00:00.51 | beek | Man, is my Perl rusty. |
00:01.22 | Katty | leifmadsen: http://www.dietandfitnesstoday.com/rda.php |
00:01.32 | Katty | leifmadsen: those numbers are very acurate. |
00:01.43 | Katty | leifmadsen: http://42ndhealthstreet.blogspot.com/2009/10/nutrient-recommendations-what-we-need.html |
00:02.14 | Katty | leifmadsen: step one of my research is done! |
00:02.17 | *** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net) |
00:02.32 | Katty | leifmadsen: step two is to keep track of what i eat tomorrow so i can see how much of what i ate. |
00:03.09 | Katty | leifmadsen: and that will be very enlightening |
00:03.18 | leifmadsen | Katty: ya, I need to keep a food log again |
00:04.04 | geneticx | hiya |
00:04.10 | Katty | leifmadsen: ever see anything called Ascorbic Acid in on an ingredient list? |
00:04.38 | leifmadsen | Katty: yes, I have seen that... let me see what that was again |
00:04.47 | Katty | leifmadsen: its a different form or alternative name for vitaminc c |
00:04.59 | leifmadsen | ah yes, I knew it was like citric acid |
00:05.44 | Katty | leifmadsen: and if you are deficient in vitamin c, you will have bleeding gums, painful joints, slow healing wounds, bruising (i have this), nose bleeds, tooth decay, loss of appetite, muscular weakness, skin hemorrahages, capillary weakness, anemai, and impaired digestion |
00:05.58 | Katty | Vitaminc C is ascorbic acid. |
00:06.15 | Katty | idk if citric acid is quite the samel. |
00:06.16 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
00:06.20 | Katty | tho citric acid probably contains vitamin c. |
00:06.30 | Katty | let's ask wikipedia! |
00:06.38 | leifmadsen | Katty: heh, that's what I did :D |
00:07.02 | Katty | jit's a food additive used for flavoring and preservation |
00:07.04 | Katty | :< |
00:07.15 | leifmadsen | I need orange juice now |
00:07.43 | *** join/#asterisk kissg_hu (n=kissg_hu@BC2492DA.dsl.pool.telekom.hu) |
00:07.48 | Katty | yes. |
00:08.05 | leifmadsen | tasty! |
00:08.09 | Katty | yes! |
00:08.11 | kissg_hu | hello there |
00:08.17 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
00:08.35 | Katty | hmm, neat. |
00:08.42 | Katty | I kept seeing Riboflavin on packages of pasta. |
00:08.47 | Katty | apparently that's vitamin b2 |
00:08.53 | Katty | niacin is b3 |
00:09.12 | Katty | i'm gettin smrt! |
00:09.23 | leifmadsen | hawt! |
00:09.40 | leifmadsen | maybe you will eventually go to school for chemistry |
00:09.52 | Katty | now this makes me wonder tho |
00:09.55 | Katty | cause salt... |
00:10.00 | Katty | you would say is sodium, right? |
00:10.16 | kissg_hu | I'd like to ask for your assistance regarding dial rules in Asterisk 1.4 |
00:10.16 | Katty | except it's sodium chloride :/ |
00:10.23 | leifmadsen | yes :) |
00:10.30 | kissg_hu | is anyone there who could help me? |
00:10.31 | leifmadsen | kissg_hu: ask away |
00:10.32 | Katty | so i wonder how you figure out the nutritional value of Salt. |
00:10.43 | leifmadsen | Katty: there isn't really a nutritional value to it |
00:10.51 | leifmadsen | Katty: lower is typically better |
00:11.17 | leifmadsen | salt causes dehydration and you to carry excess water (which seems contradictory) |
00:11.25 | Katty | it also raises blood pressure |
00:11.35 | Katty | oooh wait |
00:11.36 | leifmadsen | but kinda makes sense, since your body is trying to retain water since it is dehydrating |
00:11.39 | Katty | it has a * by it |
00:11.42 | kissg_hu | I've created a dial rule, but I'm not sure it really works the way it should |
00:11.51 | Katty | (Note: nutrients with a star indicate Adequate Intake or AI because no RDA can be established) |
00:12.21 | leifmadsen | kissg_hu: use a pastebin and show the dialplan you've created, and the console output, and explain what you expect to happen, and what is actually happening |
00:12.35 | Katty | leifmadsen: drink more OJ. |
00:12.41 | Katty | leifmadsen: and eat a tomato |
00:12.44 | leifmadsen | Katty: I try to use sea salt when I want salt for tasting |
00:12.48 | kissg_hu | leifmadsen: thanks, but it's simple and in fact, I have no problem with it :) |
00:12.51 | Katty | leifmadsen: vitamin c decreases the risk of certain cancers by 75% |
00:12.56 | leifmadsen | Katty: I love tomatoes, and am drinking OJ now -- yesterday I drank a V8 |
00:13.04 | kissg_hu | I just would like to be sure it really works the way I expect |
00:13.14 | leifmadsen | ~ask |
00:13.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
00:13.50 | Katty | leifmadsen: good news! cayenne pepper is high in vitamin c ^_- |
00:13.59 | Katty | leifmadsen: EAT MOAR HOT SAUCE |
00:14.02 | kissg_hu | First, spoken out: "Remove 06 prefix from number add 0036 instead. the next one or two numbers are local prefixes" |
00:14.10 | leifmadsen | Katty: yes, cayenne pepper is very good for you |
00:14.18 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
00:14.24 | kissg_hu | "Then, 6 or 7 numbers which are subscriber numbers" |
00:14.26 | Katty | oooh and avocado |
00:14.29 | Katty | i had some of that for dinner. |
00:14.33 | leifmadsen | 0036${EXTEN:2} |
00:14.40 | kissg_hu | the rule I created is: "06|0036+[1-99]XXXXXX" |
00:14.43 | leifmadsen | Katty: yes, avocado's have the "good" fats in them |
00:15.00 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
00:15.01 | leifmadsen | Katty: substitute for mayo on sandwiches |
00:15.10 | Katty | mayo is pretty icky |
00:15.10 | leifmadsen | Katty: mustard is also ok to have as well |
00:15.11 | kissg_hu | and this one for numbers where 7 digits identifies the subscriber: "06|0036+[1-99]XXXXXXX" |
00:15.15 | leifmadsen | Katty: ah, I love mayo :) |
00:15.20 | kissg_hu | are these okay? |
00:15.20 | Katty | it's okay with chicken salad sammiches |
00:15.29 | Katty | but have you SEEN the ingredient lists? |
00:15.30 | leifmadsen | Katty: I use it in tunafish |
00:15.36 | leifmadsen | Katty: oh, I know how bad it is :) |
00:15.38 | leifmadsen | I just like the taste |
00:15.46 | Katty | helman's real seems to be Best For You |
00:15.51 | leifmadsen | ok, off to play some flight simulator on my new 24" monitor :) reboot to windows time! |
00:15.51 | Katty | considering it's a processed food item :/ |
00:15.56 | Katty | byebye |
00:37.13 | Knightfal | hey guys is dialparties.agi specific to any particular flavor of an asterisk build out there? |
00:39.39 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
00:42.26 | *** join/#asterisk Knightfal (n=Knightfa@75.142.144.171) |
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00:58.47 | *** join/#asterisk QaDeS (n=mklaus@p4FC72958.dip0.t-ipconnect.de) |
01:01.37 | Katty | returns |
01:02.46 | Katty | pokes about for life |
01:07.26 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
01:07.31 | manxpower | Knightfal: dialedparties.agi is not part of the official Asterisk. |
01:07.55 | manxpower | It is part of FreePBX GUI |
01:10.28 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
01:12.37 | CcRnp | anyone used asterisk CEL in production ? |
01:12.51 | CcRnp | anyone used asterisk CEL in production server? |
01:26.33 | *** join/#asterisk knctrnl (n=aembrey@user-69-1-12-224.knology.net) |
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01:54.03 | knctrnl | anyone have any tips on areas to focus on preparing for DCAP? |
01:55.06 | jaytee | yeah |
01:58.04 | jaytee | be able to setup a network with an Digium analog card with 1 FXO and 1FXS, 1 Polycom phone and X-Lite with dialplan, VM and an IVR in 90 minutes for the lab, for the written test, read "the book" cover to cover and study about protocols, codecs, etc. |
01:58.36 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
01:58.51 | jaytee | and use CentOS so you're familiar with the lab distro |
01:59.03 | jaytee | at least that's what they're using in Huntsville |
02:02.51 | knctrnl | Im taking it friday |
02:03.08 | knctrnl | currently taking the advanced class in huntsville |
02:03.59 | knctrnl | Do you know how complex the IVR will be? |
02:06.57 | jaytee | had a GotoIfTime in it and a simple Press 1 to go to Bob type of menu. Do a search of sample or example dialplans on voip-info.org and you'll get some stuff that's fairly similar. |
02:07.59 | knctrnl | thats not too bad |
02:08.01 | jaytee | if they give you the option of using the T1 card or not, pass on it. it'll save you time and let you focus on the dialplan stuff and configs. |
02:10.02 | jaytee | cuz you'll need to get your 3 sip peers configured so you'll have a DAHDI analog client, a SIP client for the Polycom and a SIP client for the X-Lite softphone on the server with VM boxes and the IVR etc. If you find decent examples from googling cuz you're allowed to use the Web during the lab then you can cut and paste for your configs but you'll still end up with alot of editing. |
02:10.24 | jaytee | I ran out of time. passed the written and got 50% on the lab |
02:11.25 | jaytee | who's teaching your class? Jared? |
02:11.26 | knctrnl | web access is great |
02:11.32 | knctrnl | David Duffett |
02:11.51 | knctrnl | He is very good |
02:11.53 | jaytee | not sure if I met him. Jared Smith taught my class |
02:12.00 | knctrnl | David is from UK |
02:12.06 | [TK]D-Fender | I've met Jared twice IIRC... |
02:12.07 | knctrnl | I think Jared is out this week |
02:12.45 | jaytee | he comes to Huntsville for the classes but runs a business in Virginia or D.C. I think the rest of the time |
02:13.00 | knctrnl | I came for the Fast Start and so far after 3 days down of Advanced the only new material is PRI config |
02:14.21 | jaytee | I took the Advanced back in November. I'd been using * for about a year and a half at that point. |
02:15.01 | knctrnl | yeah I have been using for about 8 months. My main worry is syntax of course |
02:16.59 | jaytee | if the lesson plan hasn't changed much then you should already have a fair working copy that's similar to what's going to be required for a config for the lab. I think we went over IVR stuff on thursday. |
02:17.06 | Qwell | knctrnl: Did David bring candy with him? :p |
02:17.12 | Qwell | I might need to go harass him tomorrow.. |
02:17.20 | knctrnl | oh yeah |
02:17.23 | knctrnl | loads of candy |
02:17.34 | Qwell | sweet. I'll make a stop down there :p |
02:17.38 | jaytee | Qwell, have you managed to quit smoking yet? |
02:17.42 | Qwell | jaytee: meh |
02:17.50 | jaytee | same here |
02:18.04 | knctrnl | You in hsv qwell? |
02:18.09 | Qwell | knctrnl: yeah |
02:18.22 | knctrnl | im local too |
02:18.24 | knctrnl | what comp? |
02:18.25 | Qwell | knctrnl: I was upstairs yesterday with the group for lunch |
02:18.35 | Qwell | knctrnl: 2nd floor :p |
02:18.46 | knctrnl | gotcha |
02:20.23 | *** part/#asterisk knctrnl (n=aembrey@user-69-1-12-224.knology.net) |
02:20.36 | *** join/#asterisk knctrnl (n=aembrey@user-69-1-12-224.knology.net) |
02:23.11 | knctrnl | I was up there too. There were several non class folks up thre |
02:23.34 | Qwell | < non class folk |
02:24.00 | Qwell | I'm difficult to miss |
02:24.14 | knctrnl | sitting with JAN? |
02:24.21 | Qwell | mmhmm |
02:24.58 | knctrnl | I gotcha. im the big bald guy |
02:28.35 | plasmid | when i add an extension and I have a TDM400 (for my house); do I choose Generic SIP Device or Generic Zaptel Device? |
02:31.29 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
02:32.47 | p3nguin | hardwire: I don't think so. |
02:33.49 | hardwire | I bet you do. |
02:34.12 | p3nguin | jblack: Although I acknowledge that I have been unpung, was there something you needed me for? |
02:34.35 | p3nguin | hardwire: Personally, I don't like it that much. |
02:35.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
02:36.15 | plasmid | what channel is recommended for a TDM400 (This device uses zap technology. (Via DAHDI compatibility mode)) ; I am adding an extension: 100 to test. |
02:37.29 | p3nguin | After all that fighting with SIP and the conclusion was that I switched the particular client to IAX2, now the audio from said client is a loud buzzing sound. It worked well for at least one day, now not so much. |
02:40.57 | hardwire | buzzing noises are quite analog. |
02:42.12 | p3nguin | Not sure where to look for troubleshooting. I'm pretty sure the microphone on the computer is still working correctly. |
02:42.26 | jblack | p3nguin: I had lost the prompts you made me. I found them |
02:42.34 | jblack | sorry to disturb you |
02:42.41 | p3nguin | Oh, no problem. |
02:42.55 | p3nguin | Just got back in and was catching up. |
02:52.22 | *** join/#asterisk [Outcast] (n=anonymou@c-71-192-36-219.hsd1.ma.comcast.net) |
02:52.36 | [Outcast] | what is that md5 hash created from in sip registration response? |
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03:18.34 | dlynes | Anyone happen to know how many 'flavors' there are for the digium t.38 fax engine? |
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03:21.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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04:12.04 | d1ss3nt | anyone around to take a PRI question? |
04:14.09 | d1ss3nt | chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
04:14.11 | d1ss3nt | :( |
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04:57.34 | luckyaba | Nitro tha CiMien |
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05:03.32 | luckyaba | Nitro tha CiMien |
05:04.16 | knctrnl | anybody know of any vialble solutions to do conference briding wiht G722 wideband? |
05:04.38 | *** join/#asterisk Defraz (n=Defraz@192.41.16.241) |
05:04.45 | knctrnl | Also the capability do join nonG722 callers but not hurt the quality for the HD boys |
05:05.50 | *** join/#asterisk baijum (n=baiju@59.90.236.54) |
05:05.58 | [TK]D-Fender | knctrnl: look at FreeSWITCH |
05:09.08 | knctrnl | is it pretty simular to get running? |
05:09.12 | knctrnl | concepts the same? |
05:09.42 | knctrnl | would the confbrige app in 1.6.2 be workth trying? |
05:09.43 | [TK]D-Fender | knit is its own world |
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05:10.35 | [TK]D-Fender | knctrnl: *'s mixing core is SLIN, not WB |
05:10.42 | *** join/#asterisk oej (n=olle@ns.webway.se) |
05:11.50 | knctrnl | wb? |
05:12.19 | knctrnl | I was speaking with a DIgium engineer and he said that the new conf module can handel it becuse is does not mix on dahdi |
05:14.09 | [TK]D-Fender | knctrnl: Everything involving transcoding always brought it down to SLIN which would kill it |
05:14.27 | [TK]D-Fender | knctrnl: unless they are doing something VERY new.. |
05:15.00 | Corydon76-dig | [TK]D-Fender: you mean like mixing in slin16? |
05:15.39 | [TK]D-Fender | Corydon76-dig: I mean like I'ver reached about a step ahead of my experience :) |
05:15.52 | [TK]D-Fender | Corydon76-dig: Plesae do take over ;) |
05:16.06 | Corydon76-dig | Nah, you're doing fine |
05:16.14 | Corydon76-dig | I was about to go to bed, anyway |
05:22.00 | [TK]D-Fender | I'm off myself.... |
05:22.02 | [TK]D-Fender | later all |
05:30.27 | *** join/#asterisk bmg505 (n=leon@196.209.8.169) |
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05:32.05 | *** join/#asterisk mitesh (n=mitesh@210.212.5.84) |
05:33.05 | mitesh | digium tdm410, does it support gsm networks? |
05:34.58 | Corydon76-dig | Nope |
05:35.10 | Corydon76-dig | analog tdm networks only |
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05:36.21 | mitesh | Corydon76-dig, so does this mean that call from cellphones cannot be entertained nor routed to cellphones by the asterisk server |
05:38.24 | TJNII | No, it means that card can't serve as the link. |
05:39.37 | mitesh | but can the call from and to gsm network be made? |
05:43.10 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
05:44.29 | florz | no, for that you need radiation proof interface cards (and telephones) |
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05:50.20 | Corydon76-dig | mitesh: Can you change someone's mind by willing it? Or do you have to communicate with them? |
05:50.24 | ZenBSDi | Hey guys .. whats a popular web interface for asterisk? |
05:50.59 | Corydon76-dig | You asked about willing it. We're saying you have to communicate. |
05:51.55 | mitesh | Corydon76-dig, sorry, didnt get you |
05:52.35 | Corydon76-dig | The card does not directly talk to GSM networks, but it certainly can by going through the PSTN backbone |
05:52.59 | Corydon76-dig | There IS other hardware that can talk DIRECTLY to the gsm network, but the tdm card is NOT it |
05:53.42 | knctrnl | zenbsdi google Asterisk GUI or asterisk Distribution |
05:54.50 | Corydon76-dig | ZenBSDi: there's only one web interface for Asterisk... AsteriskGUI... other GUIs are for particular applications related to Asterisk |
05:55.14 | Corydon76-dig | Call centers, call conferencing engines, PBX administration... |
05:55.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
05:55.52 | Corydon76-dig | but as for Asterisk itself, only one |
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06:17.24 | mitesh | Corydon76-dig, there? |
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07:47.46 | jojo^ | Is it possible to route a SIP call with asterisk? Thus, instead of creating a new leg with Dial, just route the call like openser, with the same callid. |
07:49.09 | jojo^ | Maybe "like a SIP proxy" is what I'm looking for. |
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07:57.16 | *** part/#asterisk lftsy (n=lftsy@88.191.80.8) |
07:59.26 | mchou | jojo, use a real sip proxy like Kamailio. Asterisk is a swiss army knife but individual blades dull quickly |
08:00.24 | kaldemar | i'd put it like asterisk is a B2BUAS, not a proxy. |
08:08.36 | jojo^ | mchou, We got Kamailio in front, but due to the very flexible nature of Asterisk AGIs it would be very nice to have all logic there, instead of split logic between Kamailio and our asterisk AGI. But I guess the workable solution will be to pass the call back to Kamailio when the AGI have decided it should be forwarded (but then we need to send it back to our AGI if the forward times out and should be sent to our voicemail.. This is basically what I'm trying to avoid) |
08:08.36 | *** join/#asterisk d1ss3nt (n=nebula@s0up.digitalkharma.org) |
08:09.11 | d1ss3nt | anyone how if its possible to update to a newer firmware on a digium card? |
08:09.25 | d1ss3nt | via asterisk/zaptel, that is |
08:11.54 | ChannelZ | hmm I don't believe I've ever seen firmware for digium cards |
08:13.45 | kaldemar | d1ss3nt: ask digium sales for that. years ago you needed to send the card to digium for that. |
08:14.33 | d1ss3nt | weak sauce |
08:15.55 | d1ss3nt | Anyone ever seen this with a Digium card on a standard US PRI line? --> NOTICE[11150] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
08:16.12 | d1ss3nt | w/ intermittent alarms and call drops/resets |
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08:51.29 | mbrevda | does anything exist like chanspy, but where you can speak to BOTH parties? |
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09:06.58 | Torocatala | Hi |
09:12.01 | Torocatala | I have a weird issue whit my PBX, I made an IVR, and works fine whit softphones, but, when I use a hardware phone, and I press, 1 (or another number) the IVR do not answer, I try whit to hardware phone whit the same result |
09:12.12 | Torocatala | anyone know about this issue? |
09:15.55 | Torocatala | thx... |
09:20.51 | Torocatala | is a Matrix? |
09:21.49 | trogs | sounds like a problem with the dtmf |
09:22.13 | Torocatala | thanks! |
09:22.24 | trogs | you'll prob want to check that your phone is set to use rfc2833 mode |
09:22.29 | trogs | for dtmf |
09:22.59 | Gugge | Anyone know why Expire would show -1 when i register an realtime iax peer on my asterisk 1.4.21? http://pastebin.com/m7e113018 |
09:23.16 | trogs | you'll probably want to check it for dialing in from externally, as there might be trunk settings you need to make sure is configured correctly |
09:23.43 | trogs | rfc2833 is generally the best thing to set it to. |
09:23.52 | trogs | but really does depend on your trunk |
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09:30.24 | Torocatala | hey, thanks a lot trogs |
09:30.56 | Torocatala | I put the rfc2833 in the phone and now works fine |
09:31.02 | Torocatala | :D |
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09:39.28 | devyll | what is the recommended sollution for uploading queue_log data to mysql real time ? |
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10:18.15 | mattboll | hi, do you know some software in linux, that monitor easily blf (who is connected, incallâ¦) and which is not a webpage ? |
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10:43.25 | trogs | mattboll: you might be able to use FOP |
10:43.41 | trogs | http://www.asternic.org/ |
10:44.00 | trogs | Torocatala: good stuff, was pretty sure that would do it :) |
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10:52.33 | Grof | guys |
10:52.47 | Grof | i have two asterisk machines |
10:53.06 | Grof | while there are SIP calls between them |
10:53.17 | Grof | one of the machines shuts down |
10:53.25 | Grof | and other one just freaks out |
10:53.33 | Grof | spamming "exceptionally long queue" |
10:53.39 | Grof | is that normal? |
10:53.40 | Grof | :D |
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11:13.29 | *** join/#asterisk i-pink (n=my-pink@bzq-79-183-99-84.red.bezeqint.net) |
11:13.38 | i-pink | hii |
11:13.54 | i-pink | someone here? |
11:14.36 | i-pink | how i can change the user agate in twinkle? |
11:14.45 | *** join/#asterisk Weedle (n=Weedle@ausphreak/Weedle) |
11:14.48 | Weedle | hey peeps |
11:15.10 | Weedle | im trying to get call manager to work with asterisk, the call come in but comes up with this... |
11:15.13 | Weedle | <PROTECTED> |
11:15.33 | i-pink | ? |
11:15.45 | i-pink | you know how i can change the user agate in twinkle? |
11:16.01 | Weedle | the call manager basicly sends of the call to asterisk, doesnt register, so what do i have to do to sip.conf to allow it to pick up the call manager and select a context |
11:16.42 | *** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu) |
11:16.52 | Jankooo | Hi! |
11:22.15 | Jankooo | Is there anybody who has a little time to help me? |
11:22.21 | Jankooo | I am just installed the asterisknow under wmvare... |
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11:32.35 | tzafrir | i-pink, hi |
11:32.44 | i-pink | hii |
11:32.59 | tzafrir | what do you mean by "user agate"? |
11:34.30 | i-pink | i use twinkle for soft phone |
11:35.08 | i-pink | and i want to change the "user agate" |
11:35.22 | kaldemar | Jankooo: there is #asterisknow for asterisknow related questions. |
11:35.43 | i-pink | is send the name of the program to the server |
11:36.12 | i-pink | tzafrir ^ |
11:36.33 | kaldemar | Weedle: if the call doesn't match any peer, asterisk will look for extensions in the context defined under [general]. |
11:36.52 | i-pink | ? |
11:37.05 | Weedle | ooh cheers kaldemar |
11:37.34 | tzafrir | i-pink, "user agent" |
11:37.46 | i-pink | yes |
11:38.50 | Weedle | kaldemar nope still get.. |
11:38.51 | Weedle | [Oct 22 05:38:31] NOTICE[968]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '2222' rejected because extension not found. |
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11:41.37 | i-pink | tzafrir, "user agent " is the name of the program |
11:41.39 | tzafrir | i-pink, can't find it anywhere. What do you need it for? |
11:42.02 | i-pink | for me |
11:42.24 | tzafrir | In the worst case, I guess you can always patch the source :-( |
11:42.38 | i-pink | i start to compile it :-) |
11:43.39 | i-pink | and i try to make filter to ettercap for temporary change |
11:49.29 | kaldemar | Weedle: you don't have extension 2222 under the context asterisk is looking for. pastebin a CLI output for a call with verbosity and sip debug enabled and well take a closer look. |
11:51.43 | Weedle | i have the extention in there, ive tryed different contexts, placing it under general. |
11:53.30 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
11:53.45 | Weedle | how do i enable sip debug? |
11:54.28 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
11:55.46 | WinZ | Weedle, core set debug channel all |
11:58.18 | kaldemar | Weedle: sip set debug on |
11:58.51 | *** part/#asterisk equality4xy (n=equality@76-219-201-200.lightspeed.irvnca.sbcglobal.net) |
12:00.04 | Weedle | http://pastebin.com/m528c830d |
12:01.50 | kaldemar | still waiting for the sip debug... |
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12:16.03 | ManxPower-work | ARGH! The tech I'm working with simply won'y stop talking. |
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12:39.27 | ariel_ | Morning |
12:39.48 | Weedle | ManxPower-work slap him...with your penis |
12:42.49 | beek | Weedle: what if the tech guys likes that? |
12:43.36 | ManxPower-work | Weedle: He's 800 miles away. I don't think it would reach. |
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12:44.38 | Weedle | beek then get a chick and slap him with her tits |
12:45.09 | beek | Weedle: now *I* would like that. |
12:45.35 | Weedle | hehe |
12:45.41 | Weedle | who wouldnt |
12:46.36 | ariel_ | wow slapping by proxy |
12:46.44 | ariel_ | just does not work the same way |
12:47.04 | beek | Just borrow [TK]D-Fender's Clue-Bat (TM) |
12:47.22 | [TK]D-Fender | MY PRECIOUS!!!!!!!!!!! |
12:47.53 | [TK]D-Fender | hordes his ironwood ClueBat (tm) and sneers evilly |
12:48.08 | ariel_ | argh left my pen at home....need to raid the supply cabinet... |
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12:56.15 | scalex000 | hey, good morning guys |
12:56.39 | ariel_ | morning |
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13:04.43 | Gugge | I have a box that gives errors when asterisk sendes "Require: timer" in the 180 Ringing packets ... is there any reason to send those ... and is my box broken? |
13:05.16 | Dovid | no. from the sound of it you need dahdi (or some timing source) this is a pure guess ;) |
13:05.26 | [TK]D-Fender | Dovid: Correct. |
13:05.28 | Dovid | do u have any telephony hardware in the box ? |
13:05.41 | Dovid | TK: Thanks. I finally got something right ;) |
13:05.44 | [TK]D-Fender | Dovid: That clearly doesn't show any sense of know what you're takling about :p |
13:05.48 | Dovid | Pats himself on the bvack |
13:06.07 | [TK]D-Fender | Dovid: FAIL :D |
13:06.13 | Dovid | i am lost. |
13:06.24 | Dovid | correct was that it was a guess and NOT correct ? |
13:06.41 | [TK]D-Fender | Gugge: What is the box that gives the error to *'s 180? And what is the error? |
13:06.58 | Gugge | its some fiber box, with an ata |
13:06.58 | Dovid | talking abotu failing..... http://failblog.org/ |
13:06.59 | [TK]D-Fender | Dovid: Correct that it WAS a "pure guess". |
13:07.03 | Gugge | and it doesnt return an error to * |
13:07.05 | Dovid | haha |
13:07.11 | Gugge | the box just logs "invalid packet" |
13:07.19 | [TK]D-Fender | Gugge: Where do you see this error? |
13:07.28 | Gugge | on the console of the fiber-ata box |
13:07.35 | [TK]D-Fender | Gugge: Maybe your device doesn't care about progress indication. |
13:07.57 | [TK]D-Fender | guBut it sounds more like a nuisance than a "problem |
13:07.58 | Gugge | well, it doesnt play ringtones when it get the 180 ringing :) |
13:08.09 | Gugge | if u remove the require: timer from chan_sip.c it does :) |
13:08.34 | Gugge | i just wonder why theres even an require: timer in the ringing packets |
13:08.38 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
13:08.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
13:09.03 | Gugge | i would expect the require header only in INVITE and OK |
13:09.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:10.30 | *** join/#asterisk Skeeter- (i=Skeeter-@c216.218.2-65.clta.globetrotter.net) |
13:10.44 | Skeeter- | anyone ever worked with Polycom Spectralink wireless phones |
13:11.10 | kaldemar | the timer is an option tag that tells the other end that asterisk requires session timer support. |
13:13.58 | Katty | omnomnomnoms on breakfast burrito |
13:14.23 | beek | Good morning Katty |
13:14.43 | ariel_ | morning Katty hope your doing better this morning. Sends a hug her way.... |
13:15.00 | Katty | ariel_: yesh :> |
13:15.06 | Katty | ariel_: the world is a little brighter this morning |
13:15.09 | Katty | hugs ariel_ |
13:15.15 | Katty | good morning beeks :> |
13:15.17 | Katty | hugs on beek |
13:15.33 | beek | hugs back |
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13:19.32 | Gugge | kaldemar: thats what i figured ... but i have it set to "accept" in sip.conf ... and it still sends the require: :) |
13:19.56 | *** join/#asterisk oej (n=olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
13:19.57 | Gugge | i would expect "accept" to only accept the timer extensions, not require it |
13:20.10 | [TK]D-Fender | [09:13]* Kattyomnomnomnoms on breakfast burrito <- so much for eating "healthy" :| |
13:21.05 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
13:21.07 | Katty | oh it's quite healthy. |
13:21.13 | Katty | thankyouverymuch |
13:22.15 | galeras | Is possible to send a fax with T38 protocol over a PRI (i mean directly connected to the * box) |
13:22.26 | WinZ | http://pastebin.com/d7bd505ef -- guys, what can this error mean? Sometimes it happens on outgoing call through my trunk. The call is established, but silence only I can hear |
13:23.05 | ManxPower-work | galeras: No. |
13:23.27 | ManxPower-work | WinZ: Looks like a provider or network problem |
13:24.10 | WinZ | ok |
13:24.14 | WinZ | ManxPower-work, thank you |
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13:34.51 | tuxx- | hey guys. We have a client with an asterisk PBX. Every day they have some calldrops, and the cause seems to be the following error: 'DEBUG[26670] channel.c: Didn't get a frame from channel: Zap/30-1'. I searched google, but everyone having this problem seems to be stuck in finding an answer. Is anyone here who could point me to a solution, or maybe a workaround? |
13:37.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:40.16 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
13:40.42 | superbeef | do you think it's more CPU intensive to take calls with a T1 card and software echo cancelationation then from a IAX trunk? |
13:40.43 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
13:41.33 | angryuser | superbeef, why dont you test ? |
13:42.11 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
13:42.12 | superbeef | angryuser: I prefer to dabble in speculation |
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13:48.56 | [TK]D-Fender | superbeef: Obviously |
13:52.54 | superbeef | i'm actually in a pretty good position to test since i have a 2 PBX's connected with a T1 crossover |
13:55.32 | Katty | blinks a bit |
13:55.43 | Katty | wow...there are absolutely no nutrients in that tortilla i ate. |
13:55.46 | ManxPower-work | software EC takes CPU cycles |
13:55.47 | Katty | except for sodium. |
13:55.57 | Katty | and a smidgen of calcium |
13:56.10 | Katty | but the other stuff was good for me! |
13:56.12 | ManxPower-work | those are nutrients! |
13:56.45 | Katty | looks up whole wheat tortilla |
13:57.28 | Katty | well, it's better. |
13:57.31 | Katty | but still mostly useless. |
13:57.46 | [TK]D-Fender | superbeef: You shouldn't need EC anyway between them. |
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13:59.13 | DavidR2008 | does anyone have any information on how accurate timing has to be for iax to work well? Will an average of 99.7+ work? I'm testing dahdi dummy using dahdi_test and that's what I'm getting off of my hardware |
14:03.10 | superbeef | [TK]D-Fender: probably not.... I just have the T1 cross over so I can make sure the T1 card is setup right before I deploy a PBX |
14:03.32 | Katty | the day is young, but so far i've had absolutely no Vitamin E, b1, b2, b3, b6, b9, and very little K |
14:04.02 | [TK]D-Fender | DavidR2008: Are you using trunk mode to your provider? |
14:04.23 | DavidR2008 | no, it's between local * servers |
14:04.38 | [TK]D-Fender | DavidR2008: then timing should be irrlelvant |
14:04.54 | DavidR2008 | ok, thanks |
14:05.31 | *** part/#asterisk Boardy (n=chatzill@ip565785d1.direct-adsl.nl) |
14:06.40 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
14:06.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
14:07.28 | *** join/#asterisk cusco (n=trilili@213.63.137.210) |
14:07.30 | cusco | hi |
14:08.59 | cusco | dahdi_scan shows: http://paste.debian.net/49710/ |
14:09.11 | cusco | there should be 6 channels with 2 spans |
14:09.29 | cusco | in asterisk's cli dahdi show channels only show 30 channels |
14:09.30 | cusco | why? |
14:10.44 | kaldemar | cusco: you only configured 30 channels in asterisk |
14:11.31 | cusco | ok /etc/asterisk/dahdi-channels.conf has http://pastebin.com/f62b4b52d |
14:11.42 | cusco | what am I missing? |
14:12.57 | guax | lunch |
14:13.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:13.25 | kaldemar | cusco: is that #included in chan_dahdi.conf? |
14:14.20 | [TK]D-Fender | wonders why he doesn't SEE the primary file, or the backup of only 6 channels showing up |
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14:18.31 | tuxx- | hey guys. We have a client with an asterisk PBX. Every day they have some calldrops, and the cause seems to be the following error: 'DEBUG[26670] channel.c: Didn't get a frame from channel: Zap/30-1'. I searched google, but everyone having this problem seems to be stuck in finding an answer. Is anyone here who could point me to a solution, or maybe a workaround? |
14:18.57 | cusco | kaldemar: you got me |
14:18.59 | cusco | thanks |
14:19.17 | ManxPower-work | tuxx-: that usually means "far end hungup" |
14:19.42 | Katty | considers this lack of vitamin b |
14:19.42 | ManxPower-work | you would not even see the message if you didn't turn on DEBUG |
14:19.55 | Katty | leifmadsen: i'm lacking in B, K, and E so far this morning |
14:20.15 | Katty | leifmadsen: the morning is young, no doubt |
14:20.19 | cusco | kaldemar: chan_dahdi.conf has http://pastebin.com/f66bee108 |
14:20.25 | leifmadsen | Katty: oh snap! Hmmm... I ate my half a banana, so my K should be ok, and I had some greens+ -- I probably need a multivitamin and some fishoil though |
14:20.25 | cusco | so if I modified to look like: |
14:20.33 | ManxPower-work | Katty: you realize the eating healthy, getting exersize, not smoking, drinking, drugs, etc doesn't acutally make your live any longer. It just SEEMS longer. |
14:21.03 | *** join/#asterisk moy (n=moy@74.12.134.3) |
14:21.11 | tuxx- | ManxPower-work: we did turn on debug idd, but we called the 'far end' and he/she says the call got disconnected. and this usually happens 3 times a day |
14:21.22 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
14:21.30 | Katty | leifmadsen: judging from items high in vitamin E, i know i will have some by the end of the day. |
14:21.38 | Katty | leifmadsen: nuts, oils, spinach, sunflower seeds, whole grains |
14:21.45 | tuxx- | vitamins, eugh. |
14:21.53 | leifmadsen | Katty: nice! now I must go take some pills :) |
14:21.57 | Katty | leifmadsen: kk |
14:22.01 | ManxPower-work | zttool or dahdi_tool looked for missed IRQs/Interrupts |
14:22.02 | tuxx- | real nerds live on caffeine and pizza! |
14:22.02 | cusco | so if I modified to look like: http://pastebin.com/m10884cc3 it would be OK, right kaldemar ? |
14:22.05 | kaldemar | cusco: your dahdi-channels.conf is irrelevant because you don't include it in chan_dahdi.conf |
14:22.21 | kaldemar | cusco: yes, that's one way of solving it. |
14:22.23 | cusco | kaldemar: i would activate 2nd span that way |
14:22.45 | cusco | thats all I need, a new group and setting channel => |
14:22.51 | *** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu) |
14:22.52 | kaldemar | cusco: but your switchtype is different from dahdi-channels.conf now. |
14:23.04 | DavidR2008 | do you have to have chan_dahdi to use the dahdi dummy driver? |
14:23.12 | cusco | no |
14:23.35 | superbeef | tuxx-: you should turn on pri debugging |
14:24.02 | tuxx- | hmkay, will do. |
14:24.21 | cusco | kaldemar: so now I have to unload and load chan_dahdi.so again |
14:24.35 | cusco | (We are having calls right now, I will have to wait) |
14:24.42 | *** join/#asterisk muh-die-kuh (n=hco@muh-die-kuh.de) |
14:26.30 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-209-158-5-138.altnpa.east.verizon.net) |
14:29.19 | cusco | kaldemar: dahdi-channels.conf is not valid, it was never in use. I just did not realised it before :p |
14:29.30 | cusco | should have figured from the "context" |
14:29.51 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
14:31.39 | CGMChris | Is anyone familiar with how AGI/AMI? I am trying to place an outbound SIP call and detect when it has been answered, but cant seem to figure it out. BackgroundDetect() *appears* to be detecting rings as noise. |
14:33.06 | ManxPower-work | CGMChris: analog or PRI or SIP? |
14:33.35 | CGMChris | ManxPower-work: SIP. |
14:33.47 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
14:34.02 | [TK]D-Fender | CGMChris: AMD <- |
14:34.19 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
14:34.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:34.31 | cusco | oops |
14:34.32 | cusco | where is the changelog at? |
14:34.57 | cusco | CHANGES file has only "Functionality changes from Asterisk 1.6.1.1 to Asterisk 1.6.1.2" and previous |
14:35.05 | cusco | what are the latest changes until 1.6.1.6 |
14:35.13 | [TK]D-Fender | cusco: www.asterisk.org |
14:35.39 | cusco | im there |
14:35.55 | *** join/#asterisk baijum (n=baiju@203.123.174.162) |
14:36.24 | ManxPower-work | cusco: ChangeLog |
14:36.59 | cusco | "We couldn't find what you were looking for." |
14:37.06 | leifmadsen | what link? |
14:37.16 | leifmadsen | there are some dead links on the asterisk.org site I'm going to fix up this morning |
14:37.21 | cusco | there is no link, I just typed /Changelog |
14:37.31 | leifmadsen | heh |
14:37.36 | leifmadsen | it's in the tarball |
14:37.44 | cusco | 15:34 < cusco> CHANGES file has only "Functionality changes from Asterisk 1.6.1.1 to Asterisk 1.6.1.2" and previous |
14:37.45 | leifmadsen | see the -summary files too |
14:37.50 | cusco | ah |
14:37.55 | cusco | ls |
14:38.04 | leifmadsen | cusco: I know -- that means there haven't been any super-major-freak-out changes since those versions |
14:38.18 | *** join/#asterisk E-bola (i=bola@ip181.rev112.brygge.net) |
14:38.21 | E-bola | Hello |
14:38.22 | cusco | there must be some deserving releasing it :p |
14:38.28 | leifmadsen | ChangeLog is all changes between versions, and the summary is a summary of those changes which closed issues on the bug tracker |
14:39.15 | cusco | thanks :) |
14:40.42 | CGMChris | [TK]D-Fender: Isnt AMD only useful after the call has in fact been answered? The problem I'm having is detecting when the call is actually answered, not if it's a machine or person. I am using AMI to Originate a channel (SIP/provider/external phone #) and an application [AGI(script.php)]. The problem is that script.php begins executing as soon as the SIP channel connects (ringing), but before a person is on the line. |
14:40.49 | E-bola | I got a relatively newbie qeustion: I have a place in my dialplan where i need to call 1 phone for 10 secs, if no answer then call another phone for 10 secs, etc. I would like for the caller to hear continoues music while this system moves from phone to phone, is that possible? |
14:41.18 | [TK]D-Fender | CGMChris: if its SIP SIP is supposed to be OOB |
14:41.32 | E-bola | Does specifying m as an option restart the music on hold for each dial cmd? |
14:41.42 | [TK]D-Fender | CGMChris: Which means you're not supposed to do anything |
14:41.51 | ManxPower-work | E-bola: yes, possible, no easy. But if you don't mind the hold music starting over or don't mind a very short gap then it's easy |
14:42.30 | E-bola | ManxPower: well idealy there shouldnt be any gaps. |
14:42.54 | E-bola | They should just hear streaming music without interuptions untill a dial commands reaches an extension that picks up |
14:43.11 | ManxPower-work | E-bola: that is non-trivial unless you are good with dialplans |
14:43.17 | E-bola | :( |
14:43.29 | E-bola | It woudl have thought it was a relatively common type of behavior |
14:43.32 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
14:43.33 | E-bola | I even |
14:43.40 | ManxPower-work | E-bola: what are the two extensions? |
14:43.52 | E-bola | ManxPower: 81 and 83 |
14:44.04 | ManxPower-work | which one needs to have a 10 second delay? |
14:44.26 | E-bola | Well both, i mean it should try 81 for 10 secs, then 83 for 10 secs |
14:45.00 | CGMChris | [TK]D-Fender: My goal is to call the owner of the phone system (AMI originate) and allow them to record custom greetings for their phone system (AGI). Am I going about this the wrong way? My test call is SIP to SIP, rfc2833 for dtmfmode. |
14:45.04 | DavidR2008 | can anyone point me to instructions on how to install dahdi_dummy? |
14:45.58 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
14:45.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:46.15 | ManxPower-work | Dial(Local/8183-delay@context,20,m) then exten => 8183-delay,1,Dial(SIP/81,10) 8183-delay,n,SIP/83,10) |
14:46.23 | ManxPower-work | That will make hold music stay consistant. |
14:46.46 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
14:47.22 | ManxPower-work | Since the Local/ Dial will provide the hold music |
14:47.58 | ManxPower-work | Remember most users are so slow it takes them 10 seconds just to lift the handset. |
14:47.59 | E-bola | hmmm |
14:48.36 | E-bola | ye i see how it must work, ile try it |
14:48.54 | *** join/#asterisk hat_panda (n=pete-joh@triton.dsv.su.se) |
14:48.55 | E-bola | Im not sure i understand the Local/ part? |
14:49.26 | ManxPower-work | Local/ allows Dial to "dial" extensions instead of devices. |
14:49.35 | ManxPower-work | localchannel.txt in /path/to/src/asterisk/doc |
14:49.48 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
14:50.32 | E-bola | ohh cool, thank you |
14:50.38 | E-bola | never stumbled upon that |
14:50.50 | ManxPower-work | Its one of the most useful Asterisk features |
14:51.21 | superbeef | finally came up with a fix for IAX dropped calls between new Asterisk 1.4 boxes and old asterisk 1.2 boxes |
14:51.59 | superbeef | I ditched speex and switch to G711 |
14:52.05 | superbeef | bandwidth should be interesting |
14:52.08 | CGMChris | [TK]D-Fender: Figured it out. The system I was calling into was calling Answer() as soon as calls were received, rather than when a real person answered. My fault! |
14:52.53 | ManxPower-work | NEVER Answer() unless you have to. |
14:53.35 | p3nguin | If you're just doing a Dial(), let the phone perform the answering. |
14:53.55 | ManxPower-work | Don't use "r" option to dial either |
14:53.57 | CGMChris | Yes, I am letting it fall through to Dial now and that seems to work just fine. |
14:54.14 | ManxPower-work | "r" stands for "retard", as in "only a retard would use the 'r' option to Dial" |
14:55.19 | p3nguin | If m isn't an option, won't it ring anyway? |
14:55.25 | p3nguin | without r |
14:55.37 | *** join/#asterisk Carlos_PHX (n=Carlos@68.99.199.10) |
14:55.40 | ManxPower-work | p3nguin: correct |
14:55.49 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
14:56.26 | ManxPower-work | Well, it will send whatever is correct. ringback, your call cannot be completed as dialed, number disconnected, the cellular subscriber you are trying to reach is out of the service area, etc |
14:57.46 | *** join/#asterisk ghento (n=ghento@user146-1.wireless.utoronto.ca) |
14:57.54 | p3nguin | I need to do some testing on that, now. |
15:01.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:02.36 | ghento | Hi all. I'm just curious if anyone can recommend a method to detect voicemail? For example, if a call is answered, I'd like to use call flow A, but if I detect a voicemail, would use call flow B. |
15:03.04 | [TK]D-Fender | ghento: "core show application amd" |
15:03.08 | p3nguin | Dial(SIP/202|40|ktT) produces a ringing sound in my phone, so there's no real good reason to need to add an r in there. |
15:03.45 | ManxPower-work | p3nguin: you want BOTH the call-er and the call-ee to be able to do DTMF transfers? |
15:03.57 | ManxPower-work | what did you say the number is? |
15:04.07 | p3nguin | manxpower-work: Yeah, it's an internal to internal Dial. |
15:04.23 | ghento | D-Fender: Many thanks! |
15:04.38 | ManxPower-work | p3nguin: just making sure. There is a MASSIVE increase in using unsecured PBXs to make calls for scammers |
15:05.12 | ManxPower-work | Once a person receives their first phone bill 100x what it normally is, they usually secure their PBXs |
15:05.31 | p3nguin | manxpower-work: Ooooooh... I'll go check the dialplan for Ts in the wrong spot and remove them. |
15:06.09 | ManxPower-work | Just don't use T when a call comes from an untrusted source like the outside. |
15:06.20 | p3nguin | yeah |
15:06.36 | ManxPower-work | Or *gasp* Don't use T or t at all and use the transfer button on your phone. |
15:07.09 | p3nguin | manxpower-work: That's fine for real phones, but most of the free softphones have the xfer button blocked out. |
15:07.45 | *** join/#asterisk pirulo (n=andres@65.102.99.5) |
15:07.57 | ManxPower-work | softphone == cheap pastard |
15:07.58 | p3nguin | I'm still going to review and revise based on what you just said, though. |
15:08.11 | p3nguin | Better safe than sorry. |
15:09.12 | *** join/#asterisk gardo (n=gardo@110.55.240.37) |
15:09.28 | p3nguin | Also, do you normally "hop" into the internal context when a call comes in from outside, or do you have the Dial() commands right in the inbound context? |
15:09.54 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
15:10.24 | p3nguin | Say someone calls in from outside and enters ext202 in the IVR prompts. Seems like a perfect opportunity to send that call to internal 202. |
15:10.41 | [TK]D-Fender | O>o |
15:10.53 | ManxPower-work | I normally have an inbound context to do any pre-Dial processing (change callerID, set the ring cadence, etc), then I route the call via Goto to the correct internal extension in the dialplan for the actual Dial |
15:11.18 | ManxPower-work | if it needs to go into an IVR, I sent the call into an IVR context. |
15:12.16 | p3nguin | That's how I'm doing it, too, but I figured I would use this time to also analyze that as potentially being the wrong way. |
15:13.46 | p3nguin | exten => _2XX,n,Goto(internal|${EXTEN}|1) |
15:14.25 | p3nguin | [tk]d-fender talked me into hard-coding the internal extensions rather than having the same matching scheme in internal. |
15:14.47 | ManxPower-work | p3nguin: don't use | they will go away very soon/ |
15:14.49 | ManxPower-work | use , |
15:15.07 | p3nguin | It won't go away if I don't upgrade to 1.6. :) |
15:15.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:15.45 | superbeef | so what is more useful than show codecs to actually show what codecs i have |
15:16.01 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
15:16.50 | Qwell | superbeef: core show translations |
15:17.55 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
15:18.06 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-209-158-5-138.altnpa.east.verizon.net) |
15:18.07 | p3nguin | manxpower-work: I did recently consider converting my pipes to commas for the sake of helping others who do have 1.6 and my syntax not being compatible. |
15:18.22 | p3nguin | I might do that today. |
15:18.27 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
15:18.48 | p3nguin | Should only take couple minutes to run the replacement and then visually inspect the result. |
15:19.29 | ManxPower-work | Joy. We have one of our techs, a Broadview tech, and a verizon tech all in the phone closet all doing different things |
15:20.14 | superbeef | interesting |
15:20.15 | p3nguin | Quick! Lock the door! |
15:20.26 | superbeef | i wonder why i can't get g726 to work.. maybe because its between asteirsk 1.4 and 1.2 |
15:22.50 | *** join/#asterisk jantypas (n=jantypas@166.205.133.95) |
15:23.07 | Corydon76-dig | superbeef: some endpoints use the AAL2 alternative encoding of G726 and call it G726 |
15:23.55 | superbeef | Corydon76-dig: my 1.4 box has g726 and g726aal2 |
15:24.50 | Qwell | superbeef: You're trying to use g726 between Asterisk boxes? |
15:25.41 | superbeef | Qwell: yep |
15:25.45 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:25.46 | Qwell | why? |
15:26.18 | superbeef | because speex is causing dropped calls, and alaw has a big bandwidth footprint |
15:28.27 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
15:28.56 | superbeef | Qwell: I upgraded my main server to 1.4.current, but it still sends calls via IAX to some really old 1.2 boxes... I started getting dropped calls after a bit of utilization, and after a week of madness, it finally stopped when I quit using speex |
15:29.52 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
15:32.42 | [TK]D-Fender | [11:13]<p3nguin>exten => _2XX,n,Goto(internal|${EXTEN}|1) <--- INCLUDE |
15:33.06 | *** join/#asterisk vaflen (n=soeren@3803ds2-ns.0.fullrate.dk) |
15:33.37 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
15:33.46 | p3nguin | What's the reason to include it when it can easily just Goto the internal context? |
15:34.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
15:34.19 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
15:34.35 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
15:34.37 | vaflen | Hi. I have a question regarding Skype for Asterisk and support for Arm based systems, or rather the lack thereof. |
15:34.45 | Assid | heya |
15:34.46 | [TK]D-Fender | p3nguin: Your goto may not have a match to land on. The contexts of "internal may change. Syntax on the Goto has already changed. Whats the point of contexts when you don't use includes? |
15:35.02 | [TK]D-Fender | p3nguin: What if you have multiple patterns there? |
15:35.24 | Assid | i have a strange issue.. recently users have been noticing that the calls arent being picked up correctly.. like if i call an extension an they pick up the phone.. it continues to ring |
15:35.49 | Assid | i have a diall all policy where i am dialling the sip users all atonce.. |
15:36.08 | p3nguin | [tk]d-fender: The contents of the internal context will be the same whether I include it to Goto it. |
15:36.10 | Assid | so i am guessing the sip signalling is getting messed up.. this is also happening between individual extensions |
15:36.13 | vaflen | Is there any chance there might be support for Arm based systems coming to the Skype for Asterisk addon? |
15:36.36 | [TK]D-Fender | p3nguin: It is the equivalent of bad programming, adding complexity and risk for nothing. |
15:36.39 | Assid | can someone suggest where i can look.. this has recently started and was fine till about a month back.. and now its just happening more often |
15:36.47 | [TK]D-Fender | p3nguin: if they dial a # that is not valid you call can drop like a rock |
15:37.06 | Qwell | vaflen: You'd have to talk to Digium sales. |
15:37.23 | vaflen | Qwell: Thanks. |
15:37.25 | Qwell | vaflen: If there's enough interest, it's possible something we'd consider. |
15:38.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
15:38.38 | Assid | i hae the latest bootrom and sip |
15:38.49 | vaflen | Qwell: I have a small home office where I am currently running Asterisk on a Sheevaplug. I'd like to keep the power consumption to a minimum and the plug is great for that. Asterisk runs very well on it but it would be nice to extend the functionality to include Skype as well. |
15:41.16 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
15:42.16 | grandpapadot | Hi all. Is there an effective limit to how many peers can be dialed at the same time with Dial(SIP/100&SIP/200...), etc? |
15:42.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:44.17 | grandpapadot | .. rather, a recommended limit? |
15:44.31 | leifmadsen | grandpapadot: pretty much the limit is the number of characters, or CPU, whichever you hit first |
15:44.47 | grandpapadot | Thanks, leif! |
15:45.00 | [TK]D-Fender | grandpapadot: Yes.. you'll encounter a hard limit in the raw # of CHARS that any app can accept |
15:45.16 | [TK]D-Fender | leifmadsen: Char limi hits first by far :) |
15:45.32 | Chainsaw | *G* the char limi |
15:46.55 | leifmadsen | [TK]D-Fender: assuming you're not running on an embedded device like a Linksys or something, in which case the CPU will die first |
15:49.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:51.12 | Katty | is getting a 19" Lcd! |
15:52.43 | Katty | and i have chicken baking in the oven :> |
15:55.16 | beek | Katty: You're at work and are baking chicken? |
15:56.12 | Qwell | beek: where else would you bake chicken? |
15:56.23 | ariel_ | would love to have 2 23's widescreens on his desk... |
15:56.46 | beek | Qwell: If I did that here the vultures would start circling... |
15:56.49 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
15:57.19 | scalex000 | help, what is mean this "Spawn extension (from-zaptel, s, 2) exited non-zero on 'SIP/5229-b7c0e870<ZOMBIE>'" |
15:57.55 | p3nguin | It means that the s extension in the from-zaptel context exited at priorty 2. |
15:58.06 | scalex000 | ok |
15:58.08 | scalex000 | thanks |
15:58.35 | Katty | beek: we have a full kitchen at work |
15:58.48 | p3nguin | That's awesome! |
15:58.49 | Katty | beek: and i'm going to make some asparagus to go with it in just a little bit |
15:58.55 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-qgxwizyrdvnsbmdm) |
15:58.58 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:58.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:59.00 | p3nguin | Now you can make me some food. |
15:59.08 | Katty | mhmm |
15:59.10 | beek | Katty -- I'll hop on a plane and be right out. |
15:59.13 | Katty | k |
15:59.28 | knctrnl | j |
15:59.30 | p3nguin | I can be there in an hour if I leave.... now. |
15:59.44 | Katty | it will be out of the oven at 11:45 |
15:59.46 | Katty | so hurry it up |
16:00.16 | p3nguin | I can't do it in 45 minutes, no matter how fast I drive. |
16:00.48 | Katty | :< |
16:00.53 | Katty | well i can always give you the recipe |
16:00.55 | Katty | will that help? |
16:01.13 | p3nguin | probably |
16:01.15 | Katty | k |
16:02.19 | diatonic1 | [TK]D-Fender: I think you were right about the Redfone - This thing has been a PITA and the support is not very responsive. Shipping it back and going Sangoma |
16:07.10 | beek | When bridging Telco PRI through Asterisk to a legacy PBX via PRI do you usually find that the rx/txgain=0.0 settings adequate? I'm getting echo and wondering where to start troubleshooting. |
16:07.11 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-pmzzenjdqtezkxxb) |
16:07.15 | beek | The echo isn't consistent. |
16:07.42 | ManxPower-work | beek: echo comes from the FAR end analog loop. |
16:08.08 | ManxPower-work | So you won't get echo if the destination is VoIP, PRI, T-1, or Cell phone. |
16:08.52 | beek | ManxPower-work: That's what I always thought. I have some complaints from employees that they're hearing themselves. I thought that the far end is responsible (and I just told someone that). |
16:09.22 | ManxPower-work | beek: ALL calls that terminate in analog have echo. But without VoIP the echo happens so FAST you can't actually perceive it. |
16:09.23 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
16:09.39 | ManxPower-work | Once you put VoIP in place latency is high enough that you hear the echo that was there all along. |
16:10.07 | ManxPower-work | sort of like in a small room your voice still echos off the wall, but you can't perceive it because it happens so fast. In a very large room the delay is long enough you could herart. |
16:10.18 | ManxPower-work | .... could hear it. |
16:10.20 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
16:10.20 | diatonic1 | ManxPower-work: Is echo cancellation in a T1 card necessary if it's used to connect to PRIs or analog T1s?.. calling from IP phones internally?.. Is there anything to produce echo there? |
16:10.34 | beek | Hmmm... I have a legacy PBX connected to * via PRI to the Telco via PRI. |
16:10.36 | ManxPower-work | diatonic1: you have not been reading have you. |
16:11.09 | beek | I haven't had a time where I have had that echo problem but some of the employees have. |
16:11.10 | ManxPower-work | diatonic1: echo comes from the far end, so you can't eliminate it unless you put echo canceler on every telephone line you call. obviously not practical |
16:11.31 | ManxPower-work | beek: don't test it by calling a cell phone or a large company |
16:12.16 | beek | ManxPower-work: I told the person to let me know when she gets a call like that and I'll snoop around a bit. |
16:12.23 | beek | I'm not sure that there's anything I can do about it. |
16:12.24 | diatonic1 | ManxPower-work Thanks. It seems like I always get stuff with echo cancellation to play it safe, but I'm wondering if it's really necessary |
16:12.27 | ManxPower-work | try calling the same number that called them |
16:12.33 | ManxPower-work | diatonic1: yes it is required. |
16:13.09 | beek | ManxPower-work: I'll do that this evening.. the employee said that she called in and had echo. I'll call her this evening. |
16:13.29 | ManxPower-work | beek: the far end had echo (PSTN side)?? |
16:13.46 | beek | She claimed that she heard herself and that the inside person did not hear it. |
16:16.02 | *** join/#asterisk chazzm (n=chazz@173-24-217-9.client.mchsi.com) |
16:16.08 | ManxPower-work | and she was using an IP phone? |
16:16.25 | beek | ManxPower-work: I just double-checked with her. She called the office, our side her the echo. Thus, it's on her end. |
16:16.35 | beek | I told her to call me this evening when she gets home. |
16:17.47 | superbeef | You guys ever end up with asterisk getting an extension stuck adn thinking its busy when its not? I always end up boucning asterisk to clear it, but there's gotta be a better way |
16:18.28 | asterwiki | beek: you can also try enabling echo training 'echotraining=yes' when testing to see if that helps (or disable it its already enabled); |
16:18.54 | *** join/#asterisk JoeMoretti (n=jmoretti@76.164.171.81) |
16:19.16 | beek | asterwiki: This is HWEC... I thought that parameter was for SWEC |
16:20.03 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
16:20.25 | asterwiki | beek: that parameter is normally in your zapata.conf |
16:21.05 | beek | asterwiki: chan_dahdi.conf in this case. The docs state that this is for SWEC, which I'm not using. |
16:21.11 | beek | or am I reading that wrong? |
16:22.45 | ManxPower-work | with HWEC on digium cards I believe the echocancel= option is ignored. You should test. |
16:23.04 | ManxPower-work | Digium does a terrible, terrible, job of informing the user which EC is in use. |
16:23.08 | bpgoldsb | In voicemail.conf, if I don't configure an email address for a mailbox, will Asterisk just not send an email notification? Or will it do something bad? |
16:23.23 | beek | bpgoldsb: those are optional |
16:23.42 | bpgoldsb | So just not send? |
16:24.12 | chazzm | to use HWEC you still have to have echocancel=yes |
16:24.21 | beek | bpgoldsb: correct. |
16:24.22 | chazzm | or some number, which will just be handled as a yes for HWEC |
16:25.07 | beek | chazzm: I have echocancel=yes, but 'echotraining' is ignored if you're using HWEC according to the docs. |
16:25.11 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
16:25.36 | beek | I'm not sure about 'echocancelwhenbridged' though... Any ideas there? |
16:25.49 | chazzm | correct, but sometimes it is better to comment out echotraining when using HWEC. I think that depends on what version of zaptel you are using though |
16:26.06 | beek | chazzm: DAHDI here |
16:26.08 | chazzm | I believe echocancelwhenbridged still applies, but it is normally recommended to set it to no anyway |
16:26.20 | beek | chazzm: which is what I have set |
16:26.25 | ManxPower-work | beek: you almost never need to EC when TDM is bridged. The latency is low enough nobody should hear echo |
16:27.03 | beek | ManxPower-work: It's off, so we're good there. |
16:27.24 | beek | Ahh... the lunch bell has been rung. I'm heading out for chinese today. Thanks guys! Back later. |
16:29.25 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
16:32.27 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
16:33.15 | ariel_ | #bitos |
16:33.20 | ariel_ | argh |
16:34.06 | ariel_ | Anyone heard of BitOS's asterisk based pbx? |
16:43.30 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
16:45.00 | Gido-E | ariel_ url orso? |
16:46.01 | angryuser | is there any way to force PAP2T send external ip in Contact field instead of local ? thanks |
16:47.16 | *** join/#asterisk telnettech (n=telnette@71.5.32.196) |
16:47.51 | hardwire | url or it never happened |
16:47.58 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
16:48.01 | ariel_ | Gido-E: it's actually from a co. called bitrail.net |
16:49.08 | ManxPower-work | angryuser: why would you do that? |
16:49.47 | *** join/#asterisk andres833 (n=andres83@201.244.125.6) |
16:51.19 | angryuser | ManxPower-work, i got client <>nat <nat> server ports routed to server, client registers, but when invite comes audio is sent to local clients ip, i have specifyed external ip to pap2, but still not working, maybe you know adress of a free stun server ? |
16:51.35 | *** part/#asterisk pietro (n=pietro@nat.xsec.it) |
16:51.48 | ManxPower-work | angryuser: why not put nat=yes in the sip.conf for that peer? |
16:52.30 | angryuser | ManxPower-work, it is set |
16:52.56 | ManxPower-work | angryuser: then Asterisk will ignore the IP info inside the packet and use the info from the packet header. |
16:54.01 | angryuser | ManxPower-work, look like not all the way : Peer audio RTP is at port 192.168.1.5:16480 (if server replaces 1.5 by ext it will work) |
16:56.56 | ManxPower-work | angryuser: then you have some other problem |
16:57.00 | Gido-E | ariel_ anny clue about SVN server or track system or technical docu? |
16:57.08 | ManxPower-work | maybe the incoming call is not matching the peer you think it matches |
16:57.35 | ManxPower-work | Is your server behind nat? |
16:58.41 | angryuser | ManxPower-work, basicly he gets the invite from pap2t bu audio is sent locall (remote) ip my server is behind nat, the sip mach is good 100% |
16:59.11 | ariel_ | Gido-E: no idea, just trying to get more info about them or it, I have a customer that wants that. |
16:59.20 | ManxPower-work | so you have localnet= and externip= set? |
16:59.53 | *** join/#asterisk pta200 (n=paolo@goose.specialai.com) |
17:00.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
17:00.57 | *** part/#asterisk Joel (n=jjshoe@wsip-70-183-82-162.sd.sd.cox.net) |
17:01.01 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
17:01.03 | angryuser | ManxPower-work, yes, i am sending correct external ip when trying to reach that peer, but to incorrect peers ip |
17:01.28 | ManxPower-work | maybe SIP ALG on your router is screwing it up. |
17:01.38 | angryuser | ManxPower-work, hm, maybe |
17:03.26 | ManxPower-work | "Yahoo Offered Lap Dances At Hack Event" I guess that's one way to make an...impression. |
17:04.29 | pta200 | Why is that if you pass the meetme application the wrong conf room/pin number and it jumps to the 'h' extension after telling playing the conf-invalid file, if you try to playback any other audio file in the 'h' extension Asterisk will throw a file.c:747 ast_readaudio_callback: Failed to write frame and not play the file? I've seen this in 1.4.24-1.4.26.2 |
17:06.33 | ManxPower-work | pta200: do you have a priority after the broken meetme line? |
17:06.58 | ManxPower-work | if not, that would explain the channel hanging up |
17:08.42 | pta200 | I have a hangup priority, but the cli show asterisk going to the 'h' extension and running through priorities sequentially, I'll pull ot the hang up priority in the meetme call and try that |
17:09.49 | ManxPower-work | no, you don't have a "hangup priority" You have a hangup extension |
17:10.12 | ManxPower-work | put a line after the meetme to handle when meetme fails |
17:13.19 | pta200 | That doesn't seem to make a different app_meetme skips to the hang up extension and fails trying to play an audio file back regardless of there being a priority after the failed called to the meetme application |
17:13.25 | Chainsaw | Qwell: I've got an easy one for you, kernel 2.6.32 API change wrt driver_data: https://issues.asterisk.org/view.php?id=16114 |
17:13.32 | Chainsaw | Qwell: Patch, license OK, etc. |
17:14.09 | pta200 | This the CLI output |
17:14.50 | pta200 | DEBUG[10196]: app_meetme.c:2587 find_conf: 2222 isn't a valid conference |
17:14.50 | pta200 | -- <SIP/0004f211f05c-0a326d80> Playing 'conf-invalid' (language 'en') |
17:14.50 | pta200 | == Spawn extension (meetme-conf, login, 2) exited non-zero on 'SIP/0004f211f05c-0a326d80' |
17:14.51 | pta200 | -- Executing [h@meetme-conf:1] Playback("SIP/0004f211f05c-0a326d80", "sorry") in new stack |
17:14.51 | pta200 | WARNING[10196]: file.c:747 ast_readaudio_callback: Failed to write frame |
17:14.51 | pta200 | -- <SIP/0004f211f05c-0a326d80> Playing 'sorry' (language 'en') |
17:14.53 | pta200 | WARNING[10196]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/0004f211f05c-0a326d80 for sorry |
17:15.08 | Chainsaw | ~pb |
17:15.09 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
17:15.56 | Katty | so far i am still severely lacking in vitramin e, b1, b2, b6, b9, copper, and manganese |
17:16.28 | telnettech | katty: need to start taking your daily geritol it sounds like |
17:16.28 | ManxPower-work | n00bs flooding the channel, this looks like a Monday |
17:16.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:16.40 | Katty | telnettech: well we'll see where i'ma t after dinner. |
17:16.40 | Nugget | telnet is eeeeeeevil! |
17:17.02 | telnettech | good day (turd) nugget |
17:17.30 | ManxPower-work | Nugget: so is microsoft, but lots of poor sods still use their products. |
17:18.13 | Katty | nugget has a telnet script. |
17:18.19 | Katty | i doubt if he's really there |
17:19.43 | *** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu) |
17:20.00 | Katty | i've hit, or gone over, on the recommendation of b3, k, phosphorus, sodium, and selenium |
17:20.31 | Katty | not a surprise on the sodium there. |
17:21.00 | telnettech | everything seems to have sodium |
17:21.08 | Katty | yeah :< |
17:21.18 | [TK]D-Fender | Katty: Don't forget the arsenic, lithium, and lead! |
17:21.31 | Katty | :P |
17:21.45 | Katty | now this is interesting |
17:21.54 | Katty | a deficiency of selenium can cause dandruff |
17:22.29 | Katty | and loose skin. |
17:22.33 | Katty | looks at arm. |
17:22.49 | Katty | what does loose skin look like ^_- |
17:24.26 | [TK]D-Fender | Katty: http://farm2.static.flickr.com/1326/671376073_712b66aaaf_o.jpg |
17:24.39 | [TK]D-Fender | Katty: Oh.... and sorry :) |
17:24.56 | *** join/#asterisk jantypas (n=jantypas@42.sub-75-208-27.myvzw.com) |
17:25.12 | Katty | ohh |
17:25.13 | Katty | my. |
17:25.16 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
17:25.24 | Katty | closes tab. |
17:25.34 | Katty | well i certainly don't have that problem. |
17:25.59 | [TK]D-Fender | Katty: Now... what to take to deal with your neurosis.... hmmmm |
17:26.00 | *** part/#asterisk pta200 (n=paolo@goose.specialai.com) |
17:26.17 | Katty | well, if i had to guess. |
17:26.21 | Katty | based on current information |
17:26.24 | Katty | b1, b2, and b6 |
17:27.24 | [TK]D-Fender | facepalms |
17:27.37 | Katty | *hee* |
17:28.36 | *** join/#asterisk Blackvel (n=blackvel@84.57.75.0) |
17:30.02 | ManxPower-work | Katty: you are alive so things can't be too bad. |
17:30.09 | Blackvel | hi all. anyone with an isdn pbx (e.g euracom) AND patton smartnode media gateway 4634? I have both connected to (same) isdn ntba. now I get a problem with "busy on busy" and telco as long as patton s0 cable is plugged in (patton steals d-info information) |
17:30.31 | Chainsaw | Blackvel: I do have a Patton Smartnode 4634, but mine is attached to two regular ISDN BRIs. |
17:31.11 | Blackvel | Chainsaw: what could be the reason that patton "steals" from d-channel? |
17:31.30 | Chainsaw | Blackvel: Configuration issue, most likely. |
17:31.43 | Blackvel | i really thought both would be independent |
17:31.58 | Chainsaw | Blackvel: A configuration issue *on* the Patton. |
17:32.01 | Blackvel | when i concentrate on msdn |
17:32.04 | Blackvel | msn |
17:32.18 | Blackvel | somehow my patton sees my whole ntba bus :( |
17:32.48 | Blackvel | you do not have some other isdn equipment connected in parallel to ntba? |
17:33.14 | Chainsaw | Blackvel: No, I have two ISDN BRIs straight from British Telecom plugged into the 4634. |
17:34.31 | Chainsaw | Blackvel: I must warn you for the Patton. It likely has R4 firmware on it now. If you upgrade to R5, *nothing* will work anymore. |
17:35.05 | Chainsaw | Blackvel: Well, everything you care about like SIP will be thoroughly shafted. It seems the ISDN side remains working. |
17:35.29 | Blackvel | right. i invested ONE week in 2008 to get R5.2 build running |
17:35.43 | Blackvel | works quite well..besides "busy on busy" stealing and wrong telco message |
17:35.55 | Chainsaw | *nod* A week sounds right. I'm on R5.4 |
17:36.24 | p3nguin | How do I specify which conference number to use when using Page()? |
17:36.31 | Blackvel | my telco says on busy on busy: this dailed number is right now not assigned |
17:36.44 | Blackvel | and it is the patton which lets telco talk in that way |
17:36.52 | Blackvel | if i unplug patton everything is fine |
17:37.01 | Blackvel | well probably a little bug in config :) |
17:37.17 | Blackvel | is R5.3/R5.4 quite different? |
17:37.24 | Blackvel | is it save to upgrade? |
17:37.31 | Chainsaw | Likely. To rule out a bus termination problem you could connect only one device at a time. |
17:37.33 | *** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com) |
17:37.48 | Chainsaw | Compared to R4->R5 it is effortless, yes. |
17:38.10 | Blackvel | do you have an echo problem from time to e.g (e.g 5-10% in all cases)? |
17:38.19 | Chainsaw | No echo, no. |
17:38.26 | Blackvel | i know its always the other party (i hear me in echo locally) |
17:38.35 | [TK]D-Fender | p3nguin: You don't. Its dynamic |
17:38.36 | Blackvel | but patton / snoms can not compensate the inbound problem |
17:38.36 | Chainsaw | Neither on the Cisco handsets nor on the analog handsets (on a SmartNode 4118). |
17:38.51 | Blackvel | i really hate that...i thought even inbound can be compensated |
17:39.31 | Blackvel | how would you separate patton / euracom isdn on a nbta bus? |
17:39.42 | Blackvel | tey share d-channel? |
17:39.44 | Blackvel | tey |
17:39.46 | Blackvel | they |
17:39.57 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
17:40.01 | Chainsaw | Normally your NT1 has two outputs. |
17:40.10 | Blackvel | right |
17:40.25 | Blackvel | and they are separated? |
17:40.35 | Chainsaw | It could be one of those annoying faulty ones, in which case you'll want to use a 1-to-2 splitter on the port you were already using. |
17:40.52 | Chainsaw | But as I said, the simple test would be to disconnect the other device, connect only the Patton and then make some test calls. |
17:41.07 | Chainsaw | (I realise you may have to do that out of hours when the PBX isn't needed for other things) |
17:41.48 | *** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
17:44.07 | ryduh | hello everybody! |
17:44.22 | Chainsaw | Hi Dr. Nic.. eh, ryduh. |
17:44.51 | Blackvel | chainsaw: pardon? nt1 could be faulty? |
17:45.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
17:45.18 | Blackvel | what are those splitters doing you can use for one port? |
17:45.20 | Chainsaw | Blackvel: The secondary port on it could be. I've had that. The engineer who installed it stuck a T-splitter on the working port and left. |
17:45.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
17:45.36 | Blackvel | i guess they all share d-channel information. that is the problem in my case (i think) |
17:45.38 | louben | So Asterisk isn't supporting h323 and its newer versions anymore... Maybe are plans for the future? |
17:45.41 | Chainsaw | Blackvel: But still, I'd test *just* the Patton *without* the PBX. |
17:45.54 | Blackvel | well it works |
17:46.05 | Blackvel | it pickups my 2 business numbers (on ivr / snom) |
17:46.16 | Blackvel | also my isdn pbx works (euracom) |
17:46.31 | Blackvel | but not for busy on busy (opposite to call waiting) |
17:47.00 | Blackvel | it makes tuttuttuttuttut when msn is busy on euracom (without patton) |
17:47.45 | Blackvel | with my patton connected its no busy signal anymore but that weird telco announcement that this number is not assigned. i really dont get it ;) |
17:47.51 | DavidR2008 | I'm trying to fix this error: |
17:47.53 | DavidR2008 | Unable to support trunking on user 'asteriskm' without a timing interface |
17:47.55 | DavidR2008 | I have dahdi_dummy installed and the pseudo channel shows up when I issue a dahdi show channels, what do I need to do? |
17:48.12 | Blackvel | Chainsaw: how do you use the routing table? route <msn> ...? |
17:48.21 | Blackvel | i used that for my both msn's |
17:48.39 | Blackvel | even tried "route default none" this evening |
17:48.42 | Chainsaw | Blackvel: I'd have to look it up. I think you'll be best off contacting Patton about it. |
17:49.08 | Blackvel | i even forgot how to damn login into my patton to enable isdn tracing |
17:49.17 | Blackvel | forgot root / shell password |
17:49.18 | Blackvel | haha |
17:49.25 | Chainsaw | No configuration file backup? |
17:49.28 | Blackvel | well yes, probably |
17:49.32 | Blackvel | yes i have |
17:49.33 | Chainsaw | Just do what I did. |
17:49.37 | Blackvel | i can even login into gui |
17:49.42 | Chainsaw | Clear it out with the reset button, set a password you like. |
17:49.43 | Qwell | louben: Where did you get this information from? |
17:49.50 | Chainsaw | Then string the two configuration files together. |
17:50.03 | Chainsaw | Blackvel: If you can log into the GUI you can log into the telnet side of it. |
17:50.19 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
17:50.22 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
17:50.28 | louben | Qwell, we had a discussion with Corydon76-dig about that issue |
17:51.13 | Katty | blinks. |
17:51.27 | Katty | leifmadsen: holy crap, have you seen all the vitamins in Total? |
17:52.24 | Blackvel | chainsaw: what was the shell login user? Administrator (like gui)? |
17:52.31 | Chainsaw | Blackvel: Yes, administrator. |
17:52.40 | Chainsaw | Blackvel: It may require you to write it as admin there though. |
17:52.46 | Blackvel | ah |
17:52.50 | Blackvel | could be |
17:52.59 | Blackvel | how does your msn routing table look like? |
17:54.23 | Blackvel | you just put your business msn(n's) in which you like to pickup for incoming call? |
17:55.41 | Chainsaw | Blackvel: I do not have a degree in ISDN terminology. |
17:55.41 | leifmadsen | Katty: yes, but there is also a lot of sugar last I saw |
17:55.54 | leifmadsen | Katty: or maybe I'm thinking of Vector |
17:56.48 | Chainsaw | Blackvel: "MSN" |
17:57.29 | Blackvel | external number |
17:57.33 | Blackvel | called number |
17:57.42 | Katty | leifmadsen: i don't know. but it seems to contain all of the vitamins i'm lacking in. |
17:57.46 | Blackvel | called e164 |
17:57.55 | *** join/#asterisk d1ss3nt (n=nebula@s0up.digitalkharma.org) |
17:58.19 | d1ss3nt | can someone recommend a reliable single span T1/PRI card for asterisk that is not a digium? |
17:58.27 | leifmadsen | Katty: I'd check the sugar count on it to make sure it isn't too high -- sometimes they add a lot of that in order to make it "taste good" |
17:58.44 | Chainsaw | Blackvel: Ah, right. I check the end digits. |
17:59.11 | Chainsaw | Blackvel: And depending on whether it's a locally attached analog device, remotely attached analog device (other office) or a regular extension, send it on its way. |
17:59.39 | Katty | leifmadsen: 4.5g of sugar in 3/4c |
18:00.04 | kaldemar | d1ss3nt: sangoma ones are widely used |
18:00.06 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-haolasdondnmkbpi) |
18:00.48 | leifmadsen | Katty: that doesn't sound like too much |
18:01.10 | Katty | leifmadsen: no, it doesn't...but the real question is do i really want to stuff myself with all the additives and preservatives listed on the ingredient list. |
18:01.55 | leifmadsen | Katty: that's the key question |
18:02.10 | Katty | leifmadsen: i should probably just eat more lentils, bananas, and tuna. |
18:02.21 | leifmadsen | ideally you'd just make your own cereal -- oatmeal is good too, especially if you add some cinnamon, nuts, seeds, and berries |
18:02.30 | leifmadsen | Katty: berries too |
18:02.40 | Katty | mmmmmmmmm, berries. |
18:02.45 | leifmadsen | blueberries are amazingly good for you |
18:02.49 | Katty | yes, yes they are. |
18:02.52 | leifmadsen | anti-oxitant |
18:02.56 | thehar | mmmmm blueberries |
18:03.20 | Katty | salmon also has quite a bit of b12 |
18:03.28 | Katty | more so than tuna, actually |
18:03.54 | Katty | sounds like Salmon patties might be on the menu soon |
18:04.10 | ryduh | leifmadsen: antioxidant? |
18:04.16 | Corydon76-dig | louben: NO. What I told you is that we don't have experts on the code, so if you have trouble with compiling, you're best to ask a person who has worked on that code. |
18:04.31 | Corydon76-dig | louben: specifically, I told you to contact the person who wrote the most recent patch. |
18:04.44 | louben | Corydon76-dig, I did |
18:05.23 | Katty | ryduh: oxidation of cells leads to Free Radicals |
18:05.34 | Corydon76-dig | louben: did that work? |
18:05.36 | Katty | ryduh: bits of damaged molecules. |
18:05.47 | louben | Not really... |
18:05.53 | Katty | ryduh: specifically sells that are lacking an electron |
18:06.00 | Katty | ryduh: so they run around stealing electrons from other cells. |
18:06.11 | Qwell | Not the electrons! |
18:06.16 | Katty | yes. |
18:06.19 | ryduh | Katty: Sorry, no I was correcting his spelling/hypenation |
18:06.58 | Katty | ryduh: protect your electrons, eat a blueberry. |
18:07.13 | Jumpie | hmm im tryin to think of a really annoying wav file to play on blacklist match |
18:07.20 | louben | h323 should be mandandory for Large scale implementations |
18:07.32 | Corydon76-dig | Jumpie: tt-chickens |
18:07.35 | Chainsaw | Jumpie: Unannounced echo test. |
18:07.47 | Jumpie | Corydon76-dig im scared to ask... |
18:07.47 | louben | That is what I was told |
18:07.57 | ryduh | we-dont-have-tech-support |
18:08.02 | Chainsaw | Jumpie: I take it is some anti-telemarketing measure. |
18:08.09 | Jumpie | Chainsaw yeah |
18:08.13 | Jumpie | i have the same 5 people that call me daily |
18:08.18 | Jumpie | one is also about a studen loan of my ex wife |
18:08.18 | Corydon76-dig | louben: That's an interesting opinion |
18:08.20 | Jumpie | that is particular nasty |
18:08.27 | Chainsaw | I take unannounced echo test will freak them out more then a .wav |
18:08.33 | kaldemar | Jumpie: a simple "hello" with 20 seconds of silence and then a hangup. works like a charm every time. |
18:08.37 | Corydon76-dig | louben: it's contrary to virtually the entire voip industry |
18:08.39 | Jumpie | kaldemar lol |
18:08.49 | Jumpie | i cant find tt-chickens |
18:08.53 | Chainsaw | kaldemar: I approve! A .wav with hello and *then* the echo test. |
18:09.02 | Chainsaw | kaldemar: It'll feel like a bad line. |
18:09.15 | Corydon76-dig | Jumpie: look in the sounds-extras |
18:09.18 | Jumpie | im an idiot..how do i test what that echo test is? |
18:09.36 | kaldemar | Chainsaw: pure awesomeness |
18:09.37 | ryduh | Jumpie: you could record one saying they have been permanently blacklisted and to not bother calling back |
18:10.08 | Chainsaw | kaldemar: exten => s,n,Echo |
18:10.16 | Chainsaw | kaldemar: If they hit a # it'll continue. |
18:10.20 | Chainsaw | kaldemar: So you can stick more behind it. |
18:10.33 | Jumpie | Chainsaw....as long as it wont tie up the channel indefniitely |
18:10.37 | Jumpie | indefinitely |
18:10.38 | Corydon76-dig | Jumpie: tt-hangup is another good one |
18:10.38 | ryduh | Chainsaw: does that just echo everything they say until they hit a # |
18:10.42 | louben | So is asterisk going to stop it? |
18:10.42 | Naikrovek | it'll hang up when theyd o |
18:10.48 | Chainsaw | ryduh: It does. |
18:10.52 | Jumpie | rofl ryduh |
18:10.55 | Chainsaw | ryduh: Or until they hang up. |
18:11.00 | Corydon76-dig | Oh, and tt-chickens was renamed to nobody-but-chickens |
18:11.19 | Corydon76-dig | louben: stop what? |
18:11.38 | ryduh | Jumpie: what about a hello, Wait(10) then an Echo |
18:11.43 | louben | Corydon76-dig, supporting h323 |
18:11.46 | Jumpie | i like that ryduh |
18:11.55 | Chainsaw | ryduh: I'd start echoing immediately. Ever been on a bad speakerphone? |
18:12.01 | Corydon76-dig | louben: that depends upon what you mean by "support" |
18:12.08 | ryduh | Chainsaw: maybe? |
18:12.23 | Chainsaw | ryduh: It'll sound just the same to them. They'll be talking for up to a minute. |
18:12.25 | ryduh | brb- testing Echo |
18:12.37 | Chainsaw | ryduh: It's like a TCP tarpit for telemarketers. |
18:13.03 | louben | Corydon76-dig, I mean functionality of * with the latest features of h323 |
18:13.37 | louben | Corydon76-dig, ...that are written for openh323 |
18:13.41 | Corydon76-dig | louben: There are no plans to enhance H323, no |
18:13.42 | Jumpie | i love it |
18:13.49 | Jumpie | im tryin to think if i wanna do echo immediately or not |
18:13.58 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
18:14.02 | *** part/#asterisk diatonic1 (n=chillman@208.186.73.35) |
18:14.03 | Corydon76-dig | That could change in the next 5 minutes, 3 months, or never |
18:14.46 | louben | I see. Damn we would like to have this functionality |
18:15.06 | Corydon76-dig | Then again, this is open source. If YOU want to contribute that, then we're happy to take those contributions, as long as they fit within our architecture |
18:16.00 | ryduh | Chainsaw: Is there a way where I could use something like EAGI to autotune the caller? |
18:16.05 | ryduh | Chainsaw: I think that would be quite fun. |
18:16.07 | Corydon76-dig | That is, as a multi-protocol back-to-back-user-agent |
18:16.37 | Chainsaw | ryduh: I'm not sure, I've not used that. I like your creativity though :) |
18:17.04 | louben | Corydon76-dig, maybe. Now I'm looking forward for this |
18:17.14 | Jumpie | man..this is awesome |
18:21.02 | *** join/#asterisk bluOxigen (n=asad@119.73.68.102) |
18:21.22 | ryduh | I'm going to create a quick poll. How many of you have a dedicated server in house for asterisk? How many of you are using a VPS or dedicated host somewhere else? Obviously this is more for VoIP. |
18:21.41 | Chainsaw | ryduh: Dedicated, in-house. |
18:22.13 | *** join/#asterisk i-pink (n=my-pink@bzq-79-177-64-90.red.bezeqint.net) |
18:22.55 | i-pink | hii all |
18:23.09 | Chainsaw | Hi there. |
18:23.16 | beek | ryduh: ditto Chainsaw |
18:23.39 | i-pink | i need help with make conference room base on sip |
18:25.13 | Chainsaw | Do you mean a dial-in conference bridge? |
18:25.45 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
18:26.11 | ryduh | Failrar: where does the inspiration for your nick come from? |
18:26.24 | *** join/#asterisk aleshus (n=aleshus@81.0.207.104) |
18:27.21 | Failrar | no idea |
18:27.32 | aleshus | Hi to all.. Can anybody help me with asterisk? I need to configure it with authenticate users from LDAP.. |
18:28.12 | [TK]D-Fender | ryduh: Vast majority = inhouse |
18:28.18 | ryduh | Failrar: inspirationfail |
18:28.28 | *** join/#asterisk QaDeS (n=mklaus@p4FC72A2F.dip0.t-ipconnect.de) |
18:28.37 | Chainsaw | Mine's based on my favourite Doom II weapon. |
18:28.43 | Chainsaw | Computer games moved on but the name stuck :) |
18:28.55 | [TK]D-Fender | rider : hukt on fonix fale |
18:28.58 | asterwiki | ryduh:dedicated in-house |
18:30.09 | louben | is away: I'm busy |
18:30.38 | Chainsaw | watches to channel grind to halt as everyone waits for louben to return from his errand |
18:30.51 | aleshus | nobody have an experience? |
18:32.39 | louben | Chainsaw, sorry I'm also chatting in other networks |
18:32.52 | Qwell | louben: turn off your public away message |
18:33.27 | Blackvel | chainsaw: my experience from patton debug: it gets the call on the NT1 and disconnects it again: Cause NoRouteToDestaination |
18:33.31 | louben | sure |
18:33.42 | Blackvel | Chainsaw: to me the debug is the same (for #1 new call or #2 call where 'busy on busy' should follow |
18:34.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:34.37 | Blackvel | i am debugging layer2/layer3 |
18:35.07 | *** join/#asterisk adam0563 (n=damagele@70.103.115.194) |
18:35.07 | Blackvel | i can not find any information from euracom about "busy on busy" and an special release cause on patton :( |
18:35.59 | Blackvel | so when patton does not pickup the #2 call - euracom should send busy. still figuring out where the telco gets the returncode about a different release cause message |
18:36.08 | Chainsaw | "Busy on busy" has no meaning to me in the first place. |
18:36.35 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
18:37.14 | adam0563 | are certain channels assigned numbers? |
18:37.25 | adam0563 | or a group of channels? |
18:38.54 | p3nguin | What am I doing wrong here? http://pastebin.ca/1638014 |
18:39.42 | Blackvel | Chainsaw: "busy on busy" means when the dailed number is busy (because phone1 picked up) it should send busy instead let phone 2/3 ring |
18:39.53 | Chainsaw | p3nguin: That "unable to open pseudo device" suggests you have no DAHDI timing. |
18:40.27 | Blackvel | or let phone1 handle call waiting (knock knock) |
18:41.42 | p3nguin | chainsaw: I'm running purely SIP and IAX2, and the call to Page() is SIP to SIP. What does dahdi have to do with anything in this case? |
18:41.58 | Chainsaw | p3nguin: In order to do conferencing, a timing source is required. |
18:42.17 | Chainsaw | p3nguin: You can do that without further hardware, but you'll have to install the "dummy" driver of DAHDI (or Zaptel, if your Asterisk is that old). |
18:42.17 | p3nguin | Okay, so I should probably look into ztdummy, right? |
18:42.26 | Chainsaw | Indeed. |
18:42.53 | p3nguin | Asterisk 1.4.24.1 |
18:43.07 | Chainsaw | *waggles hand* Either one would work. |
18:43.41 | *** join/#asterisk madsara (i=madsara@2001:328:2002:f159:0:0:0:1) |
18:43.50 | kaldemar | app Page needs dahdi or zaptel timing. go for dahdi since 1.4.24.1 supports it. |
18:44.25 | madsara | Hey, what's the easiest way to prepend a wav file to a voicemail, so when it's checked a prefacing audio snippit of "This is a so-and-so-customer" plays. |
18:45.02 | ryduh | Another poll. For your dedicated inhouse server, What do you run as your OS? Centos? Or something packaged like trixbox? |
18:45.30 | madsara | ryduh: Centos |
18:46.08 | p3nguin | ryduh: FreeBSD |
18:46.11 | Qwell | ~polls |
18:46.12 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
18:46.30 | asterwiki | ryduh: Several inhouse, distributed several servers (CentOS, Ubuntu, FreeBSD) |
18:46.36 | Chainsaw | ryduh: Gentoo (non-multilib AMD64 hardened) |
18:47.23 | ryduh | Qwell: I'm not asking who runs what to then ask them a question. I'm genuinely interested in what people are running. Is that forbidden? |
18:47.40 | Blackvel | Chainsaw: is layer2 d-channel and layer3 b-channel or quite different than that? |
18:47.52 | Qwell | ryduh: If all 254 people in the channel answered, that would be bad. |
18:48.00 | Chainsaw | Blackvel: D-channel has your call setup information. |
18:48.03 | Chainsaw | Blackvel: B-channel carries voice. |
18:48.04 | [TK]D-Fender | Multiplied byt he number of questions... |
18:48.09 | *** join/#asterisk clyons (n=clyons@unaffiliated/clyons) |
18:48.36 | p3nguin | carry the 4... |
18:48.36 | ryduh | Qwell: good point. Is there a better way to ask people what they are running? Maybe linking to a poll on a website? |
18:48.44 | p3nguin | divide by Pi... |
18:49.14 | Blackvel | Chainsaw: i can debug isdn layer2 and layer3. are they both d-channel? |
18:49.16 | Qwell | ryduh: how about researching it using information that's widely available? |
18:50.06 | ryduh | Qwell: fine. I'll stop the polls. |
18:50.09 | Chainsaw | Blackvel: As I said before, I do not have a degree in ISDN. I just send calls over it to BT. And sometimes they send me some back, if they're in a good mood. That's it. |
18:50.57 | *** join/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
18:51.12 | Blackvel | Chainsaw: our philosohpy looks like to be the same :) |
18:52.09 | Blackvel | how long did it take you to get the patton setup up and running? |
18:52.15 | Chainsaw | Qwell: Did you see my bug? Or is DAHDI not your area? |
18:52.29 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
18:52.29 | Chainsaw | Blackvel: I inherited this setup, working, on R4. |
18:52.41 | Chainsaw | Blackvel: It took me about a week to convert it all over to R5.2 and have it working properly. |
18:53.59 | Blackvel | looks like 0->100 is not too bad in one week :) |
18:54.56 | Blackvel | how many people use pattons even? |
18:55.16 | Chainsaw | I think most just pop in a Sangoma card to be honest. |
18:55.43 | *** join/#asterisk TimToady_ (n=moi@adsl320-234.kln.forthnet.gr) |
18:55.51 | Chainsaw | In my case it made sense from a failover point of view. There's 2x ISDN BRI and 8x analog FXO. |
18:56.07 | adam0563 | how do I change the number that DAHDI uses? |
18:56.08 | Chainsaw | In both offices. If the connection to Asterisk fails, the calls go straight from ISDN BRI to the analog handsets. |
18:57.12 | [TK]D-Fender | adam0563: Uses for what? What kind of channel? |
18:58.00 | adam0563 | we have a few numbers assigned to us and I need to change the number that goes out on our asterisk box |
18:58.03 | Blackvel | Chainsaw: where do you connect your analog phones? fxo box? fxo connected to asterisk? |
18:58.17 | Chainsaw | Blackvel: Patton 4118 gateway. |
18:58.29 | Chainsaw | Blackvel: Both of the gateways connect to Asterisk in normal operation. |
18:58.30 | [TK]D-Fender | adam0563: what kind of channels? |
18:58.38 | Chainsaw | Blackvel: It's just the fallback path that they talk directly. |
18:59.02 | *** join/#asterisk titter (n=titter@c-98-208-156-75.hsd1.fl.comcast.net) |
18:59.04 | Blackvel | Chainsaw: i checked sangoma before and noted that the drivers didnt always enable HWEC. just wanted to get around problems with my little epia via nemiah 1gig asterisk server (maybe too small for sw echo cancel) |
18:59.19 | Chainsaw | Blackvel: ...whoa? |
18:59.56 | Blackvel | i guess patton is not a bad thing....if you have time to or support to configure. when its done it works :) |
19:00.10 | *** join/#asterisk lewellyn (n=lewellyn@greenviolet/lewellyn) |
19:00.25 | Chainsaw | Blackvel: Once you're done configuring it, it's definitely one of those "Nobody $#@* touch that. Ever!" devices, true. |
19:00.56 | p3nguin | [tk]d-fender: I still can't see any valid reason to include internal if there is a goto() that sends to call to the internal context. It just doesn't make any sense. I tried it the way you wanted me to do it and the include never gets touched because the goto() sends the call to a completely different context. |
19:01.01 | adam0563 | D-Fender: not sure what you mean by type of channel. We are connected through a T1 line so we use DAHDI |
19:01.28 | [TK]D-Fender | p3nguin: it doesn't get touched because you put an overlapping pattering in the cotext you're doing your include in <- |
19:01.35 | [TK]D-Fender | p3nguin: You are creating the conflict |
19:01.50 | [TK]D-Fender | adam0563: what SIGNALING?: |
19:02.10 | Chainsaw | hands Fender the megaphone |
19:02.25 | [TK]D-Fender | reaches for his ClueBat (tm) |
19:03.15 | adam0563 | pri_cpe |
19:03.30 | p3nguin | [tk]d-fender: If I have a Goto(), there's no logical reason to use an include. If I use an include, there's no logical reason to use Goto(). |
19:03.38 | Blackvel | so you have 4118 connected to asterisk and another patton isdn bri gw connected to NT1, asterisk and for backup directly to 4118 (over network routing)? |
19:04.01 | [TK]D-Fender | p3nguin: Are you ever going to put something in [internal] that the menu won't have access to? |
19:04.11 | [TK]D-Fender | adam0563: then jsut set the CALLErID() |
19:04.20 | p3nguin | [tk]d-fender: I doubt it. |
19:04.57 | p3nguin | [tk]d-fender: The internal context is purely for "local extensions" to be able to be dialed and reach another phone in the system. |
19:05.06 | [TK]D-Fender | p3nguin: then all you'll end up doing is running the risk of dropping calls due to lack of a pattern match on your range, and missing things that don't fit into the goto you have. |
19:05.31 | [TK]D-Fender | p3nguin: p3nguin and slowing down dialplan execution but wassting time on processing an app |
19:05.58 | p3nguin | [tk]d-fender: So are you advising that I can include internal and remove the Goto(), right? |
19:05.59 | [TK]D-Fender | p3nguin: the point of includes is so you don't run into the myriad kinds of stupidity you can create with Goto. |
19:06.26 | ryduh | Why would this context not recognize when I press an extension? http://pastebin.com/d54d1b7c8 I can press 8 and it will not recognize it |
19:06.29 | [TK]D-Fender | "Oh shit, I need extens in the 300 range! Now I haev to modify 5 IVR's!" |
19:06.57 | [TK]D-Fender | Rybad DTMFMODE on the incoming channel |
19:07.02 | [TK]D-Fender | ryduh: bad DTMFMODE on the incoming channel |
19:07.46 | *** part/#asterisk JoeMoretti (n=jmoretti@76.164.171.81) |
19:07.53 | ryduh | [TK]D-Fender: I set dtmfmode=rfc2833 in sip.conf. I'm using a call file to originate this call |
19:08.15 | [TK]D-Fender | ryduh: I don't see a call to look at with SIP DEBUG... |
19:08.17 | [TK]D-Fender | ~pb |
19:08.18 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
19:08.20 | [TK]D-Fender | ^^^^^^ |
19:08.33 | ryduh | k |
19:08.34 | p3nguin | See, for that scenario using my current configuration with the Goto() in place, I would just change exten => _[1-2]XX,n,Goto(internal,${EXTEN},1) to exten => _[1-3]XX,n,Goto(internal,${EXTEN},1) |
19:08.47 | p3nguin | Now I can dial 100s, 200s, and 300s. |
19:08.52 | [TK]D-Fender | p3nguin: and update 3-4 menus that may reference it. |
19:09.07 | p3nguin | ponders |
19:09.09 | [TK]D-Fender | p3nguin: Your way asks for trouble, wastes processing and adds work. |
19:09.28 | [TK]D-Fender | p3nguin: there isn't a single stated advantage to using a Goto, and tons of bad. |
19:09.45 | [TK]D-Fender | p3nguin: the entire point of contexts is to use includes to build a heirarchy |
19:09.47 | p3nguin | I'm trying to weigh that factor. |
19:09.54 | [TK]D-Fender | p3nguin: so that you avoid duplication. |
19:10.14 | [TK]D-Fender | p3nguin: And you're doing your best to think backwards |
19:11.34 | adam0563 | Fender: how do I set the caller ID on an outbound DAHDI call? |
19:11.46 | p3nguin | I'm doing my best to logically rationalize everything. |
19:11.48 | [TK]D-Fender | adam0563: "core show function CALLERID" <- |
19:12.06 | [TK]D-Fender | p3nguin: its the "rational" part you're having trouble with ;) |
19:12.45 | p3nguin | At least my logic is still good. |
19:13.44 | ManxPower-work | I think I may have figured out an explanation for how FreePBX works. It was written by drunken college students after they dropped some really good LSD. |
19:13.46 | [TK]D-Fender | p3nguin: or more like a well formulated bad idea ;) |
19:14.24 | [TK]D-Fender | ManxPower-work: Lick the frog instead of clicking it ;) |
19:14.28 | DavidR2008 | how reliable is res_timing_pthread? |
19:15.09 | ryduh | is it normal for AMI to take 10-15 seconds before originating a call? |
19:15.44 | *** join/#asterisk denon (i=denon@sassinak.net) |
19:15.44 | *** mode/#asterisk [+o denon] by ChanServ |
19:15.59 | adam0563 | D-Fender: exten => _801NXXXXXX,3,Set(CALLERID(num)=xxxxxxxxx) <- am I doing something wrong? |
19:16.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:16.29 | adam0563 | D-Fender, I replaced the number with the small x's |
19:16.44 | [TK]D-Fender | adam0563: Not based on that line alone. enable PRI debug and see what's getting passed <- |
19:16.53 | p3nguin | Is this for an incoming call or outgoing? |
19:17.00 | [TK]D-Fender | adam0563: And show us GOOD debug not masking ANYTHING |
19:17.24 | p3nguin | adam0563: [tk]d-fender won't call you. He promises. |
19:17.33 | adam0563 | ha ha |
19:17.45 | adam0563 | exten => _801NXXXXXX,3,Set(CALLERID(num)=18019911000) |
19:17.48 | adam0563 | that is what I have |
19:17.53 | [TK]D-Fender | adam0563: Not based on that line alone. enable PRI debug and see what's getting passed <- |
19:18.31 | p3nguin | If that's for an incoming call, you're going to be overwriting whatever the real caller ID is with that number. Was that what you wanted to do? |
19:19.43 | p3nguin | And most callerID data seems to have a 10-digit number, so drop the 1 off the front of your number. |
19:20.50 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
19:20.58 | adam0563 | it's for outgoing |
19:21.22 | CcRnp | what is the maximum sample rate asterisk can support for MOH ? |
19:21.37 | ManxPower-work | Caller*ID in NANPA (USA, Canada, a couple of other countries) is 10 digits with no quotes, dashes, or leading 1 |
19:21.46 | ManxPower-work | no spaces either |
19:22.07 | p3nguin | Okay, so that line says that any number you are calling in the 801 area code you want to set your CID to 8019911000. Is that your goal? |
19:23.20 | adam0563 | yes |
19:23.20 | adam0563 | I want to do it to every number but that's in different dial plans so I'll be modifying them later |
19:23.20 | p3nguin | ccrnp: As far as I know, it must be 8000 Hz. |
19:23.41 | ManxPower-work | There is no max, no min. It's 8000 Hz, MONO or it won't work |
19:23.47 | adam0563 | however, I think the signal is actually em_w |
19:23.55 | adam0563 | not pri |
19:24.02 | ManxPower-work | adam0563: you can't set outgoing callerid on E&M |
19:24.02 | CcRnp | Thank you guys !! appriciated |
19:24.09 | adam0563 | because I don't have pri debugging available in the asterisk cli |
19:24.30 | p3nguin | exten => _801NXXXXXX,3,Set(CALLERID(num)=8019911000) |
19:24.31 | p3nguin | exten => _801NXXXXXX,4,Dial(IAX2/account@context/${EXTEN}) |
19:24.36 | ManxPower-work | adam0563: did you install libpri before you installed Asterisk? |
19:25.24 | ManxPower-work | adam0563: You need to find out EXACTLY what kind of line you have before you do anything else. |
19:25.29 | adam0563 | I wasn't the one that installed the Asterisk machine which is why I'm having this problem. The person who set it up put like 3 asterisk installs on here so I have a tough time because everything is disorganized |
19:25.48 | ManxPower-work | adam0563: what is the signaling set to? |
19:27.47 | [TK]D-Fender | adam0563: .... where is my PRI debug? Obviously have have libpri, etc installed otherwise you wouldn't be placing calls |
19:28.02 | adam0563 | okay, correct me if I'm wrong, but my dahdi-channels.conf says pri and uses the from-pstn context, so it should be going through pri |
19:28.16 | ManxPower-work | EXACTLY what says "pri"? |
19:28.22 | adam0563 | the thing I am confused about is that there is no pri debugging in there, is that a seperate module? |
19:28.44 | p3nguin | Assuming dahdi is like SIP, the dialplan determines where calls go out. |
19:28.55 | adam0563 | http://pastebin.com/m15fef56a |
19:28.56 | ManxPower-work | adam0563: yes. It is part of the PRI support for Asterisk. Looks like you are not running a PRI. |
19:29.01 | p3nguin | The dahdi conf would just determine what happens to calls that come in. |
19:29.25 | [TK]D-Fender | adam0563: Please ignro adam0563 ENABLE IT |
19:29.32 | [TK]D-Fender | gah |
19:29.34 | [TK]D-Fender | aasdhasasjhdaslyuioweynr |
19:29.51 | ManxPower-work | adam0563: that is /etc/asterisk/chan_dahdi.conf? |
19:29.51 | [TK]D-Fender | adam0563: Ok, "pri debug span 1" |
19:29.52 | *** part/#asterisk E-bola (i=bola@ip181.rev112.brygge.net) |
19:29.52 | adam0563 | yes |
19:29.55 | [TK]D-Fender | adam0563: go to CLI. Place a call. Pastebin it. |
19:30.53 | adam0563 | http://pastebin.com/m3a0d287e |
19:31.11 | titter | if you enabled dahdi, and edited the show_dahdi.conf you will need to stop the asterisk service, and restart it |
19:32.14 | ManxPower-work | [TK]D-Fender: Best of luck helping adam0563. You'll need it. |
19:32.59 | [TK]D-Fender | adam0563: enable PRI DEBUG like I dtold you and include proof of its activation in your pastebin <- |
19:33.51 | *** join/#asterisk Hexbomber (n=Louis@S0106004854811dff.wp.shawcable.net) |
19:34.23 | adam0563 | in the asterisk CLI, I type in help and it comes up with the commands, there is nothing even related to PRI |
19:34.33 | adam0563 | therefore, when I type in PRI DEBUG it says it can't do anything |
19:35.07 | ManxPower-work | this is just too painful to watch |
19:35.09 | *** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
19:35.14 | p3nguin | hahahahaha |
19:35.23 | [TK]D-Fender | adam0563: pastebin your system.conf |
19:35.48 | [TK]D-Fender | adam0563: and show me the attempt to do the command I told you to do |
19:35.49 | adam0563 | ha ha...... |
19:35.55 | adam0563 | no system.conf |
19:35.59 | [TK]D-Fender | WTF |
19:36.07 | [TK]D-Fender | adam0563: DAHDI REQUIRES IT. |
19:36.16 | [TK]D-Fender | calls BULLSHIT |
19:36.17 | adam0563 | wait a sec |
19:36.18 | adam0563 | sorry |
19:36.26 | [TK]D-Fender | reaches for his ClueBat (tm) |
19:36.34 | beek | ducks |
19:36.36 | adam0563 | I'm working in like 20 different directories here |
19:37.14 | adam0563 | http://pastebin.com/m2f35bbc7 |
19:37.37 | [TK]D-Fender | adam0563: that is not a &#@^$ing PRI |
19:37.47 | [TK]D-Fender | e&m=1-6 |
19:38.08 | [TK]D-Fender | adam0563: So FORGET the CALLERID() function. |
19:38.24 | adam0563 | I told you earlier, there are duplicates of these files and its disorganized, yell at the guy who did this stupid system before me, it says pri in a different config file |
19:38.33 | Hexbomber | I'm going to admit, I'm a total asterisk noob. I've downloaded / installed asterisk + freepbx on my linux server, yet now I'm not sure what to do haha... I've been looking for DID's, but there are none in my area code. |
19:38.39 | [TK]D-Fender | adam0563: there is no explicit mechanism for setting CID. Check with your vendor to see if they support "wink + DTMF", etc for setting it |
19:39.01 | Hexbomber | In theory I'd like to setup a system where somone calls my DID, and is prompted to press a button, and it transfers them to various numbers based on what they enter. |
19:39.04 | [TK]D-Fender | adam0563: And system.conf begs to differ and says whats what instantly. |
19:39.09 | [TK]D-Fender | adam0563: So pay attention. |
19:39.38 | [TK]D-Fender | Hexbomber: very doable. Now go get yourself that DID |
19:40.32 | Hexbomber | D-Fender: Is it possible that a DID does not exist in my area.. I mean, I live in a fairly large city, with a decent sized area code.. but I can't find providers anywhere. |
19:41.00 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
19:41.11 | superbeef | check out this madness |
19:41.14 | superbeef | http://pastebin.ca/1638165 |
19:41.36 | [TK]D-Fender | heWhere have you looked? |
19:41.41 | [TK]D-Fender | Hexbomber: Where have you looked? |
19:42.19 | Hexbomber | flowroute, as well as a bunch of other providers that I've googled online. |
19:42.22 | doolittlework | something strange is cooking, had a power faulure system boots no problem, but for some reason the command restart now in the cli is missing, am i going bonkers i am sure there is a command "restart now"? |
19:42.23 | Hexbomber | I'm in the 204 area code. |
19:44.12 | titter | doolittlework: service asterisk stop |
19:44.18 | titter | then start it again |
19:44.31 | [TK]D-Fender | Hexbomber: Waht city |
19:44.54 | Hexbomber | Winnipeg, MB, Canada. |
19:45.06 | doolittlework | titter: did that no luck the command is still missing |
19:45.14 | titter | don't do that in the cli |
19:45.38 | [TK]D-Fender | Hexbomber: voip.ms |
19:45.55 | adam0563 | D-Fender, how can I get the number changed that it calls from? It needs to be from all numbers anyways, I'm assuming I do it somewhere in the DAHDI config files, I just need that number changed. |
19:46.07 | [TK]D-Fender | nexles.net |
19:46.13 | doolittlework | what does the syatem setting netplug do in services? |
19:46.27 | [TK]D-Fender | [15:38]<[TK]D-Fender>adam0563: there is no explicit mechanism for setting CID. Check with your vendor to see if they support "wink + DTMF", etc for setting it |
19:46.41 | [TK]D-Fender | adam0563: PAY ATTENTION |
19:47.08 | adam0563 | I got all that, but there has to be a way to just change the number. Are you telling me that there is no way to do that? |
19:47.19 | [TK]D-Fender | Hexbomber: les.net |
19:47.39 | adam0563 | when I called the provider they told me that the number on outgoing calls was through Asterisk. |
19:47.54 | *** join/#asterisk xpot-mobile (n=xpot@173.8.94.1) |
19:48.16 | titter | adam0563: if your co is allowing out bound caller id numbers, that would tell me you are doing something wrong. |
19:48.30 | [TK]D-Fender | adam0563: Feel fre to pick up some communications skills for your next call with them. |
19:48.43 | [TK]D-Fender | titter: Generic and non-applicable. |
19:49.47 | doolittlework | just loaded * 1.6 did they remove the command " restart now" in cli |
19:49.57 | titter | works for me on 1.6.10 |
19:50.15 | doolittlework | strange |
19:50.29 | leifmadsen | doolittlework: core restart now |
19:50.30 | doolittlework | i sure i used before i had a power failure |
19:51.02 | adam0563 | okay. Thanks D-Fender. I'm sorry. |
19:51.04 | *** part/#asterisk adam0563 (n=damagele@70.103.115.194) |
19:51.04 | doolittlework | @leifmadsen: ta |
19:53.00 | doolittlework | i have 2 * boxes linking with iax, was making test calls between them before power failure on one server, iax show peers os Ok on both but for some reason i can not dial of the other server |
19:56.30 | Hexbomber | Okay, so now I just fubar'd my database setup... |
19:56.45 | Hexbomber | I changed my sql password for freepbx, and now it won't let me connect. |
19:56.59 | [TK]D-Fender | ~freepbx |
19:57.00 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
19:57.01 | [TK]D-Fender | ^^^^^6 |
19:57.15 | Hexbomber | thx :) |
19:57.24 | asterwiki | is on a break and back in the forum |
20:00.23 | doolittlework | is there a way to soft hangup all busy channels at once |
20:00.42 | *** join/#asterisk andres833 (n=andres83@201.244.125.6) |
20:00.43 | Qwell | doolittlework: restart |
20:01.16 | doolittlework | lol |
20:01.33 | doolittlework | pulling the power also works |
20:03.53 | doolittlework | i was meaning a command like softhangp * |
20:05.50 | asterwiki | doolittlework: google this 'soft hangup <channel> ' |
20:06.17 | DavidR2008 | I'm seeing DTMF begin .... and DTMF end ... messages on my console, but I have verbose set to zero, where are these messages coming from? * 1.6.1 |
20:08.56 | [TK]D-Fender | checkout time, later all |
20:09.38 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
20:10.11 | coppice | Checkout time? You work in a supermarket? :-\ |
20:10.30 | brunner | I need to hire some asterisk developers. I don't suppose there would be a job site that would be better than average for finding asterisk devs, right? |
20:11.56 | DavidR2008 | what kind of work? |
20:12.29 | brunner | dialplan development, mostly, but basic knowledge of PHP and MySQL would be a huge plus |
20:13.33 | ryduh | brunner: what kind of dialplan work are you looking to have done for you? and what are you expecting to pay? |
20:13.43 | mchou | lol |
20:13.51 | mchou | negotiating already :) |
20:13.55 | *** join/#asterisk errotan (n=errotan@81.0.115.122) |
20:14.25 | brunner | the pay would be negotiated one-on-one... it's just some interactive stuff with chanspy that would allow the user to control things that are going on in a meetme |
20:14.33 | doolittlework | what is php &mysql can one eat it? |
20:14.48 | DavidR2008 | no, but it can eat you ;-) |
20:14.57 | doolittlework | lol |
20:15.48 | doolittlework | anyone here dcap cert |
20:15.50 | doolittlework | ? |
20:16.09 | ryduh | brunner: I'm a PHP/MySQL pro and just getting into asterisk. I haven't used chanspy or meetme yet. |
20:16.44 | doolittlework | free pbx has a meetme module, to get your feet wet |
20:16.47 | brunner | ryduh: I still might be able to hire you for other things, but I need an asterisk pro more urgently |
20:18.15 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
20:21.07 | ryduh | doolittlework: I haven't messed around with freepbx. From reading the ~freepbx notice I'm not sure if I want to get into it. |
20:22.11 | ryduh | If I'm comfortable developing my own dialplans, is there any need to use freepbx? |
20:22.38 | *** join/#asterisk knobo (n=user@90.149.4.182) |
20:22.48 | Qwell | ryduh: nope |
20:23.23 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
20:23.29 | eppigy | 8[] |
20:24.33 | ryduh | what is that infobot msg from? i love 8[] |
20:24.44 | eppigy | I am dave |
20:25.51 | *** join/#asterisk gardo (n=gardo@110.55.240.37) |
20:27.19 | ryduh | brunner: what are you looking to do with php aside from *? |
20:27.47 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
20:27.54 | Katty | scowls. |
20:28.00 | Katty | pouts at isp. |
20:28.07 | p3nguin | scowls too |
20:28.17 | Katty | gives p3nguin a hug :> |
20:28.36 | p3nguin | :D |
20:28.58 | p3nguin | How'd the chicken turn out? |
20:29.25 | Katty | juicy! :> |
20:29.36 | thehar | nom |
20:29.40 | Katty | and perhaps a bit too salty |
20:30.28 | p3nguin | Aww. What do you do when you put too much salt in? |
20:30.42 | Katty | yeah the recipe listed too much seasoned salt |
20:30.43 | eppigy | http://www.youtube.com/watch?v=-M49SdszkH4 |
20:30.49 | eppigy | speaking of penguins |
20:30.54 | Katty | i think i'll make my own Seasoned Salt, minus the salt. |
20:31.33 | jblack | Not a bad idea. You can get it pretty cheap in the store too |
20:31.59 | p3nguin | salt-free seasoned salt? |
20:32.23 | jblack | Usually it's called seasoning. |
20:32.24 | p3nguin | I think they call that "seasoning" |
20:32.34 | eppigy | ceasoning |
20:32.36 | Katty | :>>>>>> |
20:32.51 | jblack | No need to be rude, p3nguin. :) |
20:32.58 | Katty | i'm glad mister penguin got away from the killer whales. |
20:32.59 | *** join/#asterisk denon (i=denon@69.165.165.115) |
20:32.59 | *** mode/#asterisk [+o denon] by ChanServ |
20:33.01 | p3nguin | shakes fist at jblack |
20:33.32 | Katty | tho i am very sure that pod of whales was sad about the whole situation and dinner getting away. |
20:33.34 | jblack | flips p3nguin the bird |
20:33.48 | jblack | Get it? Penguin? bird? |
20:33.54 | ryduh | lol |
20:34.30 | *** join/#asterisk guywith2names (n=chatzill@69.70.201.246) |
20:34.43 | ryduh | guywith2names: what's your other name? |
20:35.04 | guywith2names | lol i dont think you want to know |
20:35.12 | guywith2names | its redundant |
20:35.52 | eppigy | Katty: yeah the penguin looks pretty thrilled |
20:35.56 | guywith2names | can anyone here help a noobie out? |
20:35.59 | eppigy | and ppossibly trying to blend in |
20:36.02 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-63.link.net.pk) |
20:36.10 | p3nguin | guywith2names: You'll have to actually address a specific topic. |
20:36.19 | ryduh | guywith2names: and ask a question |
20:36.21 | jblack | guywith2names: Yes. Anybody can. |
20:36.30 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
20:36.58 | guywith2names | Sure. I have access to my asterisk server from my provider through ssh. i need to be able to edit manager.conf file but i don't know how to access it |
20:37.21 | jblack | You have to ask your provider for instructions. |
20:37.21 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
20:37.43 | p3nguin | Or just use a text editor of your choice. |
20:37.52 | ryduh | guywith2names: is it in /etc/asterisk/ ? |
20:37.56 | guywith2names | yes but how do i get to it lol |
20:38.01 | Katty | eppigy: he did an excellent job. |
20:38.04 | guywith2names | i beleive so but i don't kno how to get there |
20:38.10 | Katty | eppigy: now he can take snacks home to the kids. |
20:38.11 | p3nguin | vim /etc/asterisk/manager.conf |
20:38.12 | jblack | p3nguin: Do you know where manager.conf is on that system? Whether or not he's going to require su, if he needs to run a special binary to restart *? etc etc etc |
20:38.30 | jblack | guywith2names: Ask the provider for help. Really. |
20:38.40 | jblack | p3nguin: sorry for being rude. |
20:38.44 | guywith2names | im sorry i really suck at linux... all i know is ls |
20:38.46 | guywith2names | lol |
20:38.53 | p3nguin | shakes fist at jblack again |
20:38.58 | guywith2names | let me try what penguin sugested |
20:39.13 | ryduh | guywith2names: are you on windows? If yes, have you downloaded PuTTY? If not, do you know how to open a terminal |
20:39.24 | p3nguin | I made like four assumptions when I typed that command, just so you know. |
20:39.29 | guywith2names | yes i am using putty to ssh into the server |
20:39.31 | ryduh | lol @ telling a noobie to use vim |
20:39.38 | jblack | Really bad idea guys. He should be asking for help. |
20:39.43 | kaldemar | vi is probably the last editor you want to open if you have absolutely no idea how to use it. |
20:39.44 | p3nguin | I don't know why that's funny to you. |
20:39.53 | Qwell | p3nguin: ed would have been the better choice. |
20:40.23 | jblack | sed -i the best of all |
20:40.26 | ryduh | p3nguin: just because it has somewhat of a learning curve. I guess it's not too terribly funny |
20:40.30 | guywith2names | i have some experience with linux |
20:40.49 | guywith2names | vim is an editor if im not mistakend |
20:40.54 | guywith2names | anyhow lemme try that |
20:40.56 | p3nguin | It's not that hard to press i to insert some stuff, delete or x to delete it, and ZZ to write and quit. |
20:41.04 | jblack | guywith2names: ask your provider for help. Really. You'll thank me later. |
20:41.30 | jblack | I'm not shoving you off to be mean. I really think it's appropriate for a variety of reasons. |
20:42.04 | guywith2names | jblack i made sure he made a backup before i mess around with it |
20:42.08 | ryduh | p3nguin: wow, I can't believe I forgot about ZZ. gotta start using that again |
20:42.23 | ryduh | guywith2names: who is your prodiver? |
20:42.23 | guywith2names | im just doing a little modifications |
20:42.42 | guywith2names | well its actually a friend of mine lol |
20:42.45 | kaldemar | p3nguin: it's easy, if you know what buttons to press. |
20:42.51 | jblack | Do you have control over the account asterisk is running as, either directly, or indirectly as root? |
20:43.04 | guywith2names | yes |
20:43.13 | guywith2names | i think... |
20:43.19 | *** join/#asterisk esaym153 (n=esaym153@cpe-24-174-176-203.satx.res.rr.com) |
20:43.30 | jblack | gives p3nguin a look |
20:44.11 | guywith2names | thanks for the advice, i should come here more often :D, can i get some paid private support here too? |
20:44.14 | jblack | like p3nguin said, the default directory for configuration files for asterisk is /etc/asterisk . Don't forget to tell asterisk to reload hte config file in the ami |
20:44.34 | esaym153 | howdy, does asterisk have any options to manage echo in a full voip environment (ie, no pstn or psi). I get my service from an upstream sip/iax termination.. |
20:44.36 | jblack | Definitely. There's several people that'll probably hit you up in about 5 minutes. |
20:45.12 | guywith2names | Sure i'd love for whoever offers some paid support to PM me |
20:45.16 | jblack | esaym153: in "full voip" It's handled by the endpoints. i.e. the phones. |
20:45.40 | esaym153 | hmm, then I guess my phone sucks |
20:45.49 | eppigy | Katty: yesh |
20:45.52 | jblack | That can definitely be. Is your phone made by "grandstrea" ? |
20:46.15 | jblack | btw, try playing with the volumes. Often you can mitigate or eliminate echo that way. |
20:46.54 | jblack | I meant "grandstream", btw |
20:47.18 | esaym153 | jblack: you talking to me? yes it is a grandsteam ata |
20:47.39 | jblack | esaym153: grandstream has a reputation of making the worst phones. Ever. |
20:47.52 | esaym153 | jblack: what do you recommend for an ata then? |
20:48.08 | p3nguin | ~grandsteam |
20:48.22 | jblack | Did you consider getting away from ATAs, and getting an IP phone? |
20:48.27 | p3nguin | ~gs |
20:48.28 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
20:48.49 | jblack | polycom makes phones for about $120 that are SIP devices. Actual physical phones that do tcp/ip |
20:49.05 | esaym153 | jblack: well I could get a phone, but I like my cordless handset |
20:49.13 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
20:49.32 | p3nguin | Get a cordless/wifi SIP phone. |
20:49.36 | ryduh | You can get a cordless handset for VoIP phones |
20:49.43 | jblack | Ok. You can probably get a cheap ata from cisco (aka linksys). They can be a little funky at times, but generally perform the core features well as long as you don't screw with 'em. |
20:50.14 | jblack | ryduh: Hrmm. I wonder if he meant "my cordless handset" or 'a cordless handset' |
20:50.40 | ryduh | jblack: good point. There are probably adapters? |
20:50.43 | jblack | esaym153: For $250, I picked up a linksys SPA-8000 that does 8 lines. |
20:50.47 | p3nguin | Cisco ATA 186 or Linksys SPA-3012 might be reasonable devices. |
20:51.20 | p3nguin | Sorry, SPA-3102. |
20:51.24 | esaym153 | my gs cost $25 :) |
20:51.31 | p3nguin | And look what you got! |
20:51.31 | jblack | I hear a lot of line nose on the spa-8k, but I also suspect that I have a poor ground. |
20:51.38 | jblack | esaym153: And how's it working for you? |
20:51.48 | ryduh | I wish there was a way to wirelessly use asterisk and ditch my cell phone. |
20:51.57 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
20:52.10 | p3nguin | ryduh: You could rely on wireless networking everywhere you go. |
20:52.17 | esaym153 | jblack: well it works, some people get echo though |
20:52.22 | jblack | if you can tolerate the echo on that GS you have, then all the more power to you. I was under the impression, though, that you had a problem you were trying to solve. |
20:52.29 | ryduh | p3nguin: what about areas without wifi? |
20:52.36 | p3nguin | You'll be withouta phone. |
20:52.42 | jblack | which, when you get down to it, is "your shit hardware is aqcting like shit" |
20:52.54 | esaym153 | yea |
20:54.24 | jblack | gah. I'm acting like such an asshole today. what's my problem |
20:54.40 | p3nguin | My guess is that yer an asshole. |
20:54.40 | esaym153 | jblack: you need Jesus ;) |
20:54.58 | jblack | p3nguin: Good point. |
20:55.08 | p3nguin | At least that's often why I act that way. :) |
20:55.29 | p3nguin | jblack: Something got you uptight today? |
20:55.46 | p3nguin | different from the day-to-day crap, I mean. |
20:55.56 | jblack | well, my sleep is off because I'm at war with my doctors.. maybe that's it |
20:56.17 | p3nguin | Anything preventing you from having a nap for a while? |
20:56.38 | jblack | yeah. I ate 2 dozen bbq hotwings, and need to work out before I spontaneously combust. |
20:57.05 | esaym153 | I disabled the echo cancellation on the ata I think that helped fix it. The cordless phone should have its own echo canceller I would think |
20:57.10 | p3nguin | Working out could also help relieve some tension in addition to saving your life. |
20:57.58 | jblack | yeah. |
20:58.06 | jblack | I hate this diabetes stuff. it sucks bad. |
20:58.47 | p3nguin | This may be too personal of a question, but is your body still able to produce those feel-good chemicals when you work out? |
20:59.01 | jblack | It's starting to get there. |
20:59.20 | p3nguin | Starting to not produce them or starting to get too personal? |
20:59.36 | jblack | I'm starting to get mild endorphins on occasion. |
20:59.51 | p3nguin | That's should be a good thing. |
21:00.09 | thehar | russellb: ! |
21:00.12 | jblack | yeah. finally a palpable payoff to the treadmill. :) |
21:01.01 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
21:02.10 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:02.25 | guywith2names | anyone know if its possible to inject a .wav file into a call? |
21:02.34 | jblack | sigh. |
21:02.38 | guywith2names | play a recording back to someone? |
21:02.49 | ryduh | Playback(filename) |
21:02.50 | jblack | Look at the Background() app, guywith2names . Also, consider getting the asterisk book if you want to do this yourself. |
21:03.38 | jblack | and if no one took you up on your money offer yet, hit me up after I finish my gerbil impression |
21:03.41 | guywith2names | heh i don't plan on doing anything, still waiting for someone to pm me so i can talk and maybe get some customizations done |
21:07.39 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
21:09.11 | Katty | you can also use a call file to call someone and play an audio file |
21:10.52 | *** join/#asterisk WinZ (n=winz@82.146.61.218) |
21:20.50 | *** join/#asterisk mike-ekim (n=digiport@204.13.2.30) |
21:20.58 | mike-ekim | I am getting Unable to open /dev/dahdi/ctl: No such file or directory when i try to run dahdi_tool |
21:21.01 | mike-ekim | any ideas why? |
21:21.50 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
21:23.26 | drmessano^ | dahdi not installed or running? |
21:23.35 | mike-ekim | i just did a reinstall of dahdi |
21:24.09 | mike-ekim | any specific way to check if dahdi is running? |
21:24.28 | Katty | decides to go home. |
21:25.31 | beek | Katty -- how did the chicken turn out? |
21:26.38 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
21:26.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
21:32.24 | *** join/#asterisk lucasb (n=bussey@24.67.33.55) |
21:33.33 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
21:34.35 | Chainsaw | mike-ekim: lsmod will show you. |
21:35.46 | *** join/#asterisk Number5 (n=El@j97231.upc-j.chello.nl) |
21:36.10 | ryduh | lol @ evil_gordita's nick |
21:36.30 | evil_gordita | I get that a lot. |
21:36.38 | Number5 | hello guys, does anyone know a channel about programming own voip/sip software? |
21:38.03 | ryduh | Has anyone originated calls by transferring call files via scp ? |
21:38.34 | *** join/#asterisk riddlebox (n=james@75.132.225.75) |
21:46.20 | jblack | oh that felt good. |
21:47.09 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
21:47.46 | ryduh | jblack: did you exercise? |
21:48.09 | jblack | I did. 26 minutes and 300 calories. Started feeling so good, I had to stop. :( |
21:49.07 | jblack | I can get right back to it though, as it turns out. |
21:51.40 | ryduh | I don't know much about diabetes and forgive me if it's too personal, how does diabetes affect exercise? |
21:51.42 | *** join/#asterisk djdefi (n=rtrauntv@64.136.179.53) |
21:52.36 | jblack | Your muscles use a type of sugar called glucose, which is in the blood. So, when you exercise, your blood sugar changes around a lot. Diabetics have problems controlling blood sugar. |
21:54.04 | jblack | In the case of type 2 diabetics, cells have trouble absorbing sugar, so the body compensates by making blood sugar higher. |
21:54.22 | jblack | so diabetics take something called insulin, which helps with absorption. |
21:54.55 | jblack | both insulin and exercise cause blood sugar to go down. See how they could interract? |
21:55.17 | ryduh | got it |
21:56.58 | coppice | diabetes is booming, with fantastic new opportunities for insulin and blood sugar meter sales in emerging markets |
21:57.53 | ryduh | ehh OSX is shitting it self |
21:57.53 | jblack | Yay for the widespread adoption of american food. ;) |
21:58.00 | ryduh | must restart |
21:58.03 | *** part/#asterisk ryduh (n=ryduh@204.16.143.186) |
21:58.40 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
21:59.46 | coppice | US haute cuisine, brought to you by McDonalds |
22:00.44 | jblack | sigh. Be back in another 30. |
22:00.51 | drmessano^ | That's not chicken |
22:01.11 | jblack | It's the bbq sauce that's giving me the problem. |
22:02.28 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
22:05.33 | *** join/#asterisk clive- (i=ident@dsl-242-144-214.telkomadsl.co.za) |
22:07.00 | clive- | hi, can someone please point me in the right direction on how to do a dialplan so that I can replace $EXTEN with my own value |
22:07.15 | clive- | the point is so that I can do pattern matching |
22:07.19 | *** join/#asterisk atis_home (n=atis@193.238.213.215) |
22:07.46 | jblack | clive-: reread the section on extensions, particularly _ for pattern matching |
22:07.57 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
22:08.05 | clive- | I tried set(EXTEN=${diallednumber}) but that didnt really do the trick |
22:08.24 | clive- | jblack,,, in the wiki? |
22:09.52 | leifmadsen | you can't set EXTEN |
22:09.56 | leifmadsen | it is automatically generated |
22:10.21 | ryduh | When I first visited voip-info I thought it was a fake wiki generated for advertising |
22:10.30 | leifmadsen | Use something like Set(xtn=${EXTEN}modified) |
22:11.38 | clive- | leif and then something like xtn => _1X.,1,noop etc |
22:11.41 | clive- | ? |
22:11.50 | leifmadsen | no.... |
22:11.54 | leifmadsen | what are you trying to do? |
22:12.36 | clive- | i have an agi scrip that is asking for the required number, now I want to dial that from the dialplan, with matching ability |
22:13.00 | leifmadsen | ok... so you just pattern match and use ${EXTEN} |
22:14.15 | *** join/#asterisk dkirker (n=dkirker@129.65.204.135) |
22:14.34 | clive- | problem is that the ${EXTEN} value is the access number into the system, not the number generated from the IVR (agi) |
22:16.12 | *** join/#asterisk coder2000 (n=coder200@beigetower/coder2000) |
22:16.58 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com) |
22:17.21 | clive- | I guess I could easily just calculate the complete dial string in the agi script... that may be easier ... hmm |
22:17.52 | coder2000 | I am trying to install switchvox in virtualbox 3.0.8 and found out I needed to disable apci but because I have to pass these options the root partition option is lost and I can't figure it out. Can anyone help? |
22:22.17 | Katty | :> |
22:27.26 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
22:30.53 | clive- | ok, I just did it all in the agi script... thanks for your help. |
22:32.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
22:32.26 | Katty | mmm, dinner was good. |
22:32.33 | Katty | rice, chicken, celery, onion, cheese. |
22:32.57 | Katty | and some edamame beans (= |
22:33.08 | coppice | those are components, not a product |
22:33.19 | Katty | well chicken rice and veggie casserole |
22:33.24 | Katty | and edamame beans on the side. |
22:33.28 | Qwell | coppice: You clearly are not a bachelor. |
22:33.41 | Qwell | cheese is a product. as are rice and chicken. |
22:33.51 | clive- | gnight guys..coppice ! |
22:34.09 | Katty | i often make the mistake of listing ingredients asuming people can visualize the end result in their head. |
22:34.21 | Katty | not everyone cooks as much as i do--sorry )= |
22:34.26 | [TK]D-Fender | Katty: But... I just went to the bathroom! |
22:34.43 | Katty | that's...nice? |
22:34.48 | Katty | did you wash your hands? |
22:34.59 | [TK]D-Fender | thinks of other "end results" |
22:35.08 | Katty | <PROTECTED> |
22:35.21 | [TK]D-Fender | Katty: pwned |
22:35.25 | Katty | you so did. |
22:38.08 | Katty | shower time (= |
22:38.14 | Katty | leafs. |
22:40.08 | coppice | Qwell: are you the sort who eats chicken sashimi simply to save the time and effort of cooking it? |
22:40.33 | Qwell | I don't know what sashimi is, so...yes. |
22:40.51 | [TK]D-Fender | coppice: sashimi? Nah.. that takes too long to prepare |
22:41.03 | [TK]D-Fender | chases down a live chicken and eats it live.... |
22:41.31 | [TK]D-Fender | picks the excess feathers out with a toochpick... |
22:42.52 | coppice | Qwell: you need to get past McDonalds more often |
22:43.39 | [TK]D-Fender | Qwell: You're still young enough to save yourself from ending up "tragically white"! |
22:43.53 | Kobaz | sashimi is the raw sushi |
22:44.09 | [TK]D-Fender | Kobaz: .... not quite.... |
22:44.19 | mchou | no joke, domino's has a large 1 topping for $6 special going on now |
22:44.48 | mchou | had one last night |
22:44.57 | Kobaz | s a Japanese delicacy primarily consisting of very fresh raw seafood, |
22:45.04 | Kobaz | raw fish |
22:45.11 | mchou | tasted like the cardboard box it came in |
22:45.15 | [TK]D-Fender | mchou: thats the price of a full meal at my favourite indian restaurant, only incredibly less healthy & appealing :) |
22:45.40 | coppice | mchou: that's not a cardboard box. its the 2 for 1 deal |
22:45.54 | Kobaz | [TK]D-Fender: my cousins wedding had an all you can eat sushi bar |
22:45.59 | mchou | [TK]D-Fender: really? you get a full Indian meal for $6? |
22:46.01 | Kobaz | along with all kinds of other foods |
22:46.34 | mchou | [TK]D-Fender: around here that's $10 |
22:47.11 | [TK]D-Fender | mchou: Yup.... |
22:47.30 | Kobaz | good prepared foods are expensive these days... i paid like 15 bucks at this pakistani place... the food was good, but it was like half a serving... maybe a third of one |
22:47.54 | Kobaz | and in order to get filled up on sushi, you need to order like 5 dishes |
22:47.55 | drmessano^ | I loved going to the pakistani restaurant.. Not so much being fired upon during the meal |
22:47.56 | Weedle | pakastani is a type of food ? |
22:47.59 | coppice | I've had excellent indian meals for <$6. mostly in india |
22:48.22 | mchou | drmessano^: see any Talibans over at the other table? :) |
22:48.23 | Weedle | yeah we get massive indian here for 8 bucks, so much that ylu can barely finish |
22:48.24 | Kobaz | no wait.. it was an afghani resturant |
22:48.26 | [TK]D-Fender | Kobaz: for $10 I'd be stuffed silly there... the food is awesome... I'd gladly be there paying more if the prices had to go up... |
22:48.46 | [TK]D-Fender | Kobaz: as it is its cheaper than most McDonalds trios |
22:48.50 | Kobaz | well yeah |
22:49.10 | Kobaz | mcdonalds gives you an 8000 calorie burger, with fries and a drink for like 7 bucks |
22:49.17 | drmessano^ | We get two meals out of a $15 portion of Indian food.. that includes a full order of Naan, and a Samosa per person |
22:49.27 | mchou | Kobaz: that's lame |
22:49.34 | drmessano^ | My wife and I eat for two days for $30 |
22:49.39 | mchou | Kobaz: incredibly BAD deal |
22:49.54 | Kobaz | mchou: if you need the calories from fat.. it's not bad |
22:49.59 | mchou | lol |
22:50.18 | Kobaz | although getting some $1.50 cans of corn beef would get you the same intake |
22:50.26 | drmessano^ | wants some Saag Paneer now |
22:50.32 | drmessano^ | bastards |
22:50.35 | *** join/#asterisk knctrnl (n=aembrey@nat/digium/x-awntosopzfodanov) |
22:51.10 | Kobaz | for 7 bucks, at the corner store... i get a foot long sub, a drink, and a bowl of soup |
22:51.31 | mchou | still a bad deal |
22:51.51 | Kobaz | well, it's obviously cheaper to make it yourself |
22:52.02 | Kobaz | but for across the street, it's not bad |
22:52.07 | drmessano^ | For $1.12 I get a bowl of soup in a paper bag and a badass hunk of 10 day old bread |
22:52.11 | drmessano^ | Beat that |
22:52.14 | Kobaz | haha |
22:52.23 | [TK]D-Fender | drmessano^: You know... I can't say that I've every had saag paneer.... |
22:52.29 | [TK]D-Fender | ever* |
22:52.32 | drmessano^ | [TK]D-Fender: O.O |
22:52.37 | drmessano^ | [TK]D-Fender: WHAT? |
22:52.42 | [TK]D-Fender | drmessano^: I've had jsut about everything else... |
22:52.47 | Kobaz | well for 7 bucks i can get a sack of bagels and a pound of deli meat and cheese... and make a week's worth of lunch |
22:52.50 | mchou | [TK]D-Fender: what?? That's a fake Indian place you're going to |
22:52.55 | [TK]D-Fender | drmessano^: drmessano^ what is "saag"? |
22:53.07 | Kobaz | mchou: is that a good deal?! |
22:53.09 | [TK]D-Fender | mchou: No, THEY have it... I've simply never gotten to ordering it. |
22:53.11 | drmessano^ | Spinach |
22:53.37 | [TK]D-Fender | drmessano^: that sounds like palak paneer... that is the soupy spinach & cheese |
22:53.38 | mchou | Kobaz: no. bagels are not such a great deal. Nor are cold cuts |
22:53.39 | drmessano^ | Depending where you get it, it MAY or MAY not be creamed with Coconut milk.. Where we get it, it is |
22:53.49 | Kobaz | mchou: what the hell do you eat then? |
22:53.50 | drmessano^ | Palak Paneer is the same |
22:53.52 | [TK]D-Fender | drmessano^: Perhaps a slightly diggerent blend |
22:54.06 | mchou | Kobaz: thai food |
22:54.09 | Kobaz | hehe |
22:54.19 | mchou | Kobaz: India food |
22:54.23 | mchou | Indian* |
22:54.26 | [TK]D-Fender | drmessano^: I've heard someone order saag there, but I've ordered palak myself... wonder if its a dialect thing |
22:54.26 | nix8n82 | is this like rachel ray hour? |
22:54.32 | [TK]D-Fender | different* |
22:54.33 | Kobaz | and how much is it... versus how much you get filled up? |
22:54.34 | mchou | Kobaz: Mexican Food |
22:54.37 | drmessano^ | I DONT think there is a difference between palak paneer and saag paneer, though I have wondered if thats the distinction in the creaming |
22:55.03 | Kobaz | nix8n82: no soup for you |
22:55.05 | mchou | Kobaz: I get a BBQ Pork Burrito for $4, good for 2 meals |
22:55.16 | drmessano^ | All the store brand foods are "Palak Paneer" and they all taste like shit unless I heat them with coconut milk |
22:55.32 | mchou | Kobaz: and definitely healthier than McDs and the like |
22:55.38 | [TK]D-Fender | nix8n82: http://www.youtube.com/watch?v=dpVdkHZsOW8 |
22:55.48 | Qwell | http://en.wikipedia.org/wiki/Saag |
22:55.50 | Kobaz | mchou: heh... a bbq pork burito usually is one meal for me |
22:55.50 | Qwell | lrn2wikipedia |
22:56.17 | Kobaz | mchou: and the place i get them from... they are huge... like 1.5 pounds probably alltogether |
22:56.19 | drmessano^ | I discovered that little trick after getting 10 boxes of "Palak Paneer" for $1 a box (marked down from $4) in the discount bin at the grocery store in redneckville |
22:56.45 | Kobaz | i can eat like... two footlong subs |
22:56.51 | Kobaz | and be hungry an hour later |
22:56.55 | drmessano^ | I see |
22:56.58 | drmessano^ | So both are correct |
22:56.59 | nix8n82 | lol..thanks |
22:57.06 | mchou | Kobaz: that's why the corner store aint a good deal |
22:57.07 | drmessano^ | I just never bothered to wiki it |
22:57.11 | drmessano^ | heh |
22:57.11 | Kobaz | mchou: yeah |
22:57.21 | Kobaz | mchou: or i just need to lower my metabolism |
22:57.21 | *** join/#asterisk neurosys (n=vinix@adsl-072-151-208-134.sip.mia.bellsouth.net) |
22:57.35 | Kobaz | mchou: with the soup though... i get filled up |
22:57.35 | mchou | Kobaz: you still a teenager? |
22:57.39 | Kobaz | 26 |
22:57.48 | mchou | Kobaz: close enough |
22:58.12 | Kobaz | haha |
22:58.12 | mchou | Kobaz: whait till you hit 30 |
22:58.12 | mchou | wait* |
22:58.13 | Kobaz | i dunno |
22:58.14 | [TK]D-Fender | Qwell: Per that link palak paner has always been fiercely green, not yellow. |
22:58.23 | Kobaz | with the amount of excersize i get... i'm usually losing weight |
22:58.37 | mchou | Kobaz: what you do for exercise? |
22:58.43 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
22:58.43 | [TK]D-Fender | Qwell: So does look a fair bit different |
22:58.47 | Kobaz | run, mountain bike, climb, ski |
22:59.03 | mchou | Kobaz: wher you live? CO? |
22:59.07 | Kobaz | central pa |
22:59.11 | Kobaz | i did live in co for 6 months though |
22:59.21 | Kobaz | i wanna move back at some point |
22:59.30 | [TK]D-Fender | nix8n82: You're welcome ;) |
22:59.41 | ecrane | Sorry, any rules on asterisk questions? I am having trouble finding out which version of asterisk is the official 'Stable' version. |
22:59.42 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
22:59.55 | Kobaz | asterisk... what's asterisk? |
23:00.03 | neurosys | shift + 8 |
23:00.12 | [TK]D-Fender | .. what is this "stable" of which you speak? |
23:00.13 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
23:00.15 | ecrane | yeah; it's the shift + 8 |
23:00.25 | Kobaz | there is no * |
23:00.26 | [TK]D-Fender | fetches some more spackle to hold his together |
23:00.30 | mchou | stable of horses |
23:00.49 | neurosys | stable of hookers? |
23:00.58 | [TK]D-Fender | kills mchou's equine and begins flogging..... |
23:00.58 | Kobaz | isn't it a gaggle of hookers |
23:01.11 | neurosys | geese. Gaggle of geese |
23:01.33 | [TK]D-Fender | Kobaz: Only if they're into auto-erotic asphyxiation |
23:01.34 | drmessano^ | ecrane: 1.4, 1.6.0, 1.6.1, are all "Stable" branches |
23:01.37 | ecrane | fistful of hookers? |
23:01.50 | mchou | naw, fistful of $ |
23:01.56 | mchou | wrong key |
23:02.03 | drmessano^ | Trunkful of dead hookers? |
23:02.08 | ecrane | thanks drmessano^. |
23:02.11 | Kobaz | sounds about right |
23:02.18 | neurosys | bingo |
23:02.22 | ecrane | This seems like a pretty cool channel. |
23:02.25 | [TK]D-Fender | Nothing quite as stable as a "dead stop" |
23:02.33 | drmessano^ | It's all fun and games til you get stuck figuring out where to bury the hooker |
23:02.53 | drmessano^ | Easy way to ruin a hangover |
23:03.44 | neurosys | Must always have lime in the trunk |
23:03.48 | jblack | Having sex. For a job. |
23:04.02 | Kobaz | that's the idea |
23:04.22 | jblack | sounds better than "fry cook at mcdonald's" to me |
23:04.38 | mchou | yo wanna some fries with that? |
23:04.46 | ecrane | smells worse though. |
23:04.47 | jblack | No, but I'll take some shake. =) |
23:05.07 | jblack | rimshots |
23:05.17 | neurosys | But what if its a woman who's 300+ and smells like rotten egg? Still sound like the ideal job |
23:05.40 | Kobaz | rimjob? |
23:05.47 | jblack | Not everybody is well suited for any particular job. |
23:05.49 | drmessano^ | I always find throwing a few shredded kittens in the trunk will kill the dead hooker smell |
23:05.51 | mchou | neurosys: is that years or #? |
23:06.13 | neurosys | lbs. but you get the idea ;) |
23:06.17 | jblack | drmessano^: So... your trunk smells like dead pussy, right? |
23:06.25 | drmessano^ | Rimshot? |
23:06.31 | mchou | dead pussy detail! |
23:07.02 | jblack | rimshot.. that's the drum sound that you hear after a joke. |
23:07.22 | drmessano^ | jblack: I was cueing it |
23:07.33 | drmessano^ | BA DUMP CHING! |
23:07.34 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
23:07.39 | drmessano^ | But you failed me |
23:07.41 | drmessano^ | :( |
23:07.55 | jblack | Did not |
23:08.08 | drmessano^ | The thing about hooking is, you wont ever have to worry about women over 50 propositioning you.. Thats like expecting a kid to beg for brussell sprouts |
23:08.12 | jblack | Why don't you go tweet that there's chickin in your chicken nuggets, or something |
23:08.31 | drmessano^ | jblack: That was the 17th of August |
23:08.57 | jblack | what was |
23:09.02 | drmessano^ | jblack: You've been tweetfeated |
23:09.12 | drmessano^ | The chickin the chicking nuggets |
23:09.19 | drmessano^ | Or was that on Facebook? |
23:09.20 | jblack | oh. |
23:09.37 | Curtman | Is it possible to have my fxo port monitor a line for rings, and ring my fxs extensions, but not pick up the line until/if an extension picks up? |
23:09.37 | jblack | same difference. |
23:09.37 | drmessano^ | Maybe I blogged it |
23:09.37 | drmessano^ | or Laconica |
23:09.54 | drmessano^ | or Windows Live Tweet Blog Thingo |
23:09.58 | jblack | curtman: Sure. Just do a dial without an answer. |
23:09.59 | Kobaz | Curtman: that's kinda how it works by default |
23:10.06 | Kobaz | Curtman: unless you Answer() |
23:10.23 | neurosys | Does Freenode/#asterisk have a facebook fanpage? seriously? |
23:10.34 | drmessano^ | Microsoft will eventually add a twitter like service to Windows Live... they will probably call it "Windows Live Net Send" |
23:10.42 | drmessano^ | or "Windows Live Notepad" |
23:10.47 | Kobaz | we should make an asterisk facebook game |
23:11.17 | neurosys | setup the most convoluted dialplan? What kinda game would we make :P |
23:11.19 | drmessano^ | "Kick the newb in the nuts?" |
23:11.31 | drmessano^ | Thats kinda boring |
23:11.34 | Kobaz | complete with voip packet loss, crappy telco tech support, acts of nature, and construction crews cutting your fiber |
23:11.51 | jblack | kobaz: Oh, they have that. It's caled "myspace" |
23:11.53 | Curtman | Kobaz, I've just started to play with the GUI's.. I don't know of another "default". I'm still doing a lot of reading, just wanting to know if there is light at the end of the tunnel. There's only so much free time to spend. ;) |
23:12.03 | drmessano^ | High Availability Scenarios with Grandstream ATAs? |
23:12.07 | neurosys | dont forget the homeless men pissing on your punch block in the meter rooms |
23:12.07 | drmessano^ | Now that would be a cool game |
23:12.10 | Kobaz | drmessano^: yes! |
23:12.29 | jblack | TLC could do a series called "I shouldn't be calling" |
23:12.52 | drmessano^ | "Grandstream ATA clustering solutions 2: Full Frontal Dialplan" |
23:12.55 | drmessano^ | I can see it now |
23:12.58 | Kobaz | Curtman: the gui's will only get you so far |
23:12.59 | drmessano^ | In stores next month |
23:13.16 | drmessano^ | Astercraft! |
23:13.21 | Curtman | Kobaz, I realize that. I'm just using them as a starting point to learn with. |
23:13.23 | drmessano^ | World of Astercraft! |
23:13.30 | *** join/#asterisk jadl_ (n=jadl@89.130.82.210) |
23:13.41 | Kobaz | grandstream makes the worst shit imaginable |
23:13.53 | Kobaz | it's amazing people buy their stuff |
23:13.57 | drmessano^ | spends all day in the woods sniping newbs, then heads off to slay an Avaya installer |
23:14.09 | neurosys | I finally listened to [TK]D-Fender and started replacing my ciscos with polycom |
23:14.16 | Kobaz | polycrum! |
23:14.21 | Kobaz | yeah polycom rocks |
23:14.24 | drmessano^ | Asterisk Interactive Fiction! |
23:14.26 | Kobaz | they need more buttons though |
23:14.41 | Kobaz | if polycom had like, even one more extra programmable button |
23:14.53 | Kobaz | well the 320's anyway |
23:14.54 | drmessano^ | "It is dark. You hear the ringing of a Cisco phone in the distance" |
23:14.57 | drmessano^ | > Look |
23:15.07 | drmessano^ | There is a punchblock and a grandstream ATA here |
23:15.10 | drmessano^ | > Get ATA |
23:15.19 | Kobaz | attach punchblock |
23:15.28 | drmessano^ | "Why the hell would you do that?" |
23:15.31 | Kobaz | > the puchblock awakens |
23:15.32 | ryduh | I originate a call to a PTSN # over SIP. I have my dtmfmode=rfc2833. I'd like to dial extension 102 at the destination. I try SendDTMF(1w) SendDTMF(0w) SendDTMF(2w) but it only sounds like 1 digit is being sent. is that normal? is there a better way to do that? |
23:15.34 | drmessano^ | > Kill ATA |
23:15.43 | drmessano^ | "You slay the ATA and it explodes" |
23:15.44 | Kobaz | > the pucnblock hits you with 90 volts |
23:15.57 | neurosys | uch |
23:15.57 | drmessano^ | > North |
23:16.17 | drmessano^ | "You head north and approach a scary AT&T tech in the network closet" |
23:16.27 | drmessano^ | > Kill AT&T tech |
23:16.49 | neurosys | man i miss the telnet MUDs :P |
23:16.49 | drmessano^ | "The AT&T tech smashed your head in with a buttset. You are now dead. Score 0" |
23:16.49 | Kobaz | "the AT&T network arrives" |
23:16.59 | drmessano^ | :( |
23:17.32 | jadl_ | hello, I have a problem, I have a sip voip provider Yacom and x-lite but do not work, I configured asterisk and everything is fine but I think something went wrong, when I call, the operator says the prefix is not valid |
23:17.58 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
23:18.02 | drmessano^ | "It is pitch black. You are likely to have your day eaten fighting echo on old, rotted pots lines." |
23:18.04 | Kobaz | > Look |
23:18.05 | Kobaz | "You are in the main control room, you are likely to be eaten by a grue" |
23:18.06 | drmessano^ | ^^^ thats IT! |
23:18.12 | drmessano^ | ha |
23:18.15 | jblack | Sometimes, when my head hurts really bad, I turn off the lights and wait for the grue that should surely eat me. |
23:18.16 | jadl_ | <PROTECTED> |
23:18.16 | jadl_ | can someone help |
23:18.24 | [TK]D-Fender | |<--Deeewayne has left irc.freenode.net ("Herbivores ate well cause their food didn't never run") <-- Double negative FAIL |
23:18.26 | jblack | You stole my grue line! |
23:18.34 | Kobaz | haha |
23:18.39 | drmessano^ | You stole my grue line |
23:18.47 | drmessano^ | Though i didnt use grue.. |
23:19.08 | drmessano^ | But mine was a blatant ripoff |
23:19.24 | neurosys | telenet (NUAA Attacker)....... |
23:19.28 | neurosys | c sprint |
23:19.31 | neurosys | :P |
23:20.51 | jblack | looks for a t-shirt with the grue line |
23:21.07 | jadl_ | I have read about Asterisk, I've seen videos and I downloaded some instructions and VozTelecom asterisk and nothing ... please help |
23:21.07 | drmessano^ | "You wake up. it is 4:30 in the afternoon. The smell of rotten pizza permeates the air. You're not wearing any pants, and there's poorly written dialplan glaring at you through from your 24 inch LED monitor." |
23:21.10 | drmessano^ | > whoami |
23:21.15 | *** join/#asterisk natlonehat (n=natloneh@202.170.42.67) |
23:21.16 | drmessano^ | "You are [TK]D-Fender" |
23:21.34 | drmessano^ | \o/ |
23:21.44 | neurosys | lol |
23:22.03 | [TK]D-Fender | drmessano^: http://xkcd.com/386/ |
23:22.15 | drmessano^ | One of my FAVS |
23:22.20 | cusco | hi |
23:22.28 | drmessano^ | ho |
23:22.34 | cusco | how can I logoff a manager user |
23:22.51 | cusco | if possible trough cli |
23:22.57 | cusco | asterisk's cli |
23:23.03 | jblack | hey, infocom lets you download zork 1-3 at not cost |
23:23.42 | [TK]D-Fender | cusco: iptables -..... err ;) |
23:23.42 | Jumpie | is it possible to do kinda an inverse blacklist? |
23:23.42 | Jumpie | i wanna rject all but a few i wanna specifically allow |
23:23.42 | drmessano^ | has the infocom pack with 20 games in it |
23:23.42 | cusco | lol, just logim off once |
23:23.42 | cusco | isn't it possible? |
23:23.42 | drmessano^ | HHGG ftw |
23:23.43 | [TK]D-Fender | Jumpie: You can do whatever you want. Its your dialplan. |
23:23.43 | drmessano^ | "get not tea" |
23:23.43 | neurosys | cusco:core stop now |
23:23.44 | [TK]D-Fender | Jumpie: Filter based on WEATHER if you feel like it |
23:23.45 | drmessano^ | "You pick up the not tea" |
23:23.47 | cusco | neurosys: hmm maybe later |
23:23.48 | Jumpie | i guess i meant simplistically heh |
23:23.54 | Jumpie | ah |
23:24.19 | drmessano^ | Whitelist |
23:24.21 | [TK]D-Fender | Ok, martial arts then stage time... later all |
23:24.23 | drmessano^ | then Blacklist |
23:24.28 | neurosys | l8r |
23:24.38 | drmessano^ | Whitelist has a GOTO to jump past the blacklist |
23:24.38 | ryduh | anyone for some DTMF help? |
23:24.58 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
23:25.00 | Jumpie | ah..right |
23:25.04 | jblack | [TK]D-Fender: already leave yet? |
23:25.08 | Jumpie | like network access lists somewhat |
23:25.11 | Jumpie | order is important |
23:25.30 | jadl_ | what can I do? |
23:26.00 | drmessano^ | jadl_: Anything you want |
23:26.22 | jadl_ | I don't speak english |
23:26.48 | jadl_ | I use google |
23:27.13 | jadl_ | sometimes |
23:27.39 | jadl_ | hello, I have a problem, I have a sip voip provider Yacom and x-lite but do not work, I configured asterisk and everything is fine but I think something went wrong, when I call, the operator says the prefix is not valid |
23:27.50 | jadl_ | (repeat) |
23:28.14 | jblack | I have drmessano^'s picture. Who wants to see it? |
23:28.30 | jblack | http://i258.photobucket.com/albums/hh260/lawngnome8273/untitled40.jpg |
23:28.54 | drmessano | IS that a BATMAN lunchbox? |
23:28.59 | drmessano | Oh SNAP |
23:30.03 | jblack | When 'catty' (mispelled to avoided console beeping) comes back later, I'm going to offer her "my" picture, and see what she says. :) |
23:30.54 | jadl_ | drmessano ^: you can help, or language is a problem ... |
23:31.52 | jblack | jadl_: set debug 9, and verbose 9, and take thedebugging from there. |
23:31.54 | jblack | http://media.photobucket.com/image/fat%20people/staunchusa/2ontruck.jpg |
23:32.18 | ryduh | Can I run multiple SendDTMF commands in a row? |
23:33.45 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
23:36.26 | *** join/#asterisk stew (i=1413@freenode/staff/stew) |
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23:36.39 | jadl_ | I know that there is a verbose and debug mode and I can not remember how to do what you say? |
23:37.38 | Chainsaw | jadl_: core set debug 9 |
23:37.46 | Chainsaw | jadl_: core set verbose 9 |
23:38.23 | jadl_ | thanks you |
23:38.53 | jadl_ | |
23:38.54 | jadl_ | Well, now what? |
23:39.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
23:40.23 | *** join/#asterisk RobH (n=RobH@216.38.133.254) |
23:41.57 | ecrane | Guys my compile process for asterisk is acting really weird. I did a './configure', did a 'make menuselect', 'make', 'make install' and now the program runs fine without any problems. I didn't have to chase down any dependencies or build errors... this doesn't feel right at all :<. |
23:42.29 | Chainsaw | Don't worry, it'll feel more real when you get to DAHDI. |
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23:49.13 | jadl_ | bye |
23:50.27 | jadl_ | |
23:50.27 | jadl_ | back tomorrow |
23:50.52 | jadl_ | xd |
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