| 00:07.14 | mvanbaak | paci``: try this: exten => _7.,1,Dial(sip/newz/{EXTEN:1}|60|r) |
| 00:07.25 | paci`` | mvanbaak, yea, igot it wokring |
| 00:08.06 | mvanbaak | I really like Electro Coco |
| 00:13.24 | mvanbaak | file: the commit russell did,..... it did not solve this |
| 00:13.27 | mvanbaak | I still have it |
| 00:13.49 | file | aight |
| 00:14.26 | mvanbaak | I sent my full config (cd /etc/ && tar zcvf etc_asterisk.tgz asterisk) to codefreeze |
| 00:14.56 | mvanbaak | it is my running config, without even editing passwords etc |
| 00:16.30 | mvanbaak | it's freaky to see that only part of your dialplan is loaded on fresh asterisk startup |
| 00:22.01 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.60.15) |
| 00:29.05 | eric2 | exten => _*96.,n(AVAIL),ChanIsAvail(SIP/rc-2005|s) |
| 00:29.15 | eric2 | ${AVAILSTATUS} returns 0 no matter what |
| 00:29.32 | eric2 | how can I find out if the user at extesion rc-2005 is on the phone? |
| 00:32.24 | Kage` | cmantito, you around? |
| 00:33.04 | paci`` | Kage`, guess what |
| 00:33.07 | paci`` | obnauticus = aao_pwner |
| 00:33.12 | paci`` | lawl |
| 00:33.51 | file | eric2: a SIP device is always considered reachable unless you set a call-limit and that limit is reached |
| 00:34.44 | eric2 | ok, so if I set a call limit to 1 for all devices...??? I'm trying to setup paging but keep on barging in on the phone even if someone is already using it |
| 00:34.51 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 00:34.51 | eric2 | paging over the phone speaker |
| 00:35.38 | file | if you set a call limit of 1 then the device will be considered unavailable if a call (either inbound or outbound) is in progress |
| 00:35.49 | mvanbaak | you can use the GROUP* functions for that |
| 00:35.58 | file | that is another possibility indeed |
| 00:36.18 | file | but is this what you need to accomplish what you seek? I do not know... |
| 00:36.38 | mvanbaak | probably |
| 00:36.55 | eric2 | I wnat to be able to talk to someone on their phone with an intercom type feature |
| 00:36.58 | mvanbaak | because setting call-limit to one does not allow SIP transfer etc |
| 00:39.05 | Greek-Boy | how do I use only the first digit of ${CALLERID(num)} and set the condition that it must be three digits long. Eg, if caller ID is from 2XX |
| 00:40.21 | *** join/#asterisk simNIX (n=user@82-204-21-111.dsl.bbeyond.nl) |
| 00:46.26 | eric2 | is there a way to get a substring from 'core show channels' where I'd check to see if the extension is in use...? |
| 00:49.10 | mvanbaak | no |
| 00:49.27 | mvanbaak | but it's easy to do in $random_language |
| 00:49.39 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 00:49.41 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 00:49.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 00:50.59 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 00:50.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 00:51.19 | *** join/#asterisk darius_ (n=darius@humility.bourg.net) |
| 00:51.43 | darius_ | Can anyone recommend a simple iax provider that terminates to PSTN? |
| 00:52.22 | znoG_ | anyone know what might cause an error like "app_stack.c:370 local_write: Tried to set LOCAL(dialstr), but we aren't within a Gosub routine" ? |
| 00:53.37 | file | znoG_: the LOCAL dialplan function can only be used when you are using the Gosub() application instead of Macro() |
| 00:56.26 | mvanbaak | good night all |
| 00:56.50 | darius_ | I like teliax but the toll-free and DID numbers are expensive |
| 00:58.29 | znoG_ | file: Asterisk seems to call the LOCAL dialplan function automatically when I call use Macro() .. I'm not calling LOCAL() myself .. is that normal? |
| 00:58.40 | znoG_ | s/use// |
| 00:59.43 | file | Asterisk is not doing that... have you examined the dialplan logic using dialplan show? |
| 01:04.22 | znoG_ | file: i kinda lie, it looks like pbx_ael is doing that |
| 01:04.42 | znoG_ | file: i should have mentioned I've got my dialplan setup using ael |
| 01:05.32 | znoG_ | i guess if i set it up in extensions.conf it'd work .. i was hoping i could stick to using AEL |
| 01:05.48 | file | yes, that is important to mention as AEL does stuff for you |
| 01:06.10 | znoG_ | yep, looks like it does the Set using the LOCAL() function |
| 01:06.20 | znoG_ | so I guess I have no way of getting around that, right? |
| 01:06.35 | file | I do not know, I do not know AEL |
| 01:07.17 | znoG_ | maybe if I use ${ARG1} and stuff, instead of ${str} |
| 01:07.21 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.176.208) |
| 01:07.40 | file | are you directly calling Macro? |
| 01:11.26 | znoG_ | hah, yes! it works! with ${ARG1} |
| 01:11.43 | znoG_ | yep, i'm using LDAP and calling Macro(macroname,arg1,arg2); |
| 01:12.10 | file | that is probably why... there is an AEL specific way to create/call macros that use gosub |
| 01:12.32 | darius_ | se noai |
| 01:12.32 | darius_ | ase ai |
| 01:12.36 | file | by calling Macro you bypassed that, and the logic used in the macro assumed it was going to be called with Gosub |
| 01:12.44 | file | that is probably why it did what it did |
| 01:12.59 | file | I had to guess at least. |
| 01:13.47 | znoG_ | oh, right ... you mean by using ¯o |
| 01:13.49 | znoG_ | that's the AEL way |
| 01:14.12 | znoG_ | but, since I have to call an actual Asterisk application (Macro()), I don't have much of a choice |
| 01:17.13 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM001ac3152abe.cpe.net.cable.rogers.com) |
| 01:17.36 | *** join/#asterisk eatmypiano_ (n=eatmypia@CPE00195b4be0d6-CM001ac3152abe.cpe.net.cable.rogers.com) |
| 01:18.37 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM001ac3152abe.cpe.net.cable.rogers.com) |
| 01:20.38 | JayTee52 | mmmm, tasty baby grand |
| 01:20.48 | JayTee52 | and the strings are extra chewy |
| 01:22.41 | *** join/#asterisk Olobola (i=Olobola@69.sub-75-210-170.myvzw.com) |
| 01:23.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:23.46 | Olobola | trying to get realtime up and running: find_engine: Realtime mapping for 'extensions' found to engine 'mysql' |
| 01:23.48 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 01:27.18 | cmantito | darius_: http://inerail.net |
| 01:55.52 | obnauticus | JT, are you there? |
| 02:00.22 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 02:01.08 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 02:05.26 | znoG_ | file: while we're here :) is it possible, within a macro, to play a sound file, wait for an input, do something with it .. etc? |
| 02:05.30 | znoG_ | or is that only in a context |
| 02:05.53 | *** join/#asterisk adjohn (n=adjohn@p7221-ipad87marunouchi.tokyo.ocn.ne.jp) |
| 02:12.17 | *** join/#asterisk steliosk (n=Stelios@athedsl-14339.home.otenet.gr) |
| 02:30.10 | jameswf-home | ~ch6 |
| 02:30.11 | jbot | Read about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 02:31.48 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-183-106.dsl.tul2ok.sbcglobal.net) |
| 02:36.43 | Nugget | ; Advanced Extension |
| 02:36.43 | Nugget | exten => rickroll,1,Answer() |
| 02:36.43 | Nugget | exten => rickroll,2,Playback(never-gonna-give-you-up) |
| 02:38.27 | Nivex | Nugget: if there were some way to rip out your jugular over IP, I would do so. |
| 02:39.07 | obnauticus | Nugget that's a good idea |
| 02:39.47 | obnauticus | im going to add that to my default hold music |
| 02:49.29 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 02:51.04 | jameswf-home | if you add rickroll you have to add fleamarket |
| 02:51.10 | obnauticus | ? |
| 02:51.12 | jameswf-home | ~fleamarket |
| 02:51.12 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 02:51.33 | jameswf-home | ~ronshouse |
| 02:51.34 | jbot | extra, extra, read all about it, ronshouse is http://www.youtube.com/watch?v=R6P8qssKrSw |
| 02:51.39 | obnauticus | OH GOD |
| 02:51.40 | obnauticus | WTF |
| 02:51.40 | jameswf-home | lmao |
| 02:52.29 | putnopvut | jameswf-home, there's already ~whatnow |
| 02:52.33 | putnopvut | ~whatnow |
| 02:52.33 | jbot | hmm... now is a good time to tell you that I have 6 gigabytes of data |
| 02:52.37 | putnopvut | ~nowwhat |
| 02:52.37 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 02:52.38 | putnopvut | I meant |
| 02:53.45 | jameswf-home | Between em all you would have people calling in just to be put on hol. |
| 02:53.51 | jameswf-home | *hold |
| 02:54.40 | obnauticus | Why is err, musiconhold only playing like 2 of my music files |
| 02:54.58 | jameswf-home | new telemarketer torture.... please hold while we connect you <rickroll> sorry it will be just another minute <talkin bout flea market> |
| 02:55.41 | jameswf-home | dont hang up we really want your time share |
| 02:58.21 | *** join/#asterisk Olobola (i=Olobola@69.sub-75-210-170.myvzw.com) |
| 03:05.28 | paci`` | hhhhhhhhhhhhhahahaha scientology got our pizzas in dalas |
| 03:06.20 | obnauticus | fail. |
| 03:08.04 | *** join/#asterisk graphx (n=chatzill@fl-204-215-37-142.sta.embarqhsd.net) |
| 03:08.45 | *** part/#asterisk graphx (n=chatzill@fl-204-215-37-142.sta.embarqhsd.net) |
| 03:09.03 | *** join/#asterisk graphx (n=chatzill@fl-204-215-37-142.sta.embarqhsd.net) |
| 03:15.18 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 03:27.49 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 03:29.54 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 03:36.16 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:36.16 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/03/05), Asterisk 1.4.19-rc2 (2008/03/11), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Happy birthday mvanbaak. |
| 03:39.52 | outtolunc | bing bong |
| 03:42.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:49.49 | jameswf-home | ping pong |
| 03:52.56 | TJNII | |. | |
| 03:52.56 | TJNII | | . | |
| 03:52.56 | TJNII | | .| |
| 03:52.56 | TJNII | | |. |
| 03:52.56 | TJNII | Damn. |
| 03:56.36 | Qwell | jameswf-home: I like it.. you should make that dialplan |
| 03:58.08 | [TK]D-Fender | ~rickroll |
| 03:58.09 | jbot | rickroll is, like, http://www.internetisseriousbusiness.com |
| 03:59.03 | Qwell | [TK]D-Fender: if you're gonna do it - do it right. http://linuxgeneration.org/pics/trucks/ |
| 03:59.09 | jameswf-home | someone in the trix forums asked what the name of the voicemail app is.... ummmmm asterisk |
| 03:59.11 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 03:59.18 | Qwell | ^ leet carputer |
| 03:59.45 | coppice | ~rickroll is also http://www.xkcd.com/396/ |
| 03:59.46 | jbot | okay, coppice |
| 04:00.12 | [TK]D-Fender | ~nowwhat |
| 04:00.12 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 04:00.13 | jameswf-home | lol you didnt |
| 04:00.42 | jameswf-home | ~ronshouse |
| 04:00.43 | jbot | it has been said that ronshouse is http://www.youtube.com/watch?v=R6P8qssKrSw |
| 04:07.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:10.47 | *** join/#asterisk RoyK (n=roy@ti211110a081-2200.bb.online.no) |
| 04:12.29 | *** part/#asterisk RoyK (n=roy@ti211110a081-2200.bb.online.no) |
| 04:17.11 | jameswf-home | is going to write a worm that makes all asterisk servers dial and vote for contestant 1 on American Idol so they cant loose reguardless of who 1 is...... |
| 04:17.46 | b11d | why not make it dial 1-900 numbers that you own? |
| 04:18.04 | b11d | or that a shell "holding corporation" owns that you control? :P |
| 04:18.13 | jameswf-home | I dont wanna cost anyone money... just rig the game |
| 04:18.19 | b11d | fair enough |
| 04:18.40 | jameswf-home | bet I could get people to run it willingly.. |
| 04:18.51 | b11d | i'll tie up my entire PRI for a few hours each night for you |
| 04:19.20 | b11d | . |
| 04:20.08 | jameswf-home | there must be a glich in the matrix, contestant 1 aways gets the most votes.... I could even randomize callerID if the provider allowes it |
| 04:37.56 | Olobola | can someone help with my realtime installation.. I've been trying get this running for 6 hours now. WARNING[23669]: config.c:1228 find_engine: Realtime mapping for 'extensions' |
| 04:38.09 | Olobola | mysql |
| 04:39.11 | Olobola | database is setup |
| 04:40.01 | Olobola | I went through several tutorials, everthing seems to be ok, still getting to above error |
| 04:46.32 | *** join/#asterisk VoiceCX (n=VoiceCX@216.10.136.139) |
| 04:47.16 | *** join/#asterisk joelsolanki (i=joelsola@220.224.82.7) |
| 04:47.20 | joelsolanki | Hi room |
| 04:47.33 | joelsolanki | my pbx on DMZ |
| 04:47.39 | joelsolanki | no firewall rules except DMZ |
| 04:47.53 | joelsolanki | the dsl modem does the DMZ stuff |
| 04:48.12 | joelsolanki | and extensions are also on private ip |
| 04:48.20 | joelsolanki | there is one way audion issue. |
| 04:48.25 | jer | sounds exactly like my setup, what's the issue? |
| 04:48.31 | joelsolanki | it is preety sure that it is due to NAT |
| 04:48.48 | joelsolanki | what are the things i have to set to resolve this issue |
| 04:49.41 | joelsolanki | In extension i have already set NAT=yes |
| 04:49.48 | jer | you haven't said what your issue is |
| 04:49.49 | joelsolanki | but that didn worked :( |
| 04:49.57 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.90.20) |
| 04:50.00 | joelsolanki | i told already |
| 04:50.08 | joelsolanki | thee is one way audio issue |
| 04:50.25 | jameswf-home | ~nat |
| 04:50.26 | jbot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 04:50.38 | jer | jameswf-home, beat me too it =] |
| 04:50.44 | jameswf-home | ~flood |
| 04:50.45 | jbot | [flood] spewing loads of output into a channel; *very* rude in most channels and often grounds for banning. If you want to show a lot of output to someone, ask them to join you in #flood and paste the output there. |
| 04:51.56 | joelsolanki | ok let me try to keep externalip |
| 04:58.47 | VoiceCX | is anyone having issues with PAP2s or PAP2Ts??? |
| 04:59.09 | *** join/#asterisk shtoom (n=godson@123.176.41.46) |
| 05:01.06 | VoiceCX | my adapters used to work fine and then about 2 weeks ago, they just started going to crap |
| 05:01.27 | VoiceCX | everything on the user end sounds fine, but the caller side is warbly |
| 05:01.33 | VoiceCX | anyone have any ideas |
| 05:01.47 | VoiceCX | I've already changed out cables |
| 05:02.01 | VoiceCX | defaulted the boxes and reconfigured |
| 05:02.09 | VoiceCX | upgraded the firmware |
| 05:02.28 | VoiceCX | even bought two 729A licenses from Asterisk |
| 05:02.36 | VoiceCX | nothing seems to fix the problem |
| 05:15.40 | codefreeze | znoG_: still around? |
| 05:30.27 | jblack | VoiceCX: Loaded network, perhaps? |
| 05:36.04 | TJNII | So the pap2s will run fine for 2 weeks and then die? Or they died two weeks ago and now you can't get them to work right at all? |
| 05:38.21 | VoiceCX | they haven't died |
| 05:38.41 | VoiceCX | they just sound warbly to whoever you call, or even if they call you |
| 05:39.36 | TJNII | So the pap2s will run fine for 2 weeks and then do this? Or this started two weeks ago and now you can't get them to work right at all? |
| 05:39.56 | TJNII | Is it one pap2 doing this or multiple? |
| 05:40.30 | VoiceCX | I have a PAP2-NA and a PAP2T-NA |
| 05:40.46 | VoiceCX | and they both started doing this around the same time two weeks ago |
| 05:41.11 | TJNII | Any other SIP devices doing this, or just the PAP2s? |
| 05:41.23 | VoiceCX | just the PAPs |
| 05:41.42 | VoiceCX | I have tried out softphones and I have one Grandstream GXP2000 |
| 05:41.46 | VoiceCX | they all work fine |
| 05:42.07 | TJNII | Weird. |
| 05:42.18 | TJNII | Did you do anything of noe two weeks ago? |
| 05:42.32 | TJNII | Firmware upgrade or anything of the like? |
| 05:43.09 | TJNII | s/noe/note/ |
| 05:44.17 | *** join/#asterisk VoiceCX (n=VoiceCX@216.10.136.139) |
| 05:44.34 | VoiceCX | sorry...I hit the wrong button |
| 05:53.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:04.14 | dssman | :S how can I mount my usb key to linux from shell? |
| 06:08.05 | TJNII | Really a #linux question. Take a look at the man page for mount. |
| 06:14.13 | dssman | yea man page aint helpin this n00b too much |
| 06:17.15 | TJNII | dssman: /join #linux Lots of people are awake there. I'm going to bed. :) |
| 06:19.49 | TJNII | dssman: http://forums.gentoo.org/viewtopic-t-220005-highlight-usb.html |
| 06:22.42 | *** join/#asterisk hohum (n=dcorbe@wsip-70-183-104-58.sd.sd.cox.net) |
| 06:24.32 | obnauticus | drmessano you there? |
| 06:25.55 | drmessano | Yeah |
| 06:26.43 | obnauticus | so im apperentally watching slowpoke go arcoss a post on 4chan |
| 06:26.47 | obnauticus | it's pretty entertaining. |
| 06:26.49 | obnauticus | http://img.4chan.org/b/res/58438003.html#58445069 |
| 06:30.34 | obnauticus | drmessano isn't this cool? |
| 06:40.01 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.140) |
| 06:45.49 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 06:47.36 | jameswf-home | has written a bot in php now to figure out what to do with it |
| 06:48.04 | jameswf-home | maybe it can be a rickroll bot |
| 06:48.39 | *** join/#asterisk salzh (n=salzh@124.77.139.230) |
| 07:17.58 | jameswf-home | set the bot to pipe verbose output of asterisk here :] |
| 07:20.14 | jameswf-home | that could be a handy debugger.. if freenode wouldnt dump it.... build an ircbot that creates a private room and floods CLI output |
| 07:24.53 | *** join/#asterisk svenna_ (n=svenna@p548D2BBF.dip0.t-ipconnect.de) |
| 07:27.43 | *** join/#asterisk potsboy (n=chrisg@41.246.139.61) |
| 07:32.37 | *** join/#asterisk stoffell (n=stoffell@d51A4DF41.access.telenet.be) |
| 07:38.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:41.07 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 07:55.55 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:18.54 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 08:40.21 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:41.40 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 08:54.21 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 08:55.45 | *** join/#asterisk steliosk (n=Stelios@athedsl-14339.home.otenet.gr) |
| 08:59.39 | DarKnesS_WolF | tzafrir: are u there? |
| 08:59.45 | tzafrir | yes |
| 09:13.30 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 09:19.34 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 09:24.18 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 09:45.28 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:47.39 | jql | 04:41AM <knowpaced:#Perl> "Somni" at 69.178.0.165 pasted "example session for xinming" (22 lines) at http://p3m.org/pfn/572 |
| 09:49.14 | jql | 0-0-cust672.midd.cable.ntl.com] has joined #haskell |
| 10:03.13 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.123) |
| 10:09.20 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 10:24.01 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 10:24.43 | *** join/#asterisk simbol76ss (n=simbol@host198-192-dynamic.59-82-r.retail.telecomitalia.it) |
| 10:30.28 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 10:31.43 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 10:38.14 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 10:38.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:44.29 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 10:46.02 | *** join/#asterisk steliosk (n=Stelios@athedsl-14339.home.otenet.gr) |
| 10:51.59 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 10:54.46 | *** part/#asterisk Ankleteeth (n=chatzill@main.toble.com) |
| 10:58.52 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:01.23 | *** join/#asterisk Chris-NB (n=chris@213162066133.public.t-mobile.at) |
| 11:16.10 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:20.34 | *** join/#asterisk SparFux (n=raoul@e182026002.adsl.alicedsl.de) |
| 11:20.51 | SparFux | To give user access to capi, I only have to give access to /dev/capi* , right? |
| 11:26.44 | *** join/#asterisk simNIX (n=user@82-204-21-111.dsl.bbeyond.nl) |
| 11:27.01 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 11:29.39 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:33.40 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:37.29 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:42.35 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:43.20 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 12:03.53 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 12:34.24 | *** part/#asterisk tomfmason (n=tom@unaffiliated/tomfmason) |
| 12:45.29 | *** join/#asterisk shtoom (n=godson@123.176.41.46) |
| 12:55.15 | *** join/#asterisk mcfloppy (n=Raphael@88-134-142-77-dynip.superkabel.de) |
| 12:55.18 | mcfloppy | hello |
| 13:01.41 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 13:01.45 | Alpha_AI | Hello |
| 13:01.51 | Alpha_AI | how is everyone |
| 13:02.12 | Alpha_AI | ive just installed trixbox CE through VMware player and im trying to log into the gui |
| 13:02.37 | Alpha_AI | not having much luck |
| 13:02.48 | Alpha_AI | apparently the username is: root |
| 13:02.53 | Alpha_AI | but i have no idea what the password is |
| 13:07.31 | stoffell | Alpha_AI, check on #freepbx |
| 13:18.20 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.140) |
| 13:34.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:35.36 | *** join/#asterisk A500mg (n=x@ACaen-156-1-61-167.w90-51.abo.wanadoo.fr) |
| 13:35.39 | A500mg | hello |
| 13:36.43 | A500mg | anybody can confirm to me that b410p have nothing to do with zaptel ? |
| 13:39.07 | JT | yes, uses horrible misdn drivers |
| 13:39.41 | A500mg | (i have a b410p, working good with misdn, and i need to update zaptel, no impact?) |
| 13:39.48 | A500mg | ok :) |
| 13:40.31 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 13:40.38 | hi365 | anyone good with grammer files for luminvox? |
| 13:44.16 | tzafrir | A500mg, right. A500mg the makefile in the SVN version has been updated to point to the latest mISDN |
| 13:45.17 | tzafrir | http://svn.digium.com/view/zaptel?view=rev&revision=3983 |
| 13:46.08 | tzafrir | maybe wget http://svn.digium.com/svn/zaptel/branches/1.4/Makefile |
| 13:46.35 | tzafrir | (untested) |
| 13:46.54 | hi365 | is there a speach recognotion room? |
| 13:47.34 | Alpha_AI | does asterisk have speech recognition support? |
| 13:47.40 | hi365 | yup |
| 13:47.49 | Alpha_AI | ok |
| 13:47.51 | hi365 | using addons, such as luminvox.com |
| 13:48.03 | hi365 | ops, lumenvox.com |
| 13:48.06 | tzafrir | what about sphinx2? |
| 13:48.29 | hi365 | ein li musag |
| 13:48.31 | hi365 | duno |
| 13:48.44 | hi365 | was hoping you know of some VR effort in hebrew |
| 13:50.21 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 13:54.02 | A500mg | thanks for info tzafrir |
| 14:04.50 | tzafrir | hi365, sounds like an interesting research question |
| 14:05.56 | coppice | voice recognition for anything but english is extremely patchy, though a few minority languages are well supported because one or two capable people cared |
| 14:38.30 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 14:43.11 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 14:43.16 | ice_croft | ~book |
| 14:43.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:44.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:45.06 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:49.18 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 14:50.15 | tzafrir | All those talks of telemarketers torture make me wonder |
| 14:50.39 | tzafrir | (here I hardly get those calls, anyway) |
| 14:51.15 | coppice | that can change very fast. we went from about 0 buisance calls to a deluge in a year or so |
| 14:51.28 | coppice | s/buisance/nuisance |
| 14:52.10 | tzafrir | The interesting is that PBX installers make money of both sides |
| 14:53.20 | tzafrir | I just wonder if there are some specific PBX installer who make money both from installing telemarketers centers and counter-meassures in PBXs... |
| 14:54.50 | *** join/#asterisk InsolentDreams (n=Insolent@62.157.211.194) |
| 14:58.19 | coppice | I expect they all try to |
| 14:59.40 | coppice | that is the ones who do telemarketing will definitely be trying to make money from countermeasures |
| 15:04.42 | jblack | selling to both sides at once? |
| 15:13.50 | coppice | isn't that the best way to make money? |
| 15:15.21 | coppice | equally, I'm very suspicious of anti-virus people. I know one of the early sleazy ones definitely assisted in creating a need for his wares. I kinda suspect all the others play the same game |
| 15:17.09 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.67.231) |
| 15:17.34 | *** join/#asterisk Rico29 (n=rico@ARennes-358-1-47-95.w90-49.abo.wanadoo.fr) |
| 15:17.37 | Rico29 | hi |
| 15:18.19 | Rico29 | french people here ? |
| 15:19.10 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:19.12 | Rico29 | i need help please |
| 15:19.21 | Rico29 | for configuring rtp |
| 15:19.39 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 15:21.22 | Rico29 | mmhm |
| 15:21.24 | Rico29 | ok |
| 15:21.26 | Rico29 | :( |
| 15:21.56 | tzafrir | ~ask |
| 15:21.56 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:21.57 | A500mg | oui Rico29 |
| 15:22.02 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 15:22.40 | Rico29 | ah cool |
| 15:22.47 | Rico29 | en gros |
| 15:22.59 | Rico29 | j'ai un asterisk sur un serveur dans mon LAN |
| 15:23.13 | Rico29 | un routeur qui fait passerelle vers l'extérieur |
| 15:23.20 | Rico29 | et des gens qui se connectent dessus |
| 15:23.48 | Rico29 | le problème est que le SIP passe, donc les téléphonnes sonnent |
| 15:23.56 | Rico29 | enfin, toute la sig SIP passe |
| 15:23.59 | Rico29 | mais le RTP : que dalle |
| 15:24.22 | Rico29 | donc je voulais savoir si il me fallait absolument un serveur stun ou pas |
| 15:24.31 | Rico29 | A500mg |
| 15:25.46 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 15:26.17 | A500mg | ahhhhh ben justement hier soir on faisait des tests là -dessus |
| 15:26.22 | Rico29 | :) |
| 15:26.25 | Rico29 | cool |
| 15:26.40 | A500mg | mais j'aidais juste pour les tests, c'est pas moi qu'ait trouvé |
| 15:26.45 | Rico29 | ah |
| 15:26.48 | A500mg | t'a bien mis externhost dans ton sip.conf ? |
| 15:26.49 | Rico29 | balo, ça |
| 15:26.54 | Rico29 | eu non |
| 15:26.55 | Rico29 | jcroi pas |
| 15:26.58 | A500mg | ben il faut |
| 15:27.02 | tzafrir | If you want others to help you here, then English |
| 15:27.04 | Rico29 | dan qle general |
| 15:27.08 | tzafrir | Otherwise, try #asterisk-fr |
| 15:27.10 | Rico29 | tzafrir > ok |
| 15:27.12 | A500mg | ok :D |
| 15:27.22 | Rico29 | nobody on asterisk-fr |
| 15:27.24 | Rico29 | :p |
| 15:27.30 | A500mg | me |
| 15:27.32 | tzafrir | wonders if there is actually #asterisk-fr |
| 15:27.52 | Rico29 | tzafrir > did you understand my problem ? |
| 15:29.38 | *** join/#asterisk machpo (n=machpo@ketone.jawfish.com) |
| 15:32.17 | *** join/#asterisk bantu (n=Miranda@p54A32395.dip0.t-ipconnect.de) |
| 15:35.03 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-204-112.bltmmd.east.verizon.net) |
| 15:36.43 | Rico29 | i have a problem with my asterisk server : the server is in my LAN, behin a router, and clients connect to Asterisk. Sip is OK, but RTP doesn't work, so the phone rings, but when someone answers, I can't hear him, and he can't hear me too |
| 15:36.54 | Rico29 | do I have to use a stun server ? |
| 15:38.53 | *** join/#asterisk adjohn (n=adjohn@p7221-ipad87marunouchi.tokyo.ocn.ne.jp) |
| 15:43.22 | lmadsen | Rico29: sounds like a firewall issue |
| 15:43.48 | Rico29 | my firewall is totally disabled |
| 15:44.12 | _ShrikE | Rico29: but are you using nat? |
| 15:44.22 | Rico29 | yes |
| 15:44.33 | _ShrikE | ~sipnat |
| 15:44.34 | 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 |
| 15:44.36 | Rico29 | my livebox does it |
| 15:45.15 | Rico29 | thanks, i take a look |
| 15:47.36 | Rico29 | can I replace externip=222.222.222.222 by externhost=dns_name ? |
| 15:51.10 | Rico29 | ok |
| 15:51.12 | Rico29 | it works |
| 15:51.16 | Rico29 | than you !!! |
| 15:54.52 | Rico29 | what is the aim of a stun server ? |
| 16:03.39 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 16:05.37 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 16:16.55 | Rico29 | doesn't work anymore |
| 16:16.58 | Rico29 | *argh* |
| 16:17.24 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 16:22.31 | drmessano | why doesnt it? |
| 16:23.26 | Rico29 | don't know |
| 16:23.32 | drmessano | Ok |
| 16:23.39 | drmessano | In WHAT WAY is it NOT WORKING? |
| 16:23.46 | drmessano | In other words |
| 16:23.48 | drmessano | How do you know |
| 16:23.53 | drmessano | Why do you say it isnt |
| 16:23.58 | drmessano | What indication do you have |
| 16:24.02 | Rico29 | mmh |
| 16:24.02 | drmessano | What is it doing? |
| 16:24.05 | drmessano | What is it NOT doing? |
| 16:24.22 | Rico29 | my friend juste said to me that the problem may come from him |
| 16:24.32 | Rico29 | so i wait |
| 16:24.35 | drmessano | ok |
| 16:25.05 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 16:28.03 | Rico29 | his problem is that Xlite doesn't want to send SIP sig on port 5060 |
| 16:29.42 | drmessano | Im not sure how you would even break that |
| 16:31.37 | jameswf-home | sip.conf |
| 16:31.49 | drmessano | <PROTECTED> |
| 16:31.57 | drmessano | Xlite doesnt want to send SIP sig |
| 16:32.01 | drmessano | on 5060 |
| 16:32.36 | jameswf-home | only handshales and call control go on 5060 payload goes 10000-20000 |
| 16:32.56 | Rico29 | he renounced |
| 16:32.59 | Rico29 | :) |
| 16:33.09 | drmessano | I figured as much |
| 16:33.13 | Rico29 | anybody here want to test if my serv works ? |
| 16:33.29 | Rico29 | I give him adress and ids for connecting |
| 16:34.00 | jameswf-home | has been playing with php and sockets... wonder if I can do sip diag with that method.. |
| 16:46.53 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 16:49.16 | *** join/#asterisk danilom (n=danilom@201.209.158.232) |
| 16:51.55 | JayTee52 | I've had problems with Xlite and forcing it to use 5060 for SIP. I've also had problems when a call is placed to an Xlite softphone where I get a brief delay of silence when the caller answers. |
| 16:52.37 | JayTee52 | One of these days I'm gonna try Zoiper if I ever get some time :-) |
| 16:53.09 | drmessano | I've not had the first problem with X-Lite |
| 16:53.19 | drmessano | Never had to give it a port number |
| 16:53.30 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 16:54.04 | TJNII | Why does it matter what port XLite runs on, Xlite is a client. The server port it what matters. |
| 16:54.16 | drmessano | Exactly |
| 16:54.23 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 16:54.29 | TJNII | And if you run XLite on the same box as *, of course its not going to run on 5060. 5060 will already be bound to *. |
| 16:54.50 | Rico29 | yes |
| 16:55.02 | drmessano | Isn't that kinda pointless anyway? |
| 16:55.05 | Rico29 | can somebody make tests with me ? |
| 16:55.24 | Rico29 | just try to connect on my serv, please |
| 16:56.17 | JayTee52 | Rico29, sorry but I don't have VOIP here at home. |
| 16:56.28 | Rico29 | you dont have a softphone , |
| 16:56.30 | Rico29 | ? |
| 16:56.32 | Rico29 | ok |
| 16:56.34 | Rico29 | no pb |
| 16:56.36 | JayTee52 | well, I have Ekiga |
| 16:57.01 | JayTee52 | but I don't think this version does SIP. I know the new version does. |
| 16:57.07 | Rico29 | ok |
| 16:58.01 | *** join/#asterisk stoffell (n=stoffell@d51A4DF41.access.telenet.be) |
| 16:58.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:58.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:58.27 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 16:58.53 | Yourname` | Hi. Is there a way to forcefully make Asterisk use all the CPUs? (multicore, multicpu)? |
| 16:59.18 | russellb | there is nothing you can do in that regard |
| 16:59.28 | russellb | Some aspects of Asterisk are single threaded |
| 16:59.40 | russellb | however, much of it, and all of it that does the heavy lifting, is multi-threaded |
| 16:59.50 | Yourname` | Oh yeah? I thought it was multi completely. Which ones are single? |
| 16:59.54 | russellb | so it really should utilize all CPUs under load ... depending on your usage |
| 16:59.55 | JayTee52 | Rico29, I just configured Ekiga. would you like me to place a test call? |
| 17:00.12 | russellb | SIP signaling processing, for example, is done by a single thread |
| 17:00.49 | russellb | but Asterisk isn't designed to be a SIP proxy :) |
| 17:01.06 | *** join/#asterisk graphx (n=chatzill@fl-204-215-37-142.sta.embarqhsd.net) |
| 17:01.23 | graphx | Has anyone used Druid before? |
| 17:01.26 | Yourname` | Well, the reason I ask to make it forceful is because there is one box that is singleCPU, quad core and asterisk seems to be using all the 4 cores. Then there is another box which is dual core, dualcpu, and only one cpu is being used a lot. For example, CPU 1 is 80-85% in usage, whereas the rest of the 3 are near 20-30%.. |
| 17:01.33 | *** join/#asterisk CVirus (n=GoD@82.201.222.52) |
| 17:01.47 | russellb | hm ... |
| 17:01.53 | Yourname` | So in the second box, it doesn't seem to be taking advantage of all the cores.. :S |
| 17:02.02 | russellb | how is the box being used? |
| 17:02.02 | rickross | does Asterisk 1.6.0 beta (4 or 5) require any changes to cdr_mysql.conf or any noew conf files to make CDR work? |
| 17:02.20 | graphx | Can anyone give me the your opinion on driud in difference between Druid and AsteriskNow? |
| 17:02.22 | Yourname` | For autodialing using callfiles |
| 17:02.23 | russellb | rickross: not that i remember, but if so, it would be noted in UPGRADE.txt |
| 17:03.19 | Yourname` | Actually russellb, both the boxes are being used for autodialing. Well, sort of autodialing, we callback defaulting customers with payment needed messages and regular customers with news"voices". |
| 17:03.20 | rickross | russellb: thanks, my CDR completely pooped out when I moved to 1.6.0beta4 |
| 17:03.59 | russellb | rickross: that's quite odd ... |
| 17:04.18 | Yourname` | russellb: But they do use callfiles. Trying to make it go on to manager originates tho to see the difference. |
| 17:04.44 | russellb | i see .. |
| 17:05.24 | Yourname` | russellb: jared mentioned the other day to see if irqbalance is on or off, and it was off on the box that actually was using all the cores. I didn't get a chance to check the other box on how it behaved if I turned it on there too. |
| 17:05.37 | Yourname` | russellb: You think that would have anything to do with it? |
| 17:05.40 | russellb | yeah, i would check that for sure. |
| 17:06.04 | russellb | it could be that the one core is doing all of the network packet handling |
| 17:06.58 | Yourname` | Ah.. |
| 17:07.17 | Yourname` | Is there any way I could make it uniform all over instead of letting just one core handling network packets? |
| 17:10.21 | russellb | irqbalance is what i was referring to :) |
| 17:10.41 | Yourname` | lol ok |
| 17:10.47 | Yourname` | So I'm going to disable the sucker! :) |
| 17:11.19 | Yourname` | All I'm trying to do is make asterisk to makeuseof all the cores in stores. |
| 17:13.21 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 17:14.15 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 17:18.13 | Yourname` | Does asterisk create the init script when make samples is done or something? |
| 17:18.24 | russellb | "make config" |
| 17:18.29 | russellb | as non-obvious as that sounds. |
| 17:18.56 | Yourname` | Weird, I did that. Hmm.. |
| 17:20.36 | Yourname` | Ok, I probably did something wrong because I just ran make config, and it worked; thanks russellb |
| 17:20.59 | russellb | no problem |
| 17:21.46 | rickross | russellb: found part of the CDR problem - apparently my mysql.sock file wasn't where Asterisk expected it. This is curious because it has run successfully for many moons with the exact same cdr_mysql.conf file under 1.4.x |
| 17:22.15 | rickross | if I explicitly declare the location of the sock file, then my CDR resumes |
| 17:22.19 | russellb | weird |
| 17:22.22 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 17:23.39 | Olobola | so sip users when using realtime doesn't require a 'switch' statement in sip.conf? |
| 17:23.43 | *** join/#asterisk cardiff (n=cardiff@76-10-147-182.dsl.teksavvy.com) |
| 17:24.06 | rickross | also, the "calldate" is not getting set - it is defaulting to "0000-00-00 00:00:00" |
| 17:24.10 | russellb | nope ... a switch is a dialplan concept |
| 17:24.21 | Olobola | ok, thanks |
| 17:24.59 | russellb | rickross: hm, both 1.4 and 1.6 used /tmp/mysql.sock as the default, so i don't know ... |
| 17:25.42 | rickross | russellb: mine had somehow moved |
| 17:25.44 | rickross | [root@rs1 asterisk]# locate mysql | grep sock |
| 17:26.09 | rickross | ugh - /var/lib/mysql/mysql.sock |
| 17:27.08 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:34.52 | *** join/#asterisk Bryanstein (n=Bryanste@unaffiliated/geekjuice) |
| 17:42.34 | *** join/#asterisk WindBack (n=jorge@host126.190-136-239.telecom.net.ar) |
| 17:47.27 | WindBack | hello |
| 17:54.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:58.07 | WindBack | I'm doing a basic billing: http://www.pastebin.ca/945074 I want that the DeadAGI script only executes if the channel was answered. But my problem is that: When I use a Zap channel the Dialstatus variable is always set to ANSWER even if the phone in the PSTN is not answered. Are there any way to know if the remote phone in the PSTN is answed?? |
| 17:58.57 | WindBack | Because this part of the Dialplan work perferct if I call a SIP channel |
| 17:59.34 | file | let me guess... you are dialing out using an analog phone line? |
| 18:01.25 | file | if so then you can not get accurate info such as that... they are considered answered as soon as dialed |
| 18:02.24 | WindBack | file, yes, I'm dialing out using analog phone |
| 18:02.58 | WindBack | file, Are not there any way to know?? |
| 18:03.18 | file | just the way it is. |
| 18:03.25 | WindBack | file, perhaps with another variable, I don't know?? |
| 18:03.34 | file | no, there is no variable... |
| 18:03.46 | file | chan_zap considers an analog line answered as soon as it is dialed |
| 18:05.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:05.07 | WindBack | file, Ok thank you |
| 18:06.40 | WindBack | file, So.. Does not exist any way to know when the remote phone in the PSTN is answered |
| 18:07.18 | file | not reliably. |
| 18:08.08 | *** join/#asterisk svenna_ (n=svenna@p548D2EC6.dip0.t-ipconnect.de) |
| 18:13.48 | *** join/#asterisk hohum_ (n=dcorbe@wsip-70-183-104-58.sd.sd.cox.net) |
| 18:20.03 | DarKnesS_WolF | my asterisk rulez ! |
| 18:20.11 | DarKnesS_WolF | + snom phone for sure :P |
| 18:21.29 | Yourname` | DarKnesS_WolF: Are you from Pakistan? |
| 18:22.31 | DarKnesS_WolF | Yourname`: nop egypt |
| 18:22.46 | Yourname` | ok |
| 18:22.50 | DarKnesS_WolF | Yourname`: why ? |
| 18:25.00 | Yourname` | Just wondering. |
| 18:25.35 | JayTee52 | Yourname`, are you from Pakistan? |
| 18:26.34 | JayTee52 | <PROTECTED> |
| 18:26.48 | Yourname` | JayTee52: No. |
| 18:26.53 | JayTee52 | just wondering :-) |
| 18:27.17 | *** join/#asterisk _Mongos_ (n=Mongos@mongos.devnull.ru) |
| 18:27.38 | JayTee52 | DarKnesS_WolF, so you like the snom phones? I've been thinking of giving them a try. The Grandstreams I've been using are kinda sucky. |
| 18:29.03 | plik | lol! http://www.sfw-porn.com/ |
| 18:29.55 | JayTee52 | Authorization Required |
| 18:29.55 | JayTee52 | This server could not verify that you are authorized to access the document requested. Either you supplied the wrong credentials (e.g., bad password), or your browser doesn't understand how to supply the credentials required. |
| 18:32.48 | JayTee52 | weird, but finally got the site to load |
| 18:33.06 | DarKnesS_WolF | JayTee52: yes go for snom tehy are amazign ! grandstreams sucks ! and giptel too sucks |
| 18:33.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:34.12 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 18:40.26 | rickross | anyone here using MySQL for CDR with * 1.6.0 betas? |
| 18:42.43 | *** join/#asterisk hohum (n=dcorbe@wsip-70-183-104-58.sd.sd.cox.net) |
| 18:43.31 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 18:44.23 | DarKnesS_WolF | rickross: what seems to be the problem ? |
| 18:44.27 | DarKnesS_WolF | not me but i think i might help |
| 18:45.46 | rickross | DarknesS_WolF: thanks, I have tried several different configs for the new [Aliases] section in cdr_mysql.conf, but I simply cannot get callerid recorded |
| 18:46.08 | rickross | the other columns are apprently recording correctly, but clid remains blank |
| 18:47.34 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:48.48 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:53.50 | Olobola | I'm trying to debug realtime sip, how can I view sql statement that are sent to mysql from asterisk? |
| 18:54.42 | *** join/#asterisk iPod-nano (n=david@c-68-43-60-193.hsd1.mi.comcast.net) |
| 18:55.04 | iPod-nano | I installed Asterisk via yum on Fedora. |
| 18:55.09 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:55.19 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 18:55.29 | iPod-nano | And now most of the functions don't seem to work. |
| 18:55.52 | iPod-nano | I just switched over from Ubuntu, and before that I used regular Debian. |
| 18:56.06 | iPod-nano | I kept copies of my dialplan and sip.conf. |
| 18:56.26 | iPod-nano | And just transferred them, but now most of my dialplan isn't working. |
| 18:57.17 | drmessano | Same version of Asterisk? |
| 18:57.44 | *** join/#asterisk nirz (n=nir@89-139-55-238.bb.netvision.net.il) |
| 18:57.51 | iPod-nano | I doubt it. |
| 18:58.09 | iPod-nano | I don't recall what I was running under Ubuntu. |
| 18:58.28 | iPod-nano | I'm running 1.4.17 now. |
| 18:59.06 | drmessano | Then you need to make appropriate changes |
| 18:59.15 | iPod-nano | Such as? |
| 18:59.43 | drmessano | Well, if you had 1.2 before, you need to see what changes there are between any functions you used with 1.2 and 1.4 |
| 18:59.47 | tzafrir | iPod-nano, give an example of something that doesn't work |
| 19:00.10 | iPod-nano | tzafrir, MP3Player isn't working. |
| 19:00.29 | iPod-nano | [Mar 16 14:42:04] WARNING[4778]: pbx.c:1821 pbx_extension_helper: No application 'MP3Player' for extension (default, 442, 2) |
| 19:00.38 | tzafrir | You probably need format_mp3 from asterisk-addons |
| 19:00.47 | tzafrir | do you have the module format_mp3 ? |
| 19:00.56 | iPod-nano | How do I check? |
| 19:00.57 | tzafrir | Or maybe Fedora removed it ? |
| 19:01.04 | tzafrir | show modules like mp3 |
| 19:01.22 | tzafrir | What do you need MP#Player for, btw? |
| 19:01.46 | iPod-nano | Trying to read a live stream. |
| 19:04.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:04.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:04.59 | iPod-nano | Why would such things be present in Ubuntu (and Debian) but not Fedora? |
| 19:05.05 | Yourname` | What ports need to be opened so clients behind NAT can register with Asterisk? |
| 19:06.45 | drmessano | On the client end or the Asterisk end? |
| 19:06.52 | Yourname` | Asterisk's end |
| 19:07.07 | Yourname` | Client is 10000-20000 commonly, right? |
| 19:07.23 | drmessano | 5060 UDP for SIP signalling and those defined in RTP.conf UDP |
| 19:07.48 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 19:08.25 | Yourname` | Hmm, weird. I used iptables -A INPUT -p tcp --dport 5060 -j ACCEPT and iptables --list doesn't show it, and so no one could connect. So I flushed, and evryone could connect.. even after a iptables restart I don't see 5060 anywhere in there, unless I missed something hardcore like saving? |
| 19:08.52 | drmessano | Not tcp |
| 19:08.53 | drmessano | udp |
| 19:09.05 | Yourname` | Ugh! |
| 19:09.07 | drmessano | and udp for the rtp ports too |
| 19:09.34 | drmessano | I won't "ok" 10000-20000 without you checking your rtp.conf to confirm thats what is set |
| 19:09.39 | Yourname` | But why doesn't it show though? iptables --list? |
| 19:09.39 | drmessano | But commonly, yes |
| 19:10.09 | drmessano | I dunno |
| 19:10.39 | outtolunc | has a frozen (most likely red) tongue <G> |
| 19:10.54 | outtolunc | damn that was good |
| 19:13.34 | Yourname` | Thanks, drmessano |
| 19:19.48 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 19:20.49 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:20.49 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:21.15 | outtolunc | Yourname`: you sure it the 10000 isn't there listed as ndmp |
| 19:21.25 | outtolunc | do a iptables --list -v |
| 19:22.04 | Yourname` | yh |
| 19:22.48 | outtolunc | since you /etc/services has a listing for 10000 (probably ndmp) it is just showing it.. not a range |
| 19:22.59 | Yourname` | Btw, in the CLI, sip show peers has rows Dyn/Nat/ACL/Port.. and some ports I see are over and above the 10000-20000, what does that mean? |
| 19:23.46 | lmadsen | Yourname`: those are the source ports the SIP channel is registered from |
| 19:24.09 | Yourname` | lmadsen: Ah, and so iptables allows all outgoing connections, that should be fine, right? |
| 19:24.54 | lmadsen | Yourname`: ya, it should depending on how you set it u |
| 19:24.56 | lmadsen | up* |
| 19:25.19 | Yourname` | Ok, cool. I was wondering why it's letting connections from those ports! heh |
| 19:35.15 | *** join/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
| 19:35.29 | a1fa | i come to bitch about pap2-na |
| 19:35.40 | a1fa | what a POS!! :( |
| 19:36.00 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 19:36.18 | _-Jon-_ | Hello all |
| 19:36.18 | a1fa | Hello Jon |
| 19:38.23 | _-Jon-_ | I'm having some problems after an upgrade to 1.4.18 (from 1.2). I have 2 providers (link2voip and Broadvoice), but I just realized Broadvoice isn't work. The only thing I can find is when I turn debugging on, I get this: SIP/2.0 401 Unauthorized |
| 19:41.48 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 19:54.25 | _-Jon-_ | Can anyone try to assist me with why it's saying incoming calls are not authorized? |
| 19:54.33 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:55.10 | _-Jon-_ | Also in the debug, I'm getting No user '<my cell #>' in SIP users list.. Which doesn't seem right |
| 20:02.42 | a1fa | because you did not register to the sip provider |
| 20:02.56 | a1fa | did u follow the config for broadvoice? |
| 20:03.00 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 20:09.51 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 20:14.26 | _-Jon-_ | Yes, exact same config I had before, and tried using their sample config |
| 20:15.15 | a1fa | its obviously not registering |
| 20:15.53 | _-Jon-_ | sip show registry shows registration.. |
| 20:16.28 | _-Jon-_ | and debug shows information when the call trys to come in |
| 20:17.35 | fujin | Is there anyway to show all ${} variables set on an incoming channel? |
| 20:17.48 | fujin | trying to hack w/ QM to get it to work with AQM/RQM.. Got it working for pausing. |
| 20:24.51 | *** join/#asterisk hohum_ (n=dcorbe@wsip-70-183-104-58.sd.sd.cox.net) |
| 20:25.46 | *** join/#asterisk b11d (n=no@71-221-45-36.dlth.qwest.net) |
| 20:25.49 | b11d | hello all |
| 20:29.09 | *** join/#asterisk bl4q (n=Bl@dslb-088-064-131-051.pools.arcor-ip.net) |
| 20:37.51 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 20:47.16 | *** part/#asterisk cr0n (n=adfadf@dsl-240-27-108.telkomadsl.co.za) |
| 20:56.36 | *** join/#asterisk shido6 (n=shido6@74-130-50-233.dhcp.insightbb.com) |
| 21:01.47 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 21:02.38 | axisys | so i want to purchase a small linux box and install asterisk on it? looking at linux devices.. so many choices.. anything u guys can recommend? |
| 21:02.48 | fujin | how small? |
| 21:02.51 | fujin | how many users? |
| 21:02.55 | mvanbaak | ask mcgyver |
| 21:03.04 | fujin | if it's just for your house, buy a wrt54g or something, put openwrt on it |
| 21:03.17 | mvanbaak | or get one of them soekris boxen |
| 21:03.31 | axisys | fujin: i already have a wrt54gl with kamikaze 7.07 |
| 21:03.47 | axisys | fujin: just for myself |
| 21:03.54 | axisys | fujin: home use |
| 21:03.55 | fujin | that's nice buddy, real nice |
| 21:03.56 | fujin | cares |
| 21:04.17 | axisys | fujin: ;-) |
| 21:04.39 | axisys | fujin: so i could just install asterisk on it? do i need fxo/fpo ?? |
| 21:05.54 | axisys | i see soo many suggestions in /topic |
| 21:10.16 | mvanbaak | axisys: if you want to connect to a POTS line or an analog phone you need fxo/fxp |
| 21:10.30 | mvanbaak | otherwise no need for it |
| 21:10.51 | axisys | mvanbaak: right now I have only vonage.. no regular phone line |
| 21:10.52 | mvanbaak | I have dozens and dozens of setups that dont have fxo/fxp nor pri |
| 21:11.09 | mvanbaak | most of the stuff we do is pure voip |
| 21:11.50 | axisys | mvanbaak: so I will need a voip number may be from broadvoice who allow to use my own voip device..correct? |
| 21:12.06 | axisys | mvanbaak: i want to get rid of vonage |
| 21:12.21 | mvanbaak | yeah |
| 21:13.21 | axisys | mvanbaak: and I can just get a device install linux and asterisk and get a cheap fxo/fxp card for my computer? |
| 21:13.29 | axisys | mvanbaak: am i on right track so far? |
| 21:13.47 | mvanbaak | axisys: if you go for a voip number you dont need fxo/fxp |
| 21:14.00 | mvanbaak | you only need that to connect to a landline or analog phone |
| 21:14.09 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 21:14.33 | axisys | mvanbaak: hmm.. I guess i need to understand the difference |
| 21:14.44 | mvanbaak | ~fxo |
| 21:14.44 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 21:14.52 | fujin | one implies that you need a phone line |
| 21:14.54 | fujin | one is pure voip |
| 21:15.06 | fujin | opts for pure VoIP (be it IAX2, or SIP) in most occasions |
| 21:15.23 | mvanbaak | same here |
| 21:15.25 | axisys | fujin: thats what I want too.. pure voip |
| 21:15.28 | mvanbaak | ~fxp |
| 21:15.28 | jbot | rumour has it, fxp is a program that uses FTP PASV mode to transfer between two FTP servers, ask about 'quftp' for software |
| 21:15.42 | fujin | If you've got the bandwidth; VoIP is fine |
| 21:15.55 | fujin | if you don't, use a fxo/fxs card (or a E1/T1 PRI if you can afford it..) |
| 21:16.00 | axisys | fujin: i have a cable modem with very good connection |
| 21:16.14 | mvanbaak | axisys: then go for a pure voip setup |
| 21:16.26 | mvanbaak | specially since it's for your home use |
| 21:16.44 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 21:16.46 | axisys | mvanbaak: would be nice for newbie like me to read a how-to on this.. including what parts I need |
| 21:17.00 | axisys | mvanbaak: i know systm site had one presentation.. can find it now |
| 21:17.54 | axisys | found it .. http://revision3.com/systm/asterisk/ |
| 21:18.14 | axisys | mvanbaak: any other article for newbies would be really appreciated |
| 21:18.39 | mvanbaak | axisys: |
| 21:18.44 | mvanbaak | ~book |
| 21:18.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:18.54 | mvanbaak | ^^^^^^^^ - read that |
| 21:18.58 | axisys | mvanbaak: doh! i just got the book |
| 21:19.02 | axisys | mvanbaak: hehe |
| 21:19.18 | mvanbaak | well, read it |
| 21:19.29 | axisys | mvanbaak: let me find what I need in that book as well ;-) |
| 21:19.58 | mvanbaak | read it front to back and back |
| 21:20.15 | axisys | looking thru the chapters to find what is needed |
| 21:20.21 | axisys | mvanbaak: haha |
| 21:20.38 | mvanbaak | remember it, and live by it |
| 21:20.46 | mvanbaak | it's better then the bible |
| 21:20.54 | axisys | mvanbaak: lol |
| 21:21.07 | axisys | mvanbaak: i am not into fictions :P |
| 21:21.13 | mvanbaak | lol |
| 21:21.26 | mvanbaak | that's why The Book is better |
| 21:21.29 | mvanbaak | it's real |
| 21:26.51 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 21:27.14 | axisys | ~fsx |
| 21:27.18 | axisys | ~fxs |
| 21:27.19 | jbot | from memory, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 21:27.46 | axisys | i like this channel ..almost like m-w for voip |
| 21:28.03 | *** join/#asterisk orakle (i=orakle@siteops.org) |
| 21:30.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:31.00 | axisys | what is a good fxo/fxs card or device? |
| 21:31.06 | mvanbaak | TDM |
| 21:36.46 | *** join/#asterisk WindBack (n=jorge@host69.200-117-118.telecom.net.ar) |
| 21:40.42 | defsdoor | axisys: sangoma remora |
| 21:41.53 | axisys | defsdoor: thnx |
| 21:42.08 | axisys | this presentation is very very cool http://revision3.com/systm/asterisk/ |
| 22:01.39 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.125.138) |
| 22:06.12 | Bryanstein | hey guys...I am having troubles with the old pingtel xpressa and registering with asterisk |
| 22:06.33 | Bryanstein | the user I created can reg fine with ekiga but not with my phone |
| 22:07.37 | Rico29 | maybe you can update the firmware ? |
| 22:07.42 | Rico29 | I don't know |
| 22:08.17 | Bryanstein | Rico it did that automatically when I hooked it up |
| 22:08.25 | Bryanstein | they were brand new |
| 22:08.32 | Rico29 | ah |
| 22:08.42 | Rico29 | what does the sip debug mode show ? |
| 22:08.44 | Bryanstein | it seems the phone jumps out and grabs that VxWorks os upgrade |
| 22:09.14 | Bryanstein | no matching peer |
| 22:09.40 | *** join/#asterisk IOscanner (n=IOscanne@76.187.195.124) |
| 22:09.41 | Rico29 | and with wireshark ? |
| 22:09.43 | Greek-Boy | anyone here using RSTP on their switches for redundancy? |
| 22:09.52 | Rico29 | ,op |
| 22:10.19 | Bryanstein | oh I am not capturing sip packets now Rico29 |
| 22:11.15 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-143-10.red.bezeqint.net) |
| 22:11.29 | Bryanstein | let me try a few more things then I'll report again |
| 22:11.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:11.59 | Dovid | anyone here try using OpenSBC with asterisk ? |
| 22:14.14 | Rico29 | what is openSBC ? |
| 22:14.22 | Rico29 | session border controller ? |
| 22:16.00 | Dovid | Rico29: Yes |
| 22:16.22 | Dovid | also anyone know the paramater for sip.conf so that the RTP goes form one endpoint to another right away ? |
| 22:17.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:18.30 | Rico29 | pbx.c:1821 pbx_extension_helper: No application '' for extension ( |
| 22:18.35 | Rico29 | what's the problem doc ? |
| 22:19.43 | Rico29 | it's ok |
| 22:19.48 | Rico29 | I'm just stupid :) |
| 22:20.13 | Dovid | Rico29: Please PB realavent dialplan |
| 22:20.15 | Dovid | ~pb |
| 22:20.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:20.20 | Dovid | prob. a typo ;) |
| 22:35.15 | JayTee52 | what if I flooded the channel with candy or beer? |
| 22:35.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:42.09 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 22:57.45 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 22:59.30 | hmmhesays | that would be wonderful |
| 23:05.08 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:05.49 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:13.34 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-183-106.dsl.tul2ok.sbcglobal.net) |
| 23:36.03 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 23:39.14 | *** join/#asterisk EvilDeshi (n=Skunk@75-135-93-93.dhcp.mdsn.wi.charter.com) |
| 23:39.21 | EvilDeshi | anyone know where i can download app_flite? |
| 23:40.18 | fujin | The Internet. |
| 23:41.53 | EvilDeshi | was hoping for a more specific location, I seem to only be able to find an rpm package for it and I am using debian |
| 23:42.01 | fujin | Google. |
| 23:42.06 | fujin | ~flite |
| 23:42.07 | jbot | hmm... flite is http://www.cmuflite.org/ or at http://www.handhelds.org/z/wiki/flite or lightweight speech synthesis that works on the ipaq |
| 23:42.07 | fujin | ~app_flite |
| 23:43.53 | EvilDeshi | thanks |
| 23:44.53 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 23:49.16 | *** join/#asterisk R-MAN (n=no@client-86-27-167-153.popl.adsl.virgin.net) |
| 23:49.26 | R-MAN | hey guys |
| 23:49.34 | R-MAN | wow this is my first time in the Asterisk channel |
| 23:51.22 | R-MAN | anyone around to help me real quick with asterisk 1.4 and no audio/nat problems? |
| 23:51.47 | drmessano | ~nat |
| 23:51.48 | jbot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 23:51.51 | drmessano | ~sipnat |
| 23:51.52 | 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 |
| 23:52.19 | R-MAN | ahh yes your correct there sir however ive noticed that asterisk 1.4 is completly different |
| 23:52.31 | drmessano | No it's not |
| 23:52.33 | R-MAN | for example no editing in sip.conf anyone...its really confused me |
| 23:52.40 | drmessano | Uh |
| 23:52.49 | drmessano | You must be using FreePBX then |
| 23:52.54 | R-MAN | yeah now youve got sip_general etc etc |
| 23:53.12 | R-MAN | damn..I got busted lol |
| 23:53.16 | R-MAN | sorry |
| 23:53.18 | R-MAN | yes I am |
| 23:53.19 | drmessano | Yeah, that's nothing to do with whether or not it's Asterisk 1.4 or 1.2 |
| 23:53.29 | R-MAN | oh right..you learn something new every day |
| 23:53.37 | R-MAN | so can we then assume that trixbox is POO then? |
| 23:53.40 | R-MAN | lol |
| 23:53.53 | drmessano | No |
| 23:54.00 | drmessano | Just need to edit a different file |
| 23:54.09 | drmessano | Try sip_nat.conf |
| 23:54.15 | R-MAN | ive been running 1.2 for a long time but then the hdd died so I decided to try 1.4 |
| 23:54.21 | drmessano | Thats the only difference |
| 23:54.23 | fujin | R-MAN: take it to #freepbx/trixbox/whatever |
| 23:54.32 | R-MAN | yes already created that and put the proper config in it |
| 23:54.46 | R-MAN | I even put config in the hosts file too since im testing outside my network |
| 23:55.00 | R-MAN | sorry Fujin |
| 23:55.05 | drmessano | You shouldn't be touching your hosts file |
| 23:55.28 | R-MAN | dont worry I didnt take the 127.0.0.1 |
| 23:55.32 | R-MAN | otu |
| 23:55.51 | drmessano | Pastebin your nat config pls |
| 23:55.55 | R-MAN | of what? |
| 23:56.09 | drmessano | for asterisk |
| 23:56.24 | R-MAN | sip_nat and hosts file? |
| 23:56.35 | drmessano | <R-MAN> yes already created that and put the proper config in it <---- |
| 23:56.38 | drmessano | I want to see it |
| 23:56.46 | R-MAN | roger that stand by |