj2005 | apt-get install asterisk |
| 06:22.21 | mattwj2005 | that should take care of it |
| 06:22.23 | mattwj2005 | oops |
| 06:22.28 | mattwj2005 | sorry I forget ubuntu |
| 06:22.33 | mattwj2005 | sudo apt-get update |
| 06:22.37 | mattwj2005 | sudo apt-get install asterisk |
| 06:24.48 | joe--f | i havent touched anything in the /etc/apt/sources.list, so im assuming its defaulted |
| 06:25.15 | joe--f | im getting a couple errors when doing 'asterick -cvvvvvv...', this one might be interesting? i dunno : WARNING[4455]: loader.c:414 __load_resource: chan_oss.so: load_module failed, returning -1 |
| 06:25.29 | joe--f | wait thats the wrong one |
| 06:25.31 | mattwj2005 | generally I comment out the cdrom deb and deb-src entries |
| 06:25.35 | joe--f | k |
| 06:25.38 | mattwj2005 | and uncomment everything else |
| 06:25.42 | JT | uninstall it |
| 06:25.51 | JT | wipe the modules dir |
| 06:25.53 | mattwj2005 | I don't know if I am helping |
| 06:25.58 | JT | then reinstall it |
| 06:26.28 | joe--f | JT: /usr/lib/asterisk/modules/ |
| 06:26.29 | joe--f | ? |
| 06:26.45 | JT | yes,if it has modules in it |
| 06:26.59 | joe--f | i think i've tried this too, but i'll do it again |
| 06:27.57 | JT | try to start asterisk with asterisk -cvvvvv |
| 06:28.39 | joe--f | nope... |
| 06:28.45 | joe--f | <PROTECTED> |
| 06:28.46 | joe--f | Apr 21 01:32:06 WARNING[5049]: loader.c:414 __load_resource: chan_oss.so: load_module failed, returning -1 |
| 06:28.46 | joe--f | Apr 21 01:32:06 WARNING[5049]: loader.c:554 load_modules: Loading module chan_oss.so failed! |
| 06:29.07 | joe--f | i dont have any conf's in my /etc/asterick/ anymore.. when i reinstall, it's not loading that back up |
| 06:29.30 | joe--f | s/asterick/asterisk/ |
| 06:29.54 | joe--f | hahah |
| 06:29.58 | joe--f | nice bot |
| 06:30.58 | JT | have you reinstallled? |
| 06:30.59 | joe--f | <PROTECTED> |
| 06:31.08 | joe--f | JT: i did exactly how you said it.. |
| 06:31.26 | JT | so is asterisk actually starting? |
| 06:31.45 | joe--f | 'apt-get autoremove asterisk', 'rm -rf /usr/lib/asterisk/modules', 'apt-get install asterisk' |
| 06:33.32 | joe--f | http://hashbin.com/38a.html |
| 06:33.39 | joe--f | theres the log of the install |
| 06:33.45 | joe--f | and 'asterisk -r' |
| 06:35.22 | *** join/#asterisk brussel_ (n=brussel@72.130.172.213) |
| 06:37.04 | joe--f | asterisk -cvvvvv --> http://hashbin.com/38d.html |
| 06:38.58 | joe--f | JT: no its not starting, it was before i broke it.. |
| 06:39.31 | mattwj2005 | it looks like it doesn't like oss |
| 06:40.27 | joe--f | root@lp:/# asterisk -r |
| 06:40.27 | joe--f | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 06:40.27 | joe--f | root@lp:/# ls /var/run/asterisk/asterisk.ctl |
| 06:40.27 | joe--f | /var/run/asterisk/asterisk.ctl |
| 06:40.28 | *** join/#asterisk jwalch (n=jwal@gate.nwe.de) |
| 06:40.30 | joe--f | ha |
| 06:40.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:41.42 | jwalch | corydon-w: just for your interest, i got the configuration i was describing yesterday (TE210P as ppp dial-in router with ZapRAS asterisk frontend) up and running. have been dialing in with cisco, windows etc.. at the same time, no problem |
| 06:42.26 | jwalch | corydon-w: so if there is any place for a how-to i can do that if it is wanted |
| 06:42.28 | JT | joe--f: put noload directives in modules.conf for pbx_dundi.so |
| 06:42.49 | JT | and either put a noload directive for chan_iax2.so or make an iax.conf file |
| 06:42.52 | JT | as it doesn't exist |
| 06:44.56 | joe--f | all of the /etc/asterick/*.conf's are gone.. they arent being installed anymore with apt-get.. :( |
| 06:45.27 | JT | how dumb |
| 06:45.43 | joe--f | somethings preventing it somewhere.. |
| 06:46.27 | JT | who knows |
| 06:46.36 | JT | i never use package management to install asterisk |
| 06:47.07 | joe--f | yeah.. i was being impatient today |
| 06:51.14 | justdave | Playback(you-sound-impatient) |
| 06:58.29 | voltagex | why would a hangup completely fail? Also is there a way to run commands after hangup without using the h extension? |
| 07:01.28 | voltagex | <PROTECTED> |
| 07:01.53 | *** join/#asterisk Strom_M (n=strom@217.41.233.36) |
| 07:02.20 | voltagex | hi strom |
| 07:02.30 | JT | you'll need to explain your scenario a little more |
| 07:02.32 | voltagex | maybe you can help me...this is a really strange problem |
| 07:02.35 | voltagex | oops |
| 07:02.37 | voltagex | hi JT |
| 07:05.59 | voltagex | JT: I'm trying to implement a callback for PSTN calls... http://pastebin.ca/451056 |
| 07:07.49 | JT | or you could just use my method, that works |
| 07:08.00 | voltagex | which is? |
| 07:08.32 | JT | using System to run a bash script that waits for 5 seconds, then copies a premade callfile to the spool dir |
| 07:08.48 | voltagex | but that's not the problem, it doesn |
| 07:08.51 | voltagex | doesn |
| 07:08.54 | voltagex | argh |
| 07:09.00 | voltagex | doesn't actually hang up |
| 07:09.21 | JT | what technology does the call come in over? |
| 07:09.54 | voltagex | PSTN --> sip provider |
| 07:10.42 | JT | so what do you mean it doesn't hangup? |
| 07:11.52 | voltagex | as in, Hangup() is executed, but the phone I'm using continues to ring |
| 07:12.20 | JT | what phone, you're going to have to be more specific about what fits into where |
| 07:13.40 | voltagex | I'm attempting to call asterisk using a POTS line with a cordless phone, to an Australian DID number...that works - asterisk sees the call and should then hang up. Hangup() runs but the POTS phone doesn't make the "beep beep beep" hungup noise |
| 07:14.10 | JT | try calling from a mobile phone, see if it acts the same |
| 07:14.34 | voltagex | if all that is in the dialplan is Hangup, it works |
| 07:15.00 | voltagex | <PROTECTED> |
| 07:15.04 | voltagex | works fine |
| 07:15.04 | JT | the POTS line provides a ringing indication forever, or what? |
| 07:15.13 | voltagex | yes, ringing indication forever |
| 07:15.16 | JT | hmm |
| 07:15.54 | JT | what on earth is this 09xxxx/0244xxxx,3,Wait,1 |
| 07:16.17 | voltagex | a wait line? |
| 07:16.27 | JT | you mean Wait(1) |
| 07:16.29 | voltagex | where've I screwed up |
| 07:16.30 | JT | not Wait,1 |
| 07:16.30 | voltagex | oh, shit |
| 07:16.53 | JT | fwiw, there's probably no need to wait |
| 07:17.46 | *** join/#asterisk shell (n=mmoonntt@71-211-66-25.phnx.qwest.net) |
| 07:18.27 | voltagex | <PROTECTED> |
| 07:18.27 | voltagex | <PROTECTED> |
| 07:18.27 | voltagex | <PROTECTED> |
| 07:18.46 | JT | cool, so does it work? |
| 07:18.51 | voltagex | no |
| 07:19.04 | JT | i see |
| 07:19.11 | voltagex | at the time that Gotoif is run, the pots line is still ringing |
| 07:19.15 | voltagex | so, wtf? |
| 07:19.48 | JoelSolanki | Hi all |
| 07:20.14 | *** join/#asterisk Zdrulio (n=sux_@82.119.72.130) |
| 07:20.46 | JoelSolanki | i have a call center client who wants me to disable early media on our side. |
| 07:20.55 | JoelSolanki | i m using asterisk. but i dont know what to do in asterisk. |
| 07:21.00 | JoelSolanki | can any body give me hints. |
| 07:21.43 | JoelSolanki | yesterday i searched and found that canreinvite & progressinband |
| 07:21.44 | JT | voltagex: have you tried it with just one itsp? |
| 07:22.03 | voltagex | JT: forgive me but what is an ITSP |
| 07:22.04 | JoelSolanki | can it be done in asterisk ? |
| 07:22.08 | JT | ~itsp |
| 07:22.20 | jbot | i heard itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 07:22.47 | voltagex | yes, with my only DID provider |
| 07:23.01 | voltagex | the problem is on asterisk's end, it isn't hanging up! |
| 07:23.07 | JT | voltagex: have you checked with sip debug that the correct messages are being sent? |
| 07:23.11 | JT | not necessarily |
| 07:23.18 | JT | it says it's hanging up |
| 07:24.56 | voltagex | now, where did all those 401 unauthorised's conme from??! |
| 07:29.27 | voltagex | asterisk doesn't appear to be sending a hangup at all... |
| 07:33.42 | voltagex | JT: the solution would be to have asterisk ignore " exited non-zero " |
| 07:34.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:36.23 | voltagex | any ideas JT? |
| 07:36.43 | JT | what |
| 07:36.52 | JT | ignore exited non zero, huh |
| 07:37.21 | voltagex | well when the caller hangs up the extension "exits" so nothing else gets executed |
| 07:37.43 | voltagex | <PROTECTED> |
| 07:38.04 | JT | err |
| 07:38.14 | JT | so it's not the hangup from your dialplan? |
| 07:38.21 | JT | it's help to be a LOT CLEARER in future |
| 07:38.26 | voltagex | what? |
| 07:38.35 | JT | it'd |
| 07:38.41 | voltagex | no...the hangup from my dialplan doesn't work |
| 07:38.48 | JT | grr |
| 07:38.52 | voltagex | so if I don't hang up the line rings forever |
| 07:38.53 | JT | state that clearly infuture |
| 07:38.59 | JT | bah |
| 07:39.00 | voltagex | I did! |
| 07:39.07 | JT | nope |
| 07:39.24 | JT | you should've specifically said that that Hangup we were seeing was from YOU hanging up the phone |
| 07:39.32 | JT | i thought you were just keeping the phone off hook |
| 07:39.56 | JT | we can't read your mind |
| 07:40.23 | voltagex | 1. I call asterisk. 2. Asterisk detects call and starts running dialplan. 3. Hangup runs and nothing happens. 4. POTS phone is still ringing. 5. I hang up POTS phone. 6. Asterisk sees ME hanging up and stops the dialplan. 7. Cry. |
| 07:40.28 | voltagex | sorry. |
| 07:40.35 | voltagex | above is as clear as I can be. |
| 07:42.20 | JT | is there two Hangups coming up on the CLI? |
| 07:42.23 | voltagex | no |
| 07:42.28 | voltagex | ...hang on |
| 07:42.38 | JT | then why do you say 3. Hangup runs and nothing happens. |
| 07:42.39 | voltagex | yes. but the one from asterisk doesn't work! |
| 07:42.56 | voltagex | yes, the Hangup from the asterisk dialplan is shown |
| 07:43.13 | voltagex | but it doesn't work - the pots line just rings and rings |
| 07:43.13 | JT | ok i'm totally confused |
| 07:43.17 | voltagex | so am I |
| 07:43.58 | JT | you keep contradicting yourself/being unclear |
| 07:44.04 | JT | two hangups or not? |
| 07:44.12 | voltagex | yes, 2 hangups |
| 07:44.12 | JT | if not, which one is it, and how do you know |
| 07:44.35 | JT | pastebin it |
| 07:44.38 | JT | the whole call |
| 07:44.54 | voltagex | with sip debug or not? |
| 07:45.40 | JT | may aswell switch it on i guess |
| 07:48.40 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 07:50.01 | voltagex | http://pastebin.ca/451095 |
| 07:53.10 | voltagex | with no debug: http://pastebin.ca/451096 |
| 07:57.14 | JT | voltagex: i only see 1 hangup |
| 07:57.27 | voltagex | isn't that exited non zero a hangup? |
| 07:57.46 | JT | it doesnt say hangup |
| 07:57.51 | voltagex | ok |
| 07:58.04 | voltagex | I assumed it was...sorry. |
| 08:00.01 | JT | all i was concerned about was the execution of the hangup command |
| 08:00.30 | voltagex | well that first one executes but is not received by the pots phone, that's all I know. |
| 08:01.55 | JT | it doesn't appear to send the hangup to sip |
| 08:02.13 | voltagex | so what am I doing wrong? |
| 08:02.16 | JT | so for shits and giggles, lets try waiting for 5 seconds before hangup |
| 08:02.19 | JT | Wait(5) |
| 08:04.13 | voltagex | <PROTECTED> |
| 08:04.13 | voltagex | <PROTECTED> |
| 08:04.13 | voltagex | <PROTECTED> |
| 08:04.23 | voltagex | but pots line still rings. |
| 08:04.56 | JT | try putting an Answer before the Wait |
| 08:05.29 | voltagex | err...crap, won't that make the call cost money? |
| 08:05.57 | JT | maybe, but it's just for testing |
| 08:06.56 | voltagex | damn, calling via mobile, it hangs up correctly now |
| 08:07.17 | JT | with which dialplan, Answer Wait Hangup? |
| 08:07.22 | voltagex | wait hangup |
| 08:07.59 | JT | right so might just be POTS sucking |
| 08:09.09 | voltagex | it's something to do with the goto line |
| 08:09.52 | voltagex | without the goto line, the hangup works fine on both pots and mobile |
| 08:10.06 | JT | err the callerid perhaps? |
| 08:10.32 | voltagex | that line is triggering correctly |
| 08:10.57 | voltagex | in any case, the hangup should happen before the goto |
| 08:13.34 | voltagex | maybe I should use DeadAGI to write a call file |
| 08:14.07 | JT | my method works just fine, but i've never called it from a POTS line |
| 08:14.35 | voltagex | can you pastebin your method if you didn't already? |
| 08:18.36 | JT | http://pastebin.ca/451125 |
| 08:18.39 | JT | a little simpler |
| 08:19.23 | mattwj2005 | well I am off to bed |
| 08:19.27 | mattwj2005 | have a good night everyone |
| 08:19.33 | mattwj2005 | I'll talk to you later |
| 08:19.35 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 08:20.18 | *** join/#asterisk saftsack (n=oliver@p54A7DA51.dip.t-dialin.net) |
| 08:20.32 | voltagex | dinner time here, thanks JT |
| 08:20.54 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:21.00 | *** join/#asterisk ecze (n=ecze@eczema.ecze.com) |
| 08:21.20 | ecze | hi all, |
| 08:21.42 | ecze | I have a question about asterisk 1.2 and isdn passive cards |
| 08:22.05 | ecze | I currently run a kernel 2.4.27 |
| 08:22.32 | ecze | and asterisk 1.0 was working pretty well with hisax... |
| 08:23.10 | ecze | But to support real trunk with IAX protocol with asterisk 1.2 I need to upgrade my server asterisk |
| 08:23.34 | ecze | so I try to compile de backport of asterisk 1.2 for my debian sarge |
| 08:24.03 | ecze | I have modified the Makefile under channels directory to compile chan_modem |
| 08:24.19 | JT | err |
| 08:24.21 | JT | riight |
| 08:24.31 | JT | chan_modem is well and truly deprecated |
| 08:24.51 | ecze | yep I know |
| 08:25.08 | ecze | but the problem is that I am under a 2.4 kernel |
| 08:25.19 | ecze | so no misdn available |
| 08:25.30 | JT | bristuff |
| 08:26.28 | ecze | bristuff doens't work with passive cards ( I have a lot netjet, winbond, teles ) |
| 08:26.37 | JT | also upgrading your kernel is not hard |
| 08:26.40 | JT | are you sure? |
| 08:26.50 | ecze | yes this is hard in my case |
| 08:26.51 | JT | are they hfc chipset cards? |
| 08:27.03 | ecze | no |
| 08:27.28 | ecze | all siemens type |
| 08:27.56 | ecze | I can't upgrade my kernel because I work on a kind of appliance |
| 08:28.20 | ecze | And I have a lot of function not implemented on the kernel 2.6 series |
| 08:29.02 | JT | is iax trunking that important? |
| 08:29.09 | ecze | yep |
| 08:29.13 | JT | why |
| 08:29.50 | ecze | Because this kind of appliance is used to interconnect companies |
| 08:30.17 | JT | there are other voip protocols |
| 08:30.25 | JT | and you can use iax without trunking |
| 08:30.36 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 08:30.41 | ecze | I know but the overhead is very high in this case |
| 08:31.07 | JT | the saving from iax trunking is minimal |
| 08:31.45 | ecze | when you have 30 concurrent call.... |
| 08:32.16 | JT | iax trunking sometimes has issues with that many calls anyway |
| 08:34.21 | ecze | with a codec g729, the signaling take more bandwith than the voice stream |
| 08:35.14 | jql | it's the headers that kill ya |
| 08:36.31 | JT | and 30 calls in an iax trunk is generally not recommended |
| 08:36.31 | ecze | hum |
| 08:36.37 | ecze | so IAX is not the best way to trunk calls.... ? |
| 08:36.54 | JT | you need to limit the number of calls in each trunk |
| 08:37.05 | JT | also you need a zaptel timing reference for iax trunking |
| 08:37.32 | ecze | I have a PRI from digium on one side ( the main Site ) |
| 08:37.59 | ecze | and ztdummy with USB driver on the other sides |
| 08:38.22 | JT | ztdummy is not as accurate as a real timing referenc |
| 08:38.23 | ecze | so the timing reference is not the problem. |
| 08:38.41 | JT | you think a lot of things aren't a problem, i think you should check with reality :) |
| 08:39.13 | ecze | I work with asterisk 1.0 for years without problems |
| 08:39.23 | JT | sure |
| 08:39.29 | JT | that's not under dispute |
| 08:39.51 | ecze | To prevent problems I restart asterisk each night |
| 08:40.30 | ecze | because , even with digium devices, I hace time problems after 20 days... |
| 08:40.49 | ecze | hace/have |
| 08:42.29 | *** join/#asterisk ltd (n=z@202-161-17-207.dyn.iinet.net.au) |
| 08:43.22 | ecze | The main problem is that thoses companies use internet to interconnect all PABX branches and I need to configure the QOS ( I try because of course I can control the inbound traffic from isp ) |
| 08:44.32 | ecze | and to reduce the bandwith to guanrantee a minimal delay/jitter , using less header is very usefull |
| 08:51.48 | *** join/#asterisk saftsack (n=oliver@p54A7FCFA.dip.t-dialin.net) |
| 08:52.18 | ecze | so to go back to my problem.... |
| 08:52.39 | ecze | when I use the channels, I have no response from the hisax driver |
| 08:52.49 | ecze | I dial, the chan is locked |
| 08:53.05 | ecze | I the card isn't used |
| 08:53.24 | ecze | sometimes, I have a complete kernel freeze |
| 08:54.21 | ecze | this not dangerous because all those stuffs work on a live cdrom ( The appliance don't use hard drives ) |
| 08:55.07 | ecze | but I don't well see what's going on... |
| 08:55.32 | ecze | Whats the best way to debug all the channel usage during a call ? |
| 08:56.04 | ecze | I mean a deep debug but without using gdb |
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| 09:47.32 | *** join/#asterisk fd__ (n=fd@finnishplatoon.org) |
| 09:47.50 | fd__ | mornin' |
| 09:51.48 | fd__ | anyone have time for a quick zaptel question that google hasn't been able to answer (searched for ~2hours) ? |
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| 10:20.54 | menfin | fd__, I think you must tell us your question |
| 10:25.35 | fd__ | starting to fear that what i want to achieve is c-level code :p .. |
| 10:25.57 | fd__ | in finland, we call from an e1 card, to a cellphone that is a) turned off b) not a used number |
| 10:26.09 | fd__ | both a and b return the same "all lines are busy now" error from asterisk |
| 10:26.29 | fd__ | now if we were to keep a number database, we don't know when to remove an unused number from the dbase.. |
| 10:26.47 | fd__ | zaptel status codes must differ in both instances but asterisk doesn't seem to care.. :p |
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| 10:42.23 | JoelSolanki | anybody there ? |
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| 11:04.42 | dhakatel | hi, is there any dailer in linux support g729 |
| 11:06.01 | dhakatel | join #gentoo-voip |
| 11:08.18 | *** part/#asterisk dhakatel (n=ashrar@58.65.224.5) |
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| 11:25.43 | redax | if I use a Dial() connect macro [ M(macro-name) ] how can I figure out the actual call is a Transfered call ? |
| 11:57.24 | *** join/#asterisk grok_linux (i=halb@adsl-074-245-228-071.sip.bct.bellsouth.net) |
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| 12:15.13 | *** join/#asterisk voltagex (n=voltagex@124-254-120-152-dsl.ispone.net.au) |
| 12:15.37 | voltagex | hi, any voxalot users here? I'm getting Failed to authenticate on my outgoing calls. |
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| 12:24.51 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 12:25.01 | SoftIce | anyone know offhand when brituff will be using 1.4 ? |
| 12:25.26 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:26.52 | Modcuts | Has anyone here tried using asterisk-java at all? |
| 12:27.06 | mvanbaak | SoftIce: you can already download that |
| 12:27.14 | mvanbaak | http://www.multigeeks.com/ |
| 12:27.18 | mvanbaak | oops, wrong link |
| 12:27.54 | mvanbaak | SoftIce: http://www.junghanns.net/downloads/bristuff-0.4.0-test1.tar.gz |
| 12:28.12 | SoftIce | thanks mvanbaak, just read some articile |
| 12:28.16 | SoftIce | i knew it was happening soon |
| 12:28.20 | SoftIce | never even checked the site |
| 12:29.36 | SoftIce | mvanbaak: hmm, they have not updated their site yet? |
| 12:30.01 | mvanbaak | they tend to forget that all the time |
| 12:30.31 | mvanbaak | SoftIce: you subscribed to the bristuff mailinglist ? |
| 12:31.03 | SoftIce | nope, is it worth it? |
| 12:31.30 | mvanbaak | of course ;) |
| 12:31.40 | mvanbaak | </shameless plug> |
| 12:31.50 | mvanbaak | http://lists.three-dimensional.net/mailman/listinfo/bristuff-users |
| 12:32.15 | mvanbaak | Bristuff-users list run by michiel at vanbaak.info |
| 12:32.18 | mvanbaak | ;) |
| 12:32.51 | SoftIce | mvanbaak: how much spam am i going to get? |
| 12:32.56 | mvanbaak | none |
| 12:34.06 | mvanbaak | my spamfiltering works pretty good |
| 12:34.26 | mvanbaak | http://lists.three-dimensional.net/pipermail/bristuff-users/2007-April/000036.html |
| 12:34.40 | mvanbaak | tzafrir announced the 1.4 bristuff there |
| 12:35.51 | SoftIce | mvanbaak: great, about how many mails a day do you get from them? |
| 12:39.28 | mvanbaak | gheh |
| 12:39.31 | *** join/#asterisk saftsack (n=saftsack@pD9E0619F.dip.t-dialin.net) |
| 12:39.32 | mvanbaak | like 1 every 2 weeks |
| 12:39.34 | mvanbaak | ;) |
| 12:39.40 | mvanbaak | very very low traffic at this moment |
| 12:41.35 | SoftIce | ok great |
| 12:41.38 | SoftIce | let me subscribe |
| 12:41.58 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 12:42.14 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
| 12:42.25 | Geert | in sip show peers there is an ACL entry, what is that? |
| 12:42.35 | Geert | err sip show users :) |
| 12:44.11 | redax | is there a way to call a macro when somebody make an attendant transfer? |
| 12:45.52 | mvanbaak | Geert: I wonder as well :) |
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| 12:47.23 | voltagex | hi, any voxalot users here? I'm getting Failed to authenticate on my outgoing calls. |
| 12:51.00 | DrukenLPY | morning everyone |
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| 13:01.17 | yenno | hi, does somebody know when a client needs to generate a new from tag? (;tag=......) |
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| 13:16.45 | *** join/#asterisk stees (n=steve@dyn-81-5-133-1.dsl.eclipse.net.uk) |
| 13:17.44 | stees | Hi there, I know this isn't technically an Asterisk question, but when trying to install the * Business Edition from CD I get an MD5 error on one of the RPMs |
| 13:17.57 | stees | just wondering if anyone else has noticed this? |
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| 13:25.18 | Modcuts | Does anyone know if AMI over TLS work in 1.2.17 or do i have to use 1.4? |
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| 13:36.44 | *** join/#asterisk ctp_ (n=ctp@brsg-d9bec994.pool.mediaWays.net) |
| 13:37.21 | ctp_ | hi folks. anyone here knows a sip provider offering a test account to test my isp for voip support? |
| 13:38.54 | *** join/#asterisk ariel_ (n=ariel_@70-46-87-154.ftl.fdn.com) |
| 13:39.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:39.07 | PakiPenguin | hi there |
| 13:39.14 | PakiPenguin | can i send a call to a specific agent in a queue? |
| 13:40.03 | ariel_ | you can setup priorities on agents |
| 13:40.40 | PakiPenguin | ariel_, how about i want to send the call to Agent X specifically even if agent Y is on higher priority |
| 13:40.56 | ariel_ | not via the queue it self |
| 13:42.05 | ariel_ | depending on your rules for our agents we are able to transfer to there 2nd line on the polycom's we have or dedicate routes to that 2nd line for them |
| 13:44.14 | PakiPenguin | i see |
| 13:44.19 | PakiPenguin | thanks ariel_ |
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| 14:22.58 | ariel_ | ? centos |
| 14:23.11 | ariel_ | ~ centos |
| 14:23.29 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos |
| 14:23.29 | ariel_ | forgot what the bot uses to get info |
| 14:23.53 | ariel_ | ~spinlock |
| 14:24.10 | ariel_ | ~ spinlock |
| 14:24.55 | *** join/#asterisk Corydon76-home (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 14:24.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:25.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:27.15 | [TK]D-Fender | ~centosbug |
| 14:27.17 | jbot | centosbug is probably a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 14:27.19 | [TK]D-Fender | ^^^^ |
| 14:27.59 | ariel_ | thank you [TK]D-Fender |
| 14:28.13 | ariel_ | it's been a long time since I have been on the channel. |
| 14:28.15 | [TK]D-Fender | NEXT!@!@! (c) BKW |
| 14:28.51 | [TK]D-Fender | ariel_, np, its probably been almost equally as long since I've used that notice |
| 14:29.55 | ariel_ | https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 it's the actual link to the redhat problem with it. Which is really the bottom issues. |
| 14:30.19 | redax | really, there's no way to call any macro at attended transfer ? |
| 14:30.57 | [TK]D-Fender | redax, None that you don't code into the channel drivers yourself |
| 14:32.07 | redax | aha. by the way, is it safe to call a macro from chan_sip.c ? |
| 14:32.26 | [TK]D-Fender | redax, Dunno, try and find out |
| 14:32.36 | ariel_ | you can create an extension that runs a macro for you. |
| 14:32.55 | redax | ;-) will try, that was just a conceptional question |
| 14:33.23 | redax | ariel_: hm? how do you mean that? |
| 14:34.50 | ariel_ | you create and exten => 300,1,goto(blah) |
| 14:37.18 | redax | sorry but still dont understand, how that ext300 should be called when somebody doing attendent Xfer |
| 14:39.51 | ariel_ | what do you want to do with this transfer? |
| 14:41.25 | redax | catch some data, and call an AGI |
| 14:41.29 | [TK]D-Fender | ariel_, He want to call a macro TRIGGERED by the event of an attended transfer. |
| 14:42.11 | ariel_ | you can create an ivr for that capture then send it to the extension you want. |
| 14:42.21 | [TK]D-Fender | redax, time to hack the source something nasty.... |
| 14:42.44 | redax | yeah. browsing the chan_sip for days by now |
| 14:42.58 | redax | my original problem is, the CDR |
| 14:43.09 | ariel_ | we do some transfers that require an agent ID then it looks up the info via an agi then passes it to the extension you need. It askes for all the input in the menu. |
| 14:43.32 | redax | if extA calls some Trunkcall, after he transfer the call to ExtB, it is logged very-very perversly to the CDR |
| 14:44.17 | ariel_ | I make entry's via the menu to the cdr about the calls and clean up the callerID and info I need. It's just take some thinking on how you want to set it up. |
| 14:44.52 | ariel_ | We use it for support calls to route them to the right agent and also for checking to see if customer has right to after hour support via this agi and menu |
| 14:45.45 | ariel_ | there is almost nothing you can't do with an agi and macro and extensions menu's. It's just limited to your thinking. |
| 14:45.55 | redax | ariel; no. this is a very simple pbx, some 30 extensions, 4BRI pstn. |
| 14:46.10 | redax | I simply want a correct CDR database |
| 14:46.35 | redax | and the problem when some extension calls some pstn, then he transfers to another extension |
| 14:46.37 | ariel_ | again you can write out your own cdr's via your menu |
| 14:46.41 | *** join/#asterisk af_ (n=getsmart@81-174-47-36.f5.ngi.it) |
| 14:46.58 | [TK]D-Fender | ariel_, The problem is end-to-end visibility. Trust me, transfers really screw up CDR. |
| 14:47.19 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:47.26 | redax | the blind transfer is OK, the main problem the attended transfer |
| 14:47.26 | ariel_ | yes they do I use a data input for a unic id with all my calls then recall them |
| 14:47.31 | [TK]D-Fender | ariel_, If you aren't running around screaming like the rest of us .... you clearly have failed to comprehend the depth of the problem ;) |
| 14:47.52 | ariel_ | ok I will step out of it..... |
| 14:48.46 | redax | [TK]D-Fender: if I'd call some kind of ForkCDR at the time chan_sip interconnects the cannels, even that would be sufficient |
| 14:51.50 | redax | sorry, but dont understand ariel, there's no ivr at all.. |
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| 14:57.07 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 15:00.41 | plasmid | [TK]D-Fender, 'morning or |
| 15:00.46 | plasmid | 'afternoon. |
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| 15:22.48 | *** join/#asterisk Fladimir_bg (i=Fladimir@77.70.7.91) |
| 15:23.00 | Fladimir_bg | When i set H323 phone with ooh323.conf with Trixbox 2.0, H323->SIP - there is audio and it is wotking but when i call SIP-> H323 the phone is ringing but there is no audio. Any ideas ? |
| 15:23.11 | Fladimir_bg | i created custm trunk but i heard that for Incoming call i have to put context = from-pstn but i don't know where to put this context |
| 15:24.58 | *** join/#asterisk friedrich| (n=friedric@e177244253.adsl.alicedsl.de) |
| 15:26.12 | [TK]D-Fender | ~trixbox |
| 15:26.28 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/ |
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| 15:33.10 | docelm0 | ACK! TRIXBOX SUCKS! |
| 15:35.10 | Fladimir_bg | :) |
| 15:35.52 | BSD_Tech | the new trixbox is ugly |
| 15:38.19 | Bobocop | is there any variable containing MWI text sent in sip notify? Or do I have to change it in chan_sip.c and recompile? |
| 15:41.10 | fd__ | h'loh, any idea how to extend zaptel or sip error capability from congestion,busy,chanunavail to include numbernotinuse and phoneunreachable ? |
| 15:41.34 | fd__ | now it's just chanunavail whether a cellphone is just turned off, or the number is not in use |
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| 15:42.05 | fd__ | i'd like to record my own text saying "the number you called can't be reached at this time" and "the number you called is not in use" .. |
| 15:42.31 | fd__ | i want to do this with a TDM card AND a sip-trunk to a gsm gateway |
| 15:43.01 | fd__ | tried googling but no .conf will give this to me straight, and i really don't want to edit c code and recompile (trixbox in use) |
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| 15:53.06 | *** join/#asterisk elusive (n=marcelo@201.82.48.253) |
| 15:53.58 | elusive | I have setup one asterisk here, with 2 sip accounts, and I am using 2 linksys PAP2, with g729 codecs... |
| 15:54.21 | elusive | when I call, the otherside, can hear my voice perfectly, but I the voice of the otherside, arrives like a mess here |
| 15:54.35 | elusive | any suggestion on what should I do? |
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| 15:59.25 | ctp_ | hi folks. anyone here knows how many msn's are possible using sipgate? i need 6 msn's for our purposes. |
| 16:00.37 | gambolputty | msn? |
| 16:04.34 | *** part/#asterisk Bobocop (n=Bobocop@83.168.90.41) |
| 16:04.52 | fd__ | elusive try to see it's the same codec |
| 16:05.12 | fd__ | i got a similar problem when all was not alaw, there was an ulaw component somewhere in between |
| 16:05.24 | fd__ | trashy, raspy voice, barely made out the words |
| 16:12.11 | elusive | fd__ |
| 16:12.28 | elusive | may I msg you? |
| 16:13.24 | elusive | I just want to paste 3 lines for you, to have a look |
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| 16:49.09 | *** join/#asterisk BBRodriguez (n=BBRodrig@p54B01B25.dip0.t-ipconnect.de) |
| 16:49.16 | BBRodriguez | Hi everyone |
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| 16:57.30 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 16:57.43 | kiwoneka | good afternoon to all |
| 16:58.40 | kiwoneka | i recompiled zaptel 1.4.1, now i dont have dial tone |
| 17:01.10 | THX2000 | who needs a dial tone? :P |
| 17:01.16 | kiwoneka | me |
| 17:02.32 | Modcuts | Am i right in thinking that sll and tls manager interface is not folded and part of asterisk distro at the minute? |
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| 17:10.40 | SomethingISODD | Hello all question is there anyway i could setup dids to use them for dialup internet connection?? |
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| 17:38.24 | JerJer[mobile] | moooo |
| 17:42.55 | SomethingISODD | hey JerJer[mobile] |
| 17:44.29 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
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| 17:59.57 | yenno | hey... somebody knows what's wrong with my BYE? asterisk gives me a 503 server error - http://pastebin.ca/451608 |
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| 18:07.21 | nettie | hey guys, anyone using duoBRI please? |
| 18:07.40 | nettie | www.junghanns.net |
| 18:10.37 | *** join/#asterisk Gazo (n=marcelev@S010600179a2ca219.ed.shawcable.net) |
| 18:11.39 | Gazo | Looking for a good SIP truck provider in canada, Must be able to get edmonton, alberta DID's. Anyone have any recomendations? |
| 18:15.06 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.wa.comcast.net) |
| 18:15.07 | kiwoneka | a quick question |
| 18:15.31 | kiwoneka | what is the sip notify command to reboot a polycom phone? |
| 18:16.11 | philippel | [polycom-reboot] |
| 18:16.11 | philippel | Event=>check-sync |
| 18:16.11 | philippel | Content-Length=>0 |
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| 18:16.35 | kiwoneka | got it |
| 18:16.43 | philippel | however - unless your config file for the phone is setup to alway reboot - it will only reboot if some config info changed |
| 18:17.08 | kiwoneka | sip notify polycom-check-cfg 773 |
| 18:18.00 | kiwoneka | another question |
| 18:18.14 | kiwoneka | i recompiled zaptel |
| 18:18.39 | kiwoneka | not i dont have a dial tone, and my analog line is not getting picked up |
| 18:19.13 | kiwoneka | i have configured zaptel.conf |
| 18:19.33 | kiwoneka | my modules are all loaded properly |
| 18:19.42 | kiwoneka | green lights on the card |
| 18:19.49 | kiwoneka | but no dial tone |
| 18:26.37 | Juggie | kiwoneka, did you attach the extra power to the board? |
| 18:27.36 | kiwoneka | oh yes |
| 18:27.53 | kiwoneka | this was working last night |
| 18:30.14 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 18:33.49 | kiwoneka | another simple question, |
| 18:34.30 | *** join/#asterisk steve (n=muh@bouncer.stephen.marsh.name) |
| 18:34.55 | kiwoneka | how do I increase the Verbosity level |
| 18:35.51 | kiwoneka | brb, baby is up |
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| 18:37.21 | *** join/#asterisk formulate (n=formulat@62.190.110.242) |
| 18:37.26 | formulate | hi everyone |
| 18:38.16 | formulate | i'm starting to think that digium hardware is not that great |
| 18:39.19 | gambolputty | why? |
| 18:39.49 | formulate | 8192 samples in 10223 sample intervals 75.207520% |
| 18:40.03 | formulate | older revision of card works fine |
| 18:40.09 | formulate | TE412P |
| 18:40.10 | formulate | does this |
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| 18:52.23 | isamar | hi folks |
| 18:52.32 | isamar | needing a hand with kvin's g723.1 and asterisk 1.4.1 |
| 18:52.52 | isamar | I'm using that but the audio quality is not good at all.. |
| 18:53.57 | isamar | anybody with experience on that? |
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| 19:38.38 | Strom_C | here's a stupid question: what would be an appropriate pattern match for all UK numbers as dialed from within the UK? |
| 19:38.58 | Strom_C | _0XXXXXXXXXX ? |
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| 19:45.57 | Explisit | Hello, does anyone intall asterisk-addon on debian etch/sid?? |
| 19:46.44 | reber | asterisk on debian is a pain |
| 19:48.23 | Explisit | reber: Which distribution do you suggest? |
| 19:49.06 | JunK-Y | reber: why ?? |
| 19:50.28 | reber | JunK-Y, do you use debian ? Did you tried to install app_conference ? |
| 19:50.35 | reber | Explisit, gentoo |
| 19:51.01 | JunK-Y | i use ubuntu. |
| 19:51.41 | reber | JunK-Y, Did you tried to install app_conference on your ubuntu ? |
| 19:51.53 | JunK-Y | nope. |
| 19:52.00 | reber | nothing else to say. |
| 19:52.08 | Explisit | Well reber I have install app_conference and work like charm |
| 19:52.42 | reber | Explisit, how did you installed it ? |
| 19:52.52 | JunK-Y | reber: ive install lot of other apps, and i dont think the os have something related to the apps, which is so high-level. |
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| 19:58.24 | reber | JunK-Y, "lot of other apps " hehe, do you know what app_conference is ? |
| 20:00.26 | JunK-Y | reber: no, isnt an app to call my grand-mo? |
| 20:00.37 | JunK-Y | its for conference, maybe? :) |
| 20:01.08 | reber | hehe maybe |
| 20:01.19 | JunK-Y | which is an alternative to meetme. |
| 20:01.40 | reber | installs with hacks and patches |
| 20:01.54 | reber | apt-get install app_conference ? hehe |
| 20:04.03 | reber | no ! emerge asterisk-app_conference ! And this stopped all my debian/mubuntu!@# investigations |
| 20:09.07 | Bazy | Do any of you know a romanian SIP telephony provider? www.something.ro where I can sign up for a sip account? |
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| 20:11.14 | techie | ign |
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| 20:36.41 | rikstah | Hi there, can someone tell me if there is an existing open source solution to modify asterisk realtime extensions via the web, before i write my own? Thanks |
| 20:38.03 | JerJer[mobile] | phpmyadmin |
| 20:39.03 | rikstah | yeah, obviously, but it's not as convenient |
| 20:39.19 | rikstah | it's tedious |
| 20:40.25 | JerJer[mobile] | ok then don't run realtime |
| 20:40.33 | JerJer[mobile] | use the users.conf convention |
| 20:40.57 | JerJer[mobile] | then use one of the dozens of bullshit GUIs that are out there |
| 20:41.35 | rikstah | What is your problem? What i am saying is that I can write a more convenient way of adding priorities, i just wanted to make sure I was not re-inventing the wheel. You need to relax |
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| 20:43.36 | JerJer[mobile] | ask a real question then |
| 20:43.56 | JerJer[mobile] | there are countless GUIs out there that do exactly what you have described |
| 20:44.02 | JerJer[mobile] | if they don't work for you, write your own |
| 20:44.10 | JerJer[mobile] | simple as that |
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| 20:53.16 | EzWay | would be cool to get power from usb to inject my rj45 as poe on polycom 3xx or any hardphone poe |
| 20:57.17 | JerJer[mobile] | not sure if there would be enough current |
| 20:58.09 | Juggie | not enough |
| 20:58.11 | Juggie | only 5V |
| 21:00.07 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:04.09 | rikstah | JerJer[mobile], my question was well phrased. All i was looking for was the name of one that is existing. So if you'd be kind enough to name a couple that'd be great. |
| 21:06.33 | blitzrage | JerJer[mobile]: ping! |
| 21:09.19 | JerJer[mobile] | rikstah: google |
| 21:09.26 | JerJer[mobile] | "asterisk gui" |
| 21:09.32 | JerJer[mobile] | pick one |
| 21:09.43 | JerJer[mobile] | blitzrage: pong |
| 21:09.47 | rikstah | i googled, obviously |
| 21:10.00 | rikstah | i dont want the full solution, just the web part |
| 21:10.25 | rikstah | never mind, thanks for the vaguest help possible |
| 21:10.52 | redax | you can checkout using svn |
| 21:10.55 | redax | ehh. |
| 21:10.55 | JerJer[mobile] | no problem |
| 21:11.00 | JerJer[mobile] | glad i could help |
| 21:11.32 | Juggie | hah. |
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| 21:11.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:12.04 | kiwoneka | hello |
| 21:12.07 | kiwoneka | agin |
| 21:12.12 | JerJer[mobile] | moo |
| 21:12.36 | kiwoneka | i just recompiled zaptel 1.4.1 |
| 21:12.45 | JerJer[mobile] | yay |
| 21:12.46 | kiwoneka | now i dont have a dial tone |
| 21:12.59 | *** join/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 21:13.09 | JerJer[mobile] | did you run ztcfg ? |
| 21:13.18 | kiwoneka | yes sir |
| 21:13.32 | kiwoneka | power is plugged in |
| 21:13.33 | connecta | can anyone recommend a way to configure the polycom 501's to use a shared extension for an additional line. |
| 21:13.40 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 21:13.46 | connecta | one that rings all phones |
| 21:14.26 | connecta | i all the phones to register for this line i think, rather than do a ring group. this way it will appear to them as a separate line on their phone |
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| 21:14.45 | Uatec | Greetings! |
| 21:14.50 | connecta | hi |
| 21:16.06 | redax | is it safe to run any app from channel drivers? like app=pbx_findapp("Macro"); pbx_exec( chan, app, macro_name,1); |
| 21:16.07 | redax | ? |
| 21:17.14 | JerJer[mobile] | redax: i would say no... what happens if app_macro is not loaded? |
| 21:17.53 | JerJer[mobile] | plus why make the channel driver deal with application details? let the channel driver deal with channel driver detail, then create a dialplan to sew everything together |
| 21:17.55 | Juggie | i am sure it is not good to run any dialplan app from outside the dialplan. |
| 21:18.01 | redax | ok I'd say .. app=pbx_findapp("Macro"); if( app ) { res=pbx_exec(....) |
| 21:18.16 | Juggie | seems bad to me. |
| 21:18.19 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 21:18.24 | redax | JerJer[mobile]: because of attendent transfer |
| 21:18.27 | JerJer[mobile] | sure you can trap for a null, but how is that going to be of any benefit to your app? |
| 21:18.34 | JerJer[mobile] | er |
| 21:18.36 | JerJer[mobile] | channel |
| 21:18.51 | redax | in chan_sip there's no other way |
| 21:19.14 | redax | it's simply masquerades the channels and no event at all |
| 21:19.27 | JerJer[mobile] | and? |
| 21:19.40 | redax | and CDR is looks funny |
| 21:20.22 | connecta | is anyone here familiar with the polycom .cfg files? |
| 21:20.31 | redax | extA calls some trunk number let's say TrunkX, after he Attendent xfer to extB |
| 21:20.48 | redax | you'll get 2 entries in CDR like extA calls extB |
| 21:20.48 | JerJer[mobile] | can't you fork a new CDR? |
| 21:21.11 | redax | well. why not., is that safe to do in chan_sip.c ? |
| 21:21.30 | redax | I assume yes. |
| 21:21.45 | JerJer[mobile] | i spoze if you are simply hacking things |
| 21:26.32 | redax | JerJer[mobile]: I cant find any other solution.. I think I have to add cdr forking to chan_sip.c func:attempt_transfer |
| 21:27.22 | hi365 | where does Return greturn to: gosub() or the line befor? |
| 21:27.24 | redax | before I wanted a macro stuff here, but you're right |
| 21:29.46 | rene- | hey |
| 21:29.54 | JerJer[mobile] | hoe |
| 21:29.58 | rene- | i am using nufone h323 but i get no audio at all, |
| 21:30.07 | rene- | speaking of the devil |
| 21:30.09 | rene- | heh |
| 21:30.22 | rene- | well my gsm gateway says it is transmitting RTp |
| 21:30.30 | rene- | but it isnt receiving anything at all |
| 21:30.35 | rene- | call do get established |
| 21:32.48 | rene- | i have no experience in h323 whatsoever, i have read that maybe dtmf settings can be a problem? |
| 21:32.50 | rene- | that could create incompatibilities |
| 21:32.51 | rene- | and no audio issues? |
| 21:32.57 | rene- | i have disabled my firewall so i dont think this is network related |
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| 21:34.14 | JerJer[mobile] | i'm just a sprite |
| 21:34.42 | *** join/#asterisk connecta (n=Administ@76.23.188.72.cfl.res.rr.com) |
| 21:34.42 | connecta | does asterisk support having a bridged line for multiple phones? |
| 21:35.34 | blitzrage | yay, I'm back! |
| 21:35.47 | blitzrage | connecta: it's called MeetMe |
| 21:36.05 | JerJer[mobile] | rene-: do you have a valid bindaddr set ? |
| 21:36.15 | JerJer[mobile] | valid codec ? |
| 21:36.30 | connecta | blitzrage : thanks but thats not the same |
| 21:36.36 | rene- | jerjer: it is commented out |
| 21:36.45 | rene- | i do have selected a codec |
| 21:36.58 | rene- | allow=alaw and allow=ulaw |
| 21:37.07 | connecta | hrmn, i'm reading here that asterisk does not do a simple bridged line feature where multiple phones register to 1 extension and can share it |
| 21:37.10 | connecta | thats pretty lame |
| 21:37.11 | rene- | letme try binding an ip specifically |
| 21:37.14 | connecta | cmon asterisk |
| 21:37.44 | uhb | connecta: FollowMe? |
| 21:37.47 | JerJer[mobile] | rene-: bindaddr is required |
| 21:38.01 | JerJer[mobile] | unless pcadach fixed things in that part of the code |
| 21:38.04 | JerJer[mobile] | i have no clue any more |
| 21:38.26 | connecta | uhb: are you just throwing that out there? that doesnt really have anything to do with my question does it? |
| 21:38.46 | uhb | Well, you could very well have multiple phones ring when one extension is dialed |
| 21:38.59 | connecta | yep. thats a ring group / hunt group |
| 21:40.47 | *** join/#asterisk ltd (n=z@202-161-17-207.dyn.iinet.net.au) |
| 21:41.01 | [TK]D-Fender | connecta, No, asterisk does not have proper SLA yes. |
| 21:41.03 | [TK]D-Fender | yet* |
| 21:44.17 | connecta | Just got this from juggie in the dev room. very interesting... http://www.asterisk.org/node/48342 |
| 21:44.27 | Uatec | hey, does anybody know any tool I can use to forward UDP ports from my windows machine to another machine? |
| 21:45.37 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 21:46.11 | [TK]D-Fender | connecta, From what I've seen of that patch and heard from those who've implemented it, its a real bastardization of the true method and is only a fake for lines, not phones. |
| 21:47.00 | [TK]D-Fender | connecta, Its a naty hybrid that seems to abuse presence and DISA ( a sure-fire sign that the method is not "kosher") |
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| 21:53.15 | Juggie | [TK]D-Fender, its not a patch its in 1.4.1 i thought. |
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| 21:53.21 | Juggie | if you have a problem with it, talk to russ. |
| 21:53.43 | [TK]D-Fender | Juggie, he told me direct its only staying in trunk, never release. |
| 21:53.57 | [TK]D-Fender | Juggie, because of their commit style now. |
| 21:54.23 | Juggie | "So, after enough demand for this feature, we decided that we would commit to making a way to set up an Asteirsk system to work like a key system. There was a version of this support in Asterisk 1.4.0. However, it didn't really work. So, before 1.4.1 was released, I completely re-wrote the SLA support for Asterisk. Now, it works!" |
| 21:55.13 | [TK]D-Fender | Juggie, OOPS! My bad. Forget that I said that entirely! was thinking of the DEVSTATE patch ;) |
| 21:55.38 | Juggie | [TK]D-Fender, havnt you figured out yet i am allways right :> |
| 21:55.50 | [TK]D-Fender | Juggie, Except for the "bastardized semblence of SLA" bit ;) |
| 21:56.01 | [TK]D-Fender | Juggie, Take a number on that! :D |
| 21:58.23 | Juggie | heh hot |
| 21:58.32 | Juggie | Family Guy is the top TV show in my facebook network |
| 21:59.57 | JerJer[mobile] | giggity |
| 22:00.27 | [TK]D-Fender | oooo right! |
| 22:06.43 | blitzrage | " 'Quagmires Cross Country Tour'. Doesn't 'country' have an 'o' in it?" |
| 22:06.46 | blitzrage | "Nope!" |
| 22:07.52 | Juggie | blitzrage, you fucked it up. |
| 22:08.02 | Uatec | blitzrage, when is it on? |
| 22:08.13 | Juggie | it was ' Cross Cuntry Tour' |
| 22:08.41 | blitzrage | Juggie: I know... but I was spelling it as you would have seen it in your head until Brian asked the right question |
| 22:08.59 | VJFROMGT | i just installed asterisk and restarted, when i do asterisk -r i get Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 22:09.10 | blitzrage | VJFROMGT: asteriks isn't running |
| 22:09.19 | VJFROMGT | service asterisk start ? |
| 22:09.24 | blitzrage | yes |
| 22:09.29 | Juggie | blitzrage, you mean you were being a nub messing up jokes! |
| 22:09.38 | blitzrage | run 'asterisk -c' because you have something wrong in your config which is causing asterisk to not start |
| 22:09.42 | VJFROMGT | asterisk: unrecognized service |
| 22:09.52 | VJFROMGT | is what is get when i do service asterisk start |
| 22:09.52 | blitzrage | VJFROMGT: you didn't run "make config" |
| 22:10.00 | VJFROMGT | hmm, no |
| 22:10.37 | blitzrage | shit, I burnt my veggies a bit |
| 22:10.37 | VJFROMGT | did it now,, now i get asterisk died with code 1 |
| 22:11.27 | VJFROMGT | now it keeps saying automatically tryint to start asterisk |
| 22:11.32 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 22:11.42 | VJFROMGT | Automatically restarting Asterisk. |
| 22:23.23 | *** join/#asterisk alrs_ (i=nobody@cpe-76-81-15-170.socal.res.rr.com) |
| 22:28.11 | rene- | thanks jerjer you are the man |
| 22:28.36 | rene- | i now have cheap gsm calls |
| 22:28.53 | rene- | btw is there something like an asterisk-biz irc channel? |
| 22:39.33 | *** join/#asterisk znoG (n=gs@OL132-95.fibertel.com.ar) |
| 22:41.16 | JerJer[mobile] | rene-: i accept paypal :) jj@nufone.net |
| 22:46.14 | nettie | hey guys, anyone using duoBRI please? |
| 22:46.23 | nettie | or quadBRI |
| 22:49.38 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 22:50.37 | bluelinq | hello there, can anyone provide me a sample of 7960-font.xml? |
| 22:53.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:54.23 | bluelinq | anyone home? |
| 22:54.29 | JerJer[mobile] | nope |
| 22:54.34 | JerJer[mobile] | nobody here but us chickens |
| 22:54.40 | bluelinq | bummer... |
| 22:55.03 | bluelinq | JerJer, any ideas on that file? |
| 22:55.17 | jm|laptop | hello |
| 22:55.18 | JerJer[mobile] | you try google? |
| 22:55.25 | bluelinq | done that... nothing |
| 22:55.44 | jm|laptop | does iax support srvlookup? |
| 22:59.35 | VJFROMGT | can anyone here help me install freepbx? |
| 23:00.12 | blitzrage | see #freepbx |
| 23:00.33 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 23:00.56 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 23:01.19 | VJFROMGT | not much support for centos 5 |
| 23:02.11 | JerJer[mobile] | jm|laptop: last i looked, no |
| 23:02.19 | jm|laptop | ah |
| 23:02.22 | jm|laptop | ok thanks :) |
| 23:02.33 | JerJer[mobile] | but i'm stuck on 1.2 so 1.4 might be different |
| 23:03.17 | jm|laptop | I'm 1.2, too |
| 23:04.10 | jm|laptop | IAX2/user@whatever.com points to the A record for whatever.com despite SRV dns record for _iax._udp.whatever.com ... |
| 23:08.11 | bluelinq | does anyone has a 7960 using sccp working? |
| 23:08.38 | JerJer[mobile] | bluelinq: a while ago, using one line on chan_skinny, yes |
| 23:09.14 | bluelinq | on the tftp it constantly ask for 7960-fonts.xml and keeps loopiing after that.. |
| 23:09.14 | jm|laptop | bluelinq: a guy I was just testing with has that, too |
| 23:14.24 | JerJer[mobile] | find a call manager express and tftp that file :) |
| 23:15.19 | bluelinq | why will that file be needed with asterisk? |
| 23:15.29 | JerJer[mobile] | asterisk could care less |
| 23:15.32 | JerJer[mobile] | the phone wants it |
| 23:17.24 | bluelinq | bloody phones... |
| 23:21.14 | JerJer[mobile] | have you tried feeding the phone a empty file ? |
| 23:21.26 | JerJer[mobile] | just touch 7960-font.xml |
| 23:22.01 | bluelinq | yes, it give another error... |
| 23:22.46 | JerJer[mobile] | leme see if there is anyone from cisco with a clue here |
| 23:26.59 | *** join/#asterisk brussel (n=brussel@cpe-72-130-172-213.san.res.rr.com) |
| 23:29.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:31.35 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 23:32.43 | hi365 | where does Return actualy return to: Gosub() or the line befor? |
| 23:34.25 | JerJer[mobile] | bluelinq: later on I should be able to get access to a call munger express |
| 23:36.42 | JerJer[mobile] | i found where one could download an exe that would have that file, but it requires a CCO login |
| 23:37.40 | cspot | i have cco....looking for the file(s) now |
| 23:37.57 | cspot | toss me a filename |
| 23:41.31 | JerJer[mobile] | anything localization |
| 23:41.43 | JerJer[mobile] | http://www.cisco.com/pcgi-bin/tablebuild.pl/CME-Locale |
| 23:41.45 | cspot | ok |
| 23:42.04 | hi365 | can you tell Return() to go a pririty higher that the one that sent it to Gosub()? |
| 23:43.14 | JerJer[mobile] | hi365: test it |
| 23:43.35 | hi365 | JerJer[mobile]: sure - but how? Return(-1)? |
| 23:44.54 | bluelinq | cspot can you contact me directly? |
| 23:47.28 | JerJer[mobile] | bluelinq: ok i got |
| 23:47.28 | JerJer[mobile] | stby |
| 23:52.47 | *** join/#asterisk voltagex (n=voltagex@124-254-120-152-dsl.ispone.net.au) |
| 23:53.34 | blitzrage | hi365: just like what you were doing -- just parse on the return value and have it go to a different label if you want |
| 23:54.48 | hi365 | blitzrage: im sorry, i didnt get it. would you provide an example? |
| 23:56.29 | blitzrage | Return(label_to_go_to) |
| 23:56.38 | blitzrage | Goto(${RETVAL}) |
| 23:57.07 | blitzrage | or better.... GotoIf($[${EXISTS(${RETVAL})}]?${RETVAL}) |
| 23:58.51 | hi365 | blitzrage: is retval an asterisk variable or would i have to set it myself? |
| 23:58.54 | blitzrage | I think it's RETVAL that gets set on a Return().... |
| 23:59.09 | blitzrage | might be GOSUB_RETVAL... I forget |
| 23:59.51 | blitzrage | and it might only be in the backport of app_stack.c.... I'm not totally sure :) |