| 00:00.28 | ymonsalvez | if somebody knows something I can help |
| 00:05.21 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 00:05.53 | tzafrir_home | how is your gateway connected to Asterisk? |
| 00:08.15 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 00:08.55 | justdave | it's annoying that I get booted off this channel every time there's a netsplit just because the netsplt is between me and channel services on the network :| |
| 00:09.34 | Nivex | justdave: you're just lucky I guess |
| 00:13.45 | ymonsalvez | tzafrir_home: i have gateway gsm startgate 2n connect sangoma dual for pri |
| 00:14.23 | ymonsalvez | and i have gateway gsm connect for sip protocol |
| 00:15.58 | tzafrir_home | Both PRI and SIP should give you disconnect supervision |
| 00:16.12 | tzafrir_home | Maybe this specific gateway is the problem? |
| 00:16.54 | ymonsalvez | no |
| 00:17.53 | *** join/#asterisk flujan (n=flujan@201-95-68-164.dsl.telesp.net.br) |
| 00:18.37 | ymonsalvez | tzafrir_laptop: |
| 00:18.40 | ymonsalvez | Traffic my asterisk is thirty thousand calls per day and when asterisk hangs need to cut all these calls for the tarifiquen |
| 00:19.54 | ymonsalvez | And with soft hangup delayed me a lot of slashing calls are a total of 58 channels |
| 00:20.28 | ymonsalvez | I need to cut those calls by groups |
| 00:20.38 | ymonsalvez | channels |
| 00:20.52 | ymonsalvez | no for only channel |
| 00:22.52 | killfill_ | is it possible to make a "blacklist" somehow?.. i change office, a bug number of the calls, are trying to call the old guys that owned the line... |
| 00:23.14 | [TK]D-Fender | ymonsalvez, then make a script that cycles through the channels you want like 'asterisk -rx "show channels concise"' and kill each one you feel like. |
| 00:23.43 | [TK]D-Fender | killfill_, "show function DB" , "show function CALLERID" |
| 00:23.44 | tzafrir_home | killfill_, look for "ex-girlfriend" syntax |
| 00:24.16 | *** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net) |
| 00:24.56 | codeshah | Hi guys, I am on ubuntu, and just wanted to play with asterisk to do some VOIP stuff. Do I need a dedicated box with a modem card? Or can I just use my dev machine to play around? |
| 00:25.18 | tzafrir_home | A dedicated box is generally preferred |
| 00:25.40 | tzafrir_home | Though you can't just use "a modem card" |
| 00:25.46 | killfill_ | heh.. ex-grilfriend?.. |
| 00:26.01 | JT | killfill_: google it |
| 00:26.09 | killfill_ | doing it.. :P |
| 00:26.10 | codeshah | tzafrir_home, what do you mean? |
| 00:26.37 | tzafrir_home | If you're just toying, you can use your current development box (tzafrir_laptop's laptop and this box run small-time asterisk installs) |
| 00:26.46 | [TK]D-Fender | codeshah, Means you "crappy old modem" is worthless to * and if you want to interface with physical lines you're going to have to buy real hardware |
| 00:27.04 | codeshah | [TK], yeah, thats what I figured . |
| 00:27.16 | codeshah | [TK], preferred hardware? |
| 00:27.37 | codeshah | tzafrir, ok, maybe I will have to setup another box then . |
| 00:27.59 | tzafrir_home | OTOH, you can use a SIP soft-phone. ekiga is included with ubuntu. Though I prefer twinkle if it's in the universe - much more tweakable |
| 00:28.25 | tzafrir_home | codeshah, for a toy installation? doesn't really matter |
| 00:28.41 | codeshah | tzafrir, k . what about the modem though? |
| 00:30.13 | tzafrir_home | Do you happen to have ISDN around? |
| 00:30.13 | codeshah | tzafrir, if I want to interface with phone lines . should I be going out & buying some extra hardware? That part confused me slightly in the book . |
| 00:30.13 | codeshah | tzafrir, unfortunately not . I use broadband cable... and of course have a built in modem in this dell |
| 00:30.14 | JT | codeshah: yes you need to buy the hardware |
| 00:30.14 | [TK]D-Fender | codeshah, Yes, you will need to buy special hardware to interface with your lines, etc. |
| 00:30.59 | codeshah | [TK], cool . is there a list of preferred hardware somewhere? I just want to head to the store & get something ... this is all for toy install, so want toget up and running fast . |
| 00:31.11 | tzafrir_home | There are crappy X100P cards for around 10-20$ . Nice for a toy install but both they and their drivers need some real work |
| 00:31.55 | tzafrir_home | strangely enough http://x100p.com now say they only charge 15$ for their cards (rather than the outragous 35$) |
| 00:32.09 | JT | codeshah: you can't just go to the corner store |
| 00:32.12 | tzafrir_home | Naturally they clame to be "the only genuine" and such crap |
| 00:32.13 | ymonsalvez | [TK]D-Fender: why have tested with this many thanks, he had not thought |
| 00:32.20 | JT | codeshah: most stores don't stock the gear |
| 00:32.31 | codeshah | JT, i guess online purchase eh? |
| 00:33.09 | codeshah | tzafrir, what are you using for your production system? |
| 00:33.19 | tzafrir_home | I wonder why no student took up to the task of providing drivers to any other modem... |
| 00:33.20 | codeshah | I see you can purchase total hardware packages . |
| 00:33.37 | tzafrir_home | such as? |
| 00:35.37 | codeshah | i thought I saw some setup somewhere... essentially sells the computer, setup everything for some price . but that's a packaged solution I believe . |
| 00:36.26 | codeshah | it may have been a consulting company though, that would essentially go to businesses, sellt he hardware, setup system on asterisk and so forth |
| 00:36.58 | killfill_ | ok.. i could check if the caller ID is in the blacklist database.. but when i recieve a call, and take it, how cna i add it to the BD? |
| 00:37.36 | [TK]D-Fender | killfill_, you can do something like update the "last caller" into a specific DB entry and when you dial a special exten, take that and add a blacklist entry for it |
| 00:38.51 | killfill_ | oh you mean like after the call finishes, call extension "88" and then take the last recieved call from the extension that is calling into the DB? |
| 00:38.59 | killfill_ | how do i check the last revieced call for extension X? |
| 00:39.25 | [TK]D-Fender | killfill_, everything time you CALL extension X, have it put a DB entry for itself |
| 00:39.45 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 00:39.55 | [TK]D-Fender | killfill_, YOU have to do EVERYTHING. There is nothing "built-in" about Asterisk. You write your dialplan 100% |
| 00:40.18 | killfill_ | oh sure.. im trying to feel what your telling .. :) |
| 00:40.46 | killfill_ | but the thing is how to mark that the last call was bad or not... |
| 00:40.57 | [TK]D-Fender | killfill_, Should be apparent now. You have to add a TON of DB checks around your dialplan to make this large amount of blacklisting you are planning on doing easier. |
| 00:41.26 | [TK]D-Fender | You make an exten that takes the already stored "last caller" and add it to a DB entry that you can check when calls come in. |
| 00:41.39 | [TK]D-Fender | killfill_, Basically, a few dozen lines of dialplan... |
| 00:42.08 | killfill_ | Aah.. i think i got you.. |
| 00:42.18 | kopke | setting codec g711a, it seems to works! but quite all my calls cann't connect! I will see tommorow |
| 00:42.35 | killfill_ | or i could just use CDR (already using it...) to get the last call, the put that on the "blacklist" bd |
| 00:42.50 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 00:49.52 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 00:50.04 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 00:53.21 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 00:54.24 | Sunmoon__ | hello there |
| 00:55.47 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 00:57.36 | tzafrir_home | codeshah, I tohught you were talking about a toy system. A production would be a separate system |
| 00:58.07 | tzafrir_home | If you're stressed in time, a consultant may be not such a bad idea |
| 00:58.07 | *** join/#asterisk flyman01 (n=fly@88.228.50.123) |
| 00:59.06 | flyman01 | hi all |
| 00:59.25 | Sunmoon__ | helo flyman how are u doing |
| 00:59.33 | Sunmoon__ | I am new to this chat session |
| 00:59.56 | flyman01 | i m working |
| 01:00.02 | Sunmoon__ | where are u at |
| 01:00.07 | Sunmoon__ | and what are u working omn |
| 01:00.17 | flyman01 | openser::( |
| 01:00.23 | Sunmoon__ | :) |
| 01:00.27 | flyman01 | billing platform |
| 01:00.35 | Sunmoon__ | have u checked a2billing ? |
| 01:01.33 | flyman01 | some |
| 01:01.47 | Sunmoon__ | u should chck that |
| 01:02.16 | flyman01 | how is the performans? |
| 01:02.48 | Sunmoon__ | I have heard some good reviews about it |
| 01:03.15 | flyman01 | installation is easy or hard? |
| 01:03.27 | Sunmoon__ | easy |
| 01:04.52 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 01:05.11 | flyman01 | i m looking now |
| 01:05.18 | Sunmoon__ | ok |
| 01:05.43 | flyman01 | a2biling for asterisk + openser or only asterisk |
| 01:05.50 | flyman01 | ? |
| 01:05.55 | Sunmoon__ | dont know |
| 01:06.01 | Sunmoon__ | thats for asterisk for sure |
| 01:06.57 | flyman01 | yes only asterisk |
| 01:08.46 | flyman01 | CDRTOOL for openser but some hard |
| 01:08.55 | Sunmoon__ | what do u mean |
| 01:10.09 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 01:10.57 | *** join/#asterisk huun (n=ertere@88.225.220.76) |
| 01:11.06 | huun | good evening channel |
| 01:11.15 | Sunmoon__ | good evening huun |
| 01:11.25 | flyman01 | good evening |
| 01:11.48 | stybba | good evening |
| 01:12.03 | flyman01 | huun is my friend :) |
| 01:12.05 | Sunmoon__ | good evening |
| 01:12.42 | flyman01 | sunmoon i can a litle speak english |
| 01:12.56 | Sunmoon__ | dont worry flyman same here |
| 01:13.07 | huun | aye im a noob linux user who is trying to help flyman to write i sip server ^^ |
| 01:14.37 | flyman01 | or we try openser and asterisk |
| 01:15.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 01:16.03 | flyman01 | but we have a problem for billing to openser |
| 01:16.22 | flyman01 | there is a few interface for asterisk |
| 01:21.45 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 01:26.27 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 01:26.33 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 01:33.15 | BBHoss | TJNII:that sounds like a very good idea |
| 01:34.31 | tzafrir_home | TJNII, why? What problems have you run into? |
| 01:34.49 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 01:37.19 | *** join/#asterisk macros73_ (n=cs@c-71-61-30-174.hsd1.pa.comcast.net) |
| 01:40.08 | De_Mon | apget and from scratch don't belong in the same sentance |
| 01:40.12 | De_Mon | apt-get |
| 01:44.39 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 01:46.10 | *** join/#asterisk atomicd (n=atomicd@adsl-69-109-58-155.dsl.irvnca.pacbell.net) |
| 01:56.14 | killfill_ | im seen ruby AGI scripts.. and it contains things like "say_text" |
| 01:56.51 | killfill_ | say_text is something that returns to asterisk and somehow is asterisk that render the string to voice? |
| 01:58.59 | killfill_ | the only say to render text->voice is via festival.. isnit? |
| 02:03.54 | *** join/#asterisk techie (n=techie@adsl-76-214-31-136.dsl.lsan03.sbcglobal.net) |
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| 03:31.02 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 03:31.14 | MrTelephone | anyone here going to italy for the openser training? |
| 03:38.18 | peanut- | no |
| 03:52.16 | *** join/#asterisk bmg505 (n=leon@196.209.183.44) |
| 03:53.53 | *** join/#asterisk pitbossy (n=frankjr@adsl-67-115-67-130.dsl.lsan03.pacbell.net) |
| 03:57.54 | MrTelephone | can you get x-lite for windows |
| 03:57.57 | MrTelephone | or is that for windows |
| 03:59.46 | orkid | lol |
| 03:59.54 | orkid | did you read what you said about xlite? |
| 04:00.02 | Nivex | EREDUNDANT |
| 04:02.14 | MrTelephone | hahaha |
| 04:02.15 | *** join/#asterisk stybba (n=stybba@201.204.40.106) |
| 04:02.19 | MrTelephone | yeah |
| 04:02.23 | MrTelephone | excuse me |
| 04:02.25 | MrTelephone | pardon me |
| 04:02.42 | MrTelephone | i thought x-lite was a unix client |
| 04:02.44 | MrTelephone | i have to chekc |
| 04:02.48 | MrTelephone | eyebeam keeps crashing on me |
| 04:03.09 | Nivex | x-lite was originally windows, then mac, then Linux was an afterthought iirc |
| 04:03.17 | MrTelephone | they have some code i bet... if(ondiffcomputers > 15) {crash} |
| 04:04.23 | Nivex | I thought most Windows code was if ( random() ) { crash(); } |
| 04:05.12 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:05.53 | MrTelephone | that too |
| 04:06.07 | MrTelephone | theres a whole case list of if statements that fail on a regular basis |
| 04:06.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:08.28 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:08.32 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:08.35 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
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| 04:11.34 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:12.02 | Mackes | Hey |
| 04:12.07 | coppice | if (crash) {normal_termination();} |
| 04:13.51 | coppice | there are a lot of development tools for windows where this is actually true. almost every time they terminate you get the windows XP "shall I send a bug report" thing pop up |
| 04:15.02 | atomicd | Maybe you should try Vista? |
| 04:17.57 | atomicd | I'm running Asterisk on Vista right now... |
| 04:18.17 | MrTelephone | im selling asterisk for 5 thousnad bucks and going to buy an as5350 |
| 04:18.19 | MrTelephone | <PROTECTED> |
| 04:18.20 | MrTelephone | heh |
| 04:18.39 | MrTelephone | yeah they have to stop making dll's |
| 04:18.43 | MrTelephone | stop with the dll updating |
| 04:18.53 | MrTelephone | atomicd, your full of shit |
| 04:20.02 | atomicd | I'll prove it to you... |
| 04:20.13 | MrTelephone | haha ok i beleive you then |
| 04:20.49 | MrTelephone | i need another gig of ram in this piece of crap |
| 04:22.29 | atomicd | http://www.zonespy.com/forumpics/vista-asterisk.png |
| 04:22.33 | orkid | i've never had more than 512 |
| 04:23.30 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:25.43 | [TK]D-Fender | atomicd, No, you're running a parallel process that is no no real way linked to vista. |
| 04:26.04 | [TK]D-Fender | atomicd, A sad lie which won't support Zaptel |
| 04:26.53 | MrTelephone | is that t38 in callweaver finished? |
| 04:26.58 | coppice | parallel process? he's running it in a multiverse? :-) |
| 04:27.10 | coppice | T.38 is working pretty well |
| 04:27.35 | MrTelephone | if its sitll using rtp g711 why does it work? is there lots of buffering? |
| 04:27.58 | coppice | what is using RTP G.711? |
| 04:28.09 | MrTelephone | t38 fax transmissions? |
| 04:28.27 | coppice | T.38 doesn't use RTP or G.711 |
| 04:28.48 | coppice | well, it could use RTP, but nobody ever does |
| 04:28.55 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 04:29.42 | pitbossy | Another newbie question: When using wancfg_zaptel, configing a sangoma T1 card, it asks a question about clocking. The choices are normal and master. What do the different choices mean? |
| 04:30.04 | MrTelephone | master is your using a channel bank |
| 04:30.10 | MrTelephone | normal if your connected to telco |
| 04:30.11 | ManxPower | pitbossy: normal would be for a span connecting to the telco, master would be for connecting to PBXs, channel banks, etc. |
| 04:30.16 | ManxPower | At least GENERALLY. |
| 04:30.23 | pitbossy | I get it...Thanks |
| 04:30.30 | ManxPower | for T-1/E-1, one side has to be "master" and one end "normal". |
| 04:30.42 | [TK]D-Fender | pitbossy, and fix yout span to 1,1,0 instead of 1,0,0 like you had it before |
| 04:30.50 | pitbossy | The side generating the clock signal is the master... |
| 04:31.24 | ManxPower | pitbossy: I prefer "sync signal", people confuse "clock" with "NTP" or "wall clock". Poor things. |
| 04:32.24 | Un1x_laptop | coppice you using sangoma cards with callweaver? |
| 04:32.33 | coppice | yes |
| 04:32.38 | Un1x_laptop | wich one? |
| 04:32.43 | *** join/#asterisk macros73_ (n=cs@c-71-61-30-174.hsd1.pa.comcast.net) |
| 04:32.45 | pitbossy | Fender: Fixing span? Are you referring to zaptel.conf? wancfg generated the file..I used normal for clock based on a guess. |
| 04:33.02 | coppice | which cards? A200 and A104 |
| 04:33.05 | Un1x_laptop | and are you running a fax machine aswell? |
| 04:33.28 | coppice | I don't run fax machines. I make them :-) |
| 04:33.32 | Un1x_laptop | kik |
| 04:33.34 | Un1x_laptop | lol |
| 04:33.40 | ManxPower | coppice WROTE spandsp and rxfax/txfax, I doubt he's using a "fax machine" |
| 04:33.48 | Un1x_laptop | i know he wrote them |
| 04:33.55 | Un1x_laptop | but why wouldn't he use a fax machine :)? |
| 04:34.03 | Un1x_laptop | thats like saying mark doesn't use asterisk |
| 04:34.05 | ManxPower | *shrug* I found a rock solid way of doing faxes. |
| 04:34.08 | Un1x_laptop | even tho he didn't create all of it |
| 04:34.13 | Un1x_laptop | ManxPower how? |
| 04:34.19 | [TK]D-Fender | pitbossy, yeah if you set it to Normal in wancfg it should generate 1,1,0 instead... then again I never let 3rd party scripts mess with my configs... |
| 04:34.27 | ManxPower | Un1x_laptop: no, that's like saying Mark doesn't use Cisco Call Manager. |
| 04:34.40 | Un1x_laptop | he uses cisco call manager? |
| 04:34.42 | ManxPower | Un1x_laptop: don't run the line thru Asterisk 8-) |
| 04:34.52 | ManxPower | no, he does not. |
| 04:35.05 | Un1x_laptop | anyhow ManxPower know any good t38 providers |
| 04:35.14 | Un1x_laptop | err did providers that support t38 passthrough |
| 04:35.22 | coppice | sangoma has finally got its act together and provided the ability to run fax machines, but with restrictions. you can frequency lock an A104 with hardware EC to an A200, but you can't lock the one without EC. still, at least there is some progress |
| 04:35.46 | pitbossy | Fender: Got everything configed last nice...* is running like a champ on the PRI. |
| 04:35.47 | ManxPower | Un1x_laptop: Every fax machine at every customer at every location has dedicated POTs lines direct from the telco. |
| 04:35.56 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 04:36.06 | ManxPower | Now, we do have rxfax service on their DIDs, but again, those are on PRIs. |
| 04:36.24 | Sunmoon__ | hi |
| 04:36.27 | Un1x_laptop | yes, but for example if one wants a fax phone number in lets say Washington DC |
| 04:36.27 | coppice | Manxpower: this is a sad reflection of just how bad your PBXes are :-) |
| 04:36.31 | Sunmoon__ | I am a novice to asterisk |
| 04:36.33 | Un1x_laptop | and he lived in fl it would be possible |
| 04:36.41 | Un1x_laptop | lol coppice |
| 04:36.42 | pitbossy | Fender: Thanks for your guys help last night! |
| 04:37.02 | MrTelephone | thats dumb that they had to make callweaver seperate just for t38 |
| 04:37.17 | Sunmoon__ | any help on getting started with asterisk would be greatly appreciated |
| 04:37.48 | JT | MrTelephone: you really think that's why callweaver was made? |
| 04:37.50 | ManxPower | coppice: the locations that have hardware EC seem to work fairly well for faxes, but I'd get castrated if they can't send or receive a fax, so I make SURE even if the PBX is dead, they can still send/receive faxes to the main fax machine at the office. |
| 04:37.57 | MrTelephone | thats what one of the posts said |
| 04:38.06 | JT | MrTelephone: must be true then.. |
| 04:38.23 | MrTelephone | i beleive it though because its too good of a feature not to include in asterisk |
| 04:38.30 | MrTelephone | hahha |
| 04:38.33 | MrTelephone | must be true then |
| 04:38.38 | Un1x_laptop | coppice if i choose to get a sangoma card for fax line, i would need to disable EC wouldn't I? or would i need to get them to send me the version without EC if they offer it? |
| 04:38.58 | MrTelephone | un1x, it detects fax and disabled ec |
| 04:38.58 | ManxPower | ANY decent EC should turn itself off when it detects a fas tone. |
| 04:39.00 | JT | MrTelephone: i think you'll find the actual reasons to be different |
| 04:39.03 | ManxPower | anf a fax tone too. |
| 04:39.24 | Un1x_laptop | hrmp i see |
| 04:39.43 | [TK]D-Fender | pitbossy, good to hear. |
| 04:39.48 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 04:41.04 | MrTelephone | coppice, your steve underwood? |
| 04:41.18 | MrTelephone | your famous |
| 04:41.37 | MrTelephone | to me your like a bruce willis |
| 04:41.45 | JT | you're |
| 04:41.57 | [TK]D-Fender | Yup, he sure waited long... |
| 04:42.14 | MrTelephone | mr spandsp |
| 04:42.42 | Un1x_laptop | lol |
| 04:42.55 | Un1x_laptop | whats steve doing in taiwan tho |
| 04:43.00 | Un1x_laptop | whats soo good about taiwan |
| 04:43.10 | Un1x_laptop | i never quite got that lol |
| 04:43.31 | JT | lol |
| 04:44.33 | [TK]D-Fender | Un1x_laptop, the food... |
| 04:44.47 | Un1x_laptop | :| |
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| 04:45.04 | JT | are you sure about the taiwan bit? |
| 04:45.17 | Un1x_laptop | he said he was in ttaipai last time i talked to him |
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| 04:45.48 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 04:46.46 | [TK]D-Fender | sunmoon = fast orbit |
| 04:46.56 | [TK]D-Fender | *boing* |
| 04:47.07 | JT | heh |
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| 04:48.39 | Un1x_laptop | mistake he was visiting |
| 04:48.57 | JT | Un1x_laptop: he was here before |
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| 04:49.04 | MrTelephone | coppice is in hong kong, no wonder why he is into voip |
| 04:49.20 | Un1x_laptop | lol |
| 04:49.27 | JT | Un1x_laptop: thought you meant sunmoon_ |
| 04:49.34 | MrTelephone | he had to build spandsp just to fax the united states |
| 04:49.40 | Un1x_laptop | lol |
| 04:50.56 | MrTelephone | the last of the jt's is on tv.. i mean mohicans :-/ |
| 04:51.17 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:51.34 | [TK]D-Fender | Yay, more traumatic distortions of history... |
| 04:52.51 | MrTelephone | everytime i make a spelling or grammar error he scalps me |
| 04:55.47 | coppice | MrTelephone: wny should being in HK matter? |
| 04:56.02 | MrTelephone | it doesn't |
| 04:56.53 | coppice | I think your VoIP rates are stupid, because I just pick up the phone and dial the US by the incumbant telco for less than your VoIP rates |
| 04:57.55 | MrTelephone | hong kong is probably 3 bucks a minute |
| 04:58.43 | coppice | when I pick up the phone and use the telco to dial the US I pay about 1 cent per minute, for a high quality connection |
| 04:59.08 | *** join/#asterisk chode (n=chode@pD9E8A35D.dip0.t-ipconnect.de) |
| 05:02.44 | [TK]D-Fender | ...that cuts his internet connection just as fast ;) |
| 05:02.49 | [TK]D-Fender | *click* |
| 05:05.01 | MrTelephone | does everyone in hong kong pay 1 cent a minute? |
| 05:06.29 | MrTelephone | you pay more calling in between states |
| 05:07.13 | ManxPower | MrTelephone: telecom in the rest of the world can be very different than that in the USA. |
| 05:07.54 | MrTelephone | how many people have snow blowers here? |
| 05:07.55 | MrTelephone | heh |
| 05:09.15 | MrTelephone | counterpaths website is down |
| 05:09.20 | ManxPower | Historically the USA has had pretty low calling rates and service rates, when the Bell companies were broken up, LD prices started to tumble, in the rest of the world more and more areas are deregulated in the telecom area and advances in technology allows more calls in the sam ebandwidth. |
| 05:10.23 | ManxPower | In the USA, prices are lower than they have ever been, but the local loop is usually the most expensive part of it and those prices, while they have fallen, are still fairly high in many places. |
| 05:12.17 | MrTelephone | more people are chattin on the internet too |
| 05:12.21 | MrTelephone | must be lower call volume |
| 05:12.30 | MrTelephone | send me xlite 3.0 manx :P |
| 05:12.37 | Kobaz | chattin on teh interwebs |
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| 05:13.36 | [TK]D-Fender | MrTelephone, www.xten.com |
| 05:14.47 | Kobaz | my parents had broadvoice for a while |
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| 05:15.06 | Kobaz | terrible service, and it sounded like one guy ran the whole bit |
| 05:15.28 | Kobaz | you called the support number and this one guy would always pick up, no ivr, no call queue, no nothing, just that one guy |
| 05:15.35 | MrTelephone | hahahha |
| 05:15.50 | MrTelephone | yeah it sounds like the guy is getting out of bed |
| 05:16.54 | Kobaz | hmm, xten.com now goes to a scheduled maintenance page |
| 05:16.59 | Kobaz | it was just going to broadvoice for a second there |
| 05:17.17 | MrTelephone | yeah thats what i was sayin |
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| 05:19.18 | MrTelephone | just deleted the counterpatch folder under localsettings/app data |
| 05:19.31 | MrTelephone | and now its loading |
| 05:20.44 | basskozz | Just unlocked my PAP2 v2 for use with my CentPBX, but my "Digit Map" is set for: |
| 05:20.49 | basskozz | *xxT|*1xx|[349]11|1xxx[2-9]xxxxxx|[2-9]xxxxxxT|[2-9]xxxxxxxxx|011x.T |
| 05:20.54 | Kobaz | uhh |
| 05:20.54 | basskozz | is this ok, or should I change it? |
| 05:21.21 | Kobaz | aww mrt went away |
| 05:21.31 | Kobaz | i was going to paste a wiggity comparison page |
| 05:22.21 | [TK]D-Fender | Kobaz, I dunno..... what does your ASTERISK dialplan look like? |
| 05:22.27 | [TK]D-Fender | <rhetoricalquestion> |
| 05:22.38 | [TK]D-Fender | see above |
| 05:22.41 | [TK]D-Fender | </rhetoricalquestion> |
| 05:23.12 | Kobaz | my dialplan looks nice |
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| 05:24.50 | [TK]D-Fender | Kobaz, how incredibly non-descript. I think you should look at the 2 and make sure they are appropriately matched. |
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| 05:24.59 | Kobaz | [TK]D-Fender: ? |
| 05:25.14 | atomicd | I think he's got you confused with basskozz... |
| 05:25.19 | [TK]D-Fender | Kobaz, Sry, bad aim :) |
| 05:25.21 | Kobaz | hehe |
| 05:25.23 | [TK]D-Fender | yup |
| 05:25.38 | [TK]D-Fender | basskozz, yeah, that was for you :) |
| 05:25.55 | Kobaz | i was gonna say |
| 05:26.07 | Kobaz | i was wondering why my dial plan had anything to do with poking fun at broadvoice |
| 05:26.29 | [TK]D-Fender | Kobaz, No, thats a whole schtick by itself ;) |
| 05:27.07 | basskozz | ohh... sorry, I got confused their too :p... What do you mean by Dial Plan ? |
| 05:27.25 | Kobaz | your extensions |
| 05:27.50 | basskozz | ohh, ok.. I've got 2 internal extentions and thats it really (for now ;) |
| 05:28.13 | [TK]D-Fender | basskozz, and does your PAP2 dialplan accomodate them well? |
| 05:31.07 | basskozz | well, I have to dial pound "#" after the extention to go thru, which i guess isn't that big of a deal |
| 05:32.11 | basskozz | but I haven't setup any trunks yet, but once I do I was woundering if this Digit Map would be sufficient |
| 05:32.18 | basskozz | for dialing out |
| 05:32.53 | JT | sounds like the dialplan isn't set up right if you need to press # |
| 05:36.26 | basskozz | JT: any suggestions? |
| 05:41.07 | atomicd | basskozz, the Digit Map just helps you dial things more conviently. Do you understand how it works? |
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| 05:46.08 | basskozz | atomicd: not really, I think I need to do some googling ;) |
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| 05:53.23 | atomicd | basskozz... |
| 05:53.25 | atomicd | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139414817110&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=1711054250B01 |
| 05:54.07 | atomicd | download the ATA admin guide and check out Chapter 3, Configure a Dial Plan... |
| 05:55.41 | basskozz | Thanks atomicd |
| 05:56.01 | atomicd | np |
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| 06:35.45 | CoffeeKid | I need version 1.4.11 of asterisk, I don't want the latest version. Where can i find previous versions to download? |
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| 06:37.08 | CoffeeKid | n/m, i found it! |
| 06:42.27 | CoffeeKid | how does asterisk deal with modules. How does it know what to load? |
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| 06:47.18 | tzafrir_home | http://downloads.digium.com/pub/asterisk/releases/ |
| 06:47.33 | tzafrir_home | http://svn.digium.com/svn/asterisk/tags/ |
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| 06:48.02 | tzafrir_home | CoffeeKid, modules.conf tells asterisk what to load |
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| 07:04.50 | CoffeeKid | tzafrir_home, what about modules like cdr_addon_mysql.so... Those aren't defined in the modules.conf, so how is it loaded? |
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| 07:16.13 | i3inary | Anyone ever have "[ERROR] /usr/libexec/mysqld: Table './mysql/user' happen to them? I fixed it with "find |
| 07:17.03 | i3inary | oops that got cut off "find /var/lib/mysql -name '*.MYI' -exec myisamchk -r {} \;" is what i used to fix the error |
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| 07:17.38 | CoffeeKid | sounds like your mysql.user table was corrupt |
| 07:18.09 | i3inary | yeah...i couldnt start mysqld but after i ran that repair command it starts up no prob now |
| 07:18.26 | i3inary | but now i cant add shit to it...so what are my options |
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| 07:20.43 | CoffeeKid | can't add anything to that table? |
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| 07:22.19 | i3inary | when I try to grant a user privileges i get "ERROR 2013 (HY000): Lost connection to MySQL server during query" |
| 07:23.21 | CoffeeKid | can you get into the mysql console? |
| 07:23.27 | i3inary | yes |
| 07:23.39 | CoffeeKid | do: check table mysql.user |
| 07:24.06 | CoffeeKid | if it comes back with corrupt or any error messages at all, do: repair table mysql.user |
| 07:24.36 | i3inary | ahhh "Table upgrade required. Please do "REPAIR TABLE `user`" to fix it!" im guessing i should listen to it right? |
| 07:25.54 | CoffeeKid | yes |
| 07:25.55 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 07:26.07 | CoffeeKid | then your grant statement should work fine. |
| 07:26.35 | i3inary | damn it fixes it then i try the grant again and it does the same disconnection |
| 07:26.56 | *** join/#asterisk coppice (n=chatzill@awork122012.netvigator.com) |
| 07:27.09 | i3inary | i run check again and i have some new errors |
| 07:29.01 | i3inary | my query is grant all privileges on table to blah_admin@"localhost" identified by 'blah123'; |
| 07:29.09 | i3inary | i think thats what crashing it |
| 07:29.32 | i3inary | i run it and it crashes but since repairing it the 2nd time its coming back ok each time i check it |
| 07:29.54 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 07:29.55 | i3inary | so it looks like the query is just crashing the proc and not corrupting the table now |
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| 07:37.55 | CoffeeKid | hmm, somethings not right |
| 07:38.01 | CoffeeKid | do you have the option of re-installing mysql? |
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| 07:38.17 | i3inary | i suppose so...its production but its my stuff |
| 07:38.32 | CoffeeKid | when did this start happening? when you did a mysql upgrade? |
| 07:39.03 | i3inary | yeah i did recently upgrade and this is prolly the first time since then that i tried to add a user |
| 07:39.35 | CoffeeKid | i'm not 100% sure, but some versions of mysql make you run mysql_install_db after your done |
| 07:39.38 | CoffeeKid | upgrading |
| 07:39.56 | i3inary | i see... i upgraded with yum since its centos |
| 07:40.25 | CoffeeKid | there's another program that i sometimes there, called mysql_update_permissions or something like that |
| 07:40.29 | CoffeeKid | see if thats in your path somewhere. |
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| 07:41.35 | CoffeeKid | you could also check the #mysql channel, i'd hate to tell you to try something that could make things worse.. |
| 07:42.02 | i3inary | yeah im in there...i asked in there nothing yet...no one is talking at all |
| 07:42.13 | i3inary | half hour went by already |
| 07:42.39 | i3inary | i have a dump so if i have to rebuild i can hack it i think |
| 07:43.17 | i3inary | i do have mysql_install_db |
| 07:43.22 | i3inary | not the other |
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| 08:30.06 | BBHoss | whats the best way to dial a bunch of numbers from a sql db? |
| 08:30.14 | BBHoss | i dont want code, just theory :) |
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| 08:35.23 | BBHoss | AGI or manager interface? |
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| 08:38.16 | i3inary | could use .call files |
| 08:38.38 | i3inary | script queries the db then writes the .call files and the calls will be made |
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| 08:39.00 | JT | manager interface or call files |
| 08:39.11 | JT | agi deals with existing files? |
| 08:39.50 | BBHoss | .call files? |
| 08:40.51 | BBHoss | ahh i see |
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| 08:52.17 | CoffeeKid | BBHoss, i've done something like this. Let me know if you need some help. Creating the .call files is very easy and you can do some powerful stuff. |
| 08:52.19 | JT | BBHoss: sorry |
| 08:52.24 | JT | AGI deals with existing CALLS |
| 08:52.26 | JT | brainfart |
| 08:52.50 | CoffeeKid | right, but you can use AGI within the .call file to control the call however you'd like to |
| 08:53.37 | CoffeeKid | very cool actually :) |
| 08:54.16 | BBHoss | CoffeeKid: im trying some examples on the wiki, and its telling me unknown keywork from the console |
| 08:54.30 | CoffeeKid | what does your call file look like? |
| 08:54.47 | CoffeeKid | and, what are you trying to do? |
| 08:54.51 | BBHoss | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Example1 |
| 08:55.02 | BBHoss | example 2 except with SIP instead of zap |
| 08:56.01 | BBHoss | im on 1.4.13 |
| 08:56.15 | CoffeeKid | are you using a Context: that exists in extensions.conf? |
| 08:56.35 | BBHoss | i believe so, its what my hardphone has registered |
| 08:56.39 | BBHoss | ael-demo |
| 08:56.47 | BBHoss | i can try default though |
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| 08:57.02 | CoffeeKid | k, give that a try |
| 08:57.08 | BBHoss | its like its not understanding though |
| 08:57.16 | BBHoss | Unknown Keyword ... |
| 08:57.34 | BBHoss | apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/sample.call |
| 08:57.40 | CoffeeKid | let me see exactly what your call file looks like (you can xxx out the user/pass stuff of course)... send it to me in a private message |
| 08:57.47 | BBHoss | k |
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| 09:26.49 | mvanbaak | I noticed :) |
| 09:27.51 | tzafrir | It's a pain to try to get from them a simple reply for so long |
| 09:33.51 | mvanbaak | yeah |
| 09:34.19 | mvanbaak | lucky me they are way more responsive when you contact support if you have a problem with one of their products |
| 09:35.08 | mvanbaak | hhmm, my house is a mess |
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| 10:18.08 | tzafrir | chalow looks like a nice tool for changelog rendering |
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| 10:22.59 | tzafrir | played with it a bit. I don't have time to upload the results now, but check http://rapid.tzafrir.org.il/~tzafrir/chalow_asterisk_cl.conf |
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| 10:25.52 | axscode | hi, anyone can recommend a model of cisco sip gateway? |
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| 11:03.24 | tzafrir | axscode, any cisco rep can recommend you of one. Why do you insist on cisco? |
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| 11:27.31 | Greek-Boy | i wonder if Fring works ok with asterisk |
| 11:27.44 | Greek-Boy | and if it makes a direct connection or goes through fring servers |
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| 12:53.23 | Mavvie | In the dialplan, once in a macro, is it still possible to find out which context the call came in through? |
| 12:56.24 | tzafrir | ${ORIG_CONTEXT} or something similar |
| 12:57.00 | tzafrir | this is the context that called the macro. Not necessarily where the call came through, naturally |
| 12:57.01 | Mavvie | nope. |
| 12:57.10 | tzafrir | show application macro? |
| 12:57.49 | Mavvie | MACRO_CONTEXT |
| 12:59.02 | Mavvie | this is going to be messy.... |
| 12:59.14 | Mavvie | because my macros are two or three calls deep. |
| 12:59.24 | Mavvie | let's see how I can fix this... |
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| 13:03.19 | *** join/#asterisk brainy_ (n=brainy@blub.tnib.de) |
| 13:03.24 | brainy_ | hi |
| 13:04.31 | brainy_ | I have a problem with a cisco sipgateway and asterisk .. call comes in from the Cisco SIP Gateway (Line Carrier) ... our asterisk answers and plays MOH, but the MOH has outages (like packetloss) but when i answer the call i have no problems... |
| 13:05.20 | brainy_ | i'm not sure what can cause this problem |
| 13:11.30 | puzzled | brainy_, if you don't have a digium card in that box, load the ztdummy driver |
| 13:12.25 | brainy_ | i have other SIP clients connected to that asterisk box .. they have no problems |
| 13:12.35 | brainy_ | and the ztdummy is already loaded for conferencing |
| 13:12.51 | puzzled | brainy_, afaik MoH needs a timing device to work ok, hence the ztdummy driver if you don't load any other zaptel drivers |
| 13:13.17 | puzzled | brainy_, using mpg123? |
| 13:13.22 | brainy_ | no, madplay |
| 13:13.28 | JT | brainy_: ztdummy doesn't work as well as real zap timing |
| 13:13.28 | brainy_ | mpg123 has caused to many problems on freebsd |
| 13:13.45 | brainy_ | but why do i ONLY have this problem with the cisco sip gateway? |
| 13:14.09 | brainy_ | i have Snom phones, AVM FritzBoxes, patton/inalp GW's connected to the asterisk.. everything is fine |
| 13:14.12 | puzzled | brainy_, try using an MoH file converted to wav/gsm/ulaw/alaw or whatever the codec format that you use |
| 13:14.57 | puzzled | brainy_, if the rest is ok I assume it's a funky setting on the cisco that causes it |
| 13:15.01 | brainy_ | k, i will try that |
| 13:17.31 | tzafrir | zttest checks the zaptel timing source (regardless of its exact source |
| 13:17.32 | tzafrir | ) |
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| 13:19.45 | brainy_ | puzzled: i converted it to alaw and it's still the same problem |
| 13:19.55 | Mavvie | WARNING[11173]: ast_expr2.y:742 op_minus: non-numeric argument |
| 13:19.57 | Mavvie | woops |
| 13:20.01 | Mavvie | nothing to see here. |
| 13:20.31 | *** join/#asterisk simonkern (n=simonker@p57B03EC4.dip0.t-ipconnect.de) |
| 13:20.39 | simonkern | hi |
| 13:20.57 | brainy_ | Opened pseudo zap interface, measuring accuracy... |
| 13:20.57 | brainy_ | 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% |
| 13:20.57 | brainy_ | 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.914551% |
| 13:21.05 | brainy_ | i _think_ that this is ok? |
| 13:21.11 | tzafrir | yes, it is |
| 13:21.27 | simonkern | I have a one-way audio problem between my asterisk and outgoing calls |
| 13:21.40 | brainy_ | well.. i will blame the cisco SIP gateway on that issue since everything else is working fine |
| 13:22.13 | brainy_ | simonkern: using NAT? |
| 13:22.24 | simonkern | brainy_: yes |
| 13:22.58 | brainy_ | simonkern: is the asterisk behind nat or your telephone? |
| 13:23.08 | simonkern | i've forwarded 5004-5080 tcp/udp and 10000-10100 tcp/udp |
| 13:23.36 | simonkern | the asterisk is behind the nat and i'm trying to make a call trough my sip provider |
| 13:23.57 | JT | brainy_: 99.914% is NOT okay |
| 13:24.12 | JT | simonkern: why forward tcp? |
| 13:24.17 | brainy_ | simonkern: did you change rtp.conf to that ports also? |
| 13:24.24 | simonkern | my asterisk sends the rtp stream to the provider, but the called person sends the rtp directly to me |
| 13:24.37 | simonkern | yes, i've changed the rtp.conf |
| 13:24.40 | brainy_ | simonkern: you will need 5060/udp and the udp RTP ports (from rtp.conf) |
| 13:24.46 | JT | simonkern: and why the hell did you forward all those useless ports near 5060? |
| 13:25.10 | brainy_ | JT: it is NOT ok? .. |
| 13:25.17 | brainy_ | --- Results after 27 passes --- |
| 13:25.17 | brainy_ | Best: 99.987793 -- Worst: 96.887207 -- Average: 99.870244 |
| 13:25.17 | JT | brainy_: yes, too low |
| 13:25.24 | JT | anything <99.97% is NOT ok |
| 13:25.35 | simonkern | hmm... i didn't know what to do, so i forwarded nearly everything |
| 13:25.38 | JT | that's an awful average |
| 13:25.41 | JT | and awful low score |
| 13:25.45 | JT | no wonder you have issues |
| 13:25.55 | brainy_ | JT: i think that there were always problems with that ztdummy on a freebsd, at least i already had a lot of problems |
| 13:26.18 | JT | brainy_: yeah, asterisk seems to be designed to only work properly on linux |
| 13:26.22 | brainy_ | JT: but that doesn't explain why ONLY this cisco SIP gw causing problems |
| 13:26.28 | JT | simonkern: well all the ports are very well documented |
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| 13:27.02 | JT | simonkern: you haven't even filled us in on the call scenario |
| 13:27.41 | puzzled | brainy_, dunno if that's an option but Fedora 7 and 8 have high resolution timer support in their kernel. Quite and improvement to non HRT kernels |
| 13:28.29 | coppice | hormone replacement therapy in a kernel? :-\ |
| 13:29.03 | simonkern | ok, here is the call scenario: A calls B through sip server S. S tells A that the phone is ringing. B is answering the call. B sends its rtp stream to A. A sends its rtp stream to S. |
| 13:29.11 | puzzled | coppice: man I'm slow. Initially I had no clue what you were referring too :) |
| 13:29.18 | simonkern | A can hear B, but B can't hear A |
| 13:29.20 | JT | simonkern: too many letters |
| 13:29.23 | JT | try this |
| 13:29.35 | JT | "i am trying to use a SIP ITSP on the INTERNET from BEHIND NAT" |
| 13:29.38 | JT | or similar |
| 13:29.40 | JT | not ab s |
| 13:29.44 | simonkern | ok |
| 13:30.07 | JT | i don't know if your asterisk is trying to act as sip server to the Internet or what |
| 13:30.09 | coppice | puzzled: hey, its Sunday. you can be a little slower today :-) |
| 13:30.51 | mvanbaak | it's sunday, and you are on irc. you can get a life for almost nothing at k-mart |
| 13:30.53 | puzzled | yup, I'll take that one gladly |
| 13:31.03 | JT | it's monday |
| 13:31.31 | puzzled | mvanbaak, but we don't have a k-mart here. now what? :) |
| 13:31.46 | mvanbaak | albert heijn ? |
| 13:32.10 | puzzled | heh too obvious |
| 13:32.28 | mvanbaak | dont buy one from lidl, those are cheap fakes |
| 13:32.29 | puzzled | I know, I'm going to upgrade my 7961. that should take care of the entire sunday |
| 13:32.53 | mvanbaak | puzzled: sip or skinny ? |
| 13:33.21 | puzzled | from factory default sccp to sip |
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| 13:33.39 | brainy_ | simonkern: huh, why does A send the rtp to S and not to B? |
| 13:33.45 | mvanbaak | ah. I run my 7960 and 7905 with skinny |
| 13:33.55 | puzzled | I'll leave the other one at sccp so I can play with chan_skinny and chan_sccp |
| 13:33.56 | simonkern | brainy_: i have no idea |
| 13:34.06 | JT | simonkern: what is the scenario? i'm still waiting |
| 13:34.08 | mvanbaak | chan_sccp is dead |
| 13:34.32 | puzzled | nope, next week there will be a new release that compiles against 1.2 & 1.4 |
| 13:34.37 | simonkern | my ASTERISK is in my LOCAL NET its BEHIND NAT. SIP PHONE is in the LOCAL NET. ASTERISK connected via SIP to a PROVIDER. i want to call a person at the same provider with my SIP PHONE. |
| 13:34.49 | puzzled | I'm using the 20071004 release right now on 1.4 |
| 13:34.51 | mvanbaak | puzzled: where did you get that info ? |
| 13:34.58 | JT | simonkern: there is no need to port forward then |
| 13:35.15 | puzzled | mvanbaak, https://lists.berlios.de/mailman/listinfo/chan-sccp-users |
| 13:36.20 | brainy_ | simonkern: do you have canreinvite=yes? |
| 13:36.40 | simonkern | brainy_: no, i have canreinvite=no |
| 13:36.46 | Hadi- | hi everyone |
| 13:36.48 | Hadi- | just a quick question... we have a SIP PRI connected directly to our CISCO 2800 series router... we are sending some outgoing calls from asterisk to the Cisco 2800 series.. ans we are getting a lot of Got SIP response 486 "Busy here" |
| 13:36.50 | brainy_ | simonkern: try setting it to yes |
| 13:37.07 | JT | simonkern: port forwardning is only to make asterisk act as a sip server on the Internet from behind NAT |
| 13:37.11 | JT | brainy_: wtf, why? |
| 13:37.43 | brainy_ | JT: his asterisk should send the rtp directly to the phone B and NOT to the Server S? |
| 13:37.52 | JT | brainy_: Incorrect. |
| 13:37.59 | brainy_ | oh? |
| 13:38.02 | brainy_ | sorry |
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| 13:38.04 | JT | rtp should go from ITSP to asterisk to phone |
| 13:38.08 | JT | canreinvite=no |
| 13:38.25 | JT | his phone and ITSP cannot communicate directly |
| 13:38.30 | mvanbaak | hhmm, they are forking chan_sccp again ? |
| 13:38.36 | JT | unless the phone registers direct to the ITSP |
| 13:39.28 | *** join/#asterisk macros73_ (n=cs@c-71-61-30-174.hsd1.pa.comcast.net) |
| 13:39.29 | JT | remember, his phone is behind nat |
| 13:40.03 | simonkern | this is the sip config entry for the provider: de.pastebin.ca/769477 |
| 13:40.48 | tzafrir | Hadi-, SIP or PRI? |
| 13:41.05 | JT | simonkern: where's the register entry? |
| 13:41.29 | tzafrir | SIP to a SIP/PRI gateway? |
| 13:41.38 | simonkern | i don't register me on the server, i only make outgoing calls |
| 13:41.48 | JT | simonkern: you MUST register |
| 13:41.53 | JT | and set qualify=yes |
| 13:41.56 | JT | and type=friend |
| 13:41.59 | JT | you are behind nat |
| 13:42.09 | JT | it needs to maintain a mapping on your nat device |
| 13:42.15 | JT | otherwise audio will not pass |
| 13:42.19 | simonkern | ok, i'll try this! |
| 13:45.08 | simonkern | it's the same problem |
| 13:45.24 | Hadi- | tzafrir: its a SIP PRI |
| 13:45.58 | JT | simonkern: which way is audio working? |
| 13:46.03 | JT | Hadi-: hahahahahah haha |
| 13:46.15 | JT | and faeries exist too... |
| 13:46.21 | simonkern | from the called person to me |
| 13:46.33 | JT | simonkern: is registration successful |
| 13:46.36 | JT | is it qualifying? |
| 13:46.40 | JT | what is your nat device? |
| 13:46.56 | simonkern | registration is successful |
| 13:47.04 | Hadi- | JT: ? |
| 13:47.14 | JT | Hadi-: there is no such thing as a SIP PRI. |
| 13:47.25 | simonkern | qualifying is also successful |
| 13:47.54 | simonkern | i use iptables |
| 13:48.02 | JT | simonkern: also what phone do you have? |
| 13:48.02 | Hadi- | JT: you think so? |
| 13:48.15 | JT | simonkern: get rid of all the port forwarding btw |
| 13:48.19 | JT | Hadi-: i know so |
| 13:48.26 | tzafrir | Hadi-, Is your Asterisk a SIP<->PRI gateway? |
| 13:48.32 | JT | simonkern: just make sure there's not firewalling |
| 13:48.43 | tzafrir | Or do you connect to one (of Cisco, or whatever) |
| 13:48.46 | tzafrir | ? |
| 13:49.01 | Hadi- | tzafrir: Asterisk -> Cisco 2811 -> Virtual SIP PRI |
| 13:49.14 | JT | in other words just SIP |
| 13:49.23 | Hadi- | correct |
| 13:49.28 | simonkern | i use a fritzbox with connected analog phones |
| 13:49.49 | JT | simonkern: have you deleted all the port forwards to the asterisk box? |
| 13:50.16 | simonkern | it's running on the same pc, |
| 13:50.33 | JT | so it has a public ip? |
| 13:50.40 | simonkern | yes |
| 13:51.12 | JT | simonkern: why did you have port forwarding then? |
| 13:52.04 | simonkern | forwarding is the wrong word.. i have a rule, so that the asterisk ports are not firewallt |
| 13:52.29 | JT | try this then for testing |
| 13:52.42 | JT | iptables -I INPUT 1 -j ACCEPT |
| 13:52.46 | JT | iptables -I FORWARD 1 -j ACCEPT |
| 13:52.51 | JT | iptables -I OUTPUT 1 -j ACCEPT |
| 13:54.53 | Hadi- | tzafrir: any suggestions as to why calls are being rejected |
| 13:55.15 | simonkern | still the same problem |
| 13:55.32 | JT | simonkern: please share sip.conf for the phone's entry and general |
| 13:55.47 | simonkern | ok |
| 13:55.50 | tzafrir | Hadi-, no. But I guess you should pastebin some relevant information |
| 13:55.52 | JT | also, perhaps try with a softphone instead |
| 13:56.00 | JT | it might be a problem with the fritzbox |
| 13:56.15 | tzafrir | e.g: relevant sip.conf snippets . Maybe a trace of what happens there |
| 13:57.38 | Mavvie | I believe that GotoIf(.... ? label,step) doesn't work. |
| 13:57.59 | Mavvie | :W |
| 13:58.01 | Mavvie | woops |
| 13:58.30 | tzafrir | :wq |
| 13:58.31 | tzafrir | ? |
| 13:59.03 | Hadi- | http://www.pastebin.ca/769492 |
| 13:59.09 | Hadi- | # |
| 13:59.09 | Hadi- | [IW001] |
| 13:59.25 | Hadi- | is for the Cisco 2811 |
| 13:59.34 | Hadi- | the V3 is the server sending the call |
| 14:00.25 | brainy_ | simonkern: are you using the fritzbox as router? |
| 14:00.30 | simonkern | no |
| 14:00.35 | tzafrir | BTW: are you sure about "dtmfmode=inband"? |
| 14:00.35 | simonkern | only as ata |
| 14:00.36 | Hadi- | the call is coming from <customer - V3> -> <asterisk> -> <Cisco 2811 - IW001> |
| 14:00.52 | brainy_ | simonkern: ok.. because you can't forward port 5060 via a fritzbox ;) |
| 14:00.56 | Hadi- | tzafrir: yes thats the only DTMF that works for me |
| 14:02.01 | tzafrir | So what is the exact error you get? |
| 14:02.22 | simonkern | http://de.pastebin.ca/769500 |
| 14:02.26 | Hadi- | lots of Got SIP response 486 "Busy here" |
| 14:03.14 | tzafrir | Hadi-, maybe you send the wrong number? |
| 14:03.55 | tzafrir | next thing would probably be to post a trace from a sip debug |
| 14:04.01 | tzafrir | sip set debug |
| 14:04.07 | tzafrir | call |
| 14:04.12 | simonkern | sorry, on the pastebin the qualify entry for the provider is still no |
| 14:04.35 | JT | simonkern: is it yes or not? |
| 14:04.41 | simonkern | its yes |
| 14:04.45 | JT | ok |
| 14:06.11 | JT | simonkern: interesting you're only allowing ulaw from the provider and you are in germany |
| 14:06.49 | simonkern | i can try alaw too, but ulaw is working |
| 14:07.18 | JT | well alaw is the native codec in germany |
| 14:07.26 | JT | ulaw is used in north america and japan |
| 14:07.57 | coppice | and hong kong and taiwan |
| 14:08.01 | simonkern | i've added allow=alaw |
| 14:08.38 | JT | coppice: weirdos :P |
| 14:08.47 | tzafrir | What about Japan? |
| 14:09.11 | tzafrir | oUtlaws? |
| 14:09.21 | JT | they're just insane |
| 14:09.26 | JT | they made a J1 |
| 14:09.27 | coppice | well, taiwan uses ulaw because japan uses it. how a british colony like HK came to use ulaw is more of a mystery |
| 14:09.54 | simonkern | when person b call me, everything is ok |
| 14:10.14 | *** join/#asterisk nighty__ (n=nighty@sushi.rural-networks.com) |
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| 14:14.12 | simonkern | OMG, i've used an other sip phone, and boom... it works |
| 14:14.37 | JT | didn't i suggest that about half an hour ago? ;) |
| 14:14.47 | simonkern | JT: you're right, the fritzbox causes this problem |
| 14:15.02 | JT | some devices are just defective :( |
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| 14:16.02 | coppice | J1 was not insane. it was a specific attempt at incompatibility to lock foreigners out before Japan was strong. Taiwan used J1 too, but called it T1M (T1 modified, or possibly mangled). |
| 14:19.44 | simonkern | ok thanks for help! |
| 14:19.54 | simonkern | cu |
| 14:30.47 | puzzled | JT, how true. this morning I spent some time figuring out why sip was not working. Appears the "sip helper" module in a SpeedTouch 716 adsl modem is not helping at all. Fortunately disabling the "helper" module fixed the issue |
| 14:31.11 | puzzled | sorta like the smtp fixup thingy in Cisco |
| 14:32.22 | JT | and cisco's sip fixup |
| 14:32.33 | JT | most fixups for sip seem to break it |
| 14:32.40 | puzzled | makes you wonder if they actually test that crap |
| 14:32.46 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 14:33.05 | JT | yes, but i think the test involves using complete nat unaware UAs and servers |
| 14:36.21 | puzzled | heh sure seems like it |
| 14:39.36 | brainy_ | byebye |
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| 14:48.02 | Hadi- | Can you guys recommand a good voip billing / accounting (termination and wholesale) software that supports both Asterisk and Cisco |
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| 15:36.31 | fly | hi alll |
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| 16:58.29 | riddlebox | do you need an Answer() before you have Voicemail() pick up? |
| 16:59.03 | russellb | Voicemail will answer for you |
| 16:59.14 | russellb | but sometimes it is beneficial to Answer directly |
| 16:59.20 | russellb | Answer() and then a Wait(1) |
| 16:59.25 | russellb | to ensure the audio path gets set up |
| 16:59.36 | russellb | otherwise, sometimes you get the initial audio prompts cut off at the beginning |
| 16:59.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:59.46 | [TK]D-Fender | SHHH!!!! You'll disturb the crickets! |
| 17:00.07 | russellb | sorry :( |
| 17:00.31 | riddlebox | well I have it setup right now to Answer, but I displayed my extensions.conf and someone said that I didnt need Answer in it? |
| 17:01.12 | russellb | well like i said, it's not necessary, but it doesn't hurt. and sometimes it's good to have. |
| 17:01.14 | [TK]D-Fender | riddlebox, Technically no, but you've just heard a reason for it. |
| 17:02.02 | riddlebox | as long as I am not doing anything that is looked at as wrong by having it in the dialplan |
| 17:02.42 | riddlebox | btw have you guys seen this yet, http://taa.com/amanda_products.html |
| 17:04.19 | russellb | yet another gui? how exciting, heh |
| 17:04.43 | [TK]D-Fender | russellb, You know what this means... time to update the topic ;) |
| 17:05.23 | russellb | crap, you're right |
| 17:05.23 | riddlebox | I know, I sat in on a webinar on thursday, if you want to buy their version of asterisk on a usb key, $895 |
| 17:05.42 | russellb | i'm sure asterisk itself is no different |
| 17:06.39 | Nivex | funny, I've been running Asterisk for 3 years and haven't needed a GUI |
| 17:06.41 | *** join/#asterisk mtgll (n=mtg@ool-18599c5b.static.optonline.net) |
| 17:06.59 | riddlebox | they do add things in it to integrate to their portal system, and they will strip out the voicemail and put their voicemail in |
| 17:07.42 | riddlebox | Nivex, I havent needed one and quite frankly the gui's dont seem to allow you to do half of the stuff you can do editing the conf files |
| 17:08.23 | mtgll | need some help with a tdm400 card have two fxo modules not beeing seen two fxs are being seen fine where do i start never had a problem with the modules before? |
| 17:08.28 | Corydon76-dig | and they never will... |
| 17:08.39 | [TK]D-Fender | riddlebox, Of course not. Editing the files allows you to do ANYTHING. Forget HALF, its not even a reasonable fraction. |
| 17:08.43 | aiksa[LV] | good evening everybody |
| 17:09.08 | russellb | of course, they don't intend to do everything ... |
| 17:09.10 | Corydon76-dig | mtgll: please call Digium support tomorrow morning |
| 17:09.25 | aiksa[LV] | I wanted to know where could I have some additional reading on 'data' passed to originate command through ami? |
| 17:09.35 | Corydon76-dig | mtgll: or if you bought from a reseller, please contact your reseller for support |
| 17:09.42 | russellb | the whole point is to make a defined system ... |
| 17:10.21 | mtgll | can do but thought i would ask here first have checked the conf files and all is good seems weird... |
| 17:10.45 | [TK]D-Fender | russellb, you and your silly definitions and borders! Telecom fascist! :p |
| 17:11.13 | russellb | :) |
| 17:11.20 | aiksa[LV] | is that 'data' available from dialplan afterwards? |
| 17:11.25 | [TK]D-Fender | mtgll, pastebin your configs, and the output of "ztcfg -vvvv" and "dmesg" |
| 17:11.26 | [TK]D-Fender | ~pb |
| 17:11.27 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:11.27 | *** join/#asterisk twoshadetod (n=clay@c-76-123-96-239.hsd1.fl.comcast.net) |
| 17:11.36 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:11.55 | [TK]D-Fender | aiksa[LV], What kind of "data"? |
| 17:12.27 | Corydon76-dig | Yes, but the reason why the GUI doesn't do as much is also technical... it's very difficult to abstract all implementation details into a GUI in a way that is understandable to the end user. |
| 17:12.47 | aiksa[LV] | according to documentation of starpy (AMI interface for twisted matrix framework), originate command from AMI has an additional incomming parameter data |
| 17:13.03 | Corydon76-dig | Even many of the GUIs that exist don't do a good job of the abstraction |
| 17:13.23 | Corydon76-dig | ...for the concepts that they DO implement |
| 17:13.51 | [TK]D-Fender | aiksa[LV], You can set channel variables in your originate, yes. |
| 17:14.26 | aiksa[LV] | ok. that what I wnated to find out |
| 17:14.44 | aiksa[LV] | what is the best source for ami documentation? asterisk source? |
| 17:18.09 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 17:18.19 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:18.21 | aiksa[LV] | [TK]D-Fender: although the documentation within starpy regarding that 'data' is as plain as possible. <cite>data -- data to pass to application</cite> |
| 17:19.15 | [TK]D-Fender | aiksa[LV], Oh I think thats basically just the CLI parameters when you use originate to dump the other end into an APPLICATION, and not a channel. |
| 17:20.21 | hi365 | anyone having a problem with chanspy? Im trying to only hear extensions that are on a call. using the folowing most extensions dont come on when i press * : Chanspy(SIP|b${w}) |
| 17:22.57 | aiksa[LV] | [TK]D-Fender: just understood. - thats data for the application (found it in manager documentation) |
| 17:23.26 | [TK]D-Fender | aiksa[LV], so just think of it as a fully crafted single line of dialplan. |
| 17:23.30 | mtgll | [TK]D-Fender here is the pastebin http://pastebin.com/d7250a56d |
| 17:25.11 | aiksa[LV] | [TK]D-Fender: that was not what I was trying to achieve, though. Just digged deeper into the sources of starpy manager proxy, just to find out that their online class documentation is "slightly" outofdate |
| 17:25.25 | [TK]D-Fender | mtgll, where you you experience this "not being seen" bit? |
| 17:25.54 | aiksa[LV] | their Originate now takes at least twice the parameters as given in the API documenation |
| 17:26.29 | *** join/#asterisk WindBack (n=Administ@host60.190-138-93.telecom.net.ar) |
| 17:26.51 | WindBack | I have a doubt.. |
| 17:26.51 | mtgll | when i load asterisk it only shows channel 3 and 4 under zap show channels |
| 17:27.02 | WindBack | I I have a client behind NAT |
| 17:27.28 | mtgll | and as you can see they are defined in the conf files. |
| 17:27.30 | [TK]D-Fender | mtgll, pastebin a "reload chan_zap.so" |
| 17:28.14 | [TK]D-Fender | mtgll, and "zap show channels before & after |
| 17:28.37 | hi365 | nobody using chanspy in 1.4? |
| 17:28.53 | *** join/#asterisk ManxPower (n=manxpowe@180.sub-70-221-75.myvzw.com) |
| 17:29.07 | aiksa[LV] | although on the whole I am pretty happy with starpy. Anyone else here using it? |
| 17:29.30 | WindBack | I don't undertand how can a client behind nat work, if in the firewall the any port is forwarded for that client?? |
| 17:29.33 | [TK]D-Fender | hi365, perhaps you you could pastebin something useful in showing your problem. |
| 17:29.45 | [TK]D-Fender | aiksa[LV], I'm here all the time and have never heard of it. |
| 17:30.03 | aiksa[LV] | that doesnt sound good :P |
| 17:30.14 | [TK]D-Fender | WindBack, you don't need to forward ports for any sane client behind NAT. Read up : |
| 17:30.15 | [TK]D-Fender | ~sipnat |
| 17:30.16 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:30.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:30.23 | aiksa[LV] | there is a framework for python: twisted matrix |
| 17:30.50 | aiksa[LV] | async, event based for writing internet apps |
| 17:30.59 | WindBack | [TK]D-Fender, but an UAS need the 5060 port to recive INVITE messages, or not? |
| 17:31.12 | ManxPower | WindBack: you don't understand NAT. |
| 17:31.33 | aiksa[LV] | starpy is a lib for twisted matrix consisting of protocol factory for AMI and FastAGI |
| 17:31.45 | [TK]D-Fender | WindBack, No. Read the guide, FOLLOW it and then come back and show us what you've done. |
| 17:31.59 | ManxPower | When a NAT router has a packet coming from the inside, going to the outside, it translates all the inside address information in the packet to the outside IP, then has a table to keep track of that translation for all the response packets. |
| 17:31.59 | tzafrir | If channel 3 is defined, then it can't be a matter of chan_zap breaking in the middle |
| 17:32.07 | tzafrir | Not with the current config file |
| 17:32.23 | aiksa[LV] | basicaly with it I have an high level access to the asterisk, from the env. where I can easily create xmlrpc and soap services and clients. |
| 17:32.26 | hi365 | [TK]D-Fender: im not really sure what ot pastebin - show channles hsows sip channels being used, but chanspy doesnt cycle thru them when i press * |
| 17:32.37 | ManxPower | It really is pretty much the same as using a web browser behind NAT or DNS behind NAT. |
| 17:32.40 | tzafrir | mtgll, maybe you just reloaded? try 'zap restart' or just restart asterisk |
| 17:33.00 | hi365 | with 30 users on the system, its kinda hard to find something useful to pastebin (unless you have a something specfic) |
| 17:33.19 | [TK]D-Fender | hi365, go prove your DTMF is fine and pastebin SOMETHING to see... |
| 17:33.50 | mtgll | [TK]D-Fender here you go http://pastebin.com/deca5402 |
| 17:34.42 | WindBack | ManxPower, yes, I understand that, but If I'm outside, and I want to create a new conection in the 5060 port of the client i can't, because the reouter/nat don't have the table created |
| 17:34.59 | [TK]D-Fender | mtgll, Ok, I completely don't get it... |
| 17:35.27 | [TK]D-Fender | WindBack, You don't need it. Now go foolow the guide, try stuff out, adn then come back and show us your configs |
| 17:35.48 | tzafrir | mtgll, this doesn't make sense. Why is zapata_additional.conf parsed twice (that is: #include-d twice) |
| 17:35.50 | WindBack | [TK]D-Fender, ok, thanks |
| 17:36.05 | tzafrir | we're looking at the wrong config file |
| 17:36.35 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 17:36.40 | ManxPower | WindBack: The existing connections on 5060 are used, not new ones. |
| 17:36.48 | tzafrir | grep '#include' /etc/asterisk/zapata*.conf |
| 17:36.51 | hi365 | [TK]D-Fender: doubt youll find this usefull, but then again who knows: http://pastebin.ca/769762 |
| 17:37.15 | ManxPower | WindBack: you can either argue or you can follow [TK]D-Fender's advice. Only one of those two courses of action will get your problem fixed. Can you guess which one? |
| 17:37.41 | [TK]D-Fender | &^@#%@&#% FreePBX |
| 17:37.56 | [TK]D-Fender | hi365, And why show me 10x the crap I DON'T need? |
| 17:37.57 | tzafrir | [TK]D-Fender, surely not |
| 17:38.06 | ManxPower | [TK]D-Fender: Looks like the alligators will have a good meal today. |
| 17:38.12 | mtgll | [TK]D-Fender no that was a mistake i am taking it out now dont know how it got there user error :) |
| 17:38.13 | [TK]D-Fender | tzafrir :that was for hi365 |
| 17:38.24 | tzafrir | [TK]D-Fender, noticed that the FXS channels are not zapata_additional? |
| 17:38.27 | aiksa[LV] | [TK]D-Fender: thanks, variables was the part of originate which I needed :))) |
| 17:38.31 | hi365 | like i said, i doubt youl find anyhthing usefull there |
| 17:38.32 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 17:38.53 | tzafrir | Which means that if mtgll uses freepbx, he probably won't be able to dial from those devices |
| 17:38.58 | [TK]D-Fender | tzafrir : # |
| 17:38.59 | [TK]D-Fender | #include zapata-channels.conf <--- |
| 17:39.01 | aiksa[LV] | now i can set this parameter wfrom originate, and then get back to it and find calling class from agi service |
| 17:39.05 | ManxPower | hi365: if you turned off degugging..... |
| 17:39.05 | [TK]D-Fender | tzafrir : seems fine there. |
| 17:39.31 | [TK]D-Fender | hi365, turn down the verbose and debug and just show me that CAHNNELS are up and you executing chanspy. |
| 17:39.52 | hi365 | thats easy, why dont you say so? ;-) |
| 17:39.53 | tzafrir | But zapata.conf does not #include zapata_additional.conf before zapata-channels.conf |
| 17:40.01 | ManxPower | [TK]D-Fender: what issue is hi365 having other than having too much debugging turned on? |
| 17:40.32 | ManxPower | hi365: for your future reference, the standard info to paste is verbose set to 3 and no debugging. |
| 17:40.33 | [TK]D-Fender | ManxPower, Says he can't cycle through channels pressing "*". First guess , DTMF issue. |
| 17:40.51 | ManxPower | [TK]D-Fender: Ah. or an incorrect calling of ChanSpy. |
| 17:41.13 | ManxPower | You'd think he'd have IVR issues as well as voicemail issues if that was the case. |
| 17:41.19 | [TK]D-Fender | ManxPower, possibly. would be nice if I didn't have to go crazy looking for it. |
| 17:41.31 | tzafrir | DTMF detection can also be traced using DTMF logging |
| 17:41.31 | WindBack | ManxPower, My clients work well, I only want to understand that concepts |
| 17:41.38 | hi365 | actualy i said "most extensions dont come on when i press *" some do - but not all. so abviously no dmtf issue here |
| 17:41.48 | ManxPower | [TK]D-Fender: *nod* Also would be nice if he created an extension that ONLY does the MIN required to reproduce the problem. |
| 17:41.49 | hi365 | and yes: ivr is fine vm is fine disa is fine |
| 17:42.03 | mtgll | thanks all that worked i beeter go get some glases :) |
| 17:42.03 | tzafrir | mtgll, any news? |
| 17:42.04 | *** join/#asterisk debiano777 (i=debiano@217.201.4.182) |
| 17:42.15 | ManxPower | hi365: DTMF issues are not always "works" or "doesn't work", many times it is "works 80% of the time" |
| 17:42.21 | mtgll | and learn how to spell :) |
| 17:42.22 | tzafrir | what worked? |
| 17:42.42 | ManxPower | WindBack: if you want to undetstand the concepts then you have MUCH reading to do. |
| 17:43.11 | ManxPower | you need to read the NAT RFCs first, pay special attention to UDP packets. |
| 17:43.11 | hi365 | im not arguing, but it seems to me that dmtf is not the issue (near 100% acuracy). here is my chanspy call: Chanspy(SIP|b${w}) |
| 17:43.29 | mtgll | tzafrir removed out calling zapata_additional twice |
| 17:43.43 | WindBack | ManxPower, okkk |
| 17:43.57 | WindBack | ManxPower, I'll do |
| 17:44.13 | tzafrir | mtgll, huh? that should have been harmless |
| 17:44.52 | ManxPower | hi365: remove the "b" and what is the value of the variable ${w} |
| 17:45.26 | hi365 | usualy balnk, unless hte chanspy was called with a leading 0 (555 vs. 5550) |
| 17:45.30 | mtgll | tzafrir i would have thought so also but removed it started and stopped asterisk and bingo it worked... |
| 17:45.34 | hi365 | here is an example: http://pastebin.ca/769781 |
| 17:45.35 | ManxPower | hi365: also you would prolly have to set "canreinvite=no" in sip.conf or the phones could easily send audio directly between the phones, bypassing asterisk |
| 17:45.49 | hi365 | i could only listen to SIP/212 |
| 17:47.32 | ManxPower | hi365: tjat os a TRAILING 0 and you did not say what ${w} is set to in that situation |
| 17:48.01 | hi365 | its also anoying that chanspy kepps on restarting form the begining of the channels list |
| 17:48.09 | hi365 | atm w is BLANK |
| 17:48.24 | ManxPower | so what happens when you remove both options? |
| 17:48.38 | *** join/#asterisk d3wayne (n=deeewayn@76.29.245.9) |
| 17:48.38 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 17:48.43 | hi365 | [19:44:35] <hi365> usualy balnk, <-------- i knew i said something about it! |
| 17:48.45 | hi365 | ill try |
| 17:49.46 | ManxPower | hi365: as well as the canreinvite I told you |
| 17:50.25 | hi365 | reinvites are all off |
| 17:50.59 | ManxPower | hi365: using "reinvite=no" or using "canreinvite=no"? |
| 17:51.21 | hi365 | canreinvite=no |
| 17:51.37 | ManxPower | in [general] or or in each sip peer? |
| 17:51.44 | hi365 | each peer |
| 17:52.45 | ManxPower | good! at least in 1.2, I don't think that works when in [general] |
| 17:53.11 | hi365 | chanspy worked like a charpm in 1.2 |
| 17:53.19 | hi365 | s/p/ / |
| 17:53.34 | hi365 | wtf? |
| 17:54.07 | ManxPower | hi365: welcome to the wonderful world of 1.4.x I assume you read the UPGRADE.txt and the changes files for 1.4 before installing it? |
| 17:54.21 | hi365 | of course |
| 17:54.27 | ManxPower | hi365: many people don't. |
| 17:54.37 | hi365 | (not) |
| 17:54.54 | [TK]D-Fender | ~assume |
| 17:54.55 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 17:54.56 | [TK]D-Fender | ^^^^^^^^^^^ |
| 17:55.41 | ManxPower | hi365: have you tried ExtenSpy instead of ChanSpy? |
| 17:55.57 | hi365 | ManxPower: wont i lose the incaoming (zap) calls then? |
| 17:56.22 | ManxPower | hi365: that would really depend on what you need. |
| 17:56.22 | *** join/#asterisk Nukemizer (n=Nukemize@15.249.sfcn.org) |
| 17:56.33 | ManxPower | I would assume that all incoming zap calls end up on SOME extension |
| 17:57.05 | hi365 | true. chansp (from what i understodd) will only listen in on OUTGOING calls |
| 17:57.12 | hi365 | am i wrong? |
| 17:57.17 | ManxPower | you are wrong |
| 17:57.49 | hi365 | thats good news. together with the fack that chanspy is "CURRENTLY" working, im going to thank you for your time and wish you all a good nigh |
| 17:58.04 | hi365 | i will continue test this tommorow. thanks again |
| 17:59.48 | *** join/#asterisk s1d (n=s1d_@host213-123-202-151.in-addr.btopenworld.com) |
| 18:01.39 | linagee | can i replace an "ISDN modem" with any other isdn modem, or are there unique identifiers they will have to change at the other side? |
| 18:01.42 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 18:02.09 | ManxPower | linagee: for the most part any "isdn modem" will work for data, but not for voice. |
| 18:02.38 | linagee | ManxPower: this is one of those all in one isdn modems. it has a voice port on it as well |
| 18:02.42 | ManxPower | now if you want to expand on the woefully small amount of information in your question, we might be able to expand the answer too. |
| 18:02.53 | ManxPower | linagee: having a "voice port" means NOTHING. |
| 18:02.58 | linagee | ManxPower: unfortunately, the problem is that it has a USB adapter. i want internet over ethernet, not USB |
| 18:03.37 | ManxPower | Any reason you are using an "isdn modem" instead of a "dsl modem"? |
| 18:03.38 | linagee | ManxPower: maybe it would be easier to connect an access point with a USB host controller on it, but this is in a remote site (very remote) and would require fiddling... hrm |
| 18:03.43 | ManxPower | hell, why are you using a modem at all? |
| 18:03.51 | linagee | ManxPower: latin america. ;) |
| 18:04.02 | linagee | ManxPower: no ADSL service yet. (supposed to be coming in december. argh) |
| 18:04.25 | ManxPower | linagee: your questions really don't make any sense. How are you connecting the "isdn modem" to Asterisk? |
| 18:04.26 | linagee | n/m. i guess i will have to wait. (it's been like that for years) |
| 18:04.46 | linagee | ManxPower: not connected to asterisk at all. |
| 18:04.54 | ManxPower | linagee: then why are you asking here? |
| 18:05.05 | linagee | ManxPower: except for the fact that i *want* to connect things to a hard phone which requires ethernet, not usb |
| 18:05.06 | linagee | ;) |
| 18:05.38 | linagee | ManxPower: it sucks that they took so long to provide ADSL. (i don't really get it) |
| 18:05.39 | ManxPower | A VoIP hardphone would require ethernet, an analog hardphone would not. |
| 18:05.53 | linagee | ManxPower: analog hardphone? heh |
| 18:05.57 | ManxPower | linagee: not really any difference from a networking point of view. |
| 18:06.28 | ManxPower | linagee: you have been around long enough to formulate good questions. |
| 18:08.08 | tzafrir | linagee, what's so wrong with USB? what rate is your internet connetion? |
| 18:08.14 | tzafrir | what computer is it? |
| 18:08.26 | linagee | tzafrir: right now the USB connects right to the computer |
| 18:08.41 | linagee | tzafrir: i suppose i could reshare it out using ethernet, but that's incredibly hokey |
| 18:08.46 | linagee | hoe-key |
| 18:08.57 | tzafrir | and what type of device is it? |
| 18:09.13 | linagee | tzafrir: unsure. some sort of isdn modem |
| 18:09.25 | linagee | POS modem. ;) |
| 18:09.56 | tzafrir | well, it's just like a PCI ISDN modem. With the rate of ISDN (I assume it is BRI), it really doesn't matter much either way |
| 18:10.07 | linagee | tzafrir: yes. true. slow is slow |
| 18:10.15 | linagee | best off waiting till they get ADSL. |
| 18:10.48 | tzafrir | anyway, at least you can use Asterisk with your ISDN line... |
| 18:11.16 | linagee | (this is in costa rica btw. beatiful country, horrible internet speeds. :) ) |
| 18:11.40 | linagee | oh yes, and i think voip is actually "illegal" (wtf?) |
| 18:13.04 | ManxPower | You know what Scott Adams says, right? "When technology allows us to solve all crimes, all we will learn is that EVERYONE is a criminal." |
| 18:13.44 | Maliuta | government monopoly on telecoms? combined with "interesting" laws and regulations regaurding intercepts? |
| 18:13.51 | linagee | ManxPower: it's weird though. i went into a small computer shop when we went to the large town and the computer guy was selling ATAs. when i asked him about the legality, he sort of shrugged it off. lol! |
| 18:14.12 | linagee | (vendor locked ATAs of course) |
| 18:14.20 | ManxPower | linagee: many times there are ways to work around those sorts of laws. |
| 18:14.32 | Maliuta | but hey, most people don't realise that the majority of T&C's from ISPs mean there net traffic can be intercepted anyway |
| 18:14.38 | Maliuta | and in many cases it is |
| 18:14.41 | linagee | Maliuta: can be? heh |
| 18:14.43 | ManxPower | in some countries VoIP is legal, but connecting those calls to the PSTN is not legal. |
| 18:15.08 | Maliuta | linagee: basically they can do what they want with the packets |
| 18:15.14 | ManxPower | in other places VoIP is legal if you are connecting offices of the same company, but not if you are selling the service. |
| 18:15.23 | Maliuta | transproxying would count as an interception |
| 18:15.38 | linagee | Maliuta: there was a news story a day or two ago about some coverup with the NSA basically monitoring ALL of AT&T's internet traffic |
| 18:15.52 | Maliuta | but I know of at least one company that does comercial stuff by intercepting http request |
| 18:16.29 | *** join/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 18:16.31 | linagee | Maliuta: how to "fix" the problem? |
| 18:16.34 | *** part/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 18:16.39 | *** join/#asterisk tobias__ (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 18:16.50 | linagee | Maliuta: encrypted connection between businesses and customers using some sort of "trust".. hrm |
| 18:16.51 | Maliuta | linagee: I _could_ name a number of ISP's with commercial arrangements to pass all http traffic through what is effectivley a packet sniffer, but I'd get my ass sued |
| 18:16.59 | linagee | Maliuta: lol! yup |
| 18:17.01 | Maliuta | linagee: yeah, VPN everything |
| 18:17.14 | *** join/#asterisk ToTo_ (n=ToTo@host205-134-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:17.25 | Maliuta | or go to IPv6 and use the IP level encryption |
| 18:17.27 | linagee | Maliuta: but how can i form trust relationships with each individual company's site i want to go to? heh |
| 18:17.47 | Maliuta | linagee: this is what the privacy stuff in v6 is for |
| 18:18.04 | Maliuta | and there are already "discussions" as to keying methodologies |
| 18:18.23 | Maliuta | I have made my thoughts on it known to the powers that be here in .au |
| 18:18.24 | tzafrir | THat is fine and dandy, if you have ipv6 |
| 18:18.29 | linagee | tzafrir: true |
| 18:18.32 | tzafrir | if not: use openvpn |
| 18:18.45 | linagee | tzafrir: openvpn between people you know, sure. already doing that. :) |
| 18:19.01 | linagee | tzafrir: oh shoot! they just monitored me saying that! :( |
| 18:19.12 | Maliuta | tzafrir: rollout of v6 is coming sooner than you think, Japan and Korea have timetables for all the infrastructure to be v6 within a couple of years |
| 18:19.39 | Maliuta | openvpn is one sollution, IPSec is another |
| 18:19.53 | Maliuta | there are multiple OSS vpn solutions |
| 18:20.01 | linagee | Maliuta: we "have" ipv6 here in california too. if you don't mind putting all your traffic through a tunnel that increases your latency by 100ms! |
| 18:20.15 | tzafrir | linagee, huh, I mean what can possible happe |
| 18:20.46 | Maliuta | linagee: you can get v6 through a tunnel broker no matter where you are. I am talking about native v6 support from network providers |
| 18:20.53 | linagee | Maliuta: huh??? |
| 18:21.19 | linagee | Maliuta: mbone was something cool like that and hasn't been implemented yet by any network providers. what makes you think they will do ipv6? heh |
| 18:21.36 | tzafrir | anyway, tunneling affects quality of audio... |
| 18:21.59 | Maliuta | Japan and [South] Korea both have plans for the entire infrastuctures of those countries to be dedicated IPv6 ... and soon |
| 18:22.10 | linagee | Maliuta: i think we will have large deployments of mesh wifi *long* before we have network providers doing cool stuff like ipv6 routing or mbone |
| 18:22.15 | Maliuta | it is only a matter of time |
| 18:22.42 | linagee | Maliuta: network providers will become antiquated because of their slowness to adopt new technologies |
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| 18:23.14 | Maliuta | wifi is not really going to be viable for large scale mesh networks. There are just going to be more 3G and up connections for data only purposes |
| 18:23.24 | linagee | Maliuta: why is it not viable? |
| 18:23.29 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 18:23.50 | ManxPower | Maliuta: I agree. I've use WiFi mesh networks before. Utterly horrid. |
| 18:24.01 | linagee | ManxPower: probably poorly implemented |
| 18:24.05 | Maliuta | think of the amount of saturation in the frequency range used |
| 18:24.11 | ManxPower | linagee: I don't think so. |
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| 18:24.26 | linagee | Maliuta: that's why you have to have mesh nodes that have three radios tuned to 1, 6, and 11. :) |
| 18:24.40 | linagee | Maliuta: just use ALL the bandwidth up. :) |
| 18:24.50 | ManxPower | Maliuta: in non-tropical places, most of the trees lose their leaves in the fall, that TOTALLY changes the RF characteristics of the network twice a year. |
| 18:24.55 | linagee | Maliuta: and use an intelligent routing protocol like batman instead of olsr |
| 18:25.27 | Maliuta | linagee: that's in the 2.4GHz range, which is saturated with more than just WiFi ... corldless phones, microwaves , and pretty much any consumer device that is "wireless" |
| 18:25.30 | ManxPower | linagee: you seem to be under the mistaken impression that nobody but the mesh uses WiFi. |
| 18:25.59 | linagee | ManxPower: if you have three radios tuned to every available channel, you can just use all the space |
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| 18:26.08 | Maliuta | there are so many people running WAPs that you get interference, and the range in 2.4GHz isn't sufficent for meshing anyhow |
| 18:26.10 | linagee | maybe a fourth for 802.11a |
| 18:26.22 | *** part/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 18:26.27 | linagee | have each node with all four radios. :) |
| 18:26.30 | ManxPower | linagee: yes, but then the radio has to act as a repeater on a half duplex technology. |
| 18:26.49 | linagee | ManxPower: simplex? :) |
| 18:27.14 | ManxPower | linagee: I'm a serial guy, it's half duplex, not simplex. 8-) |
| 18:27.19 | linagee | heh |
| 18:27.31 | linagee | ManxPower: we have many simplex radio stations where i work at. :) |
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| 18:27.48 | ManxPower | all radio is simplex if you are using only 1 channel. |
| 18:27.48 | linagee | ManxPower: and like half a dozen duplex ones. |
| 18:27.49 | Maliuta | ManxPower: trees aren't the only problems. most CBD's are radio black holes these days, a signal going in most likely won't come out the other side |
| 18:28.04 | ManxPower | Maliuta: CBD? |
| 18:28.13 | Maliuta | Central Business District |
| 18:28.22 | linagee | CBD's can screw themselves. :) |
| 18:28.30 | ManxPower | Maliuta: Ah. I never heard someone use that term before, except for in New Orleans. |
| 18:28.33 | linagee | they're close enough to run fiber next door anyway |
| 18:28.36 | ManxPower | most of the usa calls them "downtown" |
| 18:28.43 | Maliuta | ManxPower: think outside the US :) |
| 18:29.02 | ManxPower | Maliuta: and most of the rest of the world seems to call them "City Centers" |
| 18:29.32 | linagee | ManxPower: how about "cramped fscking expensive pieces of land" :) |
| 18:29.32 | linagee | ManxPower: where you ALWAYS have to pay for parking |
| 18:29.32 | Maliuta | ManxPower: city centre doesn't mean the same as CBD |
| 18:29.48 | linagee | Maliuta: what country are you in that has a term of "CBD"? |
| 18:29.49 | Maliuta | the CBD may not be where the city centre is |
| 18:29.53 | Maliuta | .au |
| 18:30.13 | Maliuta | it's also prevailant in the uk |
| 18:30.14 | linagee | Maliuta: what's the difference between a CBD and city centre? |
| 18:30.21 | linagee | (center) |
| 18:30.25 | ManxPower | I hope WiMax eventually comes out with a specification for WiMax on 900Mhz. That would rock. |
| 18:30.34 | linagee | ManxPower: wimax is dumb |
| 18:31.12 | linagee | ManxPower: unless the sender side is going to be available in consumer equipment, wimax will just be another highly priced 3G technology like that already sold by cell phone providers |
| 18:31.17 | ManxPower | linagee: Almost nobody seems to realize that to get the incredible range WiMax can offer, you need a LICENSED band, not an unlicensed band. |
| 18:31.38 | linagee | ManxPower: and that too. licensed band. wtf. |
| 18:31.39 | ManxPower | At least 900Mhz's RF characterists means it won't be blocked by a tree very much. |
| 18:31.44 | Maliuta | one is the centre of business activity (office blocks, stock exchanges etc.). The other is the centre of a city where people congregate for purposes other than work (shopping, cultural events etc) |
| 18:31.49 | linagee | ManxPower: just change the whole way spectrum is allocated already. :( |
| 18:32.03 | linagee | ManxPower: dynamic spectrum allocations would be a nice start |
| 18:32.06 | ManxPower | linagee: um, most of the world 900Mhz is unlicensed. |
| 18:32.15 | linagee | yes |
| 18:32.35 | linagee | ManxPower: make a radio that uses every segment of unlicensed space. that would be nice. :) |
| 18:32.35 | Maliuta | and older cordless phones are in that range aswell |
| 18:32.37 | ManxPower | WiMax at unlicensed 900Mhz would be great. |
| 18:33.00 | ManxPower | Maliuta: you forget, not everyone lives in a place with the 900Mhz area crowded. |
| 18:33.01 | linagee | oh wait, that was called UWB and has somehow been stalled for quite a while now. :( |
| 18:33.29 | Maliuta | ManxPower: it's just as polluted as the 2.4Ghz range |
| 18:33.30 | ManxPower | Where I live, even with a 14db antenna I can't find ANYTHING on 900Mhz, 2.4Mhz or 5.4Mhz. |
| 18:33.39 | linagee | ManxPower: wow |
| 18:33.50 | Greek-Boy | so if 900mhz is unlicenced how do the gsm companies sleep at night? |
| 18:33.50 | linagee | ManxPower: do you have a microwave oven at home? :) |
| 18:33.59 | ManxPower | At least in the usa 900Mhz is being vacated pretty fast by cordless phones. |
| 18:33.59 | Maliuta | ManxPower: I like to live where there are people, not in the middle of nowhere |
| 18:34.17 | linagee | Greek-Boy: different part of 900mhz. they are sliced to tiny little slivers |
| 18:34.19 | Maliuta | they are moving into the 2.4 range more rapidly |
| 18:34.20 | ManxPower | Greek-Boy: "900Mhz" is a generic term. |
| 18:35.02 | Maliuta | GSM is going away fairly quickly too |
| 18:35.12 | ManxPower | Maliuta: it's starting to get hard to even buy a 900Mhz cordless phone where I live. |
| 18:35.12 | linagee | ManxPower: it sucks to have a network that can be taken down by someone waving around a 2.4Ghz analog phone. heh |
| 18:35.22 | Maliuta | even here where the cost of a rollout is massive |
| 18:35.31 | Greek-Boy | Maliuta: What is replacing gsm? CDMA? |
| 18:35.37 | linagee | Greek-Boy: voip. :) |
| 18:35.38 | Maliuta | 3G |
| 18:35.43 | ManxPower | Heck, I HAVE two 900Mhz access points. |
| 18:35.47 | ManxPower | They are pre-Wifi |
| 18:35.47 | Greek-Boy | linagee :) |
| 18:35.51 | Maliuta | the CDMA is being shutdown |
| 18:35.57 | linagee | Greek-Boy: screw ma bell's network |
| 18:36.09 | linagee | Greek-Boy: fsck DIDs |
| 18:36.24 | Greek-Boy | linagee: I like the way you think :) |
| 18:36.26 | ManxPower | I should pull those out of storage and see if I can deploy them sometime. |
| 18:36.28 | linagee | Greek-Boy: they monitor all your voice traffic across that network anyway. so why do people still use it? :) |
| 18:36.38 | Maliuta | Greek-Boy: personally I would prefer to go to 3G for a network connection the VoIP for all my voice needs |
| 18:37.00 | linagee | Greek-Boy: when you are given a form that says "Phone Number?" put your iax address. :) |
| 18:37.01 | Greek-Boy | HSDPA is 3.5G |
| 18:37.04 | Greek-Boy | Wimax is 4G |
| 18:37.12 | ManxPower | "3G" seems to still have pretty high latency. |
| 18:37.16 | linagee | ManxPower: yes |
| 18:37.28 | linagee | ManxPower: and lots of jitter too |
| 18:37.42 | linagee | ManxPower: and no 100% guarantee of packets getting through |
| 18:37.45 | Maliuta | ManxPower: I have had limited experience with it. That is set to change in about 4 weeks time |
| 18:38.15 | ManxPower | Maliuta: my primary internet connection is via EVDO Rev. A |
| 18:38.16 | linagee | ManxPower: i've got an EVDO card and a dlink DIR evdo wireless router. :) |
| 18:38.22 | Greek-Boy | talking about all this wireless stuff. I'm in a bad situation with some mikrotik links. I have about 10 links on one tower and I'm using dual nstreme which mean I have 20 antennas. interference a big problem. the atheros wifi cards from mikrotik dont support a lot of frequencies. |
| 18:38.28 | Maliuta | the only way to "guarantee" packet deliver is with a physical connection end to end |
| 18:38.47 | ManxPower | Greek-Boy: It sucks to be you. |
| 18:38.52 | linagee | Maliuta: or to do it in software and throw up alerts when packets can't be ackknowledged |
| 18:38.56 | ManxPower | Greek-Boy: time to call in an RF Engineer |
| 18:39.12 | Maliuta | linagee: it's called TCP |
| 18:39.16 | linagee | Maliuta: true |
| 18:39.22 | linagee | Maliuta: tunnel everything through it then. :) |
| 18:39.38 | Greek-Boy | lol ManxPower. no need for RF engineer. I have a few more things to try... |
| 18:39.39 | linagee | [TK]D-Fender: wouldn't do a thing if Maliuta had correct routes set up. :) |
| 18:39.50 | linagee | [TK]D-Fender: s/routes/routing protocols that autorouted/ |
| 18:39.57 | Maliuta | [TK]D-Fender: hence the quotation marks |
| 18:40.08 | Maliuta | [TK]D-Fender: it's as close as you can get |
| 18:40.17 | ManxPower | if you have 20 antennas on one tower, you need an RF Engineer. |
| 18:40.22 | [TK]D-Fender | :D |
| 18:40.49 | Maliuta | wireless is insecure aswell as unreliable :) |
| 18:40.53 | linagee | ManxPower: naw! just lots of duct tape! :) |
| 18:41.09 | Greek-Boy | Maliuta; its just my backup links. primary links are fiber |
| 18:41.22 | Maliuta | linagee: why would you duct tape the engineer? unless he's into that kind of thing |
| 18:41.31 | linagee | Maliuta: maybe he is? |
| 18:41.40 | Greek-Boy | ManxPower: why do I need the engineer? So he can take down 15 antennnas? |
| 18:42.07 | linagee | Greek-Boy: so he can stand by to plug in new parts when they fail. (radio transmitter fail every couple years) |
| 18:42.11 | linagee | +s |
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| 18:42.18 | Maliuta | Greek-Boy: is it site to site? if so you might be able to get away with line of sight laser or microwave |
| 18:42.39 | ManxPower | so he can DESIGN the setup on the tower, rather than just throw some stuff on the tower and try to make it work "good enough" |
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| 18:43.29 | linagee | ManxPower: is there some sort of standard microwave link that the telcos deployed in the 80s/90s? |
| 18:43.49 | linagee | ManxPower: we have microwave links to our radio towers |
| 18:44.14 | Greek-Boy | ManxPower: The general rule of thumb on 5.8ghz band is keep the antennas atleast 1m apart. Keep frequencies 100mhz to 200mhz apart. |
| 18:44.22 | ManxPower | linagee: I doubt it. I suspect pretty much anything in the licensed bands were all proprietary back then. |
| 18:44.35 | ManxPower | Greek-Boy: what frequencies are you using? |
| 18:45.00 | ManxPower | Greek-Boy: those numbers also depends on transmit power, as well as how good the radios are. |
| 18:45.16 | linagee | ManxPower: true. how big are the harmonic frequencies? |
| 18:45.26 | linagee | s/how big/how much power at the/ |
| 18:45.32 | ManxPower | you might be able to get away with 100Mhz apart and 1m apart at X milliwatts, but not at Y watts. |
| 18:45.56 | ManxPower | linagee: no idea, hence my suggestion for an RF engineer. |
| 18:45.58 | Maliuta | is Y large ebough to fry small birds? |
| 18:46.31 | linagee | Maliuta: or people |
| 18:47.03 | ManxPower | I don't recall the correct term, but especially at higher frequencies the "clear space" required for "line of sight" is very (american) football shapped, not a straight line -- just one example. |
| 18:47.20 | linagee | ManxPower: american foozball! :-D |
| 18:49.24 | ManxPower | Most of my experience with RF has been in the Cable TV area, but I also had to research a long haul RF link as well. |
| 18:50.14 | ManxPower | didn't get to the stage of needing someone to DESIGN it, as the equipment was too expensive for the project. |
| 18:50.14 | Greek-Boy | ManxPower: I use 5290mhz and 5760mhz on 5ghz-turbo wifi |
| 18:50.28 | Greek-Boy | 350mw atheros cards |
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| 18:51.15 | Greek-Boy | ManxPower: the term you're referring to is the clear freshnel zone |
| 18:51.35 | ManxPower | that's right, the freshnel zone. |
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| 18:52.26 | Greek-Boy | I think that is another problem I am experiencing. |
| 18:52.34 | Greek-Boy | my freshnel zone isn't too good |
| 18:52.41 | Greek-Boy | gotta put a much taller tower |
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| 18:53.22 | Greek-Boy | maybe increase it from 24m to 50m |
| 18:53.22 | Greek-Boy | lol |
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| 18:57.04 | linagee | Greek-Boy: from 6 inches to a full 8 inches. :-D |
| 18:57.18 | Greek-Boy | lol |
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| 18:59.18 | Greek-Boy | anyone here test fring with asterisk? |
| 19:07.41 | Greek-Boy | apparantly not... |
| 19:09.44 | nestAr | slow day, everyone must be playing WoW |
| 19:13.07 | Greek-Boy | lol |
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| 19:17.28 | ManxPower | maybe if we knew what "fring" is.... |
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| 19:22.07 | Greek-Boy | fring is a free mobile service that comes with a sip client |
| 19:22.13 | Greek-Boy | it runs on most mobile phones I think |
| 19:22.18 | Greek-Boy | its a java app |
| 19:25.05 | ManxPower | As far as I can tell, anything that runs on "most mobile phones" does not run on the phone I have. |
| 19:28.03 | [TK]D-Fender | ManxPower, put the tin cup down and walk away quietly and nobody'll get hurt! |
| 19:28.29 | ManxPower | [TK]D-Fender: I keep meaning to replace it, but it's only 2 years old. |
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| 19:29.10 | asteriskguy | Hi guys |
| 19:29.25 | asteriskguy | has anyone ever got SLA to work? |
| 19:30.03 | [TK]D-Fender | ManxPower, Mine is almost the same and was a few moths on the market firther still (Motorola E815). Ma Bell here is supposed to be getting the HTC Touch (possibly the Duo), and an unlimited data plan (anything non-tethered) for $7 |
| 19:30.11 | [TK]D-Fender | asteriskguy, * does not support SLA |
| 19:31.13 | asteriskguy | 1.4.13 have sla.conf |
| 19:31.17 | asteriskguy | hey TK |
| 19:31.19 | asteriskguy | long time no see |
| 19:31.27 | *** join/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 19:31.28 | *** part/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 19:31.50 | ManxPower | asteriskguy: Shared Line Appearances, or Service Level Agreement? |
| 19:31.59 | [TK]D-Fender | asteriskguy, no fault of mine, and it may have an sla.conf but that sure as hell isn't SLA. Its a fugly hack pretending to be SLA. |
| 19:32.07 | [TK]D-Fender | ManxPower, SIP-B <- |
| 19:32.38 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 19:32.43 | asteriskguy | :) It looks like it |
| 19:32.50 | asteriskguy | So I got the trunk and stations setup |
| 19:33.02 | ManxPower | asteriskguy: which Digium card do you have? |
| 19:33.04 | asteriskguy | but somehow the hints aren't working right |
| 19:33.06 | ManxPower | ~trunk |
| 19:33.09 | jbot | hmm... trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 19:33.15 | ManxPower | ~siptrunk |
| 19:33.16 | jbot | siptrunk is, like, Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 19:33.49 | mvanbaak | trunk is also a version of asterisk in the source control system subversion |
| 19:34.07 | ManxPower | asteriskguy: you, of course, have calllimit set right? |
| 19:34.13 | [TK]D-Fender | ManxPower, thats an ACTUAL term used by those apps & SLA.CONF. no need to go postal on him :) |
| 19:34.59 | asteriskguy | ManxPower: I'm using tdm400P |
| 19:35.07 | ManxPower | Well if even Digium is using that horrid, inaccurate term, then I guess there is nothing left to do than to give up. |
| 19:35.42 | asteriskguy | but we're no longer using that card. It's still in the server but we're using a sip provider now |
| 19:35.46 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 19:35.46 | asteriskguy | for our services |
| 19:35.59 | ManxPower | asteriskguy: then you have it set up as a PEER, not a TRUNK. |
| 19:36.15 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 19:36.20 | asteriskguy | Shared Lined Appearnaces? |
| 19:36.38 | ManxPower | asteriskguy: yes. |
| 19:36.43 | ManxPower | aka Busy Lamp Field |
| 19:36.55 | asteriskguy | sorry I'm still lost |
| 19:37.07 | ManxPower | asteriskguy: what exactly do you expect SLA to do for you? |
| 19:37.13 | asteriskguy | in sla.conf there's only trunk or station |
| 19:37.41 | asteriskguy | allows me to see if the "line" is in use or not at another station |
| 19:37.54 | asteriskguy | and also be able to put a call on hold and then pick it up else where |
| 19:37.57 | ManxPower | asteriskguy: that would be a Shared Line Appearance or a Busy Lamp Field. |
| 19:38.16 | ManxPower | well, THAT would be Shared Line Appearance, not Busy Lamp Field. |
| 19:38.30 | asteriskguy | is there a place I can post my configs so you guys can take a look to see if I did anything wrong? |
| 19:38.31 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-233-224.pppoe.mtu-net.ru) |
| 19:38.34 | ManxPower | asteriskguy: so basically you are trying to emulate a 1980's Key System |
| 19:38.43 | asteriskguy | yes |
| 19:38.52 | ManxPower | asteriskguy: your search of the mailing list archives and the Wiki was not helpful? |
| 19:38.56 | asteriskguy | since asterisk doesn't have multiple parking lot |
| 19:39.10 | asteriskguy | I found sla.pdf in /asterisk/source/docs |
| 19:39.20 | asteriskguy | followed it but it seems to be missing something |
| 19:39.38 | ManxPower | asteriskguy: Really? I have multiple parking lots working. Took some hacking, but it works. |
| 19:39.39 | [TK]D-Fender | ~sla |
| 19:39.40 | jbot | rumour has it, sla is service level agreement, or shared line appearances |
| 19:39.48 | [TK]D-Fender | hrm... 1 sec |
| 19:40.14 | ManxPower | asteriskguy: then I guess you should also check the mailing list archives and the Wiki. Yes, the info in docs/ is the best place to start. |
| 19:40.35 | asteriskguy | ManxPower, it would be nice. Do you have any docs on getting multiple park lots to work? |
| 19:40.59 | ManxPower | asteriskguy: that would depend on how exactly you mean "multiple parking lots" |
| 19:41.16 | ManxPower | do you mean the same parking extension for multiple companies? |
| 19:41.26 | ManxPower | or do you mean the ability to have more than one call parked at the same time. |
| 19:41.38 | asteriskguy | neither |
| 19:41.42 | linagee | wow. heh. 1-800-GOOG-411 is so much better than 1-800-555-1212 :) |
| 19:41.48 | linagee | i love the voice of google! |
| 19:41.56 | ManxPower | Ok, how about when the call times out, it goes to different extensions? |
| 19:42.18 | ManxPower | Are you going to tell me what you mean by "multiple parking lots" or are we gong to continue this game of 20 questions? |
| 19:42.22 | asteriskguy | I would like to be able to allow certain locations to only park in certain range |
| 19:42.38 | ManxPower | An example would be nice. |
| 19:42.42 | asteriskguy | ok |
| 19:42.57 | asteriskguy | so our company have multiple branch offices |
| 19:43.06 | asteriskguy | all connected to the same * box |
| 19:43.34 | asteriskguy | problem is when a guys in CA parked a call, the guys in NY can accidentally pick it up |
| 19:43.51 | ManxPower | how many offices? |
| 19:44.02 | asteriskguy | very bad, especially when it's a sales call for that specific region |
| 19:44.05 | asteriskguy | right now 40 |
| 19:44.10 | asteriskguy | eventually 200 |
| 19:44.49 | ManxPower | that might take some dialplan magic to work right. |
| 19:44.50 | asteriskguy | of course we'll have to go into clustering asterisk when we get there but that's a future project. Right now we have about 1000 handsets from 41 locations total |
| 19:45.16 | asteriskguy | ok, is there a way to determine is there is a call parked in a certain space? |
| 19:46.03 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:47.45 | asteriskguy | ManxPower, how do I get to the mailing list archives? |
| 19:48.51 | linagee | can i get valet parking? |
| 19:49.01 | asteriskguy | nevermind, it's list.digium.com |
| 19:49.11 | asteriskguy | yeah, I saw something like valet parking |
| 19:49.19 | linagee | asteriskguy: but it costs. :-D |
| 19:49.33 | asteriskguy | but how do you incorporate it into ABE |
| 19:49.39 | ManxPower | I would love to know why my 1.4 test box does not have the park or pickup applications or funcations |
| 19:49.44 | asteriskguy | cost would not be an issue right now |
| 19:49.45 | ManxPower | ~mailinglist |
| 19:49.46 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 19:49.53 | linagee | don't tell me there's an actual metaphor for valet parking with asterisk. :) |
| 19:49.54 | *** join/#asterisk asdx (n=diego@adsl-150-171.click.com.py) |
| 19:50.19 | linagee | asteriskguy: cost is not an issue? cool. send the money over here then. :-) |
| 19:50.38 | asteriskguy | anyone know of any third party solution for * for SLA or Multiple parking lot or valet parking? |
| 19:51.53 | linagee | so "valet parking" allows user control when the call is parked? |
| 19:52.12 | linagee | can they take the call for a joy ride? |
| 19:52.20 | linagee | vroom! |
| 19:52.38 | ManxPower | asteriskguy: "show application park" and show application pickup". There are notes in the CHANGES file about those as well |
| 19:52.53 | ManxPower | well, core show application park |
| 19:53.30 | *** join/#asterisk BBHoss (n=hoss@146.229.181.183) |
| 19:53.31 | asteriskguy | I know in 1.4.13 you can park a call into a specific parking lot by Set(PARKINGEXTEN=parking#here) and then Park() |
| 19:53.59 | asteriskguy | but there's no way to tell if that spot has been taken or not |
| 19:54.09 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:54.15 | ManxPower | yes, there IS |
| 19:54.41 | ManxPower | <PROTECTED> |
| 19:54.54 | linagee | asteriskguy: if it's been taken there will be a car there |
| 19:54.54 | ManxPower | then you can park on a different exten. |
| 19:55.12 | [TK]D-Fender | small matter to make a script to cycle through for the first spot. |
| 19:55.24 | linagee | [TK]D-Fender: it should be that easy in real life |
| 19:56.50 | asteriskguy | so there's no function nor variable you can use to find out if the call failed? |
| 19:57.14 | ManxPower | asteriskguy: if the dialplan continues then the call failed. If the dialplan does not continue, then the park succeeded. |
| 19:57.28 | [TK]D-Fender | or use ChanIsAvail like the rest of us... |
| 19:57.41 | ManxPower | [TK]D-Fender: that still has a race condition |
| 19:57.57 | [TK]D-Fender | ManxPower, I'm in no hurry ;) |
| 19:58.21 | ManxPower | check with ChanIsAvail, comes back OK, before the user can park a call, someone else parked a call on that slot |
| 19:58.58 | [TK]D-Fender | ManxPower, you know with odds that low I'm in the "don't care" category :) |
| 19:59.40 | ManxPower | [TK]D-Fender: a dropped call can cost thousands of dollars in lost revenue. |
| 20:00.07 | asteriskguy | ChanIsAvail(Technology/resource[&Technology2/resource2...][|options]): |
| 20:00.18 | asteriskguy | so I'm assuming Technology = Local? |
| 20:00.20 | ManxPower | anyway, I think it's time for lunch |
| 20:00.52 | [TK]D-Fender | asteriskguy, I think you're developing a clue! z0mg! |
| 20:01.18 | asteriskguy | :) great....thanks guys...this will solve my headaches |
| 20:04.03 | Hadi- | any excel experts here :) |
| 20:04.23 | Hadi- | I have 2 questions :) |
| 20:05.34 | ManxPower | We excel at killing people asking off topic questions |
| 20:06.44 | Hadi- | 3 colums (A B and C) all have numbers in them... such as 1 905 415 |
| 20:06.56 | Hadi- | I want to combine all 3 to 1905415 |
| 20:09.38 | *** join/#asterisk l0 (n=Stuart@bigbrother.vermeulens.com) |
| 20:11.29 | [TK]D-Fender | Hadi-, Concatenate(A2,B2,C2) |
| 20:12.00 | [TK]D-Fender | Hadi-, and learn to press "F1" and look up "string functions" |
| 20:13.27 | Hadi- | perfect |
| 20:13.29 | Hadi- | thanks :) |
| 20:14.27 | Nivex | and now you must cut down the largest tree in the forest with... a herring! |
| 20:15.27 | [TK]D-Fender | Nivex, and I want ... a shrubbery! |
| 20:15.38 | [TK]D-Fender | Nivex, And a nice one this time! |
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| 21:06.00 | BBHoss | can anyone help me with dialing with iax from .call files |
| 21:06.16 | BBHoss | i need an example dial string |
| 21:06.43 | BBHoss | with SIP im doing SIP/6001 |
| 21:06.57 | BBHoss | but i need do dial through a trunk |
| 21:13.25 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 21:16.27 | [TK]D-Fender | BBHoss, here |
| 21:16.33 | [TK]D-Fender | ~jerjerguide |
| 21:16.34 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:17.04 | [TK]D-Fender | BBHoss, in there you'll see a sample dial using a peer set up of an ITSP. Do the math. |
| 21:25.25 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 21:30.07 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
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| 21:33.28 | javb | i need the app "extenspy" .. to listen on a specific exten, but i CANT find that app on my asterisk.. |
| 21:37.03 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 21:38.41 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.wa.comcast.net) |
| 21:38.51 | obnauticus | so is there any good Skype to asterisk solution out yet? |
| 21:39.25 | [TK]D-Fender | ~skype |
| 21:39.26 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 21:39.38 | obnauticus | lol |
| 21:39.39 | obnauticus | k |
| 21:39.50 | obnauticus | i knew 95% of that already :/ |
| 21:41.07 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 21:42.55 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 21:48.25 | obnauticus | [TK]D-Fender, couldn't I just get a windows computer and have it run skype 24/7 (this is a joke btw) get a USB to regular phone adapter, plug it into one of my asteris's FXO ports, and create a channel for it? |
| 21:48.33 | *** join/#asterisk basskozz (n=mike@209-6-20-97.c3-0.wrx-ubr3.sbo-wrx.ma.cable.rcn.com) |
| 21:49.11 | [TK]D-Fender | obnauticus, that would fall under the category of "ugly hack at best" <- :p |
| 21:49.18 | obnauticus | k |
| 21:49.22 | obnauticus | it would work though! |
| 21:49.23 | obnauticus | lol |
| 21:49.42 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 21:50.26 | TJNII | I want an extension like "exten => _2XXX,1,Goto(public,${EXTEN:1},1)" and I would like it to goto any extension within any context the caller is allowed. What is the best way to go about that? |
| 21:51.32 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) |
| 21:52.24 | basskozz | If I signup for a IAX Trunk for use with my CentPBX box, can I still us an extention that leads out to a PAP2 (sip) device? or must I use a SIP trunk? |
| 21:53.59 | BBHoss | yes asterisk handles that for you |
| 21:54.08 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 21:54.17 | BBHoss | iax trunks can talk to sip users/peers |
| 21:54.20 | BBHoss | and vice versa |
| 21:54.32 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 21:54.33 | basskozz | Thanks for clearing that up... and generally speaking IAX is better |
| 21:54.40 | basskozz | ? |
| 21:55.58 | [TK]D-Fender | TJNII, "INCLUDE" <- |
| 21:56.37 | [TK]D-Fender | basskozz, generally no, in specific cases yes. |
| 21:57.03 | *** join/#asterisk bkw__ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 21:57.08 | [TK]D-Fender | basskozz, only if you need to save on bandwidth having multiple simultaneous calls going through that connection. |
| 21:57.36 | *** join/#asterisk hijacked (i=BFDm@66.255.220.17) |
| 21:58.08 | basskozz | Well I am using this at home, and will probably only have 2 calls (at most) going at the same time. So I should go sip ? |
| 21:59.22 | [TK]D-Fender | basskozz, probably |
| 21:59.40 | [TK]D-Fender | basskozz, its more stable and will require less work for yoursystem |
| 22:00.06 | basskozz | roger, thanks [TK]D-Fender :) |
| 22:01.19 | *** join/#asterisk hijacked_ (i=G5h7@66.255.220.17) |
| 22:05.34 | *** join/#asterisk sx|lappy (n=sxpert@i07m-89-86-67-237.d4.club-internet.fr) |
| 22:07.06 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 22:08.38 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 22:09.08 | Sunmoon__ | hello ther |
| 22:09.15 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 22:10.36 | *** join/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 22:11.06 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
| 22:16.05 | Mackes | Quite in here |
| 22:16.08 | Mackes | Echo |
| 22:16.11 | Mackes | Echo |
| 22:16.42 | Mackes | Quite Quiet |
| 22:19.00 | *** join/#asterisk saftsack (n=saftsack@pD9E04DDB.dip.t-dialin.net) |
| 22:19.04 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net) |
| 22:20.17 | TJNII | So goto will jump to the specified extension/context if the caller is normally not allowed to call that exten via context= in the conf files, correct? |
| 22:20.22 | JT | Mackes: and repeating echo helps how? |
| 22:21.27 | JT | obnauticus: don't joke, that's roughly how most "skype solutions" work |
| 22:23.34 | *** join/#asterisk sx|lappy (n=sxpert@i07m-89-86-67-237.d4.club-internet.fr) |
| 22:28.16 | ManxPower | TJNII: correct. |
| 22:30.28 | TJNII | Rats. That's exactly what I don't want. |
| 22:30.32 | TJNII | Oh well. |
| 22:30.39 | mvanbaak | [TK]D-Fender: I think generally IAX actually _IS_ better then sip |
| 22:31.16 | [TK]D-Fender | TJNII, forget Gogo, you should be INCLUDING contexts |
| 22:31.30 | [TK]D-Fender | TJNII, Go read chapter 5 over and over again till your eyes blled |
| 22:31.32 | [TK]D-Fender | bleed* |
| 22:31.57 | JT | mvanbaak: well that's not true |
| 22:32.13 | mvanbaak | JT: notice the "I think" |
| 22:32.18 | mvanbaak | it's an opinion |
| 22:32.48 | mvanbaak | for endstations I almost always use SIP |
| 22:32.57 | TJNII | [TK]D-Fender: I'm trying to implement something akin to an area code where (code)(number) calls the same extension as just number). Include won't do what I want. Read my previous posts till your eyes bleed. |
| 22:33.08 | mvanbaak | but for connections to ITSP's or connections between asterisk boxen I do prefer IAX |
| 22:33.32 | JT | mvanbaak: and the ITSPs hate you for it ;) |
| 22:33.41 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 22:33.41 | mvanbaak | I know |
| 22:33.44 | mvanbaak | too bad |
| 22:33.46 | mvanbaak | they offer it |
| 22:33.47 | [TK]D-Fender | TJNII, then Dial into those other contexts. |
| 22:33.51 | JT | it's complete unscalable |
| 22:34.25 | mvanbaak | I hear great stories about OpenSer when it comes to scalability |
| 22:34.45 | JT | but OpenSER doesn't go IAX |
| 22:34.46 | mvanbaak | never looked into it though |
| 22:34.50 | mvanbaak | I know |
| 22:35.31 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 22:36.02 | mvanbaak | but for some reason there's always a fight between phones, asterisk and nat setups |
| 22:36.09 | mvanbaak | you get one right, the other breaks |
| 22:36.10 | ManxPower | TJNII: you almost never need Gotos to go between contexts. include => does everything most people need. |
| 22:36.16 | mvanbaak | and that's so freaking annoying |
| 22:36.42 | JT | i find if the settings are set right, sip just works |
| 22:36.59 | TJNII | ManxPower: Well, I'm trying to implement that area code in one line with a pattern match. That's why the context jump. |
| 22:37.59 | mvanbaak | JT: most of my setups are: OpenBSD nat firewall protecting asterisk |
| 22:38.10 | mvanbaak | and NAT at customer site we cannot control |
| 22:38.36 | mvanbaak | for some reason this gives trouble 60% of the time |
| 22:39.08 | mvanbaak | we forward sip and rtp ports to the asterisk box |
| 22:39.22 | mvanbaak | if the asterisk calls the phone, all goes well |
| 22:39.37 | mvanbaak | but phone calling asterisk ends up with the one-way-audio |
| 22:40.07 | JT | the phone is where with respect to *? |
| 22:40.24 | mvanbaak | out there, behind customer nat |
| 22:40.44 | mvanbaak | phone -- nat firewall -- internet -- nat firewall -- asterisk |
| 22:40.56 | ManxPower | canreinvite=no can fix many one-way audio problems |
| 22:41.02 | ManxPower | mvanbaak: why have Asterisk behind NAT? |
| 22:41.04 | [TK]D-Fender | mvanbaak, I do those all the time |
| 22:41.15 | JT | same |
| 22:41.16 | mvanbaak | ManxPower: because it's on our virtual platform |
| 22:41.25 | [TK]D-Fender | mvanbaak, get real :p |
| 22:41.39 | mvanbaak | defended by a bunch of loadbalancing openbsd boxen |
| 22:41.50 | ManxPower | mvanbaak: just remember that turning on NAT support on the phone, or SIP NAT fixup support on the NAT box frequently causes problems with Asterisk's nat=yes |
| 22:42.13 | JT | mvanbaak: remember to set externip= on the asterisk behind nat |
| 22:42.31 | mvanbaak | [TK]D-Fender: yeah yeah. some ppl actually like virtualization and stuff |
| 22:42.39 | mvanbaak | JT: we do that |
| 22:42.44 | ManxPower | This damn printer I just bought did not come with a USB cable. *whine* |
| 22:42.50 | mvanbaak | like I said, it works perfect 40% of the time |
| 22:42.57 | JT | mvanbaak: set qualify=yes |
| 22:43.03 | mvanbaak | set as well |
| 22:43.08 | JT | and make sure they register, and bring down the register times |
| 22:43.11 | mvanbaak | canreinvite=no |
| 22:43.27 | [TK]D-Fender | mvanbaak, And anyone who knows the complications and does ti anyways gets what they deserve. |
| 22:43.28 | ManxPower | qualify=yes causes many more problems than it solves, in my experience, at least until Asterisk has sip qualify smoothing. |
| 22:43.46 | [TK]D-Fender | ManxPower, they never do come with cables. |
| 22:44.07 | ManxPower | [TK]D-Fender: Ah. The last printer I bought was...um..uh... a dot matrix. |
| 22:44.20 | ManxPower | Epson, IIRC. |
| 22:44.37 | ManxPower | top of the line 24 pin 8-) |
| 22:44.45 | mvanbaak | [TK]D-Fender: we have the same setup with dedicated boxen. same trouble there |
| 22:44.55 | mvanbaak | [TK]D-Fender: it has nothing to do with virtualization |
| 22:45.42 | ManxPower | mvanbaak: sip debug would be helpful to you |
| 22:46.26 | mvanbaak | hhmm, I can try to get some extra ip space and dedicate one for asterisk and simply forward everything to the asterisk box |
| 22:47.12 | ManxPower | are IPs really that hard to get? |
| 22:47.16 | JT | forward? |
| 22:47.20 | JT | just give it a real ip |
| 22:47.27 | mvanbaak | JT: public ip is on the firewall |
| 22:47.32 | JT | but it sounds like your openbsd boxes may be screwing things up |
| 22:47.34 | JT | get more :) |
| 22:47.50 | mvanbaak | ManxPower: depends on where the box is |
| 22:48.15 | mvanbaak | some locations are not really friendly when it comes to ip space |
| 22:49.57 | mvanbaak | I'll put some time in it again later this week |
| 22:50.04 | mvanbaak | time to get some sleep now |
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| 23:27.34 | brainy_ | JT: finally i fixed my problem with that cisco sip gateway and the problems within MOH |
| 23:27.50 | JT | how? |
| 23:28.03 | brainy_ | the cisco gateway has silence supression.. resulting in stopped rtp... so asterisk didnt have a timing source |
| 23:28.14 | brainy_ | adding internal_timing=yes in asterisk.conf fixed it |
| 23:28.27 | brainy_ | and thats why i only had problems with that cisco sip gw |
| 23:28.34 | brainy_ | just because of the silence supression... |
| 23:28.36 | brainy_ | :) |
| 23:28.38 | fujin | what model cisco? |
| 23:28.47 | brainy_ | fujin: i dont know.. its from the line carrier |
| 23:29.19 | brainy_ | JT: http://bugs.digium.com/view.php?id=5374 |
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| 23:29.40 | brainy_ | JT: this report is kinda old, but describes why there is a problem |
| 23:29.51 | brainy_ | JT: now with internal timing everything is perfecrt |
| 23:29.56 | brainy_ | perfect |
| 23:31.54 | brainy_ | i spent the last days for patching asterisk, making a nice dialplan etc... |
| 23:32.03 | brainy_ | but now i'm really happy:) |
| 23:32.10 | koszik | <PROTECTED> |
| 23:32.15 | koszik | oops |
| 23:32.52 | brainy_ | looks like a cisco output ;) |
| 23:33.03 | JT | i was thinking it had something to do with silence supression |
| 23:33.05 | koszik | yes it is, and i somehow managed to paste it from screen |
| 23:33.08 | JT | i just didn't think of that angle |
| 23:33.16 | JT | so didn't mention it |
| 23:33.28 | JT | i forget MoH sometimes clocks off rtp packets |
| 23:33.33 | JT | which is weird as hell |
| 23:35.48 | brainy_ | ok, i just wanted to let you know how i fixed it :) |
| 23:35.53 | brainy_ | byebye and thanks anyway |
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| 23:46.57 | Nukemizer | Are there any examples of for creating a dialer/reminder dial ou to call customers and remind them of an appointment they have, and then prompt them to press one to confirm the appointment ? |
| 23:49.37 | fujin | no, there aren't. |
| 23:51.48 | ManxPower | there might be some wakeup call examples on the Wiki |
| 23:54.04 | Corydon76-dig | I've done one to remind people that they have voicemail |
| 23:54.21 | *** join/#asterisk drynish (n=drynish@bas7-montrealak-1128744575.dsl.bell.ca) |
| 23:54.24 | drynish | salut :) |
| 23:54.26 | drynish | hi! |
| 23:54.31 | drynish | sorry forgot it was in english :) |
| 23:54.33 | drynish | How are you? |
| 23:54.42 | Corydon76-dig | ~ask |
| 23:54.43 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:56.19 | drynish | My question is simple. I have a PAP2 from Linksys and I have the two lines plugged onto my asterisk server. the first line is sipura, the other one is sipura2. I would like to know if I should be able to make both of them ring at the same time through my dialplan: Dial(SIP/sipura&SIP/sipura2, r, 30) |
| 23:56.30 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 23:56.53 | ManxPower | drynish: other than reversing the option and the timeout and using the "r", yes. |
| 23:57.07 | ManxPower | Dial(SIP/sipura&SIP/sipura2,3o) |
| 23:57.15 | ManxPower | also do not put in extra spaces |
| 23:57.31 | drynish | ok let me see what I've put |
| 23:57.54 | drynish | Dial(SIP/sipura&SIP/sipura2,25) |
| 23:57.58 | drynish | So it should work |
| 23:58.01 | ManxPower | yup |
| 23:58.07 | ManxPower | why did you not try it before asking here? |
| 23:58.24 | drynish | Because it's not working! |
| 23:58.33 | drynish | So I'm wondering if there's any limitation to what I'm doing |
| 23:58.46 | ManxPower | drynish: no. that is a standard thing many, many, many people do every day. |
| 23:58.53 | ManxPower | so you problem is not there, it is somewhere els.e |
| 23:59.11 | drynish | At least I know :) |
