| 00:05.00 | *** join/#asterisk metafore (n=metafore@71.6.34.83) |
| 00:06.19 | *** join/#asterisk groogs (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 00:11.05 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 00:13.56 | ManxPower | CunningPike: nope |
| 00:14.14 | CunningPike | ManxPower: It's OK - I rtfm and found it |
| 00:19.46 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-16-120.cybersurf.com) |
| 00:23.03 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 00:25.16 | EmleyMoor | Anyone on FWD or a peer prepared to try my IVR? |
| 00:26.09 | EmleyMoor | Also, what's worth doing while I wait for my Digium card? Install zaptel? |
| 00:27.55 | ManxPower | Read The Book |
| 00:27.58 | ManxPower | ~book |
| 00:27.59 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:28.11 | EmleyMoor | Been reading the book for months |
| 00:33.56 | EmleyMoor | After installing the Debian zaptel package, what do I need to do? Any guide or is it just a matter of picking up in the book where the pkg leaves off? |
| 00:34.18 | ManxPower | Ask Debina |
| 00:34.21 | ManxPower | Debian |
| 00:35.34 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 00:35.39 | implicit | debina is a smart girl |
| 00:37.36 | *** part/#asterisk metafore (n=metafore@71.6.34.83) |
| 00:37.41 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:41.41 | hmmhesays | and hot |
| 00:43.26 | EmleyMoor | It looks like I have everything but the modules |
| 00:44.02 | hmmhesays | EmleyMoor: are you a bloody brit? |
| 00:44.17 | EmleyMoor | /whois hmmhesays |
| 00:44.22 | EmleyMoor | Sugar |
| 00:44.48 | EmleyMoor | Yes |
| 00:44.57 | hmmhesays | haha |
| 00:45.18 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 00:45.48 | *** join/#asterisk aao_pwner (n=_s@c-24-21-91-140.hsd1.wa.comcast.net) |
| 00:50.21 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 01:05.52 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:06.02 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:08.04 | CunningPike | Has any one had problems with the "Placed Calls" list on Polycom phones? Sometimes the phone just goes back to the idle screen when the 'Dial' softkey is pressed |
| 01:08.35 | CunningPike | And it has nothing to do with the number - two entries with the same number - one works and one doesn't |
| 01:13.33 | aptura | not me |
| 01:14.14 | aptura | is this affecting all the phones? |
| 01:16.52 | CunningPike | aptura: All the IP501s |
| 01:17.43 | aptura | mmm thats not good. Thay are all reading the same file? |
| 01:19.09 | aptura | I am sure you have done all the obvios. so you are pulling some overtime to have this resolved? |
| 01:39.32 | *** join/#asterisk BrianC69 (n=brianc@bb-205-209-95-138.gwi.net) |
| 01:39.38 | BrianC69 | hello |
| 01:40.04 | BrianC69 | wondering if i might get some help with trixbox |
| 01:57.46 | sevard | BrianC69: here? not likely. |
| 01:58.16 | BrianC69 | how come |
| 01:58.36 | sevard | because most people here don't support third party GUIs |
| 01:58.40 | sevard | ~AMP |
| 01:58.41 | jbot | methinks amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 01:58.52 | sevard | go there. |
| 01:59.29 | BrianC69 | i'd be willing to edit the stuff directly to get it working. don't mater if its freepbx or not |
| 02:00.01 | sevard | I'd suggest , no matter what your problem is, to start with plain jane asterisk. |
| 02:00.21 | sevard | if you try weeding through all that fat trixbox has installed, it'll take you eyears |
| 02:00.33 | BrianC69 | i had it working then broke something. all incoming calls are busy. any clue what conf to look at |
| 02:00.44 | mog | russellb, ! |
| 02:02.13 | sevard | BrianC69: Dude, you're not hearing what I'm saying. |
| 02:02.25 | BrianC69 | ok |
| 02:02.57 | sevard | look at your extensions.conf and whatever technology you're getting your connection by, sip or iax.comf |
| 02:03.04 | sevard | or if you have a pri zapata.conf |
| 02:03.24 | BrianC69 | thanks i will look at extenesion |
| 02:05.56 | *** join/#asterisk iaxy (i=keefejoh@gateway/web/cgi-irc/ircatwork.com/x-e0d67fe49e94980e) |
| 02:07.40 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:13.37 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 02:22.12 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 02:23.43 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:25.43 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:01.07 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:06.06 | *** join/#asterisk Hansin321 (n=chatzill@c-67-190-5-42.hsd1.co.comcast.net) |
| 03:07.02 | *** join/#asterisk GTX (i=charlie@pdpc/supporter/monthlybronze/GTX) |
| 03:07.03 | *** part/#asterisk GTX (i=charlie@pdpc/supporter/monthlybronze/GTX) |
| 03:08.33 | Hansin321 | I am looking to (maybe) create a conference bridge at work (using MeetMe of course). Does anyone know if I can set up Asterisk to appear as an MGCP endpoint (since or VoIP system is currently MGCP) on the network that I can run as the bridge (i.e. I can assign it an extension, and that will be what people call to bridge on). Thanks. |
| 03:11.29 | *** join/#asterisk justinu|laptop (n=Justin@cpe-76-170-91-201.socal.res.rr.com) |
| 03:26.28 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:36.27 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:36.57 | *** join/#asterisk avudz (i=avudz@gw.pendopo.melsa.net.id) |
| 03:37.24 | aptura | Anyone here use the snap dialer? I have mine running and works pretty good. |
| 03:42.59 | *** join/#asterisk ellisdee (n=ellisdee@cpe-72-181-175-30.houston.res.rr.com) |
| 03:43.49 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 03:53.20 | *** join/#asterisk bmg505 (n=leon@c1-211-12.rndf.isadsl.co.za) |
| 04:07.14 | *** join/#asterisk Choubaka (n=odiq@c66.110.141-241.clta.globetrotter.net) |
| 04:07.56 | Choubaka | need help to add extension for registering my gateway (I'm using openWRT and asterisk on WRT54G) |
| 04:09.49 | Choubaka | anyone who can help me plz? |
| 04:10.02 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:11.52 | sevard | Choubaka: sur |
| 04:11.52 | sevard | e |
| 04:12.05 | sevard | * on the wrt is fun. |
| 04:17.39 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 04:21.50 | *** join/#asterisk Choubaka (n=odiq@c66.110.141-241.clta.globetrotter.net) |
| 04:22.08 | Choubaka | sorry |
| 04:22.13 | Choubaka | I'm back |
| 04:23.03 | Choubaka | sevard.. can i use web gui to manage my * on my openwrt? |
| 04:26.11 | *** join/#asterisk LakeSolon (n=blake@64-83-212-33.dhcp.stcd.mn.charter.com) |
| 04:34.25 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 04:34.46 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:35.58 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 04:36.07 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 05:06.09 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 05:07.49 | *** join/#asterisk diclophis-work (n=jbardin@c-69-181-70-186.hsd1.ca.comcast.net) |
| 05:12.47 | *** join/#asterisk EFI-VJ (n=vj@ool-45756cb1.dyn.optonline.net) |
| 05:16.56 | *** join/#asterisk vjfromgt (i=vjfromgt@user-387g9ue.cable.mindspring.com) |
| 05:24.06 | *** part/#asterisk BrianC69 (n=brianc@bb-205-209-95-138.gwi.net) |
| 05:31.15 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 05:37.09 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 06:28.50 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 06:30.19 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 06:57.45 | *** join/#asterisk coppice (n=chatzill@85.157.17.210.dyn.pacific.net.hk) |
| 07:01.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:08.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:08.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 07:14.54 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:32.42 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 07:46.34 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 07:54.46 | *** join/#asterisk Choubaka (n=odiq@c66.110.141-241.clta.globetrotter.net) |
| 07:56.25 | Choubaka | got problem :S I've installed * on WRT54G with openwrt and sip.conf are well configured.. my 2 gateway are registered but I can't call the other end.. always get fast busy tone, someone? |
| 08:00.45 | Choubaka | everybody sleeping? :P hehe |
| 08:01.40 | coppice | zzzzzzzzzzZZZZZZZZZZZZ |
| 08:01.46 | Choubaka | :D |
| 08:05.45 | Choubaka | 219 users :P all sleeping |
| 08:05.48 | Choubaka | :) |
| 08:23.53 | Choubaka | so.. if someone see my quest, just answer in prv.. I'm going to sleep too :) |
| 08:34.40 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 08:39.26 | *** join/#asterisk implicit (n=implicit@ip72-197-16-68.sd.sd.cox.net) |
| 09:04.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:15.15 | *** join/#asterisk svenna (n=svenna@p548D3DC8.dip0.t-ipconnect.de) |
| 09:42.44 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 09:43.37 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-45-201.telkomadsl.co.za) |
| 09:55.08 | *** join/#asterisk lorinc (n=ang@caracas-2642.adsl.interware.hu) |
| 09:59.39 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 09:59.57 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 10:04.03 | *** join/#asterisk ManxPower (n=manxpowe@209.sub-75-203-6.myvzw.com) |
| 10:11.34 | EmleyMoor | Trying to install zaptel modules, I get install: cannot stat `doc/ztcfg.8': No such file or directory |
| 10:11.51 | EmleyMoor | Why do I get that and how do I fix it? |
| 10:12.40 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 10:13.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 10:17.14 | *** join/#asterisk af_ (n=af@ip-179-179.sn1.eutelia.it) |
| 10:20.14 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 10:25.29 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 10:41.57 | *** join/#asterisk kiddy (n=kiddy@59.93.40.107) |
| 10:43.18 | kiddy | Is anybody know how to configure the company name directory on dual server environment , so that two servers share their name directories ? |
| 10:43.19 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 11:16.09 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 11:17.38 | tRSS | hey guys. i have my asterisk machine setup like this: DSL router -> WiFi router -> asterisk machine. I have setup port forwarding from DSL router to WiFi router to Asterisk machine. A similar setup is working for ssh. but my users are not able to register with my asterisk from outside |
| 11:18.01 | tRSS | !nat |
| 11:18.04 | tRSS | ~nat |
| 11:18.09 | jbot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 11:43.30 | *** join/#asterisk knarfly (n=bcartwri@c-65-34-177-3.hsd1.fl.comcast.net) |
| 11:43.49 | knarfly | 8-) |
| 11:44.18 | knarfly | Can anyone tell me if a Pentium 3.2Ghz Celeron will make a good platform for * |
| 11:51.02 | *** join/#asterisk beyond (n=beyond@c91508a7.virtua.com.br) |
| 11:53.53 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 11:55.59 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 11:57.20 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 12:05.43 | monsted | knarfly: it's not a pentium, but it'll be fine |
| 12:06.56 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 12:12.42 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 12:20.26 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 12:21.56 | tRSS | can I telnet to port 5060 if I am checking that my asterisk is accessible behind a nat/firewall? |
| 12:21.56 | Nugget | telnet is eeeeeeevil! |
| 12:25.09 | fenlander | asterisk uses udp sip, not tcp, so no, standard telnet won't help |
| 12:26.22 | tRSS | alright, thanks |
| 12:31.12 | tRSS | fenlander: for some reason or another, my asterisk box is not accessible from outside. I have setup port forwarding for 5060 and 10000-12000 to my box from my dsl router (using dyndns). |
| 12:31.36 | tRSS | I can ssh from outside but can't register any softphone from outside? |
| 12:32.43 | tRSS | i have done some sniffing (using ethereal) and I can see that when a register is sent to asterisk, i get destination/port unreachable! |
| 12:35.17 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 12:35.44 | monsted | is your asterisk box listening on UDP/5060? |
| 12:35.55 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com) |
| 12:37.07 | tRSS | firewall on my box is completely disabled, hence, it is indeed listening on UDP 5060. infact, I can successfully register from the lan. its just the wan I am having trouble with |
| 12:41.49 | tRSS | i have done port forwarding from my dsl router to wifi router to asterisk box (which is a wired connection with my wifi box) |
| 12:46.31 | knarfly | tRSS: You can check out the post on the users forum on http:asterispbx.org |
| 12:47.19 | knarfly | tRSS: You will need to forward ports 10000 - 20000 and make some additional changes in X-Lite |
| 12:47.47 | knarfly | tRSS: http://asteriskpbx.org |
| 12:48.22 | knarfly | tRSS: Mine works fine... |
| 12:50.58 | *** join/#asterisk primeministerp (n=ppouliot@outbound.ximian.com) |
| 12:56.12 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 12:56.33 | tRSS | sorry guys, I got dropped earlier. so can anyone help my with NAT+Asterisk setup? |
| 12:57.13 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:03.52 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 13:20.34 | *** join/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 13:34.13 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 13:37.47 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:37.51 | Druken | morning everyone |
| 13:38.20 | Druken | has anyone ever had a problem with realtime iax? |
| 13:38.38 | Druken | sip and voicemail work for me without a hitch, but iax is giving me greif |
| 13:42.24 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 13:45.39 | MoutaPT | any one here has been in Asterisk Beach Camp Marbella? |
| 13:52.36 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 13:58.46 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 13:59.00 | *** part/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 14:02.26 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 14:05.44 | *** join/#asterisk jaike (i=jaike@58.69.56.216) |
| 14:07.03 | jaike | anyone know the scheduled release of 1.4 stable? |
| 14:09.01 | *** join/#asterisk coppice (n=chatzill@89.201.17.210.dyn.pacific.net.hk) |
| 14:09.55 | *** part/#asterisk jaike (i=jaike@58.69.56.216) |
| 14:26.50 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-155-98-24.red.bezeqint.net) |
| 14:26.55 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.14) |
| 14:28.43 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 14:29.21 | wiseguy_ | i want to restrict the voip phones to call only allowed networks |
| 14:29.40 | wiseguy_ | how can i do that? or where i can find more information? |
| 14:30.35 | wiseguy_ | is there other way, without context? |
| 14:33.08 | Druken | wiseguy_: define allowed networks |
| 14:33.48 | wiseguy_ | where? |
| 14:33.59 | Druken | hehehe no, i mean explain more |
| 14:34.06 | kumbalae | how do i dial with a specified DID number of my ISDN line with dial Zap ? |
| 14:34.39 | Druken | kumbalae: you don't? you set the caller id and then just dial |
| 14:35.11 | wiseguy_ | Druken: in our country there several mobile providers, and if i know the number, i can say, which network it belongs to. I want to do that some asterisk sip clients, can make calls only to one or two mobile operators |
| 14:37.07 | Druken | you'd have to have an access list or something, sounds like either context based, or a custom dialplan to check a few values |
| 14:38.22 | kumbalae | Druken, I set the callerid, but still the i get the default number of the isdn line |
| 14:38.45 | kumbalae | Druken, exten => _[6|9]X.,1,SetCallerPres(allowed) |
| 14:38.46 | kumbalae | exten => _[6|9]X.,2,SetCIDNum(64009633) |
| 14:38.46 | kumbalae | exten => _[6|9]X.,3,Dial(Zap/g1/${EXTEN}) |
| 14:38.59 | kumbalae | these are the lines doing it |
| 14:39.01 | Druken | then chances are, your telco doesn't allow you to set your CID, what type of hardware are you running? |
| 14:39.26 | kumbalae | Druken, I'm running E400P, telco said i can set my own CID |
| 14:39.48 | Druken | ok, so you have an e1 |
| 14:39.55 | kumbalae | Druken, yes |
| 14:39.59 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 14:41.44 | kumbalae | Druken, ; for zap channels I needed to put "usecallingpres=yes" in the zapata.conf to get <-- i have this line also in the zapata.conf too |
| 14:43.53 | Druken | have you gone over the zapata.conf file to make sure nothing is out of place? or possibly commented out... everything matches what your carrier expects? |
| 14:44.24 | kumbalae | Druken, yes i don't have anything that affect the did of the outgoing number |
| 14:46.36 | kumbalae | Druken, http://pastebin.ca/245851 |
| 14:46.53 | kumbalae | let me know if you find anything wrong out there or anything blocking it |
| 14:54.23 | Druken | looks fine to me.... |
| 14:55.00 | Druken | i'd be calling your carrier... have them verify you can set CID.... |
| 14:55.11 | kumbalae | you want to debug the PRI line ? |
| 14:55.21 | kumbalae | yes i called them they said, i can set any callerID |
| 14:56.32 | *** part/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 14:58.06 | *** part/#asterisk cromo (i=cromo@klej.net) |
| 14:59.37 | kumbalae | Druken, is there anyway i can query through pri line debug ? |
| 14:59.52 | kumbalae | Druken, or to analyse whether the callerid is accepted or refused ? |
| 15:44.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.159) |
| 15:44.51 | *** join/#asterisk Kamu (n=kamu@ns1.liquidexploit.com) |
| 15:44.59 | *** join/#asterisk drkludge (n=drkludge@ip68-231-40-97.ph.ph.cox.net) |
| 15:47.42 | Kamu | hey all, I was wondering on an asterisk box, could I connect a POTS line into it, as well an analog phone? so phone calls are checked by asterisk (such as diverting callerid-blocking callers to a voicebox) and then connecting the analog phone to the call? |
| 15:48.33 | Choubaka | got problem :S I've installed * on WRT54G with openwrt and sip.conf are well configured.. my 2 gateway are registered but I can't call the other end.. always get fast busy tone, someone? |
| 15:49.16 | kumbalae | Choubaka, done any ethereal capture > |
| 15:49.18 | kumbalae | ? |
| 15:49.36 | drkludge | Kamu, POTS line is connected to an FXO jack. The analog line is connected to the FXS jack. The dialplan would handle checking caller id stuff,etc. |
| 15:49.49 | Choubaka | nope kaldemar, I've got the paste of *CLI console |
| 15:50.27 | Choubaka | kumbalae: http://pastebin.ca/245694 |
| 15:50.28 | Kamu | ah I see, I am guessing special hardware is needed for the FXO and FXS ports? |
| 15:55.18 | drkludge | Kamu, digium and sangoma sells cards to go in the box while supura/linksys for example sell devices that hook up in the network |
| 15:56.13 | Kamu | what would hooking up into the network do, as apposed to straight into the box? |
| 15:56.56 | monsted | Kamu: it makes it easier to place the gateway device in a convenient place |
| 15:57.45 | Kamu | oh I see, quite so |
| 15:57.50 | drkludge | Kamu, The digium cards can be picky about the motherboard that they are inserted into. |
| 15:58.24 | drkludge | Kamu, Their cards require pci 2.2 compliance |
| 15:58.36 | Kamu | another question: could people on the network use VOIP to phone using the POTS line? |
| 15:58.51 | Kamu | sure, network does sound like a much better option |
| 16:02.28 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 16:02.40 | drkludge | Kamu, Yep. VOIP can be sip, iaxy, h.323, etc. There are analog telephone adapters that change pots stuff to ethernet |
| 16:03.41 | Kamu | nice, what happens if the POTS line is occupied? |
| 16:04.22 | Choubaka | anyone for my problem? |
| 16:05.15 | drkludge | Kamu, It's just a normal POTS line. It the kid has it locked up, then you cannot use it. You have to buy more of 'em. However, * allows you to use other providers like a VOIP company over your ISP connection. |
| 16:06.06 | Kamu | Yes :) I understand that, but will * just give an occupied message/signal? |
| 16:06.14 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:09.51 | Kamu | drkludge: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1116265541785&pagename=Linksys%2FCommon%2FVisitorWrapper <--- is that the kind of thing you could plug the POTS line into, and allow asterisk to handle it? |
| 16:10.45 | Choubaka | yes.. Kamu, or Linksys PAP2 or D-Link VTA-XX |
| 16:11.23 | Kamu | a wireless one would be best, considering the phone line isn't close |
| 16:13.01 | Kamu | thanks you two, been really helpful |
| 16:13.37 | Choubaka | u can have softphone too to do your tests |
| 16:14.07 | drkludge | Kamu, There is an asterisk book that is real helpful that pulls lots of the information together |
| 16:14.49 | Kamu | basicly I wanted teleco -> pots -> asterisk -> softphones/analog |
| 16:14.55 | Choubaka | drkludge, do you have idea for my problem? |
| 16:14.59 | Kamu | if that makes any sense |
| 16:15.04 | *** join/#asterisk implicit (n=implicit@ip72-197-16-68.sd.sd.cox.net) |
| 16:15.27 | drkludge | Choubaka, I can try |
| 16:15.51 | Choubaka | I think that a prob with extensions.conf |
| 16:15.56 | Choubaka | got problem :S I've installed * on WRT54G with openwrt and sip.conf are well configured.. my 2 gateway are registered but I can't call the other end.. always get fast busy tone, someone? |
| 16:16.21 | Choubaka | 2 gateway are d-link vta-cv |
| 16:16.28 | *** join/#asterisk Dovid (n=Dovid@line134-239.adsl.actcom.co.il) |
| 16:16.51 | Choubaka | if I do 500#.. all right.. I hear the demo |
| 16:17.17 | Choubaka | 600# is the echo test.. all right.. but I can't call ext 1234 with 8766 one |
| 16:17.34 | Choubaka | or 8766 to 1234... |
| 16:18.17 | Choubaka | here the log of CLI: http://pastebin.ca/245694 |
| 16:19.27 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:19.40 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 16:20.24 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:20.25 | Choubaka | Version: Asterisk 1.2.1 |
| 16:23.47 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 16:23.50 | Kamu | just to clear up something, when usin one of those dlink/linksys devices you CAN use a normal phone line (not voip or anything, just a plain home phone line) right? |
| 16:24.15 | Choubaka | yep no prob |
| 16:24.20 | Choubaka | When |
| 16:24.22 | InfraRed | correct |
| 16:24.27 | InfraRed | they're called ATAs |
| 16:24.34 | Choubaka | When I register to my other *@home.. all is ok.. |
| 16:24.45 | Kamu | ok, thanks :) |
| 16:24.50 | InfraRed | *@home is dead |
| 16:24.51 | Choubaka | but not on my wrt54g |
| 16:24.51 | InfraRed | isnt it |
| 16:25.01 | Kamu | also, what is with openpbx? why did they fork? |
| 16:25.29 | InfraRed | Kamu: read the wiki |
| 16:25.34 | InfraRed | www.voip-info.org |
| 16:25.38 | InfraRed | everything is there |
| 16:26.47 | Kamu | ok thanks |
| 16:26.51 | drkludge | Choubaka, Do you have some sort of outbound calling for 'em |
| 16:26.53 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-159.range86-139.btcentralplus.com) |
| 16:27.20 | Choubaka | don't understand.. I don't have any pstn in that |
| 16:28.16 | drkludge | Kamu, here's one ATA http://www.digium.com/en/products/hardware/s101i.php and another http://www.voipsupply.com/product_info.php?manufacturers_id=36&products_id=317 make sure that you get an unlocked version. |
| 16:29.08 | Kamu | awesome |
| 16:29.13 | Kamu | basicly cheapest is best ;) |
| 16:29.28 | InfraRed | wrong |
| 16:29.35 | InfraRed | you get what you pay for |
| 16:30.00 | InfraRed | except with cisco you get mediocre hardware and you lose a kidney |
| 16:30.07 | drkludge | digium uses iax protocol only while the liksys uses sip for example. |
| 16:30.10 | *** join/#asterisk Skarmeth (n=Skarmeth@201009059190.user.veloxzone.com.br) |
| 16:30.33 | Kamu | don't need all the bells and whistles, just connecting to network, speaks with * and deals with caller ids etc ;) |
| 16:30.44 | drkludge | InfraRed, well what's your favorite ATA these days? |
| 16:31.07 | InfraRed | I still have loads of ATAs coming out my ears atm |
| 16:31.19 | InfraRed | using mediatrix still |
| 16:31.37 | Kamu | send me one, if they are indeed sprouting from your ear ;) |
| 16:31.49 | Kamu | preferably cleaned, how ever |
| 16:31.51 | InfraRed | send me cash |
| 16:32.13 | Kamu | lets see how much I can get from this aussie site |
| 16:33.00 | Choubaka | drkludge.. any idea? |
| 16:33.38 | Kamu | <PROTECTED> |
| 16:33.54 | Kamu | not too bad, those PCI ones are very dear though! $700 bucks |
| 16:33.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:37.52 | drkludge | Choubaka, I don't know. It feels like you're missing context information to connect the two. |
| 16:38.07 | Choubaka | yes.. I seem to be in extensions.conf |
| 16:42.40 | *** join/#asterisk stillbourne (n=stillbou@c-67-161-131-86.hsd1.co.comcast.net) |
| 16:47.55 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:48.38 | aao_pwner | Is there any free outbound trunk service? |
| 16:48.42 | aao_pwner | Like.. |
| 16:48.48 | aao_pwner | http://www.sellvoip.net/ |
| 16:48.51 | aao_pwner | but for free |
| 16:48.53 | aao_pwner | lol |
| 16:49.25 | Kamu | what is an outbound trunk? |
| 16:49.59 | aao_pwner | erm.. |
| 16:50.02 | aao_pwner | i dont know what it's called |
| 16:50.06 | aao_pwner | an outbound "Thing" |
| 16:50.07 | aao_pwner | lol |
| 16:50.18 | aao_pwner | to connect to the worldz... |
| 16:50.32 | aao_pwner | via the interwebz |
| 16:51.26 | Kamu | if you need an outbound SIP, try voipbuster.com internetcalls.com freecalls.com/net |
| 16:51.42 | Dovid | teliax.com has been real good for me |
| 16:52.04 | aao_pwner | are they free? |
| 16:52.08 | Kamu | yessuh |
| 16:52.08 | aao_pwner | well.. |
| 16:52.09 | Dovid | :( |
| 16:52.09 | Dovid | no |
| 16:52.10 | aao_pwner | .for my pbx |
| 16:52.17 | aao_pwner | I just use skype for that shit. |
| 16:52.18 | Dovid | free = SHIT |
| 16:52.23 | Dovid | haha |
| 16:52.30 | Dovid | dont mention the s word here |
| 16:52.32 | Dovid | ~skype |
| 16:52.35 | jbot | [skype] evil. see gizmoproject.com, or offering free landlines: http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html |
| 16:52.36 | Choubaka | any tricks to make unlimites calls voip to pstn?.. any providers absolutely free? |
| 16:52.36 | aao_pwner | why? |
| 16:52.54 | Dovid | skype is evil in our eyes |
| 16:52.57 | Kamu | this is true, depends how serious the pbx is |
| 16:53.00 | infernix | Choubaka: various providers offer free dialing to landlines |
| 16:53.03 | infernix | Choubaka: voipbuster does |
| 16:53.15 | aao_pwner | Well VoIP has always been free. |
| 16:53.18 | Kamu | I beat you to it ;) |
| 16:53.23 | aao_pwner | regular phones should be completally abolished |
| 16:53.31 | Dovid | lol |
| 16:53.41 | Choubaka | infernix, no limit? |
| 16:53.42 | aao_pwner | I'm serious. |
| 16:53.46 | infernix | Choubaka: 300 minutes a week |
| 16:53.53 | infernix | then you pay |
| 16:53.59 | Dovid | no one can offer trully free |
| 16:54.02 | Dovid | calling |
| 16:54.06 | Dovid | its a loosing proposition |
| 16:54.09 | Choubaka | yeah.. the I pay... always pay:D |
| 16:54.27 | aao_pwner | how's skype do it then? |
| 16:54.32 | drkludge | Choubaka, for fast busy try this http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 16:54.34 | Kamu | bs |
| 16:54.37 | RaYmAn-Bx | infernix: is that what they are called now? :P They are such a ripoff company imho. |
| 16:54.43 | Dovid | cause skype isnt unlimited |
| 16:54.44 | Kamu | 300 minutes then you create a new account |
| 16:54.50 | Dovid | if u want that u have to pay |
| 16:54.56 | aao_pwner | I've used more than 300 miniutes out. |
| 16:54.59 | aao_pwner | And i'm good. |
| 16:55.01 | Dovid | well 300 minutes for free but they want u to pay and use them afterwards |
| 16:55.12 | aao_pwner | WEll fuck them. |
| 16:55.18 | aao_pwner | I've used more than 300 miniutes out. |
| 16:55.20 | infernix | RaYmAn-Bx: i dunno. i took a line with budgetphone for now, for both in and out. but i have read about various setups which included voipbuster so you would make use of those 300 minutes a week |
| 16:55.26 | Dovid | hell i will give u more than 500 if there is a good chance that u wull go wit me for termination |
| 16:55.40 | Dovid | on average it works to thier advantage |
| 16:55.46 | Dovid | if it didnt they wouldt offer it |
| 16:56.19 | infernix | if enough people would only dial within the free period, they'll cancel it |
| 16:56.22 | RaYmAn-Bx | infernix: I suppose they do "work"..but they keep changing name, making new companies and then gradually mess with things and start over |
| 16:56.40 | infernix | RaYmAn-Bx: well, thats not a real issue for outgoing calls |
| 16:56.49 | Dovid | from thier site |
| 16:56.51 | Dovid | The only condition is that we have said free SkypeOut within the US and Canada is guaranteed to last until the end of this year — that is, until December 31, 2006. We’re not quite sure yet what we will do after that. Maybe we extend the free period, maybe not. You’ll hear more about this towards the end of the year. |
| 16:56.52 | infernix | RaYmAn-Bx: as long as your incoming number is static, who cares really:) |
| 16:57.02 | RaYmAn-Bx | infernix: I suppose, it just requires effort :P |
| 16:57.10 | Dovid | they are trying to get many users so people will be hoked and then pay em for service |
| 16:57.16 | infernix | well you know what they say about laziness ;) |
| 16:57.36 | RaYmAn-Bx | best thing ever invented? :P |
| 16:57.45 | Damin | Skype's Free Business model is pretty smart actually.. |
| 16:58.22 | Damin | Get several millin subscribers hooked and using your service, then continue to give them the servive that costs an average of $1 / user in exchange for a $6.99 monthly subscription fee.. |
| 16:58.27 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 16:58.46 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.14) |
| 16:58.52 | aao_pwner | that's what i was thinkign they were doing |
| 16:59.33 | Damin | Think about it.. 1,000 people sign up, providing 6.9 million in revenue / month. |
| 16:59.47 | Damin | The COGS on that is perhaps 4 million / month.. |
| 16:59.56 | Damin | That's 1.9 million free and clear.. |
| 17:00.11 | Damin | Er.. 2.9 million rather.. |
| 17:01.30 | aao_pwner | Ew, i was walking back from here and when i lifted my foot my little toe got caught on the door and it like bent it perpendicular to my foot |
| 17:01.34 | aao_pwner | now it hurts like a bitch |
| 17:01.48 | Dovid | subtract what it costs to run thier network |
| 17:03.26 | aao_pwner | well right now they're probably breaking even with the skypein users |
| 17:06.28 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 17:08.50 | Damin | Dovid: That's in there.. keep in mind that they average about $1 in call costs. I'm adding in $3 MORE in overhead.. |
| 17:10.08 | Choubaka | I've found my problem thanks a lot drkludge!!!!! YAHH!! |
| 17:10.08 | Damin | Average consumer uses about 300 minutes / mo of LD.. |
| 17:10.10 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:17.21 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 17:17.59 | aao_pwner | http://www.e164.org <- what's up with that? |
| 17:18.00 | aao_pwner | what is it |
| 17:18.01 | aao_pwner | lol |
| 17:18.52 | Kamu | In a nutshell: E164.org is a public enum directory of telephone numbers that can be reached over the Internet by anyone anywhere! |
| 17:18.54 | monsted | list of phone numbers that you can reach via IP |
| 17:19.11 | tRSS | i am trying to register a softphone to my asterisk box, which sits behing a dsl router and wifi router, in this order: dsl router -> wifi router -> asterisk box. My softphone won't register from outside. |
| 17:19.11 | aao_pwner | Interestingf |
| 17:19.34 | tRSS | I have setup all the port forwarding for 5060 and rtp ports on both the routers., but can't get the softphone to register |
| 17:19.38 | drkludge | Choubaka, So what was the final resolution? |
| 17:19.48 | tRSS | although a similar setup is already working for my ssh |
| 17:20.16 | tRSS | am i missing something here? |
| 17:20.22 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:21.03 | EmleyMoor | Is there a good example about of how cadences can be used? |
| 17:21.40 | EmleyMoor | (On BT, we have single, double and triple rings for different reasons) |
| 17:22.41 | Choubaka | drkludge, missing parameters in extensions.conf |
| 17:22.47 | Choubaka | wanna have a paste? |
| 17:23.01 | EmleyMoor | I am also slightly stuck installing zaptel... a problem with a manpage I think |
| 17:23.25 | *** join/#asterisk Marshall16 (n=Marshall@d47-69-161-139.nap.wideopenwest.com) |
| 17:23.35 | drkludge | tRSS, ssh is easy because you only have to port forward one protocol |
| 17:23.36 | tRSS | can anyone help me with my asterisk+NAT issue? |
| 17:24.00 | EmleyMoor | trSS: What's the issue? |
| 17:24.18 | drkludge | tRSS, first you'd have to port forward 5060 through the two boxes to * |
| 17:24.21 | tRSS | drkludge: i understand that part, i have read that i should port forward 5060-5070 and 10000-12000 (defined in rtp.conf) at the routers to the machine |
| 17:24.44 | tRSS | which I have done. |
| 17:25.21 | tRSS | I ran ethereal, while my softphone was trying to register and amazingly, I keep getting unreachable host/port back in ethereal |
| 17:26.31 | tRSS | i have tried both ip and hostname (using dyndns) |
| 17:27.50 | drkludge | Well if ssh is working, then first try and match the routing of sip, 5060, through the three boxes. |
| 17:28.07 | drkludge | Is * the same as the ssh target? |
| 17:28.12 | *** join/#asterisk bhrobinson (n=brobinso@northtx-cust-rtr.telwestonline.com) |
| 17:29.05 | tRSS | drlukdge: i have already routed 5060 through the three boxes (infact, the softphone does register on the same lan as the server). how can i check for 5060 from the internet, because i believe sip is udp, correct? |
| 17:29.17 | tRSS | asterisk & ssh are on the same box |
| 17:29.17 | drkludge | RTP only happens after the SIP connection. That's why I am guessing that you need to look at the SIP port forwarding first. |
| 17:29.20 | bhrobinson | can anyone tell me how to pass my ANI in the dial command? |
| 17:29.51 | tRSS | drkludge: if you wish, i can give you a username to test with my asterisk box and see if it connects or not? |
| 17:30.26 | EmleyMoor | How do I determine different ring cadences on my PSTN line? |
| 17:31.44 | SwK | bhrobinson: use the SET(CALLERID(ani)=WHATEVER) function before calling dial |
| 17:32.16 | SwK | the CALLERID() function has several settings like name and num also... and dont confuse ani w/ cid num |
| 17:33.28 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:33.45 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com) |
| 17:33.55 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:34.08 | bhrobinson | SwK: thanks... I found the issue. My phone does not have a number programmed in it, so it is defaulting to the ANI number. |
| 17:39.10 | EmleyMoor | I am trying to complete the installation of zaptel on my Debian box - I have made the link to my kernel source as indicated and done the relevant make command, but make install is failing on the file doc/ztcfg.8 |
| 17:43.26 | *** part/#asterisk bhrobinson (n=brobinso@northtx-cust-rtr.telwestonline.com) |
| 17:43.33 | EmleyMoor | There is no file ztcfg.8 anywhere on the machine |
| 17:44.14 | *** join/#asterisk tRSS-other (n=tRSS@124.29.204.122) |
| 17:44.29 | tRSS-other | sorry my stupid connection died on me, drkludge |
| 17:45.22 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 17:45.50 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 17:54.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@217.54.209.1) |
| 17:54.34 | DarKnesS_WolF | just finished my 1st small asterisk review ;-0 |
| 17:54.45 | DarKnesS_WolF | http://www.eglug.org/node/2230 |
| 17:54.56 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 17:58.27 | *** join/#asterisk pavera (n=tomc@209-33-233-214.dsl.infowest.net) |
| 17:58.37 | pavera | Hello everyone, |
| 17:58.49 | pavera | I've got a quick question if anyone could help me out? |
| 17:59.55 | *** join/#asterisk Eminence (n=achin@Sears-47.resnet.brown.edu) |
| 18:00.11 | pavera | I don't know if Qwest changed something on their end, but I have a problem that the analog digium cards begin dialing before they receive dial tone from the line. |
| 18:00.36 | pavera | This results in the phone company not receiving the first digit dialed |
| 18:00.46 | Eminence | hi. i'm using asterisk-1.4.0-beta3. i've ./configured --with-zaptel, but chan_zap doesn't seemt to be getting built. any thoughts? |
| 18:01.05 | file | you installed zaptel 1.4? |
| 18:01.15 | Eminence | yeah |
| 18:01.42 | file | reran ./configure after installed? done make menuselect to confirm chan_zap will be built/dependencies are met? |
| 18:01.43 | Eminence | checked it out of svn |
| 18:01.50 | pavera | any ideas? right now I'm just prepending a digit to everything dialed out |
| 18:01.58 | pavera | but that seems like a complete hack... |
| 18:02.11 | Eminence | i haven't tried make menuselect. let me try now |
| 18:02.42 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 18:02.45 | pavera | It is happening with both a tdm2400 and a tdm400 |
| 18:02.57 | pavera | no ideas? |
| 18:03.16 | file | pavera: they don't check for dialtone before dialing afaik, but you can stick a w in front of the number... it causes a wait of 250ms |
| 18:03.54 | pavera | so the command would be dial(wBYEXTENSION)? |
| 18:04.09 | Eminence | yeah, all the dependencies are met for chan_zap and it says that it'll be built |
| 18:04.09 | file | BYEXTENSION? wow |
| 18:04.20 | file | you are using an old example |
| 18:16.51 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:16.51 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 18:16.53 | DarKnesS_WolF | ahh X100P hmm not sure i never used this card but ztcfg -vvvvv should tell u something |
| 18:17.06 | *** join/#asterisk mrbnet (n=mrbnet@corpmail1.mrbnetworks.com) |
| 18:17.50 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:17.58 | Eminence | ztcfg doesn't show anything suspecious. "Channel 01: FXS Kewlstart. \n 1 channel configured." |
| 18:31.46 | FuriousGeorge | just saw a commercial on tv for a kids quiz show, and the slogan was "smart="sqrt(cool)" and immediately what came to mind is that if cool is positive than smart must be less than half cool |
| 18:34.57 | Qwell | heh |
| 18:57.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:57.48 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 18:59.44 | DarKnesS_WolF | hmm |
| 19:00.49 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 19:06.27 | *** join/#asterisk Khisanth (n=Khisanth@pool-71-247-16-251.nycmny.east.verizon.net) |
| 19:07.51 | *** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com) |
| 19:08.22 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:11.44 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 19:12.10 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 19:14.43 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 19:15.08 | Stephnie | hi |
| 19:15.38 | *** join/#asterisk nick125 (n=ndevito@atarack/staff/nick125) |
| 19:15.44 | nick125 | hiya |
| 19:16.28 | *** join/#asterisk cryptnix (n=andrew@68-188-226-83.dhcp.bycy.mi.charter.com) |
| 19:16.31 | nick125 | I need a recommendation of an asterisk-friendly US VOIP provider, and a decent TA device for under $60 or so |
| 19:16.56 | Strom_M | TA: SPA-2002 or somesuch |
| 19:17.04 | nick125 | (local DID, areacode 505) |
| 19:17.11 | Strom_M | ah, good ol new mexico |
| 19:17.24 | Strom_M | which rate center? albuquerque? |
| 19:17.28 | nick125 | albuquerque |
| 19:17.33 | Strom_M | ok |
| 19:18.02 | Strom_M | a friend of mine claims to be having good luck with teliax |
| 19:18.07 | cryptnix | Anyone have any insight about connecting to a asterisk box sitting behind a firewall? |
| 19:18.16 | Stephnie | How to Hangup the Dialed Leg of call ??? and stay connected with asterisk... |
| 19:18.18 | Strom_M | cryptnix, open up the appropriate ports :) |
| 19:18.51 | nick125 | Strom_M: is teliax decently asterisk-friendly? |
| 19:19.03 | Strom_M | well, they speak IAX2 |
| 19:19.06 | Strom_M | is that friendly enough? |
| 19:19.32 | cryptnix | Storm_M: yeah, helpful but which ones? and on which side?! |
| 19:19.37 | nick125 | Strom_M: hah, true.. |
| 19:20.26 | Strom_M | cryptnix, well, if you give me more specifics, I can get more specific. |
| 19:20.34 | nick125 | Strom_M: do you know if they provide an interface for viewing the number of minutes used (so you don't hit the softcap and get a nice sized bill at the end of the month) |
| 19:20.37 | nick125 | ? |
| 19:20.44 | Stephnie | ??? |
| 19:20.47 | Stephnie | How to Hangup the Dialed Leg of call ??? and stay connected with asterisk... |
| 19:20.53 | cryptnix | both sides have 5060/5061 (got two lines ...) port forwarded ... |
| 19:21.02 | Strom_M | cryptnix, you're using SIP only? |
| 19:21.04 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 19:21.10 | cryptnix | Yes. |
| 19:21.30 | Strom_M | you need to open up 5060 UDP for the signaling and 10000-20000 UDP for the media |
| 19:21.31 | nick125 | Strom_M: one of my associates said he's had luck with telasip..ever heard of them? |
| 19:21.39 | Strom_M | nick125, vaguely |
| 19:24.09 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 19:24.38 | nick125 | Strom_M: looking on voip-info, it seems that alot of people were complaining about teliax and their non-existant support..hm |
| 19:24.49 | Strom_M | *shrug* |
| 19:25.17 | Stephnie | no help for me ? |
| 19:25.53 | nick125 | I get higher pings to telasip (155ms) vs. teliax (110ms) |
| 19:28.51 | *** join/#asterisk frenzy (n=frenzy@196.46.104.30) |
| 19:29.01 | frenzy | http://pastebin.ca/246017 |
| 19:29.04 | EmleyMoor | Stephnie: I can't understand what your problem is... perhaps if you explained it more clearly? |
| 19:29.21 | frenzy | trying to install Wanpipe drivers on Linux for Sangoma S518 |
| 19:29.40 | frenzy | kernel build gives errors... pls help |
| 19:30.04 | *** part/#asterisk Khisanth (n=Khisanth@pool-71-247-16-251.nycmny.east.verizon.net) |
| 19:32.24 | Stephnie | EmleyMoor: u know in Calling Cards when u want to hang up the dialed number but dont need to dial the call...u have to dial #88 |
| 19:32.34 | Stephnie | i mean #** |
| 19:32.46 | EmleyMoor | Yes, I know the sort of thing you mean... |
| 19:32.52 | EmleyMoor | Is that the exact scenario? |
| 19:33.17 | drkludge | nick125, did you try make linux26 ? |
| 19:33.19 | Stephnie | thats the what I want....Dont want to hangup the whole call...but want to hangup the dialed person and move ahead to next extension |
| 19:33.42 | drkludge | sorry that was for frenzy |
| 19:33.43 | EmleyMoor | I mean is it a calling card scenario? |
| 19:33.51 | Stephnie | yes..exactly |
| 19:34.04 | EmleyMoor | OK - can you not just send #**? |
| 19:34.25 | Stephnie | yeah... |
| 19:34.31 | drkludge | frenzy, did you try make linux26 ? |
| 19:35.05 | Stephnie | I just want this scenerio with any digits ..whether its #** or #* or **# |
| 19:35.41 | EmleyMoor | So, you make a call through asterisk to a calling card service? |
| 19:35.52 | Stephnie | no... |
| 19:36.15 | Stephnie | want to dial out from my asterisk.....but want to hangup and redial another number.... |
| 19:36.18 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 19:36.34 | EmleyMoor | Like, directly? |
| 19:36.38 | Stephnie | yes... |
| 19:36.46 | EmleyMoor | Can't you just hang up? |
| 19:36.57 | Stephnie | I can .... |
| 19:37.15 | Stephnie | it makes me dial my Asterisk again and again .... |
| 19:37.27 | EmleyMoor | "dial your Asterisk"? |
| 19:37.32 | Stephnie | it charges me for a Local Call |
| 19:37.39 | EmleyMoor | Do you mean you are calling in from outside? |
| 19:37.55 | Stephnie | yes.. |
| 19:37.57 | Stephnie | like this |
| 19:38.05 | EmleyMoor | Ah, now we get somewhere |
| 19:38.10 | Stephnie | DID number --> Asterisk --> Outgoing Carrier |
| 19:38.43 | Stephnie | I want to drop the outgoing call but remain connected with my Asterisk ...u know what I mean ? |
| 19:38.48 | EmleyMoor | I am not certain how you do it but I daresay it's possible - and its relevance to Asterisk is now clear |
| 19:39.05 | frenzy | nop |
| 19:41.14 | Strom_C | there is a way to do it; i dont remember specifics but it involves dialing out over local channels |
| 19:41.24 | frenzy | its an install script.. |
| 19:43.21 | Stephnie | yeah I have checked the features....but you cant drop the DIALED LEG OF CALL... |
| 19:44.32 | Strom_C | will you calm down? |
| 19:44.55 | Strom_C | why not set up a sip phone or something so that you don't have to use your home phone to call into your asterisk box? |
| 19:45.19 | *** join/#asterisk RoyK (n=roy@ti211310a080-4149.bb.online.no) |
| 19:45.32 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 19:53.50 | Stephnie | Strom_C : its for the clients who dont want to have IP Devices... |
| 19:59.13 | *** join/#asterisk dovid (n=Dovid@85.159.160.207) |
| 20:04.42 | *** join/#asterisk tRSS (n=tRSS@124.29.204.122) |
| 20:04.58 | *** join/#asterisk sinisterguy (n=wasted@CPE00131025897e-CM0014045b6464.cpe.net.cable.rogers.com) |
| 20:05.34 | tRSS | I have a user configured with iax. my asterisk behind a firewall and i have setup a nat for the udp port of 4569. i can hear the person calling, but he can't hear me. I am guessing this person is also natted. |
| 20:06.21 | dovid | tRSS: thier nat status should not matter if they are a sip user making call's |
| 20:06.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.14) |
| 20:06.35 | dovid | it could be a bad router |
| 20:06.37 | sinisterguy | can you set up asterisk just for sip. I need to have about 10-20 people dial into a voip conference call |
| 20:06.45 | Strom_C | sinisterguy: yes |
| 20:07.04 | tRSS | nope. they iax user calling a sip user |
| 20:07.06 | sinisterguy | Strom_C: sweet, thanks |
| 20:07.23 | Strom_C | tRSS: diagram the entire call route |
| 20:07.24 | tRSS | i am a sip user. they have iax user configured |
| 20:07.52 | dovid | ok |
| 20:07.54 | Strom_C | tRSS: SIP uses UDP 5060 for signaling and UDP 10000-20000 for media |
| 20:08.12 | dovid | so they connect to u vua IAX and u send the call out via SIP to ur provider ? |
| 20:08.13 | tRSS | User A (iax) calling me thru my asterisk box (behind a firewall and natted), whereas I am sitting on the lan as the asterisk box |
| 20:08.16 | dovid | via* |
| 20:08.41 | dovid | so u should have no NAT issues since u r on a lan |
| 20:08.50 | tRSS | nope |
| 20:08.56 | dovid | Stron_C: have u seen one way audio with IAX ? |
| 20:09.36 | tRSS | i have also have UDP 5060 open on the firewall and natted with my asterisk box. |
| 20:10.12 | Strom_C | tRSS: what about UDP 10000-20000 |
| 20:11.05 | tRSS | i don't have them configured on the firewall. i figured, that no one would be using sip from outside the firewall. only iax. SIP gave me a lot of headache in configuration, hence I decided to use IAX |
| 20:11.27 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 20:11.40 | Strom_C | tRSS: what codec is the client using |
| 20:11.46 | *** part/#asterisk sinisterguy (n=wasted@CPE00131025897e-CM0014045b6464.cpe.net.cable.rogers.com) |
| 20:12.17 | tRSS | ulaw |
| 20:12.42 | Strom_C | do you have "disallow=all" followed by "allow=ulaw" in the client's iax.conf entry? |
| 20:12.49 | tRSS | yes sir! |
| 20:13.22 | tRSS | although, i have also allowed gsm. but myself and the client are negotiating on ulaw |
| 20:13.25 | Strom_C | what happens if the client dials a recording on your box? can they hear the recording |
| 20:13.50 | tRSS | good question. i havent tried that. just give me a minute and i will ask him to do that |
| 20:14.34 | tRSS | i have mostly worked with sip, hence, i am not much experienced with IAX |
| 20:15.37 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 20:17.06 | tRSS | Strom_C: the client can hear the IVR flawlessly |
| 20:17.28 | Strom_C | can you pass audio to the asterisk box |
| 20:17.43 | Strom_C | i.e. can you make a recording on a voicemail box or something |
| 20:17.43 | tRSS | i am not sure, what you mean by that? |
| 20:18.04 | tRSS | not yet, the ivr is not setup like that yet |
| 20:18.18 | Strom_C | well, set up a voicemail box or something and try and record a messae |
| 20:18.21 | Strom_C | er, message |
| 20:18.53 | tRSS | you mean 'i' should record a message and not the client, right? since, i can hear the client, loud and clear? |
| 20:18.57 | Strom_C | yes |
| 20:19.02 | Strom_C | and the client can hear audio |
| 20:19.04 | tRSS | just give me a sec. |
| 20:19.57 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 20:26.24 | *** join/#asterisk loloski (n=hellrace@210.213.199.97) |
| 20:28.43 | tRSS | Strom_C: sorry for the delay. my box has stopped taking dtmf from my softphone. let me just fix this real quick |
| 20:29.12 | loloski | murf: i'll already upload the logs the you have requested |
| 20:29.57 | loloski | that* you have requested |
| 20:33.30 | *** join/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 20:38.41 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:40.38 | *** part/#asterisk loloski (n=hellrace@210.213.199.97) |
| 20:40.52 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:43.55 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 21:00.37 | *** join/#asterisk luke-jr_ (n=luke-jr@CPE-24-31-240-141.kc.res.rr.com) |
| 21:06.26 | tRSS | Strom_C: I, myself, started using IAX and now everything seems to be working fine. DTMF not working was xlite's fault, which I guess, I can't do much about, since I am on a mac, and now this means, I have to find another sip client for mac |
| 21:06.57 | Strom_C | or, alternatively, get a real phone :) |
| 21:07.02 | tRSS | thanks for your help though. I guess, I would still like to have sip working as well, since, a lot of people have SIP clients |
| 21:07.17 | Strom_C | well hang on man |
| 21:07.24 | Strom_C | debug your SIP issue. |
| 21:07.46 | Strom_C | isnt there an sjphone version for mac?> |
| 21:07.58 | tRSS | I would love to, but not tonight! :) I shall return, recharged and revamped tomorrow! |
| 21:08.15 | tRSS | have been working all day (specially on a saturday). |
| 21:08.16 | aydiosmio | Strom_C: yes |
| 21:08.24 | aydiosmio | used it yesterday actually |
| 21:08.28 | aydiosmio | it's junk but it works |
| 21:08.47 | tRSS | xlite crashes after 10 mins of use or so on os x |
| 21:09.24 | Nugget | I've never had that problem. Intel or PPC? |
| 21:09.27 | tRSS | found an IAX client for mac os x. seems simple and good: JackenIAX |
| 21:09.31 | tRSS | PPC |
| 21:09.34 | Nugget | weird. |
| 21:09.49 | tRSS | PB G4. 1GB , 80 GB, 15' |
| 21:09.52 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 21:10.33 | tRSS | anyhow, gotta jet guys. its 2:10 AM here in pakiland. well see you guys tomorrow! |
| 21:10.49 | tRSS | later! :) |
| 21:13.54 | drkludge | can a sip client talk directly to another sip client without using a proxy? |
| 21:14.34 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 21:14.39 | aydiosmio | drkludge: yes |
| 21:14.57 | aydiosmio | most softphones can |
| 21:15.09 | aydiosmio | hardware will depend on the frimware |
| 21:15.50 | drkludge | aydiosmio, do you have a URL? I'll I have seen so far is the Asterisk book which shows xten to asterisk. |
| 21:16.15 | drkludge | aydisomio, Do you just not configure the proxy setting then? |
| 21:16.16 | aydiosmio | what are you using? |
| 21:16.35 | drkludge | I have xten right now on Linux |
| 21:16.43 | aydiosmio | drkludge: you usually set the phone in client to client mode and dial the IP address |
| 21:16.59 | aydiosmio | oh I've never used xten |
| 21:17.02 | drkludge | aydiso, Ahh thank you. |
| 21:17.31 | drkludge | aydiosmio, What client are you using and what platform? |
| 21:17.46 | aydiosmio | I use SJPhone primarily in windows |
| 21:18.12 | drkludge | aydiosmio, Thanks. |
| 21:18.17 | aydiosmio | sure |
| 21:21.29 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:26.39 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 21:30.45 | Choubaka | question: I've got 2 * working |
| 21:30.55 | Choubaka | ans I want to link them together.. how? |
| 21:34.48 | [TK]D-Fender | Choubaka : http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 21:35.36 | Choubaka | thanks |
| 21:37.47 | murf | loloski: many thanks. Outgoing calls seem not to change state in SIP. Are you allowing reinviting? |
| 21:38.25 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 21:38.41 | murf | and loloski is gone! |
| 21:38.46 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:39.32 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:46.28 | *** join/#asterisk dorphalsig (i=jirc@pcsp168-254.supercabletv.net.co) |
| 21:46.31 | dorphalsig | Hi |
| 21:47.35 | dorphalsig | I'm trying to get a VoIP over a WiFi network (802.11n), I wanted to ask a couple of questions... |
| 21:48.05 | dorphalsig | 1) Can I have data / voice on the same network? |
| 21:48.31 | dorphalsig | 2) I read somewhere that 150ms MAX latency ... any comments? |
| 21:48.50 | dorphalsig | 3) Anybody wants to buy me a banana split? xD |
| 21:49.18 | EyeCue | well voip = voice over ip, so yes you can have data and voice on the same network |
| 21:49.43 | EyeCue | 2) you can run voice over any medium, its a matter of reliability and quality |
| 21:50.03 | EyeCue | you might want to research codecs with good loss recovery |
| 21:50.10 | EyeCue | 3) no, im busy |
| 21:50.11 | EyeCue | :] |
| 21:51.19 | aydiosmio | 150 is a general rule you realyl want less than 100ms, but even over wifi you should be able to maintain less than 50ms |
| 21:52.02 | dorphalsig | any suggestions on signal to noise ratios? |
| 21:52.16 | aydiosmio | what about them? |
| 21:52.42 | aydiosmio | loss will depend on the data speed |
| 21:52.46 | dorphalsig | I heard someehereI should try to get above -40dB |
| 21:52.58 | aydiosmio | that's pretty relative |
| 21:53.05 | dorphalsig | I was thinking of doing G729a |
| 21:53.29 | asdx | is there a client (software) for use with asterisk, i mean, for call from one client to another client... |
| 21:53.45 | aydiosmio | dorphalsig: you want a signal that will give you reliable packet delivery |
| 21:53.49 | aydiosmio | you gotta test for that |
| 21:54.10 | EyeCue | asdx, theres plenty, google iax clients |
| 21:54.17 | aydiosmio | asdx: client to client doesn't involve a PBX |
| 21:54.23 | dorphalsig | aydiosmio --> or even Eyebeam |
| 21:54.45 | dorphalsig | oops |
| 21:54.46 | asdx | i see |
| 21:54.51 | dorphalsig | adsx mot aydiosmio |
| 21:54.52 | dorphalsig | :$ |
| 21:54.52 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 21:55.27 | aydiosmio | you can call a client through asterisk sure, but "client to client" is a direct connection |
| 21:55.36 | bluregard | hi all |
| 21:56.02 | dorphalsig | hey blu |
| 21:56.37 | dorphalsig | Another question ... can anybody suggest me a good cheap ATA? (about 20 -30 units?) |
| 21:56.51 | bluregard | hey, I'm looking at the draft for IAX2 and it says it expires 7/5/2005. Is there a more up to date version I should be looking at? |
| 21:56.52 | asdx | i meant client <-> asterisk <-> client |
| 21:57.10 | dorphalsig | asdx -->Eyebeam for SIP |
| 21:57.18 | asdx | ok |
| 21:57.19 | aydiosmio | pretty much any sip softphone will work |
| 21:57.20 | asdx | thanks |
| 21:57.27 | EyeCue | asdx, depends on the protocol(s) the asterisk server is running |
| 21:57.35 | aydiosmio | er |
| 21:57.38 | aydiosmio | not sip |
| 21:57.56 | EyeCue | google '(IAX OR SIP) software clients' |
| 21:58.01 | aydiosmio | specificaly |
| 21:58.24 | dorphalsig | Just go with Eyebeam :P |
| 21:58.41 | EyeCue | asdx, also depends on what client OS youre running |
| 21:58.46 | asdx | and you can monitorear everything in asterisk? all the calls that are passing trough it, etc? |
| 22:12.02 | dorphalsig | I'mm off thnx |
| 22:24.58 | sevard | asdx: yes. |
| 22:26.02 | asdx | wow, awesome |
| 22:26.42 | asdx | is possible to call from one pbx to another one? |
| 22:27.08 | sevard | of course |
| 22:27.24 | asdx | wow :) |
| 22:27.34 | bluregard | asdx: that's kind of the idea |
| 22:27.48 | asdx | i see |
| 22:28.17 | bluregard | asdx: do you have an * machine set up? |
| 22:28.19 | EyeCue | asdx, IAX = inter asterisk exchange |
| 22:29.06 | asdx | bluregard: no but i want to set up one |
| 22:29.22 | asdx | EyeCue: i see, IAX is the protocol that asterisk uses? |
| 22:29.28 | bluregard | asdx: the only way to truly appreciate the abilities of * is to set it up and play with it. |
| 22:29.31 | asdx | iax, sip are protocols right? |
| 22:29.36 | bluregard | yes |
| 22:29.43 | *** join/#asterisk adorah (n=admin@84.94.124.95.cable.012.net.il) |
| 22:29.59 | EyeCue | asdx, yup |
| 22:30.08 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 22:30.08 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 22:30.08 | EyeCue | among others, but yeh |
| 22:30.09 | asdx | alright |
| 22:30.11 | bluregard | sip is an official protocol, IAX is draft but quite functional |
| 22:30.38 | asdx | i will install asterisk now |
| 22:30.49 | asdx | and what do i need after that, hardware? |
| 22:31.05 | bluregard | asdx: get a machine setup and a copy of the O'Reilly book on * |
| 22:31.12 | asdx | ok |
| 22:31.30 | bluregard | all you really need is to start playing with it is a network card and a softphone |
| 22:31.31 | asdx | is possible to only use software first? |
| 22:31.38 | bluregard | yeah |
| 22:31.43 | asdx | wow! nice :D |
| 22:31.45 | bluregard | that's what I'm doing right now |
| 22:32.18 | bluregard | I have soft phones set up on a few of my PC's that I use to test. |
| 22:32.28 | EyeCue | yah, ti smuch fun |
| 22:32.29 | asdx | cool |
| 22:32.31 | Qwell | Strom_C: ping |
| 22:32.42 | bluregard | I found a cheap (in every sense of the word) DID provider so I can test in/outbound calling too. |
| 22:32.59 | asdx | did provider? |
| 22:33.00 | Strom_C | Qwell: pong |
| 22:33.04 | Qwell | ! |
| 22:33.20 | Qwell | Strom_C: 8 letters, telephony device, first 4 letters are "TelT" |
| 22:33.27 | Qwell | name that company |
| 22:33.34 | bluregard | Direct Inbound Dial. It's just a phone number. |
| 22:33.44 | asdx | i see |
| 22:33.50 | Qwell | something that looks something like a modem (but isn't) - I think it connects TO CPE |
| 22:33.54 | Strom_C | TelTones? I don't know |
| 22:34.01 | Qwell | nope...don't think so |
| 22:34.12 | Strom_C | bluregard: Direct Inward Dial |
| 22:34.17 | Strom_C | INWARD, not inbound |
| 22:34.25 | Qwell | and, heh |
| 22:34.33 | bluregard | sorry |
| 22:34.40 | Qwell | I just bought a bright ugly ass orange Northern Telecom rotary wall phone... |
| 22:34.42 | Qwell | $3 |
| 22:34.51 | bluregard | I knew that wasn't right |
| 22:35.11 | Strom_C | ooh |
| 22:35.12 | Strom_C | link |
| 22:35.15 | Qwell | sec |
| 22:36.22 | asdx | bluregard: in what distro do you have asterisk installed? |
| 22:36.34 | Qwell | odd, google isn't returning much |
| 22:36.38 | bluregard | debian |
| 22:36.46 | asdx | ok |
| 22:36.53 | Strom_C | Qwell: is the dial in the handset? |
| 22:36.59 | bluregard | no wait. sorry, this one is ubuntu |
| 22:37.06 | Qwell | no |
| 22:37.20 | Qwell | it says "NE 228 A" on it |
| 22:37.24 | Qwell | but, nothing on google |
| 22:37.29 | bluregard | computers are taking over my house. I can't keep them straight anymore. |
| 22:37.50 | Strom_C | it doesn't look like this, does it? |
| 22:37.50 | Strom_C | http://www.bellsystemmemorial.com/images/1956_wall_telephone.jpg |
| 22:37.59 | Qwell | http://cgi.ebay.com/VINTAGE-ORANGE-ROTARY-DIAL-WALL-PHONE-telephone_W0QQitemZ140030993301QQcmdZViewItem |
| 22:38.12 | Qwell | umm, yep, but oranger |
| 22:38.20 | Qwell | WAY oranger, heh |
| 22:38.23 | asdx | can you put asterisk in the router, using openwrt for example? and connect a phone to it? |
| 22:38.54 | Strom_C | Qwell: awesome |
| 22:38.57 | Qwell | but, $3...hell yes |
| 22:39.08 | bluregard | asdx: no idea. Give it a try and report back |
| 22:39.10 | Qwell | lady was like "I can't guarantee it'll work - it's not even got a modern plug!" |
| 22:39.23 | Qwell | thrift stores++, heh |
| 22:39.29 | asdx | bluregard: ok |
| 22:39.37 | Qwell | (it does have RJ11, it's just inverted) |
| 22:39.39 | CharlesR | Qwell that's a 554 and it's an excellent phone |
| 22:39.49 | Qwell | it says 228 on it |
| 22:39.55 | Qwell | on mine, that is |
| 22:40.09 | Qwell | handset is a bit lighter than you'd expect though |
| 22:40.12 | bluregard | I'm still too new to * to know all the in's and out's. I'm just here to absorb. |
| 22:40.19 | Qwell | Strom_C: I'll bring it in on Thursday, if you're still coming |
| 22:40.29 | Strom_C | i certainly am |
| 22:40.33 | Strom_C | i'm drooling already :) |
| 22:40.37 | CharlesR | and notice the ridges on the left and right edges of the top - those and meant to rest the handset on when needed |
| 22:40.44 | Strom_C | btw, what should I pack? long-sleeve shirts and a coat? |
| 22:40.47 | CharlesR | few remember that detail of the 554 |
| 22:41.00 | Strom_C | CharlesR: I'm well aware of that detail of the 554 :) |
| 22:41.12 | Qwell | Strom_C: and shorts, heh |
| 22:41.17 | Qwell | Strom_C: it's been hot and cold lately |
| 22:41.19 | CharlesR | Strom_C i meant qwell might not know it |
| 22:41.22 | bluregard | are any of you from around the Chicago area? |
| 22:41.25 | Qwell | huh? |
| 22:41.31 | *** part/#asterisk clive- (n=pirch@dsl-242-165-63.telkomadsl.co.za) |
| 22:42.12 | Qwell | ohhh, I see |
| 22:43.38 | Qwell | What's the REN on this SOB? |
| 22:43.49 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 22:43.53 | CharlesR | Qwell it's 1.0 |
| 22:44.33 | Qwell | oh |
| 22:44.54 | Qwell | now how do I go about opening it? ;/ |
| 22:45.03 | bluregard | hey with the REN, you can only have 5.0 on each FXS right, not the whole system. |
| 22:45.11 | Strom_C | Qwell: there's a tab on the bottom |
| 22:45.17 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 22:45.21 | Qwell | oh |
| 22:45.35 | asdx | when i'm calling with a normal telephone to another telephone, that call is going through a pbx system right? |
| 22:45.43 | Strom_C | push the tab in and the thing should just lift off |
| 22:45.59 | Strom_C | asdx: regular residential telephones? |
| 22:46.22 | asdx | Strom_C: yes |
| 22:46.28 | Strom_C | asdx: no, you're not using a PBX |
| 22:46.32 | bluregard | regular residential just goes through the phone company's switch right? |
| 22:46.37 | Strom_C | you're using a class 5 end office switch |
| 22:47.07 | asdx | Strom_C: i see |
| 22:47.07 | Strom_C | PBX stands for "private branch exchange" |
| 22:47.11 | *** join/#asterisk knobenheimer (n=brettkoe@c-71-205-120-82.hsd1.mi.comcast.net) |
| 22:47.16 | CharlesR | i have a real PBX here at home but it is not in use (and won't be) |
| 22:47.16 | asdx | ok |
| 22:47.18 | Qwell | Strom_C++ |
| 22:47.32 | Qwell | boy that's a lot of wires :D |
| 22:47.44 | Strom_C | CharlesR: so it's a step PBX? :D |
| 22:48.03 | asdx | Strom_C: what about with cellphones systems? |
| 22:48.13 | CharlesR | no it's a NE/NT SG1A that i rescued from being junked |
| 22:48.20 | Strom_C | asdx: mobile phones don't use PBXes either |
| 22:48.46 | Strom_C | mobile phones also use class 5 central office switches |
| 22:48.52 | Qwell | 11/1969 |
| 22:49.01 | CharlesR | i'd like to find a WE 755A xbar PBX |
| 22:49.01 | Strom_C | Qwell: yum |
| 22:49.03 | bluregard | isn't Nextel's iDEN a form of VoIP? |
| 22:49.16 | bluregard | for direct connect |
| 22:49.42 | asdx | does google uses a PBX for his google talk? |
| 22:50.09 | file | bluregard: iDEN was designed for direct connect, doesn't exactly use VoIP... but the CDMA/GSM providers who do the direct connect-like thing use VoIP |
| 22:50.39 | bluregard | file: like Cingular's push-to-talk crap that never works? |
| 22:50.49 | file | bluregard: aye |
| 22:51.08 | bluregard | gotcha |
| 22:57.25 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 22:58.16 | *** join/#asterisk bobloblian (n=bob@net-252-14.northwestel.net) |
| 22:59.32 | *** join/#asterisk Supaplex (n=rfc1413@shell.aros.net) |
| 22:59.37 | *** part/#asterisk frenzy (n=frenzy@196.46.104.30) |
| 23:07.57 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 23:09.44 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 23:11.36 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 23:14.59 | Qwell | gee, you'd think a phone guy would own at least 1 rj11 cable |
| 23:15.51 | EyeCue | that would make too much sense |
| 23:15.58 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 23:16.32 | Qwell | seems wasteful to destroy a 50' cable for...6" |
| 23:18.02 | Qwell | stupid 1960s technology |
| 23:18.08 | *** part/#asterisk Eminence (n=achin@Sears-47.resnet.brown.edu) |
| 23:18.12 | Strom_C | 1960s? |
| 23:18.19 | hmmhesays | the 60's rocked man |
| 23:18.25 | hmmhesays | peace luv n shit |
| 23:18.35 | Strom_C | that phone was made in 1969, before the modular plug was invented |
| 23:18.44 | hmmhesays | show me you boobies |
| 23:18.55 | Strom_C | you're supposed to hard-wire it in, qwell :) |
| 23:19.16 | Qwell | it has a modular plug |
| 23:19.32 | Strom_C | is the modular plug on a plate that's screwed into the back? |
| 23:19.32 | Qwell | well, a jack |
| 23:19.52 | Qwell | it's a little plastic dealie that slides in the metal |
| 23:20.00 | Qwell | the handset, however, is hard wired |
| 23:20.43 | Qwell | brb |
| 23:21.30 | *** join/#asterisk asdx (n=ubuntu@200.61.236.33) |
| 23:21.58 | hmmhesays | slow day in irc |
| 23:22.07 | EyeCue | thank fuck for that |
| 23:22.10 | EyeCue | scsue my french |
| 23:22.22 | bluregard | can * do pulse dialing? |
| 23:22.52 | bluregard | rather can * hear pulse dialing from a rotary phone. |
| 23:23.22 | hmmhesays | buy a new phone |
| 23:23.53 | bluregard | yeah I'll keep that in mind next time I'm at Best Buy. |
| 23:24.16 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 23:24.30 | hmmhesays | walmart, target, kmart, walgreens, sears |
| 23:24.37 | bluregard | I was just wondering. I guess I'll keep the curiosity to a minimum |
| 23:25.11 | hmmhesays | as far as I know asterisk can't listen for pulse dialing |
| 23:26.01 | bluregard | I was wondering if * did it or if it was done in the TDM card |
| 23:26.15 | hmmhesays | tdm card might |
| 23:26.31 | hmmhesays | quick google search would probably tell you |
| 23:26.45 | bluregard | not like I'll ever plug a pulse dial phone in but you never know. |
| 23:27.03 | hmmhesays | http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing |
| 23:29.19 | bluregard | good. when machines take over the world and remove all of our phones I can still tap out my 900 numbers with some cat5 and a 9V. |
| 23:29.49 | EyeCue | laf |
| 23:30.00 | hmmhesays | dude you're not l33t if you can't whistle dtmf |
| 23:32.11 | Qwell | Strom_C: Do these NEED 4 wires, or will 2 do? |
| 23:32.49 | Qwell | I found a phone cord, but it's only 2 wires :D |
| 23:35.23 | bluregard | I can't whistle period |
| 23:37.45 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 23:40.48 | *** join/#asterisk shodan (n=shodan@ip210.99-113-216.pppoe4.joliette.intermonde.net) |
| 23:41.02 | *** join/#asterisk rustyb (i=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 23:41.41 | shodan | hmm , I just had a problem with a SPA-3k2 that made a user comment that my phone system was the "worst mistake ever bought" :( |
| 23:41.43 | shodan | damn |
| 23:42.34 | shodan | the spa-3k2 didn't hangup and a call was terminated , tying up the line for hours :\ |
| 23:45.18 | shodan | s/and/when |
| 23:45.39 | bluregard | shodan: do you know why? |
| 23:46.14 | shodan | I'm still reading the logs but I can't find out why , that's the worst, if I can't find out why then the system really is unreliable |
| 23:49.18 | Strom_C | Qwell: ? |
| 23:49.32 | Strom_C | which four wires? handset? |
| 23:52.00 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 23:53.42 | Qwell | Strom_C: the main unit |
| 23:53.56 | Qwell | this would be MUCH easier if I had wire strippers, heh |
| 23:54.36 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 23:56.17 | Strom_C | Qwell: tip connects to the terminal marked L1 |
| 23:56.22 | Strom_C | ring connects to L2 |
| 23:56.25 | Strom_C | that's it |
| 23:56.31 | Qwell | it's colored, but there are 4 |
| 23:56.40 | Strom_C | on the plug? |
| 23:56.43 | Supaplex | does asterisk maintain a modern stable release of asterisk for debian sarge? is backports.org suggested or not? |
| 23:56.44 | Qwell | yeah |
| 23:57.03 | Strom_C | red connects to L2 and green connects to L1 I'm sure |
| 23:57.07 | Strom_C | what about black and yellow? |
| 23:58.02 | Qwell | uhh... G |
| 23:58.19 | Strom_C | yeah, that's just ground |
| 23:58.21 | Strom_C | dont worry about that |
| 23:58.26 | Qwell | ok |
| 23:58.37 | Strom_C | all you need for a single-line telephone is tip and ring |
| 23:58.37 | Corydon76-home | Heh, no ground? |
| 23:59.02 | Strom_C | Corydon76-home: well, unless he's got groundstart service or something, which I doubt is the case :) |
| 23:59.16 | Supaplex | black and yellow are for a second line, or a spare set if somethings wrong with the first set (and you don't/cannot pull a new cable) |
| 23:59.23 | Corydon76-home | Does he have a ground start telephone, though? |
| 23:59.33 | Strom_C | Corydon76-home: I'm going to assume he doesn't |
| 23:59.35 | Qwell | Corydon76-home: it's going on an iaxy, heh |
| 23:59.44 | Corydon76-home | Qwell: LOL |
| 23:59.58 | Qwell | Corydon76-home: 1960s Northern Telecom, bright ass orange rotary :) |
