| 00:04.17 | digitalirony | ok, so I have another annoying issue |
| 00:04.26 | digitalirony | Retrieve conf failed to copy file(s) from a module's agi-bin dir: copy(/var/lib/asterisk/agi-bin/directory.lib.php): failed to open stream: Permission denied |
| 00:04.37 | digitalirony | permissions are all owned by asterisk user and group |
| 00:04.39 | [TK]D-Fender | ~freepbx |
| 00:04.39 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 00:04.41 | [TK]D-Fender | ^^ |
| 00:17.48 | JoshBon | Penguin: Thank you for your help, I really appreciate it. |
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| 01:30.07 | JoshBon | Penguin: Still here? |
| 01:30.12 | JoshBon | sip show channelstats |
| 01:30.28 | JoshBon | I get a packet loss of about 1.20% on each channel |
| 01:31.55 | mmlj4 | I have a TDM wildcard with 1 FXO port (I don't care about the 3 FXS ports at the moment), ran dahdi_cfg and it configured the port. dahdi show status gives me "Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)", while dahdo show channels only shows the pseudo device. lsdahdi gives me " 4 FXO FXSKS (EC: MG2 - INACTIVE) RED" |
| 01:32.07 | mmlj4 | ...and of course I can't dial out |
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| 02:23.18 | smkelly | file. |
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| 04:44.48 | D30 | hi all, is digium TDM400 card supported for dahdi? |
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| 05:21.45 | ChannelZ | yeah should be |
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| 06:33.43 | devlap | Hi. I am trying to set up outbound dialing ( I want to call my cellphone ) with ipcomms sip trunking. What do I have to do beyond what I have now to make this work http://pastebin.ca/2639713 |
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| 06:42.50 | wacomcito | good mornig !! |
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| 06:46.00 | Penguin | devlap: Do you have a SIP phone on asterisk? |
| 06:46.11 | devlap | Uhh softclient yea. |
| 06:46.39 | devlap | it works. |
| 06:46.42 | devlap | I guess. |
| 06:46.45 | Penguin | This configuration is not good. |
| 06:46.51 | devlap | Ok. |
| 06:46.51 | wacomcito | i`m working on a sms service, SIP to SIP with a DB, because maybe one SIP is offline. I need to know if an event ONCONNECT or similar can make a macro call.. |
| 06:47.02 | devlap | whats wrong with it? |
| 06:47.15 | Penguin | The context of LocalSets is to be used only for SIP phones on the system. |
| 06:47.30 | devlap | sorry? |
| 06:47.57 | devlap | Yea I get that. I just put it in the same context. |
| 06:47.57 | Penguin | You've assigned the ITSP's peer to the same context. That's wrong. |
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| 06:48.48 | devlap | Ok well its back in from-ptsn cool?> |
| 06:49.09 | devlap | I was testing something anyways. |
| 06:50.22 | devlap | So anyways, I get a number dialed is not in service IO think when im trying this. |
| 06:51.52 | Penguin | I would rather see it like this: http://pastebin.com/NBPFyCmQ |
| 06:52.51 | devlap | yea well I have sip clients in there too, so I don't think I want to add register to general. |
| 06:52.58 | Penguin | You have no choice. |
| 06:53.09 | devlap | whys that? |
| 06:53.17 | Penguin | That's where register statements go. |
| 06:53.33 | Penguin | ~book |
| 06:53.34 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 06:53.37 | devlap | So how does that get used, wouldn't it leak the password if it was on other sip clients |
| 06:53.38 | devlap | Yea |
| 06:53.40 | devlap | I have a copy. |
| 06:54.05 | devlap | I'm in chapter 7. |
| 06:54.16 | Penguin | It makes your asterisk send a SIP REGISTER to the host in the host line. |
| 06:54.26 | Penguin | err... in the register line. |
| 06:54.57 | kaldemar | register statements are not related to peers really. |
| 06:55.12 | Penguin | register => USER:PASS@2way.ipcomms.net/some-extension says to send REGISTER to 2way.ipcomms.net |
| 06:55.43 | kaldemar | registering is not a login type thing. it just lets the other end know where you are. |
| 06:56.26 | devlap | yea I get that. |
| 06:56.39 | Penguin | It has nothing to do with your phones. |
| 06:57.29 | devlap | Yea I get that too. Anyways yea the register line just had to be in the right context in the file. |
| 06:57.35 | devlap | it just dialed my cell. |
| 06:59.08 | devlap | cool thanks. Wonder if I can set callerid. |
| 06:59.48 | Penguin | You may have to add sendrpid=yes to the peer entry. |
| 07:00.37 | devlap | Worked without it. pretty amusing. |
| 07:01.16 | devlap | Now just to do incoming. I'll mess with that later. |
| 07:01.20 | devlap | Thanks for looking! |
| 07:03.30 | Penguin | I never had termination service with ipcomms, only origination. |
| 07:04.12 | devlap | I just got it for setting up a demo. |
| 07:05.16 | devlap | So you've only done outbound calling with them? |
| 07:05.36 | Penguin | No, only received calls via them. |
| 07:05.39 | devlap | Ahh. |
| 07:06.03 | devlap | Who do you recommend, for either service? |
| 07:06.12 | Penguin | I like flowroute and voip.ms. |
| 07:09.17 | wacomcito | is there any way to make a macro call when SIP has connected? |
| 07:09.40 | devlap | I wanna try T.38 |
| 07:10.39 | devlap | I don't think my current fax machines support it. |
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| 07:11.38 | D30 | hi all, want to install tdm400p card using dahdi.. |
| 07:11.47 | D30 | what to do first after inserting the card? |
| 07:12.03 | wacomcito | power off the PC |
| 07:12.05 | D30 | i run dahdi_genconf but nothing shows |
| 07:12.06 | wacomcito | :) |
| 07:12.20 | bsdice | devlap I think this new fax stuff will never replace e-mail |
| 07:12.21 | Penguin | Power it off after inserting the card? |
| 07:12.45 | devlap | bsdice, Yea but people fax stuff, we use it all the time, and I have to do something about it. |
| 07:13.07 | bsdice | devlap I know, I really like fax still as well |
| 07:13.22 | devlap | I never said I liked it. Jus that I'm responsible for making it work :P |
| 07:13.27 | wacomcito | sorry my english is poor.. before insert the card.. |
| 07:13.28 | wacomcito | xDDDD |
| 07:13.29 | D30 | Penguin: wacomcito i must rephrase myself,.. i turned it off when im inserting the card :p |
| 07:13.31 | coppice | we've moved on from fax to lies and deceit |
| 07:13.32 | devlap | I think its stupid. I scan and email everything. |
| 07:13.32 | Penguin | If your fax machine doesn't support t.38, do you want to use your fax machine or do you want to use t.38? |
| 07:13.40 | bsdice | <PROTECTED> |
| 07:13.52 | devlap | Penguin, probably need to use the existing fax. |
| 07:13.53 | D30 | now im on the cli, i run dahdi_genconf |
| 07:14.00 | D30 | but nothing shows after it.. |
| 07:14.17 | devlap | Actually. I guess we can scan/fax. |
| 07:14.22 | devlap | Inbound maybe only to email. |
| 07:14.47 | devlap | idk. |
| 07:14.56 | devlap | If we did use a fax, it would be the existing ones. |
| 07:15.04 | Penguin | That part is pretty simple. Asterisk has a couple differnet fax apps and you can easily turn the received fax into an email. |
| 07:15.08 | devlap | In which case I think you need analog thingies. |
| 07:15.59 | bsdice | I use hylafax over iaxmodem over asterisk with ulaw/alaw, works. |
| 07:16.01 | Penguin | You can get a basic ATA and connect the fax machine to the network. |
| 07:16.27 | Penguin | Turn your fax machine into a SIP fax machine and do FoIP. |
| 07:16.34 | Penguin | or FoVoIP. |
| 07:16.36 | devlap | yea. Well I gotta hit the sack. I'll look into it more tmw. See what will work with us. I'm trying to map out the use cases in my head and its just too cloudy right now. |
| 07:16.57 | devlap | yea, I think that would be a good start. |
| 07:17.01 | devlap | Thanks Penguin . |
| 07:20.14 | wacomcito | [D30] i think you must be sure yourk kernel loads module for your dahdi |
| 07:20.51 | D30 | wacomcito: how to check? |
| 07:21.13 | wacomcito | dmesg |grep "your model" |
| 07:21.24 | wacomcito | lsmod |
| 07:21.28 | wacomcito | .... |
| 07:21.38 | bsdice | I would rather use lspci to check if there is a driver loaded for that card |
| 07:22.19 | wacomcito | lspci not show loaded modules |
| 07:22.28 | wacomcito | only manufactured chipos |
| 07:22.39 | bsdice | lspci -v |
| 07:22.45 | bsdice | 06:00.0 USB controller: NEC Corporation uPD720200 USB 3.0 Host Controller (rev 04) (prog-if 30 [XHCI]) |
| 07:22.52 | bsdice | ... Kernel driver in use: xhci_hcd |
| 07:23.13 | wacomcito | ok :) |
| 07:23.14 | D30 | thanks wacomcito will try.. |
| 07:23.43 | Penguin | lspci tells the driver and the module that the driver is in. |
| 07:23.55 | Penguin | (if it is loaded and in use) |
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| 07:29.39 | D30__ | wacomcito: Penguin i can see this http://codepad.org/yTL1EzsL |
| 07:30.06 | D30__ | i check the card before installing it, it has the name Tiger Jet on it.. so im sure this is the card |
| 07:31.36 | D30 | dmesg also shows http://codepad.org/HfBnOE2K |
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| 07:33.52 | wacomcito | [D30] if we can assume your card its loaded and ok you must crate a config for this |
| 07:33.55 | D30 | i also included the some dahdi status in my box http://codepad.org/41jlAY8O |
| 07:33.59 | wacomcito | check your flie chan_dahdi.conf: |
| 07:34.57 | D30 | hmm wacomcito i only have chan_dahdi.conf.template chan_dahdi_additional.conf |
| 07:35.02 | D30 | this is a fresh install by the way.. |
| 07:35.17 | D30 | so i must create a chan_dahdi.conf file ? |
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| 07:35.33 | wacomcito | but it´s a freepbx system ? |
| 07:36.23 | wacomcito | you must edit this file chan_dahdi.conf.template for your enviroment and save as chan_dahdi.conf. |
| 07:36.44 | D30 | ahh okay wacomcito noted on that.. will try to configure this first.. thanks :) |
| 07:39.08 | wacomcito | [D30] check this another file too dahdi_channels.conf, and finaly /etc/dahdi/system.conf. |
| 07:39.32 | D30 | okay noted wacomcito , its my first time installing a digium card :p |
| 07:40.04 | kaldemar | chan_dahdi.conf is the asterisk configuration file. if you have configs in files by other names, those need to be included in chan_dahdi.conf. |
| 07:40.04 | wacomcito | you can more info about these config files googling a little., |
| 07:40.41 | D30 | okay thanks folks.. will try it first.. |
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| 07:48.12 | D30 | i found this http://www.digium.com/sites/digium/files/analog-telephony-card-4-port-user-manual.pdf from digium |
| 07:48.24 | D30 | its using dahdi also for a 4 port analog card |
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| 07:57.57 | D30 | hi one last question, :) can we generate a list of all extensions with the corresponding user (names) on it.. is it on the cdr?? |
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| 09:49.02 | Blashyrkh | what does the underscore do in front of an extension in the dialcode? |
| 09:49.56 | Blashyrkh | got it |
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| 09:56.14 | Blashyrkh | does asterisk read all .conf files when doing a dialplan reload in the /etc/asterisk directory? |
| 09:56.39 | Blashyrkh | so i could just make a new .conf file with a new context in it and goto it from anywhere else? |
| 09:58.28 | mirela666 | Blashyrkh: no, module reload will reload everything / if command haven't changed |
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| 09:58.51 | mirela666 | Blashyrkh: dialplan reload will reload only extensions.conf and all includes in it |
| 10:01.48 | Blashyrkh | so but regarding dialplan asterisk only reads extension.conf and all includes there? |
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| 10:03.26 | mirela666 | yes |
| 10:04.02 | mirela666 | if you want to use dialplan written in ael you use extensions.ael and 'ael eload' |
| 10:04.06 | mirela666 | reload* |
| 10:09.25 | kaldemar | Blashyrkh: sure, you can put dialplan code in another file. just include it in extensions conf with "#include filename". |
| 10:09.40 | kaldemar | oops, i'm echoing. |
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| 14:55.43 | Greenlight | So Asterisk stops accepting calls or seemingly doing anything. CLI lets me connect but then freezes at "core show channels". Logs show some "exceptionally long queue lengths". Classic deadlock, right? |
| 14:56.34 | WIMPy | For me the CLI usually freezes when I use the TAB key. |
| 14:56.49 | WIMPy | But works if I don't. |
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| 15:01.11 | Greenlight | Hmm I'm trying to remember exactly what I typed at the CLI now :S |
| 15:01.47 | Greenlight | Would that likely be a deadlock around channels? |
| 15:02.58 | Greenlight | Not sure how to diagnose it, as it's been stable as a rock for 50 million ish calls. |
| 15:04.05 | WIMPy | asterisk -x 'core show locks' and ... threads |
| 15:04.20 | Greenlight | And the last reference to a deadlock in any module I run, in the changelog was back in march, and reported by me |
| 15:04.24 | WIMPy | But the best way is probably knowing what happened before. |
| 15:04.42 | Greenlight | I'd need to enable debug threads though wouldn't I ? |
| 15:04.53 | WIMPy | For threads, yes. |
| 15:05.03 | Greenlight | Which is a killer for performance, no ? |
| 15:05.04 | WIMPy | I had deadlocks in 11.7. |
| 15:05.18 | *** join/#asterisk u0m3 (~u0m3@92.80.106.31) |
| 15:05.20 | WIMPy | It's not that bad in 11, yet. |
| 15:05.42 | WIMPy | But in 12 it's deadly. |
| 15:05.43 | Greenlight | Even at 500+ channels ? |
| 15:06.06 | WIMPy | Depends on how much resources you have left. |
| 15:06.18 | Greenlight | Out of interest, what were the deadlocks you saw related to ? |
| 15:06.42 | WIMPy | They happened on transfer. |
| 15:07.22 | Greenlight | Hmm |
| 15:07.25 | WIMPy | And it happened always. |
| 15:07.30 | WIMPy | Until I trid to debug. |
| 15:07.34 | Greenlight | Oh, then likely not same issue |
| 15:07.35 | WIMPy | +e |
| 15:07.44 | Greenlight | SInce this must be a edge case scenario |
| 15:08.03 | WIMPy | Since I tried to debug, it never happened again. |
| 15:08.09 | Greenlight | Oh |
| 15:08.12 | Greenlight | Typical :) |
| 15:08.13 | WIMPy | Pretty scary. |
| 15:08.16 | Chainsaw | *twilight zone theme plays* |
| 15:08.27 | WIMPy | Only if it happens sometimes. |
| 15:09.08 | Greenlight | Right, well guess if I want to have something to work on *if* it happens again, I need to enable debug threads and pray it copes okay |
| 15:09.08 | WIMPy | I had it 100% reprodicible until then. No issue on 11.7-rc1 and always a lock up on 11.7. |
| 15:09.31 | WIMPy | Maybe the locks give a usable hint. |
| 15:10.05 | Greenlight | The only changes on changelog from 11.7.0 to 11.7.0-rc1 are security and SMS |
| 15:10.38 | Greenlight | oh, sorry no, that's from rc1 to rc2 |
| 15:10.39 | WIMPy | Yes, nothing that seemed even remotely related. |
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| 15:11.57 | Greenlight | Yea, quite, nothing at all that'd you'd expect to cause a deadlock type regression. |
| 15:12.17 | Greenlight | Right .. well gonna look over what loggin I have and see if I can gleam an idea what happened. |
| 15:12.50 | Greenlight | Usually these things have a way of running perfectly for months and months and then once it does go wrong, it'll happen again and again /sigh |
| 15:13.25 | WIMPy | At least you get a chance to find the cause if it happens again. |
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| 15:17.16 | asteriskmonkey | what is the most active registrations an asterisk 11 can handle now? |
| 15:18.34 | [TK]D-Fender | I doubt that is a metric anyone has really taken independently. |
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| 15:45.39 | DevWork_ | Whats the softphone (free) that has the most features like transfer buttons and programmable buttons and is probably opensource? |
| 15:45.56 | Chainsaw | Ekiga? |
| 15:46.08 | mmlj4 | 3cx, maybe |
| 15:46.18 | Chainsaw | It's fairly GTK+/Gnome specific. Here's to hoping you can cope with that. |
| 15:46.29 | DevWork_ | Works on windows good enough. |
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| 16:12.40 | jaflong | hi can the audio from chanspy be redirected and dialed out |
| 16:15.33 | Greenlight | Sure |
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| 16:17.03 | jaflong | any tips pls? |
| 16:21.24 | [TK]D-Fender | jaflong: Chanspy is a dialplan app. Whatever is on the other end of that channel is what gets it |
| 16:22.08 | [TK]D-Fender | jaflong: You could ORIGINATE a call out, and upon answer dump them INTO Chanspy..... |
| 16:24.09 | jaflong | Dont tink I have been clear, I want to stream a copy of the audio of a channel to another destination |
| 16:24.55 | [TK]D-Fender | jaflong: there is no special hook for this. You'd have the have some other channel make the call to chanspy |
| 16:25.16 | [TK]D-Fender | jaflong: and ORIGINATE is the way to trigger * making that call out to that connecting app. |
| 16:25.27 | jaflong | I see |
| 16:25.53 | [TK]D-Fender | jaflong: Asterisk does not "stream" anything... it is a telephony engine. |
| 16:26.14 | [TK]D-Fender | jaflong: You'll need somethign else to take that and transform it the way you want. |
| 16:26.39 | [TK]D-Fender | jaflong: Perhaps a softphone with an audio-hook to a streaming app, etc |
| 16:27.44 | jaflong | The originate look the the thing I need too do. I just used wrong wording |
| 16:28.27 | Greenlight | Just use a Local channel, one side goes to the ChanSpy, the other "out" to where you need to Dial |
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| 16:31.58 | jaflong | thanks guys |
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| 16:51.40 | Greenlight | Looks like MixMonitor is chief suspect for my deadlock. |
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| 16:56.17 | *** join/#asterisk lordvadr (~lordvadr@jose-tc.ctc.biz) |
| 16:58.06 | lordvadr | So if you try to turn off parking in asterisk 12, it segfaults. Is this a known bug? Is there a work around? |
| 16:59.35 | WIMPy | The parking conf has gone elsewhere. Should get it working again if you remove it from features.conf IIRC. |
| 16:59.44 | [TK]D-Fender | What is "turn off parking"? |
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| 17:00.25 | lordvadr | either noload res_parking.so or comment out all the directives in res_parking.conf. Either one will earn you a segfault. |
| 17:00.43 | Chainsaw | [TK]D-Fender: It is a lament that parking appears to be no longer be optional in 12, like it was in 11. |
| 17:00.50 | WIMPy | Sounds evil. |
| 17:00.54 | Chainsaw | Wow. Let's try that again. |
| 17:01.02 | Chainsaw | [TK]D-Fender: It is a lament that parking appears to no longer be optional in 12, like it was in 11. |
| 17:01.30 | lordvadr | I haven't run in in a debug, but it looks like the main thread barfs when it tries to load the parked calls context, and if it doesn't exist, it gets mad. |
| 17:01.42 | [TK]D-Fender | Chainsaw: That's kinda retarded.. it was just a bunch of dialplan apps anyway... don't see why you can't ignore the context and dynamically generated extens.... |
| 17:02.13 | Chainsaw | [TK]D-Fender: It is good security practice to disable features that are not used. I am with lordvadr on this one. |
| 17:04.14 | [TK]D-Fender | not using the context sounds functional... |
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| 17:06.38 | lordvadr | Chainsaw: Agreed. |
| 17:07.01 | Chainsaw | lordvadr: It was news to me, if it doesn't turn up with a cursory search of the bug tracker I'd file a new one. |
| 17:07.32 | Chainsaw | lordvadr: Even if the Fender attitude is taken, it gives me a another patch to scavenge and apply. |
| 17:08.18 | [TK]D-Fender | "patch"? What patch? |
| 17:08.46 | [TK]D-Fender | context=blackhole |
| 17:08.48 | [TK]D-Fender | the end |
| 17:08.50 | lordvadr | Yeah, I was surprised when a google didn't come up with even some chatter. |
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| 17:10.34 | lordvadr | The problem, Fender, is that it doesn't gracefully cope with something that it allows you to do. If asterisk refused to start without that module, fine. If the module refused to load if it had no context, fine. The main thread shitting itself when it find a list of 0 items instead of 1 is bad, and could indicate a much bigger problem. |
| 17:11.19 | lordvadr | Such as an attack vector or such. |
| 17:12.53 | lordvadr | And you need more than just a context. It wouldn't run without the extensions in it. |
| 17:13.44 | DevWork_ | Ekiga seems... extremly limited |
| 17:14.58 | Chainsaw | DevWork_: Yeah, but it's free. |
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| 17:15.40 | DevWork_ | Yea, well, when I dial for example 222, it just assume it was sip:222 most softphones will atleast route it like you dialed through the normal sip channel on your pbx |
| 17:16.07 | DevWork_ | I had to manually add 222@serverip |
| 17:16.32 | DevWork_ | it lacks any kind of network configuration |
| 17:17.00 | Chainsaw | DevWork_: Well, do you want a nice one or a free one? |
| 17:17.10 | mjordan | lordvadr: please do file a bug for that. It obviously shouldn't crash. If res_parking isn't loaded then there are tons of things that won't work (for example: one touch parking...) but Asterisk should run without res_parking. |
| 17:17.10 | DevWork_ | Free doesn't mean shit. |
| 17:17.43 | Chainsaw | DevWork_: If you fail to answer such a simple question I cannot assist you. |
| 17:18.20 | DevWork_ | I want free, I said this. I never said "nice" whatever THAT means, because it is entirely subjective and I can't possibly know what you mean by nice. |
| 17:18.45 | lordvadr | mjordan: Will do. |
| 17:19.28 | mjordan | lordvadr: and thanks for testing out 12. One of the things we did try to do was to pull out parking (we had to rework the whole damn thing anyway), but parking is a bit insidious, since so many channel drivers and other things have backdoors into parking calls |
| 17:19.40 | mjordan | lordvadr: we obviously missing *something* if not having the module loaded causes explosions |
| 17:20.19 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:20.39 | mjordan | but yes, the goal is that if you don't want to have parking, you can just not load it. And, actually, the whole 'auto-generated extensions for parking' thing is actually now optional, so you can - if you want - create your own custom parking things wherever you want in the dialplan. But yeah. Exploding == bad. |
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| 17:25.22 | As001 | Hello, I set variable in dialplan and I can see that value is ok on cli, but in WaitExten(${VARIABLE}) it does not work. When I put WaitExten(5) it works but as variable it does not. Am I doing something wrong here ? |
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| 17:26.10 | [TK]D-Fender | asNot showing us |
| 17:26.21 | [TK]D-Fender | As001: Not showing us |
| 17:26.23 | [TK]D-Fender | ~pb |
| 17:26.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:26.25 | [TK]D-Fender | ^^^^ |
| 17:26.28 | WIMPy | As001: Are you sure the contents are entirely numeric? |
| 17:27.05 | As001 | i will pastebin dialplan |
| 17:28.37 | As001 | http://pastebin.ca/2640098 |
| 17:28.51 | As001 | I can see 5 in verbose output I guess it is numeric. |
| 17:29.00 | As001 | I might be wrong. |
| 17:29.18 | Blackslikz | hey guys which is the best TTS engine that is as cheap as cepstal ? |
| 17:29.41 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 17:30.08 | coppice | is cepstral cheap? it used to be, but its price seems to have drifted upwards |
| 17:31.09 | WIMPy | As001: Better to look at an actual call with verbose 3. |
| 17:31.13 | Blackslikz | @coppice it is still atleast u could get a port for 40 bucks but its not too clear for asterisk box u know |
| 17:31.29 | As001 | ok I will do it right now. |
| 17:32.05 | coppice | I thought cepstral was a lot more than $40 a port now. |
| 17:32.19 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:33.19 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 17:33.48 | DevWork_ | Hmm. So I got pretty much what I want in linphone. |
| 17:33.57 | DevWork_ | Its free, I have source, and it works. |
| 17:34.08 | Blackslikz | Nah bro thats what i got it for last month for ? Looking for something better , but cheap too |
| 17:34.22 | DevWork_ | Interface is pretty terribad, but being that I have source thats an ez change. |
| 17:35.16 | As001 | Verbose shows: WaitExten("SIP/ippbx.net2phone.com-0000e748", "5") |
| 17:35.45 | [TK]D-Fender | As001: pastebin the entire call... |
| 17:35.56 | [TK]D-Fender | As001: Do not waste time trying to describe 1 line at a time |
| 17:36.09 | Kobaz | first the call gets the lotion |
| 17:36.16 | Kobaz | and then it puts it on it's skin |
| 17:36.26 | WIMPy | Looks good enough to me. |
| 17:36.55 | WIMPy | But we don't know what the issue is. |
| 17:40.11 | As001 | http://pastebin.ca/2640104 |
| 17:40.47 | As001 | Issue is when background ends after beep it does not wait for 5 secs. |
| 17:41.05 | As001 | until i put 5 in WaitExten |
| 17:42.22 | WIMPy | What does it do? |
| 17:43.00 | *** join/#asterisk wonderworld (~ww@89.204.139.58) |
| 17:43.01 | As001 | It threat me like i didn't press anything just after beep.... |
| 17:43.23 | As001 | that was during the day... Now it seems to work after change of voip provider. |
| 17:43.42 | wonderworld | hey guys |
| 17:43.53 | As001 | maybe just dtmf issue i dunno why it did not work and now works. Only change is voip provider. |
| 17:43.58 | [TK]D-Fender | <PROTECTED> |
| 17:44.07 | [TK]D-Fender | It shoudl NOT be a PLAYBACK, it should be BACKGROUND |
| 17:44.16 | [TK]D-Fender | There's your breakage |
| 17:44.29 | As001 | ok I will change it right away. |
| 17:44.41 | [TK]D-Fender | You forced 1 thing in the background, then something that has to take up foreground |
| 17:44.49 | WIMPy | I don't see much diference for just a beep. |
| 17:45.07 | [TK]D-Fender | it's blokcing. That's the problem |
| 17:45.08 | As001 | well i could join beep to audio message in background anyway if that is an issue.. |
| 17:45.13 | [TK]D-Fender | blocking* |
| 17:45.22 | As001 | ok I understand and I will change that right away. Thanks |
| 17:45.36 | WIMPy | There's aniother playback before that. That might not be the wanted behaviour. |
| 17:45.50 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 17:50.12 | As001 | it works with 2 Backgrounds one message one Beep. It is ok. If i get issues i will join message and beep in one file and solve problem. Thanks for help. |
| 17:50.41 | lordvadr | mjordan: Issue 23297 |
| 17:50.41 | *** part/#asterisk As001 (~uros@82.117.198.142) |
| 17:50.50 | [TK]D-Fender | NEXT!@@@!@1 (c) BKW |
| 17:51.12 | WIMPy | That's the nice thing about the good old Read(). YOu can specify a list of files to play. |
| 17:54.24 | [TK]D-Fender | You can also make a single call to background and pass both.... |
| 17:54.36 | [TK]D-Fender | but lets not get fancy now or nuthin' |
| 17:56.18 | DevWork_ | I'm using ipcomms sip termination. I'm trying to set the callerid from a call, but it comes up as 209-555-1213 |
| 17:56.47 | [TK]D-Fender | Set FROM a call? |
| 17:57.06 | [TK]D-Fender | I set BEFORE I call.... |
| 17:57.07 | DevWork_ | like uhh in the dialplan. |
| 17:57.15 | DevWork_ | Im pastbining its just being slow. |
| 17:57.29 | [TK]D-Fender | devyour concept of before//after, from/to seems jumbled |
| 17:57.59 | DevWork_ | http://pastebin.com/aGG6uwqh is basically what im doing. |
| 17:58.43 | [TK]D-Fender | DevWork_: f1st : SIP DEBUG <- we can't prove what's being passed |
| 17:58.54 | DevWork_ | Yea I have sip debug on. |
| 17:59.00 | [TK]D-Fender | DevWork_: next we don't know that your provide accepts it even if you send it. |
| 17:59.22 | DevWork_ | Well I sent "EXT2" before as my caller id and it went through. |
| 17:59.25 | [TK]D-Fender | devYou showed us dialplan... not the CALL |
| 17:59.33 | [TK]D-Fender | DevWork_: Then we don't know if your peer is set up right |
| 17:59.34 | DevWork_ | Yea how do I show you a call? |
| 18:01.48 | DevWork_ | http://pastebin.com/aNkYVXcT Is my peer tho |
| 18:05.07 | DevWork_ | I checked the sip header, its sending that. |
| 18:05.14 | DevWork_ | Its like my Set doesn't take effect or something. |
| 18:06.36 | DevWork_ | I told the cli "dialplan reload" but its picking up that old callerid setting. |
| 18:07.58 | [TK]D-Fender | pastebin the call... |
| 18:08.31 | DevWork_ | I asked you how I would do that, I don't know what you want me to pastebin. |
| 18:08.59 | [TK]D-Fender | copy/paste |
| 18:09.07 | DevWork_ | What do you want copy pasted. |
| 18:09.09 | [TK]D-Fender | You ARE at asterisk CLI aren't you? |
| 18:09.13 | [TK]D-Fender | Show it executing. |
| 18:09.21 | [TK]D-Fender | core set verbose 10 |
| 18:09.25 | [TK]D-Fender | sip set debug on |
| 18:09.30 | [TK]D-Fender | core set debug 0 |
| 18:09.40 | [TK]D-Fender | call |
| 18:09.42 | [TK]D-Fender | PASTEBIN |
| 18:10.01 | DevWork_ | That I can do. |
| 18:10.11 | [TK]D-Fender | look at call = CLI |
| 18:10.37 | DevWork_ | Ok well, like I said, its already sending the from with the number showing up wrongly on the call id. I'll paste it anyways |
| 18:11.07 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.105) |
| 18:12.37 | ruben231 | hi guys i have asterisk server on my local network, and when i register my local SIP softphones from a local PC- it did register successfully - what makes it weird its generating 120ms display on asterisk CLI- but this softphones is pretty local which i expect to be like 1-5 ms in range...any idea guys..? |
| 18:13.25 | [TK]D-Fender | ruben231: if you're referring to "sip show peers", that is QUALIFY time. It has NOTHING to with PING time |
| 18:13.45 | [TK]D-Fender | it is a different OSI layer and is the LOWEST thing on the prioritization scale |
| 18:14.07 | [TK]D-Fender | This is not a measurement of actuall comm latency. Consider it only useful to see if they answer at all |
| 18:14.17 | ruben231 | <PROTECTED> |
| 18:14.21 | DevWork_ | http://pastebin.com/A0mcKQVm is the call. |
| 18:14.37 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 18:14.38 | ruben231 | coz some other softphones gets 10 ms compare to most which are 150ms all |
| 18:14.42 | DevWork_ | line 25 shows its executing that line. |
| 18:14.47 | DevWork_ | The thing is, that line isnt in the dialplan. |
| 18:14.55 | [TK]D-Fender | ruthat number is practically worthless |
| 18:15.48 | [TK]D-Fender | [13:14]DevWork_The thing is, that line isnt in the dialplan. <- it very clearly is.... |
| 18:16.04 | DevWork_ | No its not. I have te dial plan open, and its not there. |
| 18:16.23 | [TK]D-Fender | DevWork_: You showed us some tiny snippit of dialplan and we don't have a context header in there. He have no idea what else is in the file... or if you're even looking in the right one. |
| 18:16.30 | DevWork_ | yea im pasting. |
| 18:16.31 | DevWork_ | one second. |
| 18:16.42 | [TK]D-Fender | DevWork_: Or you forgot to APPLY changes you made |
| 18:17.28 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 18:18.10 | DevWork_ | http://pastebin.com/LzeZC8uR shows my dialplan. If I type dialplan reload my console says "Dialplan reloaded." Followed by " == Parsing '/etc/asterisk/extensions.conf': Found" |
| 18:18.52 | [TK]D-Fender | "dialplan show" <- |
| 18:19.22 | dan_j | Hi, can someone help me analyse a tcpdump for a call that is being transfered? for some reason, when the call is transferred, the receiving agent can only hear asterisk on hold music, while the caller can hear the agent. |
| 18:19.24 | DevWork_ | I know what it thinks is in there d-fender. I want to know why my callerid isnt. |
| 18:19.52 | [TK]D-Fender | isn't where? |
| 18:20.24 | DevWork_ | I wrote the callerid in extensions.conf. and then I reloaded the dialplan. And the old line is calling out and it thinks the old line is in there, as seen by my calling out. |
| 18:20.25 | [TK]D-Fender | DevWork_: Thought the dialplan didn't match, the call you showed DID pass the number specified |
| 18:21.01 | DevWork_ | It passed the 209 number. |
| 18:21.05 | DevWork_ | Thats from a really old config. |
| 18:21.28 | DevWork_ | That is not what I want it to, or what is in the dialplan, after I reloaded the dialplan. |
| 18:21.56 | [TK]D-Fender | does "dialplan show" reflect what is in extensions.conf right now? |
| 18:22.00 | DevWork_ | NO! |
| 18:22.10 | DevWork_ | Or asterisk wouldn't send that number... |
| 18:22.27 | [TK]D-Fender | then you've done something very wrong... |
| 18:22.39 | [TK]D-Fender | like working on the wrong folder, the wrong file, etc... |
| 18:23.00 | DevWork_ | Ok so my paste showed the directory I was in, and then I cat'd the file. |
| 18:23.07 | DevWork_ | And then I showed you want dialplan reload said |
| 18:23.18 | *** join/#asterisk mjordan (~matt@nat/digium/x-tittlhatulyifpuv) |
| 18:23.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:23.31 | DevWork_ | "Dialplan reloaded.\n == Parsing '/etc/asterisk/extensions.conf': Found" |
| 18:23.57 | [TK]D-Fender | [13:18][TK]D-Fender"dialplan show" <- |
| 18:24.28 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 18:24.30 | DevWork_ | It shows the wrong dialplan, since you insist, I will show you exactly that on a paste... one second. |
| 18:24.50 | DevWork_ | http://pastebin.com/JtexEL91 |
| 18:25.41 | [TK]D-Fender | well your "cat" showed somethign invalid.. you have your constant set but you did NOT define the [globals] context header for it |
| 18:25.45 | [TK]D-Fender | that file looks invalid |
| 18:25.48 | *** join/#asterisk jsjc (~Adium@61.Red-79-150-66.dynamicIP.rima-tde.net) |
| 18:25.54 | [TK]D-Fender | and I could trust that it wouldn't even load |
| 18:26.02 | DevWork_ | Wouldn't it make an error? |
| 18:26.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 18:26.50 | DevWork_ | Wouldn't it make an error on dialplan reload? Setting it in the general context fixed it. |
| 18:26.53 | DevWork_ | Thanks for that. |
| 18:27.00 | [TK]D-Fender | gah |
| 18:27.20 | [TK]D-Fender | I can't prove what your console settings were and getting you to show things is like pulling teeth |
| 18:27.29 | [TK]D-Fender | It's a fight ever step of the way |
| 18:27.33 | DevWork_ | I set my console settings. |
| 18:27.37 | DevWork_ | Exactly like you said. |
| 18:27.42 | [TK]D-Fender | This is a very counterproductive habit |
| 18:27.53 | [TK]D-Fender | You keep telling me.... that is not showing. |
| 18:28.02 | DevWork_ | And you never told me how to show you. |
| 18:28.04 | DevWork_ | Which I asked. |
| 18:28.09 | DevWork_ | I didn't know what you wanted. |
| 18:28.23 | [TK]D-Fender | pastebin CLI. for everything. Always. |
| 18:28.30 | DevWork_ | Ok well I've never been asked to do that before. |
| 18:28.32 | [TK]D-Fender | I want full proof, not filtered bits |
| 18:28.48 | DevWork_ | So when you say "show me your call" I didn't know. |
| 18:28.48 | [TK]D-Fender | "I have problem, here, LOOK!" |
| 18:29.37 | ruben231 | <PROTECTED> |
| 18:29.59 | [TK]D-Fender | [13:14][TK]D-Fenderruthat number is practically worthless <- |
| 18:30.32 | [TK]D-Fender | ruben231: the number is NOT a comparaison of network responsiveness. It is APPLICATION level which can prioritize OPTIOSN however low it feels like |
| 18:55.03 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
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| 19:30.10 | *** join/#asterisk Janos (~Janos@190.113.106.41) |
| 19:30.32 | Janos | hello, anyone successfully using lua as dialplan lang here ? |
| 19:30.57 | [TK]D-Fender | Maybe 3-4 people.... |
| 19:31.04 | [TK]D-Fender | ANYWHERE :) |
| 19:31.13 | Janos | ouch |
| 19:31.22 | WIMPy | was just about to ask IN HERE? |
| 19:32.30 | *** join/#asterisk YannikS (~Miranda@31-18-193-221-dynip.superkabel.de) |
| 19:32.57 | Janos | is ael more user ? i don't like the traditional asterisk lang very much and would like to use something else if possible, no offense :P |
| 19:33.08 | Janos | more used* |
| 19:34.10 | WIMPy | Many of us use their own configuration and generate standard dialplan from that. |
| 19:35.08 | [TK]D-Fender | AEL is the worst.... |
| 19:35.32 | [TK]D-Fender | it just gets parsed back to extensions.conf language and is harder to debug. |
| 19:35.38 | [TK]D-Fender | Looks pretty, works ugly |
| 19:37.38 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:38.58 | Janos | thanks for the honesty, well i tried with lua and it worked, have a couple question that might be issues, for example in the lua example file they use this 'channel.LANGUAGE():set("fr")' to set the lang, but asterisk 1.8 complains that LANGUAGE() is not a registered function, check up it looks like it's more a channel variable, but if i do channel.language:get() it does not work either ... |
| 19:39.17 | Janos | i guess i should jsut go back to standard dialplan ? |
| 19:39.49 | WIMPy | Looks like it was already replaced by CHANNEL(lang) in 1.8. |
| 19:40.18 | WIMPy | So now we know when someone last looked at the lua example. |
| 19:40.49 | Janos | WIMPy, ahh ok, that would explain the problem lol |
| 19:43.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:45.09 | Janos | ok here is another that might not be lua's fault either, when i use app.Background and app.WaitExten in my "menu" context, it works great as long as WaitExten is the last thing used in the extension for example the "s" exten. But if i put something else after the WaitExten, then it does not work and instead it executes that something always, is the "t" exten the only fallback option for WaitExten/Background ? |
| 19:46.24 | [TK]D-Fender | no, it continues on in series, and uses 'T" if there is nothing else |
| 19:47.37 | WIMPy | Err? |
| 19:48.14 | WIMPy | You might also end up at i or e. |
| 19:49.05 | [TK]D-Fender | not quite a "fallback" as it is a "fell off the trail" :) |
| 19:52.04 | Janos | so to sumarize, it makes litte or no sense to put something after Backgroud/WaitExten and instead I should rely on t/T/i and e (never heard of e before :P) |
| 19:52.10 | Janos | little* |
| 19:52.49 | WIMPy | No, you can continue after WaitExten if nothing was entered. |
| 19:53.12 | Janos | yeah but from what i see, it always continue even if i enter something |
| 19:53.54 | WIMPy | Is your DTMF working? |
| 19:54.14 | Janos | yes ... |
| 19:54.23 | WIMPy | (assuming the call was answered) |
| 19:54.33 | Janos | i get the welcome message |
| 19:54.51 | WIMPy | Then maybe we have to blame LUA. |
| 19:54.52 | YannikS | I am having problems getting a Huawei E1550 modem working with chan_dongle, it is always showing "Mode: No Service". you can see the complete device status here: http://pastebin.com/n1M54xdW |
| 19:55.03 | WIMPy | What does that say about DTMF? |
| 19:55.20 | YannikS | any ideas how to get this running? the modem got no sim lock or voice lock |
| 19:55.58 | WIMPy | YannikS: Do you know what Mode: means? |
| 19:56.02 | Janos | and i can use extensions included in my menu context from other context "to_internal" which are my internal extensions, but the single digit extensions of the menu for example 0 does not work |
| 19:56.31 | Janos | it works if i dial 00 though, it's defined as 0 not 00 |
| 19:57.43 | WIMPy | That's a little hard to debug now. Doesn't seem to make much sense at first glance. We need to see the dialplan (and hope someone understands it) and a call with verbose 3. |
| 19:57.46 | YannikS | WIMPy: I don't exactly know, but "no service" doesn't sound that good to me :-) |
| 19:58.13 | WIMPy | YannikS: GSM Registration Status: Registered, home network sounds good to me. |
| 19:59.08 | YannikS | Okay, I will try to send a sms |
| 19:59.29 | WIMPy | You try to send the messaging service? |
| 19:59.51 | YannikS | what? |
| 20:00.13 | YannikS | Sorry, I don't understand what you mean |
| 20:00.15 | WIMPy | I'd try to send an SM, not the entire SMS :-) |
| 20:00.42 | YannikS | right ;-) |
| 20:00.52 | Janos | here is the context http://pastebin.com/uTd6Wn5s, let me get the debug |
| 20:01.46 | WIMPy | Great. I wouldn't even know if the syntax makes sense. |
| 20:11.40 | Janos | no worries, thanks for the help, i'll just go back to the standard lang, /cries |
| 20:21.48 | *** join/#asterisk ghost75 (~quassel@ipservice-092-211-042-097.pools.arcor-ip.net) |
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| 20:56.19 | *** part/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 21:02.43 | *** join/#asterisk boratynskikamil (~kamilbora@109.231.38.149) |
| 21:03.56 | boratynskikamil | Mainly, is it enought to add [channels] in cham_dahdi? What is a [trunkgroup]? |
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| 21:16.07 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 21:16.46 | jeffspeff | trying to configure an aastra phone, used the web interface, getting this error in * console.... chan_sip.c:27724 handle_request_register: Registration from 'sip:2099@' failed for '173.14.143.89:50805' - Not a local domain |
| 21:16.49 | jeffspeff | any ideas? |
| 21:18.27 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 21:21.14 | *** join/#asterisk opticron (~opticron@pianoben.ch) |
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| 21:32.21 | ghost75 | there are so many nat settings now, which are the most common? |
| 21:33.10 | opticron | is there a chan_pjsip equivalent to the SIPADDHEADER channel variable that you can use in originates for chan_sip? |
| 21:41.51 | jeffspeff | ghost75, using force_rport,comedia works for all of our phones. on and off the LAN |
| 21:42.06 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 21:43.29 | ghost75 | so comedia is like "yes" |
| 21:43.55 | Penguin | No. |
| 21:44.04 | Penguin | force_rport,comedia is like yes |
| 21:44.38 | Penguin | comedia is ONLY comedia. |
| 21:44.42 | ghost75 | Setting it to force_rport forces RFC 3581 behavior and disables symmetric RTP support. Setting it to comedia enables RFC 3581 behavior if the remote side requests it and enables symmetric RTP support. |
| 21:50.17 | DevWork_ | I'm calling from a pstn to my did, which is served to asterisk via a voip sip provider. I see it come in. It looks like I am "Reliably Transmitting" to my voip provider that the call is 403 Forbidden. |
| 21:50.42 | DevWork_ | So its like im not trusting the call coming in or something. |
| 21:50.44 | *** join/#asterisk vlad_sta_ (~vlad_star@88.226.34.183) |
| 21:50.52 | Penguin | Are you requiring them to auth TO YOU? |
| 21:51.23 | Penguin | If you have set secret=yourpassword in the peer, change it from secret to remotesecret. They will never auth to you. |
| 21:51.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:51.31 | DevWork_ | I think thats what is going on, that would be the symptom right? I don't know whats standard. I only input what ipcomm said to, and my configs are otherwise barren. |
| 21:51.33 | DevWork_ | Ok |
| 21:52.00 | Penguin | Rule number 1 is that no ITSP will ever give you the correct values to configure your asterisk. |
| 21:52.00 | DevWork_ | There was also something like uhh.. anonymous=allowed or something I read about. |
| 21:52.07 | Penguin | Delete it. |
| 21:52.17 | DevWork_ | I'm not using that. |
| 21:52.25 | [TK]D-Fender | Neither is anyone else |
| 21:52.43 | Penguin | secret means you require the password and send the password. remotesecret means you send it but you do not require it. |
| 21:54.03 | *** part/#asterisk opticron (~opticron@pianoben.ch) |
| 21:55.06 | *** join/#asterisk vandyk (bd3b04b4@gateway/web/cgi-irc/kiwiirc.com/ip.189.59.4.180) |
| 21:55.28 | ghost75 | wth everyone is using secret instead remotesecret |
| 21:55.45 | Penguin | ignorance |
| 21:56.00 | ghost75 | For historical reasons, if no remotesecret is supplied for an |
| 21:56.00 | ghost75 | ; outbound registration or call, the secret will be used. |
| 21:57.00 | Penguin | I've used several ITSPs and they all give shit configuration instructions. If you don't know how to configure your own asterisk, their information works about 50% of the time on the best ones. |
| 21:57.20 | DevWork_ | Now im reading that I am 401 unauthorized to make a call out. Inbound call from them works fine. |
| 21:57.26 | vandyk | I'm trying to develop an application that will originate a call using AMI. I need to call a mobile phone as originate and another mobile phone as destination |
| 21:57.41 | ghost75 | so back to nat, how do i know which mode works best? |
| 21:57.56 | Penguin | Be sure to register to them. Many ITSPs require you to be registered before they will accept calls from you. |
| 21:58.06 | Penguin | If you are already registered, pastebin your peer entry. |
| 21:58.23 | DevWork_ | let me try to dial my cell from a softphone one sec |
| 21:59.00 | ghost75 | externhost/externip is only used when i want to use external devices right? |
| 21:59.08 | DevWork_ | Ok so SIP phone => asterisk => ipcomms to cell phone works. |
| 21:59.23 | Penguin | Good job! |
| 22:00.09 | DevWork_ | The issue is that if I call from any phone, to my ipcomms DID, that comes into asterisk fine, but then I want to Dial(mycell) and THAT isn't working. |
| 22:00.18 | DevWork_ | That's when I get the 401. |
| 22:00.32 | Penguin | You'll have to pastebin the dialplan and the peer configuration. |
| 22:00.36 | DevWork_ | ok. |
| 22:01.12 | ghost75 | what is authuser for? |
| 22:01.21 | vandyk | to originate the call I'm using this |
| 22:01.23 | vandyk | Action: Originate\r\n" |
| 22:01.23 | vandyk | Channel: Local/" + txtNumberToDial.Text + "@from-internal\r\n" |
| 22:01.23 | vandyk | Exten: " + txtNumberToDial.Text + "\r\n" |
| 22:01.23 | vandyk | Context: from-internal\r\n" |
| 22:01.23 | vandyk | Callerid:: Nomad\r\n" |
| 22:01.23 | vandyk | Priority: 1\r\n" |
| 22:01.24 | vandyk | Account: " + txtUserExtension.Text + "\r\n" |
| 22:01.24 | vandyk | Async: yes\r\n\r\n" |
| 22:01.28 | Penguin | ~pb |
| 22:01.28 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:02.17 | [TK]D-Fender | vandyk: do NOT flood in here.... |
| 22:02.38 | DevWork_ | http://pastebin.com/nmYKHkpD |
| 22:04.04 | Penguin | I don't see any sign of the VOIPOUT variable being set to anything. |
| 22:04.08 | vandyk | ok sorry |
| 22:04.10 | vandyk | http://pastebin.com/39T3j51v |
| 22:04.40 | DevWork_ | its set in [globals] as VOIPOUT=SIP/2way-outbound |
| 22:04.58 | DevWork_ | thats the only line in globals. |
| 22:06.01 | DevWork_ | And I verified just now via dialplan show globals that there is 1 variable and its set as mentioned, |
| 22:06.06 | vandyk | on Exten, should always be an extension on ASterisk server? |
| 22:06.23 | Penguin | I would much prefer the peer to look like this: http://pastebin.com/G5xCUyEC |
| 22:06.34 | [TK]D-Fender | vandyk: it's ALWAYS on your server. that is your DIALPLAN you are directing it to |
| 22:06.57 | vandyk | or could be a number to be dialed using outbound routes? |
| 22:07.23 | Penguin | Yep, that's what extensions do! |
| 22:07.31 | Penguin | They do stuff, yo. |
| 22:08.04 | Penguin | [tk]d-fender: http://stlouis.craigslist.org/cto/4277537627.html |
| 22:08.34 | DevWork_ | alright so defaultuser, disallow all is set above in general so it cascades. insecure=very? they recommended that in the guide, ill take it off. nat=no? why |
| 22:08.47 | Penguin | See rule number 1. |
| 22:08.54 | Penguin | THEY DON'T KNOW WHAT THEY ARE DOING |
| 22:08.55 | [TK]D-Fender | At least it isn't a civic with a skirt kit and big honking wing on the back |
| 22:09.09 | Penguin | Civics must not be the tits. |
| 22:09.41 | vandyk | if I want to dial to external number 5555-8988 and for number 5555-9933 and connect both on same call, how should I set on AMI ? |
| 22:09.45 | DevWork_ | I didn't ask why they wanted insecure=very. I asked why you said to set nat=no. |
| 22:09.53 | Penguin | They are not behind nat. |
| 22:10.27 | DevWork_ | so insecure=very, that means port,invite right? |
| 22:10.43 | Penguin | Yes, and that would be unnecessary for your config. |
| 22:10.46 | DevWork_ | which means I match by their ip, disregard their port, and I also don't auth incoming invites. |
| 22:11.03 | Penguin | You already don't require auth from them because you are using remotesecret. |
| 22:11.26 | DevWork_ | So remotesecret supersedes insecure=invite |
| 22:11.34 | Penguin | Totally different things. |
| 22:11.37 | DevWork_ | How do you know im not remapping their port on my side of the DMZ? |
| 22:11.51 | DevWork_ | "You already don't require auth from them because you are using remotesecret." |
| 22:12.01 | DevWork_ | Sounds like its redunant, and thus superseded no? |
| 22:12.05 | Penguin | They will never auth to you. |
| 22:12.13 | DevWork_ | I know they wont. |
| 22:12.22 | DevWork_ | insecure=invite makes it so I don't require authentication? |
| 22:12.30 | DevWork_ | sounds like without it I do. |
| 22:13.07 | Penguin | If your peer has a secret set, you are requiring auth on the INVITE. |
| 22:13.43 | DevWork_ | So, as I mentioned, insecure=invite becomes redundant when remotesecret is set? |
| 22:13.57 | Penguin | If your peer has insecure=invite, you're requiring the auth on INVITE from the secret value being set, but then turn around and disregard the requirement. |
| 22:14.39 | DevWork_ | alright ill log this call one sec. |
| 22:18.32 | DevWork_ | http://pastebin.com/QceL3Z2k |
| 22:19.00 | Penguin | I don't read debugs with munged data. |
| 22:19.37 | ghost75 | is authuser some addition to defaultuser? |
| 22:19.37 | DevWork_ | I only took out my phone numbers. |
| 22:19.49 | DevWork_ | I'm not posting these publically. |
| 22:20.04 | Penguin | and some IP addresses that I have no idea what could have been there. |
| 22:20.24 | *** join/#asterisk digiv (~digiv@as1.si.umich.edu) |
| 22:20.27 | DevWork_ | yea well they are my wan, thats the only one i took out. |
| 22:20.28 | Penguin | Maybe someone else will look at it. |
| 22:20.44 | DevWork_ | I don't see why my WAN ip is relevant. |
| 22:21.04 | ghost75 | Penguin: guess how long it took me to get password from my sip debug |
| 22:21.27 | Penguin | I would think a long time since the password shouldn't be in the sip debug. |
| 22:21.45 | ghost75 | you can brute force it |
| 22:21.47 | Penguin | Hey Doc, I'm here for my physical. No I won't take off my cloths so you can examine me. |
| 22:22.09 | DevWork_ | People scan pastebin.com for posted configs. I only took off my ip and phone number, it really doesn't change the log at all. |
| 22:22.25 | DevWork_ | assum its 4.2.2 |
| 22:23.36 | ghost75 | Penguin: the script ran 3min |
| 22:24.05 | Penguin | You should pick better passwords. |
| 22:24.15 | ghost75 | it was default from isp |
| 22:25.43 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 22:27.46 | ghost75 | so think about asking people to post their sip debug on pastebin |
| 22:28.16 | DevWork_ | I'll post mine I just have to secure my firewall first. |
| 22:28.33 | ghost75 | i mean to be carefull about what to post |
| 22:28.57 | DevWork_ | http://www.leakedin.com/page/4/ this is only one public source that scans it for leaked information |
| 22:29.44 | ghost75 | i only need to search on pastebin for sip debugs from a specific isp where the password scheme is known and u get password in no time |
| 22:32.25 | DevWork_ | http://pastebin.com/62deSxHf full log. |
| 22:32.32 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 22:33.57 | DevWork_ | Basically 401 at 128. |
| 22:34.17 | [TK]D-Fender | irrelevant |
| 22:34.28 | [TK]D-Fender | Your system responds with auth. that is a CHALLENGE |
| 22:34.31 | [TK]D-Fender | Not a refusal |
| 22:34.32 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 22:34.39 | [TK]D-Fender | They accept your call and it falls through on their end |
| 22:34.43 | [TK]D-Fender | (6030) |
| 22:34.46 | [TK]D-Fender | 603 |
| 22:35.14 | ghost75 | what means "auth" |
| 22:35.25 | DevWork_ | 603 declined? |
| 22:35.35 | DevWork_ | Ok, so why is it doing that? |
| 22:36.23 | DevWork_ | If I dial directly an outside number from my sip softphone, it goes thrugh them. |
| 22:36.58 | [TK]D-Fender | show us |
| 22:37.08 | DevWork_ | Sec. |
| 22:37.21 | DevWork_ | I did earlier btw. |
| 22:37.36 | [TK]D-Fender | link? |
| 22:38.19 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 22:38.19 | DevWork_ | Its in scroll back and I cleared, so let me make another. |
| 22:39.23 | DevWork_ | Works now. Problem wasn't the call log. Problem was the dialplan linked earlier. |
| 22:40.47 | DevWork_ | ok works good. interesting going to go test this from an outside line. |
| 22:41.35 | [TK]D-Fender | A log isn't a problem ... it may be a TRACE of one however |
| 22:42.07 | DevWork_ | Well in that case, I pasted the broken one, but no matter, it was the extensions.conf |
| 22:42.13 | DevWork_ | I changed a lot in it, reloaded, works now. |
| 22:43.59 | Penguin | Forgot the "dialplan reload" I guess. |
| 22:44.55 | Penguin | What I saw in the debug looked like good dialplan, though. |
| 22:47.04 | DevWork_ | Oh I reloaded just now, what i showed you I was running with. |
| 22:47.22 | DevWork_ | idk why it didn't work, i was tweaking something else. then I changed to a different dialout number instead of my cell because it died. |
| 22:47.31 | DevWork_ | so i reloaded |
| 22:47.42 | DevWork_ | then he asked for a trace, and it was working. |
| 22:47.52 | DevWork_ | I need to break out these scripts into includes. |
| 22:47.57 | DevWork_ | its getting a little long all in one. |
| 22:47.59 | Penguin | I guess success is a good place to stop. |
| 22:48.08 | DevWork_ | yea I commited to my git repo so all good. |
| 22:50.03 | DevWork_ | so I know im not supposed to name my phones after whos using it or where it is. But how do I have a user at extension 200, and then ring up his device, would i manually change the dialplan if they moved around or is there some app/module im not utilizing that does this? |
| 22:50.33 | DevWork_ | With the meridian system, I just go into the set, and set its extension that rings, but then set 200 is user y always. |
| 22:51.02 | Penguin | You can manually change the dialplan or you can do what I do and keep a static dialplan but change the device data in the database. |
| 22:52.06 | DevWork_ | Yea we have a company directory in a database already with their extension and everything, I would just need a table that has a link for deviceid and extension in it. |
| 22:52.19 | Penguin | Extension 1234 dials ${DB(phones/${EXTEN}/device)} |
| 22:53.17 | Penguin | The db entery for phones/1234/device = SIP/00aa11bb22cc |
| 22:53.47 | Penguin | Need to give the guy who uses extension 1234 a new phone? Change the value in the db. |
| 22:53.53 | DevWork_ | can two softphones register under the same deviceid? |
| 22:54.14 | Penguin | Not really. They _can_ do it, but only the last one gets the call. |
| 22:54.24 | DevWork_ | so both wouldnt ring. |
| 22:54.28 | Penguin | right |
| 22:54.37 | DevWork_ | not that i wanted to do that |
| 22:54.44 | DevWork_ | just curious what would happen |
| 22:55.12 | Penguin | If I need two phones on the same device, I'll append -a and -b to the base name. |
| 22:55.26 | Penguin | 00aa11bb22cc-a and 00aa11bb22cc-b |
| 22:56.03 | DevWork_ | so a register to asterisk to a entry in sip, each [entry] is a one to one mapping. |
| 22:56.03 | Penguin | Need them both to ring? Put them both in the db entry. |
| 22:56.22 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 22:56.27 | DevWork_ | and the db would return SIP/1&&SIP/2 |
| 22:56.28 | Penguin | db entry is SIP/00aa11bb22cc-a&SIP/00aa11bb22cc-b |
| 22:56.34 | DevWork_ | yea cool |
| 22:57.04 | Penguin | My extension has like eight devices in it. :( |
| 22:57.30 | DevWork_ | Now when they say secure inbound contexts, what does that entail? basically not letting them use your dialplan to bounce calls where they shouldn't go? |
| 22:57.43 | DevWork_ | like calls to china.... |
| 22:59.01 | Penguin | Mostly it means the inbound context has no way of getting to an extension in the outbound context. |
| 22:59.11 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |
| 22:59.41 | Penguin | While I generally don't give phones access to the inbound context, I can't think of how it would hurt. |
| 23:00.02 | Penguin | You just don't want random incoming calls to be able to go back outbound. |
| 23:01.45 | Penguin | It's a good idea to build hierarchical contexts so that some things can use a common context without that common context having access to any context that can access it. |
| 23:07.29 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 23:21.06 | *** part/#asterisk mjordan (~matt@nat/digium/x-tittlhatulyifpuv) |
| 23:23.09 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 23:29.29 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 23:32.49 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 23:38.27 | *** join/#asterisk posix (~pah@bl5-10-203.dsl.telepac.pt) |
| 23:38.32 | posix | hello |
| 23:40.32 | posix | i'm testing version 12 of asterisk on a quality environment, but i'm stuck with a issue regarding sip responses from asterisk being sent to the port 5060 instead of the original source port of the client |
| 23:40.54 | posix | it's basically acting as insecure=port was set |
| 23:41.16 | posix | by ignoring the port from the request came |
| 23:41.31 | Penguin | Did you look at nat=force_rport? |
| 23:41.46 | posix | little was changed in the configuration fetched from the production (versions 1.8.x) |
| 23:41.57 | posix | yes i did |
| 23:42.06 | posix | and i've tested some different settings |
| 23:42.21 | posix | with NATted and non-NATted clients |
| 23:42.49 | posix | and used nat=force_rport,comedia on some of the tests |
| 23:43.03 | posix | but I can try again, just to make sure |
| 23:44.37 | posix | nop, not solving the issue |
| 23:45.06 | posix | i'm basically requesting with an OPTIONS method now |
| 23:45.32 | posix | just to reduce everything else that might cause problems here |
| 23:45.51 | posix | but the destination port of the answer seems to be always the same |
| 23:45.54 | posix | whatever i do |
| 23:46.37 | posix | is there any default options affecting these behaviors that have been changed from 1.8 to 12? |
| 23:46.42 | posix | are* |
| 23:59.54 | *** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183) |