| 00:00.06 | [TK]D-Fender | bran: Updating your kernel <- |
| 00:00.12 | [TK]D-Fender | bran: Just like it sounded like |
| 00:00.16 | bran | i didn't really update my kernel |
| 00:00.22 | KavanS | interested to transfer *directly* to voicemail - currenlty use ** to initiate transfer - any ideas? |
| 00:00.24 | bran | im using whatever kernel that came with AsteriskNOW 1.7 |
| 00:02.38 | brycebaril | [TK]D-Fender: Right, I see the D() option, which look like it might work, but I wanted to hear someone confirm that was correct before I asked the second part of my question |
| 00:02.57 | [TK]D-Fender | bran: recompile wanpipe then |
| 00:03.22 | bran | [TK]D-Fender: i'm recompiling thru ./Setup install |
| 00:03.34 | bran | and it always fails @ the utility compile part |
| 00:03.35 | [TK]D-Fender | brycebaril: That should work... dependent on propre call-progress monitoring. |
| 00:03.51 | [TK]D-Fender | bran: Then clearly you should look at the compile failure |
| 00:04.23 | brycebaril | which is is it possible to dial a different extension after bridging for each resource on a dialstring, i.e. extension 111 for 5551212 or extension 22 for 5553434 |
| 00:06.15 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 00:06.29 | [TK]D-Fender | brycebaril: Please rephrase that...it came out a mess |
| 00:07.31 | brycebaril | I am dialing 2+ numbers simultaneously using Dial() in my example, 5551212 and 5553434. Each of them has different extensions I would like to dial depending on which number answers |
| 00:07.46 | *** join/#asterisk blaines (~blaines@71-223-168-253.phnx.qwest.net) |
| 00:09.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:10.26 | pabelanger | brycebaril: senddtmf |
| 00:10.46 | pabelanger | brycebaril: after your channel gets answered |
| 00:11.38 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:12.04 | brycebaril | which won't work since the application is inside a FastAGI app that called Dial() and Dial() is blocking? Right? (I understand that isn't best practice, I didn't write this) |
| 00:12.29 | brycebaril | Or can I use that via a macro? |
| 00:13.16 | pabelanger | brycebaril: No, once you dial your number, your context we move to the next priority. Simply call SendDTMF |
| 00:13.56 | pabelanger | brycebaril: DIALSTATUS variable |
| 00:14.34 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 00:15.32 | [TK]D-Fender | brycebaril: then dial 2 local channels each with the added DTMF to add. |
| 00:18.21 | brycebaril | So from what I understand I can set some channel variables that will indicate which extension to dial depending on which number answers, and then even though my AGI app is waiting for Dial() to return, I can put something in extensions.conf that can read those variables, know what number was connected and send the appropriate DTMF? |
| 00:19.21 | *** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 00:20.13 | [TK]D-Fender | brycebaril: My way you don't HAVE to know anything. Or check anything |
| 00:20.25 | [TK]D-Fender | BryAnd the calls are simultaneous, not sequential. |
| 00:20.58 | brycebaril | ok, what do you mean by dial two local channels then? |
| 00:25.55 | [TK]D-Fender | brycebaril: chan_local |
| 00:26.08 | [TK]D-Fender | brycebaril: Go read up on your Asterisk channel types |
| 00:36.10 | brycebaril | Thanks |
| 00:37.59 | *** join/#asterisk jhirley (~jhirley@c-98-211-237-248.hsd1.fl.comcast.net) |
| 00:44.50 | *** join/#asterisk evoltech (~evoltech@c-24-21-203-123.hsd1.or.comcast.net) |
| 00:45.18 | evoltech | Is there a way to do chat over sip with an asterisk server? |
| 00:45.56 | *** join/#asterisk sam555 (~chatzilla@c-24-21-203-123.hsd1.or.comcast.net) |
| 00:48.38 | Get_The_Fish | hey all... I'm a little confused on DAHDI on CentOS/RHEL- I shouldn't see a DAHDI process running, but I should see a kernel module (even if I am using DAHDI dummy). Is dahdi_test and timing test in asterisk the only way to see if it's "working"? |
| 00:49.23 | sam555 | is there a separate chat for asterisk now? |
| 00:50.21 | SaiSoma|AtHome | hey guys, trying to work with waitforsilence and getting what i consider unexpected results (never finishes waiting, even if i specify a timeout). here's an example without the timeout set: |
| 00:50.23 | SaiSoma|AtHome | http://pastebin.com/Tc5H1KAB |
| 00:53.54 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 00:58.35 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:58.54 | *** part/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 01:03.13 | *** join/#asterisk valajbeg (~hamo@b199c78.pptp-gw50.cable-internet.GlobalNET.ba) |
| 01:05.24 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 01:06.20 | *** join/#asterisk valajbeg (~hamo@b199c78.pptp-gw50.cable-internet.GlobalNET.ba) |
| 01:20.58 | neurosys | has anyone heard of a call being disconnected after 20 seconds of receiving it from a parked ext? |
| 01:21.01 | *** join/#asterisk b14ck (~b14ck@s66-76-50-56.lfkncmta01.lfkntx.tl.sta.suddenlink.net) |
| 01:35.07 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 01:49.50 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 01:50.34 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:10.29 | *** part/#asterisk evoltech (~evoltech@c-24-21-203-123.hsd1.or.comcast.net) |
| 02:16.17 | k-man | is a cisco IP phone model 7940 any good? |
| 02:20.23 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:21.58 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 02:22.33 | *** join/#asterisk fred1_ (~djc@65.209.147.101) |
| 02:22.36 | fenrus | i like it |
| 02:22.46 | fred1_ | <PROTECTED> |
| 02:23.01 | fred1_ | or say if I had a set of lines, and I want ring-over (eg ring first available line) |
| 02:25.02 | p3nguin | fred1_: Are you talking about a single extension ringing more than one phone? |
| 02:25.29 | fred1_ | no i only want it to ring the one first available line |
| 02:25.47 | fred1_ | like say I had a dozen phone lines as a modem pool from a telco |
| 02:25.52 | fred1_ | but only one phone # |
| 02:26.14 | ChannelZ | you could use a queue but it might be overkill |
| 02:26.14 | fred1_ | they might implement that as 'forward on busy' with each line forwarding to the next |
| 02:26.34 | fred1_ | if I just use two Dial() commands with sequenced priority, will that do what I want? |
| 02:26.53 | fred1_ | eg exten => xxx,1Dial(IAX2/peerA) |
| 02:26.58 | p3nguin | You have to use a single extension, which you said NO to already. |
| 02:27.02 | fred1_ | exten => xxx,2Dial(IAX2/peerB) |
| 02:27.02 | ChannelZ | it could, if the device doesn't implement any sort of 'call waiting' causing a second call to it to ring |
| 02:27.24 | fred1_ | this is for an incoming call. I have two iaxmodem devices setup |
| 02:27.26 | p3nguin | And so I will leave you to your own devices. |
| 02:27.34 | fred1_ | if the first is busy, I want it to ring to the second |
| 02:27.57 | fred1_ | er, missed a "," between the priority and the Dial, there |
| 02:28.02 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 02:28.11 | ChannelZ | re: dialing in sequence as you say will work, so long as peerA doesn't do call waiting and accept a second call as I said |
| 02:28.37 | fred1_ | hrm. well the peer is iaxmodem. How do I make sure it doesnt do call waiting? |
| 02:29.02 | fred1_ | or does it even support call waiting |
| 02:29.24 | fred1_ | wouldnt seem to serve much purpose for a virtual modem.. even V.92 would be pointless i thin |
| 02:29.26 | fred1_ | k |
| 02:29.51 | ChannelZ | I have no idea, this is the first time you mentioned anything about an iaxmodem |
| 02:30.27 | fred1_ | ok |
| 02:30.28 | fred1_ | thanks |
| 02:30.53 | ChannelZ | IAX2/peerA and IAX2/peerB are iaxmodems? |
| 02:31.34 | fred1_ | yes |
| 02:31.58 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 02:32.30 | ChannelZ | well my guess is they don't support CW and will just return a BUSY or CHANUNAVAIL or something |
| 02:32.30 | fred1_ | i probably used the wrong terms.. one 'extension'. but two peers. want to ring first available.. ala 'hunt group' |
| 02:32.45 | fred1_ | Seems like a reasonable guess.. I would hope, anyway |
| 02:32.56 | ChannelZ | no I get that, I just don't understand what the iaxmodems are doing.. these virtual fax or something? |
| 02:33.01 | fred1_ | yeah |
| 02:33.36 | fred1_ | pretty low traffic, low likelyhood of even two coming at the same time.. but I figure just in case |
| 02:33.57 | fred1_ | and I just wanted to prove I could do it |
| 02:34.28 | fred1_ | next I'll be running DOSEMU, setting up a whole bank of a dozen of them, and running a DOS based BBS on it! :P |
| 02:34.47 | fred1_ | virtual BBS |
| 02:34.49 | fred1_ | heh |
| 02:35.39 | fred1_ | anyway.. im also setting up to use for outbound, and I wanted to keep a free channel for inbound while a send was in progress |
| 02:36.24 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:36.51 | fred1_ | i'll just go with hoping/assuming that IM doesnt do CW |
| 02:37.01 | fred1_ | maybe I'll test that out tomorrow |
| 02:45.11 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 02:48.12 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 02:54.25 | *** join/#asterisk op3r (~op3r@182.18.192.231) |
| 02:55.50 | *** join/#asterisk resno (~bryan@cpe-098-026-005-130.nc.res.rr.com) |
| 03:04.41 | resno | i was reading about using google voice to get a free homephone. is the linksys spa-3102 the "best" ata to use? |
| 03:05.00 | fred1_ | GV doesnt offer SIP |
| 03:05.03 | fred1_ | fyi |
| 03:05.33 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 03:06.02 | resno | fred1_: using freepbx and ipkall i believe |
| 03:06.04 | fred1_ | theres some limited interaction with gizmo, but its awkward to setup |
| 03:06.12 | fred1_ | believe what? |
| 03:06.14 | fred1_ | GV doesnt offer SIP |
| 03:06.34 | resno | fred1_: this what i was reading : http://www.legitreviews.com/article/1058/1/ |
| 03:06.35 | fred1_ | oh ok.. so you plan on getting the phone service from ipkall |
| 03:06.41 | fred1_ | then getting a local number from GV |
| 03:06.44 | resno | correct |
| 03:06.46 | op3r | hello is it possible to just overwrite zaptel 1.4.21 to dahdi? |
| 03:07.01 | fred1_ | yeah I suppose that'd work |
| 03:07.15 | resno | i know its quite covoluted |
| 03:07.21 | resno | convoluted |
| 03:07.30 | fred1_ | oh I know all about convoluted |
| 03:07.35 | *** join/#asterisk denon (denon@sassinak.net) |
| 03:07.35 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:07.55 | fred1_ | as to your original Q, I have no advice as to ATA's |
| 03:08.00 | resno | but my homephone will spend more time in silence then actually doing anything |
| 03:08.13 | fred1_ | I have a spa-2000.. it works, more or less |
| 03:08.34 | fred1_ | as long as my wifi uplink stays stable, anyway |
| 03:08.56 | resno | is there any ata i should stay away from? |
| 03:09.06 | resno | i see several ones on ebay. |
| 03:09.10 | resno | that you know of |
| 03:09.15 | fred1_ | Ive no experience with anything else |
| 03:09.34 | resno | i see. thanks fred1_ |
| 03:09.34 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:09.36 | fred1_ | well, i fiddled a bit with siphon on my iphone |
| 03:09.45 | fred1_ | but that doesnt really help you, I dont think |
| 03:09.51 | resno | not at this point. |
| 03:10.43 | resno | i can test the setup without with the ata right? |
| 03:10.50 | resno | using a softphone |
| 03:11.09 | fred1_ | should be able to |
| 03:11.24 | fred1_ | assuming whatever softphone you have works with your setup |
| 03:11.40 | resno | indeed. |
| 03:12.21 | fred1_ | if ipkall offers SIP service, you may not need freepbx tho |
| 03:12.36 | fred1_ | just set the softphone to talk right to ipkall.. |
| 03:12.46 | fred1_ | unless you have some unusual requirement that wont suffice for |
| 03:12.51 | fred1_ | or you just want to play with it, of course :P |
| 03:14.34 | resno | ive been doing a lot of research so the services are running together at the moment |
| 03:16.52 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 03:21.07 | resno | fred1_: oh, this guide is using sipgate |
| 03:22.54 | *** join/#asterisk ariel_ (~chatzilla@173.151.173.110) |
| 03:28.26 | *** join/#asterisk BrendanMcc (3ce5fc51@gateway/web/freenode/ip.60.229.252.81) |
| 03:28.32 | *** part/#asterisk BrendanMcc (3ce5fc51@gateway/web/freenode/ip.60.229.252.81) |
| 03:28.34 | *** join/#asterisk BrendanMcc (3ce5fc51@gateway/web/freenode/ip.60.229.252.81) |
| 03:29.03 | BrendanMcc | hey guys - got centos 5.3 with our Asterisk box running Elastix for VoIP. I have set up port forwarding on our router and cant seem to get to the box externally on ports 22, 80, or 443. I have port forwarding succesfully working to other devices on the network.. I have firewall switched off after running 'setup' at the CLI and can successfully connect to those ports while on the local network... Have also edited /etc/hosts.allow |
| 03:30.32 | fred1_ | either your port forwarding isnt set right, or your ISP is blocking inbound on those ports |
| 03:30.52 | fred1_ | oh to other devices.. |
| 03:31.00 | fred1_ | uhm.. you can only forward a given port to ONE device |
| 03:32.50 | *** join/#asterisk sat-man (~jlupresto@74-81-241-158.static.sdyl005.digis.net) |
| 03:37.46 | BrendanMcc | yes i know, i tried 80 for one device (worked) then 80 for the centos asterisk box... doesn't work... |
| 03:39.32 | BrendanMcc | edited /etc/hosts.allow to be ALLOW=ALL... is there something I have missed |
| 03:39.32 | p3nguin | Use nonstandard ports and forward them (just for testing purposes). |
| 03:40.09 | p3nguin | I doubt the web server would be bound by tcp wrappers. |
| 03:40.37 | p3nguin | It's not uncommon for ISPs to block standard service ports. |
| 03:40.52 | BrendanMcc | ill ring bigpond now |
| 03:41.44 | p3nguin | If you can't connect to the web server on port 80, configure your router to forward port 81 to the server's port 80. Then test port 81 on the outside. |
| 03:42.27 | BrendanMcc | yes ok. |
| 03:42.28 | *** join/#asterisk blaines (~blaines@75-171-121-6.phnx.qwest.net) |
| 03:42.32 | p3nguin | Or any other unusual high port would work, too. |
| 03:42.48 | p3nguin | 8080 is a typical one to use for http. |
| 03:45.00 | BrendanMcc | its a crappy router... only has start and end ports |
| 03:46.54 | sat-man | can anyone give me a good tutorial link on how to convert .mp3 files into the right format for my moh folder? |
| 03:47.45 | p3nguin | Most of the time, there's a outside port and an inside port, so you can map ports rather than just pass them right through. |
| 03:47.53 | *** join/#asterisk simplydrew (~simplydre@pool-173-69-4-91.prvdri.fios.verizon.net) |
| 03:51.21 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 03:51.43 | *** join/#asterisk cecil_lincher (~clincher@pool-173-63-114-47.nwrknj.fios.verizon.net) |
| 03:52.48 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:56.14 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 04:15.27 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 04:18.18 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:29.25 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 04:39.06 | *** join/#asterisk zyphlar (~z@wsip-70-182-59-230.ph.ph.cox.net) |
| 04:41.19 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:50.25 | p3nguin | http://xkcd.com/600/ |
| 04:51.02 | zyphlar | hah. the latest one, dependency resolution |
| 04:51.17 | zyphlar | oh god it hurts |
| 04:52.13 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 04:59.41 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-mcfzpnqycofqzfci) |
| 05:02.47 | joako | How come if I visit asterisk.org downloads page with Internet Explorer I can't copy the URLs? |
| 05:03.12 | zyphlar | problem #1 is you're using IE ;) |
| 05:03.16 | zyphlar | lemme check |
| 05:04.35 | zyphlar | i guess IE doesn't like <a> elements being outside a <div> |
| 05:05.09 | zyphlar | this is one instance where saying "get a better browser" is actually good, non-snarky advice :) |
| 05:09.08 | ChannelZ | that doesn't make sense |
| 05:09.16 | zyphlar | sure it does |
| 05:09.24 | joako | zyphlar, Sorry I just assumed they were doing it intentionally... they used to do some nasty redirect stuff before on the downloads page |
| 05:09.36 | zyphlar | nice. maybe they are haha |
| 05:09.56 | zyphlar | <a><div>Hello!</div></a> should work though |
| 05:10.01 | zyphlar | just like <a><img></a> works |
| 05:10.18 | ChannelZ | oh you mean divs contained in an a |
| 05:10.34 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:10.47 | zyphlar | yup |
| 05:11.46 | ChannelZ | I thought you meant a's not inside divs :) I was going 'huh? when did that rule happen?' |
| 05:13.41 | zyphlar | awwwwww skee skee god damn! |
| 05:19.10 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 05:24.10 | op3r | how do you complete remove zaptel and move to dahdi? I keep on getting ztdummy: Unknown symbol zt_register |
| 05:39.11 | ChannelZ | I don't remember if zaptel has a 'make uninstall' |
| 05:39.38 | ChannelZ | otherwise really you just remove the kernel modules.. build dahdi, then reconfigure and rebuild asterisk |
| 05:40.27 | kaldemar | its makefile has "uninstall-modules" |
| 05:44.41 | ChannelZ | ta-daaa! |
| 05:45.05 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:46.33 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 05:47.10 | Martinblr | I have registerd a ATA with a Asterisk based box, when i try to call from ATA to the PBX I am getting 403 Forbidden message |
| 05:47.25 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.137.66.mtnl.net.in) |
| 05:48.34 | *** join/#asterisk AJ707 (AJ707@S0106001310779db4.vs.shawcable.net) |
| 05:54.17 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 05:57.48 | ChannelZ | you might have registered but it's not matching the peer/authing correctly on a call apparently |
| 05:59.01 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 06:03.17 | zyphlar | anyone know about ztdummy, the usb timing app? |
| 06:03.46 | zyphlar | i was having an IRQ conflict with my USB ports, so i disabled them, now it seems like i have an even worse timing issue |
| 06:04.46 | zyphlar | wondering if my digium te207p t1 card depends on usb for timing or if it truly provides its own |
| 06:06.18 | Martinblr | ChannelZ: the status is showing as registered |
| 06:06.40 | Martinblr | ChannelZ: the same thing if i tried in softphone it is working |
| 06:06.46 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 06:07.23 | ChannelZ | that only means it registered (it successfully told Asterisk its IP address) |
| 06:07.44 | ChannelZ | A call could be looking completely different to asterisk. sip set debug on and see whats going on. |
| 06:08.15 | *** join/#asterisk ming_zym (~ming_zym@123.118.80.147) |
| 06:08.58 | Martinblr | ChannelZ: In sip debug I can able to see as SIP/2.0 403 Forbidden after the SIP/2.0 100 Trying message |
| 06:10.13 | Martinblr | ChannelZ: I tried the type as peer & friend. No success |
| 06:10.15 | kaldemar | Martinblr: pastebin the sip debug for the whole call and someone will most likely tell you why it happens. |
| 06:10.42 | kaldemar | Martinblr: also enable verbosity in the output with "core set verbose 10" |
| 06:12.19 | Martinblr | My sip debug http://asterisk.pastebin.ca/1884114 |
| 06:14.19 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 06:14.34 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 06:14.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 06:16.31 | ChannelZ | Nov 30 11:11:46 WARNING[1048] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"Allo" <sip:2050@69.15.230.25:5065>;tag=as1e2809aa' |
| 06:21.43 | Martinblr | ChannelZ: yes that is the problem, but the same credentials are working in softphone |
| 06:23.50 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:24.43 | ChannelZ | well maybe that says more about the device that doesn't work |
| 06:24.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:25.26 | ChannelZ | I'm having trouble following what is going on in this dialog.. are there 2 devices besides Asterisk involved? (I'm seeing a User Agent of 'CEM ATA' and 'CudaTel' |
| 06:26.48 | Martinblr | ChannelZ: Asterisk is embedded in this device |
| 06:30.02 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 06:30.25 | *** join/#asterisk rare1980_ (~as@12.25.228.67) |
| 06:30.50 | rare1980_ | hi all on Intel Quad Core 2.33GHz system how many calls i can make at same time |
| 06:30.56 | rare1980_ | with good quality |
| 06:32.54 | ChannelZ | Martinblr: Not sure what to say, maybe the ATA is doing the MD5 hash incorrectly or something.. I'm not enough of a SIP expert to analyze this deeply |
| 06:33.37 | ChannelZ | rare1980_: totally depends on what those calls are. Analog to SIP? SIP to SIP? Analog to PRI? Any transcoding occurring? etc |
| 06:34.04 | ChannelZ | without specifics, the answer is "probably quite a lot" |
| 06:38.41 | *** join/#asterisk debuggerboy (~anish@202.83.41.95) |
| 06:38.47 | *** part/#asterisk debuggerboy (~anish@202.83.41.95) |
| 06:40.36 | kaldemar | rare1980_: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 06:41.44 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 06:42.08 | Martinblr | ChannelZ: Ok Thanks |
| 06:50.14 | rare1980_ | channelZ: those call would be from my call center to outside lines |
| 06:51.03 | Martinblr | Is there any converter tool from MP3 to GSM support in Asterisk...? |
| 06:51.29 | *** join/#asterisk BANSAL (~bansal@117.199.116.225) |
| 06:55.02 | kaldemar | Martinblr: sox is fine for that. |
| 06:59.59 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 07:00.06 | ChannelZ | rare1980_: that still doesn't say anything. What are the phones in your call center? What are the 'outside lines'? |
| 07:00.36 | ChannelZ | but alas, bed time |
| 07:05.44 | tuxx- | hey guys, somehow when i add parameter t to the dial command when dialing out on a sip trunk the call seems to fail. When i try it without parameter t, it functions properly. Does anyone have any idea whats going on? |
| 07:06.47 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:07.07 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rustljomilinckqt) |
| 07:07.20 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:07.32 | kaldemar | tuxx-: not without anything to debug. |
| 07:07.50 | tuxx- | oh w8, im fucking up. the whole siptrunk is borked :P |
| 07:08.18 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:14.59 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 07:28.52 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 07:32.27 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.138.203.mtnl.net.in) |
| 07:33.35 | *** join/#asterisk mcr_masvoz (~mcr_masvo@239.Red-80-39-76.staticIP.rima-tde.net) |
| 07:35.59 | mcr_masvoz | Hi. Can I throw some Asterisk questions there ? |
| 07:36.06 | zyphlar | nevar! |
| 07:36.48 | zyphlar | for posterity: my static problem was due to span2 on my digium card being bad. Digium support RMA'd, no problem |
| 07:41.48 | mcr_masvoz | I've a problem with playback on asterisk: I do a "PlayBack(file)" and asterisk sudenly stops playing after random seconds and asterisks is blocked until i hangup the phone. (pri environment). The PRIs are ok (verified with telco). Anybody has a similar problem? |
| 07:42.03 | mcr_masvoz | I tryed all versions of asterisk 1.6 and dahdi |
| 07:44.34 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 07:45.25 | *** part/#asterisk mcr_masvoz (~mcr_masvo@239.Red-80-39-76.staticIP.rima-tde.net) |
| 07:49.19 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 07:49.55 | *** join/#asterisk mcr_mv (~mcr_masvo@239.Red-80-39-76.staticIP.rima-tde.net) |
| 07:59.57 | DND | question guys. if my MOH in mp3 format is a large file(around 6mb), will it affect cpu usage or memory usage? |
| 08:01.09 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 08:04.04 | *** join/#asterisk sourcode (~code@ppp-61-90-14-41.revip.asianet.co.th) |
| 08:10.29 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:12.06 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 08:13.00 | mcr_mv | to much quality for telco use. Change sound quality to 16 bit , 8kHz, .... |
| 08:13.10 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 08:13.38 | mcr_mv | and use native formats |
| 08:13.41 | mcr_mv | mp3 is for CD wuality |
| 08:13.45 | mcr_mv | not for telco quality |
| 08:19.07 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 08:23.43 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:28.36 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:33.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:35.06 | *** part/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 08:37.22 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 08:41.10 | *** join/#asterisk Benwa (~Benwa@109.128.201.217) |
| 08:41.43 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:42.19 | angryuser | hello, what was the name of the web interface to manage asterisk's conference rooms ? (kick mute invite) ? |
| 08:43.53 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 08:52.26 | tzafrir_laptop | wow, infobot is slow. I ended up quoting dpkg (from #debian) instead |
| 08:57.27 | drmessano | Interesting |
| 08:57.40 | drmessano | the bench-g729 app doesn't bench generic |
| 08:58.46 | *** join/#asterisk mcr_mv (~mcr_mv@239.Red-80-39-76.staticIP.rima-tde.net) |
| 08:59.18 | Faustov | could anyone please tell me what nat=yes exactly does? |
| 08:59.47 | mcr_mv | in the sip.conf.sample is described |
| 09:00.03 | mcr_mv | it ignores the ip from SIP packet |
| 09:00.19 | mcr_mv | and only takes care the ip from IP paquet |
| 09:00.32 | Faustov | what about RTP? |
| 09:01.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:01.40 | tzafrir_laptop | angryuser, webmeetme? cmeetme? |
| 09:02.11 | mcr_mv | fausotv, please check sample conf... for more detailed explanation |
| 09:04.06 | Faustov | mcr_mv: thanks |
| 09:04.50 | *** join/#asterisk clintc (~clintc@n128-227-179-127.xlate.ufl.edu) |
| 09:04.56 | mcr_mv | u r welcome :) |
| 09:12.41 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 09:12.43 | *** join/#asterisk cfh (~luca@host88-99-dynamic.20-79-r.retail.telecomitalia.it) |
| 09:16.07 | mcr_mv | I've a problem with playback a file in asterisk. The file is a GSM 8KHz, 16bit, mono, playable without problems in sox. But when I place the file in asterisk, asterisk sudenly stops playing and stops execution until I hungup channels. Any idea what's happening? |
| 09:17.27 | cfh | hi all , I tried to use blind transfer code by asterisk ( the # ) with a patton fxs (with a dect phone connect) but it doesnt work . What can I do ? |
| 09:20.25 | cfh | mcr_mv : do you have converted the file with sox ? |
| 09:21.02 | mcr_mv | yes |
| 09:21.52 | cfh | can I see the command of the conversion ? |
| 09:23.58 | mcr_mv | whait... i think i found the problem... (ffplay has the same problem that i found on asterisk... i think that the problem is sox) |
| 09:24.42 | mcr_mv | anyway the comand is: /usr/bin/sox file.wav -r 8000 -c 1 file.gsm |
| 09:25.31 | cfh | can you try with : sox -V file.wav -t gsm -r 8000 -U -b -c 1 file.gsm |
| 09:26.24 | *** join/#asterisk BANSAL (~bansal@117.199.116.225) |
| 09:27.07 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 09:28.42 | coppice | mcr_mv: sox won't guess from the suffix that you want to convert to GSM. you need to explicitly tell it, as in cfh's command |
| 09:29.10 | mcr_mv | ok thanks. |
| 09:29.28 | mcr_mv | cfh, with your command sox said: sox sox: Bits value `-c' is not a positive integer |
| 09:29.51 | cfh | mcr_mv : sox version ? |
| 09:30.55 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 09:30.57 | mcr_mv | the correct cmd is: sox -V file.wav -t gsm -r 8000 -U -b 16 -c 1 file.gsm ... (you forget the number after '-b' option ) |
| 09:31.24 | mcr_mv | SoX v14.2.0 |
| 09:31.35 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 09:33.52 | coppice | for GSM encoding the -b is irrelevant. it will be ignored |
| 09:36.20 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 09:40.09 | *** join/#asterisk ming_zym (~ming_zym@116.69.210.245) |
| 09:44.51 | *** join/#asterisk af_ (~getsmart@78.134.22.42) |
| 09:45.38 | mcr_mv | I have the same problem with the args you proposed... i will try convert file with ffmpeg and will say you something. |
| 09:48.21 | Faustov | mcr_mv: how about canreinvite? In the sip.conf.sample it is not even mentioned (version 1.6.2.7) - but I can see it makes a difference when set to yes or no |
| 09:49.12 | kaldemar | Faustov: reinvite has been renamed to directmedia by that version. |
| 09:50.03 | Faustov | oh, thanks |
| 09:53.33 | tzafrir_laptop | coppice, sox does guess the type from the file name suffix |
| 09:53.38 | tzafrir_laptop | $ sox /usr/share/sounds/alsa/Side_Right.wav /tmp/test.gsm |
| 09:53.38 | tzafrir_laptop | sox WARN formats: gsm can't encode at 48000Hz; using 8000Hz |
| 09:54.25 | tzafrir_laptop | You can override that by explicitly stating the "file type" (-t) |
| 09:54.29 | coppice | tzafrir_laptop: is that a recent change? |
| 09:54.39 | tzafrir_laptop | Not that I know of |
| 09:55.14 | tzafrir_laptop | .gsm files have always just worked for me with sox (that is: in the last 5 years) |
| 09:58.20 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 10:01.49 | *** join/#asterisk cmn (~carlos@host155-48-dynamic.16-87-r.retail.telecomitalia.it) |
| 10:02.51 | mcr_mv | tzafrir_laptop, for me too but i tryed play a gsm file converted from sox and asterisk can't play it |
| 10:03.22 | tzafrir_laptop | was 'play' able to play it? |
| 10:03.27 | mcr_mv | i tryied to play it with ffmpeg and have the same problem. |
| 10:03.48 | mcr_mv | play is able to play it, but ffplay stops playing (like asterisk ) |
| 10:04.21 | tzafrir_laptop | That's odd. How was sox able to figure out the file characteristics? |
| 10:08.02 | mcr_mv | all attributes are ok, except lenght. cmd 'play' says << lenght: unknown >> |
| 10:08.40 | mcr_mv | (i'm looking in google how to convert files to gsm using ffmpeg ) |
| 10:14.00 | Faustov | kaldemar: once renamed canreinvite=no to directmedia=no, I get the following errors: http://pastebin.com/2wA01dWg - so it seems not only the setting was renamed, but also it does something else... |
| 10:16.25 | mcr_mv | your phone is behind a firewall/router ? did you opened the ports in the firewall/router ? |
| 10:18.25 | Faustov | it's between two servers, in both cases asterisk binds to a public IP |
| 10:20.07 | mcr_mv | Faustov, if both servers are conected to internet without routers then you don't have to use "nat=yes".. and if your machines are both asterisk is preferible the use of IAX2 protocol rather than SIP protocol |
| 10:21.07 | Faustov | mcr_mv: about the first: there is a machine that is behind a nat that connects to both, just trying to figure out the communication issue between these two |
| 10:21.24 | kaldemar | ~sipnat |
| 10:21.28 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 10:21.56 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 10:22.15 | kaldemar | Faustov: ^ the first one explains the nat parameters well |
| 10:22.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:24.59 | Faustov | kaldemar: yeah I've seen that and configured my stuff accordingly... |
| 10:29.14 | *** part/#asterisk cfh (~luca@host88-99-dynamic.20-79-r.retail.telecomitalia.it) |
| 10:42.23 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 10:46.48 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 10:53.19 | *** join/#asterisk bodie (~bodie@fcnoos-nd-fw01.freecode.no) |
| 10:53.28 | bodie | hi all |
| 10:53.40 | bodie | Someone with appconference plugin installed? |
| 10:54.41 | bodie | I was reading bunch of pages found on Google and I can't compile it. I'm quite sure that I'm missing some packages in system, but which one? READM from appconference doesn't say what's need as dependency or for compile which is strange of coursew |
| 10:55.00 | bodie | system is Ubuntu 10.04 LTS amd64 server |
| 11:00.29 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 11:01.17 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 11:12.38 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 11:16.05 | *** join/#asterisk yahh (~root@122.170.58.187) |
| 11:16.27 | yahh | Hi.. |
| 11:16.46 | yahh | Is asterisk 1.6 supports h323? |
| 11:18.40 | kaldemar | yahh: yes |
| 11:19.19 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:19.30 | yahh | thank you kaldemar |
| 11:19.55 | yahh | excly i was to use an asterisk as a gateway for h323 <=> SIP for video calling |
| 11:20.11 | yahh | can i do that with asterisk 1.6? |
| 11:20.38 | yahh | excly i wants* to use an asterisk as a gateway for h323 <=> SIP for video calling |
| 11:23.22 | kaldemar | as far as i know, the H.323 channel drivers don't support video calls, but you never it the video payload could be sent outside asterisk. |
| 11:25.25 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 11:25.28 | yahh | "you never it the video payload could be sent outside asterisk"- means ? |
| 11:25.57 | yahh | sorry but i can not understand |
| 11:26.02 | *** join/#asterisk rare1980_ (~as@12.25.228.67) |
| 11:26.16 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 11:26.36 | kaldemar | uh, what i meant was: "you never know if the video payload...". i was skipping words and threw in a typo to be more confusing. :) |
| 11:27.45 | kaldemar | and by that i mean that if the signaling could be passed through asterisk and the video stream directly between the endpoints. but in a gateway scenario, that doesn't sound like an option. |
| 11:28.07 | yahh | yes correct |
| 11:28.30 | yahh | in gateway it will go through server |
| 11:28.34 | bodie | Ok, I'm quite further, but still no go :-( http://pastebin.ca/1884235 |
| 11:29.47 | bodie | yahh: ha, you are looking for something like me :-) So I found this http://sourceforge.net/projects/appconference/files/ , but I'm not able to get it running yet http://pastebin.ca/1884235 . In OpenBSD it's available as package/port, but I need to start it on Ubuntu and it's a pain |
| 11:30.18 | *** join/#asterisk fauxalliance (~gerald@207.231.237.59) |
| 11:32.19 | yahh | bodie: compiling errors? |
| 11:32.55 | rare1980_ | on Intel Quad Core 2.33GHz .. using SIP g729 codec how many max calls i can make at same on this server with good quality? |
| 11:33.19 | bodie | yahh: yes and I don't know why, because authors of this plugin doesn't mention anywhere which libraries and dev files are needed |
| 11:33.48 | bodie | rare1980_: quite older comparison, but still something http://www.thrallingpenguin.com/articles/asterisk-solaris.htm |
| 11:36.09 | kaldemar | rare1980_: did you read the dimensioning article in voip-info? |
| 11:36.45 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 11:38.43 | *** part/#asterisk madduck (~madduck@debian/developer/madduck) |
| 11:39.28 | ujjain | is it "Dear Sir or Madam," ? |
| 11:40.22 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-cmgesbjyfedusgwy) |
| 11:41.11 | *** join/#asterisk asamoah (~caio@190.244.49.108) |
| 11:41.21 | *** join/#asterisk d00gster (~dt@94.99.193.40) |
| 11:44.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:44.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:47.36 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.155.218.mtnl.net.in) |
| 11:48.55 | tzafrir_laptop | bodie, looks like a missing include |
| 11:49.03 | bodie | ok, to avoid confusion for others. DON'T try to install appconference. You must use appkonference (with K) |
| 11:49.13 | bodie | appconference is more then 2 years old |
| 11:49.15 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 11:49.17 | bodie | appkonference is actual |
| 11:49.43 | bodie | which **** is responsible for it is a good idea for investigation |
| 11:50.01 | lost_soul | bodie: is that on openbsd, I only ask because ou referenced that earlier |
| 11:50.18 | bodie | no, on openbsd it's ok |
| 11:50.23 | lost_soul | kk, ty |
| 11:50.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 11:50.50 | bodie | it's bad in every manual on net, because it says appconference with c, which leads to sourceforge project with c where is not info that correct is with k |
| 11:50.53 | lost_soul | [TK]D-Fender: wb |
| 11:51.26 | lost_soul | bodie: ah, yea. I can see where that would cause confusion |
| 11:51.50 | lost_soul | I've never used it, though my * experience is minimal at best |
| 11:52.39 | bodie | yes, it' really confusing. Why don't they remove that page? |
| 11:53.14 | lost_soul | maybe e-mail the documents maintainer? |
| 11:56.02 | bodie | already done |
| 11:57.05 | lost_soul | Thats about all you can do I suppose. Would you mind my asking why you must deploy this on ubuntu though since you seem to prefer other operating systems? |
| 11:57.58 | lost_soul | As I said, limited * knowledge so I would like to know if their's limitations on the OBSD port I've always used |
| 11:59.38 | bodie | because company OS is just Ubuntu |
| 11:59.42 | bodie | Ubuntu everywhere |
| 12:00.07 | bodie | however |
| 12:00.10 | lost_soul | ah, okies |
| 12:01.02 | bodie | hmm, but again compilation problems |
| 12:01.19 | [TK]D-Fender | Just use ConfBridge |
| 12:02.01 | bodie | [TK]D-Fender: is it able to provide videoconferencing? |
| 12:02.50 | lost_soul | http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge |
| 12:03.13 | lost_soul | there's a page about it, looking myself. TY [TK]D-Fender |
| 12:04.34 | lost_soul | no metion of video |
| 12:05.26 | *** join/#asterisk m_tadeu (~quassel@89.180.164.45) |
| 12:06.38 | m_tadeu | hi all |
| 12:07.42 | *** join/#asterisk rsdvd (~rsdvd@5acaffdb.bb.sky.com) |
| 12:08.19 | rsdvd | hello all - can anyone help me troubleshoot why my SPA-3102 ATA does not allow me to dial out? |
| 12:10.00 | fauxalliance | hmm, is 180 beats per minute too fast for hold music? Alles naar de Klote! |
| 12:10.29 | fauxalliance | rsdvd, do you have a dial tone on it? |
| 12:10.46 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-78-30.home.otenet.gr) |
| 12:12.32 | rsdvd | the handset connect to line is working fine - can dial other phones on the pbx and other sip trunks......what I cannot get to work it using the PSTN on the 3102 to allow me to dial out. When I try to dial through thr trunk I created for it - I get "all circuits are busy now" |
| 12:13.15 | fauxalliance | rsdvd, clearly an issue with outbound routes/trunks. |
| 12:13.51 | rsdvd | yes - I guess! but I cannot see where....I have searched all day on google and I cannot understnad what I have done wrong. |
| 12:13.59 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:14.32 | bodie | done. Installed |
| 12:14.33 | rsdvd | according to the CLI it is using the right truck Dial("SIP/5001-00000033", "SIP/spapstn/1571,300,") .... but not connecting |
| 12:15.30 | lost_soul | bodie: appkonfrence? or the alternative? |
| 12:15.44 | bodie | lost_soul: appkonference |
| 12:15.56 | lost_soul | nice |
| 12:15.56 | bodie | lost_soul: I need videoconferencing |
| 12:16.08 | lost_soul | what was the issue? |
| 12:16.15 | bodie | btw Asterikast manager is really only for Asterisk 1.4.? |
| 12:16.34 | bodie | first issue was that mess with appconference and then with appkonference |
| 12:16.48 | bodie | that you can't set /usr/include/asterisk , but just /usr/include |
| 12:16.54 | bodie | but it was quite quick to find it |
| 12:17.04 | lost_soul | sweet |
| 12:17.13 | bodie | now I need to configure it so back to book from O'reilly :D |
| 12:17.23 | lost_soul | good luck m8 |
| 12:17.53 | bodie | I can't remember nearly anything from 3 day course on Asterisk :D |
| 12:18.06 | [TK]D-Fender | rsdvd: .... |
| 12:18.08 | [TK]D-Fender | ~freepbx |
| 12:18.11 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:18.11 | bodie | you know, when you don't work with something regularly |
| 12:18.12 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 12:18.14 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 12:19.07 | lost_soul | yea, I just switched to obsd 4.7 so need to resetup my *, not looking forward to it due to the same reasoning |
| 12:21.14 | bodie | yes, OpenBSD is wonderful and easy to setup/maintain/update. |
| 12:22.43 | lost_soul | indeed, my setup was messed up from the beginning so I didn't even save my config files. Asterisk seemed to be working for most of what I wanted. But trying to get the traffic to work through pf with hfsc queueing didn't go so well |
| 12:23.00 | bodie | mmmm have someone some good reading about recommended HW regarding videoconferencing? For now I will just prepare demo so it will be quite fine on AMD Athlon(tm) 64 X2 Dual Core Processor 5400+ , 2GB RAM and so on as there will be call eg. between 3 users or so just for demonstration |
| 12:23.13 | lost_soul | seems only one out of three connections were properly made, the rest would go through the bulk queue and wouldn't connect as a result |
| 12:24.09 | lost_soul | bodie:how large is this company your setting the PBX up for? |
| 12:25.27 | bodie | lost_soul: good question and I need to dig answer from my manager first :-) |
| 12:25.51 | bodie | lost_soul: you now that style - hey prepare videoconferencing demo with asterisk tomorrow |
| 12:25.57 | bodie | lost_soul: that's all you get :D |
| 12:26.26 | lost_soul | LOL |
| 12:28.01 | bodie | lost_soul: ok there was more. I asked and where I can install it? On VM? Common..... No, just grab some machine, there was one somewhere... oh here it is. Take eg. this one, you can delete it -> normal older desktop with cheap components |
| 12:28.19 | bodie | lost_soul: for me it will be fun :D |
| 12:31.54 | *** join/#asterisk rjek (~rjek@octopus.pepperfish.net) |
| 12:31.59 | rjek | Hi. Where can I download older versions of AsteriskNOW from? The website only has 1.7, and the "old downloads" section only has older versions of Asterisk, not AsteriskNOW. |
| 12:32.16 | lost_soul | bodie: yea, I enjoyed the learning and setup myself. I was inquiring since I've only set it up for myself (on a 450mhz pIII). It works but can only do like three calls at the same time |
| 12:32.37 | lost_soul | so trying to get an idea of how fast of a system you would need for large organizations |
| 12:33.11 | bodie | yep, but I'm taking it as something challenging so I can learn new stuff, but I'm not sure if they will want to wait so long :D |
| 12:33.37 | bodie | anyway some quick setup for 3 users or so will be maybe somewhat ok as I can remember from course |
| 12:33.47 | lost_soul | rjek: maybe see if AsteriskNOW has a channel. I don't think they support those platforms here, and I honestly have no clue |
| 12:34.12 | rjek | Yeah, I've asked there already and am awaiting a reply from the handful of people there. Thought I'd hedge my bets and ask here too :) |
| 12:35.13 | lost_soul | bodie: once you figure out what video equipment works best, if you can remember, please let me know. I've been pondering trying video conferencing for some time but have no equipment for it. |
| 12:35.38 | lost_soul | rjek: ok |
| 12:36.36 | bodie | lost_soul: ok. It will be more funny part as I don't know about any video equipment available here so I really don't know what they want to test :D Maybe I need to bring my webcam from home hehe and couple of people here have laptops with webcams |
| 12:37.07 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 12:37.29 | bodie | lost_soul: but it's not a real job regarding webcams. Just those with UVC, not others. UVC is working in OpenSolaris, all BSD and so on |
| 12:37.31 | lost_soul | bodie: thats actually what my main question is. Will I need specialized equipment or will any compatible webcam work. |
| 12:37.34 | *** join/#asterisk eliel (~eliels@host244.200-43-6.telecom.net.ar) |
| 12:38.03 | bodie | lost_soul: yep, but I need to dig that info first what they want to use for it |
| 12:38.19 | lost_soul | yep, sounds like an interesting project |
| 12:38.56 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 12:42.26 | bodie | lost_soul: so they will use just webcams |
| 12:49.27 | *** part/#asterisk rsdvd (~rsdvd@5acaffdb.bb.sky.com) |
| 12:56.38 | [sr] | hi people |
| 12:56.39 | [sr] | :P |
| 12:57.20 | Naikrovek | yo |
| 12:57.34 | [TK]D-Fender | -yo |
| 13:01.07 | yahh | kaldemar: openH323 is supporting video codecs i think |
| 13:01.12 | *** join/#asterisk mifadir (~Administr@dynamic.casap1-180-30-137-41.wanamaroc.com) |
| 13:01.25 | yahh | http://www.voxgratia.org/docs/faq.html#5_8 |
| 13:02.22 | yahh | how it would be to use that for gatway of h323<->SIP |
| 13:03.03 | mifadir | you must install it first with h323 support |
| 13:07.09 | *** join/#asterisk ariel_ (~chatzilla@173-127-87-196.pools.spcsdns.net) |
| 13:07.49 | yahh | right |
| 13:08.17 | yahh | but i was looking for some information before experimanting that |
| 13:09.36 | mifadir | it's not easy to install openh323 form source code you must do it with the rigth ptlib, |
| 13:09.53 | mifadir | in a first time it"s better to install it from PKG |
| 13:09.57 | mifadir | :-) |
| 13:10.11 | [sr] | hi Naikrovek [TK]D-Fender |
| 13:12.34 | yahh | :) thanks |
| 13:13.28 | yahh | that's why i am looking for information first |
| 13:14.26 | mifadir | http://yate.null.ro/pmwiki/index.php?n=Main.OpenH323 |
| 13:14.40 | mifadir | this is a comptible version |
| 13:14.51 | mifadir | http://yate.null.ro/tarballs/openh323/ |
| 13:17.26 | Naikrovek | jeepers creepers ubuntu 10 reboots fast |
| 13:17.31 | Naikrovek | oops, wrong chan |
| 13:18.34 | yahh | thanks |
| 13:18.35 | Naikrovek | (about 20 secs in a vmware vm if you're in this channel and are curious about what I consider 'fast') |
| 13:20.03 | bodie | are you using your computers or rebooting? |
| 13:20.08 | bodie | I prefer to use them ;-) |
| 13:20.15 | Naikrovek | feh |
| 13:20.19 | Naikrovek | system update required |
| 13:20.22 | Naikrovek | um |
| 13:20.26 | Naikrovek | system reboot required |
| 13:20.43 | bodie | compare it with reboot time of eg M9000 |
| 13:20.57 | bodie | then you will stop to care about speed of reboot if it's 15s or 10s :D |
| 13:21.16 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:21.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:21.57 | *** join/#asterisk orangey (~orangey@d67-193-125-203.home3.cgocable.net) |
| 13:22.10 | orangey | hello all |
| 13:22.41 | orangey | i've been googling around trying to figure out a couple of things, but haven't had any luck.. |
| 13:22.54 | orangey | Specifically, has anybody run into a voip provider that gives a DID that accepts / sends SMS? |
| 13:23.08 | orangey | I'm happy to setup my own asterisk that does the same if it's at least possible.. |
| 13:23.19 | orangey | but so far google voice is the only place where I've seen such a service |
| 13:25.00 | Naikrovek | orangey: you in the US |
| 13:25.01 | Naikrovek | ? |
| 13:25.05 | orangey | in canada |
| 13:25.23 | Naikrovek | i think you'll find it very difficult to find a voip provider that sends or receives SMS |
| 13:25.26 | orangey | which is why hacking something into gv is a bit less of an option too. |
| 13:25.27 | Naikrovek | that's a europe thing |
| 13:25.43 | orangey | Naikrovek: are there european providers that do? |
| 13:26.04 | Naikrovek | orangey: to/from european numbers, yes, but probably only if you're in europe |
| 13:26.09 | orangey | I know that the chances of a canadian provider are almost nil |
| 13:26.33 | orangey | I figured that finding a european / american service that also had canada support may happen |
| 13:26.34 | Naikrovek | manxpower (who often frequents this channel, though I've not seen him lately) knows a little about this |
| 13:27.53 | orangey | ah, awesome |
| 13:27.56 | orangey | thank you Naikrovek |
| 13:28.01 | orangey | may I ask what kind of setup you have? |
| 13:28.13 | Naikrovek | just a basic asterisk install connected to a voip provider |
| 13:28.15 | WIMPy | SMS is usually done in-band, just like fax, so I guess it's really hard to find. I haven't heard of any providers supporting it. |
| 13:28.34 | lost_soul | orangey: I looked for the same, wasn't able to find a service |
| 13:28.54 | Naikrovek | what do you want SMS service for |
| 13:29.16 | orangey | Any idea how google voice does it? |
| 13:29.24 | orangey | Well, I am about to move to a new city |
| 13:29.32 | orangey | right now, I use GV for SMS and a SIP provider for voice |
| 13:29.34 | fauxalliance | orangey, http://lists.digium.com/pipermail/asterisk-users/2007-December/202840.html |
| 13:29.58 | lost_soul | ease of use, rather than relying on instant messenging or the like it would be beneficial to have all communications run through * |
| 13:30.00 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:30.00 | orangey | it would be spectacular to have one number that can do SMS and voice.. SIP is a must, since I want to be able to route the number wherever I want, put it to asterisk, etc. |
| 13:30.04 | lost_soul | at least IMO it would |
| 13:30.16 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:31.18 | lost_soul | not to mention many people have unlimited SMS on their cellular phones, I see no down side to it |
| 13:31.42 | orangey | fauxalliance: awesome. that gives some excellent leads |
| 13:31.48 | fauxalliance | ;-) |
| 13:33.10 | fauxalliance | http://www.multitech.com/en_US/products/families/multimodemgprs/ |
| 13:33.55 | orangey | fauxalliance: I think I can send SMS pretty easily.. and receive it easily |
| 13:33.59 | orangey | but a unified number? |
| 13:34.01 | orangey | That's the trick |
| 13:34.14 | orangey | right now, I can have that with GV, but again, not in canada |
| 13:34.22 | fauxalliance | unification is indeed a magic word. |
| 13:34.39 | orangey | which actually leads to the question - is there gv voice / sms integration into asterisk? |
| 13:34.43 | fauxalliance | is in canada, with GV |
| 13:34.55 | fauxalliance | orangey, not afaik |
| 13:35.24 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-68-165.home.otenet.gr) |
| 13:35.25 | drmessano | How would you integrate it? |
| 13:35.41 | fauxalliance | orangey, i bet it would be trivial to get GV to forward txt to email and have allison dispatch it to you via cepstral... food for thought. |
| 13:36.03 | orangey | drmessano: how would you integrate which? |
| 13:36.08 | orangey | fauxalliance: probably very true |
| 13:36.14 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 13:36.20 | drmessano | [09:34] <orangey> which actually leads to the question - is there gv voice / sms integration into asterisk? |
| 13:36.26 | drmessano | What is there to integrate? |
| 13:36.27 | *** join/#asterisk neurosys (~neurosys@166.192.99.207) |
| 13:36.43 | orangey | drmessano: pick up a phone hooked into asterisk, it calls via gv |
| 13:36.58 | orangey | or femtocell asterisk -> cell phone sms? :) |
| 13:37.01 | orangey | I know.. I'm dreaming |
| 13:37.02 | *** join/#asterisk Polysics (~Luca@host236-69-dynamic.50-79-r.retail.telecomitalia.it) |
| 13:37.05 | Polysics | uh-oh |
| 13:37.10 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 13:37.27 | Polysics | looks like i got hacked :-( |
| 13:37.29 | orangey | but that's another question.. can I think about femtocells / asterisk? |
| 13:37.35 | drmessano | Considering that GV doesn't have that capability anywhere without a GUI. no |
| 13:37.37 | orangey | Polysics: what did? |
| 13:37.39 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 13:37.53 | orangey | drmessano: see SipSorcery |
| 13:37.54 | fauxalliance | orangey, that functionality is already available, i have two GV lines 'integrated' into asterisk per se... |
| 13:38.02 | orangey | fauxalliance: how? |
| 13:38.05 | fauxalliance | Polysics, chinese hackers? |
| 13:38.08 | Polysics | orangey, i had this asterisk box that was configured to use an outbound pre-paid SIP provider |
| 13:38.26 | Polysics | someone managed to start making calls and got 50 euros of traffic off to some russian numbers |
| 13:38.28 | fauxalliance | orangey, ringback, parking lots, custom extensions.. it's all in the book. |
| 13:39.01 | fauxalliance | Polysics, did you close _all_ your tcp ports? |
| 13:39.39 | Polysics | the server itself is open, it has to be a voice server for a distributed service |
| 13:39.48 | Polysics | i suppose i did something bad :-( |
| 13:39.56 | [TK]D-Fender | Polysics: Aren't your servers in that gov't controller psycho-limited network? |
| 13:40.04 | Polysics | but i also supposed people could connect only if they had a SIP account |
| 13:40.04 | fauxalliance | stupid, hacker did something bad. |
| 13:40.11 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 13:40.13 | Polysics | [TK]D-Fender, no, clients are |
| 13:40.16 | Polysics | some of the mat least |
| 13:40.20 | drmessano | orangey: It's not Sorcery, it's hacks using the "could-change-at-a-moments-notice" GV web interface. That's not even remotely reliable enough for Asterisk to touch. |
| 13:40.49 | Polysics | how do i secure a * server? |
| 13:41.04 | [TK]D-Fender | poyDuct tape <- |
| 13:41.04 | drmessano | There's a handful of AGI's out there that do the same, but nothing I would ever want to rely on |
| 13:41.12 | fauxalliance | drmessano, any sufficiently advanced technology is indistinguishable from magic - A.C. Clarke |
| 13:41.14 | orangey | drmessano: no no.. I'm talking about that service, SIPSorcery, which is a nice way of doing it |
| 13:41.18 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 13:41.34 | [TK]D-Fender | Polysics: permit/deny on SIP peers. Fail2ban for log-based protection |
| 13:42.14 | fauxalliance | Polysics, and re-consider tightening up TCP, with whitelists, or kitten sacrifice.. |
| 13:42.23 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:42.24 | fauxalliance | or CLOSE the DAMN ports. |
| 13:42.57 | orangey | kitten sacrifice definitely |
| 13:43.02 | drmessano | Well, the SipSorcery page is useless to me.. Requires Silverlight |
| 13:43.07 | fauxalliance | hahahaha |
| 13:43.11 | orangey | drmessano: same here : ) |
| 13:43.13 | orangey | it's the huge problem |
| 13:43.15 | fauxalliance | no hackers allowd. |
| 13:43.21 | orangey | the blog is more useful |
| 13:43.31 | jaytee | has anyone here tried using a Vonage softphone account as sip account with Asterisk? |
| 13:43.35 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 13:43.40 | drmessano | It's not a huge problem.. It's a good indication of the apples and oranges were at here |
| 13:43.53 | fauxalliance | jaytee, isn't that counter intuitive? |
| 13:44.02 | Polysics | [TK]D-Fender, permit/deny works on IPs? |
| 13:44.03 | [sr] | jaytee: i use the 3CX softphone and never had problems |
| 13:44.21 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 13:44.41 | orangey | drmessano: i.e., the major discussions critical of sipsorcery are about that damned interface |
| 13:44.52 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:44.53 | orangey | the only advantage being that it's a one-time setup |
| 13:44.59 | [TK]D-Fender | Polysics: yes |
| 13:45.01 | orangey | regardless, he does good thinking and work |
| 13:45.09 | drmessano | Asterisk is a telephony toolkit/engine, not a pretty colored GUI phone-app-thingo that plugs into anything the author could scrape together some halfway working code to support. The protocols that Asterisk supports all have proper application support |
| 13:45.18 | Polysics | but i can't be sure a peer will connect from the same IP |
| 13:45.51 | Polysics | and btw, how can i figure out how the attack was done? |
| 13:45.55 | drmessano | In other words, you won't see Asterisk supporting GV through some dodgy perl code wrapped in the tarball |
| 13:46.26 | jaytee | "Introducing the new and improved Pretty Colored GUI Phone-App-Thingo version 2.0 from Ronco!!!" |
| 13:46.26 | m_tadeu | is the application AgentCallbackLogin deprecated in asterisk1.6? |
| 13:46.28 | drmessano | When GV gets a proper API, then maybe |
| 13:46.36 | rrb3942 | Polysics, check your asterisk logs to see if they are full of failed registrations |
| 13:46.40 | fauxalliance | drmessano, chigger-rigged for sure. |
| 13:46.44 | jaytee | m_tadeu yes it is |
| 13:46.49 | orangey | drmessano: I appreciate that. |
| 13:47.05 | orangey | drmessano: it's the debian way : ) |
| 13:47.36 | m_tadeu | jaytee: do you know which application is replacing that functionality? |
| 13:47.54 | yahh | is it possible to use asterisk as a gateway between sip<=>h323? |
| 13:48.21 | [TK]D-Fender | [09:46]<rrb3942>Polysics, check your asterisk logs to see if they are full of failed registrations <- that's nto how these attacks work |
| 13:48.24 | [TK]D-Fender | not |
| 13:48.29 | jaytee | m_tadea, there were some AEL examples in the source docs of how to replace that functionality but not much else. |
| 13:48.42 | Polysics | [TK]D-Fender, although in this case they might |
| 13:48.55 | [TK]D-Fender | Polysics: they throw calls at yuo direct |
| 13:48.58 | m_tadeu | jaytee: ok thanks |
| 13:49.03 | Polysics | since i DO have someone trying to register all peers from 10000 to 99999 |
| 13:49.14 | Polysics | might it be part of the whole attack? |
| 13:49.23 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 13:49.31 | drmessano | I know some people are happy with routing a call via BoiledPears.com to SipSipper.net to IAXAMouse.org to some Perl Script to MyH323.net to Some App on a Windows box to TiVo to Asterisk to get a call, but most of us here are NOT |
| 13:49.50 | mifadir | ipsec Polysics :-) |
| 13:49.53 | orangey | drmessano: OK, point taken and moving on : ) |
| 13:50.01 | drmessano | That's more PBX In A Flash type crap.. |
| 13:50.02 | orangey | Let me ask about something else here.. femtocells? |
| 13:50.09 | Polysics | mifadir, what do you mean? any help is appreciated :-) |
| 13:50.17 | *** join/#asterisk Benwa (~Benwa@212.71.14.219.adsl.dyn.edpnet.net) |
| 13:50.28 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:50.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.29 | drmessano | orangey: You're sounding like a buzzword generator. Ask a full question with an application |
| 13:50.35 | orangey | heheheheh ; ) |
| 13:50.43 | orangey | well, I'm exploring, so you're right. |
| 13:50.54 | [TK]D-Fender | Polysics: Could be. |
| 13:51.00 | orangey | I basically am moving into a new city and new home with lots of time and resources |
| 13:51.10 | Polysics | i see a LOT of "maximum retries exceeded" |
| 13:51.10 | orangey | so I'm trying to figure out if I can / should do things |
| 13:51.11 | mifadir | Polysics try this url http://etel.wiki.oreilly.com/wiki/index.php/Secure_traffic_between_Asterisk_peers |
| 13:51.22 | fauxalliance | orangey, you can, and you should. |
| 13:51.27 | orangey | so, are femtocells worth thinking about? Or should I leave it? |
| 13:51.42 | Polysics | i can't use ipsec on clients |
| 13:51.47 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:51.51 | drmessano | Dunno, this is #asterisk |
| 13:51.56 | orangey | fauxalliance: do you use femtocells, or have you seen them? |
| 13:52.02 | orangey | drmessano: I'm talking about asterisk -> femtocell |
| 13:52.03 | WIMPy | orangey: Try to take a look at openbsc. |
| 13:52.09 | Polysics | can i use a system wide permit/deny mask? |
| 13:52.11 | orangey | WIMPy: I'm reading about it now |
| 13:52.20 | Polysics | that would ease up having to state a rule for each different IP |
| 13:52.29 | [TK]D-Fender | Polysics: Not sure. |
| 13:52.31 | yahh | is it possible to use asterisk as a gateway between sip<=>h323? |
| 13:52.34 | drmessano | What is Asterisk > femtocell. How does they hook together? What's the goal? |
| 13:52.47 | [TK]D-Fender | Polysics: of course you could jsut firewall that system striaght up |
| 13:52.58 | [TK]D-Fender | Polysics: if thats the angle you're going for |
| 13:53.01 | fauxalliance | orangey, we had one installed in the last office because our our strange geography. |
| 13:53.26 | Polysics | you mean, just lock out not-authorized blocks? |
| 13:53.34 | [TK]D-Fender | Polysics: Yes |
| 13:53.43 | drmessano | Last time I checked, the definition of a femtocell was mini cell site with a wired interface for transporting your call over IP |
| 13:53.52 | orangey | fauxalliance: I'll guess all you got out of it was voice? |
| 13:53.57 | drmessano | I don't see where Asterisk fits in |
| 13:54.02 | fauxalliance | or repeat to the carrier drmessano |
| 13:54.06 | bodie | yahh: http://downloads.oreilly.com/books/9780596510480.pdf ;-) |
| 13:54.11 | fauxalliance | orangey, yes |
| 13:54.33 | drmessano | fauxalliance: Wouldn't you just buy a REPEATER and skill the femtocell? |
| 13:54.38 | drmessano | skip |
| 13:55.33 | *** join/#asterisk benrometsch (~benromets@188-223-82-184.zone14.bethere.co.uk) |
| 13:55.44 | drmessano | In band repeaters have been out for years. They would be a much better option if your goal is to extend the RF path |
| 13:56.04 | benrometsch | hi - anyone got any ideas - I have asterisk 1.6 running nicely but whnever I try and get the callgroups working I lose voice from external SIP calls |
| 13:56.05 | fauxalliance | we had no network within range to repeat to... |
| 13:56.11 | orangey | fauxalliance: Did you have it running through asterisk too? I wonder about playing around with SMSs with the femtocell |
| 13:56.33 | drmessano | .... |
| 13:57.29 | fauxalliance | minds the gap. |
| 13:57.40 | orangey | ; ) |
| 13:57.42 | coppice | femtocells are only really suitable for very small monks |
| 13:57.55 | orangey | coppice: what do you mean? |
| 13:57.59 | fauxalliance | or very large houses. |
| 13:58.06 | fauxalliance | in the middle of NOWHERE> |
| 13:58.17 | *** join/#asterisk mrchrisadams (~Adium@78-105-1-158.zone3.bethere.co.uk) |
| 13:58.26 | drmessano | femtocells and Asterisk do not belong in the same conversation. They don't exist in the same application space |
| 13:58.43 | drmessano | that's like asking if I integrated my Netflix into apache |
| 13:58.47 | Polysics | [TK]D-Fender, and that is the ONLY defense available? |
| 13:58.56 | Polysics | IP-based security? |
| 13:59.43 | orangey | i'll have good reception in the place I go to. I'm trying to use it to do lots with my cell phone without going through my prohibitively expensive carrier |
| 13:59.57 | orangey | it sounds like I'm hearing that femtocells probably don't do what I'm thinking / seeking |
| 14:00.14 | orangey | really only good for voice |
| 14:00.20 | drmessano | orangey: Your femtocell is a gateway device FOR your "prohibitively expensive carrier" |
| 14:00.41 | fauxalliance | exactly... let them pay the licensing fees.. |
| 14:01.14 | orangey | drmessano: THat's not how I read it |
| 14:01.17 | benrometsch | hi - anyone got any ideas - I have asterisk 1.6 running nicely but whnever I try and get the callgroups working I lose voice from external SIP calls |
| 14:01.36 | fauxalliance | benrometsch, nat issues? |
| 14:01.40 | benrometsch | yeah I think so |
| 14:01.49 | benrometsch | but the rest of the system is workjing perfectly |
| 14:01.50 | orangey | drmessano: it's a thing your cell phone locks into that then does whatever., like routing phone calls through * or whatnot |
| 14:02.01 | benrometsch | it's only when one sip client pulls a call routed to another that I lose voice... |
| 14:02.02 | benrometsch | weird |
| 14:02.17 | fauxalliance | benrometsch, could be a codec |
| 14:02.19 | drmessano | orangey: It sounds like you're reading product brochures glued together with kiddie paste. You're asking about technologies and not applications of said technologies\ |
| 14:02.36 | benrometsch | fauxalliance: any easy way I could check that? |
| 14:03.01 | rrb3942 | Polysics: strong sip secrets and bruteforce detection, but good ACLs/Firewall rules can prevent the attackers from ever getting a chance |
| 14:03.12 | orangey | drmessano: I think it was pretty clear early on that you and I were thinking differently here. But I'm getting tons of very useful information from others, with the only cost being your irritation. I think we can leave it like that. |
| 14:03.16 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:03.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:03.19 | fauxalliance | benrometsch, take a verbose capture of the CLI result and describe for the audience a little about your particular setup... |
| 14:03.34 | Polysics | rrb3942, i do not see anyone registering though |
| 14:03.46 | Polysics | can calls be invoked even without registering? |
| 14:03.51 | fauxalliance | Polysics, apparenty |
| 14:03.55 | rrb3942 | yes |
| 14:03.56 | drmessano | Orangey, I don't think we're thinking here at all. You're throwing out buzzwords and don't seem to have any real knowledge of the technology and it's applications |
| 14:04.07 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:04.18 | rrb3942 | but they should still need to authenticate on the invite |
| 14:04.35 | orangey | drmessano: does your IRC application have ignore? |
| 14:04.37 | benrometsch | fauxalliance: http://grab.by/grabs/c05e0254f4a3e2c9e0f5044424baa8b4.png |
| 14:04.50 | drmessano | You're wanting to use Asterisk to route around your carrier using your carriers OWN GATEWAY... which is like saving your dog from the sharks by throwing it IN THE WATER |
| 14:04.54 | benrometsch | Fairly standard office setup - internal asterisk box, external VOIP provider, SIP clients inside the office |
| 14:05.08 | Polysics | rrb3942, i am stumped at how to proceed |
| 14:05.11 | fauxalliance | benrometsch, WTF is that, my poor eyes. |
| 14:05.15 | benrometsch | calling lol |
| 14:05.26 | drmessano | orangey: If that's how you're going to be, then so be it.. I am trying to waste my time offering you some insight, but if you want the middle finger, you go it |
| 14:05.32 | benrometsch | what's wrong with it?!? |
| 14:05.51 | orangey | drmessano: you're not insightful to me. Your values and direction are different and probably incompatible to mine |
| 14:06.04 | fauxalliance | ITS A SCREENCAP OF PLAIN TEXT! use pastebin |
| 14:06.17 | orangey | as indicated by the fact that I have gotten tons from other people, whereas you would have simply stonewalled me becuase I don't fit your mould of how learning should be done. |
| 14:06.33 | fauxalliance | s/your/everyones |
| 14:06.34 | [TK]D-Fender | [09:58]<Polysics>IP-based security? <- what else is there? |
| 14:06.41 | fauxalliance | did not spot socrates here |
| 14:06.49 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 14:06.59 | [TK]D-Fender | Polysics: Someone keeps lying to you.. you ignore them :) |
| 14:06.59 | fauxalliance | [TK]D-Fender, mac filters |
| 14:07.10 | Polysics | [TK]D-Fender, password-based security, which i thought DID work :-) |
| 14:07.14 | benrometsch | http://pastebin.com/ag8RkwLY |
| 14:07.18 | Polysics | i probably had bad secrets, ok |
| 14:07.27 | drmessano | orangey: I think you need to learn someting about ANY of the technology you're asking about before you start talking to me about "direction". It's clear to me you don't have any goals here other than gluing buzzwords, and if that frustrates you to hear it, you can't either stop being argumentative and listen to those that do know, or hit ignore |
| 14:07.32 | orangey | Actually, I find fauxalliance did a good job here.. It is clear that I'm a beginner and there's a gap, but still the information was delivered where appropriate. |
| 14:08.01 | Polysics | now yo ucan go leverage the cloud for added value virtualization of software as a service |
| 14:08.05 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:08.06 | orangey | drmessano: I'm not the one who's irritated. That's why I'm trying to help you too. |
| 14:08.09 | fauxalliance | with a hint of sarcasm fwiw |
| 14:08.09 | rrb3942 | if all your phones are on a lan or static IP's firewall the heck out of the system |
| 14:08.11 | orangey | I'm not here to irritate you |
| 14:08.14 | tuxx- | waarom is het zo warm |
| 14:08.16 | tuxx- | wrong chan |
| 14:08.20 | drmessano | I'm not irritated. |
| 14:08.23 | orangey | fauxalliance: I think that's part of it. |
| 14:08.36 | Polysics | rrb3942, problem is, the point of the system was to be distributed and allow operator's mobility |
| 14:08.45 | [TK]D-Fender | [10:07]<Polysics>i probably had bad secrets, ok <- If yours are weak you may not log enough failures to trigger bans, etc |
| 14:08.49 | puzzled | hi |
| 14:09.08 | Polysics | using a browser-based SIP ohone operators could log in fro manywhere |
| 14:09.34 | orangey | fauxalliance: I loved the 'mind the gap' thing. it cleverly and non-judgmentally says everything necessary |
| 14:09.46 | Polysics | [TK]D-Fender, i am afraid I need another solution |
| 14:09.50 | drmessano | Yet another clod who thinks if someone doesn't agree with him or coddle him, they're "irritated" |
| 14:10.10 | orangey | drmessano: sorry. Should I assume this is your way of conversation then? |
| 14:10.22 | fauxalliance | orangey, yes, indeed. |
| 14:10.35 | fauxalliance | or you will have to mind the boot, schwoomp. |
| 14:10.35 | orangey | heh ; ) |
| 14:10.38 | [TK]D-Fender | Polysics: Here's a thought: Rather then based on post-logging bans, enable a deny-first style firewall, and have them use a login-page from their station to ENABLE their IP for a time. |
| 14:10.44 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 14:10.56 | [TK]D-Fender | Polysics: Far more secure, and you can force-close those on timer, etc |
| 14:11.14 | Polysics | [TK]D-Fender, taht would be done at the iptables layer? or directly on *? |
| 14:11.16 | [TK]D-Fender | Polysics: Especially valid if they are using a web-phone from a host you control |
| 14:11.20 | [TK]D-Fender | Polysics: Yes |
| 14:11.35 | fauxalliance | [TK]D-Fender, OpenBSD has a wonderfully captivating authenticating gateway all because of PF. |
| 14:11.50 | fauxalliance | SSH in, or no phone for you. |
| 14:12.51 | Polysics | [TK]D-Fender, how can i figure out how the attack happened? |
| 14:12.58 | drmessano | or you could just use a femtocell > Skype > Windows Me > X-Lite > Asterisk |
| 14:13.12 | drmessano | Authenticate over RF using two-tone paging |
| 14:13.13 | benrometsch | any ideas fauxalliance ? |
| 14:13.18 | [TK]D-Fender | Polysics: unless you're loggin that kind fo activity or it is still happening.. you'll never know |
| 14:13.23 | fauxalliance | benrometsch, yeah, more log please... |
| 14:13.27 | fauxalliance | is hungy |
| 14:13.34 | benrometsch | that's all I got?!? |
| 14:13.50 | drmessano | That's what she said |
| 14:14.19 | Naikrovek | lol |
| 14:14.23 | Polysics | i do have the attacker's IP |
| 14:14.41 | fauxalliance | Polysics, what good is that. |
| 14:14.48 | drmessano | Polysics: Was the lost valued at greater than $4999 ? |
| 14:15.00 | drmessano | The FBI doesn't care otherwise |
| 14:15.04 | fauxalliance | 50> euro |
| 14:15.09 | Polysics | [TK]D-Fender, what i realyl wanted to ask is: is there any other way, otehr than bruteforcing a SIP account, to trigger a call? |
| 14:15.17 | Polysics | drmessano, 50 euros |
| 14:15.20 | drmessano | Shitty dialplan |
| 14:15.30 | [TK]D-Fender | Polysics: Yes... allowing un-authed calls in the first place |
| 14:15.34 | Polysics | but the real problem is that i can't bring the system up again unless i figure out what happened |
| 14:15.49 | Polysics | [TK]D-Fender, what controls that? maybe the problem is simply there |
| 14:16.11 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:16.12 | drmessano | Dumping all unauthed calls into the eff-me-from-behind default context |
| 14:16.19 | drmessano | Where they' |
| 14:16.21 | fauxalliance | drmessano, not worthy of being called a 'dialplan' more of a dial'hopethefuckitworks'scheme |
| 14:16.28 | drmessano | lol |
| 14:16.31 | [TK]D-Fender | Polysics: allowguest=yes in [general]. Having a context specified there and usable extensions, etc |
| 14:16.32 | drmessano | Yeah exactly |
| 14:16.33 | Polysics | and the IPs will probably be fro msome poor guy with a compromised Windows ME box anyway |
| 14:16.46 | [TK]D-Fender | Polysics: even HAVING a context named [default] dumb |
| 14:16.49 | [TK]D-Fender | = |
| 14:16.55 | drmessano | or a trixbox |
| 14:17.03 | Polysics | what can I show you so you can tell what i did wrong, if you cna? |
| 14:17.06 | Polysics | *can? |
| 14:17.10 | fauxalliance | speaking of phoning home. |
| 14:17.13 | Polysics | sip.conf and extensions.conf? |
| 14:17.19 | drmessano | I hear trixbox supports femtocells, and kills kittens |
| 14:17.42 | xheliox | only fluffy white kittens |
| 14:17.46 | timholum | does anyone know of a good phone that allow's for programable buttons? |
| 14:17.47 | fauxalliance | trixbox supports? |
| 14:18.02 | fauxalliance | i didnt think they encouraged that type of activity. |
| 14:18.06 | timholum | preferably polycom, but I am open to other sugestions |
| 14:18.08 | [TK]D-Fender | timholum: Polycom |
| 14:18.27 | Naikrovek | timholum: Polycom |
| 14:18.30 | fauxalliance | anything but grandstream or avaya. |
| 14:18.40 | drmessano | ~femtocell |
| 14:18.41 | infobot | from memory, femtocell is <define> femtocells and asterisk? GRAB TEH GLUE! |
| 14:18.45 | drmessano | gah |
| 14:19.15 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:19.28 | drmessano | ~femtocell |
| 14:19.29 | infobot | methinks femtocell is femtocells and asterisk? GRAB TEH GLUE! |
| 14:19.31 | timholum | [TK]D-Fender Naikrovek: which models allow for programmable buttons? |
| 14:19.42 | Naikrovek | all |
| 14:19.53 | Polysics | for what it's worth, the attacker's IP are chinese |
| 14:19.57 | Naikrovek | timholum: all from the 321 up |
| 14:20.03 | fauxalliance | Polysics, thay all are |
| 14:20.08 | drmessano | Polysics: That's worth nothing |
| 14:20.20 | WIMPy | Now we just need to #define GLUE LCR and the puzzle is complete. |
| 14:20.20 | drmessano | That's like saying "For what it's worth, this is made in china" |
| 14:20.24 | drmessano | It's worth about 10 cents |
| 14:20.33 | fauxalliance | thats how you know they 'are exerrent' |
| 14:20.35 | Polysics | it just reinforces my not liking the chinese :-) |
| 14:21.00 | timholum | Naikrovek: ok I currently have a 320 that is probably why mine does not have that feature :) |
| 14:21.00 | fauxalliance | prolly some script kiddie and a hole to china |
| 14:21.02 | [TK]D-Fender | timholum: You'd get a more comprehensive answer if you actually told us exactly what you had in mind instead of asking it piecemeal <- |
| 14:21.06 | fauxalliance | s/hole/tunnel |
| 14:21.10 | drmessano | Polysics: Nobody asked you to shop at Walmart or allow guests on your "call-the-world-for-free" context |
| 14:21.11 | Naikrovek | timholum: your 320 can do it, too |
| 14:21.14 | [TK]D-Fender | [10:20]<timholum>Naikrovek: ok I currently have a 320 that is probably why mine does not have that feature :) <- It does. RTFM :) |
| 14:21.29 | Naikrovek | timholum: what [TK]D-Fender said |
| 14:21.33 | timholum | :) I will have to do that |
| 14:21.39 | Polysics | drmessano, is the problem in sip.conf and/or extensions.conf? |
| 14:21.50 | fauxalliance | probably both |
| 14:21.59 | drmessano | Polysics: Sorry, app_psychic is broken. Where's the pastebin? |
| 14:22.01 | [TK]D-Fender | pummels timholum with the Polycom Adminitrators Guide: steel-cover Edition |
| 14:22.17 | fauxalliance | you can read the book, or eat it. |
| 14:22.21 | tuxx- | thats gotta hurt |
| 14:22.23 | [TK]D-Fender | whamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAM |
| 14:22.27 | tuxx- | does it have spikes [TK]D-Fender ? |
| 14:22.33 | fauxalliance | diamond plate |
| 14:22.38 | tuxx- | steel-spike-covered book |
| 14:22.56 | timholum | oh and you wanted to know what I was trying to do, I am trying to make a call center app that a user just has to press a button to call the next person on the list |
| 14:23.00 | [TK]D-Fender | tuxx-: No, I have a 2x4 with some rusty nails however :) |
| 14:23.20 | tuxx- | hehehe |
| 14:23.24 | drmessano | 99% of asterisk hacks are from shitty dialplans. The other 1% are trixboxes compromised via their Windows On Barbie subsystem using the ohgodken API |
| 14:23.26 | [TK]D-Fender | timholum: that is a boring f-ing SPEED DIAL |
| 14:23.42 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 14:23.45 | [TK]D-Fender | timholum: FFS its a 10 second job without even modding the buttons |
| 14:24.12 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 14:24.14 | [TK]D-Fender | resumes pummeling timholum with the Polycom Adminitrators Guide: steel-cover Edition |
| 14:24.23 | tuxx- | xD |
| 14:24.26 | timholum | :) |
| 14:24.31 | *** part/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 14:24.53 | drmessano | "shitty dialplans" includes dumping unauthed calls into "naughty places" in the dialplan |
| 14:24.58 | Polysics | http://pastebin.com/P8dPsbjz |
| 14:25.09 | Polysics | extensions.conf and sip.cof |
| 14:25.11 | yahh | I think asterisk can work as a gateway between SIP and H323, am i right? |
| 14:25.18 | fauxalliance | drmessano, what do you tink about asterisk barbie on solaris |
| 14:25.40 | [TK]D-Fender | context=incoming ; Default context for incoming calls |
| 14:25.47 | [TK]D-Fender | exten => _0.,1,Dial(SIP/${EXTEN:1}@sip.messagenet.it) |
| 14:25.49 | Polysics | i am using Adhearsion for al ot of things, but before we look into that, i wanted to be sure the problem was not simply there |
| 14:25.52 | fauxalliance | Polysics, or maybe you transferred that polite caller to extension 91. |
| 14:25.55 | [TK]D-Fender | Polysics: You're just F-ing ASKING for it... |
| 14:26.05 | drmessano | OUCH |
| 14:26.27 | Polysics | erm, i suppose i fucked up badly :-) |
| 14:26.28 | rrb3942 | very ouch |
| 14:26.29 | [TK]D-Fender | Polysics: Please place the abrrel firmly to your temple, stand over there on that plastic sheet and pull the trigger, k? :| |
| 14:26.33 | drmessano | Yep, that dialplan is like bending over at the nudist colony |
| 14:26.33 | kaldemar | yahh: still yes, but you might want to forget the video part. |
| 14:26.35 | [TK]D-Fender | barrel* |
| 14:26.52 | Polysics | shoudl i stand here on the big red X? |
| 14:27.05 | drmessano | Polysics, don't drop the soap |
| 14:27.14 | *** part/#asterisk bodie (~bodie@fcnoos-nd-fw01.freecode.no) |
| 14:27.20 | Polysics | ok, what do I do then? |
| 14:27.20 | fauxalliance | shall we make some swag out of it. 'Plan B' t-shirts and the like. |
| 14:27.21 | yahh | kaldemar: following link show that if i use openh323 then it is supporting video |
| 14:27.37 | yahh | http://www.voxgratia.org/docs/faq.html#5_8 |
| 14:27.39 | fauxalliance | Polysics, reboot three times in quick succession and learn how TCP/IP works. |
| 14:27.55 | Polysics | just so you know, i will be blaming an obscure bug in the exact version of * i am using |
| 14:28.22 | fauxalliance | Polysics, thats ok, asterisk knows better, and blames YOU, not even the hacker, just you. |
| 14:28.24 | drmessano | I'm sure the devs will appreciate that |
| 14:29.00 | Polysics | not in public, just with the bosses that luckily are not good with technical speak (and apparently i am not either) |
| 14:29.18 | fauxalliance | Polysics, then keep that to yourself. |
| 14:29.50 | mcr_mv | how to deactivate mmx, sse, etc. features at compiletime in asterisk ? |
| 14:29.52 | drmessano | So you want to regain confidence with your bosses by blaming the PBX instead? |
| 14:30.04 | [TK]D-Fender | Polysics: You point non-authed calls to [incoming] which automatically allows OUTGOING. in [general] do context=gofuckyourself |
| 14:30.05 | drmessano | That'll show em! |
| 14:30.26 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:30.27 | wcselby | o/ |
| 14:30.27 | Polysics | no, i want to avoid getting executed in the break room |
| 14:30.43 | Polysics | coffee machine guillotine |
| 14:30.46 | fauxalliance | drmessano, o/\o |
| 14:31.08 | *** join/#asterisk garymc (~chatzilla@host81-139-81-143.in-addr.btopenworld.com) |
| 14:31.24 | Polysics | [TK]D-Fender, i get the problem with the extension |
| 14:31.33 | kaldemar | yahh: where? it says that the OpenH323 library supports video, nothing about asterisk channel drivers supporting video. |
| 14:31.37 | Polysics | but where do calls go without a default extension |
| 14:31.46 | fauxalliance | Polysics, you don't get it. |
| 14:32.00 | fauxalliance | They go where you tell them / allow them to go. |
| 14:32.00 | yahh | kaldemar: ohh yes |
| 14:32.24 | yahh | you mean channel drivers are not supporting video |
| 14:32.58 | yahh | but without asterisks how they can use openh323 liberary? |
| 14:33.20 | yahh | using any other pbx system or switch? |
| 14:33.25 | kaldemar | yahh: yes, i said specifically "H.323 channel drivers". |
| 14:33.32 | Polysics | fauxalliance, you mean, i use the context option in the SIP accounts, not the default one |
| 14:33.43 | Polysics | can i just remove the "context" in [general]? |
| 14:33.48 | kaldemar | yahh: it's a library, any other software may use it. |
| 14:33.59 | drmessano | Polysics: ~book |
| 14:34.34 | jaytee | ~book |
| 14:34.34 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:34.37 | drmessano | Polysics: This is Asterisk 101 here. For the sake of the security of your box, I suggest a read of the book |
| 14:34.49 | yahh | kaldemar: okay |
| 14:35.01 | wcselby | Polysics - the "default" context is the default, whether you have it there or not, in the general section of sip.conf. The only thing you can do is set a new context. However, the only thing this is really going to get used for is anonymous sip calls coming in over the internet |
| 14:35.03 | yahh | thank you for your help |
| 14:35.29 | Polysics | wcselby, so i can just remove the option? |
| 14:35.40 | drmessano | NO |
| 14:35.42 | wcselby | Polysics - which, unless you have a need for, is generall a bad idea. You can set "allowguest=no" to remove that behavior |
| 14:36.03 | Polysics | by the way, fro mwhat i gather, the problem is only marginally with that |
| 14:36.15 | Polysics | it's the outbound extension that allows the attack |
| 14:36.22 | drmessano | Polysics, not true |
| 14:37.01 | Polysics | drmessano, but if the context does not have an extension taht allows calling outside, how can that be done? |
| 14:37.22 | drmessano | Polysics: You putting unauthed INCOMING calls into a context that allows OUTGOING with NO AUTH is the problem |
| 14:37.30 | wcselby | Polysics - you shouldn't have an extension that allows outbound calling in your default context. |
| 14:37.35 | drmessano | Having a nice TV isn't a problem, your FRONT DOOR IS OPEN |
| 14:37.37 | *** part/#asterisk mifadir (~Administr@dynamic.casap1-180-30-137-41.wanamaroc.com) |
| 14:37.54 | Polysics | that means that removing the extension fixes half of the problem, no? |
| 14:38.26 | Polysics | then i can just use allowguest=no in sip.conf and that should do it |
| 14:38.46 | Polysics | i did a stupid thing, but this looks like a fix |
| 14:38.55 | *** part/#asterisk rjek (~rjek@octopus.pepperfish.net) |
| 14:38.58 | Polysics | then i will learn the book by memory, but now i need to fix this :-) |
| 14:39.16 | fauxalliance | Polysics, strike that, reverse it. |
| 14:39.25 | drmessano | You need to fix this buy implementing a proper dialplan, which involves that book and not 3 lines of code |
| 14:39.29 | drmessano | by* |
| 14:40.27 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 14:40.30 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 14:41.01 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 14:41.57 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 14:42.10 | drmessano | Actually, the sample sip.conf probably has enough warnings in there to get you past this. |
| 14:45.39 | m_tadeu | please take a look at http://pastebin.com/Sy0VVByE ...thanx |
| 14:46.14 | Polysics | drmessano, i appreciate you telling me to go study, because you are right |
| 14:46.34 | Polysics | but please tell me is allowguest=no and removing that extension will allow me to get the system back up |
| 14:46.51 | Polysics | the extension isn't even used, it is probably some old test code |
| 14:47.12 | drmessano | Yes. do that and you're done |
| 14:47.28 | p3nguin | Did you read the book already? |
| 14:47.36 | drmessano | He doesn't care/want to |
| 14:47.52 | Polysics | i sure will, you can bet on it |
| 14:47.53 | p3nguin | Oh, then we probably don't care or want to help him. |
| 14:47.53 | drmessano | So to answer your question, do that and you're done |
| 14:48.02 | Polysics | but one thing is knowing things deeply, which i need to do |
| 14:48.10 | Polysics | one thing is fixing a blocking problem |
| 14:48.17 | drmessano | p3nguin: "Thank you, drive through" is more like it |
| 14:48.40 | Polysics | i do care about the book ,just i can't read a book in 30 minutes :-( |
| 14:49.00 | drmessano | No, you could have read the sip.conf sample 5 times by now, like I suggested |
| 14:49.06 | drmessano | Because even it tells you "DONT DO THIS" |
| 14:49.12 | drmessano | and documents the allowguest |
| 14:49.24 | drmessano | [10:47] <drmessano> Yes. do that and you're done |
| 14:49.33 | drmessano | ^^^ There's the McDonalds answer |
| 14:49.34 | Polysics | had missed that, thanks |
| 14:49.35 | *** join/#asterisk Footman (~Footman@gwdev.creape.unilim.fr) |
| 14:49.40 | Footman | hello |
| 14:51.37 | Footman | I've a problem with this configuration : Phones <-> PABX Siemens HiPath 3350 <-> BRI card <-> Asterisk <-> Internet |
| 14:51.50 | Footman | no problem for incoming calls |
| 14:52.04 | fauxalliance | drmessano, and by "do that" you meant read the book of course... |
| 14:52.32 | drmessano | What book? |
| 14:52.51 | fauxalliance | the one that should be in Polysics hands.. |
| 14:52.52 | p3nguin | Book? We don't need no stinking book. |
| 14:53.04 | Footman | but for external calls, I can't see callee number (Asterisk say extension s does not exist) |
| 14:53.08 | fauxalliance | nope, just a sensible dialplan.. |
| 14:53.10 | drmessano | If you looked at his dialplan, he has no interest in reading the book, nor has he ever. |
| 14:53.15 | WIMPy | Footman: What's the exact setup? What hardware, what driver and what's the issue? |
| 14:53.27 | fauxalliance | drmessano, if it makes you want to vomit, it must be 'art' |
| 14:53.41 | drmessano | lol |
| 14:54.02 | Polysics | just so you know, the book is in my shopping cart right now :-) |
| 14:54.08 | drmessano | I'm just hoping he drags this out longer so I can finish these few calls I have to make |
| 14:54.24 | Footman | WIMPy: Asterisk 1.4.21.2-BRIstuffed-0.4.0 with Junghanns duoBRI card |
| 14:54.26 | p3nguin | footman: If the call is being fed into the proper context, either make sure the numeric extension gets added to your calls OR create the 's' extension within that context. |
| 14:54.37 | Polysics | drmessano, lol, then it must be someone else's system, as my * is down :-) |
| 14:54.41 | p3nguin | polysics: It is available in PDF version for free. |
| 14:54.57 | Polysics | i like having dead trees in my hands |
| 14:55.09 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 14:55.10 | fauxalliance | Polysics, <google>filetype:pdf asterisk</google> |
| 14:55.11 | drmessano | Polysics: ... but you're DOWN |
| 14:55.13 | Polysics | and since i actually realyl need reading it, it is correct to give back :-) |
| 14:55.17 | Footman | p3nguin: the problem is that I can't launch the call via Internet if I don't have the callee number... |
| 14:55.48 | WIMPy | Footman: What's connected where and how? |
| 14:56.13 | drmessano | Polysics: Why don't you check out the PDF and fix your system first. Then worry about giving back |
| 14:56.14 | p3nguin | The other side needs to be sending the numeric extension, but it hasn't been configured to do it. |
| 14:56.28 | Polysics | if someone wants soem free calls, i will be taking up the server now and still ahve 5 euros of credit :-) |
| 14:56.55 | drmessano | Polysics: Did you fix the dialplan? |
| 14:57.17 | Polysics | removed extension, allowguest=no in sip.conf |
| 14:57.27 | Polysics | so i think there are no free calls to be had |
| 14:57.31 | Footman | I can see the callee number in the CLI with "bri intense debug span 1", for example : [Jun 16 16:17:19] VERBOSE[22386] logger.c: 1 < Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'XXXXXXXXXX' ] (XXXXXXXXXX is the callee number) |
| 14:58.11 | wcselby | did Polysics ever pastebin his extensions.conf and sip.conf for ya'll? |
| 14:58.29 | Polysics | wcselby, http://pastebin.com/P8dPsbjz |
| 14:58.46 | *** join/#asterisk tarik (~chatzilla@41.140.248.167) |
| 14:59.04 | tarik | Hi all |
| 14:59.35 | Footman | WIMPy: I have phones connected to a PABX Siemens HiPath 3350. This PABX is connected to an Asterisk server via a Junghanns duoBRI card. The Asterisk server is connected to Internet for join SIP providers. |
| 15:00.30 | WIMPy | Footman: So you're running * in NT mode? |
| 15:00.30 | wcselby | Polysics - um.... |
| 15:00.42 | Footman | WIMPy: yes, NT mode |
| 15:00.56 | WIMPy | Footman: I'd guess there's something wrong with the 'immediate' settings. |
| 15:00.59 | Polysics | wcselby, i know (actually i don't but i now understand what was wrong) |
| 15:01.17 | Footman | WIMPy: I'm in France, if it's important |
| 15:01.18 | *** join/#asterisk Glasswalker (~Glasswalk@CPE005056ad5173-CM001225e00d58.cpe.net.cable.rogers.com) |
| 15:01.57 | wcselby | Polysics - you need to take your outgoing extension out of the incoming context, create an outgoing context, and include that in your [phones] context. but then, unless it's in your agi, I don't see how you're calling your phones.... |
| 15:02.10 | Polysics | i am dialing them in the AGI |
| 15:02.14 | *** join/#asterisk b14ck (b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 15:02.25 | Polysics | but the AGI always checks for the user to be a valid SIP peer |
| 15:02.50 | Glasswalker | Hey, I have trixbox 2.6.2.3, I'm trying to connect it to my talkswitch PBX by setting up a generic SIP extension on the talkswitch, and then connecting to it as a trunk on the trixbox. The SIP extension works fine from generic SIP softphones. |
| 15:02.51 | wcselby | Polysics - but from what I can see, even with allowguest=no, you might be screwed |
| 15:03.03 | Polysics | wcselby, since i do not actually need the outgoing extension, i just removed it |
| 15:03.05 | Glasswalker | Problem is it's not working, and the talkswitch doesn't even show the asterisk box attempting to connect. |
| 15:03.25 | Glasswalker | I have asterisk connected to a voip provider with a trunk fine, and that trunk works great. |
| 15:03.33 | wcselby | ~freepbx |
| 15:03.34 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:03.35 | Glasswalker | so how do I determine if it's connecting, or what do I look for in the logs |
| 15:03.38 | wcselby | Glasswalker ^^^^ |
| 15:03.50 | Glasswalker | gotcha |
| 15:03.51 | Glasswalker | thanks |
| 15:03.54 | wcselby | Glasswalker - or try the trixbox forums |
| 15:04.23 | wcselby | Polysics - you removed the outgoing extension, but not the ability to make an outbound call. thus you need an outgoing extension |
| 15:04.39 | wcselby | at least, that's how I see it....? |
| 15:04.53 | Footman | WIMPy: this is my zaptel.conf and zapata.conf : http://pastebin.com/te4NicS7 |
| 15:04.58 | m_tadeu | I'm having some trouble with realtime queues. when I register a client the register doesn't show on the table. Posted some details in http://pastebin.com/Sy0VVByE . Can someone take a look, plz? thanx :) |
| 15:05.10 | Footman | what do you mean by immediate settings ? |
| 15:05.16 | Polysics | wcselby, how does a client initiate a call without an extension to call? |
| 15:05.27 | wcselby | Polysics - huh? |
| 15:06.03 | wcselby | Polysics - you should have separate inbound, outbound, and internal contexts. I mean, you don't have to, but it's a nice separation. you stuff too many things in one context and you open yourself to attack. |
| 15:07.23 | Polysics | the system is supposed to be like this: incoming -> IVR, internal (called "phones") => users calling each others, outgoing => calling user's cellphones when out of office |
| 15:07.24 | tarik | I'm working with Asterisk, and I've seen two missed calls, but i don't know the number/extension who called me, and when i check my reports (FreePBX) i've seen in the channel ( sip/117.41.228.242-0a511da8) |
| 15:07.36 | Polysics | wcselby, all the calling is supposedly done by AGI |
| 15:07.43 | Polysics | the extensions are just entry points |
| 15:07.46 | WIMPy | Footman: If I remember correctely there are seetings called immedieate and alwaysimmediate which you might want to play with. But meybe that was on misdn. - Too long ago. |
| 15:07.58 | WIMPy | But the HiPath will always use overlap sending. |
| 15:07.58 | tarik | what that's mean ( sip/117.41.228.242-0a511da8) ? |
| 15:08.16 | Polysics | and there is NOTHING in the AGI taht allows calling an arbitrary number |
| 15:08.18 | wcselby | tarik - it's a sip channel from a peer at 117.41.228.242 |
| 15:08.44 | tarik | But i don't know this ip |
| 15:08.49 | [TK]D-Fender | m_tadeu: Queue member table is for STATIC devices. It replaces the "member=>" lines from queuws.conf. Therefor dynamically added devices will NOT be added |
| 15:08.55 | *** join/#asterisk ruben23 (~unit41@202.137.112.11) |
| 15:09.08 | wcselby | Polysics --> exten => _0.,1,Dial(SIP/${EXTEN:1}@sip.messagenet.it) allows outgoing calls over your sip.messagenet.it trunk |
| 15:09.29 | wcselby | if someone were to dial 0123456789, you'd end up sending 123456789 to sip.messagenet.it |
| 15:09.33 | Polysics | wcselby, which is the extension I removed as it is not needed by the system, just some old code |
| 15:09.36 | wcselby | as an outbound call |
| 15:09.54 | wcselby | if you removed it, you should have taken it out of the pastebin |
| 15:09.59 | wcselby | sorry, it confused me |
| 15:10.00 | ruben23 | hi guys how do i stop running asterisk not on console... |
| 15:10.17 | wcselby | tarik - then someone has registered to your system on that IP and is using it |
| 15:10.29 | [TK]D-Fender | ruben23: conenct to it and do "core stop now" |
| 15:10.38 | m_tadeu | [TK]D-Fender: so the only way to check which agent are in the queue is with 'queue show' command? |
| 15:10.52 | [TK]D-Fender | m_tadeu: Or related AMI commands |
| 15:11.18 | m_tadeu | [TK]D-Fender: ah cool...thanx |
| 15:12.08 | ruben23 | <PROTECTED> |
| 15:12.31 | Footman | WIMPy: immediate seems to be for FXS |
| 15:12.52 | [TK]D-Fender | ruben23: Depends on your sue of a startup script. You should already know which one you may be using |
| 15:13.03 | [TK]D-Fender | ruben23: otherwise call it with asterisk -rx |
| 15:14.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:14.59 | Polysics | wcselby, current configs http://pastebin.com/bBSys8Cr |
| 15:17.17 | ruben23 | [TK]D-Fender:asterisk -rx stop now |
| 15:18.26 | ruben23 | asterisk -rx "stop now" |
| 15:18.29 | wcselby | Polysics - you should be safe for now, but please, please, please, buy the book, then download the pdf, and read that until you get the hard copy, and then read that. |
| 15:18.49 | wcselby | Polysics - then keep it on your desk, and refer back to it |
| 15:19.00 | Polysics | book is already incoming, and the PDF version already downloaded, i had an order on AMazon open anyway :-) |
| 15:19.29 | wcselby | Polysics - good luck, and welcome to #asterisk |
| 15:19.59 | Polysics | wcselby, thank you for not bashing me, although some bashing WAS in order after all :-) |
| 15:20.26 | wcselby | Polysics - it happens in here from time to time. at least [TK]D-Fender didn't pull out his ClueBat[tm] |
| 15:20.39 | quenenni2 | what does mean "the dial tone will continue after you pressed 9" in [local] ignorepat ? |
| 15:20.49 | Polysics | no, he was busy hitting someone with a Polycom manual |
| 15:20.54 | Polysics | i got lucky |
| 15:22.11 | drmessano | ~cluebat |
| 15:22.12 | infobot | *WHACK* *WHACK* *WHACK* |
| 15:23.46 | quenenni2 | nobody? |
| 15:23.48 | *** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com) |
| 15:25.27 | devdvd | anyone using the polycom 321, im having a problem getting it to connect to asterisk. It dont even seem to be trying to connect to the server. |
| 15:27.45 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 15:28.52 | [TK]D-Fender | quenenni2: Do you use Zaptel/DAHDI FXS channels? |
| 15:29.55 | quenenni2 | no i do not use zaptel... (note that i'm a total asterisk-noob) |
| 15:29.57 | Polysics | shtop the whacking, my eyes shwell up and i can't shee the conshole |
| 15:30.44 | [TK]D-Fender | quenenni2: then ignore it. It does not apply to you |
| 15:30.59 | quenenni2 | ok thanks but what does it means? |
| 15:31.36 | [TK]D-Fender | quenenni2: It means exactly what it says. If you started to dial a # that started with "9" then it would not stop the dialtone upone receipt of that first digit |
| 15:31.42 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 15:33.51 | Naikrovek | devdvd: how are you configuring the phone |
| 15:35.02 | devdvd | Naikrovek: via the web interface |
| 15:35.03 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 15:35.12 | Naikrovek | devdvd: how many phones, just 1? |
| 15:35.34 | quenenni2 | [TK]D-Fender, sorry english is not my mother tongue, what does exactly means : " then it would not stop the dialtone upon receipt of that first digit" |
| 15:36.02 | [TK]D-Fender | quenenni2: Start dialing a # with 9 and the dialtone won't stop |
| 15:36.05 | Naikrovek | quenenni2: you will hear dial tone, you will dial 9, you will continue to hear dialtone until you dial another digit |
| 15:37.59 | [TK]D-Fender | quenenni2: La tonalite que t'etends avant de composer le numero arretera-pas si il commence par un 9 |
| 15:38.03 | [TK]D-Fender | peuMieux? |
| 15:38.10 | [TK]D-Fender | quenenni2: Mieux? |
| 15:39.05 | devdvd | Naikrovek: yes, just 1 phone, ill be setting up an autoprovision server later. Just a moment and ill tell you what i got for each field. |
| 15:39.09 | devdvd | in the web gui |
| 15:40.05 | Naikrovek | even for one phone, an FTP server with configs is easier than the web gui |
| 15:40.14 | Naikrovek | and i'm not saying that to be snarky |
| 15:40.24 | Naikrovek | primary benefit of the ftp server is you can give the phone some place to put its logs |
| 15:40.29 | Naikrovek | then you can see what is going on |
| 15:40.38 | Naikrovek | s/primary/one of the/ |
| 15:40.45 | [TK]D-Fender | Users programming Polycom phones via the web interface should be dragged out and shot. Survivors should be shot AGAIN |
| 15:40.52 | Naikrovek | lol |
| 15:41.00 | Naikrovek | i won't go that far, but you're on to something there |
| 15:41.02 | quenenni2 | [TK]D-Fender, yes better... ;) but if i just press 9 i'm supposed to hear infinitly the dial tone? |
| 15:41.07 | Naikrovek | they'll want to shoot themselves by the time they're done |
| 15:41.08 | drmessano | I refuse to use any phone without a proper gopher interface |
| 15:41.43 | [TK]D-Fender | quenenni2: Not il va ignorer seulement ce premiere 9 |
| 15:41.57 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:42.01 | [TK]D-Fender | (ignorer de desactiver le ton) |
| 15:44.04 | tarik | wcselby: tarik - then someone has registered to your system on that IP and is using it =>> I can't see any extension registred in my asterisk |
| 15:44.39 | quenenni2 | [TK]D-Fender, ok i aproximately understand, i will care later... |
| 15:45.06 | *** join/#asterisk hfb (~hfb@pool-96-247-49-124.lsanca.dsl-w.verizon.net) |
| 15:46.31 | drmessano | OMG that sounds like that other guy |
| 15:46.40 | drmessano | "I know I am gonna get hacked, but lunch is getting cold" |
| 15:47.30 | [TK]D-Fender | tarik: You don't need to be registered to PLACE CALLS |
| 15:50.06 | tarik | <[TK]D-Fender : But how i can deny the person who use my asterisk ? |
| 15:51.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:51.18 | *** join/#asterisk enzo (~enzo@88.165.158.52) |
| 15:51.23 | *** join/#asterisk Z_God (~julius@wlan236159.mobiel.utwente.nl) |
| 15:51.42 | [TK]D-Fender | tarik: permit/deny masks on your SIP peers. disallowing unauthed calls. Firewall your server, etc |
| 15:51.59 | Polysics | it is a sort of common problem then :-) |
| 15:52.03 | enzo | Hi |
| 15:52.35 | [TK]D-Fender | Polysics: Yes, the Moron Virus it approaching CDC alert level Orange |
| 15:52.51 | Polysics | has it ever been lower? :-) |
| 15:53.02 | enzo | I have a strange thing, I use asterisk to automatically answer when ring begins, however asterisk waits several rings before taking the call |
| 15:53.28 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 15:53.38 | p3nguin | enzo: I don't see your dialplan. |
| 15:53.39 | enzo | I suspect some threshold |
| 15:53.51 | Corydon76-dig | enzo: DAHDI channels? |
| 15:54.25 | enzo | it's not a problem of dialplan p3nguin, I'm using my dialpan for several years. But I have changed my line in. It's now the line from an internet box |
| 15:54.50 | p3nguin | In that case, show ALL of your configuration files. |
| 15:54.53 | [TK]D-Fender | enzo: via what interface? |
| 15:55.05 | *** join/#asterisk bodie (~bodie@cm-84.215.50.129.getinternet.no) |
| 15:55.24 | bodie | nice one for telephony :-) http://www.datacenterknowledge.com/archives/2010/06/14/seamicro-unveils-its-low-power-server/ |
| 15:55.32 | enzo | I have a X101P to get the call when line is ringing |
| 15:55.45 | Corydon76-dig | enzo: set callerid=no |
| 15:55.55 | Corydon76-dig | err, usecallerid=no |
| 15:56.02 | enzo | I try |
| 15:56.26 | enzo | in the log is the called ID = unknown, you think it takes some time trying to get the caller id Corydon76-dig ? |
| 15:56.43 | Corydon76-dig | The reason it rings several times is that the callerid appears in the signal between the first and second rings, so Asterisk needs to wait for both |
| 15:57.07 | Corydon76-dig | only then is the dialplan started |
| 15:58.32 | enzo | I have put no in my zapata.conf, but asterisk waits again a long time before really getting the call |
| 15:58.48 | Corydon76-dig | enzo: did you restart? |
| 15:58.51 | enzo | yes |
| 15:59.06 | enzo | i've done /etc/init.d/asterisk restart indeed |
| 15:59.22 | Corydon76-dig | pastebin your entire zapata.conf |
| 15:59.41 | enzo | ok |
| 15:59.55 | *** join/#asterisk KnucKles_ (~bocao_198@189.89.153.211) |
| 16:01.09 | enzo | http://pastebin.com/WM43up2L Corydon76-dig |
| 16:01.12 | KnucKles_ | Hi all!! |
| 16:01.39 | KnucKles_ | Does anyone knows whats is the error PRI got event: HDLC Bad FCS on Primary D-channel of span 1? |
| 16:02.30 | Chainsaw | KnucKles_: It suggests corruption of data on your D-channel, which could have many causes. |
| 16:02.45 | Chainsaw | KnucKles_: It this a known good span? (i.e. you have had it connected to other equipment and placed calls over it?) |
| 16:03.50 | KnucKles_ | Chaninsaw: Yes.. I can send and receive calls normally but, after 3 hours I need to restart the DAHDI (dahdi restart) |
| 16:04.06 | Corydon76-dig | enzo: watch the console when you call in at verbose >= 3 |
| 16:04.22 | Corydon76-dig | enzo: does it start the simple switch immediately upon the first ring? |
| 16:04.35 | enzo | i've done this debug, i paste it |
| 16:04.48 | KnucKles_ | Chaninsaw: because the span freeze. |
| 16:05.33 | enzo | http://pastebin.com/P26Actge Corydon76-dig |
| 16:05.56 | enzo | no the simple switch is after the 5 ring or more Corydon76-dig |
| 16:06.20 | Corydon76-dig | enzo: then it's your telephony provider, not you |
| 16:06.48 | [TK]D-Fender | enzo: If you plug an analog phone in parallel, do you hearing it ring for 5 times before * pics up? |
| 16:06.56 | enzo | I try |
| 16:07.05 | KnucKles_ | Chaisaw: I'm connected on span 1 with the PSTN provider and on span 2 the PBX. The connection with PBX there is no problem. Only with PSTN |
| 16:07.28 | *** join/#asterisk orangey (~orangey@d67-193-125-203.home3.cgocable.net) |
| 16:07.35 | orangey | I think I got it!!! |
| 16:07.55 | orangey | It looks like Twilio offers DID numbers that do both SMS and voice (looks like SIP) |
| 16:09.41 | lost_soul | 3 cents per SMS, not to bad but that can certainly add up quickly |
| 16:10.28 | orangey | lost_soul: less than the 15/sms charged to me now : ) |
| 16:10.30 | *** part/#asterisk bodie (~bodie@cm-84.215.50.129.getinternet.no) |
| 16:10.37 | orangey | regardless, I can't figure out how to do sip with this thing |
| 16:11.14 | lost_soul | orangey: yea, so long as it's a cut rate from what you pay now I guess ya can't go wrong |
| 16:11.36 | orangey | indeed! |
| 16:12.00 | orangey | but obviously a nogo if no SIP. Alarmingly, it looks like THEY run Asterisk, but that their API is proprietary |
| 16:12.43 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 16:13.02 | lost_soul | orangey: the method I was looking into was to get one of those devices that incorporate GSM cellular line into asterisk, if you have a plan offering unlimited SMS it would be the cats ass, but still would need to figure out how to send the SMS through asterisk |
| 16:13.49 | orangey | I have time and desire. but the issue is that i want a google voice-like thing where a unified number receives SMS and voice, and then delegates them wherever I want |
| 16:14.09 | orangey | twilio does that |
| 16:14.17 | orangey | which is interesting |
| 16:14.18 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 16:16.45 | enzo | [TK]D-Fender: Corydon76-dig, when i plug a simple phone, it ring immediately when i call the line |
| 16:17.02 | enzo | And i can also see the caller id |
| 16:17.17 | enzo | my provider sends it when it knows it |
| 16:19.20 | *** part/#asterisk jbeez (jbeez@pool-72-78-238-135.phlapa.fios.verizon.net) |
| 16:19.43 | Corydon76-dig | enzo: Ask your reseller for hardware support |
| 16:19.52 | enzo | I'm in France by the way with an internet provider that gives me a telephony line via internet |
| 16:20.37 | [TK]D-Fender | enzo: And it takes 5 rings for "simple switch" to appear? |
| 16:20.39 | enzo | But the strange thing is that a simple telephone gets the call immediately, and everything works fine |
| 16:20.45 | enzo | yes [TK]D-Fender |
| 16:20.57 | enzo | 5 rings, 8 rings sometimes ! |
| 16:21.01 | [TK]D-Fender | enzo: Perhaps your ringing indication isn't right for your line |
| 16:21.24 | enzo | it's surely some setting to inform asterisk when to take the call |
| 16:21.48 | lost_soul | orangey: http://list.georgialibraries.org/pipermail/open-ils-dev/2009-November/005314.html |
| 16:22.40 | *** join/#asterisk cmn (~carlos@host155-48-dynamic.16-87-r.retail.telecomitalia.it) |
| 16:22.58 | lost_soul | their stating there towards the bottom that twilio uses asterisk as it's backbone, I haven't yet found concrete evidence proving this but it wouldn't surprise me |
| 16:24.11 | orangey | lost_soul: this looks like what it does is creates a front-end. It itself is not a SIP provider or whatnot |
| 16:25.03 | lost_soul | orangey: yes, basically looks like your paying for their web interface and virtual phone numbers and such |
| 16:25.06 | enzo | [TK]D-Fender: Corydon76-dig is there some setting in asterisk to detect more easily when a ring is done ? |
| 16:25.20 | orangey | lost_soul: So, the hack here would be twilio to receive SMS / voice, which then distributes to asterix / whatever. The shame is that they don't let you just link in your SIP device and cut out the middle man, as it were |
| 16:26.52 | lost_soul | orangey: well, if you can indeed confirm it's asterisk their using. You could replicate it's workings I'm sure |
| 16:27.36 | *** join/#asterisk BANSAL (~bansal@117.199.124.71) |
| 16:27.41 | lost_soul | granted, it may very well not be worth the time it would take though. But would be very interesting to do |
| 16:28.42 | orangey | lost_soul: the real problem is getting a DID that accepts SMS / voice |
| 16:28.57 | orangey | if I could do that inexpensively, I think all the pieces are in asterisk to replace this indeed |
| 16:30.03 | orangey | oh. I might have hit a showstopper: "Twilio SMS does not support toll-free phone numbers, **Canadian phone numbers**, or phone numbers in other countries" |
| 16:30.10 | orangey | and I'm trying this for a canadian phone number.. |
| 16:32.13 | *** join/#asterisk Ta^3 (~tacvbo@189.146.182.146) |
| 16:32.52 | *** join/#asterisk CrashSys (~james@office2.vicidial.com) |
| 16:33.07 | CrashSys | How do I globally change the time zone that voicemail.conf is using? |
| 16:33.14 | lost_soul | orangey: lol, always something that ends up biting ya in the ass. Been there, done that |
| 16:33.19 | CrashSys | I know I can do it user by user but is there a way to do it globally? |
| 16:34.24 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 16:34.52 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.219.69.dsl.dyn.forthnet.gr) |
| 16:36.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:41.47 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 16:43.29 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 16:43.49 | Footman | WIMPy: with immediate=no and overlapdial=no or yes in zapata.conf, and span timing to 0 in zaptel.conf, the extension called is the first number of the callee number |
| 16:44.17 | russellb | what is this zaptel that you speak of |
| 16:45.19 | Footman | for example, we call 00123456789. The first 0 permits to go out the PABX. The second 0 is called by Asterisk via the SIP provider |
| 16:45.43 | Footman | maybe I can retrieve the numbers with Read function ? |
| 16:46.05 | Footman | russellb: this is /etc/zaptel.conf |
| 16:46.26 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 16:46.56 | bran | what does DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) mean when I try to run dahdi_cfg? |
| 16:49.03 | kn0x | anyone familair with ERROR[5783]: res_config_mysql.c:1456 mysql_reconnect: MySQL RealTime: Failed to connect database server phone_dat on localhost (err 2002). Check debug for more info. |
| 16:50.09 | kn0x | if i restart asterisk it says connected to database for X minutes.. so i am assuming this happens after certain ammount of time |
| 16:50.41 | Footman | WIMPy: OK, it works with the Read command ! :) But is there no way to have directly the callee number from the PABX ? |
| 16:53.01 | lost_soul | My situation is such that I would like to run asterisk on my router machine which runs openbsd. This router is also queueing traffic via pf with ALTQ. Without the queueing everything seems to work fine but with queueing enabled calls only connect one in three times roughly the rest of the time their going to the bulk queue. What I'm wondering is whether I need to use a sip proxy to resolve this matter. |
| 16:53.35 | lost_soul | I would prefer keeping this all on the same machine, but if I must seperate them I can do so |
| 16:54.06 | *** join/#asterisk cmn (~carlos@cl-3281.ham-01.de.sixxs.net) |
| 16:55.52 | jaytee | ~itsplist-us |
| 16:55.52 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:56.18 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 17:03.43 | kn0x | lol how is teliax and broadvoice more respected than bandwidth.com |
| 17:04.44 | fauxalliance | link2voip needs more respect... |
| 17:05.18 | Qwell | fauxalliance: never heard of it. |
| 17:05.41 | Qwell | business with "2" in their name...well... |
| 17:06.17 | fauxalliance | they have great rates, and an actual support team, not a loopy IVR |
| 17:06.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:07.37 | enzo | I use asterisk 1.4, I've seen last asterisk uses dahdi instead of zaptel. Does my X100P will work with last version of asterisk ? |
| 17:07.50 | kn0x | fauxalliance: and ipsec tunnel, not bad |
| 17:08.28 | kn0x | vitelity has lost all credibility with me |
| 17:08.58 | leifmadsen | enzo: X100P is not specifically supported or many years now -- it may or may not work |
| 17:09.33 | enzo | well, I'd like to know before upgrading :) |
| 17:10.00 | leifmadsen | like I said - it is not supported |
| 17:10.13 | leifmadsen | if you want a guarantee that your card will work, use modern hardware |
| 17:10.26 | tzafrir_laptop | enzo, should generally works just as well as it did with Zaptel |
| 17:10.43 | enzo | what card would i use to replace this old x100P leifmadsen ? |
| 17:10.46 | kn0x | ick x100p ick |
| 17:10.48 | tzafrir_laptop | (not to mention some minor bug fixes) |
| 17:10.51 | fauxalliance | sketchy CID at best... yet still better than my GS... |
| 17:11.05 | tzafrir_laptop | That is to say: assuming you don't use a kernel < 2.6.9 |
| 17:11.20 | kn0x | enzo: sangoma b600 |
| 17:11.33 | enzo | it's sold by digium kn0x ? |
| 17:11.41 | kn0x | sangoma |
| 17:12.06 | leifmadsen | enzo: TDM400P with 1 FXO module |
| 17:12.27 | kn0x | digium doesnt have any fixed configuration analog cards anymroe |
| 17:12.40 | Qwell | why would you want a fixed config card? |
| 17:12.49 | enzo | I have a TDM40B, but I guess it's a very old hardware also, no more officially supported ? |
| 17:13.01 | tzafrir_laptop | kn0x, "fixed configuration analog card". That's a nice name for it :-) |
| 17:13.12 | kn0x | haha |
| 17:13.52 | kn0x | wondering how long til realtime loses connection with mysql again -_- |
| 17:15.06 | *** join/#asterisk Lord_Rahl (~quassel@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 17:15.34 | Lord_Rahl | anyone know of a way to test latency between two points? I am going to deploy asterisk I need to make sure the network can handle it |
| 17:15.46 | enzo | I could replace my X100P and TDM40B with a sangoma b600 kn0x ? |
| 17:16.20 | Qwell | enzo: or a Digium TDM410 |
| 17:16.32 | kn0x | enzo: yes they're pretty cheap and are actually 4FXO + 1 fxs |
| 17:16.38 | leifmadsen | Lord_Rahl: ping? traceroute? |
| 17:16.45 | kn0x | in a single full-length pci |
| 17:16.57 | enzo | Qwell: it's quite expensive for a simple thing |
| 17:17.02 | kn0x | Lord_Rahl: ping |
| 17:17.43 | Lord_Rahl | leifmadsen: they work I was thing something more like packet island but open source |
| 17:17.50 | enzo | 1 fxs for line in and i could connect 4 telephones on the card for internal communication, things like that kn0x ? |
| 17:17.54 | wcselby | to have manager.conf changes take effect (asterisk 1.4.x), is it sufficient to run 'module reload manager' from the asterisk CLI or do I need to completely restart asterisk? |
| 17:18.36 | kn0x | enzo: there are some chinese operations that make cheaper than sangoma, but they're definately sketchy... sangomas is a respectable company |
| 17:18.46 | kn0x | enzo: opposite fxs are for stations |
| 17:18.57 | kn0x | fxo fore lines |
| 17:19.42 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 17:20.10 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 17:20.43 | enzo | kn0x: and you know other brands like sangoma ? I mean strong (and cheaper) cards |
| 17:20.48 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 17:23.30 | kn0x | enzo: R4FXO by rhino |
| 17:23.42 | kn0x | but sangoma porbably better support |
| 17:23.49 | enzo | ok |
| 17:24.15 | enzo | I think you're french kn0x, so you may know Free provider or Bouygues Telecom right ? |
| 17:27.03 | kn0x | haha nope im in chicago |
| 17:27.13 | enzo | ok |
| 17:27.56 | enzo | the fact is my card detect badly when a call arrives |
| 17:28.05 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:28.11 | enzo | I wonder if with a modern card it will work better ? |
| 17:28.43 | kn0x | it doesn't detect ring? |
| 17:29.01 | kn0x | perhaps there are settings for ring voltage if your card is not detecting ring |
| 17:29.27 | enzo | it detects rings, but at the 4th or more... |
| 17:29.54 | mort_gib | Hi, I have an issue with a Digium Wildcard B410P, or the ISDN line |
| 17:30.37 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 17:30.44 | mort_gib | All seems to be working but when I try to place a call on the channel it's not not going, not hangup or anything just nothing happens |
| 17:31.54 | mort_gib | Mind you, the loco Telco had ISDN lines and mobiles down for two hours today, so that might be the reason |
| 17:33.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:33.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:34.37 | t_dot_zilla | is there a way to enable sip debug on only certain calls instead of every call? |
| 17:35.11 | [TK]D-Fender | t_dot_zilla: ip/peer/all. take yuor pick |
| 17:35.55 | t_dot_zilla | what is the command to turn sip debug on an ip ? |
| 17:36.04 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:36.37 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:37.24 | [TK]D-Fender | t_dot_zilla: help sip |
| 17:48.30 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.137.144.mtnl.net.in) |
| 17:51.24 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:54.53 | *** join/#asterisk Ta^3 (~tacvbo@189.146.182.146) |
| 17:57.07 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 17:57.14 | *** join/#asterisk x-demon (xdemon@2001:ba8:1f1:f0b8:216:5eff:fe00:135) |
| 17:57.31 | x-demon | hi guys. I can't configure mysql for asterisk, it says wrong database |
| 17:57.41 | x-demon | i verified details - they're OK. |
| 17:58.22 | x-demon | [Jun 16 18:58:08] WARNING[6531]: res_config_mysql.c:943 config_mysql: MySQL RealTime: Invalid database specified: 'voiceone' (check res_mysql.conf) |
| 17:58.32 | x-demon | i'm trying to install voiceone |
| 17:58.41 | x-demon | voiceone works okay, but asterisk not... |
| 18:00.42 | x-demon | even if i change details on config, it still throws absolutely same error |
| 18:01.17 | Qwell | Did you check res_mysql.conf? pastebin it |
| 18:01.17 | Qwell | ~pb |
| 18:01.18 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:01.18 | Qwell | glares at infobot |
| 18:01.58 | x-demon | i also have mysql,voiceone in extconfig |
| 18:02.01 | x-demon | oh okay, one se |
| 18:02.02 | x-demon | c |
| 18:02.59 | x-demon | Qwell, http://pastebin.ca/1884533 |
| 18:04.08 | x-demon | argh! |
| 18:04.12 | x-demon | got it |
| 18:04.46 | x-demon | [voiceone] |
| 18:04.50 | x-demon | not [general] |
| 18:05.28 | x-demon | wel, not it gives me connection refused... |
| 18:06.41 | x-demon | after setting up socket it works okay... |
| 18:09.49 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 18:12.50 | x-demon | Qwell, well i configured everything, but asterisk still refuses connections |
| 18:13.21 | *** join/#asterisk otavio (~otavio@debian/developer/otavio) |
| 18:13.57 | otavio | Hello; what can cause the call to hungup after 12 to 15min of duration? |
| 18:14.09 | Qwell | lots of thigns |
| 18:14.17 | Qwell | show logs of it happening |
| 18:19.58 | otavio | Qwell: [Jun 16 14:09:28] NOTICE[29308] chan_sip.c: Peer 'Vono' is now UNREACHABLE! Last qualify: 34 |
| 18:20.01 | otavio | [Jun 16 14:09:46] NOTICE[29308] chan_sip.c: Disconnecting call 'SIP/Vono-0000004f' for lack of RTP activity in 31 seconds |
| 18:20.26 | *** part/#asterisk ruben23 (~unit41@202.137.112.11) |
| 18:23.18 | otavio | What is RTP activity? |
| 18:23.25 | Qwell | media |
| 18:23.51 | Qwell | it means you aren't receiving anything from the other side. likely because of a firewall issue. |
| 18:24.18 | [TK]D-Fender | Qwell: I doubt that they'd stay on a call without hearing you talk for 15 minutes :0 |
| 18:24.30 | Qwell | stupid firewall that closes the port |
| 18:24.33 | [TK]D-Fender | Qwell: I'd bet on silence-suppression... |
| 18:24.37 | Qwell | not uncommon |
| 18:25.06 | otavio | [TK]D-Fender: humm but why this happens with 12 to 15min only |
| 18:25.22 | otavio | [TK]D-Fender: it is very rare to have an issue before it |
| 18:26.32 | otavio | Qwell: any idea? |
| 18:26.52 | *** join/#asterisk PMantis (~sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 18:26.54 | bran | how do I actually setup my Polycom 335 with freepbx? |
| 18:26.58 | otavio | [TK]D-Fender: in fact I was talking with a customer when the call hangup |
| 18:27.15 | otavio | [TK]D-Fender: but this happens only using SIP trunk; using PSTN it works fine |
| 18:28.18 | [TK]D-Fender | bran: There is a wonderful Administrators Guide no their site. Go get it |
| 18:28.37 | bran | going... |
| 18:29.06 | leifmadsen | holy crap, did you know in (at least 1.6.2) you can do: |
| 18:29.11 | leifmadsen | [globals] |
| 18:29.14 | leifmadsen | foo=bar |
| 18:29.28 | leifmadsen | jimmy=${GLOBAL(foo)}/pop |
| 18:29.32 | leifmadsen | <PROTECTED> |
| 18:29.39 | leifmadsen | apparently that works... I never expected it to :) |
| 18:29.51 | Qwell | neat |
| 18:29.53 | PMantis | What are the possible reasons for this: "Command 'module load cdr_addon_mysql.so' failed." |
| 18:29.56 | devmod | Somehow I got to a state that whenever I receive a call I see "Setting the marker bit due to a source update" constantly spewing on the console before dropping the call. Any idea why this happen? |
| 18:30.04 | Qwell | PMantis: what is the line immediately before that? |
| 18:30.27 | PMantis | Qwell, "Unable to load module cdr_addon_mysql.so" |
| 18:30.47 | Qwell | and before that? |
| 18:31.09 | PMantis | Qwell, Me typing the command. :) |
| 18:31.26 | Qwell | so turn up verbosity/debug |
| 18:32.27 | otavio | [TK]D-Fender: any idea how to check about silence supresion? |
| 18:33.06 | PMantis | Qwell, Hmmm, such an obvious choice. Is there a CLI command to reload asterisk.conf/ Not sure that modules reload will work. |
| 18:33.19 | bran | [TK]D-Fender: there's no specific FreePBX info in this guide :( |
| 18:35.44 | otavio | [TK]D-Fender: VAD is disabled on the phone |
| 18:36.06 | PMantis | bran, Likely someone in #freepbx has configured one before. |
| 18:36.06 | leifmadsen | PMantis: asterisk.conf has to be reloaded with a 'core restart now' I'm pretty sure |
| 18:36.44 | PMantis | leifmadsen, I was afraid of that. Gotta wait till there's no calls. |
| 18:36.51 | leifmadsen | aye |
| 18:37.15 | PMantis | Qwell, What's a good debug level? looks like the default is 3? |
| 18:37.48 | PMantis | IOW, does it go up to 10, 64, 255, 65535... ? |
| 18:37.57 | Qwell | couple billion |
| 18:37.57 | leifmadsen | 5 |
| 18:38.04 | PMantis | lol |
| 18:38.10 | leifmadsen | the answer really is 5 |
| 18:38.11 | PMantis | I was gonna try 10 |
| 18:38.13 | *** join/#asterisk Gek_ (~Gecko@rhino.biggexpress.com) |
| 18:38.14 | leifmadsen | 10 is fine |
| 18:38.17 | leifmadsen | use whatever you want :) |
| 18:38.53 | wcselby | i use 1, 3, 6, or 10 depending on what I want to see |
| 18:39.05 | wcselby | and 0 if don't want to see anything |
| 18:39.15 | Gek_ | afternoon guys. I'm doing my first asterisk (asterisknow) setup just to play it and try to do some intergration into my existing callmanager system. |
| 18:39.18 | leifmadsen | if you grep through, there is I think like 2-3 messages above 4 |
| 18:39.20 | leifmadsen | pabelanger: would remember |
| 18:39.32 | x-demon | is warnings about nonexistant columns in mysql tables critical? |
| 18:39.34 | Gek_ | before doing so, I believe that I need to get sccp working... |
| 18:39.47 | Gek_ | i'm having difficulties finding exactly what I need to get/do. |
| 18:39.50 | *** join/#asterisk gavimobile (~user@bzq-84-108-29-62.cablep.bezeqint.net) |
| 18:40.00 | Gek_ | can someone please give me a little direction? |
| 18:40.06 | Qwell | Gek_: Asterisks SCCP support doesn't include acting as an endpoint. |
| 18:40.10 | pabelanger | leifmadsen: 15 |
| 18:40.16 | gavimobile | where can I find the CREATE DATABASE script on the iso image of asterisknow? |
| 18:40.27 | gavimobile | I totally screwed up my mysql databas |
| 18:40.35 | Qwell | gavimobile: /usr/src/freepbx/ |
| 18:40.38 | Qwell | somewhere |
| 18:40.43 | gavimobile | qweel |
| 18:40.44 | gavimobile | thanks |
| 18:40.54 | PMantis | gavimobile, Or: http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 18:41.07 | leifmadsen | putnopvut: ping? |
| 18:41.10 | Qwell | actually, it's right in the top-level SQL/ subdir there |
| 18:41.15 | Gek_ | Qwell, I'm not certain by exactly what you mean. |
| 18:41.27 | Qwell | Gek_: What do you want Asterisk to do with CCM? |
| 18:41.48 | Gek_ | I'd like to try and impliment a voicemail server instead of using unity... thats one thing |
| 18:41.55 | Qwell | use SIP |
| 18:42.00 | putnopvut | leifmadsen: PONNNNg |
| 18:42.09 | Gek_ | another is maybe do some call recording and other basic things like transfers... or maybe a calling queue |
| 18:42.18 | leifmadsen | putnopvut: just curious if I'm seeing the expected behaviour here -- caller sits in a queue and hears the message stating they are first in line. However, it does not state how long the average wait time is. The 2nd (or later) caller in the queue does hear that though. Is that expected? Do you remember if that was reported and fixed in a later version of 1.6.2? |
| 18:42.52 | Gek_ | Qwell, I'm trying not to reconfigure too much on the ccm side only to try and get asterisk to fit in a little |
| 18:43.03 | Gek_ | or to start experimenting |
| 18:43.04 | leifmadsen | I'm using a pre-1.6.2.0 version on this particular clients because it "just works" :) -- we're in the process of coordinating a system update to the latest 1.6.2.x though |
| 18:43.09 | Qwell | well, unless you use some protocol that both support, it's not going to work. |
| 18:43.19 | Qwell | so...use SIP. |
| 18:43.32 | putnopvut | leifmadsen: I don't believe the hold time announcement is based on position. I think it's based on how long the caller has been in the queue, coupled with the presence of previous callers in the qeuue. |
| 18:43.41 | *** part/#asterisk gavimobile (~user@bzq-84-108-29-62.cablep.bezeqint.net) |
| 18:43.57 | leifmadsen | putnopvut: ya, seems odd -- never plays for the first caller in the queue though (in my particular setup which is pretty straight forward) |
| 18:44.01 | putnopvut | leifmadsen: so, the caller in the front may not get hold time announcements because previous callers haven't contributed to the average hold time enough to give an accurate estimate. |
| 18:44.04 | leifmadsen | the 2nd caller always hears the avg wait time, but not the first |
| 18:44.09 | putnopvut | interesting. |
| 18:44.26 | putnopvut | Which version, in particular, are you looking at? |
| 18:44.28 | PMantis | Qwell, I finally have the debug level up - picked 10, then 100... still no additional output. http://pastebin.org/336556 |
| 18:44.38 | leifmadsen | this is in a test though -- perhaps you're right about the 2nd caller not being around enough? however we left it running for a while, let the 1st caller hang up, then the 2 -> 1 never hears the avg wait time (who did previously) |
| 18:44.47 | putnopvut | leifmadsen: And if possible, test with a later 1.6.2 and see if things start working as you expect. |
| 18:44.54 | leifmadsen | putnopvut: Asterisk SVN-branch-1.6.2-r198794M |
| 18:45.12 | leifmadsen | putnopvut: ya, that is the next step here -- I'm just going to tell them we'll see if it's still an issue in the later versions and if so, report a bug |
| 18:45.18 | leifmadsen | was just curious if there was a particular reason it worked that way |
| 18:45.24 | leifmadsen | like you said, might be fixed in later versions |
| 18:45.26 | putnopvut | leifmadsen: I don't remember any issue coming up about that. But that may have been something that was fixed during my extended stay in CCBS land :) |
| 18:45.33 | leifmadsen | yep not a problem |
| 18:45.46 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:46.19 | *** join/#asterisk QaDeS (~mklaus@p54A1B13F.dip0.t-ipconnect.de) |
| 18:46.23 | gloin | bit of a puzzler here: I've got an asterisk box talking to an older shoretel system over some sip trunks. I can dial out from asterisk to shoretel using a callfile, but when I log into an asterisk extension, that extension isn't dialing out |
| 18:47.18 | putnopvut | leifmadsen: looking at the 1.6.2 tip, it does appear at a cursory glance to do what you're saying, still. |
| 18:47.35 | leifmadsen | putnopvut: oh ok, then I can file an issue about that when we update and confirm that is the case |
| 18:47.51 | putnopvut | leifmadsen: cool. |
| 18:47.54 | leifmadsen | putnopvut: I appreciate you looking into that -- I know you're crazy busy with other things |
| 18:48.12 | putnopvut | It's all good. |
| 18:48.49 | putnopvut | When it comes to things like this, I always wonder whether the decision to do such a thing was intentional. |
| 18:48.55 | leifmadsen | ya same here |
| 18:49.01 | putnopvut | Could always be an oversight though. |
| 18:49.04 | leifmadsen | I can understand not saying how many callers though |
| 18:49.13 | leifmadsen | makes no sense to know how many people are behind you :) |
| 18:49.18 | leifmadsen | but the average wait time seems to make sense |
| 18:49.19 | putnopvut | Hell, looks like 1.4 has the same behavior. |
| 18:49.33 | leifmadsen | yes I'm first in line, but am I waiting for 2 mins or 45 mins? |
| 18:49.34 | leifmadsen | :) |
| 18:51.04 | leifmadsen | client: unique test cases come up as people use the system more, that's natural |
| 18:51.15 | leifmadsen | me: ya, I guess no one tests the user experience very much :) |
| 18:54.00 | gloin | ah, dialplan weirdness |
| 18:54.04 | gloin | but this is even worse |
| 18:54.35 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:55.03 | gloin | I go straight from "SIP/trunk-0001blahblah is ringing" to "SIP/trunkblah answered" to "Executing [h@macro-dialout-trunk:1]" |
| 18:55.18 | gloin | so I can get the phone to ring, but the moment it answers the hangup gets called |
| 18:55.25 | gloin | I'm not understanding why it's going to hangup |
| 18:58.01 | gloin | any ideas? |
| 18:58.20 | idespinner | delete everything from your dialplan |
| 18:58.24 | PMantis | gloin, Might need to pose the dialplan. |
| 18:58.27 | idespinner | except the importan parts |
| 18:58.29 | PMantis | errr post |
| 19:01.13 | gloin | argh, this will be messy, freepbx... |
| 19:02.24 | Qwell | gets a mop for the blood |
| 19:03.15 | Kobaz | gloin: you'll need to paste your sip debug |
| 19:03.21 | PMantis | gloin, It's also possible that a SIP message is telling it to disconnect. Have you ran 'sip set debug on' then tried? |
| 19:03.23 | Kobaz | gloin: the phone itself is probably hanging up |
| 19:03.38 | PMantis | ods at Kobaz |
| 19:03.44 | Kobaz | the 'h' exten runs when the sip session dies |
| 19:03.45 | PMantis | err s/ods/nods/ |
| 19:03.56 | Kobaz | it's not going to get called for no reason |
| 19:04.25 | gloin | ok, let's see what this does |
| 19:04.29 | Kobaz | unless you specifically call it with a Goto ot Gosub... but you shouldn't do that |
| 19:05.49 | gloin | whew |
| 19:06.12 | gloin | debug is putting it mildly |
| 19:06.19 | PMantis | Well, I did everything that I can think of to show that the module is there, but no go: http://pastebin.org/336573 |
| 19:07.29 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 19:11.10 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 19:11.41 | citywok | I've been struggling with my phones intermittently losing their registrations to *. Check out my log/sip.conf entry. All sip.conf entries are auto-generated so they are all identical to this: http://pastebin.com/Pb2GYN2q |
| 19:15.34 | PMantis | citywok, Probably has to do with the registration timeout period on the phones. Try reducing this. |
| 19:15.35 | *** join/#asterisk MiserySoft (~elende@94.197.51.12.threembb.co.uk) |
| 19:16.09 | gloin | PMantis and Kobaz: here's the debug output up to the point where it starts executing the hangup (but after it's begun the process): http://pastebin.com/9u7pQqQw |
| 19:16.11 | citywok | register more often or register less often? I htink i already have it set pretty agressively |
| 19:16.35 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 19:16.51 | citywok | Yea, i had set the registration period to 15 seconds somewhere along the way to try and combat this I think. |
| 19:17.57 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 19:20.16 | PMantis | gloin, From what I can see the problem begins on line 553... where the 10.1.xx.xx device sends "BYE". |
| 19:20.54 | gloin | PMantis: that's the softphone |
| 19:21.43 | gloin | I see what you mean though |
| 19:21.44 | gloin | hm |
| 19:21.52 | gloin | I can call into that softphone without error |
| 19:21.53 | PMantis | gloin, @ like 492, it's answered, then I see an OK and an ACK, then BYE. |
| 19:22.34 | zyphlar | ACK might mean that the phone is choking, see if it has anything lodged in its throat |
| 19:22.52 | citywok | PMantis: additionally, looking at historical data going back the last week it appears to happen 90% of the time DURING business hours. even most of the phones are always connected. |
| 19:24.11 | citywok | it's also site independent. I've got 4 locations and they all drop randomly. I'm kind of curious if it's a netowrking issue with my provider, but there is so much QoS going on it really shouldn't be able to happen. |
| 19:25.07 | PMantis | gloin, Is there a firewall between these devices? |
| 19:25.36 | gloin | PMantis: no, but there certainly might be some network ACLS |
| 19:25.43 | gloin | logs onto the switch |
| 19:25.55 | gloin | ack, which switch |
| 19:25.57 | gloin | brb |
| 19:25.59 | PMantis | gloin, Check to be sure the RTP ports are open. |
| 19:27.48 | PMantis | gloin, Note line 473, and 493 |
| 19:28.39 | PMantis | gloin, If audio can't be established, that will cause some SIP devices to terminate the call. |
| 19:28.44 | gloin | ah |
| 19:28.50 | *** part/#asterisk MiserySoft (~elende@94.197.51.12.threembb.co.uk) |
| 19:28.58 | gloin | what the heck is at that address? |
| 19:29.00 | gloin | weird |
| 19:29.15 | PMantis | gloin, LOL, I can't answer that. :) |
| 19:29.24 | gloin | working on it heh |
| 19:29.52 | PMantis | I'm fairly confident you have an ACL issue |
| 19:30.14 | gloin | seems like the best place to start for sure |
| 19:30.24 | gloin | thanks for the help (especially the sip debug on part) |
| 19:30.35 | PMantis | gloin, Certainly! |
| 19:32.18 | PMantis | citywok, I would suggest decreasing your registration timeout on SIP links, so they register more often. Losing registration is usually because it expires before another one is sent/received. |
| 19:32.45 | *** join/#asterisk utahsaint (~utahsaint@mail.ntegratedsolutions.com) |
| 19:33.28 | PMantis | Still not sure why I can't load the mysql cdr module... Using Ubuntu 10.04 packages, BTW: http://pastebin.org/336573 |
| 19:33.46 | citywok | registertimeout defaults to 20. how high is it safe to go? I don't see any comments about that in the wiki. |
| 19:35.09 | PMantis | citywok, 20 should be OK, that causes a frequent registration (every 20 seconds). So, you said it's the PHONES that are losing registration? Hard phones? |
| 19:35.32 | citywok | Aastra hard phones and Zoiper softphones both intermittently drop out. |
| 19:36.35 | citywok | i pulled the logs for the last week and it happens every night at midnight (i have a bunch of scripts that generate a lot of load at this time) -- but other than that it doesn't happen outside of normal business hours. |
| 19:41.34 | PMantis | citywok, OK, on the Aastra phones and in the Zoiper software, set the registration expiration to something lower. |
| 19:42.04 | PMantis | This will cause then to register with * more often. |
| 19:42.29 | citywok | Interestingly i Grepped my logs for Lagged (250 lines), and for Reachable (2344 lines) -- More often they reconnect without * seeing them as lagged. |
| 19:43.13 | PMantis | If * expected to hear from a phone every 5 minutes, and it doesn't hear from one in 6, the registration is lost. |
| 19:44.05 | PMantis | citywok, Of course if the network is congested and it keeps sending registration messages and they're lost, that's a different story altogether. |
| 19:44.33 | citywok | yea, so it's the other way around most of the time then. the phone has given up on the old registration and is attempting a new one thinking it expired? |
| 19:44.41 | x-demon | asterisk not starting :( |
| 19:44.49 | x-demon | 5060 port - connection refused |
| 19:45.33 | citywok | yea, but i don't _think_ we have any packets being lost. |
| 19:45.46 | citywok | as long as the phone QoS tags the packet the network will make sure the packet gets there. |
| 19:46.10 | PMantis | citywok, You *WANT* asterisk to receive a registration request before it expects one. |
| 19:47.42 | citywok | so if * by default expects registration every 60 seconds with qualify=yes, and the phone registration period is 15 seconds (which is probably a bit too aggressive), how come the phone is effectively re-registering under a new session (indicated by teh Reachable w/out a Lagged) |
| 19:48.28 | citywok | i can see that it happens when my server is under heavy load, so if the box itself is loaded then it appears to lose registrations very frequently. (backup script Tarring several GB of files) |
| 19:48.52 | PMantis | citywok, Ahhhhhhhh, ok |
| 19:49.06 | citywok | but outside of that there shoudl never be more than 5 or 10% load on the box (that happens at midnight every day) |
| 19:49.27 | citywok | we do encode all of the recorded calls in to MP3's, but that's niced to 19 so that shouldn't interfere with asterisk |
| 19:51.00 | PMantis | citywok, Registration is simply the phone's way of saying, "I'm here, at this address and this port, if you need me" |
| 19:51.54 | citywok | Yea. Which is important for the auto-dialer which Asterisk uses to originate calls for phones (we're a call center) |
| 19:51.59 | PMantis | citywok, It can do that every 5 seconds if it wants to, but it has to be frequent enough that * doesn't assume it died or was unplugged. |
| 19:52.24 | citywok | the other thing i've noticed is the phones themselves can go "No Service" on it, and you wont be able to make outbound or take inbound calls. |
| 19:52.37 | PMantis | citywok, Ahh, cool. I used to do work for a call center. |
| 19:53.01 | citywok | Yea, it can be interesting. the asterisk replacement system i built here is about 700x better than our old Inter-Tel digital system. |
| 19:53.26 | citywok | 2 full racks of telephone gear to a pair of 1U servers (primary and standby) |
| 19:53.46 | PMantis | citywok, Yup. The one I build replaced an Interactive Intelligence system - the license fees were WAY too high. |
| 19:54.10 | citywok | If the phone has gone No Service, that seems to indicate the phone itself lost the registration if asterisk has a 10:1 ratio of Reconnects compared to the number of times it noticed a phone was gone. |
| 19:54.25 | citywok | am i understanding that right? |
| 19:54.46 | citywok | PMantis: no kidding. $6,000 for an ATM switch. We bought one on ebay for $200 and our dealer was pissed. lol. |
| 19:55.21 | citywok | i'm pretty sure our Inter-Tel deal cost several hundreds of thosuands of dollars, and i built an asterisk replacement with every feature we needed from the old system in less than 6 months. |
| 19:55.35 | PMantis | citywok, If the phone thinks there's no registration it's likely because it sent a registration request, and it never received a reply from *. WHY that happens is what has to be determined. Perhaps network congestion for the * port, system load, etc. |
| 19:57.38 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:58.19 | PMantis | citywok, So, it cost them half your annual salary + plus hardware.... and they don't have to keep paying. |
| 19:58.49 | *** join/#asterisk d00gster (~dt@94.98.25.54) |
| 19:58.55 | *** part/#asterisk d00gster (~dt@94.98.25.54) |
| 19:59.28 | citywok | yea, the 60 * $217 (Aastra 6757i) + the license for the Softphone was like a grand. |
| 19:59.40 | PMantis | citywok, Not to mention that I bet you learned a LOT in the process. :) |
| 20:00.11 | PMantis | citywok, 60 * $217 is more than a grand already. :) |
| 20:00.31 | citywok | hah, yea i hadn't worked with a shit ton of * features before this. We'd been using it as a media converter (we converted the T1s coming out of our InterTel system to SIP to cut our LD rate in half) |
| 20:00.50 | PMantis | citywok, LOL, nice |
| 20:00.56 | citywok | we also used that step to do all of our call recording, which we couldn't have on our old system without paying Inter-Tel $200,000 |
| 20:01.05 | PMantis | ugh |
| 20:01.06 | citywok | i did it with a $1000 server and $500 quad port T1 card. lol. |
| 20:01.23 | citywok | yea, they wanted 200 grand. I even built desktop screen recording :) |
| 20:01.39 | PMantis | Cool! |
| 20:01.51 | citywok | okay, so i should be looking at the * server then trying to figure out what is happening to the registartion responses. |
| 20:02.23 | *** part/#asterisk orangey (~orangey@d67-193-125-203.home3.cgocable.net) |
| 20:02.49 | PMantis | citywok, Yeah, you'll have to know when the phones are NOT registered and capture the packets just BEFORE that happens. :) |
| 20:03.12 | citywok | yea, story of my life. do the impossible. |
| 20:03.32 | citywok | either that or turn on tcpdump with a lot of disk space and keep a watchful eye to make sure it doesn't run me out of space. |
| 20:05.12 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 20:08.34 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 20:09.28 | wcselby | yawns |
| 20:09.57 | wcselby | so, i've got a box that when I sync it up with our ntp server, it's (gmt offset) hours behind |
| 20:10.10 | wcselby | this means I need to ..... update hardware clock? |
| 20:12.43 | *** join/#asterisk Shaaan (~Shaaan@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 20:13.17 | Shaaan | Hey is anyone around have a few questions on a redundant reliable tollfree did provider and a regular did provider im looking for about 10 LOCAL DIDS and 1 TollFree DID with 20 channels or so |
| 20:13.41 | *** join/#asterisk brandonf (~bran@vaoffice.inmotionhosting.com) |
| 20:13.50 | brandonf | not sure if this is * or freepbx, but having an issue with a couple specific phones on conference calls/meetme. I have both ops dialing into a meetme conference, but when my buddy tries to conference an outside number, the conference call goes active, but the original person can't hear the conferenced person (chan_dahdi.c: New owner for channel 1 is DAHDI/1-1 / app_meetme.c: Ooh, something swapped out under us, starting over / app_meetme.c: |
| 20:16.35 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:25.19 | [TK]D-Fender | checkout time, later all |
| 20:28.03 | citywok | Shaaan: a lot of people use voicepulse,flowroute, and vitelity |
| 20:28.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:28.14 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:29.47 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 20:31.26 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 20:39.35 | wcselby | anyone else running asterisk 1.4.32 right now that can help me test something? |
| 20:44.41 | wcselby | hmmmmmm |
| 20:44.56 | wcselby | i'm having this issue (https://issues.asterisk.org/view.php?id=17332), but on 1.4.32 release |
| 20:45.12 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:45.29 | wcselby | the issue is closed, as it was for a 1.6.2.8-rc1 candidate and resolved with 1.6.2.8-svn at the time. How do i reopen, or do I just create a new issue? |
| 20:47.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:47.21 | russellb | new issue |
| 20:48.07 | *** join/#asterisk Godfather_ (~aaaa@135.Red-88-11-95.dynamicIP.rima-tde.net) |
| 20:48.12 | Godfather_ | hi |
| 20:48.12 | *** join/#asterisk Tha_MAol (~mo@modemcable211.153-57-74.mc.videotron.ca) |
| 20:48.16 | wcselby | russellb - creating now |
| 20:50.24 | t_dot_zilla | i have set atxfer => # and blindxfer => ## i cannot blind transfer, it automatically does an attended transfer then looks for the extension beginning with '#' |
| 20:50.37 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 20:50.47 | t_dot_zilla | is there a problem with my setup? |
| 20:51.16 | Tha_MAol | hello. my iax2 trunk is on-line, (( 1 iax2 peers [1 online )), cli Verbosity at 10, I dial the number from cell, get busy signal, no action CLI |
| 20:51.26 | Tha_MAol | all sip extensions work fine and call each other |
| 20:51.32 | Tha_MAol | not a firewall issue. |
| 20:52.02 | Tha_MAol | extensions from different ip addresses |
| 20:52.34 | Tha_MAol | what am I doing wrong? |
| 20:53.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.56 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 20:55.01 | cusco | hi |
| 20:55.10 | cusco | in .call files, can I set Channel: Local/blah |
| 20:55.10 | cusco | ยป? |
| 20:56.08 | [TK]D-Fender | cusco: Yes |
| 20:56.45 | Tha_MAol | something with my provider perhaps? |
| 20:58.07 | *** join/#asterisk Ta^3 (~tacvbo@189.146.182.146) |
| 20:58.09 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:58.45 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:59.50 | Tha_MAol | i am not sure what to do next, any pointer would help |
| 20:59.51 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 21:00.24 | kfife | does anybody know a way to generate a list of all ASTDB's? |
| 21:00.46 | [TK]D-Fender | kfife: database show |
| 21:01.07 | kfife | right. THat's if I know the name of the ASTDB. |
| 21:01.17 | kfife | what if I want to see all families |
| 21:01.24 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 21:01.32 | [TK]D-Fender | kfife: database show |
| 21:01.36 | kfife | hmmmm |
| 21:01.44 | kfife | I see |
| 21:01.53 | kfife | got it. |
| 21:03.09 | Tha_MAol | my aix trunk configuration http://pastebin.com/2zbRtFFZ |
| 21:03.23 | Tha_MAol | iax!@#$%^^&* |
| 21:03.34 | kfife | And if I have thousands of name value pairs, and I want to find just the family names, I'd have to spool this into a file and grep it? |
| 21:03.51 | wcselby | russellb - https://issues.asterisk.org/view.php?id=17515 thanks |
| 21:05.20 | cusco | and in a .call file I can: Set: var=bla |
| 21:05.28 | cusco | several times, mre than one var |
| 21:05.28 | cusco | right? |
| 21:06.17 | citywok | kfife: DATABASE SHOW <TREE> |
| 21:06.31 | cusco | nvm |
| 21:07.01 | kfife | citywok: I LOVE your nick |
| 21:07.20 | citywok | did you need a flight? /me rotates sign. Cityairlines, how cna i help you? |
| 21:08.26 | kfife | citywok: Citybank makes citiInvestments-needs government bailout :-)\ |
| 21:09.31 | kfife | [TK]D-Fender: nothing like select distinct (family) from AstDB? :-) |
| 21:09.44 | citywok | kfife: read what i just told you :) |
| 21:12.07 | kfife | citywok: for me I observe that spews out all 10,000+ name value pairs in my AstDB, leavign those orphan families with just a few name/value pairs awash in a sea of other crap? |
| 21:12.37 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:12.40 | citywok | if you type in the name of the tree you are looking in, you will only get the pairs for that one tree |
| 21:12.51 | [TK]D-Fender | kfife: Using relational DB terms to a NON relations BDB.... |
| 21:13.06 | citywok | if you have that many orphaned keys you need to fix your code that is saving those pairs to make sure you remove them |
| 21:13.14 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:13.15 | [TK]D-Fender | kfife: You may now proceed to stab yourself with a rusty spork :p |
| 21:13.25 | kfife | [TK]D-Fender: I know. My oracle background... |
| 21:13.53 | citywok | i have like 100 orphaned values in one tree but i'm too lazy to go clean it up. from when i was writing a new piece of code and it wasn't always removing the entries. |
| 21:13.54 | kfife | citiwok: certainly I can do an asterisk -rx "..." spool it into a file then grep it, but if there's a shortcut I'd like to know it. |
| 21:14.19 | kfife | Sounds like there "ain't" |
| 21:14.51 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 21:15.43 | citywok | <PROTECTED> |
| 21:15.55 | citywok | would give me a txt file with every key for tree VMCALLID |
| 21:16.17 | citywok | then you just need to pipe each of those txt file keys in to an asterisk -rx database delete command |
| 21:16.20 | *** join/#asterisk b14ck (b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 21:16.38 | citywok | (my keys are all 15 digits long, so it's 11-25) |
| 21:19.15 | *** join/#asterisk b14ck (b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 21:19.58 | kfife | citiwok: clever. |
| 21:22.46 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 21:23.44 | citywok | you're welcome :) |
| 21:24.09 | citywok | though now that you made me write that i guess tonight i'll run it on my own database and clean it up. lol |
| 21:24.23 | [TK]D-Fender | [17:13]<kfife>citiwok: certainly I can do an asterisk -rx "..." spool it into a file then grep it, but if there's a shortcut I'd like to know it. <- plenty of BDB libs out there. its a file. Deal with it :) |
| 21:25.33 | citywok | [TK]D-Fender: can you modify it with * running or do you need to stop it? if you want to modify the db directly that is. |
| 21:25.55 | *** join/#asterisk rene- (~rene@189.221.120.191.cable.dyn.cableonline.com.mx) |
| 21:25.58 | rene- | hello guys |
| 21:26.22 | kfife | [TK]D-Fender: Can I manipulate the BDB files on disk outside of asterisk? Where does ast stick the files? |
| 21:27.18 | rene- | does anybody has a comment (good or bad) about TCAST Communications? I am planning on doing business with them but i would like to know about other people experiences with them |
| 21:27.40 | [TK]D-Fender | fkPerhaps you should look for something suspicioulsy like "astdb" |
| 21:27.47 | [TK]D-Fender | kfife: fPerhaps you should look for something suspicioulsy like "astdb" |
| 21:30.55 | kfife | Got it. Thanks. For migration purposes, does /var/lib/asterisk/astdb contain the entirety of the data structure? In other words, can I just drop that file form the losing instance to the gaining instance? (assuming identical versions of course) |
| 21:31.04 | kfife | [TK]D-Fender: ^^^^^^ |
| 21:31.23 | citywok | kfife: yes |
| 21:31.34 | kfife | [TK]D-Fender: citywok: thanks! |
| 21:31.39 | citywok | and it took a 3 line perl script to run through the txt file and remove all the entries. thanks for giving me the ambition to do it. |
| 21:31.53 | kfife | citywok: :-) |
| 21:32.14 | PMantis | Ahhh, perl. love it |
| 21:37.39 | *** join/#asterisk QaDeS (~mklaus@p54A1B13F.dip0.t-ipconnect.de) |
| 21:40.40 | kfife | Is there a way to interrogate whether there is a marked user in conference number X? |
| 21:41.06 | citywok | what's a marked user? |
| 21:41.12 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 21:41.21 | kfife | AFAIK, a marked user is the conference leader. |
| 21:41.32 | kfife | specified by option "A" |
| 21:41.58 | kfife | If option W is invoked, conference participants do not "meet" until marked user arrives. |
| 21:46.44 | kfife | citywok: I'm having nick envy. That's clever. |
| 21:46.50 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 21:49.44 | *** join/#asterisk guilhermebr (~Guilherme@189.63.81.153) |
| 21:49.48 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:50.37 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 21:51.46 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 21:55.04 | t_dot_zilla | i have set atxfer => # and blindxfer => ## i cannot blind transfer, it automatically does an attended transfer then looks for the extension beginning with '#' |
| 21:55.14 | t_dot_zilla | is their a problem with my setup? |
| 21:57.11 | [TK]D-Fender | t_dot_zilla: you're dialing to slow chances are. Or should pick another patterns |
| 21:57.17 | [TK]D-Fender | t_what phones are you using? |
| 21:58.02 | t_dot_zilla | polycoms, and i'm pressing ## very quickly |
| 21:58.33 | [TK]D-Fender | t_dot_zilla: Then you should be SHOT for even trying to use DTMF transfers |
| 21:58.44 | t_dot_zilla | why is that |
| 21:58.58 | [TK]D-Fender | t_dot_zilla: that is the equivalet to buying a Ferrari and strapping a horse to it to DRAG it around town |
| 21:59.08 | [TK]D-Fender | t_Polycoms have REAL transfer buttons. |
| 21:59.12 | [TK]D-Fender | W_T_F. |
| 21:59.13 | [TK]D-Fender | Seriously |
| 21:59.21 | t_dot_zilla | i'm aware but not all of our customers have polycoms |
| 21:59.24 | t_dot_zilla | some just ATAs |
| 21:59.31 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 21:59.36 | [TK]D-Fender | t_dot_zilla: then I guess you should give a more comprehensive answer |
| 21:59.52 | [TK]D-Fender | t_And those ATA's ALSO have their own transfer features |
| 22:00.22 | t_dot_zilla | yes but most are regular phones that don't have transfer buttons |
| 22:01.19 | [TK]D-Fender | [17:59]<[TK]D-Fender>t_And those ATA's ALSO have their own transfer features <------- |
| 22:01.39 | [TK]D-Fender | t_dot_zilla: Perhaps you should read the manuals for the equipment you run. |
| 22:01.44 | t_dot_zilla | we'd like to bypass configuring the ATAs |
| 22:01.52 | t_dot_zilla | we'd like asterisk to handle the call features |
| 22:02.11 | t_dot_zilla | the less configuration per device the better |
| 22:02.33 | [TK]D-Fender | t_dot_zilla: Horrible idea. |
| 22:02.56 | [TK]D-Fender | t_dot_zilla: And there is NOTHING to "configure" for any of those |
| 22:03.14 | t_dot_zilla | yes you have to configure the devices to connect to asterisk, that should be it |
| 22:03.44 | t_dot_zilla | i'm just trying to figure out why ## will not do a blind transfer |
| 22:05.00 | [TK]D-Fender | t_dot_zilla: did you completely restart * after changing the shortcut to that? |
| 22:05.51 | [TK]D-Fender | t_dot_zilla: These devices don't need to be configured to transfer calls. Their own native capabilities are MUCH better than * doing it and doesn't screw with DTMF in IRV's, etc |
| 22:06.17 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 22:06.21 | t_dot_zilla | no i did not restart * and that is not an option |
| 22:07.09 | *** join/#asterisk pabelanger_ (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 22:08.27 | [TK]D-Fender | t_dot_zilla: Too bad. For features.conf changes, you need to reload some nasty stuff. "reload" won't cut it |
| 22:08.46 | citywok | do a silent restart |
| 22:08.50 | citywok | somewhere along the way all your calls will end |
| 22:09.01 | [TK]D-Fender | Or block your ears so you can't hear the screaming :D |
| 22:10.21 | t_dot_zilla | [TK]D-Fender: i suspected that is why changes were not taking when we edited features.conf |
| 22:10.57 | t_dot_zilla | we're also having problem with parking calls |
| 22:11.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:11.32 | t_dot_zilla | no matter how we edited the features.conf and extensions.conf files, no changes seemed to be taking |
| 22:12.45 | [TK]D-Fender | t_dot_zilla: extensiosn.conf does take |
| 22:13.13 | t_dot_zilla | ok, thanks, good to know |
| 22:13.27 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 22:13.48 | t_dot_zilla | i'll wait till everybody's gone tonight and do some tests on our office pbx |
| 22:20.38 | *** join/#asterisk Peaceful (~Peaceful@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 22:21.43 | Peaceful | Do all digium cards in a server have to operate off of one timing source? |
| 22:21.50 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 22:22.52 | Peaceful | ...or can you tell one card to use one of its ports for its timing source, while another card uses one of its ports for a timing source? |
| 22:23.16 | Peaceful | or is timing per-port? Can each port take its own timing? |
| 22:24.29 | [TK]D-Fender | Peaceful: yea |
| 22:24.43 | [TK]D-Fender | Peaceful: primary, secondary, tertiary, etc |
| 22:25.32 | Peaceful | So, if I have Qwest, Verizon, and 2 other carriers, I could plug one PRI from each carrier into a 4-port digium card and tell each port to take timing off of what is connected to it? |
| 22:25.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 22:27.01 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:47.48 | *** join/#asterisk ikariW (~ikariW@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 22:50.08 | *** join/#asterisk b14ck (~b14ck@s66-76-50-56.lfkncmta01.lfkntx.tl.sta.suddenlink.net) |
| 22:50.31 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:52.23 | kfife | If I've found a trivial documentation bug, does that rise to the level of opening up a bugtracker issue? |
| 22:55.57 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 22:58.07 | fenrus | kfife, bugs are bugs.. :) |
| 22:58.23 | fenrus | easy tickets are good for statistics |
| 22:59.25 | p3nguin | I've considering opening tickets for grammar and such in sample configs and docs, but never did it. |
| 22:59.39 | zyphlar | Peaceful: i think the documentation says that the card will only operate on one timing, so possibly not |
| 23:00.07 | zyphlar | i think primary/secondary/etc is "which one should the card use" not "how should this span be timed" |
| 23:00.15 | Peaceful | zyphlar: What if I have two cards, then? Can I set each card to a separate timing source? |
| 23:00.35 | zyphlar | i think so, simply because there are timing sync cables necessary to sync timing between cards |
| 23:00.56 | zyphlar | pure speculation though, you might want to read the Asterisk O'Reilly book or contact your card manufacturer |
| 23:01.04 | *** join/#asterisk otavio (~otavio@debian/developer/otavio) |
| 23:01.10 | Peaceful | I've read the dang book. It's vague as heck. |
| 23:01.20 | Peaceful | and googled. |
| 23:01.49 | Peaceful | but I haven't called Digium, so that's a good suggestion. I'll see if their guys can answer that. They've had difficulty answering most of questions in the past, though. |
| 23:02.06 | Peaceful | meanwhile, if anyone _knows_ the answer, do spit it out! |
| 23:02.53 | zyphlar | yeah Digium's support is great, they spent an hour with me on the phone the other day. |
| 23:03.13 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:03.15 | zyphlar | hint: call after normal biz hours Alabama time |
| 23:03.38 | zyphlar | less call volume, i was the only caller that night |
| 23:04.45 | zyphlar | though that was support, if you're calling sales might be different |
| 23:08.08 | gloin | ah well, my problem from earlier today turned out to be the most obvious thing: a faulty softphone |
| 23:08.25 | gloin | faulty/misconfigured |
| 23:11.50 | zyphlar | the problems are always less complicated in retrospect |
| 23:12.30 | gloin | yep |
| 23:13.42 | kfife | fenrus: thanks |
| 23:41.16 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:43.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:50.56 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.136.31.mtnl.net.in) |
