| 00:09.05 | *** join/#asterisk leymoo (n=leymoo@cpc3-rdng9-0-0-cust1016.winn.cable.ntl.com) |
| 00:12.19 | *** join/#asterisk hijacked (i=VNHW@66.255.220.17) |
| 00:14.38 | *** part/#asterisk cirgal (n=robert@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 00:16.47 | Krurst | how hard would it be to set up my phones to play music when they're not being used? |
| 00:17.51 | dlynes_home | Krurst: probably not that hard, if your phones support autoanswer |
| 00:18.15 | Krurst | yep, I have intercom and paging set up. |
| 00:18.27 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:18.27 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:19.09 | dlynes_home | Krurst: Well, I would just call that phone in autoanswer mode, and then connect it to MusicOnHold(blahblah) |
| 00:19.40 | dlynes_home | Krurst: or have the user dial a special code to enable it, which would connect it to music on hold |
| 00:19.55 | dlynes_home | Krurst: and then dial another code later to disable it |
| 00:19.56 | Krurst | thats a find idea and all but what about incoming calls? |
| 00:20.15 | dlynes_home | Krurst: Well, that's something for you to figure out :) |
| 00:20.45 | Krurst | I was thinking - where does the dialtone sound come from? Is it phone generated? |
| 00:21.11 | dlynes_home | Krurst: yes, it's generally generated by the phone |
| 00:21.17 | Krurst | bugger. |
| 00:24.30 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:26.43 | *** join/#asterisk jtexter3 (n=jtexter3@12.37.168.34) |
| 00:27.05 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:33.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 00:40.07 | asdx | is the tdm400p a good card for starting? |
| 00:41.24 | Krurst | for a few analog lines yeah. |
| 00:41.26 | CoaxD | okay. where the hell does one go to get exploit code these days? Rootshell doesnt exist anymore, and i have a perfectly good remote server (that i own) that i need to root. |
| 00:42.53 | Krurst | you can also try the openvox A400P wich is a clone of the tdm400p but cheaper |
| 00:43.57 | Krurst | CoaxD: Dont that make some sort of collection of scripts and call it something these days? |
| 00:44.20 | CoaxD | Krurst: God only knows.. |
| 00:44.50 | CoaxD | There's got to be some traceroute bug or something... jeez |
| 00:48.16 | asdx | Krurst: ok |
| 00:52.35 | Krurst | thats it, it was called Metasploit. I think their motto is: "We give script kiddies their scripts" |
| 00:52.38 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 00:57.27 | CoaxD | hahaha. thanks. |
| 01:00.36 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 01:00.50 | *** join/#asterisk craigk (n=ckowald@58.174.113.53) |
| 01:04.17 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:09.37 | *** join/#asterisk dfriend (n=dfriend@69.89.168.17) |
| 01:13.11 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 01:16.49 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:17.01 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-172-112.dslextreme.com) |
| 01:20.36 | *** join/#asterisk guillote_GNU (n=guillote@host39.190-30-65.telecom.net.ar) |
| 01:26.42 | *** join/#asterisk apardo (n=apardo@34.64.220.87.dynamic.jazztel.es) |
| 01:28.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:34.31 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 01:34.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:39.09 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:39.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:41.29 | *** join/#asterisk cirgal (n=cirgal@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 01:42.46 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-8b98f2f68af39588) |
| 01:47.50 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 01:47.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:49.48 | *** part/#asterisk Qwell[0] (n=north@pdpc/sponsor/digium/Qwell) |
| 01:50.13 | Qwell | w00t :D |
| 01:50.18 | Qwell | file++ |
| 01:50.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:50.44 | jsmith | Qwell: What you w00ting about? |
| 01:50.55 | Qwell | jsmith: file helped me setup an irc proxy :) |
| 01:51.05 | jsmith | Qwell: Cool... |
| 01:51.16 | Qwell | so Qwell[] can go away soon |
| 01:51.42 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 01:51.42 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 01:51.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:52.26 | Qwell | better |
| 01:53.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:54.27 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 01:54.27 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:57.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:57.52 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 01:58.06 | *** part/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 01:58.57 | *** join/#asterisk mtaht4 (n=m@103-106-62-200.enitel.net.ni) |
| 02:08.57 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 02:15.27 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 02:16.21 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 02:16.34 | whywontitwork | is the new book released yet? |
| 02:17.44 | Qwell | very soon I think |
| 02:18.11 | whywontitwork | i heard end of aug |
| 02:18.25 | whywontitwork | cant find anything yet |
| 02:19.52 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 02:20.09 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-94-59.dslextreme.com) |
| 02:20.19 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 02:21.37 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 02:22.20 | whywontitwork | what do i need if i want asterisk to dial and extension from a webpage? |
| 02:26.57 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:28.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:29.31 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2054095512f01dda) |
| 02:29.39 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-41-246.dsl.tul2ok.sbcglobal.net) |
| 02:30.36 | kiscokid | Can * be setup so that you can retrieve your voicemail by dialing your own DID and then entering as '*' followed by your voicemail password? |
| 02:30.57 | kiscokid | or something like that? |
| 02:31.16 | Nugget | Any question that begins "Can asterisk be set up so that..." has an answer of "yes" |
| 02:31.27 | Nugget | it's just a matter of how much time and energy you have to expend. |
| 02:31.38 | kiscokid | my CEO doesn't want to have to remember a separate phone number to get his voicemail |
| 02:31.56 | hijacked | ...and whether you can get the guy that already did that to tell you how. |
| 02:31.58 | Nugget | there are thousands of ways to satisfy that request. |
| 02:32.11 | Nugget | show application voicemailmain |
| 02:32.12 | Qwell | easiest way is to buy a polycom with a messages button, heh |
| 02:32.21 | kiscokid | Nugget: yeah I figured it would be some dialpan magic |
| 02:32.52 | kiscokid | qwell this is for retrieving messages off premise |
| 02:32.58 | Qwell | oh |
| 02:33.31 | Qwell | well, check out the * exit feature of voicemail |
| 02:33.38 | kiscokid | how can you monitor for dtmf while voicemail is playing the greeting |
| 02:33.39 | Nugget | heck, you could create a custom IVR just for him and branch to it when you see the callerid is his mobile phone. |
| 02:33.41 | Qwell | let it hit vm, press *, put in your password |
| 02:34.11 | Nugget | "Press 1 to retrieve your voicemail, Press 2 to give kiscokid a raise" |
| 02:34.11 | kiscokid | qwell ok, I'll check it out |
| 02:34.25 | kiscokid | nugget: good idea |
| 02:34.48 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 02:37.37 | *** join/#asterisk Echinos (n=brian@142.46.83.12) |
| 02:38.03 | Echinos | I would like to know if anyone has a reccommend on a good distro for an *-only box... |
| 02:38.16 | dfriend | When my auto-attendant answers and a caller hangs up the channel is not being released by the PBX. Any ideas what is wrong? |
| 02:39.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:43.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:50.14 | Krurst | dfriend: is it coming in over a zap card? |
| 02:52.31 | *** part/#asterisk cirgal (n=cirgal@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 02:55.57 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:55.59 | *** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 02:56.30 | *** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 02:57.48 | *** join/#asterisk Cthalupa (n=cthulhu@cpe-76-187-118-246.tx.res.rr.com) |
| 03:11.34 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6671d773ebd31694) |
| 03:14.17 | *** join/#asterisk apardo (n=apardo@34.64.220.87.dynamic.jazztel.es) |
| 03:14.40 | Qapf | i was reading some documentation about hooking 2 asterisk boxes together and dundi sounds really cool, does anyone know of a really good tutorial for setting it up the first time? |
| 03:14.48 | dfriend | Krust Yes |
| 03:15.12 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:19.31 | Krurst | dfriend: http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision is a good place to start |
| 03:20.04 | Krurst | also look at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf and in particular busydetect |
| 03:20.25 | *** join/#asterisk Cthalupa (n=cthulhu@cpe-76-187-118-246.tx.res.rr.com) |
| 03:26.53 | dfriend | Krurst: Thanks, I have looked at both and don't seem to help. I am running loop start and the * box detects the hang-up from the VoIP caller but the PBX connected to * keeps the channel open. |
| 03:28.51 | dfriend | Krurst: It is as if the voltage sent to the PBX is either too short or too long. I am wondering if there is a setting to modify it? |
| 03:29.58 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:36.03 | Krurst | what country are you in? |
| 03:36.11 | dfriend | US |
| 03:36.31 | Krurst | can you pastebin your zapata.conf? |
| 03:45.14 | Qapf | does anyone know what the default nat keepalive timer is on a cisco 7960? i think im having an issue where the timeouts on my router and those on the phones arn't configured properly and the phones keep dropping |
| 03:46.01 | supers | 3600 seconds |
| 03:46.24 | supers | oh whoops, it should be 120 seconds |
| 03:46.53 | dfriend | Krurst: I am new to IRC and will need to figure out how to do that. |
| 03:52.38 | Qapf | supers, you know what configuration file that value lives in? i want to take it down to something a little more aggressive |
| 03:52.59 | *** join/#asterisk bmg505 (n=leon@196.209.179.92) |
| 03:57.35 | Krurst | dfriend: http://pastebin.ca/ |
| 04:05.15 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:08.43 | *** join/#asterisk SECGOD (i=SECGOD@c-24-14-83-172.hsd1.il.comcast.net) |
| 04:18.07 | *** join/#asterisk s0ck_ (n=m@unaffiliated/s0ck) |
| 04:20.59 | *** join/#asterisk darkfires (n=lwhite@xplr-ts-t11-208-114-158-128.barrettxplore.com) |
| 04:21.47 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.170.212) |
| 04:28.19 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:32.10 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-216-89.dsl.irvnca.pacbell.net) |
| 04:32.15 | BSD_Tech | hey guuys |
| 04:32.46 | BSD_Tech | couple things anyone here plan with chan_cellphone and chan_bluetooth |
| 04:33.04 | JT | plan with? |
| 04:33.11 | BSD_Tech | plan/play |
| 04:33.59 | BSD_Tech | I want to find what bluetooth adapters are supported |
| 04:34.13 | BSD_Tech | and what phones it has been tested with |
| 04:34.40 | J4k3 | it uses the regular bt stack afaik, so all bt devices that have linux support (most/all?) |
| 04:34.51 | JT | chan_cellphone is the newer one |
| 04:41.27 | BSD_Tech | ok so if linux supports the device it should work with asterisk chan cellphone |
| 04:41.33 | BSD_Tech | hmm |
| 04:42.13 | BSD_Tech | the idea is to make a office bluetooth headset compatable |
| 04:57.17 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk blinkbot2k (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 04:57.17 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.170.212) |
| 04:57.17 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk Cthalupa (n=cthulhu@cpe-76-187-118-246.tx.res.rr.com) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-94-59.dslextreme.com) [NETSPLIT VICTIM] |
| 04:57.17 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-fd9c5c872a6c7e19) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-180-14-100.red.bezeqint.net) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@123-243-144-208.tpgi.com.au) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk radicall (n=Zynth@189.168.164.235) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk slima (i=slima@unaffiliated/slima) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk remmo (n=junk@203.25.123.250) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) [NETSPLIT VICTIM] |
| 04:57.18 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk elixer (i=elixer@65.207.74.18) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk tld (n=terje@elde.net) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk NED260 (n=stefan@dhcp-089-098-165-006.chello.nl) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk Fetch (i=fetch@wintermute.cepheid.org) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a02a52c0611d0f88) |
| 04:57.19 | *** join/#asterisk chesty (n=chesty@unconcerned.org) |
| 04:57.19 | *** join/#asterisk patrick-- (i=patrick@eos.openroot.de) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk RGi_26 (i=RGi@62.97.247.44) [NETSPLIT VICTIM] |
| 04:57.19 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) [NETSPLIT VICTIM] |
| 04:57.19 | *** mode/#asterisk [+o Qwell[]] by irc.freenode.net |
| 04:57.20 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk CaT[tm] (n=cat@nessie.weebeastie.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 04:57.20 | *** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk CoaxD (i=coax@shell1.cornernet.com) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 04:57.20 | *** join/#asterisk yakkop (n=yakkop@c-69-181-237-92.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 04:57.20 | *** join/#asterisk Exstatica (i=exstatic@freenode/sponsor/velocity/exstatica) [NETSPLIT VICTIM] |
| 04:58.53 | bkruse_home | Qwell[]: do you exist in an array now? |
| 04:59.01 | bkruse_home | Qwell[0], Qwell[1], etc? |
| 04:59.07 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-94-138.dsl.peoril.sbcglobal.net) |
| 04:59.13 | WildPikachu | heh |
| 04:59.23 | Qwell | bkruse: no more Qwell[0] :D |
| 04:59.31 | bkruse_home | Qwell[]: ahh, right on |
| 05:00.25 | Qwell | Qwell[] dies tomorrow too, actually |
| 05:00.42 | bkruse_home | :[ |
| 05:00.53 | Qwell | got a nice irc proxy now.. |
| 05:02.02 | Juggie | which one |
| 05:02.09 | Qwell | bip |
| 05:05.10 | Juggie | i think i tried that and didnt like it |
| 05:05.18 | Juggie | or coudnt get it to work or something sounds famaliar |
| 05:07.35 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 05:12.04 | *** join/#asterisk wxwill (n=will@resnet2-17-137.dorm.utexas.edu) |
| 05:12.04 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:12.37 | kiscokid | can you have an asterisk character in an extension? |
| 05:13.30 | kaldemar | kiscokid: sure |
| 05:14.24 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 05:14.31 | kiscokid | kaldemar: any special syntax or can I just day exten => *22,1,DoSomething() ? |
| 05:15.05 | kiscokid | *say |
| 05:15.37 | map7 | Every time I park a call I cannot retrieve it again and it's lost in the system, which log file or command should I be using to find out where this call went? |
| 05:17.51 | fujin | anyone know what could cause crackly audio on handsets? |
| 05:18.03 | fujin | most people have been complaingin about hearing crackly audio during voicemail menus |
| 05:18.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-0ece3e54b8939d6c) |
| 05:18.21 | fujin | I'm guessing it could be network related, as I'm not running ToS/DiffServ, yet. |
| 05:18.25 | kiscokid | fujin: other ip traffic on the same vlan? |
| 05:18.34 | fujin | no, two seperate VLAN's. |
| 05:19.09 | kiscokid | what is ToS/Diffserv? |
| 05:20.19 | fujin | quality of service for switches |
| 05:20.23 | kaldemar | kiscokid: nothing special about it, just go ahead and try it. |
| 05:21.18 | kiscokid | kaldemar, ok thanks |
| 05:23.49 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 05:23.54 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:26.28 | map7 | How do I hang up a channel through the CLI if 'soft hangup' isn't working? |
| 05:27.04 | fujin | 'restart now' |
| 05:27.05 | fujin | xD |
| 05:27.23 | map7 | without throwing everyone off the system |
| 05:27.45 | map7 | three of the Zap channels are stuck |
| 05:28.16 | map7 | lost parked calls which I cannot kill |
| 05:35.40 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 05:35.45 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:37.39 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:37.54 | jarod14 | hi guys |
| 05:38.52 | Krurst | map7: what error does soft hangup give? |
| 05:39.22 | map7 | it doesn't give an error |
| 05:39.28 | map7 | it just says: Requested Hangup on channel 'Zap/4-1' |
| 05:39.32 | map7 | then never hangs it up |
| 05:39.47 | map7 | that Zap channel is stuck in Park |
| 05:40.08 | map7 | is there a way to kill all parked calls without restarting the asterisk server |
| 05:40.09 | Krurst | can you pick up the park? |
| 05:40.22 | map7 | no |
| 05:41.01 | Krurst | reload res_features.so maybe? |
| 05:41.03 | map7 | I rang in to test and parked my incoming call, straight away I lost my call |
| 05:41.59 | map7 | that didn't do much |
| 05:42.19 | Krurst | yeah it doens't look like it can unload it either |
| 05:42.27 | map7 | which log file will give me the best clue as to what's going on |
| 05:42.54 | Krurst | dmesg might have something from the zap card |
| 05:43.56 | *** join/#asterisk saftsack (n=oliver@p54A7EC2B.dip.t-dialin.net) |
| 05:49.58 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 05:52.28 | Krurst | zap restart should kill all your zap channels |
| 05:52.38 | map7 | hmmm, had to reboot the computer :( |
| 05:53.02 | map7 | we keep loosing phone calls, like 3 a month |
| 05:53.07 | Krurst | oh, ok, never mind then. |
| 05:53.21 | Krurst | do you have a timeout foryour parked calls? |
| 05:53.33 | map7 | yeah 15minutes |
| 05:53.43 | map7 | but these disappear straight away |
| 05:54.00 | map7 | you usually get music on hold, but not when it has this problem |
| 05:55.37 | Krurst | strange. is it just throught the zap card? |
| 05:56.12 | map7 | yeah I've got two TDM400 cards, and I'm using the full 8 channels |
| 05:56.17 | map7 | all together |
| 05:56.55 | map7 | I'm looking through /var/log/asterisk/full log file now around the time it stuffed up. |
| 05:57.05 | Krurst | tried the latest drivers and all that? |
| 05:57.08 | map7 | I don't know what to look for though |
| 05:57.46 | map7 | not sure, I don't normally look after this system, but we cannot get hold of the guy who does. |
| 05:57.57 | map7 | so i decided to start looking into it myself. |
| 05:58.41 | Krurst | <PROTECTED> |
| 05:59.02 | map7 | yes it does |
| 05:59.34 | map7 | i doubt it otherwise i would get nasty errors in dmesg, and I don't see anything in there |
| 06:00.03 | map7 | whilst the problem is happening people can answer phones, as long as they don't park them all is well. |
| 06:00.18 | map7 | as soon as they park a call, it's gone |
| 06:00.49 | map7 | the caller hears nothing, no music, no tone |
| 06:01.49 | Krurst | can you still park sip calls ok? |
| 06:02.38 | map7 | good point, cannot test that now until the problem happens again. I really needed to reboot though |
| 06:02.57 | map7 | i'll mark it down next time it happens which will be soon. |
| 06:03.05 | toddejohnson | I checked out asterisk-addons/trunk and can not seem to get it to compile it spews cdr_addon_mysql.c:97: error: dereferencing pointer to incomplete type. |
| 06:03.39 | Krurst | toddejohnson: did you re configure? |
| 06:03.50 | toddejohnson | yes everything checks out |
| 06:04.40 | toddejohnson | looking into the code it looks like it has to do with ast_str varibles |
| 06:06.35 | toddejohnson | full errors http://pastebin.com/d6d775ec6 |
| 06:06.40 | Krurst | hmm dies on mine too. |
| 06:08.06 | toddejohnson | i looked in bugs under category addons/* didn't see any open bugs. I would really like to get chan_mobile into my asterisk setup is there another way? |
| 06:09.23 | *** join/#asterisk saftsack (n=oliver@p54A7D060.dip.t-dialin.net) |
| 06:09.33 | Krurst | try an earlier revision? |
| 06:11.44 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:12.22 | toddejohnson | Krurst: do you run asterisk trunk on the system you tried on? |
| 06:13.11 | Krurst | nope. it requires I do, doesn't it... |
| 06:13.44 | toddejohnson | yea just thought of it when i posted that is a change in trunk not in release. |
| 06:14.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:15.41 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 06:17.39 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 06:19.16 | Krurst | hmm I seem to be getting an awesome problem with my setup. I'm running streaming music though sox for my music on hold. It seems its slowing down over time. |
| 06:19.35 | WildPikachu | ok .... asterisk is the voip platform, what does libpri do? |
| 06:20.07 | Krurst | thats for your PRI ISDN card I'd imagine. |
| 06:20.31 | WildPikachu | ok .... i see it builds a few .so's .... how are these used? :) |
| 06:20.46 | WildPikachu | and by which other package, i'm trying to understand how it all fits together |
| 06:22.30 | WildPikachu | i have zaptel, asterisk, libpri |
| 06:22.45 | Krurst | It's probably used inerfacing with a geneic pri card. |
| 06:23.06 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-b7744841a27d5be0) |
| 06:23.55 | Krurst | if you don't use ISDN, you don't need it. |
| 06:24.29 | WildPikachu | i do :o) ... i'm actually packaging rpms for a distro |
| 06:24.43 | WildPikachu | and setting up an office pbx at the same time |
| 06:27.02 | Krurst | plenty of info here: http://www.voip-info.org/wiki/view/Asterisk+PRI |
| 06:28.25 | WildPikachu | thanks man |
| 06:30.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:32.43 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 06:46.54 | *** join/#asterisk admin0 (n=admin@202.161.147.10) |
| 06:47.01 | admin0 | hi |
| 06:47.43 | admin0 | is it possible to use asterisk for callback and ivr but using a different gateway and not its own zap channels |
| 06:47.58 | Krurst | yeah |
| 06:49.30 | admin0 | where can I get more info or documentations regarding this ? |
| 06:50.37 | Krurst | http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback seems to have some info on it. |
| 06:51.06 | admin0 | thanks |
| 06:51.08 | admin0 | i will take a look |
| 06:51.27 | admin0 | thanks again |
| 06:52.05 | toddejohnson | now I can not get past res_config_mysql.c http://pastebin.com/d17ebdf75 it needs a filename as the 3rd argument |
| 06:55.48 | toddejohnson | I get res_config_mysql.c:172: error: too few arguments to function ‘ast_variable_new’ when compiling asterisk-addons from trunk |
| 06:58.20 | Krurst | aren't you only after chan_mobile? |
| 06:58.30 | toddejohnson | yea |
| 06:58.43 | Krurst | what is chan mobile anyway? is it chan cellphone renamed or something? |
| 06:58.52 | toddejohnson | chan celphone renamed |
| 06:59.31 | Krurst | could you get away with doing a make chan_mobile then? |
| 07:00.35 | toddejohnson | its not happy with that but I think i can menuselect it out |
| 07:01.12 | toddejohnson | what is res_config_mysql used for? is it used by freepbx? |
| 07:01.36 | Krurst | it's for storing all your config files in a mysql database I think. |
| 07:02.02 | FlatFoot | yep and call data 'CDR' which is what i am struggling with at the moe |
| 07:02.09 | toddejohnson | ok I will ask on freepbx |
| 07:03.01 | Krurst | no, its seperate to freepbx |
| 07:04.02 | Krurst | something about real time extensions |
| 07:07.20 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 07:07.20 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 07:13.25 | *** join/#asterisk lbow (n=lbow@196.7.14.163) |
| 07:13.43 | admin0 | it is possible to use asterisk with oracle ? |
| 07:14.03 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:14.09 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:14.23 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 07:21.20 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 07:23.27 | CCFL_Man2 | building of zaptel seems to hang at wctdm.o |
| 07:23.30 | *** join/#asterisk knobo (n=knobo@148.122.202.214) |
| 07:23.52 | knobo | how many calls can a queue in asterisk scale to? |
| 07:28.20 | Krurst | depends. http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 07:28.56 | Krurst | CCFL_Man2: what version? |
| 07:31.02 | CCFL_Man2 | the latest svn |
| 07:31.30 | Krurst | trunk or a branch? |
| 07:32.51 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:38.49 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 07:41.04 | CCFL_Man2 | Krurst: trunk |
| 07:41.39 | *** join/#asterisk dexteruk (n=dexteruk@89.253.168.92) |
| 07:41.53 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 07:42.18 | dexteruk | Problem with Asterisk 1.2 with realtime mysql access |
| 07:43.35 | dexteruk | in the res_mysql.conf i have the put in the database information but when it trys to connect it says 'MySQL RealTime: Failed to connect database server asterisk on localhost (err 2002)' |
| 07:44.04 | dexteruk | i have tested the usersname and password for the user and its fine |
| 07:48.04 | dexteruk | <PROTECTED> |
| 07:48.48 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 07:48.58 | dexteruk | <PROTECTED> |
| 07:50.43 | Krurst | does the user have acces to the databases in mysql? |
| 07:51.23 | dexteruk | yes |
| 07:51.35 | dexteruk | there is no problem with the user |
| 07:51.56 | dexteruk | if i test the user in mysql -h localhost -u asterisk -p it works |
| 07:52.13 | dexteruk | in the debug i get this which i thought was strange |
| 07:53.02 | dexteruk | Sep 22 09:16:55 VERBOSE[6346] logger.c: [res_config_mysql.so]Sep 22 09:16:55 VERBOSE[6346] logger.c: [res_config_mysql.so] => (MySQL RealTime Configuration Driver) |
| 07:53.02 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime Host: |
| 07:53.03 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime Port: 0 |
| 07:53.04 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime User: |
| 07:53.06 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime Password: |
| 07:53.08 | dexteruk | Sep 22 09:16:55 ERROR[6346] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
| 07:53.11 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime: Cannot Connect (2002): Can't connect to local MySQL server through socket '' (111) |
| 07:53.14 | dexteruk | Sep 22 09:16:55 WARNING[6346] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. |
| 07:53.17 | dexteruk | Sep 22 09:16:55 DEBUG[6346] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) |
| 07:53.20 | dexteruk | Sep 22 09:16:55 NOTICE[6346] config.c: Registered Config Engine mysql |
| 07:53.22 | dexteruk | Sep 22 09:16:55 VERBOSE[6346] logger.c: MySQL RealTime driver loaded. |
| 07:53.40 | many | whee. |
| 07:53.52 | many | flooder |
| 07:54.14 | dexteruk | oh i thought everyone was sleeping :-) |
| 07:54.31 | many | somewhere in the world someone is awake |
| 07:54.34 | zeeesh | any of my asterisk's friend hv knowledge about web-meetme? "http://www.voip-info.org/wiki/view/MeetMe-Web-Control"? i hv installed related resources but could not get success. at asterisk consloe getting msg "app_cbmysql.c:830 load_config: Successfully connected to MySQL database."? |
| 07:55.15 | kaldemar | dexteruk: you thought wrong. |
| 07:55.23 | dexteruk | Well it got your attention :-) |
| 07:55.37 | kaldemar | oh yes, in a bad way. |
| 07:56.29 | dexteruk | well its not killing anyone and this room was so quite |
| 07:57.30 | dexteruk | but do any of you have any clues to this problem? |
| 07:57.57 | dexteruk | its as if asterisk is not reading the details from the res_mysql.conf file |
| 07:58.40 | dexteruk | Krurst : Are you still there? |
| 08:05.08 | *** join/#asterisk twer (n=msimpson@61.246.220.157) |
| 08:05.30 | twer | Hey, could someone lend me a hand with zaptel.conf? |
| 08:06.50 | dexteruk | Problem with Asterisk 1.2 with realtime mysql access can anyone help |
| 08:08.10 | dexteruk | mysql is working for the CDR but not the realtime access |
| 08:08.47 | dexteruk | using the same database, did i miss someting in the compile? |
| 08:09.05 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:09.20 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 08:10.07 | tzafrir_laptop | twer, pastebin the output of cat /proc/zaptel/* |
| 08:13.30 | dexteruk | Problem with Asterisk 1.2 with realtime mysql access can anyone help mysql is working with the CDR table but not the realtime |
| 08:23.39 | *** join/#asterisk appelza (n=d@dsl-240-133-188.telkomadsl.co.za) |
| 08:24.23 | appelza | Hi guys, I've got SIP calls working..and it seems that my analog and digital cards were detected, but I can't route calls over either, how can I make sure they are detected by asterisk? |
| 08:25.23 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 08:25.59 | Krurst | try zap show status |
| 08:27.07 | appelza | ok |
| 08:27.29 | appelza | Wildcard TDM400P REV I Board 1 |
| 08:27.31 | appelza | :) |
| 08:27.37 | appelza | So, im probably just doing something wrong |
| 08:27.53 | Krurst | post your zapata.conf on pastebin |
| 08:28.07 | appelza | sec |
| 08:29.58 | appelza | not sure if it will useful (im using the AsteriskNOW distro), but I'll paste the lines which dont start with ; |
| 08:30.57 | FlatFoot | anyone know where i can get the DEBIAN version of app_addon_sql_mysql.so ? been searching to no avail |
| 08:31.05 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:31.33 | FlatFoot | also cdr_addon_mysql.so and res_config_mysql.so , doing a make on addons |
| 08:31.48 | FlatFoot | but it don't work , i have no hair left |
| 08:32.07 | Krurst | can't you compile it all yourself? |
| 08:32.25 | *** join/#asterisk appelza (n=d@dsl-240-133-188.telkomadsl.co.za) |
| 08:32.29 | appelza | sorry |
| 08:32.29 | FlatFoot | ok can i have a clue ( new to this command line stuff on debian ) |
| 08:32.42 | FlatFoot | trying to learn |
| 08:33.09 | dexteruk | Asterisk mysql is working for the CDR but not the realtime access |
| 08:33.09 | Krurst | your using debain? try http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian |
| 08:33.26 | FlatFoot | think i've been there just checking |
| 08:33.57 | Krurst | dexteruk, are the permissions on the res_mysql config ok |
| 08:34.20 | twer | bngpbx01*CLI> zap show status |
| 08:34.21 | twer | Description Alarms IRQ bpviol CRC4 |
| 08:34.21 | twer | T2XXP (PCI) Card 0 Span 1 RED 0 0 0 |
| 08:34.21 | twer | T2XXP (PCI) Card 0 Span 2 RED 0 0 0 |
| 08:34.30 | twer | that's the problem im having :( |
| 08:39.01 | twer | Okay, http://pastebin.ca/676650 thats the output |
| 08:39.20 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:39.28 | Krurst | FlatFoot: http://pastebin.ca/676652 < run those commands. see how it goes. |
| 08:39.29 | appelza | Krurst: http://pastie.caboo.se/92690 |
| 08:39.46 | appelza | I'd like to be able to route calls from sip over the tdm card |
| 08:39.56 | appelza | (if they start with 0) |
| 08:40.48 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 08:40.50 | twer | I think my zaptel.conf file is wrong |
| 08:41.34 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
| 08:41.37 | Uatec_ | Hellloooooo |
| 08:41.53 | Krurst | I gotta go, sorry guys |
| 08:42.07 | appelza | cheers |
| 08:42.21 | Krurst | appelza: you need an extention like 0|X. to dail over the FXO channel |
| 08:43.02 | FlatFoot | Kurst: sorry got called away , just gonna try now |
| 08:43.16 | tzafrir_laptop | twer, do you intentionally use just 24 of the channels in span 2? |
| 08:43.26 | tzafrir_laptop | is it T1 or E1? |
| 08:43.45 | Uatec_ | when i type iax2 show peers, i get one interesting entry: lucifer/lucifer XX.XX.XX.XX (S) 255.255.255.255 4569 (T) UNREACHABLE |
| 08:43.51 | Uatec_ | WTF is unreachable? |
| 08:44.16 | *** join/#asterisk sob0l (n=sobol@devel4.net) |
| 08:44.22 | Uatec_ | when i try to dial to that IP with those creditials i get an error message on the other asterisk server (lucifer) |
| 08:44.26 | Uatec_ | so it's definately reachable |
| 08:44.43 | Uatec_ | What is going on? |
| 08:44.45 | twer | its e1 |
| 08:44.49 | Uatec_ | IAX should not be this compliated. |
| 08:44.51 | twer | and no, |
| 08:45.24 | twer | tzafrir: We have one PRI line, (E1) with 30 channels |
| 08:45.39 | appelza | may I paste 2 lines in this channel? |
| 08:46.40 | appelza | nm, could someone please help me: my extentions.conf: http://pastie.caboo.se/92692 ; I want that extention to route sip calls to analog |
| 08:47.13 | sob0l | I have problem with cdr, there are records when duration=0 and billsec>0, is it a bug? |
| 08:47.58 | FlatFoot | Kurst: ta for that went through the commands ( most kind ) BUT the .so files did not appear in the modules dir for mysql |
| 08:50.16 | Uatec_ | How come nobody has any experience with IAX at all? |
| 08:50.28 | Uatec_ | it's one of the prime tennets of asterisk. |
| 08:51.34 | defswork | I do |
| 08:51.49 | defswork | I setup IAX between my home and a client over vpn |
| 08:52.19 | dexteruk | <PROTECTED> |
| 08:52.25 | defswork | unfortunately the laptop I ran it on at home died |
| 08:53.30 | appelza | could you someone please help me with routing my sip calls over my TDM400 card? (or guide me to a howto) |
| 08:53.49 | *** join/#asterisk gardo (n=gardo@121.97.192.1) |
| 08:54.13 | Uatec_ | defswork, do you know why when i type "iax2 show peers" i get my peer listed as UNREACHABLE ? |
| 08:54.57 | defswork | iirc mine did |
| 08:55.04 | defswork | but still routed calls ok |
| 08:55.05 | *** join/#asterisk RsaMan (n=aa@196.210.155.2) |
| 08:55.21 | Uatec_ | weird |
| 08:55.24 | Uatec_ | well ok then |
| 08:55.32 | Uatec_ | in that case i'll pass that by and go on to the next problem |
| 08:55.35 | defswork | I know that because I accidentally put the iax route into the outgoing trunks and they made outgoing calls on my home line :( |
| 08:56.33 | Uatec_ | lol |
| 08:56.57 | Uatec_ | when i dial from location A to location B i get the following on the console at location B: Aug 31 09:56:02 NOTICE[24497]: chan_iax2.c:6947 socket_read: Rejected connect attempt from xx.xx.xx.xx, who was trying to reach '115@extensions' |
| 08:57.17 | RsaMan | hello guys, still on my blind call transfer issue |
| 08:57.22 | RsaMan | i want to try upgrade my asterisk |
| 08:57.23 | RsaMan | Asterisk 1.4.10.1-BRIstuffed-0.4.0-test4 |
| 08:57.27 | Uatec_ | the username and password are right |
| 08:57.29 | defswork | Uatec_: I didn't do anything complicated like authentication |
| 08:57.31 | Uatec_ | but it's just rejecteing |
| 08:57.32 | RsaMan | i am running this version currently |
| 08:57.53 | defswork | it was private IP to private IP (via VPN) so wasn't really necessary |
| 08:57.57 | Uatec_ | hmm |
| 08:58.10 | twer | Does anyone know about the TE220? |
| 08:58.11 | RsaMan | it wont allow me to upgrade |
| 08:58.47 | RsaMan | was there any transfer issues reported in this version Asterisk 1.4.10.1-BRIstuffed-0.4.0-test4 |
| 08:58.56 | Uatec_ | well, i don't have a VPN between my two locations |
| 08:59.08 | Uatec_ | so i'm having to go over the internet, so i'm having to be secure |
| 08:59.19 | Uatec_ | how did you do it without authentication anyway? |
| 08:59.27 | Uatec_ | just didn't put in a password line in iax.conf |
| 08:59.30 | dexteruk | <PROTECTED> |
| 08:59.41 | defswork | well if you have internet to internet you can have vpn :) |
| 08:59.45 | defswork | I use openvpn |
| 08:59.55 | defswork | Uatec_: this is one side's config (/msg) |
| 09:00.42 | defswork | and that "just worked" |
| 09:01.14 | defswork | I don't have the otherside's config as the HD started grinding |
| 09:02.24 | Uatec_ | ouch |
| 09:02.41 | defswork | was a laptop I nabbed with a broken screen |
| 09:02.52 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 09:09.54 | Uatec_ | nice |
| 09:10.25 | Uatec_ | my windows 2003 server i have at home is a nabbed laptop, it's all fine, except that it's idling temperature is 54 |
| 09:14.55 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-37-19.w82-120.abo.wanadoo.fr) |
| 09:15.08 | yannj_fr | hi all |
| 09:16.00 | yannj_fr | I would like to know if anybody knows where are storedd the asterisk state tables |
| 09:21.20 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-37-19.w82-120.abo.wanadoo.fr) |
| 09:28.58 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:29.13 | *** join/#asterisk juuva (i=juuva@peili.org) |
| 09:32.41 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 09:33.10 | appelza | could you someone please help me with making sip calls (from my pc, to asterisk, then asterisk routes them over my TDM400) (or guide me to a howto) |
| 09:39.23 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:40.05 | appelza | anyone know how I can add BRIstuff to asteriskNOW? |
| 09:46.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:55.32 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 10:04.45 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 10:11.42 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 10:12.33 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.71) |
| 10:19.10 | Uatec_ | I am so close to giving up on this |
| 10:19.21 | *** join/#asterisk guillote_GNU (n=guillote@host210.200-117-50.telecom.net.ar) |
| 10:20.02 | mvanbaak | on what ? |
| 10:21.06 | Uatec_ | this iax trunking |
| 10:21.45 | appelza | :< |
| 10:22.47 | Uatec_ | this is the route that my call is supposed to take |
| 10:23.28 | Uatec_ | SPA922#1 -SIP-> asterisk1 -iax-> asterisk2 -SIP-> SPA922#2 |
| 10:24.01 | Uatec_ | but when asterisk1 tries to do Dial(IAX2/asterisk2/${EXTEN}) |
| 10:24.30 | Uatec_ | I get congestion back from asterisk2 |
| 10:24.39 | Uatec_ | and on asterisk2's cli i get: Aug 31 11:22:20 NOTICE[24497]: chan_iax2.c:6947 socket_read: Rejected connect attempt from xx.xx.xx.xx, who was trying to reach '115@extensions' |
| 10:29.23 | appelza | should an ISDN QuadBRI card be shown (if detected) with 'zap show status'? |
| 10:29.32 | appelza | if not, what command should I run to see if its working? |
| 10:29.48 | Uatec_ | what config file did you configure it in? |
| 10:29.56 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-2-53.red.bezeqint.net) |
| 10:30.26 | FlatFoot | OK this is doing my head in ....... Every asterisk-addons that i try either fails or does not contain the mysql .so files . Running debian , CAN ANYONE HELP ???? |
| 10:31.35 | FlatFoot | can't find an apt repository for addons either |
| 10:31.47 | Dovid | what do mu mean by fails. what error ? |
| 10:32.10 | FlatFoot | loads of different errors per release |
| 10:32.10 | Dovid | and what version of asterisk are you using ? |
| 10:32.20 | FlatFoot | 1.2.13 |
| 10:32.26 | Dovid | y nt 1.2.14? |
| 10:32.28 | Dovid | not* |
| 10:32.53 | FlatFoot | mainly because that was the one selected by apt-get |
| 10:32.58 | Dovid | try 1.2.24 + add on 1.2.7 |
| 10:33.19 | FlatFoot | ok i'll give it a go thanks |
| 10:33.28 | Dovid | I have never tried to compile on debian so I dont know exactly how to do it |
| 10:33.40 | Dovid | i can make u a small script that will get it and install it |
| 10:33.53 | FlatFoot | that would be very kind thank you |
| 10:34.45 | krdian_ | hi |
| 10:41.33 | *** join/#asterisk LuKinoVoIP (n=luca@gw.abanet.it) |
| 10:42.29 | Dovid | flatfoot: http://pastebin.ca/676718 |
| 10:42.38 | Dovid | hello krdian_: |
| 10:47.19 | LuKinoVoIP | hi all, any experiences in Grandstream GXW4008 FXSGW with asterisk? |
| 10:47.57 | *** join/#asterisk Strom_M (n=strom@netblock-208-127-172-112.dslextreme.com) |
| 10:48.27 | Dovid | never used that specific grandstream. what is the issue ? |
| 10:52.45 | LuKinoVoIP | in a small office with 14phones, i should route voice traffic to an asterisk server mantaining existing technology |
| 10:53.33 | Dovid | are you asking ? |
| 10:53.39 | Dovid | or u saying what u want to do? |
| 10:53.57 | LuKinoVoIP | what i want to do :-) |
| 10:54.26 | LuKinoVoIP | sorry...not good english, eheh |
| 10:54.39 | kaldemar | ~gs |
| 10:54.40 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 10:55.16 | Dovid | haha: i love jbpot |
| 10:55.30 | Dovid | ok. what do u have now ? |
| 10:55.37 | Dovid | Is it a new set up ? |
| 10:55.54 | Dovid | what kind of phones do you have now ? also what is ur native language ? |
| 10:55.56 | LuKinoVoIP | nothing, i have to decide what GW to buy |
| 10:56.03 | Dovid | there are lots of asterisk links all over. |
| 10:56.03 | LuKinoVoIP | italian |
| 10:56.10 | Dovid | ok. what r u trying to accomplish |
| 10:56.46 | Dovid | this may help you |
| 10:56.46 | Dovid | http://www.asterisk-italia.it/forum/; |
| 10:56.48 | LuKinoVoIP | i have to connect analog phones to an asterisk server |
| 10:57.05 | Dovid | ok. how many phones ? |
| 10:57.13 | LuKinoVoIP | 14 |
| 10:57.33 | Dovid | the reason I am asking is because the cards/fxs gateways cost a lot. it may be worth it getting IP phones. they will be a drop more per port |
| 10:58.09 | Dovid | unless you specifcly want to use analog phones |
| 10:58.39 | LuKinoVoIP | what's better between FXS gateways and TDM cards? |
| 10:58.58 | Dovid | sort of the same. depending on what you need. |
| 10:59.11 | Dovid | the TDM cards tend to be a bit more and you cna only have so many on one box |
| 10:59.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:59.49 | LuKinoVoIP | i can't buy new phones...employees must preserve actual phone |
| 11:00.16 | Dovid | ok. then u may want to look at the xorcom device |
| 11:00.27 | LuKinoVoIP | xorcom? |
| 11:01.22 | Dovid | http://www.xorcom.com/products/astribank |
| 11:01.46 | Dovid | http://www.xorcom.com/products/astribank/astribank_models |
| 11:02.24 | LuKinoVoIP | tx Dovid |
| 11:02.24 | Dovid | how r u going to connect to the phone system ? |
| 11:02.30 | Dovid | POTS ? BRI ? E1 ? |
| 11:02.33 | LuKinoVoIP | SIP |
| 11:02.38 | Dovid | ah ok. |
| 11:02.39 | LuKinoVoIP | k sorry |
| 11:02.49 | Dovid | so u jsut need a gateway with multiple FXS devices |
| 11:02.56 | LuKinoVoIP | yes |
| 11:03.04 | Dovid | it seems like you need this |
| 11:03.04 | Dovid | http://www.xorcom.com/products/astribank/astribank_models/astribank_xr0003 |
| 11:03.27 | Dovid | give them a call. i think that is ur best bet |
| 11:03.45 | LuKinoVoIP | thanks a lot Dovid |
| 11:04.29 | Dovid | also hav a look here: |
| 11:04.29 | Dovid | http://www.asterisk.it/ |
| 11:04.35 | Dovid | oops |
| 11:04.40 | Dovid | http://www.dailyasterisk.net/mailing-lists/ |
| 11:04.46 | LuKinoVoIP | k |
| 11:04.47 | Dovid | and |
| 11:04.47 | Dovid | http://www.dailyasterisk.net/ |
| 11:04.49 | Dovid | and |
| 11:04.55 | Dovid | http://www.asterisk-italia.it/forum/ |
| 11:05.01 | LuKinoVoIP | :-) |
| 11:05.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:06.37 | Dovid | np |
| 11:06.41 | Dovid | good luck |
| 11:07.02 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:07.39 | Dr-Linux | i got an * server console "Asterisk died with code 1" but all working fine, why is this? |
| 11:08.51 | Strom_M | Dr-Linux: you're not making much sense |
| 11:09.44 | Dr-Linux | Strom_M: hey, you are still up? |
| 11:09.59 | cpm | is making sense a prerequisite ? |
| 11:10.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:10.38 | puzzled | hi |
| 11:11.08 | Dr-Linux | opss mistake :P |
| 11:11.44 | Dr-Linux | i got a message at * server console "Asterisk died with code 1" but all working fine, why is this? |
| 11:11.48 | Dr-Linux | :P |
| 11:12.03 | Strom_M | Dr-Linux: does the message repeat? |
| 11:13.44 | Dr-Linux | Strom_M: it was being repeated. |
| 11:14.12 | Dr-Linux | Strom_M: i'm not on console, this server is located at CA |
| 11:14.17 | Strom_M | Dr-Linux: you probably ran safe_asterisk while asterisk was already running |
| 11:14.27 | LuKinoVoIP | or if you use zap devices such a E1/T1 card it's possible you must load ztcfg -vvv |
| 11:14.42 | Strom_M | Dr-Linux: who cares where the server is? that's what ssh is for |
| 11:14.43 | Uatec_ | http://www.voipuser.org/forum_topic_10797.html <-- I have posted a full report on my problem here, if anybody cares to take a look. It's an IAX 'No Authority Found' issue. |
| 11:14.51 | Dr-Linux | Strom_M: hhm.. that's what google search sounds |
| 11:15.19 | Dr-Linux | Strom_M: bcoz i can't see this message via ssh |
| 11:15.33 | Strom_M | Uatec_: yeah, you never specified a username |
| 11:16.35 | puzzled | Uatec_: http://voip-info.linuxsys.com/wiki/view/Asterisk+No+authority+found.html |
| 11:17.13 | Strom_M | Uatec_: why not just set up one "friend" entry on each box instead of making things overly complex? |
| 11:18.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:19.04 | Uatec_ | Strom_M, i've added username=..... and it's still working exactly the same |
| 11:19.51 | Strom_M | Uatec_: just try one type=friend entry rather than separate user and peer |
| 11:22.19 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 11:23.03 | Dr-Linux | anybody worked with C/C++ AGI/ |
| 11:23.05 | Dr-Linux | ? |
| 11:23.13 | appelza | ive compiled and installed both asterisk and zaptel many times before, but now any command starting with zap doesnt work (not found) |
| 11:24.04 | Uatec_ | ok, Strom_M, i've made them friends |
| 11:24.07 | Uatec_ | but i still get the same message |
| 11:24.21 | Dr-Linux | appelza: what's your user? |
| 11:24.30 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 11:24.35 | RsaMan | hi |
| 11:24.51 | RsaMan | i really need expert asterisk help..http://forums.digium.com/viewtopic.php?p=56837#56837 |
| 11:24.59 | RsaMan | still stuck on my blind transfer issue |
| 11:25.09 | Strom_M | Uatec_: pastebin what you have now |
| 11:25.39 | appelza | root |
| 11:25.51 | RsaMan | I have posted my situation http://forums.digium.com/viewtopic.php?p=56837#56837, in a nutshell , when i try do i blind call transfer the call gets dropped |
| 11:26.02 | *** join/#asterisk _WildPikachu_ (n=WildPika@about/linux/staff/wildpikachu) |
| 11:26.11 | appelza | and the zaptel module is loaded |
| 11:27.20 | RsaMan | did anyone have any similar issues ? |
| 11:27.31 | Strom_M | RsaMan: and can you dial 102 directly? |
| 11:27.50 | RsaMan | Strom_M : yes if i dial 102 on the zap channel it works |
| 11:28.14 | Strom_M | RsaMan: also, i'd advise you as a general rule never to use the "r" flag on the Dial() application |
| 11:28.45 | appelza | nm working! |
| 11:28.45 | appelza | :D |
| 11:29.23 | RsaMan | Strom_M : running this version Asterisk 1.4.10.1-BRIstuffed-0.4.0-test4 |
| 11:29.42 | *** join/#asterisk yassaccan (n=yassacca@admin186.hgo.se) |
| 11:29.57 | RsaMan | Strom_M : not sure if that makes a diff? |
| 11:30.01 | Uatec_ | http://rafb.net/p/5HCPgc26.html <-- Strom_M |
| 11:30.09 | Wonka | is there any sense in having an rtp proxy besides asterisk, like SER has? |
| 11:30.17 | Strom_M | RsaMan: not for this situation, no...but in general, it's a terrible idea to use that flag |
| 11:30.18 | Wonka | (or can have, at least) |
| 11:30.22 | Strom_M | Uatec_: ok |
| 11:30.48 | RsaMan | Strom_M : ok , understood |
| 11:31.05 | RsaMan | Strom_M : this blind call issue is killing me slowly .. |
| 11:31.12 | RsaMan | Strom_M : cant find any solution.. |
| 11:31.29 | Strom_M | RsaMan: do hookflash transfers work? |
| 11:32.11 | RsaMan | Strom_M : no, nothing happens when i flash :( |
| 11:32.20 | RsaMan | Strom_M : using a digium tdm400 card. |
| 11:32.30 | Strom_M | RsaMan: did you enable hookflash transfers? :) |
| 11:32.48 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:33.26 | RsaMan | Strom_M : transfer=yes , in zapata.conf ? |
| 11:33.31 | Strom_M | RsaMan: yes |
| 11:33.39 | RsaMan | Strom_M : then yes |
| 11:33.49 | Strom_M | Uatec_: pastebin the entirety of what you're doing now |
| 11:34.01 | RsaMan | Strom_M : is it in the right place? |
| 11:34.16 | RsaMan | Strom_M : http://forums.digium.com/viewtopic.php?p=56837#56837 |
| 11:34.40 | Strom_M | RsaMan: no |
| 11:34.52 | RsaMan | Strom_M : oh,, |
| 11:34.58 | Strom_M | you have to set that ABOVE "channel => 1" |
| 11:35.04 | Strom_M | otherwise it never gets assigned |
| 11:35.10 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:35.15 | Uatec_ | Strom_M, http://rafb.net/p/pCiPAV30.html |
| 11:35.42 | Strom_M | Uatec_: no |
| 11:35.52 | Uatec_ | what? |
| 11:35.53 | Strom_M | Uatec_: extensions.conf, console output, etc |
| 11:35.57 | Uatec_ | oh right |
| 11:36.24 | *** join/#asterisk kv0s (n=kv0s@p4FD27384.dip.t-dialin.net) |
| 11:36.34 | Uatec_ | well i posted all that was relevant on the form |
| 11:36.53 | RsaMan | Strom_M : like so ?http://pastebin.com/d45547594 |
| 11:37.38 | Strom_M | RsaMan: yes |
| 11:37.46 | RsaMan | Strom_M : i moved threewaycall=yes as well |
| 11:37.55 | Strom_M | also, don't put question marks up against the beginning of your URLs |
| 11:38.03 | Strom_M | it makes it difficult to click on them in my IRC client |
| 11:38.14 | RsaMan | Strom_M : sorry about that , i will test now thansk |
| 11:38.38 | Strom_M | Uatec_: yeah, but you've changed things |
| 11:38.52 | Strom_M | Uatec_: so I'd like to see what you have now rather than just guessing at it |
| 11:39.33 | Uatec_ | well i changed the iax.conf as i posted in rafb.net and the error messages are EXACTLY the same |
| 11:39.38 | RsaMan | Strom_M : its still now doing anything when i push flash |
| 11:39.51 | RsaMan | Strom_M : * not doing anything |
| 11:39.55 | Strom_M | Uatec_: what Dial() line are you using? |
| 11:40.08 | Strom_M | RsaMan: did you reload chan_zap.so? |
| 11:40.13 | Uatec_ | Dial(IAX2/asterisk/${EXTEN}) |
| 11:40.23 | RsaMan | Strom_M : i just typed reload |
| 11:40.27 | RsaMan | Strom_M : should do the trick |
| 11:40.33 | Strom_M | RsaMan: uh no |
| 11:40.43 | Strom_M | RsaMan: try "zap restart" |
| 11:41.04 | RsaMan | Strom_M : kk |
| 11:42.02 | Strom_M | Uatec_: also, before I forget to mention it, your inbound context includes an outbound context. that's a BAD BAD BAD BAD BAD idea |
| 11:42.13 | RsaMan | Strom_M : :( still get the same result |
| 11:42.55 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 11:43.13 | Uatec_ | what? |
| 11:43.20 | Strom_M | er |
| 11:43.22 | RsaMan | Strom_M : zap restart does something strange http://pastebin.com/d2ef96d6c |
| 11:43.25 | Strom_M | RsaMan: sorry, that was for you |
| 11:43.41 | Strom_M | im having trouble now remembering whose code i'm looking at |
| 11:43.43 | Uatec_ | lol |
| 11:43.46 | RsaMan | Strom_M : lol np |
| 11:44.01 | RsaMan | Strom_M : when i run zap restart twice it only works ? |
| 11:44.32 | Strom_M | bleh, it's too early. i'm going back to bed. |
| 11:45.25 | kv0s | Hi! |
| 11:45.56 | kv0s | My sip-trunk doesn't work for incoming calls. sip show registry says status "Request sent". What does it mean? |
| 11:47.18 | Uatec_ | kv0s, it means that you've tried to register but they've not come back to you |
| 11:47.32 | kv0s | hm |
| 11:48.55 | kv0s | how can i debug these informations? what is going wrong? |
| 11:49.04 | kv0s | i don't get any errors at console ..?!? |
| 11:49.20 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:50.01 | RsaMan | how do i debug a failed call transfer |
| 11:50.26 | RsaMan | ? |
| 11:50.36 | RsaMan | what command to i type in the cli? |
| 11:53.31 | RsaMan | trying to figure out why both calls get dropped |
| 11:53.44 | RsaMan | when i blind call transfer |
| 11:55.11 | kv0s | Uatec_: Any tipps how i can find the error why sipgate can not answer? |
| 11:56.52 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 11:58.33 | *** join/#asterisk lbow (n=lbow@dsl-241-25-146.telkomadsl.co.za) |
| 11:58.34 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 12:00.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:05.50 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 12:06.40 | masus | hi all , set(VARNAME=cat /usr/filename.inc) is something like this possible |
| 12:08.43 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:10.54 | masus | thanks :) |
| 12:11.31 | *** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk) |
| 12:15.07 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 12:17.04 | masus | set(VARNAME=cat /usr/filename.inc) ? |
| 12:17.06 | masus | :| |
| 12:17.13 | masus | is this possble any idea ? |
| 12:18.44 | tzafrir_home | masus, the SYSTEM function? something else in that general direction? |
| 12:18.59 | tzafrir_home | If all else fails, #exec |
| 12:19.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:19.11 | masus | i'll see |
| 12:22.21 | tzafrir_home | I wonder what do you people think about the comments in voip-info |
| 12:22.44 | tzafrir_home | Are they of any use? What could be done to make them useful? |
| 12:23.06 | RsaMan | Well, alot of them are questions without answers |
| 12:23.17 | RsaMan | I do not mind them very useful |
| 12:23.26 | RsaMan | i have reformulated my post http://www.asterisk.org/forum/viewtopic.php?t=17778&sid=b5eed41081008cb8863d3c70a0b9a52a |
| 12:23.27 | masus | tzafrir : system and exec is not the answer |
| 12:23.31 | elixer | tzafrir_home: turn them off, that would make them useful |
| 12:23.51 | RsaMan | still stuck with blind call transfer , but have tested some more |
| 12:23.53 | tzafrir_home | Some of them have useful information |
| 12:24.00 | RsaMan | True |
| 12:24.06 | tzafrir_home | But I think that the problem is that are not editable |
| 12:24.16 | elixer | tzafrir_home: then there contents should be moved to the page itself and then the comment deleted |
| 12:25.04 | elixer | tzafrir_home: my favorite type of comment is a comment that says something in the page is wrong, but the person can't be bothered to make the change to the page |
| 12:25.20 | tzafrir_home | kind of like the wikimedia "talk" page, right? |
| 12:25.36 | elixer | tzafrir_home: yeah |
| 12:25.44 | tzafrir_home | you do not have to be logged in to add a comment. |
| 12:26.04 | tzafrir_home | This is why they just add a comment, and don't fix, I guess |
| 12:26.05 | elixer | tzafrir_home: the wiki itself needs to be moved to mediawiki, imho, |
| 12:26.29 | elixer | tzafrir_home: but then you have other stuff _in the page_ like this: |
| 12:26.35 | elixer | Asterisk 1.2 does this and that and this and that |
| 12:26.45 | elixer | *** NOTE *** the above thing is WRONG |
| 12:27.33 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net) |
| 12:27.42 | elixer | tzafrir_home: the lack of any type of editorial system is the real short coming |
| 12:27.48 | elixer | tzafrir_home: but i'll stop complaining :) |
| 12:28.10 | [TK]D-Fender | RsaMan: have you tried copying the extens from [office] INTO [internal] and not using the "include" statement? |
| 12:28.16 | tzafrir_home | "editorial system"? what do you mean? |
| 12:28.48 | elixer | tzafrir_home: people that actively monitor changes for accuracy |
| 12:29.24 | RsaMan | tzafrir_home : i will try quickly |
| 12:29.48 | tzafrir_home | elixer, you can watch pages |
| 12:29.55 | *** part/#asterisk knobo (n=knobo@148.122.202.214) |
| 12:30.13 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-172-112.dslextreme.com) |
| 12:30.38 | elixer | tzafrir_home: true. i guess what i am saying is that there is no 'governing body' that makes sure the content that goes into the wiki isn't complete garbage. |
| 12:30.58 | RsaMan | tzafrir_home : well if i try transfer to 44 which is in context "internal" then it still does not owkr |
| 12:30.58 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 12:30.59 | RsaMan | work |
| 12:31.19 | tzafrir_home | elixer, that doesn't take software. That takes people |
| 12:31.25 | elixer | tzafrir_home: from an asterisk reference standpoint, i would say that the wiki is probably 10% useful information, and 90% crap. not to be too crude. |
| 12:31.35 | tzafrir_home | people watching pages. |
| 12:31.42 | tzafrir_home | and responding to changes |
| 12:31.44 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:31.57 | elixer | tzafrir_home: right. no one is watching changes. there is no accountability. |
| 12:32.18 | elixer | tzafrir_home: its kinda like giving asterisk SVN write access away to anyone with an e-mail address |
| 12:32.51 | tzafrir_home | well, it's a wiki. Not a software. Bugs are much easier to trace |
| 12:33.24 | elixer | tzafrir_home: well true, but that wasn't quite the point i was trying to make :-) |
| 12:34.42 | tzafrir_home | you want things to improve? pick a few pages, a small subdomain, and improve it |
| 12:34.50 | tzafrir_home | And beging watching it |
| 12:35.27 | elixer | tzafrir_home: i've done that. I put alot of work into FastAGI and have begun work on the other AGI related pages. |
| 12:49.58 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:50.55 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-109-98.lns10.syd6.internode.on.net) |
| 12:52.19 | *** join/#asterisk davixx (n=davixx@82.251.215.59) |
| 13:00.56 | appelza | are channel names variable? |
| 13:01.05 | appelza | can chan 1, be chan 2? |
| 13:04.12 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 13:04.16 | Sci_05 | morning all |
| 13:05.07 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 13:07.20 | *** join/#asterisk Woifi1988 (n=wolfgang@M1258P002.adsl.highway.telekom.at) |
| 13:08.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:19.29 | [TK]D-Fender | appelza: Channel names are named after the device that created them, plus a more or less random suffix |
| 13:24.40 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.205.69) |
| 13:26.26 | appelza | ok |
| 13:26.31 | appelza | ty |
| 13:26.47 | appelza | i have two cards, and both want to use 1 as a channel name (diff makes of cards) |
| 13:33.47 | elixer | what kind of cards? |
| 13:34.29 | elixer | T1/E1? analog? |
| 13:35.51 | appelza | its a tdm400 and a quadbri isdn |
| 13:36.20 | appelza | really struggling :< I just want to route my voip calls over the tdm but i have no idea how :/ |
| 13:36.52 | *** part/#asterisk masus (n=tet@88.248.73.2) |
| 13:36.55 | appelza | well from sip-zap even |
| 13:37.00 | appelza | and zap-sip |
| 13:38.50 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:39.53 | [TK]D-Fender | appelza: they can't fight over the DEVICENAME. And this is not "channel" we're talking about. The order your ports are in depends on how your cards initialized |
| 13:40.06 | [TK]D-Fender | appelza: pastebin "dmesg" |
| 13:40.08 | [TK]D-Fender | ~pb |
| 13:40.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:40.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 13:40.27 | [TK]D-Fender | appelza: And not knowing how to even dial out one of your interfaces is another problem altogether |
| 13:41.29 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 13:42.50 | jer | so i've got sip connection which only can hear incoming data, the other end can't hear their outgoing voice; there's no NAT at all.. what are some other possibilities? |
| 13:43.18 | elixer | jer: its nat |
| 13:43.19 | elixer | ;-) |
| 13:43.20 | elixer | kidding. |
| 13:43.21 | jer | (all other phones, set up the same, don't have a problem) |
| 13:43.34 | jer | well by the same i mean all non-unique settings i.e., extensions are different |
| 13:45.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:45.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:46.51 | s0ck | what exactly is sent to * when you hit the transfer key on your handset |
| 13:46.59 | s0ck | some kind of sip command, presumably? |
| 13:47.59 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-7e22dd224cdf006a) |
| 13:48.31 | *** join/#asterisk mog (i=mog@nat/digium/x-76aa16537ff29bb9) |
| 13:48.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:49.28 | hijacked | you could run a "sip debug" on your asterisk console. |
| 13:49.47 | s0ck | i guess |
| 13:49.56 | s0ck | trying to make it easy to park/retrieve a call |
| 13:50.22 | MihiNomenEst | shrug. |
| 13:50.29 | MihiNomenEst | *3 is how we do it here. |
| 13:50.31 | MihiNomenEst | easy enough. |
| 13:51.05 | MihiNomenEst | the problem is, some of my technicians are morons and they don't seem to realize that * calls you back to tell you where the call is in the parking lot. |
| 13:54.25 | *** join/#asterisk ganga (n=sandeep@59.95.246.210) |
| 13:55.50 | puzzled | how can I set the interval for sip re-registration? is defaultexpiry the only way I can I set it per peer too? |
| 13:57.06 | jsmith | puzzled: I'm afraid so, without changing the code. |
| 13:57.18 | jsmith | puzzled: If there's some other way, I'm not aware of it |
| 13:57.27 | puzzled | jsmith: ok, thanks |
| 13:58.21 | *** join/#asterisk ganga (n=sandeep@59.95.246.210) |
| 14:00.16 | ganga | hi |
| 14:00.30 | ganga | anyone thr? |
| 14:00.46 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:00.51 | jsmith | ganga: What's your question? |
| 14:00.56 | jsmith | ganga: There are plenty of us here :-) |
| 14:01.00 | ganga | :) |
| 14:01.11 | ganga | i am a newbie to asterisk |
| 14:01.21 | ganga | i want to know the call flows |
| 14:01.36 | ganga | i mean a document explaining the source code |
| 14:01.40 | ganga | i tried to google |
| 14:01.53 | ganga | but couldnt get any help |
| 14:01.56 | puzzled | not sure if such a document exists |
| 14:02.21 | ganga | ok |
| 14:02.27 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 14:02.27 | codefreeze | ganga: look in the doc/ dir first, there is quite a bit there. Then the voip-info wiki, the TFOT book, etc. |
| 14:02.52 | ganga | @codefreeze - tq |
| 14:03.16 | ganga | codefreeze:i tried using DOxygen |
| 14:03.20 | ganga | to study the source code |
| 14:03.36 | ganga | but i am baffled by what a channel is and how channels are bridged |
| 14:04.31 | harryr | ganga: a channel is just something that handles two directions of audio at the same time |
| 14:04.44 | codefreeze | ganga: a channel is the fundamental connection between asterisk and a device. |
| 14:05.00 | ganga | ok |
| 14:05.11 | harryr | Stream, Conduit etc. |
| 14:05.25 | ganga | so bridging channels is just like connecting 2 diff channels ? |
| 14:05.33 | codefreeze | bridge? You take the outs of one channel, and feed it to the ins of the other. And vice-versa |
| 14:05.48 | ganga | ok |
| 14:05.59 | codefreeze | With transcoding magic included |
| 14:06.02 | ganga | so a channel is created for every new call ? |
| 14:06.05 | harryr | or rather, multiple channels feeding their output to a mixer, which feeds the mixed result back to the input of the channels |
| 14:06.10 | harryr | ganga: yes |
| 14:06.14 | ganga | ohh |
| 14:06.26 | ganga | thanks for the explanation guys |
| 14:06.41 | codefreeze | When you pick up a zap telephone's handset, a channel is created |
| 14:06.43 | ganga | i will get on with the source code documentation and if i want any help i will be back |
| 14:06.52 | ganga | codefreeze:k |
| 14:08.22 | *** join/#asterisk petong (i=petong@66-117-151-141.lmi.net) |
| 14:14.10 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 14:23.26 | *** join/#asterisk PioneerVM4 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 14:23.29 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 14:23.50 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 14:23.58 | WildPikachu | is ael the new way to configure extensions? |
| 14:24.22 | PioneerVM4 | I have a PAP2T behind a router and an Asterisk box behind a firewall -- what is the minimum ports i have to open on firewall to allow outside connections from PAP2T (or other dynamic location) through? |
| 14:24.30 | PioneerVM4 | do i only need 5060 UDP? |
| 14:24.38 | PioneerVM4 | Oh this is SIP |
| 14:25.04 | codefreeze | WildPikachu: AEL is the new way to write dialplan code, if that's what you meant |
| 14:25.16 | WildPikachu | aha |
| 14:25.17 | WildPikachu | thanks |
| 14:25.44 | [TK]D-Fender | WildPikachu: It is just another way. AEL gets parsed back to standard extensions logic by its parser. You can see how it evaluates its syntax by doing "dialplan show" |
| 14:25.55 | WildPikachu | yea, i was just looking |
| 14:26.04 | *** join/#asterisk asteriskproblems (n=pbarnsle@81.171.174.178) |
| 14:26.07 | [TK]D-Fender | WildPikachu: "new" is a little subjective, but not entirely inaccurate. |
| 14:26.19 | asteriskproblems | hey Guys what call stats package(s) would you recommend? |
| 14:26.40 | [TK]D-Fender | PioneerVM4: describe the full path between * and your ATA |
| 14:27.03 | PioneerVM4 | ATA behind firewall -- assume it could be anyone travelling remotely |
| 14:27.08 | PioneerVM4 | i mean router sorry |
| 14:27.15 | PioneerVM4 | the Asterisk box is behind a firewall at my colo |
| 14:27.39 | PioneerVM4 | right now i allow multiple ports in but only for certain IPs |
| 14:27.55 | PioneerVM4 | i want to allow people to be on dynamic IPs and connect in from an ATA or software but with minimal open ports |
| 14:27.56 | [TK]D-Fender | PioneerVM4: the ATA does not need ANY ports forwarded. All it needs is "nat=yes", "canreinvite=no", and "qualify=yes" in its * sip.conf entry |
| 14:28.07 | PioneerVM4 | nono not port fw |
| 14:28.15 | PioneerVM4 | just what ports do i have to open on firewall where asterisk box is |
| 14:28.23 | PioneerVM4 | to accept a connection/registration from ATA |
| 14:28.40 | [TK]D-Fender | PioneerVM4: So * is behind a NAT of its own as well? |
| 14:28.44 | PioneerVM4 | yes |
| 14:28.51 | [TK]D-Fender | PioneerVM4: Read the full guide : |
| 14:28.53 | [TK]D-Fender | ~sipnat |
| 14:28.54 | jbot | [~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 |
| 14:30.15 | PioneerVM4 | these docs are for NAT issues, im not having those |
| 14:30.20 | PioneerVM4 | (anymore) |
| 14:30.29 | [TK]D-Fender | PioneerVM4: that is what you need to open op. |
| 14:30.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:30.45 | [TK]D-Fender | PioneerVM4: It specifies all of the ports & settings. |
| 14:30.45 | PioneerVM4 | i'm more just looking to see what port(s) does an outside of asterisk box to let people register in |
| 14:30.55 | PioneerVM4 | not port fw'd or nat issues, its more a firewall opening issue |
| 14:31.21 | styelz | 5060 for sip usually |
| 14:31.27 | [TK]D-Fender | PioneerVM : outside UA's set their OWN PORT. |
| 14:31.45 | PioneerVM4 | styelz, 5060 UDP or TCP |
| 14:31.46 | [TK]D-Fender | PioneerVM4 :Not usually 5060. |
| 14:31.55 | PioneerVM4 | DF -- im not looking for ATAs port |
| 14:31.55 | styelz | udp i think |
| 14:32.01 | PioneerVM4 | thanks styelz |
| 14:32.27 | [TK]D-Fender | PioneerVMnormally you don't need to do ANYTHING on the router. What is on the other side? |
| 14:32.29 | PioneerVM4 | DF i think you are thinking different than I -- i understand ATA uses its own port |
| 14:32.37 | PioneerVM4 | ok, lets start over, your on the wrong end |
| 14:32.44 | PioneerVM4 | ATA behind router -- forget about that |
| 14:32.51 | PioneerVM4 | ATA contacts Asterisk server behind firewall |
| 14:33.14 | PioneerVM4 | ATA communicates to * on certain ports, otherwise * cannot see the communications |
| 14:33.27 | PioneerVM4 | what ports do i have to open on the Firewall, the one that is in front of * (not the router in front of ATA) |
| 14:33.33 | styelz | i think he wanted to know what port to open on the asterisk side.. not the atas |
| 14:33.34 | PioneerVM4 | so that ATA can communicate inwards |
| 14:33.45 | PioneerVM4 | exactly styelz |
| 14:34.16 | [TK]D-Fender | PioneerVM : same ports you forward int he case of NAT |
| 14:34.27 | PioneerVM4 | right now im opening 8000-8001, 5060 and 3478 but dont think its necessary for all those |
| 14:34.33 | [TK]D-Fender | PioneerVM4: 5060,10000-20000 all UDP |
| 14:34.40 | styelz | checl rtp.conf |
| 14:34.51 | PioneerVM4 | i redirected RTP up to like 45000-50000 |
| 14:35.00 | WildPikachu | hrmmm, am I missing something? I can't seem to get asterisk to create /var/run/asterisk.ctl |
| 14:35.15 | styelz | does the dir exist? |
| 14:35.18 | PioneerVM4 | ok, so 5060 UDP is original connection in and then * tells the ATA to use the RTP ports UDP it has configured |
| 14:35.33 | styelz | i mean rw by asterisk |
| 14:36.00 | WildPikachu | nm, got it |
| 14:36.03 | WildPikachu | yea |
| 14:36.06 | WildPikachu | was no dir there |
| 14:36.09 | WildPikachu | stupid me |
| 14:36.11 | [TK]D-Fender | PioneerVM4: Yes, you need to open up all ports used by SIP & UDP |
| 14:36.14 | [TK]D-Fender | RTP* |
| 14:36.15 | PioneerVM4 | 62 |
| 14:36.18 | styelz | it's usually /var/run/asterisk |
| 14:36.21 | styelz | yea |
| 14:36.26 | PioneerVM4 | sorry numlock |
| 14:36.50 | PioneerVM4 | so all RTP ports defined in rtp.conf are what are used for call handling once a connection is established |
| 14:37.08 | [TK]D-Fender | PioneerVM4: SIP sets up the call, RTP carries the VOICE |
| 14:37.14 | *** join/#asterisk ManxPower (n=manxpowe@11.sub-70-216-146.myvzw.com) |
| 14:37.23 | PioneerVM4 | ahh and sip is thru 5060 to the * server |
| 14:37.25 | PioneerVM4 | ok got it |
| 14:37.39 | PioneerVM4 | so i guess in my case 5060 UPD and 45000-49999 UDP |
| 14:37.46 | asteriskproblems | anyone know any good call center software |
| 14:37.48 | PioneerVM4 | and * server informs ATA what RTP ports its using |
| 14:38.25 | [TK]D-Fender | PioneerVM4: Correct |
| 14:38.39 | PioneerVM4 | ok, i see 3478 was when i was using stun no longer needed |
| 14:38.49 | PioneerVM4 | i upgraded my cisco pix to 6.3(5) and it solved all my problems |
| 14:38.53 | [TK]D-Fender | asteriskproblems: Areski or Asterisk-stats. Go check the WIKI GUI list |
| 14:38.59 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-46-68.hlna.qwest.net) |
| 14:39.02 | PioneerVM4 | i use to have to use stun due to pix bugs |
| 14:39.15 | [TK]D-Fender | PioneerVM4: * neither needs nor suppotrs STUN |
| 14:39.17 | PioneerVM4 | but 6.3(5) solved everything, even the SIP options in config work now |
| 14:39.24 | PioneerVM4 | DF -- ive been thru that here tons of times |
| 14:39.24 | [TK]D-Fender | PIX = flaming piece of shit |
| 14:39.35 | [TK]D-Fender | PioneerVM4: Ton + 1 then :p |
| 14:39.38 | PioneerVM4 | PIX with earlier firmware needed stun unfortunately |
| 14:39.48 | PioneerVM4 | due to bugs it was only way to ge tit to work |
| 14:39.56 | JT | PioneerVM4: try this: [tk <tab> |
| 14:39.56 | asteriskproblems | lol D-Fender... cisco wouldnt be happy to hear you say that ;) |
| 14:39.57 | PioneerVM4 | but now that i upgraded the problem went away |
| 14:39.59 | [TK]D-Fender | PioneerVM4: No, it needed a real stack, STUSN was jsut a passable workaround ;) |
| 14:40.14 | [TK]D-Fender | asteriskproblems: Cisco couldn't care less about my opinion. |
| 14:40.28 | PioneerVM4 | well it works now with latest 6.3 firmware |
| 14:40.37 | JT | asteriskproblems: it's a well known fact in IT |
| 14:40.41 | JT | PIX == utter junk |
| 14:40.43 | PioneerVM4 | actually i think it solved all of my voip issues |
| 14:41.18 | ManxPower | You still didn't need STUN,. You just needed to disable the SIP support in the PIX, then let Asterisk's NAT support do what it's supposted to do. |
| 14:41.21 | [TK]D-Fender | PioneerVM4: Yes, now you're on your way to NEW problems! ;) |
| 14:41.29 | PioneerVM4 | manx, thats not correct in this case |
| 14:41.32 | *** join/#asterisk lbow (n=lbow@dsl-241-25-146.telkomadsl.co.za) |
| 14:41.33 | PioneerVM4 | every time i bring this up we go thru that here |
| 14:41.56 | ManxPower | well you would have had to disable the RTP fixup in addition to the SIP fixup. |
| 14:41.58 | PioneerVM4 | i disabled all sip options and it still wouldnt work, there were bugs in the firmware that have been fixed ( i read release notes) |
| 14:42.08 | PioneerVM4 | there were like 30 sip bugs |
| 14:42.21 | PioneerVM4 | anyway, moot point -- latest solved everything |
| 14:42.30 | [TK]D-Fender | PioneerVM4: Glad to hear. |
| 14:42.35 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 14:42.37 | Lucky7 | hey |
| 14:42.46 | Lucky7 | anyone here have E&M Winkstart setup? |
| 14:43.22 | JT | latest revision == place object in special container that is periodically cleared out by sanitation officers |
| 14:43.52 | Lucky7 | with E&M wink start, so i just need to set E&M=1-24 (full T1) or do i need to specify the dchan as well? |
| 14:43.53 | PioneerVM4 | i have to say i hear a lot of people complain about the pix -- i know its not perfect but as for off the shelf solution it has really worked well for me. Only problem i ever had was this sip issue and that was because i was lazy and had no upgraded firmware in like 4 years. Stability wise it has been flawless -- handling 20-30mb easily |
| 14:43.58 | ManxPower | Lucky7: we used to, but switched to PRI for obvious reasons |
| 14:44.17 | JT | PioneerVM4: much better just to setup one of those linux or bsd firewall distros |
| 14:44.26 | ManxPower | PioneerVM: you mean 20-30GB , right? |
| 14:44.32 | JT | Lucky7: E&M has no D channel |
| 14:44.37 | JT | inband signalling... |
| 14:44.39 | Lucky7 | PioneerVM2: I personally use a pix as well |
| 14:44.39 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 14:45.00 | Lucky7 | PioneerVM2: but I've been using cisco stuff for years, and I'm CCNA, so i'm a bit slanted. |
| 14:45.12 | Woifi1988 | what does "sudo make progdocs" create? |
| 14:45.17 | JT | eww, cisco junk :P |
| 14:45.26 | PioneerVM4 | no i believe i mean mbit |
| 14:45.42 | JT | Mbit/s most likely |
| 14:45.53 | Lucky7 | yea. pix505? |
| 14:46.02 | JT | even a realtek 10/100 card can handle 30Mbit/s |
| 14:46.05 | JT | chickenfeed |
| 14:46.05 | PioneerVM4 | ie: 1.54 mb = T1yes JT |
| 14:46.18 | jsmith | Woifi1988: "make progdocs" makes the Doxygen documentation for the source code... it's really only useful for Asterisk developers |
| 14:46.19 | JT | PioneerVM4: mb == millibit |
| 14:46.20 | PioneerVM4 | pix 515 |
| 14:46.36 | Woifi1988 | jsmith: thanks! |
| 14:46.44 | Lucky7 | the 515e is capable of more then that. |
| 14:47.02 | JT | yawn |
| 14:47.07 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:47.08 | Lucky7 | anyway |
| 14:47.12 | JT | overpriced defective bastardised linux boxes |
| 14:47.15 | JT | PIX... |
| 14:47.15 | [TK]D-Fender | JT : You're definitely a few bits short of a byte ;) |
| 14:47.22 | PioneerVM4 | PIx runs linux? |
| 14:47.25 | JT | yes |
| 14:47.29 | PioneerVM4 | its not overpriced if you buy it used |
| 14:47.30 | JT | a crappy cisco version |
| 14:47.31 | jsmith | [TK]D-Fender: You know, some of those 6-bit bytes... |
| 14:47.36 | JT | well then what's the point |
| 14:47.38 | PioneerVM4 | i just bought mine off ebay for 1/5 the cost |
| 14:47.40 | JT | no support contract |
| 14:47.45 | PioneerVM4 | who needs one |
| 14:47.53 | JT | if you ever need to upgrade |
| 14:48.14 | PioneerVM4 | actually you can get upgrades within the same firmware class without support |
| 14:48.17 | PioneerVM4 | for security purposes |
| 14:48.31 | JT | off the shelf linux does a much better firewalling job than PIX |
| 14:48.40 | PioneerVM4 | im sure it does |
| 14:48.47 | Dan0maN_Work | heh |
| 14:48.48 | PioneerVM4 | however it requires more maintenance |
| 14:49.09 | JT | nope |
| 14:49.12 | PioneerVM4 | as said, its not the perfect solution but it lets me not deal with yet another custom server |
| 14:49.18 | JT | just get a firewall distro that does everything |
| 14:49.27 | JT | use CF cards |
| 14:49.29 | PioneerVM4 | your a programmer i assume |
| 14:49.32 | JT | then where's the maintenance |
| 14:49.33 | JT | nope |
| 14:49.48 | PioneerVM4 | yea i looked into that, was too much trouble |
| 14:49.54 | PioneerVM4 | was going to go openbsd compact flash |
| 14:50.05 | JT | yes, and 30 SIP bugs is complete ease of mind |
| 14:50.11 | JT | i see the logic... but not really |
| 14:50.12 | PioneerVM4 | yet more things i have to research and install, my time is too valuable |
| 14:50.12 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-41-246.dsl.tul2ok.sbcglobal.net) |
| 14:50.21 | PioneerVM4 | actually the bugs are gone so no need |
| 14:50.21 | Dan0maN_Work | lol |
| 14:50.26 | PioneerVM4 | if the upgrade didnt work i wouldnt have bothered |
| 14:50.27 | JT | yet you like wasting your time on PIX problems? |
| 14:50.33 | JT | i'm sorry but that's silly |
| 14:50.34 | PioneerVM4 | first problem i ever had and its gone |
| 14:50.35 | Lucky7 | SIP Bugs? |
| 14:50.36 | ManxPower | I want to push one of my clients to move to a commercial firewall. |
| 14:50.50 | ManxPower | Then they can do their own firewall updates. |
| 14:50.51 | PioneerVM4 | the problem was due to me being lazy |
| 14:50.54 | JT | Lucky7: PIX are full of SIP bugs |
| 14:50.56 | Lucky7 | Yea, PIX's are known for royalling screwing SIP signals, if this is what your talking about |
| 14:51.05 | JT | signals? |
| 14:51.11 | Lucky7 | JT > lol, I'm fully aware |
| 14:51.15 | Lucky7 | packets signals, |
| 14:51.16 | Daviey | ManxPower: why not use a FOSS firewall with a GUI attached? |
| 14:51.16 | Lucky7 | w/e |
| 14:51.19 | JT | ... |
| 14:51.21 | Lucky7 | its frekaing early here |
| 14:51.23 | JT | you say you are CCNA |
| 14:51.28 | JT | get your terminology right |
| 14:51.32 | JT | hah |
| 14:51.37 | Lucky7 | I dont even become concious for another 4 hours. |
| 14:51.40 | Daviey | ManxPower: Ie IPCOP, Smoothwall, M0n0wall |
| 14:51.43 | JT | it's getting early |
| 14:51.57 | Dan0maN_Work | <PROTECTED> |
| 14:52.27 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 14:52.36 | lirakis | hi |
| 14:54.19 | DrAk0 | how i can set the ${STRFTIME(${EPOCH} to acts as timestamp ? |
| 14:54.31 | Lucky7 | hm. so. |
| 14:54.43 | Lucky7 | I've got a T1 with E&M winkstart |
| 14:54.50 | Lucky7 | channels 1-22 |
| 14:55.00 | Lucky7 | and i get about 2 dropped calls an hour |
| 14:55.16 | jsmith | Lucky7: What do you have set for your signalling in zapata.conf? |
| 14:55.18 | Lucky7 | which is pretty high, when I'm only making maybe 8-10 calls an hour |
| 14:55.23 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:55.24 | Lucky7 | E&M |
| 14:55.47 | jsmith | Lucky7: You could try setting your signalling to featured instead |
| 14:56.17 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:58.15 | Lucky7 | jsmith > whats the difference? |
| 14:58.34 | *** join/#asterisk chem_fun (n=spollman@c-71-205-33-130.hsd1.mi.comcast.net) |
| 14:58.56 | chem_fun | If I may, I have a question about the Digium IAXy S101I |
| 14:58.59 | PioneerVM4 | lucky what pix version u using |
| 14:59.01 | PioneerVM4 | software |
| 14:59.06 | jsmith | Lucky7: Not much, but Feature D is an extension of E&M winkstart from what I understand... I've had good luck using it in the past |
| 14:59.12 | chem_fun | The question has arose as to how it would handle a fax machine |
| 14:59.18 | jsmith | Lucky7: You can also try "em_w" as your signalling |
| 14:59.44 | jsmith | chem_fun: Faxing over a packet-switched connection is not for the faint of heart... |
| 15:00.15 | anonymouz666 | why a FXO line can't detect busy signals? |
| 15:00.49 | chem_fun | I'm new to asterisk, but I've got a swithch with QoS to give priority to the VOIP lines |
| 15:00.53 | anonymouz666 | If the far end hangs up the FXO line should detect the busy signal... |
| 15:00.55 | chem_fun | what other issues will I run into? |
| 15:02.10 | coppice | anonymouz666: * has a variety of busy tone detectors. the snag is they all suck |
| 15:02.38 | JT | anonymouz666: why should asterisk be able to detect it? |
| 15:02.51 | jsmith | anonymouz666: If the line has far-end disconnect supervision, then Asterisk will be able to tell when the far end hung up. If not, Asterisk has to guess based on the tones, and the tone detectors aren't very good. |
| 15:03.22 | anonymouz666 | fxotune on them. |
| 15:03.45 | JT | chem_fun: it not working, most likely |
| 15:06.17 | petong | howdy, I have a question about a te120p digium card and an ATT pri line |
| 15:06.33 | Lucky7 | jsmith : yea, now that i went back and double checked, I'm using em_w |
| 15:06.48 | petong | I believe I have the correct module and conf loaded |
| 15:06.55 | petong | but I see no light on the front of the card |
| 15:07.05 | petong | I am doing a line test with ATT in a bit |
| 15:07.17 | brodiem | does anyone have a PAP2T handy? I want to know if the web GUI on other's pap2ts have settings for specifying T.38. According to everything I've googled about it, they're supposed to support it but I have no T.38 options anywhere |
| 15:07.22 | JT | petong: did you run ztcfg? |
| 15:07.31 | petong | yes, |
| 15:07.40 | JT | brodiem: asterisk doesn't do T.38 |
| 15:07.49 | brodiem | JT: yes, it passes it through |
| 15:07.51 | petong | and it says span configured |
| 15:08.11 | jsmith | petong: What's the output of ztcfg -vv? |
| 15:08.17 | brodiem | JT: it just doesn't terminate/originate T.38. |
| 15:08.25 | JT | brodiem: 1.4 does passthrough, yes |
| 15:08.30 | petong | SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 15:08.37 | brodiem | right |
| 15:08.56 | petong | plus 1-23 Clear channel (Default) (Slaves: XX) |
| 15:09.02 | petong | 24 channels configured. |
| 15:09.09 | *** join/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 15:09.23 | jsmith | petong: And zttool, does it show the card. If so, is it in alarm? |
| 15:09.58 | petong | jsmith: no, zttool says OK |
| 15:10.18 | jsmith | petong: OK, what if you go into Asterisk and type "pri show span 1" |
| 15:11.04 | petong | im running asterisk 1.4 |
| 15:11.06 | [TK]D-Fender | petong: pastebin your zaptel.conf and zapata.conf |
| 15:11.08 | [TK]D-Fender | ~pb |
| 15:11.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:11.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:11.24 | petong | ok, will do |
| 15:12.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:12.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:13.30 | petong | http://pastebin.com/d68d99efd |
| 15:15.28 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:16.18 | [TK]D-Fender | petong: Everything looks fine. Now pastein from * CLI "pri show span 1" |
| 15:17.08 | petong | is the command the same in asterisk 1.4? |
| 15:17.09 | petong | No such command 'pri show' |
| 15:17.41 | petong | im looking through the help output now |
| 15:17.49 | [TK]D-Fender | petong: Ok, I'm guessing things may not have been compiled in the right order |
| 15:18.01 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:18.10 | [TK]D-Fender | petong: Stop *. then recompile (in order) libpri, zaptel, then asterisk |
| 15:18.21 | jsmith | petong: It sounds like either the chan_zap.so module didn't get compiled, or it's not being loaded due to a syntax or signalling problem |
| 15:18.28 | petong | hmm, ok |
| 15:18.29 | harryr | Is there anything special I have to do to get anybody to be able to dial into an asterisk box via SIP and go into an extension |
| 15:18.32 | petong | will try that |
| 15:18.38 | petong | thanks |
| 15:18.38 | harryr | e.g. extension@my.asterisk.example.com from any SIp phone |
| 15:24.18 | *** join/#asterisk Seb7 (n=sebast@host217-34-0-169.in-addr.btopenworld.com) |
| 15:24.46 | jsmith | harryr: Just make sure you've got an SRV record in DNS for that domain, and you're allowing unauthenticated calls, and the context contains an extension that matches the name the person will be dialing |
| 15:25.11 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 15:25.47 | harryr | jsmith: for sip.conf, i allow unauthenticated calls and control which context they go into where? |
| 15:25.59 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:26.00 | andresmujica | Hi anyone out there had worked with the sangoma t1/e1 line tapping system??? |
| 15:26.27 | andresmujica | or with the wanpipe TDM Voice API from them? |
| 15:26.37 | [TK]D-Fender | harryr: into the context specified under [general] |
| 15:26.42 | harryr | ah |
| 15:26.54 | Seb7 | Am I right in thinking that there is no Asterisk channel variable that corresponds to the channel "State" (which you can see with the DumpChan application or with show channel Xxxx from the CLI)? |
| 15:27.19 | jsmith | Seb7: No, I'm not sure it gets set as a channel variable... you *might* be able to get it from the CHANNEL dialplan function |
| 15:27.26 | Seb7 | in order to use State from the dialplan... |
| 15:27.49 | [TK]D-Fender | Seb7: You'd need to check it in an AGI or something. |
| 15:28.12 | [TK]D-Fender | Seb7: Then again, this is to check ANOTH channel than the one you're in. |
| 15:29.02 | *** join/#asterisk TheDingy (n=lboyd@houdc01.outbound.zogmo.com) |
| 15:29.23 | TheDingy | i like the rulse |
| 15:29.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:29.43 | Seb7 | I actually only want to check the current channel. It looks like the CHANNEL dialplan function would work if I were using 1.4. But nothing available on 1.2 I guess... |
| 15:30.04 | [TK]D-Fender | Seb7: It you're IN a channel, isn't its state OBVIOU? |
| 15:30.09 | [TK]D-Fender | OBVIOUS* |
| 15:30.30 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 15:30.31 | Seb7 | Er, well, not if you are in a macro, which some IFs beforehand |
| 15:31.10 | Seb7 | I could set my own variable I suppose, although that wouldn't be 100% guaranteed to be correct all the time due to timing issues. |
| 15:31.22 | s0ck | [TK]D-Fender: had problems on sip only boxes with distorted moh? |
| 15:31.34 | [TK]D-Fender | s0ck: nope |
| 15:31.52 | s0ck | apparently, lack of zaptel timing is doing it |
| 15:31.54 | TheDingy | i have a cisco 7940 sip config that works on the localnetwork, but when i place it on the wan it does nothing |
| 15:31.57 | s0ck | because i have no cards in this box |
| 15:31.57 | TheDingy | any ideas? |
| 15:32.07 | TheDingy | i have tried two different 7940's |
| 15:32.20 | s0ck | ztdummy is there but showing as unconfigured |
| 15:32.28 | s0ck | just wondering if it's worth configuring |
| 15:32.30 | TheDingy | it is getting to the asterisk box, but nothing in the log shows up |
| 15:32.53 | s0ck | ime, zaptel configured = kernel panic on reboot |
| 15:33.02 | s0ck | on an otherwise working pbx |
| 15:33.18 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:34.11 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:34.14 | harryr | [TK]D-Fender: ty for the help |
| 15:36.24 | [TK]D-Fender | Seb7: and how was this macro called, and what is it doing that you are wonding about the state of this call? Wouldn't your position in the dialplan clearly tell you whats going on.? |
| 15:38.23 | TheDingy | will domain set by dhcp make for problems when trying to register sip? |
| 15:38.29 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:39.36 | Seb7 | No, because depending on the number dialled I would have used the Ringing application, together with a Playback or not, in which case the call would still be Proceeding. |
| 15:39.58 | Seb7 | er in the case of not it would be Proceeding |
| 15:41.33 | Seb7 | And basically if the caller has been sent Alerting, I can and should play another message before disconnecting them, since I know that any PRI_CAUSE will be ignored. |
| 15:42.52 | *** join/#asterisk ToyMan (n=Stuart@pool-72-72-25-95.bstnma.east.verizon.net) |
| 15:43.44 | Seb7 | Not a huge deal - I'll set my own variable right after sending alerting, although I don't really like doing it because it's a cludge and the variable could have the wrong value if someone manages to disconnect between one and the other. |
| 15:53.00 | *** join/#asterisk konqi_ (n=konqi@217.193.163.2) |
| 15:54.15 | *** part/#asterisk simond (n=simon@208.68.95.5) |
| 15:55.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:57.58 | *** join/#asterisk ToyMan (n=Stuart@pool-72-72-25-95.bstnma.east.verizon.net) |
| 16:01.32 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 16:02.47 | CoffeeIV_ | I have an asterisk->iaxmodem->hylafax server. Every minute in the full log it says "chan_iax2.c: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 300)" How can I make that go away ? |
| 16:03.02 | *** join/#asterisk K-TA (n=ca@modemcable031.47-37-24.mc.videotron.ca) |
| 16:03.34 | K-TA | i have x100p cardd but not work on fresh install |
| 16:03.44 | K-TA | anyone can help me with this issue |
| 16:07.09 | putnopvut | CoffeeIV_: set maxregexpire in iax.conf to greater than 300 |
| 16:07.36 | [TK]D-Fender | K-TA: Soem details would be nice. |
| 16:08.33 | K-TA | i dont have sound |
| 16:08.40 | [TK]D-Fender | some* |
| 16:09.09 | K-TA | i receive inbound but i can hear him |
| 16:09.33 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:09.36 | [TK]D-Fender | K-TA: Keep going... these are very LOOSE details so far. |
| 16:09.54 | *** join/#asterisk cirgal (n=foo@wsip-70-169-190-173.sb.sd.cox.net) |
| 16:10.19 | K-TA | scuse me but i am french user |
| 16:10.35 | K-TA | i have install elastix |
| 16:11.23 | K-TA | all work but x100p card have problem |
| 16:11.47 | K-TA | when i try call with x100p to my cellphone |
| 16:12.12 | K-TA | x100p take link but no tone send |
| 16:13.24 | K-TA | i have try with trixbox but same shit |
| 16:13.33 | K-TA | maybe zaptel driver ? |
| 16:13.39 | darkskiez | is there anyway to do a pattern match hint, or do i have to enter every extension? |
| 16:14.17 | elixer | darkskiez: you have to enter every extension, iirc. |
| 16:15.42 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.12.183) |
| 16:20.25 | *** join/#asterisk michael-i (n=michael-@Le70c.l.pppool.de) |
| 16:21.52 | michael-i | i have a cosmetic problem with my voicemail setup. when a call from an external line ends up at my voicemail greeting, the internal extension is read back. is there any way to override the extension that voicemail reads back? |
| 16:22.09 | Qwell | michael-i: Have the user record their name |
| 16:22.16 | [TK]D-Fender | K-TA: maybe its the phone you are using. |
| 16:22.38 | K-TA | fender i have pap2 linksys adapter |
| 16:22.42 | [TK]D-Fender | michael-i: Or maybe, just MAYBE, a personal greeting even! |
| 16:22.58 | [TK]D-Fender | K-TA: Perhaps there is a problem with THAT, and not the X100 |
| 16:23.07 | michael-i | Qwell, that's not an option in my situation, my device does not have any permanent storage and any greetings would be lost with a reboot |
| 16:23.09 | Qwell | erm, is it the greeting that overrides exten? |
| 16:23.18 | michael-i | would be nice though! :) |
| 16:23.57 | elixer | michael-i: buy a hard drive. then have them record their name. |
| 16:23.59 | elixer | ;-) |
| 16:24.17 | elixer | (kidding) |
| 16:24.21 | [TK]D-Fender | Qwell: in order of availability : temp message, message requested by App call, name, then # |
| 16:24.23 | elixer | (is this thing on?) |
| 16:24.58 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 16:25.01 | _Raptor_ | hi |
| 16:26.48 | _Raptor_ | does anyone have an idea to this: when i try to transfer a call from my snom to someone else asterisk sends a BYE to the snom (thats ok) but it never calls the third party and the channel is simply gone? |
| 16:27.34 | michael-i | i could patch in another option (presentation extension...or something like that) but was wondering if there was already a workaround |
| 16:27.43 | michael-i | I'll keep hacking ;) |
| 16:28.42 | *** join/#asterisk Cybertoy (n=cybertoy@swillux.swill.org) |
| 16:28.46 | elixer | michael-i: the 's' option of Voicemail() won't work because it silences the instructions as well, eh? |
| 16:29.21 | michael-i | elixer, yes something more than a beep would be nice ;) |
| 16:29.28 | elixer | agreed |
| 16:30.26 | konqi_ | I can't dial out via zaptel - somebody who can help me setup my zapta.conf and the trunk in freepbx ? |
| 16:31.01 | elixer | konqi_: you can give #freepbx a try |
| 16:31.21 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 16:33.15 | konqi_ | i'll do... but i think my zapata.conf has to be corrected |
| 16:36.42 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:37.04 | *** join/#asterisk pacneil (n=pacneil@68.15.17.81) |
| 16:39.15 | *** part/#asterisk Cybertoy (n=cybertoy@swillux.swill.org) |
| 16:42.13 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 16:42.47 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 16:43.48 | [TK]D-Fender | konqi_: You are in the WRONG CHANNEL. |
| 16:44.28 | [TK]D-Fender | _Raptor_: First guess is that you are doing to transfer properly and it is hanging up on your caller. |
| 16:45.07 | konqi_ | this is where digium.com sent me |
| 16:45.11 | ramindia | can any one assists me. how best Asterisk can handle far end NAT Device. |
| 16:46.35 | _Raptor_ | [TK]D-Fender: you mean my third party is hanging up? |
| 16:46.55 | *** join/#asterisk zim (n=zim@zimonline1.demon.co.uk) |
| 16:47.12 | zim | hi all is anyone in here using asterisknow ??? |
| 16:48.14 | Cyorxamp | Hi, I am using (and need to use due to other things) 1.2.19 of asterisk, however I need a fix from 1.2.20 so my card will work... the changelog entry in 1.2.20 reads as... |
| 16:48.15 | Cyorxamp | wcte12xp.c, wctdm24xxp/base.c: Fix for when voicebus based cards |
| 16:48.15 | Cyorxamp | <PROTECTED> |
| 16:48.35 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 16:48.39 | Cyorxamp | Does anybody know which parts are this fix? so I can extract what I need and put it in 1.2.19 ? |
| 16:53.32 | [TK]D-Fender | konqi_: FreePBX is *not* supported here. |
| 16:55.52 | *** join/#asterisk drarem (n=rmcdanie@6532142hfc81.tampabay.res.rr.com) |
| 16:58.11 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 16:58.14 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-172-112.dslextreme.com) |
| 16:58.56 | elixer | Cyorxamp: you should be able to see a diff online |
| 16:59.34 | *** join/#asterisk Strom_M (n=strom@netblock-208-127-172-112.dslextreme.com) |
| 17:00.03 | *** join/#asterisk jsmith (n=jsmith@000-143-916.area3.spcsdns.net) |
| 17:00.09 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:03.12 | elixer | Cyorxamp: i'm not seeing that log message in the ChangeLog for asterisk 1.2.20 |
| 17:03.25 | elixer | Cyorxamp: that looks like a zaptel message |
| 17:03.37 | elixer | s/message/ChangeLog entry/ |
| 17:04.19 | *** join/#asterisk t3rror (n=harrison@gateway.sscgp.com) |
| 17:04.53 | codefreeze | jbot is so helpful! |
| 17:06.15 | drarem | is it possible to set up a simple phone queue on a single linux box over broadband |
| 17:06.41 | drarem | and installed phone card connected to digital line |
| 17:06.42 | elixer | codefreeze: except he keeps correcting me! ;-) |
| 17:07.41 | t3rror | could you all suggest the best setup for someone about to move away from the telco for phone service? |
| 17:07.57 | Dan0maN_Work | heh |
| 17:08.02 | t3rror | i have available naked dsl so i am trying to find the cheapest solution |
| 17:08.07 | t3rror | i have been looking at teliax |
| 17:08.31 | [TK]D-Fender | drarem: I would start with "yes", but you should clarify this "digital line" thing.... |
| 17:08.49 | [TK]D-Fender | t3rror: them or VoicePulse Connect |
| 17:09.18 | drarem | digital line is my phone provided by the cable company |
| 17:09.23 | t3rror | voicepulse lets you byod? |
| 17:10.05 | robl^ | "Voice Pulse Connect" does.. |
| 17:10.12 | drarem | as for why I need to be connected to the 'phone line', I don't know except for maybe calling the live person |
| 17:10.19 | [TK]D-Fender | t3rror: Any way you want it. provided, byod, but VPC is their direct to * service. |
| 17:10.26 | [TK]D-Fender | t3rror: They cater to us rather well |
| 17:10.29 | robl^ | not the normal consumer voiceplse |
| 17:10.47 | [TK]D-Fender | http://connect.voicepulse.com/ |
| 17:10.58 | elixer | Cyorxamp: this is commit you are interested in -> http://svn.digium.com/view/zaptel?view=rev&revision=2857 |
| 17:12.08 | drarem | and maybe to receive the call? this is all alien to me :/ I don't know what a siph is but sounds like it's some minion of the dark side |
| 17:12.44 | [TK]D-Fender | drarem: So you have an ATA fdrom your cable co giving you phone service? |
| 17:13.22 | drarem | how can i tell |
| 17:13.51 | drarem | brighthouse, they supply cable and digital |
| 17:14.31 | [TK]D-Fender | drarem: "digital" isn't a "thing", its a format of a medium |
| 17:15.04 | drarem | ok |
| 17:15.29 | [TK]D-Fender | drarem: For all I know you're talking about your ALARM CLOCK. I know *mine* is "digital". |
| 17:15.44 | [TK]D-Fender | cpm: This is a family channel! |
| 17:15.45 | drarem | lol |
| 17:15.56 | cpm | Oh, sorry, what I get for being absent minded |
| 17:17.40 | *** join/#asterisk [X-tp] (n=xtp@c-c19e70d5.015-136-6b736410.cust.bredbandsbolaget.se) |
| 17:19.03 | drarem | how does it work, I get a call in, it goes onto the computer into a queue, and calls out one by one? |
| 17:20.47 | [TK]D-Fender | drarem: At this point I'm figuring you've never even INSTALLED Asterisk |
| 17:21.10 | [TK]D-Fender | drarem: Go download Asterisk and go read THE BOOK. |
| 17:21.17 | [TK]D-Fender | ~book |
| 17:21.18 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:21.47 | elixer | keeping in mind a good chunk of it is obsolete |
| 17:21.49 | elixer | heh |
| 17:21.51 | elixer | (sorry) |
| 17:21.59 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 17:22.02 | elixer | isn't the second edition out soon? |
| 17:22.17 | drarem | it's installed along with gnuk whatever that does |
| 17:22.24 | drarem | i'll go read |
| 17:23.20 | *** join/#asterisk ToTo (n=ToTo@host72-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 17:23.45 | *** join/#asterisk alurin (i=wirus2@static-ip-193-151-98-176.promax.media.pl) |
| 17:23.45 | *** join/#asterisk ToTo (n=ToTo@host72-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 17:23.53 | alurin | hi |
| 17:24.12 | alurin | who using at320? |
| 17:26.22 | *** part/#asterisk bkw_ (n=brian@adsl-70-142-41-246.dsl.tul2ok.sbcglobal.net) |
| 17:28.54 | *** join/#asterisk MdeP (n=mdep@167-130.leased.cust.tie.cl) |
| 17:30.21 | TheDingy | ok... using a 7940 sip no problems when on the localnet but when on a wan node it gets an unautorized reply |
| 17:30.24 | TheDingy | same config both places |
| 17:30.58 | jsmith | TheDingy: Is your phone registering to an IP address, or a domain name? |
| 17:31.22 | jsmith | TheDingy: If it's registering to an IP address, I'll bet your firewall is rewriting the IP address, and hence messing up the SIP authentication |
| 17:31.41 | TheDingy | ip address |
| 17:31.53 | TheDingy | there isn't any rewriting between |
| 17:32.01 | alurin | little problem |
| 17:32.30 | alurin | I have a login my friend |
| 17:32.34 | TheDingy | i have looked at that, but can't figure out what rule would be rewriting it |
| 17:32.41 | TheDingy | jsmith you have any ideas? |
| 17:32.42 | alurin | and I want to dial to him with my ip telephone |
| 17:32.52 | alurin | We are in the same voip network |
| 17:33.14 | alurin | but How I can type login at a telephone? |
| 17:33.16 | alurin | ;p |
| 17:33.18 | jsmith | TheDingy: What type of firewall is between the phone and the Asterisk server. (Let me guess... a Cisco Pix?) |
| 17:33.20 | TheDingy | what if i cange it to register to say pbx.companyname.com would that work? |
| 17:33.30 | TheDingy | no, linux w/openvpn |
| 17:33.32 | jsmith | TheDingy: It would make it *more* likely to work |
| 17:33.39 | t3rror | is there any way to unlock the linksys SPA2002 earthlink branded ATA? |
| 17:33.39 | TheDingy | ok |
| 17:33.46 | TheDingy | let me try that then |
| 17:34.17 | alurin | guys, letters into digits? |
| 17:34.33 | jsmith | alurin: It all depends on the phone, etc. |
| 17:34.46 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-191-178.dhcp.embarqhsd.net) |
| 17:34.59 | jsmith | alurin: The easiest would be to both register to a SIP proxy (like Free World Dialup) and call each other numerically that way |
| 17:35.39 | De_Mon | OR, buy two cans of soup and a pair of shoes, tie it all togeather and be done with it |
| 17:36.11 | alurin | De_Mon, thx... |
| 17:36.35 | TheDingy | jsmith; why would it make a differance using a fqn rather than an ip? |
| 17:36.57 | TheDingy | and when you are saying domain you mean setup a dns entry correct |
| 17:36.58 | alurin | jsmith, I uses h.323 |
| 17:37.16 | jsmith | TheDingy: Because * uses a bunch of things in the SIP headers as part of the SIP authentication... one of those things is the IP address. |
| 17:37.35 | alurin | I use* |
| 17:37.36 | jsmith | TheDingy: If the IP address is getting rewritten by a firewall for NAT and/or VPN purposes, then the credentials fail to authenticate |
| 17:37.54 | jsmith | alurin: Oh, if you're using Asterisk, then simply assign extensions in the dialplan that dials the other phone |
| 17:38.00 | TheDingy | i hate vnc |
| 17:38.59 | jsmith | I hate BBQ-flavored potato chips, but I don't know what that has to do with the topic at hand :-) |
| 17:40.24 | TheDingy | lol |
| 17:40.42 | TheDingy | because i had to loginto vnc over a dumpy connection to change the dns |
| 17:40.43 | alurin | maybe somebody using at320? |
| 17:42.22 | *** part/#asterisk pgarcia (n=pedgarci@c934ac83.virtua.com.br) |
| 17:46.43 | [TK]D-Fender | Atcom = cheap garbage |
| 17:48.47 | tzanger | fuck it's a lot of work leaving a company |
| 17:48.55 | tzanger | almost better off to stay |
| 17:48.58 | De_Mon | eh? |
| 17:51.50 | CCFL_Man2 | i think i got it working |
| 17:52.03 | CCFL_Man2 | 2.6.8 kernel, built the zaptel drivers |
| 17:53.45 | *** join/#asterisk CyBeRSwOrD (n=rodo@131.178.98.181) |
| 17:54.02 | TheDingy | jsmith: if you are still around, i have done that, got by the nonlocal domain and now i just get a SIP/2.0 401 Unauthorized |
| 17:54.05 | TheDingy | on sip debug |
| 17:54.09 | CCFL_Man2 | ztdummy and zaptel modules loaded fine |
| 17:56.15 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 17:56.48 | elixer | tzanger: i feel your pain |
| 17:58.04 | Aeudian | I have a serious issue with asterisk and asterisknow, when I emulate a WAN outage (not gateway) by removing the internet feed from the router, and reboot asterisk with no internet (but still able to get to gateway) i am unable to register phones to asterisk nor am i able to register my fxo gateway? what is causing asterisk not to work without an internet feed? |
| 17:58.25 | elixer | Aeudian: dns resolution? |
| 17:58.28 | Aeudian | 0 |
| 17:58.32 | elixer | heh |
| 17:58.34 | Aeudian | i am hosting dns on the asterisk server |
| 17:58.37 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 17:58.40 | Aeudian | we thought it was that, but it still fails |
| 17:59.05 | elixer | Aeudian: nothing in the logs? |
| 17:59.08 | codec | Aeudian: i guess the asterisk is on the same subnet? |
| 17:59.09 | Zodiacal | anyone know why # (prompt for transfer) doesn't work for outgoing calls. works for incoming calls tho.. |
| 17:59.50 | elixer | Zodiacal: passing the T option to Dial()? |
| 17:59.56 | Zodiacal | yeah |
| 18:00.03 | Aeudian | same subnet and no logs, all phones/asterisk are on 192.168.30/24 and upon reboot with asterisk having no internet asterisk says all phones are unconnected and failing to register |
| 18:00.12 | Aeudian | err 192.168.3.0/24 |
| 18:00.15 | elixer | Zodiacal: ok, then i don't know |
| 18:00.16 | elixer | :) |
| 18:00.21 | Zodiacal | :) |
| 18:00.21 | *** part/#asterisk alurin (i=wirus2@static-ip-193-151-98-176.promax.media.pl) |
| 18:00.22 | Zodiacal | T and t |
| 18:01.05 | NOT_guru | i will tare your heart out and feed it to the sales people if you touch that server again |
| 18:01.12 | NOT_guru | sorry wrong window |
| 18:02.24 | Nivex | NOT_guru: s'ok, I think we've all had those moments :) |
| 18:03.03 | elixer | heh |
| 18:05.21 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 18:05.37 | Aeudian | Even when i remove the WAN from the cisco 871 router, i am unable to reload asterisk without the internet feed? |
| 18:07.37 | NOT_guru | sorry about that but help desk should never even try to log into MY servers.. its my ass if he screws it up |
| 18:07.54 | NOT_guru | anyways hows everyones day =) |
| 18:07.55 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 18:08.04 | NOT_guru | i am not THAT mean of a guy ussually |
| 18:09.07 | *** join/#asterisk SpencerBR (n=breno@201-43-78-54.dsl.telesp.net.br) |
| 18:12.19 | tzafrir_home | CCFL_Man2, you downgraded to Sarge??? |
| 18:13.46 | CCFL_Man2 | tzafrir_home: the installer would install a 2.4 kernel anyway, so i just upgraded to 2.6.8, but i had to find the debs myself |
| 18:14.22 | tzafrir_home | CCFL_Man2, that's the Sarge installer. Why would you isntall Sarge nowadays? |
| 18:14.50 | CCFL_Man2 | tzafrir_home: because there is no etch installer for netbooting on a sparc :P |
| 18:15.09 | CCFL_Man2 | i upgraded to the etch packages though |
| 18:17.42 | CCFL_Man2 | then i had to build zaptel with the instructions you gave me, so i need to find the kernel hearers and the other debs required for those |
| 18:18.24 | CCFL_Man2 | needed |
| 18:18.43 | CCFL_Man2 | so i loaded the modules without errors and i'm happy :P |
| 18:18.56 | CCFL_Man2 | because the clock on this netra is super accurate |
| 18:19.54 | CCFL_Man2 | on my emachines it would deviate atleast 7 seconds per day, this won't even deviate a tenth of a second per day |
| 18:26.40 | [TK]D-Fender | And to think NTP could solve all of that... |
| 18:26.47 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-41-246.dsl.tul2ok.sbcglobal.net) |
| 18:26.49 | Nugget | heh |
| 18:28.39 | Aeudian | Something is really messed up with 3 asterisk systems/1 asterisknow system that we have build. I have a cisco 871 router and when there is internet the asterisk system works perfectly on the network. Both the asterisk and phones are on the same network 192.168.3.0/24. Heres the problem, when i unhook the internet to test for an ISP outage, the asterisk system just halts. |
| 18:28.43 | Aeudian | Reloads take 20-30seconds and phones unregister and have a terrible time registering. I have DNS on the same server that asterisk is on as well as ntp, to point all phones and gateways to it to help alleviate an ISP outage. All configurations/settings that we have done (asterisk and phones) use IP's not names. Is there something that asterisk uses by default to check internet status thats causes the machine to not functi |
| 18:28.47 | Aeudian | I need to ensure that the system works when the internet fails so that the backup gateway works, but when asterisk fails to work, the gateway of course wont either. |
| 18:34.45 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:35.04 | flujan | hi all |
| 18:42.58 | flujan | guys, I am trying to implement a fail-over load balanced asterisk solution. |
| 18:43.18 | flujan | We reach 130 concurrent users running asterisk on a dual dual-core opteron. |
| 18:43.37 | flujan | I am having a load of 4 on the machine without monitor these users. |
| 18:43.41 | elixer | Aeudian: the only thing i could find was on the asterisk-user's list, go to the archives and search for "asterisk slows down when unplugging internet cable with VoIP lines" |
| 18:44.34 | flujan | I research a bit and find openser |
| 18:44.41 | elixer | Aeudian: message-id: 4630AB13.9020308@fgasoftware.com |
| 18:44.47 | flujan | are you guys using it for this purpose? |
| 18:58.29 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 18:59.36 | CCFL_Man2 | <PROTECTED> |
| 18:59.37 | CCFL_Man2 | Zaptel Version: 1.2.11 Echo Canceller: MG2 |
| 18:59.43 | CCFL_Man2 | nice |
| 19:01.17 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 19:01.52 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 19:02.02 | *** join/#asterisk killfill (n=killfill@pc-66-133-45-190.cm.vtr.net) |
| 19:02.19 | dijungal | what causes this eroor on the TE100P card? "HDLC Bad FCS (8) on PRimary D-Channnel for span 1" |
| 19:02.38 | Blackthorn | Hi, I've got a problem with my * server basicly stop taking/routing calls. when i log into it i just get a message repeated over and over down the screen very fast |
| 19:02.45 | [TK]D-Fender | dijungal: Manyt hing, lack of clock sync, etc. |
| 19:03.00 | [TK]D-Fender | dijungal: pastebin "ca /proc/interrupts" and "dmesg" |
| 19:03.00 | Blackthorn | [Aug 31 14:53:58] WARNING[26765]: app_dial.c:674 wait_for_answer: Unable to forward voice frame |
| 19:03.01 | [TK]D-Fender | ~pb |
| 19:03.02 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:03.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 19:03.04 | Blackthorn | anyhelp? |
| 19:03.05 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 19:04.03 | *** join/#asterisk masus (n=tet@78.162.16.99) |
| 19:04.11 | masus | hi all |
| 19:04.52 | masus | how to read a value from interbase or mssql database, not for store cdr's |
| 19:05.12 | [TK]D-Fender | masus: "show function ODBC" |
| 19:05.21 | dijungal | if i change the clocking on a span to 0 for internal clocking what do i need to restart to make take effect? |
| 19:05.24 | masus | make a query to the interbase/Firebird or mssql db |
| 19:06.17 | masus | D-Fender thanks |
| 19:06.54 | *** join/#asterisk digimania (n=none@24-119-242-84.cpe.cableone.net) |
| 19:07.24 | masus | No function by that name registered. |
| 19:07.28 | masus | :P |
| 19:07.44 | [TK]D-Fender | dijungal: stop * and redo "ztcfg -vvvv" and then restart * |
| 19:08.05 | dijungal | did that |
| 19:08.12 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 19:08.26 | dijungal | i actually ran ztcfg while * was up and i was on a call and the call dropped |
| 19:08.37 | dijungal | then i restarted * |
| 19:09.00 | Trevor_b | doesnt ztcfg take down asterisk during the cfg phase then restart it? |
| 19:09.02 | dijungal | still getting the same HDLC error tho |
| 19:09.05 | petong | jsmith: you were correct about chan_zap not being compiled. I have fixed this problem, and asterisk now loads chan_zap.so fine |
| 19:09.23 | petong | but I still have no light on the te120p card |
| 19:09.46 | digimania | does anyone know which IBM servers might work well with Asterisk and Digium cards? |
| 19:11.45 | Daviey | digimania: yeah ab i386 type |
| 19:11.52 | Daviey | s/ab/an |
| 19:11.58 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 19:12.23 | lirakis | l8r all |
| 19:12.27 | Qwell | digimania: should be fine in just about anything, as long as it's hardware compatible (ie, pci 2.2 or better) |
| 19:12.28 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 19:12.42 | Qwell | and of course, 3.3v or 5v pci, on the t1 cards |
| 19:12.58 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 19:13.14 | digimania | no, the 336, 345 and 360/365 servers do not work with Digium cards, but I don't know which do work - maybe all else |
| 19:13.26 | Daviey | digimania: It's the OS that matters more with being compatiable |
| 19:13.34 | digimania | not that's not |
| 19:13.36 | digimania | true |
| 19:14.26 | Trevor_b | Trixbox Pro (not that I am saying to use it) has a cert'd hardware list that they say does not have PCI/IRQ issues with digium hardware. |
| 19:14.40 | digimania | I was hoping to find someone actually using IBM's here to see which model |
| 19:14.46 | CCFL_Man2 | so i need the zaptel-modules and zaptel in debian i think |
| 19:15.43 | [TK]D-Fender | dijungal: please provide the information I requested in a pastebin. |
| 19:16.10 | digimania | btw, intel 915, E7221 & E7525 mb's are not compatible either |
| 19:16.24 | CCFL_Man2 | ioctl(ZT_LOADZONE) failed: Invalid argument |
| 19:16.30 | elixer | Qwell: or i could just ask here :) i can't reproduce the pgdn to "..." thing. what category were you using and what size is your terminal? |
| 19:16.53 | Qwell | elixer: it moves by 10, so you need 12 items or less |
| 19:17.04 | CCFL_Man2 | do i need ztcfg if i don't use any zaptel hardware? |
| 19:17.06 | Qwell | erm, maybe 11 or less |
| 19:17.41 | Qwell | and also, if you hit up while on the top ... line, it'll move the page up by one, but it'll keep the cursor on the ... |
| 19:17.53 | [TK]D-Fender | CCFL_Man2: You should if you run zaptel at all |
| 19:18.05 | Qwell | and I just confirmed that it happens with pgup too |
| 19:18.18 | elixer | Qwell: that's weird, you shouldn't even be able to select either of the '...' lines |
| 19:18.22 | elixer | Qwell: i'll play some more |
| 19:18.30 | Qwell | you can't select them... they can't be there when you do it |
| 19:19.02 | CCFL_Man2 | [TK]D-Fender: i'm getting this error: http://rafb.net/p/KNgnhp72.html |
| 19:19.04 | Qwell | on my screen, I have 1-10, and a ... line at the bottom. I'm on 1, hit pgdn, and it shows ..., 12-20, ..., and the cursor is on the top ... |
| 19:19.23 | Aeudian | elixer: i am having a hard time finding this typic on the site, can you point me at a link? |
| 19:19.45 | elixer | Qwell: ok, so your terminal only shows 10 items at a time |
| 19:19.52 | elixer | Qwell: i'll look into it |
| 19:19.58 | CCFL_Man2 | that normal? |
| 19:19.58 | elixer | Aeudian: ummmm, hold on |
| 19:20.15 | Qwell | 11 items, including ... lines |
| 19:20.24 | elixer | gotcha |
| 19:20.32 | Qwell | 10+1, or 9+2 |
| 19:20.38 | elixer | heh |
| 19:20.41 | elixer | or 8 + 3 |
| 19:20.49 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:20.55 | Qwell | I don't think you can get 3 ... lines :p |
| 19:20.59 | elixer | Aeudian: original was posted on april 26, 2007, does that help? |
| 19:21.13 | Aeudian | elixer: possibly let me look that way brb |
| 19:22.08 | elixer | Qwell: i see it |
| 19:22.09 | Aeudian | elixer: found it thank you |
| 19:22.13 | elixer | Aeudian: yup |
| 19:22.17 | CCFL_Man2 | because ztcfg gives me an error that it can't loadzone = us |
| 19:22.37 | Qwell | oh, you know what... |
| 19:22.44 | Qwell | elixer: 1-10, then 12-20 - it skips 11 |
| 19:22.52 | Qwell | and 21, 31, etc |
| 19:23.03 | Qwell | so it's also moving the menu up by one too many |
| 19:23.06 | elixer | Qwell: yeah |
| 19:23.18 | Aeudian | elixer: wow this is exactly my issue, word for word, almost feels that i wrote it lol |
| 19:23.23 | CCFL_Man2 | i just want ztdummy, i don't want to use any zaptel hardware |
| 19:23.25 | elixer | Aeudian: heh |
| 19:23.44 | Qwell | elixer: I definitely like it though.. I'll commit it as soon as it works right :) |
| 19:23.51 | elixer | Qwell: cool |
| 19:24.02 | elixer | Qwell: that could take WEEKS. i kid. |
| 19:24.16 | Qwell | if possible, it would also be awesome if the scroll size was either 10, or the number of visible items in the menu |
| 19:24.24 | Qwell | so, for instance, if I only had 8 lines... |
| 19:24.41 | Qwell | (which would be really rare) |
| 19:24.51 | elixer | gotcha. min(PAGE_SIZE, VISIBLE_ITEMS) |
| 19:24.54 | Qwell | right |
| 19:24.56 | *** join/#asterisk kkn088 (n=kikoun@88-136-56-85.adslgp.cegetel.net) |
| 19:25.11 | elixer | Qwell: please file a bug report for that enhancement ;-) |
| 19:25.18 | Qwell | ;0 |
| 19:25.52 | Qwell | elixer: Did you happen to run across the magic i option? heh |
| 19:26.09 | elixer | Qwell: space invaders? |
| 19:26.15 | Blackthorn | [Aug 31 14:53:58] WARNING[26765]: app_dial.c:674 wait_for_answer: Unable to forward voice frame --- anyone know what this means? it repeats continuious and shuts down the * server until restarted |
| 19:26.18 | Qwell | shhh |
| 19:26.22 | elixer | Qwell: yes. but its unplayable for me. too many refreshes. |
| 19:26.26 | Qwell | we don't talk about what it does in public :p |
| 19:26.27 | elixer | Qwell: ah. sorry. |
| 19:26.30 | elixer | heh |
| 19:26.49 | elixer | Qwell: might be better on the console, but over ssh its hard to "select" the "options" |
| 19:26.56 | elixer | wink wink |
| 19:27.11 | Qwell | yeah |
| 19:27.26 | Qwell | I think I'm the only person who's reported a bug against that "feature"... |
| 19:27.27 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 19:27.37 | Qwell | russell laughed when I told him, because he didn't realize anybody had seen it |
| 19:28.42 | Juggie | heh |
| 19:28.50 | Juggie | someone wrote space invaders into the console? |
| 19:28.58 | elixer | doh! |
| 19:29.02 | elixer | the jig is up! |
| 19:29.03 | elixer | heh |
| 19:29.22 | Qwell | Juggie: I don't know what you're talking about. |
| 19:29.54 | elixer | if the ABE folks find out, its coming out for sure |
| 19:30.30 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 19:30.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:31.32 | *** join/#asterisk centrex (i=centrex@nat/digium/x-470d713e2f66160e) |
| 19:31.42 | CCFL_Man2 | http://rafb.net/p/KNgnhp72.html <---anyone know how to fiz that? |
| 19:31.50 | elixer | Qwell: man, i should have tested on a smaller terminal. my code is a little fragile :-) |
| 19:31.55 | J4k3 | wow this kinda sucks... my trixbox box refuses to actually start asterisk today |
| 19:31.58 | J4k3 | fawk |
| 19:32.01 | J4k3 | :| |
| 19:32.04 | Qwell | J4k3: welcome to trixbox |
| 19:32.08 | J4k3 | well |
| 19:32.44 | J4k3 | seems kinda weird that it'd work fine for like 5 months then get pissy... I'm thinking its a hardware problem |
| 19:33.03 | J4k3 | I had a 'ball bearing' cpu fan start howling like a worn out sleeve fan yesterday on that box... |
| 19:33.19 | Blackthorn | [Aug 31 14:53:58] WARNING[26765]: app_dial.c:674 wait_for_answer: Unable to forward voice frame --- anyone know what this means? it repeats continuious and shuts down the * server until restarted |
| 19:33.38 | J4k3 | ohhhh, I know whats wrong |
| 19:33.51 | J4k3 | stupid mobo isn't initing the x101p anymore, bombing zaptel in the process :P |
| 19:34.27 | J4k3 | oh well, this is just a P3-700 with 192MB ram I scraped up. |
| 19:35.50 | J4k3 | yay for dead pci slots. |
| 19:37.00 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 19:39.19 | Aeudian | elixer: figured it out, it wasnt a domain name, but rather srvlookup=yes under sip.conf, obivously the asterisk system is NOT looking at the internal dns server even though the linux system is |
| 19:40.25 | elixer | Aeudian: ahhh, good. |
| 19:40.37 | elixer | Aeudian: sorry bout that. |
| 19:41.30 | Aeudian | elixer: well im just glad i figured it out, i couldnt put this system onsite knowing if the router or isp goes down the phones dont work |
| 19:44.27 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-202-145.dsl.irvnca.pacbell.net) |
| 19:46.15 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 19:47.27 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 19:48.20 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 19:48.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:48.31 | [TK]D-Fender | J4k3: running Trixbox on a crappy old computer with an X101P. You have nailed the stereotypical "cheap-ass schmuck" right on the head :) |
| 19:49.04 | etfonhomey | [TK]D-Fender, Have you consulted at any clients using "dynamic" T1's to share voice and data over the same T1? |
| 19:49.06 | elixer | Qwell: real work calls, i'll post an updated patch this weekend, unless you get bored... :-) |
| 19:49.21 | [TK]D-Fender | etfonhomey: Nope. only pure voice (partial or full) |
| 19:52.22 | Trevor_b | etfonhomey: THe only times i have dealt with those providers they handed off both PRI and T1 or used some analog breakout crap to hand the voice channels out. But in any case it was standard interconnections, not really sure what your trying to ask though. |
| 19:52.35 | etfonhomey | [TK]D-Fender, My new employer has a satellite office that currently has about 7 employees in it with 6 POTS lines and a dual-DSL Internet connection. We're looking to replace their phone system and I'm trying to keep them from going to a $10K Avaya solution. |
| 19:53.06 | Qwell | etfonhomey: Digium may do consulting for setting a data/voice T1 up |
| 19:53.23 | etfonhomey | They are currently paying $720 / month for voice and data. |
| 19:53.31 | [TK]D-Fender | etfonhomey: And you figure partial PRI w/ data will do it? |
| 19:53.39 | etfonhomey | [TK]D-Fender, yes. |
| 19:53.50 | etfonhomey | [TK]D-Fender, I don't want to deal with POTS. |
| 19:54.32 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 19:55.19 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 19:55.26 | Trevor_b | 1Mbps and 6line PRI usually does OK under 10 people as long as you dont have major bandwidth needs. |
| 19:56.28 | *** join/#asterisk heelios (n=heelios@onyx.6pixies.com) |
| 19:56.48 | Trevor_b | or at least thats the breakdown if they allow dynamic allocation on the PRI, a few of the places I dealt with had a requirment of X, Y, or Z on the PRI side, so they just didnt add 1 more line at a time, but then that all depends on the providers provisioning of your circuit. |
| 19:58.44 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 19:58.57 | etfonhomey | Trevor_b, thanks for the info!! |
| 19:59.55 | Trevor_b | etfonhomey: Ask the provider what type of handoff they will give. My experience is you will get 2 interfaces 1 for each side, but I have heard of custom hardware being in the mix, but i think that was just the CPE breakout to get the interfaces to you. |
| 20:00.15 | etfonhomey | Trevor_b, I'm soliciting quotes at the moment. |
| 20:00.35 | Trevor_b | should make that part of the RFP, description of CPE and interfaces. |
| 20:00.45 | Trevor_b | just to make sure nothing hinky shows up later from a vendor. |
| 20:01.33 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:01.35 | etfonhomey | Trevor_b, next thing I need is a local vendor in the general area of Florham Park, NJ for troubleshooting and helping me install it. |
| 20:01.37 | Trevor_b | id hate to order PRI and find the company that breaks out to analog for you ;), what a favor that is :) |
| 20:02.44 | etfonhomey | I've never done anything with asterisk other than POTS and SIP. Never messed with setting up the voice via PRI. |
| 20:02.45 | Trevor_b | etfonhomey: You will find that you probably dont need one if you have 'some' hardware experience. Or at least i mean a 'local' one. Once the system is up (and an engineer can remotely install your Linux OS for you as well) they usually can ssh into the system and test from there. |
| 20:03.19 | Trevor_b | PRI is a little tricky at first if your used to POTS, but its much nicer once your done. Less headache, or at least for me it has been. |
| 20:03.49 | etfonhomey | Well, I know how much of a headache using a POTS line was in my very first * setup. |
| 20:04.00 | Trevor_b | we actually do that for offices in other states. Just remote install the OS and setup everything, but we also manage via VPN's too. |
| 20:04.32 | Trevor_b | [TK]D-Fender: was it you i was talking to about vncconnect from lilo? |
| 20:04.44 | Trevor_b | like weeks or months ago |
| 20:05.09 | etfonhomey | I'm pretty sure once the initial config is setup, that I could handle almost all management. I could figure it out all the PRI stuff, but since it's a production system, I won't have the time to mess around. |
| 20:05.41 | Trevor_b | Digium has free installation support for their cards, use it ;) probably wont need anyone extra after that. |
| 20:05.41 | rodent|S | download, and seed. bitchez. |
| 20:05.53 | rodent|S | http://www.stonedcoder.org/tt/details.php?info_hash=da957418bff04c17aa1e357979084665643ef7e0 |
| 20:05.59 | centrex | rodent|S, quiet you! |
| 20:06.01 | [TK]D-Fender | Trevor_b: nope |
| 20:06.05 | rodent|S | blackhat_2007_audio_torrent |
| 20:06.13 | rodent|S | centrex: no u. |
| 20:11.18 | masus | how to recompile asterisk afetr install unixodbc |
| 20:12.04 | masus | or what i have to enable "show functions ODBC" |
| 20:13.22 | TheDingy | who knows about sip pretty well around here |
| 20:13.36 | TheDingy | i am having a problem with a cisco phone on a remote wan port |
| 20:14.58 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 20:15.17 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:15.18 | tzafrir_home | Please link bad voip-info pages to http://www.voip-info.org/wiki/view/PageNeedsRevision |
| 20:15.27 | Qwell | tzafrir_home: probably every page |
| 20:15.34 | tzafrir_home | So they would appear on the backlinks of that page |
| 20:15.34 | masus | :) |
| 20:15.37 | Qwell | by default, all pages should be linked from there |
| 20:16.10 | tzafrir_home | so if you care, pick a small domain, and just make sure it is OK |
| 20:16.22 | rvhi | anyone knows a good national carrier for sip trunk |
| 20:16.22 | tzafrir_home | For instance, users.conf is very badly-documented |
| 20:16.34 | rvhi | this is for commerical, so stability is critical |
| 20:16.48 | Qwell | rvhi: Get a PRI |
| 20:16.53 | mvanbaak | rvhi: define 'national' |
| 20:17.13 | rvhi | national has numbers everyone in the country |
| 20:17.26 | masus | "show function ODBC" doesn't return anything..After compile unixODBC, Do I need to load the module or something? |
| 20:17.31 | rvhi | PRI can only get local numbers |
| 20:17.42 | rvhi | too costly to get a NY number when you are at CA |
| 20:17.43 | Qwell | rvhi: no |
| 20:17.44 | Corydon76-dig | LOL |
| 20:17.50 | Qwell | You're wrong. |
| 20:17.54 | centrex | PRI can only dial local numbers where? and with what telco? |
| 20:17.57 | Corydon76-dig | func_odbc doesn't define a single function |
| 20:18.02 | [TK]D-Fender | masus: yes, you need to redo * completely |
| 20:18.15 | masus | all ? |
| 20:18.18 | rvhi | PRI, for inbound numbers |
| 20:18.19 | Corydon76-dig | func_odbc defines any number of functions, which default to start with the prefix "ODBC_" |
| 20:18.22 | masus | :S |
| 20:18.29 | mvanbaak | ./configure && make && make install |
| 20:18.31 | mvanbaak | yeah |
| 20:18.37 | Corydon76-dig | but you need to define them first in func_odbc.conf |
| 20:18.45 | masus | where is the file |
| 20:18.51 | masus | func_odbc.conf |
| 20:18.58 | mvanbaak | in /etc/asterisk |
| 20:18.58 | Corydon76-dig | in /etc/asterisk |
| 20:19.12 | masus | :S |
| 20:19.26 | mvanbaak | at least we agree :) |
| 20:19.37 | Corydon76-dig | Everybody who uses func_odbc loves the hell out of it |
| 20:19.59 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 20:20.01 | masus | i have not this file |
| 20:20.13 | masus | maybe it's come with another distribution |
| 20:20.15 | masus | i use 1.2 |
| 20:20.22 | Corydon76-dig | There's the problem |
| 20:20.31 | mvanbaak | you need 1.4 |
| 20:20.34 | masus | :S |
| 20:20.36 | Corydon76-dig | You can get func_odbc as a backport, but it's included in 1.4 |
| 20:20.56 | Corydon76-dig | http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 20:20.57 | masus | can u give me a link fr documantaton |
| 20:21.12 | Corydon76-dig | Look at the sample config |
| 20:21.19 | masus | ok thanks i'll see |
| 20:22.21 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:30.18 | masus | how to install this backport .. with svn ? |
| 20:33.45 | *** join/#asterisk Defraz (n=t0tal@66.60.111.181) |
| 20:36.44 | konqi_ | I'm getting an Error when trying to dial out "Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]" anybody knows about this? |
| 20:39.22 | *** join/#asterisk knerd2 (n=knerd4@adsl-154-240-249.jax.bellsouth.net) |
| 20:39.34 | knerd2 | hello all |
| 20:39.56 | knerd2 | im looking for a good place to start learning a2billing |
| 20:40.18 | MACscr | Has anyone tried chanskype and thought it was any good? |
| 20:43.09 | *** join/#asterisk davixx (n=davixx@lns-bzn-58-82-251-215-59.adsl.proxad.net) |
| 20:45.01 | *** join/#asterisk Star568 (n=steve@cpe-75-84-29-38.socal.res.rr.com) |
| 20:45.04 | petong | I am still having trouble with my te120p card. Can't get the light on the front to give me any indication that it is working. zap restart seems to give varying results: |
| 20:45.07 | petong | http://pastebin.com/d20c733a3 |
| 20:45.29 | petong | zaptel and zapata conf here: |
| 20:45.30 | petong | http://pastebin.com/d68d99efd |
| 20:45.42 | Star568 | How to upgrade asterisk from v1.4.4 to V1.4.11, i want to keep all the pre-settings and configs |
| 20:46.33 | petong | output of module show |
| 20:46.34 | petong | http://pastebin.com/d78af2db5 |
| 20:48.02 | masus | svn checkout http://svn.digium.com/svn/func_odbc/branches/1.2 func_odbc |
| 20:48.04 | masus | :) |
| 20:48.07 | masus | is this wrong |
| 20:49.52 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 20:50.19 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 20:51.01 | J4k3 | [TK]D-Fender: trixbox sucks, crappy old computer handles the load fine, x101p only gets used during fiber cuts and other strange telco occurances. my phones are all grandstream 101's or softphones + cheapo bt earpieces... works great 99.99% of the time. |
| 20:51.58 | mvanbaak | ~gs |
| 20:51.59 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:52.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:52.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:53.10 | J4k3 | they work, and they're half the price of the competition.... theres not much to be argued against that |
| 20:53.26 | J4k3 | if you want $150 phone quality, you better spend $150 ;) |
| 20:53.54 | J4k3 | well, mine sucked until I replaced the power supplies. |
| 20:53.56 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 20:53.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:54.12 | J4k3 | but that was zero cost, I have a pile of 5v/2A switching psus here from an old project. |
| 20:54.40 | Kurin- | Why did polycom bother to put their config files in XML format |
| 20:54.50 | Kurin- | this isn't xml |
| 20:54.54 | Kurin- | it's like the anti-xml |
| 20:55.11 | J4k3 | polycom is silly, but some people think they're the coolest thing since sliced bread. |
| 20:55.31 | Kurin- | I like the phones, more or less, but these config files are awful |
| 20:56.26 | J4k3 | like a GSM handset except... not ;) |
| 20:57.17 | MACscr | Polycoms are great IMHO, very good quality and great company backing them |
| 20:57.28 | MACscr | I hate my granstream |
| 20:57.45 | MACscr | I get better quality on my pap2 |
| 20:57.53 | J4k3 | grandstream is just stupid... the phones wouldn't be so bad if their firmware people had actually like EVER used a real phone before. |
| 20:58.11 | J4k3 | but, the damned things refuse to *not* work |
| 20:58.20 | J4k3 | so... they keep running here |
| 20:58.47 | Corydon76-dig | Because if there's one thing I want, it's a phone that stops working |
| 20:58.48 | J4k3 | I lothe the grandstream hold system... hit hold, hang up... pieceofshit starts ringing. |
| 20:59.41 | J4k3 | theres a red LED on the front... can't figure out what its used for... cuz the blue LCD backlight blinks when I have a message (which can't be seen at any sort of angle, or if theres any light in the room otherwise) |
| 20:59.51 | J4k3 | it certainly doesn't come on when I hit hold |
| 20:59.54 | J4k3 | its just... worthless ;) |
| 21:00.14 | J4k3 | buuuut, I suspect if someone hacked on their firmware some, they'd end up semi-acceptable cheapass phones |
| 21:06.30 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:14.00 | *** join/#asterisk monstertruck (n=monstert@c-66-229-137-156.hsd1.fl.comcast.net) |
| 21:14.17 | masus | i have do it :) , for users who want to know --> svn install http://svncommunity.digium.com/svn/func_odbc/1.2 |
| 21:16.06 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:16.37 | bm-chs | anyone have a few conf files I can peek at to get my lights on snom phones to light up with asterisk? |
| 21:17.05 | mvanbaak | bm-chs: there's some docs in the default sip.conf |
| 21:17.12 | mvanbaak | and have a look at voip-info.org |
| 21:17.16 | bm-chs | I've upgraded all phones to latest 6.5.10 version, so hopefully that should work. |
| 21:17.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 21:17.46 | bm-chs | "some docs in default" -- I just want to see actual conf snippets that work. |
| 21:18.33 | bm-chs | I've got this asterisknow crap and I've finally figured out editing conf files without that gui crap is the way to go . . . but I need to just get a light to freakin go on a phone. |
| 21:21.15 | *** join/#asterisk hellc2 (n=admin@85.137.120.114.dyn.user.ono.com) |
| 21:22.04 | TheDingy | any sip experts around? |
| 21:22.31 | mvanbaak | TheDingy: what's the problem ? |
| 21:22.48 | TheDingy | fighing problems with not authorized on a wan port |
| 21:22.59 | TheDingy | it doesn't look like the headers are being mangled in any way |
| 21:23.37 | mvanbaak | ??? |
| 21:24.55 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 21:25.25 | TheDingy | http://pastebin.com/d165693a0 |
| 21:25.33 | TheDingy | phone works perfect on the local lan |
| 21:29.04 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 21:29.21 | *** part/#asterisk masus (n=tet@78.162.16.99) |
| 21:31.13 | zapp-branigan | hi i have 2 fxo in the digium and i have only one line. and i use G1 to select zap, there is some option in the zaàta to detect the fxo who is not conected to the line and call for the true line ? |
| 21:33.48 | *** join/#asterisk phace (i=HydraIRC@89.146.185.82) |
| 21:33.56 | phace | hi all |
| 21:34.08 | phace | i have one question related to SIP |
| 21:34.21 | phace | i have the new lancom 1724 voip router and pbx |
| 21:34.33 | phace | i am trying to connect with sjphone on it |
| 21:34.45 | phace | i can receive calls but i cannot send |
| 21:34.54 | phace | it is always prompting me for a username/password... |
| 21:35.17 | phace | and with x-lite i cannot even receive and send calls but I am online, I can see the status on the pbx |
| 21:35.46 | Star568 | Hi all, How to upgrade asterisk from v1.4.4 to V1.4.11, i want to keep all the pre-settings and configs? |
| 21:39.36 | codefreeze | Star568: shouldn't be a problem |
| 21:40.13 | codefreeze | Star568: from 1.4.4 to 1.4.11, there's just a large number of bug fixes. |
| 21:40.55 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583575.dsl.bell.ca) |
| 21:51.37 | Star568 | codefreeze, thanks for your info. my 1.4.4 does not give me ring back on SIP leg. i plan to upgrade it 1.4.11 and see if it got fixed. so all i need is copy /etc/asterisk |
| 21:52.04 | *** join/#asterisk TheDingy (n=Linn@public-access.zogmo.com) |
| 21:54.54 | *** join/#asterisk cirgal (n=robert@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 21:56.28 | cirgal | folks, is there a way I can make * ignore that I'm making a loopback call? |
| 21:59.03 | *** join/#asterisk smultron (n=lukas@cpe-72-179-47-78.austin.res.rr.com) |
| 21:59.08 | zapp-branigan | hi i have 2 fxo in the digium and i have only one line. and i use G1 to select zap, there is some option in the zapata to detect the fxo who is not conected to the line and call for the true line ? |
| 21:59.31 | zapp-branigan | use R1 |
| 21:59.54 | phace | which softphone are good to use and are compatibile with Windows and Linux ? |
| 22:00.07 | konqi_ | With my Digium Te420 asterisk spits out an error when trying to dial out "Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]" anybody knows about this? |
| 22:00.10 | zapp-branigan | can i detect the fxo and not use ? |
| 22:00.20 | smultron | do the Polycom SoundPoint IP601s support the use of an external headset? |
| 22:01.24 | *** join/#asterisk tomcontr3 (n=tomcontr@244-161-246-201.adsl.terra.cl) |
| 22:02.11 | zapp-branigan | can i detect the fxo who not have line and not use ? |
| 22:02.30 | tomcontr3 | does anyone knows a good Digium distributor neaer Florida? |
| 22:03.59 | russellb | netxusa is in south carolina |
| 22:04.02 | mcab | smultron: yes they do |
| 22:04.12 | russellb | tomcontr3: i think there is a list on digium.com |
| 22:04.19 | tomcontr3 | thanks |
| 22:04.27 | tomcontr3 | and same prices? |
| 22:04.42 | smultron | mcab: do they require a special kind? or will a standard 2.5mm audio pin kind work? |
| 22:04.44 | russellb | should be... not sure |
| 22:04.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:05.34 | mcab | smultron: the 601s use the standard RJ9 ports (IIRC), but the new IP3x0 phones use the 2.5mm audio pin kinds |
| 22:05.55 | *** join/#asterisk ManxPower (n=manxpowe@250.sub-70-196-248.myvzw.com) |
| 22:11.05 | [X-tp] | <PROTECTED> |
| 22:11.05 | ManxPower | What a day |
| 22:11.46 | ManxPower | [X-tp]: 1) you would do it in the dialplan using complex dialplan stuff. |
| 22:12.13 | ManxPower | 2) Having your callerid restricted is done by the CARRIER. Get yourself a new carrier |
| 22:12.46 | ManxPower | for 1) look up "ex-girlfriend option" in the mailing list archives. |
| 22:12.48 | ManxPower | ~mailinglist |
| 22:12.49 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 22:12.50 | [X-tp] | Ive got one inbound account for each number in the series of 10 numbers Im using for my experiments |
| 22:13.06 | ManxPower | Why in the world would you do that? |
| 22:13.29 | ManxPower | Most carriers allow multiple numbers on the same account. |
| 22:14.05 | [X-tp] | that would certainly help me... I tried to set the fromuser to one of the other numbers but it didnt like that... |
| 22:14.22 | ManxPower | Uh, callerid is set using callerid= |
| 22:14.57 | cirgal | With respect to bridging calls and transferring calls: is the difference one of perspective? I.e., you transfer a call (you're the other call in this perspective), |
| 22:15.06 | [X-tp] | so this is not valid? "fromuser = <from_ID> : Specify user to put in "from" instead of callerid (overrides the callerid) when placing calls _to_ peer (another SIP proxy). Valid only for type=peer." |
| 22:15.09 | cirgal | and you bridge two calls (you're not the other call in this perspective) |
| 22:15.19 | smultron | mcab: what's the IP3x0 phone? |
| 22:15.26 | ManxPower | not really. It has to do with the sent userid |
| 22:15.39 | ManxPower | not the callerid |
| 22:15.47 | [X-tp] | ok |
| 22:15.51 | ManxPower | you want callerid=NAme <number> |
| 22:15.58 | [X-tp] | Ill try that then |
| 22:16.01 | ManxPower | or better yet LEAVE IT BLANK for your SIP peer. |
| 22:16.11 | ManxPower | Set it in the sip.conf entry for the calling device. |
| 22:16.17 | mcab | smultron: sorry, the new IP330 and IP320 phones they just released |
| 22:16.40 | smultron | mcab: cool, i'll look it up. |
| 22:16.48 | ManxPower | I set the callerid in one place for each sip device -- in the sip device's sip.conf sectiopn |
| 22:17.59 | ManxPower | I'm lucky because I have a DID (phone number) for each device and my carrier lets me set and correctly formatted number as the callerid |
| 22:18.41 | ManxPower | When I used a SIP carrier is my provider, all numbers were on the same account. |
| 22:19.05 | smultron | mcab: those don't seem to support 6 lines, which is what i need. too bad. |
| 22:19.05 | [X-tp] | so if I set it in the sip.conf-file for that phone, it would try and send that number to the operator if nothing else is explicitly stated in extensions.conf? |
| 22:19.07 | ManxPower | Now that I'm using PRI (VoIPoInternet was not reliable enough for my user's requirements) it is still a non-issue. |
| 22:19.17 | ManxPower | [X-tp]: that is correct. |
| 22:19.28 | [X-tp] | ok, thank you |
| 22:31.47 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 22:38.01 | *** join/#asterisk didge (n=mcveighj@bas2-barrie18-1242454602.dsl.bell.ca) |
| 22:38.11 | didge | hi. where can i find help for cmu sphinx ? |
| 22:38.19 | *** part/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 22:46.47 | *** join/#asterisk javb (n=javb@tdev213-167.codetel.net.do) |
| 22:47.43 | javb | I have an asterisk being the center of 4 PBX with asterisk, using IAX. How can make the original callerid to be sent to the destination? |
| 22:50.28 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-41-246.dsl.tul2ok.sbcglobal.net) |
| 22:52.28 | *** join/#asterisk mmmurf (n=chatzill@c-69-181-153-165.hsd1.ca.comcast.net) |
| 22:53.15 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:54.04 | mmmurf | hello... i'm trying to compile asterisk on opensolaris 10, and i am getting the message: crvs: Command not found. The full compile error is here: http://pastie.caboo.se/92900 |
| 22:54.14 | mmmurf | wondering if anyone has any idea what is going on.... |
| 22:55.32 | mmmurf | I tried googling for the error message but have not had any luck |
| 22:57.02 | *** part/#asterisk smultron (n=lukas@cpe-72-179-47-78.austin.res.rr.com) |
| 22:58.26 | *** join/#asterisk ManxPower (n=manxpowe@250.sub-70-196-248.myvzw.com) |
| 23:01.47 | CCFL_Man2 | ioctl(ZT_LOADZONE) failed: Invalid argument |
| 23:01.48 | CCFL_Man2 | Notice: Configuration file is /etc/zaptel.conf |
| 23:01.48 | CCFL_Man2 | line 231: Unable to register tone zone 'us' |
| 23:01.48 | CCFL_Man2 | ZT_DEFAULTZONE failed: Invalid argument (22) |
| 23:02.06 | CCFL_Man2 | what am i missing? |
| 23:02.40 | tzafrir_home | invalid build of ztcfg??? |
| 23:03.10 | tzafrir_home | grep zone /etc/zaptel.conf |
| 23:10.38 | CCFL_Man2 | hmm.. |
| 23:11.22 | CCFL_Man2 | loadzone=us and defaultzone=us |
| 23:15.57 | *** join/#asterisk SplasPood (n=jwb@schizophrenia.paravolve.net) |
| 23:16.25 | CCFL_Man2 | maybe i'll just try rebuilding zaptel |
| 23:20.18 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:22.26 | tzafrir_home | CCFL_Man2, no need to rebuild the whole of zaptel. maybe just ztcfg |
| 23:23.03 | tzafrir_home | this is totally unrelated to the drivers |
| 23:23.18 | tzafrir_home | this is ztcfg and libtonezone |
| 23:23.46 | CCFL_Man2 | tzafrir_home: well, i built the modules with debian, that makes a zaptel-modules .deb from zaptel-source, but i installed the zaptel package |
| 23:24.02 | CCFL_Man2 | possibly libvtonezone is missing |
| 23:24.20 | tzafrir_home | ldd ztcfg |
| 23:24.30 | tzafrir_home | I think it is statically linked to libtonezone |
| 23:24.35 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-157-172.dsl.irvnca.pacbell.net) |
| 23:25.12 | CCFL_Man2 | i don't have ldd installed |
| 23:26.30 | tzafrir_home | CCFL_Man2, do you see any message in /var/log/kern.log ? |
| 23:26.37 | tzafrir_home | when you run ztcfg ? |
| 23:26.48 | tzafrir_home | (regarding the tonezone) |
| 23:27.12 | CCFL_Man2 | yes |
| 23:27.23 | CCFL_Man2 | http://rafb.net/p/KNgnhp72.html |
| 23:27.45 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:29.13 | tzafrir_home | this is on /var/log/kern.log ? |
| 23:29.34 | CCFL_Man2 | actually, my console |
| 23:29.45 | CCFL_Man2 | let me check that |
| 23:30.13 | tzafrir_home | can you check with tail /var/log/kern.log or dmesg | tail |
| 23:33.49 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:34.57 | *** part/#asterisk didge (n=mcveighj@bas2-barrie18-1242454602.dsl.bell.ca) |
| 23:35.04 | CCFL_Man2 | http://rafb.net/p/seKA0x79.html |
| 23:35.11 | CCFL_Man2 | thats what it says |
| 23:35.58 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 23:40.21 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 23:41.18 | CCFL_Man2 | i'm not sure why that is but i know something is broke |
| 23:41.41 | *** join/#asterisk N0S3 (n=cristian@host168.190-136-201.telecom.net.ar) |
| 23:42.19 | CCFL_Man2 | i'm rebuilding zaptel 1.4 |
| 23:42.38 | CCFL_Man2 | i think the modules i'm using are 1.4 but ztcfg is 1.2 |
| 23:42.43 | TheDingy | any sip experts around? |
| 23:42.44 | CCFL_Man2 | something like that |
| 23:50.29 | mmmurf | hello... i'm trying to compile asterisk on opensolaris 10, and i am getting the message: crvs: Command not found. The full compile error is here: http://pastie.caboo.se/92900 |
| 23:51.20 | *** join/#asterisk etfonhomey_ (n=chatzill@mobile-166-214-052-243.mycingular.net) |
| 23:51.52 | CCFL_Man2 | dammity |
| 23:53.29 | *** join/#asterisk w3pog (n=pgrace@aeneas.fierymoon.com) |
| 23:54.05 | w3pog | hello.. I have a very odd problem. I had a working setup with remote phones using nat being able to make calls to our asterisk box |
| 23:54.05 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:54.05 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:54.28 | w3pog | we just changed internet providers and suddenly I'm getting 403's on the SIP INVITEs. |
| 23:54.43 | w3pog | I've checked that the firewall allows udp access through the NAT entry and I can see the calls attempting to come through |
| 23:54.49 | w3pog | but everything gets forbiddens on inbound. |
| 23:56.15 | CCFL_Man2 | you must change you7r firewall settings |
| 23:58.23 | w3pog | I'm watching through tcpdump the communication between the two points |