00:00.07 | *** join/#asterisk wepy (~wepy@pcp09870567pcs.ewndsr01.nj.comcast.net) |
00:00.10 | wepy | hello |
00:00.26 | routerheads_atho | Docelm0: Let me know what you find out. |
00:00.39 | routerheads_atho | re: pap2 |
00:00.53 | wepy | i don't understand what asterisk is for.. i read the page though |
00:01.06 | JRB | Hi - is anyone using SMS in the UK please? |
00:01.12 | JRB | I am having a nightmare getting it to work! |
00:01.15 | JamesDotCom | buzzyd: see the line that "db_class" = "DB_ser"? |
00:01.23 | JRB | there is very little docs that I can find on it |
00:01.24 | JamesDotCom | change that to "db_class" => "DB_ser" |
00:01.28 | Docelm0 | I am trying to get the firmware on both the router version and also the PAP2.. If I get the firmware I will make it freely available on my website for anyone who wants it |
00:01.29 | JamesDotCom | add the > |
00:01.46 | wepy | what can i do with asterisk :) |
00:02.09 | amer | asterisk can do your laundry |
00:02.21 | drumkilla | really?! |
00:02.27 | wepy | can i get free phone calls using my internet connection? |
00:02.28 | buzzyd | thanks James |
00:02.31 | routerheads_atho | I need to get a locked one and start playing, I will take JerJer's approach |
00:02.32 | wepy | using asterisk? |
00:02.39 | JamesDotCom | np |
00:02.40 | Docelm0 | Im sure twisted could code a module for it |
00:03.03 | Ehsan | wepy : what do u need to do with asterix ? |
00:03.03 | Docelm0 | :) |
00:03.28 | wepy | Ehsan: i don't need to do anything, i'm just curious what this is... what could i use it for |
00:03.46 | Ehsan | multifunction pbx |
00:03.55 | wepy | what does a pbx do? |
00:04.01 | wepy | (i thought it meant phone box?) |
00:04.16 | Ehsan | wepy : that's a tough question :) |
00:04.21 | wepy | hah |
00:04.26 | amer | google |
00:04.32 | PatrickDK | heh, if you don't know pbx, than you should probably give up now |
00:04.39 | amer | google "what is a pbx" |
00:04.44 | PatrickDK | it will be one hell of a learning curve |
00:04.47 | wepy | i can learn what a pbx is.. |
00:04.54 | wepy | but i'm just wondering what this is for.. |
00:05.01 | davetroy | private branch exchange |
00:05.02 | drumkilla | i bet wikipedia has an article on a PBX |
00:05.02 | robl^ | wepy: PBX it not just a social disease |
00:05.03 | PatrickDK | routing phone calls |
00:05.07 | amer | Asterisk is an IP PBX |
00:05.12 | davetroy | it's a phone system. |
00:05.14 | wepy | is it just like IP address -> phone ? |
00:05.14 | bkw_ | its more than that |
00:05.35 | davetroy | yeah, it's a way of life, a seismograph, a blender, etc... |
00:05.38 | robl^ | wepy: PBX - like a phone system in an office.. transfer calls, dial 9 for an outside line.. etc |
00:05.42 | amer | Asterisk is an answering machine |
00:05.48 | wepy | oh |
00:06.04 | routerheads_atho | I use my asterisk as a toaster ;) |
00:06.06 | PatrickDK | amer, one damn nice answering machine :) |
00:06.16 | wepy | could you use asterisk over the internet.. say, if i had it at home, and my relatives in europe had it installed..? |
00:06.19 | amer | yoo bro |
00:06.21 | wepy | then we could call for free? |
00:06.22 | *** join/#asterisk malaiwah (malaiwah@dsl6-200.express.oricom.ca) |
00:06.24 | Ehsan | anyone with Music On hold and meetme without Zaptel hardware |
00:06.42 | PatrickDK | ehsan, use ztdummy |
00:06.47 | PatrickDK | won't be as good, but should work |
00:07.22 | Ehsan | PatrickDK : used it |
00:07.26 | davetroy | any opinions as to whether 2.6 ztdummy works better than 2.4 usb ztdummy? |
00:07.27 | Ehsan | but get this error |
00:07.28 | file | meetme can't be used without a timing device |
00:07.31 | file | Music on hold can |
00:07.35 | robl^ | I use Asterisk to get rid of unwanted phone calls. match the caller ID on my "ickey list!" and Allison says "The number you have dialed, XXX-XXX-XXXX has been changed. The new number is..." and then 32 random digits. :) |
00:08.08 | wepy | hm |
00:08.16 | PatrickDK | ehsan, what error? |
00:08.24 | file | wepy: yes, you can do that |
00:08.31 | malaiwah | i have trouble with my xlite client on windows xp.. my asterisk box works right with my fxo and fxs lines and i can hear on my xlite what i say in the fxs phone.. but listening on the fxs, the voice from xlite is choppy. |
00:08.36 | file | wepy: or you can have one asterisk server, and have both phones on it |
00:08.44 | wepy | file: so asterisk is like a router for phones that use ethernet cables? :) |
00:08.48 | PatrickDK | robl, I just let the line ring, then hangup |
00:08.53 | file | it's a PBX. |
00:09.02 | wepy | or do the phones still use regular lines? |
00:09.12 | file | VoIP phones use ethernet |
00:09.17 | Ehsan | PatrickDK : just a min |
00:09.25 | PatrickDK | I don't have a min :) |
00:09.36 | wepy | does asterisk use VoIP phones, or regular phones? |
00:09.38 | file | if you Google, look at voip-info, and look at mailing list posts you'll get lots of info |
00:09.42 | *** join/#asterisk NormAst (NormAst@Toronto-HSE-ppp3676715.sympatico.ca) |
00:09.45 | wepy | ok |
00:09.56 | file | http://www.voip-info.org/ |
00:10.26 | wepy | file: so really, a regular guy at home has no use for asterisk, it's more like a router for phones in a business? |
00:10.27 | Godsey | I'm looking at Primus's offerings |
00:10.39 | file | wepy: well, you can have voicemail... and low cost VoIP service |
00:10.47 | robl^ | wepy: if you don't understand what a VoIP PBX is, that means you need t buy 4 systems from me |
00:10.58 | wepy | hehe |
00:11.16 | wepy | file: wow.. i kinda want voicemail :) |
00:11.17 | wepy | haha |
00:11.28 | wepy | file: what could i use voip for at home? |
00:11.32 | *** join/#asterisk JerJer (~mine@DSL-49-70.che.centurytel.net) |
00:11.38 | file | asterisk can e-mail you the voicemail.. and also have web access, plus phone access |
00:11.41 | file | wepy: low cost long distance? |
00:11.43 | wepy | :o |
00:11.50 | Godsey | I wish I could use Lingo w/ asterisk :) |
00:12.13 | wepy | file: um.. how? |
00:12.22 | file | Broadvoice gets you unlimited long distance for $20/mth in US and Canada... |
00:12.33 | wepy | file: the person i call must also use voip? |
00:12.35 | file | or if you don't use that much, you can use prepaid services like NuFone or Voicepulse Connect |
00:12.36 | robl^ | wepy: and the ultra cool geek factor of having a $500 VoIP phone on your desk with an LCD screen, lots of buttons, and flashing lights. |
00:12.38 | file | no - this is regular phone |
00:12.47 | Godsey | I have broadvoice right now, and it seems unreliable |
00:12.49 | wepy | hehe |
00:12.55 | file | you pay a provider and they provide you access to the regular pstn network |
00:13.07 | wepy | file: pstn ? |
00:13.17 | Godsey | I get reports about unreachable and busy from broadvoice when clearly the line it's calling isn't in use |
00:13.18 | wepy | i think i understand this now |
00:13.22 | file | 'da plain old telephone system |
00:13.23 | wepy | neat.. |
00:13.41 | robl^ | pstn = old fashioned phone network. AT&T, Bell, etc |
00:14.08 | amer | is there a thing like directed pickup in Asterisk |
00:14.17 | file | amer: what do you mean? |
00:14.24 | robl^ | amer: yes.. |
00:14.27 | miguellinux | ariel_: help!! |
00:14.33 | file | pickup groups? |
00:14.39 | amer | like can I pickup any ringing extension |
00:14.39 | miguellinux | yap |
00:14.52 | amer | yes yes |
00:14.54 | file | pickup groups, allows you to pick up other phones that are ringing in your group |
00:15.02 | amer | wo hoo |
00:15.05 | robl^ | amer: the features are there, you just have to configure it |
00:15.06 | amer | is that an ad on |
00:15.11 | wepy | so if i wanted to use asterisk for voicemail going to my email.. how would this physically be setup? |
00:15.12 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
00:15.38 | wepy | do i need 2 modems and a NIC ? |
00:15.41 | file | wepy: you'd have a PCI FXO card that goes to your current phone line, and have asterisk pick up after a number of rings... to voicemail |
00:15.54 | wepy | oh |
00:16.00 | file | wepy: the box would have to have the PCI FXO card, available from Digium, and yes connectivity to the net |
00:16.02 | Ehsan | wepy : u need asterisk installed and a fxo card |
00:16.06 | Docelm0 | Get this.. Some dude in the phillipines just told me I have to contact vontage cause they coded the firmware for the devices.. thats complete BULLSHIT! |
00:16.18 | file | Docelm0: haha |
00:16.26 | Docelm0 | Anywho.. Anyone in here know dialplans? I have a REAL small question to ask |
00:16.31 | *** join/#asterisk Moc (~mochouina@modemcable021.49-80-70.mc.videotron.ca) |
00:16.36 | file | Docelm0: ask it |
00:16.39 | Docelm0 | But I am calling vontage.. |
00:16.40 | wepy | the FXO card has 2 jacks, 1 for phone and 1 for the line? |
00:16.44 | robl^ | again.. I will say.. Asterisk is a phone system construction set.. you get all the pieces you need. You just have to put it together the way you want. You can build any phonesystem you can imagine. |
00:16.44 | script | do you guys have voip service and using asterisk? |
00:16.46 | *** join/#asterisk welby (~welby@solas.plus.com) |
00:16.53 | file | wepy: the phone is just a pass through... won't connect into asterisk, but the line one will |
00:17.02 | *** join/#asterisk nikko_ (~nikko@69.85.201.170) |
00:17.05 | davetroy | yeah, like the Lego Mindstorms Office Phone System Kit. |
00:17.05 | Moc | script, what do you mean ? |
00:17.09 | *** join/#asterisk freestyle_networ (~Paul@S0106000f6630d841.ed.shawcable.net) |
00:17.25 | Docelm0 | When the dialplan is executed and it connects.. Then is terminated under normal conditions ie.. the other party hangs up.. Can I continue to execute the dialplan and have it redial that same connection again? |
00:17.34 | freestyle_networ | annyone work with TXFAX ? |
00:17.41 | script | do you vontage or some other provider with using asterisk software? |
00:17.44 | wepy | file: so is it like phone -> FXO phone port :: FXO Line port -> phone line ? |
00:17.57 | wepy | and asterisk in the middle.. listening and managing the line..? |
00:18.00 | nikko_ | can someone possibly answer a quick asterisk upgrade question? |
00:18.06 | wepy | kinda like a big answering machine? |
00:18.09 | robl^ | wepy; FXS = phone FXO=line |
00:18.09 | file | wepy: the phone port is just there so you can call out when your computer is off |
00:18.17 | file | it won't interact with asterisk |
00:18.24 | Docelm0 | No.. this is completely unrelated.. |
00:18.27 | wepy | oh |
00:18.31 | file | asterisk, however, can use the phone line to place/receive calls and route them |
00:18.58 | wepy | file: routing phone calls? that's the voip part right? |
00:19.02 | script | so its like a router for voip? |
00:19.13 | file | it's a PBX. |
00:19.19 | file | go look at the definition of a PBX |
00:19.35 | script | because I'm configure it now with my current voip service |
00:19.51 | davetroy | more of an endpoint device than a core element, but can still be used to provide specific features in the core. |
00:20.09 | Moc | script, Nufone use * |
00:20.12 | file | script: which provider? |
00:20.24 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f8-CM00080d290642.cpe.net.cable.rogers.com) |
00:20.27 | wepy | so VoIP service would costs like $20/mo for unlimited calls to anywhere in the world? |
00:20.28 | Moc | Nufone is the most stable VOIP provider I found so far |
00:20.37 | _-Jon-_ | Any idea how I can fix this error: chan_iax2.c:5701 socket_read: Rejected connect attempt from 66.234.228.170, request '802xxxxxxx@from-pstn' does not exist |
00:20.37 | script | packet8 |
00:20.40 | file | wepy: no - US and Canada |
00:20.52 | Docelm0 | I have packet8.. I wanna hack that ATA also.. :) |
00:20.53 | wepy | file: are there any international ones? |
00:20.54 | file | script: you have to put their device onto an FXO port, you can't have asterisk connect directly |
00:21.09 | Moc | wepy, yes, if it to a VoIP termination... so IP/IP is FREE... - Bandwidth cost |
00:21.13 | script | so I don't have voip phone I have qos box |
00:21.19 | *** join/#asterisk pvinci (~root@ool-18bd103c.dyn.optonline.net) |
00:21.27 | davetroy | it's pstn orig/termination that costs $$ |
00:21.29 | script | oh ok |
00:21.40 | nikko_ | i remember something about some more stuff you have to clean up when doing an upgrade from a prettty old * version (may 2004) - could someone remind me? |
00:21.47 | nikko_ | my upgrade isn;t going to swell |
00:21.52 | wepy | why doesn't the world just get rid of phones and have special phones that use ethernet jacks.. UDP or something |
00:21.53 | Moc | _-Jon-_: well check your context on the called machine, if it allowed in iax |
00:21.56 | davetroy | nikko -- modules.con f |
00:22.02 | file | wepy: VoIP isn't 100% reliable |
00:22.08 | robl^ | Asterisk is the "operator". It connects lines and phones. Phones can be POTS phones, or VoIP of various types. Phone "lines" can be standard pstn lines, T1s, VoIP channels (SIP, SCCP, MGP, etc). Asterisks job is to do all the switching between them |
00:22.11 | file | espically with NAT and stuff |
00:22.18 | nikko_ | delete it? |
00:22.29 | script | FXO port? |
00:22.32 | wepy | file: on a scale of 1 to 10, how much does it suck (10 being good, 1 being very very sucky) |
00:22.34 | davetroy | no, /etc/modules.conf; make sure you are not loading unnecessary zaptel drivers |
00:22.41 | davetroy | it can really slow down machine |
00:22.43 | file | wepy: depends on the conditions |
00:22.45 | nikko_ | no, it's empty |
00:22.48 | wepy | file: such as? |
00:23.00 | file | script: yes, you have to connect the Packet8 ATA into asterisk via an FXO port (where you can plug a phone line in) |
00:23.10 | nikko_ | it's a zaptel problem with my t100 I think |
00:23.20 | nikko_ | ztcfg gives an error |
00:23.20 | file | wepy: low bandwidth internet connections, satellite, UDP and firewalls, NAT |
00:23.23 | wepy | maybe our networks are too damn slow.. |
00:23.26 | *** join/#asterisk jdg (~chatzilla@CA03F87B.adsl.mana.pf) |
00:23.30 | davetroy | nikko; i dunno, i have a t100 and haven't had any probs with upgrades |
00:23.41 | file | those aren't fatal but they sometimes present a challenge |
00:23.42 | davetroy | did you update zaptel and libpri both and make/make install? |
00:23.49 | wepy | how much bandwidth does a phone call take..? |
00:23.58 | file | wepy: depends on the codec (call quality) |
00:23.59 | robl^ | wepy: depends on the compression |
00:24.11 | wepy | file: the best codec available? |
00:24.14 | file | the smallest you can go is about 25kbps with G729 |
00:24.20 | wepy | so that it sounds goood.. |
00:24.22 | script | well I have my asterisk server on the same switch as my packet8 ata device |
00:24.23 | nikko_ | yep, i renamed all my source dirs and fetched new everything and did a make clean and make install |
00:24.26 | matobago | wepy:depends on the media too |
00:24.29 | wepy | can't you use ogg or something? :) |
00:24.30 | robl^ | wepy: its like watching streaming video online.. sometimes its high quality some times it sucks.. |
00:24.33 | file | the best quality codec, ULAW, is 64Kbps... I meant 25Kbps for G729 |
00:24.37 | file | kilobits... |
00:24.37 | wepy | oh |
00:24.38 | JRB | please please please can anyone point me in the right direction for * SMS in the UK? |
00:24.45 | davetroy | ogg/mp3 etc are not designed for realtime compression. |
00:24.51 | paulc | JRB: Isn't it on the wiki? |
00:24.51 | wepy | ok |
00:25.05 | JRB | paulc - not much on there |
00:25.05 | wepy | ULAW |
00:25.08 | script | but I need piece hardware to connect it right? |
00:25.11 | robl^ | wepy: you have to use a codec that Asterisk AND the phone both understand. most phones don't use ogg voribis |
00:25.22 | matobago | i think ULAW use more than 64kbps |
00:25.29 | paulc | JRB: I've seen somethign somewhere and thought it was the wiki - talked about how to register your PSTN line with BT to get SMS via FSK |
00:25.30 | wepy | sp with 25Kbps voices sound normal? |
00:25.34 | davetroy | yeah, the new cisco 7980 ogg vorbis deskphone.. ;) |
00:25.41 | file | the ULAW codec itself is 64Kbps |
00:25.46 | wepy | ok |
00:25.51 | file | with TCP/IP overhead it ends up to be 80Kbps |
00:25.55 | robl^ | davetroy: I thought it was the Snom 450? |
00:25.56 | PatrickDK | ulaw with overhead is 87kbps |
00:25.57 | wepy | 64Kbps 1 way? |
00:26.02 | PatrickDK | one way, yes |
00:26.06 | wepy | foo! |
00:26.13 | file | mmm bandwidth |
00:26.16 | file | ULAW is beautiful though |
00:26.19 | PatrickDK | gsm is about 27kbps one way |
00:26.21 | davetroy | need to release the new IAXYogg |
00:26.27 | wepy | america = cheap with the upload |
00:26.32 | PatrickDK | g729 is like 23kbps |
00:26.41 | rollotomnasi | cvs server down? |
00:26.43 | zyke | hi guys |
00:26.47 | robl^ | what about the Sipura QuickTime edition? |
00:26.51 | zyke | any one using callme files? |
00:26.56 | matobago | i think g729 is a good option |
00:27.04 | script | hey file this FXO port is piece of hardware I need to get? |
00:27.05 | PatrickDK | I perfer gsm :) |
00:27.09 | nikko_ | after my upgrade of asterisk (and reboot) ztcfg gives me :ZT-SPANCONFIG failed on span 1: Invalid argument (22) |
00:27.12 | file | script: yes |
00:27.22 | script | you have url? |
00:27.22 | robl^ | g729, sound quality, is good. but its pricey and patent encumbered |
00:27.22 | zyke | i get the call released in few seconds when using same kind of channel |
00:27.24 | file | script: Packet8 prevents you from connecting third party devices to them, so you don't abuse the service |
00:27.31 | davetroy | nikko -- that's including downloading new zaptel/libpri? |
00:27.32 | file | script: http://www.digium.com/ hardware... go look |
00:27.38 | script | thx alot man |
00:27.40 | wepy | so.. another option is to buy VoIP phones.. they could connect to your computer with a NIC? |
00:27.48 | nikko_ | yes, I fetched everything |
00:27.50 | nikko_ | today |
00:28.07 | file | wepy: they would connect into a switch, router, or hub and connect to your asterisk machine via the internet or on a local network |
00:28.14 | nikko_ | then I did a make clean; make install on zaptel, libpri and asterisk |
00:28.19 | nikko_ | in that order |
00:28.19 | matobago | someone make a g729 open source |
00:28.24 | wepy | file: now that sounds like what phones should be :) |
00:28.32 | wepy | file: do voip phones use DHCP or something? |
00:28.33 | robl^ | wepy: think of it just like a computer network.. the phones are just tiny single-purpose computers. they all plug into a switch/hub |
00:28.49 | nikko_ | now chan_zap.so fails to load |
00:28.56 | JRB | paulc: do you know if it is an app which I load separately? |
00:28.58 | PatrickDK | file, whip out that bofh |
00:29.01 | file | wepy: there are no standards for how a phone operates, except the protocol. It's up to the manufacturer to write the ability to use DHCP into the firmware |
00:29.08 | wepy | if everyone just bought voip phones.. we could all call each other with dyndns :) |
00:29.13 | file | all I see support DHCP |
00:29.23 | PatrickDK | wepy, we already do that |
00:29.31 | zyke | any one using callme files? |
00:29.32 | wepy | PatrickDK: cool! |
00:29.43 | script | man 85 bucks |
00:30.12 | paulc | JRB: Yeah, I think it is.. app_sms or something? |
00:30.18 | paulc | yeah - do "show application sms" for infor |
00:30.25 | file | script: are you looking at an FXO module? that's for use with the TDM400 card |
00:30.28 | wepy | so your phone router in that case would be asterisk? |
00:30.42 | wepy | if you had a firewall, would you just open a range of ports? |
00:30.56 | wepy | like say 8 ports for 8 incoming calls or something? |
00:31.00 | robl^ | think of Asterisk as a server.. the phones are clients. |
00:31.03 | PatrickDK | kindof |
00:31.14 | PatrickDK | 1 port for call setup, and alot of ports for audio traffic |
00:31.18 | script | file I'm just trying to get asterisk working with packet8 |
00:31.25 | wepy | cool |
00:31.26 | JRB | paulc: your application is not registered |
00:31.28 | file | script: you can't connect asterisk directly to packet8 |
00:31.38 | JRB | paulc: i am using the v1-0 release |
00:31.38 | wepy | why doesn't everyone do this :) |
00:31.48 | script | yeah I know I need to buy FXO port |
00:31.55 | wepy | people with broadband should at least.. |
00:31.55 | script | but I only see the module |
00:31.58 | paulc | JRB: If you do "show applications" is SMS listed? |
00:32.02 | malaiwah | what voip phone could i use on windows, x-lite is not to my taste... |
00:32.02 | file | look harder. |
00:32.06 | script | ight |
00:32.10 | robl^ | GalaxyVoice beats the SH*T out of Packert8 and it connects directly via SIP |
00:32.11 | JRB | paulc: no :-( |
00:32.17 | file | I see it right in front of me |
00:32.21 | davetroy | wepy the * community is growing quickly. |
00:32.22 | file | Single-Port FXO PCI interface card for interfacing with a standard analog phone line. |
00:32.26 | wepy | robl^: is GalaxyVoice a type of phone? |
00:32.28 | file | how much clearer can it be? |
00:32.40 | file | GalaxyVoice is a provider |
00:32.50 | robl^ | wepy: GV is a service provider. tehy connect me to a local phone numnber |
00:32.51 | wepy | they forward IP -> pstp? |
00:32.58 | file | yes. |
00:33.04 | wepy | hm |
00:33.16 | paulc | JRB: What's it say when you do "show version" ? |
00:33.19 | wepy | can people on cell phones dial by IP address yet? |
00:33.24 | file | wepy: haha no |
00:33.31 | robl^ | $20US per month and I get unlimited long distance |
00:33.33 | bkw_ | JRB you dont have 1.0.1 you have the old ass 1.0 |
00:34.08 | routerheads_atho | how many long distance minutes do you use a month? |
00:34.09 | JRB | bkw: does that not have SMS in it? |
00:34.18 | script | 99 bucks |
00:34.35 | PatrickDK | wepy, use e164.org |
00:34.41 | davetroy | no, that's sip:wepy@wepy@dyndns.org |
00:34.52 | wepy | ah |
00:35.04 | PatrickDK | sip:wepy@wepy.dyndns.org you mean :) |
00:35.11 | robl^ | sip:wepy@wepy@dyndns.org/911 even |
00:35.14 | wepy | oh, that will specify a specific phone at my IP? |
00:35.15 | wepy | :) |
00:35.25 | script | file 99 bucks |
00:35.32 | *** part/#asterisk matobago (~matobago@65-77-23-11.ptp.ezeronetworks.net) |
00:35.33 | JRB | bkw: does that not have SMS in it? |
00:35.47 | wepy | neato |
00:36.01 | wepy | how much is a voip phone again? |
00:36.11 | script | 20 bucks |
00:36.13 | amer | $999999 |
00:36.13 | wepy | i mean, could i get a regular phone for like $30 ? |
00:36.15 | wepy | hah |
00:36.16 | wepy | ok |
00:36.25 | file | $30? no. |
00:36.29 | PatrickDK | you can use a normal phone, and an iaxy or sipura adaptor |
00:36.39 | wepy | adapters? |
00:36.40 | PatrickDK | or just use a tdm400p card |
00:36.43 | wepy | wth |
00:36.45 | script | dude who else had packet8 again? |
00:36.56 | script | file that card is 99 bucks |
00:36.57 | PatrickDK | www.sipura.com |
00:37.02 | file | wepy: yeah... a VoIP FXS adapter... gives you a phone line connected to a VoIP provider |
00:37.06 | file | script: I know, I can read |
00:37.09 | robl^ | a VoIP phone runs from $65 to $500 |
00:37.14 | file | wepy: the Sipura for example is $94.95 for 2 ports |
00:37.15 | wepy | no i don't mean the provider.. |
00:37.19 | wepy | the physical phone |
00:37.41 | wepy | ok |
00:37.48 | JerJer | or the PAP2 for $49.99 |
00:37.56 | wepy | i should set these up for my family.. |
00:37.59 | file | JerJer: shhhhhhhh |
00:38.01 | file | JerJer: :) |
00:38.09 | davetroy | the unavailable pap-smear-na for $49.99... |
00:38.10 | file | JerJer: mine comes in tomorrow... |
00:38.11 | script | thx for your help file |
00:38.15 | wepy | christmas = linux/asterisk + voip phone -> free calls to family |
00:38.36 | wepy | don't tell me, there are wireless voip phones too right? |
00:38.46 | gambolputty | yes |
00:38.48 | robl^ | what is this PAP2, JerJer? |
00:38.50 | wepy | :o |
00:38.52 | davetroy | yeah, 802.11 sip... |
00:38.56 | wepy | cool@ |
00:39.01 | robl^ | pap2 is a new ATA? |
00:39.04 | wepy | !@ |
00:39.04 | davetroy | pap2-na = not available |
00:39.10 | wepy | is sip the protocol? |
00:39.15 | _-Jon-_ | Arg, can someone please help me with an iax problem.. I keep getting "rejected connect attempt from.." errors whenever I get a call |
00:40.02 | *** part/#asterisk mitchel (~mitchel@69-169-54-180.anhmca.adelphia.net) |
00:40.20 | wepy | also.. what about encryption? |
00:40.31 | wepy | if there are wireless voip phones, i suppose you'd want to use some protocol that encrypted it.. |
00:40.31 | Vco | anyone know the guestimated timeframe for tdm400 w/fxo backorders? |
00:40.48 | wepy | Vco: what's a tdm400 ? |
00:40.50 | file | wepy: the Sipura units support encryption, but besides that - nothing yet |
00:41.05 | Vco | or whatever the hell the digium card is.. |
00:41.08 | wepy | file: does asterisk encrypt then too? |
00:41.23 | file | wepy: nope, not that I know of |
00:41.28 | wepy | phones need linux :) |
00:41.29 | Vco | ahh..yea..tdm |
00:41.41 | file | some VoIP phones run Linux |
00:41.45 | JRB | bkw: does that not have SMS in it? |
00:42.03 | bkw_ | JRB no the old ass 1.0-stable does not |
00:42.14 | wepy | cool though |
00:42.20 | buzzyd | does transcoding take place when going from sip to iax? |
00:42.28 | PatrickDK | not really |
00:42.35 | wepy | cuz my phone line right now costs like $18/month.. then we use a phone card.. |
00:42.35 | PatrickDK | only if codec doesn't match |
00:42.41 | robl^ | tran scoding is from different compression codecs |
00:42.47 | bkw_ | http://www.asterlink.com/svp/ |
00:42.50 | wepy | so i bet voip -> provider is cheaper |
00:42.56 | buzzyd | ok thx |
00:42.58 | *** part/#asterisk zaptel (~el@200.91.70.82) |
00:42.59 | robl^ | has nothing to do with transort protcol |
00:43.29 | Docelm0 | DONT BUY THE NEW ATA! I have 2 of them and I am fighting with Linksys/Vonage to get a new firmware.. |
00:43.44 | Docelm0 | I have been on hold for over 2 hours and talked to 6 different people already |
00:43.58 | Docelm0 | But if/when I get it.. TRUST ME! I will put the firmware out there for download! |
00:43.59 | JerJer | huh? |
00:44.06 | routerheads_atho | Have you hooked them to the net? |
00:44.08 | JerJer | listen to me |
00:44.12 | JerJer | no firmware is needed |
00:44.14 | Docelm0 | Not yet |
00:44.20 | routerheads_atho | DONT! |
00:44.21 | JerJer | it is a stock SIpura inside |
00:44.22 | Docelm0 | How do I hack the linksys firmware then? |
00:44.24 | wepy | thanks for the info :) |
00:44.25 | JerJer | you don't |
00:44.46 | JerJer | Linksys simply vacuum formed some plastic around a better layed out PCB SIpura 2000 |
00:45.08 | file | JerJer: cuter eh? |
00:45.29 | *** part/#asterisk wepy (~wepy@pcp09870567pcs.ewndsr01.nj.comcast.net) |
00:45.37 | routerheads_atho | I think they look great compared to the Sipura |
00:45.41 | JerJer | hell yeah |
00:45.46 | JerJer | blue LEDs ! |
00:45.51 | Docelm0 | I have 2 of them.. How do I hack them so I can use them? |
00:45.58 | Docelm0 | And why should I not hook them up to the net? |
00:46.06 | file | if I put up a UPS release notice... that I wrote out... would UPS accept that and leave my package? |
00:46.07 | JerJer | vonage locks them down real hard |
00:46.15 | file | Docelm0: are they PAP2 or PAP2-NA? |
00:46.20 | Docelm0 | Yes.. |
00:46.23 | Docelm0 | PAP2 |
00:46.29 | JerJer | file: if you sign it, yes they will accept and leave your package at your own risk |
00:46.29 | Docelm0 | and RT31P2 |
00:46.32 | bkw_ | did you hook them to the internet first? |
00:46.34 | file | ah yes don't let them near an internet connection |
00:46.37 | Docelm0 | No |
00:46.42 | bkw_ | then you're safe |
00:46.45 | Docelm0 | Why? |
00:46.46 | bkw_ | its like easy to crack |
00:46.47 | file | JerJer: exxxxxxxxxxxxxxcellent |
00:46.54 | bkw_ | you let them get on the net they are locked down |
00:47.06 | Docelm0 | How do I get around it? |
00:47.17 | Docelm0 | What do I need to do? :) |
00:47.20 | bkw_ | cross over cable |
00:47.22 | bkw_ | tcpdump |
00:47.23 | bkw_ | and some clue |
00:47.40 | PatrickDK | clue is hard to come by these days |
00:47.42 | file | which many in this day and age lack in many regards |
00:47.46 | Docelm0 | ok I got the crossover part.. But tcpdump? |
00:47.49 | bkw_ | emerge -pv clue |
00:48.05 | robl^ | apt-get install clue |
00:48.12 | bkw_ | no clue found |
00:48.24 | file | By signing below I give permission for UPS to leave any package being delivered from Frontier PC in the shed... that'll work |
00:48.43 | JerJer | should, yes |
00:48.44 | robl^ | file: do you have the tracking #? |
00:48.51 | robl^ | I would add that if you do |
00:48.59 | file | excellent |
00:49.09 | bkw_ | OK damn it |
00:49.12 | bkw_ | you guys with two nicks |
00:49.14 | bkw_ | need to get the hell out |
00:49.24 | drumkilla | screen is your friend |
00:49.28 | bkw_ | my tab completion is messed up |
00:49.29 | file | bkw_: :) |
00:49.32 | bkw_ | drumkilla yes you know it :) |
00:49.39 | Docelm0 | Um, Im at home.. Not work.. :) |
00:49.46 | bkw_ | well pick on |
00:49.48 | bkw_ | not both boi |
00:49.49 | drumkilla | hehe |
00:49.49 | bkw_ | :P |
00:49.57 | bkw_ | s/on/one/ |
00:50.23 | file | I must go kill reality TV shows |
00:50.30 | *** join/#asterisk gregwood (~lynnwood@adsl-068-209-211-070.sip.cha.bellsouth.net) |
00:50.31 | bkw_ | I'm with ya |
00:50.44 | JRB | bwk: thanks - is there any more info on how this works exactly? |
00:50.51 | zyke | any one using callme files? |
00:51.16 | amer | hey v were hacking those linksys boxes |
00:52.12 | amer | crossover cable tcpdump, whats next |
00:52.22 | gregwood | I need help with a new asterisk install. I can connect to the server and it says it playing the demo but I get no sound. I am using iaxphone and it shows input and output. My headseat works fine and I am getting my sounds |
00:52.27 | gregwood | from windows |
00:52.35 | JerJer | amer: ethereal is more friendly |
00:52.52 | amer | what should I look for? |
00:52.57 | amer | I will use snifer pro |
00:53.42 | JerJer | it is going to request an XML file from some TFTP server |
00:53.54 | gregwood | I have also tried sjphone with sip and get the same thing |
00:53.56 | JerJer | you need to become that TFTP server and provide the unit with that XML file |
00:53.58 | kiel | bouyaka ! |
00:54.01 | kiel | hello JerJer |
00:54.15 | amer | woo hooo |
00:54.16 | amer | nice |
00:54.22 | file | become the TFTP server! become trivial! |
00:54.22 | amer | I can do that |
00:54.38 | Docelm0 | So how do I hack my Linksys devices? |
00:54.44 | file | JerJer just explained |
00:54.48 | JerJer | Docelm0: pay attention |
00:54.51 | Docelm0 | I have not connected them to the net.. |
00:55.00 | Docelm0 | I am.. But I dont see any directions.. :( |
00:55.24 | amer | connect them together with a Xover cable and wait till u hear them talk |
00:55.29 | PatrickDK | mimic their tftp server |
00:55.41 | JerJer | but do not let it talk to the real TFTP server |
00:55.47 | file | hrm I'm hungry, perhaps I should consume food |
00:56.26 | routerheads_atho | any advice on the xml file? |
00:56.32 | amer | perhaps u should send some our way too |
00:56.40 | file | amer: nah |
00:56.42 | NormAst | Anyone know how to get call forwarding working under * |
00:57.00 | JRB | upgraded to 1.0.1 and it won't start - loader.c:380 load_modules: Loading module res_parking.so failed ? |
00:57.10 | file | res_parking.so does not exist anymore |
00:57.19 | NormAst | I would hook them to a cheap hub (not a switch) using eithereal and grab packets |
00:57.19 | JerJer | rm /usr/lib/asterisk/modules/res_parking.so |
00:57.32 | NormAst | find out what the address of there TFTP server is |
00:57.41 | JRB | JerJer - you know what the prob is? |
00:57.45 | file | if it contacts Vonage at all, you're screwed so just make sure it doesn't |
00:58.03 | NormAst | Who has the firmware? |
00:58.15 | Docelm0 | How so? |
00:58.17 | file | Someone has the firmware! |
00:58.17 | Ehsan | anyone has got the Astricon 04 tutorials ? |
00:58.19 | NormAst | I hacked my primus d-link.... |
00:58.20 | file | <Someone> who me? |
00:58.21 | file | Yes you! |
00:58.23 | PatrickDK | I would recomment a firewall rule |
00:58.24 | file | <Someone> Not true! |
00:58.36 | routerheads_atho | i dont believe it is firmware, its an xml config file |
00:59.03 | NormAst | XML? Some are locked via the firmware... Primus is one of them. |
00:59.07 | PatrickDK | no other way to get the linksys one? |
00:59.18 | *** join/#asterisk Tidal (~J@202.83.64.218) |
00:59.19 | amer | NormAst what do need to know about call park |
00:59.21 | JerJer | NormAst: bleh |
00:59.45 | NormAst | amer: Need to know how to get asterisk to enable and disable call Forwarding. |
01:00.04 | PatrickDK | normast, on what phone line? |
01:00.09 | Sivana | is 'make update' broke now... will it download dev version? |
01:00.24 | NormAst | I am using a AZATEL box with *. |
01:00.44 | PatrickDK | on phones? or telco? |
01:00.50 | NormAst | On Phones.. |
01:01.01 | PatrickDK | you have to configure dialplan for that |
01:01.13 | NormAst | Oh.. Anysamples? |
01:01.15 | JRB | I have removed res_parking.so from my modules.conf file but now I get another error - chan_zap.so undefined symbol: ast_pickup_call? |
01:01.37 | JerJer | JRB: you did not recompile |
01:01.43 | JerJer | make clean install |
01:03.51 | *** join/#asterisk implicit (~implic1t@dhcp-248043.mobile.uci.edu) |
01:04.25 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
01:04.31 | JRB | did a make clean install |
01:04.39 | JRB | same error |
01:04.41 | implicit | bkw_: i would call asterisk an ip pbx, not a softswitch |
01:04.52 | JRB | should my configs from v1.0 work ok? |
01:04.56 | bkw_ | implicit it can be a softswitch |
01:05.01 | bkw_ | just not there "yet" |
01:05.14 | implicit | yeh |
01:05.20 | implicit | damn ss7 support |
01:05.30 | JRB | JerJer: did a make clean install but the error remains |
01:05.37 | implicit | wow that girl is hot, if you were here you would be come straight bkw_ |
01:06.35 | JerJer | JRB: then you do not have updated code |
01:06.47 | *** join/#asterisk john86753 (~john86753@12-218-53-187.client.mchsi.com) |
01:06.54 | *** join/#asterisk HiTech69 (hitech@34-29.202-68.tampabay.rr.com) |
01:07.21 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
01:07.22 | *** mode/#asterisk [+o twisted] by ChanServ |
01:07.47 | miguellinux | Hi I need some help with callgroups and pickupgroups |
01:08.06 | JRB | JerJer: how do I get this fixed? I did a cvs checkout -r v1-0-1 asterisk zaptel libpri asterisk-sounds asterisk-addons |
01:08.28 | JRB | then went in to /usr/src/asterisk and did make clean ; make install |
01:08.34 | implicit | hey twisted |
01:08.37 | implicit | whoats up man |
01:08.38 | twisted | hi |
01:08.46 | twisted | not sure what a whoat is |
01:08.55 | john86753 | This question may be stupid and impossible. I want to ring the phone in my house if I get a SIP phone call is this possible? I only have 1 phone line and when I do Zap/1 in my extensions I get answered immediatly |
01:09.29 | twisted | john86753, elaborate on your setup a bit |
01:09.38 | JerJer | rm -rf /usr/lib/asterisk/ |
01:09.44 | ariel_ | miguellinux, what is your question. I might be able to help. |
01:10.18 | JerJer | cvs co zaptel, libpri, asterisk ; cd zaptel && make install && cd../libpri && make install && cd ../asterisk && make install |
01:10.32 | JerJer | hmm no commas... cvs co zaptel libpri asterisk |
01:10.54 | john86753 | sure, I have 1 phone line running into my FXO card, when I recieve an internet phone call I would like to have the regular phone in my house ring. Without buying a cisco ATA device or anything like that. |
01:12.23 | bkw_ | http://www.amazon.com/exec/obidos/ASIN/B000003TWD/qid=1096593115/sr=ka-2/ref=pd_ka_2/002-3292082-3726443 |
01:12.28 | bkw_ | WTF are they smokin |
01:12.30 | bkw_ | anyone have that CD? |
01:13.25 | john86753 | twisted, does that make sense? |
01:13.33 | JRB | JerJer - can I check out the stable 1.0.1? |
01:13.48 | ariel_ | john86753, without an fxs port and a little reconfiguration of your phone lines it's not posible. |
01:14.08 | john86753 | ok, that is what I thought, no biggie |
01:14.23 | Docelm0 | What CD? |
01:14.37 | *** join/#asterisk garret_ (~garret@c-67-172-146-73.client.comcast.net) |
01:15.32 | JRB | JerJer - can I check out the stable 1.0.1? Will that work ok with SMS? |
01:17.02 | ariel_ | anyone here has connected a Mitek pbx to an asterisk box. Via it's T1 port? |
01:17.17 | JerJer | http://slashdot.org/articles/04/09/30/1916252.shtml?tid=215&tid=98&tid=1 |
01:17.44 | *** join/#asterisk wizhippo (~wizhippo@CPE0080c816780d-CM0000390aca5e.cpe.net.cable.rogers.com) |
01:18.02 | *** join/#asterisk garret_ (~garret@c-67-172-146-73.client.comcast.net) |
01:18.03 | JRB | price wars! |
01:18.06 | JRB | JerJer - can I check out the stable 1.0.1? Will that work ok with SMS? |
01:18.24 | Docelm0 | Where can I purchase the PAP's or devices like them? |
01:18.34 | JerJer | staples |
01:18.40 | JerJer | only staples |
01:18.53 | file2 | a blueberry muffin! |
01:19.11 | *** join/#asterisk JimVanM (~jimvanm@HSE-MTL-ppp77165.qc.sympatico.ca) |
01:19.39 | ariel_ | AT&T is trying to get into the market |
01:23.13 | gregwood | HELP! have installed asterisk and can connect but can't get any audio. iaxping looks good. |
01:23.35 | file2 | I'm on break. |
01:23.42 | JerJer | i'm at lunch |
01:24.23 | JRB | lunch was a long time ago for me - 2:24am Zzzzz Zzzz Zzzz |
01:27.15 | *** join/#asterisk pdk (Patrickdk@dyn-19-218.myactv.net) |
01:29.36 | Moc | JerJer, nothing new for Canadian user ? |
01:30.52 | JunK-Y | moc: how ya're adding buddies to ur list ? |
01:30.52 | JunK-Y | (on IP 500) |
01:30.52 | Moc | i dont use the buddies on the ip 500 |
01:30.52 | Moc | I dont think * support it |
01:30.59 | JunK-Y | that's bad. :( |