| 00:00.09 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 00:00.10 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:02.58 | *** join/#asterisk maverickbna (n=sentinel@wikipedia/Shadowhntr) |
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| 00:04.35 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 00:14.41 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 00:23.59 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 00:25.26 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 00:35.02 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-63709bac12011546) |
| 00:41.17 | mchou | jblack: you here? |
| 00:47.30 | *** join/#asterisk Segnale007 (n=Segnale0@host109-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 00:51.56 | [TK]D-Fender | mchou: Typical NANPA patterns : NXXXXXX , NXXNXXXXXX , 1NXXNXXXXXX, [34689]11 |
| 00:54.24 | eric_hill | To the newly joined people: Has anyone gotten an inbound ISDN PRI call to properly redirect? zapCD doesn't seem to do anything, neither does Transfer. Is Call Deflection even available? |
| 00:54.49 | tzanger | 811? |
| 00:54.58 | tzanger | there's also 310 |
| 00:55.05 | tzanger | well that's covered by NXXXXXX I guess |
| 00:55.09 | tzanger | but what's 811? |
| 00:55.12 | tzanger | and 311 |
| 00:55.33 | tzanger | there's also 0NXXNXXXXXX |
| 00:55.37 | [TK]D-Fender | eric_hill: it is possible to do 2BCT for PRI on 5ESS and DMS100 IIRC. |
| 00:55.44 | mchou | [TK]D-Fender: so here's the deal. ITSP uses 70 for voicemail access, 500 for echo test, (and a whole bunch of others that I don't know/use yet). the standard NANPA patternes you enumerated above would cover my examples? |
| 00:55.48 | tzanger | and 011. and 1011. I think |
| 00:55.49 | [TK]D-Fender | tzanger: not ear codes... |
| 00:55.55 | [TK]D-Fender | area* |
| 00:56.21 | [TK]D-Fender | mchou: Where "your samples" means "normal looking numbers on the PSTN", yeah |
| 00:57.15 | mchou | [TK]D-Fender: no, I meant "70" for vmail and "500" for echo test (my samples) |
| 00:57.56 | [TK]D-Fender | mchou: Feel free to use any value you want that doesn't conflict with your other plans |
| 00:59.55 | mchou | [TK]D-Fender: lemme rephrase, think there is a misunderstanding. I pick up my sip phone (hear dial tone), then dial 70 to access my vmail at ITSP. |
| 01:00.33 | mchou | Dont think your NANPA pattern cover that, correct? |
| 01:00.39 | [TK]D-Fender | mchou: you can do that if you want. Its an exten to setup like any other. |
| 01:00.56 | [TK]D-Fender | mchou: then again, you're using *, what on earth would be using your ITSP's VM? |
| 01:01.13 | *** join/#asterisk keith4_ (n=kbe2@207-172-236-173.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 01:01.24 | mchou | [TK]D-Fender: cause my asterisk is clearly not production use yet :) |
| 01:01.49 | [TK]D-Fender | mchou: Spend those valuable 2 minutes to add VM to you incoming :p |
| 01:01.52 | [TK]D-Fender | your* |
| 01:01.52 | mchou | not ready for prime time :) |
| 01:02.30 | keith4_ | two questions: 1) is there any way to boot people out of a conference channel? 2) is there any way to go through roll call of who's already in the conference? |
| 01:02.36 | [TK]D-Fender | mchou: Either way go make a pattern to dial out to pick up VM on their end. |
| 01:02.45 | mchou | nono, my asterisk box is gonna thrash quite a bit cause I dont know what I'm doing, testing crap |
| 01:03.05 | [TK]D-Fender | keith4 : there's a button to go into the admin menu in MeetMe when you're an admin... go try it |
| 01:03.24 | keith4_ | [TK]D-Fender: ok, thanks |
| 01:03.25 | [TK]D-Fender | mchou: And like I said, fine, go make another exten to pick up your ITSP hosted VM's |
| 01:06.32 | jameswf-home | sweet shoot em up movie is on |
| 01:07.41 | hardwire | I do enjoy that movie |
| 01:08.01 | [TK]D-Fender | jameswf-home: namely? |
| 01:09.34 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 01:09.44 | hardwire | [TK]D-Fender: "Shoot Em Up" |
| 01:10.01 | [TK]D-Fender | hardwire: Ah, literal. Never heard of it |
| 01:10.13 | hardwire | It's fun. |
| 01:10.16 | hardwire | Just saw it the other night. |
| 01:10.19 | bkruse | is watching transporter |
| 01:10.19 | hardwire | Lollerfest |
| 01:10.24 | hardwire | another good one! |
| 01:11.38 | hardwire | anybody familiar with static dead route detection and nexthopping under Linux? |
| 01:11.49 | [TK]D-Fender | BBIAB |
| 01:12.07 | *** join/#asterisk thansen|laptop (n=thansen@c-69-140-63-245.hsd1.md.comcast.net) |
| 01:15.06 | keith4_ | [TK]D-Fender: this is awesome. how'd you get to be so smart? |
| 01:15.49 | hardwire | you eat the hearts of your enemys |
| 01:17.40 | hardwire | [TK]D-Fender: has eaten quite a few foes. |
| 01:22.44 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 01:23.09 | [TK]D-Fender | The dread pirate Roberts leaves no survivors! |
| 01:25.26 | keith4_ | [TK]D-Fender: the only options I find in the admin menu that aren't in the user menu are lock/unlock, or kick the last user |
| 01:33.45 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 01:35.45 | JerJer | Anyone have any bright ideas how one can get the remote IP address a particular SIP (or IAX) endpoint coming from? we cannot trust the SIPURI channel variable, as it usually contains private IP addressing |
| 01:36.03 | JerJer | er - yeah |
| 01:36.22 | JerJer | Anyone have any bright ideas how one can get the remote IP address a particular SIP (or IAX) endpoint? (loong day :) |
| 01:36.41 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:36.59 | JerJer | ... on a per call basis |
| 01:37.29 | JerJer | seems like chan_sip needs to create another channel variable containing the remote IP addy |
| 01:45.01 | scooby2 | whats the best way to test a dialplan without having the t1's live? |
| 01:48.41 | mosty | put something in place of the t1's |
| 01:52.43 | scooby2 | would an analog telephone adaptor work or do I need something like a phone line? |
| 01:58.57 | *** join/#asterisk moy (n=moy@189.169.61.29) |
| 02:00.25 | *** join/#asterisk gramulhaozin (n=charles@c-76-110-242-178.hsd1.fl.comcast.net) |
| 02:07.27 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 02:07.45 | deeperror | ~centos52bug |
| 02:07.46 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 02:20.18 | *** join/#asterisk mmurdock (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 02:23.20 | *** join/#asterisk ReDNeQ (n=ReDNeQ@75.148.217.225) |
| 02:24.19 | ReDNeQ | sup |
| 02:31.39 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 02:32.44 | *** join/#asterisk admin0 (n=admin@bb121-7-229-228.singnet.com.sg) |
| 02:42.45 | axisys | will vonage terminate my calls ? or do I need teliax or voicepulse or one of those providers? |
| 02:43.44 | axisys | i currently am a vonage customer .. i can receive calls and can redirect to my asterisk server fine |
| 02:43.55 | axisys | so receiving calls thru asterisk is working fine |
| 02:44.25 | axisys | now I am just trying to solve the making calls from pstn to my cell phone |
| 02:44.44 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-163-192.dsl.stlsmo.sbcglobal.net) |
| 02:45.03 | axisys | according to systm.org's asterisk presentation i need to have an account with teliax or voicepulse or some other providers |
| 02:45.19 | axisys | but i need to know if vonage can also provide same service |
| 02:45.24 | axisys | anyone here knows |
| 02:45.25 | axisys | ? |
| 02:47.48 | LemensTS | axisys: you mean like sip or iax from vonage? if so im told they dont. |
| 02:50.00 | LemensTS | You can run them into an analoug card if you needed... |
| 02:50.55 | axisys | my vonage line goes to my pstn on ata |
| 02:51.02 | axisys | so that is all I need? |
| 02:51.53 | axisys | so I call from my cell to my vonage number it works fine |
| 02:51.54 | LemensTS | You you have it on a sip/analog adapter |
| 02:52.08 | axisys | i have a spa 3102 |
| 02:52.19 | axisys | its pstn like connected to my vonage adapetr |
| 02:52.49 | *** join/#asterisk admin0 (n=admin@bb121-7-229-228.singnet.com.sg) |
| 02:52.54 | axisys | that is how I were able to make call from my cell phone to my vonage number thru my asterisk server |
| 02:53.04 | axisys | so incoming is working |
| 02:53.17 | [TK]D-Fender | axisys: You're currently going through an ATA to get to Vonage through theirs. of COURSE you can continue this way Of course you've have been better off using a Vonage Soft-Phone account. |
| 02:53.29 | [TK]D-Fender | axisys: Or better yet, a decent non-encumbered ITSP |
| 02:53.54 | [TK]D-Fender | axisys: You dial OUT through the [pstn] peer you set up of course |
| 02:54.04 | axisys | [TK]D-Fender: well for now I want the technical solution.. can I call from my fxs to my vonage account? |
| 02:54.17 | [TK]D-Fender | axisys: yes, thats why you set up the SPA-3102 in the first place |
| 02:54.30 | [TK]D-Fender | axisys: Dial(SIP/pstn/123564126354126354) |
| 02:54.34 | LemensTS | axisys: yea vonage is rock solid but the price is high. I would rather not have an army of vonage boxes when you can do it by software via ITSP's. |
| 02:54.53 | [TK]D-Fender | LemensTS: Vonage IS an ITSP |
| 02:54.56 | axisys | [TK]D-Fender: ok let me try that |
| 02:55.15 | LemensTS | tk: let me rephrase, ITSP via SIP/IAX |
| 02:55.35 | [TK]D-Fender | LemensTS: Vonage uses SIP. Strike 2.... |
| 02:55.59 | axisys | [TK]D-Fender: wait! i answered the question totally wrong! |
| 02:56.12 | axisys | [TK]D-Fender: i want to make calls to my cell from my fxs |
| 02:56.14 | LemensTS | lol damn it, SIP/IAX from the ITSP server directly into your Asterisk server. |
| 02:56.15 | [TK]D-Fender | axisys: I'm not entirely surprised |
| 02:56.23 | axisys | [TK]D-Fender: sorry :P |
| 02:56.36 | [TK]D-Fender | LemensTS: Which Vonage lets you do if you ask for a soft-phone sign-in. |
| 02:56.45 | [TK]D-Fender | LemensTS: Yoooooooouuuuuuu're OUT! |
| 02:56.45 | axisys | [TK]D-Fender: how to make calls to my cell from my fxs ? |
| 02:57.00 | [TK]D-Fender | axisys: That makes even LESS sense. |
| 02:57.12 | [TK]D-Fender | axisys: Where is your cell magically coming in THROUGH? |
| 02:57.32 | axisys | [TK]D-Fender: I want to make a outgoing call to my friend |
| 02:57.50 | [TK]D-Fender | axisys: Your sense of direction is FUBAR'd. big time |
| 02:57.53 | axisys | [TK]D-Fender: from my home phone connected to fxs port |
| 02:57.58 | [TK]D-Fender | axisys: Try that from the BEGINNING. |
| 02:58.06 | LemensTS | TK: Oh yea? I didnt know that. I will set one up at work tomorrow I got a softphone account already. |
| 02:58.09 | axisys | [TK]D-Fender: ok .. here comes |
| 02:58.24 | [TK]D-Fender | axisys: No, FXS ports are what you plug a PHONE into. You are reversed |
| 02:58.42 | axisys | [TK]D-Fender: my home phone is connected to fxs of the spa 3102 |
| 02:58.44 | LemensTS | FX(STATION) |
| 02:59.10 | axisys | [TK]D-Fender: from the home phone can I call an external number ? |
| 02:59.13 | [TK]D-Fender | axisys: No, your SPA-3102 is pluggeed into your Vonage ATA via its FXO PORT <- |
| 02:59.24 | axisys | [TK]D-Fender: yes |
| 02:59.28 | [TK]D-Fender | ~fxofxs |
| 02:59.29 | jbot | extra, extra, read all about it, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 02:59.49 | axisys | i never mentioned about fxo port |
| 02:59.58 | axisys | i know fxo is connected to my vonage ata |
| 03:00.02 | [TK]D-Fender | axisys: because your teminology is all wrong |
| 03:00.13 | axisys | [TK]D-Fender: oh ok |
| 03:00.30 | axisys | home phone ---> fxs of spa 3102 |
| 03:00.34 | [TK]D-Fender | axisys: Ok, so your "home phone" is on the FXS. the FXO port leads to the Vonage ATA. |
| 03:00.42 | axisys | correct! |
| 03:01.17 | [TK]D-Fender | axisys: So you need to setup a peer for your FXS port. then process what it dials in your dialplan and the dial OUT your FXO port using your [pstn] peer |
| 03:01.21 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:01.52 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:02.36 | axisys | [TK]D-Fender: hmm.. so much make sense.. i only need to make change in the extensions.conf correct? |
| 03:03.09 | [TK]D-Fender | axisys: If you have already set up the peer entry for the FXS port, yes |
| 03:04.43 | axisys | [TK]D-Fender: i defined that in my sip.conf http://pastebin.com/f55055f8c |
| 03:05.05 | [TK]D-Fender | axisys: If 100 is it, then yup |
| 03:05.34 | axisys | [TK]D-Fender: yes |
| 03:05.56 | axisys | 100 is ponting to context internal |
| 03:06.26 | axisys | how do I tell it to use pstn to make any calls ? |
| 03:06.35 | axisys | outgoing calls that is |
| 03:07.05 | axisys | [TK]D-Fender: like this ? |
| 03:07.07 | axisys | [TK]D-Fender: exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,) |
| 03:07.17 | [TK]D-Fender | axisys: I already gave you a dial sample. |
| 03:07.24 | [TK]D-Fender | axisys: Mostly like that, yes |
| 03:07.45 | axisys | [TK]D-Fender: oh yes .. |
| 03:07.59 | [TK]D-Fender | axisys: extensions.conf is the most important part of *. Master it. |
| 03:08.18 | axisys | [TK]D-Fender: :-) |
| 03:08.29 | axisys | [TK]D-Fender: i read the asterisk oreilly book once |
| 03:08.34 | axisys | [TK]D-Fender: i have to read it again |
| 03:09.59 | scooby2 | would an analog telephone adaptor work or do I need something like a phone line to test a dial plan without the t1's? |
| 03:12.57 | [TK]D-Fender | scooby2: You have mentioned 3 things that don't belong in the same sentence |
| 03:13.16 | [TK]D-Fender | scooby2: What exactly do you want to do? |
| 03:13.33 | scooby2 | test my new asterisk setup without having to move the t1's over |
| 03:13.51 | [TK]D-Fender | scooby2: Use a soft-phone |
| 03:14.18 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 03:14.39 | scooby2 | that will let me act like I was calling into the system? |
| 03:15.04 | [TK]D-Fender | scooby2: A call is a call is a call. Doesn't matter where its from, it only matters where you send it. |
| 03:15.15 | scooby2 | ok, thanks |
| 03:15.17 | [TK]D-Fender | scooby2: EVERYTHING coming in to * is a call. |
| 03:16.25 | axisys | [TK]D-Fender: i get busy signal when calling out to my cell http://pastebin.com/f52858ace |
| 03:17.30 | ReDNeQ | hey tk; which of the pre-made iso do you favor .. elastix, trix, pbxinflash?? |
| 03:17.58 | [TK]D-Fender | axisys: SIP/2.0 401 Unauthorized <- you have not set up the auth on your [pstn] properly |
| 03:18.07 | [TK]D-Fender | ReDNeQ: NONE. |
| 03:18.22 | ReDNeQ | so you like vanilla installs? what os do you prefer |
| 03:18.35 | [TK]D-Fender | ReDNeQ: CentOS or Slackware |
| 03:19.09 | ReDNeQ | cool.. do you run freepbx at all? or just asterisk and console... |
| 03:19.29 | [TK]D-Fender | axisys: and do "set verbose 10" for next time |
| 03:19.38 | [TK]D-Fender | ReDNeQ: No FreePBX |
| 03:19.49 | ReDNeQ | man your hardcore! ;P |
| 03:19.58 | [TK]D-Fender | ReDNeQ: No, I'm running * |
| 03:20.16 | scooby2 | freepbx is bad. You can only do what they say you can do. Basically no customization. |
| 03:20.21 | ReDNeQ | no no i know that.. its just a chore for those dummies like me to fathom just running * |
| 03:20.38 | [TK]D-Fender | ReDNeQ: Then what are you doing here? |
| 03:20.55 | ReDNeQ | becuase when i have to fix something that isnt in those progs |
| 03:21.27 | ReDNeQ | actually i started in * but got lazy for a few customers because they wanted web based management of thier systems |
| 03:21.51 | ReDNeQ | i dont/havent designed anything so i chose to go through the pre-made systems on a few |
| 03:22.46 | axisys | [TK]D-Fender: http://pastebin.com/f43282e63 i am using pstn/secret .. it is same in PSTN Line in SPA |
| 03:23.40 | [TK]D-Fender | axisys: Wheres the new call? |
| 03:24.07 | axisys | [TK]D-Fender: what you mean? |
| 03:24.24 | [TK]D-Fender | axisys: the new call with proper debuig |
| 03:24.39 | axisys | [TK]D-Fender: oh ok.. let me call again.. one sec |
| 03:25.15 | *** join/#asterisk chendy (n=chatzill@58.251.229.10) |
| 03:27.03 | axisys | [TK]D-Fender: http://pastebin.com/f242166f7 (with verbosity 10) |
| 03:27.21 | axisys | ^^ full debug of new call |
| 03:28.23 | [TK]D-Fender | axisys: Same deal. Your auth is wrong. |
| 03:28.59 | axisys | but i am using the same username / password on [pstn] of sip.conf and PSTN Line on SPA |
| 03:29.06 | [TK]D-Fender | axisys: insecure=very L- remove this while you're at it |
| 03:29.33 | axisys | ok |
| 03:31.23 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 03:32.03 | axisys | I noticed on PSTN Line the Voip-TO-PSTN Gateway section the Voip Caller Method was HTTP Digest |
| 03:32.10 | axisys | just changed it to none |
| 03:32.14 | axisys | let me try again |
| 03:32.35 | axisys | Voip Called Auth Method that is |
| 03:35.53 | axisys | [TK]D-Fender: different error now http://pastebin.com/f5b4439bc |
| 03:37.12 | [TK]D-Fender | axisys: You likely haven't set up its dialplan properly (the SPA's). Go read the guides at www.voxilla.com in their forums |
| 03:37.30 | axisys | [TK]D-Fender: ok |
| 03:39.56 | axisys | i am at the wizard tool.. ip address of the adapter is the LAN ip 192.168.0.1 (default) or the WAN IP (10.0.0.1) ? |
| 03:40.46 | [TK]D-Fender | axisys: the one you've been using so far. And I've never used their wizrd. I said the guides in the forum, but whatever |
| 03:40.50 | scooby2 | whats the preferred place to buy phones and equipment from these days? |
| 03:41.39 | axisys | [TK]D-Fender: the spa 3102 has two IPS .. one on lan side and one on wan side |
| 03:41.49 | axisys | lan aka router |
| 03:42.02 | [TK]D-Fender | axisys: just use the one you've been using all this time |
| 03:42.02 | axisys | wan is the one that connects to asterisk server |
| 03:42.05 | mchou | axisys: where did you buy your SPA-3102? |
| 03:42.20 | [TK]D-Fender | scooby2: Depends where you are |
| 03:42.39 | axisys | not sure... its the linksys spa 3102 .. googled and bought it ,, do not remember |
| 03:42.52 | axisys | mchou: ^^ |
| 03:44.55 | scooby2 | usa. I always ordered from voipsupply but there are a couple places cheaper. discount voip outlet, 888voipstore, etc |
| 03:47.16 | scooby2 | cisco needs to kill off the linksys SPA941/942 |
| 03:47.17 | [TK]D-Fender | scooby2: for USA and often Canda : www.telephonydepot.com |
| 03:47.25 | [TK]D-Fender | scooby2: And why is that? |
| 03:48.28 | scooby2 | they fail quite often and I cant buy anything else until they no longer exist |
| 03:48.59 | [TK]D-Fender | scooby2: Well in North America there is very little reason to consider anything except Polycom. |
| 03:49.18 | [TK]D-Fender | scooby2: And whats this about not being able to buy anything else? |
| 03:49.28 | mchou | yeah, I dont get that |
| 03:50.03 | scooby2 | [TK]D-Fender: management wants them to be all the same |
| 03:50.19 | mchou | scooby2: that can be solved |
| 03:50.21 | mchou | :) |
| 03:50.45 | mchou | refit the whole place with polycomms |
| 03:50.46 | scooby2 | and yes Polycom is what I would prefer but when I walked into this job they already owned 400 of these Linksys phones |
| 03:50.53 | [TK]D-Fender | scooby2: Make one mistake and then sign on to support it forever. The Bush administration could use your kind of "dedication"! |
| 03:51.08 | mchou | lol |
| 03:51.27 | [TK]D-Fender | I'm sorry. Thats totally inaccurate. |
| 03:51.30 | mchou | "you break it, you buy it" |
| 03:51.35 | [TK]D-Fender | they've made a LOT more mistakes.. |
| 03:53.17 | mchou | how the shrub got elected the 2nd time way way beyond me |
| 03:53.59 | scooby2 | not like our two choices this time are much better |
| 03:54.18 | mchou | you gotta be kidding me |
| 03:54.54 | mchou | virtually anything would be an improvement over shrub |
| 03:55.43 | drmessano | McCain is what dumber than Bush |
| 03:55.44 | mchou | I don't pay my prez to talk down to me as if I'm a retard |
| 03:55.46 | drmessano | way* |
| 03:57.00 | mchou | Bernacke didnt even acknowledge the existence of the "R" word until recently |
| 03:57.09 | mchou | Talk about denial |
| 03:58.44 | jblack | On one side, a person that voted for FISA. On the other, someone that lacks diplomatic skill financial discipline. |
| 03:59.08 | jblack | We should all move to Texas and seceed. |
| 03:59.20 | mchou | haha!! |
| 03:59.30 | mchou | good riddance |
| 03:59.44 | scooby2 | I hear Canada is nice |
| 03:59.59 | jblack | I hear they're not doing as good as they used to either. |
| 04:00.00 | scooby2 | and they have real beer |
| 04:00.02 | mchou | too many smokers there :) |
| 04:00.23 | [TK]D-Fender | mchou: Banned in public places & bars actually |
| 04:00.54 | mchou | [TK]D-Fender: plenty canadians smoking out on the street |
| 04:01.02 | jblack | We need some sort of "Liberty Island", where there aren't any sorts of unnecessary laws. |
| 04:01.07 | [TK]D-Fender | mchou: Same everywhere. |
| 04:01.16 | mchou | [TK]D-Fender: no in CA |
| 04:01.17 | scooby2 | its called Libertarian |
| 04:01.24 | mchou | not* |
| 04:01.31 | jblack | No, it's not called libertarian. |
| 04:01.33 | [TK]D-Fender | RP 2008!!! |
| 04:01.45 | [TK]D-Fender | oh wait... that ship has sailed... AND 'berged |
| 04:01.54 | jblack | libertarian is too far in the opposite direction. |
| 04:01.59 | scooby2 | i'm pissed RP did not get the libertarian nomination |
| 04:02.03 | jblack | I like public fire departments. |
| 04:02.12 | [TK]D-Fender | scooby2: He wasn't RUNNING for it. |
| 04:02.25 | scooby2 | jblack: you obviously know nothing about the LP |
| 04:02.26 | jblack | I'm gonna vote for ron paul in the election. |
| 04:02.35 | scooby2 | i'm writing in RP |
| 04:03.12 | [TK]D-Fender | In an RP world, Bernanke wouldn't be a problem, only a foot-note |
| 04:03.12 | jblack | I know nothing, eh? |
| 04:03.30 | scooby2 | [TK]D-Fender: I know. Instead we have these two repubicans that want to get on the LP band wagon (Barr/Root) |
| 04:03.58 | [TK]D-Fender | scooby2: Barr means well, but has NO presence. |
| 04:04.00 | jblack | You don't consider it possible that I might know a few things about parts of the LP that you don't? |
| 04:04.30 | scooby2 | nope |
| 04:05.01 | [TK]D-Fender | scooby2: Frankly Barack is a decent enough choice in my books. I'd have preferred Kucinich or Gravel, but those weren't to be. RP was a tie with Kucinich for me. |
| 04:05.25 | jblack | I was gonna vote for Obama until FISA. I even sent him a couple hundred bucks. |
| 04:06.01 | scooby2 | Barack has not done a damn thing. He has more experience with dealing with the Mob and other crooks around here |
| 04:06.27 | mchou | well, I call that experience :) |
| 04:06.33 | jblack | lol |
| 04:06.46 | [TK]D-Fender | jblack: He has vowed to fight those particular clauses. I do wonder if he has an ace up his sleeve |
| 04:07.36 | scooby2 | he is like any other politician. He says whatever will get him the most votes and then will never do it |
| 04:07.37 | jblack | Mayhap he does, mayhap he don't. Regardless, I can't vote for someone that voted for FISA. I'm also voting against both of my senators. |
| 04:07.38 | mchou | scooby2: if you can swim with the sharks it beats being a guppy |
| 04:07.43 | jblack | Well, one of them. I don't know which one is up this time |
| 04:07.55 | axisys | [TK]D-Fender: still no luck http://pastebin.com/f1e40cd24 |
| 04:08.03 | [TK]D-Fender | scooby2: Look at Barack's legislative record for bills. thats a lot of "experience" and McSame doesn't even SHOW UP unless to "Nay" things he later takes credit for. |
| 04:08.30 | [TK]D-Fender | axisys: Go actually read the forum and keep at it |
| 04:08.47 | jblack | is out of this conversation. Too many insults |
| 04:09.02 | jblack | And oversimplifications, now that I think of it. |
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| 04:46.13 | baliktad | does * keep CDR's by default? |
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| 05:00.51 | jblack | baliktad: It did here, in csv format. |
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| 05:41.50 | ReDNeQ | anyone have experience configuring polycom ip330? |
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| 05:44.39 | webman | how can I force rtp (sip) to go via asterisk, I have canreinvite=no already ... |
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| 06:26.05 | AlexGC | good evening gentlemen |
| 06:33.25 | kash | anyone here in .jp ? |
| 06:37.22 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.3) |
| 06:38.38 | BeeBuu | how can i catch DTMF stream when i call out? |
| 06:45.10 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 06:58.04 | tzafrir_laptop | kash, not me, but ask your question anyway |
| 06:59.51 | kash | just wondering who could test cross-pacific bandwidth for me |
| 07:04.23 | dominic1 | is there any syntax highligthning for ael in any editor? |
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| 07:05.58 | tzafrir_laptop | dominic1, is C close enough? |
| 07:09.22 | ReDNeQ | anyone have a link/doc that tells you how to configure polycom phone to work with asterisk |
| 07:11.56 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 07:12.27 | dominic1 | no.... sorry |
| 07:12.34 | gones | :-D |
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| 07:52.59 | dominic1 | is it possible to write a sql statement in func_odbc.conf over more than one line? |
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| 08:16.45 | foexle | hi guys, i have a question: Need Asterisk a zaptel interface ? ... So for all cards ? I would use Astersik with BIR or PRI-cards |
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| 08:20.30 | tzafrir_laptop | foexle, what cards do you have? |
| 08:20.55 | foexle | avm |
| 08:21.20 | foexle | sec i look the specification of this card |
| 08:22.34 | tzafrir_laptop | AVM does not work with chan_zap |
| 08:22.38 | tzafrir_laptop | And it is BRI |
| 08:22.58 | tzafrir_laptop | It works with misdn and probably with chan_capi as well |
| 08:23.14 | foexle | ah ok |
| 08:23.27 | foexle | so i dont need zaptel for this |
| 08:25.44 | tzafrir_laptop | no |
| 08:26.26 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 08:26.48 | foexle | ok ty i try it with misdn :> |
| 08:27.14 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
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| 08:36.28 | roxlu | goodmorning everyone! |
| 08:37.02 | BeeBuu | good evening. roxlu |
| 08:37.12 | roxlu | I'm trying to connect to my provider but I get a "SIP/2.0 401 Unauthorized" message |
| 08:37.15 | dominic1 | morning! |
| 08:37.15 | roxlu | evening BeeBuu ;-) |
| 08:37.45 | roxlu | I did not change anything in my config.. I only changed my network (new modem/router/etc..) |
| 08:37.55 | roxlu | I can make call outside, but not "inside" |
| 08:38.53 | roxlu | Does someoen know where I can look to fix this? |
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| 08:51.45 | BeeBuu | when i call out, how can i catch the DTMF? |
| 08:55.41 | *** join/#asterisk cod3hax0r (n=hellowor@222.127.173.76) |
| 08:55.43 | cod3hax0r | hi guys |
| 08:55.54 | cod3hax0r | anyone running asterisk on openvz |
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| 09:08.29 | hi365 | can anyone recomend a gsm-sip or gsm-pci solution? |
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| 09:09.00 | tzafrir_laptop | ~seen anyone |
| 09:09.02 | jbot | anyone <n=randy@p1043-ipbf216hodogaya.kanagawa.ocn.ne.jp> was last seen on IRC in channel #debian, 54d 23h 24m 55s ago, saying: 'Nope.'. |
| 09:09.30 | tzafrir_laptop | cod3hax0r, anyone is not going to answer you, I guess. Maybe just ask? |
| 09:09.30 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 09:10.02 | dominic1 | I want to count the parameters in a cut. Is there any possibility to do this? |
| 09:10.31 | dominic1 | $var = 123, 1234, 12345 and I want that it returns 3 |
| 09:15.05 | dominic1 | FIELDQTY is my friend |
| 09:15.44 | roxlu | I've got WAN <--> ROUTER <--> ASTERISK ... Do I need to set NAT=yes in my sip.conf |
| 09:16.27 | cod3hax0r | yes to your extensions behind the router |
| 09:16.40 | roxlu | okay |
| 09:16.50 | roxlu | cod3hax0r: and as type=[??] |
| 09:17.19 | cod3hax0r | you have to put in the general settings in sip.conf nat=yes |
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| 09:19.16 | roxlu | cod3hax0r: why do I need to tell asterisk it's behind nat? (nat is used to make sure computers behind the ruter a accessible ?) |
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| 09:20.58 | roxlu | ... cod3hax0r or do I need to open more ports than 5090 ? |
| 09:22.09 | tzafrir_laptop | bristuff's dial has an option 'c' for "callback: ring once and hangup" |
| 09:22.21 | tzafrir_laptop | Any way to do it without the extra option? |
| 09:22.37 | tzafrir_laptop | callback-initiation, actually |
| 09:28.12 | cod3hax0r | take a look at the ports of rtp.conf |
| 09:28.14 | roxlu | I did: core set debug 6 |
| 09:28.19 | cod3hax0r | thats the udp ports your going to open |
| 09:28.27 | cod3hax0r | aside from 5060 |
| 09:28.35 | roxlu | Now I see: REGISTER ATTEMPT 231312312@prov.nl@IP_ADDRESS.. |
| 09:28.51 | roxlu | but the IP_ADDRESS is my old IP address... can I change that somewhere? |
| 09:31.55 | roxlu | nice it works! |
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| 09:39.07 | salzh | hi, does Asterisk support playing raw pcm audio file? |
| 09:43.33 | roxlu | I'm having a really strange problem here.. I can make outgoing calls prefectly. But incomings calls work only now and than..?? is this a typical problem? |
| 09:53.04 | mchou | roxlu: most likely a firewall issue |
| 09:53.37 | roxlu | mchou: but it works 'now and than' ? so sometimes it works ... sometimes not? how could that be a firewall thing |
| 09:54.07 | mchou | roxlu: ip connection tracking times out |
| 09:54.23 | roxlu | okay |
| 09:54.28 | roxlu | is there a fix for that? |
| 09:54.48 | mchou | roxlu: google |
| 09:55.08 | roxlu | mchou: .buth for what? |
| 09:55.16 | mchou | wtf. |
| 09:56.12 | roxlu | ;-( really do I need to look for "asterik connection tracking times out" ? |
| 09:56.16 | roxlu | or "router" ? |
| 09:56.52 | mchou | roxlu: SIP connection tracking |
| 09:57.11 | roxlu | okay |
| 09:57.13 | roxlu | I do get: |
| 09:57.25 | roxlu | <PROTECTED> |
| 09:57.44 | roxlu | and removing nat="yes" also works |
| 10:03.34 | roxlu | mchou: Could it also be something with "session-timers" in asterisk? |
| 10:05.20 | tzafrir_laptop | salzh, yes, asterisk supports that. Though only in supported file formats |
| 10:05.30 | tzafrir_laptop | see core show file formats |
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| 10:13.44 | codestr0m | anyone know a good eu based company which direct pri links and support bank draft payment. preferably in Spain |
| 10:14.20 | codestr0m | it's for a business account and the customer prefers prepay vs cc |
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| 10:37.19 | zapp-branigan | hi i have a problem i'm using sip asterisk nat , but when i do a sip set debug i not see the ip rewrite when he do a transfer then only do internal transfers, who can i solve this ? |
| 10:38.12 | zapp-branigan | i have nat=yes , rtp=no , canreinvite=yes |
| 10:38.49 | zapp-branigan | and the phone have the port nated in the router |
| 10:40.38 | codestr0m | zapp-branigan: umm.. I've no idea what you're problem is, but just a guess in the dark qualify=yes might help you |
| 10:41.24 | zapp-branigan | y have this option, |
| 10:42.30 | zapp-branigan | i have the probklem who i can reinvite in the same lan because the internal ip can see the other |
| 10:43.09 | zapp-branigan | but when is in diferent ip the asterisk send for the transfer the internal ip |
| 10:43.10 | zapp-branigan | o=root 12343 12344 IN IP4 192.168.1.150 |
| 10:43.23 | zapp-branigan | then hge cant do a transfer |
| 10:44.04 | zapp-branigan | because 192.168.1.150 must be the real ip and the prort must be the port of the client |
| 10:44.57 | zapp-branigan | but the asterisk can't detect that this a call from utside the same lan |
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| 10:53.52 | puzzled | hi |
| 10:54.10 | cod3hax0r | hi pzzled |
| 10:54.15 | puzzled | anyone know what would cause a 415 Unsupported Media Type as a response to an invite? |
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| 11:36.05 | l0verb0y | hey |
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| 12:30.39 | zippytech2 | i have 2 asterisk boxes that all of a sudden i cannot call out a tdm 400 port says TRUNK Dial failed due to CHANUNAVAIL |
| 12:31.37 | [TK]D-Fender | zippytech2: take a look in "core show channels" |
| 12:31.52 | zippytech2 | from cli |
| 12:32.03 | [TK]D-Fender | zippytech2: and then verify "zap show status" and "zap show channels" |
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| 12:33.42 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 12:33.57 | verywiseman | need softphone to download |
| 12:34.13 | zippytech2 | core shows 0 active |
| 12:34.15 | ManxPower | I'm answering (non-GUI, non-package) questions on #asterisk-cli. |
| 12:34.15 | [TK]D-Fender | verywiseman: what platform? |
| 12:34.30 | verywiseman | linux |
| 12:34.35 | [TK]D-Fender | ~ekiga |
| 12:34.36 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 12:34.40 | [TK]D-Fender | verywiseman: ^^^ |
| 12:34.59 | zippytech2 | zap shows pseudo default and zap 4 default |
| 12:35.43 | zippytech2 | the only thing that has changed is i just got on new pot line |
| 12:35.48 | zippytech2 | it's almost like it dont' detect it |
| 12:35.48 | [TK]D-Fender | zippytech2: PASTEBIN all of this and your dial attempts CLI output at verbose 10 |
| 12:35.49 | [TK]D-Fender | ~pb |
| 12:35.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:35.52 | [TK]D-Fender | ^^^^^ |
| 12:36.10 | [TK]D-Fender | zippytech2: * doesn't detect anything. You have to configure it |
| 12:38.09 | zippytech2 | well ok if i have the line in use say regular phone connected it asterk will attempt to work and you can here the people on the line but if the line is not in use it will error out |
| 12:39.00 | [TK]D-Fender | zippytech2: * should not error out. If the port is not in use by * it will assume its free |
| 12:39.17 | [TK]D-Fender | zippytech2: Now please prpvide the information I have requested. |
| 12:39.20 | [TK]D-Fender | provide* |
| 12:42.21 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 12:42.46 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 12:44.34 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:44.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 12:45.37 | oej | Afternoon |
| 12:46.34 | *** join/#asterisk zippytech2 (n=chatzill@node220.38.251.72.1dial.com) |
| 12:48.35 | k-man | i have a voip capable adsl modem that connects to a voip provider - i also want to connect asterisk to another sip provider from behind the modem |
| 12:48.40 | k-man | is that possible? |
| 12:49.03 | ManxPower | k-man: onl;y if the vendor permits that |
| 12:49.12 | [TK]D-Fender | k-man: Depends if your other device gets in the way or not |
| 12:49.21 | *** part/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 12:49.28 | ManxPower | k-man: but we need to know the actual device make/model, but don't expect you do be able to do what you want. |
| 12:50.57 | k-man | its a billion 7404 vgp-m |
| 12:51.00 | [TK]D-Fender | k-man: Consider replacing it |
| 12:51.06 | k-man | oh |
| 12:51.14 | k-man | not the answer i was hoping for |
| 12:51.16 | [TK]D-Fender | k-man: Can you forward the ports * needs on it? |
| 12:51.56 | k-man | [TK]D-Fender, i can forward most ports, but it seems i cannot forward 5060 - but its using that port already for its sip connection |
| 12:52.23 | [TK]D-Fender | k-man: Then you're screwed |
| 12:52.33 | k-man | oh well |
| 12:52.56 | k-man | i could turn off all voip on the modem and use asterisk |
| 12:53.06 | ManxPower | k-man: you could TRY. 8-) |
| 12:53.11 | k-man | yeah |
| 12:54.47 | foexle | hi guys , i must install the mISDN driver before asterisk ? if yes, why ? |
| 12:55.14 | [TK]D-Fender | foexle: Probably and its a good idea so * knows to compile support for it |
| 12:55.40 | foexle | hmm ok -.- |
| 12:55.52 | k-man | damn thats annoying |
| 12:55.52 | foexle | i've first installed asterisk and then mISDN -.- |
| 12:56.02 | k-man | i was really hoping to be able to play around with asterisk |
| 12:56.26 | k-man | [TK]D-Fender, what's a better adsl/voip option? |
| 12:56.44 | [TK]D-Fender | k-man: Just change your modem |
| 12:57.05 | [TK]D-Fender | foexle: Then just re-extract & compile * again. |
| 12:57.21 | foexle | ok |
| 12:57.24 | foexle | thx |
| 13:03.02 | *** join/#asterisk zbenjamin (n=benjamin@p54B5F634.dip.t-dialin.net) |
| 13:03.05 | zbenjamin | hi all |
| 13:03.20 | *** join/#asterisk zippytech_ (n=chatzill@node11.33.251.72.1dial.com) |
| 13:03.38 | zippytech_ | http://pastebin.com/d38e76b94 |
| 13:04.01 | zippytech_ | that is the result of the not in use call attempt |
| 13:04.28 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 13:04.49 | zbenjamin | in my company we want to give people a way to work at home. But we want them to use the telephone from the company and not their home telephone. My idea was to use asterisk as a SIP voip provider to give them access to our company ISDN stuff. |
| 13:04.55 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:04.57 | zbenjamin | Is this possible? and if yes where to start? |
| 13:05.20 | [TK]D-Fender | zippytech_: And where is the channel dump? |
| 13:05.26 | ManxPower | zippytech_ Are you using a GUI? |
| 13:05.50 | [TK]D-Fender | zippytech_: And you are picking just Zap/4 there. So no implication of uing this "new line" you were referring to. |
| 13:05.55 | [TK]D-Fender | ManxPower: CLEARLY. Just ignore |
| 13:06.06 | [TK]D-Fender | zbenjamin: Start with : |
| 13:06.08 | [TK]D-Fender | ~book |
| 13:06.09 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:06.10 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:06.15 | zippytech_ | just the pot itself |
| 13:06.18 | ManxPower | [TK]D-Fender: I'll ignore AFTER I finish the Voodoo Doll and Spells |
| 13:06.22 | [TK]D-Fender | zbenjamin: And clearly yes, its more than possible |
| 13:06.26 | zbenjamin | [TK]D-Fender, so it is possible ? |
| 13:06.30 | zbenjamin | okay ;) |
| 13:06.44 | zbenjamin | yay a free pdf |
| 13:07.05 | [TK]D-Fender | zippytech_: Go verify the port you plugged your line into, and plug a phone in parallel to test it |
| 13:07.30 | ManxPower | For people with NON-GUI questions I'll be on #asterisk-cli |
| 13:07.31 | zippytech_ | http://pastebin.com/d3343df18 |
| 13:07.33 | *** part/#asterisk ManxPower (n=manxpowe@99.sub-70-222-55.myvzw.com) |
| 13:07.57 | zippytech_ | this is what it does if the line is in use by a modem i can here it |
| 13:08.21 | zippytech_ | yes it works i can call out |
| 13:08.30 | zippytech_ | with a cheap phone |
| 13:09.12 | zippytech_ | i even changed tdm cards and computers both worked a week ago now neither work |
| 13:10.10 | verywiseman | [TK]D-Fender, thank you , but what is ^^^ meaning? |
| 13:10.21 | zippytech_ | http://pastebin.com/da5c8783 |
| 13:10.39 | [TK]D-Fender | verywiseman: Arrows pointing UP to the link for you to see |
| 13:10.40 | zippytech_ | here is the channel paste |
| 13:11.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:11.08 | [TK]D-Fender | zippytech_: What "modem"? |
| 13:11.32 | zippytech_ | i have dialup at this office that used the same line as the asterisk box |
| 13:12.08 | [TK]D-Fender | zippytech_: Guess that might cause a problem... |
| 13:12.14 | zippytech_ | if the modem is on the net there are not asterisk errors, if it is not connected to the net i get the channel problem |
| 13:13.04 | zippytech_ | i can turn it on and off no problem there but it looks like asterisk cannot see a dail tone when it's there |
| 13:13.10 | zippytech_ | just a gues |
| 13:13.51 | [TK]D-Fender | zippytech_: Disable call-progress period. |
| 13:13.56 | zippytech_ | is it posible that when the phone company installed the phone line some thing changed |
| 13:14.06 | zippytech_ | ? |
| 13:14.15 | zippytech_ | how |
| 13:14.36 | [TK]D-Fender | zippytech_: Go look in your interface, we do NOT support it here |
| 13:14.38 | zippytech_ | as both boxes worked a week ago on the old lines |
| 13:17.12 | zippytech_ | if there was a config problem with asterisk why would it matter if the line was in use |
| 13:17.44 | zippytech_ | both paste show to differnet results and only diff is the line is in use |
| 13:17.57 | zippytech_ | and the other is not |
| 13:18.47 | [TK]D-Fender | zippytech_: GO DISABLE CALL PROGRESS <- |
| 13:19.18 | *** join/#asterisk heison (n=heison@209.195.83.163) |
| 13:21.25 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:21.26 | anonymouz666 | I wonder why zaptel ALWAYS detect the WRONG path to install the kernel modules... |
| 13:21.53 | hsv-al | hello fellow internet addicts - are we all looking forward to another long&glorious week of internet/irc addiction,and ADD/ADHD fueled lust for technology :) |
| 13:22.13 | *** join/#asterisk moy (n=moy@nat/ibm/x-c8fe5b3cb97b41f8) |
| 13:22.44 | heison | ~seen bkw |
| 13:22.47 | jbot | bkw <n=bkw@tor.lindesign.se> was last seen on IRC in channel #debian, 5d 2h 58m 17s ago, saying: 'Is it possible to findout the currently used font in a terminal?'. |
| 13:23.07 | *** join/#asterisk c4t3l (n=rcallico@mail.questia.com) |
| 13:23.23 | [TK]D-Fender | anonymouz666: What OS? |
| 13:23.25 | zippytech_ | the default is diabled |
| 13:24.08 | [TK]D-Fender | zippytech_: You're going to need to doa channel dump IMMDIATELY following your dial attemtp next |
| 13:24.15 | anonymouz666 | CentOS 5 |
| 13:24.42 | [TK]D-Fender | anonymouz666: Must have done something wrong then because I've been using it for many releases and no issues whatsoever |
| 13:25.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:28.24 | tzafrir_laptop | anonymouz666, what do you mean by "wrong path"? Could you please be specific? |
| 13:28.31 | hsv-al | tzafrir |
| 13:28.36 | hsv-al | ive been using ur scripts a while |
| 13:28.43 | hsv-al | they all good, are those going in 1.6 eventually |
| 13:28.50 | hsv-al | for more "user friendly baseline install process" ? |
| 13:29.27 | tzafrir_laptop | hsv-al, at least some are |
| 13:30.05 | anonymouz666 | tzafrir_laptop: it SEEMS zaptel is not using the 'uname -r' output to install the modules in /lib/modules/kernel-version |
| 13:31.03 | tzafrir_laptop | anonymouz666, It should. Unless it gets overriden with KVERS |
| 13:31.12 | tzafrir_laptop | what distro, kernel version? |
| 13:31.26 | anonymouz666 | KVERS is the default in the Makefile |
| 13:31.55 | anonymouz666 | 2.6.18-53.el5 - CentOS 5 |
| 13:32.19 | tzafrir_laptop | Is it reproducable? |
| 13:32.40 | anonymouz666 | Zaptel is install things in: 2.6.18-8.el5 |
| 13:32.43 | anonymouz666 | installing |
| 13:33.05 | anonymouz666 | don't know if it's a bug, but I just want to understand this behaviour. |
| 13:33.31 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 13:34.31 | anonymouz666 | then... when I modprobe...ztdummy: disagrees about version of symbol zt_receive |
| 13:34.40 | *** join/#asterisk tiav (n=tiav@fw.sj.tdf-pmm.net) |
| 13:35.51 | *** join/#asterisk bronx (i=[U2FsdGV@pallas.utanet.fi) |
| 13:36.19 | *** join/#asterisk ManxPower (n=manxpowe@99.sub-70-222-55.myvzw.com) |
| 13:36.26 | *** join/#asterisk ReDNeQ (n=ReDNeQ@75.148.217.225) |
| 13:36.28 | cod3hax0r | hi guys im getting an UNABLE TO OPEN PSEUDO Device when creating a meetme conference |
| 13:36.42 | *** join/#asterisk tiav (n=tiav@fw.sj.tdf-pmm.net) |
| 13:36.55 | ManxPower | cod3hax0r: zaptel kernel modules not loaded into memory |
| 13:37.05 | ManxPower | maybe chan_zap.so is not loaded into memory? |
| 13:37.46 | cod3hax0r | ok... im using openvz... zaptel and ztdummy are loaded on the HN already... but my openvz container cant access those modules... |
| 13:38.22 | ManxPower | cod3hax0r: not our problem. Meetme requires access to the zaptel and ztdummy kernel modules. If you don't have access then you can't use MeetMe. |
| 13:38.39 | eric_hill | [TK]D-Fender: Are you a machine? :) |
| 13:38.40 | anonymouz666 | tzafrir_laptop: strange because the Makefile looks correct... using KVERS that executes 'uname -r' to determine the path |
| 13:38.43 | cod3hax0r | ok.. |
| 13:38.55 | anonymouz666 | but when you do 'make install' the path installed is incorrect |
| 13:39.13 | ManxPower | anonymouz666: for Zaptel? |
| 13:39.22 | anonymouz666 | eric_hill: we gotta use AMD application with [TK]D-Fender to discover that. |
| 13:39.42 | anonymouz666 | ManxPower: yes |
| 13:39.43 | ManxPower | anonymouz666: I can help you further on #asterisk-cli |
| 13:40.18 | [TK]D-Fender | eric_hill: Depends on your pihilosphical beliefs |
| 13:41.07 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:41.33 | c4t3l | cod3hax0r: have you checked the perms on /dev/zap |
| 13:43.29 | eric_hill | [TK]D-Fender: I believe that I've been in a short coma for the past 8 hours and you're still on here. I'm not convinced you're human. |
| 13:44.20 | [TK]D-Fender | eric_hill: And I've been OFFLINE for those hours. So from your POV I guess it would seem "constant" |
| 13:46.21 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:47.46 | eric_hill | [TK]D-Fender: Well that explains it a bit... :) |
| 13:48.44 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:53.18 | miloux | any friendly soul that could point me in the right directions when it comes to allowing calls only during certain hours on certain extensions? |
| 13:53.43 | [TK]D-Fender | miloux: "core show application gotoiftime" |
| 13:53.58 | miloux | [TK]D-Fender: Thanks! |
| 13:55.10 | miloux | so if i put 1000,1, gotoiftime <all the options needed and that certain time is not now...will it just continue with 1000,2, ? |
| 13:55.30 | eric_hill | miloux: Correct. |
| 13:55.33 | miloux | Ty |
| 13:55.38 | [TK]D-Fender | miloux: It jumpts to where you tell it to if it meets the criteria. |
| 13:59.46 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:59.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.57 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:06.54 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:07.25 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 14:07.30 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:10.00 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:13.20 | *** join/#asterisk FabiOne (n=FabiOne@host10-25-static.104-82-b.business.telecomitalia.it) |
| 14:14.09 | LemensTS | http://pastebin.com/m56aa04b8 Im getting this while trying to run 'make' on zaptel 1.4.11 on debian etch |
| 14:15.15 | tzafrir_laptop | Could you please provide a more complete trace? |
| 14:15.40 | tzafrir_laptop | Anyway, I like "make: no rule to make target `Makefile'" |
| 14:15.44 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 14:16.54 | miloux | Its probable that you dont have the headers for your kernel installed... try sudo apt-get install linux-headers-2.6.18-6-686 |
| 14:17.04 | FabiOne | hi all |
| 14:17.30 | *** join/#asterisk Segnale007 (n=Segnale0@host68-30-dynamic.19-79-r.retail.telecomitalia.it) |
| 14:17.50 | FabiOne | i need to buy a pci card to use with a pri line with 1000 numbers |
| 14:18.33 | FabiOne | i don't have idea about right price for it |
| 14:18.49 | LemensTS | miloux: they are installed, i tried to run that command again and it said they were insalled. Hey, how do i print the results of 'make' to a file so i can pastebin it |
| 14:20.06 | dr_gogeta86 | hey |
| 14:20.26 | dr_gogeta86 | anyone here is working with rtp conntrak |
| 14:20.53 | [TK]D-Fender | FabiOne: www.telephonydepot.com |
| 14:21.35 | bboschman | dr_gogeta86, is there any rtp conntrack? |
| 14:21.47 | *** join/#asterisk spokra (n=spokra@host093-179-177.sea0.speakeasy.net) |
| 14:22.03 | bboschman | afaik sip/sdp is dealing with rtp-port assignment |
| 14:22.47 | dr_gogeta86 | bboschman: essentially |
| 14:22.48 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:22.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:23.05 | dr_gogeta86 | i must put an asterisk server behind a nat |
| 14:23.36 | bboschman | therefor conntrack_rtp must listen on SIP messages - I don't think there's any module dealing with this |
| 14:24.16 | bboschman | dr_gogeta86, use rtpstart & rtpstop |
| 14:24.31 | bboschman | then you can define which ports needs to be opened/forwarded |
| 14:24.34 | [TK]D-Fender | dr_gogeta86: no need for any iptables trickery, just proper * configuration. |
| 14:24.37 | [TK]D-Fender | ~sipnat |
| 14:24.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:24.45 | [TK]D-Fender | dr_gogeta86: Go read ^^^^ |
| 14:24.59 | bboschman | [TK]D-Fender, just for me - would I have been right? |
| 14:25.06 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 14:25.52 | [TK]D-Fender | bboschman: Don't know, never needed or tried it. |
| 14:27.49 | grandpapadot | Morning, all. Implementing private DUNDi this week between roughtly 30 asterisk 1.4.current servers, any gotchas? |
| 14:27.55 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
| 14:31.53 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-124-162.pools.arcor-ip.net) |
| 14:34.58 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:35.22 | dr_gogeta86 | tnx jbot bboschman |
| 14:35.33 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:35.35 | dr_gogeta86 | [TK]D-Fender: |
| 14:38.55 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:42.14 | Blackvel | hi. who has experience using ptmp clip no screening, call transfer, call diversion, zaphfc, capi hold/ect (explicit call transfer) and telco forwarding? I am searching for a way to accept pstn calls, do IVR in Asterisk and connect the outside call back to my mobile phone (over pstn). I can not do call deflection because of IVR! And I do not want to loose the caller callerid either! (which I do when "clip no screening" is not s |
| 14:43.32 | Blackvel | is capi hold/ect sending my own phone number to the 3rd party or how can I send the 1st phone number to the 3rd party (mobile phone)? |
| 14:45.47 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 14:45.59 | *** join/#asterisk Talnakh (n=Talnakh@c180-25.i02-6.onvol.net) |
| 14:47.25 | Talnakh | hey everyone, could someone point me towards a guide on how to configure Asterisk with VoIP gateways, such as Patton Smartnode |
| 14:47.37 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 14:48.18 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 14:48.46 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:50.00 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 14:50.01 | gnorbert | Hi. I can connect to Asterisk with Linphone, but when I try to do it second time, I get the following message: Warning: UDP port 5039 seems already in use ! Cannot initialize. |
| 14:50.18 | gnorbert | Does somebody have an idea, what can be the problem? |
| 14:51.06 | Talnakh | 2 apps running at the same time? |
| 14:51.13 | [TK]D-Fender | gnorbert: "netstat -an|grep 5039". Clearly something is already using it. Go find out what |
| 14:51.23 | gnorbert | (Linphone writes it and it can't connect to an other server either. Or is it a Linphone problem? Asterisk should kick it from the conference. |
| 14:52.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:53.30 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) |
| 14:53.45 | thing1 | does anyone here have a cisco 7970 ip phone |
| 14:54.00 | thing1 | i just got one yesterday and cannot get it to load the firmware |
| 14:54.06 | gnorbert | It writes this: http://pastebin.com/d2c742866 |
| 14:54.09 | thing1 | sip firmware |
| 14:54.44 | gnorbert | Hmm, it would have been shorter to paste it direct here. :) |
| 14:56.36 | gnorbert | It was written on the linphone's computer, on the server's computer was nothing. |
| 14:56.40 | [TK]D-Fender | gnorbert: Well something is binding to it. Go find out what |
| 14:56.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:56.56 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:57.42 | gnorbert | Only Linphone can be that, but then it would mean, that * didn't kick it properly. |
| 15:00.02 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.44 | [TK]D-Fender | gnorbert: * doesn't "kick" anything. If it binds a port, then it binds a port. the fact * may reject connections from it is besides the poitn |
| 15:01.22 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-251-219.revip2.asianet.co.th) |
| 15:01.59 | gnorbert | [TK]D-Fender: The connection ended when I wrote meetme kick 1001 all in *, shouldn't have it disconnect fully? |
| 15:02.23 | [TK]D-Fender | gnorbert: * doesn't make a softphone stop BINDING its port. |
| 15:03.05 | oilinki3 | I wish to forward an call from 1 sip-number to another, both external, run by the operators. |
| 15:03.27 | gnorbert | [TK]D-Fender: Sorry, I thought it does. But then how can I make this port free? |
| 15:03.31 | oilinki3 | I can get the rinning, but no voice at all. |
| 15:03.45 | [TK]D-Fender | gnorbert: stop the application that is iusing it. |
| 15:03.48 | [TK]D-Fender | using* |
| 15:03.54 | oilinki3 | both codecs are the same g729 which has been using with both operators |
| 15:03.58 | gnorbert | I already stoped. |
| 15:04.07 | oilinki3 | what else should I take in concideration? |
| 15:04.21 | [TK]D-Fender | gnorbert: Then you have clearly failed. Look harder. We both saw that its still bound. |
| 15:04.31 | [TK]D-Fender | oilinkiSome DETAILS would be nice. |
| 15:04.36 | gnorbert | On the other hand, Asterisk sees Linphone connected after quit from Linphone. |
| 15:05.00 | gnorbert | That's, why I used kick instead of quit. |
| 15:05.42 | oilinki3 | [TK]D-Fender: asterisk running on the adsl-router. call route would be operator1->asterisk-> operator2. |
| 15:05.44 | [TK]D-Fender | gnorbert: .... your meetme kick doesn't mean ANYTHING! |
| 15:05.51 | *** join/#asterisk roxlu (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 15:05.52 | roxlu | hi |
| 15:05.59 | roxlu | is it possible to use asterisk behind nat? |
| 15:06.04 | gnorbert | [TK]D-Fender: I noticed it. :) |
| 15:06.05 | [TK]D-Fender | oilinki3: And where are they relative to your * box? |
| 15:07.03 | oilinki3 | [TK]D-Fender: my box does registration to both. one to mainly receive calls from PSTN, the other is to make calls out to the PSTN |
| 15:07.09 | hsv-al | just sneezed, and my pomegranite juice spray caused a spark to emerge from a plug socket |
| 15:07.34 | [TK]D-Fender | oilinki3: PASTEBIN a failed call with sip debug enabled. |
| 15:07.36 | [TK]D-Fender | ~pb |
| 15:07.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:07.37 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:07.52 | oilinki3 | [TK]D-Fender: so I basically want to route the calls from finnish number to a thai mobile |
| 15:09.03 | Blackvel | roxlu: yes, wiki lists some important options (port, nat=yes, externhost) |
| 15:09.39 | roxlu | Blackvel: okay thanks! Blackvel ... its working a bit .. outgoing calls work fine, but incoming call not.. is that a common problem with nat? |
| 15:09.42 | [TK]D-Fender | roxlu: Go read the GUIDE : |
| 15:09.43 | [TK]D-Fender | ~sipnat |
| 15:09.44 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:09.44 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 15:09.46 | [TK]D-Fender | ^^^^ |
| 15:12.07 | *** join/#asterisk ronr (n=ron@82-204-104-172.fttx.bbeyond.nl) |
| 15:12.25 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
| 15:12.38 | Blackvel | roxlu: if asterisk has the wrong ip that might be. do you use dyndns or something? I also do port forwarding for udp in my router |
| 15:13.28 | ronr | if I use asterisk to let for example 5 phones (polycoms) ring and one answers, the other four register the call as missed, resulting in things like 200 new missed calls showing in the display. Can I do anything about this? |
| 15:14.12 | Blackvel | is going to shoot this ivr/call diversion to mobile phone thing into knees very soon :) |
| 15:15.18 | [TK]D-Fender | ronr: No. |
| 15:15.37 | Blackvel | any german guys here know about telekom/arcor ECT (Explicit call transfer)? |
| 15:15.40 | ronr | [TK]D-Fender: ok, thx |
| 15:15.51 | oilinki3 | [TK]D-Fender: sip debug, what I got http://pastebin.com/m1b89252a |
| 15:16.28 | grandpapadot | oops.. "module unload pbx_dundi.so" segfaults asterisk 1.4.21.2 |
| 15:16.45 | [TK]D-Fender | oilinki3: Contact: <sip:4422@10.1.1.1> <-- clearly you have not followed the guide. Go read it again |
| 15:17.04 | grandpapadot | That pissed off about a hundred customers... |
| 15:17.40 | oilinki3 | [TK]D-Fender: could you give me a hint how to correct that one |
| 15:17.59 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:18.20 | [TK]D-Fender | oilinki3: Read. The. Guide. |
| 15:19.52 | ManxPower | oilinki3: if you did it right your PUBLIC IP would be there, not your private one |
| 15:20.40 | ManxPower | specifically it indicated your localnet= and externip= settings are wrong |
| 15:20.49 | oilinki3 | ok. I'll try to find where to define that |
| 15:21.00 | ManxPower | ronr: turn that feature off. |
| 15:21.08 | ManxPower | oilinki3: it should tell you in the nat guide. |
| 15:21.10 | oilinki3 | but it's not the externhost ? |
| 15:21.30 | [TK]D-Fender | oilinki3: pastebin your sip.conf masking ONLY passwords. |
| 15:21.31 | ManxPower | oilinki3: use externip= first, once you get it working you can try using externhost. |
| 15:21.38 | Blackvel | oilinki3: interesting thing. what callerid is going to be presented on the thai mobile? your asterisk pstn number or the callerid from the caller who called the finnish number? I want to do external/explicit call transfer or call forwarding too! :) |
| 15:21.46 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:22.22 | ManxPower | Blackvel: that depends on many things. |
| 15:22.27 | ManxPower | But mostly what your carrier allows |
| 15:22.37 | Blackvel | likes asterisk 1.2 :) upgrade from 1.0 to 1.2 got me externhost :) |
| 15:22.45 | oilinki3 | [TK]D-Fender: I'll try to do that. I'll first read some manuals :) |
| 15:23.03 | Blackvel | ManxPower: it's such a pain. I have a working solution with my old pbx + asterisk (to be extended as IVR) |
| 15:23.04 | Blackvel | buuuuut |
| 15:23.12 | oilinki3 | ManxPower: ok. altought my ip is changing 1-2 times a day.. |
| 15:23.34 | Blackvel | I can'T do clip no screening :( I can't send out my callerid which gets passed fine to astersk by clip no screening internal s0 bus :( |
| 15:23.37 | oilinki3 | Blackvel: on the mobile side I can just see my sip-phone number. |
| 15:23.50 | Blackvel | that suxx oilinki3 :) |
| 15:24.06 | Blackvel | i really want to see the callers of my business on my mobile |
| 15:24.09 | oilinki3 | Blackvel: I know but it's better than nothing |
| 15:24.13 | Blackvel | to be able to call them back |
| 15:24.45 | Blackvel | i mean my call diversion thing at telco side (no pickup in 20secs) shows me "forwarding..." on mobile + the external number |
| 15:25.28 | Blackvel | but of course its something different for "unanswered" calls (call deflection) than answering and then trying to forward |
| 15:25.57 | Blackvel | oilinki3: why do you let you asterisk answer the call? cheaper? it is yours mobile? |
| 15:28.00 | oilinki3 | Blackvel: I actually wish to get the transfer from the finnish number to thai mobile |
| 15:28.31 | oilinki3 | eg. my friends and family can call me with an fraction of the costs what it would do in other cases. |
| 15:28.32 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:28.42 | oilinki3 | and they do not need to my tech-experts :) |
| 15:29.22 | Blackvel | how do you call your mobile? over a thai telco sip/iax provider? |
| 15:29.43 | oilinki3 | thai sip-provider |
| 15:30.19 | Blackvel | nice |
| 15:31.14 | Blackvel | anyone using chan_capi with fritz isdn usb (instead of pci)? is it working any good? I do not have a free pci slot for testing :( |
| 15:31.24 | sulex | Hi veryone, I've an extension using MYSQL cmd(exten => 555,6,MYSQL(Query resultid_2 ${connid_2} INSERT\ INTO\ history_0\ SET\ timestamp=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}\, cli=${CALLERID(num)}\, code=${docnum}\, file=${CALLFILENAME})). I don't understand why but in the mysql table the query is stored but the timestamp value is still reported in unix time format regardless my conversion. Any idea? |
| 15:31.31 | sulex | sorry for its lenght |
| 15:33.24 | sulex | a note: in the CLI the timestamp value is reported coorectly: -- Executing MYSQL("Zap/8-1", "Query resultid_2 1 INSERT INTO history_0 SET timestamp=20080804-172145, cli=***, code=***, file=***-55-20080804-172145") in new sta |
| 15:33.28 | keith4 | why do you want to store the timestamp in something that's *not* unixtime? |
| 15:33.49 | sulex | keith4: because that field is reported as it is on a web table |
| 15:34.03 | [TK]D-Fender | sulex: doesn't LOK like unix time to me... |
| 15:34.06 | [TK]D-Fender | LOOK* |
| 15:34.22 | keith4 | imho, you should store it as unixtime, and convert it to a human-readable format when you display it |
| 15:34.49 | sulex | [TK]D-Fender: in fact, the CLI reports the query interpoled correctly but in the table is " 19910777" |
| 15:34.50 | [TK]D-Fender | sulex: 20080804-172145 <-- NORMAL. And EXACTLY what you asked for in ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} |
| 15:35.03 | [TK]D-Fender | sulex: then its your tables fault |
| 15:35.19 | sulex | it's a simple varchar(64) |
| 15:35.51 | sulex | keith4: i think i'll do it in that way... but it's weird :) |
| 15:35.52 | [TK]D-Fender | sulex: I suggest you log this more, and dump everything for us to examine. |
| 15:36.03 | sulex | ok, back in 2 mins |
| 15:37.20 | keith4 | sulex: it's weird? what you want to do is like storing 65 as the string "65" instead of as an int |
| 15:37.32 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:38.29 | sulex | keith4: it's not an int since it has to contain dashes |
| 15:38.43 | keith4 | nevermind |
| 15:38.51 | sulex | anyhow, let me put toghter a pastebin post so you can than blame better ;) |
| 15:38.52 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 15:38.59 | keith4 | it was an analogy |
| 15:41.38 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 15:41.38 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:44.21 | sulex | keith4: http://pastebin.com/d585df564 |
| 15:45.19 | keith4 | regardless of whether or not it's a good way to do it... that is rather odd |
| 15:45.47 | keith4 | you are probably going to have to log the insertion, via mysql |
| 15:45.54 | sulex | what you seggest, keep it in unixtime and make conversions on scripts? |
| 15:46.14 | Blackvel | isn't chan_capi built into bristuff 0.3.0x? |
| 15:47.32 | sulex | keith4: lol, the mysql log reports is correctly :) |
| 15:48.04 | keith4 | sounds like you're headed for #mysql then |
| 15:48.51 | Blackvel | ahhh found it. its only a patch file in patch/asterisk/optional directory :) |
| 15:49.01 | sulex | keith4: no come on, I'm not going to get into that crap. I'm gonna make perl do the conversion and take your suggestion, use unixtime and forget about it :) |
| 15:52.16 | *** join/#asterisk LND (n=Lee@fazer1-adsl.demon.co.uk) |
| 15:53.10 | LemensTS | http://lemensts.pastebin.com/m69f1c316 here is the full 'make' of the zaptel that im having problem with in debian |
| 15:55.10 | *** part/#asterisk Segnale007 (n=Segnale0@host68-30-dynamic.19-79-r.retail.telecomitalia.it) |
| 15:55.44 | sulex | keith4: I'm a fool. I've fixed it changing the table design... instead of varchar this is the way: `timestamp` datetime NOT NULL default '0000-00-00 00:00:00', |
| 15:55.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:56.05 | [TK]D-Fender | sulex: No.. it couldn't POSSIBLY be the fault of your table... |
| 15:56.23 | sulex | eh, now it records it correctly :/ |
| 15:56.28 | [TK]D-Fender | sulex: But glad you found an answer :) |
| 15:56.46 | sulex | lets' call it alternative instead of answer so :) |
| 15:57.26 | sulex | p.s.: I had to change the dialplan date format of course to let it work |
| 16:09.02 | *** join/#asterisk madison (n=madison@208.89.241.9) |
| 16:09.37 | *** join/#asterisk dr_gogeta86 (n=dr_goget@81-208-88-100.ip.fastwebnet.it) |
| 16:09.52 | dr_gogeta86 | re |
| 16:12.14 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 16:16.46 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:16.59 | eric_hill | Any thoughts about what would cause a voice channel on a PRI to take 5 to 10 seconds to establish a voice path after it was answered? |
| 16:19.44 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 16:23.34 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 16:24.03 | Qwell | eric_hill: that seems a bit off |
| 16:24.20 | pta200 | Is there a character limit on the string passed to the Dial application? |
| 16:24.26 | *** part/#asterisk madison (n=madison@208.89.241.9) |
| 16:24.29 | Qwell | pta200: yeah |
| 16:24.48 | Qwell | probably a few k. I bet Corydon76-dig would know |
| 16:25.09 | Qwell | if I had to guess, I'd say...2k? no idea though |
| 16:25.34 | pta200 | that |
| 16:25.42 | pta200 | 's what I've observed |
| 16:25.48 | Qwell | and it's not really the string passed to dial - it would be the entire extension line |
| 16:25.52 | pta200 | Channels names I'm passing to the application make the string too long |
| 16:26.02 | *** join/#asterisk Paige (n=Paige@208.89.241.9) |
| 16:26.16 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 16:26.17 | LemensTS | http://lemensts.pastebin.com/m69f1c316 here is the full 'make' of the zaptel that im having problem with in debian |
| 16:26.17 | pta200 | so the last few get cut off |
| 16:26.32 | Qwell | pta200: how long of a string are you passing? |
| 16:26.35 | Paige | hi, has anyone managed to compile asterisk-1.6-beta4 addons on a freebsd machine? |
| 16:26.49 | pta200 | I will check |
| 16:27.59 | eric_hill | @Qwell: the channel goes active as the call proceeds, a few seconds go by, then a message in the * console that says "Native bridging Zap/11-1 and Zap/1-1" and I have a voice path. |
| 16:28.18 | eric_hill | What would cause "Native bridging" to hold a few seconds before working? |
| 16:28.30 | Qwell | eric_hill: don't know.. |
| 16:28.39 | ManxPower | eric_hill: reinvites |
| 16:28.44 | Qwell | ManxPower: on zap |
| 16:29.12 | ManxPower | Perhaps a cup of coffee is in order. |
| 16:29.37 | eric_hill | I checked the Turbo Button and it was pushed in. |
| 16:29.46 | Qwell | eric_hill: that would do it |
| 16:29.51 | Qwell | (no, not really.) |
| 16:30.16 | eric_hill | @Qwell, actually I looked a bit closer and the call isn't being signalled |
| 16:30.39 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 16:30.44 | eric_hill | @Qwell, ...as answered when the handset goes offhook. How odd. |
| 16:31.54 | pta200 | string length is about 280 characters |
| 16:31.59 | pta200 | gets cut off around 244 |
| 16:46.29 | *** join/#asterisk LND (n=Lee@fazer1-adsl.demon.co.uk) |
| 16:51.44 | *** join/#asterisk deeperror (n=deeperro@76.226.177.255) |
| 16:53.34 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:54.41 | deeperror | anyone know some good reading related to running multiple servers? I've got a server in EU and US and would like to link them up to improve call quality. Any ideas? |
| 16:55.12 | [TK]D-Fender | deeperror: and how does "linking them up" have any impact on call quality? |
| 16:56.45 | deeperror | there are users (sip extensions) that connect from EU and US to only the US server now. US server gets all inbound calls from EU and US |
| 16:57.30 | deeperror | we now have a server in EU setup with asterisk as well and the idea is to maybe get EU did's pointing to that server and US did's pointing to US server but allow agents to get call from both? |
| 16:59.22 | deeperror | so i think i need ser or something in between? |
| 16:59.34 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:59.38 | [TK]D-Fender | deeperror: "Agents"? |
| 17:00.36 | deeperror | yes the users are agents that work from home using a sip soft phone |
| 17:02.22 | deeperror | there are about 50 did's with numbers from all over EU and US....some of the EU calls have lower quality as they are routed from EU server to US then an agent in spain is also connected on their broadband to a server in US to accept the call...when it seems they could connect to an EU server and this would prevent a lot of extra hops across the pond |
| 17:03.18 | dominic1 | if I have a macro with 4 args, is is possible to execute it as follow Macro(mymacro,,,1,) |
| 17:03.19 | dominic1 | ?? |
| 17:03.45 | dominic1 | or are the arguments a must? |
| 17:04.26 | deeperror | just a guess maybe use "" for empty values if it isn't working? |
| 17:06.09 | [TK]D-Fender | dominic1: Should be fine with blank args |
| 17:06.26 | dominic1 | thank you very much! |
| 17:11.40 | LemensTS | http://lemensts.pastebin.com/m69f1c316 here is the full 'make' of the zaptel that im having problem with in debian |
| 17:12.11 | *** join/#asterisk aphlix (n=aphlix@office-mt.hous.unitewireless.com) |
| 17:12.25 | aphlix | can I ask a question |
| 17:13.02 | [TK]D-Fender | aphlix: You just did. If you're lucky we might even permit another ;) |
| 17:13.05 | [TK]D-Fender | ~ask |
| 17:13.06 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:13.07 | [TK]D-Fender | ^^^^^^^ |
| 17:13.53 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 17:15.39 | aphlix | ok cool. I have a trixbox and for some reason out of the blue my dtmf tones are not being sent out to the pstn right when I am in an IVR. for instance, if I call our tech support ivr using 7777, the menus work fine but if I call it using the outside phone #, the menus don't work |
| 17:15.58 | WhiteWolf | won't get much trix support here |
| 17:17.02 | Qwell | s/much/any/ |
| 17:17.21 | aphlix | great |
| 17:17.28 | Qwell | aphlix: though, is this a toll-free provider? |
| 17:17.36 | Qwell | rather, a toll-free VoIP provider |
| 17:17.46 | aphlix | well for what it is worth, it seems to be a zaptel problem but I don't know where to begin to troubleshoot it |
| 17:17.47 | deeperror | LemensTS, did you ln -s linux with linux-headers- |
| 17:18.02 | Qwell | aphlix: PRI? |
| 17:18.12 | aphlix | no it is through regular pstn lines out of a digium 8 port fxo card |
| 17:19.05 | [TK]D-Fender | aphlix: "dtmf tones are not being sent out to the pstn". So far not a TDM problem. What are you calling FROM? |
| 17:19.46 | aphlix | you mean as far as endpoint or pbx? |
| 17:20.14 | [TK]D-Fender | aphlix: Whats on the other end of this call as you state it is TO the PSTN. |
| 17:21.10 | aphlix | well as far as the test. I would call from one of my cisco phones to one of my roll over numbers on the same pbx kinda looping back thgrough the phone company if you will |
| 17:21.43 | aphlix | but if I call ANY other IVRs in the world, I cannot get tones to work to navigate the menus |
| 17:21.55 | Qwell | sounds like the Cisco is the problem |
| 17:21.59 | [TK]D-Fender | yup |
| 17:22.10 | Qwell | dtmfmode=rfc2833 |
| 17:22.15 | [TK]D-Fender | aphlix: Go prove the Cisco itself work 100% |
| 17:22.19 | aphlix | but thats the thing, I can dial 7777 and it works just fine |
| 17:22.30 | Qwell | what is 7777? |
| 17:22.36 | LemensTS | hello |
| 17:22.36 | aphlix | yeah thats what the dtmf is set to |
| 17:22.49 | [TK]D-Fender | aphlix: makes NO sense... |
| 17:22.56 | LemensTS | -- /usr/src/linux-headers-2.6.18-6 thier is no Makefile here, but /usr/src/linux-headers-2.6.18-6-686 has a symbolic link to it...i think that is why its not working...hmm...i do i make it reinstall the linux headers, maybe it messed up the first time. |
| 17:22.56 | Qwell | dialing an extension and dialing DTMF are different animals |
| 17:22.58 | aphlix | 7777 is the internal number to call lthe main IVR menu on OUR pbx |
| 17:23.10 | aphlix | ok |
| 17:23.29 | [TK]D-Fender | aphlix: is INBOUND DTMF from TDM a problem, or is it OUTBOUND transmission? |
| 17:23.41 | aphlix | so if I try it with a different phone and get the same result, what would be the next step |
| 17:23.42 | Qwell | call from a cellphone or something |
| 17:23.55 | Qwell | something not attached in any way to your system |
| 17:24.08 | aphlix | outbound. because I can call into my pbx from my cell phone and it works fine |
| 17:24.17 | Qwell | so it's the Cisco. :D |
| 17:24.33 | [TK]D-Fender | ^ |
| 17:24.56 | Qwell | aphlix: What version of Asterisk is this? |
| 17:25.18 | aphlix | 1.2.23 |
| 17:25.20 | Blackvel | hi qwell. got asterisk 1.2.x with bristuff 0.3.0 on centos 5.2 working. :) |
| 17:25.27 | Qwell | definitely the cisco |
| 17:25.31 | Qwell | Blackvel: and? |
| 17:25.53 | aphlix | ok I will try a softphone and see if the problem persists |
| 17:25.54 | [TK]D-Fender | hands Blackvel a Kewpie doll |
| 17:26.06 | aphlix | thank you guys for all of your help. I will be back with the results shortly |
| 17:26.51 | deeperror | LemensTS, try ln -s /usr/src/linux-headers-2.6.18-6-686 /usr/src/linux |
| 17:26.55 | Blackvel | was pain in the ass with all the media/cd-rom and nfs/http network install issues :) |
| 17:27.36 | Qwell | don't do that then.. |
| 17:27.59 | Qwell | deeperror: what? no.. |
| 17:28.25 | Blackvel | you forgot? this via box with rh9 and asterisk 1.0 was way too old ... |
| 17:28.47 | Qwell | Blackvel: yet you installed Asterisk 1.2 |
| 17:29.21 | [TK]D-Fender | "Procrastination : The art of keeping up with yesterday." |
| 17:29.59 | Blackvel | well, bristuff changes note about untested functionality with asterisk 1.4. this has to be a production system |
| 17:30.17 | russellb | bristuff has some baaaad code in it |
| 17:30.27 | Qwell | LemensTS: install linux-headers-`uname -r` |
| 17:30.27 | russellb | we get bug reports that were caused by that code on a regular basis it feels like |
| 17:30.30 | russellb | so ... have fun with that |
| 17:30.31 | russellb | heh |
| 17:30.36 | Blackvel | or even the whole package mostly untested with 1.4. so what would you guys do? :) |
| 17:30.50 | Qwell | not use it |
| 17:31.44 | Blackvel | russellb: well i am aware of it a bit hehe |
| 17:31.54 | Blackvel | you US guys have luck. you don't need BRI ISDN support with NT mode |
| 17:31.56 | aphlix | ok I tried a softphone and the same thing happened |
| 17:32.08 | Blackvel | maybe I can go for chan_isdn/misdn soon |
| 17:33.19 | LemensTS | deeperror: ok tried that |
| 17:33.40 | LemensTS | Qwell: working on that, the connection where this box is at is so slooow |
| 17:35.25 | LemensTS | qwell: said i already had the newest version |
| 17:35.42 | Qwell | with the -`uname -r` ? |
| 17:35.51 | *** join/#asterisk coyote` (n=ghuih@203.184.28.239) |
| 17:36.15 | coyote` | hi guys, i have an interesting but nooblike question |
| 17:36.18 | pta200 | length of a string argument to Dial application is limited to 255 characters in case anyone else was interested |
| 17:36.42 | LemensTS | qwell: yea |
| 17:37.22 | [TK]D-Fender | pta200: Someone here modded the source to extend it to something ludicrous. Should be googleable |
| 17:37.39 | deeperror | LemensTS, on my deb boxes I would create /usr/src/linux and /usr/src/linux-2.6 both pointing to the kernel-headers-2.6.18-6-686 |
| 17:37.52 | [TK]D-Fender | coyote`: Like Nike said "Just Do It" (Its easier to get forgiveness than permission) |
| 17:38.13 | coyote` | are you talking about editbox in dialogs being 255 characters as default maximum or are you talking about programming/scripting code |
| 17:38.18 | Blackvel | who is an expert of keeping the callerid from outside pstn? under what circumstances is this going to work with sip/pstn equipment (calling my mobile phone)? |
| 17:38.43 | pta200 | cool thanks |
| 17:39.01 | [TK]D-Fender | coyote`: about the actualy core's size limit. |
| 17:39.10 | [TK]D-Fender | coyote`: Nothing interface related. |
| 17:39.48 | coyote` | ok heres my question, what i want to do is set up a viop/landline tapi server. so i can call it using voip and have it use the voice modem to dial out.. |
| 17:39.57 | coyote` | and act as a server between the 2 |
| 17:40.28 | [TK]D-Fender | coyote`: Remote soft phone, sure. "Modem" = NO. You need a upported TDM device |
| 17:40.29 | coyote` | so my noob question is can i do that with asterisk and can you direct me to any addons that are premade for that function |
| 17:40.50 | coyote` | what is a tdm device btw |
| 17:41.01 | [TK]D-Fender | coyote`: Digium, Sangoma, etc. a SUPPORTED card. |
| 17:41.09 | [TK]D-Fender | coyote`: You can't use just any old POS modem. |
| 17:41.53 | coyote` | im pretty sure theres a way to do is with that NCH program actually, im gona go check it out |
| 17:42.13 | *** join/#asterisk akio (n=Akio@cpe-72-185-91-182.tampabay.res.rr.com) |
| 17:42.54 | [TK]D-Fender | coyote`: Not with * anyways.... good luck with all that.. |
| 17:43.00 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:43.35 | akio | i just installed debian etch and a few asterisk packages, where do i go for docs on setting it up? this is my first asterisk install |
| 17:43.50 | deeperror | ~book |
| 17:43.50 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:43.54 | akio | i need a web configuration tool or something |
| 17:44.30 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 17:44.40 | [TK]D-Fender | akio: Front-ends for * are not supported here. |
| 17:44.57 | akio | ok |
| 17:45.13 | akio | it isnt a graphical setup |
| 17:46.00 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 17:46.22 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 17:46.54 | akio | is it easy enough to just edit the conf files? |
| 17:47.22 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 17:47.39 | [TK]D-Fender | akio: Is for the rest of us. Go read... THE BOOK |
| 17:47.50 | akio | the oreilly book? |
| 17:48.05 | akio | oh its free! |
| 17:48.06 | [TK]D-Fender | akio: Yes |
| 17:48.20 | akio | got it |
| 17:52.11 | LemensTS | qwell: deeperror: i uninstalled linux-headers and reinstalled them. Now the Makefile is not emtpy in the linux-headers.***-6 file...im gonna try to 'make' zaptel again... |
| 17:52.31 | LemensTS | looks like it worked! that wasy the problem...as long as makeinstall doesnt fail. |
| 17:53.28 | *** join/#asterisk bkruse (n=bkruse@conference/asterisk/x-7e360e17277046c0) |
| 17:53.28 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:56.13 | *** join/#asterisk kc2tnk (n=fskrotzk@host198.textwise.com) |
| 17:57.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:01.28 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 18:02.22 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 18:02.42 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 18:03.07 | akio | i should probably use asterisk NOW |
| 18:04.31 | *** part/#asterisk akio (n=Akio@cpe-72-185-91-182.tampabay.res.rr.com) |
| 18:04.57 | shido6 | anyone know of any really good trance music? |
| 18:05.08 | shido6 | my mp3 collection needs help |
| 18:05.20 | shido6 | starting over |
| 18:06.29 | *** join/#asterisk benneton (n=DELL@89.111.211.113) |
| 18:06.37 | benneton | hello |
| 18:06.47 | benneton | how are you guys and girls |
| 18:06.49 | benneton | ? |
| 18:06.59 | benneton | i need help, so if someone... |
| 18:07.03 | benneton | tnx |
| 18:07.19 | benneton | i'm trying to connect 3 asterisk servers |
| 18:07.23 | benneton | via IAX |
| 18:07.37 | coyote` | help me! my medication has run out |
| 18:07.41 | coyote` | lmfao |
| 18:07.51 | _Raptor_ | coyote`: yeah i need some too |
| 18:07.53 | benneton | i'm using method for 2 asterisk servers |
| 18:07.55 | _Raptor_ | PLZ!!! |
| 18:08.06 | benneton | only works with 2 servers |
| 18:08.12 | benneton | tnx |
| 18:08.44 | [TK]D-Fender | benneton: What a glorious lack of detail. PASTEBIN is your friend. |
| 18:08.45 | [TK]D-Fender | ~pb |
| 18:08.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:08.47 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:09.13 | benneton | :) sorry... usualy i don't do irc |
| 18:10.49 | [TK]D-Fender | benneton: Show us what you're doing, and the errors you get and maybe we can help you. |
| 18:11.14 | benneton | I'm using pastebin |
| 18:11.16 | benneton | :) |
| 18:15.05 | *** join/#asterisk AlexGC (n=admin@201.127.67.48) |
| 18:15.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:15.51 | keith4 | [TK]D-Fender: a "glorious" lack... nice |
| 18:17.05 | [TK]D-Fender | keith4 : "Healp pleez!!! My calls don't werk. What can be problme?!?! :) " |
| 18:18.21 | keith4 | I would jump off of a building if I tried to answer as many of those as you do |
| 18:20.10 | [TK]D-Fender | keith4 : Don't go giving me any ideas now ;) |
| 18:20.53 | keith4 | make sure it's a short building, you'll be okay |
| 18:21.13 | *** join/#asterisk km- (n=pgrace@66.92.234.252) |
| 18:21.39 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 18:21.59 | km- | Hi! I'm looking for a hard desk phone that can support two simultaneous g729 call legs. Anyone know of one? (7940/7960 only have enough juice to do one g729 leg at a time, i.e., no three way calling) |
| 18:22.14 | [TK]D-Fender | km-: Polycom should be capable. |
| 18:22.40 | [TK]D-Fender | km-: I can confirm in a few hours |
| 18:23.10 | km- | you'd go polycom over cisco in pretty much any situation, right? |
| 18:23.34 | km- | we bought an IP 3000 for the conference room and it's a pretty nice phone |
| 18:23.42 | ManxPower | km-: anyone would if they had tried several different brands of phones |
| 18:23.57 | km- | manx: you're a polycom convert too then? |
| 18:24.15 | [TK]D-Fender | km-: For everything short of connecting to a CME, yes :) |
| 18:24.20 | ManxPower | km-: have been for years |
| 18:24.21 | km- | haha |
| 18:24.31 | *** join/#asterisk Segnale007 (n=Segnale0@host68-30-dynamic.19-79-r.retail.telecomitalia.it) |
| 18:24.32 | km- | What polycom desk units do you guys like? |
| 18:24.45 | km- | we'd be likely sending them out to our remote sales droids |
| 18:24.47 | ManxPower | We tried Cisco, grandstream, polycom, and a few uncommon brands like uniden |
| 18:24.57 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:25.01 | benneton | Hi again! |
| 18:25.06 | km- | you know, I saw in home depot one day a set of uniden phones that supported SIP and came with a coupon for vonage |
| 18:25.15 | benneton | need help on this. this is my pastebin: http://pastebin.com/m607828fc |
| 18:25.16 | km- | man am I kicking myself that I didnt buy those up before they disappeared |
| 18:25.20 | benneton | THANKS |
| 18:25.51 | brodiem | any particular reason the old polycom line still costs more than the new? |
| 18:26.44 | [TK]D-Fender | km-: Uniden phones are garbage. |
| 18:27.01 | ManxPower | km-: they would have been locked devices |
| 18:27.06 | km- | ah, nuts. |
| 18:27.09 | ManxPower | [TK]D-Fender: uniden was horrid |
| 18:27.14 | [TK]D-Fender | brodiem: because suppliers that picked them up did so at old cost. |
| 18:27.17 | km- | I didnt get a chance to play with one |
| 18:27.26 | [TK]D-Fender | km-: Consider yourself lucky. |
| 18:27.29 | km- | hehe |
| 18:27.43 | km- | So anyway, what models of polycoms do you guys prefer? |
| 18:27.45 | [TK]D-Fender | km-: I have 2 UIP-200's that I'd rather not. Very glad I never have to touch them |
| 18:27.46 | brodiem | [TK]D-Fender: was it you that used those uniden clamshell style analog? |
| 18:28.03 | [TK]D-Fender | km-: Depends on intended usage & deployment |
| 18:28.18 | [TK]D-Fender | brodthem too. those were UTTER garbage. |
| 18:28.45 | [TK]D-Fender | benneton: please pastebin all 3 servers setups in one pastebin. |
| 18:28.48 | km- | intended usage is for sales droids, remote deployment, normal desk usage |
| 18:28.54 | benneton | ok |
| 18:28.59 | benneton | please wait |
| 18:29.14 | [TK]D-Fender | km-: remote deployment : one per site, or multiple per site? |
| 18:29.14 | eric_hill | benneton: waiting... |
| 18:29.20 | km- | one per site |
| 18:29.26 | km- | these are work-from-home sales dudes |
| 18:29.47 | [TK]D-Fender | km-: This is a rare point where I sometimes suggest Linksys. they are very NAT friendly. |
| 18:30.08 | km- | do you think they've got the juice to do 2x simultaneous g729 transcodes? |
| 18:30.20 | [TK]D-Fender | km-: THAT... no I don't believe so. |
| 18:30.31 | [TK]D-Fender | km-: So then I'd go for the IP320 next |
| 18:30.32 | brodiem | +1 on LS for NATing |
| 18:31.18 | [TK]D-Fender | km-: Only device I know for a fact support it are the higher-end Mediatrix ATA's |
| 18:32.02 | km- | wow, those are expensive |
| 18:32.13 | km- | huh. it has a built-in adsl modem |
| 18:32.17 | km- | that's kinda swanky |
| 18:33.35 | brodiem | km-: iirc snom will do simultaneous g729, at least the 360 anyway |
| 18:33.44 | [TK]D-Fender | km-: 1 key funky thing some can do it to be a 100% transparent ATA. It'll take a WAN IP from your modem and pass it BACK to your router/other device while stripping off the SIP invisibly for its own use |
| 18:34.03 | km- | hah WHAT |
| 18:34.05 | km- | that's awesome |
| 18:34.13 | [TK]D-Fender | km-: Ask me in about 3 hours and I'll confirm IP 301/501 for you. |
| 18:34.23 | km- | d-fender: sure thing, I appreciate it. |
| 18:34.38 | [TK]D-Fender | km-: it is. their ATA's are really complex to configure, but very powerful. |
| 18:35.34 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 18:36.37 | Paige | are there any third party apps for 1.4.20.1 that will do parkinglots? |
| 18:36.47 | grandpapadot | Hi all. I'm working on a DUNDi configuration. I've got everything setup correctly and I can do loopback lookups (i.e., to a DUNDi peer defined to itself) but whenever I try to lookup via a peer I get a ENCREJ from the DUNDi debug output. I've regenerated the keys, etc, etc. This is with 1.4.21.2, any suggestions? |
| 18:36.55 | km- | err. why would you want a third-party app to do parkinglot? |
| 18:37.05 | Paige | becauae parkinglots are in 1.6 |
| 18:37.21 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 18:37.24 | km- | call parking has been in asterisk since before 1.0 |
| 18:37.52 | km- | am I missing something? parkinglots different than call parking? |
| 18:37.53 | Paige | and i cannot get 1.6 to compile on freebsd |
| 18:38.01 | [TK]D-Fender | Paige: Perhaps you could be more specific on the defining characteristics you're looking for. |
| 18:38.02 | Paige | multiple parking |
| 18:38.20 | [TK]D-Fender | Paige: and "multiple parking" isn't too helful |
| 18:38.24 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 18:38.46 | keith4 | what's different about call parking from 1.4 to 1.6? |
| 18:39.07 | Paige | i am running a multiple tenant asterisk server and I need parking for each tenant |
| 18:39.17 | heison | ~seen bkw |
| 18:39.22 | jbot | bkw <n=bkw@tor.lindesign.se> was last seen on IRC in channel #debian, 5d 8h 14m 52s ago, saying: 'Is it possible to findout the currently used font in a terminal?'. |
| 18:39.22 | km- | ohhh. |
| 18:39.22 | km- | yeah |
| 18:39.44 | km- | I'm reading the voipinfo thing, and I see the problem, no multihoming in parking lot |
| 18:40.06 | [TK]D-Fender | Paige: Go look up "supervaletparking" |
| 18:40.20 | [TK]D-Fender | Paige: I used its predecessor under 1.2 |
| 18:40.20 | Paige | ok thanks |
| 18:40.31 | grandpapadot | No dundi experts? |
| 18:40.59 | [TK]D-Fender | grandpapadot: Sorry, not a lot of Aussie's awake at the moment ;) |
| 18:41.06 | grandpapadot | lol |
| 18:41.37 | benneton | I'm back, [TK]D-Fender |
| 18:41.39 | km- | [tk]d-fender: polycoms do presentation over sip right? |
| 18:41.45 | benneton | this is my new pastebin |
| 18:41.48 | benneton | http://pastebin.com/m5ced9ab7 |
| 18:41.50 | [TK]D-Fender | km-: ? |
| 18:41.50 | km- | err. not presentation |
| 18:41.50 | Paige | but it is supported (multiple parking) in 1.6-b4? |
| 18:41.52 | km- | presence |
| 18:41.58 | [TK]D-Fender | km-: yes |
| 18:42.15 | [TK]D-Fender | Paige: go look. And B4 is old |
| 18:43.24 | Paige | sorry, i have beta9 but the addons is still at beta4 |
| 18:43.31 | Paige | and that I cannot compile |
| 18:43.34 | Paige | the addons |
| 18:43.43 | Paige | i need mysql support |
| 18:44.03 | *** join/#asterisk defsmac (n=andy@host81-149-11-136.in-addr.btopenworld.com) |
| 18:44.41 | defsmac | hi - I've got an a500 and BRI and get noise instead of ringing when I dial in |
| 18:46.29 | eric_hill | I just noticed that IAXtel.com is down for maintenance. Any idea when it will be back online? |
| 18:46.55 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 18:48.05 | Vec | Hi, I am having crazy problem with not being able to transfer calls when dialing multiple channels at once i.e. SIP/NUM&SIP/NUM any ideas ? |
| 18:48.31 | eric_hill | Vec: Does transferring a single call work? |
| 18:48.44 | Vec | eric_hill : yes perfectly |
| 18:49.18 | eric_hill | Vec: What do the relevant Dial lines look like in your dialplan? |
| 18:49.25 | Vec | eric_hill : 1 sec |
| 18:49.36 | eric_hill | Vec: ...or are you transferring from keypresses on the phone? |
| 18:49.51 | Vec | eric_hill : transfering from keypresses on the phone no Transfer in asterisk |
| 18:50.04 | defsmac | anyone here from sangoma ? |
| 18:50.24 | eric_hill | Vec: How do you transfer two simultaneous calls with a keypress? I thought that was only a single channel transfer... |
| 18:51.42 | mog | soon eric_hill |
| 18:52.06 | eric_hill | @mog: I got that... :) |
| 18:56.38 | anonymouz666 | anyone know if there's a default variable to store RTCP stats? I can't remember |
| 18:57.15 | km- | hmm, hint priority is pretty cool. |
| 18:58.23 | Paige | is there a svn for asterisk-addons that might be newer than beta4 for 1.6? |
| 18:58.56 | Paige | res_config_mysql.c will not compile on freebsd 7.0 |
| 18:59.18 | seanbright | svn.digium.com/svn/asterisk-addons/ |
| 19:00.37 | km- | any of you guys have cool receptionist lightboard apps |
| 19:02.05 | *** join/#asterisk AlexGC (n=admin@201.127.67.48) |
| 19:02.13 | Paige | sean do you have the url? |
| 19:02.19 | Paige | seanbright, |
| 19:04.22 | *** join/#asterisk defsmac (n=andy@host81-149-11-136.in-addr.btopenworld.com) |
| 19:06.29 | keith4 | ~fop |
| 19:06.29 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 19:06.42 | keith4 | km-: like the flash operator panel? |
| 19:07.36 | km- | yeah. |
| 19:09.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:09.58 | *** join/#asterisk los415 (n=los415@LAX-DHCP-64-201-109-228.race.com) |
| 19:10.29 | los415 | hey guys anyone familier with using smdi with asterisk |
| 19:11.59 | Paige | seanbright, do you have the url to the add-ons? |
| 19:13.18 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 19:16.16 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-179-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:20.56 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:21.10 | Paige | seanbright, do you have the url to the add-ons? |
| 19:21.26 | Paige | i have tried svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and i get an error |
| 19:21.27 | seanbright | http://svn.digium.com/view/asterisk-addons/branches/1.6.0/ |
| 19:21.53 | seanbright | svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.0/ |
| 19:22.38 | Paige | it says unrecognised url scheme |
| 19:22.55 | seanbright | Paige: then something is wrong with subversion on your machine. |
| 19:23.08 | *** join/#asterisk bird_of_Luck (i=melifaro@secured.by.ipfw.ru) |
| 19:23.57 | Paige | i am trying on a gentoo machine in a moment |
| 19:27.22 | *** join/#asterisk wamty (n=jcjc@89.17.109.127) |
| 19:27.25 | wamty | So asterisk requires libslang (1)... but I have other things using libslang2. What's the workaround when v1 and v2 conflict? |
| 19:28.40 | seanbright | asterisk doesn't require libslang |
| 19:28.59 | seanbright | you're welcome. |
| 19:29.00 | seanbright | :) |
| 19:29.20 | wamty | what should I do? |
| 19:29.32 | seanbright | nothing? |
| 19:30.07 | wamty | It must be a dependency of something else that asterisk requires, then... because I tried installing asterisk and it choked when libslang was being installed. |
| 19:30.22 | defsmac | what dertermine the ringing tone played that incoming callers hear ? (on a ISDN2 BRI) |
| 19:30.35 | seanbright | wamty: asterisk doesn't try to install libslang. |
| 19:30.57 | seanbright | wamty: how are you installing? |
| 19:31.38 | wamty | packages |
| 19:31.43 | seanbright | ahh |
| 19:31.46 | wamty | It must be a dependency of something else asterisk requires. ... deja vu? |
| 19:32.26 | wamty | I'll have to go back and see what is requiring libslang. |
| 19:32.34 | seanbright | most likely libnewt |
| 19:32.45 | seanbright | which is used by menuselect if it's available |
| 19:32.56 | seanbright | but if it's not, it falls back to (n)curses |
| 19:33.40 | wamty | so newt wants libslang, but I have libslang2 for other things, and those two conflict. |
| 19:33.49 | wamty | can this be correct? |
| 19:33.52 | seanbright | wamty: sure |
| 19:33.55 | seanbright | wamty: install from source |
| 19:33.59 | seanbright | problem solved |
| 19:34.22 | deeperror | haha |
| 19:34.32 | wamty | asterisk, newt, or all of it? |
| 19:34.36 | seanbright | asterisk |
| 19:34.42 | seanbright | newt isn't required |
| 19:34.53 | seanbright | but whatever package you are using wants it to be |
| 19:35.09 | wamty | interesting |
| 19:36.48 | keith4 | what distro? |
| 19:36.53 | *** part/#asterisk defsmac (n=andy@host81-149-11-136.in-addr.btopenworld.com) |
| 19:37.08 | wamty | slack |
| 19:37.18 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 19:37.35 | keith4 | what version of asterisk? |
| 19:40.15 | wamty | I'm trying to develop it on ArchLinux, but it will ultimately be running on that other *nix I like to use. |
| 19:40.40 | wamty | seanbright: If I use the pkgbuild on Arch, newt isn't a dep. |
| 19:41.13 | wamty | If I go to compile it on the other platform, I get a problem: asterisk-1.4.20.1 depends on shared library: speex.1 - found |
| 19:41.19 | wamty | sterisk-1.4.20.1 depends on shared library: newt.51 - not found |
| 19:42.33 | wamty | ? |
| 19:43.11 | keith4 | fwiw, the debian asterisk package depends on libnewt, which depends on libslang2 |
| 19:43.35 | seanbright | wamty: you are speaking greek right now. i don't use packages because i have a penis and testicles. |
| 19:43.46 | seanbright | wamty: i build from source like nature intended. |
| 19:43.51 | seanbright | good luck and god speed. |
| 19:43.52 | *** join/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 19:45.03 | wamty | are you Sean connery? |
| 19:45.28 | AiT | I'm getting error "Signal 11" every 20 mins or so via email. This is not a new install. Google search tells me installation issues, but asterisk has been installed for over 5 months, any ideas? |
| 19:46.02 | AiT | I've done updates, but I don't know how to troubleshoot it... |
| 19:46.49 | *** join/#asterisk outtolunc (n=me@adsl-76-211-231-133.dsl.pltn13.sbcglobal.net) |
| 19:48.32 | [TK]D-Fender | AiT: Could you be a little more specific than "updates"? And perhaps yous hould watch what goes on at CLI. |
| 19:50.46 | AiT | Disconnected from Asterisk server |
| 19:50.46 | AiT | You have mail in /var/spool/mail/root |
| 19:50.56 | AiT | That's the only think it shows from the CLI |
| 19:51.16 | AiT | I've done about 30-40 module updates a few weeks back |
| 19:51.30 | [TK]D-Fender | AiT: Stop running * as a service and run it direct. |
| 19:51.44 | AiT | [TK]D-Fender, how? |
| 19:51.45 | [TK]D-Fender | AiT: and what "module" updates? |
| 19:51.53 | [TK]D-Fender | AiT: "asterisk -gvvvvvvc" |
| 19:51.57 | AiT | [TK]D-Fender, FreePBX |
| 19:52.07 | LemensTS | ... |
| 19:52.13 | [TK]D-Fender | AiT: so basically nothing that concerns us |
| 19:52.21 | AiT | Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. |
| 19:52.28 | [TK]D-Fender | AiT: So go run it direct and see what happens |
| 19:52.32 | AiT | :( Guys over at freepbx are pretty lame |
| 19:52.35 | [TK]D-Fender | AiT: well DUH, stop * first |
| 19:53.11 | AiT | [TK]D-Fender, FreePBX has all the routing, etc |
| 19:53.21 | [TK]D-Fender | AiT: Meaningless. |
| 19:53.35 | [TK]D-Fender | AiT: Go start it manually and watch |
| 19:53.47 | AiT | k |
| 19:56.52 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 20:00.36 | AiT | [TK]D-Fender, so killing FreePBX and enabling from CLI will reload all my settings? (3 active calls right now) |
| 20:01.00 | [TK]D-Fender | AiT: FreePBX isn't a PROCESS. |
| 20:01.36 | [TK]D-Fender | AiT: and "reload settings" does not apply. Stopping * will kill your calls. You'll have to write that off if you want to see whats going on |
| 20:05.34 | *** join/#asterisk trentcreek (n=trentcre@red1.cs.panam.edu) |
| 20:05.50 | AiT | [TK]D-Fender....ok, we're rolling ;) |
| 20:06.05 | trentcreek | keep them doggies tollin |
| 20:10.09 | *** join/#asterisk catenarellc (n=johanmar@41.246.140.19) |
| 20:10.26 | eric_hill | Good resources for decoding ISDN D-Channel Facility signaling hex? |
| 20:10.43 | *** join/#asterisk franhp (n=franhp@75.Red-88-24-201.staticIP.rima-tde.net) |
| 20:11.31 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 20:15.46 | *** join/#asterisk hijacked (n=argh@c-71-205-135-177.hsd1.mi.comcast.net) |
| 20:19.01 | *** join/#asterisk Yourname`` (i=yourname@unaffiliated/yourname/x-837320) |
| 20:19.25 | trentcreek | hellllllllllllllp |
| 20:19.27 | trentcreek | hellllllllpo |
| 20:19.32 | trentcreek | fender geetar |
| 20:19.35 | trentcreek | http://pastebin.com/d54c24bda |
| 20:19.36 | Yourname`` | Hi, in AMI, if a call comes in, I want to be able to see the calleridname along with the callleridnum. However, I only see calleridnum, how can I see the name too? |
| 20:19.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:20.13 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 20:20.37 | LemensTS | trentcreek: did you try what it said: ./start_asterisk start |
| 20:20.46 | [TK]D-Fender | trentcreek: ... and? |
| 20:20.51 | franhp | trentcreek, you should install pidof, which I don't know the package |
| 20:21.12 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 20:21.16 | *** part/#asterisk catenarellc (n=johanmar@41.246.140.19) |
| 20:22.10 | Qwell | franhp: sysvinit-utils |
| 20:22.25 | franhp | Qwell's answer > trentcreek |
| 20:23.42 | davevg-btwtech | trentcreek, try this: export PATH=$PATH:/sbin:/usr/sbin |
| 20:24.02 | seanbright | did someone already give him the ol' ~freepbx bit? |
| 20:24.07 | Qwell | ~freepbx |
| 20:24.08 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:24.13 | seanbright | yes... that |
| 20:24.43 | seanbright | you know what sucks? |
| 20:24.54 | seanbright | freepbx |
| 20:25.46 | AiT | [TK]D-Fender, failed :( |
| 20:26.03 | AiT | Segmentation fault (core dumped) |
| 20:26.03 | AiT | You have mail in /var/spool/mail/root |
| 20:26.09 | [TK]D-Fender | AiT: Could you be a little MORE vague? |
| 20:26.32 | seanbright | [TK]D-Fender: i'm not sure how |
| 20:26.32 | AiT | [TK]D-Fender, hehe, that's all it says in the CLI |
| 20:26.49 | [TK]D-Fender | AiT: Out of the blue? No PASTEBIN to show us? |
| 20:27.09 | AiT | sure...one sec |
| 20:27.28 | [TK]D-Fender | seanbright: </sarcasm> |
| 20:27.40 | seanbright | knows |
| 20:28.01 | keith4 | refers to earlier comment about jumping off of building |
| 20:28.13 | Qwell | ~push keith4 |
| 20:28.13 | jbot | ACTION enlists ez` to push keith4 over the edge of a very high cliff |
| 20:28.22 | [TK]D-Fender | lol |
| 20:28.44 | [TK]D-Fender | Qwell: Random nomination FTW! |
| 20:28.56 | AiT | [TK]D-Fender, didn't have the CLI logging...I guess a few minutes more before it crashes again... |
| 20:29.08 | [TK]D-Fender | heading home, BBIAB |
| 20:30.03 | trentcreek | LemensTS: yes I tried ./start_asterisk start |
| 20:30.30 | trentcreek | and I got: |
| 20:30.31 | trentcreek | ./start_asterisk start |
| 20:30.34 | Paige | has anyone managed to compile asterisk-addons-1.6-b4 on freebsd? |
| 20:30.48 | trentcreek | STARTING ASTERISK |
| 20:30.48 | trentcreek | ./start_asterisk: line 36: pidof: command not found |
| 20:30.48 | trentcreek | Asterisk is already running. /usr/sbin/safe_asterisk will exit now. |
| 20:31.08 | seanbright | ... |
| 20:31.17 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 20:31.17 | seanbright | < franhp> trentcreek, you should install pidof, which I don't know the package |
| 20:31.24 | seanbright | <@Qwell> franhp: sysvinit-utils |
| 20:31.35 | trentcreek | okay..searching now to see if it is in yum |
| 20:32.12 | trentcreek | nope |
| 20:32.17 | Qwell | they shouldn't be using pidof anyways |
| 20:32.49 | seanbright | Qwell: you know what sucks? |
| 20:32.50 | trentcreek | sounds like I have to compile it |
| 20:33.10 | Qwell | no, you don't compile pidof |
| 20:33.36 | trentcreek | change my path |
| 20:33.46 | seanbright | trentcreek: /sbin/pidof |
| 20:33.47 | seanbright | ? |
| 20:33.51 | trentcreek | export PATH=$PATH:/sbin:/usr/sbin |
| 20:34.26 | Paige | i keep running into problems with asterisk.h not being found regardless of where i copy the header file to |
| 20:34.26 | trentcreek | Ijust entered that..still same thing |
| 20:34.29 | Yourname`` | Somebody please check out my AMI question where I want to find out the CALLERID(name) |
| 20:34.33 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:34.44 | keith4 | might just be called 'sysvinit' in yum |
| 20:34.49 | keith4 | because yum is stupid |
| 20:35.14 | seanbright | trentcreek: try this: ls -al /sbin/pidof |
| 20:35.36 | keith4 | fwiw, centos5 has pidof at /sbin/pidof |
| 20:36.10 | trentcreek | lrwxrwxrwx 1 root root 8 Dec 11 2007 /sbin/pidof -> killall5 |
| 20:36.10 | seanbright | and there it is |
| 20:36.10 | franhp | Yourname``, does your line really support calleridname ? |
| 20:36.10 | seanbright | and this: |
| 20:36.12 | seanbright | echo $PATH |
| 20:36.22 | keith4 | someone didn't 'su -' |
| 20:36.26 | seanbright | apparently |
| 20:36.32 | trentcreek | oops.. |
| 20:36.39 | franhp | XD |
| 20:36.58 | keith4 | which i forget to do all the time, because debian doesn't work that way |
| 20:37.04 | *** join/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 20:37.14 | Yourname`` | franhp: Yes, it does. But I'm looking to find it in the AMI. |
| 20:37.23 | Paige | any help is greatly appreciated |
| 20:37.31 | trentcreek | OKAY GUYS..That worked su -..thanks |
| 20:37.45 | seanbright | Paige: why are you copying asterisk.h anywhere? |
| 20:38.25 | Paige | because gmake is stupid and cannot find asterisk.h in /usr/local/include/asterisk |
| 20:38.34 | seanbright | Paige: gmake is far from stupid. |
| 20:39.06 | seanbright | Paige: how are you calling ./configure in the asterisk-addons directory? |
| 20:39.07 | silvertip257 | Last month I was in here asking about asterisk and a WRT54G router. I'm trying to set up outbound/inbound to an ITSP provider (CallWithUs). I have some questions about their example config at this link (about half-page) > http://www.callwithus.com/configuration Thanks |
| 20:39.32 | Paige | ./configure --with-mysql=/usr/local |
| 20:39.57 | deeperror | silvertip257, and those questions? |
| 20:39.59 | seanbright | ./configure --with-mysql=/usr/local --with-asterisk=/usr/local |
| 20:40.31 | x86 | http://lh4.ggpht.com/abramsv/SC0sXQM9yII/AAAAAAAAQ4k/sazkB7mBxDk/s1600-h/znaki_06.jpg |
| 20:40.40 | x86 | that sign should come with ABE |
| 20:41.10 | Yourname`` | Looks like no one knows the answer to the AMI question. So, I'm going to ask my next question. I'm cascading queues, and members of both the queues are logging in via AgentLogin(). So, there's no way they can see callerids. Is there any solution where lets say queue1 is getting a cascaded call from queue2 and the meember who takes that calls gets to know that it's a cascaded call from queue2... |
| 20:41.11 | silvertip257 | deeperror: well I'm trying to use SIP for my ITSP connection/registration. I want to know why they have username:password@_username_ ... does the username after the @ send it to the sip entry named username? I find their example config confusing in that manner. |
| 20:41.12 | Yourname`` | ...and that he's NOT answering a call from queue1? (his 'home' queue?) |
| 20:41.23 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 20:41.48 | silvertip257 | deeperror: I can post my current configs if you would like |
| 20:41.58 | Paige | http://pastebin.ca/1092819 |
| 20:42.24 | seanbright | Paige: what version of asterisk is installed? |
| 20:42.31 | deeperror | i think they are referencing the friend that is being used below |
| 20:42.37 | Paige | 1.6-beta9 |
| 20:43.04 | seanbright | Paige: well then the headers that are installed aren't up-to-date |
| 20:43.14 | seanbright | Paige: looks like the 1.4 headers are installed |
| 20:43.27 | keith4 | silvertip257: deeperror is right. the [username] friend has a host defined in it |
| 20:43.28 | deeperror | silvertip257, looks like they are registering with the friend that they create below which defines the host=sip.callwithus.com never seen it done that way |
| 20:43.33 | Paige | ok. trying to build again |
| 20:43.36 | silvertip257 | deeperror: yes, I am trying to figure out how to make MY config easier to understand than the example ... |
| 20:43.57 | keith4 | you can change that to be similar to how the iax one is, but with sip.callwithus instead of iax. |
| 20:44.06 | silvertip257 | deeperror: would you look at my configs (exten + sip) b/c I'm having trouble with inbound and outbound |
| 20:44.20 | deeperror | silvertip257, you could just put the host in register if you like....i think they did that so you can change host in one spot and not have to change it in two places |
| 20:44.55 | silvertip257 | deeperror: I have it registered regularly - based on the Asterisk TFoT book and other online guides |
| 20:44.57 | seanbright | Yourname``: i think there is a way to play audio to the agent before they take the call |
| 20:45.08 | deeperror | silvertip257, pastebin? |
| 20:45.14 | silvertip257 | deeperror: working on it |
| 20:45.40 | Yourname`` | seanbright: Do tell, my friend! |
| 20:45.59 | seanbright | Yourname``: i just did |
| 20:46.11 | Yourname`` | seanbright: I mean how.. lol |
| 20:46.17 | seanbright | Yourname``: queues.conf |
| 20:46.23 | Yourname`` | announce |
| 20:46.32 | seanbright | Yourname``: announce = go-tuck-yourself-in |
| 20:46.48 | seanbright | Yourname``: where "go-tuck-yourself-in" is the sound you want played to the agent |
| 20:46.54 | seanbright | sound file* |
| 20:47.17 | anonymouz666 | No more connections allowed |
| 20:47.17 | anonymouz666 | *CLI> |
| 20:47.21 | anonymouz666 | lol |
| 20:47.30 | Paige | sean thanks, i got it to build |
| 20:47.30 | anonymouz666 | this beats all the records |
| 20:47.35 | seanbright | Paige: super duper |
| 20:47.37 | Yourname`` | seanbright: I was lookn at it last night, but thought it might just be overkill. :S But I guess this is the best way out if you're saying it |
| 20:47.53 | Paige | ls |
| 20:47.55 | Paige | oops |
| 20:48.01 | seanbright | Yourname``: i'm not sure of the best way, that was just a stab in the dark |
| 20:48.20 | Yourname`` | seanbright: Ok. Also, do you know if there's a way for me to find out the calleridname of a call via AMI? |
| 20:48.30 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 20:48.43 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:48.44 | seanbright | Yourname``: as for the AMI question, why not use the 'Getvar' action with "${CALLERID(name)}" as the variable to get? |
| 20:48.57 | silvertip257 | deeperror: here are my exten + sip configs > http://pastebin.com/d1436531c |
| 20:49.06 | Paige | seanbright, thanks again |
| 20:49.11 | seanbright | Paige: yup |
| 20:49.16 | Paige | it was stale header files |
| 20:49.32 | seanbright | Paige: yeah, i figured. |
| 20:49.40 | seanbright | Paige: you should apologize to gmake, though. |
| 20:49.49 | seanbright | Paige: he's very sensitive |
| 20:50.07 | Paige | haha |
| 20:50.15 | x86 | http://lh4.ggpht.com/abramsv/SBrALSdo3XI/AAAAAAAAP_4/tlZ5BgP9QI4/s1600-h/2007-09-04%2014-19-56.jpg |
| 20:50.17 | franhp | which would be the best way for a dialplan to ask the user for a password? We want the user to input a secret code and thus be able access the special number of the boss |
| 20:50.26 | seanbright | franhp: Read() |
| 20:50.33 | seanbright | franhp: core show application Read |
| 20:50.36 | Qwell | Authenticate? |
| 20:50.49 | seanbright | franhp: Authenticate() |
| 20:50.51 | seanbright | (heh) |
| 20:50.54 | franhp | thanks all! |
| 20:51.06 | franhp | (I knew WaitExten wasn't good enough :P ) |
| 20:51.09 | brodiem | important boss |
| 20:51.15 | [TK]D-Fender | franhp: "core show application read" |
| 20:51.52 | seanbright | [TK]D-Fender: that's what i said, man. |
| 20:52.07 | [TK]D-Fender | seanbright: I noticed. |
| 20:56.18 | x86 | http://lh5.ggpht.com/abramsv/SJZxt-fXwFI/AAAAAAAAYAM/pHK-r95llPc/s1600-h/59637_25998_5513.jpg |
| 20:56.21 | x86 | damn, i missed it |
| 20:56.37 | Yourname`` | seanbright: Thanks! |
| 20:56.46 | seanbright | Yourname``: that do the trick? |
| 20:56.57 | Paige | what is a quick and easy way to rename all foo.conf-dist files to foo.conf? |
| 20:57.07 | Paige | I forget the sed trick |
| 20:57.28 | brodiem | Paige: for i in `ls *-dist`; do newname=`echo $i | cut -f1 -d "-"`; mv $i $newname; done |
| 20:57.31 | brodiem | thats one way.. |
| 20:57.36 | Paige | thanks |
| 20:57.54 | Yourname`` | seanbright: The GetVar, yeah! :) |
| 20:58.01 | seanbright | Yourname``: super duper |
| 20:58.02 | Paige | perfect, worked :) |
| 20:58.09 | brodiem | of course;) |
| 20:58.26 | seanbright | i run those types of things twice |
| 20:58.32 | seanbright | ... echo mv $i $newname |
| 20:58.36 | seanbright | ... mv $i $newname |
| 20:58.37 | seanbright | heh |
| 21:01.48 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 21:05.17 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-73-250.32-151.iol.it) |
| 21:05.41 | JerJer | anyone know how can one get the remote IP address of a SIP peer using channel variables ? |
| 21:06.03 | *** join/#asterisk davidstrauss (n=straussd@wikimedia/davidstrauss) |
| 21:06.20 | dr_gogeta86 | hi to all |
| 21:06.24 | [TK]D-Fender | JerJer: "core show functions like SIP" |
| 21:06.57 | JerJer | ok so SIPPEER |
| 21:07.39 | x86 | http://lh4.ggpht.com/abramsv/R8-0W3H56AI/AAAAAAAAK8M/3KC0vY0RGtY/s1600-h/43_pics.jpg |
| 21:07.45 | x86 | shakes his head |
| 21:08.05 | dr_gogeta86 | hi have some problems with rtp |
| 21:08.09 | dr_gogeta86 | who can help me |
| 21:08.26 | JerJer | dr_gogeta86: help yourself by asking a specific question |
| 21:08.49 | dr_gogeta86 | i have asterisk behind a firewall |
| 21:09.09 | dr_gogeta86 | i have private ip in backend |
| 21:09.12 | [TK]D-Fender | dr_gogeta86: Read up : |
| 21:09.14 | [TK]D-Fender | ~sipnat |
| 21:09.15 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:09.16 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:09.24 | davidstrauss | I'm setting up a new AA50 appliance. I've provisioned two phones with extensions, but whenever I dial the other extension, I get a busy signal. I haven't hooked the system up to any analog or remote network services, and I'm using Polycom SountPoint IP 320 handsets. What am I doing wrong? |
| 21:09.27 | dr_gogeta86 | but it doesend work |
| 21:10.11 | [TK]D-Fender | dr_gogeta86: Read the guide |
| 21:10.16 | russellb | davidstrauss: you may want to contact digium tech support |
| 21:10.21 | JerJer | [TK]D-Fender: thanks - i knew there was an easy way |
| 21:10.42 | dr_gogeta86 | [TK]D-Fender, is possible to open rtp port dinamically |
| 21:10.55 | [TK]D-Fender | dr_gogeta86: huh? |
| 21:11.04 | *** join/#asterisk antonyo14 (n=a@206.135.21.162) |
| 21:11.16 | antonyo14 | can anyone help me with a vmx issue in that it wont let me use numbers for options, only * and 0??? |
| 21:11.27 | dr_gogeta86 | I dont'want to keep rtp ports open |
| 21:11.39 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 21:11.40 | dr_gogeta86 | but i want open it when neded |
| 21:12.48 | [TK]D-Fender | dr_gogeta86: No choice. Just do it. |
| 21:13.14 | dr_gogeta86 | any conntrack |
| 21:13.18 | dr_gogeta86 | for iptables |
| 21:13.34 | trentcreek | What is a quick way to get my "asterisk" user account to get access to all the asterisk files and folders? |
| 21:13.50 | [TK]D-Fender | trentcreek: "man chown" |
| 21:14.01 | [TK]D-Fender | antonyo14: ... "vmx"? |
| 21:14.12 | antonyo14 | voicemail |
| 21:14.28 | [TK]D-Fender | antonyo14: What options won't it let you take? |
| 21:14.38 | antonyo14 | it won't let me use 1-9 |
| 21:14.50 | antonyo14 | only * and 0 |
| 21:14.54 | trentcreek | groovy |
| 21:15.26 | *** part/#asterisk benneton (n=DELL@89.111.211.113) |
| 21:15.43 | [TK]D-Fender | antonyo14: What options is it presenting to you that you can't dial? |
| 21:17.27 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) [NETSPLIT VICTIM] |
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| 21:17.28 | antonyo14 | I want to make 9 go back to an ivr |
| 21:17.51 | antonyo14 | and it works if i use * but it is being used to go to voicemailmain |
| 21:18.24 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 21:18.40 | [TK]D-Fender | antonyo14: First you said you have VM problem, and now you say you ave IVR problem. Please make up your mind |
| 21:18.49 | *** join/#asterisk JSnoz (i=aviat200@arkana.iiens.net) |
| 21:18.49 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 21:18.56 | trentcreek | BiPolar |
| 21:19.15 | [TK]D-Fender | trentcreek: One-track mind. |
| 21:21.37 | antonyo14 | TK: I want to be able to exit Voicemail by hitting 9 |
| 21:21.44 | BBHoss | anyone know how many flops it takes for one transcoded g729 call (say ulaw to g729)? |
| 21:22.06 | antonyo14 | and return to an IVR |
| 21:22.08 | [TK]D-Fender | antonyo14: You can only do that by changing the SOURCE CODE of Voicemail itself |
| 21:22.33 | *** join/#asterisk nhuisman_work (n=nhuisman@dhcp70.IfA.Hawaii.Edu) |
| 21:22.48 | [TK]D-Fender | antonyo14: No other way. Voicemail offers * and ) for exit options. use them or not, but that what if offered by that application untill you go and recode it yourself |
| 21:23.01 | nhuisman_work | In the last week I've had two users complaining of hundreds of blank voicemail, any idea what might cause this? |
| 21:23.39 | [TK]D-Fender | nhuisman_work: Mass hallucinations. |
| 21:23.50 | nhuisman_work | [TK]D-Fender: hehe |
| 21:24.54 | mvanbaak | nhuisman_work: check the minsecs setting in /etc/asterisk/voicemail.conf |
| 21:25.07 | nhuisman_work | i have it set to 3 |
| 21:25.09 | *** join/#asterisk mgdm (n=michael@serenity.mgdm.net) |
| 21:25.20 | mvanbaak | then check your voicemail storage server |
| 21:25.27 | *** join/#asterisk rajiv_ (n=rajiv@gentoo/developer/rajiv) |
| 21:25.27 | mvanbaak | make sure there's no filesystem corruption |
| 21:25.31 | mvanbaak | I had that once |
| 21:25.36 | nhuisman_work | hmm |
| 21:25.42 | nhuisman_work | in this one vm |
| 21:25.46 | mvanbaak | resulted in like 20.000 vm's a minute here |
| 21:25.47 | nhuisman_work | i see 83 files |
| 21:25.55 | nhuisman_work | and 83 text files |
| 21:26.06 | mvanbaak | then that user has 83 voicemails |
| 21:26.15 | nhuisman_work | and the wavs are not all the same size |
| 21:26.20 | nhuisman_work | like 200bytes - 700 bytes |
| 21:26.46 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:26.47 | mvanbaak | short messages |
| 21:26.47 | [TK]D-Fender | *b00m* |
| 21:27.16 | mvanbaak | netsplit ! |
| 21:27.16 | mvanbaak | hey puzzled |
| 21:27.16 | AiT | [TK]D-Fender, hasn't crashed again...that's odd! |
| 21:27.28 | AiT | [TK]D-Fender, only other thing I did was put the hold music on the default wav from MP3... |
| 21:27.45 | seanbright | heh |
| 21:27.47 | mvanbaak | AiT: that's always a good idea |
| 21:27.52 | seanbright | might have been the problem |
| 21:28.13 | mvanbaak | yup |
| 21:28.24 | AiT | seanbright, CPU duty cycle then? Cuz top never shows more than 20% |
| 21:28.25 | mvanbaak | mp3 moh is giving me trouble in like 40% of the installs |
| 21:28.25 | [TK]D-Fender | AiT: When you want to test things.... stop moving the &^#$# target. |
| 21:28.43 | nhuisman_work | is there an easy way to really delete emails from the menus? |
| 21:28.51 | nhuisman_work | it always seems to undelete and delete the same message |
| 21:28.54 | seanbright | AiT: it may have just not liked the mp3s that were playing |
| 21:28.55 | nhuisman_work | if you just keep pressing 7 |
| 21:29.00 | [TK]D-Fender | nhuisman_work: 7676767676767676767676767676767676767676767676767676 |
| 21:29.12 | mvanbaak | what fender said |
| 21:29.14 | nhuisman_work | I thought I was doing that once, and it just kept staying on the same message |
| 21:29.17 | nhuisman_work | maybe i was doing 76 |
| 21:29.19 | nhuisman_work | err 75757577575 |
| 21:29.21 | puzzled | hi mvanbaak |
| 21:29.35 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 21:29.36 | mvanbaak | puzzled: you have that X100p setup ? |
| 21:29.47 | gaetronik | hi |
| 21:30.00 | puzzled | mvanbaak: uhm, I'm afraid I haven't had the time yet :) |
| 21:30.12 | mvanbaak | lol |
| 21:30.16 | gaetronik | is there any way to have stats on logintime pausetime exepting parsing queu_log? |
| 21:30.17 | mvanbaak | sounds like my life |
| 21:30.21 | puzzled | heh |
| 21:30.47 | mvanbaak | like, I have this skinny_realtime branch waiting for me for months now |
| 21:30.49 | [TK]D-Fender | gaetronik: Not unless you invent it in the dialplan |
| 21:31.18 | gaetronik | for logintime there is the good fiel in queue_log |
| 21:31.21 | gaetronik | put faor puse no |
| 21:31.32 | gaetronik | and i don't see how to put it in dialplan? |
| 21:31.48 | gaetronik | but for pause no* |
| 21:31.48 | puzzled | mvanbaak: how's skinny doing in trunk? is it on par feature-wise to chan_sccp? |
| 21:31.50 | mvanbaak | gaetronik: use func_odbc |
| 21:31.53 | gaetronik | fucking fingers |
| 21:31.54 | [TK]D-Fender | gaetronik: How do they get paused? |
| 21:32.08 | gaetronik | [TK]D-Fender, ami |
| 21:32.17 | [TK]D-Fender | [17:31]<gaetronik>fucking fingers <- this isn't #lesbians |
| 21:32.21 | mvanbaak | puzzled: it is if you compare it with latest chan_sccp I tested |
| 21:32.23 | JSnoz | is that possible to use s/mime cryptography with asterisk 1.4.18.* ? |
| 21:32.30 | gaetronik | [TK]D-Fender, lol |
| 21:32.30 | mvanbaak | puzzled: the chan_sccp on berlios that is |
| 21:32.37 | [TK]D-Fender | gaetronik: If you are setting it via AMI then whatever you're doing that from should do the work |
| 21:32.43 | mvanbaak | with the two patches from sgofferje to prevent segfaults |
| 21:32.56 | puzzled | mvanbaak: that's the same one I used |
| 21:32.56 | jblack | I've had it with this rhino card. once or twice a day, it congests all calls until rebooted. |
| 21:33.09 | [TK]D-Fender | JSnoz: To encrypt what exactly? |
| 21:33.21 | gaetronik | [TK]D-Fender, the daemon use the pause action |
| 21:33.39 | gaetronik | better use a Dial Local to se odbc? |
| 21:33.41 | [TK]D-Fender | gaetronik: "AMI", "daemon" WTH? |
| 21:33.50 | gaetronik | asterisk manager interface |
| 21:33.52 | gaetronik | AMI |
| 21:33.58 | mvanbaak | puzzled: then it's on par |
| 21:34.06 | puzzled | cool |
| 21:34.09 | mvanbaak | maybe two annoying issues: |
| 21:34.10 | gaetronik | i have a little daemon which put in pause agents |
| 21:34.12 | [TK]D-Fender | gaetronik: If you are running a script that pauses them, log it YOURSELF |
| 21:34.12 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:34.34 | gaetronik | [TK]D-Fender, better not add a line at this software |
| 21:34.51 | JSnoz | [TK]D-Fender: SIP headers, encrypted or signed, and SDP payloads at least |
| 21:35.05 | JSnoz | and why not rtp |
| 21:35.21 | iCEBrkr | JSnoz: Paranoid much? |
| 21:35.21 | JSnoz | I haven't found any docs about using srtp with asterisk |
| 21:35.28 | [TK]D-Fender | JSnoz: You've got the source like the rest of us... |
| 21:35.36 | mvanbaak | line buttons on the multiline phones dont have the linelabel as name on the lcd, but only the linenumber |
| 21:35.40 | JSnoz | not at all cesar_CR :), just working in security :) |
| 21:35.45 | [TK]D-Fender | JSnoz: And thats because srtp isn't supported |
| 21:35.51 | iCEBrkr | I just think it's funny that everyone is concerned about their calls being encrypted.. Meanwhile, their emails contain clear-text passwords.... |
| 21:36.04 | puzzled | mvanbaak: think I remember that one from a while back |
| 21:36.14 | mvanbaak | called number recording in the phones directory are wrong |
| 21:36.24 | JSnoz | [TK]D-Fender: ok :), but could you give me a hint ? |
| 21:36.31 | cesar_CR | JSnoz, ?? |
| 21:36.38 | JSnoz | where sould I start looking to get more information about s/mime in SIP messages ? |
| 21:36.42 | [TK]D-Fender | JSnoz: "chan_sip.c" <- |
| 21:36.57 | mvanbaak | it will put the last used exten in the dialplan there (I use a macro to dialout, so it records 's') |
| 21:37.06 | JSnoz | cesar_CR: ?? |
| 21:37.08 | puzzled | mvanbaak: chan_sccp used to have an issue too but they fixed it. is that code of any help? |
| 21:37.21 | mvanbaak | cant use it |
| 21:37.27 | mvanbaak | because it's not disclaimed |
| 21:37.28 | JSnoz | thanks [TK]D-Fender :) |
| 21:37.36 | puzzled | mvanbaak: ah yes |
| 21:37.53 | mvanbaak | the chan_sccp developers should join chan_skinny |
| 21:38.21 | mvanbaak | ah well |
| 21:38.32 | puzzled | mvanbaak: how do you get an understanding of the protocol? got a ccm to play with? |
| 21:38.39 | mvanbaak | we have some ppl using chan_skinny that also have a callmanager setup |
| 21:38.45 | [TK]D-Fender | mvanbaak: the search for them keeps returning "empty set" :p |
| 21:38.47 | mvanbaak | so they do all the packetdumping for us |
| 21:39.01 | mvanbaak | puzzled: some users have |
| 21:39.06 | puzzled | nice |
| 21:39.19 | mvanbaak | puzzled: and we ask them for dumps from time to time if we need to fix something. and they provide them |
| 21:39.38 | mvanbaak | good example: http://bugs.digium.com/view.php?id=13182 |
| 21:40.39 | mvanbaak | [TK]D-Fender: that's because you have to follow all the forks ;) |
| 21:41.00 | mvanbaak | latest version that actually gets some commits is chan_sccp2b on sf.net ;) |
| 21:41.12 | mvanbaak | or however they call it |
| 21:41.19 | puzzled | yeah naming is confusing |
| 21:41.34 | mvanbaak | it's another fork of chan_sccp, which is itself a fork of chan_skinny |
| 21:41.38 | puzzled | that bug looks good. more like those and issues can get flushed out asap |
| 21:41.47 | mvanbaak | puzzled: yup |
| 21:41.55 | mvanbaak | I'm working on that one right now |
| 21:42.26 | puzzled | would be nice if you could register a phone on one of their ccm's so you can make the trace yourself |
| 21:42.29 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) [NETSPLIT VICTIM] |
| 21:42.29 | *** join/#asterisk un\a\ffiliate (n=un@unaffiliated/unafilliate) |
| 21:42.29 | *** join/#asterisk hardwire (n=hardwire@rdbk-12639.mtaonline.net) [NETSPLIT VICTIM] |
| 21:42.29 | *** join/#asterisk ddunavant (n=David@75.145.240.14) [NETSPLIT VICTIM] |
| 21:42.29 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) [NETSPLIT VICTIM] |
| 21:42.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 21:42.30 | *** join/#asterisk kash (n=kash@2001:470:1f04:457:0:0:0:2) [NETSPLIT VICTIM] |
| 21:42.30 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) [NETSPLIT VICTIM] |
| 21:42.30 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) [NETSPLIT VICTIM] |
| 21:42.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:43.00 | mvanbaak | puzzled: or get ccmse myself |
| 21:43.11 | [TK]D-Fender | mvanbaak: Isn't it ironic? Don't you think? |
| 21:43.45 | mvanbaak | [TK]D-Fender: yup. specially now that chan_skinny is almost on par but totally disclaimed and shipped with core asterisk |
| 21:44.20 | mvanbaak | once my branch is finished it will be even better. RT in chan_skinny is something I really want to happen |
| 21:44.26 | mvanbaak | if only my days were 72 hours |
| 21:44.50 | puzzled | mvanbaak: ccmse is? cisco cm sucky edition? |
| 21:45.01 | mvanbaak | ah well, maybe when I'm in phoenix I can sit down with Qwell and get it done |
| 21:45.18 | mvanbaak | puzzled: some version of ccm that runs on linux and is for small setups |
| 21:45.25 | mvanbaak | it's the cheapest version |
| 21:45.27 | Qwell | mvanbaak: won't be there |
| 21:45.36 | mvanbaak | Qwell: too bad |
| 21:45.53 | mvanbaak | dude, you have to |
| 21:45.56 | mvanbaak | I'll be there |
| 21:45.59 | puzzled | I thought all ccm ran on linux these days after cisco threw out m$ |
| 21:46.17 | mvanbaak | puzzled: yeah, but the full ccm is more expensive |
| 21:46.25 | mvanbaak | ccmse has some limitations |
| 21:46.36 | mvanbaak | but it's close enough to ccm to get the stuff correct |
| 21:46.53 | puzzled | but I guess it's still not $99 |
| 21:47.00 | mvanbaak | no idea |
| 21:47.15 | mvanbaak | I forgot to download it when my CCO account was still active |
| 21:47.28 | puzzled | heh |
| 21:47.40 | puzzled | mvanbaak: http://telecom.paginamarkt.nl/cisco-call-manager-incl-cisco/203961.html |
| 21:47.47 | mvanbaak | I did not have a linux box back then |
| 21:47.50 | mvanbaak | only OpenBSD |
| 21:48.03 | mvanbaak | nowedays I have an ubuntu box for testing etc |
| 21:49.17 | mvanbaak | puzzled: nice |
| 21:49.29 | mvanbaak | care to sponsor it ? |
| 21:49.31 | mvanbaak | hides |
| 21:49.34 | puzzled | heheh |
| 21:49.44 | mvanbaak | whehehe |
| 21:49.55 | puzzled | searches his pockets for some coins |
| 21:49.56 | puzzled | there |
| 21:50.03 | mvanbaak | wow thanks |
| 21:50.13 | puzzled | I remember seeing ccm offered on various torrents |
| 21:50.19 | baliktad | I'm having trouble finding information on a web interface to *'s voicemail system - is this included by default or only available as a separate addon? |
| 21:50.20 | puzzled | while searching for legal stuff off course |
| 21:50.22 | mvanbaak | I still have some 0.50 and 0.20 euro coins left from my sailing trip |
| 21:50.32 | mvanbaak | 0.50 for a shower, 0.20 for hot water to do dishes |
| 21:51.08 | mvanbaak | wow, 5.20 euro in total ! |
| 21:51.12 | mvanbaak | I'm rich ! |
| 21:51.25 | mvanbaak | buys a pack of smokes |
| 21:51.30 | mvanbaak | 1.20 euro left |
| 21:51.37 | mvanbaak | maybe that's enough for ccm |
| 21:51.44 | *** join/#asterisk davidstrauss_ (n=straussd@wikimedia/davidstrauss) |
| 21:52.04 | mvanbaak | holy crap! that reminds me ..... |
| 21:52.35 | mvanbaak | 4 hours on O'hare international |
| 21:52.47 | mvanbaak | is that enough to have a shitload of smokes ? |
| 21:52.49 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 21:52.59 | puzzled | mvanbaak: can't smoke in the airport |
| 21:53.30 | drmessano | Sure you can |
| 21:53.37 | drmessano | Find a security guard |
| 21:53.47 | drmessano | Offer to split the joint with him |
| 21:53.55 | drmessano | "Hey bro, wanna fire one up?" |
| 21:54.06 | drmessano | He's never gonna see you again, so he'll go for it |
| 21:54.12 | mvanbaak | cool |
| 21:54.26 | drmessano | If he doesnt |
| 21:54.32 | Qwell | yeah... he does that, and we may never see mvanbaak again either :p |
| 21:54.38 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 21:54.39 | drmessano | Be all like "Just testing out homeland security, NARC" |
| 21:54.55 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 21:55.03 | [TK]D-Fender | baliktad: There is one included crappy script thats virtually undocumented. |
| 21:55.42 | *** join/#asterisk cmantito (n=gphreak@pool-96-248-64-222.cmdnnj.fios.verizon.net) |
| 22:00.20 | mvanbaak | I'll just wait 18 hours ..... |
| 22:00.22 | mvanbaak | damn |
| 22:02.58 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 22:03.11 | mvanbaak | Qwell: if you're not there, can you put a sccp loaded phone in the van that's going there ? |
| 22:03.13 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 22:03.32 | Qwell | there's a van? |
| 22:03.39 | *** part/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 22:03.46 | mvanbaak | I dont know |
| 22:04.07 | mvanbaak | ah well, just stuff it in Kevin's bag |
| 22:04.18 | file | I nominate... jpeeler to take one |
| 22:04.41 | mvanbaak | I cant |
| 22:04.49 | mvanbaak | checked AMS policy |
| 22:04.57 | mvanbaak | it will get me in big trouble |
| 22:05.28 | Qwell | mvanbaak: I'll see what I can do |
| 22:05.32 | silvertip257 | hello [TK]D-Fender |
| 22:06.06 | mvanbaak | with a bit of luck I'll have CCM on my laptop in a xen instance by then |
| 22:06.15 | mvanbaak | how cool will that be |
| 22:06.16 | jpeeler | doesn't know what he just got voluntold for, but sure |
| 22:06.43 | Qwell | jpeeler: smuggle a phone to Astricon for mvanbaak :D |
| 22:06.54 | mvanbaak | jpeeler: you have to take a cisco phone with sccp firmware on it to astricon |
| 22:07.14 | mvanbaak | file: will you be there ? |
| 22:07.14 | jpeeler | what am i missing here |
| 22:07.19 | file | I will be. |
| 22:07.25 | mvanbaak | nice |
| 22:07.31 | mvanbaak | and bkruse ? |
| 22:07.35 | *** join/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 22:07.36 | jpeeler | yes, i am fortunately coming |
| 22:07.41 | Qwell | jpeeler: mvanbaak can't legally bring one into the country |
| 22:07.54 | nny_1 | any reason why rxgain and txgain adjustments with a digium card don't seem to make a difference? |
| 22:08.09 | jpeeler | oh, that's kind of crazy |
| 22:08.11 | nny_1 | i have set both to -8.0 (from default 0) and it seems to seill be 4600 or so |
| 22:08.15 | [TK]D-Fender | nny_1: Describe exactly how you're testing your changes. |
| 22:08.44 | nny_1 | [TK]D-Fender: having client xfer me (incoming call on channel 23) to the Milliwatt app |
| 22:08.54 | nny_1 | using ztmonitor 23 -vv |
| 22:09.14 | [TK]D-Fender | nny_1: Continue... |
| 22:09.30 | silvertip257 | I've been working on an asterisk + WRT54 setup. I am unable to make inbound/outbound calls through my ITSP. LAN extn calls work. Config / debug here > http://pastebin.com/d29d3c7b0 Thank you. |
| 22:10.01 | nny_1 | adjusting txgain to -8.0, stopping asterisk, stopping zaptel, starting zaptel starting asterisk |
| 22:10.11 | nny_1 | I understand that a reload is supposed to work, tried that too |
| 22:10.20 | nny_1 | Rx: 0 ( 0) Tx: 4435 ( 4435) |
| 22:10.21 | [TK]D-Fender | silvertip257: Looking for 17177734469 in default , SIP/2.0 404 Not Found |
| 22:10.28 | nny_1 | is my current ~ value |
| 22:10.44 | [TK]D-Fender | silvertip257: Dialplan error. There is no match for 17177734469 in [default] |
| 22:10.55 | silvertip257 | [TK]D-Fender: yes, I noticed that, but wasn't sure what I did wrong... |
| 22:11.11 | nny_1 | *adjusting txgain=-8.0 in zapata.conf before the channels= line |
| 22:11.12 | silvertip257 | [TK]D-Fender: should I be using that on my inbound calls instead of the "s" ? ... |
| 22:11.19 | [TK]D-Fender | silvertip257: DIALPLAN ERROR. no match for the EXTENSION dialed by the incoming call |
| 22:11.22 | *** join/#asterisk mgdm (n=michael@serenity.mgdm.net) |
| 22:11.24 | silvertip257 | yes |
| 22:11.35 | [TK]D-Fender | silvertip257: Well thats what they are sending the call to. What do you think? |
| 22:11.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:11.53 | silvertip257 | thanks for the reassurance |
| 22:12.00 | silvertip257 | will try and reload |
| 22:12.04 | [TK]D-Fender | nny_1: You do need to reload chan_zap.so to take effect. That is certain. Shouldn't require anything else. |
| 22:12.11 | mvanbaak | jpeeler: if I bring one with me, I have to pay extra taxes for my laptop and phone |
| 22:12.16 | mvanbaak | roughly 200 USD |
| 22:12.26 | nny_1 | [TK]D-Fender: ok that will keep me from having to stop start the rest |
| 22:12.29 | jeev | Fender |
| 22:12.33 | jeev | i mean mr fender |
| 22:12.34 | Qwell | mvanbaak: oh.. that still sucks |
| 22:12.46 | jpeeler | mvanbaak: i assume that is from your laws |
| 22:12.47 | nny_1 | [TK]D-Fender: shouldn't -8.0 make a big difference in the TX value with Milliwatt? |
| 22:12.58 | mvanbaak | no, it's us import stuff |
| 22:13.11 | [TK]D-Fender | nny_1: Probably. |
| 22:13.20 | [TK]D-Fender | nny_1: Sounds like a time to take it up with Digium support... |
| 22:13.27 | nny_1 | [TK]D-Fender: gotcha |
| 22:13.29 | lesouvage | mvanbaak: you are going to astericon on the US? |
| 22:13.35 | mvanbaak | same as the rule: I can take a fullsize laptop and a minisized laptop. but two fullsize laptops means extra pay |
| 22:13.38 | mvanbaak | lesouvage: yup |
| 22:14.25 | mvanbaak | Qwell: you have to go as well |
| 22:14.31 | jpeeler | mvanbaak: sounds like the US is just trying to make some money |
| 22:15.08 | mvanbaak | yup |
| 22:15.16 | lesouvage | mvanbaak: That is great for you, last time we had contaact you wish you could. |
| 22:15.28 | AiT | [TK]D-Fender, crashed... http://pastebin.com/d4839d960 |
| 22:15.39 | mvanbaak | lesouvage: yeah. ask my creditcard how it's possible ;) |
| 22:16.11 | *** join/#asterisk davidstrauss (n=straussd@wikimedia/davidstrauss) |
| 22:16.35 | [TK]D-Fender | AiT: Guess you'd better submit the core dump info. |
| 22:16.51 | AiT | [TK]D-Fender, how? |
| 22:16.59 | lesouvage | mvanbaak: just see it as an investment ;-) . It certainly is good for your burn out rate, although the USA is getting cheaper every day. |
| 22:17.11 | [TK]D-Fender | AiT: it goes to a file normally. go find it. |
| 22:17.20 | AiT | ok |
| 22:17.22 | eric_hill | Do Polycom phones have a "fallback" mode in the event they lose connectivity with a central SIP registrar? |
| 22:17.36 | baliktad | [TK]D-Fender thanks, I guess I would be OK with just vm -> email but I can't find good information on what the prerequisites are (for a stock *nix machine) |
| 22:17.37 | [TK]D-Fender | eric_hill: Yes |
| 22:17.57 | [TK]D-Fender | baliktad: a sendmail compatible binary. |
| 22:18.33 | eric_hill | [TK]D-Fender, any particular Polycom I should be looking at? |
| 22:18.45 | [TK]D-Fender | eric_hill: For that functionality, any of them |
| 22:18.48 | mvanbaak | lesouvage: ticket is 650 euro |
| 22:18.57 | mvanbaak | 150 usd/night for the hotel |
| 22:19.47 | eric_hill | [TK]D-Fender, how about in general? What's a "wow" phone for the boss? 501? 330/320? |
| 22:19.59 | [TK]D-Fender | eric_hill: 670 |
| 22:20.31 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 22:20.40 | eric_hill | [TK]D-Fender, gracias. |
| 22:21.25 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 22:21.53 | silvertip257 | I changed the "s" in incoming to my DID number - still get the 404 error |
| 22:22.49 | [TK]D-Fender | silvertip257: And you felt no need to show any of it. niiiice.. |
| 22:22.59 | silvertip257 | [TK]D-Fender: pastebin coming your way |
| 22:23.16 | silvertip257 | [TK]D-Fender: > http://pastebin.com/d28de3aac |
| 22:23.28 | silvertip257 | nvm that's the old one ... err |
| 22:23.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:24.35 | silvertip257 | [TK]D-Fender: up to date > http://pastebin.com/d37c3c92f |
| 22:24.45 | lesouvage | mvanbaak: so with an all access registration it is around EUR 1.300,- ! |
| 22:25.00 | mvanbaak | yup |
| 22:25.16 | [TK]D-Fender | silvertip257: and now you're showing me a SIP REGISTRATIOn problem : SIP/2.0 401 Unauthorized |
| 22:25.40 | jblack | That's almost 300 dollars worth of fun. It must be neat as all getout |
| 22:25.51 | lesouvage | mvanbaak: I have a credit card too, it is very tempting just to register and go. |
| 22:26.18 | mvanbaak | lesouvage: credit cards in .nl (by rabobank) can be charged for 2500 euro |
| 22:26.50 | [TK]D-Fender | silvertip257: And while you're at it, you're looking in the wrong dialplan CONTEXT. exten=> 1717+NXXXXXX,1,Goto(voipout|BYEXTENSION|1) <-- and this isn't appttern match until you prepend it with a "_" and "+" is counted as a LITERAL char in there |
| 22:28.38 | lesouvage | mvanbaak: that leaves some nice drinking budget. |
| 22:28.42 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:29.15 | mvanbaak | yup |
| 22:29.25 | mvanbaak | well, I bought a thinkpad |
| 22:29.37 | lesouvage | mvanbaak: I will seriously think about going to Glendale. Iet you know! |
| 22:29.37 | mvanbaak | and my car borked down so had to pay 700 euro there |
| 22:29.51 | bkruse | mvanbaak: No :[ |
| 22:29.59 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 22:30.30 | mvanbaak | bkruse: moron ! |
| 22:30.32 | mvanbaak | ;) |
| 22:31.00 | lesouvage | mvanbaak: just returned from two weeks Isla La Palma Spain so I need some financial recovery too. |
| 22:31.36 | mvanbaak | lesouvage: if you're going, let me know |
| 22:31.53 | mvanbaak | btw, lufthansa flight on 9/22 ;) |
| 22:32.03 | bkruse | bkruse: I cannot go! It's not my fault! :/ but it's ok |
| 22:32.20 | mvanbaak | Lufthansa 9154 |
| 22:32.40 | mvanbaak | bkruse: let me rebook my flight |
| 22:33.08 | mvanbaak | instead of PHX -> WashingtonDC -> AMS use PHX -> DGM -> AMS |
| 22:33.09 | mvanbaak | ;) |
| 22:35.06 | *** join/#asterisk davidstrauss (n=straussd@wikimedia/davidstrauss) |
| 22:35.09 | mvanbaak | ah well |
| 22:35.25 | mvanbaak | at least I can feed file some muffins |
| 22:36.01 | mvanbaak | cat /dev/muffin >> /dev/file |
| 22:36.50 | [TK]D-Fender | silvertip257: (buffer overflow) |
| 22:36.55 | [TK]D-Fender | mvanbaak: rather ;) |
| 22:37.07 | [TK]D-Fender | mvanbaak: followed by a core-dump ;) |
| 22:37.45 | silvertip257 | [TK]D-Fender: k |
| 22:38.00 | mvanbaak | lol |
| 22:38.14 | mvanbaak | gdb /usr/sbin/asterisk core.file |
| 22:38.15 | mvanbaak | bt |
| 22:38.39 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 22:38.41 | mvanbaak | #0 in file.c in function eat_muffin(): too many muffins |
| 22:40.10 | AiT | [TK]D-Fender, another dump...I cannot find where the file is stored |
| 22:40.55 | mvanbaak | in /tmp ? |
| 22:40.55 | [TK]D-Fender | AiT: mvanbaak here just alluded as to where it might be |
| 22:41.09 | mvanbaak | [TK]D-Fender ;) |
| 22:44.43 | silvertip257 | [TK]D-Fender: not sure where it's fscked, but well that's my experience with this ... SIP/2.0 401 Unauthorized |
| 22:45.09 | [TK]D-Fender | silvertip257: thats a separate problem of having the wrong auth on your register |
| 22:45.26 | nny_1 | [TK]D-Fender: hmm so Digium says that hardware echo cancel does not use any settings in zapata.conf other than echocancel=yes, including echocancel=(TAPS) or echotraining timing (which makes sense as far as software - hardware goes). They also say txgain and rxgain have no effect. Does that sound kosher to you? |
| 22:45.36 | [TK]D-Fender | silvertip257: You still have your more immediate problem of the call attemt that IS trying to come in not matching. |
| 22:45.57 | silvertip257 | [TK]D-Fender: the info I have works fine for proper login to the control panel/account thing |
| 22:46.28 | [TK]D-Fender | nny_1: That EC just uses that flag, yes. As for Gains not being configurable, thats either BS (incorrect) or BS (crap) |
| 22:46.30 | *** join/#asterisk underdawg (n=steve@unaffiliated/underdawg) |
| 22:46.36 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 22:46.53 | [TK]D-Fender | silvertip257: "control panel/account thing" ? Huh? |
| 22:47.07 | silvertip257 | [TK]D-Fender: account set up for the ITSP |
| 22:48.02 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:48.26 | [TK]D-Fender | silvertip257 : your register line should also be at the END of everything else under [general] |
| 22:48.51 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 22:49.15 | mvanbaak | hhmm |
| 22:49.20 | mvanbaak | register => bla |
| 22:49.25 | mvanbaak | bindport 5060 |
| 22:49.28 | mvanbaak | ;) |
| 22:51.41 | AiT | [TK]D-Fender / mvanbaak don't see my core dump file in /tmp or /usr/sbin :( |
| 22:52.43 | mvanbaak | AiT: find / -name '*core*' |
| 22:53.16 | *** join/#asterisk Mez (n=mez@ubuntu/member/mez) |
| 22:54.02 | AiT | nice! ;) |
| 22:54.25 | Mez | Hey there, I have an issue in getting my SIPphone to register with asterisk... (softphone and hardphone) - I believe that my ISP is blocking udp 5060 (grr!) any ideas how to get past this? |
| 22:54.54 | mvanbaak | setup a vpn to your asterisk server |
| 22:55.22 | Mez | mvanbaak, I cant really do that... |
| 22:55.28 | nny_1 | [TK]D-Fender: heh reasoning was "it's a digital circuit" |
| 22:55.55 | Mez | I'm actually trying to use a service provider... which stopped working, so trying to setup asterisk to work with it (even if it does mean dialing 0 firsT) |
| 22:56.40 | Mez | does anyone actually know how I can test if my ISP is blocking? |
| 22:56.40 | [TK]D-Fender | nny_1: Gains are still pertinent. digital circuits can still run "hot" |
| 22:57.17 | [TK]D-Fender | Mez: have someone call you |
| 22:57.30 | Mez | [TK]D-Fender, I cannot register... |
| 22:57.53 | [TK]D-Fender | Mez: PASTEBIN is your friend. As is SIP DEBUG. |
| 22:57.55 | [TK]D-Fender | ~pb |
| 22:57.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:57.57 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 22:58.03 | Mez | so I cant get someone to call me... I need to find out whether from here, I can use udp on the right port |
| 22:58.09 | nny_1 | [TK]D-Fender: it appears the echo is only from "local calls". I am cranking the gain down in 2.0 increments and the issue seems* to be subsiding.. |
| 22:58.14 | nny_1 | I <3 echo |
| 22:58.16 | Mez | [TK]D-Fender, what does pastebin have to do with anything ? |
| 22:58.41 | [TK]D-Fender | Mez: Show us the traffic so we can see what's going on |
| 22:59.06 | AiT | [TK]D-Fender, should the core dump be in raw text? I've got a bunch of 20MB files in /etc/asterisk |
| 22:59.30 | Mez | [TK]D-Fender, got a linux program? or should I just log my network traffic? |
| 23:00.04 | [TK]D-Fender | Mez: SIPDEBUG from * CLI |
| 23:01.14 | Mez | http://www.beyondlogic.org/solutions/sipdebug/sipdebug.htm ? |
| 23:01.22 | mvanbaak | AiT: a core dump is NOT in plain text |
| 23:01.38 | mvanbaak | it's in some binary format. lookup the format in google |
| 23:01.40 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
| 23:02.33 | underdawg | ping me |
| 23:02.41 | mvanbaak | pong |
| 23:02.57 | nhuisman_work | so If I want to find out what is leaving hundreds of blank messages what should I do? Look at the call logs? |
| 23:03.34 | [TK]D-Fender | Mez: Go to * CLI and type "sip debug" <---- |
| 23:03.45 | mvanbaak | look at the logfiles, and have a look at the msgXXX.txt files |
| 23:03.48 | eric_hill | nhuisman_work: set the minimum voicemail message length to 4 seconds |
| 23:04.06 | Mez | [TK]D-Fender, on my server? |
| 23:04.07 | nhuisman_work | it's set at 3 |
| 23:04.16 | silvertip257 | [TK]D-Fender: now it says > Looking for s in default |
| 23:04.22 | Mez | [TK]D-Fender, the issue is, I don't know if it's getting through to my server.. |
| 23:04.24 | nhuisman_work | i'm wondering what is calling |
| 23:04.40 | nhuisman_work | it's leaving hundreds of blank voicemails on one vm box in a period of a few hours. |
| 23:04.42 | mvanbaak | nhuisman_work: the .txt files in the vm's directory will have that info |
| 23:04.59 | [TK]D-Fender | Mez: turn is on so YOU CAN SEE. |
| 23:06.23 | nhuisman_work | it looks like somehow my macro is looping |
| 23:07.02 | nhuisman_work | luckily i redirect my console asterisk log to the real log so i'm checking tha tout |
| 23:07.11 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 23:07.27 | mvanbaak | logs ftw |
| 23:07.33 | Mez | [TK]D-Fender, nothing except the normal registration to my outbound PTSN provider |
| 23:07.52 | [TK]D-Fender | nhuisman_work: And maybe you can show us your macro so we can see if you've done anything outright silly |
| 23:08.02 | [TK]D-Fender | Mez: ...PASTEBIN |
| 23:08.03 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 23:08.30 | Mez | [TK]D-Fender, calm down.... and dont treat me like an idiot... |
| 23:08.38 | mvanbaak | ~pb |
| 23:08.39 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:08.50 | nhuisman_work | [TK]D-Fender: yeah taking a look at it now, I actually had copied this macro from someone elses config, probably a mistake. |
| 23:09.38 | mvanbaak | nhuisman_work: gheh, dont blindly copy. read it, understand it, look for flaws, and if you're happy use it |
| 23:10.20 | nhuisman_work | mvanbaak: i didn't blindly copy it, but I think there is a special case in which it loops and won't leave a voicemail properly. checking what that is now. |
| 23:10.46 | irieKen | Anyone know why outgoing calls would be fine, but when people call me, they can hear me but they break up? |
| 23:11.47 | mvanbaak | irieKen: NAT involved ? |
| 23:12.06 | nhuisman_work | irieKen: could be echo problems |
| 23:12.07 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 23:12.08 | irieKen | mvanbaak: Nope; goes directly through a Zap to the PSTN. |
| 23:12.26 | nhuisman_work | that doesn't really make sense though if it's only one way |
| 23:12.46 | irieKen | nhuisman_work: The echo idea doesn't make sense if it's only one way? |
| 23:13.09 | nhuisman_work | yeah I think echo would break up the call both when they call you and when you call them |
| 23:13.15 | nhuisman_work | unless it is going through a different circuit |
| 23:13.27 | irieKen | I believe that it's going through the same circuit. |
| 23:13.41 | legis | Hi, is there a module in asterisk for it to dial a list of numbers from file/database and play a sound when the call its answered? |
| 23:14.15 | nhuisman_work | irieKen: You might try opening the firewall all the way |
| 23:14.21 | [TK]D-Fender | <PROTECTED> |
| 23:14.34 | [TK]D-Fender | legis: go lookup "call files" and AMI originate" on the WIKI |
| 23:14.35 | nhuisman_work | irieKen: err nm i see it's through zap |
| 23:14.37 | [TK]D-Fender | ~wikis |
| 23:14.37 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 23:15.21 | legis | [TK]D-Fender: I see, thanks. |
| 23:16.21 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:17.47 | irieKen | nhuisman_work: Yeah; it's confusing the hell out of me. We also have another issue; when calling out from the Polycom units, there is a delay between when the callee picks up, and when they can hear us. |
| 23:18.07 | *** join/#asterisk BhaalWK (n=bhaal@203-217-12-194.perm.iinet.net.au) |
| 23:18.45 | Mez | [TK]D-Fender, I've actually used netcat to confirm that my ISP is blocking port 5060 :( |
| 23:19.05 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 23:20.31 | Mez | now I need to work out how to change it so that it works on a nice address |
| 23:20.52 | [TK]D-Fender | Mez: What are you looking to connect to? |
| 23:21.00 | Mez | [TK]D-Fender, to sipgate.co.uk |
| 23:21.15 | [TK]D-Fender | Mez: change your "bindport" in sip.conf [general] |
| 23:21.16 | Mez | [TK]D-Fender, but, it seems my ISP is blocking a load of ports in the 5000's |
| 23:21.38 | Mez | [TK]D-Fender, I know, but then I have to figure out which options to change on my phone ;) |
| 23:22.15 | *** join/#asterisk blq (i=Bl@dslb-088-065-172-144.pools.arcor-ip.net) |
| 23:22.38 | *** join/#asterisk blq (i=Bl@dslb-088-065-172-144.pools.arcor-ip.net) |
| 23:25.07 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 23:25.13 | heedly | Hi |
| 23:26.22 | heedly | I do something like DetectBackground, and when it jumps to the talk extension I do a Dial at which point you can no longer hear incoming sound. |
| 23:27.28 | nhuisman_work | is there anyway to print out all the key value pairs in the astdb ? |
| 23:27.30 | AiT | [TK]D-Fender, some exciting information ;) (learned more today about my asterisk box and linux than every before thanks to you!). Here's one of the dump files: http://pastebin.com/m4e96ce75 |
| 23:28.13 | mvanbaak | nhuisman_work: on the cli: database show |
| 23:28.20 | nhuisman_work | mvanbaak: ah thanks |
| 23:30.01 | nhuisman_work | the sip registry uses astdb eh |
| 23:30.04 | nhuisman_work | i see lots of entries from it |
| 23:30.07 | mvanbaak | yup |
| 23:30.45 | nhuisman_work | k screw this macro, no one is using the forwarding feature it was setup to supply |
| 23:30.58 | nhuisman_work | they can use the phone forward if they want one |
| 23:32.34 | [TK]D-Fender | nhuisman_work: And after all this still don't see fit to show us what you've done so we can help fix it. |
| 23:33.01 | nhuisman_work | [TK]D-Fender: was paste binning as we speak |
| 23:34.35 | nhuisman_work | http://pastebin.com/m3d04ce46 |
| 23:34.38 | nhuisman_work | old macro is the first one |
| 23:35.49 | [TK]D-Fender | nhuisman_work: s,3 should be hangup. if you dial a local channel, make sure you don't come BACK. |
| 23:36.06 | nhuisman_work | [TK]D-Fender: It looks like it bugs if the message is too short |
| 23:36.33 | [TK]D-Fender | nhuisman_work: And you go from voicemail right into ANOTHER dial. WHT? |
| 23:36.36 | nhuisman_work | [TK]D-Fender: which is probably cause by that line you just mentioned |
| 23:36.59 | nhuisman_work | that's used by the goto statement |
| 23:37.26 | nhuisman_work | so at line 8 it checks if there is a forward astdb flag set, if there is it dials that number |
| 23:37.30 | drmessano | In general, when provisioning with XML, and you send a small set of parameters to a device, is the idea "Default everything and then add these" or are the other parameters left alone? |
| 23:37.33 | nhuisman_work | otherwise it goes voicemail at line 10 |
| 23:37.49 | nhuisman_work | drmessano: that's how mine is. |
| 23:38.02 | [TK]D-Fender | nhuisman_work: here's what do do. At the start of your macro set a counter. then check your forwarding. if forwarded, increment the counter, and OVERRIDE the exten to dial (arg1). The check if you've exceeded a max-forward limit |
| 23:38.22 | [TK]D-Fender | nhuisman_work: Do NO bother recursing the dialplan as this seems limited to forwarding to extensions. |
| 23:38.49 | drmessano | nhuisman_work: Which one, I mentioned two scenarios? |
| 23:38.59 | [TK]D-Fender | nhuisman_work: Then once you've hit the end of your forwarding chain, actual just do your boring single dial |
| 23:40.05 | nhuisman_work | drmessano: I thought it was, there is a set of defaults, if you change them then the defaults over overridden. |
| 23:40.07 | AiT | [TK]D-Fender, can you take a look at that core dump when you have a chance? http://pastebin.com/m4e96ce75 |
| 23:40.10 | nhuisman_work | drmessano: but don't take my word for it. |
| 23:40.28 | [TK]D-Fender | AiT: Can't help you with that. Sorry |
| 23:40.30 | drmessano | But does it change to defaults every time? |
| 23:40.41 | nhuisman_work | drmessano: oh you mean does it stored on the device? |
| 23:40.51 | drmessano | Grrr |
| 23:41.10 | drmessano | No |
| 23:42.19 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 23:42.22 | *** part/#asterisk Paige (n=Paige@208.89.241.9) |
| 23:42.29 | drmessano | Question was.. and I have no idea if this varies across brands of phones, when you apply an XML and address say 3 or 4 parameters ONLY, does the phone WIPE to default then apply the 3 or 4 or does it ONLY CHANGE the 3 or 4 and leave the rest however they are currently set.. |
| 23:42.43 | nhuisman_work | drmessano: i see |
| 23:43.03 | *** join/#asterisk LemensTS (n=matthew@adsl-70-130-242-118.dsl.stlsmo.swbell.net) |
| 23:43.40 | nhuisman_work | drmessano: sorry I'm not really sure, I might have run up against that situation but I've since forgotten how it works. |
| 23:43.54 | nhuisman_work | it's probably safe to say there is no standward |
| 23:43.54 | nhuisman_work | heh |
| 23:45.26 | mvanbaak | drmessano: the cisco phones will leave the stuff you did not mention in the xml alone |
| 23:45.37 | drmessano | I see |
| 23:45.39 | mvanbaak | for example: |
| 23:45.53 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 23:46.01 | mvanbaak | set 'cmm server fallback' using the phone menu |
| 23:46.12 | mvanbaak | give it an xml file with only the main ccm |
| 23:46.20 | mvanbaak | it will leave the fallback setting alone |
| 23:46.24 | mvanbaak | it wont reset it |
| 23:46.47 | mvanbaak | unless you give it an empty value in the xml |
| 23:46.49 | nhuisman_work | mvanbaak: that seems to be the way my phones have responded to xml changes as well. |
| 23:47.14 | mvanbaak | yeah |
| 23:47.15 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 23:47.20 | mvanbaak | and it makes sense to me |
| 23:47.45 | mvanbaak | 'if provisioning does not overwrite it, take the stuff from RAM' |
| 23:47.46 | drmessano | Just means having to address the full set of XML |
| 23:48.04 | mvanbaak | drmessano: that's always the best way to do it |
| 23:48.08 | mvanbaak | just to make sure |
| 23:48.27 | mvanbaak | nhuisman_work: you still at work at 2 AM ? |
| 23:48.33 | drmessano | Got any XML examples for PAP2s? lol |
| 23:48.38 | mvanbaak | ah, hawaii |
| 23:48.42 | nhuisman_work | ;) |
| 23:49.01 | nhuisman_work | plus, where the heck are you? |
| 23:49.08 | mvanbaak | in .nl |
| 23:49.11 | nhuisman_work | ah |
| 23:49.14 | mvanbaak | yes, it's 2 AM here |
| 23:49.39 | mvanbaak | I should go to bed |
| 23:49.43 | mvanbaak | 4 hours of sleep |
| 23:49.50 | mvanbaak | and I'm getting drunk |
| 23:50.02 | mvanbaak | that doesn't mix |
| 23:50.02 | nhuisman_work | hehe |
| 23:50.08 | nhuisman_work | the best kind |
| 23:50.12 | mvanbaak | yeah |
| 23:50.19 | drmessano | HA |
| 23:50.21 | mvanbaak | I'll regret it tomorrow morning at 6 AM |
| 23:50.28 | drmessano | Get drunk at work, come home and party |
| 23:50.49 | mvanbaak | ah well |
| 23:51.01 | mvanbaak | I'm thinking bout giving up my job anywayz |
| 23:51.19 | Yourname`` | seanbright: You there? |
| 23:51.25 | drmessano | Good idea |
| 23:51.43 | drmessano | Im hoping my wife gets an awesome high paying job so I can be a stay at home asterisk |
| 23:51.46 | Yourname`` | What's up drmessano.. haven't seen your cheery self in here for some time |
| 23:52.21 | drmessano | Been busy running over kids in my SUV.. time consuming |
| 23:52.27 | nhuisman_work | drmessano: I'm not sure my dream life would be living and breathing asterisk |
| 23:52.27 | nhuisman_work | heh |
| 23:52.58 | mvanbaak | drmessano: I'm waiting for that as well |
| 23:53.05 | irieKen | Anyone know why I would get stuttering ONLY when someone calls me? When I call out, everything is fine. And, they can hear me, whether I call out or they call me... But for some reason, I hear stuttering when they call me. |
| 23:53.12 | mvanbaak | drmessano: only difference is: I'll never have kids |
| 23:53.22 | nhuisman_work | irieKen: is it every call, even say a call from your cel |
| 23:53.23 | nhuisman_work | l |
| 23:53.36 | nhuisman_work | mvanbaak: castration sucks eh? |
| 23:53.40 | nhuisman_work | jk :P |
| 23:53.42 | irieKen | nhuisman_work: less stuttering from cell phone, but it's still there. |
| 23:53.44 | mvanbaak | no, it rox |
| 23:53.53 | nhuisman_work | or did you actually do that? |
| 23:54.01 | mvanbaak | yup |
| 23:54.13 | Yourname`` | ah, must be a hard process. Come to Toronto, I'll show you all those loud mouth noisy teens and you can have a good go at 'em |
| 23:54.23 | mvanbaak | wouldn't have married my wife if she wanted kids |
| 23:54.25 | nhuisman_work | I might store a bunch of my little wonders in cold storage |
| 23:54.33 | nhuisman_work | if I were to do that |
| 23:55.02 | mvanbaak | nhuisman_work: just wank off in front of your freezer |
| 23:55.04 | mvanbaak | ;) |
| 23:55.10 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 23:55.11 | irieKen | nhuisman_work: So, any ideas?:) |
| 23:55.19 | mvanbaak | make sure you've opened the door ;) |
| 23:55.29 | nhuisman_work | mvanbaak: can use it to glue my papers on |
| 23:55.37 | mvanbaak | lol |
| 23:55.48 | mvanbaak | ok, now it's really time to get some sleep |
| 23:55.56 | mvanbaak | before I say more stupid stuff |
| 23:56.31 | Yourname`` | Meanwhile, Agent 101 signed into helpq, Agent 102 signed into testq. These two queues cascade each other, helpq overflows into testq and vice versa. But both these agents will get confused as to where these calls are coming from (they're on AgentLogin()) How can I make them know that the next call they're getting is not from their "home" queue.. but an overflow from another queue? seanbright... |
| 23:56.33 | Yourname`` | ...said announce, but I'm looking to see if there's another alternative. Because helpq gets calls from inbound DIDs as well, one DID is for helping A product and another DID is for helping B product. |
| 23:56.34 | mvanbaak | off to bed here |
| 23:56.36 | mvanbaak | see you tomorrow |
| 23:56.38 | mvanbaak | latero |
| 23:56.38 | Yourname`` | Any suggestions? |
| 23:56.50 | Yourname`` | mvanbaak! wait |
| 23:56.52 | Yourname`` | lol |
| 23:57.37 | nhuisman_work | irieKen: is there anyway you can bypass asterisk and use a normal phone to test the line. |
| 23:57.46 | irieKen | Yeah. Sounds fine. |
| 23:57.50 | mvanbaak | Yourname``: use Set(CALLERID(name)=<queue>) |
| 23:58.13 | Yourname`` | mvanbaak: aha! That's kind of what I wanted to do. However, they are on AgentLogin().. |
| 23:58.16 | irieKen | nhuisman_work: Yeah, it sounds fine when the call is made directly. It's either a hardphone or asterisk probelm. |
| 23:58.47 | mvanbaak | Yourname``: exten => bla,n,Set(CALLERID(name)=testq) |
| 23:58.49 | nhuisman_work | irieKen: you could try using a soft phone |
| 23:59.01 | mvanbaak | exten => bla,n,Queue(testq,,20) |
| 23:59.26 | mvanbaak | exten => bla,n,Set(CALLERID(name)=helpq) |
| 23:59.28 | Yourname`` | mvanbaak: AgentLogin() keeps the agent logged in constantly.. meaning the Agent is on a constant "call" with Asterisk and Asterisk delivers queue calls WITHIN that bridge.. |
| 23:59.35 | irieKen | nhuisman_work: yeah, let me load one up. |
| 23:59.40 | mvanbaak | exten => bla,n,Queue(helpq,,20) |
| 23:59.43 | Yourname`` | mvanbaak: So there's no way the call 'rings' on the phone due to AgentLogin() |
