00:06.31 | jpsharp | You have to pay for it somehow. Either by putting up with ads or paying a per month/per minute charge. |
00:18.08 | jgowdy | You have to pay to reach the PSTN |
00:18.19 | jgowdy | If you want to call someone else in a peer to peer fashion |
00:18.37 | jgowdy | You can do that for free using just Internet |
00:18.48 | jgowdy | But that's not calling a phone number on the PSTN |
00:19.09 | jgowdy | That's starting a SIP session with another IP address, followed by the associated RTP stream |
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00:34.38 | gusto | ok |
00:34.44 | gusto | this g722 is really cool |
00:35.10 | gusto | i thaught it were bad, but then i noticed, that i still had echo cancellator on and that is famous for worsen the quality |
00:35.27 | gusto | but now with disabled echo cancellator it's very good |
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00:39.02 | gusto | one can listen radio through it |
00:39.36 | gusto | i am testing codecs now that way, i have a line-out from a radio (dab+) and i am looping it back through asterisk with app_echo |
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01:52.58 | ZenMaster | So, people who use TrixBox, use something like Vici dial to create a outbound campaign? |
01:54.02 | ZenMaster | I am just testing and learning. I have a TrixBox machine setup, and wanted to give setting up a predictive enviroment a go. Then eventually live call transfers. |
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01:56.17 | darrenlooby | If my asterisk is running from a database. And I'm using the queues table - how do I get the AgentCalled event to trigger? It doesn't seem to be doing it. |
01:56.33 | darrenlooby | I'm in 1.8.7.1 |
02:16.25 | ZenMaster | My other question is, would a crm allow agents to dispositio calls and what not? Lead, Hungup, DNC? |
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03:06.30 | UnixDev | is there some way to use page without dahdi on asterisk 1.8? I see this has been done in asterisk 11, but I don't think its ready for prime time yet |
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03:31.28 | UnixDev | what version of asterisk would this commit effect? http://svnview.digium.com/svn/asterisk?view=revision&revision=358730 ? |
03:37.20 | jpsharp | Asterisk 11 eventually. Or "trunk" if you pull the source tree from SVN. |
03:41.23 | UnixDev | ahh, got it |
03:41.32 | UnixDev | any chance that can be back ported to 1.8 ? |
03:51.49 | jpsharp | Looks to be possible, but I can't guarantee it. |
03:52.00 | ZenMaster | Anyone have expierence with astercc? |
03:52.42 | UnixDev | jpsharp: I can back port it if it will go into trunk… another option is to create app_pageconf |
03:53.04 | UnixDev | jpsharp: that way it would not break existing installations using 1.8… what do you think? |
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03:54.52 | jpsharp | ZenMaster: I used it a few years back and it was kinda kludgey, but that was a few years ago. So, grain of salt and all that. |
03:55.25 | ZenMaster | I was just trying to set something up for prdictive with trix box as a test. |
03:55.34 | ZenMaster | I got it to install, and now I can't login wierd. |
03:55.38 | jpsharp | UnixDev: If you're worried about breaking existing installations, I'd go with the separate app. |
03:55.40 | ZenMaster | Might be a sign for me to leave this office. |
03:55.58 | ZenMaster | I just want to see what it looks like. |
03:56.00 | UnixDev | jpsharp: I'm just worried about getting the patch into trunk, I dislike maintaining my own patchsets |
03:58.08 | ZenMaster | Going home. |
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04:25.11 | linocisco | hi all |
04:25.51 | linocisco | i have Dlink DI-524 wifi router. Can I get asterisk firmware for that?? most said WRT54G of Linksys is compatible with asterisk firmware but I dont know about my Dlink router |
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05:02.47 | kaldemar | linocisco: asterisk is not a firmware. if for example openwrt supports your router, you can run openwrt on it and install asterisk in openwrt. |
05:03.12 | linocisco | kaldemar, how can I check? |
05:03.51 | kaldemar | openwrt project has a device list on their web page |
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05:39.49 | KNERD | how to get the jabber and gtalk modules to load upon start? |
05:43.54 | kaldemar | KNERD: you've done something that prevents them from loading. see that you don't have noloads in asterisk's modules.conf. |
05:44.57 | KNERD | okay..let me look.. |
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05:46.18 | KNERD | kaldemar: yeah, that was it..thanks |
05:46.40 | As001 | Hello, 4 of my agent channels hang in cli in state down and Application none at cli. I can not hangup them with hangup request. I restarted Asterisk and they have gone away but I wonder why would some channel get in that state ? |
05:47.14 | As001 | All of them had elapsed time unbelivable big like 200 hours etc... |
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06:02.45 | Jo_Bell | Hi. If I set the destination of 2 audio devices to be a single SIP device URI, will asterisk mix the audio and send it in a single channel? |
06:02.50 | Auckla | Anyone awake tonight? :D |
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06:11.09 | kaldemar | Jo_Bell: asterisk will only audio from two channels if you connect the channels to a conference. |
06:11.30 | kaldemar | s/only/mix/ |
06:12.42 | bulkorok | hi |
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06:28.26 | Auckla | Can't seem to get astercrm to allow me to login via web interface? |
06:28.35 | Auckla | Anyone have any expierence with that? |
06:42.09 | KNERD | Is there a CURRENT method to setup GV? It seems the ones a year or older just outright fail, and I cannot find one newer |
06:43.49 | Auckla | I just don't get it. :&*( |
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06:53.20 | ChannelZ | Well the GV integration isn't very old relatively speaking |
06:56.08 | KNERD | well, it has not functioned in the past 1.5 years..I am trying again...still fails |
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07:06.16 | Jo_Bell | infboot and kaldemar: ok got the idea |
07:07.25 | kaldemar | infobot is a bot, not a person. |
07:15.20 | Jo_Bell | kaldemar: fine. :-) |
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07:23.32 | khurram | hi every one |
07:24.08 | khurram | a newbie just need initial understanding about asterisk.. |
07:24.49 | khurram | do i have to have some telephone line connection to run asterisk server |
07:27.36 | Samuel_CTX | khurram: no, you can use everything with SIP phones (and trunks to VOIP providers) |
07:28.20 | khurram | VOIP provider so i need to have VOIP service first from some one right ? |
07:29.10 | khurram | how can i know if my table phone is SIP phone ? do i require to buy hardware to convert this to SIP ? |
07:29.19 | kaldemar | khurram: no. you don't need any service or special hardware unless you want to connect to public telephony network. |
07:29.59 | kaldemar | khurram: what is your table phone? |
07:30.11 | khurram | the normal landline phone |
07:30.39 | kaldemar | then it is not a VoIP phone. |
07:31.17 | kaldemar | but you could get an ATA that speaks SIP to asterisk and lets you connect an analog telephone to it. |
07:31.33 | khurram | ATA ? |
07:31.42 | kaldemar | ~ata |
07:31.42 | infobot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
07:32.28 | kaldemar | FXS is for connecting phones, FXO is for connecting to an analog line. |
07:34.22 | khurram | <PROTECTED> |
07:34.31 | khurram | hmmm |
07:36.17 | khurram | so if some dials my extention then i can programme Asterisk to pick the phone after 5 dial tones and play the message i have recorded and then email me or sms me about that i have missed a call on my landline from bla with recorded mesg bla bla. Am i right? |
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07:39.17 | khurram | kaldemar: thanks i watched the video and it is quite impressive but i think the turorials are missing for new developers like me from php background |
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07:41.40 | kaldemar | the video? the tutorials? |
07:43.21 | kaldemar | ~book |
07:43.22 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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08:10.02 | *** join/#asterisk bikcmp (jason@april-fools/winner/thesecondest) |
08:11.38 | bikcmp | i'm configuring asterisk 1.8 on a debian server. |
08:11.40 | bikcmp | outgoing calls work fine. |
08:11.42 | bikcmp | incoming calls are either "nobody hears anything", or "caller hears person but can't speak" |
08:11.51 | bikcmp | i'm almost positive it's NAT. |
08:12.23 | bikcmp | <PROTECTED> |
08:12.25 | bikcmp | eout reached on transmission 6a966a664f191e641e0ed79b03f0f85c@66.54.140.46 for s |
08:12.30 | kaldemar | ~sipnat |
08:12.30 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
08:12.59 | bikcmp | yeah. |
08:13.03 | bikcmp | kaldemar: we've been trying that. |
08:13.18 | kaldemar | what exactly? |
08:13.26 | bikcmp | kaldemar: give me 10 secs, i'll pull the config out on you |
08:14.21 | darrenlooby | If my asterisk is running from a database. And I'm using the queues table - how do I get the AgentCalled event to trigger? It doesn't seem to be doing it. |
08:16.15 | bikcmp | kaldemar: okay |
08:16.18 | bikcmp | kaldemar: i've enabled nat=yes |
08:16.36 | bikcmp | kaldemar: set qualify=yes |
08:16.57 | bikcmp | kaldemar: localnet=192.168.1.1/255.255.255.0 |
08:16.59 | kaldemar | ~pb |
08:16.59 | 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. |
08:17.04 | bikcmp | kaldemar: that'll work then |
08:17.11 | bikcmp | kaldemar: how much you want? |
08:17.13 | bikcmp | the whole file? |
08:17.25 | bikcmp | sip.conf, that is? |
08:17.41 | kaldemar | [general] and the peer for ipkall |
08:18.15 | kaldemar | a sip debug of a call would tell more about what is wrong. |
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08:20.18 | bikcmp | kaldemar: mind if i pm it? |
08:20.34 | bikcmp | i think that i removed the secrets/passwords, but just in case. |
08:21.12 | kaldemar | go ahead |
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08:22.23 | kaldemar | externaddr/externip is missing from your configs. |
08:23.41 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.202) |
08:23.53 | kaldemar | also, change the provider entry from fried to peer. |
08:24.35 | kaldemar | # is not a comment character in config files, use ; with canreinvite=no. |
08:24.47 | ziz212 | hi friends, how could we mute all except one who speak in a conference call ? |
08:24.53 | bikcmp | ok |
08:25.19 | bikcmp | kaldemar: ok |
08:25.24 | bikcmp | kaldemar: so what should canreinvite be? |
08:25.30 | bikcmp | nonat? |
08:26.08 | kaldemar | more like "no" in your case. |
08:26.56 | kaldemar | also, put your register statements under above all peer definitions or they will not work. |
08:27.24 | bikcmp | put the register statements? |
08:27.30 | bikcmp | kaldemar: what do you mean by register statements? |
08:27.31 | kaldemar | they belong under [general] |
08:27.55 | kaldemar | the "register => ..." lines |
08:28.18 | bikcmp | ok |
08:28.28 | bikcmp | now, do you still need a log? |
08:29.12 | bikcmp | kaldemar: where should externip be |
08:29.26 | kaldemar | externip/externaddr goes under [general]. |
08:29.37 | kaldemar | the sample sip.conf is very helpful with these things. |
08:29.54 | bikcmp | kaldemar: yeah |
08:30.06 | bikcmp | kaldemar: just trying to double check stuff, since i've literally spent hours on this. |
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08:30.25 | bikcmp | externaddr isn't needed if externip is given |
08:30.26 | bikcmp | right? |
08:32.28 | kaldemar | externip is the old name for externaddr. their meaning is the same. i'd use the latter. |
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08:33.45 | bikcmp | kaldemar: ok |
08:33.49 | bikcmp | kaldemar: just modified what you told me |
08:33.50 | bikcmp | still nothing |
08:33.52 | bikcmp | same issue. |
08:33.58 | bikcmp | i have the sip debug logs, but god there are a lot |
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08:35.15 | kaldemar | pastebin your current configs and the output |
08:35.25 | bikcmp | sure |
08:35.33 | bikcmp | well |
08:35.34 | bikcmp | darn |
08:35.39 | bikcmp | kaldemar: here's the problem. |
08:35.55 | bikcmp | kaldemar: the `sip set debug` command actually put out /so much output/ that it's no longer in my backlog. |
08:36.57 | bikcmp | kaldemar: want just the config? |
08:38.05 | *** join/#asterisk din3sh (~din3sh@41.212.203.212) |
08:38.10 | din3sh | hello all |
08:39.46 | kaldemar | start with the config. if you scrollback is not enough, use "asterisk -vvvr | tee /tmp/sip_debug.txt" for example to get it to a file. |
08:40.30 | bikcmp | kaldemar: i'll throw you the config in hopes you see the problem |
08:42.42 | bikcmp | kaldemar: pm. |
08:42.57 | din3sh | anyone ever implemented click-to-call on lotus notes working with asterisk? |
08:44.01 | bikcmp | kaldemar: you get my pm's? |
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08:50.20 | ziz212 | can someone hlep me in my question? Can I repost my question? |
08:54.29 | din3sh | ziz212: dont ask if you can ask a question dude |
08:54.45 | din3sh | post your question, if someone can help, he will |
08:55.15 | ziz212 | ok i will repost it.. hi friends, how could we mute all except one who speak in a conference call ? |
08:56.36 | din3sh | what if everybody is talking at the same time |
08:56.40 | din3sh | how will you decide which one should be talking, and which muted? |
08:57.33 | kaldemar | ziz212: which conference app are you using? |
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09:15.01 | Lantizia | Lo, for some reason *8 group call pickup isn't working - it just hangs up (both peers are in the same call/pickup group), if I turn on sip debug on the peer dialing *8 I see it does dial *8@<ip of pbx> in the To: line - but ultimately then checks if *8 is in from-internal context... but it's an internal function isn't it?! (yes pickupexten is set to *8 in features.conf). Any ideas? |
09:18.23 | Lantizia | I can provide logs! Theres like nothing on the web I can find about this - this is day 4 of me asking in this channel |
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09:19.50 | wdoekes | Lantizia: perhaps features.conf / res_features is not loaded properly? check the core show features (or something) and make sure it lists *8 |
09:20.15 | Lantizia | nah it is - i watched asterisk start up this morning ... both the app and the config files are loaded |
09:20.16 | kaldemar | "features show" |
09:20.42 | Lantizia | yeah it shows |
09:20.49 | wdoekes | (b.t.w. the relevant bit is *8@<ip> in the request-uri line (the first one, after INVITE)) |
09:21.18 | Lantizia | umm i think i may have found it - they're in call/pickup group 71 - is the max 63? |
09:21.23 | bikcmp | s/53 |
09:21.24 | bikcmp | 4/54 |
09:21.30 | wdoekes | yes it is |
09:21.44 | Lantizia | (btw I've been saying they're in 71 since thurs in here and no one has picked up on it :P) |
09:21.48 | Lantizia | why 63?! of all numbers |
09:21.55 | wdoekes | bits |
09:22.02 | Lantizia | is 0 allowed? |
09:22.08 | wdoekes | probably |
09:22.14 | Lantizia | non-integers? :P |
09:22.29 | wdoekes | floats? no |
09:22.36 | Lantizia | damn |
09:22.40 | wdoekes | 0.71 |
09:22.54 | Lantizia | 0.71 ? |
09:22.57 | wdoekes | ;) |
09:23.05 | Lantizia | yeah a non-integer :P |
09:25.26 | Lantizia | see our system is multi-tennant - the callgroup number is the room number |
09:25.32 | Lantizia | from about 1 to 200 ish rooms |
09:25.44 | Lantizia | each room a business - wish we could increase the number of call groups |
09:27.00 | wdoekes | Lantizia: see the Pickup application and friends |
09:27.21 | Lantizia | oh i know i could do it with dialplan - just a load of work is all |
09:27.45 | wdoekes | perhaps it's easier if you just decrease the number of rooms |
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09:31.26 | Jo_Bell | All. Where can I get 1 / a few examples on getting a number of SIP devices into a conference using Asterisk? |
09:31.44 | Lantizia | wdoekes, how about if I put each tenant in their own context? do you think it's possible to write a *8 function that'd only pick up the oldest call in that context? |
09:32.46 | kaldemar | Jo_Bell: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
09:35.02 | wdoekes | Lantizia: PICKUPMARK sounds like what you want.. |
09:40.52 | Lantizia | wdoekes, err reading - not altogether following but reading |
09:41.17 | Lantizia | this looks like it has nothing to do with contexts though |
09:43.02 | wdoekes | I didn't say that. contexts are used when the device calls.. you could set the to_be_picked_up_mark there, but you could do that using a setvar= on the device too |
09:43.31 | Lantizia | bleh?! |
09:43.34 | wdoekes | you'd still need to set the right pickupmark on incoming calls |
09:44.55 | ziz212 | I am using elastix and I am trying to find out conference module in elastix? I really having problem to find out the name of it |
09:45.19 | wdoekes | ~elastix |
09:45.19 | infobot | rumour has it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
09:46.49 | ziz212 | Ok pls let me now the conference module to try for my feature? |
09:48.05 | ziz212 | Let say if I am using asterisk without any other add-ons |
09:48.29 | Lantizia | wdoekes, ok i'm reading about another idea - something about as of * 1.6.2 chan_sip uses the PickUp app by putting a hint in the context? but it doesn't say much more |
09:48.40 | Lantizia | something about notifycid |
09:49.01 | kaldemar | ziz212: meetme, confbridge, app_conference or app_konference. all have different syntax. ask the elastix people. |
09:50.35 | ziz212 | ok thanks for the help but how could I use that unmute for active spaker and mute or others? Which module support it? |
09:51.07 | Lantizia | he aint giving up is he? :D |
09:53.39 | kaldemar | yes, it supports it. |
09:56.50 | wdoekes | ziz212: meetme has MeetmeMute etc.. manager commands |
09:57.26 | ziz212 | Thanks for the help. I can try out that and see |
09:58.04 | Lantizia | wdoekes, think PickupChan may help me? |
09:58.27 | wdoekes | Lantizia: if I give you a solution that I think works, why are you trying to find others? |
09:59.00 | Lantizia | i'm not sure it does - or at least I don't understand it if it does |
09:59.05 | wdoekes | pickupchan might work if you your room-people share the same prefix [room71-account1] |
09:59.32 | Lantizia | nah that's not always true |
10:00.19 | Lantizia | ok so pickupmark - are you saying when a call comes in and a peer starts ringing - a special pickupmark variable is set for that one instance of it ringing? |
10:00.45 | Lantizia | and the *8 dialplan would grep all pickupmark's in the list for those starting with particular digits that correspond to that context? |
10:01.29 | wdoekes | Set(_PICKUPMARK=71) ; Dial(SIP/room71-account1) |
10:01.47 | Lantizia | then later *8 does what? |
10:01.50 | wdoekes | *8 => Pickup(${to_be_picked_up_mark}@PICKUPMARK) |
10:02.09 | Lantizia | what is 'to be picked up mark' ? |
10:02.13 | wdoekes | where setvar=to_be_picked_up_mark=71 in [room71-account2] |
10:02.56 | Lantizia | so pickup 71@71 ?! |
10:03.13 | wdoekes | I will leave you with 'core show application' and google |
10:03.22 | Lantizia | yeah that still makes zip all sense |
10:03.39 | Lantizia | and whats more doesn't record how old a call is - it won't pick up the oldest one |
10:05.20 | Lantizia | is PICKUPMARK a list? thus when it picks up 71@PICKUPMARK - it finds the earliest 71 in that list and removes it when picked up? |
10:05.44 | Lantizia | and presumably when that call ends (if not picked up with *8) it removes it from the list also? |
10:06.29 | wdoekes | so many questions |
10:06.32 | wdoekes | have you tried it? |
10:06.43 | Lantizia | am I at least CLOSE with that line of thinking? |
10:07.01 | wdoekes | I don't know about the pickup order, the rest looks ok |
10:07.16 | Lantizia | so it is a list variable then? |
10:07.51 | wdoekes | PICKUPMARK in @PICKUPMARK isn't a variable, technically |
10:08.03 | wdoekes | but you don't need to know that |
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10:11.12 | Lantizia | so perhaps I could set accountcode in each peer to be like a 'customer number' (usually room number) and put something generic in somewhere to use accountcode as pickupmark |
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10:37.41 | vahonen | Hi all. I'm running Asterisk 1.6.2.20 on CentOS 5.8, with 2.6.18-308.8.2.el5xen. When trying to load app_meetme.so, that came when installing asterisk-dahdi, I get this; "Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: undefined symbol: __ast_cond_wait". Any hints...? |
10:42.45 | din3sh | anyone ever implemented click-to-call on lotus notes working with asterisk? |
10:48.31 | Jo_Bell | in my /etc/asterisk, i don't see confbridge.conf. Do I have to install anything more ? |
10:49.00 | wdoekes | vahonen: then that app_meetme wasn't compiled for the right asterisk version |
10:50.45 | kaldemar | Jo_Bell: what version are you using and how did you install asterisk? |
10:53.16 | Jo_Bell | kaldemar: i got the tar files for libpri, dahdi and asterisk from site and did make install on all |
10:53.31 | Jo_Bell | how to check the version from the CLI prompt? |
10:53.57 | Jo_Bell | it shouldnt be any old version. i took the packages in the last 10 days or so. |
10:56.25 | kaldemar | core show version |
10:57.13 | Jo_Bell | kaldemar: 1.8.13.0 is the version |
10:58.04 | kaldemar | look under configs/ in the source package. |
10:58.16 | kaldemar | "make samples" is used to copy the sample configs under /etc/asterisk |
10:58.48 | kaldemar | see the README |
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11:00.10 | kaldemar | the confbridge app in asterisk 10 is more advanced than the one in 1.8. |
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11:02.05 | kaldemar | Jo_Bell: looks like the confbridge in 1.8 does not even use confbridge.conf. it is entirely different. |
11:05.35 | Jo_Bell | kaldemar: oh. for upgrading to asterisk 10, what should I do? |
11:08.24 | Jo_Bell | can i just take the asterisk 10 tar file and do a make install? |
11:08.45 | kaldemar | do a make uninstall in the 1.8 source dir first. |
11:09.55 | kaldemar | grab the tar, exctract it, run the configure script, make menuselect to select features (if you know what you want), make, make install. |
11:10.27 | Jo_Bell | i have 3 folders one for asterisk, one for dahdi, and one for libpri. i assume i have to uninstall only the asterisk..is it correct? |
11:11.19 | kaldemar | yes |
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11:13.55 | Jo_Bell | kaldemar: where to get the 10 source tar? |
11:14.02 | Jo_Bell | site itself? |
11:15.06 | kaldemar | http://www.asterisk.org/downloads |
11:15.24 | din3sh | is certified asterisk free? |
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11:17.27 | kaldemar | din3sh: http://www1.digium.com/en/products/asterisk/certified-asterisk |
11:20.24 | din3sh | thnks |
11:21.50 | Lantizia | yes but it's kind of pointless |
11:22.04 | Lantizia | i.e. pointless if all you need is something free :) |
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11:31.10 | vahonen | "Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: undefined symbol: __ast_cond_wait", solved. Asterisk was version 1.6, but meetme (asterisk-dahdi installed with yum) was 1.8. Removed a few repos and reinstalled asterisk-dahdi and now it's working. |
11:31.32 | kaldemar | Lantizia: it's an LTS version with digium phone support and it goes through more testing than the mainline releases. |
11:32.34 | Lantizia | kaldemar, i already know what it is |
11:32.54 | Lantizia | i spoke at length about it with digium - time I'll never get back |
11:34.32 | Lantizia | it's only purpose is for a completely custom production-level sytem - itsp grade - and you've got BAGS AND BAGS of cash for support |
11:35.14 | Lantizia | using it on an every day phone system is nuts - cheaper (if you want support from digium) to use the regular LTS |
11:39.00 | Jo_Bell | kaldemar: towards the end of configure, i see that many checks are saying no starting from "checking for resample_open in -lresample", Similarly ptlib.h usability and presence returned no. checks for sqlites also returned no.. |
11:39.09 | Jo_Bell | any corrective measures needed? |
11:40.22 | kaldemar | Jo_Bell: at least sqlite3 development files are something you do need. |
11:41.58 | Jo_Bell | package name? sqlite3-src? do you know? |
11:44.14 | kaldemar | Jo_Bell: what OS are you on? |
11:47.07 | Jo_Bell | ubuntu.. i use apt-get |
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11:47.25 | kaldemar | Jo_Bell: libsqlite3-dev |
11:47.31 | Jo_Bell | thanks |
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12:30.39 | Jo_Bell | kaldemar: when I did a fresh configure, make and make install. I am not getting any conf files in /etc/asterisk as I used to get with previous version of asterisk. |
12:30.48 | Jo_Bell | should I do make samples to get those? |
12:32.01 | kaldemar | you should have the old ones under /etc/asterisk unless you removed them. "make samples" will put the sample config files there, yes. |
12:34.09 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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13:01.15 | bulkorok | can somebody help with res_fax!? I have to send a CED in T38 but asterisk never sends it on ReceiveFax... |
13:01.35 | s[x] | Hey all, Anyone here tied a Samsung OfficeServ into an Asterisk box |
13:02.35 | [TK]D-Fender | s[x], Got another question about it? |
13:02.43 | [TK]D-Fender | s[x], Considerig just how low your odds are |
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13:03.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
13:03.35 | s[x] | [TK]D-Fender: Tell me about it, Figured id take the shot. Was purely hoping to extend to the SIP client base using a trunk with asterisk instead of direct communication with the PBX |
13:03.55 | s[x] | SIP Client Licenses are $250 a pop on the Samsung |
13:04.14 | [TK]D-Fender | s[x], Ok, that's your goal, it still isn't a question. |
13:04.36 | s[x] | [TK]D-Fender: Well for starters, is it at all possible |
13:04.54 | [TK]D-Fender | s[x], They both talk SIP apparently... so "why not"? |
13:04.55 | Chainsaw | Who drained all the anger out of Fender? |
13:05.46 | s[x] | [TK]D-Fender: Well how would one go about Client on Asterisk box talking to Phone on PBX |
13:06.12 | [TK]D-Fender | s[x], have you read the manual for your PBX? Start there |
13:06.21 | [TK]D-Fender | s[x], *'s isde is just like everything else |
13:06.25 | [TK]D-Fender | side |
13:08.53 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
13:12.22 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
13:12.26 | v0lZy | hello |
13:12.43 | v0lZy | i have a rather silly question i think |
13:13.13 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
13:13.48 | v0lZy | I have this situation where I'm using VoIP as my main line and classic ISDN as a backup line |
13:14.17 | v0lZy | However, this backup ISDN line is also my main phone number for incoming calls |
13:14.44 | v0lZy | Now I'm at a risk that whenever my internet drops and my users switch to dial outside our company with the ISDN provider... |
13:15.11 | v0lZy | they clog up my ISDN line with outgoing calls |
13:15.23 | v0lZy | which then prevents incoming calls... |
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13:15.53 | v0lZy | now I know I can disable outgoing calls |
13:16.10 | v0lZy | but can I restrict the amount of channels for outgoing calls to make reservations for incoming calls? |
13:16.21 | v0lZy | like.... max 2 outgoing calls at the same time |
13:16.25 | v0lZy | can this be done? |
13:16.32 | kaldemar | v0lZy: GROUP functions |
13:17.30 | WIMPy | That's whay call waiting is a really important feature. |
13:18.02 | v0lZy | what does call waiting do? |
13:18.28 | v0lZy | I mean... im talking about ISDN lines which i need to keep free so people can dial in and out |
13:18.32 | WIMPy | It will give you the incomming calls even if all channels are in use. |
13:18.44 | v0lZy | how can it do that? |
13:19.08 | WIMPy | The most basic feature. |
13:19.52 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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13:20.20 | [TK]D-Fender | v0lZy, "core show function GROUP" <------- |
13:20.46 | [TK]D-Fender | v0lZy, COUNT how many you want to allow out in that scenario and choose when to cut them off |
13:23.01 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.13) |
13:23.02 | v0lZy | I have no idea how to use this |
13:23.15 | v0lZy | It just give me a description of what it is |
13:23.52 | v0lZy | I imagine theres a config file i should edit? |
13:24.16 | WIMPy | Yes. Your extensions.conf. |
13:25.18 | v0lZy | i imagine this is under where i define my provider? |
13:26.39 | [TK]D-Fender | v0lZy, your DIALPLAN |
13:27.58 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
13:29.34 | kaldemar | v0lZy: Set(GROUP()=isdn) would put a channel into a group called "isdn". NoOp(${GROUP_COUNT(isdn)}) would return the number of channels in the group. the latter can be used for example in a GotoIf to decide whether a call is allowed or not. |
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13:31.03 | leifmadsen | wdoekes: ping |
13:32.35 | v0lZy | so everytime my patteren matches, i first need to run it through this application? |
13:32.44 | leifmadsen | yes |
13:32.52 | leifmadsen | you need to check and evaluate |
13:32.54 | bobb_WU | i cant get record to work and the CLI isn't telling me anything |
13:33.30 | bobb_WU | Record(/var/lib/asterisksounds/en/prompt:gsm) |
13:33.37 | [TK]D-Fender | bobb_WU, And you aren't showing us anything or providing details. Fix that and we'll see what we can do for you. |
13:33.43 | leifmadsen | bobb_WU: what version? |
13:33.43 | bobb_WU | directory has 777 |
13:34.05 | bobb_WU | 1.8 - guess i need to update this one too |
13:34.19 | leifmadsen | bobb_WU: you're using old syntax -- run, 'core show application record' |
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13:34.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
13:34.35 | bobb_WU | thanks very much! |
13:36.42 | bulkorok | I use receivefax and asterisk has to send a CED tone to sender, but it doesn't... I use T38 anyone knows why it's not sending!? |
13:37.08 | kaldemar | bobb_WU: ":gsm" <-- what is that supposed to be? |
13:38.36 | coppice | bulkorok: if you are using spandsp it sends the CED signal in T.38 mode. I don't know about the digium FAX module. however. most of the things you send the T.38 CED indicator to ignore it |
13:39.02 | *** join/#asterisk bchia (~Adium@nat/digium/x-euhqiylbgeoagvtw) |
13:39.22 | bobb_WU | its just our chosen format |
13:39.42 | [TK]D-Fender | bobb_WU, http://pastebin.com/LsjuFZvv |
13:39.47 | [TK]D-Fender | bobb_WU, no ":" |
13:40.07 | [TK]D-Fender | bobb_WU, Always read the instructions for your app when it doesn't do what you figured it should have |
13:40.24 | bulkorok | coppice: I use digium fax for asterisk... and the provider from where the faxes are send wants the ced ... |
13:41.40 | bobb_WU | i'm a googler and voip-info is the main result and its super out of date apparently |
13:41.46 | anonymouz666 | bulkorok: I am just curious... why did you choose digium fax for asterisk instead spandsp stuff? digium offers support for that module? |
13:41.55 | bulkorok | yeap |
13:42.00 | [TK]D-Fender | bobb_WU, CLI instructions = your reality |
13:42.21 | bobb_WU | now that i know how to look them up in the CLI i will check it in the future |
13:42.27 | bulkorok | spandsp and digium faxforasterisk are segfaulting asterisk... now I have digium support :-) |
13:42.28 | bobb_WU | however its still not working with the new format |
13:42.46 | [TK]D-Fender | bobb_WU, And you aren't showing us... |
13:43.08 | bobb_WU | cause its on a different computer and it takes me a minute to post pastebins |
13:43.27 | [TK]D-Fender | bobb_WU, Then save it up for us. |
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13:45.00 | bobb_WU | ahhhh i think i just remembered something |
13:45.06 | leifmadsen | kaldemar: that :gsm stuff is the old syntax on like Asterisk 1.2 |
13:45.15 | leifmadsen | I suspect it was found on voip-info or something |
13:45.28 | bobb_WU | so its playing back but not saving- i have to move the file, right? |
13:45.47 | bobb_WU | yeah that was VOIP-info and it said for 1.2 |
13:46.18 | bobb_WU | nvm - i will use the k flag |
13:46.41 | [TK]D-Fender | bobb_WU, No, you don't hav to move the file |
13:46.49 | bulkorok | if I would at least understand the source of res_fax.c... it seems like with T38 there will never be a ced tone generated... |
13:47.07 | [TK]D-Fender | bobb_WU, You can record wherever your * user has rights to. You can playback any file * can erad and has rights to. |
13:47.14 | [TK]D-Fender | read* |
13:47.36 | coppice | bulkorok: the CED signal comes from the actual FAX engine, not from res_fax |
13:48.38 | bobb_WU | well my script is still broken but the recording ability is working again. thanks guys :) |
13:48.47 | bulkorok | coppice: I see the ced stuff only in res_fax.c .... |
13:48.59 | Katty | hello my asterisk does not work at all how to fix plz. |
13:49.51 | Faustov | kill it with fire |
13:50.19 | [TK]D-Fender | ^^^ |
13:50.22 | [TK]D-Fender | Correct answer |
13:50.37 | kaldemar | oh beautiful. my polycom hung up a call when i sent it a BYE from a different machine from my asterisk. which asterisk did not react to and left the channel up. |
13:50.43 | Katty | i don't like that answer, i want a new one |
13:51.01 | coppice | bulkorok: oh, the are fudging in an actual audio tone when going into T.38 mode |
13:51.20 | kaldemar | another call with malformed tags put the polycom to a state that it responds to any message with server error and does not hang up the call even when it's supposed to. |
13:53.38 | bulkorok | coppice: this is how it should be I think... but the ced is not send... |
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14:01.53 | asteriskATmarmuD | hi guys, I can't find the mistake in this one "$AGI->exec("Dial", "SIP/101");" why isn't a call placed? the agi hangs there and nothing happens. any help is highly appreciated |
14:02.34 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
14:02.37 | _zoom_ | hi, |
14:02.56 | _zoom_ | am looking for a commercial Contact center based on asterisk |
14:02.57 | _zoom_ | ? |
14:03.12 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
14:04.05 | _Corey_ | _zoom_: I can help you with that... I'll send you a PM |
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14:05.47 | n3hxs | Katty, you must take that crank like thing and insert it into the front of your PBX, then wind it up. It must have run down overnight. |
14:08.20 | [TK]D-Fender | asteriskATmarmuD, perhaps you should show us the full call with AGI debug.... |
14:09.03 | [TK]D-Fender | asteriskATmarmuD, As well as your AGI |
14:09.42 | asteriskATmarmuD | <PROTECTED> |
14:10.10 | [TK]D-Fender | maybe... I also don't know if you even declared the var in that call... |
14:10.44 | asteriskATmarmuD | <PROTECTED> |
14:11.06 | [TK]D-Fender | asteriskATmarmuD, Start showing us. We aren't going to take your word for it... |
14:11.30 | asteriskATmarmuD | <PROTECTED> |
14:13.10 | Katty | n3hxs: i like it :> |
14:13.21 | Katty | i also like caffeine. |
14:13.32 | Katty | ALL THE CAFFEINES. |
14:13.35 | asteriskATmarmuD | [TK]D-Fender: here is the AGI: http://pastebin.com/XGwT9tia |
14:14.26 | asteriskATmarmuD | [TK]D-Fender: line 185 should initiate the call |
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14:17.30 | carrar | I gots the caffeinesers |
14:17.42 | Katty | glomps carrar |
14:18.09 | carrar | you glomper you! |
14:18.35 | carrar | I roasted 3 cups of coffee beans yesterday too |
14:19.05 | carrar | thats the best way to drink coffee |
14:19.11 | carrar | fresh roasted beans |
14:20.13 | Chainsaw | We have pre-roasted beans in the office. But the grinding & brewing happens as you request it. |
14:21.18 | carrar | Should put a roaster next to it |
14:21.29 | carrar | smoke up the whole office with coffee bean smells |
14:21.43 | carrar | I did it in my office (room at home) once |
14:21.51 | carrar | smelt like a coffee roaster for 2 week |
14:21.53 | carrar | s |
14:22.04 | carrar | was awesome |
14:25.00 | leifmadsen | Katty: plz 2 b helping! my astrix system crashy! |
14:25.11 | Katty | pamples it. |
14:25.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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14:27.03 | asteriskATmarmuD | ok, a general question. If I want to automatically call a list of numbers, may I use AGI and call the perl-agi-script directly from the command line? |
14:29.28 | [TK]D-Fender | "help channel originate" <- |
14:29.53 | [TK]D-Fender | Katty, PAMPLEMOUSE! |
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14:58.52 | bulkorok | coppice: I tried with res_fax_spandsp... this on is sending the ced tone! |
14:59.49 | coppice | a lot of people move from the digium FAX to spandsp to fix problems :-) |
15:00.19 | bulkorok | :) |
15:00.26 | coppice | I've also helped people who were paying for digium FAX, but actually using spandsp :-\ |
15:01.06 | bulkorok | this is my situation now... but maybe I can use SendFax with digium fax and spandsp for receiving |
15:01.26 | coppice | why use digium FAX at all? |
15:02.02 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
15:02.41 | bulkorok | asterisk segfaults when I use spandsp... so I hoped that it doesn't when using digium fax... but asterisk is segfaulting with it too :-( |
15:03.09 | bulkorok | but... I opend a ticket to digium and mybe it's fixed soon... for bith techs! |
15:03.12 | bulkorok | both |
15:03.44 | ChannelZ | I've used FFA, not extensively, but haven't had any problems |
15:04.53 | bulkorok | we will receive about 2.000 - 3.000 faxes per month with this... so I have to test very hard... |
15:04.56 | coppice | ChannelZ: how many 100's of k of FAXes do you handle each day? |
15:07.22 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
15:07.59 | bulkorok | k guys... got to go... I opened another ticket to digium... we will see waht the y say about that :-) |
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15:38.38 | din3sh | anyone ever implemented click-to-call on lotus notes working with asterisk? |
15:43.05 | [TK]D-Fender | din3sh, Told you yesterday, TAP is GENERIC and there links ont he WIKI for tools for this |
15:43.08 | [TK]D-Fender | TAPI* |
15:43.10 | [TK]D-Fender | ~wikis |
15:43.11 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
15:44.18 | din3sh | already looked stuff up on it |
15:44.39 | din3sh | i actually wanted to find somebody who already implemented it via lotusscript |
15:44.41 | leifmadsen | I suspect near-zero people here have done lotus notes integration |
15:44.55 | din3sh | i guess too |
15:45.10 | leifmadsen | lotus notes is a fairly specialized app that not a lot of people use |
15:45.12 | coppice | half of them haven't even heard of Lotus Notes :-) |
15:45.20 | din3sh | lol |
15:45.21 | leifmadsen | exactly :) |
15:45.27 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
15:45.33 | leifmadsen | I only know one person who has ever used it, and they do it all backwards in Sarnia |
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15:47.28 | din3sh | a tricky project, they want their lotus notes to be integrated with pbx, through click-to-dial, and also missed call status on their mail |
15:49.00 | leifmadsen | sounds like you have some reading up to do on lotus notes |
15:49.15 | Chainsaw | You could always hire an IBM contractor/integrator. |
15:49.23 | din3sh | trying to install a trial version |
15:49.51 | din3sh | Chainsaw:yeah would have to consider that too |
15:50.18 | din3sh | aparently avaya pbx already has some of tool working with lotus |
15:52.02 | din3sh | irrespective of lotus mail, how to implement missed call alert on mail? |
15:52.32 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
15:52.46 | Chainsaw | din3sh: Asterisk supports IMAP as a voicemail backend, you may be able to do something with that. |
15:55.13 | din3sh | can i use ${DIALSTATUS} - NOANSWER to trigger some kind of send mail application? |
15:55.22 | nny | have a strange issue, not sure what to look at explicitly yet. We have an office with some phones that are remote (over VPN). They can't use the phone's conference feature. - When they hit join it dies. The local phones perform this function normally. I am setting up a test now, not sure if i'll see anything since the "join" is on the phone's side. I have directmedia=no if that helps. I think the problem is the phones can't handle the media streams effectively |
15:55.34 | Chainsaw | din3sh: Sure. |
15:56.01 | nny | din3sh: you can send an email during a call using System or Shell (for example) |
15:56.15 | nny | we have used sendmail to send texts from the cli before screwing around |
15:57.33 | din3sh | will try that |
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15:59.17 | din3sh | users also want to logoff/lock their phones when they lock their computers (CTRL+ALT+DEL) in windows |
15:59.31 | din3sh | any idea on how to go about implementing such a feature? |
16:00.00 | nny | din3sh: look into asterisk management interface |
16:00.28 | nny | din3sh: everything you're trying to do will require a connection to it to handle those types of things. You could have windows log off also log an agent off, for example |
16:01.18 | *** join/#asterisk General_Z0D (~z0d@216.255.243.34) |
16:01.24 | nny | din3sh: lockign the phone itself is however a feature of the phone and I don't think many phones have an API to interpret remote calls to do so. Agent log off would render the phone useless unless someone logged in, however. Just shooting from the hip |
16:01.29 | [TK]D-Fender | <din3sh> a tricky project, they want their lotus notes to be integrated with pbx, through click-to-dial, and also missed call status on their mail <- missedcall via e-mail is 1 trival line of dialplan. |
16:01.39 | General_Z0D | cany anyone recommend a good working open source billing system for asterisk |
16:01.59 | [TK]D-Fender | General_Z0D, a2billing |
16:02.47 | nny | din3sh: you also may look at something like this http://www-10.lotus.com/ldd/nd85forum.nsf/Customer/6900587585d8221105257677001ec69a?OpenDocument |
16:02.52 | General_Z0D | thanks fender |
16:03.05 | General_Z0D | aint that star2billing now? |
16:03.18 | din3sh | [TK]D-Fender: missedcall via e-mail is 1 trival line of dialplan. how!? |
16:03.38 | [TK]D-Fender | din3sh, "man sendmail" |
16:04.13 | din3sh | how to send details about the calling number/etc? |
16:05.34 | din3sh | nny: thnks , already checked that srcipt, trying to download lotus notes to test it |
16:07.45 | [TK]D-Fender | din3sh, Include the CALLERID obviously |
16:10.02 | din3sh | will try it |
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16:12.59 | General_Z0D | does 10.5 of * have realtime support like in 1.4 it has been some timesince I used it |
16:15.14 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
16:15.16 | nny | oops |
16:15.32 | nny | din3sh: did you get my last message? "not sure if TAPI is supported natively in lotus notes, but it's the common phone system API. If it isn't native, you're better off using scripting to call the AMI instead" |
16:18.55 | [TK]D-Fender | General_Z0D, * doesn't tend to LOSE functionality over time... |
16:19.22 | [TK]D-Fender | Actually you're better off with AMI altogether |
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16:25.31 | din3sh | nny: thanks, looking into that |
16:27.07 | ThinkGNU- | Hitting an interesting issue: if I set timeout for more than 20 seconds when dialing a local channel I can't get the call to go through the rest of the dialplan rules for that dialplan |
16:28.42 | nny | teliax.. thoughts? |
16:30.15 | [TK]D-Fender | ThinkGNU-, Show us |
16:31.15 | [TK]D-Fender | nny, They are (were) OK, but not the strongest on any one side.Flakey on rarer occasions, but otherwise decent enough... |
16:31.47 | ThinkGNU- | Okay [TK] |
16:40.25 | ThinkGNU- | I'm trying to narrow down the conditions under which I'm able to reproduce the problem and then get the logs from that |
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16:46.07 | ThinkGNU- | http://pastebin.com/z68JcVsd |
16:46.18 | ThinkGNU- | If my timeout is 25 or greater I seem to have this problem |
16:47.23 | Qwell | what timeout? |
16:47.37 | ThinkGNU- | The timeout on the Dial application that calls the local channels |
16:47.56 | Qwell | I see no such timeout in your log |
16:48.12 | ThinkGNU- | I agree |
16:48.23 | ThinkGNU- | If I set the timeout to 23 seconds for example then the timeout works |
16:48.35 | ThinkGNU- | If set to 25 or greater then the call just ends |
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16:50.34 | [TK]D-Fender | ThinkGNU-, Dialplan please.... |
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16:51.24 | ThinkGNU- | Sure |
16:51.53 | carrar | Shouldn't ThinkGNU be using sipwitch? |
16:52.00 | carrar | :) |
16:52.29 | [TK]D-Fender | And I don't see that call that CALLS the Local channel |
16:52.39 | [TK]D-Fender | I certainly don't trust that at all |
16:52.49 | [TK]D-Fender | [Jul 10 10:41:41] -- Executing [channel_2@office-reception:1] Verbose("Local/channel_2@office-reception-bc |
16:53.01 | [TK]D-Fender | First line shows I'm already missing the greatest limiting factors |
16:53.39 | ThinkGNU- | http://pastebin.com/NAZjnx5Z |
16:54.59 | [TK]D-Fender | [Jul 10 10:42:05] == Spawn extension (open, office-day, 2) exited non-zero on |
16:55.22 | ThinkGNU- | What's causing that to happen just by changing the timeout number |
16:55.23 | [TK]D-Fender | Call starts at 41, ends at 05. that's ... 25 seconds <------------- |
16:55.23 | ThinkGNU- | ? |
16:55.28 | [TK]D-Fender | same => n,Dial(Local/channel_1@office-reception&Local/channel_2@office-reception/n,25,Tk) |
16:55.34 | [TK]D-Fender | YOUR 25 second ilmit was reached |
16:55.37 | [TK]D-Fender | limit* |
16:55.58 | ThinkGNU- | right but why won't it move along to the next priority in the dialplan? |
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16:57.04 | [TK]D-Fender | I see limits logs there. I don't what else you've cut off... |
16:57.52 | *** join/#asterisk nny (~Scott@174.107.223.14) |
16:57.53 | [TK]D-Fender | I also only see the nested channels die, not the main |
16:58.00 | [TK]D-Fender | My trust level is shinking fast |
16:58.12 | carrar | throws TK a rope |
16:58.25 | carrar | tied it to the timeout!! |
16:58.28 | [TK]D-Fender | hangs ThinkGNU- with it |
16:58.31 | [TK]D-Fender | :D |
16:58.33 | ThinkGNU- | Thanks |
16:58.34 | nny | [TK]D-Fender: thanks. Looking for a secondary provider, striking out here and there. If you have any other suggestions, feel free. I think i've gone through all the ones the channel suggests |
16:58.55 | ThinkGNU- | Sorry I'm obviously missing what's going on here and I don't mean to agitate you guys |
16:59.04 | [TK]D-Fender | nny, vitelity/voip.ms pretty solid, les.net |
17:00.03 | carrar | ThinkGNU, at the time time, does it jump to the t priority? |
17:00.10 | carrar | ThinkGNU, at the timeout time, does it jump to the t priority? |
17:00.18 | carrar | err |
17:00.22 | carrar | t extension I mean |
17:00.23 | [TK]D-Fender | Of course not. |
17:00.33 | [TK]D-Fender | this ian app timeout, not an IVR |
17:00.37 | carrar | oh |
17:00.39 | carrar | I missed that |
17:00.40 | nny | [TK]D-Fender: thanks |
17:00.44 | ThinkGNU- | yeah, it's just ending |
17:01.05 | ThinkGNU- | I have a log where I set the app timeout to 23 seconds and everything executes beautifully |
17:01.10 | ThinkGNU- | 24 works fine |
17:01.14 | ThinkGNU- | 25 and greater no luck |
17:01.16 | [TK]D-Fender | I still don't see the ENTIRE call |
17:01.31 | ThinkGNU- | I'll get an entire call, sorry TK |
17:01.41 | [TK]D-Fender | Entire call + MORE |
17:01.44 | [TK]D-Fender | both sides |
17:01.53 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
17:01.57 | [TK]D-Fender | I want to make sure that things have a chance to die out |
17:02.17 | ThinkGNU- | okay, gimme just a second and I'll have that for you |
17:07.20 | ThinkGNU- | http://pastebin.com/vz0L1X5A |
17:07.28 | ThinkGNU- | That's everything it logged for that call TK |
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17:09.34 | [TK]D-Fender | Local/channel_1@offic |
17:09.34 | [TK]D-Fender | e-reception <-- missing the /n on it |
17:09.38 | [TK]D-Fender | 2nd one had it |
17:09.51 | [TK]D-Fender | [Jul 10 11:02:35] -- Called channel_1@office-reception |
17:09.52 | [TK]D-Fender | [Jul 10 11:02:35] -- Called channel_2@office-reception/n |
17:10.27 | ThinkGNU- | lemme fix that real quick |
17:11.14 | [TK]D-Fender | Jul 10 11:02:59] == Spawn extension (open, office-day, 2) exited non-zero on 'SIP/wyomingcom1-0000021f' |
17:11.21 | [TK]D-Fender | there's the outer channel dying |
17:12.12 | ThinkGNU- | exiting non-zero isn't what it should be doing right? |
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17:12.58 | [TK]D-Fender | ThinkGNU-, One serious consideration is that you are getting a call from an ITSP .... THEY may lok at this and go "Hey, MY client isn't responding" ... and GIVE UP |
17:13.08 | [TK]D-Fender | ThinkGNU-, You should probably ANSWER the call first <- |
17:13.24 | ThinkGNU- | That's a good point |
17:13.52 | ThinkGNU- | I knew I was overlooking something simple here |
17:18.28 | ThinkGNU- | Let me make some adjustments and see what happens. Thank you for helping out guys. Obviously I still have a lot to learn here but I am determined to learn. |
17:18.50 | [TK]D-Fender | ThinkGNU-, Well you just got a good lesson in tracing things back to the start |
17:19.03 | [TK]D-Fender | ThinkGNU-, And isolating each breaking point |
17:19.03 | ThinkGNU- | Yes I did and thank you so much |
17:19.23 | ThinkGNU- | This is the kind of experience I can definitly use. |
17:21.53 | ThinkGNU- | [TK]D-Fender, that did it. I ANSWER the call first and then no problems. Thank you again for all your help guys. One day, when I don't suck as bad I hope to be able to contribute here in the channel and answer questions. |
17:25.53 | [TK]D-Fender | ThinkGNU-, You're welcome. |
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17:28.21 | bobb_WU | how do you reload just the dialplan? and is there a good doc for asterisk CLI commands? |
17:29.24 | outtolunc | help and help dialplan |
17:30.37 | nny | what is the suggested method for using one touch recording (automon) with meetme? I see options for dial and queue, but not for the application. Do i have to use Local? |
17:31.30 | bobb_WU | ok and how do i execute a System() command after the user hangs up after a recording? is there a trick for that? i'll pastebin my current code if need be |
17:31.54 | ectospasm | bobb_WU: put System() in the 'h' extension |
17:32.25 | ectospasm | 'h' executes after the call has hung up. |
17:32.28 | bobb_WU | so i can move that to behind the Hangup() one too? |
17:33.00 | ectospasm | what do you mean,"behind?" |
17:34.47 | bobb_WU | i have a hangup in all my extensions, i thought maybe i could add a backend action after that executes |
17:36.07 | [TK]D-Fender | no |
17:36.11 | bobb_WU | pastebin.com/KbHaiyPw |
17:37.35 | bobb_WU | returnCall.sh writes a few lines to a {exten}.call file then moves it to /var/spool/asterisk/outgoing |
17:37.47 | bobb_WU | it works fine from bash and the message is there to playback |
17:38.19 | bobb_WU | running 1.8 on this server |
17:38.44 | ectospasm | Hangup() isn't necessary unless you want to force a hangup in every context/extension. The 'h' extension executes *after* Hangup(), or the far end hangs up, so you can do your call postprocessing there. |
17:39.06 | bobb_WU | (that's what i meant by behind) thanks |
17:39.20 | ectospasm | exten => h,n,... ; put your post prcessing here. |
17:40.21 | bobb_WU | so not exten = > 8051,h ? |
17:40.40 | bobb_WU | i only need this number to postprocess |
17:41.31 | [TK]D-Fender | NO |
17:41.34 | [TK]D-Fender | ther is no "chaining this |
17:41.47 | [TK]D-Fender | You need to integrate this into it |
17:42.10 | nny | i'm just gonna use the local channel for recording meetme on demand, let me know if that's improper :) |
17:43.33 | nny | hmm maybe not, seems using local to call meetme supercedes the admin and user menu :S |
17:43.48 | nny | recording worked, however.. lol |
17:46.07 | bobb_WU | "ther is no "chaining this" can you clarify? its still not running the script on hangup |
17:46.37 | bobb_WU | i tried exten = > 8051,h,System() and exten => h,n,System() |
17:46.59 | [TK]D-Fender | "h" is NOT a priority |
17:47.26 | [TK]D-Fender | and exten => h,n,System() doesn't tell me you have a priority 1 in the first place |
17:47.46 | [TK]D-Fender | and from what you're showing you also seem to be suggesting calling that with no parameters |
17:48.12 | bobb_WU | it has parameters |
17:48.16 | bobb_WU | did you see the pastebin? |
17:48.51 | nny | so I guess my actual question to channel is "can I use local to record meetme (with the W option) and still pass key pressed to Meetme for the admin/user functions? |
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17:49.06 | [TK]D-Fender | Your pastebin shows your trying to use "h" as a PRIORITY |
17:49.14 | [TK]D-Fender | it is not |
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17:58.15 | leifmadsen | bobb_WU: uhhh, note that 8051,h,Foo() is invalid |
17:58.38 | bobb_WU | yeah i think i understand |
17:59.05 | bobb_WU | its h,1,Foo() but that means i force an action on every exten in the context |
17:59.13 | bobb_WU | so i'm giving it its own context |
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18:00.23 | bobb_WU | and it worked |
18:00.30 | bobb_WU | ish |
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18:05.23 | bobb_WU | System(/asteriskScripts/returnCall.sh ${CALLERID(num)}) doesn't seem to be working. its being called with the correct data as seen by logging output |
18:06.23 | [TK]D-Fender | We aren't seeing anything |
18:06.34 | leifmadsen | what isn't working? |
18:06.46 | bobb_WU | its not calling the script |
18:06.54 | [TK]D-Fender | We aren't seeing anything |
18:07.22 | leifmadsen | does somethign simple like, System("touch /tmp/this_worked") work? |
18:10.51 | bobb_WU | ok http://pastebin.com/95eBpZwk |
18:11.28 | bobb_WU | i'm not seeing yep and there is nothing under ./calls/ |
18:11.33 | bobb_WU | *Yep |
18:12.10 | [TK]D-Fender | empty pastebin |
18:12.55 | bobb_WU | http://pastebin.com/95eBpZWK |
18:13.05 | bobb_WU | its case sensitive- hmmm |
18:14.11 | [TK]D-Fender | You never EVER use "cp" to put a call file in the spool folder |
18:14.31 | bobb_WU | what should i use instead? |
18:14.33 | outtolunc | mv |
18:15.01 | [TK]D-Fender | And I don't see proof that * can read that script you are colling. that the paths are right.. that * is defaulting to the FOLDER yout hinit it is since yuo're creating files relative to where you ARE |
18:15.04 | bobb_WU | k |
18:15.20 | [TK]D-Fender | This is a massive failure in debugging practice |
18:16.10 | [TK]D-Fender | "you think"* |
18:16.25 | [TK]D-Fender | gah, typing going down the tubes again |
18:17.21 | bobb_WU | well how about it. its working correctly! |
18:17.55 | bobb_WU | i made the paths absolute and changed cp to mv |
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18:30.49 | UnixDev | would it be possible to move on this request? (https://issues.asterisk.org/jira/browse/ASTERISK-18068) .. its already been done in trunk (https://reviewboard.asterisk.org/r/1754/) and looks pretty trivial to back port |
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18:43.34 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
18:43.38 | Katty | hello my asterisk does not work at all how to fix plz. |
18:43.56 | Katty | it makes a gurgle noise. |
18:44.01 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
18:44.05 | drmessano | You make a gurgle noise |
18:44.11 | drmessano | Its hawt too |
18:44.14 | Katty | that's a drinking game. |
18:44.24 | Katty | i saw it at hastings the other night. |
18:44.34 | drmessano | Thats also AFTER a drinking game |
18:44.35 | *** join/#asterisk rcaskey (~Rob@dumbledore.athenshousing.org) |
18:44.47 | _Corey_ | Katty: sounds like a plumbing issue... pour Drano into system |
18:44.55 | Katty | oh wait, no. it's called "Gargle" |
18:45.07 | Katty | _Corey_: *hee* |
18:45.11 | malcolmd | you mean the scary things on top of old buildings? |
18:45.12 | drmessano | lol |
18:45.17 | Katty | adds to list. |
18:46.09 | Katty | i'm knitting a fabulious little thing today. |
18:46.14 | Katty | and by little i mean a 28 hour project. |
18:46.50 | [TK]D-Fender | 28 HOURS LATER |
18:46.58 | Katty | YES. |
18:47.00 | [TK]D-Fender | pictures zombies with scarves |
18:47.21 | Katty | i have some smaller projects in the making too. |
18:47.24 | Katty | BACON STRIPS. |
18:47.25 | slav3_kitten | Katty, trade you :D i've got roughly 2 or 3 days left on the ddrescue of my ipod :( |
18:47.34 | drmessano | The only thing I ever spent 28 hours on was my best friends mom |
18:47.50 | slav3_kitten | drmessano, was she worth it |
18:47.57 | Katty | how does any knitting project the size of an ipod take 2 to 3 days? |
18:48.17 | drmessano | slav3_kitten: I had to start giving her CPR after 14 hours, so not really |
18:48.22 | Katty | oh. ddrescue. |
18:48.24 | [TK]D-Fender | Katty, AND JD. That's SMART. |
18:48.28 | slav3_kitten | :( |
18:48.30 | Katty | i totally just skipped that word. |
18:48.31 | slav3_kitten | that sucks |
18:48.41 | Katty | [TK]D-Fender: that's a terrible idea. |
18:48.58 | slav3_kitten | lol @ Katty |
18:49.18 | slav3_kitten | yea when i was on vacation last month mister ipod started making funny noises an losing bits of songs |
18:49.38 | [TK]D-Fender | Katty, The EMT guys are locals here... one of their runs was on a Taco Bell that's within walking distance of my place |
18:49.38 | Katty | :< |
18:50.05 | Katty | I guess they call them Volcano Tacos for a reason. |
18:50.22 | Katty | slav3_kitten: i hope it completes ok :< |
18:50.44 | drmessano | I thought they called them Volcano tacos because of what happens 24 to 36 hours after |
18:50.45 | slav3_kitten | me too, or at least lets me recover file names |
18:51.02 | slav3_kitten | then i can spend a week having dad search the attic for my CD collection |
18:51.30 | Katty | your dad has better things to do. |
18:51.32 | Katty | like your mom. |
18:51.36 | Katty | search for it yourself! |
18:51.36 | drmessano | HA LOL |
18:51.54 | slav3_kitten | Katty, you figure out how to climb a ladder one legged and on crutches an i'll give you a cookie |
18:52.11 | drmessano | One legged AND on crutches? |
18:52.13 | Katty | ask your girlfriend. |
18:52.21 | Katty | accepts cookie. |
18:52.24 | slav3_kitten | what girlfriend |
18:52.25 | drmessano | lol |
18:52.36 | sp00kz | hop |
18:52.38 | slav3_kitten | you see the channel we are all currently in? |
18:52.47 | Katty | yes. |
18:52.49 | Katty | danny is married. |
18:52.50 | slav3_kitten | what part of that makes you think i have a girlfriend or boyfriend for that matter |
18:52.54 | Katty | and i have a gentleman friend. |
18:53.00 | drmessano | I think Katty meant to say your mom has a peg leg with a kickstand |
18:53.04 | Katty | i refute your logic and replace it with my own. |
18:53.08 | slav3_kitten | meh |
18:53.21 | slav3_kitten | i've been single for like *does math* 10 years |
18:53.31 | drmessano | Yes, Katty has a special gentleman friend |
18:53.31 | Katty | why. |
18:53.35 | Katty | drmessano: yes. |
18:53.36 | drmessano | and she's got a beverage here |
18:53.41 | Katty | YES. |
18:53.45 | Katty | it's a lovely beverage. |
18:54.06 | Katty | ...you can't have it. either of them. |
18:54.09 | slav3_kitten | Katty, before i got disabled 5 years ago i was regularly dating to find a woman into computers, and electronics |
18:54.14 | drmessano | slav3_kitten: Theres no reason to be single with lonely married women and zaftig girls all over the internet |
18:54.24 | slav3_kitten | now i can't even get dates |
18:54.32 | Katty | oh i see. |
18:54.36 | drmessano | Its like this |
18:54.53 | Qwell | Aren't you married? You aren't allowed to give dating advice. :p |
18:54.54 | Katty | have you been /trying/? |
18:54.59 | drmessano | "Hi" "Are you married?" "No?" "Are you Fat?" "Yes? W00t" |
18:55.06 | slav3_kitten | Katty, i have |
18:55.12 | Katty | well keep trying dear. |
18:55.21 | drmessano | That should be a macro in your favorite IM client |
18:55.29 | Katty | a good friend of mine is paralyized from the waist down due to a mountain biking accident where he flipped over the bars. |
18:55.34 | slav3_kitten | eh, lately i've decided my money would be better spent on computer and electronics bits |
18:55.43 | slav3_kitten | i'm terribly sorry to hear that |
18:55.46 | Katty | and he has a lovely lady in his life now thanks to him being persistant. |
18:55.49 | drmessano | Check out myspace, Meetme.com, and Yahoo. Look for anyone married or fat. Easy targets |
18:55.50 | slav3_kitten | i broke my spine a little bit at work |
18:56.00 | Qwell | Katty: persistent ~= creepy |
18:56.06 | Katty | no not in a creepy way. |
18:56.08 | Katty | just kept looking |
18:56.14 | Katty | didn't give up. |
18:56.19 | Katty | not all stalkery over a person or anything. |
18:56.34 | Qwell | stalks Katty |
18:56.43 | drmessano | Just keep asking the Doctor Messano 2 key questions and you will find love |
18:56.46 | Qwell | (in a non-creepy way) |
18:56.48 | slav3_kitten | is his lovely lady from a slovic nation that has good prices an low shipping? |
18:57.01 | drmessano | "Are you married?" "Are you fat?" <-- All you ever need to know |
18:57.02 | Katty | no. |
18:57.08 | Katty | she's an adorable little brunette with a heart of gold. |
18:57.20 | Katty | and happens to like dr who. |
18:57.23 | slav3_kitten | i never said that mail order brides can't be high quality... |
18:57.25 | Katty | no it's not me. |
18:57.31 | Katty | they can be. |
18:57.33 | Katty | i know one. |
18:57.37 | Katty | she just wanted to start a family. |
18:57.40 | drmessano | Oh, and if you get to step 2.. Ask them "So are you happily married or just married?" <---- SRSLY |
18:57.58 | slav3_kitten | yea that's my main problem these days Katty |
18:58.08 | slav3_kitten | i'm near 30 and i want nothing to do with having a family or starting one |
18:58.17 | Katty | oh that's not a problem. |
18:58.26 | Katty | i'm getting close to 30 and want none of that family crap either. |
18:58.28 | slav3_kitten | women in my age group seem to want to start families and that crap |
18:58.34 | Katty | i disagree. |
18:58.37 | slav3_kitten | so, want to get some coffee? |
18:58.48 | Katty | drmessano: did you see that comment? |
18:58.52 | Katty | drmessano: he has no idea. |
18:59.02 | drmessano | Yes I did |
18:59.09 | Katty | it's amusing |
18:59.22 | slav3_kitten | what don't i have any idea about? |
18:59.22 | drmessano | Let me explain the age groups... |
18:59.24 | Katty | slav3_kitten: and this is why -> https://sphotos.xx.fbcdn.net/hphotos-ash4/c0.0.403.403/p403x403/376956_10100311924647677_407737399_n.jpg <- posted 30 minutes ago. |
18:59.36 | Katty | slav3_kitten: i HAVE a coffee. iced coffee. |
18:59.39 | slav3_kitten | ... |
18:59.42 | slav3_kitten | freaky! |
18:59.44 | drmessano | 18-25 Either concerned about school or want a family, NOW |
18:59.53 | Katty | slav3_kitten: no it's actually quite tasty. not freaky at all. |
18:59.54 | drmessano | 25-33 Happily Single |
19:00.01 | Katty | drmessano: YES. |
19:00.13 | drmessano | 33-39 Exiting first marriage, looking for good guy. PRIME REAL ESTATE |
19:00.22 | Katty | that's not true. |
19:00.26 | Katty | i already did my graceful exit. |
19:00.34 | slav3_kitten | you were married? |
19:00.36 | drmessano | 39 to 45 -- Doesnt matter, they will do anything in bed tho |
19:00.37 | Katty | nearly. |
19:00.40 | slav3_kitten | oh good |
19:00.48 | slav3_kitten | dodged that bullet |
19:00.53 | drmessano | Ok well |
19:00.58 | Katty | drmessano: agreed. one of my friends really enjoys that 40-50 age group. |
19:01.02 | drmessano | 33-39 is also "STILL HAPPILY SINGLE" |
19:01.02 | slav3_kitten | so what you're saying drmessano is aim for 40 |
19:01.14 | Katty | slav3_kitten: yes. |
19:01.24 | Katty | slav3_kitten: as my friend says, they cook clean, and enjoy sex! |
19:01.36 | jpsharp | Where does that biological clock of "I HAVE TO PROCREATE NOW" come into play? |
19:01.49 | drmessano | Nah.. You will find LOTS of very cool, awesome women who married some guy too young and now want something real in the 30 to 40 group |
19:01.55 | Katty | jpsharp: i don't know. i've not ran into it yet. |
19:02.01 | Qwell | jpsharp: 16, if TV has taught me anything. |
19:02.10 | Katty | this is why i love Qwell |
19:02.13 | slav3_kitten | Katty, pm? |
19:02.26 | jpsharp | No, that's "I HAVE TO BE AN ATTENTION WHORING TEENAGER NOW" |
19:02.27 | Katty | slav3_kitten: sure! |
19:03.02 | Katty | jpsharp: they are annoying :< |
19:03.09 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
19:03.20 | *** join/#asterisk PipBoy (PipBoy@ip067.216.145.100.nas.net) |
19:03.24 | drmessano | Second time around can be completely awesome for a lot of women. I keep telling my best friend that. He's 35 and wants a damn 35 yr old puritan virgin that goes to church 3 times a week OR NOTHING. Says hes doomed to never find anyone. I keep telling him to look for a nice second-time-around gal who is completely awesome and sick of the meathead she married after high school |
19:04.03 | Katty | ...goes to church? :< |
19:04.06 | Katty | yuck. i do not approve. |
19:04.11 | Katty | but that's ok, i dont' have to! |
19:04.25 | drmessano | The whole "All the good ones are taken" thing is bullshit. Nobody stays in bad marriages anymore, and there's lots of great gals going back into circulation |
19:05.13 | PipBoy | quick question. When a person calls in over a SIP trunk.. And pressed a digit in an IVR.. is it still considered DTMF? |
19:05.29 | drmessano | Of course it is |
19:05.59 | PipBoy | Ah, figured as much... just trying to tweak a pbx to be a little more dtmf sensitive... not picking up a lot of key presses |
19:06.10 | PipBoy | thanks |
19:06.34 | jpsharp | If its coming in over SIP, you should use something like SIP INFO or RFC to transport DTMF rather than trying to detect the tones. |
19:07.06 | jpsharp | rfc2833 |
19:07.17 | Katty | PipBoy: I LOVE PIPBOYS |
19:07.23 | Katty | PipBoy: in reference to fallout. |
19:07.32 | PipBoy | fer leveling n stuff |
19:07.33 | PipBoy | lol |
19:07.37 | Katty | i don't want to set the world on fire |
19:07.51 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
19:08.08 | PipBoy | hmmm.. thanks jpsharp . I need to look into that, as I have no idea what my system is currently doing |
19:10.32 | PipBoy | But yea, I believe my system must be doing DTMF. You know the whole "press * to start recording" I have to constantly mash * lol... Customer is kinda annoyed :P |
19:10.50 | jpsharp | inband DTMF over SIP sucks, especially if you're running something other than G711 codec. |
19:12.13 | Katty | tell them to take their stress out on the * button! |
19:12.15 | Katty | problem solved! |
19:12.47 | PipBoy | yea * is the modern digital bubble wrap! |
19:12.48 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.12) |
19:13.10 | jpsharp | If I did that, my phone would be a shattered, broken shell of itself...lying bleeding in a corner. |
19:13.46 | PipBoy | having "Nam" flashbacks lol |
19:13.53 | PipBoy | YOU DONT KNOW!! YOU WERNT THERE! |
19:14.05 | Katty | puts phone into therapy |
19:14.34 | slav3_kitten | refrains from inserting a holodisk into PipBoy |
19:14.46 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
19:15.03 | PipBoy | lol... hey now! i dont go that way |
19:16.06 | Katty | why not. |
19:16.12 | Katty | you should try it. you might like it. |
19:16.17 | PipBoy | lolol |
19:16.31 | Katty | i bet you didn't like vegetables as a kid either! |
19:16.48 | PipBoy | walks away slowly and starts googling RFC's |
19:16.51 | ThinkGNU- | Did I switch channels somewhere down the line? |
19:17.31 | Katty | yes. |
19:17.45 | ThinkGNU- | lol |
19:18.10 | ThinkGNU- | You know, PipBoy still might not like vegetables |
19:18.20 | PipBoy | its true |
19:18.27 | PipBoy | my diet is beer and potatos |
19:18.47 | Katty | potato is a vegetable. |
19:18.47 | drmessano | Are you irish? |
19:18.51 | Katty | and beer is made from a plant. |
19:18.53 | PipBoy | canadian :P so close |
19:19.06 | Katty | technically you could be a vegetarian. |
19:19.10 | ThinkGNU- | cows are made from plants too I think, indirectly |
19:19.13 | drmessano | 4 course meal to an irishman.. 3 beers and a potato |
19:19.14 | Katty | unless you consider yeast an animal. |
19:19.19 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
19:19.23 | Qwell | Katty: I could totally be a vegetarian. |
19:19.35 | Qwell | Just as long as they change the classification of meat to be a vegetable |
19:19.41 | Katty | :P |
19:19.42 | Qwell | that's my only hangup |
19:20.01 | Katty | lately i've been enjoying burgers. |
19:20.23 | Katty | and leinenkugel |
19:20.36 | drmessano | Someone told me I don't eat enough greens, so I put some green M&Ms in my bacon, ham, and sausage omelette the other night |
19:20.52 | Katty | you should have put green onions on top |
19:20.58 | Qwell | cilantro |
19:21.11 | Katty | cilantro is evil. it makes me sneeze and turn redish. |
19:21.17 | Qwell | lame |
19:21.18 | drmessano | I'm saving the green onions for Bulgogi |
19:21.37 | Katty | Bulgogi? |
19:22.06 | Katty | BUGATTI?! |
19:22.14 | drmessano | No |
19:22.20 | Katty | sad :< |
19:22.29 | Katty | danny you should get me a bugatti. |
19:22.34 | Katty | they're only 1.5mil. |
19:22.35 | drmessano | http://en.wikipedia.org/wiki/Bulgogi |
19:22.43 | Katty | i found a picture. it looks tasty. |
19:23.10 | drmessano | I make mine with beef, sesame oil, soy sauce, and sugar. It's amazing |
19:23.35 | drmessano | Drop it on some rice, and top it with sesame seeds and green onion |
19:23.52 | Katty | i think you're avoiding the bugatti topic. |
19:24.07 | drmessano | Let me just say this |
19:24.19 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
19:24.42 | drmessano | That's 1.4-ish million more than I have spent on women who I have done the naked roly poly with |
19:24.50 | drmessano | So.. There's qualifications here |
19:24.51 | Katty | derik promised me a bugatti. |
19:25.00 | Katty | as soon as he can afford to get himself one too. |
19:25.23 | drmessano | Very nice |
19:25.27 | drmessano | That seems reasonable |
19:25.31 | Katty | i agree. |
19:26.13 | drmessano | I would be happy with a nice second hand job right now for her |
19:26.25 | drmessano | Its either new car, or clunker |
19:26.30 | drmessano | No in-between |
19:26.50 | Katty | second hand job? |
19:26.53 | Katty | she doesn't work? |
19:27.48 | drmessano | Well, "job" in the objective sense. We need a better car.. Something second hand |
19:28.36 | Katty | ah i see. |
19:29.06 | drmessano | Buying a second hard car is like buying software.. If you're not gonna buy the real deal, expect someone to lie to you about how awesome it is |
19:29.21 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
19:29.22 | ThinkGNU- | You just have to know what you're looking at |
19:29.22 | drmessano | "Dude, this is JUST LIKE PHOTOSHOP... $19.99" |
19:29.36 | Katty | i agree. |
19:29.39 | Katty | how much are you willing to spend? |
19:30.00 | drmessano | I do know what I am looking at.. The problem is the misrepresentation and the time consumption test driving crap |
19:30.38 | Katty | tell me your price range, dagnabbit. |
19:30.47 | drmessano | Probably 6-8K |
19:30.58 | Katty | and are there any requirements? |
19:31.02 | Katty | automatic... 4 doors |
19:31.09 | Katty | must have backseat. |
19:31.15 | drmessano | Automatic, 4 doors.. all must work |
19:31.39 | Katty | have you considered salvage? |
19:32.21 | drmessano | We spent a 2 weeks looking for the POS she drives now.. Can't tell you how many $200 cars there are out there with $2000 price tags |
19:32.39 | drmessano | "You didnt specify you needed a transmission with it" |
19:33.27 | ThinkGNU- | who needs those? |
19:33.35 | Katty | ya get a rotobox ;) |
19:33.41 | ThinkGNU- | I agree with you on the overpriced used cars though |
19:33.55 | Katty | i'll sell you mine for that much! |
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19:33.57 | *** mode/#asterisk [+o sruffell] by ChanServ |
19:34.09 | Katty | there's nothing wrong with it, other than it's not as shiny as a 350z |
19:34.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
19:34.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
19:35.57 | drmessano | I had a guy try to sell me a 2000 KIA Sephia.. It looked like it spent 3 months underwater. Tires were bald.. He wanted $1200 for it. |
19:36.14 | drmessano | Oh, and it was in the bulletproof vest section of town |
19:47.23 | rcaskey | drmessano, here in the states I highly recommend if you can at all afford it sucking it up and buying a new Elantra, Sonata, or Optima. Used cars are overpriced now. |
19:48.14 | Katty | i disagree. |
19:48.39 | Katty | i do agree that some people don't know how to properly price a used car. |
19:49.05 | rcaskey | It's not that, it's that if they do they will sell it for more than you should pay |
19:49.14 | rcaskey | because there are too many broke folks that have to buy a used car |
19:50.15 | drmessano | Thats true |
19:50.52 | rcaskey | drmessano, the trick is to find a base Hyundai model - you don't want the GLS |
19:51.34 | drmessano | Well, if we buy new, it's going to be for me.. and she'll get my car. That being said, I would need an SUV or large crossover.. |
19:51.49 | rcaskey | drmessano, kids or tools or dogs or what? |
19:52.03 | drmessano | Tools, equipment, dead bodies |
19:52.13 | rcaskey | no truck? |
19:52.53 | drmessano | Need it covered.. Dont want to deal with a pickup with some add-on cover... though I had considered it once |
19:53.17 | rcaskey | would you want a white panel van? they are cheap |
19:53.32 | *** join/#asterisk jm_s2s (~jm_work_s@static-72-64-129-133.tampfl.fios.verizon.net) |
19:54.13 | drmessano | Nope.. I dont need a work vehicle.. I need a car that can hold my EMS and Fire gear, tools, and occasionally large pieces of equipment |
19:54.49 | rcaskey | how large? would a small SUV be big enough? |
19:54.59 | *** join/#asterisk chasing`Sol (~cS@41.232.122.158) |
19:55.04 | drmessano | Yeah, like I said.. SUV or large crossover |
19:55.32 | rcaskey | I'd look at hte Kia Sorento and the Hyundai Santa Fe |
19:56.19 | drmessano | I was thinking about a Sportage or a Sorento |
19:56.22 | rcaskey | if I was buying for my wife however I'd buy the Murano because it's pretty |
19:57.23 | rcaskey | if you can go that small you might see if you could do with an Elantra Hatchback, the seats fold down |
19:57.39 | drmessano | She's getting a Rondo in the deal.. Which kinda stinks.. She doesn't need that kind of space. She really needs some little thing she can drive back and forth to work and the mailbox. Just a pain to find something that isnt crap |
19:58.18 | drmessano | We BOTH don't want to buy her a new car.. But almost have no choice but to give her the Rondo |
19:58.33 | rcaskey | if shes doing low MPG a 15 year old crown vic in good condition will probably keep runnin |
19:58.39 | jpsharp | A minivan. I love our Dodge Caravan. |
19:59.07 | rcaskey | Not touching a Dodge |
20:02.22 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
20:02.29 | drmessano | I thought about buying an older camry, but yeah, the pricing is insane |
20:02.52 | drmessano | "Blue book is $4000, but I put a new head gasket on there, so I want $6500" |
20:02.59 | ThinkGNU- | I have over 200k on my 2003 Chryser(dodge) minivan |
20:03.04 | [TK]D-Fender | rondo is actually pretty shitty on storage... |
20:03.10 | ThinkGNU- | Chrysler* I meant |
20:03.18 | drmessano | I have fit a LOT in my Rondo |
20:03.23 | *** join/#asterisk acidjazz (acidjazz@notchill.com) |
20:03.31 | [TK]D-Fender | impractical width, etc. |
20:03.39 | drmessano | Well, yes |
20:03.41 | acidjazz | what is the best service these days out there for bulk sms/phone calls (automated not people) |
20:03.49 | [TK]D-Fender | maybe witht he rear seat folded out, but base... blows |
20:03.49 | drmessano | It's defintely not EASY to pack, at all |
20:04.27 | [TK]D-Fender | Because of all the musical gear I bring around I needed to be abe to fit my keyboard hidden well away in the trunk. |
20:05.05 | drmessano | I wont let my wife pack it up when we go on trips.. She leaves it all outside and I make it all fit. Just used to how to maneuver crap in there |
20:05.06 | [TK]D-Fender | Ended getting a 2004 Altima. a decade worth of that family was well designed for port access size. |
20:05.13 | ThinkGNU- | Ford is building some really good vehicles now |
20:05.18 | drmessano | Nice |
20:06.11 | UnixDev | ford: fix or repair daily |
20:06.22 | ThinkGNU- | old thought there UnixDev |
20:06.33 | ThinkGNU- | Ford's stuff, as well as Chrysler's stuff has gotten really good recently |
20:06.37 | jaytee | ford: first on race day |
20:06.51 | UnixDev | found on road dead |
20:07.44 | ThinkGNU- | the new ecoboosts are pretty badass |
20:08.18 | UnixDev | all new cars a nice.. i just don't personally like ford.. but you are right about them getting better recently |
20:08.41 | acidjazz | hey all what is the best service these days out there for bulk sms/phone calls (automated not people) |
20:08.47 | ThinkGNU- | I didn't like fords before but their stuff is improving so much |
20:08.54 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
20:10.08 | citywok | Is there a way to set SQL server options using res_odbc/cdr_adaptive_odbc? like SET ANSI_NULLS, etc. |
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20:15.53 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.6.0 (2012/07/10), 1.8.14.0 (2012/07/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
20:20.46 | *** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net) |
20:21.51 | Nivex | ETA for new packages on http://packages.asterisk.org/ from today's release? (looks like pabelanger is the maintainer?) |
20:22.51 | mjordan | Nivex: given that I just hit the "send e-mail" button, some patience would be appreciated :-) |
20:23.25 | Nivex | mjordan: most impressive :) though the package that's in there is still 1.8.11.x, so... |
20:23.44 | mjordan | for debian, yes. pabelanger does not maintain the debian/ubuntu stuff for Asterisk |
20:24.22 | Nivex | ah, so he's been framed :P (looking at the Maintainer field in Packages) |
20:24.55 | mjordan | Nivex: that should be updated :-) |
20:25.40 | mjordan | Nivex: what link are you looking at that shows that? |
20:25.47 | Nivex | http://packages.asterisk.org/deb/dists/squeeze/main/binary-i386/Packages |
20:25.54 | mjordan | ah |
20:26.02 | mjordan | well, that is accurate as of 1.8.11.1 |
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20:41.09 | pabelanger | Nivex, Ya, Digium no longer signs my paychecks, so I've had to let them slide. However, if you are experienced in maintaining them, I'm sure access could be provided. There are some technical hurtles but nothing serious |
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20:51.33 | [TK]D-Fender | pabelanger, How long have been out? |
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20:52.30 | pabelanger | 3months? |
21:01.00 | [TK]D-Fender | ah |
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21:28.22 | Sean-Der | What is a good way to dump incoming traffic? I having a server set up in a DNAT, I have my externip set, nat=yes, qualify=yes but I can't seem to get anything to happen |
21:28.50 | Sean-Der | I tried to use tcpdump to dump all trafic, but I don't see any requests from a softphone coming in |
21:29.44 | Sean-Der | Is there a better way to debug this? I have a feeling that my traffic is not even making it to the server, it would just help my process if I could be sure I was at least getting the invites coming in |
21:29.59 | Sean-Der | I also have localnet set |
21:36.00 | Sean-Der | I also have sip debug set on |
21:36.08 | Sean-Der | And not seeing anything? |
21:38.01 | [TK]D-Fender | You settings hardly matter for seeing nothing. that means packets just aren't reaching you at all |
21:46.12 | Sean-Der | [TK]D-Fender: Am I doing it correctly though? I was doing a tcpdump on 5060-5070, will that catch the traffic. I have sip bound to 5060 and my softphone sflphone has just the host and auth details |
21:48.36 | [TK]D-Fender | All you should need is "sip set debug on |
21:49.18 | Sean-Der | Yep thats what I did. Thanks for the help [TK]D-Fender I am gonna mess with my router some more I am just learning as I go |
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22:03.19 | m_tadeu | hi everyone...I'd like to know something about the bridge event for the manager...is the callerid1 always the caller and the callerid2 always the receiver? |
22:04.36 | mjordan | they're not the caller ID |
22:04.43 | mjordan | er, nm |
22:04.44 | mjordan | :-P |
22:06.13 | mjordan | m_tadeu: that isn't anything that enforces that relationship |
22:06.59 | mjordan | in general, as its currently implemented, callerid1 will refer to the caller, and callerid2 will refer to the callee |
22:07.34 | mjordan | but it might be dangerous to make that assumption, as that's currently an implementation decision. What you do know is that two channels have been bridged, who those channels are, and their caller IDs |
22:09.24 | m_tadeu | mjordan: thanx for the explanation. My problem here is that I need to know who is receiving the call. is there a way to know this on a manager? |
22:10.25 | mjordan | I'd listen for a Dial event in conjunction with a Bridge event |
22:10.36 | mjordan | the Dial event will guarantee who the caller is in that relationship. |
22:10.53 | mjordan | When the Channel in a Dial event is in a Bridge event, you know that the other channel was the callee. |
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22:14.00 | m_tadeu | I was hoping for a more explicit way to do it :) but that will do the job |
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