| 00:00.11 | [TK]D-Fender | Faiz: Do yourself a favour and just re-dl and compile the complete one |
| 00:00.18 | SaiSoma | yea, what fender said |
| 00:00.22 | svm_invictvs | Hm. |
| 00:00.24 | Faiz | ok, should i uninstall my current version? |
| 00:00.25 | SaiSoma | it won't overwrite your config |
| 00:00.27 | SaiSoma | nope |
| 00:00.29 | svm_invictvs | What's everybody's thoughs on Teliax? |
| 00:00.33 | Faiz | ok |
| 00:00.46 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 00:00.54 | Faiz | side question: |
| 00:01.02 | SaiSoma | Bene Tleilax? |
| 00:01.07 | SaiSoma | ;) |
| 00:01.19 | Faiz | say i wanted to downgrade to 1.6.0.6, how would i go about doing this? |
| 00:01.24 | SaiSoma | same thing |
| 00:01.25 | Faiz | svn the version tag? |
| 00:01.27 | [TK]D-Fender | svm_invictvs: they're OK. Sometimes spotty, but a good dead for a bunch of things |
| 00:01.38 | SaiSoma | Faiz: compile, shutdown asterisk, install, start asterisk |
| 00:01.39 | svm_invictvs | Yeah. |
| 00:01.44 | svm_invictvs | [TK]D-Fender: Their prices are good. |
| 00:01.49 | Faiz | ah, ok, cool. |
| 00:01.50 | [TK]D-Fender | Faiz: Just go onto the digium server and download that specific version |
| 00:01.56 | *** part/#asterisk nixer (n=nixer@78.154.216.86) |
| 00:01.57 | Faiz | gotcha, thanks :) |
| 00:02.11 | SaiSoma | Faiz: how's yoru project coming? Your prof happy with it so far? |
| 00:02.24 | Faiz | agh, things were fairly good, but i have to implement fax right now |
| 00:02.44 | Faiz | and the latest versions of asterisk, 1.6.1 onward are terrible with it |
| 00:03.06 | Faiz | i'm just worried about what i'll be doing next since i'm already overwhelmed |
| 00:03.26 | Faiz | c'est la vie |
| 00:03.46 | SaiSoma | Faiz: how are they terrible with it? I'm using 1.6.1.0 and a handytone 486 ATA with no problems? |
| 00:03.52 | Faiz | and women! agh, women women women women women, a never ending tale of ambivalence |
| 00:03.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:03.59 | SaiSoma | not using t.38, just passthrough and no problems at all |
| 00:04.06 | Faiz | for some reason, 1.6.1.1 doesn't seem to be registering the fax modules |
| 00:04.24 | SaiSoma | and 1.6.0.6 did? |
| 00:04.24 | Faiz | when i downloaded the specific ones from the digium server and put them in the modules directory |
| 00:04.40 | Faiz | someone earlier in here said it worked with 1.6.0.6, so i was going to try that |
| 00:04.42 | SaiSoma | oh, you're doing something fancy, not normal analog fax machine stuff:) |
| 00:04.43 | Faiz | but my internet died |
| 00:05.03 | Faiz | oh no, its analog fax |
| 00:06.15 | SaiSoma | really? Are you using a sip ata adapter or something else? |
| 00:06.38 | Faiz | as of right now, i don't have a fax machine set up, since i wanted to make sure the fax modules were read by asterisk before i proceeded |
| 00:06.50 | Faiz | but it would be a standard fax machine hooked into the PCI card |
| 00:07.02 | Faiz | using "fax over asterisk" |
| 00:07.07 | SaiSoma | you know, maybe I'm doing it wrong, but I treat my fax like an analog extension |
| 00:07.14 | SaiSoma | and it's worked perfectly for a few months now |
| 00:07.32 | Faiz | ah |
| 00:08.00 | Faiz | that's great, hopefully i'll be able to make some progress by the end of tonight |
| 00:08.56 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 00:09.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 00:09.25 | buttons840 | in the dialplan, can i reference multiple variable in the same line IE. someapp(${var1}{var2}) what is the syntax? |
| 00:10.23 | buttons840 | reads the book |
| 00:10.33 | SaiSoma | buttons840: I've done this like somaapp(${var1}${var2}) |
| 00:10.42 | SaiSoma | may not be the only way, but it has worked for me in macros |
| 00:10.50 | buttons840 | i'll try it |
| 00:10.51 | buttons840 | thx |
| 00:10.54 | SaiSoma | *nod* |
| 00:23.33 | Faiz | i ran "make" under 1.6.0.6, and I get the following output: |
| 00:23.59 | Faiz | erm, i get an error as it compiles app_dahdiras.o |
| 00:24.12 | Faiz | "/usr/include/dahdi/user.h:766: error: expected specifier-qualifier-list before â__s32â" |
| 00:25.10 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 00:26.59 | SaiSoma | this a fresh download of 1.6.0.6? |
| 00:28.13 | Faiz | yes, but forget it, i'm going to try 1.6.1.0 since it seemed to be an issue with that branch and was resolved in a later release |
| 00:30.29 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 00:32.44 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 00:34.04 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 00:37.32 | *** join/#asterisk [D-Ster-] (n=ster@pool-70-109-24-51.atl.dsl-w.verizon.net) |
| 00:50.06 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM] |
| 00:50.32 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) [NETSPLIT VICTIM] |
| 00:51.58 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM] |
| 00:51.58 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:52.25 | *** join/#asterisk subl (n=sublime@2001:470:1f0f:da:224:1dff:fe1e:15c) |
| 00:52.25 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM] |
| 00:52.34 | *** join/#asterisk Corydon76-dig (i=four@pdpc/supporter/bronze/Corydon76-home) |
| 00:52.34 | *** join/#asterisk snowboarder04 (n=un@94-193-80-153.zone7.bethere.co.uk) [NETSPLIT VICTIM] |
| 00:52.34 | *** join/#asterisk [Outcast] (n=asdf@ip202-27-193-197.cust.kci.net.nz) [NETSPLIT VICTIM] |
| 00:52.34 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) [NETSPLIT VICTIM] |
| 00:52.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 00:52.35 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 00:52.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk macko (n=yoavw@corky.net) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) [NETSPLIT VICTIM] |
| 00:52.35 | *** join/#asterisk korcan (n=korcan@99.23.50.73) [NETSPLIT VICTIM] |
| 00:52.36 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) [NETSPLIT VICTIM] |
| 00:52.36 | *** join/#asterisk AdvoWork (n=AdvoWork@unaffiliated/advowork) [NETSPLIT VICTIM] |
| 00:52.36 | *** join/#asterisk RaNaji (n=WinNT@rrcs-76-79-99-213.west.biz.rr.com) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk dongs (n=lol@l212168.ppp.asahi-net.or.jp) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 00:52.46 | *** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk MT`AwAy (n=MagicalT@shigoto.ookoo.org) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 00:52.46 | *** join/#asterisk NoxIn- (n=noxin@zorlit.org) [NETSPLIT VICTIM] |
| 00:52.46 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) [NETSPLIT VICTIM] |
| 00:52.46 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 00:55.27 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com) |
| 00:56.24 | *** join/#asterisk Hatrix (n=Hatrix@62.43.132.100.dyn.user.ono.com) |
| 01:04.06 | *** join/#asterisk metfan2007 (n=metfan@189.146.149.29) |
| 01:04.21 | metfan2007 | hi all |
| 01:08.14 | metfan2007 | i have a question, i want to do an IAX trunk between 2 ast boxes, but in one side I cannot forward any ports to the ast box, and the ISP does not provides any special IP to that router, is a shared external IP, how can I get around this? I can make calls form ast box A to box B, (box B with no problems, fixed external IP and port forwarded) but no in the other direction |
| 01:12.43 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 01:14.00 | thansen | does http://www.voip-info.org/wiki/view/record+file not set the RECORDED_FILE variable? |
| 01:15.24 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 01:17.34 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 01:21.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-xltgaqqbzmuvymoh) |
| 01:25.43 | *** join/#asterisk jtodd (i=ecskzlcb@ns.fox-den.com) |
| 01:25.43 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:30.56 | *** join/#asterisk NaPs (n=Unknown@tecknet.org) |
| 01:34.38 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 01:35.15 | [TK]D-Fender | metfan2007: run a vpn between them originated & maintained by the troublesome side |
| 01:37.14 | [TK]D-Fender | thansen: What variable? That page makes no such reference |
| 01:38.37 | thansen | [TK]D-Fender: the RECORDED_FILE variable like Record() would do |
| 01:39.01 | metfan2007 | [TK]D-Fender: OK, I can try that, does the register => line helps? maybe register the trouble side to the "good" one? |
| 01:40.36 | *** join/#asterisk Kumbang (n=lebah@167.205.24.69) |
| 01:42.49 | KavanS | I have call forwarding setup...but sometimes it would be nice to know who is calling...does anyone have a suggestion to change the outgoing callerid to the number that's calling me? |
| 01:42.55 | KavanS | by legal means of course... |
| 01:43.11 | KavanS | aren't their some voip services that you can specify something like that? |
| 01:43.49 | *** join/#asterisk NaPs (n=Unknown@82.67.228.63) |
| 01:43.56 | *** join/#asterisk apeiron (i=apeiron@isuckatdomains.net) |
| 01:46.27 | *** join/#asterisk QaDeS (n=mklaus@p4FC728A8.dip0.t-ipconnect.de) |
| 01:54.01 | *** join/#asterisk ethicx (n=ethicx@adsl-10-114-195.mia.bellsouth.net) |
| 01:56.04 | *** join/#asterisk ingenius (n=alektro@186.136.0.175) |
| 01:56.06 | ethicx | Hei folks. I'm having this interesting problem. Asterisk Server is behind NAT (port forwarding), client sip is outside behind another NAT. The client successfully registers with the server. When I make a call from client side it rings on the other end but I don't get audio. Just when Im placed on hold Im able to hear something. does this sound like a NAT problem to you guys? |
| 01:56.17 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 01:59.50 | ethicx | anyone? |
| 02:03.36 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:14.04 | *** join/#asterisk kihote (n=chatzill@118.69.75.193) |
| 02:14.50 | [TK]D-Fender | thansen: Well it doesn't say it does that. So it won't. Besides YOU named the file when you started the recording. You should already know it |
| 02:15.03 | [TK]D-Fender | ethicx: READ : |
| 02:15.05 | [TK]D-Fender | ~sipnat |
| 02:15.06 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:15.22 | [TK]D-Fender | ethicx: And yes, it certainly does. |
| 02:17.01 | ethicx | I think I've read those guides like 3 times by now =D but still breaking my head over it...thx though |
| 02:19.29 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 02:21.02 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 02:21.16 | thansen | [TK]D-Fender: yes, thanks...I just wanted to access that variable POST execution of the script, but I'll just have to set another variable in the script with file name |
| 02:22.19 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:23.59 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 02:30.00 | *** join/#asterisk [Outcast] (n=asdf@ip202-27-193-197.cust.kci.net.nz) |
| 02:37.30 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 02:42.10 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:45.23 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 02:48.46 | *** join/#asterisk sah-work (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net) |
| 02:53.58 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 03:11.42 | *** join/#asterisk xpot-mobile (n=james@173-14-225-198-Utah.hfc.comcastbusiness.net) |
| 03:14.14 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 03:30.02 | joako | How can my system uptime be 20 hours but asterisk claim it is up 1 day 5 hours? |
| 03:39.46 | *** join/#asterisk SoCal (n=ster@pool-70-109-24-51.atl.dsl-w.verizon.net) |
| 03:40.48 | SoCal | ~pbx |
| 03:40.48 | infobot | pbx is probably a Private Branch eXchange |
| 03:41.24 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 03:44.37 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 03:49.11 | *** join/#asterisk elfguy516 (n=elfguy51@32.136.160.104) |
| 03:52.42 | *** part/#asterisk elfguy516 (n=elfguy51@32.136.160.104) |
| 04:06.54 | *** join/#asterisk kihote (i=76454bc1@gateway/web/freenode/x-aghykynfejcfobmx) |
| 04:08.18 | *** join/#asterisk jtodd (i=b8krfmt0@ns.fox-den.com) |
| 04:08.18 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 04:11.38 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:16.30 | *** join/#asterisk moy (n=moy@67.201.69.2) |
| 04:19.13 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 04:19.37 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 04:22.11 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 04:32.03 | *** join/#asterisk kihote (i=76454c3a@gateway/web/freenode/x-rypqxnyappqrvgas) |
| 04:35.38 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 04:37.07 | zchaos | Time is 12:36am, computer has been up for 1w 4d 1h 3m 10s |
| 04:37.52 | kihote | Time |
| 04:47.37 | *** join/#asterisk _gm (n=_gm@203.215.176.22) |
| 04:47.43 | *** join/#asterisk Thazza (n=me@ntechm.lnk.telstra.net) |
| 04:54.45 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 05:06.41 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 05:13.11 | *** join/#asterisk K3rN3L (n=dam@infapen.com) |
| 05:13.27 | K3rN3L | good night from Mèxico everybody |
| 05:13.32 | K3rN3L | some body help me? |
| 05:13.55 | K3rN3L | how i can do a message repeat and repeat and repet in asterisk? |
| 05:14.00 | K3rN3L | for example |
| 05:14.07 | K3rN3L | Hello Hello Hello Hello Hello |
| 05:15.05 | denon | show application goto |
| 05:15.05 | denon | (that's for you, K3rN3L) |
| 05:16.50 | K3rN3L | check denon |
| 05:18.42 | [TK]D-Fender | denon only accepts CASH |
| 05:19.32 | K3rN3L | se [TK]D-Fender hi dude |
| 05:30.36 | denon | or paypal! |
| 05:32.27 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 05:33.03 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 05:33.04 | Rob3Rt | Time |
| 05:33.06 | Rob3Rt | Thu Aug 06 15:32:53 2009 |
| 05:33.08 | Rob3Rt | :p |
| 05:33.25 | denon | gday mate |
| 05:46.54 | *** join/#asterisk sah-work (n=Bawbatos@99-40-7-14.lightspeed.sntcca.sbcglobal.net) |
| 05:47.39 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 05:49.26 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 05:53.46 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/x-tiplibxahjqrites) |
| 05:55.55 | *** join/#asterisk manxpower (n=eric@69.73.94.162) |
| 05:57.26 | *** join/#asterisk uruhux (n=uruhu@ip01-140.dsl.i-set.ru) |
| 05:57.51 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
| 05:59.38 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 06:03.01 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 06:03.46 | fiddur | Hi. Anyone here uses php for AMI? The AsteriskManager package on pear doesn't handle events, and the link to StarAstAPI (from http://www.voip-info.org/wiki/view/Asterisk+manager+Examples ) is broken. I finally found the package on web.archive.org, and it's very much better... but I don't know how to contact the author, since the domain starutilities.com obviously is dropped and he had his email address there... Anyone knows of S. A. Kamran? |
| 06:06.06 | *** join/#asterisk J4zen (n=j4zen@95-36-106-201.dsl.alice.nl) |
| 06:07.52 | J4zen | Hi there, i've got a simpel question.. im pretty sure this isn't possible but here goes anyway: Is it possible to send an incoming fax to two addresses, for example; It first goes to Hylafax and then has to go to an actual faxmachine. This way you should get a digital log of the fax as well as a hardcopy at the faxmachine. |
| 06:07.55 | J4zen | Any way to achieve this? |
| 06:09.52 | fiddur | J4zen: shouldn't be any problem.. One solution would be to recieve it in asterisk, and then send it to the two destinations... |
| 06:10.36 | fiddur | otoh, if you recieve it in asterisk, you probably wouldn't need hylafax; just copy the recieved file |
| 06:13.13 | *** join/#asterisk TimRiker (i=timr@bzflag/projectlead/TimRiker) |
| 06:14.06 | TimRiker | what's the status on ipv6 with asterisk? (sip in particular) |
| 06:14.56 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 06:15.00 | manxpower | TimRiker: it should be listed in the UPGRADE file in the 1.6.x source directory |
| 06:15.54 | TimRiker | k. currently on 1.4.x, reading the changelogs on the web site and didn't see it mentioned there. perhaps I just have not found it yet. :) |
| 06:18.12 | manxpower | TimRiker: I doubt 1.4 has any of that support. 1.6 is what you want to look at |
| 06:18.29 | manxpower | If it's not listed there then it's not there. |
| 06:18.47 | J4zen | fiddur: If you sent it to two destinations, will it actually open two channels for the incoming/outgoing audio? |
| 06:18.52 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 06:23.37 | *** join/#asterisk xrmx__ (n=rm@87.13.48.2) |
| 06:24.53 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 06:33.51 | *** join/#asterisk Defraz (n=tim@c72co-edge-router.fuzecore.com) |
| 06:38.30 | TimRiker | I can't seem to find anything that says that ipv6 is supported. :( I see on the asterisk-video list that various things with video are broken in 1.6 too. |
| 06:38.42 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 06:39.53 | Qwell | TimRiker: there is a branch with support for IPv6, but it's rather out of date |
| 06:40.50 | Qwell | TimRiker: That's something I'm personally really interested in it, but haven't been able to find the time to look at it at all... I want it to be on the top of my list. ;/ |
| 06:40.53 | TimRiker | Qwell: you mean the http://www.asteriskv6.org/cgi-bin/moin.cgi/AsteriskIPv6 stuff, yes? |
| 06:41.19 | Qwell | yeah |
| 06:41.38 | Qwell | there's a branch in the asterisk svn though |
| 06:41.59 | TimRiker | would seem that with linphone and a few others supporting ipv6, sip support should not be all that hard to do. |
| 06:42.47 | TimRiker | is the branch in svn still at 1.4 or as someone updated it? |
| 06:42.54 | Qwell | it would be trunk |
| 06:42.56 | TimRiker | s/as/has/ |
| 06:43.09 | Qwell | I have no idea how far out of date it is though |
| 06:43.26 | TimRiker | nods |
| 06:43.40 | Qwell | http://svn.digium.com/svn/asterisk/team/group/v6/trunk/ maybe? |
| 06:44.42 | fiddur | J4zen: If you first recieve it with RecieveFax, and then send it with SendFax twice, yes. http://www.voip-info.org/tiki-index.php?page=Asterisk T.38 ...off course, that's easiest if you're using a 1.6.X asterisk |
| 06:45.48 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:50.12 | J4zen | fiddur: Thanks |
| 06:50.48 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:52.32 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:53.30 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 07:05.12 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.69) |
| 07:15.05 | *** join/#asterisk agx (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:15.54 | agx | i've a Digium TE card PRI at span 2 the timing value in zaptel.conf should be 1 or 2 ? i mean: span=2,1,0 or span=2,2,0 ? span=1 is an FXO card instead |
| 07:21.34 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 07:23.51 | *** join/#asterisk justanothercoder (n=devilz@121.243.137.253) |
| 07:30.10 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:32.07 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 07:32.13 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:40.26 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 07:41.24 | *** join/#asterisk DarkRift (n=dark@65.92.165.121) |
| 07:45.15 | *** join/#asterisk Kumbang (n=ahoy@rusnas.paume.itb.ac.id) |
| 07:57.11 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:00.35 | *** join/#asterisk Psychobilly (n=moi@adsl176-124.kln.forthnet.gr) |
| 08:01.56 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:02.41 | *** join/#asterisk ascent (n=ascent@schoot.org) |
| 08:03.31 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 08:05.15 | ascent | if I have two SIP (hard)phones behind (different) nats, trying to call eachother, and asterisk on a non-natted internet host, what do I need to do to make it work over NAT ? |
| 08:05.43 | ascent | Do I need configchanges for asterisk, on the phone, on the nat routers? |
| 08:12.52 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:21.11 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
| 08:25.10 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 08:27.55 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:29.30 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 08:42.01 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-29.twcny.res.rr.com) |
| 08:43.57 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a1d1:3adb:cbb9:73c1) |
| 08:46.41 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 08:48.22 | *** join/#asterisk Zylogue (n=L4SMB@ip68-12-149-243.ok.ok.cox.net) |
| 08:50.35 | Zylogue | I am a bit confused on what hardware I need for asterisk. I am only wanting to set up a basic SOHO solution to provide a few extensions (only two phones in the house), and voicemail for 5 people. At this time I have one POTS line into the house. Is it needed? |
| 08:52.18 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:56.32 | kaldemar | Zylogue: only if you want PSTN connection using the line |
| 08:58.02 | Zylogue | at this time, I can only get one POTS line into the house, but I would really like to have the ability to support two seperate phone numbers for the house. One for business and one for family/personal. |
| 08:58.34 | Zylogue | what is a good way to handle that? I'm in Oklahoma, US, so providers may be different. |
| 08:59.01 | defsdoor | Zylogue: IVR |
| 08:59.16 | box2 | you wouldn't need much in the way of powerful computer for a setup like that |
| 08:59.27 | Zylogue | defsdoor, yes, an IVR is one of the items I will want to set up. |
| 08:59.37 | box2 | you could use a Linksys router to handle it |
| 08:59.57 | box2 | except for your line cards |
| 09:00.02 | Zylogue | box2, great. From what I have been reading I will need IP phones? |
| 09:00.32 | box2 | you can use analog phones if you get a card with FXS slots |
| 09:00.44 | Zylogue | box2, 'line cards'? are those used to interface the computer with physical phones? |
| 09:00.44 | box2 | s/slots/jacks |
| 09:01.01 | Zylogue | FXS jacks, OK. |
| 09:01.51 | box2 | if you want IP phones, they just connect them to your network, no extra hardware needed on your * server end |
| 09:02.17 | Zylogue | box2, that is good to hear. That will let me add phones in other rooms, as needed. |
| 09:02.19 | mort_gib | Plus, they give much more functionality |
| 09:02.50 | Zylogue | now for a hard question: who do I get phone numbers from? Especially if I am wanting to escape the local telco? |
| 09:03.14 | mort_gib | Where are you based?? |
| 09:03.29 | Zylogue | $55.00/month to a single phone line with no 'features' is way too expensive. |
| 09:03.33 | Zylogue | Oklahoma, US |
| 09:04.06 | Zylogue | but I have cable internet at 20Mb down, 2Mb up |
| 09:04.09 | mort_gib | You can get DDI's from a lot of providers, meaning that no phone line is required... |
| 09:04.38 | Zylogue | DDI's, OK. I will look that up... |
| 09:04.38 | mort_gib | You have to play around with a NAT a bit as SIP generally don't like NAT |
| 09:04.39 | mort_gib | ~sipnat |
| 09:04.40 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 09:05.05 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:05.13 | box2 | voip phone providers are ok, i havn't used one that was super great though |
| 09:05.22 | mort_gib | That way, you can use an Alix board with a 4gb Sandisk III card and get Snom 300 or Polycom 330's |
| 09:05.29 | box2 | you should look for comparisons of the providers online somewhere |
| 09:05.42 | Zylogue | I've got a linksys router running dd-wrt. |
| 09:05.47 | Zylogue | will that help? |
| 09:05.55 | mort_gib | box2: True, but there is a price/quality balance to explore |
| 09:06.08 | box2 | yea, that's not something i've explored much heh |
| 09:06.10 | mort_gib | Zylogue: Whatever you are comfortable with |
| 09:06.25 | Zylogue | mort_gib, that price/quality quoteint is why I'm wanting to tell the local telco to 'get off' |
| 09:06.57 | mort_gib | box2: I have been looking into this and have found decent enough providers, though NOT to all destinations. |
| 09:07.16 | Zylogue | mort_gib, like who? |
| 09:07.25 | Zylogue | Vonage? |
| 09:07.49 | mort_gib | I use voipon.co.uk for European calls... |
| 09:08.36 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-29-183.dsl.lsan03.sbcglobal.net) |
| 09:08.59 | mort_gib | They suck for calls to Spanish mobiles |
| 09:09.21 | mort_gib | UK calls are great, Germany is ok, Switzerland not |
| 09:09.43 | mort_gib | box2: Is right when he says that it differs.... |
| 09:10.24 | Zylogue | I appreciate the details. I will have to look into this deeper. |
| 09:10.36 | Zylogue | you guys have been great. |
| 09:11.41 | Zylogue | to summarize: If possible go with IP phones, so I can avoid specialized hardware in the server, and I need to locate VOIP service providers to compare/contrast service quality/price for my needs |
| 09:12.03 | agx | hello, i've a Digium TE card PRI at span 2 the timing value in zaptel.conf should be 1 or 2 ? i mean: span=2,1,0 or span=2,2,0 ? span=1 is an FXO card instead. |
| 09:12.05 | ascent | Hmm |
| 09:12.14 | ascent | Does asterisk support messaging, and if so, from what version? |
| 09:12.42 | box2 | Zylogue: sounds about right |
| 09:12.59 | box2 | i like the Polycom phones |
| 09:13.47 | mort_gib | Polycoms are nice, I can't decide between Polycoms and Snoms |
| 09:13.54 | tzafrir_laptop | agx, 1 |
| 09:14.21 | agx | tzafrir_laptop: ty mate!! i own you 3 beers at least so far :-P |
| 09:14.50 | agx | mort_gib: Polycom is the best, Snom was good before they starting putting "made in china" on their product |
| 09:15.33 | agx | ascent: MESSAGE header can be sent directly using sipsak utility; asterisk 1.4 does not route message: there is a patch for it too somewhere; dunno about 1.6 |
| 09:15.44 | mort_gib | agx: Yeah, I had to return a few with faults, but lusers like them better.... |
| 09:16.00 | Zylogue | box2, mort_gib, thansk again for your answers. |
| 09:16.14 | mort_gib | Zylogue: np |
| 09:16.21 | box2 | np |
| 09:16.30 | ascent | agx: ok, thanks. Let me check the changelog for 1.6 |
| 09:16.31 | Zylogue | at work (Dell computers) we use Avaya IP phones. |
| 09:16.42 | agx | mort_gib: i'm fine with IP310 POE, my coworkers like Snom320 |
| 09:17.29 | mort_gib | agx: See I have done both Polycom and Snom installs, but with the Snoms users get into the project faster... They simply like the website |
| 09:17.53 | mort_gib | That said, the Polycoms are better quality, and sound is simply just a little bit better |
| 09:18.51 | mort_gib | -And don't get me started on handsfree.... |
| 09:18.58 | agx | mort_gib: you can have better quality on Snom using the Klarvoice handset :) the problem now i think they put too much things in firmware |
| 09:19.10 | defsdoor | uses aastra almost exclusively |
| 09:19.17 | agx | i like that handset |
| 09:19.35 | defsdoor | have one polycom conference station and it was a bitch |
| 09:19.51 | mort_gib | I haven't tried the "Klearvoice" yet |
| 09:20.12 | box2 | "kan joo heer mi nao?" |
| 09:21.33 | mort_gib | defsdoor: How come?? |
| 09:21.53 | mort_gib | I have 8 of them things running, just fine |
| 09:22.00 | defsdoor | mort_gib: stupid web configuration with "obscure" fields |
| 09:22.18 | mort_gib | mind you, DON'T use the webinterface for configuration |
| 09:22.42 | defsdoor | this was a single handset on a site full of aastras - I didnt want to get into provisioning it properly |
| 09:22.56 | defsdoor | especially as there was little help on how to do it on line |
| 09:22.59 | mort_gib | Pity, you really should have |
| 09:23.16 | agx | box2: ROFL |
| 09:23.17 | Zylogue | as I look for IP phones, what features should I be looking for? The Polycom 320 is one I found on ebay. Does SIP matter with asterisk? |
| 09:23.20 | mort_gib | Or paid someone in here to help out... |
| 09:23.49 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 09:23.59 | mort_gib | Zylogue: ANY sip compliant IP handset will do, you choice is how much to spend |
| 09:24.15 | mort_gib | Zylogue: Remember never be cheap :-) |
| 09:25.46 | Zylogue | mort_gib, define 'cheap' |
| 09:25.58 | *** part/#asterisk agx (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:25.59 | viraptor | Zylogue: stay away from grandstreams ;) |
| 09:26.02 | mort_gib | Greandstream# |
| 09:26.15 | viraptor | ~gs |
| 09:26.16 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 09:26.32 | mort_gib | viraptor: I fully agree |
| 09:26.50 | mort_gib | How are the Astras these days?? |
| 09:26.57 | ascent | Hm let's drop the messaging idea for a while :) |
| 09:27.13 | defsdoor | mort_gib: I can't fault the aastras I have used |
| 09:27.30 | defsdoor | 9133i, 480i, 57i etc.. so far |
| 09:27.38 | mort_gib | defsdoor: Ok, never gave them a fair chance :-) |
| 09:27.55 | mort_gib | Anyway, gottago |
| 09:32.43 | Elwell | don |
| 09:35.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:50.05 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:52.07 | TSM | has anyone had to work out disconnect tone sequences for SPA adapters? |
| 10:06.21 | *** join/#asterisk Micc (n=dotirc@174-21-26-36.tukw.qwest.net) |
| 10:06.39 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 10:07.17 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 10:09.17 | *** join/#asterisk qdk (n=qdk@0x573d8cf5.bynqu1.dynamic.dsl.tele.dk) |
| 10:16.21 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:48.09 | *** join/#asterisk c4rg (i=crg@lagoon.freebsd.lublin.pl) |
| 10:49.38 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a1d1:3adb:cbb9:73c1) [NETSPLIT VICTIM] |
| 10:49.38 | *** join/#asterisk cnu (n=cnu@63.80-203-44.nextgentel.com) |
| 10:49.38 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:0) [NETSPLIT VICTIM] |
| 10:49.38 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 10:56.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:57.38 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 11:06.30 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 11:10.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 11:17.00 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-xsnevculhhtsogkx) |
| 11:38.51 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
| 11:39.02 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:46.25 | TSM | can asterisk send back the display name of a number that you are calling, our polycom phones only show the extention and not the name when we call out |
| 11:48.44 | synthetic | over isdn or pots line |
| 11:48.49 | synthetic | or to outher voip phone |
| 11:49.53 | TSM | internal voip phone to asterisk, calling internal extention and send back the name |
| 11:50.43 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 11:57.25 | synthetic | what do you have set in sip.conf |
| 11:57.33 | synthetic | you can used callerid=asreceived |
| 11:59.16 | jkroon | is it possible to make voicemail* apps auth against some key in the astdb? |
| 11:59.28 | jkroon | i see stuff for odbc based auth, but not astdb. |
| 12:02.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:07.50 | *** join/#asterisk Yoe (n=wouter@samba.grep.be) |
| 12:08.08 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 12:08.21 | ascent | yoe: Are you the same Wouter Verhelst that posts to planet.debian.org? :) |
| 12:08.27 | Yoe | ascent: yes :-) |
| 12:08.29 | ascent | fun :) |
| 12:08.54 | ascent | Your posts keep hilighting because we share first names and I have an irc bot that aggegates RSSfeeds :) |
| 12:09.20 | Yoe | I was going to ask: I'm currently having issues with a Linksys SPA3102 connected to an asterisk box |
| 12:10.12 | Yoe | it works when calling; but when the system needs to go to voicemail, the caller will hear dialtones for a very short moment, and the recording does not contain any voice; instead, it contains 'occupied' beeps |
| 12:10.23 | Yoe | any hints? |
| 12:12.08 | box2 | up up down down left right left right b a b a select start |
| 12:12.41 | Yoe | box2: er, if you need more information, I'm happy to provide it, but this is less than useful... |
| 12:12.50 | box2 | heh sorry, i couldn't resist |
| 12:12.59 | box2 | i have no helpful information for you |
| 12:14.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 12:15.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.52 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 12:16.21 | tzafrir_laptop | Yoe, I figure you could start with a dialplan trace of things |
| 12:16.28 | tzafrir_laptop | core set verbose 3 |
| 12:16.45 | tzafrir_laptop | and then look at the CLI |
| 12:19.53 | Yoe | it could of course be that I did something wrong; the setup is that we have a exten => s,1,Dial(SIP/201&SIP/202&SIP/203,30) |
| 12:20.12 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:20.56 | ariel_ | Morning |
| 12:20.56 | Yoe | after that it does a "Voicemail(800,u)" |
| 12:21.28 | TSM | synthetic: at the moment each extention has callerid=device <2200> depending on extention |
| 12:21.51 | Yoe | so there's just one voicemail box for everyone, which makes more sense than having one per person (there are only three extensions, and they're sitting at the same 3-sided desk...) |
| 12:22.30 | synthetic | i beleive you needs quotes for "device" |
| 12:22.38 | tzafrir_laptop | Yoe, well, I don't clearly understand what you try to do. Maybe start by a pastebin of the relevant dialplan? |
| 12:22.49 | Yoe | tzafrir_laptop: sure, hang on |
| 12:23.27 | ariel_ | has found that people don't like sharing vm accounts in the long term..... |
| 12:23.49 | Yoe | tzafrir_laptop: http://paste.debian.net/43478/ |
| 12:24.23 | Yoe | the voicemail setup could probably be easier, since "busy" is never going to happen anyway and we don't really handle anything else |
| 12:24.51 | Yoe | I copied this from somewhere, not sure where exactly |
| 12:25.31 | TSM | synthetic: i still dont understand how that would return the name of the phone i am calling if its in the extentions list, the called party see my extention name fine, i want to see who i am calling and not just the number i dialed |
| 12:26.02 | tzafrir_laptop | Yoe, for starters, it is recommended not to use 'default' |
| 12:26.23 | Yoe | oh? why's that? |
| 12:26.43 | tzafrir_laptop | that is: leave it for non-functional incoming calls and such |
| 12:26.44 | ariel_ | it's open |
| 12:27.01 | tzafrir_laptop | but that's really a minor point |
| 12:27.31 | tzafrir_laptop | next step would be to see what actually happens |
| 12:27.57 | tzafrir_laptop | in the CLI (asterisk -r) - |
| 12:28.02 | tzafrir_laptop | core set verbose 3 |
| 12:28.04 | Yoe | ah, with the verbose thing? |
| 12:28.07 | Yoe | right, hang on |
| 12:28.13 | tzafrir_laptop | and then you should be able to see a trace of the call |
| 12:28.46 | Yoe | but there's nothing that jumps out to you as stupid or the cause of this problem, I take it? |
| 12:29.23 | ariel_ | basic ring group, leave message setup, just inbound rule should be more like exten => X.,goto(blah) to catch numbers in the default |
| 12:29.50 | tzafrir_laptop | s/X./_X./ , of course |
| 12:30.19 | Yoe | right, though since it's coming from an ATA with no DID setup, that shouldn't be a problem |
| 12:31.40 | Yoe | mm, that can't happen right now. It's at a customer, and they tell me they'd rather I wait a bit, since they're expecting some rather important phone |
| 12:31.44 | Yoe | will try in half an hour or so |
| 12:33.51 | *** join/#asterisk propellerhead (n=yogurt2u@host159.200-117-121.telecom.net.ar) |
| 12:34.03 | ariel_ | your trying this on the actual customer? wow, vm on your desktop and try it out for yourself....testing lab...hint... |
| 12:35.19 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 12:37.04 | Yoe | the setup was done yesterday, so it's okay if there are a few kinks in the thing still. |
| 12:37.49 | Yoe | and it's only the voicemail that's broken at this point, anyway. |
| 12:37.53 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 12:41.06 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:41.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:44.47 | [TK]D-Fender | ~cpid |
| 12:44.48 | infobot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 12:52.45 | jkroon | hi all - is it possible to store VM passwords in astdb? |
| 12:59.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:01.35 | *** join/#asterisk Pan3D (n=Pan3D@streaming.sensoryresearch.net) |
| 13:02.32 | box2 | does anyone have experience playing music through a softphone? |
| 13:03.18 | Yoe | box2: er, you just add an extension that says "MusicOnHold()" when you dial it. Then you dial that extension from your softphone. Done. |
| 13:03.36 | Yoe | (of course, that does require you to set up a working MOH setup, but other than that...) |
| 13:03.41 | box2 | i mean the other way around |
| 13:03.59 | box2 | from the softphone to * |
| 13:05.27 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 13:05.36 | leifmadsen | jkroon: not without hacking the voicemail code I don't think |
| 13:05.52 | ascent | yoe: sounds like fun, let's try. |
| 13:06.21 | leifmadsen | it'll sound better if you can use G.722 |
| 13:06.28 | leifmadsen | otherwise, stick to G.711 |
| 13:07.25 | [TK]D-Fender | jkroon: You can bypass *'s use of PW's and auth entry to it yourself. |
| 13:07.52 | [TK]D-Fender | jkroon: There are other options depending on what you intend to actually DO with it... |
| 13:09.09 | [TK]D-Fender | box2: Whats there to know? Set your recording source to something thats outputting music. |
| 13:09.28 | jkroon | leifmadsen, thanks for the book btw. |
| 13:09.40 | [TK]D-Fender | box2: on Creative Labs cards you have a "What U Hear" mixer source you can use as "Mic" |
| 13:10.15 | jkroon | ok, so what are my options? basically I want to allow a user to change his/her password but I from time to time regenerate voicemail.conf and don't want to reset all the passwords when I do. |
| 13:10.22 | leifmadsen | jkroon: glad you enjoy it! |
| 13:10.27 | jkroon | I also use #include in voicemail.conf which doesn't do so well. |
| 13:10.36 | box2 | [TK]D-Fender: hmmm |
| 13:10.54 | box2 | none of my softphones give me options for anything other than my HDA-Intel mic port |
| 13:11.22 | leifmadsen | jkroon: check voicemail.conf.sample |
| 13:11.25 | leifmadsen | ; If you need to have an external program, i.e. /usr/bin/myapp |
| 13:11.25 | leifmadsen | ; called when a voicemail password is changed, uncomment this: |
| 13:11.25 | leifmadsen | ;externpass=/usr/bin/myapp |
| 13:12.00 | jkroon | leifmadsen, i was hoping to avoid external programs. |
| 13:12.13 | leifmadsen | jkroon: sorry -- those are your options |
| 13:12.17 | jkroon | how do i make voicemail auth against the astdb to being with? |
| 13:12.24 | [TK]D-Fender | jkroon: Maybe you should make your "recreation" script smarter <- |
| 13:12.24 | leifmadsen | you don't |
| 13:12.32 | jkroon | leifmadsen, that's cool: _hoping_ to avoid :) |
| 13:12.54 | [TK]D-Fender | jkroon: and as I said you can do your OWN auth OUTSIDE of Voicemail() |
| 13:12.54 | leifmadsen | it either auths against voicemail.conf (non-included file) or it auths against IMAP or ODBC |
| 13:13.01 | leifmadsen | or that |
| 13:13.24 | leifmadsen | play the prompts and authentication outside of Voicemail() |
| 13:13.34 | leifmadsen | in the dialplan, as [TK]D-Fender is suggesting |
| 13:13.37 | jkroon | [TK]D-Fender, before passing into VoiceMail() ... Mr [TK]D-Fender - I do believe you are the most creative person here. thanks. |
| 13:13.41 | leifmadsen | (which allows you to use astDB) |
| 13:14.00 | leifmadsen | [TK]D-Fender: you should write a book or something |
| 13:14.28 | [TK]D-Fender | is Zen Master of "Or Something". |
| 13:14.33 | [TK]D-Fender | ~[TK]D-Fender |
| 13:14.34 | infobot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 13:14.37 | [TK]D-Fender | And that :) |
| 13:18.10 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 13:19.05 | jkroon | lol |
| 13:19.14 | coppice | writing books is easy..... until someone mentions plagarism |
| 13:19.15 | *** join/#asterisk kron4eg (n=kron@unaffiliated/fixer/x-387521) |
| 13:19.27 | jkroon | either way [TK]D-Fender - what is blatantly obvious to you takes some of us weeks. |
| 13:19.36 | jkroon | thanks. |
| 13:21.11 | [TK]D-Fender | coppice: Stealing material from one person = plagiarism. Stealing from 100 people = research :p |
| 13:21.52 | leifmadsen | [TK]D-Fender: amen! |
| 13:22.00 | leifmadsen | ...errr. Wait. |
| 13:22.56 | [TK]D-Fender | Ramen! Ancient prayer of the noodle! |
| 13:24.28 | eppigy | TRABAJO |
| 13:24.31 | box2 | lol |
| 13:25.09 | coppice | Ramen is just a Japanese ripoff of something from a place that would never copy |
| 13:28.39 | *** join/#asterisk ruben23 (n=ADMIN@122.55.48.243) |
| 13:29.49 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:29.49 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:33.05 | *** join/#asterisk simprix (n=simprix@mail.menufocus.com) |
| 13:33.13 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:40.22 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 13:40.28 | boch | hi all |
| 13:41.24 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 13:45.35 | boch | is it possible to list a db family through the AMI ? |
| 13:46.00 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 13:47.42 | [TK]D-Fender | boch: Yes |
| 13:48.10 | boch | [TK]D-Fender, what command? cause there is a DBGet and DBPut, but nothing like DBShow |
| 13:49.35 | [TK]D-Fender | boch: COMMAND <- |
| 13:50.29 | boch | ok im gonna try, thanks |
| 13:54.29 | *** join/#asterisk KazaLite (n=KazaLite@94-193-98-124.zone7.bethere.co.uk) |
| 13:54.44 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:55.17 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 13:55.17 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:58.44 | ariel_ | is it not database show |
| 14:00.49 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:01.57 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:03.08 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 14:03.41 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:03.50 | LemensTS | hey guys. any reason http://www.serverloft.com/dedizierte-server/server-details.php?products=3 wouldnt work on calling hundreds of people and playing pre recorded messages in g729 format (handing off to itsp in g729 also) |
| 14:04.55 | *** join/#asterisk btsteve (n=tstevens@67.201.69.212) |
| 14:06.11 | Yoe | LemensTS: if it's pre-recorded in the correct format, then you have almost no processing power requirements |
| 14:07.56 | LemensTS | Yea i pre-record it. But didnt know if how many simultaneous calls i could do on the server with these specs |
| 14:08.18 | ariel_ | LemensTS: issue is how many calls per sec your going to be doing, how manny channels and license for g729 are you going to run, then next is your handing it off via sip? with canreinvite=yes ? |
| 14:08.26 | LemensTS | http://www.serverloft.com/dedizierte-server/server-details.php?products=0 they got one with a xeon also....im not much of a hardware guy |
| 14:08.48 | LemensTS | ariel_: we will pass it off to itsp via g729 so we wont need licenses |
| 14:09.01 | ariel_ | you will to play the message |
| 14:09.13 | LemensTS | yes handing off sip, what difference will canreinvite=yess make? |
| 14:09.39 | Yoe | if you're going to be playing prerecorded messages on a quadcore, I'm pretty sure your network is going to be the bottleneck, not the processor |
| 14:09.45 | LemensTS | ariel_: i got a couple g729 licenses to record the messages in g729 format |
| 14:09.56 | Yoe | compare it to a webserver serving static pages -- that's pretty much the same thing |
| 14:10.18 | ariel_ | does not matter each channel is going to be connected to the asterisk system while playing the message |
| 14:10.29 | [TK]D-Fender | LemensTS: Reinvite does not exist, you have no transcode load, you could probably pass a hundred or few calls at a time on it. |
| 14:10.45 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:10.54 | [TK]D-Fender | [10:10]<ariel_>does not matter each channel is going to be connected to the asterisk system while playing the message <- way off base here... |
| 14:11.11 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:11.17 | ascent | What does "rc_avpair_new: unknown attribute 1490026597" mean ? |
| 14:11.29 | ascent | Google doesn't show too much. Something about cdr.conf but that's commented out |
| 14:11.35 | ariel_ | ok |
| 14:11.53 | LemensTS | tkd-fender: what does reinvite matter |
| 14:13.27 | LemensTS | tkd-fender && ariel_: so is ariel wrong? i talked to a digium tech and they said if i recorded the message in g729 i could pass it off to the itsp in g729 without needing a licnese...so im not sure now... |
| 14:13.44 | [TK]D-Fender | LemensTS: it DOESN'T. there is no reinvite, there is no bridged call. |
| 14:14.05 | LemensTS | ok gotcha. |
| 14:14.06 | [TK]D-Fender | LemensTS: And there is no licensing issue or load while running. |
| 14:14.09 | ariel_ | LemensTS: it appears that I am wrong |
| 14:14.12 | [TK]D-Fender | CRAZY PEOPLE |
| 14:14.23 | LemensTS | god is great, beer is good, and people are crazy |
| 14:15.39 | LemensTS | Last question. opteron or xeon |
| 14:15.55 | JT | c2d ;) |
| 14:16.45 | [TK]D-Fender | LemensTS: YES |
| 14:16.57 | [TK]D-Fender | LemensTS: C2Q <- |
| 14:17.21 | [TK]D-Fender | (until the six-core's get released) |
| 14:17.22 | JT | xeon probably has a slight performance edge on opteron, but a major fail on power use |
| 14:17.26 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 14:17.41 | JT | until they get rid of FBDIMM in their xeon chipsets, they will always be at a disadvantage |
| 14:17.48 | JT | FBDIMM uses like 70W alone |
| 14:19.07 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:20.58 | LemensTS | c2d is that a xeon quadcore x2 ? |
| 14:21.36 | JT | core 2 duo |
| 14:21.48 | JT | desktop cpu |
| 14:22.09 | LemensTS | you reccomend that over a xeon? |
| 14:22.33 | JT | depends what you're doing |
| 14:22.43 | [TK]D-Fender | JT: ... Oh shit, RUN!! http://www.environmentalgraffiti.com/featured/terrifying-underwater-encounters-bengali-white-tiger/14065 |
| 14:23.04 | LemensTS | just want to play a message to as many people as possible. no transcoding. handing off via si to itsp |
| 14:23.04 | JT | i like to think more about doing things in clusters of cheap power efficient commodity hardware these days |
| 14:23.28 | ascent | [tk]d-fender: nice link :) |
| 14:23.49 | LemensTS | jt: i dont have to pay the bill |
| 14:25.11 | JT | a bit short sighted but ok ;) |
| 14:25.23 | JT | don't get me wrong, big expensive servers could be the answer |
| 14:29.59 | box2 | [TK]D-Fender: aaahhh i see now what you meant before |
| 14:30.12 | box2 | the capture port in alsamixer does what i think you were saying |
| 14:30.32 | box2 | you have led me to the most awesome and sometimes slightly evil of revelations |
| 14:31.06 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:31.06 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:31.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:37.39 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 14:37.49 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 14:40.52 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 14:48.45 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:48.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:51.03 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 14:52.16 | Naikrovek | hey dudes; i have two offices, workers in remote office connect to asterisk server in our office; i want to put a server in their office and have their server serve their phones. any clue how that would be configured? |
| 14:52.40 | [TK]D-Fender | Naikrovek: * there is the same as anywhere else. |
| 14:52.49 | Naikrovek | i'm guessing that there would be a trunk set up between the servers, with some routing between to direct calls to the appropriate server |
| 14:53.05 | [TK]D-Fender | Naikrovek: You've apparently already done it once... |
| 14:53.21 | [TK]D-Fender | Naikrovek: for that, yes, 1 peer/side could o. |
| 14:53.31 | Naikrovek | well i inherited this asterisk server, but i love it |
| 14:54.49 | Naikrovek | so i came into an already-working asterisk config, but need to put a server at the remote office |
| 14:55.20 | Naikrovek | then connect the two so when i dial an extension over there, it goes through my pbx, into theirs, then to whatever extension |
| 14:55.35 | Naikrovek | think i've figured it out in my head but need to try it with some virtual machines to make sure. |
| 14:59.56 | *** join/#asterisk jtodd (i=o9le8yt4@ns.fox-den.com) |
| 14:59.56 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:01.38 | *** join/#asterisk andres833 (n=andres83@190.156.199.40) |
| 15:02.15 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:04.37 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:08.02 | Naikrovek | forgive the stupid question here but what would the requirements be for an asterisk box serving about 60 phones, with possibly about 20-30 simultaneous calls (between extensions) at any time |
| 15:08.21 | Naikrovek | polycom phones, I assume you won't need much if the phones are configured correctly and no transcoding is required |
| 15:08.22 | [TK]D-Fender | Naikrovek: Any P4 could do this easy |
| 15:08.34 | Naikrovek | perfect |
| 15:08.36 | Naikrovek | thanks again |
| 15:09.06 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:09.11 | Naikrovek | need to find a voip provider for that office, too... |
| 15:09.21 | Naikrovek | so they don't suck up our bandwidth anymore |
| 15:09.24 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 15:11.52 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:11.53 | *** join/#asterisk nauticalthinker (n=mratliff@phoenix.warren-wilson.edu) |
| 15:12.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:12.59 | nauticalthinker | hey guys! I'm in the middle of trying to get asterisk to connect to an f9600 system via a t1 (TE121) card. Alarm stays on red no matter what config I put in |
| 15:13.30 | nauticalthinker | Does anyone have any infput on getting these two systems working together? |
| 15:13.32 | *** join/#asterisk bhodder (n=blake@142.166.111.186) |
| 15:13.44 | [TK]D-Fender | nauticalthinker: T1 is T1... so your settings are wrong... |
| 15:13.57 | [TK]D-Fender | nauticalthinker: Would be nice to know what the F9600 is expecting and seeing your configs... |
| 15:13.59 | [TK]D-Fender | ~pb |
| 15:14.00 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:14.01 | [TK]D-Fender | ^^^^^^^^^ |
| 15:14.07 | [TK]D-Fender | nauticalthinker: PASTEBIN is your friend... |
| 15:14.41 | bhodder | hi I am trying to set an IVR that I have pre-recorded on my laptop and sent to the asterisk server but it will not play the file. can anyone help? |
| 15:21.02 | bhodder | I have done it once but am failing to do it the second time. I have tried to take the .wav and convert it to .ulaw or .gsm but it still will not work. |
| 15:22.37 | [TK]D-Fender | bhodder: PB the attempt. |
| 15:24.23 | bhodder | oh sorry, must clearify that the conversion worked but would not play on the IVR |
| 15:26.00 | bhodder | <[TK]D-Fender>: do you want me to PB the conversion attempt? |
| 15:26.13 | [TK]D-Fender | bhodder: both |
| 15:26.59 | bhodder | ok |
| 15:27.45 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 15:35.40 | Psychobilly | modprobe -r zaptel |
| 15:35.40 | Psychobilly | zaptel: Device or resource busy |
| 15:35.51 | [TK]D-Fender | Psychobilly: Might want to stop *... |
| 15:35.53 | Psychobilly | any ideas to unload zaptel module without rebooting? |
| 15:35.54 | bhodder | <[TK]D-Fender>: I was about to PB it however I just got it to work thanks anyways |
| 15:36.07 | Psychobilly | [TK]D-Fender it is stopped and all other modules are unloaded |
| 15:36.30 | Psychobilly | nothing is running and lsof doesnt return anything * or zaptel related |
| 15:36.42 | [TK]D-Fender | Psychobilly: Ok, outside of my scope then... |
| 15:36.51 | Psychobilly | :< |
| 15:36.52 | *** join/#asterisk ddickenson_ (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 15:37.18 | tzafrir_laptop | Psychobilly, you can't unload zaptel if one of the dependent module uses it |
| 15:37.24 | tzafrir_laptop | or if Asterisk keeps it open |
| 15:37.31 | Psychobilly | no other module is loaded |
| 15:37.37 | tzafrir_laptop | stop asterisk |
| 15:37.46 | tzafrir_laptop | btw: /etc/init.d/asterisk stop |
| 15:37.47 | Psychobilly | it is stopped, just told it |
| 15:37.49 | tzafrir_laptop | err |
| 15:37.54 | tzafrir_laptop | btw: /etc/init.d/zaptel stop |
| 15:37.58 | Psychobilly | same |
| 15:38.02 | tzafrir_laptop | unloads all dependning modules |
| 15:38.16 | Psychobilly | i did |
| 15:38.19 | Psychobilly | grrrr |
| 15:38.20 | tzafrir_laptop | do you have zttest or whatever running ? |
| 15:38.25 | Psychobilly | nope |
| 15:38.45 | ddickenson_ | exten => s,n, Set(VMStatus=$["${DB(users/${UserID}/vm)}" = "1"]) |
| 15:38.50 | tzafrir_laptop | lsmod | grep ^zaptel |
| 15:39.04 | ddickenson_ | crap.. if I used that in a macro how do I get VMstatus in ASTDB? |
| 15:39.37 | Psychobilly | lsmod | grep ^zaptel |
| 15:39.37 | Psychobilly | zaptel 236768 2 |
| 15:39.55 | tzafrir_laptop | Psychobilly, ok. so the reference count is 2 |
| 15:40.07 | tzafrir_laptop | ls /dev/zap |
| 15:40.30 | Psychobilly | lost of nodes there |
| 15:40.44 | Psychobilly | its an old slack machine with static dev nodes |
| 15:41.03 | tzafrir_laptop | What version of zaptel is it? |
| 15:41.05 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:41.08 | [TK]D-Fender | ddickenson_: VM status isn't in AstDB <- |
| 15:41.23 | Psychobilly | 1.4.12.1 |
| 15:41.43 | ddickenson_ | oh... well what sets that variable? |
| 15:42.24 | [TK]D-Fender | ddickenson_: "core show functions like VM" |
| 15:42.39 | ddickenson_ | The idea of this large macro (that I clearly didn't write) is partially to send calls to voicemail only if they are subscriber |
| 15:43.34 | [TK]D-Fender | ddickenson_: You seem dangerously ignorant of your own operating environment... |
| 15:43.57 | box2 | danger is a sexy middle name |
| 15:44.10 | ddickenson_ | valid observation... I've been kinda thrown into this and having to learn as I go on soon to be production systems |
| 15:45.08 | ddickenson_ | why do you think I'm on here so much! once I figure out what I'm doing a bit better I'll try to contribute |
| 15:46.46 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.29 | box2 | softphone soundboard, success |
| 15:47.32 | box2 | yatta! |
| 15:47.41 | box2 | chun li victory pose |
| 15:50.03 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 15:50.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 15:52.26 | Psychobilly | im rebooting, i hope this old shit boots up again :< |
| 15:54.25 | *** join/#asterisk btsteve (n=tstevens@204.10.23.244) |
| 15:55.13 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 15:55.43 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 15:56.15 | muiro | does anyone here use any software to analyze CDRs? |
| 15:56.27 | Psychobilly | ouuff |
| 15:56.32 | Psychobilly | it did |
| 16:01.15 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:02.41 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 16:02.51 | *** join/#asterisk jtodd (i=ac3eyqd3@ns.fox-den.com) |
| 16:02.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:06.27 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:06.47 | *** join/#asterisk moy (n=moy@c-98-193-24-21.hsd1.il.comcast.net) |
| 16:09.12 | ascent | Can anyone quicly try to see if sip:wouter@schoot.org rings at my phone please ?:) |
| 16:10.05 | box2 | yep it works |
| 16:10.10 | box2 | yw |
| 16:10.14 | ascent | thanks for testing mate. |
| 16:11.11 | ascent | I just lost my voip virginity because of you ;-) |
| 16:12.09 | *** join/#asterisk DigitalDaz21 (n=digitald@78-105-169-114.zone3.bethere.co.uk) |
| 16:12.10 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 16:13.53 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 16:14.09 | bsilberman | question re adding to the mysql db in asterisk 1.4 |
| 16:14.21 | bsilberman | what do I need to do to get the call-limit parameter added? |
| 16:14.36 | bsilberman | keeps choking on it... the rest of the line inserts just fine. |
| 16:14.50 | box2 | ascent: lol |
| 16:15.11 | bsilberman | mysql> INSERT INTO sip (name,host,nat,type,cancallforward,canreinvite,context,secret,disallow,allow,username) VALUES ("6533934","dynamic","no","friend","yes","yes","collectors","6533934","all","g729","6533934"); |
| 16:15.37 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.108.61) |
| 16:15.41 | DelphiWorld | hello all |
| 16:15.54 | DelphiWorld | please could anyone point me to any SIP Provider that use TLS? |
| 16:16.10 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 16:16.15 | _Raptor_ | hi |
| 16:16.47 | _Raptor_ | i have a register entry (register = |
| 16:17.39 | *** join/#asterisk QaDeS (n=mklaus@p4FC72923.dip0.t-ipconnect.de) |
| 16:17.39 | _Raptor_ | sry. i have a register entry (register => 091XXXX800@arcor) and a section named [arcor] with all details. everything works fine but the console shows [Aug 6 18:10:36] WARNING[3400]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'arcor' |
| 16:17.45 | _Raptor_ | any ideas? |
| 16:20.19 | WindowsUser | register => you need full host |
| 16:22.20 | bsilberman | anyone re the mysql query? |
| 16:22.28 | *** join/#asterisk [D-Ster-] (n=ster@pool-70-109-24-51.atl.dsl-w.verizon.net) |
| 16:25.06 | *** join/#asterisk c4rg (i=crg@lagoon.freebsd.lublin.pl) |
| 16:25.40 | muiro | does anyone know of any CDR analysis tools? |
| 16:28.03 | *** join/#asterisk ethicx (n=chatzill@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 16:28.07 | _Raptor_ | WindowsUser: but the example does not say so: ;register => 2345:password@sip_proxy/1234 |
| 16:29.14 | WindowsUser | What works > the comments ;) |
| 16:29.55 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:35.14 | styelz | its just a warning, wouldnt worry about it |
| 16:35.44 | styelz | maybe you have srvlookup=yes |
| 16:36.43 | styelz | <PROTECTED> |
| 16:42.22 | bpgoldsb | I'm trying to use ParkAndAnnounce. When I specify the return context, I get 'Warning: Return Context Invalid, call will return to default|s'. But when I setup a test exten and go to that context directly, it works fine. Anyone have any ideas why? |
| 16:43.10 | *** join/#asterisk btsteve (n=tstevens@67.201.69.212) |
| 16:47.24 | *** part/#asterisk ruben23 (n=ADMIN@122.55.48.243) |
| 16:49.25 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 16:49.53 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 16:51.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:52.15 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:55.27 | _Raptor_ | styelz: yes i have. but this is only the name, not the host. there is also a host entry in the [xxxxx] section |
| 16:55.51 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 16:56.53 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:57.59 | [TK]D-Fender | bpgoldsb: PASTEBIN <- |
| 16:58.12 | [TK]D-Fender | muiro: Asterisk-stat |
| 16:58.44 | [TK]D-Fender | _Raptor_: Register statement has precised NOTHING to do with your peer entry |
| 16:59.23 | [TK]D-Fender | _Raptor_: register => user:pass@serveriporfqdn/extentodial |
| 16:59.34 | _Raptor_ | [TK]D-Fender: so how can i set options like canreinvite and so on and trigger a register |
| 16:59.47 | lmadsen | I really wish Asterisk-stat would let you filter or sort on LastApp or LastData |
| 16:59.51 | muiro | [TK]D-Fender: does that have to plug into the management interface or can I feed it a custom CDR from a database? |
| 17:00.00 | [TK]D-Fender | _Raptor_: reinvite is a peer option, nothing to do with registering |
| 17:00.03 | [TK]D-Fender | ~sipregister |
| 17:00.04 | infobot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 17:00.23 | [TK]D-Fender | muiro: I runs off MySQL, and optionally PG as well |
| 17:00.38 | dandre | Hello, |
| 17:00.40 | *** join/#asterisk QaDeS_ (n=mklaus@p4FC7320C.dip0.t-ipconnect.de) |
| 17:00.59 | [TK]D-Fender | _Raptor_: and you don't trigger a register. * jsut does it. and does it on expiry |
| 17:01.36 | Kobaz | if you do a sip reload, i think it forces a register |
| 17:01.45 | [TK]D-Fender | Kobaz: Normally, yes |
| 17:01.48 | _Raptor_ | yes it does |
| 17:02.07 | DigitalDaz21 | Hi all, I have a provider where on a hardware phone Proxy Server Address:=A Registrar Server Address:=A and Outbound Proxy:=B. Everything works fine. Unfortunately "A" does not resolve by DNS and this seems to cause a problem for asterisk, does anyone know how to create a register string that works for this scenario? |
| 17:02.07 | Kobaz | what about abnormally? |
| 17:02.10 | dandre | I have an extension that calls queue application. If I use Dial (Local/thatextension/n) I never go to the next step in my dialplan |
| 17:02.12 | _Raptor_ | thx for your help, i have to leave now, i will try again later |
| 17:03.08 | Kobaz | dandre: queue will hijack the call until it meats the conditions to come out of the queue |
| 17:03.12 | Kobaz | meets |
| 17:03.44 | dandre | yes but the queue timesout and the call is hungup |
| 17:04.09 | [TK]D-Fender | dandre: And the queue plays music, etc which means the call is ANSWERED |
| 17:04.21 | Kobaz | dandre: send the Queue application the 'c' option |
| 17:04.24 | Kobaz | <PROTECTED> |
| 17:04.29 | [TK]D-Fender | dandre: And what did you do taht would make a Dial that is ANSWERED continue after it terminates? |
| 17:04.40 | [TK]D-Fender | reloads chan_rhetoricalquestion.so |
| 17:04.58 | dandre | ok |
| 17:05.11 | [TK]D-Fender | Kobaz: CLOSE |
| 17:05.25 | [TK]D-Fender | Kobaz: Nifty idea, but not where he wants to continue. |
| 17:06.02 | Kobaz | hold on... so he wants to go into a queue and dial a phone? |
| 17:06.36 | [TK]D-Fender | Kobaz: Read it a few more times :) |
| 17:06.38 | Kobaz | heh |
| 17:07.11 | dandre | I want to continue the dialplan if nobody answered the call |
| 17:07.43 | Kobaz | that's what queue does |
| 17:08.15 | dandre | yes but in that case i use dial(Local/queueextension/n) |
| 17:08.26 | dandre | so must be impossible |
| 17:08.33 | Kobaz | nothing is impossible |
| 17:08.38 | [TK]D-Fender | dandre: Standard TIMEOUT parm does that already |
| 17:08.40 | dandre | as [TK]D-Fender said |
| 17:08.58 | [TK]D-Fender | dandre: if you want to continue WITHING the local channel. |
| 17:09.09 | Kobaz | you don't need a local channel |
| 17:09.12 | Kobaz | you can use one if you want |
| 17:09.16 | [TK]D-Fender | dandre: If you want to continue on the outer, look at your Dial() |
| 17:09.27 | [TK]D-Fender | dandre: Of course I see no point to the local channel yet... |
| 17:09.34 | BlargMaN00 | if i use 'include => xxxx' in a context does it get evaluated before or after the dialplan in the context?? |
| 17:09.50 | Kobaz | BlargMaN00: after |
| 17:09.56 | *** join/#asterisk errotan (n=errotan@5403E45E.catv.pool.telekom.hu) |
| 17:12.15 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 17:13.54 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 17:14.03 | dandre | ok here http://pastebin.fr/5262 is what I am trying to do |
| 17:14.20 | brimstone | has anyone seen polycoms cutting off the time when using the idledisplay microbrowser? |
| 17:14.53 | dandre | if the queue timesout I never reach the netx Dial(Local/456/n) step |
| 17:15.10 | [TK]D-Fender | dandre: http://pastebin.fr/5263 |
| 17:15.39 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 17:16.10 | dandre | ok so it is not possible to use dial(Local...) is that case |
| 17:16.28 | [TK]D-Fender | dandre: Sure it is... its also a complete waste of time, effort, and CRD records. |
| 17:16.31 | [TK]D-Fender | CDR* |
| 17:16.38 | Kobaz | dandre: yeah it's an unneeded step |
| 17:17.01 | [TK]D-Fender | SEVERAL |
| 17:17.37 | dandre | in the real situation I use the dial(local...) because à have set some vaariables that I need |
| 17:18.18 | [TK]D-Fender | dandre: So set them there... |
| 17:19.52 | dandre | can I use a macro for instance my-queue instead of queue cmd? |
| 17:20.16 | [TK]D-Fender | dandre: HUH?! |
| 17:20.48 | [TK]D-Fender | dandre: Macro is jsut a way to get more dialplan. Again you keep adding useless steps in |
| 17:21.31 | dandre | that macro could set those variables so I don't have to duplicate there setting |
| 17:22.04 | [TK]D-Fender | dandre: You are tripping through Hypothetical Land. Show us something concrete |
| 17:23.33 | Kobaz | more cowbell |
| 17:23.36 | dandre | ok unfortunatly I must go, will show you more details tomorow |
| 17:24.07 | [TK]D-Fender | Kobaz: I'm already on guitar & keys... |
| 17:24.28 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 17:24.39 | Kobaz | do de do |
| 17:28.32 | jaytee | I went to see Crosby, Still and Nash last night. It was excellent |
| 17:28.37 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:29.04 | Kobaz | [Aug 6 13:21:26] WARNING[31243]: ael/pval.c:2521 check_pval_item: Warning: file /etc/asterisk/ael/hosted_dnc.ael, line 104-104: application call to GoSub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! |
| 17:29.12 | beek | hi jaytee -- I'm envious |
| 17:29.14 | Kobaz | i've been ignoring that error for the most part, but |
| 17:29.29 | jaytee | beek, be more envious. I won the tickets, box seats |
| 17:29.43 | beek | jaytee: Okay... now I'm GREEN with envy. |
| 17:29.45 | Kobaz | gosub *is* the intended behavior... unless there is a gosub in ael... hmm, lemme look |
| 17:30.10 | jaytee | beek, and envriromentalists the world over are thankful to you! |
| 17:30.35 | codefreeze-lap | Kobaz: dep. on version, macro calls are done with gosub. |
| 17:30.57 | beek | reaches over and powers down his unused PC |
| 17:31.10 | Kobaz | codefreeze-lap: yeah... but macros are suppposed to be depricated in 1.6 |
| 17:31.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:31.43 | Kobaz | does Macro in 1.6 actually use a gosub internally? |
| 17:31.49 | Kobaz | or is it still the same behavior |
| 17:31.59 | *** join/#asterisk Psychobilly (n=moi@79.103.38.123) |
| 17:32.16 | [TK]D-Fender | Kobaz: Same. Deprecation means it should jsut be GONE in the next |
| 17:33.14 | Kobaz | yeah |
| 17:33.19 | Kobaz | just making sure |
| 17:33.36 | Kobaz | so what should i use instead of GoSub in ael (since it complains) |
| 17:33.50 | Kobaz | even the wiki says to use gosub in ael |
| 17:34.22 | lmadsen | [TK]D-Fender: incorrect -- we don't remove functionality from Asterisk anymore, even if it is deprecated. |
| 17:34.30 | lmadsen | Macro == Macro |
| 17:34.42 | lmadsen | Kobaz: ^^ |
| 17:34.46 | Kobaz | yeah |
| 17:35.00 | lmadsen | (AEL is a different beast) |
| 17:35.07 | [TK]D-Fender | lmadsen: "Issues with commitment" :p |
| 17:35.16 | Kobaz | maybe i should just switch to lua |
| 17:35.20 | codefreeze-lap | Kobaz: AEL used Macro() to implement AEL macros in 1.4; in 1.6, AEL uses gosub underneath |
| 17:35.35 | [TK]D-Fender | lmadsen: I presume you trash things that dies when the core falls out from under it, right? |
| 17:35.47 | lmadsen | I don't know what that means |
| 17:36.00 | Kobaz | so like, it's old, rotting, and falling apart |
| 17:36.04 | Kobaz | would you still keep it? |
| 17:36.25 | lmadsen | just because you provide the functionality doesn't mean it is maintained, or even enabled by default |
| 17:36.29 | [TK]D-Fender | lmadsen: Well sure some apps can remain, but if the underlying bits change, that could break things. In those cases would it not be normal to drop it entirely? |
| 17:36.37 | lmadsen | it just means it exists for people for backwards compatibility reasons |
| 17:36.55 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:38.03 | Naikrovek | if i have two asterisk servers that i want to serve different extensions within the same phone system, do i use a trunk to connect them or something else? |
| 17:38.20 | Kobaz | so if i change a GoSub to Macro in ael.. the generated dialplan is Macro) |
| 17:38.22 | Kobaz | () |
| 17:38.32 | Kobaz | 13:35 <codefreeze-lap> Kobaz: AEL used Macro() to implement AEL macros in 1.4; in 1.6, AEL uses gosub underneath |
| 17:38.41 | Kobaz | so now, I'm not sure what you mean by that |
| 17:38.44 | lmadsen | huh? |
| 17:38.59 | lmadsen | it means if you do 'gosub' in AEL in 1.4, it is using Macro(). In 1.6, it uses GoSub() |
| 17:39.02 | lmadsen | (dialplan applications) |
| 17:39.14 | lmadsen | AEL gets "compiled" down into dialplan when it is loaded into memory |
| 17:39.18 | Kobaz | yeah i know |
| 17:39.22 | Kobaz | i'm looking at the resulting dialplan |
| 17:39.32 | lmadsen | right |
| 17:39.35 | Kobaz | and using Macro() in ael, in 1.6.0.10... makes a Macro() in the dialplan |
| 17:39.38 | Kobaz | not a gosub |
| 17:39.46 | lmadsen | that's odd |
| 17:39.52 | lmadsen | it should be using GoSub() |
| 17:39.57 | Kobaz | well it's not, heh |
| 17:40.13 | lmadsen | shrugs |
| 17:40.15 | lmadsen | I don't use AEL |
| 17:40.21 | lmadsen | I just use dialplan |
| 17:40.36 | Kobaz | context foo { _X! => { Macro(dialOut,foo); } } |
| 17:40.41 | Kobaz | <PROTECTED> |
| 17:40.52 | Kobaz | so you like pain and suffering then? |
| 17:40.52 | Kobaz | hehe |
| 17:41.00 | lmadsen | Kobaz: sounds like you do -- my stuff works |
| 17:41.04 | Kobaz | I got annoyed with dialplan after using it for 15 minutes |
| 17:41.12 | lmadsen | I've used it for 6 years. I'm good. |
| 17:41.13 | codefreeze-lap | uh, AEL has only one func def/call mech, macro(), and ¯oname() --- it compiles these into Macro() defs and calls. Case is important in the keywords. |
| 17:41.17 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 17:41.21 | Kobaz | lmadsen: oh no, there's no issues at all |
| 17:41.36 | Kobaz | lmadsen: i'm just wondering how to fix some warnings |
| 17:41.38 | lmadsen | use what works. I'll stick with dialplan. |
| 17:41.44 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 17:42.01 | Kobaz | lmadsen: it's like going back to qbasic... so... heh |
| 17:42.03 | Kobaz | oh well |
| 17:42.03 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 17:42.08 | lmadsen | shrugs |
| 17:42.14 | lmadsen | like I said, just use whatever you want |
| 17:42.17 | Kobaz | yeah |
| 17:42.19 | ariel_ | likes dial plan, as he is not a programmer... |
| 17:42.55 | eppigy | I WILL PROGRAM ABYTHING |
| 17:42.58 | Kobaz | okay |
| 17:43.01 | Kobaz | i see what you mean now |
| 17:43.06 | [TK]D-Fender | is a programmer and realizes that AEL only takes AWAY control from him.... |
| 17:43.32 | Kobaz | ¯o() calls, are converted into GoSub()'s instead of Macro() |
| 17:43.42 | Nugget | I write my dialplans in Lua so that I can maximize the code sharing in between my phone system and my world of warcraft addons. |
| 17:43.57 | Kobaz | [TK]D-Fender: how does ael take away control? you can still use all the constructs you can in dialplan, it's just more like a structured programming langauge than dialplan is |
| 17:44.05 | Kobaz | if (foo) { do stuff } |
| 17:44.11 | Kobaz | is *much* easier to read than |
| 17:44.27 | Kobaz | exten => 123123,GotoIf(1>5:4:8) |
| 17:44.51 | Kobaz | or whatever the syntax is, haven't used it in a while |
| 17:44.56 | lmadsen | who uses numbers anymore? |
| 17:44.57 | lmadsen | :) |
| 17:45.01 | ariel_ | I do |
| 17:45.03 | lmadsen | priority labels ftw |
| 17:45.08 | Kobaz | even if you use labels |
| 17:45.19 | Kobaz | it's still. exten => 123123,GotoIf(1>5:dostuff:dootherstuff) |
| 17:45.37 | lmadsen | I'm still a masochist and have adapted to reading dialplan quite quickly |
| 17:45.39 | Kobaz | and then you have linear blocks of code that are not always straightforward to follow |
| 17:45.40 | ariel_ | still has many 1.09 and 1.2's out there that need them.... but slowly moving off them... |
| 17:46.02 | *** join/#asterisk acanales (n=acanales@adsl-99-40-45-60.dsl.sndg02.sbcglobal.net) |
| 17:46.18 | Kobaz | but yeah... whatever works... but... heh |
| 17:46.26 | lmadsen | can't believe he's still having this conversation |
| 17:47.02 | Kobaz | so how bout that <insert sports team> |
| 17:47.30 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 17:47.41 | [TK]D-Fender | Kobaz>is *much* easier to read than <- not for me. And it wastes gusub nesting levels, adds bloat, another layer that can have bugs, etc. Adds nothing. I don't care how it renames the way I get places. |
| 17:47.43 | ariel_ | football starts this weekend.... |
| 17:48.03 | neurosys | sorry, this may be a dumb question, but on the digium site, I tell if Asterisk is GPL v2 or v3. The fact that it doesnt state it, would that mean its v3? |
| 17:48.12 | [TK]D-Fender | Kobaz: You'll run into possble things like AEL delimiters preventing you from doing stuff you could do directly in extensions.conf |
| 17:48.15 | lmadsen | GPLv2 |
| 17:48.20 | lmadsen | neurosys: check the code |
| 17:48.26 | [TK]D-Fender | Kobaz: ALSO, good luck with AEL in a DB :p |
| 17:48.32 | Kobaz | [TK]D-Fender: yeah i mean... there are some weird things here and there |
| 17:48.49 | neurosys | lmadsen, thanks :) |
| 17:48.53 | Kobaz | [TK]D-Fender: yeah I don't use ael from a db... i do generate dialplan from postgres.. but the dialplan that gets made is generally a bunch of gosubs |
| 17:48.57 | [TK]D-Fender | Kobaz: Adds nothing I care about. All it can do is fake out things. And debugging it is a bitch because of the back-compiling. |
| 17:49.20 | bpgoldsb | [TK]D-Fender, What parts would you like me to Pastebin? My ParkAndAnnounce call? |
| 17:49.36 | Kobaz | i've written some big stuff in ael, and haven't had any trouble debugging |
| 17:49.44 | lmadsen | this is a dumb conversation |
| 17:49.45 | *** join/#asterisk scunizi (n=scunizi@adsl-99-40-45-60.dsl.sndg02.sbcglobal.net) |
| 17:50.05 | bpgoldsb | (sorry it's a bit late, was eating some delicious mexican food) |
| 17:50.08 | lmadsen | Kobaz likes AEL, and I respect his decision. EOL. |
| 17:50.10 | [TK]D-Fender | bpgoldsb: the call, dialplan, configs, etc |
| 17:50.35 | [TK]D-Fender | lmadsen: I respect Kobaz, but his decisions still suck :p |
| 17:50.38 | Kobaz | hehe |
| 17:50.45 | neurosys | heh |
| 17:51.00 | [TK]D-Fender | :D |
| 17:51.15 | lmadsen | [TK]D-Fender: so does your mom |
| 17:51.43 | [TK]D-Fender | lmadsen: but.... I just want... |
| 17:51.43 | lmadsen | decides to interject some immaturity into the room |
| 17:52.35 | [TK]D-Fender | ... |
| 17:52.44 | bpgoldsb | [TK]D-Fender, http://pastebin.com/m692a2921 |
| 17:53.02 | lmadsen | [TK]D-Fender: ! ! ! |
| 17:53.02 | [TK]D-Fender | waits for the rest |
| 17:53.05 | [TK]D-Fender | \o/ |
| 17:55.26 | bpgoldsb | [TK]D-Fender, http://pastebin.com/m359163a9 (added the error message) |
| 17:56.58 | [TK]D-Fender | bpgoldsb: And quite true, the context you see listed in the error does NOT exist.... |
| 17:57.07 | [TK]D-Fender | bpgoldsb: look at it VERY CLOSELY |
| 17:58.50 | bpgoldsb | I give up. I don't see any typos etc |
| 17:58.57 | bpgoldsb | Got a hint? |
| 17:58.59 | [TK]D-Fender | -- Return Context: (parkedcallstimeout,0,0) ID: 112 <--- what context do you see? |
| 17:59.12 | [TK]D-Fender | bpgoldsb: Hint.... its between the brackets <- |
| 17:59.44 | bpgoldsb | parkedcallstimeout,0,0 |
| 17:59.48 | [TK]D-Fender | bpgoldsb: Do you see a context in your dialplan whit COMMAS in it? |
| 18:00.01 | [TK]D-Fender | with* |
| 18:00.15 | [TK]D-Fender | bpgoldsb: ParkAndAnnounce(announce:template|timeout|dial|[return_context]) <--- HMMMM |
| 18:00.20 | bpgoldsb | I did try that. |
| 18:00.25 | bpgoldsb | Er, wait. |
| 18:00.52 | bpgoldsb | ParkAndAnnounce(announce:template,timeout,dial[,return_context]): |
| 18:00.58 | [TK]D-Fender | :) |
| 18:01.02 | bpgoldsb | Thats what I have for core show application parkandannounce |
| 18:01.04 | [TK]D-Fender | Actually reading instructions! |
| 18:01.24 | [TK]D-Fender | bpgoldsb: Yes... do YOU see it asking for an EXTENSION, or PRIORITY? |
| 18:01.34 | bpgoldsb | Ah. |
| 18:01.45 | [TK]D-Fender | bpgoldsb: :p |
| 18:01.53 | bpgoldsb | I guess I just assumed context would allow for context + priority |
| 18:01.56 | bpgoldsb | + exten |
| 18:01.59 | [TK]D-Fender | loves wathing people invent syntax... |
| 18:02.06 | [TK]D-Fender | watching* |
| 18:02.52 | bpgoldsb | Heh, everyone's a noob at some point |
| 18:03.25 | [TK]D-Fender | bpgoldsb: Well there is a difference between not knowing things exist, and having the instructions, and adding parameters that clearly aren't listed ;) |
| 18:03.47 | bpgoldsb | unless you misunderstand the definition of a context ;) |
| 18:03.47 | [TK]D-Fender | bpgoldsb: Big Print wins this fight, time for the war!!!! |
| 18:04.00 | [TK]D-Fender | bpgoldsb: Not going there! |
| 18:04.38 | bpgoldsb | [TK]D-Fender, Thanks for the help :) |
| 18:05.59 | [TK]D-Fender | bpgoldsb: You're welcome |
| 18:06.57 | [TK]D-Fender | "Vonage Announces First Profit Ever" --- http://www.pcmag.com/article2/0,2817,2351215,00.asp |
| 18:07.00 | [TK]D-Fender | LOL!!! |
| 18:07.17 | lmadsen | wow... they finally made money> |
| 18:07.22 | *** join/#asterisk jtodd (i=gmw4428m@ns.fox-den.com) |
| 18:07.22 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:10.28 | Kobaz | haha |
| 18:10.30 | Kobaz | wow |
| 18:13.05 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 18:13.34 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:13.39 | ariel_ | wow vonage head office is in NJ HolmDel...... |
| 18:18.01 | nauticalthinker | I've made progress...current I can call from the f9600 to Asterisk, the phone rings, but hear static and then it drops |
| 18:18.29 | nauticalthinker | I get this in the cli: Ring/Off-hook in strange state 6 on channel 1 |
| 18:22.28 | *** join/#asterisk Mango (n=Mango@76-10-187-135.dsl.teksavvy.com) |
| 18:23.03 | Mango | Hello. I'm still pondering the idea of MWI propagation. |
| 18:23.20 | Mango | I've been a programmer for years but I have never written an Asterisk module. |
| 18:23.34 | Mango | How easy would it be to write a module that captured SIP NOTIFY messages so I could parse them? |
| 18:23.53 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 18:24.42 | bpgoldsb | [TK]D-Fender, return_context: The goto-style label to jump the call back into after timeout. Default <priority+1>. This doesn't say it has to be a context. And in fact, the examples on http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce show you should be able to do context+exten+priority |
| 18:25.22 | bpgoldsb | I could be misunderstanding it, but I would like to be sure. |
| 18:26.35 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:27.01 | timeshell_atwork | What's better on a TDM analog card? Hardware or software echo cancellation? |
| 18:27.36 | ariel_ | hardware |
| 18:28.15 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:30.10 | *** join/#asterisk btsteve (n=tstevens@67.201.69.212) |
| 18:30.15 | [TK]D-Fender | bpgoldsb: WIKI = LOL |
| 18:35.20 | ethicx | anyone has a Linksys WIP330 with asterisk? I'm thinking of buying one.. |
| 18:35.38 | [TK]D-Fender | ethicx: EW!!!!!!! |
| 18:36.13 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:36.19 | ethicx | lol |
| 18:36.22 | ethicx | really |
| 18:36.25 | ethicx | ok |
| 18:36.37 | [TK]D-Fender | ~wifivoip |
| 18:36.38 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 18:36.46 | ethicx | good to know.. before spending bucks =D |
| 18:37.21 | defsdoor | gets on fine with a nokia n95 with voip over wifi |
| 18:38.21 | timeshell_atwork | ariel_ Problem is we're experiencing what sounds like vox issues with our polycoms. When we talk on our side, it's like those push to talk thingies where you then can't hear the other side until you stop talking |
| 18:39.24 | timeshell_atwork | ethicx I"m using a Snom M3 DECT w/VoIP base.. Not too bad. |
| 18:39.26 | [TK]D-Fender | ethicx: Note "Notice" that... and Linsys SPA / PAP2 series is fine |
| 18:39.29 | [TK]D-Fender | don't* |
| 18:39.41 | citywok | has anybody played with asterisk in Hyper-v? |
| 18:39.56 | [TK]D-Fender | Linksys* |
| 18:40.12 | ethicx | ur tha MAN |
| 18:40.23 | ariel_ | Polycom 8020/8030 are good wifi phones but they require an SVP server... |
| 18:40.43 | Elwell | needs to find the correct settings for SPA3102 and his local telco |
| 18:43.15 | *** join/#asterisk qdk (n=qdk@0x573d8ce3.bynqu1.dynamic.dsl.tele.dk) |
| 18:45.29 | [TK]D-Fender | If you're cordless itdoesn't matter where the base is. Of course you might want a phone with an extra charging stand |
| 18:45.41 | [TK]D-Fender | ethicx: and stop using IRC NOTICEs for this |
| 18:46.43 | ethicx | got it. |
| 18:47.30 | ariel_ | Elwell: at home I am using an SPA3102 with a Vtech wireless phones. 3 phones one main base and 2 remote charging base, And I know it's working great due to wife does not know she is on voip service for 90% of her calls.. |
| 18:48.21 | Elwell | ariel_: yeah I had it set fine for UK but I havent seen swiss settings yet |
| 18:48.54 | ariel_ | can't help you there, I am in the US and it's very much default settings... |
| 18:50.11 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 18:51.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:55.00 | box2 | sip over wifi is a sad thing :( |
| 18:55.17 | ariel_ | why? |
| 18:55.26 | box2 | i'm getting huge amount of packetloss |
| 18:55.32 | box2 | can't get a DTMF tone through |
| 18:56.48 | ariel_ | ahh, We have installed some really nice Wifi phones on 3 ships now, We have 7 more ships with orders to do... All on Polycom 8020 with an SVP. 80 to 150 phones per ship depending on size, works really well...... |
| 18:58.19 | box2 | hmmm |
| 18:58.31 | box2 | you don't get radio collision with lots of phones on at once? |
| 18:58.45 | TSM | is anyone up on UK PRI settings in *, having issue with correct CLI |
| 18:59.02 | ariel_ | no we move between 8 to 20 AP's around the ship without issues. |
| 18:59.24 | box2 | hmmm |
| 18:59.41 | *** join/#asterisk propellerhead (n=yogurt2u@host68.190-31-73.telecom.net.ar) |
| 19:00.15 | ariel_ | got to go, bbl |
| 19:03.15 | ethicx | ariel_ what model of vtech phones you got at home? |
| 19:07.14 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 19:12.12 | generalhan | hey guys, quick question ... can i use priority labels in a GoTo() from a different context? like Goto(context,s,n(start)), i use them in the same context all the time with just Goto(start), but never tried from a different context |
| 19:14.04 | *** join/#asterisk btsteve (n=tstevens@67.201.69.212) |
| 19:14.37 | timeshell_atwork | Is there a way to disable hardware echo cancellation without removing the echo cancellation module from the TDM card? |
| 19:14.45 | Naikrovek | citywok: i have in vmware |
| 19:15.03 | Naikrovek | have a hyper-v server, but it's just about out of memory |
| 19:15.17 | timeshell_atwork | In favor of using software cancellation. |
| 19:15.26 | kaldemar | generalhan: you can use goto like Goto(context,extension,label) |
| 19:15.54 | generalhan | kaldemar: perfect, thats what i needed to know ! |
| 19:15.56 | kaldemar | generalhan: using your example, Goto(context,s,start) |
| 19:15.56 | generalhan | thanks |
| 19:19.30 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 19:21.36 | *** join/#asterisk andres833 (n=andres83@190.156.199.40) |
| 19:25.20 | *** join/#asterisk acanales (n=acanales@adsl-99-40-45-60.dsl.sndg02.sbcglobal.net) |
| 19:28.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:30.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:34.16 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 19:36.22 | *** join/#asterisk MACscr1 (n=Mark@98.214.100.212) |
| 19:37.11 | MACscr1 | ok, so digium is beta testing a skype addon, its free for now, but the question is, how much on Sept 1st? |
| 19:38.28 | Psychobilly | i dont think ti will be ready to be sold on sept 1 :P |
| 19:38.44 | Corydon76-dig | MACscr1: Nobody who knows is actually here |
| 19:38.46 | MACscr1 | well, I more meant after the beta period |
| 19:38.51 | MACscr1 | yeah, figured |
| 19:39.00 | Corydon76-dig | MACscr1: for that matter, we don't even know if it's set in stone yet |
| 19:39.20 | *** join/#asterisk cb` (n=cb@72.37.252.50) |
| 19:39.23 | *** join/#asterisk tharrison (n=chatzill@static-67-62-126-158.t1.cavtel.net) |
| 19:39.24 | Corydon76-dig | Whatever the price currently is upstairs, it could still change |
| 19:39.40 | *** part/#asterisk cb` (n=cb@72.37.252.50) |
| 19:40.06 | MACscr1 | of course, was more just hoping for an idea |
| 19:40.37 | tharrison | Anyone ever have a Polycom 4000 conf phone flake out on them? |
| 19:41.37 | tharrison | The polycom just immedately flashes lights when I power it on. not the normal boot sequence I expect from a polycom |
| 19:41.58 | Corydon76-dig | MACscr1: you could try calling Digium sales |
| 19:42.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:44.05 | TSM | the the polycom show any boot screen? |
| 19:45.24 | tharrison | TSM: nope. Just immedately starts flashing red lights. |
| 19:45.53 | tharrison | TSM: I've handled plenty of 500's before. This is my first 4000. I think its dead, but I'm not sure. |
| 19:46.22 | tharrison | (immedately == 0 seconds after power) |
| 19:46.38 | *** join/#asterisk NickRios05 (n=unicall2@static-200-105-140-237.acelerate.net) |
| 19:46.42 | TSM | tharrison: weird, prolly dead then, was it working before? |
| 19:46.48 | *** join/#asterisk rseste (i=c93f9ac2@gateway/web/freenode/x-iwmephmpgvmcrwzj) |
| 19:47.18 | TSM | tharrison: usualy they should show up a booting screen after flashing lights, but you know that from past poly experiance |
| 19:47.34 | tharrison | It was a few weeks ago (in a under-used conf room). When I went to check it recently, it was doing this fast flashing light thing, so I unplugged it until I I had time to look at it . |
| 19:48.17 | tharrison | yeah, but with the boot stuff, you get status on the lcd screen ususally. |
| 19:49.04 | tharrison | Nothing is showing on this phone's lcd. |
| 19:49.41 | tharrison | I was kinda holding out hope that maybe there was a firmware reset or something I could do. |
| 19:53.47 | MACscr1 | uh, is it me or is the "voice of asterisk" just sound like its digitally created. Is it? |
| 19:54.26 | kaldemar | no, it's not. |
| 19:56.30 | *** join/#asterisk rgsteele||work (n=rgsteele@207.106.239.81) |
| 19:56.52 | rgsteele||work | Can you designate a zaptel card interface (for POTS lines) as an 'outbound' interface only? |
| 19:56.54 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 19:56.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:57.10 | citywok | ~phones |
| 19:57.10 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:58.17 | [TK]D-Fender | rgsteele||work: Calls get processed in your dialplan however you tell them to. |
| 19:58.34 | jaytee | rgsteele||work, not really. you'd have to tell your telco you don't want calls incoming to that line. you could set the incoming context to just hangup andy incoming calls but that wouldn't block it from recieving them |
| 19:58.58 | [TK]D-Fender | rgsteele||work: YES, see above |
| 19:59.24 | *** join/#asterisk edguy3 (n=edguy@c-98-221-27-224.hsd1.nj.comcast.net) |
| 19:59.45 | [TK]D-Fender | jaytee: SILLY GOOSE |
| 20:00.13 | ariel_ | tharrison: there is a setting you can do on that phone to reset to factory default. you might have to get that from Polycom. We had one do the same thing many years ago. |
| 20:00.20 | jaytee | [TK]D-Fender, you know what I meant |
| 20:00.50 | [TK]D-Fender | jaytee: Who said he had to hangup? :) |
| 20:01.09 | tharrison | ariel_: usually those depend on the phone working enough so you can do a special keystroke or three finger salute. |
| 20:01.41 | tharrison | TSM: ariel_: I just found someone else that described the same problem... of course, there was no solution listed: http://www.fixya.com/PostAnswer.aspx?thid=1054398&prdid=521003&ref=unsl |
| 20:01.55 | ariel_ | tharrison: if it's the same 4000 no there is a way to hold some of the keys down together on power up for it. |
| 20:02.03 | jaytee | [TK]D-Fender, well he doesn't but you don't want any incoming call attempts if you want dedicated outbound only trunks so the best thing is to order a plain POTS line that is non-DID. |
| 20:02.14 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 20:02.34 | tharrison | ariel_: ok.... I'll go see if the polycom 4000 manual has anything. |
| 20:04.36 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 20:07.26 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:08.24 | *** join/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 20:08.49 | dustybin | are there any bsd alternatives to asterisk? |
| 20:10.41 | Nugget | asterisk. |
| 20:10.52 | Nugget | or do you mean licensing? |
| 20:11.59 | dustybin | nope its ok |
| 20:12.32 | Nugget | asterisk will run fine on bsd as long as you don't care (much) about hardware support for stuff like PRI cards. |
| 20:13.03 | Nugget | I prefer to just suck it up and deal with linux crappiness just because it's the path of least resistance, but you can go the bsd route and it'll work mostly well |
| 20:13.24 | dustybin | if i use asterisk at home, could i put on some of my favourite tracks while my friends are on hold? |
| 20:13.32 | Nugget | for a strictly voip task it's fine |
| 20:14.02 | Nugget | or if you opt for external hardware (like a linksys pap for pots connectivity, instead of a card) |
| 20:14.39 | dustybin | also, does asterisk support conference calls, ie. could 3 of my friends call me at the same time, and we all speak together? |
| 20:14.45 | Nugget | yes on both. |
| 20:14.51 | dustybin | bloody heck :D |
| 20:15.15 | dustybin | is asterisk CPU hungry? |
| 20:15.16 | Nugget | assuming voip. obviously if you're talking about plugging asterisk into a phone line then that phone line isn't going to be able to do two calls at once. |
| 20:15.27 | dustybin | yes VOIP |
| 20:15.48 | afink | no not cpu hungry |
| 20:15.57 | dustybin | double ace :D |
| 20:16.15 | dustybin | i am going to setup a small part-time business, i will need some kind of system to record calls |
| 20:16.24 | dustybin | this sounds perfect |
| 20:16.31 | afink | it does that too |
| 20:16.37 | dustybin | ace :D |
| 20:17.20 | dustybin | i wonder if there is some kind of 'pay as you go' VOIP service for UK users |
| 20:17.20 | generalhan | now that i have my GotoIfTime stuff all setup, i had a different thought ... is there a way to run a macro under time contraints ? like MacroIfTime LOL |
| 20:18.22 | [TK]D-Fender | ~itsplist-uk |
| 20:18.23 | infobot | from memory, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 20:18.25 | [TK]D-Fender | dustybin: ^^^^^ |
| 20:18.34 | dustybin | thanks :D |
| 20:18.51 | [TK]D-Fender | generalhan: No. |
| 20:19.42 | dustybin | feels excited |
| 20:19.44 | generalhan | :( im trying to shorten up my holiday time check context. instead of having a context for each holiday i wanted to do a generic holiday context using variables for which holiday it is. |
| 20:19.50 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:20.32 | kaldemar | generalhan: core show application ExecIfTime |
| 20:21.05 | [TK]D-Fender | Hrm.... overlooked one... |
| 20:21.25 | generalhan | kaldemar: looking at it now, thanks ! |
| 20:22.04 | citywok | What is the recommendation for a cordless phone? I see that aastra is high on the list of recommendations, and they have their 480i CT which you can use a cordless handset with -- what kind of coverage does it get? 200' from the base station through a handful of hollow walls okay? |
| 20:23.26 | [TK]D-Fender | citywok: More than... just remember that the handset is married to the base. BOTH will ring |
| 20:24.03 | citywok | that's not an issue, as long as the volume on the base can be turned down so as to not annoy people there if the receptionist has walked off and isn't answering |
| 20:24.27 | citywok | if we married two handsets (says it supports 4?) would they all then be the same as well? so all 4 would ring? |
| 20:24.44 | dustybin | can some cellphones connect into a wireless network and communicate with asterisk? |
| 20:24.48 | generalhan | kaldemar: what is an example of an 'application' that would run if the designated time is a match ? |
| 20:25.05 | Chainsaw | dustybin: I've seen some cellphone/DECT hybrids, but they're rare. |
| 20:25.11 | dustybin | ok |
| 20:25.44 | Chainsaw | dustybin: Some cellphones have WiFi however, and I suppose you could install a SIP application on them. |
| 20:25.57 | dustybin | aye ok |
| 20:26.06 | [TK]D-Fender | checkout time, later all |
| 20:26.39 | rgsteele||work | Later TK |
| 20:26.40 | ariel_ | tzafrir_laptop: how much ram does the new liveCD 2.0 require? |
| 20:27.07 | kaldemar | generalhan: any dialplan application, such as Macro |
| 20:27.19 | generalhan | ariel_: thanks for the heads up on those Xorcom devices! mine should be here on monday for initial testing ! |
| 20:27.50 | ariel_ | Nice, I just got a 2 PRI/8 FXO unit for testing... |
| 20:28.11 | generalhan | ariel_: lol, just got the 2PRI/8FXS model |
| 20:28.37 | jaytee | generalhan, there is also GotoIfTime for time based decisions in your dialplan |
| 20:29.22 | ariel_ | I have a 8 FXO and a 24 FXS unit already in production, They work |
| 20:29.39 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 20:29.56 | generalhan | jaytee: i cant pass variables down to another context with GotoIfTime |
| 20:30.46 | *** join/#asterisk many (n=many@dslb-094-217-197-170.pools.arcor-ip.net) |
| 20:31.11 | generalhan | kaldemar: so, ExecIf(*|*|*|*?Macro(whatever,Variable1,Variable2) <-- just like that ? that seems to simple, like i should have been using this ages ago |
| 20:31.33 | generalhan | opos, ExecIfTime, rather than ExecIf - sorry |
| 20:31.44 | jaytee | generalhan, ah ok. didn't see that requirement further back. you can pass variable values to another context or channel by prefacing them with an underscore when you set their value or reset their value and a double underscore will inherit until the call is terminated regardless of what context it goes to. |
| 20:33.12 | kaldemar | generalhan: yes, just like that. add another ) in the end to correct the syntax. |
| 20:33.35 | generalhan | kaldemar: wow, i really cant believe i didnt find out about this sooner. anyway thanks for the help ! |
| 20:38.55 | *** join/#asterisk SBBsupport (n=chatzill@208.50.100.60) |
| 20:39.56 | *** join/#asterisk andres833 (n=andres83@190.156.199.40) |
| 20:41.20 | *** join/#asterisk Carlos_PHX (n=carlos@ip68-3-162-244.ph.ph.cox.net) |
| 20:41.59 | *** join/#asterisk talntid (n=eric@66-208-251-170-Washington.hfc.comcastbusiness.net) |
| 20:42.40 | s14ck | \/hola |
| 20:42.46 | talntid | is ChanSpy included by default on Asterisk? or do I need to configure it? |
| 20:42.53 | s14ck | <PROTECTED> |
| 20:44.00 | kaldemar | talntid: yes it is included by default and yes you need to configure it. |
| 20:44.55 | talntid | okies :) |
| 20:45.12 | talntid | can you point me in the right direction? I added "exten => 556,1,ChanSpy(scan) " to my extensions.conf |
| 20:46.13 | talntid | then, it just does a lot of beeps |
| 20:46.23 | Carlos_PHX | According to the (very old) Wiki entry, chanspy is not included by default in versions after 8/2004. |
| 20:47.43 | *** join/#asterisk dandate2 (n=mangy@173-11-82-122-SFBA.hfc.comcastbusiness.net) |
| 20:47.55 | dandate2 | if i'm going to create my pbx to record calls for a call-center will it require a raid array? |
| 20:47.55 | kaldemar | talntid: "core show application ChanSpy" in CLI will show you the syntax and options |
| 20:48.14 | Carlos_PHX | dandate2: How many concurrent calls? |
| 20:48.21 | dandate2 | up to 30 channels at a time |
| 20:48.23 | Carlos_PHX | And what CODEC? |
| 20:48.23 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 20:48.24 | dandate2 | so 15 |
| 20:48.28 | dandate2 | g729 or gsm |
| 20:48.39 | Carlos_PHX | That's no problem on a single SATA drive. |
| 20:48.43 | dandate2 | k |
| 20:49.02 | Carlos_PHX | I do 60+ GSM like that. |
| 20:49.06 | talntid | extenspy is more useful i think |
| 20:49.12 | dandate2 | do you prefer GSM to g729? |
| 20:49.22 | dandate2 | i didn't pay for the license yet so mabyei should switch heh |
| 20:49.27 | Carlos_PHX | I have no preference other than no license for GSM. |
| 20:49.33 | Carlos_PHX | 729 sounds better. |
| 20:49.35 | Carlos_PHX | Is smaller. |
| 20:49.52 | dandate2 | hard for me to tell my reps are in the phillipines they all sound like hell anyway |
| 20:49.57 | dandate2 | were powering this on a comcast cable modem... |
| 20:49.58 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.162.64) |
| 20:50.08 | dandate2 | anyone know where i can rent a pbx hooked up to t1? |
| 20:50.13 | Carlos_PHX | Well then, just use 8k.... :-p |
| 20:50.25 | Carlos_PHX | Rent a PBX...?? |
| 20:50.32 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 20:50.41 | kaldemar | Carlos_PHX: the wiki is wrong on that one then |
| 20:50.41 | dandate2 | yeah noone rents em out? |
| 20:50.55 | dandate2 | i know like a pbx in a data center hooked up to t1 |
| 20:50.58 | Carlos_PHX | What do you mean by rent? Move it to your location? |
| 20:51.02 | [TK]D-Fender | Why does it require T1? Voice? Data? |
| 20:51.04 | Carlos_PHX | Oh, you want a SIP provider? |
| 20:51.04 | dandate2 | noo |
| 20:51.15 | dandate2 | its voice for call center, 20 agents |
| 20:51.23 | dandate2 | currently using comcast but its crap |
| 20:51.26 | Carlos_PHX | So like we have Asterisk boxes with PRIs in data centers and we sell minutes. |
| 20:51.37 | dandate2 | i like the way the ebay and paypal customer support reps sound tho heh |
| 20:51.50 | Carlos_PHX | Or you saying you want to rent a dedicated Asterisk server? |
| 20:51.59 | dandate2 | yes thats correct, i wouldn't want to pay for minutes i have DID already |
| 20:52.19 | Carlos_PHX | WTF? So you're going to point the DID at a TDM phone? |
| 20:52.34 | dandate2 | sorry TDM? |
| 20:52.44 | Carlos_PHX | Regular digital line. PRI/T1 |
| 20:53.06 | dandate2 | is that not advisable? its cheap did thats for sure |
| 20:53.14 | Carlos_PHX | How cheap? |
| 20:53.15 | dandate2 | didforsale.com |
| 20:53.23 | dandate2 | $8/mo for unlimmited miniutes 20 channels |
| 20:53.25 | dandate2 | HELLLLLA cheap |
| 20:53.30 | dandate2 | i couldn't even get that price at didx |
| 20:53.44 | Carlos_PHX | So you just need a server to point them to? |
| 20:53.47 | dandate2 | yes |
| 20:53.48 | Carlos_PHX | Then you don't need T1. |
| 20:53.56 | dandate2 | well wait heres the thing |
| 20:53.59 | dandate2 | all my reps are remote |
| 20:54.01 | dandate2 | theres no local office |
| 20:54.08 | Carlos_PHX | Ah, see. |
| 20:54.17 | Carlos_PHX | And all your time is inbound? |
| 20:54.20 | dandate2 | yes |
| 20:54.25 | dandate2 | unless they have to call back a declined sale |
| 20:55.07 | Carlos_PHX | $8/mo for 20 concurrent unlimited you say. |
| 20:55.12 | dandate2 | hellllla cheap |
| 20:55.16 | Carlos_PHX | Not possible. |
| 20:55.20 | dandate2 | i'm serious |
| 20:55.24 | dandate2 | theres a $5 setup charge |
| 20:55.26 | Carlos_PHX | I mean, sure, they will sell it, but not possible to make money at it. |
| 20:55.40 | dandate2 | these guys making no money |
| 20:55.57 | Carlos_PHX | They're charging a fraction of what we pay our wholesale carriers. |
| 20:56.25 | dandate2 | the 1800 # is $5/mo + 3 cents a min |
| 20:56.35 | Carlos_PHX | That's normal. |
| 20:56.39 | Carlos_PHX | We pay way less. |
| 20:56.54 | Carlos_PHX | Nobody can do 20 concurrent for $8/mo. |
| 20:57.01 | dandate2 | no no not concurrent |
| 20:57.04 | dandate2 | 20 channels |
| 20:57.07 | dandate2 | u get 10 concurrent i believe |
| 20:57.10 | Carlos_PHX | The difference is? |
| 20:57.14 | dandate2 | heh |
| 20:57.23 | dandate2 | just look at their website |
| 20:57.30 | dandate2 | no money |
| 20:57.56 | Carlos_PHX | I don't dispute that they advertise it, but it's not possible to deliver that profitably. |
| 20:58.58 | Carlos_PHX | I could switch a few thousand numbers to them and make a killing, but then I'd be screwed when they go under. |
| 20:59.06 | dandate2 | heh |
| 20:59.14 | dandate2 | mabye they make money off blackbilling |
| 20:59.23 | dandate2 | they are just waiting to find out that i am using the service for business use |
| 20:59.28 | dandate2 | i don't even know if that is against the TOS |
| 20:59.41 | dandate2 | i didn't see anything about it but i did see it in other providers |
| 20:59.46 | Carlos_PHX | Must be ok, what residential customer could have 10 calls? |
| 20:59.50 | dandate2 | they siad they would bill u $100 per day lol |
| 21:00.17 | dandate2 | they've been nice to me, they saw my 1000 calls per day and told me i was popular |
| 21:03.17 | Carlos_PHX | So anyway you need a server with a static IP. You can buy a server and send it to colo, or rent a server in colo, or get a "virtual" server. The latter option only works if they can give you real-time performance. |
| 21:03.45 | Carlos_PHX | Or just buy minute usage from a provider like us or thousands of others. |
| 21:07.12 | talntid | anyone here interested in helping me get chanspy or extenspy working? it's a call center environment, and I'm just not sure what I'm doing wrong |
| 21:08.01 | Carlos_PHX | Are the channels SIP, or Zap, or...? And what's the error or result? |
| 21:08.28 | talntid | I am not getting an error, or I do not know how to read it. When I dial the exension, it just beeps, like it's scanning around |
| 21:08.35 | talntid | they are SIP phones |
| 21:08.42 | talntid | but they go out over a zaptel card |
| 21:08.52 | talntid | sangoma, to be exact |
| 21:09.05 | Carlos_PHX | Have you tried a specific SIP and/or Zap channel? |
| 21:09.21 | talntid | i tried SIP/htdsk5001 |
| 21:09.27 | Carlos_PHX | exten => _*XXX,1,ChanSpy(DAHDI/${EXTEN:2},q) |
| 21:09.29 | talntid | but, not sure if that's what it needs |
| 21:10.09 | Carlos_PHX | htdsk5001 is the SIP registration name? |
| 21:10.37 | talntid | here's an example: htdsk5015/htdsk5015 10.21.5.218 D 5060 OK (54 ms) |
| 21:10.45 | talntid | from sip show peers |
| 21:11.42 | talntid | this * server, is linked to a dedicated * server, that interfaces with the sangoma |
| 21:11.50 | Carlos_PHX | You probably want extenspy. I've only used Chanspy on Zap/DAHDI channels. |
| 21:12.03 | talntid | ah, ok :) |
| 21:12.12 | talntid | i'v been googling both, just been trying to figure out what's possible |
| 21:12.20 | Carlos_PHX | In theory chanspy should work, but it's poorly documented from what I've seen. |
| 21:12.30 | talntid | i concur with that. heh |
| 21:12.37 | Carlos_PHX | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy |
| 21:12.40 | dandate2 | carlos when you said colo did you mean colorado? |
| 21:12.59 | Carlos_PHX | Yes, the servers stay nice and cool there. |
| 21:13.00 | talntid | no, he meant colocation |
| 21:13.05 | talntid | lol |
| 21:13.05 | Carlos_PHX | It's really the only place to run server. |
| 21:13.11 | dandate2 | lol |
| 21:13.18 | dandate2 | yes that is what i was looking for |
| 21:13.23 | dandate2 | a static ip server for rent colo |
| 21:13.40 | Carlos_PHX | Colocation is where you can send a server to live in a facility with dedicated power, bandwidth, etc. |
| 21:13.52 | dandate2 | where do i find that? are any of them specifically geared for voip? |
| 21:13.54 | Carlos_PHX | You can often rent the server itsefl too. |
| 21:14.12 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 21:14.13 | dandate2 | yes i may need a colo |
| 21:14.14 | Carlos_PHX | We are, but we're not cheap, and I get the impression you're trying to do this cheap. |
| 21:14.28 | dandate2 | no this is like emergency backup or full on upgrade |
| 21:14.41 | dandate2 | see my current system is alright but i'm moving to phillipines on the 14th and leaving my wife here to guard the server |
| 21:14.51 | dandate2 | once she finds out i got 3 more wives she might try to shut my business down in revenge |
| 21:14.53 | talntid | Carlos_PHX: extenspy isn't very well documented very well either, it seems |
| 21:14.57 | *** join/#asterisk pwden (n=domin8@ool-ad03dcac.dyn.optonline.net) |
| 21:15.03 | talntid | unless you know something I don't :) |
| 21:15.04 | dandate2 | so imma need a colo ready at any given moment to just swithc my DID over to |
| 21:15.12 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 21:15.21 | dandate2 | but if the price is right i'll replace my pbx with a colo just for the t1 |
| 21:15.32 | Carlos_PHX | Ok, now I feel like I'm being trolled. |
| 21:15.33 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:15.40 | talntid | by me? |
| 21:15.42 | Carlos_PHX | Four wives? |
| 21:15.52 | dandate2 | yeah thats how u run a business in todays market |
| 21:15.53 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 21:15.59 | dandate2 | u get wives cuz they work for free |
| 21:16.10 | dandate2 | check out what happened to british airways, they told all their workers they are gunna work for free for 1 month lol |
| 21:16.29 | Carlos_PHX | Well, thanks for the amusement, gotta go see a customer. |
| 21:16.35 | dandate2 | =) |
| 21:16.52 | Carlos_PHX | Anyway if you want to drop me a line for a quote, carlos@televolve.com |
| 21:16.57 | dandate2 | ok |
| 21:25.41 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:25.54 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-d501d80be968079a) |
| 21:28.59 | Defraz | seem to be seeing this a lot " channel.c: Didn't get a frame from channel" and I am getting some dropped calls. |
| 21:29.56 | Defraz | Any ideas where to look, it isn't every call just started happening more lately. |
| 21:42.07 | timeshell_atwork | Why does DAHDI say that == Everyone is busy/congested at this time (1:0/0/1) when the lines are not in use!? |
| 21:44.29 | TSM2 | wat type of lines |
| 21:46.09 | [TK]D-Fender | timeshell_atwork: Would be nice to see your configs and the SOURCE of that error... |
| 21:46.23 | timeshell_atwork | Yah |
| 21:46.25 | timeshell_atwork | just a sec |
| 21:46.39 | timeshell_atwork | It's the same problem I had that I thought I had fixed. |
| 21:46.44 | timeshell_atwork | I'm investigating another option |
| 21:47.19 | timeshell_atwork | crap |
| 21:47.26 | timeshell_atwork | Now I'm panicking |
| 21:47.27 | *** join/#asterisk Alfio (n=Amunoz@adsl-51-26.tricom.net) |
| 21:53.06 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:53.07 | timeshell_atwork | [TK]D-Fender http://pastebin.com/m5705e2aa |
| 21:54.10 | timeshell_atwork | Here's the interesting thing [TK]D-Fender. I haven't changed the configuration. I just rebooted the server. |
| 21:54.10 | *** join/#asterisk manxpower (n=eric@69.73.94.162) |
| 21:54.48 | timeshell_atwork | It's the same configuration |
| 21:55.04 | drmessano | You need to rebuild DAHDI |
| 21:55.14 | timeshell_atwork | drmessano I did |
| 21:55.16 | timeshell_atwork | 4 times |
| 21:55.37 | timeshell_atwork | I tried rebuilding 2.1.0.4 twice and 2.2.0.2 twice |
| 21:55.43 | drmessano | What Distro? |
| 21:55.48 | timeshell_atwork | CentOS |
| 21:56.02 | timeshell_atwork | 64 bit |
| 21:56.23 | drmessano | rpm -q kernel |
| 21:56.26 | drmessano | rpm -q kernel-devel |
| 21:56.30 | drmessano | uname -r |
| 21:56.41 | drmessano | rpm -e your old kernel and kernel-devel |
| 21:56.50 | drmessano | make sure you have the new kernel-devel |
| 21:56.55 | drmessano | and rebuild DAHDI |
| 21:56.56 | timeshell_atwork | kernel-2.6.18-53.el5 |
| 21:56.57 | timeshell_atwork | kernel-2.6.18-128.1.10.el5 |
| 21:57.16 | timeshell_atwork | wtf |
| 21:57.30 | timeshell_atwork | 2.6.18-128.1.10.el5 |
| 21:57.48 | drmessano | What about it? |
| 21:57.56 | timeshell_atwork | kernel-devel-2.6.18-53.el5 |
| 21:57.58 | timeshell_atwork | kernel-devel-2.6.18-128.1.10.el5 |
| 21:57.59 | timeshell_atwork | lol |
| 21:58.00 | drmessano | which -devel do you have? |
| 21:58.01 | timeshell_atwork | nothing |
| 21:58.04 | timeshell_atwork | I have 2 |
| 21:58.13 | drmessano | ok |
| 21:58.24 | timeshell_atwork | So, it's confusing dahdi? |
| 21:58.28 | timeshell_atwork | That's really scarry |
| 21:58.29 | drmessano | rpm -e kernel-2.6.18-53.el5 |
| 21:58.50 | drmessano | rpm -e kernel-devel-2.6.18-53.el5 |
| 21:59.07 | drmessano | make sure you ./configure dahdi |
| 21:59.13 | timeshell_atwork | yah |
| 21:59.34 | drmessano | Then you'll be fine |
| 21:59.55 | timeshell_atwork | no ./configure for dahdi-linux |
| 21:59.57 | drmessano | Undo anything you tried to fix in your panix |
| 21:59.58 | timeshell_atwork | just dahdi-tools |
| 22:00.00 | drmessano | panic |
| 22:00.06 | timeshell_atwork | Haven't done nothing |
| 22:00.10 | drmessano | I use the bundle |
| 22:00.37 | timeshell_atwork | BUt I have a back up of my last good config anyway |
| 22:02.15 | *** join/#asterisk jtodd (i=w7btnmwa@ns.fox-den.com) |
| 22:02.15 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:02.36 | timeshell_atwork | drmessano No joy :( |
| 22:02.59 | manxpower | It's pretty easy to think you applied a change, but discover the next time you reboot that the change did not apply until the reboot and then you realize you screwed something up. |
| 22:03.25 | timeshell_atwork | Means? |
| 22:03.36 | manxpower | I had that happen recently with a customer. He realized the wrong IP address was in the boot scripts. |
| 22:04.06 | manxpower | After he had to drive 15 miles into the office to figure out why the server did not come back online. |
| 22:04.26 | timeshell_atwork | So, you're telling me to reboot first? |
| 22:04.28 | timeshell_atwork | :p |
| 22:05.15 | manxpower | no, reboot last, after you 've made all other changes, just to make sure. |
| 22:05.21 | timeshell_atwork | Stopping the dahdi and asterisk services not good enough? |
| 22:07.31 | manxpower | You sure have a lot of faith in being able to REALLY TEST it without a reboot. |
| 22:08.06 | timeshell_atwork | https://issues.asterisk.org/view.php?id=15099 |
| 22:08.11 | timeshell_atwork | What do you make of that? |
| 22:08.34 | timeshell_atwork | How about power off the machine for good measure? |
| 22:08.58 | manxpower | looks to me like dahdi is not starting on boot |
| 22:09.07 | timeshell_atwork | Nope |
| 22:09.09 | timeshell_atwork | It starts |
| 22:09.38 | drmessano | "This looks like a support request. Please use the asterisk-users mailing list for this kind of support. Thanks!" |
| 22:09.51 | manxpower | timeshell_atwork: I've seen issues that happen only during a cold boot and issues that happen only during a warm boot. |
| 22:10.35 | manxpower | Mostly with older cards and older drivers. Sangoma drivers are known to under some situations cause a kernel panic when you try to warm boot. |
| 22:10.54 | drmessano | You could always wipe the source directory for DAHDI, download and install again |
| 22:11.40 | timeshell_atwork | Dahdi status after boot http://pastebin.com/m42c8b6c |
| 22:11.46 | manxpower | Timeshell call me! I want to make free phone calls thru your system. Can you see the security issue in the dialplan you attached to your bug? |
| 22:12.15 | timeshell_atwork | What/ |
| 22:12.17 | timeshell_atwork | ? |
| 22:12.17 | manxpower | exten => _X.,1,Dial(dahdi/1/${EXTEN},10,tTm) T + t = allow either side to do a transfer using # |
| 22:12.21 | timeshell_atwork | That wasn't my bug |
| 22:12.46 | timeshell_atwork | I was referring to the timing issue that guy was bringing up |
| 22:12.51 | timeshell_atwork | That's not me. |
| 22:13.45 | timeshell_atwork | So, in my pastebin, is it normal for it to show the channels "In Use" |
| 22:13.47 | timeshell_atwork | ? |
| 22:14.39 | manxpower | In use means "Used by asterisk, can't be used by anything else" |
| 22:14.50 | timeshell_atwork | k |
| 22:14.52 | manxpower | It does not mean "off hook or on a call" |
| 22:15.02 | timeshell_atwork | still have the same issue however. |
| 22:15.11 | timeshell_atwork | Says everyone busy/congested |
| 22:15.14 | timeshell_atwork | Lines aren't in use |
| 22:16.43 | timeshell_atwork | It's the same bloody config as 7 days ago that worked! |
| 22:16.46 | manxpower | What is the value of HANGUPCAUSE and DIALSTATUS |
| 22:16.53 | timeshell_atwork | just a sec |
| 22:21.08 | timeshell_atwork | Back. Where do I find those vars? |
| 22:23.38 | manxpower | Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) as the priority after the dial. See channelvariables.txt in the Asterisk souce directory for information on all channel variables |
| 22:25.16 | *** join/#asterisk Psychobilly (n=moi@adsl282-123.kln.forthnet.gr) |
| 22:33.52 | timeshell_atwork | <PROTECTED> |
| 22:37.28 | manxpower | That is what I would expect with "unable to create channel DAHDI" |
| 22:37.41 | timeshell_atwork | Ok |
| 22:37.51 | timeshell_atwork | WHy can't it create channel DAHDI? |
| 22:38.18 | manxpower | what happens when you do a "module unload chan_dahdi.so" and then "load module chan_dahdi.so" |
| 22:39.15 | manxpower | sorry, "module load chan_dahdi.so" |
| 22:39.36 | timeshell_atwork | same thing |
| 22:40.34 | manxpower | Same thing as what? |
| 22:40.36 | timeshell_atwork | Is 2.1.0.4 or 2.2.0.2 better than the other? |
| 22:40.49 | timeshell_atwork | I don't get a message when I do that and calls still fail with the same error |
| 22:41.09 | manxpower | You should get a message about reading chan_dhadi.conf or something like that |
| 22:41.09 | timeshell_atwork | All the status messages seem to suggest that the dahdi drivers are available. |
| 22:41.15 | timeshell_atwork | Incoming calls appear to work. |
| 22:41.56 | timeshell_atwork | It's only outgoing calls that don't work |
| 22:43.36 | timeshell_atwork | Oddly enough 1 of the lines has just started working |
| 22:43.40 | timeshell_atwork | only 1 |
| 22:43.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:45.30 | timeshell_atwork | No, I didn't get a message like that, but when I tried to load it again it said it was already loaded. |
| 22:46.52 | timeshell_atwork | That's really strange. The second channel in the group of 3 is now accepting outbound calls whereas the other 2 channels are still reporting busy |
| 22:47.34 | timeshell_atwork | Oooooo |
| 22:47.35 | timeshell_atwork | Hold on |
| 22:47.52 | timeshell_atwork | I called it from outside |
| 22:47.57 | timeshell_atwork | And THEN it started working |
| 22:48.20 | timeshell_atwork | I'm gonna try calling them all and see what that does |
| 22:48.42 | timeshell_atwork | OMG |
| 22:48.44 | timeshell_atwork | That's it!! |
| 22:49.01 | *** part/#asterisk manxpower (n=eric@69.73.94.162) |
| 22:49.12 | timeshell_atwork | After an incoming call comes on the channel in question it's working!! |
| 22:49.18 | timeshell_atwork | manxpower ^^^^ |
| 22:49.40 | timeshell_atwork | What's that all about? |
| 22:52.00 | timeshell_atwork | [TK]D-Fender HEY ^^^^ I need input on 4 lines above. |
| 22:52.52 | timeshell_atwork | The dahdi channel isn't init'ing for outgoing calls until a call comes IN on it. |
| 23:06.57 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 23:12.34 | *** part/#asterisk korihor (n=korihor@190.77.83.180) |
| 23:12.49 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
| 23:15.24 | Faiz | After downgrading Asterisk to version 1.6.1.0, the fax modules now load properly, but for some reason.. I can't receive or make outgoing calls from my analog phone |
| 23:15.27 | Faiz | i get the message: WARNING[3235]: app_dial.c:1518 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 23:15.46 | timeshell_atwork | omg |
| 23:15.49 | timeshell_atwork | lol!!! |
| 23:15.54 | timeshell_atwork | That's what I was just working on |
| 23:16.06 | Faiz | same branch version? |
| 23:16.09 | timeshell_atwork | 1.6.0 |
| 23:16.25 | timeshell_atwork | My problem was I couldn't make outgoing calls |
| 23:16.36 | Faiz | ah |
| 23:16.44 | timeshell_atwork | I discovered if I called INTO the line first it would init the channel and afterward I could make outgoing |
| 23:16.56 | timeshell_atwork | Are you using 64 bit? |
| 23:17.01 | Faiz | no, 32 bit |
| 23:17.04 | timeshell_atwork | hmmm |
| 23:17.22 | Faiz | it worked fine before, with my current extensions.conf setting |
| 23:17.33 | timeshell_atwork | Yah, I had the same issue too |
| 23:17.41 | timeshell_atwork | Which dahdi version are you using? |
| 23:18.01 | Faiz | tools 2.2.0, dahdi version 2.2.0.2 |
| 23:18.06 | timeshell_atwork | DId you change your chan_dahdi.conf or your dahdi/system.conf or anything else dahdi? |
| 23:18.20 | Faiz | no |
| 23:18.33 | timeshell_atwork | What happens for incoming calls? |
| 23:18.38 | Faiz | it doesn't recognize them either |
| 23:18.44 | Faiz | as if the line doesn't exist |
| 23:18.57 | Faiz | for outgoing, it also displays "status = "CHANUNAVAIL" |
| 23:19.04 | timeshell_atwork | yah |
| 23:19.11 | timeshell_atwork | CentOS? |
| 23:19.13 | Faiz | yep |
| 23:19.16 | timeshell_atwork | Hmmm |
| 23:19.48 | Faiz | is it related to the channels? or the TRUNK variable in extensions? |
| 23:20.14 | timeshell_atwork | I can't help much today, but I'm gonna be investigating this more tomorrow. I think it may be a dahdi bug |
| 23:20.22 | timeshell_atwork | dahdi driver |
| 23:20.26 | Faiz | ah.. |
| 23:20.31 | Faiz | this is interesting because |
| 23:20.41 | Faiz | 1.6.1.1 works fine with incoming/outgoing, but doesn't support fax properly |
| 23:20.52 | Faiz | so i downgraded to 1.6.1.0, fax works, calls dont.. heh |
| 23:20.57 | timeshell_atwork | Hmm |
| 23:21.05 | timeshell_atwork | Still works if you upgrade back to 1.6.1.1? |
| 23:21.07 | Faiz | (my .conf files were configured with respect to 1.6.1.1) |
| 23:21.11 | Faiz | possibly, |
| 23:21.23 | Faiz | should i try 1.6.1.2? |
| 23:21.33 | timeshell_atwork | I don't know. I haven't used 1.6.1.x series yet |
| 23:21.41 | Faiz | indeed |
| 23:22.13 | timeshell_atwork | I'm trying to stabilize my current production server. I've had hard enough time doing it with 1.6.0's |
| 23:22.15 | timeshell_atwork | :p |
| 23:23.00 | timeshell_atwork | Anyway, I'm sorry, but I have to go. I have a prior engagement otherwise I'd help further. |
| 23:23.03 | Faiz | heheh, completely understandable |
| 23:23.16 | timeshell_atwork | (limited as that may be) :p\ |
| 23:23.23 | Faiz | no problem good sir |
| 23:23.34 | Faiz | if you come to a general conclusion, please let me know? |
| 23:23.44 | Faiz | or if i do, i will let you know as well |
| 23:23.46 | timeshell_atwork | For certain. |
| 23:23.51 | timeshell_atwork | I'm on daily. |
| 23:24.01 | Faiz | great, i'll be on all night after 10 pm |
| 23:24.03 | timeshell_atwork | Usually in the #asterisk-gui channel though |
| 23:24.06 | Faiz | ah |
| 23:24.15 | Faiz | nice meeting you, have a great day :) |
| 23:24.21 | timeshell_atwork | later.... afk |
| 23:28.05 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:32.06 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:32.52 | svm_invictvs | Is "fax" a special extension? |
| 23:32.57 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 23:32.59 | svm_invictvs | Like, for fax detect? |
| 23:42.36 | *** join/#asterisk alunca (n=alun@c-24-130-216-30.hsd1.ca.comcast.net) |
| 23:43.12 | alunca | My Aserisk cannot ping google.com ... and nslookup not working. please help |
| 23:44.43 | Psychobilly | edit your /etc/resolv.conf |
| 23:46.47 | alunca | Psychobilly I just changed dns1 and dns2, do I need to change domain? |
| 23:47.04 | thansen | is it possible to have a shared voicemail box? |
| 23:51.56 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 23:53.41 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 23:55.48 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 23:58.05 | LiNeTuX | Hey, any Aastra folks in here? I'm trying to find how complex the SIP secret can be... my searches are coming up dead, and trial & error isn't going so well either... |
| 23:59.20 | alunca | Psychobilly after edit the file, resolv.conf, is there anyother command to reboot asterisk without reboot? |
| 23:59.48 | Psychobilly | reload or restart now in asterisk cli |