| 00:01.37 | *** join/#asterisk brack101 (~dude@66-189-216-156.dhcp.yakm.wa.charter.com) |
| 00:01.41 | brack101 | Hey |
| 00:01.46 | brack101 | Anyone tried https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ? |
| 00:03.02 | atan | brack101, I was about to =D |
| 00:03.08 | brack101 | Cool |
| 00:03.18 | brack101 | I'm trying to decide whether to pay for a pbxes account to get that |
| 00:03.29 | brack101 | Right now I have gizmo5 and it works ok but there's a couple things that suck about it |
| 00:04.01 | brack101 | Not being able to dial out for instance |
| 00:14.02 | *** join/#asterisk tmberg (tmberg@unaffiliated/tmberg) |
| 00:16.00 | *** join/#asterisk trollboy (~trollboy@wsip-24-234-128-35.lv.lv.cox.net) |
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| 00:26.05 | atan | wtf? chan_sip.c:13413 check_auth: username mismatch, have <222>, digest has <s> |
| 00:33.15 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:34.18 | atan | chan_sip.c:21219 handle_request_invite: Failed to authenticate device "222" |
| 00:34.33 | atan | pfft. I beg to differ. =| it connects just calls don't work :P blahhhh |
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| 01:23.41 | zleslie | I am compiling from source now, and I get the make menuselect, and I want to enable a module in there. It tells me that AJI depends on iksemel, so I yum install iksemel, then ./configure agains. Stills says depends on. What gives? |
| 01:24.26 | WIMPy | Maybe there's a -dev package you need? |
| 01:25.32 | zleslie | ahh, right. Libraries |
| 01:27.18 | ChannelZ | headers and link libraries |
| 01:28.07 | zleslie | do most people just make install, or roll packages? |
| 01:28.25 | ChannelZ | make install |
| 01:29.17 | zleslie | Anybody run asterisk at home? |
| 01:29.26 | DeHackEd | <-- |
| 01:29.44 | ChannelZ | ... me does |
| 01:29.53 | zleslie | ncie |
| 01:30.35 | zleslie | Asterisk seems pretty light weight. Is it better to run as a dedicated system or appliance like a firewall? |
| 01:30.43 | zleslie | Currently I have it running in a vm |
| 01:31.02 | DeHackEd | it needs a responsive CPU. that's really the big issue |
| 01:31.19 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:31.22 | ChannelZ | If you're not handling hundreds of calls it's not a big deal |
| 01:32.09 | zleslie | I have been wanting to build asterisk for years, but never got around to it. Now I am in a position where I do not like my current voip carrier for work and thinking if it can be replaced with asterisk |
| 01:32.25 | zleslie | rather, if I have the patience and the will to replace it |
| 01:32.25 | ChannelZ | You wouldn't be the first |
| 01:32.37 | zleslie | lots of learning involved |
| 01:32.46 | *** join/#asterisk saizai (~saizai@unaffiliated/saizai) |
| 01:32.47 | zleslie | and I really dont want to get some web ui |
| 01:32.59 | saizai | how much RAM should I be expecting asterisk to take when idling / extra per call? |
| 01:35.02 | atan | Okay. Two phones, and the calling phone sends video to the called phone .But the called phone cant send video back to it... ? Is there something I might have forgotten here |
| 01:35.29 | atan | http://pastie.org/1371705 is my config |
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| 01:37.21 | zleslie | atan: just currious, which client? |
| 01:37.33 | atan | Nortel IP1535 |
| 01:43.51 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 01:44.09 | atan | Err, found a hitch on the phone |
| 01:47.22 | Silicium | hmm general question |
| 01:47.36 | Silicium | is it possible to use a ADSL Modem for providing DSL? |
| 01:48.10 | Silicium | so if i have 200m phone cable, can i just use two modems and two PPPoE devices (cleint and server_ |
| 01:48.13 | Silicium | ? |
| 01:48.21 | WIMPy | Silicium: No |
| 01:48.29 | Silicium | damn |
| 01:48.37 | Silicium | what do i need? |
| 01:48.38 | DeHackEd | you can for SDSL modems, but not ADSL. you need a DSLAM |
| 01:48.45 | Silicium | dslam is to expencive |
| 01:48.55 | WIMPy | You can get modem pairs to do that however. |
| 01:49.14 | Silicium | hmm ok |
| 01:49.48 | WIMPy | DSLAMS need not be expensive any more, but for a single line a pair of modems might be more sensible. |
| 01:50.14 | Silicium | i need it for my private environment :) |
| 01:50.14 | DeHackEd | what situation are you in that you have 200m of phone cable but couldn't get cat5? |
| 01:50.51 | Silicium | i want to terminate my ADSL connection in the basement |
| 01:51.00 | Silicium | and then to pppoe again to my flat |
| 01:51.36 | WIMPy | You are 200m above the basement? |
| 01:51.42 | Silicium | no :) |
| 01:51.49 | Silicium | maybe 100 |
| 01:52.13 | Silicium | the entry from is in another building |
| 01:52.35 | WIMPy | And your DSL wouldn't take the extra 200m? |
| 01:52.47 | Silicium | i loosing 2mbit |
| 01:53.12 | Silicium | also it would be cool to trunk in the basement |
| 01:53.28 | *** part/#asterisk JRCormier (sir@142.167.208.32) |
| 01:53.36 | Silicium | its just an idea :) |
| 01:54.59 | WIMPy | DeHackEd: Are you sure that works with SDSL modems, BTW? |
| 01:55.21 | WIMPy | The old ADSL modems sooem to do SDSL as well. |
| 01:55.28 | DeHackEd | I work for an ISP. might vary by manufacturer but we have identical modems on each side, so long as they're configured right |
| 01:55.33 | WIMPy | seem |
| 01:55.36 | DeHackEd | then again these things look old |
| 01:56.04 | WIMPy | I guess the problem is tat you usually have no way to configure them. |
| 01:56.43 | Silicium | haha |
| 01:56.45 | Silicium | ebay: |
| 01:56.52 | WIMPy | Or at least no documentation available. |
| 01:56.56 | Silicium | did you mean "islam"? |
| 01:57.31 | WIMPy | Have faith in your data. |
| 01:58.31 | ChannelZ | And if it strays, blow up the shit out of it |
| 01:59.20 | WIMPy | ChannelZ: You misunderstood. That's just pushing to make it go faster. |
| 01:59.57 | ChannelZ | How many AAIIIAAIIAAIIIAAIIIAIIAIII's does it take for gigabit? |
| 02:00.00 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
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| 03:02.55 | shmaltz | :-h |
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| 03:07.44 | WIMPy | Silicium: I just read about 10base-S. |
| 03:08.00 | Silicium | mh |
| 03:18.53 | zleslie | Anyone use google voice with asterisk? |
| 03:19.44 | zleslie | I am able to dial out calls, but I don't know how to make google let me receive calls. Which number do I dial? |
| 03:21.05 | drmessano | The google voice number |
| 03:21.57 | zleslie | then do si Configure google for forward to chat? |
| 03:22.52 | drmessano | yes |
| 03:24.50 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 03:25.26 | zleslie | Will I need to forward ports through the firewall? |
| 03:26.31 | *** join/#asterisk Leddy (leddy@aqua.surgehost.net) |
| 03:28.37 | drmessano | No |
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| 03:53.34 | zleslie | When I call my google voice number, it runes like three times before my logs say that it answered the call. Then once it answers, it rungs two more times, and then google sends the call to google voicemail. I never see the call come in on my extension though |
| 03:54.16 | zleslie | http://pastebin.com/M2d7JtQi |
| 03:55.22 | drmessano | zleslie, are you running 1.8.1? |
| 03:55.55 | zleslie | Asterisk SVN-branch-1.8-r298099 |
| 03:56.00 | zleslie | nope |
| 03:56.24 | zleslie | Should I be |
| 03:56.27 | zleslie | ? |
| 03:57.53 | drmessano | No, you need post-1.8.1, which you have |
| 03:59.01 | drmessano | Why don't you remove the SendDTMF and test |
| 03:59.09 | dan__t | hrm. so would I be wise to create a macro for voice mailboxes? Assigning an extension of 9NNNN or something, where NNNN is their regular extension? |
| 03:59.28 | zleslie | what is SendDTMF? |
| 03:59.52 | drmessano | The 4th line in your pastebin |
| 04:00.02 | zleslie | yes, but what does it do? |
| 04:00.06 | drmessano | Well, 5th.. but anyway |
| 04:00.33 | zleslie | for a config change, is service asterisk reload sufficient? |
| 04:00.34 | drmessano | it sends a DTMF digit |
| 04:00.41 | drmessano | dialplan reload |
| 04:11.44 | zleslie | failing back to exten 's' <-- this is expected, correct? |
| 04:12.54 | zleslie | I see the call come in noticable faster now. |
| 04:18.55 | zleslie | Spawn extension (local, s, 5) exited non-zero on 'Gtalk/+1293874923874" |
| 04:19.32 | zleslie | How can I tell why? The dialplan seems to answer the call, but does not Dial the sip user I specify. |
| 04:21.34 | drmessano | I guess you could paste more of your dialplan and we could look.. Like extensions.conf, sip.conf, with passwords masked |
| 04:23.03 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 04:25.18 | zleslie | http://pastebin.com/NXQ3FFRN |
| 04:31.54 | drmessano | SIP/zach instead of SIP/zach@local for your S destination |
| 04:32.28 | zleslie | Don't I tell it what context its in? |
| 04:34.38 | zleslie | If I make the change on like 19 like you suggest, it only works if I also comment out line 15, where teh gtalk_incomming is declared. |
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| 04:57.25 | *** join/#asterisk `Onyx (~Bowowow@unaffiliated/onyx/x-893823) |
| 04:58.21 | `Onyx | Hi. I have fail2ban running, and it's working. However, I've been asked if there was any possible way to hit back at people who attempt to hack the PBX? |
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| 04:58.54 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:58.55 | `Onyx | Or, is there any way that I could change the PBX so that it's not open to random scans from people |
| 04:59.44 | WIMPy | That obviousely depend on what you want it to do. |
| 05:00.47 | `Onyx | Well he wants it to be detrimental to their system |
| 05:01.00 | `Onyx | I think it'd be better just to avoid it all |
| 05:01.11 | `Onyx | And make it invisible to scans |
| 05:01.13 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 05:01.40 | WIMPy | If you want to be reachable by everyone there is no hiding. |
| 05:02.05 | WIMPy | If you don't, don't. |
| 05:02.52 | `Onyx | Well is there anything you could suggest? |
| 05:03.29 | WIMPy | changing the port. only allowing known peers via iptables. |
| 05:04.10 | `Onyx | Pretty sure that's already been done |
| 05:04.27 | `Onyx | Well yeah, I'm certain it's been done |
| 05:04.46 | WIMPy | So why do you get scanned then? |
| 05:05.12 | `Onyx | We get scanned, but we don't get hacked |
| 05:05.24 | `Onyx | They try to |
| 05:05.34 | `Onyx | But they can't because of the port change I'd assume |
| 05:05.48 | WIMPy | If you don't allow connections from everywhere the is nothing to scan. |
| 05:06.17 | WIMPy | So what kind of scan are you talking about? |
| 05:08.14 | `Onyx | Forgive me for my lack of knowledge here... but the kind of scan that people would do do see if your system was open to exploit and make phone calls using your box |
| 05:08.43 | WIMPy | There is only one thing to avoid that: Pull the plug. |
| 05:08.56 | `Onyx | That's pretty much what I thought |
| 05:08.58 | ChannelZ | I usually see people trying to register as 1000s of extensions |
| 05:09.39 | `Onyx | So basically, what my boss should do is just let it go, and be happy that fail2ban is working? |
| 05:10.04 | `Onyx | There's no extra steps that could be taken (apart from things like turning it off, etc) |
| 05:10.24 | WIMPy | If you don't need to be reachable by eveyone, you don;t need fail2ban. |
| 05:10.34 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 05:11.04 | ChannelZ | I don't think you ever really answered that question - is your Asterisk publically accessable for a reason? |
| 05:11.05 | `Onyx | We do indeed need it |
| 05:11.10 | WIMPy | And if you do, there's the usual argument of the potential to shoot yourself ion the foot with it. |
| 05:11.31 | `Onyx | Yes, ChannelZ, it is |
| 05:11.45 | ChannelZ | Then there's not much you can do beyond what you are. |
| 05:12.03 | `Onyx | Thought that was the case |
| 05:12.07 | `Onyx | Cheers for the affirmation |
| 05:12.10 | ChannelZ | "Striking back" at these hackers might feel good but it's just a waste of your own time and bandwidth |
| 05:12.19 | `Onyx | That's exactly what I said |
| 05:12.26 | ChannelZ | If after a few failed attempts they get blocked by the firewall, they give up and move along |
| 05:12.36 | `Onyx | And it potentially will make things worse |
| 05:12.46 | WIMPy | You could use something a little more clever thatn fail2ban. but indeed, you will have to live with ppl trying to abuse your service in the end. |
| 05:13.21 | `Onyx | What other programs are there? |
| 05:13.34 | WIMPy | nfi. I did my own. |
| 05:13.42 | `Onyx | Oh right |
| 05:14.37 | WIMPy | And I always thought it was a little shitty because it might take a few ms to react. But compared to fail2ban is seems ace :-) |
| 05:15.18 | `Onyx | Is yours a public program? |
| 05:15.46 | WIMPy | I thought about sending unknown packets to userspace for approval. But I didn;t find it important enough so far. |
| 05:16.15 | `Onyx | Fair enough |
| 05:16.36 | `Onyx | Well either way, that's pretty cool that you made your own |
| 05:16.37 | WIMPy | No, and as the whole programm isn't much more than custom configuration that doesn;t seem to make much sense. |
| 05:16.41 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 05:17.39 | WIMPy | The important ting about any firewall solution is to configure it to your usage scenario. |
| 05:19.08 | `Onyx | Yeah of course |
| 05:27.25 | ChannelZ | and to use real fire |
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| 05:41.17 | *** join/#asterisk Jinxed- (c3e5f234@gateway/web/freenode/ip.195.229.242.52) |
| 05:42.35 | Jinxed- | any ideas on what could cause something where a sip device is able to make calls out, but when you try to call the device it is busy? |
| 05:43.05 | *** join/#asterisk glam (~glam@183.13.27.4) |
| 05:43.40 | WIMPy | The registration failed? |
| 05:45.45 | Jinxed- | WIMPy: what is the command sip show registration |
| 05:45.48 | Jinxed- | or something like that |
| 05:46.04 | WIMPy | Sip show peers |
| 05:46.13 | Jinxed- | hmm alright |
| 05:46.21 | WIMPy | sip show registrations is for outbound registrations. |
| 05:47.09 | Jinxed- | WIMPy: do you know a good resource for asterisk debugging? |
| 05:47.11 | kaldemar | Jinxed-: is it really busy? what do you see in the CLI when calling the device? |
| 05:48.47 | Jinxed- | not sure... im not on location and they don't have internet there right now |
| 05:48.58 | Jinxed- | so i was going to download some info and try and see if me and my buddy can figure it out |
| 05:49.40 | kaldemar | all you can do is guess if you can't access the system. |
| 05:50.39 | Jinxed- | kaldemar: I will be able to access the system when i get there |
| 05:50.49 | Jinxed- | i just want to know what to look for before i do |
| 05:51.37 | Jinxed- | it is a simple network we are just learning asterisk with |
| 05:51.43 | Jinxed- | with like 3 or 4 phones |
| 05:55.39 | Jinxed- | is there a good resource or anything with how to debug simple problems |
| 05:56.21 | WIMPy | Turn up verbose at the *CLI and see what it has to tell. |
| 05:57.36 | *** join/#asterisk uzy (~me@unaffiliated/uzy) |
| 05:59.56 | Jinxed- | so just do sudo asterisk -vvvvvvvvvvvvv fr |
| 06:00.02 | Jinxed- | sorry sudo asterisk -vvvvvvvvvvvvvvvv r |
| 06:00.03 | uzy | morning all. a little off-topic but google hasn't been much of a help. Thought some telecom gurus could come handy. Need to know what is a HALF-CIRCUIT ? is it a leased line kind of thing or what ? |
| 06:01.02 | *** join/#asterisk t3hrealadamd (6c117281@gateway/web/freenode/ip.108.17.114.129) |
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| 06:08.07 | glam | uzy: fractional......depending on how many channels you're buying off an T1 or E1 |
| 06:09.36 | uzy | "For international call centers, bandwidth is available on half-circuit basis as well as on public Internet backbone" hmm. I am trying to make sense of you definition in context of this statement. |
| 06:12.21 | uzy | glam: We use E1's. So your what ur saying is if I buy half the channels of E1 ~1Mb. that would be a half circuit thing ? |
| 06:13.45 | glam | uzy: i've never come across that term, but 'fractional' T1 or E1 means you don't buy the full capacity of the entire trunk but ie. 20 channels off an E1 or 15 channels off an T1 |
| 06:14.52 | uzy | glam: okay. agreed. now you know what I was a bit confused with that statement.. thanks. |
| 06:15.27 | glam | uzy: best to check with your carrier on what it means.... |
| 06:15.37 | uzy | yeah. okay |
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| 06:16.29 | b14ck | Hi all :o |
| 06:16.56 | t3hrealadamd | This may sound dumb, but if you have two DHCP servers on the same subnet with different scopes, one with reservations for the phones and one without, how do you guarantee the phones will get the IP from the right DHCP server? |
| 06:17.25 | t3hrealadamd | As in, if you have reservations for phones on one, but not the other, how do you guarantee the phones won't grab an address from the one with no reservations? |
| 06:17.40 | WIMPy | You don't |
| 06:17.46 | ChannelZ | don't make the pools cross |
| 06:17.59 | glam | t3hrealadamd: that's why there's VLAN |
| 06:18.06 | ChannelZ | and don't run two DHCP servers on the same subnet |
| 06:18.17 | t3hrealadamd | if the scopes don't cross it's fine |
| 06:18.27 | t3hrealadamd | I suppose VLANs may be the way to go |
| 06:18.33 | WIMPy | Don't have more than one DHCP server on one network unless they mirror each other or you can make one ignore requests the other would anser. |
| 06:18.40 | ChannelZ | (though depending on what you're running there are some settings for 'backup' DHCP servers, where if one notices a client not getting a response, it will jump in) |
| 06:18.52 | WIMPy | Why do you want two DHCP servers? |
| 06:19.16 | t3hrealadamd | one on the router for PCs, one locally on the asterisk box for phones |
| 06:19.49 | WIMPy | Configure one of them to do what you want and shut down the other. |
| 06:20.45 | t3hrealadamd | Well, the one that hands out DHCP for the computers is just a simple home router. I don't think they'll want to shut it down in lieu of another. |
| 06:20.56 | t3hrealadamd | It's probably going to have to be static on the phones or VLANs |
| 06:21.17 | t3hrealadamd | No problems, just wanted to know if that was possible. Thanks for the input. |
| 06:21.28 | WIMPy | It's not your router then? |
| 06:21.33 | t3hrealadamd | No, it is not. |
| 06:21.52 | WIMPy | ugly |
| 06:21.57 | t3hrealadamd | But we're a single home network spread across three apartments, and we're going to put in VoIP phones to save on text messages and phone calls, etc. |
| 06:22.07 | t3hrealadamd | The three apartments are not conjoining. |
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| 07:42.26 | schmidts | good morning |
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| 07:57.10 | dan__t | 'morning. |
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| 07:59.06 | dan__t | I'm starting to dive in to setting up voicemail. I could set up each individual voice mailbox, but is it more common practice to maybe create a macro or set and incorporate the extension number in to the voicemail context itself? So say I have extension 123, and I wanted that extension to have a voice mailbox of 7123, can I effectively use 7${EXTEN} in a macro or set? I might not be explaining this example with perfect syntax, but I can figure that |
| 07:59.06 | dan__t | out eventually - however, is the logic sound? Is this "how it can be done"? |
| 08:00.45 | dan__t | guess that would be ${MACRO_EXTEN} etc etc |
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| 08:11.40 | kaldemar | dan__t: no need for macro there. |
| 08:11.51 | dan__t | no? |
| 08:14.54 | dan__t | why not? |
| 08:15.17 | kaldemar | because you can do it with an extension without a macro. |
| 08:15.53 | dan__t | Guess I didn't want to have to explicitly define each mailbox by hand so long as I knew the extension of a keyset. |
| 08:16.13 | dan__t | Keyset having an extension of 123, I want to assume that the mailbox will always be 7123 etc etc |
| 08:16.38 | kaldemar | you can do pretty much whatever you want. have 9XXX as the voicemail number and take the XXX as the mailbox number, have a static number and use caller id for the mailbox, take it from a config or whatever. what do you want to do? |
| 08:17.02 | dan__t | oh, right |
| 08:17.25 | dan__t | I guess, "I don't know". I'm going through examples to see what's actually *possible*, but you're right, anything is possible. I'm kind of overwhelmed. |
| 08:17.45 | dan__t | So setting a simple goal of creating an implied mailbox based on a given extension was one goal |
| 08:18.01 | dan__t | But I didn't think about incorporating extension pattern matching, that's a good way then. |
| 08:18.26 | dan__t | _9${EXTEN} etc etc |
| 08:19.07 | *** join/#asterisk Dovid (~Dovid@96.242.61.139) |
| 08:20.38 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:24.14 | dan__t | alright, I'll run with that, thank you. |
| 08:25.16 | dan__t | re: CDR, if I use service from a VoIP provider or something, do they bill me using billsec or duration? I know billsec is actual 'billsec', but do they just give me the difference in 'duration' for free? How does that work? |
| 08:25.33 | dan__t | I guess to ask if *I* was billing someone, which would I use? duration or billsec? |
| 08:27.17 | *** join/#asterisk SeTTleR (~bernd@p5DDEE38F.dip.t-dialin.net) |
| 08:30.23 | joeyjones | dan__t: i'm not sure what those are, but it's billed per minute, so it's likely to be cieling(seconds/60) |
| 08:31.21 | dan__t | oh ok, understood. |
| 08:32.16 | dan__t | ALright I gotta crash out before I don't wake up for work tomorrow. Thanks again for the help. Have a good one. |
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| 08:35.03 | cmnky | what up joeyjones |
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| 08:38.47 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 08:38.58 | joachim_- | Any1 got a working example of using SendFax? |
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| 09:02.23 | dandre | hello |
| 09:03.13 | dandre | does anyone hav experience with LG-Nortel SIP phone serie 8800 and specially 8815? |
| 09:03.56 | *** join/#asterisk Tim_Toady (~moi@77.49.253.53.dsl.dyn.forthnet.gr) |
| 09:07.48 | DND | guys anyone knows about a2billing here? im just wondering if i change the context to a2billing, i cannot use asterisk shortcuts like call transfer(##) and call forward(*73) |
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| 09:16.07 | cmnky | anyone know of any good quality, budget priced, bluetooth sets, that are concealed better than the star trek phaser looking things everyone wears ? |
| 09:17.22 | DND | cmnky is it for mobile? |
| 09:17.58 | cmnky | for cell phone / laptop(voip) use |
| 09:18.49 | DND | i just watched "knight and day" movie and this girl is wearing a plantronics headset |
| 09:19.08 | DND | with just a very thin tube for the mic |
| 09:19.23 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 09:19.58 | cmnky | was just wondering if i could get something that worked, that wasnt so obvious, or if i have to walk around in public displaying my borg adapter like everyone else ;) |
| 09:20.28 | DND | if you really want to have a very small headset, look for a headset that can do "Bone-Conducting" |
| 09:20.44 | DND | meaning the headset will listen on the vibrations on your bone near the ear |
| 09:20.51 | cmnky | sure ... but it needs to work ... i have to be able to make calls ;) |
| 09:21.22 | cmnky | surgical implantation of a magnet on my eardrum isnt an option ... that violates the budget priced condition ;) |
| 09:21.35 | DND | http://onlygizmos.com/worlds-smallest-bluetooth-headset/2010/04/ |
| 09:21.48 | DND | no surgery needed for bone conducting |
| 09:22.23 | cmnky | where does it say thats "bone conducting" ? |
| 09:22.55 | DND | that's not bone conducting |
| 09:23.08 | DND | i just searched for the smallest headset :D |
| 09:23.17 | DND | since you wanted concealed |
| 09:23.29 | DND | here's the one i saw in knight and day: http://www.plantronics.com/north_america/en_US/products/mobile/bluetooth-headsets/discovery-975 |
| 09:23.31 | cmnky | i searched for "micro bluetooth" ... apparently i should have picked "mini bluetooth" |
| 09:24.02 | cmnky | yeah, newegg has that one ... plantronics is well known in the phone world |
| 09:24.17 | DND | personally i wanted at least to have the mic stretching out from my cheek a little because i dont want to look dumb thinking i was talking to myself :D |
| 09:24.49 | cmnky | the scare factor of a guy talking to himself strangely appeals to me ;) |
| 09:28.14 | cmnky | i would imagine i could get creative and add my own mic tube to one |
| 09:28.17 | DND | http://www.amazon.com/INVISIO-G5-Compact-Bluetooth-Wireless/dp/B000X747NQ |
| 09:28.57 | petern_ | can't have much of a battery life though |
| 09:29.16 | DND | invisio b3 is also small |
| 09:29.36 | DND | 20 hours talk time is not bad :D |
| 09:30.40 | DND | http://www.amazon.com/INVISIO-Q7-Conduction-Bluetooth-Wireless/dp/B0013LF8DY |
| 09:30.41 | cmnky | http://www.amazon.com/Plantronics-Discovery-975-Bluetooth-Headset/dp/B002JIMZU4/ref=pd_cp_cps_2 |
| 09:30.50 | DND | anyway my query has been buried :D |
| 09:30.53 | DND | will bump it |
| 09:31.12 | DND | guys anyone knows about a2billing here? im just wondering if i change the context to a2billing, i cannot use asterisk shortcuts like call transfer(##) and call forward(*73) |
| 09:31.34 | cmnky | has no clue about a2billing |
| 09:33.13 | DND | cmnky got the links? those are my dream headsets :D small and concealed :D |
| 09:33.30 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 09:33.45 | cmnky | the 975's ? i just pasted it |
| 09:34.31 | cmnky | newegg wanted way more |
| 09:36.23 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-97.dsl.net.pk) |
| 09:37.02 | mechbangirc | hi can i have a local loop sort of trunk. that is one asterisk box with a sip trunk pointing to itself? |
| 09:37.18 | cmnky | DND, invisio M-series are the "Tactical, discrete, bone conductors" |
| 09:38.27 | cmnky | M3 and M4 models are rated down to 60 meters underwater ... now were talking ;) |
| 09:39.00 | petern_ | 2.4GHz underwater? |
| 09:39.23 | cmnky | petern_, i doubt it ... the point is that its waterproof |
| 09:41.24 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:43.13 | Weazel | anyone knows why when i hit * asterisk disconnects the call and doesn't show me anything in verbose ? or Sip debug ? |
| 09:43.35 | Weazel | it seems as if the call was disconnected normally |
| 09:45.40 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 09:45.46 | kaldemar | Weazel: do you have the h option in app Dial? |
| 09:46.09 | Weazel | i might, i see the h on the verbose but i dunno where it set... |
| 09:46.22 | kaldemar | show your dial statement. |
| 09:47.45 | Weazel | noob here, can't really know how to show the dial statement |
| 09:50.40 | kaldemar | the line in your dialplan that dials. something like exten => ...Dial() |
| 09:50.58 | kaldemar | dialplan = extensions.conf |
| 09:52.23 | Weazel | there are a lot of Dial options in there |
| 09:54.29 | kaldemar | did you configure asterisk by hand or use a GUI? |
| 09:55.04 | Weazel | gui |
| 09:55.15 | kaldemar | which one? |
| 09:55.18 | Weazel | freepbx |
| 09:56.20 | kaldemar | ask in #freepbx how to deal with that. |
| 09:57.20 | Weazel | thx |
| 10:06.46 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 10:08.24 | FlashDeluxe | Hi! I have a short questions: if i make a include => context_bla in context_blabla, does asterisk lookup the configured numbers in both contexts? |
| 10:10.19 | kaldemar | yes. first in the current context, then the included one. |
| 10:10.20 | schmidts | an include really means that all extensions from the included context are also available in this context, so the answer is just Yes |
| 10:10.25 | kaldemar | or included ones. |
| 10:10.51 | FlashDeluxe | thanks, thats all i wanted to know :) |
| 10:19.45 | cmnky | hmm ... think i might try this lg HBM-235 headset ... $6.00 after rebate |
| 10:24.13 | hrhrhr | is it me |
| 10:24.22 | hrhrhr | or is the transfer option gone from x-lite 4? |
| 10:31.29 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
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| 10:47.42 | joachim_- | Do I have to use Hylafax for sending fax through an email address? |
| 10:50.46 | *** join/#asterisk Godfather_ (~godfather@193.153.128.7) |
| 10:59.19 | joachim_- | Hmmm.. Then I got another nut to crack! Is there anyway I can tell Asterisk where to look for the digit sounds when setting up Queues? Other than using language= Ive got a Realtime ODBC setup for my Queues, there I can specify the Queue messages, but NOT the Digit sounds. ARGH! |
| 11:00.59 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 11:01.28 | DND | cmnky i think underwatermeans you can use it to communicate provided that your head will be dry the whole time :D |
| 11:02.32 | cmnky | DND, i was thinking more technical, like, 2.4GHz is absorbed heavily by water |
| 11:03.15 | cmnky | but the M3/M4 are just earpieces/mics ... not radios/bluetooth |
| 11:04.37 | FlashDeluxe | hi! i have an application which should log into asterisk via manager.conf. The username and password is correct in manager.conf. But if i try to start the application i get the error utils.c:1178 ast_careful_fwrite: fwrite() returned error: Broken pipe in asterisk |
| 11:04.42 | FlashDeluxe | any suggestions? |
| 11:05.02 | DND | but the earpiece looks tad ugly especially the borwn/orange color :D |
| 11:05.26 | DND | broken pipe = connection? |
| 11:06.14 | cmnky | im going with the $6.00 newegg special for now |
| 11:06.18 | ectospasm | DND: broken pipe is UNIX parlance that some part of an Input/Output pipeline failed, thus the entire command (in this case, the connection) failed. |
| 11:06.35 | FlashDeluxe | mhh but i can ping the server and the username/password is right and there are no firewall rules |
| 11:07.43 | FlashDeluxe | i updated my asterisk server from 1.4 to 1.8 with new dialplan etc, maybe the commands in the php script is deprecated? |
| 11:13.20 | kaldemar | FlashDeluxe: what else do you see in the CLI? |
| 11:13.36 | FlashDeluxe | thats all |
| 11:14.37 | kaldemar | no logon/logoff prints? |
| 11:15.02 | FlashDeluxe | oh |
| 11:15.16 | FlashDeluxe | yes there is something, i will pastebin it |
| 11:15.36 | kaldemar | while you're at it, describe what your script does (or pastebin it too). |
| 11:15.55 | FlashDeluxe | http://pastebin.de/12778 |
| 11:16.46 | FlashDeluxe | heres the script |
| 11:16.48 | FlashDeluxe | http://pastebin.de/12779 |
| 11:17.19 | FlashDeluxe | this is the part which should connect |
| 11:19.00 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e7.bcn.adamo.es) |
| 11:19.44 | FlashDeluxe | the application in general is an application wich has a large telephonebook and if you click on a number, asterisk sould dial the number and connect it with a sip client |
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| 11:49.51 | joachim_- | Hmmm.. Then I got another nut to crack! Is there anyway I can tell Asterisk where to look for the digit sounds when setting up Queues? Other than using language= Ive got a Realtime ODBC setup for my Queues, there I can specify the Queue messages, but NOT the Digit sounds. ARGH! |
| 11:55.03 | Dovid | anyone know the propper filter for wireshrk to sort by protocol sip ? |
| 12:01.49 | *** join/#asterisk Tim_Toady (~moi@188.4.76.200.dsl.dyn.forthnet.gr) |
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| 12:12.45 | tzafrir_laptop | sip? sip+rtp? |
| 12:12.53 | tzafrir_laptop | Dovid, ==^ |
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| 12:28.38 | Faustov | .seen gr0mit |
| 12:28.48 | Faustov | ~seen gr0mit |
| 12:28.59 | infobot | gr0mit <~tim@extrt.txrx.org.uk> was last seen on IRC in channel #asterisk, 103d 22h 25m 9s ago, saying: 'anyone in 3rd Arrondisement who can try and sort out a customer's network for me?'. |
| 12:33.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:39.50 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 12:40.20 | grEvenX | hi, anyone know how to get information about which party hung up the call in dialplan? (HANGUPCAUSE doesn't say anything about which party) |
| 12:42.49 | kaldemar | grEvenX: see Dial option g. |
| 12:43.32 | spenguin[work] | hey, ill be getting an E1 'spillover' line, which would go into the 2nd port of the dualport PRI card |
| 12:43.44 | spenguin[work] | does this dahdi config look ok |
| 12:44.44 | grEvenX | kaldemar: well, how to know which party hang up.... g only means you can execute logic after the call has hung up... |
| 12:45.08 | spenguin[work] | ttp://pastebin.com/Z6N2VueS |
| 12:45.10 | spenguin[work] | oops |
| 12:45.11 | spenguin[work] | http://pastebin.com/Z6N2VueS |
| 12:46.14 | spenguin[work] | currently I have 1 E1 line hooked |
| 12:46.17 | spenguin[work] | which works fine |
| 12:49.56 | kaldemar | grEvenX: well, if the caller hangs up, the execution will not continue in the next priority, right? |
| 12:50.23 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 12:53.34 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 12:54.19 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:54.58 | evangelion | hello, how does codec negotiation work in asterisk 1.6.2? does it loog for supported codec on called side before sending "SIP OK" message to the calling one? |
| 12:59.37 | kaldemar | make a call with sip debug and verbosity enabled and you'll see how it works. it will print capability messages that have the common codecs. |
| 13:01.59 | Chainsaw | Digium, your MySQL on the issue tracker blew up. |
| 13:03.47 | evangelion | kaldemar: if possible, i would prefere to avoid to spend $HOURS on reverse engineering |
| 13:07.21 | kaldemar | it will hardly take hours. more like minutes. |
| 13:08.32 | evangelion | well, no! |
| 13:13.58 | *** join/#asterisk jeffik (~chatzilla@69-165-175-200.dsl.teksavvy.com) |
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| 13:25.02 | Chainsaw | evangelion: Not to mention there's no reverse engineering involved at all. |
| 13:25.22 | Chainsaw | evangelion: Asterisk is an open source product. You can see the C source code for every single thing it does. |
| 13:27.46 | evangelion | Chainsaw: ack! anyway chan_sip.c ( where I SUPPOSE the algorithm is ) is 26250 lines long |
| 13:28.33 | kaldemar | and a sip debug for a call is significantly shorter. |
| 13:29.11 | Chainsaw | kaldemar: Agreed. So multiple avenues have now been presented to solve this problem. |
| 13:29.16 | evangelion | kaldemar: ack, but it would not be accurate enough imho! |
| 13:29.19 | Chainsaw | evangelion: We're here to help you help yourself. |
| 13:29.37 | Chainsaw | evangelion: If you want us to the job for you and rub your back... that costs money. |
| 13:29.42 | Pan3D | rome wasn't build in a day |
| 13:29.49 | Pan3D | built* |
| 13:30.45 | kaldemar | evangelion: in what sense is it not accurate? |
| 13:32.25 | evangelion | Chainsaw, kaldemar thank you for your replies! I understand your points and I think you're both right, anyway i'm looking for an YES/NO answer |
| 13:33.48 | Pan3D | ... |
| 13:33.59 | Pan3D | kids today |
| 13:38.28 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 13:38.44 | schmidts | evangelion your answer is Yes, does this help you? |
| 13:39.19 | nny | assuming wctc4xxp module only needed when you have a transcoding card? |
| 13:41.53 | nny | really odd, just updated asterisk and dahdi (1.4.38 and 2.4.0) on a system, and it drops calls whenever a transfer is attempted, Spawn extension (sip, 13, 0) exited non-zero on 'DAHDI/22-1' |
| 13:45.11 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 13:48.58 | evangelion | schmidts: yes thank you |
| 13:51.43 | evangelion | schmidts: are you sure http://pastebin.com/nqx1CkNr ? |
| 13:51.59 | evangelion | lines in the link above are from sip.conf |
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| 13:56.46 | nny | anyone here that can advise or help with this issue? It's debilitating at this time in the morning, thanks |
| 14:02.26 | joachim_- | Hmmm.. Then I got another nut to crack! Is there anyway I can tell Asterisk where to look for the digit sounds when setting up Queues? Other than using language= Ive got a Realtime ODBC setup for my Queues, there I can specify the Queue messages, but NOT the Digit sounds. ARGH! |
| 14:07.34 | Katty | Dear Universe, you have officially gone Too Far. What were you thinking? 8F degree weather. and snow? have you lost your sanity? This is unacceptable and I will be speaking with your lawyer. That's right, you're going to pay for EVERYTHING, including this cough syrup I just had to buy. PS - do we /really/ have to do this /every/ year? seriously? you just never learn.... Love, Katty |
| 14:08.17 | Nivex | Have you considered moving to a warmer climate? |
| 14:08.54 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 14:12.25 | Katty | yes. |
| 14:12.30 | Katty | i have also considered not bitching about things. |
| 14:12.38 | Katty | didn't like either thought process. |
| 14:12.41 | Nivex | :) |
| 14:17.07 | Jasnejac | Katty: now you can take pretty pictures for your Xmas cards |
| 14:17.40 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:18.38 | Katty | oh yeah |
| 14:18.46 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 14:19.05 | Katty | Dear everyone, wednesday is the VERY LAST DAY to sign up for the christmas card exchange. please see me for details. |
| 14:22.50 | schmidts | evangelion atleast it takes you how long to find this by yourself? |
| 14:24.09 | evangelion | schmidts: ~20 minutes |
| 14:24.25 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 14:25.49 | schmidts | evangelion to be honest there was no sure yes or no answer thats why we told you to lock for it ;) |
| 14:27.57 | evangelion | schmidts: nice! thank you! |
| 14:31.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 14:31.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:31.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:39.09 | *** join/#asterisk boch (c829e3f5@gateway/web/freenode/ip.200.41.227.245) |
| 14:40.35 | boch | hi all |
| 14:41.46 | boch | is there a way to know which channel is an agent going to be linked before answering the call? im listening events on AMI, but the Newchannel with state Ring has no info about the other channel, maybe a channel variable? |
| 14:42.36 | tuxx- | Hi guys, i have added an *code to my features.conf in the context [applicationmap]. But when i do a dial, with option tTkK, and i press the dtmf sequence (*38), it doesnt call the macro i assigned to this feature. How is this possible? Other features in my application map tend to work brilliantly, just this one app with *38 doesn't work properly. Also, i removed the standard *38 for callparking in my features.conf. |
| 14:43.55 | boch | tuxx-: reload config, or check if you set well DYNAMIC_FEATURES variable, and check features show output on console |
| 14:44.59 | tuxx- | just read about DYNAMIC_FEATURES variable, i think thats my bug. Thanks for the fast reply! :) |
| 14:45.47 | tuxx- | works like a charm :-D |
| 14:46.21 | boch | good to know |
| 14:46.53 | joachim_- | How can i execute a command in the dialplan when calling out? I need to send a NoOp when outgoing Call is Answered |
| 14:47.06 | joachim_- | is that possible? |
| 14:47.27 | schmidts | joachim_ take a look at the M option of the Dial app |
| 14:47.42 | boch | joachim_-: Dial() has an option to run a macro when call is answered, check that out |
| 14:47.48 | joachim_- | schmidts: Thank you very much! |
| 14:48.00 | schmidts | your welcome ;) |
| 14:48.18 | boch | schmidts: are you familiar with queues? |
| 14:49.03 | schmidts | boch only a little bit |
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| 15:08.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 15:09.35 | WIMPy | I have an audio problem depending on combination of channeltypes. Does someone have an idea how I can find out what's going wrong? |
| 15:10.45 | WIMPy | I'm using sip, iax and lcr and get gabled audio from lcr to iax, all other directions are ok. It's all alaw only. |
| 15:10.54 | Jinxed- | What can cause 1 way audio? I have a very simple system (basically 4 phones). |
| 15:11.13 | WIMPy | Jinxed-: NAT? |
| 15:11.43 | Jinxed- | WIMPy: no nat used |
| 15:11.55 | Jinxed- | it is basically just the simple type of tutorial setups |
| 15:12.23 | WIMPy | All phones and Asterisk on the same LAN? |
| 15:12.56 | Jinxed- | the phones that aren't working are |
| 15:12.59 | Jinxed- | on the same lan |
| 15:14.57 | WIMPy | And the server? |
| 15:15.05 | Jinxed- | yep |
| 15:15.06 | Jinxed- | same lan |
| 15:15.14 | Jinxed- | all on the 10.1.40.x/24 |
| 15:15.46 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 15:16.47 | leifmadsen | who wants a new set of release candidates for Asterisk?! |
| 15:17.25 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 15:18.02 | leifmadsen | Jinxed-: all SIP? what does sip.conf look like? Have you defined localnet and/or externip? is asterisk listening and replying on the same subnet? |
| 15:18.49 | leifmadsen | that's where I would start looking. I would also do a 'rtp set debug on' to make sure audio is flowing in both directions. That would be dependent upon asterisk being in the media path (which it doesn't necessary need to be) |
| 15:21.27 | Jinxed- | leifmadsen: http://pastebin.com/5FEy90z7 |
| 15:21.53 | leifmadsen | Jinxed-: you should have masked the passwords |
| 15:22.26 | Jinxed- | leifmadsen: it is just a test |
| 15:22.41 | leifmadsen | Jinxed-: perhaps you can try adding canreinvite=no to the top. Also you might want to change bindaddr= to just the IP address on the same subnet as the phones to force asterisk to only listen and respond on a single interface |
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| 15:23.33 | Jinxed- | leifmadsen: well all of my phones work fine except for the 5800M phone |
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| 15:23.48 | Jinxed- | it is the only one that has one way audio |
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| 15:24.27 | leifmadsen | could be a phone configuration perhaps... not sure. |
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| 15:24.43 | leifmadsen | like i said, try rtp set debug on and make sure you get audio flowing through asterisk. Could try canreinvite=no too |
| 15:25.33 | WIMPy | Ok, I have to add lcr-iax-iax gives no audio at all, while lcr-iax-sip works again. |
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| 15:27.10 | Weazel | argh i'm getting the weirdest thing, i have an asterisk machine which is connected via Sip trunk from provider at one and, and to an IAX trunk to another asterisk machine, now whenever an external call goes into IVR it can't hit 3,6,9 dtmf coz it will double them as in 99 33 66... anyone has a clue ? |
| 15:27.22 | leifmadsen | WIMPy: I don't know what chan_lcr is... ? |
| 15:27.42 | WIMPy | leifmadsen: One of the ISDN options. |
| 15:27.50 | bkruse | waves to leifmadsen |
| 15:27.57 | russellb | bkruse: ! |
| 15:28.16 | leifmadsen | bkruse: OMG HAI! |
| 15:28.25 | leifmadsen | WIMPy: ah that's why I don't know it then :) |
| 15:28.26 | WIMPy | lcr-iax-iax-lcr and lcr-iax-iax-sip are woking as well. |
| 15:29.11 | WIMPy | Wait lcr-iax-iax not working at all might be a networking issue. |
| 15:29.17 | bkruse | !!! russellb leifmadsen: How goes it? |
| 15:29.57 | russellb | goooooooood :-) and yourself? |
| 15:30.09 | leifmadsen | bkruse: more good than bad :) |
| 15:30.35 | WIMPy | So it looks like it is only the combination of lcr and iax, while interestingly any further conversion makes it work again. |
| 15:30.52 | WIMPy | That seems pretty strtange to me. |
| 15:31.35 | WIMPy | But I think I had the same or a very similar situation way back when I used misdn. |
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| 15:35.58 | bkruse | I heard the book was going well too! #facebook-stalk. Things are going pretty good over here, just working, the usual :P |
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| 15:40.54 | timholum1 | is it a security risk to have calltokenoptional=0.0.0.0/0.0.0.0 in my iax.conf? |
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| 15:58.27 | mintee | I'm getting errors all the sudden regarding my zonemessages in voicemail.conf |
| 15:58.30 | mintee | but they are default |
| 15:58.35 | mintee | eastern = America/New_York,'vm-received' Q 'digits/at' IMp |
| 15:58.41 | mintee | ... |
| 15:58.43 | mintee | military = Zulu,'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' |
| 15:59.01 | mintee | [Dec 13 10:58:18] WARNING[20160]: app_voicemail.c:8927 load_config: Invalid timezone definition at line 3098 |
| 16:02.11 | mintee | weird... i replaced the comma with a pipe and it works fine. |
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| 16:11.14 | leifmadsen | bkruse: ya going pretty well so far! we're getting reasonably close to done |
| 16:11.32 | leifmadsen | gonna be well over 600 pages this time around (and we removed the appendices) which means we're going to be pretty much double the content over the last book |
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| 16:15.08 | _pepo_ | hi friends |
| 16:15.17 | devmod | Hello, is there a way to send an INFO msg to an existing channel from asterisk ? |
| 16:15.24 | leifmadsen | mintee: that's because pipes have been replaced with commas in newer versions of asterisk (per the CHANGES file) |
| 16:16.07 | leifmadsen | devmod: SendDTMF() ? |
| 16:16.16 | _pepo_ | I am looking for a open sip phone to use with the logo of my telco (under Windows), What do you recommend me? |
| 16:16.42 | devmod | leifmadsen, isnt that restricted to digits + A-D ? |
| 16:17.35 | leifmadsen | devmod: perhaps I misunderstand what you're trying to do -- I was presuming you meant sending DTMF using the SIP INFO method |
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| 16:18.02 | devmod | leifmadsen, Oh I need to send an INFO msg containing 'ALERT' to an endpoint |
| 16:18.05 | leifmadsen | devmod: if you're talking about constructing some other SIP packet, that's not what asterisk does. You'd be further ahead iwth something like sipsak |
| 16:18.26 | leifmadsen | devmod: what is it you're trying to do? |
| 16:18.51 | devmod | leifmadsen, this specific endpoint supports some sort of alerting - lights flash |
| 16:19.07 | devmod | it responds to an INFO msg containing 'ALERT' on it |
| 16:19.16 | leifmadsen | I think you'll have to do that outside of Asterisk |
| 16:19.27 | leifmadsen | you can't just construct a SIP packet and send it from the dialplan |
| 16:19.47 | devmod | leifmadsen, Oh ok thanks |
| 16:19.53 | leifmadsen | try sipsak |
| 16:20.17 | leifmadsen | you could trigger that from an AGI() or something, so then you'd be effectively sending what you want by triggering dialplan that way |
| 16:20.38 | devmod | That would work but I need to insert it into an existing call |
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| 16:22.00 | leifmadsen | devmod: so you'd have to store some information about the call in a channel variable in order to know where to send it and with what information. You could potentially get the data from AMI as well |
| 16:22.14 | leifmadsen | it'll all work, you just need to be clever enough to figure out where to get the data from |
| 16:22.26 | devmod | leifmadsen, I see |
| 16:22.50 | leifmadsen | the AGI() could even be triggered through features.conf if you want to trigger something from the dialpad on your phone during a call |
| 16:22.58 | leifmadsen | it depends what you're trying to do and when |
| 16:23.05 | devmod | right |
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| 16:25.15 | mintee | leifmadsen, yes, i understand that, but the comma's didn't work |
| 16:25.27 | mintee | it doesn't matter now anyway |
| 16:27.56 | leifmadsen | mintee: oh you went the other way -- ya in voicemail.conf there is still some pipe usage for some reason |
| 16:28.09 | leifmadsen | well, I know the reason :) |
| 16:28.15 | mintee | ya, standards... You gotta love them. :P |
| 16:28.27 | leifmadsen | the best part about standards is there are so many to choose from! :) |
| 16:28.35 | leifmadsen | (credit: jsmith) |
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| 16:32.59 | evangelion | does 1.6.2 support the preferred_codec_only feature? |
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| 16:48.07 | ningia | hi all |
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| 17:11.12 | paulc | c |
| 17:17.24 | dan__t | Hello. |
| 17:19.11 | dan__t | I've been reading a lot but might not fully understand how this process works. I've been hacking on this Polycom 601 for a while, and I'd like to make some action happen - another context be used instead of the default context for ringing this phone - when a certain button is pressed on the phone. I'm thinking I want to use that DND button as an example because it works in a similar manner. My question is, when DND is pressed, does asterisk first se |
| 17:19.11 | dan__t | nd the call to the phone and THEN the phone tells asterisk that its in DND mode, or does the phone actively tell asterisk that its in DND mode when that key is pressed? |
| 17:20.25 | dan__t | If the former, does DIALSTATUS just become DONTCALL or something, and I handle that logic in my dialplan? |
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| 17:23.13 | WIMPy | That depends on the phone and how it is configured. And your dilplan. |
| 17:23.25 | dan__t | I could go that route, or just program a button to dial a special extension to enable/disable some functionality in Asterisk. |
| 17:23.32 | dan__t | Ok, I get that. |
| 17:23.48 | dan__t | Its kind of a loaded question - I wanted to figure this out but might not understand it enough just yet. |
| 17:25.06 | WIMPy | Eiter the phone gets the call and replies that it doesn't want it or it could dial some feature code when you (de)activate the feature. |
| 17:25.32 | dan__t | Ok. |
| 17:25.53 | dan__t | Is there a preferred method, or does it again just depend on the phone, how its configured, and my dialplan? |
| 17:26.03 | dan__t | I can program a polycom button to send a feature code, that's no problem. |
| 17:26.35 | WIMPy | You can do it what ever way you like and your phone supports. |
| 17:26.42 | dan__t | Cool. |
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| 17:30.51 | dan__t | I'll play around with it and see what works best. Thank you. |
| 17:49.13 | dan__t | Thanks, WIMPy. |
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| 18:03.34 | bmoraca_work | anyone have experience getting the time to set properly on an SPA 504g phone? No matter what I do, it always ends up an hour fast. am I missing something? |
| 18:11.22 | russellb | does it have a daylight savings option or something? |
| 18:18.12 | bmoraca_work | it does, and I've set it properly, and it does nothing |
| 18:18.21 | carrar | means you are working so fast that you have completed todays work and are working on tomorrows work load |
| 18:18.23 | bmoraca_work | it still has the wrong time. as we're not in DST time right now |
| 18:18.36 | carrar | err next hours work load |
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| 19:13.45 | marl_scot | anyone know why a sip phone would show 'inuse' under core show hints, when it isnt being used? It was working fine for most of the day, but about 1500, it seamed to lock on 'InUse' and rebooting the phone has made no differance |
| 19:14.17 | marl_scot | any know if its posible to get * (1.8) to re-check all the attached sip phones for there hint status? |
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| 19:16.07 | leifmadsen | marl_scot: you have callcounter=yes enabled in sip.conf? |
| 19:16.42 | marl_scot | yes |
| 19:17.20 | marl_scot | strange thing was it was working fine yesterday, and most of today, and all the other phones are working ok |
| 19:24.46 | WIMPy | Maybe 'sip reload' helps? |
| 19:25.03 | marl_scot | nope :( |
| 19:25.27 | WIMPy | Any zombies? |
| 19:25.47 | marl_scot | zombies? |
| 19:26.02 | WIMPy | Undead channels |
| 19:26.36 | marl_scot | am showing one active channel |
| 19:27.00 | marl_scot | strange, as there shouldnt be any calls on the system just now |
| 19:27.31 | WIMPy | Can you 'channel request hangup ...' it? |
| 19:27.33 | marl_scot | and its for the extension that is showing inuse |
| 19:28.04 | marl_scot | didnt hang it up :( |
| 19:28.26 | WIMPy | Bad channel. |
| 19:28.41 | WIMPy | Just out of interest: Where is it stuck? |
| 19:29.22 | marl_scot | what do you mean? as in the 'state' of the channel? |
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| 19:29.35 | WIMPy | Application |
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| 19:29.53 | marl_scot | application says '(none)' |
| 19:29.55 | WIMPy | It must be somewhere in the dislplan if it exists. |
| 19:30.15 | WIMPy | No channel connected to nothing. |
| 19:30.15 | marl_scot | SIP/210-000001dc *8275@internal:2 Up (None) |
| 19:30.31 | marl_scot | where *8XXX calls the pickup app |
| 19:31.16 | marl_scot | exten => _*8XXX,1,Pickup(${EXTEN:2}) / exten => _*8XXX,n,Hangup() |
| 19:32.34 | marl_scot | where : exten => 275,1,Dial(SIP/211&SIP/202&SIP/201) / exten => _2XX,1,Dial(SIP/${EXTEN}) |
| 19:32.48 | marl_scot | could it be because i dont have a 'hangup()' for exten 275 ? |
| 19:33.55 | WIMPy | No, you're in *8... |
| 19:34.22 | WIMPy | So it's Pickup or some other sort of sip glitch. |
| 19:34.34 | marl_scot | ok, just thought as *8XXXX is trying to pickup ext 275 |
| 19:34.36 | WIMPy | But I can't remeber having had a stuck sip channel. |
| 19:34.56 | marl_scot | is there a way to force the channel to be cleared without restart * ? |
| 19:35.48 | WIMPy | ich channel request hangup won;t do it, I think there are no more options. |
| 19:35.55 | WIMPy | if |
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| 19:36.34 | WIMPy | Do yu have 'core show locks'? |
| 19:37.15 | marl_scot | nope |
| 19:37.55 | WIMPy | Yes, that has to be explicitly enabled IIRC. |
| 19:41.27 | marl_scot | just out of interest, when i restart *, its takes some time for the phones to sort themselves out, i take it this is down to the 'session-expires' value? |
| 19:42.10 | WIMPy | Are you using realtime? |
| 19:42.36 | marl_scot | nope, static files |
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| 19:43.19 | WIMPy | Thought so. But in that case they should continue to work immediately. |
| 19:43.51 | marl_scot | ah, posibly something on the phones themselves then :( |
| 19:44.10 | marl_scot | ok, looks like im goingt o have to restart * :( |
| 19:44.55 | WIMPy | It they foind out about the situation at all I'd expect them to retry within a few seconds. |
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| 19:56.06 | jsolares | yikes, 1 call and valgrind says possibly lost: 54,500,602 bytes in 67,947 blocks |
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| 20:09.42 | jsolares | what happens with memory allocated inside an AGI, if the agi leaks does that bloat asterisk's memory usage? |
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| 20:32.02 | marl_scot | one posibility for my stuck channel, may have been a rouge asterisk process running :( |
| 20:32.33 | marl_scot | looks like i may have run /etc/init.d/asterisk start or restart, at some point as a non prov user |
| 20:32.46 | marl_scot | cant be positive, but will keep an eye out :) |
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| 20:57.04 | marl_scot | thank [TK]D-Fender for helping track down my stuck channel problem :) |
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| 21:06.06 | *** mode/#asterisk [+b *!*chatzilla@216.191.106.*] by russellb |
| 21:06.09 | *** kick/#asterisk [[TK]D-Fender!~russellb@asterisk/digium-open-source-team-lead/russellb] by russellb ([TK]D-Fender) |
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| 21:22.34 | afink | Hey guys is there anyway I can have asterisk record verbose level 3 to file? |
| 21:22.49 | afink | for an extended period of time like 24 hrs? |
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| 21:41.42 | citywok | afink: logger.conf ? |
| 21:43.37 | afink | awesome thank you |
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| 21:44.58 | citywok | yw |
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| 21:49.29 | afink | citywok: already had the way I needed plenty of data to work with |
| 21:50.38 | afink | whats the general consensus on upgrading to 1.8? Is it ready yet? |
| 21:51.30 | citywok | i'm using it on one system that's about to go in to production and haven't found any issues yet. |
| 21:52.39 | citywok | well, except queue reload members is broken, but it appears as though it's been like that since 1.6.x |
| 21:53.09 | afink | hmm, ok I'm just doing super basic stuff. So hopefully it should be ok |
| 21:55.24 | marl_scot | im running 1.8 and it looks stable (4 port BRI with 14 SIP clients) |
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| 22:49.00 | atan | Anyone try http://sip-communicator.org/? |
| 22:58.50 | carrar | SPAM |
| 22:58.54 | atan | bs |
| 22:59.03 | atan | LEGIT LINK FOUND ON GOOGLE |
| 22:59.07 | carrar | look at the topic |
| 22:59.07 | atan | now go die, plskthx |
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| 23:22.33 | pabelanger | Can chan_oss work with access to /dev/dsp? |
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| 23:51.01 | voxter | Does anyone have a good reference to how asterisk is doing codec selection? Ive got a setup wherein an asterisk PBX (customer premise) has an extension specifically configured to allow only g.711u, and on its sip "trunk" back to the ITSP(also me) it allows both g729 and ulaw, as most of their phones use g729 normally. |
| 23:51.50 | voxter | I thought that if the ulaw-forced extension made a call, asterisk would attempt to use ulaw the whole way through on further calls, unless it had to fall back, but it appears to be doing ulaw only to that extension, and then picking g729 from the pbx to the itsp, because g729 is listed first on the allow= (i think this is the reason, anyways) |
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