| 00:38.52 | *** join/#asterisk aruntomar (~Thunderbi@49.248.152.20) |
| 01:03.20 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 01:03.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 01:06.28 | *** join/#asterisk Alex_Bkash (b4eacf31@gateway/web/freenode/ip.180.234.207.49) |
| 01:20.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.36) |
| 01:24.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:34.24 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 01:35.10 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 01:41.52 | *** join/#asterisk g_r_eek (~g_r_eek@78-27-146.adsl.cyta.gr) |
| 02:08.50 | *** join/#asterisk kuruption (kuruption@vato.is.a.big.black.cock.addikt.org) |
| 02:09.33 | *** join/#asterisk zendel (~zendel.fe@chatswood.au.ulterius.net) |
| 02:17.43 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 02:54.37 | *** join/#asterisk dgeary2 (~debian@120.21.78.52) |
| 02:59.31 | *** join/#asterisk dgeary2 (~debian@120.21.78.52) |
| 03:00.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.198) |
| 03:12.20 | *** join/#asterisk dgeary2 (~debian@120.21.78.52) |
| 03:23.24 | *** join/#asterisk Alex_Bkash (cbdf5c4a@gateway/web/freenode/ip.203.223.92.74) |
| 03:24.48 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.198) |
| 03:41.51 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-sgwvexqqpxticvue) |
| 04:21.09 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 04:22.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.198) |
| 04:46.16 | *** join/#asterisk phyu (~kvirc@tok69-5-82-235-151-229.fbx.proxad.net) |
| 05:03.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:13.06 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 05:36.36 | j4jackj | Hello everyone |
| 05:39.47 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 05:44.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.9) |
| 05:46.19 | phix | j4jackj: hai hai |
| 05:47.31 | j4jackj | Anyone want to call the worst conference room ever? |
| 05:51.24 | phix | Not particulary |
| 05:51.51 | ChannelZ | it's called "tech support" |
| 05:52.08 | j4jackj | ChannelZ: it isn't |
| 05:52.16 | j4jackj | It's actually not the worst |
| 05:52.34 | j4jackj | I'm talking about room 1, 71@99199.11.127 |
| 05:52.55 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 05:59.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 05:59.40 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 06:12.05 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 06:25.18 | *** join/#asterisk jsjc (~Adium@221.Red-83-41-76.dynamicIP.rima-tde.net) |
| 06:31.43 | ChannelZ | Interesting IP. |
| 06:53.18 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 06:57.16 | j4jackj | ChannelZ: I missed a dot |
| 06:57.20 | j4jackj | 99.199.11.127 |
| 06:57.57 | ChannelZ | yah I figgered that out |
| 06:58.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.105) |
| 06:58.55 | j4jackj | 71@99.199.11.127 |
| 07:12.36 | ChannelZ | bad MOH |
| 07:14.49 | j4jackj | not really |
| 07:15.06 | j4jackj | ctually the best I've heard, and hey, it's asterisk default |
| 07:15.31 | j4jackj | Can you call it again, I'm in the room now |
| 07:17.48 | j4jackj | ChannelZ: can you call again? |
| 07:19.30 | ChannelZ | well that was exciting |
| 07:19.44 | j4jackj | Sorry, Linphtone assploded |
| 07:19.52 | j4jackj | Call again, I can talk this time |
| 07:20.03 | ChannelZ | well it also just disconnected me |
| 07:20.09 | j4jackj | Oh. |
| 07:20.18 | j4jackj | Do you want an account to register to? |
| 07:20.34 | j4jackj | Or is t fine? |
| 07:20.56 | j4jackj | ChannelZ: it should waork this time... |
| 07:21.01 | ChannelZ | all I can do is play you sounds at the moment, my audio setup is... well not. No mic |
| 07:21.39 | j4jackj | Well you can still play sounds and I'll say if it worked. |
| 07:22.03 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:22.29 | j4jackj | Also, buy a mic or use the one in your webcam. |
| 07:22.38 | j4jackj | Yea, it's b0rked |
| 07:22.44 | ChannelZ | I have a quite expensive mic. I just have my audio interface torn apart |
| 07:23.03 | j4jackj | Can you hear me when I say stuff though? |
| 07:23.10 | ChannelZ | I heard just some room tone/static from you.. after a few seconds it started playing MOH, did that for a few seconds, then ends the call. |
| 07:24.06 | j4jackj | Yea, try again. i'm debugging my server setup |
| 07:24.12 | ChannelZ | have you done local echo tests with your softphone? |
| 07:24.19 | j4jackj | I think I have |
| 07:24.23 | j4jackj | It does work |
| 07:25.28 | j4jackj | Yup, echo test works |
| 07:25.48 | j4jackj | Now can you call into the conference? |
| 07:26.39 | ChannelZ | back to music,. |
| 07:26.46 | ChannelZ | aaaand it hung up on me. |
| 07:29.40 | j4jackj | Why does it hang up on you? |
| 07:29.45 | j4jackj | This may clue you in |
| 07:29.55 | j4jackj | [Aug 26 00:25:56] WARNING[10026]: chan_sip.c:3685 retrans_pkt: Hanging up call 3c19336547cad79b7c9aeaab7f6a8fc1@173.160.35.173:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 07:30.04 | j4jackj | My unreliable Internet |
| 07:30.04 | ChannelZ | dunno. I wasn't looking at the SIP dialog. |
| 07:30.15 | j4jackj | My error message is shown |
| 07:30.46 | ChannelZ | Is your asterisk behind a firewall? |
| 07:33.25 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 07:34.20 | j4jackj | hi dash_ |
| 07:34.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:35.16 | j4jackj | Hi ChannelZ - |
| 07:35.34 | j4jackj | ChannelZ: You have (1) unread memo from me. |
| 07:36.05 | j4jackj | Thank you. |
| 07:36.07 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 07:38.15 | ChannelZ | not sure what your issue is.. SIP dialogs obviously working since I can setup the call in the first place |
| 07:38.41 | j4jackj | Yes, and then it times out |
| 07:38.56 | j4jackj | Sorry bout the on hold |
| 07:39.24 | ChannelZ | Oh you have config issues. |
| 07:39.45 | j4jackj | Howso? |
| 07:39.55 | j4jackj | did it asplode? |
| 07:40.39 | ChannelZ | your box is sending out its LAN IP for starters |
| 07:41.28 | ChannelZ | you didn't config localnet and externaddr correctly (or externip, I don't remember when it changed) |
| 07:42.04 | j4jackj | Oh. |
| 07:42.14 | j4jackj | localnet does not include the outernet |
| 07:42.30 | ChannelZ | eh? |
| 07:43.45 | ChannelZ | localnet=192.168.0.0/16 externaddr=99.199.11.127 |
| 07:46.32 | j4jackj | How did you get in there? |
| 07:46.39 | j4jackj | I have those |
| 07:46.48 | j4jackj | And many more, on channel four.... |
| 07:46.53 | j4jackj | :D |
| 07:47.13 | j4jackj | I dded nat=yes <-- will that help? |
| 07:47.36 | ChannelZ | not really, my IPs are fine. |
| 07:47.48 | j4jackj | Yes they are. |
| 07:48.02 | j4jackj | However, I'm behind a NAT |
| 07:48.08 | ChannelZ | But your box is convinced it's 192.168.1.101 so after the initial INVITE that's where mine starts sending replies and RTP |
| 07:48.46 | ChannelZ | what does 'sip show settings' show under Network Settings |
| 07:49.55 | j4jackj | http://sprunge.us/IWSP |
| 07:50.48 | j4jackj | Is that read yet? |
| 07:51.16 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 07:52.09 | j4jackj | ChannelZ: is everything correct or wrong? |
| 07:52.21 | ChannelZ | well it seems right |
| 07:52.53 | ChannelZ | Dunno if you have something else in sip.conf breaking it or if it's a quirk or something else with 1.8, I don't know. |
| 07:53.01 | j4jackj | also, you can try call me by IPv6, 71@[2001:470:b308:cafe:20c:f1ff:fea0:239a] |
| 07:53.09 | j4jackj | I know, the infamous 470 |
| 07:53.28 | j4jackj | If you have IPv6 that is |
| 07:53.29 | ChannelZ | no |
| 07:53.38 | j4jackj | mightbe a 1.8 quirk |
| 07:54.17 | j4jackj | did you hear sth? |
| 07:54.25 | ChannelZ | no just a wierd humming |
| 07:54.49 | j4jackj | Oh. |
| 07:54.55 | j4jackj | Can you help me then? |
| 07:55.18 | ChannelZ | http://pastebin.com/tnPhu2Dc |
| 07:55.20 | j4jackj | Because I am in the DMZ behind a NAT and I've set everything as it is meant to be set. |
| 07:55.36 | ChannelZ | see line 45 |
| 07:55.47 | ChannelZ | your box isn't putting its external IP in. |
| 07:56.01 | ChannelZ | Why I am uncertain. You can pb your entire sip.conf I guess |
| 07:56.11 | ChannelZ | just XX out secrets |
| 07:56.41 | ChannelZ | but I can't mess with this much longer, I need to go to bed. work tomorrow (well today) |
| 07:56.41 | j4jackj | I saw tht |
| 07:58.18 | j4jackj | I've pulled all the secrets out using sed. http://sprunge.us/KVSC |
| 08:00.15 | ChannelZ | ugh you should pull out all the commented examples too |
| 08:00.16 | j4jackj | So what I ask is the problem? |
| 08:00.28 | j4jackj | Yes I should |
| 08:00.34 | ChannelZ | dunno haven't finished scrolling through this mess yet |
| 08:00.43 | j4jackj | But they help me see where the **** I am going. Without them i am blind. |
| 08:00.48 | j4jackj | In the sip.conf |
| 08:00.53 | ChannelZ | initial thought is your udpbindaddr is ipv6, dunno if that causes chaos |
| 08:01.04 | j4jackj | It's :: |
| 08:01.04 | j4jackj | Which can do IPv4 |
| 08:01.07 | j4jackj | In linux |
| 08:01.15 | ChannelZ | hint: copy all the default configs into a "dist" directory. Then you have them for reference. |
| 08:01.40 | ChannelZ | yes but possibly broken in asterisk, I have no idea. Only a random guess. |
| 08:01.42 | j4jackj | Example IPv6-mapped IPv4: ::ffff:63c7:b7f |
| 08:02.43 | j4jackj | Redone with the IPv4 only listener |
| 08:03.13 | *** part/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 08:05.30 | ChannelZ | now it's showing the right IP. |
| 08:05.46 | j4jackj | Oh ok |
| 08:06.36 | ChannelZ | hasn't dumped me. |
| 08:06.45 | ChannelZ | I want to kill myself with this music |
| 08:08.08 | ChannelZ | ok that's enough. I need to poop. |
| 08:08.19 | j4jackj | Can you stay on the line with it turned down |
| 08:09.36 | j4jackj | Did I just make him poo himself? |
| 08:09.50 | j4jackj | I will never get to sleep. Thank you ChannelZ |
| 08:14.15 | *** join/#asterisk CeBe (~CeBe@port-92-206-22-4.dynamic.qsc.de) |
| 08:16.17 | j4jackj | CeBe: I just made ChannelZ poo himself= |
| 08:18.16 | *** join/#asterisk vlad_starkov (~vlad_star@176.110.120.2) |
| 08:18.22 | j4jackj | hi vlad_starkov |
| 08:20.39 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 08:20.40 | j4jackj | ChannelZ: I promise I won't hang up before you call again. That was probably the worst mental image I got in my life. |
| 08:22.31 | *** join/#asterisk vlad_sta_ (~vlad_star@176.110.120.2) |
| 08:23.15 | j4jackj | Everyone here is aware I just had the worst mental image of my life? :( |
| 08:23.28 | j4jackj | My hold music made someone poop themselves. I bet I will get a surprise in the mail, fecal vendetta style. |
| 08:29.45 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:29.52 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 08:31.01 | *** join/#asterisk l0c0 (~root@host-212-68-194-46.brutele.be) |
| 08:33.37 | j4jackj | Anyone want to call me? |
| 08:34.08 | j4jackj | 8j4jackj@99.199.11.127 or 71@99.199.11.127 (for conference) |
| 08:37.23 | CeBe | j4jackj: what are you talking about? |
| 08:37.37 | j4jackj | CeBe: my asterisk exchange |
| 08:37.57 | *** join/#asterisk vlad_starkov (~vlad_star@176.110.120.2) |
| 08:38.26 | j4jackj | Hi vlad_starkov |
| 08:41.18 | *** join/#asterisk vlad_starkov (~vlad_star@176.110.120.2) |
| 08:41.39 | j4jackj | Hi vlad_starkov |
| 08:46.26 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:46.41 | j4jackj | Hi c0rnoTa |
| 08:49.03 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:02.28 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-fkklgbklqaakjpmc) |
| 09:06.49 | cusco | hi j4jackj |
| 09:08.43 | j4jackj | hi cusco |
| 09:08.50 | j4jackj | wanna call? |
| 09:09.21 | cusco | no |
| 09:09.25 | j4jackj | why not? |
| 09:09.32 | j4jackj | I'm debugging my asterisk |
| 09:09.39 | j4jackj | and it's someone to talk to |
| 09:09.41 | j4jackj | if it works |
| 09:10.00 | cusco | I'm going rush off to work |
| 09:10.02 | cusco | brb |
| 09:10.11 | j4jackj | :D |
| 09:11.39 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-220-156.pools.arcor-ip.net) |
| 09:11.41 | j4jackj | I'm very lonely and want to debug my Asterisk. My mum is threatening to disown me. |
| 09:12.58 | j4jackj | ChannelZ: ? |
| 09:16.15 | apb1963_ | I'd help for a few minutes if I knew why I couldn't call sip addresses :) |
| 09:16.23 | apb1963_ | s/couldn't/can't/ |
| 09:16.29 | j4jackj | apb1963_: I could help |
| 09:16.41 | apb1963_ | that would be appreciated |
| 09:16.45 | j4jackj | Check your client. |
| 09:17.10 | j4jackj | What is it asking for? |
| 09:17.31 | j4jackj | apb1963_: is the problem in Asterisk or in your client? |
| 09:17.50 | apb1963_ | I'm assuming asterisk, but I have no idea |
| 09:18.15 | j4jackj | can you pastebin your logs when you try to call the ceho test? |
| 09:18.57 | j4jackj | or better yet when asterisk tries to call 71@99.199.11.127? |
| 09:19.17 | apb1963_ | when I try to call 71@99.199.11.127 I get the following message: |
| 09:19.28 | j4jackj | Yes? |
| 09:19.36 | apb1963_ | That's what I get. |
| 09:19.47 | j4jackj | Hmm |
| 09:19.52 | j4jackj | 02:18 < apb1963_> when I try to call 71@99.199.11.127 I get the following message: |
| 09:19.55 | apb1963_ | That's what I said |
| 09:20.01 | apb1963_ | Yes |
| 09:20.02 | j4jackj | Hmm |
| 09:20.07 | j4jackj | Interesting |
| 09:20.07 | apb1963_ | Tjat |
| 09:20.10 | apb1963_ | That's what I said |
| 09:20.17 | apb1963_ | Very |
| 09:20.25 | j4jackj | It never logged. |
| 09:20.28 | apb1963_ | More frustrating than interesting actually |
| 09:20.30 | apb1963_ | Correct |
| 09:20.33 | j4jackj | Have you tried restarting asterisk? |
| 09:20.35 | apb1963_ | No message |
| 09:20.40 | apb1963_ | Yes |
| 09:21.00 | j4jackj | Have you tried a factory reset (reinstall from source or DEB package)? |
| 09:21.03 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 09:21.17 | apb1963_ | Yes |
| 09:21.32 | apb1963_ | However... now that I look at my client logs... it says Forbidden |
| 09:21.37 | j4jackj | Are you sure you're single or non-NATted |
| 09:21.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.242) |
| 09:21.46 | apb1963_ | 403: Forbidden |
| 09:21.54 | j4jackj | I don't log a 403 error |
| 09:22.04 | apb1963_ | not in asterisk.. in the client |
| 09:22.09 | j4jackj | I know |
| 09:22.17 | j4jackj | Have you tried a different phone? |
| 09:22.20 | *** join/#asterisk suneye (~atcmmi@119.123.220.188) |
| 09:23.12 | j4jackj | Ex. If you use Ekiga try Linphone and if you use Linphone try Jitsi |
| 09:23.47 | apb1963_ | zoiper says: bearer capability not authorized |
| 09:24.10 | j4jackj | Try a diferent client |
| 09:24.22 | apb1963_ | That's the second client |
| 09:24.43 | j4jackj | Tried Ekiga? |
| 09:24.55 | apb1963_ | what I think it's trying to say, is that asterisk is not configured properly to allow it |
| 09:25.00 | *** join/#asterisk suneye (~atcmmi@119.123.220.188) |
| 09:25.02 | j4jackj | Not true |
| 09:25.06 | apb1963_ | ok |
| 09:25.14 | apb1963_ | I'm all up for a better idea |
| 09:25.28 | j4jackj | It's that the sip receiver server in your client is not able to process the request |
| 09:25.37 | j4jackj | (Of course that would be a 503) |
| 09:25.46 | j4jackj | \But it's 403 because of a specific reason. |
| 09:26.00 | apb1963_ | sip....receiver.....server.... in the client..... |
| 09:26.03 | j4jackj | It's not authenticated to process the reque.st |
| 09:26.08 | j4jackj | apb1963_: yes there is |
| 09:26.59 | j4jackj | A sip receiver server (in my vocabulary) is a miniserver that does SIP call negotiation |
| 09:27.10 | j4jackj | It's a 'satellite' of the sip server. |
| 09:27.16 | j4jackj | It's not a node |
| 09:27.26 | j4jackj | It's an endpoint on a star topology |
| 09:27.42 | j4jackj | With the center being the proxy |
| 09:30.00 | j4jackj | apb1963_: ? |
| 09:30.23 | apb1963_ | Yes? |
| 09:30.38 | apb1963_ | ekiga won't register at all |
| 09:30.57 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:79c8:9f0c:6ed2:f687) |
| 09:32.00 | j4jackj | apb1963_: what about linphone? |
| 09:32.22 | apb1963_ | The last time I installed linphone & jitsi they both had issues |
| 09:32.25 | j4jackj | How about I give you an acct on my server for testing |
| 09:32.44 | j4jackj | You need a stun server with linphone |
| 09:32.47 | apb1963_ | linphone wouldn't work at all... jitsi... ok, it worked for the most part but the ring made me crazy |
| 09:33.06 | j4jackj | linphone is configurable. use it to your advatage |
| 09:33.20 | apb1963_ | I like phonerlite |
| 09:33.23 | apb1963_ | mostly |
| 09:34.05 | j4jackj | Meh |
| 09:34.08 | apb1963_ | ekiga complains about transport errors |
| 09:34.11 | j4jackj | I can only support Ekiga and Linphone |
| 09:34.20 | j4jackj | Ekiga is playing bluff |
| 09:34.28 | j4jackj | I think it needs some help |
| 09:34.35 | j4jackj | Or try linphone |
| 09:34.44 | apb1963_ | linphone never worked for me |
| 09:35.01 | apb1963_ | it pops up a window that's blank. And that's all she wrote. |
| 09:35.24 | j4jackj | Hmm |
| 09:35.28 | tparcina | Is there a "grep like" command in * cli? |
| 09:35.34 | j4jackj | I don't think so |
| 09:35.42 | j4jackj | IANAE though. |
| 09:35.45 | apb1963_ | yes... it's called grep |
| 09:35.53 | tparcina | Like sip show peers | grep 777 |
| 09:36.00 | apb1963_ | yep |
| 09:36.01 | kaldemar | tparcina: use the shell one. as in "asterisk -vvvvr | grep something" |
| 09:36.14 | apb1963_ | asterisk -rx "sip show peers" | grep 777 |
| 09:36.28 | apb1963_ | or what kaldemar said ? |
| 09:36.46 | tparcina | kaldemar: apb1963_: I know about that one, just I would prefer something from * CLI, if it's possible. |
| 09:36.50 | kaldemar | or that. depends on what you want to grep. |
| 09:37.23 | kaldemar | tparcina: in asterisk you can show all or one. that's it. |
| 09:37.46 | j4jackj | I'm just helping you test it. |
| 09:37.47 | tparcina | kaldemar: Thank you. Aldo this would be nice option. :) |
| 09:38.16 | j4jackj | apb1963_: check PM |
| 09:39.52 | apb1963_ | registered |
| 09:40.05 | j4jackj | Yup |
| 09:40.38 | apb1963_ | so... what have we learned? |
| 09:40.43 | j4jackj | Now, if you think it's on a lark to do so, call 71 (the conf prefix 7 followed by conf no 1) |
| 09:40.54 | j4jackj | apb1963_: the port for your cli and serv were conflicting |
| 09:41.01 | j4jackj | they are the same computer I think- |
| 09:41.10 | apb1963_ | I can register to my own asterisk just fine |
| 09:41.34 | apb1963_ | just not with ekiga... which I'm not using now |
| 09:41.38 | j4jackj | apb1963_: also, you registered with an IP not in the public internet. now please use a stun serwver. |
| 09:41.50 | apb1963_ | I did? that's interesting |
| 09:41.55 | kaldemar | stun is not needed. |
| 09:42.01 | j4jackj | I can't accept people registering without a STUN server |
| 09:42.04 | apb1963_ | I don't see how I could have |
| 09:42.08 | j4jackj | Unless they are in the pub internet |
| 09:42.18 | j4jackj | apb1963_: 'you don't have it set in phonerlite |
| 09:42.25 | j4jackj | kaldemar: you lie |
| 09:42.32 | kaldemar | j4jackj: no, i don't. |
| 09:42.35 | apb1963_ | you're saying you see me come in with a 192.x.x.x address? |
| 09:42.37 | j4jackj | yes, you do. |
| 09:42.41 | j4jackj | apb1963_: yup |
| 09:42.51 | j4jackj | kaldemar: stun is needed with IPv4 and NAT |
| 09:42.54 | apb1963_ | how is that possible? |
| 09:42.59 | j4jackj | thus, kaldemar, you lie. |
| 09:43.05 | j4jackj | apb1963_: you're not usingSTUN. |
| 09:43.08 | kaldemar | j4jackj: you're full of bs. |
| 09:43.12 | j4jackj | kaldemar: pronk |
| 09:43.23 | apb1963_ | what is the address you see j4jackj? |
| 09:43.34 | j4jackj | apb1963_: 192.168.0.100 |
| 09:43.39 | j4jackj | that will fail instantly |
| 09:43.41 | apb1963_ | that's the one |
| 09:43.48 | j4jackj | it will send to a non-existent address. |
| 09:44.14 | apb1963_ | yeah... I don't see how it gets past my router... let alone my ISP's router. |
| 09:44.23 | j4jackj | apb1963_: can you set up stun in PhonerLite? |
| 09:44.28 | apb1963_ | I dunno |
| 09:44.37 | apb1963_ | I mean it has the option |
| 09:44.38 | j4jackj | Try using another client then. |
| 09:44.46 | apb1963_ | I just don't know what server to use |
| 09:44.51 | j4jackj | Can you tell it to use tun server stunserver.org ? |
| 09:44.59 | j4jackj | *stun server |
| 09:45.01 | kaldemar | apb1963_: it doesn't. |
| 09:45.15 | apb1963_ | kaldemar: it what doesn't what? |
| 09:45.32 | kaldemar | apb1963_: get past your router as it would look like coming from a private address. |
| 09:45.38 | j4jackj | apb1963_: plon him for your own good, he's full of it |
| 09:45.52 | kaldemar | apb1963_: the sip message might have private addresses in it, but that is not an issue is asterisk is configured properly. |
| 09:46.17 | apb1963_ | oic ... that makes sense |
| 09:46.28 | kaldemar | ~sipnat |
| 09:46.28 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 09:47.02 | apb1963_ | so I added the stun server |
| 09:47.07 | j4jackj | yes? |
| 09:47.09 | apb1963_ | just to make j4jackj happy |
| 09:47.18 | kaldemar | good luck. |
| 09:47.32 | apb1963_ | it's only in use for his address |
| 09:47.38 | j4jackj | BTW that's not for when your asterisk is in your network, remove it once you can get thru to 71 |
| 09:47.55 | apb1963_ | it's profile dependent |
| 09:47.57 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:48.02 | j4jackj | OK good |
| 09:48.03 | apb1963_ | so it's setup for your asterisk, but not for mine |
| 09:48.12 | j4jackj | <PROTECTED> |
| 09:48.17 | j4jackj | still same old |
| 09:48.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.242) |
| 09:48.44 | apb1963_ | how about now? |
| 09:49.15 | j4jackj | bork bork |
| 09:49.32 | apb1963_ | I don't speak dog |
| 09:49.42 | j4jackj | apb1963_: I meant it broke |
| 09:49.49 | apb1963_ | what broke? |
| 09:49.54 | j4jackj | As in didn't unregister |
| 09:49.58 | j4jackj | Or reregister |
| 09:50.00 | apb1963_ | looks registered here |
| 09:50.28 | j4jackj | It's not registred crrectly. |
| 09:50.35 | apb1963_ | call is proceeding |
| 09:50.49 | apb1963_ | and now as you can see... we're connected |
| 09:53.08 | *** join/#asterisk g_r_eek (~g_r_eek@78-42-169.adsl.cyta.gr) |
| 09:54.26 | j4jackj | Sorry, my mum is winding me up |
| 09:54.34 | *** join/#asterisk LooserOuting (~LooserOut@ip-176-198-132-85.unitymediagroup.de) |
| 09:55.16 | apb1963_ | lol |
| 09:55.17 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-jmqnsinvlivjstdf) |
| 09:55.23 | apb1963_ | I kinda figured there was something going on |
| 09:55.51 | apb1963_ | I just figured it was a kid you were bawling out. lol |
| 09:56.15 | j4jackj | btw you can likely hear my mu |
| 09:56.17 | j4jackj | m |
| 09:56.34 | apb1963_ | now I can |
| 09:57.12 | apb1963_ | I think I saw this on tv |
| 09:57.25 | apb1963_ | Monty Python or Bennie Hill |
| 09:57.31 | j4jackj | no |
| 09:58.36 | apb1963_ | I might be witness to a murder |
| 09:59.55 | apb1963_ | jeez |
| 10:00.44 | apb1963_ | wow. that was a knock down drag'em out screaming match |
| 10:00.58 | apb1963_ | I haven't heard something like that since I was 3 |
| 10:04.33 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 10:09.32 | *** join/#asterisk atcmmi (~atcmmi@121.34.41.107) |
| 10:12.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:30.05 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:42.40 | [TK]D-Fender | [05:42]j4jackjkaldemar: stun is needed with IPv4 and NAT <- incorrect |
| 10:44.15 | kaldemar | [TK]D-Fender: he's not around for cluebat anymore. |
| 10:44.28 | [TK]D-Fender | Yeah, missed the departure... |
| 10:44.45 | [TK]D-Fender | And his config has mistakes.... in addition to having all the sample crap in there |
| 10:44.55 | [TK]D-Fender | And his "scrub" .... was partial |
| 10:47.50 | kaldemar | but he has a sip receiver miniserver! |
| 10:55.11 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xhoqzukoxaysbktn) |
| 10:59.14 | *** join/#asterisk orn (~orn@2a01:8280:101:0:a905:7dec:177a:b66e) |
| 10:59.53 | orn | How is astcanary generally used? Do people just create a script that will run astcanary after asterisk has been started? Or is there some default way of running it after install asterisk? |
| 11:01.26 | orn | can't find much about it when googling |
| 11:01.39 | *** join/#asterisk g_r_eek (~g_r_eek@78-37-53.adsl.cyta.gr) |
| 11:01.43 | orn | (other than intended use) |
| 11:07.39 | *** join/#asterisk Joram (~joram@141.136.124.2) |
| 11:08.35 | *** join/#asterisk banane_ (528bc51a@gateway/web/freenode/ip.82.139.197.26) |
| 11:10.04 | banane_ | hi everyone, could someone please tell me if there´s another way to resume a dialplan after hangup? I´m currently using dial(g) but the context is only resumed if the called party hangs up, i would like it to also resume if the calling party hangs up |
| 11:10.05 | fredericve | orn, generally it is started with an init script |
| 11:10.51 | kaldemar | banane_: hangup extension |
| 11:10.53 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 11:11.30 | Joram | Hello, i have a question: my provider does not support the change of the caller id. So what i wanted to make was that asterisk first calls a list of required people. An user answers his phone and the asterisk server hangs up. Then the call is transfered to the user so he will see the number of the customer on his screen. The problem is that after i do a Hangup() i cannot transfer the call anymore. |
| 11:11.43 | orn | fredericve: Does * come with any pre-definied inits for that, as far as you know? I installed from source. |
| 11:12.00 | fredericve | what distro are you running on? |
| 11:12.55 | fredericve | example init scripts are available in the contrib/init.d directory in the source |
| 11:13.24 | banane_ | kaldemar: is the hangup extension generally called after every call or do i have to specify another parameter for this to be called? |
| 11:14.09 | kaldemar | banane_: it is executed in the current context for the call, if it exists. you don't need to specify any additional parameters for it. |
| 11:14.18 | *** join/#asterisk bmm (~bram@141.136.124.2) |
| 11:14.50 | *** join/#asterisk g_r_eek (~g_r_eek@78-37-53.adsl.cyta.gr) |
| 11:15.43 | fredericve | orn: There doesn't seem to be any code for starting the canary in those init scripts, but adding that is trivial |
| 11:16.11 | orn | fredericve: Sorry, was AFK. Am running on ubuntu server |
| 11:16.40 | banane_ | kaldemar: i only have to trigger this for calls on specific extensions, should i just create a variable to identify the specific calls or do you have a more elegant idea in mind? |
| 11:16.50 | orn | fredericve: What I'm wondering is if using astcanary is advisable, since very few people seem to be using it -- judging from the discussion (or lack thereof) online |
| 11:17.33 | banane_ | ok stupid question i could just switch the called party id... thanks for your help kaldemar |
| 11:18.17 | fredericve | orn: I asked the same question once. astcanary is useful if you also run other applications than asterisk that may not be as critical as asterisk |
| 11:19.34 | fredericve | If there's another process hogging resources, asterisk may suffer from it and this could result in bad quality calls, slow dialplan execution, etc. |
| 11:20.00 | fredericve | That said, I personally don't use astcanary |
| 11:20.48 | fredericve | orn: and for the init script, the debian packaging in wheezy (asterisk 1.8.13.1) contains an init script with astcanary included |
| 11:21.02 | fredericve | debian ships it with astcanary enabled by default |
| 11:21.31 | orn | fredericve: I understood it conversely -- I understood it as such that if some asterisk threads are hogging the CPU, astcanary will reduce asterisk's priority, thus allowing you to access the machine and kill off the threads without power cycling the machine |
| 11:22.28 | orn | Thanks a lot for the info fredericve |
| 11:23.14 | Joram | Hello, i have a question: my provider does not support the change of the caller id. So what i wanted to make was that asterisk first calls a list of required people. An user answers his phone and the asterisk server hangs up. Then the call is transfered to the user so he will see the number of the customer on his screen. The problem is that after i do a Hangup() i cannot transfer the call anymore. |
| 11:26.19 | Greenlight | I don't understand the question. AFter you hangup the call, you can't do something else with it. |
| 11:26.43 | orn | Joram: After you hangup the channel, you can't do anything further with the call. Also, how can you transfer the call with a different caller-id, if your provider doesn't support changing the caller-id, even if you call the user first? |
| 11:27.55 | Joram | ok, so is it even possible to first get an user to answer, kill that call and than transfer the original call? |
| 11:28.29 | Greenlight | Yea, I guess with a bit of trickery, but why. |
| 11:29.17 | Greenlight | What are you actualy trying to do ? |
| 11:30.05 | Joram | because i cannot change the caller id, the incoming call must be answered by any of the employees but the employee must see the number of the client |
| 11:30.47 | Greenlight | So, as orn, asked, how are you going to make the employee see the numbre when the calls transferred ? |
| 11:31.57 | Joram | when i transfer(Application_Transfer) the call the number is visible to the employee but i cannot control the call anymore |
| 11:32.14 | Greenlight | Right, that means you CAN change the callerid ... |
| 11:33.22 | Joram | No, the Transfer function sends a redirect to the user and then the user calls the other number(that is how i thought how it worked) |
| 11:33.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:33.47 | Greenlight | YHour emplyee is a SIP phone ? |
| 11:33.59 | fredericve | orn: correct, but asterisk is a realtime application. Running astcanary is only useful when running asterisk in realtime priority. you would only run asterisk in realtime priority when there's other stuff that might influence it. |
| 11:34.03 | Joram | no regular cell phone |
| 11:34.16 | Greenlight | And your caller is dialling on, on SIP or ISDN ? |
| 11:34.56 | Joram | external number provided by the sip provider |
| 11:35.22 | fredericve | orn: if you run asterisk in normal priority, you would still be able to access the console when it's gone insane |
| 11:36.31 | Greenlight | That doesn't make any sense. *you* must be calling out to the employees phone over SIP. Unless your ITSP is allowing some odd redirect to any number and then billing you for it. Can we see a call where this happens? |
| 11:37.43 | Joram | yes, one minute |
| 11:38.47 | Greenlight | Or, after the transfer has happened, and they are still on the phone. What does a "core show channels" show? |
| 11:39.41 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:40.14 | orn | fredericve: All righty. Then I have no need for astcanary :D |
| 11:41.37 | Joram | 0 active channels |
| 11:41.37 | Joram | 0 active calls |
| 11:41.37 | Joram | 2 calls processed |
| 11:41.40 | orn | Joram: There's a good chance you can change the caller-id number if you insert a Diversion header. |
| 11:41.58 | *** join/#asterisk ziegleka (~ziegleka@keeper.etnetera.cz) |
| 11:42.25 | orn | Joram: Try adding a line like this to your dialplan (and change where needed): exten => x,n,SIPAddHeader(Diversion: <sip:YOURPHONENUMBER@YOURIPADDRESS>) before calling dial after you change the caller-id number |
| 11:42.29 | orn | see if you can make the call then |
| 11:42.41 | ziegleka | Hi, can someone help me with SIP Message Method? |
| 11:42.43 | Greenlight | Joram: I guess they are allowing the redirect on the nextwork side then. |
| 11:44.50 | Joram | ok, but will it be possible to first answer a call, hangup and then transfer it? |
| 11:44.58 | [TK]D-Fender | I fail to see why Transfer() is being used at all... |
| 11:45.20 | Greenlight | Although we can present any callerid we want, what we do with a "support" number which forwards to some mobiles out of hours, is prior to the caller being connected, we have an IVR "read out" the telephone number that's calling so they know who it is |
| 11:45.37 | Joram | Transfer wil redirect the call from the client directly to the employee so the employee will be able to see the correct phone number |
| 11:45.41 | Greenlight | [TK]D-Fender: Seems his ITSP allow a SIP Redirect to transfer the call at the network side |
| 11:45.59 | Greenlight | Thus mainining the original callerid. |
| 11:46.24 | [TK]D-Fender | the channel has the proper CallerID when it comes in in the first place |
| 11:46.24 | Joram | the changing of the caller id is blocked by the provider for security reasons |
| 11:46.32 | [TK]D-Fender | Since when is it changing? |
| 11:46.34 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 11:46.51 | [TK]D-Fender | incoming channel sets the callerid |
| 11:47.11 | Joram | i cannot change the outgoing caller id |
| 11:47.12 | Greenlight | He wants to sent the call to a mobile phone. |
| 11:47.25 | Greenlight | BUt, still have the original callerid. His ITSP don't allow that. |
| 11:47.42 | orn | Joram: Try adding a diversion header. You might well be able to make the call with the original caller-id if that header is present. |
| 11:48.59 | Joram | orn: i will give it a try |
| 11:49.10 | [TK]D-Fender | probably screwed then |
| 11:49.13 | [TK]D-Fender | use another provider |
| 11:49.53 | Greenlight | +1 |
| 11:50.07 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 11:52.03 | Greenlight | Joram: It's going to be rather tricky to do what you want from the dialplan directly. |
| 11:52.46 | [TK]D-Fender | If he says his provider doesn't permit it ... then it doesn't permit it. |
| 11:53.09 | [TK]D-Fender | Doesn't sound like "They're lying and there is a way around and we just want to make it difficult" |
| 11:53.46 | Joram | orn: no it doesn work |
| 11:54.15 | Greenlight | It works if he does a SIP Rediect |
| 11:54.26 | Greenlight | aka Transfer |
| 11:54.27 | [TK]D-Fender | Joram: If they don't let ou set it... then they don't let you set it.... |
| 11:54.47 | [TK]D-Fender | Greenlight: So what's the deal? |
| 11:54.48 | Greenlight | eg, They will allow the call to be transferred at *their* end. Which is odd. |
| 11:55.06 | Greenlight | eg, the call disappears completely from his system. |
| 11:55.27 | [TK]D-Fender | So what is missing here? |
| 11:55.31 | orn | Joram: Okay... maybe they only allow 302 redirects or something. |
| 11:55.43 | Joram | yes probably |
| 11:56.25 | Greenlight | Some ITSP's are funny about allowing any user to present ANY callerid they want, potentially ones they don't own. However, a SIP redirect, they know this is the real callerid. |
| 12:08.34 | [TK]D-Fender | heading to the office... |
| 12:13.46 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.27) |
| 12:16.56 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.27) |
| 12:27.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:27.52 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:34.48 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:38.46 | *** join/#asterisk dongola7 (~dongola7@50-197-3-153-static.hfc.comcastbusiness.net) |
| 12:41.22 | *** join/#asterisk Draecos (~Draecos@124-148-226-67.dyn.iinet.net.au) |
| 12:43.07 | *** join/#asterisk acidfu (~nib@unaffiliated/acidmen) |
| 12:51.22 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 12:58.05 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:01.45 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 13:02.03 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 13:13.51 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-oyorqhwbgwbicnxo) |
| 13:13.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:29.20 | fredericve | are there any tricks to get Cisco 7940 phones to register to asterisk? |
| 13:29.50 | fredericve | asterisk 1.8.13.0 that is |
| 13:30.11 | jmetro | can you see it trying to register and fail |
| 13:30.17 | fredericve | currently the phone sends a register, and asterisk sends 401, unauthorized |
| 13:30.51 | jmetro | sounds like a config issue on your end almost |
| 13:32.10 | *** join/#asterisk serafie (~erin@nat/digium/x-hboozhhimunxddxd) |
| 13:33.20 | jmetro | check sip.conf |
| 13:33.27 | jmetro | make sure user/pass match |
| 13:37.06 | [TK]D-Fender | fredericve: IIRC you need to set nat=force_rport |
| 13:37.08 | [TK]D-Fender | for your peer |
| 13:37.51 | *** part/#asterisk Joram (~joram@141.136.124.2) |
| 13:37.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:37.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:40.13 | *** join/#asterisk af_ (~getsmart@93-43-17-167.ip89.fastwebnet.it) |
| 13:41.42 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:41.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.03 | ziegleka | Hi, can someone help me with SIP Message Method? |
| 14:05.51 | *** join/#asterisk Katty (~Angela@68-184-14-250.dhcp.stls.mo.charter.com) |
| 14:06.25 | WIMPy | What kind of message? |
| 14:06.26 | *** join/#asterisk davlefouAMD (~david@197.15.217.90) |
| 14:06.32 | WIMPy | ~ask |
| 14:06.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:08.41 | ziegleka | WIMPy: SIP Extension for Instant Messaging, MESSAGE method |
| 14:09.16 | ziegleka | rfc3428 |
| 14:12.27 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.182) |
| 14:12.54 | ziegleka | the MESSAGE method is not in Allow list and I get SIP/2.0 405 Method Not Allowed |
| 14:13.43 | Qwell | ziegleka: Are you using a version of Asterisk that supports MESSAGE? |
| 14:14.19 | igcewieling | *blink* Someone who does not call SIP MESSAGE the term "SMS" |
| 14:14.24 | igcewieling | unpossible! |
| 14:14.53 | igcewieling | ziegleka: If you are not using Asterisk 11 you have no hope to do what you want. |
| 14:15.30 | ziegleka | Qwell: How can I it find out? |
| 14:15.51 | igcewieling | ziegleka: remember back to when you installed it. |
| 14:15.57 | ziegleka | igcewieling: Asterisk 1.8.20.0 |
| 14:15.58 | [TK]D-Fender | ziegleka: How do you NOT know what version you are running? |
| 14:15.59 | igcewieling | "core show version" will tell you. |
| 14:16.06 | [TK]D-Fender | \ziUpgrade |
| 14:16.12 | [TK]D-Fender | ziegleka: Upgrade |
| 14:16.14 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:19.51 | ziegleka | I'm trying to realize this http://docs.flashphoner.com/display/WCS/Instant+Messaging |
| 14:21.06 | WIMPy | What else do you need to know? |
| 14:21.07 | ziegleka | [TK]D-Fender: I know what version I'm running but I don't know which features/extensions this version supports.. I'm new in Asterisk |
| 14:22.15 | WIMPy | You have just been told that you requite at least version 11. |
| 14:22.22 | ziegleka | WIMPy: Do you know some repo with Asterisk for CentOS6 |
| 14:22.49 | WIMPy | no |
| 14:24.40 | ziegleka | WIMPy: ok |
| 14:24.44 | ziegleka | thank you all |
| 14:24.46 | [TK]D-Fender | ziegleka: SIP Message is supported in * 10+. * 10 is EOL, so basically it's time to upgrade to * 11 |
| 14:25.27 | [TK]D-Fender | ziegleka: AsteriskNOW's repo is for C6 |
| 14:25.49 | ziegleka | [TK]D-Fender: ?? |
| 14:26.07 | WIMPy | 10 already did it? |
| 14:26.27 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.182) |
| 14:27.08 | igcewieling | there is little 11 does which 10 does not. |
| 14:27.42 | ziegleka | http://packages.asterisk.org/centos/6/asterisk-11/x86_64/RPMS/ is OK? |
| 14:28.09 | [TK]D-Fender | ziegleka: Any reason why it wouldn't be? |
| 14:28.37 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:28.39 | ziegleka | No, I ask only, if it is ok for production use? |
| 14:28.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:28.52 | [TK]D-Fender | it's full release. People use it in production |
| 14:29.06 | igcewieling | I feel no pre-buit RPMS are ready for production. Install from source. |
| 14:32.58 | jmetro | ^^ |
| 14:35.08 | [TK]D-Fender | Well that is a singular opinion, but it is released to the general public for a stable branch.... |
| 14:37.50 | igcewieling | one of the problems with pre-built RPMS is you are mostly on your own if you need support. |
| 14:38.08 | ziegleka | once more thank you all, I'm going to upgrade ... exists some page with simple overview what features/extensions support which version of Asterisk |
| 14:38.42 | [TK]D-Fender | upgratde.txt |
| 14:38.47 | *** join/#asterisk Katty (~Angela@68-184-14-250.dhcp.stls.mo.charter.com) |
| 14:38.52 | [TK]D-Fender | upgrade.txt |
| 14:38.55 | Katty | good morning, cupcakes! |
| 14:38.58 | [TK]D-Fender | and the official WIKI |
| 14:39.28 | Katty | hugs [TK]D-Fender |
| 14:39.29 | mjordan | ziegleka: https://wiki.asterisk.org/wiki/display/AST/New+in+11 and https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 |
| 14:39.49 | [TK]D-Fender | Katty: mew. |
| 14:40.12 | Katty | [TK]D-Fender: did you have a nice weekend? |
| 14:40.34 | [TK]D-Fender | Katty: ok I guess... we'll see tomorrow post-x-ray |
| 14:40.41 | Katty | nods |
| 14:41.17 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 14:50.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 14:52.51 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 14:53.21 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:53.34 | Tuju | any idea's why my pstn-trunk does not ring at asterisk end, regardless that sip show registry - shows that it's registered? |
| 14:53.50 | Tuju | i can call outbound just fine. |
| 14:54.52 | tuxx- | did you check with `sip set debug on` if theres any sip coming in from the trunk>? |
| 14:56.07 | Tuju | i did do sip set debug peer <trunk> and it's all silent. |
| 14:56.24 | Tuju | feels like it's not considered being registered in pstn-side. |
| 14:56.24 | tuxx- | maybe your provider isnt routing the number to the correct trunk? |
| 14:56.39 | Tuju | it works if i hook the line into cisco desktop phone. |
| 14:56.57 | igcewieling | sounds like a NAT issue |
| 14:57.01 | tuxx- | yep |
| 14:57.03 | [TK]D-Fender | that doesn't say that * is configured correctly or the rest of your networking is proper for it |
| 14:57.10 | igcewieling | Tuju: use wireshark to verify you are receiving the packets |
| 14:57.14 | [TK]D-Fender | So go see if * is properly registering |
| 14:57.15 | Tuju | that's true that those are in different networks. |
| 14:57.27 | [TK]D-Fender | Forget wireshark |
| 14:57.30 | Tuju | the asterisk, my end is in public network and there is no nat whatsoever. |
| 14:57.31 | [TK]D-Fender | get it straight from * |
| 14:57.45 | Tuju | maybe i should especially say so, in sip.conf |
| 14:57.51 | igcewieling | [TK]D-Fender: (10:56:07 AM) Tuju: i did do sip set debug peer <trunk> and it's all silent. |
| 14:58.15 | jmetro | i hate that NAS's are not computers. They are beautiful at what they do, but desperately need a hard interface incase net locks up |
| 14:58.18 | igcewieling | hence the wireshark, since wireshark is a commonly known tool, maybe he'll accept that the packets are not getting to Asterisk and move on. |
| 14:58.57 | igcewieling | jmetro: *nod* Naval Air Bases have a TERRIBLE UI. |
| 14:59.18 | jmetro | NAB's? |
| 14:59.29 | igcewieling | ..er.. Naval Air Stations |
| 14:59.49 | igcewieling | I think armys have bases, navys have stations. |
| 15:00.00 | Tuju | nope, no luck with nat this-n-that |
| 15:00.04 | igcewieling | If people should stop killing each other we would need neither. |
| 15:00.13 | jmetro | I think Naval Bases are a thing too |
| 15:00.17 | Tuju | i tried to look into fw logs and there is nothing suspecting either. |
| 15:00.22 | [TK]D-Fender | tutShow us your system registering. |
| 15:00.32 | [TK]D-Fender | Tuju: Show us your system registering. |
| 15:00.56 | Tuju | [TK]D-Fender: sip debug messages? |
| 15:01.07 | [TK]D-Fender | clearly |
| 15:02.18 | Tuju | those do not appear right now |
| 15:02.32 | Tuju | sip show registry lists that this line is 'registered' |
| 15:02.47 | Tuju | can i somehow trigger it re-register? |
| 15:03.03 | igcewieling | Tuju: if you are not seeing sip debug when asterisk registers to your provider then you are using the command wrong. |
| 15:03.17 | [TK]D-Fender | Tuju: "sip reload |
| 15:03.25 | Tuju | i've used that, yes |
| 15:03.34 | Tuju | and i used 'sip set debug perustele' |
| 15:03.40 | Tuju | that's the name of the line i'm using. |
| 15:03.50 | Tuju | and i used 'sip set debug peer perustele' |
| 15:05.21 | igcewieling | try debugging by IP |
| 15:06.08 | [TK]D-Fender | no, stop restricting COMPLETELY |
| 15:06.17 | [TK]D-Fender | "sip set debug on" <- |
| 15:07.22 | Tuju | i try that |
| 15:07.50 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 15:08.24 | Tuju | cannot see any register messages to that line |
| 15:08.29 | Tuju | that makes sense then |
| 15:08.46 | Tuju | although there is quite a flow of stuff going in screen |
| 15:09.04 | [TK]D-Fender | and you did a "sip reload" after that? |
| 15:11.30 | Tuju | yes |
| 15:12.27 | Tuju | sip show registry |
| 15:12.30 | Tuju | 79.134.121.233:5060 N 35893157xXxX 105 Registered Mon, 26 Aug 2013 18:00:35 |
| 15:12.46 | Tuju | sip show channels - does not list it doing REGISTRY like other devices. |
| 15:13.11 | [TK]D-Fender | show us |
| 15:13.14 | [TK]D-Fender | ~pb |
| 15:13.14 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 15:13.15 | [TK]D-Fender | ^^^ |
| 15:14.14 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:18.32 | Tuju | http://pastebin.ca/2437980 |
| 15:18.53 | Penguin | If you see that it is currently registered, that means a REGISTER packet was sent at some point. The sip debug will show those packets. |
| 15:19.21 | Tuju | what i could see from that flood, there were no packets with 'perustele' |
| 15:19.31 | Tuju | which implies that that's the problem. |
| 15:20.43 | Tuju | what i understood, registration packets are not visible with 'debug peer ' becuase it hooks the debug messages to account that is not know before registration. |
| 15:21.40 | Tuju | [Aug 26 18:20:59] NOTICE[9916]: chan_sip.c:21353 handle_response_register: Outbound Registration: Expiry for 79.134.121.233 is 120 sec (Scheduling reregistration in 105 s) |
| 15:21.40 | [TK]D-Fender | tutI do not see you sitting at * CLI issuing the 2 commands you were giving and waiting for output. |
| 15:21.46 | [TK]D-Fender | Tuju: I do not see you sitting at * CLI issuing the 2 commands you were giving and waiting for output. |
| 15:22.03 | Tuju | sip reload and set debug on ? |
| 15:22.19 | [TK]D-Fender | REVERSE order |
| 15:22.24 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 15:22.53 | Tuju | but that would mean that i paste some 20 phones messages here which are at least irrelevant and maybe some information that i don't want to publish. |
| 15:23.00 | igcewieling | what is the point of doing a sip reload to trigger a registration if sip debug is off? |
| 15:23.15 | jmetro | cause its fun. |
| 15:23.15 | igcewieling | Tuju: who told you this would be easy. It isn't. |
| 15:23.35 | igcewieling | Tuju: consider trying to solve this when you have less traffic on your PBX. |
| 15:24.35 | navaismo | *or use sngrep* some people said its awesome.. i cant tell that since my 14" cant support it |
| 15:25.33 | jmetro | or |
| 15:25.35 | jmetro | do it |
| 15:25.41 | jmetro | and then debug the log files rather than console |
| 15:26.49 | Penguin | I don't see why debug by IP is such a problem. |
| 15:27.34 | igcewieling | Penguin: maybe the packets are coming from a different address then he thinks. |
| 15:28.18 | igcewieling | I don't understand why this is such a big deal. If you can't get SIP debug without lots of drama, what happens when he has to do something complicated? |
| 15:28.47 | Penguin | failure |
| 15:29.12 | igcewieling | indeed. |
| 15:30.16 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 15:30.26 | Tuju | http://pastebin.ca/2437990 there should be register and it appears to me be successfull. |
| 15:31.27 | Katty | hugs chuckf |
| 15:31.46 | chuckf | hugs Katty right back |
| 15:31.59 | Katty | how're you dear? |
| 15:32.19 | *** join/#asterisk bondar (~bondar@197.156.132.62) |
| 15:32.51 | chuckf | I'm doing well. Celebrated the Mrs' birthday over the weekend |
| 15:32.52 | [TK]D-Fender | Tuju: Now show us a call attempt |
| 15:33.08 | chuckf | and lots of work to look forward to this week |
| 15:33.09 | Tuju | [TK]D-Fender: can i put that peer filter on? |
| 15:33.16 | [TK]D-Fender | Tuju: No |
| 15:33.52 | chuckf | Katty: how are things in your world? |
| 15:35.50 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.182) |
| 15:35.50 | Katty | chuckf: good good. made a chocolate espresso cheesecake this weekend which was fun! looking forward to the weekend, but you already knew that (= |
| 15:37.20 | *** join/#asterisk bondar (~bondar@197.156.132.62) |
| 15:37.29 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:37.30 | igcewieling | <-- going camping this weekend |
| 15:38.16 | Tuju | HA!!! 'perustele' rejected because extension not found in context 'plop'. |
| 15:39.01 | igcewieling | Tuju: that would have shown up even without sip debug, you must not have noticed it before. |
| 15:39.14 | igcewieling | ....not noticed... |
| 15:39.23 | [TK]D-Fender | Tuju: And that is precisely what you told them to dial in your REGISTER statement |
| 15:39.40 | Tuju | ---- now it works ------- tralla lalla laaa !!!! ------ |
| 15:39.48 | [TK]D-Fender | register => 35893157xXxX:yyyyYYYYYY@79.134.121.233/perustele |
| 15:40.14 | Tuju | yep, i never configured that trunk before so i was not so sure what to put into dialplan. |
| 15:40.26 | Tuju | Haaaa, tastes so sweeeeeeeet. |
| 15:40.33 | Penguin | If you ask them to send your calls to extension 'perustele', be prepared to expect them to send to that extension. |
| 15:40.35 | Tuju | really sweeet. |
| 15:41.00 | *** join/#asterisk natschil (~nathanael@stgt-5f70f023.pool.mediaWays.net) |
| 15:41.04 | Tuju | now i need to call someone and ask 'em to call back.... |
| 15:41.20 | igcewieling | sounds like you need to read sip.conf.sample and the Asterisk book. |
| 15:41.54 | AllanButton | oh my turn |
| 15:41.56 | AllanButton | ~book |
| 15:41.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:42.48 | Tuju | made a call, and it worked. now he called back and it worked fine! :) |
| 15:44.28 | igcewieling | Does anyone happen to know what an "acceptable level of noise (in dBm)" on an analog line (according to telco standards) would be? |
| 15:45.04 | igcewieling | The only one I could find is published by Telcordia, so it will cost the GDP of a small country to get a copy of it. |
| 15:46.21 | Tuju | so nice to use self-hacked system for calls. :) |
| 15:46.44 | Penguin | Self-hacked? Self-assembled, maybe. |
| 15:46.46 | coppice | acceptable noise varies a lot |
| 15:47.04 | dpeloquin | i think +/- 2db, igcewieling |
| 15:47.58 | *** join/#asterisk Dovid (~Dovid@69.127.106.197) |
| 15:48.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:48.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:48.35 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 15:51.48 | *** join/#asterisk tharkun (~0@unaffiliated/tharkun) |
| 15:53.08 | tharkun | Good $(date +%P) If I need to link two remote offices which allready have a VPN between them is it more efficient to route traffic through it or use the open internet for that? |
| 15:53.53 | [TK]D-Fender | clearly more efficient for open internet. the question is how desperate you are for that 0.01% |
| 15:55.11 | tharkun | [TK]D-Fender: If ain't broken I will rather not fix it. I can live with the difference :D |
| 15:55.12 | Penguin | 1/10000 (one ten-thousandth) isn't much. |
| 15:56.13 | Greenlight | I would have thought the overhead higher, for such small packets |
| 15:56.19 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:57.09 | igcewieling | Greenlight: depends on the VPN I suppose. |
| 15:57.50 | Greenlight | Yea, IPSEC i'd have thought to be most efficient, but still not as much as 1/10000. OpenVPN less efficient, I'd have imagined. |
| 15:58.34 | Greenlight | Would be interesting to see some numbers for VPN usage with SIP traffic. |
| 16:05.33 | Tuju | ouki douki, gotta go. thanks to all. |
| 16:05.40 | *** part/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 16:09.42 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:5c42:5b4f:cf7a:3318) |
| 16:10.34 | coppice | igceweiling: the spec for exchange line cards is -68dBm, which is not far above the quantisation noise. I'm not sure what the spec for the customer end of a line |
| 16:12.17 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 16:13.31 | igcewieling | coppice: thanks. |
| 16:36.49 | *** join/#asterisk slidesinger (~slidesing@c-69-141-78-33.hsd1.nj.comcast.net) |
| 16:37.21 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 16:38.32 | natschil | Hello. Does there exist somewhere an option to tell asterisk to verify client side tls certificates? |
| 16:42.10 | jmetro | problem with cisco phones |
| 16:42.14 | jmetro | they keep coming us as "everyone busy here" |
| 16:42.21 | jmetro | when dialed direct |
| 16:44.12 | ChannelZ | SIP? SCCP? |
| 16:45.14 | jmetro | SIP of course |
| 16:45.41 | ChannelZ | well we need to see a SIP debug or verbose console or _something_ to make any determination |
| 16:45.53 | jmetro | two different cisco phones that each register to 2 unique lines [4 unique lines total] |
| 16:46.02 | jmetro | im not sure when it happens is the problem |
| 16:46.11 | jmetro | they work great on reboot, come back the next day and theyre busy and cant dial out |
| 16:47.25 | jmetro | oh, busy on reboot fresh too now. Huh |
| 16:47.44 | ChannelZ | well then capture some SIP debug next time it happens. Look at the sip registry and see what it says. Are their IPs changing but not re-registering, etc.. there's 50 different reasons |
| 16:49.54 | jmetro | trying a factory first |
| 16:50.20 | igcewieling | jmetro: you have a NAT port translation timing out. set qualify=yes and qualifyfreq=30 |
| 16:50.45 | jmetro | Oh, i like that answer, and its something i forgot to check that happens to be an issue on our network occasionally. |
| 16:53.55 | jmetro | oh i love sip prune realtime |
| 16:56.59 | cusco | why jmetro ? |
| 16:57.16 | cusco | talking about realtime |
| 16:57.45 | cusco | how do you people handle... Local members in realtime queues ... I mean, a call comes member is busy, and then member becomes available |
| 16:57.54 | cusco | and I must issue a queue shoiw <queue> |
| 16:58.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-oyorqhwbgwbicnxo) |
| 16:58.01 | cusco | so the member status is updated ? |
| 16:58.27 | jmetro | hm |
| 16:58.34 | jmetro | i dont think i do realtime queues. |
| 16:58.40 | cusco | hm |
| 16:58.52 | jmetro | the queue i have that works is defined in queues.conf and people login dynamically yes |
| 16:58.55 | jmetro | but its a ringall |
| 16:59.05 | jmetro | anything else is ridic |
| 17:00.17 | cusco | well we have had realtime queues before I started on asterisk.. I made a simple script that runs every X secs, performimg a core show channels, and if there is a call queue'ing not bridged, script issues a queue show <queue> |
| 17:00.37 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.182) |
| 17:01.03 | cusco | I guess this is a common issue, I made this script ages ago and just never cared about it since |
| 17:01.15 | cusco | I've been meaning to ask.. but.. well.. I'm asking now :p |
| 17:05.42 | *** join/#asterisk coolacid (~CoolAcid@unaffiliated/coolacid) |
| 17:05.58 | *** part/#asterisk coolacid (~CoolAcid@unaffiliated/coolacid) |
| 17:08.58 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 17:11.31 | Katty | hello my asterisk does not work at all how to fix plz is urgent thx |
| 17:11.54 | jmetro | Katty: you need to fix your intertubes, your asterisk capacitor seems to break on a weakly basis and might need a new shot of RF232 |
| 17:12.08 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:12.45 | Katty | oh noes the intertubes are broken!!! |
| 17:12.57 | cusco | put some tape on it |
| 17:13.10 | drmessano | Hello, this is off-topic but I have an Android phone that has a SIP client on it that connects to Asterisk and I am stuck on level 147 on Candy Crush Saga. Plz Help? |
| 17:13.15 | Katty | hi danny |
| 17:13.24 | drmessano | hi |
| 17:13.28 | cusco | lol |
| 17:13.30 | cusco | drmessano: use iphone |
| 17:13.33 | *** join/#asterisk brandor (~bran@vaoffice.inmotionhosting.com) |
| 17:13.37 | cusco | and it fixes your problems |
| 17:13.41 | Katty | ^- that. |
| 17:13.55 | drmessano | I would never own an Android phone. That was a complete and total troll |
| 17:13.59 | jmetro | yeah mainly because you wont be able to do anything since you'll be broke |
| 17:14.05 | drmessano | But CCS is still hard as hell on the iPhone |
| 17:14.07 | jmetro | [and the iphone will be a brick] |
| 17:14.18 | cusco | btw, there is a jb app for iphone, taht allows you to buy upgrades in apps... for free |
| 17:14.21 | Katty | candy crush saga seems likea huge waste of time |
| 17:14.21 | drmessano | How do you brick an iphone? |
| 17:14.28 | cusco | so candy crush, can buy stuff to pass levels easy |
| 17:14.29 | cusco | :) |
| 17:14.36 | drmessano | lol I have seen that app |
| 17:14.40 | drmessano | errr |
| 17:14.41 | jmetro | iphones come pre-bricked, it is their standard state of operation |
| 17:14.42 | drmessano | HAVENT |
| 17:14.45 | cusco | drmessano: send it to my address, I'll break it for you |
| 17:14.46 | drmessano | I need to check Cydia |
| 17:14.55 | cusco | drmessano: need a repo, but its iap free |
| 17:15.15 | coppice | candy crush has reduced the average IQ of Asia by at least 10 points |
| 17:15.17 | drmessano | jmetro, You mean Android phones.. You're thinking about the fact that they last like 3 months |
| 17:15.25 | drmessano | I've never seen someone kill an iphone |
| 17:16.06 | drmessano | "Why are there no upgrades for my Android phone?" "You won't have it long enough to see one" |
| 17:16.16 | jmetro | drmessano: lasting 3 months seems to be the average lifecycle of most apple devices in general.. i've had garbage low-end motorola phones last years until i dropped it on concrete |
| 17:16.21 | coppice | the failure rate of older iphones was very low, but the 4S and 5 are terrible |
| 17:16.46 | drmessano | jmetro, you have no idea what you're talking about. |
| 17:17.01 | brandor | Anyone know of a plugin/functionality to dynamically set a caller id based on the number dialed (like a lookup from a db). Upgrading our freepbx/asterisk from an ANCIENT version and the fixlocalprefix script i tied into previously is now deprecated |
| 17:17.23 | jmetro | i guess all my experience in retail, consumer electronics, consumer IT, and business IT doesnt qualify me :| |
| 17:17.24 | drmessano | Most people sell Apple devices at the end of 2 years, whereas most Android owners are happy to be eligible for an upgrade after swapping out phones 6 times |
| 17:17.45 | drmessano | jmetro, Your experience is only 100% different than anyone elses |
| 17:17.59 | drmessano | Because i've never heard or experienced such a thing |
| 17:18.00 | jmetro | youre the only person i've ever met that claims apples last longer than 6 months |
| 17:18.11 | Greenlight | My 3 year old iPhone will be going to a happy new owner when I eventually do upgrade. |
| 17:18.14 | drmessano | You dont know many people then |
| 17:18.15 | jmetro | including family-owned apple devices with planned obsolecense |
| 17:18.23 | drmessano | Again, you dont know many people |
| 17:18.28 | Greenlight | Now you've met two of us :) |
| 17:18.49 | drmessano | I can go swipe 20 Apple devices from offices right now, most of them are at least a year old. Some are pushing 3 years |
| 17:18.52 | drmessano | So whatever |
| 17:19.03 | igcewieling | all cell phones suck. |
| 17:19.37 | coppice | drmessano: most of the the 3 year old one will be fine. its the ones a few months old which will need repair |
| 17:19.52 | drmessano | But I can't say there's an Android device in the building older than 3 months at this point. I have a few users with models they've had for 2 years, but they're on their 4th or 5th device |
| 17:20.25 | drmessano | Most Android phones have shorter lives than boxes of Kleenex |
| 17:20.27 | coppice | my Nexus S is nearly 3 years old. I've never had a problem with it |
| 17:21.08 | jmetro | motorola defy going on ... what, 4 years? samsumg exhibit at 2 years, 2 galaxy S's since launch, 2 galaxy S3's for half a year.. no breaks or replacements. |
| 17:21.11 | jmetro | thats my family atm |
| 17:21.40 | jmetro | the defy has a cracked screen from dropping it on its edge on concrete, still works, full touch capability on the cracks too |
| 17:22.36 | drmessano | I feel sorry for your family. What antivirus are they running on their Android devices? Norton? McAfee? |
| 17:22.44 | jmetro | you run antivirus? lols |
| 17:22.54 | drmessano | Where did I say that |
| 17:22.55 | jmetro | ^ mac user |
| 17:23.04 | drmessano | [13:22:36] <drmessano> I feel sorry for your family. What antivirus are they running on their Android devices? Norton? McAfee? <--- |
| 17:23.07 | drmessano | Maybe re-read that |
| 17:23.45 | jmetro | 1. id never run antivirus, including on PC's., 2. smartphone antivirus? really? 3.mac users would think they need to buy antivirus, 4. apple troll is a troll |
| 17:24.05 | drmessano | I wouldn't know, I am not a Mac user |
| 17:24.15 | drmessano | 2. Android has more malware than windows |
| 17:24.17 | drmessano | 3. Uhhh |
| 17:24.21 | drmessano | 4. Profit |
| 17:24.44 | jmetro | since launch of android i've never seen a single peice of malware. |
| 17:24.51 | drmessano | Are you serious? |
| 17:24.55 | jmetro | mmmmhm |
| 17:24.57 | drmessano | Have you been living under a rock? |
| 17:25.07 | jmetro | obviously not if i've got smartphones bro |
| 17:25.07 | Qwell | Some people don't drool all over their phones while browsing the Play Store. |
| 17:25.09 | drmessano | You have all this experience, and you've never heard of Android malware? |
| 17:25.10 | Katty | jmetro IS the rock. |
| 17:25.16 | igcewieling | drmessano: maybe he doesn't install every shiny thing he sees on the internet? |
| 17:25.23 | Katty | monsieur Qwell |
| 17:25.28 | Qwell | Katty: WHAT |
| 17:25.30 | jmetro | i've heard of it, but it seems like garbage that dumb users buy antivirus to prevent. |
| 17:25.39 | Katty | Qwell: congratulations on your engagement |
| 17:25.49 | [TK]D-Fender | [13:24]drmessano2. Android has more malware than windows <- "cmon, troll if you will for the other stuff, but this is just wrong... |
| 17:26.00 | drmessano | igcewieling, the lack of knowledge of such is baffling. You would almost never need to read anything on the internet regarding smartphones to not be aware of the proliferation of Android malware |
| 17:26.07 | igcewieling | Qwell: condolences on your engagement |
| 17:26.15 | drmessano | I am beginning to think *I* am being trolled |
| 17:26.18 | *** part/#asterisk brandor (~bran@vaoffice.inmotionhosting.com) |
| 17:26.25 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-220-156.pools.arcor-ip.net) |
| 17:26.32 | igcewieling | drmessano: I know there is Android Malware, no I've never encountered Android Malware. |
| 17:26.42 | jmetro | Android runs Linux.. unlesss you directly install the script on your phone to run "rm -rf *" |
| 17:27.25 | drmessano | wow |
| 17:27.36 | jmetro | i would be intrigued to see android malware, i cant imagine what it would do |
| 17:27.37 | drmessano | It's a bit easier than that |
| 17:27.42 | [TK]D-Fender | iPhone 4 is 3 years old now and I have several in the field. They all work. My Android phones also all work with 1 having physical issues.... |
| 17:27.42 | jmetro | like, flash ads on the screen or something ? lol |
| 17:27.44 | coppice | you hear lots of rumours about android malware, but I've never seen an infected phone |
| 17:28.22 | jmetro | Tons of antivirus, but its just hype. |
| 17:28.54 | drmessano | I've had 2 infected devices come across my desk. Both of the infections were basically stealing addressbook data and passwords |
| 17:28.57 | jmetro | Im thinking dr messano works in an office with a bunch of loose nuts behind the keyboard and thats why theyve gotta go iphone in the first place. |
| 17:29.00 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.182) |
| 17:29.06 | drmessano | jmetro, hardly |
| 17:29.15 | drmessano | jmetro, your ignorance is amazing though |
| 17:29.30 | jmetro | i guess all the other people who disagree with you are ignorant too. |
| 17:29.37 | jmetro | So i'll be happily and proudly ignorant. |
| 17:29.39 | drmessano | I dont see anyone disagreeing |
| 17:30.07 | drmessano | Except maybe [TK]D-Fender calling me out of my "More malware than windows" comment. That was actually expected |
| 17:30.21 | coppice | ios apps can read the contacts just as easily as android apps. |
| 17:30.27 | [TK]D-Fender | I have factors that favour each platform both for business and personal use. I dislike Apple's walled-ed garden and lack of customization, and dislike Android makers lack of updates and lower average stability. |
| 17:30.53 | jmetro | I've definitely seen garbage terrible android phones that i would never suggest be purchased or supported |
| 17:30.59 | jmetro | but thats the glory of an open market. |
| 17:31.20 | igcewieling | I won't buy apple for a couple of reasons, but mostly because you can't remove the battery |
| 17:31.51 | coppice | you can't remove the battery from an increasing number of things |
| 17:32.24 | [TK]D-Fender | Given I've got iPhone4's in the field still working fine, I don't really care about the battery deal... Unless you're a heavy user and can't charge through the day I can't picture my guys swapping... |
| 17:32.29 | drmessano | I've seen most Android phones are garbage terrible and I would never suggest purchasing. That's why i've nudged everyone towards iPhone, which isn't easy in a BYOD world.. but when they stop having to visit AT&T or Verizon once a month for support, they thank me |
| 17:32.32 | jmetro | People like the business model of "Distribute something very cheap to produce, with as simple of an interface as possible so the dumb user doesnt get confused and angry" which is why things look more apple-like |
| 17:33.16 | [TK]D-Fender | jmetro: It's not the Apple is "simple" (though it is an aspect), so much as that it is consistent |
| 17:34.09 | igcewieling | coppice: apparently people don't remove the battery from their cell phone when they are buying drugs or meeting up with the mistress |
| 17:34.10 | [TK]D-Fender | that vision has it's downside though. Including lack of WiFi standards support, no alternative input options, no custommization, etc |
| 17:34.10 | jmetro | a glass figurine locked in a cabinet is sure consistent and perfect, as long as noone interacts with it, it will never break |
| 17:34.29 | jmetro | the perfect phone, no users allowed. |
| 17:34.34 | igcewieling | [TK]D-Fender: I want a removable battery for PRIVACY |
| 17:34.50 | module000 | jmetro: preferrably we'd all force our 'users' into a trash compactor, and get on with doing productive things |
| 17:35.19 | jmetro | igcewieling: iphones collect GPS data at all times even with GPS turned off. connecting the phone to any mac or PC with Itunes installed will dump al lthe GPS data to it which you can then view. |
| 17:35.21 | [TK]D-Fender | igcewieling: Don't worry, the transmitter implanted in your molars is still sufficient ;) |
| 17:35.39 | igcewieling | jmetro: even with the battery removed? |
| 17:36.06 | igcewieling | [TK]D-Fender: I thought I might be overly paranoid until the Snowden leaks. 8-| |
| 17:36.06 | jmetro | battery removal shuts it off of course, but you cant on the iphone. |
| 17:36.23 | igcewieling | jmetro: exactly why people should not buy iphones |
| 17:36.27 | jmetro | with no battery there is no way to power the GPS collection device. |
| 17:36.35 | drmessano | Yeah, because who wants an iOS device with a working MS Exchange client when you can get an Android device that may scramble its ActiveSync settings every few months, requiring the account to be setup again. Silly damn users wanting email |
| 17:36.49 | drmessano | But at least they can make calls, most of the time between reboots |
| 17:37.20 | jmetro | i guess i'm one of the lucky few who have never had activesync decide to bork my settings. |
| 17:37.27 | jmetro | and one of the lucky few who never gets viruses |
| 17:37.36 | jmetro | and one of the lucky few who doesnt have their phone spontaneously combust |
| 17:37.40 | jmetro | either that, or your users are bad. |
| 17:37.41 | igcewieling | my android crashes about as often as my Win 98. |
| 17:37.47 | [TK]D-Fender | drmessano: That is why I've got iPhones here.. our Shitty Lotus Domino uses ActiveSync tech for PDA sync and Apple's natice support w/ profile works instantly without an extra install and the default user experience is superior |
| 17:38.04 | jmetro | [TK]D-Fender: THE NAME OF THE DEVIL ! THOU SHALT NOT SPEAK LOTUS's NAME |
| 17:39.02 | cusco | so are we past iphone stuff? |
| 17:39.13 | Katty | i was wondering that earlier |
| 17:39.15 | Qwell | gives cusco a stick too |
| 17:39.16 | Katty | but i don't think so, not yet |
| 17:40.28 | drmessano | I wasnt finished hearing about jmetro's years of experience and never having a bad device cross his lap. :( |
| 17:40.39 | drmessano | I was intrigued |
| 17:40.52 | jmetro | Actually just deployed about 50 devices, mix of iphone and android, through verizon to one of our clients |
| 17:40.52 | drmessano | I skipped lunch for it |
| 17:40.53 | Katty | you being intrigued leads to bad things. |
| 17:41.18 | drmessano | Katty, I know :) |
| 17:41.25 | jmetro | The iphones gave us the most problems because every user had to enter a million questions to get the phones started, including security and billing information. |
| 17:41.30 | Katty | drmessano: thank goodness you've been married off :P |
| 17:41.36 | Katty | drmessano: or you'd really have no manners! |
| 17:41.50 | drmessano | Katty, who says I have any now? |
| 17:42.05 | drmessano | Katty, The Doctor LIVES! |
| 17:42.30 | Katty | i heard that, but still haven't caught up on dr. who |
| 17:42.53 | jmetro | Katty: Old or new series? Plz say tom baker. |
| 17:43.08 | Katty | well, i mean i haven't caught up on the last season or two |
| 17:43.12 | Katty | so i'm not going to watch the new bits |
| 17:43.18 | Katty | BUT, i have seen some of the older ones. |
| 17:43.27 | Katty | the very very first ones, in black and white |
| 17:43.54 | jmetro | oh lawd, the first doctor. I started at tom baker myself. He's the most prolific doctor and its when things started getting interesting. |
| 17:44.27 | Katty | so tom baker is the one with the very young...uhh, daughter i think it was? |
| 17:44.46 | Katty | seems like there was an episode early on about cave men |
| 17:45.05 | jmetro | Tom baker has a companion that looks exactly like Rose Tyler. |
| 17:45.10 | Katty | then they discover the dalek city |
| 17:45.12 | jmetro | hes the iconic doctor |
| 17:45.32 | Katty | ah right. the one with the exceptionally nice scarf. |
| 17:45.49 | Katty | no i've not watched those |
| 17:46.00 | jmetro | Yes, scarfman |
| 17:46.25 | Katty | i should watch them. |
| 17:46.33 | jmetro | Netflix has a surprising amount of the old stuff. |
| 17:46.37 | Katty | but i'm not starting over with the very first black and white ones. |
| 17:46.44 | Katty | those put me to sleep |
| 17:46.50 | jmetro | yeah, no guilt there. |
| 17:47.05 | Katty | there's also game of thrones to catch up on |
| 17:47.25 | Katty | battlestar galactic, and star trek too. |
| 17:47.36 | Katty | but...i just can't be bothered with watching tv lately. perhaps when it starts getting too cold to be outside |
| 17:47.53 | Katty | and when i've ran out of JR ward novels. |
| 17:47.55 | jmetro | Netflix on the screen while you work :3 |
| 17:48.46 | Katty | been too busy at work lately for that. but it comes in spurts. |
| 17:49.01 | drmessano | I could probably skip everything except Breaking Bad right now |
| 17:49.03 | Katty | surely it will calm down here shortly |
| 17:49.15 | drmessano | Thank goodness my wife watches MTV reality crap on demand and I can skip it |
| 17:49.30 | Katty | that doesn't sound pleasant. |
| 17:49.37 | Katty | but we all have our strange quirks |
| 17:49.41 | jmetro | jersey shore x.x |
| 17:49.46 | drmessano | I don't need an excuse not to watch TV when "Road rules/Real World Challenge" is on. Blech |
| 17:50.04 | Katty | i don't know what those are |
| 17:50.19 | igcewieling | I hope this customer who thinks we are their IT department gets pissed off and leaves. |
| 17:50.33 | tm1000 | igcewieling: I think you can just uninstall the dahdi module instead of just moving that file |
| 17:50.43 | drmessano | "Becky like totally snuggled with Shawn and like he's a rival but like he's so hot so like OMG I am going to kick Tara's ass and then like totally win tomorrow with Steve" |
| 17:50.51 | drmessano | MINDSPLODE |
| 17:51.02 | jmetro | drmessano: a million times less brainhurty than jersey shore / toddlers in tiaras. |
| 17:51.06 | Katty | i don't speak highschool. |
| 17:51.25 | Katty | my brain didn't parse most of that sentance |
| 17:51.34 | Katty | tho i'm guessing there's some sort of dramatic love triangle going on. |
| 17:52.19 | drmessano | Well, we have an agreement that I don't even tolerate Jersey Shore. I am fine doing work on my laptop while she watches some of that crap.. Even Grey's Lobotomy, but.. Jersey Shore is for when I am at the Gym or out working on a project and she's home alone |
| 17:52.36 | drmessano | Some. Things. Are. Just. Intolerable. |
| 17:52.49 | jmetro | I'm glad my futurewife prefers Buffy the Vampire slayer over reality TV |
| 17:52.58 | tzanger | futurewife. lol |
| 17:53.20 | Katty | you know i tried to watch buffy again about 2 or 3 years ago |
| 17:53.26 | Katty | i used to like it. |
| 17:53.33 | Katty | now it's all just annoying. weird how things change :/ |
| 17:53.36 | jmetro | start at season 2 or 3 at least, to skip the bad. |
| 17:53.39 | drmessano | She said to me the other day... "Guess I am going to GTL" I had to ask.. "You know, from Jersey Shore... Gym, Tan, Laundry". I asked her to please stop |
| 17:55.24 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 17:55.51 | Katty | jmetro: you're probably right. |
| 17:56.00 | Katty | jmetro: i ended up finding a little series called Lost Girl |
| 17:56.05 | jmetro | ^ Yes |
| 17:56.06 | Katty | jmetro: it was a suitable alternative |
| 17:56.11 | jmetro | lost girl is also fun. |
| 17:56.22 | drmessano | Buffy seemed pretty cool when I was introduced to that one really hot chick |
| 17:56.31 | Katty | yeah |
| 17:56.33 | Katty | the brunette |
| 17:56.35 | Katty | what was her name? |
| 17:56.39 | drmessano | Eliza Dushku |
| 17:56.45 | drmessano | PreeeeeeOW |
| 17:56.55 | jmetro | oh, Faith? |
| 17:56.59 | jmetro | she was not pretty in the series :| |
| 17:57.01 | Katty | yes. she was quite lovely |
| 17:57.17 | Katty | and then there was that cute little thing in lost girl |
| 17:57.20 | Katty | oh what was her name... |
| 17:57.31 | jmetro | Katty: all of them, or the gothy one. |
| 17:57.31 | Katty | the human |
| 17:58.13 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 17:59.06 | jmetro | Kenzi was the one i like <.< |
| 18:01.04 | Katty | yes. kenzi. |
| 18:01.08 | Katty | the human. lol |
| 18:01.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:01.41 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 18:02.22 | jmetro | Nothing beats "Once More, with Feeling" from Buffy |
| 18:03.52 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:03.54 | *** join/#asterisk deegen (~deegen@S01060023bee90320.gv.shawcable.net) |
| 18:04.20 | [TK]D-Fender | jmetro: "Dr Horrible's Sing Along Blog" <- |
| 18:05.57 | ChannelZ | ...was OK but overrated I think |
| 18:07.46 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 18:11.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:11.11 | jmetro | ^ ChannelZ got it |
| 18:11.45 | *** join/#asterisk vvac (~vvac@178235025201.wroclaw.vectranet.pl) |
| 18:11.52 | jmetro | Mm, beefaroni |
| 18:12.06 | vvac | Hello all |
| 18:12.41 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 18:12.59 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
| 18:13.02 | vvac | Anyone eager to help solving a dahdi issue? |
| 18:13.24 | navaismo | ~ask |
| 18:13.24 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:13.26 | jmetro | just ask |
| 18:15.57 | igcewieling | nobody WANTS to help, we are all here because of OCD or court a order. |
| 18:16.23 | jmetro | well, i'm here because i usuall need someone who gets more sleep to fix my problems |
| 18:17.17 | vvac | I've got a four span E1 digium card with four E1's connected so total amount of dahdi channels is 120 but I very often get a CHANUNAVAIL when dialing this extension |
| 18:17.43 | vvac | the message is something like: evertyone busy/congested 1 0:0:1 |
| 18:18.44 | vvac | of course there ARE channels available "dahdi show channels" and "pri show channels" shows that currently used is only 10-15 of 120 |
| 18:19.08 | vvac | would you be se king and point me what could be a cause of such behavior? |
| 18:19.11 | vvac | kind* |
| 18:19.28 | igcewieling | vvac: get the HANGUPCAUSE when the call fails. Then send that to your telco. |
| 18:20.29 | igcewieling | you can also put the output of a failed call along with the pri debug of the failed call on a pastebin and hope someone wants to work through all the data |
| 18:20.53 | ChannelZ | and are you dialing a group? does it contain all the channels? |
| 18:21.05 | vvac | yes all channels are in the same group |
| 18:21.41 | ChannelZ | is it any number? |
| 18:22.14 | vvac | if you're asking a group number it is g0 |
| 18:22.28 | ChannelZ | no I mean it's dialing any phone number fails when it gets into that state |
| 18:23.24 | ChannelZ | (not to say that 'busy/congested' is a "failure", it can also just be a busy number, yeah?) |
| 18:24.25 | jmetro | true, and having 120 lines available does not mean you have 120 available endpoints |
| 18:24.38 | vvac | I investigated that when it gets a busy the message is: "evertyone busy/congested 1 0:1:0" and then dialstatus is set to "BUSY" |
| 18:25.29 | igcewieling | vvac: neither of those messages mean much. you just said the dialstatus was CHANUNAVAIL not BUSY. |
| 18:25.33 | vvac | but in decribed situation message has "1" at the end and dialstatus says "CHANUNAVAIL" |
| 18:25.53 | [TK]D-Fender | vvac: We'd have to see configs, the actual call debug and channel dumps |
| 18:26.24 | vvac | I will provide |
| 18:26.48 | ChannelZ | just got "I will survive" stuck in his head |
| 18:29.29 | vvac | which files do you need to see? |
| 18:30.29 | *** join/#asterisk vvac (~vvac@178235025201.wroclaw.vectranet.pl) |
| 18:34.34 | vvac | http://pastebin.com/vWcT2L9a -> this is chan_dahdi.conf and i'm working on a dumps. Let me know if something else will be necessary |
| 18:36.45 | jmetro | hm, im probably doing this wrong |
| 18:36.57 | cusco | vvac: are the PRI connections from the same telco? |
| 18:36.59 | cusco | all 4? |
| 18:37.06 | jmetro | Registering a phone to the same extension multiple times..gives them more lines right |
| 18:37.08 | vvac | yes |
| 18:37.25 | igcewieling | frame slips would likely show up as HDLC abort errors in the CLI |
| 18:38.13 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 18:38.36 | igcewieling | jmetro: technically doing that causes a migrane, but most phones just ignore the extra registrations and register only once, others require you set only one line to register and tell that line to use more than one button(polycom style) |
| 18:39.32 | jmetro | igcewieling: right, thats what i thought, but at the same time i was headaching over "how does asterisk know its not busy if theres only one registration D=" |
| 18:39.43 | jmetro | but then realized i have about 100 phones doing this with 4 lines and 1 register. |
| 18:39.44 | igcewieling | jmetro: Asterisk does not |
| 18:40.24 | [TK]D-Fender | jmetro: What is your definition of "busy"? |
| 18:40.30 | igcewieling | jmetro: it just sends the call to the phone, if the phone accepts the call great. if not, then great |
| 18:40.43 | jmetro | [TK]D-Fender: on the phone or dialing |
| 18:40.57 | jmetro | and the phone decids "i've got multiple line appearances, im accepting the call" |
| 18:41.00 | [TK]D-Fender | jmetro: On a single registration this is perfectly easy |
| 18:41.00 | igcewieling | jmetro: how does Asteirsk know your SIP device cant handle 100 calls. |
| 18:41.23 | jmetro | i blame exhaustion for my question. sorry. |
| 18:41.24 | [TK]D-Fender | jmetro: Busy means you're doing something. Doesn't mean you can't do MORE |
| 18:41.39 | jmetro | i realized i literally have all my aastras doing 4 lines with 1 register already. |
| 18:46.54 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 18:48.27 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 18:52.00 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 18:55.16 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 18:56.19 | vvac | here's dump of a failed call |
| 18:56.22 | vvac | http://pastebin.com/ELackWHq |
| 18:58.07 | vvac | every of those calls set hangupcause 1 or 31 |
| 18:59.05 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 18:59.40 | vvac | and in peak hours it occurs more than 32 time per minute |
| 19:00.00 | j4jackj | apb1963_: sorry I cut you off but actually my b**** of a mum did. |
| 19:01.21 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 19:03.42 | [TK]D-Fender | vvac: I'd ask your provider and include a call with PRI debug enabled |
| 19:04.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:06.06 | vvac | hmm ok |
| 19:07.44 | navaismo | the reason is because cause 1 --> Unallocated (unassigned) number and 31 ---> normal. unspecified. |
| 19:07.48 | navaismo | http://networking.ringofsaturn.com/Routers/isdncausecodes.php |
| 19:08.22 | vvac | yes I know, but i checked those numbers and you can normally call them by the mobile |
| 19:08.41 | vvac | moreover when I tried again from asterisk the call was successful |
| 19:09.02 | vvac | it occurs randomly |
| 19:10.56 | navaismo | i guess your mobile provider is different than your e1 |
| 19:11.07 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 19:11.35 | vvac | that's true but as I wrote i tried again from asterisk just a while after I got an error and call was successful |
| 19:11.49 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 19:11.50 | navaismo | thats why you need to ask your telco, |
| 19:12.04 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 19:12.06 | vvac | will do :) |
| 19:12.15 | navaismo | asterisk only send the call request, if your telco connect it or not is another issue |
| 19:13.12 | vvac | tell me if my suspicions are possible |
| 19:15.06 | vvac | Is it possible that asterisk "see" a single E1 channel as ready and tries to put a call there but in fact another call at this channel has just ended and it's resources hasn't been freed yet? |
| 19:15.31 | [TK]D-Fender | that does not look like "glare" |
| 19:16.33 | vvac | is there any way to force a timeout between asterisk try to use the same channel again? |
| 19:16.41 | [TK]D-Fender | nope |
| 19:16.44 | vvac | :/ |
| 19:18.00 | j4jackj | My mum threatened to dsown me last night over the fact that I was using the computer 6 hours late. She knows if there's a teen and a toddler in the same house she should expect chronic total sleep deprivation. |
| 19:18.54 | vvac | It tries to send all traffic to the first 20 channels and another 100 stays untouched... |
| 19:20.17 | vvac | ok thank you for your time |
| 19:21.18 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 19:21.37 | navaismo | try with G0 and see if the bejaviour its the same |
| 19:21.53 | navaismo | h*** |
| 19:21.59 | j4jackj | what? |
| 19:22.18 | navaismo | h instead j |
| 19:22.20 | j4jackj | navaismo: why the hasteriskasteriskasterisk? |
| 19:22.32 | jmetro | <PROTECTED> |
| 19:22.38 | jmetro | try with h0 and see if the behavior is the same. |
| 19:22.43 | j4jackj | use s/bej/bah/ |
| 19:22.44 | Penguin | uh |
| 19:22.49 | j4jackj | *s/bej/beh |
| 19:22.51 | Penguin | G0 rather than g0 |
| 19:23.25 | jmetro | oh, G instead of g0 |
| 19:23.50 | Penguin | G starts at the opposite end of the channel group. |
| 19:24.01 | navaismo | j4jackj: i know howw to use the s/ but i dont want it |
| 19:24.14 | j4jackj | navaismo: baad |
| 19:24.31 | navaismo | mieh |
| 19:27.47 | jmetro | Tacos. |
| 19:28.33 | j4jackj | jmetro: 71@99.199.11.127 |
| 19:30.23 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:31.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.92) |
| 19:32.06 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.92) |
| 19:35.52 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 19:37.41 | boom^time | So I'm trying to build a simple IVR to record, then give the option to listen, rerecord, or save. So I need a temporary audio file that will be created with the record app that will then replace the persistent one on a save |
| 19:38.02 | j4jackj | ... |
| 19:38.04 | boom^time | Is there a good function/app for moving a file or doing this? |
| 19:38.09 | j4jackj | That can be done without a tempfile |
| 19:38.21 | cusco | monitor |
| 19:38.25 | cusco | record |
| 19:38.43 | boom^time | Record immediately saves to a file on # |
| 19:38.59 | cusco | youcan specify the filename |
| 19:39.27 | boom^time | I'm going to research monitor now thanks. |
| 19:39.29 | [TK]D-Fender | and then mv it after |
| 19:39.52 | [TK]D-Fender | monitor is to record a bridged call. |
| 19:40.10 | [TK]D-Fender | If you want to just let the caller make a recording then Record() is what you want |
| 19:40.16 | boom^time | Okay so I'm better off with record then |
| 19:40.32 | boom^time | right but what I'm asking is what is the best way to move the recording afterwards |
| 19:40.55 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.92) |
| 19:41.15 | boom^time | I can record to tmp.gsm, then on a save I'd need to move it to something like recording.gsm |
| 19:42.40 | navaismo | system |
| 19:42.42 | dongola7 | boom^time: take a look at http://pastebin.com/GeL8Ndzb. I did something similar using Record and System in the dialplan. |
| 19:43.12 | boom^time | Okay so you do have to use system. I was hoping their was an option that didn't require running system commands. But that'l work. |
| 19:43.18 | boom^time | there* |
| 19:43.23 | *** join/#asterisk clive- (~pirch@ti-226-252-24.telkomadsl.co.za) |
| 19:43.38 | boom^time | Thanks everybody. |
| 19:43.46 | navaismo | I use record, if they like nothing to do if not jump again to record again |
| 19:44.44 | boom^time | Yeah I'm making it a bit more complicated. If they try to rerecord over the original and they decide they just can't get anything better then they can still back out |
| 19:44.57 | boom^time | to keep the original. |
| 19:45.16 | [TK]D-Fender | That's not complicated |
| 19:45.24 | Penguin | That won't be too hard. Just cp the original once and save it always. |
| 19:45.34 | boom^time | More so than navaismo's method I mean. |
| 19:46.06 | *** join/#asterisk LonghornWork (~kvirc@74.115.99.1) |
| 19:46.17 | Penguin | Maybe a few more lines of dial plan, but nothing too serious. |
| 19:46.23 | boom^time | Right |
| 19:46.41 | boom^time | I suppose 'complicated' wasn't the right word. |
| 20:06.10 | ghost75 | is there an easier way than this: baikal carddav -> mysqldump of vcard data -> convert vcard to ldif -> import ldif to ldap -> create xml out of ldap |
| 20:06.49 | cusco | to be honest we have no such feature, however every time we record calls, use mixmonitor(), and it has argument at the end, for a shell command to be run. So the call is recorded to ramfs, then moved to its finel destination |
| 20:07.15 | cusco | ghost75: are you asking in the correct channel? |
| 20:07.33 | cusco | it does seem awfully complicated. do those conversions work flawlessly? |
| 20:07.35 | ghost75 | xml to be shown on cisco phone :) |
| 20:08.42 | ghost75 | ldif i got already, next step is to import it to openldap |
| 20:10.24 | cusco | thing is, do you need it on ldap ? or just the xml? |
| 20:10.52 | ghost75 | not now, maybe later |
| 20:17.17 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 20:18.34 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 20:23.20 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 20:24.53 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:26.33 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:31.15 | *** join/#asterisk Alex_Bkash1 (b4eaf8fd@gateway/web/freenode/ip.180.234.248.253) |
| 20:32.48 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 20:32.54 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 20:33.01 | dwayne | anyone ever use Asterisk (1.8.23.0) real-time and see INVITE retransmissions even after receiving a 100 response? |
| 20:34.29 | igcewieling | dwayne: did you read the doc referenced in the error message? |
| 20:34.43 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 20:38.10 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 20:40.31 | *** join/#asterisk SuperNull (~YoMommaSo@24-148-101-238.ip.mhcable.com) |
| 20:40.34 | dwayne | igcewieling, mostly but thanks for repointing me to that. I thought it may be related to the timert1 setting but its not. thanks |
| 20:40.57 | igcewieling | retransmissions are almost always a networking or NAT issue. |
| 20:42.34 | SuperNull | Hey guys, im noticing that my realtime user table .. keeps clearing the IP info .. way before registration time this is on 1.8.21 has anyone seen this ? |
| 20:59.53 | igcewieling | anyone know what this warning means? [Aug 26 16:59:21] WARNING[19510][C-00000302]: channel.c:3632 ast_waitfordigit_full: The FD we were waiting for has something waiting. Waitfordigit returning numeric 1 |
| 21:05.06 | cusco | file descriptor on wait for digit? |
| 21:05.07 | cusco | I have no idea |
| 21:05.29 | cusco | seemed that the file descriptor is taken? |
| 21:08.42 | igcewieling | If I knew, I'd not be asking. |
| 21:09.41 | SuperNull | TO THE SOURCE NEO! ;) |
| 21:10.37 | SuperNull | i always get stuck finding my solutions in the source.. its painful sometimes. |
| 21:13.14 | igcewieling | seems to happen mostly when I throw a couple of hundred short dtmfs at Asterisk |
| 21:15.10 | igcewieling | appears to require a restart of Asterisk to fix |
| 21:15.17 | SuperNull | randomly generated dtmfs ? |
| 21:16.48 | SuperNull | ive gotten stuck doing asterisk ghetto fixes.. at this point we have accepted that MWI wont work without at least openais.. and even still maybe not. we offload MWI to an external port and require public ip to the sip device. yey. |
| 21:17.27 | igcewieling | SuperNull: no, sending ASCII encoded into DTMF for a "fun project" |
| 21:17.40 | SuperNull | at least its just 'play. |
| 21:17.52 | igcewieling | SuperNull: MWI always works for us. |
| 21:18.03 | igcewieling | SuperNull: not entirely. I'm actually using it for something. |
| 21:18.13 | SuperNull | using realtime peers .. and a remote vm with some 1.4 and some 1.8 ? (vm server being 1.4) |
| 21:18.22 | igcewieling | ah, realtime. |
| 21:18.24 | SuperNull | ;) |
| 21:18.32 | SuperNull | realtime breaks pretty much any fun. |
| 21:18.45 | igcewieling | they should have called is realpainintheass |
| 21:18.55 | SuperNull | the traditional method was to use polling .. but that doesn't seem to work like it did in 1.2 properly. |
| 21:18.57 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 21:18.59 | igcewieling | SuperNull: WHY are you using realtime? |
| 21:19.03 | SuperNull | well. |
| 21:19.19 | SuperNull | 6000 or so sip lines.. someone thought asterisk was a solution to a class 5 sip switch. |
| 21:19.39 | igcewieling | is still waiting for a reason |
| 21:20.02 | SuperNull | multiple servers all using the same peer database.. with integration via mysql so forth.. |
| 21:20.08 | SuperNull | the database.. peer registration |
| 21:20.27 | igcewieling | there is a mostly valid reason. 8-| |
| 21:20.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.85) |
| 21:20.37 | SuperNull | yeah. |
| 21:20.48 | SuperNull | i think you advised me you were doing hard exported sip.conf files ? |
| 21:20.55 | igcewieling | though #exec /path/to/dbtosip.php might work |
| 21:21.05 | igcewieling | yup. |
| 21:21.08 | SuperNull | or opensips ;) |
| 21:21.09 | SuperNull | lol |
| 21:21.13 | SuperNull | what eva. |
| 21:21.33 | igcewieling | we left all our peers in the sippeers table, but export it to sip.conf instead of using realtime |
| 21:21.41 | SuperNull | i hear freeswitch/opensips can dominate the service provider market for stuff like this.. but obviously vastly more convoluted configuring it. |
| 21:22.38 | igcewieling | we run 1 - 2 million calls through asterisk per month |
| 21:23.15 | SuperNull | when does that #exec get ran .. on reload/startup only ? |
| 21:24.59 | *** join/#asterisk g_k (4245e770@gateway/web/freenode/ip.66.69.231.112) |
| 21:26.54 | g_k | hi folks.. What does "Probation passed - setting RTP source address to ..." mean? I changes from asterisk 1.8 to 11, and I started seeing this. Things work otherwise. |
| 21:28.35 | igcewieling | SuperNull: any reload |
| 21:28.39 | igcewieling | any sip reload at least |
| 21:30.25 | SuperNull | do you trigger a reload remotely when something gets added or do you auto-reload blindly on X interval.. ? |
| 21:30.43 | jmetro | i dont sip reload, i use realtime like a boss |
| 21:30.53 | jmetro | and sip prune peers for necessary fields [allow, disallow, password changes, etc] |
| 21:31.07 | SuperNull | jmetro do you have remote VM working ? |
| 21:31.11 | SuperNull | without openais ;) |
| 21:31.18 | jmetro | remote VM as in? |
| 21:31.43 | jmetro | the VM's are stored in the db with all the realtime stuff. |
| 21:31.47 | SuperNull | asterisk1 goes to asterisk2 for all voicemail and uses the externnotify to trigger it on the proper server |
| 21:31.52 | SuperNull | so your using odbc based storage vmail ? |
| 21:31.56 | jmetro | yus |
| 21:32.11 | SuperNull | does that 'just work' on multiple servers sharing it ? |
| 21:32.21 | jmetro | we have 8 production atm, yes. |
| 21:32.31 | SuperNull | hurm. |
| 21:32.34 | igcewieling | SuperNull: our peers don't register |
| 21:32.54 | igcewieling | SuperNull: you can use manager to connect to asterisk and issue a sip reload |
| 21:33.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.11) |
| 21:33.24 | *** part/#asterisk navaismo (~navaismo@189.241.19.115) |
| 21:33.31 | SuperNull | jmetro are you using any workarounds to make MWI work ? or does it check the odbc table completely instead of the /var/spool/asterisk/voicemail/context/yomommas#/INBOX/msgfilesbitch0001.txt |
| 21:33.33 | igcewieling | our peer add and remove script pokes all our asterisk server when something changes |
| 21:33.37 | jmetro | sip reload is nasty, deregisters things |
| 21:33.46 | SuperNull | yeahhh on realtime.. |
| 21:33.57 | jmetro | i dont have to reload anymore though |
| 21:34.17 | jmetro | SuperNull: i believe mwi relies on the odbc database as wel |
| 21:34.25 | jmetro | SuperNull: something in INBOX = red light on |
| 21:34.37 | jmetro | [we have a web interface that does the same thing for visual voicemail too ] |
| 21:34.38 | SuperNull | yeah.. some of these phones accept a vm count tho :-/ |
| 21:35.00 | jmetro | well yeah |
| 21:35.09 | SuperNull | jmetro is this on asterisk 10+ ? |
| 21:35.12 | jmetro | they just count(where dir=%inbox% |
| 21:35.27 | jmetro | this is ... 11 |
| 21:35.44 | jmetro | Connected to Asterisk SVN-branch-11-r378219 |
| 21:35.49 | *** join/#asterisk g_k (~kyriazis@cpe-66-69-231-112.austin.res.rr.com) |
| 21:35.51 | SuperNull | okay im running latest 1.8 due to some database issues (unable to create views, unable to modify table easily to work with 11) |
| 21:36.07 | SuperNull | in production with thaT? ! MADNESS ;) |
| 21:36.20 | jmetro | <.< |
| 21:36.29 | jmetro | "views" |
| 21:36.31 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 21:36.32 | jmetro | psh |
| 21:36.58 | SuperNull | alright, i will look into the ODBC voicemail if it works this nicely. maybe i will have an extra server (old voicemail) once this is over lol |
| 21:37.20 | jmetro | we have a SQL box that runs everything and all the asterisk boxes look at it |
| 21:37.37 | jmetro | working on offloading SIP too |
| 21:37.58 | jmetro | al in virtual machines |
| 21:39.35 | SuperNull | my goal would be honestly.. to have asterisk do voicemail period. have opensips handle direct registrations .. asterisk for media server related stuff. take the 'simple stuff' off asterisk. |
| 21:40.13 | jmetro | kamailio registration mwi and hints, asterisk media and routing. |
| 21:40.45 | SuperNull | routing .. ? |
| 21:40.57 | jmetro | dialplan |
| 21:40.59 | SuperNull | asterisk isn't super suited for LCR is it ? |
| 21:41.15 | SuperNull | mm im thinking to hard. |
| 21:41.22 | jmetro | call routing =p |
| 21:41.24 | SuperNull | i dont like the 'hacks' :-/ |
| 21:42.09 | jmetro | hacks? |
| 21:42.30 | SuperNull | our current system. |
| 21:44.04 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 21:47.14 | jmetro | ah |
| 21:47.14 | jmetro | yeah |
| 21:47.15 | jmetro | 1.98 |
| 21:47.17 | jmetro | 1.89* |
| 21:47.45 | jmetro | youget the point |
| 21:47.57 | Penguin | 1.89 what? |
| 21:48.30 | jmetro | i will make a penguin sandwich. |
| 21:48.34 | Penguin | 1.89 cats? |
| 21:48.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:49.54 | Penguin | If it's 1.89 cats, I want to know what you did with the other 11% of a cat. |
| 21:50.13 | Qwell | Penguin: You're looking at it the wrong way. |
| 21:50.48 | Qwell | The real question you should be asking, is what happened to the rest of the other 3 cats. |
| 21:50.48 | Penguin | I should wonder what he's doing with the 89% of the cat? |
| 21:51.22 | Qwell | The 1.89 is only by weight. |
| 21:58.58 | g_k | Hi folks. A question: What does "Probation passed - setting RTP source address to …" mean? I upgraded from asterisk 1.8 to 11, and I started seeing this. Otherwise, things work. |
| 22:04.14 | g_k | c'mon.. somebody must know.. :-) |
| 22:04.42 | file | there's a probationary period of time before Asterisk blocks other sources of media to the RTP port |
| 22:05.23 | g_k | can you define "other sources of media"? Do you mean other media types, or other clients? |
| 22:05.43 | file | other sources that may be sending packets to the port |
| 22:05.57 | file | ie: an attacker trying to inject audio |
| 22:07.45 | g_k | and after that probationary period is done, other sources are allowed to? |
| 22:07.45 | g_k | It seems to happen right after a connection is started |
| 22:07.45 | g_k | I'm getting 2 printouts, one for each end of the connection |
| 22:08.03 | file | no, other sources aren't allowed to |
| 22:08.33 | file | once probation is up it ignores packets from any other source than the one it has locked on to |
| 22:09.03 | Penguin | When the connection starts, there is a probation period. When the probation is over, the source address is set and no others are allowed. |
| 22:09.09 | g_k | so, before the period is over, it's trying to figure out which one is the right one, and then afterwards, it locked to the right one |
| 22:09.14 | file | yes. |
| 22:09.24 | g_k | ah, ok. thanks.. |
| 22:09.37 | g_k | another question that I have is about passing dtmf... |
| 22:09.55 | g_k | while I have an active call (to an ivr system) |
| 22:10.37 | g_k | if I pass the "Tt" parameters to Dial(), then it seems that dtmf key presses don't always go through. If I don't include "Tt", then they get passed rock solid. |
| 22:10.56 | Penguin | Do you know what T and t do? |
| 22:11.50 | g_k | yes, it has to do with call transfer. I have set up call transfer (both blind and attended) key sequences, and call transfer works. |
| 22:12.34 | [TK]D-Fender | g_k: What are you using for phones? |
| 22:12.42 | g_k | ht502 |
| 22:12.52 | Penguin | By having some Dial options, you are forcing asterisk to stay in the media path. Without seeing your settings, I would guess that you are otherwise allowing directmedia. |
| 22:13.41 | [TK]D-Fender | g_k: then use the ATA's transfer features and not Asterisk's |
| 22:14.03 | Penguin | If your direct media works fine but when asterisk is in the media path then you see problems, consider your dtmfmode setting that you are using. Maybe it's wrong. |
| 22:14.30 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-20-241.home.otenet.gr) |
| 22:14.41 | g_k | I have asterisk behind NAT, and I think direct media will not work (if I understand it right) |
| 22:15.36 | g_k | How will the phone's transfer features will work, if asterisk is on the path? |
| 22:15.47 | Penguin | SIP |
| 22:15.53 | Penguin | SIP and media are not the same. |
| 22:16.17 | g_k | because of NAT, I'm pretty sure that asterisk is in the loop to shuffle packets around (i.e. no direct connection) |
| 22:16.57 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ttvkeumvnxtsegfk) |
| 22:16.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:17.07 | g_k | what would be the best directmedia setting for my setup? |
| 22:17.30 | g_k | (I don't set it in sip.conf) |
| 22:18.48 | [TK]D-Fender | no <- |
| 22:20.03 | g_k | fender: if I use the device's transfer feature, I have to set it for all the devices that I have. It seems better to have it centralized on asterisk |
| 22:20.23 | Penguin | It's built in. It's not something you have to set or not set. |
| 22:20.45 | g_k | penguin: can you please describe what would make a "wrong" vs "right" dtmf setting? |
| 22:21.11 | Penguin | I prefer rfc2833. |
| 22:22.29 | [TK]D-Fender | DTMF based transfers = ass |
| 22:22.32 | g_k | so, when the ht502 sets up the call (ie. when I dial the number), there are no problems. The problem is only when I press dtmf during the call. |
| 22:23.02 | g_k | when do the ht502's dtmf settings apply? Do they apply in both situations, or only for some cases? |
| 22:23.07 | [TK]D-Fender | g_k: while in setup it is just audio from the phone to the HT |
| 22:23.16 | [TK]D-Fender | DTMF is AFTER the call is answered |
| 22:23.22 | [TK]D-Fender | for the mode to * |
| 22:23.27 | g_k | oh, i see |
| 22:23.36 | Penguin | Not only are they "ass," but the T _and_ t settings used together allow BOTH SIDES of the call to initiate a transfer. |
| 22:23.51 | [TK]D-Fender | phone is always "inband:" because that's what an analog phone is. SIP to Asterisk is another matter |
| 22:24.01 | [TK]D-Fender | Penguin: Indeed far worse |
| 22:24.05 | g_k | Ok, I'll try switching it around. I currently have SIP INFO followed by rfc2833 followed-by in-audio |
| 22:24.49 | Penguin | Followed by? |
| 22:25.16 | Penguin | I didn't know you got to choose a backup plan for dtmf. I thought you set the mode and that's the end of it. |
| 22:25.18 | g_k | ht502 has a "preferred dtmf order" |
| 22:25.26 | g_k | priorities |
| 22:25.48 | Penguin | Oh, I see. Set it on asterisk and set the device to the same value or to auto. |
| 22:25.56 | g_k | so, I assume dtmf setting is negotiated between ht502 and asterisk. Where do I set it on asterisk? |
| 22:26.09 | Penguin | It goes in the peer entry for the device. |
| 22:26.12 | g_k | there is no auto. :-) |
| 22:28.26 | g_k | so rfc2833 is the preferred method? |
| 22:28.51 | Penguin | I prefer it, but some people like the other modes. |
| 22:29.18 | g_k | hmm.. what would make one setting a good choice vs a bad choice, though? |
| 22:29.42 | g_k | putting it another way, what would make SIP INFO not work? |
| 22:29.48 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:fd9b:e5a4:e673:8ba0) |
| 22:29.49 | g_k | or be a bad choice |
| 22:31.40 | Penguin | What codec are you using? |
| 22:32.34 | g_k | ulaw |
| 22:32.45 | g_k | rfc2833 has problems |
| 22:34.11 | Penguin | I don't know what the advantage or disadvantage of using INFO would be, but if you don't use ulaw or alaw, you cannot reliably transmit dtmf inband. |
| 22:34.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.47) |
| 22:35.25 | g_k | but why would the "Tt" dial() options affect dtmf detection? |
| 22:36.15 | [TK]D-Fender | they don't |
| 22:36.32 | [TK]D-Fender | They STEAL DTMF for functionality instead of passing them on immediately |
| 22:36.41 | g_k | but that's where my problem is… If I don't specify "Tt", then I don't have any issues with dtmf |
| 22:37.03 | [TK]D-Fender | And it may do a weak detection on an attempt to steal that might have succeeded at the far end. |
| 22:37.05 | Penguin | It's not the T and t together that causes it. |
| 22:37.57 | j4jackj | ... |
| 22:38.18 | j4jackj | i use g722, will dtmf work ? |
| 22:39.26 | g_k | fender: if dtmf was passed inbound, I can understand that.. But it's send through rfc2833. |
| 22:41.02 | g_k | penguin: sure it may not that both T and t are specified. it could be only one of them. The end effect though is that one of them is causing an issue. |
| 22:42.36 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:44.52 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:45.25 | *** join/#asterisk Meaw (~salem@146.185.62.88) |
| 22:47.32 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 22:49.16 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 22:49.42 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 22:50.59 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:50.59 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 22:55.05 | g_k | just did some dtmf debug, and with rfc2833, the correct keys are passed through the destination (enabled full logger) |
| 22:55.11 | g_k | I'm stumped. :( |
| 22:58.57 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 23:02.16 | j4jackj | hi sruffell |
| 23:02.39 | sruffell | is nervous |
| 23:03.05 | j4jackj | why? |
| 23:03.11 | j4jackj | am I on a no fly list |
| 23:03.21 | paulc | maybe your reputation precedes you ;-) |
| 23:03.42 | sruffell | heh..no. It's just that when someone says hi to me here it's normally because there is some issue that I need to look at. |
| 23:04.06 | sruffell | so if that's not the case, then hi! |
| 23:04.57 | WIMPy | has issues |
| 23:05.53 | [TK]D-Fender | #psychology <- |
| 23:09.10 | *** join/#asterisk Alex_Bkash1 (b4eaf8fd@gateway/web/freenode/ip.180.234.248.253) |
| 23:19.04 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:fd9b:e5a4:e673:8ba0) |
| 23:21.55 | carrar | you mean |
| 23:21.57 | carrar | ##psychology |
| 23:27.48 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 23:33.59 | j4jackj | sruffell: I was just saying 'hi' because you're in the channel |
| 23:35.07 | [TK]D-Fender | j4jackj: 182 to go |
| 23:35.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.233) |
| 23:36.10 | sruffell | :) thanks |
| 23:39.28 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
| 23:43.47 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |