| 00:00.09 | *** part/#asterisk ToR\L (i=toril@cpe-24-58-23-240.twcny.res.rr.com) |
| 00:01.01 | clyrrad | devel, try envelope=off |
| 00:01.11 | clyrrad | or envelop=on |
| 00:01.35 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:01.56 | newtoasterisk | clyrrad: it is |
| 00:02.31 | *** join/#asterisk Smi|k (n=smilk@netblock-72-25-103-142.dslextreme.com) |
| 00:02.50 | clyrrad | then check your outboud dial syntax is correct |
| 00:03.11 | devel | no joy, clyrrad. thanks anyway, i'll try again at a later date. |
| 00:03.34 | clyrrad | devel, the manual sais that you can override that feature with envelope=on/off |
| 00:03.41 | clyrrad | notsure why it does not work for you |
| 00:03.50 | clyrrad | it works on 1.2 stable |
| 00:04.02 | devel | yeah, maybe i'm too old. |
| 00:04.17 | newtoasterisk | clyrrad: I believe it is |
| 00:04.46 | clyrrad | devel did you say you were running HEAD? |
| 00:05.44 | willt[work] | im so close I can taste it LOL |
| 00:06.07 | willt[work] | can someone please look at this http://pastebin.ca/45543 and tell me why it is seeing "129" as the user?? |
| 00:07.33 | *** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 00:07.49 | clyrrad | will what are you trying to do? |
| 00:07.57 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:08.17 | willt[work] | Im sending calls from a partysip server to my * box and then out to my LD provider |
| 00:08.40 | willt[work] | For some reason * is seeing 129 as the user intead of 206.81.96.73 as the peer |
| 00:09.01 | clyrrad | is 129 a phone? |
| 00:09.21 | willt[work] | 129 is a phone/extension thats on my partysip server |
| 00:09.29 | willt[work] | 129 is also a phone on my * box |
| 00:09.35 | willt[work] | differnt phones though |
| 00:10.10 | willt[work] | if I comment out my 129 on my * box, * will see 206.81.96.73 as the peer and handle the call corectly |
| 00:10.17 | clyrrad | well I am not farmiliar with a partysip server, do you * extensions have to match? |
| 00:10.19 | clyrrad | ah... |
| 00:10.21 | clyrrad | so try this |
| 00:10.35 | clyrrad | on the * box try changing 129 to 1290 |
| 00:10.49 | clyrrad | then you can just strip off the trailing 0 when you need {EXTEN:1} |
| 00:11.18 | blitzrage | clyrrad: he's matching in sip.conf -- not extensions.conf |
| 00:11.36 | willt[work] | hey blitzrage I almost have it figured out! lol |
| 00:11.40 | blitzrage | :) |
| 00:11.59 | willt[work] | it's amazing what rtfm can acomplish! |
| 00:12.11 | blitzrage | *gasp* :) |
| 00:12.14 | willt[work] | lol |
| 00:12.34 | willt[work] | it's weird i don't know why it's picking up 129 from the other system |
| 00:12.57 | blitzrage | probably because that's what its requesting in the INVITE |
| 00:14.05 | *** join/#asterisk ebag (n=gabe@adsl-69-239-166-49.dsl.renocs.pacbell.net) |
| 00:14.09 | willt[work] | in my sip headers I have two INVITES one from partysip and then one from the phone itself. Does * try to match on all of them? |
| 00:16.45 | willt[work] | anyone? |
| 00:17.54 | asterboy | Where is a good example on the net of setting up a Polycom phone to Asterisk that can then call out/receive calls on wcfxo? |
| 00:18.16 | blitzrage | asterboy: asterisk documentation doesn't really work like that |
| 00:18.36 | blitzrage | asterboy: you need to learn basics and fundamentals and put them together -- rarely will you find documentation that specific |
| 00:18.54 | asterboy | ya, I figured that from voip-info.org |
| 00:19.12 | asterboy | ok, so how about getting a polycom phone to register with asterisk? |
| 00:19.28 | clyrrad | Anyone know of a good resource that shows how to get call details out of asterisk into a C module that you can use to do CDR billing? |
| 00:19.40 | asterboy | been using sip but it wants to verify the username and pass. |
| 00:19.59 | asterboy | need asterisk to be a sip server. |
| 00:20.15 | hardwire | talk to me |
| 00:21.24 | clyrrad | Anyone here written a module for Asterisk? |
| 00:21.42 | myconid | Can asterisk scale to a 100 seat company, with 5 remote sites over T1 lines w/ 20 users each |
| 00:22.19 | ambriento | sure myconid |
| 00:22.23 | X-Rob | why not stick asterisk boxes at each of the 5 sites? |
| 00:22.32 | X-Rob | will be easier on bandwidth for calls within the same site. |
| 00:22.32 | myconid | X-Rob: reliability? |
| 00:22.41 | X-Rob | nah, just cheaper 8) |
| 00:22.41 | myconid | we are looking at a ip pbx.. for ~80k |
| 00:22.50 | myconid | a concern with * is hardware failure |
| 00:23.01 | X-Rob | well don't use cheap hardware then |
| 00:23.02 | X-Rob | or |
| 00:23.07 | X-Rob | use cheap hardware, but buy two of everything |
| 00:23.18 | myconid | can * do that somehow? |
| 00:23.26 | myconid | all the phones have both servers |
| 00:23.26 | X-Rob | do what? |
| 00:23.28 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 00:23.28 | X-Rob | no |
| 00:23.33 | X-Rob | it's called a hot spare |
| 00:23.34 | myconid | all the interface lines come into both servers somehow |
| 00:23.42 | X-Rob | if it fails, unplug it and plug the other one in |
| 00:23.44 | myconid | this is a medical services company.. |
| 00:23.48 | myconid | it cant really go down |
| 00:23.50 | X-Rob | so buy decent harwdare |
| 00:24.45 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 00:24.48 | myconid | do I run straight asterisk.. or go with a package? |
| 00:25.08 | ambriento | decent hardware with spare parts |
| 00:25.16 | ambriento | like redundant power supplies |
| 00:25.16 | myconid | Software wise.. |
| 00:25.28 | _Sam-- | if you dont want to be a phone guy for the rest of your useful waking hours, just hire a phone guy to do it. |
| 00:25.29 | myconid | ambriento: dell poweredge servers.. |
| 00:25.29 | ambriento | RAID with hot swap |
| 00:25.48 | myconid | _Sam--: I can do basic setup.. i dont think it will change that much once its setup honestly.. |
| 00:25.49 | ambriento | I dont know if Dell is th best choice |
| 00:25.55 | myconid | _Sam--: and job security is a nice word. |
| 00:26.01 | myconid | ambriento: what do you suggest? |
| 00:26.06 | _Sam-- | there will be no security, if your crap dont work. |
| 00:26.15 | asterboy | maybe I should setup polycom to asterisk with IAX and then extensions can take me out via wcfxo? |
| 00:26.19 | _Sam-- | i think its a tall order to come in and try to setup a 5 location pbx with no experience |
| 00:26.24 | _Sam-- | but im not saying it cant be done |
| 00:26.26 | myconid | _Sam--: I have basic experience |
| 00:26.29 | asterboy | unless someone has a url to setup asterisk as a sip server. |
| 00:26.31 | ambriento | My concern is about the onboard disk controllers |
| 00:26.31 | myconid | _Sam--: and it would be a slow rollout |
| 00:26.52 | myconid | ambriento: the dell cerc cards are crap.. i usually run 3ware escalades |
| 00:27.11 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 00:27.12 | asterboy | voip-info.org mentions asterisk as sip server but has no example sip.conf files. |
| 00:27.18 | asterboy | anyone know of any? |
| 00:27.32 | myconid | _Sam--: I have setup a small company with two sites over a gigabit wireless link (ie, pretty much a big lan) with one server and 40 cisco phones |
| 00:27.34 | asterboy | anyone setup asterisk as a sip server? |
| 00:27.42 | ambriento | well, if you can afford that, is nice cause dell has that on-site support and switch parts |
| 00:27.43 | willt[work] | blitzrage: do you know a way I can have asterisk fall through to the second SIP entry and not match the phone |
| 00:27.57 | _Sam-- | make samples? |
| 00:28.04 | asterboy | ls |
| 00:28.15 | _Sam-- | be cafeful though |
| 00:28.22 | _Sam-- | 3) "make samples" |
| 00:28.22 | _Sam-- | <PROTECTED> |
| 00:28.35 | myconid | ambriento: I don't want SCSI drives.. and for low end servers, dell is relaly the only sata player. |
| 00:28.43 | myconid | ambriento: everything compaq/hp comes with scsi backplanes |
| 00:28.45 | AlexCTI | Hi, someone can help me to set barge in to extensions? |
| 00:28.46 | blitzrage | willt[work]: what do you mean? Asterisk will match from bottom of the file to the top -- so switching the order might help |
| 00:28.59 | _Sam-- | myconid : what is the wireless hardware and distance between the two locations? |
| 00:29.01 | ambriento | if you have asterisk source, just go to configs/ directory and all the samples will be there |
| 00:29.27 | myconid | _Sam--: bridgewave.. ~2 miles |
| 00:29.56 | clyrrad | Any of you guys written your own modules for asterisk? |
| 00:29.56 | _Sam-- | i been messing with some orthogon systems...but over much larger distances |
| 00:30.07 | myconid | they were rated to 40 miles I think |
| 00:30.11 | willt[work] | blitzrage: it is at the bottom and I just tried the top |
| 00:30.18 | _Sam-- | my buddy just setup 111km / 40mbps |
| 00:30.22 | _Sam-- | over water though |
| 00:30.28 | willt[work] | blitzrage: I was talking about int the SIP headers |
| 00:30.28 | myconid | nice. |
| 00:31.01 | myconid | anyways.. the company is willing to pay 80k for a full ip pbx rollout |
| 00:31.09 | myconid | and I think we can do asterisk with quality hardware for like 30k |
| 00:31.13 | myconid | (cisco phones, dell servers). |
| 00:31.19 | asterboy | has anyone setup asterisk to act as a sip server? |
| 00:31.28 | _Sam-- | asterboy : probably everyone here. |
| 00:31.35 | willt[work] | myconid: will they buy a 7 series bmw with the change ? :) |
| 00:31.36 | ambriento | I dont think you need Cisco phones either :) |
| 00:31.45 | myconid | ambriento: i like the interface :D |
| 00:31.54 | asterboy | can you point me to a URL so I can get some info on it? |
| 00:32.00 | myconid | but im just sketchy about relying on asterisk.. |
| 00:32.06 | myconid | ive had some issues with it before.. |
| 00:32.06 | ambriento | sure its cool :) but its a litle it more expensive :) |
| 00:32.08 | asterboy | voip-info.org has it buried somewhere I'm sure. |
| 00:32.20 | myconid | Does digium do call up support? |
| 00:32.28 | willt[work] | I have tried a few ip phones. I like cisco the best so far. quality seems to be better |
| 00:32.30 | ambriento | I rather sangoma cards |
| 00:32.54 | ambriento | they that AFT104D, with hardware echo cancellation that is pretty nice |
| 00:32.56 | myconid | we pay ~200 for cisco IP phones |
| 00:32.58 | AlexCTI | Someone has setup a chanspy()? |
| 00:33.03 | ambriento | they have* |
| 00:33.04 | myconid | so theyre actually the cheapest |
| 00:33.12 | myconid | cheaper than polycom |
| 00:33.18 | willt[work] | that is pretty cheap |
| 00:33.18 | ambriento | alexcti, clyrrad just did that |
| 00:33.21 | *** join/#asterisk Qwellj2me (n=Qwell@unaffiliated/qwell) |
| 00:33.24 | willt[work] | is that refurb? |
| 00:33.27 | _Sam-- | i use chanspy |
| 00:33.30 | _Sam-- | works fine |
| 00:33.53 | AlexCTI | chaspy is not part of asterisk? |
| 00:33.54 | myconid | willt[work]: discount from cisco directly because of the industry |
| 00:34.05 | AlexCTI | is it a separate package? |
| 00:34.07 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 00:34.12 | willt[work] | nice |
| 00:34.16 | _Sam-- | its part of asterisk |
| 00:35.04 | AlexCTI | Because I set it as incoming chanspy(scan) and beep,beep .... but i never get hear anything |
| 00:35.17 | _Sam-- | need to set some spygroups |
| 00:35.20 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 00:35.48 | myconid | I can setup two asterisk servers.. each with a T1 card.. and run each T1 of 24 lines into the servers.. and have redundancy that way cant I? |
| 00:35.49 | ambriento | AlexCTI, try CLI> show application chanspy |
| 00:35.56 | myconid | and have a frontend server |
| 00:35.57 | AlexCTI | ok. |
| 00:35.58 | _Sam-- | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy |
| 00:36.01 | myconid | that just forwards sip |
| 00:36.05 | _Sam-- | follow the directions...they work |
| 00:36.07 | *** join/#asterisk `Consultant (n=darkguar@68-232-73-89.chvlva.adelphia.net) |
| 00:36.14 | `Consultant | Hello |
| 00:36.16 | AlexCTI | oki.. I'll do that |
| 00:36.18 | `Consultant | I have a few questions |
| 00:36.26 | *** join/#asterisk redondos_ (n=redondos@190.48.36.29) |
| 00:36.36 | ambriento | as a consultant, you should have the answers |
| 00:36.39 | ambriento | :) |
| 00:36.49 | myconid | lol |
| 00:37.07 | `Consultant | that is what I'm paid for, but when I dont have the answers I know enough to ask around |
| 00:37.24 | ambriento | myconid, I didn't get it yet |
| 00:37.27 | myconid | `Consultant: please paypal #asterisk $50 for the first 5 minutes, $10 every minute thereafter. |
| 00:37.36 | `Consultant | lol |
| 00:37.39 | ambriento | :) |
| 00:37.57 | ambriento | myconid, 2 servers, same facility |
| 00:38.12 | myconid | ambriento: just trying to figure out how to make this reliable |
| 00:38.20 | `Consultant | okay, I know almost nothing about VOIP |
| 00:38.26 | myconid | ambriento: where reliable means can lose an entire trunk/server/router/switch |
| 00:38.38 | ambriento | go ahead and ask, don't need to justify anything :) |
| 00:38.53 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:39.01 | ambriento | myconid, I see.. |
| 00:39.10 | asterboy | how does * acting as a SIP server verifiy the username/pass of a sip client? |
| 00:39.25 | `Consultant | but need info on ways to set up a 125 line in house VOIP system, that will allow for expansion, and regular phone number dialing. They want everything in house |
| 00:39.41 | myconid | ambriento: mental health.. if the pbx goes down.. and someone cant call during an emerency.. a savings of $60k is nothing compared to the cost of a life.. situatio |
| 00:39.42 | myconid | n |
| 00:40.00 | myconid | `Consultant: whats a 125 line/ |
| 00:40.06 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 00:40.12 | SwK | 125 station most likely |
| 00:40.28 | myconid | wassthat |
| 00:41.05 | SwK | 125 phones on the pbx |
| 00:41.56 | myconid | those a !usa thing? |
| 00:42.27 | SwK | nope |
| 00:42.36 | SwK | thats just a pbx thing |
| 00:42.50 | myconid | Are those the phones you have when you run a standard PBX? |
| 00:43.35 | myconid | ambriento: my cisco switches apparently can do HA Sip switching |
| 00:43.40 | X-Rob | 125 lines is what, 8 T1's? |
| 00:43.54 | X-Rob | 120 is 4 E1's |
| 00:43.59 | X-Rob | so it's 6 T1's or so. |
| 00:44.07 | _Sam-- | less if you could use g729 |
| 00:44.57 | X-Rob | Sam, an E1 contains 30 B channels and 1 D channel |
| 00:45.00 | [av]bani | whats a good win32 softphone? |
| 00:45.04 | SwK | 125 stations doesnt mean 125 Channels to the PSTN |
| 00:45.14 | X-Rob | he said '125 line' |
| 00:45.15 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 00:45.19 | X-Rob | not '125 station' |
| 00:45.31 | X-Rob | so it's prolly 500 or so stations |
| 00:45.31 | newtoasterisk | Could anyone take a look at my, very simple, sip.conf /extensions.conf and see where I have gone wrong with my outbound dialing? |
| 00:45.41 | _Sam-- | i have a t1...i can have about 50 lines or more on it...using g729, smartass. |
| 00:46.04 | X-Rob | Sam, do you know what B channels and D channels are? |
| 00:46.14 | _Sam-- | uh yea...not all t1's have b or d channels. |
| 00:46.16 | _Sam-- | just PRI |
| 00:46.20 | willt[work] | grrrr |
| 00:46.33 | X-Rob | Goodo. So. As I said. 30 B channels per E1. 24(?) per T1. |
| 00:46.38 | X-Rob | or is it 26? |
| 00:46.59 | _Sam-- | you dont need a b channel to have a phone call on a t1 |
| 00:47.05 | _Sam-- | i speak SIP to the CO |
| 00:47.18 | asterboy | So asterisk must act as a SIP server without formal need to run any specific modules. |
| 00:47.24 | X-Rob | good on you. |
| 00:47.36 | Smi|k | is there any HUD projects similar to fonalitys new dewal |
| 00:47.38 | _Sam-- | go f' yourself |
| 00:47.40 | *** part/#asterisk _Sam-- (n=sam@mail.kneedraggers.com) |
| 00:47.46 | X-Rob | heh |
| 00:48.13 | Mw3 | lol |
| 00:48.24 | X-Rob | Anyway, `Consultant seems to have buggered off |
| 00:48.38 | blitzrage | X-Rob: 23 on a T1 with one data channel for a total of 24 |
| 00:48.45 | AlexCTI | -Sam- I don't find thw way to create groups to chanspy, do yo have info about it? |
| 00:48.47 | X-Rob | Ah, there ya go. Thanks, blitzrage. |
| 00:49.03 | blitzrage | X-Rob: np :) |
| 00:49.14 | X-Rob | I knew it was some wierd arse number. |
| 00:49.14 | asterboy | Can someone who has setup * to act as a sip server please give me some hints or a URL? |
| 00:49.20 | X-Rob | ~docs |
| 00:49.22 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:49.25 | X-Rob | asterboy ^^^ |
| 00:49.37 | asterboy | voip-info.org is useless. |
| 00:49.42 | asterboy | digium same |
| 00:49.48 | X-Rob | *blink |
| 00:49.49 | X-Rob | ( |
| 00:49.52 | asterboy | astmasters mentions but nothing |
| 00:50.00 | asterboy | oreilly maybe...checking. |
| 00:50.20 | X-Rob | 'voip-info.org is useless' |
| 00:50.22 | X-Rob | *news flash* |
| 00:50.46 | justinu | what is up? |
| 00:51.01 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 00:51.08 | wunderkin | AlexCTI, its not very complicated, did you read: show application chanspy? |
| 00:51.20 | newtoasterisk | Could anyone take a look at my, very simple, sip.conf /extensions.conf and see where I have gone wrong with my outbound dialing? |
| 00:51.34 | X-Rob | newtoasterisk, pastebin it - www.pastebin.ca |
| 00:51.34 | willt[work] | Ok I understand that asterisks looks for s [user] entry before it looks for a peer to match on host. But I want it to match peers first! Is there a way to do this?? |
| 00:51.56 | ManxPower | willt, why? |
| 00:52.01 | asterboy | Holy fuck...as usual, gonna have to do this myself. |
| 00:52.13 | blitzrage | asterboy: you get what you pay for in here |
| 00:52.20 | asterboy | no doubt...0 |
| 00:52.21 | ManxPower | [user] is for INCOMING, username= is for OUTGOING |
| 00:52.53 | willt[work] | ManxPower: I am sending request from a partysip server and it's matching on the user the phone is setup as |
| 00:52.57 | asterboy | just surprised with 272 users, not one know a fucking thing about * sip serving. |
| 00:53.15 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:53.20 | clyrrad | Anyone here using billing on thier * server? If so what are you using? DId you do it using AGI, or a C module? |
| 00:53.20 | ManxPower | willt[work], well stop providing the same auth info |
| 00:53.25 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 00:53.38 | X-Rob | asterboy, I've given you the voip-info.org clue |
| 00:53.42 | willt[work] | ManxPower: lol.. im migrating off of talkisp/partysip |
| 00:53.48 | X-Rob | If there's somtehing _specific_ you want to know, feel free to ask |
| 00:53.50 | blitzrage | ManxPower: sometimes you have to match on the peer for incoming when the other end can't really auth to you... |
| 00:53.53 | ManxPower | You can fight asterisk all you want, asterisk will always win. |
| 00:53.54 | asterboy | voip-info.org has nothing on sip server |
| 00:53.55 | blitzrage | ManxPower: but you already knew that :) |
| 00:54.03 | X-Rob | asterboy, uh, sip.conf? |
| 00:54.04 | asterboy | points you to express routing |
| 00:54.06 | ManxPower | blitzrage, that's what permit= and deny= is for 8-) |
| 00:54.12 | willt[work] | lol aint that the truth |
| 00:54.47 | justinu | i'm sure a lot of people here know about your topic |
| 00:54.52 | justinu | they might just not feeling like talking |
| 00:54.54 | blitzrage | ManxPower: true -- but I rarely use type=user anymore -- almost everything is just a type=peer |
| 00:55.22 | ManxPower | blitzrage, I use: peer/user for gateway devices, friend for phones |
| 00:55.27 | asterboy | voi-info.org/sip.conf only gives examples for sip client config...nothing on sip server. |
| 00:56.06 | X-Rob | asterboy, try asking a reasonable, verbose question. |
| 00:56.16 | X-Rob | people might not treat you like a cock then. |
| 00:56.16 | willt[work] | im close to just setting up ser or a second asterisk box for this migration!! |
| 00:56.23 | X-Rob | becase that's what you're appearing ot be right now. |
| 00:56.25 | asterboy | well, IAX is pretty cool, I'll setup the Polycom for that and divvy out/in calls there. |
| 00:56.40 | blitzrage | ManxPower: I don't think I have a single friend entry anywhere :) I use peer for friend now |
| 00:57.02 | blitzrage | or separate peer / user entries |
| 00:57.03 | willt[work] | well your peers are your friends right? |
| 00:57.03 | asterboy | What is unreasonable about asking for a URL to setup * as a sip server? Holy fuck, now I am a cock! |
| 00:57.10 | ManxPower | blitzrage, I do things the Old Fashioned Way until peer/user are actually removed |
| 00:57.24 | X-Rob | Wow. |
| 00:57.32 | ManxPower | asterboy, because Asterisk is NOT a SIP server. |
| 00:57.35 | X-Rob | You know, I haven't /ignored someone in years. |
| 00:57.38 | asterboy | huh? |
| 00:57.41 | blitzrage | asterboy: yeppers -- you are. Read the docs just like everyone else -- actually, when I started I didn't even HAVE documentation, so suck it up |
| 00:57.47 | asterboy | * can be a sip server |
| 00:58.12 | ManxPower | asterboy, As close as Asterisk is would a B2BUA or something like that. |
| 00:59.13 | ManxPower | blitzrage, is there info on setting up Asterisk as a server for SIP devices to register and route calls to in the new asterisk book? |
| 00:59.32 | asterboy | I just want my Polycom to register with * . |
| 00:59.48 | ManxPower | asterboy, so what is the SPECIFIC problem? |
| 00:59.55 | X-Rob | There we go |
| 00:59.57 | X-Rob | A specific, verbose, question |
| 01:00.11 | Ariel_ | polycoms works with asterisk. Great phones |
| 01:00.25 | X-Rob | type=friend in sip.conf |
| 01:00.27 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 01:00.34 | ManxPower | X-Rob, my internet service will prolly go out from the storm just as I'm typing the critical answer to him. |
| 01:00.35 | X-Rob | (although, you may want to use two) |
| 01:00.49 | asterboy | As said before, when I try to initiate the SIP session, I'm getting Username/Auth name mismatch. |
| 01:00.59 | willt[work] | thanks guys you rock! |
| 01:01.02 | X-Rob | ManxPower, he's actually asked a question now, so he might be able to get some help |
| 01:01.15 | asterboy | x-rob, honestly I asked this before. |
| 01:01.19 | ManxPower | X-Rob, I'll just post the top part of my sip.conf |
| 01:01.20 | willt[work] | ill just use deny= on entries I don't want to match on! |
| 01:01.20 | X-Rob | asterboy, so, what does your sip.conf say? |
| 01:01.21 | Ariel_ | asterboy, did you setup the phone via ftp setup or via it's web gui |
| 01:01.24 | blitzrage | ManxPower: in TFOT? probably not step-by-step, but the info is there to teach you how :) |
| 01:01.37 | X-Rob | (for that device) |
| 01:01.39 | asterboy | via ftp |
| 01:01.54 | asterboy | sip.conf talks about sip server but provides no exampls. |
| 01:01.58 | X-Rob | OOh, speak to Ariel_, he's clued on polycoms. |
| 01:02.16 | X-Rob | asterboy, type = friend. |
| 01:02.20 | Ariel_ | X-Rob, really now... |
| 01:02.22 | asterboy | yes |
| 01:02.23 | X-Rob | that lets you send and receive calls |
| 01:02.24 | blitzrage | or [TK]D-fender |
| 01:02.26 | asterboy | type = friend |
| 01:03.14 | ManxPower | asterboy, http://pastebin.ca/45551 |
| 01:03.28 | ManxPower | that is a WORKING sip.conf with a polycom phone |
| 01:04.00 | asterboy | excellent! |
| 01:04.02 | ManxPower | the phone, of course has to be configured to send the correct username/secret |
| 01:04.05 | Ariel_ | wow a mac address setup |
| 01:04.20 | asterboy | I'll digest that and work on it. |
| 01:04.25 | asterboy | At least its a bone to chew on. |
| 01:04.26 | X-Rob | there's only 16 billion IP addresses on the net |
| 01:04.28 | X-Rob | I'll find it |
| 01:04.29 | ManxPower | the username/secret for the first line is 0004f200cf0c-a/0004f200cf0c-a |
| 01:04.38 | willt[work] | blitzrage: thanks for putting up with me.. I owe you a pint or two :) |
| 01:04.53 | ManxPower | Guess what the 2nd line is configured to send 0004f200cf0c-b/0004f200cf0c-b as the username/secret |
| 01:04.57 | blitzrage | willt[work]: too bad i stopped drinking 3 days ago :D |
| 01:05.03 | willt[work] | LOL |
| 01:05.06 | asterboy | do you need the context to have the mac or is that just for ease of debug? |
| 01:05.27 | ManxPower | X-Rob, The IP address of that phone is 172.16.7.126 |
| 01:05.30 | blitzrage | ManxPower: you don't program right? |
| 01:05.39 | X-Rob | asterboy, context= is where the numbers go when you dial them on the phone |
| 01:05.45 | X-Rob | ManxPower, fwor. L33t. |
| 01:05.57 | *** join/#asterisk agentofbsd (n=sdfsfd@202.147.31.212) |
| 01:05.59 | asterboy | pardon, I meant the lables. |
| 01:06.07 | ManxPower | asterboy, the [whatever] is what asterisk matches on for calls from the phone to asterisk |
| 01:06.09 | asterboy | [mac] |
| 01:06.10 | X-Rob | I'm gunna h4xx0r that right after I finish on 127.1.37.1 |
| 01:06.22 | ManxPower | blitzrage, I don't admit I can program |
| 01:06.34 | asterboy | ok, that is what I thought, just didn't know if there might be some significance to that. |
| 01:07.02 | ManxPower | we use the MAC as the user/secret because it makes it SO much easier to work with and makes us remember that A DEVICE AND SIP ENTRY IS NOT AN EXTENSION. |
| 01:07.15 | asterboy | I like that |
| 01:07.28 | asterboy | makes perfect sense |
| 01:07.32 | ManxPower | and really, if someone has hacked into our network enough to see MAC addresses then we're screwed anyway |
| 01:07.39 | X-Rob | heh |
| 01:07.48 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 01:08.01 | X-Rob | good point. If your switch is 0wn3d then you're pretty much stuffed8) |
| 01:08.03 | ManxPower | since you can only see a MAC address on the same subnet, not across routers |
| 01:08.35 | ManxPower | asterboy, for phones you always want the [whatever] to be the same as username=whatever for that sip.conf entry |
| 01:08.45 | *** join/#asterisk tikola (n=tikola@203.118.179.11) |
| 01:08.50 | tikola | hi guys |
| 01:08.52 | ManxPower | as you can see from my pastebin |
| 01:09.11 | *** join/#asterisk newtoasterisk (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 01:09.48 | tikola | small question if i may. it seems when my iax phones "refresh" every 60 seconds, theres like a clicking/static sound on the phone. anyway to fix it? |
| 01:10.32 | AlexCTI | wunderkin: U there? |
| 01:10.55 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:11.16 | AlexCTI | The only thing that I don't understand well is teh group part? I dont see any sample as a guide |
| 01:11.35 | asterboy | ok, working on mine now. |
| 01:11.38 | asterboy | thx |
| 01:12.50 | wunderkin | AlexCTI, ? you are wanting to set spy groups? show application chanspy g(grp) - Match only channels where their ${SPYGROUP} variable is set to 'grp' .. so.. you have to set the SPYGROUP variable on the call.. |
| 01:13.44 | wunderkin | ChanSpy(|qg(type2-32)) Set(SPYGROUP=type2-32) |
| 01:13.53 | ManxPower | blitzrage, I do a little bit of perl and c and php programming, but I have to have a much stronger reason than mere money |
| 01:13.55 | AlexCTI | I see |
| 01:14.36 | asterboy | thats all it was for crying out loud! |
| 01:14.56 | asterboy | dam [lable] has to be the same as username...christ! |
| 01:15.08 | asterboy | no wonder I have grey pubic hair. |
| 01:15.23 | ManxPower | asterboy, TECHNICALLY it does not, but unless you want to learn all things SIP, just assume it is true. |
| 01:15.51 | asterboy | good point |
| 01:16.13 | Hmmhesays | damn it is nice jamming with a singer |
| 01:16.15 | Hmmhesays | i fucking hate singing |
| 01:16.42 | X-Rob | he gave up |
| 01:16.52 | X-Rob | couldn't tell the difference between lines and xtns |
| 01:17.29 | ManxPower | asterboy, you can think of it (from the asterisk perspective) as [blah] is the username asterisk expects the device to send, and username=blah is what asterisk will send to the device. |
| 01:17.29 | ManxPower | Many SIP providers use DIFFERENT usernames for incoming/outgoing and that's why I ALWAYS have seperate peer and user entries for providers or other gateways. |
| 01:19.15 | asterboy | ok, thats a good tip |
| 01:21.46 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 01:22.44 | justinu | any AMI gurus around? |
| 01:22.47 | justinu | manager interface |
| 01:22.58 | X-Rob | Oooh |
| 01:23.44 | asterboy | ~ami |
| 01:23.50 | X-Rob | I blew up a 48 port 3250TG switch (48 10/100 + 2 gig) (cost ~$700) and they're replacing it with a 3550TG (l3, faster, cost $1200) |
| 01:24.36 | asterboy | guess jbot isn't even a noob at AMI |
| 01:24.48 | justinu | heh |
| 01:25.30 | asterboy | I like CLI for management myself. |
| 01:27.06 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
| 01:27.23 | X-Rob | but I'm going to need to buy another one now - it has 'single ip management' |
| 01:27.35 | X-Rob | so all switches have the same IP address and you can manage them from anywhere |
| 01:27.38 | X-Rob | that's clever. |
| 01:27.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 01:28.21 | justinu | dlinks are pretty cool, until you have to reboot them :( |
| 01:29.25 | X-Rob | I've got a couple, and really have never had to reboot 'em |
| 01:29.37 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 01:29.44 | X-Rob | the one that I sent off had actually had a reasonably direct lightning strike |
| 01:29.46 | justinu | cool |
| 01:29.50 | X-Rob | but they fixed it under warranty |
| 01:29.55 | X-Rob | which i thought was nice |
| 01:30.00 | justinu | i like them too... i'm just hoping this is not going to happen very often |
| 01:30.16 | X-Rob | (especially since the insurance claim bought them a new one, and I now get the old broken one8) |
| 01:30.33 | justinu | isn't that like insurance fraud? :P |
| 01:30.34 | X-Rob | ..which is no longer broken, and is better than it ever was |
| 01:30.38 | X-Rob | no |
| 01:30.40 | X-Rob | it was broken |
| 01:30.45 | X-Rob | they needed a new one urgently |
| 01:30.48 | X-Rob | they got a new one urgently |
| 01:30.53 | justinu | ic |
| 01:30.56 | X-Rob | they threw out the old one |
| 01:30.58 | X-Rob | I intercept it |
| 01:31.06 | X-Rob | that's all 8) |
| 01:31.09 | X-Rob | intercepted |
| 01:31.57 | xtrvd | intercepted: A new word for dumpster diving. |
| 01:32.15 | X-Rob | *grin* |
| 01:32.36 | X-Rob | the cat has _finally_ found a comfy place to sit on this desk |
| 01:33.33 | justinu | mine prefers the top of the stereo amplifier |
| 01:33.37 | *** join/#asterisk shell (i=shell@201.132.18.159) |
| 01:33.45 | justinu | it used to like the tops of the CRTs |
| 01:33.50 | justinu | but those went away about a year ago |
| 01:34.22 | blitzrage | I can't imagine how bad a cat sitting on top of a monitor would be for it :) |
| 01:34.29 | justinu | it eventually died |
| 01:34.35 | Zipper_32 | The CRT or the Cat? |
| 01:34.37 | justinu | the crt |
| 01:34.38 | justinu | :P |
| 01:34.41 | Zipper_32 | hehe |
| 01:34.44 | blitzrage | lol |
| 01:34.57 | justinu | i was hoping, because I wanted this dual 20" LCD setup |
| 01:35.00 | justinu | and I got it |
| 01:35.13 | Zipper_32 | The cat's life insurance? |
| 01:35.25 | Zipper_32 | *ahem* sorry. |
| 01:36.09 | justinu | cat is right in the middle of a dream, with it's eyes open |
| 01:37.26 | X-Rob | this one starts purring and dribbling when she's in the middle of a good dream |
| 01:37.33 | justinu | some of them do that |
| 01:38.18 | Zipper_32 | Could somebody perhaps take a look at why my Zap drivers aren't compiling... I haven't been able to figure out why all day: http://pastebin.com/600844 |
| 01:38.27 | Zipper_32 | And I've been a typical male, not asking for help. |
| 01:39.28 | X-Rob | Zipper_32, someone recently posted that |
| 01:39.31 | X-Rob | it's a centos bug |
| 01:39.36 | Zipper_32 | BAH! |
| 01:39.45 | Zipper_32 | Well, that makes me feel a lot better... |
| 01:40.06 | X-Rob | http://bugs.digium.com/view.php?id=6696#bugnotes |
| 01:40.09 | Zipper_32 | You see, I swear I followed the EXACT same steps on my .32 kernel, and then .34 rolls around and 'whammy' |
| 01:41.12 | Zipper_32 | Thanks X-Rob, =) |
| 01:41.19 | X-Rob | Fix is: |
| 01:41.28 | X-Rob | edit 2.6.9-34.EL-i686/include/linux/spinlock.h |
| 01:41.41 | X-Rob | change 'rw_lock_t' to 'rwlock_t' |
| 01:42.13 | Zipper_32 | <3 |
| 01:44.23 | *** part/#asterisk woolbeo (n=woolbeo@toby.stoneflytech.com) |
| 01:47.36 | X-Rob | ManxPower, pfft. I can't afford cisco these days |
| 01:47.38 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:47.45 | X-Rob | although I've got some 3810's in the shed |
| 01:47.55 | X-Rob | anyone want some MC3810's? |
| 01:48.03 | ManxPower | X-Rob, eBay |
| 01:48.06 | De_Mon | shed? |
| 01:48.06 | X-Rob | Yeah |
| 01:48.20 | X-Rob | I probably should do that before they depreciate away to nothing |
| 01:48.22 | ManxPower | we get Cat 550xs off eBay and the cards for them |
| 01:49.00 | CrashHD | where can I find documentation on the iax config options? |
| 01:49.05 | CrashHD | voip-info just lists them |
| 01:49.07 | Zipper_32 | Yes X-Rob, people like me need cheap switches. |
| 01:49.08 | CrashHD | doesn't say much about them |
| 01:49.11 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 01:49.55 | X-Rob | Zipper_32, Dlink is your friend then. |
| 01:50.42 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:51.11 | X-Rob | Geez. |
| 01:51.15 | ManxPower | CrashHD, iax.conf.sample |
| 01:51.21 | ManxPower | Just like every other asterisk config file. |
| 01:51.27 | CrashHD | ya I know |
| 01:51.33 | CrashHD | didn't see the auth= explanation |
| 01:51.33 | X-Rob | D-link are _so_ cheap. 16 ports of unmanaged gig, AU$400 |
| 01:51.55 | ManxPower | Ah. We don't need GigE |
| 01:52.12 | X-Rob | ManxPower, you _do_ need GigE, you just don't know it yet |
| 01:52.17 | *** part/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:52.19 | X-Rob | I've got gig into the file server, and gig into my PC |
| 01:52.21 | X-Rob | *fwor* |
| 01:52.21 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:52.24 | ManxPower | CrashHD, if it's not in the .sample config file then it does not exist |
| 01:52.27 | X-Rob | (only have 2 gig ports in the switch) |
| 01:52.47 | ManxPower | X-Rob, our users are morons and can hardly turn on their computer, they don't use much bandwidth. |
| 01:52.57 | CrashHD | ok ManxPower thanks |
| 01:52.58 | ManxPower | Heck most of our users don't even have access to the file servers |
| 01:53.03 | X-Rob | Heh |
| 01:53.08 | X-Rob | Well. _YOU_ need gig then |
| 01:53.11 | X-Rob | stuff everyone else |
| 01:53.12 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 01:53.25 | ManxPower | X-Rob, Why? |
| 01:53.38 | X-Rob | coz how else can you get your porn off the fileserver quickly? Duh. |
| 01:53.45 | ManxPower | if we had bandwidth issues we'd be hearing it on the VoIP stuff on the local lan. |
| 01:53.58 | X-Rob | Not need in a technical sence. |
| 01:54.38 | ManxPower | Ah. The modern sense of the word "need", where in old times the word "want" was used. |
| 01:54.45 | X-Rob | That's the one. |
| 01:54.58 | X-Rob | When you work in IT, the terms become interchangeable. |
| 01:55.09 | X-Rob | because beancounters never give you money for stuff you 'want' |
| 01:55.17 | ManxPower | X-Rob, anything that causes more work does not get purchased. |
| 01:55.38 | X-Rob | that's not a bad theory |
| 01:55.52 | ManxPower | we run the IT department of a US$550 million/year company with 16 or so locations with 2 full time IT staff and a consultant. |
| 01:55.54 | ManxPower | X-Rob, anything that causes more work does not get purchased. |
| 01:56.04 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:56.09 | X-Rob | you said that. I said that's not a bad theory. |
| 01:56.30 | ManxPower | I sent in a request to the Guinness Book of World Records about that but they never responded. |
| 01:57.20 | CrashHD | why did the new iax jitter buffer start being default? |
| 01:57.26 | CrashHD | s/why/when |
| 01:57.28 | ManxPower | X-Rob, we are in the process of downsizing services MIS provides. |
| 01:57.49 | ManxPower | We are phasing out desktop support for the agents, they also no longer get access to the file servers |
| 01:59.24 | X-Rob | that's a damn good idea. |
| 01:59.43 | twisted[asteria] | hooah |
| 02:02.37 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 02:02.58 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 02:03.03 | ManxPower | CrashHD, prolly in 1.2.0 |
| 02:03.11 | ManxPower | UPGRADE.txt is your frield. |
| 02:03.36 | *** join/#asterisk liew123 (n=goh_mail@60.49.6.190) |
| 02:03.40 | X-Rob_ | Mmm, frield. |
| 02:03.55 | CrashHD | upgrade.txt |
| 02:03.58 | CrashHD | I'll have to remember that |
| 02:03.59 | CrashHD | thanks |
| 02:05.18 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 02:05.24 | liew123 | hi any can help, where the asterisk database store at? |
| 02:07.11 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 02:07.35 | project_2501 | I just upgraded to asterisk 1.2.5 and now it won't start |
| 02:07.45 | project_2501 | <PROTECTED> |
| 02:08.01 | project_2501 | <PROTECTED> |
| 02:08.32 | X-Rob_ | project_2501, you didn't do what the install said |
| 02:08.42 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 02:08.44 | X-Rob_ | it said 'These modules were not installed by 1.2. Make sure they work' |
| 02:08.50 | X-Rob_ | the chan_modem ones definately don't. |
| 02:09.56 | project_2501 | hmmmm, I never seen anything in the README about that |
| 02:09.58 | *** join/#asterisk the_magic_bean (n=the_magi@c-68-58-159-114.hsd1.in.comcast.net) |
| 02:10.16 | project_2501 | I'll check it again |
| 02:11.20 | X-Rob_ | when you did 'make install' it told you that |
| 02:11.30 | ManxPower | no, UPGRADE.txt |
| 02:11.41 | ManxPower | /var/lib/asterisk/astdb |
| 02:12.20 | project_2501 | yah actually it did say something at the very end |
| 02:12.44 | ManxPower | project_2501, perhaps you should read UPGRADE.txt |
| 02:13.05 | project_2501 | ok I'll check it out |
| 02:13.10 | ManxPower | I'm reading the section about chan_modem now |
| 02:13.18 | the_magic_bean | can anyone offer some advise on my extensions.conf, I am trying to make a DID that automatically calls out to another number when it receives a call. However, it makes the call and when the channel is established about 3 seconds later the channel is dropped. |
| 02:13.26 | *** part/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 02:14.23 | ManxPower | exten => 9035551212,1,Dial(Zap/5559284) |
| 02:14.26 | ManxPower | There ya go |
| 02:14.47 | ManxPower | actually it would be |
| 02:14.51 | ManxPower | exten => 9035551212,1,Dial(Zap/g1/5559284) |
| 02:15.25 | project_2501 | it says "All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated |
| 02:15.26 | project_2501 | <PROTECTED> |
| 02:15.41 | project_2501 | <PROTECTED> |
| 02:15.41 | project_2501 | <PROTECTED> |
| 02:15.42 | ManxPower | project_2501, that's pretty clear. 8-) |
| 02:15.49 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 02:15.56 | ManxPower | project_2501, just noload them and be done with it |
| 02:16.00 | the_magic_bean | any reason that would be better than this? # |
| 02:16.00 | the_magic_bean | [depression] |
| 02:16.00 | the_magic_bean | # |
| 02:16.00 | the_magic_bean | exten => s,1,Dial(SIP/lightbound/1234567,10) |
| 02:16.00 | the_magic_bean | # |
| 02:16.00 | the_magic_bean | exten => s,2,Voicemail(u502) |
| 02:16.02 | the_magic_bean | # |
| 02:16.04 | the_magic_bean | exten => s,3,Hangup |
| 02:16.06 | the_magic_bean | # |
| 02:16.08 | the_magic_bean | exten => s,102,Voicemail(b502) |
| 02:16.10 | the_magic_bean | # |
| 02:16.11 | ManxPower | the_magic_bean, do not flood the channel |
| 02:16.12 | the_magic_bean | exten => s,103,Hangup |
| 02:16.15 | ManxPower | ~pastebin |
| 02:16.16 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 02:16.16 | the_magic_bean | # |
| 02:16.18 | X-Rob_ | ~pb |
| 02:16.20 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:16.20 | the_magic_bean | <PROTECTED> |
| 02:16.20 | the_magic_bean | # |
| 02:16.21 | ManxPower | the_magic_bean, Ah, you have SIP questions |
| 02:16.22 | the_magic_bean | [default] |
| 02:16.24 | the_magic_bean | # |
| 02:16.26 | the_magic_bean | ;Depression Hurts Phone Number |
| 02:16.28 | project_2501 | how do I no load it? in modules.conf? |
| 02:16.28 | the_magic_bean | # |
| 02:16.30 | the_magic_bean | exten => _1235554567,1,Goto(depression,s,1) |
| 02:16.32 | the_magic_bean | sorry that was ugly |
| 02:16.39 | ManxPower | project_2501, /etc/asterisk/modules.conf |
| 02:16.59 | the_magic_bean | ManxPower, yes i think so, cause the channel is established and it works fine for a few sec |
| 02:17.20 | ManxPower | the_magic_bean, perhaps you have a NAT or reinvite problem |
| 02:17.38 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:17.41 | the_magic_bean | ManxPower, we don't have any NAT |
| 02:18.18 | the_magic_bean | ManxPower, however the phones are on a lan port on the asterisk box, while the trunk SIP connected is via a wan port on the asterisk box |
| 02:18.38 | X-Rob_ | ok |
| 02:18.46 | X-Rob_ | The trick is with the new rhel stuff, Don't use GCC4 |
| 02:18.47 | ManxPower | the_magic_bean, and all the internal machines are on public ip addresses? |
| 02:18.50 | AlexCTI | Hi, i'm having a hardtime with the chanspy() function, someone can help me? |
| 02:19.16 | the_magic_bean | ManxPower, no private IP's, but there is no NAT between them and the asterisk, because it is on the same network |
| 02:19.20 | AlexCTI | I only get the beep, beep, beep ..... |
| 02:19.39 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
| 02:19.47 | ManxPower | the_magic_bean, no, asterisk is on two networks. |
| 02:19.52 | ManxPower | the_magic_bean, just turn off reinvites |
| 02:20.06 | ManxPower | AlexCTI, and NOTHING on the CLI? |
| 02:20.14 | the_magic_bean | ManxPower, Thanks, i will take a look at that |
| 02:20.36 | AlexCTI | yes.. onthe CLI y shows: - Executing ChanSpy("Zap/49-1", "scan|bg(10001)") in new stack |
| 02:20.49 | AlexCTI | ans then Playing 'beep' (language 'en') |
| 02:21.25 | ManxPower | AlexCTI, your chan prefix is "scan" |
| 02:22.16 | AlexCTI | I used scan, and Nothing |
| 02:22.20 | AlexCTI | and I get the same thing |
| 02:22.45 | ManxPower | ChanSpy(,b) |
| 02:22.46 | AlexCTI | And the active call is on Zap/1-1 |
| 02:22.48 | ManxPower | does that work? |
| 02:23.16 | AlexCTI | I put ChanSpy(scan|bg(10001)) |
| 02:23.37 | AlexCTI | with | i never used "," |
| 02:24.15 | bkw_ | woooo |
| 02:24.30 | X-Rob_ | uhoh. bkw_'s a ghost. |
| 02:24.30 | ManxPower | | and , are pretty much the same in this case. |
| 02:24.38 | russellb | they are the exact same. |
| 02:24.50 | russellb | "," gets translated into "|" before it even makes it to the application. |
| 02:25.00 | ManxPower | russellb, They are not exactly the same. |
| 02:25.08 | russellb | i know |
| 02:25.11 | russellb | but in this case they are |
| 02:25.19 | ManxPower | Yes, in this case they are. |
| 02:25.24 | russellb | :) |
| 02:25.24 | AlexCTI | so, let me putr the , |
| 02:26.23 | AlexCTI | :-S the same.. I just get the beep .. beep.. beep.. beep |
| 02:27.28 | clyrrad | AlexCTI are you just trying to listen on a specifc extension? |
| 02:27.38 | clyrrad | what are you trying to listen on? |
| 02:28.03 | bkw_ | just reading the lumenvox press release |
| 02:28.21 | bkw_ | its something that will be dead soon |
| 02:28.35 | AlexCTI | I'm try to do that i get any briged channel, on the system, but if you have the option to choose specific one... is welcome |
| 02:29.08 | X-Rob_ | bkw_, link? |
| 02:29.13 | clyrrad | I can show you how to do it with SIP and IAX, im sure it shoudl be the same for Zap (but not sure) |
| 02:29.23 | ManxPower | AlexCTI, you can work from this starting point then |
| 02:29.25 | clyrrad | what are you listening on SIP or IAX or ZAP? |
| 02:29.26 | AlexCTI | oki |
| 02:29.30 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 02:29.35 | AlexCTI | excellent |
| 02:29.59 | ManxPower | bkw_, that's why I don't follow press releases |
| 02:30.11 | ManxPower | I'll read the reviews when it's working |
| 02:30.21 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-49-128.cybersurf.com) |
| 02:31.01 | AlexCTI | So the incoming calls that what I want to hear, |
| 02:31.11 | AlexCTI | i just have SIP and ZAP |
| 02:31.37 | AlexCTI | incoming and outgoing by zap and the extensions are sip |
| 02:31.39 | clyrrad | Ok, how many digit extensions are your phones? |
| 02:31.49 | AlexCTI | 3 digits |
| 02:31.52 | clyrrad | perfect |
| 02:31.56 | clyrrad | exten => _*27XXX,n,ChanSpy(SIP) |
| 02:32.15 | clyrrad | in this case, you would dial *27101, where the extension you want to listen to is '101' |
| 02:32.35 | ManxPower | "show application chanspy" |
| 02:32.43 | ManxPower | or "show applications like spy" |
| 02:32.47 | clyrrad | while listening to that channel you can do something like 120#, which will alow you to start listening on extension 120 |
| 02:32.58 | clyrrad | same for IAX, just exten => _*27XXX,n,ChanSpy(IAX) |
| 02:33.03 | ambriento | don't forget the /${EXTEN:3} |
| 02:33.10 | clyrrad | I would assume it would be the same for ZAP |
| 02:33.23 | clyrrad | ambriento, like that he wont need the EXTEN:3 |
| 02:33.36 | AlexCTI | I think so.. let me put on my system.. |
| 02:34.13 | ambriento | that way he wont listen to a specific channels |
| 02:34.17 | ambriento | channel* |
| 02:34.23 | project_2501 | boo-yah, got 'er all working now |
| 02:34.36 | ambriento | we 'll scan all the channels starting with SIP |
| 02:34.37 | project_2501 | damn it feels good |
| 02:35.12 | clyrrad | ambriento, if he does the dial plan like exten => _*27XXX,n,ChanSpy(SIP) it will start on the XXX channel that was entered |
| 02:35.57 | clyrrad | and then he can change to any other channel (scan using *) or choose a specific one like (101#) |
| 02:36.49 | redondos | What does this mean? Mar 13 23:36:18 NOTICE[26503]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'from-sip' |
| 02:37.07 | clyrrad | it means it cant find the from-sip context |
| 02:37.40 | ManxPower | you specified context=from-sip in sip.conf, but never specified [from-sip] in extensions.conf |
| 02:37.45 | ambriento | you're rite clyrrad :) |
| 02:37.47 | redondos | sweet, thank you. |
| 02:38.22 | clyrrad | ambriento, yea i was playing with it for a while today and found thats how it works, at first its a bit strange until you see how it actually works |
| 02:38.25 | AlexCTI | clyrrad, i put on my did chanspy(zap|b) and it works, so to leave with the option o choose how it should be? |
| 02:38.49 | ManxPower | chanspy(zap|bo) |
| 02:39.05 | clyrrad | yep |
| 02:41.25 | AlexCTI | Clyrrad, in what part must be the XXX, if I leave the chanspy() function as incoming call? |
| 02:41.28 | redondos | How can I use ekiga to connect to Asterisk? I always get "user rejected the call" and the asterisk console shows nothing about it. (actually, it shows nothing at all) |
| 02:41.37 | AlexCTI | I mean starting in my DID |
| 02:41.56 | clyrrad | AlecCTI exten => _*27XXX,n,ChanSpy(SIP) |
| 02:42.26 | AlexCTI | Oki |
| 02:42.33 | clyrrad | that means you will dial *27101 to listent to extension 101, *27200 would listen to extension 200 etc |
| 02:43.03 | clyrrad | you can think of it as *27 envokes the application passing whatever extension was stored in XXX (in very simple terms) |
| 02:43.24 | ambriento | clyrrad, I'm still have some doubt about it |
| 02:43.27 | AlexCTI | thanks... i'm setting up now |
| 02:43.34 | clyrrad | np |
| 02:43.50 | bkw_ | guess its time to crack open sphinx :P |
| 02:43.54 | ambriento | lets think togheter |
| 02:44.35 | ambriento | ChanSpy(Something) will spy on any channel starting with Something |
| 02:45.09 | clyrrad | if the dial plan is as we have said yes |
| 02:45.11 | ambriento | if I dial 1234#, it will look for Something/1234 |
| 02:45.26 | clyrrad | while in the ChanSpy application yes thats how it works |
| 02:45.27 | bkw_ | its designed to do like Agent/ |
| 02:45.31 | bkw_ | so you can dial agent id's |
| 02:45.46 | bkw_ | it will also scan all SIP/ you put tha tin also |
| 02:46.10 | clyrrad | yes if you do just exten => _*27,n,ChanSpy(SIP) |
| 02:46.19 | clyrrad | that will scan all SIP starting with one first |
| 02:46.33 | clyrrad | but you can still change to another SIP channel by 1234# |
| 02:46.43 | bkw_ | FreeSwitch will have the same feature but in a better form |
| 02:46.59 | ambriento | well |
| 02:47.02 | bkw_ | I have come up with some neat ways of doing scanning |
| 02:47.05 | bkw_ | muhahaha |
| 02:47.06 | bkw_ | and call recording |
| 02:47.26 | bkw_ | some that don't involve touching the channel or latching onto it like you do in asterisk |
| 02:47.27 | clyrrad | you dont like ChanSpy and Monitor? |
| 02:47.40 | ambriento | any kind of listen to and talk to only one party? |
| 02:47.51 | bkw_ | ambriento, thats what I want |
| 02:47.55 | bkw_ | and we will have muhahahaha |
| 02:47.58 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 02:48.02 | ambriento | that would be awesome |
| 02:48.13 | ambriento | hahaha |
| 02:48.13 | bkw_ | Asterisk isn't getting that feature.. we are writing it for FreeSwitch |
| 02:48.20 | ambriento | muhahaah |
| 02:48.23 | clyrrad | oh that could be neat yea, so you could both tell the support agent what to tell the customer whith out the customer hearing |
| 02:48.28 | clyrrad | that could be useful |
| 02:48.31 | ambriento | what freeswitch is? |
| 02:48.36 | bkw_ | #freeswitch |
| 02:48.45 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 02:48.55 | bkw_ | we still use and develope internally for Asterisk |
| 02:49.17 | bkw_ | we still have this stack of patches we are waiting to give back once the outstanding ones we have now are done on the bug tracker |
| 02:49.35 | ambriento | I'll red it later |
| 02:49.47 | ambriento | its late in here must go take some sleep :) |
| 02:49.51 | ambriento | read* |
| 02:50.03 | ambriento | cya guys |
| 02:50.05 | ambriento | take care |
| 02:50.11 | clyrrad | later |
| 02:51.25 | [hC] | did freeswitch start as an asterisk fork? or a complete rewrite? |
| 02:51.54 | bkw_ | complete rewrite |
| 02:51.56 | bkw_ | ground up |
| 02:52.25 | bkw_ | brb shower time |
| 02:55.55 | *** join/#asterisk jradford (n=jradford@hoss.npl.com) |
| 02:57.47 | *** part/#asterisk jradford (n=jradford@hoss.npl.com) |
| 02:59.38 | Mavvie | hmm... funny: |
| 02:59.41 | Mavvie | Feb 14 18:29:05 DEBUG[883]: Launching 'NoOp' |
| 02:59.46 | Mavvie | The funny part of it is... |
| 02:59.49 | Mavvie | Tue Mar 14 13:59:48 EST 2006 |
| 03:00.01 | bkw_ | is your system clock wrong? |
| 03:00.10 | Mavvie | nope |
| 03:00.18 | bkw_ | smells like an off by one bug |
| 03:00.45 | Mavvie | other funny thing is that the last line in the log file says: |
| 03:00.53 | Mavvie | Mar 14 14:00:06 DEBUG[15416] chan_sip.c |
| 03:01.43 | *** join/#asterisk Qwellj2me (n=Qwell@unaffiliated/qwell) |
| 03:01.48 | *** join/#asterisk file[laptop] (n=jcolp@adsl-69-215-40-45.dsl.chcgil.ameritech.net) |
| 03:04.11 | Mavvie | oh |
| 03:04.14 | Mavvie | wrong machine |
| 03:04.43 | Mavvie | that one started december two years ago. |
| 03:06.16 | *** join/#asterisk coppice (n=chatzill@211.155.168.153) |
| 03:10.07 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
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| 03:12.46 | redondos | Why would asterisk refuse to let phones to register? |
| 03:12.58 | Mavvie | redondos: wrong authentication. |
| 03:13.28 | Mavvie | wonder what $decode does do in Mirc. |
| 03:14.13 | Mavvie | wonder what's happening on #Manila. |
| 03:14.28 | Mavvie | do I really want to get involved in script-kiddie teasing? |
| 03:14.57 | *** join/#asterisk tainted- (n=identd@ppp-71-134-51-107.dsl.irvnca.pacbell.net) |
| 03:15.09 | encode | anyone tried out http://www.ekiga.org/ yet? |
| 03:15.17 | redondos | encode: I am trying to use it. |
| 03:15.56 | encode | redondos: is it not easy to set up (i just read about it on slashdot, took a quick look at their website) |
| 03:15.57 | redondos | Please kickban agentofbsd, it's a infected with a worm spreading dubious mirc commands by pm. |
| 03:15.57 | encode | yes |
| 03:15.57 | encode | i just noticed |
| 03:16.01 | redondos | encode: I mean... yeah, it's fine. But it's not working as expected. |
| 03:16.09 | encode | ok |
| 03:16.31 | redondos | It sats "failed security check" whenever I try to make a call... who knows why,. |
| 03:16.41 | Mavvie | agentofbsd: let me guess, you wanted to become an operator on a certain channel and somebody invited you by pasting a line with a lot of random numbers in it? |
| 03:16.48 | redondos | Anyway, I'm a little new(b), so it's probably my fault. |
| 03:17.04 | redondos | great |
| 03:17.12 | encode | Mavvie: lol |
| 03:17.39 | Mavvie | still don't understand what that .load does. |
| 03:18.12 | encode | i havent got a clue either, but it's not gonna affect irssi |
| 03:19.26 | Mavvie | oh, .load is /load, but then in mirc scripting. |
| 03:19.30 | Mavvie | anyway, he's gone and I don't care. |
| 03:20.08 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 03:20.09 | redondos | Mar 14 00:19:51 NOTICE[26503]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 03:20.12 | redondos | What's that all about? |
| 03:22.23 | *** part/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
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| 03:40.36 | asterboy | Hey guys, how do I get my incoming fxo to ring a specific line on my Polycom 600? Right now it rings the first line even though my Dial(SIP/polycom_line2,10,t) is setup in extensions.conf. |
| 03:42.00 | asterboy | I would have thought it went by whatever labels/username matched in the phone setup to the Dial(SIP/label |
| 03:42.37 | talljon84 | I'm using AOH 2.7 and want to change the manager user "admin" password. If I do this, where do I need to update the new password to maintain functionality? |
| 03:43.25 | asterboy | pwd |
| 03:44.25 | PBXtech | why wont asterisk turn on the msg light on a cisco when i have VM? |
| 03:44.52 | PBXtech | * says i have VM |
| 03:45.45 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 03:49.01 | CoffeeIV | I am looking for an IAX termination/origination service. I tried to sign up for IAX.cc but they require paypal. Any other options out there ? |
| 03:49.40 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 03:52.05 | Zipper_32 | PBXtech: Configure the mailbox # in the sip.conf ? |
| 03:52.44 | PBXtech | says 5265@default |
| 03:53.22 | PBXtech | which is right |
| 03:55.39 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
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| 04:09.15 | asterboy | How do you setup an outbound SIP connection for a Polycom phone? |
| 04:09.23 | mwgbc | I'm having problems compiling zaptel. I have done it successfully on a different system, but I'm not sure what I'm doing wrong now. I am setting up * on a new installation of Debian 2.4.27-2-386 I put the output from make @ http://pastebin.ca/45565 |
| 04:10.49 | asterboy | mwgbc, any troubles compiling other stuff? Looks like some kind of incompatibility...not sure what though. |
| 04:11.16 | *** join/#asterisk xyklopz (n=xyklopx@216-91-89-21.biltmorecomm.com) |
| 04:11.39 | xyklopz | anyone had any success with pyAstre? |
| 04:11.41 | mwgbc | asterboy, no other problems so far. |
| 04:12.39 | asterboy | Do a http://clusty.org search on the first error message to see if anyone else is reporting it. |
| 04:14.07 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
| 04:15.10 | iq | mwgbc, seems like non zaptel issue |
| 04:15.33 | asterboy | Ok, so I have an extension "poly" registered with my * box, my extensions.conf file has exten = > 1,2,Dial(Zap/1) pointed from sip.conf, not sure what I'm missing but can't make an outbound sip call. |
| 04:16.04 | iq | asterboy, can you dial 500 from the phone? |
| 04:16.55 | asterboy | yes, but it does nothing |
| 04:17.09 | iq | asterboy, you do not hear anything or it fails? |
| 04:17.51 | asterboy | I'm missing the connection between the pots phone line and the SIP extension on the phone. |
| 04:17.59 | asterboy | do not hear anything. |
| 04:18.27 | iq | asterboy, using supported codecs on phone? |
| 04:18.28 | asterboy | when I type in a full number like 15551231234 it give the fast beep. |
| 04:18.35 | iq | asterboy, on same subnet? |
| 04:18.46 | asterboy | yes and I can dial the extension from another phone |
| 04:19.14 | asterboy | its a config issue with SIP somewhere on outbound calling. |
| 04:19.18 | asterboy | I should do a pastebin. |
| 04:19.50 | asterboy | The extension registers on * |
| 04:20.03 | asterboy | I can dial the extension from a pots line. |
| 04:20.12 | asterboy | Just can't dial out. |
| 04:20.54 | asterboy | How does the context work from sip.conf to extensions.conf to the Dial plan Dial(SIP,ZAP/1) ? |
| 04:20.59 | iq | asterboy, i'm sure you would've checked the context and everything |
| 04:21.39 | asterboy | wouldn't be the first time I missed some [label] necessary somewhere. |
| 04:22.26 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 04:24.17 | asterboy | Polycom Phone [poly] ---> sip.conf [poly] ----> extensions.conf [poly] ----> Dial(Zap/1)? |
| 04:25.59 | *** join/#asterisk Nivex (i=kjotte@user-0ce2nsu.cable.mindspring.com) |
| 04:29.54 | *** join/#asterisk apardo (n=apardo@87.218.45.83) |
| 04:30.29 | sevard | Grr, can anyone help me transcode an x-lite client conf to a cornfedsip_cli config |
| 04:31.42 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
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| 04:39.41 | tdonahue-laptop | hi all |
| 04:40.00 | *** join/#asterisk ravsi (n=ravsi@pool-71-108-205-127.lsanca.dsl-w.verizon.net) |
| 04:40.03 | tdonahue-laptop | can anyone recommend a good radius server for CDR collection? |
| 04:43.16 | asterboy | ok, gotit |
| 04:43.18 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:43.23 | asterboy | it was the dial plan. |
| 04:43.38 | asterboy | needed a "_9." instead of just a number. |
| 04:43.46 | *** join/#asterisk PBXtech (i=PBXtech@227.sub-70-213-223.myvzw.com) |
| 04:44.13 | asterboy | Need to readup on the differences with "_9." and just "9,1,Whatever" |
| 04:45.01 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:46.47 | asterboy | How do I make it so that I don't need the "_9." |
| 04:46.48 | asterboy | ??? |
| 04:46.59 | asterboy | Just want to pickup the line and dial straight out. |
| 04:47.35 | tdonahue-laptop | asterboy, try 1NXXNXXXXXX,1,Whatever |
| 04:47.49 | *** join/#asterisk forao (n=fjaosdfj@ool-4354d6b4.dyn.optonline.net) |
| 04:51.06 | nayyares | i am using AMP to configure extesions, what i will write in NAT to allow NATing, it has "never" by default? |
| 04:52.23 | willt | nayyares: I think it's just nat=yes |
| 04:52.31 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 04:53.05 | nayyares | willt, i have to write something in sip_nat.conf ? |
| 04:53.25 | willt | I don't think so |
| 04:53.33 | willt | I think just in sip.conf |
| 04:54.11 | willt | wait are you talking about for your asterisk box or one of your phones connecting? |
| 04:54.48 | asterboy | is there something where I can dial whatever? |
| 04:54.51 | nayyares | yes |
| 04:54.55 | asterboy | like xxx-xxxx |
| 04:55.06 | asterboy | I don't want any restrictions. |
| 04:55.39 | nayyares | i want to give access to xten softphone from my branch office to call me, i am at asterisk server. |
| 04:55.59 | willt | is the phone behind nat or the asterisk server? |
| 04:56.12 | nayyares | willt, both are behind NAT |
| 04:56.25 | nayyares | i.e. server, branch office PC |
| 04:56.40 | willt | oh.. :) well for the phone you want nat=yes in your sip.conf |
| 04:57.39 | willt | then edit sip_nat.conf |
| 04:57.44 | nayyares | i have configured it. |
| 04:58.01 | willt | and add externip = 1.2.3.4 where 1.2.3.4 is the public external ip of your asterisk box |
| 04:59.20 | nayyares | nat=1 or nat=yes are samething? |
| 04:59.41 | willt | I dunno I use nat=yes :) |
| 05:00.16 | tdonahue-laptop | asterboy, i have 3 entries in my gateway context. "1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@carrier)" "NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@carrier)" and "NXXXXXX,1,Dial(Zap/g1/${EXTEN})" |
| 05:01.02 | tdonahue-laptop | i personally shy away from the . wildcard in the dialplan... it seems to always catch stuff i don't necessarily want it to. |
| 05:01.05 | asterboy | ah ok. |
| 05:01.11 | asterboy | so it will pick up whatever. |
| 05:01.22 | ravsi | do you guys know if musiconhold a) requires mpg123 b) if I have to compile it or will a package work? |
| 05:01.27 | asterboy | I'd like just an "*" wildcard or something. |
| 05:01.30 | asterboy | but that will do. |
| 05:01.34 | nayyares | willt, do i need to restart asterisk service? how can i? |
| 05:02.00 | tdonahue-laptop | yes, you could theoretically use . but the law of unintended consequences will always rear its ugly head when you do |
| 05:02.01 | willt | just connect to the cli and do a reload |
| 05:02.08 | *** join/#asterisk FuriousGeorge (n=Brian@ool-43536ea8.dyn.optonline.net) |
| 05:02.57 | nayyares | ok |
| 05:03.00 | *** join/#asterisk JPey (n=jpoliver@r200-125-63-3-dialup.adsl.anteldata.net.uy) |
| 05:03.36 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 05:08.04 | asterboy | tdonahue, that still does not allow me to dial out. |
| 05:08.11 | asterboy | I still need the _9 |
| 05:08.17 | asterboy | then dial 9 to get out. |
| 05:08.58 | *** join/#asterisk jmacz (n=jmacz@201.244.197.240) |
| 05:10.08 | willt | anyone using the RealTime stuff in a production enviroment? |
| 05:10.38 | X-Rob_ | iaxtel uses realtime |
| 05:12.49 | willt | interesting.. just trying to get a feel for it |
| 05:13.09 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 05:13.14 | jsaunders | http://www.voip-info.org/wiki/view/Asterisk+timer |
| 05:13.50 | jsaunders | Is this true? I have to do a hack to get MeetMe working w/ out Zaptel hardware? |
| 05:14.43 | jsaunders | And which is prefered method, ztdummy? |
| 05:14.46 | willt | iaxtels website is down :( |
| 05:17.31 | NewSole | Anyone want to buy some TE411P Cards and TDM400 Cards.... |
| 05:18.21 | ManxPower | jsaunders, ztdummy |
| 05:19.46 | X-Rob_ | willt, oh, sorry, I thought you were asking for product proof rather than instructions. There's nohting on their website that says they're using it, but they are. |
| 05:20.11 | ManxPower | X-Rob_, I was not aware that iaxtel was even running 1.2 |
| 05:20.51 | X-Rob_ | ManxPower, I'm not sure they're running 1.2, but they definately are using realtime. (Well, they were, about a year ago) |
| 05:21.24 | willt | X-Rob: hmm ok.. I was just wondering if it is suitable to use in a production environment yet |
| 05:21.53 | NewSole | willt> we use it |
| 05:22.09 | willt | NewSole: how many users do you have on your system? |
| 05:22.33 | FuriousGeorge | ~realtime |
| 05:22.35 | jbot | well, realtime is http://www.voip-info.org/wiki-Asterisk+RealTime |
| 05:22.35 | NewSole | about 2-300 calls at once |
| 05:22.54 | willt | is that one machine? |
| 05:23.08 | NewSole | Quad P4 |
| 05:23.12 | asterboy | ok, so when I pickup a line on my polycom, I get a dial tone...then I have to dial 9 to get an outside line and then I get another dial tone...then I can dial. |
| 05:23.18 | asterboy | How do I get rid of that 9. |
| 05:23.19 | asterboy | ??? |
| 05:23.25 | willt | NewSole: cool |
| 05:23.45 | ManxPower | asterboy, EVERYTHING in the polycom dialplan is configured ON THE POLYCOM |
| 05:24.02 | willt | NewSole: would it be better to run two dual machine or 4 single machines than a quad machine? |
| 05:24.03 | ManxPower | The call should not even be getting to Asterisk until you are fully finished dialing the number |
| 05:24.44 | NewSole | well we dont do any transcoding.... |
| 05:25.29 | FuriousGeorge | jbot: no, realtime is a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 05:25.30 | jbot | okay, FuriousGeorge |
| 05:25.47 | FuriousGeorge | damn straight |
| 05:25.52 | NewSole | lol |
| 05:25.59 | NewSole | get it right jbot |
| 05:26.11 | FuriousGeorge | ~realtime |
| 05:26.12 | jbot | realtime is, like, a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 05:26.30 | FuriousGeorge | much better :) |
| 05:27.06 | willt | nice |
| 05:27.30 | FuriousGeorge | ~nj |
| 05:27.31 | jbot | it has been said that nj is home to the sopranos. Fogedaboudit! |
| 05:27.42 | NewSole | willt.... all our audio files are g729/g723 and since we got rid of the crappy digium cards everything purrs.... now we need the memory leaks fixed in asterisk |
| 05:27.57 | FuriousGeorge | NewSole: got a sangoma? |
| 05:28.05 | NewSole | nope |
| 05:28.09 | tsume | new jersey is a nasty place |
| 05:28.09 | FuriousGeorge | cb? |
| 05:28.16 | NewSole | VegaStream |
| 05:28.31 | FuriousGeorge | tsume: way to generalize a place that is hundreds of square miles |
| 05:28.34 | willt | NewSole: how well does the g729 work for you quality wise? |
| 05:29.15 | tsume | NewSole: what is wrong with Digium cards, except the overuse of irq access? :) |
| 05:29.38 | NewSole | vegaStream can encode/decode 4 T1's with no audio interuptions |
| 05:29.48 | NewSole | tsume.... ALOT |
| 05:30.02 | tsume | NewSole: really? what would you use for POTS lines? |
| 05:30.12 | NewSole | VegaStream |
| 05:30.20 | tsume | never heard of it |
| 05:30.27 | NewSole | it has built in T1's |
| 05:31.36 | NewSole | http://www.vegastream.com/vega400.asp |
| 05:32.58 | FuriousGeorge | newsole, what did you use before? tdm2400? |
| 05:33.18 | NewSole | TE410's and TE411's |
| 05:33.28 | NewSole | they were CRAP |
| 05:34.52 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:36.09 | NewSole | thats why I am looking to sell them.... I have 6 in total...... |
| 05:36.11 | FuriousGeorge | when they say the sangoma a201 has 128ms EchCan time, doesnt that seem a bit long? |
| 05:36.36 | NewSole | vega is instant |
| 05:36.48 | ManxPower | FuriousGeorge, Yeah. |
| 05:36.54 | NewSole | it was worth the money |
| 05:37.06 | FuriousGeorge | that leaves, what, like 370ms of additional lag b/t isp, voip provider etc |
| 05:37.18 | [hC] | They do infact have 128ms echo can, as opposed to digiums what.. 64ms? |
| 05:37.40 | NewSole | digiums has no echo can |
| 05:37.53 | [hC] | when you add the echocan module, i mean. to the tdm400p |
| 05:38.12 | FuriousGeorge | NewSole: but these are different animals, no? for someone who has to interface with, say 5 pots lines, you cant justify the cost of using a PRI |
| 05:38.34 | FuriousGeorge | [hC]: tdm2400, you mean right? |
| 05:38.39 | [hC] | no, tdm400p. |
| 05:38.51 | [hC] | \the 2400 is what, a 12 t1 capable card? or 24 or something? |
| 05:38.58 | [hC] | the tdm400p is the 4 port modular card. not t1 |
| 05:39.05 | [hC] | fxo/fxs |
| 05:39.05 | NewSole | we bough 4 of the TE411P with echo and it did not wark at all |
| 05:39.07 | FuriousGeorge | [hC]: 2400 is 24 fxs/fxo |
| 05:39.15 | [hC] | Ah. tdm400p is 4 port fxs/fxo |
| 05:39.25 | FuriousGeorge | wher as the tdm400 is only 4 |
| 05:39.30 | [hC] | Im quite happy with my sangoma a102u |
| 05:39.31 | FuriousGeorge | *where |
| 05:39.59 | [hC] | for t1. |
| 05:40.08 | [hC] | dont believe ive had echo issues with it |
| 05:40.15 | [hC] | i def. have with the tdm400p, from time to time, on fxo lines |
| 05:40.33 | FuriousGeorge | t1 is digital, so i heard there was no echo |
| 05:40.37 | [hC] | problem is for 2-6 line offices, something better is not always attainable due to price |
| 05:40.40 | FuriousGeorge | i guess i heard wrong |
| 05:40.54 | NewSole | never tried sangoma... but we had a demo vega sent to us... and after 1 week we bought 10 of them |
| 05:41.16 | [hC] | hm.. that would make sense, i suppose... however they have echo can on their t1 card, and tout it quite heavily... so i suppose its possible? |
| 05:41.18 | [hC] | I dont know. |
| 05:41.32 | [hC] | these vegas, how many ports? and whats the cost? |
| 05:42.01 | NewSole | 4 T1's on each.... and I can buy for 7k |
| 05:42.10 | [hC] | gasp |
| 05:42.13 | NewSole | Quad T1 |
| 05:42.24 | [hC] | my dual t1 sangoma was like... 800 bucks? |
| 05:42.32 | [hC] | and its got zero issues that ive had so far |
| 05:42.46 | [hC] | aside from linux's serial console support sucking ass. |
| 05:42.47 | NewSole | ya but does it encode g729/g723 off the pri |
| 05:42.51 | [hC] | and crashing the sangoma driver |
| 05:42.57 | NewSole | and do t38 faxing |
| 05:43.02 | [hC] | hah.. no.. |
| 05:43.05 | [hC] | t38 i believe it may |
| 05:43.10 | [hC] | g729 encoding on board though, no |
| 05:43.45 | NewSole | sangoma with g729/g723 cost 2200 for single T1 |
| 05:43.52 | [hC] | dont know if that justifies the addidional 700x price hike. |
| 05:44.10 | [hC] | pardon me, 7x |
| 05:44.23 | NewSole | ya but vega offers you somthing sangoma dont |
| 05:44.28 | [hC] | which is? |
| 05:44.39 | [hC] | other than what youve already told me |
| 05:44.45 | NewSole | if it breaks in 4 years.... |
| 05:44.56 | NewSole | they will replace it in 24 hours free |
| 05:45.07 | ravsi | is there any reason why background would not work? it just playing the file without stopping or doing anything when I ask for a extention , it behaves just like playback |
| 05:45.09 | [hC] | if my sangoma breaks in 4 years i can replace it myself in 24 hours and still save money |
| 05:45.13 | NewSole | and live 7/24 support |
| 05:46.50 | FuriousGeorge | [hC]: i dont know if you can call it a 7X price hike when the single T1 with the same features costs 2200 |
| 05:47.15 | FuriousGeorge | he'd need four of those for 8800 to match, am i wrong? |
| 05:47.43 | NewSole | yup |
| 05:47.53 | [hC] | and ive easily got nenad on the phone to fix something immediately for me |
| 05:48.06 | [hC] | Well, what Im saying is |
| 05:48.13 | [hC] | I paid $800 for my a102u |
| 05:48.23 | [hC] | so far the only difference i see is native g729 and (maybe) t.38 |
| 05:48.32 | NewSole | yes... but then u tax your server to transcode...... |
| 05:48.37 | [hC] | i dont know if id even pay sangoma an extra $1400 for those features. |
| 05:48.45 | [hC] | i agree its a nice feature |
| 05:48.53 | [hC] | just dont know if $1400 for native transcoding is well spent |
| 05:49.00 | [hC] | Add another cpu for 1/3 of that? heh |
| 05:49.17 | [hC] | i see the benefits of native transcoding dont get me wrong |
| 05:49.24 | [hC] | just seems like theres gotta be something else to warrant such a price spik |
| 05:49.25 | [hC] | e |
| 05:49.38 | NewSole | yes but if you dont have to transcode... you can multiply your call count by almost 200% |
| 05:51.02 | NewSole | and for us... our colocation is cheap but we pay for BW.... |
| 05:51.05 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:51.20 | NewSole | and all our device use g729/g723 |
| 05:51.31 | NewSole | so why not use them |
| 05:52.05 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 05:52.39 | NewSole | so anyone want to buy some TE411P Echo Cards and some TE410P non echo cards.... |
| 05:53.44 | NewSole | 6 Echo (800$each) and 2 non echo (400$each).... |
| 05:54.59 | NewSole | thats pretty bad.... enve the asterisk people dont want the digium crap.... lol |
| 05:56.31 | willt | is there a better way to do a call group then Dial(SIP/100&SIP/101) ? |
| 05:56.57 | FuriousGeorge | use a variable |
| 05:57.11 | FuriousGeorge | or group |
| 05:57.21 | FuriousGeorge | dial(zap/G1) |
| 05:57.31 | FuriousGeorge | well, thats not the same |
| 05:57.45 | NewSole | ya but you can not group sip devices |
| 05:58.28 | willt | so using & will not work? |
| 05:58.34 | FuriousGeorge | it will |
| 05:58.48 | FuriousGeorge | he means in a call group, but thats not quite what you want |
| 05:58.52 | *** join/#asterisk tuxinator_linux (n=zarina@166.214.53.89) |
| 05:59.19 | FuriousGeorge | there is another type of group besides that one and pickup groups though |
| 05:59.33 | FuriousGeorge | ~group |
| 05:59.34 | jbot | adduser <USER> <GROUP>; read the adduser man page. |
| 05:59.43 | willt | I just want to be able to rin several phones at once |
| 05:59.55 | willt | s/rin/ring/ |
| 06:00.15 | willt | jbot is too cool LOL |
| 06:00.18 | FuriousGeorge | ~jbo5 |
| 06:00.21 | FuriousGeorge | ~jbo6t |
| 06:00.24 | FuriousGeorge | ~jbot |
| 06:00.25 | jbot | somebody said jbot was only marginally useful at best, or a silly little bugger |
| 06:00.38 | FuriousGeorge | ~slap jbot |
| 06:00.39 | jbot | ACTION slaps jbot, keep your grubby fingers to yourself! |
| 06:00.53 | NewSole | FuriousGeorge.... |
| 06:01.32 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 06:01.43 | NewSole | who will I talk to to submit the modules I have... its a realtime provisioning system to ring multi phones at once via exten |
| 06:02.04 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:03.03 | NewSole | its modules.... FindChannel FindVoiceMail it allows you to group calls SIP and IAX and group Voicemail |
| 06:03.58 | FuriousGeorge | NewSole: im no expert, but i just ask mog everything since he works at digium |
| 06:04.09 | FuriousGeorge | hes not here :) |
| 06:04.11 | NewSole | Well its 1am time for bed... |
| 06:04.27 | FuriousGeorge | i'd be interested to know what comes of that |
| 06:04.49 | russellb | anything you would like to submit should be posted to bugs.digium.com |
| 06:04.53 | NewSole | well I may post it on bugs... |
| 06:04.59 | russellb | :) |
| 06:05.07 | FuriousGeorge | yeah listen to russel :) |
| 06:05.12 | FuriousGeorge | leave mog alone |
| 06:05.18 | NewSole | lol |
| 06:05.22 | russellb | that's all he would say ... |
| 06:05.41 | russellb | and then he'd say, now be patient and get in line |
| 06:05.43 | *** join/#asterisk freat (n=freat@h-72-244-84-46.chcgilgm.covad.net) |
| 06:06.23 | FuriousGeorge | yeah, i always harass him about asterisk-xmpp |
| 06:06.24 | russellb | hehe, we have a good problem of receiving an extremely high number of contributions |
| 06:06.36 | MikeJ[Laptop] | russellb, anyone hear from file? |
| 06:06.44 | russellb | I talked to him earlier, yes |
| 06:06.53 | russellb | his flight was delayed, wasn't supposed to get there until after 1 AM |
| 06:07.31 | ravsi | anyone know why background would just play a file? |
| 06:07.48 | russellb | isn't that what it is supposed to do? |
| 06:07.50 | ravsi | and not do anything |
| 06:07.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:07.59 | ravsi | I type in all sorts of exten |
| 06:08.07 | ravsi | and it just plays through |
| 06:08.11 | ravsi | no response |
| 06:08.16 | russellb | then it's not receiving your dtmf |
| 06:08.56 | puzzled | morning |
| 06:09.15 | MikeJ[Laptop] | you can make background just play a file and ignore dtmf |
| 06:09.25 | ravsi | mm |
| 06:09.30 | MikeJ[Laptop] | it's an option... it becomes playback |
| 06:09.31 | ravsi | I just have a basic setup |
| 06:09.44 | ravsi | but its coming from a sip phone |
| 06:09.51 | ravsi | might that have anything to do with it? |
| 06:09.55 | russellb | yes |
| 06:10.02 | MikeJ[Laptop] | ravsi, what dtmf method is the sip phone set up for |
| 06:10.13 | ravsi | let me check |
| 06:10.14 | MikeJ[Laptop] | and what do you have asterisk configured for that phone for dtmf |
| 06:10.23 | russellb | yay SIP for having *4* ways to send dtmf |
| 06:10.28 | MikeJ[Laptop] | 4? |
| 06:10.33 | MikeJ[Laptop] | try ... ummm |
| 06:10.43 | MikeJ[Laptop] | is it 6 or 7.. I can neverr remember |
| 06:11.25 | ravsi | in the phone settings its "in-audio" |
| 06:11.33 | russellb | RFC2833, Inband, SIP INFO, and some newer method that uses XML ... |
| 06:11.33 | MikeJ[Laptop] | ravsi, that's inband |
| 06:11.45 | MikeJ[Laptop] | russellb, NOTIFY |
| 06:11.52 | russellb | ah. |
| 06:11.52 | MikeJ[Laptop] | there are some more... |
| 06:11.54 | russellb | lol |
| 06:12.07 | MikeJ[Laptop] | ravsi, use 2833 if it's got it.. |
| 06:12.12 | MikeJ[Laptop] | info is good too |
| 06:12.19 | ravsi | ok |
| 06:12.21 | MikeJ[Laptop] | just make sure to set asterisk up to the same |
| 06:12.25 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:12.29 | MikeJ[Laptop] | what kind of device |
| 06:12.30 | MikeJ[Laptop] | ? |
| 06:12.53 | ravsi | budge tone 100 |
| 06:13.02 | MikeJ[Laptop] | yeah.. 2833 |
| 06:13.11 | ravsi | just set it |
| 06:13.17 | MikeJ[Laptop] | yay grandstreamm.. |
| 06:13.41 | ravsi | and it also needs to be set in sip.conf somewhere as well correct? |
| 06:13.52 | ravsi | or is 2833 defualt or somthing? |
| 06:14.15 | MikeJ[Laptop] | put it on the friend or whatever you have setup for the phone is sip.conf, yes |
| 06:16.13 | *** join/#asterisk bweschke (n=bweschke@196.sub-70-198-213.myvzw.com) |
| 06:16.54 | jsaunders | Can anyone help w/ this zaptel compiling error? http://pastebin.ca/45579 |
| 06:17.08 | X-Rob_ | jsaunders, you using centos? |
| 06:17.13 | jsaunders | yep |
| 06:17.15 | X-Rob_ | (without even clicking on the link) |
| 06:17.21 | X-Rob_ | Didn't I tell you how to fix this earlier? |
| 06:17.26 | jsaunders | Negative |
| 06:17.44 | MikeJ[Laptop] | heh |
| 06:17.48 | ravsi | does dtmfmode=auto work at all? |
| 06:18.06 | MikeJ[Laptop] | at all yes, completely, no |
| 06:18.16 | MikeJ[Laptop] | hard set it to 2833 if you can |
| 06:18.20 | ravsi | ok |
| 06:18.27 | X-Rob_ | edit /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h |
| 06:18.33 | MikeJ[Laptop] | because it will keep a dsp from being allocated |
| 06:18.40 | X-Rob_ | change rw_lock_t to rwlock_t |
| 06:18.51 | jsaunders | k |
| 06:19.04 | X-Rob_ | line 407 |
| 06:19.32 | puzzled | X-Rob_: is that with the latest CentOS kernel? |
| 06:19.39 | X-Rob_ | puzzled, yeah |
| 06:19.49 | puzzled | ah ok, that's why I have not seen it |
| 06:19.51 | jsaunders | By golly it worked. |
| 06:20.04 | jsaunders | Go X-Rob. |
| 06:20.06 | jsaunders | Tnx mang |
| 06:20.35 | ravsi | thanks for the help |
| 06:20.35 | ravsi | all |
| 06:20.35 | ravsi | it works |
| 06:20.35 | X-Rob_ | ~centosbug |
| 06:20.39 | jbot | i guess centosbug is Edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t' |
| 06:20.39 | puzzled | X-Rob_: thanks for the tip. will add another ifdef to my zaptel.spec file |
| 06:20.49 | MikeJ[Laptop] | I love it when I help! |
| 06:20.51 | MikeJ[Laptop] | :D |
| 06:21.01 | MikeJ[Laptop] | X-Rob_, heh |
| 06:21.15 | X-Rob_ | It's going to be a bloody common question |
| 06:21.32 | Zipper_32 | X-Rob_ has that answer on copy and paste... he answered it for me earlier tonight. |
| 06:22.13 | X-Rob_ | jbot, no, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 06:22.19 | jbot | okay, X-Rob_ |
| 06:22.20 | puzzled | nice asterisk demo box: http://www.theinquirer.net/?article=30258 |
| 06:22.24 | X-Rob_ | ~centosbug |
| 06:22.30 | jbot | rumour has it, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 06:22.51 | *** join/#asterisk kilobit2001 (n=locid@206-248-130-227.dsl.teksavvy.com) |
| 06:22.59 | X-Rob_ | Zipper_32, the bot knows it now 8) |
| 06:23.02 | MikeJ[Laptop] | forget centosbug... just do it as centos :P |
| 06:23.06 | X-Rob_ | hehehe |
| 06:23.14 | X-Rob_ | ~centos |
| 06:23.19 | jbot | methinks centos is better than Fedora Core |
| 06:23.24 | X-Rob_ | heh |
| 06:23.29 | X-Rob_ | most things are better than f |
| 06:23.29 | X-Rob_ | c |
| 06:23.33 | X-Rob_ | so that's not hard. |
| 06:23.44 | X-Rob_ | jbot, no, centos is better than Fedora Core (not that that's hard) |
| 06:23.48 | jbot | X-Rob_: okay |
| 06:23.48 | puzzled | what's with the gcc4 thingy |
| 06:24.01 | kilobit2001 | ne1 knows how to build an advanced hosted pbx with asterisk? with web driven extension management |
| 06:24.01 | X-Rob_ | puzzled, other crap in their kernel source won't build with gcc4 |
| 06:24.07 | MikeJ[Laptop] | jbot, no, centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 06:24.09 | jbot | MikeJ[Laptop]: okay |
| 06:24.10 | puzzled | I never had an issue compiling zaptel on centos 4.2 with gcc4 |
| 06:24.13 | Zipper_32 | puzzled: Use the same version of gcc as used to build the kernel... don't use the new gcc4 |
| 06:24.17 | MikeJ[Laptop] | :P |
| 06:24.23 | X-Rob_ | heh |
| 06:24.29 | MikeJ[Laptop] | ~centos |
| 06:24.31 | jbot | [centos] better than Fedora Core except for that silly bug, see ~centosbug for details |
| 06:24.33 | puzzled | Zipper_32: and how do I find that out? |
| 06:24.37 | MikeJ[Laptop] | hehe |
| 06:24.40 | MikeJ[Laptop] | ok.. fun |
| 06:24.44 | MikeJ[Laptop] | and now.. goodnight |
| 06:24.48 | Zipper_32 | puzzled: cat /proc/version |
| 06:24.50 | puzzled | nite MikeJ[Laptop] |
| 06:25.26 | X-Rob_ | puzzled, it's a shitfight with the new kernel |
| 06:25.27 | X-Rob_ | it's crap |
| 06:25.31 | kilobit2001 | this irc channel really sucks,.,, |
| 06:25.38 | X-Rob_ | no idea how it snuck through RH's QC |
| 06:25.40 | *** join/#asterisk mujjoo (n=mujjoo@c-67-172-211-223.hsd1.tx.comcast.net) |
| 06:25.44 | mujjoo | hello all |
| 06:25.53 | puzzled | Zipper_32: see it. thanks |
| 06:25.54 | mujjoo | have a question hope one of you can help |
| 06:25.57 | X-Rob_ | kilobit2001, you must have filtering turned on. Try Alt-F4 to fix it |
| 06:26.05 | mujjoo | my calls get hung up with the following |
| 06:26.35 | kilobit2001 | as usual.... filled with half_a$$ comment.... instead of useful infos. |
| 06:26.43 | puzzled | X-Rob_: never knew about the gcc4 issue and frankly never had issues too but nevertheless will spin some new rpms built with gcc-3.4.4 |
| 06:26.55 | X-Rob_ | puzzled, it's fine with the previous release |
| 06:26.55 | Zipper_32 | kilobit2001: Because your comment was incredibly useful. |
| 06:27.05 | X-Rob_ | I've been using Gcc4 forever. |
| 06:27.10 | puzzled | X-Rob_: release = kernel release? |
| 06:27.12 | X-Rob_ | but their new kernel is totally broken |
| 06:27.13 | X-Rob_ | yeah |
| 06:27.16 | puzzled | ah ok |
| 06:27.54 | mujjoo | NOTICE[3714] pbx_spool.c: Call failed to go through, reason 3 |
| 06:28.05 | mujjoo | anybody know what reason code 3 stands for |
| 06:28.21 | X-Rob_ | #define I_HATE_YOU 3 |
| 06:28.29 | X-Rob_ | There you go. |
| 06:29.03 | X-Rob_ | mujjoo, check /var/log/asterisk/full for what really happened. |
| 06:29.05 | mujjoo | ok you can hate me but why am i getting that |
| 06:29.05 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 06:29.12 | *** join/#asterisk tuxinator_linuxM (n=zarina@166.214.244.23) |
| 06:29.13 | mujjoo | well i am looking at that |
| 06:29.34 | mujjoo | it dials the number and after a couple of rings it gets a notify on the sip channel saying no RTP |
| 06:29.37 | mujjoo | then the call hangs up |
| 06:29.46 | mujjoo | i am using .call files to generate calls... |
| 06:29.58 | kilobit2001 | the way I see.... extensions.conf is only good for really simple ca routing.... if you are to build an advanced call routing, beyond simple call forwarding... |
| 06:29.58 | mujjoo | but if i dial using a regular phone this issue doesnt happen |
| 06:30.30 | puzzled | kilobit2001: you can do a ton of advanced call routing in extensions.conf |
| 06:30.38 | Abydos313 | anyone got a sec to talk about ata adapters |
| 06:31.10 | Zipper_32 | Abydos313: I've heard of analog telephone adapters, but never analog telephone adapter adapters... |
| 06:31.18 | puzzled | Abydos313: sure. my sipura spa-3000 is still in its box.how about yours :) |
| 06:31.19 | Abydos313 | heh |
| 06:31.33 | mujjoo | This used to work in 2.2@home release |
| 06:31.36 | Abydos313 | my card didn't go thru so i'm getting ready to reorder right now |
| 06:31.37 | Zipper_32 | My linksys PAP2 is still in its box too... |
| 06:31.38 | mujjoo | but now it doesnt :( |
| 06:31.56 | puzzled | mujjoo: iirc there is an A@H mailing list |
| 06:31.58 | kilobit2001 | how do you build extensions.conf with a web script? |
| 06:32.33 | Abydos313 | just making sure that the spa3k is the best choice. also looking at pap2-na and a few other sipuras |
| 06:32.36 | mujjoo | ok...so no one can help :( |
| 06:32.39 | puzzled | kilobit2001: check out asterisk@home. and if you can't get it to work, bug the A@H people and not here |
| 06:33.43 | X-Rob_ | Oi, puzzled, don't fob him off on me |
| 06:33.48 | mujjoo | Mar 14 00:52:04 VERBOSE[4124] logger.c: -- SIP/telasip-gw-fe1d is ringing |
| 06:33.48 | mujjoo | Mar 14 00:52:04 DEBUG[2754] chan_sip.c: Checking device state for peer telasip-gw |
| 06:33.48 | mujjoo | Mar 14 00:52:04 DEBUG[2754] devicestate.c: Changing state for SIP/telasip-gw - state 6 (Ringing |
| 06:33.49 | mujjoo | ) |
| 06:33.49 | mujjoo | Mar 14 00:52:04 DEBUG[4127] app_queue.c: Device 'SIP/telasip-gw' changed to state '6' (Ringing) |
| 06:33.49 | mujjoo | Mar 14 00:52:14 DEBUG[3139] chan_sip.c: = No match Their Call ID: 399e310e445515f44ae29f4d25652 |
| 06:33.51 | mujjoo | 2b4@67.172.211.223 Their Tag as7d49fb73 Our tag: as4fca6531 |
| 06:34.02 | mujjoo | see how the tag changes |
| 06:34.06 | mujjoo | sorry did not mean to flood |
| 06:34.16 | puzzled | pastebin please |
| 06:34.33 | puzzled | X-Rob_: hehe |
| 06:35.16 | kilobit2001 | a@h -- why did i not think of that?? |
| 06:35.51 | Zipper_32 | Is there any software out there that can validate a dialplan?... |
| 06:36.00 | Zipper_32 | A web validator perhaps? |
| 06:36.22 | puzzled | not that I know off |
| 06:37.08 | Zipper_32 | hmm, somebody should make a dialplan rip off of http://validator.wc3.org , it sure could help a lot of people with syntax errors. |
| 06:37.58 | Mavvie | Zipper_32: verbosity of 3 should give you enough warnings if it's wrong. |
| 06:38.01 | Zipper_32 | err... http://validator.w3.org/ |
| 06:38.07 | Mavvie | Zipper_32: what you're thinking about is like an ASIC tester. |
| 06:38.34 | Mavvie | Zipper_32: where you put random values on the inputs which should give proper values on the outputs. |
| 06:39.12 | Zipper_32 | Mavvie: I know if it's wrong... but I would like to know what it takes to be 'right'.... for example, I had a dialplan from 1.0.4, and a number of parts needed to be changed for it to be compatable with 1.2.X |
| 06:39.17 | kilobit2001 | zipper-- the dialplan is hand built? |
| 06:39.24 | Zipper_32 | kilobit2001: Yes, |
| 06:39.38 | kilobit2001 | thats really not the way to go. |
| 06:40.59 | Zipper_32 | Any other recommendation? |
| 06:41.36 | puzzled | Zipper_32: read the UPGRADE.txt file in the docs section |
| 06:42.01 | Zipper_32 | puzzled: Well yes, I did that in order to change the dialplan... |
| 06:42.23 | Abydos313 | screw it, i reordered the spa3k :)) |
| 06:42.36 | X-Rob_ | kilobit2001, I see you're a retard. |
| 06:42.52 | X-Rob_ | ARe you going to be hanging around in this channel for a while, or is this just a getting-less-stupid exercise? |
| 06:43.09 | puzzled | lol |
| 06:43.14 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:43.20 | Zipper_32 | I get less stupid on a regular basis. |
| 06:43.23 | puzzled | morning oej |
| 06:43.47 | oej | Morning! |
| 06:44.47 | kilobit2001 | x-rob-- as i said... half_a$$ comments... no real infos |
| 06:45.08 | Zipper_32 | kilobit2001: I'm still waiting for a suggestion, since my way is 'not the way to go' |
| 06:45.51 | Zipper_32 | kilobit2001: Because you seem to be contributing to the 'half_a$$ comments' yourself. |
| 06:46.19 | Zipper_32 | As much as I enjoy constructive criticism, the 'constructive' part seems to be lacking at the moment. |
| 06:47.30 | Mavvie | wonder why voip-info.org switches to edit mode when I double click on a word (to get it into my clipboard) |
| 06:47.39 | puzzled | kilobit2001: judging from the tone and content of your questions/comments I would expect such a response. there is no handholding here. first read all available material yourself. then asking informed questions will result in reasonable answers |
| 06:49.04 | liew123 | May I know who have experince using Java Asterisk AGI? |
| 06:49.46 | Mavvie | ast_cdr_setvar: Attempt to set a read-only variable!. |
| 06:49.48 | Mavvie | *growl* |
| 06:50.49 | kilobit2001 | liew123 -- there is no hand holding here... read all the info you can. |
| 06:50.50 | Mavvie | brilliant. |
| 06:51.46 | liew123 | okie |
| 06:52.13 | puzzled | liew123: not me but there is a mailing list here: http://www.asteriskjava.org/latest/ |
| 06:52.15 | Zipper_32 | X-Rob_ held my hand several times today. I think we had a moment. |
| 06:52.47 | puzzled | hehe |
| 06:53.17 | kilobit2001 | again... half_a$$ comments.... then a "hehe" from someone else. |
| 06:53.43 | Mavvie | kilobit2001: nice out-of-context-quoting. |
| 06:55.00 | Zipper_32 | kilobit2001: I was hoping that my satirical response of your uncooperative behaviour would at least brighten your mood. But to call it 'half_a$$'ed... I worked hard at it. |
| 06:56.22 | kilobit2001 | child |
| 07:01.17 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 07:06.39 | *** part/#asterisk mujjoo (n=mujjoo@c-67-172-211-223.hsd1.tx.comcast.net) |
| 07:07.51 | liew123 | puzzled: thanks I have subcribe I just think to know how to pass paramter to agi. Any way I will try my best find it. |
| 07:08.06 | puzzled | liew123: there are agi examples in the asterisk src and there is info at voip-info.org and the asterisk book |
| 07:12.14 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 07:12.29 | jsaunders | Can someone help w/ this MeetMe related problem? http://pastebin.ca/45595 |
| 07:12.31 | jsaunders | plz |
| 07:12.36 | liew123 | puzzled: ic |
| 07:14.28 | puzzled | jsaunders: search the list archives for TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN. there was talk about that a while back. look at line 8. seems a var is empty which shouldn't be |
| 07:21.03 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net) |
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| 07:50.43 | Mavvie | oej: at this moment I'm fighting with CDRs. Give me a weekend. |
| 07:50.56 | oej | :-) |
| 07:51.00 | oej | Will do |
| 07:51.15 | oej | Just wanted it not to disappear to the bottom of the bug tracker |
| 07:51.27 | Mavvie | oej: biggest problem for "let's fix this quickly" is btw that my test machine has become a production machine. |
| 07:51.41 | Mavvie | so I can only do this in the weekends and then in the evenings. |
| 07:51.53 | Mavvie | or during weekdays and then between 00:00 and 05:00 |
| 07:52.01 | Mavvie | don't you love live? |
| 07:52.04 | Mavvie | life |
| 07:54.41 | oej | That happens |
| 07:54.56 | oej | My old development machine got so many users that I had to |
| 07:55.08 | oej | move somewhere else to be able to go wild with the code |
| 07:55.25 | Mavvie | that reminds me, maybe I should move it to a xen-based host. |
| 07:55.47 | Mavvie | wonder if they can access the zaptel cards. |
| 08:00.14 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 08:00.21 | masked | hello |
| 08:00.34 | masked | whats the default sip audio port(s)? |
| 08:01.38 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 08:01.50 | Mavvie | masked: there is a range for that, defined in asterisk/rtp.conf |
| 08:02.12 | Mavvie | I don't know if there is an IANA approved range for it. |
| 08:02.22 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 08:04.19 | masked | ok but im not using asterisk at the minute |
| 08:04.33 | Mavvie | aha. |
| 08:04.35 | masked | just interested in the range so i can set my QoS |
| 08:05.23 | masked | can you post me a list? |
| 08:05.33 | *** join/#asterisk azpbxguy (n=chatzill@ip68-2-209-210.ph.ph.cox.net) |
| 08:07.11 | masked | Mavvie, ? |
| 08:07.29 | *** part/#asterisk azpbxguy (n=chatzill@ip68-2-209-210.ph.ph.cox.net) |
| 08:07.44 | *** join/#asterisk tuxinator_linux (n=zarina@142.131.190.116) |
| 08:07.44 | masked | brb |
| 08:09.05 | *** join/#asterisk Altair256 (n=icechat5@tn-greenback1a-30.rhmdky.adelphia.net) |
| 08:09.07 | Mavvie | masked: do you have a single phone or a range? |
| 08:09.10 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 08:09.38 | *** join/#asterisk Qwell[laptop] (n=north@unaffiliated/qwell) |
| 08:09.47 | tuxinator_linux | Hey, it's the funny Qwell[laptop] |
| 08:10.04 | Qwell[laptop] | indeed |
| 08:10.16 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 08:11.45 | Mavvie | masked: do you have a single phone or a range? |
| 08:11.47 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-178.claranet.co.uk) |
| 08:11.55 | Mavvie | masked: for example, my grandstream has a single port. |
| 08:12.01 | Mavvie | masked: but my asterisk server has a whole range. |
| 08:13.30 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 08:13.34 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
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| 08:20.19 | masked | Mavvie, a singal phone |
| 08:20.55 | masked | Mavvie, that is directly connected to my itsp, so im unsure what port ill end up using :S |
| 08:21.10 | masked | single* |
| 08:21.11 | Mavvie | masked: aha, that way. |
| 08:21.19 | Mavvie | masked: my best guess would be 5004 |
| 08:21.36 | Mavvie | mostly because that's the default on my grandstream |
| 08:22.14 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:22.21 | masked | i thought they started in the 10000's? |
| 08:22.33 | masked | its a zyxel wifi voip phone |
| 08:23.28 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:26.27 | Zeeek | hej! |
| 08:27.36 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 08:30.30 | *** join/#asterisk rharfield (i=Dacat@68-190-18-68.dhcp.mtgm.al.charter.com) |
| 08:30.49 | *** part/#asterisk Altair256 (n=icechat5@tn-greenback1a-30.rhmdky.adelphia.net) |
| 08:31.39 | rharfield | Is anyone familiar with the COP (Control Operator) functions of app_rpt? I'm having problems getting cop,6 to work. |
| 08:35.07 | *** join/#asterisk Qwell[] (n=Qwell[]@unaffiliated/qwell) |
| 08:35.14 | *** join/#asterisk core-ix (n=ivo@pirus.securax.be) |
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| 08:39.48 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 08:43.18 | nayyares | hi |
| 08:43.21 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 08:45.05 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 08:46.10 | nayyares | if we add extension using AMP, where it store this configuration, i mean file name? |
| 08:48.13 | tzafrir | nayyares, basically in a "flat" table in the mysql db |
| 08:48.30 | tzafrir | The config files are basically generated from those tables |
| 08:49.40 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 08:51.54 | Zeeek | m |
| 08:57.22 | nayyares | i have configured SIP behind NAT by redirecting all in comming traffic to port 5060,10000-20000 to my Asterisk server, same configuration is done at our branch office, but when i try to call branch office it give me "TIME OUT" error? |
| 09:16.34 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
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| 09:33.13 | zoa | hey ho |
| 09:34.29 | Zeeek | ya yo |
| 09:37.07 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 09:39.34 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 09:39.49 | *** join/#asterisk ^HeLL^ (n=admin@232.Red-83-42-51.dynamicIP.rima-tde.net) |
| 09:39.54 | ^HeLL^ | hello all |
| 09:39.59 | brookshire | hey |
| 09:40.08 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 09:42.53 | zoa | hey brookie |
| 09:48.30 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 09:48.40 | *** part/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 09:49.35 | brookshire | zoa |
| 09:49.37 | brookshire | ! |
| 09:49.44 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 09:49.44 | brookshire | did you see digium's new website? |
| 09:49.49 | brookshire | qwell! |
| 09:49.53 | brookshire | goto bed :) |
| 09:50.25 | Qwell[] | umm |
| 09:50.27 | Qwell[] | no bed |
| 09:50.30 | Qwell[] | VON! :p |
| 09:50.32 | brookshire | lol |
| 09:50.48 | brookshire | qwell: we put new website up for von |
| 09:50.51 | brookshire | go look :) |
| 09:50.54 | brookshire | http://www.digium.com |
| 09:51.01 | Qwell[] | lookin |
| 09:51.02 | Qwell[] | g |
| 09:51.08 | brookshire | are you drunk? |
| 09:51.09 | zoa | looking good! |
| 09:51.14 | Qwell[] | tomorrow |
| 09:51.36 | coppice | the girl with blue eyes and black hair looks really fake |
| 09:51.46 | Qwell[] | pretty... |
| 09:51.51 | brookshire | they are green silly |
| 09:51.52 | brookshire | :) |
| 09:51.54 | Qwell[] | site, that is |
| 09:52.48 | brookshire | zoa: http://www.digium.com/en/supportcenter/resources/community.php |
| 09:52.50 | zoa | its not a good idea to host kiddie porn on the website |
| 09:52.53 | Qwell[] | digium.com account != asterisk.org account? |
| 09:53.04 | brookshire | huh? |
| 09:53.12 | zoa | j/l |
| 09:53.13 | zoa | j/k |
| 09:53.29 | zoa | where is this girl with the blue eyes ? |
| 09:53.54 | brookshire | i guess that was the old girl |
| 09:54.04 | brookshire | we're done with her, on to the new :D |
| 09:54.23 | puzzled | wot, no blonde |
| 09:54.26 | brookshire | no.. digium.com account does is not an asterisk.org site |
| 09:54.50 | brookshire | okay.. i'm tired |
| 09:54.52 | brookshire | i goto bed now |
| 09:55.15 | coppice | if you want a blonde, I can pose |
| 09:55.28 | brookshire | :) |
| 09:55.29 | glm2k | lol |
| 09:55.39 | glm2k | you have silver grey hair coppice, no can do lol |
| 09:55.42 | brookshire | send in your pic :D |
| 09:55.56 | glm2k | or was it golden grey? |
| 09:55.57 | puzzled | coppice: hehe |
| 09:56.26 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net) |
| 09:56.46 | coppice | there is a bit of grey, but most of it is still blonde |
| 09:57.19 | coppice | its scarcity in the central area, rather than colour, which has really changed it |
| 09:57.28 | glm2k | i hear ya hehe |
| 09:58.01 | glm2k | real men lose hair due to extra high levels of testosterone :) |
| 09:58.54 | glm2k | there was speculation that it was radiation from monitors but meh. monks lost hair in the middle ages... |
| 09:59.26 | puzzled | thanks <$deity> for tft then |
| 09:59.26 | coppice | due to their intense sexual activity, no doubt |
| 09:59.36 | glm2k | rotfl! |
| 10:01.00 | coppice | "Hey monk. What sexual activity have you had today?" |
| 10:01.02 | coppice | "Nun" |
| 10:01.09 | puzzled | hahaha |
| 10:01.31 | brookshire | :/ |
| 10:02.15 | glm2k | hahaha |
| 10:04.40 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 10:04.52 | Faithful | I don't know what I have done but now I keep getting like 4sec dropouts over Zap incoming??? |
| 10:05.14 | Faithful | I have been messing with my extension.conf but thats all basically |
| 10:08.00 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 10:08.03 | zoa | coppice: lol |
| 10:08.05 | zoa | ah too bad |
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| 10:17.50 | Zeeek_ | wtf |
| 10:18.28 | ambriento | companenents ?? |
| 10:18.49 | Zeeek_ | I'm confused |
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| 10:29.25 | MGSsancho | 2:30am in los angeles |
| 10:29.29 | MGSsancho | time for me to sleep |
| 10:32.50 | *** join/#asterisk fulgas (n=fulgas@209.8.233.207) |
| 10:35.27 | Aze` | Anyone have idea when digium distrib bri cards ? |
| 10:36.38 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:36.40 | backblue | hi |
| 10:36.55 | Zeeek_ | lo |
| 10:37.33 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 10:38.15 | astra^^ | can anyone tell me wher do i gt to buy licence for g729 .exact link |
| 10:38.22 | astra^^ | ? |
| 10:38.37 | Zeeek_ | digium.com |
| 10:39.07 | astra^^ | how much do i have to pay for tat i have 30 lines.. ? |
| 10:39.12 | Zeeek_ | <PROTECTED> |
| 10:39.33 | Zeeek_ | astra^^ write and ask them |
| 10:41.16 | astra^^ | k.. |
| 10:42.55 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
| 10:43.02 | Strom_C | I'm the square from Adventure |
| 10:44.13 | *** join/#asterisk Lino` (n=Lino@i577BDFBF.versanet.de) |
| 10:47.44 | Zeeek_ | in Hollywood squares? |
| 10:48.31 | Strom_C | sigh, no |
| 10:48.39 | Strom_C | obviously you never had an atari 2600 |
| 10:49.01 | Zeeek_ | no, TRS-80 |
| 10:51.04 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
| 10:51.09 | blkremedy | hello |
| 10:51.12 | Lino` | hi |
| 10:51.16 | Lino` | ~seen Possible |
| 10:51.24 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 3d 22h 32m 3s ago, saying: 'I guess not'. |
| 10:51.24 | Zeeek_ | jello |
| 10:51.26 | Strom_C | waffle giblets |
| 10:51.43 | Zeeek_ | gibson onions |
| 10:51.56 | RoyK | ~seen me |
| 10:51.57 | jbot | me <n=SportChi@ca-redbch-cuda1-c3b-a-40.stmnca.adelphia.net> was last seen on IRC in channel #bz-inc, 11d 19h 32m 30s ago, saying: 'autojoin :)'. |
| 10:52.07 | RoyK | ~seen elvis |
| 10:52.08 | jbot | elvis <n=sdad@ipd50a583c.speed.planet.nl> was last seen on IRC in channel #debian, 30d 21h 17m 25s ago, saying: 'is there a combined package on debian to install all perl modules?'. |
| 10:52.37 | Zeeek_ | ~seen the_troubles_i've_seen |
| 10:52.39 | jbot | Zeeek_: i haven't seen 'the_troubles_i've_seen' |
| 10:52.48 | Zeeek_ | muhahahah |
| 10:52.48 | blkremedy | question...I'm in the military stationed in Japan using a@h 2.4. I'm having trouble finding weather in txt format for festival. Can someone point me in the right direction? |
| 10:53.06 | RoyK | ~seen Zeeek_'s brain |
| 10:53.07 | jbot | i haven't seen 'zeeek_'s brain', RoyK |
| 10:53.59 | Zeeek_ | blkremedy someone wrote an app already |
| 10:54.40 | Lino` | hmmm |
| 10:54.50 | Zeeek_ | try looking here: http://www.voip-info.org/tiki-index.php?page=asterisk+at+home+festival+weather+configuration |
| 10:56.11 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 10:59.22 | blkremedy | Thanks I'll check it out |
| 10:59.54 | RoyK | interesting.... etiga.org |
| 11:00.45 | puzzled | RoyK: you mean ekiga.org? |
| 11:00.54 | RoyK | erm. yes |
| 11:00.55 | RoyK | sori |
| 11:06.16 | Mavvie | RoyK: let me know if you get opal compiled :-/ |
| 11:06.18 | *** join/#asterisk vira (n=drake@c-24-125-100-198.hsd1.va.comcast.net) |
| 11:06.35 | vira | how can i use the asterisk console as a softphone? |
| 11:11.11 | astra^^ | anyone knows wher i can host my asterisk. any data center.. ? |
| 11:11.22 | Lino` | which part of the world? |
| 11:11.45 | astra^^ | us? |
| 11:11.53 | Lino` | ah kk i dunno |
| 11:12.01 | Strom_C | astra^^, I have one in colo |
| 11:12.05 | RoyK | Mavvie: ? |
| 11:12.16 | astra^^ | ur charge..? |
| 11:12.28 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
| 11:12.31 | Strom_C | you'll have to speak English; I don't know what you mean |
| 11:12.32 | astra^^ | specifications..? |
| 11:12.44 | puzzled | vira: you need to have sound support on the pc where you built asterisk and run it. if you do not have the Dial command in the asterisk console than you do not have it compiled in or your box does not have sound support |
| 11:12.45 | Lino` | price...? |
| 11:13.04 | astra^^ | yes. |
| 11:13.41 | Strom_C | .......? |
| 11:13.47 | puzzled | no. |
| 11:13.47 | Strom_C | something resembling complete sentences would be nice |
| 11:14.00 | Zeeek_ | I think 2 |
| 11:14.09 | puzzled | Zeeek: that would be 42 |
| 11:14.10 | Zeeek_ | but not |
| 11:14.19 | vira | puzzled: ah, i do have sound support, but i installed asterisk via the debian stable package (which includes version asterisk version 1.0.7) |
| 11:14.23 | Zeeek_ | char(@) + 2 |
| 11:14.34 | puzzled | vira: I don't know anything about debian |
| 11:14.38 | astra^^ | actually i need to host a server in some data center coz here b/w is touchin the stars |
| 11:14.40 | Zeeek_ | IntVal('@') |
| 11:14.45 | puzzled | segfault |
| 11:15.06 | Zeeek_ | there are lists on the wiki |
| 11:15.16 | astra^^ | k' |
| 11:15.34 | vira | ok, anyone know whether the most recent asterisk works ok in debian? |
| 11:17.08 | Strom_C | vira, works for me |
| 11:17.31 | vira | strom, you running 3.1 stable? |
| 11:18.56 | Strom_C | I have a client running asterisk stable on sarge stable |
| 11:19.01 | Strom_C | no problems |
| 11:19.33 | Zeeek_ | ok, all the girls have left the office : should I upgrade from 1.2 to 1.2.5 ? |
| 11:19.38 | vira | strom: cool, thanks |
| 11:19.46 | tzafrir | vira, what do you want to know about Debian? |
| 11:20.07 | *** part/#asterisk X-Gen (n=x-gen@dsl-145-243-163.telkomadsl.co.za) |
| 11:21.12 | vira | tzafrir: i'm trying to run asterisk on an old laptop that is too slow for both asterisk and a softphone, so i'm trying to use asterisk itself to dial and all... but it seems the debian stable asterisk package does not include the 'dial' command |
| 11:21.39 | Strom_C | vira, oh hell, install debian stable and then compile asterisk stable from source |
| 11:21.49 | tzafrir | vira, "dial" is part of chan_oss or chan_alsa |
| 11:22.10 | tzafrir | Those are not loaded by default. What kernel do you have? |
| 11:22.23 | vira | when i tried "load chan_alsa" it did this: |
| 11:22.26 | vira | <PROTECTED> |
| 11:22.28 | tzafrir | (not loaded by default as they tend to get the sound card exclusivly) |
| 11:22.29 | vira | <PROTECTED> |
| 11:22.43 | vira | but then nothing for a few minutes, so i hit ^C to kill it |
| 11:22.47 | Strom_C | do you have ALSA installed? |
| 11:22.57 | vira | i do have alsa, and sound does work elsewhere |
| 11:23.12 | vira | kernel is 2.6.8-2-386 |
| 11:23.13 | tzafrir | Without that module loaded, can you play sounds? |
| 11:23.19 | Strom_C | well, I guess you could always wave a dead hooker at it and see if that works |
| 11:23.33 | tzafrir | using , e.g., play? |
| 11:23.36 | vira | i can play sounds with mpg321 outside of asterisk |
| 11:38.30 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 11:40.35 | *** join/#asterisk vimman (n=codeshep@125.16.130.66) |
| 11:42.53 | vimman | is there a billing program that I can interface with asterisk to show call charges and records.. |
| 11:46.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:46.16 | vira | oh sweet, it works |
| 11:46.25 | vira | with the usb headset anyway |
| 11:48.12 | vira | wow asterisk is hella lightweight |
| 11:50.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:50.44 | vira | i heard that if a dns request on an iax address ever fails, it will never work again.. is that true? |
| 11:51.04 | Zeeek_ | explain? |
| 11:51.28 | vira | like if it tries to register with digium.com, but then the lookup fails |
| 11:51.54 | Zeeek | and you mean fails until a reboot? |
| 11:52.18 | Zeeek | or restart |
| 11:52.24 | vira | i dunno |
| 11:52.39 | vira | that is the extent of the rumor that i heard :) |
| 11:52.49 | Zeeek | well, in the old days, the dns lookups took place once at startup. I don't think that is the case nowadays |
| 11:53.02 | vira | old days = 1.0.7? |
| 11:53.13 | Zeeek | probably |
| 11:53.22 | vira | ah... that's what i'm using here |
| 11:53.30 | Zeeek | you'd have to actually read the changes file to know |
| 11:54.02 | Zeeek | never work again is probably not an accurate expression of reality though |
| 11:54.22 | vira | ok |
| 11:54.33 | Zeeek | "never work again" = "weapons of mass destruction" |
| 11:54.46 | vira | well something like that seems to have happened on my netbsd installation of asterisk... all of a sudden all the iax stuff just stopped working for no reason |
| 11:54.57 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:55.01 | Zeeek | all accounts ya mean? |
| 11:55.03 | vira | yeah |
| 11:55.13 | Zeeek | can you ping them? |
| 11:55.16 | vira | yup |
| 11:55.21 | Zeeek | did you restart? |
| 11:55.23 | vira | yes |
| 11:55.25 | Zeeek | or reload iax2 ? |
| 11:55.26 | vira | also rebooted |
| 11:55.39 | Zeeek | did you replace the mobo and CPU and add RAM? |
| 11:55.46 | Zeeek | well, then... |
| 11:56.03 | vira | lol |
| 11:56.17 | vira | i'm not there to debug it right now.. i'll mess with it later |
| 11:56.22 | Zeeek | but I don't go with never again. I'd imagine "until restart" maybe |
| 11:56.53 | Zeeek | for example, I had an IAXy on a dynamic ip and this was with * 0.9 or something |
| 11:57.07 | Zeeek | if the ip changed, naturally it became unreachable |
| 11:57.11 | backblue | oej: ping? |
| 11:57.31 | Zeeek | haven't seen him today |
| 11:58.02 | vimman | hey.. I had compiled asterisk 1.2.5 in my machine.. how do I get it up.. what should I do next.. some links.. |
| 11:58.16 | Zeeek | Starter tutorial: |
| 11:58.16 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 11:58.17 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 11:58.17 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 11:58.17 | Zeeek | THE reference of the moment: |
| 11:58.17 | Zeeek | http://www.asteriskdocs.org |
| 11:58.50 | Zeeek | all very old links |
| 11:59.02 | vimman | thanks a lot Zeeek |
| 11:59.24 | PakiPenguin | Zeeek, sup? |
| 11:59.39 | Zeeek | picking my nose, mostly |
| 12:00.08 | Zeeek | but not in front of the webcam |
| 12:00.08 | PakiPenguin | :p |
| 12:00.27 | Zeeek | which actually is focused on me right now |
| 12:01.03 | RoyK | if phone A rings in the office, is it somehow possible to pickup that from somewhere else in case phone A's owner is out? |
| 12:01.11 | Zeeek | heh, password protected |
| 12:01.20 | PakiPenguin | Zeeek, hacked :p |
| 12:01.25 | Zeeek | damn! |
| 12:02.09 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 12:02.59 | Zeeek | fortunately we have 2 ip and the camera isn't on the same one as me |
| 12:03.27 | PakiPenguin | :) |
| 12:04.11 | Zeeek | I have a bass player friend whi looks exacly like Ben Laden. I really must get a phone of us together |
| 12:04.48 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 12:05.10 | PakiPenguin | foto :p |
| 12:05.27 | Zeeek | not phone, photo indeed |
| 12:09.09 | Zeeek | Paris |
| 12:17.39 | *** join/#asterisk htims (n=pd@Vf802.v.pppool.de) |
| 12:18.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 12:19.39 | *** join/#asterisk ckruetze (n=ckruetze@62.214.75.2) |
| 12:20.15 | Zeeek | hey there, hi there, ho there |
| 12:20.22 | ckruetze | hi |
| 12:21.01 | ckruetze | I had a look at the Digium booth at CeBit yesterday - I think I saw one booth somewhere that was smaller then the Digium one. |
| 12:21.28 | Zeeek | The Microsoft open source booth? |
| 12:21.33 | I-MOD | lol |
| 12:21.45 | Zeeek | nyuk, nyuk |
| 12:22.16 | Zeeek | I'm so bored I laugh at my own jokes |
| 12:22.28 | Zeeek | I gotta get outta here |
| 12:22.53 | Zeeek | somebody please, say something hilarious |
| 12:22.56 | PakiPenguin | Zeeek, fix my server :p |
| 12:23.11 | Zeeek | I can install Win 3.11 |
| 12:23.17 | Zeeek | that should fix it |
| 12:23.35 | PakiPenguin | good! |
| 12:23.36 | PakiPenguin | :) |
| 12:23.37 | PakiPenguin | hehe |
| 12:23.56 | ckruetze | Zeeek: Actually I think Microsofts open source corner was slightly larger |
| 12:24.02 | Zeeek | the sky is blue, but it's like 3°C. That sucks |
| 12:24.24 | Zeeek | ckruetze I hear it's huge in CVhina :) |
| 12:25.04 | Zeeek | I'm so border I'm reading the spamcop auto reports |
| 12:25.19 | Zeeek | "She wants it bigger" |
| 12:27.56 | ckruetze | Zeeek CeBit is huge, over 6000 exhibitors and over 300.000 sqm exhibition space |
| 12:28.19 | Zeeek | almost considered going this time, but still havent ever been |
| 12:29.19 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:30.44 | ckruetze | You are missing out |
| 12:31.34 | PakiPenguin | ckruetze, any live feed? |
| 12:32.52 | Zeeek | Luis Garr said he didn't hear the engine but heard the splash as the plane "kind of landed into the water." |
| 12:34.46 | ckruetze | PakiPenguin I don't know, probably, but since I went I didn't look for any |
| 12:35.19 | Zeeek | did you bring home any neat stuff like a mouse flashlight or screwdriver sets? |
| 12:35.51 | Zeeek | Digium has nice pens and screwdrivers |
| 12:36.00 | Zeeek | and calculators |
| 12:37.06 | Zeeek | for Katty: http://ak.imgfarm.com/images/today/creators/bmp/bmp0314g.gif |
| 12:37.56 | RoyK | hehehehehe |
| 12:38.16 | Zeeek | now for RoyK : http://ak.imgfarm.com/images/today/creators/wio/wio0314g.gif |
| 12:40.41 | RoyK | :P |
| 12:41.51 | Zeeek | and in the "she's a real hog" department: http://ak.imgfarm.com/images/gossip/celebrities/0214spears.jpg |
| 12:41.53 | *** join/#asterisk htims (n=pd@Vf802.v.pppool.de) |
| 12:41.57 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 12:42.22 | PakiPenguin | omfg imgfarm is blocked too |
| 12:42.48 | Zeeek | office or govt ? |
| 12:43.07 | Zeeek | http://thecloak.com |
| 12:43.40 | PakiPenguin | govy |
| 12:43.43 | PakiPenguin | govt* |
| 12:46.22 | Zeeek | try this one: http://www.wireimage.com/ |
| 12:47.15 | vira | hmm |
| 12:48.26 | vira | anyone see this message before? |
| 12:48.34 | vira | Mar 14 07:44:43 WARNING[6954]: chan_alsa.c:704 alsa_indicate: Don't know how to display condition 14 on ALSA/hw:1,0 |
| 12:50.18 | vira | this is when i'm trying to dial an 800 number through free world dialup.. i can dial FWD users fine |
| 12:50.43 | vira | but when i try the 800 number, it says "Console call has been answered" but doesn't ring or anything |
| 12:54.37 | fugitivo | ~seen coppice |
| 12:54.45 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 2h 53m 43s ago, saying: '"Nun"'. |
| 12:55.08 | mutilator | man last night sucked |
| 12:55.11 | mutilator | power went out at 11 |
| 12:55.14 | mutilator | still isn't back on |
| 12:55.14 | *** join/#asterisk shiznatix (n=Bambr@213-35-239-33-dsl.end.estpak.ee) |
| 12:55.22 | mutilator | my alarm was a freezing cold house |
| 12:59.02 | tzafrir | vira, any chance that alsa is busy or whatever by another app (or maybe another asterisk instance)? |
| 13:04.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:04.40 | *** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
| 13:05.31 | pycsusz | Hi Everybody! Somebody can help me in "Asterisk to Asterisk on SIP" topic? |
| 13:05.35 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:05.40 | Zeeek | ask away! |
| 13:06.41 | pycsusz | well I would like a call to transfer from X asterisk server to Y asterisk server |
| 13:07.02 | pycsusz | Y asterisk server register perfectly to X asterisk server |
| 13:07.08 | mutilator | uh |
| 13:07.10 | mutilator | use iax for that.. |
| 13:07.21 | Zeeek | switch |
| 13:07.33 | pycsusz | ok thanx |
| 13:07.41 | Zeeek | or hire someone in the third world to run the servers |
| 13:07.46 | *** join/#asterisk lunaphyte_ (n=lunaphye@c-71-193-101-146.hsd1.mi.comcast.net) |
| 13:07.59 | FlyboySR22 | Good Morning Everyone |
| 13:07.59 | mutilator | man |
| 13:08.01 | pycsusz | :) |
| 13:08.05 | mutilator | wind out here is horrible still |
| 13:08.10 | mutilator | was 40mph last night |
| 13:08.15 | mutilator | and it;s still really strong |
| 13:08.17 | Zeeek | and temperature? |
| 13:08.18 | FlyboySR22 | mutilator, where are you at..? |
| 13:08.19 | vira | tzafrir, no chance.. it works fine when i dial a FWD user or something |
| 13:08.20 | mutilator | looks like siberia |
| 13:09.12 | mutilator | 23F |
| 13:09.23 | mutilator | wubdis 20, gusting to 30mph |
| 13:09.26 | mutilator | winds* |
| 13:09.47 | mutilator | i still don't have power at home |
| 13:09.54 | mutilator | gf is sittin there freezin her butt off |
| 13:10.14 | Zeeek | well there are things you could do to keep warm |
| 13:10.25 | mutilator | we did those things last night after it went out |
| 13:10.37 | Zeeek | only for a few minutes though |
| 13:10.39 | *** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au) |
| 13:10.48 | mutilator | well it was going on before and after |
| 13:10.48 | mutilator | so |
| 13:10.52 | Zeeek | heh |
| 13:11.29 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 13:12.22 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:12.35 | Zeeek | A Broadway play for geeks: "CAT5!" |
| 13:12.45 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:13.29 | Zeeek | ok |
| 13:13.35 | Zeeek | no drum roll |
| 13:13.49 | mutilator | bah dum pshhhh |
| 13:14.43 | Zeeek | I think it's about 38°F here |
| 13:14.57 | Zeeek | and I'm starting to think about going out |
| 13:16.03 | Zeeek | hmmmm they claim it's 42 Latest 7 AM (12) Mar 14 42 (6) 21 (-6) 30.18 (1022) E 6 |
| 13:17.11 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:18.07 | oej | backblue: pong |
| 13:18.19 | Zeeek | bad lag problem ! |
| 13:18.31 | Zeeek | oej is (all but) UNREACHABLE |
| 13:18.43 | Zeeek | 6355463524ms |
| 13:19.07 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-79.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 13:22.55 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 13:23.29 | RoyK | Zeeek: 73 days of lag? |
| 13:25.02 | Zeeek | <PROTECTED> |
| 13:26.19 | oej | Back |
| 13:26.27 | oej | :-) |
| 13:28.53 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:28.53 | ^HeLL^ | w |
| 13:30.28 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 13:30.43 | kippi | to make a ringing group, do I need to create a queue? |
| 13:31.12 | kippi | sorry, hunt group |
| 13:31.33 | jsharp | One extension rings multiple phones simultaneously? No. |
| 13:31.45 | *** join/#asterisk ddanier (n=ddanier@p5498FB57.dip.t-dialin.net) |
| 13:31.50 | kippi | one DDI that rings 5 phones |
| 13:32.11 | mutilator | just put each node in your dial |
| 13:32.15 | jsharp | exten => 1234,1,Dial(SIP/foo1&SIP/foo2&SIP/foo3&SIP/foo4&SIP/foo5) |
| 13:32.22 | mutilator | ya |
| 13:32.23 | mutilator | like that |
| 13:32.26 | kippi | that simple? |
| 13:32.32 | jsharp | That makes extenion 1234 ring foo1 through foo5 simultaneously. |
| 13:32.35 | jsharp | That simple |
| 13:32.51 | jsharp | And the first person to answer the call gets the call and nobody else. |
| 13:33.41 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 13:36.46 | ddanier | I have one question about the licensing of asterisk: As it is dual-licensed I may use the GPL-code and change it. Now if I send a patch to Digium with my changes (which are released under the GPL as they must be), is it possible for Digium to include this code in their commercial version of asterisk? |
| 13:36.58 | tzanger | ddanier: no |
| 13:37.06 | tzanger | ddanier: you need to disclaim any code you wish to contribute back |
| 13:37.39 | tzanger | ddanier: Digium cannot take your GPL code/patch and use it for ABE, and instead of trying to juggle this, Digium will simply refuse your code into Asterisk, which makes sense. |
| 13:37.53 | ddanier | So they have no real use of the GPL-released code if someone takes it and adds nice features.... |
| 13:38.18 | ddanier | I thought there might be some license-extension they use to make the changes code somehow more useable :) |
| 13:38.20 | tzanger | ddanier: the disclaimer basically states that you give Digium a permanent "license" to not only your code, but also allows them to sell your code as part of Asterisk |
| 13:38.48 | tzanger | ddanier: it's not that they don't have a use for it; they are simply trying to protect their ability to sell Asterisk |
| 13:39.41 | ddanier | Of course, thats why I thought there might be some extension to the GPL, that says you have to desclaim any code you change back to them... |
| 13:40.01 | ddanier | But I found nothing and thought it is somehow unusable for them if someone changes their code.... |
| 13:40.26 | tzanger | isn't that exactly what I said? Any code you want included into mainstream Asterisk must be disclaimed, granting Digium a permanent, royalty-free license to do what they want with the code you want included. |
| 13:40.31 | ddanier | Thanks for the clarification ;-) |
| 13:41.02 | ddanier | Of course, you said that, I was just suprised :) |
| 13:41.12 | tzanger | You're free to make your GPL code available but if it's not disclaimed it'll never get into Asterisk :-) |
| 13:43.44 | *** join/#asterisk tengulre (n=tengurle@222.90.175.75) |
| 13:46.03 | *** join/#asterisk htims (n=pd@Vc9c4.v.pppool.de) |
| 13:53.31 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 14:02.47 | *** join/#asterisk Altair256 (n=icechat5@mail.clccorp.com) |
| 14:04.13 | *** join/#asterisk jmacz (n=jmacz@201.244.197.240) |
| 14:05.56 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:09.56 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 14:17.21 | jsharp | glorp |
| 14:20.37 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 14:24.13 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:24.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:27.20 | Hmmhesays | i said this girl gotta go she's a head full of crazy |
| 14:30.31 | bkw_ | Hmmhesays, yo |
| 14:31.05 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 14:31.39 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
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| 14:39.08 | *** join/#asterisk razu_ (n=razu@193.40.101.34) |
| 14:39.30 | razu_ | have anyone any experience with elevator phonesystems over voip ? |
| 14:40.21 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 14:41.25 | MikeJ[Laptop] | don;t think that is legal inthe US |
| 14:41.42 | Hmmhesays | reminds me of that freaky x files last night |
| 14:41.45 | MikeJ[Laptop] | regulations on elevator phones are strict |
| 14:41.45 | tzanger | elevator phone system? |
| 14:41.52 | tzanger | oh emergency phones |
| 14:41.56 | MikeJ[Laptop] | yeah |
| 14:42.07 | Hmmhesays | he could be in bumfarkegypt |
| 14:42.14 | MikeJ[Laptop] | last building I worked on elevator phones, they were static analog from telco |
| 14:42.22 | MikeJ[Laptop] | couldn't even hit a phone system |
| 14:42.37 | MikeJ[Laptop] | and they autocalled the elevator company when they went off hook |
| 14:43.10 | MikeJ[Laptop] | so if you can' hook them to a phone system.. I seriously doubt you can use voip |
| 14:43.22 | Hmmhesays | the phone in our old building was a regular telco line |
| 14:43.29 | Hmmhesays | could dial anywhere you wanted |
| 14:43.43 | jsharp | There's a hotel in Decatur, TX who's elevator phone is tied to their phone system. If you call the hotel and the front desk never answers, the call gets routed to an elevator. |
| 14:43.43 | Hmmhesays | hell you could call it |
| 14:43.45 | Altair256 | but did it dial through your PBX, or direct connect to the telco? |
| 14:44.23 | Hmmhesays | no pbx, go off hook, dialtone from the telco |
| 14:44.37 | Altair256 | are you in the USA? |
| 14:44.55 | Hmmhesays | Aye |
| 14:45.21 | Hmmhesays | until north dakota officially becomes part of canada anyway |
| 14:45.42 | Altair256 | I'm trying to find what the federal regulations are, but I have to agree with MikeJ... I believe they are required to be direct connect to the telco |
| 14:45.42 | mutilator | revolution! |
| 14:45.49 | *** join/#asterisk websae (n=icechat5@h4608253e.area4.spcsdns.net) |
| 14:46.14 | Hmmhesays | I'd be more inclined to believe it was state based |
| 14:46.41 | Altair256 | very probable |
| 14:46.44 | backblue | http://lists.digium.com/pipermail/asterisk-dev/2006-March/019262.html -> anyone? |
| 14:46.45 | *** join/#asterisk bweschke (n=bweschke@232.sub-70-198-227.myvzw.com) |
| 14:46.58 | websae | anyone here own a nationwide VoIP company with e911? |
| 14:47.26 | exonic | websae, I work for one. There are e911 solutions via SIP |
| 14:48.28 | exonic | websae, we're not interested in providing that service yet. |
| 14:49.03 | websae | you have e911 via SIP |
| 14:49.07 | Hmmhesays | yeah it is state based |
| 14:49.08 | websae | but you can't provide to multiple DIDs? |
| 14:49.15 | willt | isn't that federaly mandated or something? |
| 14:49.31 | Altair256 | exonic, out of curiosity, why were packet8 and vonage, etc, required by the FCC to have e911 service? |
| 14:49.41 | *** join/#asterisk noky (n=Noky@200.69.211.18) |
| 14:49.45 | noky | hi! |
| 14:49.56 | Hmmhesays | now they are |
| 14:49.56 | websae | you have to have e911 |
| 14:49.59 | websae | it's federal |
| 14:49.59 | noky | i have a question... call parking is the same that call hold ? |
| 14:50.05 | Altair256 | no noky |
| 14:50.09 | exonic | websae, http://www.affinityvoiptelecom.com/ is a example of a a IAX/SIP e911 provider |
| 14:50.09 | websae | FCC requires it now |
| 14:50.16 | Altair256 | call parking is putting the call on a "temp extension" |
| 14:50.22 | exonic | websae, they also provide DIDs |
| 14:50.29 | Altair256 | and then anyone can call that extension to "pick up" the call |
| 14:50.37 | noky | ok.. i will investigate this point... |
| 14:50.46 | Hmmhesays | anyone have an mp3 of hed pe "boom" |
| 14:50.59 | Altair256 | so the FCC mandate only requires VoIP providers that provide DIDs to also provide e911 service? |
| 14:51.23 | noky | is call hold supported by asterisk? |
| 14:51.26 | websae | any VoIP company that provides termination |
| 14:51.30 | Altair256 | yes noky |
| 14:51.35 | noky | thanks Altair256 |
| 14:51.49 | Hmmhesays | last time I looked you can't dial 911 from voipjet |
| 14:52.02 | websae | that's correct Hmmhesays |
| 14:52.33 | Hmmhesays | I think the law says something like any company providing traditional style phone service |
| 14:52.48 | Hmmhesays | i dunno last time I read it was a lot of hangovers ago |
| 14:53.30 | razu_ | Altair256 : yes it called the main base fine ... but it coulnd make its data connection over voip ... or smthing like that... and i'm not in the usa :) |
| 14:54.08 | noky | the character ";" is a comment of a asterisk's file configuration? |
| 14:54.17 | websae | <--------------------is E911 service provider, and sells E911 service to providers :) |
| 14:54.48 | willt | must be nice |
| 14:55.20 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:55.59 | Altair256 | noky, in most places either a ; or a # represents a comment |
| 14:56.35 | noky | thanks again |
| 14:57.31 | *** join/#asterisk nagl (n=nagl@137.208.4.180) |
| 14:58.23 | Altair256 | noky, btw most hard phones have a "hold" button, but you will have to manually setup Call Parking on the extensions config file(s) |
| 14:58.54 | Altair256 | AAH2.2-2.6 have it built in, and AAH2.7 as well but 2.7 changed the format |
| 14:58.59 | noky | ok |
| 15:00.20 | willt | websae: whats the link to your site? |
| 15:01.42 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 15:01.58 | cytrak | any ideas on IAX sofphone for linux ? |
| 15:02.27 | tzanger | ok wtf |
| 15:02.27 | tzanger | <PROTECTED> |
| 15:02.31 | tzanger | <PROTECTED> |
| 15:02.54 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 15:03.14 | ahattar | hi everyone |
| 15:03.30 | astra^^ | cytrak:firefly |
| 15:03.46 | *** part/#asterisk babyhuey (n=justin@ip-131-123-81-11.housing.res.kent.edu) |
| 15:03.49 | astra^^ | cytrak:http://www.tucows.com/preview/344929 |
| 15:03.54 | ahattar | question can I connect avaya ip hardphone to asterisk PBX? |
| 15:04.00 | tzanger | I am guessing that SIP/214 transfered the call to SIP/211 but why would I get a 500 back? |
| 15:04.16 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfliq.dialup.mindspring.com) |
| 15:04.23 | cytrak | that's for windows |
| 15:04.40 | cytrak | ast_freak: cubix is better .. same people of firefly |
| 15:04.49 | cytrak | I need for linux |
| 15:05.15 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 15:05.32 | backblue | cytrak: http://kiax.sourceforge.net/en_news.html |
| 15:05.54 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfliq.dialup.mindspring.com) |
| 15:05.56 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:06.05 | kippi | anyone got anyideas why when I ring in to my asterisk box it dosn't show the number you are calling from? |
| 15:06.06 | backblue | cytrak: sorry, http://iaxclient.sourceforge.net/iaxcomm/ |
| 15:06.09 | *** join/#asterisk ibob63 (n=hp@bb-87-82-29-147.ukonline.co.uk) |
| 15:06.48 | tzanger | kippi: ring it how |
| 15:07.13 | kippi | tzanger: ring a DDI that goes to a SIP extension |
| 15:07.39 | tzanger | kippi: so you have a DID with some VOIP provider? Do they provide Caller*ID? |
| 15:07.50 | Altair256 | cytrak, 1 sec |
| 15:07.55 | ahattar | question can I connect avaya ip hardphone to asterisk PBX? |
| 15:08.08 | kippi | tzanger: its a ISDN line, asterisk says Accepting call from '' |
| 15:08.13 | jsharp | What protocol does the avaya use? |
| 15:08.16 | Altair256 | I can't see why not ahattar |
| 15:08.47 | ahattar | it is not a fully SIP compatbile phone |
| 15:08.51 | Altair256 | cytrak, try http://www.asteriskguru.com/tools/idefisk_beta.php |
| 15:08.57 | tzanger | kippi: ok, do you have Caller*ID provisioned on your PRI/BRI? |
| 15:09.08 | Altair256 | what is the model number ahattar |
| 15:09.18 | ahattar | 4621 |
| 15:09.24 | tzanger | kippi: do a "pri debug span 1" and see what is coming from the telco |
| 15:09.34 | threepoint141592 | O_o |
| 15:09.47 | tzanger | (it'll be noisy, but you will be able to see if they're sending you any identification IEs |
| 15:11.15 | kippi | tzanger: I can't see the incoming number |
| 15:11.34 | tzanger | kippi: it's unlikely that the telco is providing that to you then |
| 15:12.44 | kippi | ah ok then, there is no config to turn it on? |
| 15:12.50 | astra^^ | Mar 14 20:37:17 NOTICE[10068]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: |
| 15:12.55 | astra^^ | y do i get tis msg |
| 15:13.10 | ahattar | Altair256: avaya phone http://www.avaya.com/gcm/master-usa/en-us/products/offers/4600_series_ip_telephones.htm&View=ProdOverview |
| 15:14.37 | Juggie | astra^^, seems prettty self explanatory to me |
| 15:14.43 | astra^^ | heheheh |
| 15:19.20 | *** join/#asterisk pengyong (n=lala@222.185.19.51) |
| 15:19.45 | backblue | anyone know a god grafical admin interface for asterisk? |
| 15:19.48 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:19.53 | *** join/#asterisk medusaXX (n=medusaxx@p54A986A8.dip0.t-ipconnect.de) |
| 15:21.03 | Altair256 | blackblue AMP is useful |
| 15:21.12 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:22.23 | *** join/#asterisk bit123 (i=bit123@222.165.171.41) |
| 15:22.23 | Altair256 | ahattar, where are you getting the information that the Avaya 4600 series phones are not fully SIP compliant? |
| 15:22.46 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 15:22.52 | bit123 | using IAX can calls be blind/consultation transfered ? |
| 15:22.54 | kippi | I have a 4620 running SIP and it is all working fine |
| 15:23.21 | LostFrog | Is there a way to kill sip calls from *? |
| 15:23.35 | [TK]D-Fender | backblue : How big a setup are you looking to manage and what do you expect the GUI to do for you? |
| 15:23.50 | Altair256 | yes LostFrong, but I do not know the command. Let me see if I can look it up for you |
| 15:24.07 | astra^^ | how do i see cdr in asterisk..? |
| 15:24.37 | LostFrog | I know zap is zap destroy channel. |
| 15:24.45 | jsharp | soft hangup |
| 15:24.57 | [TK]D-Fender | astra^^ : /var/log/asterisk/cdr-csv |
| 15:25.00 | LostFrog | Thank you. |
| 15:25.23 | astra^^ | i mean grafical.. |
| 15:25.23 | tzanger | kippi: if it isn't coming from the telco, they need to provision the line for it. There is nothing you can do on your end. |
| 15:25.33 | astra^^ | graphical |
| 15:25.33 | Altair256 | graphical even? ^^ |
| 15:25.36 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:25.50 | [TK]D-Fender | astar` : Areski |
| 15:25.57 | [TK]D-Fender | astra^^ rather... |
| 15:25.59 | Altair256 | astra^^ AMPhttp://sourceforge.net/projects/amportal/ |
| 15:26.08 | Hmmhesays | i like the new freepbz |
| 15:26.12 | Hmmhesays | *freepbx even |
| 15:26.16 | kippi | tzanger: just phoning them now |
| 15:26.21 | Altair256 | isn't freepbx just a repack of AMP? |
| 15:26.41 | MikeJ[Laptop] | what is freepbx? |
| 15:26.49 | Altair256 | http://wiki.freepbx.org/tiki-index.php |
| 15:26.52 | *** join/#asterisk [swb] (n=swb@cornelyn.force9.co.uk) |
| 15:26.57 | Hmmhesays | the new incarnation of amp |
| 15:27.12 | [swb] | Does anyone know if you can do a GoSub from an AGI |
| 15:27.13 | Altair256 | what's wrong with AMP the way it is? |
| 15:27.15 | backblue | [TK]D-Fender: i just want to do like asterisk -r but for remote asterisks. |
| 15:27.34 | Altair256 | [swb] depends if the language you are in supports gosub |
| 15:27.45 | [TK]D-Fender | backblue : That has nothing to do with GUI's..... |
| 15:27.50 | [swb] | Altair256, I mean a dial plan gosub |
| 15:27.57 | MikeJ[Laptop] | [swb], macros |
| 15:28.08 | [TK]D-Fender | backblue : You can just log in to * remotely and get CLI.... |
| 15:28.12 | [swb] | MikeJ[Laptop], I am trying to mvoe from a macro |
| 15:28.19 | cytrak | any other iax clients for linux besides kiax ? |
| 15:28.20 | [swb] | to make it easier to read/write to mysql |
| 15:28.25 | kippi | is there away so that when you pick up your phone it rings an number/extension? |
| 15:28.25 | backblue | [TK]D-Fender: login? ssh? |
| 15:28.33 | MikeJ[Laptop] | [swb]. how is that easier? |
| 15:28.45 | [swb] | from an AGI I am in python |
| 15:28.47 | [TK]D-Fender | backblue : yup. |
| 15:29.29 | Altair256 | [swb] if you are in python, use functions |
| 15:29.30 | [TK]D-Fender | kippi : depends on the phone. That is nick-named the "bat-phone" effect. Zap analog channels can do that as well as sever SIP phones. |
| 15:29.48 | Hmmhesays | do you know how cool it would be to make a phone call adam west? |
| 15:30.23 | Altair256 | the other option would be to do the entire thing in AEL, [swb] |
| 15:30.24 | [swb] | Altair256, yeah I suppose I could |
| 15:30.44 | Altair256 | put all the logic into your Python script and out of the dialplan |
| 15:30.52 | Altair256 | I'm a phpagi fan myself |
| 15:30.59 | [swb] | but I already have the dialplan code there that I wanted to call, I just wondered if it was possible to do EXEC Gosub context|exten|priotiry |
| 15:31.03 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:31.05 | [swb] | yeah |
| 15:31.09 | [swb] | that might be an idea |
| 15:31.34 | [TK]D-Fender | Altair256 : Got a sample script & the supporting files you could give me as a learning primer by any chance? ;) |
| 15:31.34 | [swb] | Altair256, where would I get beginners info for AEL |
| 15:31.35 | FuriousGeorge | wtf, one of my * boxes is suddenly not detecting a hangup |
| 15:31.41 | FuriousGeorge | from a sip peer |
| 15:31.41 | Altair256 | if you are making a phone "application", it makes sense (to me) that the entire application should in fact be "in the application" |
| 15:31.45 | [TK]D-Fender | [swb] : AEL = waste of time. |
| 15:31.53 | FuriousGeorge | and all my zapchannels are now "off hook" |
| 15:31.55 | FuriousGeorge | hmm |
| 15:32.11 | MikeJ[Laptop] | ael is the same as dialplan with a different presentation |
| 15:32.20 | Altair256 | [TK]D-Fender - I can point you to the phpagi website... lol |
| 15:32.36 | [TK]D-Fender | Altair256 : Ummm.. sure I guess :) |
| 15:32.36 | Altair256 | [TK]D-Fender - if you are using AAH, then you also need to fix the phpagi.conf file in /etc |
| 15:32.52 | Altair256 | otherwise festival won't work from phpagi |
| 15:33.03 | [TK]D-Fender | Altair256 : ME?!?! AMP?!?! Ick! I teach/consult * for CLI :) |
| 15:33.21 | Altair256 | lol |
| 15:33.23 | [TK]D-Fender | I just never got off my ass as far as external scripting is concerned :) |
| 15:33.28 | Altair256 | http://phpagi.sourceforge.net/ |
| 15:33.37 | Altair256 | they have example scripts as well |
| 15:33.52 | Altair256 | the class is very easy to use |
| 15:34.03 | Altair256 | and php lends itself fairly well to this type of programming |
| 15:35.01 | *** join/#asterisk sag_ich_nicht (i=bitch2k@85-124-38-82.dynamic.xdsl-line.inode.at) |
| 15:35.17 | [TK]D-Fender | Altair256 : Looking for a sample now... if its there its not linked in an obvious way (or I'm just blind today) |
| 15:35.23 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 15:35.38 | [TK]D-Fender | Altair256 : I do PHP already for basic web stuff so I'll be just fine with it I'm sure... |
| 15:36.06 | *** part/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 15:36.11 | [TK]D-Fender | Altair256 : I think I found some... |
| 15:36.31 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131) |
| 15:37.54 | Altair256 | [TK]D-Fender if you click on the documentation... |
| 15:37.56 | Fedoracore6 | hai all |
| 15:38.07 | Hmmhesays | survival of the sickest is a fantastic song |
| 15:38.07 | Altair256 | the choose phpagi.php at the bottom... you'll see some examples |
| 15:38.13 | astra^^ | weather tis amp will go with the present runnin asterisk.. ? |
| 15:38.32 | [TK]D-Fender | Altair256 : Yup, I've got samples :) Now I just need to read up on the class-calls and get coding. Thanks for the link. |
| 15:38.40 | Altair256 | np |
| 15:39.08 | Altair256 | if you want festival (text to speech) to work from phpagi, make sure you set a phpagi.conf file that correctly points to the text2wave exec on your box |
| 15:40.01 | noky | asterisk can traslate sip2h323 ? |
| 15:40.44 | Altair256 | noky, I'm fairly certain asterisk itself can bridge most protocols out there |
| 15:40.58 | noky | thanks |
| 15:41.20 | [TK]D-Fender | Altair256 : I never tried installing festival before actually. Are the WIKI instructions relatively easy to follow or is it a harrowing experience? I'm running Slackware with a full load of dev tools so compiling "shouldn't" be a problem. |
| 15:42.01 | Altair256 | I'm a firm believer in riding on the back of giants... |
| 15:42.14 | Altair256 | AAH in single server setups is the way to go |
| 15:42.50 | Altair256 | if you plan on having multiple servers in your VoIP infrastructure, then roll your own. Otherwise... the package is "too easy" |
| 15:43.48 | [TK]D-Fender | Altair256 : You aren't seriously propsing that as my solution to getting a working Festival install now are you? |
| 15:44.00 | Altair256 | >.> |
| 15:44.27 | Altair256 | I think it's a solution to a "bigger" problem... lol |
| 15:44.33 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 15:44.39 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 15:45.05 | Altair256 | I can't guide you where I haven't been |
| 15:45.17 | [TK]D-Fender | Altair256 : What bigger problem, so far I don't have any, just a single question (we'll discount the PHP-AGI one since thats solved with *1* link).... |
| 15:45.26 | Altair256 | lots of places use festival, so I have to imagine the instructions on the wiki work |
| 15:45.27 | [TK]D-Fender | Altair256 : Ok, fair enough :) |
| 15:45.32 | Ateboy | Hi everyone... Just one question: How do I tell * to add a "*67" in front of all outgoing calls (but not internal calls) so that people don't know which one of my pstn lines I'm using to call? |
| 15:45.59 | willt | why don't you just set the outgoing caller id? |
| 15:46.02 | Altair256 | set your outbound route as *67|. |
| 15:46.16 | Altair256 | err.. wait |
| 15:46.19 | RoyK | Ateboy: exten => _XXXXXXX.,1,Dial(zap/1/*65${EXTEN}) |
| 15:46.20 | Altair256 | I have that backwards |
| 15:46.29 | ManxPower | Dial(Zap/g1/*67${EXTEN}) of course. |
| 15:46.30 | Altair256 | RoyK has it |
| 15:46.42 | ManxPower | now, if you are using amp or something like that then this is the wrong place to ask/ |
| 15:47.03 | Altair256 | ManxPower - We aren't all CLI elitest ;) |
| 15:47.24 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 15:47.24 | ManxPower | ~amp |
| 15:47.26 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 15:47.41 | [TK]D-Fender | ManxPower : Doesn't there have to be a pause for the2nd dial-tone? |
| 15:48.11 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.31) |
| 15:48.17 | ManxPower | [TK]D-Fender, that would depend. The guy was too lazy to say what tech he's using. |
| 15:48.25 | iCEBrkr | werd. |
| 15:48.32 | Ateboy | Thanks Royk, but I'm using an ATA to go out |
| 15:48.44 | FuriousGeorge | very weird: when calling out, zap channels are unavail (off hook), but they work fine when calling in? |
| 15:48.56 | ManxPower | Ateboy, using non-analog means none of our advice will work |
| 15:49.00 | [TK]D-Fender | ManxPower : You are taking on BKW's bad habits.. you're only missing his patented "NEXT!!!!"... |
| 15:49.17 | Altair256 | FuriousGeorge - could be an outbound route issue |
| 15:49.19 | ManxPower | [TK]D-Fender, I'm still on my first cup of coffee. |
| 15:49.32 | FuriousGeorge | Altair256: as in my telco is screwing up? |
| 15:49.34 | [TK]D-Fender | ManxPower : Using non-analog he could just use Set(CALLERID()) for that matter! |
| 15:49.50 | [TK]D-Fender | Hmmhesays : "EVERYTHING ABOUT YOU! |
| 15:49.56 | Altair256 | no, as in your outbound route for your calls |
| 15:50.01 | [TK]D-Fender | Hmmhesays : An oldie I got to play. |
| 15:50.08 | Altair256 | you said inbound works, so your trunks must be setup right |
| 15:50.20 | *** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84) |
| 15:50.28 | FuriousGeorge | Altair256: im not sure what you eman by that. its zap. the only ourbound route is dial(zap/g1) |
| 15:50.44 | [TK]D-Fender | God I can't wait till Ekiga comes out with a Win32 build.... |
| 15:50.47 | Hmmhesays | [TK]D-Fender yeah I gotta fake the solo to it, I really don't care to learn it note for note |
| 15:50.50 | ManxPower | [TK]D-Fender, then it would be something like Dial(SIP/*67@sippeer,,D(${EXTEN}) |
| 15:50.59 | ManxPower | or even |
| 15:51.02 | FuriousGeorge | but zap show channel 3 (or 4) shows them as "off hook" |
| 15:51.03 | Altair256 | 1 sec, let me look up what I'm trying to say |
| 15:51.05 | *** join/#asterisk nagl (n=nagl@137.208.4.186) |
| 15:51.06 | ManxPower | Dial(SIP/*67@sippeer,D(${EXTEN}) |
| 15:51.09 | Altair256 | oh.. |
| 15:51.10 | Altair256 | hrm |
| 15:51.15 | FuriousGeorge | so asterisk doesnt even TRY to dial out there |
| 15:51.16 | ManxPower | I dunno, I see no reason to hide my callerid |
| 15:51.29 | Altair256 | do you have a PRI, or an fxs bank? |
| 15:51.39 | FuriousGeorge | fxs |
| 15:51.42 | FuriousGeorge | tdm400 |
| 15:51.49 | FuriousGeorge | fxo is actually what aint working |
| 15:51.54 | Altair256 | I bet you have it as fxs_ls? |
| 15:51.59 | ManxPower | FuriousGeorge, What is your Dial line? |
| 15:52.01 | FuriousGeorge | no sir |
| 15:52.03 | Altair256 | change it to ks |
| 15:52.07 | FuriousGeorge | already is |
| 15:52.12 | Altair256 | dang v.v |
| 15:52.28 | willt | www.voip-info.org sucks! |
| 15:52.37 | iCEBrkr | ha |
| 15:52.37 | kippi | willt: why? |
| 15:52.40 | ManxPower | willt, Tell us something we don't already know. |
| 15:52.41 | Altair256 | what's wrong Willt? |
| 15:52.46 | willt | lol |
| 15:52.52 | *** part/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 15:52.52 | *** join/#asterisk bmrocha (n=bruno@82.102.1.42) |
| 15:52.52 | willt | there server is always timming out |
| 15:52.55 | FuriousGeorge | exten => s,1,chanisavail(${ARG2}) |
| 15:52.56 | FuriousGeorge | exten => s,2,dial(${ARG2}/${ARG1},,T) |
| 15:52.59 | Altair256 | yeah, that part does suck willt |
| 15:53.09 | Hmmhesays | use google cache and stfu |
| 15:53.15 | [TK]D-Fender | Hmmhesays : You know I don't remember that song even HAVING a solo.... playing the MP3 now :) |
| 15:53.22 | ManxPower | [TK]D-Fender, use groups |
| 15:53.36 | Ateboy | the thing is that my entry for the outgoing dial is "exten => s,1,Dial(SIP/2998/${ARG1})" |
| 15:53.41 | [TK]D-Fender | ManxPower : .. huh? |
| 15:53.46 | Hmmhesays | right before it goes into the shuffle at the end |
| 15:53.59 | FuriousGeorge | ManxPower: thats my dial above, but its not gettiong past the chanisavail, as the status of zap 3 and 4 is offhook, despite ks siognalling |
| 15:54.03 | FuriousGeorge | just started happening |
| 15:54.05 | Hmmhesays | "look again cause I ain't wearing no frown" |
| 15:54.14 | ManxPower | [TK]D-Fender, group=1 before your channel defs in zapata.conf then Dial(Zap/g1/${EXTEN}) or whatever it is |
| 15:54.15 | Ateboy | shoud I put the *67 in front of "${ARG1} ? |
| 15:54.28 | ManxPower | Ateboy, only if you are using analog fxo card. |
| 15:54.46 | Ateboy | I'm using a SPA-3000 ATA |
| 15:54.55 | ManxPower | then you are not using an analog fxo card. |
| 15:54.59 | ManxPower | Dial(SIP/*67@sippeer,D(${EXTEN}) |
| 15:55.12 | [TK]D-Fender | Hmmhesays : Sounds like an easy one, just spend an hour on it.... |
| 15:55.12 | ManxPower | I think that should be ,,D |
| 15:55.20 | FuriousGeorge | ManxPower: you taking to me or [TK]D-Fender? |
| 15:55.27 | FuriousGeorge | s/taking/talking |
| 15:55.36 | Hmmhesays | [TK]D-Fender yeah its not bad, theres just a few anchor notes you have to hit |
| 15:55.36 | ManxPower | FuriousGeorge, you for the group stuff. |
| 15:55.42 | [TK]D-Fender | ManxPower : Doesn't sound like I am the person who you should be answering... I only doubted having to have a pause betweent he * code... |
| 15:56.05 | FuriousGeorge | ManxPower: they are already grouped. this worked yesterday |
| 15:56.25 | ManxPower | I still can't understand people's fetish for using ChanIsAvail |
| 15:56.30 | ManxPower | at least on zap ports |
| 15:56.57 | wunderkin | some people like it rough, manx |
| 15:56.59 | FuriousGeorge | you rather i use the priority+101? |
| 15:57.10 | FuriousGeorge | either way the dial is gonna fail as the channel status is offhook? |
| 15:57.17 | ManxPower | FuriousGeorge, no, I would rather use groups and do away with all that extra crap |
| 15:57.24 | FuriousGeorge | ManxPower: i do have groups |
| 15:57.29 | FuriousGeorge | i have 2 zap channels |
| 15:57.32 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 15:57.41 | ManxPower | I use neither priority+101 nor Chanisavail |
| 15:57.49 | FuriousGeorge | so the group gets filled up pretty quick, and i gotta use chan_iax2.so to make a local call |
| 15:57.56 | Ateboy | ManxPower: what is the D for? |
| 15:58.09 | ManxPower | Ateboy, "show application dial" |
| 15:58.22 | FuriousGeorge | ManxPower: what happens when someone tryies to dial and all your zap channels are in use (since groups dont work across techs last i heard)\ |
| 15:58.27 | ManxPower | FuriousGeorge, why not just check the value of DIALSTATUS to decide what to do? |
| 15:58.36 | FuriousGeorge | ManxPower: fair enough |
| 15:58.40 | ManxPower | Granted it is a little more complicated, but much more flexible |
| 15:58.58 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 15:59.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:59.17 | FuriousGeorge | ManxPower: yeah ill give that a whirl. lemme see if asterisk-1.2.5 fixes the offhook thing for now |
| 15:59.38 | willt | Does anyone know why A@H/AMP ringgroups are so complicated? |
| 15:59.39 | ManxPower | FuriousGeorge, If you try doing something like Dial(Zap/g1/${EXTEN}) and all ports are in use then Dial will exit and set a value of CHANUNAVAIL for DIALSTATUS and you can decide what to do from there. |
| 15:59.46 | Hmmhesays | they aren't |
| 15:59.46 | ManxPower | ~amp |
| 15:59.47 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
| 15:59.53 | asteriskmonkey | anyone every experience echo cancellation issues with asterisk and digium hardware on an AMD CPU ? |
| 15:59.58 | willt | im not using amp lol |
| 16:00.07 | FuriousGeorge | ManxPower: yeah i know what you mean, im gonna try that |
| 16:00.07 | willt | I was comparing ring groups |
| 16:00.14 | ManxPower | asteriskmonkey, Every one has experienced echo. |
| 16:00.22 | Ateboy | ManxPower: reading |
| 16:00.25 | [TK]D-Fender | asteriskmonkey : Not BECAUSE of AMD hardware.... |
| 16:00.55 | asteriskmonkey | ok explain this then .. te110p and te406p digium cards echo badly.. stick in bottom of the line sangoma and things run next to perfect. |
| 16:01.27 | Altair256 | IRQ signalling issue. do a cat /proc/interrupts |
| 16:01.34 | ManxPower | asteriskmonkey, I can't explain it without extensive research using the exact motherboards you are using. |
| 16:01.43 | Altair256 | if any of your cards are on a shared IRQ, then you'll have issues |
| 16:01.44 | asteriskmonkey | did :P even ran zttest was at 99.98 so irq was good |
| 16:02.08 | ManxPower | Perhaps the digium cards are not synching correctly, dropping data, or are sharing interrupts. All of these things would really screw up echocan. |
| 16:02.18 | [TK]D-Fender | asteriskmonkey : I had *2* TE405P's fron your company and have horrible echo on my Intel Xeon Supermicro server..... I'm not blaming Intel or Supermicro..... |
| 16:02.27 | asteriskmonkey | :) |
| 16:02.29 | asteriskmonkey | lol |
| 16:02.42 | ManxPower | [TK]D-Fender, we could not even get asterisk to work on a Xeon Supermicro server. |
| 16:02.47 | [TK]D-Fender | My A104d is churning away without flaws. |
| 16:02.47 | asteriskmonkey | well when the support staff says change the motherboard i kinda think a bit |
| 16:02.50 | *** join/#asterisk MarioGamboa (n=yyyy@201.123.66.152) |
| 16:02.59 | ManxPower | HDLC Abort errors all the time, changed out the motherboard and all worked well. |
| 16:03.28 | willt | ManxPower: Supermicro is a problem all on it's own! |
| 16:03.31 | wunderkin | ManxPower, "ALL" the time? or frequently? |
| 16:03.36 | [TK]D-Fender | asteriskmonkey : They had me disable NICs in bios ruibuild kernels, and all sort of other stuff to try to get them to cooperate to no avail. |
| 16:03.41 | ManxPower | wundaboy, anytime IDE access happened. |
| 16:03.45 | *** part/#asterisk vimman (n=codeshep@125.16.130.66) |
| 16:03.46 | ManxPower | So, yes, all the time |
| 16:04.02 | asteriskmonkey | man that is whacked they need to make there cards like sangoma does |
| 16:04.12 | wunderkin | well, i was referring to setting the card to 33mhz, i had the same problem |
| 16:04.19 | wunderkin | the new cards are forced to that now |
| 16:04.47 | [TK]D-Fender | asteriskmonkey : You need to what what you say given where you are now :) |
| 16:04.53 | [TK]D-Fender | and I don't mean jsut in channel :) |
| 16:05.27 | wunderkin | that was on a dual xeon supermicro mb, the bus was set to auto of course and it wouldnt work until i forced it to 33mhz, kpfleming caught that one for me |
| 16:05.33 | brettnem | what company does asterboy work for? |
| 16:05.50 | brettnem | er asteriskmonkey rather.. damn tab completion |
| 16:06.05 | asteriskmonkey | force the bus to 33mhz? what was the symptoms when your buss was wrong |
| 16:06.36 | wunderkin | asteriskmonkey, i would get hdcl aborts and all kinds of other errors scrolling the screen |
| 16:06.40 | wunderkin | errr |
| 16:06.47 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 16:06.56 | wunderkin | damn brettnem, your typos are rubbing off on me :P |
| 16:07.01 | RoyK | methinks someone should test out some nukes soon - and why not in israel? http://news.bbc.co.uk/2/hi/middle_east/4804424.stm |
| 16:07.04 | asteriskmonkey | ah k. i know i have no conflicts and all the tests pan out so thats why its bothering me badly |
| 16:07.22 | asteriskmonkey | i like digium im just frustrated becuase no one seems to have an answer |
| 16:08.47 | *** part/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 16:08.50 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 16:10.04 | Fedoracore6 | hal all i try run my agi script but have some error like this... |
| 16:10.05 | Fedoracore6 | <PROTECTED> |
| 16:10.06 | Fedoracore6 | Failed to execute '/var/lib/asterisk/agi-bin/password.agi': No such file or directory |
| 16:10.28 | masonf | is there a way to make a zaptel card only pick up after a sip phone answers? I want my regular phones to ring while I integrate asterisk into my system |
| 16:10.34 | Fedoracore6 | what the error mean |
| 16:10.34 | iCEBrkr | Fedoracore6: password.agi needs to be chmod +x |
| 16:10.39 | Altair256 | do a chown asterisk:asterisk password.agi |
| 16:10.46 | iCEBrkr | Fedoracore6: and should probably be owned by asterisk |
| 16:11.01 | Altair256 | then make sure it's +x |
| 16:11.05 | willt | Fedoracore6: make sure it can find perl or whatever is running the script |
| 16:11.13 | Altair256 | yeah, iCEBrkr and I are on the same page... lol |
| 16:11.16 | Fedoracore6 | ls -l /var/lib/asterisk/agi-bin/password.agi |
| 16:11.17 | Fedoracore6 | <PROTECTED> |
| 16:11.17 | iCEBrkr | :) |
| 16:11.22 | Fedoracore6 | like this right |
| 16:11.56 | Altair256 | the chmod should be 744 when it's done |
| 16:12.11 | Fedoracore6 | hemm |
| 16:12.22 | willt | Altair256: why do others need read access why not 700 ? |
| 16:12.40 | Fedoracore6 | so what i must do i type chmod +x /var/lib/asterisk/agi-bin/password.agi |
| 16:12.45 | Fedoracore6 | but still can runn |
| 16:12.57 | willt | Fedoracore6: is this using perl? |
| 16:13.02 | Altair256 | because I don't SSH into the box with the asterisk account? |
| 16:13.06 | [TK]D-Fender | masonf : Easily done. |
| 16:13.09 | Ateboy | ManxPower: Using exten => s,1,Dial(SIP/*67@2998,D/${ARG1}) does not work (can't even call) |
| 16:13.14 | Fedoracore6 | hem AGI |
| 16:13.26 | iCEBrkr | Fedoracore6: Maybe you should learn how chmod and chown work first?? |
| 16:13.36 | willt | Fedoracore6: what is the line at the top of the file? |
| 16:13.38 | Altair256 | how are you calling it from your exten list? |
| 16:13.45 | willt | iCEBrkr: :) |
| 16:13.48 | Ateboy | ManxPower: Using exten => s,1,Dial(SIP/*67@2998,,D/${ARG1}) sends the call through, but doesn't seem to send the *67 as I can see the CID on the recipient's side |
| 16:14.07 | Fedoracore6 | #!/usr/local/bin/php -q |
| 16:14.19 | willt | Fedoracore6: can you ls -la /usr/local/bin/php ? |
| 16:14.23 | Altair256 | change it to #!/usr/bin/php -q |
| 16:14.29 | Ateboy | ManxPower: wait, my mistake |
| 16:14.37 | [TK]D-Fender | exten => s,1,Dial(SIP/2998/*67,D/${ARG1}) |
| 16:14.37 | *** join/#asterisk fulgas (n=fulgas@209.8.233.207) |
| 16:14.46 | [TK]D-Fender | Ateboy : see above |
| 16:15.00 | [TK]D-Fender | Ateboy : think that'll do it :) |
| 16:15.38 | masonf | [TK]D-Fender: do you have a tutorial on how to do this? |
| 16:15.39 | Ateboy | I'll try it |
| 16:15.47 | Fedoracore6 | ls: /usr/local/bin/php: No such file or directory |
| 16:15.47 | Fedoracore6 | ls: ?: No such file or directory |
| 16:15.58 | Altair256 | try ls /usr/bin/php |
| 16:15.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 16:16.05 | Altair256 | that's the next most likely location |
| 16:16.07 | willt | Fedoracore6: where is php installed? find / -name php |
| 16:16.17 | iCEBrkr | 'which php' |
| 16:16.26 | RoyK | locate php |
| 16:16.28 | RoyK | :P |
| 16:16.30 | iCEBrkr | RoyK: ew |
| 16:16.34 | willt | which only works if it's in path |
| 16:16.43 | willt | find / |grep php |
| 16:16.45 | iCEBrkr | willt: well if it's CLI, it should be in the path |
| 16:16.52 | Fedoracore6 | [root@asterisk ~]# ls /usr/bin/php |
| 16:16.52 | Fedoracore6 | /usr/bin/php |
| 16:16.52 | Fedoracore6 | [root@asterisk ~]# |
| 16:16.54 | RoyK | willt: find / -iname '*php*' |
| 16:16.59 | willt | depends who installedit :) |
| 16:17.02 | iCEBrkr | lol |
| 16:17.03 | Altair256 | see, /usr/bin/php |
| 16:17.06 | Fedoracore6 | ok if i type like this the gree word come |
| 16:17.09 | RoyK | rm /usr/bin/php |
| 16:17.15 | iCEBrkr | lol |
| 16:17.15 | [TK]D-Fender | masonf : no magic tutorial for it, just don't put an "answer" in front of your "dial"'s |
| 16:17.19 | Altair256 | Fedoracore6, change the top line... remove the local part |
| 16:17.27 | willt | grep php /dev/hda |
| 16:17.30 | Fedoracore6 | <PROTECTED> |
| 16:17.33 | Fedoracore6 | like this |
| 16:17.41 | Altair256 | don't listen to RoyK |
| 16:17.46 | Altair256 | RoyK |
| 16:17.55 | Altair256 | RoyK - that's just terrible >.< |
| 16:18.04 | RoyK | :) |
| 16:18.07 | RoyK | me |
| 16:18.07 | RoyK | ? |
| 16:18.10 | Fedoracore6 | hehehe |
| 16:18.14 | Fedoracore6 | thank you all |
| 16:18.21 | Fedoracore6 | i try first |
| 16:18.25 | Altair256 | that fix it? |
| 16:18.42 | willt | Fedoracore6: you should read up on basic linux administration |
| 16:19.00 | Fedoracore6 | yes willt |
| 16:19.17 | ibob63 | I am so frustrated by me asterisk server / nat issues. I think I really need an expert to come an debug my installation. Can anyone point me the direct of someone (not a company) in the UK who can show me the ropes? |
| 16:19.42 | Fedoracore6 | <PROTECTED> |
| 16:21.00 | ^HeLL^ | ibob63 : why not a company? |
| 16:21.50 | *** join/#asterisk razu (n=razu@62.65.35.162) |
| 16:22.04 | ibob63 | because I prefer to work with people rather than salesmen :) |
| 16:22.35 | ^HeLL^ | behind of salesmen there are "people" :) |
| 16:22.46 | Hmmhesays | ibbo63 whats wrong |
| 16:22.48 | [TK]D-Fender | ibobo : Just pastebin your sip.conf so we can take a look... |
| 16:22.51 | [TK]D-Fender | ~pb |
| 16:22.53 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:22.58 | Hmmhesays | or pay me, because i'm broke |
| 16:24.18 | ibob63 | the thing is I don't know whether is my computer / router / or asterisk confs are the problem. |
| 16:24.33 | wasim | all of the above! |
| 16:24.57 | jsharp | I recommend a large sledgehammer |
| 16:25.07 | Altair256 | ibob63, sounds like you should hire a professional network/telephony consultant to come by for a few hours |
| 16:25.10 | [TK]D-Fender | ibob63 : Just pastebin your config and we'll tell you if its * |
| 16:25.54 | ibob63 | I'll pastebin all my configs. If only I could pastbin my whole computer. |
| 16:26.16 | bronze | ibob63: you can |
| 16:26.30 | [TK]D-Fender | ibob63 : sip.conf will suffice |
| 16:26.31 | eric_hill | You don't want to pastebin your computer, you want to wastebin your computer. There's a difference. |
| 16:26.52 | *** join/#asterisk donsapo (n=donsapo@host178.201-252-145.telecom.net.ar) |
| 16:27.23 | bronze | ibob63: you pastebin your computer by giving someone ssh access to it. Arrange a payment to them first, oh and check references.. :-) |
| 16:27.37 | donsapo | Hello, I have a problem and no idea how to solve it, maybe one of you can give me a clue, |
| 16:27.49 | [TK]D-Fender | donsapo : fire away... |
| 16:27.57 | donsapo | :) |
| 16:27.58 | ^HeLL^ | donsapo : tell us |
| 16:28.08 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 16:28.21 | mocker | Has anyone seen the error "Don't know what to do if second ROSE component is of type 0x6" |
| 16:28.34 | mocker | It's happening after RxFax tries (and fails) to write it's tif file. |
| 16:28.40 | donsapo | when I execute dial command with a dialstring like SIP/phobos|30|Ttr||, |
| 16:29.01 | donsapo | after the call finishes the variable ANSWEREDTIME is set correctly, |
| 16:29.16 | ibob63 | okay. here is my sip.conf: http://pastebin.com/601494 |
| 16:29.26 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 16:29.34 | CrashHD | hey fella |
| 16:29.35 | CrashHD | s |
| 16:29.40 | donsapo | however, if the dialstring is like SIP/1002@127.0.0.1|30|Ttr, |
| 16:29.41 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:29.44 | CrashHD | Received mini frame before first full voice frame |
| 16:29.45 | donsapo | the same variable is not set |
| 16:29.51 | CrashHD | any ideas on how to fix that? |
| 16:30.02 | donsapo | and, phobos is extension 1002 at my server |
| 16:30.05 | ManxPower | CrashHD, you don't, that's normal |
| 16:30.16 | CrashHD | oh |
| 16:30.27 | ManxPower | donsapo, you do not want to dial to an IP address and you don't want to dial to yourseld. |
| 16:30.36 | ibob63 | bascially, when someone phones my number: 1-302-294-7127 the phone rings, asterisk answers it and then fails to make the bridge. |
| 16:30.45 | CrashHD | they should stop calling it a warning then lol |
| 16:30.52 | Fedoracore6 | hemm i try run my password.agi but have problem like this its i must declere my vairiable |
| 16:30.56 | Fedoracore6 | http://pastebin.com/601495 |
| 16:31.27 | [TK]D-Fender | ibob63 : Immediate problem : All of your localnet clauses are commented out and are essential. |
| 16:31.34 | donsapo | the problem is that I am testing a2billing, which asks for the number to be dialed in an IVR menu, |
| 16:31.35 | Altair256 | Fedoracore6, you need to put phpagi.php in the same directory as this file |
| 16:31.38 | donsapo | and then dials it, |
| 16:31.42 | [TK]D-Fender | ibob63 : Fix those, restart * and try it out. |
| 16:31.50 | ManxPower | donsapo, Dial(Local/extension@context) |
| 16:31.54 | Fedoracore6 | why i can open web site voip.org |
| 16:32.04 | donsapo | suppose I'd like to have an music-on-hold extension and charge for it |
| 16:32.07 | donsapo | oohhh... |
| 16:32.12 | donsapo | ok, I'll give it a try, |
| 16:32.17 | donsapo | thanks, ManxPower |
| 16:32.54 | ManxPower | donsapo, but it still seems more complicated then you need to be. |
| 16:32.56 | Altair256 | Fedoracore6, not sure |
| 16:33.05 | Fedoracore6 | yes i not sure |
| 16:33.08 | *** join/#asterisk Angeljarod (n=jerome@comtepouest69.net8.nerim.net) |
| 16:33.09 | ManxPower | Why not just make the IVR run on a matching dialed string |
| 16:33.13 | Altair256 | Fedoracore6, looks like it's missing an include file that would have all those constants declared |
| 16:33.13 | ibob63 | D-Fender: what should the localnet be? |
| 16:33.17 | Angeljarod | hi there |
| 16:33.47 | [TK]D-Fender | ibob63 : Those should descibe all subnets that * doesn't need to forge your public IP for (those behind your NAT normally) |
| 16:33.48 | ManxPower | ibob63, you didn't read any docs, did you |
| 16:33.59 | CrashHD | how can I reload codecs.conf without dumping the system? |
| 16:34.07 | ManxPower | CrashHD, "reload" |
| 16:34.18 | CrashHD | won't that dump the current calls? |
| 16:34.23 | CrashHD | or is that just restart? |
| 16:34.25 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 16:34.27 | [TK]D-Fender | ManxPower : Go finish your coffee, bile-person! |
| 16:34.41 | ManxPower | CrashHD, no, reload does not dump calls. |
| 16:35.03 | CrashHD | sweet |
| 16:35.05 | CrashHD | thank you |
| 16:35.30 | x86 | morning |
| 16:35.36 | Fedoracore6 | PHP Notice: Use of undefined constant extension - assumed 'extension' in /var/lib/asterisk/agi-bin/password.agi on line 47 |
| 16:35.48 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmul.dialup.mindspring.com) |
| 16:35.50 | Fedoracore6 | where i can find constanst |
| 16:36.01 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfmul.dialup.mindspring.com) |
| 16:36.47 | donsapo | ManxPower, I tried like you said, still no ANSWEREDTIME |
| 16:37.31 | *** join/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 16:37.34 | *** part/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:37.39 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:40.24 | donsapo | ManxPower, I tried like you said, still no ANSWEREDTIME |
| 16:40.51 | donsapo | -- AGI Script Executing Application: (Dial) Options: (Local/1002@sip|30|Ttr||) |
| 16:40.57 | _Paulo_ | Fedoracore6, try $extension |
| 16:41.02 | donsapo | -- Called 1002@sip |
| 16:41.02 | donsapo | <PROTECTED> |
| 16:41.02 | donsapo | <PROTECTED> |
| 16:41.02 | donsapo | <PROTECTED> |
| 16:41.20 | donsapo | <PROTECTED> |
| 16:41.21 | donsapo | <PROTECTED> |
| 16:41.21 | donsapo | <PROTECTED> |
| 16:41.21 | donsapo | <PROTECTED> |
| 16:41.27 | Altair256 | -.- |
| 16:41.32 | Altair256 | pastebin |
| 16:41.36 | ManxPower | doesn't whatever billing package you are using have example configs? |
| 16:41.39 | [TK]D-Fender | donsapo : Use pastebin next time please.... |
| 16:41.40 | [TK]D-Fender | ~pb |
| 16:41.45 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:41.53 | donsapo | I am sorry, don't know what is pastebin |
| 16:42.02 | donsapo | ok |
| 16:42.34 | SplasPood | So if you all had to suggest/manage a firewall+qos device for a customer, and you were looking for something in the $300-500 range, what would you all suggest? |
| 16:42.52 | Altair256 | do you need multiple networks? |
| 16:42.56 | Altair256 | or just a single LAN and a WAN? |
| 16:42.59 | ManxPower | SplasPood, I would suggest looking at a different customer 8-) |
| 16:43.20 | SplasPood | Altair256: Just a single |
| 16:43.29 | Altair256 | consider a Linksys for half the price |
| 16:43.32 | SplasPood | the ones with more complicated setups can spend $$$, we're... |
| 16:43.36 | SplasPood | Linksys? |
| 16:43.38 | Altair256 | or go with a Netscreen/Watchguard SOHO unit |
| 16:43.39 | SplasPood | some $40 POS |
| 16:43.51 | Altair256 | no, the $150 ones |
| 16:43.56 | Altair256 | 1 sec, I'll get you a link |
| 16:43.58 | SplasPood | looks like there are no options inbetween ghetto linksys and $500 or so |
| 16:44.01 | Altair256 | they have a business line of products now |
| 16:44.20 | Altair256 | don't get the ghetto Linksys, look at their business line |
| 16:44.40 | Altair256 | http://www.linksys.com/servlet/Satellite?c=L_Product_C1&childpagename=US%2FLayout&cid=1117775454480&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 16:44.52 | SplasPood | there;s http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416833192&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 16:45.03 | Altair256 | http://froogle.google.com/froogle?q=RV082+&btnG=Search+Froogle |
| 16:45.13 | SplasPood | but does it do QoS? |
| 16:45.24 | Altair256 | lemme check |
| 16:45.34 | Altair256 | I'd also look at VLAN tagging |
| 16:45.45 | donsapo | ok, what I get is in http://pastebin.ca/45690 |
| 16:46.13 | SplasPood | Altair: basically I just wanna be able to insure that SIP traffic going out the WAN is given priority over any other data |
| 16:46.16 | Altair256 | even the el-cheapo linksys $40 WRTG54's do QoS, SplasPood |
| 16:46.23 | SplasPood | Altair; since when? |
| 16:46.35 | SplasPood | Altair: with hacked up firmware, sure.. |
| 16:46.38 | Altair256 | bah.. I have packet8 at my house and one of those rot-gut bottom of the barrel linksys WRTG v5 |
| 16:46.44 | Altair256 | no.. stock firmware |
| 16:46.59 | SplasPood | umm... WRT54G never had QoS |
| 16:47.02 | SplasPood | maybe NOW it does |
| 16:47.07 | Altair256 | no kidding, I bought it, set up MAC QoS, set my DTA as the highest priority... problem gone |
| 16:47.11 | SplasPood | but no model I ever purchased came that way |
| 16:47.11 | donsapo | any clue? |
| 16:47.19 | Altair256 | I even hammer my connection with bittorrent traffic |
| 16:48.08 | sevard | Anyone know much about cornfedsip? I have a 133mhz laptop that i'm trying to get it to run on :/, I can connect to the asterisk server but everything is really choppy. Audio sent and audio recieved is choppy. I can record wav files using sox ag 8000hz and play them and it sounds great. I thought this would be a limitation of the CPU but barely 10% of the CPU is used |
| 16:48.23 | bronze | Altair256: Which model WRTG? |
| 16:48.57 | salviadud | guys, if i want to use mixmonitor to record calls on every changing filenames, say {DATE}{TIME}.gsm |
| 16:49.04 | salviadud | what are variable names i can use? |
| 16:49.36 | Altair256 | the dumbed down v5 does |
| 16:50.14 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:50.20 | medusaXX | how can i forward a call from incoming capi to sip without answering the call before the actual talk starts |
| 16:50.34 | SplasPood | Altair256: Do they even sell WRT54GS anymore? |
| 16:50.48 | SplasPood | hrm, guess they do |
| 16:50.54 | medusaXX | at the moment, i am answering the call and then dialing the sip number |
| 16:51.00 | SplasPood | so you're saying R5 WRT54GS ships /w QoS? |
| 16:51.09 | Altair256 | 1 sec |
| 16:51.13 | Altair256 | at work... had a question |
| 16:51.20 | SplasPood | k |
| 16:51.23 | medusaXX | so the caller pays for the call although there is only a ringtone |
| 16:52.58 | Hmmhesays | time to try out lcdial |
| 16:52.59 | Hmmhesays | wooo |
| 16:53.00 | Altair256 | SplasPood I have V5 of this : http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1127782957298&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 16:53.07 | Altair256 | I have setup 7 of these in the last 4 weeks |
| 16:53.16 | Altair256 | 1 at my house |
| 16:54.00 | Altair256 | pulling up the userguide now to show you where the setting is |
| 16:54.16 | Altair256 | actually, I'll just VNC into my house and pull up the config window |
| 16:54.27 | SplasPood | Altair256: hrm, doesn't say QoS anywhere, but I'll take your word for it |
| 16:54.57 | Altair256 | click on the userguide page... |
| 16:55.04 | Altair256 | wait for the PDF to load and do a search for QoS |
| 16:55.07 | mutilator | i found it jus fine |
| 16:55.24 | *** part/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 16:55.58 | Altair256 | SplasPood - page 52 of the UserGuide PDF is where I set mine up. I can't remember if I did MAC QoS, or port QoS |
| 16:56.01 | bronze | http://www.linksys.com/servlet/Satellite?blobcol=urldata&blobheadername1=Content-Type&blobheadername2=Content-Disposition&blobheadervalue1=text%2Fplain&blobheadervalue2=inline%3B+filename%3Dwrt54g_ver%252C9.txt&blobkey=id&blobtable=MungoBlobs&blobwhere=1130793011719&ssbinary=true |
| 16:56.10 | bronze | search for qos in the page |
| 16:56.18 | bronze | its definitely in there. |
| 16:56.30 | Altair256 | 4.00.7 seems they "updated" the QoS features |
| 16:57.08 | Altair256 | I'm telling you, my calls went from scratchy with bad latency... (because of bittorrent downloads) to crystal clear |
| 16:58.02 | Altair256 | I picked up all the ones I've setup at Wal-Mart -.- I didn't have time to order them |
| 16:58.28 | ibob63 | D-Fender: I have modified the localnet to include my localnetwork 192.168.3.0 and also external ip address. you can see it here http://pastebin.com/601506 Still the phone doesn't connect properly. |
| 16:58.33 | Altair256 | alright, I'm going to lunch. bbl |
| 16:59.48 | salviadud | what filetypes does mixmonitor use? |
| 17:00.12 | asteriskmonkey | if you get openwrt on the router you can do just about anything :D |
| 17:01.42 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 17:02.08 | *** join/#asterisk nusse (i=nusse@mega2000.de) |
| 17:02.26 | zoa | hey hoooo |
| 17:02.30 | nusse | good evening |
| 17:03.03 | ^HeLL^ | evening? where? |
| 17:03.27 | *** join/#asterisk bmg505 (n=leon@dsl-146-14-212.telkomadsl.co.za) |
| 17:03.36 | FuriousGeorge | is there anyway to force a recheck when a zap channel is shown as "off hook" |
| 17:03.52 | nusse | when i dial blaa@host.tld on my phone, asterisk does not try to reach blaa at hst.tld but blaa@localhost, how do i fix that? |
| 17:04.02 | ManxPower | FuriousGeorge, why do you think the port if off hook? |
| 17:04.08 | ManxPower | "zap show channel X" |
| 17:04.18 | masonf | how does asterisk detect a hangup? |
| 17:04.22 | FuriousGeorge | yeah |
| 17:04.28 | ManxPower | masonf, battery drop |
| 17:05.02 | Fedoracore6 | hai all i try change to extensions variable but still have error like this |
| 17:05.03 | Fedoracore6 | http://pastebin.com/601592 |
| 17:05.04 | *** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 17:05.18 | Fedoracore6 | hem.. any suggestion |
| 17:05.59 | masonf | When I hang up the call remains on the line for a bit. |
| 17:07.09 | ManxPower | Fedoracore6, we have no idea what password.php is, how it works, or what it's supposed to do, or how you are supposed to use it. |
| 17:07.33 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 17:07.38 | ManxPower | perhaps you can get support from whoever wrote password.agi |
| 17:08.04 | Fedoracore6 | oic |
| 17:08.26 | Fedoracore6 | hem... i wanna do user add the password same into my database |
| 17:09.40 | Fedoracore6 | the hold code like this http://pastebin.com/601600 |
| 17:09.46 | jarrod | hey I'm getting these Avoiding deadlock errors that seem to be hosing asterisk |
| 17:10.02 | jarrod | it doesnt crash so I don't get a core file... but it stops calls |
| 17:11.37 | noky | hi |
| 17:11.57 | FuriousGeorge | ManxPower: |
| 17:13.53 | noky | must i put in sip.conf all sip's users of my phones? |
| 17:13.53 | ManxPower | noky, usually yes |
| 17:13.54 | noky | but.. i have so much users... |
| 17:13.54 | jarrod | WARNING[23080] channel.c: Avoided deadlock for '0x828a048', 10 retries! |
| 17:13.54 | caio1982 | noky: try a database backend then |
| 17:14.07 | [TK]D-Fender | Fedoracore6 : "extension" isn't a constant it should be a STRING to match the array |
| 17:14.15 | [TK]D-Fender | $exten= $agi['extension']; |
| 17:14.20 | noky | ok... mmm.. i must to add this interface between a DB and Asterisk for then... no? |
| 17:14.23 | noky | not? |
| 17:14.24 | [TK]D-Fender | assuming thats even the valid parameter. |
| 17:14.52 | riksta | noky: voip-info.org search for realtime |
| 17:15.22 | SplasPood | anyone fammilar with asterisk's behavior when it has no callerid number specified by the client? It seems to be generating one from the [context] in sip.conf ?? |
| 17:15.44 | FuriousGeorge | ManxPower: http://pastebin.ca/45693 |
| 17:15.46 | noky | ? |
| 17:15.53 | cytrak | hey guys I got a wireless client that for some reason cannot connect to * ... I'm using iax |
| 17:15.54 | FuriousGeorge | hangup and not detected |
| 17:15.57 | noky | thanks |
| 17:16.04 | cytrak | all wired clients work fine |
| 17:16.34 | [TK]D-Fender | SplasPood : if a user account in sip.conf has "callerid" set in its definition * ignores whats sent by the phone. |
| 17:16.39 | FuriousGeorge | well, hangup and not set to offhook |
| 17:16.50 | FuriousGeorge | s/onhook/offhook |
| 17:17.05 | ManxPower | FuriousGeorge, if it's not detected then your telco is not dropping battery |
| 17:17.05 | SplasPood | [TK]: In this case asterisk has no callerid= setting, and the client isn't sending a valid CID, so its taking the NUMBERS from the [context] in sip.conf and making THAT the caller id |
| 17:17.12 | SplasPood | I cannot find this behavior discussed or documented anywhere |
| 17:17.26 | noky | but.. i only know if Asterisk have some configuration in sip.conf that allow to interact with a DB that storage all sip's users... because, in other form, i must to code... |
| 17:17.27 | Juggie | is voip-info down? |
| 17:17.31 | [TK]D-Fender | SplasPood : makes no sense... |
| 17:17.38 | SplasPood | ie if its [sipuser12345] the CID would be 12345 |
| 17:17.39 | FuriousGeorge | ManxPower: could internal wiring problems cause that? |
| 17:17.40 | SplasPood | yes I know |
| 17:17.43 | [TK]D-Fender | Splas : sure yoy didn't set it somewhere in the device? |
| 17:17.48 | SplasPood | but this happens with both 1.0 and 1.2 |
| 17:17.48 | ManxPower | FuriousGeorge, prolly not |
| 17:17.49 | SplasPood | SURE. |
| 17:17.53 | ManxPower | give a min |
| 17:17.58 | FuriousGeorge | and if its not dropping battery, how come asterisk detected a hangup at the top |
| 17:17.59 | [TK]D-Fender | SplasPood: INSANITY.... |
| 17:18.06 | SplasPood | [TK]D-Fender: TOtal. |
| 17:18.09 | FuriousGeorge | then zap show channel that hungup and its offhook |
| 17:18.09 | [TK]D-Fender | Juggie : Apparently |
| 17:18.15 | jarrod | What is happening when Asterisk is Avoiding a deadlock on channels |
| 17:18.18 | mocker | Can a TE410P card have more than one primary timing source? i.e. primary for span one, and another primary for span two? |
| 17:18.36 | ManxPower | FuriousGeorge, it's detecting the SIP hangup. But as I said, as soon as my FXO is available I'll look at it. |
| 17:19.04 | ManxPower | the hookstate is prolly only for FXS ports, as it says on the output |
| 17:19.04 | jarrod | mocker each pri can have have different timing sources |
| 17:19.05 | ManxPower | mocker, no. |
| 17:19.05 | jarrod | one can be external, and another internal |
| 17:19.09 | mocker | Conflicting responses.. :) |
| 17:19.14 | ManxPower | Digium cards can only have one primary timing source per card. |
| 17:19.19 | jarrod | oh |
| 17:19.25 | jarrod | :) |
| 17:19.26 | jarrod | digium sucks |
| 17:19.37 | ManxPower | i.e. only 1 span can be the primary timing source |
| 17:19.55 | jarrod | give me some software dsps and a single primary timing source and buggy software! |
| 17:20.04 | jsharp | You can have multiple timing sources, but they are set in priority levels if one span goes down. |
| 17:20.21 | SplasPood | [TK]D-Fender: Beyond finding some mention of this behavior, it'd be nice if there was a way to say "Use this callerid= assuming the client sends nothing" |
| 17:20.33 | SplasPood | I suppose I can do it in the dialplan, but... |
| 17:20.42 | asterboy | Getting an echo on my SIP connection, are there any settings to help cancel this? |
| 17:20.55 | noky | can i ? |
| 17:20.56 | [TK]D-Fender | SplasPood : You could, but frankly I feel that callerid is best set in the sip.conf definition direct... |
| 17:21.03 | asteriskmonkey | anyone play with asterisk and h323 |
| 17:21.10 | jarrod | yes i ran openh323 |
| 17:21.14 | mocker | ManxPower: So our config of http://pastebin.com/601621 makes no sense at all then? |
| 17:21.16 | [TK]D-Fender | asterboy : SIP to SIP calls? |
| 17:21.18 | jarrod | but just to interface to ld carriers |
| 17:21.20 | asterboy | Getting an echo on my SIP connection, are there any settings to help cancel this? |
| 17:21.20 | SplasPood | [TK]D-Fender: Yea, but what if I have a customer using us for termination that wants to pass their own CID |
| 17:21.22 | jarrod | no GateKeeper |
| 17:21.26 | ManxPower | noky, yes, see the "realtime" options in 1.2 |
| 17:21.29 | asteriskmonkey | jarrod: so you have to download a seperate package and install it? |
| 17:21.32 | noky | thanks ManxPower |
| 17:21.42 | SplasPood | [TK]D-Fender: 99.9% of our customers are end-users, thus we set the CID.. And thus I've never noticed this before :) |
| 17:21.46 | ManxPower | asterboy, no, echo has to be canceled at the PSTN interface |
| 17:21.50 | noky | thanks riksta too |
| 17:21.52 | salviadud | why can't i get mixmonitor to record mp3's? |
| 17:21.52 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 17:21.53 | fu3 | allo! |
| 17:21.54 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 17:21.54 | [TK]D-Fender | Splas : Ok, I've done what you've described and never see that before... friggen wierd... and you says its been documented? |
| 17:22.14 | asterboy | ok, so the wcfxo can have something setup? |
| 17:22.17 | fu3 | is the wiki down or is it just me? |
| 17:22.18 | jarrod | manx: your services are invaluable... if you never receive praise then let me tell you, thanks. |
| 17:22.25 | [TK]D-Fender | fu3 : Salut mon ostie! |
| 17:22.27 | SplasPood | [TK]D-Fender: No documentation that I've been able to find... But I've been able to duplicate the behavior over and over and on both 1.2 as well as 1.0 |
| 17:22.33 | [TK]D-Fender | fu3 : its down.... |
| 17:22.45 | ManxPower | asterboy, Um, you mean like echocancel=256 and echotraining=900 ? |
| 17:22.54 | [TK]D-Fender | Splas : You sure its not THEIR end? |
| 17:22.57 | ManxPower | jarrod, I'd rather receive cash 8-) |
| 17:22.59 | asterboy | yes , I think I've seen those settings before. |
| 17:23.00 | fu3 | trop mauvais |
| 17:23.06 | ManxPower | But I am available for short term consulting |
| 17:23.08 | SplasPood | [TK]D-Fender: "Their" in my testing case is me |
| 17:23.11 | asterboy | do they go in zapata.conf? |
| 17:23.11 | riksta | noky: ok |
| 17:23.16 | SplasPood | [TK]D-Fender: so yea I'm sure |
| 17:23.21 | ManxPower | asterboy, read the docs |
| 17:23.39 | [TK]D-Fender | fu3 : c'est l'enfer sans avoir nos ressources disponibles a tout-le-temps... |
| 17:23.40 | asterboy | ok, thx |
| 17:23.49 | ManxPower | hell, just the zapata.conf.sample should be enough, other docs would be even more helpful |
| 17:23.50 | ManxPower | ~docs |
| 17:23.58 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:23.58 | [TK]D-Fender | :/ |
| 17:24.33 | fu3 | :) |
| 17:24.36 | fu3 | I hate French |
| 17:24.45 | fu3 | I had to take it for 14 god damned years |
| 17:25.02 | [TK]D-Fender | fu3 : I work in french so its just a fact of life. Nothing against it personally.... Quebec politics on the other hand.... |
| 17:25.08 | jsharp | I'll let you be my translator when I have to call France Telecom. |
| 17:25.11 | fu3 | ugh.. thats where im coming from. |
| 17:25.19 | jarrod | is there any docs on deploying asterisk servers in a provider model... with failover, cluster, and scalable environments? |
| 17:25.22 | fu3 | nothing against the people (not the 51% that said no to seperation). |
| 17:25.28 | ManxPower | FuriousGeorge, the hookstate is not valid for fxo ports. |
| 17:25.36 | cytrak | is there a way to increase the Unreachable thing on asterisk ? |
| 17:25.37 | ManxPower | my fxoports show offhook and they are on hook |
| 17:25.50 | FuriousGeorge | yeah, i JUST noticed that |
| 17:25.54 | ManxPower | that's prolly why it says "Hookstate (FXS only): Offhook |
| 17:25.55 | ManxPower | " |
| 17:25.55 | cytrak | If I don't use the phone for like 2 minutes I get unreacheble |
| 17:26.00 | FuriousGeorge | the difference between now and before is that its working thogh |
| 17:26.04 | ManxPower | jarrod, the wiki |
| 17:26.09 | fu3 | is down |
| 17:26.25 | fu3 | and I need to learn how to properly handle these DID's ive got. |
| 17:26.25 | ManxPower | cytrak, set qualify=no |
| 17:26.37 | cytrak | but i like that |
| 17:26.37 | fu3 | come on.. google cache.. |
| 17:26.38 | FuriousGeorge | ManxPower: at one point i restarted the server while i was mulling and outbound started working again for zap |
| 17:26.48 | cytrak | it tells me when some one is online |
| 17:26.57 | asterboy | holyl crap, voip-info.org is timing out |
| 17:27.06 | jarrod | manx: which wiki is that |
| 17:27.08 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 17:27.14 | fu3 | Zut! Zut! Zut! |
| 17:27.15 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
| 17:27.34 | asterboy | voip-info.org is down for me...anyone else reporting same? |
| 17:27.36 | ManxPower | cytrak, ALL qualify does is send a SIP OPTIONS packet and measure how long it takes to get a response. |
| 17:27.44 | fu3 | yes |
| 17:27.45 | fu3 | its down |
| 17:27.58 | fugitivo | down |
| 17:28.02 | fu3 | perhaps the topic should be modified? |
| 17:28.05 | asterboy | digium shoudl have the skinny on that echocancellation. |
| 17:28.24 | fu3 | the google cache of the DID wiki page is weak :? |
| 17:28.39 | CoffeeIV | I wish to find a IAX2 termination/origination service that will hopefully will do faxes, I was about to try iax.cc but they require paypal. Any suggestions ? |
| 17:28.46 | hfb | The other day at a local Linux User Group meeting, a member did a demo of Asterisk. Now I'm curious and have questions. |
| 17:29.05 | fu3 | ask |
| 17:29.12 | asteriskmonkey | is voip-info down? |
| 17:29.14 | fu3 | yes! |
| 17:29.15 | hfb | Does anyone now anything about Strata Systems? More specifically the phones? |
| 17:29.18 | ManxPower | YES VOIPINFO IS DOWN |
| 17:29.32 | fu3 | god dammit |
| 17:29.32 | ManxPower | would an op change the /topic |
| 17:29.32 | jarrod | IS VOIP INFO DOWN!?! |
| 17:29.32 | jarrod | hahah |
| 17:29.37 | fu3 | Holy SHIT!! THE WIKI IS DOWN!!! |
| 17:29.38 | fu3 | RUN!!! |
| 17:29.51 | hfb | I'm curious if I'm able to use the phones some how with an Asterisk based system. |
| 17:29.52 | asteriskmonkey | damnit just when i needed to look up h323 support too :P |
| 17:29.52 | fu3 | Anyone see that SNL sketch where the teleprompter went down, and the newspeople started freaking out? |
| 17:29.58 | fu3 | Thats whats happening here. |
| 17:30.06 | jarrod | asteriskmonkey: you need openh323 |
| 17:30.08 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Yes, voip-info.org is currently down. |
| 17:30.11 | jarrod | thats not at voip-info |
| 17:30.20 | asteriskmonkey | yes was looking for some more detail install instructions :P |
| 17:30.29 | jarrod | you need pwlib and openh323 |
| 17:30.42 | asteriskmonkey | so what just download make install and im done? |
| 17:30.46 | asteriskmonkey | cant be that easy |
| 17:30.56 | jarrod | hey.. when it says voip-info.org is currently down does that mean I can't browse to it? |
| 17:30.57 | jarrod | haha |
| 17:31.07 | ManxPower | asteriskmonkey, There IS NOTHING easy about h323 |
| 17:31.14 | asterboy | that does it, can't use asterisk now that wiki is down, my whole system is melting...help |
| 17:31.18 | jarrod | asteriskmonkey: what are you gunna use it for |
| 17:31.19 | ManxPower | First off there are at least 5 H323 drivers for Asterisk |
| 17:31.20 | hfb | I did order a TDM400P with both FXS and FXO for testing. |
| 17:31.23 | file[laptop] | russellb: !!! |
| 17:31.26 | asteriskmonkey | jarrod: ld provider |
| 17:31.28 | russellb | file[laptop]: !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 17:31.52 | fu3 | hfb.. I like my TDM400P but it DID have hardcore echo problems. |
| 17:32.12 | file[laptop] | russellb: I made it! but my luggage isn't here yet. |
| 17:32.37 | russellb | file[laptop]: i'm sorry :* |
| 17:32.38 | *** join/#asterisk Derkommissar (n=Alberto@66.64.215.6.nw.nuvox.net) |
| 17:32.38 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 17:32.42 | russellb | er, :( |
| 17:32.52 | hfb | fu3, Not sure about that yet. I have the card ordered, but I do remember seeing something in the faq's about that. |
| 17:32.59 | Derkommissar | I have a perl script, an agi. that always works and does what its supposed to |
| 17:33.06 | Derkommissar | but it creates a core dump |
| 17:33.17 | gambolputty | Can the on/off pulse width in the SendDTMF command be changed? |
| 17:33.34 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 17:33.59 | Derkommissar | :-/ |
| 17:34.02 | fu3 | hfb.. thats ok. Hopefullt it wont even affect you. |
| 17:35.11 | Derkommissar | also im experiencing something wierd, new sessions of asterisk are starting, i though it was because i was using safe asterisk, so i stopped using it, and i find sometimes duplicate sessions of asterisk other than the one i mannually started myselt |
| 17:35.39 | FuriousGeorge | someone mentioned new echocan in asterisk 1.2 where is that line i gotta uncomment? in asterisk.c? |
| 17:35.53 | fu3 | THE CLAN OF THE HAND WILL LIVE FOREVER |
| 17:36.05 | salviadud | clan of the hand? |
| 17:36.11 | salviadud | as in the hand that strokes? |
| 17:36.13 | fu3 | I'm just cracking up thinking about an old SNL sketch |
| 17:36.16 | salviadud | or jacks? |
| 17:36.18 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 17:36.39 | salviadud | it's been a while since i see some SNL... |
| 17:36.48 | fu3 | same here.. i guess we're not missing much. |
| 17:37.14 | Fedoracore6 | [TK]D-Fender thank bro ... i maded :) |
| 17:37.17 | *** join/#asterisk sag_ich_nicht (i=bitch2k@85-124-13-18.dynamic.xdsl-line.inode.at) |
| 17:37.22 | Derkommissar | here is a trace of that core dump by an agi,,,, it works and it does everything its suposed to |
| 17:37.29 | Derkommissar | but still leaves a core dump |
| 17:37.31 | Derkommissar | http://pastebin.com/601648 |
| 17:37.37 | hfb | fu3, Hmmm. I just read the faq's. I wonder if using separate cards (non-zaptel) would work? One for FXS and one for FXO. |
| 17:37.44 | fu3 | probably not. |
| 17:37.54 | fu3 | get the single card first |
| 17:38.06 | fu3 | I wouldnt start speculating on the issues until you're up against them |
| 17:38.17 | hfb | Yeah, I already did that. One card ordered. |
| 17:38.21 | [TK]D-Fender | Fedoracore6 : Next time don't just post the error, post the SOURCE of the error as well... |
| 17:38.28 | fu3 | thats what I did. One TDM400P with two FXO and two FXS ports. |
| 17:38.42 | fu3 | With that, you'll be able to test analog phones and VoIP phones. |
| 17:39.28 | hfb | fu3, First I have to make sense of everything. :) |
| 17:39.38 | fu3 | yeah.. i hear that. |
| 17:39.45 | Derkommissar | this is wierd |
| 17:39.47 | fu3 | take your time.. ask the questions you need to ask. |
| 17:39.56 | jsharp | four days of Asterisk and skiing. Heee. |
| 17:40.45 | fu3 | When I asked my telco about how many DID digits were being sent, they said "cease the whole trunk" |
| 17:40.50 | Derkommissar | anyone here experienced the same problem |
| 17:40.52 | fu3 | does that make ANY sense to anyone? |
| 17:41.02 | Derkommissar | im not running safe_asterisk |
| 17:41.10 | Derkommissar | and asterisk still starts new sessions |
| 17:41.18 | fu3 | How can I find out how many digits are being received by my PBX? |
| 17:41.19 | Derkommissar | i dont understand |
| 17:41.31 | fu3 | I dont see anything with debug, and I dont see anything come across the dump from the T1 card. |
| 17:41.32 | Derkommissar | http://pastebin.com/601659 |
| 17:41.37 | jsharp | Send some calls to it? |
| 17:41.41 | jsharp | Oh. |
| 17:41.43 | *** part/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
| 17:42.08 | ManxPower | Derkommissar, define "new session" |
| 17:42.25 | Derkommissar | im going to paste 2 lines |
| 17:42.27 | Derkommissar | root 22940 75.8 33.9 401252 350780 ? Sl Mar13 949:34 asterisk -vvgc |
| 17:42.27 | Derkommissar | root 12610 0.0 28.1 342228 291340 ? S 10:27 0:00 asterisk -vvgc |
| 17:42.39 | ManxPower | Derkommissar, you do not understand threads and PS. |
| 17:43.05 | jsharp | fu3: You running PRI or robbed-bit T1? |
| 17:43.05 | Derkommissar | well i understand this much |
| 17:43.05 | x86 | where can i get a list of country code numbers to ISO abbreviations? |
| 17:43.05 | x86 | like +1 => US |
| 17:43.06 | x86 | etc |
| 17:43.06 | fu3 | robbed-bit T1 (I'm almost certain) |
| 17:43.08 | ManxPower | they are difrerent threads. It is not an issue. It is normal. Some versions if "ps" print out the threads, others only print out the processes. |
| 17:43.28 | Derkommissar | well for a whole day it was just one process |
| 17:43.30 | jsharp | If you look in the debug log file, you should be able to see what digits are being dialed into the zaptel channel |
| 17:43.37 | Derkommissar | and it keeps adding at least one a day |
| 17:43.45 | fu3 | hmm.. i push the debug info to the console, and do not see anything. |
| 17:43.46 | Derkommissar | untill the machine runs out of resourses |
| 17:43.46 | FuriousGeorge | its bad if the wctdm is sharing an irq w/ usb, even if usb isnt being used in the box, right? |
| 17:43.50 | fu3 | i'll do it again, just to make sure. |
| 17:43.52 | fu3 | i'll also check that log. |
| 17:43.53 | ManxPower | Derkommissar, and then I'll bet you started using MoH, which, of course, would require a second processes, so would AGIs and System |
| 17:44.06 | Derkommissar | and im not running safe_asterisk |
| 17:44.13 | Derkommissar | No im not loading MOH |
| 17:44.15 | ManxPower | Derkommissar, that has NOTHING to do with it. |
| 17:44.23 | jsharp | fu3: And what signalling are you running on your T1 channels? |
| 17:44.30 | ManxPower | ALL safe_asterisk is, is a shell scropt |
| 17:44.33 | Derkommissar | well at first i though it was safe_asterisk's fault |
| 17:44.39 | ManxPower | Derkommissar, what specific PROBLEM are you having? |
| 17:44.51 | ManxPower | What doesn't work? |
| 17:45.01 | Derkommissar | new threads of asterisk start |
| 17:45.02 | fu3 | fu3.. e&m wink |
| 17:45.03 | fu3 | wow |
| 17:45.10 | fu3 | jsharp.. e&m wink :) |
| 17:45.14 | Derkommissar | slowly untill the machine is out of resourses |
| 17:45.15 | jsharp | uh. |
| 17:45.26 | ManxPower | Derkommissar, What version of Asterisk? |
| 17:45.27 | Derkommissar | in a couple of days it will be 5 |
| 17:45.37 | jsharp | Is that what your telco says to run? And are they sending e&m wink with DTMF or MF? |
| 17:45.45 | fu3 | DTMF |
| 17:45.47 | ManxPower | Derkommissar, sounds to me like you are using AGIs that do not exit properly |
| 17:45.48 | Derkommissar | SVN-branch-1.2-r10137 |
| 17:45.54 | fu3 | yes, its what they say to run. |
| 17:46.19 | Derkommissar | Manx, you may have hit the nail on the head |
| 17:46.27 | Derkommissar | i use a lot of agi |
| 17:46.34 | Derkommissar | but 1 script only |
| 17:46.39 | Derkommissar | and it always exits |
| 17:46.44 | ManxPower | Are you catching SIPHUP to exit the scropt? |
| 17:46.54 | ManxPower | if you are not catching it, that could be the problem |
| 17:47.08 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:47.13 | Derkommissar | for some reason the script is leaving a bunch of core dumps |
| 17:47.18 | Derkommissar | no i dont think im |
| 17:47.40 | Derkommissar | :-/ |
| 17:47.41 | ManxPower | Derkommissar, You need to fix the core dumps first. |
| 17:47.54 | Derkommissar | they are not verry usefull |
| 17:48.02 | Derkommissar | not giving me much info to go by |
| 17:48.23 | Derkommissar | http://pastebin.com/601648 |
| 17:48.34 | Derkommissar | i dint think one problem had anything to do with an other |
| 17:48.45 | Derkommissar | thats the bt of a coredump |
| 17:48.52 | ManxPower | Derkommissar, running asterisk in the foreground will allow you to see the STDERR of the script |
| 17:49.21 | fu3 | can I see an example of an extensions.conf in which someone is actually using DID? |
| 17:49.55 | FlyboySR22 | anyone still having problems with NuFOne...? |
| 17:49.57 | ManxPower | fu3, exten => _1XX,1,Dial(Zap/g1/${EXTEN}) |
| 17:50.01 | Derkommissar | let me see |
| 17:50.08 | Derkommissar | manx here is the perl script |
| 17:50.13 | ManxPower | that is for our 3-digit DIDs and they all get sent to the PBX |
| 17:50.17 | Derkommissar | http://pastebin.com/601685 |
| 17:50.24 | fu3 | and all of your Did's start with 1 right? |
| 17:50.26 | Derkommissar | i do a proper exit at the end of the script |
| 17:50.29 | *** join/#asterisk chr|s_ (n=chris@217.171.52.110) |
| 17:50.29 | fu3 | and the last two digits are variable? |
| 17:50.34 | chr|s_ | hey peeps |
| 17:51.01 | ManxPower | Derkommissar, looks like you are catching the signals using the callback, so there must be some other problem with your script. |
| 17:51.07 | ManxPower | fu3, correct |
| 17:51.28 | fu3 | thanks |
| 17:51.57 | ManxPower | fu3, DIDs are simple, E&M Wink might not be so simple. |
| 17:52.09 | Derkommissar | $AGI->setcallback(\&callback); |
| 17:52.20 | Derkommissar | ops that was supose to be once |
| 17:52.20 | *** join/#asterisk gniretar_work (n=mark@gateway.meteor-web.com) |
| 17:52.23 | gniretar_work | hi all |
| 17:52.33 | Derkommissar | the coredump is because of that agi command ? |
| 17:52.37 | Derkommissar | i dont use it at all |
| 17:52.39 | gniretar_work | hey, is it jsut my connection or is the VOIP wikki down? |
| 17:52.55 | fu3 | what might be unsimple about E&M Wink? |
| 17:52.59 | ManxPower | Derkommissar, I doubt it. that callback is what allows your processes to exit if the user hangs up. |
| 17:53.05 | ManxPower | gniretar_work, THE WIKI IS DOWN |
| 17:53.17 | Derkommissar | i see |
| 17:53.24 | *** join/#asterisk Utah_Dav1 (n=boucha@0-1pool139-79.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:53.24 | ManxPower | Derkommissar, so your AGI problem is somewhere else in your script |
| 17:53.28 | gniretar_work | lol, i see it in the topic now |
| 17:53.29 | gniretar_work | lol |
| 17:53.40 | salviadud | just search on google |
| 17:53.42 | salviadud | and hit |
| 17:53.44 | salviadud | cached |
| 17:53.44 | *** part/#asterisk Utah_Dav1 (n=boucha@0-1pool139-79.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:53.48 | Derkommissar | :-/ |
| 17:53.58 | salviadud | that's what i did |
| 17:54.12 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-79.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:54.14 | gniretar_work | hey, then i guess i'l just ask here. my telco is saying I am only sending a local extension for my CID info. How do i get Asterisk to send my 800 number and the correct CIDNUM? |
| 17:54.23 | gniretar_work | er, the 800 num and the right CIDNAME |
| 17:54.29 | Derkommissar | what can i do to catch where the problem is |
| 17:54.41 | Derkommissar | the dificult part is that the script does do what is suposed to |
| 17:55.01 | brettnem | show function CALLERID.. I think |
| 17:55.29 | Derkommissar | i route all my calls trough it |
| 17:56.01 | Derkommissar | http://pastebin.com/601700 |
| 17:57.07 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:58.21 | Derkommissar | ManxPower, can i msg you? |
| 17:58.47 | mutilator | no |
| 18:03.17 | ManxPower | Derkommissar, you may /msg me if you provide a credit card for billing my private consulting time. Otherwise, talk on the channel. |
| 18:03.42 | ManxPower | I don't know what you expect me to do. I'm certinally not going to debug your AGI script. |
| 18:05.40 | willt | what does pedantic=yes do? |
| 18:05.47 | brettnem | wahoo.. what have I been missing?! |
| 18:05.47 | file[laptop] | willt: lots |
| 18:06.02 | fu3 | hahahaha.. why am I having these problems understanding how DID works in asterisk. |
| 18:06.02 | willt | well it just broke my outbound calling |
| 18:06.04 | fu3 | it should be easy |
| 18:06.07 | ManxPower | willt, what does sip.conf.sample say it does? |
| 18:06.10 | fu3 | and it will be, once im done. |
| 18:06.11 | file[laptop] | then don't use it |
| 18:06.17 | file[laptop] | you shouldn't use it under normal circumstances |
| 18:06.28 | ManxPower | fu3, Asterisk thinks of a DID as someone picking up the line and dialing the digits |
| 18:06.35 | file[laptop] | it's for trained professionals who have a reason to use it |
| 18:06.52 | fu3 | oh really? i thought it tagged the numbers somehow and just sent them to asterisk. |
| 18:06.56 | fu3 | that makes way more sense. |
| 18:07.13 | brettnem | what?! |
| 18:07.14 | *** join/#asterisk lorinc (n=ang@caracas-2036.adsl.interware.hu) |
| 18:07.20 | brettnem | you guys are crazy |
| 18:07.35 | file[laptop] | yes I admit I'm crazy |
| 18:07.36 | mutilator | pooooooo |
| 18:07.38 | file[laptop] | what'cha gonna do about it? |
| 18:07.38 | ManxPower | fu3, well with E&M/Wink getting the digits into asterisk is somewhat more complicated, but once Asterisk gets the digits it's just like someone picked up the line and dialed the digits. |
| 18:07.54 | fu3 | cool.. that made a bunch of things "click" for me. |
| 18:07.56 | ManxPower | willt, what does sip.conf.sample say pedantic=yes does? |
| 18:08.09 | austinnichols101 | dtmf is like actually dialing the digits and is sent as audible beeps in the voice portion of the channel. DID is delivered as data attached to the call outside of the voice portion. |
| 18:08.34 | willt | ManxPower: Enable slow, pedantic checking for Pingtel and multiline formatted headers for strict SIP compatibility (defaults to "no") |
| 18:08.39 | ManxPower | austinnichols101, that is bullshit |
| 18:08.43 | austinnichols101 | but from the asterisk perspective you don't really care how the digits get there physically |
| 18:08.46 | austinnichols101 | how's that? |
| 18:08.49 | ManxPower | On PRI the info is delivered as data |
| 18:08.54 | austinnichols101 | right |
| 18:08.57 | ManxPower | on all other DID types it's delivered as DTMF or pulse. |
| 18:09.07 | ManxPower | and E&M Wink IS NOT A PRI |
| 18:09.07 | austinnichols101 | DID != DTMF |
| 18:09.16 | noky | if i change the /etc/asterisk/sip.conf i must to restart the daemon asterisk ? |
| 18:09.31 | ManxPower | noky, maybe. try a realod chan_sip.so |
| 18:09.35 | fu3 | hmm.. the telco says that I am getting DID and it's using DTMF signalling. |
| 18:09.54 | noky | thanks |
| 18:10.00 | ManxPower | fu3, DID = Direct Inward Dial. DID can be any number of signalling and digit types. |
| 18:10.03 | Aurs | "sip reload" |
| 18:10.13 | fu3 | ok.. that makes sense. |
| 18:10.37 | austinnichols101 | well stated |
| 18:10.52 | fu3 | so, when my asterisk box gets "1234" from the CO, it will dial based on that as though I picked up a phone and dialed 1234 right here in my office. |
| 18:10.56 | fu3 | neat. |
| 18:10.58 | ManxPower | a Direct Inward Dial call can be delivered on a PRI (out of band), FXS (w/DTMF), E&M Wink (DTMF or pulse) and several other ways I'm too lazy to look up. |
| 18:11.10 | SplasPood | I'd be willing to give some cash to anyone that can explain this asterisk caller id from the [sipcontext] behavior to me.. |
| 18:11.11 | ManxPower | fu3, correct. |
| 18:11.15 | fu3 | thanks.. that was well explained |
| 18:11.17 | file[laptop] | why do something one way, when you can do it tons of other ways too? |
| 18:11.57 | ManxPower | SplasPood, file it as a bug, then be prepared to defend the bug. Expect it to take several days for anyone to admit it's a bug. |
| 18:12.22 | SplasPood | Manx: Well I don't think it's a bug.. seems to me like this is a "feature" just can't seem to find any mention of it |
| 18:12.33 | ManxPower | If it gets closed, then post to asterisk-dev mailing list -- also be prepared to defend your post. |
| 18:12.37 | SplasPood | How does one specify the caller id in X-Lite ? |
| 18:12.39 | brettnem | ManxPower, "MF" |
| 18:12.48 | SplasPood | Display Name seems to send the "name" portion |
| 18:13.01 | fu3 | i have one line thats DID, 2632902 - so I would have exten => _263XXXX,1,Dial(SIP/voip1) |
| 18:13.01 | ManxPower | SplasPood, it's either a bug in the code, or a bug in the documentation, either way it's a bug. |
| 18:13.04 | ManxPower | brettnem, *nod* |
| 18:13.14 | fu3 | and that should dial my voip phone when 263XXXX comes across the link |
| 18:13.23 | ManxPower | fu3, that would depend on how many digits your telco is sending you |
| 18:13.31 | fu3 | ahh.. and they said "cease the whole trunk" |
| 18:13.33 | fu3 | whatever that means |
| 18:13.40 | Hmmhesays | anyone using lcdial right now? |
| 18:14.21 | *** join/#asterisk ckruetze (n=ckruetze@i577A5578.versanet.de) |
| 18:14.28 | ckruetze | Hi |
| 18:15.01 | fu3 | hi |
| 18:15.08 | ckruetze | When will the Digium BRI card be released? |
| 18:15.22 | ckruetze | Or is it out already and I can buy it? |
| 18:15.23 | Abydos313 | when it works |
| 18:15.25 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 18:15.26 | Abydos313 | haha, just kidding |
| 18:15.38 | fu3 | why not ask Digium? |
| 18:15.43 | Fedoracore6 | now i try use the code update , but when i run asterisk the system say "valid mysql result" its i script update wrong |
| 18:15.54 | Fedoracore6 | http://pastebin.com/601726 |
| 18:16.06 | ManxPower | ckruetze, I don't think Digium is even working on a BRI card. |
| 18:16.24 | kippi | hey |
| 18:16.32 | kippi | I am getting this erro |
| 18:16.32 | kippi | r |
| 18:16.33 | kippi | Mar 14 18:14:06 WARNING[3452]: file.c:583 ast_readaudio_callback: Failed to write frame |
| 18:16.45 | kippi | anyideas what this error is? |
| 18:17.04 | ManxPower | kippi, that usually means the caller hung up |
| 18:17.11 | kippi | ah ok |
| 18:17.14 | Abydos313 | maybe you're missing the @ |
| 18:17.22 | kippi | so nothing to worry about? |
| 18:17.23 | ckruetze | ManxPower According to the Von newsletter I got just now, they do |
| 18:18.17 | ManxPower | ckruetze, I don't believe it. |
| 18:18.35 | ManxPower | ckruetze, where on the Digium web site is it talked about? |
| 18:19.03 | *** join/#asterisk LoonaTick (i=LoonaTic@ipd50aa84e.speed.planet.nl) |
| 18:19.08 | LoonaTick | hi |
| 18:19.13 | fu3 | hi |
| 18:19.15 | *** join/#asterisk ToTo (n=ToTo@host146-88.pool8256.interbusiness.it) |
| 18:19.17 | ckruetze | ManxPower, nowhere |
| 18:19.23 | ManxPower | ckruetze, exactly |
| 18:19.28 | LoonaTick | what's the best way to collect 5 digits entered by DMTF |
| 18:19.42 | ManxPower | LoonaTick, "show applications like dtmf" |
| 18:20.02 | LoonaTick | 0 applications found |
| 18:20.05 | ManxPower | sorry, "show applications like read" |
| 18:20.12 | LoonaTick | thanks :) |
| 18:20.18 | ManxPower | LoonaTick, then your asterisk install is screwed up. |
| 18:20.22 | ckruetze | ManxPower, "03/14/06 - Digium Announces its First Basic Rate Interface (BRI) Card at CeBIT and VON" |
| 18:20.42 | ManxPower | ckruetze, URL? |
| 18:20.46 | *** join/#asterisk Los415 (n=los@ssf-office.corp.race.com) |
| 18:20.56 | ManxPower | ckruetze, call digium to ask about it. |
| 18:20.58 | RoyK | hm |
| 18:21.03 | ManxPower | obviously nobody here knows anything about it. |
| 18:21.03 | RoyK | i wonder what chipset |
| 18:21.06 | LoonaTick | ManxPower: This is to read a file or variable, i would like to read the user input entered by dmtf |
| 18:21.14 | LoonaTick | i saw its possible with waitexten |
| 18:21.17 | LoonaTick | or authenticate |
| 18:21.23 | LoonaTick | but don't think any of those match my needs |
| 18:21.25 | ManxPower | LoonaTick, there are several ways |
| 18:21.36 | LoonaTick | should I install a different (thid-party?) application to do this? |
| 18:21.50 | ManxPower | LoonaTick, what does "show application read" tell you? |
| 18:21.59 | LoonaTick | -= Matching Asterisk Applications =- |
| 18:21.59 | LoonaTick | <PROTECTED> |
| 18:22.00 | LoonaTick | <PROTECTED> |
| 18:22.00 | LoonaTick | <PROTECTED> |
| 18:22.03 | ManxPower | LoonaTick, no, you need to learn more about how to get information from the CLI |
| 18:22.17 | ManxPower | LoonaTick, and "show application read" will give you the detailed docs for this |
| 18:22.42 | ManxPower | ya know like "this application reads dtmf from the caller and puts it into a variable. |
| 18:22.43 | LoonaTick | aaah ok |
| 18:22.52 | LoonaTick | sorry man |
| 18:23.47 | LoonaTick | show application like read didn't ring a bill that it could also read dtmf input |
| 18:24.06 | LoonaTick | by the way, i would like to thank you guys for helping me out with the E1 card |
| 18:24.27 | LoonaTick | after all, the reason it didnt work was two gigabit network interfaces |
| 18:25.06 | ManxPower | noky, it would be "show applicationS like read" will give you the summary of all apps that match. |
| 18:25.23 | ManxPower | then you do "show application whatever" to show the detailed docs for the whatever application |
| 18:25.50 | zoa | ManxPower: afaik its going to be one that does echo cancellation etc on board |
| 18:26.12 | zoa | so it would be a higher class card than the existing ones from junghanns etc |
| 18:26.22 | ManxPower | zoa, I'll stick to T-1 ports and Tellabs |
| 18:26.41 | zoa | well, i you only had a bri at home you wouldnt :) |
| 18:26.58 | zoa | bri is the standard phone line in germany i htink |
| 18:26.59 | ManxPower | zoa, true. But I suspect the BRI card will be expensive. |
| 18:27.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:27.27 | zoa | probably yes |
| 18:29.01 | ckruetze | And I just bought a new BRI card from Beronet :( |
| 18:29.23 | noky | i have a kphone an i couldn't connect to my asterisk via a sip's users... appears: 2006-03-14 14:54:02 NOTICE[5644]: chan_sip.c:10854 handle_request_register: Registration from '"Pirulo" <sip:prueba@172.16.210.161>' failed for '172.16.210.107' - Username/auth name mismatch |
| 18:29.30 | noky | and* |
| 18:29.50 | noky | and the username and password is ok =( |
| 18:31.57 | Altair256 | how do I specifiy different settings for my different cards in zapata.conf? |
| 18:32.05 | Altair256 | ie, I have a TE110P and an X100P |
| 18:33.43 | ManxPower | noky, what is the username? |
| 18:34.25 | [TK]D-Fender | noky : Apparently not... |
| 18:34.42 | chr|s_ | http://www.weebls-stuff.com/toons/ultimate+showdown/ << you have to see / hear this!! |
| 18:35.02 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 18:35.20 | Zodiacal | anyone know how to make my phones earpiece not click like call waiting when a second call comes in on another line? i still want the phones base to still ring tho.. |
| 18:35.42 | ManxPower | then you need a [prueba] section for that device. |
| 18:36.10 | ManxPower | Zodiacal, perhaps you might give us a few details like what phone you are using, is it analog or SIP, if it's analog are you using a Digium card or a ATA |
| 18:36.18 | Zodiacal | if i disable the call waiting feature of the phone, it stops the clicking but it also stops ringing.. how come my phone (cisco 7960) thinks another line ringing is call waiting? |
| 18:36.20 | Zodiacal | SIP |
| 18:36.31 | Zodiacal | digium card analog Fxo cards |
| 18:36.56 | ManxPower | Zodiacal, you mean a 2nd call on you ANALOG PSTN line? |
| 18:37.00 | Zodiacal | yea |
| 18:37.04 | Zodiacal | i don't have call waiting service |
| 18:37.06 | ManxPower | Asterisk does not support that |
| 18:37.33 | salviadud | not yet |
| 18:37.36 | Zodiacal | :( |
| 18:37.37 | ManxPower | Zodiacal, please put down the booze and step away from the keyboard. You are giving 2 conflicting answers to each of my questions. |
| 18:37.48 | Zodiacal | i am? |
| 18:37.50 | ManxPower | ManxPower Zodiacal, you mean a 2nd call on you ANALOG PSTN line? |
| 18:37.50 | ManxPower | Zodiacal yea |
| 18:37.50 | ManxPower | Zodiacal i don't have call waiting service |
| 18:38.00 | ManxPower | do you or do you not have call waiting service on your analog line? |
| 18:38.07 | Zodiacal | 2nd call on you ANALOG PSTN line(S)? |
| 18:38.17 | *** part/#asterisk warp (n=warp@goomba.frob.nl) |
| 18:38.19 | Zodiacal | no call waiting service from my phone co |
| 18:38.23 | Zodiacal | just multiple lines |
| 18:38.23 | ManxPower | Zodiacal, a 2nd call on the same line is called "call waiting" |
| 18:38.27 | Zodiacal | with multiple fxo modules |
| 18:38.31 | Zodiacal | differnt line |
| 18:38.47 | ManxPower | I give up. getting information from you is like getting blood from a republican. |
| 18:39.22 | Zodiacal | manxpower i don't have call waiting service from my phone co, i just have 6 phone lines |
| 18:39.42 | Zodiacal | if im on the phone and a second call comes in, i want my phone to ring, but not click like call waiting does |
| 18:39.45 | Zodiacal | and thats whats happening |
| 18:39.49 | Zodiacal | it rings and clicks |
| 18:39.52 | Zodiacal | i want to remove the clicks |
| 18:39.54 | Zodiacal | but leave the ringing |
| 18:40.22 | Zodiacal | i hope that makes sence |
| 18:40.56 | Zodiacal | manxpower make sence? |
| 18:41.33 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 18:42.09 | gaspiz | hi I just installed festival and it creates a lot of warnings like |
| 18:42.23 | gaspiz | Mar 14 07:32:19 WARNING[25732]: utils.c:619 tvfix: warning negative timestamp 0.-1 |
| 18:42.24 | bkw_ | Zodiacal, you want the phone you're already on to ring? |
| 18:42.56 | Zodiacal | yes |
| 18:42.58 | gaspiz | any ideas about the asterisk + festival issue? |
| 18:43.00 | kippi | is anyone using gandstream GXP-200's with the new firmware? |
| 18:43.04 | bkw_ | you need to go back to school and learn how phones work |
| 18:43.10 | bkw_ | thats not like um POSSIBLE |
| 18:43.16 | Altair256 | I RMA'd both my GXP-2000's |
| 18:43.19 | Altair256 | I think they're trash |
| 18:43.22 | fu3 | unless his phone supports multiple lines |
| 18:43.28 | Zodiacal | cisco 7960 |
| 18:43.32 | fu3 | but to ring down a line already in use.. not possible |
| 18:43.33 | Zodiacal | bkw umm, it does it right now |
| 18:43.34 | bkw_ | you can't it will beep |
| 18:43.36 | Altair256 | ordered the Linksys/Sipura SPA-941 in their place |
| 18:43.38 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 18:43.46 | [TK]D-Fender | Altair256 : Better off with a Polycom IP 501 |
| 18:43.49 | Altair256 | I've never been happier... lol |
| 18:43.50 | Zodiacal | it just clicks too |
| 18:43.51 | bkw_ | its a 7960 a second call on the phone while you're talking will BEEP |
| 18:43.57 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 18:44.11 | tasat | hi all, anyone have experience using sox to convert from gsm to mp3? |
| 18:44.13 | bkw_ | I always get a peep |
| 18:44.17 | Altair256 | why isthat, [TK]D-Fender? |
| 18:44.21 | kippi | Altair256: ah ok, trying to default it, but where you are ment to enter the MAC, its not there |
| 18:44.23 | bkw_ | tasat, you coudl say HI first you rude bastard |
| 18:44.24 | SplasPood | Altair256: Yea, Polycom phones are very nice |
| 18:44.25 | fu3 | Grr.. Is there no way to see what numbers Asterisk is receiving from the CO? Or a way to determine that the CO is NOT sending the DTMF signals down the line? |
| 18:44.27 | Zodiacal | bkw it clicks in the earpeice and rings, but if i disable callwaiting feature on the phone itself, it does nothing |
| 18:44.30 | bkw_ | tasat, first compile it with MP3 support |
| 18:44.38 | bkw_ | Zodiacal, as it should |
| 18:44.44 | Zodiacal | but it doesn't beep |
| 18:44.49 | Zodiacal | or ring or anything |
| 18:44.55 | bkw_ | you need to talk to your vendor |
| 18:44.57 | Zodiacal | they goto voice mail |
| 18:45.02 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 18:45.07 | bkw_ | then setup more than one line apperance |
| 18:45.08 | Altair256 | have either of you used the SPA-941? |
| 18:45.10 | bkw_ | and dial them all |
| 18:45.32 | Altair256 | they Polycom IP 501 is $200 and the 1 year warranty is more |
| 18:45.39 | jarrod | what is mgcp condition 14 and why doenst asterisk know how to indicate it? |
| 18:45.42 | [TK]D-Fender | Altair256 : SPA-941 uses SLA for "buddy watch" which isn't supported by *, Plycom provides much better use of LCD and superior sound on both handset ond speakerphone, on the 501 you have 3 potentially fully independant lines capable of handling up to 24 calls EACH, and feels more solid for starters. Also the 501 has 2 ethernet ports. |
| 18:45.47 | SplasPood | Altair: I can get the 501 for $160 |
| 18:45.56 | [TK]D-Fender | Altair256 : You can get the IP 501 for $170USD at www.atacomm.com |
| 18:46.13 | [TK]D-Fender | SplasPood : Where? |
| 18:46.22 | jarrod | what do you mean each line can support 24 calls? |
| 18:46.31 | Altair256 | [TK]D-Fender, have you actually ever held a 941, or are you basing your info on the 841? |
| 18:46.32 | tasat | bkw_: I said hi.... I do have it compiled for mp3 support -- the problem I've got is in the conversion the audio seems sped up and out of sequence.... are you aware of additional settings needed to make a clean conversion? |
| 18:46.32 | SplasPood | [TK]D-Fender: Well I have a sales person over at moredirect, but I'm sure via other vendors as well |
| 18:46.46 | Altair256 | and what is SLA for "buddy watch"? Is that similar to BLF? |
| 18:46.48 | LoonaTick | in asterisk, DBPut writes to a BDB file. Where can I find this file, and can an external application write to this file (would asterisk notice?) |
| 18:46.49 | [TK]D-Fender | jarrod : you can shuffle between multiple calls on a given line key with the cursor arrows in call-waiting like manner. |
| 18:46.57 | tasat | bkw_: and thanks for responding |
| 18:47.00 | LoonaTick | or is there a better way to communicate with the 'outside world' from within the pbx? |
| 18:47.09 | [TK]D-Fender | Splas : got a base link for me to see how their pricing "starts"? |
| 18:47.23 | [TK]D-Fender | Altair256 : I owned and just SOLD my SPA-941 |
| 18:47.24 | noky | the user sip |
| 18:47.29 | noky | and pass sip is ok |
| 18:47.34 | [TK]D-Fender | Altair256 : and run an all-Polycom shop here. |
| 18:47.38 | Altair256 | interesting [TK]D-Fender |
| 18:47.39 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:47.43 | SplasPood | [TK]D-Fender: you gonna be purchasing a few, or just for yourself? If the former I can hook you up with my sales contact |
| 18:47.46 | [TK]D-Fender | I am not Sipura-free :) |
| 18:47.59 | noky | must i configurate something about sip.conf for only allow register a user ? |
| 18:48.06 | SplasPood | [TK]D-Fender: Any solution for the 7 monitored lines limit? |
| 18:48.10 | [TK]D-Fender | SplasPood : from USA it'd be just personal, but I'm opening up my consulting business.... |
| 18:48.38 | ManxPower | LoonaTick, you do not want external apps to write to the database file |
| 18:48.42 | SplasPood | [TK]D-Fender: Dunno if they do direct end-user sales... I don't see why not... Not sure if they list pricing on the website tho |
| 18:48.45 | noky | sorry |
| 18:48.46 | [TK]D-Fender | SplasPood : Its in the works, Polycom is removing their artificial limit very soon, and SIP-B support is in progress for * (+/- Summer 06) |
| 18:48.53 | noky | configurate something more |
| 18:48.55 | ManxPower | SplasPood, that limit is a prolycom issue |
| 18:48.59 | noky | about the filename sip.conf |
| 18:49.01 | SplasPood | [TK]D-Fender: Yea I heard the summer 06 date... Can't wait |
| 18:49.06 | SplasPood | Manx: I know. |
| 18:49.18 | [TK]D-Fender | SplasPood : Like everyone.... you'll HAVE to. |
| 18:49.18 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 18:49.21 | kardecallan | hello, |
| 18:49.24 | LoonaTick | ManxPower: ok, what would you suggest instead to validate a code and perform some code? |
| 18:49.28 | gaspiz | hi, when I try to call festival from agi it gives me a lot of warnings like" negative timestamp 0.-1" . Any ideas? |
| 18:49.34 | LoonaTick | sorry for my probably stupid questions, im quite new to all this |
| 18:49.48 | *** join/#asterisk andrew_p (n=andrew@ip.85.202.191.14.dyn.sub-9.broadband.voliacable.com) |
| 18:49.52 | andrew_p | hello |
| 18:49.56 | ManxPower | LoonaTick, use an external database in your own AGI script |
| 18:50.13 | LoonaTick | thanks, i'll look in to AGI documentation :) |
| 18:50.18 | andrew_p | what beginners guide to asterisk would you recommend, please? :) |
| 18:50.23 | [TK]D-Fender | Altair256 : So if you can reverse your order I'd highly suggest it... |
| 18:50.37 | Altair256 | Only ordered 1 so far |
| 18:50.40 | Altair256 | still not convinced |
| 18:50.41 | ManxPower | ~docs |
| 18:50.47 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:50.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:50.53 | Altair256 | not sure why I should pay more for the Polycom IP 501 |
| 18:51.13 | GerbilNut | andrew_p, "Building Telephony Systems with Asterisk" by PACKT Publishing was quite useful |
| 18:51.14 | andrew_p | thanks a lot, jbot and ManxPower |
| 18:51.25 | Altair256 | afk, /msg me if you need me |
| 18:51.38 | andrew_p | GerbilNut: great, will search that book or publication :) |
| 18:52.11 | [TK]D-Fender | Altair256 : LIke I said, superior call handling, physical & sound quality, 2nd ethernet port, better use of LCD display, and direct presence support. Also better access to call features while on-line (transfer soft-keys etc are annoying on the 941) |
| 18:52.46 | [TK]D-Fender | Anyways, thats it for now... |
| 18:52.55 | tasat | bkw_ : still there? |
| 18:52.57 | ManxPower | [TK]D-Fender, the 301 has 2 ethernet ports |
| 18:53.31 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
| 18:54.20 | [TK]D-Fender | ManxPower : well aware.... |
| 18:54.31 | willt | how does the polycom compare to like a 7960 in terms of sound quality? |
| 18:54.42 | [TK]D-Fender | willt : pretty much the same. |
| 18:54.50 | willt | nice |
| 18:55.04 | kardecallan | I have a system with all the signalling R2 implemented MFC-Br using D/300 plates. I now want to incorporate this signalling in the Asterisk to make use of the VOIP. Somebody power to help me? |
| 18:55.08 | ManxPower | willt, similar, but you don't have to pay extra for the power supply or for the SIP firmware (you have to pay extra for both of those with Cisco) |
| 18:55.10 | fu3 | does asterisk have to have a exten => s,1,BLAH in the default context in order to work properly? |
| 18:55.21 | ManxPower | fu3, no |
| 18:55.26 | ManxPower | unless you need it, of course. |
| 18:55.32 | fu3 | I didnt think so, but its bitching about none. |
| 18:55.44 | fu3 | I seriously think my CO isnt sending me ANY digits with these DIDs |
| 18:55.45 | [TK]D-Fender | willt : Polycom is cheaper and more "open" than Cisco providing standard PoE and other options for a better value to * types... |
| 18:55.48 | ManxPower | exten => s is only run when asterisk gets no dialed digits |
| 18:55.56 | fu3 | ahh ok.. then that confirms it |
| 18:56.00 | willt | hmm maybe ill order up a polycom and check it out |
| 18:56.01 | fu3 | my CO isnt sending me shit |
| 18:56.02 | digime | polycom 500/501 is great |
| 18:56.07 | digime | we want to set up different ringtones so when we receive personal calls to our extensions its different than the main queue ringtone. we already set it up on asterisk; how do we make the change on a poly501? |
| 18:56.13 | [TK]D-Fender | fu3 : what is "default context" ? |
| 18:56.15 | *** join/#asterisk ebag (n=gabe@adsl-69-239-166-49.dsl.renocs.pacbell.net) |
| 18:56.19 | fu3 | [default] |
| 18:56.21 | fu3 | :) |
| 18:56.22 | willt | I tried a sipura handset and did not like the sound quality at all |
| 18:56.49 | andrew_p | thanks again for links and good luck! |
| 18:56.51 | *** part/#asterisk andrew_p (n=andrew@ip.85.202.191.14.dyn.sub-9.broadband.voliacable.com) |
| 18:57.07 | bkw_ | bbl |
| 18:57.13 | ManxPower | fu3, maybe asterisk is just not SEEING the digits |
| 18:57.23 | fu3 | that is very possible |
| 18:57.24 | ManxPower | like if you did something stupid like use immediate=yes |
| 18:57.29 | fu3 | nope |
| 18:57.33 | fu3 | i read about that already :) |
| 18:58.11 | [TK]D-Fender | fu3 : no context should ever be named [default] :/ |
| 18:58.19 | fu3 | well, im just trying to get this working |
| 18:58.28 | fu3 | this will not reflect my production dialplan |
| 18:58.42 | ManxPower | my default context is called INVALID |
| 18:59.30 | fu3 | Hmm.. I dont know what to do next. When I ask how many digits im getting, the telco says "cease the trunk" and nothing more. Asterisk shows no digits, and there is nothing coming across my T1 when i run a trace onit. |
| 18:59.36 | fu3 | although I do know that calls ARE making it across.. |
| 19:00.45 | fu3 | the telco says that my T1 is SuperFrame and that it's D4 and B8ZS, even though I heard that b8zs isnt compatible with D4 and that I should be using AMI, blah blah blah |
| 19:00.52 | fu3 | grrrrr :) |
| 19:01.01 | [TK]D-Fender | I prefer to name each "general" context based on its origin like [misc-sip], [misc-iax], etc.... |
| 19:01.07 | fu3 | I cant wait until im seasond enough to weed through the bullshit |
| 19:01.20 | ManxPower | fu3, in the USA most T-1s are ESF/B8ZS |
| 19:01.20 | fu3 | yeah.. thats fine.. i'll happily debate context layouts with you another time :) |
| 19:01.25 | kardecallan | Sorry! I have difficulty to write English. |
| 19:01.26 | fu3 | yeah.. this is NOT esf. |
| 19:01.38 | ManxPower | then it would be D4/AMI |
| 19:01.44 | fu3 | ok.. ive set it to D4/AMI |
| 19:01.56 | SplasPood | ManxPower: http://bugs.digium.com/view.php?id=6450 that seems semi-related to what I'm seeing.. (I opened my own bug as well) |
| 19:02.07 | fu3 | but the problem still exists in that Asterisk is not seeing the digits sent by the CO, or the CO isnt sending me anything. |
| 19:02.45 | ManxPower | fu3, "asterisk -cvvvddd" |
| 19:02.55 | fu3 | they say "cease the trunk" -- what the fuck does that mean? even the telco cant explain it. |
| 19:03.01 | fu3 | oh |
| 19:03.03 | fu3 | -ddd eh |
| 19:03.04 | fu3 | brb |
| 19:03.07 | backblue | voip-info down? |
| 19:03.17 | fu3 | yes |
| 19:03.19 | ManxPower | backblue, look at the /topic |
| 19:03.28 | backblue | :o |
| 19:03.33 | backblue | ManxPower: sorry. ;) |
| 19:03.37 | fu3 | ManxPower.. still nothing. |
| 19:03.38 | eric_hill | "cease the trunk" probably means "seize the trunk", meaning bring the trunk online. |
| 19:03.52 | fu3 | debug info is there, but no digits.. it doesnt even say that it got none, or what it did reeive. |
| 19:04.02 | eric_hill | Does the channel go high? |
| 19:04.12 | eric_hill | i.e. does asterisk see an inbound call? |
| 19:04.15 | fu3 | yes |
| 19:04.23 | kardecallan | here in Brazil, the signalling R2MFC causes problem because ISDN is used an Asterisk |
| 19:04.33 | fu3 | if I have exten => s,1,Dial(SIP/snom) it WILL route all incoming calls to my desk phone. |
| 19:04.42 | fu3 | but when I try to break it up by the specific number, it faisl |
| 19:04.43 | fu3 | fails |
| 19:04.56 | ManxPower | fu3, if exten => s works then no did will work |
| 19:05.03 | eric_hill | Can you post a debug of the channel during an incoming call (pastebin.com)? |
| 19:05.09 | fu3 | i took out that line ManxPower |
| 19:05.12 | backblue | voip-info down, when i must want it! argggg |
| 19:05.25 | fu3 | right now i've got exten => _29XX,1,Dial(SIP/snom) |
| 19:05.36 | fu3 | and asterisk complains about no "s" in default context. |
| 19:06.00 | backblue | fu3: because you do not have s extension? |
| 19:06.03 | fu3 | correct |
| 19:06.05 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 19:06.26 | fu3 | I understand that it will only default to "s" if no digits are received by Asterisk |
| 19:06.32 | Katty | Hmmhesays: you around? |
| 19:06.36 | eric_hill | (Potentially talking out of my arse) Digit signalling happens in-channel, meaning that you HAVE to answer the call, then listen for digits... |
| 19:06.49 | fu3 | hmm.. maybe |
| 19:06.54 | fu3 | exten => s,1,Answer |
| 19:06.56 | Katty | [TK]D-Fender: or maybe you? (= |
| 19:07.04 | fu3 | doh |
| 19:07.07 | eric_hill | Unless you are on a PRI, you have to sieze the trunk (i.e. answer it) then Listen for (typically 4) digits. |
| 19:07.07 | fu3 | i dont even know what im talking about |
| 19:07.15 | [TK]D-Fender | Katty: mew. |
| 19:07.20 | *** join/#asterisk Iam8up|lappy (n=user@cpe-71-65-112-38.woh.res.rr.com) |
| 19:07.27 | fu3 | eric_hill.. that makes total sense with what im seeing here. |
| 19:07.39 | Katty | [TK]D-Fender: hewwo. think you could give me a hand with diagnosis? |
| 19:07.39 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 19:07.39 | fu3 | so.. how?! |
| 19:07.39 | [TK]D-Fender | Katty : do my best... |
| 19:07.42 | Katty | [TK]D-Fender: or are ya busy doin somethin else? (= |
| 19:07.42 | *** join/#asterisk WereTiger (n=WereTige@CPE00062586f64e-CM0014e8271804.cpe.net.cable.rogers.com) |
| 19:08.01 | Iam8up|lappy | does anyone know of something (preferablly linux) that can be used to test out a network for voip? specifically sip and rtp traffic? |
| 19:08.13 | eric_hill | Maybe Answer the call then Goto a context that has exten => _XXXX,1,Dial(${EXTEN}) in it? |
| 19:08.14 | ManxPower | eric_hill, he thinks he's using E&M Wink |
| 19:08.27 | fu3 | thats what the telco says |
| 19:08.29 | Katty | [TK]D-Fender: k. so, what's happen..sometimes when we call out, we get only static (like one of the phone lines isn't plugged into our cards.) and other times, when people call us..they hear what they think is a fax sync. |
| 19:08.35 | Iam8up|lappy | i'd also be nice to have some numbers (jitter, latency, packet information) if possible |
| 19:08.41 | Katty | [TK]D-Fender: the call them comes to us and we can answer it and such... |
| 19:08.54 | Katty | [TK]D-Fender: have any idea where to start with this? |
| 19:09.03 | [TK]D-Fender | Katty : What kind of interface? And do you suspect the lines, not the phones? |
| 19:09.15 | eric_hill | E&M still requires the trunk to go "high" to accept the call. |
| 19:09.22 | eric_hill | (Right?) |
| 19:09.22 | Katty | [TK]D-Fender: we have a t1 that goes into a channel bank and then we get 8 analog lines from that. |
| 19:09.42 | Katty | [TK]D-Fender: when we take one of the analog lines and plug it into a regular ol phone, and call that line, there's no fax sync noise. |
| 19:09.43 | ManxPower | eric_hill, yes, but asterisk handles that, not Asnwer |
| 19:09.57 | [TK]D-Fender | Katty: So you get static on in/out calls to various outside #'s from various inside phones? |
| 19:10.05 | kardecallan | Is there anybody that can help me in the implementation of the register/line signalling R2MFC to be used in Asterisk? |
| 19:10.13 | eric_hill | ManxPower - k. Again, I'm an asterisk newbie, but I've been around traditional PBXes for years. |
| 19:10.47 | Katty | [TK]D-Fender: the staticy fax/modem noise doesn't happen all the time, but outgoing calls will get static from any phone...and all sorts of people calling us report that faxy sync noise when calling in. |
| 19:10.51 | ManxPower | eric_hill, chan_zap handles the wink and digit collection below the level of extensions.conf |
| 19:11.06 | Katty | [TK]D-Fender: the incoming syncish noise doesn't happen all the time either. |
| 19:11.20 | Katty | [TK]D-Fender: i never get any static when calling echo test or any of our extensions |
| 19:11.22 | ManxPower | Katty, are you sure you have the timing correct? |
| 19:11.34 | [TK]D-Fender | Katty : So basically its the line(s). Have you stress tested each one individually to see if its global to your lines or just a specific one? |
| 19:11.50 | Katty | [TK]D-Fender: stress tested? |
| 19:11.54 | ManxPower | channel banks would by default get their timing from the T-1, so you would want the first digit of your span to be 0 |
| 19:12.01 | Katty | [TK]D-Fender: you mean keep dialing out on the same line over and over again? |
| 19:12.23 | ManxPower | or maybe the 2nd digit, whatever digit indicats timing sync |
| 19:12.28 | *** join/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 19:12.33 | [TK]D-Fender | Katty : as in pump a few calls out on eash line one by one to see if its just 1 line or all of them. |
| 19:12.34 | _Paulo_ | how can I "flash" an unicall channel? |
| 19:12.45 | Katty | [TK]D-Fender: it's not just line one. |
| 19:13.32 | Katty | [TK]D-Fender: i can try to tie up each line one by one and see if i can tell which card maybe. |
| 19:13.46 | [TK]D-Fender | Katty : Ok, so its the whole setup then. What interface cared is the CB on? does it have its own IRQ. any devices in the server that may conflict? Pastebin /cat/proc/interrupts and "ifconfig" |
| 19:14.10 | [TK]D-Fender | Katty : if you're pretty sure its not a specific line we'll look elsewhere first |
| 19:15.25 | Katty | [TK]D-Fender: http://pastebin.com/602142 |
| 19:15.29 | drewr | Where in the asterisk documentation does it describe throttling data for voice QOS? |
| 19:16.10 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 19:16.16 | WereTiger | isn't that handled by the network? |
| 19:16.23 | Katty | [TK]D-Fender: these cards are 4 port digium cards that take analog phone lines...we have 2 cards. |
| 19:16.38 | eric_hill | fu3,ManxPower: Maybe this will help? http://tinyurl.com/focu7 |
| 19:16.55 | drewr | WereTiger: Doesn't Asterisk control the network in a VoIP gateway? |
| 19:17.15 | Katty | [TK]D-Fender: the only other thing that i've noticed is a couple power outages and i had to fsck the hard drive.. |
| 19:17.19 | [TK]D-Fender | Katty : that TDM sharing with your USB = BAD... |
| 19:17.21 | SplasPood | drewr: QoS is enforced by the network |
| 19:17.34 | SplasPood | drewr: asterisk will just set proper tags on it's packets |
| 19:17.43 | Katty | [TK]D-Fender: hmm, yes it is. they need their own irqs. |
| 19:17.44 | drewr | SplasPood: Hmm, ok. |
| 19:17.48 | Katty | [TK]D-Fender: how do i get it on its own irq? |
| 19:18.03 | WereTiger | 802.1p (VLAN tags) give QOS. Asterisk tags, network enforces. |
| 19:18.30 | drewr | So in order to properly segment voice and data traffic, you still need something else. |
| 19:19.12 | WereTiger | you need (correct me if I'm wrong here guys) Layer 3 switches that support 802.1q and 802.1p |
| 19:19.16 | [TK]D-Fender | Katty : thats always the challenge.... 1st big suggestion : CHEAT. Put an analog splitter on each incoming line and when you find you are on a call with static, plug in the phone in parallel with that line. then disconnect the line from the digium card and resume "stand-alone" on the Channel Bank with the phone and see if the static continues. |
| 19:20.12 | Katty | [TK]D-Fender: that's not quite sinking in. |
| 19:20.19 | [TK]D-Fender | Katty : IRQ control is a funny thing... you need to go into your BIOS and see if you can dedicate it an IRQ based on slot or device. if not, start disabling everything you don't require until you run out of those options. Then try changinging slots. |
| 19:20.29 | drewr | It seems as though Asterisk can do something on its own to keep data from overtaking voice. |
| 19:21.04 | WereTiger | drewr: if you find documentation on that, send it my way :) |
| 19:21.06 | justinu | herro #asterisk |
| 19:21.31 | [TK]D-Fender | Katty : put an RJ11 splitter on the analo line between the channel bank and the TDM card on all ports. when you get static, ID the line then plug the analog phone inline with the digium card. both should be on the CB at the same time. then stay on the parallel phone and disconnect the TMD card. thIf the static is gone then its the TDM card at fault. |
| 19:22.02 | Katty | [TK]D-Fender: i've already determined that |
| 19:22.16 | Katty | [TK]D-Fender: i unhooked line 1 and plugged it into a regular analog phone. there was no static. |
| 19:23.14 | [TK]D-Fender | Katty : to be a good test you need to hook the phon in only once static is in progress so you can isolate it. The problem is that this is "intermittent" |
| 19:23.24 | [TK]D-Fender | So you need to pin it down "live" |
| 19:23.53 | Katty | [TK]D-Fender: alrighty. |
| 19:25.30 | fu3 | yeah lads |
| 19:25.36 | fu3 | the CO isnt sending me any digits :( |
| 19:25.40 | fu3 | or im still configuring this wrong |
| 19:25.45 | fu3 | exten => s,1,Answer |
| 19:25.58 | fu3 | exten => s,2,goto(extensions,${EXTEN},1) |
| 19:26.11 | fu3 | and asterisk still reports that it cannot find "s" in extensions context. |
| 19:26.25 | fu3 | which, as far as I know, means that there are no digits for it to pass as the ${EXTEN} |
| 19:26.39 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:26.41 | PakiPenguin | evening |
| 19:26.44 | fu3 | hi |
| 19:26.46 | eric_hill | Does debugging the T1 show the handshake? |
| 19:27.01 | fu3 | when I run a trace on the T1, it shows me NOTHING |
| 19:27.05 | fu3 | :( |
| 19:28.15 | eric_hill | set verbose 10 doesn't show you anything on an inbound call? |
| 19:28.23 | eric_hill | That's not good.... |
| 19:28.25 | fu3 | verbose 10 where? |
| 19:28.40 | eric_hill | Asterisk console CLI>set verbose 10 |
| 19:28.40 | fu3 | im using wanpipemon's "raw hex trace" |
| 19:28.44 | fu3 | hmm.. i'll try that |
| 19:28.55 | salviadud | hey, i just made a call with mixmonitor, and it cuts off early |
| 19:29.05 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:29.06 | salviadud | had it happened to someone over here? |
| 19:29.44 | fu3 | it shows me stuff eric_hill.. but no digits or dtmf info whatsoever. |
| 19:29.54 | eric_hill | Can you put that @ pastebin |
| 19:30.21 | *** join/#asterisk irieKEN (i=irieKEN@adsl-69-225-126-17.dsl.sndg02.pacbell.net) |
| 19:30.22 | fu3 | I can try |
| 19:30.22 | fu3 | brb |
| 19:30.23 | *** part/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 19:30.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:31.22 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 19:32.17 | [TK]D-Fender | fu3 : What kind of signalling do you have on that T1? |
| 19:32.47 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 19:32.56 | [TK]D-Fender | <PROTECTED> |
| 19:33.34 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:34.00 | eric_hill | I think he said E&M wink. Based on ManxPower's comment, chan_zap handles the win start automatically. |
| 19:34.17 | eric_hill | s/win/wink |
| 19:34.27 | [TK]D-Fender | eek |
| 19:34.35 | [TK]D-Fender | fu3 : Go PRI! |
| 19:34.37 | justinu | lol, there's a bug in E&M unless you're running fairly new asterisk |
| 19:34.38 | justinu | i fixed it |
| 19:34.47 | justinu | has to do with the dtmf inpulsing |
| 19:35.14 | _Paulo_ | when I use hungup, * will do an incoditional jump to the h extension?? |
| 19:36.45 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 19:36.47 | [TK]D-Fender | justinu : So inbetween "winks" is when it sends the DTMF for DID purposes? |
| 19:37.15 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 19:37.21 | justinu | CO should go off hook, asterisk should wink. CO should outpulse DTMF |
| 19:37.26 | justinu | usually there isn't a 2nd wink |
| 19:37.36 | justinu | brb |
| 19:38.38 | gaspiz | hi, anyone experienced the following problem? dialing a sip friend (he is logged in) and asterisk still shows no route to destination? |
| 19:38.43 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:38.48 | gaspiz | sometimes it works and sometimes not |
| 19:39.01 | [TK]D-Fender | gaspiz : NAT factor in there somewhere? |
| 19:39.09 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 19:39.20 | gaspiz | no |
| 19:39.55 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:40.12 | [TK]D-Fender | gaspiz : So the phones are both local to *? |
| 19:40.54 | gaspiz | i'm trying the following case: user calling himself |
| 19:41.26 | gaspiz | the call gets to asterisk fine , when it tries to dial itself it says no route to destination |
| 19:41.29 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 19:42.01 | justinu | reboot |
| 19:42.12 | fu3 | sorry guys.. email server needed a reboot. |
| 19:42.14 | fu3 | anyway. |
| 19:42.19 | fu3 | signalling.. E&M Wink on that T1. |
| 19:42.44 | fu3 | [TK]D-Fender.. oh :) hmm.. then how can I get the DID number being called into ${EXTEN} ? |
| 19:42.47 | justinu | is your CO receiving the wink? |
| 19:42.59 | Katty | justinu: mew. |
| 19:42.59 | fu3 | I dont know.. i guess so? |
| 19:43.02 | justinu | you should get them on the phone and ask them |
| 19:43.08 | justinu | katty: meow! |
| 19:43.12 | Katty | justinu: how beith? |
| 19:43.16 | willt | whats the best format to store audio in for an auto atendant menu ie Press 1 for blah 2 for blah blah |
| 19:43.20 | Hmmhesays | wow ubuntu is messed up when it comes to libs |
| 19:43.32 | Katty | Hmmhesays: hey you! |
| 19:43.32 | Katty | Hmmhesays: i heard you were rockin out the other day |
| 19:43.32 | justinu | katty: wedding in 33 days |
| 19:43.34 | justinu | :/ |
| 19:43.36 | justinu | :) |
| 19:43.36 | Katty | justinu: :>>>>>>>>>>>>>>>>>>>> |
| 19:43.37 | justinu | :( |
| 19:43.44 | Katty | justinu: calm down, they're just butterflies. |
| 19:43.44 | Hmmhesays | yeah |
| 19:43.46 | justinu | heh |
| 19:43.47 | Hmmhesays | want to see? |
| 19:43.51 | gaspiz | <[TK]D-Fender>: any ideas why? |
| 19:43.55 | Katty | justinu: breathe! |
| 19:44.03 | Hmmhesays | http://66.173.103.100:4080/pm/jpg |
| 19:44.04 | Katty | justinu: if it goes insane, remember, there's always annullment. |
| 19:44.07 | justinu | lol |
| 19:44.10 | justinu | that won't happen |
| 19:44.12 | Katty | justinu: you don't have to go through divorce legalities. |
| 19:44.18 | Katty | justinu: just pretend it never happened. |
| 19:44.21 | Katty | justinu: good :> |
| 19:44.28 | justinu | brb, reboot |
| 19:44.30 | *** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:44.42 | Katty | Hmmhesays: m'kay |
| 19:45.01 | Katty | Hmmhesays: oh. |
| 19:45.16 | Katty | Hmmhesays: check that url and get back with me, kthx. |
| 19:45.44 | iDunno | evening :) |
| 19:45.56 | Katty | afternoon! (= |
| 19:46.09 | iDunno | :) |
| 19:46.12 | *** join/#asterisk jskcr (n=zaphod@9-pool1.ras14.floca.alerondial.net) |
| 19:46.29 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
| 19:46.36 | fu3 | what variable in the extensions.conf carries the number being dialed? |
| 19:46.38 | Katty | tuxinator_linuxM: allo, duke! |
| 19:46.45 | Katty | tuxinator_linuxM: how's ye ole chap eh? |
| 19:47.37 | Katty | :< |
| 19:47.38 | gaspiz | d it never happened. |
| 19:47.39 | gaspiz | <Katty> justinu: good :> |
| 19:47.55 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:47.57 | Katty | gaspiz: that's what i said, bunny bread. |
| 19:47.59 | gaspiz | sorry wrong paste |
| 19:48.07 | gaspiz | :) |
| 19:48.09 | Katty | (= |
| 19:48.22 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 19:48.31 | WereTiger | bunny bread? *blinks* |
| 19:48.42 | WereTiger | *disturbing mental imagery* |
| 19:48.50 | Katty | WereTiger: freak. |
| 19:49.06 | WereTiger | you read my mind :) |
| 19:50.40 | eric_hill | fu3: ${EXTEN} |
| 19:51.33 | eric_hill | i.e. exten => _2XXX,1,Dial(Zap/g1/${EXTEN}) ; Means that a user dialing 2345 goes out Zap group 1 x2345. |
| 19:52.08 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:52.09 | fu3 | alright.. so this proves that the issues im having deal with the T1 and not my extensions.conf |
| 19:52.09 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@142.131.190.116) |
| 19:52.14 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
| 19:52.26 | justinu | re |
| 19:52.45 | [TK]D-Fender | fu3 : No, you shouldn't need to use "s" and should be doing it like eric_hill just showed. |
| 19:52.59 | fu3 | yep. that makes sense with what im seeing. |
| 19:53.28 | fu3 | it sure looks like my CO isnt sending the digits then |
| 19:53.33 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 19:53.40 | fu3 | or it's not using E&M wink, or I have zapata.conf configured wrong. |
| 19:53.48 | justinu | it's simple, get on the phone and ask the CO if they get your wink |
| 19:53.56 | fu3 | oh yeah.. thats simple. |
| 19:54.02 | fu3 | let me just call my buddies at the CO. |
| 19:54.02 | fu3 | :| |
| 19:54.12 | gaspiz | i'm dialing a sip friend (he is logged in) and asterisk still shows no route to destination. the wierd thing is that this sometimes work and sometimes doesn't, had anyone heard or expirienced this problem? |
| 19:54.19 | justinu | open a trouble ticket with your telco |
| 19:54.20 | Katty | tuxinator_linuxM: :> |
| 19:54.22 | Katty | file[laptop]: :>> |
| 19:54.25 | eric_hill | Get a new CO. The two CO's I use in town are very responsive :) |
| 19:54.26 | justinu | they will eventually put you intouch with the CO staff |
| 19:54.30 | loud | Hi, anyone using P0S3-08-2-00 on 7960s? |
| 19:54.50 | eric_hill | loud: Only with Call Manager Express <ick> |
| 19:55.04 | loud | ah |
| 19:55.09 | loud | ill remove the file then |
| 19:55.10 | loud | hah |
| 19:56.04 | eric_hill | I haven't tried any 7940 or 7960 phones with Asterisk yet. The 7905's work great though. |
| 19:56.10 | RoyK | Katty: http://ak.imgfarm.com/images/today/creators/bmp/bmp0314g.gif |
| 19:56.19 | loud | althought, it said : SIP Flash Image for 7940/7960 IP Phone v8.2(0)- Non-CallManager |
| 19:57.45 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-74-170.aubnin.fios.verizon.net) |
| 19:58.10 | fu3 | ok |
| 19:58.31 | fu3 | i fired an email out to my qwest service rep, and my intertech rep basically telling them that screw your guessing.. VERIFY what im asking. |
| 19:58.42 | justinu | fuck the service tech |
| 19:58.49 | justinu | call the trouble ticket toll free |
| 19:58.55 | justinu | service rep, that is |
| 19:58.55 | *** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz) |
| 19:58.55 | fu3 | no.. the service rep is the only guy I know there who talks to me straight |
| 19:59.01 | fu3 | I CANNOT CALL THEM DIRECT AND OPEN A TICKET!!!!! |
| 19:59.09 | justinu | why is that? |
| 19:59.21 | fu3 | I work for the state of MN, which means I have to submit reqests to InterTech (state agency), who can contact Qwest with trouble tickets. |
| 19:59.25 | justinu | oh |
| 19:59.28 | fu3 | it fucking sucks hard!! |
| 19:59.37 | justinu | well there's no need to shout |
| 19:59.37 | fu3 | I know :) |
| 19:59.37 | fu3 | sorry |
| 19:59.38 | noky | i can register an user sip!!!! |
| 19:59.38 | noky | :D |
| 19:59.39 | fu3 | it's frustrating is all |
| 19:59.45 | justinu | anyways, i was going to say... |
| 19:59.51 | justinu | i'm not convinced E&M wink works in asterisk |
| 19:59.57 | fu3 | oh |
| 19:59.57 | justinu | if you have trouble, ask them to change to E&M immediate |
| 19:59.57 | jsharp | It does. |
| 20:00.25 | jsharp | They're not running something strange like Feature Group D on your line, are they? |
| 20:00.30 | fu3 | no idea.. |
| 20:00.32 | fu3 | i'll ask. |
| 20:00.34 | *** join/#asterisk redondos (n=redondos@200-112-136-108.bbt.net.ar) |
| 20:00.41 | justinu | FGD would be good, imo |
| 20:00.45 | justinu | more info is nice |
| 20:00.54 | eric_hill | Did you ever post the "set verbose 10" results to pastebin? |
| 20:01.20 | redondos | Hello. Calling from SIP 203 to AIX 202 gives this output. Can you please tell me what the error is? http://pastebin.com/602225 |
| 20:01.20 | fu3 | eric_hill.. no.. sorry. |
| 20:01.25 | justinu | fu3: perhaps what you could do is use ztmonitor to record the CO inpulsing |
| 20:01.33 | fu3 | its all console, I dont really want to manually type it all out. |
| 20:01.43 | justinu | i could listen to that recording and tell you if they're sending MF or DTMF |
| 20:01.52 | fu3 | justinu.. the calls come in on random channels, i cant get ztmonitor to monitor ALL of them can I? |
| 20:01.56 | justinu | or nothing, which is likely the case |
| 20:02.10 | justinu | shouldn't be random... least idle perhaps? |
| 20:02.12 | justinu | most idle? |
| 20:02.12 | eric_hill | Copy paste, man, copy paste :) Start up an SSH server on your asterisk box and use putty to get into it from your desktop |
| 20:02.13 | fu3 | the telco has supposedly verified that I AM using dtmf. |
| 20:02.30 | fu3 | no.. whenever i call a specific number, it comes in on a random ds0 channel |
| 20:02.34 | justinu | we're not concerned about what you're sending... we need to know what they send |
| 20:02.39 | fu3 | sometimes its zap8 sometimes zap21 sometimes zap1 |
| 20:03.01 | fu3 | sorry.. I mean that the T1 was using dtmf pulsing. |
| 20:03.14 | fu3 | im about to paste!!! |
| 20:03.15 | fu3 | runk signaling = wink |
| 20:03.15 | fu3 | trunk pulsing = DTMF |
| 20:03.15 | fu3 | signaling = they didn't give me that (based on what you have working on your equipment, I assume it's SF (Super Frame - D4 and B8ZS) |
| 20:03.15 | fu3 | digits to be sent from C.O. = cease the whole trunk |
| 20:03.22 | fu3 | thats exactly what I got from the telco. |
| 20:03.30 | justinu | cease the whole trunk? |
| 20:03.33 | justinu | i think they meant seize |
| 20:03.36 | fu3 | i think so too |
| 20:03.37 | justinu | dude |
| 20:03.40 | justinu | that's fucked up |
| 20:03.53 | fu3 | I am really learning that.. |
| 20:04.03 | justinu | anyways, ask them for 10 digit outpulsing |
| 20:04.24 | jsharp | Something in the back of my head is telling me that when you tell * to use E&M wink, it wants inpulsed digits as MF. I remember having a similar problem a few years ago. |
| 20:04.35 | justinu | it can accept either |
| 20:04.38 | eric_hill | I think the tech's comments are getting lost in the translation through your intertech folks. Sounds a lot like "innatech" to me... |
| 20:04.45 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 20:05.06 | fu3 | yeah.. believe me.. InterTech's WAN guys are TOP NOTCH, but.. FUCK does their voice support LICK BALLS. |
| 20:05.07 | _Paulo_ | Seems that I cant use the flash command on an unicall channel... |
| 20:05.35 | justinu | fu3: "digits to be sent from co = seize the whole trunk" sounds suspicious to me |
| 20:05.42 | chr|s_ | fu3, you pay extra for that service? |
| 20:05.43 | eric_hill | Seriously, don't hold back... Tell us how you REALLY feel... |
| 20:05.48 | fu3 | same here.. when I immediately asked for clarification on that, they went silent. |
| 20:05.56 | RoyK | fu3: spamming the channel? |
| 20:05.59 | chr|s_ | my telco piss me off, but they don't make up for it with any ball licking. *sigh* |
| 20:06.00 | fu3 | chr|s_.. well the taxpayers of the state of MN are. |
| 20:06.04 | fu3 | RoyK?? |
| 20:06.05 | *** join/#asterisk Dr-Linux (n=nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 20:06.09 | justinu | anyways, i think you could start ztmonitor on all channels |
| 20:06.21 | fu3 | i'll try |
| 20:06.28 | justinu | since you probably have no T1 test set, it's the only way to know what's happening |
| 20:06.54 | fu3 | ztmonitor 1-24 -v ? |
| 20:07.00 | justinu | there's a way to make it record to a file |
| 20:07.24 | fu3 | i see that now |
| 20:07.47 | fu3 | god dammit no /dev/dsp :) |
| 20:07.48 | fu3 | brb |
| 20:09.15 | *** join/#asterisk htims (n=pd@Vc9c4.v.pppool.de) |
| 20:10.01 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 20:11.22 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:11.23 | a1fa | tada |
| 20:11.27 | a1fa | upgraded to 1.2.5 |
| 20:11.28 | a1fa | ;P |
| 20:11.33 | a1fa | smells like teen spirit |
| 20:11.33 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 20:11.58 | chr|s_ | Here we are now, entertain us! |
| 20:12.40 | ibob63 | everning all |
| 20:12.42 | a1fa | <fart> |
| 20:13.38 | _Thor | Hello everybody |
| 20:14.07 | redondos | hello |
| 20:14.27 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 20:14.57 | _Thor | has anyone mastered the problem of sip extensions un-registering? |
| 20:15.34 | _Thor | these extensions keep unregistering every 10 seconds! |
| 20:15.55 | a1fa | hehe |
| 20:15.57 | a1fa | common issue |
| 20:16.00 | a1fa | got a firewall? |
| 20:16.02 | a1fa | got a nat? |
| 20:16.04 | _Thor | no |
| 20:16.19 | a1fa | no what? |
| 20:16.29 | redondos | Where can I read about setting up IVRs? |
| 20:16.40 | _Thor | no firewall, if so, it will be a router/firewall, right? |
| 20:16.56 | a1fa | yes |
| 20:16.58 | _Paulo_ | how can I pickup, hangup, wait 750ms and pickup again the same unicall channel? |
| 20:17.02 | a1fa | _Thor : make sure you got |
| 20:17.05 | a1fa | nat = yes; |
| 20:17.06 | a1fa | canreinvite=no |
| 20:17.06 | a1fa | qualify=2000 |
| 20:17.10 | _Thor | no, unless it is within the router |
| 20:17.16 | a1fa | well nat=yes |
| 20:17.29 | _Thor | I have canreinvite=yes |
| 20:17.49 | a1fa | try it with ni |
| 20:17.58 | _Paulo_ | I tried the Flash command, but it doesnot work for unicall channels. |
| 20:18.07 | _Thor | and qualify =300 |
| 20:18.10 | a1fa | canreinvite is only if you have direct communication between the pbx and client |
| 20:18.50 | _Thor | qualify is the number of seconds it will take to re-register? |
| 20:19.13 | _Paulo_ | I tried the "h" extension, but the command Hungup seems to go to "h"without generating a drop call event. |
| 20:19.52 | _Thor | a1fa: in a sipura/linksys, have you tried setting the "registry expires" real low? |
| 20:21.54 | chr|s_ | ok in #java they are talking about sticking things in their ass, and how it isn't *that messy*, seriously, freenode... lol |
| 20:22.14 | a1fa | _Thor : no |
| 20:22.14 | a1fa | why |
| 20:22.33 | a1fa | chr|s_ : shit man. thats nothing... wait till you hear what they talk in #gays |
| 20:22.53 | fu3 | fuck |
| 20:22.55 | chr|s_ | alfa, object oriented programming? |
| 20:22.56 | redondos | ...Where can I read about setting up IVRs? |
| 20:22.58 | fu3 | how come I joined that channel? |
| 20:23.01 | chr|s_ | lol |
| 20:24.40 | fu3 | ok.. well im going to wait patiently for the responses to my emails before going any further with this setup. |
| 20:25.19 | justinu | lol |
| 20:25.25 | justinu | typical state govt employee :P |
| 20:25.26 | *** join/#asterisk brl4n (n=none@ip68-7-243-20.sd.sd.cox.net) |
| 20:25.32 | *** join/#asterisk backblue (n=moo@87-196-66-128.net.novis.pt) |
| 20:25.32 | fu3 | well. what should I do>? |
| 20:25.36 | justinu | in the private sector, we bother people until shit gets done |
| 20:25.43 | justinu | if I have to call you 50 times a day, I will |
| 20:25.45 | fu3 | I have bothered people.. |
| 20:25.52 | fu3 | and im waiting on their responses |
| 20:25.55 | justinu | anyways, i'm just giving you shit |
| 20:25.58 | fu3 | fair enough :) |
| 20:26.05 | fu3 | i give enough of it to people around where I work :) |
| 20:26.18 | lemmy | hi, is anybody using app_conference with * 1.2? i built the .so file and put it into the asterisk modules dir. after an asterisk restart i can use Conference(..) in my dialplan. but when i call the exten i hear nothing and get alot of debug on the console. any hints? |
| 20:27.01 | *** part/#asterisk brl4n (n=none@ip68-7-243-20.sd.sd.cox.net) |
| 20:28.20 | *** join/#asterisk ToTo (n=ToTo@host146-88.pool8256.interbusiness.it) |
| 20:28.52 | a1fa | justinu : i kick and slap and scratch until shit gets done |
| 20:29.02 | a1fa | and then i start yelling |
| 20:29.07 | Katty | how girly. |
| 20:29.16 | Hmmhesays | usr/src/linux-headers-2.6.12-10-386/scripts/gcc-version.sh: line 12: gcc-3.4: command not found |
| 20:29.23 | fu3 | What should I work on while this problem continues? |
| 20:29.29 | a1fa | lol |
| 20:29.30 | Hmmhesays | what is up with that |
| 20:29.32 | Katty | Hmmhesays: didja ever fix that url? |
| 20:29.33 | justinu | slap? |
| 20:29.37 | chr|s_ | fu3 level 10 of neverball |
| 20:29.38 | a1fa | Hmmhesays : you need build esentials |
| 20:29.42 | chr|s_ | in hard mode, I am stumped |
| 20:29.45 | Hmmhesays | what am I missing on that one |
| 20:29.47 | mocker | Would the span timing sources effect zttest scores? |
| 20:29.50 | fu3 | ok |
| 20:29.55 | a1fa | you are missing gcc |
| 20:29.56 | Hmmhesays | I thought I got everything |
| 20:29.57 | fu3 | i dont know what that is, but im down for it |
| 20:30.08 | Hmmhesays | bhwwhaah |
| 20:30.09 | chr|s_ | install neverball, it is fun |
| 20:30.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:30.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:30.10 | a1fa | Hmmhesays : apt-get install build-essential |
| 20:30.43 | Hmmhesays | i was missing g++ |
| 20:30.45 | Hmmhesays | rookie mistake |
| 20:30.53 | Katty | anthm: (= |
| 20:30.59 | Hmmhesays | http://66173.103.100:4080/pm.jpg |
| 20:31.01 | anthm | hi |
| 20:31.01 | Hmmhesays | whoa |
| 20:31.12 | Hmmhesays | http://66.173.103.100:4080/pm.jpg |
| 20:31.13 | a1fa | they call it a nub mistake |
| 20:31.22 | a1fa | :P |
| 20:31.26 | Hmmhesays | i avoid that cause a nub i am not |
| 20:31.37 | a1fa | ok |
| 20:31.41 | Katty | Hmmhesays: i see you're as furry as usual. |
| 20:32.02 | Hmmhesays | yeah |
| 20:32.03 | *** join/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 20:32.08 | a1fa | Katty : he got his ass eaten by mamals :P |
| 20:32.09 | loko | file[laptop] you here? |
| 20:32.11 | Hmmhesays | and i still get teh same error |
| 20:32.26 | Katty | Hmmhesays: don't worry, nothing that a razor can't fix ^_^ |
| 20:32.39 | eric_hill | Hmmhesays: apt-get install gcc make automake |
| 20:32.39 | loko | If anyone uses BroadVoice BYOD, can you please PM me? |
| 20:32.47 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 20:32.50 | a1fa | i am |
| 20:32.51 | a1fa | :P |
| 20:32.58 | a1fa | loko : ask here.. |
| 20:33.20 | a1fa | eric_hill : how about you apt get build-essential nub |
| 20:33.21 | loko | Is there any qay around a $40 activation fee for BYOD |
| 20:33.22 | fu3 | neverball looks hard :) |
| 20:33.24 | loko | any way. |
| 20:33.29 | a1fa | no |
| 20:33.36 | a1fa | gay? |
| 20:33.39 | eric_hill | He said it didn't work. |
| 20:33.43 | a1fa | i didnt have to pay $40 |
| 20:33.47 | a1fa | it was free |
| 20:33.54 | loko | its $40 now |
| 20:33.55 | a1fa | it was like $9.99 to activate it |
| 20:34.05 | loko | and i wonder if it is $40 per line or just per the account - I want multiple lines |
| 20:34.07 | a1fa | really? |
| 20:34.14 | a1fa | per account probably |
| 20:34.15 | a1fa | let me check |
| 20:34.24 | noky | 2006-03-14 17:03:12 WARNING[7119]: config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 20:34.32 | noky | anybody know what is it ? |
| 20:34.42 | justinu | www.cepstral.com seems down |
| 20:34.58 | noky | i'm trying to follow the: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 20:34.58 | a1fa | wow |
| 20:35.00 | a1fa | gays |
| 20:35.08 | a1fa | they did set a $40 activation fee |
| 20:35.12 | a1fa | they must be loosing money |
| 20:35.20 | loko | if you try to add another line do they want to charge you $40 again? |
| 20:35.25 | loko | after you log into your account |
| 20:35.37 | a1fa | no |
| 20:35.50 | a1fa | its $40 per account |
| 20:35.55 | a1fa | you get 3 lines per account |
| 20:35.58 | a1fa | and you can not add more lines |
| 20:36.03 | a1fa | you need to get a different account |
| 20:41.45 | a1fa | unless they fixed this |
| 20:41.45 | Hmmhesays | gcc-version.sh doesn't exist in that path |
| 20:41.45 | loko | ill need to find a different provider then i guess |
| 20:41.45 | loko | someone that will allow my Cisco phone to connect to them |
| 20:41.45 | a1fa | vonage is doing it now |
| 20:41.46 | a1fa | vonScuksage! |
| 20:41.46 | a1fa | vonage sucks |
| 20:41.46 | a1fa | broadvoice sucks |
| 20:41.46 | a1fa | everybody sucks |
| 20:41.47 | fu3 | yay!! |
| 20:41.47 | fu3 | the wiki is back! |
| 20:41.47 | a1fa | what wiki |
| 20:41.47 | fu3 | the voip-wiki |
| 20:41.48 | chr|s_ | http://en.wikipedia.org/wiki/Illegal_prime |
| 20:41.48 | fu3 | read the topic |
| 20:41.48 | fu3 | err voip-info |
| 20:41.48 | a1fa | oh |
| 20:41.48 | a1fa | it worked for me |
| 20:41.49 | a1fa | chr|s_ : omfg.. you solved the mistery of the universe |
| 20:41.49 | noky | in http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip "put the following in res_mysql.conf " |
| 20:41.49 | noky | i don't have this file... |
| 20:41.49 | a1fa | noky : make it |
| 20:42.03 | noky | nice |
| 20:42.03 | a1fa | noky : touch /etc/asterisk/res_mysql.conf |
| 20:42.12 | a1fa | noky : pending that you have mysql support compiled with * |
| 20:44.42 | noky | it's work! |
| 20:44.42 | noky | thanks a1fa :) |
| 20:44.42 | loko | a1fa vonage is allowing your own device now? |
| 20:44.42 | loko | a1fa why does boradvoice suck |
| 20:44.45 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
| 20:46.24 | Hmmhesays | wrong version of gcc for this kernel |
| 20:46.37 | lemmy | has app_conference some requirements regarding voice codecs? I get some warnings "no translator path from unknown to alaw". |
| 20:47.37 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 20:47.48 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org/ |
| 20:48.24 | willt | yes whats wrong with broadvoice? |
| 20:49.12 | redondos | How can I reduce the verbosity level? |
| 20:49.21 | a1fa | willt: for starters, the quality is crappy.. most of the time it doesnt work |
| 20:49.34 | a1fa | willt: it drops my calls, you cant reach support when you need them |
| 20:49.39 | loko | a1fa who do you use as a provider |
| 20:49.44 | a1fa | willt: it only supports ulaw |
| 20:49.48 | I-MOD | redondos: set verbose <1-10> |
| 20:49.49 | a1fa | loko : broadvoice |
| 20:49.54 | redondos | I-MOD: Thank you very much. |
| 20:50.18 | noky | can i configurate my asterisk to talk h323 with a GNU GK ? |
| 20:50.20 | willt | a1fa: where are you out of? |
| 20:50.38 | a1fa | CSA |
| 20:50.43 | a1fa | CST |
| 20:50.45 | a1fa | sorry |
| 20:50.53 | a1fa | South |
| 20:50.54 | loko | a1fa hmm then why do you use them if they suck? |
| 20:51.00 | a1fa | loko : they are cheap |
| 20:51.19 | a1fa | if they only let me over-ride CID on my own.. i'd love em |
| 20:51.24 | *** join/#asterisk NexGen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net) |
| 20:51.26 | willt | a1fa: I am in California and they sounded ok to me although I have tested them too much |
| 20:51.37 | *** join/#asterisk AlexCTI (n=alex@64.251.9.44) |
| 20:51.55 | willt | unlimited calling is pretty nice |
| 20:51.57 | redondos | Where can I read about the syntax for the extensions? I know what XX mean but how about _? |
| 20:53.01 | a1fa | willt: they sound "ok" |
| 20:53.01 | a1fa | they sound like ulaw.... |
| 20:53.14 | a1fa | i mean wtf.. but when you call some1 else.. over-seas.. i hear clicking sounds and shit like that |
| 20:53.20 | a1fa | i hear delay and etc |
| 20:53.28 | *** join/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 20:53.50 | willt | do the usa calls sound good though? |
| 20:53.54 | drewr | WereTiger: http://lnk.nu/voip-info.org/8l9.php |
| 20:54.09 | lemmy | or is app_conference just working with iax2 and no sip? %) |
| 20:54.09 | drewr | WereTiger: The "Multi-link PPP" section. |
| 20:54.36 | drewr | You can reallocate channels for voice and data, giving voice preference. |
| 20:54.37 | loko | a1fa if price didnt matter, but BYOD did, who would you recommend |
| 20:55.12 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 20:56.04 | willt | I need to convert some pcm sound files to gsm can anyone recomend a decent program to do so? |
| 20:57.18 | redondos | How would I record a digital receptionist message for different extensions without having to change the user/extension number on my softphone all the time? |
| 20:57.20 | jbalcomb | willt does sox do that? |
| 20:58.23 | willt | jbalcomb: hmm maybe.. I was going to use windoez but I guess I can do that :) |
| 20:58.28 | a1fa | willt: they are ok> |
| 20:58.35 | jbalcomb | redondos set up a dictate on *71{EXTEN} that uses the extention to name the faile? |
| 20:58.51 | a1fa | chr|s_ : what is this illegal prime bullshit? |
| 20:58.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:59.00 | AlexCTI | Some one can recomend me a very good firewall and friendly, that work with Asterisk? |
| 20:59.33 | redondos | jbalcomb: Sounds good... how would the line look like more or less? |
| 20:59.39 | jbalcomb | AlexCTI Watchgard SOHO, Smoothwall on a PC, or maybe just ipchains on the server itself? |
| 20:59.44 | a1fa | chr|s_ : a prime hash of the program, that turns out to be illegal? |
| 20:59.51 | a1fa | lol |
| 20:59.56 | jbalcomb | redondos im not sure |
| 21:00.16 | AlexCTI | Thnks, I'll take a look |
| 21:00.26 | jbalcomb | AlexCTI anyone firewall should work just fine with Asterisk though. Really no interoperability. |
| 21:00.52 | redondos | jbalcomb: all right... what should I read to be able to do that? |
| 21:01.04 | a1fa | http://www.cs.cmu.edu/~dst/DeCSS/Gallery/css_descramble.mp3 |
| 21:01.06 | a1fa | muhahaha |
| 21:01.15 | jbalcomb | redondos exten => *71,2,Record(/var/lib/asterisk/sounds/{$EXTEN}:gsm) maybe? |
| 21:01.16 | Dr-Linux | does asterisk support .vox format audio files? |
| 21:01.35 | NexGen | just checking I need to open ports 5060, and ports 10K-20K udp for sip, correct? |
| 21:01.36 | jbalcomb | redondos are you turkish? |
| 21:01.50 | russellb | Dr-Linux: nope |
| 21:02.49 | redondos | jbalcomb: Argentinian |
| 21:02.52 | Dr-Linux | russellb: actually i can change the .vox files to .gsm but after chaning the files voice quality is not good? :S |
| 21:03.13 | redondos | jbalcomb: thanks a lot, but how will that associate the recording with *other* extensions? |
| 21:03.16 | jbalcomb | NexGen: UDP5060SIP |
| 21:03.16 | jbalcomb | UDP5036IAX |
| 21:03.16 | jbalcomb | UDP4569IAX2 |
| 21:03.17 | jbalcomb | UDP20000 to 21000RTP |
| 21:03.27 | russellb | Dr-Linux: actually, i lied. we do support .vox :) |
| 21:03.29 | russellb | I forgot ... |
| 21:03.49 | Dr-Linux | russellb: why you lied? :S |
| 21:03.51 | Dr-Linux | its okey |
| 21:04.01 | russellb | because I forgot what vox was at first :-p |
| 21:04.33 | Dr-Linux | but what i'd need to place files in /var/lib/asterisk/sounds/.vox files ? |
| 21:04.34 | jbalcomb | redondos change the *71 so that you can dial *71XXXX or whatever and then dial *711234 if you want to record for the extension 1234 |
| 21:04.44 | jbalcomb | ~vox |
| 21:05.28 | jbot | hmm... vox is a power hungry fiend who likes to slay his users |
| 21:05.28 | jbalcomb | russellb isnt vox 'voice activation |
| 21:05.28 | redondos | jbalcomb: so instead of *71 I should use *71XXXX? |
| 21:05.29 | redondos | jbalcomb: please bare with me :) |
| 21:05.36 | *** join/#asterisk Cyphon (n=Cyphon@ip68-225-173-236.om.om.cox.net) |
| 21:05.38 | Dr-Linux | russellb: i know asterisk has a module for .vox files |
| 21:05.41 | Cyphon | My time is wrong on the caller id output from Asterisk |
| 21:05.42 | Cyphon | If I type date at Linux CLI it shows the correct time and CST, which is correct |
| 21:05.52 | NexGen | jbalcomb i have a sip trunk comming from a provider, so do i need to open the rtp ports? |
| 21:05.56 | Cyphon | What is wrong? |
| 21:06.01 | jbalcomb | redondos yeah, that is correct. i'm don't really know how to do anything in Asterisk I just know the theory |
| 21:06.02 | russellb | jbalcomb: no, vox is the vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file. Vox a very old file t |
| 21:06.09 | russellb | crap, i meant jbot |
| 21:06.12 | Dr-Linux | but what i will need, if i place files in sounds/dir and they work? |
| 21:06.19 | russellb | jbot: vox is the vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file. Vox a very old file type and |
| 21:06.26 | jbot | ...but vox is already something else... |
| 21:06.42 | russellb | i copied that from a web site :-p |
| 21:06.52 | jbalcomb | NexGen i think so but you could test it without them open to be sure |
| 21:07.12 | redondos | jbalcomb: How come you don't actually do things after reading? :) |
| 21:07.30 | Dr-Linux | russellb: you didn't answer me? |
| 21:07.42 | russellb | ok, now he knows what it is |
| 21:07.51 | russellb | Dr-Linux: you don't need anything extra |
| 21:07.56 | Dr-Linux | russellb: you might know, you work for digium, i think |
| 21:07.58 | russellb | Dr-Linux: just make sure you have format_vox.so loaded |
| 21:08.21 | Dr-Linux | russellb: you sure? |
| 21:08.24 | jbalcomb | redondos I am a knowledge hound but have no use for most of what i learn. between work/school/business/friends/family/ladies i don't have any time to try stuff at home. |
| 21:08.32 | russellb | Dr-Linux: yes |
| 21:08.40 | Dr-Linux | thanks |
| 21:08.49 | willt | What is the prefered format for recording prompts? gsm? |
| 21:08.55 | redondos | jbalcomb: Ok... but you help in an Asterisk channel. It's just strange. Hey, thanks for helping btw. |
| 21:08.55 | jbalcomb | willt yes |
| 21:09.12 | Dr-Linux | russellb: vox.so module comes with asterisk source or .. its in addones? :S |
| 21:09.17 | jbalcomb | redondos well, i do asterisk at work but i've only just started really |
| 21:09.21 | russellb | it's in the main asterisk tree |
| 21:09.32 | jbalcomb | ~vox |
| 21:09.44 | jbot | [vox] a format that most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to ... |
| 21:09.44 | redondos | k |
| 21:09.48 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 21:10.05 | russellb | heh |
| 21:10.18 | Dr-Linux | russellb: actually we are using .vox file for our Dialogic boards, and that's sounds is pretty neat and clean |
| 21:10.32 | russellb | cool |
| 21:12.18 | russellb | ~vox |
| 21:12.21 | jbot | well, vox is a format that most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. vox files are supported by Asterisk using the format_vox.so and codec_adpcm.so modules. |
| 21:12.23 | Hmmhesays | is there a global canreinvite=no? |
| 21:12.26 | russellb | therrrre we go. |
| 21:14.18 | Dr-Linux | russellb: codec_adpcm.so is also needed ? |
| 21:14.18 | russellb | yes, unless the other leg of the call is also using adpcm |
| 21:15.15 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-79.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:15.21 | Dr-Linux | russellb: if other is not using adpcm? then i'd not work for them? |
| 21:15.35 | Dr-Linux | russellb: we want to use .vox files for IVR |
| 21:15.40 | russellb | correct, unless you have codec_adpcm.so, then it will be able to translate it into other formats |
| 21:16.17 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 21:17.05 | Dr-Linux | great |
| 21:17.09 | *** join/#asterisk _octothorpe (n=octothor@unaffiliated/octothorpe) |
| 21:17.09 | Dr-Linux | russellb: but why always converted files to .gsm not work good? :S |
| 21:17.11 | rpm | is there a zaptel/wctdm driver for freebsd/openbsd or is that not going to happen? |
| 21:17.12 | Dr-Linux | do i use wrong converter? or what happend |
| 21:17.12 | jsharp | Yes, there is one. |
| 21:17.28 | Dr-Linux | btw, i use wavepad |
| 21:17.44 | russellb | Dr-Linux: i don't know. you probably do lose some quality making that conversion ... |
| 21:17.51 | redondos | Asterisk answers every incoming call on my FXO card. Is it possible to disable that temporarily, keeping Asterisk running so I continue configuring it? |
| 21:18.02 | jsharp | http://www.voip-info.org/wiki-FreeBSD+zaptel |
| 21:18.13 | Octothorpe | redondos, yes |
| 21:18.29 | [TK]D-Fender | redondos : remove "s" from your incoming context or point to another one that is a dead-end |
| 21:18.32 | Dr-Linux | russellb: thanks |
| 21:18.42 | russellb | you are welcome |
| 21:19.35 | Dr-Linux | some of my client using x-lite, and they are behind the NAT, they were working fine .. |
| 21:20.14 | Dr-Linux | but they network is DHCP now, so problem is that, after sometime their phone auto disconnect after some time:S |
| 21:22.05 | justinu | i'm playing a prompt, and using waitexten to collect a digit |
| 21:22.15 | justinu | i have an exten => 1 |
| 21:22.20 | justinu | it gets called when they press 1 |
| 21:22.27 | justinu | i also have an i, and t |
| 21:22.51 | justinu | my question is this: when the user presses anything other than one, asterisk jumps to i immediately |
| 21:23.03 | justinu | but they press 1, it takes a few seconds before it jumps to 1 |
| 21:23.07 | justinu | why? |
| 21:23.46 | jsharp | Cause its timing out waiting for the user to dial more digits. |
| 21:24.26 | Cyphon | My time is wrong on the caller id output from Asterisk |
| 21:24.28 | Cyphon | If I type date at Linux CLI it shows the correct time and CST, which is correct |
| 21:24.34 | justinu | jsharp: there's no valid extension other than 1 |
| 21:25.05 | justinu | jsharp: and if they dial 2, as the first digit, it goes to invalid /immediately/ |
| 21:25.36 | [TK]D-Fender | justinu : autofallthrough=yes = ICK. |
| 21:25.43 | justinu | ah |
| 21:25.44 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
| 21:26.00 | [TK]D-Fender | So Waitexten = nearly worthless command. |
| 21:26.04 | [hC] | Any of you guys tried any of the recent wifi phones that have hit the market? utstarcom or linksys, or anyone? |
| 21:26.26 | [av]bani | [hC]: they all suck |
| 21:26.46 | [hC] | Urg. Please dont say that! :P I have a client who is begging for a wifi phone or 5. |
| 21:26.56 | [hC] | Do any of them suck less than others? what are their biggest problems? |
| 21:27.01 | justinu | hmm... i turned autofallthru off, and it's the same |
| 21:27.02 | Cyphon | lol |
| 21:27.14 | [hC] | they're basically going to be used for nurses walking around, who use them essentially as pagers |
| 21:27.18 | [hC] | to read caller id, etc. |
| 21:28.13 | Cyphon | Can anyone help me with my simple time problem |
| 21:28.28 | [TK]D-Fender | justinu : pastebin the dialplan |
| 21:28.30 | justinu | i heard doctors are the worst clients |
| 21:28.34 | *** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211) |
| 21:29.16 | Cyphon | Is their a timezone setting in Asterisk, or why is my caller id two hours behind |
| 21:29.30 | [TK]D-Fender | justinu : No, salesmen are the worst clients ; Doctors are the worst PATIENTS ;) |
| 21:30.09 | justinu | [TK]D-Fender: http://pastebin.com/602406 |
| 21:30.21 | fu3 | IT WORKS!!!!!!!!!!!!! |
| 21:30.23 | fu3 | finally |
| 21:30.25 | fu3 | DID works!!!!! |
| 21:30.25 | the_magic_bean | anyone know if there is a way to change where soft buttons are on a Cisco 7940 or 7960? so one does not have to press the more button to get to transfer |
| 21:30.42 | fu3 | the telco gave me 7 digit pulsing and it's working fine! |
| 21:31.22 | austinnichols101 | the_magic_bean: not with cisco sip - I think that there's some sort of capability with the skinny stuff |
| 21:31.23 | [TK]D-Fender | justinu : make an "s" exten in there, set your timeouts and ensure auto-fallthrough is off. |
| 21:31.38 | justinu | which timeout do I need... digit timeout? |
| 21:31.42 | justinu | response timeout? |
| 21:32.01 | the_magic_bean | austinnichols101, ahh, ok thanks, yeah using SIP images here |
| 21:32.06 | [TK]D-Fender | justinu : I do : |
| 21:32.07 | [TK]D-Fender | Set(TIMEOUT(digit)=3) |
| 21:32.15 | [TK]D-Fender | Set(TIMEOUT(response)=15) |
| 21:32.20 | *** part/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 21:32.27 | *** join/#asterisk tfrevor (n=tfrevor@66.196.198.54) |
| 21:32.33 | justinu | should I not be using waitexten to collect the digit? |
| 21:32.37 | X-Rob_ | ~centosbug |
| 21:32.49 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 21:33.05 | tfrevor | Afternoon, folks. Do we have anybody here to help out with a DLink DVG-1120S connected to an asterisk box? |
| 21:33.11 | *** join/#asterisk firejon (n=firejon@206-169-48-226.gen.twtelecom.net) |
| 21:33.29 | riksta | anyone know where i can get a yum repo so i can install asterisk on FC4 x86_64 box? thanks |
| 21:34.21 | X-Rob_ | riksta, build it from source. |
| 21:34.30 | tfrevor | I have a DVG-1120S (converted from an 1120M). Connects just fine. Only, after initial connection, asterisk isn't recognizing any o the DTMF tones. Kinda kills transfers, dictation, etc... Any suggestions? |
| 21:34.55 | [TK]D-Fender | justinu : no need |
| 21:35.04 | [TK]D-Fender | just use standard "s" logic. |
| 21:35.15 | justinu | k, back to the drawing board |
| 21:35.19 | justinu | [TK]D-Fender: thx man |
| 21:35.22 | X-Rob_ | tfrevor, I'm guessing your device isn't doing DTMF detection properly. See if you can find something about DTMF 'inband' and set it to 'SIP-INFO' or 'RFC' |
| 21:35.33 | Katty | hi lads. |
| 21:35.44 | tfrevor | I do have it set for the RFC. Still no go. |
| 21:35.51 | firejon | does anyone know how to use hints in ael? |
| 21:36.22 | fu3 | cya lads |
| 21:36.22 | fu3 | im out for the night |
| 21:36.29 | fu3 | thanks for * |
| 21:36.50 | Hmmhesays | gmail is down |
| 21:36.50 | Hmmhesays | lovely |
| 21:36.58 | X-Rob_ | really? |
| 21:37.00 | X-Rob_ | fwor. |
| 21:37.08 | Katty | Hmmhesays: :< |
| 21:37.12 | justinu | gmail is down more and more often lately |
| 21:37.45 | chr|s_ | justinu, IT IS INDEED! |
| 21:37.48 | tfrevor | Specifically, the setting in the Telelphony configuration is "DTMF Relay" and I have it set for RFC2833 |
| 21:38.02 | Katty | justinu: they'r eprobably working bugs out or making it better |
| 21:38.07 | [hC] | Any of you guys at von? |
| 21:38.17 | chr|s_ | 'its free' and 'its beta' dont cut it |
| 21:38.22 | [hC] | I remembered to check the site today to see if i should go |
| 21:38.24 | [hC] | and it starts today |
| 21:38.24 | [hC] | :S |
| 21:38.32 | chr|s_ | They got me to use this app for a reason, to make ad money, and I accepted on the grounds it would work |
| 21:38.33 | redondos | [TK]D-Fender, Octothorpe: Sorry, I had to leave for a second. So you're saying that I should remove 's' from my incoming context? |
| 21:38.38 | chr|s_ | I got Opps Try Again |
| 21:38.42 | chr|s_ | 30 minutes before a presentation |
| 21:38.59 | chr|s_ | the longest 30 minutes trying to get this damn thing emailed so I could pick it up at the other place... ffs! |
| 21:39.58 | Octothorpe | the is what [TK]D-Fender suggested, he is usually right, I use a different approach, but they probably both work |
| 21:40.08 | redondos | What's your approach? |
| 21:40.16 | redondos | I'd like to learn every possible POV. |
| 21:40.21 | justinu | katty: i wish I could share your optimism :\ |
| 21:40.54 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 21:40.55 | *** join/#asterisk apardo (n=apardo@87.218.44.116) |
| 21:40.56 | Octothorpe | redondos: I actually direct it to a seperate context, here are the 2 lines that context includes |
| 21:43.02 | [TK]D-Fender | redondos : Just change the context in zapata.conf to one that doesn't exist. easiest way |
| 21:43.13 | [TK]D-Fender | other than renaming it in extensions.conf by 1 char :) |
| 21:43.22 | Octothorpe | actually 1 line |
| 21:43.25 | Octothorpe | exten => s,1,NoOp(Do Nothing) |
| 21:43.34 | justinu | [TK]D-Fender: got it behaving correctly now, thanks |
| 21:44.05 | Octothorpe | redondos: then nothing else in that context (pstn-no-answer) |
| 21:44.25 | [TK]D-Fender | justinu : glad to help |
| 21:44.30 | GerbilNut | so who is running on a Soekris system with a CF for the disk? |
| 21:44.40 | austinnichols101 | I want a gumstix |
| 21:45.34 | austinnichols101 | "is that an asterisk in your pocket"? |
| 21:46.34 | austinnichols101 | tks |
| 21:46.49 | a1fa | ok |
| 21:46.51 | a1fa | this is so weerd |
| 21:46.59 | a1fa | my remote sip client is dropping out |
| 21:47.03 | a1fa | she cant dial |
| 21:47.07 | a1fa | all she gets is a busy signal |
| 21:47.15 | a1fa | i see her login, 20s after, she is unreachable |
| 21:47.19 | jbalcomb | What is this about? pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 |
| 21:47.25 | a1fa | nat=yes;canreinvite=no;qualify=2000 |
| 21:47.34 | *** join/#asterisk convey (n=test@66.55.43.2) |
| 21:47.44 | convey | Can anyone recommend a good SBC? |
| 21:48.33 | austinnichols101 | a1fa: do a sip debug peer and watch what happens with the options messages being generated by the qualify |
| 21:48.40 | [TK]D-Fender | jbalcomb : Sounds like a frame synch issue. Possibly your card isn't clocking stable to the telco. |
| 21:49.02 | austinnichols101 | a1fa: see if they're retrying |
| 21:49.15 | justinu | i was impressed with jasomi |
| 21:49.18 | justinu | but that ain't cheap |
| 21:49.29 | jesster | we use Kagoor's .. seem fine |
| 21:49.43 | a1fa | yeah |
| 21:49.43 | *** part/#asterisk WereTiger (n=WereTige@CPE00062586f64e-CM0014e8271804.cpe.net.cable.rogers.com) |
| 21:49.43 | a1fa | they are retrying |
| 21:49.45 | jbalcomb | [TK]D-Fender hrmm. thats odd. i think thats the span to our Telrad handling the faxes. |
| 21:50.02 | austinnichols101 | a1fa: then port 5060 isn't port-mapped or something to the far end |
| 21:50.27 | a1fa | ok she is reachable |
| 21:50.44 | a1fa | i am seeing destryoing call constantly.. why is that |
| 21:50.46 | austinnichols101 | so then qualify never gets an response to the options and asterisk assumes that she's no longer reachable |
| 21:51.12 | a1fa | call 603: declined |
| 21:51.35 | austinnichols101 | a1fa: you can also take a look at the addresses being used within the options messages - are the pointing to the right remote ip? |
| 21:51.49 | convey | what does an entry level Jasomi/Kagoor cost approximately? |
| 21:51.54 | a1fa | yeah |
| 21:51.58 | a1fa | she is behind a nat |
| 21:51.59 | austinnichols101 | a1fa: could be that they're not being sent to the right destination (private address) |
| 21:52.10 | a1fa | router issue |
| 21:52.13 | austinnichols101 | right |
| 21:52.17 | a1fa | she may need to update the router |
| 21:52.29 | austinnichols101 | or make sure that 5060 is mapped in the router |
| 21:52.36 | a1fa | why am I constantly seeing destroying call and no body is calling anybody |
| 21:52.46 | a1fa | inbound 5060? |
| 21:52.49 | a1fa | why do you need inbound |
| 21:52.58 | redondos | [TK]D-Fender: I followed your advice because it looked like the simplest, until I actually learn what everything does. Ok so in zapata.conf I've got a [channels] section with "context=from-pstn", which I changed to 'from-pstn2' that doesn't exist. My goal is to prevent Asterisk to 'pick up the phone' when the line rings. Will * still answer even if there's no such context? |
| 21:52.59 | tfrevor | alfa: Is it your asterisk box behind the NAT or the SIPphone? |
| 21:53.06 | a1fa | the SIP phone is |
| 21:53.12 | a1fa | asterisk is directly on the net |
| 21:53.12 | austinnichols101 | because the SIP options message is sent from the asterisk server to the remote phone on 5060 by default |
| 21:53.39 | austinnichols101 | so if the remote phone is behind a firewall the options message will never reach the phone |
| 21:53.59 | a1fa | well, once the phone establishes a connection, everything will pass back |
| 21:54.03 | a1fa | stateful firewall |
| 21:54.07 | austinnichols101 | nope |
| 21:54.14 | tfrevor | Alfa: Austin's right with that. I have a SIP phone behind my firewall. Have to have both 5060 and the 10000-20000 range open. |
| 21:54.16 | a1fa | i am running the same setup at home |
| 21:54.19 | austinnichols101 | put it in the dmz and you'll see |
| 21:54.32 | a1fa | how come I dont have the same problem @ home? |
| 21:54.48 | a1fa | same setup |
| 21:54.48 | tfrevor | Your firewall may not be blocking those ports. |
| 21:54.49 | austinnichols101 | what phone @ home? |
| 21:55.00 | a1fa | Linksys |
| 21:55.05 | a1fa | SIPura |
| 21:55.07 | austinnichols101 | k - you running keepalive? |
| 21:55.17 | a1fa | yeah? |
| 21:55.44 | austinnichols101 | ok - so what will happen is that if you turn on qualify=yes or qualify=2000 for your home phone the options messages will bounce the same way you're seeing now |
| 21:55.57 | a1fa | nah |
| 21:55.59 | austinnichols101 | but the phone keeps sending traffic because of the keepalive so asterisk sees traffic |
| 21:56.03 | loko | a1fa have you looked at telesip at all? |
| 21:56.06 | a1fa | right |
| 21:56.09 | *** join/#asterisk forao (n=fjaosdfj@ool-4354d6b4.dyn.optonline.net) |
| 21:56.15 | a1fa | austinnichols101 : the same thing should happen for this phone |
| 21:56.31 | a1fa | Destroying call '0033998a56d956672b1edc8526bcc80c@IP |
| 21:56.32 | a1fa | lol |
| 21:56.39 | a1fa | why am I seeing this shit? |
| 21:56.55 | austinnichols101 | a1fa: may be a phantom call from earlier |
| 21:57.09 | a1fa | Retransmitting #4 (no NAT) to 84.114.177.167:42214: |
| 21:57.11 | a1fa | here is the problem |
| 21:57.13 | a1fa | no NAT |
| 21:57.18 | [hC] | I believe you see destroying call messages from qualify, do you not? |
| 21:57.20 | a1fa | ?? |
| 21:57.21 | austinnichols101 | a1fa: here's another cut at an explanation |
| 21:57.22 | *** join/#asterisk Lino` (n=Lino@i577BDFBF.versanet.de) |
| 21:57.42 | austinnichols101 | when you have keepalive, there's a SiP packet sent out from the remote site every so often |
| 21:58.06 | austinnichols101 | if you have iptables as the firewall then it will keep the connection open waiting for a response for 60 seconds by default |
| 21:58.14 | austinnichols101 | then it will close out the connection |
| 21:58.23 | a1fa | i have iptables @ the pbx level |
| 21:58.32 | a1fa | it has been opened for that host |
| 21:58.33 | austinnichols101 | shouldn't matter |
| 21:58.38 | a1fa | including UDP |
| 21:59.21 | *** join/#asterisk JayPhillips (n=bgates@66-100-35-20-static.dsl.oplink.net) |
| 21:59.24 | a1fa | 201/201 84.114.177.167 D N 42214 UNREACHABLE |
| 21:59.48 | austinnichols101 | if I DON'T have keepalive on the remote site, then the phone registers and I'm fine for about 60 seconds and then I go unreachable. The reason is that the qualify=yes SiP options packets going from the asterisk box never reach the remote phone and then the remote phone never responds |
| 21:59.58 | justinu | a1fa: don't worry about that destroying call message |
| 22:00.08 | justinu | it's a debug message that asterisk shouldn't bother you with |
| 22:00.15 | austinnichols101 | I spent about 5 freakin days driving everyone here crazy trying to understand that behavior |
| 22:00.17 | justinu | it doesn't mean anything important |
| 22:00.46 | a1fa | the problem is |
| 22:00.51 | a1fa | it thinks 201 is not behind nat |
| 22:00.57 | austinnichols101 | a1fa: do a port map on the remote fw and you'll see the retries stop |
| 22:01.02 | a1fa | Retransmitting #4 (no NAT) to 84.114.177.167:42214: |
| 22:01.03 | austinnichols101 | from the server perspective it is not |
| 22:01.30 | austinnichols101 | and things will 'magically' start working |
| 22:01.44 | a1fa | hm |
| 22:01.46 | a1fa | that makes me mad |
| 22:01.55 | a1fa | i didnt have to this here |
| 22:01.58 | *** part/#asterisk lemmy (n=lemmy@developer.g2gui.net) |
| 22:02.10 | a1fa | i wonder why is her phone crapping out like that |
| 22:02.14 | *** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca) |
| 22:02.15 | austinnichols101 | this is why I don't test with xten. xten is smart about how it tries to determine it's own addy, nat, etc. and is deceptive. it'll work when other phones won't |
| 22:02.46 | a1fa | yeah |
| 22:02.48 | a1fa | i love xten :p |
| 22:03.51 | austinnichols101 | what's also deceptive is that you can have a setup that appears to be working perfectly but then you check the sip debug peer and you'll see the qualify retries. Yeah - it works, but it's not set up right. |
| 22:03.54 | Seldon1975 | xten killed my father |
| 22:03.54 | a1fa | i see her trying to establish a call |
| 22:03.56 | a1fa | she sents an invite |
| 22:04.02 | austinnichols101 | and many people are comfortable with that |
| 22:04.31 | GerbilNut | can you playback multiple files with one Playback(filename,filename2,filename3)? |
| 22:04.34 | a1fa | Found user '201' |
| 22:04.56 | firejon | is is possible to use hints in AEL>? |
| 22:05.06 | austinnichols101 | a1fa: the best thing I've found so far on asterisk NAT is http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html and it's really WAY too confusing. |
| 22:05.34 | a1fa | Reliably Transmitting (NAT) to 84.114.177.167:42376: |
| 22:05.34 | a1fa | SIP/2.0 404 Not Found |
| 22:05.35 | austinnichols101 | I'm going to write something up because this topic keeps reappearing |
| 22:05.36 | a1fa | there |
| 22:05.42 | a1fa | lol |
| 22:05.45 | a1fa | 404 what is not found? |
| 22:05.59 | a1fa | oh shit |
| 22:06.01 | a1fa | my number :P |
| 22:06.02 | a1fa | lol |
| 22:06.15 | austinnichols101 | SIP showing it's http roots |
| 22:06.22 | a1fa | ok this has been fixed |
| 22:06.30 | a1fa | still the unreachable error is haunting me |
| 22:07.09 | austinnichols101 | try opening the sip and rtp ports, get things working and then close them down |
| 22:07.09 | a1fa | i am guessing the router is flaky |
| 22:07.25 | a1fa | i dont have the access to the remote router |
| 22:07.31 | a1fa | and no body there that can work it |
| 22:07.44 | a1fa | 201/201 84.114.177.167 D N 42214 UNREACHABLE |
| 22:07.46 | a1fa | ok |
| 22:07.52 | a1fa | its says NAT -> N |
| 22:07.58 | a1fa | does that mean NO, or NAT |
| 22:08.00 | austinnichols101 | don't rely on that - it's misleading |
| 22:08.04 | loko | Does anyone know of a reference of telephone zones online (no phonebook handy) |
| 22:08.38 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 22:08.40 | austinnichols101 | loko: damnit - you made me type google |
| 22:09.18 | loko | =) |
| 22:09.21 | loko | i can't find any thru google |
| 22:09.36 | loko | atleast of my city |
| 22:10.13 | [hC] | alfa: as long as nat=yes is specified in your sip peer, it will use nat |
| 22:10.26 | austinnichols101 | a1fa: good question - let me see if I can find anything. I didn't look at it like that before |
| 22:11.33 | a1fa | [hC] : it is |
| 22:11.48 | a1fa | but for some reason that peer is dropping out |
| 22:12.07 | *** join/#asterisk ghotiboy1 (n=ghotiboy@24-176-0-219.dhcp.klmz.mi.charter.com) |
| 22:12.36 | ghotiboy1 | hi...i have a problem with an iax2 trunk btw 2 servers |
| 22:12.40 | ghotiboy1 | i control both |
| 22:12.48 | a1fa | ok.. no voice |
| 22:12.55 | ghotiboy1 | one side registers fine and i can call |
| 22:13.13 | ghotiboy1 | the other side says unreachable and i cannot call |
| 22:13.23 | a1fa | omfg |
| 22:13.27 | a1fa | this is so annoying |
| 22:13.36 | [hC] | alfa: try setting maxexpirey and defaultexpirey to 60 |
| 22:13.37 | a1fa | my mom can call even though she is unreachable |
| 22:13.44 | [hC] | your nat timeout on your router may be doing it |
| 22:13.45 | a1fa | but no voice |
| 22:13.48 | a1fa | ok |
| 22:13.54 | a1fa | in sip.conf, right? |
| 22:13.59 | austinnichols101 | a1fa: fix the port on the router and it'll work just fine |
| 22:14.00 | [hC] | you will always be able to call out, but inbound will not work since asterisk deems you unreachable |
| 22:14.01 | [hC] | yes. |
| 22:14.36 | ghotiboy1 | ok...so, if i set my host in the dmz it still doesn't work |
| 22:14.40 | a1fa | ok |
| 22:15.10 | a1fa | Mar 14 22:15:06 NOTICE[3578]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 0 |
| 22:15.16 | a1fa | it went unreachable instantly |
| 22:15.53 | a1fa | with that value of 60 |
| 22:16.01 | a1fa | what does that prove? |
| 22:16.20 | [hC] | not sure. |
| 22:16.30 | *** join/#asterisk SparFux (n=player@e182026245.adsl.alicedsl.de) |
| 22:16.32 | [hC] | usually what i see with a router that is expiring your session out of its nat tables is that if you reboot the device |
| 22:16.38 | [hC] | it will work for 5 minutes or so |
| 22:16.40 | [hC] | then become unreachable |
| 22:16.52 | [hC] | changing that to 60 usually cures it, since alot of nat routers have a 60 second expiry time |
| 22:17.52 | a1fa | ok |
| 22:17.52 | austinnichols101 | a1fa: I think you're right. The 'NAT' column in sip show peers is normally blank and then shows 'N' if nat=yes |
| 22:17.54 | a1fa | i was able to reach her |
| 22:18.03 | a1fa | she is reachable |
| 22:18.05 | austinnichols101 | a1fa: kind of confusing |
| 22:18.06 | a1fa | but she cant hear me |
| 22:18.19 | a1fa | Mar 14 22:17:32 NOTICE[3620]: chan_sip.c:9697 handle_response_peerpoke: Peer '201' is now REACHABLE! (160ms / 2000ms) |
| 22:18.50 | a1fa | Mar 14 22:18:36 NOTICE[3620]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 16 0 |
| 22:19.14 | jarrod | why do I have tons of these 'Avoiding initial deadlock' errors |
| 22:19.31 | austinnichols101 | [hC] the options messages from qualify=yes are failing for him (going into retries) and never reaching the remote side |
| 22:19.39 | [hC] | Ah |
| 22:19.44 | [hC] | What nat router are you using, alfa? |
| 22:19.55 | [hC] | Let me take a stab at a dlink? |
| 22:19.55 | austinnichols101 | gotta open up that remote port :) |
| 22:20.00 | a1fa | netgear |
| 22:20.06 | a1fa | austinnichols101 : i will do that tomorrow |
| 22:20.09 | [hC] | wgt624? |
| 22:20.13 | [hC] | firmware upgrade it |
| 22:20.18 | a1fa | i will do that too |
| 22:20.24 | [hC] | it is absolutely crucial |
| 22:20.35 | [hC] | i had old firmware on one that wouldnt pass RTP at all |
| 22:20.41 | [hC] | as soon as i upgraded it, it worked flawlessly |
| 22:20.50 | *** join/#asterisk BrianUT (n=list@c-67-166-96-54.hsd1.ut.comcast.net) |
| 22:22.26 | a1fa | wgr614x2 |
| 22:22.51 | a1fa | v2 |
| 22:23.50 | *** part/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 22:24.03 | a1fa | i will see if my mom can pull a remote update |
| 22:25.02 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:25.21 | Beirdo | arp arp! |
| 22:25.22 | a1fa | ok |
| 22:25.25 | a1fa | i have an idea |
| 22:25.29 | Nivex | arp rarp! |
| 22:25.33 | a1fa | arp r0x |
| 22:25.38 | a1fa | arabs |
| 22:26.43 | *** join/#asterisk jhiver (n=jhiver@office.gossamer-threads.com) |
| 22:26.49 | jhiver | good afternoon everybody |
| 22:26.56 | Katty | hihi |
| 22:27.04 | a1fa | hi |
| 22:27.07 | a1fa | katty sup babe |
| 22:27.08 | a1fa | :p |
| 22:27.13 | *** join/#asterisk ToTo (n=ToTo@host146-88.pool8256.interbusiness.it) |
| 22:27.27 | Beirdo | watch it, a1fa... you may well get your ass kicked for that |
| 22:27.58 | Beirdo | ahoy there Katty :) |
| 22:28.32 | a1fa | i love katty man |
| 22:28.38 | a1fa | she like family |
| 22:28.43 | a1fa | i name my kids after her |
| 22:29.06 | tsume | mm, she is a hot one ;) *meow* |
| 22:29.15 | a1fa | same here :P |
| 22:29.17 | a1fa | makes two of us |
| 22:29.19 | a1fa | muhahaha |
| 22:29.48 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda3-24-51-50-156.lndnnh.adelphia.net) |
| 22:30.23 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 22:32.27 | ravenpi | Okay -- I'm bidding against a 3Com system, and he's got some 3102's with PoE, full-duplex speaker, and dual 10/100's. He's selling them for about $260 -- which, if I bid the Polycom 501's, eats up a good chunk of my margin. Suggestions for a less-expensive (but decent) phone that supports those features? |
| 22:32.33 | *** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 22:33.17 | ManxPower | ravenpi, no. |
| 22:33.25 | xmark | I'm trying to use a cordless phone on my FXS channell When I dial it just keeps dialing and dialing and dialing sound familiar to anyone? |
| 22:33.30 | ManxPower | find a better price on the polycoms |
| 22:33.53 | Katty | tsume: you probably don't have a clue what i look like |
| 22:36.00 | justinu | ravenpi: you can supposedly get 501s w/ PoE for about 210 |
| 22:36.26 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 22:36.55 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 22:36.57 | ravenpi | justinu: actually, PC Connection has 'em for $201 (plus S&H), but I was hoping for some add'l wiggle room. |
| 22:36.59 | generalhan | whats up guys |
| 22:37.01 | tsume | Katty: like Laura Croft, right? ;) |
| 22:37.40 | Katty | i do have a lara croft picture... |
| 22:37.58 | Katty | but i no longer have the uberlong hair. |
| 22:38.00 | tsume | *lara, oops ;) |
| 22:38.03 | tsume | Katty: awww |
| 22:39.21 | generalhan | guys im having some issues (well i think at least) ... can some one please take a look at these WARNING messages that im getting in the CLI ?? ::http://generalhan.pastebin.ca/45754:: |
| 22:39.37 | *** join/#asterisk Aurs (n=Aurs@a217-118-41-157.bluecom.no) |
| 22:39.49 | SparFux | general: I take a look! |
| 22:39.54 | Seldon1975 | you know it's time to find a new job when you think to yourself "I'm going to go to the kitchen and make a cup of tea, because that will use up some of my work hours for today" |
| 22:40.12 | SparFux | general: please post again the url without :: |
| 22:40.20 | generalhan | http://generalhan.pastebin.ca/45754 |
| 22:40.34 | Aurs | Seldon1975: surprised that you are even able to get _a_ job, if it takes hours to make a cup of tea ;) |
| 22:40.35 | SparFux | got it already... |
| 22:40.53 | Seldon1975 | Aurs: well, you have to grow the tea leaves,... |
| 22:41.04 | Aurs | Seldon1975: hehe, indeed |
| 22:41.29 | a1fa | ok |
| 22:41.32 | a1fa | i did remote upgrade |
| 22:41.32 | Beirdo | heh |
| 22:41.35 | a1fa | router remote upgrade |
| 22:41.38 | SparFux | gerneal: sorry, no idea what the problem is. seems like software bug? |
| 22:41.45 | Beirdo | well, time for me to make some boxes and fill em with books |
| 22:41.52 | a1fa | its working like a charm |
| 22:41.57 | Beirdo | moving-- |
| 22:42.02 | generalhan | hmm |
| 22:42.21 | AlexCTI | Hi everyone, I have a Asterisk working as VoIP gateway with my PBX, using PRIs Lines PBX PRI Trnks--> Zaptel Trnks --> VoIP, which with manually calls works fine, but if I put a dialer, it send the events to fast to the PBX and it cannot get the answer line and make the PBX with out control the call, so there is any way to make the the answer event, busy event reply with a little delay, in order to the PBX dialer recognize that? |
| 22:42.27 | generalhan | anyone ever seen this before ?? i came back from vacation and now my CLI scrolls about 100 of these messages ever 30 minutes :: channel.c:784 channel_find_locked: Avoided initial deadlock for '0xb7d12330', 10 retries! |
| 22:42.38 | SparFux | like asterisk detected a programming fault and prevented such "deadlock"? |
| 22:42.49 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 22:43.28 | a1fa | damn it |
| 22:43.33 | a1fa | she is unreachable again |
| 22:43.34 | a1fa | god dman it |
| 22:43.58 | ManxPower | generalhan, unless you indicate you have a PROBLEM, those messages are normal and should be ignored |
| 22:44.00 | eric_hill | Have you tried sniffing the traffic between the two devices with Ethereal? |
| 22:44.30 | *** join/#asterisk qai (n=sydsyd65@209-161-237-114.dsl.look.ca) |
| 22:44.34 | generalhan | ManxPower: thank you thats what i was looking for ... there are no ISSUES. but i havent changed my verbosity and ive never seen it before |
| 22:44.39 | ghotiboy1 | [hC]: I have a dlink...problem with those and asterisk? |
| 22:44.46 | generalhan | so i wanted to be sure that it wasnt causeing an issue |
| 22:45.00 | ManxPower | generalhan, is it a WARNNIG or a NOTICE? |
| 22:45.19 | generalhan | WARNING |
| 22:45.25 | ManxPower | Odd. |
| 22:45.30 | generalhan | i agree |
| 22:45.38 | generalhan | but everything seems to be owrking the same as it was when i left |
| 22:45.50 | qai | ghotiboy1 - I've used a dlink and it didn't cause any problems. wasn't all the fun to set up though |
| 22:46.29 | [hC] | ghotiboy1 not specifically i wouldnt say, ive just had nat problems off and on with anything but linksys |
| 22:46.32 | ghotiboy1 | qai - did you have to do something special? |
| 22:46.56 | ghotiboy1 | one end is linksys...no prob...other end is dlink...unreachable |
| 22:47.11 | a1fa | ghotiboy1 : update firmware |
| 22:47.12 | generalhan | ManxPower: if you have a sec im also getting a WARNING on one of my B Channels in the CLI that i have never seen before :: http://generalhan.pastebin.ca/45755 :: |
| 22:47.13 | ManxPower | older linksys routers would reboot when it saw some types of SIP packets. |
| 22:47.14 | a1fa | i have the same problem here |
| 22:47.15 | ghotiboy1 | i have the asterisk box in the dmz |
| 22:47.18 | qai | Do you consider opening firewall up special? - |
| 22:47.21 | ghotiboy1 | alfa: at the latest |
| 22:47.33 | a1fa | i have the same problem |
| 22:47.48 | qai | ghotiboy1 - I didn't try DMZ - that caused some headaches. I've kept codecs to ulaw |
| 22:47.55 | ManxPower | generalhan, I have never seen that error |
| 22:48.01 | AlexCTI | Someone knows is asterisk has a PRI Detector tool? I need syncronze zapel and my PBX. |
| 22:48.07 | ghotiboy1 | codec is ulaw |
| 22:48.14 | SparFux | Can I have TeamSpeak connects to Asterisk? |
| 22:48.27 | generalhan | hahahaha !!! i am the best !!! i leave for a week and come back with errors that even YOU GUYS havent seen before ??!! |
| 22:48.31 | ManxPower | SparFux, does TeamSpeak interoperate with other SIP devices? |
| 22:48.35 | generalhan | why do i suck !! |
| 22:48.49 | SparFux | Manx: not sure... |
| 22:48.51 | a1fa | ok |
| 22:48.57 | a1fa | that maxexpiry=60 fixed it now |
| 22:48.57 | a1fa | :P |
| 22:49.01 | SparFux | Manx: let me guess: if so, it can connect? |
| 22:49.02 | ManxPower | SparFux, then we cannot answer your question |
| 22:49.09 | BrianUT | that would be neat if it worked with teamspeak though. |
| 22:49.11 | [hC] | alfa: thought it might. |
| 22:49.12 | ghotiboy1 | alfa: you are using sip? |
| 22:49.16 | CoffeeIV | I want an IAX2 termination/origination provider, that at least mostly works with faxes -- I started to sign up at iax.cc, but they required paypal. Any other fax friendly options out there ? |
| 22:49.19 | ghotiboy1 | i am trying to use iax |
| 22:49.26 | the_magic_bean | anyone have experience with dialplan.xml for cisco phones? Specifically, our voice mail is accessed by dialing *97, however putting *97 in the dailplan of course is read as a wild card, anyone know if they have an escape char? |
| 22:49.59 | a1fa | yes |
| 22:50.08 | [hC] | alfa: typo it before, or didnt apply it earlier? |
| 22:50.16 | a1fa | it didnt apply |
| 22:50.20 | a1fa | i uncommented it this time |
| 22:50.23 | a1fa | and it works great |
| 22:50.29 | I-MOD | the_magic_bean: * isnt a wildcard in the asterisk dialplan |
| 22:50.41 | a1fa | <PROTECTED> |
| 22:50.55 | *** join/#asterisk ToTo (n=ToTo@host146-88.pool8256.interbusiness.it) |
| 22:51.01 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 22:51.02 | qai | If anyone here has a working connection with Project Gizmo I'd like to speak with them. Perhaps best to msg me. |
| 22:51.04 | ghotiboy1 | should i try something similar to alfa? |
| 22:51.10 | a1fa | yes, try it |
| 22:51.24 | ghotiboy1 | does that option work with iax? |
| 22:51.28 | a1fa | i guess |
| 22:51.38 | a1fa | Mar 14 22:51:12 NOTICE[3620]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 161 |
| 22:51.42 | a1fa | god damn it |
| 22:52.02 | eric_hill | the_magic_bean: http://pastebin.com/602589 |
| 22:52.24 | a1fa | god damn it |
| 22:52.25 | austinnichols101 | a1fa: I feel your pain |
| 22:52.29 | a1fa | i am going to set that value to 25 |
| 22:52.39 | a1fa | see if that helps |
| 22:52.47 | a1fa | Mar 14 22:52:31 NOTICE[3620]: chan_sip.c:9697 handle_response_peerpoke: Peer '201' is now REACHABLE! (160ms / 2000ms) |
| 22:52.48 | a1fa | :) |
| 22:52.49 | a1fa | hahaha |
| 22:52.49 | the_magic_bean | erick_hill, thanks much. |
| 22:52.55 | eric_hill | hth |
| 22:52.57 | a1fa | its going up&down like crazy biznatch |
| 22:53.29 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:53.35 | a1fa | awesome |
| 22:53.37 | a1fa | cross-atlantic |
| 22:53.38 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 22:53.51 | a1fa | Mar 14 22:53:35 NOTICE[3620]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 160 |
| 22:53.54 | Alric | The n priority has to be one of the single nicest features to ever be coded :) |
| 22:54.13 | qai | agree |
| 22:54.41 | ghotiboy1 | any special advice for getting iax2 to work through NAT? it works great on one side and not the other |
| 22:55.18 | SparFux | Manx: looked it up. No, I don't think asterisk can communicate with teamspeak. Teamspeak is proprietary as it seems and it doesn't use sip or any other good protocol. |
| 22:56.08 | CoffeeIV | ghotiboyl: isn't just a matter of forwarding a single UDP port ? |
| 22:56.29 | a1fa | Mar 14 22:55:43 NOTICE[3943]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 0 |
| 22:56.29 | a1fa | damn it dude |
| 22:56.29 | a1fa | this is herpes |
| 22:56.29 | a1fa | it wont go away |
| 22:56.36 | glm2k | SparFux: you're correct. |
| 22:56.48 | SparFux | glm2k: yes, unfortunately :-( |
| 22:56.50 | *** join/#asterisk taec_home (n=gash@86.16.58.66) |
| 22:57.22 | eric_hill | alfa: Have you tried sniffing the traffic with Ethereal? Is your asterisk box getting a RST from a router somewhere? |
| 22:57.23 | *** join/#asterisk qaica (n=sydsyd65@209-161-237-114.dsl.look.ca) |
| 22:57.24 | BrianUT | ventrilo is similar to teamspeak but I think you'll run into the same problem |
| 22:57.25 | ghotiboy1 | CoffeeIV: you would think that wouldn't you...forwarded 4569...unreachable...changed it to 100...unreachable...put system in dmz...unreachable |
| 22:58.02 | taec_home | Can anyone point me in the right direction on how to get an inbound DID to forward to an external number |
| 22:58.39 | ManxPower | taec, exten => 5551212,1,Dial(Zap/g1/5553333) |
| 22:58.39 | qaica | taec_home.. did you try setting the inbound number to an extension that is forwarded to the external number? |
| 22:58.43 | ManxPower | there you go |
| 22:59.18 | ManxPower | a1fa, good god man, what do you have your qualify= set to? |
| 22:59.28 | taec_home | qaica: how to get the extension to forward to another number? We don't want a phone at the other end of it, otherwise, I'd simply set the forward on the phone. I guess that's what I'm asking, how to forward an extension permanently to another number. |
| 22:59.47 | ManxPower | taec_home, you already have your answer |
| 23:00.35 | *** join/#asterisk hollymolly (n=sydsyd65@209-161-237-114.dsl.look.ca) |
| 23:01.00 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 23:01.51 | taec_home | OK, on setting up an extension it dials SIP/<EXTNO> where <EXTNO> is the SIP extension number. What should I be looking to replace that with for an external no? I know I'm bitterly close but I'm not familiar enough with asterisk syntax from here |
| 23:02.08 | ManxPower | Oh, SIP! |
| 23:02.10 | ManxPower | you didn't say SIP, did you? |
| 23:02.11 | taec_home | Our asterisk guru is out for the week :) |
| 23:02.36 | a1fa | hahaha |
| 23:02.39 | a1fa | damn dog |
| 23:02.45 | ManxPower | taec, paste a line that would dial out if you were not forwarding |
| 23:02.50 | a1fa | its working |
| 23:02.55 | a1fa | after i port forwarded 5060 |
| 23:03.21 | ManxPower | a1fa, 5060 is what SIP uses. |
| 23:03.29 | ManxPower | why were you forwrding the IAX2 port? |
| 23:03.51 | a1fa | i didnt |
| 23:04.01 | *** join/#asterisk areski (n=areski@76.Red-83-57-44.dynamicIP.rima-tde.net) |
| 23:04.10 | ManxPower | oh, sorry, that was someone else. |
| 23:04.12 | taec_home | ManxPower: I think that's what i'm looking for :) |
| 23:04.19 | SparFux | Is there a channel on irc to meet people actually wanting to do voip to other people on this channel? |
| 23:04.23 | ManxPower | Unless Asterisk is behind NAT you should not need to forward any ports |
| 23:04.31 | ManxPower | SparFux, not really. |
| 23:04.40 | SparFux | Manx: I'll have to start one! |
| 23:04.43 | ManxPower | I don't want to talk to any of you anyway. |
| 23:05.02 | SparFux | Would be fun! Why just chatting about stuff, let's all join a conference :^) |
| 23:05.15 | hollymolly | My apologies for asking this again - I've been disconnected three times so am not sure if anyone has replied.. Can someone pm me if they have project gizmo working with their asterisk system.. thank |
| 23:05.16 | Meaty | ;lj |
| 23:05.36 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:06.09 | a1fa | austinnichols101 : i port forwarded 5060 |
| 23:06.12 | a1fa | and it is working |
| 23:06.17 | austinnichols101 | a1fa: kewl |
| 23:06.20 | a1fa | but thats not a perfect solution |
| 23:06.26 | a1fa | people can call you |
| 23:06.29 | ManxPower | Unless Asterisk is behind NAT you should not need to forward any ports |
| 23:06.32 | a1fa | sip://ip:5060/ |
| 23:06.42 | hollymolly | a1fa - why not port forward 5004-5082 |
| 23:06.43 | a1fa | ManxPower : the router is all messed up |
| 23:06.51 | ManxPower | a1fa, That's pretty obvious |
| 23:06.53 | a1fa | hollymolly : why? |
| 23:06.53 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 23:07.04 | a1fa | 201/201 84.114.177.167 D N 5060 OK (158 ms) |
| 23:07.11 | a1fa | suprisingly.. cross-atlantic connection |
| 23:07.15 | a1fa | 158ms |
| 23:07.18 | a1fa | it sounds perfect |
| 23:07.21 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:07.27 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 23:07.28 | austinnichols101 | sorry a1fa |
| 23:07.33 | hollymolly | I dunno - just sounds like a good idea from all the stuff I read about ports to forward for SIP |
| 23:07.33 | a1fa | a bit of muffle.. but thats because of the bt101 headset |
| 23:07.46 | SparFux | alfa: where are you located? |
| 23:08.05 | a1fa | USA |
| 23:08.17 | austinnichols101 | a1fa: what really sucks is if you want two have two phones at that remote site |
| 23:08.24 | SparFux | alfa: germany here. Would be cross-atlantic too :-) |
| 23:08.34 | a1fa | nice |
| 23:08.39 | a1fa | my sister is in austria |
| 23:08.41 | austinnichols101 | a1fa: you end up having to set the second phone up on 5061, etc. |
| 23:08.42 | a1fa | so i sent her my phone |
| 23:08.48 | SparFux | alfa: austria is great! |
| 23:08.51 | a1fa | so we can talk |
| 23:08.55 | a1fa | she is only 150ms away now |
| 23:08.56 | a1fa | :P |
| 23:09.00 | SparFux | I've been to austria two weeks ago. |
| 23:09.10 | a1fa | i've been there last year |
| 23:09.12 | a1fa | i loved it |
| 23:09.16 | a1fa | vienna |
| 23:09.26 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 23:09.28 | SparFux | alfa: it's really nice. I have been to Ischgl. |
| 23:09.53 | a1fa | time to go home |
| 23:10.02 | a1fa | time to ride my little ninja back to its garrage |
| 23:10.19 | a1fa | bye guys |
| 23:10.23 | a1fa | thanks for the help |
| 23:11.35 | riksta | can anyone recommend a medium level IP phone (not too budgety like grandstream) |
| 23:11.45 | Alric | Polycom! |
| 23:11.46 | riksta | multiple lines |
| 23:12.20 | SparFux | Has anybody ever used capicommand(ect) successfully? |
| 23:13.49 | jarrod | anyone heard of a silini softswitch? |
| 23:13.56 | jarrod | i don't know if im spelling it properly |
| 23:14.19 | SparFux | Or to put it another way: How does an ISDN telephone send digits once a connection has been established? |
| 23:14.30 | SparFux | Does it use DTMF? |
| 23:15.48 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:16.38 | austinnichols101 | a1fa: keep the shiny side up! |
| 23:16.56 | austinnichols101 | riksta: check out the aastra 9133i or linksys 942 (my new fav) |
| 23:17.43 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 23:17.49 | diclophis | howdy all |
| 23:18.02 | diclophis | so why would i be getting hangups after ~3seconds of connection? |
| 23:18.23 | diclophis | Timed out looking for connect acknowledge -- is the error i see in pri debug |
| 23:18.44 | austinnichols101 | diclophis: nat involved? |
| 23:19.30 | diclophis | no.. these are ISDN pris |
| 23:19.42 | _Thor | austinnichols101: excuse me, what did a1fa meant by sip://ip:5060/ ??? |
| 23:20.01 | austinnichols101 | _Thor: I missed that part of the conversation |
| 23:20.10 | *** join/#asterisk doce (i=doce@66.238.251.135.ptr.us.xo.net) |
| 23:20.18 | austinnichols101 | stupid HIPS knocking down my IRC connection |
| 23:20.20 | _Thor | austinnichols101: he said he had made a port foward |
| 23:20.29 | doce | Any Free bsd guru's in here? |
| 23:20.45 | _Thor | austinnichols101: can't ask him because he is gone :( |
| 23:20.52 | kippi | how easy is it to get asterisk to connect to another asterisk box? using SIP? |
| 23:20.56 | austinnichols101 | yes - problem was that his remote phone was going unreachable after a short period of time |
| 23:21.32 | austinnichols101 | we did a sip debug peer on the server and saw that the qualify=yes optioins messages were retrying which indicated that port 5060 wasn't port-mapped on the remote firewall |
| 23:21.34 | _Thor | yes, that I got, it's because I've been pulling my hair with the exact same problem overseas |
| 23:21.43 | austinnichols101 | he opened up 5060 and things started working |
| 23:21.54 | austinnichols101 | _Thor: been there, done that for about a week straight |
| 23:22.29 | _Thor | austinnichols101: but still don't get where the sip://ip:5060/ comes from |
| 23:22.31 | diclophis | so if i have 4 spans, 2 of which are trunked together, and the other two are independant |
| 23:22.37 | diclophis | what would my config look like? |
| 23:23.15 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:23.18 | diclophis | so.. on the 3rd span would i have span=3,1,0,esf,b8zs; bchan=49-71; dchan=72 ? |
| 23:23.21 | austinnichols101 | damnit - this sucks! |
| 23:23.34 | _Thor | austinnichols101: but still don't get where the sip://ip:5060/ comes from |
| 23:23.37 | austinnichols101 | _thor: not sure - didn't see when he said that |
| 23:23.57 | austinnichols101 | he may have been reading the debug options messages |
| 23:24.52 | Zodiacal | anyone know if i can create a macro that parks a call and tells me the park # and binds the transfer. all in one key press during the call? |
| 23:25.12 | Zodiacal | where would i begin? whats the transfer procedure for asterisk? |
| 23:25.13 | _Thor | austinnichols101: ok, thanks |
| 23:25.23 | Zodiacal | transfer() :) |
| 23:25.35 | Zodiacal | i guess asterisk calls them applications? |
| 23:26.03 | diclophis | is this ok? Scheduling timer at 0 sample intervals |
| 23:26.40 | *** join/#asterisk TokyoJimu (n=jimmy@198.51.175.64) |
| 23:26.59 | AlexCTI | Someone can tell me how i stop the zaptel in order to make an update of version? |
| 23:27.52 | glm2k | AlexCTI: rmmod zaptel |
| 23:28.17 | TokyoJimu | MeetMe suddenly stopped working, sending all calls to a busy signal. Debug doesn't tell me much. Just "channel_find_locked: Avoiding initial deadlock" and then "Requested indication 5". |
| 23:28.17 | glm2k | AlexCTI: actually you should rmmod your card's module first |
| 23:28.32 | AlexCTI | Thnks, so after all off just the normal make clean && make && mae install? |
| 23:29.24 | AlexCTI | how is that? i'm using to up it modprobe wct4xxp |
| 23:30.14 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 23:33.27 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:34.06 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:36.46 | *** join/#asterisk mogorman (n=mogorman@206.166.206.34) |
| 23:36.51 | mogorman | anyone here at von |
| 23:37.43 | doce | Not I.. Im stuck at work.. ugh! |
| 23:40.02 | [hC] | i was asking earlier too |
| 23:40.08 | [hC] | I only today checked to see when it WAS. |
| 23:40.10 | [hC] | :| doh. |
| 23:42.47 | TokyoJimu | Last time I went to VON it was all over-hyped VoIP. |
| 23:43.00 | TokyoJimu | VoIP for VoIP's sake. |
| 23:43.15 | mogorman | all conferences are a little over hyped |
| 23:43.27 | diclophis | hey.. so has anyone done a resporg change before? |
| 23:43.35 | mogorman | ? |
| 23:44.11 | diclophis | like to port a 1800 number from ATT to MCI ? |
| 23:47.08 | rharfield | I was wondering if anyone was familiar with the Control Operator (COP) functions in the app_rpt module that could help me figure out why cop,6 won't function correctly. |
| 23:48.42 | [hC] | I was most shoked at entrance fees for VON |
| 23:48.46 | [hC] | almost 2k a person?? |
| 23:48.56 | RoyK | usd 2k? |
| 23:52.19 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 23:52.38 | kippi | how can I connect a SIP extension to another box? SIP/1001@10.69.69.20 ? should that work |
| 23:52.53 | [hC] | Yah |
| 23:54.10 | RoyK | kippi: better code the sip peer in sip.conf and dial SIP/peer/${EXTEN} |
| 23:58.22 | *** join/#asterisk cthompson (n=ct@cpe-65-189-12-4.cinci.res.rr.com) |
| 23:59.29 | kippi | RoyK: it seems to say that it is dialing but then I am getting this error http://pastebin.com/602687 |
