| 00:17.00 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-27.tricom.net) |
| 00:33.56 | *** join/#asterisk propellerhead (n=yogurt2u@host19.190-230-40.telecom.net.ar) |
| 00:38.17 | *** join/#asterisk sjobeck (n=Adium@137.118.193.9) |
| 01:05.04 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 01:07.48 | ricko73 | question about moving from zaptel to dahdi |
| 01:08.04 | ricko73 | does the channel name in the dialplan change from Zap/1 to ??/1 |
| 01:08.40 | [TK]D-Fender | ricko73: DAHDI/ |
| 01:09.21 | ricko73 | gotcha. seems with 1.4 that Zap/1 still works |
| 01:13.30 | [TK]D-Fender | ricko73: There is a compatibility flag that can be set to allow use of Zap |
| 01:17.05 | ricko73 | either way, thanks. we're adding support for dahdi to Astlinux so I'm just updating our documentation |
| 01:22.10 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:24.03 | russellb | both work in 1.4 for dialing, only DAHDI in 1.6 |
| 01:24.20 | russellb | the option is for what the channel names will show up as in the channel name |
| 01:24.43 | ricko73 | russellb: thanks for the clarification |
| 01:24.45 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 01:24.48 | russellb | np |
| 01:25.12 | ricko73 | dahdi is pretty much required for the newer analog cards right? |
| 01:25.52 | ricko73 | I could never get this 8 port card working (properly) until I started using dahdi (today) |
| 01:25.55 | russellb | well, yeah, it's very strongly recommend for everything now, though, since zaptel isn't being maintained |
| 01:26.05 | russellb | i'm not surprised |
| 01:31.04 | *** part/#asterisk sjobeck (n=Adium@137.118.193.9) |
| 01:36.47 | coppice | I'm never surprised when stuff doesn't work :-) |
| 01:37.35 | russellb | trollllllll |
| 01:37.55 | russellb | :-p |
| 01:38.18 | russellb | I guess you didn't specify anything more than "stuff", which is pretty vague. |
| 01:39.22 | coppice | anything at all that ever works should be a source of celebration |
| 01:42.53 | coppice | most people take this attitude with relationships working, hence they have wedding anniversaries |
| 01:44.28 | hesco | I'm building a new server, do I need libpri if I have no PRI hardware in the mix? |
| 01:45.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:45.25 | hesco | also, are there any recipes for building an asterisk installation for AWS' EC2 ??? |
| 01:46.35 | russellb | hesco: no libpri needed |
| 01:46.52 | russellb | and for the latter, i don't know, *refers you to google* |
| 01:47.04 | WindowsUser | EC2 recipie + asterisk guide that says you dont need libpri |
| 01:50.45 | ecret | Can someone recommend a cheap DID service? Mine is down and i need a quick 1800#. Thanks. |
| 01:51.41 | WindowsUser | ~itsp-us |
| 01:51.55 | hesco | We're paying $3 a month at diamondcard.us |
| 01:51.58 | WindowsUser | hrm, i thought that was the botflag |
| 01:52.03 | hesco | not for 800 but a local number |
| 01:52.24 | hesco | I found this recipe for a Fedora install: http://voxilla.com/2009/02/13/asterisk-amazon-ec2-1178 |
| 01:52.59 | hesco | I've ben building a Debian Lenny AMI image myself, though, but I hope this might be helpful. |
| 01:53.18 | hesco | thanks russellb on the PRI question. |
| 01:53.28 | russellb | np. |
| 01:54.10 | WindowsUser | ecret: also check these out: vitelity.com les.net voip.ms |
| 02:10.57 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:35.49 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-70-199.kgldga.dsl.dynamic.tds.net) |
| 02:37.03 | phunyguy | quick question folks: I know you guys may get this one all the time. I understand that you cannot use a regular old modem with asterisk, why is that? |
| 02:38.23 | phunyguy | is it because something has to take the incoming voice signal and convert it to be able to send out the line to PTSN? |
| 02:39.35 | coppice | I understand that you cannot use a <arbitrary unrelated product> with asterisk, why is that? |
| 02:39.57 | jaytee | ~101 |
| 02:39.58 | infobot | i heard 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:40.06 | jaytee | ~book |
| 02:40.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:40.46 | jaytee | some basics in the first link and some good info in the beginning of the book about traditional phone systems and lots more on voip and asterisk |
| 02:43.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:52.38 | phunyguy | wow |
| 02:52.54 | eppigy | fungi |
| 02:52.58 | phunyguy | was just a simple question, and how is that an "arbitrary unrelated product" |
| 02:53.10 | phunyguy | im not looking to read a technical manual |
| 02:53.15 | eppigy | u a arbitrary |
| 02:53.20 | phunyguy | what is the point of IRC if nobody wants to chat? |
| 02:53.25 | eppigy | lets chat |
| 02:53.28 | eppigy | how r u |
| 02:53.31 | phunyguy | a/s/l? |
| 02:53.34 | phunyguy | (lol) |
| 02:53.39 | eppigy | 12/f/ca |
| 02:53.43 | phunyguy | mmmm |
| 02:53.48 | phunyguy | xD |
| 02:53.48 | eppigy | =^.^= |
| 02:53.58 | phunyguy | anyways |
| 02:54.02 | phunyguy | was just a simple question i asked |
| 02:54.09 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:54.14 | phunyguy | didnt think i had to read a tech manual to find the answer. |
| 02:54.25 | eppigy | ou should anyway |
| 02:54.28 | eppigy | learnign is fun |
| 02:54.54 | phunyguy | ok should I read the entire bible because I wanted to know what 1 page said? |
| 02:54.59 | phunyguy | comon now. |
| 02:55.06 | eppigy | no |
| 02:55.12 | eppigy | scan the index |
| 02:55.16 | eppigy | flip to the proper page |
| 02:56.40 | phunyguy | i just wanted to know why you can't connect an asterisk box to my phone line going out to the phone company, and make a call with lets say a software based ip phone |
| 02:56.53 | [T]ank | what causes the dst field in the cdr to not be the priority that answered the call. I have exten => 801XXXXXXX,1,Answer() as the first line. Then later on it does a goto and sends it off to another extension. But in the cdr it shows the other extension as the dst instead of the 801XXXXXXX |
| 02:56.56 | phunyguy | should be a simple answer |
| 02:57.21 | phunyguy | (with a v.92 modem or something) |
| 02:57.33 | phunyguy | was just curious what the modem lacks |
| 02:57.47 | phunyguy | "the ability to _________" |
| 02:57.51 | [T]ank | phunyguy: what are you trying to accomplish? |
| 02:57.56 | phunyguy | im not |
| 02:58.10 | phunyguy | was just curious and googling only told me that it wont work |
| 02:58.12 | phunyguy | but now why |
| 02:58.16 | phunyguy | not* |
| 02:58.34 | [T]ank | once again.. what are you trying to do |
| 02:58.38 | coppice | a modem is a thing that whistles down the phone. if you want to whistle, its perfect for the job |
| 02:58.42 | [T]ank | i got here after you posted your question |
| 02:58.46 | phunyguy | ok |
| 02:59.27 | phunyguy | simple asterisk setup connected to the phone company (analog - regular old service), with software phones around the house |
| 02:59.38 | [T]ank | phunyguy: just get a sip provider and do away with the need for the modem |
| 03:00.04 | phunyguy | i dont want to ditch the phone service |
| 03:00.09 | [T]ank | will cost you less money |
| 03:00.17 | phunyguy | *sigh* |
| 03:00.21 | phunyguy | im not spending anything |
| 03:00.30 | phunyguy | was just curious as to what the modem lacks |
| 03:00.37 | eppigy | the modem is not made fior it |
| 03:00.40 | [T]ank | wow... how are you getting free phone from a pots carrier? |
| 03:00.41 | eppigy | it is very simple |
| 03:00.52 | eppigy | a modem is made to convert binary data |
| 03:00.57 | eppigy | into analog waveform |
| 03:01.10 | phunyguy | not digital audio... |
| 03:01.12 | eppigy | if you want to use pots |
| 03:01.12 | phunyguy | right? |
| 03:01.21 | eppigy | get an analog interface |
| 03:01.24 | eppigy | from digium |
| 03:01.27 | eppigy | or someone similiar |
| 03:01.39 | coppice | if you are talking about a winmodem, the software can be completely replaced, and the harwdare can do the job. the problem there is just that nobody has bothered to write drivers |
| 03:02.15 | [T]ank | seriously, how are you getting free pots... I want int. |
| 03:02.16 | eppigy | GET TO IT |
| 03:02.16 | [T]ank | in |
| 03:02.24 | jaytee | TRABAJO! |
| 03:02.27 | eppigy | DONDE |
| 03:02.32 | jaytee | hehe |
| 03:02.33 | eppigy | SI ME GUSTA |
| 03:04.10 | eppigy | owned |
| 03:04.33 | eppigy | asking too many questions, nullrouted |
| 03:04.35 | hesco | in make menuconfig, what means Module Embedding ??? |
| 03:04.39 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-70-199.kgldga.dsl.dynamic.tds.net) |
| 03:04.42 | phunyguy | for petes sake. |
| 03:04.52 | eppigy | hesco: you can choose to have loadable modules |
| 03:04.58 | eppigy | or embedded modules |
| 03:05.08 | phunyguy | wow, last line I got was "eppigy> or someone similiar" |
| 03:05.17 | hesco | embedded in the built binary of asterisk, I presume ? |
| 03:05.29 | phunyguy | so digium... not looking to spend a ton of money. |
| 03:05.31 | hesco | or in the kernel ? |
| 03:05.39 | [T]ank | then you did not see me asking how you are getting free POTS |
| 03:05.45 | phunyguy | im not getting free pots |
| 03:05.51 | eppigy | hesco: asterisk |
| 03:05.56 | hesco | thanks |
| 03:05.57 | eppigy | they are not kernel modules |
| 03:05.59 | phunyguy | but i like having a phone line that works if my power goes out |
| 03:06.01 | eppigy | np |
| 03:06.14 | coppice | he's getting free flowers, and he's looking for free pots to put them in |
| 03:06.21 | eppigy | zing |
| 03:06.23 | eppigy | ! |
| 03:06.29 | phunyguy | troll? lol |
| 03:06.31 | [T]ank | phunyguy: Ahhhh, you said you werent paying anything for it |
| 03:06.33 | jaytee | I want a pebble bed reactor in my basement....but first I want a basement :-) |
| 03:06.39 | phunyguy | OH, confusion |
| 03:06.45 | phunyguy | meant im not spending any money |
| 03:06.48 | phunyguy | like |
| 03:06.52 | eppigy | you must spend money |
| 03:06.53 | phunyguy | not buying more hardware |
| 03:06.58 | phunyguy | :) |
| 03:07.01 | eppigy | unless you can build what you need |
| 03:07.12 | phunyguy | you mention getting a SIP provider |
| 03:07.16 | phunyguy | which is possible... |
| 03:07.23 | phunyguy | but would that still need more hardware? |
| 03:07.27 | eppigy | well your internet will probably not be accesible |
| 03:07.31 | eppigy | in the event of power outtage |
| 03:07.56 | eppigy | unless you want to buy some 2200va backups batteries |
| 03:08.00 | eppigy | 8[] |
| 03:08.03 | phunyguy | well i wanted to keep the service so i can take my regular 1970 analog phone, plug it in and make a call if the power goes out |
| 03:08.14 | phunyguy | :) |
| 03:08.43 | phunyguy | hurricane comes through, knocks out power, and my wife |
| 03:08.47 | phunyguy | might need to call an ambulance |
| 03:08.49 | phunyguy | ;) |
| 03:09.06 | eppigy | that is touching |
| 03:09.15 | phunyguy | but yeah |
| 03:09.22 | eppigy | I remember when I got my first gun |
| 03:09.26 | phunyguy | still have a pending question... |
| 03:09.28 | phunyguy | :) |
| 03:09.31 | eppigy | I lived in the ghetto |
| 03:09.38 | phunyguy | <phunyguy> but would that still need more hardware? |
| 03:09.43 | phunyguy | (SIP Provider) |
| 03:09.43 | eppigy | and I had a nightmare some dudes broke in and raped my gf in front of me |
| 03:09.44 | carrar | he ghetto |
| 03:09.45 | carrar | the |
| 03:10.02 | eppigy | I went out and purchased an ak-47 clone the next day |
| 03:10.09 | phunyguy | ak-74? |
| 03:10.09 | phunyguy | lol |
| 03:10.17 | eppigy | sl95-mb |
| 03:10.22 | phunyguy | ahhh |
| 03:10.48 | eppigy | slr 95-mb rather |
| 03:10.59 | eppigy | bulgarian clone |
| 03:11.18 | phunyguy | *sigh* |
| 03:13.11 | eppigy | with this many questions |
| 03:13.17 | eppigy | you need to do due diligence |
| 03:13.23 | eppigy | and research onm your own |
| 03:13.39 | phunyguy | this many? meaning more than 1? |
| 03:13.44 | phunyguy | because I only had 2 |
| 03:13.44 | eppigy | yes |
| 03:13.46 | phunyguy | lol |
| 03:13.55 | eppigy | I find out everything on my own |
| 03:14.11 | eppigy | and when I have trouble finding and answer |
| 03:14.17 | eppigy | I just search harder |
| 03:14.22 | phunyguy | ok, one more. You got a good recipe for blueberry crumble? |
| 03:14.26 | phunyguy | xD |
| 03:14.27 | eppigy | I will ask advice for like what works well for people |
| 03:15.04 | phunyguy | and my last question was just a yes/no question.. |
| 03:15.06 | coppice | phonyguy: Put them in liquid nitrogen, and then hit them |
| 03:15.24 | phunyguy | mmmm... liquid notroge |
| 03:15.25 | phunyguy | n |
| 03:15.33 | phunyguy | nitrogen* |
| 03:15.35 | phunyguy | wow nevermind |
| 03:15.39 | phunyguy | screwed that one up |
| 03:15.40 | phunyguy | lol |
| 03:17.15 | phunyguy | does it require additional hardware. |
| 03:17.20 | phunyguy | thats all i want to know |
| 03:17.29 | eppigy | The thing is |
| 03:17.30 | phunyguy | i will probably never do this, but i was just curious |
| 03:18.10 | eppigy | this is like me asking someone how do I set up a site to site vpn |
| 03:18.35 | eppigy | and they were like well I mean you shoudl decide a few things, likew which encryption algorithm you will use |
| 03:18.38 | phunyguy | Install OpenVPN, set up certs, start services, set up routing |
| 03:18.39 | phunyguy | done. |
| 03:18.46 | eppigy | and then I replied, "What is encryption." |
| 03:18.47 | phunyguy | :) |
| 03:18.48 | eppigy | ? |
| 03:19.01 | eppigy | You do not know the basic fundamentals |
| 03:19.08 | eppigy | surrounding the topic |
| 03:19.53 | eppigy | and you should remedy that |
| 03:20.01 | eppigy | rather than expecting other people |
| 03:20.09 | eppigy | to explain it to you |
| 03:22.17 | *** join/#asterisk Defraz (n=tim@24-117-236-174.cpe.cableone.net) |
| 03:24.02 | *** join/#asterisk R0b3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 03:24.06 | R0b3Rt | God Damn. |
| 03:24.10 | R0b3Rt | Ive got some sip issues. |
| 03:24.12 | R0b3Rt | New user. |
| 03:24.30 | R0b3Rt | Created a sip extension and it connects fine its a soft phone. |
| 03:24.49 | R0b3Rt | I detailed my config to connect to my voip provider and no dice at all, it doesnt even appear to register. |
| 03:24.56 | R0b3Rt | or appear to try to register. |
| 03:25.00 | R0b3Rt | ports are free. |
| 03:28.56 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-70-199.kgldga.dsl.dynamic.tds.net) |
| 03:29.04 | phunyguy | getting real tired of this thing crashing |
| 03:30.18 | phunyguy | look I know I don't know a lot... but i merely was just curious and wanted to see what was possible before I start doing any major research on it |
| 03:30.37 | R0b3Rt | is there a question? |
| 03:30.59 | phunyguy | yes but every time i ask them, i get the book thrown at me |
| 03:31.06 | R0b3Rt | also, why is exetel secretly hiding thier sip server from me, and me only ? :p[ |
| 03:31.20 | R0b3Rt | phunyguy, ask a question, or im gonna throw a book at u |
| 03:31.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:31.26 | phunyguy | i already did |
| 03:31.31 | phunyguy | then PC crashed |
| 03:31.41 | phunyguy | lol |
| 03:31.55 | R0b3Rt | dude are u an exetel customer ? |
| 03:32.04 | R0b3Rt | repeat the question please i didnt see it. |
| 03:32.15 | R0b3Rt | shit my bad |
| 03:32.17 | R0b3Rt | wrong chan |
| 03:32.48 | phunyguy | i asked why a regular v.92 modem cant be used with asterisk, got the answer to that after some book dodging, then was told to get a SIP provider so i wouldnt have to worry abotu a modem, and then i asked if that would still require more hardware. |
| 03:33.17 | phunyguy | then i got punched in the face right before my PC crashed |
| 03:33.30 | coppice | you were given the answer to your question about modems |
| 03:33.43 | phunyguy | thats what i said. |
| 03:35.07 | *** join/#asterisk CryWolf (n=freedomb@mn01.freedombi.com) |
| 03:45.02 | eppigy | RAGEQUIT |
| 03:45.04 | R0b3Rt | k |
| 03:45.07 | R0b3Rt | ya |
| 03:45.19 | R0b3Rt | lulz |
| 03:45.34 | R0b3Rt | so anyway i need a demo dial plan for my voip provider |
| 03:45.34 | eppigy | THIS IS GOING IN HIS BLOG |
| 03:45.41 | R0b3Rt | coz that shit just aint registering |
| 03:45.47 | R0b3Rt | yes its going in his blog :p |
| 03:46.00 | R0b3Rt | YOU BET YOUR ASS ITS GOING IN HIS BLOG |
| 03:46.07 | R0b3Rt | and in his myspace too. |
| 03:47.05 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
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| 03:49.58 | *** part/#asterisk ArchGT2 (n=ArchGT@190.149.2.89) |
| 03:50.42 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 03:50.55 | brunner | can the digium cards handle more than one D channel per PRI? |
| 03:51.36 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-70-199.kgldga.dsl.dynamic.tds.net) |
| 03:54.19 | eppigy | brunner: I think you can have mor than 23 channels per d-channel but why have more than one d-channel per 23 b channels? |
| 03:54.34 | coppice | that isn't really a card issue. Its a software issue. I don't know if the software provides for multiple D channels |
| 03:54.40 | brunner | I know it's possible, but do the digium cards support it? |
| 03:54.42 | brunner | ah |
| 03:54.43 | brunner | okay |
| 03:54.45 | brunner | thanks |
| 03:55.03 | eppigy | well you can set which channels are bearer |
| 03:55.06 | eppigy | barer? |
| 03:55.10 | eppigy | and which are delta |
| 03:55.12 | eppigy | ion the config |
| 03:55.13 | eppigy | in |
| 03:56.01 | brunner | would the call setup be faster between two PRIs on the same switch than it would be between two PRIs on different carriers? |
| 03:57.04 | eppigy | I WOUDL ASSUME SO |
| 03:57.07 | eppigy | whoops |
| 03:58.01 | WindowsUser | there shouldn't be too too much lag on the pstn |
| 03:59.15 | brunner | WindowsUser: how many seconds would a typical call setup take from PRI to PRI? |
| 03:59.44 | brunner | I know when I'm dialing from VoIP providers over SIP it's about 6 seconds between the execution of the DIAL string and a ring on the other end |
| 03:59.56 | WindowsUser | wow |
| 04:00.16 | brunner | I know it seems instant, but have you ever actually measured it? |
| 04:00.22 | brunner | this is over several different providers |
| 04:01.30 | WindowsUser | hehe its like 1.5 seconds for the call to get to the next town, the ringing is out of sync a lot of the time :) |
| 04:01.58 | brunner | hmm |
| 04:02.14 | *** join/#asterisk ArchGT2 (n=ArchGT@190.149.2.89) |
| 04:02.24 | brunner | so 1.5 is a more reasonable expectation for PRI to PRI on the same switch? |
| 04:02.43 | WindowsUser | yea |
| 04:02.46 | brunner | how many call setups or teardowns can take place at one time before a D channel gets clogged? |
| 04:03.51 | brunner | I'd think it'd be a lot, but the existence of dual-D-channel configurations makes me think the bottleneck must be reachable if it's going nuts |
| 04:04.23 | coppice | where have you seen dual D channel setups? |
| 04:04.35 | jaytee | never seen one myself |
| 04:04.37 | brunner | I haven't seen them in person. I've just heard of htem |
| 04:04.50 | jaytee | never heard of one before now |
| 04:05.07 | russellb | never heard of it, either. |
| 04:05.23 | coppice | much more common is sharing 1 D channel across multiple links (NFAS), even for situations handling huge numbers of 30s calls |
| 04:06.15 | jaytee | I've seen that in a call center with AT&T megacomms |
| 04:06.24 | brunner | hmm |
| 04:06.48 | brunner | what if they're rapid 2s calls? |
| 04:08.03 | WindowsUser | what are you going to do in 2 seconds? "die bitch *click*" |
| 04:08.19 | brunner | no. the calls would never be answered. |
| 04:08.54 | coppice | a lot of systems uses calls of only 2 or 3 seconds, like voting systems |
| 04:10.21 | brunner | yup |
| 04:10.32 | brunner | call one number for option 1, call second number for option 2 |
| 04:10.37 | brunner | I just need to know what number is calling |
| 04:11.58 | brunner | do you think an RBOC would flip out, or cut off my service, if I consistently saturated a PRI with that behavior? or wouldn't they care if I was paying for a PRI? |
| 04:12.41 | WindowsUser | can I somehow use CHANNEL(audioreadformat) from an agi or will I have to Set something to it before I go in to the agi? |
| 04:12.48 | coppice | a lot of public switches can't cope with that, if its over several T1/E1s. their maximum call rates are surprisingly low |
| 04:13.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:14.49 | brunner | coppice: you mean it could crash a Class 5 switch?? |
| 04:15.03 | coppice | yep. seen it done |
| 04:15.10 | brunner | what if all the calls were originating from the same switch? |
| 04:15.34 | brunner | I've crashed a switch before, but it was a class 4.. I figured calls coming to and from the same switch might be easier on it |
| 04:15.49 | coppice | I expect it would naturally limit the call rate to something it can cope with |
| 04:15.59 | brunner | you'd think |
| 04:17.29 | brunner | so if I telco got pissed off, would they just disconnect the PRI, or what? |
| 04:18.26 | coppice | in the situations where I've seen this happen there were just lots of panicky phone calls trying to find out what happened |
| 04:19.37 | brunner | how many PRIs were involved? |
| 04:20.25 | coppice | I'm not sure. they were competitors systems. the first one I remember was actually when they asked the telco to do a flood test of their new setup :-) |
| 04:21.01 | brunner | heh |
| 04:21.14 | coppice | that was reputed to have taken down more than one switch, and blacked out a large part of east Kowloon for a while |
| 04:21.25 | brunner | wow. |
| 04:24.06 | WindowsUser | hey if the telephone switches go down, how do people call to complain? :) |
| 04:24.52 | coppice | what do you think cell phones are for? |
| 04:25.16 | coppice | on the plus side, that was one of the best laughs I've ever had |
| 04:29.46 | brunner | on a completely different subject, can T-Mobile @Home be used with Asterisk? |
| 04:31.55 | coppice | I guess "laugh" brought T-Mobile to mind |
| 04:32.19 | brunner | what's wrong with T-Mobile? |
| 04:32.28 | brunner | their general cell service, not the SIP stuff |
| 04:32.44 | coppice | dunno, but few people say anything nice about them |
| 04:33.01 | brunner | I've been a customer of theirs for 10 years, and I couldn't be more pleased |
| 04:33.32 | brunner | the coverage is superb in the southeast, the customer service is great, the price is the lowest |
| 04:33.48 | coppice | people complain a lot about them messing around with the software in cellphones |
| 04:33.53 | brunner | oh |
| 04:33.54 | brunner | yeah |
| 04:34.02 | brunner | that's why I buy unbranded phones |
| 04:34.16 | brunner | but I was doing that anyway, before they started fucking with the software a few years ago |
| 04:34.50 | coppice | I don't live in the US, but from the complaints I see it looks like any cellular operator make suck or be OK, depending on the state you live in |
| 04:35.02 | brunner | well, the G1 was the first branded phone I've bought in several years, but I promptly rooted it and put the Google Developer firmware on it |
| 04:35.27 | brunner | I really have had excellent luck with them |
| 04:35.56 | brunner | and what other carrier sells a phone that you can compile linux software on natively? |
| 04:36.35 | coppice | anyone supplying android phones, I guess :-\ |
| 04:36.53 | brunner | I don't think any other carriers are even selling android phones right no |
| 04:36.54 | brunner | now* |
| 04:37.01 | brunner | and t-mobile is about to come out with a second one |
| 04:37.03 | coppice | vodaphone are |
| 04:37.31 | coppice | vodafone |
| 04:37.51 | WindowsUser | how many gsm carriers does the us have? I think canada is down to just one (Rogers/Fido) |
| 04:38.20 | russellb | t-mobile, at&t |
| 04:38.58 | coppice | and how many non-gsm networks? |
| 04:39.53 | *** join/#asterisk iamamoron (n=iamamoro@210.238.181.188) |
| 04:39.56 | iamamoron | hi there |
| 04:40.21 | russellb | verizon, sprint, and some smaller ones, i think .. alltel ... |
| 04:40.24 | iamamoron | so you know where can I find price sip-based speaker? |
| 04:41.30 | iamamoron | my purpose is only for voice broadcasting |
| 04:41.43 | brunner | russellb: alltel is verizon now |
| 04:41.51 | russellb | ah. i can't keep up :-) |
| 04:42.00 | brunner | there are basically two of each, and some really small ones, and a bunch of resellers |
| 04:42.23 | brunner | I mean, two GSM (AT&T, T-Mobile), and two CDMA (Verizon/Sprint) |
| 04:42.45 | brunner | and then there's iDEN, which probably isn't even worth mentioning |
| 04:43.50 | coppice | Someone should write the story of iDEN and the fall of the house of Motorola :-) |
| 04:45.51 | brunner | there are technically exactly 542 wireless carriers in the US if you count all the entities that are incorporated for specific states |
| 04:46.19 | brunner | like verizon, who has a separate freakin OCN for every state |
| 04:46.47 | coppice | that makes a lot of business sense |
| 04:47.37 | brunner | I believe it, but I don't know exactly why |
| 04:48.58 | WindowsUser | they can hit up every state for tax credits? |
| 04:50.11 | coppice | look how many local operators have been bought and sold and bought and sold. fully integrating them cuts flexibility |
| 04:51.14 | brunner | yeah |
| 04:51.17 | brunner | makes sense |
| 04:55.00 | brunner | will most carriers let you send full CID over a PRI? |
| 04:55.10 | brunner | if you request it |
| 04:57.15 | WindowsUser | full cid? as in name as well as the number? |
| 04:57.23 | brunner | no |
| 04:57.28 | brunner | the 10 digit number |
| 04:57.32 | brunner | instead of the last 4 or whatever |
| 04:58.10 | WindowsUser | ? CID is usually 10 digits |
| 04:58.29 | brunner | oh |
| 04:58.38 | brunner | so it's standard when you have a PRI to send all ten? |
| 04:58.44 | brunner | on outbound calls? |
| 04:58.53 | WindowsUser | yea |
| 04:58.57 | brunner | cool |
| 05:03.46 | brunner | how did companies like Illuminet become LIDB owners? |
| 05:06.52 | *** join/#asterisk kombi (n=kombi@cpe-68-175-101-211.nyc.res.rr.com) |
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| 05:12.30 | kombi | problem: *box1 is behind port forwarding which cannot forward rtp 10000-20000. *box2 is freely accessible. Can outside phones link to *box1 over *box2 somehow? (over iax maybe?) |
| 05:16.22 | WindowsUser | either iax or symetric rtp |
| 05:18.01 | WindowsUser | nat=yes in sip.conf maybe? |
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| 05:25.00 | *** join/#asterisk iamamoron (n=iamamoro@210.238.181.188) |
| 05:25.03 | iamamoron | hi there |
| 05:25.04 | iamamoron | Grandstream BudgeTone BT-201 |
| 05:25.17 | iamamoron | is there any speaker out? |
| 05:25.42 | iamamoron | i am looking for an ip based speaker? |
| 05:25.45 | iamamoron | any ideas? |
| 05:30.37 | coppice | most of us can speak as well as type |
| 05:30.47 | coppice | ~modems |
| 05:30.47 | infobot | rumour has it, modems is something you can not use as an fxo interface under asterisk |
| 05:31.17 | coppice | ~modems is rumour has it, modems is something you can not use as an fxo interface under asterisk. See http://www.soft-switch.org/cards.html#modems |
| 05:31.18 | infobot | ...but modems is already something else... |
| 05:31.33 | coppice | ~forget modems |
| 05:32.22 | coppice | ~no, modems is rumour has it, modems is something you can not use as an fxo interface under asterisk. See http://www.soft-switch.org/cards.html#modems |
| 05:32.52 | coppice | ~modems |
| 05:32.53 | infobot | extra, extra, read all about it, modems is something you can not use as an fxo interface under asterisk |
| 05:33.02 | coppice | ~no, modems is rumour has it, modems is something you can not use as an fxo interface under asterisk. See http://www.soft-switch.org/cards.html#modems |
| 05:33.41 | coppice | ~modems is also See http://www.soft-switch.org/cards.html#modems |
| 05:33.42 | infobot | okay, coppice |
| 05:33.48 | coppice | ~modems |
| 05:33.49 | infobot | somebody said modems was something you can not use as an fxo interface under asterisk. See http://www.soft-switch.org/cards.html#modems |
| 05:34.03 | coppice | stupid bot |
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| 06:03.20 | kombi | what is code 102 "recovery on timer expiry" in zoiper? |
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| 06:20.13 | kombi | what is code 102 "recovery on timer expiry" in zoiper? |
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| 06:36.29 | kombi | <PROTECTED> |
| 06:37.48 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 06:48.50 | howie | so im trying to get asterisk up and running any good books on setting it up? |
| 06:50.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:54.10 | coppice | ~thebook |
| 06:54.11 | infobot | somebody said thebook was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 06:54.55 | howie | ok sweet i got that one |
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| 09:30.02 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.104.179) |
| 09:30.06 | DelphiWorld | hello |
| 09:30.17 | DelphiWorld | please any Flash/JAVA IAX2 Appelet? |
| 09:30.53 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 09:34.42 | DelphiWorld | java iax appelet please |
| 09:56.04 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 09:56.07 | Rob3Rt | WHAT UP |
| 09:56.16 | Rob3Rt | Ive got inbound dial plans sorted |
| 09:56.23 | Rob3Rt | but how to dial out through my voip provider ? |
| 09:56.36 | Rob3Rt | That is to say that I can answer if I call into my voip provider |
| 09:56.43 | Rob3Rt | what is it that Im looking for ? |
| 09:58.50 | tzafrir_laptop | DelphiWorld, searching for "java iax" gives me several hits. I haven't tried any of them |
| 10:01.08 | DelphiWorld | tzafrir_laptop: i found only javaiaxappelet in sourceforge, but no download only test page |
| 10:18.52 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 10:23.39 | Rob3Rt | Ive got inbound dial plans sorted |
| 10:23.44 | Rob3Rt | but how to dial out through my voip provider ? |
| 10:31.38 | DelphiWorld | Rob3Rt: do you have a trunk? |
| 10:32.30 | DelphiWorld | Rob3Rt: you must have a trunk and a Inbound/outbound route |
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| 10:50.33 | chutkin | 1 |
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| 12:08.43 | advorak | howdy! |
| 12:12.36 | advorak | is having trouble with sccp and cisco 7961 phones ... I've found a lot of forums with people having problems, but no solutions ------ |
| 12:13.00 | advorak | everything seems to work, except I can hear no sound nor transmit any sound from the 7961's .. |
| 12:13.21 | advorak | of course my iaxy works perfectly ... but not the sccp / 7961s .. |
| 12:14.10 | advorak | any pointers would be greatly appreciated :-) |
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| 14:13.46 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
| 14:19.02 | pulpster | hello - I have asterisk 1.6 and this problem: http://www.trixbox.org/forums/vendor-forums-non-certified/linksys-cisco/desperate-help-needed-cisco-7911-phone |
| 14:19.15 | pulpster | please help |
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| 14:34.27 | *** join/#asterisk Dustan (n=dustan@CPE-65-29-46-193.wi.res.rr.com) |
| 14:35.32 | Dustan | is this a support channel only or is it open to general questions? |
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| 14:39.10 | Dustan | I operate a small advertising business and am looking to possibly offer 800 numbers to local businesses, is this something asterisk can/is used for? |
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| 14:44.01 | mbrevda | Dustan: yes |
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| 14:46.12 | Dustan | can I get a 1-10 on difficulty to administrate a asterisk server? I've only ever run a lamp + email. |
| 14:49.18 | mbrevda | ~book |
| 14:49.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:49.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:49.42 | mbrevda | all you ransweres are there. like anything else - once you know how to do it, it really not a big deal |
| 14:50.03 | mbrevda | *your answeres |
| 14:50.04 | Dustan | <PROTECTED> |
| 14:51.25 | Dustan | appreciate your time mbrevda, have a good day. |
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| 15:39.10 | Rob3Rt | exten => _X.,1,Dial(SIP/from-3cx/2000) |
| 15:39.36 | Rob3Rt | this is a forwarded extension anyway |
| 15:39.52 | Rob3Rt | so when 1000 dials in this plan comes into effect, |
| 15:40.06 | Rob3Rt | but it then cant dial out to 2000, coz its already in use. |
| 15:40.13 | Rob3Rt | Is there a workaround ? |
| 15:42.46 | guax | thats nonsense, what are you trying to do? |
| 15:48.26 | Rob3Rt | when 1000 dials into this context, i want it to transfer the call to an internal extension, 2000 |
| 15:48.41 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 15:49.13 | Rob3Rt | why so narky? I think the concept is correct, isn't it ? |
| 15:49.28 | *** join/#asterisk PhunTelTek (i=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 15:50.01 | Rob3Rt | im saying, that 1000 calls in, i ask it to call 2000, but it says CHANUNAVAIL |
| 15:50.19 | Rob3Rt | im presuming becuase 1000 is already ringinig, that it cant call out. |
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| 15:51.50 | [TK]D-Fender | Rob3Rt: Presuming means you aren't even looking at SIP debug to see whats going on. |
| 15:52.19 | Rob3Rt | i am. |
| 15:52.25 | [TK]D-Fender | Rob3Rt: And 1000 generates the error... then is PLACING THE CALL |
| 15:52.27 | Rob3Rt | at verbose 3 |
| 15:52.40 | [TK]D-Fender | RobThe failure is * calling whatever it is callin |
| 15:52.48 | [TK]D-Fender | Rob3Rt: Verbose is NOT SIP DEBUG |
| 15:52.49 | Rob3Rt | hmm |
| 15:53.01 | [TK]D-Fender | Rob3Rt: "sip set debug" |
| 15:53.05 | Rob3Rt | im running sip debug as well |
| 15:53.08 | Rob3Rt | yep |
| 15:54.10 | Rob3Rt | like this for example, exten => _X.,1,Dial(SIP/from-exetel/${EXTEN},60) the call is placed but cant be routed, the error is no such host from-exetel <-- but thats a valid context. |
| 15:54.12 | Rob3Rt | wierd. |
| 15:54.21 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 15:54.30 | [TK]D-Fender | Rob3Rt: PASTEBIN is your friend... |
| 15:54.39 | [TK]D-Fender | ~pb |
| 15:54.40 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 15:54.42 | [TK]D-Fender | ^^^ |
| 15:55.00 | Rob3Rt | heh |
| 15:55.26 | Rob3Rt | win32 doesnt allow pasting of debug afaik |
| 15:56.05 | PhunTelTek | hi IRC people. Been a while. |
| 15:56.17 | [TK]D-Fender | Rob3Rt: uh...huh.... |
| 15:56.52 | Rob3Rt | ok |
| 15:56.58 | tzafrir_laptop | Rob3Rt, if you can't copy it directly., get it from the logs |
| 16:00.37 | Rob3Rt | Extensions, where the pain lies, looking for logs. http://pastebin.com/m5577198 |
| 16:00.51 | Rob3Rt | And Thanks in advance . |
| 16:01.27 | PhunTelTek | I have trixbox CE 2.8.0, is this the appropriate place to deal with issues on this pbx? |
| 16:01.33 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:946c:8331:bb15:b8cf) |
| 16:01.54 | Qwell | PhunTelTek: problems with trixbox are usually caused by their configs |
| 16:01.58 | Qwell | PhunTelTek: So, no. |
| 16:03.30 | PhunTelTek | I'm sure it's a config issue. it worked before i changed the AMPMGRPASS. when i changed it back it worked fine. |
| 16:04.15 | Rob3Rt | [TK]D-Fender, Relevent Log, http://pastebin.com/m5163ae11 - Extensions, where the pain lies, looking for logs. http://pastebin.com/m5577198 |
| 16:06.19 | *** join/#asterisk Alfio (n=Alfio@190.94.60.57) |
| 16:06.55 | Rob3Rt | so I've got it messed up a bit, but i don't think it's impossible |
| 16:08.11 | [TK]D-Fender | Rob3Rt: No such host means exactly that... |
| 16:08.43 | [TK]D-Fender | PhunTelTek: Do, definitely not supported here. |
| 16:09.07 | [TK]D-Fender | Rob3Rt: And I don't see you PB-ing your configs anywhere or the COMPLETE call attempt |
| 16:09.58 | Rob3Rt | thats not a host anyway, thats a context |
| 16:10.01 | Rob3Rt | right ? |
| 16:10.44 | PhunTelTek | thanx anyway. I'll hunt for trixbox channels |
| 16:12.53 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:13.50 | [TK]D-Fender | Rob3Rt: No, its a peer entry of your sip.conf <- |
| 16:15.03 | PhunTelTek | they have 28 people in the trixbox channel compared to 242 here! :-) |
| 16:16.19 | Rob3Rt | HMM |
| 16:18.57 | Rob3Rt | progress |
| 16:19.06 | Rob3Rt | failed to autenticate to 100 |
| 16:28.32 | Rob3Rt | no progreess |
| 16:28.44 | Rob3Rt | got a basic layout i could follow ? |
| 16:29.49 | [TK]D-Fender | Rob3Rt: What part of "show us the dead body" aren't you getting? |
| 16:30.06 | [TK]D-Fender | Rob3Rt: This isn't "Guess how my config should look and do my job for me". |
| 16:30.39 | *** join/#asterisk jitter (i=jitter@62.120.225.128) |
| 16:30.45 | Rob3Rt | Alright |
| 16:30.52 | Rob3Rt | Ill paste the entire log. |
| 16:33.41 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:33.51 | *** join/#asterisk Alfio (n=Amunoz@190.94.60.57) |
| 16:34.03 | Rob3Rt | Its not logging the call mate |
| 16:34.09 | Rob3Rt | but the debug shows it |
| 16:34.25 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:35.12 | *** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 16:36.26 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:38.14 | [TK]D-Fender | Rob3Rt: I'm not seeing the entire call, and I'm not seeing your configs. |
| 16:38.52 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:47.41 | tzafrir_laptop | If I call (IAX) from from server A to server B and then route the call back to server A |
| 16:48.03 | tzafrir_laptop | (from and to two phones connected to server A) |
| 16:48.20 | tzafrir_laptop | Is Asterisk smart enough to remove the excess loop? |
| 16:48.47 | *** join/#asterisk sjobeck (n=Adium@137.118.193.9) |
| 16:48.52 | tzafrir_laptop | I recall that it was, and that there was an option to disable that. But can't find it |
| 16:49.25 | *** part/#asterisk sjobeck (n=Adium@137.118.193.9) |
| 16:52.39 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:53.49 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 17:02.36 | [TK]D-Fender | tzafrir_laptop: "transfer=yes" IIRC |
| 17:04.25 | jaytee | yeah, that should work. I use an IAX trunk to route calls to an IVR running on another * box and then back to whatever number is associated with the caller's menu choice and it transfers the call back to my primary * server. |
| 17:12.07 | Rob3Rt | told ya its not loggin the calls |
| 17:15.10 | [TK]D-Fender | Rob3Rt: You had CLI output before and I've asked for your configs about 3 times now. |
| 17:17.58 | *** join/#asterisk Carlos_PHX (n=carlos@65.126.36.98) |
| 17:18.09 | *** part/#asterisk Carlos_PHX (n=carlos@65.126.36.98) |
| 17:22.52 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:24.58 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
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| 17:57.14 | *** join/#asterisk ToTo (n=ToTo@host130-150-dynamic.17-79-r.retail.telecomitalia.it) |
| 17:58.15 | *** join/#asterisk ToTo (n=ToTo@host130-150-dynamic.17-79-r.retail.telecomitalia.it) |
| 18:04.06 | KavanS | is there anyway to use the Read command and the Festival command on the same line? |
| 18:04.51 | KavanS | normally you can Read(var,soundfilename,digits) |
| 18:04.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:05.06 | KavanS | would it make sense to just use Read right after the Festival command...on separate lines? |
| 18:09.21 | *** join/#asterisk DarkLogik (n=darklogi@darklogik.com) |
| 18:14.38 | [TK]D-Fender | KavanS: Huh? |
| 18:14.50 | [TK]D-Fender | KavanS: What do you mean "separate lines"? |
| 18:15.05 | KavanS | [TK]D-Fender, I'm prompting the user to enter information with Festival...and so I'm used to using the "Read" command with a pre-recorded GSM |
| 18:15.13 | KavanS | what would be the preferred method to accomplish the same with Festival |
| 18:15.30 | [TK]D-Fender | kavauSE FESTIVE, READ WITHOUT PLAYING ANOTHER WASTED PROMPT |
| 18:15.38 | [TK]D-Fender | darn caps |
| 18:16.02 | KavanS | hrm |
| 18:16.11 | KavanS | I'm not sure I understand |
| 18:17.47 | KavanS | hrm, looks like I can just use text2wave command and just create the sound files manually |
| 18:19.14 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 18:21.50 | *** join/#asterisk tamiel (n=tamiel@ip-224.net-81-220-88.toulouse.rev.numericable.fr) |
| 18:21.52 | [TK]D-Fender | KavanS: Festival plays your TTS prompt. You tell Read NOT to prompt. Whats so difficult? |
| 18:24.29 | Corydon76-dig | [TK]D-Fender: I think the difficulty is that Festival doesn't allow DTMF to be entered during playback |
| 18:25.12 | [TK]D-Fender | Corydon76-dig: Nobody say that was an issue :) |
| 18:25.42 | [TK]D-Fender | Corydon76-dig: All I her is "get festival to play prompt, then read, but I don't want to pass another prompt" |
| 18:25.45 | [TK]D-Fender | hear* |
| 18:26.15 | Corydon76-dig | Some people aren't good at English |
| 18:26.52 | [TK]D-Fender | Corydon76-dig: And until I have good reason to think otherwise I try not to add stuff to their demands that they didn't say. Works out for the best :) |
| 18:27.10 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 18:27.22 | dwery | hello. anyone is using espeak/mbrola with * 1.6 ? |
| 18:28.51 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:32.02 | *** join/#asterisk SkramX (n=SkramX@173-134-0-182.pools.spcsdns.net) |
| 18:33.00 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
| 18:33.26 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 18:34.16 | Zuchmir2 | hi, is there any way to connect * with MagicJack? (I get a "SIP/2.0 400 Bad Request" response when i try an outgoing call using instructions from: http://revolution.hackthisbox.com/joo/component/content/article/1-latest-news/39337 ) |
| 18:35.04 | [TK]D-Fender | Zuchmir2: MJ has recently blocked people trying to use * with their service as I hear. |
| 18:36.08 | Zuchmir2 | my account worked right out of the box with X-Lite, but w/* it's giving trouble, i even changed the User-Agent to MJ |
| 18:43.37 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:946c:8331:bb15:b8cf) |
| 18:48.40 | *** join/#asterisk psilikon (n=psilikon@118-243.187-72.tampabay.res.rr.com) |
| 18:50.29 | Defraz | Wonder how they know the difference between the two if the user-agent were the same? |
| 18:51.14 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:55.41 | Zuchmir2 | yeah, that was what i was wondering... :-( |
| 18:56.04 | Zuchmir2 | it allows "registration", and all incoming calls route to * |
| 18:57.25 | brunner | does anyone here know any good Java programmers? |
| 18:57.36 | *** join/#asterisk jjnw-wibble (n=jjnw-wib@82-69-3-154.dsl.in-addr.zen.co.uk) |
| 18:57.52 | brunner | I'd like to hire pick someone's brain for a quick second and hire them, if they're available, for a quick project |
| 18:58.22 | brunner | I have most of the code written, but I don't know enough about Java to know if this is the best way to do it |
| 18:59.04 | gr0mit | http://www.portech.com.tw/p3-product1_1.asp?Pid=14 <--- aanyone got any experience of one of these? |
| 18:59.06 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 19:04.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:11.01 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:20.03 | dwery | gr0mit: bought a 372 some times ago, tested for a while, seemed to work |
| 19:23.56 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 19:27.10 | *** join/#asterisk artemmakhutov (n=artemmak@ip-95-222-243-116.unitymediagroup.de) |
| 19:30.42 | blackest_mamba | Holy crap - if anyone needs a wireless sip phone, DON'T but the cicsco/linksys WIP310. |
| 19:30.49 | blackest_mamba | Major POS. |
| 19:32.29 | artemmakhutov | Hello, I am trying to extend chan_mobile with busy detection, but I can not make the channel to go into busy ... I am trying to call "ast_queue_control(pvt->owner, AST_CONTROL_BUSY);" but it seems to get ignored and I am always getting CHANUNAVAIL instead of BUSY. Can somebody help me? |
| 19:35.33 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 19:35.43 | apeiron | blackest_mamba, wow that is one expensive phone. o_O |
| 19:36.09 | apeiron | notes that the *warranty* is more than he paid for his last normal cordless phone |
| 19:36.27 | apeiron | (normal == legacy) |
| 19:45.19 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:45.55 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 19:53.29 | blackest_mamba | Yes, and it's a major hunk of crap. |
| 19:53.59 | blackest_mamba | I have three. Cisco has acknowledged that it is a piece of crap and is no longer selling them. |
| 19:54.13 | blackest_mamba | They say that they are planning a replacement, but who knows when that will be. |
| 19:54.32 | blackest_mamba | Now I have to fight with Cisco to get my money back. :( |
| 19:57.54 | KavanS | hrm...I'm running a macro and it's exiting non-zero |
| 19:57.59 | KavanS | is there anything I can do about this? |
| 20:00.56 | dwery | artemmakhutov: better to ask in #asterisk-dev |
| 20:02.33 | [TK]D-Fender | KavanS: Perhaps you should WAKE UP and lrean to pastebin stuff. |
| 20:02.37 | [TK]D-Fender | learn* |
| 20:05.49 | *** join/#asterisk BadHAL (n=nn@173-112-189-186.pools.spcsdns.net) |
| 20:06.29 | gr0mit | dwery, still in use? |
| 20:06.51 | dwery | gr0mit: no, it was just a test |
| 20:07.17 | gr0mit | ok, what happened to the unit? |
| 20:07.24 | WindowsUser | blackest_mamba: is there any wifi sip phones that dont suck? |
| 20:08.51 | jaytee | Polycom Spectralink models tend to be a bit pricey but most of the feedback I've heard about them is positive |
| 20:09.03 | dwery | gr0mit: should be somewhere in my room ;) |
| 20:09.29 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 20:10.24 | gr0mit | lot of money for a door stop |
| 20:11.22 | dwery | gr0mit: It wasn't so pricey, I neede it for a demo |
| 20:12.07 | dwery | gr0mit: there's a chance it mught be put in operation the next week |
| 20:12.32 | gr0mit | cool |
| 20:12.36 | dwery | gr0mit: I will know tomorrow, drop me anote if you want |
| 20:13.18 | gr0mit | which coutry are u iun? |
| 20:13.24 | dwery | .it |
| 20:13.26 | gr0mit | u in i mean! |
| 20:13.27 | dwery | ;) |
| 20:13.31 | gr0mit | aah ok |
| 20:14.05 | gr0mit | here i is £275 |
| 20:14.13 | gr0mit | seems expensve |
| 20:14.47 | gr0mit | like 320 euro |
| 20:14.48 | dwery | gr0mit: I bought it on ebay. I think it was shipped from china or taiwan |
| 20:14.57 | gr0mit | + VAT |
| 20:15.15 | dwery | it was the cheapest sip/gsm device avaialble at that tim. I think it was two years ago |
| 20:17.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:17.40 | dwery | the web interface wasn't particularly friendly, but it worked |
| 20:18.05 | gr0mit | does it count mins/month? |
| 20:18.18 | gr0mit | so it stops after using your free minutes? |
| 20:22.34 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:23.45 | KavanS | WindowsUser, iphone with siphon is working ok for me |
| 20:23.51 | KavanS | WindowsUser, I'd not give it to users though... |
| 20:32.33 | *** join/#asterisk Aiatek (n=Amunoz@190.94.61.34) |
| 20:34.04 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:34.39 | KavanS | I need to evaluate if a variable is null....can someone give me an example gotoif statement that would evaluate a null value? |
| 20:34.46 | Zuchmir2 | found the error in my sip.conf (had wrong credentials!), magicJack now works! will have to see if they disconnect me due to * |
| 20:34.49 | KavanS | i.e. if it is null, then goto line x |
| 20:35.00 | KavanS | any other values goto line y |
| 20:35.21 | KavanS | I'm reading through voip-info's cmd gotoif, but I can't seem to find the correct syntax |
| 20:37.05 | wdoekes | null? wouldn't that be empty? |
| 20:37.14 | wdoekes | I believe it is if the source is odbc |
| 20:37.47 | [TK]D-Fender | Someone doesn't understand * expressions it seems. |
| 20:37.52 | KavanS | hrm, let me look if empty is the right keyword |
| 20:38.05 | wdoekes | which would then yield GotoIf($["${myvar}"=""]?nullsomewhere:notnullsomewhereelse) |
| 20:38.12 | [TK]D-Fender | Not that there isn't also a FUNCTION for this either... |
| 20:39.47 | KavanS | ok....reading on expression syntax on voip-info.org |
| 20:40.13 | Zuchmir2 | [TK]D-Fender: thanks for your pointers |
| 20:45.32 | KavanS | [TK]D-Fender, thanks for the heads up...I'll read some more |
| 20:45.46 | KavanS | wdoekes, thanks as well...def, good to be put in the right direction |
| 20:47.59 | [TK]D-Fender | kavaMore like handed you the answer |
| 20:48.31 | KavanS | [TK]D-Fender, roger that |
| 20:48.51 | KavanS | [TK]D-Fender, are you from canada? |
| 20:50.07 | KavanS | ok back to ssh for me...thx |
| 20:52.56 | Xetrov` | [TK]D-Fender: dude youre pretty condescending |
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| 21:04.06 | [TK]D-Fender | Xetrov`: Oh? I just stated that he was handed the EXACT answer |
| 21:05.02 | [TK]D-Fender | Xetrov`: Now had I said "a 3rd grader should know this" then that would be codecending. So if you're going to slam me pick the right point. |
| 21:10.57 | rob0 | strives to be conascending |
| 21:11.27 | Alfio | :) |
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| 21:14.13 | *** mode/#asterisk [+o mog] by ChanServ |
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| 21:26.17 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 21:30.00 | Linuturk | I'm looking to replace a Xilinx Corporation Wildcard TE405P/TE410P with a newer card with hardware echo cancel |
| 21:30.06 | Linuturk | what would you recommend for a t1 PRI |
| 21:30.06 | *** join/#asterisk PhunTelTek (i=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 21:30.09 | Linuturk | ? |
| 21:30.27 | Linuturk | I guess the better question is, what's the difference between all the TE cards? |
| 21:30.40 | Linuturk | http://www.digium.com/en/products/digital/ << there |
| 21:30.50 | Linuturk | 4 ports needed |
| 21:31.08 | Linuturk | TE412P and TE407P seem like my two options |
| 21:32.01 | Linuturk | looks like the only difference is the voltage of the PCI slot . . . |
| 21:33.02 | *** join/#asterisk voip_troll (n=voip_tro@96.51.229.227) |
| 21:34.14 | Alfio | Linuturkyou answered yourself |
| 21:35.47 | rob0 | Talk about conascending! |
| 21:36.13 | rob0 | Well, that is, it might be, if I knew what it meant. :) |
| 21:39.14 | [TK]D-Fender | Linuturk: Sangoma A104d |
| 21:45.08 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:46.41 | Linuturk | how do I tell what voltage my pci slot takes? |
| 21:46.51 | Linuturk | or, should I pick the lower number? lol |
| 21:47.12 | Linuturk | [TK]D-Fender: I like digium cards :) |
| 21:47.29 | Linuturk | they haven't failed me yet, except for this one that got hit by a power surger |
| 21:47.31 | Linuturk | surge* |
| 21:50.08 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 21:55.34 | [TK]D-Fender | Linuturk: You read your mobo's MANUAL |
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| 21:57.39 | *** join/#asterisk advorak (n=advorak@c-69-181-129-41.hsd1.ca.comcast.net) |
| 21:58.03 | advorak | howdy, y'all! |
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| 22:01.11 | advorak | is having trouble with sccp and cisco 7961 phones ... I've found a lot of forums with people having problems, but no solutions ------ |
| 22:01.24 | advorak | rything seems to work, except I can hear no sound nor transmit any sound from the 7961's .. |
| 22:01.24 | advorak | <PROTECTED> |
| 22:01.57 | russellb | advorak: what channel driver are you using? |
| 22:02.00 | advorak | (everything is behind the same router on the same network .. |
| 22:02.06 | advorak | russellb: chan_sccp .. |
| 22:02.20 | russellb | chan_skinny has come a long way, I would suggest trying that. |
| 22:02.28 | russellb | at least in 1.4, and if you have trouble there, try chan_skinny in 1.6 |
| 22:02.35 | artemmakhutov | @advorak try using the newest chan_sccp-b from svn, they have fixed a bug |
| 22:02.35 | russellb | people actively use it/support it/improve it now a days |
| 22:02.35 | advorak | ok, I'll try that :-) |
| 22:02.44 | advorak | oh I meant I'm using chan_sccp-b |
| 22:02.48 | artemmakhutov | and make sure that you use protocolversion=17 |
| 22:02.57 | *** join/#asterisk PhunTelTek (n=PhunTelT@cpe-76-188-233-188.neo.res.rr.com) |
| 22:03.14 | advorak | ok, I shall try those suggestions, thanks :-) |
| 22:03.46 | artemmakhutov | 1st try out protocolversion=17 if this does not help try using the svn version |
| 22:03.58 | advorak | ok |
| 22:04.29 | artemmakhutov | and do not use chan_sccp-b with asterisk 1.6.1 ... use asterisk 1.4 or 1.6.0 or the 1.6.2 beta ... |
| 22:04.59 | artemmakhutov | the phones will "hang" after some time with 1.6.1 |
| 22:10.18 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:23.05 | voip_troll | Is it possible to list someone a dial-by-name directory without them having VM assigned? |
| 22:24.11 | [TK]D-Fender | voip_troll: They need an entry in voicemail.conf but disassociate the term "assigned" |
| 22:24.49 | [TK]D-Fender | voip_troll: nothing is "assigned". VM boxes don't have any explicit relationship to any specific device. |
| 22:25.12 | voip_troll | yea, my bad, still stuck in Broadworks terminology... |
| 22:32.01 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 22:32.47 | ariel_ | hello folks |
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| 22:57.23 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 22:57.26 | *** mode/#asterisk [-b *!comfrey@*] by [TK]D-Fender |
| 22:57.42 | *** mode/#asterisk [-b *!*@66.166.226.6] by [TK]D-Fender |
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| 23:24.23 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 23:30.15 | kmem | hello |
| 23:30.32 | PhunTelTek | hi |
| 23:30.47 | kmem | how goes it phun |
| 23:31.16 | PhunTelTek | it's going... |
| 23:31.31 | kmem | i hear ya |
| 23:31.51 | kmem | is this a newb friendly channel? |
| 23:32.36 | PhunTelTek | dunno, I'ver been here a few hours. |
| 23:33.42 | PhunTelTek | what's your issue? |
| 23:34.02 | kmem | ok, well I'm having some trouble getting my ata to register with a new pbxinaflash installation |
| 23:35.50 | kmem | I've got a strange unit a dlink VTA |
| 23:36.37 | kmem | is there a log I can look at to see whats going on? |
| 23:36.58 | coppice | when did a D-Link VTA become strange? :-\ |
| 23:38.21 | kmem | I am using 1.08NA firmware on it and the web administration seems very finicky. |
| 23:38.47 | coppice | crappy firmware == industry standard |
| 23:41.14 | kmem | I'm prolly just doing something dumb. |
| 23:42.31 | PhunTelTek | i'm not familiar with pbxinaflash |
| 23:43.29 | kmem | coppice do you use a VTA |
| 23:44.57 | [TK]D-Fender | D-Link = amongst the worst of the larger brands |
| 23:45.25 | [TK]D-Fender | kmem: Enable SIP DEBUG at * CLI and look at whats actually happening |
| 23:45.30 | kmem | I wish I had bought a PAP, it looks much better from screenshots |
| 23:46.38 | [TK]D-Fender | kmem: Hindsifght is 20/20 |
| 23:46.43 | [TK]D-Fender | kmem: Hindsight is 20/20 |
| 23:47.07 | coppice | its best to choose the most widely used option if you want help, and currently that's the PAP2T |
| 23:47.50 | *** join/#asterisk Rob3Rt (n=admin@87.37.96.58.static.exetel.com.au) |
| 23:48.13 | kmem | this thing was dirt cheap, however after unlocking it I an very dissapointed. It worked ok with gizmo + grandcentral for incoming |
| 23:48.14 | coppice | although many of its supposed heavy users seem convinced it has features which it doesn't :-\ |
| 23:48.35 | kmem | the thing cant even pass aplha characters in the username |
| 23:48.44 | kmem | well anything except a-d |
| 23:49.10 | [TK]D-Fender | kmem: the fact you had to unlock it says something... |
| 23:49.19 | kmem | do you mean crossflashing it with a different firmware? |
| 23:49.33 | [TK]D-Fender | kemas in "dead-end piece of shit that will cause plenty of grief down the road" |
| 23:50.03 | kmem | well not enough, I cant get the damn thing to even register with asterisk... I'd like to blame it on the ATA but I'm convinced its cockpit error |
| 23:50.27 | [TK]D-Fender | kmem: SIP DEBUG <----- |
| 23:50.43 | kmem | i totally agree. |
| 23:51.10 | Alfio | if you want quality buy grandstream <------------------- hehehehehehehehhehehhehe |
| 23:51.41 | kmem | I gotta figure out how to get into the asterisk CLI :( told ya I was a noob |
| 23:51.52 | kmem | -vr |
| 23:51.55 | kmem | i think |
| 23:52.36 | kmem | thanks fender I'm looking at it now |
| 23:53.06 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 23:57.35 | kmem | it says "really destroying SIP dialog" at the end |
| 23:58.20 | kmem | but it says SIP/2.0 200 OK |
| 23:58.32 | [TK]D-Fender | kmem: PASTEBIN is your friend <- |
| 23:58.34 | [TK]D-Fender | ~pb |
| 23:58.35 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 23:58.54 | kmem | thanx |