| 00:00.24 | Chainsaw | bcrisp: Nope, just the CLI. |
| 00:00.56 | bcrisp | Chainsaw: thats all im using for now too |
| 00:01.33 | *** join/#asterisk lordmortis (n=lordmort@203-59-59-187.dyn.iinet.net.au) |
| 00:03.16 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 00:11.32 | p3nguin | chainsaw: What model of Siemens DECT SIP phone do you have? |
| 00:11.44 | Chainsaw | p3nguin: C475 IP I think. Let me check. |
| 00:12.23 | Chainsaw | p3nguin: Webpage says yes. C475. |
| 00:13.29 | p3nguin | I currently have a 2.4 GHz (regular cordless) SIP phone at home, and I had been considering an upgrade. |
| 00:13.30 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 00:15.35 | Chainsaw | p3nguin: I'm having some NAT issues which I blame on my ISP-supplied router (that I can't exchange). |
| 00:15.43 | Chainsaw | p3nguin: But with a better router, it was working perfectly. |
| 00:16.21 | bcrisp | hiya p3nguin |
| 00:16.29 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 00:16.35 | p3nguin | Google product search doesn't know anything about that model number. :( |
| 00:16.39 | p3nguin | hi |
| 00:16.58 | bcrisp | p3nguin guess who had his first successful SIP convo? |
| 00:17.52 | bcrisp | thanks for your help yesterday |
| 00:18.10 | *** join/#asterisk BadHAL (n=nn@24-179-237-8.dhcp.eucl.wi.charter.com) |
| 00:18.24 | p3nguin | Oh, I couldn't figure out who you were... |
| 00:18.40 | p3nguin | Norway, right? |
| 00:18.54 | bcrisp | nah... i was just a newb asking questions |
| 00:19.00 | p3nguin | oh |
| 00:19.09 | bcrisp | 2nd day linux user |
| 00:19.22 | p3nguin | Ah right, toresbe is still here. That's where he is. |
| 00:23.11 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 00:23.11 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.11 (2009/11/30), 1.6.0.19 (2009/11/30), 1.4.27.1 (2009/11/30), *-Addons 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 00:23.21 | bcrisp | now i really need to go learn more about queues |
| 00:24.07 | p3nguin | That's pretty simple, really. Once you read the section about it, dive in and configure it. If you have any trouble, just ask for help here. |
| 00:24.36 | bcrisp | i stumbled on agents.conf in one of the appendices .. it has more info than i had anticipated which is good |
| 00:25.10 | p3nguin | Are you going to have agents logging on and off of the system? |
| 00:25.33 | bcrisp | im not quite sure.. i need to set up a small / medium call center with queueing ability |
| 00:26.10 | bcrisp | it sounds like that is probably correct |
| 00:26.53 | p3nguin | Agents' ability to log on and off is good when they are not always present during call-in hours. You don't want a person's phone ringing with an important call if he is not at work. |
| 00:27.03 | bcrisp | absolutely |
| 00:27.43 | bcrisp | it would also be nice to have some sort of visual feedback for the queues |
| 00:28.03 | bcrisp | would i need to delve into something offered via asterisk gui for that? |
| 00:28.42 | p3nguin | I don't think asterisk-gui offers that, but I don't know. I've never used the gui, so I don't know what it offers. |
| 00:29.16 | p3nguin | I've seen some third-party apps for that sort of thing, though. |
| 00:29.29 | bcrisp | im not sure if freepbx offers that sort of thing |
| 00:29.42 | bcrisp | but i do want to keep things as simple as possible |
| 00:29.56 | bcrisp | i suppose i could always just write a web app that taps into mysql |
| 00:30.12 | [TK]D-Fender | bcrisp: AMI <--- |
| 00:30.30 | bcrisp | ~AMI |
| 00:30.31 | infobot | rumour has it, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 00:30.40 | bcrisp | thanks fender |
| 00:30.45 | [TK]D-Fender | bcrisp: Easy to write your own. I have my phones display live stats on their screens monitoring 2 queues and 4 agents |
| 00:31.09 | bcrisp | nice |
| 00:31.16 | Katty | will be shutting down ferret cam temporarily while letting critters out |
| 00:31.57 | bcrisp | ill probably head over to the dev room once i need to interface with our crm system |
| 00:32.17 | [TK]D-Fender | bcrisp: Doubtful |
| 00:32.37 | [TK]D-Fender | bcrisp: AMI is pretty basic stuff. You only need dev if you intend or changing raw * source |
| 00:32.46 | bcrisp | oh i see |
| 00:33.57 | bcrisp | oh wow... AMI does look great |
| 00:37.45 | [TK]D-Fender | bcrisp: Look at FOP as a sample of what you can do with AMI and then realize you could do a lot better yourself for your specific needs if you're competant |
| 00:37.47 | [TK]D-Fender | ~fop |
| 00:37.48 | infobot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 00:37.55 | [TK]D-Fender | (latter) |
| 00:38.22 | bcrisp | excellent |
| 00:39.55 | bcrisp | does Festival work well? |
| 00:39.55 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 00:40.25 | [TK]D-Fender | bcrisp: Not bad. Cepstal is noticably better but costs per channel of simultaneous use |
| 00:40.37 | [TK]D-Fender | Off to martial arts and then stage time. Later all |
| 00:40.42 | bcrisp | thanks |
| 00:40.45 | bcrisp | cya |
| 00:41.31 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 00:45.15 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:47.37 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
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| 00:54.59 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 00:55.56 | Katty | ferret cam's back on, but no audio at ryan's request (= |
| 00:56.39 | *** join/#asterisk ming_zym (n=ming_zym@114.251.86.0) |
| 00:57.26 | jaytee | got a link? |
| 00:58.56 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 00:59.40 | Chainsaw | jaytee: http://www.ustream.tv/channel/squirrel-critter-cam |
| 01:01.18 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 01:01.27 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam |
| 01:01.37 | jaytee | lol, Riddick looks like he's having fun |
| 01:01.56 | Katty | yep (= |
| 01:02.05 | Katty | he was raised with the lil critters |
| 01:02.55 | coppice | Riddick? Isn't he that bad telecoms guy from "The Chronic Calls of Riddick"? |
| 01:02.57 | Chainsaw | Yeah, our shepherd dog would have gotten hungry by now. |
| 01:03.06 | jaytee | drmessano ping |
| 01:03.21 | Chainsaw | This one's more of a team player :) |
| 01:04.36 | Katty | riddick's perfectly fine around them |
| 01:09.53 | Chainsaw | Katty: Yeah, I've seen another one that grew up with a cat. They got along great :) |
| 01:10.31 | Chainsaw | Best dogs ever though. They say golden retrievers are more intelligent, but I wouldn't trust a retriever to guard my house. |
| 01:11.03 | Katty | nods |
| 01:11.13 | Katty | i enjoy having riddick here. he's good company, and a good watcher. |
| 01:11.59 | Chainsaw | :) |
| 01:12.53 | Katty | they keep getting on the laptop and bumping the camera. |
| 01:12.55 | *** join/#asterisk _pepo_ (n=_pepo_@190.12.5.106) |
| 01:13.23 | _pepo_ | hi friendss |
| 01:13.47 | TommyBotten | But we've just met |
| 01:14.43 | _pepo_ | when I "transfer" one call to another extension, I free the channel so and I can receive another call, or not? |
| 01:14.50 | Chainsaw | Katty: But they're having a blast though :) |
| 01:15.00 | Katty | yep. |
| 01:15.31 | Katty | oh. |
| 01:15.35 | Katty | pippin likes hiding toys. |
| 01:15.38 | Katty | goes to find one |
| 01:16.48 | Katty | was it on camera? |
| 01:17.02 | jaytee | yeah, what's the white ferret's name? |
| 01:17.05 | LemensTS | I closed xlite, which disconnected the sip account with asterisk that it was using. But im getting many of these a second in asterisk sip debug even tho the phone is offline http://pastebin.ca/1701020 |
| 01:17.07 | Katty | Merry |
| 01:17.14 | jaytee | :-) |
| 01:18.57 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 01:21.48 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:22.49 | p3nguin | _pepo_: Check "core show channels" before you transfer the call, then check it again after it's done transferring (after it's answered and you hang up your phone). |
| 01:23.13 | _pepo_ | tnx, ill try it |
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| 01:45.08 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 01:46.32 | Defraz | how can I slow down the time between having to dial ## |
| 01:46.33 | Godfather_ | how can i see the latency of a peer? |
| 01:46.41 | Defraz | some of my users can't seem to hit it fast enough. |
| 01:48.09 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
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| 02:08.02 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:08.39 | *** join/#asterisk makafre (n=makafre@76-10-134-32.dsl.teksavvy.com) |
| 02:09.17 | makafre | hey guys, remind me how to create peers using patterns in sip.conf, I have to create 60 of them and I rather use patterns |
| 02:16.18 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 02:21.18 | Tech_Travis | makeafre:[sets](!) for template then (sets) next to the entries. |
| 02:29.42 | *** join/#asterisk wolfeyes (i=291347d1@gateway/web/freenode/x-cxxussdlkweobkwm) |
| 02:38.11 | makafre | Tech_Travis: thanks! |
| 02:40.20 | wolfeyes | is anyone here using asterisk as a business by selling it to companies? |
| 02:46.11 | *** join/#asterisk syadnom (n=syadnom@host-98-127-86-3.bln-mt.client.bresnan.net) |
| 02:46.31 | syadnom | hello |
| 02:47.55 | *** part/#asterisk gewt (i=b4@ip-67-202-107-221.chicago-il.gewt.net) |
| 02:48.21 | syadnom | I've googles til my fingers were raw and cant find any solid answers to a question. I just see a lot of back and forth in forums. is SCA or shared extensions a possiblility? not SLA, but SCA. registering multiple devices to a single extension and having it behave. |
| 02:50.48 | syadnom | no one awake here huh? |
| 02:51.34 | ChannelZ | or of those who don't know the answer |
| 02:51.45 | syadnom | its just very quiet |
| 02:51.57 | syadnom | not a lot of banter :) |
| 02:51.57 | *** join/#asterisk lordmortis (n=lordmort@203-59-59-187.dyn.iinet.net.au) |
| 02:52.10 | ChannelZ | we're all busy fapping |
| 02:53.07 | syadnom | a circle fap? or fapping in the cloud as it may be? |
| 02:53.27 | syadnom | cloud=trend new name for the internet |
| 02:54.37 | ChannelZ | with the assistance of, let's say |
| 02:55.04 | wolfeyes | lol |
| 02:55.21 | wolfeyes | Amd I thought you all were hiding. |
| 02:56.10 | syadnom | anyone worked with sip proxies at all? specifically proxy functions, not whole PBX functions. |
| 02:56.49 | syadnom | im looking at creating(or modifying) a sip proxy to allow multiple registrations on extensions and to push hints out to registered devices. |
| 02:58.01 | syadnom | the idea is to have a sip proxy handle shared call appearance by registering itself to an asterisk system on a specific extension and then accepting registrations from devices on some alternate port (so it can sit on the asterisk server). |
| 02:58.51 | Kobaz | syadnom: multiple registratios to the same sip user will cause the calls to get routed to who ever registered most recently |
| 03:00.19 | wolfeyes | Ok I am a beginner here , let me get this straight. Asterisk is a linux programmed. But can it link with ms users? <ost of my companies here ms o/s's and i dont want to get through the book only to find it can't do what I want! |
| 03:00.32 | syadnom | yes, that is why some intermediate needs to be involved (or asterisk needs to accept multiple regs per extension). The SIP proxy would recieve the calls as it is the only "device" registered to the extension. It then has devices registered to it on some port like 65060. when the proxy receives that call, it can push ringing out to each device registered to it on that extension. |
| 03:00.49 | Kobaz | i know polycom phones support sla but asterisk doesn't support it |
| 03:01.07 | wolfeyes | Most* |
| 03:01.28 | syadnom | wolfeyes, standard tcp/udp networking so it can talk with computers of any OS that handle tcp/udp(windows, mac, linux, *bsd, etc etc etc) |
| 03:01.31 | Kobaz | syadnom: you would have to make sure the phone supports what you're trying to do |
| 03:01.32 | ChannelZ | Why not just dial multiple extensions |
| 03:01.40 | wolfeyes | excellent |
| 03:01.55 | wolfeyes | and asterisk grows some more |
| 03:02.08 | Kobaz | or you could just take the easy way, and use what's available |
| 03:02.08 | Kobaz | heh |
| 03:02.36 | syadnom | Kobaz, actually, the phone should just see the sip proxy as a standard sip proxy running on a different port, which all phones I have seen do. The phone should be un-aware. It would be required that the phone can receive hints about hold, busy, etc but that is pretty standard |
| 03:03.11 | syadnom | Kobaz, what is available is parking lots and ring groups, which add complication for users, daily |
| 03:03.36 | wolfeyes | well... I wan the best way, not only where money making is concerned but, being able to get the most from the products, hardware and software, and so making my service to my client a step above the competition. |
| 03:04.02 | syadnom | wolfeyes, what are you trying to do? |
| 03:05.09 | wolfeyes | pabx phone systems for companies to be able to talk using voip syadnom , preferably even per second billing. |
| 03:05.09 | syadnom | Asterisk isnt really a goldmine for MAKING money in that the market for selling and installing phone systems is pretty saturated. Asterisk is very good as SAVING money for your company and yourself, which does of coarse mean making money. |
| 03:05.30 | syadnom | wolfeyes, its a busy market, with lots of vendors. |
| 03:05.55 | wolfeyes | I want to resell it to companies, in my town is basically doesn't exist yet. |
| 03:06.18 | wolfeyes | vox only and that is just for house consumers |
| 03:07.18 | syadnom | #1 piece of advice, get very very familiar with asterisk and dont skimp on hardware. sangoma for POTS lines, server systems not desktops to run the software on, aastra or polycom phones.... |
| 03:07.56 | syadnom | pbx in a flash is very nice and works very very well for me. I also like elastix. they both use the freepbx interface which is very nice for configuring and maintaining asterisk systems |
| 03:11.07 | wolfeyes | ok great ty that sounds awesome syadnom |
| 03:11.46 | syadnom | the last thing you want is calls (from cell phones) all day long that service is down at a business. big no no. |
| 03:12.37 | wolfeyes | Now that would be against the objective |
| 03:13.19 | syadnom | I maintain over 30 asterisk systems in production. i have learned a lot of lessons |
| 03:15.15 | *** join/#asterisk haryv (n=ltech@174.1.121.39) |
| 03:20.43 | *** join/#asterisk darkadan (n=darkadan@190.96.171.203) |
| 03:21.39 | darkadan | HI |
| 03:22.23 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-245-105.dsl.lsan03.sbcglobal.net) |
| 03:26.42 | wolfeyes | That sounds pretty awesome syadnom |
| 03:28.02 | ChannelZ | OH HAI! |
| 03:32.36 | *** part/#asterisk lanning (n=lanning@208.87.235.224) |
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| 03:40.28 | *** join/#asterisk joobie (n=joobz@mx01.anric.com.au) |
| 03:40.37 | joobie | guys is there a way to reload just voicemail.conf? |
| 03:40.45 | bcrisp | voicemail reload |
| 03:41.26 | joobie | 1.4 |
| 03:41.29 | joobie | doesn't work |
| 03:41.34 | bcrisp | oh sorry |
| 03:42.47 | ChannelZ | module reload app_voicemail |
| 03:43.08 | joobie | thanks |
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| 03:47.12 | dudee | Hi I have a issue, hope somebody here could assist me, I have setup a ring group and a person can ring into that ring group , everything there is okay. But lets say if it rings and somebody on DAHDI/1 was on the line, it would not ring DAHDI/1, only /2 /3 /4 and so... how can /1 pick up the phone with out going to /2 desk to pick it up ? ( ** ) didnt work , *8 gave a busy / engaged tone. please advice. thank you |
| 03:47.46 | syadnom | call waiting. |
| 03:48.23 | syadnom | most phones will roll a call from the line1 button up to line2 if it is not occupied. call waiting causes this to happen. |
| 03:49.03 | ChannelZ | I have a *7xxx extension for picking up other extensions -- Pickup(${EXTEN:2}@internal) |
| 03:49.07 | syadnom | dudee, freepbx or plain asterisk? what phones do you use? |
| 03:49.38 | dudee | syadnom: freepbx, analog |
| 03:49.48 | dudee | using TDM2400p for it |
| 03:50.39 | dudee | Call waiting is enabled |
| 03:50.44 | dudee | should it be disabled? |
| 03:51.12 | syadnom | dudee what phones? |
| 03:51.24 | ChannelZ | POTS phones sounds like |
| 03:51.40 | dudee | syadnom: analog phones.... using TDM2400p . |
| 03:51.46 | dudee | yeap, POTS phone |
| 03:51.56 | dudee | its a mix environment with POTS and SIP trunks |
| 03:52.48 | syadnom | no, which devices, list aastra, polycom, grandstream |
| 03:53.54 | dudee | syadnom: i wouldn't know for sure, if i have to make a guess, i would say polycom |
| 03:54.21 | dudee | ChannelZ: with that , it should pick up lines from another extension, how about the general incomming line? is there a way to grab it like *8 ? |
| 03:54.46 | *** part/#asterisk darkadan (n=darkadan@190.96.171.203) |
| 03:55.11 | syadnom | You should be able to enable call waiting which would have the call roll to the next available line button for pickup. either that or *8 for direct call pickup. |
| 03:56.26 | ChannelZ | I'm not really sure what your dialplan is doing, if *8 doesn't do it (is it turned on in features.conf ?) |
| 03:57.06 | bcrisp | can i change the "comedian mail" intro when using the voice mail menu? |
| 03:57.28 | dudee | syadnom: lets say DAHDI/1 was one the phone while the phone was ringing, and he puts down the phone. DAHDI/1 will not ring any more, just the remaining extensions will ring. and i would either like /1 to ring or would like to give /1 the option of picking up the call without running to another desk ( like *8) |
| 03:57.31 | jaytee | yes |
| 03:57.39 | jaytee | record your own |
| 03:57.47 | dudee | but when i tested with *8, it gave a busy / engaged tone |
| 03:58.05 | dudee | yes *8 is turned on |
| 03:58.09 | Kobaz | hmm |
| 03:58.21 | Kobaz | i think meetme's L option is a little borken |
| 03:58.27 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 03:58.38 | Kobaz | the second arg is supposed to be recurring warning track... i put it to 5000ms, but it never plays |
| 03:58.43 | dudee | pickupexten=*8 |
| 03:58.58 | ChannelZ | hmm |
| 03:59.31 | dudee | Kobaz: L option? |
| 03:59.57 | Kobaz | er, the third option is recurring warning.. the second is a final warning... which doesn't play |
| 04:00.03 | Kobaz | yeah... conference limit |
| 04:01.18 | dudee | Kobaz: does my problem fall under conference? |
| 04:01.56 | Kobaz | what? |
| 04:02.10 | Kobaz | oh, i got it working, never mind... is misread how it works |
| 04:02.13 | ChannelZ | dudee: So you're not trying to *8 on a SIP phone? |
| 04:02.46 | dudee | ChannelZ: not sip, ZAP extension |
| 04:03.34 | ChannelZ | ok just making sure your phone didn't have a dialplan of it's own |
| 04:03.56 | ChannelZ | To be honest I dunno as I don't use any of the feature codes |
| 04:04.24 | Kobaz | feature codes in features.conf? |
| 04:04.25 | dlynes | Are asterisk queues really as complicated as the asterisk queues tutorial makes them out to be? It didn't even seem to be using agent queues, either |
| 04:04.55 | Kobaz | dlynes: it's not really all that complicated, but it's kinda limiting i've found... so i wrote my own queue |
| 04:05.35 | dlynes | Kobaz: oh...so that's why orderlyq.com's written their own, too? |
| 04:06.46 | p3nguin | dlynes: Where I come from, queues are queues... no such thing as "agent queues." |
| 04:06.46 | Kobaz | dlynes: probably |
| 04:06.46 | Kobaz | like, the built in asterisk queue is really inefficient |
| 04:06.46 | p3nguin | You can assign agents as queue members or you can assign SIP channels as members. |
| 04:06.55 | Kobaz | it will make callers wait for no reason, when there are plenty of agents available to take a call |
| 04:07.09 | dlynes | Kobaz: oh...that's bad |
| 04:07.19 | ChannelZ | not if you don't like people |
| 04:07.35 | Kobaz | it's not like, extra time in minutes, but several seconds here, several seconds there |
| 04:07.42 | dlynes | Kobaz: could explain why I've called in to a few companies that seem to be using asterisk, and i get put on hold for about 2 or 3 minutes, even though they don't have any calls |
| 04:07.53 | Kobaz | and when there's sla agreements that callers will be picked up in 4 rings, those seconds add up |
| 04:08.10 | p3nguin | How would you know if they have calls or if they're in the can? |
| 04:08.20 | dlynes | p3nguin: well...there's that, too |
| 04:08.25 | p3nguin | Maybe they didn't have time to answer you for three minutes. |
| 04:08.27 | Kobaz | yeah, the can is the place to be |
| 04:08.48 | dlynes | I don't even think i wanna know what they were doing in the can for that long |
| 04:08.48 | Kobaz | the biggest problem with asterisk queues, is you cannot play tracks, and ring phones at the same time |
| 04:08.56 | dlynes | play tracks? |
| 04:09.04 | dlynes | you mean bet on the ponies? |
| 04:09.08 | ChannelZ | haha |
| 04:09.09 | p3nguin | hmm |
| 04:09.12 | Kobaz | 'thank you for holding, your call is important to us'.... music.... 'thank you for holding' |
| 04:09.20 | dlynes | oh |
| 04:09.22 | dlynes | that sucks |
| 04:09.29 | Kobaz | so, while the track is playing |
| 04:09.34 | Kobaz | it's not ringing any phones |
| 04:09.34 | p3nguin | Uh, why doesn't that work? |
| 04:09.35 | dlynes | why can't you do moh and ring at the same time? |
| 04:09.36 | Kobaz | when it should be |
| 04:09.36 | ChannelZ | I haven't played with queues too much but doesn't it dump them into a MOH context? |
| 04:09.44 | Kobaz | you can do moh and ring |
| 04:09.47 | Kobaz | that's no problem |
| 04:10.02 | Kobaz | but then if you pop out of moh to play a track, no more ringing |
| 04:10.02 | ChannelZ | and there's a periodic announce function |
| 04:10.12 | Kobaz | and if you play periodic announce, no more ringing |
| 04:10.20 | dlynes | Kobaz: to play a track? you mean the periodic announce function? |
| 04:10.24 | Kobaz | yeah |
| 04:10.35 | Kobaz | there's several track options |
| 04:10.43 | Kobaz | position in queue, periodic, welcome, etc |
| 04:10.50 | dlynes | ah...so what happens when it's your turn to ring the phone, in the middle of the track? |
| 04:10.51 | Kobaz | if you are playing any of them, no ringing |
| 04:10.53 | dlynes | you lose your spot in line? |
| 04:11.03 | Kobaz | you don't lose it, it's just delayed |
| 04:11.09 | dlynes | oh |
| 04:11.13 | dlynes | nothing wrong with that, then |
| 04:11.24 | Kobaz | which is fine for like, most casual call centers |
| 04:11.54 | dlynes | as opposed to? |
| 04:12.02 | Kobaz | but when you're setting up a queue for a call center that needs to answer as many calls as possible, as fast as possible |
| 04:12.12 | ChannelZ | apparently one where you hear a lot of ringing instead |
| 04:12.14 | Kobaz | then not ringing phones is a problem |
| 04:13.02 | p3nguin | What type of a ring strategy do you have? |
| 04:14.01 | dlynes | yeah...i'm being asked to set up a system where every extension has the possibility for their own private queue |
| 04:14.20 | Kobaz | p3nguin: i keep ringing until someone picks up |
| 04:14.28 | dlynes | very fun...got the job today...need to have it ready for tomorrow |
| 04:14.32 | p3nguin | That's not a strategy. |
| 04:14.36 | Kobaz | dlynes: i've been asked that too |
| 04:14.46 | Kobaz | p3nguin: oh, it's ringall |
| 04:14.47 | p3nguin | ringall, for example, is a strategy. |
| 04:14.57 | Kobaz | yeah i know the strategy options |
| 04:15.16 | Kobaz | 2 people in a room, taking 4-5 calls at a time |
| 04:15.35 | bcrisp | sounds like hell |
| 04:15.37 | Kobaz | hehe |
| 04:15.43 | Kobaz | they don't seem to mind |
| 04:15.45 | dlynes | 4 to 5 calls at a time on an analog phone? |
| 04:15.47 | p3nguin | How many people can a person talk to at a given moment? |
| 04:16.00 | p3nguin | I'm thinking... one. |
| 04:16.04 | Kobaz | polycom phones |
| 04:16.15 | p3nguin | The rest need to either be on hold or in a queue. |
| 04:16.19 | dlynes | ah...thought polycom could only handle 2 or 3 at a time? |
| 04:16.22 | Kobaz | yeah |
| 04:16.27 | Kobaz | pickup, hello, please hold |
| 04:16.31 | Kobaz | pickup, hello, please hold |
| 04:16.34 | jaytee | I've been working on a weird problem with a few of my Polycom phones that happened today |
| 04:16.35 | Kobaz | pickup, hi, how may i help you |
| 04:16.38 | *** join/#asterisk mintos (n=mvaliyav@117.206.18.61) |
| 04:16.52 | p3nguin | Wouldn't a queue save the trouble of picking up at all? |
| 04:16.53 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 04:16.56 | Kobaz | i still have one last problem with polycom phones |
| 04:17.05 | Kobaz | p3nguin: can't.. the calls *must* be picked up in 4 rings |
| 04:17.10 | p3nguin | I mean, that IS the purpose of the queue. |
| 04:17.21 | *** join/#asterisk geneticx (n=geneticx@adsl-149-108-244.mia.bellsouth.net) |
| 04:17.24 | dlynes | p3nguin: they don't want to lose customers |
| 04:17.27 | jaytee | some of my 550's and 330's would ring but when the user picked up the handset the phone would keep ringing until they pressed the line key |
| 04:17.42 | jaytee | and then often they'd only have one way audio. |
| 04:17.43 | Kobaz | my polycom problem is when you ring the phone for more than 60 seconds, the phone drops the call |
| 04:17.45 | dlynes | p3nguin: some customers will hang up and go elsewhere if they're put on hold |
| 04:17.53 | p3nguin | I don't see the difference between going to a queue and waiting your turn and getting an answer and immediately being put on hold. |
| 04:18.11 | ChannelZ | yeah I'm missing the point as well |
| 04:18.17 | p3nguin | dlynes: Then his callcenter is going to lose customers, cause he's wanting to do that to ALL his callers. |
| 04:18.24 | ChannelZ | Sounds like you just need a secretary |
| 04:18.25 | Kobaz | it's not my call center |
| 04:18.27 | jaytee | Kobaz, do you have a timeout set in your Dial statement? |
| 04:18.33 | Kobaz | jaytee: of course not, heh |
| 04:18.37 | Kobaz | jaytee: the polycom drops the call |
| 04:18.38 | dlynes | p3nguin: none of them are getting put on hold immediately, though |
| 04:18.49 | dlynes | p3nguin: there's some human interaction first |
| 04:18.52 | p3nguin | (2216.27) <Kobaz> pickup, hello, please hold |
| 04:19.03 | dlynes | p3nguin: i.e. 'hello' |
| 04:19.12 | ChannelZ | Playback(hello-please-hold) |
| 04:19.14 | ChannelZ | done |
| 04:19.16 | ChannelZ | :P |
| 04:19.16 | p3nguin | You've obviously missed something. |
| 04:19.17 | Kobaz | there's more to it then that |
| 04:19.18 | Kobaz | haha |
| 04:19.29 | dlynes | I like Channelz's option :) |
| 04:19.34 | p3nguin | channelz: No doubt. |
| 04:19.37 | Kobaz | basically they are answering for multiple companies |
| 04:19.45 | Kobaz | some are more important than others |
| 04:19.56 | p3nguin | Playback(please-hold), then go into the friggin' queue. |
| 04:20.00 | Kobaz | heh |
| 04:20.01 | dlynes | Kobaz: throw them into less important queues, then |
| 04:20.02 | Kobaz | like i said |
| 04:20.03 | Kobaz | SLA |
| 04:20.23 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:20.26 | p3nguin | Create more queues on new extensions. |
| 04:20.36 | dlynes | Kobaz: i.e. implement weighted queues |
| 04:20.46 | Kobaz | wouldn't work |
| 04:20.53 | dlynes | why? |
| 04:20.58 | Kobaz | these people, *must* be picked up |
| 04:21.02 | p3nguin | Not if you refuse to consider it, anyway. |
| 04:21.26 | Kobaz | contractual obligation |
| 04:21.27 | ChannelZ | I'm a customer. I call. What exactly is supposed to happen to me? |
| 04:21.29 | p3nguin | If someone answered my call and then put me on hold, I would probably hang up. |
| 04:21.33 | dlynes | Kobaz: i meant the vip's go straight to the attendant...all others get thrown into weighted queues |
| 04:21.49 | dlynes | p3nguin: you're kidding, right? |
| 04:21.56 | Kobaz | well, we do have two seperate 'queues' |
| 04:22.11 | p3nguin | channelz: ringing... Hi, please hold. music... |
| 04:22.19 | Kobaz | the not-important-at-all people will sit listening to music for as long as it takes to answer them |
| 04:22.31 | Kobaz | but then the important people, there can be 2,3,4,5,6 calls at the same time |
| 04:22.47 | dlynes | Kobaz: then you need to hire more call center agents :) |
| 04:22.48 | Kobaz | but anyways |
| 04:22.50 | Kobaz | heh |
| 04:22.54 | Kobaz | it's not my call center! |
| 04:23.00 | p3nguin | If you're getting put on hold for an undetermined length of time while answering other calls, what the fuck is the difference between that and sitting in a queue for the same length of time? |
| 04:23.01 | Kobaz | it's my customer |
| 04:23.04 | Kobaz | i just write the code |
| 04:23.18 | dlynes | Kobaz: they're paying yoiu enough...hire some more agents for them :) |
| 04:23.21 | heliosj | sighs |
| 04:23.22 | Kobaz | p3nguin: no difference. but... sla requires it |
| 04:23.28 | p3nguin | A good consultant would have explained the logic to the customer. |
| 04:24.22 | Kobaz | anyways |
| 04:24.23 | Kobaz | heh |
| 04:24.35 | Kobaz | the system i finally came up with, works very well, and fits all their needs exactly |
| 04:24.49 | p3nguin | dlynes: No, I'm not kidding. I would rather sit in the queue with the periodic announcement with estimated time before answer as opposed to getting a person and them throwing me on hold without even bothering to talk to me. |
| 04:25.00 | Kobaz | and it's a mish mosh of infinite loops, local channels, and music on hold |
| 04:25.09 | Kobaz | and some background originate's |
| 04:25.22 | dlynes | p3nguin: that hello, please hold sequence is very common for super busy offices |
| 04:25.36 | Kobaz | p3nguin: depending on who is calling in, it might be more than a hello... it just depends |
| 04:25.52 | p3nguin | dlynes: And queues are a better solution for a high call volume. |
| 04:26.03 | Kobaz | p3nguin: depending on your needs |
| 04:26.20 | dlynes | Kobaz: and wants |
| 04:26.31 | p3nguin | Why discriminate? Give everyone the same call treatment. |
| 04:26.40 | Kobaz | i agree, it's a better technical solution... but whether it's the correct solution, is up to the business requirements |
| 04:26.49 | Kobaz | p3nguin: becasue some people are paying more than others |
| 04:26.51 | dlynes | Kobaz: the customer unfortunately is always right...you can argue with them till the cows come home, but ultimately if you don't want to play ball their way, they'll hire someone else |
| 04:26.58 | Kobaz | p3nguin: and some people are paying a *lot* more than others |
| 04:27.23 | bcrisp | hmm |
| 04:27.26 | dlynes | Kobaz: and so you're stuck making asterisk work like a legacy system, because the customer doesn't want to adjust their way of thinking |
| 04:27.30 | p3nguin | So you're going to manually screen the calls. |
| 04:27.33 | Kobaz | dlynes: precisely |
| 04:27.50 | bcrisp | id rather them go through a maze than have to screen them myself |
| 04:27.57 | Kobaz | dlynes: we replaced their 10 year old siemens system.. and the new one does exactly the same thing, with some bonus features |
| 04:27.59 | ChannelZ | There's no point in arguing this. Kobaz has a system of 'rudeness with a slight human touch' and it seems to work for him and his customer |
| 04:28.06 | Kobaz | hehe |
| 04:28.18 | dlynes | Kobaz: yeah...i've had to deal with the same bs, catering to ex-nortel customers |
| 04:28.21 | p3nguin | I'd prefer to give my "better" clients a "better" extension with a "better" representative. |
| 04:28.28 | dlynes | Kobaz: that brought me through asterisk blf hell |
| 04:28.41 | Kobaz | blf... |
| 04:28.46 | Kobaz | i love when the light is still flashing madly |
| 04:28.53 | Kobaz | but the phone stopped ringing in the other room |
| 04:28.55 | Kobaz | and it's still flashing |
| 04:28.56 | dlynes | Kobaz: blf is a four letter word in our office |
| 04:29.13 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:29.39 | dlynes | well, there's that, and the random crashing of phones with all the blf lights lit up, and no longer responding to calls |
| 04:29.55 | Kobaz | ChannelZ: it's not really rudeness... without going into an hour long discussion of how their business works... it's a system that works for everyone, my customer, and her customers |
| 04:29.56 | dlynes | really fun stuff...and the odd random rebooting after it gets all blf confused, too |
| 04:30.17 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 04:30.21 | Kobaz | the most annoying thing is... |
| 04:30.27 | Kobaz | as soon as i start to really trust asterisk |
| 04:30.34 | p3nguin | Okay, off of the business models... I have a question about conditional branching using a GotoIf(). |
| 04:30.38 | Kobaz | it will crash and drop a dozen calls |
| 04:30.50 | p3nguin | exten => 201,n,GotoIf($["${DIALSTATUS}" = "NOANWSER"]?:followme,201,1) |
| 04:30.58 | Kobaz | ael is your friend |
| 04:31.02 | dlynes | Kobaz: damn...but then again...I've never fully trusted asterisk |
| 04:31.05 | p3nguin | That line is after the Dial() for 201, obviously. |
| 04:31.15 | p3nguin | uh, sip/201, I mean. |
| 04:31.30 | Kobaz | if ("${DIALSTATUS}" == "NOANWSER") { &followme(); } |
| 04:31.49 | *** join/#asterisk psykon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 04:31.51 | p3nguin | Now, to me, that says if the dialstatus IS NOT NOANSWER, followme. |
| 04:31.54 | Kobaz | i got sick and tired of programming in 'basic' |
| 04:32.25 | dlynes | p3nguin: wrong |
| 04:32.26 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 04:32.28 | dlynes | p3nguin: it's not C/C++ |
| 04:32.43 | p3nguin | GotoIf(expression?destination1:destination2) |
| 04:32.58 | p3nguin | If true, go to destination1. |
| 04:33.02 | Kobaz | ael! |
| 04:33.03 | ChannelZ | is followme really an extension label? |
| 04:33.04 | dlynes | p3nguin: if the condition is true, it goes to followme, else it goes to 201,1 |
| 04:33.11 | psykon | i have *got* to figure out why this box keeps loosing it's connection to the net after a reboot before it gets put in production. I took the same drive and put it into a pIII and it runs better... |
| 04:33.15 | p3nguin | channelz: No, it is a context. |
| 04:33.28 | Kobaz | if ("${DIALSTATUS}" == "NOANWSER") { &followme(); } <---- isn't that much better |
| 04:33.29 | p3nguin | dlynes: No, that's incorrect. |
| 04:33.41 | dlynes | p3nguin: no, it is not.....it'll go to n(followme) |
| 04:33.53 | ChannelZ | what he said |
| 04:34.00 | ChannelZ | GotoIf jumps to labels |
| 04:34.04 | p3nguin | dlynes: If the status is false, it goes to context followme, exten 201, priority 1. |
| 04:34.30 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 04:34.45 | dlynes | p3nguin: GotoIf(condition?[labeliftrue]:[labeliffalse]) |
| 04:35.08 | dlynes | p3nguin: that's straight from core show application gotoif |
| 04:35.23 | Kobaz | noone cares about ael :( |
| 04:35.39 | dlynes | p3nguin: and as I said and ChannelZ said also, it jumps to the label, followme, not the context |
| 04:36.01 | p3nguin | The destinations can each be one of the following: ... ... # |
| 04:36.02 | p3nguin | A context, extension, and priority label, such as incoming,123,weasels |
| 04:36.27 | p3nguin | context incoming, exten 123 ... |
| 04:36.32 | dlynes | Kobaz: I've just never used it because in earlier versions of 1.4, it wasn't terribly stable, and I havent' bothered to check to see if it's stablized yet |
| 04:36.44 | p3nguin | Same as mine, but I have a priority NUMBER of 1 instead of a label. |
| 04:37.05 | p3nguin | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-3.html |
| 04:37.26 | *** join/#asterisk torriem (n=torriem@c-67-186-209-155.hsd1.ut.comcast.net) |
| 04:37.37 | Kobaz | dlynes: it's really great in 1.6 |
| 04:38.04 | Kobaz | i haven't touched extensions.conf in like a year |
| 04:38.32 | dlynes | p3nguin: have you tried it? |
| 04:38.34 | p3nguin | Is it okay to put a label on priority 1? |
| 04:38.38 | drmessano | AEL seems pointless |
| 04:38.48 | dlynes | p3nguin: I'm reading the docs too, but in reality it doesn't perform that way |
| 04:39.01 | Kobaz | drmessano: well, i'm slowly converting most of my ael to agi |
| 04:39.22 | Kobaz | drmessano: but if you don't want/need to write agi, it's much much easier to write complex dialplan in ael, than in extensions.conf |
| 04:39.24 | dlynes | p3nguin: i'm guessing what they're calling context for a single argument is really a label |
| 04:39.34 | *** join/#asterisk DaveCanoe (n=Dave@strike-dsl.dclg.ca) |
| 04:39.39 | Kobaz | it's programming in a linear style versus structured |
| 04:39.42 | p3nguin | dlynes: No, they mean a CONTEXT in the extensions.conf file. |
| 04:39.43 | dlynes | p3nguin: label, then extension,priority, then context,extension,priority |
| 04:39.55 | dlynes | p3nguin: depending on whether you give 1, 2 or 3 arguments |
| 04:40.13 | p3nguin | exten => 201,n,GotoIf($["${DIALSTATUS}" = "NOANWSER"]?:followme,201,1) |
| 04:40.39 | dlynes | p3nguin: oh shit...nvm....I just realized they left the first operator value out |
| 04:40.40 | p3nguin | exten => 201,n,GotoIf($["${DIALSTATUS}" = "NOANWSER"]?[gohereiftrue]:followme,201,1) |
| 04:40.47 | ChannelZ | either the docs lie or it doesn't like your ?: - what actually happens with the above? |
| 04:41.12 | p3nguin | channelz: ?: is just leaving out the iftrue part. |
| 04:41.16 | ChannelZ | I know |
| 04:41.19 | dlynes | ChannelZ: well nothing will happen with the above, because it'll never match |
| 04:41.33 | dlynes | ChannelZ: DIALSTATUS will never have a value of NOANWSER |
| 04:41.41 | p3nguin | Here's the problem... |
| 04:41.58 | dlynes | It should be NOANSWER, not NOANWSER |
| 04:42.15 | p3nguin | Oh, a typo? |
| 04:42.19 | dlynes | YES |
| 04:42.25 | p3nguin | THANK YOU! |
| 04:42.29 | p3nguin | Let me correct that. |
| 04:42.30 | Kobaz | hehe |
| 04:42.30 | dlynes | LOL!!! |
| 04:42.46 | p3nguin | I could not figure out WTF was wrong. |
| 04:42.54 | dlynes | Kobaz: btw....ael wouldn't help him out with that :) |
| 04:43.12 | dlynes | Kobaz: i guess ael2 is pretty stable now, too? |
| 04:43.19 | ChannelZ | nice catch |
| 04:43.40 | Katty | what the frell |
| 04:43.58 | dlynes | Katty! what are you still doing up???? Go to bed! :) |
| 04:43.59 | ChannelZ | let me aks you somethin |
| 04:44.04 | Katty | ^_- |
| 04:44.16 | Katty | i am pretty tired. |
| 04:44.43 | p3nguin | Okay, changed to: exten => 201,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?:followme,201,1) |
| 04:45.03 | p3nguin | Now it should evaluate the DIALSTATUS and compare it to NOANSWER. |
| 04:45.04 | dlynes | claps for p3nguin |
| 04:45.12 | p3nguin | If it is NOANSWER, nothing should happen. |
| 04:45.24 | p3nguin | If it is not NOANSWER, it should followme, right? |
| 04:45.33 | ChannelZ | supposedly |
| 04:45.42 | p3nguin | checks to see what REALLY happens. |
| 04:45.43 | dlynes | p3nguin: correct...go to extension 201, priority 1, in context followme |
| 04:46.09 | dlynes | p3nguin: btw...DIALSTATUS sometimes equals "" |
| 04:46.14 | Kobaz | dlynes: well ael is more like a real modern language |
| 04:46.19 | dlynes | p3nguin: you might want to watch out for that |
| 04:46.20 | p3nguin | Let me throw in a verbose() to see what the DIALSTATUS is just to be sure. |
| 04:47.35 | dlynes | I guess verbose() is a new application? |
| 04:47.40 | dlynes | I don't remember seeing it before |
| 04:48.18 | p3nguin | Okay, DIALSTATUS was NOANSWER and it went to the next priority, which was Hangup(), so it worked okay. |
| 04:48.38 | psykon | p3nguin, do you usually use freepbx in production or do you use a pure asterisk setup? |
| 04:48.39 | Kobaz | why would you not want to do the followme if you got a noanswer |
| 04:48.40 | p3nguin | Now I need to move that : to the other side of the followme,201,1. |
| 04:48.55 | p3nguin | psykon: I've never touched FreePBX. |
| 04:49.08 | psykon | p3nguin, what about trixbox? |
| 04:49.11 | p3nguin | kobaz: No clue. I saw that on the wiki. |
| 04:49.19 | p3nguin | psykon: Never touched that, either. |
| 04:49.29 | Kobaz | i would think anything other than an ANSWER, would be sent to the followme |
| 04:49.30 | dlynes | p3nguin: fwiw, in practice, when DIALSTATUS has a value of "", it seems to be the same as CHANUNAVAIL |
| 04:49.46 | p3nguin | kobaz: I was trying to evaluate it and could not figure out what it was supposed to be doing. |
| 04:50.14 | Kobaz | but then, you'll hang up on ANSWER anyway, unless you pass in the g option, |
| 04:50.23 | Kobaz | well, hang up after the call is done |
| 04:50.55 | Kobaz | NOANSWER means it rang and rang, but noone picked up |
| 04:51.22 | p3nguin | Where's the list of possible DIALSTATUSes? |
| 04:51.30 | Kobaz | core show application Dial |
| 04:51.53 | psykon | p3guin, oh wow. yeah I have been wondering how to formulate a question that is basically pbxinaflash vs. pure asterisk without looking like a complete troll. I am rolling out a 15 channel system soon and a buddy talked me into pbxinaflash... but I wonder about efficiency |
| 04:52.19 | Kobaz | efficiency of the system is going to be negligable between all the above |
| 04:52.28 | Kobaz | efficiency of administration on the other hand, will vary |
| 04:52.56 | Kobaz | but, we roll our own gui and db backend, so i prefer plain asterisk |
| 04:55.45 | p3nguin | Okay, I changed that so that if the DIALSTATUS is NOANSWER, it can go to the followme context to try to follow me, but what would be a reasonable thing to do when the DIALSTATUS is anything else? Continue in the dialplan, possibly end up in voicemail? |
| 04:56.40 | p3nguin | I'm not using g in the Dial() command, by the way. |
| 04:57.12 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 04:57.24 | dlynes | p3nguin: as I stated earlier though, there's an additional DIALSTATUS that's not documented by Dial()...there's also an empty DIALSTATUS |
| 04:58.23 | p3nguin | dlynes: I really only care if the call was not answered... so it can followme. A null DIALSTATUS would never evaluate true anyway. |
| 04:58.25 | dlynes | p3nguin: what would be reasonable is all dependent on your particular situation |
| 04:59.02 | p3nguin | Without the g option in the Dial() command, when the call is ended, the dialplan should end, right? |
| 04:59.15 | p3nguin | With g, that's when it keeps going. |
| 04:59.18 | dlynes | p3nguin: it depends |
| 04:59.30 | dlynes | p3nguin: what's autofallthrough= set to? |
| 04:59.43 | p3nguin | yes |
| 04:59.54 | dlynes | Then yeah...it should keep going |
| 05:00.15 | dlynes | Unless you get a dialstatus of ANSWERED...then it'll execute the 'h' extension |
| 05:00.51 | makafre | hey guys, anybody knows how to configure a linksys rtp 300 to connect with asterisk? |
| 05:01.11 | dlynes | makafre: fill in the fields with appropriate values? |
| 05:01.21 | dlynes | makafre: and then click 'save'? |
| 05:01.48 | makafre | dlynes: and click confirm as well :- ) |
| 05:02.50 | p3nguin | Okay, so the typo made the evaluation always false. So I fixed the typo (which I totally could not see), and changed the if true/if false destinations, now it behaves like I expected. |
| 05:03.10 | p3nguin | If was the fact the the evaluation was always false that I brought up the topic. |
| 05:03.35 | dlynes | makafre: it seems to be quite similar to the Linksys/WRP400 |
| 05:03.46 | p3nguin | I could not figure out how it could always be false when the DIALSTATUS was exactly what I was comparing it to. |
| 05:04.06 | dlynes | makafre: try the following link for the WRP400 manual: http://www.google.ca/url?sa=t&source=web&ct=res&cd=8&ved=0CCAQFjAH&url=http%3A%2F%2Fwww.linksys.com%2Fservlet%2FSatellite%3Fblobcol%3Durldata%26blobheadername1%3DContent-Type%26blobheadername2%3DContent-Disposition%26blobheadervalue1%3Dapplication%252Fpdf%26blobheadervalue2%3Dinline%253B%2Bfilename%253DWRP400_V10_UG_B-WEB.pdf%26blobkey%3Did%2 |
| 05:04.08 | dlynes | 6blobtable%3DMungoBlobs%26blobwhere%3D1193800771924%26ssbinary%3Dtrue%26lid%3D8910889895B05&ei=0ZcYS5-EHo_gnAf0wJzaAw&usg=AFQjCNE94EaU-CXY0i3-rAi6QTspYRODtg&sig2=bGFEv1e_4uJOqm-mEGj5Hw |
| 05:04.10 | dlynes | erm |
| 05:04.53 | dlynes | makafre: Just paste those two lines together for the url |
| 05:06.08 | makafre | hehe |
| 05:06.16 | makafre | let me try to fetch this |
| 05:06.19 | p3nguin | lol |
| 05:06.24 | p3nguin | tinyurl! |
| 05:06.46 | p3nguin | Does vim have a spell checker? :D |
| 05:12.37 | *** part/#asterisk torriem (n=torriem@c-67-186-209-155.hsd1.ut.comcast.net) |
| 05:13.31 | dlynes | p3nguin: yes |
| 05:14.15 | dlynes | p3nguin: http://www.vim.org/scripts/script.php?script_id=135 |
| 05:15.06 | dlynes | p3nguin: or better yet, so you can pick your own: http://bit.ly/5bl0FG |
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| 05:27.46 | dlynes | Kobaz: I'm guessing multi-tenanted queues are also possible? |
| 05:33.59 | voipmonk | thats a nice word, multi-tenanted |
| 05:34.05 | voipmonk | can I use that? :) |
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| 05:45.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:45.55 | Kobaz | dlynes: yeah, you can do like... queue companya_foo, and queue companyb_foo |
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| 06:11.58 | *** join/#asterisk dudee (n=Karan@cm9.sigma53.maxonline.com.sg) |
| 06:12.15 | dudee | Hi I have a issue, hope somebody here could assist me, I have setup a ring group and a person can ring into that ring group , everything there is okay. But lets say if it rings and somebody on DAHDI/1 was on the line, it would not ring DAHDI/1, only /2 /3 /4 and so... how can /1 pick up the phone with out going to /2 desk to pick it up ? ( ** ) didnt work , *8 gave a busy / engaged tone. please advice. thank you |
| 06:12.23 | dlynes | Kobaz: yeah...i see that, now |
| 06:12.26 | dlynes | Kobaz: thanks |
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| 06:21.56 | p3nguin | dudee: There's no call waiting option? |
| 06:24.54 | ChannelZ | pete and repeat... |
| 06:25.09 | p3nguin | Been here before? |
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| 06:25.22 | ChannelZ | little earlier. These are analog phones |
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| 06:27.19 | ChannelZ | dudee: you say you have a ring group, do you have a pickup group defined for the applicable channels? Lets see some configs |
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| 06:27.59 | p3nguin | This should be a little better than the previous configuration: http://pastebin.com/d3b62590e |
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| 06:29.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 06:30.06 | pif | hi, how do I 'reload' in 1.6 ? |
| 06:30.15 | pif | I tried "core reload now" |
| 06:30.25 | p3nguin | "restart when convenient" |
| 06:30.31 | pif | not restart |
| 06:30.33 | p3nguin | "restart now" |
| 06:30.42 | pif | i want reload |
| 06:30.43 | p3nguin | There is no "reload" to my knowledge. |
| 06:30.59 | p3nguin | You can do "module reload <some module>" I think. |
| 06:31.06 | pif | ok |
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| 06:44.59 | Micc | Are there any good free cdr analyzers? |
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| 07:30.27 | MAbbas | Hi All, I want to know from where following command pulls information about queue stats |
| 07:30.33 | MAbbas | show queue <queue-name> |
| 07:30.59 | MAbbas | DB or somewhere else? |
| 07:37.27 | p3nguin | I guess app_queue.so probably loads them into memory. |
| 07:37.44 | p3nguin | But you should be using "queue show" and not "show queue" |
| 07:39.14 | MAbbas | p3nguin>: is there way I can access them in my AGI script? |
| 07:43.48 | p3nguin | I'm sure there's a way. |
| 07:43.55 | p3nguin | Maybe the AMI can help. |
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| 07:48.06 | ChannelZ | MAbbas: Didn't we go over this last night? |
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| 09:13.39 | pif | has the command "core reload now" disappeared in 1.6.x ? |
| 09:14.18 | ChannelZ | AFAIK yes (I don't run 1.6) |
| 09:15.09 | kaldemar | pif: has there been such a command in some version? use "module reload". |
| 09:15.23 | *** part/#asterisk fiddur (n=fiddur@dhcp01.textalk.com) |
| 09:17.07 | ChannelZ | 1.4 had just 'reload' (not 'core reload') |
| 09:18.05 | pif | oh well |
| 09:19.30 | ChannelZ | oh well what? |
| 09:19.46 | trogs | just a "reload" should work |
| 09:19.53 | trogs | in 1.6 anyway |
| 09:20.13 | pif | *smacks forehead* |
| 09:21.42 | pif | in my 1.6.2.0 it works, not my client's 1.6.2.0 , go figure |
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| 09:21.59 | pif | features.conf ? |
| 09:22.02 | kaldemar | trogs: not unless it's defined in cli_aliases.conf and the alias module exists |
| 09:22.15 | pif | now you're talking |
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| 09:22.48 | kaldemar | res_clialiases that is |
| 09:22.51 | trogs | i don't have that config file, so it's whatever the default is |
| 09:23.35 | kaldemar | trogs: check again |
| 09:23.50 | pif | reload=module reload |
| 09:24.06 | kaldemar | reload is mapped to "module reload" in cli_aliases.conf |
| 09:24.30 | kaldemar | without it, you get "No such command" |
| 09:24.36 | trogs | well, i don't have that file |
| 09:24.44 | trogs | this is on 1.6.0.1 btw, ancient :) |
| 09:27.01 | pif | someone using wcb4xxp ? |
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| 09:34.16 | schajee | Q: Is it possible to send captured IVR data to a softphone via SIP? In my case my application receives a call request (SIP) and displays the captured IVR data. |
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| 09:58.04 | mbrevda | im using this in a system command, should I have []'s somewhere? where? --var="${foo}?${foo}:${bar}" |
| 09:59.29 | kaldemar | mbrevda: is that supposed to be an if statement? |
| 09:59.42 | mbrevda | yup |
| 09:59.52 | kaldemar | core show function IF |
| 10:00.32 | kaldemar | an expression is something inside $[] |
| 10:00.36 | mbrevda | kaldemar: in as much as conditional statments are if statemnets. im not actualy suing the if function, im trying to use a conditional operator |
| 10:01.35 | kaldemar | Set(var=${IF($[insert expression here]?{foo}:${bar})}) |
| 10:01.53 | mbrevda | hmm |
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| 10:11.48 | mbrevda | kaldemar: its condition?true:false right? |
| 10:12.24 | kaldemar | yes, just like core show function IF tells you |
| 10:13.49 | mbrevda | --status="${IF($[${foo}]?${foo}:${bar})}" <--- never gettinging anything when foo is blank |
| 10:18.40 | kaldemar | it's not a valid expression then |
| 10:19.09 | mbrevda | how would you right it then? |
| 10:19.27 | mbrevda | *write |
| 10:21.15 | kaldemar | for example --status="${IF($["${foo}x" = "x"]?${bar}:${foo})}" |
| 10:21.42 | kaldemar | what do you want to test? for foo existance? |
| 10:22.18 | mbrevda | yup |
| 10:23.08 | kaldemar | there's a function for that too, --status="${IF($[${EXISTS(${foo})}]?${foo}:${bar})}" |
| 10:23.53 | mbrevda | nice! thanks! |
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| 11:00.09 | Omorika | chan_dahdi.c:12244 dahdi_pri_error: ACK received for '0' outside of window of '5' to '6', restarting |
| 11:00.25 | Omorika | what's the meaning of this message? |
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| 11:18.10 | gestahlt | Hi |
| 11:18.15 | gestahlt | I need LAN Phones |
| 11:18.40 | gestahlt | I need to have the functionality to see on which lines are busy |
| 11:18.50 | gestahlt | Do you guys have any recommentation? |
| 11:19.05 | gestahlt | Like AMPPortal, just on the phone itself |
| 11:20.34 | gestahlt | I also need a FXS Card with 8 Ports |
| 11:21.58 | Gido-E | gestahlt allot of peole on this channel are living in the usa |
| 11:22.03 | Gido-E | so they are sleeping. |
| 11:22.03 | sun28 | gestahlt: Linksys SPA921/SPA941 |
| 11:22.29 | sun28 | living in russia, 4 PM |
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| 11:23.29 | gestahlt | sun: Can i use this for the reception? They need to see which ppl are currently having a call |
| 11:23.53 | gestahlt | sun: and is there also something i have to configure in asterisk for that? |
| 11:23.55 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 11:24.02 | sun28 | asterisk -rx 'sip show inuse' |
| 11:24.21 | sun28 | gestahlt: yeah of course |
| 11:24.37 | gestahlt | sun28: wonderful, thank you |
| 11:25.07 | sun28 | will be try a FXS/FXO gateway, not PC card |
| 11:25.36 | sun28 | PC card like mamonth stonyshit |
| 11:26.15 | sun28 | see Audiocodes |
| 11:26.54 | sun28 | this is not cheap, but is the best way |
| 11:27.31 | bbt | getting a multifunction laser printer that has fax - what would the best way to get the fax to work on it via asterisk? |
| 11:28.00 | Gido-E | bbt spa3102 orso. |
| 11:28.05 | gestahlt | sun28: Do you have any recommentation on the brand? |
| 11:28.06 | sun28 | bbt: UDPTL |
| 11:28.14 | sun28 | gestahlt: Audiocodes |
| 11:29.16 | sun28 | bbt: oh, you gest a hardware |
| 11:32.12 | bbt | i have a broadband modem Billion 7404vnpx which has two voip ports. but i am routing the voip to the asterisk box. can i still make use of one of them FXS ports? |
| 11:35.29 | gestahlt | sun28: That idea with a FXS gateway is great |
| 11:35.54 | gestahlt | sun28: i found a Linksys SPA8000 for this |
| 11:36.16 | gestahlt | sun28: cheaper than those cards and also easier to deploy |
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| 11:38.03 | gestahlt | sun28: Okay, now i need a internal S0 Bus or Gateway and im fine |
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| 11:45.48 | gestahlt | are those netcologne HFC-S cards for around 20 ⬠usable? |
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| 12:26.30 | *** join/#asterisk Spiky-{roam}- (n=spiky@195.234.242.5) |
| 12:27.37 | Spiky-{roam}- | hi all. new to asterisk. can someone tell me that if i use the yum install for centos/rhel, do i end up with everything that asteriskNow has? |
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| 12:32.04 | Superbartt | Spiky-{roam}- not sure, but why would you want to use yum? just compile it :x |
| 12:32.08 | kaldemar | Spiky-{roam}-: no. |
| 12:32.49 | kaldemar | Spiky-{roam}-: and there's #asterisknow for questions related to it. people here don't generally use asterisknow. |
| 12:42.46 | Spiky-{roam}- | Superbartt: since i'm new to it all, i thought it might be benefitial to see what it does before i expend the effort to hack my own |
| 12:47.18 | kaldemar | are you looking into using asterisknow or just asterisk? |
| 12:47.39 | Spiky-{roam}- | kaldemar: i dont plan on using asteriskNow. it was a question about asterisk install |
| 12:47.59 | Spiky-{roam}- | kaldemar: looking to use *, not Now |
| 12:48.13 | Spiky-{roam}- | but will check out Now just for a demo |
| 12:48.22 | kaldemar | asterisknow is a distro with asterisk, freepbx and other stuff. you won't get the gui with just asterisk. |
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| 12:51.29 | Spiky-{roam}- | kaldemar: right, being a distro is a problem for me, that's why i need the plain * |
| 12:51.36 | Spiky-{roam}- | i need it to run on centos |
| 12:52.09 | kaldemar | it will |
| 12:52.25 | Spiky-{roam}- | so without the freepbx and gui, how do ppl here use it? |
| 12:52.37 | Spiky-{roam}- | you guys just configure it using CLI? |
| 12:52.41 | kaldemar | they edit the configuration files. |
| 12:52.46 | Spiky-{roam}- | i see |
| 12:52.48 | Spiky-{roam}- | thanks |
| 12:52.50 | Superbartt | CLI, configuration files, and knowledge :p |
| 12:52.53 | Superbartt | !book |
| 12:53.04 | kaldemar | with a GUI you lose an enormous amount of flexibility and features. |
| 12:53.05 | Superbartt | !asterisk-book |
| 12:53.09 | Spiky-{roam}- | hmmm, that means a learning curve |
| 12:53.10 | kaldemar | ~book |
| 12:53.11 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:53.18 | Superbartt | ah, that was the command :p |
| 12:53.20 | kaldemar | ~answers |
| 12:53.21 | infobot | hmm... answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:53.54 | Spiky-{roam}- | how much is oreilly paying you guys :D |
| 12:54.03 | Superbartt | with those information the learning curve droppes alot :p |
| 12:54.14 | kaldemar | nada. the book is available as a free PDF too. :) |
| 12:54.29 | Superbartt | well nothing, the book is free of charge, you can get it in html and pdf, oreilly just has a print out of it for the people who prefer that:) |
| 12:55.02 | Spiky-{roam}- | oh great. thanks guys |
| 12:55.49 | Superbartt | just take a week on your attic or dark cellar, read the book and in the mean time build your system with the new found knowledge, in the end you can do a little asterisk :p |
| 12:56.49 | Spiky-{roam}- | just to confirm my understanding... freepbx IS the gui, and nothing but? |
| 12:56.56 | kaldemar | yes |
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| 12:57.06 | kaldemar | or A gui |
| 12:57.18 | Spiky-{roam}- | got it. cheers |
| 12:57.27 | kaldemar | there are others also |
| 12:57.40 | Superbartt | there are multiple available, asterisk-gui, freepbx, trixbox, etc :p |
| 12:57.52 | Spiky-{roam}- | info overload! |
| 12:57.55 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 12:57.59 | Spiky-{roam}- | one is enough for now :) |
| 12:58.04 | Superbartt | welcome to asterisk ;) |
| 12:59.43 | Spiky-{roam}- | and i can install freepbx myself on centos too, right? |
| 13:00.16 | kaldemar | right |
| 13:01.01 | Spiky-{roam}- | i have a mitel 5224 (i believe a SIP phone) it will just PnP with this * setup, am i right in thinking that? |
| 13:01.39 | kaldemar | if it uses standard SIP, most likely. |
| 13:02.11 | Spiky-{roam}- | kaldemar: cool. now just need to find the time to play with it all |
| 13:02.22 | Spiky-{roam}- | thanks for the book recommendation |
| 13:12.58 | *** join/#asterisk c0rnoTa (n=c0rnoTa@109.188.9.105) |
| 13:12.59 | *** part/#asterisk c0rnoTa (n=c0rnoTa@109.188.9.105) |
| 13:14.32 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 13:17.57 | *** join/#asterisk viraptor (n=viraptor@212.11.80.34) |
| 13:18.00 | *** join/#asterisk michael-i (n=michael-@141.41.40.153) |
| 13:18.14 | *** join/#asterisk shinao1 (n=shinao1@41.219.229.196) |
| 13:19.42 | viraptor | hi, could someone tell me what does asterisk really do after receiving NEW and before ACCEPT (assuming ip-based auth) - I've got a problem where 1000th or so call takes up to 10 seconds to send back ACCEPT |
| 13:27.29 | mtltemplar | hello all. i need some help with a digium/zaptel problem. Im getting yellow errors on all my spans after a power outage and I cant get the card out of that mode. has anyone here ever dealt with that? |
| 13:28.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:53.05 | fcois93 | which module have I to load to have the 'console' command ? |
| 13:53.11 | fcois93 | in the CLI |
| 13:54.31 | Pan3D | morning |
| 13:55.50 | kaldemar | fcois93: chan_oss or chan_alsa |
| 13:56.12 | fcois93 | kaldemar: thank you |
| 13:56.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:56.38 | *** join/#asterisk wolfeyes (i=2913c2cf@gateway/web/freenode/x-eqzwzlomaivjklge) |
| 13:57.48 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 13:57.54 | drfreeze | Good morning |
| 13:58.34 | wolfeyes | Good day all |
| 14:02.38 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 14:07.51 | Pan3D | wolfeyes: there is a band called wolfeyes from where I am, but I think their music would scare you. good morning :) |
| 14:08.50 | viraptor | trying again :) could someone tell me what does asterisk really do after receiving NEW and before sending ACCEPT (assuming ip-based auth, no pass) - I've got a problem where 1000th or so call takes up to 10 seconds to send back ACCEPT |
| 14:09.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.21 | wolfeyes | ha ha ha ha Pan3D |
| 14:10.55 | wolfeyes | I think it might |
| 14:10.57 | wolfeyes | lol |
| 14:12.10 | *** join/#asterisk ManxPower-work (n=EWieling@84.sub-70-223-191.myvzw.com) |
| 14:12.25 | ManxPower-work | ~answers |
| 14:12.26 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 14:13.51 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:15.48 | wolfeyes | Anyone here from South Africa working with Asterisk? |
| 14:16.19 | gr0mit | i have a customer in Durban, does that count? |
| 14:16.32 | wolfeyes | lol |
| 14:16.43 | wolfeyes | and you are based where? |
| 14:17.05 | wolfeyes | If I may ask? |
| 14:18.25 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) |
| 14:18.46 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:18.52 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 14:21.12 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:22.30 | voipmonk | what are you looking to do wolfeyes? |
| 14:23.34 | ManxPower-work | ~ask |
| 14:23.35 | 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. |
| 14:24.25 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
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| 14:47.01 | Katty | http://www.filedropper.com/angiecoupon <- free 16 oz international delight creamer. |
| 14:48.54 | *** join/#asterisk c0rnoTa (n=c0rnoTa@109.188.9.105) |
| 14:48.55 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
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| 14:51.14 | *** join/#asterisk smooth_penguin (n=smoove@59.95.17.195) |
| 14:51.45 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
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| 14:58.27 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 14:59.50 | *** join/#asterisk gscmans (n=guna@212.36.53.178) |
| 15:01.31 | gscmans | Hi i need advise on Ast* 1.4 jitter buffer configuration for <some VoIP channel>==>*==>SIP; the SNOM300 phones seems not be able to handle huge jitter(640ms) from other channel |
| 15:04.27 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 15:06.25 | Amorsen | It's hardly surprising that a phone doesn't play well with 640ms jitter, is it? |
| 15:08.26 | gscmans | yeah but unfortunately one of the source due from other channel create this jitter due to issue with recorder |
| 15:08.45 | gscmans | so i am trying to smooth out the jitter in Ast* |
| 15:10.06 | gscmans | i used jbforce=yes on sip.conf |
| 15:10.53 | gscmans | with jbmaxsize = 3000 and jbresyncthreshold = 5000 |
| 15:11.02 | gscmans | but it is not making much of differencec |
| 15:11.17 | gscmans | this jitterbuffer configuration in Ast is a big confusion anyway |
| 15:12.42 | *** join/#asterisk VaGoNeTaS (n=debian@200.111.138.170) |
| 15:12.47 | VaGoNeTaS | hello everybody |
| 15:13.00 | Zeeek | quiet nights |
| 15:13.18 | VaGoNeTaS | i trying to restrict outgoing calls at the office by hours |
| 15:13.27 | VaGoNeTaS | but by some reasson i dont know, is not working |
| 15:13.31 | VaGoNeTaS | http://www.pastebin.ca/1701882 |
| 15:13.43 | VaGoNeTaS | any ideas? |
| 15:15.10 | Corydon76-dig | VaGoNeTaS: looks like it's working to me |
| 15:15.31 | VaGoNeTaS | yep, but look down |
| 15:15.33 | VaGoNeTaS | at the CLI |
| 15:15.41 | VaGoNeTaS | the call is not going through |
| 15:15.52 | Corydon76-dig | You forget that you've CHANGED the extension with the Goto |
| 15:15.59 | kaldemar | VaGoNeTaS: you're going to s |
| 15:16.15 | Corydon76-dig | Then extension is NO LONGER 9... It's now "s" |
| 15:16.32 | VaGoNeTaS | but i need to call outside not to s |
| 15:16.35 | VaGoNeTaS | hmmm |
| 15:16.45 | VaGoNeTaS | how should i do it ? |
| 15:16.47 | Corydon76-dig | Then don't goto s,1 |
| 15:17.02 | VaGoNeTaS | where should i go? |
| 15:17.07 | Corydon76-dig | Probably what you want is miniout,${EXTEN},1 |
| 15:17.28 | VaGoNeTaS | let me give it a try |
| 15:17.30 | VaGoNeTaS | just a sec |
| 15:17.30 | kaldemar | better way would be to use includes instead of goto. |
| 15:17.40 | Corydon76-dig | But you'll need to change those s extensions |
| 15:17.48 | Corydon76-dig | kaldemar: not with GotoIfTime |
| 15:18.12 | Corydon76-dig | kaldemar: There's a really bad edge case with timed includes |
| 15:18.39 | kaldemar | Corydon76-dig: what is it? |
| 15:19.03 | Corydon76-dig | kaldemar: What happens if the time changes between 2 steps? |
| 15:19.30 | kaldemar | depends on how the includes and time are handled |
| 15:19.51 | kaldemar | does it leave a minute in between? |
| 15:20.12 | Corydon76-dig | kaldemar: what you're executing could suddenly disappear out from under you, and you could be nowhere or in the middle of a completely unrelated dialplan |
| 15:20.44 | Corydon76-dig | Timed includes don't include any smarts for existing channels |
| 15:21.12 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:21.13 | [TK]D-Fender | VaGoNeTaS: http://www.pastebin.ca/1701907 |
| 15:21.14 | Corydon76-dig | One second, the dialplan is there, the next, poof! |
| 15:21.20 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:21.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:21.31 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 15:21.41 | bcrisp | mornin |
| 15:22.02 | *** join/#asterisk jpcansa (n=jpbenavi@200.91.99.148) |
| 15:22.13 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 15:22.18 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:22.35 | kaldemar | Corydon76-dig: well, the engine is obviously FUBAR then when it comes to this. |
| 15:22.55 | Corydon76-dig | kaldemar: that's why GotoIfTime is a much better option |
| 15:24.01 | kaldemar | i've always hated goto's, they make control much harder in complex dialplans. |
| 15:24.46 | bcrisp | spaghetti code anyone? |
| 15:25.16 | [TK]D-Fender | With MEATBALLS! |
| 15:25.28 | VaGoNeTaS | ok, it worked |
| 15:25.34 | VaGoNeTaS | but, i had an little issue |
| 15:25.40 | bcrisp | i about had a heart attack when i saw "Gosub" |
| 15:25.43 | VaGoNeTaS | i have* |
| 15:26.00 | Corydon76-dig | bcrisp: what's wrong with Gosub? |
| 15:26.05 | kaldemar | gosub at least has return, so it can be used properly. |
| 15:26.16 | bcrisp | Corydon76-dig: nightmares of BASIC |
| 15:26.39 | Corydon76-dig | bcrisp: If you have a better name for it, I'm all ears |
| 15:27.15 | coppice | its the Stopsubs that really get you |
| 15:27.17 | Corydon76-dig | but "gosub" is a very universally recognized name for a command |
| 15:27.17 | bcrisp | Corydon76-dig: its not a biggy |
| 15:27.45 | Corydon76-dig | I try to choose names that need less explanation |
| 15:27.45 | Amorsen | One school of dial plan design is to have exten => _X!,1,Agi(something) as the complete dial plan... |
| 15:27.57 | bcrisp | Corydon76-dig: yes i think most programmers would understand that command |
| 15:28.58 | Corydon76-dig | Amorsen: One school of editors is to consume all available memory... |
| 15:29.07 | Amorsen | I've never actually used AGI's, but sometimes that idea seems quite seductive. |
| 15:29.23 | VaGoNeTaS | this little issue |
| 15:29.24 | VaGoNeTaS | http://www.pastebin.ca/1701913 |
| 15:29.40 | VaGoNeTaS | i need the 9 prefix |
| 15:29.51 | VaGoNeTaS | with the _X., goes through |
| 15:30.02 | VaGoNeTaS | but with _9Z., i got busy/congested |
| 15:30.03 | VaGoNeTaS | :S |
| 15:30.05 | VaGoNeTaS | very weird |
| 15:30.17 | Amorsen | Which number are you dialing? |
| 15:30.21 | VaGoNeTaS | 93658567 |
| 15:30.32 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:30.47 | *** join/#asterisk wolfeyes (i=2911de9e@gateway/web/freenode/x-arcxbrcrxvvxxuje) |
| 15:30.51 | Amorsen | Which context is having the problem? |
| 15:30.51 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:30.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:30.52 | VaGoNeTaS | the original number is 3658567 |
| 15:30.56 | VaGoNeTaS | Pruebas |
| 15:31.00 | Naikrovek | dialing 9 is such an old idea. why do people still set this up |
| 15:31.07 | VaGoNeTaS | is not my idea |
| 15:31.13 | VaGoNeTaS | is my boss idea |
| 15:31.19 | VaGoNeTaS | already told him |
| 15:31.29 | Naikrovek | 9 was in place when i started here but i added support to dial without the 9 and now everyone is happy |
| 15:31.31 | VaGoNeTaS | but he still wants the damn 9 before |
| 15:31.32 | ManxPower-work | Naikrovek: because it's a good idea |
| 15:31.51 | Gido-E | why is it a good idea? |
| 15:31.53 | Katty | shivers |
| 15:31.55 | Amorsen | VaGoNeTaS: Do you remember to remove the 9 when doing the actual dial? |
| 15:31.57 | Naikrovek | dialing 9 to get some imaginary outside line is a good idea? |
| 15:31.59 | kaldemar | deja vu |
| 15:32.10 | VaGoNeTaS | Amorsen : the 2nd plan |
| 15:32.16 | VaGoNeTaS | has _9Z |
| 15:32.18 | bcrisp | i dont like the 9... we get accidental 911s a lot |
| 15:32.27 | VaGoNeTaS | so 9 plus the 3658567 |
| 15:32.28 | bcrisp | (and ya, the cops show up) |
| 15:33.00 | Amorsen | VaGoNeTaS: _9Z and _X leaves the same in ${EXTEN} |
| 15:33.23 | VaGoNeTaS | so what should i have on my dialplan in order to work with the 9 |
| 15:33.30 | VaGoNeTaS | please update my pastebin |
| 15:33.37 | Naikrovek | 9|NXXXXXX |
| 15:33.49 | VaGoNeTaS | Naikrovek |
| 15:33.49 | [TK]D-Fender | ^ huh? |
| 15:33.51 | Amorsen | You probably want ${EXTEN:1} in the Dial DAHDI thing |
| 15:34.03 | VaGoNeTaS | take a look to my pb |
| 15:34.04 | VaGoNeTaS | http://www.pastebin.ca/1701913 |
| 15:34.12 | [TK]D-Fender | VaGoNeTaS: Indeed YOU are leaving what you consider to be a "prex" int he number you actually dial. |
| 15:34.14 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:34.21 | [TK]D-Fender | prefix* |
| 15:34.29 | [TK]D-Fender | [10:33]<Amorsen>You probably want ${EXTEN:1} in the Dial DAHDI thing <---------- |
| 15:34.31 | VaGoNeTaS | my boss wants that 9 prefix |
| 15:34.37 | Naikrovek | we get that |
| 15:34.40 | Gido-E | VaGoNeTaS explain him click2dial. |
| 15:34.47 | Gido-E | maybe he will change his thoughts |
| 15:34.53 | bcrisp | for basic call center functionality, should I be looking into using agents? |
| 15:34.55 | [TK]D-Fender | VaGoNeTaS: I'm pretty sure you want to StRIP the 9 off of the exten to get the # you want to actually call out to |
| 15:35.00 | Amorsen | VaGoNeTaS: Your boss wants it, but your service provider doesn't. So hide it from your service provider with :1 |
| 15:35.19 | [TK]D-Fender | VaGoNeTaS: DAHDI doesn't want to see the 9 in front. REMOVE IT FIRST |
| 15:35.56 | Amorsen | bcrisp: Depends whether you prefer the "agents" to have phones ringing or be constantly connected |
| 15:36.04 | *** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl) |
| 15:36.12 | Amorsen | Either has its advantages |
| 15:36.27 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 15:36.45 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 15:36.55 | VaGoNeTaS | it worked |
| 15:36.56 | VaGoNeTaS | with the :1 |
| 15:37.08 | bcrisp | Amorsen: i see, right now I have a very basic setup using SIP + softphones only .. I want to play with queuing calls that are made to extension X |
| 15:37.16 | VaGoNeTaS | Amorsen : it worked |
| 15:37.21 | VaGoNeTaS | thank you for the advice |
| 15:37.22 | Amorsen | Glad to hear it |
| 15:37.37 | VaGoNeTaS | xD |
| 15:37.52 | bcrisp | Amorsen: I don't mind having the system ring the person |
| 15:37.59 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 15:38.16 | Amorsen | bcrisp: Then stick with putting the SIP devices in the queue directly |
| 15:38.47 | bcrisp | Amorsen: ok, so i would just look at using the Queue app int he dialplan? |
| 15:38.55 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 15:39.22 | Amorsen | bcrisp: That is true whether or not you used Agents |
| 15:39.46 | Amorsen | The only difference is whether you add Agent/whatever or SIP/whatever into queues.conf (or AddQueueMember) |
| 15:40.06 | bcrisp | Amorsen: ok wonderful, i think that was my main question - whether i list them in queue.conf, thanks I'll check it out |
| 15:40.18 | [TK]D-Fender | [10:39]<Amorsen>The only difference is whether you add Agent/whatever or SIP/whatever into queues.conf (or AddQueueMember) <- or whatever else.... |
| 15:40.51 | [TK]D-Fender | Since you are looking to queue "extensions" this is a fixed call list. Says "static members" to me. |
| 15:41.06 | *** join/#asterisk bleger (n=yvannepr@topp148-office-nyc.openplans.org) |
| 15:41.12 | bcrisp | ok |
| 15:41.24 | bcrisp | putty keeps kicking me out grr |
| 15:42.18 | [TK]D-Fender | bcrisp: keepalive <- |
| 15:42.24 | bcrisp | ty |
| 15:44.21 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:45.46 | bcrisp | fender, im familiar with using static members in code :) but I'm not sure of the asterisk translation |
| 15:46.36 | bcrisp | right now i just have a handful of softphone clients, set up in extensions.conf with extension-like uniuqe IDs |
| 15:46.45 | bcrisp | or sip.inf |
| 15:46.57 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 15:47.43 | bcrisp | to testing queueing, im building a basic menu with a few options, one of which sends the call on to a queue of which these softphone clients should belong |
| 15:48.11 | bcrisp | wow A+ on my grammar |
| 15:49.12 | bcrisp | ill research queues.conf |
| 15:50.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:53.09 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:53.43 | Zeeek | In one hour on VoIP Users Conference we're talking about how VoIP is used on social networks. Welcome to join us, call in to the conference: http://vuc.me and join #vuc here on IRC |
| 15:53.47 | bcrisp | For English press one, for Broken English press 2 |
| 15:54.23 | bcrisp | vuc.me ... |
| 15:54.40 | Zeeek | well, there was no .you TLD :) |
| 15:54.54 | *** join/#asterisk bobg (n=bobg@ool-4576d9c2.dyn.optonline.net) |
| 15:55.02 | bcrisp | hehe |
| 15:56.52 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 15:58.44 | *** join/#asterisk tzafrir (n=tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 16:05.39 | *** join/#asterisk gavimobile (n=user@bzq-84-108-106-34.cablep.bezeqint.net) |
| 16:07.30 | gavimobile | do I need any hardware to create a group voice mailbox answering system with asterisknow ? |
| 16:07.30 | *** join/#asterisk saftsack (n=oliver@p4FF0CB6B.dip.t-dialin.net) |
| 16:07.58 | Gido-E | gavimobile no |
| 16:08.24 | Gido-E | ok, with asterisk. no clue about asterisknow |
| 16:08.46 | Gido-E | isn't there #asterisknow? |
| 16:09.00 | gavimobile | Gido-E: can I use my same phone provider or must I get voip |
| 16:09.02 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 16:09.47 | bcrisp | for monitor formats, would you guys suggest gsm or wav? |
| 16:09.58 | Zeeek | where is TK-DFender ??? |
| 16:10.44 | Gido-E | * [TK]D-Fender * |
| 16:11.18 | Gido-E | gavimobile how are you connected? |
| 16:11.22 | [TK]D-Fender | looks around |
| 16:11.25 | [TK]D-Fender | Dunno.... |
| 16:11.27 | gavimobile | what I really am tring to get at over here Gido-E is if I can use asterisk with a 56k modem |
| 16:11.38 | kaldemar | gavimobile: no |
| 16:11.39 | Zeeek | you're too quiet |
| 16:11.40 | Gido-E | gavimobile i don't think so. |
| 16:11.40 | bcrisp | they still make those? |
| 16:11.40 | [TK]D-Fender | gavimobile: No, your modem is not usable |
| 16:11.50 | Zeeek | what is a dialplan? |
| 16:11.55 | Gido-E | you need an fxo card. |
| 16:12.01 | [TK]D-Fender | gavimobile: So if yuo want to use your physical phone line you're going to need a hardware interface |
| 16:12.02 | Zeeek | and how does asterisk work? |
| 16:12.20 | [TK]D-Fender | Zeeek: It works HARD! |
| 16:12.55 | bcrisp | allowing callers to join an empty queue is evil is it not, fender? |
| 16:12.56 | Gido-E | Zeeek there is a free pdf book from Oreilly, available. |
| 16:12.57 | Zeeek | me too. And in 30 minutes we're warming up #vuc and you all can come over there for our weekly conference - about presence |
| 16:13.20 | [TK]D-Fender | bcrisp: Not necessarily |
| 16:13.22 | Zeeek | Yes I've hear of this book. It's called The Swiss Army or something |
| 16:13.25 | gavimobile | fxo card... im gona research that right now.. I knew I needed some type of hardware.... anyways the way im connected is with a rj11 to a modem. this modem has a rj45 port which is not in use. and the modem is coaxil or fieroptic to the provider. |
| 16:13.42 | bcrisp | "Your estimated wait time is now infinity" :) |
| 16:13.46 | Gido-E | Asterisk, The future of telephony |
| 16:14.11 | bcrisp | ~infinity |
| 16:14.12 | infobot | extra, extra, read all about it, infinity is ibot: message for ibot: infinity |
| 16:14.14 | Zeeek | there is no future of telephony |
| 16:14.22 | [TK]D-Fender | Gido-E: .... Zeeek Has been using * for a LONG time... |
| 16:14.23 | Zeeek | ~future |
| 16:14.24 | infobot | it has been said that future is bright |
| 16:14.41 | Zeeek | stop spoiling everything :) |
| 16:14.44 | [TK]D-Fender | "warning : do not look into the laser with your remaining good eye" |
| 16:14.51 | Gido-E | [TK]D-Fender i only try to answer his questions. |
| 16:14.51 | Zeeek | ~Zeeek |
| 16:14.52 | infobot | you are probably someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:15.04 | [TK]D-Fender | Gido-E: It wasn't a real question. |
| 16:15.25 | Zeeek | Gido-E: thans for trying to be helpful, we are just kidding around |
| 16:15.32 | Gido-E | [TK]D-Fender for me it is already past 5 pm :-). |
| 16:15.38 | Gido-E | Zeeek :-), no problem. |
| 16:15.50 | Zeeek | being hlpful on IRC is a brand new concept, not everyone understands it |
| 16:16.23 | Zeeek | Here is it exactly 17:16, in 30 minutes the wideband g722 bridge goes up for #vuc |
| 16:16.41 | Zeeek | in 45 minutes the old rickety g711 bridge goes up |
| 16:16.53 | Katty | oh look |
| 16:16.55 | Katty | it's Zeeek |
| 16:17.02 | [TK]D-Fender | gavimobile: On the cheap side : Linksys SPA-3102 |
| 16:17.06 | Zeeek | we love you all and welcome anyone interested in VoIP to come and met guests and some Digium people |
| 16:17.10 | gavimobile | Gido-E: I searched google for fxo cards and this was one of the first results that came up.. how to use a pci fax modem as a fxo card http://voip.weblogsinc.com/2005/07/14/use-a-v92-modem-as-an-fxo-card-on-asterisk/ |
| 16:17.18 | Zeeek | ohhhhhhh {{{{{Katty}}}}} |
| 16:17.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.28 | Zeeek | we especially love Katty |
| 16:17.39 | Zeeek | and welcome her on our vuc |
| 16:17.53 | gavimobile | [TK]D-Fender: I have an spa 2100 will that do? |
| 16:17.59 | Katty | i should not be loved anymore than anyone else. |
| 16:18.02 | bcrisp | fender, rather than listing all members of the queue im setting up, is it possible to unite them under one name? |
| 16:18.23 | bcrisp | i.e. sip/someone, sip/someoneelse |
| 16:18.46 | ManxPower-work | gavimobile: Go right ahead and use the device, but nobody will be able to help you if you have problems |
| 16:18.48 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 16:18.58 | gavimobile | why not? |
| 16:19.23 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 16:19.28 | gavimobile | the 3102 isa more supported model? |
| 16:19.54 | [TK]D-Fender | gavimobile: No |
| 16:19.59 | Zeeek | so for now I bid you all, including ManxPower-work a wonderful day |
| 16:20.02 | gavimobile | ... |
| 16:20.05 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:20.06 | ManxPower-work | gavimobile: no. random cards that claim to be V92 modems working with Asterisk |
| 16:20.08 | [TK]D-Fender | gavimobile: The 2100 lets you plug PHONES in. You need to plug a LINE in. |
| 16:20.21 | Gido-E | gavimobile, i would choose something like tdm400 from digium. |
| 16:20.38 | ManxPower-work | I would pick something from Sangoma |
| 16:20.57 | gavimobile | Gido-E: and this will allow me to create a inbound call center |
| 16:20.59 | gavimobile | ? |
| 16:21.01 | Katty | orders ryan's xmas present :> |
| 16:21.13 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:21.21 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733516.dsl.bell.ca) |
| 16:22.13 | [TK]D-Fender | gavimobile: Hard to call a system with ONE LINE as a "call center" |
| 16:22.21 | [TK]D-Fender | gavimobile: I think you are overselling yourself. |
| 16:22.30 | gavimobile | these are some featurews im looking for. auto answring, voice mailboxs,voice on hold |
| 16:22.45 | TheDavidFactor | is there a way to see group counts from the CLI? |
| 16:23.09 | gavimobile | [TK]D-Fender: possibly... I am not firmiliar with this technology |
| 16:24.13 | gavimobile | it makes no sense to me, because before I was told I dontneed any hardware. neither have I ever seen a company with a scomputer in their office dedicated to telephone |
| 16:25.17 | gavimobile | I do see however cabinets with a bunch of wires with 4 colorful wires in them |
| 16:25.37 | bcrisp | ill try that later.. |
| 16:25.43 | [TK]D-Fender | gavimobile: * can talk to various techs, including VoIP services. Indeed if you use an ITSP, you don't need extra hardware. |
| 16:25.47 | bcrisp | can i reload queue individually from CLI? |
| 16:26.17 | bcrisp | similar to reloading voicemail with voicemail reload |
| 16:26.18 | [TK]D-Fender | gavimobile: YOU on the otherhand said "I have a physical line I want to use". ONE LINE. Well if thats what you want to use, well guess what, you can't plug your line into a piece of SOFTWARE. |
| 16:26.35 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:27.19 | gavimobile | is this a bad idea to do it this way. if not please explain\ |
| 16:27.34 | *** join/#asterisk MAbbas (i=Jinbaba@115.186.30.38) |
| 16:27.53 | Gido-E | gavimobile for me it is a natural thing, to have a computer fixxing your phone shizzle. |
| 16:28.05 | [TK]D-Fender | gavimobile: It isn't a bad idea because the scope of what you expect is not clearly defined. |
| 16:28.20 | [TK]D-Fender | gavimobile: Is this line you are lready have and are already paying for ENOUGH? |
| 16:28.36 | [TK]D-Fender | gavimobile: We can't tell you what your NEEds are. We can only tell you what solutions can FIT them. |
| 16:28.38 | Gido-E | gavimobile, How is your landline connected PSTN, BRI or PRI? |
| 16:28.51 | Gido-E | i assumed PSTN |
| 16:29.24 | MAbbas | Hi everyone, Can I issue AMI commands in AGI script? |
| 16:29.26 | kaldemar | don't mix PSTN and POTS. :P |
| 16:29.42 | kaldemar | MAbbas: sure |
| 16:30.07 | MAbbas | How come? any hint? |
| 16:30.11 | gavimobile | Gido-E: I would have to call my provider and find out.. |
| 16:30.43 | [TK]D-Fender | MAbbas: .... Script = programming language. AMI = socket conenction to *. Go get PROGRAMMING already. |
| 16:31.02 | [TK]D-Fender | MAbbas: Go learn the language you are programming in. |
| 16:31.10 | gavimobile | [TK]D-Fender: I know my needs.. I want a simple automated answering phone to answer calls when im in and out of the office |
| 16:31.12 | ManxPower-work | MAbbas: exactly the same as if you are not using a CHI script. |
| 16:31.24 | [TK]D-Fender | MAbbas: The book also has code samples in various languages, as does the WIKI, and its highly Google-able. |
| 16:31.32 | Gido-E | gavimobile go do that. If you figured it out, you can find hardware that fits your need. |
| 16:32.06 | [TK]D-Fender | gavimobile: And what will calls come IN on? |
| 16:32.14 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:32.18 | gavimobile | Gido-E: ok ill do that |
| 16:32.24 | ManxPower-work | gavimobile: that is not your "needs" that is a "vague statement of your desires". |
| 16:32.58 | Katty | yay |
| 16:33.01 | ManxPower-work | your needs would be "IVR, inexpensive calling, 8 people with 1/2 of them being on the phone at the same time" |
| 16:33.07 | Katty | ryan's helicopter should be here in 1 week (= |
| 16:33.24 | Katty | it's surprising that hong kong can ship that to the usa so quickly |
| 16:33.31 | gavimobile | ManxPower-work: I think ivr |
| 16:33.32 | bcrisp | how do i refresh the queue.conf ? |
| 16:33.37 | bcrisp | queues.conf * |
| 16:33.43 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:34.05 | gavimobile | im the only employee.. and I need to service clients offices with pc repair |
| 16:34.12 | ManxPower-work | you THINK ivr? I think I'm dieing of cancer, but that means nothing until I'm sure. |
| 16:34.14 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:34.14 | gavimobile | I don't have anyone in the office when im out |
| 16:34.26 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 16:34.36 | [TK]D-Fender | gavimobile: So vfar I don't see how you need more than an ANSWERING MACHINE. |
| 16:34.51 | gavimobile | ManxPower-work: these are new terms to me however I was using a program called nch ivr which was almost taking care of what I need |
| 16:35.26 | gavimobile | [TK]D-Fender: they can leave a message if im not there |
| 16:35.31 | ManxPower-work | gavimobile: you should go read up on telephony then come back |
| 16:35.39 | gavimobile | however I would like the call possibly to forward to my cellphone |
| 16:35.42 | [TK]D-Fender | gavimobile: yeah... I think that's what ANsWERING MACHINES do... |
| 16:36.10 | [TK]D-Fender | gavimobile: You aren't describing the conditions for these scenarios you know.... |
| 16:36.27 | gavimobile | ManxPower-work: I will do that. but don't you think there is something more specific that I should be reading up? or telephony in general |
| 16:36.33 | Gido-E | gavimobile if you have only one incomming line, what to use to call you ? |
| 16:36.48 | bobg | I have a digium TDM400 connecting a wireless analog phone to * and voip service from axvoice. A couple times a day a call is interrupted by a steady high pitch, loud tone and we have to hang up. Any idea on how I can track down what is causing this? |
| 16:36.57 | gavimobile | [TK]D-Fender: please excuese the ignorance.. I need to start somewhere |
| 16:37.10 | bcrisp | got it |
| 16:37.13 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 16:37.21 | bcrisp | gavimobile, you could try doing what i did |
| 16:37.25 | bcrisp | im a first timer too |
| 16:37.28 | [TK]D-Fender | gavimobile: You can't describe the actions you want it to do in any detail. This isn't even a "how do I?" question |
| 16:37.55 | bcrisp | the oreilly book helps a lot |
| 16:37.56 | [TK]D-Fender | gavimobile: this is a "I want it to do X when the user does this, Y when they do that. I want it to do Z based on the time of day, etc" |
| 16:38.14 | [TK]D-Fender | gavimobile: this is the description of your NEEds, and has NOTHING to do with the "HOW" |
| 16:38.18 | gavimobile | well I was explainging hoping someone will tell me which hardware I could look at which might be right for me |
| 16:38.30 | gavimobile | and then I was gona read up about this hardware |
| 16:38.32 | bcrisp | lol the default moh is hillarious |
| 16:38.32 | [TK]D-Fender | gavimobile: Again, what are your calls coming IN on? |
| 16:39.15 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:39.15 | gavimobile | like what connection? |
| 16:39.28 | Gido-E | gavimobile, you are great :-) |
| 16:39.28 | [TK]D-Fender | gavimobile: Thats a start. |
| 16:39.43 | bcrisp | i didnt buy any hardware.. just leased a linux server from 1and1 |
| 16:40.02 | bcrisp | i cant recommend that tho.. i haven't gone far enough with it yet tho |
| 16:40.28 | gavimobile | right now I don't have anything but a pc phone and voice fax modem |
| 16:40.45 | gavimobile | that's whati was using till now, and now I want to advance |
| 16:40.51 | [TK]D-Fender | gavimobile: a modem by itself does do anything until you PLUG IT INTO SOMETHING. |
| 16:41.01 | [TK]D-Fender | gavimobile: Please grab a coffe and try to catch up. |
| 16:41.18 | [TK]D-Fender | gave precisely what will delive the phone calls to your room? |
| 16:41.32 | gavimobile | from phone modem to inut on modem and out to phone |
| 16:41.56 | jaytee | plugs in his soldering iron and prepares to cobble an FXO channel bank together out of old sound cards and a bunch of modems |
| 16:42.00 | [TK]D-Fender | gaveSo you have 1 phone line with an RJ-11 connection? |
| 16:42.09 | [TK]D-Fender | ~savemoney |
| 16:42.10 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 16:42.12 | [TK]D-Fender | jaytee: FTW! |
| 16:42.20 | gavimobile | that is almost correct |
| 16:42.33 | [TK]D-Fender | gavimobile: gavimobile Clarify the "almost" then |
| 16:42.37 | gavimobile | rj11 connection from a modem which has either fiber optic or coaxil |
| 16:42.47 | gavimobile | to provider |
| 16:43.11 | [TK]D-Fender | gavimobile: Same shit. The plug YOU use on the back is RJ11, correct? |
| 16:43.22 | gavimobile | yup |
| 16:43.27 | gavimobile | correct |
| 16:43.47 | [TK]D-Fender | gavimobile: Is this SINGLE line enough for your needs? If a call comes in on it I don't see how you can selectively FORWARD a call out to your cell. |
| 16:43.54 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 16:44.19 | [TK]D-Fender | gavimobile: Call comes in, but you can't bridge it to the outside all by itself so far |
| 16:44.32 | *** join/#asterisk ryduh (n=ryduh@204-16-143-186.utilitytelephone.net) |
| 16:44.43 | gavimobile | [TK]D-Fender: but if I get another line than I can do this? |
| 16:45.13 | [TK]D-Fender | gavimobile: Call comes in on 1 line, gets processed by *, you choose to call out the other line and bridge them. |
| 16:45.54 | gavimobile | until now no hardware is needed. correct? |
| 16:46.06 | gavimobile | just another line |
| 16:46.10 | gavimobile | <PROTECTED> |
| 16:46.13 | ManxPower-work | .part #telephony101 |
| 16:46.29 | gavimobile | conofigured correctly of course |
| 16:46.51 | [TK]D-Fender | gavimobile: Again, you need an INTERFACE to plug those LINES into. * is SOFTWARE. How do you shove an RJ11 plug into SOFTWARE? |
| 16:47.10 | gavimobile | you don't |
| 16:47.13 | *** join/#asterisk luminblade (n=luminfor@cpe-24-28-78-39.austin.res.rr.com) |
| 16:47.16 | [TK]D-Fender | gavimobile: And to repeat from earlier, your modem is worthless. |
| 16:47.24 | [TK]D-Fender | gaveSo yes you need HARDWARE |
| 16:47.30 | gavimobile | right.. |
| 16:48.33 | gavimobile | is this hardware which we plug the lines into the same hardware which will allow me to do the other tasks I wish to do |
| 16:48.52 | Gido-E | gavimobile but you could look for sip accounts. |
| 16:48.56 | [TK]D-Fender | gavimobile: the hardware is to let * do what you want it to do. |
| 16:48.59 | luminblade | is there any way to determine/record post-dial-delay (pdd) in Asterisk 1.4 or 1.6? |
| 16:49.15 | gavimobile | [TK]D-Fender: now its coming together |
| 16:49.21 | gavimobile | thans for your paitence |
| 16:49.48 | gavimobile | so the next step after I get another line would be to find the right hardware.. |
| 16:50.15 | fcois93 | how to ask some digits and save its in a var ? |
| 16:50.34 | gavimobile | can you offer a tip [TK]D-Fender |
| 16:50.50 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 16:51.01 | *** part/#asterisk ryduh (n=ryduh@204-16-143-186.utilitytelephone.net) |
| 16:51.35 | [TK]D-Fender | gavimobile: So far you don't have 2 physical lines. You should look at the cost of adding another. And the cost of said hardware. You should take a look at various suitable itsP's to see if that is a better soluion for you. |
| 16:51.50 | [TK]D-Fender | gavimobile: Tah means looking at the grade of internet connection you have as well |
| 16:51.55 | [TK]D-Fender | that* |
| 16:52.03 | [TK]D-Fender | ~itsp |
| 16:52.04 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:53.43 | gavimobile | [TK]D-Fender: I believe in my country there are 2 telephone providers excluding voip |
| 16:54.14 | gavimobile | and I wouldn't feel very comfortable using voip as a main way to receive phone calls |
| 16:54.20 | [TK]D-Fender | gavimobile: I'm telling you to look at the VoIP possibilities and see how that compares to the physical line based solution. |
| 16:55.15 | gavimobile | [TK]D-Fender: im sure the voip will be cheaper, but I also do not with the lack of quality with voip |
| 16:56.43 | *** join/#asterisk silverglade00 (n=mdkelly@anubis.ddns.ualr.edu) |
| 16:57.06 | Naikrovek | lack of quality? |
| 16:57.14 | Naikrovek | jumps in the middle of a conversation again |
| 16:57.30 | Naikrovek | also, i think we had this conversation the other day |
| 16:57.45 | Naikrovek | voip can do as good as or better than analog phones |
| 16:57.50 | [TK]D-Fender | Naikrovek: Don't forget your galoshes |
| 16:58.04 | Naikrovek | forgot his galoshes. |
| 16:58.06 | gavimobile | I've tried companies like cordiaip and vonage, and the service is terrible |
| 16:58.12 | Naikrovek | dammit, conversation all over my pants |
| 16:58.47 | Naikrovek | you've either been unlucky or have the most precise ears in the world, my vonage works and sounds amazing |
| 16:58.51 | gavimobile | since then I have never really liked using voip |
| 16:59.35 | Naikrovek | well in most nations, the core telephone network is voip anyway |
| 16:59.41 | Naikrovek | so you're using it whether you think you are or not |
| 16:59.47 | Naikrovek | it's the last mile that's analog these days |
| 16:59.50 | Naikrovek | the rest is voip |
| 17:00.05 | [TK]D-Fender | Last building my company was in had corroding copper lines... nasty static and the drain on them caused telco switch to not provide tone at times, etc... |
| 17:00.18 | [TK]D-Fender | Naikrovek: Or straight digital tDM |
| 17:00.27 | Naikrovek | those problems don't exist in voip |
| 17:00.36 | Naikrovek | yes, perhaps i should rephrase and say "digital" |
| 17:00.38 | Naikrovek | rather than voip |
| 17:00.56 | Gido-E | voip is good. |
| 17:01.03 | Naikrovek | very |
| 17:01.26 | Naikrovek | in the US anyway |
| 17:01.37 | [TK]D-Fender | VoCP iz teh shizn1t y0 |
| 17:01.38 | Naikrovek | gavimobile isn't in the US i don't think so it may be different there |
| 17:01.41 | luminblade | is there any way to determine/record post-dial-delay (pdd) in Asterisk 1.4 or 1.6? |
| 17:01.43 | Gido-E | in the netherlands, there are more people connected with voip than analoge |
| 17:01.44 | gavimobile | well lets say the power goes out |
| 17:02.01 | gavimobile | Naikrovek: I was gona mention that |
| 17:02.23 | luminblade | the power goes out, and everyone with corded phones will still be able to use them... which is how many people these days? |
| 17:02.23 | Naikrovek | gavimobile: you have the server, core network, and PoE switch on a UPS. PoE powers the phones |
| 17:02.23 | Gido-E | gavimobile you are from? |
| 17:02.33 | gavimobile | usa but I am in israel |
| 17:02.59 | Gido-E | ok, in a warzone :-) |
| 17:03.04 | gavimobile | lol |
| 17:03.19 | gavimobile | miami was worse |
| 17:03.51 | gavimobile | murderers and theft everyday |
| 17:04.09 | Gido-E | ok. Whe don't have gun's here. It is verry illegal to have them :-) |
| 17:04.27 | gavimobile | smart way of living |
| 17:04.34 | Naikrovek | i was in israel once. a more racist place i can't imagine |
| 17:04.37 | bcrisp | i woke up, went to the garage and found a .44 slug lodged in the door frame of my car |
| 17:04.42 | Naikrovek | they don't like black people in israel AT ALL |
| 17:04.50 | Naikrovek | at least not the folks i've met |
| 17:05.31 | gavimobile | well im american :-D |
| 17:05.33 | bcrisp | ok how do i find where the default moh is defined? |
| 17:05.36 | gavimobile | born and raised |
| 17:05.46 | Naikrovek | i wasn't saying you're racist |
| 17:05.49 | gavimobile | let me repronounce that |
| 17:05.54 | gavimobile | ameri-can |
| 17:05.54 | Naikrovek | just that it was really weird |
| 17:06.12 | gavimobile | Naikrovek: there's racism everywhere |
| 17:06.16 | Naikrovek | yes |
| 17:06.19 | Gido-E | bcrisp see musiconhold.conf |
| 17:06.22 | gavimobile | it sucks |
| 17:06.24 | Naikrovek | but not like that, not that i've seen |
| 17:06.26 | Naikrovek | yes it does |
| 17:06.31 | bcrisp | ty gido |
| 17:06.45 | Naikrovek | it makes no sense, which seems to be the requirement for many habits to endure |
| 17:06.58 | *** part/#asterisk ManxPower-work (n=EWieling@84.sub-70-223-191.myvzw.com) |
| 17:07.06 | Naikrovek | if it makes sense, it's cast aside, if it makes no sense, it's tradition by god and i'm keeping it |
| 17:07.43 | gavimobile | anyways.. so [TK]D-Fender so assuming I want to stay with the same itsp which hardware would I might want to consider |
| 17:09.30 | bcrisp | Gido-E: looked in musiconhold.conf - it points out the default directory, but is there a file somewhere that physcially states what file name is used for what? |
| 17:10.46 | bcrisp | oh |
| 17:10.51 | bcrisp | it just randomly picks them from the directory? |
| 17:10.56 | Gido-E | bcrisp yep |
| 17:11.06 | Gido-E | in cli type: moh show files |
| 17:11.06 | *** join/#asterisk Tim_Toady (n=moi@188.4.44.157.dsl.dyn.forthnet.gr) |
| 17:11.12 | *** part/#asterisk viraptor (n=viraptor@212.11.80.34) |
| 17:11.24 | bcrisp | Gido-E: is the default by sort order? |
| 17:12.30 | Gido-E | i think random. |
| 17:12.41 | Gido-E | but, also checkout: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf |
| 17:12.47 | bcrisp | it always plays the same thing lol |
| 17:13.23 | Gido-E | bcrisp ad random = yes to the context |
| 17:14.25 | gavimobile | [TK]D-Fender: I would really apreciate you hardware suggestion and why |
| 17:14.51 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:16.16 | bcrisp | ok renaming the annoying song makes everything silent |
| 17:16.23 | bcrisp | so im guessing its set somewhere to call it out by name |
| 17:16.32 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 17:16.40 | eppigy | yes |
| 17:17.19 | bcrisp | ooh im getting really choppy playback of the music |
| 17:17.21 | Gido-E | bcrisp that can!, check your dialplan. |
| 17:17.41 | bcrisp | Gido-E it seems to be working now.. just took a while for the new music to start playin |
| 17:17.51 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 17:18.54 | CGMChris | Anyone familiar enough with Wireshark to know if -1800ms clock drift is normal for a 50 second SIP call? |
| 17:22.43 | sysreq | hi! we just upgraded our servers from 1.4.26.3 to 1.4.27.1, and since then we seem to have a problem with calls being made/received by customers using x-lite.. we've had a couple customers use zoiper instead, and the issue doesn't seem to arise.. a packet capture shows a BYE packet sent by the softphone, although mid-call and not requested by the user (by hanging up).. |
| 17:23.02 | sysreq | i was wondering if anyone noticed the same problem |
| 17:23.20 | bcrisp | nat issue? |
| 17:24.12 | sysreq | most of our customers are behind nat, and we've had customers who had problems with x-lite use zoiper instead, and it 'fixed' the problem |
| 17:24.46 | bcrisp | xlite itself is nat.. i just remember having to add nat=yes in config |
| 17:25.01 | bcrisp | but im a total newb, ill let the gurus answer |
| 17:25.51 | bcrisp | im getting " Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available" warnings for various mappings |
| 17:26.04 | bcrisp | the mysql service is running so im not sure what the dealio is |
| 17:27.00 | kaldemar | bcrisp: you don't have res_config_mysql installed or loaded. |
| 17:27.09 | kaldemar | it's in asterisk-addons |
| 17:27.28 | bcrisp | hmm im pretty sure i went through all that setup |
| 17:27.33 | bcrisp | but ill look again thanks |
| 17:29.31 | [TK]D-Fender | gavimobile: With an ITSP you don't NEED hardware |
| 17:30.08 | [TK]D-Fender | gavimobile: ITSP = VoIP. * Speaks multiple VoIP protocols. Yuo have an internet connection. Packets flow. End of story |
| 17:32.10 | *** join/#asterisk makafre (n=makafre@64.86.141.133) |
| 17:33.23 | makafre | hi guys, Im looking for monthly/unlimited plans to the US, any good recommendations? |
| 17:33.49 | bcrisp | im declaring newb immunity before asking this question... |
| 17:34.24 | bcrisp | im sure i followed the instructions when making / install of asterisk addons |
| 17:34.38 | bcrisp | can't find the damn thing tho |
| 17:35.15 | bcrisp | in the filesystem |
| 17:35.30 | Gido-E | bcrisp /usr/src normally? |
| 17:36.09 | bcrisp | thanks Gido-E you're right |
| 17:36.12 | bcrisp | appreciate it |
| 17:44.07 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 17:45.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:47.07 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 17:52.17 | ecrane | Ok, SIP nomenclature question; I have seen message numbers like 100,200,404, etc. What would you call these? SIP "message ids"? SIP "codes"? |
| 17:53.31 | Tim_Toady | responce codes |
| 17:53.35 | ChannelZ | response codes |
| 17:53.41 | ChannelZ | jinx |
| 17:54.03 | ecrane | Thanks! |
| 17:54.50 | ChannelZ | in features.conf, pickupexten -- are you supposed to dial the pickupexten and then the extension number or something, or shouldn't it just pick up a ringing phone in the same pickup group when you dial the feature code? |
| 17:55.36 | CGMChris | Take a SIP from your DID and pass it to me with RTP and UDP. |
| 17:56.36 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:58.57 | *** join/#asterisk stevetotaro (n=Steve@c-68-49-104-232.hsd1.md.comcast.net) |
| 17:59.33 | *** part/#asterisk gavimobile (n=user@bzq-84-108-106-34.cablep.bezeqint.net) |
| 18:00.21 | ChannelZ | ah callgroups have to match. |
| 18:01.39 | ruben23 | hi can i firewall my asterisk server havng public ip directly connected to WAn..? |
| 18:01.57 | [TK]D-Fender | ruben23: "man iptables" |
| 18:01.57 | ian6 | sure, but thanks for asking permission first. |
| 18:02.47 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 18:03.52 | ruben23 | [TK]D-Fender: ok, do it need to be on a separate hardware or just built in on the asterisk server itself..? |
| 18:04.15 | makafre | ruben23: I can send you an example by email |
| 18:04.39 | ian6 | ruben23: you can do either. |
| 18:04.46 | [TK]D-Fender | ruben23: Every PC can have its own firewall, and you can do wahtever you want |
| 18:05.12 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 18:05.42 | ruben23 | makafre:thanks please, 3trglobal.it@gmail.com |
| 18:07.07 | *** join/#asterisk TimToady_ (n=moi@79.103.4.32.dsl.dyn.forthnet.gr) |
| 18:07.31 | makafre | ruben23: sent ! |
| 18:08.00 | ruben23 | makafre:thanks a lot, you used this on your asterisk also.? |
| 18:08.20 | makafre | yes, on asterisk |
| 18:08.39 | makafre | CentOS |
| 18:08.49 | ruben23 | ok thanks again |
| 18:08.53 | makafre | np |
| 18:15.05 | luminblade | anyone know if it's possible to determine PDD in aterisk 1.4 or 1.6? i saw a patch online for version 1.2, somehow that doesn't inspire confidence. |
| 18:17.19 | Kobaz | so i'm having an audio problem... http://www.kobaz.net/misc/fscked.wav |
| 18:18.01 | Kobaz | it has the sound like the cpu is overloaded and is slacking on transcoding |
| 18:18.11 | hardwire | Kobaz: neat.. is there an audio fscker? :) |
| 18:18.13 | Kobaz | *but*... the entire stream is ulaw, endpoint to endpoint |
| 18:18.15 | hardwire | fsck.sox? |
| 18:18.18 | hardwire | :P |
| 18:18.23 | Kobaz | heh |
| 18:18.36 | Kobaz | now, this problem only happens asterisk -> asterisk |
| 18:18.57 | Kobaz | if i load up a soft phone directly to asterisk box B, using the same codec, it sounds fine |
| 18:19.20 | Kobaz | but, IP phone -> asterisk A -> asterisk B... sounds like that wav file |
| 18:19.54 | Kobaz | so something is wrong with asterisk A, but i can't figure out what |
| 18:21.06 | ian6 | I didn't listen to it, but maybe the link between A and B is congested, has bad jitter/latency, etc. |
| 18:21.56 | Kobaz | nope |
| 18:22.17 | Kobaz | ip phone,asterisk A, is on network A... and asterisk B is on network B, across the interwebs |
| 18:22.28 | Kobaz | if i load up a soft phone on network a, and connect to asterisk B... it sounds perfect |
| 18:22.30 | hardwire | sip? |
| 18:22.34 | Kobaz | iax |
| 18:22.41 | hardwire | change it to sip and see what happens |
| 18:22.44 | Kobaz | heh, yeah |
| 18:22.49 | hardwire | you're comparing apples to oranges a bit |
| 18:22.49 | Kobaz | doing that now |
| 18:22.55 | Kobaz | hardwire: not really |
| 18:23.13 | hardwire | Kobaz: if you want to do a proper diagnostic, you are. |
| 18:23.41 | Kobaz | iax soft phone -> asterisk b (works fine). phone -> iax ip gateway -> asterisk A -> astrisk b (sounds shitty) |
| 18:23.57 | Kobaz | phone -> iax ip gateway -> asterisk b (sounds fine) |
| 18:24.09 | Kobaz | take asterisk A out of the picture... you have the source of the problem |
| 18:25.44 | Kobaz | replace one piece at a time, keeping the same setup... it's apples and apples |
| 18:25.44 | hardwire | oh.. iax soft phone. :) |
| 18:25.44 | eppigy | NEVER GONNA DANCE AGAIN |
| 18:25.48 | hardwire | Kobaz: I thought it was sip.. sorry |
| 18:27.11 | Katty | HELLO THAR |
| 18:27.16 | Katty | eppigy: i took a picture today. |
| 18:27.17 | eppigy | hy |
| 18:27.21 | Katty | eppigy: i saw it, and thought of you |
| 18:27.21 | eppigy | oh nice |
| 18:27.23 | eppigy | link plz |
| 18:27.24 | Katty | uploads. |
| 18:27.26 | eppigy | :> |
| 18:27.32 | Kobaz | soooo |
| 18:27.34 | Kobaz | the weird thing |
| 18:27.42 | Kobaz | is i had this problem before, and i restarted asterisk, and it fixed it |
| 18:27.49 | Kobaz | but now, that's not fixing it |
| 18:32.05 | Katty | imagebin sure is slow today |
| 18:32.11 | Katty | http://imagebin.org/74092 |
| 18:32.11 | eppigy | :< |
| 18:32.29 | eppigy | lol |
| 18:32.34 | eppigy | yes that is me 100% |
| 18:32.42 | Katty | i just know you so well |
| 18:32.47 | Kobaz | splena, ick |
| 18:32.55 | Kobaz | splenda |
| 18:33.00 | Katty | Kobaz: well, ick, but not as ick as some sweeteners. |
| 18:33.08 | Kobaz | splenda gives me headaches |
| 18:33.12 | Katty | nods |
| 18:33.17 | Katty | it can most certainly do that |
| 18:33.18 | eppigy | yesh you do |
| 18:33.21 | eppigy | LUNCH TIME |
| 18:33.32 | Katty | kbai |
| 18:33.49 | eppigy | squishugs Katty |
| 18:34.10 | Katty | dont' eat poorly! |
| 18:35.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:35.39 | eppigy | :< |
| 18:36.32 | Katty | so febreze supposedly makes an allergen reducer. i'm calling the vet to see if that might help riddick |
| 18:37.02 | Katty | eppigy: that doesn't mean it can't be tasty |
| 18:37.10 | Katty | eppigy: just don't eat poorly. |
| 18:37.17 | Katty | eppigy: NO WHITECASTLE |
| 18:37.21 | [TK]D-Fender | Katty: maybe your dog is allergic to ferrets ;) |
| 18:37.30 | Katty | [TK]D-Fender: actually, i think he might be. sadly. |
| 18:37.42 | Katty | [TK]D-Fender: he seemed to get worse when we started putting him in the basement while we're gone at work |
| 18:37.58 | Katty | [TK]D-Fender: there's nothing different about the basement than the rest of the house, as far as i know, other than the ferrets |
| 18:38.16 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 18:38.29 | Katty | it could be any number of things...an additive in his dog food, mylaundry detergent... |
| 18:39.16 | Katty | at least the antihistamine seems to be helping |
| 18:40.16 | Katty | hmm. vet recommends some sort of allergy reducer by Pledge |
| 18:40.29 | Kobaz | the fender |
| 18:40.32 | Kobaz | my savior |
| 18:40.34 | Kobaz | [TK]D-Fender: :) |
| 18:40.48 | Kobaz | [TK]D-Fender: http://www.kobaz.net/misc/fscked.wav |
| 18:41.07 | Katty | i just hate to spray things around. if he already has allergies then who knows what else he's allergic too |
| 18:41.17 | Katty | my luck he'd be allergic to the anti-allergen |
| 18:41.23 | *** join/#asterisk zippoman (i=48817f53@gateway/web/freenode/x-byvlganvfglnixjk) |
| 18:41.33 | Kobaz | [TK]D-Fender: that's from... ip phone -> asterisk a -> asterisk b (recording being done on asterisk a)... the audio is comming from asterisk b |
| 18:42.13 | Kobaz | it's not network... there's no jitter, packetloss, etc... i can hook up a phone directly to asterisk B, and everything sounds perfect |
| 18:42.26 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-djnlbfknycqdhyps) |
| 18:42.47 | Katty | Kobaz: just curious, but could you describe the audio problem you're having? |
| 18:42.53 | Katty | i assume it's audio related. |
| 18:43.03 | Kobaz | okay |
| 18:43.14 | Kobaz | ever set up asterisk on a really slow box... say 200mhz |
| 18:43.20 | Kobaz | and you're transcoding from ulaw to gsm |
| 18:43.24 | Kobaz | that's what it sounds like |
| 18:43.35 | Katty | i can't say i've ever tried to run asterisk on something that old. |
| 18:43.38 | Kobaz | really choppy, and consistantly choppy, since the cpu can't keep up |
| 18:43.46 | Katty | ouch )= |
| 18:43.48 | Kobaz | but this is a 2ghz box |
| 18:43.56 | Katty | oh. |
| 18:44.02 | Katty | ponders |
| 18:44.04 | Kobaz | so i shouldn't be having that problem... AND, there is no transcoding, since everything is ulaw |
| 18:44.12 | *** join/#asterisk Delido1983 (n=Delido19@dslb-084-063-216-190.pools.arcor-ip.net) |
| 18:44.35 | Katty | is it any audio? |
| 18:44.35 | Katty | phone calls, recordings, playback() |
| 18:44.46 | Kobaz | yes, any audio from asterisk a, to b |
| 18:45.01 | Kobaz | well, recieving audio from b, actually |
| 18:45.06 | Kobaz | make a call from a to b, and get audio from b |
| 18:45.29 | Katty | if you record a conversation from phone -> asterisk b, what's the audio sound like? |
| 18:46.06 | Kobaz | it's perfect |
| 18:46.17 | Katty | like just a Record(foo) Playback(foo) sort of thing |
| 18:46.17 | *** part/#asterisk Delido1983 (n=Delido19@dslb-084-063-216-190.pools.arcor-ip.net) |
| 18:46.20 | *** join/#asterisk steve745 (n=steve@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 18:46.30 | Katty | and if you go from b -> a -> phone? |
| 18:46.35 | Katty | what's that sound like |
| 18:46.59 | Kobaz | not sure. i can try |
| 18:47.13 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:47.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:49.36 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-155-79.dsl.stlsmo.sbcglobal.net) |
| 18:51.21 | LemensTS | How do you guys feel about doing tftp to configure phones if your an itsp |
| 18:51.48 | Katty | LemensTS: well. |
| 18:52.02 | Katty | LemensTS: with the right securities it might be okay |
| 18:52.16 | Katty | LemensTS: are your phones lan? wan? |
| 18:52.19 | Katty | LemensTS: vpn? |
| 18:52.20 | LemensTS | wan |
| 18:52.22 | LemensTS | no vpn |
| 18:52.25 | Katty | hmm |
| 18:52.29 | Katty | i wouldn't |
| 18:52.30 | *** join/#asterisk Alagar (n=Administ@122.164.32.160) |
| 18:52.36 | Katty | not unless you can lock them down to a static public ip |
| 18:52.43 | *** join/#asterisk gscmans (n=guna@cpc1-haye16-2-0-cust91.haye.cable.virginmedia.com) |
| 18:52.43 | Katty | or mac address |
| 18:52.48 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 18:52.53 | *** part/#asterisk luminblade (n=luminfor@cpe-24-28-78-39.austin.res.rr.com) |
| 18:53.25 | LemensTS | I got 40 sipura's with keep-alive enabled, and it needs turned off. So without physically logging in to clients computers then logging into the phone and fixing it, or sending them another ata and have them send the old one back im kinda screwed |
| 18:53.37 | Katty | nods |
| 18:53.56 | Katty | yeah i understand. |
| 18:54.06 | Katty | but i'd still be careful |
| 18:54.26 | Katty | i know with our polycom setup, the phone's cfg files have authentication information in them |
| 18:54.33 | LemensTS | I figured if i fix it i should put them on a boot server or something in case of future problems. I looked into this before and didn't do it because of security concerns |
| 18:54.34 | ChannelZ | you can't provision them? |
| 18:54.38 | Katty | and that's not something you want available to big wide world (= |
| 18:55.07 | Katty | LemensTS: do they have a firewall of some sort that would accept incoming connections from your static ip address? |
| 18:55.15 | LemensTS | ChannelZ: isnt that where they get the config from tftp? |
| 18:55.17 | Katty | LemensTS: and no where else |
| 18:55.40 | LemensTS | Katty: no just peoples houses on isp modems |
| 18:55.49 | ChannelZ | yea.. at least with the Linksys SPAs (which came from Sipura) you can send them configs that only change one or two things |
| 18:56.03 | Katty | ouch |
| 18:56.34 | Katty | i'm not too familiar with the tftp security options |
| 18:56.47 | Katty | for stuff like that i've always used just a cheap linksys firewall at the user's home |
| 18:56.57 | ChannelZ | but you either have to tell the person how to login to the phone and set up a provisioning URL, or send them a template URL they need to type in and replace with their phone's address to do it, etc. |
| 18:56.59 | Katty | and then setup a few rules to help prevent intruders and nosey people |
| 18:57.14 | Katty | as far as i know, ftp authentication isn't encrypted |
| 18:57.16 | ChannelZ | Which at that point it's probably just as easy to tell them how to turn off keepalive :) |
| 18:57.50 | Katty | so something sitting around with a packet sniffer like wireshark... |
| 18:58.09 | Katty | i've always locked down to static ips |
| 18:58.15 | LemensTS | ChannelZ: Ill read into that provisioning, is that considerably safer than tftp? |
| 18:58.31 | Kobaz | it's all plain text |
| 18:58.38 | Kobaz | ftp/tftp/http |
| 18:58.46 | Katty | nods |
| 18:58.49 | LemensTS | Katty: i cant do a firewall at each clients house, dont got enough man power |
| 18:59.14 | ChannelZ | tftp has no authentication, that's the point.. it's 'trivial' |
| 18:59.40 | Katty | i don't know that the information on the tftp server is really all that trivial tho |
| 18:59.55 | Katty | guess it depends on what's there, and what's being sent back and forth |
| 19:00.01 | ChannelZ | you don't need to put the auth info into the provisioned configs |
| 19:00.24 | Katty | if it's something trivial like, logo or directory or whatever...that's not such a big deal |
| 19:00.59 | Katty | but the last thing you want is someone who's gotten ahold of extension login information using your server and registering as a client |
| 19:01.36 | Katty | LemensTS: what about on your side, only allowing certain mac addresses to connect? |
| 19:01.55 | Katty | wouldn't have to really worry about static IPs that way |
| 19:02.24 | Katty | it might be a little time consuming, but at least unauthorized network devices couldn't be poking about in your stuff |
| 19:02.31 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 19:03.16 | LemensTS | Katty: is that doable filtering mac address's? I checked before and was told you couldn't |
| 19:04.07 | Katty | that's probably something at the firewall level |
| 19:04.10 | Katty | not the ftp server level |
| 19:04.44 | ChannelZ | No MAC on tcp/ip |
| 19:05.28 | Katty | ChannelZ: how do wireless access points do it then? |
| 19:05.31 | *** join/#asterisk ManxPower-work (n=EWieling@84.sub-70-223-191.myvzw.com) |
| 19:05.40 | ManxPower-work | ~answers |
| 19:05.41 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 19:05.54 | ChannelZ | ARP maps mac addresses to IP addressess.. which works on a local network |
| 19:06.47 | LemensTS | ChannelZ: yep thats what i was told |
| 19:06.48 | ChannelZ | (since ethernet adapters, etc. work on MAC addresses) |
| 19:06.57 | LemensTS | no mac on tcp |
| 19:07.17 | Katty | bummer. |
| 19:07.24 | ChannelZ | You can make the config that the phone fetches have the MAC in the filename, or come up with something completely random and unique for each client |
| 19:07.39 | ManxPower-work | The MAC of an ethernet packet would normally be the MAC of the last device that routeed the packet |
| 19:09.37 | LemensTS | http://www.voip-info.org/wiki/view/sipura+mass+deployment |
| 19:10.22 | ChannelZ | precisely |
| 19:10.32 | LemensTS | For fetching the file with the MAC in the file name, it still has to go thru tftp from what that link says, right? |
| 19:10.49 | LemensTS | Or is it just the pre-provisioning that uses TFTP |
| 19:10.50 | ChannelZ | yes |
| 19:12.22 | ChannelZ | locally I have a 'bootstrap' provision file just called spa.cfg - so if I setup a new phone, I just punch in the url and have it load spa.cfg. Inside spa.cfg it re-sets the provisioning URL to one that includes the MAC of the phone. So then the phone fetches it's own custom config. |
| 19:13.07 | LemensTS | ChannelZ: ok gotcha, so is that custom config for the phone on the TFTP server? If so cant anyone view/download it |
| 19:13.35 | ChannelZ | they can download it if they know the exact filename |
| 19:13.48 | ChannelZ | In my case it's all LAN though |
| 19:14.50 | ChannelZ | But you could make the filename the user's DID number combined with the MAC, so in order for someone to 'steal' the config (assuming you were wanting to put auth info into the config and as such wouldn't want just anyone to see it) they'd have to know the exact MAC and phone number in order to get anything interesting.. |
| 19:15.02 | ChannelZ | assuming they know something about your system in the first place to figure out how to construct the filename |
| 19:16.02 | mtltemplar | hey anyone on yet that knows anything about zaptel/digium cards? i would put the whole data point in here but I'd rather see if anyone's home first :) |
| 19:16.04 | LemensTS | ChannelZ: that is a good idea. Im gonna set this up tonite. Last question. What tftp server do you recommend |
| 19:16.09 | bcrisp | can anyone suggest an irc channel for mysql? :) |
| 19:16.18 | LemensTS | #mysql |
| 19:16.22 | ChannelZ | I'm just using the packaged one in ubuntu |
| 19:16.28 | mtltemplar | lol lemensts |
| 19:16.30 | [TK]D-Fender | mtltemplar: lOTS OF US KNOW LOTS OF THINGS |
| 19:16.39 | bcrisp | hehe |
| 19:16.52 | bcrisp | my asterisk mysql config is all funked up |
| 19:17.01 | mtltemplar | I bet you do. Just checking to see if anyone's home first. I asked yesterday and this morning and no responses so i wanted to see first. |
| 19:17.04 | bcrisp | im used to sql server / oracle |
| 19:17.11 | LemensTS | ChannelZ/Katty: Thanks for the help and advice |
| 19:17.24 | ChannelZ | It's really pretty stupid. You run it from inetd or somesuch and tell it what directory it's allowed to play in. It's all abstracted, not like FTP where you can 'cd' around the filesystem or even 'ls' |
| 19:18.06 | mtltemplar | so basically, i had a power outage and both of my asterisk machines went down hard and dirty. now their te405 cards are coming up with all 4 spans getting a yellow alarm. rebooting, etc, doesnt fix it |
| 19:18.06 | [TK]D-Fender | mtltemplar: And you've still not asked a specific question. |
| 19:18.23 | mtltemplar | well i did, but not recently. now i have. thanks :) |
| 19:18.23 | zippoman | is there any documentation on getting asterisk to grab numbers to dial from a mysql database? |
| 19:18.30 | ChannelZ | your system is screw |
| 19:18.42 | [TK]D-Fender | ChannelZ: He's gone BTW. |
| 19:18.57 | LemensTS | zippoman: mysql in asterisk-addons |
| 19:18.58 | ChannelZ | who? |
| 19:19.05 | [TK]D-Fender | zippoman: Grab what numbers from a DB? |
| 19:19.11 | [TK]D-Fender | ChannelZ: trupsalms |
| 19:19.11 | LemensTS | zippoman: look for mysql on voip-info |
| 19:19.13 | ChannelZ | zippoman: are you a telemarketing robodialer? |
| 19:19.30 | ChannelZ | [TK]D-Fender: gone like dead? |
| 19:19.33 | LemensTS | zippman: if its complicated shoot for php-agi |
| 19:19.48 | ManxPower-work | ChannelZ: We tossed the last telemarketer into the swamp for the alligators to have. |
| 19:19.52 | [TK]D-Fender | ChannelZ: Gone like "all accounts on FreeNode deleted" |
| 19:20.00 | Katty | i can't believe i'm sitting in ustream.tv support channel helping people. |
| 19:20.06 | Katty | i must really hate myself. |
| 19:20.12 | zippoman | ChannelZ, haha no |
| 19:20.20 | ChannelZ | [TK]D-Fender: Learned his lesson then! hehe |
| 19:20.28 | denon | Katty: how much help could people need on ustream? |
| 19:20.33 | denon | plug in camera, hit broadcast |
| 19:20.41 | [TK]D-Fender | mtltemplar: what versions of * & zaptel/DAHDI? |
| 19:20.41 | zippoman | [TK]D-Fender, phone numbers haha |
| 19:20.42 | denon | tweak throughput settings a bit |
| 19:20.43 | denon | bam |
| 19:21.17 | [TK]D-Fender | ChannelZ: Not necessarily, but the lesson was printed out, taped out a ClueBat (tm) and used to bludgeon him a bit... |
| 19:21.19 | zippoman | ChannelZ, its for my parents they are elderly and have trouble remembering numbers |
| 19:22.08 | [TK]D-Fender | zippoman: No, there is no documentation specifically for your application, but the dialplan apps that allow DB access are all documented jsut like every other dialplan app |
| 19:22.37 | zippoman | [TK]D-Fender, yes I know theres nothing specific I wasn't asking for anything specific |
| 19:22.47 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:22.53 | ChannelZ | so then the answer is just "yes" |
| 19:23.47 | LemensTS | zippoman: how do you 'see' this working? |
| 19:24.25 | [TK]D-Fender | LemensTS: With his "eyes" ;) |
| 19:24.31 | Katty | I FEEEEEL SO UNSURE |
| 19:24.45 | Deeewayne | O.O |
| 19:25.06 | Deeewayne | waves to Katty |
| 19:25.07 | zippoman | I input numbers into a database....they select from a drop down on the web who they want to call and hit submit and it calls them then calls the number listed in the database under that ID |
| 19:25.29 | LemensTS | zippoman: look for web dialer |
| 19:25.29 | mtltemplar | * is 1.4.21.2 and im not sure on zaptel. actually i am using zapata i suppose but im not sure im even using it properly. |
| 19:25.34 | ChannelZ | So anyone use SPA*s? |
| 19:25.44 | mtltemplar | im not the one who set the systems up so im scrounging (sp?) here |
| 19:26.09 | Katty | hi Deeewayne (= <3 |
| 19:26.25 | LemensTS | zippoman: think they go thru that in the asterisk agi book actually |
| 19:26.30 | LemensTS | phpagi |
| 19:27.03 | [TK]D-Fender | zippoman: So far I don't see where * is doing the lookup |
| 19:27.29 | [TK]D-Fender | LemensTS: And I fail to see where AGI comes into play here either... |
| 19:27.40 | [TK]D-Fender | ChannelZ: Yes, what of them? |
| 19:27.41 | LemensTS | tkd-fender: ami i mean |
| 19:27.45 | ChannelZ | zippoman: yah you can do all that with a web app that talks AMI to * to dial the number. * need not know about mysql at all |
| 19:28.05 | LemensTS | tkd-fender: not sure what i was thinking |
| 19:28.19 | ChannelZ | [TK]D-Fender: Is the general idea to just wipe out all of these 'vertical service codes' (make them blank) to disable them or is there some global switch I"m not seeing? |
| 19:28.50 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 19:29.00 | [TK]D-Fender | ChannelZ: just disable each and thats it... though there are a bunch I'd leave... |
| 19:29.12 | zippoman | ChannelZ, I tried something with asterisk manager api to see if I could just input the numbers without having them in a database it didn't work out |
| 19:29.23 | Katty | you know what i hate. having to pee. |
| 19:29.27 | Katty | it's so ANNOYING |
| 19:29.29 | Kobaz | heh |
| 19:29.49 | [TK]D-Fender | zippoman: And you haven't given details as to your approach and you didn't show us your erros so we could help fix them |
| 19:29.57 | [TK]D-Fender | zippoman: Also you could use "call files", or call Originate from * CLI, etc. |
| 19:30.05 | ChannelZ | zippoman: Well how you keep track of these numbers is up to you -- database or not, statically built into your web page or whatever |
| 19:30.23 | [TK]D-Fender | zippoman: either way the DB has nothing to do with * |
| 19:30.39 | Kobaz | katty: the perfect computer chair: https://my.estoresnw.com/kap/ProductImages/C-toilet-computer-C.jpg |
| 19:30.40 | zippoman | yeah its just a way of storing things long term |
| 19:31.28 | ChannelZ | thats fine.. but you can write all that outside of * |
| 19:32.18 | eppigy | Katty: best chicken salad sammich ever |
| 19:32.26 | zippoman | yeah I was just wondering how asterisk would grab them from the database I see res_obdb.conf |
| 19:32.38 | eppigy | Katty: AS I TAKE YOUR HAND |
| 19:32.43 | eppigy | AND LEAD YOU TO THE DANCE FLOOR |
| 19:32.45 | ChannelZ | you don't need to have * even touch the database! |
| 19:33.16 | ChannelZ | You write a bit of code through whatever this interface is and simply pass the number you're dialing to * as a static string (from it's point of view) |
| 19:33.33 | [TK]D-Fender | zippoman: Where is * ever called upon to look anything up? Why should it have to? You pick it off a page and the RESULT is available. |
| 19:33.44 | zippoman | yeah I see that |
| 19:34.20 | zippoman | Ok so how would you tell asterisk to call the first number then after the person has picked up to call the original number |
| 19:34.23 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 19:34.30 | Kobaz | zippoman: originate |
| 19:34.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:34.49 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 19:34.54 | LemensTS | zippoman: you should just get a free php web dialer, and modify it to work with a drop down box instead of the dialer buttons. |
| 19:35.13 | [TK]D-Fender | zippoman: AMI / CLI Originate, or a call-file |
| 19:35.19 | LemensTS | zippman: than all u gotta do is connect the ami |
| 19:35.52 | zippoman | I tried this $call = $asterisk->originateCall('s', "SIP/voicepulse-primary/$party1", 'hookup', $from, 1, 30000, $party2); |
| 19:35.59 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 19:36.26 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 19:36.31 | Kobaz | dialplan Originate() would be nice if you could backgroud it instead of waiting for an answer |
| 19:36.38 | LemensTS | pastebin the whole file |
| 19:36.42 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 19:36.43 | ChannelZ | that has no meaning to us, what is origiateCall() doing? |
| 19:36.46 | zippoman | yeah |
| 19:36.53 | [TK]D-Fender | pastebin the CLI output of the failed attempt. |
| 19:36.55 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 19:38.55 | zippoman | LemensTS, http://pastebin.com/m5f315dd |
| 19:38.58 | [TK]D-Fender | zippoman: What language? What library provides that method? |
| 19:39.23 | [TK]D-Fender | zippoman: http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI_AsteriskManager.html#methodOriginate |
| 19:39.35 | [TK]D-Fender | zippoman: I see Originate, not OriginateCall |
| 19:39.39 | Katty | Kobaz: that's just gross. |
| 19:40.01 | Katty | Kobaz: there's not even a fuzzy bath mat in there, come on now |
| 19:40.16 | [TK]D-Fender | [14:36]<Kobaz>dialplan Originate() would be nice if you could backgroud it instead of waiting for an answer <- it DOES... there is no waiting for an answer, it spawns the new call and moves on. |
| 19:40.19 | Katty | eppigy: oh? where did you go? mcalister's? they have pretty good chicken salad sammiches |
| 19:41.02 | eppigy | folks |
| 19:41.13 | eppigy | Katty: time can never mend |
| 19:41.19 | eppigy | the careless whispers of a good friend |
| 19:41.23 | Kobaz | [TK]D-Fender: hmm... when i played with it... it waited |
| 19:41.28 | Kobaz | i think that's in the docs too |
| 19:41.47 | zippoman | [TK]D-Fender, it works though it connects to $party1 then hungup before calling $party2 |
| 19:41.48 | Katty | eppigy: Folks Southern Kitchen Restaurant? |
| 19:41.55 | eppigy | yesh |
| 19:42.28 | ChannelZ | now I'm starving |
| 19:42.29 | Kobaz | [TK]D-Fender: This application will block until the outgoing |
| 19:42.30 | [TK]D-Fender | zippoman: where's your dialplan and CLI debug? I'd expect SIP debug as well |
| 19:42.30 | Kobaz | call fails or gets answered. |
| 19:42.41 | [TK]D-Fender | Kobaz: Retarded... |
| 19:42.45 | Kobaz | [TK]D-Fender: yes i know |
| 19:42.46 | [TK]D-Fender | Kobaz: Don't need it anyway |
| 19:42.48 | Katty | eppigy: http://www.youtube.com/watch?v=3354flS1KJs |
| 19:42.58 | zippoman | [TK]D-Fender, let me fetch it |
| 19:43.04 | Kobaz | [TK]D-Fender: yes, of course you can to TrySystem(asterisk -rx "originate...") |
| 19:43.09 | Kobaz | [TK]D-Fender: but why should you have to |
| 19:43.28 | [TK]D-Fender | Kobaz: You shouldn't... thats why I called it "retarded" |
| 19:43.30 | Kobaz | haha |
| 19:43.31 | Kobaz | yeah |
| 19:43.43 | *** join/#asterisk smooth_penguin (n=smoove@59.95.17.195) |
| 19:43.50 | eppigy | Katty: this is intense |
| 19:44.12 | Katty | not really. |
| 19:44.19 | eppigy | oh |
| 19:44.20 | Katty | the trailer for the prosecution of george w bush was intense tho |
| 19:44.50 | eppigy | lol |
| 19:45.33 | *** join/#asterisk syntax_erorr (n=syntax@cblmdm72-240-127-126.buckeyecom.net) |
| 19:45.40 | syntax_erorr | Hi all |
| 19:46.00 | Katty | eppigy: ryan ordered an Epipe the other night |
| 19:46.03 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 19:46.18 | syntax_erorr | I'm new to asterisk and was wondering if you can use this software with regular modems? Or does it need special hardware? |
| 19:46.25 | [TK]D-Fender | syntax_erorr: Latter |
| 19:47.04 | syntax_erorr | can you point me in the right direction as to what I should be looking for? Its PBX stuff I'm assumeing |
| 19:47.28 | Kobaz | syntax_erorr: fxo/fxs devices... sangoma.com |
| 19:47.39 | syntax_erorr | Thanks |
| 19:47.47 | Kobaz | syntax_erorr: and also audiocodes, linksys, make external gateways |
| 19:47.53 | [TK]D-Fender | syntax_erorr: Digium & Sangoma are the 2 largest manufacturers of internal cards for * and those likely to provide support. |
| 19:48.15 | Kobaz | there's also rhino... but don't buy anything from them |
| 19:48.16 | [TK]D-Fender | syntax_erorr: Mediatrix does gateways as well |
| 19:48.49 | bcrisp | do i need to manually create the database for asterisk's use? |
| 19:49.10 | Kobaz | rhino's firmware is hit or miss if it works correctly |
| 19:49.11 | [TK]D-Fender | bcrisp: What database? |
| 19:49.12 | Katty | eppigy: http://www.youtube.com/watch?v=vO3pGoedz1E <- review & photo -> http://www.farseegroup.com/UploadFiles/DSE601B-1.jpg |
| 19:49.20 | bcrisp | for queues etc |
| 19:49.43 | [TK]D-Fender | bcrisp: Yes, but there are some quick sample scripts you can adapt in the source tarball |
| 19:49.50 | syntax_erorr | thanks for the info guys |
| 19:49.52 | bcrisp | oh ok |
| 19:49.54 | bcrisp | thanks |
| 19:50.29 | Naikrovek | when did polycom's 335 phone come out? I don't think i've seen that before |
| 19:50.40 | Katty | eppigy: i don't know the model number of it, but ryan got an e-cig which is about the same size as a standard pen. he says it's the best vapor/flavor wise so far. |
| 19:50.51 | Katty | eppigy: he had a flavor called Java Jolt the other nigth...didn't smell too shabby |
| 19:51.12 | [TK]D-Fender | Naikrovek: About 2 months ago IIRC |
| 19:51.20 | Naikrovek | i'm going to start buying those |
| 19:51.33 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.138.161.dsl.dyn.forthnet.gr) |
| 19:51.34 | Naikrovek | dual port switch and g722 |
| 19:51.40 | Naikrovek | nice! |
| 19:52.20 | hardwire | does anybody have utstarcom iAN-02EX firmware hanging around? |
| 19:53.05 | [TK]D-Fender | Naikrovek: Backlit as well |
| 19:53.12 | [TK]D-Fender | Naikrovek: Big selling point |
| 19:58.14 | bcrisp | ya.. my office has motion sensor lights .. when my lights go off and the phone rings i cant tell who it is till i answer : / |
| 20:00.49 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 20:02.24 | syntax_erorr | sorry for the newb question, but how does this setup work? I'd buy an interface card which could be connected to a regular phone line then be able to output different lines through a PBX system? |
| 20:03.12 | [TK]D-Fender | syntax_erorr: * will use those lines to whatever you want. * can be used to MAKE a PBX. |
| 20:03.20 | [TK]D-Fender | syntax_erorr: Do you already ahve another PBX system? |
| 20:03.53 | syntax_erorr | We have a system that a local company set up for us and it is complete poo poo we want a system we manage our self |
| 20:04.10 | [TK]D-Fender | syntax_erorr: What kind of "system"? |
| 20:04.44 | eppigy | Katty: im quiting next week :D |
| 20:04.52 | *** join/#asterisk Optic (n=Optic@miso.capybara.org) |
| 20:05.01 | syntax_erorr | Its an automated attendant / voice mail system |
| 20:05.32 | bcrisp | ~future of telephony |
| 20:05.39 | bcrisp | ~future |
| 20:05.40 | infobot | future is, like, bright |
| 20:05.57 | ChannelZ | ~book |
| 20:05.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:06.11 | bcrisp | syntax_erorr - read that online pdf ebook |
| 20:06.15 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 20:06.21 | bcrisp | it has a great overview |
| 20:06.32 | [TK]D-Fender | syLets try this again. What MODEL of PBX? |
| 20:07.36 | syntax_erorr | ok great, thanks for the link |
| 20:08.05 | syntax_erorr | Its a toshiba system I don't know the model off the top of my head and its in a bulding across town |
| 20:08.16 | Optic | hmm, having some strangeness, have you seen this one before? |
| 20:08.16 | Optic | app_voicemail.c: Unable to stream password file |
| 20:08.18 | [TK]D-Fender | syntax_erorr: How many lines, how many phones? |
| 20:08.38 | syntax_erorr | we have 3 voice lines, and about 12 phones |
| 20:09.24 | [TK]D-Fender | syntax_erorr: Still want to use analog lines? |
| 20:10.39 | syntax_erorr | We were planing on keeping the analong lines |
| 20:10.49 | [TK]D-Fender | syntax_erorr: then http://www.telephonydepot.com/Catalog/Sangoma/B600D-Analog-Voice-Card |
| 20:11.03 | [TK]D-Fender | syntax_erorr: + Appropriate Polycom handsets |
| 20:12.11 | Naikrovek | backlit as well |
| 20:12.13 | Naikrovek | ooh |
| 20:12.19 | Naikrovek | likes. |
| 20:12.25 | syntax_erorr | I really appreciate the help fender, thanks |
| 20:13.22 | [TK]D-Fender | syntax_erorr: NP. This is a "rip & replace" scenario |
| 20:15.29 | syntax_erorr | Yeah the company that put our current phone system in, made a mess of our phone situation. To answer a call on a wireless handset you have to press 3 different buttons, we can't change our own greetings (we have to pay them to change it) |
| 20:15.44 | bcrisp | yikes |
| 20:16.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.45 | Optic | my * voicemail seems to be broken :( |
| 20:17.52 | bcrisp | each time i write my question it sounds worse than the last try |
| 20:19.08 | bcrisp | i have SIP working pretty good with softphones. If i want to then be able to handle communication with outside phones (pstn?) can i use some 3rd party service ? - i can't modify the hardware on the asterisk server |
| 20:19.51 | bcrisp | (and i re-invoke my newb immunity) |
| 20:20.20 | Naikrovek | bcrisp: yes |
| 20:20.26 | Optic | any of you have any idea what might be going on here? |
| 20:20.26 | Optic | http://pastebin.com/m724aa1dd |
| 20:20.31 | [TK]D-Fender | bcrisp: What "hardware'? |
| 20:20.32 | Naikrovek | ~itsplist_us |
| 20:20.35 | Optic | this used to work but recently stopped, something must have changed :( |
| 20:20.39 | Naikrovek | ~itsplist-us |
| 20:20.40 | infobot | [itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:20.45 | bcrisp | i can't install any IO cards etc on the server |
| 20:20.46 | Naikrovek | bcrisp: ^^^^ |
| 20:21.22 | LemensTS | i like voipinvite and voipinnovations... |
| 20:21.24 | [TK]D-Fender | bcrisp: What do you WANT to use as connectivity goes? |
| 20:21.39 | [TK]D-Fender | Optic: Its not finding the vm box... |
| 20:21.53 | bcrisp | need to be able to handle incoming calls from the outside, and then route those to either analog or sip phones |
| 20:21.59 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:21.59 | [TK]D-Fender | syntax_erorr: Congratulations on finding and exploring the * option then. |
| 20:22.10 | Optic | d-fender: yeah, it seems to be in /etc/asterisk/voicemail.conf... |
| 20:22.12 | [TK]D-Fender | bcrisp: Lets try this again... |
| 20:22.15 | bcrisp | haha |
| 20:22.23 | [TK]D-Fender | bcrisp: What is the call coming in on? |
| 20:22.59 | bcrisp | : | |
| 20:23.00 | [TK]D-Fender | Optic: PASTEBIN your complete call attempt and your configs. |
| 20:23.15 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 20:23.17 | Optic | configs are freepbx so they're messsy |
| 20:23.17 | bcrisp | the server hosting asterisk doesn't have telephony hardware |
| 20:23.18 | [TK]D-Fender | ~pb |
| 20:23.19 | 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 |
| 20:23.20 | [TK]D-Fender | ^^^^^^^^ |
| 20:23.25 | [TK]D-Fender | bcrisp: ..... |
| 20:23.35 | [TK]D-Fender | bcrisp: Doesn't answer my question. |
| 20:23.36 | bcrisp | i need to be able to have people call in from analog devices from all over |
| 20:23.50 | Naikrovek | bcrisp: we understand this |
| 20:23.55 | [TK]D-Fender | bcrisp: Where the caller is isn't important. how the call is delivered to YOU is |
| 20:24.01 | bcrisp | ok... |
| 20:24.05 | Naikrovek | those voice providers give you service without the need for hardware, which is why we're telling you this |
| 20:24.08 | bcrisp | thats the part im having trouble with |
| 20:24.17 | [TK]D-Fender | bcrisp: Allow me to translate. |
| 20:24.18 | bcrisp | ok THATS what im asking about :) |
| 20:24.24 | bcrisp | Naikrovek |
| 20:24.33 | [TK]D-Fender | bcrisp: WHAT KIND OF FUCKING WIRE GETS YOU THAT CALL INTO THE DAMN BUILDING!? :P |
| 20:24.37 | bcrisp | lol |
| 20:24.38 | Optic | i suspect something has gone bad with permissions or asterisk's home directory or something ;) |
| 20:24.44 | Naikrovek | he has analog now i think or nothing |
| 20:24.55 | Naikrovek | he wants voip incoming, but doesn't know if such a thing exists |
| 20:24.58 | Naikrovek | i think |
| 20:25.01 | bcrisp | yes i WANT voip incoming |
| 20:25.08 | Naikrovek | or didn't know until very recently |
| 20:25.09 | [TK]D-Fender | Naikrovek: He never said that |
| 20:25.13 | [TK]D-Fender | (till now) |
| 20:25.15 | bcrisp | haha |
| 20:25.18 | Naikrovek | [TK]D-Fender: i know but i infered it |
| 20:25.27 | Naikrovek | usually i infer incorrectly but not this time |
| 20:25.31 | bcrisp | im showing signs of stigmata.... |
| 20:25.41 | Naikrovek | he was having trouble putting it into words i think is all |
| 20:25.44 | bcrisp | yes |
| 20:25.51 | [TK]D-Fender | Naikrovek: If you're gonna put words in there mouths, lace it with cyanide first! |
| 20:25.54 | bcrisp | its kind of like beating your kid because he can't pronounce "yellow" |
| 20:25.56 | Optic | d-fender: here is a complete call to freepbx *97 |
| 20:25.58 | Optic | http://pastebin.com/m57c1466e |
| 20:26.02 | Naikrovek | [TK]D-Fender: i did. around dinnertime he's gonna fall over |
| 20:26.09 | [TK]D-Fender | ~freepbx |
| 20:26.10 | 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 |
| 20:26.12 | [TK]D-Fender | Optic: ^^^^^^^^^^^^^^ |
| 20:26.16 | Optic | yeah |
| 20:26.18 | Optic | sorry :( |
| 20:26.28 | Naikrovek | it's cool, they're just more equipped to help you than we |
| 20:26.35 | Naikrovek | the #freepbx guys i mean |
| 20:26.51 | Optic | I strongly suspect an asterisk problem |
| 20:27.01 | Optic | but I will ask there |
| 20:27.07 | bcrisp | Naikrovek - ok, so if i need incoming voip, whats the terminology im looking for -- a voice provider? |
| 20:27.17 | Naikrovek | bcrisp: that's the terminology I use |
| 20:27.54 | Naikrovek | bcrisp: the afore-mentioned companies (/msg infobot itsplist-us) can get you set up |
| 20:28.29 | bcrisp | thank you.. i guess somehow i missed it .. yes i see it up there now |
| 20:28.34 | bcrisp | appreciate it |
| 20:28.38 | Naikrovek | any time |
| 20:28.42 | Naikrovek | nevermind our attitudes |
| 20:28.50 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:28.51 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 20:28.51 | Naikrovek | we're real people with real jobs which are sometimes in furiating |
| 20:28.57 | Naikrovek | infuriating* |
| 20:29.07 | bcrisp | you mean like having to learn linux, asterisk, mysql, perl in one week? |
| 20:29.14 | Naikrovek | that would qualify |
| 20:29.18 | bcrisp | ok thats me |
| 20:29.25 | bcrisp | :) |
| 20:29.37 | Naikrovek | hang out, keep reading, and in a couple weeks you'll be able to help people with their own questions |
| 20:30.08 | bcrisp | thanks for your patience, and D-Fender, as long as your nick includes "Fender" i'll still be nice to u |
| 20:30.08 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 20:30.38 | [TK]B-Bender | htm. |
| 20:30.40 | [TK]B-Bender | hrm. |
| 20:30.46 | bcrisp | oh no you didn't |
| 20:30.47 | eppigy | HTMMMM |
| 20:30.48 | Naikrovek | Nah. |
| 20:31.19 | bcrisp | i really like xchat more than mirc |
| 20:31.24 | Naikrovek | yeah |
| 20:31.32 | Naikrovek | get a good theme on there and it's even better |
| 20:32.09 | bcrisp | last month i had to learn silverlight and wcf |
| 20:32.14 | bcrisp | : / |
| 20:32.20 | Naikrovek | you are overworked my man |
| 20:32.49 | bcrisp | nah its all good.. i like the satisfaction when i actually get things rolling |
| 20:33.04 | DocAwesome | has anyone used IAX2 channels as dynamic members in a queue? The reason I ask is that I'm writing some documentation and have never used IAX2 peers as queue members, and I'm wondering how others have handled the lack of device state information? |
| 20:33.07 | bcrisp | i wonder if i should steer clear of perl tho in all honesty |
| 20:33.15 | DocAwesome | at least this is what looks to be the problem in my test environment |
| 20:33.37 | Naikrovek | bcrisp: perl is awesome, and is only as hard as you make it |
| 20:34.22 | [TK]D-Fender | DocAwesome: It has the same devicestate possibilities as SIP AFAIK |
| 20:34.48 | [TK]D-Fender | DocAwesome: Basic presence is the same and isn't really tech dependent. its based on the generic driver model & channels |
| 20:35.02 | DocAwesome | I don't think that is the case |
| 20:35.09 | [TK]D-Fender | Optic: And your PB didn't have an error that i could see in it, nor your configs |
| 20:35.21 | [TK]D-Fender | DocAwesome: Show me :) |
| 20:35.25 | smooth_penguin | /] |
| 20:35.30 | Optic | d-fender: i know, it's a really quiet failure |
| 20:35.41 | Optic | I think app_voicemail is not finding the user |
| 20:35.42 | DocAwesome | [TK]D-Fender: you show me where device states exist for IAX2 |
| 20:35.50 | [TK]D-Fender | applauds Optic ..... with 1 hand |
| 20:36.05 | Optic | but from the error messages, even in the debuggingest mode, I can't see why :( |
| 20:36.18 | [TK]D-Fender | DocAwesome: you can do hints on them just the same. Same state <- |
| 20:36.18 | *** join/#asterisk southtel_ (n=slester@68-114-19-101.dhcp.gwnt.ga.charter.com) |
| 20:36.36 | [TK]D-Fender | Optic: What error messages? the PB I asked you for didn't HAVE any. |
| 20:36.57 | Optic | http://pastebin.com/m724aa1dd |
| 20:37.05 | Optic | this one, from before, from the full log |
| 20:37.37 | southtel_ | Has anyone ever seen an * system intermittently not get CID? |
| 20:37.56 | [TK]D-Fender | Optic: I said show me a COMPLEtE call with the errorand your CONFIGS |
| 20:38.07 | [TK]D-Fender | southtel_: Off flakey analog lines, sure |
| 20:38.26 | Naikrovek | Optic: if you give him what he wants he'll know how to fix it, almost certainly. he's smart. |
| 20:38.42 | southtel_ | Ouch, the answer I didn't want to hear. |
| 20:38.52 | Optic | complete call, from which log? |
| 20:39.17 | [TK]D-Fender | Optic: * CLI |
| 20:39.24 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 20:39.28 | southtel_ | Anything I can do to improve that? tx/rx gain perhaps? |
| 20:39.35 | Optic | that's what was in the last paste :( what configs would you like to see? |
| 20:39.37 | [TK]D-Fender | southtel_: Yup |
| 20:39.46 | [TK]D-Fender | Optic: voicemail.conf |
| 20:39.55 | southtel_ | D-Fender: bumpt them up perhaps? |
| 20:40.04 | [TK]D-Fender | southtel_: or down. |
| 20:40.31 | southtel_ | That covers the options...I'll try those, thanks. |
| 20:41.27 | DocAwesome | [TK]D-Fender: I'm not sure what you're expecting me to show you |
| 20:41.31 | DocAwesome | [TK]D-Fender: if you add a 'hint' to the dialplan for an IAX2 peer it still shows up as Unavailable or Unknown |
| 20:42.33 | [TK]D-Fender | DocAwesome: News to me.... |
| 20:42.42 | [TK]D-Fender | DocAwesome: Tons of samples showing its use |
| 20:42.49 | DocAwesome | where are these examples? |
| 20:43.43 | Optic | here is voicemail.conf |
| 20:43.44 | Optic | http://pastebin.com/d44cd6cd1 |
| 20:45.08 | bcrisp | would it be fair to say that i'm looking for a pstn gateway service? |
| 20:45.38 | Katty | i'm hungry |
| 20:45.48 | bcrisp | i just ate some gyros |
| 20:45.56 | Katty | what's a gyro |
| 20:46.06 | eppigy | man i love gyros |
| 20:46.09 | bcrisp | greek |
| 20:46.10 | eppigy | joly cow |
| 20:46.16 | eppigy | lamb |
| 20:46.32 | Katty | hmm. |
| 20:46.36 | Katty | what's it look like |
| 20:46.54 | [TK]D-Fender | DocAwesome: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions <- show just about every other channel type reporting back, and I've seen too many 3rd party samples with it, but never directly attempted it... I may just... |
| 20:47.02 | bcrisp | pita with lamb/beef inside |
| 20:47.06 | eppigy | Katty: http://www.propertiesinparadisevi.com/shared/genericrestaurant/gyro.jpg |
| 20:47.15 | eppigy | with a really good sauce |
| 20:47.34 | bcrisp | tsaziki or however it is spelled |
| 20:47.35 | eppigy | http://www.cafemed1.com/images/gyro.jpg |
| 20:47.37 | eppigy | OMNOM |
| 20:48.18 | bcrisp | ~vanity request |
| 20:48.26 | bcrisp | anyone know what that is? |
| 20:48.36 | [TK]D-Fender | DocAwesome: We know it also works for Zap...It'd be kind sad if IAX2 were broken in this capacity |
| 20:49.32 | VaGoNeTaS | g2g |
| 20:50.02 | Optic | d-fender: any ideas? |
| 20:50.19 | bcrisp | im looking at juntionnetworks' pstn gateway.. looks reasonable |
| 20:52.23 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 20:54.37 | Katty | eppigy: that looks pretty good |
| 20:54.57 | heliosj | Anyone using Asterisk as a radio repeater? |
| 20:55.26 | Katty | eppigy: i had an overly ripe bartlette pear and a banana |
| 20:55.27 | DocAwesome | heliosj: I'm sure someone is |
| 20:55.29 | [TK]D-Fender | heliosj: Only those HAM freaks who haven't discovered BACON yet :p |
| 20:55.47 | heliosj | [TK]D-Fender: I'm a cannibal. |
| 20:55.57 | heliosj | DocAwesome: I'm sure someone is too. :p |
| 20:56.06 | Katty | you're going to die from various bacterial infections |
| 20:56.19 | heliosj | Gotta die somehow. |
| 20:56.26 | Katty | i can think of better ways to die |
| 20:56.27 | [TK]D-Fender | must remember to pick up some fava beans and a nice bottle of chianti on the way home... |
| 20:56.33 | heliosj | lol |
| 20:56.47 | Katty | i'm making 3lbs of taco meat tonight |
| 20:56.59 | Qwell | that's one big taco! |
| 20:57.06 | bcrisp | im making shabu shabu |
| 20:57.06 | Katty | Qwell: not really. |
| 20:57.28 | Katty | Qwell: making Tacos, Taco Salad, and Idaho tacos. hopefully it'll last the two of us at least through the weekend |
| 20:57.29 | p3nguin | bcrisp: How do you pronounce gyro? :) |
| 20:57.34 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:57.36 | Katty | guyro? |
| 20:57.39 | bcrisp | p3nguin: "yeeeero" |
| 20:57.48 | Katty | like Nero? |
| 20:57.53 | bcrisp | Yeero |
| 20:57.54 | Katty | sitting on a hill. while his city burned. |
| 20:57.55 | p3nguin | gives thumb up |
| 20:58.02 | DocAwesome | [TK]D-Fender: your link only shows SIP channels being used |
| 20:58.05 | [TK]D-Fender | Katty: guy-row |
| 20:58.18 | Katty | [TK]D-Fender: you-row |
| 20:58.19 | bcrisp | p3nguin how do you pronounce tabesasenakerebanaranai |
| 20:58.30 | p3nguin | bcrisp: I don't |
| 20:58.31 | Katty | ye ole ro? |
| 20:58.33 | [TK]D-Fender | DocAwesome: I mentioned that... I don't see formal docs with it... |
| 20:58.36 | Katty | or yero nero |
| 20:58.43 | [TK]D-Fender | Katty: Sure |
| 20:58.45 | p3nguin | http://upload.wikimedia.org/wikipedia/commons/c/cb/Gyros.ogg |
| 20:59.02 | Katty | ah |
| 20:59.03 | Katty | rightr. |
| 20:59.12 | Katty | irc's not the best place for pronunciation :P |
| 21:00.11 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 21:00.22 | bcrisp | it really bugs me when people say "disorientated" when pronouncing disoriented |
| 21:00.32 | p3nguin | bcrisp: I only asked because so many people want to pronounce it like gyro in gyroscope. |
| 21:00.44 | DocAwesome | I say it like "euro" |
| 21:00.58 | voipmonk | the gee is an eich |
| 21:00.59 | bcrisp | p3nguin do you think gyroscope is actually meant to be pronounced that way? :) |
| 21:01.22 | p3nguin | probably not |
| 21:01.32 | bcrisp | im not sure, but somehow i doubt it |
| 21:01.34 | Katty | p3nguin are you guys calling for snow this weekend? |
| 21:01.59 | p3nguin | katty: Sunday night we might get some. |
| 21:02.10 | Katty | do you know which direction it's headed? |
| 21:02.57 | p3nguin | I honestly do not remember, but probably north and east if it's a normal front. |
| 21:03.20 | Katty | hmmmm. |
| 21:03.21 | Katty | k |
| 21:03.25 | bcrisp | its 60s here today.. burr |
| 21:03.33 | p3nguin | It was down to 21 degrees overnight last night. |
| 21:03.45 | p3nguin | only 31 right now. |
| 21:03.53 | bcrisp | scottsdale az here |
| 21:04.28 | Katty | it's 35F here |
| 21:04.41 | p3nguin | My mother lives in the Florida panhandle, and she said it's been getting into the 30s down there, even. |
| 21:04.48 | Katty | 48% humidity and the pressure is falling |
| 21:04.58 | LemensTS | Global warming |
| 21:05.20 | Katty | i'm gonna miss polar bears |
| 21:06.07 | p3nguin | I should be old and possibly dead by the time they are all gone. |
| 21:06.12 | carrar | yeah |
| 21:06.16 | Katty | i hope so |
| 21:06.22 | Katty | well i hope they're not gone |
| 21:06.27 | carrar | polar bear meat tast good |
| 21:07.16 | carrar | There is a large polar bear at the Juneau Airport |
| 21:07.26 | carrar | second floor |
| 21:07.35 | Katty | i don't suppose anyone know hows to build a window feeder tray, that does not use suction cups. |
| 21:07.41 | carrar | I have a picof it someplace |
| 21:07.51 | bcrisp | duct tape |
| 21:07.58 | Katty | ^_- |
| 21:08.00 | Katty | no |
| 21:08.04 | bcrisp | superglu? |
| 21:08.25 | Katty | no |
| 21:08.29 | Katty | some sort of window mount |
| 21:08.43 | Katty | needs to support around...10lbs i think |
| 21:08.46 | carrar | http://pics.osburn.com/photo/1635/original |
| 21:09.14 | Katty | carrar: that makes me sad. |
| 21:09.27 | carrar | that thing his huge |
| 21:09.31 | carrar | is |
| 21:09.54 | Katty | yeah, but, it's a polar bear |
| 21:09.57 | Katty | what do you really expect |
| 21:10.17 | carrar | I expected it to smile and hand me a coke |
| 21:10.19 | Katty | how do you keep window boxes attached to a window? |
| 21:11.39 | bcrisp | the window or the sill? |
| 21:12.08 | Katty | idk how they attach |
| 21:12.15 | Katty | but i think i want a tray that's mounted as a window box |
| 21:12.36 | eppigy | it might snow tonight and/or tommorow morning :> |
| 21:12.43 | Katty | :< |
| 21:12.44 | Katty | i do not like snow. |
| 21:12.47 | Katty | it is wet. and cold. |
| 21:12.54 | eppigy | i see it so rarely it is a :> for me |
| 21:13.14 | eppigy | for my dog it is a :< |
| 21:15.08 | *** join/#asterisk galeras (n=galeras@190.146.13.27) |
| 21:15.50 | galeras | Is there any way to disable manager logging? |
| 21:24.48 | LemensTS | Maybe in manager.conf ? |
| 21:25.42 | Katty | eppigy: i am very surprised yourpuppy does not like snow. |
| 21:25.48 | Katty | eppigy: riddick goes absolutely mad in the white fluff |
| 21:26.25 | Katty | also, i am hungry again :< |
| 21:26.56 | [TK]D-Fender | chackout time, later all |
| 21:33.06 | voipmonk | i just ate 6 donuts |
| 21:33.29 | Optic | yow |
| 21:33.58 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 21:34.29 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 21:38.40 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:40.51 | Katty | i guess i could make a tray stand. |
| 21:43.38 | Pan3D | are there any public ENUM DB? or are they all for-profit? |
| 21:46.19 | p3nguin | ~enum |
| 21:46.20 | infobot | i heard enum is http://www.voip-info.org/wiki-Enum |
| 21:46.24 | ecrane | E164.org although I've never used it |
| 21:47.09 | LemensTS | voipmonk: how much did you smoke |
| 21:47.18 | eppigy | Katty: it makes his paws cold :< |
| 21:48.45 | Katty | eppigy: get him booties |
| 21:48.45 | p3nguin | If I'm using VoIP to an ITSP only, with no PSTN connectivity on my side, is enum something I need to bother learning about? |
| 21:48.55 | Pan3D | ecrane: thanks mang |
| 21:49.05 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 21:49.21 | Katty | do you think hard drive magnets could penetrate a window? |
| 21:49.26 | Katty | maybe i could mount a squirrel feeder tray with those |
| 21:49.42 | *** join/#asterisk kW (n=root@unaffiliated/kw) |
| 21:49.55 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 21:50.08 | ecrane | Can it hold the weight of the feeder + feeder food + a squirrel hanging off of it? |
| 21:50.26 | Katty | a regular hard drive magnet could |
| 21:50.27 | Naikrovek | Katty: they're very strong up close, but not very strong at all past about 5mm distance |
| 21:50.31 | Katty | but...i'm not sure about through the window |
| 21:50.39 | kW | Hello, does anybody know how to the incoming caller ID be properly scoped (currently local calls, national calls or international calls are all without prefixes (e.g. "0", "00" or "+")) |
| 21:50.44 | Katty | well, scratch that idea then. |
| 21:50.59 | Naikrovek | probably through a window, yes, but it won't grip the window tightly enough to prevent it from sliding down |
| 21:51.01 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 21:51.03 | ecrane | And how strong is the window ^^. |
| 21:51.15 | Katty | another good question |
| 21:51.20 | Katty | ij ust don't want to have to drill INTO the house |
| 21:52.02 | Katty | short of putting a table directly up against the window, and placing a tray on top of it...i'm pretty much out of ideas |
| 21:52.31 | kW | basically, nationalprefix and internationalprefix does not work |
| 21:52.46 | eppigy | that is a good idea |
| 21:53.26 | p3nguin | katty: http://ep.yimg.com/ca/I/incentrakiii_2080_14473549 |
| 21:53.52 | Katty | oh man |
| 21:54.00 | *** join/#asterisk DMeloUK (n=DominicM@127.159.119.70.cfl.res.rr.com) |
| 21:54.02 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 21:54.07 | Katty | that'd work |
| 21:54.28 | Katty | what's that like some sort of shower curtain rod thing |
| 21:54.29 | p3nguin | Or if you want a really bad-ass feeder, http://ep.yimg.com/ca/I/incentrakiii_2080_7195229 |
| 21:54.38 | DMeloUK | anyone had any experience with gagaphone? |
| 21:54.41 | p3nguin | Looks like a regular dowl rod to me. |
| 21:54.44 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 21:54.44 | Naikrovek | you can mount to glass using those 3M command strips |
| 21:54.52 | Naikrovek | you can stick anything to anything with thsoe |
| 21:54.53 | Katty | hmm. |
| 21:54.56 | *** join/#asterisk steve745 (n=steve@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 21:54.57 | Katty | also a good idea |
| 21:55.03 | Katty | but what kind of tray can be attached to those |
| 21:55.17 | *** join/#asterisk [Ryto] (n=Email@82-169-89-194.ip.telfort.nl) |
| 21:55.20 | Katty | i do like the rod idea, but i'm not sure how to design something which would be squirrel friendly |
| 21:55.34 | Katty | i'm guessing the bottom portion of the tray would have to have holes in it |
| 21:55.34 | Naikrovek | just the strip i'm talking, will stick to anything flat & clean |
| 21:55.40 | p3nguin | How does the squirrel get to the feeder? |
| 21:56.01 | Katty | they can reach from the porch floor |
| 21:56.19 | ecrane | If there is food in the feeder, the squirrel will find a way to get there. |
| 21:56.35 | Katty | it's keeping the feeder attached to the window while the squirrels are on it, that's the problem |
| 21:56.43 | Katty | because my fat furry little friends are HUGE |
| 21:56.49 | [Ryto] | Within a few minutes, a sures comes here named EDRO who needs help setting up an asterisk setup.. Who could help him? |
| 21:56.52 | Katty | the typical feeder won't hold them |
| 21:57.00 | [Ryto] | *user |
| 21:57.16 | Katty | there's also the possibility that the racoon who's been lurking about will get into it as well |
| 21:57.27 | ecrane | http://www.youtube.com/watch?v=nWU0bfo-bSY |
| 21:57.29 | Katty | there needs to be a way to detach the ray for cleaning |
| 21:58.14 | *** join/#asterisk stevetotaro (n=Steve@c-68-49-104-232.hsd1.md.comcast.net) |
| 21:58.23 | [Ryto] | anyone? ^^ |
| 21:59.02 | Katty | i like the idea of that rod thing, but i don't have anything that can drill a hole into the tray |
| 21:59.05 | p3nguin | katty: I think I would go with the dowl rod idea if the feeder is compatible. |
| 21:59.13 | Katty | i'll be building the tray |
| 21:59.17 | Katty | it needs to be quite large |
| 21:59.34 | Katty | 2ft by 1ft by 3inch deep |
| 21:59.36 | p3nguin | You can get dowls at least 1 inch in diameter, so that should hold. |
| 22:00.16 | p3nguin | You really don't even have to use down rod. You could cut a strip of wood 1x1 and serve the same purpose. |
| 22:00.19 | Katty | p3nguin: yes, i think you're right. |
| 22:00.30 | p3nguin | err, dowl rod |
| 22:00.38 | Katty | i just don't know how to get the hole into the tray i'd build |
| 22:00.57 | p3nguin | Attach it externally instead of through a hole. |
| 22:01.09 | *** part/#asterisk [Ryto] (n=Email@82-169-89-194.ip.telfort.nl) |
| 22:01.14 | Katty | with what? |
| 22:01.24 | p3nguin | screws, nails |
| 22:01.29 | p3nguin | liquid nails |
| 22:01.39 | Katty | 3m stickies! |
| 22:01.50 | p3nguin | What will you use to frame the feeder? |
| 22:01.55 | eppigy | SMALL ANIMALS |
| 22:01.58 | Tech_Travis | Katty: lag bolts? |
| 22:01.59 | Katty | nothing |
| 22:02.01 | Katty | it's just a tray |
| 22:02.06 | p3nguin | oh hmm |
| 22:02.08 | Katty | a tray with...sticks on either end |
| 22:02.13 | Katty | which will fit into the window sill |
| 22:02.21 | Katty | i hope. |
| 22:02.39 | Katty | i think a hole in the bakc of the tray would allow more weight tho |
| 22:03.16 | Katty | if i could somehow cut groves into the bottom sections of the board, i could slide plexiglass in as the bottom |
| 22:03.21 | Katty | would make it much easier to clean |
| 22:03.59 | p3nguin | You can make grooves using a regular circular saw with the blade set very shallow. |
| 22:04.00 | Katty | or really, i could just make the entire tray of out plexiglass |
| 22:04.07 | Katty | yeah but i don't have a circular saw :< |
| 22:04.15 | Katty | i have a hammer! and lowes. |
| 22:04.15 | p3nguin | ah, that poses a problem. |
| 22:04.31 | Katty | i think lowes can cut a hole into boards for me tho |
| 22:04.40 | p3nguin | I guess it's safe to assume that you also do not have a table saw. |
| 22:04.58 | Katty | no |
| 22:05.01 | Katty | i have a hammer. |
| 22:05.08 | Katty | well, i do have a drill |
| 22:05.21 | p3nguin | I should put together a little kit and make it so you only need a hammer to assemble it. :) |
| 22:05.25 | Katty | there's probably some attachment of ryans that can cut a hole. |
| 22:05.42 | Katty | digs up mspaint |
| 22:07.37 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:12.35 | Katty | uploads |
| 22:13.01 | Katty | http://imagebin.org/74111 |
| 22:13.38 | Katty | i guess the tray could be cut to fit directly into the tray |
| 22:13.53 | Katty | it would help protect the wood at the bottom if liquidy substances got in there |
| 22:15.03 | CGMChris | Do I need ztdummy installed for SIP to operate properly? |
| 22:16.07 | DMeloUK | I cannot figure out how to route this incoming DID |
| 22:16.38 | DMeloUK | anyone have any exp with gagaphone.com trunks ? |
| 22:17.04 | Katty | i bet lowes would sell me a kitchen drawer and drill holes into it for me |
| 22:19.50 | DMeloUK | katty - I helped my wife with a similar project - we ended up framing the shelf and then running the feeder in front of the window |
| 22:20.11 | DMeloUK | like putting up shelves |
| 22:20.29 | DMeloUK | one on either side and then the feeder shelf sits on top |
| 22:21.32 | Katty | DMeloUK: do you think you could find a photo of it? |
| 22:23.01 | LemensTS | Anyone ever get hit for a rogue 911 call |
| 22:24.09 | DMeloUK | katty - I will have a look |
| 22:24.23 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 22:25.01 | DMeloUK | but the basic premise is that for small animals over 7 - 9 llbs - you need two support points |
| 22:25.04 | Katty | p3nguin: you know a lot of wooden trays already have 'handles' which a dowl rod would easily fit through |
| 22:25.20 | *** join/#asterisk manxpower (n=ewieling@84.sub-70-223-191.myvzw.com) |
| 22:25.25 | DMeloUK | you would need to do some kind of crossbar |
| 22:25.31 | DMeloUK | two rods |
| 22:25.37 | DMeloUK | like an X |
| 22:25.52 | mchou | LemensTS: what's a rouge 911 call? |
| 22:26.02 | fo0bar | LemensTS: we used to have a cisco callmanager system that, for whatever stupid reason, would dial 911 if you dialed 9 or 91 and then waited too long to finish dialing. we'd get police to the office every few months at least |
| 22:26.05 | Katty | DMeloUK: why? |
| 22:26.06 | mchou | LemensTS: is it a prank call? |
| 22:26.18 | DMeloUK | but that would still make it front heavy |
| 22:26.22 | fo0bar | if that's what you mean by "rogue" |
| 22:26.33 | DMeloUK | the drawer would tilt forward when weight was applied |
| 22:26.44 | DMeloUK | forward and down |
| 22:26.50 | Katty | hmm, point. |
| 22:27.00 | Katty | well |
| 22:27.05 | Katty | it will be sitting against the bottom of the window sill |
| 22:27.07 | Katty | so that will help |
| 22:27.21 | Katty | i can make it a bit more narrow too |
| 22:27.30 | DMeloUK | makes it a bit easier |
| 22:27.34 | DMeloUK | enlarge the sill |
| 22:27.42 | DMeloUK | frame at the sides |
| 22:27.42 | p3nguin | dmelouk: Did you get the proxy and registration information from the ITSP already? |
| 22:27.43 | Katty | i could also use some sort of a chain with a hook |
| 22:27.59 | Katty | and connect a second dowl rod in the front with the chain tot he ones in the back |
| 22:28.14 | DMeloUK | p3nguin - yes I have the reg info but it refuses to route on incoming did |
| 22:28.17 | Katty | no, that wouldn't work |
| 22:28.26 | p3nguin | dmelouk: Does the call reach your box at all? |
| 22:28.34 | DMeloUK | p3nguin |
| 22:28.40 | DMeloUK | yes plays "not in service" |
| 22:28.50 | p3nguin | That's not on your box, I'm sure. |
| 22:29.16 | DMeloUK | if I edit the incoming did to blank the calls come through |
| 22:29.43 | DMeloUK | its when I try to get clever with the did's it defaults to the not in service message |
| 22:29.45 | Katty | okay, what if i put 3m sticky hooks on the top of the window sill, and then hung the tray |
| 22:29.51 | p3nguin | Okay, so debug the call and see what is going on when a call comes in. Adjust your dialplan as needed. |
| 22:30.00 | DMeloUK | tried that |
| 22:30.13 | DMeloUK | looked at from, x-did, etc.. |
| 22:30.23 | DMeloUK | none of them seem to route |
| 22:30.48 | DMeloUK | did a comparison between a working one and this and nothing stood out |
| 22:30.53 | p3nguin | Make it so the call is not coming in correctly, then get a sip debug and put it in a pastebin. |
| 22:31.09 | DMeloUK | k minute |
| 22:31.33 | DMeloUK | katty frame out the sill and just put the tray on top loose |
| 22:31.52 | DMeloUK | more versatile that way |
| 22:32.02 | DMeloUK | and you wouldn't have to bother with tools to take it out to clean |
| 22:32.45 | Katty | that would require modifying the house |
| 22:32.56 | Katty | which is what i'm trying to avoid |
| 22:33.40 | Katty | i'm starting to wonder if heavy duty velcro would work |
| 22:33.51 | Katty | along with the rod |
| 22:34.04 | p3nguin | dmelouk: I have to take off for maybe 30 minutes, so that will give you time to "sip set debug" and make a failed call. I'll check for the pastebin when I get back. |
| 22:37.28 | DMeloUK | thanks p3nguin |
| 22:37.38 | DMeloUK | http://www.pastebin.ca/1702419 |
| 22:37.57 | dlynes | If I have asterisk bound to two ip addresses for sip, I can set 'fromaddress=xx.xx.xx.xx' to send packets out through a specific interface, right? |
| 22:41.29 | *** join/#asterisk hluesea (n=hulusika@88.239.46.254) |
| 22:48.26 | dlynes | DocAwesome: did issue 13114 ever get added to asterisk 1.6? |
| 22:51.49 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) |
| 22:51.58 | DocAwesome | M13114 |
| 22:52.30 | DocAwesome | dlynes: I don't know what you mean |
| 22:52.45 | DocAwesome | that was a feature request |
| 22:52.48 | DocAwesome | I don't see a patch for it |
| 22:52.59 | dlynes | DocAwesome: ah, ok...so it never got added, then....thanks |
| 22:53.37 | dlynes | DocAwesome: there was some talk in that feature request about there being a branch that had some code in it for that feature, but it being lost or something |
| 22:53.56 | vlt | Hello. I try to assign the output of a shell command to a variable. What is wrong with following line? "Set(VAR_1=${System(/etc/asterisk/script.sh ${CALLERID(num)})})" I get "ERROR[2587]: pbx.c:1403 ast_func_read: Function System not registered" |
| 22:54.15 | DocAwesome | hmm, never heard of it |
| 22:54.16 | dlynes | vlt: system is an application, not a function |
| 22:54.31 | bcrisp | "received SIP subscribe for peer without mailbox" |
| 22:54.49 | vlt | dlynes: How do I have to rewrite it that it works? |
| 22:55.18 | dlynes | vlt: all system() will return is failure or success |
| 22:56.57 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 22:57.13 | dlynes | bcrisp: do you have a question, or did you just accidentally paste your clipboard contents? |
| 22:57.49 | vlt | dlynes: Will it return failure when my script returns non-zero? And how to assign this value to a variable? |
| 22:57.58 | bcrisp | dlynes i did accidentally paste it |
| 22:58.24 | bcrisp | dlynes: but ya.. im receiving that warning and am not sure what it means |
| 22:58.53 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:58.56 | p3nguin | It means that you didn't configure a mailbox for the peer who is asking for mailbox status. |
| 22:59.04 | dlynes | bcrisp: it means you've got a mailbox= entry in your sip config, but you don't have a matching entry in yoru voicemail.conf file |
| 22:59.06 | bcrisp | ok .. i used extensions for the vm boxes |
| 22:59.12 | p3nguin | dmelouk: "Looking for s in from-trunk-sip-GagaPhonePeru" |
| 22:59.28 | bcrisp | oh ok.. let me look at sip.conf |
| 22:59.34 | dlynes | vlt: you probably want TrySystem, instead |
| 22:59.43 | p3nguin | bcrisp: and voicemail.conf |
| 22:59.49 | vlt | dlynes: Thank you. |
| 22:59.50 | dlynes | vlt: that'll also return APPERROR, if your application returned an error |
| 23:00.08 | dlynes | vlt: but nothing in asterisk will buffer stdout afaik |
| 23:00.19 | *** join/#asterisk doctorray (n=ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 23:00.31 | bcrisp | p3nguin: in voicemail.conf i have them setup as extension #s, under the SIP conf do i need to just say mailbox=#? |
| 23:00.50 | p3nguin | dmelouk: There's no phone number being sent in your packets. I just went through this same thing on my box with sipgate. I changed my register statement to include the phone number, and the problem was fixed. |
| 23:00.57 | vlt | dlynes: true/false is enough. For things I need to use output from I use ${CURL ...} ;-) |
| 23:01.07 | Katty | haha i love my coworkers |
| 23:01.30 | p3nguin | bcrisp: 1234 => 4242,Example Mailbox,root@localhost |
| 23:01.31 | Katty | metal condute, in brackets, attached to the back of the feeder |
| 23:01.33 | dlynes | bcrisp: in sip.conf (under your sip user configuration), add mailbox=mymailbox, and in voicemail.conf, add in mymailbox => myvmpassword,...... |
| 23:01.34 | Katty | it will hold tons of weight |
| 23:01.39 | Katty | cheers |
| 23:01.40 | p3nguin | bcrisp: There is no "extensions" in voicemail.conf. |
| 23:01.44 | Katty | goes to lowes |
| 23:01.55 | p3nguin | bcrisp: The 1234 is the mailbox number. |
| 23:02.02 | bcrisp | yes |
| 23:02.10 | bcrisp | i have that all setup |
| 23:02.14 | hardwire | hihihihihihihihihihihihihihihihihihihihihihihi |
| 23:02.16 | bcrisp | in sip.conf under the sip user |
| 23:02.16 | DMeloUK | p3nguin they advocated that with register statement /loginID |
| 23:02.17 | p3nguin | bcrisp: Create a mailbox=1234 in sip.conf for a peer, and then 1234 is that peer's voicemail number. |
| 23:02.21 | dlynes | bcrisp: if you did, you wouldn't be getting an error |
| 23:02.23 | DMeloUK | I tried that no luck |
| 23:02.37 | dlynes | bcrisp: either that, or you haven't reloaded your voicemail.conf file |
| 23:02.39 | bcrisp | mailbox=351 (since 351 is entered in voicemail.conf as a mailbox #) |
| 23:02.46 | bcrisp | ok cool thanks guys |
| 23:02.47 | dlynes | bcrisp: module reload app_voicemail.so |
| 23:02.49 | p3nguin | You have to reload. |
| 23:02.51 | bcrisp | i just added the sip.conf part so thank you |
| 23:02.52 | bcrisp | yep |
| 23:02.57 | DMeloUK | (loginid:password@sip.gagaphone.com/LoginID). |
| 23:03.10 | p3nguin | dmelouk: Is loginid the phone number? |
| 23:03.15 | DMeloUK | no |
| 23:03.28 | p3nguin | dmelouk: Change it to /phonenumber. |
| 23:03.29 | DMeloUK | should I change it to /DID and try routing on that? |
| 23:03.30 | doctorray | hi all, my asterisk box has been developing a mild 'chirping' in the background of calls as of late; haven't changed config much and it's been doing it since I upped to 1.6.x; I'm currently on 1.6.1.11... ring any bells for anyone?? :) two ip phones, and a 4 port digium analog card to connect to the phone company |
| 23:03.39 | dlynes | bcrisp: if you're using an older version of asterisk, you might need to unload and then load, instead of reloading (some older versions of asterisk didn't properly reinitialize the voicemail information) |
| 23:03.52 | p3nguin | dmelouk: That's what fixed it for me. Without /phonenumber, it was hitting the 's' extension. |
| 23:04.08 | bcrisp | dlynes: i used sip reload and voicemail reload |
| 23:04.16 | p3nguin | dmelouk: I even put the wrong DID number on there at first, so it was routing to the WRONG extension. |
| 23:04.20 | bcrisp | dlynes: i think that fixed things, thanks |
| 23:04.30 | dlynes | bcrisp: yeah...if you've got the 'voicemail reload' command, it's a non-issue...you're running a newer version, then |
| 23:04.54 | bcrisp | dlynes: cool thanks |
| 23:05.10 | dlynes | bcrisp: it was mostly an issue on very early 1.4 versions, 1.2 and 1.0 |
| 23:05.30 | p3nguin | Usually that error arises when you have mailbox= in the sip peer, but don't have the voicemail configured for that mailbox number. |
| 23:05.41 | bcrisp | in my case it was opposite |
| 23:05.47 | bcrisp | i didn't have it in the sip.conf |
| 23:05.51 | p3nguin | Well that doesn't make any sense. |
| 23:05.55 | bcrisp | hehe |
| 23:06.07 | p3nguin | Since it's the sip peer that determins what mailbox a user has, and all. |
| 23:06.18 | bcrisp | maybe i have them configured incorrectly |
| 23:06.25 | DMeloUK | p3nguin - alas no joy :( |
| 23:06.28 | dlynes | p3nguin: actually, it does make sense |
| 23:06.38 | dlynes | p3nguin: he had it configured in his phone, but not sip.conf |
| 23:06.42 | p3nguin | oh |
| 23:06.57 | bcrisp | right, the phone is asking for all kinds of stuff and i think asterisk is like 'wtf?' |
| 23:06.58 | p3nguin | I thought * was telling the client to check a certain mailbox, which was not configured. |
| 23:07.11 | dlynes | p3nguin: nah...it was a client/asterisk issue |
| 23:07.39 | dlynes | p3nguin: most of the time i ignore shit like that, because some phones don't allow you to turn the voicemail option off |
| 23:07.54 | p3nguin | dmelouk: sip reload, if you didn't already. |
| 23:07.59 | dlynes | p3nguin: mediatrix fxs gateways are a prime example |
| 23:08.51 | p3nguin | dmelouk: Check sip show registry and sip show peers to make sure the number is right. |
| 23:09.17 | p3nguin | dmelouk: Then do another debug on a failed call. |
| 23:10.19 | bcrisp | will x-lite notify you of voicemail? |
| 23:11.01 | bcrisp | i guess we'll find out.. coworker is leaving mea voicemail now |
| 23:11.14 | p3nguin | I think I remember MWI in X-Lite. |
| 23:11.36 | dlynes | Is something like ${DB(${ARG2}/option1} valid? |
| 23:11.49 | bcrisp | ah xlite is showing a little envelope icon saying "messages waiting" |
| 23:11.56 | p3nguin | I really can't stand X-Lite, so I don't know what all it has. |
| 23:11.58 | dlynes | bcrisp: congratulations |
| 23:12.07 | bcrisp | too bad you cant click on it :/ |
| 23:12.22 | dlynes | bcrisp: it might even tell you _how many_ messages are waiting, too |
| 23:12.28 | bcrisp | yeah it does |
| 23:12.42 | bcrisp | i left my damn mic at home so this is not as fun |
| 23:13.27 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 23:16.19 | bcrisp | thx for your help p3nguin / dlynes |
| 23:17.42 | DMeloUK | oh my god |
| 23:17.50 | DMeloUK | p3nguin |
| 23:18.07 | DMeloUK | I have to add two incoming routes for each line |
| 23:18.14 | DMeloUK | 1 with the username, 1 with the DID |
| 23:18.25 | p3nguin | I have my doubts about that. |
| 23:18.55 | DMeloUK | when I tryi it with just the DID, it gives fail on the username, just the username, gives fail on the DID |
| 23:19.10 | p3nguin | You only need to create one peer/friend for that ITSP, and you need to have one register statement in the general section. |
| 23:19.13 | vlt | dlynes: Thanks again. TrySystem() and the parsing ${SYSTEMSTATUS} works fine :) |
| 23:19.30 | vlt | s/the/then/ |
| 23:19.45 | dlynes | vlt: not a problem...I had the same issue the other day...that's why i know about it |
| 23:19.57 | DMeloUK | p3nguin - can you provide an example of that? |
| 23:19.58 | dlynes | vlt: I was spawning somethign I wanted to parse the error code from |
| 23:20.15 | dlynes | vlt: I think trysystem/tryexec are both pretty new |
| 23:20.32 | p3nguin | dmelouk: Yes, if you give me a few minutes. |
| 23:20.41 | vlt | dlynes: But works on an old 1.2 install here. |
| 23:20.51 | bcrisp | do they speak english in "what" ? |
| 23:20.52 | bcrisp | :) |
| 23:20.58 | DMeloUK | p2nguin - thanks! |
| 23:23.45 | bcrisp | ok, if i want to record my own call menus will wav work, or do i need to convert them to gsm? |
| 23:24.15 | russellb | 8 kHz wav is fine. |
| 23:24.19 | russellb | mono. |
| 23:25.10 | vlt | bcrisp: `sox in.wav out.ul` makes it very easy. |
| 23:25.35 | p3nguin | dmelouk: http://pastebin.com/d5f560e75 |
| 23:26.19 | bcrisp | (i read thatthx vlt |
| 23:26.40 | bcrisp | yikes... just meant "thanks" |
| 23:26.49 | p3nguin | dmelouk: Without the /DIDnumber on the register statement, there was no phone number being sent, so calls would look for 's' just like yours. |
| 23:27.08 | p3nguin | dmelouk: I added /DIDnumber, and now calls go to the correct extension. |
| 23:27.10 | DMeloUK | p3nguin - so with multiple DID's from the same provider |
| 23:27.40 | p3nguin | I... don't know about that, because I only have one DID from each of my providers. |
| 23:27.42 | DMeloUK | I just need to have mutiple register => lines and not separate trunks |
| 23:28.10 | bcrisp | thast my next hurdle |
| 23:28.53 | DMeloUK | p3nguin - did you get my msg? |
| 23:30.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:34.01 | *** join/#asterisk deitarion (n=deitario@moinmoin/fan/deitarion) |
| 23:36.13 | deitarion | Does anyone here have experience using the Asterisk GUI (It's just called asterisk-gui-2.0-1 by BAPS ipkg) with handwritten config files? I'm wondering how well it reacts to being relegated to a mere monitoring GUI for handwritten config files. |
| 23:37.03 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:38.30 | DMeloUK | deitarion - my experience with the gui is limited to the aa50 appliance |
| 23:38.48 | DMeloUK | anything offbook there invalidates the warranty :) |
| 23:40.14 | deitarion | Heh. I'm running an IP01 (open-source hardware, uClinux-based, no GUI by default, same size and shape as a Linksys PAP2, Blackfin chip capable of at least 4 to 8 calls according to my estimates) |
| 23:41.09 | deitarion | If you decide to look it up, look up IP04. I just only needed one FXO/FXS slot and was low on cash. |
| 23:41.58 | *** join/#asterisk syadnom (n=syadnom@host-98-127-86-3.bln-mt.client.bresnan.net) |
| 23:42.35 | DMeloUK | deitarion - could you perhaps use some of the monitoring code from the freepbx front page :) |
| 23:44.30 | deitarion | DMeloUK: Point. I haven't tried using the freepbx package instead of the asterisk-gui one. |
| 23:45.29 | DMeloUK | the system status page is handy |
| 23:45.52 | bcrisp | i wonder if i can use my altigen ip phone with asterisk |
| 23:47.36 | deitarion | To use a little 'net vernacular, "Woh!" |
| 23:48.02 | deitarion | (FreePBX is a lot heavier than asterisk-gui according to the package manager) |
| 23:48.25 | DMeloUK | yes |
| 23:48.39 | DMeloUK | but I am talking about just the framework and the status page |
| 23:48.50 | DMeloUK | you wanted to monitor it right? |
| 23:49.21 | deitarion | DMeloUK: Yeah. I'm just testing out FreePBX because I never thought to try it. |
| 23:49.54 | DMeloUK | if you want to be real light on the box - how about snmp? |
| 23:49.55 | DMeloUK | http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131 |
| 23:51.03 | dlynes | vlt: oh |
| 23:52.15 | vlt | dlynes: ? |
| 23:53.34 | dlynes | vlt>dlynes: But works on an old 1.2 install here. |
| 23:54.59 | vlt | dlynes: Yes, "1.2.23-BRIstuffed-0.3.0-PRE-1y-j" compiled back in 2007 :) |
| 23:56.11 | bcrisp | im still contemplating asterisk gui |
| 23:56.51 | DMeloUK | does anyone have any feedback regarding asterisk and atom processors? |
| 23:57.02 | DMeloUK | I am looking at building small branch boxes for a client |
| 23:57.27 | DMeloUK | and was wondering if a dual atom 1.6 would be enough |
