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| 00:05.35 | *** join/#asterisk crazyx__ (n=crazyx@41.249.253.247) |
| 00:05.46 | crazyx__ | hello everybody |
| 00:06.06 | crazyx__ | it's the first time I come to this room for help so sorry if i'm doing the things good |
| 00:06.21 | ManxPower | ~ask |
| 00:06.22 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:06.33 | crazyx__ | i just wana no what the 102 cause hangup ( Channel 0/31, span 2 got hangup, cause 102 |
| 00:06.33 | crazyx__ | ) refers to |
| 00:06.41 | crazyx__ | please. |
| 00:07.52 | crazyx__ | sorry, i just wanna know what the 102 causes hangup mean or refers to ( Channel 0/31, span 2 got hangup, cause 102 |
| 00:07.52 | crazyx__ | ) |
| 00:07.59 | crazyx__ | please if someone can help me |
| 00:08.40 | crazyx__ | no one? |
| 00:09.43 | crazyx__ | :'( |
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| 00:14.03 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-78-138.nycmny.east.verizon.net) |
| 00:14.04 | crazyx__ | ok another question : i got troubles on connecting a intel dialogic DMV1200_A/4E1 (on pri_cpe mode) to a Digium TE410P. I check everything on dialogic card (Q.931 timers, protocol (qsig) etc..) and put everything the same on asterisk but i got some errors and then calls drops. The dialogic is on a predictive dialer ACD and the error is BAD FCS (8), write to XXX failed unknow error 500 making the span "holder" going down. Any advice from someone please? |
| 00:14.38 | crazyx__ | or like now, i can't get call etablished on span 2, all calls hangup with 102 errors. |
| 00:14.45 | crazyx__ | please... :'( |
| 00:15.42 | jaytee | crazyx__, what is the signalling set to on the Digium TE410P's side in zapata.conf or system.conf? |
| 00:15.54 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 00:17.11 | crazyx__ | jaytee : system.conf : span=1,1,0,ccs,hdb3 |
| 00:17.15 | Steve_J-obs | hello everybody!!! |
| 00:17.19 | rene- | hey, is using local channels as dynamic queue members still the recommended approach or is logging in SIP devices directly ok in 1.6? |
| 00:17.25 | jaytee | crazyx__, sorry, I meant chan_dahdi.conf if you're using dahdi. |
| 00:17.25 | rene- | hello Steve |
| 00:17.47 | Steve_J-obs | hello |
| 00:17.51 | crazyx__ | span2,2,0,ccs,hdb3, as the dialogic is configured |
| 00:18.11 | jaytee | crazyx__, if your dialogic is set to use pri_cpe mode signalling then your digium side needs to be set to pri_net signalling and you need to supply timing. |
| 00:18.22 | jaytee | so the second one in that line should be a 0 |
| 00:18.36 | crazyx__ | jaytee : group = 1 |
| 00:18.36 | crazyx__ | signalling = pri_net |
| 00:19.07 | jaytee | crazyx__, but you're not supplying timing and the dialogic is expecting a timing source from the net side |
| 00:19.08 | crazyx__ | jaytee : i got the same troubles with span1,1 span2,2 span3,3 span4,4 |
| 00:19.13 | crazyx__ | span1,1 span2,2 |
| 00:19.18 | crazyx__ | span1,1 span2,0 |
| 00:19.23 | crazyx__ | i got the same troubles |
| 00:19.45 | crazyx__ | less on span1,1 span2,0 |
| 00:19.56 | jaytee | crazyx__, what is span1,1 and span2,2? |
| 00:20.04 | crazyx__ | but give me ten minutes i can retry and give u a feedback |
| 00:20.18 | jaytee | crazyx__, are these both located in the same facility? |
| 00:20.30 | crazyx__ | i mean span1,1,0,css,hdb3 and span2,2,0,css,hdb3 |
| 00:20.38 | crazyx__ | jaytee : oups what's facility ? |
| 00:20.49 | jaytee | crazyx__, in the same building |
| 00:20.53 | crazyx__ | yeah |
| 00:20.58 | crazyx__ | in the same technical room |
| 00:21.03 | jaytee | are you using crossover cables? |
| 00:21.11 | crazyx__ | yeah E1 cross cables |
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| 00:21.20 | crazyx__ | it's working well with quintum before |
| 00:21.28 | crazyx__ | but now the provider takes his hardware |
| 00:21.44 | crazyx__ | and i have to do with the TE410P :( |
| 00:22.18 | jaytee | try changing it from span=1,1,0,ccs,hdb3 to span=1,0,0,ccs,hdb3 |
| 00:22.33 | crazyx__ | ok |
| 00:22.41 | crazyx__ | i do it, launch a test |
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| 00:22.47 | crazyx__ | and give u the feedback |
| 00:23.09 | crazyx__ | thanks a lot i'm trying for more than two weeks it can works 2 days without troubles and sometimes 10 mn not more |
| 00:23.14 | jaytee | the second 0 means YOU supply timing to the dialogic. the fcs errors means a loss of timing sync on the d channels. |
| 00:23.26 | crazyx__ | ok |
| 00:23.29 | crazyx__ | i try |
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| 00:23.40 | crazyx__ | thanks a lot for taking times to answer me jaytee |
| 00:23.51 | jaytee | crazyx__, remember you need to restart zaptel or dahdi, whichever one you're using |
| 00:24.31 | crazyx__ | yeah ok i'm restarting it |
| 00:24.48 | Ryushin | How does someone break leaving a voicemail message and go back to the main menu? |
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| 00:26.26 | crazyx__ | jaytee one question before starting the test : is it better to erase the customised Q.931 from chan_dahdi.conf before trying or not? |
| 00:27.08 | jaytee | crazyx__, pastebin your chan_dahdi.conf file |
| 00:27.12 | jaytee | ~pb |
| 00:27.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 00:27.21 | beek | evening jaytee |
| 00:27.51 | jaytee | evening beek |
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| 00:28.55 | crazyx__ | jaytee http://pastebin.com/d5c5105dc |
| 00:32.00 | crazyx__ | jaytee http://pastebin.com/m3b5c98a0 |
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| 00:34.22 | crazyx__ | jaytee something wrong with the config? |
| 00:34.47 | crazyx__ | jaytee i'm restarting the dialogic too... just one moment before i can start the rest |
| 00:34.49 | crazyx__ | test |
| 00:35.10 | jaytee | crazyx__, just hold on a darn minute |
| 00:36.53 | crazyx__ | yep |
| 00:37.43 | jaytee | crazyx__, what country are you in? |
| 00:38.31 | jaytee | france? |
| 00:38.44 | crazyx__ | jaytee in morroco, but the system calling in france, us, canada, uk, and italia |
| 00:39.06 | crazyx__ | the dialogic is on us, and i can't change nothing on the dialogic |
| 00:39.13 | j_o_e | hello, I have a budgetone 200 hardphone. It's able to receive calls and everything works perfectly. However, when I dial out voices are not audible and all I hear on the other end is fast clicking/popping noises (sort of like the predator) ... How can I troubleshoot this? My theory is that it's vocoder related and I tried unsuccessfully to force ulaw |
| 00:39.28 | jaytee | crazyx__, switch to the PM window |
| 00:41.04 | crazyx__ | PM window ? oups what's it ? |
| 00:41.10 | crazyx__ | ah ok |
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| 01:03.34 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 01:03.38 | shmaltz | hi everyone |
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| 01:07.56 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 01:14.16 | shmaltz | ~ping |
| 01:14.17 | jbot | ~pong |
| 01:14.25 | shmaltz | ~anyone here? |
| 01:14.40 | shmaltz | ~anyone? |
| 01:14.41 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
| 01:14.56 | shmaltz | ~ok |
| 01:14.56 | jbot | fine |
| 01:15.09 | shmaltz | ~stupid |
| 01:15.10 | jbot | well, stupid is http://fun.drno.de/pics/english/bart.gif |
| 01:15.22 | shmaltz | ~tv |
| 01:15.23 | jbot | No TV and no beer makes purl something-something. |
| 01:15.55 | jaytee | "Sorry, but we're all busy and can't come to the chat at the moment. Please leave a message at the beep and we'll return your ping as soon as we can!" |
| 01:16.21 | jaytee | BEEP |
| 01:16.53 | shmaltz | hi jaytee, was just interested to know what you busy with, you can get back to me at /msg shmaltz if no answer leave a msg with nickserv |
| 01:17.14 | shmaltz | or is it memeserve? |
| 01:17.17 | shmaltz | checking |
| 01:17.55 | shmaltz | it's memoserv |
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| 01:37.44 | shmaltz | ~sleep |
| 01:37.45 | jbot | it has been said that sleep is overrated, and a poor substitute for caffeine. |
| 01:38.03 | shmaltz | ~really |
| 01:38.04 | jbot | REALLY! |
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| 01:52.14 | hardwire | really. |
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| 02:00.34 | hardwire | e164.org called me 16 times |
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| 02:49.34 | kb3ien | little fuzzy here, this should allow all of the loopback class A only pright (in iax.conf in the context of a peer) deny=0.0.0.0/0.0.0.0 |
| 02:49.34 | kb3ien | permit=127.0.0.0/255.0.0.0 |
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| 03:12.37 | k-man | is it possible to set up a voicemail box with more than one email address? so it will send the messages to multiple recipients? |
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| 03:20.50 | rob0 | It should be trivial in your MTA to set up an alias to go to multiple recipients. |
| 03:21.24 | k-man | rob0: ok - didn't realise that was the approach |
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| 03:32.10 | crazyx__ | hello, please a question : if i'm using a TE410P and it's not a cross over cable, then it can work for some minutes or hours before going down? |
| 03:34.08 | crazyx__ | i mean can it work with a non cross cable E1 and after making problems? and how it's a cross over E1 cable |
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| 03:37.42 | crazyx__ | no one ? |
| 03:38.28 | shmaltz | crazyx__, you there? |
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| 03:38.52 | *** part/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 03:38.59 | crazyx__ | shmaltz i'm there |
| 03:39.01 | *** join/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 03:39.16 | shmaltz | a T/E1 will NOT work with a cross over cable if it needs a striaght thru, or with a striaght thru if it needs a cross over EVER |
| 03:40.22 | shmaltz | however if you are using a device (CSU/DSU or EC) between telco and/or asterisk that when it's down it bridges to the 2 sides it MIGHT but SHOULDN'T break it b/c it wants a cross over in that config |
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| 03:41.10 | shmaltz | crazyx__, you got that answer? |
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| 03:42.37 | shmaltz | the basic idea is that Net equipment and end equipment need to have the layer1 connection so that the tx pair of the net is rx pair on the end, and viceversa for the rx to tx |
| 03:42.42 | crazyx__ | shmaltz i'm trying to understand u |
| 03:43.10 | crazyx__ | i'm trying to get work a dialogic DMV1200(cpe) with a TE410P(net) |
| 03:43.15 | shmaltz | if both are hardwired to the same pair so that the 1st pair (blue) is expected to be TX on both sides then you need a crossover |
| 03:43.32 | shmaltz | crzyx__, then you need a crossover cable |
| 03:43.45 | crazyx__ | yeah ok that what i understand |
| 03:43.51 | crazyx__ | from what u said before |
| 03:43.57 | shmaltz | crazyx__, are you using a crossover? |
| 03:44.22 | crazyx__ | it's linking (green) and work for some minutes or some hours |
| 03:44.29 | crazyx__ | before HDLC erros |
| 03:44.42 | crazyx__ | so i think it's crossed |
| 03:44.50 | shmaltz | then you are getting layer2 errors not layer1 errors |
| 03:45.22 | crazyx__ | PRI got event: HDLC Abort (6) on Primary D-channel of span 2 |
| 03:45.28 | shmaltz | crossover or straight thru problems would leave you in red state indicating a layer1 error |
| 03:45.48 | shmaltz | crazyx__, what happens after those errors to your phone calls? |
| 03:45.53 | crazyx__ | ok. i don't know what's the layer 2 .. |
| 03:45.58 | crazyx__ | i'm lost. |
| 03:46.05 | shmaltz | Layer1=Physical |
| 03:46.16 | shmaltz | ~iso/osi |
| 03:46.33 | crazyx__ | ok |
| 03:46.38 | crazyx__ | PRI got event: HDLC Abort (6) on Primary D-channel of span 2 |
| 03:46.41 | shmaltz | ~iso osi model |
| 03:46.49 | shmaltz | ~iso osi |
| 03:46.56 | shmaltz | hey jbot cant you answer? |
| 03:47.08 | shmaltz | ~wiki iso osi model |
| 03:47.29 | shmaltz | ~wiki osi model |
| 03:47.46 | crazyx__ | hum... |
| 03:48.12 | crazyx__ | ok |
| 03:48.17 | crazyx__ | thanks u shmaltz |
| 03:48.27 | crazyx__ | i'm going to make some research |
| 03:48.30 | crazyx__ | thanks u |
| 03:48.31 | shmaltz | basicly for the purpose of this conversation, layer1 is the wiring |
| 03:48.48 | shmaltz | while anything higher than layer one is configuration and/or software/drivers |
| 03:49.02 | shmaltz | crazyx__, don't leave yet |
| 03:49.19 | shmaltz | once you get that HDLC error what happens to your phones? |
| 03:49.29 | shmaltz | crazyx__, -v? |
| 03:49.33 | shmaltz | what version? |
| 03:49.54 | shmaltz | ~wiki the reader |
| 03:50.24 | crazyx__ | it work sometimes for a whole day |
| 03:50.31 | crazyx__ | and sometimes for 10 mn |
| 03:50.40 | shmaltz | crazyx__ it works after the HDLC errors? |
| 03:50.42 | crazyx__ | all the calls in the channels is drop |
| 03:50.56 | shmaltz | when you get the error all calls drop? |
| 03:51.01 | crazyx__ | asterisk 1.6.0.1 dahdi |
| 03:51.09 | shmaltz | why dahdi? |
| 03:51.13 | crazyx__ | all the calls on the specified span yeah |
| 03:51.31 | crazyx__ | juste because it's working better on dahdi |
| 03:51.34 | crazyx__ | on zaptel |
| 03:51.45 | crazyx__ | 10 mn and error |
| 03:51.58 | shmaltz | and does it stay down? if yes till when? restart? |
| 03:52.11 | crazyx__ | checking the logs |
| 03:52.32 | crazyx__ | restart |
| 03:52.33 | crazyx__ | <PROTECTED> |
| 03:52.55 | crazyx__ | <PROTECTED> |
| 03:52.55 | crazyx__ | [Mar 4 04:26:53] WARNING[5031]: chan_dahdi.c:3347 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! |
| 03:52.57 | crazyx__ | and then |
| 03:53.02 | crazyx__ | == Primary D-Channel on span 2 up |
| 03:57.39 | andresmujica | time sync source probably |
| 03:58.04 | shmaltz | andresmuijca, didnt' even think about it |
| 03:58.08 | shmaltz | you might be right |
| 04:00.10 | ManxPower | crazyx__: are you using sangoma? |
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| 04:02.05 | shmaltz | ManxPower no it's a TE4xx from digium |
| 04:04.04 | jaytee | the card it connects to is a Dialogic |
| 04:04.14 | jaytee | and he's definitely having timing issues |
| 04:05.10 | shmaltz | jaytee, he says that he can set T timers and q. timers but not clocking source |
| 04:05.48 | jaytee | on the Dialogic? yeah, I guess but I've never dealt with a Dialogic card. |
| 04:06.33 | shmaltz | anyone here watched The Reader? |
| 04:06.54 | jaytee | no, new show? |
| 04:07.05 | ManxPower | HDLC issues are caused by a variety of problems. |
| 04:07.28 | ManxPower | The most common is a basic hardware issue with interrupts being locked by device/driver/etc. |
| 04:07.36 | ManxPower | the less common are caused by telco line errors |
| 04:09.46 | jaytee | ManxPower, he's connecting them in the same building |
| 04:10.14 | andresmujica | the abort(6) error is weird |
| 04:11.03 | jaytee | and the HDLC Aborts FCS errors don't happen constantly like with a line problem but intermittently which sounds more like an interrupt issue but he ran dahdi_test and it kept coming back 99.x%. never under 99.something. |
| 04:11.40 | andresmujica | maybe the USB is sharing the irq? or the ethx? |
| 04:12.13 | jaytee | I was thinking the NIC earlier. He said it craps out under load |
| 04:12.30 | jaytee | and he's running realtime against a remote SQL server. |
| 04:12.47 | crazyx__ | two days ago |
| 04:12.55 | crazyx__ | it work the whole day without troubles |
| 04:13.00 | crazyx__ | before the two days |
| 04:13.03 | crazyx__ | a lot of problems |
| 04:13.11 | andresmujica | hmmm that could be the issue... some big load stressing the card and irq's fleeing around.. |
| 04:13.18 | crazyx__ | and today a lot of problem (not more than then minutes) |
| 04:13.38 | crazyx__ | for the sql, is the ACD that's connected to, to read number to send it by the dialogic to TE410P |
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| 04:14.03 | andresmujica | not that much load.. how many calls? it's only 1 E1 port used right? |
| 04:14.04 | crazyx__ | for interrupts i check i never loose then |
| 04:14.28 | crazyx__ | no |
| 04:14.33 | crazyx__ | i try to make work the 4 |
| 04:14.41 | crazyx__ | but now i'm trying just 2 |
| 04:14.49 | crazyx__ | <PROTECTED> |
| 04:14.49 | crazyx__ | <PROTECTED> |
| 04:14.49 | crazyx__ | <PROTECTED> |
| 04:14.50 | crazyx__ | <PROTECTED> |
| 04:14.50 | crazyx__ | <PROTECTED> |
| 04:14.50 | crazyx__ | <PROTECTED> |
| 04:14.52 | crazyx__ | <PROTECTED> |
| 04:14.54 | crazyx__ | sorry |
| 04:14.57 | crazyx__ | a mistake |
| 04:14.58 | brunner | how can I determine what kind of phone line a UK number is attached to? I mean, are there special area codes for land lines, mobile phones, national, personal, etc? |
| 04:15.23 | crazyx__ | the /proc/interrupts : http://pastebin.com/m5dec6d44 |
| 04:15.34 | crazyx__ | thanks to everybody |
| 04:15.45 | crazyx__ | really thanks to you |
| 04:16.06 | andresmujica | it's a dell? |
| 04:16.40 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:16.44 | crazyx__ | HP pro liant DL380 |
| 04:16.52 | crazyx__ | but the dialogic is on a Dell server |
| 04:17.30 | andresmujica | nope, i've seen weirdo things with dell.. not with hp.. hmm maybe with hp too... :P |
| 04:18.11 | toddejohnson | Anyone know how to do alert-info for both grandstream and polycom phones at the same time? |
| 04:18.39 | shmaltz | crazyx__, read also this: |
| 04:18.40 | shmaltz | http://www.asteriskguru.com/archives/asterisk-users-hdlc-abort-6-error-vt35785.html?highlight=hdlc+abort++error |
| 04:19.08 | andresmujica | whih distro is it? |
| 04:19.11 | shmaltz | toddejohnso, are you trying to send the exact same type of alert info? |
| 04:19.13 | aaroneous | hey.. I am filling out the paperwork to order PRI service for my company.. have a few questions about some technical details.. anyone knowledgeable who can help? |
| 04:19.14 | crazyx__ | ok 'im going to read |
| 04:19.15 | andresmujica | which |
| 04:19.16 | crazyx__ | on a debian |
| 04:19.29 | shmaltz | aaroneous, JUST ASK |
| 04:20.14 | toddejohnson | shmaltz, I'm trying to make a FreePBX ring group ring different than normal calls. |
| 04:20.25 | shmaltz | toddejohnson, before dialing the phones just do 2 lines of alert info |
| 04:20.27 | shmaltz | it should work |
| 04:20.47 | crazyx__ | shmaltz, about the link of asteriskguru, i don't have on it IDE devices, but SCSI RAID 15000tr/mn |
| 04:20.48 | toddejohnson | shmaltz, thanks |
| 04:20.49 | shmaltz | since the Polycom will ignore what the gs expects and viceversa |
| 04:20.57 | aaroneous | okay.. first off, it asks if I want ESF/B8ZS framing or SF/D4AMI framing |
| 04:21.14 | shmaltz | aaroneous, esf/b8zs |
| 04:21.17 | shmaltz | next? |
| 04:21.39 | aaroneous | k.. so that will be good, for, say, an audiocodes or cisco media gateway, right? |
| 04:22.07 | shmaltz | aaroneous, well I assumed you mean for digium hardware |
| 04:22.12 | andresmujica | crazyx: most probably sas, no? |
| 04:22.15 | shmaltz | but anyhow I don't see why not |
| 04:22.21 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:22.29 | aaroneous | shmaltz.. yeah I assume that's the more modern framing |
| 04:23.00 | aaroneous | next it says "DID" "DOD", or "Combo" |
| 04:23.37 | andresmujica | but your problem i would say is a timing issue, if * is the net, maybe the dialogic box is not configured for receive the time sync, or viceversa maye your * is expecting the source from the dialogic... |
| 04:23.38 | shmaltz | you using this both ways? |
| 04:23.38 | aaroneous | I want DNIS and DIDs, but I would assume DOD is the very purpose of most phone lines, so it's slightly confusing |
| 04:23.42 | aaroneous | yes |
| 04:24.58 | shmaltz | then I guess it's combo |
| 04:25.02 | shmaltz | but ask them to make sure |
| 04:25.58 | crazyx__ | andresmujica, what's sas :$ |
| 04:26.04 | aaroneous | yeah I will in the morning.. just trying to get through as much of this now so that I can turn it around quickly and get off our terrible SIP origination experience as quickly as possible |
| 04:26.30 | crazyx__ | andresmujica the dialogic is cpe, and TE410P is network |
| 04:26.35 | crazyx__ | ok |
| 04:26.42 | crazyx__ | what i can do for time sync? |
| 04:26.47 | aaroneous | finally, "DNIS Digits"? |
| 04:27.08 | aaroneous | and "Block Third-Party"? |
| 04:27.44 | shmaltz | DNIS, I usualy ask for 4 digits |
| 04:28.05 | shmaltz | I don't know what the "Block Third-Party" is for |
| 04:28.21 | aaroneous | any disadvantage to asking for 10 digits? |
| 04:28.22 | *** join/#asterisk scientes (n=scientes@75-165-95-28.tukw.qwest.net) |
| 04:28.38 | shmaltz | aaroneous, not at all, but why? |
| 04:28.46 | shmaltz | are you going to have 10 digit extensions? |
| 04:29.11 | aaroneous | well we have overlapping area codes here.. just on the off chance that we get two DIDs that share the same last 7 digits.. |
| 04:29.12 | shmaltz | usualy you'd make sure the last 4 of the phone numbers match the extensions in which case your dp will look way simpler |
| 04:29.25 | aaroneous | yeah I am going to try to do that anyway |
| 04:29.37 | shmaltz | aaroneous, then you ask 10 digit DNIS just on that second block of DIDs |
| 04:29.52 | aaroneous | ah I see.. |
| 04:30.15 | aaroneous | why not just ask for it on everything and keep things uniform? |
| 04:30.50 | shmaltz | b/c they might change PSTN dialing to 14 digits some time in the future |
| 04:31.08 | aaroneous | okay |
| 04:31.32 | aaroneous | hopefully telephone "numbers" will be a dead concept by that point :> |
| 04:31.44 | shmaltz | anyone here watched The Reader? |
| 04:32.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 04:32.17 | shmaltz | aaroneous, no what for? |
| 04:32.18 | Damin | Hmm.. is T.38 passthrough supported behind NAT? |
| 04:32.21 | shmaltz | I need to make a living :P |
| 04:32.33 | shmaltz | Damin, it's a codec it is |
| 04:33.30 | shmaltz | is starting to watch The Reader don't interuppt unless your phones are down :P |
| 04:33.41 | aaroneous | shmaltz, you'll make a living the same way, just with SIP or XMPP identities instead of this antiquated notion of having to remember a string of arbitrary numbers for every contact in your address book :> |
| 04:34.32 | shmaltz | aaronewous, I still have to remeber his name that gets translated thru my PIM to an IP or 10 digit phone number or WORSE to an IP6 address |
| 04:34.53 | shmaltz | it's all the same |
| 04:35.03 | shmaltz | who nowadays remembers phone numbers? |
| 04:35.12 | shmaltz | it's all address books in your cell phone or computer |
| 04:35.16 | *** join/#asterisk iamy_china (n=iamy_chi@123.121.182.242) |
| 04:35.17 | aaroneous | well hopefully your DNS server will have a little more functionality by then :> |
| 04:35.32 | shmaltz | i'm the only nut around here that doesn't really use the address book function of my cell phone |
| 04:36.21 | shmaltz | it's the same stupidity, I still have to remeber something, I'd rather it be a 10 digit phone number that has some Geoinfo to it then a 32 bit number or IP6 address |
| 04:36.24 | aaroneous | cool you probably have a better ability to remember phone numbers than the rest of us as a consequence of that |
| 04:36.52 | shmaltz | well, I know my customers better by their extension numbers than by their names |
| 04:37.00 | shmaltz | and some even by their IP addresses |
| 04:37.11 | aaroneous | I don't think anyone will be asking you to associate IP addresses (v4 or v6) with users.. if they are, they aren't doing a very good job at running the core services of their network |
| 04:37.27 | shmaltz | exactly my point |
| 04:38.03 | shmaltz | all your remembeer is a contact that goes by a common name (i.e. Bill Clinton) |
| 04:38.11 | shmaltz | you then use either the Yellow Pages |
| 04:38.17 | aaroneous | but anyway.. D-channel assignment? |
| 04:38.18 | shmaltz | or your MS Outlook |
| 04:38.23 | shmaltz | or your cell phone |
| 04:38.29 | shmaltz | or your whatever to get: |
| 04:38.36 | shmaltz | A. Their phone number |
| 04:38.42 | shmaltz | B. Their fax number |
| 04:38.43 | *** join/#asterisk hapoteh (i=hapoteh@yossman.net) |
| 04:38.52 | shmaltz | C. Their email address |
| 04:38.57 | shmaltz | D. using DNS their IP address |
| 04:39.04 | shmaltz | 24 of course |
| 04:39.16 | shmaltz | aaroneous, you in the states? |
| 04:39.21 | aaroneous | okay yeah I figured it just seemed weird that it's even an option.. |
| 04:39.24 | aaroneous | yes, NYC |
| 04:39.25 | shmaltz | T1 or E1 |
| 04:39.35 | shmaltz | good you'll be using channel 24 |
| 04:39.38 | aaroneous | :> |
| 04:40.18 | aaroneous | do they really need to know PBX make and model? does that determine protocol or is this optional info? |
| 04:40.33 | aaroneous | I guess I could put something like "Cisco 3640" |
| 04:40.49 | shmaltz | aaroneous, this for a Cisco? or Asterisk? |
| 04:40.58 | shmaltz | yes put the make and model, it helps them alot |
| 04:41.19 | shmaltz | aaroneous, you getting my private msgs? |
| 04:41.32 | aaroneous | asterisk is going to be our PBX, but it is going to be doing SIP trunking to the Cisco 3640 |
| 04:41.40 | aaroneous | shmaltz, oh sorry.. yes.. |
| 04:45.31 | *** join/#asterisk iamy_china (n=iamy_chi@123.121.182.242) |
| 04:45.45 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 04:46.28 | brunner | is there a prepaid voip provider that allows you to use an unlimited number of concurrent channels? |
| 04:47.00 | shmaltz | burnner, teliax, nufone and tons of others |
| 04:47.05 | brunner | thanks |
| 04:49.34 | Damin | shmaltz: What the hell does "it's a codec it is" |
| 04:49.36 | Damin | mean? |
| 04:50.03 | shmaltz | Damin, if your sound works behind NAT then t.38 should work as well |
| 04:50.09 | shmaltz | it's just a codec just like ulaw |
| 04:50.44 | Damin | shmaltz: What are you smoking? T.38 isn't a codec.. it's a protocol.. |
| 04:51.58 | shmaltz | Dmain, do a show codecs on your CLI |
| 04:52.11 | shmaltz | Damin, it's a codec |
| 04:53.18 | Damin | What codec? Doesn't appear in my codec list.. |
| 04:53.25 | Damin | What codec number is it? |
| 04:53.34 | shmaltz | Don't know |
| 04:53.49 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-24f5c11b6563bc6c) |
| 04:53.50 | shmaltz | I am still on 1.2 no T.38 for me |
| 04:54.03 | shmaltz | it' doesn't really matter since it's part of the RTP |
| 04:54.59 | Damin | shmaltz: So on what box did you see T.38 show up in the "show codec" list? |
| 04:55.10 | shmaltz | Damin, I didn't |
| 04:55.27 | shmaltz | but I understood that thats how it was implemented from the dev list |
| 04:55.28 | shmaltz | http://www.voip-info.org/wiki/view/T.38 |
| 04:55.51 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 04:56.47 | Damin | shmaltz: Asterisk 1.4 implemented T.38 Passthrough... not a T.38 "codec". |
| 04:57.06 | Damin | shmaltz: it allows endpoints that speak T.38 to bridge their traffic.... |
| 04:57.18 | shmaltz | Damin, I know |
| 04:57.36 | shmaltz | which is a descriptor in the codec as far as I understand |
| 04:57.49 | shmaltz | I just assumed it would show up in codecs |
| 04:57.58 | *** join/#asterisk erth64net (n=erth64ne@c-76-105-207-9.hsd1.or.comcast.net) |
| 04:58.24 | Damin | From the document that you just pointed me to: "From version 1.4, Asterisk supports T.38 negotiation for SIP users, and the related passthrough of UDPTL T.38 data. This allows many T.38 nodes to communicate through an Asterisk box. Asterisk 1.4 does not, however, understand the T.38 protocol. It cannot terminate T.38 calls, or act as a T.38 PSTN gateway without external support - i.e. by passing the T.38 data to something which can perform those functions |
| 04:59.34 | *** join/#asterisk dan__t (n=dant@ns1.hitb.net) |
| 04:59.38 | dan__t | 'evening. |
| 04:59.54 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:00.03 | shmaltz | Damin, I know that, I remeber when it was announced |
| 05:00.07 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 05:00.10 | shmaltz | however it's still part of the RTP |
| 05:00.11 | dan__t | Any AGI hackers around? More specifically, any that do it in PHP? I'm trying to figure out some better debugging techniques for debugging AGI in real-time. |
| 05:00.12 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 05:00.41 | *** part/#asterisk toddejohnson (n=toddejoh@69.220.214.70) |
| 05:02.13 | [TK]D-Fender | dan__t: Show your debug, scripts, etc, and ask the questions you have to ask and see what happens |
| 05:02.32 | dan__t | Yeah, well, they're more questions of the general type. Good practice etc etc. |
| 05:02.45 | shmaltz | [TK]D-Fender don't you love it when ppl ask if they can ask a question? |
| 05:02.57 | shmaltz | did they ask permission to ask THAT question? |
| 05:03.45 | [TK]D-Fender | shmaltz: Like certain people evading a ban asking if they can come back :) I've seen the answer in the reason code for their summary kick-ban :) |
| 05:04.40 | Damin | Hmmm.. [2009-03-04 00:04:20] NOTICE[21057]: rtp.c:1285 ast_rtp_read: Unknown RTP codec 100 received from '207.166.196.242' |
| 05:06.42 | [TK]D-Fender | dan__t: Good practice? General coding rules apply. Nothig to comment on. |
| 05:06.42 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 05:06.42 | dan__t | Got it. |
| 05:06.42 | dan__t | Guess that's something I need to improve on as well heh. |
| 05:07.40 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 05:12.07 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 05:17.21 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 05:22.14 | Damin | <PROTECTED> |
| 05:22.23 | Damin | T.38 passtrhough works.. |
| 05:31.59 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 05:32.35 | shmaltz | bye guys gtg |
| 05:32.37 | shmaltz | l8r |
| 05:34.00 | crazyx__ | guy, plz, did someone ever try to put Intel dialogic DMV1200A/4E1 (cpe) with TE410P (net) without HLDC error ? |
| 05:34.09 | crazyx__ | anyone had experience on this? |
| 05:35.27 | stabler | ~sipnat |
| 05:35.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:35.48 | stabler | bot still down? |
| 05:35.50 | stabler | ~book |
| 05:35.51 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:35.57 | stabler | oh |
| 05:35.57 | stabler | woo |
| 05:36.47 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 05:42.57 | *** join/#asterisk Badrobot- (i=BadRobot@cpe-76-173-233-75.socal.res.rr.com) |
| 05:43.18 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 05:48.14 | hapoteh | i'm struggling to get a gs102 to register on my lan |
| 05:49.02 | hapoteh | i am following the book, chapter 4 sip.conf example |
| 05:49.21 | hapoteh | but phone says unregistered and sip show peers tells me: |
| 05:49.34 | hapoteh | gs102/gs102 (Unspecified) D 0 Unmonitored |
| 05:50.00 | hapoteh | am i missing something obvious here? |
| 05:51.38 | [TK]D-Fender | hapoteh: wel have no idea what you may have misconfigured. You need to PASTEBIN the SIP debug of your failed register attempts so taht we can see what kind of errors are being thrown off. |
| 05:51.40 | [TK]D-Fender | ~pb |
| 05:51.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 05:54.41 | hapoteh | [TK]D-Fender: i'm totally new to this, where is the sip debug info? |
| 05:55.42 | [TK]D-Fender | hapoteh: "sip set debug on" at * CLI and pastebin the output. |
| 05:56.48 | hapoteh | i do 'sip set debug' but nothing comes out after that, even if i do sip reload |
| 05:57.11 | hapoteh | if i do 'sip set debug on' it gives me usage |
| 05:57.22 | hapoteh | telling me to run it either on ip or peer |
| 05:57.27 | hapoteh | or nothing |
| 05:57.45 | hapoteh | no more debug output |
| 05:57.56 | k-man | how do i make an extension to log into my voicemail directly? without having to type the voicemail box number or pw? |
| 05:59.28 | [TK]D-Fender | hapoteh: Restart your phone |
| 05:59.39 | [TK]D-Fender | k-man: Is your phone PSYCHIC?! |
| 06:00.11 | [TK]D-Fender | k-man: "core show application voicemailmain" <- |
| 06:00.35 | k-man | [TK]D-Fender: its my home asterisk server - i only have one mailbox |
| 06:01.26 | hapoteh | oooh psychic phones??!! /me wants. |
| 06:01.45 | k-man | thanks [TK]D-Fender |
| 06:02.28 | hapoteh | [TK]D-Fender: i restarted the phone but nothing |
| 06:02.39 | hapoteh | no output in the cli |
| 06:05.06 | [TK]D-Fender | hapoteh: then either you have a networking/firewall issue, or you have misconfigured your phone and it isn't even trying to talk to your * server |
| 06:06.52 | hapoteh | hmm. |
| 06:07.00 | hapoteh | phone and * server are on a switch |
| 06:07.10 | hapoteh | and the * server isn't running any firewall |
| 06:07.24 | hapoteh | so i guess i misconfigured the phone |
| 06:07.35 | hapoteh | i gave it std port and the correct ip |
| 06:08.16 | hapoteh | investigates |
| 06:08.24 | hapoteh | thanks for the pointers [TK]D-Fender |
| 06:08.45 | [TK]D-Fender | hapoteh: Go verify that and that hapoteh Wait, try "sip set debug" |
| 06:08.51 | [TK]D-Fender | hapoteh: without "on" |
| 06:08.59 | [TK]D-Fender | hapoteh: I think the syntax may be alittle off |
| 06:10.55 | hapoteh | yeah i did that |
| 06:11.04 | hapoteh | gifthorse*CLI> sip set debug |
| 06:11.04 | hapoteh | SIP Debugging re-enabled |
| 06:11.11 | hapoteh | but i think you're right |
| 06:11.14 | hapoteh | the phone is misconfigured |
| 06:11.26 | hapoteh | I saw 192.168.6.3 for sip server |
| 06:11.29 | [TK]D-Fender | hapoteh: Well if you still see nothing... still double check iptables, etc |
| 06:11.31 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 06:11.33 | hapoteh | it should have been 192.168.69.3 |
| 06:11.38 | [TK]D-Fender | that'd do it :) |
| 06:11.44 | hapoteh | no iptables :) |
| 06:11.47 | hapoteh | running on fbsd |
| 06:11.47 | Qwell | So, who wants to contribute to the "buy Qwell a new video card" fund? :P |
| 06:11.54 | Qwell | (kidding, of course) |
| 06:11.58 | *** join/#asterisk The_LightSide (n=dgush@dsl-241-44-37.telkomadsl.co.za) |
| 06:12.14 | hapoteh | Qwell: if you paypal me shipping i have an ati rage pro 3d 8meg card you can have.... |
| 06:12.21 | Qwell | heh |
| 06:12.34 | stabler | lol |
| 06:12.56 | stabler | 8meg FTW |
| 06:13.01 | k-man | if i want asterisk prompts in a different language, and I only use sip, do I just need to put language=au in sip.conf? will that cover voicemail prompts also? |
| 06:13.40 | stabler | k-man: do you have that critter running smoothly now? |
| 06:13.41 | [TK]D-Fender | k-man: Yes |
| 06:13.55 | k-man | stabler: yeah, pretty damn well actually |
| 06:14.25 | k-man | stabler: once i ironed out the problems wich mostly were to do with my vsp not telling me they had turned off the DID |
| 06:14.54 | crazyx__ | please, is there someone experienced before interco E1 between dialogic Intel (cpe) and TE410P (net) ? i got lot of troubles like Bad HDLC errors and i don't know what to do |
| 06:15.13 | k-man | i noticed today a little bit of disturbed playback of voicemail prompts - not sure why that was |
| 06:15.18 | k-man | but it was fairly minor |
| 06:15.34 | hapoteh | [TK]D-Fender: now i have it connecting sort of |
| 06:15.36 | k-man | stabler: and now that its all running, i'm really impressed with it |
| 06:15.37 | [TK]D-Fender | crazyx__: pastebin "dmesg" , "ztcfg -vvvv", and "cat/proc/interrupts" |
| 06:15.38 | hapoteh | still unmonitored |
| 06:15.39 | [TK]D-Fender | ~pb |
| 06:15.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 06:15.41 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 06:15.50 | [TK]D-Fender | hapoteh: unmonitored is fine. |
| 06:15.52 | crazyx__ | ok [TK]D-Fender |
| 06:15.59 | k-man | stabler: and thanks again for helping me out |
| 06:16.04 | hapoteh | ok sweet |
| 06:16.11 | hapoteh | i think it's bed time |
| 06:16.13 | [TK]D-Fender | hapoteh: restart your phone. If you don't see packets then... still other core issues |
| 06:16.16 | hapoteh | but this is a good first step then |
| 06:16.19 | hapoteh | packets? |
| 06:16.27 | hapoteh | i see things like: <--- SIP read from 192.168.69.10:5060 ---> |
| 06:16.32 | hapoteh | on the CLI debug |
| 06:16.35 | stabler | k-man: nice.. good to hear |
| 06:16.42 | hapoteh | about ever ... 5 or 10 secs |
| 06:17.30 | k-man | stabler: i'm goint to set up voicemail as our answering machine too - i might even go for the "press 1 for jason, 2 for X, 3 for Y" thing |
| 06:17.51 | stabler | nice |
| 06:18.00 | crazyx__ | [TK]D-Fender : http://pastebin.com/m18cbf556 |
| 06:18.07 | [TK]D-Fender | hapoteh: WE should be seeing those things... in full |
| 06:18.14 | k-man | we just had a baby though - so its very hard snatching a few minutes here and there to work on asterisk |
| 06:19.27 | [TK]D-Fender | crazyx__: Wow, TC400 as well? |
| 06:20.35 | crazyx__ | [TK]D-Fender yeah cause E1 is G711 and i'm using the TE410P just for translating to SIP from E1, and then G729 |
| 06:21.10 | [TK]D-Fender | crazyx__: ok, I see nothing inherently wrong. New thing to try : start up with NOAPIC |
| 06:21.32 | crazyx__ | [TK]D-Fender the dialogic card is on a ACD predictif dialer so many channels to compress |
| 06:21.33 | crazyx__ | ok |
| 06:21.34 | *** join/#asterisk The_LightSide (n=lightie@dsl-241-44-37.telkomadsl.co.za) |
| 06:21.38 | [TK]D-Fender | crazyx__: That has helped others in the past. the TE410 is a fidgety card. |
| 06:22.06 | crazyx__ | [TK]D-Fender i try now and make a test and come back with informations. Many thanks for ur help |
| 06:22.20 | [TK]D-Fender | crazyx__: Hope you get things working better soon |
| 06:22.43 | *** join/#asterisk erth64net (n=erth64ne@96-25-65-141.war.clearwire-wmx.net) |
| 06:23.05 | crazyx__ | [TK]D-Fender i hope i'm working on it for more than 2 weeks, got Quintum DX before but i loose them... i hope it will work |
| 06:23.51 | stabler | k-man: babies.. not important... asterisk.. important! |
| 06:24.20 | crazyx__ | [TK]D-Fender stupid question : how to start with NOAPIC on Debian? |
| 06:24.32 | *** join/#asterisk maddog01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 06:24.59 | [TK]D-Fender | crazyx__: No clue... try asking in #debian :) |
| 06:25.10 | crazyx__ | lol ok |
| 06:25.19 | [TK]D-Fender | ok, bed time, later all |
| 06:30.07 | *** join/#asterisk mahiti-irc (n=mahiti1@203.145.183.210) |
| 06:30.34 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 06:31.04 | stabler | ~sipnat |
| 06:31.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:35.34 | mcnobody | codefreeze-lap: Thanks. |
| 06:36.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:36.13 | *** join/#asterisk Shaun222 (n=shaun@ip68-5-154-128.oc.oc.cox.net) |
| 06:37.03 | Shaun222 | i want to allow people with no user/pass to connect to my asterisk server via sip and be put into a certian context, where can i find info on this |
| 06:37.18 | drmessano | allowguest |
| 06:37.28 | crazyx__ | hum.. the same |
| 06:38.03 | *** join/#asterisk intralanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 06:38.14 | k-man | stabler: yeah - but you know- i can't just let him cry |
| 06:38.33 | stabler | k-man: i kid |
| 06:38.44 | k-man | stabler: i know :) |
| 06:39.01 | crazyx__ | <crazyx__> please, is there someone experienced before interco E1 between dialogic Intel (cpe) and TE410P (net) ? i got lot of troubles ( i tried NOAPIC mode |
| 06:39.04 | crazyx__ | no changes |
| 06:39.04 | k-man | stabler: are you in AU? |
| 06:39.14 | stabler | no |
| 06:39.16 | Shaun222 | anybody recommend a sip client for windows thats free? |
| 06:39.17 | stabler | US |
| 06:39.26 | stabler | Shaun222: x-lite |
| 06:39.31 | drmessano | Shaun222: X-lite.. many are |
| 06:39.39 | drmessano | Google is your friend |
| 06:39.49 | stabler | ~softphone |
| 06:39.49 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 06:39.52 | k-man | are any of the free ones any good? |
| 06:40.01 | stabler | Shaun222: ^^^^^^^^^^^^^^^^ |
| 06:40.02 | drmessano | Um yeah |
| 06:40.15 | drmessano | Look at stabler with the TK action |
| 06:40.22 | stabler | :D |
| 06:40.25 | drmessano | Trigger, arrows |
| 06:40.39 | stabler | drmessano: there are no other options |
| 06:40.53 | drmessano | Well, yes.. there is level 2 |
| 06:40.59 | drmessano | ~softphone |
| 06:41.00 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 06:41.06 | drmessano | READ ^^^^^^^^^^^^^^^^^ |
| 06:41.15 | *** join/#asterisk RobertLaptop (n=rmiddle@173-100-33-237.pools.spcsdns.net) |
| 06:41.19 | drmessano | CAPS ^^^^^^^^^^^^^^^^^^ <--- |
| 06:41.22 | stabler | k-man: yes! i love x-lite |
| 06:41.43 | stabler | drmessano: you could point down |
| 06:41.46 | stabler | drmessano: VVVVVVVVVVVVVVVVVVVVv |
| 06:42.31 | drmessano | Hmmm |
| 06:42.36 | drmessano | X-Lite is pretty decent |
| 06:47.06 | Shaun222 | ok, i set allowguest=yes. |
| 06:47.11 | Shaun222 | now trying to connect using x-lite |
| 06:47.21 | Shaun222 | x-lite wont allow me to not specify a user though. |
| 06:51.12 | drmessano | Holy cow |
| 06:51.24 | drmessano | You cant register to asterisk without registering |
| 06:52.04 | drmessano | YOu said you wanted them to "connect" to asterisk, which means to 99.999999% of us, accept a call from a non-authenticated peer |
| 06:52.20 | drmessano | Now you want to register without registering.. |
| 06:52.40 | drmessano | What are you trying to accomplish here? |
| 06:53.28 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 06:56.22 | Shaun222 | ok, maybe i got it wrong. |
| 06:56.30 | Shaun222 | This is what i want to do. |
| 06:57.18 | Shaun222 | i want my customers to be able to configure there sip devices to connect to my asterisk server so that when they dial 0 or any number they basically get thrown into the IVR context. |
| 06:57.46 | Shaun222 | this way international customers with softphones and whatever can call into the phone system with out being charged a ton. |
| 06:58.44 | Shaun222 | hmm, kinda got it working |
| 06:59.06 | Shaun222 | created a friend entry in the sip.conf with allowguest=yes and no secret. |
| 06:59.14 | Shaun222 | default context is ivr. |
| 07:01.38 | stabler | Shaun222: sounds alittle unsecure =/ |
| 07:01.55 | Shaun222 | stabler: what problems do you see? |
| 07:02.28 | stabler | Shaun222: well.. i suppose to you make sure everything is setup nicely youll be fine |
| 07:02.31 | stabler | just be careful |
| 07:02.50 | stabler | *if |
| 07:03.10 | Shaun222 | stabler: this user will be put into the same context that calls from the T1 are put into |
| 07:03.31 | Shaun222 | if they can jump out of it, then i would assume anybody calling in on the T1 could also. |
| 07:03.32 | stabler | so that means i can login |
| 07:03.34 | stabler | and make calls |
| 07:03.36 | stabler | lol |
| 07:03.37 | k-man | if you modify voicemail.conf, do you have to reload anything in the CLI? |
| 07:03.42 | stabler | yea |
| 07:03.44 | k-man | for the changes to take effect? |
| 07:03.49 | stabler | k-man: yes |
| 07:03.56 | k-man | stabler: what do I reload? |
| 07:04.11 | Shaun222 | stabler: you can log in, but the context your in only would only dump you into the IVR, no dial out from there... |
| 07:04.36 | Shaun222 | my context for internal is totally seperate from my context for calling comming from the world. |
| 07:04.59 | stabler | Shaun222: oh ok... so outside mistry users can dial out? if that true then youre pretty safe for the most part |
| 07:05.13 | stabler | *mystery |
| 07:05.48 | stabler | k-man: i norally just do a full reload |
| 07:05.54 | k-man | ok, tahnks |
| 07:05.55 | stabler | by typeing: reload |
| 07:06.07 | Shaun222 | ivr context does the norm, feeds them some options and dumps them into a queue() |
| 07:06.10 | k-man | oh - thats nice. I thought I had to quit to do that |
| 07:06.21 | k-man | thanks stabler. GTG |
| 07:06.37 | stabler | k-man: np... later |
| 07:07.09 | stabler | k-man: nice thing about "reload" is it doesnt drop calls |
| 07:07.29 | stabler | or interrupt anything important |
| 07:07.49 | k-man | stabler: ah - thats good. thanks bye |
| 07:09.40 | Shaun222 | ahh, it's working. sweet... |
| 07:10.00 | Shaun222 | course i wonder how it likes having multiple people using the same sip config? |
| 07:10.16 | stabler | heh |
| 07:10.22 | stabler | it may be fine |
| 07:10.55 | *** part/#asterisk dan__t (n=dant@ns1.hitb.net) |
| 07:11.00 | stabler | sleep time... |
| 07:11.01 | stabler | laterz |
| 07:11.04 | drmessano | later |
| 07:11.17 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:13.27 | Shaun222 | well that was easy... |
| 07:13.45 | Shaun222 | think i'll make my username somthing different than just cuest though. |
| 07:13.52 | Shaun222 | that way the bots out there dont go messin |
| 07:16.15 | *** join/#asterisk ShadowGear2009 (n=Cliff@bas12-montreal02-1242554295.dsl.bell.ca) |
| 07:16.43 | *** join/#asterisk _omer (n=_omer@119.152.52.57) |
| 07:16.46 | _omer | hello |
| 07:17.06 | _omer | Does Asterisk support MESSAGE request ? |
| 07:18.47 | kaldemar | no |
| 07:18.52 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 07:19.23 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:19.31 | _omer | not even in Asterisk Latest Versions ? |
| 07:19.38 | ShadowGear2009 | hey guys, i'm new to asterisk and am trying to set up a home PBX. So far all i got working is asterisk intercepts outgoing calls and plays back tt-monkeys file |
| 07:20.07 | ShadowGear2009 | here is were it gets weird, the quality of the playback is not so great |
| 07:20.15 | ShadowGear2009 | how do i improve it? |
| 07:20.56 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 07:21.33 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 07:22.30 | kaldemar | ShadowGear2009: are you using gsm encoded sound files and gcc >4.2? there's an issue with the gsm playback. |
| 07:23.11 | ShadowGear2009 | i think i might be. How do i go about testing with other files? |
| 07:23.18 | *** join/#asterisk raasdnil (n=mikel@60-240-58-103.tpgi.com.au) |
| 07:23.37 | kaldemar | download some other package from here: http://downloads.digium.com/pub/telephony/sounds/ |
| 07:23.55 | *** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) |
| 07:24.27 | kaldemar | the sounds are in /var/lib/asterisk/sounds/<lang>/ by default |
| 07:24.57 | ShadowGear2009 | thanks i'll try that |
| 07:25.14 | raasdnil | evening all. |
| 07:26.29 | raasdnil | I have a TDM2400 full of FXSs and I have a TDM800 full of FXOs in the same box. If I land a fax on the 2400 and pipe it through to a PSTN on the 800, am I going to run into any problems with the fax via Asterisk? Or will that be totally fine (like I am expecting) ? |
| 07:26.50 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 07:27.16 | brunner | How can I record the audio of an unanswered/unsupervised outbound call? |
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| 07:41.48 | mahiti-irc | brunner : what setup you have ? |
| 07:42.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:48.07 | k-man | i'm trying to make a dial plan to dial my voicemail directly to leave a message - i did this but i just get engaged signal: exten => 775,VoiceMail(777@default) |
| 07:48.50 | k-man | and in voicemail.conf i have 777 => 1234,jason,jason@myemailaddress |
| 07:49.05 | k-man | am I missing something here? |
| 07:50.45 | brunner | mahiti-irc: I'm using the Manager API to generate outbound calls over a SIP or IAX trunk (depending on the call) |
| 07:53.30 | kaldemar | k-man: put a priority in your extension |
| 08:03.56 | k-man | kaldemar: ah! thanks |
| 08:05.59 | k-man | kaldemar: yeah, that fixed it |
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| 09:08.43 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 09:08.58 | angryuser | hello |
| 09:09.32 | *** join/#asterisk mahiti-irc (n=mahiti1@121.243.168.201) |
| 09:10.30 | angryuser | when i have "Remote host can't match request BYE to call" it means that remote host has remover allready his this call leg ? |
| 09:10.43 | angryuser | removed* |
| 09:12.51 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:18.20 | *** join/#asterisk nima0102 (n=nima@91.98.205.100) |
| 09:21.43 | fcois93 | the end of my make in asterisk_src:http://pastebin.com/d645d653f |
| 09:21.49 | fcois93 | I have an error |
| 09:22.32 | nix8n82 | how many channels can IAX2 handle at one time? |
| 09:22.57 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 09:23.14 | mosty | nix8n82, as many as your server/network can handle |
| 09:24.47 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:28.08 | nix8n82 | do you think 1 quad core with 4 gigs of ram with enough network bandwidth can handle 250 calls at once? |
| 09:29.09 | mosty | probably, depending on transcoding and other factors |
| 09:29.10 | nix8n82 | would sip be better? and would iax2 be the best if going to another asterisk box? |
| 09:29.45 | mosty | most setups use sip to end users and iax to other asterisk servers |
| 09:30.57 | nix8n82 | why is that a benefit? for iax2 to servers and not to users? |
| 09:31.46 | mosty | because there are no good phones that use IAX |
| 09:32.34 | mosty | iax is more efficient and flexible than sip, but almost nothing supports iax besides asterisk |
| 09:34.42 | nix8n82 | thank you very much for the info mosty |
| 09:35.39 | mosty | no problem |
| 09:36.19 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 09:39.35 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 09:41.12 | *** join/#asterisk sypher (n=sypher@83-103-99-37.ip.fastwebnet.it) |
| 09:41.17 | fcois93 | nix8n82: lokkatithat http://www.transnexus.com/White%20Papers/Performance_Test_of_Asterisk_v1-4.htm |
| 09:41.31 | sypher | hi guys |
| 09:41.51 | sypher | was wondering if someone had a link to a guide that would help me configure asterisk to use a betamax service ... im googling but still no real luck ... |
| 09:44.31 | mosty | sypher, should be the same as any other SIP provider i imagine? |
| 09:51.19 | sypher | mosty, hum ... yeah. |
| 09:51.30 | sypher | mosty, next hint would be "rtfm"? ... |
| 09:52.48 | mosty | sypher, well yeah, the book would be a good start |
| 09:52.50 | mosty | ~thebook |
| 09:52.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 09:52.55 | *** join/#asterisk Pazzo (n=ugelt@sadsl-246059.rol.raiffeisen.net) |
| 09:53.04 | sypher | thank you |
| 09:54.00 | sypher | actually, have another question (and im a total newbie here) ... a "fax" is just a call right? ... i mean .. if i successfully integrate hylafax and asterisk; and configure asterisk to use a sip provider ... i should get charged as a normal call for each fax, that correct? |
| 09:54.40 | angryuser | nix8n82: well, i still dont like the idea to have 1 thread for all channels, which is the case of iax |
| 09:55.48 | mosty | sypher, fax over sip is a waste of time, until asterisk has full t.38 support. i advise that you give up on that idea |
| 09:56.15 | sypher | mosty, ugh ... why? can you elaborate? |
| 09:56.21 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 09:56.39 | mosty | sypher, jitter breaks fax |
| 09:56.51 | sypher | jitter? |
| 09:57.06 | angryuser | sypher: the data is passed by audio, and minimal delay breaks it |
| 09:57.18 | LuisTorres | Hi.., Im getting some issues on Chanspy. When the remote party is putted on hold and then retrieved, I can not hear anymore both parts. do you know what could it be? |
| 09:57.30 | mosty | sypher, in short, fax was not designed for voip, and it does not work reliably |
| 09:58.03 | sypher | angryuser, mosty, ok. understood that ... so, if i were in the need to send thousand of faxes, what would you do ? have any suggestion ? |
| 09:58.13 | mosty | sypher, get PRI |
| 09:58.29 | angryuser | sypher: i am using hylafax with bri lines and multitech modems |
| 09:58.50 | angryuser | sypher: directly connected to fxs pci card |
| 09:59.15 | sypher | angryuser, im searching hylafax + asterisk to reducing costs ... what are your costs for 1 ? |
| 09:59.33 | angryuser | sypher: what costs ? |
| 09:59.43 | sypher | for sending it ... oh nevermind. |
| 09:59.49 | sypher | it depends on your carrier i suppose ... |
| 10:00.35 | mosty | i have used hylafax with eicon diva cards and PRI lines, works well |
| 10:01.03 | angryuser | sypher: compare prices, some providers are able to provide mail2fax pdf2fax gateways, is more effective sometimes |
| 10:01.21 | sypher | angryuser, yeah, thats what we're using now .... |
| 10:01.44 | sypher | i started searching for asterisk because it seemed to cut costs by 50/60% .. . |
| 10:02.02 | sypher | (with voip) ... but if you guys telling me all the faxes will break .. its not worth it ... |
| 10:02.58 | mosty | sypher, you might be able to find a sip provider that supports t.38, but you will need something other than asterisk on your end |
| 10:03.02 | LuisTorres | Anyone know any issue with Chanspy? |
| 10:04.04 | LuisTorres | if Im listen two partys and one of the party is putted on hold, when is retrieved I cant ear that party anymore |
| 10:04.23 | mosty | LuisTorres, i suggest that you submit a bug report |
| 10:04.44 | LuisTorres | mosty: ok thank you |
| 10:05.10 | angryuser | sypher: if you are so in need of that faxinf over ip, try to search som gateway with t38 support and a provider with t38 support |
| 10:05.39 | sypher | angryuser, im reading around right now ... seems like i can use asterisk + betamax in G711 ... |
| 10:05.59 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:06.51 | angryuser | LuisTorres: i dont remember exactly, just make sure that when the person is put on hold there is no another channel created when of - hold button is pressed, just look at sip debug |
| 10:07.57 | LuisTorres | angryuser: thank you for you answer. On the web I found on the bug report that 1.4.19 had some issues like this. But supposed to be fixed on the further versions. Im using 1.4.21 and it seems still happen |
| 10:08.58 | angryuser | LuisTorres: that's why i am happy with 1.4.18.1 here |
| 10:09.13 | LuisTorres | ahah really? no bugs on chanspy? |
| 10:10.00 | angryuser | LuisTorres: well i dont use it too much ;) but in general this release was among most stables |
| 10:10.17 | LuisTorres | angryuser: ehe Cheers |
| 10:12.12 | *** join/#asterisk SparFux (n=raoul@e182020199.adsl.alicedsl.de) |
| 10:12.50 | SparFux | Hi! Is there some kind of technical term for sending an arbitrary CallerID out as a private person, not one of the own MSNs? Like Fake-CallerID or whatever? |
| 10:13.27 | mosty | sparfux: "unauthenticated" callerid perhaps? |
| 10:13.35 | SparFux | Yes, like that. |
| 10:14.13 | mosty | or "arbitrary" |
| 10:14.35 | angryuser | SparFux: CID ? |
| 10:14.38 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:15.02 | SparFux | I think CID = CallerID. |
| 10:15.19 | angryuser | SparFux: just tell me what do you want to send |
| 10:17.16 | SparFux | angryuser: I am using Alice-DSL german phone provider service. I can call my own Cellphone which is also an Alice phone. And what I want to do is route calls to my home box via asterisk to the cell phone with the CallerID remaining. But what happens is that the CallerID is set to the first MSN I have at Alice, I think because they change it for some obvious reason. I wonder wether there is some code I can type into my Telephone which will tur |
| 10:19.49 | angryuser | SparFux: you just need to verify if you provider is sending you the Caller id and set it before calling your cell phone, also if your provider permit that |
| 10:20.32 | SparFux | Hm... yes. |
| 10:21.02 | SparFux | I am afraid they don't permit it, but perhaps I can switch it on somehow. So I am doing some searches on the net, but I cannot find anything useful. |
| 10:21.06 | angryuser | SparFux: generally providers do not let you set any called id you want, but only from your trunk |
| 10:21.30 | angryuser | but only from your trunk DID's you got |
| 10:22.24 | mosty | SparFux, if they don't permit it, you're out of luck unfortunately |
| 10:22.58 | SparFux | mosty: obviously, yes. |
| 10:24.13 | angryuser | SparFux: the feature you might like for a remplacement is a simple follow me, let say the person is calling you, you answer with * then say, "we are location your called person, please tell me your name" then register it, then * call you, play the recording and propose the choice 1 Accept the call 2 Reject to voicemail |
| 10:24.56 | SparFux | Hehe, yes, nice idea :-) |
| 10:24.58 | mosty | sparfux: you might be able to hack your asterisk box to call your mobile, play a message "call from <say the original callerid>, press 1 to accept" |
| 10:25.03 | mosty | and then connect the two |
| 10:25.10 | SparFux | Mosty: I read about this on voip-info.org I think. |
| 10:25.14 | mosty | ahh, too slow |
| 10:25.46 | SparFux | Yes, slow, but a nice idea for a workaround crap restrictionns :-) |
| 10:26.25 | angryuser | SparFux: our provider is letting us set ani cid we want , but we are using it everyday |
| 10:28.18 | angryuser | SparFux: it is easyer when our provider are us ;) |
| 10:31.16 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 10:32.48 | scruz | hello everyone |
| 10:34.17 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:34.27 | stmaher | hi guys.. |
| 10:34.47 | stmaher | is there a way in the CLI in asterisk to show what lines are used by Dahdi? |
| 10:34.57 | stmaher | like a more verbose output.. |
| 10:35.09 | stmaher | the dahdi commands channels and channel are not really what im looking for |
| 10:35.43 | angryuser | stmaher: analog ? |
| 10:35.48 | stmaher | yep |
| 10:36.01 | stmaher | its an aex800 |
| 10:36.49 | andresmujica | zap show channels, then zap show channel Zap/1-1 (or with dahdi) ... |
| 10:37.37 | stmaher | ok.. but they dont show lines in use? |
| 10:38.15 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:38.41 | angryuser | stmaher: what i did to see what ports are used is a simple check script (ifavail) for each port and asign to one exten, call it and see status |
| 10:39.35 | stmaher | angryuser cool.. would you have a copy of that script? |
| 10:39.35 | andresmujica | ohh in use.. upps |
| 10:39.37 | angryuser | stmaher: took me 2 min |
| 10:40.11 | SparFux | angryuser: what is you? I mean, your provider? |
| 10:40.35 | SparFux | angryuser: the Dial command actually has "follow me" feature, it is option 'p'. |
| 10:41.58 | angryuser | stmaher: it's not a script jsut a simple use of ChanIsAvail for each zap port, look at core show application ChanIsAvail |
| 10:42.25 | stmaher | AHhhhhhhhhhhhhhhhhhhhhhhh |
| 10:42.27 | stmaher | thank joo! |
| 10:42.37 | angryuser | SparFux: i ment that it helps to be your own provider |
| 10:43.01 | SparFux | Ah, I see. :-) |
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| 10:49.56 | vlt | tzafrir_laptop: iaxmodem and hylafax works perfectly. Thank you. |
| 10:51.19 | tzafrir_laptop | stabler, for starters: lsdahdi |
| 10:51.25 | tzafrir_laptop | (in the linux command line) |
| 10:52.00 | tzafrir_laptop | http://docs.tzafrir.org.il/dahdi-linux/#_procfs_interface_proc_dahdi |
| 10:52.48 | tzafrir_laptop | And in the asterisk CLI: dahdi show channel 4 |
| 10:53.08 | tzafrir_laptop | rather than: dahdi show channel DAHDI/4-1 |
| 10:53.46 | k-man | in this bit of code from ATFOT book, http://pastebin.ca/1352618 what is the purpose of line 5? |
| 10:56.11 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 10:56.56 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:57.49 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
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| 10:59.14 | SparFux | If I call multiple extensions in one Dial() command, can I CLIR the CID on one of the called extensions in sip/extension1&capi/extension2&sip/extension3 ? |
| 11:01.04 | Jacke | sure you can |
| 11:01.08 | mosty | k-man, perhaps if there is no voicemail account it goes back to incoming? |
| 11:01.35 | Jacke | build yourself a nice little local extension like _CLIDX., |
| 11:02.09 | Jacke | set clir inside it, then make it call your SIP/{$EXTEN} or whatever you want. |
| 11:02.40 | Jacke | an make your original Dial call Local/CLID{$EXTEN} |
| 11:05.08 | LuisTorres | can anyone tell me that this issue (0012837) is resolved on 1.4.23? About chanspy |
| 11:09.26 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 11:17.14 | SparFux | jacke: Ah, yes, I can even Dial(extension@context) I guess. |
| 11:18.09 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 11:19.18 | SparFux | jacke: oh no, then I would have to use Goto() |
| 11:20.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:21.47 | SparFux | no, I don't get it. |
| 11:24.31 | SparFux | I am stupid, of course it works, with the local/ channel type. yes. |
| 11:26.30 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 11:28.22 | scruz | tzafrir_laptop: you're on my server??!?!??!?! |
| 11:29.02 | tzafrir_laptop | scruz, no |
| 11:29.08 | scruz | yes you are |
| 11:29.23 | scruz | saw tzafrir in the zaptel README |
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| 11:39.36 | scruz | this voip thing is getting troublesome |
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| 11:49.58 | The_LightSide | Hi all, does anyone have feedback on issue 0014112? |
| 11:50.12 | The_LightSide | thread deadlocks |
| 11:50.24 | AndyT | anyone have a click2dial script for Asterisk? |
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| 11:54.12 | scruz | i've had to give up on building dahdi on switch to zaptel |
| 11:55.13 | scruz | and zaptel gives the same error |
| 11:57.36 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 11:57.52 | mosty | scr, what error? |
| 11:58.53 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:01.33 | scruz | no rule to make target * |
| 12:01.48 | mosty | what command are you running? |
| 12:01.55 | scruz | * being a placeholder for the particular dahdi/zaptel package |
| 12:01.59 | mosty | can you pastebin it? |
| 12:02.01 | scruz | make |
| 12:04.16 | scruz | mosty: zaptel: http://pastebin.com/d5f459627 |
| 12:04.27 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:06.00 | mosty | is /usr/src/kernels/2.6.18-92.el5-x86_64 a proper kernel source tree? |
| 12:07.36 | phix | hey, I am still getting IRC timing issues |
| 12:07.40 | phix | IRQ even |
| 12:07.44 | phix | on my TDM card |
| 12:07.53 | scruz | mosty: i'm a windows guy. i've no idea what a 'proper' kernel tree is, but i assume it is. i installed the sources from the CentOS DVD as the server has no internet access as yet |
| 12:08.06 | phix | How can I tell Linux or my mobo to not assign any other device except the TDM on a certain IRQ? |
| 12:08.27 | phix | can I do this with apic? (even though it only work when apic is disabled since by default apic doesnt work) |
| 12:08.35 | scruz | mosty: dahdi-linux: http://pastebin.com/d7f6ce568 |
| 12:08.35 | mosty | scruz, if you change into that directory, can you do "make modules" ? |
| 12:08.51 | scruz | in the source directory? |
| 12:09.20 | mosty | in the kernel dir |
| 12:09.25 | scruz | make modules fails |
| 12:10.02 | scruz | http://pastebin.com/dcd9aeee |
| 12:11.19 | mosty | i suspect that's your problem |
| 12:11.38 | mosty | but i have no experience with centos, so i don't know how their kernel source is setup |
| 12:14.54 | tzafrir_laptop | scruz, no. No need to run 'make modules' there |
| 12:16.22 | scruz | since i already did, can i do 'make distclean'? |
| 12:16.26 | tzafrir_laptop | make -C /lib/modules/2.6.18-92.el5/build ARCH=x86_64 SUBDIRS=/root/software packages/zaptel-1.4.12.1/kernel |
| 12:17.09 | tzafrir_laptop | The problem is the space in the PWD |
| 12:17.18 | scruz | ack |
| 12:17.47 | scruz | thanks |
| 12:18.03 | scruz | moved the folder. will redo and revert |
| 12:18.25 | scruz | it buildeth! |
| 12:18.35 | scruz | hugs tzafrir_laptop |
| 12:19.11 | scruz | i didn't for one moment think that was the problem |
| 12:19.42 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 12:20.02 | scruz | in what order should i build libpri, dahdi and asterisk? it seems i build wanpipe last |
| 12:21.48 | mosty | libpri, dahdi, asterisk, wanpipe (from memory) |
| 12:22.14 | scruz | thanks mosty |
| 12:22.50 | *** join/#asterisk The_LightSide (n=lightie@dsl-241-44-37.telkomadsl.co.za) |
| 12:23.59 | mosty | you can possibly build wanpipe before asterisk, depending on what parts of wanpipe you want to compile |
| 12:25.42 | scruz | i don't actually know what parts i need |
| 12:25.54 | scruz | except i need to get a sangoma E1 card working with asterisk |
| 12:26.05 | k-man | mosty: ok, makes sense, thanks |
| 12:26.08 | mosty | in that case, you can compile wanpipe before asterisk |
| 12:27.39 | mosty | if you use the BRI cards from sangoma, wanpipe compiles an asterisk module, and therefore needs to be built after asterisk |
| 12:28.11 | scruz | i'll compile it after asterisk then |
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| 12:29.05 | scruz | mosty: once again, thanks a lot |
| 12:30.51 | scruz | hmm, make install for dahdi tried to download a file. when i downloaded it on another machine and upacked it, it gives me a .bin file. do i 'chmod +x' it? |
| 12:31.05 | scruz | the tarball is named dahdi-fw-oct6114-064-1.05.01.tar.gz |
| 12:32.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:33.23 | mosty | that file is probably a firmware image- you don't have to run it |
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| 12:34.53 | scruz | ok. i'll have to work on granting that server internet access, as the 'make install' for dahdi failed because of the download |
| 12:35.06 | scruz | be ack in a few. gone to lunch |
| 12:35.11 | scruz | *back |
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| 12:54.37 | scruz | ok...back |
| 12:55.08 | k-man | can anyone recommend a simple audio player for playing emailed voicemail messages in windows? |
| 12:55.08 | k-man | it seems overkill to load windows media player just to hear a 10k wav file |
| 12:55.08 | tzafrir_laptop | vlc? |
| 12:55.10 | k-man | tzafrir_laptop: yeah, might try that |
| 12:55.39 | drmessano | Thats better? |
| 12:56.03 | drmessano | VLC seems like overkill as compared to Windows Media Player |
| 12:56.30 | Gido-E | drmessano ? |
| 12:56.33 | phix | vlc is great |
| 12:56.39 | scruz | i'd recommend sox. create a batch file, drag and drop |
| 12:56.47 | phix | it actually comes with codecs for watching movies :) |
| 12:56.58 | k-man | vlc doesnt seem to like wav49 |
| 12:57.02 | phix | instead of having to download them all the time |
| 12:57.17 | phix | of corse when media player downloads some it only grabs the audio codec and not the video or vise versa |
| 12:57.25 | phix | I hate microsoft |
| 12:57.26 | scruz | phix: so does mplayer |
| 12:57.36 | scruz | and the kmplayer |
| 12:57.37 | k-man | i hate ms too |
| 12:57.40 | phix | scruz: arn't we talking about vlc? |
| 12:57.42 | drmessano | Ok, skipping the fanboyism, you may as well use Media Player, since the core is embedded in the OS anyway |
| 12:57.46 | *** join/#asterisk erth64net (n=erth64ne@96-25-65-141.war.clearwire-wmx.net) |
| 12:57.49 | drmessano | Loading VLC is too much |
| 12:57.50 | Gido-E | is a real Microsoft lover :-) |
| 12:57.50 | scruz | vlc != mplayer |
| 12:58.02 | Gido-E | mplayer is a great hack |
| 12:58.11 | Gido-E | vlc is verry nice programmed |
| 12:58.13 | k-man | i hate mediaplayer |
| 12:58.29 | Gido-E | mplayer != mediaplayer |
| 12:58.35 | phix | :) |
| 12:58.42 | drmessano | Lame |
| 12:59.08 | tzafrir_laptop | Lame is an encoder. Not a player |
| 12:59.16 | drmessano | No, the responses |
| 12:59.26 | phix | Gido-E: that statement is wrong if I do this before hand --> mplayer = null; mediaplayer = null; |
| 12:59.33 | scruz | sox/win32 doesn't come with mp3 support though |
| 12:59.50 | scruz | at least not the binary from sourceforge |
| 13:00.17 | tzafrir_laptop | THose poor win32 folks with their limited system. |
| 13:00.25 | tzafrir_laptop | almost feels sorry for them |
| 13:00.32 | Gido-E | tzafrir_laptop same for me. |
| 13:00.35 | scruz | tzafrir_laptop: why thank you |
| 13:01.44 | drmessano | scruz: Isnt it great how the original question about windows users playing their voicemail will become a windows/linux discussion, and maybe at some point a linux distro war? |
| 13:04.24 | k-man | i guess windows media player isn't so bad if you change the skin to classing |
| 13:04.41 | scruz | drmessano: it's human to act emotional and lump together (perceived) ralted stuff |
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| 13:04.55 | scruz | *related |
| 13:05.14 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:05.46 | k-man | it would be nice if thunderbird could play .wavs directly actually |
| 13:06.55 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 13:10.19 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:10.35 | scruz | [TK]D-Fender: o/ |
| 13:10.54 | [TK]D-Fender | Sombody give this guy a hand! |
| 13:11.44 | stintel | :D |
| 13:12.02 | phix | ok so any way |
| 13:12.07 | phix | yay [TK]D-Fender! you are here |
| 13:12.24 | phix | I want to find out what my phone providers ring / cadence thingy is |
| 13:12.40 | [TK]D-Fender | phix distintive ring is for cheap bastards. |
| 13:12.43 | scruz | [TK]D-Fender: it's a wave :) |
| 13:12.45 | phix | it says in the asteriek info website that this is logged in debug / verbose mode |
| 13:12.46 | [TK]D-Fender | phix: get a metronome :p |
| 13:13.02 | [TK]D-Fender | cues the band |
| 13:13.05 | phix | [TK]D-Fender: a what? well I have distintive ring setup on my phone providers end |
| 13:13.18 | phix | [TK]D-Fender: I also want my ring to be the same as my providers ring any way |
| 13:13.25 | phix | currently it isn't apparantly |
| 13:14.17 | phix | I started asterisk console thingy with -rddddvvvv but it doesnt tell me the int settings for the ring |
| 13:15.07 | phix | I am also getting some echo (I can hear my self) when the FXS and FXO are bridged |
| 13:15.31 | phix | and not to mention IRQ issues still --> http://rafb.net/p/0PWX3711.html |
| 13:21.21 | k-man | is it possible to get asterisk to email you missed calls as well as voicemails? |
| 13:22.27 | mosty | k-man, yes but most phones can display that themselves |
| 13:24.08 | k-man | mosty: but if i want to do it, whats the approach? |
| 13:25.06 | mosty | use System() to send an email, or write an AGI script if you prefer |
| 13:25.29 | k-man | ok - thanks |
| 13:26.03 | k-man | night all |
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| 13:31.59 | vncsnvs | hello! anyone uses QueueMetrics? |
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| 13:34.34 | phix | [TK]D-Fender: I will talk to you later :) night |
| 13:34.41 | Gido-E | :-) |
| 13:34.47 | Gido-E | night it is 2pm here :-) |
| 13:35.18 | [TK]D-Fender | phix: later |
| 13:35.26 | AndyT | anyone have a click2dial script? |
| 13:35.36 | [TK]D-Fender | Gido-E: Night is 6pm everywhere else :) |
| 13:36.09 | [TK]D-Fender | AndyT: http://www.google.ca/search?hl=en&q=asterisk+%22click+to+dial%22+script&btnG=Google+Search&meta= |
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| 13:51.04 | shazaum | yo guys |
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| 13:52.20 | vncsnvs | wich is the best call center app 4 asterisk? (open-source) |
| 13:53.35 | Gido-E | vncsnvs how do you mean? |
| 13:55.03 | ManxPower | There's too much money and the applications are too complex for there to be many Open Source call center applications. |
| 13:55.41 | Gido-E | what is wrong with asterisk? |
| 13:56.03 | ManxPower | phix: you have some hardware issue. |
| 13:56.16 | ManxPower | Gido-E: Asterisk is not a call center application |
| 13:56.41 | ManxPower | Asterisk is a telephony toolkit that lets you build PBXs, dialers, call center, etc. |
| 13:56.58 | vncsnvs | Gido-E, I need a call center application to work with asterisk, open source preference |
| 13:57.52 | ManxPower | vncsnvs: if you cant' find anything open source, you can contact my employer sales@asteriasgi.com for information on my employer's commercial call center application (Assist is the name) that runs on top of Asterisk. |
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| 13:58.30 | Gido-E | vncsnvs /msg vncsnvs for how manny people? |
| 13:58.55 | shazaum | anyone know tell me, where in a call cdr records in the DB? |
| 13:59.04 | vncsnvs | Gido-E, 50+ |
| 13:59.11 | rgsteele||work | Hey folks. If I have an asterisk server behind a natted firewall, and a remote location with several sip phones also behind a natted firewall, how will inbound calls work to the remote phones? E.g., even if the SIP ports are open on the firewall, how will it know which IP to hit behind that firewall? |
| 13:59.41 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-72-72-236-164.altnpa.east.verizon.net) |
| 14:00.01 | rgsteele||work | All the * box will be able to see is the public IP for the remote firewall. And, port forwarding seems hacky, it doesn't scale. |
| 14:00.04 | vncsnvs | shazaum, did not understand |
| 14:00.12 | ManxPower | ~sipnat |
| 14:00.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:00.46 | shazaum | vncsnvs, eu tb nao |
| 14:00.50 | *** join/#asterisk brutuz (n=brutuz@ip67-88-58-242.z58-88-67.customer.algx.net) |
| 14:01.09 | vncsnvs | shazaum, lol! :D |
| 14:01.09 | brutuz | any idea on this? chan_ooh323.c:90: warning: initialization from incompatible pointer type |
| 14:01.36 | tzafrir_laptop | brutuz, is that the first warning? |
| 14:01.47 | brutuz | thats the first warning |
| 14:01.48 | rgsteele||work | ManxPower: Thanks. |
| 14:03.56 | *** join/#asterisk ingenius (n=alektro@host62.190-224-108.telecom.net.ar) |
| 14:04.31 | ManxPower | rgsteele||work: you should only have to port forward on the server NAT, not the client NAT |
| 14:05.00 | brutuz | tzafrir_laptop: any idea how to fix that? |
| 14:09.56 | Katty | brushabrushabrusha |
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| 14:14.46 | Katty | hugs jaytee |
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| 14:26.13 | shazaum | is away: Trabalhando - Working... |
| 14:26.54 | dandre | Hello, |
| 14:27.06 | shazaum | is back (gone 00:00:53) |
| 14:27.08 | *** join/#asterisk kerx (n=kerx@adsl-69-104-17-222.dsl.irvnca.pacbell.net) |
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| 14:27.28 | dandre | hhas anyone trie sangoma B700 hybrid card? |
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| 14:31.25 | shazaum | is away: Trabalhando - Working... |
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| 14:36.20 | rdahlin_1 | is there someone that can help me go get Asteriskgui 2 's timespan to work with asterisk 1.4 ??? I just get error's in the message-log Is the latest versions ofh A-GUI incompatible with Asterisk V1.4 and only men't to be used with Asterisk V1.6 ? |
| 14:36.31 | jaytee | hugs Katty |
| 14:39.11 | *** join/#asterisk ManxPowerAsteria (n=Administ@router.asteriasgi.com) |
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| 14:40.11 | jaytee | ManxPowerAsteria, got a quick question for you if you have a second |
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| 14:41.37 | jaytee | ManxPowerAsteria, on my TE212P card if I'm running both spans as pri_cpe and take timing from my telco should I have both span definitions use a 1 for timing or should I set one of the spans to a 2 instead? |
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| 14:43.09 | ManxPowerAsteria | jaytee: you can't have more than one primary sync source on a Digium card. Use 1 for the first span, 2 for the 2nd span. |
| 14:43.22 | jaytee | ManxPowerAsteria, thanks |
| 14:44.00 | ManxPowerAsteria | I think Sangoma does not have this limitation. However, you very seldom need different sync sources for t-1s plugged into a multi port card |
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| 14:44.27 | ManxPowerAsteria | jaytee: does one of those spans have fax on them? |
| 14:44.44 | jaytee | ManxPowerAsteria, I used to have one span setup as pri_net using 0 going to my Nortel pbx and the other setup as pri_cpe with a 1 going to the telco and had no problems, now I have an occassional HDLC Abort(6) |
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| 14:45.23 | jaytee | ManxPowerAsteria, yes we have some fax devices that will use either span depending on whether they send a fax or recieve one. |
| 14:45.24 | ManxPowerAsteria | HDLC abort is almost always an interrupt/driver/hardware issue. Once in a while line problems can cause it. |
| 14:45.47 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:46.17 | jaytee | I only had 1 event in the last 24 hours for each span at the same time about 1AM this morning. |
| 14:46.27 | ManxPowerAsteria | jaytee: I recommend using whatever span handles the most faxes be the primary sync source. clock slips on voice are hardly noticable, but they can cause issues with faxes. |
| 14:46.39 | ManxPowerAsteria | jaytee: that could easily be the telco doing stuff |
| 14:47.06 | jaytee | ManxPowerAsteria, yeah, I thought maybe they might be doing circuit testing and that could have caused it. |
| 14:47.16 | ManxPowerAsteria | jaytee: that is what I think. |
| 14:47.36 | ManxPowerAsteria | see if zttool shows IRQ misses, if so you should deal with it before it becomes a big issue |
| 14:47.49 | jaytee | ManxPowerAsteria, I have both spans set as 1 for timing at the moment so I'll change one of them to a 2 and do a restart after hours. |
| 14:49.08 | jaytee | ManxPowerAsteria, can zttool be run while the system is in service or should I do it after hours? |
| 14:49.18 | ManxPowerAsteria | jaytee: HDLC Abort means "Got corrupted data on the D-channel, no idea why" |
| 14:50.22 | ManxPowerAsteria | that corruption could be caused by noise or issues on the T-1 loop, it could be some other card or controller locking interrupts for longer than the card can buffer. GigE, RAID, and video can all cause this (usually only the ones built into the motherboard) |
| 14:51.23 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 14:51.26 | eppigy | hello |
| 14:51.28 | eppigy | i am dave |
| 14:51.32 | jaytee | It's a Dell 2950 Quad Core XEON with 2 mirrored SAS drives and 2 gigE nics |
| 14:51.43 | eppigy | nice |
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| 14:53.34 | ManxPowerAsteria | jaytee: sounds like devices on the server may be locking interrupts for a long time to improve performance of those devices. |
| 14:56.34 | jaytee | ManxPowerAsteria, if I run zttool during the day will it cause problems with the spans? |
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| 14:57.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:58.39 | ManxPowerAsteria | jaytee: as long as you don't loop the spans it should be OK. It mostly just shows you the /proc/zaptel stuff in an easy to see form. |
| 14:59.58 | jaytee | ManxPowerAsteria, cool. thanks for your input and advice! always appreciated |
| 15:00.52 | jaytee | hmmm, both spans show no alarms and no missed interrupts |
| 15:00.56 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 15:01.00 | jaytee | must have been the telco |
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| 15:07.46 | shazaum | is back (gone 00:36:20) |
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| 15:12.04 | dandre | has anyone tried sangoma B700 hybrid card? |
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| 15:28.56 | littlerock | I connect avaya SES with asterisk using SIP trunk, I can call avaya's phone from asterisk, howoever I can not call asterisk's phone from avaya |
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| 15:47.34 | mosty | do the calls come in to the asterisk server? |
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| 16:03.45 | *** join/#asterisk Aaron-- (n=Aaron@c-71-63-159-191.hsd1.mn.comcast.net) |
| 16:04.40 | Aaron-- | I'm looking for a good (cheap but reliable) voip provider for a home line that won't get used much. So far vitelity doesn't look too bad. Anyone have thoughts? |
| 16:06.15 | jaytee | Aaron, how's it going? |
| 16:06.25 | jaytee | ~itsplist-us |
| 16:06.32 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:06.40 | Aaron-- | yay, hi John! |
| 16:06.46 | jaytee | hehehe |
| 16:06.55 | Kobaz | Aaron--: voicepulse |
| 16:06.59 | jaytee | Aaron, the Nortel is completely disconnected |
| 16:07.10 | Aaron-- | wow! that's so cool. |
| 16:07.26 | jaytee | yep! |
| 16:07.45 | Kobaz | Aaron--: voicepulse gives you 4 channels per account, which is cool... not sure about other providers |
| 16:07.47 | Aaron-- | what are you going to do with all that extra space now? =) |
| 16:07.51 | GameGamer43 | Aaron: flowroute works great too |
| 16:07.57 | Aaron-- | kobaz: thanks |
| 16:08.05 | thehar | flowroute is awesome |
| 16:08.10 | jaytee | Aaron--, dunno, we'll just leave it for extra headroom |
| 16:08.39 | jaytee | still trying to get Crystal Catering to migrate to Time Warner so we can get rid of that damn Litespan rack |
| 16:09.04 | Aaron-- | you actually did it. got rid of the Nortel completely. |
| 16:09.08 | Aaron-- | you're my hero |
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| 16:09.57 | Assid | anyone here heard of novatelnetworks? |
| 16:10.20 | jaytee | yep, got 2 2950 quad xeons, rysyncing the config files for the dialplans and if one has problems I just swap T1 cables, restart and I'm back up and running in less than 10 minutes |
| 16:10.40 | Aaron-- | nice |
| 16:10.44 | nix8n82 | thanks fcois93 and angryuser |
| 16:10.47 | Aaron-- | so what was the total hardware cost? |
| 16:11.17 | jaytee | Aaron--, ballpark, around 30K including servers, T1 cards, phones and ATA's. |
| 16:11.28 | fcois93 | nix8n82: no problems |
| 16:11.54 | Aaron-- | with all the phones? that's awesome |
| 16:12.00 | riddlebox | jaytee, you could use heartbeat as well and i think there is a t1 box that "splits" to 2 cards for failover |
| 16:12.13 | jaytee | roughly 1/5th of what they spent on the Nortel system total |
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| 16:12.54 | jaytee | riddlebox, I've looked into it and probably will head that way in the future. Trying to pry money out of these assclowns is a tough job, just ask Aaron, we used to work together |
| 16:13.10 | [TK]D-Fender | jaytee: Quick, BURN IT before they have second thoughts! |
| 16:13.16 | Aaron-- | true that. He's a miracle worker for getting the 30k out of them. |
| 16:13.31 | jaytee | [TK]D-Fender, hahaha, though has crossed my mind |
| 16:13.39 | Aaron-- | but they would have given $250k directly to cisco |
| 16:13.44 | Aaron-- | lol |
| 16:13.56 | riddlebox | lol |
| 16:14.13 | riddlebox | heartbeat and rsync are two great tools |
| 16:14.17 | jaytee | Aaron--, yeah and Will kept going back on his word about getting the second and third cards for testing and redundancy until I threatened to just up and quit. |
| 16:14.50 | Aaron-- | jesus |
| 16:14.52 | jaytee | and of course blaming it on Claudia when it was really him not having a pair |
| 16:15.20 | Aaron-- | and whose fault would it be if the card went bad and you didn't have a spare? |
| 16:15.29 | Aaron-- | Claudia's, of course |
| 16:15.30 | Aaron-- | right? |
| 16:15.52 | jaytee | Aaron--, well, mine of course!!! if anything breaks it's automatically my fault. mine and Brent's. :-) |
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| 16:16.16 | jaytee | "This is another fine mess you've gotten us into, Stanley" |
| 16:16.25 | *** part/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 16:16.31 | Aaron-- | "you really threw me under the bus here John" |
| 16:16.47 | jaytee | hahahaaha, I wish I could throw him under a Caterpillar |
| 16:17.27 | riddlebox | where are you guys located? |
| 16:17.39 | jaytee | I'm in Indy, Aaron's in Minneapolis |
| 16:17.48 | Kobaz | Aaron--: heh, i've been looking at vitelity... looks like the nickle and dime you to death |
| 16:17.56 | Assid | anyone heard of novatel networks? |
| 16:18.01 | Assid | wondering how they are |
| 16:18.14 | jaytee | Aaron, did you decide to go with the Linksys SPA 2102? |
| 16:18.23 | Aaron-- | kobaz: thanks for the feedback. I'm looking at flowroute and voicepulse |
| 16:19.06 | Aaron-- | jaytee: I went with an 1001 |
| 16:19.22 | Aaron-- | saved $30, I don't really need more than one port |
| 16:19.42 | jaytee | Aaron--, cool! |
| 16:19.51 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:19.54 | jaytee | Aaron--, did you buy it online? |
| 16:20.02 | Aaron-- | yeah, ebay |
| 16:20.08 | riddlebox | jaytee, not far from me illinois |
| 16:20.21 | jaytee | riddlebox, what part? |
| 16:20.38 | riddlebox | by stl |
| 16:23.03 | riddlebox | we do work for teledata here |
| 16:23.09 | *** part/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 16:23.27 | riddlebox | i want to work for someone who sells asterisk |
| 16:23.31 | *** join/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 16:23.33 | jaytee | so do I |
| 16:23.53 | jaytee | not many opportunities there unless you make your own |
| 16:24.04 | riddlebox | yeah |
| 16:24.16 | riddlebox | i sold 1 on my own |
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| 16:26.02 | jaytee | cool, all the markets are moving upwards today |
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| 16:26.48 | Aaron-- | yay |
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| 16:30.02 | jaytee | yay? |
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| 16:30.23 | macarthy | hello all |
| 16:30.41 | jaytee | oh, on the markets. I wonder if we've seen the bottom or if we haven't even begun to see the really ugly side of this yet. |
| 16:30.45 | *** join/#asterisk qdk (n=qdk@213.173.230.10) |
| 16:31.26 | riddlebox | yeah i sold it to an insurance company who loves it and said every new office will have asterisk |
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| 16:32.52 | Aaron-- | we'll meet the real markets when economists realize there is no american economy and we've been printing money for years with no resources or gnp to back it up |
| 16:33.01 | Shaun222 | i have a screening system, basically a caller comes in, and a sub screen runs on them. Then the caller is put into a queue. After that the queue has local/extensions in it that use dial with another sub that screens the callee with options. I'm having a few problems. First problem is that if sombody answers a call, it continues to ring the other extensions until they choose a option that bridges the call. |
| 16:33.08 | Aaron-- | woops |
| 16:33.10 | jaytee | Aaron--, shhhh! |
| 16:33.22 | jbjuly | what are the most used python asterisk API? |
| 16:33.48 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 16:34.00 | riddlebox | i just wrote my own python app |
| 16:34.16 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 16:34.38 | riddlebox | for mythtv and asterisk to work together |
| 16:34.49 | jaytee | jbjuly, py-Asterisk is one of the most common, but you'd have known that if you tried to Google it. |
| 16:34.50 | macarthy | very newbie question, I want to build a service that people can call a local number for private meetings , I need to support about 100 concurrent user , in several meetings , where should i start looking on the hardware, hosting and software sides? |
| 16:36.29 | Kobaz | adevc: pastebin your pg_hba.conf |
| 16:36.29 | macarthy | what kind of service. product am I looking to buy from the telco for that can of service? |
| 16:36.31 | Kobaz | er |
| 16:36.37 | jaytee | ponders what to have for lunch |
| 16:36.47 | macarthy | *kind of |
| 16:37.32 | Aaron-- | jimmy john's |
| 16:37.36 | Aaron-- | it's delicious |
| 16:37.47 | jaytee | macarthy, are you saying that each "meeting" will have 100 concurrent users? or 100 concurrent users spread amongst multiple meetings. |
| 16:37.59 | jaytee | Aaron--, hehehe |
| 16:38.06 | macarthy | jaytee: situation b |
| 16:38.11 | jaytee | but they never have horseradish sauce |
| 16:38.24 | [TK]D-Fender | macarthy: 4 PRI's , an appropriate card with HWEC, 1 large server w/ 4 gig, RAID5 SAS ought to do. |
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| 16:39.20 | jaytee | macarthy, what [TK]D-Fender said, although if you have the bucks you might split it out into a couple servers. |
| 16:40.29 | macarthy | really new to telco stuff, so what should I be asking the telcos to support a box like that ? |
| 16:41.14 | jaytee | macarthy, 4 T1 spans configured as PRI will give you 96 channels or simultaneous calls. |
| 16:41.15 | macarthy | or can you point me to any telco who have a package /service where I can place a box like that ... |
| 16:41.54 | macarthy | ah ok .. now i'm getting somewhere I understand :-) |
| 16:42.36 | macarthy | jaytee: are there options to host that in a telco, so the T1s aren't needed? |
| 16:43.03 | macarthy | in the cloud as we have to call these days |
| 16:43.11 | jaytee | macarthy, you'd have to check with whatever telcos are in your area, it varies |
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| 16:43.56 | macarthy | jaytee: thanks |
| 16:44.22 | jaytee | Aaron--, hey brother, I'm gonna run to lunch. I'll chat with ya soon, good talking to ya! :-) |
| 16:44.31 | Aaron-- | ok, I'll be around |
| 16:44.37 | jaytee | cool! |
| 16:44.45 | jaytee | bbiab |
| 16:44.54 | jaytee | macarthy, you're welcome |
| 16:44.55 | Aaron-- | later |
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| 16:48.28 | Aaron-- | is there a sip phone that I can put on a usb stick? |
| 16:51.05 | mvanbaak | puts his snom m3 on his usbstick |
| 16:51.17 | mvanbaak | also tries with the grandstream videophone |
| 16:51.28 | mvanbaak | Aaron--: both can be put on a usb stick |
| 16:51.49 | mvanbaak | ;) |
| 16:51.52 | Aaron-- | har har har |
| 16:51.53 | Shaun222 | on a dial(CHAN,30,U(sub,s,1)) what happens in the sub if the callee hangs up? |
| 16:52.09 | Shaun222 | it doesnt look like h, in the sub is running. |
| 16:52.11 | mvanbaak | sorry, couldn't resist |
| 16:52.21 | Shaun222 | looks like it just dumps back to the context that ran dial() |
| 16:52.31 | Aaron-- | =) |
| 16:52.32 | mvanbaak | had a long and difficult day |
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| 17:06.23 | tzafrir_laptop | oh, nice. let's see if you can pass the test: soxexam(7) |
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| 17:13.08 | seanmh | what version of asterisk is asterisk business edition 2.1.1 ? |
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| 17:19.40 | SparFux | I now have the idea to manage my callerID problem. My ISP sets it to the first MSN on my bri, when I use an invalid MSN for outgoing calls. So I have to make asterisk do the same and set it to an MSN of my choice and not to the first, as I don't want this. It should be a simple case statement. |
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| 17:20.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:21.59 | *** join/#asterisk michaely (n=Mike@207.114.199.107) |
| 17:22.09 | michaely | is Asterisk Business Edition C.2.1.1 Asterisk 1.4 or 1.6? |
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| 17:23.51 | putnopvut | michaely: it's based off the 1.4 branch of Asterisk. |
| 17:24.08 | michaely | putnopvut: thanks |
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| 17:29.12 | hardwire | oy oy you lucky people! |
| 17:29.58 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 17:31.58 | *** part/#asterisk michaely (n=Mike@207.114.199.107) |
| 17:33.57 | *** join/#asterisk jozza (n=chatzill@84-255-254-28.static.t-2.net) |
| 17:34.50 | BlargMaN00 | Is there a way that when you dial an extension, you can test to see if it is ringing before actually dialing the extension?? |
| 17:35.06 | jozza | hi, can anyone answer an AMI question? |
| 17:35.49 | jozza | there should be a way actualy |
| 17:36.38 | hardwire | BlargMaN00: explain more |
| 17:37.06 | jozza | there is afunction to check for extension status, isn't there? |
| 17:37.44 | BlargMaN00 | basically, in the dialplan, I want to test to see if an extension is currently ringing before i actually Dial() the extension... |
| 17:37.58 | hardwire | ahha! |
| 17:38.05 | hardwire | Use groups |
| 17:39.06 | hardwire | BlargMaN00: http://www.pastebin.ca/1352810 |
| 17:39.12 | hardwire | I use that in my dialplans and in my queues |
| 17:39.25 | hardwire | I dial Local/0026@ring-once |
| 17:39.43 | hardwire | somebody may be able to refactor that completely. |
| 17:39.48 | BlargMaN00 | lemme check it out real quick... thx |
| 17:40.01 | hardwire | BlargMaN00: all other methods are hit and miss. |
| 17:40.02 | [TK]D-Fender | BlargMaN00: Yes, set a hint on that device, and test for it. |
| 17:40.18 | hardwire | or there's always that. |
| 17:40.32 | *** join/#asterisk KU0N (n=kuon@217.144.50.4) |
| 17:40.33 | hardwire | heh |
| 17:40.45 | BlargMaN00 | what would be the best way to test for the hint?? |
| 17:40.58 | SparFux | Is there a way to span exten lines on multiple lines? |
| 17:41.11 | [TK]D-Fender | BlargMaN00: DEVICE_STATE() , IIRc |
| 17:41.18 | hardwire | SparFux: ? |
| 17:41.27 | [TK]D-Fender | SparFux: No. |
| 17:41.33 | BlargMaN00 | ahhh... |
| 17:41.40 | SparFux | I have a very long line which is simply unreadable. |
| 17:41.43 | hardwire | [TK]D-Fender: does that work for remote extensions and agents? |
| 17:41.57 | BlargMaN00 | excellent... I knew you guys would be good for something... lol 8-)~ |
| 17:42.06 | [TK]D-Fender | hardwire: what is a "remote extension"? got a GPS attached? |
| 17:42.10 | hardwire | yes@ |
| 17:42.17 | [TK]D-Fender | hardwire: then MAYBE |
| 17:42.23 | hardwire | heh |
| 17:42.43 | hardwire | I was thinking more like in my case where I don't want to set a bunch of logic on a remote site where I use asterisk as a local gateway |
| 17:42.47 | hardwire | for simple simple dialplans. |
| 17:42.57 | hardwire | but I can call to it via sip/remote/exten |
| 17:43.14 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:43.14 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:43.22 | hardwire | I use groups for that.. to make sure I don't flood calls from my switch to their switch |
| 17:43.28 | SparFux | I have to do this, but on one line it is a nightmare coding: http://pastebin.com/d10dd75f9 |
| 17:43.34 | [TK]D-Fender | hardwire: Your sample doesn't actually directly answer his request |
| 17:43.47 | hardwire | [TK]D-Fender: it was just an example. |
| 17:43.57 | hardwire | I'm not challenging you mr super samurai |
| 17:44.09 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 17:44.09 | hardwire | ... I haven't forgotten what you called me. |
| 17:44.11 | [TK]D-Fender | hardwire: hard so i should give you an example of some PHP to draw a pie chart for your request, right? ;) |
| 17:44.24 | hardwire | [TK]D-Fender: oh.. oh..! My example is the shiz! |
| 17:44.30 | [TK]D-Fender | hardwire: What... peasant? :p |
| 17:44.38 | hardwire | it directly addresses his issews. |
| 17:44.51 | hardwire | just.. it doesn't use hints |
| 17:44.51 | [TK]D-Fender | hardhe said RINGING. Not "limit to one" |
| 17:45.07 | hardwire | I knew what he *meant*! |
| 17:45.10 | [TK]D-Fender | hardwire: You're cound be in any state. |
| 17:45.10 | hardwire | lul |
| 17:45.14 | [TK]D-Fender | sould* |
| 17:45.17 | [TK]D-Fender | hfdgfaklgfdasd |
| 17:45.19 | hardwire | lets talk this to the judge. |
| 17:45.23 | hardwire | BlargMaN00: ! |
| 17:45.36 | hardwire | admits that [TK]D-Fender is much more literal than I |
| 17:45.43 | hardwire | and therefore better at this crap |
| 17:46.09 | hardwire | BlargMaN00: did you want one call at a time? or only one line ringing at a time? |
| 17:46.38 | jozza | does anyone know anything about ami? i have a question |
| 17:47.11 | [TK]D-Fender | ~ask |
| 17:47.12 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:47.13 | hardwire | [TK]D-Fender does. |
| 17:47.23 | hardwire | [TK]D-Fender has candy in his pockets.. go get him. |
| 17:47.34 | jozza | ok |
| 17:47.49 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 17:48.17 | docid | ok...so heres what im doing, ive got 2 multitech 2410s hooked up to one asterisk box in an attempt to simulate 2 T1s. the dual T1 card in the other asterisk box i have set up to pass through the calls back out the other t1 for now, i get the connection through back to the first asterisk box, but no audio gets through |
| 17:48.38 | rene- | docid: can you setup calls? |
| 17:48.55 | rene- | or you mean you just got green lights on both ends |
| 17:49.31 | [TK]D-Fender | covers hardwire in HP sauce and pushes him into Kisa's cage at jaytee's office |
| 17:49.45 | docid | no, i monitor both asterisk boxes clis and i see the call passed and connected, and it plays what i have setup for it, it just doesnt have any sound on the headset |
| 17:49.54 | docid | handset |
| 17:50.04 | rene- | oh got you |
| 17:50.05 | rene- | hmm |
| 17:50.10 | hardwire | hmm. |
| 17:50.12 | BlargMaN00 | hardwire: sorry, reading up on hints and everything... I'm still learning all this, and making sure I get everything right so that I don't have to bug you guys too much... |
| 17:50.18 | *** join/#asterisk cguerrero (n=cguerrer@200.34.66.137) |
| 17:50.22 | hardwire | BlargMaN00: good man! |
| 17:50.26 | rene- | so you ve got two asterisk boxes? |
| 17:50.28 | *** join/#asterisk NotForResale-US (n=nrds@24.102.131.63.res-cmts.sm.ptd.net) |
| 17:50.34 | docid | yes, |
| 17:50.38 | *** join/#asterisk CRC-error (n=Yoni@85.64.204.4.dynamic.barak-online.net) |
| 17:50.40 | BlargMaN00 | hardwire: I try! |
| 17:50.51 | rene- | how are you linking the asterisk boxes? voip or t1? |
| 17:50.51 | hardwire | jozza: did you PM [TK]D-Fender? |
| 17:51.06 | jozza | no |
| 17:51.07 | docid | basically this is for testing to see if the main asterisk box can pass through most calls and catch the ones ment for internal lines, or voicemail |
| 17:51.09 | [TK]D-Fender | jozza: Not if you know whats good for you :D |
| 17:51.19 | jozza | i didnt |
| 17:51.20 | hardwire | jozza: whats your AMI question? |
| 17:51.31 | rene- | docid: but how are the asterisk systems linked? |
| 17:51.51 | docid | the asterisk boxes arnt exactally linked, one is connected via sip to 2 multitech 2410s, those 2410s talk to th other asterisk box via t1 |
| 17:51.53 | rene- | a T1 crossover? |
| 17:51.54 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 17:51.56 | rene- | ok |
| 17:51.58 | jozza | its about the function ast_manager_register_hook and whats it supposed to hook? |
| 17:52.22 | rene- | id say your issue is on the multitech -first asterisk box side |
| 17:52.42 | docid | im looking at it |
| 17:52.51 | rene- | discard network issues and codec issues |
| 17:52.58 | rene- | are they on the same network? |
| 17:53.18 | docid | are what on the same network? the 2 * boxes? |
| 17:53.28 | rene- | the multitech and the asterisk box is connected to |
| 17:53.39 | docid | they are on seperate subnets, and the whole point of the test is to test t1 passthrough capability |
| 17:53.49 | jozza | hardwire? |
| 17:54.00 | docid | the multitechs are on our voip device subnet, but its all routed correctly |
| 17:54.36 | hardwire | jozza: whats your situation? |
| 17:54.50 | rene- | id say if you can, tackle one problem at a time |
| 17:55.03 | rene- | put both the multitech and the asterisk box on the same subnet |
| 17:55.12 | rene- | to avoid routing, firewalling and nat issues |
| 17:55.21 | jozza | its about the function ast_manager_register_hook and whats it supposed to hook? |
| 17:55.41 | BlargMaN00 | ok........ question on the whole hint thing.... (And yes... I want to support multiple calls... WINNER: [TK]D-Fender) |
| 17:55.42 | rene- | something is wrong there, meaning call signalling usually will work but you might get one way audio or no audio |
| 17:56.08 | Sargun | Will FXS cards from the US work in China? |
| 17:56.37 | BlargMaN00 | If an extension is currently talking on the phone, and another call is ringing into the extension, which devstate will win?? ringing or in use?? |
| 17:57.02 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 17:57.05 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 17:57.36 | hardwire | jozza: it's a hook for a main process to use when loading a module. |
| 17:57.37 | hardwire | afaik |
| 17:57.48 | hardwire | jozza: whats your situation? |
| 17:58.23 | rene- | is local channels still the preferred way to do dynamic member queues? or can one just login the device to the queue |
| 17:58.32 | Greek-Boy | How do I prevent indications from generating tones for a particular outgoing trunk only? So tones come from the telco instead of asterisk... |
| 17:58.33 | jozza | i thought that i could hook to the manager_event somehow and make handle the status messages myself also |
| 17:58.47 | [TK]D-Fender | hardwire: :p |
| 17:58.52 | [TK]D-Fender | VICTORY IS MINE! |
| 17:58.59 | [TK]D-Fender | </stewie> |
| 17:59.00 | hardwire | aww |
| 17:59.35 | jozza | you know, if i wanted to make my own module like csta |
| 17:59.47 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 17:59.57 | hardwire | jozza: I thought there were module skels for people that wanted to get going with the code |
| 18:00.05 | rene- | asterisk 1.6 queue will run a macro on connect so no real need for an local channel context for agents in my situation, so if i could get rid of local channels and clean up redundant information from logs, cdrs and ami that would be sweet |
| 18:00.06 | *** part/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 18:00.08 | hardwire | jozza: whats it got to do wi.. haha |
| 18:00.11 | hardwire | ami.. not ami! |
| 18:00.14 | hardwire | I get it now |
| 18:00.26 | jozza | ok |
| 18:00.52 | hardwire | was just converting some d4,ami to esf,b8zs |
| 18:00.57 | hardwire | I was totally lost |
| 18:01.18 | BlargMaN00 | [TK]D-Fender: So what exactly is the difference between DEVICE_STATE() and Devstate()??? |
| 18:01.19 | jozza | because ami seems to be the center of event reporting, i though it would be the best to start from there |
| 18:01.43 | hardwire | jozza: I'd pop into #asterisk-dev |
| 18:01.46 | hardwire | mebbe |
| 18:01.56 | jozza | ok, let me see |
| 18:02.00 | [TK]D-Fender | BlargMaN00: Renaming, IIRC |
| 18:02.05 | Greek-Boy | so nobody knows the answer to my indications question? |
| 18:02.24 | BlargMaN00 | [TK]D-Fender: So nothing really except the name... excellent... |
| 18:02.27 | hardwire | Greek-Boy: nobody may be working on it? |
| 18:02.45 | [TK]D-Fender | Greek-Boy: if * is generating tones its because progress is passed OOB to * anyways and the telco isn't sending it as audio |
| 18:02.51 | hardwire | Greek-Boy: explain your situation. |
| 18:02.56 | [TK]D-Fender | Greek-Boy: Chicken & egg. |
| 18:03.07 | hardwire | heh |
| 18:03.21 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 18:04.04 | rene- | can anybody shred some light regarding local channels and queues on 1.6? |
| 18:04.39 | hardwire | rene-: with respect to? |
| 18:04.48 | SparFux | What is wrong with this conditional? http://pastebin.com/d33d3b8fa Log says: GotoIf("ALSA/default:1", "0?msn_na:msn_ok") in new stack |
| 18:05.31 | Greek-Boy | hardwire: Its quite simple, I have a PRI to the telco but they have different ringback tones. * generated its in own tone and in most cases the * users dont know if the dialed phone rang or not |
| 18:05.36 | hardwire | rene-: is something failing? |
| 18:06.21 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:06.41 | hardwire | Greek-Boy: I dunno how correct I am in saying this but you should think of asterisk as a telephone network switch that receives tones, interps, and relays. |
| 18:07.00 | hardwire | much like most telephone network switches that end up converting signaling |
| 18:07.21 | hardwire | so the PRI you have with the telco gives you several different indications back for a ring state? |
| 18:07.43 | hardwire | if so.. talk to their manager. |
| 18:07.50 | [TK]D-Fender | Greek-Boy: Only reason for * to generate tones is beacuse it thinks it has to. Like in cases where it answered the incoming channel, or you are forcing ringing. |
| 18:08.18 | hardwire | so I'm off on that one too? |
| 18:08.22 | hardwire | I'm 0/2! |
| 18:08.38 | hardwire | should go to his peasant box and sit and think for a while. |
| 18:09.05 | BlargMaN00 | hardwire: it'll be alright in the morning... lol |
| 18:10.37 | Greek-Boy | [TK]D-Fender and hardwire: I'll look into it more and see if I'm going wrong somewhere |
| 18:11.19 | rene- | hardwire: to if its still recommended to use local channels bounded to devices as dynamic queue members instead of just adding the devices to the queue, |
| 18:11.38 | rene- | taking advantage of the 1.6 queue ability to execute macros on connect |
| 18:12.03 | hardwire | rene-: oh.. neat.. |
| 18:12.11 | hardwire | I didn't know you could even do that iin 1.6 |
| 18:12.19 | rene- | heh |
| 18:12.21 | hardwire | I just use local channels in 1.2/1.4 and it's not a prob bob |
| 18:12.25 | Greek-Boy | [TK]D-Fender: Will Asterisk always generate the tone for ougoing calls on SIP channels? |
| 18:12.34 | rene- | yeah but it tends to add to much stuff to ami and cdr |
| 18:12.43 | hardwire | rene-: agreed |
| 18:12.56 | hardwire | it makes cdr parousers confused. |
| 18:12.59 | [TK]D-Fender | Greek-Boy: * will generate tones for any OOB signal. |
| 18:13.15 | [TK]D-Fender | Greek-Boy: tech agnostic |
| 18:14.01 | rene- | do u think that logging in the devices and using hints would make for reliable queue performance as in not presenting calls to an already busy member and so on |
| 18:14.48 | hardwire | yes |
| 18:15.02 | hardwire | i don't use hints.. I use groups.. because I only want agents handling one call at a time |
| 18:15.10 | hardwire | the phones ring more.. but the agents are happier.. |
| 18:15.46 | hardwire | we went from a situation before I was hired where agents were answering 4 lines at a time.. putting each one on hold.. then coming back to them. |
| 18:15.52 | hardwire | which meant they were making more work for themselves |
| 18:16.09 | hardwire | I disabled that by not letting more than one call per all queues to hit some agents. |
| 18:17.36 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 18:19.11 | hardwire | rene-: I work for a sat tv installer company.. and we had a windy weekend in the area |
| 18:19.22 | *** join/#asterisk uluatu (n=uluatu@200.195.162.210) |
| 18:19.30 | hardwire | a crudload of dishes fell off everybodies houses.. apparently last time this happened our entire dispatch center threatened to quit |
| 18:19.44 | hardwire | but now that it's one call after the other.. nice and orderly.. everybody was super relieved |
| 18:19.49 | rene- | call groups? |
| 18:19.53 | hardwire | yaer |
| 18:19.56 | rene- | those are good too |
| 18:20.08 | hardwire | I meant using group and group counts |
| 18:20.14 | rene- | yes |
| 18:20.27 | hardwire | I don't use call groups because we don't have that diverse of a situation. |
| 18:20.29 | rene- | ups heheh |
| 18:20.41 | rene- | group counts |
| 18:20.44 | rene- | that is the term |
| 18:20.46 | hardwire | yar |
| 18:21.00 | hardwire | anyways.. [TK]D-Fender is eyeing me.. I should go. |
| 18:21.11 | rene- | heh |
| 18:21.15 | [TK]D-Fender | RUN FORREST RUN!!! |
| 18:21.53 | hardwire | bully |
| 18:22.28 | hardwire | I set up some daily reporting of some stuff and now my boss is all hot and bothered and wants every call in the company audited and reported to him via email |
| 18:22.30 | hardwire | so.. bbl |
| 18:23.24 | tzafrir_laptop | hardwire, hmm.... any chance he wants an SMS for every email? |
| 18:23.49 | hardwire | tzafrir_laptop: if he's not smart enough to forward to his phone.. then I quit. |
| 18:24.00 | hardwire | is trying to move along anyways. |
| 18:24.03 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:25.00 | tzafrir_laptop | hardwire, anyway, isn't it exactly what a mixmonitor script is for? |
| 18:25.33 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 18:25.43 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:25.44 | hardwire | tzafrir_laptop: I do that for another company.. super awesome many call recording of doom |
| 18:25.48 | hardwire | I love having 8 gigs of ram |
| 18:26.09 | hardwire | tzafrir_laptop: but no.. this one simply looks at userfields set when people progress some ivr.. and as queues progress from tier to tier |
| 18:26.34 | hardwire | then reports missed calls hourly.. so that our CSR can call back and get people hooked up.. as well as daily reports of all action and what not. |
| 18:26.52 | hi365_m | seem like iaxmodem is causing my iax stuff to get messed up? could it be a port issue? (i.e. bot asterisk and iaxmodem's are set to the same port and the same system) |
| 18:26.54 | hardwire | just using a shell script and mysql html output |
| 18:27.14 | hardwire | I should add my recording solution to the voip-info.org wiki |
| 18:27.17 | hardwire | it's the bees knees |
| 18:29.10 | Daejeo | asterisk handling a lot calls on freesd os http://netmedia.paichai.ac.kr/test2.avi |
| 18:29.13 | hardwire | records to memory then processes each memory stored recording one at a time using the postpone program. uses buffer to slow down the speex encoding to reduce cpu spikes (went from 96% cpu time to 10%) |
| 18:29.34 | hardwire | buffer and postpone are some of my favorite programs.. evar! |
| 18:33.59 | CRC-error | Hi all, I'm new to Asterisk - Tomorrow I have a test of Installing Asterisk system on a server - This is a test for getting into a new job. |
| 18:34.44 | *** join/#asterisk seanmh (n=johndoe@209-193-76-148.mammothnetworks.com) |
| 18:34.48 | CRC-error | So I'm trying to learn some basics before the test... I was wounder what are the major diffarents between Asterisk 1.4 to 1.6? I browsed the Asterisk web site without seeing the diffarents so far... |
| 18:39.04 | [TK]D-Fender | CRC-error: then you're in serious trouble for not knowing the docs are in the source tarball |
| 18:39.42 | [TK]D-Fender | CRC-error: CHANGES & ChangeLog.txt |
| 18:39.56 | [TK]D-Fender | CRC-error: and UPGRADE-* |
| 18:40.45 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 18:41.41 | Aaron-- | so you're trying to fake your way into a job that requires asterisk knowledge? |
| 18:41.53 | Aaron-- | good luck =) |
| 18:42.44 | nullable_type | Hey guys, in dial plan how do i do this? while in an exten line i want to switch to another extension. Just like as a user entered 99 while WaitExten()... do i need to use Dial(Local/99@test)? Is there any better way |
| 18:43.11 | nullable_type | o do u set ${Exten} |
| 18:43.14 | [TK]D-Fender | nullable_type: Goto() |
| 18:43.23 | *** join/#asterisk intralanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 18:43.33 | nullable_type | right! Thanks D-Fender you are the man :) |
| 18:44.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:45.23 | Chuggs | Does anyone know how to get rid of this; "WARNING[15743]: translate.c:645 __ast_register_translator: plc_samples 160 format f" |
| 18:45.28 | CRC-error | Aaron--, No I'm not... I have knowledge in the VOIP world, I have knowledge in SIP & trying to earn some new information about Asterisk :) |
| 18:46.19 | ManxPowerAsteria | CRC-error: read the included docs and the Book |
| 18:46.53 | Aaron-- | good call. |
| 18:47.05 | CRC-error | Thank you :) |
| 18:52.13 | SuPrSluG | what's a good way to get sip debug and agi debug info from the cli? can I enable it in logger and get the sam info? |
| 18:52.28 | SuPrSluG | sam = same |
| 18:55.22 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 19:02.31 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 19:02.55 | deadpigeon | I know this isn't the right channel, hell I don't think there is one. Atleast this is telephone related. Anyone familiar with GR-303 services? |
| 19:03.12 | *** join/#asterisk [netman] (n=netman@10.Red-88-23-116.staticIP.rima-tde.net) |
| 19:05.17 | deadpigeon | Nobody eh? |
| 19:06.24 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 19:06.58 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:14.24 | *** join/#asterisk _BBV_ (n=boris@213.138.71.254) |
| 19:14.57 | SuPrSluG | how to capture cli output? |
| 19:17.21 | _BBV_ | asterisk -x "show chanels" |
| 19:17.55 | _BBV_ | asterisk -rvvvvvx 'iax2 show registry'|grep Registered |wc -l |
| 19:18.01 | [TK]D-Fender | _BBV_: ... |
| 19:18.15 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 19:18.17 | [TK]D-Fender | _BBV_: this isn't the shell you're looking for... |
| 19:18.26 | [TK]D-Fender | pulls the old Jedi Mind Trick |
| 19:18.39 | eppigy | BRILLIANT |
| 19:18.45 | _BBV_ | i show how capture cli result |
| 19:19.16 | [TK]D-Fender | _BBV_: And... how's that working for you? |
| 19:22.43 | russellb | [TK]D-Fender: what's wrong with it? |
| 19:22.54 | *** join/#asterisk FunkyGMT (n=Hayes@74.57.74.26) |
| 19:23.49 | [TK]D-Fender | russellb: Aside from the speeling earors and the over-specific nature? ;) |
| 19:24.15 | [TK]D-Fender | russellb: That isn't "just CLI output", those are target tasks, not a general dump for logging / review |
| 19:24.26 | russellb | Mr. Pedantic! |
| 19:24.49 | russellb | but sure ... i suppose if you wanted to capture cli output for an entire CLI session ... asterisk -r | tee cli_output.txt |
| 19:24.52 | [TK]D-Fender | russellb: it pays off. Constantly. |
| 19:25.03 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 19:25.11 | russellb | the previous is a good example, but only if you want the output for a single command. :-) |
| 19:25.12 | [TK]D-Fender | russellb: I also like my "#" in sip.conf with that |
| 19:25.20 | russellb | heh |
| 19:25.34 | [TK]D-Fender | russellb: and the first sample could use a spelling fix |
| 19:25.57 | russellb | heh, that's the command to show what designer purses asterisk has available |
| 19:26.37 | [TK]D-Fender | #5 Alive! |
| 19:31.50 | hardwire | I usually have to pass asterisk -x 'blah' | strings |
| 19:31.51 | hardwire | hah |
| 19:33.21 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 19:41.38 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:45.34 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 19:45.42 | brunner | does anyone here have NuFone? |
| 19:47.10 | BlargMaN00 | [TK]D-Fender: is there anyway to use variables with hints??? e.g. exten => 16XX,hint,SIP/${EXTEN} |
| 19:49.10 | BlargMaN00 | quit |
| 19:52.13 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 19:52.19 | *** mode/#asterisk [+o Mog] by ChanServ |
| 19:53.18 | *** join/#asterisk DarkRift (n=dark@65.92.166.68) |
| 19:54.56 | [TK]D-Fender | BlargMaN00: in 1.6 (or 1.6.1 Yes, IIRC) |
| 19:58.03 | *** join/#asterisk action_one (n=betatest@41.205.210.146) |
| 19:58.10 | action_one | hi all |
| 19:58.25 | action_one | can someone help me please ? |
| 19:58.31 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 19:59.13 | action_one | a have this message when i receive an h323 call : WARNING[5642] channel.c: No translator path exists for channel type SIP (native 65535) to 0 |
| 19:59.46 | action_one | but when a do the test from openphone that work perfecly |
| 20:00.20 | [TK]D-Fender | action_one: go enable H.323 debug to see what codecs are being offered |
| 20:00.32 | Shaun222 | when i caller hangs up during a sub shouldnt the h extension run? |
| 20:00.33 | [TK]D-Fender | action_one: And look at your SIP peer to see which ones it supports. |
| 20:00.37 | Shaun222 | in that sub |
| 20:00.40 | [TK]D-Fender | action_one: there is clearly a mismatch |
| 20:00.58 | [TK]D-Fender | Shaun222: What is happening? |
| 20:02.15 | Shaun222 | when the callee hangs up right a sub is running it's just exiting that sub, not going to h. |
| 20:02.33 | Shaun222 | basically the queue calls a local/exten which dials a agent using a gosub |
| 20:02.55 | Shaun222 | if the local/exten hangs up while that sub is running the caller exits the queue. |
| 20:03.24 | Shaun222 | from what i can tell it's because the sub just exits and doesnt go to h which would set GOSUB_RESULT=CONTINUE |
| 20:04.11 | BlargMaN00 | [TK]D-Fender: 1.6.1-rc1 |
| 20:04.24 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:05.09 | action_one | <[TK]D-Fender> i activate the debug |
| 20:05.45 | action_one | <[TK]D-Fender> and where can i see the codecs offered ? |
| 20:07.03 | action_one | <[TK]D-Fender> another things i have this issue when i upgraded the freepbx to the new version 2.5 |
| 20:07.50 | lesouvage | When starting my irc client I see the message "your forward and reverse dns don't match". On my sip trunk I miss the incoming audio stream during a phonecall. Can there be a relation and if yes is this my cable company to blame? |
| 20:07.52 | *** join/#asterisk harry__ (n=h@imperialglamour.com) |
| 20:07.54 | action_one | <[TK]D-Fender> i use currently the version 1.2.27 of asterisk |
| 20:08.23 | jaytee | that's ok, I'm still running Windows 95 |
| 20:08.59 | *** join/#asterisk KC42 (n=kevinc@dsl-146-63-58.telkomadsl.co.za) |
| 20:09.10 | harry__ | I'm trying to convert mp3 -> alaw. atm I'm using this sox(1) commando: `sox foo.mp3 -r 8000 -A foo.al` - but at some frequencies of the mp3 file the resulting alaw is kinda broken (you know, distorted, slow). any tricks w/ sox to convert all mp3 files to some alaw that asterisk will play nicely? |
| 20:10.44 | BlargMaN00 | [TK]D-Fender: you there?? |
| 20:12.06 | action_one | how can i verify if my ooh323 module is ok and the channel.c is ok |
| 20:15.06 | Shaun222 | [TK]D-Fender: ahh, figured out a workaround. |
| 20:15.51 | Shaun222 | first thing i do in the sub needs to be to Set(GOSUB_RESULT=CONTINUE) that way if it craps out for any reason asterisk see's CONTINUE and goes on. |
| 20:18.31 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:18.54 | Shaun222 | ok now for problem number two. I have a queue with two weights for agents. The first weight calls 10 phones, and using dial() with a gosub. The problem is that when one person out of those 10 phones picks up a call and goes through the gosub the rest of the phones still ring. But whats weird is as soon as they time out the queue doesnt go on to dial agents from the next weight until that sub exits. I want it to ring all phones from the |
| 20:19.01 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 20:19.51 | *** join/#asterisk rdahlin_2 (n=rdahlin_@81-233-49-160-no58.tbcn.telia.com) |
| 20:20.06 | rdahlin_2 | Hello... |
| 20:20.45 | action_one | hello |
| 20:20.48 | keith4 | Hello |
| 20:21.19 | action_one | please i need somme help with the new ooh323 |
| 20:21.25 | keith4 | we sound like a Metro PCS commercial |
| 20:22.00 | rdahlin_2 | Do you know if there is a uncompability between Asterisk GUI 2 and Asterisk 1.4... i'm thinking about time intervals ans incoming calling rules... |
| 20:22.12 | rdahlin_2 | ans = and |
| 20:22.59 | [TK]D-Fender | action_one: type "help' at CLI and see what H.323 debug options you ahve |
| 20:23.22 | [TK]D-Fender | rdahlin_2: Ask in #asterisk-gui , it isn't supported here |
| 20:23.45 | [TK]D-Fender | BlargMaN00: Yes, and I've answered you on this. Go try and see what happens |
| 20:24.00 | rdahlin_2 | [TK]D-Fender: OK. thanks... |
| 20:24.21 | [TK]D-Fender | harry__: any issue jsut letting * playt he MP3's? |
| 20:26.42 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 20:26.48 | BlargMaN00 | [TK]D-Fender: I tried, and it doesn't seem to work... when I 'core show hints' it just shows 16XX and always says it's idle... |
| 20:26.48 | vncsnvs | any con about 1.6.0.6? |
| 20:27.10 | harry__ | [TK]D-Fender: 95% of my users are on cell phones, so they don't get any quality increase w/ mp3's, so I though I'd spare some CPU on playing straight alaws |
| 20:27.13 | [TK]D-Fender | vncsnvs: unload chan_echo.so |
| 20:27.30 | [TK]D-Fender | harry__: use 8 to convert them and save yourself the effort |
| 20:27.31 | [TK]D-Fender | * |
| 20:27.46 | [TK]D-Fender | BlargMaN00: PASTEBIN is your friend. use it. |
| 20:27.48 | [TK]D-Fender | ~pb |
| 20:27.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 20:28.35 | harry__ | [TK]D-Fender: so, just playing them straight? same with WAV files? how much of a CPU hog will it be.. |
| 20:29.10 | [TK]D-Fender | harry__: I said CONVERT, not transcde live |
| 20:29.35 | harry__ | uh, I didn't know I could use asterisk for that |
| 20:30.30 | *** join/#asterisk t_corr (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:31.09 | t_corr | if i were having problems with call quality using sip, where the audio would cut out for a split second over and over, where would be the best place to look? the rtp? |
| 20:34.26 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 20:40.24 | vncsnvs | ateh logo |
| 20:40.25 | *** part/#asterisk vncsnvs (n=vncsnvs_@189.27.17.115.dynamic.adsl.gvt.net.br) |
| 20:42.21 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 20:42.47 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 20:45.38 | *** join/#asterisk e4 (n=adunlop@rrcs-76-79-48-214.west.biz.rr.com) |
| 20:45.39 | brad_mssw | I've got a phone line coming into Zap/g1 (a digium TDM400P), I want it to make it a dial-out only line, and ignore incoming calls ... is there some flag I can set to tell it not to go to any context? |
| 20:47.30 | [TK]D-Fender | brad_mssw: No. |
| 20:47.41 | [TK]D-Fender | brad you send it into a context that will not answer the call. |
| 20:48.35 | Shaun222 | [TK]D-Fender: anyway to get the ip of the sip connection in the extensions.conf? |
| 20:49.16 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:49.26 | [TK]D-Fender | Shaun222: "core show functions like SIP" <- go read the list a bit |
| 20:49.40 | e4 | I've got an IAX2 configuration that's not connecting properly. It's sending packets out, doing the poke, pong, ack, but no registration is occurring. Any recommendation for a place to start troubleshooting? |
| 20:50.41 | Shaun222 | lol, wth... what greating is the default greating for voicemail... i've set unavailible and busy and it still does the stupid default anounce. |
| 20:51.07 | Shaun222 | i just want one fricken greeting, not a busy, unavailible, taking a crap, etc... :)) |
| 20:52.23 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 20:54.31 | Shaun222 | exten => _X!,1,Set(CALLERID(num)=${SIPCHANINFO(recvip)}) |
| 20:54.32 | Shaun222 | sweet. |
| 20:54.53 | [TK]D-Fender | Shaun222: "core show application voicemail" <- |
| 20:55.56 | Shaun222 | which greeting is it playing by default, the temp? |
| 20:56.50 | [TK]D-Fender | Shaun222: temp overrides the other 2 |
| 20:57.02 | harry__ | [TK]D-Fender: meh, I might be stupid, but all my google results returns something w/ sox(1) |
| 20:58.23 | [TK]D-Fender | harry__: CLI |
| 20:58.59 | Shaun222 | [TK]D-Fender: you see any problems with creating a sip account for guest calls? Basically i created a sip context for username=customer that dumps into my IVR (same context that incomming PRI calls dump too), i set allowguest=yes and no secret. SO far it's working great, not sure how multiple connections on the same context will be handled though. |
| 20:59.26 | Shaun222 | the idea is to allow international customers the ability to connect into our * server and press any key and get dumped into the IVR |
| 20:59.29 | NotForResale-US | can anyone tell me what? y0 b0x is 0wned? is when i boot up? it tells me that cannot find asterisk |
| 20:59.30 | *** join/#asterisk gulden (n=gulden@av2-84-90-24-170.netvisao.pt) |
| 20:59.40 | [TK]D-Fender | Shaun222: Every call is jsut a call. |
| 21:00.24 | [TK]D-Fender | NotForResale-US: means somebody probably hacked your box and raped it both ways, no K-Y |
| 21:00.24 | NotForResale-US | damn |
| 21:00.28 | NotForResale-US | guess i have to reload the backup |
| 21:00.38 | harry__ | [TK]D-Fender: checked that, still no luck. |
| 21:00.40 | eppigy | lol |
| 21:00.51 | eppigy | 0wnz0r3d |
| 21:00.51 | NotForResale-US | i have a 1tb server that backs up my Asterisk box bit by bit |
| 21:00.51 | *** part/#asterisk gulden (n=gulden@av2-84-90-24-170.netvisao.pt) |
| 21:00.54 | harry__ | thinks he is missing something obvious here. |
| 21:01.02 | [TK]D-Fender | harry__: "help convert" |
| 21:01.05 | *** join/#asterisk stbuehler (n=stbuehle@lighttpd/stbuehler) |
| 21:01.34 | harry__ | meh, one module missing. found it. and thank you very much, as always. |
| 21:07.23 | *** join/#asterisk ingenius (n=alektro@host62.190-224-108.telecom.net.ar) |
| 21:09.12 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 21:10.57 | hardwire | never fear! hardwire is here! |
| 21:11.37 | [TK]D-Fender | flees in terror |
| 21:11.46 | hardwire | now that the air is cleared. |
| 21:12.10 | eppigy | allo |
| 21:16.57 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 21:17.40 | e4 | Iptables + selinux is your friend. |
| 21:17.40 | [T]ank | does anyone know of a usb connected ATA? I am looking for a hardware solution similar to magic-jack or http://www.bestbuy.com/site/olspage.jsp?skuId=7790946&type=product&id=1142291023440 |
| 21:18.06 | mvanbaak | [T]ank: what you want to do ? |
| 21:18.37 | *** join/#asterisk double_cheesburg (n=chatzill@ip68-98-36-177.ph.ph.cox.net) |
| 21:18.54 | double_cheesburg | What command tells you the version of dahdi you are running? |
| 21:19.10 | [T]ank | i have built a "work at home agent" solution for my office, but many of these agents that use it are not savvy enough to add a device into their network in addition to their computer. |
| 21:19.25 | [T]ank | they dont have routers or switches for example |
| 21:19.26 | thehar | then teach them. |
| 21:19.32 | thehar | jetblue does it just fine. |
| 21:19.44 | thehar | as do many others. |
| 21:19.51 | [T]ank | i agree.... |
| 21:19.56 | [T]ank | thinking outside the box |
| 21:20.05 | thehar | diagrams printed on papers are best. |
| 21:20.21 | e4 | [T]ank: Sounds like you found a part-time job if they won't/can't/can't be bothered to learn. |
| 21:20.38 | mvanbaak | [T]ank: just give them an usb headset and install a softphone |
| 21:20.41 | mvanbaak | tadaaaa |
| 21:21.05 | [T]ank | ... back to my question... does anything like that exist that is not specific to skype? |
| 21:22.00 | [TK]D-Fender | double_cheesburg: "dahdi_cfg -vvvv" |
| 21:23.39 | mvanbaak | [T]ank: you want a device that is plugged into the usb port and uses the pc's internet connection to connect to a sip server ? |
| 21:23.49 | double_cheesburg | [TK]D-Fender : My output says http://www.pastebin.ca/1352976 I'm running dadhdi-tools version 2.0, but says nothing about Dahdi-Linux. Should I take this to mean I'm running 2.0? |
| 21:24.24 | [TK]D-Fender | double_cheesburg: And the more userland stuff... not sure |
| 21:24.37 | [TK]D-Fender | double_cheesburg: check the "--help" for the given app, etc |
| 21:24.55 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 21:25.25 | hardwire | e4? |
| 21:25.38 | e4 | yes? |
| 21:25.42 | [T]ank | mvanbaak: correct. I am finding a bunch out there made for skype. |
| 21:25.51 | nullable_type | I want to ask the user "Press 1 to continue". which method is more reliable using WaitExten() or Get Data to compare if 1 was entered? |
| 21:26.02 | [TK]D-Fender | [T]ank: jsut a USB headset. No magic here. Loitech makes plenty |
| 21:26.04 | mvanbaak | [T]ank: I am not aware of such a device |
| 21:26.06 | [TK]D-Fender | logitech* |
| 21:26.15 | hardwire | e4: random iptables/selinux plug? |
| 21:26.35 | e4 | @ NotForResale-US |
| 21:26.39 | hardwire | [T]ank: usb headset? |
| 21:26.43 | hardwire | no go? |
| 21:26.57 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 21:27.01 | mvanbaak | hardwire: he doesn't want a softphone |
| 21:27.05 | hardwire | ah |
| 21:27.14 | nullable_type | D-Fender >> Can you look at my question? |
| 21:27.15 | mvanbaak | basically an ATA that connects over usb instead of lan |
| 21:27.18 | hardwire | [T]ank: yer just gonna have to learn em up |
| 21:27.36 | jayrod422 | im trying to compile in cdr_odbc but for some reason asterisk wont build the module.. i have unixodbc and freetds install and can do a isql asterisk.. any ideas? |
| 21:27.40 | [TK]D-Fender | mvanbaak: No such thing that I can think of. |
| 21:27.51 | [TK]D-Fender | mvanbaak: Get him a bloody ATA or SIP phone |
| 21:27.54 | hardwire | [T]ank: if you can find an ATA with a 2 port switch in it.. that may make it easier.. if the remote agents are using router/firewalls. |
| 21:27.54 | [T]ank | this looks promising http://www.amazon.com/VOIP-Phone-Adapter-Support-Skype/dp/B000A4XQR0 |
| 21:27.57 | mvanbaak | indeed |
| 21:28.19 | e4 | mvanbaak: I'm assuming that what he wants involves setting up NAT on the machines in question and using the headset as an ethernet over usb device or something like it. |
| 21:28.27 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 21:28.32 | [TK]D-Fender | [T]ank: Extremely doubtful |
| 21:28.49 | bullium | is there a setting to make the system accept key presses while the system voice is giving options after you first login to voicemail? exp. you call in and have to wait for her to finish talking before it will accept your input...I'm connected in through a T1 card from a definity PBX |
| 21:28.50 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 21:28.56 | [TK]D-Fender | [T]ank: its just a FXS matched SOUND CARD. |
| 21:29.05 | mvanbaak | [T]ank: Just teach them how to use a softphone with an usb headset, or use an ATA or use a sip hardphone |
| 21:29.17 | mvanbaak | eeeeeew, fxs sound cards ....... |
| 21:29.23 | mvanbaak | that's _NOT_ going to work |
| 21:29.37 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 21:29.53 | [TK]D-Fender | bullium: Nowhere should you have to wait |
| 21:30.25 | bullium | the system simply doesn't notice the keypresses until after the system has finished talking |
| 21:30.28 | e4 | [T]ank: An alternative solution is hooking up a router to those offices/lan's and using regular voip phones. That would be cheap and simple. |
| 21:30.28 | hardwire | [T]ank: I deal with some extremely wonky people myself |
| 21:30.55 | hardwire | [T]ank: I usually just forward to their land line through a local PRI. It's a shame I can't fire people who can't deal. |
| 21:30.57 | bullium | [TK]D-Fender: sorry: the system simply doesn't notice the key presses until after the system has finished talking |
| 21:31.16 | [TK]D-Fender | bullium: Sorrier still that I see no debug and am out of time. |
| 21:31.17 | hardwire | [T]ank: then I make sure they press an ack key before I transfer the call |
| 21:31.19 | [TK]D-Fender | Heading home, BBIAB |
| 21:32.03 | hardwire | bullium: using Background or Playback? |
| 21:32.34 | hardwire | ^^ http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Background |
| 21:32.37 | bullium | hardwire: I'm not sure I understand your question |
| 21:33.00 | bullium | hardwire: I'll look at the link |
| 21:33.01 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:34.29 | *** join/#asterisk StanManCan (n=stan_man@S010600195b3059b4.gv.shawcable.net) |
| 21:34.43 | StanManCan | How do you setup your outgoing callerID number in Asterisk ? |
| 21:35.02 | hardwire | bullium: ah.. that may not be what you need |
| 21:35.32 | bullium | hardwire: OK, then what do I need :) |
| 21:36.32 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:37.08 | BlargMaN00 | [TK]D-Fender: OK.. back to grinding wheel... You want me to put that output from 'core shoe hints' on pastebin?? |
| 21:37.14 | BlargMaN00 | crap... he left |
| 21:41.15 | *** join/#asterisk GameGamer43 (n=GameGame@nat/digium/x-9dcc18159de973d4) |
| 21:41.22 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 21:41.46 | BlargMaN00 | can anyone else help me out with my hint issue? |
| 21:44.18 | mvanbaak | hey lesouvage |
| 21:44.31 | mvanbaak | you in hannover for the cebit ? |
| 21:45.58 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 21:47.38 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 21:47.54 | jplank | anyone have any experience with any multi tenant guis for *? |
| 21:48.24 | mvanbaak | jplank: they all suck++ |
| 21:49.28 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 21:49.38 | jplank | really? |
| 21:49.44 | *** join/#asterisk ingenius (n=alektro@host62.190-224-108.telecom.net.ar) |
| 21:49.56 | nullable_type | exten => s,1,WaitExten() ; Wait for 9 to be entered. |
| 21:49.56 | nullable_type | exten => 9,1,DoSomethingWith9() |
| 21:49.56 | nullable_type | exten => [012345678],1,DoSomethingWithOtherNumbers() |
| 21:49.56 | nullable_type | Do you guys know why i get a warning when a # like 7 is entered? I was hoping it will go to Line 3 |
| 21:50.23 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:51.32 | jplank | one of our uses for asterisk today is as a hosted PBX. We currently do everything manually (the best way to go), but my boss wanted me to look into some gui setups, so he can have non-tech people do adds moves and changes and whatever |
| 21:51.53 | Corydon76-dig | nullable_type: you forgot to prefix the pattern with a '_' ? |
| 21:52.14 | lesouvage | mvanbaak: unfortunately not, you are? |
| 21:52.15 | nullable_type | oh ya u mean like _[0123] |
| 21:52.28 | Corydon76-dig | nullable_type: correct |
| 21:52.36 | nullable_type | oh thanks! |
| 21:52.37 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:52.39 | *** join/#asterisk intralanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 21:53.02 | nullable_type | Cory >> but if i had a wildcard pattern and a exact match pattern both with priority 1. Which will it go to |
| 21:53.03 | t_corr | Hmm. After a Dial() in extensions.conf, should I need to test to see what the results were? (Busy, etc) and play the appropriate tone? |
| 21:53.21 | mvanbaak | lesouvage: nope. not going there this year |
| 21:53.30 | mvanbaak | lesouvage: I have real work to do |
| 21:53.53 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 21:53.58 | lesouvage | jplank: scopserv is, within the limitation of a gui, a good working solution, but it is licensed . |
| 21:54.30 | lesouvage | mvanbaak: enjoying the city? |
| 21:54.58 | [TK]D-Fender | t_corr: ${DIALSTATUS} |
| 21:55.14 | *** join/#asterisk Non-ICE (n=non-ice@ti231120a080-0479.bb.online.no) |
| 21:56.06 | t_corr | Should I have to do a Playtones, though? |
| 21:56.46 | t_corr | Or maybe a better question would be: When Dial()ing out, when no one picks up in the limit (currently 60s), what's the best thing to do? |
| 21:56.56 | mvanbaak | lesouvage: yeah, a LOT |
| 21:56.58 | t_corr | right now i just Hangup, but that confuses my users |
| 21:57.29 | Non-ICE | i'm trying to get asterisk running to my sip provoider using only static NAT mappings.... only thing i'm having trouble getting to work is audio in.... having nated 1024-1036, 3478-3479,5060-5070,10000-10007,16384-32767,48000-64000, still no luck.... |
| 21:57.46 | [TK]D-Fender | t_corr: Playback(slackers_arent_answering) |
| 21:57.51 | t_corr | lol |
| 21:58.04 | t_corr | << is rwaite, in case anyone cares |
| 21:58.09 | t_corr | or remembers me. |
| 21:59.38 | Non-ICE | is recieveing audiostreams totally unpredictable? |
| 21:59.42 | mvanbaak | Playback(tt-weasels) |
| 21:59.48 | Non-ICE | regarding to tcp/ip ports? |
| 22:00.43 | t_corr | or i think maybe i will just not set a limit, if they want to sit there for 5 minutes, let them, and then they can hang up when they're ready |
| 22:01.34 | BlargMaN00 | [TK]D-Fender: ok, so i'm not making any headway on this whole hint thing... I have a wildcard dialplan setup (_16XX) and I can't get the exten => 16XX,hint,SIP/${EXTEN} to work... any ideas?? |
| 22:02.19 | [TK]D-Fender | BlargMaN00: pastebin "dialplan show" and your subscription attempt |
| 22:02.35 | ManxPowerAsteria | you cant's wildcard hints last I heard. |
| 22:02.44 | stbuehler | hi, i am trying to get capi working via misdn (avmfritz pnp, 2.6.28.7, patched with std2kern + some extra work); the output of misdnportinfo looks fine (one Port listed) |
| 22:02.46 | ManxPowerAsteria | that may have changed in 1.6 |
| 22:02.48 | stbuehler | but capiinfo still shows "capi not installed - No such device or address (6)" (capitutils 1:3.9.20060704-3.6 from debian) |
| 22:03.34 | t_corr | hmm i just noticed that i am using 'r' in my Dial() |
| 22:03.43 | t_corr | maybe that is why i am running into these problems |
| 22:04.57 | BlargMaN00 | [TK]D-Fender: ok... I highlighted the pertinent parts... http://pastebin.com/d741e5fcf |
| 22:05.58 | BlargMaN00 | [TK]D-Fender: how do i show subscription attempts?? |
| 22:06.31 | [TK]D-Fender | BlargMaN00: enable SIP DEBUG and watch the subscription attempt |
| 22:07.04 | t_corr | OK, if I remove the 'r' option from Dial() it seems like the channel is silent until I see an 'Is Ringing' on the console |
| 22:07.23 | t_corr | ponders which one would be better |
| 22:10.17 | nullable_type | Does Dialplan has commenting syntax other than ; ? Like /****/ |
| 22:10.19 | *** join/#asterisk crazyx__ (n=crazyx@41.249.253.247) |
| 22:11.22 | BlargMaN00 | [TK]D-Fender: ok... I think I got everything you asked for in there... http://pastebin.com/d783008d8 |
| 22:13.22 | nullable_type | D-Fender ==> Dialplan supports comments with syntax /****/ instead of ; ? |
| 22:13.42 | russellb | it does not. |
| 22:13.57 | russellb | multi-line comments are done using ;-- some stuff \n more stuff \n whatever --; |
| 22:14.00 | Corydon76-dig | No, if you want multiline comments, use ";--" and "--;" to end |
| 22:14.17 | thehar | russellb beat you |
| 22:14.29 | Corydon76-dig | I think my explanation was more clear |
| 22:14.35 | thehar | haha |
| 22:15.09 | t_corr | wouldn't harhar be more appropriate |
| 22:15.16 | thehar | no |
| 22:15.18 | russellb | your statement claimed that both ";--" and "--;" were used to end |
| 22:15.25 | thehar | gets popcorn for the fight |
| 22:15.30 | russellb | that's not very clear :-p |
| 22:15.45 | Corydon76-dig | thehar sounds much like "the whore" |
| 22:15.59 | Corydon76-dig | intentionally |
| 22:16.13 | thehar | tilghman: learn to be nice. |
| 22:16.18 | thehar | haha |
| 22:16.18 | t_corr | the tortoise and the whore? |
| 22:16.20 | Corydon76-dig | Well? |
| 22:16.24 | thehar | My name is Harley. |
| 22:16.29 | thehar | my nickname is the har |
| 22:16.32 | thehar | or harles |
| 22:16.38 | thehar | i don't like harles. so it's thehar |
| 22:16.39 | t_corr | harles in charges? |
| 22:16.41 | thehar | haha yes |
| 22:16.47 | t_corr | heh |
| 22:20.42 | crazyx__ | hello everybody. please how can i set on asterisk the Q.931 TEI retry timer ? thanks by advance |
| 22:21.56 | StanManCan | Anybody have any information on changing the outgoing callerID number? |
| 22:22.04 | *** join/#asterisk mphill (n=mphill@174.37.19.92-static.reverse.softlayer.com) |
| 22:22.36 | StanManCan | I have multiple phone numbers and owuld like to be bale to control which one it appears I'm calling from |
| 22:22.50 | StanManCan | business 1's line.. business 2's line.. |
| 22:24.04 | *** join/#asterisk davevg (n=davevg__@nj-67-76-177-147.sta.embarqhsd.net) |
| 22:26.08 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 22:29.13 | crazyx__ | please is there somebody know how can i set on asterisk the Q.931 TEI retry timer ? thanks by advance ! |
| 22:29.58 | ManxPowerAsteria | StanManCan: change it in the dialplan |
| 22:33.24 | *** part/#asterisk harry__ (n=h@imperialglamour.com) |
| 22:38.48 | nullable_type | Is there a way i can get call duration in the DialPlan? without looking at CDR Database? |
| 22:39.16 | *** join/#asterisk dwery1 (n=dwery@nslu2-linux/dwery) |
| 22:39.55 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 22:40.25 | dwery1 | hello. I'm trying to compile wanpipe 3.5.0.27 but got an error: 'struct device' has no member named 'priv' . anyone knows if there's a patch? |
| 22:41.21 | dwery | s/struct device/struct netdevice/ |
| 22:41.22 | codefreeze-lap | nullable_type: how about CDR(duration) ? ${CDR(duration)} |
| 22:41.58 | codefreeze-lap | It'd only be useful in the h exten |
| 22:42.08 | nullable_type | thank you so much, I was just looking at wiki about that |
| 22:42.12 | codefreeze-lap | If you set endbeforehexten |
| 22:42.35 | codefreeze-lap | And in xfers, parks, etc, you can kiss it goodbye |
| 22:42.53 | nullable_type | yes, But if i was trying to bridge calls, CDR(duration) gives the last call duration rite? |
| 22:43.22 | codefreeze-lap | It gives you whatever is sitting in the CDR on the current channel |
| 22:43.49 | nullable_type | ohh ok thanks |
| 22:44.20 | *** join/#asterisk ^Bloo (n=who@cuervo.unwiredbuyer.com) |
| 22:44.21 | codefreeze-lap | DIAL sets some vars; see the docs for the Dial() app |
| 22:44.36 | codefreeze-lap | nullable_type: That might be easier |
| 22:45.20 | codefreeze-lap | You might be able to fetch that info after dial returns, nullable_type |
| 22:45.23 | nullable_type | From what i read i can set the vars there but to find the actual duration i use CDR(duration) right |
| 22:45.33 | nullable_type | how can i fetch them back? |
| 22:45.43 | nullable_type | oh u mean the vars that was set, nm |
| 22:46.46 | nullable_type | ${ANSWEREDTIME} |
| 22:46.46 | *** join/#asterisk itakinet (n=chatzill@adsl-065-005-186-231.sip.asm.bellsouth.net) |
| 22:46.49 | nullable_type | that one? |
| 22:48.25 | codefreeze-lap | nullable_type: yeah, just checked the code, that and DIALEDTIME |
| 22:48.45 | nullable_type | thanks |
| 22:48.55 | codefreeze-lap | same diff as billsec and duration |
| 22:49.28 | nullable_type | Is there any example app on how can i bill if i am briding two channels? |
| 22:49.32 | nullable_type | example DialPlan |
| 22:49.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:50.48 | kb3ien | anyone seen the linkydink PAP2t (NA version) work fine for outbound, but loose audio in both directions for inbound? |
| 22:50.55 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:52.41 | kb3ien | seems to be a history of people writting about this particular product doing this, but no solutions. |
| 22:55.30 | crazyx__ | if i get my E1 card (TE410P) working well with no problems with span1,1,0... and span2,2,0... , but when i plug the third one with span3,3,0... i got errors, what's the better way to stop having this issues ? span3,0,0 ?? |
| 23:02.57 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 23:03.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:03.25 | joako | would their be a problem to have asterisk listen on multiple ports on the same ip address? e.g. 5060-5069? |
| 23:08.13 | edoceo | What's the terms to Google so * understands when users 'Say One' rather than 'Press One' ? |
| 23:09.00 | nullable_type | Hey guys, Can i set multiple variables using exten => s,1,Set(Var1=0,Var2=0) ? |
| 23:10.04 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 23:10.23 | edoceo | nullable_type: I always have multiple lines - for readabilty |
| 23:10.31 | edoceo | s,n,Set(Var=foo) |
| 23:10.38 | edoceo | s,n,Set(Var=bar) |
| 23:10.52 | nullable_type | ohh |
| 23:11.23 | nullable_type | thanks guys |
| 23:11.56 | dwery | anyone is using the wanpipe drivers with a recent kernel? (2.6.28/29) |
| 23:22.55 | *** join/#asterisk Aaron-- (n=Aaron@c-71-63-159-191.hsd1.mn.comcast.net) |
| 23:23.24 | Aaron-- | is there speech-to-text magic I can use to give people an email transcription of their voicemail? |
| 23:24.23 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 23:25.14 | *** part/#asterisk ManxPowerAsteria (n=Administ@router.asteriasgi.com) |
| 23:26.55 | joako | edoceo: LumenVox? |
| 23:27.44 | ^Bloo | S2T of freeform text is gonna be garbage, tho.. have fun. |
| 23:28.48 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 23:31.08 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-196-188.lns10.mel6.internode.on.net) |
| 23:32.59 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 23:33.35 | *** join/#asterisk SkykingOH (n=skyking@207.58.236.226) |
| 23:34.03 | SkykingOH | Having a bit of trouble with some extension syntax |
| 23:34.14 | SkykingOH | exten => _X!,n,GotoIf($["${CALLERID(num):0:2}" != ",1"]?nodigitstrip) |
| 23:34.26 | SkykingOH | Is evaluating false and jumping to nodigitstrip |
| 23:34.28 | jaytee | Aaron, speech to text? |
| 23:34.46 | Daejeo | Aaron: he meant- tts |
| 23:34.59 | jaytee | no, I think he meant the other way |
| 23:35.09 | Daejeo | speeh to text- ASR |
| 23:35.47 | jaytee | you'd have a hell of a time getting Lumenvox to handle that |
| 23:35.56 | SkykingOH | The caller ID is beginning with the ,1 I want to strip tested with a noop - any assistance appreciated |
| 23:36.01 | *** join/#asterisk DarkRift (n=dark@65.92.166.68) |
| 23:36.25 | *** join/#asterisk ingenius (n=alektro@host62.190-224-108.telecom.net.ar) |
| 23:37.22 | Daejeo | Aaron: you can use voicexml browser |
| 23:38.21 | jaytee | for real speech to text you need something like Dragon Dictate which I don't think they have a linux version and it requires training the engine so random voices would throw it off. |
| 23:38.39 | Aaron-- | hmm |
| 23:39.29 | Aaron-- | I meant speech to text, like simulscribe |
| 23:39.30 | jaytee | and Lumenvox works on keyed grammars so it recognizes random voices but the grammar has to be built, it ain't free and the 12,000 vocabulary engine license per port is a bit daunting in price |
| 23:39.35 | hardwire | dwery: checked with sangoma? |
| 23:39.48 | jaytee | Aaron--, never tried simulscribe |
| 23:40.06 | dwery | hardwire: haven't called them yet. I managed to compile the drivers bt it seems I'm having problems with the utilities |
| 23:40.19 | Aaron-- | I've got it on my cell phone. voicemail forwards to them, then they email me a transcription of the voicemail, and a wav |
| 23:40.31 | hardwire | dwery: what kind of issues? |
| 23:40.44 | ^Bloo | even the very best ASR engines cannot do freeform (ie grammarless, untrained) speech recognition well. Especially over a phone-quality audio. |
| 23:41.01 | jaytee | ^Bloo, very true |
| 23:41.12 | ^Bloo | it's like 60% recognition or some such |
| 23:41.12 | dwery | hardwire: the ./Setup dahdi compilation script failed when building the utilities, so I'm now trying to understand how to do the same things manually. |
| 23:41.16 | Aaron-- | so there's no easy way to do it. works for me. |
| 23:41.22 | ^Bloo | yeh, sorry |
| 23:41.36 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 23:41.37 | ^Bloo | the services that do it actually use slave-labor-rate callcenters |
| 23:41.58 | ^Bloo | they pre-process it with asr, but they touch all of the messages |
| 23:42.29 | Daejeo | ^Bloo: try my ASR |
| 23:42.31 | hardwire | dwery: did you just change the kernel? |
| 23:42.37 | Daejeo | it does work |
| 23:42.48 | Aaron-- | I guess I could forward it all to spinvox if I really cared |
| 23:42.55 | dwery | hardwire: no, first compilation, I've just received my sub fxo device ad my kernel is .29-rc6 |
| 23:43.18 | hardwire | ah ok.. so the kernel part of it seems moot. |
| 23:43.29 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 23:43.32 | dwery | hardwire: But I managed to compile the kernel part |
| 23:43.37 | hardwire | I'd report your dahdi version, asterisk version, kernel version, to sangoma |
| 23:44.05 | hardwire | wanpipe is kinda a naughty word around here. |
| 23:44.06 | hardwire | tee hee |
| 23:44.26 | hardwire | well.. probably not |
| 23:44.31 | hardwire | but I like chaos.. so lets pretend. |
| 23:44.39 | dwery | give me another usb fxo device and I will be happy to use it :) I'll try to contact them tomorrow.. |
| 23:45.01 | jpcansa | how can i make dahdi to start on boot on opensuse? |
| 23:46.00 | hardwire | jpcansa: dahdi-tools-2.1.0.2/dahdi.init |
| 23:46.17 | dwery | hardwire: I think I ust need to generate the configuration files now |
| 23:47.45 | Aaron-- | daejeo: thanks, I'll take a look at voicexml |
| 23:53.17 | jaytee | brb |
| 23:55.47 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
