| 00:00.30 | jeffspeff | yes, that is set |
| 00:02.30 | Tim_Toady | jeffspeff: t[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[Cry this in ur dialplan, comment out the wait() and set the delay in answer() like this Answer(4000) thats 4 sec delay after answering |
| 00:02.38 | Tim_Toady | wtf |
| 00:02.45 | Tim_Toady | sorry for that |
| 00:03.01 | jeffspeff | lol, that's ok |
| 00:03.01 | Tim_Toady | jeffspeff: try this in ur dialplan, comment out the wait() and set the delay in answer() like this Answer(4000) thats 4 sec delay after answering |
| 00:03.05 | jeffspeff | i'll try that |
| 00:03.15 | jeffspeff | this is what i'm currently getting when the call comes in |
| 00:03.17 | jeffspeff | <PROTECTED> |
| 00:03.17 | jeffspeff | nnel 'SIP/teliax-00000009' status is 'UNKNOWN' |
| 00:06.03 | jeffspeff | Tim_Toady, i tried that, it still failed, "auto fallthrough, channel ...." but i didn't get a time out error this time |
| 00:06.06 | ChannelZ | Isn't the fax tone 1100Hz? |
| 00:06.14 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:06.36 | jeffspeff | im not sure... why? |
| 00:06.47 | Tim_Toady | jeffspeff: have u checked that the fax actually sends the cng tone? send the cal to a phone answer it and wait to hear the tone |
| 00:07.07 | Tim_Toady | yes its 1100 |
| 00:07.10 | jeffspeff | ok, be right back |
| 00:07.36 | ChannelZ | I"m actually making a little test and I can't get it to work either so far. I wonder if it's busted in 1.8.5 |
| 00:08.45 | Tim_Toady | i dont have 1.8 in production yet, 1.6.2 seems to detect fax fine |
| 00:10.37 | ChannelZ | Hmm nevermind, it just worked that time |
| 00:11.35 | jeffspeff | Tim_Toady, ChannelZ, I just attempted to send my office a fax, and it indeed makes loud beeping sounds at me |
| 00:12.25 | Tim_Toady | u got the call through sip? using that same asterisk box? |
| 00:12.35 | jeffspeff | no, analog box |
| 00:13.15 | jeffspeff | but if i uncomment the part in the incoming context for phone calls, it will actually dial and send the fax on through to my other desk phone |
| 00:13.48 | jeffspeff | the asterisk box is still in development, we are going to replace our old phone line system with it |
| 00:14.03 | jeffspeff | so i have 2 phones, one old phone lines, and one sip |
| 00:14.44 | jeffspeff | ChannelZ, you said you just got yours to detect a fax? are you using sip? |
| 00:15.55 | jeffspeff | would a sip debug help you help me figure this out? |
| 00:18.21 | jeffspeff | be right back, gotta fix somebodies computer problem... gotta love offices |
| 00:18.54 | ChannelZ | no, not SIP unfortunately. I can rig it up and try it from my home system in a minute |
| 00:25.40 | jeffspeff | ok, back... had to reset his domain password for the 2nd time today. |
| 00:25.52 | Tim_Toady | \m/ |
| 00:26.35 | jeffspeff | is there anywhere specific with the res_digium_fax or res_fax stuff where I have to explicitly tell it to detect the fax or tell it to use g.711? |
| 00:27.15 | Tim_Toady | the detection has nothing to do with the fax module, its done in the sip channel |
| 00:27.29 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 00:27.57 | Tim_Toady | is that line dedicated to fax only? if yes skip the detection and sent it straight to the fax etxtension |
| 00:28.54 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 00:28.55 | Tim_Toady | actualy do it anyway to see if fax is going to work, fax over voip over the internet is a hairy story |
| 00:29.38 | ChannelZ | hmm doesn't want to work via SIP for me. |
| 00:30.50 | jeffspeff | Tim_Toady, ok, i'll change my sip.conf to go straight into the fax-rx context |
| 00:32.15 | jeffspeff | how do i specify in context=fax-rx not to start with s,1 but to start with receive,1? |
| 00:33.50 | Tim_Toady | dont know, i would just add s,1,goto(blah) as a quick dirty fix |
| 00:33.52 | ChannelZ | depends on how the call gets there |
| 00:35.17 | ChannelZ | oh wait a minute, I put the faxdetect in the wrong place in my sip.conf |
| 00:35.43 | ChannelZ | ok it actually did work here |
| 00:36.00 | jeffspeff | ChannelZ, where? |
| 00:36.09 | ChannelZ | here on my systems |
| 00:36.52 | ChannelZ | I built a little test from my home box to here.. made an exten at home to PlayTones the fax tone. Had it dial via SIP my box here at work. |
| 00:37.11 | ChannelZ | which did detect properly and jump to the fax extension and ring my actual fax machine |
| 00:37.24 | jeffspeff | where did you specify faxdetect? in general or under your sip providers context? |
| 00:37.38 | ChannelZ | in general (I was just doing an 'anonymous' call) |
| 00:42.13 | jeffspeff | i'm having a weird dialplan issue here, give me a sec to test this |
| 00:42.54 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:50.22 | jeffspeff | ChannelZ, Tim_Toady, ok, so with the system setup to throw the call immediately into the fax context, i end up with this error |
| 00:50.24 | jeffspeff | Channel 'SIP/teliax-00000010' FAX session '2' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
| 00:53.16 | jeffspeff | ChannelZ, Tim_Toady, here's a pastebin of the call flow/output from console http://pastebin.com/Auav9Fke |
| 01:00.07 | Tim_Toady | i cant really remember what a T1 timeout is, i think it means it failed to start a fax session and get data |
| 01:00.12 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:02.01 | Tim_Toady | generally speaking, apart from your detection problems, fax over voip over the inetrnet is a recepie for disatser, it might work right like 1% of the time, you might have to rething your setup |
| 01:02.08 | Tim_Toady | at least try t38 for fax |
| 01:02.41 | jeffspeff | my provider only supports incoming t.38, not outgoing |
| 01:02.53 | Tim_Toady | but i would suggest to pass ur fax traffic over your land lines, analog or digital |
| 01:03.25 | jeffspeff | which, doesn't make any sense to me at all, but i'll google the heck out of this error and figure it out... at least I'm getting an error now, instead of it just telling me it failed |
| 01:04.08 | jeffspeff | Tim_Toady, ChannelZ, thanks for the help. I've been at this office for 12 hours now, i'm going home... gonna take a crack at this again tomorrow. |
| 01:04.27 | Tim_Toady | good luck |
| 01:09.12 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:11.15 | rjune | What tool will tell me if a port in a digium card is off hook? |
| 01:13.32 | Tim_Toady | dahdi show channel x |
| 01:13.54 | Tim_Toady | will show u something like: Hookstate: Offhook |
| 01:15.13 | rjune | in asterisk or in bash? |
| 01:15.43 | Tim_Toady | asterisk cli |
| 01:16.25 | Tim_Toady | or from ur shell running: asterisk -rx 'dahdi show channel x' |
| 01:16.27 | rjune | offhook is hung up, right? |
| 01:17.02 | Tim_Toady | onhook is hangup |
| 01:17.11 | rjune | so offhook is in use? |
| 01:17.34 | WIMPy | It tells you where your handset is. |
| 01:17.40 | Tim_Toady | yes, u can say so |
| 01:17.46 | WIMPy | On the hook or elsewhere. |
| 01:18.11 | WIMPy | And if you place a brick on th ehook, you fooled it. |
| 01:18.14 | rjune | WIMPy, thanks. |
| 01:18.15 | rjune | LOL |
| 01:18.49 | rjune | Hookstate (FXS only): Offhook <-- that thinks it's an FXS port that is in use, ya? |
| 01:19.11 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 01:23.08 | rjune | guys? |
| 01:23.12 | WIMPy | yes |
| 01:24.21 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:26.01 | rjune | yes it thinks it's an FXS port that's in use? |
| 01:27.49 | WIMPy | yes |
| 01:28.50 | rjune | FXS goes to device, FXO goes to phone network, right? |
| 01:29.13 | WIMPy | I tend to mix them up, but that sounds correct. |
| 01:29.37 | rjune | then I'll confirm with google |
| 01:30.13 | WIMPy | NT/TE is much easier. |
| 01:30.21 | rjune | NT / TE? |
| 01:30.35 | WIMPy | digital |
| 01:30.49 | rjune | is there a command to query the cards from the digium card? |
| 01:31.15 | rjune | I believe that card has all FXO ports. |
| 01:31.40 | WIMPy | No idea if dahdi_tool, dahdi_hardware or dahdi_scan can read that. I've never seen anything analog. |
| 01:32.08 | rjune | I'm trying to figure out why it goes off hook, but nothing happens |
| 01:32.42 | rjune | I can dial out, and watching it with dahdi_monitor, I see my voice going out, but i get no audio back. I don't know if I have a dial tone or anything |
| 01:33.03 | rjune | dahdi_scan, claims all ports are fxo |
| 01:33.26 | rjune | which confuses me because asterisk says they're fxs |
| 01:33.44 | rjune | and all in use. even though I know they are not. |
| 01:34.12 | WIMPy | Maybe because you're not reading carefully. FXO ports use FXS signalling and vice versa. But that definitely IS confusing. |
| 01:35.01 | rjune | oh jesus. |
| 01:35.20 | WIMPy | "Analog is evil" |
| 01:35.42 | rjune | bad labels are evil. |
| 01:36.08 | WIMPy | That as well. |
| 01:36.22 | rjune | ok, so I can see the audio, pretty sure it's trying to dial, Let me save the audio and replay it to confirm. |
| 01:42.32 | rjune | WIMPy, audio file has no dialtone, six pulse tones, a pause, then a seventh. I'm wondering if it's not dialing the right number |
| 01:42.40 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 01:42.56 | a1fa | is there an esy way to tell why am I getting fauke auth reject for a peer ? |
| 01:43.45 | WIMPy | rjune: IF an FXO port doesn't detect power, it should show an alarm, I guess. |
| 01:44.01 | WIMPy | a1fa: Start a softphone and try to register with random data? |
| 01:44.09 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:44.47 | a1fa | WIMPy: its rejecting my broadvoice.. same configuration works on another device; except this one is proxied |
| 01:44.57 | a1fa | proxy allows all connections |
| 01:47.15 | a1fa | i think it has to do something with the proxy |
| 01:48.20 | a1fa | and now my sip is timing out again |
| 01:48.33 | rjune | WIMPy, thank you for the help. |
| 01:49.47 | a1fa | of course, its bypassing outboundproxy again |
| 01:51.24 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 01:52.02 | p3nguin | I keep getting APPERROR when trying to receive fax with res_fax (Fax for Asterisk). What should I do to correct it? I don't know where to begin. |
| 01:56.45 | rjune | damn, dialing the right number. |
| 01:58.00 | rjune | Is there a way to put a line onhook when it's offhook? |
| 02:02.52 | rjune | asterisk has the channels as offhook, I have no idea why. the were all confirmed as getting dialtones, so the line is good, it's the card. |
| 02:08.46 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 02:14.39 | a1fa | i honestly think this is ignoring outboundproxy |
| 02:16.58 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:17.38 | maxagaz | is there a way to know the phone number associated to an FXO card port on a remote asterisk server ? |
| 02:18.59 | rjune | maxagaz, only have it call out |
| 02:19.23 | rjune | I don't know how to do it yet, but there does seem to be a way to have asterisk call out. |
| 02:23.45 | a1fa | **shrug** |
| 02:23.48 | a1fa | register => 8012567983@sip.broadvoice.com:Za!fFGfgZ1:8012567983@sip.broadvoice.com@192.168.1.1 |
| 02:23.55 | a1fa | is this how you reference outbound proxy in register? |
| 02:24.31 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:53.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:54.34 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 02:56.40 | ChannelZ | rjune: The hook status only applies to FXS (phones) |
| 02:57.24 | rjune | ChannelZ, I found this. |
| 02:57.30 | rjune | thank you |
| 02:57.53 | rjune | I'm still left with these ports appear to get no dialtone |
| 02:58.02 | rjune | and they don't actually call out. |
| 02:59.15 | ChannelZ | The lines are functional when connected to a standard phone? |
| 02:59.30 | rjune | yea |
| 02:59.36 | rjune | and inbound calls get routed properly |
| 03:00.09 | ChannelZ | You probably just need to put a delay in on your dial, some telcos don't provide the dialtone immediately and it'll start dialing before the remote end is even paying attention |
| 03:00.30 | ChannelZ | put a w or two in front of your number |
| 03:00.43 | ChannelZ | like Dial(DAHDI/1/w5551212) |
| 03:00.58 | rjune | ok, let me try |
| 03:01.08 | *** part/#asterisk GreyFoxx (~greg@mythtv/developer/GreyFoxx) |
| 03:02.14 | ChannelZ | you can verify it by using dahdi_monitor to record, do a dial, and then listen to the resulting file and hear what is going on down the line. Though even if you just dial through * you'd probably eventually get a click and a message from the telco about your number not being able to be completed as dialed, etc. |
| 03:04.11 | rjune | should I hear a dialtone? |
| 03:04.33 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:06.23 | ChannelZ | yes |
| 03:06.34 | *** join/#asterisk turtlefence (~turtlefen@2406:9800:1999:1:ad8c:971d:5705:157b) |
| 03:06.38 | ChannelZ | (if you're recording the monitor, that is) |
| 03:06.59 | ChannelZ | If not it's possible your signalling is wrong. 99% of the time it's kewlstart though. |
| 03:07.13 | rjune | FXS kewlstart is what I have |
| 03:07.17 | rjune | for signalling |
| 03:07.18 | turtlefence | hey anybody know if any software for listening .sln files on a mac? |
| 03:07.29 | turtlefence | *of |
| 03:08.55 | ChannelZ | rjune: dunno for sure for your country |
| 03:09.09 | rjune | USA |
| 03:09.27 | ChannelZ | oh. duh I was reading the server you're on :) |
| 03:09.39 | ChannelZ | What telco? |
| 03:10.00 | rjune | Uhm, dunno |
| 03:10.05 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 03:10.20 | ChannelZ | really? |
| 03:10.49 | rjune | really. |
| 03:11.10 | ChannelZ | Well you must pay for the things, or did you just find cables hanging down from the ceiling |
| 03:11.12 | rjune | I wasn't involved with the initial setup, and I'm trying to help somebody out |
| 03:11.25 | rjune | Neither. |
| 03:11.36 | rjune | is 1k miles away |
| 03:11.39 | ChannelZ | hmm ok. But you say calling in works? |
| 03:11.42 | rjune | yea |
| 03:11.56 | rjune | give a sec, I'll link the .wav recording |
| 03:11.58 | ChannelZ | and does it detect hangups OK? |
| 03:12.54 | rjune | www.bravegnuworld.com/dial.wav |
| 03:13.15 | maxagaz | how to make a callut from the console using a given port of an FXO card ? |
| 03:14.00 | rjune | <PROTECTED> |
| 03:14.13 | rjune | I believe it's detecting that correctly |
| 03:14.18 | ChannelZ | channel originate DAHDI/1 extension 444 |
| 03:14.39 | ChannelZ | would connect DAHDI/1 to extension 444 which presumably does something interesting |
| 03:14.44 | rjune | LOL |
| 03:14.56 | ChannelZ | there are others too, do 'help channel originate; |
| 03:15.00 | ChannelZ | s/;/'/ |
| 03:15.11 | rjune | there's no examples. which can be invaluable in learning. |
| 03:15.40 | ChannelZ | rjune: what is that wav supposed to be? |
| 03:16.00 | rjune | That's me dialing out on the DAHDI/1 |
| 03:16.04 | rjune | channel |
| 03:16.06 | ChannelZ | it's just silent |
| 03:17.04 | rjune | there's a series of DTMF tones, then me saying three two one and hanging up |
| 03:17.11 | rjune | Odd. I can here it |
| 03:17.12 | ChannelZ | oh. Stupid POS Quicktime in the browser is jacking it up |
| 03:17.16 | rjune | LOL |
| 03:18.05 | ChannelZ | hmm. It's like your telco isn't getting the signal |
| 03:18.26 | rjune | and yet, I can dial into the system without any problems |
| 03:18.32 | ChannelZ | are you sure you're dialing out the right channel? Are all lines connected? |
| 03:18.39 | rjune | yes. |
| 03:18.44 | rjune | that was verified today |
| 03:18.52 | rjune | the all lines connected with dialtones |
| 03:19.01 | turtlefence | whats the issue? |
| 03:19.18 | ChannelZ | Are these at a home or a business or.. |
| 03:19.25 | rjune | ports 1-6 were connected, port five had no dialtone. we swapped five and six, so now ports 1-5 are connected with a dialtone |
| 03:19.30 | rjune | is a business |
| 03:19.41 | rjune | turtlefence, outbound dialing isn't working |
| 03:19.45 | ChannelZ | It's possible they're using ground start I suppose |
| 03:19.55 | ChannelZ | But it's rare that I know of. |
| 03:20.32 | rjune | I'm suspecting a driver issue. was working with a zapata driver on fedora 6 |
| 03:21.06 | rjune | is now dahdi on centos 5.6 |
| 03:21.15 | rjune | everything else worked without a hitch. |
| 03:22.24 | ChannelZ | seems odd |
| 03:22.52 | rjune | turtlefence, asterisk says it's dialing the outside number, and even does. but the line has no dialtone, and the line ignores DTMF, as the number doesn't get called |
| 03:23.08 | rjune | ChannelZ, I agree. if were easy, I would have likely solved it already |
| 03:24.18 | turtlefence | does the line accept incoming calls? |
| 03:24.20 | ChannelZ | I meant more that it seems odd it would be a driver problem, I can't imagine what. |
| 03:24.23 | rjune | turtlefence, yum |
| 03:24.25 | rjune | yup |
| 03:24.39 | rjune | ChannelZ, everything worked until we changed software |
| 03:24.49 | rjune | drivers seem the most likely culprit to me. |
| 03:24.58 | rjune | either that or some esoteric config option |
| 03:25.21 | turtlefence | does the line function properly with a standard pstn phone? |
| 03:25.27 | ChannelZ | pb your dahdi/system.conf and asterisk/chan_dahdi.conf |
| 03:25.42 | rjune | turtlefence, yes, confirmed today. |
| 03:26.02 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 03:26.41 | rjune | http://pastebin.com/8KqJLZvL <-- system.conf |
| 03:27.10 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 03:27.53 | rjune | http://pastebin.com/JTgp6V2e <-- chan_dahdi.conf |
| 03:28.49 | rjune | they're both generated, by a program written by someone who knows more about it than me. |
| 03:29.34 | ChannelZ | uhm.. did chan_dahdi.conf get cut off? I don't actually see it consuming any of the channels. |
| 03:30.17 | rjune | http://pastebin.com/TUVS1bCf <--dahdi-channels.conf |
| 03:30.32 | rjune | no, it includes this file, which looks like that. |
| 03:30.49 | rjune | 6 entries, only difference is channel number |
| 03:30.57 | ChannelZ | which is only defining channel 1 then.. though your paste above seemed to indicate you were trying to use channel 3 |
| 03:31.00 | rjune | that what you were looking for? |
| 03:31.13 | ChannelZ | nevermind if they really are all defined |
| 03:31.17 | rjune | there are six entries there, only copied the first |
| 03:31.31 | rjune | yeah, I got lazy on that. |
| 03:31.50 | rjune | outbound is always trying to use channel one |
| 03:33.44 | ChannelZ | well the config seems to be OK near as I can tell. Puzzled |
| 03:34.10 | rjune | I'm going to try to replicate the problem here |
| 03:34.17 | rjune | but yeah, I'm stumped. |
| 03:34.42 | rjune | and tomorrow I'm going to ask them to swap the hard drive out to confirm the old one still works. |
| 03:35.05 | rjune | I can get at both systems, so any configuration option you think might be it, I can pull |
| 03:35.10 | ChannelZ | did you copy configs over from the old install or was this a complete rebuild? |
| 03:35.26 | dwayne | pastebin dahdi_cfg -vv, dialplan, and cli output of a call |
| 03:35.54 | rjune | ChannelZ, complete rebuild |
| 03:36.10 | rjune | Newer OS, Newer Asterisk |
| 03:36.24 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:36.47 | ChannelZ | Change your config to fxsgs/fxs_gs just for the hell of it |
| 03:36.51 | rjune | http://pastebin.com/TjfDCJeA |
| 03:37.12 | rjune | I'll try that, then finish getting the info |
| 03:37.37 | rjune | dwayne, dialplan is right, if I send it out a sip trunk, it's happy. |
| 03:38.12 | rjune | In fact, they're doing that now. they have to dial everything as long distance, I'm covering the calls. |
| 03:38.51 | ChannelZ | Did they have an old PBX prior to asterisk? |
| 03:39.15 | rjune | Yes, an older version of Asterisk with Zaptel drivers running on Fedora 6 |
| 03:39.19 | rjune | ChannelZ, Which config settings goes to fxsgs? |
| 03:39.33 | *** join/#asterisk MefhigosetH (~Mefhigose@186.129.193.86) |
| 03:39.53 | ChannelZ | dahdi/system.conf is fxsgs and fxs_gs is your chan_dahdi.conf or whatever it was |
| 03:40.00 | ChannelZ | the included one |
| 03:40.25 | rjune | fxsks=1 < that becomes fxsgs=1 ? |
| 03:40.32 | Deeewayne | no cli output ? |
| 03:40.46 | rjune | Deeewayne, I'm getting there. |
| 03:40.56 | ChannelZ | yes |
| 03:41.23 | ChannelZ | Deeewayne: he's not even getting a dialtone on the line as recorded via dahdi_monitor -f |
| 03:41.40 | rjune | ChannelZ, need to kick anything for the dahdi/system.conf change? |
| 03:41.57 | ChannelZ | yes |
| 03:41.58 | p3nguin | I keep getting APPERROR when trying to receive fax with res_fax (Fax for Asterisk). What should I do to correct it? I don't know where to begin. |
| 03:42.10 | ChannelZ | stop asterisk, stop/start DAHDI and then fire * back up |
| 03:42.29 | ChannelZ | It's a long-shot but I'm curious |
| 03:45.22 | rjune | ChannelZ, fxs_gs gets ring, but no answer |
| 03:45.51 | ChannelZ | what do you mean |
| 03:46.12 | dwayne | p3nguin, what res_fax tech module are you using ? |
| 03:46.44 | p3nguin | FAX For Asterisk Components: Applications: 1.4_1.3.0 Digium FAX Driver: 1.4_1.3.0 (optimized for i686_32) |
| 03:47.03 | rjune | ChannelZ, I have to call in to trigger the outbound dial. |
| 03:47.15 | rjune | I'm not local. |
| 03:47.22 | p3nguin | I hope that's the answer you were looking for. |
| 03:47.38 | dwayne | p3nguin, module show like fax |
| 03:47.52 | p3nguin | res_fax.so Generic FAX Applications 1 |
| 03:47.56 | p3nguin | res_fax_digium.so Digium G.711 FAX Technology (optimized f 0 |
| 03:47.59 | ChannelZ | oh.. ok not that then |
| 03:48.09 | rjune | ChannelZ, I have ssh access |
| 03:48.14 | rjune | so I can get at the cli |
| 03:49.39 | ChannelZ | So you are dialing in on chan 3 to try a dial out on chan 1? |
| 03:49.59 | rjune | pretty much yea |
| 03:52.37 | rjune | http://pastebin.com/1aUe33S5 out bound dial |
| 03:53.33 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
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| 03:56.25 | rjune | dwayne, theres the outbound dial, last bit is native bridging DAHDI/0 and DAHDI/1 |
| 03:57.09 | rjune | I'll confirm the old system still works, if so I'm guessing drivers. |
| 03:57.19 | rjune | or some other software. |
| 03:59.34 | dwayne | rjune, at this point you should be using a 3-5 line dialplan and start ruling large chunks of your solution out until you narrow it down to something you can focus on |
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| 05:31.26 | DND | hi guys i need help how can i create a call forward command? |
| 05:31.54 | DND | sorry not call forward, i mean call transfer |
| 05:32.42 | irroot | DND explain please ?? there a couple options |
| 05:33.07 | DND | ok i want to use ## to transfer a call to another extension |
| 05:33.27 | DND | we prefer a blind transfer |
| 05:33.36 | irroot | ah thats via dial features the tT options and /etc/asterisk/features.conf |
| 05:33.46 | irroot | blind or atten possible |
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| 05:34.00 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:34.01 | DND | but how can i assign ## to transfer? |
| 05:35.11 | *** join/#asterisk manji (~manjiki@ppp-2-84-9-60.home.otenet.gr) |
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| 05:43.22 | p3nguin | I'd much rather use the Transfer button on the phone to transfer a call. |
| 05:43.58 | p3nguin | But if you're going to do DTMF transfers, configure it in features.conf and use the t and/or T options in your Dial(). |
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| 05:45.58 | turtlefence | Hi peoples just wondering what you would use to listen to the native format asterisk file (.sln) |
| 05:46.08 | irroot | also # is a "special" button often a phone will interpret it ie the linksys uses this as "complete" |
| 05:46.25 | irroot | and default # is hangup the "hH" options |
| 05:47.08 | irroot | use something like sox to write it to a wav file and anything can play it |
| 05:47.24 | turtlefence | cheers |
| 05:47.36 | turtlefence | was hoping to be able to play it as a sly but meh |
| 05:47.39 | turtlefence | that works |
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| 05:51.56 | DND | or maybe *9 or something |
| 05:52.18 | irroot | DND i use *1 / *2 |
| 05:52.40 | DND | we are using x-lite at the moment and the onboard freepbx call transfer is not doing its job so i need to create a custom call transfer |
| 05:53.21 | irroot | http://pastebin.com/C7YYFJyz <- here is my features.conf snipet |
| 05:53.24 | DND | the reception has snom phone because she will do a lot of transfers but for us normal users we dont need much of call transfer |
| 05:53.46 | DND | hmm i see |
| 05:53.50 | DND | i will try that :D |
| 05:53.58 | DND | thanks irroot |
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| 06:06.43 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:06.44 | schmidts | good morning |
| 06:06.57 | irroot | morning ... |
| 06:07.56 | p3nguin | turtlefence: Did you try `play' already? |
| 06:08.08 | turtlefence | huh? |
| 06:08.12 | turtlefence | i sox'd em |
| 06:08.45 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:08.49 | p3nguin | Yeah? `play' comes from sox. play yourfile.sln |
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| 06:11.18 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 06:18.02 | ChannelZ | #@?! |
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| 06:19.34 | turtlefence | yeah I got the files from a customers remote asterisk box wanted to play on my machine |
| 06:19.43 | turtlefence | thanks for the new knowledge tho :) |
| 06:20.02 | p3nguin | I really think play will play it. |
| 06:23.17 | turtlefence | yeah I'm not working on a cents machine and i need to listen to them over and over again to make a crappy transcript |
| 06:23.30 | turtlefence | kinda blows |
| 06:23.59 | p3nguin | I have no clue what a cents machine or the number of times you need to listen to it has to do with anything. |
| 06:24.13 | turtlefence | dw |
| 06:24.30 | p3nguin | 'play' is part of sox. It'll play as many times as you want it to play. |
| 06:24.53 | p3nguin | while true; do play myfile; done |
| 06:25.13 | ChannelZ | I'm assuming if he's transcribing he needs control |
| 06:28.35 | turtlefence | yeah |
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| 06:29.31 | jmls | morning all |
| 06:31.07 | jmls | does anyone know how this works ? http://sailpbx.com/mediawiki/index.php/ZTP |
| 06:31.41 | jmls | I'm curious about the SIP Multicast listener as I want to be able to auto-provision SPA5xxx phones |
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| 06:58.06 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 06:58.14 | jacc0 | good morning all |
| 07:00.28 | *** join/#asterisk skyion (~bradley@41.162.7.154) |
| 07:01.33 | skyion | Hi There, I know SIP has progressinband, though does anyone know if IAX has an equivalent that will produce inband ringing? |
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| 07:03.33 | hurdman | hi |
| 07:03.39 | jacc0 | morning |
| 07:04.04 | jacc0 | @skyion: I guess it's exacly the same for iax; there isn't mutch difference between them |
| 07:05.29 | jacc0 | @skyion: I use asterisk in realtime and all my sip and iax accounts are in one and the same table |
| 07:05.48 | jacc0 | correct me if i'm wrong |
| 07:06.54 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:07.28 | skyion | will try that jacc0, is it's done in the general config, not per peer? or either.... |
| 07:07.37 | skyion | will try both... |
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| 07:13.09 | ChannelZ | Yay, my interwebs connection at work came back... just in time for me to have to go to bed. :| |
| 07:14.54 | irroot | hehe anyone want to come to south africa as a linux/asterisk techie have 2 positions |
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| 07:18.42 | skyion | irroot > some of us are already here :) |
| 07:18.54 | irroot | hehe |
| 07:19.07 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 07:19.11 | irroot | want a job :P got 1 in durbs and one in jozi |
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| 07:28.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 07:33.48 | *** join/#asterisk nfi|ermes (~sdff@host25-9-static.224-95-b.business.telecomitalia.it) |
| 07:36.33 | nfi|ermes | Asterisk 1.6.2.6 + dahdi 2.4.0 + libpri . dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! |
| 07:36.42 | *** join/#asterisk xpot (~xpot@66.219.225.48.provo.dynamic.broadweavenetworks.net) |
| 07:36.45 | nfi|ermes | please help debug |
| 07:37.12 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 07:37.47 | nfi|ermes | [Aug 4 09:37:09] WARNING[11135]: app_dial.c:1745 dial_exec_full: Unable to crea te channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 07:37.53 | jacc0 | irroot: 5for at least 65.000 euro a year I will come |
| 07:38.29 | jacc0 | :) |
| 07:41.22 | irroot | jacc0 mmm |
| 07:41.50 | jacc0 | I don't know what they pay in africa |
| 07:41.54 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:43.07 | jacc0 | I 15 years of experiance in telecom, software dev. and application security |
| 07:43.19 | jacc0 | *I have |
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| 07:45.11 | irroot | cost of living is much cheaper so its hard to say |
| 07:45.20 | jacc0 | true |
| 07:45.30 | jacc0 | 60.000 will do |
| 07:46.10 | jacc0 | I'll be making 50.000 right now |
| 07:46.24 | jacc0 | 10.000 a year to move to africa :P |
| 07:46.30 | irroot | lol |
| 07:51.45 | nfi|ermes | can you help me understand where is my problem please ? |
| 07:52.38 | irroot | nfi|ermes <- what is channel 34 |
| 07:53.47 | nfi|ermes | i thought 34 was a error code |
| 07:54.16 | ChannelZ | he never said anything about channel 34..? |
| 07:54.51 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 07:55.10 | ChannelZ | nfi|ermes: what version of Libpri |
| 07:55.43 | nfi|ermes | 1.4.11.5 |
| 07:56.42 | *** join/#asterisk gg0 (~gg0@unaffiliated/gg0) |
| 07:56.47 | ChannelZ | hmm |
| 07:59.14 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:59.16 | Polysics | hello |
| 07:59.26 | Polysics | i am using realtime sip peers on asterisk 1.8.5 |
| 07:59.38 | jacc0 | me to |
| 07:59.42 | Polysics | on mysql, and sometimes the new users added to the table do not work right away |
| 07:59.58 | Polysics | having to sip reload or restart * at every new user sort of defeats the point .-D |
| 07:59.59 | beardy | So you press Enter instead |
| 08:00.03 | beardy | to get it working? |
| 08:00.17 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 08:00.18 | Polysics | beardy, i suppose you are not talking to me :-D |
| 08:00.25 | beardy | is |
| 08:00.31 | ChannelZ | I feel like I walked in on a private conversation |
| 08:00.53 | jacc0 | I have to add some user; let me see if I get the same thing |
| 08:00.59 | Polysics | anyway, is there a setting that disables that kind of caching? |
| 08:02.34 | jacc0 | I have this in my sip.conf : http://pastebin.com/NL1aBP51 |
| 08:03.04 | ChannelZ | rtcachefriends or something? |
| 08:09.53 | *** join/#asterisk syntaxx (~patvan@unaffiliated/syntaxx) |
| 08:10.16 | jacc0 | i didn't need a reload |
| 08:11.29 | irroot | sip prune realtime / sip show peer .... load |
| 08:11.35 | irroot | those help lots |
| 08:14.36 | syntaxx | until now i cant seem to enable the online status on my sip client.. all users shows offline any idea? is this an asterisk configuration problem? |
| 08:16.57 | jacc0 | @Polysics: let me know what you find out |
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| 08:17.54 | Polysics | i will test it a little later |
| 08:18.05 | Polysics | seems like it is rtcachefriends, from the docs |
| 08:18.37 | jacc0 | yes, could be, it's in my sip.conf |
| 08:18.58 | jacc0 | but you also have: |
| 08:18.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:18.59 | jacc0 | 2.rtcachefriends=yes |
| 08:18.59 | jacc0 | 3.rtcache=yes |
| 08:18.59 | jacc0 | 4.rtcachepeers=yes |
| 08:19.33 | Polysics | i only have rtcachefriends in my sip.conf |
| 08:19.47 | jacc0 | let me know if that fixes your problem |
| 08:19.56 | irroot | working on a customer its so old it still has a 2.6.0 linux kernel module folder |
| 08:20.13 | jacc0 | this is what is set in my sip.conf: http://pastebin.com/NL1aBP51 |
| 08:20.19 | jacc0 | no need to relead after adding a user |
| 08:23.11 | Polysics | asterisk 1.8? |
| 08:24.29 | jacc0 | yes; 1.8.5 |
| 08:24.46 | jacc0 | with my app_originate patch |
| 08:34.35 | *** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
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| 08:38.42 | Morlac | guys |
| 08:39.12 | jacc0 | hello |
| 08:39.32 | Morlac | need a bit of an advise regarding the requirements for a site with 10 E1 connection and 3000 extensions in terms of CPU/RAM requirements.. Any ideas? |
| 08:39.50 | Morlac | anyone with such an experience? |
| 08:43.19 | schmidts | jacc0 i have some news for you |
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| 09:30.18 | tzanger | hmm, is there any way to execute a script (or do anything in the dialplan) when a SIP phone registers to my asterisk box? |
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| 09:41.54 | jacc0 | @tzanger: not that I know of, but it's an interesting feature |
| 09:45.49 | schmidts | tzanger i have read something but i didnt remeber exactly :( |
| 09:46.05 | schmidts | maybe search in the sip.conf.sample and also extension.conf.sample but i might be wrong |
| 09:46.53 | hurdman | hello, i try to have a very simple sip server on a public (debian) server. I have ever made that conf on my local network, without any router or firewall, but now , on my public server, sip don't work |
| 09:47.00 | Polysics | can i hang up a call with AMI if i know the UniqueID? |
| 09:47.05 | hurdman | any idea ? |
| 09:49.08 | jacc0 | open udp 5060 and 10000-20000 |
| 09:51.39 | tzanger | schmidts: checking sip.conf.sample isn't too enlightening; I've been running asterisk since 1.0. :-) |
| 09:51.57 | tzanger | schmidts: but like you,s omething is tweaking at my brain that says this is possible |
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| 09:53.40 | hurdman | jacc0: hi, i have allow all from my ip |
| 09:57.39 | hurdman | jacc0: i have flush and restart my firewall |
| 09:57.43 | hurdman | now it"s ok, thanks |
| 10:00.50 | schmidts | tzanger ;) |
| 10:03.16 | maxagaz | is there a way to know from the console which ports of my fxo card has a working cable plugged ? |
| 10:03.39 | maxagaz | s/has/have |
| 10:05.16 | schmidts | tzanger which asterisk version do you use? |
| 10:05.28 | schmidts | cause i just take a look at the code if i find something in there :D |
| 10:05.44 | tzanger | schmidts: nah, it's not that important; I can do the same |
| 10:05.58 | tzanger | was just wondering if I had to write an AMI "watcher" that allowed me to do that or not |
| 10:06.05 | schmidts | you know if you have something interesting and also want to find if its possible ;) |
| 10:06.20 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 10:06.23 | schmidts | tzanger as i have looked at handle_request_register it looks like you have too |
| 10:07.11 | tzanger | what I *would* like to find is a nice AMI library. The only one I have really found for C that is even somewhat usable is libami and it's ... wanting. |
| 10:07.33 | jacc0 | maybe if you use a realtime setup you can make a database trigger |
| 10:09.13 | maxagaz | in zaptel.conf, all ports of my fxo card is shown, how the pci port of each card ? |
| 10:09.33 | maxagaz | corresponding to what lspci gives me |
| 10:10.57 | tzanger | jacc0: yeah, but I'm not going ot use realtime. :-) |
| 10:11.15 | tzanger | maxagaz: I don't understand the question |
| 10:11.27 | tzanger | lspci won't show you anything other than which pci slot the card is in |
| 10:11.37 | schmidts | tzanger i know what was on my mind, the regcontext stuff |
| 10:12.42 | tzanger | schmidts: ahh yes |
| 10:12.47 | tzanger | I'd forgotten about that |
| 10:12.48 | schmidts | ; If regcontext is specified, Asterisk will dynamically create and destroy a |
| 10:12.49 | schmidts | ; NoOp priority 1 extension for a given peer who registers or unregisters with |
| 10:12.49 | schmidts | ; us and have a "regexten=" configuration item. |
| 10:12.56 | schmidts | but i guess this will not help you ;) |
| 10:13.02 | tzanger | schmidts: I wonder if I could abuse that |
| 10:13.06 | maxagaz | tzanger, exact, so what's the link between the slot number and the name given in zaptel.conf ? |
| 10:13.27 | tzanger | maxagaz: there is no real link; generally "earlier" cards will be enumerated first. that's about it. |
| 10:13.44 | maxagaz | tzafrir_laptop, okay |
| 10:13.52 | maxagaz | tzanger, okay |
| 10:14.08 | tzafrir_laptop | tzanger is back :-) |
| 10:14.11 | schmidts | tzanger ok with regcontext the function register_peer_exten is called but this really adds only a noop extension to this context |
| 10:14.18 | tzanger | tzafrir_laptop: let the confusion resume! |
| 10:14.54 | tzanger | schmidts: yes, I do realize. I was thinking to abuse it with an auto-running context or even a local/ type thing with a callfile to periodically scan |
| 10:15.15 | maxagaz | tzanger, and from asterisk, I have no way to know which port is usable ? |
| 10:15.39 | maxagaz | tzanger, (I mean, on which port a cable is plugged) |
| 10:16.07 | tzanger | maxagaz: I believe that for fxo you might get something useful out of chanisavail. I seem to remember that FXO ports went into red alarm if they were unconnected. |
| 10:16.34 | irroot | tzanger yes loss of voltage show red alarm |
| 10:16.54 | tzanger | I've been largely telco-free for quite some time now |
| 10:18.00 | maxagaz | tzanger, it seems I can't use this command in the console |
| 10:18.58 | tzanger | which command? |
| 10:19.11 | maxagaz | tzanger, chanisavail |
| 10:20.35 | maxagaz | tzanger, I'd like to get the list of available channels either in a bash or in the asterisk console |
| 10:22.55 | tzafrir_laptop | maxagaz, any chance you'll switch to dahdi? |
| 10:23.16 | *** join/#asterisk garymc (~chatzilla@host109-155-153-229.range109-155.btcentralplus.com) |
| 10:23.41 | maxagaz | tzafrir_laptop, I will, but not in the coming weeks |
| 10:23.46 | tzafrir_laptop | maxagaz, and in any case, what cards do you have? |
| 10:24.12 | maxagaz | tzafrir_laptop, Wildcard TDM800P |
| 10:24.37 | Polysics | anyone knows how the Hangup command from AMI is supposed to work? |
| 10:24.55 | Polysics | i tried hanging up one of the channels i DO have in a call, but it says "no such channel" |
| 10:25.02 | Polysics | how could i know what it is expecting? |
| 10:25.13 | Polysics | i tried hanging up SIP/10005 |
| 10:25.20 | tzafrir_laptop | in DAHDI those cards properly report RED alarm . So in Zaptel, but potentially buggy (added around 1.4.11, fixed in 1.4.12.1 or so) |
| 10:25.42 | tzafrir_laptop | If this works, you just put all the channels in a group, and dial to that group |
| 10:25.57 | maxagaz | tzafrir_laptop, I'm using Asterisk 1.4.25 |
| 10:26.03 | tzafrir_laptop | The ones that are RED never get dialed into |
| 10:26.22 | tzafrir_laptop | I was referring to versions of Zaptel |
| 10:27.13 | tzanger | maxagaz: it appears that you didn't compile chanisavail in then |
| 10:28.32 | maxagaz | tzafrir_laptop, tzanger, I got it |
| 10:28.58 | maxagaz | tzafrir_laptop, tzanger, the RED label is in zapata-channels.conf |
| 10:29.08 | maxagaz | tzafrir_laptop, tzanger, thanks a lot for your help |
| 10:29.45 | tzafrir_laptop | that label probably happens to be there because it was RED when that file was generated |
| 10:29.55 | tzafrir_laptop | It doesn't mean anything beyond that |
| 10:30.14 | maxagaz | tzafrir_laptop, indeed, it's not completely correct... |
| 10:30.16 | tzanger | maxagaz: in that case, you could cat /proc/zap/1 etc |
| 10:30.20 | tzanger | and the status will be there |
| 10:30.41 | tzanger | er /proc/zaptel/1, I am used to just typing /proc/zap<tab> :-) |
| 10:31.23 | maxagaz | tzanger, that's what I was looking for! :-) |
| 10:31.31 | maxagaz | tzanger, thank you so much |
| 10:35.11 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:36.25 | *** join/#asterisk gokulnath (~gokulnath@115.111.177.122) |
| 10:36.46 | garymc | having trouble recieveing calls to my polycom phone, now iam at home. i can make outgoing calls, but no one can call me |
| 10:36.54 | garymc | I have opended all relevant ports |
| 10:38.00 | garymc | i know there is a file I need to check, pretty sure i sorted it a while back |
| 10:42.12 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 10:44.09 | Polysics | can i speak into a channel with AMI? |
| 10:45.16 | Polysics | to, say, warn of credit that is ending |
| 10:45.53 | irroot | Polysics you can do that with Dial command the L option |
| 10:46.35 | Polysics | irroot, i am not using dial + L to hadle stuff, i built an external EventMachine timing server that checks the credit in real-time |
| 10:46.51 | irroot | ah ok |
| 10:47.12 | irroot | then you may need to look at a dialplan function that creates a frame hook ?? |
| 10:48.29 | Polysics | if i can't get it to work any other way, i will just play some DTMF tones |
| 10:50.35 | *** join/#asterisk BuenGenio (BuenGenio@119.237.220.25) |
| 10:56.53 | nfi|ermes | [Aug 4 09:37:09] WARNING[11135]: app_dial.c:1745 dial_exec_full: Unable to crea te channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 10:58.01 | nfi|ermes | Asterisk 1.6.2.6 + dahdi 2.4.0 + libpri |
| 11:01.10 | nfi|ermes | please help, i can't receive and make calls |
| 11:01.40 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 11:03.36 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:07.34 | Polysics | PlayDTMF isn't working either |
| 11:07.45 | Polysics | is htere a way to play ANY audio in a call using AMI? |
| 11:23.02 | *** join/#asterisk cookieliu (cookieliu@27.115.97.6) |
| 11:38.59 | Sylnai | is there any reason I get tons of warnings when registering a non-dynamic peer? |
| 11:40.14 | jacc0 | non-dynamic peer shouldn't register |
| 11:40.38 | Sylnai | why not? |
| 11:41.18 | jacc0 | the registration of a client is to tell where the client can be found |
| 11:41.34 | jacc0 | as it's not dynamic it's not needed |
| 11:43.07 | Sylnai | ah, fair enough, so registering wouldn't add any extra security anyway |
| 11:45.01 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 11:46.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 11:47.25 | jacc0 | no it doesn't |
| 11:48.31 | *** join/#asterisk garymc (~chatzilla@host109-155-153-229.range109-155.btcentralplus.com) |
| 11:54.13 | kaii | in fact, it adds little security |
| 11:54.55 | kaii | registration means authentication, which is not needed when peer is configured with static ip/port |
| 11:55.25 | kaii | if the ip/port gets hijacked, malicious peer can call without digest authentication |
| 11:56.03 | kaii | when registration times out, malicious dynamic peer needs credentials before allowed doing anything |
| 12:01.54 | irroot | been working on my call centre dialer script window allow creating a script in HTML outputing / saving the result in XML with feedback elements that corespond to dynamic table in SQL for storing results ... can also load a "CSV" file into the system it will dynamically create a contacts table with the info supplied displayed to the agent |
| 12:02.38 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:12.00 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 12:13.09 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:24.51 | *** join/#asterisk rjune (~rjune@oh-67-77-122-143.sta.embarqhsd.net) |
| 12:26.06 | *** join/#asterisk jglauche (~joaz@p5DC930E6.dip.t-dialin.net) |
| 12:26.20 | jglauche | wonders why this channel is +r |
| 12:26.31 | rjune | dwayne, ChannelZ : Found my problem. the Channel 1 FXO controller appears to be bad. When I cut channel one out of the dahdi config, eeverything dials like it's supposed to. |
| 12:31.24 | *** join/#asterisk Manu18 (~megairc@lau18-1-82-246-196-145.fbx.proxad.net) |
| 12:31.26 | PoWeRKiLL | Hi |
| 12:31.29 | Manu18 | bonjour |
| 12:31.30 | nfi|ermes | hi |
| 12:31.35 | PoWeRKiLL | bonjour manu |
| 12:32.58 | Manu18 | PoWeRKiLL ? |
| 12:36.52 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:37.26 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 12:39.33 | *** join/#asterisk mpe_ (~mpe@94.127.50.104) |
| 12:43.33 | Polysics | ok, no way to play audio via AMI in an existing call |
| 12:43.40 | Polysics | asterisk 1.8 |
| 12:43.53 | Polysics | i tried the few methods floating around and none work |
| 12:45.26 | *** join/#asterisk dhartman (~dhartman@wilug/newlug/ricko73) |
| 12:45.41 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 12:56.30 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 12:58.32 | volker- | bye |
| 12:58.35 | *** part/#asterisk volker- (~volker@84.201.4.133) |
| 12:58.35 | jacc0 | bye |
| 12:59.04 | jacc0 | how do I show all availabkle variables in dialplan again? |
| 13:00.53 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:00.58 | jacc0 | never mind |
| 13:01.05 | jwiggins | core show globals ? just guessing as I don't have a session open |
| 13:11.45 | engrxyz | hi, i received an INVITE with a SDP message something like " Media Attribute (a): fmtp: 18 annexb=no". I wish to reply it with "Media Attribute (a): fmtp:18 annexb=no"...any tips how how to do this? |
| 13:12.00 | *** join/#asterisk Cadey (~x@62.84.178.107) |
| 13:12.43 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 13:12.52 | Cadey | Hi guys, I have installed a fresh 1.8 today and for some reason asterisk wont start on boot, I can start it with "service asterisk start" and I have ran chkconfig asterisk on |
| 13:13.07 | Cadey | any ideas why this would happen? I dont recall having this issue with 1.6 |
| 13:13.24 | nfi|ermes | ERROR[4122]: chan_dahdi.c:12280 dahdi_pri_error: Unable to receive TEI from network in state 2! |
| 13:13.36 | nfi|ermes | i m going crazy |
| 13:13.58 | Jcook_5xData | anyone know of a directory GUI app for asterisk |
| 13:14.20 | Katty | gooooooooooooood morning you crazy kids! |
| 13:15.10 | Cadey | hums may be iits SElinux as ive left it on |
| 13:16.25 | irroot | Katty morning ... hey you luffs us bit crazy |
| 13:17.05 | Katty | 'course i does. |
| 13:17.09 | Katty | crazy folks are the best. |
| 13:17.12 | tzanger | morning katty |
| 13:17.17 | Sylnai | Cadey: have you tried launching it manually with asterisk -cvvvvvvvvvvvv? |
| 13:17.21 | Katty | hugs irroot |
| 13:17.22 | Manu18 | ya des francais pour un leger soucis de config |
| 13:17.24 | Katty | hugs tzanger |
| 13:17.28 | Katty | tzanger: good morning sweet pea! |
| 13:17.37 | tzanger | how's things/ |
| 13:18.18 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 13:19.20 | *** join/#asterisk frawd (~francois@79.Red-81-38-142.dynamicIP.rima-tde.net) |
| 13:19.36 | nfi|ermes | can anyone help mke with this problem ? |
| 13:19.46 | nfi|ermes | i think it's related to libpri version |
| 13:21.06 | irroot | katty you help make a nuclear reactor in my kitchen ?? |
| 13:21.14 | *** join/#asterisk BuenGenio (~BuenGenio@cm61-10-82-188.hkcable.com.hk) |
| 13:26.09 | *** join/#asterisk Hanumaan (~Hanumaan@dhcp208-34.rz.uni-freiburg.de) |
| 13:26.22 | nfi|ermes | no one ? |
| 13:27.04 | Manu18 | sd-23905*CLI> sip show channels |
| 13:27.05 | Manu18 | Peer User/ANR Call ID Format Hold Last Message Expiry Peer |
| 13:27.05 | Manu18 | 0 active SIP dialogs |
| 13:27.07 | irroot | nfi|ermes what version of libpri / asterisk / dahdi you on |
| 13:27.23 | nfi|ermes | i tried all libpri version |
| 13:27.30 | nfi|ermes | from 11.2 to 12 |
| 13:27.38 | irroot | currently using |
| 13:27.44 | irroot | and the config pb them |
| 13:27.47 | nfi|ermes | asterisk is 1.6.2.6 |
| 13:28.36 | nfi|ermes | DAHDI Version: 2.4.0 Echo Canceller: MG2 |
| 13:28.47 | nfi|ermes | ERROR[4122]: chan_dahdi.c:12280 dahdi_pri_error: Unable to receive TEI from network in state 2! |
| 13:30.05 | *** join/#asterisk garymc (~chatzilla@host109-155-153-229.range109-155.btcentralplus.com) |
| 13:30.33 | nfi|ermes | sip extensions are working good but i can't receive and make calls outside through isdn channel |
| 13:31.03 | irroot | what card ?? and config |
| 13:31.24 | Katty | tzanger: thing are good! generally speaking |
| 13:31.31 | nfi|ermes | pci:0000:02:02.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card |
| 13:31.34 | Katty | tzanger: i was getting dumb, but then discovered it was from a proper lack of sleep |
| 13:31.49 | Katty | tzanger: and after altering some rather poor sleeping habits, my brain has decided to join me again! |
| 13:31.57 | tzanger | Katty: lucky you. I haven't figured out my source of dumb yet |
| 13:32.01 | nfi|ermes | signalling = bri_cpe_ptmp |
| 13:32.20 | Katty | tzanger: :< |
| 13:32.32 | Katty | irroot: now why do you want a nuclear reactor in your kitchen dear |
| 13:32.51 | Katty | irroot: nuclear fushion is HOT and will melt your stainless steel silverware. |
| 13:33.00 | irroot | katty someone in SF got bored and was arested for trying to split atoms |
| 13:33.06 | Katty | irroot: the kitchen is hardly a palce to put an 11000F photosphere |
| 13:33.26 | Katty | really? |
| 13:33.32 | Katty | i didn't realize that atom splitting was a crime. |
| 13:33.57 | Katty | i bet that'd be fun on an interview |
| 13:34.05 | Katty | so your resume says that you were arrested...tell me what's going on... |
| 13:34.08 | irroot | Katty think it was the posession of radium and uranium that did him in |
| 13:34.13 | Katty | "i got locked up for nuclear fission" |
| 13:34.19 | nfi|ermes | when i try do do an incoming call i receive: [Aug 4 15:33:38] WARNING[3358]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 13:34.21 | irroot | hehe apparently it was a "hobby" |
| 13:34.35 | irroot | nfi|ermes output of lsdahdi |
| 13:35.24 | Katty | i wonder how someone gets uranium |
| 13:35.32 | nfi|ermes | http://pastebin.com/DErdGMdV |
| 13:35.44 | Katty | go to the store.. "i'd like a pound of uranium please" |
| 13:36.53 | irroot | nfi|ermes set verbose 3 and past a trace |
| 13:37.01 | Katty | oh, this is interesting. 27.3% of the uranium came from Kazakhstan |
| 13:37.19 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-176-204-91.red.bezeqint.net) |
| 13:37.28 | Katty | oh here we go. $59/lb |
| 13:37.49 | irroot | well there is a uranium mine not far from here but dont think they will sell me any :P |
| 13:37.58 | Katty | i would certainly hope not |
| 13:38.25 | Katty | it's a toxic metal! |
| 13:38.37 | Katty | and i'd like to see your kidney, brain, liver, and heart intact |
| 13:39.10 | tzanger | Katty: so is mercury but you would not believe how much of that stuff I played with as a kid |
| 13:39.12 | nfi|ermes | irroot, http://pastebin.com/AxS0E8pa |
| 13:39.14 | tzanger | ... with a hammer, no less |
| 13:39.20 | chuckf | Its jus the government trying to supress us again.... |
| 13:40.14 | tzanger | bah. I showed my wife the cupcake vaginas I found online. she just looked at them and said "gross" and walked away |
| 13:40.25 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 13:41.03 | irroot | nfi|ermes check the chan_dahdi conf and what channels are group 1 ?? |
| 13:42.29 | irroot | nfi|ermes some providers insist that you set the caller id correctly ?? |
| 13:43.26 | *** join/#asterisk coppice (~chatzilla@116.92.28.83) |
| 13:43.55 | Katty | :> |
| 13:43.59 | Katty | hugs coppice |
| 13:43.59 | *** join/#asterisk johnnyasterisk (~johnnyast@62.231.40.234) |
| 13:44.24 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:44.57 | nfi|ermes | in chan_dahdi.conf i have usecallerid=yes |
| 13:47.50 | jaytee | so where's this info on the guy in SF doing fission with uranium from Kazakhstan? |
| 13:49.52 | chuckf | has got to stop letting the wife drive... |
| 13:50.00 | chuckf | http://rumors.automobilemag.com/that%E2%80%99ll-buff-right-out-bentley-azure-causes-expensive-crash-in-monte-carlo-64521.html |
| 13:52.28 | nfi|ermes | irroot , now i m using libpri 1.4.12 |
| 13:52.43 | nfi|ermes | and dahdi 2.4.0 |
| 13:53.53 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:54.20 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:54.49 | irroot | http://www.escapistmagazine.com/forums/read/7.304580-Man-Arrested-For-Trying-to-Split-the-Atom-at-Home |
| 13:54.58 | raden | Naikrovek, morning bro |
| 13:55.44 | Naikrovek | sup |
| 13:56.03 | raden | another day man |
| 13:56.25 | raden | have to go run like 2400 ft of cat5 today from building to building , going to be fun |
| 13:57.12 | Naikrovek | weeeeee |
| 13:57.21 | raden | lol |
| 13:57.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:57.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.07 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:02.05 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 14:02.16 | Katty | hi Naikrovek |
| 14:02.17 | Katty | hi raden |
| 14:02.29 | Naikrovek | hiya katty |
| 14:02.45 | raden | Heya Katty :) :) :) |
| 14:02.51 | raden | hugs Katty |
| 14:04.03 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:04.35 | Katty | hugs raden |
| 14:05.28 | schmidts | Hi Katty |
| 14:05.36 | Katty | schmidts: hello. |
| 14:08.25 | nfi|ermes | irroot i tried with usecalleridno but it doesn t work the same |
| 14:09.17 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 14:09.47 | beek | waves to Katty |
| 14:12.03 | *** join/#asterisk DanFromUK (DanFromUK@2.26.253.177) |
| 14:12.55 | DanFromUK | Hi all. Quick question, can "pickupgroup" contain alphanumeric characters? or just numbers? |
| 14:13.50 | irroot | DanFromUK only some numbers |
| 14:14.28 | DanFromUK | thanks, is there any maximum to the length or size of the number? |
| 14:16.26 | *** join/#asterisk radic (~radic@tmo-096-180.customers.d1-online.com) |
| 14:17.27 | Katty | BEEKERS! |
| 14:17.29 | Katty | hugs beek |
| 14:19.42 | nfi|ermes | is dahdi 2.4.1 compatible con HFC cards ? |
| 14:21.10 | WIMPy | nfi|ermes: Partially. See http://voice.yeti.dk/Asterisk_vs_ISDN |
| 14:21.32 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 14:21.37 | raden | How is Katty today |
| 14:22.37 | WIMPy | irroot: What's you idea about access to your misdn fork? There was a discussion about that issue on th ML. Would be nice to tell them where they can get a working copy. |
| 14:25.00 | tzanger | what do y'all recommend for a C API to the manager interface? I'm using libami but it doesn't give me feelings of active development and future-proofisity |
| 14:25.09 | WIMPy | irroot: Do you have that on your list? |
| 14:25.25 | asterisk-Tester | hi |
| 14:25.30 | irroot | Wimpy hang on |
| 14:25.56 | irroot | should be open check |
| 14:26.08 | WIMPy | tries |
| 14:26.56 | WIMPy | Nope. Still asks for login. |
| 14:27.18 | irroot | mmm check apache conf :P |
| 14:28.15 | WIMPy | authorization failed: Could not authenticate to server: rejected Basic challenge. |
| 14:28.57 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 14:29.02 | Katty | raden: in a lovely mood :> |
| 14:29.17 | Katty | raden: working on a couple certs this morning. sbs 2008 and CCNT |
| 14:29.30 | raden | Sweet :) |
| 14:29.33 | raden | CCNT :) ? |
| 14:29.35 | tzanger | http://www.intuitivecreations.com/contributions/AMS/ami.php <-- that's what I've been using for an AMI access library for Asterisk |
| 14:29.44 | raden | if your in a loving mood want to go to dinner :P :) |
| 14:29.50 | tzanger | Cisco Certified Noogie Target |
| 14:29.52 | raden | lovely |
| 14:29.56 | raden | stupid auto correct |
| 14:30.16 | raden | Katty, how the CCNT going im working on my CCNA |
| 14:32.29 | irroot | WIMPy now |
| 14:32.53 | WIMPy | Nope. |
| 14:32.54 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 14:33.03 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:fc19:61e3:a256:5c42) |
| 14:33.15 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:33.22 | WIMPy | svn co http://pbx.distrotech.co.za/svn/mISDN/trunk/ |
| 14:33.29 | WIMPy | Or am I on the prong place? doing ' |
| 14:33.30 | WIMPy | ' |
| 14:33.34 | WIMPy | Err. |
| 14:33.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:34.21 | irroot | its ok just took time to restart apache |
| 14:34.37 | irroot | it should only be asking auth for https |
| 14:34.54 | irroot | ill fix it soon |
| 14:35.03 | WIMPy | Yes. Working now. |
| 14:35.07 | irroot | for now its not asking pw |
| 14:35.08 | WIMPy | Great. |
| 14:36.25 | *** join/#asterisk wonderworld (~ww@port-92-201-142-91.dynamic.qsc.de) |
| 14:38.34 | schmidts | irroot see pm message |
| 14:48.40 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 14:48.42 | *** join/#asterisk mtbf (~in9836@89-76-9-252.dynamic.chello.pl) |
| 14:54.24 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 14:59.00 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:01.57 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 15:03.00 | jacc0 | office hours are over!! Yeah!! see you all later. |
| 15:03.22 | *** join/#asterisk DanFromUK (DanFromUK@2.26.253.177) |
| 15:03.52 | gg0 | hi |
| 15:04.05 | WIMPy | lo |
| 15:04.15 | ruyo | http://pastebin.com/Spf5aThk - This is a rather problematic machine, considering it's only running asterisk, right? (check out cpu wait) |
| 15:04.34 | DanFromUK | hi all, I operate a hosted pbx server and have an issue with call pickup. The maximum seems to be 63. How do people deal with this to ensure one tenant can't pick up calls from another tenant? I thought I could just use the client's ID number, but they are always larger than 63. |
| 15:05.22 | ruyo | Can mISDN cause that kind of wait or is it probably something like bad HDD? |
| 15:06.01 | gg0 | here meetme-only pbx. just added a UK eutelia trunk but I got invalid conf number messages calling it from italy. DTMF issue |
| 15:06.22 | WIMPy | ruyo: Unless theres something going seriousely wrong, it must be something else. |
| 15:06.44 | gg0 | already tried all dtmfmode |
| 15:07.40 | ruyo | It's probably I/O then. I'll try replacing the HDD. |
| 15:08.03 | WIMPy | That doesn't seem to have much to do. |
| 15:08.12 | ruyo | I'm guessing this can cause Asterisk to behave badly, even on SIP calls. |
| 15:08.19 | ruyo | Hmm. |
| 15:09.02 | WIMPy | Or are you using an USB adapter? |
| 15:10.32 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 15:10.47 | ruyo | No, it's a B400P. |
| 15:11.47 | _abc_ | Ding. Hello. I would like to experiment with passing iax2 traffic over a ssh tunnel. This does not seem to have been done before. ssh does not provide forwarding of udp but there is this evil hack involving remote udp dns access due to c**p isp politics. Could it be this simple? http://www.qcnetwork.com/vince/doc/divers/udp_over_ssh_tunnel.html |
| 15:11.49 | WIMPy | iotop might help. |
| 15:12.43 | WIMPy | _abc_: openvpn? |
| 15:12.48 | _abc_ | no |
| 15:13.10 | _abc_ | just plain ssh into an asterisk server which is being set up/checked and a iax phone to it would be nice |
| 15:13.15 | _abc_ | I mean over the ssh |
| 15:13.48 | _abc_ | The question was, barring latency and other quality related issues, would there be a problem to do what that dns hack does for iax2? |
| 15:14.00 | _abc_ | I am willing to try this, but is there any info? |
| 15:14.18 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:14.34 | WIMPy | Surely easier than a ssh tunnel and 4 NCs converting tcp to UDP and viceversa at both ends. |
| 15:14.48 | ruyo | top shows asterisk using between 0 and 6%, sometimes mISDNstack jumps up but for a small amount of time, iotop shows pretty much only kjournald also only from time to time. |
| 15:15.09 | ruyo | (load average is around 0.6) |
| 15:15.17 | WIMPy | Well, with tcp you better not have any packet loss. |
| 15:16.36 | WIMPy | Oh, misdn1? |
| 15:16.44 | ruyo | Yes, mISDN1. |
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| 15:17.15 | ruyo | But I have several installations with the same hardware without any problem. |
| 15:18.13 | ruyo | htop shows a gray bar going up to 100% of the CPU sometimes. I believe that's the cpu wait. |
| 15:18.52 | Katty | raden: it's boring :< |
| 15:19.09 | Katty | raden: my brain keeps rebeling after about 30 or 45 minutes, and i have to go work on something else. |
| 15:19.32 | raden | Katty, LOL |
| 15:19.36 | WIMPy | If you have the same stuff working, it might be easier to try to find the difference. |
| 15:19.43 | raden | I have resorted to audio CD's for my CCNA |
| 15:20.00 | Katty | i don't seem to learn well from audio cds |
| 15:20.05 | raden | I do better implementing to learn |
| 15:20.10 | Katty | i read material, and make flash cards at the same time |
| 15:20.14 | Katty | yeah, me too |
| 15:20.16 | raden | Katty, I only do when I'm in the car |
| 15:20.20 | Katty | i also do tutorial write ups |
| 15:20.29 | raden | Katty, thats a good idea I will have to try that |
| 15:20.46 | raden | audio out of car i get like 10% of it |
| 15:20.51 | raden | in the car i get like 60% of it |
| 15:20.54 | raden | kinda nuts |
| 15:21.01 | Katty | that is nuts |
| 15:21.02 | tzanger | raden: my bro completed his CCIE/CCNE (I can never remember) a couple years back. he is going for his top-level Juniper cert now |
| 15:21.07 | raden | I leave it playing in background when im at home as well |
| 15:21.16 | tzanger | I remember him playing around with virtualized cisco routers |
| 15:21.19 | raden | tzanger, Nice :) |
| 15:21.34 | raden | I just want the cert to say hey I know stuff |
| 15:21.36 | raden | LOL |
| 15:21.55 | tzanger | Katty: I can't learn stuff like that from audio cds. I need a person to help me learn properly. I am a horrible book-learner unless I already have a good base knowledge |
| 15:22.00 | raden | Katty, the way audio helps me the best is the stuff I do catch I didnt really know |
| 15:22.11 | raden | so when I read then I have a better understanding of it etc... |
| 15:22.13 | ruyo | WIMPy, hai, the problem is that I don't find any, apart from the dialplan and things alike. That's why I was wondering about hardware. |
| 15:22.16 | tzanger | Katty: I listen to TED talks when driving long distances. it's a perfect mesh. I can't listen to them without driving though, heh |
| 15:22.21 | Katty | tzanger: it probably helps that i'm ocd about stuff |
| 15:22.26 | raden | and I write down stuff I dont understand and look it up etc... |
| 15:22.34 | raden | kind of a chaotic way of learning but seems to work |
| 15:22.36 | Katty | tzanger: my documentation is PRISTINE |
| 15:22.45 | Katty | tzanger: every little note and tidbit has to be documented. |
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| 15:23.00 | Katty | tzanger: otherwise the writeup is incomplete, and irritates me |
| 15:23.49 | tzanger | nice, you should scan a few pages to show me, I always enjoy seeing how others organise their thoughts and see if I can adopt their techniques for myself |
| 15:24.15 | Katty | tzanger: http://42ndgeekstreet.blogspot.com/ |
| 15:24.21 | tzanger | personally when I am going somewhere with the fam I am anal about writing todo lists and crossing shit off. my wife laughs at me and gets frustrated when I won't budge until some item is crossed off but it's saved our asses a few times |
| 15:24.37 | Katty | tzanger: http://42ndgeekstreet.blogspot.com/2011/01/setting-up-ssh-keys-between-two-servers.html <- there's a basic one |
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| 15:25.01 | tzanger | Katty: interesting, you use a Q&A kind of format |
| 15:25.18 | raden | Nice Katty :) |
| 15:25.19 | Katty | tzanger: those are flash cards |
| 15:25.31 | Katty | tzanger: every single...worthwile fact...is documented |
| 15:25.36 | Katty | tzanger: which then turns into a flash card |
| 15:25.42 | Katty | tzanger: which is is then memorized |
| 15:25.52 | tzanger | interesting. you're even more anal about that kind of stuff than me |
| 15:25.57 | Katty | there is a ubuntu 11.04 writeup there too |
| 15:26.03 | WIMPy | ruyo: Do yu have any unneccessary stuff loaded? |
| 15:26.04 | raden | Nice |
| 15:26.07 | tzanger | 11.04 was a disappointment for me |
| 15:26.12 | raden | will be spending some time on there this weekend :) |
| 15:26.18 | Katty | i rather liked it |
| 15:26.22 | raden | Im moving to ubuntu I think |
| 15:26.27 | raden | just seems to work |
| 15:26.28 | tzanger | 10.10 was pretty good, I'm unhappy that I upgraded. lots of regressions on notebooks |
| 15:26.33 | Katty | having used debian almost exclusively, it was kinda of nice to see the ubuntu software package manager thingy |
| 15:26.34 | raden | opensuse becoming a PITA |
| 15:26.50 | raden | Later peeps |
| 15:26.56 | Katty | but yes, i'm very anal about documentation |
| 15:26.57 | raden | gotta go run cabling |
| 15:26.58 | tzanger | yeah, I don't touch RPM as a matter of principle |
| 15:27.11 | ChannelZ | raden: have fun |
| 15:27.25 | Katty | if i forget everything i've learned, i want a reference to go back and look at, all inclusive, so i can do it exactly as it should be done |
| 15:27.48 | Katty | if i'm ever in a car wreck and lose my mind, i can still do my job! theoretically. |
| 15:28.04 | tzanger | Katty: exactly. My documentation is more "stream of thought" or "stream of consciousness" type writing, and if they're electronic the checklists are in the form of [ ] some item and then I use / for partially complete and X for complete |
| 15:28.09 | Katty | most of my writeups are in text documents...haven't tansfered much to the blog yet |
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| 15:28.32 | Katty | hey as long as it works for you, that's what counts |
| 15:28.40 | tzanger | I tend to use pencil and paper for my checklists though, they get a little... interestingly organized as they're written |
| 15:29.00 | Katty | i use my blackberry. |
| 15:29.04 | Katty | there are currently.. *checks* |
| 15:29.20 | Katty | 167 lists |
| 15:29.21 | ruyo | WIMPy, unlikely. Pretty much a base Debian installation with compiled asterisk, asterisk-addons, DAHDI, mISDN and apt-getted mysql, php... |
| 15:29.39 | ruyo | Ah, and iaxmodem and hylafax. |
| 15:29.43 | Katty | i would be lost without my phone |
| 15:29.51 | Katty | probably have withdrawl too |
| 15:30.17 | WIMPy | ruyoL I was more thinking of hardware support. |
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| 15:31.52 | tzanger | I have been playing with rememberthemilk and evernote... I don't like both for different reasons but they aren't bad. |
| 15:32.13 | Katty | nods |
| 15:32.22 | Katty | haven't tried either of those. i just use the basic note thingy |
| 15:32.29 | Katty | blackberry + outlook integration with BES |
| 15:32.52 | Katty | grocery list, account login information, various recipes |
| 15:33.06 | *** join/#asterisk bmg505 (~leon@196-209-99-219.dynamic.isadsl.co.za) |
| 15:33.15 | Katty | gear lists for camping, vacation, and astronomy night |
| 15:33.34 | Katty | ie ven have a document with lists of christmas gifts |
| 15:33.51 | Katty | i write down ideas all through the year, then review the list ;) |
| 15:34.09 | Katty | currently the boyfriend is getting a kayak and a boba fett hoodie from thinkgeek |
| 15:34.35 | Katty | horay for lists tho! i'd be lost without them |
| 15:34.46 | Katty | i don't understand how people organize their thoughts without them )= |
| 15:35.12 | WIMPy | Katty: In almost no time. |
| 15:35.52 | Katty | that'd be a good read. |
| 15:36.15 | Katty | the organization of thoughts of type a and type b people. |
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| 15:37.00 | ruyo | WIMPy, I can only think of the ethernet module. I'll have a look at that. |
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| 15:53.16 | Katty | i want something bad for me, for lunch |
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| 16:01.50 | rjune | ChannelZ, you catch my message last night? |
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| 16:15.02 | irroot | katty im having pizza without the base .... |
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| 16:20.05 | PoWeRKiLL | hi coppice |
| 16:20.09 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 16:20.10 | PoWeRKiLL | you there ? |
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| 16:20.45 | _abc_ | Guys is there some older version of asterisk which sent bad audio packets (type 0) in iax channels? I seem to have this problem with one. |
| 16:20.56 | _abc_ | I mean, is this a known problem? |
| 16:20.58 | p3nguin | What version? |
| 16:21.39 | _abc_ | 1.6.1.x |
| 16:21.55 | p3nguin | Now I know the branch... so what version are you using? |
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| 16:28.21 | ChannelZ | rjune: no sorry |
| 16:28.58 | rjune | Port 1 on the digium card was out. |
| 16:29.10 | ChannelZ | out? |
| 16:29.12 | rjune | I disabled that this morning, and everything started working fine. |
| 16:29.21 | rjune | out, dead, kaput. an ex port. |
| 16:29.27 | ChannelZ | oh. hmm |
| 16:29.31 | rjune | No longer pining for the fjords |
| 16:29.56 | rjune | which is cool, because they have a dead line too that apparently nobody knew about. |
| 16:30.04 | ChannelZ | wonder if it's a bad cable/connection or the port is dead? |
| 16:30.27 | rjune | Cable is possible. |
| 16:31.03 | _abc_ | okay that old thing (1.6.x) gives me this IaxWrapper::event_text() Message: Type=1 Message=Bad or incomplete voice packet. Unable to decode. dropping in kiax and also iaxcomm connects and no sound. Any idea if iax was flawed in 1.6.1.x ? |
| 16:31.18 | ChannelZ | have 'em peek in the hole and make sure none of the pins have gotten mushed together. Glad you figured it out though |
| 16:31.31 | rjune | ChannelZ, thanks again for the help. do you have an email address I can send a beer to? |
| 16:31.34 | ChannelZ | Told you I didn't think it was drivers but you never know :) |
| 16:31.48 | ChannelZ | Heh I'm good but thanks |
| 16:31.58 | rjune | Everything was working with the old system. I never suspected hardware. |
| 16:33.07 | ChannelZ | It certainly makes it more difficult when it happens to fail fail at the same time as another event |
| 16:33.14 | rjune | No kidding. |
| 16:33.15 | ChannelZ | s/fail fail/fail/ |
| 16:34.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:34.41 | ChannelZ | I've certainly chased my own tail more than once with those kind of things. It's a diversion! |
| 16:35.02 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 16:35.10 | rjune | I need to find out how much their phone bill cost me for a day |
| 16:36.01 | ChannelZ | How many 1-900-HOT-SEXY numbers did they call? :D |
| 16:38.40 | nfi|ermes | here's my pri debug : http://pastebin.com/Xhk3Sa1t |
| 16:41.11 | WIMPy | nfi|ermes: You are not receiving anything. So there must be somethign wrong at a very low level. |
| 16:41.34 | _abc_ | http://translate.google.com/translate?hl=en&ie=UTF8&prev=_t&sl=auto&tl=en&twu=1&u=http://www.mail-archive.com/pld-users-pl%40lists.pld-linux.org/msg02435.html has anyone seen this? I have the same error message, minus the codec error. |
| 16:42.22 | ChannelZ | Pickup() doesn't give a damn about callgroups and pickupgroups does it? That's only for the features.conf pickup right? |
| 16:46.00 | irroot | WIMPy got the httpd.conf sorted open for non https and require pw only for https |
| 16:49.24 | WIMPy | Does that mean I should try if it still works? :) |
| 16:50.36 | WIMPy | Oh, yes. Now I get a 'forbidden'. That was better earlier. |
| 16:52.01 | nfi|ermes | WIMPy, but if i type: pri show span 1 i receive : http://pastebin.com/B1Uifcqv |
| 16:52.27 | rjune | ChannelZ, thankfully none. |
| 16:52.42 | rjune | I had limited it to about $30, so that wouldn't have lasted long. |
| 16:53.00 | WIMPy | Have you tried the other zaphfc version? Maybe it's a bug/incompatibility. |
| 16:53.57 | nfi|ermes | the zaphfc i use is the one included in dahdi 2.4.0 |
| 16:54.03 | nfi|ermes | which should i try ? |
| 16:54.05 | WIMPy | There have been others. not receiving anything. But I don't remember if we found out what was the cause. |
| 16:55.20 | WIMPy | There is no zaphfc in dahdi. |
| 16:56.31 | nfi|ermes | dahdi-linux-complete-2.4.0+2.4.0/ |
| 16:57.10 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:57.21 | nfi|ermes | ls /usr/src/ASTERISK/dahdi-linux-complete-2.4.0+2.4.0/drivers/dahdi/zaphfc/ |
| 16:58.45 | nfi|ermes | version is 1.42 |
| 16:59.16 | WIMPy | That does not exist in the official dahdi. |
| 17:00.38 | WIMPy | You can find links to two zaphfc versions at the end of http://voice.yeti.dk/Asterisk_vs_ISDN |
| 17:00.48 | WIMPy | Where did you get that dahdi version? |
| 17:01.38 | nfi|ermes | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ |
| 17:04.50 | WIMPy | If I download there, I can't find any zaphfc in there. |
| 17:05.40 | nfi|ermes | in 2.4.1 is not there but in 2.4.0 there is |
| 17:06.41 | WIMPy | No. There is no zaphfc in 2.4.0+2.4.0, either. |
| 17:07.05 | nfi|ermes | you are right |
| 17:07.18 | nfi|ermes | so i don't know where it comes from |
| 17:07.29 | nfi|ermes | i built it a long time ago |
| 17:08.00 | nfi|ermes | anyway , svn co http://zaphfc.googlecode.com/svn/branches/2.4 zaphfc-2.4 doesn t work |
| 17:08.36 | WIMPy | Then try the other. |
| 17:09.32 | nfi|ermes | ok git works |
| 17:10.03 | WIMPy | Where did the googlecode version fail? |
| 17:10.27 | WIMPy | Just out of interest. |
| 17:12.15 | WIMPy | is off for a beer. CUL. |
| 17:12.31 | nfi|ermes | no |
| 17:12.33 | nfi|ermes | wimpy |
| 17:12.36 | nfi|ermes | too cool |
| 17:12.38 | nfi|ermes | it worked |
| 17:13.31 | nfi|ermes | thx so much |
| 17:15.14 | nfi|ermes | can i offer you the beer ? |
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| 17:17.22 | ChannelZ | What do it do? |
| 17:22.17 | Katty | sprawls |
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| 17:45.47 | nfi|ermes | i upgraded asterisk form 1.6 to 1.8 and now pickup doesn t work. Anything changed for this ? |
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| 18:34.38 | leifmadsen | nfi|ermes: what version of 1.8? |
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| 18:41.35 | nny | when using TrySystem, the syntax is TrySystem(command (${passed_variable})) or TrySystem(command ${passed_variable})? |
| 18:45.05 | philippel_mac | question on attended transfer and channel variables |
| 18:45.34 | philippel_mac | I noticed that if you look at the channel after an attended transfer you will see any channel variables that were created by the original call AND the same named variables |
| 18:45.50 | philippel_mac | that were created by the initiation of the attended transfer duplicated in the channel |
| 18:45.57 | philippel_mac | when you do a core show channel |
| 18:46.13 | philippel_mac | from the dialplan though (such as with a dynamic application) |
| 18:46.35 | philippel_mac | it will retrieve only the value set by the caller who initiated the transfer |
| 18:46.55 | philippel_mac | is there any way that it can get to the "phantom/other" variable from the original call? |
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| 19:04.35 | *** join/#asterisk thecardsmith (~doug@pool-72-92-150-82.burl.east.myfairpoint.net) |
| 19:05.34 | thecardsmith | little telephony question out of curiosity... how does the PLMN or PSTN "know" what phone numbers can receive a text message? |
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| 19:09.52 | IAXAlexG | any chance there is someone here that is familiar with the (byte) to use to represent amr in a iax full frame voice packet? In all the captures i have seen ... they use g711 (0x04); g711a(0x08) or someother codec (h223) (0x00)... cant find amr though... |
| 19:10.44 | IAXAlexG | the pdf i have has the media format as 4 bytes long... 00000004 is 711u and 00000800 is amr.. but i cant represetn that amr number in a single byte |
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| 19:20.35 | IAXAlexG | no takers? |
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| 19:23.55 | DanFromUK | hi |
| 19:24.14 | DanFromUK | is there a technical reason for the 63 pickupgroup limit? |
| 19:27.29 | Janos | hi, i have a problem that started recently on my asterisk, checking the logs i'm seeing things like "chan_sip.c: Peer 'xxxx' is now Lagged. (2016ms / 2000ms)" which i think could be the cause, now the question is, is there any other reason for a phone to lag other than network timings ? i have not done extensive network testing yet, but initial tests show acceptable ping times > 5ms, the problem i have is some calls get established but do not have audio on a |
| 19:27.29 | Janos | ny direction |
| 19:27.57 | WIMPy | thecardsmith: That depends on the network. The PLMN know the termonal capabilities. On the PSTN you need to register to a database. |
| 19:28.58 | Katty | i want to know why this basic telecommunications book, which is part of the CCNT certifiation, is primarily analog technologies |
| 19:29.19 | Katty | the copyright is 2003 |
| 19:29.34 | Katty | there's should be 1 chapter dedicated to analog |
| 19:29.39 | Katty | and it should say OBSOLETE. |
| 19:30.07 | Katty | then there should be 1 chapter on digital |
| 19:30.11 | Katty | and it should say MOSTLY obsolete! |
| 19:30.24 | Katty | let's get on with some real technology! |
| 19:30.41 | IAXAlexG | so is there any chance someone knows where to find a capture of an iax call with AMR? |
| 19:30.42 | Katty | but noooo, let's read about technology from the 1980s. *snore* |
| 19:30.50 | _Corey_ | Katty: Those who don't learn from history are doomed to repeat it, right? :) |
| 19:31.05 | Katty | i don't think we're oging to Repeat Analog |
| 19:31.38 | _Corey_ | lol, probably not |
| 19:31.42 | Katty | now if i was planning to invade russia |
| 19:31.47 | Katty | you may have a valid point |
| 19:32.07 | Katty | instead i would loan them tons of money |
| 19:32.11 | WIMPy | Katty: Now you know how I feel since I (try to) use Asterisk. |
| 19:32.18 | Katty | and then when they can't pay me back, require them to sell me all of their public business |
| 19:32.27 | *** join/#asterisk gxdssoft (~gxdssoft@190.81.6.83) |
| 19:32.35 | Katty | much easier than running through the snow. |
| 19:33.00 | Katty | WIMPy: i don't really understand the Value of knowing exactly how Analog works |
| 19:33.03 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-176-204-91.red.bezeqint.net) |
| 19:33.16 | Katty | WIMPy: right down to how it packs 4 calls, on different frequency, onto the same line |
| 19:33.45 | Katty | WIMPy: that's not my world :< |
| 19:34.11 | WIMPy | Analog is evil. |
| 19:35.05 | Katty | yes. |
| 19:35.10 | Katty | it seems very crude. |
| 19:35.19 | Katty | maybe that's why we have to learn about it, so that we can appreciate how awesome voip is |
| 19:35.42 | WIMPy | VOIP is mostly shit as well. |
| 19:36.03 | Katty | fiber is certainly much better than copper |
| 19:36.13 | WIMPy | The days when telephone systems used to work are over. |
| 19:36.36 | _Corey_ | lol |
| 19:36.43 | Katty | what do you mean Work |
| 19:36.59 | WIMPy | Do what you expect them to do. |
| 19:37.17 | Katty | i'm a bit confused. |
| 19:37.21 | Katty | mine does what i expect it to do |
| 19:37.30 | Katty | and if it doesn't, it's usually because my logic is off |
| 19:37.34 | Katty | or i've gone and typod something |
| 19:38.44 | WIMPy | For normal telephony, Asterisk works a lot worse that the old plastic PBX. But that doesn't matter. The telcos stuff works a lot less than it did 15 years ago, as well. |
| 19:38.54 | WIMPy | (or rather: as bad) |
| 19:39.02 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 19:39.08 | Katty | i wasn't in telecom 15 years ago. |
| 19:39.34 | Katty | and in fact, the only experience i have with older technology are those pesky tdm cards |
| 19:39.38 | Katty | and echo issues |
| 19:39.49 | WIMPy | Then you will have to visit a museum to see how things can actually work. |
| 19:40.15 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 19:41.22 | Katty | that would be much more fun than reading the CCNT book |
| 19:42.41 | _Corey_ | I do a 4-hour "Telecom History" session several times a year for new hires... it's absolutely amazing to me how some (smart) engineers are so clueless how the PSTN worked (and works) |
| 19:43.51 | WIMPy | I came across an online museum about two weeks ago. Klicked my way for two evenings. Pretty mind boggling to see how things used to work and how tittle time has gone since then. |
| 19:43.56 | mpe | hi do any of you know where I can find "tables.sql" for sqlite |
| 19:43.56 | mpe | in res_config_sqlite they mention that there exist a file with all the tables need |
| 19:43.56 | mpe | 00055 * To use res_config_sqlite for static and/or RealTime configuration, refer to the |
| 19:43.57 | mpe | 00056 * Asterisk documentation. The file tables.sql can be used to create the |
| 19:43.57 | mpe | 00057 * needed tables. |
| 19:43.57 | mpe | I have tried to use this structure, but i is not compatible with sqlite |
| 19:43.57 | mpe | https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure |
| 19:44.18 | _Corey_ | WIMPy: Yeah, look at some of the step switching videos on Youtube, pretty cool content out there |
| 19:45.11 | Katty | WIMPy: link? |
| 19:45.18 | WIMPy | The sheer amount of work required to get things going was huuuuge. All that soldering and so on. |
| 19:46.18 | WIMPy | http://bayern-online.com/v2261/kategorie.cfm/203/Bilder.html |
| 19:46.27 | WIMPy | It's german. |
| 19:46.45 | _Corey_ | This one is good: http://techchannel.att.com/index.cfm?SearchTag=Archives |
| 19:47.07 | singler | mpe: try searching asterisk source for .sql files |
| 19:47.22 | singler | there may be in docs/contrib or something like that |
| 19:48.02 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-176-204-91.red.bezeqint.net) |
| 19:48.10 | nny | um 10.10.212.37    101ef5324fb 00:00:03 0000000104 0000000000 ( 0.00%) 018756 0000000112 0000274790 (245348.21%) 000000 |
| 19:48.32 | nny | why would sip show channelstats give bad results? Is it reliable normally or not at all? |
| 19:49.15 | nny | or is that just saying the JITTER is really freakin bad |
| 19:49.37 | Janos | analog is the real thing, just a bit too complex for most folks (including myself) and so voip came to be, the fact that voip oriented pbx suck at handling it does not mean it's evil :P |
| 19:50.05 | nny | So sayeth Janos. Ohhmmmmmmmmmmmmmmmmmmmm |
| 19:51.08 | WIMPy | No, Digital is good, but realtime over IP is as we know it and SIP for telephony is really bad. |
| 19:52.25 | leifmadsen | How can you say, "really bad" when it works for many many people? |
| 19:52.51 | WIMPy | It does? |
| 19:52.56 | mpe | thanks singler I will try |
| 19:53.09 | WIMPy | VOIP != telephony |
| 19:53.39 | WIMPy | Sure you can use SIP to set up VOIP (with all the known issues). But Normal telephony offers a lot more than that. |
| 19:54.23 | WIMPy | And with SIP completely unusable in that area it comes to all those mant extensions that may be supported or not. Pretty random. |
| 19:54.42 | WIMPy | s/mant/many/ |
| 19:54.49 | leifmadsen | WIMPy: you're going to have to elaborate because I use VoIP (with SIP signalling) in many places |
| 19:55.03 | leifmadsen | you must mean a specific area, and not in general |
| 19:56.09 | _Corey_ | Well, from my perspective SIP isn't the end-all be-all of VoIP. It still has a ways to go... |
| 19:56.33 | WIMPy | Both basic signalling and features. |
| 19:57.11 | WIMPy | There are no reliable standards. |
| 19:57.36 | anthm | digital is no walk in the park, debugging ISDN dialects or timing errors over TDM is almost as frustrating as SIP is but SIP still takes the cake in the annoying category with more and more work-around specs added to it constantly, that time has mostly passed for ISDN but it's had its day in the sun to torture ppl too. |
| 19:57.41 | WIMPy | Or even different ones. |
| 19:57.52 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.12 | WIMPy | anthm: Yes but the days of different protocols have been over for about 15 years, unless you're sitting in North America, where VOIP may seem like a way out of that. |
| 19:59.33 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 19:59.50 | _Corey_ | anthm: I'm inclined to agree... "The grass is always greener" as they say. I can (vividly) recall some carrier timing train-wrecks that put anything I've ever had to fight w/SIP on to shame... |
| 20:00.47 | WIMPy | Yes, I've seen them messing their stuff up quite badly as well, but for the user it just works. |
| 20:01.04 | jaytee | yeah |
| 20:01.40 | anthm | I think in the end you learn that Telephony in general, like all things used by everyone to the point that they are not allowed to move forward without clinging to legacy, is maddening and most of use who work with it are batty. |
| 20:02.41 | anthm | err most of us |
| 20:02.47 | coppice | the legacy stuff worked better, so don't knock it too much |
| 20:03.01 | jaytee | I tried setting SIP session timers because I had two calls over a month period where the call lasted 6 hours and 1 sec (Flowroute maximum) so I set up the session timers two days ago and started getting dropped calls after about 20 minutes duration. Now I'm trying to figure out what settings I can set to allow a call to go on for a maximum of 2 hours without getting terminated. |
| 20:04.16 | anthm | produces a copy of "everything I need to know about Telephony was developed in the early 70's" |
| 20:04.35 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 20:04.48 | _Corey_ | coppice: I blame the incompetents and lazy goofballs everyone seems to be hiring now more than the technology :) |
| 20:05.10 | anthm | coppice, i mean legacy in expectations, like preserving exact behaviors etc |
| 20:06.12 | anthm | like needing a familiar dial tone and dtmf sounds on a sip phone or cars with pretend needles for speedometer |
| 20:06.13 | WIMPy | Well, I certainly expect to not get less features. |
| 20:06.52 | anthm | or a particular series of buttons to push an exact certain way that is never permitted to change |
| 20:08.27 | anthm | jaytee, session timers was designed for that but they tend to get messed up by nat and backfiring on you |
| 20:09.19 | WIMPy | jaytee: Set TIMEOUT(absolute) |
| 20:09.51 | WIMPy | Or Dial(,,L) |
| 20:10.40 | coppice | anthm: you want funky dial tones? like everyone now seems to have a funky ring tone? I long for the good old days |
| 20:13.05 | anthm | coppice, I didn't say it was impractical to expect the legacy, just that we are lunatics for being the ones to provide it to them ;) |
| 20:14.07 | coppice | anthm: don't get me wrong. I'm all on favour of confusing the heck out of people. |
| 20:15.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:20.45 | Janos | i think VoIP is to network bandwidth/communications what a programming language is to RAM, when you have 8GB who cares if you program in Java, .NET, Python or whatever, that does not mean assembler is not better in every way that actually matters |
| 20:23.08 | _Corey_ | Janos: Not really... VoIP is significantly more efficient with "bandwidth/communications" than its predecessors and uses far less bandwidth than most things on networks |
| 20:23.58 | tzafrir_laptop | Janos, huh? VoIP can be much more sensitive to the latency (and jitter) in the network |
| 20:25.01 | rjune | Janos, I think you don't understand networks very well. |
| 20:25.44 | rjune | or programming for that matter. |
| 20:27.56 | WIMPy | People didn;t want ATM because of the overhead. Now with RTP/UDP/IP we produce much more overhead and have no real QoS. |
| 20:28.23 | jaytee | hahaha, the DIAL timeout default is 136 years!!! So who tested that and verified it? :-) |
| 20:28.30 | WIMPy | And then for all the DSL users the whole thing actually even sits on top of an ATM link. |
| 20:31.50 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 20:33.05 | Janos | yes VoIP might be more efficient about traffic but make it play nice with a bunch of other things and it just breaks cuz there is no real control of the communications layer, just like there is no real control of the system memory with some of the high level languages, but rjune might be right, maybe i don't have a clue |
| 20:33.40 | WIMPy | VOIP is not efficient at all. |
| 20:34.13 | _Corey_ | on a lighter note... our voice helpdesk just got one of the best tickets in a while: |
| 20:34.17 | _Corey_ | "Ok, read now, laugh in a minute. I have a user, XXXXX Extension XXXX, that wants a smiley face at the end of his display name. No lie. " |
| 20:35.33 | WIMPy | Yes. Names are one of the good points in SIP. |
| 20:35.36 | nny | i know this isn't asterisk related *at all* (well it is for what I am doing) but I can't figure out my fail |
| 20:35.46 | nny | i have a new cron dir http://pastebin.com/V7DeBma8 using http://pastebin.com/SqSnNqN1 but i get http://pastebin.com/KPqUs9gD can anyone assist me with what I may be doing wrong? |
| 20:35.52 | nny | hides |
| 20:37.29 | WIMPy | Why don't you use an endless loop? |
| 20:38.08 | WIMPy | while true;do sleep 20;start_something;done |
| 20:38.36 | nny | WIMPy: hmm this script is setup to run once per cycle.. lot more involved on that end |
| 20:41.40 | nny | i see it |
| 20:41.48 | *** join/#asterisk luisfelice (~luisfelic@190.79.25.6) |
| 20:41.50 | nny | my format for the subdir job is all wrong |
| 20:42.09 | WIMPy | 3 forks less per minute :-) |
| 20:45.25 | nny | WIMPy: well, the issue is the subdir/dialer.cron is formatted like a crontab (wrong on my part) I want to call the script (which dies when done) 3 times in intervals of 20 and then die. |
| 20:45.30 | nny | /bin/bash /usr/local/bin/dialer.sh && /bin/sleep 20 && /bin/bash /usr/local/bin/dialer.sh && /bin/sleep 40 && /bin/bash /usr/local/bin/dialer.sh is ham handed |
| 20:45.54 | nny | but would work :P |
| 20:46.40 | WIMPy | But each minute, or did I get that wrong? |
| 20:47.00 | nny | yeah once on the minute, once at :20, :40 and then starts over |
| 20:47.22 | nny | gonna make a clean call |
| 20:47.29 | nny | the &7 stuff is garbage |
| 20:47.29 | WIMPy | So that's just every 20s, forever. |
| 20:47.31 | nny | &&* |
| 20:47.35 | nny | for i do something etc |
| 20:47.50 | nny | until the cron job is removed |
| 20:47.52 | nny | :) |
| 20:48.00 | WIMPy | sure |
| 20:48.42 | nny | lemme make a real call, just user error with cron atm :\ |
| 20:48.42 | luisfelice | Hi, I am trying to use the transfer medianly setting between two IAX2 phones, but when any of these two phones calls the other one I can see full activity on the switch on the Server and phones status light, I thought that I should have activity on the Phones ports and maybe a little on the server port, any idea? |
| 20:51.57 | WIMPy | luisfelice: Do you have any features enabled, that require Asterisk to stay in the media path, i.e. anything controlled via DTMF? |
| 20:52.55 | luisfelice | WINPy: I don't think so, but can you give me something to check? |
| 20:53.08 | WIMPy | Your Dial() Options. |
| 20:56.35 | nny | WIMPy: lol for now /bin/bash /usr/local/bin/dialer.sh & { /bin/sleep 20 && /bin/bash /usr/local/bin/dialer.sh; } & { /bin/sleep 40 && /bin/bash /usr/local/bin/dialer.sh; } :) |
| 20:56.49 | nny | WIMPy: refine a bit later |
| 20:58.03 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:58.25 | luisfelice | WIMPy: is using the macro macro-stdexten |
| 20:58.53 | luisfelice | WIMPy: to call between extensions |
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| 20:59.04 | WIMPy | There must be at least a hundred versions of that. |
| 20:59.23 | WIMPy | But I guess most of them will have at least a t in the options. |
| 21:00.04 | WIMPy | That will require Asterisk to stay in the media path. |
| 21:01.59 | luisfelice | WIMPy: Yes I am reading tTkK options, so I believe this could be the problem |
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| 21:11.40 | fireman_biff | Hi, one of my PRIs is showing as "provisioned, down, active", but there are no alarms on it. The other PRI on that box is fine. What should I check? |
| 21:12.26 | beek | fireman_biff: That your provider actually has that PRI hot |
| 21:13.17 | fireman_biff | beek: the lights are lit up like normal on both, could it still be on their end? |
| 21:14.15 | beek | yes |
| 21:14.42 | fireman_biff | alright, thanks |
| 21:17.43 | Bezzie | Evening all, running into an odd bug with asterisk 1.6... (Havn't tried in 1.8, as I was getting crashing issues with asterisk-gui on 1.8.) ... Just curious if anyones seen it or if theres a workaround known: It appears that say my net connect goes out and my box cannot perform dns lookups (timing out on doing them), asterisk while attempting to nslookup for registering my trunk, will cease to handle |
| 21:17.45 | Bezzie | ALL sip traffic |
| 21:17.47 | Bezzie | even local extensions |
| 21:17.49 | Bezzie | im finding it quite odd |
| 21:17.58 | Bezzie | with sip debug on, it just... stops |
| 21:18.01 | Bezzie | its like it's in sleep mode :P |
| 21:18.12 | Bezzie | Then once it's done, it will allow it, |
| 21:18.52 | Bezzie | running the stock package in debian 's repository. I know its a bit outdated. :P |
| 21:20.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:23.38 | Bezzie | thinks he just found a workaround by enabling dnsmgr |
| 21:23.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 21:23.53 | Bezzie | bingo |
| 21:25.53 | Bezzie | resolved one of my issues :P |
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| 22:03.06 | nny | anyone know of an up to date sound list for asterisk? |
| 22:03.55 | nny | nm |
| 22:04.00 | nny | wiki seems good enough |
| 22:06.14 | ChannelZ | or 'ls' |
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| 22:20.24 | ChannelZ | Ugh. Someone got some crazy salad dressing or something for lunch, it smells like throw-up. This is worse than burnt popcorn |
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| 22:38.11 | Naikrovek | anyone here use EC2 w/asterisk |
| 22:41.15 | Naikrovek | having a small network issue, it seems. wondering if anyone has seen it before. |
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| 22:53.52 | *** join/#asterisk midnightwalker05 (c8578026@gateway/web/freenode/ip.200.87.128.38) |
| 22:54.43 | midnightwalker05 | hello somebody can help me please, My asterisk server was working fine but then we switched isps now i get this error http://pastebin.com/tiqq65FU |
| 22:55.14 | midnightwalker05 | am not sure if it has something to do with that but i get 403 RTTS-Invalid host |
| 22:55.21 | midnightwalker05 | any ideas please?? |
| 22:59.47 | nny | can you make read playback multiple files? or just 1 |
| 23:00.45 | ChannelZ | midnightwalker05: does your ITSP know you on a static IP? |
| 23:01.04 | ChannelZ | or by a hostname which no longer resolves correctly to you? |
| 23:01.41 | ChannelZ | nny: yes, with & - see 'core show application Read' |
| 23:01.49 | nny | ChannelZ: derpt hanks |
| 23:02.07 | midnightwalker05 | ChannelZ: nop, the last connection was also an adsl connection with dynamic IP |
| 23:02.19 | nny | reading the wiki is for chumps! |
| 23:02.25 | nny | like me :( |
| 23:03.51 | ChannelZ | midnightwalker05: well you'd have to ask them then why they're refusing but it feels like they don't know who you are. |
| 23:04.28 | midnightwalker05 | ChannelZ: so am getting rejectect by the provider right?? it has nothing to do with my astbox .. |
| 23:05.04 | ChannelZ | Yes they are rejecting you.. maybe because of your config, but maybe not. |
| 23:05.36 | ChannelZ | They're not asking for auth which is interesting, which is what leads me to believe they have no idea who you are in the first place and are just rejecting the call outright |
| 23:06.19 | ChannelZ | Do you register with them, and it works? |
| 23:06.34 | midnightwalker05 | ChannelZ:yup, i get peer registration with them |
| 23:09.10 | midnightwalker05 | ChannelZ: well thank you cuz I was confused i thought that the call was getting rejected inside of the box, but if my provider is rejecting me i guess I can check with their logs whats going on |
| 23:09.11 | ChannelZ | Since you appear to be behind NAT did you change your sip.conf |
| 23:09.35 | midnightwalker05 | ChannelZ: yes I enabled nat=yes |
| 23:09.46 | ChannelZ | but your externip |
| 23:10.06 | ChannelZ | They might also just be rejecting you because your username is kind of junk.. you didn't change *anything* about your Asterisk config, only switched ISPs? |
| 23:11.10 | midnightwalker05 | nop |
| 23:11.42 | midnightwalker05 | on host i got dynamic |
| 23:11.47 | midnightwalker05 | on sip.conf |
| 23:11.49 | ChannelZ | hmm. Well without knowing their requirements, nothing looks "wrong".. you send an invite, and they say "uhh, no." |
| 23:12.38 | midnightwalker05 | ok i was not able to read that right i thought that my astbox was not even sending the call |
| 23:13.18 | midnightwalker05 | I couldnt find any info on this error SIP/2.0 403 RTTS-Invalid host |
| 23:13.43 | midnightwalker05 | thats the answer am getting from them right? |
| 23:14.16 | ChannelZ | 403 is 'Forbidden'. That they say 'Invalid host' is what suggests to me they are confused by your new IP or whatever. |
| 23:14.49 | ChannelZ | But yes that response is coming from 70.42.72.49 |
| 23:17.37 | midnightwalker05 | ChannelZ: ok thank you very much, Ill send them an email to check whats going on there |
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