| 00:16.49 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 00:16.49 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 00:17.58 | *** join/#asterisk linuxviewer (n=example@68.110.115.3) |
| 00:21.09 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:35.11 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 00:41.11 | *** join/#asterisk [8none1] (n=[8none1]@70.216.106.183) |
| 00:42.47 | *** join/#asterisk MrMeek (i=MrMeek@74.223.252.162.nw.nuvox.net) |
| 00:46.38 | *** join/#asterisk butchevans2001 (n=butcheva@host-72-51-244-77.newwavecomm.net) |
| 00:49.31 | butchevans2001 | I know this is not a zultys support site. I am having trouble with my zultys 4x4 and not sure what term to google. When I make a call, some systems with a "press x to reach person.." do not recognize the digit I press. It seems to me that the problem is not the DTMF, but more related to the fact that the button press is too short. what is the name of the setting that I am looking for |
| 00:50.06 | butchevans2001 | this does not occur everywhere, by the way. Just some places. |
| 00:52.18 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:52.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:53.33 | phliver | Anybody in here familiar with the cisco 7905 phones? I just received one and I'm wondering if there is supposed to be anything on the screen when you plug it in. |
| 00:54.35 | phliver | There is a red button and green button that stay lit up. Nothing shows up on the screen and I think its DOA. I know my 7960s at least show status when they start. |
| 00:55.03 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 00:56.47 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
| 01:02.33 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:10.14 | *** join/#asterisk ethicx (n=geneticx@adsl-2-59-65.mia.bellsouth.net) |
| 01:16.30 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:20.07 | *** join/#asterisk s14ck (n=s14ck@190-76-114-69.dyn.movilnet.com.ve) |
| 01:20.08 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
| 01:23.08 | *** join/#asterisk mumtazah (n=mumtazah@36.98.48.60.wmu01-home.tm.net.my) |
| 01:26.20 | *** join/#asterisk jcape (n=jcape@adsl-75-21-78-108.dsl.chcgil.sbcglobal.net) |
| 01:28.35 | *** join/#asterisk jcape (n=jcape@adsl-75-21-78-108.dsl.chcgil.sbcglobal.net) |
| 01:30.36 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 01:31.02 | *** join/#asterisk max_ep (n=max@c-98-220-76-124.hsd1.in.comcast.net) |
| 01:31.41 | max_ep | What's the best/cheapest way to get an FXO (Or an FXO -> SIP gateway)? |
| 01:31.56 | max_ep | I'm looking at this not really for asterisk, but to use a Softphone with my landline |
| 01:35.22 | *** join/#asterisk scalex000 (n=chatzill@200.88.91.73) |
| 01:35.29 | scalex000 | good evening |
| 01:35.40 | scalex000 | TK: Are you there? |
| 01:37.48 | scalex000 | Hi everyone!!! I setup a SIP, I need to pickup the line before dial, How can I do it? |
| 01:43.46 | KyleK | ? |
| 01:44.15 | KyleK | call whatever it is without a number? |
| 01:44.40 | scalex000 | which function (Dial?) |
| 01:45.32 | KyleK | well what line are you picking up before you dial? |
| 01:45.59 | scalex000 | kylek: let me explain for you. |
| 01:46.21 | KyleK | kk |
| 01:46.51 | scalex000 | kylek: I setup a Sip with a linkys PAP, this line I connect to Analog Station on my BCM (note: I have problem to get VOIP Trunk in my country) |
| 01:48.01 | KyleK | whats a BCM? |
| 01:48.17 | scalex000 | kylek: so I want to from another extension to pickup this SIP to dial internal and outbound |
| 01:48.22 | scalex000 | Nortel BCM |
| 01:48.36 | scalex000 | Business Comunication Manager |
| 01:49.08 | KyleK | PAP is an ATA with an FXS for running a handset normally? |
| 01:49.11 | scalex000 | yes |
| 01:49.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:50.19 | KyleK | Dial(SIP/username@PAP.ip.address) should do it |
| 01:50.58 | KyleK | then your BCM should detect the rings and pick up |
| 01:51.04 | scalex000 | ok |
| 01:51.16 | scalex000 | I understand I little, let me try it |
| 01:51.19 | scalex000 | thanks |
| 01:56.55 | KyleK | max_ep: cheapest FXO's tend to be FXO->SIP gateways but I'm not aware of any concensus on who makes the best ones |
| 02:02.33 | *** join/#asterisk jcape (n=jcape@adsl-75-21-78-108.dsl.chcgil.sbcglobal.net) |
| 02:08.04 | *** join/#asterisk Deeewayne (n=dwayne@71-80-105-185.dhcp.jcsn.tn.charter.com) |
| 02:08.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:09.46 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 02:10.26 | raden_work | is there a way sip.conf if lets inbound fails to qualify it can try inbound1 |
| 02:10.37 | KyleK | hey if I dial SIP/trunk/a&SIP/trunk/b how can i tell who picked up in the CDR's? |
| 02:11.48 | KyleK | raden_work: what do you mean? if it fails to register at inbound.bwankers.edu it'll try inbound1.bwankers.edu? |
| 02:12.09 | raden_work | yeah |
| 02:14.55 | KyleK | I have no idea on that :-/ |
| 02:20.36 | max_ep | KyleK: I'm seeing the x100 pci card as fairly cheap... but there are VERY mixed reviews on that. |
| 02:20.56 | max_ep | Most of the places selling any of these things have very questionable return/support policies :/ |
| 02:25.33 | KyleK | do you gamble? |
| 02:27.52 | KyleK | cost/return ratio sucks but the odds are probably a bit better |
| 02:34.33 | KyleK | anyone using a DECT SIP phone? |
| 02:36.07 | *** join/#asterisk WindBack (n=quassel@200-122-74-15.cab.prima.net.ar) |
| 02:37.34 | WindBack | I had an unespected crash on my asterisk server 1.6.0.13. What file can i see to see what happened? I'm looking in the /var/log/asterisk/messages but i dont see anything extrange |
| 02:38.56 | ethicx | Hello, I need to install a digium card TDM400P with the latest DAHDI drivers, but I have never configured one of these cards before and I have found a bunch of guides with zaptel drivers but no dahdi guides, anyone might point me in the right direction? |
| 02:47.26 | KyleK | ethicx: isn't dahdi an evolution of zaptel? should be similar options |
| 02:50.12 | ethicx | yes it is, both some options change i.e. echo cancelation.. |
| 02:50.44 | ethicx | I guess trial and error is the best guide =D |
| 02:50.48 | thehar | ~dahdi |
| 02:50.49 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 02:52.51 | russellb | zomg thehar |
| 02:53.11 | thehar | russellb: ohhai thar |
| 02:53.25 | russellb | h4x0rz pbx02 |
| 02:53.31 | thehar | please noes :( |
| 02:53.50 | thehar | tonight is the first night since production that i haven't gotten paged |
| 02:53.55 | thehar | i need a break! |
| 02:54.55 | thehar | i went to the bar yesterday.. got really shitty drunk with friends.. thank god i can drunk asterisk! |
| 02:55.50 | thehar | russellb: i might not be going to astricon :( |
| 02:58.32 | russellb | thehar: sadness |
| 02:58.54 | thehar | russellb: i have it budgeted but there are new policys for conferences that require us to book the hotel on our own cards and get reimbursed. |
| 02:59.33 | russellb | credit card! |
| 02:59.36 | thehar | lol |
| 02:59.43 | thehar | i'm gay. i like to shop. |
| 02:59.50 | thehar | therefore there is not $600 lying around for a hotel |
| 03:00.51 | russellb | stolen credit card! |
| 03:01.23 | thehar | haha |
| 03:01.28 | thehar | steals russellb's credit card. |
| 03:07.05 | Nugget | 08-Oct-2007 15:54 <russellb> i just don't like people getting credit they don't deserve, IMO |
| 03:07.21 | thehar | lol |
| 03:08.20 | Nugget | rust never sleeps and the internet never forgets. |
| 03:10.49 | butchevans2001 | <PROTECTED> |
| 03:11.12 | butchevans2001 | any help would be greatly appreciated |
| 03:11.19 | russellb | Nugget: ! |
| 03:12.04 | russellb | butchevans2001: um, contact zultys? |
| 03:12.19 | russellb | or drop it off the room and then install asterisk? |
| 03:12.23 | russellb | s/room/roof/ |
| 03:13.16 | butchevans2001 | zultys may be of assistance. I simply thought someone here may know what this would be called. or offer an idea for what to google |
| 03:22.12 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 04:03.35 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@bas2-toronto44-1176437585.dsl.bell.ca) |
| 04:03.43 | darkdrgn2k3 | Hey, how do you specify a domain for a sip trunk? |
| 04:15.17 | phix | ok I bought a Snom300 :) |
| 04:15.23 | phix | $50 AUD |
| 04:18.15 | phix | should I commit suicide yet [TK]D-Fender ? |
| 04:21.42 | *** join/#asterisk hadding (n=kyle@207.177.231.9) |
| 04:24.19 | hadding | "Add the following lines to your extension config file in the same context as your SIP phone. " |
| 04:24.34 | hadding | whats the config files name? |
| 04:26.00 | *** join/#asterisk Micc (n=Micc@c-71-231-123-28.hsd1.wa.comcast.net) |
| 04:26.50 | carrar | ~book |
| 04:26.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:30.39 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 04:32.18 | p3nguin | hadding: Ordinarily, the extensions are configured in extensions.conf. |
| 04:33.42 | p3nguin | darkdrgn2k3: host= |
| 04:34.10 | p3nguin | darkdrgn2k3: Is that what you are asking about? |
| 04:34.15 | darkdrgn2k3 | p3nguin: is the host= not the proxy |
| 04:34.31 | darkdrgn2k3 | p3nguin: my sip provider gave me a username AND a domain |
| 04:34.55 | p3nguin | darkdrgn2k3: What is the "domain"? Maybe I can figure out what it is based on the name. |
| 04:35.19 | darkdrgn2k3 | p3nguin: the sip provider provides sip lines for a comapny i work for |
| 04:35.23 | darkdrgn2k3 | the domain is the companie's name |
| 04:35.51 | p3nguin | darkdrgn2k3: Maybe they expect you to use that in realm= ? |
| 04:35.55 | *** part/#asterisk butchevans2001 (n=butcheva@host-72-51-244-77.newwavecomm.net) |
| 04:36.02 | darkdrgn2k3 | hmmm worth a try |
| 04:36.28 | darkdrgn2k3 | i managed to get teh sip registered.... |
| 04:36.32 | darkdrgn2k3 | jsut cant seem to make outbound calls |
| 04:36.52 | p3nguin | I wouldn't think they would need to tell you what your company name is, though. |
| 04:37.14 | p3nguin | What happens when you dial out? |
| 04:37.26 | darkdrgn2k3 | p3nguin: the sip provider usses a nortel MCS, and i think they segregate differnt customers using "domains" |
| 04:37.33 | darkdrgn2k3 | all circutes busy |
| 04:38.03 | darkdrgn2k3 | oh yeh i tested this account using SJPhone, and it does work |
| 04:39.47 | p3nguin | Without seeing sip.conf and possibly a debug of a failed call, there's not much I can tell you. |
| 04:40.12 | darkdrgn2k3 | ok well i am using elastix (thought it be easier to get my feet wet) |
| 04:40.27 | p3nguin | I don't even know what that is. |
| 04:40.45 | darkdrgn2k3 | its a packaged asterisk distro |
| 04:40.46 | darkdrgn2k3 | based on centos |
| 04:41.04 | darkdrgn2k3 | my peer details/registry string: http://pastebin.ca/1570471 |
| 04:43.12 | p3nguin | I don't know how critical it is, but I put my register string in the [general] context. |
| 04:43.24 | p3nguin | And you haven't even defined a context for your peer at all. |
| 04:43.51 | p3nguin | And realm probably should be specified in [general] instead of your peer context. |
| 04:44.36 | darkdrgn2k3 | just fyi screen shot of sjphone's config screen -> (http://imagebin.ca/view/Wy1bpGDa.html) |
| 04:45.22 | darkdrgn2k3 | and sip show registry shows that its been registered |
| 04:45.24 | darkdrgn2k3 | proxy.oneconnect.ca:5060 name1@compa 93 Registered Fri, 18 Sep 2009 00:59:59 |
| 04:46.50 | p3nguin | What about sip show peers? |
| 04:47.02 | p3nguin | I can't imagine it can work without a context. |
| 04:48.22 | darkdrgn2k3 | first of all sip_additional.conf -> http://pastebin.com/d45bb4c89 |
| 04:48.59 | darkdrgn2k3 | sip show peers returns |
| 04:49.00 | darkdrgn2k3 | XC/username 205.205.231.22 5060 Unmonitored |
| 04:49.36 | darkdrgn2k3 | and my connection in from my soft client |
| 04:49.40 | p3nguin | It looks like you started to make a context for mycompany at the bottom of sip_additional. |
| 04:49.56 | p3nguin | I'm sure if you clean up the mess, it will work. |
| 04:50.09 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 04:50.26 | darkdrgn2k3 | thats for incomming though no? |
| 04:50.53 | p3nguin | You need to have a peer for outbound calling. That's where your problem was, right? |
| 04:51.34 | darkdrgn2k3 | yes |
| 04:51.48 | darkdrgn2k3 | right now i dont really care about incomming.. |
| 04:52.04 | p3nguin | So clean up the stray lines in sip.conf and move them into the appropriate context in sip_additional.conf. |
| 04:52.44 | darkdrgn2k3 | in sip.conf |
| 04:52.52 | darkdrgn2k3 | does #include mean include or commented out incldue |
| 04:53.18 | p3nguin | #include means it will be included. ; is a comment mark. |
| 04:53.24 | darkdrgn2k3 | ok |
| 04:53.38 | darkdrgn2k3 | so looks liek sip.conf just includes a bunch of files.. |
| 04:53.58 | p3nguin | That's not uncommon, actually. |
| 04:54.08 | darkdrgn2k3 | yeh.. just looking at the general area |
| 04:54.11 | p3nguin | sip_additional, sip_custom, etc. |
| 04:55.13 | darkdrgn2k3 | so far general is |
| 04:55.13 | darkdrgn2k3 | http://pastebin.com/m39eeddcc |
| 04:55.35 | darkdrgn2k3 | so what stray lines am i looking up to clean up? |
| 04:56.18 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 04:56.25 | hadding | Add the following lines to your extension config file in the same context as your SIP phone. |
| 04:56.31 | hadding | exten => 33,1,Answer exten => 33,2,AGI(cidspoof.agi) |
| 04:56.37 | hadding | where do i put this? |
| 04:56.40 | p3nguin | Well, I asked for sip.conf and get pieces of it, so I really have no idea how you have it configured. I could only assume that the first paste doesn't even have a context for it, so those need to be removed and put into the correct context in sip_additional.conf. |
| 04:57.07 | darkdrgn2k3 | p3nguin: let me try to compile the whole file from the includes.. seems to only be 4 files |
| 04:57.10 | darkdrgn2k3 | 1 sec |
| 04:57.10 | p3nguin | hadding: It goes in extensions.conf in the same context that you use for dialing to and from your phone(s). |
| 04:57.17 | darkdrgn2k3 | feels like a pre processor in a compiler :-D |
| 04:57.51 | p3nguin | darkdrgn2k3: If you would just paste the entire sip.conf and edit your passwords, that would make my life a lot more pleasant. |
| 05:00.33 | darkdrgn2k3 | ok |
| 05:00.36 | darkdrgn2k3 | i THINK i got it all |
| 05:00.36 | darkdrgn2k3 | http://pastebin.com/m479191a7 |
| 05:00.45 | darkdrgn2k3 | 1 larger merged file :) |
| 05:00.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:01.17 | darkdrgn2k3 | 1111 is an extension |
| 05:01.29 | darkdrgn2k3 | XC is the outboud trunk |
| 05:01.40 | darkdrgn2k3 | company.com is sopposed to be the inbound but im not using it yet |
| 05:02.17 | p3nguin | XC is the one you're having trouble with? |
| 05:02.25 | darkdrgn2k3 | yes |
| 05:03.04 | darkdrgn2k3 | XC is setup as the only outbout route |
| 05:03.12 | p3nguin | Was the realm=company.com in their before we started this? |
| 05:03.31 | darkdrgn2k3 | no] |
| 05:03.37 | darkdrgn2k3 | added it as per your suggestion earlier |
| 05:03.49 | darkdrgn2k3 | no change in status |
| 05:04.30 | darkdrgn2k3 | peer details and registry information adapted from post http://forum.voip.com/topic/2818/trixboxasterisk-config-for-sip-trunk/ |
| 05:04.53 | p3nguin | My suggestion was to use the domain name for your realm, not to include realm in your context for that ITSP. And that was before I got very much info about the problem. |
| 05:05.10 | darkdrgn2k3 | 10-4 |
| 05:05.45 | p3nguin | Did XC not give you the configuration for their service? |
| 05:06.34 | darkdrgn2k3 | no, |
| 05:06.42 | darkdrgn2k3 | they are running a nortel mcs |
| 05:07.02 | darkdrgn2k3 | it usses hardware phones or a "propriatery" softphone |
| 05:07.35 | darkdrgn2k3 | however i had luck getting sjphone to work |
| 05:07.40 | darkdrgn2k3 | debug log from a successfull call: http://pastebin.ca/1570487 |
| 05:08.21 | p3nguin | If they provide SIP termination, I don't see why they need to have any special equipment. |
| 05:09.03 | darkdrgn2k3 | p3nguin: you dont, but they dont provide sip details outside of the config for their soft phones :-S |
| 05:12.00 | p3nguin | If you've registered correctly (which it seems that you have) and you have them as a peer (which it seems like you do), then my next thought is the dialplan. Did you use the softphone with Asterisk, or directly to XC? |
| 05:12.13 | Micc | Is there a way to set the language for the voicemail app? |
| 05:12.26 | Micc | in the dialplan |
| 05:12.28 | darkdrgn2k3 | im trying to do it from the soft phone |
| 05:12.39 | darkdrgn2k3 | (dont know how to test xc dirrectly) |
| 05:12.55 | p3nguin | What about the successful call? |
| 05:13.26 | darkdrgn2k3 | it was using sjphone configured directly to the xc sip |
| 05:13.29 | darkdrgn2k3 | not via asterisk |
| 05:13.57 | p3nguin | Okay, so let's look at your dialing rules. |
| 05:14.05 | darkdrgn2k3 | log from an unsucessful cll from asterisk: |
| 05:14.05 | darkdrgn2k3 | http://pastebin.ca/1570492 |
| 05:16.30 | darkdrgn2k3 | outbout dial plan |
| 05:16.30 | darkdrgn2k3 | http://pastebin.ca/1570495 |
| 05:16.44 | darkdrgn2k3 | (took the 9 out of the default 9 for outbound plan) |
| 05:18.16 | hadding | ; Example "main menu" context with submenu |
| 05:18.17 | hadding | ; |
| 05:18.17 | hadding | ;exten => s,1,Answer |
| 05:18.36 | hadding | I change that to exten => 33,1,Answer |
| 05:18.38 | hadding | ? |
| 05:18.53 | hadding | Thats the right line, correct? |
| 05:19.01 | p3nguin | I doubt it. |
| 05:19.13 | p3nguin | I don't even know what you're trying to do. |
| 05:19.38 | hadding | <PROTECTED> |
| 05:19.42 | hadding | exten => 33,1,Answer exten => 33,2,AGI(cidspoof.agi) |
| 05:20.10 | p3nguin | You've pasted that like four times, now. |
| 05:21.18 | p3nguin | It's funny how you can paste it four times here, but you're totally incapable of pasting it just once into extensions.conf. |
| 05:21.39 | hadding | Well sorry, I dont now were I put it in there |
| 05:21.49 | p3nguin | Even though I told you? |
| 05:22.19 | p3nguin | How many contexts do you have? Put it in the context where your phones dial each other. |
| 05:24.00 | darkdrgn2k3 | p3nguin: so any ideas? |
| 05:26.19 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 05:26.28 | p3nguin | darkdrgn2k3: No, I gave up trying to determine the problem based on small pieces of config files. |
| 05:27.13 | darkdrgn2k3 | 1) thanx for telling me |
| 05:27.32 | darkdrgn2k3 | 2) log seems to indicate the call IS going out ot the sip trunk, to the dail plan may not be the problem |
| 05:27.50 | p3nguin | Well, I did ask for the files rather than small particles of them. So there's not much I can do about that. |
| 05:28.01 | darkdrgn2k3 | what files did u ask for |
| 05:28.07 | darkdrgn2k3 | i compiled the show sip.conf file |
| 05:28.17 | darkdrgn2k3 | i must of missed smethign else? |
| 05:28.20 | darkdrgn2k3 | show=whole |
| 05:30.15 | darkdrgn2k3 | *sigh* why do i feel so hopelessly lost in this asterisk world :-S |
| 05:30.18 | p3nguin | Originally I wanted to see all the configurations. |
| 05:30.49 | darkdrgn2k3 | can i msg you? |
| 05:30.55 | p3nguin | There's so much crap in that elastix thing that I doubt I would be able to understand it anyway. |
| 05:31.05 | p3nguin | I would prefer if you don't. |
| 05:31.10 | darkdrgn2k3 | okie then |
| 05:31.24 | darkdrgn2k3 | just dont wanan speak in certain terms on an open channel |
| 05:31.36 | darkdrgn2k3 | but oviosuly you reached your limit, regardless thakn you for all your insite. |
| 05:31.40 | darkdrgn2k3 | ....every little bit helps... |
| 05:32.09 | darkdrgn2k3 | jsut tell me this. from asterisk console, is there a way to test the sip directly? |
| 05:32.54 | p3nguin | help originate |
| 05:33.28 | darkdrgn2k3 | hmm thank you |
| 05:33.31 | darkdrgn2k3 | :) |
| 05:33.45 | p3nguin | That elastix thing reminds me a lot of PiaF. |
| 05:34.09 | darkdrgn2k3 | p3nguin: well once i figured out the basic config of teh sip, im hopeing to start a clean nix box with not bs.. pure config files |
| 05:34.18 | p3nguin | I tried it once. I had to dump it rather quickly due to all the mess in the configs. |
| 05:34.24 | darkdrgn2k3 | p3nguin: just figured a gui will give me an idea of what CAN be done in the ned |
| 05:35.22 | darkdrgn2k3 | this was sopposed to be a fun little project, but its turning out into a bit of a headach because of the nortel domain bs... :-S |
| 05:36.01 | p3nguin | I bet if I had the credentials, I could plug them into my sip.conf and extensions.conf and make calls on their system. |
| 05:36.07 | *** join/#asterisk fiddur (n=fiddur@dhcp08.textalk.com) |
| 05:36.40 | p3nguin | Meaning, I bet it's all the crap in elastix that is causing the problems. |
| 05:36.57 | darkdrgn2k3 | ok forget about elastix, lets talk theory |
| 05:37.17 | darkdrgn2k3 | i have a proxy server FQDN, username, "domain", extension |
| 05:37.47 | darkdrgn2k3 | i create a context in sip.conf, and register in [general] |
| 05:37.48 | darkdrgn2k3 | right/ |
| 05:38.50 | p3nguin | I put my register string in the [general] context, but I don't think it's required to go there... then a context at the bottom [myitsp] host= username= secret= |
| 05:39.11 | darkdrgn2k3 | where does the domain go then? |
| 05:39.15 | darkdrgn2k3 | host is proxyu server |
| 05:39.22 | darkdrgn2k3 | username is username or username@domain.com? |
| 05:40.24 | p3nguin | Depends on how they have it set up on their end, but probably just the user name. |
| 05:40.59 | p3nguin | http://pastebin.com/d14008b76 |
| 05:41.15 | p3nguin | This is my context for my outbound calling. |
| 05:41.25 | p3nguin | without the register string |
| 05:41.41 | darkdrgn2k3 | the od part is looking a differnt IP phones |
| 05:41.47 | darkdrgn2k3 | some have a "domain" filed others do nto |
| 05:41.53 | darkdrgn2k3 | (soft phones) |
| 05:42.25 | p3nguin | My register string is this: register => 105:sekrit123@sip.us4.voip.ms:5060 |
| 05:43.05 | darkdrgn2k3 | curerntly mine is username@domain:secret@proxy.oneconnect.ca/phonenumber |
| 05:43.10 | p3nguin | Between the pasted context and the register string (along with a dial plan in extensions.conf), I can make calls. |
| 05:43.26 | darkdrgn2k3 | hmm |
| 05:43.37 | darkdrgn2k3 | *sigh* |
| 05:43.41 | darkdrgn2k3 | well back to the drawing board |
| 05:43.52 | darkdrgn2k3 | i dont think its the dial plan cause i didnt change anythign in there |
| 05:43.58 | darkdrgn2k3 | (except remove the 9 requirement) |
| 05:44.22 | p3nguin | Now another ITSP uses phonenumber:secretpass@theirdomain.com/phonenumber |
| 05:44.40 | p3nguin | The phone number is the username. |
| 05:44.49 | darkdrgn2k3 | yeh, not in this case |
| 05:44.50 | p3nguin | The username is the phone number. Whichever way you want to look at it. |
| 05:44.59 | darkdrgn2k3 | but i see what you eman |
| 05:45.06 | darkdrgn2k3 | in my case i have a username username |
| 05:46.28 | p3nguin | Then to call out, I have an extension: exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms) |
| 05:47.42 | p3nguin | So if I dial 18002223333 on my phone, it's really dialing SIP/18002223333@voipms. Where voipms was the context I showed you in the pastebin. |
| 05:48.32 | darkdrgn2k3 | i see |
| 05:48.53 | darkdrgn2k3 | well im 100% use that the route is correct, i set a password for the route and it asked me for it |
| 05:49.28 | p3nguin | It looked like you registered and also peered with them, so there has to be something else going on. |
| 05:50.08 | darkdrgn2k3 | im wondering if its the peer details.. |
| 05:50.08 | the_5th_wheel | Morning all. Can anyone possibly tell me what could cause the transmit clock to be unstable on a Digium E1 Card? |
| 05:50.45 | darkdrgn2k3 | what does "user=phone" mean |
| 05:52.07 | p3nguin | It's a setting that some providers require. |
| 05:52.11 | p3nguin | Mine don't. |
| 05:53.27 | p3nguin | I don't know the actual meaning of it, though. |
| 05:53.44 | darkdrgn2k3 | if my peer details were incorrect |
| 05:53.46 | p3nguin | username= is for the authentication to the outbound proxy. |
| 05:53.51 | darkdrgn2k3 | would it cause this behaviour |
| 05:54.54 | darkdrgn2k3 | ok so whats "fromdomain" |
| 05:54.59 | darkdrgn2k3 | and "fromuser" |
| 05:55.09 | [TK]D-Fender | where is the real debug & configs? |
| 05:55.57 | p3nguin | And there's the debug king, now. :) |
| 05:57.48 | p3nguin | Here's the debug I got from him earlier: http://pastebin.ca/1570492 |
| 05:58.25 | darkdrgn2k3 | and im actualy registered now |
| 05:58.31 | p3nguin | The configs seem to be scattered about. |
| 05:58.49 | darkdrgn2k3 | <PROTECTED> |
| 05:59.01 | p3nguin | I thought it registered ealier, too. |
| 05:59.32 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 05:59.34 | darkdrgn2k3 | p3nguin: it did but not when i lased talked to TK :-S |
| 05:59.37 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:59.38 | darkdrgn2k3 | SIP.CONF -> http://pastebin.com/m479191a7 |
| 05:59.49 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 05:59.54 | [TK]D-Fender | Real call. Don't mask ANYTHING |
| 06:00.37 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:00.43 | [TK]D-Fender | And real configs masking ONLY the password |
| 06:01.14 | p3nguin | I feel like I went down this road once already. :/ |
| 06:02.27 | darkdrgn2k3 | p3nguin: you did, once again dont want to make to much of this info public as it IS a work account :-S |
| 06:03.21 | p3nguin | It's just that playing the Secret Squirrel game bugs the piss outta me and most other people that are genuinely trying to solve a problem. |
| 06:03.56 | [TK]D-Fender | DarkgThey 404 the # you dial. I'm betting they insist on having a "1" in front of the # |
| 06:04.05 | [TK]D-Fender | darkdrgn2kSo try adding it and test |
| 06:04.50 | darkdrgn2k3 | same thing |
| 06:05.02 | [TK]D-Fender | darkdrgn2k3: Show me... |
| 06:05.05 | p3nguin | I don't know everything there is to know about Asterisk and the components, so only receiving tidbits of the configuration here and there wear on me quickly. |
| 06:05.53 | [TK]D-Fender | darkdrgn2kand generally you don't need "realm=company.com" |
| 06:06.05 | p3nguin | That's still in there? |
| 06:06.06 | darkdrgn2k3 | SAME LINK |
| 06:06.09 | darkdrgn2k3 | soryr |
| 06:06.13 | darkdrgn2k3 | and i removed that already |
| 06:06.40 | p3nguin | That was added due to miscommunication on my part. |
| 06:06.40 | darkdrgn2k3 | sorry the only line i removed |
| 06:06.52 | darkdrgn2k3 | and misunderstanding on mine |
| 06:07.05 | p3nguin | It happens. We fix it and move on. |
| 06:08.55 | *** join/#asterisk [netman] (n=netman@216.Red-88-17-240.dynamicIP.rima-tde.net) |
| 06:10.37 | [TK]D-Fender | darkdrgn2k3: I'm not seeing * calling out... |
| 06:10.46 | darkdrgn2k3 | sorry i redid the log again just now |
| 06:11.52 | darkdrgn2k3 | ooo wait |
| 06:11.54 | darkdrgn2k3 | i have an idea |
| 06:12.05 | [TK]D-Fender | darkdrgn2k3: I'm not seeing * calling out... <------- AGAIN |
| 06:12.12 | darkdrgn2k3 | yeh .. |
| 06:12.14 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-zfxwhcogjyyipdra) |
| 06:12.29 | darkdrgn2k3 | dail plan was NXXNXXXXXX only |
| 06:12.57 | darkdrgn2k3 | so dailing the one didnt hit the plan |
| 06:13.04 | darkdrgn2k3 | so is it "1NXXNXXXXXX" |
| 06:13.49 | [TK]D-Fender | that shoud work....of course I didnt' tell you to dial anything different on your PHONE. |
| 06:13.56 | [TK]D-Fender | I jsut said change what you dial to your PROVIDER |
| 06:14.17 | darkdrgn2k3 | um |
| 06:14.29 | darkdrgn2k3 | i dont undersatnd |
| 06:14.37 | darkdrgn2k3 | im using a sip phone to the asterisk server |
| 06:15.13 | darkdrgn2k3 | ok new log is up |
| 06:15.41 | darkdrgn2k3 | same problem |
| 06:15.53 | [TK]D-Fender | darkdrgn2k3: .... * doesn't care what you dial. It matches it and then does whatever you tell it you. You could dial 123 and it will dial 456 because it FEELS LIKE IT |
| 06:16.09 | [TK]D-Fender | darkdrgn2k3: I'm not seeing * calling out... <------- AGAIN |
| 06:16.15 | p3nguin | heh |
| 06:16.16 | [TK]D-Fender | Holy F-ING SHIT |
| 06:16.23 | darkdrgn2k3 | ok im so lost |
| 06:16.56 | [TK]D-Fender | Wait.. |
| 06:16.57 | [TK]D-Fender | there it is.. |
| 06:17.16 | darkdrgn2k3 | is releaved hes not THAT big of a dumbass |
| 06:17.38 | darkdrgn2k3 | i dont know if its relevent |
| 06:18.08 | darkdrgn2k3 | but the to line in the dump i did with a working sip point to the provider didnt have @proxy.oneconnect.com but rather the company domain |
| 06:18.12 | [TK]D-Fender | darkdrgn2kin your peers add :'canreinvite=no" |
| 06:18.26 | [TK]D-Fender | darkdrgn2k3: Apply and LASt try |
| 06:18.39 | [TK]D-Fender | I think this might do it |
| 06:19.30 | darkdrgn2k3 | no luck |
| 06:19.32 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 06:19.37 | darkdrgn2k3 | but what about what i said? think that may be something |
| 06:19.47 | darkdrgn2k3 | the log shows phonenumbeer@proxy.oneconnet.ca |
| 06:19.58 | darkdrgn2k3 | the debug of a working sip phone use phonenumber@company.com |
| 06:20.38 | [TK]D-Fender | ok, too late for me to continue at this point,. |
| 06:20.42 | [TK]D-Fender | maybe tomorrow |
| 06:20.46 | darkdrgn2k3 | 10-4 |
| 06:21.06 | p3nguin | That's a good observation. Try making those changes. |
| 06:21.34 | darkdrgn2k3 | p3nguin: how would i make those changes though? |
| 06:21.46 | darkdrgn2k3 | the only place i specify proxy.oneconnect.ca is in host= |
| 06:22.02 | p3nguin | You have fromdomain=company.com? |
| 06:22.07 | darkdrgn2k3 | yes |
| 06:22.11 | darkdrgn2k3 | is there a todomain? |
| 06:22.46 | p3nguin | I don't think so. |
| 06:23.05 | p3nguin | Do you have domain= ? |
| 06:23.24 | darkdrgn2k3 | no |
| 06:23.40 | p3nguin | Maybe that's why they told you the domain. |
| 06:24.04 | darkdrgn2k3 | no effect :-S |
| 06:24.12 | darkdrgn2k3 | still @proxy.oneconnect.ca |
| 06:24.27 | p3nguin | Are you reloading after each change? |
| 06:24.31 | darkdrgn2k3 | i was reading that domains are used to segregate users |
| 06:24.36 | darkdrgn2k3 | so yeh.. |
| 06:24.41 | darkdrgn2k3 | yep |
| 06:25.33 | p3nguin | Would you be willing to let me try to use the information on my Asterisk box? |
| 06:26.17 | p3nguin | If you feel it's not a good idea, I totally understand. |
| 06:26.56 | darkdrgn2k3 | can i msg u the 411 |
| 06:27.13 | p3nguin | just one second |
| 06:27.42 | p3nguin | Send me a test message to make sure you can get through to me. |
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| 07:02.35 | dustybin | what would the asterisk CLI look like with 50 busy phones connected to it? |
| 07:02.45 | dustybin | imagines a constant stream of data |
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| 08:33.47 | jozza | hi i have a sound problem with ooh323 |
| 08:34.11 | jozza | remote side ooh323 is sending out rtp data on wrong detination port |
| 08:34.57 | jozza | how can i change the rtp ports ooh323 should use |
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| 09:49.30 | jozza | ok, its not this problem |
| 09:49.44 | jozza | one side is not sending audio at all |
| 09:50.15 | elred_ | sound like a nat problem |
| 09:50.25 | jozza | hmm |
| 09:50.42 | jozza | ooh323 uses rtp.conf for audio, doesnt it? |
| 09:50.52 | jozza | sip works fine |
| 09:51.05 | elred_ | do you use reinvite ? |
| 09:51.10 | jozza | no |
| 09:51.13 | elred_ | (canreinvite=yes in sip.conf) |
| 09:51.19 | elred_ | ok, then i don't know |
| 09:51.53 | elred_ | if you don't use reinvite and both legs are able to connect to your Asterisk RTP/SIP port |
| 09:52.06 | elred_ | and you have no firewall rules or others network problem |
| 09:52.11 | elred_ | then it's probably a codec problem |
| 09:52.20 | elred_ | well, maybe try in a first time to change the codec |
| 09:52.20 | jozza | yea, weird |
| 09:52.25 | jozza | i did |
| 09:52.27 | elred_ | to see if it works this way |
| 09:52.31 | jozza | alaw and ulaw |
| 09:52.43 | jozza | the config is: |
| 09:52.58 | elred_ | oh you'r trying to use h323, not sip. sorry |
| 09:53.06 | elred_ | i don't know anything at all with h323 |
| 09:53.27 | jozza | sip phone-->asterisk--inernet-ooh323-->asterisk-->sip phone |
| 09:53.52 | jozza | there is sound in one way left to right |
| 09:54.10 | elred_ | do some sniff on the inernet |
| 09:54.15 | jozza | i checked the tcpdump and nothing is going from one side |
| 09:54.17 | elred_ | maybe there is ICMP reply |
| 09:54.20 | elred_ | ok |
| 09:54.55 | jozza | connect works fine, the call stays connected |
| 09:55.24 | jozza | its weird |
| 09:55.34 | elred_ | nothing is weird, just complex |
| 09:56.01 | elred_ | i am not the man to help you right now, i have no clue to h323 |
| 09:56.06 | *** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net) |
| 09:56.23 | elred_ | but i think if you use some pastebin with a maximum of CLI's debug info and config file others people could better help you |
| 09:56.25 | jozza | ok,thx |
| 10:00.48 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:01.51 | garymc | Yo! |
| 10:12.11 | kaldemar | jozza: do you have a specific reason for using ooh323 between two asterisks? |
| 10:12.17 | jozza | yea |
| 10:12.45 | jozza | production will run between alcatel and asterisk over h323 |
| 10:13.05 | jozza | but i'm first trying to solve it between two asterisks |
| 10:13.32 | jozza | the problem will be the same on both i presume |
| 10:13.51 | jozza | i have a log from ooh323 |
| 10:14.33 | jozza | capabilitytable entry says Unsupported cap type encountered |
| 10:14.40 | jozza | but makes no sense |
| 10:14.55 | jozza | codecs are the same on both sides |
| 10:19.15 | jozza | aha |
| 10:19.17 | jozza | hmm |
| 10:19.45 | jozza | one side is sending traffic out to ip that is local on the other side |
| 10:19.52 | jozza | hmm |
| 10:20.33 | jozza | it is not sending to the external ip of the other side |
| 10:20.42 | jozza | why could that be |
| 10:22.32 | Chainsaw | Gary! ltns. |
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| 10:47.17 | jozza | i dont understand why one side of ooh323 sends rtp data to internal ip of the other side, when it should be sending to its external ip |
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| 10:57.22 | kaldemar | jozza: the signaling messages probabaly have the internal address. |
| 10:57.34 | jozza | yea but why |
| 10:57.50 | jozza | one side is sending correctly to the other side |
| 10:57.56 | jozza | but vice versa not |
| 10:58.17 | kaldemar | are they both behind a nat? |
| 10:58.22 | jozza | yea |
| 10:58.41 | jozza | we are talking about ooh323 |
| 10:58.53 | jozza | just a mention |
| 10:59.03 | kaldemar | is it always the same side that sends to the internal address, regardless of which side initiates the call? |
| 10:59.09 | jozza | yea |
| 10:59.20 | kaldemar | yes, that i figured out already. :) |
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| 11:02.08 | jozza | hmm |
| 11:02.14 | jozza | what could it be? |
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| 11:07.39 | jozza | any idea? |
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| 12:13.43 | jozza | does anyone have an idea why would one side of ooh323 send rtp to an internal ip of another asterisk, when it should be sending it to its external ip |
| 12:14.20 | jozza | only first side does that |
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| 12:15.50 | jozza | rtp debug shows packets being sent to an internal ip of the remote asterisk |
| 12:17.12 | jozza | call control packets work fine |
| 12:17.49 | *** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net) |
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| 12:18.22 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:26.21 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 12:26.41 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
| 12:32.15 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 12:32.34 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 12:33.56 | *** join/#asterisk [8none1] (n=[8none1]@c-76-22-141-39.hsd1.tn.comcast.net) |
| 12:34.17 | garymc | GOD my BT BROABAND HUB 2.0 is a peice of SH*T! |
| 12:34.33 | garymc | you open ports. they stay open for one call then it all messes up! |
| 12:34.48 | garymc | Anyone recommend a decent router? |
| 12:35.21 | garymc | that will open ports UDP 5060 and 10000-20000 and keep them open? |
| 12:35.39 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 12:35.44 | leifmadsen | never had an issue with linksys |
| 12:35.53 | garymc | what model linksys |
| 12:35.55 | garymc | ? |
| 12:36.14 | garymc | they expensive? |
| 12:36.34 | *** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net) |
| 12:37.13 | casnik | your average Linksys wrt54g type router |
| 12:37.23 | casnik | I flash mine with DDWRT |
| 12:37.35 | casnik | so then it becomes a Linux router |
| 12:37.41 | garymc | ? |
| 12:38.12 | casnik | they cost like 40$ in the US |
| 12:38.26 | zamba | yeah, and they're sweet |
| 12:38.45 | zamba | though i flash mine with openwrt |
| 12:38.53 | casnik | or openwrt also |
| 12:38.56 | casnik | yeah |
| 12:39.30 | jozza | confirmed |
| 12:39.36 | jozza | they are sweet |
| 12:42.07 | Naikrovek | yeah they're inexpensive |
| 12:42.32 | zamba | i think i've purchased close to 30 in total |
| 12:42.44 | jozza | hehe, me about 6 |
| 12:42.45 | zamba | the flexibility is awesome |
| 12:42.59 | troffasky | what DSL router would you recommend for running *wrt? |
| 12:43.08 | casnik | garymc, but yeah , you should look into that if your having problems like you describe |
| 12:43.19 | zamba | troffasky: using pppoe? |
| 12:43.20 | Naikrovek | gotta make sure that it'll work with your provider. |
| 12:43.34 | Naikrovek | garymc: do you have DSL or Cable |
| 12:43.35 | troffasky | zamba, no, garymc will be using PPPoA with BT |
| 12:43.48 | garymc | yes PPoA |
| 12:43.52 | garymc | sucks |
| 12:44.02 | Naikrovek | A? what's the A? |
| 12:44.08 | troffasky | ATM |
| 12:44.08 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:44.27 | Naikrovek | asynchronous transfer mode? |
| 12:44.33 | troffasky | the very same |
| 12:44.43 | Naikrovek | over cable? over dsl? |
| 12:45.02 | troffasky | BT only supply DSL for home users, so yes, DSL |
| 12:45.05 | Naikrovek | because linksys routers don't have a modem, they're just a modem |
| 12:45.07 | troffasky | hence my asking which router for DSL... |
| 12:45.11 | Naikrovek | just a router |
| 12:46.01 | Naikrovek | okay so if you have a single device that the phone line goes into and your computer connections come out of, you'll need to get two devices |
| 12:46.10 | Naikrovek | you'll need your own router, and you'll need your own modem |
| 12:46.48 | troffasky | surely there's a device with a DSL interface that can run *WRT? |
| 12:47.00 | afink | would it be ok to install asterisk-addons on a production asterisk box? Am I going to screw anything up by doing so? |
| 12:48.11 | Naikrovek | troffasky: i don't know of any |
| 12:48.18 | Naikrovek | afink: shouldn't have any issues |
| 12:48.25 | Naikrovek | afink: but get a second opinion |
| 12:48.56 | troffasky | I'm guessing if there was it would be based on the AR7 chipset |
| 12:49.04 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 12:49.15 | kaldemar | troffasky: http://wiki.openwrt.org/oldwiki/tableofhardware |
| 12:49.22 | *** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net) |
| 12:49.29 | afink | ok thanks. I'll get a backup b4 I do it |
| 12:52.22 | troffasky | kaldemar, hmm yeah categoryar7device is emtpy, I'll take this on over to #openwrt |
| 12:53.53 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 12:56.13 | kazaa_lite | how can i build a single module instead of rebuilding the whole asterisk? |
| 12:57.36 | Naikrovek | kazaa_lite: why don't you just build the whole asterisk then just copy the module you want into the existing installation |
| 12:57.54 | Naikrovek | that should work as long as the version you're running and the version you're building are the same i think |
| 12:58.14 | kazaa_lite | sorry... should have asked in asterisk-dev |
| 13:03.18 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:03.18 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 13:03.26 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 13:03.59 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:04.18 | *** join/#asterisk Meaty (n=meaty@office.abi.ca) |
| 13:04.52 | Katty | :>>>>>>>> |
| 13:05.06 | Katty | have you looked outside?! |
| 13:05.09 | Katty | it's like paradise!!! |
| 13:05.28 | leifmadsen | Katty: and FRIDAY |
| 13:05.29 | kaldemar | well, not here.. |
| 13:05.41 | Katty | sends sunshine and fluffy white clouds kaldemar's way |
| 13:06.02 | Katty | leifmadsen: oh yes, and friday :> |
| 13:06.40 | kaldemar | is obliged |
| 13:07.11 | Katty | so i watched a really old movie last night called Mega Shark Giant Octopus |
| 13:07.51 | Katty | despite its cheesiness, and awful graphics, i rather liked it. |
| 13:08.10 | Katty | http://www.imdb.com/title/tt1350498/ |
| 13:08.20 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:08.22 | *** join/#asterisk propellerhead (n=yogurt2u@host206.190-30-180.telecom.net.ar) |
| 13:08.25 | Katty | hugs anonymouz666 |
| 13:08.29 | anonymouz666 | hello |
| 13:08.41 | anonymouz666 | Katty !!!! |
| 13:08.54 | Naikrovek | deborah gibson is in that movie |
| 13:08.54 | Naikrovek | that says a lot about it, unfortunately |
| 13:09.08 | Katty | stop being so negative. |
| 13:09.18 | Katty | how can you be negative on such a wonderful morning? |
| 13:09.31 | Naikrovek | it's pretty easy at this point |
| 13:09.38 | Katty | well perk up buttercup! |
| 13:09.40 | Katty | hugs Naikrovek |
| 13:09.45 | Katty | you're gonna bring people down. |
| 13:09.46 | Naikrovek | thanks |
| 13:10.06 | Naikrovek | i'm not going to bring anyone down that wouldn't be down on their own anyway |
| 13:10.32 | Katty | yes you are. |
| 13:10.43 | Katty | just trust me on that one. |
| 13:11.00 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:11.20 | Katty | hi _ShrikE!!! :> |
| 13:11.22 | Katty | hugs on _ShrikE |
| 13:11.22 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 13:12.01 | dustybin | anyone here use: static=yes |
| 13:12.21 | Naikrovek | yeah |
| 13:12.29 | Naikrovek | wait maybe |
| 13:12.30 | dustybin | *no |
| 13:12.33 | dustybin | <PROTECTED> |
| 13:12.45 | dustybin | anyone here use: static=no |
| 13:12.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:13.35 | Katty | hi fender. |
| 13:14.03 | [TK]D-Fender | Katty: Mew. |
| 13:14.04 | _ShrikE | hugs Katty |
| 13:14.42 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:15.11 | *** join/#asterisk matrix1233 (n=Administ@41.226.30.102) |
| 13:15.16 | matrix1233 | hi |
| 13:16.16 | matrix1233 | i have a question |
| 13:16.23 | matrix1233 | i wanna to spy my TRUNK |
| 13:16.35 | matrix1233 | outgoing sip/trunk |
| 13:16.47 | matrix1233 | i cant found the method to do that |
| 13:17.13 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 13:17.49 | Katty | matrix1233: chanspy? (= |
| 13:18.13 | leifmadsen | you want to.... spy.... on a .... channel? :) |
| 13:18.16 | DigitalFlux-AFK | Hi Asteriskians :) |
| 13:18.17 | leifmadsen | use ChanSpy! :) |
| 13:18.22 | leifmadsen | (as per Katty) |
| 13:18.29 | matrix1233 | i want to spy in all my trunk |
| 13:18.37 | leifmadsen | matrix1233: same thing |
| 13:18.42 | DigitalFlux-AFK | I need to ask about the behaviour of Asterisk queues |
| 13:18.49 | anonymouz666 | I don't know in which version we have this fixed but some version of chanspy add delay in the spy |
| 13:18.52 | matrix1233 | so if there are 20 active channel in outgouing i will spy all |
| 13:19.00 | leifmadsen | matrix1233: that is a conference |
| 13:19.05 | kaldemar | matrix1233: they're just channels like any others |
| 13:19.12 | DigitalFlux-AFK | I installed Elastix, added a test queue |
| 13:19.24 | DigitalFlux-AFK | and installed the call center module too |
| 13:19.30 | leifmadsen | I don't know that you can spy on all channels at the same time, other than to create something that is a script, that will attach to all channels, and then put them into a conference bridge that you listen to |
| 13:19.45 | DigitalFlux-AFK | when an agent connects to the queue, then login, he stays on the line |
| 13:19.52 | DigitalFlux-AFK | until someone one cals |
| 13:19.54 | DigitalFlux-AFK | calls * |
| 13:20.01 | DigitalFlux-AFK | this is not the behaviour i want |
| 13:20.20 | DigitalFlux-AFK | i want the agent to login, then when the any one calls, the phone will ring |
| 13:20.21 | kaldemar | hearing 20 conversations is just noice. :) |
| 13:20.26 | DigitalFlux-AFK | is that possibele ? |
| 13:20.43 | leifmadsen | DigitalFlux-AFK: that's because Elastix is using AgentLogin() -- you need to modify it to use different dialplan functions |
| 13:20.48 | [TK]D-Fender | DigitalFlux-AFK: ... |
| 13:20.52 | dustybin | all i need now is people to ring |
| 13:20.52 | [TK]D-Fender | ~freepbx |
| 13:20.53 | infobot | [~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 |
| 13:20.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 13:20.55 | dustybin | ponders |
| 13:21.06 | matrix1233 | chanspy work like that .; 999,chanspy(sip/200) to cans spy the extent 200 ans haow can i spy the trunk ?? i tested sip/trunk but nothin |
| 13:21.08 | leifmadsen | DigitalFlux-AFK: it is possible with Asterisk, but Elastix is doing something you don't want, so you should ask the Elastix people |
| 13:21.28 | leifmadsen | matrix1233: you can't spy on multiple calls simultaneously unless you do some trickery |
| 13:21.30 | Katty | listens to some Eagles. |
| 13:21.50 | Katty | leifmadsen: on a dark desert highway |
| 13:21.57 | dustybin | listens to a dial tone |
| 13:21.57 | leifmadsen | matrix1233: you could attach to SIP/200, and SIP/201, and SIP/203, and use the AMI to send those spied channels to a MeetMe() |
| 13:22.01 | Katty | dances with leifmadsen |
| 13:22.02 | [TK]D-Fender | leifmadsen: channel dump + Originate + MeetMe :) |
| 13:22.10 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 13:22.13 | leifmadsen | [TK]D-Fender: I don't want to meet your Mom! |
| 13:22.23 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:22.34 | [TK]D-Fender | Katty: Next wekk I'm performing "Boys Of Summer"... more likely a Don Henly version vs the Ataris |
| 13:22.38 | [TK]D-Fender | leifmadsen: I just want... |
| 13:22.41 | leifmadsen | ! ! ! |
| 13:22.45 | leifmadsen | \o/ |
| 13:22.50 | [TK]D-Fender | leifmadsen: z0mg HI!! |
| 13:23.15 | leifmadsen | Katty: you can checkout any time you like, but you can never leave ... do do do |
| 13:23.41 | matrix1233 | leifmadsen: i have done a script that originate a liste of number to an IVR and i d'ont success to spy beacuse i don't have a real extention like sip/200 |
| 13:23.50 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 13:24.05 | Katty | leifmadsen: welcome to the hotel california! such a lovely place! |
| 13:24.06 | *** join/#asterisk oej_ (n=olle@132.177.253.250) |
| 13:24.09 | leifmadsen | matrix1233: you will need to be clever in how you solve the problem, but I gave you the general concept of how I would approach it |
| 13:24.57 | dustybin | looks forward to going home and editing some audio for menu |
| 13:25.07 | [TK]D-Fender | matrix1233: problem is you aren't looking at a proper channel list. |
| 13:26.13 | matrix1233 | [TK]D-Fender: yes i have just to spy an active channel on the trunk |
| 13:26.20 | dustybin | has anybody here replaced every single atserisk audio sample with pwn voice? |
| 13:26.27 | dustybin | *own |
| 13:27.05 | [TK]D-Fender | dustybin: module unload godilovethesoundofmyownvoice.so :p |
| 13:27.49 | [TK]D-Fender | dustybin: Actually I've heard from many people that I've got a great voice for TV, IVR's, and advertising |
| 13:27.54 | matrix1233 | leifmadsen: yes i see the concept but i can"t add ani extention to the Meetme beacause i have just the 900 (IVR) and the list on Tel to call |
| 13:28.14 | [TK]D-Fender | dustybin: As I did the menus for someone's IVR I'm hosting, first as a temporary, then left as permanent. |
| 13:28.25 | dustybin | ace |
| 13:28.42 | [TK]D-Fender | matrix1233: huh? |
| 13:28.55 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 13:29.04 | leifmadsen | dustybin: I haven't, but I planned to at some point :) |
| 13:29.05 | dustybin | i need to chop up some audio to create my menu, i have found a nice english female voice |
| 13:29.59 | Katty | oh boy! here we go!!! |
| 13:30.02 | Katty | calls Symantec |
| 13:30.10 | matrix1233 | [TK]D-Fender: i have a script that do an originate to call many number via trunk and to redirect all call to my IVR .. and now i wanna to spy the active call |
| 13:30.22 | Chainsaw | dustybin found Amy? :) |
| 13:30.45 | dustybin | Chainsaw: ICE something website |
| 13:30.59 | *** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk) |
| 13:31.08 | [TK]D-Fender | matrix1233: So like I said, go dump the list of open channels and originate a call to Chanspy and dump them into a MeetMe |
| 13:31.26 | [TK]D-Fender | Chainsaw: If you seek Amy? |
| 13:33.42 | matrix1233 | [TK]D-Fender: thanks i will see what i can do .. |
| 13:33.48 | Katty | this poor symantec guy. |
| 13:33.58 | Katty | i bet he hates me :< |
| 13:34.53 | dustybin | i am using these sounds: http://www.enicomms.com/cutglassivr/ |
| 13:37.17 | Katty | it's going to be alone morning i'm afraid. |
| 13:38.06 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 13:39.07 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:39.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:40.26 | DigitalFlux-AFK | leifmadsen: Sorry man, |
| 13:40.40 | DigitalFlux-AFK | leifmadsen: So i should talk to Elastix guys to change how it handle this |
| 13:40.55 | [TK]D-Fender | DigitalFlux-AFK: Yes, or in #freepbx |
| 13:40.56 | leifmadsen | yep, since the people in here are just #asterisk vanilla people |
| 13:41.19 | [TK]D-Fender | is a Rocky Road #asterisk person.... you'll never go back to vanilla :D |
| 13:41.27 | DigitalFlux-AFK | leifmadsen: Hope they will respond then ! |
| 13:41.30 | leifmadsen | now I want icecream |
| 13:42.15 | DigitalFlux-AFK | leifmadsen: [TK]D-Fender : Okay, thanks guys |
| 13:44.20 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 13:44.28 | *** join/#asterisk pzn (n=pzn@189.35.28.96) |
| 13:45.11 | pzn | In asterisk out-of-the-box configuration, it I call to an extension and it is busy, what should I dial to "request a callback" when the extension is not busy anymore? |
| 13:45.27 | pzn | s/it I call/if I call/ |
| 13:45.28 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:46.44 | afink | all of a sudden I am getting no audio between sip peers. this config has been working for months. only think I did today was reboot a vpn firewall |
| 13:47.00 | afink | think =thing |
| 13:48.10 | [TK]D-Fender | pzn: Out of the box... there is no functionality at all |
| 13:48.30 | *** join/#asterisk Deeewayne (n=dwayne@adsl-072-156-219-201.sip.bna.bellsouth.net) |
| 13:48.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:48.41 | [TK]D-Fender | pzn: It is your job to code absolutely everything in the dialplan yourself |
| 13:49.57 | pzn | [TK]D-Fender: ok, I'll search for how to do this. I tought that the demo that comes with it had this |
| 13:50.44 | afink | On my problem above I went into SIP.conf and changed both peers to qualify=no restarted * then changed them back to qualify=yes and it solved the problem. Is there any thing I can do to keep this from happening? |
| 13:51.11 | [TK]D-Fender | pzn: No... such a thing would require a monitoring script which would be specific to the dialplan and device setups of the system it was written around |
| 13:51.13 | afink | Is there a place where I can set the qualify interval? |
| 13:51.27 | [TK]D-Fender | afink: chan_sip.c |
| 13:51.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:51.45 | afink | ;) um ok.. |
| 13:53.11 | afink | change it then re-compile? |
| 13:53.29 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 13:53.33 | kaldemar | afink: qualifyfreq in sip.conf |
| 13:54.17 | afink | thanks guys |
| 14:00.02 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:02.10 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 14:02.26 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:03.57 | *** join/#asterisk witchdoc (n=witchdoc@unaffiliated/witchdoc) |
| 14:03.59 | witchdoc | hello |
| 14:10.26 | *** join/#asterisk MaliutaLap (n=biteme@220.253.91.219) |
| 14:11.05 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 14:13.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.34 | Katty | bruce springsteen was pretty hot back in the 70s |
| 14:15.46 | Kobaz | okay so |
| 14:15.58 | Kobaz | i have an iax peer that went unreachable two days ago |
| 14:16.27 | Kobaz | i can ping the ip... i can send udp packets on port 4569, and asterisk gets them |
| 14:16.30 | Kobaz | authentication is correct |
| 14:16.41 | Kobaz | i didn't change anything |
| 14:16.54 | Kobaz | iax2 debug doesn't show anything interesting |
| 14:17.08 | Katty | can it call out still? |
| 14:17.18 | Katty | the iax peer |
| 14:20.08 | Katty | who was the lead singer of Guns N Roses? |
| 14:20.27 | IBC_jkenney | who is bright eyed and bushy tailed this morning? |
| 14:20.28 | Faustov | axel rose |
| 14:20.29 | Katty | and does that funny little dance in sweet child of mine. |
| 14:20.29 | IBC_jkenney | dahdi_call: Unrecognized prilocaldialplan TON modifier |
| 14:20.32 | IBC_jkenney | Axel Rose |
| 14:20.45 | Katty | that funny little dance is hilarious. |
| 14:21.03 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
| 14:21.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:21.23 | Amorsen | If any of you is running recent 1.6.x, can you try core show device2extenstate? |
| 14:21.25 | IBC_jkenney | I was hot 10 years ago too whats ur point |
| 14:21.38 | IBC_jkenney | alas now i'm a fat old man |
| 14:21.40 | TSM | does anyone use the polycom 320/330/321/331 phones, i just need to know if they show to the switches as full power devices or if they correctly assign power level, it will determine what switches i will buy |
| 14:21.45 | Katty | IBC_jkenney: we still love you. |
| 14:21.49 | Amorsen | According to the changelog I should have that command, but it doesn't work |
| 14:21.49 | Katty | IBC_jkenney: no matter how old you get. |
| 14:21.54 | Katty | hugs IBC_jkenney |
| 14:22.13 | IBC_jkenney | thanks katty |
| 14:22.26 | Katty | i dated a guy who looked almost exactly like david gilmour in the 60s. |
| 14:22.28 | Katty | it was crazy. |
| 14:23.07 | [TK]D-Fender | TSM: They report based on model, and you want the 321/331 not 320/330. the +1 is a higher RAM/CPU upgrade which will ensure longer supportability |
| 14:23.21 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 14:23.22 | Amorsen | Noone running 1.6? |
| 14:23.34 | IBC_jkenney | ok so i could never get my question answered either the digium lurkers don't know or don't care to answer. And nobody has the dilemma |
| 14:24.23 | coppice | Katty: you must be old. even I was too young to date in the 60s |
| 14:24.26 | IBC_jkenney | Here is my issue i run out of real time mysql DB but want to use hints |
| 14:24.27 | TSM | [TK]D-Fender: yeh i am ordering the 321/331, i just have a few 330s left, i was reading that the 330/320 did not report back to the switch its power profile and so the switches just put full power assignment on that port, the higher model phones do though, this is concerning |
| 14:24.33 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 14:24.44 | Kobaz | Katty: i can call out. using zoiper |
| 14:24.49 | coppice | Katty: or did you mean Dave Gilmour in his 60s? |
| 14:24.53 | Amorsen | Ah, devmode only |
| 14:24.54 | Kobaz | Katty: but the trunk doesn't show as reachable in asterisk |
| 14:24.58 | Katty | http://photos-a.ak.fbcdn.net/photos-ak-sf2p/v288/159/32/227700809/n227700809_433440_2503.jpg <- guy i dated, other girl is not me. |
| 14:25.07 | Katty | http://www.morethings.com/music/pink_floyd/david-gilmour-100.jpg <- david gilmour. |
| 14:25.16 | [TK]D-Fender | TSM: these are the sort of things they correct with firmware updates.... look at the changelogs and you'll see various other models being updated for their reporting |
| 14:25.22 | Katty | coppice: no, david gilmour in his late 20s. |
| 14:25.25 | IBC_jkenney | naa now come on katty where is your photo |
| 14:25.36 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:25.56 | Katty | i don't give them out. |
| 14:26.03 | Katty | cause i get like 20 private messages. |
| 14:26.04 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 14:26.07 | Katty | it's irritating. |
| 14:26.07 | IBC_jkenney | who said give it share it |
| 14:26.13 | leifmadsen | prefers Doug Gilmour |
| 14:26.16 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 14:26.22 | Kobaz | Katty: one asterisk is 1.6.0.15 and one is 1.6.0.13 |
| 14:26.23 | IBC_jkenney | oh is katty just that hot |
| 14:26.27 | Katty | no. |
| 14:26.31 | Katty | i'm a 50 year old male. |
| 14:26.33 | Kobaz | Katty: i have calltoken support disabled |
| 14:26.40 | IBC_jkenney | lol |
| 14:26.58 | Katty | Kobaz: hrmm :< |
| 14:27.05 | coppice | Katty: if you dated Dave Gilmour in the 60s you are older than that |
| 14:27.31 | Katty | i just have a fascinating with David Gilmour. |
| 14:27.31 | Faustov | 50 year old males are definitely not hot! |
| 14:27.32 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:27.32 | Katty | coppice: i did not date david gilmour. |
| 14:27.32 | coppice | I am |
| 14:27.35 | IBC_jkenney | and if you dated Dave Gilmour and are male |
| 14:27.38 | Faustov | notes down to remember to avoid that age |
| 14:27.55 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:27.57 | IBC_jkenney | anyway enough on katty's personal life lol |
| 14:28.05 | Katty | yes. |
| 14:28.07 | Katty | that's quite enough. |
| 14:28.16 | coppice | though I'm hot mostly because the air con is off |
| 14:28.28 | Katty | Kobaz: so let me get this straight. your iax peer cannot dial? or it can dial meetme but not through a dahdi channel? |
| 14:28.29 | Amorsen | creates #asterisk-dating |
| 14:28.30 | chuckf | I knew I should have shown up earlier.... |
| 14:29.09 | chuckf | missed Katty's personal life details:( |
| 14:29.30 | Katty | >.< |
| 14:29.31 | Katty | NEXT!!! |
| 14:29.35 | Katty | help Kobaz. |
| 14:30.33 | Amorsen | Is there a way to see the state of a SIP device directly, instead of the extensionstate shown with core show hints? |
| 14:30.53 | Katty | have you tried binoculars? |
| 14:31.24 | [TK]D-Fender | Amorsen: extensionstate is something completely different. Hints referencing the device is jsut about all there is |
| 14:31.33 | IBC_jkenney | ok now can i get my question answered |
| 14:31.40 | Katty | 8 ball. |
| 14:31.40 | IBC_jkenney | can I run hint from the DB Dialplan |
| 14:31.43 | IBC_jkenney | in real time |
| 14:31.56 | leifmadsen | you mean realtime dialplan? |
| 14:31.58 | leifmadsen | I don't think so |
| 14:32.07 | IBC_jkenney | err |
| 14:32.31 | Katty | i miss file. |
| 14:32.46 | Katty | file: :< |
| 14:34.30 | IBC_jkenney | I wonder if their are any digium lurkers brave enough to present themselves |
| 14:34.31 | IBC_jkenney | lol |
| 14:34.45 | Amorsen | [TK]D-Fender: Ok, but how can I see what the state of the underlying devices is? |
| 14:34.53 | Katty | IBC_jkenney: i would pester Qwell |
| 14:35.02 | Katty | IBC_jkenney: Qwell is not pestered enough. |
| 14:35.13 | [TK]D-Fender | Amorsen: "core show hints" |
| 14:35.42 | IBC_jkenney | I will do that |
| 14:35.48 | Katty | oh man. i need an itty bitty screw driver. :< |
| 14:35.57 | Katty | grumbles. |
| 14:36.09 | Amorsen | Right, but that maps some device states to some extension states. Or the other way around? |
| 14:36.38 | *** join/#asterisk manxpower (n=EWieling@24.42.221.26) |
| 14:37.01 | Katty | hi manx! |
| 14:37.36 | Amorsen | I'm trying to squash bug 15852, or at least find some more information. |
| 14:37.53 | manxpower | Amorsen: try #asterisk-dev? |
| 14:38.12 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:38.15 | Amorsen | manxpower: Good idea |
| 14:39.20 | Kobaz | hmm |
| 14:39.22 | ccesario | hello ... somebody have idea about theses messages ? http://pastebin.com/m62168a92 |
| 14:39.28 | Kobaz | my trunk was reachable for a little while |
| 14:39.30 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 14:39.47 | Kobaz | this is so god damn frusterating |
| 14:40.31 | Katty | hugs Kobaz |
| 14:40.39 | Katty | i am sorry that you are frustrated :< |
| 14:40.56 | Faustov | I just gave away 1000 pounds on futures |
| 14:41.00 | Faustov | do I get a hug too? :> |
| 14:41.14 | Katty | what is this Futures thing |
| 14:41.33 | Kobaz | heh |
| 14:41.59 | Faustov | equity contract |
| 14:42.25 | Kobaz | Katty: in tcpdump i see the iax packets getting to the other box just fine |
| 14:42.42 | Katty | Kobaz: hrmmmm. |
| 14:42.48 | Kobaz | lemme turn off qualify and try to make a call from asterisk |
| 14:42.51 | Katty | sounds very fishy |
| 14:42.52 | Kobaz | calls work fine from zoiper |
| 14:43.01 | Katty | is there natting involved? |
| 14:43.10 | TSM | [TK]D-Fender: well its been some time and seems there is nothing in the changelog about it, i have no proof of this, its just what i read, thats why i was wandering if someone had the real info, im about to buy the switches & phones today |
| 14:43.41 | Kobaz | [Sep 18 10:43:25] WARNING[15992]: app_dial.c:1499 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 14:43.46 | *** join/#asterisk oej_ (n=olle@132.177.253.250) |
| 14:44.45 | Katty | i would have expected no route to destination |
| 14:44.45 | [TK]D-Fender | Kobaz: what do you see int he actual debug? |
| 14:46.32 | Kobaz | [TK]D-Fender: nothing |
| 14:46.36 | Kobaz | [TK]D-Fender: iax2 debug shows nothing |
| 14:46.51 | Kobaz | no packets on port 4569 are going out when dialing either |
| 14:47.01 | [TK]D-Fender | Kobaz: see that the module is indeed loaded |
| 14:47.17 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:47.24 | Kobaz | yeah it's loaded |
| 14:48.14 | jaytee | http://www.collegehumor.com/article:1791636 :-) |
| 14:48.30 | Kobaz | http://pastebin.ca/1570869 |
| 14:48.35 | Kobaz | iax2 show peers... etc |
| 14:48.38 | Kobaz | it's very much loaded |
| 14:50.49 | Katty | stupid question. |
| 14:50.54 | Katty | but you did change zoiper to use 4569, right? |
| 14:51.47 | Kobaz | yeah |
| 14:51.49 | Kobaz | well |
| 14:51.51 | Kobaz | that's the default |
| 14:52.04 | Katty | ^_- |
| 14:52.07 | Katty | looks at her zoiper. |
| 14:52.23 | [TK]D-Fender | Kobaz: Show us |
| 14:52.23 | Kobaz | the crazy thing, is i can make calls with zoiper |
| 14:52.26 | Kobaz | but not with asterisk |
| 14:52.28 | Kobaz | show you what? |
| 14:52.48 | Katty | my zoiper defaults to 5060 |
| 14:52.52 | Katty | oh wait, you're using iax. |
| 14:52.54 | Kobaz | that's sip |
| 14:53.10 | Kobaz | heh |
| 14:53.32 | Katty | try iaxcom real quick |
| 14:53.38 | Katty | eliminate a zoiper problem. |
| 14:54.36 | Kobaz | zoiper problem? |
| 14:54.39 | Kobaz | it works in zoiper |
| 14:55.00 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:55.16 | Naikrovek | i wish my IRC client would color code names differently. All 3 of you have the same color, and two of you both have the same first letter in your name and the same length of name |
| 14:55.31 | Kobaz | heh |
| 14:59.40 | phix | hi |
| 15:00.03 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.36 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:01.53 | manxpower | ccesario: what version of Asterisk are you using? |
| 15:02.07 | Katty | Kobaz: but asterisk can't call zoiper |
| 15:02.21 | ccesario | SVN-branch-1.6.2-r219376M :D |
| 15:02.33 | Katty | Kobaz: right? |
| 15:03.07 | *** join/#asterisk s14ck (n=jtorres@ccscliente156.ifxnetworks.net.ve) |
| 15:03.14 | manxpower | ccesario: Have you considered using a released version of Asterisk? |
| 15:04.33 | ccesario | manxpower, yea. :D ... but openr2 is in 1.6.2 release ... :) |
| 15:05.35 | moy | ccesario: but you can use rc2 |
| 15:05.53 | ccesario | moy, yes I know... :) |
| 15:06.46 | ccesario | but previous versions tha I tested have any problems with transfer :).. and in this release is working perfect :) |
| 15:07.16 | moy | ok, though, as a general rule, debug message are nothing to worry about for users, is just avoiding a deadlock as the message says :) ... any *real* problem or just worried about the msg? |
| 15:07.40 | manxpower | the avoiding deadlock message pretty much means "I'll be crashing soon" |
| 15:08.57 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:09.23 | [TK]D-Fender | <PROTECTED> |
| 15:10.14 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 15:10.31 | Kobaz | yeah |
| 15:10.48 | Kobaz | i'm working on like 50 things |
| 15:11.07 | ccesario | moy, today I get a lot theses messages in full log... and my asterisk was down :/ |
| 15:11.49 | Kobaz | http://pastebin.ca/1570879 |
| 15:11.53 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:13.05 | moy | well, that is a very very very generic debug message that all it says is that some other component in Asterisk is holding the lock for long time ( several milliseconds ). What do you mean by "down"? the process was stopped? |
| 15:13.36 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 15:14.00 | moy | besides, is not an issue to worry to much about since it seems after about 10 retries was able to hold the lock, which means was just delayed more than desirable, but still succeeded |
| 15:14.14 | Kobaz | [TK]D-Fender: i have two boxes with the same peer config |
| 15:14.31 | moy | current hard-coded limit to give up is more than 100 tries, iirc |
| 15:14.35 | Kobaz | [TK]D-Fender: i'm diffing both setups... the only thin different is the call-limit... and the name in [] |
| 15:14.39 | Kobaz | [TK]D-Fender: one works.. one doesn't |
| 15:14.41 | Kobaz | aughh |
| 15:14.52 | Kobaz | lemme do some more tcpdumping |
| 15:15.10 | *** join/#asterisk mumtazah (n=mumtazah@45.100.48.60.wmu01-home.tm.net.my) |
| 15:15.16 | Kobaz | so this has to be some sort of network problem |
| 15:15.31 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:16.03 | *** join/#asterisk s14ck (n=jtorres@200.62.25.156) |
| 15:16.10 | ccesario | moy, the process was running, but the peers dont get make calls ... :/ ... and full log ... with a "lot" theses messages |
| 15:17.05 | ccesario | I enabled DEBUG_THREADS as russellb suggest to ... and I'll try investigate :/ |
| 15:17.13 | moy | yep |
| 15:17.16 | moy | that's the best to do |
| 15:17.35 | ccesario | until this momment no "locks" |
| 15:18.49 | Zeeek | lox and bagels |
| 15:20.18 | Kobaz | [TK]D-Fender: http://pastebin.ca/1570882 iax2 debug on the remote end |
| 15:20.28 | *** join/#asterisk oej (n=olle@132.177.253.250) |
| 15:20.32 | Kobaz | when placing a call from the box that can't |
| 15:22.17 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 15:22.40 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 15:23.40 | *** part/#asterisk matrix1233 (n=Administ@41.226.30.102) |
| 15:25.50 | Zeeek | It's Friday: In 30 minutes the VoIP Users Conference (was Asterisk Users Conference). Call in and join us. Numbers and SIP URI at http://VUC.me #voip-users-conference on IRC. |
| 15:26.37 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 15:28.01 | ayeso | Im having an issue that periodically appears, I have an * server all SIP, on occasion it will stop releasing calls, if I do a `sip show channels` I will see channles in the BYE state that will not release, has anyone seen this\ have any advice? |
| 15:28.15 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:28.18 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:28.28 | *** join/#asterisk rofcorp (n=rfletche@190.140.107.122) |
| 15:29.18 | rofcorp | where asterisk saves the extensions sip numbers? |
| 15:29.37 | kaldemar | extensions are in extensions.conf by default |
| 15:30.09 | rofcorp | freepbx saves the extensions in that file? |
| 15:30.21 | kaldemar | no, i don't think so. |
| 15:30.38 | kaldemar | for freepbx questions, #freepbx will be a better place |
| 15:30.45 | jaytee | I believe it's users.conf |
| 15:30.51 | rofcorp | exellent, thanks |
| 15:31.02 | dustybin | what does this do: |
| 15:31.10 | dustybin | extenpatternmatchnew= |
| 15:31.10 | kaldemar | users.conf, extensions? |
| 15:32.17 | kaldemar | it's a new algorithm for pattern searching. |
| 15:32.35 | kaldemar | see extensions.conf.sample for more |
| 15:34.06 | Kobaz | [TK]D-Fender: i fixed it |
| 15:34.16 | Kobaz | [TK]D-Fender: the router at the other end had a bad mac entry for the host |
| 15:34.35 | [TK]D-Fender | Kobaz: yuck... |
| 15:34.44 | [TK]D-Fender | Kobaz: horrible |
| 15:35.17 | Zeeek | and for VoIP Users COnference questions, use #voip-users-conference |
| 15:35.36 | Kobaz | i knew something was crazy somewhere |
| 15:35.39 | Kobaz | i didn't change any configs |
| 15:35.42 | Zeeek | come to Astricon and meet all the great VUC peeps |
| 15:35.55 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 15:36.12 | [TK]D-Fender | ayeso: Do they pose a real problem? I'm presuming that there is no actually call channel open for them... |
| 15:36.55 | dustybin | pattern searching in a dial plan? i cannot figure that out |
| 15:37.23 | kaldemar | dustybin: did you read the sample conf? |
| 15:38.42 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:41.27 | Katty | ever get really really naseous all of a sudden for no particular reason |
| 15:41.36 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 15:41.36 | Katty | ugah |
| 15:42.50 | dustybin | kaldemar: yes, it made no sense |
| 15:42.58 | rofcorp | there is a chat room related to skype for asterisk? |
| 15:43.00 | wcselby | anyone here using bandwidth.com as a sip trunk provider with asterisk? |
| 15:44.58 | *** join/#asterisk smth (n=mike__@199.84.137.3) |
| 15:45.02 | giovani | `/join #freeswitch |
| 15:45.07 | giovani | whoops :) |
| 15:46.10 | wcselby | omg |
| 15:46.11 | dustybin | is it worth opening yourself up to the skype network? |
| 15:46.12 | IBC_jkenney | anyone use the SPA962's is there a good help guide for population of the directory in a full deployment env |
| 15:46.19 | wcselby | CIO at my client is describing his intestinal issues |
| 15:46.20 | kaldemar | dustybin: well, chances are that i don't make any more sense, but: it's an algorithm for finding the right extension when a call lands in the dialplan. the old one isn't that great performance wise, so a new one was made. does that satisfy you? |
| 15:46.32 | *** join/#asterisk shinao1 (n=shinao1@41.219.226.125) |
| 15:47.09 | dustybin | kaldemar: its either the right extension or not ? |
| 15:47.12 | dustybin | 1 or 0 |
| 15:47.16 | dustybin | no pattern matching? |
| 15:48.02 | kaldemar | it's the operation of choosing the right extension. |
| 15:48.02 | smth | hey, how to have asterisk to generate ring back tone (response 180 not 183) before answer the channel? |
| 15:48.32 | dustybin | kaldemar: i see, this is something to do with the engine |
| 15:48.39 | dustybin | maybe make it work faster? |
| 15:48.49 | kaldemar | exactly. |
| 15:48.55 | dustybin | ok |
| 15:49.03 | dustybin | is it worth turning it on |
| 15:49.41 | kaldemar | look at the speedup table and judge for yourself. and notice that it may be unstable. |
| 15:50.20 | dustybin | eeeeek |
| 15:50.25 | dustybin | hates unstable |
| 15:50.35 | dustybin | leaves unstable for ubuntu users :P |
| 15:51.28 | ayeso | [TK]D-Fender: Once it starts no calls are released so the active call count grows untill * crashes |
| 15:51.44 | Naikrovek | stupid pharmacist. I'm male. MUST you put pregnancy warnings on the bottle that cover up important information on the label? EVERY time they do this. |
| 15:52.18 | ayeso | How can I get the output of 'sip set debug on' to go to the log file? I have: messages => notice,warning,error,debug in logger.conf... no luck |
| 15:52.53 | smth | [TK]D-Fender, how to have asterisk to generate ring back tone (response 180 not 183) before answer the channel and dial out? |
| 15:53.18 | kaldemar | ayeso: grab it with other means, such as script or something else that saves stdout to a file. |
| 15:53.57 | smth | ayeso, asterisk -rx "sip debug" > log.txt |
| 15:54.14 | ayeso | smth: Ill give it a try, thanks. |
| 15:54.40 | ayeso | kaldemar: Id have to write an expect script for that with what im trying to do... not a problem just want an easier way. |
| 15:56.46 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 15:57.56 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 15:58.19 | RoyCrowder | what kind of load does the Asterisk MySQL CDR module put on the IO, etc. Also, what about over the network to a MySQL db on another machine? |
| 15:58.43 | RoyCrowder | over using the standard CDR text file |
| 15:58.59 | Naikrovek | RoyCrowder: very little. |
| 15:59.26 | RoyCrowder | so on a production machine with 200 simultaneous calls, little to no difference? |
| 15:59.31 | Naikrovek | it does one, maybe 2 inserts at the end of each call, that's it |
| 15:59.57 | jdnWEST | Question, anyone know if there is a way to have SSL/encryption or any other security on a SIP trunk to vitelity or another carrier? |
| 16:00.07 | RoyCrowder | so is it recommended over the text file method? |
| 16:00.34 | troffasky | CDR is generated on events so it's call setup/teardown that will determine DB load |
| 16:00.37 | Naikrovek | RoyCrowder: if you have requirements for a lot of custom searches, DB is recommended i think yes |
| 16:00.42 | troffasky | what do you want to do with the CDR? |
| 16:01.01 | RoyCrowder | billing primarily and portal access to customers to view calls, etc |
| 16:01.01 | Naikrovek | jdnWEST: if you set up a VPN, yes. I don't think SIP handles encryption natively. |
| 16:01.05 | *** join/#asterisk bradleyprice86 (n=bradleyp@74.11.1.34) |
| 16:01.08 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 16:01.18 | troffasky | RoyCrowder, hell yeah you want DB rather than text! |
| 16:01.26 | RoyCrowder | roger that, thanks much |
| 16:02.01 | Naikrovek | RoyCrowder: there are single system MySQL servers that can handle 10,000 inserts per second. |
| 16:02.03 | *** join/#asterisk SebastianS (n=schu@adsl-dyn14.78-98-105.t-com.sk) |
| 16:02.42 | Naikrovek | your 200 simultaneous calls system will make mysql use less than 1% of CPU and 1% of network capacity, or some other small number |
| 16:02.43 | jdnWEST | slaps RoyCrowder around a bit with a large trout |
| 16:03.48 | jdnWEST | Naikrovek: is it normal for sip providers to offer ipsec? |
| 16:04.02 | Qwell | jdnWEST: some require it |
| 16:04.12 | Naikrovek | jdnWEST: there's your answer |
| 16:06.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:07.02 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 16:07.34 | bmoraca | so, i've got a bit of a connundrum |
| 16:08.27 | bmoraca | i have a paging speaker that's sip driven that is configured to automatically answer any call that comes in...i need to play a sound to that while another set of phones is ringing, like an external ringer |
| 16:08.52 | bmoraca | any ideas? |
| 16:09.38 | Naikrovek | you have a sip speaker, you ring a paging group, and you want it to play a sound while other phones make their sound |
| 16:09.58 | bmoraca | somewhat |
| 16:10.00 | Naikrovek | k |
| 16:10.01 | Naikrovek | no idea |
| 16:10.03 | Naikrovek | heh |
| 16:10.20 | bmoraca | the key that makes it difficult is that the paging speaker will always automatically answer the call, and there is no way to change that behavior. |
| 16:10.48 | Naikrovek | and the other phones ring once then answer? or you have to wait for people to pick up |
| 16:10.55 | bmoraca | no |
| 16:11.01 | bmoraca | the other phones ring until people pick them up |
| 16:11.10 | Naikrovek | ah |
| 16:11.51 | troffasky | so, you want to play a sound to the paging speaker, then stop once another extension is picked up? |
| 16:11.54 | bmoraca | i'm thinking i can use local channels, but i'm not sure how to keep the sip channel alive while the local channel answers |
| 16:11.57 | Naikrovek | oh so you may have someone who should answer the phone away from their desk, and you want the speaker to tell them when the phone rings via the sip speaker |
| 16:12.05 | bmoraca | troffasky: yes |
| 16:12.12 | troffasky | effectively making the paging speaker into a great big ringer for the other phones? |
| 16:12.17 | bmoraca | indeed |
| 16:12.32 | bmoraca | i'd like to do it without using AGI... |
| 16:12.34 | manxpower | why not spend the money to get a real external ringer |
| 16:12.54 | troffasky | because he's using asterisk, which is flexible enough to work with what he has :-) |
| 16:13.10 | bmoraca | manxpower: because they have the paging speaker already for paging purposes, and it doesn't really make sense to have to have two speakers mounted in their shop area |
| 16:14.08 | troffasky | can we assume the paging speaker doesn't have an option to just ring built into it? |
| 16:14.28 | bmoraca | yes |
| 16:14.45 | bmoraca | because it does not, and the company that makes it has no desire to put that feature in (i've asked many times) |
| 16:15.08 | troffasky | I guess they want you to buy one of their ringers and install that too |
| 16:15.20 | bmoraca | they don't make a ringer, strangely |
| 16:15.22 | troffasky | have you looked at the Playback() command? |
| 16:15.34 | bmoraca | yes, but that doesn't exactly do what i need it to do |
| 16:15.43 | bmoraca | unless i create another call channel of some kind |
| 16:18.07 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 16:18.49 | *** part/#asterisk shinao1 (n=shinao1@41.219.226.125) |
| 16:23.39 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 16:26.11 | bmoraca | hmmm |
| 16:26.14 | bmoraca | well, i'm halfway there |
| 16:27.08 | bmoraca | i can get the speaker to answer and play tt-monkeys while my ring group is ringing...but how to shut it off when the call is answered... |
| 16:27.35 | manxpower | use an agi to use the manager interface to kill the channel? |
| 16:28.43 | bmoraca | hrm |
| 16:28.59 | bmoraca | when the sound stops playing, the paging speaker picks up the call...that's not good |
| 16:31.13 | *** part/#asterisk manxpower (n=EWieling@24.42.221.26) |
| 16:31.15 | *** join/#asterisk torrancew (n=torrance@70.164.20.45) |
| 16:31.59 | torrancew | hi all, i'm a sysadmin, and was just told that our company would like a pbx server, and they'd like me to implement. i know asterisk is what i want, but i've never configured it or any other pbx before. |
| 16:32.07 | *** join/#asterisk jtodd (i=zoebh1h7@ns.fox-den.com) |
| 16:32.07 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:32.15 | torrancew | can anyone point me at a good guide? |
| 16:32.20 | Qwell | ~book |
| 16:32.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:32.21 | Katty | infobot: thebook |
| 16:32.21 | infobot | somebody said thebook was 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://astbook.asteriskdocs.org |
| 16:32.47 | torrancew | cool |
| 16:32.53 | torrancew | what about a list of compatible hardware? |
| 16:32.54 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
| 16:32.58 | torrancew | any good reference point on that |
| 16:33.39 | Qwell | torrancew: just about anything SIP, and anything supported by DAHDI (such as Digium hardware) |
| 16:33.56 | torrancew | and what about voice modems for pstm? |
| 16:33.59 | torrancew | pstn* |
| 16:34.17 | Qwell | You don't use voice modems. You use telephony hardware (which is where DAHDI comes in) |
| 16:34.36 | torrancew | oh ok |
| 16:34.41 | torrancew | excellent |
| 16:34.45 | torrancew | thanks for the quick help guys |
| 16:34.47 | Katty | idly wonders what a voice modem is |
| 16:34.54 | torrancew | i'm sure i'll be back once this starts to get implemented |
| 16:34.58 | *** join/#asterisk dymaxion (n=dymaxion@host86-172-50-54.range86-172.btcentralplus.com) |
| 16:36.42 | Katty | prods backup exec. Run Faster Please |
| 16:40.51 | dymaxion | wonders if it's better NOT to use Asterisk GUI and just use the command line? what think ye? |
| 16:41.19 | jaytee | ever since we replaced our PowerVault DLT Tape Libraries with e-sata removable drive units we gained a significant speed improvement in backups and restores with Backup Exec |
| 16:42.02 | *** join/#asterisk [netman] (n=netman@216.Red-88-17-240.dynamicIP.rima-tde.net) |
| 16:46.43 | *** join/#asterisk errotan (n=errotan@5403E60D.catv.pool.telekom.hu) |
| 16:48.21 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 16:48.23 | *** join/#asterisk zamba (i=marius@flage.org) |
| 16:49.03 | [TK]D-Fender | jaytee: DLT? ancient... what rates were you getting? |
| 16:49.57 | jaytee | [TK]D-Fender, glacial....don't recall the actual througput figures of MB/sec but it was sooooo slow |
| 16:50.54 | Naikrovek | I have a d2d2t system that is goofy fast |
| 16:51.07 | Naikrovek | disk to disk to tape, for those who don't kn ow |
| 16:51.44 | [TK]D-Fender | jaytee: I get about 1800mb/s with my LTO3 & BE |
| 16:51.54 | Naikrovek | nice |
| 16:51.57 | Xetrov` | how do you have your ddt setup? |
| 16:52.13 | [TK]D-Fender | Naikrovek: I want to cache my backups the same to have a "last backup" on HD for local speed and mirroring |
| 16:52.45 | Naikrovek | cybernetics device, which is a SAN, and I use Windows shadow copy to keep 30-days of backup. then I use the tape to backup the shadow copies |
| 16:52.59 | Naikrovek | all of this via MS data protection manager |
| 16:53.06 | Naikrovek | which is very nice |
| 16:53.17 | Naikrovek | but i could have used Amanda probably |
| 16:53.33 | Kobaz | i thought only bug fixes where supposed to go in the 1.6.0.x tree |
| 16:53.49 | Naikrovek | so users can restore any file they delete, or view previous versions for 30-days back all by themselves |
| 16:53.59 | Naikrovek | if they want to go back farther i gotta bust out the tapes |
| 16:55.22 | jaytee | yeah, we use shadow copy and can go back 30 days |
| 16:55.42 | Naikrovek | it's an extremely handy feature, the shadow copies |
| 16:55.56 | Naikrovek | some nut here thought it would be a good idea to rearrange the folder structure |
| 16:55.58 | jaytee | one of MS's better ideas |
| 16:56.00 | Naikrovek | deleted gigs of stuff |
| 16:56.16 | Naikrovek | jaytee: not sure it was their idea but their implementation is very good |
| 16:56.47 | Naikrovek | and the shadow copies use very little disk space |
| 16:56.58 | bmoraca | how can i get asterisk to treat quotation marks as literals? |
| 16:57.03 | Naikrovek | i could probably go back a year, if i could store more than 64 FS snapshots |
| 16:57.22 | bmoraca | for instance, exten => s,1,System(/usr/sbin/asterisk -rx 'originate SIP/200 extension PR-200@pager-ringer-ring') doesn't work because asterisk does not pass the quotation marks |
| 16:57.47 | Naikrovek | bmoraca: you may have to escape them, not sure. instead of ' use \' |
| 16:58.04 | Naikrovek | bmoraca: firmware works great, btw |
| 16:58.10 | bmoraca | good |
| 16:58.23 | Naikrovek | no features that my users would notice so i'm not in a hurry to upgrade them |
| 16:59.59 | bmoraca | excaping the quotes worked |
| 17:00.03 | Naikrovek | yay me |
| 17:00.14 | bmoraca | and external ringer works as well |
| 17:00.15 | bmoraca | woo |
| 17:00.24 | Naikrovek | nice |
| 17:00.28 | bmoraca | five lines |
| 17:01.42 | *** join/#asterisk Arun_Krishnakuma (n=chatzill@dxb-b126613.alshamil.net.ae) |
| 17:01.50 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 17:03.02 | *** part/#asterisk Arun_Krishnakuma (n=chatzill@dxb-b126613.alshamil.net.ae) |
| 17:03.25 | dustybin | I need to figure out some basics of extensions.conf |
| 17:03.47 | Naikrovek | what you wanna know |
| 17:03.55 | Naikrovek | i'm not a font of extensions.conf knowledge but maybe i can help |
| 17:04.00 | dustybin | is [global] a context what is static? |
| 17:04.09 | dustybin | ie. its for global configuration of the dial plan? |
| 17:04.35 | Naikrovek | you've reached the limit of my extensions.conf experience. good luck with that. |
| 17:04.39 | Naikrovek | heh |
| 17:04.50 | dustybin | Naikrovek: please dial again using the # key |
| 17:05.41 | dustybin | [globals] and [general] what is the difference? |
| 17:06.12 | bmoraca | dustybin: general is just a context...globals are global dialplan variables that are constant throughout all contexts |
| 17:06.31 | dustybin | right i see, so [general] is something just made up by the user |
| 17:06.36 | bmoraca | yep |
| 17:06.43 | dustybin | right...... I seeeeeee |
| 17:06.43 | Naikrovek | see now i knew that |
| 17:06.51 | Naikrovek | contexts can be named just about anything |
| 17:07.00 | dustybin | apart from [globals] |
| 17:07.09 | Naikrovek | there are keywords |
| 17:07.13 | Naikrovek | that you can't sue |
| 17:07.14 | wonderworld | is there a way to run an extension on asterisk startup? i want to set some variable ${DB(blah/blah)} ? |
| 17:07.16 | Naikrovek | characters you can't use |
| 17:07.21 | dustybin | so logically, you would put, autofallthrough=yes in [globals] ? |
| 17:07.40 | [TK]D-Fender | wonderworld: AstDB is persistent. |
| 17:08.09 | *** join/#asterisk Arun_Krishnakuma (n=Arun_Kri@dxb-b126613.alshamil.net.ae) |
| 17:08.26 | *** part/#asterisk Arun_Krishnakuma (n=Arun_Kri@dxb-b126613.alshamil.net.ae) |
| 17:08.33 | dustybin | is [default] also a made up context by the user? |
| 17:08.48 | bmoraca | dustybin: every context is just made up |
| 17:08.58 | dustybin | bmoraca: apart from [globals] !! |
| 17:09.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:09.13 | bmoraca | globals isn't necessarily a context, though. |
| 17:09.20 | dustybin | it looks like one!!! |
| 17:09.25 | dustybin | how would i know? |
| 17:09.31 | bmoraca | no it doesn't...the syntax is entirely different |
| 17:09.39 | dustybin | [globals] is used by the application for global settings of the dial plan |
| 17:10.01 | dustybin | bmoraca: yes i understand that, but if you see [globals] on its own, its hard to know |
| 17:10.10 | wonderworld | [TK]D-Fender: tnx |
| 17:10.10 | bmoraca | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 17:10.23 | wonderworld | [TK]D-Fender: even if asterisk crashes? |
| 17:10.25 | dustybin | question no.2 |
| 17:10.33 | dustybin | if i put everything in a [context] |
| 17:10.47 | [TK]D-Fender | wonderworld: Yes |
| 17:10.53 | dustybin | would you need to put something outside of the [context] so it gets activated? |
| 17:11.10 | [TK]D-Fender | dustybin: "activated"? huh? |
| 17:11.35 | dustybin | does asterisk start at the top of extensions.conf and run downwards |
| 17:11.42 | p3nguin | Sounds like a marketing term |
| 17:11.52 | dustybin | i thought things in [context] can only be activated if required |
| 17:12.01 | dustybin | ie |
| 17:12.05 | p3nguin | Extensions are matched, not activated. |
| 17:12.34 | dustybin | exten => 12345678,n(public),GotoIfTime(07:00-23:30|mon-sun|*|*?open,s,1) |
| 17:12.51 | dustybin | [open] context will be activated if its open |
| 17:12.53 | p3nguin | If you dial a number and it matches an extension, Asterisk runs whatever command or application it has been programmed to run. |
| 17:12.56 | Naikrovek | yeah, a lookup table of sorts is created i think. so when a call somes in for an extension, matching rules are found, and the call is passed through those rules in the order specified |
| 17:12.56 | dustybin | otherwise it will not get activated |
| 17:13.13 | p3nguin | There is no "activation" |
| 17:13.19 | p3nguin | It either matches or it doesn't. |
| 17:13.22 | dustybin | ok |
| 17:13.24 | [TK]D-Fender | dustybin: No, that extension's APPLICATION will only jump if it matches |
| 17:13.29 | *** part/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:13.30 | dustybin | ok |
| 17:14.21 | [TK]D-Fender | dustybin: it isn't a switch. this is programming 101. You do an "IF" in any language and you are expressly doing the actions you determine in the "THEN" part attached to it... |
| 17:14.21 | p3nguin | If you call 12345678 between 07:00-23:30 on mon-sun, you will jump to the [open] context to the s extension, priority 1. |
| 17:14.34 | dustybin | right it makes sense now |
| 17:14.50 | [TK]D-Fender | p3nguin: Assuming that line wasn't the only one in that context ;) |
| 17:14.50 | dustybin | asterisk can match with: context=BLAH from the sip.conf or iax.conf |
| 17:14.58 | dustybin | thats what creates the match |
| 17:15.07 | [TK]D-Fender | dustybin: No. |
| 17:15.09 | p3nguin | no |
| 17:15.17 | [TK]D-Fender | dustybin: * LOOKS in the context you tell it to |
| 17:15.20 | dustybin | ok |
| 17:15.27 | dustybin | this makes more sense now |
| 17:15.33 | p3nguin | The contexts for phones basically tells Asterisk where to start looking. |
| 17:15.34 | [TK]D-Fender | dustybin: And on match process that exten there |
| 17:15.42 | dustybin | extensions.conf doesnt do anything on its own |
| 17:15.57 | p3nguin | Other than set variables, correct. |
| 17:16.12 | dustybin | it can only perform tasks if sip.conf or iax.conf have been told to LOOK at a certain context |
| 17:16.35 | [TK]D-Fender | dustybin: Some channel has to care about it. |
| 17:16.40 | p3nguin | Your peers (phones) have to start looking in a context. |
| 17:16.42 | dustybin | yep |
| 17:16.46 | dustybin | i seeee |
| 17:16.58 | dustybin | 1) iax / sip |
| 17:17.03 | *** join/#asterisk ezzadin (n=esharafi@modemcable138.151-70-69.static.videotron.ca) |
| 17:17.05 | dustybin | 2) then look at dialplan |
| 17:17.21 | wonderworld | dustybin: you can include contexts into others to twist your brain even more :) |
| 17:17.33 | dustybin | eeek |
| 17:17.39 | [TK]D-Fender | dustybin: channel drive configs (zip.conf, iax.conf, etc) configure devices so * can get and/or places calls to/from them. Calls incoming on those devices and mathing an appropriate entry will land on the context you specified. |
| 17:17.48 | p3nguin | For example, all my phone peer contexts have dialplan contexts of "phones". There's a [phones] context in extensions.conf, and that is where matching will begin when I dial a number. |
| 17:18.10 | dustybin | i seeee |
| 17:18.24 | ezzadin | Hi all.. first time here... I'm having an issue with loading Digium TE100P board... |
| 17:18.32 | dustybin | so a channel is a entry in either sip or iax config? |
| 17:18.42 | p3nguin | I don't actually have any extensions listed in the [phones] context, but I do have some includes to other contexts that will match things dialed. |
| 17:18.47 | dustybin | i have 1 channel what speaks to my VOIP provider |
| 17:18.55 | Naikrovek | man lots of people come in with digium board problems... |
| 17:18.55 | dustybin | i have another channel what speaks to my polycom |
| 17:18.58 | Naikrovek | ezzadin: what's the problem |
| 17:19.32 | wonderworld | dustybin: and you will probably have a context defined like "from-pstn" "incoming" "sipprovider" or whatever. this context is for people calling you. It tells asterisk what to do, depending on the number that has been called. |
| 17:19.58 | dustybin | yes indeed |
| 17:20.01 | [TK]D-Fender | dustybin: No. You have a peer entry that can be used to call your phone if it's registered, and can auth calls coming frmo it. |
| 17:20.09 | dustybin | ok |
| 17:20.16 | p3nguin | dustybin: I read the O'Reilly asterisk book and understood all about the contexts within a few hours. |
| 17:20.28 | dustybin | :( |
| 17:20.35 | p3nguin | You should read it, too. |
| 17:20.41 | dustybin | p3nguin: i didnt find it clear at all |
| 17:20.41 | Naikrovek | oh let's not pick on him, christ |
| 17:20.42 | wonderworld | it's fun to read actually |
| 17:21.02 | Naikrovek | dustybin: print that chapter out, or buy the book, and just look at it every time you take a dump |
| 17:21.03 | [TK]D-Fender | dustybin: a channel is an active call between * and a device in progress. Do not confuse this with the device's entry in a config file |
| 17:21.04 | ezzadin | I have recently installed 1.4.23.1 with dahdi 2.1 and freepbx 2.5... I have all the setting.. so it should work just fine... each time I use /etc/init.d/dahdi start it just loads the dummy and not the board |
| 17:21.09 | Naikrovek | it'll be clear in no time |
| 17:21.12 | wonderworld | i like the foreword. it's so true. |
| 17:21.17 | dustybin | Naikrovek: LOL |
| 17:21.25 | [TK]D-Fender | looks for some more nits |
| 17:21.32 | Naikrovek | nits? |
| 17:21.34 | Naikrovek | lice? |
| 17:21.43 | p3nguin | nits in which to pick, I presume. |
| 17:21.51 | wonderworld | nuns in tight skirts? |
| 17:21.55 | dustybin | what is the difference between friend and peer ? are there other names? |
| 17:21.58 | dustybin | type= ? |
| 17:22.04 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 17:22.05 | [TK]D-Fender | wonderworld: Bad habit :) |
| 17:22.17 | [TK]D-Fender | is just swimming in puns today |
| 17:22.31 | Naikrovek | dustybin: user= call maker, peer = call recipient, friend = both |
| 17:22.32 | Naikrovek | ... i think |
| 17:22.46 | hesco | If I know a ${CHANNEL}, how do I test to see if there is still an active call on it? |
| 17:22.48 | wonderworld | i trashed a PAP2T today and attached the faxes directly to the line. |
| 17:22.54 | dustybin | Naikrovek: so i could call everything 'friend' ? |
| 17:23.00 | wonderworld | i did try for a day to make it work |
| 17:23.01 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.112.210) |
| 17:23.07 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.112.210) |
| 17:23.34 | wonderworld | no way |
| 17:23.40 | p3nguin | Straight from the sip.conf commenting: |
| 17:23.41 | p3nguin | ; type = user a device that authenticates to us by "from" field to place calls |
| 17:23.45 | p3nguin | ; type = peer a device we place calls to or that calls us and we match by host |
| 17:23.48 | p3nguin | ; type = friend two configurations (peer+user) in one |
| 17:23.57 | wonderworld | probably there is one, but i didn't find out |
| 17:24.20 | Naikrovek | ezzadin: hang out in #freepbx and ask there maybe. hang out in here also and ask again when the channel dies down |
| 17:24.26 | [TK]D-Fender | dustybin: in 1.4+ this started changing. peer vs user use IP vs username for authe, etc |
| 17:24.27 | Naikrovek | ezzadin: those in the know are busy nitpicking |
| 17:24.50 | [TK]D-Fender | hesco: huh? |
| 17:25.36 | [TK]D-Fender | ezzadin: modprobe t100p |
| 17:25.36 | p3nguin | sip show channel <channel> maybe? |
| 17:25.40 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 17:25.42 | [TK]D-Fender | ezzadin: modprobe dahdi |
| 17:25.50 | [TK]D-Fender | ezzadin: dahdi_cfg -vvvv |
| 17:26.17 | dustybin | i have read this 15 times, and still [FAIL] to understand: |
| 17:26.18 | dustybin | type = user a device that authenticates to us by "from" field to place calls |
| 17:26.41 | ezzadin | modporbe t100p doesnt work... I tried modprobe wct1xxp but no luck |
| 17:26.56 | hesco | [TK]D-Fender: I'm writing a dialer application, the channel for the agent is stored in the database. If the agent is still on the last call, I don't want to dial, but once they get off the phone, I do want to dial. So I need to test if their bridged call is still active. |
| 17:27.02 | p3nguin | dustybin: What are you trying to configure, phone or trunk? |
| 17:27.05 | dustybin | the communcations between asterisk and my VOIP provider = peer ? |
| 17:27.15 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 17:27.18 | dustybin | the communcations between asterisk and my polycom = friend ? |
| 17:27.23 | ezzadin | this is what I get when I do dahdi_cfg DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 17:27.29 | dustybin | p3nguin: phone |
| 17:27.36 | ezzadin | I see the channels of course... 1 to 23 and D chan 24 |
| 17:27.48 | [TK]D-Fender | hesco: show the DB |
| 17:27.52 | p3nguin | dustybin: Use friend for the phone. You will want to make and receive calls. |
| 17:27.58 | Naikrovek | ezzadin: you may get faster response by calling digium |
| 17:28.19 | [TK]D-Fender | ezzadin: I seem to recall that the T100/E11 we removed from DAHDI. They may only be supported by Zaptel |
| 17:28.20 | hesco | ~paste |
| 17:28.21 | infobot | methinks paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 17:28.30 | [TK]D-Fender | ezzadin: call for confirmation |
| 17:28.30 | wonderworld | hesco: you could grep from "sip show channels" |
| 17:28.42 | wonderworld | if you find him, he is still on the line |
| 17:28.54 | [TK]D-Fender | E100* |
| 17:29.00 | p3nguin | if you know the channel, why not sip show channel <channel>? |
| 17:30.01 | wonderworld | even better... |
| 17:30.09 | ezzadin | # Digium E100P: PCI single-port E1 |
| 17:30.09 | ezzadin | wct1xxp |
| 17:30.32 | ezzadin | this is in /etc/dahdi/modules |
| 17:30.39 | hesco | [TK]D-Fender: http://bin.cakephp.org/view/1479800303 |
| 17:30.42 | [TK]D-Fender | ezzadin: and when you try to modprobe it? |
| 17:30.43 | ezzadin | so it should support it ..no? |
| 17:30.53 | hesco | channel | SIP/hugh_desk-08424ff0 |
| 17:30.56 | ezzadin | it work without any issue |
| 17:31.08 | ezzadin | modprobe wct1xxp and modprobe dahdi |
| 17:31.11 | ezzadin | return no error |
| 17:31.34 | [TK]D-Fender | hesco: No, the ASTERISK source that shows you their status |
| 17:31.47 | *** join/#asterisk Matrix_Maddy (n=Arun_Kri@dxb-b126613.alshamil.net.ae) |
| 17:31.47 | [TK]D-Fender | ezzadin: then dahdi_cfg -vvvv |
| 17:32.41 | wonderworld | ezzadin: do an lsmod | grep wct1 to verify, that the module is loaded |
| 17:33.01 | hesco | not sure I follow what you mean by: 'ASTERISK source that shows you their status' |
| 17:33.18 | *** part/#asterisk Matrix_Maddy (n=Arun_Kri@dxb-b126613.alshamil.net.ae) |
| 17:33.21 | hesco | p3nguin: *CLI> sip show channel SIP/hugh_desk-08424ff0 |
| 17:33.22 | hesco | No such SIP Call ID starting with 'SIP/hugh_desk-08424ff0' |
| 17:33.24 | ezzadin | [TK]D-Fender: when I do cfg -vvv I get all 24 channels, but at the end I get this error: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 17:33.55 | hesco | is that what you mean? |
| 17:33.58 | ezzadin | this is result of lsmod: # lsmod | grep wct1 |
| 17:33.58 | ezzadin | wct1xxp 12772 0 |
| 17:33.58 | ezzadin | dahdi 182564 2 dahdi_dummy,wct1xxp |
| 17:34.31 | wonderworld | hesco: thats how you could find out if that person is still using his phone |
| 17:34.44 | hesco | I can do that from the *CLI>; but am not sure how to call that from a perl script. |
| 17:35.00 | [TK]D-Fender | hesco: What is supposed to poll on-call status? How is it supopsed to associate a device to an "agent" (No, I don't trust any stock definition for what you may be meaning) |
| 17:35.07 | wcselby | wow |
| 17:35.08 | hesco | the perl script runs as root from the agi-bin/ |
| 17:35.21 | wcselby | some company is rerouting a/c ducts in our server / switch rooms |
| 17:35.22 | wonderworld | hesco: perl_system_command("asterisk -rx \"sip show channels $user\""); |
| 17:35.24 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 17:35.38 | wcselby | somehow they unplugged the UPS that controls half the switches on our 6th floor |
| 17:35.39 | [TK]D-Fender | hesco: AMI, or asterisk -rx "some command" |
| 17:35.40 | hesco | thank you wonderworld. |
| 17:35.46 | wonderworld | i am not sure how to run system commands from perl. |
| 17:35.55 | hesco | I can handle that part |
| 17:36.03 | Naikrovek | wonderworld: backticks or perl system() function |
| 17:36.07 | wcselby | we find out because someone comes and says "hey all the phones are dead upstairs" |
| 17:36.40 | wonderworld | hesco: i think the hard part would be to check that often enough |
| 17:36.47 | hesco | My database notes which channel an agent is connected with, per the db table I pasted. |
| 17:36.52 | Naikrovek | wcselby: fire that person who unplugged it. of course no one will fess up |
| 17:36.53 | wonderworld | are you doing a threaded script or will you use cronjobs? |
| 17:37.12 | hesco | But the agent is supposed to stay on the line, while the dialer finds the next number to bridge |
| 17:37.45 | wonderworld | well, cron wouldn't be possible.. |
| 17:37.49 | hesco | so the real question I guess is: is the channel to the last connected phone number still active |
| 17:39.45 | hesco | I'm spawning a forked process on every start up of the dialer, I think, not sure, that was last week's issue. |
| 17:39.57 | wcselby | Naikrovek - outside company brought in to do the work. They said "oh yeah I think we may have bumped that cable" but they won't fess to anything more than that |
| 17:40.00 | wonderworld | ok, it should work then with sip show channel if you already have the current channel for the user |
| 17:40.03 | hesco | I would have to restudy the code to be sure. |
| 17:40.14 | wcselby | of course, there were five angry techs and a VP of the company staring him down |
| 17:40.25 | wonderworld | i was wondering about how you would trigger those checks |
| 17:40.56 | Naikrovek | effing cowards. i always do what I can to fire people who don't own up to their mistakes |
| 17:41.07 | Naikrovek | if you admit the mistake then you've reached my good side |
| 17:41.23 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 17:41.35 | wonderworld | i am not sure if it would perform well in perl if you check that for many agents in a "one check per agent" way |
| 17:41.47 | wcselby | Naikrovek - i'd prefer someone who admitted their mistake over someone who covers it up |
| 17:42.05 | [TK]D-Fender | I'd prefer someone who can FIX the mistake quietly and quickly... |
| 17:42.16 | [TK]D-Fender | Or better still, the one that doesn't MAKE them in the first place... |
| 17:42.25 | Naikrovek | wcselby: exactly |
| 17:43.17 | Naikrovek | [TK]D-Fender: everyone makes mistakes |
| 17:43.29 | Naikrovek | but better yet are those that make mistakes, find them, and fix them all on their own |
| 17:43.54 | [TK]D-Fender | Naikrovek: pffft... MORTALS.... |
| 17:43.54 | Naikrovek | because then you know you've got a person who is learning AND working. people who make no mistakes don't do a lot of work |
| 17:45.38 | wcselby | no audio on a sip call - nat issue? |
| 17:45.45 | wcselby | generally, I mean |
| 17:45.46 | Naikrovek | most likely |
| 17:45.48 | Naikrovek | yes |
| 17:46.00 | Katty | what is ACORN? |
| 17:46.07 | Naikrovek | usually the problem is unidirectional though |
| 17:46.19 | Katty | and don't tell me Association of Community Organizations for Reform Now |
| 17:47.09 | wcselby | Katty - a group that's in a lot of trouble for some things their workers have been caught on tape doing |
| 17:54.59 | Naikrovek | i need to find a graphic designer. the in-house one we have is terrible. but he knows how to crop in photoshop so he is all "I'm the graphic designer." he also pretends to be the accountant, the receptionist, the newsletter copywriter, and only guy who knows anything in the entire office |
| 17:55.22 | Naikrovek | he's terrible at everything he does |
| 17:55.25 | Naikrovek | wtf |
| 17:55.50 | p3nguin | Everything? Well, mom always said not to limit myself to just one field. |
| 17:56.04 | Naikrovek | well if he were good at ONE of the things maybe he'd be tolerable |
| 17:56.33 | Naikrovek | he designed this: http://www.autorobinc.com/ "that's the best I've ever done. I'm really proud of that one." |
| 17:56.56 | wonderworld | lol |
| 17:57.53 | wonderworld | it always looks better when you use a nice framework and template |
| 17:58.17 | Naikrovek | we need new logos (two companies here) just new graphical design theme in general. I need a designer that can show me something, unpaid, so I can go to mgmt and say "yo, our guy is terrible, we need to hire this guy." |
| 17:58.17 | wonderworld | it's plain stupid to do this by hand |
| 17:58.52 | Naikrovek | he spent like 2 weeks trying to figure out a JS incompatability between IE and firefox. I told him at least 5 times that people have figured these thigns out, and to use google. "no, this one is new. I've never seen it before." |
| 17:59.38 | wonderworld | i suck at the gfx part. so i stopped doing it. |
| 17:59.46 | Naikrovek | see i wish he'd do that. |
| 17:59.52 | Naikrovek | he sucks, but doesn't know it |
| 18:00.00 | wonderworld | don't be too cruel |
| 18:00.10 | Naikrovek | oh i wish i could show you our newsletter |
| 18:00.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:00.13 | Naikrovek | it is painful |
| 18:00.14 | p3nguin | naikrovek: Get in touch with Michael Allton (Lake St. Louis) - http://www.stadiastudio.com/ |
| 18:00.20 | Naikrovek | thanks, p3nguin |
| 18:01.09 | Katty | i'm hearing things. randomly, i think my phone is ringing |
| 18:01.13 | Katty | phantom rings |
| 18:01.29 | p3nguin | Not sure if he'll fit your needs, but it can't hurt to lay out what you're looking for and see if he's interested. |
| 18:01.40 | Naikrovek | Katty: my phones do that if you miss a call and don't clear it out. every like 4 minutes it'll sound like it's ringing for 0.2 seconds |
| 18:01.49 | Naikrovek | p3nguin: yeah. |
| 18:01.51 | Katty | well mine doesn't do that |
| 18:01.55 | Katty | i'm hearing things. literally. |
| 18:02.00 | Naikrovek | p3nguin: check these out: http://abduzeedo.com/tags/logo-design |
| 18:02.46 | Katty | needs another vacation, possibly scuba diving. |
| 18:03.04 | wonderworld | i worked at a company a while ago. the boss did the website. they sell hardware. they didn't install any shop software. the boss is updating the "shop" without ordering function by hand in html. |
| 18:03.31 | Naikrovek | oh man |
| 18:03.33 | wonderworld | that's really sick. they still do it |
| 18:03.45 | wonderworld | it's full of 404's, typos etc |
| 18:03.49 | Naikrovek | yeah |
| 18:03.49 | wonderworld | and it's a lot of text |
| 18:04.08 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a9ae:d5ea:f6d8:8c57) |
| 18:04.11 | Naikrovek | you can always spot the mgmt quality and/or company budget via its website |
| 18:04.32 | Naikrovek | well it seems that way anyway |
| 18:04.47 | wonderworld | actually you can spot if the boss is surfing the web or not |
| 18:05.03 | wonderworld | if he would, he would have realized, that there are better ways to do it |
| 18:05.32 | wonderworld | some managments with people 40+ just missed the net/web |
| 18:05.51 | friartuck | Naikrovek I worked for a crap company that had a sweet website.... |
| 18:05.57 | Naikrovek | lol yeah when i worked at yahoo, we were trying to sell advertisements to aol in australia |
| 18:06.15 | Naikrovek | I went with a sales guy and they told us (this is 2001, btw) that they wanted to see what other companies were doing |
| 18:06.50 | Naikrovek | they AOL(!) sent emails to each other by sharing a single phone line and aksing if anyone was dialled in |
| 18:06.59 | Naikrovek | in 2001 |
| 18:07.00 | wonderworld | lol |
| 18:07.01 | Naikrovek | in sydney |
| 18:07.13 | Naikrovek | no servers, no ethernet, no admin |
| 18:07.17 | Naikrovek | AOL! |
| 18:07.23 | Katty | has anyone ever heard of the Poseidon hotel? |
| 18:07.25 | wonderworld | dialing into AOL? |
| 18:07.42 | Naikrovek | no, this WAS AOL in australia. |
| 18:07.44 | Naikrovek | their offiec |
| 18:07.46 | Naikrovek | office |
| 18:07.49 | wonderworld | i mean the AOL office guys. were they dialing into AOL? |
| 18:07.52 | Naikrovek | yes |
| 18:07.55 | wonderworld | hehe |
| 18:07.57 | friartuck | nice |
| 18:08.02 | Naikrovek | they had a SINGLE dial up line |
| 18:08.04 | wonderworld | AOL sucked, sucks, will suck |
| 18:08.07 | Naikrovek | some 40 people shared it |
| 18:08.08 | friartuck | that was their "VPN" |
| 18:08.18 | dustybin | check out my secure menu: http://paste.debian.net/46934/plain/46934 |
| 18:08.31 | Naikrovek | a SINGLE dial up line.. "is anyone logged in" tick tick tick tick "john is!" |
| 18:09.19 | wonderworld | i don't know if they did it in the US as well. in germany they spammed everyone with their free CD'S. everyone had 5 or 6 of them. |
| 18:09.28 | wonderworld | SO environmently unfriendly |
| 18:09.31 | Naikrovek | yes |
| 18:09.32 | ayeso | AGI is bad on a production system |
| 18:09.33 | Naikrovek | i had dozens |
| 18:09.48 | dustybin | ayeso: im using it at home, why is it bad? |
| 18:10.08 | Naikrovek | dustybin: it can't handle more than a few simultaneous connections, from what i've heard |
| 18:10.18 | dustybin | right ok |
| 18:10.32 | wonderworld | what would be the right way to do it? have some program talk to asteisk via AMI? |
| 18:10.59 | Naikrovek | apparently people write proxies. proxy maintains a single connection to AGI, then other AGI applications connect to the proxy |
| 18:11.10 | Naikrovek | someone told me they wrote one in C |
| 18:11.13 | Naikrovek | but i forget who |
| 18:11.28 | ayeso | dustybin: If you use a server as a proxy for RTP (peers cannot reinvite) and you have alot of traffic, the agi scripts will have to fight for CPU time to run, they are seperate processes. this will cause your audio to sound like shit.. use fastagi instead of learn to code in C. |
| 18:12.09 | dustybin | eeek |
| 18:12.14 | ayeso | s/instead of/instead or/ |
| 18:12.50 | ayeso | thank you infobot. |
| 18:12.50 | infobot | ayeso: my pleasure |
| 18:12.56 | dustybin | well, this is for me only, nobody else |
| 18:13.01 | ezzadin | [TK]D-Fender: there are not interrupts.. I also called Digium and drivers for the board is still included... |
| 18:13.18 | ayeso | dustybin: youll be cool then, just keep it in mind for the future, you can really screw yourself that way. |
| 18:13.26 | dustybin | ok :D |
| 18:13.33 | ezzadin | does it have anything to do with the fact that I'm using FreePBX |
| 18:18.13 | wonderworld | whats fastagi? agi that uses less resources? |
| 18:18.57 | Naikrovek | i believe it's an AGI proxy as I described earlier. let's see what infobot says: |
| 18:18.59 | Naikrovek | ~fastagi |
| 18:19.00 | infobot | fastagi is probably Implements (agi://) over TCP socket using port #4573 by default. |
| 18:19.02 | [TK]D-Fender | wonderworld: JFGI :) |
| 18:19.17 | wonderworld | thank you infobot |
| 18:19.17 | infobot | de rien, wonderworld |
| 18:19.20 | wonderworld | hehe |
| 18:19.31 | wonderworld | ok, so it's agi over the network? |
| 18:19.45 | wonderworld | like running the scripts on another machine... |
| 18:20.11 | wonderworld | yes. it is. that's cool |
| 18:20.54 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 18:20.56 | wonderworld | but buying an extra server for your agi-scripts is not very cost-effective. |
| 18:21.51 | hesco | well from my tests, `my $channel_report = system("asterisk -rx \"sip show channel $AGI{'channel'} \"");` returns only 0, for success, to the scalar, $channel_report. I'm going to need more to go on, I'm afraid. But I'm late for my conference call so will return to this later. |
| 18:24.33 | wonderworld | hesco: try "asterisk -rx blah" in BASH first to make sure it's working at all. |
| 18:25.41 | wonderworld | maybe there is another command in perl for getting back output from a system command? sorry, don't know perl. |
| 18:25.52 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 18:28.42 | [TK]D-Fender | wonderworld: Oh? Certainly can be.. |
| 18:29.03 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 18:29.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:29.04 | [TK]D-Fender | wonderworld: What happens when the task you want to places a nasty load on the server? |
| 18:29.29 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 18:29.38 | wonderworld | sorry, didn't understand... |
| 18:30.37 | witchdoc | wonderworld: asterisk -rx is for "dirty hacks" - if you want an API better the agi or the manager interface to use |
| 18:30.38 | kaldemar | wonderworld: qx, iirc |
| 18:30.42 | ayeso | wonderworld: its more cost efective than crashing asterisk on your production box with AGI. |
| 18:31.06 | kaldemar | you don't get output with system in perl, if i'm not completely wrong |
| 18:31.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:31.51 | wonderworld | i have been doing it for an app. it works fine but i am sure it's not the best way to to it. |
| 18:32.00 | kaldemar | yes, qx. don't use system. |
| 18:32.11 | wonderworld | but that box is 95% idle all the time. |
| 18:32.27 | kaldemar | system only issues the command, it doesn't return any output. |
| 18:32.33 | witchdoc | wonderworld: hey, if the box is *not* idle, then its broken - if you use asterisk |
| 18:32.47 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.137) |
| 18:32.51 | *** join/#asterisk Matrix_Maddy (n=chatzill@dxb-b126613.alshamil.net.ae) |
| 18:33.12 | witchdoc | wonderworld: with high load (cause of many context switches or whatever) i got asterisk systems with load of 26 |
| 18:33.19 | *** join/#asterisk mvanbaak_ (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:33.37 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.137) |
| 18:34.06 | *** join/#asterisk csmyth (n=csmyth@ext-52.sagetelecom.net) |
| 18:34.25 | wonderworld | i am still looking for the perfect asterisk hardware. should be small and very stable. best would be fanless, 1 pci slot, no moving parts at all for a fair price. |
| 18:34.55 | [TK]D-Fender | wonderworld: Would you like fries with that, sir? |
| 18:35.26 | p3nguin | When using Asterisk, if I have a low number of phones doing a moderate amount of calling... if I multiply the number of phones in use by, say, 10 fold, does Asterisk's memory usage, processor usage, or both increase? |
| 18:35.40 | witchdoc | wonderworld: hmm. asus pundits - two pci or pci-e ports, very cheap |
| 18:35.58 | wonderworld | witchdoc: i'll have a look. thanks. |
| 18:35.58 | [TK]D-Fender | p3nguin: Of course... |
| 18:36.14 | witchdoc | wonderworld: but not fanless |
| 18:36.20 | p3nguin | [tk]d-fender: I think you missed the point of the question. |
| 18:36.31 | *** join/#asterisk misteranonymous (n=mister@udp115943uds.hawaiiantel.net) |
| 18:36.43 | p3nguin | I'm wanting to know: RAM, CPU, both |
| 18:37.01 | witchdoc | p3nguin: for what kind of pstn? |
| 18:37.08 | p3nguin | witchdoc: none |
| 18:37.10 | wonderworld | thats not so bad. it NEEDS to be reliable because people are used to it, that phones just work. they worked for the last 100 years. now if you sell them a pbx and that goes down, they have no tolerance. |
| 18:37.12 | [TK]D-Fender | p3nguin: Both |
| 18:37.14 | p3nguin | SIP/IP only |
| 18:37.15 | [TK]D-Fender | p3nguin: Both |
| 18:37.41 | wonderworld | they tolerate a windows box crashing once a day, but they EXPECT a phone to work. |
| 18:37.44 | [TK]D-Fender | p3nguin: Calls take ram & CPU. Instructions and data to store & process. Of course its "both" |
| 18:37.55 | Naikrovek | p3nguin: both will increase, but i don't think it's precicely linear. |
| 18:38.07 | misteranonymous | "jabber test" yields "'Unable to find client 'asterisk'! |
| 18:38.07 | misteranonymous | " i don't have asterisk as client in jabber.conf anyone know what might be up? |
| 18:38.29 | witchdoc | wonderworld: yes, pbx customer ar not stressless |
| 18:38.45 | [TK]D-Fender | wonderworld: My new work server locks out the PRI every 2 hours +/- and I have a monitoring script to check for the overruns and restart things that checks every 2s to minimise downtime and automate getting back up |
| 18:38.51 | wonderworld | thats why i am looking for a stable little box that makes not too many problems |
| 18:39.34 | [TK]D-Fender | Time/Last Error [14:36:59 / 14:36:55] Error Count [397] | |
| 18:39.37 | witchdoc | [TK]D-Fender: i had one problem box, with zombie sip channels after a while and monitor it with nagios |
| 18:39.38 | [TK]D-Fender | \o/ |
| 18:40.00 | [TK]D-Fender | Since 14:00 which is the laster reset. when the overruns hit 2000+ the script restarts things |
| 18:40.18 | witchdoc | uhhhh |
| 18:40.40 | witchdoc | [TK]D-Fender: witch PRI hardware you use? |
| 18:41.04 | [TK]D-Fender | witchdoc: its the PCI bus on thi s bleeding edge new Dell R610.... |
| 18:41.35 | [TK]D-Fender | witchdoc: &^$#ing PITA and I'm very close to doing a roll-back to old hardware on Sunday night. I would be bald were I to be a hair-puller |
| 18:41.54 | [TK]D-Fender | as it is I'm near burnout which fortunately means that I'm too tired do blast people... |
| 18:42.39 | witchdoc | [TK]D-Fender: i use hp proliant server and sangoma hardware. works well. no interrupt problems and now fax and echo fear |
| 18:43.19 | witchdoc | [TK]D-Fender: get some free time, thats better for you, really. |
| 18:44.00 | AeroCloud | Question: When using DIAL() within AGI, is there a way to check if an IAX2 connection is available before doing the dial? I want to loop the IAX2's before sending a dial |
| 18:44.31 | witchdoc | p3nguin: the question is the concurrent calls on this, and if you must translate codecs. |
| 18:45.33 | witchdoc | p3nguin: if you have 20 sip clients a celeron 1.?ghz and 1gb ram works well with asterisk 1.4 |
| 18:45.38 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 18:47.46 | AeroCloud | anyone know ^ how would I know if all trunks on that particular IAX2 gateway is busy? |
| 18:48.57 | [TK]D-Fender | AeroCloud: probably by the DIALSTATUS returned by your dial |
| 18:49.11 | wonderworld | hmmm.. iax2 show channels maybe ? |
| 18:49.13 | AeroCloud | I tried dialstatus, but it returns -1 when checking on IAX2/NUMBER |
| 18:49.34 | AeroCloud | it only works for the current AGI channel they used to dial in on |
| 18:49.39 | [TK]D-Fender | AeroCloud: Impossible. DIALSTATUS doesn't return a number |
| 18:49.49 | AeroCloud | it did for me, watch |
| 18:50.02 | [TK]D-Fender | AeroCloud: and AGI isn't a channel, and IAX2 doesn't carry "trunks" |
| 18:50.03 | p3nguin | waits for aerocloud to pull a rabbit from his hat |
| 18:50.14 | AeroCloud | let me find my logs |
| 18:50.15 | [TK]D-Fender | p3nguin: That trick again? that'll never work! |
| 18:50.40 | [TK]D-Fender | Nothing up my sleeve |
| 18:50.42 | [TK]D-Fender | ... |
| 18:50.44 | [TK]D-Fender | PRESTO!!!! |
| 18:50.51 | [TK]D-Fender | gets eaten by a lion |
| 18:51.00 | AeroCloud | AGISCRIPT.php: This is a test: -1 |
| 18:51.06 | AeroCloud | the -1 is what it returned |
| 18:51.07 | Naikrovek | that shows nothing |
| 18:51.12 | Naikrovek | pastebin the code |
| 18:51.22 | [TK]D-Fender | AeroCloud: thats what AGI returned which means precisely nothing |
| 18:51.32 | [TK]D-Fender | AeroCloud: that is not the DIALSTATUS VARIABLE |
| 18:51.34 | AeroCloud | $chan_status = $agi->channel_status($dial_out_usa); |
| 18:51.34 | AeroCloud | <PROTECTED> |
| 18:52.02 | Katty | wow. |
| 18:52.08 | Katty | for about 250 million dollars, i can buy an island. |
| 18:52.16 | Katty | what a bargin! |
| 18:52.23 | AeroCloud | thats the code, 2 lines ^ |
| 18:52.31 | [TK]D-Fender | AeroCloud: meaningless. |
| 18:52.56 | AeroCloud | if I do not specify the $dial_out_usa, which is IAX2/numbers |
| 18:53.03 | AeroCloud | then it returns my current channels status |
| 18:53.09 | AeroCloud | that part works perfectly |
| 18:53.26 | AeroCloud | but trying to get a status on a trunk seems impossible |
| 18:53.32 | [TK]D-Fender | AeroCloud: And unless that's a DeadAGI then it'll likely always report "up" |
| 18:53.41 | [TK]D-Fender | AeroCloud: and that is not the result of a DIAL. |
| 18:53.46 | AeroCloud | its AGI, not DeadAGI |
| 18:53.55 | [TK]D-Fender | AeroCloud: There is no "status" execpt as a returned result for a Dial() |
| 18:54.00 | AeroCloud | I need it to know if anyone else is using the trunk |
| 18:54.05 | AeroCloud | before I send a dial to it |
| 18:54.16 | [TK]D-Fender | AeroCloud: "core show application chanisavail" |
| 18:54.25 | AeroCloud | once I do dial, if all trunks are busy, it hangs up |
| 18:54.38 | AeroCloud | ok will look ^ |
| 18:54.38 | [TK]D-Fender | AeroCloud: It only hangs up if you coded it to. |
| 18:55.00 | AeroCloud | so I should put a loop around the dial ? |
| 18:55.08 | [TK]D-Fender | AeroCloud: If Dial reports back that it couldn't dial, call processing continues. if it stops, its because you didn't give it anything more to do |
| 18:55.24 | [TK]D-Fender | AeroCloud: You need to determine for yourself what actions you want to take |
| 18:55.52 | [TK]D-Fender | AeroCloud: we aren't here to guess what you want, only to advise paths for you to acheive them |
| 18:55.58 | AeroCloud | while ($status = "BUSY") { do crap; } ? |
| 18:56.30 | AeroCloud | ok I will do a test.. and see if this works.. appreciate the help :) |
| 18:57.05 | [TK]D-Fender | while ($status = "whatever") { do whatever you feel like doing in a loop } |
| 18:58.01 | AeroCloud | so $var = $agi->exec("Dial"); ? |
| 18:59.15 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 18:59.20 | AeroCloud | $chan_status = $agi->exec('Dial',"$dial_out/$target,-1,oL(".($max_minutes*60000).",60000,30000)M(custom-cdr)"); |
| 18:59.20 | AeroCloud | <PROTECTED> |
| 19:00.36 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:01.53 | dustybin | why is this required in my settings for my phones: requirecalltoken=auto |
| 19:01.58 | dustybin | if i dont put that it errors |
| 19:02.01 | [TK]D-Fender | AeroCloud: No, that is not the DIALSTATUS CHANNEL VARIABLE |
| 19:02.19 | [TK]D-Fender | AeroCloud: And I gave you another app so you aren't actually placing calls to await failure |
| 19:02.52 | IBC_jkenney | i have a soundpoin 7000 sitting on my desk that i was told has to be provisioned in 1 hour for a conference call |
| 19:02.57 | IBC_jkenney | no ida how to go about it |
| 19:03.26 | IBC_jkenney | is it the same as the 550's |
| 19:03.44 | [TK]D-Fender | IBC_jkenney: for everythign that matters, yes. |
| 19:03.55 | [TK]D-Fender | IBC_jkenney: consider of course the # of line-keys |
| 19:04.02 | *** join/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 19:04.42 | Kobaz | Remote-Party-ID: "foo" <sip:@192.168.15.137>;privacy=full;screen=no |
| 19:04.49 | Kobaz | what does privacy=full mean? |
| 19:05.06 | pittmodal | hello all. I've heard that asterisk uses slinear audio internally - what is the default bit and sample rate? From some of the audio that I'm getting it looks like 8k/32bit. Is this configurable? |
| 19:05.47 | [TK]D-Fender | pittmodal: 8k/16bit, not configurable |
| 19:05.48 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 19:06.10 | [TK]D-Fender | pittmodal: and it uses it as an intermediary for transcoding / conferencing, etc |
| 19:06.53 | CrashSys | Anyone know if there's a way to get Zaptel 1.4.12.1 to compile on Kernel 2.6.31? |
| 19:07.37 | witchdoc | CrashSys: uhhh - no. |
| 19:08.03 | CrashSys | awww, come on man, I was hoping for a miracle... |
| 19:08.06 | [TK]D-Fender | CrashSys: upgrade time... |
| 19:08.07 | dustybin | [TK]D-Fender: why havent the whole of public PSTN upgraded the sound to 16-bit ? |
| 19:08.26 | [TK]D-Fender | CrashSys: Well... I can make you disappear.... |
| 19:08.28 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:08.31 | dustybin | PSTN are using 8-bit sounds in 2009 ... odd |
| 19:08.32 | CrashSys | claps |
| 19:08.32 | pittmodal | [TK]D-Fender: thanks for the info |
| 19:08.34 | AeroCloud | Fender: |
| 19:08.36 | AeroCloud | $x = 0; |
| 19:08.36 | AeroCloud | <PROTECTED> |
| 19:08.36 | AeroCloud | <PROTECTED> |
| 19:08.36 | AeroCloud | <PROTECTED> |
| 19:08.36 | AeroCloud | <PROTECTED> |
| 19:08.37 | AeroCloud | <PROTECTED> |
| 19:08.37 | [TK]D-Fender | makes room on his shelf next to Jimmy Hoffa... |
| 19:08.39 | AeroCloud | <PROTECTED> |
| 19:08.41 | AeroCloud | better? |
| 19:08.51 | [TK]D-Fender | AeroCloud: No... |
| 19:08.54 | CrashSys | you are missing a ' |
| 19:08.57 | [TK]D-Fender | AeroCloud: Do not spam like that in here |
| 19:09.06 | AeroCloud | sorry figured 6 lines was ok |
| 19:09.15 | AeroCloud | 7 ^ |
| 19:09.35 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:09.50 | CrashSys | Guess I have to start testing newer versions of asterisk and see which one bombs the least... and I had just gotten 1.4.21.2 mature with all the patches :( |
| 19:10.16 | pittmodal | is there a way to discover the bit/sample rate of an audio file from the command line? can sox do it? |
| 19:10.34 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:10.48 | pittmodal | nevermind. "file" seemed to do it just fine |
| 19:10.51 | spck | anyone have a solution for the changing number of columns returned from a REALTIME query? apparently anything that is null doesn't get returned, creating a variable number of columns |
| 19:12.13 | *** join/#asterisk AbsurdHero (n=chatzill@c-24-99-221-202.hsd1.ga.comcast.net) |
| 19:13.48 | dustybin | 'One of the major advantages of IAX2 over SIP is the IAX2 jitterbuffer' |
| 19:13.50 | dustybin | interesting |
| 19:14.55 | CrashSys | sounds like old documentation |
| 19:15.20 | p3nguin | I'm pretty sure that SIP has a jitterbuffer, too. |
| 19:15.28 | wonderworld | i prefer the no-NAT advantage |
| 19:15.28 | CrashSys | 1.4+ has jitterbuffer |
| 19:15.29 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:15.59 | p3nguin | wonderworld: Can you explain what you mean by that? |
| 19:16.25 | CrashSys | SIP is not designed to traverse a NAT, IAX is... |
| 19:16.44 | dustybin | will SIP eventually be phazed out? |
| 19:16.49 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 19:16.49 | ayeso | spck: if your talking about the realtime querys in the source code, the (SENTINAL) argument is what is telling it to do a "select *". |
| 19:16.51 | CrashSys | Probably not |
| 19:16.54 | dustybin | why not |
| 19:17.15 | dustybin | there is no logic to SIP, IAX wins |
| 19:17.18 | CrashSys | Because it's already the incumbent standard? |
| 19:17.23 | dustybin | hmm ok |
| 19:17.47 | ayeso | dustybin: I disagree. |
| 19:17.55 | dustybin | why? |
| 19:18.06 | CrashSys | So until you convince the Verizon's and Level3's and Global Crossings of the world to use something other then sip you are probably going to be stuck with it for a long time... |
| 19:18.53 | *** join/#asterisk Inode (n=Inode@unaffiliated/inode) |
| 19:19.09 | wonderworld | p3nguin: sure, you have to traverse NAT with IAX, but you have less problems with it |
| 19:19.15 | ayeso | ^^ and many other things.. |
| 19:19.34 | dustybin | SIP is full of [FAIL] |
| 19:19.48 | ayeso | dustybin: Maybe for you. |
| 19:20.00 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 19:20.03 | CrashSys | The RTP part of SIP is full of fail :) |
| 19:20.13 | CrashSys | SIP for what it is isn't too bad... |
| 19:20.30 | Naikrovek | if it were done on a single port it would have been better i think |
| 19:20.58 | p3nguin | Weird. I use only SIP on a daily basis. |
| 19:20.59 | dustybin | Naikrovek: if it was done on a single port, and didn't mind NAT it would be awesome! |
| 19:21.12 | kaldemar | well RTP isn't a part of SIP. most implementations just happen to use it for many things. |
| 19:21.13 | ayeso | Most carriers want to have signaling on one IP address and RTP on another... SIP makes this possible. |
| 19:21.15 | AeroCloud | got it :) use while ($x<count($dial_out) && ($chan_status['data']=="BUSY" || $chan_status['data']=="CONGESTION" || $chan_status['data']=="CHANUNAVAIL")) { |
| 19:21.22 | *** part/#asterisk Inode (n=Inode@unaffiliated/inode) |
| 19:21.42 | CrashSys | The good part about SIP is that's it's a protocol for other protocols... that bad part about SIP is the other protocols :D |
| 19:21.43 | AeroCloud | it looped the dials and worked :) |
| 19:22.07 | Naikrovek | all ya gotta do is convince cisco that SIP needs to go away AND convince them that IAX2 is superior and that should bring just about everyone else around |
| 19:22.31 | ayeso | IAX2 is good for small implimentations. it cannot scale the way SIP can. |
| 19:22.36 | dustybin | who created IAX2? Microsoft? |
| 19:22.43 | Naikrovek | dustybin: mark spencer |
| 19:22.45 | kaldemar | dustybin: ... |
| 19:22.49 | dustybin | *only* *jokin* :P |
| 19:23.01 | p3nguin | I'm kind of glad that Cisco supports SIP. I wasn't looking forward to figuring out SCCP with Asterisk. |
| 19:23.06 | CrashSys | yeah, but Crisco alreadt lost to SIP... if you remember SCCP/H323/MGCP |
| 19:23.09 | Naikrovek | p3nguin: that's good |
| 19:23.21 | ayeso | CrashSys: MGCP is still widely used |
| 19:23.23 | seanbright | is there any way to find out who owns a particular toll free? |
| 19:23.40 | jaytee | has anyone using a Linksys SPA-2102 ever seen both fxs ports act like they were the same number for callerid even though the web management interface shows them configured properly? this ATA worked fine up till today. |
| 19:23.41 | [TK]D-Fender | seanbright: My telco tells me no |
| 19:24.03 | CrashSys | ayeso: As legacy, sure... haven't seen much of it being deployed lately though... |
| 19:24.16 | [TK]D-Fender | jaytee: debug time... |
| 19:24.23 | ayeso | CrashSys: used alot for SS7 to SIP |
| 19:24.34 | jaytee | [TK]D-Fender, nothing's changed in my asterisk configs |
| 19:24.42 | dustybin | iax2 supports encryption? |
| 19:24.42 | seanbright | [TK]D-Fender: thanks |
| 19:24.43 | dustybin | http://www.voip-info.org/wiki/view/IAX+encryption |
| 19:25.04 | [TK]D-Fender | jaytee: look at the calls |
| 19:25.41 | dustybin | if i put in: encryption=yes in my iax.conf [general] section, will the voice data between my home asterisk box and the VOIP provider be encrypted? |
| 19:25.41 | jaytee | [TK]D-Fender, yeah. probably the ATA flaking out though |
| 19:26.09 | CrashSys | dustybin: Only if your provider also puts encryption=yes |
| 19:26.19 | dustybin | CrashSys: I havent a clue if they do |
| 19:26.24 | p3nguin | dustybin: That would be bad. Unless the provider wants to encrypt. |
| 19:26.27 | CrashSys | Then i'd say no |
| 19:26.27 | wonderworld | dustybin: thats cool. didn't know that. i tunneled it thru ssh for testing. that didn't work well. UDP over TCP sucks. |
| 19:26.31 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 19:26.36 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:27.10 | AbsurdHero | Does anyone know if there's something specific about the Sonicwall TZ series that would cause intermittent 4-5 second periods of silence? |
| 19:27.16 | CrashSys | So i've been slacking in my following of Asterisk v.1.6... what's the difference between the 1.6.0 and 1.6.1 tree? |
| 19:27.41 | ayeso | CrashSys: .1 |
| 19:28.01 | CrashSys | AbsurdHero: Turn off all SIP helper applications... they almost always break things in my experience... other then that it may be handling UDP poorly... |
| 19:28.38 | CrashSys | ayeso: But is there a functional difference between them? Trying to figure out why two tree's are being maintained... |
| 19:28.51 | AbsurdHero | We generally enable Consistent NAT and disable SIP Transformations, along with putting in an access rule to make sure it allows the traffic |
| 19:29.04 | dustybin | i wonder who wrote www.voip-info.org, they must get a _shit_ load of hits |
| 19:29.06 | ayeso | CrashSys: I really dont know, was being a smart ass.. Its documented with the release. |
| 19:29.46 | dustybin | i guess if you was running 2x asterisk boxes, you could encrypt the traffic between both boxes |
| 19:29.57 | dustybin | but then again, would there be any point |
| 19:30.22 | wonderworld | sure. nobody can spy. |
| 19:31.06 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 19:31.24 | Naikrovek | it's probably some simple xor "encryption" though |
| 19:31.24 | [TK]D-Fender | AbsurdHero: You should be just fine then |
| 19:31.56 | Naikrovek | without exchanging public keys outside of the normal channel traffic (unsnoopably) then it probably won't be very secure |
| 19:32.55 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:33.09 | Naikrovek | though i guess SSH does that |
| 19:33.35 | Naikrovek | suppose that's why they're public |
| 19:33.37 | Naikrovek | nevermind me |
| 19:34.09 | Naikrovek | so long as IAX uses asynchronous encryption of any reasonable quality i think it would be secure |
| 19:34.21 | Naikrovek | beats pgpphone probably as well |
| 19:35.41 | Naikrovek | (in terms of convenience) |
| 19:36.30 | *** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 19:38.54 | Naikrovek | is uh, is this thing on? awful quiet awful fast... |
| 19:39.11 | wcselby | no one can see what you're typing |
| 19:39.29 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 19:39.34 | p3nguin | wcselby: did naikrovek type something? |
| 19:39.35 | Naikrovek | i thought so. now i think i should maybe start calling people names. |
| 19:39.37 | *** join/#asterisk jtodd (i=xtufs3zt@ns.fox-den.com) |
| 19:39.37 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:39.45 | CrashSys | I'm still trying to figure out why 1.6.0 and 1.6.1 are different |
| 19:39.55 | wcselby | ~versioning |
| 19:39.59 | Naikrovek | do you mean "how" they're different? |
| 19:40.28 | wcselby | http://www.asterisk.org/node/48602 |
| 19:40.30 | [TK]D-Fender | ~asteriskversioning |
| 19:40.31 | infobot | from memory, asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 19:41.14 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 19:41.34 | vader-- | Do you guys know if it's possible to reset a TDM2400 while an asterisk box is up? |
| 19:41.57 | vader-- | i think one of the channels is stuck |
| 19:42.17 | vader-- | off the hook |
| 19:45.01 | [TK]D-Fender | vader--: You cannot |
| 19:45.17 | [TK]D-Fender | vader--: Well.. not without losing other calls on Zap/DAHDI |
| 19:45.50 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 19:46.06 | CrashSys | Well, i'm trying to figure out why there are two branches of 1.6 being maintained... |
| 19:46.20 | Naikrovek | i used to work at an isp that would have modems that would just not hang up |
| 19:46.55 | Naikrovek | so the tv repair guy built a giant panel of pushbuttons, one for each phone line. hundreds and hundreds of pushbuttons. and lights, if a call is in progress, there's a light |
| 19:47.03 | [TK]D-Fender | ~asteriskversioning |
| 19:47.04 | infobot | asteriskversioning is probably Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 19:47.06 | [TK]D-Fender | CrashSys: ^^^^^^^ |
| 19:47.16 | CrashSys | Ahhhso |
| 19:47.39 | dustybin | i remember _AGES_ ago, before i barely new what asterisk was, there was a big thing in news about serious security issues, do you remember those times? |
| 19:47.42 | Naikrovek | so if we saw that there was a particular modem that was always up, we'd jsut short the two wires so it could hang up, and if we wanted to leave it "offline" we'd leave the button pushed |
| 19:47.47 | Naikrovek | worked great |
| 19:48.01 | Naikrovek | when we replaced the modem, we'd pull the button out and it would receive calls again |
| 19:48.02 | [TK]D-Fender | dustybin: Welcome to "all the time" |
| 19:48.07 | dustybin | :P |
| 19:48.20 | [TK]D-Fender | Naikrovek: I normally push buttons. You pull them? |
| 19:48.31 | wcselby | i'm sending a private IP address from my asterisk server to my ITSP (bandwidth.com) in my SIP packets...how do I change that? http://pastebin.com/d278188de |
| 19:48.35 | Naikrovek | security issues are like "trouble in the middle east" always present |
| 19:48.54 | Naikrovek | [TK]D-Fender: yes, they were push, pull, no springs. push it to close the circuit, pull it to open it |
| 19:48.58 | [TK]D-Fender | ~sipnat |
| 19:48.59 | infobot | [~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 |
| 19:49.01 | [TK]D-Fender | wcselby: ^^^ |
| 19:49.21 | [TK]D-Fender | Naikrovek: Go whole hog next time... get the springs ;) |
| 19:49.37 | wcselby | with one of the nat commands |
| 19:49.48 | wcselby | yeah, thanks [TK]D-Fender :) |
| 19:49.55 | wcselby | I knew there was a link for that, thought I had saved it |
| 19:50.25 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:50.25 | Naikrovek | [TK]D-Fender: hehe, well they were chosen so you could see by looking at the switch whether or not the circuit was shorted. |
| 19:50.30 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:50.44 | [TK]D-Fender | Naikrovek: I thought thats what the light was for... |
| 19:50.58 | Naikrovek | no, light showed that a call was connected to that modem |
| 19:51.36 | Naikrovek | no call, no light |
| 19:51.39 | *** join/#asterisk mvanbaak_ (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:51.55 | Naikrovek | so there was no visual indication that a button was pushed unless you looked at the switch panel from an angle |
| 19:52.07 | Naikrovek | once though, a guy quit. on his way out he pushed every single button in |
| 19:52.16 | Naikrovek | all calls dropped |
| 19:52.31 | Naikrovek | this is also how the system admin would prepare to do server maintenance |
| 19:52.41 | p3nguin | naikrovek: I think the discrepancy is that of a push-on, push-off switch, and a plunger style switch. |
| 19:52.46 | Naikrovek | yes |
| 19:52.53 | Naikrovek | this was push on pull off |
| 19:53.44 | De_Mon | hrm, what was that command to convert an object to a simple array? ls|select filename -simple or something like that |
| 19:54.08 | Naikrovek | that looks almost like powershell |
| 19:55.04 | *** join/#asterisk el_critter (n=critter@190.78.48.45) |
| 19:55.19 | De_Mon | ahh -expandproperty |
| 19:55.48 | Naikrovek | De_Mon: what shell is that |
| 19:56.21 | De_Mon | powershell |
| 19:56.40 | De_Mon | wtf, expandproperty exists in v1 and v2, but doesn't work as expected in v1 :( |
| 19:56.41 | Naikrovek | why did you ask that in here? |
| 19:56.48 | De_Mon | looks around |
| 19:57.06 | De_Mon | whoops, wrong room |
| 19:57.09 | Naikrovek | hah |
| 19:57.27 | Naikrovek | is there a #powershell or soemthing |
| 19:57.34 | De_Mon | ya, its even on this network |
| 19:57.44 | Naikrovek | fetches the channel list |
| 19:57.51 | Naikrovek | cool |
| 19:58.02 | Naikrovek | powershell is nice. objects > plain text |
| 19:58.27 | De_Mon | if you have to be in windows, it's extremly nice |
| 19:58.45 | De_Mon | i pitty the sysadmins that had to use batch and vbscript |
| 19:58.51 | el_critter | Hi, sometimes when I call via PSTN (Digium TMP400P) and I hang up the call, it seems like asterisk hangs but the card keeps open the line. |
| 19:58.51 | Naikrovek | yeah |
| 19:58.56 | ayeso | I cant imagine a situation where I would have to use windows... |
| 19:58.56 | De_Mon | (or should I say have) |
| 19:59.05 | el_critter | Is that a known situation? |
| 19:59.07 | Naikrovek | el_critter: someone else was just asking about that |
| 19:59.17 | p3nguin | I'm confused with the correlation of "having to be in Windows" and "nice." |
| 19:59.29 | ayeso | p3nguin: Agreed |
| 19:59.49 | De_Mon | p3nguin it goes along the same lines of "if life gives you lemons, make lemon-aid" |
| 19:59.51 | el_critter | Naikrovek: really? I've been fighting with that for a couple of days but I can't find the solution. |
| 20:00.02 | Naikrovek | p3nguin: there are some things in windows that are very nice |
| 20:00.10 | wonderworld | i need windows to cash in with pokerstars. |
| 20:00.17 | Naikrovek | p3nguin: like not having to fiddle with my laptop when it comes out of standby |
| 20:00.22 | ayeso | Naikrovek: name 1. |
| 20:00.45 | De_Mon | oh brother. /me wanders away from windows hate group |
| 20:00.53 | wcselby | lol @ De_Mon |
| 20:01.51 | ayeso | I dont hate windows, its just not usefull for anything. |
| 20:01.56 | Naikrovek | ayeso: proper font hinting. Office. remote administration en masse. user compatability. hardware compatability. powershell. development software. |
| 20:02.04 | Naikrovek | it's plenty useful in the enterprise |
| 20:02.13 | ayeso | na |
| 20:02.16 | Naikrovek | ya |
| 20:02.28 | ayeso | just need people that actually know how to use/admin *nix |
| 20:03.15 | Naikrovek | my CEOs and CFO will *NEVER* learn to use unix, and that is not anything they should be bothered with |
| 20:03.37 | ayeso | Were talking about end users? |
| 20:03.48 | ayeso | I was taking about admins and servers. |
| 20:03.54 | p3nguin | ayeso: It bugs the hell out of me when these people try to pull the "You don't know how to use cPanel? What kind of sysadmin are you?" crap. |
| 20:04.00 | Naikrovek | well for servers it's more or less a different issue |
| 20:04.02 | ian6 | admins provide services to end users. |
| 20:04.05 | ian6 | try not to forget that. |
| 20:04.08 | Naikrovek | depends on what you're running |
| 20:04.25 | ayeso | they can use windows all day if they want, thats what help desk guys are for. |
| 20:04.40 | Naikrovek | servers are indeed a different issue. but most of my servers are windows |
| 20:04.46 | Naikrovek | though i do have serveral linux boxes |
| 20:04.53 | Naikrovek | virtual boxes but they count |
| 20:05.43 | ayeso | I dont want to have to click on things to get something done. |
| 20:06.06 | Naikrovek | that's what's so nice about powershell |
| 20:06.16 | Naikrovek | powershell can do MORE than the gui |
| 20:06.27 | ayeso | *sigh* |
| 20:06.30 | Naikrovek | and it's entirely cmdline and/or scripted |
| 20:06.43 | wonderworld | is it some kind of BASH-clone or is it a whole different thing? |
| 20:06.50 | Naikrovek | all different |
| 20:06.54 | ayeso | its a wanna be *nix shell |
| 20:06.59 | Naikrovek | no it isn't |
| 20:07.05 | Naikrovek | listen for a moment |
| 20:07.20 | p3nguin | places his ear by the speaker |
| 20:07.36 | Naikrovek | linux shells deal with text. text->pipe->process-> text |
| 20:07.36 | [TK]D-Fender | De_Mon: if life hands you lemons, ask the the tequila and salt... |
| 20:07.36 | p3nguin | There's a slight buzzing. |
| 20:07.51 | Naikrovek | text in text out |
| 20:07.53 | Naikrovek | which is awesome |
| 20:07.54 | Naikrovek | okay |
| 20:08.09 | De_Mon | [TK]D-Fender that works too! |
| 20:08.12 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 20:08.33 | Naikrovek | but powershell deals with objects. ls (which exists in powershell) does not output text, it outputs an object, which you can convert to text (and is the default behavior) |
| 20:08.45 | ayeso | Naikrovek: Use python |
| 20:08.59 | Naikrovek | omfg can i finish please |
| 20:09.05 | De_Mon | drags Naikrovek away from the conversation kicking and screaming |
| 20:09.07 | p3nguin | lol |
| 20:09.11 | ayeso | Naikrovek: everything is an object in python, even a string of text. |
| 20:09.16 | Naikrovek | great |
| 20:09.21 | Naikrovek | but i didn't get to the meat of the point y et |
| 20:09.27 | Naikrovek | and now you can live in ignorance because i'm done |
| 20:09.40 | ayeso | your not going to sell me on a windows shell. no matter what you say. |
| 20:10.04 | Naikrovek | not trying to sell you on it |
| 20:10.07 | De_Mon | ayeso what do you call someone who makes up their mind before hearing the evidence? |
| 20:10.07 | Naikrovek | trying to inform you |
| 20:10.09 | Naikrovek | but you're not having it |
| 20:10.11 | ayeso | Naikrovek: ill never need it. ill use python if i have to deal with windows. |
| 20:10.11 | Naikrovek | so i won't try |
| 20:10.34 | p3nguin | de_mon: an elitest? |
| 20:10.52 | [TK]D-Fender | el_critter: ... |
| 20:10.54 | [TK]D-Fender | ~cds |
| 20:10.54 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 20:10.56 | [TK]D-Fender | ^^^^^^^^ |
| 20:11.10 | [TK]D-Fender | el_critter: ask your telco to ensure this service is active on your lien |
| 20:11.40 | *** join/#asterisk Hadding (n=chatzill@207.177.231.9) |
| 20:11.51 | Naikrovek | things like powershell will almost certainly never be seen on any open system because of people like ayeso |
| 20:12.06 | Naikrovek | despite their fundamental superiority over text-based shells |
| 20:12.25 | p3nguin | I doubt that he single-handedly prevented such a thing. |
| 20:12.25 | ayeso | Naikrovek: Python |
| 20:12.29 | Naikrovek | ayeso: Scala |
| 20:12.31 | p3nguin | There has to be a bigger reason. |
| 20:12.46 | Naikrovek | p3nguin: cooperation is required and ayeso can't even be bothered to HEAR THE OTHER SIDE |
| 20:12.57 | Naikrovek | OR read it |
| 20:13.19 | ayeso | Naikrovek: I used to be a windows admin, i know all about it dude.. it sucks learn *nix and youll understand. |
| 20:13.30 | Naikrovek | i know unix |
| 20:13.34 | ayeso | sure |
| 20:13.39 | Naikrovek | been using it for damn near 20 years |
| 20:13.39 | Naikrovek | daily |
| 20:13.47 | Naikrovek | but powershell is superior to *sh |
| 20:13.52 | ayeso | Naikrovek: you use KDE or Gnome? |
| 20:13.55 | Naikrovek | even pyhon |
| 20:13.59 | Naikrovek | i don't use a gui |
| 20:14.02 | ayeso | good |
| 20:14.03 | Naikrovek | on linux |
| 20:14.07 | ayeso | was a trick question |
| 20:14.10 | Naikrovek | kde and gnome are both terrible |
| 20:14.18 | ayeso | thats the right answer |
| 20:14.30 | ayeso | what distro do you run? |
| 20:14.31 | [TK]D-Fender | I use a GUI... its a great way to have lots of text shells open on a huge screen simultaneously :p |
| 20:14.32 | p3nguin | I hope you don't think kde and gnome are the only choices. |
| 20:14.35 | ayeso | ubuntu? |
| 20:14.37 | Naikrovek | ayeso: several different ones |
| 20:14.41 | Naikrovek | depends |
| 20:15.01 | ayeso | p3nguin: only choices for people that claim to be *nix admins that are not |
| 20:15.04 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 20:15.10 | darkdrgn2k3 | hello world |
| 20:15.35 | bmoraca | wow...if this dude's epeen gets any bigger, we'll need a bigger IRC room |
| 20:15.42 | Naikrovek | ubuntu server, opensuse, centos, rhel, and (my favorite because it's not linux) freebsd |
| 20:16.00 | ayeso | Naikrovek: anyway this an asterisk board, im done, your right powershell is superior to all other shells past present and future. I hope you write great scripts with it. |
| 20:17.34 | De_Mon | can we all hug now? |
| 20:17.34 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 20:17.34 | Naikrovek | not better than future shells i'm sure but, believe it or not, it is well worth looking at for any windows admin because there are a hell of a lot of things that can only be done in that shell. |
| 20:17.34 | Naikrovek | i'm not hugging him but i'm done arguing |
| 20:17.41 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 20:17.48 | darkdrgn2k3 | Naikrovek: i think that was ment as a way to CLOSE the conversation, not contiue it.... |
| 20:18.01 | De_Mon | uh huh |
| 20:18.02 | ayeso | ^^^ |
| 20:18.13 | Naikrovek | it's done |
| 20:18.16 | *** join/#asterisk gardo (n=gardo@110.55.42.65) |
| 20:19.06 | el_critter | [TK]D-Fender: I have hangup on polarity change but does it supervises battery cuts too? |
| 20:19.17 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 20:19.39 | Naikrovek | el_critter: telco tell you that's what you have? |
| 20:20.41 | el_critter | Naikrovek: Let's assume that talking with my telco is near impossible. Because it is. |
| 20:23.20 | el_critter | I've monitored via wctdm module debug option and in dmesg I usually see this messages: |
| 20:23.20 | el_critter | [95331.351761] 23757837 Polarity reversed (-1 -> 1) |
| 20:23.20 | el_critter | [95331.367761] NO BATTERY on 1/3! |
| 20:23.20 | el_critter | [95331.687744] 23757921 Polarity reversed (1 -> -1) |
| 20:23.20 | el_critter | [95331.703736] BATTERY on 1/3 (-)! |
| 20:24.07 | *** part/#asterisk Matrix_Maddy (n=chatzill@dxb-b126613.alshamil.net.ae) |
| 20:25.03 | Naikrovek | that looks like a battery cut and a polarity reversal |
| 20:25.47 | el_critter | and those are totally different events? |
| 20:26.00 | Naikrovek | each can occur, or both can occur |
| 20:26.29 | [TK]D-Fender | el_critter: what signalling are you using exactly? |
| 20:26.56 | el_critter | fxsks |
| 20:27.40 | wcselby | <Naikrovek> that looks like a battery cut and a polarity reversal <--- what was your first clue? :P |
| 20:27.54 | Naikrovek | wtf is this? pick on naikrovek day |
| 20:28.06 | wcselby | haha, damn, I wasn't trying to pick on him |
| 20:28.16 | wcselby | i thought he was trying to be funny |
| 20:28.19 | wcselby | oh well |
| 20:28.43 | bmoraca | he left? |
| 20:28.43 | el_critter | wcselby: "oh well"? Now you help me! |
| 20:28.50 | darkdrgn2k3 | whats teh big voip provider in europe? |
| 20:28.52 | el_critter | :) |
| 20:29.25 | wcselby | el_critter - i know nothing about what you're trying to do |
| 20:29.40 | el_critter | kidding |
| 20:32.43 | [TK]D-Fender | checkout time, later all |
| 20:33.20 | csmyth | this is fun. let's talk about vi vs. emacs now! |
| 20:34.58 | bmoraca | nano > all |
| 20:35.10 | wcselby | vim-enhanced |
| 20:35.33 | [8none1] | bmoraca: your sign is backwards. should be nano < all |
| 20:35.34 | p3nguin | nano? hahahaha |
| 20:36.03 | csmyth | yeah IMO nano is for folks who would rather be using notepad LOL |
| 20:36.38 | wcselby | i worked with a guy who loved joe |
| 20:36.46 | wcselby | joe the editor |
| 20:36.50 | wcselby | not the plumber |
| 20:36.52 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 20:36.54 | darkdrgn2k3 | csmyth: you like talking religion dont you? |
| 20:37.26 | darkdrgn2k3 | csmyth: if you ask me good old cat >filename rox! |
| 20:37.30 | csmyth | darkdrgn2k, yeah i think it is funny how folks get up in arms about technology |
| 20:37.39 | csmyth | copy con prn |
| 20:37.42 | [8none1] | wcselby: joe his life partner that's an editor? |
| 20:37.42 | darkdrgn2k3 | hahah |
| 20:37.54 | darkdrgn2k3 | i remember that command :-P |
| 20:38.03 | darkdrgn2k3 | dir /s > prn |
| 20:38.09 | darkdrgn2k3 | oo the paper i wasted :-D |
| 20:38.17 | csmyth | yup i suck at DOS but i remember that one |
| 20:38.36 | darkdrgn2k3 | haha i had a challange proposed to me by some one to hack into a machine... |
| 20:38.45 | darkdrgn2k3 | the owner of the machine knew i was comming so he piped all his logs to lp0 |
| 20:38.56 | darkdrgn2k3 | worst part of it was i foudn out :-D |
| 20:39.09 | csmyth | while true; do ps -ef; done |
| 20:39.32 | csmyth | or |
| 20:39.41 | darkdrgn2k3 | csmyth: i remember in school one day... we had an exam.. teacher put down while (2) { ... instead of while (1) { ... |
| 20:39.46 | csmyth | while true; do echo ehatever | su -; done |
| 20:39.46 | p3nguin | `yes' |
| 20:39.50 | darkdrgn2k3 | more then 1/2 the class was like OMG what does while 2 do! |
| 20:40.14 | darkdrgn2k3 | worst part was it was an "advanced" c class :-S |
| 20:42.41 | csmyth | i had a basic course in c once where i took an incomplete because i couldn't find a bug in my own program - i had a test in a loop that was like "if( x=100 )" |
| 20:42.50 | dustybin | is .sln format u-law ? |
| 20:42.58 | darkdrgn2k3 | awww.. |
| 20:43.07 | [8none1] | I remember using edlin in DOS 3.22, now that's a real editor |
| 20:43.18 | csmyth | yeah i found it next semester and got my "B"...*whew* |
| 20:44.05 | csmyth | yeah edlin was hard...i used that before i knew any better. i think i switched to notepad on win3.1 LOL |
| 20:48.09 | el_critter | brb |
| 20:48.15 | darkdrgn2k3 | csmyth: could of been worst. could be using debug |
| 20:54.22 | IBC_jkenney | what is everyone using for asterisk click to dial on a linux env? |
| 20:54.27 | ayeso | I miss the dail-up first class BBS systems |
| 20:54.32 | ayeso | Hermes II |
| 20:54.40 | IBC_jkenney | iniquity |
| 20:54.57 | csmyth | lol /join #nostalgia |
| 20:55.36 | ayeso | no one was in that channel. |
| 20:55.44 | IBC_jkenney | used to run a board was the first kid on the block with a 1 gig hd |
| 20:55.46 | IBC_jkenney | lol |
| 20:56.29 | ayeso | ANSI graphics were awesome. |
| 20:58.56 | Hadding | So a context is like [here]? |
| 20:59.04 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
| 20:59.16 | ayeso | Hadding: that woudl be the 'here' context yes |
| 20:59.46 | Hadding | So whatever is between brackets |
| 20:59.49 | Hadding | gotcha |
| 21:00.14 | ayeso | Hadding: best to name them something that makes scene in my opinion... like [phones] [inbound] [outbound] etc....... |
| 21:00.59 | csmyth | ayeso, like you're some sort of expert. loser. |
| 21:01.00 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 21:01.25 | ayeso | Hadding: contexts have extensions, extensions have priorities. if a call is in a context, it cannot go to another context unless you tell it to. |
| 21:01.45 | ayeso | csmyth: na I only know what you showed me. |
| 21:02.02 | Hadding | Ok in extensions.conf I need to add lines in the same context as my SIP phone, whats the name of this context? |
| 21:02.56 | ayeso | Hadding: look in sip.conf. your phones should have "context={something}" this is where they go. |
| 21:03.10 | Katty | boingboingboing |
| 21:03.15 | Katty | points at time |
| 21:03.17 | Katty | boingboingboing |
| 21:05.22 | p3nguin | You're STILL on this? I told you two days ago what to do with those two lines you wanted to keep pasting in here. |
| 21:06.16 | Hadding | Cant find where in extensions.conf |
| 21:08.48 | ayeso | Hadding: patebin your extensions.conf and your sip.conf |
| 21:09.06 | Hadding | How do I patebin |
| 21:09.31 | ayeso | here is one of many: http://pastebin.ca/ |
| 21:13.15 | bmoraca | does anyone have a good wav file of a generic ringing sound that's maybe 15 seconds long? |
| 21:23.50 | Qwell | Record() a Playtones(ring) ? |
| 21:24.02 | Qwell | 15 seconds would be silly. You want full cycles |
| 21:24.14 | *** join/#asterisk ewheeler (n=ewheeler@pool-151-202-75-199.ny325.east.verizon.net) |
| 21:24.33 | bmoraca | it's for an external ringer of sorts |
| 21:25.31 | bmoraca | and there are no cycles, because it's never actually ringing |
| 21:25.33 | *** join/#asterisk wr| (n=wr@p54BE2EBC.dip.t-dialin.net) |
| 21:26.45 | IBC_jkenney | why not use the one in asterisk |
| 21:26.50 | IBC_jkenney | the sounds dir |
| 21:26.53 | IBC_jkenney | in /var/spool/asterisk/sounds |
| 21:26.59 | IBC_jkenney | in /var/spool/asterisk/sounds/ringing |
| 21:29.07 | bmoraca | interestingly, i don't have that file in my sounds dir |
| 21:29.34 | hesco | ~paste |
| 21:29.35 | infobot | hmm... paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 21:30.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 21:31.32 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 21:32.46 | *** join/#asterisk d4rkstar (n=bruno@ip-43-88.sn1.eutelia.it) |
| 21:33.47 | wr| | hi, compiling chan_capi gives me "error: asterisk/channel_pvt.h: No such file or directory" and "error: âstruct ast_channelâ has no member named âpvtâ" (a hundred times), anyone any idea what to do? |
| 21:35.10 | *** join/#asterisk jtodd (i=qsozo9yj@204.91.156.197) |
| 21:35.10 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:37.05 | Katty | bmoraca: did you find a ringing noise? |
| 21:39.09 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:41.49 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:42.54 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
| 21:46.14 | Qwell | wr|: The author of that module needs to update it for the version of Asterisk you're compiling against. |
| 21:47.30 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:47.41 | wr| | Qwell: yeah, I assumed. just thought maybe it's a known problem and there's a quick hack I could do to fix it. |
| 22:03.51 | *** join/#asterisk ewheeler_ (n=ewheeler@pool-151-202-75-199.ny325.east.verizon.net) |
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| 22:22.17 | *** join/#asterisk d4rkstar (n=bruno@ip-43-88.sn1.eutelia.it) |
| 22:31.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:34.30 | *** join/#asterisk mmanders (n=max@client-80-5-175-229.glw-bng-011.adsl.virginmedia.net) |
| 22:36.10 | mmanders | So, I've recently set up FreePBX at work (UK using ISDN30). I'm getting the odd dropped call; coinciding with this in the logs are lines pertaining to 'DTMF digit f' |
| 22:36.26 | mmanders | Does this mean anything to anyone? What is DTMF digit f (I thought it only went up to D) |
| 22:36.35 | mmanders | And why would this be dropping calls |
| 22:39.34 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@CPE0080c6fe36d6-CM001cea39706c.cpe.net.cable.rogers.com) |
| 22:39.39 | darkdrgn2k3 | p3nguin: i just got the sip to work! it was "host=domain" and "outboundproxy=sip provider" |
| 22:44.30 | *** join/#asterisk Badrobot- (n=Badrobot@76.173.229.89) |
| 22:46.21 | p3nguin | darkdrgn2k3: You thought it might have been that way, but I thought you tested that and it didn't work. |
| 22:48.02 | darkdrgn2k3 | p3nguin: i know.. but today it decided to work ?!?!?! go figure |
| 22:48.11 | darkdrgn2k3 | p3nguin: i didnt event rebot! |
| 22:48.36 | darkdrgn2k3 | p3nguin: maybe i was just to dam tiered :-S now to figure out the incomming part.. |
| 22:49.36 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 22:50.50 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 22:57.24 | darkdrgn2k3 | if the "to" line of an incomming call looks like "6474321456@pritrunkdomain" |
| 22:57.45 | darkdrgn2k3 | when configuring an inbound number do i do a DID 6474321456@pritrunkdomain or just 6474321456 |
| 22:58.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:03.51 | *** join/#asterisk propellerhead (n=yogurt2u@190.246.145.56) |
| 23:15.47 | *** join/#asterisk matrix1233 (n=Administ@41.226.30.102) |
| 23:15.56 | matrix1233 | hello |
| 23:16.24 | matrix1233 | in dialplan how can i dial a IVR (900 for example) ?? |
| 23:19.07 | matrix1233 | any one here |
| 23:20.30 | [TK]D-Fender | matrix1233: No such thing as "dial an IVR". You dial an extension and it does whatever you coded for it in your dialplan. |
| 23:22.08 | matrix1233 | [TK]D-Fender: i tryed Dial(900) it's not work |
| 23:22.40 | *** join/#asterisk smps (n=maher@193.170.53.51) |
| 23:22.58 | [TK]D-Fender | matrix1233: Of course not. Dial() is meant t =o call a DEVICE. an IVR is simply DIALPLAN. You are already IN the dialplan. |
| 23:23.03 | [TK]D-Fender | to* |
| 23:24.30 | matrix1233 | [TK]D-Fender: sorry am newer in asterisk .. can you write me an example to can use the IVR (900) |
| 23:25.13 | [TK]D-Fender | MatWhat IVR 900? Where did you even create this thing? PASTEBIN is your friend.... |
| 23:25.14 | [TK]D-Fender | ~pb |
| 23:25.15 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 23:25.17 | [TK]D-Fender | ^^^^^^^^^ |
| 23:26.50 | matrix1233 | [TK]D-Fender: :) am testing with switchvox and i have a webinterface to can create IVR , extention ... |
| 23:26.59 | darkdrgn2k3 | [TK]D-Fender: can i ask is this a valid incomming call: http://pastebin.ca/1571335 |
| 23:27.07 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 23:27.20 | [TK]D-Fender | matrix1233: GUI's are NOT supported in here. Please see the topic for a list of other related channels |
| 23:28.12 | [TK]D-Fender | darkdrgn2k3: that isn't a call.. that is an ACK packet to "God knows what SIP comm" |
| 23:28.44 | darkdrgn2k3 | wierd when i call in, thats the only thing that shows up on console |
| 23:28.53 | darkdrgn2k3 | i guess my user details are not configured properly yet then :-S |
| 23:29.11 | matrix1233 | [TK]D-Fender: ok.. now am trying to change the extention.conf without interface |
| 23:29.57 | [TK]D-Fender | matrix1233: You do not do this with Switchvox. your changes will get blown away. Their interface is designed in a way that is hostile to outside mods. |
| 23:30.19 | [TK]D-Fender | darkdrgn2k3: I did not say that. What I said is you showed nothing of value. |
| 23:30.27 | darkdrgn2k3 | ok sorry |
| 23:30.30 | darkdrgn2k3 | posted whole log |
| 23:30.30 | darkdrgn2k3 | http://pastebin.ca/1571341 |
| 23:30.34 | [TK]D-Fender | darkdrgn2k3: That final packet is not identifiable as anything at all |
| 23:31.27 | [TK]D-Fender | darkdrgn2k3: Looking for 6472892222 in xcopper.com (domain 70.33.240.53) ----- SIP/2.0 404 Not Found |
| 23:31.42 | darkdrgn2k3 | dmmm |
| 23:31.50 | darkdrgn2k3 | thanx didnt see that before |
| 23:31.53 | [TK]D-Fender | darkdrgn2k3: you should pay more attention to teh CONTEXT you specified, and the exten its looking for. This is a dialplan error |
| 23:35.21 | matrix1233 | [TK]D-Fender: yes i know .. but i have to do that beacause i have a spetial work to do with switch |
| 23:37.25 | *** join/#asterisk incorrect (n=incorrec@cpc3-cmbg8-0-0-cust1020.cmbg.cable.ntl.com) |
| 23:37.54 | [TK]D-Fender | matrix1233: Again, GUI's are not supported in here, go ask in their channel |
| 23:40.36 | matrix1233 | [TK]D-Fender: ok thanks, i found the solution using Goto |
| 23:40.40 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 23:51.57 | *** join/#asterisk R-Guy (n=ron@secure.starsolutions.com) |
| 23:52.24 | R-Guy | Anyone here use VoipJet for outbound? |
| 23:53.29 | dustybin | at long last, i now have a fully working dialplan |
| 23:54.01 | dustybin | and i now understand how things work :D |
| 23:57.59 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
