| 00:00.50 | NovceGuru | em0 |
| 00:01.39 | jblack | Hard core. It went from melodramatic, to "I hate the world songs". to "I fucking hate this person songs", to "there's no hope for the world" to "there's no hope for me". Really kinda spooky. |
| 00:02.13 | jblack | It's like when my tivo spent 2 weeks thinking I was gay. |
| 00:03.12 | jblack | ohhh, nifty thought. How about an app that rates people that call. Then, depending on your mood, it screens based on rating. :) |
| 00:05.42 | *** part/#asterisk AJayMN (n=mybed@66-188-94-131.dhcp.mdsn.wi.charter.com) |
| 00:06.55 | coppice | jblack you got that from a weird al song, didn't you? |
| 00:07.16 | jblack | The first two are entirely true, the third one I made up all on my own. |
| 00:08.12 | jblack | I thought the tivo/gay thing was a running gag, but it actually happened to me. I fell asleep while watching the home improvement channel. For the next two weeks, everything tivo suggestion included a lisp. |
| 00:08.57 | coppice | youtube comes up with some really funky suggestions for me. If it ever recommend a Celine Dion video, I think I'll stop using it |
| 00:09.29 | jblack | bleh youtube. The only way my kid talks to me is links to youtube. |
| 00:09.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:10.42 | jblack | wtf? "I am a shark, if I stop moving, I will die. If you hold me, I'll just dissapear" |
| 00:11.09 | coppice | sounds like someone reworking a Woody Allen line |
| 00:11.55 | jblack | Shark Serenade, Vermillion Lies. THe main instrumentals are a xylophone and a vibraphone. |
| 00:13.53 | coppice | look on the bright side - at least its not repeating the top 10 all day |
| 00:14.00 | jblack | That and some suzanne vega b tracks would make for great "go away" hold music. ;) |
| 00:14.18 | jblack | Goo point. |
| 00:14.39 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 00:15.08 | jblack | LOL. last.fm thinks I should live. It went to enya, then "What is love" as per SNL. :) |
| 00:17.59 | *** join/#asterisk Micho123 (n=mcho123@94.187.66.218) |
| 00:18.30 | Micho123 | Hi all, WHn trying to start asterisk I got the following error...Starting asterisk: /usr/sbin/safe_asterisk: line 75: ulimit: open files: cannot modify limit: Operation not permitted |
| 00:18.37 | Micho123 | Any ideas about this issue? |
| 00:18.55 | [netman] | man ulimit |
| 00:19.47 | Micho123 | [netman], I used ulimit -n to increase the number of open files for root...but still have the same issue |
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| 00:21.52 | jblack | Micho123: Sounds like you're trying to raise the limits above what's possible. |
| 00:22.22 | [netman] | maybe /usr/sbin/safe_asterisk is not being executed with enough privileges |
| 00:22.54 | Micho123 | [netman], maybe you are right |
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| 00:23.01 | Micho123 | I'll check this |
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| 01:02.03 | f0ner00t | Hmmp |
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| 01:30.46 | f0ner00t | Okay. I just want to hook up sip... To my ipkall / gatorphone acct via asterisk can someone help me |
| 01:32.27 | Linuturk | waves @ russellb |
| 01:33.35 | *** join/#asterisk jaxxan (n=jaxxan@bankoh-m39s4.blueskynet.as) |
| 01:33.40 | jaxxan | hey guys |
| 01:34.51 | jameswf | speaking of gay i am watchinf Family guy starwars episode |
| 01:34.55 | jaxxan | i'm running 1.4.21.1 and I have asterisk-addons-1.4.7, but I need the MYSQL Application, but i'm not sure how to get that compiled in now. i guess it was removed from the asterisk-addons? any pointers? |
| 01:35.39 | jaxxan | i see a http://svn.digium.com/svn/asterisk-addons/branches/1.4/cdr/cdr_addon_mysql.c |
| 01:35.47 | jaxxan | how do i install that? |
| 01:36.19 | jaxxan | just svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.4/cdr/ cdr_addon_mysql ??? |
| 01:36.26 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 01:37.06 | raden_work | anyone have any opinions on gafachi ? |
| 01:39.43 | jameswf | bless you |
| 01:40.27 | russellb | waves back to Linuturk |
| 01:40.51 | Linuturk | passes russellb frito chili pie |
| 01:40.58 | russellb | osnap! |
| 01:41.10 | Linuturk | ;p |
| 01:41.12 | russellb | that was a yummy lunch :-) |
| 01:41.18 | russellb | wonders what you guys get tomorrow ... |
| 01:41.23 | Linuturk | yep. Haven't had that in a while |
| 01:41.42 | Linuturk | that's the fun. it's always a suprise |
| 01:42.09 | russellb | heh |
| 01:43.52 | Linuturk | the wife is going stir crazy in the hotel :) |
| 01:44.04 | Linuturk | it's slightly amusing |
| 01:44.57 | jaxxan | anyone know how to install the mysql application in asterisk 1.4 ? |
| 01:45.09 | jameswf | practice |
| 01:45.21 | jameswf | make menuselect |
| 01:45.48 | jaxxan | jameswf: are you talking to me? |
| 01:46.25 | jameswf | jaxxan, if it works yes otherwise no |
| 01:46.53 | f0ner00t | Can someone help me configure my asterisk box to work with IPKALL / Gatorphone? |
| 01:49.17 | jameswf | f0ner00t, try http://tinyurl.com/dmyrym |
| 01:50.17 | f0ner00t | jameswf: Thank you noted. Goodnight see ya all tommorrow! |
| 01:50.31 | jameswf | uhhh lol |
| 02:01.03 | Nasra | need to find someone to help me findout what is going on with my asterisk box ....it does not want to start ....please take a look at my pastebin thanks.... http://www.pastebin.ca/1372429 |
| 02:01.03 | Nasra | <PROTECTED> |
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| 02:21.26 | phix | Nasra: hello sir |
| 02:22.15 | phix | Nasra: you forgot the sudo before asterisk -rvvdd |
| 02:22.36 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:22.39 | phix | a normal user doesn't gave access to the asterisk console |
| 02:23.13 | phix | I though this line would of given it away --> # |
| 02:23.15 | phix | <PROTECTED> |
| 02:23.24 | phix | (Permission denied) |
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| 02:26.26 | Nasra | phix: |
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| 02:27.34 | jameswf | eeek dont run asterisk as root |
| 02:28.35 | scooby2 | why the heck is the audio on this sangoma card so choppy |
| 02:28.40 | Nasra | phix: is now connectd |
| 02:29.22 | jameswf | scooby2, its sangoma... |
| 02:31.59 | scooby2 | jameswf: lol, I have not had a lot better luck with digium gear |
| 02:32.25 | jameswf | okay maybe its you |
| 02:32.27 | jameswf | :) |
| 02:32.34 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 02:33.11 | kn0x | quick question |
| 02:33.11 | scooby2 | finally took buying new hardware to get the digium to work |
| 02:33.36 | kn0x | anyway to use zaptel channels as a dial-up terminal? |
| 02:39.47 | kn0x | i need to dial in and do remote admin without dedicating a trunk to a modem |
| 02:40.08 | phix | jameswf: ? |
| 02:40.20 | kn0x | is there a way to hook this up without running through an fxs? |
| 02:40.22 | phix | jameswf: sudo -u asterisk asterisk -rvvvvvvdddddd |
| 02:41.03 | jameswf | proper setup is key no sudo |
| 02:41.20 | phix | sudo == proper setup |
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| 02:44.00 | jameswf | phix http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root |
| 02:44.30 | KavanS | sudo? |
| 02:44.50 | jameswf | ~sudo |
| 02:45.08 | jbot | somebody said sudo was (SUperuser DO) better than su, according to talon.It is able to give limited super user privileges to specific users, or can allow you to do silly things like run X apps with root perms, or good in scripts with "username ALL = NOPASSWD: /some/program", or http://www.aplawrence.com/Basics/sudo.html, or good for ordering sandwiches, or not pseudo |
| 02:45.25 | jaxxan | so make menuselect shows the app_addon_sql_mysql with XXX next to it and down below it says Depends on mysqlclient(E) |
| 02:45.27 | jaxxan | what is that? |
| 02:45.52 | jameswf | jaxxan, means your missing myswlclient |
| 02:45.59 | jameswf | *mysql |
| 02:46.00 | jaxxan | i have mysql mysql-devel and mysql-server installed |
| 02:46.38 | jameswf | jaxxan, mysql-client |
| 02:47.03 | jaxxan | ummmm |
| 02:47.14 | jaxxan | yum install mysql-client doesn't show anything |
| 02:48.07 | jameswf | yum list mysql* |
| 02:48.59 | jaxxan | the only ones that aren't installed are mysql-bench and mysql-test |
| 02:49.03 | jaxxan | all the others are already installed |
| 02:49.31 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 02:49.38 | jaxxan | it's weird, i never had this problem with 1.2 |
| 02:49.57 | jameswf | jaxxan, try yum provides mysqlclient |
| 02:50.26 | jaxxan | downloading new filelists |
| 02:51.00 | infinity1 | a sip dial takes about 30 seconds to completely fail when the sip provider is unavailable. how can i make it fail faster so it gets to the failover trunk faster? |
| 02:52.29 | jaxxan | libmysqlclient ? |
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| 02:53.45 | jameswf | try it |
| 02:54.12 | jaxxan | nothing |
| 02:54.26 | jaxxan | the libmysqlclient is included in all the other mysql packages that I already have installed |
| 03:03.50 | stabler | ~sipnat |
| 03:03.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 03:32.02 | phix | jameswf: but it is ok to run the client as root? |
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| 03:54.49 | jameswf | remember what you run as root can do whatever it wants |
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| 04:33.20 | stabler | does any one run a pfsense firewall? |
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| 06:16.36 | hope4every1 | Hi every1 |
| 06:16.53 | hope4every1 | I have set up a vicidial and trixbox |
| 06:17.23 | hope4every1 | but my extension not ringing when i login the vicidial |
| 06:19.43 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:19.50 | hope4every1 | can u help me with information regarding setting up vicidial with trixbox |
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| 09:02.31 | brunner | can anyone help me find a co-location provider in Mexico that offers TDM connectivity to the PSTN? |
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| 09:19.52 | yang | brunner: what about www.voip.ms ? |
| 09:21.25 | brunner | those rates are pretty good, thanks |
| 09:27.10 | yang | Yes, I use it they are ok |
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| 10:04.29 | k-man | in an spa942, to transfer a call, do you hit transfer, dial the extension and then hang up? |
| 10:06.22 | *** join/#asterisk dr_gogeta86 (n=fisgro@81-208-88-100.ip.fastwebnet.it) |
| 10:07.28 | k-man | oh - when all else fails, read the manual |
| 10:07.42 | k-man | apparantly you have to press transfer again to complete the transfer |
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| 10:37.15 | bobsaccamano | hi im trying to write a dial plan for international dialing..so i have a SIP no and a country code..ONLY when i dial the country code prefix i should be able to call this sip number |
| 10:37.32 | bobsaccamano | so i employed a simple pattern matching extension |
| 10:37.37 | bobsaccamano | like so exten=>_011XXXXXXXXXXXXXXX,1,Dial(SIP/544444444444444/${EXTEN:3},20) |
| 10:37.49 | bobsaccamano | is this the right way? |
| 10:39.51 | Bladerunner05 | hello using Dial command I need to specify to call user1 for 15 seconds and if no answer or busy try user2 and so on... but if I do Dial(SIP/user1&SIP/user2,15) it call simultaneously user1 and user2.... |
| 10:40.44 | mort_gib | Bladerunner05: Use followme or DIALSTATUS |
| 10:41.13 | Bladerunner05 | mort_gib my u explain me a little bit with an example ? |
| 10:41.36 | mort_gib | Bladerunner05: Eh, no, the book is full of examples |
| 10:42.02 | mort_gib | Bladerunner05: DIALSTATUS allows you to check if the call was picked up after 15 sec |
| 10:43.08 | mort_gib | Bladerunner05: Followme works different, but might be what you want |
| 10:43.59 | mort_gib | bobsaccamano: Are you trying to place the call via a SIP account, or to dial a SIP phone?? |
| 10:47.35 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:48.00 | bobsaccamano | dial a sip phone |
| 10:48.09 | bobsaccamano | mort_gib, it worked |
| 10:48.31 | mort_gib | bobsaccamano: Ok :-) |
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| 10:52.17 | defswork | anyone heard of swyx ? |
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| 10:56.15 | defswork | had a call from a company that have been quoted for swyx in a small call centre - they quoted for 1*ISDN2 and 1*ADSL for VOIP and they will be having a minimum of 10 simultaneous outgoing calls 99% of the time |
| 10:56.39 | defswork | sounds like trouble to me - 10 calls over VOIP over a single ADSL |
| 10:56.57 | defswork | I've told her she should look at E1 |
| 10:57.16 | mort_gib | Depends on the ADSL, but if they think they can do that while lusers are surfing they are in for a nasty surprise |
| 10:57.56 | defswork | plus no SLAs on ADSL |
| 10:58.05 | defswork | could be down for days |
| 10:58.05 | mort_gib | Exactly |
| 10:58.36 | mort_gib | ADSL sucks for VOIP, unless you can reserve a circuit ONLY for VOIP, and even then.... |
| 10:58.46 | defswork | she want's crm integration though - to instigate calls from something called Zoho |
| 10:59.25 | mort_gib | Hey, have a look at SugarCRM, I stumbled over some integration with Asterisk the other day... |
| 10:59.28 | defswork | I'll see if I can downgrade that requirement to a would-be-nice-to-have |
| 11:00.15 | mort_gib | defswork: You don't have to do that! I have all my clients (being able to) call directly from Outlook |
| 11:00.21 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 11:00.30 | mort_gib | defswork: Then use SugarCRM on the back of Outlook |
| 11:00.53 | defswork | she's already using zoho though - I don't think moving is an option |
| 11:01.02 | defswork | zoho integrates with outlook though |
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| 11:01.40 | mort_gib | defswork: There you go! |
| 11:04.04 | mort_gib | http://www.voip.com.sg/asterisk/voip_asterisk_outlook_dialer.html |
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| 11:05.26 | defswork | I need to play with that |
| 11:05.40 | defswork | I hate outlook though - is there a thunderbird equiv too ? |
| 11:06.05 | mort_gib | defswork: I don't know, I don't use Outlook or Thunderbird for that matter |
| 11:06.11 | mort_gib | My users do though |
| 11:06.14 | defswork | yeah there is - it's called snap |
| 11:06.56 | mort_gib | Yeah?? url?? |
| 11:09.05 | mort_gib | Windows only though |
| 11:09.21 | mort_gib | Snom handsets allows you to use CTI |
| 11:09.38 | mort_gib | popup included (they drive me nuts, but some like them) |
| 11:10.07 | mort_gib | http://www.snom.com/en/products/software/cti-solution/ |
| 11:13.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:15.22 | defswork | hmm |
| 11:15.26 | defswork | I've never used snom |
| 11:17.10 | mort_gib | defswork: They are nice... |
| 11:20.12 | *** join/#asterisk thelordmortis (n=lordmort@119.11.33.194) |
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| 11:28.38 | diyanat | hi |
| 11:29.16 | diyanat | i am trying to compile zaptel-1.2 on slackware 12.2 getting this error http://pastebin.com/m53a8af3b |
| 11:29.46 | diyanat | too few arguments to function 'device_create' |
| 11:30.55 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 11:31.50 | diyanat | the kernel is 2.6.27.7 |
| 11:35.24 | *** join/#asterisk DelphiWorld (n=Miranda@41.221.19.173) |
| 11:37.58 | Gopaul | i m getting this error while trying to reach a extension in legacy pbx [Mar 26 15:41:03] WARNING[8820]: app_dial.c:1480 dial_exec_full: Unable to create channel of type 'DAHDI' (ca use 0 - Unknown) |
| 11:37.59 | Gopaul | <PROTECTED> |
| 11:37.59 | Gopaul | <PROTECTED> |
| 11:38.26 | Gopaul | from the legacy pbx the incoming call is coming in |
| 11:47.04 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:51.03 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:51.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:52.04 | *** join/#asterisk j_kroon (n=jkroon@dsl-244-11-59.telkomadsl.co.za) |
| 11:52.11 | j_kroon | hi all, i'm hoping for some advice here. |
| 11:55.40 | j_kroon | I'm bridging between a SIP channel (GXP2010 and SNOM320 atm, but also a Linksys SPA, also with GS BT200 and GXP280) and a DAHDI channel. asterisk 1.6.0.3 with dahdi 2.1.0.3. What I'm getting is that it seems that some frames are dropped. |
| 11:56.32 | j_kroon | if I didn't know that asterisk doesn't support silence supression I would have guessed that I can blame that. |
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| 12:11.40 | SuPrSluG | j_kroon, GSbt200's had a nasty frimware bug that created broadcast storms. make sure you update frimware if it's available. or try taking that off the network. |
| 12:11.49 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
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| 12:17.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:19.14 | j_kroon | internal timing? |
| 12:20.01 | j_kroon | SuPrSluG, we try to keep the firmware of the phones up to date ... we've learned the hard way that it's a good idea. but we do run local update servers so that we can at least run a screening check on the firmware before pushing it to clients. |
| 12:20.12 | *** join/#asterisk djin (n=djin@a82-95-147-11.adsl.xs4all.nl) |
| 12:21.50 | fiddur | Hi. Is there a way to verify a config-file before applying it? We had quite a horrible episode here when users.conf happened to get faulty and a sip reload removed all users! |
| 12:24.08 | SuPrSluG | fiddur, use a test box on the network |
| 12:25.47 | *** join/#asterisk rempek (n=mike@aoip1.informatik.fh-kl.de) |
| 12:26.18 | [TK]D-Fender | fiddur: No. |
| 12:26.58 | rempek | hello, i've got a question concerning creating users via the asterisk manager. is it possible (and if yes, how?) to append a template to a new user? |
| 12:26.59 | fiddur | Hmm, then I guess it should be added a verify-command besides all reload-commands... |
| 12:27.52 | fiddur | SuPrSluG: Yes, that's an option to work around it... I was hoping the verification, that obviously is done (since there was an error-message in the debug-log) could be done weparately |
| 12:28.03 | fiddur | separately |
| 12:28.30 | [TK]D-Fender | fiddur: "verify" needs a basis of comparison... what is 8 supopsed to to even judge on? |
| 12:28.51 | [TK]D-Fender | rempek: Normally, no AMI is not a complete tool for remotely configuring * |
| 12:28.58 | [TK]D-Fender | * |
| 12:29.09 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:29.28 | tzafrir_laptop | [TK]D-Fender, check syntax? |
| 12:30.05 | tzafrir_laptop | fiddur, not exactly possible beyond very simplistic tests |
| 12:30.24 | *** join/#asterisk harryv (n=harry@67.207.147.205) |
| 12:30.34 | *** join/#asterisk cm_5_1_2 (n=cm_5_1_2@58.252.229.215) |
| 12:30.38 | tzafrir_laptop | unless you want to make some strange assumptions |
| 12:30.41 | harryv | one of my te121 cards says this in zttool: IRQ Misses: 69791 |
| 12:30.42 | [TK]D-Fender | tzafrir_laptop: well in retrospect to "removed all users", that could be a wrong "include", etc, though no [general] might cause taht as well.... syntax can be fine and yous till fubar yourself. Guess its a question of what kind of screwup you expect to save yourself from. |
| 12:30.44 | harryv | what does that mean? |
| 12:31.29 | SuPrSluG | fiddur, if it's a production environment having a virtual machine or something for testing is a wise move. Or deal with the chance something might not work the way it's supposed to and experiment at convenient times |
| 12:32.29 | fiddur | [TK]D-Fender: It can verify on form... in this case, it was either an encoding error or a character inserted before the first context marking, that sip reload complained about and disregarded the rest of the file. |
| 12:33.37 | [TK]D-Fender | fiddur: That sounds like something that should simply be fixed as a "bug" |
| 12:33.47 | rempek | "Normally no"...so that means there is a (ugly ;) )way to do so? Or definitely impossible? |
| 12:34.13 | [TK]D-Fender | rempek: incredibly ugly. |
| 12:34.24 | [TK]D-Fender | rempek: Nearly epic |
| 12:34.28 | rempek | :D |
| 12:34.46 | [TK]D-Fender | rempek: Stop trying to shove the square block through the round hole... |
| 12:34.59 | rempek | (guess it can't be much worse than how i'm doing it at the moment) |
| 12:35.27 | [TK]D-Fender | rempek: there is very little in life so bad that it can't get worse... |
| 12:36.03 | rempek | true. true. |
| 12:36.40 | tzafrir_laptop | rempek, you can use UpdateConfig and such to manipulate Asterisk config files |
| 12:37.00 | tzafrir_laptop | One thing that this does not play nicely with is configuration tamplates |
| 12:37.51 | fiddur | SuPrSluG: We have a development server too, but haven't used it to verify conf-changes... perhaps we'll do that, and add some auto-tests that verify the sanity of the environment (certain phones exists, lines registered etc)... or, move more to realtime db :) |
| 12:38.02 | *** join/#asterisk diyanat (n=diyanat@59.93.69.230) |
| 12:38.04 | tzafrir_laptop | rempek, so if you have: |
| 12:38.15 | tzafrir_laptop | [template](!) |
| 12:38.19 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 12:38.23 | tzafrir_laptop | foo = bar |
| 12:38.23 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:38.33 | tzafrir_laptop | [a](template) |
| 12:38.57 | tzafrir_laptop | in the manager you'll see section [a] as having 'foo = bar' |
| 12:39.12 | rempek | so, i see 2 possibilities...which one would be better: just put the attributes from template under each entry, or manipulate the conf file via a shell script and reload the conf file? |
| 12:41.30 | [TK]D-Fender | rempek: teplates are a nifty idea, but I'm of a mind that everything should be done in each entry explicitly. |
| 12:41.42 | *** join/#asterisk djin (n=djin@a82-95-147-11.adsl.xs4all.nl) |
| 12:42.32 | rempek | ok, thank you both for your help/advise. |
| 12:42.47 | diyanat | can anyone please help with this zaptel-1.2 compile error on slackware 12.2 kernel 2.6.27 http://pastebin.com/m53a8af3b |
| 12:43.15 | Bladerunner05 | on 1.4.x wich is the sip show peers command ? |
| 12:44.40 | jaytee | huh? |
| 12:44.42 | diyanat | sip show peers |
| 12:44.56 | tzafrir_laptop | diyanat, shortest path would be to use zaptel 1.4 ... |
| 12:45.34 | tzafrir_laptop | Or backport the fixes from there, which I expect wouldn't be difficult |
| 12:45.39 | diyanat | tzafrir i have to use zaptel-1.2 only for vicidial |
| 12:45.45 | lftsy | Hello, is it possible on asterisk to control the use of symmetric codecs in one SIP channel? |
| 12:46.06 | tzafrir_laptop | why does it depend on zaptel 1.2? |
| 12:46.26 | diyanat | performance with zaptel-1.4/asterisk-1.4 is not so good for some reasons |
| 12:46.53 | diyanat | system load is more with 1.4 |
| 12:46.59 | lftsy | or to enforce g711 for download and g729 for upload for example.. |
| 12:47.01 | tzafrir_laptop | Asterisk 1.2 can work with zaptel 1.4 and I know of no performance regressions with zaptel 1.4 |
| 12:47.15 | diyanat | http://www.eflo.net/VICIDIALforum/viewtopic.php?t=4329 |
| 12:48.08 | tzafrir_laptop | And that's asterisk 1.4.10 from 1.5 years ago |
| 12:48.35 | [TK]D-Fender | And we're up to what... 1.4.24 now? |
| 12:48.39 | [TK]D-Fender | +50% |
| 12:48.46 | diyanat | let me try 1.4 and see |
| 12:53.12 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 12:53.23 | pukkita | Hi |
| 12:54.25 | fiddur | I added a queue-setting to disregard queue members penalty if very few queue members are logged in... I haven't gotten any response on it, so I thought maybe I should get some more voices for it here! :) http://bugs.digium.com/view.php?id=14559 |
| 12:54.38 | fiddur | Very few = configurable ammount of course |
| 12:56.00 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:01.36 | [TK]D-Fender | fiddur: Sounds like a good idea to me... |
| 13:03.02 | fiddur | [TK]D-Fender: Yay :) There should be a voting or something in the bug-tracker... |
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| 13:18.18 | *** join/#asterisk filo1234 (n=filo@unaffiliated/filo1234) |
| 13:18.35 | filo1234 | hi all, for to use asterisk i need an service voip from provider? |
| 13:18.39 | filo1234 | or only a simple adsl connection? |
| 13:19.38 | [TK]D-Fender | filo1234: You don't need either. Its a question of what you WANT |
| 13:20.00 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 13:20.01 | [TK]D-Fender | filo1234: Service of any kind usually doesn't come free. |
| 13:21.14 | filo1234 | [TK]D-Fender: hi, i want test asterisk only for internal comunication now, using traditional phones |
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| 13:22.29 | filo1234 | is possible to do that? |
| 13:23.24 | [TK]D-Fender | filo1234: Sure |
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| 13:24.39 | filo1234 | [TK]D-Fender: thanks, where i can find some wiki for to configure it? |
| 13:24.47 | [TK]D-Fender | ~book |
| 13:24.47 | jbot | [~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 |
| 13:24.51 | [TK]D-Fender | ^^ first thing to read. |
| 13:25.05 | [TK]D-Fender | filo1234: there is a WIKI but it should be used as a 3rd place reference only |
| 13:25.09 | [TK]D-Fender | ~wikis |
| 13:25.10 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:27.31 | filo1234 | [TK]D-Fender: can you say me and follow me step by step for to configure one interal phone? :) |
| 13:27.44 | jaytee | hahaha |
| 13:27.51 | filo1234 | :P |
| 13:28.13 | [TK]D-Fender | filo1234: No. but here is a "primer" guide that can show you what a very minimal setup could look like : |
| 13:28.17 | [TK]D-Fender | ~jerjerguide |
| 13:28.18 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 13:29.05 | filo1234 | [TK]D-Fender: ok i'm reading |
| 13:30.12 | [TK]D-Fender | filo1234: Also note this is referring to the first release of the 1.4 series which is not "current'. disregard the install process and follow those in the tarball of whatever version you download. |
| 13:31.55 | filo1234 | [TK]D-Fender: i have installed asterisknow |
| 13:32.06 | filo1234 | 1.5 |
| 13:32.14 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:32.31 | [TK]D-Fender | filo1234: GUI's are not supported here. Please refer tot he channel topic for related channels. |
| 13:33.01 | jaytee | filo1234, try #asterisknow for gui questions |
| 13:33.05 | [TK]D-Fender | filo1234: If thats the route you're going to take |
| 13:33.07 | filo1234 | argh sorry |
| 13:33.47 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 13:34.09 | jaytee | gui versions of asterisk are for girlymen |
| 13:35.01 | tzafrir_laptop | or for clients |
| 13:35.22 | jaytee | yeah, that too |
| 13:35.38 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
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| 13:37.08 | [TK]D-Fender | I am Hans and he is Franz, and we are here to ... PUMP YOU UP! |
| 13:37.32 | DeeJayTwo | IS there a known security issue that can lead to authentify to an asterisk system knowing only the username and not the password? Assuming the password is not trivial. |
| 13:37.46 | russellb | no |
| 13:37.49 | Kobaz | [TK]D-Fender: haha, i remember that |
| 13:38.42 | russellb | DeeJayTwo: unless someone got a capture of a call being authenticated. it doesn't take that long to brute force the md5 hash ... |
| 13:38.53 | [TK]D-Fender | submits "authentify" as a new word to Websters |
| 13:38.54 | russellb | that's not asterisk specific, though. |
| 13:39.04 | DeeJayTwo | russellb : ok |
| 13:39.06 | DeeJayTwo | thanks |
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| 13:43.16 | harryv | what does it mean when the irq for one of my digium cards constantly increases |
| 13:43.23 | harryv | I tried moving it to another pci slot |
| 13:43.27 | harryv | pxi express even |
| 13:45.38 | coppice | I guess it means its working |
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| 13:53.38 | *** join/#asterisk mocker (n=kyle@shell.mocker.org) |
| 13:54.40 | mocker | I'm looking for a way to retrieve the email address set in voicemail.conf for an extension back into a variable in my dialplan. Is there a function that will pull that based on extension? |
| 13:55.50 | yang | mocker: are you asking about voicemail delivery into your mailbox? |
| 13:55.54 | *** join/#asterisk Marquel (n=Marquel@port-5413.pppoe.wtnet.de) |
| 13:56.04 | Marquel | morning. |
| 13:56.33 | mocker | yang: Nope, I want to write a dialplan app that if I hit *EXTENSION while on a call will deliver a recording of that call to the address associated with that extension. |
| 13:57.08 | mocker | So I'm on a call and I hit *1234, in voicemail.conf my extension is 1234 w/ my email associated. I'd like to deliver the recording to that email address. |
| 13:57.08 | Marquel | once more with an evil question: is it possible to connect a voicemailbox with a busy lamp? |
| 13:57.28 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 13:58.56 | harryv | I have a VoiceGuide with a Dialogic D/300JCT-E1 card connected to my asterisk box w/ TE121 with PRI_NET signalling and a span looking like this: 1,0,0,ccs,hdb3,crc4 -- it drops all the time. any ideas for better config? |
| 13:59.42 | [TK]D-Fender | Marquel: Wht kind of "lamp"? |
| 14:00.14 | Marquel | [TK]D-Fender: busy lamp as on the grandstream gxp-20x0 series. |
| 14:00.46 | [TK]D-Fender | mocker: You could always parse it out yourself, but there is no internal * way. You could also set a var in the SIP peer if there is one related to the box. |
| 14:01.07 | [TK]D-Fender | Marquel: Is there somethign wrong with the normal MWI lamp on those? |
| 14:01.32 | mocker | [TK]D-Fender: Ok, just making sure before I wrote some sort of lookup table. :) |
| 14:01.36 | mocker | Thansk |
| 14:01.37 | Marquel | [TK]D-Fender: no, but not sufficient with more than one mailbox ;) |
| 14:01.39 | mocker | Thanks even. |
| 14:01.52 | *** join/#asterisk Laureano (n=Laureano@190.245.121.21) |
| 14:01.57 | [TK]D-Fender | Marquel: You can configure * to monitor multiple boxes. |
| 14:03.02 | Laureano | If I set the debug level of a DAHDI kernel module in 1 (or any other value different from zero), I should see the debugging information in the /var/log/messages file, right? |
| 14:03.10 | Marquel | [TK]D-Fender: as someone said yesterday: w/ * it's almost impossible to find something impossible. now the remaining question is: how ;) |
| 14:03.18 | [TK]D-Fender | mocker: Of course this is easier to do if your VM config is in a DB... then you could jsut func_odbc it or something |
| 14:03.21 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 14:03.40 | [TK]D-Fender | Marquel: add 2 mailbox lines to your SIP peer |
| 14:03.45 | [TK]D-Fender | Marquel: Its that easy |
| 14:03.48 | mocker | [TK]D-Fender: Agreed. |
| 14:03.56 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:05.25 | mocker | [TK]D-Fender: I'll probably switch just for that reason. |
| 14:05.39 | [TK]D-Fender | mocker: How big a setup? |
| 14:05.58 | mocker | [TK]D-Fender: Probably 80 users. |
| 14:06.06 | [TK]D-Fender | mocker: might be worth it... |
| 14:06.19 | mocker | [TK]D-Fender: Yeah, and it's not that hard to switch. |
| 14:06.26 | [TK]D-Fender | mocker: But for the intended scal I'd probably sooner make an AGI for it |
| 14:06.49 | mocker | scal? |
| 14:07.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:07.52 | [TK]D-Fender | scale* |
| 14:08.10 | [TK]D-Fender | mocker: 1-letter typo... geez! Get some intuition! |
| 14:08.31 | mocker | hah |
| 14:08.55 | mocker | I was trying to come up with how this related to some new ical format I hadn't heard of. |
| 14:09.58 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 14:10.00 | Gopaul | i get this error chan_dahdi.c:10059 dahdi_ss7_error: !! Unable to handle message of type 0x3 on CIC 1 when dialing a outbound number from my asterisk machine |
| 14:11.48 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
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| 14:12.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:13.21 | tzafrir_laptop | Laureano, not sure. Some of the debug messages are of priority "debug" and thus won't show up in /var/log/messages |
| 14:13.36 | tzafrir_laptop | On Debian / Ubuntu you'll see them in /var/log/kern.log |
| 14:13.47 | tzafrir_laptop | also see the output of dmesg |
| 14:15.18 | Laureano | Thank you tzafrir_laptop. I will check that files and dmesg. |
| 14:15.31 | mocker | [TK]D-Fender: Actually this seems like an even better solution: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg199590.html |
| 14:15.38 | mocker | just plop it in a voicemail box |
| 14:18.24 | Kobaz | http://pastebin.com/m613563e9 |
| 14:18.25 | Kobaz | is that a dahdi problem, or a wanpipe problem |
| 14:18.25 | Kobaz | when i restart wanpipe and it loads the firmware and etc, it's waiting for /dev/zap to come up |
| 14:18.25 | Kobaz | i symlinked /dev/dahdi to /dev/zap, and now the script is happy |
| 14:18.59 | Kobaz | i dont see a "Waiting for..." in the wanrouter startup script though |
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| 14:33.50 | Kobaz | what the hell |
| 14:33.51 | Kobaz | dev {/apps} root# /etc/init.d/asterisk start |
| 14:33.51 | Kobaz | Starting Asterisk PBX: The canary is no more. He has ceased to be! He's expired and gone to meet his make! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off thtwig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleding choir invisible!! THIS is an EX-CANARY. (Reducing priority) |
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| 14:44.37 | Kobaz | now that everyon's back |
| 14:44.37 | Kobaz | maybe not |
| 15:48.23 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 15:48.23 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.24 (2009/03/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 15:49.35 | mort_gib | http://www.pbx.in/voip-info/wiki/view/Asterisk+deadlock.html yes this is exactly what I have!! |
| 15:50.28 | c0rnoTa | hello all |
| 15:57.00 | c0rnoTa | still getting "astobj2.c: user_data is NULL" in asterisk 1.6.1.0-rc2 |
| 15:57.56 | c0rnoTa | it's a bug, but I couldn't describe it |
| 15:58.44 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 15:58.45 | c0rnoTa | and didn't know how to reproduce |
| 15:58.55 | vader-- | hey do you guys know if its possible to reset a vonage cisco ata? |
| 15:59.09 | vader-- | I have an old one i want to use now at work to play around with |
| 15:59.36 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:00.06 | c0rnoTa | but it appears when I'm absent on my work place |
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| 17:32.27 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.24 (2009/03/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 17:34.01 | *** join/#asterisk swathanthran (n=user@unaffiliated/shyam-k/x-8459115) |
| 17:34.28 | Defraz | what protocol or packet type does the qualify feature use? |
| 17:34.37 | TommyBJ | fiddur: AJAM. Is it simple to handle? |
| 17:34.46 | Defraz | when qualifing a sip device or phone |
| 17:34.59 | TommyBJ | Defraz: It goes through the session layer. |
| 17:35.39 | fiddur | TommyBJ: http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) if you just activate the http-server and go in to the host:8080/static/ajamdemo.html |
| 17:35.39 | *** join/#asterisk sinein (n=kvirc@adsl-68-255-97-56.dsl.chcgil.ameritech.net) |
| 17:35.40 | Defraz | okay I have like 5 phones that when qualify is on they don't work butwhen I turn it off for the extensions they work and ring |
| 17:35.45 | Defraz | they can make calls out no problem. |
| 17:36.11 | ncom | TommyBJ and UQlev thanks that did the trick :) |
| 17:36.15 | Defraz | I am wondering if I am blocking something on the ASA, I have a differenet route for each network. |
| 17:36.18 | Mark21 | UQlev: can you reach the sip server (asterisk) on ip 195.184.64.194? |
| 17:36.55 | TommyBJ | fiddur: Thank you. Much appreciated :-) |
| 17:37.04 | sinein | hey guys , is the backbone of the cell phone system wired through the public telecoms switching network? - meaning that at any point in time a cell phone conversation will run through physical transmission mediums? |
| 17:37.19 | UQlev | Mark21: Host 195.184.64.194:5060: (ICMP timeout) SIP timeout |
| 17:37.29 | Mark21 | sinein: yes |
| 17:37.32 | fiddur | TommyBJ: or even /asterisk/manager?action=login&username=foo&secret=bar and then run the command there MeetmeList |
| 17:37.38 | Mark21 | UQlev: strange :S |
| 17:38.10 | UQlev | Mark21: I can get sip from sip1.netphonecom.nl |
| 17:38.18 | sinein | Mark21: what if I have a conversation in the same cell? or not far away? |
| 17:38.51 | Mark21 | sinein: than it often stays on the system you are connected through |
| 17:39.29 | Mark21 | UQlev: that is another system from someone else i am normally connected to |
| 17:39.37 | sinein | so would a call in that scenario be wired or would it all be wirless thorugh the base station antenna? |
| 17:39.51 | Mark21 | UQlev: wireless |
| 17:40.28 | sinein | thanks :) |
| 17:40.56 | Mark21 | sinein: wireless |
| 17:41.09 | Mark21 | UQlev: what could be the problem? |
| 17:41.14 | UQlev | Mark21: you may use smap to check accessibility to your sip |
| 17:41.45 | UQlev | Mark21: I guess problem with firewall/router if any in between |
| 17:42.10 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 17:42.17 | Mark21 | the firewall looks correct and i will shut it down to test if it is the firewall |
| 17:42.58 | vader-- | Does anyone know how to reset a vonage cisco ATA? I used to have vonage and now i don't, i want to use the old ATA with my asterisk system at work? Can't seem to find anyway to just remove the password and reset the unit to factory settings. |
| 17:43.01 | UQlev | Mark21: then shut it down |
| 17:43.38 | [TK]D-Fender | vader--: http://www.google.ca/search?hl=en&q=unlock+vonage+cisco+ata&btnG=Google+Search&meta= |
| 17:43.49 | Mark21 | UQlev: well i cannot find an option to shut it down |
| 17:43.51 | drmessano | Google.ca? |
| 17:43.54 | drmessano | Communist |
| 17:43.57 | Vec | Where can I get a sample mp3 file for MOH in asterisk ? |
| 17:44.02 | [TK]D-Fender | Socialist |
| 17:44.04 | Vec | I free one I can download ? |
| 17:44.16 | drmessano | <-- Obamanist |
| 17:44.16 | UQlev | Mark21: is it centos? |
| 17:44.17 | [TK]D-Fender | vec* COMES with samples |
| 17:44.58 | Vec | [TK]D-Fender : they are WAV ? |
| 17:45.05 | vader-- | tkd i checked those articles out |
| 17:45.11 | [TK]D-Fender | Vec: Many formats availabl;e |
| 17:45.14 | jaytee | <-- Paulist |
| 17:45.19 | [TK]D-Fender | RP 2008!!!! |
| 17:45.21 | Vec | Vec : yeh, well can't find mp3 :P |
| 17:45.27 | jaytee | He Knew!!! |
| 17:45.37 | [TK]D-Fender | Vec : What do you really need MP3 for? |
| 17:45.44 | jaytee | He Warned Us But No One Listened! |
| 17:46.14 | jaytee | for all those Neil Sedaka tunes he wants to use for MOH |
| 17:46.28 | drmessano | Ron Paul/Steve Jobs 2012 "I want to believe" |
| 17:46.31 | TommyBJ | fiddur: Sorry to bother you. But do you know how the MeetMeList command works? |
| 17:46.37 | TommyBJ | fiddur: It's lacking documentation. |
| 17:46.46 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 17:46.57 | Vec | [TK]D-Fender : I am having a strange problem, when I use Playback everything works fine, when I use MOH, the sound breaks up every once and a while, this has something to do with silence detection on a ISDN->SIP gateway I have, I am trying to isolate the reason why it is fine on Playback and not with MOH, so I wanted to test with mp3? |
| 17:47.04 | fiddur | TommyBJ: try manager show command MeetmeList |
| 17:47.23 | [TK]D-Fender | Vec: if ti "playback"'s OK, then its not the file. |
| 17:47.30 | [TK]D-Fender | Vec: because everything transcodes the same. |
| 17:47.48 | [TK]D-Fender | Vec: Unless of course you're using an external app that's FUBAR-ing things. |
| 17:47.52 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 17:47.54 | [TK]D-Fender | Vec: at which point use Native MoH |
| 17:48.04 | pukkita | hiya all |
| 17:48.06 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 17:48.32 | pukkita | having issues with line supervision with thomson st2030, it worked fine with a 1.2.13 server but not with a 1.4.18 one :( |
| 17:48.44 | Mark21 | UQlev: debian |
| 17:48.46 | pukkita | fw 1.64.2 |
| 17:48.49 | Mark21 | but iptables is stopped |
| 17:48.58 | jplank | stupid hackers from china trying to break into my DNS servers! |
| 17:48.59 | Mark21 | and it stills seems not to work correctly |
| 17:49.03 | fiddur | TommyBJ: hmm, since "MeetmeList will follow as separate events, followed by a final event called", I guess the demo manager-interface is insufficiant... I guess a simple application is better then. |
| 17:49.25 | TommyBJ | fiddur: Perl it is then? :) |
| 17:49.42 | UQlev | Mark21: "netstat -an | grep 5060" |
| 17:50.58 | UQlev | jplank: dns-cache poisoning |
| 17:51.14 | jplank | no, they were trying to root the box |
| 17:51.22 | jplank | with a brute force |
| 17:51.26 | Mark21 | UQlev: |
| 17:51.26 | Mark21 | tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN |
| 17:51.26 | Mark21 | udp 109548 0 195.184.64.194:5060 0.0.0.0:* |
| 17:51.33 | Mark21 | that look normall to me |
| 17:51.35 | jplank | little did they know, root only has local login rights |
| 17:51.49 | fiddur | TommyBJ: In my choice it would be :) But, I've developed the Asterisk::Manager quite a lot lately, and I think the published version is quite crappy... but it might work for a really simple app. |
| 17:51.52 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 17:51.55 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 17:52.32 | TommyBJ | fiddur: Is it in CPAN? |
| 17:52.42 | UQlev | Mark21: yes it looks normal |
| 17:53.04 | *** join/#asterisk bob_slacker (n=bob_slac@189.27.12.11) |
| 17:53.20 | UQlev | jpcansa: you mean dictionary ssh attack? |
| 17:53.26 | bob_slacker | hack me! my ip: 127.0.0.1 |
| 17:53.32 | fiddur | TommyBJ: the old version is, yes... |
| 17:53.59 | UQlev | jplank: you mean dictionary ssh attack? |
| 17:54.08 | TommyBJ | fiddur: Old? ... is there a new one available? |
| 17:54.31 | jplank | yes |
| 17:54.35 | jplank | who would of thunk it |
| 17:55.07 | drmessano | Hack me first: 192.168.1.2 |
| 17:55.19 | fiddur | TommyBJ: Well, I haven't released the modified version I use locally... but I think it will be an alternative, not a replacement, since it relies on more external modules, like Coro and AnyEvent... |
| 17:55.30 | UQlev | jplank: those are not obviously chinese, scans are from zombies |
| 17:55.34 | fiddur | TommyBJ: If you want it I can mail it or something... |
| 17:55.34 | Gopaul | ERROR[7175]: chan_dahdi.c:10059 dahdi_ss7_error: !! Unable to handle message of type 0x3 on CIC 1 i am getting like this errror in asterisk |
| 17:55.37 | jplank | seems to be a compromised computer trying to hack me - 219.134.242.67 |
| 17:56.10 | *** join/#asterisk pdani (n=pdani@catv-89-133-156-227.catv.broadband.hu) |
| 17:56.11 | Vec | [TK]D-Fender : its rether odd, because I am not using an external app but with MOH it doesn't work but with playback it does. |
| 17:56.16 | Mark21 | UQlev: can you now connect to 195.184.64.194 (i did try telnet and i did get a connection) |
| 17:56.24 | jplank | UQlev: exactly |
| 17:56.25 | UQlev | jplank: is it your 1st look into ssh.log? ;) |
| 17:56.29 | smooth_penguin | hey how do I make a sip call to an extention from the asterisk console? |
| 17:56.42 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 17:56.44 | drmessano | UQlev: First hack attempt, I would guess |
| 17:56.49 | smooth_penguin | and do I need a sip client if I want one on the asterisk server? |
| 17:56.58 | smooth_penguin | or can i use asterisk itself? |
| 17:57.08 | [TK]D-Fender | Vec: Would be nice to see... |
| 17:57.17 | jplank | UQlev: of the sort. They were hitting 22, and 21 at the same time |
| 17:57.32 | _pepo_ | what do you think? Can be Asterisk my voicemail system for x>120000 users? |
| 17:57.57 | bob_slacker | _pepo_, sure |
| 17:57.58 | UQlev | jplank: welcome to jungles of internet :)) |
| 17:58.12 | jplank | the kick in the ass was they were using the username root on port 21 to :P |
| 17:58.30 | bob_slacker | jplank, change ssh telnet ports |
| 17:58.47 | _pepo_ | what do you think that is better, use voicemail.conf, MySQL or Pgsql? |
| 17:58.47 | UQlev | Mark21: I can ssh but I can't sip |
| 17:58.48 | jplank | can't unfortunately, would screw up our NOC |
| 17:59.31 | Gopaul | ERROR[7175]: chan_dahdi.c:10059 dahdi_ss7_error: !! Unable to handle message of type 0x3 on CIC 1 Why is this error? |
| 17:59.45 | jplank | i'm sorry I should elaborate why I find this surprising. I have a sandbox on the same subnet that wasn't touched |
| 17:59.56 | drmessano | Are we really dedicating this much bandwidth to someone trying to brute force SSH? Can we talk about something less common, like Internet Explorer vulnerabilities or "Holy crap, firefox just crashed".. pleeeeease |
| 18:00.08 | Mark21 | UQlev: did you ever see that before? |
| 18:00.13 | TommyBJ | fiddur: That would be very nice |
| 18:01.17 | UQlev | Mark21: on my servers I would guess only routers because I can disable firewall |
| 18:01.20 | drmessano | ZOMG someone is trying to connect to my mail server and open relay some spam.. ZOMG |
| 18:01.32 | _pepo_ | what do you think that is better for voicemail: voicemail.conf, MySQL or Pgsql? |
| 18:02.18 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:02.56 | UQlev | Mark21: but I have never used VPS, I have 1 server colocated with ISP and all others in my office |
| 18:02.59 | jplank | well drmessano, you shouldn't be running an open mail relay |
| 18:03.12 | *** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) |
| 18:03.21 | ScriptFanix | hi |
| 18:03.24 | drmessano | jplank: Duh |
| 18:03.52 | Vec | When using chanspy(), normally, (i.e. not in whisper or private whisper mode), can the person being spied on here the person spying ? i.e. is it like a forced conference ? |
| 18:04.26 | UQlev | drmessano: it is constantly happens to any smtp server in the world |
| 18:04.33 | Gopaul | how to solve the dadhi ss7 error from my asterisk console, it with some cic error |
| 18:04.39 | drmessano | wow |
| 18:05.19 | drmessano | Is there anyone in here, currently, with a sense of humor |
| 18:05.24 | smooth_penguin | hey how do I call any extention from the asterisk console? |
| 18:05.24 | drmessano | nm |
| 18:05.53 | [TK]D-Fender | smooth_penguin: "help console dial" |
| 18:05.59 | smooth_penguin | ok |
| 18:06.04 | UQlev | smooth_penguin: dial 1234 |
| 18:06.25 | smooth_penguin | No such command 'dial 999' (type 'help dial 999' for other possible commands) |
| 18:06.43 | smooth_penguin | its console dial |
| 18:06.45 | smooth_penguin | ah |
| 18:08.03 | japherwocky | drmessano: this channel is for srs business only, pls |
| 18:08.19 | drmessano | japherwocky: No kidding.. |
| 18:08.33 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:08.58 | drmessano | japherwocky: Like ZOMG, someone put bad code on a website that I just visited and it tried to do bad stuff, or, "People try to hack SSH? ZOMG WHY: |
| 18:09.07 | vader-- | anyone else have any ideas how i can unlock this vonage cisco ata 186? |
| 18:09.09 | drmessano | Guess I misread the sign on the door |
| 18:09.35 | jplank | drmessano: I wasn't serious |
| 18:10.00 | ScriptFanix | I have some strange problem with a queue. The call shall timeout after 30 sec, but it rings for more than one minute http://pastebin.com/d31732b53 (caller hanged up) |
| 18:10.34 | *** join/#asterisk Strom (n=stromcar@netblock-75-79-35-51.dslextreme.com) |
| 18:10.42 | smooth_penguin | hrm I cant seem to be getting any sound from asterisk |
| 18:10.47 | smooth_penguin | when I can hear my mic |
| 18:11.10 | smooth_penguin | others on other extentions can hear me |
| 18:11.16 | smooth_penguin | but I cant hear them |
| 18:11.20 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:11.26 | smooth_penguin | I just did a tt-monkeys test |
| 18:11.47 | UQlev | smooth_penguin: did you try echo test? |
| 18:11.56 | smooth_penguin | UQlev, yep even that |
| 18:12.00 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 18:12.02 | smooth_penguin | no echo |
| 18:12.10 | smooth_penguin | but others get it |
| 18:12.19 | smooth_penguin | Im on the same box as asterisk |
| 18:12.28 | UQlev | smooth_penguin: are your client in the same subnet with asterisk? |
| 18:12.35 | smooth_penguin | UQlev, yes |
| 18:12.39 | drmessano | lol |
| 18:12.56 | smooth_penguin | its odd |
| 18:13.09 | smooth_penguin | I can play music fine so it isnt my snd card |
| 18:13.30 | UQlev | smooth_penguin: I just wonder how you client and astaris can listen the same ports e.g. 5060 |
| 18:13.42 | drmessano | holy crap |
| 18:13.56 | drmessano | [14:06] <smooth_penguin> hey how do I call any extention from the asterisk console? [14:06] <UQlev> smooth_penguin: dial 1234 |
| 18:13.59 | smooth_penguin | UQlev, I had changed the port on the client to 5061 |
| 18:14.01 | Gopaul | anybody has ss7 error |
| 18:14.07 | smooth_penguin | then I even tried calling via the console |
| 18:14.35 | Strom | Is the polycom autoprovisioning stuff in 1.6 worth using at all, or is it too flaky and unreliable to waste my time with? (in any event, it's documented like shit) |
| 18:14.39 | tzafrir_laptop | 'console dial' uses the local sound card and microphone |
| 18:14.58 | tzafrir_laptop | 'originate' tells Asterisk to generate a call (with your chaice of device) |
| 18:16.03 | Defraz | when asterisk qualifies a registration using the qualify=yes is that icmp or what? |
| 18:16.52 | vader-- | anyone else have any ideas how i can unlock this vonage cisco ata 186? |
| 18:17.13 | UQlev | Defraz: it seems not icmp because on my box icmp is blocked |
| 18:17.16 | drmessano | vader--: Console? |
| 18:17.37 | Defraz | hmm wonder what it uses. |
| 18:17.47 | UQlev | Defraz: probably same udp |
| 18:17.54 | Defraz | 5060? |
| 18:18.11 | UQlev | I didn't debug it |
| 18:18.13 | smooth_penguin | damn I cant hear the monkeys :| |
| 18:19.05 | Strom | Defraz: it sends a SIP OPTIONS message to the phone and times how long a response takes (IIRC) |
| 18:19.06 | [TK]D-Fender | smooth_penguin: Just because you can hear your mic doesn't mean the rest of your mixer settings are sane. |
| 18:20.44 | smooth_penguin | [TK]D-Fender, I can also play mp3 fine |
| 18:21.15 | [TK]D-Fender | smooth_penguin: And you're not showing me whats happening. |
| 18:22.00 | [TK]D-Fender | smooth_penguin: console dial SHOULD fail because * requires complete control over your sound card and if other stuff si working, the console SHOULDN'T |
| 18:23.37 | smooth_penguin | so no other app should be able to use the snd card if * is running? |
| 18:24.12 | Laureano | tzafrir_laptop, in the man page of dahdi_scan there is a reference to the file /dev/zap/ctl. Shouldn't be /dev/dahdi/ctl? (phrase "Requires read access to /dev/zap/ctl") |
| 18:26.01 | [TK]D-Fender | smooth_penguin: One or the other |
| 18:26.09 | smooth_penguin | hrm |
| 18:26.35 | [TK]D-Fender | smooth_penguin: Go use a soft-phone, and show us a failed call, and no audio can be a lot of things including firewalls, etc |
| 18:26.37 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:30.13 | tzafrir_laptop | Laureano, what are you talking about? I don't see that. http://docs.tzafrir.org.il/dahdi-tools/man/dahdi_scan.8.html |
| 18:30.57 | tzafrir_laptop | That is to say: fixed. thanks :- ) |
| 18:32.29 | *** join/#asterisk bbryant (n=Brett_Br@209.192.100.193) |
| 18:33.06 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 18:33.58 | Laureano | tzafrir_laptop, glad to hear that is fixed. Thanks to you. |
| 18:36.34 | *** join/#asterisk ew01f (n=chatzill@201.170.12.47) |
| 18:36.57 | Strom | Is the polycom autoprovisioning stuff in 1.6 worth using at all, or is it too flaky and unreliable to waste my time with? (in any event, it's documented like shit) |
| 18:38.10 | smooth_penguin | [TK]D-Fender, http://pastebin.com/m53a5fa29 |
| 18:38.14 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:38.20 | smooth_penguin | 999 is my tt-monkeys extention |
| 18:38.41 | smooth_penguin | I can see the audio levels in ekiga moving , but cant hear anything |
| 18:38.49 | smooth_penguin | <PROTECTED> |
| 18:38.50 | smooth_penguin | <PROTECTED> |
| 18:39.00 | smooth_penguin | those are the two lines I get in the asterisk console |
| 18:39.03 | smooth_penguin | when i call up |
| 18:39.07 | Strom | smooth_penguin: are you running the softphone on the same machine as you're running asterisk on? |
| 18:39.13 | [TK]D-Fender | smooth_penguin: then ekiga / your audio has a problem |
| 18:39.17 | smooth_penguin | Strom, yes |
| 18:39.29 | Strom | smooth_penguin: *facepalm* don't do that :P |
| 18:39.29 | [TK]D-Fender | Strom: Long time no see... |
| 18:39.39 | Strom | likewise, [TK]D-Fender |
| 18:39.42 | smooth_penguin | Strom, i cant even call via console :\ |
| 18:39.43 | [TK]D-Fender | Strom: Not normally a problem.... |
| 18:39.57 | smooth_penguin | this is a stupid issue |
| 18:40.15 | smooth_penguin | asterisk is working fine |
| 18:40.18 | [TK]D-Fender | smooth_penguin: Go call ekiga.net 's echo service and test things. Check your firewalls |
| 18:40.34 | smooth_penguin | ah good idea |
| 18:43.42 | Strom | so...i'm guessing the lack of responses means that the polycom autoprovisioning is a waste of time |
| 18:44.51 | Strom | *cricket noises* |
| 18:45.34 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 18:48.29 | *** join/#asterisk qdk (n=qdk@213.173.230.10) |
| 18:51.07 | ncom | I am using asterisk in conjunction with freepbx, where would I implmement jitterbuffer? |
| 18:52.18 | vader-- | anyone have any ideas how i can unlock this vonage cisco ata 186? |
| 18:53.10 | ncom | anyone with suggestions pls PM me, I have to walk away for a few |
| 18:55.06 | [TK]D-Fender | Strom: huh? |
| 18:55.10 | jpcansa | is it normal that a IAX client keeps registering on different ports? http://pastebin.com/m49b640fc ? |
| 18:56.28 | Strom | [TK]D-Fender: building a new asterisk box after mostly ignoring my old one for a year, and i'm trying to figure out whether to use the polycom autoconfig feature or to just provision them the old-fashioned way |
| 18:56.48 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 18:56.49 | [TK]D-Fender | Strom: Not familiar with "autoconfiguring" |
| 18:57.12 | Strom | new feature in 1.6, supposedly |
| 18:57.25 | Strom | there's shit for documentation, though |
| 18:57.25 | [TK]D-Fender | Strom: Oh that crap? :p Just build your provisioning for you |
| 18:57.49 | [TK]D-Fender | Strom: curious parallel :p |
| 18:57.54 | Strom | ? |
| 18:58.04 | [TK]D-Fender | Strom: It sucks, and so does its docs |
| 18:58.17 | [TK]D-Fender | Strom: just builds provisioning files to be fed over HTTP |
| 18:58.18 | Strom | well, yeah, that was generally the conclusion i was making |
| 19:01.38 | Qwell | Strom: randomly found something you did the other day... |
| 19:01.56 | Qwell | Strom: http://www.woz.org/seg/parodies.html |
| 19:02.06 | Qwell | "wtf? is that...Strom?" |
| 19:02.15 | *** join/#asterisk path_ (n=path_@242-125-21-190.adsl.terra.cl) |
| 19:02.47 | Strom | Qwell: hehe, that was a long time ago :P |
| 19:03.50 | Qwell | Strom: You'll be happy to know that I got there from the best page in the universe. |
| 19:04.18 | Qwell | (through a maze of links at 2am) |
| 19:05.17 | Strom | I see |
| 19:11.15 | *** join/#asterisk clive- (i=ident@dsl-242-179-98.telkomadsl.co.za) |
| 19:12.07 | clive- | hi, any cisco guys around? I am trying to figure out how to get this cisco router to send sip calls to asterisk |
| 19:15.22 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 19:16.02 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.75) |
| 19:18.31 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 19:20.16 | Mark21 | is the following normal? netstat -an | grep 5060 |
| 19:20.17 | Mark21 | udp 9180 0 0.0.0.0:5060 0.0.0.0:* |
| 19:21.12 | Qwell | Mark21: sure, what's wrong with it? |
| 19:21.32 | bmoraca | clive-: you're trying to trunk from CCM or CME to asterisk? |
| 19:22.47 | Mark21 | Qwell: from outside no one can connect |
| 19:23.27 | Qwell | where is "outside"? |
| 19:23.28 | Mark21 | the configs did work on an other server without any problems |
| 19:23.35 | Qwell | another server that wasn't behind NAT? |
| 19:23.46 | Mark21 | Qwell: multiple networks on the internet |
| 19:24.02 | Mark21 | the server and the system trying to connect aren't behind NAT |
| 19:24.08 | Mark21 | both have public ips |
| 19:24.11 | Qwell | iptables? |
| 19:24.26 | Mark21 | Qwell: i did: iptables -X and iptables -F |
| 19:24.30 | Mark21 | it is on Debian |
| 19:24.32 | Qwell | default policy? |
| 19:24.57 | Mark21 | Qwell: where can i check that? i can put the output from iptables-save online if that helps |
| 19:25.06 | *** join/#asterisk qdk (n=qdk@0x55817ca1.terminal.tdcmobil.dk) |
| 19:25.08 | Qwell | iptables -L |
| 19:25.19 | Qwell | and also -L -t nat |
| 19:25.32 | Mark21 | policy ACCEPT |
| 19:25.44 | Mark21 | for all 3 options |
| 19:25.48 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-127-rbry-bb-1.dynamic.isadsl.co.za) |
| 19:25.48 | Mark21 | and no other lines |
| 19:25.58 | Mark21 | it is not behind NAT |
| 19:26.19 | Mark21 | also policy ACCEPT |
| 19:26.34 | Qwell | How is the server connected to the internet? |
| 19:26.55 | Mark21 | it is connected to a switch and from there to a router with multiple transits (all wired) |
| 19:27.15 | Qwell | and is the router correctly forwarding stuff on? |
| 19:27.24 | Qwell | (router = NAT btw) |
| 19:28.00 | Mark21 | router != NAT |
| 19:28.10 | Mark21 | the router is just a BGP router like all networks use |
| 19:28.29 | Qwell | NAT doesn't imply RFC 1918 |
| 19:28.33 | clive- | bmaraca , I am not sure what CCM or CME are, but its PSTN----E1--(cisco)----SIP-----Asterisk |
| 19:28.36 | [TK]D-Fender | Mark21: so enable SIP debug and look for packets |
| 19:28.44 | clive- | bmaraca , I am not sure what CCM or CME are, but its PSTN----E1--(cisco)----SIP-----Asterisk |
| 19:28.49 | clive- | oops |
| 19:29.07 | Mark21 | [TK]D-Fender: packets are send by the asterisk server but nothing comes back for as far as i can see (also not from a server on the same switch) |
| 19:29.17 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 19:29.32 | [TK]D-Fender | Mark21: PASTEBIN is your friend.... |
| 19:29.44 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:29.48 | [TK]D-Fender | ~pb |
| 19:29.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:30.04 | [TK]D-Fender | Mark21: Show us and maybe we'll have something to comment on. |
| 19:30.27 | bmoraca | clive-: CCM and CME are different Call Manager implementations that may be available on your router depending on the model and what IOS it has loaded. Without that, you can't use a Cisco router as a gateway for SIP-to-PSTN traffic |
| 19:30.28 | Mark21 | [TK]D-Fender: sip debug info is now created in a logfile and i will put that logfile online |
| 19:30.46 | [TK]D-Fender | Mark21: Live CLI only w/ SIP DEBUG |
| 19:31.13 | Mark21 | that information is also in the logfile available [TK]D-Fender |
| 19:31.37 | jaytee | god I hate Windows |
| 19:32.16 | jaytee | and the new Dell Optiplex SX-755 |
| 19:32.33 | jaytee | and the Intel Trusted Computing Initiative chipset |
| 19:32.52 | jaytee | and overcooked spinach |
| 19:32.58 | Strom | jaytee: at this point, you might as well open a boutique |
| 19:33.04 | jaytee | and people that talk in the theater |
| 19:33.18 | bmoraca | TCPA needs to die |
| 19:33.42 | jaytee | it needs to be stabbed to death with a rusty dull spork |
| 19:33.50 | Mark21 | [TK]D-Fender: http://yourpaste.net/1799/ << output from the cli |
| 19:34.30 | bmoraca | Mark21: what's your SIP.conf look like? |
| 19:34.57 | bmoraca | (otherwise known as "pastebin your sip.conf") |
| 19:35.02 | *** join/#asterisk dni (n=humberto@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 19:35.29 | [TK]D-Fender | Mark21: So where is this Cisco in three? |
| 19:35.43 | [TK]D-Fender | there* |
| 19:35.45 | bmoraca | [TK]D-Fender: different person |
| 19:36.07 | [TK]D-Fender | oops |
| 19:36.16 | [TK]D-Fender | Mark21: [Mar 26 20:32:10] Reliably Transmitting (NAT) to 194.221.62.198:5060: <-- shouldn't be NAT <- |
| 19:36.47 | bmoraca | that's why i wanted to see his sip.conf...he's specifying that the peers are behind/his server's behind a NAT and it's fubaring |
| 19:37.02 | Mark21 | bmoraca: moment, i will upload it somewhere after i have changed the passwords to 12345678 |
| 19:37.07 | bmoraca | Polycom phones don't care about that, incidentally, they can figure it out for themselves...but other phones i've notice have problems. |
| 19:37.13 | *** part/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net) |
| 19:37.32 | bmoraca | Mark21: no problem with obfuscating your sensitive info |
| 19:39.30 | Mark21 | other info is no problem ;) |
| 19:39.41 | [TK]D-Fender | Mark21>Qwell: from outside no one can connect |
| 19:39.47 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:39.56 | [TK]D-Fender | Mark21: I frankly don't see an incoming register from anything remote there. |
| 19:41.31 | [TK]D-Fender | Mark21: What kind of devices should be trying to reg with your *? |
| 19:47.23 | Mark21 | http://89.248.170.34/sip.conf << my sip.conf |
| 19:48.19 | Mark21 | [TK]D-Fender: another asterisk server, x-lite clients (2), a few hardphones (cisco or linksys) and 2 ata boxes (linksys pap2t) |
| 19:48.22 | [TK]D-Fender | Mark21: ; route = Assume NAT, don't send rport <- bad |
| 19:48.45 | bmoraca | Mark21: if your system truely is on the public internet, nuke your localnet statements and change all your nat settings to "no" |
| 19:48.48 | [TK]D-Fender | Mark21: Plus all of the stuff in [general] yet under your REGiSTER lines gets ignored |
| 19:49.09 | [TK]D-Fender | Mark21: Andtrash all that sample commentary |
| 19:49.13 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 19:49.14 | *** join/#asterisk bartpbx (n=bartpbx@62.216.165.71) |
| 19:49.20 | bartpbx | hello |
| 19:50.23 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 19:51.24 | [TK]D-Fender | Mark21: And Your SIP debug from earlier didn't seem to have any remote devices trying to call in or register.... |
| 19:51.38 | Mark21 | they cannot connect somehow |
| 19:51.53 | Mark21 | but i know they are trying to connect |
| 19:52.07 | [TK]D-Fender | Mark21: we see other traffic comgin in.. that tells me you configured those other devices wrong or they have different issues |
| 19:52.12 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 19:52.18 | [TK]D-Fender | Mark21: just that it isn't * side |
| 19:52.33 | [TK]D-Fender | Mark21: you have no packets from them, but do from others. So THEY are bad. |
| 19:52.37 | *** join/#asterisk neurosys (n=neurosys@173.9.159.182) |
| 19:52.51 | ncom | I am using asterisk in conjunction with freepbx, where would I implement jitterbuffer? |
| 19:53.24 | *** join/#asterisk [psy] (n=psy0rz@lounge.datux.nl) |
| 19:53.24 | [TK]D-Fender | ncom: Depends on the protocol |
| 19:53.30 | ncom | IAX2 |
| 19:53.49 | [TK]D-Fender | ncom: In your IAx2 peer then normally. |
| 19:54.46 | ncom | should i enable it on trunks too? |
| 19:55.15 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:55.48 | [TK]D-Fender | ncom: http://bugs.digium.com/file_download.php?file_id=5250&type=bug |
| 19:57.09 | [psy] | so i made a automated testscript with sipp and asterisks, to detect regressions and glibc/gcc problems. where should i post a link so people can use it? |
| 19:57.41 | [psy] | we already found out asterisks with glibc2.3.2 has problems :) |
| 19:58.48 | Mark21 | [TK]D-Fender: why did it work earlier today and not now? they are connecting to the right IP (and why is another asterisk server not connecting?)? |
| 19:59.10 | bmoraca | Mark21: did you fix those various issues we pointed out in sip.conf? |
| 19:59.19 | [TK]D-Fender | Mark21: Nothing for us to see. Packets aren't coming in. Look at the OTHER end. |
| 19:59.49 | [TK]D-Fender | bmoraca: While those points aren't good, they're not really relevent either... |
| 20:00.08 | [TK]D-Fender | bmoraca: No incoming communications is the other end's fault |
| 20:00.13 | bmoraca | [TK]D-Fender: they could be. any potential error could be bad. |
| 20:00.13 | Mark21 | [TK]D-Fender: I am ALSO looking at the other end (at the asterisk server) and also no information at that system to see |
| 20:00.25 | Mark21 | or it is dropped somewhere between both point |
| 20:00.26 | Mark21 | or it is dropped somewhere between both points |
| 20:00.33 | [TK]D-Fender | Mark21: Could be |
| 20:00.41 | [TK]D-Fender | Mark21: Some ISP's filter SIP |
| 20:00.54 | elred | Hello, do you guys see any solution to Dial all SIP registered phone from an AGI ? |
| 20:00.55 | [TK]D-Fender | Mark21: Could be anything (esp as we see nothing) |
| 20:01.12 | Mark21 | [TK]D-Fender: they are both on the same switch (same subnet) so that shouldn't be the problem |
| 20:01.14 | [TK]D-Fender | elred: how many? |
| 20:01.16 | bmoraca | [TK]D-Fender: i've noticed that SIP Debug Peer doesn't work if asterisk doesn't have an IP to reference, and then nothing happens. if the server isn't properly configured for inbound connections, it may not know about a peer's attempts to register |
| 20:01.16 | elred | like, how to retrieve the current registered extension so that I can build my Dialstring |
| 20:01.25 | elred | [TK]D-Fender : not more than a dozen |
| 20:01.50 | [TK]D-Fender | bmoraca: Who said he we restricting debug to a specific peer? I certainly wouldn't |
| 20:02.22 | [TK]D-Fender | elred: "core show application chanisavail" <- |
| 20:02.49 | [TK]D-Fender | s/we/was |
| 20:02.58 | elred | [TK]D-Fender : thx I will try that |
| 20:03.22 | [TK]D-Fender | bmoraca: that would be a completely retarded thing to do in trying to debug this |
| 20:03.30 | bmoraca | true |
| 20:03.43 | *** join/#asterisk sasargen (n=chatzill@174-153-61-162.pools.spcsdns.net) |
| 20:03.52 | [TK]D-Fender | bmoraca: lets hope its not the case |
| 20:04.49 | SomethingIsodd | HEllo everyone.. does anyone here know anything about Phonexpress 4.30 i am trying to figure out what i need to do to convert it over to asterisk |
| 20:08.13 | jameswf | Fonality givet Asterisk 1.6 + Dahdi their ready for primetime approval: http://trixbox.org/devblog/trixbox-28-beta |
| 20:08.54 | Qwell | jjshoe: |
| 20:09.07 | Qwell | Did they use my SRPMs? |
| 20:09.08 | [TK]D-Fender | SomethingIsodd: and what is this "Phonexpress"? |
| 20:09.12 | Qwell | jameswf: ^^ |
| 20:10.01 | jameswf | i havent pulled it post went up <1hr ago |
| 20:10.07 | SomethingIsodd | [TK]D-Fender its the pbx that was installed in the office i just took over they have that and adc Higain converters for the phones |
| 20:10.13 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 20:10.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:10.16 | SomethingIsodd | i want to take this to the next level and have it voip |
| 20:10.59 | ncom | [TK]D-Fender: I put the jitterbuffer in iax_custom.conf like this: http://www.mibbit.com/pb/tW2tc2 . the command "iax2 show netstats" says there is 340 ms of jitter. what is normal? |
| 20:11.01 | [TK]D-Fender | SomethingIsodd: So far I see nothing taht is compatible in this setup |
| 20:11.04 | jameswf | no repo as far as i can see |
| 20:11.32 | kb3ien | any setup guides out there for dlink 1402S ? |
| 20:12.07 | SomethingIsodd | [TK]D-Fender i know nothing about Phonexpress, but it looks like its just running off a computer i guess what i am asking is if anyone knows if i can format the server and put asterisk on it and have it just feed the higain adc converts. |
| 20:12.15 | SomethingIsodd | and have it convert digital to analog |
| 20:12.26 | jjshoe | Qwell no idea |
| 20:12.35 | Qwell | jameswf: there's no download link? |
| 20:12.45 | [TK]D-Fender | SomethingIsodd: That hardware is not supported by * |
| 20:12.47 | Qwell | jjshoe: figured out what samy has been working on :p |
| 20:13.12 | SomethingIsodd | oh too bad ok so i can the only way is to get a channel bank |
| 20:13.12 | jjshoe | Qwell yeah apparently |
| 20:13.42 | jameswf | http://master.dl.sourceforge.net/sourceforge/asteriskathome/trixbox-2.8.0-BETA.iso |
| 20:14.54 | jjshoe | Qwell good business move, let the trixtards find the bugs, then moved it over to pbxtra |
| 20:14.59 | jjshoe | move* |
| 20:15.21 | [TK]D-Fender | SomethingIsodd: Hardly worth it. Channelbank and card will typically cost as much as decent SIP phones, only less functional |
| 20:15.59 | jjshoe | Qwell arguably also what tbpro was good for, except they never bothered unless they had paid for support, which was as close to 0 without being 0. |
| 20:16.21 | SomethingIsodd | [TK]D-Fender only issue is i have 32 offices |
| 20:16.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:16.39 | SomethingIsodd | do you still think it would be better to go sip phones instead |
| 20:16.41 | leifmadsen | hmmmmmmmmm.... am I just forgetting a way of doing this from the dialplan, or is using the AMI the only way of doing this; I want to call multiple people and put them into a meetme conference (all of them, not just the first to answer) |
| 20:17.05 | leifmadsen | basically, I want to call a Local channel that calls many SIP channels, and puts them all into a conference room. |
| 20:17.13 | [TK]D-Fender | SomethingIsodd: Cost per port <- thats the issue |
| 20:17.41 | SomethingIsodd | is there 32 channel... channel banks? |
| 20:17.53 | Qwell | SomethingIsodd: E1 channel bank |
| 20:17.57 | Qwell | well, 31 |
| 20:18.09 | SomethingIsodd | anyidea what hey run at for price? |
| 20:18.15 | [TK]D-Fender | leifmadsen: System(asterisk -rx "originate .......") for each |
| 20:18.27 | leifmadsen | dirty... :) |
| 20:18.35 | [TK]D-Fender | SomethingIsodd: very pricey for E1. Even at T1 SIP phones are cheaper |
| 20:18.45 | [TK]D-Fender | SomethingIsodd: You old system is effectively trash |
| 20:19.03 | [TK]D-Fender | SomethingIsodd: Only value left to squeeze out of it is selling the whole mess |
| 20:19.12 | [TK]D-Fender | leifmadsen: Just the way you like it! |
| 20:19.33 | leifmadsen | lol |
| 20:20.16 | leifmadsen | [TK]D-Fender: oh! there is an Originate() application in 1.6 |
| 20:20.18 | leifmadsen | perfect! |
| 20:21.23 | [TK]D-Fender | interesting... |
| 20:22.01 | [TK]D-Fender | leifmadsen: Not in 1.6.0 AFAICT |
| 20:22.41 | SomethingIsodd | hanks [TK]D-Fender |
| 20:24.25 | leifmadsen | [TK]D-Fender: it exists in the calendaring branch |
| 20:24.33 | leifmadsen | which is what I'm testing |
| 20:24.52 | [TK]D-Fender | leifmadsen: ok/fine/sure .... you elitist bastard :p |
| 20:25.06 | Cody | I'm having a problem with inbound calls on a new systems I'm setting up. Whenever I dial the number I get a message 'The number you have dialed is not in service. Please check the number and try again.' I know it's originating from my trixbox, but I'm not sure why. I've posted the log at http://pastebin.com/d4aeecad3 my trunk settings (censored) at http://codybaker.org/postdir/broadvox_sip_setting.pdf and inbound route at http://codybaker.org |
| 20:25.11 | [TK]D-Fender | *sigh* |
| 20:25.59 | [TK]D-Fender | Cody: The call is coming in, you have configured your routes wrong |
| 20:26.13 | [TK]D-Fender | Cody: And FreePBX is not supported here. Go ask in their channel. |
| 20:26.17 | [TK]D-Fender | #freepbx |
| 20:26.34 | Cody | ok, thank you |
| 20:28.15 | [TK]D-Fender | checkout time, BBIAB |
| 20:37.50 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 20:43.34 | jameswf | cox hates me I cant seem to pull over 200K/s from source forge |
| 20:43.41 | *** part/#asterisk [psy] (n=psy0rz@lounge.datux.nl) |
| 20:45.22 | *** part/#asterisk jstoker (n=jstoker@88-107-140-165.dynamic.dsl.as9105.com) |
| 20:46.03 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:46.53 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 20:48.55 | *** join/#asterisk monstertruck (n=monstret@201.255.213.233) |
| 20:49.07 | monstertruck | hello children |
| 20:49.25 | monstertruck | I just finished seting up an spa3102, its wrking fine, except that it dials an extra zer at the end |
| 20:49.32 | monstertruck | zero* |
| 20:49.49 | monstertruck | cant find anything about that issue, has anyone seen it before? |
| 20:49.52 | _pepo_ | I am going to use Asterisk like voicemail system for x>120000 users... What do you think is the best distro that I can use? |
| 20:51.47 | lidocaine | windows |
| 20:52.48 | Qwell | _pepo_: whatever you're familiar with. |
| 20:53.05 | _pepo_ | :) |
| 20:53.42 | _pepo_ | I use Debian and CentOS...Do you have any preference based in somthing? |
| 20:54.25 | russellb | if you use CentOS, you can run Asterisk off of official binary packages from Digium :-) |
| 20:54.56 | jameswf | I dont have it yet but one of my guys says packager: Jason Parker |
| 20:55.35 | *** join/#asterisk jblack (n=jblack@pool-71-173-9-11.sctnpa.east.verizon.net) |
| 20:56.19 | russellb | I've heard that guy is a jerk |
| 20:56.21 | russellb | runs from Qwell |
| 20:56.23 | monstertruck | anyone? |
| 20:56.40 | Qwell | jameswf: orly? |
| 20:56.53 | jameswf | Qwell, http://pastebin.com/m4f06677f |
| 20:57.24 | Qwell | that's epic |
| 21:01.41 | stabler | does anyone have an asterisk box behind a pfsense firewall? |
| 21:03.07 | UQlev | stabler: pf-firewall is the worst choice for sip |
| 21:03.34 | stabler | =/ |
| 21:04.31 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 21:04.43 | UQlev | stabler: are these 2 on the same box? |
| 21:04.52 | stabler | no |
| 21:05.00 | stabler | im running a pfsense box |
| 21:05.04 | stabler | as my gateway |
| 21:05.27 | stabler | pfsense is an OS |
| 21:05.49 | stabler | used to firewall/route/load balance |
| 21:06.32 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 21:07.05 | UQlev | stabler: and you want to use sip across the pfsense? |
| 21:07.16 | stabler | yes |
| 21:07.50 | stabler | im having some issues getting my asterisk box to register my voip provider though it |
| 21:08.21 | UQlev | is your voip provider sip or iax? |
| 21:08.46 | jplank | _pepo_: was this a serious questions? _pepo_: I am going to use Asterisk like voicemail system for x>120000 users... What do you think is the best distro that I can use? |
| 21:09.36 | stabler | UQlev: sip |
| 21:09.49 | _pepo_ | What do you think is the best distro that I can use? |
| 21:10.25 | jplank | 120000 users and your worried about the distro?? |
| 21:11.20 | UQlev | stabler: if it is the only sip device behind your firewall it is all right if you portforward udp 5060,10000-20000 to * box |
| 21:11.45 | stabler | ok |
| 21:11.47 | stabler | sounds good |
| 21:11.50 | stabler | ill give it a whirl |
| 21:12.03 | UQlev | jplank: 120K users is quite a lot even for ordinary mail-server |
| 21:12.04 | *** join/#asterisk monstertruck (n=monstret@201.255.213.233) |
| 21:13.18 | jplank | 120k users is insane. If you have 120k users you want to host their voicemail for, you should be more worried about the platform, redundency, and how your going to cluster the boxes, the actual distro should be the easy part |
| 21:13.58 | jplank | I could see someone trying to configure 120k users on one box. The voicemail.conf would make my head spin |
| 21:13.59 | UQlev | jplank: freebsd is the best distro for any internet server |
| 21:14.31 | jplank | I'm not getting sucked into a freebsd/linux argument |
| 21:14.54 | jplank | OS/2 FTW! |
| 21:16.58 | KavanS | lol |
| 21:17.01 | KavanS | DOS FTW! |
| 21:17.04 | timeshell_atwork | OS What? |
| 21:17.08 | jjshoe | with dahdi, when doing a dial, is it no longer Zap? Dahdi/group/num ? |
| 21:17.21 | timeshell_atwork | slaps jplank around with a large trout |
| 21:17.47 | Qwell | jjshoe: s/zap/dahdi/ |
| 21:18.04 | Qwell | jjshoe: see above btw |
| 21:18.31 | jjshoe | Qwell how far above? |
| 21:18.53 | Qwell | http://pastebin.com/m4f06677f |
| 21:18.55 | Qwell | ^^ |
| 21:19.07 | Qwell | jameswf confirmed :p |
| 21:20.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:21.19 | jplank | is asterisk business edition still based off of 1.2? |
| 21:21.37 | Qwell | jplank: C.x is 1.4-based |
| 21:21.39 | *** join/#asterisk path_ (n=path_@167-108-21-190.adsl.terra.cl) |
| 21:21.42 | Qwell | B.x was 1.2-based |
| 21:21.45 | jjshoe | Qwell hahahahhahaha |
| 21:21.55 | jjshoe | Qwell have you started a war in the forum yet? |
| 21:22.04 | Qwell | I have not. |
| 21:22.06 | Qwell | I expected it. |
| 21:22.14 | jjshoe | awe shame |
| 21:22.41 | jjshoe | if only you liked to troll as much as me ;) |
| 21:23.03 | jplank | qwell, they are making us purchase it to become digium certified (either that or switchvox) so we are getting the 10 license version. Anything cool with it, or should I file it with all the other BS stuff we are required to buy to become "official resellers" |
| 21:23.08 | Corydon76-dig | jjshoe: join linux-elitists and troll Rick Moen, then |
| 21:23.41 | Qwell | or troll Theo |
| 21:23.48 | Qwell | he deserves a good trolling now and again |
| 21:24.01 | Corydon76-dig | Theo isn't as brash as Moen |
| 21:24.04 | Qwell | jplank: I've not heard of that requirement... I have no idea. |
| 21:24.09 | jplank | 1ABEC10SIL = 1.4? |
| 21:24.12 | Qwell | Corydon76-dig: but you can't argue that he doesn't deserve it :p |
| 21:24.29 | Corydon76-dig | Qwell: Theo has calmed down in his old age |
| 21:24.42 | Qwell | Corydon76-dig: yet you still can't argue that he doesn't deserve it :p |
| 21:25.20 | Corydon76-dig | Qwell: I dunno, Moen actively goes out and looks for fights |
| 21:25.36 | Qwell | who's Moen? |
| 21:25.42 | Corydon76-dig | Theo may be a braggart about OpenBSD, but he keeps to his own lists |
| 21:25.55 | Corydon76-dig | Qwell: THAT is a very good question |
| 21:26.38 | Corydon76-dig | Qwell: Apparently, he fancies himself a Linux elitist, but other than some rough documentation heavily abusing the reader, I haven't found evidence that he's really done anything of note |
| 21:27.46 | Corydon76-dig | Nothing that bears out his contempt for others, anyway |
| 21:30.07 | jjshoe | Corydon76-dig I'm in linux-elitists over on efnet ;) |
| 21:30.17 | Qwell | jjshoe: that's #linuxhelp |
| 21:30.41 | jjshoe | Qwell exactly :) |
| 21:31.32 | jjshoe | used to be such a busy channel... |
| 21:32.20 | Corydon76-dig | jjshoe: http://zgp.org/mailman/listinfo/linux-elitists |
| 21:32.35 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:32.46 | jjshoe | changes to dahdi-channels... can they be picked up without dropping active calls using dahdi? |
| 21:33.13 | Corydon76-dig | Depends on the type of change |
| 21:33.28 | jjshoe | changing a context |
| 21:33.35 | Corydon76-dig | Yes, that works |
| 21:33.41 | jjshoe | just reload? or? |
| 21:33.52 | Corydon76-dig | Most changes work, as long as they don't have anything to do with signalling |
| 21:34.01 | Corydon76-dig | Yep, reload |
| 21:34.05 | mvanbaak | Theo ftw ! |
| 21:35.05 | jjshoe | Corydon76-dig thanks |
| 21:35.29 | mvanbaak | and Theo has a problem now |
| 21:35.48 | mvanbaak | the .eu reseller is years behind with paying for the cd-sets and t-shirts |
| 21:38.22 | mvanbaak | http://archives.neohapsis.com/archives/openbsd/cvs/2009-03/0568.html |
| 21:40.29 | mvanbaak | zzzz time |
| 21:41.25 | *** join/#asterisk path_ (n=path_@167-108-21-190.adsl.terra.cl) |
| 21:54.21 | hardwire | yo |
| 21:55.01 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-4f5e332ae5493df1) |
| 21:56.25 | *** join/#asterisk stabler (n=seedbox@cpe-24-210-38-135.columbus.res.rr.com) |
| 21:57.03 | rbd | hey guys, sanity check here...if i have format_mp3 installed, asterisk should support mp3 playback through something like Playback(/tmp/mysong.mp3) for instance ...and AGI as well, right? |
| 21:59.29 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 22:00.59 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 22:05.12 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 22:05.21 | f0ner00t | Hello. |
| 22:05.32 | f0ner00t | Can someone give me the link again how to configure asterisk w/ IPKALL. |
| 22:06.34 | *** join/#asterisk Ether_Man (i=Ether_Ma@h229n2fls310o1101.telia.com) |
| 22:06.54 | *** join/#asterisk m0s3s (i=44bd7212@gateway/web/ajax/mibbit.com/x-8b51f79410d7f786) |
| 22:08.32 | f0ner00t | Anybody? |
| 22:16.18 | Khratos | rbd, partially right, I think you should not use the '.mp3' part |
| 22:16.27 | bkw_ | I can agree with that |
| 22:16.28 | bkw_ | it won't scale |
| 22:20.52 | beek | Anyone on the channel have any horror stories around running Asterisk on an HP ProLiant ML150 G5? |
| 22:25.27 | *** join/#asterisk qdk (n=qdk@213.173.230.10) |
| 22:29.02 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:29.02 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:30.00 | f0ner00t | Should I be able to telnet 127.0.0.1 5060 if asterisk is running? |
| 22:30.29 | Qwell | f0ner00t: no |
| 22:30.41 | russellb_ | Not unless you are running asterisk 1.6, and SIP TCP is enabled |
| 22:32.05 | f0ner00t | What do I have to do to get ipkall working. I already configured sip.conf and extensions.conf? |
| 22:34.47 | *** part/#asterisk harryv (n=harry@67.207.147.205) |
| 22:39.50 | f0ner00t | Anybody? |
| 22:41.27 | naxxfish | do you need a register => line for that? |
| 22:42.32 | *** join/#asterisk hi365 (n=hi365@85.130.230.240) |
| 22:42.37 | f0ner00t | Umm how would I register the line? |
| 22:42.49 | *** join/#asterisk hadronzoo (n=user@ppp-70-250-188-181.dsl.rcsntx.swbell.net) |
| 22:43.25 | Corydon76-dig | f0ner00t: clearly, not. Go look in sip.conf and pay special attention to lines that contain "register =>" |
| 22:43.50 | f0ner00t | Corydon76-dig: May I pm you for a second? |
| 22:43.53 | Corydon76-dig | and then read the surrounding text |
| 22:43.54 | f0ner00t | Or acutally. |
| 22:43.54 | Corydon76-dig | NO |
| 22:43.58 | f0ner00t | I'll pastebin it. |
| 22:44.53 | f0ner00t | Corydon76-dig: http://pastebin.com/m6da8da23 |
| 22:44.59 | f0ner00t | Can you look at that link please?] |
| 22:45.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:45.32 | Corydon76-dig | Okay, look in iax.conf, paying special attention to lines that contain "register =>" |
| 22:46.01 | f0ner00t | Ok.. Thank you |
| 22:46.20 | f0ner00t | What else needs to be contained in iax.conf besides register = > |
| 22:47.02 | Corydon76-dig | A context for the trunk |
| 22:48.06 | f0ner00t | Do you think it would be better just to edit the samples or write my own .conf? |
| 22:48.31 | Corydon76-dig | You'd probably be better off editing the samples |
| 22:48.59 | Corydon76-dig | Or more accurately, reading the samples and basing your own config off that |
| 22:49.19 | f0ner00t | Yea Thats what I'm doing. I'm intergrating pieces of the sample into the main. |
| 22:52.37 | f0ner00t | Corydon76-dig: I guess I'm just not confortable with the conf files yet. |
| 22:52.44 | _pepo_ | Please, How do I disable option 8 from voicemail? |
| 22:54.14 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 22:56.09 | Corydon76-dig | _pepo_: first, learn C. I recommend Practical C Programming from O'Reilly. |
| 22:57.06 | _pepo_ | I know C ... so can be disabled just by code? |
| 22:58.52 | Corydon76-dig | Correct |
| 23:01.08 | Khratos | brb |
| 23:03.18 | _pepo_ | is voicemail.c the file to be modified? |
| 23:05.28 | Qwell | _pepo_: apps/app_voicemail.c |
| 23:06.26 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 23:07.15 | Corydon76-dig | Qwell: we taking bets on whether he goes stark raving mad before he accomplished what he set out to do? |
| 23:12.55 | _pepo_ | I saw ... I think that is the function vm_execmain |
| 23:20.20 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 23:23.25 | _pepo_ | a lot of thanks |
| 23:32.16 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 23:32.33 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:32.54 | *** join/#asterisk Steve_J-obs (n=Chris123@pool-68-239-44-216.bos.east.verizon.net) |
| 23:35.00 | *** join/#asterisk Magicblaze007 (n=sony@fl-67-233-204-182.dhcp.embarqhsd.net) |
| 23:36.27 | Steve_J-obs | hello everybody!!!! |
| 23:36.58 | k-man | hi Steve_J-obs |
| 23:37.14 | Steve_J-obs | Anyone who says hi to me gets 90% Apple stocks |
| 23:37.21 | Steve_J-obs | for the next 1 minute |
| 23:37.30 | Steve_J-obs | 90% off |
| 23:37.33 | *** join/#asterisk t0rrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:37.47 | k-man | Steve_J-obs: great! where do I sign up? |
| 23:37.57 | Steve_J-obs | no sign up |
| 23:38.20 | Steve_J-obs | ok, time is up! |
| 23:38.22 | Steve_J-obs | sorry |
| 23:40.04 | *** join/#asterisk shindig (n=matt@138.172.188.72.cfl.res.rr.com) |
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