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| 00:44.49 | inetd50 | hi |
| 00:44.58 | inetd50 | any freebsd users of asterisk here? |
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| 01:16.31 | ruben23 | hi guys, i have an asterisk server which on publci ip and dont used NAT, but when i call client cant hear me at all..any ide what could be casuing it. |
| 01:18.11 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:18.33 | Inssomniak | Does asterisk need a timing source (dahdi dummy) for just bridging RTP streams? (no transcoding) |
| 01:19.08 | jaytee | Inssomniak, it shouldn't. |
| 01:20.11 | jaytee | and versions 1.6.2 and higher don't require dahdi dummy as long as the server hardware can support high performance timing. |
| 01:28.16 | beek | evenin' jaytee |
| 01:28.51 | jaytee | evenin' beek |
| 01:29.52 | Inssomniak | ok Im just trying to find out if my crackling audio is because of trying to use dahdi-dummy in vmware esxi, but if rtp bridging doesnt use it then problem could be somewhere else |
| 01:30.39 | jaytee | "crackling audio" is usually jitter |
| 01:31.44 | WIMPy | It will use it if available. |
| 01:32.17 | WIMPy | You need some timing source and dahdi would be the preferred choice. |
| 01:32.20 | Inssomniak | rtp bridging will use dahdi-dummy if its available? |
| 01:33.01 | jaytee | is every call affected? extension to extension or just extension to external sip? |
| 01:33.08 | WIMPy | yes |
| 01:33.52 | Inssomniak | not every call is affected, and I cant say for sure that if its just extension to extension or external->extension |
| 01:34.11 | Inssomniak | but an affected call will remain crackly the whole time and a clear call will remain clear the whole time |
| 01:34.15 | jaytee | really? my understanding was that dahdi-dummy or zaptel dummy drivers were only needed for applications like MeetMe and not for normal calls endpoint to endpoint |
| 01:34.39 | WIMPy | Correct :-) |
| 01:34.56 | WIMPy | MeetMe requires dahdi (or zaptel) timing. |
| 01:34.56 | Inssomniak | OK now Im confused. |
| 01:35.20 | WIMPy | Everything else requires *some* timing source. |
| 01:35.58 | WIMPy | If dahdi is available it will be first choice over timerfd and others internal to Asterisk. |
| 01:36.34 | Inssomniak | so a simple call that asterisk is bridging RTP for without doing any transcoding still will use dahdi-dummy (or some timing source) |
| 01:37.17 | WIMPy | It will use dahdi if available, something else otherwise. |
| 01:38.11 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 01:38.25 | Inssomniak | ok. I wasnt sure. So a poor dahdi_test result can cause poor audio quality in a plain old call |
| 01:38.28 | jaytee | I typically install dahdi even if I don't have the hardware that needs it just in case I want to add a card in the future I won't have to recompile everything |
| 01:38.53 | WIMPy | Inssomniak: Correct |
| 01:39.06 | Inssomniak | Best: 99.972 -- Worst: 99.942 -- Average: 99.951728, Difference: 99.992644 |
| 01:39.33 | jaytee | those are fine stats and don't indicate a timing problem |
| 01:39.58 | WIMPy | Should be good enough, yes. |
| 01:40.05 | Inssomniak | (this is esxi and I am moving it to dedicated hardware), I do have problem with recordings and voicemail |
| 01:40.06 | WIMPy | If it's correct. |
| 01:40.46 | WIMPy | What does 'timing test' give you? |
| 01:41.05 | jaytee | Inssomniak, what kind of server hardware are you running it on and how many other virtual machines are competing for that server's resources? |
| 01:42.15 | Inssomniak | jaytee right now the server has one other VM and its a router, no other vms installed, its a dell r710 server, 12 gigs, 2x 2.13ghz xeons |
| 01:42.32 | Inssomniak | how do I run this "timing test" command? |
| 01:42.41 | WIMPy | At *CLI |
| 01:43.29 | jaytee | Inssomniak, then the hardware isn't overworked |
| 01:44.54 | Inssomniak | I cant run timing test, command not found at *CLI |
| 01:45.08 | WIMPy | What Version are you running? |
| 01:46.46 | Inssomniak | Asterisk SVN-branch-C.3-duckpond-r292720M (switchvox box I managed to get CLI access to for troubleshooting) |
| 01:46.51 | Inssomniak | no idea what version that is |
| 01:47.53 | WIMPy | Neither do I, but it's 8964 updates behind. |
| 01:47.58 | Inssomniak | lol |
| 01:48.10 | Inssomniak | yea the whole thing seems kinda old |
| 01:49.40 | Inssomniak | I should just stop trying to chase problems down that may just dissapear once I have it off of a VM |
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| 01:50.11 | Inssomniak | I read that a dahdi test result of anything less than 99.975 was going to be problems but you guys say it should be OK? |
| 01:50.30 | WIMPy | Yes, that seems to be a bit hit or miss. We've had ppl running Asterisk fine in VMs, others had trouble. |
| 01:51.20 | WIMPy | It shouldn't cause obviously bad audio. |
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| 01:51.48 | WIMPy | It will probably be obvious when listening to music. |
| 01:51.58 | Inssomniak | lots of trouble recording non-garbled IVR recordings |
| 01:52.05 | Inssomniak | have to try several times often to get a good one |
| 01:52.12 | WIMPy | obhearious :-) |
| 02:04.38 | Inssomniak | internal_timing=yes |
| 02:04.55 | Inssomniak | does that mean its not using the dahdi dummy module for timing? |
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| 02:24.08 | AlcornTech | I have an Asterisk system that is answering from a SIP provider and passing the calls to another server via Dial(Zap/g1/other_sys_num). Is there a way to monitor or have asterisk print the passed DTMF to the CLI ? |
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| 02:25.59 | carrar | Sounds Illegal! |
| 02:26.42 | AlcornTech | carrar: no, just a ghetto 1994 way of using asterisk to forward SIP channels thru a diallogic card to a VB4 program :-\ |
| 02:26.58 | carrar | turn on debuging |
| 02:27.08 | AlcornTech | already did "set verbose 100" |
| 02:27.28 | AlcornTech | we're talking about a 2004 CVS release. |
| 02:27.34 | *** part/#asterisk Howie69 (~Howie69@65.111.172.187) |
| 02:35.49 | AlcornTech | ah... I imagine its fairly difficult to log DTMF if its inband rather than rfc2833? :) |
| 02:43.12 | pabelanger | func_odbc.conf: Anybody using readsql=SELECT LAST_INSERTED_ID(); ? |
| 02:54.50 | pabelanger | Heh, deadlock in the middle of playing a prompt; Asterisk 1.8.1 |
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| 03:19.21 | csnook | anyone know a TCPA scrubbing service that doesn't charge an arm and a leg for the DNC scrubbing I don't need? |
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| 03:25.02 | JaSW- | does anyone know if running asterisk -g could cause the system to hang after a while? |
| 03:36.08 | pabelanger | JaSW-: shouldn't. You can try: CLI> core show locks |
| 03:36.13 | pabelanger | to see if you have a deadlock |
| 03:39.18 | JaSW- | i dont have a core show locks command |
| 03:39.28 | JaSW- | hmm |
| 03:39.39 | *** part/#asterisk k-man (~jason@unaffiliated/k-man) |
| 03:39.54 | JaSW- | Asterisk1*CLI> core show locks |
| 03:39.54 | JaSW- | No such command 'core show locks' (type 'core show help core show' for other possible commands) |
| 03:40.54 | JaSW- | nm figured it out thanks |
| 03:44.01 | pabelanger | JaSW-: Ya, you need to recompile to DEBUG_LOCKS enabled |
| 03:45.55 | pabelanger | So, anybody explain this? I'm calling back to back ODBC functions in my dialplan, the first function works fine, but the second one is failing. Even tho I pass different arguments to function 2, it seems to still be using arguements from the 1st? |
| 03:46.12 | pabelanger | If that is hard to follow, see the following link: http://asterisk.pastebin.com/sjyC8RKj |
| 03:46.20 | pabelanger | Has dialplan and debug output |
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| 04:00.22 | JaSW- | pabelanger: Ive never used ODBC functions in a dialplan |
| 04:00.41 | JaSW- | [Jan 11 21:42:01] WARNING[20776] tcptls.c: Accept failed: Too many open files |
| 04:00.41 | JaSW- | [Jan 11 21:42:01] WARNING[20781] pbx_spool.c: Unable to open directory /var/spool/asterisk/outgoing: Too many open files |
| 04:00.41 | JaSW- | [Jan 11 21:42:01] WARNING[20766] asterisk.c: Accept returned -1: Too many open files |
| 04:00.44 | JaSW- | any idea? |
| 04:04.25 | atan | Close some files. |
| 04:06.51 | JaSW- | hehe |
| 04:06.57 | JaSW- | i'm looking into it :) thanks |
| 04:07.21 | p3nguin | There ought to be some type of wiki that describes that problem and how to fix it. |
| 04:07.37 | p3nguin | It makes me think your ulimit is not where it should be. |
| 04:07.56 | JaSW- | i set the ulimit to 65000 a few weeks back and still get this |
| 04:08.27 | JaSW- | i was getting a seg fault before the ulimit, but now asterisk just hangs and errors out |
| 04:09.27 | p3nguin | I'd think unlimited would be a fine idea. |
| 04:09.47 | JaSW- | ok i'll try that |
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| 08:12.06 | schmidts | good morning |
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| 08:14.13 | FlashDeluxe | hi @all! i got a question if i get a call, i want to see the number of the caller +a digit infront of that number in my display :) how do i do it? i already tried it but it doesn`t work, here is an output of my extensions.conf http://pastebin.de/13772 and here is an output of the messages i get in the cli http://pastebin.de/13771 |
| 08:15.05 | FlashDeluxe | as you can see, the callerid is set, but it doesn`t appear in the Answer() application, can anybody tell my why? |
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| 08:17.39 | ChannelZ | use NoOp to output it |
| 08:18.04 | schmidts | Flashdeluxe hello, i think you mix Callerid(num) and EXTEN. this is not the same |
| 08:18.28 | ChannelZ | NoOp(${CALLERID(all)}) |
| 08:21.10 | kaldemar | FlashDeluxe: your phone probably shows the name part of caller id, and you are prefixing the number part with the 0. |
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| 08:21.59 | kaldemar | FlashDeluxe: and, the answer application is not supposed to show anything. |
| 08:29.20 | FlashDeluxe | klademar: ok, so do i understand it right, i did set it up right, but the pone doesn`t show the right things? |
| 08:29.52 | ChannelZ | your problem is you're answering and then dialing |
| 08:30.41 | ChannelZ | well.. it could be anyway depending on your sip config |
| 08:30.54 | ChannelZ | so I take it back. It could also be your phone |
| 08:31.34 | ChannelZ | but have you also looked at your variables? I think that was sort of your original question |
| 08:33.56 | FlashDeluxe | i do not have much variables and i was wondering if the CALLERID variable is the right way to do it? |
| 08:34.09 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 08:34.35 | ChannelZ | it is (it's actually a function) but you are setting them to other variables, have you checked to see if those are even right? IE you're maybe erasing the caller ID |
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| 08:37.23 | FlashDeluxe | i am absolutely sure that the other variables are right, in the cli i see that the callerid is set with the prefix, but asterisk is answering the number without the prefix, so i am wondering if set callerid is the right way to do it |
| 08:40.03 | kaldemar | FlashDeluxe: what do you mean answering the number without the prefix? if you mean "DAHDI/i1/02282xx-1", it is the channel name and it is not supposed to change. |
| 08:40.29 | FlashDeluxe | klademar: oh, i didn`t know that |
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| 09:10.59 | c0rnoTa | Hello everyone |
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| 09:11.43 | c0rnoTa | I have problems with reloading module chan_skinny.so on asterisk 1.4.35 |
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| 09:13.39 | c0rnoTa | couldn't find any information in bugtracker for that issue |
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| 09:19.07 | kaldemar | c0rnoTa: what kind of problems? |
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| 09:23.44 | c0rnoTa | kaldemar: when i'm trying to reload chan_skinny ro apply new settings from configuration file, i'v got : ast_module_reload: The module 'chan_skinny.so' was not properly initialized. Before reloading the module, you must run "module load chan_skinny.so" and fix whatever is preventing the module from being initialized. |
| 09:24.17 | c0rnoTa | kaldemar: but module loaded and some active channels over skinny protocol exists in my system |
| 09:27.04 | c0rnoTa | kaldemar: i'm trying to unload module, but it gives me "loader.c:502 ast_unload_resource: Unload failed, 'chan_skinny.so' is not loaded." |
| 09:28.36 | c0rnoTa | kaldemar: "core show channeltypes" shows loaded skinny protocol |
| 09:32.22 | eject_ck | kaldemar: hello today :) |
| 09:33.11 | eject_ck | Fixed my strange problem with registration by upgrading to 1.6.2 :) |
| 09:33.36 | kaldemar | what was is? |
| 09:33.39 | kaldemar | it.. |
| 09:34.01 | kaldemar | c0rnoTa: sounds strange. did it work normally at some point? |
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| 09:35.53 | schmidts | c0rnota try to start asterisk direct from the console and search if you see something when its loaded the first time, |
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| 09:39.04 | c0rnoTa | kaldemar: i think, it work normally for official skinny channel: calls are going on cp7921, no problems with sound, transfers and cp7911, cp7961 doesn't work :) so, I think it working fine. |
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| 09:40.48 | c0rnoTa | schmidts: i'll try to start asterisk with noload => chan_skinny.so and then module load chan_skinny in console to find some errors |
| 09:43.18 | c0rnoTa | schmidts: kaldemar: module loaded without any errors or warnings with verbose level 3 and debug level 1 |
| 09:43.49 | schmidts | c0rnota and when you try to reload it, you get the error again? |
| 09:45.55 | c0rnoTa | schmidts: yeap, It gives me "ast_module_reload: The module 'chan_skinny.so' was not properly initialized." |
| 09:47.08 | schmidts | c0rnota what do you see with module unload? |
| 09:47.31 | c0rnoTa | schmidts: ast_unload_resource: Unload failed, 'chan_skinny.so' is not loaded. |
| 09:48.07 | schmidts | c0rn0ta and module show? |
| 09:48.43 | c0rnoTa | schmidts: and i can make another module load chan_skinny, and it will be loaded. but list of devices will have duplicates. and calls wouldn't go correctly. Phone will not ring on incoming call. |
| 09:49.13 | c0rnoTa | schmidts: module show: "chan_skinny.so Skinny Client Control Protocol (Skinny) 5" |
| 09:49.30 | schmidts | the 5 is the use counter, right? |
| 09:49.37 | c0rnoTa | schmidts: yeap |
| 09:50.01 | schmidts | maybe its just a wrong warning, instead of module couldnt be unload cause its in use, you get the above?? |
| 09:50.54 | c0rnoTa | schmidts: i don't think so, because "show channels" is clear |
| 09:51.49 | hrhrhr | 1.6.2.15 is the latest 1.6? |
| 09:51.53 | schmidts | c0rnota ok then there could be a bug with the use counter. something increase the use counter, exit and doesnt decrease it |
| 09:52.22 | c0rnoTa | schmidts: yeap.. looks like you are right |
| 09:52.24 | schmidts | hrhrhr 1.6.2.16 will be released, maybe today or tomorrow |
| 09:52.42 | schmidts | c0rnota which version do you use? |
| 09:53.39 | c0rnoTa | schmidts: it's "Asterisk 1.4.35 built by root" |
| 09:55.05 | c0rnoTa | schmidts: I haven't found any changes in changelog of asterisk 1.4 wired with skinny |
| 09:55.32 | schmidts | c0rnota does it work with an older version? |
| 09:56.19 | schmidts | has to go afk for 30 mins |
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| 09:58.50 | c0rnoTa | schmidts: on asterisk 1.4.33.1 it doesn't work too |
| 10:02.52 | hrhrhr | cheers schmidts |
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| 10:53.46 | schmidts | c0rnota do you know a version when this worked? even if there was no recent change to skinny maybe its an old bug ;) |
| 10:54.23 | c0rnoTa | schmidts: no, i don't know |
| 10:54.44 | c0rnoTa | schmidts: i think it's an old bug. |
| 10:54.57 | schmidts | c0rnota ok :D i will take a look at the code if i can find any place where the use counter is not decreased before a return |
| 10:55.17 | c0rnoTa | schmidts: thanks a lot! |
| 10:55.44 | c0rnoTa | schmidts: i think most of all use chan_sccp. But it crashs some time :) |
| 11:00.31 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 11:01.00 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 11:08.37 | *** join/#asterisk atommat (~atommat@pushbutton.plus.com) |
| 11:09.09 | atommat | hi, any BRI people here? |
| 11:10.22 | Chainsaw | atommat: As in Basic Rate Interface? |
| 11:11.16 | WIMPy | atommat: ~ask |
| 11:11.22 | WIMPy | or http://voice.yeti.dk/AvI |
| 11:11.47 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:12.24 | *** join/#asterisk Poincare (~jefffnode@ip-188-118-11-57.reverse.destiny.be) |
| 11:12.49 | atommat | tnx |
| 11:13.17 | WIMPy | Err |
| 11:13.22 | WIMPy | ~ask |
| 11:13.22 | 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. |
| 11:13.41 | WIMPy | That way :-) |
| 11:16.08 | atommat | i have many questions, the first off which is the timing issue across the 4 spans. I have the 1st set 1 and the remaining spans as zero. The auto config sets them to 1234 respectively. wihch is correct? |
| 11:16.58 | WIMPy | Maybe you tell us what you've got first. |
| 11:18.39 | WIMPy | The timing source should be any interface connected to the PSTN. |
| 11:18.52 | kaldemar | atommat: zero means that the span provides timing for the other end, a non-zero digit means that the span takes timing from the other end, smaller digits take preference. there's no yes/no answer to your question as is. |
| 11:22.24 | kaldemar | there is a pretty good explanation in the sample system.conf. |
| 11:22.57 | *** join/#asterisk cynicismic (~rclark@188-223-82-64.zone14.bethere.co.uk) |
| 11:24.24 | WIMPy | And the whole thing is pretty pointless. |
| 11:26.50 | coppice | its pretty important |
| 11:27.44 | WIMPy | From my experience the timing in no where near good enough for that to make a difference. |
| 11:28.32 | atommat | We have a asterisk server with a digium b410p card. we have an inbound route set up as a test. it goes straight to the holding music. the dahdi is reporting a healthy connection but the spans keep going up and down and the sound when dialling in is crackled and poor. I think it my be a timing issue but im not entirely certain. |
| 11:29.28 | atommat | occasionally, we do get a clear sound but it then reverts to either a poor one again or the spans go down |
| 11:29.47 | coppice | you'd better make sure you sync to a clock from the PSTN, and provide the clock for any downstream ports, like a PBX. all the ports from the PSTN will normally be in sync, as they come from the same atomic clock |
| 11:30.01 | WIMPy | Might be interrupt problems. |
| 11:30.51 | WIMPy | If they come from different carriers they can be out of sync if you've go bad luck. |
| 11:30.52 | Chainsaw | leifmadsen: Sorry I had to leave suddenly yesterday. |
| 11:31.05 | atommat | by this you mean using this span=1,1,0,ccs,ami rather than this span=1,0,0,ccs,ami |
| 11:31.26 | Chainsaw | leifmadsen: To recap, I have a _#TLIO#4XXXX extension that claims not to exist. However, I can set hints on it. |
| 11:31.28 | coppice | all the carriers should be in sync. if they aren't a modem won't work on a call between those carriers |
| 11:31.36 | atommat | all four spans are from bt |
| 11:32.24 | atommat | I have tried reducing the spans to 2 and testing this but the result is the same. could it be related to echo cancellation? |
| 11:32.32 | WIMPy | coppice: a. "should be", yes, b. modems often don't work any more anyway because someone uses voip in between. |
| 11:33.18 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 11:33.20 | WIMPy | atommat: Does the card have an IRQ on it's own or is it shared? |
| 11:33.42 | Chainsaw | leifmadsen: http://www.pastie.org/1455817 |
| 11:34.31 | atommat | im not sure how to answer that but dahdi_scan report each spa as having irq=217 |
| 11:34.36 | atommat | *span |
| 11:35.16 | leifmadsen | dies |
| 11:35.39 | Chainsaw | revives leifmadsen with an automatic defibrillator |
| 11:35.44 | Chainsaw | leifmadsen: LIVE! |
| 11:35.50 | Chainsaw | leifmadsen: And tell me about my dialplan issue. |
| 11:35.59 | leifmadsen | ok gotta go |
| 11:36.09 | WIMPy | atommat: Err, is that an PCI-e card? |
| 11:36.59 | atommat | " is a half-length universal 3.3V and 5.0V 32-bit PCI 2.2 card" |
| 11:37.06 | atommat | B410p |
| 11:37.37 | WIMPy | That makes a no. Are you using some virtualisation? |
| 11:38.01 | atommat | no its running on a p4 |
| 11:38.08 | atommat | no virtualisation |
| 11:38.10 | *** join/#asterisk ChrisH (~dg7pc@dslb-092-072-131-173.pools.arcor-ip.net) |
| 11:38.10 | Chainsaw | Yuck, a P4 :( |
| 11:38.15 | atommat | lol |
| 11:38.24 | *** join/#asterisk stix_ (~stix@firewall.o4.dk) |
| 11:38.29 | atommat | i know, is that gonna stop the show? |
| 11:38.39 | Chainsaw | atommat: Anyhow, could we see /proc/interrupts on pastebin please? |
| 11:39.16 | Chainsaw | atommat: And the output of dmesg as well please. Also on pastebin, it will be a lot. Don't chop anything off. |
| 11:39.20 | WIMPy | I've only seen that kind or IRQ numbers for MSIs so far. |
| 11:39.36 | atommat | ok, gimme a few mins |
| 11:39.36 | Chainsaw | WIMPy: It could be behind a DMAR as well. |
| 11:39.41 | atommat | thanks forthe help btw |
| 11:39.56 | Chainsaw | WIMPy: So it would be useful to see the full listing. |
| 11:40.27 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:40.27 | Chainsaw | WIMPy: To keep your mind occupied while you wait... http://www.pastie.org/1455817 |
| 11:42.19 | stix | Hi guys. I have this asterisk which is behaving very strange. On inbound calls, callers are presented with a Playback of a file. This file plays fine most of the times, but sometimes there is no sound at all. I have tried calls via two providers, both SIP and IAX - when there is no sound, it occurs on both providers. The system has lots of free resources - what can be wrong here? |
| 11:42.29 | atommat | http://www.pastie.org/1455844 |
| 11:42.53 | Chainsaw | stix: Do you Answer() the channel before you start playback? |
| 11:43.06 | stix | Chainsaw, yes I do |
| 11:43.15 | atommat | http://www.pastie.org/1455847 |
| 11:44.11 | stix | this happens in periods of up to 5 minutes at the time |
| 11:44.34 | atommat | dial 02070173150 to hear the crackling |
| 11:44.35 | Chainsaw | WIMPy: Not on MSI. Level-triggered interrupts though. That's rather rare. |
| 11:44.37 | atommat | uuk |
| 11:44.38 | atommat | uk |
| 11:46.08 | WIMPy | atommat: Why do you load vzaphfc? And why do you even have it? |
| 11:47.42 | atommat | its out of the box |
| 11:47.49 | atommat | not sure what its doing |
| 11:47.55 | atommat | should it be removed? |
| 11:48.34 | WIMPy | What box was that? |
| 11:48.56 | atommat | no i mean the elastix install must have installed it |
| 11:49.15 | WIMPy | It *might* try to access the same hardware. It might be worth a try to get it out of the way. |
| 11:49.54 | atommat | you mean our hardware? |
| 11:50.13 | WIMPy | The b410p |
| 11:50.34 | atommat | do you need access to our server? fine by me |
| 11:51.15 | WIMPy | What versions of Asterisk and dahdi did you get with elastix? |
| 11:51.49 | WIMPy | If you want to use ISDN it's worth using recent versions. |
| 11:52.13 | atommat | latest version. downloaded and installed only a couple of days ago |
| 11:52.51 | WIMPy | What versions of _Asterisk_ and _dahdi_ did you get with elastix? |
| 11:52.53 | atommat | http://www.pastie.org/1455875 |
| 11:54.13 | atommat | ill paste up some more of the dahdi config so you can verify its correct |
| 11:55.03 | WIMPy | Should work, but with ISDN I'd go for Asterisk 1.8. |
| 11:55.58 | atommat | system.conf http://www.pastie.org/1455885 |
| 11:56.17 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:56.51 | atommat | chan_dahdi.conf http://www.pastie.org/1455892 |
| 11:57.58 | atommat | dahdi-channels.conf http://www.pastie.org/1455897 |
| 11:59.03 | WIMPy | All ports are connected to the PSTN? |
| 11:59.19 | atommat | yup, all 4 are green and reported as OK |
| 11:59.41 | WIMPy | It was more about what is connected. |
| 12:00.00 | atommat | yeh theyre all connected up |
| 12:00.01 | WIMPy | Ok, you should set them all as timing source in system.conf then. |
| 12:00.10 | WIMPy | 0 is appropriate for NT interfaces. |
| 12:00.11 | atommat | the BT equipment is also solid greeen light |
| 12:00.36 | atommat | ok so set all of timings to 1234 respectively? |
| 12:00.46 | WIMPy | yes |
| 12:00.52 | atommat | disco |
| 12:01.03 | atommat | ill give it a go, fingers crossed |
| 12:02.24 | atommat | other than this timing issue can you see any other issues with the config? |
| 12:02.35 | WIMPy | And just as an aside: You should never use the "default" context. |
| 12:02.41 | WIMPy | no |
| 12:03.35 | atommat | is this likely to affect the bri |
| 12:03.37 | atommat | ? |
| 12:03.42 | atommat | or the sound quality? |
| 12:04.13 | WIMPy | The timing could affect ports 2-4. |
| 12:05.10 | WIMPy | The context might give you some unexpected dialpaln behaviour but not voice quality issues, off course. |
| 12:06.09 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 12:06.43 | atommat | ok so sometimes when we dial in, we get a good quality sound. if we then dial in with another phone then yeh span 2 - 4 is bad. when span 1 becomes available, the sound returns to bad quality |
| 12:07.25 | atommat | is it worth reinstalling the sytem from scratch? |
| 12:08.05 | WIMPy | That could have been the timing thing then. |
| 12:08.16 | ironm | Good morning :) - is there an asterisk command like "traceroute" to check routing for a particular target phone number? - thank you in advance for any hinsts. |
| 12:09.36 | schmidts | ironm nope but thats a nice idea :D |
| 12:10.28 | kaldemar | ironm: if you use DUNDi, yes. |
| 12:10.36 | ironm | thank you schmidts :) - the last part of the log entry looks like: [Fwd: pers:69062 (EXCHANGE03) FAILED] |
| 12:11.03 | ironm | thank you kaldemar - I have to check what DUNDi is at all |
| 12:11.32 | ironm | well .. the last part of the log entry: "SIP/74519a0-0970c500","SIP/acc04trunk-09faf810","Hangup","","2011-01-11 08:50:08",,"2011-01-11 08:50:16",8,0,"FAILED","BILLING","CR=MCI IPAddr=" |
| 12:11.59 | *** join/#asterisk salimb (~chatzilla@81.5.139.17) |
| 12:12.30 | ironm | all other calls (other target numbers in AU) are routed correctly |
| 12:12.36 | schmidts | kaldemar would dundi tell you the context + extension local or only on another server? |
| 12:13.08 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 12:13.30 | atommat | so now with the timing as 1234 respectively, after a reboot we get a good sound on dialing in. if we dial in on another line simultaneously, we get a bad sound. |
| 12:14.11 | *** join/#asterisk guilhermebr (~Guilherme@189.50.117.126) |
| 12:14.30 | ironm | !dundi |
| 12:14.52 | atommat | could the card be at fault? |
| 12:15.01 | atommat | or dare i say it, the line? |
| 12:15.24 | kaldemar | schmidts: both if configured in a strange way. other than that, remote. |
| 12:15.58 | WIMPy | atommat: I'm running low on ideas. The sound is good on all ports now, but only with one call at a time? |
| 12:16.34 | atommat | no only the 1st dial in after a reboot is good. it then turns sour. no calls thereafter are good |
| 12:16.52 | atommat | what was the deal with the IRQ? |
| 12:17.08 | WIMPy | Oh |
| 12:17.25 | atommat | im not sure what this is but theyre all set to 217 |
| 12:17.57 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.24.173) |
| 12:18.00 | DelphiWorld | hey all |
| 12:18.16 | DelphiWorld | can someone please give me dahdi svn repo? |
| 12:18.41 | DelphiWorld | tzafrir_laptop: arround? |
| 12:18.54 | kaldemar | DelphiWorld: http://svn.digium.com/svn/dahdi/ |
| 12:19.14 | DelphiWorld | kaldemar: thanks a lot |
| 12:19.17 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 12:19.18 | tzafrir_laptop | DelphiWorld, dahdi-linux or dahdi-tools? |
| 12:19.19 | DelphiWorld | kaldemar: so svn co http://svn.digium.com/svn/dahdi/trunk? |
| 12:19.31 | DelphiWorld | tzafrir_laptop: preferably both |
| 12:19.35 | schmidts | c0rnota still here? |
| 12:19.36 | kaldemar | DelphiWorld: take a look at the insides of the directory. |
| 12:19.45 | c0rnoTa | schmidts: yeap |
| 12:19.56 | kaldemar | DelphiWorld: you'll find dahdi-linux, dahdi-tools etc. under it. |
| 12:19.57 | tzafrir_laptop | svn co http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux |
| 12:20.17 | tzafrir_laptop | svn co http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools |
| 12:20.20 | DelphiWorld | thanks a lot tzafrir_laptop |
| 12:20.57 | kaldemar | Chainsaw: i can't reproduce your dialplan issue. neither with 1.8.1 nor 1.8.1.1. |
| 12:21.03 | schmidts | c0rnota what i can see in code, if reading and parsing the config fails you get exactly the error you see, but the usecounter should be decreased at all. |
| 12:21.24 | Chainsaw | kaldemar: That's because you haven't actually subscribed to a TLIO hint. |
| 12:22.20 | WIMPy | atommat: Can you describe the bad audio? |
| 12:22.27 | Chainsaw | kaldemar: Otherwise you would see that either calling or the hint breaks. |
| 12:22.50 | c0rnoTa | schmidts: is any way exist there to see if config parsing fails? |
| 12:23.27 | atommat | can you imagine listening to an fm radio and then switching on a hairdryer. its crackled and distrorted. |
| 12:23.37 | WIMPy | c0rnoTa: Turn up verbose and debug and do a reload. |
| 12:23.43 | schmidts | c0rnota you should see warnings or notice if this happens |
| 12:23.48 | Chainsaw | kaldemar: You would add a buddy like *TLIO#45999 on a phone that is SIP/45999. |
| 12:23.49 | atommat | as if someone is rustling plastic sheets over it |
| 12:24.06 | Chainsaw | atommat: Yes, I've heard that noise. |
| 12:24.25 | Chainsaw | atommat: That's the noise of bad interrupt delivery. Either you have too many cards in one mainboard... |
| 12:24.34 | Chainsaw | atommat: Or your mainboard sucks. With the latter more likely. |
| 12:24.46 | WIMPy | atommat: Hmm, that would fit to a bad timing issue. |
| 12:24.49 | Chainsaw | atommat: Upgrade the BIOS to the newest available, failing that, replace the mainboard with something else. |
| 12:24.57 | atommat | we have another board we can try |
| 12:24.57 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:25.22 | Chainsaw | atommat: Go for that, don't spend any time on software until you get clear audio. |
| 12:25.23 | atommat | yeh, im convinced its timing |
| 12:25.32 | WIMPy | still wonders why it works for onw call. |
| 12:25.51 | WIMPy | atommat: Have you tried a 2nd call with the first still up? |
| 12:25.59 | atommat | yup, its bad |
| 12:26.10 | atommat | the CLI just reports spans going up and down |
| 12:26.20 | schmidts | wimpy memory leaks, or IRQ blocked or something like this |
| 12:26.21 | atommat | for no reason |
| 12:26.22 | WIMPy | And the first gets bad then as well? |
| 12:26.55 | atommat | ok chaps, i will do a board switch and see how we get on. thanks for the input |
| 12:27.05 | atommat | serisouly, really helpful, thanks |
| 12:27.15 | WIMPy | Well, spans going down when idle is normal. |
| 12:33.41 | Chainsaw | kaldemar: It's solved now thanks to jkroon. |
| 12:33.57 | Chainsaw | kaldemar: I was using equal context & subscribecontext. Separating hints out fixed things. |
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| 12:50.39 | WIMPy | atommat: Just to make sure: The Asterisk box is the only thin connected to those lines? |
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| 12:55.06 | *** part/#asterisk WonTu (~WonTu@p57B5411A.dip.t-dialin.net) |
| 12:58.16 | cynicismic | any ideas on where to start debugging an issue with incoming calls? - From the asterisk side I can hear the voice of the caller, but they can't hear me |
| 12:58.20 | cynicismic | outgoing calls work fine.. |
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| 13:48.50 | inetd50 | hi |
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| 13:53.20 | kaldemar | cynicismic: what technology are you using? SIP? |
| 13:53.21 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 13:54.16 | cynicismic | kaldemar: yeah - SIP with a 7960 phone |
| 13:54.45 | cynicismic | connected to an upstream sip provider |
| 13:58.27 | cynicismic | 1.6.2.5-0ubuntu1.1 |
| 13:58.33 | *** join/#asterisk atommat_ (~atommat@92.27.148.57) |
| 14:01.12 | kaldemar | cynicismic: is your asterisk behind a NAT? |
| 14:01.42 | cynicismic | kademar: yep, behind nat - was all working a couple of weeks ago and nothing has changed since then |
| 14:04.03 | *** join/#asterisk SeTTleR (~bernd@p5DDEF3B8.dip.t-dialin.net) |
| 14:04.19 | kaldemar | cynicismic: and you have nat=yes and externip or externhost set under [general]? |
| 14:04.30 | Katty | wibbles |
| 14:05.04 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:05.24 | Katty | today is not a good day |
| 14:07.58 | cynicismic | kaldemar: yeah - I've got both of those set fine |
| 14:08.12 | cynicismic | annoying thing is that the issue seems to be intermittent |
| 14:08.14 | *** join/#asterisk RickCogley (~anonymous@ntkngw479129.kngw.nt.ftth.ppp.infoweb.ne.jp) |
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| 14:15.07 | z4nD4R | hi all |
| 14:16.35 | Katty | hi |
| 14:17.02 | z4nD4R | i need in my extensions set somthing like default gw, so. if somebody call wich is not local a send it on special adres ( pstn ) |
| 14:18.41 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 14:19.11 | kaldemar | cynicismic: remove the other one if you have both. and try to get a sip debug of a call with unidirectional audio. |
| 14:22.38 | kaldemar | z4nD4R: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 14:24.36 | chuckf | Morning Katty. Sorry to hear that you're down this morning |
| 14:24.48 | E-bola | Wherent there some web service that could test the quality of your VOIP connection? |
| 14:25.29 | Katty | chuckf: thanks |
| 14:25.53 | Chainsaw | hugs Katty |
| 14:28.08 | chuckf | if I can do something to cheer you up Katty you let me know and I'll consider it:) |
| 14:28.28 | eject_ck | I have SIP peer registered, there is no active calls, when I'm trying to make call to that phone - I'm getting Got SIP response 480 "Temporarily not available" back |
| 14:30.16 | kaldemar | eject_ck: check if the phone has a "do not disturb" mode on. |
| 14:31.42 | cynicismic | kaldemar: removing/adding nat=yes doesn't make a difference, but dropping externip makes audio break completely |
| 14:32.06 | Katty | hugs Chainsaw |
| 14:32.10 | Katty | chuckf: just need time, i think |
| 14:32.18 | cynicismic | kaldemar: how do I get a sip debug - asterisk -rvvv ? |
| 14:32.27 | Chainsaw | cynicismic: sip set debug on |
| 14:32.57 | Chainsaw | cynicismic: Or instead of "on", you can specify something like "peer 1234" or "ip 1.2.3.4" |
| 14:33.57 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 14:34.49 | E-bola | http://pastebin.com/BWDnNSdT |
| 14:34.53 | omani | is it possible to send an email for incoming faxes, but to especially only the user who was called on his faxnumber? |
| 14:35.13 | E-bola | Do anybody know why i get this in my console everytime somebody calls in and get music on hold? |
| 14:36.00 | E-bola | there's no quality problem or anything |
| 14:36.06 | E-bola | its just anoying getting all these warnings |
| 14:36.34 | Chainsaw | E-bola: Because the hold music is being generated in ulaw and your channel is alaw-only. |
| 14:36.48 | omani | http://pastebin.com/5qw9PNJk here is my pastebin for my problem |
| 14:36.50 | Chainsaw | E-bola: It is pretty rare to find ulaw/alaw-only devices, so it is best to allow both codecs. |
| 14:37.29 | omani | or to my config |
| 14:38.49 | *** join/#asterisk wuwu191280 (~wolfgang@90.146.26.10) |
| 14:39.22 | wuwu191280 | hi all - i do have a problem with cli channel redirect |
| 14:40.07 | wuwu191280 | i am using the correct syntax - the destination does exists - the source channel does exist - but as soon as i want to redirect - it will hangup my channel |
| 14:40.18 | wuwu191280 | the channel is in state ringing - not in state up |
| 14:40.26 | *** join/#asterisk millerx (~millerx@rrcs-24-73-84-250.se.biz.rr.com) |
| 14:40.39 | wuwu191280 | asterisk is version 1.6.2.15 |
| 14:40.49 | wuwu191280 | i have taken a look at main/pbx.c |
| 14:41.08 | wuwu191280 | and have found the function ast_async_goto |
| 14:41.37 | wuwu191280 | there you can read: |
| 14:41.39 | wuwu191280 | <PROTECTED> |
| 14:41.39 | wuwu191280 | <PROTECTED> |
| 14:41.39 | wuwu191280 | <PROTECTED> |
| 14:41.41 | wuwu191280 | <PROTECTED> |
| 14:42.05 | wuwu191280 | so - if i understand it right - do redirect the channel to the new destination - and then hang it up.... |
| 14:42.14 | wuwu191280 | in 1.8 the same code is there |
| 14:42.20 | wuwu191280 | i don't understand it |
| 14:43.00 | pabelanger | ~pb |
| 14:43.00 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:43.07 | pabelanger | wuwu191280: ^^ for the next time |
| 14:44.28 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 14:44.29 | wuwu191280 | ok |
| 14:44.31 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:44.43 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:46.32 | cynicismic | kaldemar: http://pastebin.com/4AvjA1Q1 |
| 14:48.03 | cynicismic | not sure if the line 'transmitting (no NAT) to 88.208.249.40:5060' is a good thing. |
| 14:49.45 | E-bola | Chainsaw: The only explanation would then be if this provider limits me to alaw, but i think that woudl be a bit odd |
| 14:49.49 | wuwu191280 | i have found the answer to my question - in the next prerelease notes.... ;-) |
| 14:50.00 | wuwu191280 | http://www.asterisk.org/node/51537 |
| 14:50.14 | Chainsaw | E-bola: Something is limiting you, yes. Worth investigating whether it's your end or their end. |
| 14:50.20 | E-bola | ChanServ: Is "the music beign generated in ulaw" because thats my #1 codec of choice on my server |
| 14:50.40 | E-bola | cuz then maybe there would be an easy way to just "generate it in alaw" for this specific channel |
| 14:50.46 | ChrisH | I managed to connect a Nortel unistim IP Phone. The phone rings upon incoming call and I can talk to the caller. But I cannot call out or transfer a call. I do not see any indication in any logs why it fails. |
| 14:50.49 | omani | is it possible to send an email for incoming faxes, but to especially only the user who was called on his faxnumber? |
| 14:50.56 | omani | http://pastebin.com/5qw9PNJk here is my pastebin to my config |
| 14:51.04 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-wgagexuwquiqrurk) |
| 14:51.15 | E-bola | Chainsaw: i have to different providers one via iax and one via sip, its only the sip one that generates these warnings, when going through the same dialplan |
| 14:52.00 | Chainsaw | E-bola: Well, this hold music. Where is it coming from? |
| 14:52.30 | E-bola | ChanServ: its an mp3 file on my server |
| 14:53.03 | z4nD4R | register => USERID:PW@voiptalk.org/USERID if i have 3 gsm number's i muss 3x registered? for each number? |
| 14:53.20 | E-bola | but the anoying thing is everything is ulaw on my server, including allt he recorded voiceprompts |
| 14:53.40 | SaiSoma | Grrrr. Hey guys, my ATAs for fax stopped working for incoming faxes when I moved from 1.6.x (don't remember what version) to 1.8.0 (which was like 2-3 months ago) and I'm just now getting told about it. |
| 14:54.08 | SaiSoma | anyone tell me what I have missed in the readme's maybe? |
| 14:54.13 | Chainsaw | E-bola: Then I'd see if the other end can do ulaw. Or if you change provider to one that supports ulaw. |
| 14:54.33 | Chainsaw | E-bola: Because it does sound like you have a fairly ulaw-centric setup that will take quite some effort to change. |
| 14:55.02 | E-bola | Ya i should probably call them, im just wondering if maybe they are just forcing alaw, because they can see i support both. I wonder would would happen if i removed allow=alaw |
| 14:55.10 | E-bola | would=what |
| 14:55.17 | Chainsaw | E-bola: Only one way to find out... |
| 14:56.32 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:56.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:56.37 | z4nD4R | somebody to help with pstn... because .. i am confused... |
| 14:58.21 | kaldemar | cynicismic: it's all fine until "-- Native bridging SIP/hostcomm-00000017 and SIP/rclark-00000018". you need to disable re-invites. |
| 14:58.56 | kaldemar | cynicismic: defining localnet might also help. |
| 15:02.20 | Chainsaw | z4nD4R: "PSTN" can mean a lot. Please elaborate. |
| 15:03.28 | z4nD4R | Chainsaw: ok.. i have asterisk .. and i buy 3 number ( VOX number ), this 3 number are gsm numer |
| 15:03.34 | pabelanger | Can I access asterisk.conf variables from a dialplan? |
| 15:03.57 | cynicismic | kaldemar: localnet is defined ok |
| 15:04.01 | z4nD4R | and i want to connect my asterisk peers to the 3 gsm numbers |
| 15:04.08 | cynicismic | i'll take a look at re-invites |
| 15:04.11 | kaldemar | pabelanger: func AST_CONFIG |
| 15:05.25 | kaldemar | cynicismic: first it setups the call ok but then it tries to setup the media path between the endpoints, but it goes wrong because asterisk sends the private address of the phone in the SDP. |
| 15:06.29 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 15:09.14 | millerx | Would it be possible to add an overhead paging system on an spa8800 fxs port? I have a viking PI-1A interface, but when I page the fxs port with this interface, I just hear ringing on the loud speaker. |
| 15:09.28 | z4nD4R | Chainsaw: and i want to connect my asterisk peers to the 3 gsm numbers |
| 15:11.52 | pabelanger | kaldemar: Thanks, guess that is an option. Though, I'm not sure I want to open asterisk.conf each time a new call comes in. |
| 15:12.14 | pabelanger | I'm actually surprised that ${ASTETCDIR} don't exist by default |
| 15:12.26 | kaldemar | pabelanger: true. don't know a better way though. |
| 15:12.48 | pabelanger | kaldemar: Guess it's time for a patch :) |
| 15:17.46 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:18.11 | tzafrir_laptop | pabelanger, asterisk.conf is not guaranteed to be in ASTCONFDIR |
| 15:19.37 | pabelanger | tzafrir_laptop: Yup, another reason against AST_CONFIG |
| 15:19.40 | cynicismic | kaldemar: thanks for the pointers - will do some more digging around later on |
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| 15:26.13 | omani | how can I give user asterisk permission to send mail via mutt? |
| 15:26.21 | stix | I have this asterisk which is behaving very strange. On inbound calls, callers are presented with a Playback of a file. This file plays fine most of the times, but sometimes there is no sound at all. I have tried calls via two providers, both SIP and IAX - when there is no sound, it occurs on both providers. The system has lots of free resources - what can be wrong here? |
| 15:26.29 | omani | user asterisk is alrdy in group mail |
| 15:34.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:34.58 | pabelanger | kaldemar: patch written :) |
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| 15:51.31 | mechbangirc | hi how much cpu power do i need to get 2000 sip calls working without any transcoding (g729-g729) in asterisk? thanks |
| 15:54.12 | mechbangirc | i did a little research and found out that 1 call needs 10Mhz without transcoding on SIP? comments plz |
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| 15:59.21 | mechbangirc | hello |
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| 16:20.37 | tuxxie | is it possible to check an extension status to if the extension as an active call via the asterisk api? |
| 16:22.37 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
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| 16:27.08 | debayan | Is it legal to route incoming PSTN calls to Google Talk or Skype in India? |
| 16:27.27 | JustinCampbell | debayan: sure, why not? |
| 16:27.35 | JustinCampbell | once someone calls you, its up to you what to do with it |
| 16:27.56 | E-bola | debayan: You expect to find a legal expert on an IRC channel? |
| 16:28.14 | debayan | JustinCampbell: err, no. In india you cant then use voip to send the call to an indian number. |
| 16:28.23 | debayan | you can route it to a non indian number |
| 16:29.05 | JustinCampbell | im pretty sure in the US you can do whatever you want with your call |
| 16:29.11 | debayan | E-bola: well i dont but last time I was here I did find someone who gave me good legal advice. |
| 16:29.11 | JustinCampbell | Dell routes me to India all the time ;) |
| 16:29.23 | debayan | JustinCampbell: lol. thats right. |
| 16:29.34 | E-bola | debayan: I hope for your sake you dont base anything serious on advise from random people on irc :) |
| 16:29.36 | JustinCampbell | actually Cisco does the most |
| 16:29.49 | debayan | E-bola: hehe, ya well I do. |
| 16:30.12 | E-bola | debayan: stupid of you then :) |
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| 16:37.11 | debayan | E-bola, In India the governement is trying to replace Windows with Ubuntu in schools, but we hit a wall when it comes to support. If people could just call and touch base with some person, it will help things a lot. There are so many people who want to volunteer and help out but it needs a platform. |
| 16:37.40 | debayan | So what if people call this number, and they get redirected to a random volunteer expert |
| 16:37.53 | debayan | ofcourse this routing can be done intelligently. |
| 16:38.02 | *** join/#asterisk vAd0r (~vAd0r@unaffiliated/vad0r) |
| 16:38.18 | vAd0r | how do i reset my trunk to my carrier. I keep getting busy signal when calling |
| 16:38.18 | debayan | based on what registration data the volunteer expert has given to our system. |
| 16:39.26 | debayan | I am thinking about this particular domain, but potentially this can become a voice network of domain experts in any field. |
| 16:46.49 | *** join/#asterisk af_ (~getsmart@78.134.22.194) |
| 16:48.54 | raden | Katty, :D |
| 16:58.08 | *** join/#asterisk atommat (~atommat@pushbutton.plus.com) |
| 17:00.09 | atommat | ok, new board hasnt solved the issue. still getting crackled audio. anyone remember me from earlier? it may be that i need some support. is there an official support from digium or asterisk? |
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| 17:10.15 | tvc123 | atommat: I would start here http://www.digium.com/en/supportcenter/ |
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| 17:21.34 | Linuturk | is queue stats on asteriskguru the best stat program? |
| 17:25.43 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 17:27.20 | _Corey_ | Linuturk: Queuemetrics is the most common package we use |
| 17:27.46 | _Corey_ | Asternic is also decent, though not as robust |
| 17:27.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-013.mobile.uci.edu) |
| 17:28.54 | _Corey_ | Queuemetrics is comparatively expensive but very good. I'm not familiar with whatever asteriskguru has |
| 17:28.56 | Linuturk | Asternic does the fop too |
| 17:29.01 | _Corey_ | Yeah, same guys |
| 17:29.14 | Linuturk | I've only got 1 queue to pull stats on |
| 17:29.23 | Linuturk | so, queuemetrics might be overkill |
| 17:29.41 | _Corey_ | yeah, depends on the analytics you're looking for |
| 17:29.51 | _Corey_ | queuemetrics gives you a lot of per-agent stuff |
| 17:30.00 | _Corey_ | it's licensed based on agents |
| 17:30.22 | _Corey_ | hope that helps |
| 17:30.45 | Linuturk | it does |
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| 17:48.42 | vAd0r | how do i reset my trunk to my carrier. I keep getting busy signal when calling |
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| 17:49.24 | DelphiWorld | slaps neurosys around a bit with a large trout |
| 17:54.06 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.24.173) |
| 18:10.21 | SaiSoma | hey guys, anyone using ATA's with fax machines on 1.8? |
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| 18:14.19 | singha850 | #citrix |
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| 18:26.46 | z4nD4R | pls help: on asterisk i set register => user:pass@domain/user - but remote server resopnse with 401 Unauthorized, but if i set this on snom370 all is ok.. somebody? |
| 18:35.24 | Katty | hi raden |
| 18:35.56 | bn-7bc | z4nD4R: : gmm I had tthisproblem with a sopprovider i used once, For som reson the plocked everything wit uasrring asterisk, so i added : useragent = XAVi/EG101-1.6.7 in the general section of sip.conf, that did the trick |
| 18:38.53 | z4nD4R | i tryit thx |
| 18:38.53 | raden | hi Katty :) :) :) : ) |
| 18:38.55 | raden | hows life ? |
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| 18:47.11 | bn-7bc | z4nD4R: did it work? |
| 18:47.23 | z4nD4R | w8 |
| 18:52.41 | Katty | raden: not good. |
| 18:52.52 | Katty | raden: boyfriend and i went our seperate ways yesterday. |
| 18:53.19 | Katty | raden: sadly, it was not mutual, and he was not happy with my decision |
| 18:54.54 | z4nD4R | bn-7bc: 1.6.7 is what? |
| 18:58.27 | raden | Katty, I sorry to hear that :( |
| 18:58.45 | raden | Katty, did not know you had a boyfriend ( lucky dude ) :P |
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| 19:05.08 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
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| 19:08.03 | raden | Katty, what is the purpose of being with someone ? |
| 19:08.33 | z4nD4R | Q: i have 3 gsm numbers. What i muss set on asterisk site, if i want to connect with my 2 peers? |
| 19:10.06 | bn-7bc | z4nD4R: not shpre te software versjn i guess |
| 19:10.22 | z4nD4R | bn-7bc: ? |
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| 19:13.16 | bn-7bc | z4nD4R: sorry, broken keyboard. I guess the 1.6.7 is the software version |
| 19:14.08 | z4nD4R | sip show registry |
| 19:14.08 | z4nD4R | Host dnsmgr Username Refresh State Reg.Time |
| 19:14.08 | z4nD4R | xxxx.cz:5060 N 222361101 105 Registered Thu, 13 Jan 2011 19:22:35 |
| 19:14.16 | z4nD4R | that means i am registered? |
| 19:16.30 | *** join/#asterisk eject_ck (~eject_ck@83-218-247-23.dynamic.vega-ua.net) |
| 19:19.28 | bn-7bc | yes,so now try making a call to that did |
| 19:19.57 | bn-7bc | that is the did assignied to rhet account |
| 19:21.08 | z4nD4R | bn-7bc: but the same user muss i have on my asterisk? |
| 19:24.46 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-009.mobile.uci.edu) |
| 19:26.23 | bn-7bc | z4nD4R: hmm arer these 3 gsm numbers on the same account or om 3different (yhat so do you hace one or three usermame/pawword pairs for rhem? |
| 19:26.31 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 19:28.41 | z4nD4R | in order |
| 19:30.29 | z4nD4R | 1. i have 3 gsm numbers... each number have username/pass on remote server ( remote.something ) |
| 19:30.29 | z4nD4R | 2. i want that if somebody call one of this 3 nuimber, call peer on my asterisk |
| 19:30.49 | z4nD4R | remote.something ... is the pstn gateway... |
| 19:31.25 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 19:31.41 | z4nD4R | somebody? ^ |
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| 19:33.17 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 19:34.35 | tuxxie | I am looking for a way to see if and extension is on the phone using perl. Can that be done? |
| 19:34.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 19:34.55 | russellb | yup. |
| 19:35.08 | russellb | Use the AMI, use the GetVar action to read EXTENSION_STATE(<exten>) |
| 19:35.34 | tuxxie | cool thanks. |
| 19:36.33 | Katty | what does "black hole of drama" mean? |
| 19:39.01 | raden | ? |
| 19:39.18 | raden | Katty, never ending drama ? |
| 19:39.29 | raden | consuming drama |
| 19:39.38 | raden | drama queen |
| 19:39.52 | raden | just my interpretation matters the context I guess |
| 19:47.08 | ChrisH | I have a question on dialplan/extensions.conf; How to I rewrite the "target number" to get numbers without area-code, with area code, and international style into a common way to proceed. I would like to have +4912345678 and 012345678 be handled the same way without duplicating the lines |
| 19:49.13 | wdoekes2 | exten => 0Z!,1,Goto(+49${EXTEN:1},1) |
| 19:50.51 | russellb | tuxxie: or you might want to use DEVICE_STATE() instead. If you use EXTENSION_STATE(), you'll need to make sure your dialplan has a hint for that extension. |
| 19:51.03 | russellb | exten => 6098,hint,SIP/myphone |
| 19:52.49 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:53.29 | Katty | mmmk. |
| 19:54.05 | ChrisH | wdoekes2: something similar to exten => _ZXX.,?,Goto(+49123${EXTEN},1) should rewrite 45678 to +4912345678 given 123 is the local area code? |
| 19:54.41 | ChrisH | May with the help of ! it could be shorter |
| 19:55.05 | wdoekes2 | that would do |
| 19:55.10 | *** join/#asterisk Preytell (~jerry.win@68-188-27-90.static.stls.mo.charter.com) |
| 19:56.06 | Preytell | Is there a way to troubleshoot random dropped calls? I have a executive that complains that his calls are dropped several times a day, but I don't see abnormalities in the logs. |
| 19:56.09 | ChrisH | wdoekes2: Thanks. |
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| 20:02.15 | raden | Katty, ? |
| 20:04.33 | Katty | raden: -> |
| 20:04.41 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 20:07.01 | Preytell | hmm, looks like I need to look deeper into the Busydetect, busycount, and callprogress stuff. |
| 20:26.10 | ChrisH | exten => _ZXX!,1,Goto(+492173${EXTEN},1) |
| 20:26.25 | ChrisH | ups |
| 20:31.07 | z4nD4R | i need help... i set my trunk like this.. http://www.voiptalk.org/products/asterisk-sip-trunk.html .. but i have 3 users, not one... somebody??? ( for 1 peers finction wery well ) |
| 20:42.52 | z4nD4R | i need help... i set my trunk like this.. http://www.voiptalk.org/products/asterisk-sip-trunk.html .. but i have 3 users, not one... somebody??? ( for 1 peers finction wery well ) |
| 20:44.50 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:52.21 | ChannelZ | then you need to make additional peers with different names |
| 20:53.15 | z4nD4R | ChannelZ: yes... understand.. but how define it in extensions? |
| 20:53.36 | ChannelZ | Dial(SIP/whatever/5555555) |
| 20:53.43 | ChannelZ | Dial(SIP/whateverelse/5555555) |
| 20:54.31 | ChannelZ | no different than dialing anything else |
| 20:55.03 | z4nD4R | <PROTECTED> |
| 20:55.04 | z4nD4R | exten => _[*#0-9]!,1,Ringing |
| 20:55.04 | z4nD4R | exten => _[*#0-9]!,2,Dial(SIP/${EXTEN}@pstn) |
| 20:55.28 | z4nD4R | and one regisrtation record |
| 20:59.31 | z4nD4R | ChannelZ: i dont know if u know what i want |
| 20:59.55 | *** join/#asterisk Corydon76-home (gold@c-69-137-80-31.hsd1.tn.comcast.net) |
| 20:59.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:08.04 | z4nD4R | ChannelZ: i need to have 3 peer and each go out in difrent trunk |
| 21:17.30 | *** join/#asterisk cache_surplus (~unixrealm@71-83-188-135.dhcp.lnbh.ca.charter.com) |
| 21:18.24 | erinspice | :460 |
| 21:19.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:20.40 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 21:21.51 | russellb | erinspice: orly! |
| 21:22.06 | erinspice | Nah, doesn't work. :) |
| 21:22.44 | russellb | erinspice: boo |
| 21:22.51 | Tozz_ | Little question ;) I have a application in features.conf, that runs a macro on pressing #1. This works if I dial out, but not if I call my SIP phone frmo my mobile phone. What am I doing wrong? |
| 21:22.57 | russellb | erinspice: there are much worse things to accidentally type into IRC. |
| 21:23.18 | russellb | Tozz_: not setting the DYNAMIC_FEATURES variable in the right place |
| 21:23.39 | Tozz_ | russellb: well, pressing the #1 does work when I press it on my mobile phone (which is not really what I want ;)) |
| 21:24.01 | Tozz_ | so it seems the dynamic feature is set, but only works on the caller's device, not the callee |
| 21:24.23 | russellb | k, so you have it set on the wrong channel |
| 21:24.29 | Tozz_ | probably ;) |
| 21:24.31 | russellb | only set it on the channel you want to be able to execute the feature |
| 21:25.03 | Tozz_ | mm, how do I do that? I now set it before the Dial(SIP/phone) |
| 21:25.13 | Tozz_ | (for both in- and outgoing calls) |
| 21:25.22 | russellb | execute a macro on the called channel and set it there |
| 21:25.32 | russellb | there is an option to Dial() that lets you specify a macro to run |
| 21:25.36 | russellb | M() probably, I don't remember |
| 21:25.42 | Tozz_ | ah, thnx. I'll check it out |
| 21:26.25 | Tozz_ | so what do I do in my Macro? only set dynamic_features? |
| 21:27.58 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 21:29.03 | Tozz_ | ah great |
| 21:29.07 | Tozz_ | it works. thanks! |
| 21:32.27 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 21:37.33 | citywok | what headsets do people use for call centers? |
| 21:37.47 | citywok | plugging directly in to the RJ9 headset jack on the phone. |
| 21:38.42 | ChrisH | citywok: depends on the phone.... The Phones we have at the office know about headsets and so they are direct connected. |
| 21:38.51 | Tozz_ | we have Plantronics |
| 21:39.36 | ChrisH | At home I use an Avaya Box to connect the Headset (Plantronics or Avaya) to my ISDN Phone. |
| 21:39.43 | citywok | chrish we have aastra's, and they're supposed tow ork with headsets directly. |
| 21:40.01 | ChrisH | Both ways do work well. |
| 21:40.53 | Chainsaw | russellb: Is there a reason why the fix for #18185 is not in 1.8.1.1 please? |
| 21:41.07 | ChrisH | citywok: Given that I would try out before ordering a larger amount of headsets. Some like one-ear Headsets others two-ear headsets.... I prefer 2 ear ones |
| 21:41.27 | citywok | yea, been there done that. plan to go through a couple. which ones do you use now / do they work well? |
| 21:41.39 | russellb | Chainsaw: because it went in too late? |
| 21:42.20 | russellb | we tagged 1.8.2, hasn't been announced/published yet, should be soon |
| 21:42.34 | Chainsaw | russellb: Okay, cool. Rolling an 1.8.1.1-r2 in Gentoo for now. |
| 21:43.31 | ChrisH | citywok: At the office we have older Plantronics and the newer ones are Sennheiser. At home I have older Avaya and Plantronics.... and a horrible cheap PC Headset |
| 21:44.10 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 21:44.29 | russellb | 1.8.2 has already been tagged though ... |
| 21:44.46 | russellb | hates when distros package patched/non-standard releases |
| 21:44.48 | Chainsaw | russellb: Sure, but I need working blind transfers by tomorrow. |
| 21:44.48 | citywok | we've got a large number of older plantronics H connector headsets, but with the old vista amps we have it sounds bad. lol. |
| 21:45.03 | Chainsaw | russellb: I hate it when upstream puts out broken releases. |
| 21:45.13 | Chainsaw | russellb: We all have our cross to bear. |
| 21:45.22 | russellb | I hate it when users complain about the software they got for free |
| 21:45.33 | citywok | russellb: f***** asterisk pomfs, it's terrible. |
| 21:45.46 | citywok | russellb: no way am i going to help contribute to this pos open soruce project! |
| 21:46.06 | russellb | citywok: wrong button to push |
| 21:46.21 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:47.38 | ChrisH | citywok: Just checked I have such a Vista M21, branded Avaya and the sound is fine with the attached Headsets... I only needed to modify the cable from the M21 to the phone. |
| 21:48.02 | citywok | i've got 10 year old vista amps, lol. |
| 21:48.25 | citywok | a couple hundred of them from our old phone inter-tel digital phone system. |
| 21:48.45 | ChrisH | Mine is only 8 years old... similar source where mine is from :) |
| 21:53.06 | citywok | what cable did you ahe to modify & how? |
| 21:53.18 | citywok | s/ahe/have/ |
| 21:54.26 | ChrisH | citywok: From the Vista to the phone. I needed to find out the correct cabeling and crimp'ed the cable correct, |
| 21:54.41 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
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| 22:01.26 | *** join/#asterisk bittwist (~USER@unaffiliated/bittwist) |
| 22:01.35 | bittwist | oh dear |
| 22:03.19 | *** join/#asterisk share (share@unaffiliated/share) |
| 22:03.53 | share | :) |
| 22:03.55 | *** part/#asterisk share (share@unaffiliated/share) |
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| 23:03.07 | z4nD4R | [Jan 13 22:57:28] WARNING[16175]: chan_sip.c:13392 check_auth: username mismatch, have <trunk-01>, digest has <222361103> |
| 23:03.07 | z4nD4R | <PROTECTED> |
| 23:03.45 | pabelanger | z4nD4R: what version of Asterisk? |
| 23:04.51 | Tozz_ | z4nD4R: u probably need to set fromuser= |
| 23:04.53 | Tozz_ | in sip.conf |
| 23:06.01 | z4nD4R | pabelanger: 1.8.0 |
| 23:06.06 | *** join/#asterisk Inssomniak (~dave@199.19.173.132) |
| 23:06.45 | pabelanger | z4nD4R: Then I believe it is a bug. You could confirm by ~pb a SIP debug log |
| 23:06.54 | z4nD4R | Tozz_: i have fromuser=222361101 on [trunk-01] and fromuser=222361102 on [trunk-02] |
| 23:07.44 | Inssomniak | Is there something obviously odd about my setup? Co-located asterisk in a data center, 100meg link, IPsec tunnels from 4 branches to router via DSL, tied to asterisk server. Less than half of available bandwidth used on tunnels. |
| 23:08.51 | WIMPy | Inssomniak: Only voip traffic or other data as well? |
| 23:09.44 | Inssomniak | inter branch data but only about 200k tops |
| 23:10.57 | Tozz_ | z4nD4R: mm, sorry. I still run 1.4. But I had you rproblem with 1.6, which was solved with fromuser= |
| 23:11.03 | Tozz_ | I have no other ideas ;) |
| 23:11.49 | WIMPy | Inssomniak: Even with <50% of the available bandwidth in use you can get choppy sound it you don't have working TC/QOS. |
| 23:11.57 | *** join/#asterisk wwztw (~Pk@201.82.51.230) |
| 23:12.49 | wwztw | anyone familiar with v23 modem ? |
| 23:14.23 | WIMPy | wwztw: I've used one, >20 years ago. |
| 23:15.17 | wwztw | Im having a hard time to identify why a mg is ignoring the v23 signaling |
| 23:15.45 | wwztw | do you know were I can find the initial 1300hz with ans to a v23? |
| 23:15.57 | wwztw | *1300hrz tone |
| 23:16.06 | z4nD4R | pabelanger: her is mi log http://pastebin.com/EammHZD6 |
| 23:16.11 | WIMPy | find? |
| 23:16.30 | wwztw | yes I need an audio file |
| 23:16.36 | wwztw | with the tone |
| 23:17.39 | WIMPy | Start audacity and click Generate->Tone |
| 23:17.47 | wwztw | the mg is completely ignoring the v23 tone without the ans |
| 23:18.10 | wwztw | i need the ans, but I cant find the specification for the ans anywhere |
| 23:18.55 | pabelanger | z4nD4R: No, you look good. Here is the issue I was talking about: http://lists.digium.com/pipermail/asterisk-users/2010-November/255744.html |
| 23:19.10 | pabelanger | I confused your WARNING message |
| 23:21.35 | z4nD4R | solution? |
| 23:27.13 | *** join/#asterisk mahlon (mahlon@martini.nu) |
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| 23:57.38 | Beltechs | does asterisknow come with arecord and ast-playlinein ? If not how do I install? |