| 00:01.14 | Kobaz | anyone doing polycom blf using <attendent> |
| 00:04.48 | *** join/#asterisk ruied_ (~ruied@po-217-129-154-119.netvisao.pt) |
| 00:05.39 | *** join/#asterisk ruied_ (~ruied@po-217-129-154-119.netvisao.pt) |
| 00:05.57 | *** join/#asterisk QbY (~kelvin@73.61.137.138) |
| 00:15.54 | Naikrovek | not with attendant, i don't think |
| 00:16.05 | Naikrovek | but my receptionist uses blf extensively |
| 00:19.09 | *** join/#asterisk kaushal (~kaushal@115.118.253.143) |
| 00:19.11 | kaushal | Hi |
| 00:19.40 | kaushal | Anyone here have ever used http://pycall.org/ ? |
| 00:28.51 | QbY | kaushal: Looks like it generates the call file and throws it in your outbound |
| 00:29.16 | kaushal | QbY: can i pastebin the pycall script ? |
| 00:29.35 | QbY | i'm not the boss of paste bin.. i'd assume so.. |
| 00:29.42 | p3nguin | haha |
| 00:30.03 | kaushal | QbY: i mean i have created pycall script |
| 00:30.08 | QbY | anyone here work with an outbound call center/dialer? |
| 00:30.20 | kaushal | p3nguin: hi |
| 00:30.26 | p3nguin | waves |
| 00:30.31 | QbY | ok. |
| 00:30.52 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 00:31.04 | p3nguin | What's your question about dialers? |
| 00:31.34 | QbY | we now provide service to dialers.. big change for us.. so i'm looking for dialers who need lower cost termination. |
| 00:32.19 | carrar | i can dial pretty fast |
| 00:32.27 | carrar | I can even use both hands to dial |
| 00:32.28 | kaushal | QbY: http://pastebin.ubuntu.com/735809/ |
| 00:32.42 | QbY | carrar: well, we take slower dialers too |
| 00:32.44 | QbY | ;) |
| 00:33.05 | p3nguin | People give me funny looks when I use two hands to dial on my big ol' desk phone. |
| 00:33.14 | kaushal | can someone please guide me about |
| 00:33.22 | carrar | turn left |
| 00:33.28 | kaushal | http://pastebin.ubuntu.com/735809/ |
| 00:33.57 | QbY | kaushal: so what you wanna do? |
| 00:34.09 | kaushal | QbY: it does not dial out |
| 00:34.16 | SeRi | carrar, LOL |
| 00:34.22 | SeRi | hola p3nguin |
| 00:35.01 | QbY | kaushal: well.. is this the entire program? 17 lines? |
| 00:35.05 | kaushal | yes |
| 00:35.27 | QbY | well, i don't know python, but somehow you'd need to write the data into a .call file. |
| 00:35.38 | QbY | then MOVE it mv to /var/spool/asterisk/outgoing |
| 00:35.46 | kaushal | yeah i have done it |
| 00:35.55 | QbY | asterisk isn't picking up the file? |
| 00:36.13 | kaushal | but via pycall it does not work |
| 00:36.23 | QbY | cries. |
| 00:36.25 | kaushal | manually it defintely works |
| 00:36.39 | kaushal | QbY: nevermind |
| 00:36.51 | QbY | if asterisk doesn't pick it up, then the file would have to be remaining in the spool. |
| 00:36.52 | QbY | example in |
| 00:36.56 | QbY | paste one of those |
| 00:37.17 | kaushal | QbY: is there a way to dial out 300 numbers using call files ? |
| 00:37.24 | QbY | yes. |
| 00:37.28 | QbY | however, i would caution you on that |
| 00:37.31 | p3nguin | seri: ¿cómo es? |
| 00:37.45 | QbY | first of all, Asteirsk is going to pick them all up at once.. messy. |
| 00:37.48 | SeRi | que? |
| 00:37.53 | SeRi | p3nguin, ^^ |
| 00:37.54 | SeRi | lol |
| 00:37.59 | p3nguin | lo siento |
| 00:38.02 | SeRi | whats going on p3nguin! |
| 00:38.05 | kaushal | QbY: i need that behaviour |
| 00:38.09 | QbY | so, timestamp them.. only a few a minute.. |
| 00:38.19 | p3nguin | I'm just watching some ghost story crap on tv. |
| 00:38.23 | kaushal | pick them all at once |
| 00:38.25 | QbY | most asterisk installations which i have seen, will crap -- opening 300 channels at once |
| 00:38.41 | kaushal | QbY: Any working example please ? |
| 00:38.43 | QbY | most providers wouldn't take 300 calls at once. most limit you to 50 CPS.. |
| 00:38.50 | SeRi | p3nguin, I see. I just put in Three Amigos (Clasic) for the kids. They are laughing there ass off. |
| 00:38.53 | p3nguin | I typically don't do 300 channels at once, but I have a lot less system resources than many. |
| 00:39.00 | kaushal | 50 CPS .... ? |
| 00:39.05 | QbY | Calls Per Second |
| 00:39.12 | p3nguin | calls ... what he said. |
| 00:39.12 | QbY | the number of calls you can initiate at once. |
| 00:39.18 | kaushal | ok |
| 00:39.36 | QbY | takes one for the team…. |
| 00:39.39 | QbY | kaushal: PM me. |
| 00:39.55 | SeRi | you go QbY! |
| 00:40.04 | QbY | … you guys better think of us when you need low cost termination.. |
| 00:40.40 | p3nguin | What's the company name? |
| 00:40.43 | SeRi | :) |
| 00:40.44 | carrar | Just sent the change time to a future time |
| 00:40.45 | SeRi | yes |
| 00:40.49 | carrar | so they all don't go at once |
| 00:40.49 | QbY | Altus Carrier Services Company |
| 00:40.50 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 00:40.53 | QbY | www.altuscarrier.com |
| 00:41.08 | p3nguin | I'm sure you've told me before, but I have a tendency to forget things I don't use regularly. |
| 00:41.15 | QbY | hahahhaa |
| 00:41.17 | QbY | so do i |
| 00:41.22 | SeRi | bookmarks www.altuscarrier.com |
| 00:41.37 | SeRi | QbY, Thanks. |
| 00:41.52 | p3nguin | Oh, I think you were just the person/company I was looking for a few days ago. |
| 00:42.13 | QbY | whoo hoo.. |
| 00:42.14 | QbY | here i am |
| 00:42.30 | p3nguin | I couldn't remember the name, but I was going to go through logs to find it. Now I don't have to. |
| 00:42.58 | QbY | something told me to come lurk today |
| 00:43.00 | SeRi | one stone two birds ;) |
| 00:43.26 | SeRi | It only took one shoot. |
| 00:43.28 | SeRi | nice. |
| 00:43.50 | QbY | https://plus.google.com/106532311325970234139/about |
| 00:43.54 | QbY | ^^ Circle us |
| 00:43.56 | p3nguin | I think the last time you were on, the site wasn't really ready for production yet. |
| 00:44.40 | SeRi | pizza is in |
| 00:45.02 | carrar | What, altus isn't on facebook? |
| 00:45.05 | carrar | or twitters? |
| 00:45.06 | QbY | p3nguin: Ah.. For subscriber self sign up.. Nope, that goes live this weekend.. |
| 00:45.14 | carrar | How can you stay in biz!! |
| 00:45.21 | QbY | http://facebook.com/AltusCarrier http://twitter.com/AltusCarrier |
| 00:45.24 | carrar | hahah |
| 00:45.45 | QbY | asterisk won't touch it until the time is reached |
| 00:45.47 | QbY | wrong win |
| 00:45.48 | carrar | The page you requested was not found. |
| 00:45.50 | p3nguin | That's why I was looking for the name a few days ago... I was wanting to see if that stuff was there yet. Now I can watch for it. |
| 00:45.59 | carrar | oh woops |
| 00:46.21 | QbY | p3nguin: PM Me |
| 00:46.38 | QbY | everything you *need* we deliver.. FTP pulls of CDRs or SCP Pushes |
| 00:46.44 | QbY | we have APIs for everything else. |
| 00:46.49 | carrar | companies that have all those social pages need to get some real work to do |
| 00:47.09 | QbY | hahahaha.a |
| 00:47.12 | QbY | we like social |
| 00:47.13 | QbY | fun |
| 00:49.19 | p3nguin | I did it. |
| 00:49.36 | carrar | paste tense? |
| 00:49.49 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:51.08 | p3nguin | Yes. I PMed him. Past tense. |
| 00:56.07 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 00:59.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:59.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:06.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:06.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:12.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:12.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:15.01 | ruied_ | what is the best way to send fax from email? ictfax ? |
| 01:21.51 | QbY | efax.com |
| 01:21.59 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 01:25.47 | sawgood | Is there a way to determine the DEVICE_STATE of a peer from the CLI? |
| 01:27.16 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:35.02 | ruied_ | sawgood, core show channels ??? |
| 01:39.04 | ruied_ | with that you can see if it's ringing / up, or you can use hints in the dialpan and than 'core show hint <extension>' |
| 01:40.48 | *** join/#asterisk [1]SnakeDoctor (~mtaylor1@206-188-70-99.cpe.distributel.net) |
| 01:48.30 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 01:49.14 | dijib | SeRi, r0m|u you around? |
| 01:49.30 | dijib | and p3nguin is the Voip users confrence going on? |
| 01:49.36 | dijib | or anybody for that matter? |
| 01:49.44 | dijib | rest assured im not drinking tonight |
| 01:49.53 | p3nguin | It starts at 12 EST and usually lasts 1-3 hours. |
| 01:49.59 | dijib | oh thats it? |
| 01:50.00 | p3nguin | So no, it's not going on right now. |
| 01:50.12 | dijib | so it closes when the last man leaves? |
| 01:50.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 01:50.45 | p3nguin | There's a silence timeout on the conference bridge. When it has been silent for I think five minutes, it closes. |
| 01:50.53 | dijib | crappy then |
| 01:51.01 | dijib | 2 bad.. im having a good day |
| 01:51.16 | *** join/#asterisk aglenday (~aglenday@59.167.161.74) |
| 01:51.30 | dijib | this chan is dead eh |
| 01:51.31 | p3nguin | But if a bunch of people want to talk, there's enough of us in here to set up a conference mesh to support a bunch of people. |
| 01:51.41 | dijib | this is true |
| 01:51.51 | dijib | anybody down to conf? |
| 01:52.03 | dijib | anybody other than p3 |
| 02:09.46 | SeRi | dijib, I am in waz up |
| 02:09.50 | SeRi | you at the chan? |
| 02:09.56 | dijib | yessum i am |
| 02:10.01 | dijib | p3nguin, you down? |
| 02:20.53 | *** join/#asterisk mintos (~mvaliyav@114.143.165.8) |
| 02:24.33 | sawgood | ruied: ty! |
| 02:30.47 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 02:30.52 | justdave | so I see the userland dahdi stuff is in EPEL6.... do they have the kernel module somewhere, too? |
| 02:31.49 | SeRi | p3nguin, you in? |
| 02:41.48 | SeRi | http://pastebin.com/cgUVkkuj |
| 02:47.14 | p3nguin | I see you didn't take the section of dial plan that I spent my time on. |
| 02:47.23 | p3nguin | the one with gsm and disa. |
| 02:48.41 | [1]SnakeDoctor | quit |
| 02:50.49 | p3nguin | I suppose I have to write it again. |
| 02:53.25 | SeRi | p3nguin, Yes is there not on that dial plan though |
| 02:53.35 | p3nguin | What? |
| 02:53.52 | SeRi | what I posted thats abckup of my original dialplan. |
| 02:54.31 | p3nguin | What was the purpose of pasting it? |
| 02:54.41 | SeRi | jump in I all ways keep backup for every modyfication |
| 02:54.51 | SeRi | o for dijib |
| 02:54.57 | SeRi | dijib wanted to see it |
| 02:55.21 | p3nguin | Most people want to see current working dial plan rather than old backup stuff that could be broken. |
| 02:56.28 | SeRi | he does not need it to work. He just wanted to see it. |
| 02:56.33 | p3nguin | haha |
| 02:56.35 | SeRi | :/ |
| 02:56.37 | p3nguin | Doesn't need it to work. |
| 02:56.38 | p3nguin | That's great! |
| 02:56.47 | SeRi | call in |
| 02:56.54 | p3nguin | Maybe later. |
| 02:56.59 | p3nguin | Maybe sooner. |
| 02:57.37 | SeRi | :) |
| 02:58.01 | SeRi | we talking about dial plans. Your expertise is most need it. |
| 02:58.16 | SeRi | I told him about how you fix everything pretty much |
| 02:59.36 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:00.03 | SeRi | p3nguin, is that you with the rave? |
| 03:00.09 | SeRi | trance* |
| 03:00.43 | p3nguin | Maybe. That's why I recommend having callers start out muted. |
| 03:00.57 | SeRi | ROFL!!!!!!!!!!!! |
| 03:00.58 | p3nguin | So you'll have to mute me or kick me. |
| 03:01.02 | SeRi | hahaahha |
| 03:01.21 | p3nguin | 'Cause I'm not even on there. I just connected my conf to your conf. |
| 03:02.19 | SeRi | Hey fine with me I like the music |
| 03:02.30 | SeRi | lol |
| 03:02.35 | SeRi | he is trying hard to kick you! |
| 03:02.38 | SeRi | LOL |
| 03:02.55 | p3nguin | You'd be better off muting me. |
| 03:03.16 | SeRi | he cant. he is trying |
| 03:03.21 | p3nguin | At least I won't hammer your bandwidth trying to reconnect if you just mute me and leave me. |
| 03:03.39 | p3nguin | He needs to call his admin conf number instead of the regular conf number. |
| 03:04.02 | SeRi | he did and is not working |
| 03:04.11 | SeRi | you p2wn3d hes conf |
| 03:04.20 | SeRi | own3d |
| 03:04.22 | SeRi | lol |
| 03:04.50 | SeRi | he needs your help. I think |
| 03:04.55 | p3nguin | heh |
| 03:05.02 | SeRi | lol |
| 03:05.39 | nobodyshome | i like the tunes |
| 03:05.45 | nobodyshome | better then my moh |
| 03:05.50 | nobodyshome | you can talk over it |
| 03:05.51 | SeRi | 2 |
| 03:05.55 | SeRi | +1 |
| 03:08.12 | SeRi | dijib, mute it |
| 03:09.00 | dijib | mute his music |
| 03:09.10 | dijib | my admin extension isnt working |
| 03:09.44 | SeRi | p3nguin, you disconnected? |
| 03:14.19 | SeRi | p3nguin, you in? |
| 03:15.48 | SeRi | p3nguin, do you still have the pastebins from yesterday? |
| 03:17.21 | p3nguin | Which one? |
| 03:17.51 | SeRi | all of the ones you did for me. |
| 03:18.03 | SeRi | I have the ones from your examples. I lost the ones you did for me. |
| 03:18.34 | p3nguin | I think they probably had expiration. |
| 03:18.35 | SeRi | I am sort of adding some of the stuff to secure it a bit more. |
| 03:18.43 | p3nguin | I can write it up again. |
| 03:19.26 | SeRi | I am more after the disa if you can. I am going to play with it again. to see if it works as "is suppose too" tech support email me thismorning |
| 03:22.46 | p3nguin | There's your new gsm inbound context. |
| 03:23.18 | dijib | http://pastebin.com/YTmwzHw1 eat this. |
| 03:23.57 | p3nguin | line 20 is still wrong. |
| 03:24.33 | p3nguin | options *m* AND *r* |
| 03:24.35 | p3nguin | together |
| 03:24.37 | p3nguin | WRONG |
| 03:24.49 | SeRi | Thanks p3nguin! |
| 03:25.02 | p3nguin | Remove the r. If you want ringing, remove the m also. |
| 03:25.24 | p3nguin | If you want music instead of ringing, leave the m. |
| 03:25.47 | dijib | i think think this is reminence of a test |
| 03:25.50 | dijib | that failed |
| 03:26.22 | p3nguin | Take out r. |
| 03:26.38 | p3nguin | With the m, you'll have music. If you want ringing, take out m also. |
| 03:27.05 | p3nguin | Same on line 37. |
| 03:27.10 | p3nguin | and 43. |
| 03:27.22 | p3nguin | and 56. |
| 03:27.22 | p3nguin | and 62. |
| 03:27.27 | p3nguin | and 68. |
| 03:27.30 | p3nguin | and 74. |
| 03:27.34 | p3nguin | and 80. |
| 03:27.39 | p3nguin | and 86. |
| 03:27.45 | p3nguin | and 92. |
| 03:27.53 | p3nguin | and 98. |
| 03:28.05 | p3nguin | And I'm tired of listing them all. |
| 03:28.16 | p3nguin | Just fix them all. |
| 03:28.23 | SeRi | lol damn! |
| 03:32.29 | SeRi | p3nguin, jump in! |
| 03:32.31 | SeRi | lol |
| 03:32.33 | SeRi | :) |
| 03:36.19 | dijib | http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge |
| 03:40.06 | dijib | p3nguin, are you accusing me of copy pasta? |
| 03:40.07 | SeRi | p3nguin, is there a way to tunr announcement on in a conf? |
| 03:40.10 | p3nguin | I see they say to use # to access the menu. |
| 03:40.22 | p3nguin | But core show application ConfBridge incorrectly says * |
| 03:40.23 | dijib | i do? |
| 03:40.41 | dijib | DTMF isnt being picked up |
| 03:40.45 | p3nguin | Announcement on? Don't use option q. |
| 03:41.31 | p3nguin | I guess by announcement, you mean enter sound. |
| 03:42.24 | dijib | are you still listening? |
| 03:42.35 | dijib | im not using the q option. |
| 03:43.22 | p3nguin | Maybe it's just broken. |
| 03:43.50 | p3nguin | Or maybe you have to set the sound? |
| 03:44.29 | p3nguin | I don't know what the CONFBRIDGE_JOIN_SOUND actually controls. |
| 03:44.37 | SeRi | p3nguin, I mean like when somebody joins to announce in the chann that somebody join in. |
| 03:44.47 | p3nguin | I would have thought there would be a default sound. |
| 03:44.59 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 03:46.22 | *** join/#asterisk master_of_master (~master_of@p57B52439.dip.t-dialin.net) |
| 03:47.31 | p3nguin | Try putting this right before you run ConfBridge()... Set(CONFBRIDGE_JOIN_SOUND=confbridge-join) |
| 03:47.38 | p3nguin | I don't know if that will do it or not. |
| 03:48.16 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 03:56.17 | dijib | hell - - o |
| 03:57.41 | p3nguin | Do you hear the sound? |
| 03:57.51 | SeRi | rofl |
| 03:57.52 | p3nguin | That's the Set(CONFBRIDGE_JOIN_SOUND=confbridge-join) thing |
| 03:57.53 | SeRi | yes |
| 03:57.59 | p3nguin | the confbridge-join sound |
| 03:58.05 | p3nguin | But... |
| 03:58.15 | p3nguin | I don't know if it plays to the callER or into the conf. |
| 03:58.27 | p3nguin | I just started using ConfBridge recently. |
| 03:58.36 | SeRi | I can hear a soft beep in |
| 03:58.47 | p3nguin | Always used MeetMe until you started having problems and I suggested ConfBridge. |
| 03:58.57 | dijib | http://pastebin.com/wL2eNAaC |
| 04:00.30 | p3nguin | If you've added the Set(), call in again or redirect one of the active channels, and see if it plays the sound. |
| 04:00.57 | p3nguin | You can redirect your current channel back to 2663,1 and it will be like you dialed in again. |
| 04:01.09 | p3nguin | channel redirect ... |
| 04:02.25 | SeRi | bad ass! |
| 04:04.37 | dijib | heh |
| 04:04.40 | dijib | did i cut you off too? |
| 04:05.39 | dijib | channel redirect SIP/asterisk.serveirc.com-0000002e SIP/2663,1 |
| 04:05.50 | p3nguin | no... |
| 04:06.22 | p3nguin | channel redirect SIP/the-phone-you-are-using conferences,2663,1 |
| 04:06.33 | p3nguin | channel redirect SIP/the-phone-you-are-using conference,2663,1 |
| 04:06.43 | p3nguin | I don't know if you are using conference or conferences |
| 04:08.10 | dijib | I> channel redirect SIP/asterisk.serveirc.com-0000002e SIP/2663,1 |
| 04:08.21 | dijib | channel redirect SIP/asterisk.serveirc.com-0000002e SIP/conference/2663,1 |
| 04:08.26 | dijib | channel redirect SIP/asterisk.serveirc.com-0000002e conference/2663,1 |
| 04:08.37 | p3nguin | nope |
| 04:08.40 | p3nguin | (2206.33) <p3nguin> channel redirect SIP/the-phone-you-are-using conference,2663,1 |
| 04:08.43 | p3nguin | ^ |
| 04:09.14 | *** join/#asterisk wepy (~wepy@ip72-192-221-88.dc.dc.cox.net) |
| 04:09.20 | wepy | hi |
| 04:09.44 | SeRi | hola |
| 04:10.12 | wepy | are there any good sites that talk about setting up asterisk |
| 04:10.29 | wepy | a lot of the google results i get are for sites that are cluttered with many many ads... |
| 04:10.37 | dijib | how do i book? |
| 04:10.40 | dijib | ~book |
| 04:10.40 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 04:10.49 | p3nguin | channel redirect <TAB> |
| 04:11.01 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 04:11.16 | wepy | thanks |
| 04:11.16 | p3nguin | If you press the tab key, it will show you all currently open channels. |
| 04:11.25 | p3nguin | Choose yours. |
| 04:12.15 | dijib | channel redirect SIP/anonymous.invalid-0000002c conference/2663,1 |
| 04:12.23 | wepy | also, i've read that SIP supports TLS now (sometimes called SIPS?). Are there any PTSN services that actually support this? |
| 04:12.31 | p3nguin | That would have been mine, I'd guess. |
| 04:12.32 | wepy | i guess it's termination services |
| 04:12.45 | p3nguin | But you sent it to an invalid location. |
| 04:12.47 | dijib | channel redirect SIP/anonymous.invalid-0000002c conference,2663,1' |
| 04:15.49 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 04:20.52 | SeRi | ~awk |
| 04:20.53 | infobot | [awk] the unix hacker's first scripting language |
| 04:20.58 | SeRi | ~sed |
| 04:20.58 | infobot | somebody said sed was the GNU Stream Editor. URL: ftp://ftp.gnu.org/pub/gnu/ MANPAGE:http://www.gnu.org/software/sed/manual/ |
| 04:21.13 | SeRi | Thank you boot |
| 04:21.16 | SeRi | bot* |
| 04:29.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:36.13 | ChannelZ | OMG Comcast's music on hold is going to make me kill myself |
| 04:36.52 | dijib | vyatta.org |
| 04:39.00 | SeRi | ChannelZ, comcast? |
| 04:40.15 | SeRi | astlinux.org |
| 04:44.46 | *** join/#asterisk radic (~radic@dslb-178-002-229-221.pools.arcor-ip.net) |
| 04:46.32 | ChannelZ | cable/internet company |
| 04:47.10 | ChannelZ | Their MOH was one song, only like 20 seconds long. Over and over and over and over |
| 04:51.30 | *** join/#asterisk timahvo1 (~rogue@197.176.107.196) |
| 04:52.32 | ChannelZ | brb |
| 04:55.44 | p3nguin | dijib: You should also set CONFBRIDGE_LEAVE_SOUND to confbridge-leave |
| 04:57.35 | *** join/#asterisk jkroon (~jkroon@dsl-241-251-222.telkomadsl.co.za) |
| 04:58.56 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 05:02.14 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 05:08.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:15.00 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 05:15.02 | dijib | operatorchan.org |
| 05:15.42 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 05:15.47 | dijib | http://operatorchan.org/k/src/k313531_50bmg%20deagle.jpg |
| 05:17.27 | Kobaz | anyone have a download link for the 4.0.0 polycom firmware |
| 05:18.03 | *** join/#asterisk mintos (~mvaliyav@114.143.165.8) |
| 05:26.52 | dijib | https://plus.google.com/115135339292268300308/posts/MCpnZpV1SLD |
| 05:30.19 | dijib | http://cosketch.com/Rooms/gvxmjky |
| 05:35.51 | p3nguin | dijib: PoP = Point of Presence |
| 05:40.57 | SeRi | ~pop |
| 05:40.57 | infobot | somebody said pop was (Point Of Presence) This is a local telephone number through which you can access your ISP. The largest national ISPs have POPs all over the country. |
| 05:46.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 06:00.04 | p3nguin | (2255.44) <p3nguin> dijib: You should also set CONFBRIDGE_LEAVE_SOUND to confbridge-leave |
| 06:00.08 | dijib | same => n,Set(CONFBRIDGE_JOIN_SOUND=confbridge-join); |
| 06:00.09 | dijib | same => n,Set(CONFBRIDGE_LEAVE_SOUND=confbridge-leave); |
| 06:04.12 | p3nguin | I have no idea where you sent me. |
| 06:04.19 | dijib | monkeys |
| 06:04.23 | dijib | sorry |
| 06:04.31 | p3nguin | I'm gone. |
| 06:04.36 | dijib | i need to know anonymous.invalid |
| 06:04.39 | SeRi | dial back in |
| 06:04.40 | p3nguin | You must have had a typo. |
| 06:04.49 | SeRi | lol |
| 06:05.25 | *** part/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 06:06.14 | *** join/#asterisk pietro (~pietro@88-149-224-186.dynamic.ngi.it) |
| 06:07.19 | dijib | n,ConfBridge(${EXTEN},cMs |
| 06:07.27 | dijib | ) |
| 06:17.38 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 06:34.19 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:34.26 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 07:01.30 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:05.04 | nobodyshome | so how do you play that? |
| 07:07.24 | p3nguin | nobodyshome: channel originate Local/2663@conference application Playback silence/2&vm-goodbye |
| 07:10.17 | p3nguin | nobodyshome: channel originate Local/2663@conference application Playback silence/10&tt-allbusy |
| 07:12.00 | p3nguin | nobodyshome: channel originate Local/2663@conference application Congestion 10 |
| 07:34.56 | nobodyshome | http://cosketch.com/Rooms/gvxmjky |
| 07:49.16 | *** join/#asterisk Pio (~pio@reyes.longstair.com) |
| 07:56.58 | SeRi | sip originate jelapanos |
| 07:58.16 | p3nguin | channel originate tortillas/jalapenos application MixMonitor somefilename |
| 07:59.25 | nobodyshome | <PROTECTED> |
| 08:06.14 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 08:08.55 | p3nguin | seri: http://www.backupschedule.net/backupschedules/towerofhanoi.html |
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| 08:39.26 | p3nguin | |
| 08:39.28 | p3nguin | |
| 08:54.02 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
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| 08:56.18 | *** part/#asterisk Ast001 (~uros@cable-89-216-180-158.dynamic.sbb.rs) |
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| 11:18.12 | Madkiss | Hi there! |
| 11:19.23 | Madkiss | I have just developed an OCF resource agent for asterisk. It's available on https://github.com/fghaas/resource-agents/commits/asterisk and I'm looking for more comments on it :) |
| 11:24.26 | irroot | Madkiss you could put it on the asterisk reviewboard ?? |
| 11:26.14 | Madkiss | I was not sure that's the appropriate place for it because that resource agent is supposed to go into the resource-agents package, not into the asterisk source itself |
| 11:31.27 | *** join/#asterisk brezular (~brezular@adsl-dyn60.78-98-49.t-com.sk) |
| 11:40.47 | irroot | Madkiss ah you have a point |
| 12:10.45 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.224.253.dsl.dyn.forthnet.gr) |
| 12:36.15 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:39.42 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 12:39.58 | [sr] | update topic to 10.0.0-rc1 |
| 12:39.59 | [sr] | :p |
| 12:43.32 | irroot | sr lol |
| 12:43.38 | irroot | indeed |
| 12:44.38 | [sr] | ) |
| 12:44.40 | [sr] | :) |
| 12:44.42 | *** join/#asterisk francisvgarcia (~networker@190.122.115.211) |
| 12:45.15 | francisvgarcia | Hi guys |
| 12:45.51 | [sr] | ai |
| 12:46.04 | francisvgarcia | out of the cisco ip phones |
| 12:46.25 | francisvgarcia | which phones has the best sound quality, I mean speaker and handset |
| 12:47.08 | francisvgarcia | I have only worked with grandstream, which has a nice quality but not even compared with the cisco |
| 12:47.29 | irroot | francisvgarcia grandstream is shocking quality |
| 12:47.59 | heffer | what about snom? anyone used that brand before? |
| 12:49.35 | WIMPy | heffer: Like them the mos so far. |
| 12:49.52 | WIMPy | Which does not mean I really like them. |
| 12:50.03 | irroot | snom is really great the new v7/8 is best so far problems with <=6 |
| 12:50.09 | francisvgarcia | there are many brands I have not used before like Aastra, Snom, Polycom. But I dont know how good the sound quality are |
| 12:50.25 | irroot | they autoconfig easiest of all models i have used |
| 12:50.27 | heffer | from what i have seen on the web i think they seem quite nice. they are IPv6 capable and the design is quite nice |
| 12:50.59 | irroot | polycom has the best sound quality i find but did not like there licence they introduced |
| 12:51.26 | irroot | heffer snom is a euro centric design and runs linux |
| 12:51.44 | irroot | polycom / cisco are more a us centric design |
| 12:51.46 | heffer | from what i understood cisco devices are supposed to be used with other cisco telephony products. but as my presence here might suggest i prefer open source :D |
| 12:52.05 | irroot | yealink has is also available with HD audio |
| 12:52.06 | francisvgarcia | I got a Linksys SPA941, but the sound sucks |
| 12:52.27 | irroot | heffer CISCO phones are a BITCH to work with |
| 12:52.29 | heffer | irroot: i use Cisco phones(as a user) at work but i think i will recommend snom to my fathers business |
| 12:52.37 | *** join/#asterisk timahvo1 (~rogue@41.90.91.181) |
| 12:53.04 | heffer | at home i use a Siemens phone, which also is a bitch to work with :D |
| 12:53.04 | [sr] | howdy WIMPy |
| 12:53.36 | irroot | francisvgarcia heffer the SPA range are the Sipura phones Linksys bought out now Cisco too great phones and ATA's i recomend the ATA's not much experiance on the handsets |
| 12:53.37 | francisvgarcia | someone got me a yealink phone one time, but speaker sound was sucking too |
| 12:55.20 | irroot | i support Yealink/Linksys/Snom/Polycom for provisioning and management from GUI |
| 12:55.22 | francisvgarcia | this is because, at least in my country, people prefer voice quality over features. And specially business managers prefer use the speaker that the handset |
| 12:56.05 | irroot | francisvgarcia polycom will be best option from my expereriance |
| 12:56.56 | francisvgarcia | and what about web auto provisioning? are they grandstream like? |
| 12:58.48 | irroot | francisvgarcia it is auto mated via a config on server the grandstream does not auto provision its manually provisioned via access to device |
| 12:59.23 | irroot | the polycom admin interface is shocking in this regard but the net provisioning is great |
| 13:02.10 | francisvgarcia | irroot: at least in the new GS, the GXP1450, you can define by DHCP a web url where the phone is going to download the config file from. |
| 13:03.39 | irroot | francisvgarcia its about time they woke up i started of with GS and ditched them ASAP when phones that could be provisioned came out |
| 13:04.41 | puzzled | hi |
| 13:04.53 | irroot | puzzled o/ |
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| 14:31.52 | tm1000 | irroot: Is your provisioner freely available? |
| 14:33.08 | irroot | tm1000 it is however its not so plug and play have some custom hacks we ship it as a distro on flash |
| 14:33.30 | irroot | you can look at the scripts and bits its in PHP |
| 14:33.52 | irroot | all working off 1.8/10 asterisk realtime |
| 14:34.11 | irroot | the db schema is bit non standard though |
| 14:34.42 | irroot | im working on making it standard |
| 14:35.43 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 14:38.38 | tm1000 | irroot: Website? Or place to look at it? |
| 14:38.54 | irroot | pbx.distrotech.co.za/svn |
| 14:39.02 | irroot | all me bits are there tm1000 |
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| 15:38.23 | mrtall | hey everybody... |
| 15:39.37 | mrtall | i am new to asterisk, i installed it and it connects to my sipgate account. i can call any normal numbers but i can not get called. |
| 15:40.06 | mrtall | can somebody help me with this topic? |
| 15:40.26 | WIMPy | What happens? |
| 15:42.01 | mrtall | i get "person you have called is temporarly not available" |
| 15:42.14 | mrtall | do i have to open ports? |
| 15:43.05 | WIMPy | At least 5060 udp |
| 15:43.10 | mrtall | ok thx |
| 15:43.15 | mrtall | i'll try |
| 15:45.50 | mrtall | ahh |
| 15:45.54 | mrtall | [Nov 12 16:45:19] NOTICE[14188]: chan_sip.c:20785 handle_request_invite: Call from '1294235' to extension '1294235' rejected because extension not found in context 'default'. |
| 15:46.24 | mrtall | i think i'm getting closer? |
| 15:46.43 | carrar | You are |
| 15:47.02 | WIMPy | Yes, just create that extension. |
| 15:49.14 | mrtall | well... i dont really understand what to do |
| 15:49.36 | WIMPy | extensions go to extensions.conf. |
| 15:50.12 | WIMPy | For general stuff try the |
| 15:50.17 | WIMPy | ~book |
| 15:50.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 15:50.48 | WIMPy | or http://das-asterisk-buch.de/ But that's no as up to date |
| 15:51.14 | mrtall | i work with das-asterisk-buch... |
| 15:51.19 | mrtall | ahh well |
| 15:51.32 | mrtall | maybe i get it now... |
| 15:52.47 | *** join/#asterisk kaushal (~kaushal@14.99.138.131) |
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| 15:54.36 | mrtall | it works... :) thank you! but there is no sound... in each direction |
| 15:54.57 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:55.36 | WIMPy | That's probably a firewall or NAT issue. How / from where do you try to connect? |
| 15:55.54 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 15:56.47 | mrtall | i have a local server in my private house |
| 15:56.51 | mrtall | with centos |
| 15:56.54 | mrtall | and shorewall |
| 15:57.17 | mrtall | the hello world thing worked pretty well |
| 15:57.24 | WIMPy | Is your Asterisk behind te firewall? |
| 15:57.32 | mrtall | yes |
| 15:57.43 | WIMPy | Then see |
| 15:57.48 | WIMPy | ~sipnat |
| 15:57.48 | 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, canreinvite, externhost or externip, and localnet. |
| 15:58.37 | mrtall | thx! gonna work it through :) |
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| 17:25.19 | mrtall | WIMPy do you speak german? |
| 17:26.53 | WIMPy | Yes. And tehre's an #asterisk-de as well. |
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| 18:22.09 | ruied | is there a wy so I can use BLF to check if a phone is not registered (having the light off) ? |
| 18:23.27 | WIMPy | The device state will tell you if it's reachable. |
| 18:23.54 | WIMPy | Might work better if you have qualify on. |
| 18:29.26 | ruied | WIMPy, I would like to check if the phone is registered by the BLF at phone's extension module, for instance, having the light of that extension off if phone not registered |
| 18:30.31 | WIMPy | I'm not sure what happens if registration times out. But you definitely get the stat if you have qualify enabled. |
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| 18:35.14 | ruied | ok, I'm looking at my sip.conf, I have that in some peers, going to add in [general] and make some tests |
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| 18:45.56 | *** join/#asterisk rollo-bolo (~rawle@99.110.90.181) |
| 18:52.16 | *** join/#asterisk rollo-bolo (~rawle@99.110.90.181) |
| 18:54.24 | rollo-bolo | running asterisk 1.8.7.1 and needing fax for asterisk. I am getting undefined symbol: ast_fax_tech_unregister. is there a fix for this? |
| 18:55.34 | Kobaz | anyone have a download link for the 4.0.0 polycom firmware? |
| 18:56.08 | Kobaz | rollo-bolo: the fix would be to compile which ever module provides ast_fax_tech_unregister |
| 18:57.12 | p3nguin | Have you installed res_fax during build/install of Asterisk? Did you then get the free fax for asterisk module, install it, and register it? |
| 18:57.29 | p3nguin | The fax for asterisk module is res_fax_digium. |
| 18:57.38 | rollo-bolo | yes i resistered |
| 18:57.56 | rollo-bolo | running 1.8 so don't need res_fax just res_fax_digium |
| 18:58.15 | p3nguin | false |
| 18:58.34 | p3nguin | You need res_fax to give you the fax applications SendFAX() and ReceiveFAX(). |
| 18:58.42 | p3nguin | And then res_fax_digium to make it go. |
| 18:58.53 | rollo-bolo | i think this is a bug because another use inthe digium forum has the same issue running the latest asterisk |
| 18:59.09 | rollo-bolo | hmm |
| 18:59.11 | ChannelZ | You don't have any old modules lying around do you? |
| 18:59.14 | ChannelZ | It kind of sounds familiar |
| 18:59.20 | p3nguin | They're doing it wrong. I use 1.8.7.1 with res_fax and res_fax_digium with no problem. |
| 18:59.24 | ChannelZ | Either that or a version mismatch |
| 18:59.41 | p3nguin | It's very likely a version mismatch. |
| 18:59.42 | rollo-bolo | well the selector does not have res_fax. just rex_fax_digium |
| 19:00.00 | p3nguin | I said you have to install res_fax when you install asterisk. |
| 19:00.13 | rollo-bolo | o so it does work p3nguin? ok good |
| 19:00.17 | p3nguin | It's enabled or disabled in make menuselect. |
| 19:00.39 | p3nguin | Alternative to fax for asterisk, you can use app_fax and spandsp. |
| 19:00.57 | rollo-bolo | i think it had XXX in the makemeun. i'll double check |
| 19:01.19 | p3nguin | Yes, app_fax will be XXX until you satisfy the requirements of it. |
| 19:01.29 | rollo-bolo | aaaaahhhh |
| 19:01.53 | p3nguin | I think you have to have spandsp before you can enable app_fax, but I'm not 100% on that... since I use fax for asterisk (res_fax and res_fax_digium). |
| 19:03.49 | rollo-bolo | so again just compile in asterisk source (once i meet the requirements) |
| 19:04.32 | ChannelZ | if you go the spandsp/app fax route |
| 19:04.35 | p3nguin | There's probably a wiki page talking about how to use spandsp and app_fax. |
| 19:04.54 | rollo-bolo | rather go the other route |
| 19:04.56 | p3nguin | I'm satisfied with the fax for asterisk solution. It works well for me. |
| 19:05.23 | p3nguin | Just make sure you have the right module. The module selector should give you the right one. |
| 19:05.27 | ChannelZ | in that case you need to check that res_fax is built and then DL the correct version of res_fax_digium |
| 19:06.04 | ChannelZ | 1.8.4 in your case |
| 19:06.10 | rollo-bolo | well looks like i need the correct ver of res_fax. at site only 1.4 and 1.6 was there |
| 19:06.22 | ChannelZ | http://downloads.digium.com/pub/telephony/fax/res_fax_digium/ |
| 19:06.35 | p3nguin | Again, res_fax is part of your asterisk source, and it is enabled/disabled in make menuselect. |
| 19:06.43 | p3nguin | It's not something you go get from some place. |
| 19:06.53 | rollo-bolo | k |
| 19:07.36 | rollo-bolo | b back then |
| 19:07.44 | p3nguin | You can find out if you already have it. find /usr/lib/asterisk/modules -name \*fax\* |
| 19:08.01 | p3nguin | I do it, and I see: |
| 19:08.02 | p3nguin | /usr/lib/asterisk/modules/res_fax_digium.so |
| 19:08.02 | p3nguin | /usr/lib/asterisk/modules/res_fax.so |
| 19:08.07 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 19:08.19 | rollo-bolo | res_fax_digium.so ---comes up |
| 19:08.33 | ruied | rollo-bolo, module show like res_fax.so |
| 19:08.34 | rollo-bolo | i know res_fax is missing. i now know i need it |
| 19:08.46 | ChannelZ | check your build |
| 19:08.46 | p3nguin | So you installed the ffa module from digium, but you don't have res_fax enabled in menuselect. |
| 19:09.03 | rollo-bolo | correct |
| 19:09.08 | p3nguin | If you haven't deleted your source tree, it will be an easy fix. |
| 19:09.16 | rollo-bolo | ok |
| 19:09.26 | p3nguin | Just go back to the source directory, make menuselect, enable that one, save/exit, make. |
| 19:09.35 | p3nguin | It should only build the new module. |
| 19:09.46 | rollo-bolo | it had XXX the last i looked |
| 19:09.47 | p3nguin | Then you can either make install again, or you can copy the module manually. |
| 19:09.53 | p3nguin | res_fax? |
| 19:09.57 | rollo-bolo | yes |
| 19:10.02 | p3nguin | app_fax is usually the one that is marked out. |
| 19:10.10 | rollo-bolo | ooooooo |
| 19:10.35 | p3nguin | Look in the resource modules section, not the applications section. |
| 19:12.11 | rollo-bolo | ok recompiling |
| 19:12.32 | rollo-bolo | yes it's *. but i think i deleted it originally |
| 19:12.36 | rollo-bolo | standby |
| 19:13.41 | rollo-bolo | sweeeeeeetttttt |
| 19:13.47 | rollo-bolo | works |
| 19:13.50 | p3nguin | Nice. |
| 19:13.53 | rollo-bolo | i know what i did |
| 19:14.32 | rollo-bolo | i deleted the module res_fax after having the wrong red_fax and res_fax_digium |
| 19:14.56 | ChannelZ | I wondered, since it defaults to being built and has no dependencies that I know of |
| 19:14.57 | rollo-bolo | thx p3nguin |
| 19:16.02 | p3nguin | After adding res_fax and restarting asterisk, "fax show capabilities" shows the right information? |
| 19:21.32 | *** join/#asterisk dlublink (~david@206-248-174-128.dsl.teksavvy.com) |
| 19:21.48 | dlublink | Can I use a channel variable in the dynamic features.conf to choose the agi server ? |
| 19:23.20 | dym | ~book |
| 19:23.20 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 19:58.30 | *** join/#asterisk Ast001 (~uros@cable-89-216-185-211.dynamic.sbb.rs) |
| 19:59.24 | irroot | hi folks what up |
| 19:59.41 | Ast001 | hello does Asterisk 1.4.42 support [mappings] in cdr_manager.conf ? |
| 20:01.58 | irroot | nope dont look like it |
| 20:02.40 | irroot | 1.6.2 does |
| 20:04.34 | Ast001 | oh NO... I can't register with 1.6.2. and I cant use mappings in manager in 1.4.42 ... |
| 20:05.01 | irroot | Ast001 no help changing password ?? |
| 20:05.47 | Ast001 | I registered from 1 attempt with 1.4.42 with "userid@ims.blabla":password@domain |
| 20:06.04 | p3nguin | Quotes are not valid in register statements. |
| 20:06.06 | Ast001 | They dont work until monday and i am sure they wont change my password.. |
| 20:06.29 | Ast001 | so how can I register 381117152550@ims.telekomsrbija.com as auth id |
| 20:06.36 | irroot | Ast001 it can be fixed will need to see what issue is first |
| 20:07.07 | Ast001 | issue is @ is problem for 1.6.2 |
| 20:07.17 | irroot | p3nguin problem is the "@" in password |
| 20:07.23 | irroot | ive had this before got password changed |
| 20:07.37 | p3nguin | I don't see an @ in the password. |
| 20:07.47 | Ast001 | its in authid |
| 20:07.49 | irroot | Ast001 is it resolved in 1.8 ? |
| 20:08.01 | Ast001 | no I tried with 1.8.7.1 and nothing wrong password in cli |
| 20:08.24 | Ast001 | p3nguin I don't have @ in password i have @ in authid |
| 20:08.36 | p3nguin | It looks like you don't. |
| 20:08.38 | irroot | appologieses |
| 20:08.39 | Ast001 | and with 1.4 I can register with "" and authid inside |
| 20:08.54 | p3nguin | It looks like you have an authentication for a proxy. |
| 20:08.56 | Ast001 | with 1.8 it is bad from field.. |
| 20:09.26 | Ast001 | yes I have outbound proxy |
| 20:10.12 | p3nguin | It's valid in 1.8 to have user@domain:authuser:password@itsp.net |
| 20:10.38 | p3nguin | I may have the order of that mixed up... |
| 20:10.51 | p3nguin | user@proxy:password:authuser@provider |
| 20:10.53 | p3nguin | there |
| 20:11.22 | Ast001 | user@proxy :password:authuser:provider ? |
| 20:11.34 | Ast001 | I can try that |
| 20:11.35 | p3nguin | user@proxy:password:authuser@provider <---- this |
| 20:11.48 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 20:11.55 | Ast001 | ok i will try that |
| 20:12.33 | Ast001 | and write result here |
| 20:12.47 | ruied | how can I check the available variables from cli ??? like ${CALLERID(num)} |
| 20:12.57 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:13.37 | p3nguin | I personally use a string like that. 3145551212@proxy.provider.com:secret:3145551212@sip.provider.com/3145551212 |
| 20:13.42 | p3nguin | It's silly, but it works. |
| 20:13.56 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 20:14.21 | p3nguin | You can even specify a port for the proxy if necessary. |
| 20:15.01 | Ast001 | ok i am recompiling asterisk with 1.8.7.1. I will try that. Thanks |
| 20:15.12 | SeRi | wow what a day. |
| 20:15.17 | SeRi | waz up p3nguin! |
| 20:15.25 | p3nguin | You stayed up too late, didn't you?! |
| 20:15.28 | Ast001 | if it works you need to open paypal account :) so I can put some $ on it |
| 20:15.36 | p3nguin | Already have one. |
| 20:15.40 | Ast001 | ok |
| 20:16.22 | irroot | Ast001 look at 1.8.8-rc3 its got some good fixes in there esp some deadlocks |
| 20:16.23 | SeRi | p3nguin, LOL! Right when we finish the chat I hit the bed. kids woke me up right a 7AM for the park. lol |
| 20:17.00 | Ast001 | ok irroot if I manage to register I will do it |
| 20:17.27 | SeRi | just got back about 1hr a go and finish the rest of the house chores... Now getting the bbq ready for the fight tonight :) |
| 20:17.28 | p3nguin | I went off to bed within 15 minutes of your departure, and was up by 9. |
| 20:18.44 | SeRi | damn. exhausting :) lol |
| 20:18.57 | SeRi | Arch day tomorrow! |
| 20:18.59 | SeRi | w00t! |
| 20:19.04 | SeRi | lol |
| 20:24.30 | Ast001 | -- Got SIP response 503 "CSCF Server Internal Error" back from 10.0.0.2:5060 |
| 20:25.03 | ChannelZ | WHARRRFF! |
| 20:25.12 | SeRi | waz up ChannelZ |
| 20:26.10 | ruied | is there any equivalent to ${DIALEDPEERNAME} |
| 20:26.10 | Ast001 | p3nguin do you have some idea about it ? |
| 20:26.57 | p3nguin | I'm not familiar with that response. |
| 20:30.38 | ruied | .... I meant ${DIALEDPEERNUMBER} , it seems not to be working... |
| 20:30.59 | p3nguin | What is a dialed peer number? |
| 20:32.06 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 20:32.41 | ruied | p3nguin, I want to store records with the name: ${CALLERID(num)}_TO_<dialed number> |
| 20:33.13 | p3nguin | The dialed number has nothing to do with the peer you're attempting to reach. |
| 20:33.24 | p3nguin | The dialed number is probably going to be ${EXTEN} |
| 20:34.38 | ruied | p3nguin, hehe, you are right, I just have to pass that as an arg to the macro... |
| 20:39.46 | p3nguin | Is ${MACRO_EXTEN} no good? |
| 20:40.37 | p3nguin | It would be ${ARG1} inside the macro context. |
| 20:42.04 | Ast001 | this approach does not work.Does anyone know how can I pass @ in register statemant in sip.conf in situation where userid is pure number and authid is number @ domain ? |
| 20:48.07 | ruied | p3nguin, I think so. Is there a way to check the available variables from asterisk cli? |
| 20:48.39 | *** join/#asterisk Netgeeks-laptop (~chris@gw1.netgeeks.net) |
| 20:49.21 | rollo-bolo | how can i correct ERROR[22679]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/sip.broadvoice.com-00000002' in T.38 mode |
| 20:50.10 | p3nguin | Turn off t38 mode in sip.conf, since they apparently don't support it. |
| 20:50.21 | rollo-bolo | k |
| 20:50.46 | p3nguin | ruied: I don't know of any way to show every variable that you're trying to use, no. |
| 20:53.05 | ruied | ok, I was searching for something like 'core show functions' |
| 20:53.36 | p3nguin | But those are functions, not variables. |
| 20:54.37 | p3nguin | If you want to see variables that a particular application sets, you can probably find the ones significant to such application with core show application <whatever application you're wanting to use> |
| 20:59.53 | ruied | yes, but not always. I was working with MixMonitor() and having some problems finding the variables that I wanted... |
| 21:00.54 | ruied | they were not directed related to mixmonitor... |
| 21:01.03 | ruied | directly |
| 21:02.28 | p3nguin | Then you don't understand what variables do. |
| 21:05.11 | p3nguin | As far as I know, there are no variables FOR MixMonitor that it uses internally. To my knowledge, it only sets the MIXMONITOR_FILENAME based on the file name provided as app data. |
| 21:05.16 | [TK]D-Fender | [15:39]p3nguinIs ${MACRO_EXTEN} no good? <- virtually worthless |
| 21:05.50 | [TK]D-Fender | p3nguin: May as well pass it as the 1st arg. Otherwise you become dependand on where you call the macro from... and there goes the reusability |
| 21:06.51 | p3nguin | It was my understanding that knowing where the macro was run from was his intention. |
| 21:07.30 | [TK]D-Fender | Yes, and passing it as ARG1 is just as easy and you know the same answer... only you get to call it from places you want to fake, etc |
| 21:07.36 | p3nguin | He wanted to know the dialed number. |
| 21:07.52 | ruied | I do, what I was trying to get was variables from asterisk, and not from application, like ${CALLERID(num)} and so... |
| 21:08.06 | [TK]D-Fender | Maybe the exten you're on isn't it anyway... and if it is where you are.... doesn't stop you from passing it as a parm... |
| 21:08.08 | [TK]D-Fender | m,akes assumptions. |
| 21:08.16 | [TK]D-Fender | Basically... no plus, only potential minus |
| 21:08.37 | p3nguin | CALLERID(num) is not a variable. it's a function. |
| 21:08.40 | p3nguin | (like I said earlier) |
| 21:09.29 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
| 21:10.39 | ruied | p3nguin, yes, you are right, now I understand what you have said... |
| 21:11.24 | kriegerod | is that normal that asterisk refuses to transcode .gsm audio file if remote side of call wants ulaw only? |
| 21:12.43 | p3nguin | I would have thought it would transcode just fine. Always has for me. |
| 21:15.33 | *** join/#asterisk JuanCri (~JuanCri@ec2-184-73-226-120.compute-1.amazonaws.com) |
| 21:29.00 | kriegerod | damn, it does with Playback() app, but doesnt with Read() |
| 21:29.15 | kriegerod | i mean it transcodes, or not |
| 21:38.00 | *** join/#asterisk irroot (~irroot@197.111.205.64) |
| 21:45.20 | p3nguin | You're saying a ulaw call won't record in gsm? |
| 21:48.20 | ruied | I'm trying to make a record on demand by pressing *1, but I would like to make it with BLF key switching RED when it's recording, can I use in features.conf something like: 'automon => *1,1' and 'automon*1,n,Set(DEVICE_STATE(Custom:lamp1)=INUSE)' like in extensions.conf? Maybe it's just a stupid idea... |
| 21:48.41 | rollo-bolo | p3nguin -> you say fax works do you have hardware or writing to a file? |
| 21:49.01 | p3nguin | If I had hardware, I wouldn't be using fax for asterisk. |
| 21:49.15 | rollo-bolo | :) |
| 21:49.31 | rollo-bolo | i am having trouble writing to a file. |
| 21:49.37 | p3nguin | Fax call comes in, ReceiveFAX() writes a TIFF, which I convert to a PDF and email to my email address. |
| 21:50.01 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 21:50.13 | SeRi | p3nguin, would it be better to route 911 calls threw my gsm gateway or threw voip.ms? |
| 21:50.36 | rollo-bolo | im having trouble receiving. using broadvoice. i turned T.38 off |
| 21:50.49 | p3nguin | seri: That would be your decision. Which one do you feel is more reliable? |
| 21:51.08 | p3nguin | seri: If the internet is down, you can't use voipms to send a 911 call out... |
| 21:51.20 | SeRi | excellent point. |
| 21:51.24 | p3nguin | but can you send out a 911 call over gsm if the internet is down? |
| 21:51.30 | SeRi | Yes |
| 21:51.54 | SeRi | I tested the fail over and all cals go out ok if there is no internet |
| 21:51.59 | SeRi | calls* |
| 21:52.02 | p3nguin | You could use voipms as the primary and the gateway as secondary. |
| 21:52.09 | SeRi | threw the gsm gateway |
| 21:52.14 | SeRi | excellent |
| 21:54.05 | rollo-bolo | i have the receive faxed in home dir. should i mode to like /var/spool/asterisk? i'm getting permssion denied issued but i have 777 on all... |
| 21:59.28 | p3nguin | rollo-bolo: http://pastebin.com/6RQV9nEx |
| 22:00.04 | rollo-bolo | thx |
| 22:04.05 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 22:04.07 | p3nguin | That's just how I handle my faxes. |
| 22:04.52 | rollo-bolo | ok. im trying |
| 22:06.51 | *** join/#asterisk Ast001 (~uros@cable-89-216-185-211.dynamic.sbb.rs) |
| 22:07.16 | Ast001 | Can you explain me this ? The thing that saved me is the fact the last part of the register string can be a context. In that context you can define all kinds of things like fromdomain. It can give you the same registration SIP packet as in 1.4. |
| 22:07.41 | Ast001 | it is from this forum http://forums.digium.com/viewtopic.php?t=73703 and it seems some people had same problem like i have |
| 22:08.56 | Ast001 | I really need registration SIP packet as in 1.4 on 1.8. asterisk |
| 22:09.09 | p3nguin | What you just said makes no sense to me. |
| 22:09.55 | Ast001 | One customer says he put context in registration string and in context put fromdomain and he got registration packet as in 1.4 |
| 22:10.32 | p3nguin | Sounds like a ridiculously uninformed statement on his part, then. |
| 22:10.33 | WIMPy | Maybe you need to set your system name/domain? |
| 22:11.55 | Ast001 | you can read that link to see what is the problem. |
| 22:12.21 | Ast001 | set system name to what ? |
| 22:12.42 | WIMPy | Your IP if you want that to be sent. |
| 22:13.03 | WIMPy | But it's just a guess. I'm not the SIP guy. |
| 22:13.35 | WIMPy | I'm just reading about OpenHorst. Extremely interesting. |
| 22:14.03 | *** join/#asterisk timahvo1 (~rogue@41.90.91.181) |
| 22:23.31 | ChannelZ | OpenHorst? |
| 22:25.45 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:30.23 | ChannelZ | I have no idea if this works but did you change the @ in the username to %40 and try that? I thought parts of the SIP RFC state certain characters must be encoded in SIP URIs |
| 22:40.20 | SeRi | p3nguin, you in? |
| 22:40.51 | p3nguin | Sure. |
| 22:41.07 | Ast001 | <PROTECTED> |
| 22:41.13 | Ast001 | *main |
| 22:41.49 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:41.58 | SeRi | p3nguin, pb coming up. |
| 22:42.23 | p3nguin | If it doesn't work, then how am I able to authenticate to my proxy using the string I listed earlier? |
| 22:43.11 | Ast001 | it does not work i just saw in sip debug what it sends as username |
| 22:43.14 | SeRi | p3nguin, I did as instructed last night about the qualify which is now disable and for fail over I change it like this: http://pastebin.com/q26FKcMS |
| 22:43.38 | Ast001 | and it is just number without @domain |
| 22:43.41 | *** join/#asterisk tick (tick@80.54.23.253) |
| 22:44.01 | SeRi | for got SIP in line 11 :) |
| 22:44.59 | p3nguin | You can also check if a channel can be estabished before dialing each peer. |
| 22:46.46 | ChannelZ | Ast001: What is it you want the From: line to look like? |
| 22:48.42 | Ast001 | i want to be able to send registration like asterisk 1.4 can in maner "username" : secret @ domain . com (without spaces of course where username is number @ somedomain.com) |
| 22:49.15 | Ast001 | to send all of that as username not just number like 1.6 and 1.8 do |
| 22:49.33 | Ast001 | and I can not put " in 1.6 and 1.8 |
| 22:49.48 | p3nguin | If I use 3145551212@proxy.provider.com:secret:3145551212@sip.provider.com/3145551212 my From: in a REGISTER looks like From:<sip:3145551212@proxy.provider.com> |
| 22:50.15 | Ast001 | your 3145551212@proxy.provider.com is not your username it is username@proxy |
| 22:51.05 | p3nguin | My user ID is only 3145551212. |
| 22:51.21 | Ast001 | yes and mine is number@domain |
| 22:51.23 | SeRi | Thanks p3nguin ill do that. |
| 22:51.26 | SeRi | working on it |
| 22:51.50 | Ast001 | or 381117152550@ims.telekom.srbija.com |
| 22:52.35 | ChannelZ | in 1.8 if I do register => Bob@foo:secret@mysiphost.com I get From: <sip:Bob@foo>;taqg=xxxxx |
| 22:53.02 | p3nguin | That's exactly what I said. |
| 22:53.02 | Ast001 | and on 1.4 i can put that in " " and it can register while on 1.6 and 1.8 i get error... |
| 22:54.34 | Ast001 | I believe this is problem Authorization: Digest username="381117152550", realm="ims.telekomsrbija.com", algorithm=MD5, ... |
| 22:54.49 | Ast001 | username should be number@domain not just number. |
| 22:56.13 | Ast001 | from and to are ok I believe but username is not |
| 22:58.23 | Ast001 | you can see comunication here http://pastebin.com/5PHaxNpb |
| 23:02.22 | ChannelZ | change your syntax up: |
| 23:03.05 | ChannelZ | register => username@domain:secret:authuser@sipdomain |
| 23:03.43 | ChannelZ | IE add :authusername after your secret with whatever you want it to be |
| 23:04.09 | Ast001 | ok |
| 23:04.14 | ChannelZ | register => Bob@testy:mysecret:Bob@testy@mysiphost.com |
| 23:05.09 | ChannelZ | I think that seems to give the right results; Digest username="Bob@testy" |
| 23:05.58 | Ast001 | ChannelZ you are the king |
| 23:06.04 | Ast001 | it registered... |
| 23:06.19 | ChannelZ | Hurray! More ale, serving wench! |
| 23:06.29 | SeRi | p3nguin, I ran in to some issues but I have to go to start the bbq. people are starting to arrive. I put everything back. Ill hit you up tonight if you are still around. |
| 23:06.40 | SeRi | cya guys! |
| 23:06.53 | ChannelZ | Bring me back a chicken wing |
| 23:07.03 | SeRi | no ribs? |
| 23:07.12 | SeRi | some beer? or whiskey? |
| 23:07.17 | Ast001 | chicken ? you deserve more |
| 23:07.18 | SeRi | :P |
| 23:07.24 | ChannelZ | Dry maybe.. I'm not actually a fan of BBQ sauce |
| 23:07.38 | SeRi | cya guys! |
| 23:27.29 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:37.28 | *** join/#asterisk cerberus_za (~coert@8ta-151-33-39.telkomadsl.co.za) |
| 23:46.27 | Ast001 | good night and thanks for help especially to ChannelZ |
| 23:51.24 | p3nguin | I tell him the syntax for the register statement, and he thanks someone else. Interesting. |
| 23:51.58 | WIMPy | Use coulours next time ;-) |
| 23:52.07 | WIMPy | -u |
| 23:52.30 | p3nguin | *shrug* |
