| 00:00.16 | Duckbizkit | he has it set up to calculate the amount of silence on the greeting and route the call based on the return....this is the voice patch, the other path is the answering machine path |
| 00:00.30 | Qwell | fa: no offense, but you need to figure some of this stuff out on your own |
| 00:01.18 | outtolunc | well he/you need to read that doc <G> |
| 00:01.20 | fa | Qwell i don't understand all.. my english is little |
| 00:02.03 | Qwell | fa: Start at google.com, work from there |
| 00:02.14 | outtolunc | duck: i have to head home, but if both of you read that doc, everything you need is in there |
| 00:02.23 | sjaak538 | does anybody know how I can test if I have an ISDN cable connected to my server dial/zp/g1 doesn't give response and dial in on the regular phone net doesn't give any response it's to far away from home to look |
| 00:02.24 | Duckbizkit | the GotoIf doc? |
| 00:02.49 | fa | Qwell i was there.. many times ;] |
| 00:02.50 | outtolunc | http://www.dynx.net/ASTERISK/DOCS/RTF/agentlogin3.rtf |
| 00:03.10 | *** join/#asterisk planetWayne (~wayne@cpc1-lich2-5-0-cust62.brhm.cable.ntl.com) |
| 00:03.15 | Duckbizkit | k |
| 00:03.47 | PTG123 | anyone in here know a good provider for newyork? |
| 00:04.34 | Delmar | well, buggered if I know. this echo thing just wont piss off. |
| 00:05.26 | Delmar | I have tried echotraining=800... I have tried messing with the txgain, I have even modified the READ_SIZE in chan_zap.c ..... |
| 00:06.31 | brc_ | woohoo! new powerbooks =) http://www.apple.com/powerbook/ |
| 00:08.41 | ManxPower | Delmar, You almost certinally have an impedence problem. |
| 00:09.49 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 00:09.50 | Duckbizkit | coffee and keyboards don't mix |
| 00:09.54 | Duckbizkit | bleh |
| 00:11.12 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 00:11.40 | kaitseb | I am calling pstn->iax2 to client that has lost internet access, how do I limit the time * asterisk is waiting for answer ? |
| 00:12.03 | ManxPower | qualify=yes in iax.conf |
| 00:12.51 | kaitseb | ManxPower: thanks, will check that |
| 00:13.15 | kaitseb | ManxPower: does it increas the load on the server or sth like that? |
| 00:13.45 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 00:14.22 | Primer | Anyone here use xten and have a problem with it where it complains about codecs? I can't recall the exact error, as it was reported by a user |
| 00:15.34 | Primer | in asterisk, it reports: chan_sip.c:2764 process_sdp: No compatible codecs! |
| 00:15.55 | file[laptop] | SDP doesn't match what you have defined in the entry |
| 00:16.10 | Delmar | ManxPower yeah i think i do have some kinda problem huh. I dunno what the hell I can do to fix it. :( |
| 00:16.15 | Primer | the user tells me he has to select a different codec, try again, then select the same codec as before, and try again...I'm presuming this is an xten issue |
| 00:16.37 | Delmar | I tried to measure the line, but I can't even seem to do that... I used to be able to do that sorta stuff. |
| 00:16.46 | Delmar | ah I have an idea. |
| 00:16.58 | Delmar | perhaps the DSL filter is messing the line up. |
| 00:16.59 | Mother_ | in xten you can enable/disable codecs and also set the precedence - I have it set to accept remote (i.e. *'s) setting |
| 00:17.15 | Mother_ | then you can just allow/disallow in sip.conf |
| 00:17.28 | Primer | Mother_: and xten (err x-lite, I guess) works fine for you, I presume? |
| 00:17.37 | Delmar | I might kill my net connection and strip the line back to nothing..unplug everything... and then measure the line and also see if the echo goes away. |
| 00:17.46 | Mother_ | Primer: yes |
| 00:17.55 | *** join/#asterisk rene- (~rene-@201.137.86.219) |
| 00:17.55 | Delmar | im getting seriously anoyed with this crap tho. grr. |
| 00:18.09 | Primer | I have this in sip.conf: disallow=all allow=ulaw...now that I think about it, it may be left over from when I was testing |
| 00:18.24 | Primer | should I perhaps allow all? |
| 00:18.31 | Delmar | you would think that the cards could measure and report the line conditions. |
| 00:18.41 | Mother_ | Primer: try to allow all if you need to debug other things, then restrict to the codec you want |
| 00:18.51 | Primer | ok, will do that...thanks |
| 00:19.07 | rene- | Hi |
| 00:19.23 | rene- | where does one gets chan_spy? |
| 00:19.51 | Delmar | probably gonna get disc. ttyl :P |
| 00:19.59 | Mother_ | Primer: in advanced settings -> Codec order, set Yes to Use remote preferred Codec....... |
| 00:20.09 | rene- | it used to be in Mantis |
| 00:21.11 | Primer | Mother_: excellent. Thanks |
| 00:21.19 | Mother_ | np |
| 00:21.40 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 00:21.58 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 00:22.03 | Primer | I really should get these people to use an IAX client |
| 00:22.06 | Primer | most are behind NAT |
| 00:22.12 | Primer | and I presumed that was their issue at first |
| 00:22.57 | kaitseb | ManxPower: I read client has to responde with ping to the poke packets when qualify is on, we have some clients not responding. Is there any other way? |
| 00:23.28 | sjaak538 | Nobody knows how to test ISDN cable connection |
| 00:24.13 | JunK-Y | sjaak538: whatcha mean? |
| 00:24.15 | Primer | tell me something. When I have an extenstion that looks like this: exten => s,1,Dial(SIP/${ARG1},30,tr), does asterisk act as an intermediary or does it attempt to tell the caller the IP address of the receiver and them let them try to negociate their own independent connection? |
| 00:24.28 | JunK-Y | just try to make a zap call on it???? |
| 00:25.24 | Mother_ | Primer: SIP gets the endpoints in touch, then RTP is direct between them |
| 00:26.06 | Primer | Mother_: then what happens when both endpoints are behind NAT and neither NAT is configured to redirect port traffic? |
| 00:26.17 | Mother_ | Primer: failure |
| 00:26.21 | Primer | lovely |
| 00:26.27 | Mother_ | :D |
| 00:26.35 | Mother_ | go tell the people that invented NAT hehe |
| 00:26.39 | sjaak538 | I have a server with asterisk and my provider told me that they have connected a ISDN cable to my server |
| 00:26.42 | *** join/#asterisk IPSo (~ipso@d207-81-249-35.bchsia.telus.net) |
| 00:27.08 | sjaak538 | with ISDN card but I can't get contact |
| 00:27.11 | IPSo | Anyone know why when I'm on the phone, and someone else dials my ext, it immediately cuts off my current call, and switches to the incoming call? |
| 00:27.16 | Mother_ | if you look at NAT in the wiki there's a good paper on the subject, nothing too easy to do sadly |
| 00:27.24 | Primer | well, theoretically, if client A, upon receiving the IP address and port of client B, initiated a connection to that IP and port, and client B did the same, it _should_ work |
| 00:27.29 | sjaak538 | Dial/zap/g1 no response |
| 00:27.49 | sjaak538 | Dial my number from the ISDN line |
| 00:27.53 | sjaak538 | Nothing |
| 00:27.59 | *** join/#asterisk JerJer (~JerJer@dsl-107-53.che.centurytel.net) |
| 00:28.03 | Mother_ | Primer: the problem is that RTP can take place on any arbitrary UDP port, unless limits can be imposed |
| 00:28.06 | Primer | asterisk to client A: talk to client B at IP X and port Y. Then asterisk to client B: talk to client B at IP Z, port Y |
| 00:28.23 | sjaak538 | How can i test if my provider did connect the ISDN line to my ISDN card ??? |
| 00:28.26 | Primer | Mother_: right, if asterisk (or the SIP protocol) for provide for referals |
| 00:28.29 | Primer | this could work |
| 00:28.37 | Mother_ | I am actually going to try this with a Cisco phone behind NAT against * behind a different nat |
| 00:28.41 | Primer | where asterisk brokers the conenction, rtp port and all |
| 00:29.05 | Mother_ | Primer: yes, that's what I will try to do, limit the UDP port range to say 10 ports, then map those on both NATs |
| 00:29.08 | sjaak538 | Junk-Y a made a zap call but nothing |
| 00:29.20 | Primer | Mother_: that's still a crappy hack, but it should work |
| 00:29.52 | Mother_ | yep |
| 00:29.52 | sjaak538 | ztcfg response okay |
| 00:30.49 | sjaak538 | asterisk startup okay, so everything looks okay |
| 00:31.28 | sjaak538 | lspci and cat /proc/interupts are okay |
| 00:31.45 | sjaak538 | all looks fine |
| 00:32.25 | fa | how can i run that from a php agi script? - AppendCDRUserField |
| 00:33.17 | JunK-Y | fa: see agi command: EXEC |
| 00:33.55 | planetWayne | hello all, has anyone got any recorded samples from 'Festival' at all? |
| 00:34.28 | Qwell | my festival acts all weird, heh |
| 00:34.32 | Nugget | your site was very helpful when I was first exploring asterisk. thanks. |
| 00:34.38 | fa | 1junkthanks |
| 00:34.43 | fa | JunK-Y thanks |
| 00:34.45 | planetWayne | hay no worries :) |
| 00:34.46 | Qwell | if I play a long string of text, its like the narrator is running out of breath. heh |
| 00:35.05 | xkev | qwell, that's festival |
| 00:35.13 | planetWayne | good to see that people are making use of it :) |
| 00:35.24 | xkev | use punctuation :) |
| 00:35.30 | Sedorox | gJan 31 17:34:26 WARNING[4992]: chan_iax2.c:7430 load_module: Unable to open IAX timing interface: Device not configured any clues where I have to set that up? |
| 00:35.31 | *** join/#asterisk TedC (~ted@gray.impulse.net) |
| 00:35.39 | Qwell | xkev: I did. (I think?) |
| 00:35.45 | Qwell | Is that "supposed to happen"? |
| 00:36.23 | Primer | damn, so I had setup my family to use xlite and asterisk to talk to relatives in Brazil, and they had a few problems...now one of them switched everyone else to skype |
| 00:36.32 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 00:36.35 | Primer | except I have a sipura |
| 00:36.39 | Primer | and a cordless phone |
| 00:36.41 | Primer | fags |
| 00:37.19 | Mother_ | hahaha |
| 00:37.31 | PTG123 | Qwell, hey you still looking for #s? |
| 00:37.41 | Mother_ | the only thing that really scares me about skype is all that UDP traffic passing by your box.... |
| 00:37.42 | Qwell | PTG123: might be. another time perhaps |
| 00:37.55 | PTG123 | Qwell, heh weren't you the one asking before? |
| 00:37.55 | Qwell | gotta head out for a bit...maybe a while longer, we'll see |
| 00:38.00 | xkev | need chan_skype |
| 00:38.05 | Qwell | PTG123: yeah, was. I'll be interested when I get back, heh |
| 00:38.09 | Primer | Mother_: what traffic? |
| 00:38.16 | Primer | I mean, other than the voice traffic? |
| 00:38.16 | PTG123 | Qwell, oh ok np message me found someone :) |
| 00:38.22 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543956pcs.mainf01.in.comcast.net) |
| 00:38.28 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 00:38.30 | JohnJacob | hey can anyone help me with ztdummy? |
| 00:38.30 | Qwell | ahh, I've got about 10 minutes. You free right now? |
| 00:38.30 | JohnJacob | I' |
| 00:38.34 | Mother_ | Primer: skype uses your PC as node on their network, i.e. you route other people's traffic |
| 00:38.40 | Primer | no |
| 00:38.41 | PTG123 | Qwell, sure |
| 00:38.46 | Primer | I can't believe that |
| 00:38.56 | Mother_ | and if you have a phat pipe, you can end up becoming a supernode and routing a LOT of traffic |
| 00:39.00 | JohnJacob | m running 2.6 and it seems to be fine... modprobe zaptel and ztdummy don't error out at all |
| 00:39.08 | Primer | heh, I have 2 ds3s here |
| 00:39.08 | JohnJacob | but the darn device files don't appear |
| 00:39.13 | Primer | soon to be 3 |
| 00:39.17 | JohnJacob | the /proc/zap is there |
| 00:39.20 | syslod | Sup ppl. |
| 00:39.25 | JohnJacob | and so it /proc/zap/1 |
| 00:40.02 | ManxPower | Isn't there a README.Liux26 or something like that in the Zaptel source? |
| 00:40.04 | JohnJacob | anyone know if I can create the device files for ztdummy myself? |
| 00:40.08 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-58-62.dsl.pltn13.pacbell.net) |
| 00:40.10 | JohnJacob | Ya, I read that |
| 00:40.36 | JohnJacob | I did what it said :) |
| 00:40.47 | netsurfer | JohnJacob - make the whole package.. its just as simple |
| 00:41.07 | bonbon-home | has anyone else seen a problem where sip calls originated from asterisk get cut off after around 30 seconds? |
| 00:41.14 | netsurfer | rene- I dont know, but id like to find it |
| 00:41.22 | ScythelX | could someone take a look at this and maybe tell me what could be wrong?? http://pastebin.ca/5007 |
| 00:42.13 | silik0n | its using a /ca website? |
| 00:42.14 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 00:42.23 | silik0n | err .ca website that is |
| 00:42.35 | Primer | hrmm chan_skype would be a cool idea |
| 00:42.43 | AgiNamu | $80 a sqft to have space in the Asterisk Pavilion!!! |
| 00:42.53 | AgiNamu | holy shit! ... that's a lot. |
| 00:43.00 | AgiNamu | Pulver must be making a KILLING off of VON |
| 00:43.01 | ScythelX | chan_sip.c:2551 parse: Too many SIP headers... |
| 00:43.08 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 00:43.09 | ScythelX | thats what it says about 50 times |
| 00:43.09 | MrEntropy | yo |
| 00:43.16 | AgiNamu | tu |
| 00:43.18 | ScythelX | chan_sip.c:2385 find_call: Call missing call ID from '24.181.176.62' |
| 00:43.24 | AgiNamu | el/ella |
| 00:43.50 | netsurfer | lmao pastebin.ca using postgresql |
| 00:43.53 | ManxPower | From CNET: " Misprinted 800 number in some versions of Intuit's TurboTax software sends customers to phone sex operation." |
| 00:44.06 | tzanger | ha |
| 00:44.08 | netsurfer | an 800 phonesex number? where ! ? |
| 00:44.17 | netsurfer | :oP |
| 00:44.21 | AgiNamu | netsurfer, then you "press to accept charges" and get billed $5000 :D |
| 00:44.28 | netsurfer | lmao |
| 00:44.29 | tzanger | you'll notice though that it's not postgres that fell down, it's the name server :-) |
| 00:44.41 | syslod | One of those dial the 1800 to hear the speal then you get hit with a charge. |
| 00:44.50 | Nugget | ManxPower: when I worked at a direct mail firm we screwed that up once. Back when 877 was brand-new as a toll-free number. |
| 00:45.05 | Nugget | We accidently published 800-somenumber instead of 877-somenumber |
| 00:45.10 | Nugget | (from habit, I guess) |
| 00:45.30 | Nugget | we ended up having to buy out the porn number for two months for some absurd amount of money |
| 00:45.37 | tzanger | hahaha |
| 00:45.44 | Mother_ | LOL |
| 00:45.44 | Nugget | and presumably the bank's heloc department got a few calls from pervs looking for pr0n |
| 00:45.52 | tzanger | Nugget: who got their ass beat for that one |
| 00:46.03 | Nugget | it was an expensive mistake for us. :) |
| 00:46.03 | ManxPower | "I thought it was going to take me to India, and then I got...this," he said. "I thought I dialed the wrong number. They don't usually talk to you like that at a tax operation." |
| 00:46.29 | netsurfer | lol |
| 00:46.31 | Mother_ | I had something like that, the phone company gave us a special number which they then assigned to some sex phone line, and we had bussiness cards etc. printed |
| 00:46.38 | netsurfer | yeah right, not "usually" lol |
| 00:46.39 | AgiNamu | "Welcome to 1800phonesex. In a moment, we'll transfer you to a 400lb transvestite that just happens to have a voice of a 14 year old. He'll pretend to be a foxy blonde named Bambi. We will charge you for this "service"" |
| 00:46.44 | Mother_ | it was hard to explain to costumers |
| 00:46.50 | tzanger | AgiNamu: <shudders> |
| 00:47.03 | outtolunc | ours is 'still' on some of the old ATT billing docs |
| 00:47.11 | AgiNamu | Mother_ a "special" number? |
| 00:47.12 | outtolunc | (after 5+ years) |
| 00:47.32 | xkev | att can't bill right to save their life |
| 00:47.56 | AgiNamu | And some poor secretary somewhere is wondering why guys keep on asking or "where do you want it"? when she says "Where should I direct your call?" |
| 00:48.20 | Nugget | hah |
| 00:48.33 | AgiNamu | I wrote a rating engine in a few hours in C#. I jsut had to cut out all the more difficult shit, like time-based billing and so on |
| 00:48.46 | AgiNamu | read the CSV from asterisk, do some DB lookups, save to SQL via bulk insert |
| 00:48.49 | Mother_ | AgiNamu: here we have different prefixes, 900 is toll-free for the caller, 902 is local rates but from any national number |
| 00:49.06 | AgiNamu | where's "here"? |
| 00:49.15 | Mother_ | AgiNamu: we took a 902 number, so that clients anywhere (Spain) could call us at local rates |
| 00:49.36 | AgiNamu | is Japan for 2 cents a good deal? |
| 00:50.36 | Nugget | AgiNamu: are you doing e164.org enum lookups? |
| 00:51.31 | AgiNamu | nugget, I just strip the international prefix, extract the country code, then search for the most specific match inside that country code |
| 00:52.35 | AgiNamu | seemed pretty straightforward. I'll admit, I have a very limited need (just um, billing customer calls ingoing and outgoing) |
| 00:52.39 | Mother_ | I'm out before I fall asleep on the keyboard |
| 00:52.49 | Mother_ | square marks on forehead don't look fashion |
| 00:52.51 | AgiNamu | but sure as hell not gonna pay $10,000 for it |
| 00:52.54 | AgiNamu | mother_ since when1? |
| 00:53.13 | Mother_ | since when what? |
| 00:53.23 | AgiNamu | since when are square marks not in fashion? |
| 00:53.24 | Nugget | *nod* |
| 00:53.27 | Mother_ | HAHAHA |
| 00:53.33 | Mother_ | OK then :) |
| 00:53.41 | Mother_ | take care all |
| 00:53.42 | AgiNamu | Nuggest, what's a e164 org enum lookup anywyas? |
| 00:53.53 | *** part/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 00:53.58 | outtolunc | you need a split keyboard with padding in the center <G> |
| 00:54.05 | AgiNamu | an MS Natural KB :) |
| 00:54.10 | AgiNamu | ... bit a bit larger. |
| 00:54.17 | ManxPower | The Geek-proof keyboard. |
| 00:54.20 | AgiNamu | I just keep a pillow under my desk and get on the floor. |
| 00:54.59 | Nugget | AgiNamu: it's a way for your asterisk server to know if there's a direct SIP or IAX route to any given phone number. |
| 00:55.16 | outtolunc | ok, raise your hands.. how many of you have a mp3 with 'typing' you loop <G> |
| 00:55.22 | AgiNamu | Oh I see.... Nope. I just terminate via TDM |
| 00:55.32 | Nugget | for instance, you can call me at +1 512 249-7218, but I'd rather you call me at sip:nugget@slacker.com -- e164.org is a registry of those lookups. |
| 00:55.41 | AgiNamu | outtolunc huh? |
| 00:55.45 | file[laptop] | slacker - indeed... |
| 00:55.55 | outtolunc | so you play it when you crawl under the desk |
| 00:55.57 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 00:56.00 | AgiNamu | lol |
| 00:56.08 | AgiNamu | Oh that. yea, that's a cool idea. |
| 00:56.21 | AgiNamu | I guess we could implement that and save a few bucks here and there. |
| 00:56.28 | Nugget | it works really well. |
| 00:56.34 | Nugget | there are over 100,000 numbers in the registry |
| 00:56.44 | Nugget | it's dns based. quite fast |
| 00:56.51 | AgiNamu | yea, that's one thing we'll do eventually probably. |
| 00:56.55 | AgiNamu | Right now, I'm getting our system done |
| 00:57.03 | AgiNamu | not even doing call forwarding, voicemail, nothing |
| 00:57.09 | Nugget | heh |
| 00:57.11 | Delmar | well, i stripped the line back to nothing and tested the X100 plugged directly in.. no DSl filters.. not even any internet at all... and still there is an echo. I even changed the polarity of the connection ...and still there is this festering bloody self-echo on the sip client side. |
| 00:57.13 | Nugget | chop chop! |
| 00:57.15 | AgiNamu | Just a simple sign up, get DIDs, provision, and bill |
| 00:57.19 | AgiNamu | Yep. I got 1 month |
| 00:57.25 | AgiNamu | I figure I'll be done in about 2-3 weeks |
| 00:57.25 | RaYmAn-Bx | Nugget: the thing that's lacking a bit is proper voip provider usage...(I.e. the providers adding their numbers) |
| 00:57.31 | AgiNamu | and have a week to test and play around |
| 00:57.36 | AgiNamu | before our official "alpha" launch |
| 00:57.55 | JunK-Y | someone knows how can i know on which span is a specific bchannel? |
| 00:58.02 | AgiNamu | RaYmAn-Bx -- I'd assume that'd have to be done for it to make much sense.... |
| 00:58.14 | AgiNamu | if it's only an opt-in thing... |
| 00:58.16 | RaYmAn-Bx | AgiNamu: afaik presently people add their own numbers |
| 00:58.18 | Nugget | there's a facility in e164.org for that, but I'm not familiar with how it works. |
| 00:58.22 | AgiNamu | Also, all our clients are behind nat |
| 00:58.29 | Nugget | I'm not a provider, so I ignored that part. :) |
| 00:58.42 | AgiNamu | so we'd have to open our server to dial our local extensions |
| 00:58.48 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 00:58.52 | AgiNamu | and I prefer to keep it as locked down as possible. |
| 00:59.16 | AgiNamu | anywyas... just making a list of my "3.2 cent residential flat plan" countries |
| 00:59.26 | AgiNamu | wondering if I should throw Japan in.... |
| 00:59.33 | Nugget | hai! |
| 00:59.38 | Nugget | nihon wa suteki desu ne |
| 01:00.16 | AgiNamu | simimassen. watashiwa nihongoga wakarimassen. |
| 01:00.35 | Nugget | pity :) |
| 01:00.37 | AgiNamu | yea |
| 01:00.43 | AgiNamu | I'm stdying korean... and then mandarin |
| 01:00.57 | AgiNamu | what did you write anyways? |
| 01:01.07 | Nugget | yes! japan is cool, I hope you agree. |
| 01:01.10 | AgiNamu | :) |
| 01:01.27 | AgiNamu | Yea.. I just hired a guy today... and he loves anime and has 1000+ CDs |
| 01:01.37 | Nugget | scary |
| 01:01.40 | AgiNamu | Gonna add "copying DVDs" to his job desc. |
| 01:01.55 | Nugget | I've never watched anime. doesn't look like my sort of thing. |
| 01:01.55 | AgiNamu | CDs -- i.e., DivX |
| 01:02.07 | AgiNamu | i used to be really into it... not as much now |
| 01:02.45 | Nugget | http://slacker.com/photos/tokyolaunch/IMG_2215 <-- that's the most japanimation I can stomach :) |
| 01:02.48 | AgiNamu | it plays off of common fantasies: Girls with robots. Girls with nice tits. Girls in robots with nice tits. Big weapons. Girls with big tits and weapons. etc.e tc. |
| 01:02.53 | *** join/#asterisk lcstyle (~Lc@adsl-9-68-57.mia.bellsouth.net) |
| 01:02.57 | rene- | netsurfer: http://bugs.digium.com/file_download.php?file_id=3776&type=bug |
| 01:03.07 | rene- | do u think this is it? |
| 01:03.10 | outtolunc | closest i came to anime was flippin a comic really fast after a few hits <G> (obviously many many years ago) |
| 01:03.18 | Nugget | I like the strongbad email about japanimation version of strongbad |
| 01:03.25 | tzanger | ugh |
| 01:03.26 | tzanger | I hate strongbad |
| 01:03.31 | tzanger | ST00PID |
| 01:03.33 | AgiNamu | i dont get strongbad. |
| 01:03.46 | Nugget | strongbad is way funnier than userfriendly, at least. |
| 01:03.49 | AgiNamu | my connex must be fux0red. |
| 01:03.55 | Nugget | but I agree, it's mostly just obtuse and obscure references |
| 01:03.56 | *** join/#asterisk cupcakes (~mike@81-86-133-210.dsl.pipex.com) |
| 01:04.13 | cupcakes | hi room |
| 01:04.16 | AgiNamu | userfriendly? isn't that the one that goes like "Hey, look at John." "Hey John, what are you doing?" "Getting real close to my monitor to see my FTP speeds" |
| 01:04.21 | lcstyle | hi all |
| 01:04.27 | AgiNamu | I like sexylosers. THATS funny. |
| 01:04.42 | Nugget | AgiNamu: yeah, you got it. |
| 01:04.53 | Nugget | and then the last panel is "linux. huh huh. cool. linux. huh huh" |
| 01:05.04 | cupcakes | quick question if anyone would be so kind: does Asterisk do anything useful running on a standard PC with one modem & soundcard? |
| 01:05.06 | AgiNamu | lol |
| 01:05.07 | lcstyle | does asterisk support any digital phones?.. |
| 01:06.37 | AgiNamu | OMFW, My ISP sucks LOLZ fuck! ... it even makes me type like an aol kiddie. |
| 01:07.08 | AgiNamu | lets see.... 8% packet loss.... DNS doesnt work... yep, everything is within normal operational paramteres. |
| 01:07.26 | file[laptop] | Dellllllllllll e-mail me back paleeeeeeeeeeeeeeeeez |
| 01:07.39 | *** join/#asterisk jero (~boo@modemcable040.12-81-70.mc.videotron.ca) |
| 01:08.07 | AgiNamu | Hmm... 50ms from me to my ISPs nameserver... I wonder if they are adding lag on purpose? |
| 01:08.14 | Nugget | lcstyle: not only does asterisk support digital phones, but it also has a website that says it supports digital phones right on the front page. |
| 01:09.21 | Nugget | file needs to buy a powerbook. |
| 01:09.22 | cupcakes | Nugget - would you mind answering my question too? |
| 01:09.29 | file[laptop] | I ordered Friday at 10AM, and here it is almost Tuesday and not a peep |
| 01:09.32 | file[laptop] | even though they charged me |
| 01:09.38 | Nugget | cupcakes: I suspect the answer is no. |
| 01:09.47 | Nugget | but I don't know, so I ignored you |
| 01:09.48 | outtolunc | file: what you gettin now? |
| 01:09.59 | file[laptop] | outtolunc: I bought a new computer. |
| 01:10.14 | file[laptop] | 64-bit goodness |
| 01:10.18 | AgiNamu | EMT? |
| 01:10.18 | Nugget | cupcakes: unless you have high speed internet or fxo hardware, asterisk will be of little interest to you |
| 01:10.21 | cupcakes | Nugget - ok, thanks. I suspect same but their site is over-techie (not often I think that!) |
| 01:10.23 | outtolunc | sweet |
| 01:10.35 | file[laptop] | yesssss |
| 01:10.37 | AgiNamu | Nugget... or unless you have a local PBX |
| 01:10.52 | file[laptop] | AgiNamu: yes. |
| 01:11.06 | Nugget | AgiNamu: I was talking specifically to cupcakes. I figured if "a single modem and a sound card" was the description, a local pbx was not likely in the picture. |
| 01:11.13 | AgiNamu | oh lol |
| 01:11.18 | cupcakes | well I have ADSL internet here in the UK. but no, no PBX |
| 01:11.24 | AgiNamu | file, which OS will you use? |
| 01:11.40 | AgiNamu | I ordered RHEL 3.1, 32-bit for my new Xeon... cause I didnt wanna be running into issues |
| 01:11.40 | Nugget | cupcakes: with adsl it becomes more interesting. |
| 01:11.43 | file[laptop] | AgiNamu: trying out 64-bit XP, and whatever Linux distributions I can find |
| 01:11.48 | jero | hurd |
| 01:11.59 | cupcakes | I'm interested because of bellster.net (with which maybe my basic setup *would* be useful...) |
| 01:11.59 | AgiNamu | 64-bit windows will be fun.... Especially when more apps run oin .NET |
| 01:12.06 | AgiNamu | and .NET 2.0 is out and has a 64-bit clr |
| 01:12.09 | file[laptop] | indeed |
| 01:12.17 | AgiNamu | bellster..arrg! |
| 01:12.24 | Nugget | all 64 bit does is make your cache half as effective. :) |
| 01:12.33 | cupcakes | heh did I say something bad AgiNamu |
| 01:12.47 | AgiNamu | nugget... well, only if you double ALL your 32bit ints |
| 01:12.54 | AgiNamu | in windows, for instance, only pointers are double sized |
| 01:13.14 | AgiNamu | How many registers are added in EMT64? |
| 01:13.20 | AgiNamu | I'm guessing not 128 like Itanium :) |
| 01:13.49 | Nugget | s/ium/ic/ |
| 01:13.59 | cupcakes | if you're concerned I'm just here to get freephonecalls then don't worry :) |
| 01:14.06 | AgiNamu | cupcakes, I think bellster is silly, yet good publicity for Pulver. |
| 01:14.14 | AgiNamu | I also think bellster is a way to get in trouble fast |
| 01:14.18 | AgiNamu | IF it takes off |
| 01:14.42 | AgiNamu | there's no such thing as free |
| 01:14.51 | cupcakes | I'm still reading up on it, so maybe this is a silly question: why would it get you in trouble? |
| 01:15.02 | AgiNamu | there may be CHEAP (like sub 1 cent to Singapore) |
| 01:15.03 | cupcakes | no of course not |
| 01:15.04 | AgiNamu | but not free |
| 01:15.27 | AgiNamu | cupcakes: You share your phone with strangers around the world. First, you are most likely violating your contract with your telco. |
| 01:15.29 | cupcakes | like I say - it's not the freeness or cheapness that intrigues me, it's the system and hardware/software |
| 01:15.33 | Nugget | AgiNamu: free is e164.org :) |
| 01:15.40 | AgiNamu | Second, some dude signs up and calls someone named bush |
| 01:15.48 | cupcakes | telco contract - agreed |
| 01:15.55 | AgiNamu | and tells him he's gonna stick a sharp object up his bell...err, whatever. something bad. |
| 01:16.17 | AgiNamu | SS tracks you at home, takes you in as a terror suspect (if you arent white) |
| 01:16.24 | cupcakes | yes gotcha, apologies I hadn't even started to think it all through ;) |
| 01:16.39 | outtolunc | and all your toys end up in pieces <G> |
| 01:16.59 | AgiNamu | nope. most people haven't because it's "cool". It's "peer to peer" telephony. fuck. "p2p" fucking overused term |
| 01:17.04 | cupcakes | perhaps they'll introduce a ebay-style feedback system. but without listening into peoples' calls... hmm no maybe not |
| 01:17.33 | AgiNamu | Note to people: The *entire fucking Internet* is peer to peer at one level or another! |
| 01:17.50 | AgiNamu | cupcakes, perhaps, some people will use it, then someone will get hurt. then less people will use it. |
| 01:17.50 | Nugget | AgiNamu: except for those poor suckers who use nat. |
| 01:18.07 | cupcakes | heh I can understand your reaction - guess you've had a lot of people asking about this here recently. |
| 01:18.07 | AgiNamu | Nugget: Almost everyone in Guatemala is behind NAT |
| 01:18.14 | AgiNamu | at least 1 level, sometimes 2 or more |
| 01:18.14 | Nugget | wow, that sucks. |
| 01:18.24 | AgiNamu | cupcakes, nope. |
| 01:18.46 | *** join/#asterisk Rick_Hunter (~rhunter@02-033.008.popsite.net) |
| 01:18.50 | AgiNamu | I just don't like publicity stunts for Pulver, esp. when he's charging something like $60 a square foot for floor space at VON |
| 01:18.51 | cupcakes | well maybe better get ready - bellster is just starting to get blogged |
| 01:19.00 | AgiNamu | yea, I might write up about it. |
| 01:19.26 | tzanger | I respect what Jeff Pulver's done, but I really don't like FWD nor Bellster, and the customer service I received on the WiSIP was horrible |
| 01:19.42 | cupcakes | thanks for the information; Asterisk interests me in any case so I hope to have a play with it when time permits |
| 01:19.49 | outtolunc | agi: i know i'm getting old but you said $80/per not too long ago, which is it? |
| 01:19.56 | AgiNamu | I got WiSIP/WiIAX/WiH323/WiMGCP for you. PA168-based FXS port + 900Mhz phone from local store. |
| 01:20.10 | AgiNamu | outtolunc, it's $80 at asterisk pavilion |
| 01:20.12 | Delmar | ok, can anyone tell me what line conditions in terms of voltage and impedence, that the X100 typically work best at....?? |
| 01:20.13 | AgiNamu | And $60 base cost |
| 01:20.15 | outtolunc | k |
| 01:20.22 | AgiNamu | in other words, damn expensive :) |
| 01:20.28 | outtolunc | nods |
| 01:20.28 | AgiNamu | we wanted a small table for brochures... forget it |
| 01:20.40 | outtolunc | sublet <G> |
| 01:21.13 | outtolunc | 6"x6" space on the digium table is what <G> |
| 01:21.32 | silik0n | $34.95 |
| 01:23.12 | outtolunc | Booth(s): 901/a/b/c/d/e |
| 01:23.43 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 01:24.44 | outtolunc | (which is 50x30) |
| 01:25.09 | ManxPower | Delmar, 600 Ohms |
| 01:26.04 | Delmar | ManxPower cheers. |
| 01:26.14 | Delmar | thats aparantly the target for here in NZ also. |
| 01:26.58 | Delmar | so, assuming that TelecomNZ have things about right...I'm going to focus on other things.... |
| 01:27.07 | Delmar | ManxPower ah something I just thought of..... |
| 01:27.19 | outtolunc | so who all is going to spring VON in san jose? |
| 01:27.52 | cupcakes | thanks folks, bye |
| 01:27.53 | Delmar | would it make a huge difference .. if the X100's did not have their own IRQ's exclusivly? I see they are both sharing by the looks if things... |
| 01:28.21 | Delmar | looks like one card is sharing IRQ with the vga, and another is sharing with a network controler. |
| 01:28.47 | Delmar | is that likely to cause any issues? |
| 01:29.53 | outtolunc | the x100p is *known* to have issues if sharing an IRQ |
| 01:29.57 | Delmar | ive never liked irq sharing .. ever.. so i might just fix that. |
| 01:30.11 | Delmar | ahh right... well its gonna get fixed anyway... echo or no. |
| 01:30.32 | outtolunc | you 'have' to fix that or echo will *remain* an issue |
| 01:30.57 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 01:31.20 | outtolunc | there are probably thousands of ML archive posts about it |
| 01:31.20 | Delmar | heh. right. |
| 01:31.29 | Delmar | maybe this is the fix... |
| 01:32.05 | *** join/#asterisk JSabines (JSabines@201.129.81.39) |
| 01:35.31 | AgiNamu | shit nugget, you look just like this guy i knew down here (he was from ohio) |
| 01:36.49 | Delmar | damn this motherboard. grrrr. Display Controller and X100 keep getting the same damn IRQ. |
| 01:37.40 | Luhiwu | Delmar, did you try changing slots? |
| 01:37.49 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 01:40.00 | Delmar | haha i knew that while I had my screw driver and head inside the case doing that... that i would turn around and see someone saying that :P oldest trick in the book to force IRQ change. |
| 01:40.58 | ta[i]nted | how do i make * dial an extension using Dial()? |
| 01:41.19 | ta[i]nted | let's say i want to dial extension 333 at number 123-4567 |
| 01:41.35 | tzanger | ta[i]nted: w |
| 01:42.38 | ta[i]nted | w |
| 01:42.39 | Delmar | so u want it to dial 123-4567 via a pstn/pots .. then dial someones extension (333) ? |
| 01:42.48 | ta[i]nted | Delmar yea |
| 01:42.52 | outtolunc | w=wait=pause |
| 01:42.58 | Delmar | I guess u could throw in some pauses |
| 01:43.08 | Delmar | then dial the 333 after a few secs |
| 01:43.35 | ta[i]nted | i just put w or p into the dial string? |
| 01:43.41 | outtolunc | w |
| 01:43.42 | Delmar | well.. irq issue all sorted... now for the echo test. |
| 01:44.17 | outtolunc | manx got that wiki marco handy <G> |
| 01:44.24 | outtolunc | er macro |
| 01:48.08 | ta[i]nted | so like Dial(IAX2/provider/1234567ww333) ?? |
| 01:48.16 | Delmar | wow. ok the echo is still there.. faintly.. but man that is a MASSIVE improvement. calls in both directions sounding really good. perhaps some fine tuning to polish off the remaining echo but.. fark me.... IRQ sharing sure doesnt work nice with X100's at all. |
| 01:48.40 | tzanger | IRQ sharing oesn't helpo much at all for anything involving latency |
| 01:48.43 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 01:49.16 | Delmar | yeah. im glad i thought to take a peek at things outside of asterisk itself. |
| 01:51.45 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 01:53.55 | tzanger | are bantam jacks just 1/4" mono jacks? |
| 01:54.19 | mrgoby | bantam |
| 01:54.33 | mrgoby | sounds like a starwars creature |
| 01:54.44 | tzanger | reminds me of a chicken |
| 01:54.54 | mrgoby | bantam desert people |
| 01:56.01 | mrgoby | cool... unfortunately i dont have my login info... can i get that emailed to me ? |
| 01:56.09 | mrgoby | whoops |
| 01:56.13 | mrgoby | nobody saw nuthin |
| 01:56.57 | Sedorox | weeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 01:57.08 | *** join/#asterisk santiago (~santiago@63.245.86.97) |
| 01:57.23 | mrgoby | jbot dance |
| 01:57.24 | jbot | ACTION becomes steve ballmer |
| 01:57.57 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 01:58.11 | mikegrb | :O |
| 01:58.45 | Sedorox | thats great |
| 01:59.07 | luisgrin | :) |
| 02:00.37 | sivana | !thwack mrgoby |
| 02:00.44 | sivana | ~thwack mrgoby |
| 02:00.46 | jbot | ACTION beats mrgoby on the leg with a AS/400 |
| 02:02.15 | *** join/#asterisk Dagrim (~junglesto@dagrim.user) |
| 02:02.27 | Dagrim | Well everybody.. I figured 'it' out FINALLY |
| 02:02.37 | sivana | does anyone know the difference between Cisco switches, standard or enterprise -- is it just the IOS? ie. Cisco 2924 comes in both standard/enterprise but are the same lookking |
| 02:03.56 | Delmar | ok thats strange.... all I have done is sort out those IRQ's.. and now... the busydetect/hangup detection has stopped working... |
| 02:04.41 | Dagrim | I had never been in my "basement" before.. seemed pretty skerry.. lol.. no lights.. and uhh yea .. Found my incoming cable lines.. and yea.. the make date printed on them was from like.. 1983 .. it was the WAY wrong guage.. way to small.. and it looked like someone had took scissors to it several times.. like sliced peices off .. lol |
| 02:06.03 | Sedorox | hmmmm |
| 02:06.43 | Dagrim | But I ran a new cable like 100 ft. or something.. lol.. and its FINE. I jus wanna kick myself in the a$$. |
| 02:07.03 | Delmar | ok this is ghey. i had the busydetection working mint before... grrr. |
| 02:07.47 | silik0n | whats up bitches? |
| 02:07.53 | silik0n | misfire |
| 02:07.54 | silik0n | hah |
| 02:07.59 | Delmar | lol |
| 02:08.03 | Delmar | :O |
| 02:08.11 | Delmar | ,,!,,(_),,!,, |
| 02:08.33 | ChulJin | tainted: I had to do something similar...Dial (IAX2/whatever/9878255www123) may not work |
| 02:08.37 | *** join/#asterisk iMediax (~user@00045a809589.click-network.com) |
| 02:09.02 | ChulJin | because the whole '9878255www123' choked my ITSP |
| 02:09.21 | Delmar | (sigh). this is tedious.. one thing starts working better.. another thing breaks. for fuk sakes. |
| 02:10.01 | Dagrim | yea |
| 02:10.05 | ChulJin | I did: Dial(IAX2/whatever/9878255,30,rD(www123)) |
| 02:10.17 | Sedorox | <PROTECTED> |
| 02:10.17 | Sedorox | Jan 31 19:10:02 NOTICE[6544]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
| 02:10.18 | Sedorox | ?? |
| 02:10.36 | Sedorox | that mean the times off on the machine? |
| 02:11.27 | ManxPower | Sedorox, It means your machine is not quite fast enough to schedule tasks as fast as it should. |
| 02:11.46 | Sedorox | its 380mhz apparently |
| 02:13.04 | Delmar | ManxPower, I had things detecting busy/hangup tone before.. and and now it doesnt.. apart from busydetect=yes and busycount=7 in my zapata.conf what other settings could effect signal/tone detection? it was working before I sorted out the IRQ sharing which really helped my echo.. now its broken :( |
| 02:15.35 | ChulJin | oh my |
| 02:15.41 | ChulJin | iaxtel seems to be back for some reason |
| 02:16.17 | Dagrim | where did it go? |
| 02:16.46 | ChulJin | I heard some time ago it had gone now...and I didn't press the point...but I just noticed it seemed to be back |
| 02:16.52 | ChulJin | oops, gone now=gone down |
| 02:17.45 | techie | hmrrm |
| 02:18.52 | MrEntropy | how can i allow a user, once connected through a zap card to be prompted for a number, a pin for example, i want the name of the command (i think this would be in extentions) that allows for a user to type that in? |
| 02:19.02 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 02:20.21 | KalD|Work | MrEntropy, do this: exten => s,1,Playback(vm-exten) exten => XXXX,1,Goto(exten-${EXTEN},1) |
| 02:20.26 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 02:20.57 | wasim | MrEntropy: show application Authenticate, you could also do this with an AGI |
| 02:21.08 | Delmar | uh damn it. that stupid echo is back again. this is bull shit. |
| 02:21.32 | Sedorox | Question...... if I have a extention/sip phone on server B and server A hots voicemail.. can I have it goto the voicemail system on A if it doesn't get picked up in X time...? |
| 02:21.55 | wasim | Sedorox: yep |
| 02:22.16 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 02:22.35 | Sedorox | the same way... or...? |
| 02:22.37 | KalD|Work | After the dial() on system B - put in dial(iax2/vm-exten-for-user@boxA) |
| 02:23.19 | KalD|Work | so when the first dial times out it calls the 2nd dial which goes to the vm mailbox on the other system |
| 02:23.20 | Sedorox | the vmail extention is the same as the extention.. so will that still work? |
| 02:23.56 | Sedorox | ok.. the extention is 2000... so I would do... Dial(iax2/2000@<server A>) ? |
| 02:24.01 | KalD|Work | then do dial(iax2/${EXTEN}@boxA) |
| 02:24.03 | KalD|Work | correct |
| 02:24.09 | wasim | as long as boxA knows 2000 exists in voicemail.conf |
| 02:24.19 | KalD|Work | then on server a setup exten for 2000 that just goes to vm |
| 02:25.00 | Sedorox | ok... because right now I have 2000 on the one box but wanna move it... so as 2 I have Voicemail(u2000)... |
| 02:25.53 | Sedorox | lets try this... |
| 02:26.11 | Darwin35 | we need g729 for fbsd |
| 02:26.17 | Darwin35 | with otu linux emu |
| 02:27.27 | greg_work | how do i get parameters passed to AGI? AGI(myscript|123) doesn't ever seem to get anything there |
| 02:28.09 | JunK-Y | greg_work: its | yes. |
| 02:28.15 | KalD|Work | greg_work they will be passed in as normal command line args... try AGI(script,1,2,3) |
| 02:28.28 | JunK-Y | im using | all days long. |
| 02:29.08 | greg_work | KalD|Work, ahh, ok tahnks |
| 02:29.16 | greg_work | i was looking for them in the AGI request headers |
| 02:29.26 | JunK-Y | ~agi api |
| 02:29.27 | jbot | it has been said that agi api is at http://home.cogeco.ca/~camstuff/agi.html |
| 02:29.28 | *** part/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 02:30.00 | JunK-Y | greg_work: seee Passing arguments to your AGI script on that url. |
| 02:30.28 | MrEntropy | KalD|Work: what does "exten-${EXTEN}" do? |
| 02:30.47 | *** join/#asterisk DaGrim85 (~junglesto@dagrim.user) |
| 02:31.35 | KalD|Work | MrEntropy, that would goto exten-2000 |
| 02:31.38 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 02:31.58 | MrEntropy | KalD|Work: why 2000? |
| 02:32.16 | KalD|Work | isnt that the ext you used for an example? |
| 02:32.24 | MrEntropy | ....no |
| 02:32.33 | MrEntropy | i never mentioned anything about 2000 |
| 02:32.41 | *** join/#asterisk scubasteve (~steve@rdu88-251-252.nc.rr.com) |
| 02:32.51 | scubasteve | Anyone want to see inside a Sipura SPA-841? http://www.miselconsulting.com/?page=841 |
| 02:32.54 | MrEntropy | Sedorox did though =) |
| 02:33.04 | JunK-Y | ~agi api |
| 02:33.05 | jbot | [agi api] at http://home.cogeco.ca/~camstuff/agi.html |
| 02:33.17 | Sedorox | lol |
| 02:35.55 | DaGrim85 | finally |
| 02:36.06 | DaGrim | guess I coulda pinged em.. bah |
| 02:36.30 | DaGrim | Soo yea.. this unixodbc has been compiling forever.. X) |
| 02:37.27 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 02:38.01 | MrEntropy | sorry, the X hung, again...i think it might be ram though |
| 02:38.23 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
| 02:38.29 | DaGrim | I said forget X some time ago.. lol |
| 02:38.38 | DaGrim | Its really neat.. but to much resources. |
| 02:38.54 | DaGrim | to run * on at the same time |
| 02:39.08 | MrEntropy | haha, i'm definately not doing that |
| 02:39.16 | MrEntropy | this is my desktop machine |
| 02:39.55 | DaGrim | yea.. I wish I had another box to run x on.. lol |
| 02:40.03 | DaGrim | but I use * for my stuff.. ya know. . |
| 02:40.31 | *** join/#asterisk Jeet (~manjitr@207.168.236.99) |
| 02:40.46 | Jeet | good evening everyone |
| 02:41.04 | MrEntropy | Jeet: morning =) |
| 02:41.10 | Sedorox | *sighs* |
| 02:42.03 | Jeet | ok i am thinking of dumping broadvoice. |
| 02:42.13 | DaGrim | really? |
| 02:42.20 | DaGrim | I was thinking about trying them.. |
| 02:42.23 | DaGrim | =\ |
| 02:42.26 | Jeet | any suggestions to a VoIP provider with a las vegas DID |
| 02:42.34 | Jeet | yeah too many problems. |
| 02:42.46 | DaGrim | Yea I need a local dids for Southern Illinois.. haha.. |
| 02:42.58 | DaGrim | like thats gonna happen anytime soon |
| 02:43.08 | Jeet | have to restart my asterisk server every 24 hours othewise BV just forwards incoming calls to VM |
| 02:43.21 | DaGrim | I bet the closest I could .. maybe get would be St. Louis? |
| 02:43.29 | DaGrim | Thats an hour and a half.. |
| 02:43.57 | Jeet | yeah the number i got from BV has starts with 991.. that's a lot confusing. |
| 02:44.10 | DaGrim | yea i bet that pisses 911 off |
| 02:44.10 | DaGrim | lol |
| 02:44.37 | Jeet | yeah there others too .. but i got a good number with this. |
| 02:44.39 | DaGrim | cuz you know.. trailing = * |
| 02:45.02 | Jeet | did you try any of the providers. |
| 02:45.06 | DaGrim | but i guess 1+ would prevent that.. but if someone wasnt all there.. yea.. heh |
| 02:45.09 | shido6 | we will have IL soon |
| 02:45.10 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 02:45.14 | DaGrim | NuFone is great for toll free's |
| 02:45.25 | Jeet | i have heard a lot of voicepulse .. but they don't have local DID |
| 02:45.29 | ChulJin | dagrim: (618)215-xxxx |
| 02:45.31 | ChulJin | ? |
| 02:45.47 | DaGrim | ChulJin: Yep |
| 02:45.51 | DaGrim | ChulJin: =) |
| 02:45.55 | ChulJin | livevoip.com |
| 02:46.02 | DaGrim | Hmmm.. THANK YOU! ;) |
| 02:46.10 | shido6 | keep watching our site, once we get the front end back up IL will be active shortly |
| 02:46.38 | DaGrim | sweeet.. Where are you located? |
| 02:46.49 | ChulJin | me? los angeles |
| 02:47.00 | shido6 | we're a Michigan based company |
| 02:48.11 | DaGrim | oh cool I used to live 5 minutes to the border, n. indiana.. it was south bend.. I drove all over michigan many times just for the view. |
| 02:48.19 | DaGrim | it was fall.. |
| 02:48.28 | DaGrim | and beautiful |
| 02:48.56 | ChulJin | DaGrim: 'Small World!' etc. I grew up in LaPorte. |
| 02:49.02 | DaGrim | Been there.. |
| 02:49.06 | DaGrim | Like 500x |
| 02:49.11 | DaGrim | Thats basically michigan city |
| 02:49.15 | DaGrim | right? |
| 02:49.30 | DaGrim | but La Porte is like before m. city.. thats the exit u gotta take.. |
| 02:49.34 | ChulJin | hehe right |
| 02:49.35 | DaGrim | To get there from the int. |
| 02:49.44 | Sedorox | Jan 31 19:48:59 WARNING[7067]: chan_iax2.c:5510 socket_read: Call rejected by 64.251.71.178: No authority found |
| 02:49.45 | Sedorox | ? |
| 02:50.10 | file[laptop] | I love it when people end up going to providers that we service... and then they end up paying my pay check... |
| 02:50.18 | file[laptop] | tickles me to no end |
| 02:51.01 | ChulJin | DaGrim: that's right, one exit before MC on either the 94 or the tollroad |
| 02:52.14 | blitzrage | blah |
| 02:52.22 | file[laptop] | hi blitzrage!!! |
| 02:52.26 | blitzrage | y0 file[laptop] |
| 02:52.27 | DaGrim | =) |
| 02:52.35 | blitzrage | I hate math |
| 02:52.44 | file[laptop] | awwwwwwww |
| 02:52.46 | blitzrage | and I hate even more at not being good at it |
| 02:52.54 | file[laptop] | I bet it hates you too |
| 02:52.55 | blitzrage | I'm good at everything! :) |
| 02:53.02 | DaGrim | Im not good at it because im not interested in it. |
| 02:53.15 | mrproper_ | anyone had any luck with fedora core3 and Fritz BRI card? |
| 02:53.16 | DaGrim | It makes me zzZzZzZZzzz |
| 02:53.18 | blitzrage | I'm interested... but it just takes me so long to process it for some reason |
| 02:53.30 | blitzrage | FC3 < shit |
| 02:53.33 | DaGrim | I mean I can do it.. dont get me wrong. I just hate it. |
| 02:53.38 | *** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
| 02:53.52 | DaGrim | haha.. havent tried it man. |
| 02:53.59 | *** part/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
| 02:54.04 | *** join/#asterisk El_Presidente (Martin@p508C99D1.dip0.t-ipconnect.de) |
| 02:54.08 | El_Presidente | hello |
| 02:54.13 | El_Presidente | anyone here? |
| 02:54.14 | blitzrage | its the President! |
| 02:54.18 | El_Presidente | yep :) |
| 02:54.18 | blitzrage | no, everyone left |
| 02:54.22 | El_Presidente | shit :) |
| 02:54.27 | blitzrage | I'm serious |
| 02:54.28 | blitzrage | its just me |
| 02:54.31 | El_Presidente | well |
| 02:54.35 | El_Presidente | oh |
| 02:55.02 | El_Presidente | maybe you can answer me a question about this (i dont find words for it) great program |
| 02:55.22 | El_Presidente | its about hardware compatibility |
| 02:55.41 | El_Presidente | i have an intel D/300-E1 card is it compatible? |
| 02:55.55 | blitzrage | I'm going to go ahead and say no :) |
| 02:56.12 | blitzrage | unless it has its own drivers which interface with Zaptel |
| 02:56.36 | El_Presidente | well on the site i find the intel D/300JTC-E1 |
| 02:56.39 | El_Presidente | as compatible |
| 02:56.46 | blitzrage | what site? |
| 02:56.48 | El_Presidente | its nearly the same card |
| 02:56.49 | blitzrage | and compatible with what? |
| 02:56.56 | El_Presidente | asterisk |
| 02:57.05 | blitzrage | well then try it |
| 02:57.17 | blitzrage | thats the only way you're going to know |
| 02:57.29 | El_Presidente | http://www.asterisk.org/index.php?menu=hardware |
| 02:57.30 | Sedorox | Jan 31 19:57:21 WARNING[7067]: chan_iax2.c:5510 socket_read: Call rejected by 64.251.71.178: No authority found |
| 02:57.33 | Sedorox | anyway to fix that? |
| 02:57.47 | El_Presidente | i will blitzrage |
| 02:58.01 | El_Presidente | is there an ebuild for gentoo? |
| 02:58.03 | Jeet | just sent a email to broadvoice to cancel me account.. let's see if they ever respond |
| 02:58.15 | El_Presidente | im not running my linux sys right now to check it |
| 02:58.24 | Sedorox | for asterisk there is El_Presidente |
| 02:58.27 | blitzrage | Sedorox: sounds username/secret is wrong |
| 02:58.30 | El_Presidente | ty Sedorox |
| 02:58.36 | blitzrage | El_Presidente: build from source |
| 02:58.56 | Sedorox | blitzrage: in iax.conf? |
| 02:59.01 | blitzrage | yah |
| 02:59.32 | El_Presidente | ok blitzrage thats what i had in mind |
| 02:59.32 | ChulJin | El_Pres: mentioned in the handbook, but not that particular page, is that dialogic cards are supported only with non-cheap, separate drivers. |
| 02:59.44 | Sedorox | I don't think so... everything should be right.... orrrrr..... when I do Dial(IAX/2000@smart-serv.net) should I have a username/pass in there somewhere? |
| 03:00.00 | El_Presidente | ChulJin, hm where is that handbook located? |
| 03:00.09 | ChulJin | http://www.digium.com/handbook-draft.pdf |
| 03:00.10 | El_Presidente | and how much are these drivers? |
| 03:00.13 | El_Presidente | ah |
| 03:00.23 | El_Presidente | ty |
| 03:01.02 | ChulJin | it doesn't say... |
| 03:01.29 | ManxPower | Contact Digium directly. |
| 03:01.50 | El_Presidente | i see |
| 03:02.01 | ChulJin | what ManxPower said. |
| 03:02.19 | El_Presidente | well maybe i should stay with ctade then, because i thought i would have a "free" alternative |
| 03:02.32 | El_Presidente | but i will ask for pricing @digium |
| 03:04.36 | *** join/#asterisk PyroSteve (~steve@ip68-227-149-247.no.no.cox.net) |
| 03:04.40 | PyroSteve | hey guys |
| 03:04.56 | PyroSteve | im installing a new asterisk sever soon |
| 03:05.27 | PyroSteve | im going to be registering with a few sip servers |
| 03:05.44 | PyroSteve | i need to do some useage balenceing between my sip accounts |
| 03:06.04 | PyroSteve | ive read thru the news ground archvies with very little luck |
| 03:06.25 | PyroSteve | im having trouble figuring out the logic behind what I need done |
| 03:06.46 | PyroSteve | i think i would need to use databases but im not sure |
| 03:06.48 | PyroSteve | any help ? |
| 03:06.58 | JunK-Y | some1 has PRI here to make some tests? |
| 03:07.03 | Nugget | hey, cool. we've got a PyroSteve to go with our ScubaSteve. |
| 03:07.10 | PyroSteve | heheh |
| 03:07.13 | PyroSteve | funny |
| 03:07.53 | PyroSteve | Im am Pyrotechnician |
| 03:08.08 | PyroSteve | I am ... sorry |
| 03:08.15 | DaGrim | lol |
| 03:08.47 | PyroSteve | any tip on how to balence my useage across a few sip <-> pstn gateways ? |
| 03:10.07 | PyroSteve | for example instead of having 4000 minutes used on one service provider, i would want 2000 minutes across two sip services |
| 03:11.49 | Jeet | what's the minimum for voipjet ? i heard they don't provide accounts to individuals ?? or am i mistaken ? |
| 03:12.36 | pulu | Jeet: the first deposit i did there was for $5 |
| 03:13.11 | Jeet | pulu: how is their service ?? support and the overall voice quality ? |
| 03:13.58 | mrproper_ | anyone had any luck with fedora core3 and Fritz BRI card? |
| 03:14.16 | pulu | Jeet: I haven't used it much but it was good enough at $5 that I put in another $120... They're who I'm going to use for sending calls for my customers, but I'm pretty small time... the voice quality sounds perfect to me.. |
| 03:14.50 | greg_work | so, i'm doing it. agi voicemail is about 15% functional compared to Comedian :p |
| 03:14.56 | Jeet | pulu: do they provide dids ? |
| 03:14.56 | pulu | they were down for about 20 min last week sometime, and i noticed at the same time they were down broadvoice was down as well, so i assume that's who broadvoice is using too... |
| 03:15.06 | pulu | Jeet: no, termination only |
| 03:15.17 | JunK-Y | greg_work: ya've fixed ur problem with passing args to ur agi? |
| 03:15.24 | bjohnson | Sedorox: do you mean IAX2? |
| 03:15.27 | greg_work | JunK-Y, yeah, back when we were talking :) |
| 03:15.38 | Jeet | pulu: are you using boradvoice ? |
| 03:15.38 | pulu | PyroSteve: i'm sure you could use a gotoif somehow to switch between them |
| 03:16.08 | greg_work | was looking in the agi headers, didnt even think to look at argv .. been a long, long while since i've done cgi programming |
| 03:16.11 | pulu | i have a $10/mo all in state account that i use to talk to my family in the us for free... but not really otherwise |
| 03:16.31 | Sedorox | bjohnson: yes |
| 03:16.51 | Jeet | pulu: well i am using broadvoice too.. but i have to restart my asterisk every day otherwise all incoming calls go straight to voicemail |
| 03:16.52 | bjohnson | PyroSteve: use the cdr and make something that checks usage before making a call |
| 03:17.24 | bjohnson | PyroSteve: odd to even out the calls between voip providers .. normally you would max out one and then use the other |
| 03:17.34 | Sedorox | bjohnson: I just get a unauthorized.... |
| 03:18.06 | pulu | Jeet: voicemail on your asterisk machine? Sometimes it seems to lose the registration but it's only happened once or twice (whwen someone tried to call and they got an unavail message) |
| 03:18.15 | bjohnson | Sedorox: try just dialing 2000 and put the username/secret in iax.conf |
| 03:18.36 | mtqh | kram: do you use an IDE when you program or just VI |
| 03:18.41 | Jeet | pulu: exactly incoming calls go directly to BV voicemail.. it has happened a lot to me.. (almost every day) |
| 03:18.42 | kram | nedit & make |
| 03:19.11 | Sedorox | bjohnson: I have the user/secret in in aix.conf |
| 03:19.25 | Jeet | pulu: i just sent them a cancellation request anyways |
| 03:19.33 | Chuji | hmm, can anyone get to www.ecost.com? |
| 03:19.41 | pulu | Jeet: people talk about some kind of patch that you need if you're natted but the machine that use with it isn't and it's supposed to be included in the cvs anyway... |
| 03:19.48 | bjohnson | Sedorox: type=peer? |
| 03:20.07 | MrEntropy | Chuji: yes |
| 03:20.11 | Jeet | pulu: my machine has a public IP |
| 03:20.15 | Sedorox | type=friend |
| 03:20.27 | bjohnson | Sedorox: qualify=yes? |
| 03:20.28 | Chuji | MrEntropy : Hmm, I get a 500 internal server error |
| 03:20.47 | Sedorox | [ss-server] |
| 03:20.47 | Sedorox | <PROTECTED> |
| 03:20.47 | Sedorox | ;auth=md5 |
| 03:20.47 | Sedorox | <PROTECTED> |
| 03:20.47 | Sedorox | <PROTECTED> |
| 03:20.47 | Sedorox | <PROTECTED> |
| 03:20.49 | Sedorox | ; defaultip=64.251.71.178 |
| 03:20.51 | MrEntropy | Chuji: maybe your isp has a transparent proxy |
| 03:20.51 | Sedorox | <PROTECTED> |
| 03:20.54 | Sedorox | <PROTECTED> |
| 03:20.55 | Sedorox | <PROTECTED> |
| 03:20.57 | Chuji | ~pastebin |
| 03:20.58 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:20.58 | Sedorox | ; allow=all |
| 03:20.59 | Sedorox | ops...sorry |
| 03:21.05 | Sedorox | yea... didn't mean to here... |
| 03:21.15 | pulu | I talk to my wife about an hour a day at least, and my dad talks to my mom probably 2 hrs /wk so that's like $8-10/wk if I was using voipjet and I wouldn't have a local number for my wife to call, I haven't found any other providers that match up to that |
| 03:21.22 | bjohnson | put the host and the username in iax.conf |
| 03:21.30 | ManxPower | Shall we string 'im up or let it slide this time? |
| 03:21.36 | pulu | but yeah, it's got issues that i wouldn't want to use it for my customers |
| 03:21.43 | bjohnson | Sedorox: you're connectiong to a voip provider? |
| 03:21.51 | Sedorox | no.. two asterisk boxes.. |
| 03:21.54 | pulu | the sound quality isn't near as good as voipjet fo rme |
| 03:21.59 | Chuji | pulu : just get locla did's in there lata |
| 03:22.03 | Juggie | i saw someone pastebin this, "GotoIf("IAX2/teliax/4", "1?voice|1:answering|1") " earlier today, anyone have any docs on this voice detection? |
| 03:22.11 | pulu | Chuji: from who? |
| 03:22.16 | Chuji | pulu : You can add bv did's for like $3 per month |
| 03:22.22 | bjohnson | Sedorox: one side should have a host= and the other can be dynamic |
| 03:22.33 | pulu | Chuji: oh, that's what im doing, i thought you meant from someone else |
| 03:22.34 | bjohnson | both sides should have username and secret in the iax.conf entry |
| 03:22.53 | Chuji | pulu : Ohh, only caught the tail end, sorry |
| 03:22.58 | Delmar | ok guys.. this echo thing.. i just had 2 calls come in.. and both times the echo started out really loud.. then the self-echo eventually dropped right back.. so the echo cancellation is there.. just not working too shit hot. |
| 03:23.01 | Nugget | iax peering never made sense to me until I stopped using type=friend |
| 03:23.13 | Sedorox | Server A and B... my phone is connected to A.. there is another on B.. but we want voicemail only on A... so I was told to have the second option as Dial(IAX2/2000@severA).. where 2000 is the extention and voicemail extention |
| 03:23.22 | Nugget | save yourself grief -- do it right from the beginning. |
| 03:23.24 | DaGrim | Where does * defautly (on slack 9.0) store the modules? (rather the files i have to copy from the MyODBC-3.51.10-pc-linux-i686 package??) |
| 03:23.41 | DaGrim | its like.. .so and .la files? |
| 03:23.43 | Sedorox | Nugget: I followed a guide from voip info site.... |
| 03:23.50 | DaGrim | theres no makefile, config, anything but those |
| 03:23.56 | rene- | Hi |
| 03:24.11 | Juggie | nugget/bj/manx either of you know how "1?voice|1:answering|1" operates? its obviously voice detection but whats the syntax |
| 03:24.22 | Nugget | no clue, sorry. |
| 03:24.24 | bjohnson | Juggie: I doubt that does anything that you think it does .. voice is likely a variable |
| 03:24.39 | rene- | has anyone found anything on the whereabouts of chan_spy? |
| 03:24.43 | bjohnson | or a context |
| 03:25.01 | Juggie | i thought it may be voice detection vs answering machine? |
| 03:25.06 | Juggie | which would be useful |
| 03:25.17 | bjohnson | voice detection for what? |
| 03:25.26 | Juggie | voice or answering machine |
| 03:25.32 | Juggie | to let you know if u have a live caller or not |
| 03:25.36 | Sedorox | 78: No authority found .... yay.... |
| 03:25.41 | bjohnson | err .. you getting many calls where there isn't a voice? |
| 03:25.51 | Juggie | it would be for when you use asterisk to place calls |
| 03:25.54 | Juggie | not receive them |
| 03:26.25 | bjohnson | exactly how would you tell if it's a voice or a recorded voice? |
| 03:26.36 | Juggie | i have no idea |
| 03:26.38 | Juggie | but its possible |
| 03:26.43 | Juggie | there are a number of factors |
| 03:26.52 | Juggie | big one being a normal voice would just say hello |
| 03:26.56 | Juggie | while a recording would go on and on |
| 03:27.07 | bjohnson | Sedorox: you get auth errors .. look at your auth settings |
| 03:27.10 | Juggie | dialogic does voice/answering machine detection etc. |
| 03:27.12 | Delmar | and two, a normal voice would say things in response to what you say.... |
| 03:27.20 | bjohnson | Sedorox: as I said, you need a username= on both sides |
| 03:27.23 | Delmar | and three, a normal voice wouldnt repeat itself exactly the same way a second time. |
| 03:27.24 | Delmar | :P |
| 03:27.54 | bjohnson | Sedorox: one side can be dynamic but the other side needs a sattic hostname setting. The dynamic side should register to the static side |
| 03:28.46 | *** join/#asterisk ZX81 (matt@222-152-158-141.jetstream.xtra.co.nz) |
| 03:29.11 | Sedorox | ok.. so... say Server A is set dymanic.. I don't need it to send a register request.. currect? |
| 03:29.47 | bjohnson | server has a dynamic IP or the iax.conf on server a is set to dynamic? |
| 03:29.47 | ZX81 | anyon manage to solve problems with no audio on Yellow Dog Linux 4.0? |
| 03:30.33 | bjohnson | Sedorox: a register command tells another machine what IP address you have. |
| 03:31.04 | bjohnson | Sedorox: since you have the one you flooded showing host=dynamix, I assume the other one has a register command |
| 03:31.22 | Sedorox | well both had dymanic set.. and both were sending register |
| 03:31.48 | *** join/#asterisk bratner (~bman@bzq-179-152-71.pop.bezeqint.net) |
| 03:32.07 | ZX81 | . |
| 03:32.32 | Juggie | when a client registars with asterisk, is the password encrypted when sent? |
| 03:32.39 | bjohnson | Sedorox: I don't think that works well |
| 03:32.50 | Sedorox | well... |
| 03:33.08 | bjohnson | Sedorox: I think one side needs a fqdn |
| 03:33.13 | Sedorox | I just changed it.. the one side has host=and IP.. and the server doesn't send register... |
| 03:33.26 | Sedorox | I'm just doing IPs.. does it need to be fqdn? |
| 03:33.34 | bjohnson | are they static IPs? |
| 03:34.24 | ZX81 | :) |
| 03:34.25 | ZX81 | ok |
| 03:34.38 | ZX81 | who s Kristion Kielhofner |
| 03:34.40 | ZX81 | :) |
| 03:36.41 | ZX81 | ~ping |
| 03:36.42 | jbot | pong |
| 03:38.06 | ZX81 | jbot: a bit slow today arent you? |
| 03:38.43 | ZX81 | so |
| 03:38.54 | ZX81 | nobody running on YDL on Mac? |
| 03:40.00 | DaGrim | ZX81: Where would I copy the .so.1 files for MyODBC for asterisk to? |
| 03:40.07 | ZX81 | dunno |
| 03:40.08 | ZX81 | :) |
| 03:40.11 | DaGrim | I know the .so ones go in /usr/lib/asterisk/modules |
| 03:40.13 | ZX81 | bkw'd know |
| 03:40.13 | DaGrim | ugh |
| 03:40.14 | ZX81 | :) |
| 03:40.16 | ZX81 | yeah |
| 03:40.18 | DaGrim | is he here? |
| 03:40.22 | ZX81 | /usr/lib |
| 03:40.23 | ZX81 | or |
| 03:40.27 | ZX81 | /usr/local/lib |
| 03:40.28 | ZX81 | maybe |
| 03:40.30 | ZX81 | dunno |
| 03:40.34 | ZX81 | ~seen bkw_ |
| 03:40.37 | jbot | bkw_ <~brian@65.38.28.146> was last seen on IRC in channel #asterisk, 10d 4h 33m 38s ago, saying: 'you have got to be joking me right'. |
| 03:40.37 | DaGrim | hmm my * wont work now |
| 03:40.37 | DaGrim | lol |
| 03:40.41 | DaGrim | because of it |
| 03:40.42 | ZX81 | hmm |
| 03:40.54 | ZX81 | 10 days? |
| 03:41.42 | *** join/#asterisk dca_ (~teliax@c-67-166-37-218.client.comcast.net) |
| 03:42.29 | Juggie | seems to be a long time for him |
| 03:42.52 | DaGrim | yea |
| 03:42.54 | DaGrim | really |
| 03:43.04 | ZX81 | ~yeah |
| 03:43.05 | jbot | well, yeah is YEAH |
| 03:43.13 | ZX81 | he's usually here every day |
| 03:43.13 | Juggie | ~unf |
| 03:43.14 | jbot | unf is probably the Universal Noise of Frustration, it is also washort's favourite word, after heh . something else related to pr0n movies |
| 03:43.29 | Juggie | ~doop |
| 03:47.32 | chesty | i'm using linphone to test with asterisk, during a call, the speech starts cutting out and becomes silent. usually in the exact same time into the call. tcpdump looks good, no dropped packets or big delays, but it does show icmp dest unreachable coming from the linphone host at the time of break up. any ideas? is there a better linux softphone around? |
| 03:49.07 | *** join/#asterisk scuba_laptop (~steve@rdu88-251-252.nc.rr.com) |
| 03:49.49 | scuba_laptop | Anyone know how to do a firmware upgrade on an spa-841 without the windows app they provide? |
| 03:50.35 | scuba_laptop | I see tftp requests for spa841.cfg, but don't have it :) |
| 03:53.26 | Juggie | check for an example in the documentation |
| 03:54.05 | scuba_laptop | Juggie.. Documentation? Where? |
| 03:54.17 | Juggie | er, documentation for the phone? |
| 03:54.30 | Juggie | i am only guessing ive only worked with cisco and mitel sip phones |
| 03:54.42 | shido6 | hrmm |
| 03:54.59 | scuba_laptop | Juggie: I ordered 2. One was dead and neither box had any paper other than the box itself. |
| 03:55.29 | Juggie | scuba_laptop, pdf online? |
| 03:55.43 | Juggie | which phone is it |
| 03:55.50 | Juggie | spa841 hold on |
| 03:56.02 | Jeet | i paid 10$ to voipjet and it just credited 9.7 |
| 03:56.05 | techie | pr0n affecting Qos..not good |
| 03:56.27 | *** join/#asterisk qwerp (~abc@219.95.105.74) |
| 03:56.37 | scuba_laptop | juggie: There's a user pdf but nothing in there of use. |
| 03:56.39 | qwerp | harlo... |
| 03:57.07 | scuba_laptop | Jeet: VoipJet charges you their paypal fees ... in violation of paypal policy... |
| 03:57.11 | qwerp | i had a CDR to fix, but i dunno how... |
| 03:57.15 | qwerp | anyone can help? |
| 03:57.46 | JunK-Y | qwerp: more details? |
| 03:57.57 | qwerp | if i ring a group of callers.. |
| 03:58.16 | JunK-Y | explain more if ya want help. |
| 03:58.17 | Jeet | scuba: but in their FAQ they brag about being a reputable company and not breaking any law. |
| 03:58.23 | qwerp | example exten => 1234,1,Dial(SIP/1&SIP/2&SIP/3) |
| 03:58.24 | cbachman | Jeet: complain to PayPal, vocally. They aren't supposed to do that. |
| 03:58.33 | scuba_laptop | Jeet: Donno what to tell you...:-) |
| 03:58.34 | qwerp | if SIP/2 pickup, |
| 03:58.57 | JunK-Y | so ? |
| 03:59.08 | scuba_laptop | qwerp, when some calls 1234 .. it calls 3 sip phones... |
| 03:59.17 | qwerp | in the cdr, it will show exten 1234 |
| 03:59.24 | scuba_laptop | qwerp .. whomever picks up "wins" and the others stop ringing. |
| 03:59.32 | JunK-Y | which is true. |
| 03:59.40 | randu | did they credit you 9.70 or did they receive 9.70? |
| 03:59.44 | scuba_laptop | greg_work: My other /nick won't time out :-) (ScubaSteve) |
| 03:59.45 | qwerp | wat i wan is to show SIP/2 |
| 03:59.55 | Juggie | scuba_laptop, is there a download config option? |
| 03:59.58 | qwerp | anyway i can do that? |
| 04:00.13 | JunK-Y | when doing a Dial, it's generate a new cdr, no? |
| 04:00.15 | scuba_laptop | greg_work: One of the 841's was a dud...Am trying to upgrade the firmware but have no info on how to do it (no windows here) |
| 04:00.23 | dontmsgme | Anyone knwo what this means Jan 31 19:53:33 WARNING[-193356880]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x94475a4 (len 446) to 68.121.22.193 returned -1: Bad file descriptor |
| 04:00.36 | scuba_laptop | Juggie: Yes, but it doesn't say how or where. it gives you a spot for firmware keys and a minimum and maximum upgrade version... |
| 04:00.47 | MrEntropy | type 'user' means the given UA can only place calls on asterisk, yes? |
| 04:00.55 | qwerp | i generates a cdr when u issue ResetCDR(w) |
| 04:00.57 | scuba_laptop | mrentropy: Correct |
| 04:01.15 | greg_work | scuba_laptop, theres a .bin file, i thought you could just submit it somewhere in the web interface? |
| 04:01.29 | Juggie | that sucks anyways, u want to get tftp working |
| 04:01.33 | scuba_laptop | greg_work: News to me. I'll look around some more.. sure are a lot of config options. |
| 04:01.51 | scuba_laptop | Juggie: I put the bin file in my tftp root, just like my ciscos want... it didn't pick it up. |
| 04:02.37 | greg_work | scuba_laptop, i dont see it? hm |
| 04:02.49 | *** part/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 04:03.41 | MrEntropy | scuba_laptop: so why does it work the other way around? i have SER forwarding to asterisk, it works if ser is set in sip.conf as a peer, but it doesn't when set to user...? |
| 04:03.48 | scuba_laptop | greg_work: It's not there.. |
| 04:03.55 | greg_work | scuba_laptop, i was actually going to write an app to configure the 841's for me.. so i could just put in the mac address and extension, and then have it find it (using rarp) and send the SIP login, dialing directory, set all the config options.. |
| 04:04.12 | scuba_laptop | greg_work: Spiffy! |
| 04:04.27 | greg_work | scuba_laptop, i can imagine what they'll say.. "oh, you have to upgrade to the newer firmware to get the upgrade page" |
| 04:04.29 | scuba_laptop | MrEntropy: No idea... |
| 04:04.42 | Juggie | scuba_laptop, http://www.phoneboy.com/blog/archives/2005/01/provisioning_si.html |
| 04:04.46 | scuba_laptop | greg_work: Yeah, was just thinking that... |
| 04:04.48 | scuba_laptop | sweet |
| 04:04.49 | MrEntropy | peculiar =/ |
| 04:05.40 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:05.57 | scuba_laptop | Juggie: Interesting, if this is true then I'm screwed :-) |
| 04:06.17 | Juggie | scuba_laptop, did u put the latest firmware on the tftp? |
| 04:06.32 | scuba_laptop | Juggie: yes, but all the phone did was look for spa841.cfg |
| 04:06.34 | scuba_laptop | which doesn't exist |
| 04:06.39 | Juggie | try this, http://www.phoneboy.com/bin/view.pl/Voip/SipuraUpgrade |
| 04:06.41 | scuba_laptop | which I don't have any idea what it would contian |
| 04:07.01 | scuba_laptop | yay upgrade w/o windows |
| 04:07.24 | Juggie | i realize its for a dif phone |
| 04:07.26 | Juggie | but it may apply |
| 04:07.57 | scuba_laptop | Juggie: Gonna try this. Crazy enough to work :) |
| 04:09.20 | Juggie | if it does add it to the wiki page for the phone :) |
| 04:09.49 | scuba_laptop | ok well for grins tried /upgrade on the phone with the browser... |
| 04:09.53 | Delmar | anyone have any thoughts or comments regarding the #define AGGRESSIVE_SUPPRESSOR in zconfig.h to try and get better echo cancellation? |
| 04:09.54 | scuba_laptop | It said upgrading on the phone |
| 04:10.05 | scuba_laptop | all it did was try to get spa841.cfg again by tftp, don't have it. |
| 04:11.04 | Juggie | does the phone webserver have a page which says provision phone? |
| 04:11.21 | scuba_laptop | specifying the whole thing, tftp://host/firmware.bin makes it pull the right bin file |
| 04:11.59 | Juggie | well, you should follow instructions :) |
| 04:12.25 | scuba_laptop | Well.. it pulled the firmware, didn't complain.. but version shows the old one still. |
| 04:12.46 | Delmar | Juggie sure.. you try and find the "ANY" key. |
| 04:12.55 | DaGrim | How do I save a page to disk via lynx? |
| 04:12.59 | DaGrim | lol |
| 04:13.07 | Juggie | just use wget |
| 04:13.10 | DaGrim | im trying to save app_dbodbc.c .. lol |
| 04:13.12 | DaGrim | k |
| 04:13.23 | scuba_laptop | DaGrim, let us know if you don't have wget or GET |
| 04:13.24 | DaGrim | Juggie: nice.. thanks for the tip |
| 04:13.25 | Delmar | yep. wget is your friend. |
| 04:13.30 | DaGrim | ;) |
| 04:13.38 | DaGrim | it works |
| 04:13.58 | Delmar | if I dont fix this echo cancellation problem... im going to start bashing things. |
| 04:14.03 | Juggie | scuba_laptop, did the phone do any kind of upgrade process? |
| 04:14.21 | Delmar | i have 8 more X100's in the drawer... i can afford to waste a few. |
| 04:14.23 | Juggie | did u set upgrade enable = yes |
| 04:14.28 | Juggie | on the web page. |
| 04:14.36 | scuba_laptop | Juggie: It said it was upgrading on the screen, pulled the firmware... No change in version # tho. |
| 04:14.44 | scuba_laptop | Upgrade is enabled in the firmware. |
| 04:14.46 | Juggie | did it reboot? |
| 04:14.59 | scuba_laptop | Juggie: Yes, then I power cycled it just for fun, no difference. |
| 04:15.04 | DaGrim | ok now how do i save in vi? |
| 04:15.05 | DaGrim | lol |
| 04:15.13 | Juggie | ew, vi |
| 04:15.13 | scuba_laptop | I've done a million 7960 upgrades and fw changes... |
| 04:15.13 | Juggie | i forget |
| 04:15.15 | scuba_laptop | :w |
| 04:15.23 | scuba_laptop | will write to the file you are editing |
| 04:15.28 | DaGrim | yea. i know but this screwy installer for myodbc did it and wants me to save it now |
| 04:15.28 | scuba_laptop | make sure you hit escape then :w |
| 04:15.29 | DaGrim | lol |
| 04:15.37 | DaGrim | ok.. thanks |
| 04:15.37 | scuba_laptop | .. then enter |
| 04:15.40 | Juggie | scuba_laptop, have you noticed the 7960 wont cross subnets for tftp? |
| 04:15.48 | DaGrim | gotcha |
| 04:15.59 | scuba_laptop | Juggie: I want to say I've upgraded across the internet more than once. |
| 04:16.05 | Delmar | hey has anyone here done any hardware level messing around with an Snom type phone? i have an snom100 and would love to get it working.. it crashed during firmware update and never booted again.. must be a way to clock in to the altera max chip and feed the firmware in again somehow...? |
| 04:16.21 | Juggie | if i run my tftp on the same subnet all is well, but across subnets it will see the tftp server, download configs, but it wont do an upgrade from it |
| 04:16.23 | Delmar | there is a JTAG port on the max chip. |
| 04:16.27 | scuba_laptop | Delmar as snom people. |
| 04:16.32 | scuba_laptop | JTAG, cool. |
| 04:17.21 | Juggie | that being said scuba_laptop, a mitel 5055 i have works fine across subnets for tftp, but that phone has other problems |
| 04:17.33 | Delmar | heh. i WAS in touch with Snom.. and they were even going to send me a free board .. in a sort of.. " stop fucking with our hardware and sending us hi-rez photos of our bits".... but .. i think they were pulling my leg.. they sent me a package alright... it had 10 of their software CD's in it is all. lol. |
| 04:17.35 | scuba_laptop | Juggie: I think the elusive spa841.cfg is needed... |
| 04:17.56 | Juggie | hrmf. |
| 04:17.59 | scuba_laptop | Juggie: I should have just sucked it up and ordered 7940 or 7960's |
| 04:18.14 | scuba_laptop | They work wonderfully in * |
| 04:18.18 | Juggie | scuba_laptop, i hear the mitel 5215's are good too |
| 04:18.27 | Juggie | i have a ton at work but they are not dualmode (aka no sip) |
| 04:18.35 | Juggie | we are getting more soon so i told the guys to make sure they are dual mode |
| 04:18.38 | scuba_laptop | Haven't tried any Mitel endpoints. |
| 04:19.04 | Juggie | the 5055 seems to work ok, i had no problem with conferencing, transfering, music on hold etc on thep hone |
| 04:19.14 | scuba_laptop | Cool. |
| 04:19.19 | Juggie | the only extra thing i had to do was hunt down a SNTP server (simple network time protocol) |
| 04:19.24 | Juggie | so i could have the mitel set its phone time. |
| 04:19.39 | ManxPower | pool.ntp.org |
| 04:19.50 | scuba_laptop | Run your own :) |
| 04:20.15 | Juggie | ManxPower, cant access external networks like that. |
| 04:20.31 | Qwell | port forward to it. heh |
| 04:20.53 | Juggie | its worth noteing that it need Sntp |
| 04:20.54 | Juggie | not just NTP |
| 04:21.07 | Juggie | i had to dig for like 30min to find a SNTP server. |
| 04:21.33 | Sedorox | Quick Q.. I finally got my problem solved... is there a way... that when when I call through Server A to Server B to see is X person ix avail. and they aren't.. it kicks back to server A... is there a way to then remove Server B from the loop (not have the call keep going through server B? I guess.. how would I do extention transfers... in extentions.conf... |
| 04:21.55 | *** join/#asterisk dano_ (~dano@70.57.156.97) |
| 04:22.23 | MrEntropy | discounting the weirdness of this question, is there a function i can use in the dialplan that gives me the position of a certain character/number in an extention, so that I may then use it in a substring()? |
| 04:22.48 | Juggie | MrEntropy, what are you trying to accomplish? |
| 04:23.28 | dano_ | OK - bottom line. * rocks. It just saved my butt. If any of the devs are on the channel - thank you. Big time. |
| 04:23.55 | MrEntropy | Juggie: i'm trying to stash messages in the sip message and decode them on asterisk...=D |
| 04:23.59 | Qwell | I'd bet donations are accepted |
| 04:24.13 | scuba_laptop | Juggie thanks for your help... will have to mess with it more tomorrow.. gotta hit the sack :) |
| 04:24.14 | scuba_laptop | Nite all |
| 04:24.29 | *** join/#asterisk kimosabe (~natt@dsl-200-78-71-49.prod-infinitum.com.mx) |
| 04:24.40 | Juggie | MrEntropy, i'll ask one more time |
| 04:24.44 | Juggie | what are you trying to accomlish |
| 04:24.46 | kimosabe | how can i see the interupts for my fxo card on freebsd |
| 04:24.59 | Juggie | *accomplish |
| 04:25.04 | MrEntropy | i just said |
| 04:25.13 | Juggie | yes |
| 04:25.14 | Juggie | i can see that |
| 04:25.18 | Juggie | but whats your reason behind it |
| 04:25.21 | Juggie | what is your overall goal |
| 04:25.37 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 04:26.08 | MrEntropy | an added layer of authentication between SER and asterisk, since SER cannot initiate an auth |
| 04:27.22 | *** join/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 04:27.23 | Juggie | so your making people do what, put a password in their phone number? |
| 04:27.36 | MrEntropy | nope, SER will insert that automagically |
| 04:27.51 | QRPartner | Hello, anyone familiar with the alarm debounce setting when loading the digium T1 card? |
| 04:28.04 | Juggie | oh, because SER cant auth against asterisk? |
| 04:28.14 | MrEntropy | yes |
| 04:28.18 | JunK-Y | QRPartner: red/yellow alarms? |
| 04:28.24 | Juggie | why dont you just lock down to a certain ip? |
| 04:28.33 | MrEntropy | Juggie: spoofing |
| 04:28.35 | QRPartner | Red... Trying to get rid of them |
| 04:29.04 | JunK-Y | T100P? |
| 04:29.10 | Juggie | MrEntropy, spoofing will only get you so far, you wont be able to receive the responses, therefore nothing will happen |
| 04:29.22 | JunK-Y | just plug a t1 cable, with the dchan, it supposes to be cleared. |
| 04:29.28 | Juggie | i could send data to asterisk and pretend i'm you, but i would never get hte answers. |
| 04:29.32 | QRPartner | JunK-Y: Yes |
| 04:29.41 | JunK-Y | so whats wrong? |
| 04:29.54 | Jeet | livevoip has better international rates tahan voipjet |
| 04:30.08 | MrEntropy | Juggie: that's a really low level spoof, you 'can' do a spoof so classy as to fool routers into forwarding you traffic meant for someone else |
| 04:30.09 | QRPartner | ->JunK-Y I keep getting alarms and the card keeps resetting. |
| 04:30.21 | Jeet | is livevoip's call quality comparable to voipjet |
| 04:30.27 | Qwell | PTG123: Ended up only being a few hours. |
| 04:30.35 | Juggie | MrEntropy, possibly... but that would be hard... anyways |
| 04:30.36 | PTG123 | wow |
| 04:30.40 | *** join/#asterisk Guest^DJ (some@211.24.146.10) |
| 04:30.40 | PTG123 | good deal :) |
| 04:30.47 | MrEntropy | Juggie: very hard |
| 04:30.48 | Qwell | oh, didn't think you were around, heh |
| 04:30.53 | MrEntropy | Juggie: but not impossible =/ |
| 04:30.55 | Qwell | Find anything out? |
| 04:30.57 | Juggie | is your password going to be variable, or fixed lenght? |
| 04:31.03 | PTG123 | yah |
| 04:31.07 | PTG123 | message me |
| 04:31.07 | JunK-Y | can ya reboot the machine? sometime, my T410P stays on green light even if no cable are plugged. |
| 04:31.14 | JunK-Y | dunno what happened exactly. |
| 04:31.26 | MrEntropy | Juggie: i'd like variable, but i already know what you'll suggest if fixed length |
| 04:31.42 | Juggie | MrEntropy, asterisk i think internally could only handle fixed lenght |
| 04:31.56 | QRPartner | -->JunK-Y: We have two T1s, it only does it on the one with higher volume. The Provider is the same and they say the lines are setup exactally |
| 04:32.06 | Jeet | POLL : LiveVoIP or VoIPJET ?? |
| 04:32.11 | MrEntropy | Juggie: might have to make an AGI then to do it for me |
| 04:32.21 | Juggie | that being said, you could call a perl script to process the string, and then return with two variables, one if the pass is correct, and one with the real dialstring |
| 04:32.24 | Qwell | Jeet: livejet |
| 04:32.32 | kimosabe | how can i check the interupts on freebsd |
| 04:32.52 | Juggie | MrEntropy, i've begun using php with asterisk for that, much easier then perl :) |
| 04:32.53 | Jeet | Qwell: 404 not found |
| 04:33.18 | *** join/#asterisk thetalon (~Ari@pcp05736786pcs.norstn01.pa.comcast.net) |
| 04:33.27 | MrEntropy | Juggie: well i don't know perl at all, i know php and C/C++...and bash, but i wouldn't trust a bash script with that |
| 04:33.27 | tessier | Jeet: VoIPVoIP |
| 04:33.39 | Qwell | tessier: sure, take the easy one :p |
| 04:33.49 | Juggie | MrEntropy, then look at using php |
| 04:33.52 | Juggie | let me get you the link |
| 04:35.05 | MrEntropy | you mean the voip-info link? |
| 04:35.09 | Juggie | MrEntropy, look at http://www.voip-info.org/wiki-Asterisk+AGI+php |
| 04:35.13 | Juggie | and look at #12 |
| 04:35.25 | Juggie | i started with that example and have had good success since that |
| 04:36.04 | greg_work | MrEntropy, what are you trying to do? |
| 04:36.53 | Juggie | MrEntropy, you should be able to pass in the dialed string easily enough and then parse it from there and accomplish what you want. |
| 04:37.15 | MrEntropy | sure, i just thought something like that was written =) |
| 04:37.52 | kimosabe | is any one runnig freebsd and asterisk |
| 04:38.24 | *** join/#asterisk robf (~robf@SP3-24.207.240.3.charter-stl.com) |
| 04:38.26 | Juggie | let me check the dialplan functions again |
| 04:40.23 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:40.38 | Juggie | yah i dont think u can search for a value in a string |
| 04:40.39 | Juggie | not that i see |
| 04:40.45 | Juggie | use php :) |
| 04:41.40 | *** join/#asterisk |Blaze| (dirc@d142-59-247-192.abhsia.telus.net) |
| 04:42.00 | *** part/#asterisk rene- (~rene-@201.137.86.219) |
| 04:43.14 | DaGrim | Can someone that updated recently give me line 30 of their Asterisk Makefile? |
| 04:43.24 | DaGrim | I changed something on accident and didnt realize it |
| 04:43.24 | DaGrim | blah |
| 04:43.35 | DaGrim | I have $(CC) $(SOLINK) -o $@ $< -lodbc |
| 04:43.46 | Qwell | cvs diff |
| 04:43.52 | DaGrim | on line 30.. and i get this error on make: Makefile:30: *** missing separator. Stop. |
| 04:43.57 | DaGrim | ok.. |
| 04:44.04 | DaGrim | thanks |
| 04:44.22 | DaGrim | holy hell that was neat. |
| 04:44.23 | Qwell | I don't know the cvs syntax, but you should be able to do cvs help diff |
| 04:44.27 | DaGrim | Im learning so much tonight.. lol |
| 04:44.35 | robf | 'missing separator' almost always means you have spaces for the indention instead of a tab character |
| 04:44.35 | DaGrim | i just typed cvs diff and it fixed that line.. lol |
| 04:44.38 | Nugget | "cvs diff <filename>" is sufficient. |
| 04:44.46 | Qwell | it shouldn't fix that line... |
| 04:44.49 | Nugget | it won't fix anything, but it'll tell you what's different. |
| 04:44.52 | Qwell | yeah |
| 04:44.53 | robf | it didn't fix anything, it just showed you what you did |
| 04:44.59 | DaGrim | oooh .. still neato.. lol |
| 04:45.00 | *** join/#asterisk Rick_Hunter (~rhunter@05-019.008.popsite.net) |
| 04:45.15 | greg_work | theres some interesting looking sound files in cvs.. teletubbie-murder.gsm |
| 04:45.22 | robf | if you just want current cvs back, delete it and 'cvs update' |
| 04:45.32 | greg_work | lyrics-louie-louie.gsm |
| 04:45.45 | Qwell | those are in asterisk-sounds |
| 04:46.05 | DaGrim | ok.. |
| 04:48.11 | *** join/#asterisk WizardWlf (~shawn@wrt54g.djernes.org) |
| 04:49.03 | WizardWlf | anyone know if there is a way to build zaptel / ztdummy into a 2.4 or 2.6 kernel, (not as modules) |
| 04:53.21 | JunK-Y | WizardWlf: i dont think so. |
| 04:53.29 | datareactor | WizardWlf why you want them not as module |
| 04:53.44 | JunK-Y | datareactor: i was to ask the same question :) |
| 04:53.52 | Qwell | anythings "possible", heh |
| 04:55.10 | datareactor | <PROTECTED> |
| 04:56.10 | file[laptop] | BoRiS: poke |
| 04:57.27 | *** join/#asterisk shaZwaz (~lukyali@203.81.196.167) |
| 04:57.54 | shaZwaz | morning ppl |
| 04:59.37 | datareactor | shaZwaz g morning |
| 05:01.49 | WizardWlf | very simply a hosting provider I use www.linode.com uses UML (user mode linux) but their security policy will not allow moduler kernels. So it ineeds to be build it |
| 05:02.46 | WizardWlf | datareactor: do you have a patch for that |
| 05:03.18 | datareactor | WizardWLF No :( |
| 05:03.55 | *** join/#asterisk Inv_arp (junya@adsl-3-247-162.mia.bellsouth.net) |
| 05:04.09 | WizardWlf | anyone have a hosting provider that doesn't cost an arm and a leg that lets you use modular kernels |
| 05:05.20 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 05:06.19 | postel | WizardWlf: linux setups of hosting providers suck bigtime, use a colo, bring your own box |
| 05:06.54 | Nugget | linux is poo. |
| 05:07.14 | postel | well, aint poo, its doing rather ok |
| 05:08.05 | postel | i managed AIX for 500 users, i KNOW what poo looks like |
| 05:08.24 | Nugget | I'm not saying that AIX isn't *also* poo. :) |
| 05:09.41 | postel | if you give it time and set it up the proper way its as good as solaris </flamebait> ^_^ |
| 05:10.18 | Nugget | I'm in favor of using the right OS for any given job. I just haven't found a job yet where Linux is the best OS. |
| 05:10.30 | Nugget | so maybe "poo" is a bit strong, but it sure isn't all that great |
| 05:10.47 | WizardWlf | I would if I cold afford it. most colo's want over $150 a month, the projects I am working on right now have to pay me anthing let alone pay for that high of price. anyone knof of a good low cost colo |
| 05:11.10 | Nugget | it's not a great server, it's not a great firewall, and it sure isn't a great desktop. it's not a great gaming machine, and it's not a great embedded os. |
| 05:11.37 | Nugget | it's just "fine" at all those things, and I hate settling for "fine" |
| 05:11.40 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:11.48 | Nugget | (well, it's far worse than "fine" for gaming) |
| 05:13.19 | greg_work | how can an AGI cause * to jump to n+101 priority ? |
| 05:14.09 | postel | Nugget: well, its *almost* stable (if you stay away from unstable trees) but then you get the nasty bugs of the exploited old versions, its doing alright in the desktop area (fbsd is MILES better but thats a different story altogether) gaming is not there yet but we're getting somewhere.. hardware support has gone a long way, memory handling the same, monolithic kernel OSes are considered obsolete.. well.. |
| 05:14.20 | postel | wait a sec.. it IS poo |
| 05:14.21 | postel | :P |
| 05:14.25 | Nugget | yay! :) |
| 05:14.45 | JunK-Y | greg_work: ${PRIORITY} |
| 05:15.55 | greg_work | JunK-Y .. what do you do with it? |
| 05:16.06 | JunK-Y | change it. |
| 05:16.14 | JunK-Y | but not sure why ya want to do it exactly. |
| 05:16.14 | *** join/#asterisk DaGrim (~junglesto@dagrim.user) |
| 05:16.37 | greg_work | oh nm, theres a priority command (in phpagi anyways) |
| 05:16.47 | greg_work | i'm rewriting voicemail as agi |
| 05:17.02 | greg_work | but i want to emulate comedian so it'll work as a drop-in replacement |
| 05:17.06 | file[laptop] | WizardWlf: I found one for $79.95... |
| 05:17.15 | JunK-Y | any goal to this except losing performances? |
| 05:17.19 | file[laptop] | don't remember it though, so good luck |
| 05:17.30 | greg_work | yes, getting features that are in CVS only into stable release |
| 05:17.42 | DaGrim | Who was talking about /usr/src/asterisk/asterisk-sounds ?? |
| 05:17.44 | greg_work | making it easier to modify |
| 05:17.45 | DaGrim | Is that where it was? |
| 05:17.52 | DaGrim | I wanna hear em.. lol |
| 05:18.01 | blankman | Anyone on use the PGSQL application? |
| 05:18.03 | Qwell | DaGrim: get the asterisk-sounds package, or get it from cvs |
| 05:18.08 | Qwell | cvs co asterisk-sounds |
| 05:18.10 | DaGrim | ahh cool.. thanks |
| 05:18.14 | Qwell | cvs has much more |
| 05:18.31 | DaGrim | Wish I woulda known it was so extensive before.. |
| 05:18.33 | DaGrim | lol |
| 05:18.38 | blankman | I am having a devil of a time getting to work with this stored proc ... can't figure out why it won't use "get" the result set :-( |
| 05:18.39 | JunK-Y | blankman: i do. |
| 05:18.40 | DaGrim | err rather.. so handy |
| 05:19.04 | blankman | JunK-Y, have you used a none sql language stored proc. with it successfully? |
| 05:19.28 | DaGrim | Qwell: theres like 1000+ sounds in there =P .. freakin awesom |
| 05:19.36 | JunK-Y | a none sql language ? im using pg/plsql. |
| 05:20.05 | blankman | JunK-Y, can I get you to go to #pgsql-prob so that I don't take up everyone else time on the list for this? |
| 05:20.14 | blankman | I am not sure that everyone cares :-) |
| 05:20.26 | Qwell | DaGrim: after you get it from cvs, run a make install |
| 05:20.43 | DaGrim | Qwell: ok.. I just updated everything else anyways .. |
| 05:20.58 | Qwell | then, you can do what I did. :P `find /var/lib/asterisk/sounds/ -name '*.gsm' -exec play {} \;` |
| 05:21.00 | Qwell | heh |
| 05:21.24 | DaGrim | whats that do? |
| 05:21.28 | Qwell | it actually took quite a while to go through them all. I was exhausted afterwards, heh |
| 05:21.32 | Qwell | plays them all... |
| 05:21.35 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 05:21.35 | DaGrim | ahh Im on putty ;( |
| 05:21.44 | *** join/#asterisk rumba (~ropawa@cs68201148-205.sw.rr.com) |
| 05:21.50 | DaGrim | I really wish I could do that.. lol |
| 05:22.19 | DaGrim | i dont even have the sound enabled on my * box.. dont run x on it either.. |
| 05:22.25 | DaGrim | just sits there.. lol |
| 05:25.46 | DaGrim | Wish I could have it play the whole directory back thru an exten |
| 05:25.47 | DaGrim | hrm |
| 05:26.06 | Nugget | write an AGI :) |
| 05:26.12 | Qwell | make it play them with moh |
| 05:26.13 | Qwell | heh |
| 05:26.22 | DaGrim | can you do gsm with that? |
| 05:26.25 | Qwell | dunno |
| 05:26.35 | DaGrim | hmmm i think its only mp3 |
| 05:27.13 | DaGrim | ill just dl the directory and use winamp skrew it |
| 05:32.32 | Guest^DJ | ~seen XZ81 |
| 05:32.36 | jbot | Guest^DJ: i haven't seen 'xz81' |
| 05:32.54 | Guest^DJ | ~seen ZX81 |
| 05:32.55 | jbot | zx81 <matt@222-152-158-141.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 1h 49m 42s ago, saying: 'he's usually here every day'. |
| 05:32.55 | freat | oh well... hehe so much for running an untested script on a box in another state... |
| 05:33.26 | freat | at least I can tell someone what to do at the console to preserve my uptime ;) |
| 05:34.40 | *** join/#asterisk CpuID (~nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 05:34.41 | *** part/#asterisk JunK-C (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 05:34.59 | DaGrim | Hey.. whats a free GSM Player for windoze? |
| 05:35.20 | Qwell | heh |
| 05:35.42 | freat | DaGrim: I think there's a plugin for winamp |
| 05:36.07 | DaGrim | really? sweet.. thx |
| 05:36.28 | DaGrim | I have like 1500 new asterisk-sounds I really wanna hear.. heh |
| 05:36.43 | DaGrim | i had never downloaded that pkg before |
| 05:37.06 | Qwell | DaGrim: There are a bunch in there that you'll probably be able to use |
| 05:37.31 | DaGrim | awesome =) |
| 05:37.52 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 05:37.53 | DaGrim | Yea I only had like less than a hundred before I got the addon.. |
| 05:37.53 | DaGrim | lol |
| 05:38.27 | Qwell | I especially enjoy the concatination of "press 1", "press 2", etc. much smoother |
| 05:39.22 | *** join/#asterisk simon_ca (sedgett@S01060004e23873e1.vc.shawcable.net) |
| 05:39.24 | DaGrim | Hmmm yea |
| 05:39.38 | DaGrim | Can you do that on the same Playback() or Background() |
| 05:39.46 | DaGrim | with just a comma? |
| 05:39.48 | Qwell | its one sound now |
| 05:39.57 | DaGrim | Oooooh. gotcha |
| 05:41.45 | DaGrim | asterisk-friend.gsm ... AWWW |
| 05:41.47 | DaGrim | hehe |
| 05:42.16 | DaGrim | yay my names in ther |
| 05:42.29 | Guest^DJ | hi, is anyone using handytone 486 with * ? |
| 05:42.32 | Qwell | yeah, mine too...(both of them). I was excited, heh |
| 05:42.54 | Qwell | being that there are only like 5 names, its pretty nice that its there |
| 05:43.11 | niZon | http://www.loligo.com/asterisk/sounds/allison-smith1/carried-away-by-monkeys.gsm |
| 05:43.13 | niZon | o.O |
| 05:43.28 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 05:43.38 | Nugget | tt-allbusy is my favorite allison sound clip. |
| 05:44.18 | shido6 | that tickles can be annoying if you have it as a part of your t or i exten lines |
| 05:45.15 | DaGrim | Haha.. gambling-drunk |
| 05:45.18 | DaGrim | sweeeeet |
| 05:46.12 | Qwell | yeah, that ones nice |
| 05:46.31 | Qwell | Playback(north) -- Playback(teletubbies-murder), is my personal favorite though |
| 05:46.43 | niZon | http://www.loligo.com/asterisk/sounds/allison-smith1/wolverine-hunting.gsm < um? |
| 05:46.48 | DaGrim | doh lol |
| 05:46.50 | Qwell | works great, for if I don't answer |
| 05:47.48 | *** join/#asterisk doushanes (~Owner@c-67-184-189-220.client.comcast.net) |
| 05:48.05 | doushanes | What's up all |
| 05:49.54 | *** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net) |
| 05:50.41 | doushanes | Do I need to order caller ID, call waiting, three-way calling from the phone company if I want to utilize the features in asterisk? |
| 05:51.13 | shido6 | you need to order caller id |
| 05:51.22 | shido6 | thats about it |
| 05:51.26 | xkev | where does -g switch dump the core to? |
| 05:51.47 | shido6 | wherever you ran it from |
| 05:51.52 | shido6 | from whatever dir you ran it from |
| 05:52.00 | xkev | gah then my -U switch prevented it from writing the core |
| 05:54.13 | Nukemizer | I am trying to get xorcoms distribtion to work and i can not seem to find and answer to the error message that I get. I am guessing it thinks i have a pri card, I am only using a tdm card with 2 FXS and 2 FXO |
| 05:56.02 | Nukemizer | http://pastebin.ca/5064 |
| 05:56.16 | Nukemizer | if anyone can help me locate this |
| 05:56.53 | simon_ca | anyone registered for von in march? |
| 05:56.55 | brett_ | re all |
| 05:57.05 | brett_ | is anyone here running * on osx? |
| 05:57.58 | shaZwaz | Nukemizer: put noload app_rxfax.so noload app_txfax.so in modules in conf |
| 05:58.36 | *** join/#asterisk WildPikachu (~nkukard@wildpikachu.user) |
| 05:58.59 | WildPikachu | anyone here using isdn internal modem + capi? |
| 05:59.06 | Nukemizer | Shazwaz, thanks Trying now :) |
| 06:01.36 | DaGrim | Is there a SayLetter() ? |
| 06:01.51 | shaZwaz | its SayAlpha |
| 06:01.55 | DaGrim | ok.. thanks |
| 06:01.55 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 06:02.30 | Grooby | hello hello |
| 06:04.07 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 06:04.29 | letherglov | doushanes, yes |
| 06:04.38 | letherglov | caller id is fairly pricey for a residential pots line too |
| 06:04.47 | letherglov | it's something like $7-8/mo from SBC |
| 06:04.57 | letherglov | considering the damn like is $10 |
| 06:05.02 | letherglov | it's a good percentage |
| 06:05.20 | `Sauron | Anyone ever had a problem where a spa-1001 gives you a fast busy w/o it actually hitting *? |
| 06:05.34 | letherglov | `Sauron, must be unable to register |
| 06:05.51 | `Sauron | I only get it if I try calling an international number |
| 06:05.57 | `Sauron | so it's registering alright |
| 06:05.59 | letherglov | you need to change the sip dial plan |
| 06:06.01 | Grooby | how do I get my spa-2000 to do call waiting w/ asterisk? |
| 06:06.18 | `Sauron | My dialplan is down to: (*x.|x.) |
| 06:06.22 | Grooby | when i use my spa 2000, and then from other line to dial my spa2k extension, i get the message the line is busy.... |
| 06:06.38 | `Sauron | on the sipura |
| 06:06.48 | letherglov | `Sauron, ok, I give up |
| 06:06.53 | `Sauron | It's not even hitting * |
| 06:06.58 | `Sauron | Pshaw. |
| 06:07.36 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 06:07.54 | Nukemizer | shaZwaz: Thank you that did it for me ! I am up and runing |
| 06:08.37 | *** part/#asterisk WizardWlf (~shawn@wrt54g.djernes.org) |
| 06:10.11 | MrEntropy | mohmp3 comes with the mpg123 package? |
| 06:10.27 | letherglov | make mpg123 will download and build the correct version |
| 06:11.37 | Grooby | does anyone know? |
| 06:13.59 | PyroSteve | my scroll back buffer wont let me go back very far |
| 06:14.10 | PyroSteve | did anybody reply to my questions |
| 06:14.16 | PyroSteve | about useage balenceing |
| 06:14.17 | PyroSteve | ? |
| 06:14.57 | PyroSteve | i need to be able balence the useage between my sip service providers |
| 06:15.18 | PyroSteve | ive read thru the news ground archvies with very little luck |
| 06:16.09 | PyroSteve | any tip on how to balence my useage across a few sip <-> pstn gateways ? |
| 06:16.46 | `Sauron | Hum di dum. |
| 06:16.55 | Grooby | lol |
| 06:17.03 | Grooby | lots of questions w/ no answers |
| 06:17.08 | PyroSteve | hehe |
| 06:17.30 | *** join/#asterisk blehhh (~james@dsl-202-173-158-49.vic.westnet.com.au) |
| 06:17.37 | blehhh | chan_iax2.c:3905 register_verify: No registration for peer |
| 06:17.38 | Grooby | anyone here uses Spa2k? |
| 06:17.45 | blehhh | would anyone know why these messages keep coming up? |
| 06:17.48 | `Sauron | Just 1. |
| 06:17.49 | `Sauron | 1k |
| 06:17.52 | blehhh | i have no idea how to fix it |
| 06:17.57 | Grooby | does 1k support call waiting? |
| 06:18.00 | `Sauron | and I found out apparently why it wasn't working |
| 06:18.00 | blehhh | and i've looked through mailing list etc |
| 06:18.01 | `Sauron | Grr. |
| 06:18.04 | `Sauron | Works now |
| 06:18.12 | Grooby | congrad sauron |
| 06:18.15 | `Sauron | Grooby: I think it should, not sure tho |
| 06:18.23 | Grooby | did you get it to work? |
| 06:18.45 | `Sauron | Haven't tried yet |
| 06:18.51 | Grooby | i dunno what I need to do to get spa2k to do call waiting |
| 06:19.02 | Grooby | use the spa2k line, then try to call it and get the busy message |
| 06:19.12 | `Sauron | Beats me |
| 06:19.17 | `Sauron | I think that happens to me as well |
| 06:19.44 | `Sauron | does feature call waiting |
| 06:21.04 | simon_ca | pyrosteve: i use random and goto... |
| 06:21.35 | simon_ca | pyrosteve: to be tricky you could set a var based on firsta nd second choice and check the dialstatus to try the other provider idf the first fails... |
| 06:22.24 | greg_work | how can i look at the contents of *'s db? |
| 06:22.38 | greg_work | nm |
| 06:22.47 | `Sauron | greg |
| 06:22.52 | `Sauron | Did you get fwd stuff working? |
| 06:23.12 | greg_work | i have outgoing calls i think, but incoming doesnt seem to wok. no clue why |
| 06:23.20 | `Sauron | weird |
| 06:23.24 | greg_work | work, rather |
| 06:23.29 | Jeet | do i need to re-register with broadvoice every 15 minutes? if so how ? |
| 06:23.37 | `Sauron | Jeet: Err, don't think so |
| 06:23.44 | PyroSteve | simon_ca: hey ... thanks |
| 06:24.02 | PyroSteve | simon_ca: i just simply needed a little kick start in the ass |
| 06:24.15 | `Sauron | Anyone know how if anything needs to be configured in * for call waiting? |
| 06:24.32 | Jeet | sauron : every day incoming calls are forwarded to BV voicemail. and once i restart asterisk everything is back to normal |
| 06:24.51 | simon_ca | yrosteve: see http://www.edgett.bc.ca/simonsays/archives/000613.html - don't think i've changed it much since that post |
| 06:28.28 | PyroSteve | simon_ca: thanks |
| 06:28.37 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:31.19 | *** join/#asterisk simon_ca (sedgett@S01060004e23873e1.vc.shawcable.net) |
| 06:32.16 | *** join/#asterisk DaGrim (~junglesto@dagrim.user) |
| 06:33.34 | DaGrim | Does anybody know where Im supposed to copy the /install/MyODBC-3.51.10-pc-linux-i686/lib/*.so and *.la files to for this package.. seems thats all they have included.. and dont tell you where to put them. and Im trying to install it along with ..unixODBC-2.2.10 .. to work with * .. but .. Im getting this error when I try to load * now: loader.c:301 __load_resource: libodbc.so.1: cannot open shared object file: No such file or directory |
| 06:33.37 | DaGrim | then it crashes. |
| 06:33.53 | DaGrim | unixODBC worked fine.. |
| 06:33.57 | Grooby | sauron, figured out I need to enable CW |
| 06:33.59 | DaGrim | and I used the --prefix= |
| 06:34.07 | `Sauron | Grooby, how? |
| 06:34.08 | Grooby | for each extension..but would like to have it enabled by default |
| 06:34.29 | `Sauron | huh? |
| 06:34.37 | Grooby | I use amp, and in the extension.conf, it has the code dbput(CW/exten=ENABLED) |
| 06:34.51 | Grooby | just tried it with sjphone (it wasn't doing call waiting before either) and now it works |
| 06:35.01 | Grooby | but my spa2k is acting weird with it's dialplan |
| 06:35.04 | Grooby | need to fix that |
| 06:35.09 | Grooby | can't do *70 |
| 06:35.16 | `Sauron | Hum di dum. |
| 06:35.27 | DaGrim | =\ .. dam |
| 06:35.30 | `Sauron | Ah. |
| 06:35.54 | `Sauron | Grooby: I have the fix-it-all dialplan for the sipura's :) |
| 06:36.11 | Grooby | what do you use? |
| 06:36.13 | `Sauron | (*x.|#x.|x.) |
| 06:36.14 | `Sauron | :) |
| 06:36.17 | Grooby | roflmao |
| 06:36.20 | Grooby | I guess I could use that |
| 06:36.38 | `Sauron | shrug, I just let * take care of dialplan stuff |
| 06:36.48 | Grooby | that's a good point |
| 06:36.56 | `Sauron | and greg's canadian. :p |
| 06:37.14 | `Sauron | j/k |
| 06:37.37 | Grooby | that didn't work |
| 06:37.39 | Grooby | very wierd |
| 06:37.42 | `Sauron | Grooby: in your dbput thing, is "exten" the extension, or the word "exten" |
| 06:37.45 | DaGrim | `Sauron: you know anything about MyODBC? just installing it? heh =\ |
| 06:37.51 | `Sauron | DaGrim: Nope. |
| 06:37.54 | Grooby | it's the extension |
| 06:37.57 | Grooby | something is funking |
| 06:38.00 | DaGrim | or at least disabling it to load on startup |
| 06:38.00 | `Sauron | Humhum. |
| 06:38.06 | Grooby | *7 is not getting passed through |
| 06:38.06 | DaGrim | I dont even want it at this point |
| 06:38.11 | Grooby | is it my dmtf setting? |
| 06:38.20 | `Sauron | Could be |
| 06:38.29 | Grooby | how's your setup? |
| 06:38.30 | Grooby | inband? |
| 06:38.33 | Grooby | or auto? |
| 06:38.35 | `Sauron | Umm |
| 06:38.43 | `Sauron | sipura->* is rfcwhatever |
| 06:38.51 | `Sauron | *->bv is now rfcwhatever too |
| 06:38.54 | `Sauron | 3288 |
| 06:38.56 | `Sauron | err |
| 06:38.58 | `Sauron | 2388 |
| 06:39.12 | Grooby | interesting |
| 06:39.22 | Grooby | broadvoice right? |
| 06:39.26 | Grooby | i thought broadvoice uses inband |
| 06:39.29 | `Sauron | I used to use inband, until I found out all of it can handle rfc2388 |
| 06:39.34 | `Sauron | That's what they said |
| 06:39.41 | Grooby | interesting...... |
| 06:39.48 | `Sauron | well, there's a guy who was in here who claims they do rfc2388 |
| 06:39.51 | `Sauron | so I switchd |
| 06:39.53 | `Sauron | switched |
| 06:39.55 | `Sauron | and it works |
| 06:40.00 | `Sauron | so far, anyway |
| 06:42.29 | Grooby | it use to work |
| 06:42.31 | Grooby | not anymore |
| 06:42.32 | Grooby | :( |
| 06:42.47 | freat | ahhhhh |
| 06:49.05 | DaGrim | Damnit.. Im getting errors on * load now.. How do I take it back to CVS-curr? |
| 06:49.45 | DaGrim | Im tired of this.. I just want it to go back to the way it was.. w/ out trying to install libodbc ..that isnt there. |
| 06:50.02 | DaGrim | If it aint one thing.. its another.. lol |
| 06:53.44 | `Sauron | Hum. |
| 06:53.49 | freat | I think I've finally gotten my QoS script working well |
| 06:53.53 | `Sauron | I'd suggest looking for "unixodbc" as the package |
| 06:53.59 | `Sauron | it's not called libodbc :) |
| 06:54.05 | freat | VoIP + Video + Citrix |
| 06:54.30 | DaGrim | loader.c |
| 06:54.51 | DaGrim | where does it look for .so.1 or .so files.. and what is the file it defines that? |
| 06:54.56 | DaGrim | in.. |
| 06:55.56 | DaGrim | loader.c:301 __load_resource: libodbc.so.1: cannot open shared object file: No such file or directory |
| 06:56.30 | DaGrim | I havent been able to run * all night.. |
| 06:56.30 | DaGrim | what am I doing wrong? |
| 06:57.11 | *** join/#asterisk three55ml (~who@cs666898-229.austin.rr.com) |
| 06:57.52 | three55ml | Does anyone have any experience with FAX over IP? I know Vonage does it, anyone gotten it to work? |
| 06:57.59 | three55ml | I've tried ulaw and alaw with no success. |
| 06:58.03 | Grooby | ok |
| 06:58.05 | Grooby | i fixed my problem |
| 06:58.29 | Grooby | i resetted my spa2k couple days ago and it added bunch of stuff in the provisional tab that i don't use...took everything out |
| 06:58.30 | Grooby | hehehe |
| 06:59.42 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 07:00.56 | DaGrim | So my shit doesnt work.. cant I just rm * in /usr/src/asterisk |
| 07:01.01 | DaGrim | and do cvs update |
| 07:01.06 | DaGrim | and it will re-download it all? |
| 07:01.52 | DaGrim | i must have moved file(s) into /usr/src/asterisk at one point or another that is causing a problem |
| 07:03.30 | kram | ~ |
| 07:04.26 | *** join/#asterisk alakdan (~alakdan@210.213.196.101) |
| 07:05.05 | shaZwaz | any ideas on initializing a global from * DB on reload/startup |
| 07:06.32 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 07:09.36 | DaGrim | finally |
| 07:09.39 | DaGrim | whew |
| 07:11.03 | alakdan | any one subscribed to nufone? Im just wondering how many will be able to 'call' simultaneously the DID number? I mean if one customer dialed the DID number and gets routed to our asterisk box , will another customer be able to call the same DID number withou getting a busy signal? |
| 07:12.24 | DaGrim | alakdan: as many as you want.. |
| 07:12.25 | DaGrim | alakdan: but you will need plenty of b/w |
| 07:12.46 | DaGrim | alkadan: yea it uses diff channels |
| 07:13.04 | datareactor | i get this error when try to register to go2call |
| 07:13.06 | datareactor | chan_sip.c:4001 sip_reg_timeout: Registration for |
| 07:13.06 | datareactor | 023 'mylogin@sip01.go2call.com' timed out, trying again |
| 07:13.16 | alakdan | DaGrim: so its not something like a normal phone where if the phone is in 'use' others will get a busy tone? |
| 07:13.54 | DaGrim | not at all |
| 07:13.59 | datareactor | here is my setting http://pastebin.ca/5070 |
| 07:14.14 | DaGrim | It can definetly handle many many simeltanious (sp?) calls |
| 07:14.19 | *** join/#asterisk makamani (~user@pub-nms.stcl.com) |
| 07:14.21 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:14.59 | makamani | what does the below mean: NOTICE[4956]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'local' |
| 07:15.31 | DaLion | makamani: you need [local] in your extensions.conf |
| 07:15.44 | makamani | it is there, is there a special way to load it? |
| 07:15.45 | DaLion | or sned to a different one with @somewhere |
| 07:15.48 | alakdan | DaGrim: Do I need a diguim hardware for that? or a asterisk box will suffice? |
| 07:15.52 | DaLion | where is call from ? |
| 07:16.05 | DaLion | iax? |
| 07:16.08 | DaGrim | alkadan: You can run it over a T1.. |
| 07:16.21 | DaLion | makamani ? |
| 07:16.40 | tafazzi | Morning (italy) |
| 07:16.41 | makamani | from extension |
| 07:16.52 | DaLion | like if you send to that * box with DIAL(IAX2/blah/Exten |
| 07:17.16 | DaLion | try IAX2/somserver/EXTEN@local |
| 07:17.24 | DaLion | bbl8 |
| 07:19.22 | *** part/#asterisk doushanes (~Owner@c-67-184-189-220.client.comcast.net) |
| 07:20.14 | greg_work | is the only way to set voicemail options like "saycid" in voicemail.conf? ie, the actual voicemail user can't do it from the menu? |
| 07:23.07 | freat | dammit!!! |
| 07:23.31 | freat | I've been sitting here, soaking my network to test QoS... and the cord to my handset was loose |
| 07:23.32 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 07:23.42 | freat | I thought I was dropping audio |
| 07:23.45 | freat | lol |
| 07:24.00 | makamani | hi DaLion, we are trying a normal extension to extensino |
| 07:24.02 | makamani | no Ip calling |
| 07:26.11 | datareactor | can anyone look into my configs http://pastebin.ca/5070 unable to reg to go2call |
| 07:36.06 | *** part/#asterisk chesty (chesty@unconcerned.org) |
| 07:45.54 | *** join/#asterisk zoose (~kvirc@dsl-136.116.240.220.dsl.comindico.com.au) |
| 07:51.01 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 07:51.01 | makamani | i am trying to setup an asterisk pbx with voicetronix card openswitch12 |
| 07:51.01 | makamani | no IP comms right now |
| 07:51.01 | makamani | just plain analog handsets to analog handsets with analog trunk lines |
| 07:51.01 | *** join/#asterisk UrBaNLeGeNd (~root@202.61.44.3) |
| 07:51.02 | makamani | anybody there or am i lagging a lot |
| 07:53.15 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 07:54.11 | *** part/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 07:54.18 | DaGrim | yay |
| 07:54.27 | DaGrim | it all works again.. |
| 07:54.36 | DaGrim | and i have bandwith again |
| 07:54.40 | *** join/#asterisk UrBaNLeGeNd (~cron@202.61.44.3) |
| 08:07.01 | DaGrim | l8r all |
| 08:09.13 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 08:09.54 | firestrm | is it me, or are people who are expert at debian arrogant.. |
| 08:10.18 | djin | It's just you ;) |
| 08:11.10 | firestrm | i asked on #debian how to install with unstable, and was basicly told if i have to ask ,im not good enough to run unstable.. |
| 08:11.17 | shaZwaz | any ideas how to load globals from *DB on startup/reload |
| 08:12.09 | djin | firestrm, that's cold. |
| 08:12.35 | djin | Why run unstable btw, better go with testing. |
| 08:14.19 | fa | anyone use postgresql with cdr and ast_data |
| 08:14.22 | fa | ? |
| 08:14.28 | firestrm | djin, i need to upgrade to exim 4 before anyone on #exim will tell me how to add a blacklist to my mailserver.. |
| 08:14.37 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
| 08:15.27 | datareactor | can anyone check http://pastebin.ca/5070 i am unable to reg to go2call |
| 08:15.50 | djin | Why not just upgrade Exim :? |
| 08:16.09 | DaGrim | go2call? |
| 08:16.33 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 08:16.43 | firestrm | djin, not that easy, configs are different. and im unknowlegable enough that i cant build them without the aid of the system tool |
| 08:16.47 | mAsH` | morning all |
| 08:17.00 | firestrm | datareactor, are you behind a nat? |
| 08:18.01 | firestrm | i think i might just get rid of exim and use qmail or postfix |
| 08:18.02 | datareactor | firestrm no nat |
| 08:18.19 | firestrm | and sip show registry =? |
| 08:18.21 | djin | don't use qmail, go for postfix. |
| 08:18.52 | firestrm | djin, thanks for the suggestion.. i will try postfix first |
| 08:18.55 | fa | so what with that postgresql, anybody use? |
| 08:18.56 | *** join/#asterisk porche (~a@81.215.118.183) |
| 08:18.57 | porche | hi |
| 08:19.16 | shaZwaz | hi porche |
| 08:19.21 | wasim | wroom wroom porche |
| 08:19.28 | shaZwaz | wasim |
| 08:19.28 | datareactor | firestrm Host Username Refresh State |
| 08:19.28 | datareactor | sip01.go2call.com:5060 mylogin 120 Request Sent |
| 08:19.41 | djin | firestrm, you'll like it. Just some config files and lots of configuration options. |
| 08:19.43 | porche | hi wasim |
| 08:19.47 | porche | hi shaz |
| 08:19.55 | porche | guys for sure, got a question |
| 08:20.09 | firestrm | datareactor, that looks like a firewall problem.. try nat=yes just for shits and giggles |
| 08:20.18 | porche | on a x100P, *, cannot detect the hang up before the timeouts |
| 08:20.18 | shaZwaz | wasim any ideas on loading globals from *DB on startup/reload automatically |
| 08:20.34 | porche | is this normal? |
| 08:20.41 | datareactor | ok |
| 08:20.47 | jetscreamer | you run * on sparc, firestrm ? |
| 08:20.49 | wasim | porche: depends on the line signalling |
| 08:21.08 | porche | you mean zaptel.conf |
| 08:21.11 | wasim | shaZwaz: no, haven't fiddled with that |
| 08:21.16 | wasim | porche: yes, zaptel.conf and the telco |
| 08:21.23 | makamani | anybody having a sample extensions.conf for analog handsets only. i dont seem to get hold of the configuration right |
| 08:21.23 | firestrm | jetscreamer, no my mailserver/webserver does, i have a spare ultrasparc that i was going to try it on though |
| 08:21.28 | porche | hms got it, for loadzone |
| 08:21.30 | *** join/#asterisk Martohtar (Martohtar@194.19.32.100) |
| 08:21.42 | makamani | every digit i dial keeps saying cannot find extension context |
| 08:21.45 | jetscreamer | ah |
| 08:21.48 | makamani | every digit i dial keeps saying "cannot find extension context" |
| 08:21.49 | porche | I could not find one for turkey, where/what must I look for? |
| 08:21.49 | DaGrim | lol.. anybody else ever use... wtf is <cmd> ?? |
| 08:21.59 | fa | can anyone paste me a cdr_pgsql.conf standard file, i did a mistke and delete it |
| 08:21.59 | *** join/#asterisk meppl (~mephisto@p54853A08.dip.t-dialin.net) |
| 08:22.05 | meppl | guten morgen |
| 08:22.14 | porche | morgen meppl |
| 08:22.29 | meppl | good morning porche |
| 08:22.45 | datareactor | still can reg to go2call http://pastebin.ca/5070 |
| 08:23.46 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:23.55 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 08:24.31 | DaGrim | datareactor: what is go2call? lol |
| 08:25.09 | firestrm | datareactor, are you running uptables.. (this one bit me in the ass once) |
| 08:26.00 | datareactor | no iptables ethereal shows 216.52.153.209 SIP Request: REGISTER sip:sip01.go2call.com |
| 08:26.04 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:26.53 | firestrm | datareactor, i have no doubt that you data is getting there, i think the problems are in the return.. |
| 08:27.34 | firestrm | datareactor, do a sip debug, and carefully walk though the messages.. |
| 08:27.35 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 08:28.01 | datareactor | firestrm ok |
| 08:28.59 | firestrm | datareactor, im willing to guess that the return is missing.. which means, isp block, nat block, firewall block.. |
| 08:29.09 | mAsH` | sorry, onyone can help me |
| 08:29.21 | mAsH` | when i try to start * i get this msg |
| 08:29.22 | mAsH` | Illegal instruction (core dumped) |
| 08:29.37 | firestrm | mAsH` running g729? |
| 08:30.14 | mAsH` | no...it's first time that i try to start *, it's a new installation |
| 08:30.22 | firestrm | mAsH`, i found it does this when you try to run a loadable module at was built for another processor |
| 08:31.11 | firestrm | hmm... very wierd.. asterisk, usually runs nicely aout of the box |
| 08:31.44 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 08:32.08 | firestrm | mAsH`, i would try a make clean;make on the source again.. and watch the warnings.. maybe some clue |
| 08:36.05 | datareactor | firestrm i forgot to mention this error SIP Status: 405 Method Not Allowed |
| 08:36.30 | *** part/#asterisk DaGrim (~junglesto@dagrim.user) |
| 08:37.19 | firestrm | datareactor, that tells me that the server doesn't like you for some reason.. do you have a bindaddress = anywhere? |
| 08:37.19 | mAsH` | well,i'm trying |
| 08:39.09 | datareactor | firestrm no i am try to use it now |
| 08:40.05 | firestrm | datareactor, try bindaddress=0.0.0.0 |
| 08:40.44 | *** join/#asterisk cjk (~cjk@80.92.75.32) |
| 08:40.57 | cjk | hi, what do you suggest as a stun server? |
| 08:41.12 | *** join/#asterisk MuchToDo (~jack@82-32-5-69.cable.ubr01.azte.blueyonder.co.uk) |
| 08:41.40 | firestrm | cjk, i dont know.. thats a minefield i havent crossed yet.. |
| 08:42.13 | datareactor | firestrm it is already bindaddr = 0.0.0.0 |
| 08:42.25 | firestrm | hmmm... |
| 08:42.47 | firestrm | and you sure thay you have the correct username/passwd |
| 08:43.06 | datareactor | firestrm yes |
| 08:43.22 | firestrm | other than that.. im stumpped.. but thats the nature of SIP.. i only use it when no choice.. |
| 08:43.39 | firestrm | iax= channel joy |
| 08:44.40 | firestrm | you may have to contact go2call.. i did with terracall, it helped as they could see in their logs why it was dieing |
| 08:44.44 | fa | are anybody use ast_data ? |
| 08:46.19 | datareactor | firestrm i just called my cell from go2call dialer account is ok thanks for the help |
| 08:46.33 | fa | reload |
| 08:46.48 | firestrm | datareactor, its working now? |
| 08:47.26 | firestrm | sounds like SIP, works one minuit, broken the next.. |
| 08:50.40 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 08:52.19 | MuchToDo | hm |
| 08:52.22 | MuchToDo | is SIP that bad? |
| 08:52.41 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:52.44 | MuchToDo | I'm just gettting started, so should I concentrate on IAX2 from the start? |
| 08:52.51 | *** join/#asterisk porch2000 (~a@dsl81-215-30391.adsl.ttnet.net.tr) |
| 08:52.54 | Zeeek | sounds like a good plan |
| 08:52.59 | Zeeek | except... |
| 08:53.09 | Zeeek | if you want hardware phones |
| 08:53.09 | datareactor | firestrm its not working with asterisk :( |
| 08:53.20 | *** join/#asterisk LarsAC (~chatzilla@134.130.124.227) |
| 08:53.28 | Zeeek | GOOD ${LocalTimeExpression} |
| 08:53.56 | datareactor | Zeeek hi can you check my config http://pastebin.ca/5070 |
| 08:54.10 | firestrm | MuchToDo, iax i had set up the first time in 5 min, Sip took me 2 weeks |
| 08:55.00 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 08:55.00 | firestrm | Zeeek, iaxy's work great.. |
| 08:55.15 | datareactor | firestrm :) |
| 08:55.20 | jetscreamer | see, i can just sit here and pick your brains, no hands required :) |
| 08:55.23 | LarsAC | someone using hfc drivers (zaphfc) ? |
| 08:55.26 | cervajs | me |
| 08:55.32 | cervajs | larsac: me |
| 08:55.35 | UrBaNLeGeNd | anyone using areskicc app? |
| 08:55.49 | LarsAC | cervajs: have you tried florz' patches ? or the vihai drivers ? |
| 08:56.03 | Zeeek | I have an IAXy |
| 08:56.17 | datareactor | LarsAc yes |
| 08:56.21 | fa | ay one is recording all calls? |
| 08:56.37 | cervajs | larsac: vihai |
| 08:56.56 | LarsAC | cervajs: I didn't get it to compile this morning |
| 08:56.57 | firestrm | Zeeek, my info on iaxy's comes second hand, but from a trusted source.. Im buying 6 on them tomorrow.. |
| 08:57.14 | LarsAC | cervajs, datareactor: I have lots of problems regarding stability on my SMP machine |
| 08:57.19 | Zeeek | it has its faults, see the mailing list for a detailed view of those |
| 08:57.34 | cervajs | larsac: i have 1.0.3 + bristuff 0.2.0RC3 + vihai-0.31 |
| 08:57.40 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 08:57.41 | Zeeek | most important you need a proper 1200 ma power supply |
| 08:57.52 | firestrm | Zeeek, they are nice because you just have to wire the building with cat 5, no cat 3 and put iaxy's where you want hard phones |
| 08:58.16 | firestrm | Zeeek, they are that power hungry? wow |
| 08:58.20 | Zeeek | I think if they are all on the same side of the LAN they would work very well but are not cost effective |
| 08:58.32 | fa | how to record call.. after make dial? |
| 08:58.41 | Zeeek | yes and if you use a ps that isn't quite up to it, they'll work but die intermittently |
| 08:58.56 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 08:59.10 | *** join/#asterisk RoyK (~roy@i171219.dsl.tjukkband.no) |
| 08:59.11 | firestrm | Zeeek, switching or linear wallwart? |
| 08:59.40 | MuchToDo | The other thing I've found is that VOIP hardware is not really readily available in the UK. |
| 08:59.42 | Zeeek | I'm using a Radio SHack 1500ma supply |
| 08:59.55 | Zeeek | yes it is |
| 08:59.56 | MuchToDo | There's only a couple of suppliers and the prices seem way higher than they should be :/ |
| 09:00.04 | Zeeek | it's just horribly expensive |
| 09:00.11 | Zeeek | yup |
| 09:00.13 | firestrm | zeeek, i have a bunch of 1.5a switchers |
| 09:00.31 | Zeeek | try one out - if it doesn't burn it out, you're ok :) |
| 09:01.15 | cervajs | larsac: i have single processor, athlon |
| 09:01.47 | LarsAC | cervajs: tried bristuff 0.2.0-RC5 and vihai 0.3.3 this morning, but it gave compiler errors about invalid lvalues... |
| 09:01.48 | kks | Hello, i want to implement G.729 in my *, voiceage and www.readytechnology.co.uk, which one is better. |
| 09:02.23 | cervajs | larsac: i can try this evening |
| 09:02.37 | *** join/#asterisk pranav (dawda_pran@203.115.89.135) |
| 09:02.59 | cervajs | larsac: which version of * ? |
| 09:03.13 | pranav | hello is there any one |
| 09:03.46 | LarsAC | cervajs: RC5 of bristuff uses 1.0.5 I guess |
| 09:03.51 | LarsAC | cervajs: gcc is 3.3.5 |
| 09:04.02 | Zeeek | pranav millions of us |
| 09:04.25 | pranav | hi zeeek |
| 09:04.55 | pranav | thank for that |
| 09:05.07 | kks | Hello, i want to implement G.729 in my *, but i'm bit blur who should i get the codec and who should i pay for the license? |
| 09:05.13 | pranav | i am facing a problem in making calls |
| 09:05.25 | Zeeek | kks http://www.digium.com |
| 09:05.35 | Zeeek | they sell license for $10 ea |
| 09:05.43 | Zeeek | you need one for each channel |
| 09:05.57 | pranav | i have a sipura device(spa/2000) and 2 lines attached to it |
| 09:06.03 | Zeeek | for example, I bought 4, that's good enough for two concurrent calls |
| 09:06.24 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:06.37 | tzafrir | Someone asked me earlier about purchaing Diguim hardware in Israel, so: http://mirror.hamakor.org.il/archives/linux-il/01-2005/13861.html |
| 09:06.49 | pranav | i have registered both the phones i.e I am getting the dialtone in both the phones , but iam not able to call between the 2 |
| 09:07.02 | kks | if i have 30 for E1 card, so i have get $300? |
| 09:07.17 | firestrm | Zeeek, i found it quite annoying that digium's g729 register util requires an internet connection.. you have to haul the box in, expose it to the net, register , then go install the thing |
| 09:07.22 | Zeeek | pranav dialtone is probably generated by the phone and doesn't mean anyhting |
| 09:07.35 | datareactor | kks 10$ for each concurrent seesion |
| 09:07.59 | Zeeek | no $10 per channel - you'd need two if both ends use 729 IIRC |
| 09:08.08 | pranav | no after i had made some mistake in sip.conf which i rectified, only after that the phones registered and the dialtone came |
| 09:08.11 | fa | any one is recording calls to wav file? |
| 09:08.29 | datareactor | right Zeeek |
| 09:09.16 | Zeeek | pranav maybe but I believe it is still generated by the phone. Anyway, "sip show peers" |
| 09:09.39 | firestrm | need sleep, gnite all.. |
| 09:09.45 | Zeeek | s'long! |
| 09:09.53 | *** part/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 09:09.59 | pranav | ya i did that it shows both the lines(spa2000 and spa2001) and a long list |
| 09:11.08 | Delvar | morning |
| 09:11.16 | datareactor | firestrm gnight |
| 09:11.52 | pranav | tell me i have given the extension 2000 and 2001 to the 2 phones, to make call what exactly i have to dial? |
| 09:12.13 | wasim | pranav: 2000,1,Dial(SIP/phone1) |
| 09:12.16 | fa | Executing Monitor("SIP/1001-9653", "wav|/work/www/htdocs/inez/mp3/asterisk-recording") in new stack |
| 09:12.20 | fa | <PROTECTED> |
| 09:12.23 | fa | why it's not recording anything ? |
| 09:12.26 | fa | only create a file |
| 09:13.26 | datareactor | fa is no file is created ? |
| 09:13.52 | fa | no. file is created, but empty |
| 09:14.05 | pranav | thanks mr. wasim but can u tell me where to put this command |
| 09:14.09 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:14.09 | fa | w8. i will install sox. |
| 09:14.12 | wasim | pranav: extensions.conf ... |
| 09:14.43 | fa | sox is needed right? |
| 09:15.06 | kks | Zeeek, so i have to pay Digium, Intel IPP lib, what about voiceage? |
| 09:15.18 | datareactor | fa i think it is not needed but not sure |
| 09:15.35 | fa | datareactor so what other may be bad? |
| 09:16.08 | fa | And i want to record only that files.. which is accept by person how i call.. if no answer. i don't want to record |
| 09:18.32 | pranav | ya mr wasim in my extensions.conf i have context=[from-sip],exten=>2000,1,Dial,sip/spa2000|30|t |
| 09:19.01 | pranav | still do i need the add the line you told me before |
| 09:20.22 | Zeeek | kks you're prolly best off just dealing with Digium on that |
| 09:20.29 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 09:21.14 | kks | Zeeek, thanks |
| 09:21.35 | Zeeek | np |
| 09:21.50 | Zeeek | wasim mon ami - comment va ? |
| 09:23.10 | pranav | can any one you tell me where the problem is and why am i not able to connect between two phones |
| 09:25.33 | *** join/#asterisk djin (~marius@217-19-18-130.dsl.cambrium.nl) |
| 09:26.02 | pranav | hi djin |
| 09:26.17 | Zeeek | pranav what is CLI output when you dial? |
| 09:27.01 | pranav | i am not getting anything on the cli prompt |
| 09:27.18 | Zeeek | ???? |
| 09:27.31 | Zeeek | you say the phones are registered? |
| 09:27.56 | Zeeek | Is there a send button on the phone (like * or #)? |
| 09:28.10 | Zeeek | int's an ATA, not a phone in fact right |
| 09:28.15 | pranav | yes there are both these buttons |
| 09:28.37 | Zeeek | so presumably you are hitting SEND |
| 09:28.57 | Zeeek | in other words, how does the phone know you have finished the number? |
| 09:29.05 | Zeeek | (I never had an ATA) |
| 09:29.18 | Zeeek | maybe it's in the Sipura dialplan? |
| 09:29.35 | pranav | when i hit this button (*,#)it says cannot find extension context=[from-sip] |
| 09:29.50 | Zeeek | pranav if you think your * config is ok, try asking on the freeworldialup.com forum |
| 09:29.58 | Zeeek | who says that? |
| 09:30.01 | Zeeek | CLI? |
| 09:30.11 | pranav | yes |
| 09:30.15 | pranav | in the cli |
| 09:30.21 | Zeeek | try dialing 1234567 |
| 09:30.30 | Zeeek | and tell me what the CLI outputs |
| 09:31.22 | pranav | nothi8ng comes on the cli when i dial 1234567 |
| 09:31.47 | Zeeek | ok now try 2000# or * |
| 09:31.47 | pranav | should i paste my extensions.conf on pastebin |
| 09:31.52 | Zeeek | not yet |
| 09:32.08 | pranav | ok |
| 09:32.41 | pranav | still nothing comes |
| 09:33.20 | pranav | actually when i press 2 at that time only some other tone starts to ring |
| 09:33.49 | Zeeek | anyone answer? |
| 09:35.15 | pranav | it shows this error pbx.c:1335 pbx_extension_helper: cannot find extension context '[from-sip]' |
| 09:35.29 | pranav | this is exact what it shows |
| 09:35.35 | Zeeek | so it is telling you that it doesn't find the context |
| 09:35.59 | Zeeek | is there a [from-sip] context in extensions.conf ? |
| 09:36.18 | pranav | yes there is |
| 09:36.33 | Zeeek | since the last restart or reload extensions command? |
| 09:36.41 | mAsH` | onyone can help me pls? |
| 09:36.45 | Zeeek | show dialplan from-sip |
| 09:36.59 | pranav | its there in extensions.conf as well as sip.conf |
| 09:37.00 | Zeeek | people with ticks in the pseudo never get help :) |
| 09:37.18 | Zeeek | pranav SHOW DIALPLAN from-sip |
| 09:37.59 | pranav | i dint get you mr.Zeeek |
| 09:38.07 | Zeeek | type that command |
| 09:38.21 | pranav | ok |
| 09:38.40 | Zeeek | does it show the extensions for 2000 and 2001 ? |
| 09:39.09 | pranav | no it says there is no existence of 'from-sip' context |
| 09:39.26 | Zeeek | and what do you suppose that means? |
| 09:39.30 | mAsH` | i get this message just i try to start * |
| 09:39.49 | Zeeek | try typing extensions reload and then the show dialplan again |
| 09:39.59 | mAsH` | root@lite:/etc/asterisk# asterisk -vvvvgc |
| 09:39.59 | mAsH` | Illegal instruction (core dumped) |
| 09:39.59 | mAsH` | root@lite:/etc/asterisk# |
| 09:40.13 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
| 09:40.14 | pranav | ok |
| 09:40.17 | xpasha | hi |
| 09:40.34 | pranav | ok i did that but again it shoiws the same thing |
| 09:40.36 | Zeeek | pranav if that works you really did put from-sip context in |
| 09:40.47 | xpasha | who could say why asterisk queue don't write short calls? |
| 09:40.55 | xpasha | i mean monitor function |
| 09:40.56 | Zeeek | ah then you do NOT have the context in extensions.conf - maybe you are working in the wrong .conf file? |
| 09:41.35 | pranav | no in my extensions.conf i have put context=[from-sip] |
| 09:41.39 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 09:41.52 | Zeeek | ah. PLease go read this: |
| 09:42.03 | Zeeek | http://asteriskdocs.org |
| 09:42.21 | pranav | ok is there a serious error |
| 09:42.25 | Zeeek | go to the site, find "read online as one HTML page" and search for context |
| 09:42.37 | pranav | ok |
| 09:42.39 | Zeeek | I can't help you if you won't go read that document |
| 09:42.46 | Zeeek | see you later |
| 09:42.57 | pranav | fine i'll read this document and then talk to you |
| 09:43.00 | datareactor | i get SIP Status: 405 Method Not Allowed when trying to reg to go2call |
| 09:43.43 | Zeeek | pranav and others: please make a bookmark to http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html |
| 09:44.20 | Zeeek | Go to above page and search for dialplan |
| 09:44.37 | Zeeek | "The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.... " |
| 09:44.58 | Zeeek | The beauty and verity of that phrase makes it eternal |
| 09:46.39 | djin | Spoken like a true profet, Zeeek. |
| 09:46.47 | Zeeek | my new macro |
| 09:46.49 | Zeeek | "The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. " |
| 09:46.49 | Zeeek | Get http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html |
| 09:46.49 | Zeeek | Search for "dialplan" |
| 09:47.19 | Zeeek | I think I'll refine that URL to http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 09:47.27 | Zeeek | that way they don't even have to search |
| 09:48.12 | djin | CTRL-F on the page itself ;) |
| 09:48.29 | Zeeek | yes that's what I usually do, but there is that direct anchor |
| 09:48.37 | djin | ah, ok. |
| 09:49.00 | Zeeek | Nothing is more irritating that hearing "I did read the docs" when the question is covered in simple language there |
| 09:49.29 | Zeeek | now if the question about about polarity reversal or something that's different |
| 09:49.45 | djin | include PM in annoying behavior. |
| 09:49.52 | Zeeek | ya |
| 09:49.59 | Zeeek | that's why I have queries disabled |
| 09:50.10 | Mavvie | (somebody here with a linux version of x-lite for download?) |
| 09:50.24 | djin | yeah, was looking for that as well. |
| 09:50.27 | Zeeek | is there one? |
| 09:50.36 | djin | not sure if it violates beta-licence |
| 09:50.48 | Mavvie | there has been one. |
| 09:50.49 | djin | yes, Zeeek. There is a beta. |
| 09:51.03 | djin | Read about release around 14 Feb. |
| 09:51.05 | Zeeek | wow that'd be slick |
| 09:51.38 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 09:52.10 | djin | Zeeek, do you have experience with manager.conf? |
| 09:52.29 | Zeeek | a little - I think we already discussed this |
| 09:52.51 | djin | perhaps, but I learned in the meantime ;) |
| 09:53.13 | Zeeek | no I mean I think your question wasn't within my grasp :) |
| 09:53.25 | djin | No, different question. |
| 09:53.37 | Zeeek | well whatever it is, let's get it out of the bag! |
| 09:54.18 | djin | Is it correct that a manager is nog bound to a context and therefor it's possible he has more rights that you would like him to? |
| 09:54.23 | djin | nog=not |
| 09:54.45 | jerlique | Hi - can I use a cisco 1600 as a "channel bank" |
| 09:54.45 | Zeeek | I think the manager can do anything, ya |
| 09:54.51 | djin | Wanted to handle managers the same as SIP users. |
| 09:55.01 | djin | AstTAPI is too powerfull this way. |
| 09:55.22 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 09:55.27 | Zeeek | I use manager to send commands on the web server that's all |
| 09:55.49 | djin | If one sets up a 'from' and 'to' to external numbers, only the 'to' gets logged. |
| 09:56.09 | *** join/#asterisk shaZwaz (~lukyali@203.81.196.167) |
| 09:59.04 | *** join/#asterisk olivier_ (~olivier_@AMarseille-152-1-1-32.w82-127.abo.wanadoo.fr) |
| 09:59.04 | Zeeek | sorry I can't answer that one |
| 09:59.41 | olivier_ | morning ppl :) |
| 10:00.01 | djin | Zeeek, thanks. It's pretty specific. |
| 10:00.49 | fa | any one records a call to file? |
| 10:01.47 | Zeeek | olivier_ that was a short stay |
| 10:04.35 | fa | how can i record to mp3 files |
| 10:04.36 | fa | Feb 1 11:04:19 ERROR[10419]: format_mp3.c:299 mp3_rewrite: I Can't write MP3 only read them. |
| 10:04.39 | fa | ? |
| 10:05.06 | djin | fa, * can only decode MP3 |
| 10:05.17 | tzafrir | write to a different format and convert? |
| 10:07.06 | tzafrir | why, a patent issue? |
| 10:07.07 | fa | gsm is good format? |
| 10:08.20 | djin | tried wav and covert to mp3 using LAME. Result a 'metallic' sound. |
| 10:08.33 | djin | I stopped there ;) |
| 10:08.35 | tzafrir | generally , yes. Natually if you create a file you need to be able to use it |
| 10:09.23 | makamani | can i have passwords to dial certain extensions and record dtmf tones |
| 10:09.31 | makamani | meaning can i setup passwords... |
| 10:10.30 | fa | in which variable i have a id of call - unique ? |
| 10:10.39 | fa | i will re3cording to gsm for now |
| 10:11.34 | djin | makamani, Authenticate(1234) ?? |
| 10:12.23 | djin | fa, ${UNIQUEID} ?? |
| 10:12.33 | makamani | ok djin, will try |
| 10:13.20 | fa | good |
| 10:13.21 | fa | exten => _8.,1,Monitor(gsm,/work/www/htdocs/inez/mp3/${TIMESTAMP}/${UNIQUEID},m) |
| 10:13.22 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 10:13.27 | djin | makamani, it ask for a code and continues (1234) and continues when correct. |
| 10:13.40 | kks | Zeeek, do u know how to handle incoming SIP call with IP (no exten), how is the dialplan look like? |
| 10:13.43 | makamani | thank you djin |
| 10:14.07 | Zeeek | kks I don't understand your quest |
| 10:14.42 | djin | fa, uniqueid is build up with a timestamp. |
| 10:14.56 | djin | might help . . |
| 10:15.04 | Zeeek | you mean name@yourserver.com ? |
| 10:15.46 | kks | i wan to allow another SIP server machine able to make use of my zap channel without given them an extension |
| 10:16.08 | Zeeek | they have to have a context |
| 10:16.23 | Zeeek | so they have an entry in sip.conf yes? |
| 10:16.29 | kks | no. |
| 10:16.48 | kks | they are not asterisk |
| 10:17.00 | Zeeek | ZAP channel for calling where? On PSTN? |
| 10:17.05 | kks | yup |
| 10:17.15 | Zeeek | with no authentication? |
| 10:17.18 | kks | yes |
| 10:17.26 | *** join/#asterisk mutombo_ (~muto@reverse-213-146-112-84.dialin.kamp-dsl.de) |
| 10:17.28 | Zeeek | so you want to run your pOTS line open on the net? |
| 10:17.30 | kks | maybe filter on ip? |
| 10:17.36 | Nix | kks: I suggest you dont tell anyone your IP... |
| 10:17.42 | Nix | except for me :-D |
| 10:17.43 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 10:17.49 | djin | kks, what's your ip? |
| 10:17.57 | Zeeek | kks you'll have to set up a peer or friend in sip.conf |
| 10:18.14 | Zeeek | host=ip.adr.ess.here |
| 10:18.29 | Zeeek | that will allow one host to connect |
| 10:18.52 | kks | Zeeek, ok |
| 10:18.59 | Zeeek | "The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 10:19.42 | djin | ah, the bible again. |
| 10:20.02 | djin | the one book thans answers almost all questions ;) |
| 10:20.20 | Zeeek | "The first thing tht needs to be done is setup the general settings. Much like IAX this allows you to make settings that all sip connections will use." http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN607 |
| 10:20.30 | Zeeek | hey a typo! |
| 10:23.34 | Zeeek | djin actually it used to answer more questions but they changed the approach to be more clear with a more limited scope |
| 10:23.59 | djin | nobody is reading docs anyway. |
| 10:24.18 | djin | why read when you can PM? |
| 10:24.40 | djin | just here to server. |
| 10:24.42 | djin | serve. |
| 10:25.24 | Zeeek | It's sometimes hard to determine whether it's bloody laziness or a person has a real problem reading and understanding English |
| 10:25.42 | Zeeek | ONe of the guys that wrote asteriskdocs.org is very nice to beginners on the ML |
| 10:26.13 | Zeeek | he feels (and I agree) that asterisk should welcome all necwcomers into the community and try to help them as much as we can. BUT: |
| 10:26.27 | Zeeek | They do have to make the effort to read the first docs |
| 10:26.43 | Zeeek | I have my list distilled down to about four great ones |
| 10:27.06 | djin | I agree to that. No need to explain the basics, that is very well covered in docs. |
| 10:27.50 | Zeeek | Questions about what hardware do I need oth should be answered by those experienced with the situation. I would always try to help people who want to do what I have successfully done, for example |
| 10:27.53 | fa | good exten => _8.,1,Monitor(gsm,/work/www/htdocs/inez/mp3/${TIMESTAMP}/${UNIQUEID},m) |
| 10:27.56 | fa | ? |
| 10:27.58 | djin | * takes time, but for me is quite rewarding so far. Hence hosting a mailing search and help a little here. |
| 10:28.05 | mutombo_ | i have a little understandingquestion, when i want to connect analog phones to asterisk, i need some of these Wildcard TDM400P? |
| 10:28.22 | djin | mutombo, yes |
| 10:28.24 | Zeeek | fa - besides getting aon a lot of folks nerves - your directory TIMESTAMP exists? |
| 10:28.27 | mutombo_ | what when i need more than 40 phones connected? |
| 10:28.38 | djin | use FXS channel banks. |
| 10:28.39 | fa | Zeeek i need no timestamp.. i need that variable - 20040201 ? |
| 10:28.41 | fa | which is it? |
| 10:28.46 | Zeeek | mutombo_ ah non that won't give you enough lines |
| 10:29.10 | Zeeek | fa remove ${TOIMESTAMP}/ |
| 10:29.19 | mutombo_ | someone know how many of this modules i can plug onto this cards? |
| 10:29.30 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 10:29.33 | Zeeek | remove that and the slash at the end (trailing slash) |
| 10:29.35 | djin | TDM? -> 4. |
| 10:29.38 | fa | Zeeek and what add? i want to save gsm in folder of that day |
| 10:29.46 | mutombo_ | k thx |
| 10:29.55 | Zeeek | don't add anything fa just try what I said and report backif success |
| 10:30.14 | fa | Zeeek ${TIMESTAMP:8} - sth like that.. by i want first 8 char.. not from 8 char |
| 10:30.19 | djin | mutombo, mind the difference between FXO and FXS. |
| 10:30.23 | Zeeek | mutombo if you want that many phones you may want to use SIP pones? |
| 10:30.37 | fa | Zeeek what try? |
| 10:30.46 | Zeeek | fa ? |
| 10:30.55 | mutombo_ | its about switching an existing office the phones and lines are all there already |
| 10:30.55 | Zeeek | fa read |
| 10:30.58 | Zeeek | fa read this |
| 10:31.04 | mutombo_ | but thats a good idea to start new :) |
| 10:31.18 | Zeeek | fa read this line: REMOVE the TIMESTAMP variable and the slash |
| 10:31.36 | djin | mutumbo, connecting 40 analog phones might be a wasted investment. |
| 10:31.42 | fa | Zeeek and only that, right: exten => _8.,1,Monitor(gsm,/work/www/htdocs/inez/call/${UNIQUEID},m) |
| 10:31.53 | Zeeek | mutombo_ yes a common situation - but to collect 40 phones you'd need other hardware, noit my area :) |
| 10:32.02 | djin | functionality is limited and I would advise to investigate VOIP-phones options. |
| 10:32.12 | Zeeek | fa try it and don't speak until you have the results |
| 10:32.33 | fa | result is that file asterisk-11078-1107253926.0.gsm |
| 10:32.34 | djin | but |
| 10:32.46 | fa | but i want to have that file in catlogu with name 200402021 |
| 10:32.52 | fa | 20040201 |
| 10:33.18 | Zeeek | fa what does TIMESTAMP give you ? |
| 10:33.21 | djin | what's 20040201? |
| 10:33.32 | Zeeek | feb 1 2004 :) |
| 10:33.45 | djin | someone needs a date -s |
| 10:33.48 | Zeeek | what format is timestamp someone? |
| 10:33.53 | RaYmAn-Bx | or jan 2 2004 depending on where you are from :P |
| 10:33.54 | fa | Zeeek timestamp give me that 20050201-113337 |
| 10:34.24 | Zeeek | fa ok ${EXTEN:1:8}${}UNIQUEID |
| 10:34.35 | Zeeek | no slash between the two |
| 10:35.02 | Zeeek | and the ${UNIQUEID} written <<<------ like this |
| 10:35.20 | Zeeek | fa ${EXTEN:1:8}${UNIQUEID} |
| 10:35.27 | fa | Zeekand that back me |
| 10:35.42 | fa | ""//work/www/htdocs/inez/call/08001211asterisk-11078-1107254109.6- |
| 10:35.49 | fa | where the number was 0800121121 |
| 10:36.05 | fa | i know |
| 10:36.09 | fa | TIMESTAMP:1:8 |
| 10:36.09 | fa | ;] |
| 10:36.16 | Zeeek | fa I think you need a consultant, honestly |
| 10:36.21 | Nugget | 11078 is probably your unix PID and 1107254109 is the current time in unix epoch format. |
| 10:36.52 | Zeeek | TIMESTAMP must be in unix native - I 've never used it |
| 10:36.58 | Zeeek | ya |
| 10:37.33 | Nugget | it's trivial to transform that epoch format into a human-readable form if that's what you're trying to do. |
| 10:37.42 | Zeeek | in extensions.conf |
| 10:37.44 | fa | reload |
| 10:38.24 | fa | exten => _8.,1,Monitor(gsm,/work/www/htdocs/inez/call/${TIMESTAMP:0:8}/${UNIQUE$ |
| 10:38.28 | fa | it's what i need |
| 10:39.12 | Zeeek | uh hello olivier_ |
| 10:39.27 | Zeeek | so use it |
| 10:40.18 | xpasha | so guys who can tell why asterisk doesnt record calls in queues? |
| 10:40.32 | xpasha | when call is short the file contents header only |
| 10:40.39 | olivier_ | Hi zeek |
| 10:40.58 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 10:41.08 | fishboy1669 | ello |
| 10:41.13 | Zeeek | fishy ! |
| 10:41.20 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:41.49 | fishboy1669 | hi zeek |
| 10:41.53 | fishboy1669 | hows thinks |
| 10:42.05 | Zeeek | gut |
| 10:42.16 | fishboy1669 | das ist |
| 10:42.22 | djin | sehr gut? |
| 10:42.27 | fa | I have a little echo with call on ZAP. |
| 10:42.29 | fa | how to delete that |
| 10:42.50 | Zeeek | fa would you consider doing some research on your own ? |
| 10:43.07 | fa | Zeeek I was findinf, but i didn't found |
| 10:43.10 | djin | if it doesn't take too much of your time . . . |
| 10:43.22 | Nugget | ich bin schlafrig. |
| 10:43.22 | mAsH` | i get always the same message when i try to start * |
| 10:43.25 | mAsH` | [res_adsi.so] => (ADSI Resource) |
| 10:43.25 | mAsH` | <PROTECTED> |
| 10:43.25 | mAsH` | Illegal instruction (core dumped) |
| 10:43.25 | mAsH` | root@lite:~/asterisk-1.0.5# |
| 10:43.31 | Zeeek | there are 20 PAGES of echo cancellation stuff on the wiki |
| 10:43.35 | mAsH` | anyone known help me? |
| 10:43.36 | djin | fa, come on. 4 minutes search? |
| 10:44.07 | Zeeek | mash - you tried noload= on siome modules? |
| 10:44.17 | fishboy1669 | fa |
| 10:44.18 | djin | fa, don't PM me. |
| 10:44.20 | olivier_ | <mAsH`> you really need adsi ? |
| 10:44.30 | fishboy1669 | zapata.conf |
| 10:44.32 | mAsH` | no |
| 10:44.33 | Zeeek | fa I understand you are not perfect with English but there is a limit to the help you can expect here |
| 10:44.38 | fishboy1669 | echotraining=yes |
| 10:44.42 | mAsH` | i alredy removed it |
| 10:44.52 | mAsH` | and i get the same errore later |
| 10:44.52 | fishboy1669 | echocancancel=tes |
| 10:44.54 | fishboy1669 | yes |
| 10:45.01 | fishboy1669 | rxgain= |
| 10:45.03 | fa | I have that echotraining=800 |
| 10:45.03 | Zeeek | mash I'm trying to rememebr when I had that probelm... |
| 10:45.05 | fishboy1669 | txgain= |
| 10:45.14 | mAsH` | tnx Zeeek ;) |
| 10:45.18 | djin | mAsH, I don't think it has anything to do with adsi. |
| 10:45.22 | fishboy1669 | make sure u put them in that order |
| 10:45.27 | fishboy1669 | tweek your gains |
| 10:45.33 | Zeeek | fishboty is the resident echo reduction expert here - he will solve your problem |
| 10:45.35 | fishboy1669 | put max one way then max other |
| 10:45.38 | Zeeek | fishbooty |
| 10:45.39 | mAsH` | so do i djin |
| 10:45.41 | fishboy1669 | lol |
| 10:45.48 | fishboy1669 | took me 3 days lol |
| 10:45.55 | fa | rxgain=0.7 |
| 10:45.55 | fa | txgain=0.7 |
| 10:45.57 | fa | good? |
| 10:46.03 | fishboy1669 | im 0.0 |
| 10:46.05 | Zeeek | fish you solved it? Go tell the 20,000 posters on the ML how! |
| 10:46.06 | fishboy1669 | 3.0 |
| 10:46.15 | fa | rxgain=0.0 |
| 10:46.15 | fa | txgain=3.0 |
| 10:46.16 | fa | ? |
| 10:46.21 | fishboy1669 | yes |
| 10:46.29 | fishboy1669 | main thing is the order |
| 10:46.37 | Zeeek | oh? |
| 10:46.49 | fishboy1669 | make sure u put the signalling=fxs_ks at the end of the file |
| 10:46.54 | fa | now it's not working |
| 10:47.07 | Zeeek | hmmmmmm |
| 10:47.07 | fishboy1669 | keep tweekeng |
| 10:47.07 | fa | signalling = bri_cpe_ptmp |
| 10:47.08 | fa | is bad? |
| 10:47.21 | Zeeek | drums stop. Bad. |
| 10:47.31 | fishboy1669 | anyone know about using a sip to h323 converter? |
| 10:50.53 | *** join/#asterisk meppl (~mephisto@p54853A08.dip.t-dialin.net) |
| 10:51.55 | Nix | don't even think about it with asterisk |
| 10:52.42 | Nix | check out yate for that.. it has very stable h323 support, reasonable SIP and can do rtp sip-h323 bypass ;-) |
| 10:54.02 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 10:54.08 | fishboy1669 | cheers nix |
| 10:56.12 | *** join/#asterisk shaZwaz (~lukyali@203.81.196.167) |
| 10:56.20 | shaZwaz | ~ping |
| 10:56.21 | jbot | pong |
| 10:56.45 | tzanger | ping..... KABOOM! |
| 10:57.33 | shaZwaz | Zeeek how can I load values from astdb at startup/reload |
| 10:57.33 | lters | any likely date of the next cvs to stable date? |
| 10:58.23 | Mavvie | "when it's ready" |
| 11:00.40 | lters | and, when will that be :) |
| 11:01.53 | shaZwaz | maybe drumkilla can tell |
| 11:05.28 | *** join/#asterisk pranav (dawda_pran@203.115.89.135) |
| 11:06.04 | pranav | hi |
| 11:07.11 | pranav | hi |
| 11:07.28 | shaZwaz | hi hi |
| 11:08.43 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.11) |
| 11:09.29 | datareactor | can anyone check what wrong with my config http://pastebin.ca/5070 unable to register to go2call |
| 11:09.47 | djin | anyone Realtime experience icw MySQL? |
| 11:10.43 | pranav | hi shazwas |
| 11:10.57 | pranav | ca we talk |
| 11:11.39 | djin | never mind, got my answer. |
| 11:11.50 | fa | can any give my his IAX2.. |
| 11:11.52 | shaZwaz | pranav: abt ? |
| 11:11.53 | fa | i will call for test |
| 11:12.33 | olivier_ | <fa> you have a digium iax sample in your extension.conf |
| 11:12.51 | Zeeek | datareactor |
| 11:13.03 | fa | olivier_ i want to call some alive |
| 11:13.07 | fa | anyo ne use zaphfc? |
| 11:13.13 | olivier_ | fa something like that : Dial(IAX2/guest@misery.digium.com/s@default) |
| 11:13.47 | fishboy1669 | is this still current? |
| 11:13.49 | fishboy1669 | http://www.voip-info.org/tiki-print.php?page=Asterisk+Download |
| 11:13.56 | fishboy1669 | as the latest stable release |
| 11:14.08 | Mavvie | what does it say there? |
| 11:16.17 | fishboy1669 | last modification: Monday 20 of December, 2004 |
| 11:16.30 | fishboy1669 | for the webpage |
| 11:16.37 | fishboy1669 | just checking to see if that is current |
| 11:17.52 | Delmar | ffs this X100 echo is still driving me nuts. and now to top it all off.. the busy/disconnect tone detection seems to be intermittant... so if someone say.. rings and leaves a message on voicemail then hangs up, the X100 wont hangup. |
| 11:19.21 | djin | Does anyone use Asterisk Realtime? |
| 11:19.34 | djin | just a quick question. |
| 11:22.35 | Zeeek | not that quick apparently |
| 11:22.43 | Zeeek | coffee, anyone? |
| 11:22.48 | olivier_ | yep ! |
| 11:23.00 | Zeeek | nespresso! |
| 11:23.09 | olivier_ | one sugar please :) |
| 11:23.18 | djin | ah, Zeek was waiting for a response ;) |
| 11:23.22 | tzafrir | anything I can download? |
| 11:23.59 | djin | I use * CVS (v1-0) and res_mysql is missing in asterisk-addons. |
| 11:24.22 | djin | I downloaded latest CVS and there it is! |
| 11:24.34 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 11:24.39 | djin | Can this version be use with v1-0? |
| 11:24.56 | fa | i have that error |
| 11:24.57 | fa | Feb 1 12:24:36 ERROR[6988]: chan_zap.c:6416 mkintf: Unable to open channel 1: No such device or address |
| 11:25.01 | fa | here = 0, tmp->channel = 1, channel = 1 |
| 11:25.03 | fa | Feb 1 12:24:36 ERROR[6988]: chan_zap.c:10063 setup_zap: Unable to register channel '1-2' |
| 11:25.07 | fa | Feb 1 12:24:36 WARNING[6988]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 11:25.09 | fa | Feb 1 12:24:36 WARNING[6988]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 11:25.13 | fa | but i have loaded zaphfc module |
| 11:25.34 | mAsH` | fa |
| 11:25.38 | fa | mAsH` |
| 11:25.40 | mAsH` | ztcfg -v |
| 11:25.50 | fa | SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) |
| 11:25.50 | fa | 3 channels configured. |
| 11:25.54 | *** join/#asterisk festr_ (~festr@ns.regnet.cz) |
| 11:25.56 | mAsH` | try now |
| 11:25.57 | festr_ | hello |
| 11:26.06 | fa | 3 channels configured.thanks |
| 11:26.07 | fa | now good |
| 11:26.28 | mAsH` | ;) fa |
| 11:26.58 | mAsH` | i have always the same problem :/ |
| 11:27.08 | mAsH` | Illegal instruction (core dumped) |
| 11:27.14 | mAsH` | any idea? |
| 11:28.09 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 11:28.49 | thieumS | which addon do you propose to perform prepaid calls (mysql) |
| 11:30.52 | festr_ | pls help. i'm installing some next 3xFXS 1xFXO pci TDM-400p, cat /proc/zaptel/1 1,2,3WCTDM/0/0...4 WCTDM/0/3 FXSKS. ztcfg is OK. when loading asterisk: Unable to open channel 1: No such device |
| 11:30.56 | Delmar | mAsH`.. when exactly do u get a core dump? |
| 11:31.12 | festr_ | signalling=fxo_ks channel=>1 |
| 11:31.33 | Delmar | festr signalling for FXO is reverse. |
| 11:31.34 | mAsH` | just i try to start asterisk |
| 11:31.59 | mAsH` | there are the default conf file :/ |
| 11:32.00 | Delmar | mAsH` what kernel? |
| 11:32.04 | mAsH` | 1.0.5 |
| 11:32.10 | mAsH` | 2.4.26 |
| 11:32.17 | Delmar | hrm. |
| 11:32.30 | Delmar | i run 2.6 kernels myself but still.. |
| 11:32.36 | mAsH` | VIA C3 processor on mini-itx |
| 11:32.44 | festr_ | Delmar: Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 11:32.44 | Delmar | so you downloaded latest CVS? |
| 11:32.49 | mAsH` | yes |
| 11:32.53 | festr_ | Delmar: this is output from ztcfg |
| 11:33.10 | festr_ | Delmar: signalling=fxo_ks channel=>1 |
| 11:33.21 | *** part/#asterisk Nix (~Nix@81.213.125.220) |
| 11:33.23 | Delmar | channel 1 is your FXO port? |
| 11:33.46 | festr_ | Delmar: no 1-3 is FXS |
| 11:34.12 | festr_ | Delmar: /etc/zaptel.conf |
| 11:34.12 | festr_ | fxsks=4 |
| 11:34.13 | festr_ | fxoks=1-3 |
| 11:34.34 | Delmar | heh. yet, ztcfg -v shows what? :P |
| 11:35.41 | festr_ | Delmar: 4 channels configured. |
| 11:35.53 | Delmar | Channel 01: FXO Kewlstart (Default) (Slaves: 01) ...channel 1 = FXO |
| 11:35.59 | festr_ | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 11:36.29 | festr_ | sorry channel 1 = FXS (for phones) |
| 11:36.38 | festr_ | so ztcfg is right |
| 11:36.42 | festr_ | it is reverse |
| 11:36.56 | festr_ | i have the same configuration on other machine and there is no problm |
| 11:37.01 | Delmar | yep. |
| 11:37.25 | festr_ | but when starting asterisk and chan_zap.so, Feb 1 12:36:41 WARNING[2259]: chan_zap.c:769 zt_open: Unable to specify channel 3: No such device |
| 11:37.29 | Delmar | ok, so channels 1-3 are not loading but channel 4 is? |
| 11:37.33 | festr_ | channel 1 and 2 too |
| 11:37.38 | festr_ | Delmar: yes |
| 11:37.52 | festr_ | Delmar: but proc/zaptel/1 is right, ztcfg show no errors |
| 11:38.25 | Delmar | in /etc/zaptel.conf .. what do u have? |
| 11:38.34 | Delmar | use pastebin.com if need be. |
| 11:39.32 | MuchToDo | Anybody here in the UK? |
| 11:39.44 | Delmar | at this time of year? no way.. lol |
| 11:39.53 | MuchToDo | :) |
| 11:40.18 | MuchToDo | I'm looking around for VOIP hardware suppliers, but not finding much. |
| 11:40.47 | MuchToDo | Looking at ebay.com, there's loads of zaptel cards (and clones) around, but not much on ebay.co.uk |
| 11:41.00 | MuchToDo | I've found a couple of online stores that look ok, but the prices seem fairly high.. |
| 11:41.01 | libpcp | what does it mean by Warning, flexibel rate not heavily tested! |
| 11:41.31 | InfraRed | means the rates are not tested, heavily |
| 11:41.45 | fishboy1669 | hi guys |
| 11:42.05 | fishboy1669 | anyone got there * box behind a nat firewall and tried to connect from external using sip? |
| 11:42.12 | fishboy1669 | how do i do it? |
| 11:42.45 | MuchToDo | fishboy1669: i'm trying that at the moment |
| 11:43.08 | MuchToDo | not sure if it's the firewall at home or the one at the external site which is causing a problem. |
| 11:43.09 | fishboy1669 | how is it going? |
| 11:43.16 | festr_ | Delmar: fxoks=1-3 and fxsks=4 |
| 11:43.21 | MuchToDo | not sure really - only just started! |
| 11:43.34 | MuchToDo | still setting up port forwards etc |
| 11:43.47 | fishboy1669 | the issue i found was that the outside device gets given the internal ip address of the * box and therefore cant route back |
| 11:43.51 | MuchToDo | looks like it will be easier with IAX2 than with SIP |
| 11:43.56 | fishboy1669 | need it with sip |
| 11:44.47 | fishboy1669 | hint from my setup use nat=yes canreinvite=no qualify=10000 in sip.conf |
| 11:45.10 | fishboy1669 | and set up 5060 portforward in the fw to the * box ip |
| 11:45.22 | MuchToDo | need to try mine from an external site which isn't firewalled before I can say if it's working or not |
| 11:45.26 | fishboy1669 | the sip bit works and the phones connect but there is no sound |
| 11:45.32 | MuchToDo | but i'm betting i'll get the same problem :) |
| 11:45.32 | Delmar | festr_ so u jsut have like.. 4 lines... loadzone, defaultzone, fxsks= and fxoks= ? |
| 11:45.52 | fishboy1669 | im at work behind silly fw here so cant test with u sorry |
| 11:46.05 | fishboy1669 | anyone else any experience with sip via nat |
| 11:46.07 | fishboy1669 | ??????? |
| 11:46.30 | Delmar | yep fishboy1669. |
| 11:46.52 | Delmar | does the firewall let u connecy outbound to any destination addy/port ? |
| 11:46.59 | festr_ | Delmar: exactly |
| 11:47.19 | Delmar | festr_ ok ... gotta be /etc/asterisk/zapata.conf then dude.. |
| 11:47.41 | Delmar | whats the output of ztcfg -vv ? |
| 11:47.49 | Delmar | it lists all 4 ports as they should be right? |
| 11:48.18 | festr_ | Delmar: eys |
| 11:48.19 | festr_ | yes |
| 11:48.30 | festr_ | Delmar: ztcfg is right, /proc/zaptel/1 is right too |
| 11:49.00 | fishboy1669 | delmar can u explain further |
| 11:49.04 | festr_ | Delmar: zapata.conf is signalling=fxo_ks |
| 11:49.04 | festr_ | channel=>1 |
| 11:49.13 | fishboy1669 | it lets me portforward external port to internal ip |
| 11:49.37 | Delmar | fishboy1669 nah u dont need any port forwarding really. |
| 11:49.58 | Delmar | what SIP client are u using.. hardware or software? |
| 11:50.17 | *** join/#asterisk waddy (waddy@66.90.92.190) |
| 11:50.31 | fishboy1669 | both |
| 11:50.42 | fishboy1669 | xlite for testing but also ip [ |
| 11:50.44 | fishboy1669 | phone |
| 11:50.50 | Delmar | I have a solution here that is working.. like... xlite SIP client ===> Nat FW Router ===> Internet ====> Nat Router ===> Asterisk. |
| 11:50.51 | fishboy1669 | delmar how do i do it then |
| 11:51.08 | fishboy1669 | yay thats exactly what i want |
| 11:51.12 | Delmar | but when i replace the xlite end with say.. a budgetone 101/102... no go. |
| 11:51.41 | fishboy1669 | aha thats cos xlite has nat fw delection |
| 11:51.41 | Delmar | fishboy1669 you need to use stun/proxy etc. |
| 11:51.57 | Delmar | yeah xlite seems to work mint. |
| 11:52.04 | Delmar | budgetone are budget. |
| 11:52.08 | festr_ | Delmar: i'm stupid |
| 11:52.13 | festr_ | Delmar: i didnt plug the power |
| 11:52.19 | festr_ | LOL :) |
| 11:52.31 | fishboy1669 | oh does that mean i need a server with external ip for the stun |
| 11:52.32 | fishboy1669 | ? |
| 11:52.33 | Delmar | power to what dude? lol |
| 11:52.45 | fishboy1669 | my phones i can set up stun but dont want more servers |
| 11:53.09 | mAsH` | Delmar:always the same error, i have also recompiled :/ any ideas? |
| 11:53.31 | festr_ | Delmar: power to TDM400P |
| 11:53.46 | festr_ | Delmar: it has external power |
| 11:53.57 | Delmar | oh, the main trick with the Nat and Asterisk.. is that the Asterisk box needs to be like.. DMZ'd for calls in both directions to work nice.. that is... ie. an Alcatel you set a "defserver" pointing to the Asterisk. |
| 11:54.02 | fishboy1669 | delmar how do i get the phone to transvers the nat? |
| 11:54.41 | fishboy1669 | my * has internal ip and portforward 5060 from the external of the nat to the internal * ip |
| 11:54.43 | Delmar | xlite or something else fishboy1669? cuz i know xlite works.. and i know budgetone wont... anddont know anything more... |
| 11:54.59 | fishboy1669 | i have tried with xlite |
| 11:55.08 | fishboy1669 | the call connects but no sound |
| 11:55.14 | fishboy1669 | do u use stun server |
| 11:55.19 | Delmar | well, port forwarding to the Asterisk is pointless because you are going to DMZ to the Asterisk box, so ALL ports will forward to it. |
| 11:55.26 | fishboy1669 | or is your * box on dmz or what |
| 11:55.56 | fishboy1669 | ok so all ports go to the * box but the * box is still registering with the internal ip |
| 11:56.01 | Delmar | or, you can just port forward udp/5060/4569/5004 etc, and not do DMZ... and u will get calls in one direction but notht he other.. ( cant remember which way it was ). |
| 11:56.23 | fishboy1669 | so when a call is made the external phone tryes to return packets to an internal ip address and routing wont allow this |
| 11:56.33 | fishboy1669 | how do i get round this |
| 11:56.44 | Delmar | oh there was something else that i played with.... |
| 11:56.45 | fishboy1669 | does your * box have internal ip |
| 11:56.57 | fishboy1669 | aha that sounds like the key what was it |
| 11:56.58 | Delmar | in my sip.conf i set externip = mythingie.dyndns.org |
| 11:57.12 | fishboy1669 | aha bingo that sounds like what i need |
| 11:57.17 | fishboy1669 | cheers ill read up on it |
| 11:57.19 | Delmar | yep. could be. |
| 11:57.29 | Delmar | google is your friend :P |
| 11:57.34 | Delmar | hey but ill tell you what.... |
| 11:57.56 | Delmar | after all my fuckin around with NAT.. and getting REALLY anoyed at it.. there are a couple of things I have decided.... |
| 11:58.39 | Delmar | 1. IAX2 rules.. so biff any softphone that uses SIP out the window... IAX2 and NAT get along nice... 2. Get a better DSL modem or whatever.. and get your linux box on a public IP running Asterisk.... and no more headaches. :P |
| 11:59.34 | fishboy1669 | but if the * is on public ip how do i set up the internal phones to access it? |
| 12:02.25 | thetalon | fishboy, use SER with the NAT proxy |
| 12:02.36 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:02.48 | thetalon | it will rewrite SIP headers for your NATed clients |
| 12:03.32 | *** join/#asterisk Underground (~undergrou@202.147.174.177) |
| 12:03.49 | *** join/#asterisk meppl (~mephisto@p54853A08.dip.t-dialin.net) |
| 12:03.51 | fishboy1669 | but then i need an extra server im trying to get out of needing that |
| 12:04.00 | fishboy1669 | chees for the though thetalon |
| 12:04.06 | Delmar | fishboy1669 im not sure what u are meaning there... umm... if there are clients on a remote network behind a nat device... and your * is public.. they will have less issues than if the * box is also behind nat... |
| 12:04.10 | fishboy1669 | thought though |
| 12:04.16 | Underground | I'm in SSH....how do I run Xwindow from SSH? |
| 12:04.39 | Delmar | and if the * box has a public IP on one interface, and internal address on the other.. and is acting as your firewall/router AND Asterisk... your internal devices will have no issues. |
| 12:05.43 | Delmar | Underground, dude.. that was funny. :P |
| 12:05.51 | Underground | no no Delmar :p |
| 12:06.34 | Underground | my network administrator set my linux to boot from SSH....and I want to in Red Hat Windows |
| 12:06.45 | Delmar | LOL |
| 12:06.50 | Delmar | u are killin me. |
| 12:06.54 | Underground | :D |
| 12:07.13 | Underground | I may sound dump to you b'coz I dont know even the ABC of linux |
| 12:07.20 | Underground | :) |
| 12:07.38 | Underground | come on Delmar. stop enjoying my problem......let me know! |
| 12:07.47 | shaZwaz | Underground: startX |
| 12:07.54 | Underground | let me check. |
| 12:08.20 | Delmar | are u on console or in an SSH session from your windows box? |
| 12:08.27 | OhMyAchinLap | ssh -X |
| 12:08.46 | Delmar | cuz u aint gonna run no Xwindows in an SSH shell application i can tell u now. |
| 12:08.54 | Underground | startx . . works! |
| 12:08.59 | Underground | thanks Shazwaz |
| 12:09.02 | shaZwaz | :) |
| 12:09.03 | *** part/#asterisk Underground (~undergrou@202.147.174.177) |
| 12:09.19 | Delmar | lol. thats messed up. Xwindows in an SSH hahaha. whatever. |
| 12:10.05 | shaZwaz | Underground better read the docs next time |
| 12:10.11 | OhMyAchinLap | heh |
| 12:10.23 | OhMyAchinLap | aren't there some windows ssh clients that have X clients as well? |
| 12:10.45 | Delmar | OhMyAchinLap well, after seeing that... im thinking there must be. lol |
| 12:10.59 | tzafrir | OhMyAchinLap, cygwin has both ssh and X |
| 12:11.03 | OhMyAchinLap | hehe |
| 12:11.08 | Delmar | anyways. im gonna hit the sack and get some Z's and fight with echo cancellation tomorrow. |
| 12:11.08 | tzafrir | http://cygin.com |
| 12:11.12 | OhMyAchinLap | i think he was just confusing a command line with ssh |
| 12:11.14 | tzafrir | http://cygwin.com |
| 12:11.44 | Delmar | cygwin is a pain in the arse. |
| 12:11.46 | Delmar | :P |
| 12:11.48 | Delmar | night. |
| 12:11.52 | fa | how to make simple chack if isdn channel is free, and is not change to other.. befor make dial |
| 12:11.53 | tzafrir | (and it is an X serve, not a client) |
| 12:13.31 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:13.36 | shaZwaz | any idea how values from astdb can be loaded at startup |
| 12:13.45 | puzzled | morning |
| 12:13.54 | shaZwaz | howdy puzzled |
| 12:14.14 | puzzled | shaZwaz: in a dialplan use dbget. don't know how to do it at actual * startup |
| 12:14.31 | shaZwaz | not the dialplan puzzled |
| 12:14.54 | shaZwaz | suppose I wanna intialize a global from astdb |
| 12:15.26 | puzzled | no idea |
| 12:15.35 | shaZwaz | may be some script |
| 12:15.58 | *** part/#asterisk thetalon (~Ari@pcp05736786pcs.norstn01.pa.comcast.net) |
| 12:16.58 | *** join/#asterisk pranav (dawda_pran@203.115.89.135) |
| 12:17.00 | jerlique | can someone tell me what the easiest way to get a incoming call into * (without hardware). I've been tring with iaxtel for days and cannot get it to work! |
| 12:17.34 | fa | I call somebody firefly -> * ->isdn -> cellular phone |
| 12:17.39 | fa | and sb. dont accept my caLL |
| 12:17.48 | fa | why firefly show my.. in call woith.. still |
| 12:20.49 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 12:21.20 | jerlique | Why do I get this error:chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame |
| 12:22.04 | *** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com) |
| 12:23.28 | pranav | hi fa |
| 12:23.33 | puzzled | jerlique: usually google has answers to these kind of answers e.g. http://lists.digium.com/pipermail/asterisk-users/2004-March/040212.html |
| 12:24.07 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
| 12:26.27 | pranav | hello jluk |
| 12:26.34 | pranav | r u there |
| 12:27.48 | jerlique | Yes, I saw that, however I was wondering if there were any advancements.. |
| 12:32.06 | Zeeek | http://willypick.mindsay.com/?date=2005-02-01 |
| 12:32.27 | jerlique | sorry I jumped the gun a little. The email I saw had something different on it.... I'll reread this one. |
| 12:34.49 | Jas_Williams | jerlique: Do you whant a hand to get it working |
| 12:36.24 | *** join/#asterisk X-Gen (~x-gen@rrba-146-67-74.telkomadsl.co.za) |
| 12:37.06 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 12:37.12 | X-Gen | hey all |
| 12:37.43 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 12:38.33 | Jas_Williams | Afternoon X-Gen |
| 12:38.36 | jerlique | yeah, if you dont mind |
| 12:39.18 | X-Gen | Jas_Williams: how did u know it was afternoon ? ;) |
| 12:39.37 | Zeeek | yo Jas |
| 12:39.45 | Jas_Williams | X-Gen: It is here ;-) |
| 12:39.52 | X-Gen | hehehe, same here |
| 12:39.59 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 12:40.01 | Zeeek | samr here! |
| 12:40.58 | Jas_Williams | States must still be asleep lazy not getting up until the middle of the afternoon ;P |
| 12:42.29 | datareactor | Zeeek can you help i am unable to reg to go2call |
| 12:43.01 | Zeeek | Yes |
| 12:43.10 | Zeeek | I tried to answer you before |
| 12:44.05 | datareactor | Zeeek sorry i was disconnected here is my http://pastebin.ca/5070 |
| 12:44.35 | Zeeek | Yes I saw that - but it seems you are timing out. Are you sure the server is up? |
| 12:45.18 | pranav | hello |
| 12:45.31 | datareactor | ZeeeK i can ping it |
| 12:46.56 | Zeeek | are you behind NAT? |
| 12:47.28 | datareactor | Zeeek i using public ips |
| 12:48.01 | fa | I want to writre my own manager of dialplan in PHP + postgres + perl or python to generate dialplans to file |
| 12:48.18 | fa | can anybody propose me some good scratch |
| 12:49.22 | Zeeek | data the config files look right to me |
| 12:49.44 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 12:50.20 | Zeeek | assuming you are unisng the right user/pass of course |
| 12:50.43 | datareactor | Zeeek this error i got through ethereal 216.52.153.209 -> x.x.x.x SIP Status: 405 Method Not Allowed |
| 12:50.43 | Zeeek | but you will want to remove the 1 from the first part I think, no? |
| 12:50.45 | fa | Zeeek can we query for moment? |
| 12:51.05 | Zeeek | no fa I'm a nice boy I do queries |
| 12:51.33 | mutombo_ | have another small question, how do i route calls from the sipphone to my isdnline |
| 12:51.42 | Zeeek | no fa I'm a nice boy I do queries |
| 12:51.43 | mutombo_ | tried something like that: |
| 12:51.44 | Zeeek | plus I seriously want you to go find some documents study until you undertsand them |
| 12:51.45 | mutombo_ | exten => _XXXXXXX,1,Dial(CAPI/${EXTEN}) |
| 12:53.00 | Zeeek | data did you read my last about extension? Do all the extension start with 1? |
| 12:54.52 | datareactor | Zeeek only if person want to dial internationally they dial 1 + country + city code number |
| 12:54.52 | fa | How can i switch call to first free consultant on IAX ? |
| 12:55.20 | Zeeek | fa would you mind leaving say 15 minutes between questions? |
| 12:55.30 | fa | no ;[ |
| 12:55.36 | Zeeek | you are probably being ignored by 19% of the people here now |
| 12:55.42 | Zeeek | 90% |
| 12:55.51 | fa | nooo |
| 12:55.55 | fa | i want to know |
| 12:55.55 | datareactor | :) |
| 12:56.06 | sambal | 89% |
| 12:56.09 | Zeeek | which explains the LACK OF ANSWERS |
| 12:56.23 | cypromis | fa: 1. don't query |
| 12:56.30 | Zeeek | and will soon result in embarassing profaity and vulgarity on the channel |
| 12:56.34 | cypromis | 2. try to use one of the allready existing sql features |
| 12:56.35 | cypromis | :D |
| 12:56.44 | Zeeek | 2. shut the fuck up for 15 minutes |
| 12:56.50 | *** join/#asterisk allgood (~allgood@200-101-232-031.fnsce7004.dsl.brasiltelecom.net.br) |
| 12:56.50 | fa | cypromis nice.. to see you |
| 12:56.55 | allgood | hi all |
| 12:57.06 | sambal | 3. google is your friend, 4. www.voip-info.org also |
| 12:57.07 | cypromis | if you check the makefile in the apps subdirectory |
| 12:57.10 | allgood | i'm still looking for supervised transfer support... |
| 12:57.16 | cypromis | there is a app for making sql requests to postgresql right there |
| 12:57.21 | Zeeek | don't wate your time with links... he ignores them |
| 12:57.22 | cypromis | which you can use out of the dialplan |
| 12:57.27 | cypromis | so you don't need the file in between |
| 12:57.29 | allgood | is there a way to do that when the phone (sip, iax2, or other) supports only blind transfers? |
| 12:57.31 | fa | cypromis z baza daje rade. ale mam troszeczke jeszcze problem z jakoscia polaczenia dzwoniac na ISDN. |
| 12:57.41 | cypromis | hmmm |
| 12:57.42 | kajtzu | in english, please |
| 12:57.46 | cypromis | you are using cheap HFC-S cards |
| 12:57.47 | cypromis | and |
| 12:57.52 | cypromis | you get what you pay for :D |
| 12:58.07 | Zeeek | no better to keep it in Polish :) |
| 12:58.09 | fa | cypromis jeszcze mam taka sprawe. ze jak dzwonie do kogos na komorke, i ktos odrzuci to ja w firefly mam nadal call with.. a przeciez kolo odrczuicl polaczenie. |
| 12:58.14 | sambal | :D |
| 12:58.17 | fa | ok.now english. |
| 12:58.21 | sambal | hmmmm /ignore .... ;) |
| 12:58.35 | Zeeek | done. |
| 12:59.20 | datareactor | Zeeek it seems to me i am not passing correct parameter to go2call 216.52.153.209 -> x.x.x.x SIP Status: 405 Method Not Allowed |
| 12:59.29 | sambal | Zeeek: do you know if somebody is working on that zaptel bug, that you have to reset the module when there is noise on the line? |
| 12:59.42 | Zeeek | if ((question_repeated>=20) && (links_given) ) continue; |
| 12:59.53 | Zeeek | dunno at all sambal |
| 12:59.55 | cypromis | fa: if somebody rejects your call on the mobile network |
| 13:00.01 | cypromis | it is forwarded to voicemail normally no ? |
| 13:00.05 | cypromis | so you are still in a call |
| 13:00.10 | cypromis | and for firefly bugs |
| 13:00.14 | cypromis | contact virbiage |
| 13:00.15 | cypromis | ;) |
| 13:00.20 | allgood | does anybody knows how (if possible, of course) can I make supervised transfers with phones that doesn't have this function? |
| 13:00.30 | allgood | is there a workaround on asterisk for this? |
| 13:00.31 | fa | cypromis no. i have busy signal then on regular phone. |
| 13:00.45 | Zeeek | allgood the new versions will do this I hear |
| 13:00.51 | Zeeek | maybe need HEAD? |
| 13:00.55 | cypromis | allgood: there is a patch |
| 13:00.59 | cypromis | check out bugs.digium.com |
| 13:01.06 | Zeeek | ah - for 1.0.3? |
| 13:01.06 | allgood | cypromis, will look |
| 13:01.17 | cypromis | you meant 1.0.5 I hope |
| 13:01.17 | cypromis | no |
| 13:01.20 | Zeeek | btw is 105 working well now? |
| 13:01.26 | cypromis | but you are welcome to do a backport |
| 13:01.26 | cypromis | :D |
| 13:01.32 | Zeeek | heh, not |
| 13:01.42 | Zeeek | you using 105? |
| 13:01.59 | allgood | cypromis, searched for 'transfer' on bugs.digium.com... 0 results... |
| 13:02.12 | Zeeek | I keep seeing that 1.0.5 has problems |
| 13:02.21 | cypromis | what kind of problems ? |
| 13:02.45 | Zeeek | dialing? |
| 13:02.53 | fa | cypromis why you don't answer on phone... noo |
| 13:03.12 | Zeeek | the word STABLE means to me "this sucker is working perfectly now" |
| 13:03.21 | Makenshi | when i update * (from cvs), how do i stop it renaming all my config files when run make install? |
| 13:03.29 | Makenshi | last time it renamed them all to *.old |
| 13:03.35 | Zeeek | don't make config |
| 13:03.48 | Makenshi | i didnt afair |
| 13:03.56 | Makenshi | oh well i can just back them up |
| 13:04.01 | Zeeek | I've never had that happen - it would be irritating though |
| 13:04.01 | Makenshi | now im prepared |
| 13:04.05 | allgood | cypromis, can you point me directly to the patch that makes this? |
| 13:04.08 | cypromis | fa: I run a company and am most of the time on the phone |
| 13:04.37 | cypromis | makenshi: sounds like you did make samples as well |
| 13:04.46 | cypromis | allgood: hmmm I think it is closed |
| 13:04.48 | Zeeek | ooops I said make config |
| 13:04.52 | fa | cypromis bedziemy mogli jakos pogadac, wiem ze troche mecze, ale musze troche rzeczy ustalic... i juz sobie bede robil samemu dalej. |
| 13:04.52 | cypromis | untag the closed exclusion box |
| 13:04.52 | Zeeek | meant samples |
| 13:04.55 | cypromis | and it will find it |
| 13:05.12 | allgood | transfer |
| 13:05.17 | Makenshi | cypromis, oh, yeah, that must be it :> |
| 13:05.18 | allgood | ops... wrong window... damn mouse |
| 13:07.41 | allgood | cypromis, will it be 'attended pound transfers 2005 style'? |
| 13:08.37 | cypromis | probably |
| 13:08.37 | cypromis | :D |
| 13:08.39 | fa | cypromis hm? |
| 13:09.07 | allgood | cypromis, how will this work with sip and iax2 clients? just press the pound key and the extension to redirect? |
| 13:09.10 | Zeeek | allgood you don like parking? |
| 13:09.35 | cypromis | Zeeek: parking is a feature only US people like |
| 13:09.35 | cypromis | ;) |
| 13:09.45 | allgood | Zeeek, I'm starting a project to sell some asterisk servers, to people that use simple PABX... |
| 13:09.47 | cypromis | everywhere else in this world people are used to assistet transfers |
| 13:10.48 | allgood | unnassisted transfers are possible with clients that doesn't support them (xlite... for instance)? |
| 13:10.50 | Zeeek | I think parking is kinda cool -but never use it) |
| 13:11.10 | allgood | Zeeek, I didn't undesrtood yet what exactly are call parking |
| 13:11.38 | Zeeek | parking is when you allow the Tt in DIal and hit # |
| 13:11.53 | *** join/#asterisk RoyK (~roy@dsl-40-122.kunde.brednett.no) |
| 13:12.05 | Zeeek | it sticks the call at 701 by default and anyone can pick it up |
| 13:12.23 | *** join/#asterisk dktele (~sil@212.130.42.35) |
| 13:12.46 | allgood | Zeeek, then I can call the destination and tell him to pick the parked one? |
| 13:12.47 | Zeeek | so you could fake an attended transfer using parking tho it's a PITA |
| 13:12.53 | Zeeek | yea |
| 13:12.58 | Zeeek | egg act ly |
| 13:13.09 | allgood | Zeeek, any secure option to avoid the wrong person to pick the call? |
| 13:13.22 | Zeeek | jeeze what kind of a l=place do you work? |
| 13:13.25 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 13:14.10 | Zeeek | "pr0n line 701, pr0n..." |
| 13:14.28 | Zeeek | "this is pr0n, how can I help uyou?" |
| 13:14.54 | allgood | Zeeek, didn't understood yet |
| 13:14.59 | allgood | Zeeek, will look at it |
| 13:15.03 | Zeeek | hey how about this? You could fake attended transfer by programming a bunch of extensions |
| 13:15.12 | Makenshi | ive seen parking used in the uk, but rarely |
| 13:15.54 | Zeeek | the main utility of it is when you want to park the call to move somewhere else |
| 13:15.57 | Makenshi | could transfer them to a meetme perhaps |
| 13:16.01 | Makenshi | but then its stuck |
| 13:16.59 | Zeeek | if the patch is for 1.0.5 why is it a patch? Or is that a dumb question? |
| 13:17.11 | Zeeek | since I've heard about it for a while |
| 13:17.44 | allgood | Zeeek, I have another problem... xlite doesn't support transfers... How do I use this with asterisk? |
| 13:18.09 | Zeeek | X-Heavy will do transfers I think |
| 13:18.22 | Zeeek | I believe you can buy it for something like $8 |
| 13:18.33 | darkskiez | allgood, allow the # key to be used for transfers |
| 13:18.47 | allgood | Zeeek, x-heavy... you mean x-pro? |
| 13:18.55 | Zeeek | the only problem with that is if you ever need the # key when on the phone :( |
| 13:18.57 | *** join/#asterisk ZX81 (~ZX81@222-152-158-141.jetstream.xtra.co.nz) |
| 13:19.04 | Zeeek | pro, heavy, same thing :) |
| 13:19.14 | allgood | Zeeek, x-pro is us$ 50 |
| 13:19.25 | Zeeek | although the list price is high, it is on sale somewhere cheap |
| 13:19.35 | Zeeek | now I have to try to rememebr where, right? |
| 13:19.41 | Zeeek | Would you pay under $10? |
| 13:19.42 | allgood | Zeeek, right! |
| 13:19.53 | allgood | Zeeek, my customers will! :-D |
| 13:20.01 | Zeeek | someplace like voxilla I'm guessing |
| 13:20.08 | Zeeek | shit you cust can pay $50! |
| 13:20.22 | Zeeek | that would be per license though |
| 13:20.25 | allgood | Zeeek, but I really liked the "iax phone" soft... but it have some flaws |
| 13:20.57 | allgood | Zeeek, at us$ 50 per extension... they will prefer to keep using their current PABX |
| 13:21.27 | Zeeek | close! : Only US $39 with $10 nikotel credit included. |
| 13:21.41 | allgood | Zeeek, ouch |
| 13:21.46 | Zeeek | how many extensions? |
| 13:21.56 | allgood | Zeeek, above 50 |
| 13:22.31 | allgood | Zeeek, I'll keep a look on 'iax phone' and other iax free soft phones... |
| 13:22.58 | *** part/#asterisk datareactor (datareacto@203.81.192.33) |
| 13:23.15 | Zeeek | IAXPhone is pretty good IMO |
| 13:23.22 | allgood | how do I enable transfers with the pound key? |
| 13:23.52 | allgood | i'm using 1.0.2 (debian package) |
| 13:24.10 | Zeeek | Show application dial |
| 13:24.15 | Zeeek | see T and t |
| 13:24.30 | allgood | Zeeek, on conf files? |
| 13:24.41 | Zeeek | CLI type show application dial and read |
| 13:25.16 | Zeeek | I swore someone on FWD said X-Pro was sold for $10 legitimately somewhere |
| 13:25.32 | Zeeek | prolly with a subscription to some voIP service |
| 13:25.58 | allgood | Zeeek, I saw one that was locked to a sip provider... |
| 13:26.04 | Makenshi | anyone using eyebeam yet? |
| 13:26.09 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:26.44 | Zeeek | <PROTECTED> |
| 13:26.50 | Zeeek | <PROTECTED> |
| 13:27.03 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:28.01 | Zeeek | another thing I'v seen discussed on FWD forum is eyebeam |
| 13:28.26 | fishboy1669 | anyone know how to restrict an extention from ringing out on certain sip nubers? |
| 13:28.40 | fishboy1669 | wupee got my * accepting sip from external |
| 13:28.43 | fishboy1669 | life is so good |
| 13:28.49 | fishboy1669 | and im off snowboarding next week |
| 13:28.54 | fishboy1669 | bring it on |
| 13:29.25 | bjohnson | fishboy1669: put the restricted numbers in a different context and don't let the restricted users access it |
| 13:29.35 | allgood | Zeeek, so, I must put T on the extensions.conf Dial commands? |
| 13:29.57 | Zeeek | allgood no you must read the output of show application Dial |
| 13:30.26 | allgood | Zeeek, is a man page... :-D ... but I didn't understood where I use the T parameter |
| 13:30.29 | Zeeek | but then any time you type # you will hear "transfer?" |
| 13:30.55 | Zeeek | allgood in that case you need to carefully read this: |
| 13:31.00 | allgood | Zeeek, I think that this will not be a problem for my customers |
| 13:31.04 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN510 |
| 13:31.30 | Zeeek | if you don't know where the arguments go in Dial you really need to study up a little |
| 13:32.33 | allgood | Zeeek, will look at it |
| 13:33.19 | allgood | Zeeek, thx for your help |
| 13:33.36 | Zeeek | allgood: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x518.html search for dial() |
| 13:36.08 | allgood | Zeeek, I didn't understood where the dial is used for outgoing calls... |
| 13:36.17 | Zeeek | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 13:36.46 | allgood | Zeeek, but I think I'm annoying you too much with newbie questions... please forgive me |
| 13:36.56 | Zeeek | Dial will be used to ring both a distant AND a local phone if you think about it |
| 13:37.11 | Zeeek | no I think you need to go read one of the 5 docs I just posted |
| 13:37.22 | allgood | Zeeek, I'll read them |
| 13:37.25 | allgood | Zeeek, thx |
| 13:37.26 | Zeeek | promise? |
| 13:37.33 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 13:37.40 | fa | I want switch call to first free iax consultant, how can i do that? |
| 13:37.40 | allgood | Zeeek, hehe... yes... I promise |
| 13:37.53 | allgood | Zeeek, just got it... |
| 13:38.13 | Zeeek | A list of asterisk commands: |
| 13:38.13 | Zeeek | http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands |
| 13:38.16 | fa | reload |
| 13:38.24 | allgood | t on incoming Dial() and T on outgoing |
| 13:38.42 | Zeeek | since there is also a 't' priority, it IS confusing |
| 13:39.05 | *** join/#asterisk nicolasg (~nicolasg@host-112.6.60.66-ta.adsl.netizen.com.ar) |
| 13:40.07 | nicolasg | Hello all |
| 13:40.14 | fishboy1669 | hi |
| 13:40.16 | george_ | Anyone using raw audio files for MOH? I have it working on one machine, but not on another... |
| 13:40.27 | Luhiwu | hi nicolasg (hola) |
| 13:40.29 | fishboy1669 | poo spoke to soon i can dial out sip but not in |
| 13:40.31 | fishboy1669 | arse |
| 13:40.49 | nicolasg | hola luhiwu |
| 13:40.56 | george_ | fishboy1669: check your registration table |
| 13:42.13 | bjohnson | fa: you could ring all phones and let one pick up (by using & in the dial string .. read the dial command docs) .. or you could use call queues and agent logins (this requires a lot of reading) |
| 13:42.33 | fishboy1669 | as in sip show peers |
| 13:42.35 | fishboy1669 | there all there |
| 13:42.36 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 13:42.41 | cypromis | hi mr flashpanel |
| 13:42.44 | fishboy1669 | is cos im trying to go through nat |
| 13:42.53 | nicolasg | hi cypromis |
| 13:43.13 | george_ | fishboy1669: you migth want to have the sip client unregister and reregister |
| 13:43.25 | fishboy1669 | ok |
| 13:43.29 | george_ | I'm not a wizard at this, but I've done some * <--> SIP stuff... |
| 13:43.44 | fishboy1669 | via nat? |
| 13:43.54 | Jas_Williams | fishboy1669: No try sip show registry |
| 13:44.12 | fishboy1669 | blank |
| 13:44.42 | george_ | I have avoided doing SIP through a NAT. |
| 13:44.54 | fishboy1669 | i cant :( |
| 13:44.57 | Jas_Williams | fishboy1669: Then Asterisk has not registered to any external sip providers is this correct ? |
| 13:45.31 | fishboy1669 | yes but it shows the peers |
| 13:45.59 | nicolasg | I have a question for fellow asteerisk users... I would like your opinion before bothering a bug marshall |
| 13:46.09 | fishboy1669 | xlight - nat - * - ip phone |
| 13:46.51 | nicolasg | When you receive a call (your phone is ringing), would you like to have the callerid of the remote party available before picking up the phone? |
| 13:47.23 | george_ | fishboy1669: I'm afraid I can be of little help w/ SIP and NAT.... sorry |
| 13:47.36 | nicolasg | or is it ok to have the callerid AFTER picking up and the channels are bridged? |
| 13:47.37 | Zeeek | geaorge_ |
| 13:47.46 | Zeeek | george_ |
| 13:48.02 | Zeeek | http://willypick.mindsay.com/?entry=10 |
| 13:48.18 | bjohnson | nicolasg: before |
| 13:48.33 | nicolasg | I think the same |
| 13:48.41 | Zeeek | nicolasg you DO have the callerid before answering |
| 13:48.43 | nicolasg | but it seems that the latest asterisk does not |
| 13:48.58 | allgood | Zeeek, just tested the T here... worked... thx for your help and directions |
| 13:49.01 | nicolasg | zeeek: not in many places |
| 13:49.03 | Zeeek | great! |
| 13:49.06 | george_ | Zeeek: thanks, I'll note that |
| 13:49.12 | Zeeek | where are you nico? |
| 13:49.16 | nicolasg | zeeek: what version of asterisk are you using? |
| 13:49.21 | Zeeek | 1.0.3 |
| 13:49.28 | fishboy1669 | georege cheers anyway |
| 13:49.32 | nicolasg | zeeek: ok, you have the callerid, try 1.05, and you wont |
| 13:49.44 | Zeeek | why would I try 1.0.5? It isn't ready yet |
| 13:50.04 | nicolasg | look at bug 3471, and you will find that you wont have it anymore |
| 13:50.05 | Zeeek | at least that's what I keep hearing *here* |
| 13:50.19 | nicolasg | zeeek, you are right |
| 13:50.27 | nicolasg | But the comments on bug 3471 disturbs me |
| 13:50.28 | Zeeek | How would asterisk have it for use with dialplans? |
| 13:50.42 | bjohnson | I'm compiling a list of special * extensions to avoid duplicating in a dial plan .. what is the one for parking a call? #71? |
| 13:51.17 | tzafrir | 700 |
| 13:51.43 | Zeeek | configurable |
| 13:51.54 | fishboy1669 | is anyone here from the uk and is selling theses systems as there main concernc |
| 13:52.12 | Zeeek | I think bonbon does |
| 13:52.22 | Zeeek | haven't seem him for a long time tho |
| 13:52.36 | Zeeek | fish check out the Biz list |
| 13:52.55 | fishboy1669 | where is biz list? |
| 13:53.05 | Zeeek | it's the biz mailing list |
| 13:53.22 | Zeeek | http://lists.digium.com/mailman/listinfo/ |
| 13:53.30 | fishboy1669 | oh im dum do i get details of * site |
| 13:53.33 | fishboy1669 | aha beet me to it |
| 13:53.34 | fishboy1669 | cool |
| 13:53.37 | fishboy1669 | cheers zeek |
| 13:54.44 | nicolasg | zeeek: it seems that Cordyon76 thinks that the callerid is useful only after the call is bridged |
| 13:55.02 | Zeeek | that's terrible! I'll never update! |
| 13:55.10 | Zeeek | never, ever, ever! |
| 13:55.22 | Zeeek | bad devel guy, bad! |
| 13:55.52 | Zeeek | can't be right. The whole pont of CID is to see who's calling |
| 13:56.03 | nicolasg | zeeek, you are right |
| 13:56.14 | nicolasg | zeeek, the flash panel will not be that usefull anymore! |
| 13:56.37 | Zeeek | I use a popup window on all our maxhines |
| 13:56.47 | Zeeek | my wife would be really angry if that goes away |
| 13:57.04 | nicolasg | Maybe I did not get the new idea of callerids.. |
| 13:57.07 | mAsH` | anyone installed * with VIA C3 processor? |
| 13:58.17 | fa | How to check if IAX phone is "in call" ? |
| 13:58.27 | george_ | that change to CID seems nonsensical... are you sure that's the "new way"? |
| 13:58.39 | nicolasg | it seems |
| 13:59.10 | george_ | that's nutz. Like zeeek said, that defeats the purpose of having CID... |
| 13:59.12 | nicolasg | At least, I did not find a manager event that lets me extract the callerid before the calls are bridged |
| 13:59.26 | nicolasg | The callerid is allways your own |
| 13:59.45 | george_ | doesn't it get bridged when it starts ringing, or is bridging connecting the audio streams? |
| 14:00.12 | darkskiez | hmm, is it a fault of asterisk or the 7960 or my configuration, that it doesnt say the name of the person i'm calling on the phone. |
| 14:00.12 | Zeeek | why does it say "fixed in CVS" |
| 14:00.26 | nicolasg | no, it is bridged when you pick up and the audio is connected both ways |
| 14:00.47 | PoWeRKiLL | Hi all, Salut Zeeek :) |
| 14:00.59 | Zeeek | Hi PoW |
| 14:01.19 | nicolasg | zeeek: because the calleridnmae was right |
| 14:01.40 | Zeeek | hmmmm this can't be right. This is only in the manager stuff |
| 14:02.18 | Zeeek | are you saying the CID doesn't work even on the phone? |
| 14:02.41 | nicolasg | I believe that it brakes also the lists of last callers from some ip phones |
| 14:02.56 | nicolasg | but not the callerid on the display on my budgetone |
| 14:02.56 | Zeeek | is CID displayed on the phone? |
| 14:03.08 | nicolasg | yes, on the phone it is displayed |
| 14:03.31 | Zeeek | ok let me put it this way: what does NoOp(${CALLERID}) show in the first priority of an extension? |
| 14:03.43 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 14:03.43 | Zeeek | it still works, right? |
| 14:04.05 | Zeeek | this is about something in the manager interface, correct? |
| 14:04.28 | nicolasg | zeeek: When I found this last week |
| 14:04.45 | nicolasg | the show channels displayed the wrong callerid (in my opinion) |
| 14:05.06 | nicolasg | before the change, if A called B, and I did a show channel B, I see A as the callerid |
| 14:05.10 | bjohnson | just before I start a wiki page .. is there one that summarizes * default system extensions and discusses dialplan organization? I haven't found one .. but thought I'd ask |
| 14:05.15 | Zeeek | He seems to be saying you can get it if you need it |
| 14:05.27 | Zeeek | bjohnson go for it! |
| 14:05.36 | nicolasg | zeeek: who? |
| 14:05.43 | nicolasg | zeeek: cordyon? |
| 14:05.46 | Zeeek | Corydon76 |
| 14:05.48 | Grooby | go bjohnson go! |
| 14:05.53 | nicolasg | zeeek: he says that you can get it with the LINK event |
| 14:05.57 | Zeeek | See the manager Link event. You can then either note the previous manager event describing the callerid of each channel or you can request information on the bridged channel, via "Command: show channel $chan". |
| 14:06.09 | Grooby | and while at it, can someone explain to me if there was a default call waiting setup in asterisk? |
| 14:06.15 | nicolasg | zeeek: thats when you PICK UP the phone |
| 14:06.20 | Grooby | i.e. disabled or enabled? |
| 14:06.26 | darkskiez | is there a way i can dump all variables? |
| 14:06.30 | nicolasg | zeeek: not when is ringing |
| 14:06.32 | fa | how can i check the status of iax2 user - is it "in call" |
| 14:06.41 | Zeeek | nicolas why do you care about this? Does it break something you wrote? |
| 14:06.59 | bjohnson | Grooby: a lot of basic things like that are hard to find info about unless someone points you in the right direction .. need a doc that gives a brief overview of those things |
| 14:07.15 | nicolasg | zeeek: I'm not the only one |
| 14:07.23 | Zeeek | bjohnson I approve of your initiative - GO FOR IT! |
| 14:07.24 | nicolasg | zeeek: there are mails in the users lists, in the devel list |
| 14:07.24 | bjohnson | fa: iax2 show channels I think |
| 14:07.37 | nicolasg | zeeek: it breaks the callerid on my flash operator panel |
| 14:07.51 | nicolasg | zeeek: I can fix it if I can get the callerid before the LINK event |
| 14:08.09 | shido6 | mornin |
| 14:08.24 | Zeeek | I see. As I said, I use something else for callerid |
| 14:08.47 | Zeeek | shido6 are you in UK by any chance? |
| 14:08.53 | nicolasg | zeeek: what do you use? It depends on the manager interface? |
| 14:08.56 | fa | bjohnson I want to switc incomaing call to first free consultant - is good way to check status of iax user and then switch or play massage and hangup? |
| 14:09.05 | Zeeek | no I don like the manager interface much |
| 14:09.13 | fa | bjohnson i want to do that in agi with php. |
| 14:09.46 | shido6 | not in the uk |
| 14:09.48 | shido6 | whats up? |
| 14:09.51 | shido6 | I have a POP in the uk |
| 14:10.15 | Makenshi | The JANET voice advisory group is kicking off now, got dates for the first meeting |
| 14:10.26 | nicolasg | zeeek: ok.. believe me, you will have problems, using the manager or not |
| 14:10.33 | george_ | anyone here use raw audio files for MOH? |
| 14:10.53 | Jas_Williams | Makenshi Joint Academic Network ? |
| 14:11.17 | Zeeek | nicolas if NoOp can print the variable CALLERIDNUM that's good enuf for me |
| 14:11.23 | vaewyn | fa: You want a call queue... no reason to do it yourself... http://www.voip-info.org/wiki-Asterisk+call+queues |
| 14:11.38 | *** join/#asterisk E|nyPRI_ (~E_nste_N@205-200-64-180.static.mts.net) |
| 14:11.49 | Zeeek | vaewyn links are lost on him - been there all day with that |
| 14:11.57 | Grooby | bjohnson, yeah...i was pulling my hair out on why my phone wasn't doing call waiting til I saw the dial plan to enable it (i am still using *@home) |
| 14:12.04 | Grooby | so now i am trying to figure out how to get that set to enable by default |
| 14:12.22 | vaewyn | Zeeek: heh... ohh well... wortha try |
| 14:12.26 | E|nyPRI_ | anyone know any providers using sip in the 1c/minute range, wherer you can set callerid? |
| 14:12.27 | Zeeek | sure |
| 14:12.43 | fa | vaewyn thanks for help |
| 14:12.46 | Zeeek | shido6 I was confusing you with someone else, sorry :) |
| 14:12.59 | Makenshi | Jas_Williams, yes |
| 14:14.01 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
| 14:14.13 | fa | vaewyn but i have in database a skill groups.. base on callrdid number.. and i want only to switch to other consultant, but from that same skill group (i must make a select to datbase) |
| 14:14.47 | vaewyn | fa: then set up a queue for each group and be done with it |
| 14:14.54 | nicolasg | zeeek: I'll bet calleridnum wont display the correct value, I will try later today. |
| 14:15.17 | *** join/#asterisk datareactor (datareacto@203.81.215.183) |
| 14:15.33 | Zeeek | if it doesn't that should be fixed. I can't believe tho that they would think CID isn't needed before answering |
| 14:16.10 | Godsey | I just got an email from sun |
| 14:16.13 | nicolasg | I think the same |
| 14:16.24 | Godsey | granting commercial licence for solaris 10 unlimited |
| 14:16.28 | Godsey | wonder if everyone got that |
| 14:16.41 | vaewyn | Godsey: Yep... |
| 14:16.53 | vaewyn | funny thing... we already had that being an educational institution |
| 14:16.53 | *** join/#asterisk Caede (~chatzilla@204.94.248.81) |
| 14:17.02 | vaewyn | morons |
| 14:17.09 | Godsey | nifty :) |
| 14:17.23 | Godsey | I hope this dvd release includes ZFS |
| 14:17.44 | Godsey | I only wish zfs was a cluster enabled fs :) |
| 14:18.02 | *** join/#asterisk mcisse__ (~mcisse@ARennes-303-1-4-211.w80-11.abo.wanadoo.fr) |
| 14:18.56 | Godsey | time to go buy a crap load of ram and setup some containers tonight :) |
| 14:20.04 | Godsey | Directory Server may contain, at no charge, up to an aggregate |
| 14:20.04 | Godsey | maximum of |
| 14:20.04 | Godsey | 200,000 Entries, across any and all Directory Instances running |
| 14:20.04 | Godsey | Enterprise Wide. |
| 14:20.04 | Godsey | <PROTECTED> |
| 14:20.07 | Godsey | 10 |
| 14:20.13 | Godsey | oops didn't know it wouldn't one line sorry |
| 14:20.31 | Godsey | it goes on to say it excludes solaris 10 entries that don't define users |
| 14:20.39 | Godsey | wonder if that means posixAccount schema stuff :) |
| 14:24.36 | *** join/#asterisk El_Presidente (Martin@p508C9D02.dip0.t-ipconnect.de) |
| 14:24.37 | El_Presidente | hi |
| 14:25.21 | fa | hi |
| 14:27.46 | Godsey | Hello Mr. Koehler! |
| 14:27.52 | El_Presidente | :) |
| 14:28.11 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:28.15 | El_Presidente | a well informed person :) |
| 14:28.39 | Godsey | I have a great friend who works at zeus.de :) |
| 14:28.47 | El_Presidente | ah |
| 14:28.57 | El_Presidente | :) |
| 14:29.05 | *** join/#asterisk darwin_35 (~mrverizon@pa-robinson1b-88.pit.adelphia.net) |
| 14:29.17 | El_Presidente | is someone using intel dialogic cards with asterisk? |
| 14:29.18 | darkskiez | Hmm, is it a fault of 1) asterisk 2) the 7960 or 3) my configuration, that the phone does not display the name of the called person |
| 14:29.23 | Godsey | only a few days ago he said he was pissed that so many americans think Schroder(sp?) runs .de :) |
| 14:29.35 | darwin_35 | ok having a issue when I load asterisk 1.0.5 it spawns 12 times |
| 14:30.07 | El_Presidente | yes godsey mr. schroeder is the apocalypse of germany |
| 14:30.15 | darkskiez | darkskiez, is it a problem? |
| 14:30.16 | darwin_35 | is this a know issue ? |
| 14:30.17 | darkskiez | duh |
| 14:30.51 | darkskiez | darwin_35, is it an issue ? |
| 14:31.17 | *** join/#asterisk drfc (~drfc@8.10.2.4) |
| 14:31.27 | drfc | heya |
| 14:31.38 | nicolasg | darkskiez http://bugs.digium.com/bug_view_page.php?bug_id=0003471 |
| 14:31.42 | darwin_35 | I dont find any mention in the bug reports |
| 14:31.58 | darkskiez | darwin_35, why is it a problem? |
| 14:31.59 | nicolasg | darkskiez: IMHO the callerid is totally broken |
| 14:32.00 | drfc | anyone run into this problem http://pastebin.ca/raw/4423 |
| 14:32.14 | darwin_35 | it should only spawn 1 process |
| 14:32.20 | *** part/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 14:32.33 | darkskiez | nicolasg, thats in the manager though, no ? |
| 14:32.54 | darwin_35 | yet its spawning 12 |
| 14:33.06 | darkskiez | darkskiez, why should it only run 1 process? |
| 14:33.14 | darkskiez | darwin_35, why should it only run 1 process? |
| 14:33.32 | nicolasg | darkskiez: I think is the calleridname on any place |
| 14:33.38 | freat | good morning! |
| 14:33.45 | *** join/#asterisk datareactor (datareacto@203.81.215.183) |
| 14:34.08 | darwin_35 | by norm it should only run 1 process that all it did before |
| 14:34.15 | *** join/#asterisk nestAr (nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 14:34.20 | nestAr | man, freenode' |
| 14:34.23 | nestAr | is weird |
| 14:34.35 | darkskiez | nicolasg, the callerid works, but if i dial say 3000 on my phone, it just says 3000, i wanted to know if that should resolve into the name of the destination, or how that should happen |
| 14:35.15 | fa | can somebody helop with confiugre agents queues? |
| 14:35.42 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 14:35.46 | nicolasg | darkskiez: if the name is not provided, you can use agi to set the calleridname |
| 14:35.49 | sjaak538 | Hello can i manage the order of using codecs I like to use first G729 and then G911 |
| 14:35.51 | thieumS | plz help , amaflags=omit doesn't work |
| 14:35.55 | randu | hello! Is the Festival Server hard to get up and running for asterisk? |
| 14:36.59 | shido6 | yes |
| 14:37.01 | shido6 | you can |
| 14:37.04 | shido6 | sjaak538 |
| 14:37.16 | darkskiez | nicolasg, right, say i call from "Dave <3002>" to "Bob <3001>", dave dials 3001, and his phone says 3001, bob gets a call from "Dave <3002>" ok. but can Dave get the callerid of Bob on the phone to find out the name of the extension he has just called. |
| 14:37.23 | sjaak538 | but how just in order in your conf file ?? |
| 14:37.26 | shido6 | dark - is that what you WANT it to do? |
| 14:37.33 | shido6 | disallow=all |
| 14:37.35 | shido6 | allow=g729 |
| 14:37.37 | shido6 | allow=ulaw |
| 14:37.45 | shido6 | if your device supports g729 |
| 14:37.50 | shido6 | it will use g729 |
| 14:37.54 | sjaak538 | That's all |
| 14:37.56 | shido6 | if it supports ulaw |
| 14:37.59 | shido6 | it will use ulaw |
| 14:38.04 | sjaak538 | easy |
| 14:38.09 | sjaak538 | Great thanks |
| 14:38.17 | darkskiez | shido6, i want to know if the phone should resolve the callerid of the phone it connected too. |
| 14:38.42 | shido6 | does the phone you are calling have some special number? |
| 14:38.42 | nicolasg | darkskiez: hmm, I do not think that you can feed the callerid of the caller |
| 14:38.49 | bjohnson | let me know if I've missed any special default extensions http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Planning |
| 14:38.56 | darkskiez | Like a SetCalledId rather than SetCallerId |
| 14:38.57 | shido6 | are you using LDAP? |
| 14:40.00 | bjohnson | sjaak538: order is set in the general section .. individual sections can limit which codecs are used |
| 14:40.18 | darkskiez | shido6, its a demo deployment with 2x Cisco 7960s. All I want to do is get the calling phone to resolve the number it has dialled into the name, rather than just the receiving phone getting the callerid of the sender. |
| 14:40.25 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 14:40.59 | sjaak538 | Thanks bjohnson and shido I'm going to test this |
| 14:40.59 | shido6 | err |
| 14:41.03 | shido6 | u can set the calleridname |
| 14:41.07 | shido6 | of the dialing phone |
| 14:41.09 | bjohnson | darkskiez: what does that mean? |
| 14:41.09 | shido6 | in sip.conf |
| 14:41.12 | allgood | Zeeek, just printed the hichhikers guide... :-D |
| 14:41.15 | shido6 | or on a per call basis |
| 14:41.21 | allgood | Zeeek, killed some pine trees |
| 14:41.26 | bjohnson | darkskiez: you just want the incoming callerid to show up on the extensions? |
| 14:41.43 | bjohnson | allgood: Don't Panic |
| 14:41.49 | darkskiez | I want reverse callerid, to resolve the name of the person you called. |
| 14:42.09 | darkskiez | like exists on every other phone system i've used. |
| 14:42.15 | allgood | I have another question... |
| 14:42.25 | bjohnson | darkskiez: you'll have to look it up somehow and then use setcallerid |
| 14:42.31 | allgood | here on brazil, caller id is simple DTMF tones sent to the line before the first rink |
| 14:42.45 | allgood | is there a way to make asterisk generate this tones on fxs interfaces? |
| 14:42.46 | darkskiez | bjohnson, that sets the ID of the caller though |
| 14:43.25 | fa | bjohnson Can i define a groups of users in iax.conf? |
| 14:43.32 | bjohnson | darkskiez: I've never seen anything other than displaying the number you called .. unless you have a db to look it up in |
| 14:43.40 | bjohnson | fa: I don't know |
| 14:43.45 | darkskiez | bjohnson, i have a db, i dont know how to send it to the phone. |
| 14:43.45 | bjohnson | fa: I don't think so |
| 14:43.54 | shido6 | reverse callerid? |
| 14:44.04 | *** join/#asterisk kant (~bernd@63.245.57.70) |
| 14:44.05 | shido6 | are you dialing by name? |
| 14:44.27 | darkskiez | On other phone systems when i've dialled a number it turns it into the name.. |
| 14:44.50 | fa | bjohnson I want to switch call to first free consultant. But from specific group.. no all |
| 14:44.56 | nicolasg | darkskiez: I think about one way on doing it, but its akward, and it will not be comfortable to users |
| 14:45.25 | darkskiez | hmmm |
| 14:45.35 | bjohnson | darkskiez: never heard of it .. you want to feed callerid back to the originating internal phone .. don't know if it can be done |
| 14:45.45 | fa | bjohnson do you know how to do that? |
| 14:46.00 | darkskiez | bjohnson, never used any other phone exhange? |
| 14:46.10 | darkskiez | even 10+ year old ones do that. |
| 14:46.11 | bjohnson | fa: you could ring all phones and let one pick up (by using & in the dial string .. read the dial command docs) .. or you could use call queues and agent logins (this requires a lot of reading) |
| 14:46.43 | bjohnson | darkskiez: yes .. a few key systems and a number of pbxs units at other locations |
| 14:46.57 | bjohnson | darkskiez: our Nortel CICS does not do that |
| 14:47.47 | fa | bjohnson i define a queues with [support] timeout = 15 and member => IAX2/me |
| 14:47.56 | fa | but id didn't calling to me |
| 14:48.07 | bprice20 | can I do a register using realtime configuration engine? |
| 14:48.14 | bjohnson | I haven't used queues |
| 14:48.27 | bprice20 | or do i need to leave register statements in static config files like sip.conf or iax.conf |
| 14:48.37 | bprice20 | bjohnson how are ya man |
| 14:49.03 | bjohnson | fine .. don't know realtime .. but you could use a dyndns service to avoid register statements at all |
| 14:49.09 | bprice20 | bjohnson, i have realtime working now |
| 14:49.14 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 14:49.31 | bprice20 | I'm registering with the termination provider |
| 14:49.54 | shido6 | what the ? |
| 14:49.57 | bprice20 | bjohnson, no big deal i'll leave the registers in sip.conf |
| 14:49.58 | bjohnson | oh .. guess you can't change that unless you can convince them to .. |
| 14:49.59 | shido6 | bprice20 what? |
| 14:50.11 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:50.26 | Zeeek | allgood I printed that doc up last year and brought it with me on the raod. Read and reread it 20 times at least until I knew what was in it and understaood everything I needed to understand |
| 14:50.35 | shido6 | as long as the register statement is valid |
| 14:50.53 | bprice20 | shido6, broadvoice requires that i register with them in my sip.cnf, but i am trying to move everything over to realtime |
| 14:51.18 | allgood | Zeeek, can you explain a little more your idea of how to simulate attended transfers with a bunch of extensions? |
| 14:51.19 | bprice20 | I have to leave the register statements in sip.conf and put the info for the clients i the mysql db for realtime |
| 14:51.23 | bprice20 | its working so far |
| 14:51.23 | bjohnson | darkskiez: try again later to see if anyone knows .. also check the wiki and mailing list archives .. but I don't remember seeing anything like that |
| 14:51.46 | Zeeek | allgood actually it would work best with a phone that had programmable buttons |
| 14:52.04 | Zeeek | In fact maybe IAXPhone |
| 14:52.14 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:52.16 | Zeeek | if that will dial when "off hook" |
| 14:52.17 | allgood | IAXPhone is a good option |
| 14:52.38 | Zeeek | will it dial the number on the button while you are on channel in a call? |
| 14:53.02 | allgood | Zeeek, do not know this |
| 14:53.30 | djin | does anyone know what happens if two SIP are registered under the same login at the same time? Do incoming calls route to both phones, of the first or last logged in? |
| 14:53.31 | *** join/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 14:53.52 | Makenshi | djin, only one may be registered at any time |
| 14:53.53 | allgood | Zeeek, I believe that the programable buttons of iaxphone is only iax addresses... |
| 14:54.01 | *** part/#asterisk darwin_35 (~mrverizon@pa-robinson1b-88.pit.adelphia.net) |
| 14:54.02 | allgood | Zeeek, not sequences of DTMF |
| 14:54.07 | djin | So the second fails lo login? |
| 14:54.56 | Makenshi | djin, when another terminal logs in, the new registration replaces the old |
| 14:54.58 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 14:55.04 | shido6 | djin |
| 14:55.04 | shido6 | no |
| 14:55.09 | Zeeek | djin what you wanna do is have multiple accounts |
| 14:55.15 | shido6 | whoever registers last |
| 14:55.18 | shido6 | gets the call |
| 14:55.25 | shido6 | better answer still |
| 14:55.29 | shido6 | sip show peers |
| 14:55.34 | djin | Zeeek, I know. Better use callgroup. |
| 14:55.34 | shido6 | whoevers ip is there |
| 14:55.37 | shido6 | will get the call |
| 14:55.46 | shido6 | you can use the "&" |
| 14:55.50 | shido6 | and call groups are better yes |
| 14:55.54 | ManxPower | It's a very very bad thing to have two devices/lines register as the same account. |
| 14:56.06 | djin | I was just asked this question and was wondering what would happen. |
| 14:56.08 | Zeeek | oh oh - you woke up ManxPOwer! |
| 14:56.09 | shido6 | there he is |
| 14:56.21 | shido6 | take your cape off ManxPower :) |
| 14:56.30 | djin | Average user do very, very bad things, ManxPower ;) |
| 14:56.39 | Zeeek | kickin ass and takin names :) |
| 14:56.45 | randu | ManxPower: Have you had any experience with Festival, is it hard to install and configure for asterisk? |
| 14:56.50 | Zeeek | drums stop. no good. |
| 14:56.59 | shido6 | festival works , its easy to setup follow the directions to the T |
| 14:57.19 | djin | Could fa stop sending PM'es? |
| 14:57.19 | Zeeek | Manx used to have a page on that stuff - very good info |
| 14:57.23 | kant | Accepted AUTHENTICATED TBD call from 10.2.2.202 <--- What does TBD mean?! |
| 14:57.39 | shido6 | means u have an IAX device registered |
| 14:57.39 | Zeeek | fa has been told the same things all day by everyone |
| 14:57.49 | shido6 | or someone took an IAXy offhook |
| 14:57.49 | shido6 | :) |
| 14:58.00 | djin | I know, it was nog the first time, Zeeek. |
| 14:58.33 | Zeeek | which is why he's on about 90% of the ignore lists of the planet |
| 14:58.33 | Zeeek | slash ignore pseudo |
| 14:59.07 | djin | FA stands for commercials with bare chested women here, so it's hard to ignore, Zeeek. |
| 14:59.26 | Zeeek | yea I love those! WHere do they find such lovely breats? |
| 14:59.33 | Zeeek | S |
| 14:59.48 | djin | Well, they don't keep rubbing up like fa does ;) |
| 14:59.49 | Zeeek | music hath charmz to soothe |
| 15:00.14 | Zeeek | drums stop. very, very bad. |
| 15:00.52 | fa | djin i dont understand you |
| 15:01.17 | Zeeek | Drums stop VERY BAD!! Bass solo :( |
| 15:01.43 | Grooby | no guitar? |
| 15:01.45 | Grooby | or piano? |
| 15:01.50 | *** join/#asterisk inspired (mikael@host-81-191-114-81.bluecom.no) |
| 15:01.51 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 15:01.54 | Zeeek | when drums stop, very bad. |
| 15:01.57 | jero | hello |
| 15:02.00 | fa | djin what? |
| 15:03.01 | *** part/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 15:03.23 | Grooby | i am off |
| 15:03.26 | Grooby | see you all later |
| 15:03.30 | datareactor | is ping should be open to register to voip provider like go2call |
| 15:03.31 | Grooby | and sank you all for da help! |
| 15:03.38 | Zeeek | goodbye and a special wave of the hand bye ye to fa |
| 15:03.39 | jero | Anyone knows if I can lease for example just 4 channels of a T1 to Bell in Canada ? |
| 15:03.46 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 15:03.51 | Zeeek | Starter tutorial: |
| 15:03.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 15:03.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 15:03.51 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 15:03.51 | Zeeek | THE reference of the moment: |
| 15:03.51 | Zeeek | http://www.asteriskdocs.org |
| 15:03.56 | shido6 | jero |
| 15:03.58 | shido6 | where are you |
| 15:03.58 | *** join/#asterisk escualis (~cfuentea@192-146.adsl.cust.tie.cl) |
| 15:03.59 | shido6 | ? |
| 15:03.59 | Zeeek | ^^^^^^^^^ for fa ^^^^^^^^^^^^ |
| 15:04.03 | jero | shido6, in Montreal |
| 15:04.14 | shido6 | setup is gonna hurt |
| 15:04.16 | escualis | hello :-) |
| 15:04.28 | jero | gonna hurt? |
| 15:04.28 | Zeeek | *enog* |
| 15:04.59 | shido6 | setup price |
| 15:05.01 | shido6 | for a T |
| 15:05.08 | shido6 | unless you're at a colo |
| 15:05.17 | jero | a colo ? |
| 15:05.31 | shido6 | colocation facility |
| 15:05.39 | jero | for servers or phone lines? |
| 15:05.42 | fa | i add that to queses.conf |
| 15:05.43 | fa | [support] |
| 15:05.43 | fa | music = mptrzy |
| 15:05.43 | fa | timeout = 5 |
| 15:05.43 | fa | member => IAX2/inezk |
| 15:05.45 | *** join/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 15:05.45 | fa | strategy = ringall |
| 15:05.50 | fa | and in extensions.conf i am running |
| 15:05.51 | mtqh | fa: USE PASTEBIN |
| 15:06.02 | datareactor | Zeeek why are after fa ;) |
| 15:06.09 | fa | exten => s,3,Answer |
| 15:06.10 | fa | exten => s,4,Queue(support) |
| 15:06.12 | fa | good? |
| 15:06.15 | shido6 | no, fa |
| 15:06.18 | shido6 | use pastebin.ca |
| 15:06.25 | darkskiez | bjohnson, chan_sccp has a setCallerParty command. I want this for SIP. I'm sure when you call someone on a software sip phone it displays the name of the person you are calling. |
| 15:06.27 | fa | ok. in future i will use pastebin, sorry |
| 15:06.33 | mtqh | fa: Thank you |
| 15:06.36 | ManxPower | He floods the channel. He doesn't listen. He wants his hand held all the time. |
| 15:06.37 | fa | but can sb help me with that? |
| 15:06.43 | fa | i want to call a IAX2/me |
| 15:06.48 | kant | Do register => ... statements need to be right after the [general] section in sip.conf or can I place them anywhere? |
| 15:06.55 | ManxPower | What is there to like? |
| 15:06.57 | shido6 | anywhere in [general] |
| 15:07.19 | fa | ManxPower no |
| 15:07.28 | ManxPower | His skills with English also make things more difficult, but that's about the only thing we can't really blame him for. |
| 15:07.34 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 15:07.35 | kant | Say I have [general] [sip-proxy] and then register => ... |
| 15:07.40 | datareactor | :) |
| 15:07.40 | kant | Will that work? |
| 15:08.05 | ManxPower | kant, register will only work under [general] not in any other sections. |
| 15:08.37 | fa | blame - what that means? ;] |
| 15:08.56 | datareactor | ManXPower can you help me setting * from go2call |
| 15:09.00 | kant | Aww shucks, I wanted to have it in different sections. Anywho... |
| 15:09.02 | ManxPower | datareactor, No. |
| 15:09.05 | shido6 | the register is in [sip-proxy] then and not in [general] |
| 15:09.12 | shido6 | which will not worky |
| 15:09.33 | datareactor | ManxPower Ok |
| 15:09.42 | fa | I have only Executing Queue("Zap/1-1", "support||||5") in new stack |
| 15:09.42 | fa | <PROTECTED> |
| 15:09.45 | fa | and that is all |
| 15:11.26 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:11.54 | kant | shido6: But it seems to work since 'sip show registry' shows them as registered. |
| 15:12.33 | shido6 | um |
| 15:12.36 | shido6 | sip show peers |
| 15:12.42 | shido6 | asterisk uses PEERS to reach devices |
| 15:12.53 | shido6 | what does sip show peers show you? |
| 15:13.12 | *** join/#asterisk makamani (~user@pub-nms.stcl.com) |
| 15:13.47 | makamani | is it possible to provide users of extensions a PIN and then each PIN would be allowed to call specific number pattern? |
| 15:14.09 | kant | It shows their Status as UNKNOWN. |
| 15:14.11 | *** join/#asterisk tty74it (~tiziano@151.11.170.2) |
| 15:14.28 | makamani | e.g. user A has pin 1234 and with pin 1234 you are allowed to dial local extensions only and other number not starting with 90 |
| 15:14.58 | fa | it's working ;] nice. |
| 15:15.10 | makamani | i believe Authenticate() would auth a user but then how would the limitation be put in place |
| 15:15.23 | datareactor | congrt fa |
| 15:15.31 | fa | ;] |
| 15:16.35 | fa | thanks |
| 15:16.54 | fa | but i still don't know how to create a iax2 users group |
| 15:17.13 | tty74it | Hi i've a problem with iaxy , i can't use efficently flash button on my analog phone, one time works and the other times doesen't work. I suppose that is a flash time problem but i 've chanche a lot of time settings but without succes. I use italy analog phone attachedt to iaxy |
| 15:18.26 | *** join/#asterisk RoyKa (~roy@193.213.34.92) |
| 15:18.26 | shido6 | um |
| 15:18.26 | shido6 | tty74it |
| 15:18.26 | shido6 | are you using users and peers |
| 15:18.26 | shido6 | or friends |
| 15:18.26 | shido6 | in your iax.conf? |
| 15:18.28 | shido6 | for this iaxy |
| 15:18.34 | shido6 | be honest |
| 15:18.34 | tty74it | friends |
| 15:18.38 | shido6 | why? |
| 15:18.50 | shido6 | create a user |
| 15:18.52 | shido6 | and create a peer |
| 15:18.56 | fa | is it possible to create a groups of iaxy users/friends.. and latter call them member => Iax/@groupname |
| 15:19.07 | tty74it | i read the digium manula for config |
| 15:19.26 | shido6 | yes fa but thats a Manxpower question |
| 15:19.43 | fa | shido6 and who know the answer? |
| 15:20.06 | shido6 | manxpower |
| 15:20.14 | shido6 | let me look at the corp pbx |
| 15:20.20 | fa | ok |
| 15:20.22 | *** join/#asterisk Casper_UA (~casper@ragu.bestnet.kharkov.ua) |
| 15:20.31 | fa | ManxPower alive? |
| 15:20.32 | bjohnson | makamani: yes .. likely with a database system |
| 15:20.46 | tty74it | My problem is: when i wanto to transfer call with analog phone i press flash hook and not all times it work and i can tranfer to anoter phone |
| 15:20.46 | Casper_UA | ~seen eKo1 |
| 15:20.51 | jbot | eko1 <~bernd@63.245.57.70> was last seen on IRC in channel #asterisk, 16h 57m 2s ago, saying: 'Say, do all contexts where calls are made need a t extensions?'. |
| 15:20.52 | shido6 | u can do that fa using Agenets |
| 15:20.56 | shido6 | in queues.conf |
| 15:21.16 | fa | Agents.. but how descipe agents <-> iax friends? |
| 15:21.25 | shido6 | grr |
| 15:21.30 | shido6 | from now on |
| 15:21.43 | shido6 | repeat after me, I will never use friends when Im using iax |
| 15:21.48 | Casper_UA | tty74it: you may need to try with another phone to do it |
| 15:21.49 | tzanger | shido6: :-) |
| 15:21.51 | shido6 | I will only use users and peers |
| 15:22.01 | darkskiez | whats wrong with friends |
| 15:22.05 | shido6 | friends are evil |
| 15:22.09 | fa | shido6 ok. ;] |
| 15:22.13 | shido6 | you cannot rely on them to come through 100% |
| 15:22.14 | fa | shido6 wat is differents ? |
| 15:22.31 | tty74it | Casper_UA is a phone problem? |
| 15:22.32 | shido6 | the difference is friends doesnt always work and creates more problems than solutions |
| 15:22.36 | shido6 | so become a loner |
| 15:22.39 | shido6 | and use users and peers |
| 15:22.49 | fa | ok |
| 15:23.14 | bjohnson | makamani: you could grab the PIN with read (which assigns it to a variable), do a dbget to find the group that the PIN belongs to and assign that to another variable, then use gotoifs to direct the call to other extensions that controls what extension matching is included |
| 15:23.16 | *** join/#asterisk jarnaud (~jarnaud@65.217.47.11) |
| 15:23.29 | jarnaud | Hi all |
| 15:24.23 | bjohnson | darkskiez: type=friends causes problems for callerid settings, authentication (if different incoming and outgoing auth needed .. eg FWD), cdr accountcodes, etc |
| 15:24.50 | tty74it | in yaxi provisioning file conf there is a flash time configuration? |
| 15:24.54 | bjohnson | anything that would require a different config for incoming and outgoing calls with the same "friend" |
| 15:25.00 | darkskiez | rename type=friend type=eviltwin |
| 15:25.04 | makamani | ok bjohnson |
| 15:26.23 | bjohnson | darkskiez: it's simpler for newbies .. but you will quickly learn to separate them. Once you get that far .. you won't use friend again |
| 15:27.37 | *** join/#asterisk educa (~educasoft@d51A56826.access.telenet.be) |
| 15:28.09 | tty74it | I suppose that iaxy isn't a working 100% telephone adapter , i can't use it in a production enviroment |
| 15:28.23 | educa | Hi there. Is there anyone here who could tell me what SDK I could use to make a h323 or a SIP softphone ? |
| 15:28.29 | makamani | bjohnson, if I want to relax rules and set the limitation depending on the originating extension, then I believe that i need to use gotoifs with the variable name that has originating port right? |
| 15:29.28 | bjohnson | yes I think you could do that. I can't rmember the variable name that contains the caller device name though |
| 15:29.53 | makamani | bjohnson, as far as on exists then it's pretty sorted out. thank you for the idea |
| 15:29.58 | Casper_UA | tty74it: maybe... flash timings may differ |
| 15:31.12 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 15:32.36 | *** join/#asterisk RoyKa (~roy@193.213.34.92) |
| 15:32.59 | shido6 | um |
| 15:32.59 | shido6 | STOP |
| 15:33.03 | shido6 | dont knock the iaxy |
| 15:33.08 | shido6 | till you've made a user and peer in iax.conf |
| 15:33.17 | shido6 | pastebin.ca your iax.conf |
| 15:33.23 | shido6 | the IAXy works and works well :) |
| 15:33.40 | shido6 | you also need to be sure you extend the limit of the dhcp lease time as well for the iaxys |
| 15:38.45 | shido6 | where's the pastebin tty7it? |
| 15:39.25 | cypromis | iaxies are perfect |
| 15:39.30 | cypromis | for putting them on the floor |
| 15:39.36 | cypromis | and than jumping a lot on them |
| 15:40.25 | shido6 | ? |
| 15:40.40 | shido6 | I fax through mine, and use it with my cordless |
| 15:40.49 | cypromis | yeah you live in bandwdith paradise |
| 15:41.02 | shido6 | whats wrong cypromis? |
| 15:41.18 | cypromis | I am trying to understand the use of a remote device |
| 15:41.25 | cypromis | that increases the cost of a call |
| 15:41.30 | cypromis | that's all |
| 15:41.31 | cypromis | :D |
| 15:41.34 | shido6 | hehe |
| 15:42.22 | cypromis | and if abusing bandwidth allready |
| 15:42.34 | cypromis | why only supporting an exclusively north american centric codec ? |
| 15:42.46 | cypromis | besides there being no european power supply delivered with it |
| 15:42.51 | cypromis | and some other topics |
| 15:43.04 | cypromis | but anyway, that's not for #asterisk |
| 15:43.05 | cypromis | :D |
| 15:44.14 | djin | if you do a CVS checkout with '-r v1-0', do you get a CVS version of the most current stable release (1.0.5)? |
| 15:44.52 | ManxPower | djin, yes |
| 15:44.59 | djin | cool, thanks. |
| 15:45.04 | ManxPower | Well, 1.0.5 plus any updates since 1.0.5 was released. |
| 15:46.08 | djin | doublechecking, because I thought Asterisk Realtime was final. |
| 15:47.57 | ManxPower | Realtime is only available in CVS-HEAD, not 1.0.x CVS STABLE |
| 15:48.01 | *** join/#asterisk Rick_Hunter (~rhunter@07-037.008.popsite.net) |
| 15:48.44 | djin | That what I just found out. Not focussing on Realtime for now ;) |
| 15:49.07 | BoRiS | MWI with realtime needs fixing :-p |
| 15:49.25 | *** join/#asterisk miketal (~tal@snert3.tal.de) |
| 15:49.33 | miketal | hi all ! |
| 15:50.21 | miketal | question: is it possible to log all the asterisk manager events in a seperate file ? |
| 15:51.04 | *** part/#asterisk escualis (~cfuentea@192-146.adsl.cust.tie.cl) |
| 15:51.25 | djin | miketal, I would settle for a full logging of manager events. |
| 15:51.30 | bjohnson | does anyone here handle a file server system which is accessed by users at multiple locations? I'd like to discuss concepts related to userid mapping and how it might be tied into an * system |
| 15:52.13 | _-Jon-_ | Has anyone had any experieces with livevoip? |
| 15:54.26 | file[laptop] | BoRiSSSSSSSSSSSSSSSSSS |
| 15:55.06 | BoRiS | file!!!!!!!!!!!!!!!!!!!!!!! |
| 15:55.10 | BoRiS | Wassssssssssssssssssssup? |
| 15:55.23 | *** join/#asterisk RoyKa (~roy@dsl-40-122.kunde.brednett.no) |
| 15:58.31 | ManxPower | Ugh. It's been raining here for 3 days. |
| 15:58.44 | shido6 | water main break here |
| 15:58.46 | shido6 | brown water |
| 15:58.53 | shido6 | yum |
| 15:59.46 | shido6 | does it take out chemicals or organisms? :) |
| 15:59.48 | file[laptop] | BoRiS: grumbling as no e-mail from Dell, and pondering going upstairs |
| 15:59.50 | ManxPower | Customer: AOL is blocking our e-mail! What can be do to fix it? Me: Read the fucking instructions that AOL provides in the bounce message. |
| 16:00.58 | ManxPower | shido6, Both IIRC |
| 16:01.24 | darkskiez | what does the VXML_URL sip header do? |
| 16:05.32 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:05.38 | vaewyn | arrghh... I wish the IP-500 was autosense PoE like the 600 |
| 16:06.21 | ManxPower | vaewyn, Well you need the correct cable. |
| 16:06.57 | vaewyn | ManxPower: that's what I wish... the IP-600 you don't even neeed a cable |
| 16:07.07 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 16:07.32 | vaewyn | ManxPower: is there any electronics in the cable even? or are they just flipping the PoE pair for the other polarity? |
| 16:07.52 | ManxPower | vaewyn, I don'tknow. |
| 16:09.17 | vaewyn | It's 4 diodes and a LM chip people!.... less than .50$ |
| 16:09.23 | miketal | question: is it possible to log all the asterisk manager events in a seperate file ? |
| 16:10.12 | *** join/#asterisk porche (~a@dsl81-215-30391.adsl.ttnet.net.tr) |
| 16:10.20 | porche | hi |
| 16:10.34 | porche | got a problem on X100P, hang up detection, |
| 16:10.42 | porche | where must I look for? |
| 16:10.43 | shido6 | takes forever to hangup? |
| 16:10.51 | shido6 | 10 minutes instead of a few seconds? |
| 16:10.53 | porche | yep shido almost |
| 16:10.56 | shido6 | did we talk on the phone? :) |
| 16:11.05 | porche | 10 mins or similar sometimes |
| 16:11.21 | mtqh | porche: busycount=8 in zapata should do it |
| 16:11.24 | porche | :) no it's not me shido |
| 16:11.29 | shido6 | hehee |
| 16:13.08 | miketal | question: is it possible to log all the asterisk manager events in a file ? |
| 16:13.10 | mtqh | has anyone had any luck with the new fxo module from digium....Did I get a bad one? |
| 16:13.16 | miketal | no one knows ? :/ |
| 16:13.22 | shido6 | whats wrong with yours? |
| 16:13.23 | mtqh | miketal: use perl |
| 16:13.56 | miketal | i have some manager events in /var/log/asterisk/event_log |
| 16:14.22 | miketal | but i want all events, so there must be a config file ... ( ill hope so) |
| 16:14.45 | bjohnson | for anyone interested in helping to plan out a dialplan for multiple office locations. Some review and comment are needed. http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Planning |
| 16:19.59 | mAsH` | anyone can help me? |
| 16:20.05 | mAsH` | i get this error |
| 16:20.22 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 16:20.23 | mAsH` | Feb 1 16:22:04 ERROR[1096]: chan_zap.c:9429 setup_zap: Unknown signalling method 'bri_cpe_ptmp' |
| 16:20.23 | mAsH` | Feb 1 16:22:04 ERROR[1096]: chan_zap.c:9071 setup_zap: Signalling must be specified before any channels are. |
| 16:20.23 | mAsH` | Feb 1 16:22:04 WARNING[1096]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 16:20.23 | mAsH` | <PROTECTED> |
| 16:20.23 | mAsH` | <PROTECTED> |
| 16:20.24 | mAsH` | Feb 1 16:22:04 WARNING[1096]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 16:22.18 | ManxPower | mAsH`, This is a Feequently Asked Question. None of the 167 results for your google search of site:lists.digium.com Loading module chan_zap.so failed were helpful? |
| 16:22.51 | mAsH` | opss...thanks |
| 16:22.54 | ManxPower | The first error messag is what you care about. |
| 16:23.09 | Zeeek | bjohnson you're gonna think I'm nuts but I don't understand the first sentence! |
| 16:23.15 | vaewyn | Polycom gurus... how well is the configuration file documented by polycom? |
| 16:23.21 | mAsH` | but it's strange because on another pc it works fine :/ |
| 16:23.24 | Zeeek | err of this section |
| 16:23.25 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:23.27 | ManxPower | looks like you didn't install zapbri or zaphfc before you built Asterisk |
| 16:23.30 | Zeeek | "Here are some of the __extensions patterns that I wanted to avoid: " |
| 16:23.47 | ManxPower | Or tou didn't install libpri |
| 16:24.13 | Zeeek | bjohnson you mean you wanted those to reamin reserved and not be used in a dialplan? |
| 16:24.18 | mAsH` | i did it :/ |
| 16:25.05 | bjohnson | Zeeek: what line do you mean? |
| 16:25.16 | Zeeek | "Here are some of the __extensions patterns that I wanted to avoid: " |
| 16:25.46 | Zeeek | I'm glad you're doing this page but the sentences aren't good for quick reading IMO - no offense meant |
| 16:25.57 | bjohnson | right .. I guess I should edit to say .. standard extension patterns that I wanted to keep |
| 16:26.00 | *** join/#asterisk Flatcat (~ScaredyCa@84.119.133.131) |
| 16:26.15 | Zeeek | something like that would be immediately clearer, you see? |
| 16:26.25 | Zeeek | and the very last sentence: |
| 16:26.33 | Zeeek | Note: any pattern matching that ends in a "." (meaning any number of any digits) should be in a context that is included. This is only only way to control the order of pattern matching! |
| 16:26.46 | *** join/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 16:26.49 | ManxPower | Or just avoided all togather when you can |
| 16:26.54 | Zeeek | I know what this means but if I didn't already know that would be some weird shit |
| 16:27.21 | ManxPower | Using . causes a delay in having your call processed by DigitTimeout seconds. |
| 16:27.42 | vaewyn | Anyone have a Hitachi IP-5000 yet? |
| 16:28.23 | Zeeek | and for the France toll free unfortunately it's way more complicated that this but the basic is 33800 however I'm not sure if they work from "outside" |
| 16:28.52 | Zeeek | there are a zillion 08XXX numbers with about a half zillion tolls |
| 16:28.53 | file[laptop] | vaewyn: not I but people have been in here... and it supposedly works nifty like |
| 16:28.55 | *** join/#asterisk kingcobra (~martin@214.35.233.64.transedge.com) |
| 16:29.00 | Zeeek | from 0 to 50c/min or more |
| 16:29.23 | vaewyn | file[laptop]: I would assume significantly better than the wisip?!? ;P |
| 16:29.39 | bjohnson | Zeeek: I don't call France toll free numbers but picked that up on the mail list and thought I'd include it .. maybe I just delete for now |
| 16:29.45 | ManxPower | countries with variable length dialing plans frequently need . |
| 16:29.52 | Zeeek | yes |
| 16:29.53 | *** join/#asterisk flowed (~flow@pD9EF04D7.dip.t-dialin.net) |
| 16:29.55 | flowed | hi |
| 16:30.01 | file[laptop] | vaewyn: yes. |
| 16:30.05 | vaewyn | countries with variable length dialing plans should be shot :P |
| 16:30.07 | *** join/#asterisk zno (~chatzilla@160.79.174.101) |
| 16:30.13 | ManxPower | vaewyn, Yeah! |
| 16:30.32 | Zeeek | yes |
| 16:30.33 | vaewyn | even the US 7 vs 10 thing is getting old |
| 16:30.37 | ManxPower | Maybe the USA will invade them and force a civilized dialing plan on them! |
| 16:30.42 | Zeeek | yes |
| 16:30.46 | flowed | how to configure my extensions.conf for a default/standard phone for all accounts? |
| 16:31.01 | flowed | when no other extension is given |
| 16:31.16 | bjohnson | flowed: s? |
| 16:31.17 | Zeeek | [default] ? |
| 16:31.51 | *** part/#asterisk Flatcat (~ScaredyCa@84.119.133.131) |
| 16:32.25 | flowed | i mean when somebody calls me to bring it to a standard phone.... |
| 16:32.31 | flowed | my english is bad :) |
| 16:32.56 | bjohnson | Zeeek: some of the goals and concepts included on that page are a little hard to describe and are only partially possible now .. but I'm trying to future proof as much as possible |
| 16:33.03 | flowed | u know when dont have given a extension for this number, ip |
| 16:33.04 | bjohnson | flowed: s? |
| 16:33.10 | Zeeek | good idea tho bjohnson |
| 16:33.20 | *** join/#asterisk Mike (~mike@201.129.119.248) |
| 16:33.25 | Zeeek | flowed : |
| 16:33.26 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 16:33.28 | Mike | what codecs does sipphone support? |
| 16:33.36 | bjohnson | flowed: check examples in extensions.conf or on wiki |
| 16:33.38 | Zeeek | that has the answer |
| 16:33.48 | flowed | ok, thx |
| 16:34.09 | bjohnson | Mike: check the sipphone site/docs/help files |
| 16:34.10 | *** join/#asterisk Fanguin (~Fanguin@p508188F0.dip0.t-ipconnect.de) |
| 16:34.12 | Zeeek | above link The special 's' extension |
| 16:34.20 | djin | Zeeek, they should make that URL as topic. |
| 16:34.30 | bjohnson | Mike: almost everything supports ulaw .. try that first |
| 16:34.35 | Zeeek | it is a particularly powerful piece of prose |
| 16:34.45 | bjohnson | djin: yes |
| 16:35.22 | bjohnson | do only channel mods have ability to change topic? Maybe they could add the wiki url |
| 16:35.27 | randu | Hello! Our company is two companies in 1 ie two 1-800 numbers going to be coming into asterisk. I want to use the same voice mail boxes for each. How would I be able to decipher via the email received with the voice message which 1-800 number the message came in on? |
| 16:35.28 | Zeeek | where's fa? |
| 16:35.32 | mtqh | hehe, I just figured out that you can do something like exten => asdfasdf,1,do sometihng |
| 16:35.41 | mtqh | and the asdfasdf is the extention |
| 16:35.42 | Casper_UA | ~docs |
| 16:35.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:35.44 | bjohnson | randu: callerid? |
| 16:35.49 | flowed | exten => _X.,1,Dial(SIP/snom190) this dont work for me as standard phone :( |
| 16:35.53 | RoyK | ~lart bjohnson |
| 16:35.58 | djin | fa is reading, I hope. |
| 16:36.02 | Zeeek | flowed did you go read the link? |
| 16:36.03 | mtqh | flowed us a lowercase x |
| 16:36.18 | shido6 | flowerd |
| 16:36.23 | shido6 | flowed what the heck is that |
| 16:36.30 | shido6 | dial anynumber and get snom190? |
| 16:36.30 | bjohnson | flowed: exactly |
| 16:36.33 | shido6 | is that what you want? |
| 16:36.49 | flowed | no :) |
| 16:36.51 | flowed | ok |
| 16:36.52 | mtqh | shido6: his nick? |
| 16:36.55 | Zeeek | he want to reinvent 's' using fallthrough without understanding dialplans |
| 16:36.56 | shido6 | thats what you're telling asterisk |
| 16:36.58 | bjohnson | shido6: no .. he wants to go read the wiki page and learn about the s extension |
| 16:37.02 | randu | bjohnson: that is the caller id of the person that called, I want the 1-800 number that it rang in on, ie Company Line 1 or Company Line 2 |
| 16:37.04 | shido6 | LOL |
| 16:37.07 | shido6 | hehe |
| 16:37.13 | zno | my office just went live with asterisk, ditching our old phone system |
| 16:37.15 | zno | so far so good |
| 16:37.24 | mtqh | zno: party time... |
| 16:37.29 | Zeeek | zno watch out for the millenium bug |
| 16:37.47 | zno | these sipura 841s are really a bang for the buck |
| 16:37.51 | Zeeek | or worse, the asterisk valentines bug |
| 16:38.05 | bjohnson | randu: depends on the voip provider .. maybe they can use feed you the 800 as the extension (like FWD comes in on your accound # as the extension) |
| 16:38.15 | Zeeek | on Feb 14th asterisk autocalls the ugly chick in copies and imitates your voice asking her for a date |
| 16:38.17 | *** join/#asterisk allgood (~allgood@200-101-232-031.fnsce7004.dsl.brasiltelecom.net.br) |
| 16:38.28 | allgood | i'm back! :-D |
| 16:38.29 | Zeeek | Valentine's bug |
| 16:38.43 | allgood | can anybody tell me when 1.0.5 was branched from HEAD? |
| 16:38.46 | bjohnson | "begging" for a date |
| 16:38.59 | allgood | hi zeeek ... |
| 16:39.02 | bjohnson | allgood: about 5 days ago |
| 16:39.15 | ManxPower | allgood, 1.0.5 was never branched from HEAD. |
| 16:39.18 | bjohnson | allgood: check the cvs mailing list archives .. should be in there |
| 16:39.19 | Zeeek | it is especially easy since everyone seems to use 2000 as a start to internal ext |
| 16:39.21 | allgood | I was hopping that the atxfer feature was on 1.0.5... just installed the .deb package of it |
| 16:39.27 | ManxPower | it was taken as a snapshot or -r v1-0 |
| 16:39.41 | ManxPower | allanon, NO NEW FEATURES ARE ADDED TO 1.0.5 |
| 16:39.52 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166196242.nb.aliant.net) |
| 16:39.58 | ManxPower | NO new features are added to any 1.0.x release. |
| 16:40.18 | Zeeek | I heard one new feature is that CALLERID is b0rken? |
| 16:40.25 | Zeeek | just a rumor |
| 16:40.30 | bjohnson | well .. if bugs are features .. then removing bugs ? |
| 16:40.38 | allgood | so... 1.0.5 wasn't branched from HEAD... |
| 16:40.53 | allgood | on Jan 21, HEAD already had the attended transfer feature |
| 16:41.18 | Manipura | ~seen cyberstuph |
| 16:41.19 | jbot | Manipura: i haven't seen 'cyberstuph' |
| 16:41.36 | ManxPower | Zeeek, Caller*ID NAME, right? |
| 16:41.42 | allgood | ManxPower, I think I'll have to compile it.... :-D |
| 16:41.45 | *** join/#asterisk LarsAC (~chatzilla@pD95009B2.dip0.t-ipconnect.de) |
| 16:42.04 | Zeeek | perhaps Manx |
| 16:42.08 | randu | bjohnson: ok thanks I will research some more. |
| 16:42.15 | Zeeek | but now I'm too scared to upgrade |
| 16:42.37 | Zeeek | Why upgrade if there are no new features and it works fine |
| 16:42.51 | ManxPower | Zeeek, No reason to. |
| 16:42.51 | Zeeek | and it may break stuff you need ? :) |
| 16:42.54 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:42.58 | ManxPower | 1.0.x is for bugfixes only. |
| 16:43.02 | kant | You upgrade to fix the old features that don't work properly. |
| 16:43.14 | Zeeek | there are apparently none that I use |
| 16:43.23 | *** part/#asterisk djin (~marius@217-19-18-130.dsl.cambrium.nl) |
| 16:43.31 | kant | Sure there are, you just haven't run into them yet. |
| 16:43.40 | Zeeek | OMG my beer can is empty |
| 16:43.56 | Zeeek | that is a seriousbug needing an upgrade |
| 16:44.05 | bjohnson | Zeeek needs a beer can bug fix |
| 16:44.19 | Zeeek | number 3471 on Mantis |
| 16:44.31 | Zeeek | no that's taken |
| 16:44.50 | bjohnson | I just CANNOT figure out sound cards on linux!! |
| 16:44.52 | file[laptop] | po, tat, oe |
| 16:45.00 | Zeeek | potato |
| 16:45.03 | *** join/#asterisk jackflash (~jf@cpc2-rdng8-4-0-cust187.winn.cable.ntl.com) |
| 16:45.03 | file[laptop] | oh god my dad came online |
| 16:45.16 | ManxPower | bjohnson, What's to understand. run alsa-config and be done with it |
| 16:45.21 | Zeeek | better than coming on the phone |
| 16:45.51 | ManxPower | file, I like my parent's computer illitercy just the way it is. 8-) |
| 16:46.04 | Grooby | soundcard on linux? |
| 16:46.15 | Grooby | why do we need soundcards? |
| 16:46.23 | Grooby | pc speaker's beep and boop is perfect! |
| 16:46.27 | ManxPower | Yeah. Why do you need your sound card to work? |
| 16:46.27 | kant | Grooby: Good for testing. |
| 16:46.33 | wankel | grooby: to listen to calls people make through * |
| 16:46.42 | Grooby | oooooh |
| 16:46.45 | Grooby | interesting.... |
| 16:46.48 | LarsAC | hi |
| 16:46.49 | Grooby | someone got a howto? |
| 16:46.59 | kant | Full Dobly Digital AC3 sorround sound phone sex. |
| 16:47.13 | ManxPower | kant, that image just ruined my life. |
| 16:47.14 | bjohnson | ManxPower: I don't seem to have that command .. let me check another desktop |
| 16:47.25 | bjohnson | nope |
| 16:47.26 | ManxPower | bjohnson, What DISTRO are you using? |
| 16:47.35 | bjohnson | fc2 and fc3 |
| 16:47.41 | ManxPower | try sndconfig |
| 16:47.52 | fa | what? |
| 16:48.08 | bjohnson | Grooby: I want to feed sounds to the MOH of my Nortel CICS from the server soundcard |
| 16:48.25 | ManxPower | Zeeek, try "man -k alsa" |
| 16:48.26 | bjohnson | ManxPower: nada |
| 16:48.30 | file[laptop] | you people are absolutely crazy |
| 16:48.34 | file[laptop] | crazy as a coconut |
| 16:48.37 | Grooby | ahhhhh |
| 16:48.44 | ManxPower | file, pot. kettle. black. |
| 16:48.47 | Grooby | not listening to your boss having phone sex huh? |
| 16:48.48 | Zeeek | I don need no stinking sound on linux mon... |
| 16:48.57 | file[laptop] | ManxPower: condensed. |
| 16:49.04 | fa | What is the better in fast ethernet IAX or SIP? and use alaw, right? this is fasethernet and 4-5 clients in LAN |
| 16:49.06 | bjohnson | ManxPower: seems man -k alsa gives me info on fc3 but not on the fc2 machine |
| 16:49.26 | ManxPower | bjohnson, then alsa is not installed on the fc2 machine |
| 16:49.38 | ManxPower | fa, it does |
| 16:49.41 | Zeeek | alsa is schmaltzy |
| 16:49.43 | ManxPower | fa, it does NOT matter. |
| 16:50.08 | fa | ManxPower when, that have a matter? |
| 16:50.26 | bjohnson | it has alsa-lib-1.0.3a-2 and alsa-utils-1.0.3-1 |
| 16:51.03 | ManxPower | bjohnson, rpm -qil alsa-utils-1.0.3-1 | less |
| 16:51.53 | LarsAC | someone using florz' patch for the hfc drivers ? |
| 16:52.42 | allgood | anybody knows if the patch from http://bugs.digium.com/bug_view_page.php?bug_id=0003241 can be applied against 1.0.5? |
| 16:52.45 | file[laptop] | I'm blinded, BLINDED! |
| 16:53.07 | bjohnson | ManxPower: nothing that looks like a config script |
| 16:53.58 | bjohnson | ManxPower: found system-config-soundcard - A graphical interface for detecting and configuring soundcards .. but I don't have a graphical system on that machine |
| 16:54.16 | fa | I want to make a groups of iax2 users, i must us agents to that? |
| 16:54.19 | ManxPower | try running it anyway. |
| 16:54.28 | Grooby | alsa-mixer? |
| 16:54.34 | Grooby | for volume control |
| 16:54.43 | ManxPower | bjohnson, A lot of the RH/Mandrake config scripts support console as well as X |
| 16:54.53 | file[laptop] | 26 minutes have passed since food was ordered. |
| 16:55.07 | Beirdo | Mmmm. food |
| 16:55.17 | ManxPower | if you were running Mandrake "urpmi -y alsaconf" would tell you which RPM to install |
| 16:55.20 | fa | last food, i see 18hours ago |
| 16:59.06 | george_ | anyone here familiar w/ kphone? |
| 16:59.21 | bjohnson | system-config-soundcard requires a currently running X server. |
| 16:59.27 | zno | where can I download ringtones? |
| 16:59.49 | bjohnson | alsamixer showed the volumes down .. I upped them but still no output from mpg123 anyfile.mp3 |
| 17:00.21 | george_ | I'm not getting any message-waiting indication and I don't know if I should expect one... |
| 17:03.23 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 17:03.25 | LarsAC | florz: ping |
| 17:03.39 | Grooby | bjohnson, are they muted? |
| 17:03.54 | Derkommissar | why doesnt VoiceMailMain2(s${CALLERIDNUM}) |
| 17:03.59 | Derkommissar | work anymore ? |
| 17:04.21 | *** join/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 17:04.28 | Derkommissar | Feb 1 06:02:23 WARNING[6253]: pbx.c:1294 pbx_extension_helper: No application 'VoiceMailMain2' for extension (from-ip, 199, 1) |
| 17:04.49 | blitzrage | VoicemailMain |
| 17:05.08 | blitzrage | file[laptop]: !!! |
| 17:05.26 | file | blitzrage!!! |
| 17:05.28 | Derkommissar | but voicemailmain doesnt reconize the user by the callerid number |
| 17:05.33 | fa | can somebody help me with aganets. I have a iax users. now i am logging by firefly as user/peer for iax2. how can i login as agent? |
| 17:05.34 | Derkommissar | i always use voicemailmain2 |
| 17:05.35 | Grooby | spanish flea! |
| 17:05.45 | blitzrage | lol |
| 17:05.54 | *** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com) |
| 17:06.16 | *** join/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 17:07.20 | bjohnson | god DAMN .. Grooby gets the prize |
| 17:07.22 | Derkommissar | without voicemailmain2, how does voicemailmain reconize the user by callerid, without asking for the password ? |
| 17:08.09 | blitzrage | voicemailmain == voicemailmain2 |
| 17:08.13 | *** join/#asterisk magictux (~mark@rrcs-24-123-52-170.central.biz.rr.com) |
| 17:08.20 | bjohnson | Derkommissar: I thought they were the same command |
| 17:08.50 | Derkommissar | I though VoiceMailMain asked for the mailbox and password |
| 17:09.06 | bjohnson | Derkommissar: just password if you specify the mailbox |
| 17:09.14 | Derkommissar | :-/ |
| 17:09.24 | Derkommissar | how can i set it not to ask anything |
| 17:09.27 | Grooby | w00t |
| 17:09.43 | Derkommissar | i remember just having VoiceMailMain2(s${CALLERIDNUM}) |
| 17:09.50 | Derkommissar | and that would be it |
| 17:10.04 | bjohnson | Derkommissar: I don't know if that's possible/wise .. maybe set the password to nothing? |
| 17:10.24 | *** join/#asterisk |dennis| (~dennis@vsat-148-64-30-39.c050.t7.mrt.starband.net) |
| 17:10.43 | bjohnson | Derkommissar: show application VoiceMailMain |
| 17:10.51 | bjohnson | Derkommissar: If the mailbox is preceded by 's' then the password check will be skipped |
| 17:11.37 | Derkommissar | nope its not working rigth now |
| 17:13.31 | *** join/#asterisk dtwilson_ (~dave@cpc1-blfs2-5-0-cust219.blfs.cable.ntl.com) |
| 17:13.57 | dtwilson_ | hello all |
| 17:14.05 | dtwilson_ | having some issues with iax2 here |
| 17:14.30 | dtwilson_ | namely iaxphone -> nat -> internet -> nat -> asterisk |
| 17:14.51 | dtwilson_ | have forwarded ports at each end |
| 17:15.03 | dtwilson_ | but still can't seem to register |
| 17:15.09 | dtwilson_ | any ideas? |
| 17:16.40 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:19.23 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:20.05 | randu | When I call in to asterisk box it always cuts off the first few seconds of the greeting. ie Welcome to Company Name. The Greeting Starts at Name. Any ideas why? |
| 17:20.22 | `Sauron | randu: sip connection? |
| 17:20.31 | |dennis| | need help>using sarge with kernel 2.6.8-10> I am having problems compiling zapatel... |
| 17:20.34 | bjohnson | dtwilson: works for me all the time (only servers with dynamic IPs need to register .. only needs forwarded ports at the side that does not register) |
| 17:20.36 | `Sauron | There was a thing on the tips and tricks page talking about adding a 1 second delay to the beginning |
| 17:20.38 | `Sauron | so |
| 17:20.44 | `Sauron | exten => whatever,1,Wait(1) |
| 17:21.00 | `Sauron | exten => whatever,2,VoiceMail(blah) |
| 17:21.05 | `Sauron | or whatever your second step is |
| 17:21.27 | `Sauron | actually |
| 17:21.32 | `Sauron | 2nd step is answer |
| 17:21.35 | `Sauron | blah |
| 17:21.37 | `Sauron | 1. answer |
| 17:21.38 | `Sauron | 2. wait |
| 17:21.40 | `Sauron | 3. greeting |
| 17:21.43 | outtolunc | should be ,1,answer ,2,wait(x) ,3,playback |
| 17:21.54 | outtolunc | hehe |
| 17:22.20 | |dennis| | need help>using sarge with kernel 2.6.8-10> I am having problems compiling zapatel... |
| 17:22.35 | Mike | hey guys im trying to conect my GS with ilbc to my asterisk server and call 1800 numbers using sipphone with ulaw and i hear terrible noice and get errors like Feb 1 11:20:50 NOTICE[14197]: chan_sip.c:2773 process_sdp: No compatible codecs! any ideas? |
| 17:22.36 | `Sauron | http://www.voip-info.org/wiki-Asterisk+tips+answer-before-playback |
| 17:22.51 | greg_work | |dennis|, what does it say |
| 17:22.58 | `Sauron | ^^^ for randu |
| 17:23.29 | ManxPower | Mike, remove all bandwidth= lines Make sure you have only disallow=all and allow=ulaw OR allow=ilbc NO OTHER ALLOW LINES |
| 17:23.32 | greg_work | |dennis|, if you want the lazy way, put deb http://updates.xorcom.com/rapid sarge main in /etc/apt/sources.list, they have zaptel packages for 2.6.8 |
| 17:23.36 | randu | `Sauron: Thanks! |
| 17:23.46 | dtwilson_ | bjohnson: I've forwarded udp 4569 to our * box in the office and have udp 4569 forwarded on my home oruter to my workstation - I don't see any IAX2 traffic occurring on the * box whatsowever |
| 17:23.47 | `Sauron | no prob |
| 17:24.05 | `Sauron | I'm just glad I'm actually able to help, even if I've used * less than a week :) |
| 17:24.17 | dtwilson_ | am thinking I must have something wrong in my iax.conf but have been searching to no avail for the past few hours |
| 17:24.23 | shido6 | um |
| 17:24.25 | |dennis| | greg_work thanks but i prefer to compile from scratch......shall let you know of the error in a min... |
| 17:24.32 | Mike | ManxPower, on the sipphone context? |
| 17:24.33 | shido6 | 4569 and 5036 |
| 17:24.38 | shido6 | if youre gonna forward |
| 17:24.46 | shido6 | but u really shouldnt have to if u use register and host=dynamic |
| 17:24.49 | ManxPower | Mike, in [general] in sip.conf of course. |
| 17:24.55 | dtwilson_ | shido : surely 5036 is only for IAX? |
| 17:24.56 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 17:24.59 | greg_work | how do you control how many calls an IAX trunk can handle before being congested? |
| 17:25.03 | shido6 | thats what you trying to use |
| 17:25.05 | shido6 | isnt it? |
| 17:25.13 | Mike | ManxPower, can i paste you what i have? |
| 17:25.14 | dtwilson_ | IAX2 sorry shido |
| 17:25.16 | ManxPower | greg_work, "show application setgroup" also see the Wiki |
| 17:25.17 | shido6 | i came in on the tail end |
| 17:25.27 | ManxPower | Mike, I would never use a piece of shit GS phone. |
| 17:25.27 | shido6 | u need to open 5036 and 4569 |
| 17:25.30 | dtwilson_ | yeah no probs :) |
| 17:25.36 | shido6 | and why are you not using iax2? |
| 17:25.49 | dtwilson_ | do i need a register line pointing to each iax client? |
| 17:26.02 | dtwilson_ | i am using iax2 |
| 17:26.11 | Mike | ManxPower, its the only phone supporting ilbc for the moment |
| 17:26.21 | ManxPower | dtwilson, Devices with dynamic IP addresses register to devices with static IP address. |
| 17:26.31 | Zeeek | dtwilson : |
| 17:26.32 | ManxPower | Mike, if you cannot follow my simple instructions I cannot help you further. |
| 17:26.33 | Zeeek | http://willypick.mindsay.com/?entry=10 |
| 17:26.41 | dtwilson_ | well all points have static ips |
| 17:26.49 | ManxPower | dtwilson, Then don't bother to register! |
| 17:26.58 | Zeeek | The asterisk config that dare not speak its name: Double NAT! |
| 17:27.10 | bjohnson | dtwilson: which side has a dynamic IP address? |
| 17:27.23 | Mike | ManxPower, i was talking about GS in general, its the only phone supporting ilbc |
| 17:27.23 | ManxPower | Mike, Once you get it working with your GS phone you can modify things to not break all your other phones. |
| 17:27.55 | Juggie | whats going on with * and silence detection these days? |
| 17:28.05 | dtwilson_ | bjohnson: neither side has dynamic - although I want to treat the two client ends as dynamic so they can work from anywhere else as well - one is a laptop |
| 17:28.19 | Zeeek | dtwilson see my link |
| 17:28.28 | bjohnson | dtwilson: one side HAS to have a fqdn or static IP |
| 17:28.35 | Mike | - Got SIP response 488 "Not Acceptable Media" back from 198.65.166.131 |
| 17:28.51 | bjohnson | dtwilson: only dynamic IPs need to register |
| 17:28.59 | ManxPower | sounds like your GS phone is not using ilbc |
| 17:29.11 | outtolunc | juggie what cvs you using? |
| 17:29.18 | dtwilson_ | Zeek: perfect thanks - I've two wrt54g's at the client ends :) |
| 17:29.18 | ManxPower | dtwilson, Um, if the IP address chages then it's DYNAMIC. |
| 17:29.47 | bjohnson | dtwilson: if a server is receiving a register, it need host=dynamic in the conf .. otherwise host=whatever the fqdn or IP of the other machine is |
| 17:29.48 | shido6 | gs ilbc is crappy |
| 17:29.51 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:29.54 | shido6 | use g729 or ula |
| 17:29.55 | shido6 | w |
| 17:30.08 | dtwilson_ | bjohnson thanks, yeah i understand that |
| 17:30.17 | ManxPower | I am Ula of the Tribe Codec! |
| 17:30.25 | shido6 | heh |
| 17:30.30 | bjohnson | dtwilson: next, authentication .. if you want 2 way traffic .. each side has to authenticate against the other |
| 17:30.51 | bjohnson | dtwilson: esiest is to use the same user/secret on both machines |
| 17:31.05 | bjohnson | dtwilson: start with type=friend on both machines |
| 17:31.35 | greg_work | ManxPower, thanks |
| 17:32.36 | ManxPower | "I don't want to be a part of any organization that would have me as a member." --WC Fields (I think) |
| 17:32.46 | bjohnson | just so I'm clear .. codec quality vs reduced bandwidth order preferences seem to be ulaw, g729 if available or gsm, speex |
| 17:33.57 | mutilator | MMMMmmmmm |
| 17:33.57 | *** part/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 17:34.01 | mutilator | gatorade goood |
| 17:35.07 | Zeeek | ~seen JudgeCrater |
| 17:35.13 | jbot | i haven't seen 'judgecrater', Zeeek |
| 17:35.18 | Zeeek | heh |
| 17:35.30 | Zeeek | neither has anyone else for about 100 years |
| 17:35.43 | Zeeek | ~seen TheLight? |
| 17:35.44 | jbot | Zeeek: i haven't seen 'thelight' |
| 17:35.50 | Zeeek | that's obvious |
| 17:36.20 | fa | Zeeek IAX user can by autlogin as agent? |
| 17:37.32 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 17:38.48 | |dennis| | greg_work > firstly, I changed the line KERNEL_SOURCE?=/lib/modules/`uname -r`/build to KERNEL_SOURCE?=/usr/src/linux, /usr/src/linux points to /usr/src/kernel-source-2.6.8 which contains the actualsrc files. on compiling i get these errors> http://www.shc.edu.bz/dennis/error.txt |
| 17:40.07 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:40.39 | Mike | ManxPower, i think the problem is the bridge call |
| 17:40.47 | Mike | because using iax2 --> sip |
| 17:40.52 | Mike | doenst make the strange noise |
| 17:40.57 | Mike | but when is sip --> sip |
| 17:41.00 | Mike | it does |
| 17:41.58 | |dennis| | please help!> firstly, I changed the line KERNEL_SOURCE?=/lib/modules/`uname -r`/build to KERNEL_SOURCE?=/usr/src/linux, /usr/src/linux points to /usr/src/kernel-source-2.6.8 which contains the actualsrc files. on compiling i get these errors> http://www.shc.edu.bz/dennis/error.txt |
| 17:43.35 | Juggie | outtolunc, i am on the stable branch, not CVS... is there some silence suppresion stuff in CVS? |
| 17:43.38 | greg_work | if there are any devs around ... ast_variable_retrieve() just grabs info from an ast_config struct, and doesn't look at the database at all, does it? |
| 17:43.57 | fa | is it possible to specify in queues.conf recording a call, not in agents? |
| 17:44.04 | bjohnson | ~seen mycrack |
| 17:44.05 | jbot | bjohnson: i haven't seen 'mycrack' |
| 17:44.13 | bjohnson | and you're not going to either |
| 17:44.21 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 17:44.50 | Zeeek | eewwwww |
| 17:45.15 | *** join/#asterisk zno (~chatzilla@160.79.174.101) |
| 17:45.16 | Zeeek | should be yourcrack anyway |
| 17:46.07 | greg_work | is there a way to set voicemail options like saycid, envelope, from within VoicemailMain()? (ie, the user can do it in IVR) |
| 17:46.07 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:46.56 | greg_work | |dennis|, i had that same problem the other day .. |
| 17:47.18 | *** join/#asterisk zoa (~zoa@ip-212-239-162-97.dsl.scarlet.be) |
| 17:47.30 | greg_work | it was something really simple (i forget now).. you just need to find the error that's just before all those dereferencing lines |
| 17:48.09 | |dennis| | thanks greg..shall see i can get the pull it out..:) |
| 17:48.46 | greg_work | if you're using ssh, just increase your scrollback.. or you can output to a file |
| 17:49.03 | greg_work | or a gui shell |
| 17:49.36 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 17:52.18 | |dennis| | increased scroll back to 5000 lines but still looking...need to increase it....:) |
| 17:53.47 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 17:53.58 | LUTOR_ASI | hi, |
| 17:54.44 | PBXtech | why does it show "from Asterisk" with co callerID? can that be changed |
| 17:54.49 | zoa | it can be changed |
| 17:54.52 | zoa | in sip.conf i thin |
| 17:54.53 | zoa | k |
| 17:54.54 | *** join/#asterisk jlewis (~jlewis@solo.atlantic.net) |
| 17:54.57 | LUTOR_ASI | i have a connection from asterisk to Broadvoice.com, and i get a voice delay of 5 seconds..anyone can help me... |
| 17:55.23 | PBXtech | useragent? |
| 17:56.13 | greg_work | |dennis|, actually i think i had to hit ctrl+c to get it :p |
| 17:56.18 | greg_work | took me a couple tries |
| 17:56.45 | PBXtech | zoa, do you know what the line is? |
| 17:57.09 | jlewis | if I want to have callerid on internal (ext to ext) calls be the person's name and extension, but on calls out to the PSTN have it be our 800 number, whats the best way to do that?...set internal callerid in each person's sip.conf entry, and then setCIDNum(800number) in our outbound dialing contexts? |
| 17:57.48 | *** join/#asterisk cjk (~cjk@80.92.75.32) |
| 17:59.26 | zoa | no sorry |
| 17:59.39 | randu | LUTOR_ASI: I would suggest a different proxy. I am not sure if that will help or not |
| 18:01.34 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 18:01.38 | PakiPenguin | hello everyone |
| 18:01.44 | DrmC | jlewis if that works yes |
| 18:01.45 | *** join/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 18:02.19 | PakiPenguin | how to accept calls from one ip , like i have a gateway , i just want to accept incoming calls from that gateway , without any login / password |
| 18:02.22 | DrmC | is it even possible to change the CIDnumber on non trunked PSTN circuits? |
| 18:02.34 | |dennis| | greg_work > In file included from /usr/src/phone/zaptel/zaptel.c:40: |
| 18:02.35 | |dennis| | /usr/src/phone/zaptel/zconfig.h:10:27: linux/version.h: No such file or directory |
| 18:02.35 | |dennis| | /usr/src/phone/zaptel/zconfig.h:66:41: missing binary operator before token "(" |
| 18:02.58 | |dennis| | I did a ln -s /usr/src/linux linux in the zapatel dir... |
| 18:03.50 | *** part/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 18:06.17 | ctooley | Ok, this is getting really irritating. My extensions.conf keeps getting overwritten on some kind of schedule with older data |
| 18:06.27 | greg_work | |dennis|: you have linux-kernel-headers installed? thats looking for /usr/include/linux/version.h |
| 18:06.31 | dtwilson_ | hmmm isp having problems atm which is the cause of my earlier iax2 woes |
| 18:06.41 | DrmC | PakiPenguin its my understanding that you need a login .. however it doesnt need a pass ( i could be wrong on this ) |
| 18:06.43 | dtwilson_ | server cant see the clients |
| 18:06.47 | ctooley | with asterisk running I make changes, run reload, then run "show channels" and the changes show up |
| 18:06.54 | greg_work | |dennis|: i'm not sure why you had to link your kernel source like that, you shouldn't have to |
| 18:07.34 | ctooley | But, later the changes get reverted in both the running dialplan and the extensions.conf |
| 18:08.21 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 18:08.58 | *** join/#asterisk echion (~rickard@c213-100-37-165.swipnet.se) |
| 18:09.02 | DrmC | ctooley do you have any sort of config UI installed? |
| 18:09.11 | echion | hi, is there a way that ztcfg could damage a pri card? |
| 18:09.11 | Nugget | ctooley: I trust you're aware of the "writeprotect=yes" directive for extensions.conf? |
| 18:09.33 | DrmC | echion describe 'damage' |
| 18:09.40 | ctooley | DrmC, Nugget No config UI and "writeprotect" is set to yes |
| 18:09.43 | echion | i ran a ztcfg -vvvv on a card, the machine hanged and now the pri just won't go up again |
| 18:09.49 | Nugget | strange |
| 18:10.09 | ctooley | [general] |
| 18:10.09 | ctooley | static=yes |
| 18:10.09 | ctooley | writeprotect=yes |
| 18:10.14 | ctooley | very 1st 2 lines |
| 18:10.17 | ctooley | err 3 |
| 18:10.19 | echion | the driver says yallow alarm on span x |
| 18:10.23 | DrmC | echion card type? |
| 18:10.44 | echion | driver is wct4xxp |
| 18:11.09 | LUTOR_ASI | ok, thanks, randu.. |
| 18:11.18 | DrmC | i assume you have done the normal reseating of everything |
| 18:11.44 | echion | have a suggestion of how i get it running again? |
| 18:12.03 | *** join/#asterisk RoyK (~roy@i171219.dsl.tjukkband.no) |
| 18:12.14 | *** join/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 18:12.36 | fa | how to load PGSQL? |
| 18:12.44 | DrmC | well as i was eluding to i would start by checking layer1 .. reseat cards and connectors etc |
| 18:13.02 | ctooley | DrmC, Nugget, any suggestions? |
| 18:13.06 | DrmC | fa what do you mean load |
| 18:13.36 | fa | use.. module PGSQL |
| 18:13.43 | echion | so pull the cables from the pri, run ztcfg again? |
| 18:13.49 | DrmC | ctooley are you sure you are not using any sort of auto config generator or GUI? |
| 18:13.51 | ctooley | fa: do you mean how do you take your current configuration files and populate a PGSQL database with the configuration? |
| 18:14.07 | fa | ctooley that's right. yes. |
| 18:14.07 | ctooley | DrmC, nothing that I'm aware of. |
| 18:14.29 | DrmC | echi yes and first reseat the pri card in the mother board |
| 18:14.35 | ctooley | fa: I know there's a MySQL database script, it could probably be easily modified to handle PGSQL |
| 18:14.35 | fa | ctooley ? |
| 18:14.43 | echion | DrmC, how do I do that? |
| 18:14.58 | fa | ctooley but i read on mailing list, that PGSQL exzists, but i don't know where to download it |
| 18:14.59 | DrmC | unscrew it remove it replace it re screw it |
| 18:15.01 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 18:15.11 | echion | DrmC, ah oh well :-) |
| 18:15.22 | DrmC | =] |
| 18:16.44 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 18:17.00 | buddah | when i do a reload in asterisk, will that clear out any active calls? |
| 18:17.08 | DrmC | yes |
| 18:17.11 | buddah | hmm |
| 18:17.17 | buddah | what about just extensions reload |
| 18:17.19 | buddah | that do it too? |
| 18:17.21 | DrmC | er |
| 18:17.30 | DrmC | reload may not .. a restart will |
| 18:18.01 | buddah | i know restart will |
| 18:18.08 | zoa | reload does not |
| 18:18.12 | zoa | but if you have lotsa users |
| 18:18.14 | buddah | just thinking reload is doing that, got a customer complaining about dropped calls |
| 18:18.16 | zoa | it might cause some glitches |
| 18:18.18 | buddah | and it happens around that |
| 18:18.23 | buddah | when i'm reloading |
| 18:18.36 | buddah | 54 users considered a lot? |
| 18:18.52 | DrmC | how many active calls at once? |
| 18:18.54 | zoa | no |
| 18:18.56 | buddah | k |
| 18:19.20 | buddah | anywhere from 25-54 active calls at once |
| 18:19.22 | buddah | generally |
| 18:19.55 | *** join/#asterisk jjg (tink@216.253.86.223) |
| 18:20.18 | jjg | anyone tried creating a nufone account lately? |
| 18:20.44 | jjg | it doesn't appear to be correctly adding accounts, it won't move pass the "debit my card" step |
| 18:21.40 | buddah | they were having issues yesterday |
| 18:21.45 | buddah | maybe they still are |
| 18:22.11 | RyanE | I got a Perl error yesterday starting registration, so definitely problems |
| 18:24.05 | outtolunc | jjg: ouch |
| 18:24.14 | outtolunc | paypal one is screwed also <G> |
| 18:24.53 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 18:26.03 | jarrod | whats a good provider that i can use with asterisk(iax/sip) that i can receive did's in multiple states from? |
| 18:26.13 | *** join/#asterisk JakBeatZ (~JakBeatZ@216.7.194.254) |
| 18:26.29 | JakBeatZ | Folks, is it possible to force asterisk to unregister a SIP peer? |
| 18:27.02 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 18:27.20 | Zeeek | jarrod a long list has been prepared on the wiki |
| 18:27.35 | Zeeek | voicepulse nufone voipjet... |
| 18:28.07 | jarrod | i didnt see where to receive multiple dids in other states from nufone |
| 18:28.10 | jarrod | but thank you for the others |
| 18:28.24 | machinehd | building a new server for * can anyone recommend a good sata raid card? |
| 18:28.27 | Zeeek | erm maybe voipjet doesn't do DID now - but check Voicepulse |
| 18:28.30 | wasim | machinehd: 3ware |
| 18:28.59 | machinehd | wasim, do they have good linux drivers? |
| 18:29.30 | wasim | machinehd: prolly the best out there |
| 18:29.49 | jarrod | Please enter the serial number and MAC address found on the bottom of your VoicePulse-Ready SPA. |
| 18:30.02 | jarrod | voice pulse requires a voicepulse ready spa |
| 18:30.05 | file[laptop] | you want voicepulse connect |
| 18:30.11 | file[laptop] | http://connect.voicepulse.com/ |
| 18:30.14 | jarrod | werd |
| 18:30.16 | machinehd | I'm pretty noobish with server hardware. If I want hotswappable drives, what else do I need? |
| 18:30.17 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-8-169.w82-122.abo.wanadoo.fr) |
| 18:30.33 | outtolunc | ~seen jerjer |
| 18:30.35 | jbot | jerjer <~JerJer@dsl-107-53.che.centurytel.net> was last seen on IRC in channel #asterisk, 2d 16h 58m 52s ago, saying: 'then go have a coke and a smile'. |
| 18:30.37 | wasim | machinehd: hotswappable drive bays |
| 18:31.01 | wasim | machinehd: and hotswap support, which is pretty flaky, iirc |
| 18:31.09 | machinehd | wasim, so just make sure I get a case that supports hotswap drive bays? There's no other card required? |
| 18:31.38 | wasim | machinehd: unless you're doing scsi |
| 18:31.56 | *** join/#asterisk DaLion (anon@Toronto-HSE-ppp3771251.sympatico.ca) |
| 18:33.41 | wasim | machinehd: something like http://www.pogolinux.com/storage/index.html |
| 18:35.32 | jarrod | dcd |
| 18:36.10 | machinehd | wasim, great thanks. So hotswap is flaky atm? |
| 18:36.23 | wasim | machinehd: unless you've got good drivers |
| 18:37.03 | Zeeek | I'll have to remember to tell people to go the the bottom of the home page and click voicepulse connect |
| 18:37.14 | Zeeek | I know they have limited area codes though |
| 18:37.28 | machinehd | wasim, any idea if fedora/hotswap work well together? |
| 18:37.52 | wasim | machinehd: its prolly a kernel option, not a fedora specific at all, its all linux after all |
| 18:38.13 | Damin | Strange question.. |
| 18:38.25 | Damin | Does anyone know if Port Adapters in a VIP2-50 card are hot swappable? |
| 18:38.30 | *** join/#asterisk juice (~juice@mo-65-41-219-110.dyn.sprint-hsd.net) |
| 18:39.27 | Manipura | Since its a pain in the ass for me to test, I thought I'd ask first... Anyone have any problems with nat='yes' when they aren't in a nat? |
| 18:40.05 | multrix | does it exist big racks of 24 or more ATA converters ?? |
| 18:40.36 | multrix | for exemple a customer wants to keep its traditional phones, but wants to put its PABX out |
| 18:40.42 | vaewyn | multrix: called a channel bank |
| 18:40.59 | wasim | uberata! |
| 18:41.01 | multrix | rather than putting a little SIP ATA on each desk, a big one in the central |
| 18:41.07 | vaewyn | They come with T1 E1 or SIP interfaces |
| 18:41.16 | wasim | or IAX, we wish ... |
| 18:41.17 | multrix | vaewyn: where could I find this ? :) |
| 18:42.01 | vaewyn | multrix: look for stuff made by channel access... and devices like adit 600 and such |
| 18:42.20 | vaewyn | 'adit 600' on ebay will usually give you a couple good models to look at |
| 18:43.01 | tzanger | multrix: if you don't need 2 T1s worth of channels and/or don't need FXO, look for "Access Bank I" (substutute I for 1, also try ABI or AB1) |
| 18:44.26 | multrix | actually, I want to put for exemple 24 analog phone, and make them work like IP phones |
| 18:44.49 | multrix | a think with one RJ45 and 24 RJ11 practically |
| 18:45.32 | vaewyn | There are a couple newer channel banks that talk SIP... but you won't find one used... and they are $$$$ |
| 18:45.37 | *** join/#asterisk nicolasg (~chatzilla@ip-189.houseware.com.ar) |
| 18:46.23 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 18:46.36 | *** join/#asterisk freat[laptop] (~freat[lap@65.170.62.117) |
| 18:47.12 | *** join/#asterisk xilch (~xilch@66.239.17.228.ptr.us.xo.net) |
| 18:49.49 | PakiPenguin | hello everyone |
| 18:49.55 | PakiPenguin | can anyone help me with some iax problem |
| 18:50.18 | zno | is there an example of setting up a hunt group anywhere? |
| 18:50.30 | PakiPenguin | i see requests coming in from my client ( showing IAX requests ) onto the server's console ( iax debug ) , but the client never registers :( |
| 18:51.04 | *** join/#asterisk HD (~Henk@82-136-197-93-mx.xdsl.tiscali.nl) |
| 18:52.21 | Luhiwu | PakiPenguin, does the iax server have more than one IP address? |
| 18:53.38 | *** join/#asterisk SirPrize (~blah@host-83-146-24-114.bulldogdsl.com) |
| 18:53.47 | wasim | zno: group=2 |
| 18:54.02 | wasim | zno: dial(zap/g2) |
| 18:54.12 | PakiPenguin | Luhiwu : yeah |
| 18:54.31 | PakiPenguin | Luhiwu: i just need 4569 in/out UDP/TCP from my firewall right? |
| 18:55.07 | wasim | PakiPenguin: just udp |
| 18:55.29 | *** join/#asterisk tty666 (1001@200.184.153.54) |
| 18:55.30 | PakiPenguin | wasim: tcp's open too , just incase :p |
| 18:55.33 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 18:55.33 | tty666 | hi all |
| 18:55.33 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.rr.com) |
| 18:55.58 | wasim | PakiPenguin: that's like keeping a spare can of diesel, in your petrol car |
| 18:56.22 | DaLion | yo |
| 18:56.24 | zno | if a hunt group is called, is there anyway of setting callerID to say that the person is dialing the hunt group? |
| 18:56.32 | DaLion | Paki yeah |
| 18:56.34 | PakiPenguin | i know , still cant take any chances , wasim , you had micronet @ isb? |
| 18:56.43 | DaLion | lol |
| 18:56.44 | tty666 | someone know DND i have try using with sip and BT-100, i'am dial *78, *79 and receive 404 |
| 18:56.46 | |Vulture| | zno: huh? |
| 18:56.55 | wasim | zno: setcallerid("Group") |
| 18:57.43 | wasim | PakiPenguin: had ... as in? |
| 18:57.43 | PakiPenguin | wasim: you still do? i thought you guys moved to lhr , i have micronet :( |
| 18:57.43 | |Vulture| | zno: but that will kill your inbound CID |
| 18:57.43 | zno | like let's say someone dials "Tech Support" whichi is a hunt group I'd like it to say something like From:555-1212 To: "Support" or something like that |
| 18:57.43 | DaLion | iax uses ONLY udp |
| 18:57.47 | DaLion | 4569 |
| 18:57.50 | DaLion | you should be fine |
| 18:57.55 | wasim | PakiPenguin: no, i never had micronet in Isloo, since hungama was so handy |
| 18:57.59 | zno | I guess I can get the callerid and append it |
| 18:58.09 | PakiPenguin | wasim: dialup :o :o!!! |
| 18:58.21 | PakiPenguin | http://pastebin.ca/5096 <-- iax debug |
| 18:58.28 | wasim | PakiPenguin: umm ... no, broadband and colo |
| 18:58.28 | |Vulture| | zno: then you want their CID # to be correct, but the CID Name to be "Support"? |
| 18:58.36 | Zeeek | "I wouldn't leave my kids alone with him, but I know he's innocent," |
| 18:58.59 | zno | well calleriD From the caller to be the same, but adding to which group it's going |
| 18:59.08 | DaLion | in a dialplan sits illegic to dial IAX for xx secs then try something else right ? if client needs iax temrination |
| 18:59.09 | |Vulture| | zno: SetCIDName("Support") |
| 18:59.10 | zno | for example if someone is in multiple hunt groups |
| 18:59.23 | zno | ah thanks |
| 18:59.42 | |Vulture| | zno: you could also make different extensions on the phones depending on where it is coming from |
| 18:59.47 | *** join/#asterisk ennuyeux7 (~ennuyeux7@83.146.53.34) |
| 18:59.53 | |Vulture| | like an extension for interal and external... |
| 19:01.01 | PakiPenguin | it works!!! YAY! |
| 19:03.04 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 19:04.53 | DaLion | the hell awas that |
| 19:05.22 | Darwin35 | net hickup |
| 19:05.29 | DaLion | btw u have to play moh fo ragents ? |
| 19:05.39 | DaLion | kind of sucks unless its radio |
| 19:05.57 | SirPrize | What file do I edit to change the format of the announcements made when you reach voicemail on Asterisk ? |
| 19:07.12 | calisto | wasim: hows it going with customs |
| 19:07.12 | ManxPower | SirPrize, For the most part you don't. Remove the file types you do not want to use. |
| 19:07.19 | wasim | calisto: success! |
| 19:08.06 | SirPrize | ManxPower: Specifically, I want to stop the voicemail system from announcing that an extension number is involved |
| 19:08.08 | calisto | wasim: good stuff.... when you thibk we might see reviews etc.. on list or elsewhere |
| 19:08.20 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.OneCall.Net) |
| 19:08.33 | ManxPower | SirPrize, Then record the busy and unavail message. |
| 19:08.51 | |Vulture| | yea peopl are dumb... |
| 19:09.05 | ManxPower | It only plays "the person at extension...." when the user was too lazy to record a custom busy and unavail greeting. |
| 19:09.09 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 19:09.34 | ManxPower | Directory() Uses the "Record your name" stuff to play back people's names. |
| 19:09.52 | DaLion | curently redirirecting home phone to my pbx... fuck those bad calls |
| 19:09.53 | calisto | wasim: you worried now the pa186 stuff has iax firmware |
| 19:09.57 | SirPrize | ManxPower: Ah, I see. thanks. |
| 19:10.00 | Delvar | nn all |
| 19:10.29 | ManxPower | calisto, when he ships this farfon's in qty they will blow away the PA186 stuff. |
| 19:10.48 | DaLion | Zap line dialpaln is 75 lines |
| 19:10.57 | DaLion | for one exten |
| 19:11.15 | calisto | manxpower: when !!!! .... if he's not missed the boat |
| 19:11.33 | ManxPower | calisto, nobody knows. |
| 19:12.07 | ManxPower | There was supposed to be a phone from te guys that do firefly too that never actually shipped in qty |
| 19:12.30 | Zeeek | looked good, too |
| 19:12.36 | file[laptop] | sounded good, too |
| 19:12.42 | Zeeek | but not even an email back fromthem after like 1 tyear |
| 19:13.14 | calisto | manxpower: it actually sounded better than farfon but no show |
| 19:13.22 | jpmcallister | What is the best way to distribute around 300 extensions fro * to common fxo telephones? |
| 19:13.46 | vaewyn | jpmcallister: channel banks |
| 19:14.23 | calisto | manxpower: i think farfon has missed the market by 6-9 months given when I first saw the details iax is now becoming more accepting and i don't think it will be too long before the medium level players get in on the act |
| 19:14.26 | jpmcallister | vaewyn: could you point me some documentation where I can study more about this? I'm new to telco, and never heard of it |
| 19:14.27 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
| 19:14.40 | wasim | calisto: yep |
| 19:14.56 | vaewyn | jpmcallister: there are tons of them... google for 'channel bank' |
| 19:15.05 | wasim | calisto: no, we're glad pa168 is doing iax stuff, the more the merrier |
| 19:15.14 | b0ef | I need a gsm phone that I can hook up to my computer. Anyone know of such a phone or even what the interface is called? |
| 19:15.26 | greg_work | damn, why do so many people have large numbers of analog phones? i though most phone systems used proprietary digital phones |
| 19:15.31 | vaewyn | jpmcallister: they normally take 24,48 or 96 analog connections and jam either T1/E1 or SIP network connections out |
| 19:15.36 | jpmcallister | vaewyn: thats is the problem, I'm lost with the tons of site googles returns |
| 19:15.41 | ManxPower | I never thought a phone like the PA186 would be the first to market with an IAX phone. |
| 19:15.45 | |Vulture| | greg_work: yea 300 regular phones is crazy for a business |
| 19:15.56 | calisto | wasim: so why if i was buying 1000 would I buy farfon given that should iax not make it the pa186 devices still do sip |
| 19:16.02 | vaewyn | jpmcallister: check around for the likes of 'adit 600' and such |
| 19:16.17 | jpmcallister | vaewyn: hmm, and could you indicate some vendors? |
| 19:16.23 | RyanE | if I have an wcfxo card on the same line as existing analog phones, is there a way for Asterisk to tell if an analog phone has been picked up? I want to start out just using it for IVR at home, and if I pick up a phone, I don't want Asterisk to go to voicemail. |
| 19:16.54 | Nugget | RyanE: just suck it up and put that phone on an FXS interface. :) |
| 19:16.56 | ManxPower | RyanE, No. Only that it stopped ringing |
| 19:17.14 | jpmcallister | vaewyn: many tanx, I'll check that |
| 19:17.34 | greg_work | |Vulture|: i just can't see how it would get to that point (unless theres a lot of PBXs that just use analog phones? i don't have a ton of experience with them). a few years ago our company had like 2 phones on every desk (well, there were like 3 or 4 desks).. it was crazy |
| 19:18.23 | greg_work | that was before i actually worked here fulltime (its a family business) .. but i said "seriously, time to get a phone system." "but this works just fine for us" "no, no it doesnt. you're getting a phone system" |
| 19:18.36 | tty666 | someone know DND i have try using with sip and BT-100, i'am dial *78, *79 and receive 404 |
| 19:18.48 | Zeeek | Manxpower which leads to taling on the phone while asterisk phone rings for another 10 secods |
| 19:19.01 | Zeeek | very uncomfortable if in the same offcie |
| 19:19.17 | Zeeek | nice typing tonight - and I haven't had a second six pack yet |
| 19:19.58 | ManxPower | Zeeek, I got tired of argueing with newbies. I just tell them how/what can be done and I let them go thru the living hell of finding out exactly why whatever it is is such a bad idea. |
| 19:20.37 | |Vulture| | greg_work: I've been working with small offices installing * and it has worked out weel going from merridian systems to Polycom + * |
| 19:20.37 | Zeeek | yes that'll work :) |
| 19:20.57 | calisto | manxpower: kudos people who don't RTFM need a good kick in the right direction |
| 19:21.09 | Zeeek | As search marketing gets more and more competitive, and pay per click costs |
| 19:21.09 | Zeeek | rise, blogging and news feeds will become a key marketing strategy. |
| 19:21.27 | Zeeek | good think there is an asterisk news RSS |
| 19:21.54 | zno | |Vulture| is the demand big for small businesses for a * solution? |
| 19:22.24 | calisto | zno: my guess is that at present thats the best target for * |
| 19:22.27 | greg_work | |Vulture|: we just got a bunch of spa-841's. i'm still doing some stuff with * before we roll it out though |
| 19:22.43 | Zeeek | Being a small business and knowing a lot of others... I can say YES |
| 19:23.04 | zoa | anyone here with cisco 7960 ? |
| 19:23.24 | zno | yeah asterisk is perfect for our 10 person company |
| 19:23.25 | calisto | zoa: i wish.. mway to many $$$$ |
| 19:23.25 | Zeeek | What is needed is more nice small hardware servers with swap in/out cards |
| 19:23.42 | Zeeek | even better for 2-3 people who move around a lot |
| 19:23.57 | Zeeek | I can answer my office phone at home or from another country |
| 19:24.08 | greg_work | what do you mean "swap in/out cards" ? hotswappable pci? |
| 19:24.24 | Zeeek | we're lucky the cellphones are ubiquitous too so breaking up means nothing now :) |
| 19:24.44 | Zeeek | no just a stock of the main boards - someone will need to service these things |
| 19:25.13 | Zeeek | my own box for example, sure I'll go buy what I need when the CPU dies or the mobo or whatevr but the avgerage owner can't |
| 19:25.35 | Zeeek | and will need someone to come in and make that swap in a couple of hours max |
| 19:25.58 | Zeeek | if the business is heavily phone related they may need a swap in BOX |
| 19:27.10 | *** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com) |
| 19:27.34 | Zeeek | the problem is that even with our humble 2xFXO + 1 TDM400 system, if the hardware bites it, I don't have spares |
| 19:27.58 | Zeeek | and if I did, it'd still take some time to reconnect those lines |
| 19:28.05 | greg_work | hey, can anyone here suggest a good network monitoring system? I want something simple, where I can just plug in a script as a check for something .. ie, is asterisk running, is this system pingable, etc... that sends me an alarm when theres a problem |
| 19:28.09 | wasim | Zeeek: a TDM400 on its own is really not much use |
| 19:28.35 | ctooley | greg_work, worked a lot with Nagios in the past but it's kind of complex to set up initially |
| 19:28.35 | Zeeek | not when it's sitting outside a box gathering dust, no |
| 19:28.44 | greg_work | yeah, i dont like the setup of nagios at all |
| 19:29.14 | greg_work | i'd rather have a web-based setup. its just easier to manage |
| 19:29.27 | *** join/#asterisk kingcobra (~martin@214.35.233.64.transedge.com) |
| 19:29.29 | ctooley | greg_work, you can use Nagat |
| 19:29.30 | wankel | thbbt. nagios is nothing. you should try setting up openview :P |
| 19:29.53 | ctooley | it hasn't been updated recently but it works pretty well for the 1.X versions |
| 19:29.54 | greg_work | file-based is fine, except the way nagios does it you have to relate service records or hosts or something, and theres just too many different files to edit and keep track of compared to how easy it should be. |
| 19:30.04 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 19:30.14 | ctooley | wankel, we're not all masochistic |
| 19:30.24 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 19:30.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:30.42 | wankel | we're still using nagios for some stuff here. we've had to hack the thing to pieces to get it to scale over home network size. |
| 19:30.50 | wankel | but it works nicely for small networks |
| 19:30.56 | Zeeek | are there cordless phones that can use headsets? |
| 19:31.02 | Zeeek | stupid question I know |
| 19:31.11 | wankel | yes |
| 19:31.17 | Zeeek | name it pls |
| 19:31.18 | wankel | at least we got rid of openview |
| 19:31.24 | Zeeek | oops |
| 19:31.24 | wankel | uh... siemens. |
| 19:31.24 | bjohnson | b0ef: 1. there is a cell phone adapter on voxilla 2. a bluetooth phone can be used with a bluetooth usb dongle (I think) 3. you can get a gsm fixed terminal |
| 19:31.35 | Zeeek | they make cordless with headset plug? |
| 19:31.39 | wankel | yes |
| 19:31.40 | greg_work | ctooley: yeah, even that is not that great. you know, if you're going to make it webbased, you could at least put better names. i dont understand half those options |
| 19:31.45 | wankel | lots of people do |
| 19:32.00 | Zeeek | see my base is on another floor |
| 19:32.10 | wankel | greg: maybe you still wouldn't understand them if they changed the name :) |
| 19:32.14 | Zeeek | but the cordless phone is to be used by someone who wants to be able to type |
| 19:32.17 | greg_work | (and yeah, i can rtfm. but its just one more thing to remember that you really shouldn't have to. its so easy to put descriptive names when you already have the gui .. ) |
| 19:32.46 | greg_work | wankel: maybe not. but its just bad design |
| 19:32.57 | greg_work | i say this as a web developer |
| 19:33.02 | zno | Zeeek: check out plantronics |
| 19:33.06 | zno | very expensive but cool |
| 19:33.15 | wankel | zeeek: siemens makes very nice multi-extension cordless systems. you can have multiple cordless sets (up to 8, i think) hanging off of one base station. each cordless set has its own charging cradle. |
| 19:33.21 | Zeeek | phones? I know they make headsets |
| 19:33.28 | wankel | but almost any nice cordless handset has a headset jack |
| 19:33.49 | Zeeek | really? We have old seimens phones now - they haven't lasted that long |
| 19:34.04 | wankel | my gigaset system is... i dunno, i guess about 5 years old now. |
| 19:34.05 | Zeeek | "Balloon shooter game.Up to 15 hours talk time and 250 hours standby time" handy! |
| 19:34.08 | ctooley | greg_work, yeah, well, it's also not being developed anymore. |
| 19:34.20 | Zeeek | ours are about 3-4 and they're worn out |
| 19:34.29 | Zeeek | balloon shooter game :) |
| 19:34.31 | wankel | the handset or just the batteries? |
| 19:34.37 | wankel | i had to replace the rechargables on mine twice |
| 19:34.39 | Zeeek | handset - buttons |
| 19:34.44 | Zeeek | display one one |
| 19:34.47 | Zeeek | on one |
| 19:34.49 | wankel | huh. haven't had any problems like that with mine. |
| 19:35.00 | wankel | i've only used the gigaset ones, though |
| 19:35.25 | Zeeek | these are gigasets |
| 19:36.24 | Corvin | hi have someone problems with zap channnel I mean silent pops in background? |
| 19:36.43 | bjohnson | Zeeek: what country? is $50USD too much? |
| 19:36.46 | b0ef | bjohnson: nice, I'll check it out; I will need to use my gsm phone to make voice calls aswell through asterisk preferably. Also to make connect my computer to the internet |
| 19:36.59 | Zeeek | No but I'm in France |
| 19:37.03 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.90) |
| 19:37.10 | Zeeek | Gigaset S440 |
| 19:37.11 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 19:37.14 | Zeeek | looking at that |
| 19:37.19 | *** part/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 19:38.04 | vaewyn | I'm a genius in france |
| 19:38.05 | bjohnson | Zeeek: how is shipping from UK? Look at the bluetooth headsets for non-bluetooth phones. I can walk you through a couple of options but my shopping has been in the states |
| 19:38.08 | vaewyn | :} |
| 19:38.20 | Zeeek | we don wan no stinking bluetooth |
| 19:38.35 | Zeeek | too much technology blur already |
| 19:38.53 | multrix | Zeeek: laisse tomber la voip en france c mort ;) |
| 19:38.57 | bjohnson | Zeeek: local discount store down the street has wired phones with callerid and headset jack for $20 .. through in a BT headset and adapter for $20-$50 and you have an ear mounted wireless rig |
| 19:39.06 | Zeeek | multrix pas du tout - tres actif |
| 19:39.15 | multrix | juste une blague ;) |
| 19:39.20 | Zeeek | multrix http://wengo.fr |
| 19:39.36 | Zeeek | oh but what sucks in Frane is the small range of phone hardware |
| 19:39.42 | multrix | Zeeek: oui j'aime bien wengo je vais voir :p |
| 19:39.51 | Zeeek | very limited comared to any store in the US |
| 19:39.54 | bjohnson | Zeeek: just got my first bt hardware and it is a pretty slick concept |
| 19:40.22 | Zeeek | "rig" ya I like that |
| 19:40.38 | Zeeek | wengo marche bien avec asterisk |
| 19:40.51 | bjohnson | wengo? |
| 19:40.54 | Zeeek | vaewyn what you're Jerry Lewis? |
| 19:40.59 | Zeeek | JerJer Lewis? |
| 19:41.01 | *** part/#asterisk Flyboy6440 (~Bobo@192.76.82.90) |
| 19:41.14 | Zeeek | wengo 6eu/month unlimited fixed dialing in France |
| 19:41.38 | Zeeek | fairly decent if you need to talk to France a lot :) |
| 19:41.50 | multrix | Zeeek: ouai mais moi j'ai un compte sip gratuit vers tous les fixes en france, mais aussi dans le monde, alors wengo sur le principe ca allait, mais kan j'ai eu ca j'ai abandonné l'inscription :p |
| 19:42.02 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 19:42.02 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:43.32 | multrix | Zeeek: et c ke du bonheur :p ca marche niquel, j'ai deja essayé vers la france, pologne, canada, norvege, republique czech :p |
| 19:43.45 | Zeeek | multrix if you want to monologue, do it in English |
| 19:43.58 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 19:44.36 | multrix | Zeeek: you work in voip ? |
| 19:44.48 | Zeeek | no |
| 19:44.59 | multrix | just a user |
| 19:45.02 | multrix | enduser |
| 19:45.14 | Zeeek | yes but we endusers are the most important |
| 19:45.30 | fa | ;] |
| 19:45.35 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 19:45.37 | tessier_ | Hello all! |
| 19:45.42 | fa | hello you |
| 19:45.43 | Zeeek | without us, nothing means anything - it's all just a cry in space -et dans l'espace, personne ne vous entend CHIER |
| 19:45.55 | Zeeek | muhahaha |
| 19:46.00 | tessier_ | Any good way to tell what channels are live conversations and which ones are hung? It would be nice if show channels showed how long the channel had been up. |
| 19:46.03 | fa | ;] |
| 19:46.13 | fearnor | endusers suck |
| 19:46.17 | fearnor | meh |
| 19:46.33 | fearnor | unfortunately they pay money. |
| 19:46.50 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
| 19:47.28 | multrix | Zeeek: I think that 2005 will be a very important year in voip actually ! but I hope that free software like Asterisk will win and stay open source ! |
| 19:47.35 | buddah | anyone used a linksys pap2 phone adapter? |
| 19:47.44 | *** join/#asterisk three55ml (~who@cs666898-229.austin.rr.com) |
| 19:48.02 | wasim | i hope someone makes the uberata in 2005 |
| 19:48.06 | Zeeek | multrix I'm sure asterisk will unless there is a hostile takeover by France Télécom |
| 19:48.08 | file | yes, there's a pap2-na sitting right beside me |
| 19:48.12 | file | in all it's pretty form |
| 19:48.24 | bjohnson | Zeeek: http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=42404&item=6362668921&rd=1 |
| 19:48.36 | three55ml | Anyone work with IAXClient at all? |
| 19:48.38 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 19:48.46 | fa | three55ml yes. I, little. |
| 19:48.47 | fearnor | uberata heh |
| 19:48.54 | buddah | file: where in the config do you put in the sip server address? |
| 19:48.56 | Zeeek | down with bluetooth |
| 19:49.03 | three55ml | Or any IAX2 in gerneral I guess. I pretty much always get a delay from anywhere from a few seconds to 10 minutes in actually registering with the server. |
| 19:49.05 | file | buddah: this is a pap2-na right? |
| 19:49.06 | fearnor | i hope someone makes a decent and reasonably-priced ip phone |
| 19:49.10 | buddah | oh, na? |
| 19:49.12 | *** join/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 19:49.12 | buddah | lemme look |
| 19:49.14 | fearnor | or an ip phone with opensores firmwarez |
| 19:49.17 | Grooby | zeeek, you use BT headset? |
| 19:49.20 | file | if it's a PAP2 it's locked... |
| 19:49.22 | multrix | Zeeek: do you run an asterisk at home ? |
| 19:49.34 | three55ml | I'm about to get out a packet sniffer and see if it's actually sending out the registration requests, because Asterisk with IAX debugging on shows nothing. |
| 19:49.37 | buddah | just says pap2 |
| 19:49.41 | Zeeek | at the office - two lines FT three FXS in theoffice |
| 19:49.45 | buddah | 'no |
| 19:49.46 | bjohnson | buddah: then it's locked |
| 19:49.49 | buddah | 'its pap2-na |
| 19:49.51 | Zeeek | and SIP and IAX outside |
| 19:49.53 | buddah | checked the device |
| 19:49.57 | buddah | the pap2 is the vonage right? |
| 19:50.09 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 19:50.10 | Grooby | so who here has used BT headset with softphones? |
| 19:50.10 | file | if it's an NA, you would go to the web server on it's IP, go to the line and fill out the details in the Proxy and Registration section |
| 19:50.14 | bjohnson | buddah: yes |
| 19:50.16 | tessier_ | I would love to see someone mass-produce an open IP phone. |
| 19:50.20 | Grooby | my BT headset w/ my ecellphone sucks |
| 19:50.27 | buddah | yeah |
| 19:50.27 | tessier_ | Unfortunately the investment is more than anyone here is going to front. |
| 19:50.37 | bjohnson | Grooby: just got a test rig (supposed to be present for wife) |
| 19:50.40 | three55ml | tessier: The Snom ones have opensource firmware I believe, but a lot of people don't like them for various reasons. |
| 19:50.41 | Zeeek | where is the best store to buy phones in paris? |
| 19:50.47 | buddah | file[laptop]: in the sip section? |
| 19:50.49 | tessier_ | three55ml: I know, I have a few. |
| 19:50.51 | multrix | Zeeek: IP phones ? |
| 19:50.51 | buddah | grr |
| 19:50.56 | Zeeek | no analmog |
| 19:50.58 | buddah | file: in sip section? |
| 19:51.01 | Grooby | bjohnson, let me know how that goes....i got 2 BT headsets here..haven't had the time to play with it |
| 19:51.01 | tessier_ | analmog? |
| 19:51.06 | file | Line 1 -> Proxy and Registration |
| 19:51.07 | buddah | file: or line 1? |
| 19:51.09 | buddah | ok |
| 19:51.09 | file | Outbound Proxy |
| 19:51.12 | multrix | Zeeek: analog phones steel exist ??? what is it ? :D |
| 19:51.15 | three55ml | fa: Any idea with the IAX troubles? |
| 19:51.19 | file | is the address of your asterisk box... |
| 19:51.20 | buddah | yeah i got the ip in proxy |
| 19:51.21 | Zeeek | I have a free FXS line |
| 19:51.29 | buddah | set to register |
| 19:51.36 | fa | Zeeek when to use & when Dial, and when to use queue, hu? |
| 19:51.36 | buddah | got the user id/pass set in it |
| 19:51.39 | bjohnson | Grooby: played with it for about an hour yesterday .. works well .. will take some practice getting on |
| 19:51.44 | buddah | says it cant connect to login server |
| 19:51.45 | file | then it should work fine |
| 19:52.02 | multrix | ah yes I remember this think bell invented in 1876...... but I thought we could only find this in museums !!! :p |
| 19:52.08 | Grooby | so the voice quality is good? when I use mine with my SE 600, i get echos left and right |
| 19:52.10 | bjohnson | Grooby: based on that I've ordered a BT usb and BT headset for myself to play with |
| 19:52.24 | fa | three55ml no |
| 19:52.36 | fa | bjohnson when to use & when Dial, and when to use queue, hu? |
| 19:52.36 | *** join/#asterisk bacondoublechz (~bacon@69-162-37-142.stcgpa.adelphia.net) |
| 19:52.41 | buddah | i've noticed how many things that should 'work fine' but i cant get to work |
| 19:52.49 | buddah | heh |
| 19:52.52 | file[laptop] | is it behind NAT? |
| 19:52.52 | Zeeek | multrix aside from toy SIP phones or expensive Cisco jobs, regular phones are always useful, especially if asterisk pbx hardware dies |
| 19:53.06 | bjohnson | Grooby: Jabra BT200 for non-bluetooth phones (includes a small receiver that plugs into 2.5mm headset jack) .. seems to work well .. good voice quality |
| 19:53.06 | buddah | yeah |
| 19:53.11 | buddah | i can get it from outside nat |
| 19:53.17 | file[laptop] | set nat=yes in it's entry in sip.conf |
| 19:53.23 | buddah | k |
| 19:53.24 | bjohnson | fa: I still don't use queues |
| 19:53.24 | Grooby | ok..i'll have to yank out my other BT headset and try it out now |
| 19:53.26 | Grooby | :-D |
| 19:53.31 | multrix | Zeeek: wow I couldn't imagine if some asterisk were out of order !!! |
| 19:53.40 | file[laptop] | globally too... yeah globally, won't hurt |
| 19:53.51 | Zeeek | shit happens multrix, shit happens |
| 19:54.26 | multrix | Zeeek: shit happens but It's my work so it has no consequence ! |
| 19:54.46 | Zeeek | are you Batman or RObin then? |
| 19:55.04 | multrix | Zeeek: redundancy is my work |
| 19:55.06 | tzanger | hahaha |
| 19:55.17 | tzanger | stealth asterisk installs rock |
| 19:55.53 | Zeeek | redundancy is my work |
| 19:56.03 | bjohnson | Grooby: found a cheap US source of BT hardware if interested. Not sexiest things though. I've ordered a couple to test |
| 19:56.23 | `Sauron | Zeeek: I didn't hear it the first 20 times, could you repeat it again? |
| 19:56.24 | `Sauron | ;) |
| 19:56.31 | multrix | Zeeek: me too, repeat ! |
| 19:56.35 | Zeeek | redunancy... |
| 19:56.35 | `Sauron | bjohnson: What kind of BT hardware? |
| 19:56.42 | Zeeek | get it? |
| 19:56.45 | ManxPower | tzanger, Putting an Asterisk server between the PBX and the T-1's from the telco |
| 19:56.46 | Grooby | bjohnson, i bought my GF a BT that she can't use it |
| 19:56.50 | Grooby | so i am gonna use that |
| 19:56.50 | Grooby | :-D |
| 19:56.54 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 19:57.11 | `Sauron | I have this overwhelming urge to /msg dontmsgme |
| 19:57.18 | file[laptop] | `Sauron: so do I |
| 19:57.21 | *** part/#asterisk Caede (~chatzilla@204.94.248.81) |
| 19:57.21 | Zeeek | redunancy...siemens S100 looks nice - 6 days of autonomy |
| 19:57.34 | ManxPower | `Sauron, /msg me instead 8-) |
| 19:57.51 | multrix | tzanger: about stealth asterisk install, is it possible to deal with ip phones anyway without reconfiguring pbx ? :) |
| 19:57.59 | bjohnson | `Sauron: headset $10, usb dongle $16, usb phone line adapter $10 (this thing is interesting since it plugs directly into phone line .. could plug directly into a fxs) |
| 19:58.28 | bjohnson | `Sauron: shipping $12 .. I ordered a set of 3 to check quality |
| 19:58.46 | `Sauron | bjohnson: Hum.. usb FXS or usb FXO interface? |
| 19:58.59 | `Sauron | err |
| 19:59.02 | bjohnson | neither |
| 19:59.13 | `Sauron | yeah, I'm trying to figure out it's purpose |
| 19:59.16 | Nugget | I wish there was a good solution for a wireless headset on my cisco 7960. |
| 19:59.18 | shido6 | Sauron, eat any hobitses lately? |
| 19:59.37 | `Sauron | shido: Lots. I ate them for lunch. |
| 19:59.49 | ManxPower | Best with just a little garlic butter, huh? |
| 19:59.50 | bjohnson | sorry .. not usb phone line adapter .. BT phone line adapter . |
| 19:59.58 | `Sauron | Err, oh. Duh. |
| 20:00.25 | tzanger | ManxPower: been there, done that. works *great* |
| 20:00.34 | `Sauron | Now if chan_bluetooth could only support headset profiles, and not just handsfree |
| 20:00.35 | tzanger | multrix: what do you mean? |
| 20:00.46 | greg_work | so i'm writing this new voicemail agi replacement, which is pretty much a drop-in replacement for existing app_voicemail (uses same mailbox structure/format).. but i'm trying to figure out how to do config options. the existing app_voicemail rewrites voicemail.conf when you change your password, and doesnt allow you to change options. i'm going to have menu options to control "say caller id" "say envelope" etc.. should I make those |
| 20:00.46 | greg_work | wrote voicemail.conf, or set database variables? the latter is better technically, but will be incompatible with app_voicemail, and you couldn't look at voicemail.conf to see/set mailbox options for that user |
| 20:01.07 | bjohnson | `Sauron: http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=44999&item=5747305573&rd=1 |
| 20:01.15 | multrix | tzanger: example : PBX with 10 phones numbers : 10 to 20 |
| 20:01.18 | `Sauron | greg: Use ast_data, and store all the info in a database? |
| 20:01.29 | multrix | tzanger: you put a stealth asterisk |
| 20:01.35 | greg_work | i was also worried about locking issues.. if two people change their password at the same time, for example, one won't work |
| 20:01.37 | bjohnson | `Sauron: what does it support? I thought I'd use one with iaxcomm if I can get it working |
| 20:01.42 | multrix | and you want to add ip phones with 21 22... etc... |
| 20:01.58 | multrix | is it possible without configuring pbx ? |
| 20:02.29 | `Sauron | hum |
| 20:02.38 | `Sauron | bjohnson: So it's a BT FXO interface |
| 20:02.42 | `Sauron | Interesting. |
| 20:02.48 | greg_work | `Sauron: what would happen is in voicemail.conf you'd have a line like voicemail => 123,.....,saycid=yes but if the user has changed saycid, it will be off in the database (ie, voicemail.conf would say it's on) .. maybe that isn't even a big deal |
| 20:03.03 | bjohnson | `Sauron: yeah I guess .. with pass through |
| 20:03.04 | *** join/#asterisk xai (~pasta@user-0vvdb42.cable.mindspring.com) |
| 20:03.04 | tzanger | multrix: depends on the PBX. typically not |
| 20:03.20 | tzanger | multrix: becaues the PBX users (10-20) will want to hit an IP user and the PBX will need to know how to get there... |
| 20:03.23 | `Sauron | greg: When you use ast_data to pull info from pgsql/mysql, it doesn't use the file anymore |
| 20:03.35 | greg_work | the password is more of an issue, i'm not sure it's a good idea NOT to write voicemail.conf to change the password (in case other programs are reading voicemail.conf to get the password .. would also definately break app_voicemail) |
| 20:03.44 | greg_work | what is ast_data ? |
| 20:03.45 | InfraRed | 8 |
| 20:03.54 | `Sauron | search voip-info for it |
| 20:03.56 | greg_work | i was talking about db_put() |
| 20:04.02 | bjohnson | `Sauron: we have Nortel handsets at office with headset jacks (RJ11?) .. thought I'd see if I could rig it up for a BT headset |
| 20:04.12 | multrix | tzanger: I should say to the pbx " if 10 to 20 call 20 to 30, send this call to E1" ? |
| 20:04.19 | `Sauron | greg: I dunno 'bout db_put |
| 20:04.31 | Bonbon | anyone know what sort of indexing should be put on the realtime iax / sip tables? |
| 20:04.43 | Bonbon | i.e. what do the update queries look like? |
| 20:05.30 | greg_work | `Sauron: ah, but that's not in * right now. one of my main things of rewriting voicemail is to get the new features into 1.0.5 stable |
| 20:05.38 | tzanger | multrix: if your PBX will let you do that, sure |
| 20:05.40 | tzanger | multrix: most will NOT |
| 20:05.57 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 20:06.37 | `Sauron | greg: Hum, I see. |
| 20:06.50 | greg_work | ie, mine will have an OPTIONAL busy greeting (if it's not there, it'll say the unavailable, not the canned busy greeting), temporary greeting, all recordings will be temporary at first (right now, whle you're recording a new greeting, if someone goes to your vm they hear it, even if you haven't "saved" it yet) .. same with messages |
| 20:07.18 | greg_work | also some basic things.. better handling of press 0 for operator (ie, press it at anytime). press * during your vm message to login |
| 20:07.40 | multrix | tzanger: so it's impossible to have a transition solution with a stealth Asterisk |
| 20:08.05 | greg_work | i think i'll use db to store most options, and voicemail.conf will just set the defaults. |
| 20:08.06 | tzanger | multrix: with stealth, yes-ish... like I said it depends on the KSU/PBX |
| 20:08.12 | multrix | What do you think is the best solution for putting some ip phones on a network and migrate progressively the others on asterisk |
| 20:08.18 | greg_work | except for password.. voicemail.conf can store the pwd.. |
| 20:08.22 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:08.29 | kant | Has anyone here configured a Clipcomm CG-410? |
| 20:08.31 | tzanger | multrix: TDM440P with analog "extension" ports on the KSU/PBX |
| 20:08.36 | tzanger | i.e. Norstar ATAs |
| 20:08.54 | `Sauron | Y |
| 20:09.09 | *** join/#asterisk falcoIT (~uncle@host6-67.pool8248.interbusiness.it) |
| 20:09.24 | vaewyn | greg_work: I am working on voicemail improvements as well... but mine stem towards removing the dialplan part of it from the C and having a dialplan based voicemail |
| 20:09.51 | falcoIT | Hi.. i have some "dummy" questions to start and build up my first ASterisk based PBX... |
| 20:09.52 | greg_work | vaewyn: oh yeah? how far along are you? and how are you doing actual recordings etc? |
| 20:10.19 | falcoIT | can someone help me? |
| 20:10.25 | firestrm | `Sauron, ive never used postfix , im worried im about to jump into a 3 day adventure to hell (thats what converting from sendmail to exim was) |
| 20:10.44 | `Sauron | firestrm: It'll be at most a 6 hour adventure |
| 20:10.48 | `Sauron | And it's nowhere near hell. |
| 20:10.58 | firestrm | i hope so.. |
| 20:10.59 | Nugget | postfix isn't too hard to get going. |
| 20:11.03 | Nugget | the docs are great |
| 20:11.24 | *** join/#asterisk vaewyn (freeman@mail.deltamach.com) |
| 20:11.30 | falcoIT | yeah... postfix is not sush a hell :) |
| 20:11.32 | vaewyn | arggh... bad internet.... bad! |
| 20:11.39 | firestrm | Nugget, i agree, its just that i have to learn a new config, and then customize .. never easy |
| 20:11.46 | greg_work | falcoIT: go check voip-info.org, google, and just ask otherwise .. theres LOTS of stuff on voip-info |
| 20:11.48 | Nugget | sure, sometimes that's easy. |
| 20:11.52 | greg_work | vaewyn: oh yeah? how far along are you? and how are you doing actual recordings etc? |
| 20:11.54 | Nugget | postfix is just average difficulty. |
| 20:11.57 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 20:12.01 | ariel_ | I have a question about CDR's. When asterisk dial an extension via a macro for follow me or to try more then one device it does not make any cdr entry's is there any fix for this? |
| 20:12.29 | Pinhole | Is there an equivalent in agi to the cli "sip show peer xxx" ? |
| 20:12.33 | firestrm | Nugget, what is it more like in config, sendmail or exim? |
| 20:12.36 | falcoIT | who can tell me the difference between a FXO and a FXS? |
| 20:12.51 | Nugget | I'm not familiar with exim, so I dunno, but it's not much like sendmail at all. |
| 20:12.51 | blitzrage | ~fxo |
| 20:12.52 | jbot | it has been said that fxo is foreign exchange office - or the type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx |
| 20:12.52 | bjohnson | tzanger: I can't get my fxo on the Nortel ATA to detect dtmf tones |
| 20:12.52 | multrix | falcoIT: www.voip-info.org can tell you, google.com also |
| 20:12.55 | blitzrage | ~fxs |
| 20:12.56 | jbot | hmm... fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 20:12.56 | Nugget | I came to postfix from qmail. |
| 20:13.11 | bjohnson | tzanger: so that might be limited in functionality |
| 20:13.12 | firestrm | ok here goes.. |
| 20:13.20 | vaewyn | greg_work: about 75ish% ... I just use the built in Record() and such... My philosophy is that 90% of the voicemail functions we already have in extension capable functions... we should be using them instead of doing dual development |
| 20:13.55 | blitzrage | vaewyn: plus not having to update code in 2 places when you change one of them |
| 20:14.14 | bjohnson | vaewyn: greg_work: either of you doing ldap saving of vm passwords? |
| 20:14.21 | greg_work | vaewyn: hm, yeah.. how do you do processing of inbox? thats about the only complicated bit |
| 20:14.41 | greg_work | bjohnson: no, i'm actually just getting to that part (was just asking about how I should do it) |
| 20:15.03 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 20:15.23 | greg_work | i think anyway I do it now, i'll write it as a set of functions in an included file .. so if, for example you want to use ldap, you just include vm_ldap.inc instead of vm_voicemailconf.inc |
| 20:15.49 | vaewyn | greg_work: I have made a couple functions that are exposed so extensions can move stuff between mailboxes... the indexing inside the folders is done with variables in the extension logic |
| 20:15.57 | tzanger | bjohnson: hmm |
| 20:15.57 | buddah | file: so its not behind nat now, the proxy stuff is setup, sip/extensions.conf are setup, and registration is still failing, any ideas? |
| 20:16.05 | bjohnson | I don't know best way to do it .. but needs a ldapget AND a ldapput |
| 20:16.22 | falcoIT | so, in a standard PC, i need AT LEAST a FXO card to connect to the telco plug and a FXS card for each "old style" phone, right? |
| 20:17.18 | greg_work | vaewyn: what do you mean by 'functions' .. macros? |
| 20:17.30 | bjohnson | I think it is the nortel hardware. a user on another extension cannot hear my button presses when I push buttons during an internal call between handsets. (Nortel M7208 and M7310 on a Nortel CICS) |
| 20:17.51 | greg_work | falcoIT: sortof. you can get cards that have multiple ports (ie, tdm400p has 4 ports, each can be fxs or fxo) |
| 20:18.02 | buddah | anyone else familiar with linksys pap2-na's? |
| 20:18.12 | greg_work | falcoIT: you can also get sip ATA's, like the SPA-2000 that has two fxs ports and plugs into ethernet |
| 20:18.43 | greg_work | falcoIT: you can also get T1 cards (and 4-span t1 cards) that you can hook into channel banks that provider fxs/fxo ports, each T1 gives you 24 ports |
| 20:19.16 | bjohnson | falcoIT: cheapest solution is a softphone + voip provider for incoming and outgoing |
| 20:19.26 | vaewyn | greg_work: sorry... no... applications... like 'hasvoicemail' and such... I am writing more powerful ones that are exposed so the dialplan can do this crud directly |
| 20:19.34 | greg_work | ahh, ok. |
| 20:19.55 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 20:20.01 | wolfson | anyone know the legality of pre-recorded solicitation to a business. I know its illegal to a residence. Some company keeps calling us with recorded messages with a CID of all 0's. |
| 20:20.03 | greg_work | thats not a bad idea actually. can i see what you have so far? |
| 20:20.28 | bjohnson | wolfson: depends on country, state, etc |
| 20:20.32 | vaewyn | greg_work: I will put you on the list to notify when I have it ready for alpha |
| 20:20.44 | wolfson | under US federal law |
| 20:20.54 | firestrm | When Postfix sees an address with only one component in the hostname, should it append .$mydomain? .. i have answer y/n |
| 20:20.55 | greg_work | i'm doing it phpagi, and i'm probably 50%. if you have something that will work though, then maybe i shuoldn't bother developing this |
| 20:20.56 | bjohnson | wolfson: zapateller or IVR is most practical defence |
| 20:20.57 | falcoIT | greg_work: thanks, so a sip ATA can convert a standard phone, fax or whatever to a VOIP device, but to have asterisk dialogue with my standard telco line i need a FXO card. right? |
| 20:21.04 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 20:21.10 | firestrm | im think y, but not sure |
| 20:21.19 | greg_work | falcoIT: you can also get SIP fxo adapters |
| 20:21.22 | bjohnson | wolfson: I think you can request to be put on a no-call list .. good luck finding anyone that follows it though |
| 20:21.28 | BoRiS | Yippy, The T.38 bounty is now $3000USD. |
| 20:21.31 | wolfson | bjohnson: I know that, I can filter them, but if its illegal, they will be in the area doing a seminar, and would like look into legal action |
| 20:21.37 | fearnor | zippy, boris |
| 20:21.39 | greg_work | i think hte spa-3000 provides one fxo and one fxs |
| 20:21.40 | fearnor | now go fucking code it. |
| 20:21.41 | fearnor | :) |
| 20:21.42 | wolfson | bjohnson: do not call does not apply to businesses |
| 20:21.44 | bjohnson | falcoIT: a SPA 3000 has one fxo and one fxs |
| 20:21.53 | BoRiS | you first fearnor |
| 20:22.01 | fearnor | i just put 1k$ on it |
| 20:22.05 | fearnor | heh |
| 20:22.18 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-8-169.w82-122.abo.wanadoo.fr) |
| 20:22.20 | falcoIT | <bjohnson>: i know that softphone will be the cheapest solution.. but i would like to set up a dedicated PC with its own asterisk to start learning things, not to make phone calls ;) |
| 20:22.30 | *** join/#asterisk Rick_Hunter (~rhunter@06-097.008.popsite.net) |
| 20:22.47 | bjohnson | falcoIT: then you don't need any fxo or fxs hardware |
| 20:22.51 | BoRiS | oh...lol |
| 20:22.56 | multrix | falcoIT: I'm at the same way as you, trying asterisk on a PC :) |
| 20:23.04 | fearnor | coppice should finally notice it ;) |
| 20:23.05 | greg_work | vaewyn: any chance I can just see it now? i'm just curious how it looks etc. maybe be able to help you out too |
| 20:23.31 | redder86 | $3000 for T.38. Hehe. Not even close to what it's worth. |
| 20:23.35 | greg_work | i'm aiming to have mine beta-status by the end of the week, btw |
| 20:23.53 | multrix | do somebody have a sample web interface to a asterisk box ? :) |
| 20:24.27 | *** join/#asterisk scubasteve (~steve@office37.neonova.net) |
| 20:24.35 | fearnor | red: its actually probably simple enough to do. |
| 20:24.41 | fearnor | all the *hard* code is already has been done |
| 20:24.44 | fearnor | by coppice |
| 20:24.49 | redder86 | Steve already is progressing towards T.38 support anyway, and I don't think that he's motivated by bounties. |
| 20:24.51 | fearnor | whats left is asstricks interface. |
| 20:24.59 | fearnor | red: indeed, unfortunately, he isn't ;) |
| 20:25.19 | redder86 | Yes, I'd venture to bet that 90% of the work is done. |
| 20:25.25 | greg_work | multrix: amportal is .. ok. but it doesnt configure everything, and it does restrict you a bit in what you can do (well, unless you can program) |
| 20:25.31 | fearnor | mpd/ |
| 20:25.32 | fearnor | nod. |
| 20:26.00 | BoRiS | redder86: And where can we see steve's latest T38 implimentation? |
| 20:26.02 | falcoIT | bjohnson: in your opinion, to get things together for the first time, in addition to a PC, Asterisk and a bag full of time and enthusiasm, would you suggest me a SPA3000 or an internal card to the PC providing FXO and FXS? |
| 20:26.11 | wasim | \ |
| 20:26.15 | redder86 | BoRiS: http://www.opencall.org |
| 20:26.30 | fearnor | yeah. |
| 20:26.43 | multrix | falcoIT: I would buy an Ip phone.... I think it's better ! |
| 20:26.53 | greg_work | falcoIT: what are you trying to do? |
| 20:27.01 | `Sauron | Someone needs to make cheap(er) fxo hardware |
| 20:27.06 | redder86 | coppice mentioned here last week that the Class 1 modem emulation was done. |
| 20:27.12 | bjohnson | falcoIT: what country? |
| 20:27.17 | redder86 | it's supposedly part of unicall |
| 20:27.25 | redder86 | no docs, though. |
| 20:27.44 | redder86 | He did give a brief instruction on how to do it here in IRC, though. |
| 20:27.53 | redder86 | I haven't had time to try it yet, though. |
| 20:27.54 | greg_work | falcoIT: if you're planning on having multiple fxo ports, there not much point in a SIP adapter |
| 20:28.10 | bjohnson | `Sauron: X100P clone for $20 .. SPA 3000 for $100 .. channel bank for approx $50 per port |
| 20:28.11 | fearnor | red: i was thinking of implementing "ghetto.38" |
| 20:28.25 | redder86 | what's that? |
| 20:28.26 | fearnor | instead of doing proper t.38, just translate datastream into AT-class1-commands ;) |
| 20:28.42 | `Sauron | bjohnson: the spa gear isn't FXO |
| 20:28.45 | redder86 | oh, yeah, I think you and I discussed that before |
| 20:28.48 | fearnor | yeah |
| 20:28.49 | fearnor | nod. |
| 20:28.57 | fearnor | but unfortunately i have more money than time |
| 20:29.04 | fearnor | i'd rather throw 1k$ to steve ;) |
| 20:29.05 | bjohnson | `Sauron: yes it is |
| 20:29.12 | bjohnson | `Sauron: I have 3 SPA 3000 |
| 20:29.14 | `Sauron | hum, ah |
| 20:29.29 | bjohnson | `Sauron: one fxo and one fxo |
| 20:29.33 | BoRiS | Hmmm, now how to convince Steve to continue his T.38 implimentation |
| 20:29.46 | redder86 | I wonder if you could just save yourself the grand and wait it out. I don't think $1K more is going to move it along any faster. |
| 20:29.48 | file | BoRiS: what's up? |
| 20:29.50 | bjohnson | falcoIT: what country? |
| 20:29.52 | `Sauron | I see. |
| 20:29.54 | fearnor | red: dunno. |
| 20:29.55 | fearnor | heh |
| 20:30.10 | multrix | bjohnson: where do you find channel bank equivalent to lots of SPA3000 together ?? :) |
| 20:30.12 | fearnor | i'm hoping bounty with a fixed deadline will spur things along ;) |
| 20:30.16 | `Sauron | bjohnson: I hear a lot of people having problems with the x100p clones |
| 20:30.20 | `Sauron | dum di dum |
| 20:30.20 | falcoIT | bjohnson: italy and switzerland |
| 20:30.32 | Nugget | yay switzerland. |
| 20:30.38 | bjohnson | multrix: ebay is likely target .. Adit600 for mixed fxo and fxs according to tzanger |
| 20:30.41 | Beirdo | `Sauron: you have any alternative? :) |
| 20:30.46 | BoRiS | file: not too much, waiting for an email and need to call atacomm |
| 20:30.56 | redder86 | fearnor: I did some recent fax development for a customer, and $3000 is pretty low. |
| 20:30.58 | file | BoRiS: I too am waiting for emails, yet they are not coming |
| 20:31.05 | fearnor | no doubt, redder. |
| 20:31.29 | redder86 | if the bounty were $10K then I'd probably stop and do it myself. |
| 20:31.43 | fearnor | instead, you just bullshit on irc ;) |
| 20:31.46 | bjohnson | falcoIT: look into voip providers .. they can get you on and off the pstn. Don't know those countries but France has like 9 euro per month unlimited |
| 20:31.47 | redder86 | hehe |
| 20:32.04 | bjohnson | falcoIT: why buy fxo or fxs at all if you don't need them |
| 20:32.30 | redder86 | I lurked here for a long time trying to understand Asterisk. I find that I have to lurk around in project discussion groups for a while long enough to understand what I want to know about the project. |
| 20:32.43 | fearnor | yeah, and use it :) |
| 20:32.44 | bjohnson | Beirdo: alternative to X100P clone? I love my SPA 3000s |
| 20:32.46 | redder86 | Asterisk isn't very well documented - just lots and lots of stuff scattered around. |
| 20:32.47 | falcoIT | bJohnson: to make practice into configuing and using them |
| 20:32.52 | redder86 | so I had to lurk here to learn. |
| 20:32.52 | InfraRed | knowledge by osmosis |
| 20:32.55 | fearnor | red: its all in the source |
| 20:33.13 | redder86 | fearnor: I've read some of the source. It's pathetically documented. Nearly no comments whatsoever. |
| 20:33.14 | multrix | redder86: I agree !!! |
| 20:33.29 | fearnor | redder: hey, if you could read it, it wouldn't be called CODE |
| 20:34.17 | Beirdo | bjohnson: Hmm, I hadn't considered that possibility. |
| 20:34.28 | *** join/#asterisk ckruetze (~ckruetze@i3ED61843.versanet.de) |
| 20:34.32 | falcoIT | grek_work:now just making practice.. first production system must be a voip pbx on a side of a standard pbx, mapping some internal numbers to external numbers via several voip providers around the globe |
| 20:34.34 | redder86 | I can read c/c++, but reading c and reading english comments is a world of difference |
| 20:34.42 | bjohnson | falcoIT: what do you plan to use * for? In US you can get 800 number for incoming and pay nufone $0.02/minute for incoming and outgoing .. like 5000 minutes for what you'd pay in hardware |
| 20:34.59 | bjohnson | Beirdo: I think your friend has them too |
| 20:35.03 | Beirdo | Yep |
| 20:35.06 | Beirdo | he does |
| 20:35.39 | bjohnson | I bought from voxilla to get the free broadvoice month .. but I ended up paying it to UPS instead |
| 20:35.51 | Beirdo | heh |
| 20:35.54 | fearnor | heh |
| 20:35.59 | fearnor | cheap people pay twice |
| 20:36.10 | *** join/#asterisk andersee (~andersee@codepoet.org) |
| 20:36.20 | Beirdo | yeah, nothing lilke the $50-75 UPS bend-over-and-take-it fee... oh... "Brokerage" |
| 20:36.28 | bjohnson | fearnor: errmm .. the X100Ps I have are from Digium |
| 20:36.41 | firestrm | `Sauron, that was painless.. im up and running with postfix already.. |
| 20:36.45 | bjohnson | fearnor: so are the S100U peices of sheit |
| 20:37.21 | fearnor | i sense a pattern. :P) |
| 20:38.15 | bjohnson | (with a quick followup with a second S100U) |
| 20:38.29 | redder86 | fearnor: normally when one writes code it is important to put comments into the code that allow people who are reading it understand what "x" and "y" and various functions do. Reading code without documentation like this pretty much requires a cover-to-cover read before you get it. Instead, it would be nice to just look at one code file and read it and understand about that channel or that application or whatever, rather than needing to read eve |
| 20:38.30 | BoRiS | I hate the brokerage fee's |
| 20:38.33 | bjohnson | the USB cable acts as a stabilizer |
| 20:39.17 | redder86 | fearnor: I've tried reading spandsp, too, and it's pretty much the same story. |
| 20:39.23 | firestrm | BoRiS, clear it yourself.. if you have a customs office nearby |
| 20:39.27 | kant | die die die you PoS |
| 20:39.31 | fearnor | redder: /* you are not expected to understand this */ |
| 20:39.42 | andersee | Any thoughts on the "OEM X100P - FXO PCI Card" -- http://www.digitnetworks.com/store/product_info.php?products_id=28 |
| 20:39.43 | fearnor | and in general, understanding spandsp is impossible if you dont have dsp background |
| 20:39.50 | andersee | any good? |
| 20:40.03 | BoRiS | firestrm: And how do you do that if UPS or FedEx automatically does it for you? |
| 20:40.07 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 20:40.21 | redder86 | fearnor: spandsp does T.30, and I understand T.30, I shouldn't need to know DSP in order to look at the T.30 portion of spandsp. |
| 20:40.33 | fearnor | ah true |
| 20:40.37 | Beirdo | BoRiS: how's life in winterpeg? |
| 20:40.41 | ardor | i cut my finger |
| 20:40.59 | fearnor | another thing is, i think that code would benefit from 'coroutines' |
| 20:41.06 | tzanger | coroutines? |
| 20:41.08 | *** join/#asterisk cripito (~ncripito@68.216.32.158) |
| 20:41.09 | fearnor | because of the way datapumps are working |
| 20:41.10 | tzanger | I read that as croutons |
| 20:41.12 | falcoIT | bjohnson: i need a PBX that is able to receive calls from several VOIP providers, forward them following specific rules to numbers internally to an office PBX, or externally to mobile phones. And viceversa |
| 20:41.22 | fearnor | aka 'duffs device' (google it). cute :P |
| 20:41.43 | BoRiS | Beirdo: Life is great....The weathers getting warmer but that usually means we can expect a snow storm sometime soon. :-p |
| 20:42.01 | file[laptop] | andersee: see other channel. |
| 20:42.22 | Beirdo | heh. Yeah, we even got dumped on in Toronto lately. Fun to see the city slickers panic |
| 20:42.50 | |Vulture| | is there a command like zapbarge but you can actually talk? |
| 20:43.45 | BoRiS | Beirdo: lol |
| 20:44.26 | Beirdo | I grew up in the snow belt north of here, so I have no patience for people who think that 1/2" of snow in Toronto constitutes an emergency :) |
| 20:44.43 | tzanger | hahaha |
| 20:44.45 | tzanger | I live in a snow belt |
| 20:44.56 | BoRiS | Berido: You know how it is...WHen it snows, people panic. |
| 20:45.03 | *** part/#asterisk Corvin (~zbysio@chello084010031149.chello.pl) |
| 20:45.08 | *** join/#asterisk LarsAC (~chatzilla@pD95009B2.dip0.t-ipconnect.de) |
| 20:45.20 | BoRiS | They drive at 20km/h |
| 20:45.20 | tzanger | BoRiS: you're from the TO area, no? |
| 20:45.27 | Beirdo | yeah, so true, BoRiS |
| 20:45.48 | BoRiS | tzanger: No, Winnipeg... I was in toronto last summer for a few days (my first time there) |
| 20:46.00 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 20:46.39 | falcoIT | a question for you all: any IP phone running linux onboard? (maybe with more than a LAN port and possibly with an DLS modem)? |
| 20:47.08 | Nugget | why would that matter? who cares what os is inside the phone? |
| 20:47.34 | falcoIT | you can customize it... installing postfix for example :) |
| 20:47.40 | Beirdo | hehe |
| 20:47.48 | Beirdo | you want postfix on yer phone?! |
| 20:47.51 | Beirdo | heh |
| 20:48.04 | Nugget | I'll run linux on my phone just as soon as I'm finished porting linux to my dishwasher. |
| 20:48.06 | Beirdo | geeks never fail to surprise me |
| 20:48.33 | tzanger | ahh |
| 20:48.44 | falcoIT | i was kidding, but i saw out there on the net a phone doing such things. |
| 20:48.46 | tzanger | falcoIT: wasim's working with some people on one I think |
| 20:48.47 | tzanger | I THINK |
| 20:49.25 | falcoIT | the real need is to cipher the call |
| 20:49.32 | Beirdo | oooh, trilingual political wallops |
| 20:49.36 | tzanger | falcoIT: do it at the edge |
| 20:49.53 | tzanger | you're already in an open office -- they can hide a mic in the false cieling |
| 20:50.08 | falcoIT | tzanger: where is the edge, for you? |
| 20:50.17 | tzanger | falcoIT: firewall/router |
| 20:50.37 | falcoIT | tzanger: that's not a ciphered call... but a vpn |
| 20:50.50 | tzanger | falcoIT: depends. :-) |
| 20:51.03 | falcoIT | if we are in a 10 ppl office, 9 ppl would be able to listen. |
| 20:51.05 | tzanger | falcoIT: all your phones would talk back to a central * box (guessing) - you can encrypt from there too |
| 20:51.12 | tzanger | falcoIT: exactly -- what's the point |
| 20:51.24 | tzanger | unless you're in a sealed office it's overkill |
| 20:51.27 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-143-246.dsl.scarlet.be) |
| 20:51.31 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
| 20:51.46 | *** join/#asterisk AnaKali (Serge@126-239.243.81.adsl.skynet.be) |
| 20:52.14 | falcoIT | tzanger: that's what my customer require me. that's privacy. why do people use PGP instead of simple SSL SMTP ? |
| 20:52.20 | sjaak538 | somebody knows something about ISDN hfc |
| 20:52.25 | tzanger | falcoIT: much different. |
| 20:52.45 | tzanger | falcoIT: but yeah if your customers demand that then it's your job as a manager of expectations to either provide it or convince them it's not a great idea |
| 20:52.58 | AnaKali | hello all, Is anyone know a good motherboard in supermicro or asus for TDM400P and asterisk in P4 ? |
| 20:53.22 | falcoIT | tzanger: i do not think it's a stupid idea. |
| 20:53.26 | *** join/#asterisk n4y (~tmalkut@170.orasoft.net.pl) |
| 20:53.54 | tzanger | falcoIT: ok, fair enough. I didn't say it was stupid, just unreasonable unless the physical security is also there |
| 20:54.13 | falcoIT | tzanger: physical security is ok.. but the electronic no |
| 20:54.20 | tzanger | falcoIT: when I was a kid I used a he-ne laser to listen on conversations by bouncing the beam off a window of a nearby house... |
| 20:54.41 | falcoIT | tzanger: did it work? |
| 20:54.44 | tzanger | falcoIT: of course |
| 20:55.05 | falcoIT | tzanger: always thought it was a fake of Hollywood |
| 20:55.08 | tzanger | but that's what I'm saying -- unless the physical security is tight enough ot prevent an eavesdropper from doing something like that, encrypting the *phones* is a little silly |
| 20:55.19 | kant | How does bouncing a beam of a window allow you to hear the conversation!? |
| 20:55.45 | tzanger | falcoIT: nope -- take a he-ne laser and aim it at a windowpane. you need to be pretty much on the same level as the window and a "straight" shot |
| 20:56.02 | *** join/#asterisk darby_t (~tom@dnw41.neoplus.adsl.tpnet.pl) |
| 20:56.03 | tzanger | the window vibrates with teh sound in he room whcih is picked up by a receiver mounted on top of the laser and demodulated |
| 20:56.13 | zoa | send me your setup tzanger |
| 20:56.15 | zoa | sounds like fun |
| 20:56.16 | zoa | :) |
| 20:56.17 | tzanger | admittedly it's not 5.1 digital surround audio but it's very intelligble |
| 20:56.28 | tzanger | double-pane glass attenuates it quite a bit |
| 20:56.53 | kant | But how does the laser detect the vibration? |
| 20:57.01 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 20:57.06 | falcoIT | tzanger: you must be exactly in front of the glass! |
| 20:57.07 | tzanger | I was just a pizza-faced 13 year old with a he-ne laser and some electronics... imagine what someone who really wanted your encrypted conversations would do |
| 20:57.19 | tzanger | falcoIT: across the street, or a few houses away, yes |
| 20:57.32 | tzanger | you need clear, direct shot to window and back otherwise the beam bounces off away from you |
| 20:57.34 | falcoIT | but on the normal to the glass |
| 20:57.35 | tzanger | you need a dead-on shot |
| 20:57.40 | *** part/#asterisk n4y (~tmalkut@170.orasoft.net.pl) |
| 20:57.41 | tzanger | yes normal -- that's teh word I was looking for, thanks |
| 20:57.46 | Nugget | heh |
| 20:57.54 | tzanger | kant: amplitude modulation -- ever make a solar phone? exact same idea |
| 20:58.15 | tzanger | Me ---- | window |
| 20:58.17 | falcoIT | tzanger: solar phone? |
| 20:58.19 | tzanger | if I was above or below it wouldn't work |
| 20:58.22 | tzanger | Me |
| 20:58.24 | tzanger | <PROTECTED> |
| 20:58.26 | tzanger | (shit) |
| 20:58.30 | fearnor | laser doesn't "detect" vibration. glass vibrates by itself. |
| 20:58.45 | outtolunc | http://www.global-security-solutions.com/LaserRoomMonitor.htm has a nice explanation |
| 20:58.47 | fearnor | laser will bounce slightly due to vibration, resulting in less/more signal |
| 20:58.50 | techie | solar phone...nice |
| 20:59.00 | `Sauron | Humm. |
| 20:59.00 | tzanger | trying to find a circuit online now |
| 20:59.04 | kant | But I'm guessing the amplitude of the light bouncing back at you changes as the glass vibrates. |
| 20:59.07 | zoa | tzanger: send me your tools :p |
| 20:59.10 | `Sauron | Anyone know of working linux sip softphones? |
| 20:59.15 | modulus_ | linphone |
| 20:59.37 | tzanger | kant: the amplitude is modulated by the vibration... the received signal's strength varies as teh vibration |
| 20:59.44 | `Sauron | Hum. |
| 20:59.59 | fearnor | ok |
| 21:00.02 | fearnor | i hate dell officially |
| 21:00.07 | kant | tzafrir: that's what I said. |
| 21:00.13 | fearnor | to get anything resolved it takes 10 phone calls |
| 21:00.16 | tzanger | solar phone: basically take a paper tube, cover one side with aluminum foil shiny side out and take or build an AM radio you can hook a photodiode to the AM demodulator |
| 21:00.25 | falcoIT | tzanger: back to the encryption, since the customer wants it, i'll do it. in reality, my boss wants it.. i want the money from him, his customer wants the solution from my boss... i try to make everyone happy. :) |
| 21:00.31 | tzanger | aim the foil so that the sun reflects off of it to the diode... now speak into the tube |
| 21:00.40 | `Sauron | fearnor: You just don't know what to tell them... |
| 21:00.47 | tzanger | falcoIT: yup that's how it generally works |
| 21:00.50 | tzanger | follow the money :-) |
| 21:01.34 | tzanger | falcoIT: you can bring up with them the ability to eavesdrop without the phone (mic in cieling, bugged room, laser listener, etc.) and ask if they really do want to pay for each phone being encrypted instead of all calls out of the building being encrypted |
| 21:01.44 | bjohnson | `Sauron: I use iaxcomm and linephone |
| 21:01.50 | _Vile | the worst problem I've ever had with dell, is that they are dell. |
| 21:01.57 | tzanger | falcoIT: also if you can find some phones that do SRTP or eIAX2 you might be able to pursuade them ot drop it when you present the price |
| 21:01.58 | bjohnson | `Sauron: got kphone to run but couldn't figure out the config |
| 21:02.04 | `Sauron | Hum. |
| 21:03.10 | `Sauron | I need to set up the DNS records for * on my network |
| 21:03.12 | `Sauron | dum di dum |
| 21:03.22 | `Sauron | rather, for sip |
| 21:03.32 | bjohnson | falcoIT: big question is .. what security do they have NOW? |
| 21:03.44 | falcoIT | tzanger: the problem on the router or firewall is that it cannot handle ciphered calls if both parties are not in two different offices... imagine two brokers moving with the car and going into customers offices and need to talk together. |
| 21:04.00 | falcoIT | bjohnson: NOW they use email. Too slow. |
| 21:04.19 | fearnor | dell is great pricing and all |
| 21:04.31 | bjohnson | gotta be a government agency to throw the bucks at internal office call encryption |
| 21:04.32 | fearnor | its just that i spend an hour a week on the phone with dell to fix things they screwed up |
| 21:04.37 | shido6 | brb |
| 21:04.38 | tzanger | falcoIT: yes if Room 101 calls Room 102, it'll be unencrypted... but see my previous messages about the "who cares" aspect |
| 21:04.52 | bjohnson | dell has a good deal on system with 17" LCD now |
| 21:04.54 | falcoIT | bjohnson: webmail on a HTTPS connection. |
| 21:04.58 | tzanger | falcoIT: hell if Room 101 calls a cell phone or other client that odesn't have VOIP it'll become unencrypted at the hopoff |
| 21:05.01 | bjohnson | $800 I think for a P4 |
| 21:05.03 | fearnor | heh tzanger |
| 21:05.07 | _Vile | not bad |
| 21:05.11 | fearnor | "room 101" always recalls 1984 for me |
| 21:05.12 | tzanger | falcoIT: it's possible to solve these problems but again, at what cost |
| 21:05.15 | tzanger | fearnor: :-) |
| 21:05.15 | _Vile | I need a new monitor |
| 21:05.22 | falcoIT | bjohnson + tzanger: they are NOT internal calls :) |
| 21:05.28 | buddah | i need a new workstation for work |
| 21:05.32 | _Vile | hmm |
| 21:05.33 | buddah | whopping 200mmx |
| 21:05.40 | _Vile | you should buy the dell $800 |
| 21:05.42 | tzanger | falcoIT: so Office 1 calling Office 2? What's wrong with eIAX2? |
| 21:05.43 | buddah | no |
| 21:05.44 | _Vile | you'll get a new workstation |
| 21:05.44 | djin | buddah, Mac Mini |
| 21:05.46 | buddah | my boss should buy the dell |
| 21:05.47 | _Vile | i'll get a monitor |
| 21:05.50 | _Vile | yes |
| 21:05.52 | _Vile | convince him |
| 21:05.53 | _Vile | :) |
| 21:05.54 | file[laptop] | Dude, I just bought a Dell! |
| 21:06.03 | file[laptop] | and they haven't even e-mailed me yet |
| 21:06.05 | tzanger | my general opinion is that brokers talk loud enough and brash enough that everyone within 500 meters can hear them clearly anyway |
| 21:06.12 | buddah | yeah we have 6 of the dell 17" lcds in our ups room |
| 21:06.15 | buddah | just sitting there |
| 21:06.22 | buddah | from clients we gave them boxes, just no moniters |
| 21:06.32 | falcoIT | tzanger: where do you live? in which country? |
| 21:06.34 | _Vile | so you could ship a couple and noone would notice? |
| 21:06.35 | _Vile | :) |
| 21:06.38 | tzanger | falcoIT: Canada |
| 21:06.39 | djin | lemme sent you my address. |
| 21:06.41 | buddah | i think my boss has a master plan |
| 21:06.47 | falcoIT | tzanger: never meet a swiss banker? |
| 21:06.48 | _Vile | *shrug* worth a shot |
| 21:06.52 | djin | Flight Simulator? |
| 21:06.59 | tzanger | falcoIT: admittedly not |
| 21:07.02 | bjohnson | buddah: send one to me |
| 21:07.04 | buddah | Feb 1 13:01:14 NOTICE[1315]: rtp.c:317 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
| 21:07.09 | tzanger | hahaha |
| 21:07.10 | buddah | why do i keep getting that message? |
| 21:07.12 | tzanger | 16:10 < falcoIT> tzanger: never meet a swiss banker? |
| 21:07.14 | tzanger | 16:10 < bjohnson> buddah: send one to me |
| 21:07.14 | buddah | have had it for like a week |
| 21:07.19 | fearnor | buddah: turn off rfc3389 on your client. |
| 21:07.22 | fearnor | VAD. |
| 21:07.27 | buddah | i didnt know it was on |
| 21:07.32 | buddah | thought we were running 2833 |
| 21:07.33 | _Vile | http://lists.digium.com/pipermail/asterisk-users/2005-January/080769.html |
| 21:07.37 | _Vile | buddah /|\ |
| 21:07.52 | buddah | k |
| 21:08.06 | buddah | ahh |
| 21:08.07 | buddah | its ss |
| 21:08.08 | _Vile | Turning off silence suppression |
| 21:08.11 | _Vile | ;) |
| 21:08.11 | buddah | someone told me that before |
| 21:08.19 | buddah | my memory is bad |
| 21:08.24 | falcoIT | tzanger: when he visites a customer, and has to call a broker in the US to sell properties or discuss about assets, he does not trust the local phone company, he does not use the cell phone, now he uses a webmail on a https connection. |
| 21:08.35 | pointer-gaim | if you term T1s to a cisco 5x00...and link it with * via SIP....rx_fax should still work, right? |
| 21:08.39 | _Vile | you should replace your memory |
| 21:08.41 | _Vile | :) |
| 21:08.43 | buddah | yeah |
| 21:08.44 | pointer-gaim | rather than a bunch of zaptel interfaces in * |
| 21:08.48 | buddah | find a brain on the black market |
| 21:08.51 | buddah | just hotswap it |
| 21:08.56 | falcoIT | tzanger: but many of our customers cannot use the keyboard... more than a keystroke per second. |
| 21:09.06 | falcoIT | tzanger:so... voip |
| 21:09.06 | tzanger | falcoIT: sounds excellent. an iaxclient-enabled softphone on his laptop that enables encryption back to the central VOIP box |
| 21:09.09 | mikegrb | pointer-gaim: you could terminate it to a digium t1 card |
| 21:09.24 | mikegrb | pointer-gaim: they don't just make fxo fxs cards |
| 21:09.39 | pointer-gaim | mikegrb: I was hoping to reuse existing resources... |
| 21:09.45 | Delmar | does anyone here know where I can find some detailed information that explains what the different echo cancelation algs actually do? there are a bunch that can be chosen in zconfig.h .... ECHO_CAN_STEVE, ECHO_CAN_STEVE2, ECHO_CAN_MARK, ECHO_CAN_MARK2, ECHO_CAN_MARK3 |
| 21:09.53 | pointer-gaim | mikegrb: ie I can still use them for dialup and the like |
| 21:10.00 | falcoIT | tzanger: no laptop. my boss wants a phone. for that reason i asked a phone that includes a linux OS |
| 21:10.15 | tzanger | falcoIT: give him a VAIO and call it a phone. :-p |
| 21:10.22 | falcoIT | tzanger:at 13 you used laser... at 56 they cannot send an SMS! |
| 21:10.25 | mikegrb | pointer-gaim: but yes, sip over lan should be fine for rx_fax |
| 21:10.44 | _Vile | http://216.239.63.104/search?q=cache:rUKajDN59iYJ:lists.digium.com/pipermail/asterisk-users/2002-September/004761.html+ECHO_CAN_STEVE+ECHO_CAN_MARK&hl=en |
| 21:10.57 | _Vile | Delmar /|\ can help a little bit |
| 21:11.24 | Delmar | mint. reading that now. cheers :P |
| 21:11.44 | _Vile | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg30000.html <- helps too |
| 21:12.01 | bjohnson | falcoIT: a HP ipaq running familiar linux, openvpn or ipsec, and linphone? |
| 21:12.20 | bjohnson | you can even get wifi or BT support |
| 21:12.24 | _Vile | I've had success w/ the mark series |
| 21:12.28 | _Vile | but I flipped to PRI |
| 21:12.31 | _Vile | so *shrug* |
| 21:12.37 | _Vile | haven't gotten to test mark3 |
| 21:12.54 | falcoIT | tzanger: Motorola A768 is a choice... but i want a phone with a LAN connector :) |
| 21:13.07 | bjohnson | falcoIT: I think the pa168 units run linux |
| 21:13.22 | Delmar | _Vile i dont see mark3 in my cvs version.... |
| 21:13.41 | *** join/#asterisk adker (~adker@70-97-140-150.dsl1.glv.ny.frontiernet.net) |
| 21:13.51 | Delmar | _Vile still dont see any information which gets into a little bit of detail about exactly what the alg is doing... |
| 21:14.23 | dan2 | anybody have music on hold setup here? |
| 21:14.24 | tzanger | falcoIT: that's a nice phone, and its VPN would work to encrypt your traffic |
| 21:14.24 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 21:14.26 | Delmar | _Vile, im inclined not to waste my time commenting out mark2 and aggressive code, to test the others when by the sounds of it they are not as good... |
| 21:15.09 | Delmar | I dont think I am going to be able to use an X100 solution then, and I'm not buying a TDM400P just yet. |
| 21:15.11 | bjohnson | falcoIT: nope .. maybe no |
| 21:15.23 | Delmar | these cards need to seriously get a bit cheaper. |
| 21:15.31 | shido6 | they are cheap |
| 21:15.34 | shido6 | compared to brooktrout |
| 21:15.41 | jarrod | i just received mine |
| 21:15.43 | jarrod | they work great |
| 21:15.44 | bjohnson | Delmar: start manufacturing some that we can buy for $1 ea |
| 21:15.52 | jarrod | 1400 is not expensive for 4 t1 channels |
| 21:15.58 | multrix | do you know VoIP phone with gigabit ethernet ??? |
| 21:16.00 | shido6 | then how will Asterisk get supported when u buy from the cheap guy? |
| 21:16.07 | jarrod | why do you need gigabit on a phone |
| 21:16.19 | bjohnson | I think most are 10Mb |
| 21:16.29 | Delmar | yeah.. i know.. if I take a look around.. I know that they are very well priced if not under priced as they are. |
| 21:16.48 | jarrod | if you can stream over 512k from your phone i would be impressed |
| 21:16.57 | Delmar | but I guess everyone wants to get things as inexpensive as they can :P |
| 21:17.30 | Delmar | if u are doing more than 512k something is wrong :P |
| 21:17.39 | jarrod | exactly |
| 21:17.47 | `Sauron | Hum. |
| 21:17.53 | `Sauron | T.38 must be popular |
| 21:17.57 | `Sauron | well, sort of |
| 21:17.59 | Delmar | nice big fat codec that sounds great like.. ulaw, and I measure that at about 150kbit/sec full-d. |
| 21:18.08 | `Sauron | people willing to pay $8k+ for getting it into * |
| 21:18.12 | `Sauron | but nobody's written it yet |
| 21:18.29 | Delmar | i dont know anything about t.38 whats that all about dude? |
| 21:18.36 | `Sauron | fax over IP |
| 21:18.38 | falcoIT | tzanger: and do u suggest such motorola with a softphone running on it? |
| 21:18.49 | Delmar | how does it compare to ilbc and g729 etc? |
| 21:18.53 | jarrod | what do you guys currently use for receiving faxes? |
| 21:19.07 | `Sauron | jerrod: I hacked together a server in linux |
| 21:19.09 | Nugget | a fax machine. |
| 21:19.19 | jarrod | fax machine with an ata device? |
| 21:19.19 | Delmar | oh speaking of g729... does anyone here use g729 on their * ? Licenced or.. "other" ? :) |
| 21:19.21 | tzanger | falcoIT: that's about all you can do there, no? |
| 21:19.27 | mutilator | anyone here used a nortel m7324 with a CAP addon module? |
| 21:19.29 | jarrod | i use 'licensed' |
| 21:19.31 | `Sauron | But we're talking bout switching to * with the SpanDSP stuff |
| 21:19.33 | mutilator | phone.. |
| 21:19.45 | `Sauron | Delmar: It's just $10 |
| 21:19.52 | `Sauron | Don't be cheap |
| 21:19.54 | Delmar | yep i know that much. |
| 21:19.59 | Delmar | and I would rather buy it... |
| 21:20.09 | Delmar | i have the haxed one working tho... just for testing.. |
| 21:20.16 | Delmar | and if that is g729.. i wont buy it. |
| 21:20.21 | Delmar | its the same size as ilbc. |
| 21:20.29 | Delmar | so what i was really wanting to ask is.... |
| 21:20.59 | Delmar | does anyone know if there is a huge difference in codec performance with g729 non-licenced ... vs the proper licenced one. |
| 21:21.38 | Delmar | and in non-licenced i mean.. the suposedly fully working g729 codec that is a bit of a breach of copyright that is.. floating around. |
| 21:21.50 | jarrod | pay a simple $10 per license |
| 21:22.19 | Delmar | sure.. i would.. but ... if the g729 i use is the same as the legit one... i wont.. ill use ilbc. |
| 21:22.59 | Delmar | i have dumped g729 now anyway... because it was chewing up the same amount of data as ilbc. |
| 21:23.10 | Delmar | so why pay $10/channel when ilbc is the same? |
| 21:23.14 | falcoIT | bjohnson + tzanger: thanks anyway for your help :) in the future, if you have some time to get some money configuring *... just drop me a line |
| 21:24.17 | fa | ale one hear sth about skype - > * |
| 21:24.50 | Delmar | skype are cheeky. they bounce/relay data off your client. |
| 21:25.06 | *** join/#asterisk WifiFred (~wififred@apollo.bcwireless.net) |
| 21:25.28 | Delmar | and I dare to ask.. whos codec they have stolen or modified for their own use :P... G729. |
| 21:25.33 | Delmar | :P |
| 21:25.34 | zoa | Delmar: i tried |
| 21:25.44 | zoa | i dont think its a big difference |
| 21:25.51 | Delmar | zoa tried what sorry? |
| 21:25.57 | zoa | the codec stuff |
| 21:26.00 | Delmar | g729 legit vs non-legit? |
| 21:26.01 | zoa | i compared em once |
| 21:26.16 | zoa | intel vs digium |
| 21:26.23 | Delmar | oh yep. |
| 21:26.24 | zoa | intel claims its very optimized but its not |
| 21:26.37 | Delmar | but its not a big diff? |
| 21:26.49 | Delmar | can u remember usage figures? |
| 21:27.06 | *** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 21:27.24 | Delmar | i mean.. the legit g729 codec claims 8-12kb + overhead. so should be 16kb/sec easy. well its no where near that. |
| 21:27.58 | Delmar | its more like 32-40kbit per leg. |
| 21:29.01 | zno | the new polycom sip speakerphone is $1100 |
| 21:29.14 | zno | the h323 one is $600 |
| 21:29.16 | zno | wow |
| 21:29.30 | BurnedOutGeek | Anyone here been successful in actually getting fax to passthru a Sipura device with 711u? |
| 21:29.47 | modulus_ | fax via ulaw? |
| 21:29.48 | modulus_ | wow |
| 21:29.51 | Delmar | ok i just retested it .. g729 to * while running iptraf on the console.. 54kbit/sec total. thats double what it should be. |
| 21:30.07 | redder86 | BurnedOutGeek: yes, disable all the fax features |
| 21:30.20 | BurnedOutGeek | redder86: really? |
| 21:30.25 | redder86 | really |
| 21:30.46 | BurnedOutGeek | and thats through Asterisk? |
| 21:30.52 | *** part/#asterisk darby_t (~tom@dnw41.neoplus.adsl.tpnet.pl) |
| 21:31.02 | fafnir | no its through your mom |
| 21:31.08 | Delmar | * can receive faxes and drop them to an image file.. you can then process them to email or whatever..... greg_work told me about it the other day... aparantly its rally simple. |
| 21:31.18 | redder86 | PSTN -> X100P -> Asterisk -> SIP -> SPA-2000 -> fax modem |
| 21:31.26 | Delmar | i had the lines he gave me to do it in notepad but.. wife locked up my box. lol. |
| 21:31.34 | redder86 | Delmar: that's spandsp |
| 21:31.47 | Delmar | ? |
| 21:32.09 | redder86 | Delmar: you're talking about txfax/rxfax from spandsp that you can build into Asterisk |
| 21:32.20 | BurnedOutGeek | yeah.. Ive seen spanDSP, but I need support for standard fax machines |
| 21:32.44 | Delmar | BurnedOutGeek for reception or sending? |
| 21:32.48 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 21:33.01 | BurnedOutGeek | delmar: both if possible |
| 21:33.11 | JonR800 | BurnedOutGeek: he meant turn off all the fax features on the sipura. |
| 21:33.11 | BurnedOutGeek | even if the speed needs to come down to <9600 |
| 21:33.19 | JonR800 | they seem to break faxing.. surprisingly. |
| 21:33.23 | _-Jon-_ | Hey is anyone else having problems with BroadVoice? I can't seem to make any calls through them and I'm getting registration timed out messages in CLI |
| 21:33.31 | JonR800 | you should be able to get 14.4kps reliably. |
| 21:33.32 | BurnedOutGeek | yeah. Thats really odd. I will give that a try |
| 21:33.53 | Delmar | BurnedOutGeek u havent messed with the echo cancelation stuff in zconfig.h ? |
| 21:34.04 | redder86 | yeah, the SPA-2000 has about a half-dozen "fax" features and a number of "echo" features and other things. On the port where the fax device is plugged-in, disable all of those. |
| 21:34.30 | Delmar | if u turn on aggressive for mark2 u need to turn on the 2100hz thingie that turns off the aggressive echo cancelation because aggressive and faxes dont work. |
| 21:34.36 | BurnedOutGeek | delmar: no. At this point, I am gion straight IP... no zap channels |
| 21:34.41 | MuchToDo | Hi all |
| 21:34.55 | Delmar | ah right'o |
| 21:34.57 | BurnedOutGeek | redder86: I will definately try that today |
| 21:34.59 | redder86 | BurnedOutGeek: where are the fax endpoints? |
| 21:35.03 | buddah | what dtmf mode can i use with g729? the 'auto' setting on this pap2-na isnt working, and you cant use inband, there are info and avt left |
| 21:35.07 | buddah | any suggestion as to which? |
| 21:35.25 | MuchToDo | I'm just getting started with asterisk, and have configured iaxtel, but every time I make a call I get "All circuits are busy now." |
| 21:35.31 | BurnedOutGeek | PSTN -> Internet -> Asterisk -> Sipura 2000 |
| 21:35.43 | MuchToDo | am I just being unlucky? Unfortunately I don't really know any numbers to call.. |
| 21:35.48 | MuchToDo | apare from the echo test |
| 21:35.48 | redder86 | BurnedOutGeek: PSTN -> Internet ? |
| 21:36.26 | Delmar | So, I ask this question every day.... so bear in mind that there is much I have already tried but.. does anyone here have any fresh ideas for me to try to get rid of this damn self-echo I'm getting during calls via my X101P card? I'm pretty much out of ideas and getting pretty bummed about it. |
| 21:36.30 | BurnedOutGeek | redder86: meaning that its coming into the asterisk box via IP and not Zap |
| 21:36.47 | redder86 | BurnedOutGeek: you're not likely to have much success |
| 21:37.05 | redder86 | BurnedOutGeek: or rather, you're likely to have an annoyingly large number of problems. |
| 21:37.11 | BurnedOutGeek | oh? Even though its all 711? |
| 21:37.27 | redder86 | BurnedOutGeek: not because of G.711, but because of the internet |
| 21:37.45 | BurnedOutGeek | hmmm. unfortunately, you might just be right :) |
| 21:37.49 | redder86 | BurnedOutGeek: fax is much like data calls, and there is a lot of stuff crammed into milliseconds of audio |
| 21:37.55 | BurnedOutGeek | right |
| 21:38.08 | redder86 | BurnedOutGeek: any little glitches in the audio stream (and there will be some over the internet) will affect faxing |
| 21:38.09 | fa | anyone make a callback for callular phones? |
| 21:38.10 | fa | with AGI ? |
| 21:38.39 | `Sauron | Are people having good luck with the digit networks x100p cards? |
| 21:38.44 | Delmar | NO |
| 21:38.46 | Delmar | :P |
| 21:39.05 | `Sauron | Aww, poor widdle Delmar |
| 21:39.49 | Delmar | well.. not too bad.. only 2 problems I am having is.. some echo I cant seem to remove.. and the busy/hangup tone detection works but not 100% of the time. more like 0%. |
| 21:39.53 | Delmar | 70% even. |
| 21:40.04 | MuchToDo | Anybody wanna be my guinea pig for my first VOIP call..? :) |
| 21:40.23 | `Sauron | MuchToDo: You could call me, but I won't answer for 40 minutes |
| 21:40.29 | buddah | heh |
| 21:40.33 | `Sauron | :) |
| 21:40.36 | _-Jon-_ | Is anyone else having problems with BroadVoice? |
| 21:40.40 | Delmar | you could call me and I will get my cat to answer :P |
| 21:40.43 | MuchToDo | `Sauron: would it go to voicemail or anything? |
| 21:40.46 | `Sauron | I spoke with Greg last night, though |
| 21:40.51 | `Sauron | MuchToDo: Yup |
| 21:40.54 | MuchToDo | anything but this damn "All circuits are busy" message.. |
| 21:41.07 | `Sauron | Jon: It worked fine for me last night |
| 21:41.11 | randu | lol |
| 21:41.21 | `Sauron | I can have Muchtodo call me, and tell me if it works now. :) |
| 21:41.28 | _-Jon-_ | Hmm I don't know about last night but mine sure isn't working now |
| 21:41.32 | MuchToDo | Sounds good! |
| 21:41.33 | _-Jon-_ | Let me know |
| 21:41.35 | `Sauron | muchtodo: you got FWD, or just regular number? |
| 21:41.37 | MuchToDo | although, I've only set up iaxtel.. |
| 21:41.49 | *** join/#asterisk IcePick__ (~me@12.10.168.165) |
| 21:41.50 | Delmar | i thought iaxtel were poked? |
| 21:41.56 | MuchToDo | poked? |
| 21:41.57 | IcePick__ | hi all |
| 21:42.03 | randu | my BV is working fine now |
| 21:42.03 | Delmar | stuffed, buggered, hosed, history. |
| 21:42.03 | MuchToDo | I could set up FWD now, I suppose? |
| 21:42.09 | MuchToDo | Oh I see |
| 21:42.18 | MuchToDo | would that explain "All circuits are busy"? :) |
| 21:42.28 | _-Jon-_ | randu, it wasn't earlier? |
| 21:42.54 | IcePick__ | How do I set up to transfer to a diffrent context? OR how do I set the TRANSFER_CONTEXT |
| 21:42.58 | randu | _-Jon-_ yea It was then too I just wanted to let ya's know that BV was working on my end. |
| 21:43.21 | _-Jon-_ | Oh okay. It's not working through * or if I set my ATA to use it directly.. wtf |
| 21:44.43 | randu | if It was working before and not now trying rebooting asterisk server |
| 21:44.50 | randu | sometimes I find that helps |
| 21:45.09 | `Sauron | BV must be working |
| 21:45.15 | `Sauron | I can call my voicemail |
| 21:45.18 | Delmar | if the ATA isnt working bypassing * then its not the * box. |
| 21:45.40 | Delmar | `Sauron but can u place a call? their supply routes might be shagged. |
| 21:46.15 | _-Jon-_ | hmm, maybe it's the particular server I use? I think I"m on 147.135.0.128 |
| 21:46.22 | `Sauron | humm |
| 21:46.33 | `Sauron | Jon, btw - the IP's on the voip-info pages are not all correct |
| 21:46.48 | `Sauron | I need to finish editing the "how to connect sip to broadvoice page |
| 21:46.54 | _-Jon-_ | Sauron, I got mine from the email they sent out with the patch information |
| 21:47.01 | `Sauron | Jon: Ah. |
| 21:47.01 | randu | yea it is not recommended that you use the ipaddresses |
| 21:47.30 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 21:47.37 | `Sauron | here we go |
| 21:47.43 | `Sauron | gunna fix the voip-info page |
| 21:48.44 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 21:48.48 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 21:49.01 | dan2 | anybody have musiconhold setup with custom? |
| 21:49.29 | randu | IcePick__: exten => 705,2,Dial(SIP/${EXT705}@broadvoice,15,m) The @ broadvoice is the different context I think.... |
| 21:49.35 | IcePick__ | anyone know how to set ${TRANSFER_CONTEXT} |
| 21:49.58 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 21:50.32 | *** join/#asterisk florz (nobody@odnb-d9baa5a9.pool.mediaWays.net) |
| 21:50.45 | florz | LarsAC: pong |
| 21:51.01 | LarsAC | re florz |
| 21:51.06 | `Sauron | Dum di dum. |
| 21:51.08 | LarsAC | meine mail gelesen ? |
| 21:51.10 | ManxPower | Cool! Lots of cheap bastards hitting my former-web site URL today. |
| 21:51.18 | randu | lol |
| 21:51.20 | IcePick__ | randu - that is for a dial - I need to let my queue people transfer calls out of thier context. |
| 21:51.26 | `Sauron | What url? |
| 21:51.31 | florz | LarsAC: jepp, mom noch eben ... ;-) |
| 21:52.10 | falcoIT | tzanger:are you there? |
| 21:52.12 | randu | oh |
| 21:52.25 | randu | ok sorry I am to much of a newbie to answer that :-) |
| 21:53.05 | *** part/#asterisk El_Presidente (Martin@p508C9D02.dip0.t-ipconnect.de) |
| 21:53.34 | IcePick__ | When My que people hit # to transfer - the transfer stays in in their que context - I read about the ${TRANSFER_CONTEXT}, but no examples on how to set it, or use it. |
| 21:54.00 | `Sauron | woot |
| 21:54.06 | `Sauron | Dear SPA Line 1: |
| 21:54.12 | `Sauron | Just wanted to let you know you were just left a 0:09 long message |
| 21:54.22 | `Sauron | ... |
| 21:54.23 | `Sauron | :) |
| 21:54.55 | *** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net) |
| 21:55.35 | dan2 | is there anyway to test the status of musiconhold online |
| 21:55.41 | dan2 | erm or from asterisk console |
| 21:55.48 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 21:56.43 | tzanger | falcoIT: yes |
| 21:56.47 | jjg | anyone know why nufone account creation is broken? |
| 21:56.48 | ManxPower | dan2, "show applications" will give you a hint. |
| 21:56.58 | `Sauron | dan2: Y'all having problems today? |
| 21:57.11 | `Sauron | lots of people getting all trunks are busy |
| 21:57.20 | `Sauron | s/trunks/circuits |
| 21:57.41 | dan2 | ManxPower: how does that help at all? |
| 21:58.48 | mutilator | cya lata all |
| 21:59.04 | ManxPower | dan2, You might see the application that plays the music on hold. |
| 21:59.08 | IcePick__ | ${TRANSFER_CONTEXT} - Anyone? Anyone? |
| 21:59.20 | jjg | can anyone recommend a better provider than nufone? |
| 21:59.24 | *** join/#asterisk shodan (~shodan@216.113.99.175) |
| 21:59.36 | dan2 | ManxPower: I already know that, I'd like to see how many people who are on hold |
| 21:59.49 | mrempire667 | hello guys I want to install the Bri stuff, the bristuff install scriipt doesnt work |
| 21:59.53 | ManxPower | dan2, Oh! No way to do that. |
| 22:00.06 | falcoIT | tzanger: i'm on the private session.. :) |
| 22:00.09 | ManxPower | not unless you use the manager interface or something. |
| 22:00.26 | ManxPower | IcePick__, README.variables anyone |
| 22:00.40 | fa | ManxPower do you know how to make callback on zap channel, for callular phone |
| 22:00.44 | mrempire667 | the script looks on the ftp server for version 1.03 but the server has version 1.0.5 |
| 22:01.04 | shodan | what is the prefered (royalty free) codec to compress voice with asterisk ? |
| 22:01.20 | ManxPower | shodan, That depends on what you need? |
| 22:01.42 | ManxPower | LAN: ulaw or alaw. WAN: Speex, iLBC, or GSM |
| 22:01.44 | IcePick__ | ManxPower - read that - set that - looks like this exten => 129,1,SetVar(TRANSFER_CONTEXT=transferree) |
| 22:01.44 | `Sauron | Manx: ulaw, right? ;) |
| 22:02.06 | shodan | ManxPower, something around 64kbps |
| 22:02.15 | IcePick__ | ManxPower - also tried the static context - it wont transfer out of the current context. |
| 22:02.18 | MuchToDo | hmm... I seem to get "All circuits are busy now" no matter what I try to dial |
| 22:02.23 | MuchToDo | :( |
| 22:02.34 | shodan | these 3 (Speex, iLBC, or GSM) are royalty free ? |
| 22:02.40 | dan2 | ManxPower: something is wrong, I'm not getting audio from asterisk and I've like tripple checked it |
| 22:02.44 | `Sauron | Hehn. I remember when IBM was bragging they'd gotten voice/video streams down to 64kbit each direction |
| 22:03.07 | `Sauron | muchtodo: try calling again? |
| 22:03.14 | `Sauron | I connected my debug console |
| 22:03.27 | `Sauron | and, are you using fwd sip or iax2? |
| 22:03.52 | MuchToDo | fwd |
| 22:03.57 | `Sauron | sip or iax? |
| 22:04.07 | *** join/#asterisk delphiuk (~delphi@host217-44-235-22.range217-44.btcentralplus.com) |
| 22:04.45 | delphiuk | hi, I am trying to build cvs asterisk and am getting the following error: configure: error: termcap support not found |
| 22:04.54 | delphiuk | any ideas? I am using debian sarge |
| 22:05.05 | ManxPower | install termcap |
| 22:05.22 | MuchToDo | I enabled iax at first, then diabled it until I can at least get sip working |
| 22:05.35 | `Sauron | Use iax |
| 22:05.41 | `Sauron | for fwd, anyway |
| 22:05.45 | Manipura | iax rules |
| 22:05.51 | MuchToDo | ok, i'll enable it again and do the configs |
| 22:05.52 | buddah | so for linksys pap2's i can configure them same way i'd do a spa2000? |
| 22:06.06 | `Sauron | If you need, I can send example config |
| 22:06.14 | dan2 | ManxPower: any idea on why I'm not getting sound via musiconhold |
| 22:06.14 | shodan | because I just figured something , why use pci cards for pots input/output , ethernet is a lot more flexible |
| 22:06.17 | `Sauron | I hate FWD's website. Silly flash kills me |
| 22:06.36 | Mike | ManxPower, could you make the sound of your spa841 louder? |
| 22:06.39 | ManxPower | dan2, assuming you are using mpg123 0.59r, no. |
| 22:06.46 | vaewynAFK | delphiuk: apt-get install libncurses5-dev |
| 22:06.52 | Mike | ManxPower, is it a good choise for a office this phone |
| 22:06.53 | Mike | ? |
| 22:06.53 | shodan | like I could use a DS80C400 , a microcontroller with a ethernet port and an ADC and DAC and use that to input/output phone calls |
| 22:06.56 | ManxPower | Mike, There's a volume button |
| 22:07.04 | dan2 | ManxPower: I'm not, I'm using a custom player that is spitting out 8khz, mono streams on stdout |
| 22:07.12 | Mike | ManxPower, you said it was not enough that it was to low |
| 22:07.18 | ManxPower | dan2, Then don't ask me. |
| 22:07.24 | *** join/#asterisk Borgon (~Borgon@default-ip-teleglobe.shellfusion.net) |
| 22:07.30 | dan2 | anybody using custom musiconhold handlers |
| 22:07.34 | Mike | ManxPower, my question is if this phone is ok for a office |
| 22:07.35 | Mike | ? |
| 22:07.36 | ManxPower | Mike, the MICROPHONE volume. I have not resolved the issue |
| 22:07.40 | zno | can I define globals in queues.conf? |
| 22:07.44 | ManxPower | Mike, Ask me again in a month. |
| 22:07.52 | *** join/#asterisk riksta (~rick@81-178-224-251.dsl.pipex.com) |
| 22:07.58 | Borgon | hello |
| 22:08.06 | Mike | ManxPower, the speaker mic? |
| 22:08.07 | Juggie | ManxPower, do you know anything about silence detection being implemented? |
| 22:08.11 | ManxPower | Mike, Yes. |
| 22:08.18 | ManxPower | Juggie, It is not implimented. |
| 22:08.19 | Mike | ok |
| 22:08.32 | Mike | ManxPower, what about the buttons do they feel better than GS |
| 22:08.32 | Mike | ? |
| 22:08.33 | jjg | is FWD better than nufone? |
| 22:08.35 | jjg | "better" |
| 22:08.37 | Juggie | ManxPower, is there any ongoing work in the CVS? |
| 22:08.42 | ManxPower | Mike, I would never use a GS. |
| 22:08.45 | sivana | ~seen jerjer |
| 22:08.47 | jbot | jerjer <~JerJer@dsl-107-53.che.centurytel.net> was last seen on IRC in channel #asterisk, 2d 20h 37m 4s ago, saying: 'then go have a coke and a smile'. |
| 22:08.52 | ManxPower | Mike, The phone is only $80. Go buy one for testing. |
| 22:09.07 | Mike | ManxPower, ok |
| 22:09.12 | ManxPower | Juggie, Not that I am aware of. |
| 22:09.20 | vaewyn | Get an IP-300 before a GS... is only 130$ |
| 22:09.26 | Borgon | Hello, I am a total newbie to this technology is there a faq? Can i just run asterisk with a voip provider, make calls andi can spoof ids? Or do i need to get a pbx? |
| 22:09.31 | vaewyn | and you don';t have to add the lead sheet :P |
| 22:09.34 | ManxPower | Juggie, Or at least nothing that I know of that was NOT discussed on the mailing lists. |
| 22:09.36 | delphiuk | vaewyn: thanks, that appears to have fixed it |
| 22:09.43 | vaewyn | delphiuk: no prob |
| 22:10.17 | `Sauron | Borgon: You can't spoof CID information |
| 22:10.25 | `Sauron | but you can just hook it up to a voip provider |
| 22:10.40 | IcePick__ | how new is the ${TRANSFER_CONTEXT}? - that var is not in my README.variables |
| 22:10.54 | ManxPower | IcePick__, Then where did you learn about it? |
| 22:11.11 | `Sauron | I think I've recompiled * at least daily the last 5 days |
| 22:11.16 | IcePick__ | in the wiki |
| 22:11.25 | IcePick__ | it has a note about it under transfer command. |
| 22:11.27 | ManxPower | IcePick__, The Wiki has been known to be wrong. |
| 22:11.27 | Borgon | `Sauron, so i wouldnt need extra hardware? maybe a mic or something? |
| 22:11.37 | `Sauron | borgon: Depends on what you want to do with it |
| 22:11.45 | `Sauron | you could use a softphone on your laptop, and be set |
| 22:11.47 | ManxPower | IcePick__, submit a bug about the variable not being in README.variables then |
| 22:11.57 | IcePick__ | hrm |
| 22:12.33 | ManxPower | IcePick__, It's pretty clear nobody here has ever heard of it. |
| 22:12.34 | jjg | is there a forum somewhere that people discuss providers? and issues related to? |
| 22:12.34 | shodan | ManxPower, how much cpu power does it take roughly to use Speex/iLBC/GSM ? |
| 22:12.39 | Borgon | `Sauron, i jsut want to make and receive calls, i dont need anything sophisticated or professional |
| 22:12.41 | bjohnson | `Sauron: sure you can spoof CID |
| 22:12.42 | zno | can variables be defined in queues.conf? |
| 22:12.46 | IcePick__ | true |
| 22:12.51 | IcePick__ | guess I'll try later. |
| 22:12.54 | `Sauron | bjohnson: If your voip provider doesn't override it |
| 22:13.00 | ManxPower | shodan, I'm sorry I'm out of magical fairy dust to feed my own personal Asterisk Oracle. |
| 22:13.02 | `Sauron | I'm assuming most do. |
| 22:13.24 | bjohnson | shodan: 1 horsepower per call |
| 22:13.31 | shodan | :\ |
| 22:13.36 | bjohnson | `Sauron: not from the few I've checked |
| 22:13.38 | `Sauron | half a donkey power |
| 22:13.49 | `Sauron | bjohnson: I'll have to check when I get home. |
| 22:13.49 | bjohnson | that's a STEED |
| 22:13.50 | MuchToDo | wow I think it's working! |
| 22:14.00 | `Sauron | and muchtodo just left me a message |
| 22:14.05 | bjohnson | Borgon: the answer is yes |
| 22:14.06 | `Sauron | well |
| 22:14.09 | `Sauron | he hung up on me |
| 22:14.10 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 22:14.10 | `Sauron | :( |
| 22:14.12 | MuchToDo | I didn't actually say anything ;) |
| 22:14.15 | bjohnson | prank call |
| 22:14.22 | `Sauron | I need to fix my spa-1001 |
| 22:14.28 | fa | bjohnson how can i give a dialtone from some zap channel to some user? |
| 22:14.31 | `Sauron | <PROTECTED> |
| 22:14.31 | `Sauron | <PROTECTED> |
| 22:14.31 | `Sauron | <PROTECTED> |
| 22:14.33 | Borgon | `Sauron, since when someone cant spoof cid with asterisk? |
| 22:14.46 | MuchToDo | ok I'll speak :) |
| 22:14.47 | shodan | any idea if a 75mhz 8051 CPU can do any of them ? |
| 22:14.48 | bjohnson | fa: a fxs provides a dial tone |
| 22:14.57 | `Sauron | anyone know why the spa-1001 keeps returning busy after like 4/5ths of a ring? |
| 22:15.05 | bjohnson | fa: DISA I think also gives a dial tone |
| 22:15.08 | fa | fxs? but i want to do that in extensions/context.. i am working on callback |
| 22:15.13 | greg_work | if only you could spoof over pots.. |
| 22:15.14 | fa | simple callback for cellular phone |
| 22:15.19 | MuchToDo | there, message left! |
| 22:15.25 | `Sauron | awww |
| 22:15.32 | bjohnson | fa: I haven't done that .. I saw a few examples in the wiki though |
| 22:15.40 | ManxPower | `Sauron, I guess you didn't read my postings about config the SPA-841 on the mailing list? |
| 22:16.12 | `Sauron | Manx: Not yet :) |
| 22:16.16 | ManxPower | Oh! SPA-1001. Nevermind. |
| 22:16.21 | `Sauron | Bah. |
| 22:16.23 | bjohnson | shodan: check the wiki .. a P100 with 16M RAM is in use for up to 2 concurrent calls |
| 22:16.24 | `Sauron | Indeed |
| 22:16.55 | bjohnson | the outrage! |
| 22:16.56 | shodan | 16m ram , ouch that's a lot :\ but I'll check the wiki then |
| 22:17.19 | Borgon | can i run asterisk and use it good via vmware? |
| 22:17.20 | LarsAC | florz: noch da ? |
| 22:17.24 | bjohnson | `Sauron: I had a problem that I didn't figure out with my SPAs .. I turned off call waiting on them |
| 22:17.28 | ManxPower | shodan, Asterisk does everything in software and so needs lots of CPU |
| 22:17.33 | bjohnson | Borgon: no |
| 22:17.39 | florz | LarsAC: Jo, lese gerade nochmal Deine Mail, bin gleich soweit ... :-) |
| 22:17.40 | `Sauron | bjohnson: I'll have to try that |
| 22:17.54 | `Sauron | which is sad, I'd like to have call waiting working |
| 22:17.56 | greg_work | ManxPower: what about it? |
| 22:18.06 | Borgon | bjohnson, why is that? |
| 22:18.25 | LarsAC | oki |
| 22:18.37 | shodan | I don't want to run asterisk on a 8051 , make a FXS<->ethernet bridge |
| 22:18.47 | `Sauron | someone said to try auth=md5 in sip.conf |
| 22:18.52 | shodan | a 10$ alternative to a TDM400P |
| 22:18.52 | greg_work | ManxPower: oh, this? http://lists.digium.com/pipermail/asterisk-users/2005-January/086199.html |
| 22:18.54 | delphiuk | um, now I get further compiling, but get this problem? |
| 22:18.56 | delphiuk | cannot find -lssl |
| 22:19.01 | greg_work | `Sauron: you're still having problems? |
| 22:19.12 | `Sauron | greg: with it returning a busy, yes |
| 22:19.17 | harryvv | del, did not include ssl library? |
| 22:19.21 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 22:19.51 | greg_work | `Sauron: what does it do exactly? |
| 22:20.22 | `Sauron | greg: I get someone dialing in, the default context is set to ring the spa-1001 (ext 100) |
| 22:20.35 | bjohnson | `Sauron: is it intermittent or a constant problem? |
| 22:20.43 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 22:20.49 | `Sauron | Seems to be an all the time problem |
| 22:20.55 | `Sauron | greg: then * gives me this: |
| 22:20.57 | greg_work | sip.conf: [119] username=119 type=friend secret=123 port=5060 nat=never mailbox=119 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no |
| 22:21.06 | florz | Is there anyone who feels offended if we speak German in here? |
| 22:21.17 | buddah | ich habe kein beine |
| 22:21.19 | bjohnson | I was getting that with my SPA 3000 |
| 22:21.28 | bjohnson | florz: it's all Greek to me |
| 22:21.39 | greg_work | exten=>119,1,Dial(SIP/119,15,tr) |
| 22:21.41 | bjohnson | `Sauron: the error kept changing on me |
| 22:21.47 | florz | buddah: wierd buddah, you are =:-) |
| 22:22.02 | buddah | lol |
| 22:22.06 | bjohnson | `Sauron: but never success .. even when calls answered by the fxo on the same device |
| 22:22.11 | Borgon | Is it something new that asterisk cant spoof cids or it never could? |
| 22:22.18 | florz | LarsAC: Du hast da einen SMP-Kernel laufen, ja? |
| 22:22.45 | bjohnson | Borgon: depends on your service provider. * can set the CID name and number .. but it may not go through the phone system |
| 22:22.53 | `Sauron | greg: http://www.pastebin.com/236451 |
| 22:23.13 | `Sauron | And anyone else who's interested in the spa ring busy problem |
| 22:23.17 | LarsAC | florz: ja |
| 22:23.19 | buddah | Ich bin wirklich nicht alle dass unheimlich, war das ein Zitat von einem Film |
| 22:23.47 | florz | buddah: But you are unheimlich wierd indeed =:-) |
| 22:23.58 | harryvv | bj, mirad of reasons why people do not reveal there identity |
| 22:24.09 | srt | florz: schoen dich mal zu sehen - dein hfc patch hat mir sehr geholfen :) |
| 22:24.19 | Juggie | is anyone running a cisco and the 7.3.0 firmware? |
| 22:24.46 | florz | srt: Schoen - geht gerade darum, dass das wohl nicht bei allen so ist ;-) |
| 22:24.57 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 22:25.16 | Mother_ | Juggie: yes |
| 22:25.27 | Borgon | bjohnson, aah so it depends on the provider ok |
| 22:25.53 | `Sauron | MuchToDo: you need to set CID data on your fwd line |
| 22:26.09 | srt | ich hatte vorher probleme mit 2.6er kernel, die hat er behoben. aber ne smp maschine hab ich nicht |
| 22:26.54 | fa | anyone use DISA for callback? |
| 22:27.11 | srt | fa: yes |
| 22:27.17 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 22:27.25 | `Sauron | time to go home |
| 22:27.27 | `Sauron | wavewave |
| 22:27.30 | shido6 | ZzzzZ |
| 22:27.34 | shido6 | i need cookies |
| 22:27.36 | shido6 | or a doughnut |
| 22:27.43 | shido6 | something sweet |
| 22:27.55 | HitTop | hi all~ |
| 22:28.01 | delphiuk | vaewyn: I get a bit further, but am now getting this problem: cannot find -lssl |
| 22:28.43 | ManxPower | delphiuk, Guess what! You need to install the openssl libs |
| 22:29.12 | HitTop | i have a question about connecting to asterisk that is behind nat from outside... |
| 22:29.40 | HitTop | is there any sol'n for it? or do i need to try out SER? |
| 22:29.45 | delphiuk | ManxPower: :) ok, thanks. any idea of package name? |
| 22:29.52 | Mother_ | openssl |
| 22:30.03 | ManxPower | delphiuk, Read the bottom section of this document, the part where it tells you what you need to install to build zapte and/or asterisk, http://www.asteriskpbx.org/index.php?menu=download |
| 22:30.17 | ManxPower | delphiuk, I am not your Linux Support Consultant. |
| 22:30.57 | Godsey | just got off the phone w/ polycom for an RMA :) |
| 22:31.13 | Godsey | they really don't like dealing w/ end users :) |
| 22:31.18 | delphiuk | ManxPower: ok, i apologise, I should have spotted that document first... |
| 22:31.49 | Mother_ | Godsey: that's why they want their dealers certified & tied etc. so that they get the heat |
| 22:32.11 | Godsey | ya I'm becoming a certified voip person |
| 22:32.20 | Godsey | just so I can ask them stupid questions :) |
| 22:32.37 | Mother_ | certified by Polycom? |
| 22:32.40 | Godsey | ya |
| 22:32.45 | Godsey | voip partner |
| 22:32.50 | Mother_ | OK |
| 22:33.10 | Juggie | this mitel 5055 sucks ass |
| 22:33.13 | Juggie | freezes all the time |
| 22:33.14 | Mother_ | it's sad that all these manufacturers are going into world domination mode |
| 22:33.18 | Juggie | has anyone triede these? |
| 22:33.27 | Godsey | I think it sucks alright |
| 22:33.30 | Godsey | it's a f'n phone |
| 22:33.49 | Godsey | he hinted that no dtmf after placing a call on hold is a problem |
| 22:33.53 | Mother_ | Godsey: yeah, they make it look like they're selling you a space shuttle |
| 22:34.03 | Godsey | but also sugested it's because of an unsupported platform (asterisk) |
| 22:34.29 | Mother_ | oh! yes! I had that - the local Polycom branch also said "what is that?" |
| 22:35.04 | Godsey | so I finally said oh it happens w/ our Sylantro software! |
| 22:35.06 | Mother_ | and all this crap about their phones being certified with such and such PBX systmes etc. |
| 22:35.22 | Godsey | he said "I'm sure Sylantro has that fixed now, lets call them" :) |
| 22:35.24 | Mother_ | s/systmes/systems |
| 22:35.33 | Mother_ | hehe |
| 22:35.37 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 22:35.39 | Godsey | http://forms.polycom.com/audio_files/techpartners.htm |
| 22:35.44 | Godsey | I just picked it at random :) |
| 22:35.52 | Godsey | did you find a fix for it? |
| 22:35.59 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:36.05 | Mother_ | it's all one big pile of dung, and the products and support is just as crappy, but they get to make money from the costumers *and* the dealers |
| 22:36.10 | cjk | hi, is there any win-sip-soft client out that is eays to use (like sky...) and does not use alof of ressources |
| 22:36.18 | PBXtech | is the supervised xfer stuff in the features.conf flakey for everyone else? |
| 22:36.45 | Godsey | I think I'm getting budgetone phones for home |
| 22:36.50 | Godsey | at least you get support w/ them :) |
| 22:36.54 | Mother_ | lol |
| 22:36.55 | Godsey | or maybe even iaxys |
| 22:37.25 | Mother_ | I looked at IAXy but the problem is the codecs, unless you have a phat pipe or dedicated DSL etc. |
| 22:37.34 | Mother_ | if only it had GSM... |
| 22:38.23 | ManxPower | Mother_, The IAXy has like 4k of flash and 4k of RAM. Not a lot to work with. I don't know what the CPU is, but it's too slow for GSM. |
| 22:38.53 | Godsey | oh |
| 22:38.54 | Mother_ | ManxPower: indeed, I'm not knocking on it as such, just making a wish :) |
| 22:38.55 | mishehu | guess that means we can't use g729a or speex on it either ;-) |
| 22:38.59 | Godsey | well then SPA-2100 |
| 22:39.08 | Godsey | but last i looked they were still coming soon :) |
| 22:39.10 | ManxPower | This has all been talked about on the mailing lists. |
| 22:39.22 | Godsey | I don't care about bandwidth tho |
| 22:39.26 | pulu | Has anyone used those new ATA's with taht chinese chipset that do GSM, ilbc, etc and can be flashed for iax? |
| 22:39.32 | pulu | they're cheaper than iaxy's, too |
| 22:39.36 | Mother_ | nope |
| 22:40.07 | Godsey | I don't care so much about price as long as it's under $120/port |
| 22:40.36 | Godsey | I'm leary of vendor lockin |
| 22:40.48 | Godsey | and if I get digium cards I'm locked to asterisk right? |
| 22:40.53 | jjg | any nufone reps here? |
| 22:41.39 | echion | anyone can tell me something about this error message? |
| 22:41.42 | Mother_ | Godsey: I can't see anything wrong with that :) |
| 22:41.50 | echion | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 22:41.50 | echion | FATAL: Error running install command for wct4xxp |
| 22:41.51 | Mother_ | better than being locked to Cisco |
| 22:41.52 | echion | ?? |
| 22:42.08 | Godsey | well cisco works better for queues and agents at the moment |
| 22:42.16 | ManxPower | Godsey, Think of it like marriage. It either will be OK or you won't realize how big of a mistake you've made until it's far, far too late. |
| 22:42.21 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:42.29 | Mother_ | lol |
| 22:42.37 | ManxPower | echion, you read the Zaptel readme? |
| 22:42.50 | Godsey | I was going to look at sipX but gave up quickly |
| 22:43.20 | blitzrage | how odd... I'm having things sound like they're a skipping record on my IAX2 connection |
| 22:43.28 | blitzrage | this is how it, this is how it, this is how it sounds |
| 22:43.42 | Godsey | Mother: what did you do to fix dtmf after hold? :) |
| 22:43.44 | echion | ManxPower, the ones following the driver are quite brief? |
| 22:44.06 | Mother_ | Godsey: nothing, I've not had that problem :) |
| 22:44.08 | pulu | blitzrage: that's cool |
| 22:44.12 | blitzrage | pulu: not really |
| 22:44.15 | pulu | chan_maxheadroom |
| 22:44.20 | blitzrage | pulu: I'm trying to talk to someone on customer support |
| 22:44.31 | Godsey | oh |
| 22:44.52 | blitzrage | it was working great for like 10 mins... then all of a sudden started having a problem |
| 22:44.59 | shido6 | ruh roh |
| 22:45.01 | Godsey | we're working on becoming linksys sp |
| 22:45.04 | blitzrage | and I normally don't have this problem.... |
| 22:45.15 | blitzrage | wonder if upgrading would help |
| 22:45.17 | zoa | all nufone support go to #nufone |
| 22:45.17 | Godsey | we just want to buy pap2-na adaptors for $50 :) |
| 22:45.57 | Mother_ | a nice new bussiness would be to be a middleman between all these hardware companies and people wanting to be partners etc. |
| 22:45.59 | buddah | i got one right here |
| 22:46.02 | buddah | and i cant get it to work |
| 22:46.04 | Mother_ | Cert-A-Go |
| 22:46.05 | buddah | well |
| 22:46.07 | buddah | outgoing is fine |
| 22:46.13 | redder86 | zoa: but JerJer isn't even there |
| 22:46.14 | buddah | but it wont register, dtmf dont work, wont take incoming |
| 22:46.24 | Godsey | we just want to sell the pap2-na to our customers :) |
| 22:46.24 | andersee | anyone know if the wcfxo driver works on powerpc? |
| 22:46.28 | zoa | shido6 is |
| 22:46.36 | redder86 | who is that? |
| 22:46.42 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 22:46.42 | zoa | nufone guy |
| 22:47.28 | blitzrage | damnit... all that time wasted |
| 22:47.31 | dan2 | zoa: nah, Jerjer is Nufone guy |
| 22:47.40 | Mother_ | are there any USB to FXS adapters which are not either tied to Skype, tied to some propietary software, or suck? |
| 22:47.48 | blitzrage | just about to get the number for my problem, and it starts skipping like a record again |
| 22:48.53 | *** part/#asterisk andersee (~andersee@codepoet.org) |
| 22:50.05 | zoa | if i say shido6 is a nufone guy |
| 22:50.11 | zoa | try to at least believe me |
| 22:50.16 | zoa | or talk to him |
| 22:50.19 | zoa | and find out yourself |
| 22:50.21 | `Sauron | BWAHAHA |
| 22:50.41 | mikegrb | heh |
| 22:51.03 | ManxPower | Shido is either EMPLOYED by NuFone or works for NuFone as a CONSULTANT. Either way, reporting a problem to him may get something done. |
| 22:51.23 | ManxPower | But perhaps sending an e-mail via official support channel will be better. Send e-mail to support@nufone.net |
| 22:51.59 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:51.59 | ManxPower | And why is it that only people I'm not attracted to actually contact me via that site. |
| 22:52.06 | file | shido works for NuFone... |
| 22:52.07 | shmaltz | hi everybody |
| 22:52.10 | file | well, shido6 |
| 22:52.14 | shido6 | wha? |
| 22:52.30 | ManxPower | shido6, someone claims the automated sign up or automated DID adding is broken. |
| 22:52.35 | jjg | it is |
| 22:52.38 | jjg | last night too |
| 22:52.40 | shmaltz | I'm running CVS 12-21 does anybody know of a cisco blindxfer problem? |
| 22:52.42 | file | which happened because the site moved boxes |
| 22:52.47 | file | AND |
| 22:52.47 | shido6 | that may be , we're fixing it - we had a drive failure |
| 22:52.49 | file | it even clearly states |
| 22:52.58 | shido6 | Jeremys been working his tail off getting things back in order |
| 22:53.04 | file | We are currently updating our system. This will not effect operation of your account. |
| 22:53.04 | file | However, oddities may occur with this site. |
| 22:53.06 | jjg | maybe put something on the site, or disable the page? not a bad idea |
| 22:53.15 | file | note that, "However, oddities may occur with this site." |
| 22:53.24 | jjg | would take like ~ 5 seconds |
| 22:53.43 | jjg | i don't feel all cozy about putting my credit card into a system that says oddities may occur |
| 22:54.03 | harryvv | I am configuring x-lite for asteriks and have completed it but getting one warning when asterisk is started. What is a common misconfiguration when there is a notice of Registration from name <sip:username@ipaddressofasteriskserver> failed for clientipaddress ? |
| 22:54.05 | file | nobody is forcing you to |
| 22:54.28 | jjg | did I say that someone was forcing me? |
| 22:54.35 | shmaltz | I'm running CVS 12-21 does anybody know of a cisco blindxfer problem? |
| 22:54.39 | ManxPower | harryvv, That means you don't have a [username] in sip.conf |
| 22:54.53 | harryvv | I created one. |
| 22:54.56 | harryvv | ohh |
| 22:55.05 | harryvv | you mean context? |
| 22:55.07 | jjg | what i'm getting at is that it is stupid to leave the page up that allows someone to put in credit card info if the site is broken |
| 22:56.02 | zoa | unless everyone is trying to fix it |
| 22:56.05 | zoa | and it almost works |
| 22:56.05 | denon | welcome to the interweb |
| 22:56.06 | harryvv | so you mean rigt before secret=secretpassword should be username=thisname? |
| 22:56.20 | jjg | zoa : heh, yeah right |
| 22:56.27 | denon | besides, there's a 99% chance your credit card number will go via cleartext email to someone in norway |
| 22:56.27 | ManxPower | a username=username and a [username] |
| 22:56.36 | tessier_ | oh crap, I just moved the whole intarweb into the trash! help!111!! |
| 22:56.40 | harryvv | okay |
| 22:56.41 | ManxPower | [robertdobbs] |
| 22:56.43 | ManxPower | type=friend |
| 22:56.51 | ManxPower | username=robertdobbbs |
| 22:56.54 | tessier_ | context=subgenious |
| 22:56.58 | harryvv | I see |
| 22:58.54 | blitzrage | ManxPower: pretty sure username= is just used for the initial registration and not subsequent registrations (which is why [username] needs to be the username) |
| 22:59.36 | *** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com) |
| 23:00.48 | ManxPower | blitzrage, As I understand it, username=ernie is only used for outgoing connections. [ernie] is using for incoming connections. Perhaps the reverse. |
| 23:00.48 | ManxPower | Therfore for type=friend they should be the same. |
| 23:00.55 | ManxPower | I HAVE been wrong before, however. |
| 23:01.07 | Nugget | say it ain't so. |
| 23:01.08 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 23:01.43 | blitzrage | ManxPower: apparently oej went through the code (this is SIP I'm talking about), in that username=ernie is only used for the first registration, and [ernie] is used for subsequent registrations (which is why [ernie_blah] can't be different from username=ernie), but I have also been wrong before :) |
| 23:01.48 | blitzrage | I need to clear that up with OEJ |
| 23:02.13 | ManxPower | blitzrage, Either way would be consistant with needing them to be both the same. |
| 23:02.37 | tzanger | evening blitzrage |
| 23:02.42 | buddah | anyone know how i setup the sip.conf entry for a pap2-na? |
| 23:02.49 | buddah | still not able to get it to register |
| 23:03.35 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 23:04.51 | ManxPower | blitzrage, seems like a horribly overly complex way of doing it. |
| 23:08.18 | *** join/#asterisk mChicago (~jt@81-178-211-22.dsl.pipex.com) |
| 23:08.41 | mChicago | good morning all |
| 23:09.03 | buddah | mornin |
| 23:09.26 | *** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net) |
| 23:09.33 | mChicago | is this a good place to ask about installing windows [washes mouth out] versions of asterisk? |
| 23:10.39 | Mavvie | there are a lot of worser places to ask it. |
| 23:10.50 | harryvetch | when that happens we will get 10 times the amount of noobs poping in here :) |
| 23:11.43 | ManxPower | mChicago, Are you familiar with a Klingon Pain Stick? Well you get jabbed with one for every question about Asterisk and Windows. |
| 23:11.46 | blitzrage | ManxPower: yes you're right. The thing that sucks is that I can't do something like, [service-provider-A] and username=blitzrage |
| 23:11.49 | blitzrage | tzanger: y0! |
| 23:11.54 | harryvetch | It just has not caught on that well...yet. |
| 23:12.21 | ManxPower | I STILL think the whole Win32Asterisk is still an April Fool's joke. |
| 23:12.38 | mChicago | ManxPower: if its anyhting like the "persuadatron" then it only works in close quaters and i know im safe in my flat. .Wait! you mean theres a real windows asterisk not just one through coLinux ? |
| 23:13.03 | stevekstevek | mChicago: why??? |
| 23:13.03 | `Sauron | WOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOT |
| 23:13.16 | greg_work | stevekstevek: thats exactly what i was thinking |
| 23:13.18 | tessier_ | How's RealTime coming along? The comments on the RealTime wiki page don't sound promising. Are those people clueless about RealTime? |
| 23:13.23 | ManxPower | mChicago, I have no idea. I'm afraid to even look at it. |
| 23:13.25 | `Sauron | Alternately, how about a cattle prod? |
| 23:13.42 | buddah | ooooh |
| 23:13.46 | buddah | cattle prods |
| 23:13.47 | harryvetch | manxpower, got a notice that my sip client x-lite is now REAchable via asterisk |
| 23:14.19 | ManxPower | That I want is a Nerf Bat. |
| 23:15.07 | harryvetch | its got another problem though It was not set as host=dynamic because its not a dynamic ip address. |
| 23:15.15 | `Sauron | I called my mommy through * to wish her a happy birfday. |
| 23:15.15 | harryvetch | and its complaining about it. |
| 23:15.18 | *** join/#asterisk Asta2 (~123@66.180.175.16) |
| 23:15.33 | ManxPower | harryvetch, Sure you can. host=dynamic really should be renamed host=willregisterwithis |
| 23:15.54 | harryvetch | I put the clients static ipaddress in it. |
| 23:16.25 | harryvetch | probebly not the thing to do. |
| 23:16.25 | blitzrage | no MWI in Realtime |
| 23:16.25 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-6-3.d4.club-internet.fr) |
| 23:16.25 | harryvetch | no dhcp on this network |
| 23:17.40 | Asta2 | hi there, I'm a newbiew in Voice IP and here goes the question: is it possible to arrange some kind of a "forward" a phone call connecting 2 sips if 1 of them has a PSTN number and the other can make outbound calls to PSTN |
| 23:18.38 | *** part/#asterisk shodan (~shodan@216.113.99.175) |
| 23:18.51 | blitzrage | Asta2: yep |
| 23:19.09 | Godsey | what is the new ATA device that does iax called? |
| 23:19.52 | Mother_ | ATAiwanese thing |
| 23:19.52 | buddah | so i have 2 polycom ip 500s, setup to work with a static ip |
| 23:19.57 | buddah | everything works fine |
| 23:19.57 | Asta2 | blitzrage: any directions on where I should look to understand how to do it? |
| 23:20.02 | buddah | clients want to put them behind a router |
| 23:20.10 | buddah | all i gotta do is turn dhcp on and they should work fine right? |
| 23:20.57 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 23:21.17 | *** join/#asterisk darkskiez (~mhb@host-84-9-81-116.bulldogdsl.com) |
| 23:21.20 | dan2 | hmmm |
| 23:21.27 | dan2 | I'm not getting any rtp when people are put on hold |
| 23:21.31 | dan2 | anybody have any idea |
| 23:22.47 | Godsey | my setup breaks kinda when people are put on hold too :) |
| 23:23.10 | *** join/#asterisk RoyK (~roy@i171219.dsl.tjukkband.no) |
| 23:23.33 | dan2 | Godsey: oh? |
| 23:23.35 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 23:23.42 | LUTOR_ASI | hi guys. |
| 23:24.03 | LUTOR_ASI | somebody knows how to change a codec in the dialplan??? |
| 23:25.34 | Godsey | I have Polycom IP-300 phones |
| 23:25.36 | dan2 | LUTOR_ASI: you can't |
| 23:25.44 | Godsey | when people use hold, the phones no longer send dtmf |
| 23:25.44 | jarrod | setvar(SIP_CODEC=ulaw) ? |
| 23:25.52 | Godsey | so you can't do # transfer etc... |
| 23:26.13 | dan2 | Godsey: my phones aren't receiving rtp when they're on hold |
| 23:26.38 | Godsey | I don't know if mine are, or even how to check :) |
| 23:26.54 | Godsey | I can't figure out queues for the life of me |
| 23:27.05 | Godsey | and call waiting goes off in tech's ears constantly :) |
| 23:27.25 | buddah | fuck |
| 23:27.32 | buddah | polycoms arent working in dhcp |
| 23:27.45 | buddah | not connecting to boot server |
| 23:27.54 | Godsey | my polycoms boot w/ dhcp |
| 23:28.01 | Godsey | are you using tftp or ftp server? |
| 23:28.02 | buddah | yeah i had them setup |
| 23:28.03 | buddah | ftp |
| 23:28.11 | buddah | they were working fine |
| 23:28.25 | buddah | using static |
| 23:28.41 | buddah | but now they got put behind a router |
| 23:29.06 | Godsey | routers don't forward broadcasts |
| 23:29.09 | LUTOR_ASI | jarrod: did you ever probe that...!? (SIP_CODEC=ulaw)? |
| 23:29.14 | buddah | ok |
| 23:29.17 | buddah | so that means i gotta do what? |
| 23:29.26 | harryvetch | when I registered with voipjet I got my iax info. no sip. is that the way it is? |
| 23:29.28 | jarrod | i use it, yes |
| 23:29.30 | jarrod | setvar |
| 23:29.34 | Godsey | I'm not sure :) |
| 23:29.40 | Godsey | maybe setup a dhcp proxy on router? |
| 23:29.52 | KalD|Work | Is silent supression a function of the codec or protocol? or both? |
| 23:29.54 | Godsey | I would run dhcp on that router |
| 23:30.13 | harryvetch | anyone here run voipjet |
| 23:33.30 | terrapen | i tried voipjet |
| 23:33.34 | terrapen | on their free trial |
| 23:33.37 | terrapen | and was not impressed |
| 23:33.45 | terrapen | and now they are spamming me |
| 23:35.00 | |Vulture| | lol I enjoy broadvoice, nufone, and vpc |
| 23:36.45 | bjohnson | I use voipjet some |
| 23:37.08 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:37.35 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:38.35 | sjaak538 | before voipjet was better then now. |
| 23:39.00 | KalD|Work | anyone had any luck w/ latest CVS on OSX? |
| 23:39.45 | harryvetch | bjohnson, voipjet does not except iax ? |
| 23:40.07 | sjaak538 | iax2 |
| 23:40.55 | mishehu | I normally have a decent experience using voipjet. |
| 23:40.56 | harryvetch | okay going to setup my iax for voipjet. I did not know the register => statment needs to go at the top after [general] |
| 23:42.05 | sjaak538 | On some route voipjet is okay |
| 23:42.22 | mishehu | blah. another skype article on /. |
| 23:42.28 | mishehu | I don't know why anybody hypes that program. |
| 23:42.39 | redder86 | yeah, one would think that Skype were paying the /. editors |
| 23:44.18 | redder86 | slashdot has been the home page on one of my browsers for over a year. The content has become so dull that I will change it around whenever I find a suitable replacement. (I like my homepage to have news on it. So although I use Google, I want some news there.) |
| 23:45.22 | Godsey | news.google.com :P |
| 23:45.30 | mishehu | occaissional /. has a good article, but even a few of the ones dedicated to asterisk were lame. |
| 23:45.50 | Godsey | I normally read theregister for news |
| 23:46.52 | terrapen | hahah |
| 23:46.59 | Nugget | heh |
| 23:46.59 | terrapen | oops |
| 23:47.14 | *** join/#asterisk K-Sensei (~K-Sensei@user-37ka4b7.dsl.mindspring.com) |
| 23:47.28 | blitzrage | <PROTECTED> |
| 23:47.42 | mikegrb | yes |
| 23:47.47 | mikegrb | welcome to two years ago |
| 23:47.49 | mikegrb | :D |
| 23:49.27 | Nugget | at least michael finally got fired from /. |
| 23:49.34 | K-Sensei | Okay, any NMI gurus here? I have gone through everything I could find on google. I installed the T100P on a RedHat FC3 box, and now the system is unstable. It looks like, firstly, the t1xxp module is being probed, but "kobject_register failed for t1xxp (-17)". And every now and again I get "Uhuh. NMI received for unknown reason 35 on CPU 0. -- Dazed and confused, but trying to continue." |
| 23:49.36 | Nugget | that's a ten-fold improvement, imho. |
| 23:49.52 | buddah | jesus christ, we've been waiting for 3 weeks for DIDs from some company |
| 23:49.56 | buddah | and they still havent delivered |
| 23:49.58 | buddah | pathetic |
| 23:50.28 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 23:50.31 | fearnor | buddah: which company? ;) |
| 23:50.35 | buddah | empwoer |
| 23:50.39 | buddah | err |
| 23:50.39 | buddah | empower |
| 23:50.48 | fearnor | heh |
| 23:50.49 | modulus_ | how much per month does a toll free number cost from nufone? |
| 23:50.56 | fearnor | mod: 0 |
| 23:50.57 | terrapen | hmmm |
| 23:51.00 | terrapen | Skype for OS X |
| 23:51.08 | terrapen | does Skype have spyware? |
| 23:51.19 | fearnor | not yet. |
| 23:51.45 | postel | No, but its made by the kazaa ppl, and thats food for thought |
| 23:52.33 | terrapen | heh, true |
| 23:52.39 | Nugget | phones for pirates. |
| 23:52.53 | Delmar | i hate skype. |
| 23:53.04 | buddah | how do i setup voicemail so that its 1 mailbox that 2 phones share? |
| 23:53.05 | Delmar | they are deceitful slappers. |
| 23:53.34 | redder86 | Godsey: http://news.google.com/?ned=us&topic=t |
| 23:53.41 | redder86 | unfortunately, they too are reporting about Skype there |
| 23:54.04 | stevekstevek | it's annoying that they automatically publish your directory entry.. |
| 23:56.50 | stevekstevek | skype has 11 users with "asterisk" in their "skype name" |
| 23:57.02 | fearnor | skype is hype. |
| 23:57.12 | riksta | you mean shite |
| 23:57.48 | buddah | with polycoms how do you answer call waiting? |
| 23:58.41 | crash3m | press the down arrow |
| 23:58.44 | crash3m | then hit 'answer' |
| 23:59.34 | buddah | nice |