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| 03:51.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.1.0 (2012/12/10), 10.11.0 (2012/12/10), 1.8.19.0 (2012/12/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 04:07.36 | morfin | hello |
| 04:07.48 | morfin | is there way to monitor user status in Asterisk? |
| 04:07.59 | morfin | like away and reason: WC |
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| 08:37.11 | PbxMan | morning |
| 08:42.22 | ChannelZ | meh |
| 08:43.57 | Wiretap | counterintuitive fact: you won't be getting SIP media through a Juniper JunOS firewall without DISABLING NAT in asterisk... |
| 08:44.21 | Wiretap | the ALG in JunOS does all the SBC itself |
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| 08:47.11 | x1user | Hi all. Is there any software that can help me to test all possible cases of my dialplan and find if there is something wrong? |
| 08:47.27 | Wiretap | yeah |
| 08:47.29 | Wiretap | core set verbose 10 |
| 08:48.17 | x1user | I mean more like fuzzing. |
| 08:48.44 | x1user | I have 2k lines dialplan and i want to be sure that every context is working properly. |
| 08:49.18 | kaldemar | x1user: that's really for you to do. no software can really know what your dialplan is expected to do. |
| 08:49.41 | x1user | i had to ask. thanks anyway :) |
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| 08:50.16 | schmidts | good morning |
| 08:58.39 | davlefou | bonjour, |
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| 09:53.03 | verywiseman | what is the "transfer key" that is used with call parking? |
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| 11:11.19 | [sr] | hi WIMPy |
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| 11:12.52 | [sr] | how to configure to point to point BRI connection? |
| 11:12.59 | [sr] | using just bri_cpe doesn't work |
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| 12:26.32 | tompaw | If I'm doing [channel originate SIP/blah extension blah@super-extension], is there any way to send some extra data to super-extension? |
| 12:26.46 | tompaw | Like one extra value... |
| 12:28.01 | *** join/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 12:28.41 | tompaw | If not, can I somehow set that value prior to calling the channel originate command? |
| 12:30.16 | tompaw | Hm... I think I can merge it into the exten, like 'FOO-BAR' and then explode via '-' in the extension. |
| 12:30.44 | tompaw | Sorry, in the context. |
| 12:31.13 | kaldemar | extension sounds right. |
| 12:32.06 | tompaw | I mean explode the ${EXTEN} in the context ;) |
| 12:32.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:32.48 | tompaw | (where context == dialplan) |
| 12:33.05 | kaldemar | still, an extension is where you do the stuff. |
| 12:33.31 | tompaw | Yep, so this CUT()-based approach, are there any reasons not to do it that way? |
| 12:34.17 | kaldemar | if you use CLI originate, you have no other way. |
| 12:34.59 | passerby | I have a question about chan_dongle... I use E173 dongles in only GSM mode, but when I'm trying to connect more than 5 devices, they begin to behave strangely. Is there problem with power or something? Is anyone had similar problems? log output: [Dec 17 13:51:47] ERROR[1904] chan_dongle.c: [dongle7] timedout while waiting 'OK' in response to 'AT' After that message device is disconnected and connected again and so on |
| 12:35.16 | tompaw | kaldemar: thanks. |
| 12:35.52 | passerby | but if I connect only 5 devices, they works good |
| 12:39.44 | passerby | hub with external power did not help =( |
| 12:42.09 | ghost75 | wasnt there a gotosubif or something? |
| 12:42.33 | tompaw | Ok, so here's my approach at CUT() http://bpaste.net/show/WU8gx4F55CfRI5hRhZ4i/ |
| 12:44.24 | passerby | ghost75, is it question to me? |
| 12:44.29 | ghost75 | no |
| 12:44.33 | passerby | k |
| 12:45.20 | kaldemar | ghost75: "core show applications like GoSub" |
| 12:45.57 | ghost75 | ah gosubif it was |
| 12:48.16 | passerby | if anyone had experience with module chan_dongle.so please respond |
| 12:48.48 | ghost75 | copy protection in asterisk oO |
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| 12:58.41 | ghost75 | are ami commands displayed in cli? |
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| 13:06.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:06.54 | kaldemar | displayed as in when/how? |
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| 13:16.53 | itamarjp | hello guys |
| 13:17.09 | itamarjp | where I can get help for setting up a digium card with r2 protocol ? |
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| 13:25.18 | ghost75 | i am sending something over ami but dont see anything in cli except login/logoff from ami |
| 13:26.32 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 13:28.22 | gavimobile | when my agents are busy in a call from the queue and a new call comes in to the queue, the busy phones get notified that a new call is coming in. is disabling call waiting the only way do stop this? I've tried using strategy=rrmemory && ringinuse=no but they don't solve my issue |
| 13:29.09 | leifmadsen | gavimobile: no, it sounds like your device state isn't working, thus the queue doesn't know if the phone is busy or not |
| 13:29.14 | leifmadsen | you likely need callcounter=yes in sip.conf |
| 13:30.32 | gavimobile | leifmadsen: for my peer setting or for the general settings? |
| 13:30.51 | leifmadsen | look at the sip.conf.sample file and read the documentation for that option |
| 13:30.52 | kaldemar | ghost75: that's normal until you actually do something with AMI that causes output. |
| 13:30.53 | leifmadsen | it'll become clear |
| 13:31.15 | ghost75 | mhh hard to troubleshoot if no see |
| 13:31.48 | kaldemar | ghost75: add debug to your script or what ever uses AMI. |
| 13:36.55 | gavimobile | leifmadsen: what does this mean? To enable callcounters, you use the new |
| 13:36.55 | gavimobile | ; "callcounter" setting (for extension states in queue and subscriptions) |
| 13:37.26 | leifmadsen | gavimobile: it means callcounter is a new option name for the previous option name described in that same text |
| 13:37.44 | gavimobile | but I don't use call-limit |
| 13:37.51 | leifmadsen | that's fine |
| 13:37.52 | gavimobile | im using 1.8 btw |
| 13:37.56 | leifmadsen | I know |
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| 13:38.13 | leifmadsen | just enable callcounter in order to allow tracking of device state for sip peers |
| 13:39.03 | gavimobile | I did. I added it into my general section and it doesn't sove my problem |
| 13:39.56 | leifmadsen | then you'll need to be more specific about your problem |
| 13:40.24 | leifmadsen | did you follow the queue documentation in asterisk the definitive guide? |
| 13:40.38 | leifmadsen | if you step through it, it'll get a queue up and running and teach you what parts you need involved to make a queue work |
| 13:40.38 | gavimobile | leifmadsen: ok, let me check the verbose to see if it shows inuse when placing a tell call |
| 13:40.49 | gavimobile | leifmadsen: thanks, ill try again |
| 13:40.54 | gavimobile | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html |
| 13:41.05 | gavimobile | An Introduction to Device State |
| 13:42.55 | ghost75 | not even getting error from ami command |
| 13:43.28 | tompaw | Is it CONFBRIDGE(bridge,record_file)=/blah or CONFBRIDGE(bridge,record_file)="/blah"? |
| 13:44.10 | leifmadsen | tompaw: it shouldn't really matter -- but are likely valid, but the first probably more valid |
| 13:44.24 | leifmadsen | double quotes in asterisk dialplan are just literal characters |
| 13:44.25 | gavimobile | I made a test call from peer1 to peer 2 than ran from cli queue show queuename. here is my output. it shows that the caller is not in a call but they are still in a call right now http://pastebin.com/8kn6ZYnV |
| 13:44.25 | tompaw | leifmadsen: thanks. The problem is neither works :P |
| 13:44.34 | tompaw | got it. |
| 13:44.55 | gavimobile | mainQueue has 0 calls (max unlimited) in 'ringall' strategy |
| 13:45.05 | gavimobile | im going to try to change it to rrmemory |
| 13:45.09 | [TK]D-Fender | gavimobile, Show us where you logged those members in in the first place... |
| 13:45.24 | [TK]D-Fender | gavimobile, Strategy has nothing to do with state |
| 13:45.32 | [TK]D-Fender | gavimobile, That change is pointless |
| 13:45.41 | [TK]D-Fender | gavimobile, Show us where you logged them in. |
| 13:46.02 | gavimobile | [TK]D-Fender: thanks, not sure what you mean by where they were logged in |
| 13:46.17 | [TK]D-Fender | gaivShow us where & how your MEMBERS were added to the queue |
| 13:46.23 | [TK]D-Fender | gavimobile, ^ |
| 13:46.45 | gavimobile | [TK]D-Fender: ok |
| 13:47.43 | gavimobile | [TK]D-Fender: here is my queues.conf, since my queue members don't login to receive the call, I added them on the bottom of the file to receive the call http://pastebin.com/sa22eqfV |
| 13:48.15 | [TK]D-Fender | gavimobile, member=Local/110@motekPC <- this is NOT a SIP device. Chane_local knows NOTHING about the call some SIP device is making |
| 13:48.27 | gavimobile | [TK]D-Fender: bingo |
| 13:48.34 | gavimobile | ok, I know what to do |
| 13:48.39 | [TK]D-Fender | gavimobile, You did not tell the queue to look at the SIP device |
| 13:48.43 | gavimobile | it needs to be using the sip channel |
| 13:49.34 | [TK]D-Fender | gavimobile, You either need to change and put the SIP device directly as a member or tell it to use that device for state checks |
| 13:50.18 | gavimobile | SIP/0000FFFF0000 (In use) has taken no calls yet |
| 13:50.22 | gavimobile | ;-D |
| 13:50.24 | gavimobile | lets test this out |
| 13:50.31 | [TK]D-Fender | gavimobile, read the sample config for the static way of formatting that line and "core show application addqueuemember" for the dialplan way |
| 13:50.49 | gavimobile | its workinh |
| 13:51.10 | gavimobile | instead of using local, I changed them to SIP/mypeer |
| 13:51.25 | gavimobile | 1 down 2 to go |
| 13:51.30 | gavimobile | 1 down 1 to go* |
| 13:51.57 | gavimobile | yesterday I was playing with the calendar module. im able to see my upcomming events, but im not getting a call when the event arrives |
| 13:52.07 | gavimobile | nor do I see any logging information in cli |
| 13:53.22 | gavimobile | I create an event, I view the event through the cli, and I see the reminder time. but I never get a call |
| 13:56.15 | *** join/#asterisk serafie1 (~erin@nat/digium/x-cvienvakradydvum) |
| 13:56.44 | tompaw | If I set CONFBRIDGE(bridge,record_file) to anything it doesn't even start the recording stack. Any idea what might be causing it? |
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| 14:00.12 | gavimobile | ? |
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| 14:06.27 | gavimobile | bingo! |
| 14:06.32 | gavimobile | it works on local channel but not sip |
| 14:10.30 | gavimobile | thanks folks! |
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| 14:16.10 | ghost75 | any reason to not work? http://pastebin.com/62cjBbf0 |
| 14:16.24 | tompaw | Here's the CONFBRIDGE issue log: http://bpaste.net/show/dUvbamMF7TvvLOiH7Wt9/ |
| 14:16.42 | tompaw | If I provide the record_file value, MixMonitor Recording doesn't start. |
| 14:17.02 | tompaw | I tried filename with and without extension, both relative and full paths, none helps. |
| 14:18.06 | [TK]D-Fender | ghost75, We don't see what you are populating those vars with, or your dilaplan which it references, etc. Also, Channel => "Local/$argext\@$argcontextlocal", <--- why are you escaping the @? |
| 14:18.32 | ghost75 | its perl, otherwise will treat as something else |
| 14:22.40 | *** join/#asterisk DanFromUK (5189c338@gateway/web/freenode/ip.81.137.195.56) |
| 14:23.09 | DanFromUK | Hi, with cmd controlplayback, how much does the recording jump by when the user presses the rew or ff keys? |
| 14:24.04 | DanFromUK | Actaully, never mind. I just saw the option. |
| 14:24.20 | tompaw | This whole feature is fucked up. On the website https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 you say record_file's value should be *path*, and then in description " the specific name of the recorded file can be set using this option". |
| 14:24.27 | tompaw | Make up your mind then, is it a path or a filename? |
| 14:25.34 | [TK]D-Fender | tomaw, What version are you running? |
| 14:26.34 | tompaw | [TK]D-Fender: 11.0.1 |
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| 14:28.38 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:28.47 | ghost75 | [TK]D-Fender the strange thing is i didnt change the script, its like it forgot something |
| 14:29.03 | [TK]D-Fender | tompaw, pastebin your bridge config |
| 14:29.04 | tompaw | debug doesn't say anything about it. |
| 14:29.19 | tompaw | [TK]D-Fender: it's a default config with recording changed to yes |
| 14:29.25 | [TK]D-Fender | tompaw, pastebin your bridge config |
| 14:29.26 | tompaw | let me strip it off comments and paste |
| 14:29.32 | tompaw | let me strip it off comments and paste |
| 14:31.19 | tompaw | [TK]D-Fender: http://bpaste.net/show/lJ5oaIh5d6156G1yFvCh/ |
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| 14:33.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:34.20 | *** part/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 14:35.42 | [TK]D-Fender | tomaw, Test with another caller in conference |
| 14:37.52 | tompaw | [TK]D-Fender: http://bpaste.net/show/DLVtg2DIsLwUwW1utkds/ << in this case both caller had the record_file set to the same value |
| 14:37.59 | tompaw | s/caller/callers/ |
| 14:38.24 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 14:41.53 | [TK]D-Fender | tomaw, So far not seeing anything wrong.... still reading... |
| 14:42.23 | [TK]D-Fender | tompaw, there was a bug on this supposedly solved for 11.0.1 which you say you're using... |
| 14:42.51 | tompaw | [TK]D-Fender: I debug'd the whole call, but there's nothing there, even the filename is not mentioned in debug. |
| 14:43.07 | tompaw | [TK]D-Fender: Maybe I'll try upgrading to 11.1.. |
| 14:43.27 | [TK]D-Fender | tomaw, Well if you ended the conf, see there's no file under the monitor folder, no errors... well... not looking like it's doing its job |
| 14:43.38 | [TK]D-Fender | tompaw, Certainly a good idea. |
| 14:44.13 | tompaw | [TK]D-Fender: I checked that, too, no file, no subdirs, nothing, I guess the next step would be to trace it, which probably would take me 2 months ;) |
| 14:44.22 | tompaw | Thanks, will let you know how it goes on 11.1 |
| 14:44.50 | leifmadsen | has anyone run into an issue where a dialplan with Answer() exists then goes to a Queue(), and an answered call produces a CDR, but one that falls through the Queue() to say Voicemail() produces no CDR? |
| 14:47.02 | [TK]D-Fender | leifmadsen, Queue() itself should answer the call (unless flagged otherwise), but certainly hitting VM should do it regardless... |
| 14:51.37 | *** join/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 14:52.06 | *** part/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 14:53.22 | Penguin | As I remember it, app_queue does not have an implicit answer. |
| 14:55.09 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 14:56.20 | leifmadsen | "...where a dialplan with Answer() exists then goes..." |
| 14:56.39 | leifmadsen | I'm implicitly answering the call |
| 14:56.48 | Penguin | You're explicitly answering it. |
| 14:56.49 | mjordan | leifmadsen: are the Agents busy? |
| 14:57.06 | leifmadsen | mjordan: no, just falls through on timeout, then goes to Voicemail() |
| 14:57.37 | mjordan | k. If it falls through on time out, then it shouldn't affect the disposition of the CDRs. I'd expect you'd have an Answer, prior to going into Voicemail. |
| 14:57.58 | mjordan | but CDRs in Queue are certainly a mess. |
| 14:58.02 | leifmadsen | the Answer() is earlier in the dialplan -- does that matter? |
| 14:58.24 | leifmadsen | I thought as long as the channel was answered at any period of time a CDR would be produced? |
| 14:58.28 | Penguin | I would have thought that the Answer() before any of it would start CDR at that point. |
| 14:58.32 | mjordan | only if something exists between the Answer and the Queue. You could try using a ForkCDR to see if something else is setting a disposition. |
| 14:58.59 | leifmadsen | mjordan: hmmm interesting. Well there is Answer(), Background(), then Queue(), then Voicemail() |
| 14:59.10 | mjordan | I'd expect that to produce a CDR regardless. |
| 14:59.18 | leifmadsen | indeed |
| 14:59.25 | mjordan | leifmadsen: what version is this? |
| 14:59.32 | leifmadsen | OOOOOLD BASTARD VERSION |
| 14:59.34 | mjordan | ah |
| 14:59.37 | leifmadsen | 1.4.43 I think |
| 14:59.37 | mjordan | heh |
| 14:59.51 | mjordan | yeah, all bets are off then. I know that works in 1.8 - tested it recently |
| 14:59.57 | leifmadsen | indeed |
| 15:00.17 | leifmadsen | oh there is even an Answer() before Voicemail() |
| 15:00.23 | leifmadsen | so ya, wtf |
| 15:00.52 | [TK]D-Fender | leifmadsen, Either way you do directly hit VM. I can't imagine a reason that it shouldn't guarantee one.... |
| 15:02.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:02.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:04.32 | WIMPy | seems to remember the need for an explicit Answer() before VoiceMail(). |
| 15:04.48 | Penguin | Me too. And he has at least three. |
| 15:05.02 | Penguin | BackGround() answers, Answer() answers, etc. |
| 15:05.03 | WIMPy | Should be enough :-) |
| 15:06.21 | tompaw | Which file stores the menuconfig selections? |
| 15:07.08 | WIMPy | menuselect.makeopts |
| 15:07.45 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 15:08.49 | tompaw | Thanks! |
| 15:17.11 | *** join/#asterisk vimreaper (~vimreaper@rrcs-70-62-43-252.central.biz.rr.com) |
| 15:18.05 | tompaw | [TK]D-Fender: same thing on 11.1 :/ |
| 15:18.25 | vimreaper | hey guys i have a question.. if port 5060 is closed to udp but open to tcp will the phone still register, or does 5060 have to allow udp traffic as well for registration? |
| 15:18.57 | [TK]D-Fender | vimreaper, Is your client using TCP? Did you configure your peer to use it? |
| 15:19.05 | [TK]D-Fender | vimreaper, Otherwise, no. |
| 15:19.05 | WIMPy | SIP is usually UDP, but it can use TCP if configured that way. |
| 15:19.27 | vimreaper | ok, as far as i know they didnt configure it to use tcp |
| 15:19.27 | Penguin | Unless you have configured Asterisk to use TCP for SIP and your phone uses TCP, it isn't going to work. |
| 15:19.32 | vimreaper | so thats most likely the problem |
| 15:19.35 | vimreaper | thank you! |
| 15:20.09 | tompaw | F**ing hell, already wasted 3 hours on this damned thing. Anyone here using CONFBRIDGE with * 11? |
| 15:20.39 | WIMPy | tompaw: Yes |
| 15:20.51 | tompaw | WIMPy: are you by any chance using custom file names for recordings? |
| 15:21.13 | WIMPy | Nope. Nopt using recordings so far. |
| 15:21.20 | tompaw | :< |
| 15:24.06 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:24.06 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:27.07 | tompaw | http://bpaste.net/show/FExOFBagh3yVz4R25fB6/ < Added bridging.c debug. It's helpful as hell. |
| 15:27.43 | file | <PROTECTED> |
| 15:28.13 | tompaw | file: so what should I seek in the debug log? |
| 15:29.01 | file | if I were debugging it I'd look for the logic that determines if recording should occur and add stuff around there... |
| 15:29.24 | tompaw | file: I would love to do that, but where to look for that logic if not in debug 99? |
| 15:29.39 | tompaw | you don't mean app_confbridge.c, do you? |
| 15:29.48 | file | I do, and you'd have to look at the code itself |
| 15:29.57 | tompaw | I hate mondays. |
| 15:29.58 | file | there's not debug at every line of execution |
| 15:30.33 | tompaw | How did this thing make it to production code anyway... |
| 15:31.28 | file | it was written, put up for code review, and then ultimately committed |
| 15:31.31 | *** join/#asterisk greenwolf (42570085@gateway/web/freenode/ip.66.87.0.133) |
| 15:31.47 | file | it's entirely possible that it's something unique to how you are doing things that is causing the issue which others have not encountered |
| 15:32.05 | file | or it could be a feature that nobody normally touches |
| 15:33.59 | tompaw | file: sorry, I'm just full of negative energy because of it. |
| 15:34.11 | tompaw | Recording confereces doesn't sound like very exotic feature. |
| 15:37.32 | [TK]D-Fender | tomaw, As a sanity check, try recording with the automatic name selection instead of specifying it.... |
| 15:37.47 | tompaw | [TK]D-Fender: http://bpaste.net/show/FExOFBagh3yVz4R25fB6/ ... |
| 15:37.50 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 15:37.57 | *** part/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 15:37.59 | tompaw | first log shows that it works if there is no name given. |
| 15:38.04 | tompaw | line 7 |
| 15:38.33 | tompaw | And the recording works perfectly fine. |
| 15:39.19 | [TK]D-Fender | tompaw, Sounds like a very isolated aspect that's broken then... Tried with file extension? |
| 15:39.49 | tompaw | [TK]D-Fender: yes, tried both with and without: extensions, full paths, quotas. |
| 15:40.24 | [sr] | WIMPy: are you there? |
| 15:40.33 | [TK]D-Fender | tompaw, :/ Might see about opening up a new issue on the tracker for it. You've done a lot of work collecting debug and should help them to find out what's going on... |
| 15:40.39 | tompaw | Looking at the code, app_mixmonitor would throw a log message if app_confbridge didn't provide it with a file name. |
| 15:40.47 | tompaw | As per app_mixmonitor.c, line 955 |
| 15:40.51 | WIMPy | [sr]: Somehow :-) |
| 15:41.04 | tompaw | So it looks like it doesn't even TRY to record it. |
| 15:41.48 | [sr] | WIMPy: need your help-... problem is on the telco side this is contigured as PtP instead of PTmP, but when i configure this to bri_cpe cannot work |
| 15:42.36 | WIMPy | [sr]: That just means that you cannot connect multiple devices. |
| 15:42.40 | *** join/#asterisk serafie (~erin@nat/digium/x-ryuflrnvtkxxsbjc) |
| 15:42.54 | tompaw | [TK]D-Fender: Whoa!!!! |
| 15:42.56 | tompaw | I think I got it. |
| 15:43.01 | tompaw | https://code.asterisk.org/code/browse/asterisk/trunk/apps/app_confbridge.c?u=3&r=378002 |
| 15:43.04 | tompaw | check this out |
| 15:43.05 | tompaw | line 438 |
| 15:43.11 | tompaw | 437 actually |
| 15:43.18 | tompaw | if (!(ast_strlen_zero(conference_bridge->b_profile.rec_file))) { |
| 15:43.39 | tompaw | the actual variable name is "record_file" |
| 15:43.48 | tompaw | not "rec_file" |
| 15:44.21 | [TK]D-Fender | tompaw, Sounds like a quick fix to test... |
| 15:46.01 | [sr] | WIMPy: even having kjust the asterisk pbx, it doesn't work :( |
| 15:46.22 | WIMPy | What does (not) happen? |
| 15:46.27 | [sr] | WIMPy: i did the test with the telco on the phone, and they change it to PTmP and works fine |
| 15:48.17 | [sr] | WIMPy: whem i try to connect just have a isdn cause 27 |
| 15:49.39 | WIMPy | WHICH I GUESS IS AN INTERNALLY GENERATED ONE. |
| 15:49.41 | WIMPy | oops |
| 15:50.35 | [sr] | WIMPy: hum |
| 15:54.16 | file | tompaw, add a call to Set(CONFBRIDGE(bridge,record_conference)=yes) before the record_file one |
| 15:57.14 | [sr] | WIMPy: how could i colve this? |
| 15:57.18 | tompaw | file: ok |
| 15:57.38 | [sr] | WIMPy: what you're saying is that i can only have the asterisk pbx connected to the NTBA? i already tried but same result and isdn cause |
| 15:58.34 | WIMPy | [sr]: That's the rater obvious difference between point-to-point and point-to-multi-point. |
| 15:58.53 | WIMPy | Do you see any communication with pri debug? |
| 15:59.30 | tompaw | file: whoa, now it works. but why?? |
| 15:59.44 | file | because it's not a combined end result |
| 15:59.58 | file | the dynamic bridge profile created using CONFBRIDGE overrules |
| 16:00.03 | tompaw | Are you saying it ovewrites both or nothing? |
| 16:00.28 | [sr] | WIMPy: understand, but with JUST asterisk doesnt work anyway. i think i'm going to ask to change this to PTmP |
| 16:00.31 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 16:00.45 | file | I don't understand the question |
| 16:00.53 | WIMPy | [sr]: So you wat to connect something else in parallel? |
| 16:00.53 | tompaw | file: thanks a lot, much appreciated. I'd never work it out probably. |
| 16:01.13 | [sr] | WIMPy: forget, its working :D |
| 16:01.24 | file | if you create a dynamic bridge using CONFBRIDGE then that is used, not what is configured in the .conf file |
| 16:01.27 | tompaw | file: I meant the CONFBRIDGE() overwrite replaces the whole structure of bridge profile? |
| 16:01.32 | [sr] | WIMPy: w8, brb give me a sec i'll give you the result |
| 16:01.35 | file | yes |
| 16:04.23 | tompaw | file: is that the intended behavior? Because in my opinion it's not very intuitive and definitely lacks a warning in the docs. |
| 16:04.49 | file | that's the way it's written, if you want to provide some documentation tweakage to reflect that then cool |
| 16:05.03 | tompaw | I understand. Thanks again, back to work! |
| 16:06.29 | [TK]D-Fender | tompaw, So CLI was all-or-nothing for parms? |
| 16:07.13 | tompaw | [TK]D-Fender: the dialplan function CONFBRIDGE(), yeah. |
| 16:07.44 | [TK]D-Fender | tompaw, It was something I was going to suggest as the next sanity check, but was called off here... |
| 16:07.50 | [TK]D-Fender | tompaw, Glad you found it though... |
| 16:08.53 | tompaw | [TK]D-Fender: so am I :) You wouldn't believe how big of a show stopper that was for me. |
| 16:09.41 | *** part/#asterisk PbxMan (c335d91f@gateway/web/freenode/ip.195.53.217.31) |
| 16:09.42 | [TK]D-Fender | tompaw, Actually I could..... |
| 16:10.58 | *** join/#asterisk Xaviertoor (~Xaviertoo@189.52.87.66) |
| 16:11.04 | tompaw | [TK]D-Fender: I was actually looking for where in the code is the b_profile initiated, so maybe after a few days I'd work it out ;) |
| 16:11.09 | Xaviertoor | anybody help-me |
| 16:11.21 | Xaviertoor | I have a problem com sip account |
| 16:11.34 | [TK]D-Fender | ~ask |
| 16:11.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:11.34 | Xaviertoor | NOTICE[100113][C-00000046]: chan_sip.c:10416 process_sdp: No compatible codecs, not accepting this offer! |
| 16:11.38 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 16:11.49 | [TK]D-Fender | Xaviertoor, Like it says... "no compatible codecs" |
| 16:12.07 | [TK]D-Fender | Xaviertoor, * and your device can't agree on codecs |
| 16:12.10 | Xaviertoor | alaw and ulaw |
| 16:12.18 | Xaviertoor | no compatible |
| 16:12.21 | [TK]D-Fender | Xaviertoor, PASTEBIN the entire call attempt. |
| 16:12.24 | [TK]D-Fender | ~pb |
| 16:12.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:12.29 | [TK]D-Fender | ^^^ |
| 16:12.29 | tompaw | Xaviertoor: you have to make sure that the codecs your client (sip-phone) is offering are the same as defined in user's profile. |
| 16:12.36 | [TK]D-Fender | Xaviertoor, With SIP DEBUG enabled naturally... |
| 16:12.37 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 16:13.56 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.101) |
| 16:16.48 | Xaviertoor | [TK]D-Fender, http://pastebin.com/YzdbUkQw |
| 16:17.41 | [TK]D-Fender | Xaviertoor, We clearly don't see an entire call in there especially not the part where there is an actual error message. |
| 16:18.28 | [TK]D-Fender | Xaviertoor, -- Called SIP/ciao/556181482053 <- ther is no SIP debug for this. Do not restrict to your PHONE. It's the OTHER side that has the issue |
| 16:19.39 | tompaw | I always found wireshark more handy than asterisks log messages. |
| 16:20.50 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 16:23.25 | [TK]D-Fender | tompaw, Asterisk will show other things that raw SIP won't ... like proof of what peer it THINK's it should be matching, contexts used, etc. |
| 16:23.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:23.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:24.22 | tompaw | [TK]D-Fender: I agree, but it's not easy for a newcomer to capture it. |
| 16:24.57 | [TK]D-Fender | tompaw, "copy all to buffer" -> pastebin. 10 second job. |
| 16:27.52 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 16:28.39 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 16:28.50 | tompaw | [TK]D-Fender: the problem is, it usualy doesn't fit on the screen. I know it sounds stupid, but it's not. Plus if you run asterisk in screen, it's not even scrollable. |
| 16:32.57 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-philadelphia.hfc.comcastbusiness.net) |
| 16:33.15 | kaldemar | tompaw: asterisk -vvvr | tee /tmp/ast_out.log |
| 16:34.40 | tompaw | kaldemar: I have ways to capture it NOW. But in the early days, it was problematic. The guys who's been asked for a call log has now been quiet for over 20 minutes. |
| 16:36.02 | *** join/#asterisk Claies (~chatzilla@rrcs-97-76-29-141.se.biz.rr.com) |
| 16:36.44 | Claies | hello, anyone available to help me with an unusual issue? |
| 16:37.02 | file | if you ask a question or describe the problem someone may respond |
| 16:37.03 | tompaw | We've all been waiting for you the whole day! |
| 16:37.21 | file | except me, I know absolutely nothing |
| 16:37.57 | tompaw | He's probably gonna ask you to seek answer in the source code. |
| 16:38.17 | Claies | I am trying to install goautodial following their instructions at http://goautodial.org/projects/goautodialce/wiki/64bit |
| 16:39.28 | Claies | their instructions are straightfoward, and I receive no errors through any of the yum commands |
| 16:39.46 | Qwell | huh, they use my packages. Neat. |
| 16:39.49 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 16:39.59 | Claies | however, upon reboot, I continually receive fatal: Module dahdi not found. |
| 16:40.39 | Claies | I have tried every option that I can identify, and even tried to download and install the latest dahdi package and install from source |
| 16:41.14 | Qwell | Claies: Their packages are out of date. |
| 16:41.44 | Qwell | amateurs >.> |
| 16:41.58 | Claies | I kept the kernel at 2.6.18-308.16.1.el5 because the packages wouldn't compile using the kernel from yum update |
| 16:42.20 | Qwell | uname -r |
| 16:42.35 | Claies | and I am aware that the packages are keeping asterisk 1.4, but that is a requirement for vicidial (they have not updated their conferencing options) |
| 16:42.53 | tompaw | I think stuff like that should be delivered as virtual appliance. |
| 16:43.00 | tompaw | But what do I know... |
| 16:43.07 | Claies | I don't have that option in this environment |
| 16:44.03 | Claies | thing is, I should be able to just follow their instructions and use the packages from their guide, and it should load even if they are out of date, no? |
| 16:44.18 | Qwell | Not if you don't have a module for your currently running kernel, no. |
| 16:44.20 | Claies | as long as I keep the kernel matching their packages, at least |
| 16:44.23 | Qwell | Go ask them for help. |
| 16:45.28 | Claies | I wouldn't even be messing with this if it weren't for the client's need for vicidial and vicidial's refusal to move past 1.4 |
| 16:45.39 | Qwell | mutters something about them keeping my name as the packager on the RPMs. |
| 16:45.47 | Qwell | BAD. Why do people do that? |
| 16:46.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:46.04 | file | because you are awesome? |
| 16:46.36 | Qwell | and of course there's no email address where I can yell at them |
| 16:47.52 | Qwell | Claies: Nobody here can help you with their brokenness. |
| 16:48.54 | Claies | ok so I guess I'll try another distro that has vicidial and see what happens (this one is opensuse) |
| 16:49.39 | Claies | I hate being stuck on 1.4.... anyone know of an autodialer like vici but current 10/11? |
| 16:52.20 | Qwell | meh, autodialers |
| 16:52.27 | Qwell | good luck getting much help |
| 16:52.43 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 16:52.45 | [TK]D-Fender | tompaw, don't use screen then. And newb won't even know of its existance |
| 16:52.49 | drfreeze | Hi |
| 16:52.50 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 16:53.44 | tompaw | [TK]D-Fender: the newb you were trying to help went silent for a reason. |
| 16:53.46 | drfreeze | I just installed a new router and have 4 out of 8 phones not registering |
| 16:54.13 | drfreeze | they are connecting to the ftp server and downloading the config files |
| 16:54.34 | drfreeze | And I can ping the phones |
| 16:54.53 | drfreeze | when I reboot the phones I see: Registration failed User: 677, Error Code:480 Temporarily not available |
| 16:55.46 | tompaw | Qwell: what's wrong with autodialers? They're quite fun to work with. I set one up for a company in Spain and it was working fine for a long time. But then it must've called the king of Spain on a crapper or something, because out of a sudden the company went down. |
| 16:56.13 | Qwell | Everything. |
| 16:56.47 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:56.47 | morfin | haha |
| 16:56.57 | [TK]D-Fender | <tompaw> [TK]D-Fender: the newb you were trying to help went silent for a reason. <- Sure there's always a reason. Like being distracted by others. Falling asleep. Extended bathroom break. Zombie apocalypse. I'm not one to start guessing as to why. |
| 16:57.14 | drfreeze | not sure that means anything because I see the same message on the phones that are working |
| 16:57.16 | tompaw | I wish I could experience the Z-day for meself. |
| 16:58.49 | *** join/#asterisk wasabi (~wasabi@ubuntu/member/wasabi) |
| 16:59.23 | wasabi | Howdy. So, can one declare a peer but with a different extension name than that which will be used by the host during registration? |
| 16:59.37 | wasabi | For instance: Registration from '<sip:RtcApplication-62788b8e-ec69-49d7-a2ca-5a9dbd5f5254@ad.isillc.com>' failed |
| 16:59.47 | wasabi | The last thing I want to do is actually have to enter that a dozen times in extensions.conf |
| 17:00.23 | [TK]D-Fender | wasabi, What does your peer have to do with "extensions.conf"? |
| 17:00.58 | wasabi | When using Dial, don't you have to enter the name of the peer to dial with? |
| 17:02.54 | [TK]D-Fender | wasabi, what is an extension name where you are dialing? |
| 17:03.14 | wasabi | Huh? |
| 17:03.24 | wasabi | SIP/peername/extension? |
| 17:04.10 | morfin | is there way to monitor users statuses |
| 17:04.18 | morfin | like Away, Online etc? |
| 17:06.15 | [TK]D-Fender | wasabi, that is usually used for inter-server (ITSP, etc), style calling. Registration has nothing to do with that however. |
| 17:06.46 | wasabi | ... okay. |
| 17:06.54 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 17:07.05 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 17:09.18 | wasabi | Let me rephrase. I have an Asterisk box. It has three peers. It does nothing except route calls between the three peers, but only in certain allowable directions. Each peer has a context. Depending on the pattern of the number dialed, it gets sent to a specific next peer. That's all I'm doing. Some of those peers register. Some don't. My understanding was that to route a call to a peer, I simply needed to Dial(SIP/peername/${EXTEN}). So I'm doing tha |
| 17:09.49 | wasabi | And I'd rather it not be, as that name is silly and long. |
| 17:10.09 | Qwell | So don't use a silly and long name? |
| 17:10.30 | Qwell | or make it a variable? |
| 17:10.31 | wasabi | Cool. So how do I do that? |
| 17:10.36 | wasabi | A variable. That might work. |
| 17:11.39 | *** join/#asterisk af_ (~getsmart@78-134-67-168.v4.ngi.it) |
| 17:12.40 | [TK]D-Fender | wabYou have to name the peer.... |
| 17:13.19 | wasabi | Uh huh. My question was how do I name it somethign different from what it registers as |
| 17:13.32 | wasabi | And if I cannot, then that is the answer to my question. |
| 17:13.35 | [TK]D-Fender | wasabi, as for "silly and long".... if you have only 3 peers, why did you name them something silly and long? You could have called them A, B, and C, for all it matters. |
| 17:13.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:13.58 | [TK]D-Fender | wasabi, You don't name anything with a register |
| 17:14.32 | [TK]D-Fender | wasabi, username= <--- this is the username to match. "defauluser=" <- newer versions use this syntax |
| 17:14.36 | wasabi | Ahh! |
| 17:14.37 | [TK]D-Fender | defaultuser* |
| 17:15.02 | [TK]D-Fender | [whateveryouwantthistobe] defaultuser=thisnameislongandsilly |
| 17:17.47 | wasabi | Hmm. Still getting 'no matching peer found' |
| 17:19.14 | [TK]D-Fender | wasabi, that is an inbound call.. you were just asking about outbound (Dial) |
| 17:19.24 | [TK]D-Fender | wasabi, You seem to be reversing yourself. |
| 17:19.40 | [TK]D-Fender | wasabi, And should be showing us configs & debug if you're hoping for our input. |
| 17:19.52 | [TK]D-Fender | ~pb |
| 17:19.53 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:19.54 | [TK]D-Fender | ^ |
| 17:20.17 | wasabi | I'm pretty sure registration has a lot to do with placing outboudn calls towards that which has registered. |
| 17:21.47 | [TK]D-Fender | wasabi, The peer you dial out has no relationship to a register on the calling server |
| 17:21.56 | *** join/#asterisk aleek (~aleeksand@153.19.49.249) |
| 17:22.23 | wasabi | Then how does it determine what IP to send the INVITE to? |
| 17:22.33 | [TK]D-Fender | wasabi, HOST= |
| 17:22.39 | wasabi | And if it is dynamic? |
| 17:22.54 | [TK]D-Fender | wasabi, Then your description of which side the register was on is vague |
| 17:23.02 | [TK]D-Fender | wasabi, AAnd you're still getting a mismatch. |
| 17:23.11 | [TK]D-Fender | wasabi, Pastebin is your friend.... |
| 17:23.22 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 17:23.22 | wasabi | Some of my config: http://pastebin.com/fucSy19c |
| 17:23.29 | *** join/#asterisk navaismo (~Administr@189.191.20.105) |
| 17:23.47 | wasabi | I'm really just trying to get the registration message from the Lync machine to work, instead of just failing. |
| 17:24.31 | [TK]D-Fender | wasabi, so far we aren't even talking about "dialing" yes then. That part was distracting... |
| 17:24.43 | [TK]D-Fender | wasabi, You should be showing us debug to match this. |
| 17:24.44 | wasabi | I agree. Which is why I never mentioned it until asked. |
| 17:25.05 | wasabi | [Dec 17 11:17:34] NOTICE[12715]: chan_sip.c:25005 handle_request_register: Registration from '<sip:RtcApplication-62788b8e-ec69-49d7-a2ca-5a9dbd5f5254@ad.isillc.com>' failed for 'XXX.XXX.XXX.143:59348' - No matching peer found |
| 17:25.08 | [TK]D-Fender | <wasabi> Let me rephrase. I have an Asterisk box. It has three peers. It does nothing except route calls between the three peers, but only in certain allowable directions. Each peer has a context. Depending on the pattern of the number dialed, it gets sent to a specific next peer. That's all I'm doing. Some of those peers register. Some don't. My understanding was that to route a call to a peer, I simply needed to Dial(S |
| 17:25.08 | [TK]D-Fender | IP/peername/${EXTEN}). So I'm doing th <- no, this is what you asked just walking in the door |
| 17:25.21 | [TK]D-Fender | wasabi, pastebin the complete debug and do not filter it. |
| 17:25.28 | aleek | hello! I am researching open video conference server's. I need to implement one to my IMS network. I've found ConfBridge plugin for Asterisk. Is it working? Is it is dev or stable state? |
| 17:25.42 | [TK]D-Fender | wasabi, host=XXX.XXX.XXX.XXX <- and you CAN'T register. you specified a HOST. |
| 17:25.58 | [TK]D-Fender | wasabi, You were just mentioning "dynamic" before and clearly it isn't. |
| 17:26.07 | [TK]D-Fender | wasabi, You are giving us contradictory information. |
| 17:26.09 | wasabi | Fender, you are reading WAAAY to much into what I am saying. |
| 17:26.18 | wasabi | I never, at all, said that my situation was dynamicl. |
| 17:26.39 | [TK]D-Fender | <wasabi> And if it is dynamic? <- stop dropping misleading bits in. |
| 17:26.52 | wasabi | Because I'm trying to get how it functions clarified. |
| 17:27.07 | [TK]D-Fender | wasabi, Well you specified a host. They cannot register because of it. |
| 17:27.21 | wasabi | So that's it, a registration cannot be filtered by host? |
| 17:27.44 | [TK]D-Fender | wasabi, If you wish to RESTRICT the peer then use permit/deny |
| 17:27.59 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 17:28.27 | wasabi | Okay. Well, you answered my question. I'll just ignore the error message then. |
| 17:29.48 | [TK]D-Fender | wasabi, Your registering system may have an issue with thata... like not thinking it can pass you calls ever. |
| 17:30.16 | wasabi | It seems to work fine. I was only annoyed by the error message. Registration doesn't seem to actually have to succeed. |
| 17:31.25 | [TK]D-Fender | wasabi, I'd take that error message as a clear sign it didn't register.... |
| 17:31.35 | wasabi | uh huh. |
| 17:32.09 | Penguin | I'll make it clear: host=dynamic means the peer must register to you. host=ipaddr mean the peer is not allowed to register to you. |
| 17:32.56 | wasabi | Well, now that I understand that host=ipaddr means registration is simply not permitted, I know what the problem is. I'd like something that allows it to register anyways, but leaves it as unrequired, with a hard coded IP anyways. |
| 17:32.59 | [TK]D-Fender | Penguin, He seemed to have gotten that after my explanation. |
| 17:33.20 | wasabi | But since it doesn't seem to matter, and there seems to be no such option, I'm going to ignore it. |
| 17:33.25 | Penguin | host=dynamic and an ACL |
| 17:33.40 | [TK]D-Fender | wasabi, There is defaultip IIRC. |
| 17:34.01 | wasabi | Oh my. |
| 17:37.13 | wasabi | defaultip doesn't seem to do the proper thing with inbound calls. They go to the default context. |
| 17:37.35 | Penguin | Peer matching? |
| 17:38.08 | Penguin | If calls are not going to the context defined for the peer, it didn't match your peer definition. |
| 17:38.24 | wasabi | Uh huh. |
| 17:38.24 | Penguin | Are you using type=peer or type=friend? |
| 17:38.42 | [TK]D-Fender | friend <- |
| 17:38.43 | wasabi | I have tried both. |
| 17:39.04 | wasabi | It's unclear whether defaultip is compatible with 'friend', as the documentation for 'defaultip' states that only 'peer' is allowed. |
| 17:39.25 | [TK]D-Fender | wasabi, Then maybe that's what you should use. |
| 17:39.28 | Penguin | It is my guess that you have no reason to NOT use type=peer. |
| 17:39.47 | wasabi | Which, as mentioned, I have tried. And, as I also mentioned, it does not match the peer correctly then. |
| 17:39.58 | Penguin | Unless you have some special case where you have to match user name rather than host IP address, peer is usually the correct value. |
| 17:40.14 | Penguin | Does the host register to you? |
| 17:40.43 | wasabi | I don't care. |
| 17:40.52 | wasabi | I know the hosts's IP address. |
| 17:40.59 | Penguin | It matters. |
| 17:41.25 | Penguin | If the host registers, you will use host=dynamic if you want it to work right. |
| 17:41.44 | Penguin | If it doesn't register, you'll use the IP address if you want it to work at all. |
| 17:41.50 | wasabi | So I'd be trading a useless error message for a situation where registration is required, and yet need not be. |
| 17:43.26 | Penguin | I think trading an error message for correct functionality would be desired. |
| 17:44.08 | Penguin | If you don't want the peer to register to you, turn off registrations in it. |
| 17:44.20 | [TK]D-Fender | Penguin, Wasn't sure if "defaultip" was a fallback for inbound on host=dynamic |
| 17:44.43 | wasabi | I cannot turn off registrations in it. |
| 17:44.44 | wasabi | =) |
| 17:44.49 | Penguin | I know defaultip works with defaultport, but I haven't seen a use for those settings. |
| 17:44.53 | [TK]D-Fender | wasabi, then set it up right to accept the register |
| 17:45.10 | Penguin | It's a simple operation to allow the registration. |
| 17:45.18 | Penguin | host=dynamic. done. |
| 17:45.45 | Penguin | And once it registers, then calls from it will happily match the peer entry and you can send calls to it. |
| 17:47.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:47.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:47.54 | wasabi | There actually a defaultport equivilent? |
| 17:48.03 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 17:48.08 | Penguin | There is actually a defaultport parameter. |
| 17:48.09 | Molo | leifmadsen: you mentioned a public preview of the 4th edition. where should i be looking for that when it becomes available? |
| 17:48.22 | leifmadsen | ofps.oreilly.com |
| 17:48.42 | Molo | ok, that is what I thought. Thank you |
| 17:51.27 | [TK]D-Fender | Shouldn't need defaultport. port is already set explicitly. |
| 17:56.40 | *** join/#asterisk hdiogenes (~humberto@189.124.196.68) |
| 18:05.09 | cusco | hi |
| 18:05.20 | *** join/#asterisk parasitodelsur (~wtf@23.30.88.89) |
| 18:05.23 | cusco | what would be the most common reason for a call to hangup at 32 seconds? |
| 18:05.38 | cusco | sip call |
| 18:06.14 | wasabi | I had that problem with Lync. |
| 18:06.23 | WIMPy | No RTP? |
| 18:06.34 | cusco | but rtp is present in those first 32seconds |
| 18:06.35 | [TK]D-Fender | That'd be it |
| 18:06.36 | *** join/#asterisk vinhdizzo (~vinh@70.98.241.200) |
| 18:06.44 | cusco | Sent RTP packet to 10.10.10.34:12962 (type 00, seq 063964, ts 3351776, len 000160) |
| 18:06.54 | WIMPy | IIRC rtptimeout default to 30s +1s off-by-one-error. |
| 18:07.09 | WIMPy | Or no response to a SIP packet. |
| 18:07.15 | cusco | there is also sip |
| 18:07.18 | cusco | there is a BYE |
| 18:07.23 | cusco | just when the call hangs up |
| 18:07.29 | cusco | before that I read several 200 OK |
| 18:07.35 | WIMPy | From which end? |
| 18:07.47 | cusco | from the guy that keeps getting calls droped |
| 18:07.55 | cusco | Im calling him now, normally a queue calls him |
| 18:08.03 | [TK]D-Fender | cusco, Where is that send from?] |
| 18:08.16 | WIMPy | BTW: [sr]: You asked for a second, two hours ago? Did you get it working? |
| 18:08.20 | cusco | ah from asterisk's ip |
| 18:08.23 | cusco | there is no direct media |
| 18:08.35 | [TK]D-Fender | cusco, * would hang up on no INBOUND RTP..... |
| 18:09.10 | [TK]D-Fender | cusco, Why would * hang on its own transmission? |
| 18:09.47 | Penguin | narcissism? |
| 18:10.07 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:10.08 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:10.30 | cusco | [TK]D-Fender: wel yes that is what I am trying to figure out |
| 18:11.01 | [TK]D-Fender | cusco, Let us know when you've found it... |
| 18:11.14 | cusco | 12-17 18:10:47] VERBOSE[23345] chan_sip.c: Scheduling destruction of SIP dialog 'NThhZjQ0N2FiM2Q1ZDljMTFmOGIyY2JiOTVkNDRjMDM.' in 6400 ms (Method: ACK) |
| 18:11.32 | cusco | I got the BYE |
| 18:11.39 | cusco | érr |
| 18:12.05 | parasitodelsur | Penguin: Comcas is now allowing me to bring my own modem. |
| 18:12.40 | parasitodelsur | Going to do some research and see what I am goin for. |
| 18:14.13 | Penguin | parasitodelsur: I prefer Motorola SURFboard modems. |
| 18:14.44 | parasitodelsur | Penguin: you mention one in specific some time ago. |
| 18:16.15 | cusco | [TK]D-Fender: http://paste.debian.net/216820/ I still can't figure but I see a read from MY IP, emtpy |
| 18:17.01 | Penguin | If you need D3, I'd look at something such as the 6120 or 6220. |
| 18:17.16 | Penguin | The 6220 is a voice-enabled modem, so it has a battery backup built in. |
| 18:17.59 | [TK]D-Fender | cusco, Funny I only see half a call worth of debug there... |
| 18:18.26 | cusco | becuase there are many ongoing calls and its though to get it all |
| 18:18.46 | Penguin | I use a 6220 and I like it. |
| 18:19.13 | [TK]D-Fender | cusco, When we tell you you should be looking ... showing us that you aren't looking any making excuses about it isn't likely to get you far. |
| 18:19.22 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:19.49 | cusco | ok wait |
| 18:33.30 | cusco | [TK]D-Fender: the call is here somewhere, let me see where it starts http://62.28.187.252/geada/call.txt |
| 18:35.21 | wasabi | Hmm. I assume there is a way to set up a set of exten => statements that can be reused from multiple context's? |
| 18:36.07 | cusco | [TK]D-Fender: after the line 2012-12-17 18:21:02] VERBOSE[12229] chan_sip.c: |
| 18:36.20 | cusco | there is another line: 2012-12-17 18:21:02] VERBOSE[12229] chan_sip.c: |
| 18:36.30 | cusco | let me pastebinit to be able to state the line numbers |
| 18:36.50 | cusco | Length of code is not allowed to exceed 150kB gah |
| 18:37.22 | Penguin | If you mean multiple contexts, you have three options: include one context in another, duplicate the extensions into other contexts, or set up subroutines which utilize those extensions. |
| 18:38.23 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
| 18:39.00 | cusco | [TK]D-Fender: http://pastebin.com/3K15VA2L it starts at line 870 |
| 18:39.49 | wasabi | Didn't know you could include one context in another. |
| 18:40.34 | Penguin | [context2] |
| 18:40.34 | wasabi | Neato. |
| 18:40.36 | Penguin | include => context1 |
| 18:40.55 | Penguin | Now everything in context1 is available from context2. |
| 18:43.17 | [TK]D-Fender | cusco, Again, only half the call (if even that) in there. |
| 18:43.28 | [TK]D-Fender | moves on to other matters |
| 18:44.56 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 18:46.43 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 18:48.33 | cusco | are you sure? |
| 18:48.35 | cusco | :/ |
| 18:48.41 | cusco | it wa not intentional |
| 18:49.57 | *** join/#asterisk n8ideas (~joshua@65.112.207.46) |
| 18:50.26 | n8ideas | Can anyone tell me if/why periodic announcements don't play in realtime queues? Do I have field names wrong? |
| 18:51.33 | Qwell | We don't know - do you? |
| 18:53.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ithraakdfgqefloh) |
| 18:53.32 | cusco | n8ideas: they do, we here, use them |
| 18:54.57 | n8ideas | Would it be possible to grab a schema snapshot or let me know how you have those fields named? |
| 19:00.22 | cusco | n8ideas: http://paste.debian.net/216832/ |
| 19:02.19 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:03.40 | n8ideas | thank you so much |
| 19:04.19 | n8ideas | and what version are you on? |
| 19:11.50 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 19:11.51 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.108) |
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| 20:09.04 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 20:12.11 | *** part/#asterisk navaismo (~Administr@189.191.20.105) |
| 20:17.26 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 20:17.39 | Katty | HI LADS. |
| 20:19.52 | leifmadsen | Katty: HI MADAM |
| 20:20.12 | leifmadsen | fyi, in case anyone feels like doing an early review of Asterisk: TDG 4e it's online: http://ofps.oreilly.com/titles/9781449332426/index.html |
| 20:21.24 | leifmadsen | Molo: ^^^^ |
| 20:21.45 | WIMPy | leifmadsen: Are we supposed to bitch again? ;-) |
| 20:21.53 | leifmadsen | about what? |
| 20:21.58 | leifmadsen | the errors? yes :) |
| 20:22.17 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 20:26.26 | leifmadsen | https://twitter.com/leifmadsen/status/280768893386104832 |
| 20:26.31 | leifmadsen | be sure to retweet! :) |
| 20:32.14 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 20:32.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:32.32 | russellb | leifmadsen: you link here? |
| 20:33.06 | russellb | 4th edition of "Asterisk: The Definitive Guide", getting finished up this month, ready for review here: http://ofps.oreilly.com/titles/9781449332426/index.html |
| 20:33.15 | russellb | enjoy :) |
| 20:33.32 | tompaw | Is that book free? |
| 20:34.21 | tompaw | It either is or those previews are really long :F |
| 20:40.56 | russellb | :) |
| 20:41.02 | russellb | we make it available via a creative commons license |
| 20:41.12 | russellb | but you can buy it in print form (or ebook form) as well |
| 20:41.16 | russellb | if you like it, we appreciate it ;) |
| 20:41.27 | russellb | this is early access right now, it's not quite done |
| 20:41.33 | Qwell | russellb: You earn those nickels, yo. |
| 20:41.38 | russellb | yeah man |
| 20:41.56 | Qwell | Buy 30, and they get a beer (it's a bud light, but a beer all the same)! |
| 20:42.03 | russellb | that's right |
| 20:42.04 | russellb | better than nothing |
| 20:42.07 | Qwell | barely |
| 20:42.08 | tompaw | russellb: looks very nice, the structure of it. |
| 20:42.11 | WIMPy | And when looking at How to install, it seems to have a sponsor. |
| 20:42.15 | russellb | tompaw: thanks! |
| 20:42.27 | tompaw | Not very often you see free stuff of good quality available on .com domains these days. |
| 20:42.35 | russellb | WIMPy: hm? |
| 20:43.08 | WIMPy | Using dahdi is only one of many ways to get hardware going. |
| 20:43.28 | Qwell | There are 3 people that use anything else. |
| 20:43.31 | russellb | please leave comments on the site for that kind of stuff |
| 20:43.46 | russellb | not assuming it's perfect, that's why we have this web site for collecting feedbcak |
| 20:45.49 | tompaw | From my personal experience, I would like MeetMe to be replaced by ConfBridge in a book I'm readying. |
| 20:45.52 | tompaw | Reading even. |
| 20:45.59 | tompaw | MeetMe brings painful memories. |
| 20:46.31 | Qwell | Did v3 even document MeetMe? |
| 20:46.48 | leifmadsen | yes it did |
| 20:46.52 | leifmadsen | because it was 1.8 based |
| 20:46.55 | leifmadsen | so we didn't use confbridge |
| 20:46.55 | Qwell | oh |
| 20:47.03 | WIMPy | has successfully forgotten about the existance of MeetMe. |
| 20:47.04 | leifmadsen | this version of the book is confbridge preferred |
| 20:47.16 | leifmadsen | Qwell: as you know. v 10 had the new confbridge :) |
| 20:47.25 | leifmadsen | and we don't update the book against non-LTS |
| 20:47.32 | Qwell | leifmadsen: yeah, I forgot it was 1.8 |
| 20:47.34 | [TK]D-Fender | Does this version properly and simply cover basic NAT settings, etc? |
| 20:47.49 | Qwell | [TK]D-Fender: Have you written that section for them? :p |
| 20:48.09 | [TK]D-Fender | Qwell, I did. It was on my blog whose post had been used in here for years :) |
| 20:48.16 | [TK]D-Fender | about half a decade actually... |
| 20:48.23 | Qwell | submit it - it can't just be pulled from somewhere |
| 20:48.34 | tompaw | I clicked through a few chapters and I'm sure I noticed a few MeetMe's |
| 20:48.46 | [TK]D-Fender | They could have just written it themselves.... |
| 20:48.55 | [TK]D-Fender | Rather than copy mine verbatim... |
| 20:49.20 | [TK]D-Fender | This is probably the most common thing to see newbs struggle with walking in the door. |
| 20:49.22 | tompaw | NAT for Asterisk: don't use it, go VPN. |
| 20:49.25 | tompaw | :> |
| 20:49.37 | [TK]D-Fender | And should be the first thing they see when reading about SIP in the book. |
| 20:50.11 | leifmadsen | well we still talk about MeetMe |
| 20:50.16 | leifmadsen | we didn't just ignore that it exists |
| 20:50.33 | WIMPy | Unless you know about the NAT stuff, you can easily shoot yourself in the foot by using a VPN. |
| 20:50.37 | leifmadsen | plus people upgrading or maintaining older systems will still have to interact with MeetMe |
| 20:50.41 | leifmadsen | WIMPy: +1 |
| 20:50.48 | leifmadsen | VPNs are not the solution |
| 20:51.30 | WIMPy | they can help, but unless you know how things work, it can also become worse. |
| 20:51.33 | tompaw | WIMPy: how come? I had all sorts of issues with MeetME/ConfBridge @ 1.8 and 10.x, like dead cli, dead monitor, dead asterisk, dead server (sic!). |
| 20:52.06 | tompaw | Switching everyone to nat=no and setting up a proper ipsec network in the call centres sorted it out like a magic touch! |
| 20:52.12 | leifmadsen | http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html#ch07_network_address_translation |
| 20:52.59 | WIMPy | I'n not sure how NAT issues could kill your Asterisk. |
| 20:53.21 | leifmadsen | this was the sort of positive and useful discussion I was hoping for! |
| 20:53.49 | tompaw | I don't either. Last summer I spent months here trying to find out. Even recently with 10.x and confbridge I experienced the same (like CLI waiting for some shit to time out and being dead in meantime). |
| 20:54.03 | tompaw | With the NAT gone, so are ALL these issues. |
| 20:54.23 | WIMPy | That sounds very strange to me. |
| 20:55.06 | WIMPy | I've had lots of deadlocks in the 1.4 and especially 1.6 days. But since then it has been working quite well. |
| 20:55.12 | tompaw | Not to mention the client (Zoiper/Bria) end. Same story. Some stacks of software dying, requiring OS restart or at least a process kill. I honestly don't know which parts and I don't even care. |
| 20:55.13 | WIMPy | Unless I try to use skinny. |
| 20:55.22 | tompaw | No NAT == no problems. |
| 20:55.37 | WIMPy | OTOH it is good to have a way to kill it for testing :-) |
| 20:56.16 | WIMPy | Just wait a few decades and the VOIP stuff might become stable as well :-) |
| 20:56.32 | tompaw | It is now! Because it doesn't use NAT ;) |
| 20:58.44 | *** join/#asterisk lorsungcu (~anonymous@67.136.169.134) |
| 20:58.44 | WIMPy | As far as zoiper goes, I had to switch it from IAX to SIP because of audio issues depending on the bridged channeltype. |
| 20:58.44 | WIMPy | Got absolutely NFI what's going on there. |
| 20:58.44 | tompaw | Got absolutely NFReason to even consider IAX ;) |
| 21:00.09 | WIMPy | IAX is surely inspiring a lot more confidence. |
| 21:03.24 | tompaw | So is the season finale of Dexter, speak laters! |
| 21:18.15 | *** join/#asterisk jsjc (~Adium@229.Red-2-136-114.dynamicIP.rima-tde.net) |
| 21:25.06 | [TK]D-Fender | checkout time, BBIAB |
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| 21:50.36 | Claies | anyone that is an expert with dial plans? |
| 21:51.50 | Claies | I have a dialplan that looks like exten => _91NXXNXXXXXX,2,Dial(${MEGAPATH}/${EXTEN:1},,tTor) |
| 21:52.38 | Claies | it is executing as Dial("SIP/8011-0000005a", "SIP/918136900063@megapath||To") which seems correct |
| 21:53.24 | Claies | however, my provider is needing the phone number passed instead of a station id, and I can't see where the dialplan specifies the first half of the output |
| 21:53.57 | Claies | basically I can't seem to figure out where to change the "SIP/8011-0000005a" part of this output |
| 21:55.15 | Claies | anyone have any ideas? |
| 21:56.33 | tzanger | you don't change that part of the output |
| 21:56.42 | tzanger | it sounds like your provider wants a specific SIP From: header |
| 21:56.53 | tzanger | and that's not so much the dialplan as it is the peer's sip.conf entry |
| 21:57.38 | tzanger | I have to go, but it would be helpful if you pasted the relevant (and scrubbed of login/password and IP information) parts of sip.conf and a sip debug output on pastebin |
| 21:57.46 | Claies | ok is that a setting that can be entered from account entry variables? |
| 21:57.56 | tzanger | just replace the login name with LOGIN and the password with PASSWORD and the ip with IPADDR or something so that we can clearly see what was replaced |
| 21:58.16 | tzanger | would also be helpful to state who the provider is and if calls ever worked through them |
| 21:58.24 | Qwell | megapath, I'm sure |
| 21:58.25 | tzanger | anyway, I ahve to run, hopefully that info will help someone else here to help you |
| 21:58.45 | tzanger | Qwell: nonsense. I regularly name my extensions to throw off those who will come after me. :-) |
| 21:59.39 | Claies | I'm not editing the sip.conf directly, I have a carrier entry with account entry lines |
| 22:00.44 | Claies | so it sounds like I'm missing an account entry line, but I can't seem to identify what it might say |
| 22:01.20 | Claies | I'll do a bit more research and if I can't come up with anything I'll set up a pastebin with the debug |
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| 22:53.37 | skirmisha | guys |
| 22:53.55 | skirmisha | is it possible to use sin_addr in pbx.c ? |
| 22:55.37 | [TK]D-Fender | patience[-1] |
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| 23:24.30 | zopsi | has anyone worked with digium D70's? I'm looking for a way to create a custom app that displays speech to text info on the screen (for captioned calls for people with hearing loss). |
| 23:29.57 | [sr] | WIMPy: its working with PtP, anyway sometimes i get a PRI Error on span 3: Received MDL/TEI managemement message, but configured for mode other than PTMP! |
| 23:30.00 | [sr] | WIMPy: any idea? |
| 23:30.12 | [sr] | WIMPy: everything seems to be working fine anyway |
| 23:33.33 | *** join/#asterisk frek818 (~Adium@111.68.38.82) |
| 23:35.06 | WIMPy | [sr]: Looks like the other end is ptmp. |
| 23:39.09 | WIMPy | That was a hell of a second, BTW. |
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