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| 01:17.05 | zopsi | Any of you seen pots lines display merely "asterisk" on the CID display? I'm using a standard Digium 4-port fxo card. Is that just a default set somewhere when the telco provides no data? |
| 01:19.25 | zopsi | It is only happening on some calls (i.e. calls that should say Blocked, Private, or Unavailable). |
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| 02:00.15 | bsdice | zopsi in sip.conf [general] try to set |
| 02:00.20 | bsdice | sdpowner=Administrator |
| 02:00.22 | bsdice | callerid=Administrator |
| 02:00.26 | bsdice | see if it changes anything |
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| 02:05.30 | bsdice | oh, pots, might need some callerid= in its config file too |
| 02:07.46 | zopsi | yeah... |
| 02:08.04 | zopsi | we have a hybrid system. Thanks for the tip though I think I can set it in dahdi config or dahdi channel. |
| 02:08.27 | zopsi | Just don't know if it is possible to do all three of those on POTS like the telco does. |
| 02:09.01 | zopsi | bsdice |
| 02:09.46 | zopsi | Is ARI new in 12 or was it in 11? |
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| 02:26.47 | mjordan | zopsi: very new in Asterisk 12 |
| 02:27.00 | gnudna | Hi guys was wondering is there a way to stop calls from being recorded using a key combination? |
| 02:27.11 | gnudna | this is for a queue |
| 02:28.12 | zopsi | Thanks mjordan. One more question, I have a few Digium D70s and I was wondering whether 12 currently supports or will support the DPMA. |
| 02:28.31 | mjordan | zopsi: it will :-) |
| 02:28.38 | mjordan | stay tuned |
| 02:30.30 | zopsi | mjordan: Awesome, we are looking to upgrade at some point to 12 for ARI and other improvements. Any ETA on that being even in an unstable svn repo? I'm supposed to have this system in place by mid February. |
| 02:31.22 | mjordan | unfortunately, I don't have a firm ETA on the DPMA. Being a commercial product, we want to make sure we test the hell out of it before releasing it, and it hasn't gone through all of the testing. That aside, it is being worked on - but I don't want to give out dates and not hit them. |
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| 02:32.35 | gnudna | is what im looking for even possible? |
| 02:37.26 | *** part/#asterisk mjordan (~matt@nat/digium/x-yhcijyubkqxbkmwr) |
| 02:41.54 | gnudna | just stumbled across the feature i was looking for. |
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| 02:43.53 | gnudna | is there a difference between automon and automixmon |
| 02:44.03 | gnudna | they seem to do the same thing |
| 02:44.15 | gnudna | according to the notes in features.conf |
| 02:45.01 | zopsi | Do you use freepbx? |
| 02:45.35 | gnudna | no straight up asterisk from debian repo |
| 02:46.29 | zopsi | Just making sure. |
| 02:47.20 | gnudna | i believe i found how to enable this option seems simple enough |
| 02:47.23 | zopsi | Is this helpful at all? http://answers.oreilly.com/topic/2741-how-to-record-phone-calls-using-asterisk/ |
| 02:47.45 | gnudna | i just read that |
| 02:47.58 | gnudna | maybe i missed the description about automon vs automixmon |
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| 02:48.31 | gnudna | ok automixmon it is ;) |
| 02:49.39 | zopsi | Someone more knowledgeable than me will probably be able to answer your question more thoroughly just trying to help in the meantime. |
| 02:51.07 | gnudna | thanks zopsi |
| 02:51.42 | gnudna | i found the same link a few min earlier but i thank you for even answering ;) |
| 02:51.58 | zopsi | the magic of Google haha. |
| 02:52.03 | gnudna | i an asterisk rookie at best |
| 02:52.08 | gnudna | an = am |
| 02:52.54 | gnudna | btw i choose straight asterisk vs freepbx and the like cause i want to learn the low level stuff |
| 02:53.58 | gnudna | to be honest except for dialplan the other configs are pretty straight forward. the ones i have used anyways ;) |
| 02:58.03 | gnudna | thanks again |
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| 03:25.02 | Gokee2 | Hello everyone, I upgraded my Asterisk install and am having some issues. I am getting warnings when I load dahdi like the following "WARNING[13258]: chan_dahdi.c:12661 mkintf: Attempt to configure channel 1 with signaling Unknown signalling -1 ignored because it is already configured to be FXS Kewlstart". Is there any way to figure out where its finding this "-1" signalling? I don't see it in any of my dahdi configuration files. |
| 03:26.42 | Gokee2 | I tried reforming all my configuration (Which I have been using since the zaptel days) to fit with the new upgrade as dahdi broke. I took the example config file from the 4th addition Asterisk the definitive guide |
| 03:27.26 | Gokee2 | And I have got most things working again but my fax line refuses to work and I would like to get rid of all the warnings so I can cleanly load dahdi |
| 03:27.44 | ChannelZ | chan_dahdi.conf |
| 03:28.25 | Gokee2 | ChannelZ, I took all my signaling out of that file after seeing the warnings as its already in the dahdi system config |
| 03:28.25 | ChannelZ | is what is loaded when you start Asterisk. You might have something wierd in it, or perhaps it doesn't match what the hardware is configured for, per /etc/dahdi/system.conf |
| 03:29.12 | ChannelZ | system.conf sets the hardware. chan_dahdi is needed to configure asterisk to know about the channels. |
| 03:29.25 | Gokee2 | ChannelZ, http://pastebin.com/8qbrJknV is what mine looks like currently |
| 03:29.40 | Gokee2 | Should I go ahead and uncomment the signaling again? |
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| 03:31.19 | ChannelZ | That's your chan_dahdi.conf? |
| 03:31.56 | Gokee2 | ChannelZ, Yes |
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| 03:32.38 | ChannelZ | Are you trying to use users.conf? |
| 03:32.54 | Gokee2 | No... |
| 03:33.22 | ChannelZ | ok then this is all wrong |
| 03:33.57 | Gokee2 | Ah.... I had an old config that was much longer but dahdi was no longer loading |
| 03:34.04 | ChannelZ | You said you 'updated asterisk', does that mean this was all working sometime before? |
| 03:35.01 | Gokee2 | Yes, on Debian Squeeze, I updated the entire system to Wheezy pulling in Asterisk 1.8 replacing 1.6 |
| 03:35.05 | Gokee2 | It was 1.4 before that |
| 03:35.52 | ChannelZ | and no backups of your configs? Cuz you were either using users.conf before or else I don't see how this config would have worked |
| 03:37.40 | ChannelZ | The syntax in this with the "dahdichan" lines is a bit special, used in conjunction with users.conf -- something I never use -- I can help you fix the chan_dahdi.conf the normal way but if the rest of your setup IS such that it is supposed to be using users.conf, then it won't help much |
| 03:37.59 | Gokee2 | Okay, one sec let me upload the old one |
| 03:38.19 | Gokee2 | It might be a better base if this one is for users |
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| 03:47.41 | Gokee2 | ChannelZ, Okay, this was my chan_dahdi http://pastebin.com/73AeCg5p with dahdi_channels http://pastebin.com/yLFwXbur and this is the errors I get trying to load that config http://pastebin.com/J7FK5qCt |
| 03:48.43 | Gokee2 | I started messing with them because dahdi was not working at all. Turned out to be the dahdi modules was not loaded in the kernel so it was probably a bad idea to mess with those files |
| 03:49.27 | Gokee2 | Now its just got signalling complaints |
| 03:50.37 | Gokee2 | Oh and for good measure this is what my system config looks like http://pastebin.com/CWt9YbnE |
| 03:53.39 | ChannelZ | is this only on a reload? because it all looks fine and those warnings are normal on a live system |
| 03:54.04 | Gokee2 | Oh? They are? Okay |
| 03:57.05 | ChannelZ | yes because you can't change the signalling config live, so it tells you that and then proceeds to whine about ignoring parts of the config it can't change (and acknowledges that they haven't changed anyway) |
| 03:57.27 | Gokee2 | That one has got echocancel=yes but if I do a dahdi show channel 1 it says Echo Cancellation: 1 taps (unless TDM bridged) currently OFF Is that going to kick in when a call comes in? |
| 03:59.17 | Gokee2 | Okay it did change to on when I called in |
| 03:59.36 | Gokee2 | But only 1 tap... Is that normal? I thought it would need something like 256? |
| 04:00.22 | ChannelZ | Yeah I think default is 128 |
| 04:01.29 | ChannelZ | although for olsec dunno.. |
| 04:02.20 | Gokee2 | Is it best to make changes in dahdi-channels.conf or chan_dahdi.conf? |
| 04:04.48 | ChannelZ | well chan_dahdi.conf is the actual file that gets read. Then you are #including dahdi-channels.conf so they all get mashed into one, albeit sequentially. |
| 04:05.02 | ChannelZ | I don't use dahdi-channels, I have everything in one place. |
| 04:06.06 | ChannelZ | So it's not a matter of 'is it best', but more 'however you ultimately want to do it' |
| 04:06.37 | Gokee2 | Ah okay, I think this is the way the original book suggested, its been a few years |
| 04:09.58 | ChannelZ | the auto-config script will write it's output into dahdi-channels.conf so as not to destroy your chan_dahdi.conf accidentally if you ran it |
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| 04:17.44 | Gokee2 | ChannelZ, Hmmm, is oslec a good echo canceler to use or should I use something else? I am still getting lots of echo |
| 04:18.22 | ChannelZ | I use mg2 just fine, never tried oslec |
| 04:20.15 | Gokee2 | ChannelZ, Do I need to restart dahdi or such to make a new one take affect? Can I just change oslec to mg2 in the dahdi system config to try it? |
| 04:22.23 | ChannelZ | yeah need to restart dahdi and asterisk |
| 04:28.45 | Gokee2 | ChannelZ, Ah, a restart of Dahdi cleared it all up. Working much better now, thanks! |
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| 06:12.42 | chare | what does google voice dropping xmpp mean for asterisk |
| 06:15.58 | ChannelZ | "adios" |
| 06:16.05 | [TK]D-Fender | BAI BAI |
| 06:16.24 | [TK]D-Fender | no XMPP = no protocol to talk to * = no more GV on * |
| 06:16.42 | chare | but theres going to be a replacement api right? |
| 06:16.53 | chare | RIGHT? |
| 06:17.07 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:17.44 | chare | I was going to go through the trouble of figuring out how to use google voice as a freed landline and you guys are saying thats going away for good? |
| 06:17.45 | ChannelZ | Well there is one. Hangouts. Based on who knows, but it doesn't matter because it's closed. |
| 06:18.06 | chare | so now there is no way to get a free landline with google voice? |
| 06:18.45 | ChannelZ | You still can, you just can't hook into it other than Google's own software anymore. |
| 06:18.50 | [TK]D-Fender | BAI BAI <---- |
| 06:19.04 | [TK]D-Fender | Free lunch time is almost over. |
| 06:19.30 | chare | ChannelZ what are you saying as an alternative instead |
| 06:19.46 | ChannelZ | paying |
| 06:19.53 | chare | no |
| 06:20.14 | ChannelZ | Then talking to no one, I guess. |
| 06:20.33 | ChannelZ | At least you'll have more free time |
| 06:21.20 | chare | tell me theres a replacement api |
| 06:21.44 | chare | why the hell would google get rid of xmpp with no replacement planned |
| 06:22.06 | ChannelZ | there's not, Google is doing their own propriatary thing (or maybe not completely, but cutting off external access at minimum) |
| 06:22.40 | ChannelZ | Which is not to say they might not re-open it in the future, but I tend to doubt it |
| 06:23.42 | chare | wait isn't it the case that XMPP support for Google Chat, the service used by Google Voice for VoIP calling, is going away, and being replaced by Hangouts, so just use the hangouts protocol instead of xmpp |
| 06:24.13 | ChannelZ | Hangouts is everything.. voice, video, chat.. all integrated into Google Plus and Android apps and Chrome and shit |
| 06:24.38 | chare | but I'm saying in theory that asterisks could have a hangouts addon to make this work? |
| 06:24.41 | chare | is that possible? |
| 06:24.45 | ChannelZ | Underneath bits of it might still BE XMPP, who knows.. that isn't the point |
| 06:25.29 | ChannelZ | It's possible if they published it and allowed 3rd parties to connect. They aren't. |
| 06:26.03 | WIMPy | Or you do some reverse engeneering. |
| 06:26.26 | ChannelZ | Come May or whenever it is they say they're turning it off, the servers will be gone.. or stop accepting connections, or be encrypted needing a certificate you don't have, etc. Same result either way |
| 06:26.55 | chare | god damn it are you seriously saying that google has no hangouts public API |
| 06:27.18 | ChannelZ | Over and over |
| 06:28.22 | ChannelZ | I mean they have an API for things INSIDE Hangouts, plugins and crap, but dunno to what degree (if any) it could be hijacked to be useful in this regard |
| 06:28.49 | ChannelZ | But they have no API for writing your own hangouts client so to speak.. that I'm aware of |
| 06:28.52 | [TK]D-Fender | Google does not have to care about offerit it as a public API |
| 06:29.00 | [TK]D-Fender | They could choose to license it |
| 06:29.06 | ChannelZ | sure |
| 06:29.06 | [TK]D-Fender | They could leave it 100% private |
| 06:29.16 | chare | theres an ios app that does google voice calls free, so I can could continue to use free calls but I would be fixed to using a clunky iphone for all calls right? |
| 06:29.44 | [TK]D-Fender | Guess how long that iOS app will continue working... |
| 06:29.53 | chare | its google's own app |
| 06:30.00 | chare | so its gotta be there forever |
| 06:30.09 | ChannelZ | hahaha just like Wave! |
| 06:30.12 | [TK]D-Fender | Or until they change their mind. |
| 06:30.34 | chare | wave was just stupid, didn't solve any new problems, don't know why people expected it to take off |
| 06:31.02 | ChannelZ | http://www.wordstream.com/articles/retired-google-projects |
| 06:31.12 | ChannelZ | Pick whichever you like better |
| 06:31.34 | chare | https://itunes.apple.com/us/app/google-voice/id318698524?mt=8 |
| 06:31.39 | chare | thats going to be aroud |
| 06:31.46 | WIMPy | So what new problem does GV solve? |
| 06:32.24 | ChannelZ | Whether it's that or Hangouts, yes you'll still be able to access your Google Voice until they kill that too (or as TK says, change their minds.) But no 3rd-party access, a-la Asterisk. |
| 06:33.08 | [TK]D-Fender | Google doesn't want to get abused but cheap-shit calling-card companies using PBX's to get unlimited free termination. |
| 06:33.25 | [TK]D-Fender | So BAI BAI integration. |
| 06:33.31 | chare | I don't understand the terminology |
| 06:33.35 | chare | explain what you mean |
| 06:33.53 | ChannelZ | Basically do what you're trying to do |
| 06:34.12 | [TK]D-Fender | This (like Skype) is left for cheap chumps who don't mind the inconvenience of using their proprietary tools just to place 1 simple call at a time |
| 06:34.12 | chare | hey google voice charges for international calls, they're not going ripped off |
| 06:34.16 | ChannelZ | hook your phone system up to GV for free instead of paying an ITSP or a telco etc |
| 06:34.57 | [TK]D-Fender | They off just enough to stop other startups and not enough to lose (too much) on the deal |
| 06:35.02 | [TK]D-Fender | offer* |
| 06:35.39 | [TK]D-Fender | And then let the rest of the "high-end" users whine about not getting more free stuff. |
| 06:36.08 | chare | you guys are so negative in predictions |
| 06:36.25 | ChannelZ | s/negative/realists/ |
| 06:37.02 | [TK]D-Fender | A realist looks like a pessimist but actually has a clue about what's going on. |
| 06:39.08 | ChannelZ | I don't much care about GV; I liked Talk for the techno-nerd angle, that people could call my company from their GMail or whatnot. Same with Skype before MS killed integration on that. |
| 06:39.57 | chare | god damn it I don't want to pay verizon or at&T |
| 06:40.02 | chare | fucking monopolies |
| 06:40.25 | ChannelZ | I wish ENUM or whatever it's called would take off for bridging VoIP<->PSTN a bit more transparently |
| 06:40.55 | ChannelZ | Then don't. Get an ITSP. |
| 06:41.12 | [TK]D-Fender | [01:39]charegod damn it I don't want to pay verizon or at&T [01:39]charefucking monopolies<- I know dozens of ITSPs. Who said all that is left is Verizon and AT&T? |
| 06:41.28 | WIMPy | The NSA woll make sure ENUM doesn't get implemented in any relevant product. |
| 06:41.32 | [TK]D-Fender | You just seem to be whining about losing a free ride |
| 06:41.41 | ChannelZ | I pay $1.50/mo plus like $.01/min in the US which is fine because I don't use the phone a lot at home. But I pay per year less than I paid for 1 month from the phone company. Win. |
| 06:41.57 | chare | DAMN RIGHT I'M WHINING ABOUT FREE RIDE |
| 06:42.12 | ChannelZ | Get an Obamaphone |
| 06:43.16 | chare | look this is what is going to happen |
| 06:43.25 | chare | there is going to be an asterisk hangout addon |
| 06:43.29 | chare | to make things work again |
| 06:43.35 | chare | then we don't need to pay for landlines |
| 06:43.43 | chare | then we are going to setup a cjdns wireless meshnet |
| 06:43.52 | chare | so we can avoid paying fucking mobile phone plan |
| 06:44.34 | WIMPy | do it |
| 06:44.35 | ChannelZ | Money-farting unicorns first |
| 06:47.37 | chare | why are you guys being negative again |
| 06:47.57 | WIMPy | What's negative about "do it"? |
| 06:48.40 | WIMPy | We are jut startingto set up a mesh network here as well. |
| 06:48.46 | ChannelZ | You can already free your ass off with SIP or other protocols. But connecting to the PSTN is not free. |
| 06:49.09 | [TK]D-Fender | SOMEBODY is paying for that. |
| 06:49.36 | [TK]D-Fender | Your free app is going to come with advertising.. then time limits, and all sort of other fun things in any sense of time. |
| 06:50.12 | chare | google voice lets you connect to PSTN for free, but NOT free for international calls |
| 06:50.17 | chare | i'm willing to pay for international calls |
| 06:50.32 | ChannelZ | SO GET AN ITSP |
| 06:51.03 | ChannelZ | What are Goooooogle's rates for say China? |
| 06:51.05 | [TK]D-Fender | It's not that we're "negative". It's that you delusionally optimistic verging on manic psychotic. |
| 06:51.18 | [TK]D-Fender | Oh ... and nobody cares what you are personally willing to pay. |
| 06:51.21 | ChannelZ | Not less than anyone else's I imagine. |
| 06:51.40 | chare | so then you agree its a workable model channelz |
| 06:51.48 | chare | no need to be pessimistic about google voice then |
| 06:52.07 | [TK]D-Fender | You are not a relevant market share. There's got to be money in there somewhere and your free lunch doesn't fit the big picture |
| 06:52.08 | ChannelZ | It's not pessimism, it's reality. |
| 06:52.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.156) |
| 06:52.47 | [TK]D-Fender | [01:51]chareno need to be pessimistic about google voice then <- Do you know how many free rides I've seen go away? lots. |
| 06:52.59 | chare | STOP THE FREE |
| 06:53.01 | chare | ITS NOT FREE |
| 06:53.05 | chare | INTERNATIONAL CALLS COST MONEY |
| 06:53.08 | chare | SO STOP CALLING IT FREE |
| 06:53.18 | [TK]D-Fender | that is PARTLY free. |
| 06:53.31 | [TK]D-Fender | and now they are taking THAT part oaway from "lets let busniess abuse this |
| 06:53.42 | chare | what abuse? |
| 06:53.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.156) |
| 06:53.51 | [TK]D-Fender | CALLING CARD COMPANIES |
| 06:54.58 | [TK]D-Fender | "Hey lets chain a bunch of these together to pum thousands of calls from India to America for free! |
| 06:55.01 | chare | ok you're going to have to explain calling card company abuse |
| 06:55.05 | [TK]D-Fender | Yeah... SOMEBODY is losing money there. |
| 06:55.17 | chare | i'm not familiar with how it works |
| 06:55.29 | [TK]D-Fender | Google is paying for that usage. |
| 06:55.37 | chare | how does calling card company abuse work? |
| 06:55.38 | WIMPy | You will be once they used your server to do it :-) |
| 06:56.22 | [TK]D-Fender | Well that's something else. |
| 06:56.36 | chare | explain how the abues works |
| 06:56.40 | [TK]D-Fender | I just did |
| 06:56.51 | chare | what is "chain" |
| 06:56.58 | [TK]D-Fender | HOPOK UP |
| 06:57.05 | [TK]D-Fender | Set up 1 stupid PC with 10000 logins |
| 06:57.08 | chare | I'M NOT AN EXPERT AT THIS STUFF |
| 06:57.11 | chare | EXPLAIN IN ENGLISH |
| 06:57.13 | [TK]D-Fender | pass a shitload of calls through it |
| 06:58.34 | [TK]D-Fender | So they make it hard to do so by cutting off open access APIs |
| 06:59.02 | [TK]D-Fender | And then eventually even the free little stuff simply becomes "cheap" one you get "addicted" to their service |
| 06:59.09 | ChannelZ | At the end of the day you are not entitled to free calls. Interstate or international or otherwise. It costs Google money to do what they do. It's their toys, they can do what they want to. |
| 06:59.13 | [TK]D-Fender | Like the one you are now desperately clinging to. |
| 06:59.24 | *** join/#asterisk Hilverdays (~jhilverda@office.voys.nl) |
| 06:59.28 | [TK]D-Fender | It's sad to see someone get so thoroughly played and not get it... |
| 07:00.28 | chare | I'M USING GMAIL FREE |
| 07:00.29 | chare | YOU CAN'T SAY THAT WILL END |
| 07:00.39 | chare | THERE I REFUTED YOUR GENERALIZATION |
| 07:00.52 | Cynagen | it's not free... |
| 07:01.00 | [TK]D-Fender | Like those people who got a FREE DID with Google... guess what.. you gave that # to too many of your contacts. They EXPECT to reach you on it now. It WAS a free ride... until NOW and the price goes up. You'd change... but no, you'll complain first.. and then you'll pay because it's still less than the pain of trying to leave let alone what the alternatives might be. |
| 07:01.13 | Cynagen | there's ads inserts along the sides, and there's datamining |
| 07:01.17 | chare | what is a "DID" |
| 07:01.19 | ChannelZ | No one has made a generalization. |
| 07:01.22 | [TK]D-Fender | [02:00]chareTHERE I REFUTED YOUR GENERALIZATION <- with a shittier one of your own... and no basis of comparison |
| 07:01.29 | Cynagen | you're just feeding them information about you in return for your "free" service |
| 07:01.40 | [TK]D-Fender | I *have* seen a ton of "free termination services" DIE |
| 07:01.44 | chare | if the calls stop being free I'll switch |
| 07:02.01 | Cynagen | I sense Fender is readying his "fuck this" button (aka, disconnect) |
| 07:02.13 | chare | who the hell is this new cynagen guy joining the conversation |
| 07:02.24 | Cynagen | i'm an old lurker |
| 07:02.32 | ChannelZ | Oh you're making new friends now |
| 07:02.34 | Cynagen | i got in here back when gvoice was new |
| 07:03.10 | Cynagen | and support totally sucked and you had to use a hacked up script method to make calls |
| 07:03.38 | [TK]D-Fender | Similar to Skype... who also killed their public API. |
| 07:03.49 | [TK]D-Fender | Guess what... THAT free ride died |
| 07:03.51 | ChannelZ | Well they never really had one either |
| 07:03.59 | Cynagen | i digress, i just wanted to mention that you're sitting here arguing about 'free' shit online, it's not truly free |
| 07:04.02 | ChannelZ | That was *licensed* and they still killed it! |
| 07:04.04 | Cynagen | somebody is footing a bill |
| 07:04.15 | [TK]D-Fender | Now to connect with normal systems you have to use a Business account. |
| 07:04.22 | Cynagen | and you're contributing one way or another, whether you realize it or not |
| 07:04.59 | chare | look I'm willing to go through the trouble of setting up asterisk to connect with google voice, how many people are willing to do that, not a lot, so the free loader problem isn't a big problem |
| 07:05.03 | Cynagen | once that sinks in, a lot of why those "free" services die, makes a lot more sense, they didn't really demand your portion, they merely asked for it, and you, being the dick you are, free loaded |
| 07:05.03 | ChannelZ | Wasn't the original impetus for GV to help train their voice recognition? |
| 07:05.35 | *** join/#asterisk bengoodman (~bengoodma@nc-76-5-182-52.dhcp.embarqhsd.net) |
| 07:06.06 | Cynagen | i'm tired, there's my $.02 |
| 07:06.11 | WIMPy | Yes, the NSA will pay more for transcrips than for recordings. |
| 07:06.20 | [TK]D-Fender | ChannelZ: You mean NSA Voice-Fingerprinting Profiler 1.0! |
| 07:06.23 | WIMPy | +t |
| 07:06.25 | Cynagen | whoever is op, feel free to boot me when you wanted |
| 07:06.30 | Cynagen | want* |
| 07:06.37 | Cynagen | shit i'm tired |
| 07:06.49 | WIMPy | Voice figerprinting is older than that. |
| 07:06.51 | [TK]D-Fender | Cynagen: You're not crossing any lines here so far... |
| 07:07.09 | [TK]D-Fender | WIMPy: Asterisk didn't start with whole numbers either... it's all marketing ;) |
| 07:07.12 | chare | look all those people who have those obihai or whatever they are called, are going to flood the asterisk forum for a hangout plugin replacement |
| 07:07.22 | chare | i have confidence this is just a temporary problem |
| 07:08.07 | [TK]D-Fender | They are moving to hangouts. There is no API. Keep praying |
| 07:08.15 | ChannelZ | I have an investment opportunity for you then |
| 07:08.20 | ChannelZ | Please send me all your personal information |
| 07:08.31 | [TK]D-Fender | And obihai users are in for a rude awakening |
| 07:09.02 | [TK]D-Fender | ChannelZ: Like my Arizona ocean-front real-estate business? |
| 07:09.13 | ChannelZ | Even better! |
| 07:09.17 | [TK]D-Fender | \o/ |
| 07:09.18 | Cynagen | when i saw those obihai's |
| 07:09.24 | Cynagen | i was wondering how they were pulling it off |
| 07:09.36 | Cynagen | i figured it was the same API asterisk was hooking |
| 07:09.40 | [TK]D-Fender | Cynagen: Well XMP is there... for now... and rthe clock is ticking |
| 07:09.57 | Cynagen | meh |
| 07:10.00 | Cynagen | i have a backup plan |
| 07:10.04 | Cynagen | it's not a great one mind you |
| 07:10.07 | [TK]D-Fender | Think their access is special? Think they'll get any special privileges? |
| 07:10.12 | Cynagen | it's actually a really shitty one |
| 07:10.13 | ChannelZ | Tin cans and LOTS of string |
| 07:10.22 | Cynagen | not that shitty |
| 07:10.24 | chare | http://www.dslreports.com/forum/r28323794- |
| 07:10.25 | Cynagen | tyvm |
| 07:10.26 | chare | eat that |
| 07:10.30 | ChannelZ | Free as a naked hippy! |
| 07:10.33 | chare | someone knows how to make hangouts work |
| 07:10.35 | chare | EAT THAT |
| 07:10.44 | Cynagen | eat your own offspring |
| 07:10.48 | Cynagen | please stop reproducing |
| 07:11.39 | chare | thats right |
| 07:11.44 | chare | read it and weep |
| 07:11.46 | Cynagen | i fucking hate 1-UP people, i used to be one, i hated myself, then i learned it's not about 1-UP'ing everyone, it's about actually 1-UP'ing everyone else |
| 07:11.52 | Cynagen | just... spare me |
| 07:11.58 | [TK]D-Fender | chare: See how long that lasts... |
| 07:12.00 | chare | SO YOU CONCEDE |
| 07:12.06 | Cynagen | can somebody post me this guy's fucking hostname? |
| 07:12.07 | [TK]D-Fender | chare: No. |
| 07:12.12 | Cynagen | i'm gonna block him through my bouncer |
| 07:12.21 | Cynagen | i don't even want his messages being delivered to my client anymore |
| 07:12.28 | chare | hey i'm not hiding behind a proxy |
| 07:12.37 | Cynagen | neither am i dbag |
| 07:12.44 | Cynagen | my bouncers run on my router |
| 07:12.55 | ChannelZ | Until Google changes the protocol. Or the encryption key. Or whatever. No one said it's impossible, but it'll probably turn into some cat & mouse bullshit where it breaks every 4 days... if you want to depend on that level of 'service', well more power to you |
| 07:12.57 | Cynagen | i shut down my pc nightly, so i wanted to stay in the rooms |
| 07:13.06 | Cynagen | local bouncers |
| 07:13.14 | Cynagen | stop fucking assuming you know everything |
| 07:13.28 | ChannelZ | try /ignore |
| 07:13.33 | chare | Did google ever announce that they're trying to stop these obihai devices? |
| 07:13.45 | ChannelZ | oh lord |
| 07:13.48 | Cynagen | again, i don't want his messages passing through the router anymore |
| 07:14.01 | Cynagen | his messages should not be transmitted through my local network |
| 07:14.18 | [TK]D-Fender | chare: Do they have to say what segment they're after? |
| 07:14.38 | Cynagen | obihais are the worst offenders |
| 07:14.48 | [TK]D-Fender | chare: Keep praying. You do that... An sure, use your obihai... doesn't get the call to Asterisk in any convenient way necessarliy. |
| 07:14.51 | Cynagen | the product alone is purposefully designed to freeload |
| 07:15.09 | WIMPy | chare: Why don't you just wirte a nice letter to google and ask them your questions? |
| 07:15.14 | ChannelZ | wanders off to have a Magic Jack if you know what I mean |
| 07:15.15 | [TK]D-Fender | chare: Expecting the ride to last forever is a fool's game. |
| 07:15.53 | chare | hey they just seem to be saying we are consolidating everything realitem into google hangouts, where does consolidation mean kill off free voice |
| 07:16.01 | ChannelZ | snickers - [Jan 9 00:07:53] WARNING[15931]: res_xmpp.c:3766 xmpp_client_thread: JABBER: socket read error |
| 07:16.28 | Cynagen | right there |
| 07:16.36 | Cynagen | that's where it says "kill off free voice" |
| 07:16.38 | ChannelZ | And for the last time, no one said they were killing off GV |
| 07:17.06 | ChannelZ | They might, who knows. But for now they are restricting access to it. |
| 07:17.25 | Cynagen | anyone remember when they found the SIP server addresses for that one day? |
| 07:17.31 | Cynagen | remember how fast that shit got closed up? |
| 07:19.52 | chare | anyway going to take a break to play starcraft 2 |
| 07:20.03 | ChannelZ | For free I bet |
| 07:21.02 | [TK]D-Fender | And... way past bed time. Later all |
| 07:21.05 | ChannelZ | Oooh now I'm getting /msg insults! |
| 07:21.39 | [TK]D-Fender | The best kind.... |
| 07:21.55 | ChannelZ | <chare> you failing college and girlfriend cheated on you? |
| 07:22.09 | ChannelZ | Oh, snap! |
| 07:25.24 | chare | 6 months from now we'll have voice with hangoutsFREE. |
| 07:25.31 | chare | and i'll be back to prove you all wrong |
| 07:29.59 | chare | no response? |
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| 07:34.59 | kaldemar | a blog post about rigging another pc running hangouts to asterisk with a modem and a soldering iron does not count. |
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| 09:01.09 | chare | question guys |
| 09:01.17 | chare | how do you do voicemail with asterisk |
| 09:04.43 | Chainsaw | chare: There's an app for it. |
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| 09:16.55 | ChannelZ | voicemail.conf |
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| 09:55.46 | Blashyrkh | what does Set(CALLERID(pres)=35) do?, i cant find i list of numbers i can enter here or what they do |
| 09:58.28 | WIMPy | You're not supposed to use numbers. See 'core show function CALLERPRES'. |
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| 10:16.39 | weinerk | hi. WaitforSilence is killing the CLI interface .. same old as here: http://www.eflo.net/VICIDIALforum/viewtopic.php?p=22927 |
| 10:16.39 | weinerk | Is there any solution to stop it from dumping on console? |
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| 10:56.26 | *** join/#asterisk Moz (Moz@cpc1-yead1-0-0-cust45.7-1.cable.virginm.net) |
| 10:57.02 | Moz | Hi, any asterisk developers in the channel who can provide some paid for development? |
| 10:58.09 | WIMPy | You will have to be a LOT more specific to hope for an answer. |
| 10:58.57 | Moz | I'd like to validate that a dialled number is a possible valid number using libphonenumber and if the number is not valid respond with a 484 error |
| 10:59.16 | Moz | If it is a valid possible number then continue with a dialplan as normal |
| 10:59.35 | Chainsaw | So you want a custom application to be implemented. |
| 10:59.58 | Chainsaw | Presumably you aim to have it send upstream for inclusion. |
| 11:00.04 | Chainsaw | sent, even. |
| 11:00.38 | Moz | You mean to be added to the Asterisk source? |
| 11:00.41 | Chainsaw | Yes. |
| 11:00.42 | WIMPy | phone number validation is not possible. |
| 11:00.43 | WIMPy | Except for certain ranges of numbers. |
| 11:00.50 | Chainsaw | WIMPy: Yes, some countries have those annoying "open" number plans. |
| 11:01.03 | zamba | i'm trying to run music on hold using an internet stream, and getting lots of the following: |
| 11:01.10 | Moz | Chainsaw - I wasnt planning on it but that's a good idea to add it back into the source |
| 11:01.25 | Chainsaw | Moz: It is good practice. It will help ensure two things. |
| 11:01.30 | WIMPy | Well, it's pretty helpfull if you want DID. |
| 11:01.30 | zamba | res_musiconhold.c:659 monmp3thread: poll() failed: Interrupted system call |
| 11:01.34 | Moz | WIMPy - libphonenumber will validate enough for me |
| 11:01.36 | Chainsaw | Moz: Continued maintenance and forward porting when the Asterisk structure changes. |
| 11:01.47 | Chainsaw | Moz: And contributions from other users. |
| 11:02.11 | Chainsaw | zamba: Generally that suggests that the binary you called exited. |
| 11:02.14 | Chainsaw | zamba: And as such, it was restarted. |
| 11:02.50 | Moz | Chainsaw: OK thanks. Sounds like a good plan. Can you provide help with getting it running or recommend anyone that can? |
| 11:03.16 | zamba | Chainsaw: /usr/bin/wget -q -T 120 -O - http://scfire-mtc-aa02.stream.aol.com:80/stream/1048 | /usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -20 - |
| 11:03.19 | Chainsaw | Moz: There are quite a few consultants that lurk here. |
| 11:03.21 | zamba | Chainsaw: that's all there is in it |
| 11:03.41 | Chainsaw | Moz: You may have to be a bit more specific about where you are, what you aim to pay as an hourly rate, etc. |
| 11:04.27 | zamba | ah, got it working |
| 11:04.33 | zamba | wrong url :) |
| 11:04.44 | Chainsaw | zamba: You've made it impossible for errors to be reported this way, indeed. |
| 11:05.13 | Chainsaw | zamba: But the "interrupted system call" can be read as "hey it stopped, I kicked it". |
| 11:05.39 | Chainsaw | zamba: I get it once every 12 minutes, once my music on hold track has reached the end. |
| 11:07.14 | Moz | Chainsaw: I'm in the UK but the development can be done remotely. I'm using Asterisk with A2Billing and the validation check should be done before the call gets to A2Billing. I hope it'll only take a day programming so I could pay $180 to get it implemented. |
| 11:07.43 | Chainsaw | Forgive me, but it seems odd for someone in the UK to pay in dollars. |
| 11:08.04 | Moz | Chainsaw - Well I figured £110 and then converted it to dollars as a lot of people seem to work in dollars |
| 11:08.07 | zamba | Chainsaw: i've made it impossible? what do you mean? |
| 11:08.26 | Chainsaw | zamba: The way you pipe that wget into the madplay means that any errors will not be logged or output. |
| 11:25.08 | chare | developers FIX GOOGLE VOICE integration now that xmpp is gone |
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| 11:46.40 | Greenlight | Is there an easy or slick way to do like a reverse pattern match on an extension. For example, I want to block (Hangup) all internation calls, *except* those to one particular country ? |
| 11:46.55 | Chainsaw | Greenlight: So you need a negation operator in the extension match. |
| 11:47.02 | Greenlight | Yes |
| 11:47.58 | Chainsaw | Greenlight: The up to date documentation is here, and I see no negation support mentioned: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 11:48.15 | Greenlight | Yup already got that on my screen... |
| 11:48.25 | Greenlight | Was hoping I was missing something :) |
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| 11:48.43 | Chainsaw | Greenlight: Preliminary google results are not encouraging. |
| 11:48.53 | Greenlight | Yea. Ah well, perhaps was a long shot :) |
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| 11:49.21 | Chainsaw | Greenlight: The match order is quite rigidly defined though. Does that not help? |
| 11:49.46 | Greenlight | Not that I can see |
| 11:50.01 | Greenlight | WEll, unless I repeat a load of dialplan |
| 11:50.22 | Greenlight | Guess I'll extract that to a sub and do it that way |
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| 11:51.28 | Chainsaw | Greenlight: It looks like the most specific pattern will always win, so if you have the specific allow and a generic deny, the allow will win for your special country code. |
| 11:51.37 | Greenlight | Yea exactly |
| 11:51.57 | Greenlight | At the moment we block all international calls, and allow the rest |
| 11:52.13 | Greenlight | But seesm they want to be able to call Ireland |
| 11:52.31 | Greenlight | So I need to match ireland first, and then hit my "allowed" section of the dialplan |
| 11:52.39 | Chainsaw | Ah yes, +353 |
| 11:52.41 | Greenlight | Yea |
| 11:52.49 | Chainsaw | Always a fun one. Everybody seems to assume a country code is 2 digits. |
| 11:53.02 | Greenlight | Trust the Irish |
| 11:53.15 | Greenlight | Mind, great country, was over there not so long ago |
| 11:53.22 | Chainsaw | Oh yes, I've lived in Dublin for two years. |
| 12:02.47 | anonymouz666 | mine is 2 digits |
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| 12:17.24 | skrusty | is the *11 documentation for AMI events complete? It seems to be missing a number of events. Or is it still a work in progress? |
| 12:19.26 | *** join/#asterisk qoottaa (~Adium@83.69.225.78) |
| 12:22.45 | qoottaa | hello all! |
| 12:23.35 | qoottaa | anybody use cdr-stats for statistic? |
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| 12:40.41 | zamba | Chainsaw: what's the better way? |
| 12:41.47 | *** join/#asterisk infobot_ (~infobot@rikers.org) |
| 12:41.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 12:42.56 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 12:42.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 12:59.18 | Greenlight | Oh joy, second segfaut this week.... |
| 13:01.33 | skrusty | heh :) |
| 13:05.43 | Greenlight | Bloody net-snmp again |
| 13:06.00 | Greenlight | Ironic that it's there to monitor the health of the * box, and it's what actually kills it |
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| 13:10.15 | Greenlight | Damn, thought I'd fixed this by upgrading net-snmp earlier in week. |
| 13:16.20 | skrusty | Greenlight :) |
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| 13:17.44 | IMS77 | Hi, question: how in my dialplan can I force to test if a peer is registered? I've used qualifyfreq to reduce the time between 2 checks presence but I'm using a softphone on a smartphone and so I'm killing my battery (and pollute my network)! |
| 13:17.44 | IMS77 | I've tested with MessageSend (outside a call) => i've got a failure when the peer is not registred but Asterisk do not update its status. Any ideas how can I do? |
| 13:19.08 | [TK]D-Fender | IMS77: "core show function SIPPEER" |
| 13:21.31 | Greenlight | What I don't get is that Asterisk had been up over 100 days prior to the segfault earlier this week, and had been fine. And now it dies a few days apart. |
| 13:22.01 | IMS77 | SIPPEER will give me the current status like "show sip peers", right? so if Asterisk does not detect the peer is not registered (with qualify) how to force it to update its status |
| 13:22.24 | [TK]D-Fender | IMS77: READ THE INSTRUCTIONS |
| 13:22.46 | [TK]D-Fender | IMS77: and "qualify" does not "detect registration" |
| 13:23.19 | kaldemar | IMS77: qualify does not check if a peer is registered. use a combination of dynamic and ip fields of SIPPEER for that. |
| 13:23.48 | IMS77 | no it will send OPTION and if the peer is not available => change the status to UNAVAILABLE. When I read instruction is see "Doesn't work with RealTime." |
| 13:24.04 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 13:25.08 | qoottaa | hello all, which static use for call-center? |
| 13:25.26 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:25.37 | [TK]D-Fender | qoottaa: Please rephrase your question. |
| 13:25.43 | IMS77 | what do you mean by dynamic and ip fields |
| 13:26.57 | [TK]D-Fender | IMS77: look at the IP. Not having a valid IP listed = not registered. |
| 13:27.03 | kaldemar | IMS77: core show function SIPPEER |
| 13:27.05 | qoottaa | need reporting module for call-center, i try asternic and other, but there small of analitics |
| 13:27.17 | [TK]D-Fender | IMS77: Having an IP but a negative status = not going to get called anyway |
| 13:27.33 | [TK]D-Fender | qoottaa: QueueMetrics |
| 13:27.34 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 13:27.46 | qoottaa | yep |
| 13:28.26 | Moz | Greenlight - have you tried using something like a2billing? It would be overkill but you could install a2billing and then send all calls there and only have one route open which is ireland |
| 13:28.29 | qoottaa | but… цу |
| 13:28.42 | qoottaa | * but we stopped on cdr-stats |
| 13:29.04 | [TK]D-Fender | qoottaa: That made no sense... |
| 13:29.25 | Greenlight | Moz: It's okay, we stoted it now, just extracted our "allowed" code out, and then had both default and ireland goto that path |
| 13:30.42 | qoottaa | [TK]D-Fender: it module has poor documentation's |
| 13:31.19 | [TK]D-Fender | qoottaa: It's documentation is fine and as it is a PAID solution they also include installation support |
| 13:32.43 | [TK]D-Fender | qoottaa: and there are manuals on their site for this. |
| 13:32.48 | [TK]D-Fender | qoottaa: http://queuemetrics.com/path_support.jsp |
| 13:32.57 | qoottaa | [TK]D-Fender: hehe… paid solutions it's not interestingly :( |
| 13:33.35 | [TK]D-Fender | qoottaa: If you want "good" with lots of reports... get ready to pay for it. |
| 13:33.40 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 13:35.00 | qoottaa | [TK]D-Fender: yep, "good" reporting system worth the money |
| 13:35.51 | qoottaa | but can u give advice which opensource solutions can help me |
| 13:36.38 | [TK]D-Fender | ~wikis |
| 13:36.38 | infobot | well, wikis is VoIP Wiki covering FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners. For Asterisk, see ~asteriskwiki |
| 13:36.43 | [TK]D-Fender | ^^^ They have a list |
| 13:36.47 | [TK]D-Fender | Go look for yourself. |
| 13:37.00 | [TK]D-Fender | I use Asternic myself and am happy with it. |
| 13:37.22 | [TK]D-Fender | If you're not, I'm not going to guess which other ones won't make you happy. |
| 13:37.22 | qoottaa | u have a call-center? |
| 13:37.29 | [TK]D-Fender | Obviously |
| 13:37.58 | qoottaa | heh |
| 13:39.07 | IMS77 | Qualify does not detect if a peer is registered but with the use of qualifyfreq when the peer is not available the status (displayed with sip show peers) change to UNREACHABLE. With a softphone application on a smartphone maybe the peer is reachable maybe not. The registration status can be OK even if the application has gone (depends on the timeout). I search the good method to force Asterisk to refresh its status or to directly have the reachable statu |
| 13:39.27 | qoottaa | thx |
| 13:39.53 | [TK]D-Fender | IMS77: If it times out... it has FAILED to respond. You can't make a phone respond. |
| 13:40.16 | [TK]D-Fender | IMS77: And if it's lagged out * is not going to send it calls or messages |
| 13:40.59 | IMS77 | no i would like to say the registered timeout (can be 1 hour) |
| 13:41.35 | IMS77 | peer seems is registered but we can't call it. |
| 13:41.51 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 13:42.20 | [TK]D-Fender | [08:40][TK]D-FenderIMS77: And if it's lagged out * is not going to send it calls or messages <--------- |
| 13:42.46 | [TK]D-Fender | [08:27][TK]D-FenderIMS77: Having an IP but a negative status = not going to get called anyway |
| 13:42.54 | IMS77 | I want to get the reachable status to be able to wake up the application on the smartphone using a specific development |
| 13:43.57 | [TK]D-Fender | IMS77: There is no "wakeup my phone" command. |
| 13:44.32 | [TK]D-Fender | IMS77: It is failing qualify. get it to stop failing |
| 13:45.17 | *** join/#asterisk Draecos (~Draecos@58-7-60-116.dyn.iinet.net.au) |
| 13:45.19 | [TK]D-Fender | IMS77: qualify IS the problem. If you don't want it to stop * from being able to try calling or messaging the device... then DISABLE IT. You placed this restriction on yourself. |
| 13:45.26 | IMS77 | not in Asterisk but I have a specific development which can do "wake up!" |
| 13:47.01 | IMS77 | qualify with qualifyfreq) is a good solution. The only problem it check every xx second if the peer is reachable or not => on a smartphone it has a cost |
| 13:47.10 | [TK]D-Fender | IMS77: then call your wakeup script |
| 13:47.56 | IMS77 | I would like to be able to send an OPTION (like qualifyfreq does) on demand |
| 13:47.58 | saint_ | hi all - Is there a way to find what line is screwed up in my extension.conf , when I get the error message in /var/log/asterisk/message: [2014-01-09 08:46:40] WARNING[27644] pbx.c: Error in extension logic (missing '}') |
| 13:48.41 | [TK]D-Fender | [08:47]IMS77I would like to be able to send an OPTION (like qualifyfreq does) on demand <- you're not going to get * to do that. |
| 13:49.05 | [TK]D-Fender | IMS77: and if you have the ability to push triggers out the device... then make it RE-REGISTER instead |
| 13:49.32 | [TK]D-Fender | IMS because sending it an OPTIONS packet externally is not going to make * process that answer if it didn't originate it |
| 13:50.02 | [TK]D-Fender | saint_: You'd see based on the rest of the lines that are getting processed |
| 13:51.15 | saint_ | [TK]D-Fender it gives me 4 useless lines |
| 13:52.03 | [TK]D-Fender | saint_: Show us the full execution |
| 13:52.10 | [TK]D-Fender | saint_: and the dialplan to match |
| 13:52.55 | IMS77 | I want to RE-REGISTER only when needed not before every dial/call. And because the peer can have a state OK I don't know how long to wait before to be able to dial. |
| 13:53.01 | saint_ | [TK]D-Fender those are the 4 lines out of message: http://pastebin.com/aE2X4MY2 |
| 13:54.54 | saint_ | [TK]D-Fender i guess sometimes it just takes to come here and talk with someone who ask you to pastebin error messages. because i just found it. i had no clue I had a function FILTER |
| 13:55.19 | saint_ | but I found it in my conf. and I found the missing parenthesis ... damn it.. |
| 13:55.26 | *** part/#asterisk qoottaa (~Adium@83.69.225.78) |
| 13:56.13 | [TK]D-Fender | IMS77: There is no option to force * to keep trying forever to qualify. It failed. It's there on purpose. |
| 13:56.21 | [TK]D-Fender | IMS77: If you don't like it, stop doing it. |
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| 13:58.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:02.03 | saint_ | so if I use Digium phones with DPMA , do I have to add the option t and/or T for a Digium phone to blind transfer an incoming call internally / externally |
| 14:02.06 | saint_ | ? |
| 14:02.38 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:02.46 | [TK]D-Fender | saint_: No. |
| 14:03.12 | [TK]D-Fender | saitIt's a SIP phone like any other.... real phones have real transfer options and don't need crappy DTMF call-features for things like that |
| 14:03.42 | [TK]D-Fender | saint_: Almost all ATA's also have their own methods for this... most using the hook-flash, etc. |
| 14:04.11 | [TK]D-Fender | saint_: that leaves just about the only thing needing it being direct POTS/PSTN connections like DAHDI FXS and the PSTN-side |
| 14:04.56 | saint_ | [TK]D-Fender ok, because when i receive an incoming call, if i pick up with a D70 , then hit transfer / dial_internal_x / hit transfer again, it will hang up on me |
| 14:05.08 | saint_ | [TK]D-Fender it only works if the internal extension i called picks up first |
| 14:05.40 | [TK]D-Fender | saint_: Read the dons on the precise order to do your transfers properly |
| 14:05.42 | [TK]D-Fender | docs* |
| 14:06.24 | saint_ | [TK]D-Fender i m only using the TRANSFER key on the D70. One would think it would work out of the box .. |
| 14:06.43 | [TK]D-Fender | [09:04]saint_[TK]D-Fender ok, because when i receive an incoming call, if i pick up with a D70 , then hit transfer / dial_internal_x / hit transfer again, it will hang up on me <- this sounds like a blind transfer which WOULD hang up on you .... and the transferree continues on on their own. |
| 14:07.00 | [TK]D-Fender | saint_: I'm thinking your timing is off in the process |
| 14:07.23 | saint_ | [TK]D-Fender when I said it hangs up on my , my bad. It hangs up on the person I am trying to transfer.. i m making tests myself here.. sorry |
| 14:07.43 | [TK]D-Fender | saint_: You're not using it properly then |
| 14:07.49 | [TK]D-Fender | saitread the user guide |
| 14:08.47 | IMS77 | [TK]D-Fender:Thanks for your help. I 've search a solution to my problem on many forum but the only working solution i found untill now to get the reachable status of a peer is to set a short qualifyfreq. It's working but it's a bad solution => ping (sending OPTION) every xx seconds to my smartphone, kill my battery and doesn't work in all the cases. I'm here to get an other idea. I've try MessageSend(). When the peer is unreachable, i got a failure reply |
| 14:09.29 | [TK]D-Fender | IMS77: STOP USING QUALIFY THEN |
| 14:09.43 | [TK]D-Fender | IMS77: If you want to complain that you end up enreachable.... STOP DOING IT |
| 14:10.16 | Katty | morning |
| 14:10.56 | [TK]D-Fender | IMS77: MessageSend(). When the peer is unreachable, i got a failure reply <--- You could also just queue up your message and resend it after it DOES come back anyway |
| 14:11.16 | [TK]D-Fender | IMS77: These problems are entirely your own doing. |
| 14:13.31 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:18.58 | IMS77 | Maybe I'm not clear. A must call B on a smartphone via Asterisk. B is registered but reachable or not. I have to do the call in any case. So if the B is not reachable (application not started, ip changed because of the 3G) => i search a way to detect that B is not reachable. If so I can ask him to wake up and re-register. When I have to detect that the new registration is done to be able to intiate the call. That's all. I'm using Asterisk for all the sip |
| 14:20.42 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 14:22.48 | [TK]D-Fender | IMS77: STOP USING QUALIFY THEN. |
| 14:23.02 | [TK]D-Fender | IMS77: The fact you are USING it is why it won't call. DISABLE IT. |
| 14:23.13 | IMS77 | ok |
| 14:23.29 | [TK]D-Fender | IMS77: If you disable it ... Asterisk WILL just try and send the call/message\ |
| 14:24.41 | *** part/#asterisk lkthomas (~lkthomas@n218250150072.netvigator.com) |
| 14:25.14 | saint_ | what is the best way to access voicemail calling the asterisk from outside , without a welcome ivr ? Is there a way to redirect to voicemailmain if a dtmf is entered while the phone rings or the voicemail greetings plays ? |
| 14:26.58 | [TK]D-Fender | there are 2 options to exit VoiceMail. |
| 14:27.40 | [TK]D-Fender | There is also an option to abort a dial if you really want to go that way. |
| 14:27.46 | [TK]D-Fender | but I wouldn't bother |
| 14:28.32 | saint_ | [TK]D-Fender i m just looking for a way to dial in from my cell phone, and have access to my voicemail if i need to |
| 14:29.06 | [TK]D-Fender | saint_: You've already listed 2 valid ways. Exit VoiceMail to enter the admin... or make an ivr option for it. |
| 14:29.28 | [TK]D-Fender | saint_: Or setup a DID you can identify to auto-acceppt based on callerID... |
| 14:29.44 | [TK]D-Fender | saint_: Or whatever |
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| 14:31.05 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 14:33.19 | saint_ | [TK]D-Fender but how do you exit the voicemail. i tried to have operator = yes and use the extension 'o' to redirect, but it does not catch my dtmf .. |
| 14:34.46 | [TK]D-Fender | saint_: Then either your DTMF itself has issues, or you didn't put it in the right place |
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| 15:03.39 | MacroMan | Is there a database or community of people offering commercial Asterisk programming on the net anywhere? |
| 15:04.06 | *** join/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 15:05.18 | DelphiWorld | lol ifdowneth0 |
| 15:05.30 | DelphiWorld | someone please may give me the current svn trunk or ast? |
| 15:05.40 | ifdowneth0 | lol its my afk name |
| 15:05.41 | ifdowneth0 | ;) |
| 15:06.20 | ifdown-wlan0 | nullroute ifdowneth0 |
| 15:07.25 | DelphiWorld | guys, svn repo please? didnt find it... |
| 15:08.38 | Qwell | http://www.asterisk.org/downloads/source-code |
| 15:09.32 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.180) |
| 15:10.20 | DelphiWorld | Qwell: thank Quntom well :) |
| 15:12.30 | IMS77 | [TK]D-Fender: i've tryed without qualify and re-read the doc => it's not how i have to work. Without qualify i don't know if a peer is registered or not => ok i can dial but if the peer is not reachable dial will fail. I just need to know before to dial if really a peer is reachable. If not i wake it up and must wait a new registration. When registered, i dial. |
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| 15:12.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:13.06 | [TK]D-Fender | IMS77: Without qualify i don't know if a peer is registered or not <- qualify does NOT prove registration |
| 15:13.19 | [TK]D-Fender | IMS77: you know it's registered if it has an IP <- |
| 15:13.34 | [TK]D-Fender | IMS77: And that it's registration period has not EXIRED |
| 15:14.15 | IMS77 | ok and how to know if the peer is reachable? |
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| 15:17.28 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:22.39 | IMS77 | You can consider that the peer is not reachable when in the process of dialing the first INVITE doesn't receive a 200 OK (after a timeout). But the dial command will only return a failure status after the ringing timeout and you can't make the difference between both. You can't know if it's the INVITE which fail or if it's the user who didn't reply to the incoming call. |
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| 15:33.18 | Ibrahim22 | Hi, my sip peers don't have the default 5060 rport assigned to them, but rather a random one between 5000-50000 |
| 15:33.57 | Ibrahim22 | This is causing pjsip to not register the contact address properly, and giving them the status "Unknown". Anybody else ran into this problem? |
| 15:40.23 | mirela666 | Hi, does anyon know how to call and give parameters to ODBC_custom function when I want to insert 3 values? |
| 15:40.25 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-cwccchjkjndnmiqv) |
| 15:40.26 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:40.52 | mirela666 | Set(ODBC_ins_language()=${ARRAY(${CDR(uniqueid)},${digit},${CHANNEL(language)})}); |
| 15:41.06 | mirela666 | isn't this with Array corect way? |
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| 15:43.25 | Blashyrkh | is there a way to trace ss7 packets |
| 15:43.47 | Blashyrkh | in asterisk |
| 15:46.35 | newtonr | Blashyrkh, I've not used ss7 stuffs personally, but a quick google found http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+commands |
| 15:47.34 | newtonr | possibly the "ss7 dump start ..." |
| 15:48.24 | Blashyrkh | it was ss7 set debug on linkset |
| 15:49.05 | [TK]D-Fender | [10:22]IMS77You can consider that the peer is not reachable when in the process of dialing the first INVITE doesn't receive a 200 OK (after a timeout). But the dial command will only return a failure status after the ringing timeout and you can't make the difference between both. You can't know if it's the INVITE which fail or if it's the user who didn't reply to the incoming call. <-... |
| 15:49.06 | [TK]D-Fender | ...there is no "what I can". Asterisk only considers things unreacahble that is checks up on with qualify. For everything else it WILL simply call out. |
| 15:49.13 | newtonr | Blashyrkh, ah for console debug, cool |
| 15:49.39 | *** join/#asterisk SGjunior (~sgjunior@out-pq-169.wireless.telus.com) |
| 15:49.44 | [TK]D-Fender | IMS77: You do not get to jsut randomly change *'s behavior for not getting a 200 (or some such) via any standard configuration. That would require a substantial re-write of the Dial application itself. |
| 15:49.55 | [TK]D-Fender | IMS77: And you are mixing what you WANT with what Asterisk DOES. |
| 15:50.31 | [TK]D-Fender | IMS77: "qualify=no" = Asterisk WILL dial. If it gets a failure you'll get a code for why. |
| 15:55.38 | *** join/#asterisk marlinc (~marlinc@ip565fa73c.direct-adsl.nl) |
| 15:56.13 | IMS77 | I understand your position. I'm just saying that when the invite doesn't reply 200 ok it's quite different if the user doesn't pickup the call. Asterisk doens't give us the posibility to make the difference and my trying to find an other way. Modifying Asterisk is not a good and easy option. Maybe I will have to implement a sip stack in my AMI development, I hope no. |
| 15:56.19 | IMS77 | thanks anyway |
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| 15:57.34 | heliar | Hi. |
| 15:59.23 | navaismo | wasa |
| 16:01.03 | [TK]D-Fender | IMS77: "Asterisk doens't give us the posibility to make the difference and my trying to find an other way" <- wrong. If the phone doesn't respond at all it will NOT respond "NO ANSWER" |
| 16:01.15 | [TK]D-Fender | IMS77: You'll have your status |
| 16:06.29 | ipengineer | Does anyone know if support was ever added for CHANNEL(peername) in PJSIP? |
| 16:07.28 | file | peername? no |
| 16:07.48 | file | endpoint, yes |
| 16:13.44 | navaismo | awesome trying to install asterisk on virtualbox with debian 7 and get--->illegal Instruction. See why I hate Debian :P |
| 16:14.51 | newtonr | navaismo, i installed it on debian 7.2 on virtualbox and had no issue |
| 16:19.55 | navaismo | i guess is a hardware issue now is compiling without BUILD_NATIVE |
| 16:26.19 | navaismo | yep disabling build_native works |
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| 17:03.15 | mjordan | ipengineer: yes it was |
| 17:03.19 | mjordan | but it isn't that :-) |
| 17:04.19 | mjordan | ipengineer: use CHANNEL(endpoint) to get the name of the endpoint, then use PJSIP_ENDPOINT to get more details |
| 17:04.20 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_CHANNEL |
| 17:04.26 | mjordan | and https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_ENDPOINT |
| 17:05.00 | mjordan | and I just now totally saw that file already responded |
| 17:06.53 | mjordan | really needs to fix the formatting of the CHANNEL documentation |
| 17:15.16 | newtonr | navaismo, i didn't have to disable build_native. what version of Asterisk, and what system options did you select for your virtualbox system? and was it debian 32bit or 64bit? |
| 17:17.30 | navaismo | Asterisk version 11, debian 32bits, PAE/NX only |
| 17:17.56 | navaismo | 11.6.0 |
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| 17:31.25 | newtonr | navaismo, interesting, i had asterisk version 1.8, debian 32bits, VT-x/AMD-V, Nested Paging. didn't do PAE/NX |
| 17:31.42 | anonymouz666 | north america freezing and here 40+ celsius. Nice. |
| 17:32.47 | navaismo | :S |
| 17:34.51 | navaismo | newtonr, I guess some times the VMs do weird things |
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| 17:39.43 | jesselang | Hello, I'm using Asterisk 1.6 with DAHDI. I'm sending calls from Freeswitch to Asterisk, then to the T1 circuits. After some time sending calls, the call count on Asterisk is 10 channels higher than on Freeswitch. Can someone help me troubleshoot why these calls aren't tearing down? |
| 17:44.24 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 17:44.51 | gmalsack | ARGGG!!!! I HATE DISTRIBUTED DEVICE STATE!!!!!! |
| 17:44.56 | gmalsack | p.s. i'm bald now.... |
| 17:45.36 | *** join/#asterisk SGjunior (~sgjunior@out-pq-169.wireless.telus.com) |
| 17:46.16 | gmalsack | I've got xmpp debug enabled, and can see messages leaving pbx1 heading to my pubsub server, but never see them on pbx2.... anyone else ever setup tigase for distributed device status? |
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| 17:51.36 | anonymouz666 | gmalsack: please don't hate DDS, that won't help yuo |
| 17:51.56 | anonymouz666 | are you distributing over LAN or WAN? |
| 17:51.57 | ChannelZ | Don't hate the player, hate the game |
| 17:52.00 | gmalsack | anonymouz666: k |
| 17:52.04 | gmalsack | lan |
| 17:52.31 | gmalsack | anonymouz666: lan |
| 17:53.06 | gmalsack | oops shit, have to pickup someone from work at noon.... lol brb |
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| 18:31.27 | newtonr | gmalsack, i found openfire much easier to use |
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| 18:42.43 | newtonr | jesselang, Do the calls involve AGI scripts that may be still running? The channels may still then show up in various CLI commands. What call count are you looking at? Also, you may consider upgrading, as the latest 1.6 was End of Life more than one and a half years ago. |
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| 18:43.14 | jesselang | newtonr, I'm using the packages from Debian stable. |
| 18:46.00 | jesselang | newtonr, No AGI scripts are involved. Asterisk is simply serving as a SIP -> DAHDI gateway. |
| 18:46.19 | newtonr | jesselang, and where are you looking for call counts? |
| 18:46.38 | jesselang | core show channels count |
| 18:46.57 | jesselang | And in Freeswitch: show calls count |
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| 18:47.36 | jesselang | When the calls stop, there will be 6 or more active calls on Asterisk when Freeswitch shows 0. |
| 18:48.09 | jesselang | I'm going FS -> Asterisk -> DAHDI |
| 18:48.55 | newtonr | can you pastebin the "core show channels" output when it is showing active calls? |
| 18:49.14 | newtonr | Specifically when it is showing active calls that you feel shouldn't be there |
| 18:51.22 | jesselang | newtonr, sure thing. Pretty boring though. They all say AppDial((Outgoung Line)) on one leg, and my Dial() string on the other leg. |
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| 18:52.41 | newtonr | jesselang, pastebin it, and "sip show channels" if they are SIP, plus a "sip show channel <channelname>" for two channels in one pair of legs |
| 18:57.41 | newtonr | jesselang, oh and "core show channel <channelname>" for those same two legs |
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| 19:03.06 | jesselang | newtonr, it's possible my theory is getting beat up by an ugly gang of facts. |
| 19:03.42 | jesselang | Eventually most all of the calls did finally end. |
| 19:04.07 | jesselang | Now I'm trying a config change to see if it makes a difference. |
| 19:04.13 | newtonr | jesselang, well that is good |
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| 19:08.39 | jesselang | newtonr, any idea why call tear down would take a while like that? |
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| 19:10.04 | jesselang | It is still happening with my change, so I'll go through the steps of gathering the information. |
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| 19:11.25 | gmalsack | newtonr: damn we are already using openfire.... lol didn't know it did pubsub |
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| 19:13.07 | leifmadsen | gmalsack: hells ya |
| 19:13.33 | ipengineer | I am trying to figure out why I can make and receive calls and have no issues with Media, but any media coming from asterisk destined to the PSTN is not working. So If I take a DID point it to a device everything works fine. If I point that to Playback and audio file I get no audio |
| 19:14.04 | newtonr | gmalsack, and when you set it up with openfire you should write a little guide for wiki.asterisk.org since I forgot to do so a while back. :) |
| 19:21.36 | anonymouz666 | ipengineer: is this asterisk 12? |
| 19:21.41 | ipengineer | Yes |
| 19:21.48 | anonymouz666 | scared |
| 19:22.34 | ipengineer | I am curious if it has to do with the fact that my trunk to flow route is running over chan_sip while everything else is through PJSIP. Although again what is weird is I can have it Dial a device and not run into any problem. |
| 19:24.22 | gmalsack | newtonr: leifmadsen: ok, I'll get it done, then write a wiki and send it to leif for editing before I publish! :-D |
| 19:24.40 | leifmadsen | sounds right |
| 19:24.41 | leifmadsen | :D |
| 19:25.38 | leifmadsen | newtonr: did you get openfire and DDS to work? |
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| 19:26.04 | leifmadsen | Marquis42: openfire won't work with distributed device state (XMPP) will it? |
| 19:26.11 | leifmadsen | I thought it required collection nodes? |
| 19:26.34 | leifmadsen | or perhaps that is not the case since we're going to try and do that at Thinking Phones without the collection nodes I guess |
| 19:26.52 | newtonr | leifmadsen, yeah it worked, i set it up several months back. |
| 19:27.06 | Marquis42 | Yeah, it'll work without collection nodes on OF AFAIK. |
| 19:27.31 | Marquis42 | If you want/need collection nodes your only option is Tigase, though. |
| 19:30.44 | newtonr | gmalsack, ^ more info for the wiki :D |
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| 19:36.29 | kristoffelloos | I'm looking into Opennms to monitor some asterisk servers, who has done this for real on his server, can you compile the snmp module for asterisk and get everything up and running without losing any concurrent calls?? |
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| 20:13.02 | drmessano | kristoffelloos, why would you lose concurrent calls? |
| 20:13.04 | drmessano | Load? |
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| 20:49.10 | Katty | looks in |
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| 20:56.11 | Katty | hi n3hxs |
| 20:56.42 | n3hxs | Hi, How are you. |
| 20:56.58 | Katty | oh just peachy. how're you dear? |
| 20:57.36 | n3hxs | Finally home again. |
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| 20:58.16 | n3hxs | Katty: Left for Europe Dec 1 just got in the house |
| 20:58.36 | Katty | n3hxs: always feels good to get home. have you collapsed on your bed yet? |
| 20:59.33 | n3hxs | Not yet, but it sure looks good. Especially since I am running 6 hours later than Eastern time. |
| 21:00.59 | Katty | nods |
| 21:01.29 | n3hxs | Did you have a good Christmas & Newyear? |
| 21:02.12 | Katty | yes, it was quite lovely! bit stressful, as usual...but still wonderful. what about yours? hopefully not too overwhelming? |
| 21:03.20 | n3hxs | Christmas with my daughter and 3 grand kids, New Year's eve in Frankfurt as we flew back to DC on the 1st. |
| 21:03.38 | n3hxs | 5 days in Napa CA with the other daughter, then finally home. |
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| 21:04.17 | n3hxs | Looked at your birdies a few times while away. |
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| 21:16.04 | Katty | n3hxs: sounds like you had a great christmas!!! |
| 21:16.19 | Katty | n3hxs: and hopefully many many more to come :> |
| 21:17.05 | n3hxs | Katty: Yes, went to several Christmas markets in Germany Austria and the Czech Republic. |
| 21:17.24 | Katty | sounds tasty! |
| 21:18.05 | bsdice | Gingerbread-Powerrun :) |
| 21:18.08 | n3hxs | Yes, one could gain quite a bit of weight with the wine, beer and all the other goodies. |
| 21:20.43 | Katty | doner kebabs ^_^ |
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| 21:29.20 | navaismo | does anyone knows or have a tip to solve when a cisco 7912 stop requesting the tftp? |
| 21:30.20 | bsdice | turn off, turn on again :) |
| 21:30.29 | bsdice | <PROTECTED> |
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| 21:37.36 | Ryushin | I've been trying to google this for the last two hours. Running 11.4.0. When inside voicemail, option 3, Advanced Options does not do anything. Pressing 3 does not do anything at this point. I'm at a bit of a loss right now. |
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| 22:27.16 | wmchris | hi i have one question - every time i make a call via SIP Gateway to my ISDN Phone it displays the correct number in the display for one second and then it switches immediately to the extension ID - is there any way to fix this behavior? |
| 22:27.38 | wmchris | i already set exten = s,1,Set(CALLERID(name)=${CALLERID(num)}) (this way it displays the number the first few secs) |
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| 23:09.36 | file | who remembers, back in the day, when there was no ./configure? those were the times |
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| 23:30.54 | mjordan | file: heresy |
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