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| 01:53.37 | tonikasch | bye |
| 01:54.48 | drkat | hello |
| 02:05.49 | apb3304 | Hello |
| 02:07.27 | drkat | hows it going |
| 02:07.48 | apb3304 | fine |
| 02:07.51 | apb3304 | how's it going with you? |
| 02:08.06 | drkat | well it's Saturday, but it's going |
| 02:08.19 | apb3304 | yes it is |
| 02:08.48 | drkat | I was reading the asterisk book looks pretty neat |
| 02:09.26 | apb3304 | Great! Maybe you can answer a question for me? |
| 02:09.42 | drkat | Maybe?? heh |
| 02:09.45 | drkat | ill try |
| 02:09.49 | apb3304 | In fact... everyone is invited to try :) |
| 02:10.09 | *** join/#asterisk jsjc (~Adium@139.Red-83-59-182.dynamicIP.rima-tde.net) |
| 02:10.31 | apb3304 | I'm getting a SIP/2.0 484 Address Incomplete error. And, I don't know how to fix it. Any help? Thank you |
| 02:10.44 | drkat | what does you debug look like? |
| 02:11.29 | WIMPy | apb3304: You should find out what a valid number is supposed to look like. |
| 02:12.08 | drkat | whats the invite look like before the 484 |
| 02:12.42 | drkat | and what type of handset? |
| 02:15.05 | apb3304 | invite looks fine to me :) |
| 02:15.28 | apb3304 | plain ol' ordinary POTS phone. |
| 02:16.03 | drkat | what's your call flow look like? who is sending the 484 |
| 02:16.34 | apb3304 | my voip provider |
| 02:17.15 | drkat | Ok so you send an INVITE to a POTS # @ your voip provider.. and then they send back a 484 or is there other messaging in between |
| 02:18.05 | apb3304 | I get an invite... 2 404's not found and 2 acks |
| 02:18.41 | drkat | have you opened a ticket with your sip provider? |
| 02:18.58 | apb3304 | no support... free provider |
| 02:19.11 | drkat | I see.. |
| 02:19.17 | drkat | do other numbers work or is it all numbers |
| 02:19.45 | apb3304 | only have 1 number with that provider.... my other provider works. |
| 02:20.12 | apb3304 | i don't want to use just one provider... as that means 1 point of failure. |
| 02:20.16 | drkat | Ok. Well to be honest without seeing the full call I'm not gonna be much help.. you're getting a 404 not found as well |
| 02:20.34 | apb3304 | different results from different tools |
| 02:20.35 | drkat | How many digits are you sending to the provider |
| 02:20.45 | apb3304 | I don't u nderstand the question |
| 02:20.49 | apb3304 | I dial a DID |
| 02:20.53 | apb3304 | with my POTS phone |
| 02:21.09 | apb3304 | the provider forwards that call to me at asterisk |
| 02:21.22 | drkat | So it's inbound call failure |
| 02:21.27 | apb3304 | yes |
| 02:21.41 | WIMPy | So it is somethign your Asterisk sends to your ITSP on an incomming call? |
| 02:22.08 | WIMPy | Then you need to configure the right extension or register the right extension. |
| 02:22.09 | drkat | it looks like this I think... PSTN -> SIP TRUNK -> ASTERISK -> |
| 02:24.01 | drkat | does that look correct flow apb? |
| 02:24.02 | apb3304 | yes.. |
| 02:24.16 | drkat | Ok so your asterix is sending the 484 back to the SIP provider/> |
| 02:24.17 | drkat | ? |
| 02:25.04 | apb3304 | invite comes from provider |
| 02:25.11 | apb3304 | 404 comes from my asterisk |
| 02:25.36 | apb3304 | address incomplete comes from my provider |
| 02:25.36 | drkat | Ok well 404 not found is from your pbx... |
| 02:26.06 | drkat | so are you sure the number dialed is a valid number setup on your pbx |
| 02:26.35 | apb3304 | well... therein lies part of the problem... no I'm not sure. :) |
| 02:31.57 | apb3304 | getting the address incomplete even without a call |
| 02:36.25 | apb3304 | open to further suggestions... :) |
| 02:37.01 | WIMPy | You need to tell us the full story. including a debug log of a failed call. |
| 02:37.23 | apb3304 | Happy to |
| 02:38.00 | drkat | yeah were gonna need more info |
| 02:38.24 | apb3304 | except it's not just a failed call.... as mentioned I get the address incomplete error with no call whatsoever |
| 02:39.02 | drkat | are you getting the 484 unsolicited? |
| 02:39.15 | WIMPy | That might be just an options packet that can safely be ignored. |
| 02:40.38 | apb3304 | Yes, it is in fact an options packet going by... those come when I have qualify = yes and go away when I have qualify = no. |
| 02:41.22 | WIMPy | That's abolutely ok. Asterisk only cares for the fact that it receives a reply, not what it looks like. |
| 02:41.41 | apb3304 | Not sure which way it's supposed to be set... someone helping me more or less insists it should be set to yes. |
| 02:41.52 | apb3304 | I on the other hand think no is more correct. |
| 02:41.58 | apb3304 | But he's the expert, I'm a beginner |
| 02:42.01 | apb3304 | OK |
| 02:42.19 | apb3304 | then that leads to the next roblem |
| 02:42.20 | WIMPy | If you set it to yes, Asterisk will send an options packet regularly to check if the peer is alive. |
| 02:42.29 | apb3304 | O I C |
| 02:42.40 | apb3304 | well it's already re-registering every 2 minutes |
| 02:42.47 | apb3304 | is that normal? |
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| 02:43.29 | WIMPy | Seems to be quite often. But maybe they think it's a good idea for users with dynamic IPs. |
| 02:43.57 | apb3304 | I see. Interestingly, my dynamic IP seems to be pretty static :) |
| 02:44.10 | WIMPy | But with that register interval, an extra qualify seems rather pointless. |
| 02:44.23 | apb3304 | That's what I was thinking |
| 02:44.31 | apb3304 | But had no experience to know any better |
| 02:44.57 | apb3304 | So, the real problem is that I get a busy signal when I try to call in. |
| 02:45.23 | drkat | which is your 404 |
| 02:45.25 | WIMPy | Yes, 404, so the extension they tried to send the call to doesn;t exist. |
| 02:45.44 | drkat | Do you have a translation between the extension and the DID? |
| 02:45.46 | WIMPy | But you should get a message saying exactely that along with what was sent. |
| 02:47.19 | apb3304 | I believe I do have a xlation between ext and DID... but not sure it's right. |
| 02:47.39 | apb3304 | message saying... what now? |
| 02:47.51 | chris_n | with "qualify=yes" is "SIP/2.0 503 Unable to load gateways" a valid answer to an OPTIONS packet? |
| 02:48.52 | chris_n | it seems that shortly after this I get " NOTICE[4101] chan_sip.c: Peer 'foobar'' is now UNREACHABLE! Last qualify: XX" |
| 02:51.18 | apb3304 | is anyone able to use antisip as a trunk? |
| 02:52.39 | WIMPy | I like that name. |
| 02:54.30 | apb3304 | i believe there's no transalation since we are using g711 only |
| 02:55.42 | drkat | if I send an invite to to a 10 digit number and the SIP provider passes that call to your PBX and you only have say x444 built.. you need to map the DID to the extension in order for the call to be setup |
| 02:59.19 | apb3304 | you're referring to the dialplan |
| 02:59.43 | apb3304 | inbound route is set to any/any |
| 02:59.54 | WIMPy | Or your register statement. |
| 03:00.35 | WIMPy | What inbound route? |
| 03:01.03 | apb3304 | from the provider... antisip |
| 03:02.38 | drkat | to make this easier.. what URI is invited during the incoming call .. so when the SIP provider sends the invite to your pbx what is the number they're inviting |
| 03:03.25 | WIMPy | it's all in the call rejected message. |
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| 03:15.27 | SeRi | xnt14: WIMPy what is going on |
| 03:15.37 | SeRi | wtf |
| 03:15.39 | SeRi | lol |
| 03:15.46 | SeRi | why did that handle got mixed there |
| 03:16.02 | WIMPy | Which one? ;-) |
| 03:16.09 | SeRi | xnt14 |
| 03:16.20 | SeRi | I meant to ask you |
| 03:16.35 | WIMPy | Tech the cat not to walk across the keyboard :-) |
| 03:16.42 | WIMPy | I'm trying to do a conference system. |
| 03:16.49 | SeRi | lol |
| 03:17.37 | SeRi | Which one you are using? MeetMe? ConfBridge? |
| 03:17.41 | drkat | qq who here is using asterisk in a professional environment |
| 03:18.17 | SeRi | drkat: define professional |
| 03:18.30 | WIMPy | ConfBridge. I've been dahdi free since it was possible. |
| 03:18.32 | drkat | SMB or Enterprise |
| 03:18.37 | SeRi | You mean in what grade? enterprise? small office? |
| 03:18.40 | drkat | 100+ users |
| 03:18.42 | SeRi | Ahhhh |
| 03:19.00 | SeRi | I am using it in several SMB |
| 03:19.29 | *** join/#asterisk radic (~radic@dslb-178-002-225-088.pools.arcor-ip.net) |
| 03:19.48 | SeRi | one site has 25+ users |
| 03:20.39 | drkat | coo.. |
| 03:21.02 | SeRi | no ent here |
| 03:21.32 | drkat | Are you a consultant that does installations? or ? |
| 03:23.02 | SeRi | nope... My brother is a consultant and all this sites are his. I just did it for him as a learning curve. |
| 03:23.27 | Mr_Break_it | or maybe an installer that does consulting, or an constaller that does insulting? |
| 03:23.38 | SeRi | lol |
| 03:23.44 | drkat | heh |
| 03:25.08 | SeRi | WIMPy: I been using confbridge as well. Works very well |
| 03:25.57 | WIMPy | I have a very interesting feature, but that might be located in the channel rather than in ConfBridge. |
| 03:26.17 | SeRi | sense the cheaptards at work did not want to pay for a business bridge I had to open up my own. |
| 03:26.32 | SeRi | WIMPy: what is that? :) |
| 03:27.33 | WIMPy | Under some circumstances messages to the user are played to all users. Like the menu or the You have been (un)muted. |
| 03:30.21 | SeRi | ah I see |
| 03:31.15 | SeRi | http://www.amazon.com/Jetway-NF9HQL-525-Quad-LAN-Mini-ITX-Motherboard/dp/B009TFIW84 |
| 03:31.20 | SeRi | My next board |
| 03:31.27 | SeRi | affordable and nice. |
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| 03:48.02 | apb3304 | ok! We've been able to eliminate the rejected message... turns out it was the registration string that was retarded. But..... I'm still getting a busy signal. The invite comes from antisip, and then my pbx responds UNAUATHORIZED |
| 03:48.42 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 03:49.16 | WIMPy | You probably need 'insecure=invite'. |
| 03:50.19 | apb3304 | got insecure=invite,port... originally had insecure=very |
| 03:50.39 | apb3304 | it's in both peer and user contexts |
| 03:51.02 | apb3304 | on the incoming trunk |
| 03:53.40 | apb3304 | I take it back. 'We made so many changes that line got deleted |
| 03:53.50 | apb3304 | That was the ticket I think |
| 03:54.04 | apb3304 | still testing... but thank you! |
| 03:56.04 | jpsharp | I'm using ConfBridge. I want to play a sound to all of the conference members except the one who is currently joining, but without turning on the announce_join_leave and prompting for a name. I don't see a way to do it, but I might be missing something. Ideas? |
| 03:56.50 | jpsharp | Or, failing that, send a sound to the conference before running ConfBridge on the inbound channel. |
| 03:57.04 | WIMPy | I do it with originate to a local channel. But the joining user will hear hist announcement as well, unless you wait a little befor sending him to the conference. |
| 03:59.17 | jpsharp | I could do that. |
| 03:59.32 | jpsharp | Ugly, but doable. |
| 03:59.52 | WIMPy | I don't see any other way. |
| 04:00.01 | WIMPy | But it works so far. |
| 04:00.28 | jpsharp | ponders patching app_conference to allow that. |
| 04:00.42 | WIMPy | And not too bad. If you announce the user count to the joining party, the two announcements will happen simultameousely which is quite an advantage. |
| 04:01.16 | WIMPy | I did it because I needed to use a pre-recorded username. |
| 04:03.43 | jpsharp | The originate trick does what I need. At least for the moment. |
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| 07:58.02 | jeffspeff | i'm looking for something like this http://www.broad-tel.com/products/phoneadapter.php but from a company that's still open |
| 08:02.10 | jeffspeff | i should clarify, i'm referring to the first item/device on the page |
| 08:02.23 | pppingme | why usb based? why not something ethernet based? |
| 08:03.27 | jeffspeff | pppingme, i don't need a box that requires any other configuration settings, or network stuff or additional power source. the usb just works for what i need |
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| 08:05.19 | pppingme | it just leaves a dependency on a cmoputer, and the possiblity of future issues with getting drivers, since it essentially emulates a sound card to the computer. |
| 08:06.17 | jeffspeff | it does what i'm looking for |
| 08:07.58 | TheTaLlesT | im having a problem with executing sox from asterisk. its running sox after the call is over but i get a empty file. the wav files are good and if i run sox in the terminal it produces a good file. any thoughts? |
| 08:10.18 | pppingme | jeffspeff as far as I know its pretty much an OEM item, so you're still left up to the software around it, and for that reason, its just simply not a retail item. |
| 08:10.59 | pppingme | just go get a majicjack and break it down |
| 08:11.10 | jeffspeff | i never said i was looking for a retail item |
| 08:11.57 | pppingme | if you're needing a real qty of them, like 10,000 of them, you could probably do that.. |
| 08:14.59 | jeffspeff | yeah, i'm gonna go immediately buy 10k of a product to carry my brand name that's completely untested with my system |
| 08:15.46 | ChannelZ | TheTaLlesT: without knowing or seeing what you're actually doing, the obvious things to look for are whether you're specifying complete paths to things since Asterisk's running environment is probably different than yours |
| 08:15.52 | ChannelZ | Same goes for permissions |
| 08:16.36 | TheTaLlesT | full paths |
| 08:20.07 | TheTaLlesT | in my extensions.conf i have exten => h,1,System(/etc/asterisk/encoderecording.sh ${FILENAME}) and in the script i have /usr/bin/sox /var/spool/asterisk/monitor/$1-out.wav /var/spool/asterisk/monitor/$1-in.wav --channels 2 --combine merge /var/spool/asterisk/monitor/$1.ogg mixer 1,0,0,1 |
| 08:21.42 | TheTaLlesT | also have rm /var/spool/asterisk/monitor/$1-out.wav /var/spool/asterisk/monitor/$1-in.wav in the script |
| 08:21.53 | ChannelZ | if you're sure it's even executing, perhaps redirect stderr to a temp file in your script and see what it's saying to you |
| 08:22.32 | TheTaLlesT | it creates a file 2.8K |
| 08:22.55 | ChannelZ | You said earlier it was empty. |
| 08:23.21 | TheTaLlesT | its always that size and it does not play |
| 08:23.26 | ChannelZ | Anyway this doesn't sound like an asterisk problem |
| 08:24.09 | TheTaLlesT | odd thing is if i run that script as the user asterisk it works |
| 08:25.35 | pppingme | jeffspeff if the manufacturer thinks you're serious, they'll probably get you a sample. |
| 08:26.08 | pppingme | or, back to majicjack, just get one of theirs, once you open it, you'll see who makes it, then order direct from them |
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| 09:50.29 | _zoom_ | hi, |
| 09:51.39 | _zoom_ | would anyone please recommend a call center solution based on asterisk? |
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| 12:43.58 | hyphenex | Howdy. Just wondering. I'm thinking of getting a speech menu thing happening. For instance, a blind lady who is elderly and can't use the speed dials that well could just say the name of the person she wants to call. Is there a free speech recognition engine when choosing between about 8 items? |
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| 16:46.38 | dfgas-cr48 | what file do i need to change so my ivr loads my wav files as wav files and not trying to load my wav files as ulaw files |
| 16:47.24 | WIMPy | None. It's always selected automatically. |
| 16:47.36 | WIMPy | Never specify the files extension in the dialplan. |
| 16:49.00 | dfgas-cr48 | i had to do this last time on my last install |
| 16:49.09 | dfgas-cr48 | [2012-11-04 10:41:31] WARNING[29275]: file.c:958 ast_streamfile: Unable to open custom/ivrwelcome (format 0x4 (ulaw)): No such file or directory |
| 16:49.20 | dfgas-cr48 | the file is there as .wav |
| 16:50.05 | WIMPy | Is it readable by Asterisk? |
| 16:50.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:50.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:50.54 | *** join/#asterisk wonderworld (~w@dsdf-4db50c89.pool.mediaWays.net) |
| 16:52.00 | ScytheBlade1 | okay, so I'm pretty new to this, so feel free to bust out the RTFM wrench if needed, but.. |
| 16:52.36 | WIMPy | ~book |
| 16:52.36 | 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 |
| 16:52.41 | WIMPy | ^^ That's our RTFM |
| 16:52.42 | ScytheBlade1 | I can't seem to get external SIP connectivity working. I've setup two asterisk servers, and connected one device to each server. They can't call each other -- I keep getting an error stating that a call to the extension was rejected because the extension is not found in the context |
| 16:52.47 | ScytheBlade1 | Yup, I've been reading through that |
| 16:53.04 | ScytheBlade1 | All that it really says, as far as I can tell, is to set allowguest and configure the unauthenticated context |
| 16:53.06 | SeRi | pastebin your peers |
| 16:53.24 | WIMPy | Ok, that message also tells you what extension is missing from what context. |
| 16:53.47 | ScytheBlade1 | Right, which makes logical sense |
| 16:53.52 | SeRi | WIMPy: do you ever sleep? |
| 16:53.59 | ScytheBlade1 | as I'm dialing userB@domainB.com from userA@domainA.com |
| 16:54.08 | ScytheBlade1 | And I'm not cross-defining the domains in the dialplan |
| 16:54.23 | ScytheBlade1 | (I didn't think that you really could, without a SIP trunk) |
| 16:54.25 | SeRi | suspects WIMPy is a hibrid |
| 16:54.39 | ScytheBlade1 | sec on the peers |
| 16:54.41 | WIMPy | SeRi: Yes. I did until half an hour ago or so :-) |
| 16:55.15 | SeRi | ah. I was starting to wonder is you where like part droid or something. |
| 16:55.23 | SeRi | lol |
| 16:55.48 | WIMPy | SIP is SIP. Trunks don't exist. But I'm not sure what you intended (not) to do there. |
| 16:56.31 | ScytheBlade1 | again, new to this, feel free to bust out the wrench :) |
| 16:56.37 | ScytheBlade1 | http://pastebin.com/BWWKPrum |
| 16:57.43 | WIMPy | Looking nice. |
| 16:58.20 | ScytheBlade1 | Whenever I actually make the call though |
| 16:58.25 | ScytheBlade1 | using a full sip: URI |
| 16:58.25 | ScytheBlade1 | Call from 'tablet' (10.254.254.88:37233) to extension 'phone' rejected because extension not found in context 'LocalSets'. |
| 16:58.33 | ScytheBlade1 | (and vice-versa) |
| 16:59.30 | ScytheBlade1 | what am I missing here? |
| 17:00.02 | WIMPy | Thet's on the box where tablet is registered, I assume? |
| 17:00.10 | ScytheBlade1 | Correct |
| 17:00.29 | ScytheBlade1 | Call from 'phone' (10.254.254.86:58832) to extension 'tablet' rejected because extension not found in context 'LocalSets'. |
| 17:00.34 | WIMPy | Then you didn;t tell that box what/where "phone" is. |
| 17:01.09 | ScytheBlade1 | Right. I got that from the error message. But I'm dialing 'sip:phone@domainA' |
| 17:01.12 | ScytheBlade1 | Not just 'phone' |
| 17:01.37 | ScytheBlade1 | I'm assuming that I don't have to define the entire internet in my dialplan to make this work :) |
| 17:03.17 | WIMPy | That's an issue with "tablet". It doesn't call phone@theotherbox, but calls phone@itsbox. |
| 17:03.26 | ScytheBlade1 | Okay, that helps quite a bit |
| 17:04.05 | WIMPy | Not all phones are good at calling full URLs. Some only will do it if not configured for any server. |
| 17:04.24 | ScytheBlade1 | Both of these 'phones' are http://code.google.com/p/csipsimple/ |
| 17:04.32 | ScytheBlade1 | (fwiw) |
| 17:05.02 | WIMPy | Don't know that one. |
| 17:05.09 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.128.148) |
| 17:05.39 | SeRi | ScytheBlade1: I use csipsimple working ok for me |
| 17:05.43 | SeRi | I use it over VPN |
| 17:06.09 | ScytheBlade1 | SeRi: good to know. At the moment all of this stuff is behind NAT, but is accessible over ipv6 (at least in theory) |
| 17:06.15 | ScytheBlade1 | SeRi: any special config that you had to do? |
| 17:06.49 | SeRi | Nothing just se it as a normal peer |
| 17:07.04 | SeRi | s/se/set/ |
| 17:07.13 | ScytheBlade1 | yup that's what I did.. |
| 17:09.50 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 17:11.21 | ScytheBlade1 | okay, anyone aware of a softphone that *works* with ipv6? |
| 17:11.31 | ScytheBlade1 | All of these do in theory... and in practice they don't |
| 17:13.29 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 17:16.59 | SeRi | none that I know of... Have you try Jitsi? |
| 17:17.03 | SeRi | Maybe jitsi does |
| 17:17.34 | ScytheBlade1 | I've tried blink, linphone, and csipsimple so far |
| 17:17.36 | ScytheBlade1 | Let me try that one |
| 17:20.52 | SeRi | let me know if it worked. I am curious.... |
| 17:22.33 | ScytheBlade1 | sure |
| 17:22.55 | ScytheBlade1 | I think that part of my problem is that while I have the SRV records defined, they point to hosts that only have AAAA records, and no just regular A records |
| 17:23.32 | ScytheBlade1 | when I configured a SIP peer across the both of them, that was not a problem -- it picked up the hosts just fine |
| 17:23.40 | ScytheBlade1 | but all of the clients are a bit... more fun |
| 17:24.24 | WIMPy | If you want more fun, set up DUNDi and let the Asterisk boxes handle it. |
| 17:25.23 | ScytheBlade1 | Heh. Would it work on a v6 only box? |
| 17:26.19 | WIMPy | Doesn't matter. |
| 17:26.39 | WIMPy | It's there so the Servers can exchange information about what extensions they know. |
| 17:28.24 | ScytheBlade1 | Jitsi signed in properly and let me call another peer on the same installation no problem |
| 17:28.32 | ScytheBlade1 | (v6 only) |
| 17:28.42 | ScytheBlade1 | That's -way- more than any other client has managed to do out of the box |
| 17:29.41 | ScytheBlade1 | Calls to another domain are failing with the same unknown extension in context error, though. |
| 17:36.45 | SeRi | ScytheBlade1: Do you have the domain extensions under misc calls? |
| 17:37.18 | ScytheBlade1 | I don't have even the slightest clue what you're asking about, heh. Is this in asterisk or the SIP client? |
| 17:37.30 | *** join/#asterisk adolfomaltez (~taro@200.49.191.8) |
| 17:37.37 | SeRi | asterisk |
| 17:37.59 | ScytheBlade1 | Is there a page in the book where I could read up on that? |
| 17:38.02 | *** part/#asterisk LiuYan1 (~LiuYan@222.125.128.148) |
| 17:38.04 | SeRi | on sec |
| 17:39.44 | SeRi | under sip.conf -> [general] I have a context of "misc_calls" you can call it what ever... This is for IP calling. when somebody IP calls me it rotes through the misc_calls |
| 17:40.15 | ScytheBlade1 | http://pastebin.com/BWWKPrum |
| 17:40.16 | SeRi | ie: sip.mydomain.com |
| 17:40.18 | *** join/#asterisk blee (~blee@67.8.200.237) |
| 17:40.42 | ScytheBlade1 | I'm following the book pretty closely... context=unauthenticated |
| 17:40.56 | ScytheBlade1 | and then I've defined an extension under unauthenticated as well |
| 17:40.59 | SeRi | I see. so is from asterisk to asterisk..... |
| 17:41.04 | SeRi | no ip calling |
| 17:41.08 | SeRi | Mhhhhh |
| 17:41.20 | ScytheBlade1 | not with an IP address, no, I'm only using DNS labels here |
| 17:41.20 | SeRi | I have one setup that way bvetween two asterisk server but is IAX2 |
| 17:41.31 | ScytheBlade1 | hrm not quite |
| 17:41.58 | ScytheBlade1 | it is asterisk to asterisk, but I suppose a better way of thinking of it would be server A to server B without any custom config to glue them together |
| 17:42.30 | ScytheBlade1 | I want server A to be sitting somewhere on the internet and server B to be sitting somewhere else on the internet, and then I want people to be able to call in/out of them from the general internet |
| 17:43.24 | ScytheBlade1 | I currently can't call *anything* external to each asterisk installation :) |
| 17:43.45 | ScytheBlade1 | (as it whines about not knowing the extension in the context) |
| 17:44.16 | ScytheBlade1 | I suppose it would help knowing that this is asterisk-10.9.0-1.fc17.x86_64 |
| 17:44.27 | SeRi | ScytheBlade1: one sec |
| 17:45.47 | ScytheBlade1 | SeRi: no worries. thank you for your help. |
| 17:47.49 | SeRi | ScytheBlade1: http://pastebin.com/CR73qzkT <- my sip.conf for my peers |
| 17:48.28 | ScytheBlade1 | I think I'm phrasing my question / problem poorly |
| 17:48.44 | ScytheBlade1 | I can't call externally to $randomSIPaddress |
| 17:49.12 | ScytheBlade1 | when I try to, it says that the extension is unknown in the context |
| 17:49.20 | ScytheBlade1 | Internally, I can call between peers no problem |
| 17:49.44 | ScytheBlade1 | Now I'm pretty sure that I don't have to define the entire internet in sip.conf :) (or setup dundi) |
| 17:49.45 | WIMPy | There isn't much to be done if the phone sends the call to the wrong box. |
| 17:49.59 | SeRi | ScytheBlade1: What I gave you is a perr shaking between two asterisk server to call each other remotely. |
| 17:50.04 | ScytheBlade1 | I get that. But so far, quite literally EVERY client that I use has managed to send it 'to the wrong box' |
| 17:50.41 | ScytheBlade1 | SeRi: right, I follow. But how would I call random box Y without a peer definition? |
| 17:50.46 | ScytheBlade1 | I must be missing something stupid here |
| 17:50.49 | WIMPy | Try to not configure the client at all. |
| 17:50.57 | ScytheBlade1 | That's pretty much what I did with Jitsi. |
| 17:51.04 | ScytheBlade1 | Actually, sec, let me wireshark this. |
| 17:52.00 | ScytheBlade1 | The SIP headers are properly qualified with the @domain |
| 17:52.52 | SeRi | ScytheBlade1: when calling random boxes I dal ext@domain |
| 17:53.08 | ScytheBlade1 | That's what I'm dialing too |
| 17:53.16 | SeRi | in my context I define that as a extension locally |
| 17:53.22 | SeRi | as so: |
| 17:54.35 | SeRi | http://pastebin.com/kqtBnMd6 |
| 17:54.40 | SeRi | ScytheBlade1: ^^ |
| 17:55.11 | WIMPy | That's where dundi comes in handy. |
| 17:56.36 | ScytheBlade1 | okay, let me try hard-coding one, just to test |
| 17:57.30 | SeRi | ScytheBlade1: make sure the destination has a matching context |
| 17:57.33 | ScytheBlade1 | That context matches the context= within sip.conf/[general], correct? |
| 17:57.40 | SeRi | This where I have define a misc_context |
| 17:58.11 | SeRi | ScytheBlade1: Yes. and make sure that that context is in your dial plan |
| 17:58.57 | *** join/#asterisk gown (~phonetic@gateway/tor-sasl/martinphone) |
| 18:01.29 | ScytheBlade1 | (again, thanks for the help, it is appreciated) |
| 18:01.38 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 18:01.39 | ScytheBlade1 | http://averageurl.com/asterisk/ <-- all configs here |
| 18:02.30 | WIMPy | You can use includes. |
| 18:02.41 | ScytheBlade1 | Hrm? |
| 18:02.59 | SeRi | WIMPy: is correct. |
| 18:03.02 | SeRi | is cleaner |
| 18:03.02 | WIMPy | Desfine your extensions in the most unprivileged context and include that in the more privileged ones. |
| 18:03.45 | ScytheBlade1 | At this point I'm not too worried about that, I'm just trying to get A to talk to B without hard-coding anything specific to either server or using DUNDi |
| 18:04.00 | ScytheBlade1 | Eventually I'll be loading most of the config into an LDAP backend and going from there |
| 18:10.25 | SeRi | ScytheBlade1: any luck? |
| 18:10.32 | ScytheBlade1 | Not really, sec |
| 18:11.22 | ScytheBlade1 | I just tested calling sip:500@ekiga.net via Jitsu, and captured it in wireshark. The From: header is correct (@domain), and the To: header is also correct ("To: <sip:500@ekiga.net>"), but asterisk didn't care: |
| 18:11.26 | ScytheBlade1 | Call from 'tablet' ([2601:7:70c0:1:a1f1:e3b:b2b3:f152]:28902) to extension '500' rejected because extension not found in context 'LocalSets'. |
| 18:11.49 | SeRi | asterisk did care |
| 18:12.00 | ScytheBlade1 | well it cared, it just derped from what I can tell :) |
| 18:12.01 | SeRi | something is wrong with your dial plan context |
| 18:12.23 | ScytheBlade1 | Okay. Considering the simplicity (the current version hasn't changed since I uploaded a copy), what am I missing? |
| 18:12.28 | ScytheBlade1 | It something absent? |
| 18:12.41 | SeRi | ext 500 is not in your dial plan man |
| 18:12.50 | ScytheBlade1 | .. I get that |
| 18:13.11 | SeRi | It should be desktop@domain |
| 18:13.18 | SeRi | or tablet@domain |
| 18:13.54 | ScytheBlade1 | is there some generic entry that I need to add to the dialplan to make it lookup and call remote domains automatically? |
| 18:14.43 | ScytheBlade1 | Because the *complete config* is up there, and when I try to call 500@ekiga.net, asterisk is just reading it as '500' instead of what I would expect '500@ekiga.net' -- despite the wireshark capture confirming that 500@ekiga.net is being sent over the wire |
| 18:15.57 | WIMPy | Asterisk is not a SIP proxy. If it receives a call, it assumes that it's for itself. |
| 18:16.37 | WIMPy | It should be possible by parsing the SIP header, however. |
| 18:16.50 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:17.02 | *** part/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:17.17 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 18:17.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:28.34 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 18:32.27 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
| 18:41.53 | dijib | dead in here |
| 18:42.25 | jpsharp | I killed everyone. |
| 18:42.48 | SeRi | dijib: waz up dude! |
| 18:43.18 | WIMPy | points at SeRi. He, you forgot to kill someone. |
| 18:44.16 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 18:45.32 | SeRi | lol |
| 18:51.12 | dijib | SeRi: ! DUDE |
| 18:51.16 | dijib | nothing much man |
| 18:51.31 | dijib | im still breathing dont worry about me WIMPy |
| 18:51.47 | dijib | my asterisk needs some rehab though. |
| 18:52.07 | dijib | after i finish this LMV rebuild with a new 3TB drive |
| 18:52.20 | dijib | pvmove takes FOREVER |
| 18:52.41 | dijib | still approx 600GB to go. |
| 18:52.52 | dijib | whats going on in the states today? |
| 18:52.59 | dijib | brb coffee run |
| 18:56.40 | SeRi | dijib: your conf is up? did you fix it? |
| 18:58.34 | dijib | is it? |
| 18:58.40 | dijib | might be |
| 18:58.55 | dijib | what was the exten? |
| 18:59.07 | dijib | 6633 2663 something like this? |
| 18:59.19 | dijib | if you tell me the extension i will fix it right now. |
| 18:59.35 | SeRi | ok one sec |
| 18:59.51 | SeRi | 2663 |
| 19:03.01 | SeRi | dijib: looks like your domain does not resolve anymore |
| 19:03.16 | dijib | no? |
| 19:03.18 | dijib | should |
| 19:03.59 | dijib | try emintech as the child domain instead of dijib |
| 19:05.15 | dijib | follow me? |
| 19:05.33 | SeRi | dijib: look at your pm |
| 19:07.33 | dijib | looked |
| 19:07.37 | dijib | here he come |
| 19:07.39 | dijib | s |
| 19:09.58 | dijib | p3nguin: where you at? |
| 19:33.58 | dijib | WIMPy: what you doing? |
| 19:35.46 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 19:39.46 | *** join/#asterisk echo083 (~adam@gateway/tor-sasl/echo083) |
| 19:39.49 | echo083 | i received tons of calls from 5550000 101 201 are they default internal numbers ? |
| 19:51.00 | [TK]D-Fender | there is no "defualt", there is only what you put in your configs |
| 19:51.29 | dijib | echo083: blacklist them |
| 19:51.45 | echo083 | really the destination was "s" |
| 19:51.57 | echo083 | you think they are phone spam |
| 19:55.21 | [TK]D-Fender | I think we should see more... |
| 20:02.01 | echo083 | i'm using the freepbx distribution |
| 20:02.17 | echo083 | but i had no answer on the channel #freepbx |
| 20:02.29 | echo083 | please prepend my nickname so that it rings |
| 20:08.35 | echo083 | do you know if in the freepbx distribution they define default internal numbers ? |
| 20:09.41 | *** join/#asterisk af_ (~getsmart@78.134.25.32) |
| 20:13.43 | SeRi | echo083: for freepbx please go to #freepbx |
| 20:13.56 | echo083 | i had no answer |
| 20:14.03 | echo083 | that's why i asked here |
| 20:14.10 | echo083 | did you read me ? |
| 20:14.46 | WIMPy | The thing is... |
| 20:14.51 | WIMPy | ~2nd |
| 20:14.51 | infobot | hmm... 2nd is #asterisk is not Tier 2 FreePBX/Trixbox Support! |
| 20:19.57 | [TK]D-Fender | They do not. Show us your actual EVIDENCE |
| 20:19.59 | dijib | mailcmd = /usr/sbin/ssmtp -t |
| 20:22.26 | echo083 | ok ok |
| 20:22.30 | echo083 | thanks for your help |
| 20:22.54 | echo083 | my evidence is in the freepbx call logs |
| 20:23.05 | echo083 | to answer you if you were talking to me :p |
| 20:23.18 | [TK]D-Fender | is waiting.... |
| 20:32.24 | *** join/#asterisk jschimno (~jschimno@107.195.30.90) |
| 20:34.16 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:35.19 | jschimno | I am running asterisk 1.8.11 on deb with Cisco SPA50XG phones. Anyone out there with experience with * and these phones? |
| 20:37.11 | [TK]D-Fender | ~ask |
| 20:37.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:00.41 | *** join/#asterisk jschimno (~jschimno@107.195.30.90) |
| 21:06.22 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:14.04 | *** join/#asterisk bombev (~kudev@37-143-192-141.gabrovo.ddns.bulsat.com) |
| 21:15.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:26.36 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:36.48 | *** join/#asterisk Lantizia (~lantizia@cpc6-stok15-2-0-cust8.1-4.cable.virginmedia.com) |
| 21:37.27 | Lantizia | Is there some way I can use the cdr database to work out what the highest number of simultaneous calls (inbound or out) there has been in a month? |
| 21:37.50 | Lantizia | It'd help me work out how many channels we should really be paying out for |
| 21:41.58 | ectospasm | I'm sure you could write a script to do that. |
| 21:42.49 | ectospasm | just cycle through the database and count everything |
| 21:42.57 | ectospasm | Lantizia: ^ |
| 21:43.17 | Lantizia | well the cdr will only have start/end times - i'd need something that can work out what overlaps |
| 21:44.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:44.46 | ectospasm | Lantizia: just cycle through time... add a |
| 21:45.09 | ectospasm | add one to the tally when a call starts... subtract when one ends. |
| 21:45.36 | ectospasm | keep a value that notes the high water mark |
| 21:45.47 | ectospasm | write a script that will do it for a 24hr period |
| 21:46.00 | ectospasm | then run that script for every day in a month you want to check |
| 21:47.03 | Micc | anyone know of a quick way to end a conbridge without using AMI. I want to hangup on everyone and end a confbridge from the dialplan. |
| 21:51.45 | Micc | could I mark a user with end when marked leaves then enter them and then hangup on that channel? |
| 21:58.38 | Lantizia | ectospasm, might be easy to say but not to do |
| 21:59.22 | ectospasm | Lantizia: should be straigtforward |
| 21:59.51 | ectospasm | pick a light day (like Sunday) and do it by hand |
| 22:00.05 | Lantizia | the point is not to pick a light day, i need it for a month |
| 22:01.26 | ectospasm | Lantizia: start small first |
| 22:01.36 | ectospasm | you can just run the script for every day in a month |
| 22:01.41 | ectospasm | let the computer do your work |
| 22:03.28 | ectospasm | then, feed your script a particular month, and have it churn through the database and generate a report. |
| 22:04.00 | ectospasm | may take a couple of minutes to run, depending on how the underlying script is written |
| 22:04.14 | ectospasm | I'd think it'd take a couple of hours to write the script first |
| 22:05.14 | Lantizia | well i don't have that time sorry |
| 22:06.48 | ectospasm | your loss |
| 22:07.13 | ectospasm | depending on how your CDR is stored, it may just be the appropriate SQL |
| 22:07.27 | ectospasm | ...which still could take some time to write. |
| 22:20.47 | *** join/#asterisk nicknam12322 (021d147d@gateway/web/freenode/ip.2.29.20.125) |
| 22:22.28 | Lantizia | ectospasm, http://www.mail-archive.com/asterisk@uc.org/msg05559/maxcallcount.pl |
| 22:22.35 | Lantizia | I knew someone had got around to doing it :P |
| 22:28.37 | ectospasm | there ya go |
| 22:41.35 | *** join/#asterisk nicknam12322 (021d147d@gateway/web/freenode/ip.2.29.20.125) |
| 22:43.15 | *** join/#asterisk elico (~Thunderbi@bzq-109-64-221-76.red.bezeqint.net) |
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| 22:47.01 | [sr] | hi |
| 22:47.02 | [sr] | guys |
| 22:47.20 | WIMPy | hi [sr] |
| 22:47.21 | [sr] | i'm still going crazy about hardware echo canceller |
| 22:47.40 | [sr] | card always says on load: NOTE: hardware echo cancellation has been disabled |
| 22:47.44 | [sr] | hi WIMPy |
| 22:47.50 | WIMPy | I heard some rumours this is perfectely normal. |
| 22:48.13 | [sr] | well i have on the channels hwec |
| 22:48.17 | [sr] | instead of mg2 |
| 22:50.40 | WIMPy | I guess I wouldn't bother with HWEC any more. |
| 22:51.38 | WIMPy | So how good is openvoxes customer support? |
| 22:52.28 | [sr] | WIMPy: well can't tell, started with this issues on saturday... let see starting tomorow.. |
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| 22:53.43 | [sr] | damn undock's the # window and don't know how to dock it again grrrrrr |
| 22:53.45 | [sr] | ok never mind |
| 22:54.17 | [sr] | WIMPy: if i change this to hwec,chan-chan , it doesn't even recognize the chans |
| 22:55.40 | [sr] | is that telling me that hwecho is not working? |
| 22:59.18 | [sr] | i think this will be the last openvox card's i'll ever buy |
| 22:59.20 | [sr] | damn |
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| 23:23.39 | [sr] | back |
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| 23:28.20 | ChannelZ | front |
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