| 00:00.25 | *** join/#asterisk _af (unknown@69.26.168.58) |
| 00:01.01 | Sauron | Hum. That's annoying. |
| 00:01.33 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 00:02.59 | timecop | uh |
| 00:03.01 | timecop | except one problem |
| 00:03.07 | *** join/#asterisk Robot_ (~robot_@pool-71-113-23-113.sttlwa.dsl-w.verizon.net) |
| 00:03.08 | timecop | when you mute xlite/eyebeam on osx |
| 00:03.12 | timecop | it mutes the system volume |
| 00:03.13 | timecop | lol @ that |
| 00:04.21 | Nuxi | sounds like a mac |
| 00:05.07 | _af | is there a preferred OS for asterisk, silly question i know, but i must ask ;] |
| 00:05.19 | file[laptop] | Linux |
| 00:05.25 | cluecon | Debian, Solaris, Fedora, anything but windows. |
| 00:07.08 | dalabera | anyone here using Email to Fax?? if so in what format you send the file format on the email? |
| 00:07.10 | _af | mind if i continue to ask silly questions, im not sure about the question policy of the channel is why im asking, im used to being in channels where if your not that experienced in the channel topic, they beat you ;] |
| 00:07.16 | dalabera | anyone here using Email to Fax?? if so in what format you send the file format on the email? |
| 00:07.33 | [hC] | twisted: can you get that headset to work with a 7960? |
| 00:07.39 | `Sauron | Hum |
| 00:07.42 | [hC] | twisted: even more preferrably, with a lifter? |
| 00:07.43 | `Sauron | the spa1k1 is weird |
| 00:07.52 | `Sauron | if you have a message, it'll ring half a ring every so often |
| 00:07.55 | `Sauron | creepy |
| 00:08.23 | [hC] | you can shut that off |
| 00:08.28 | file[laptop] | dalabera: PDF or TIFF |
| 00:08.29 | Nuxi | hey, * runs on windows too. |
| 00:08.31 | `Sauron | hum |
| 00:08.33 | `Sauron | That'd be nice |
| 00:08.34 | `Sauron | how? |
| 00:09.04 | [hC] | i forget exactly where but i think its in regional or something.. at the bottom of the page there is a value for ring length on VMWI (voice mail waiting indicator) |
| 00:09.07 | [hC] | <PROTECTED> |
| 00:09.09 | [hC] | set it to 0 |
| 00:09.32 | Nuxi | http://www.asteriskwin32.com go cygwin! (not everything works and it's slow) |
| 00:09.57 | dalabera | file[] I'm trying to send an image from the windows imaging software and when it send the fax it's received in blank pages |
| 00:10.00 | `Sauron | VMWI Ring Splash Len: |
| 00:10.02 | cluecon | _af: we won't beat you too severely. |
| 00:10.04 | `Sauron | on the "User 1" page |
| 00:10.09 | `Sauron | yum. |
| 00:10.13 | _af | haha thanks cluecon ;] |
| 00:10.23 | timecop | fuck cygwin |
| 00:10.33 | timecop | if anyone actually had a clue they'd do a proper port to win32 |
| 00:10.36 | cluecon | Nuxi: he said preferred. windows is not preferred. |
| 00:10.38 | `Sauron | hC, thanks |
| 00:10.52 | Nuxi | cygwin brings the stability of windows to a unix like environment. lol ;) |
| 00:11.07 | timecop | another words makes windows crash and slow? |
| 00:11.20 | cluecon | Nuxi: that would actually work as a selling point to some people. |
| 00:11.26 | _af | anyone had much luck/experience with the intel 537 chipset modems? i bought a few to test out |
| 00:11.26 | Nuxi | or makes your unix-like environment crash and run slow. |
| 00:11.37 | timecop | that already does that by itslef |
| 00:12.06 | timecop | \\golf has been up for: 267 day(s), 18 hour(s), 29 minute(s), 51 second(s) |
| 00:12.10 | timecop | ^^ windows machine which is used daily |
| 00:12.14 | *** join/#asterisk tPO (~tPO@195.82.106.196) |
| 00:12.29 | cluecon | timecop: what version of windows and what does that machine do? |
| 00:12.30 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
| 00:12.38 | Nuxi | XP SP2 |
| 00:13.07 | `Sauron | hum |
| 00:13.10 | Weezey | g729 licenses... If I call into asterisk then make a call out to another SIP device which has 729, how many licenses do I need? 1 or 2? |
| 00:13.22 | timecop | XP pro, and it runs some cpu-intensive calculations and occasionally used as a file server |
| 00:13.26 | `Sauron | Man, it's like all the austinites fell off the planet today |
| 00:14.00 | cluecon | timecop: more details? sounds like it doesn't really do much. |
| 00:14.11 | timecop | Weezey: if you call from a sip device which sues 729 and you call into device which uses 729 and there is no codec translation going on, you need 0 licenses |
| 00:14.46 | Weezey | for some reason I tried that and it didn't work though. |
| 00:15.05 | timecop | another words if your voip provider accepts 729 and you connect from some sip phone that does 729 in hardware, it should just pass it through wihtout any conversion |
| 00:15.08 | timecop | cluecon: no, it does |
| 00:15.19 | timecop | cluecon: the calculations are important, so are the files which are copied to/from it. |
| 00:15.23 | Weezey | timecop: interesting. thanks |
| 00:15.36 | timecop | there's also a wince 3 development environment on it but thats only used like once a month |
| 00:15.46 | timecop | to do a new build of some PDA stuff used here |
| 00:16.02 | jacks | is there like a module for asterisk wich enables it to communicate with like a nokia mobile phone (trough a datacable) |
| 00:16.15 | timecop | no? |
| 00:16.20 | cluecon | ttimecop: the calcs are important? what kind of calcs? I can do important calcs on my cell phone and I can store files on a usb thumb drive. |
| 00:16.22 | Ariel_ | kernel-source is like 40mgs for CentOS argh |
| 00:16.53 | timecop | Ariel_: as opposed to kenrel sores for lunix 2.4.31 which is like 30megs? and 2.6 which is ~40? |
| 00:17.04 | timecop | cluecon: thats none of your business, really. |
| 00:17.14 | timecop | cluecon: fact is, its being used actively every day. |
| 00:17.21 | Ariel_ | I did not expect it to be that big |
| 00:18.26 | *** join/#asterisk Leomar (Leomar@p04.sa02.gti.procergs.com.br) |
| 00:18.36 | *** join/#asterisk \usr\sbin (~squirrelv@202.57.81.146) |
| 00:18.59 | \usr\sbin | hello everybody |
| 00:19.12 | \usr\sbin | how do i setup caller id on my * |
| 00:19.15 | cluecon | a windows machine being actively used for something more than a file server will quickly die due to inherent memory leaks in the windows runs. I should know. I can usually run my machine for about 2 weeks before it requires a reboot. |
| 00:19.19 | *** join/#asterisk Morex (~blah@host81-157-123-74.range81-157.btcentralplus.com) |
| 00:19.24 | Morex | Hello all |
| 00:19.43 | timecop | cluecon: thats because you're an idiot admin |
| 00:19.43 | Leomar | Hi ... anybody from Brazil ? |
| 00:19.44 | Morex | Can I ask a favour? |
| 00:19.56 | cluecon | timecop: what does admin have to do with a windows box having memory leaks? |
| 00:19.59 | Morex | I need someone from outside the UK to call two numbers, for testing... |
| 00:20.11 | Ariel_ | \usr\sbin, via sip iax2 or pots |
| 00:20.16 | Morex | Would be immensely grateful. |
| 00:20.24 | timecop | Morex: perhaps, what are they? |
| 00:20.40 | cluecon | timecop: tell me how it is possible to eliminate something that is tied directly into the OS. |
| 00:20.41 | \usr\sbin | hi ariel, any howtos? im tryin to test with my sip phone xten, would that be ok? |
| 00:22.13 | Ariel_ | \usr\sbin, use the power of the wiki:::http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID |
| 00:22.31 | \usr\sbin | thanks. ill try |
| 00:23.04 | timecop | cluecon: by having a clue |
| 00:23.08 | timecop | (which I guess you dont) |
| 00:23.19 | \usr\sbin | Ariel, Be aware that setting fromuser= in sip.conf will overide SetCallerID! |
| 00:23.28 | timecop | because I got a machine that's up for > 250 days only because of a 8 hour power outage at the time which killed all the building backup power. |
| 00:23.40 | \usr\sbin | how do i do that? is that the CallerID config? |
| 00:23.44 | timecop | which would otherwise been > 300-400 days |
| 00:23.57 | timecop | and its running windows |
| 00:24.02 | timecop | and its actively used daily |
| 00:25.09 | cluecon | timecop: you claim that you have a machine that is used for something more than a file server and it is able to perform the tasks required of it without a performance loss yet you can't specify what it is actually doing. Any machine will run forever if it doesn't have to do anything. |
| 00:25.30 | Weezey | timecop: you just saved me a buncha bucks. |
| 00:25.52 | jacks | hmm how do i get dftm input from a user into asterisk? |
| 00:26.03 | KrimHum | cluecon: You never asked. |
| 00:27.04 | cluecon | KrimHum: yes i did. to quote 'more details?' and 'what kind of calcs?' |
| 00:27.18 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
| 00:27.25 | *** join/#asterisk DannyF (~dannyf@dialup-82-96-28-103.rixtele.com) |
| 00:27.26 | timecop | Weezey: :D |
| 00:27.54 | timecop | Weezey: got it working in passthrough? |
| 00:27.54 | timecop | jacks: you jsut do, depends onyour device. |
| 00:28.21 | MRH2 | hey |
| 00:28.38 | cluecon | jacks: take a look at http://www.voip-info.org/wiki-DTMF to get started. |
| 00:29.53 | MRH2 | ne1 know if there is a way to enter a cli command from the dialplan in extensions.conf? |
| 00:30.49 | cluecon | MRH2: what command? |
| 00:31.44 | MRH2 | I would like it to execute CLI > agent logoff AGENT/2000 |
| 00:31.58 | *** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc) |
| 00:32.29 | cluecon | MRH2: how is the agent logging in? |
| 00:32.32 | [hC] | lol |
| 00:32.41 | [hC] | that is probably the most backwards approach ive ever seen to anything |
| 00:32.48 | MRH2 | agentcallbacklogin |
| 00:34.17 | MRH2 | (with password though) |
| 00:34.44 | cluecon | MRH2: who would be executing this command> |
| 00:36.01 | MRH2 | hoping to run it by calling an extension |
| 00:36.14 | cluecon | MRH2: take a look at http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin it covers logging off as well as logging on. |
| 00:37.24 | MRH2 | yep i have read it but I understand u need the password to logoff an agent |
| 00:37.46 | cluecon | so you want to do it without the password? |
| 00:38.09 | MRH2 | yep agent has password but logoff without password |
| 00:38.34 | tzanger | anyone here any good with VHDL/verilog? |
| 00:40.39 | cluecon | MRH2: http://www.voip-info.org/wiki-Asterisk+cmd+RemoveQueueMember might work. |
| 00:41.01 | MRH2 | ideally looking for something like... exten => 123, 1, magicapp (agent logoff AGENT/${AGENTBYCALLERID_CALLERIDNUM}} |
| 00:41.33 | timecop | tzanger: why |
| 00:42.06 | tzanger | just have some beginner questions |
| 00:42.29 | tzanger | I've done all manner of hardware design but nothing in VHDL or verilog and the language is confusing |
| 00:42.50 | MRH2 | I think i tried remove queue memeber .. will give it another go alothough agent is a queue member |
| 00:42.50 | *** part/#asterisk Morex (~blah@host81-157-123-74.range81-157.btcentralplus.com) |
| 00:43.16 | MRH2 | sorry not a queue member |
| 00:43.29 | *** join/#asterisk popooya (~popooya@364a7d22cb284cef.session.tor) |
| 00:44.30 | *** join/#asterisk esilberb (~er@209.227.180.13) |
| 00:44.55 | MRH2 | damn cluecon is gone :( |
| 00:44.59 | esilberb | hey all!!!!! did any1 ever deal with a fax passthrough issue?? |
| 00:45.51 | esilberb | ?? |
| 00:46.22 | newl | No, 1 is a lonely number. |
| 00:47.00 | esilberb | newl - do you get the feeling we're alone in here? |
| 00:47.17 | Ariel_ | esilberb, your not along. But what do you mean fax passthrough? |
| 00:48.08 | Ariel_ | esilberb, quickley I am about to leave to get some din din for the wife. |
| 00:48.14 | esilberb | i have a running asterisk which is connected as a client to a SIP server. When i connect a fax machine, i want the SIP session to use a special codec and not the one used for phone conversations |
| 00:48.23 | *** part/#asterisk tPO (~tPO@195.82.106.196) |
| 00:48.36 | *** part/#asterisk poillet (~pn@20-239-126-200.fibertel.com.ar) |
| 00:48.45 | esilberb | i looked this up on the Internet and there is SOME material on the subject but i couldn't figure it out |
| 00:48.52 | Ariel_ | setup a different context and account settings |
| 00:49.31 | esilberb | but how do i recognize that the machine is a fax? the zaptel driver can definately do that but i don't know how to configure asterisk |
| 00:49.41 | Ariel_ | if you user sip1 as peer for your dialing out copy it to sip2 change the codec there and then put your fax in a seperate context. |
| 00:49.58 | file[laptop] | you can't detect it's a fax. |
| 00:50.34 | esilberb | i can. the zaptel driver can do that but asterisk has to answer the call first and receive a DTMF tone indicating this is a fax |
| 00:50.41 | Ariel_ | esilberb, there is a sip add-on program on the wiki from nwdetec something that might help |
| 00:50.54 | esilberb | you got a URL? |
| 00:51.05 | Ariel_ | esilberb, no don't use it just read about it |
| 00:51.10 | Ariel_ | look it up on the wiki |
| 00:51.18 | esilberb | nwdetec you say? |
| 00:51.21 | file[laptop] | ah it's coming in via zaptel? |
| 00:51.25 | timecop | esilberb: what |
| 00:51.32 | timecop | esilberb: yo ucan tell zaptel to do fax detectio nthen use spandsp |
| 00:51.37 | timecop | on the "fax" extension |
| 00:51.38 | esilberb | yes and going out though the network usinf SIP |
| 00:51.41 | timecop | fax -> rxfax(foo) |
| 00:51.43 | timecop | or whatever |
| 00:51.44 | timecop | what |
| 00:51.50 | timecop | fax over voip = not happening |
| 00:51.54 | file[laptop] | or you can dial somewhere... like... wait for it... a SIP device! |
| 00:51.55 | esilberb | exactly - the fax extension! what' sthat all about? |
| 00:52.08 | timecop | esilberb: just use it, thats where asterisk will route shit it thinks is a "fax" |
| 00:52.10 | file[laptop] | if a fax is detected on a zaptel channel, it'll go to the fax extension... |
| 00:52.14 | timecop | if you enable fax detection in zaptel/zapata.conf |
| 00:52.20 | Ariel_ | you can sent a fax extension to any device |
| 00:53.41 | Ariel_ | sent/send |
| 01:00.46 | tzafrir_laptop | is there any RTFM on the meaning of the 'span' directive in zaptel.conf? |
| 01:01.28 | tzanger | tzafrir_laptop: that's already in zaptel.conf |
| 01:01.36 | tzanger | the description (a very good one, IMO) |
| 01:11.38 | tzafrir_laptop | OK, Debian's packaging managed to delete this file. Will be properly packed in the next version :-( |
| 01:16.01 | *** join/#asterisk drbrown_ (~chatzilla@oh-65-40-73-223.sta.sprint-hsd.net) |
| 01:16.33 | MRH2 | ne1 know when the 406P / 411P cards will be available? |
| 01:16.41 | drbrown_ | is anyone familier w/ the fxotune utility |
| 01:18.36 | drbrown_ | maybe not |
| 01:18.58 | drbrown_ | how is MRH2? |
| 01:19.13 | Weezey | timecop: what would I need g729 licenses for? routing into voicemail or to Zap channels? |
| 01:19.32 | MRH2 | tired , horny..the usual how is dr brown? |
| 01:21.50 | MRH2 | do u happen to know nething about these 406/411 cards? |
| 01:22.33 | MRH2 | as per http://www.lightreading.com/document.asp?doc_id=75224&site=supercomm&WT.svl=wire1_2 |
| 01:24.29 | MRH2 | well i gtg have fun |
| 01:24.38 | *** part/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
| 01:30.33 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:32.34 | timecop | Weezey: yes, for example if you had somethign that required codec conversion |
| 01:32.45 | timecop | Weezey: zap -> voip (g729) would require 1 license per channel |
| 01:34.50 | timecop | how many hours am I looking at for compiling gay323 on a gay6-450 with 128megs of memory? |
| 01:40.20 | *** join/#asterisk int19h (~tirath@60-240-230-104-vic-pppoe.tpgi.com.au) |
| 01:40.31 | Qwell | timecop: depends |
| 01:41.40 | timecop | depends on what fag |
| 01:41.42 | timecop | i just pasted the specs |
| 01:42.00 | Qwell | your mom |
| 01:42.13 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 01:42.13 | *** part/#asterisk int19h (~tirath@60-240-230-104-vic-pppoe.tpgi.com.au) |
| 01:42.17 | MikeJ[Laptop] | depends on if you can stop being a bitch long enough to type make ;) |
| 01:43.21 | *** join/#asterisk int19h (~tirath@60-240-230-104-vic-pppoe.tpgi.com.au) |
| 01:44.21 | *** join/#asterisk juice (1000@mo-65-173-76-96.dyn.sprint-hsd.net) |
| 01:46.17 | Optic | moo moo |
| 01:46.49 | *** join/#asterisk SwK (~SwK@12-219-156-206.client.mchsi.com) |
| 01:51.08 | *** join/#asterisk cryptnix (~cryptnix@24-231-209-5.dhcp.bycy.mi.charter.com) |
| 01:51.20 | cryptnix | whats AMP's user/pass on asterisk@home? |
| 01:51.25 | drbrown_ | are any of you guys familier with the fxotune command? |
| 01:51.33 | cryptnix | the admin/password combo don't seem to work |
| 01:51.38 | nn | huh? |
| 01:51.53 | drbrown_ | I am uncertain as to whether or not I am using it correctly |
| 01:52.09 | drbrown_ | fxotune -i 2 |
| 01:52.17 | Optic | moo moo |
| 01:56.57 | *** join/#asterisk hermie (~nick@24-236-167-53.dhcp.bycy.mi.charter.com) |
| 01:57.24 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 01:57.53 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:00.00 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 02:00.05 | SpaceBass | evening folks |
| 02:00.28 | MikeJ[Laptop] | maybe where you are... did you ever think, maybe everyone does not live where you do |
| 02:00.36 | MikeJ[Laptop] | wow.. how egocentric |
| 02:00.49 | MikeJ[Laptop] | ;) |
| 02:00.53 | SpaceBass | everyone doesnt live in Virginia? |
| 02:00.54 | SpaceBass | WHAT? |
| 02:03.49 | SpaceBass | anyone know about cisco PoE and a way around it? :) |
| 02:03.58 | Strom_C | a way /around/ PoE? |
| 02:04.37 | SpaceBass | i mean... I've seen re-wire diagrams for regular poe injectors... wondering if anyone has tried it |
| 02:07.55 | SwK | there are cisco poe to .3af adapters out there |
| 02:08.43 | SpaceBass | i already ordered a power cube |
| 02:09.01 | SpaceBass | but I have another 7940 coming with out a cube and have a spare injector (linksys)... was hoping to use it |
| 02:09.38 | Strom_C | well, it's worth a shot |
| 02:09.47 | Strom_C | just requires you making a new cable |
| 02:09.51 | SpaceBass | tried it, no dice... |
| 02:10.03 | SpaceBass | thats why i was wondering if anyone had done it successfully |
| 02:10.12 | SpaceBass | rather ambigious request |
| 02:10.35 | Strom_C | *shrug* |
| 02:10.46 | Strom_C | i've got a separate power adapter I use for my 7960 |
| 02:11.17 | Strom_C | perhaps one day I will get lucky and a catalyst switch with PoE will fall into my lap...but until that day the extra adapter is fine for me :) |
| 02:11.17 | SpaceBass | how is the 7960? wondering if I'm going to regret not getting it over the 7940 |
| 02:11.29 | Strom_C | having six line keys is very useful |
| 02:11.34 | SpaceBass | i bet |
| 02:11.46 | Strom_C | I've got two extensions on one asterisk box and four on a completely different box |
| 02:11.52 | SpaceBass | really? |
| 02:11.57 | Strom_C | yes |
| 02:11.57 | SpaceBass | <PROTECTED> |
| 02:12.22 | SpaceBass | can you DND one line and not the others? |
| 02:12.30 | SpaceBass | (or one box and not the other) |
| 02:12.43 | Qwell | Strom_C: sip? |
| 02:12.48 | Strom_C | of course |
| 02:12.50 | Strom_C | re sip |
| 02:12.55 | Strom_C | havent played with dnd |
| 02:13.18 | SpaceBass | speaking of sip... what is the process to get the firmware? |
| 02:13.36 | Qwell | get a ccp(?) account |
| 02:13.45 | Strom_C | my process is asking my company's network guys if they can download the firmware for me :) |
| 02:13.54 | Qwell | there is always that, yeah |
| 02:13.56 | SpaceBass | ccp? |
| 02:14.03 | Qwell | dunno, is that what its called? |
| 02:14.09 | SpaceBass | Strom_C ... i like that process |
| 02:14.18 | Optic | what's the process to get polycom firmware? :) |
| 02:14.24 | SpaceBass | is that the support agreement licence? |
| 02:14.56 | SwK | Optic: go ook thru the wiki its out there |
| 02:15.02 | Optic | ok |
| 02:15.20 | Strom_C | for polycom firmware, I think you have to wear the special robe, wave the special dead chicken, and then hop on one foot thirty times whilst repeating your MAC address in hex, octal, and Russian. |
| 02:15.23 | Optic | we have an ip500 for evaluation at work |
| 02:15.26 | Optic | it seems very nice |
| 02:15.45 | SpaceBass | Strom_C funny, I thought that was the cisco process |
| 02:15.51 | Optic | we have 14 sipura |
| 02:16.00 | SpaceBass | well, I guess the cisco also requires a dna sample |
| 02:16.25 | Strom_C | SpaceBass: if that's the cisco process, my coworkers are really good at doing it without getting up or changing clothes :) |
| 02:16.41 | Qwell | Strom_C: practice |
| 02:16.48 | SpaceBass | Strom_C well, I think if you are a cisco partner or have a support agreement its cake.. but if you are a home user its not easy |
| 02:16.51 | Strom_C | hehe probably |
| 02:17.00 | SpaceBass | and practice |
| 02:17.03 | SpaceBass | :) |
| 02:17.10 | Strom_C | SpaceBass: yeah, that's probably one of the benefits of being a huge cisco customer |
| 02:17.52 | SwK | Optic: if it already has SIP firmware you really dont need to change it up... just get the configs and admin manual from the link on the wiki |
| 02:17.56 | SpaceBass | Strom_C I used to work for a plat partner... was nice... everyone had cisco switches at home, etc |
| 02:18.14 | Strom_C | hehe |
| 02:18.31 | SwK | or paypal me $50 and its mac address and I'll have you a config in 15 minutes |
| 02:18.35 | SpaceBass | and now I work for a fortune 15 company as a drone with nothing to do with IT... so glad I got that MCSE under my belt |
| 02:18.50 | SpaceBass | SwK steep! |
| 02:19.10 | SwK | hey payday is still a few days off and I need smokes and booze |
| 02:19.37 | SpaceBass | roger the booze |
| 02:19.45 | SwK | besides I'm a whore, not a cheap whore ;) |
| 02:20.15 | SpaceBass | LOL |
| 02:21.56 | SpaceBass | arrruuggg I just want to play with this 7940... |
| 02:25.09 | loud | most people in here got a cco account, we are just lazy to download the firmware for you. |
| 02:25.25 | Qwell | cco...thats the one |
| 02:25.28 | SpaceBass | loud thats even more reassuring |
| 02:25.30 | SpaceBass | :) |
| 02:25.43 | SpaceBass | anyhoo with out the power cube, its a moot point |
| 02:25.47 | Optic | spacebass... as in the fish? |
| 02:26.02 | SpaceBass | as in the insturment |
| 02:26.38 | Qwell | fish would have been cooler |
| 02:27.04 | SpaceBass | hummm |
| 02:28.01 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
| 02:28.09 | mike^^ | if WaitExten(2) times out .. will it change the EXTEN? |
| 02:29.04 | mike^^ | no one awake who knows? |
| 02:29.09 | SpaceBass | what would it change to? |
| 02:29.18 | SpaceBass | <-- more curious than knowing |
| 02:29.26 | MikeJ[Laptop] | mike^^, your being a rude ass |
| 02:29.42 | mike^^ | im very sorry :( |
| 02:29.43 | mike^^ | im in a rush |
| 02:29.46 | mike^^ | im about to elave because of gf ;) |
| 02:29.48 | mike^^ | u know how that goes |
| 02:30.00 | loud | gf first, * later. |
| 02:30.02 | mike^^ | spacebase: in other words if i call my own phone and voicemail picks up within 2 secs if i choose the right exten |
| 02:30.03 | mike^^ | thats why |
| 02:30.04 | MikeJ[Laptop] | naw... I'm married ;) |
| 02:30.08 | mike^^ | loud:e xactly |
| 02:30.12 | Qwell | loud: you must be new here :P |
| 02:30.21 | mike^^ | i just wanna know if WaitExten(2) and no one pushes a button what it will do to EXTEN |
| 02:30.26 | MikeJ[Laptop] | Qwell, hehe |
| 02:30.33 | mike^^ | actually.. girls always first |
| 02:30.35 | mike^^ | or no love |
| 02:30.36 | mike^^ | ;) |
| 02:30.39 | MikeJ[Laptop] | nothing I can think of |
| 02:30.52 | MikeJ[Laptop] | o wait... |
| 02:30.54 | SwK | mike^^: what does show application waitexten tell you? |
| 02:31.03 | Qwell | why not like...try it |
| 02:31.04 | MikeJ[Laptop] | you said waitexten |
| 02:31.04 | SpaceBass | Funny... I had to write my wife a cover letter today just so she'd promise to spend 10 minutes learning "my geeky phone thingy" |
| 02:31.39 | SwK | SpaceBass: you should have held out for sex |
| 02:31.55 | SwK | seeing all us married guys know we never get any |
| 02:32.08 | SwK | (unless its newly weds) |
| 02:32.13 | SpaceBass | SwK nawww thats standard procedure stil... |
| 02:32.26 | SpaceBass | 1 year... still pretty 'active' |
| 02:32.34 | MikeJ[Laptop] | what, sex with SwK's wife? |
| 02:32.37 | SwK | yeah well that'll change soon |
| 02:32.39 | SwK | hah |
| 02:32.42 | SpaceBass | lol |
| 02:32.44 | SpaceBass | :) |
| 02:32.44 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:32.56 | SpaceBass | can the display on the 7940/60 display porn? |
| 02:33.01 | SpaceBass | xml maybe |
| 02:33.02 | MikeJ[Laptop] | yes |
| 02:33.13 | SwK | SpaceBass: why not.. i've seen it displaying penguins |
| 02:33.19 | SpaceBass | exten => 69,s,1 |
| 02:33.32 | loud | if you like black/white bmp porn |
| 02:33.59 | Qwell | 7970 |
| 02:34.05 | loud | ah |
| 02:34.14 | SpaceBass | best arguement for a 7970 ever |
| 02:34.22 | loud | do they support sip now ? |
| 02:34.31 | SpaceBass | I have 2 iPicasso phones with color touch screens... got them on ebay... apparently no firmware |
| 02:34.38 | MikeJ[Laptop] | for crappy color instead of crappy b/w porn? |
| 02:34.45 | Qwell | MikeJ[Laptop]: yes |
| 02:35.13 | MikeJ[Laptop] | that is somthing being married is much better for than ip phones |
| 02:35.13 | SpaceBass | how big is the display on that wifi cisco |
| 02:35.23 | MikeJ[Laptop] | size dosn't matter |
| 02:35.27 | SpaceBass | lol |
| 02:35.34 | MikeJ[Laptop] | at least that is what my wife keeps saying |
| 02:35.40 | MikeJ[Laptop] | hehe |
| 02:35.53 | SpaceBass | i believed that until i found "mr. pinky"... now we just dont discuss size |
| 02:35.59 | SpaceBass | too much info? maybe |
| 02:36.36 | MikeJ[Laptop] | hehe |
| 02:36.44 | SpaceBass | <cricket sounds> |
| 02:36.45 | cryptnix | Here's a dumb question but I've been reading the "Setup your own IP-PBX" and ... well when I try to connect using XLite it keeps saying Login Failed! |
| 02:36.45 | MikeJ[Laptop] | I was joking, so ummm.. yeah |
| 02:36.48 | SpaceBass | quiet in here |
| 02:37.08 | SpaceBass | cryptnix which one? Kerry Grahams (or what ever) |
| 02:37.10 | MikeJ[Laptop] | cryptnix, that's cuz you are not logging in correctly |
| 02:37.34 | cryptnix | oh c'mon guys lol |
| 02:37.51 | SpaceBass | cryptnix, dumb question, but I assume you are using Asterisk@home |
| 02:38.02 | cryptnix | Yep |
| 02:38.26 | SpaceBass | cryptnix, did you set up the extension in AMP? |
| 02:38.32 | cryptnix | Yes, I did |
| 02:38.38 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:38.41 | MikeJ[Laptop] | did you set it up right? |
| 02:38.41 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:38.44 | SpaceBass | whats the extension number? |
| 02:39.25 | dasuberdavid | oh |
| 02:39.28 | dasuberdavid | oh |
| 02:40.09 | SpaceBass | <cricket sounds> |
| 02:40.11 | cryptnix | SpaceBass: Yes, I did. |
| 02:40.14 | cryptnix | I mean |
| 02:40.21 | cryptnix | how much is there to screw up in AMP |
| 02:40.36 | Qwell | cryptnix: more then in * |
| 02:40.39 | cryptnix | haha |
| 02:40.45 | SpaceBass | cryptnix in my expirence its pretty stright forward, but some stuff is so "masked" that it can be cumbersome |
| 02:41.10 | cryptnix | the error thats on it says 403 Forbidden |
| 02:41.32 | SpaceBass | cryptnix in theroy all you need to do with x-lite is add the in and out server, the user and extenison (one in the same) and password(secret) |
| 02:41.59 | SpaceBass | sometimes x-lite doesnt take well to changes and needs to be restarted... noticed that on OS X more than windows |
| 02:44.05 | *** join/#asterisk joe (~jsauer@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 02:44.06 | cryptnix | ah |
| 02:44.11 | SpaceBass | clear as mud |
| 02:44.16 | cryptnix | i git 'r done |
| 02:44.22 | cryptnix | :-/ |
| 02:44.31 | SpaceBass | git er done! |
| 02:44.54 | cryptnix | Yeah |
| 02:44.56 | cryptnix | shexy |
| 02:44.59 | cryptnix | :) I love it |
| 02:46.58 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
| 02:47.06 | SpaceBass | whats the deal with video over sip? does it require anything server side or is it just added packets from client to client |
| 02:48.26 | cryptnix | heh |
| 02:48.32 | *** join/#asterisk mrproper_ (mrproper@61.95.55.251) |
| 02:48.36 | cryptnix | I want to hear the hold music it has on the system... |
| 02:49.10 | mrproper_ | when specifying load => chan_iax2.so in the modules.conf starting asterisk reports: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature |
| 02:49.27 | SpaceBass | cryptnix dont bother... get a really bad new age Cd... its the same |
| 02:49.37 | cryptnix | lmao |
| 02:49.41 | cryptnix | i still wanna listen to it |
| 02:49.42 | cryptnix | :D |
| 02:50.18 | SpaceBass | get xlite working? |
| 02:50.33 | SpaceBass | assign another exension and call it and put yourself on hold |
| 02:51.08 | MikeJ[Laptop] | mrproper_, remove all your modules and re-do the make install |
| 02:51.10 | mrproper_ | but if i dont specify in the modules.conf when it starts iax it gets: unable to bind port |
| 02:51.21 | cryptnix | another q SpaceBass ... whats the maintenance passwd on AMP |
| 02:51.36 | SpaceBass | cryptnix for amp? |
| 02:51.39 | MikeJ[Laptop] | mrproper_, then start asterisk back up |
| 02:51.40 | cryptnix | Yeah. |
| 02:51.41 | SpaceBass | user: maint pass: password |
| 02:51.54 | SpaceBass | typically |
| 02:52.25 | MikeJ[Laptop] | p ass word |
| 02:53.34 | *** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
| 02:53.59 | SpaceBass | ohhh just tried IAXcom for os x ... much nicer than x-lite |
| 02:55.17 | SpaceBass | now I have a use for my spare bluetooth headset |
| 02:55.33 | SpaceBass | and I'm off |
| 02:55.42 | outtolunc | was looking for one |
| 02:55.51 | SpaceBass | bluetooth headset? |
| 02:55.51 | outtolunc | spare=ebay <G> |
| 02:56.00 | outtolunc | hehe |
| 02:56.01 | outtolunc | yeah |
| 02:56.18 | SpaceBass | i had a first gen jabra that I didnt like so I got a sony ericson |
| 02:56.41 | SpaceBass | but my kitchen computer has bt and a softphone.... |
| 02:56.53 | outtolunc | there are a few $50 ones that 'supposedly' work nowdays.. i'll give one of those a try soon |
| 02:57.12 | SpaceBass | they were pricy for a while... havent checked lately |
| 02:57.43 | SpaceBass | now I'm going to ogle the flat screen i hung (un level) in my bedroom |
| 02:57.58 | outtolunc | nods <G> i was given a gift card for office depot, walked around for hours, all i could find was a double prices bt card |
| 02:58.09 | outtolunc | er priced |
| 02:58.18 | outtolunc | so that's what i got <G> |
| 03:02.39 | outtolunc | even the dvd-r's were double what i usually pay |
| 03:03.02 | outtolunc | i'd never 'shop' there, if i didn't have too |
| 03:03.36 | outtolunc | (as it was it took about 4 months for me to say screw it and go use the damn card) |
| 03:07.25 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3084.nb.aliant.net) |
| 03:08.49 | SpaceBass | office depot? |
| 03:08.54 | *** join/#asterisk mxmasster (~maxc@pool-71-106-161-147.lsanca.dsl-w.verizon.net) |
| 03:09.01 | SpaceBass | i'm always blown away by retail when I froogle... |
| 03:09.13 | mxmasster | shido: are you awake? |
| 03:09.35 | SpaceBass | but I'm some what impatient and willing to pay $10 more to but something in town if I can |
| 03:09.56 | cryptnix | hmm SpaceBass what about the weather feature? |
| 03:09.57 | cryptnix | ;-) |
| 03:10.11 | SpaceBass | cryptnix *61 |
| 03:10.14 | SpaceBass | or is it 62 |
| 03:10.26 | cryptnix | *61 |
| 03:10.28 | *** join/#asterisk mariogp (~caro@201.133.253.90) |
| 03:10.30 | cryptnix | but its just hanging up on me |
| 03:10.31 | cryptnix | hmm |
| 03:10.41 | SpaceBass | you getting nyc weather or nothing |
| 03:10.46 | cryptnix | nothing |
| 03:10.54 | SpaceBass | cryptnix have you used the Command line interface yet? |
| 03:11.05 | cryptnix | umm thats kinda vague |
| 03:11.12 | cryptnix | to asterisk? |
| 03:11.18 | SpaceBass | SSH into the asterisk box (requires downloading putty for windows) |
| 03:11.24 | cryptnix | yeah... |
| 03:11.32 | cryptnix | I got it :-) |
| 03:11.37 | SpaceBass | log in *cough* root and run 'asterisk -r' |
| 03:11.45 | SpaceBass | then call weather and see what scrolls by |
| 03:11.49 | SpaceBass | sometimes it can be telling |
| 03:12.52 | cryptnix | well |
| 03:12.55 | SpaceBass | with the weather script I've found it is processor intensive b/c of all the text-to-speach and it takes a second to do the FTP download... so somtiems you just have to wait on the line a second |
| 03:12.56 | cryptnix | *gulp* |
| 03:13.26 | cryptnix | all i see is asterisk1*cli |
| 03:13.46 | SpaceBass | ok, once you see that, call weather and watch the screen |
| 03:14.06 | cryptnix | oh |
| 03:14.09 | cryptnix | LOL |
| 03:14.15 | mariogp | hi i newbie in this i have a little question is posible with E1 of digium to have MFC/R2 works |
| 03:14.36 | anthm | generate 1x it in cron every hour into a raw sln |
| 03:14.46 | cryptnix | ok, what am i looking for? |
| 03:15.01 | cryptnix | ah, its a 2.5Ghz box with a gig of ram |
| 03:15.08 | SpaceBass | cryptnix copy and paste it into a form at www.pastebin.org |
| 03:15.25 | valence | ~pastebin |
| 03:15.26 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:15.27 | cryptnix | naw, i lie :) 300mhz with 98 megs of ram |
| 03:15.33 | cryptnix | its amazing seeing this thing do this |
| 03:15.38 | SpaceBass | oops pastebin.ca |
| 03:15.43 | SpaceBass | thanks jabot |
| 03:15.52 | cryptnix | wow |
| 03:15.57 | cryptnix | i'm not that incredibly new to this |
| 03:16.00 | cryptnix | -- Executing Answer("SIP/200-1042", "") in new stack |
| 03:16.00 | cryptnix | -- Executing AGI("SIP/200-1042", "weather.agi") in new stack |
| 03:16.00 | cryptnix | -- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi |
| 03:16.00 | cryptnix | -- AGI Script weather.agi completed, returning 0 |
| 03:16.00 | cryptnix | -- Executing Hangup("SIP/200-1042", "") in new stack |
| 03:16.02 | cryptnix | == Spawn extension (from-internal, *61, 3) exited non-zero on 'SIP/200-1042' |
| 03:16.04 | SpaceBass | ouch |
| 03:16.04 | cryptnix | -- Executing Macro("SIP/200-1042", "hangupcall") in new stack |
| 03:16.06 | cryptnix | -- Executing ResetCDR("SIP/200-1042", "w") in new stack |
| 03:16.08 | cryptnix | -- Executing NoCDR("SIP/200-1042", "") in new stack |
| 03:16.10 | cryptnix | -- Executing Wait("SIP/200-1042", "5") in new stack |
| 03:16.12 | cryptnix | -- Executing Hangup("SIP/200-1042", "") in new stack |
| 03:16.14 | cryptnix | == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/200-1042' in macro 'hangupcall' |
| 03:16.16 | valence | STOP |
| 03:16.18 | cryptnix | == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-1042' |
| 03:16.20 | cryptnix | fuck! |
| 03:16.22 | cryptnix | sorry |
| 03:16.24 | SpaceBass | cryptnix in the future use www.pastebin.ca please |
| 03:16.25 | SpaceBass | :) |
| 03:16.26 | cryptnix | musta hit ctrlv on the browser |
| 03:16.28 | cryptnix | :( |
| 03:16.30 | cryptnix | I did! |
| 03:16.39 | cryptnix | I hit CTRLV in the browser on accident |
| 03:16.42 | cryptnix | http://pastebin.com/300102 |
| 03:16.44 | cryptnix | lol |
| 03:16.49 | cryptnix | didn't mean to i swear |
| 03:16.54 | cryptnix | Went to fast ... |
| 03:17.21 | SpaceBass | dont give fuck all about dropping the damn f-bomb |
| 03:17.23 | *** join/#asterisk illumiantus_ (~illuminat@cpe-65-185-103-95.woh.res.rr.com) |
| 03:17.44 | illumiantus_ | can anyone help me set up a SIP trunk? |
| 03:17.48 | JerJer | fuck the fucking fuckers |
| 03:18.08 | cryptnix | illumiantus_: http://asteriskathome.sourceforge.net/handbook/index.html |
| 03:18.12 | SpaceBass | cryptnix assume your calling from exten 200 |
| 03:18.17 | cryptnix | Yep |
| 03:18.57 | SpaceBass | looks like its working... do you not hear a thing? |
| 03:19.06 | cryptnix | nope just immediately hangs up |
| 03:19.16 | Nivex | ooh I had a fun idea for weather. for those living near a decent sized airport, get a radio scanner that does AM and tune it to the automated weather for the airport. http://www.faa.gov/asos/map/map.htm |
| 03:19.26 | SpaceBass | cryptnix what version of *@hom |
| 03:19.27 | SpaceBass | e |
| 03:19.27 | Nivex | no text to speech drain and its realtime |
| 03:19.55 | SpaceBass | Nivex now, how is that as much fun |
| 03:20.09 | cryptnix | just downloaded it off of sourceforge as of yesterday |
| 03:20.10 | cryptnix | the iso |
| 03:20.10 | SpaceBass | how about www.weather.com ? faster... but still not as much fun |
| 03:20.37 | SpaceBass | cryptnix gotcha, the older .8 never worked right for me but the latest does... |
| 03:21.24 | valence | ~aah |
| 03:21.26 | jbot | aah is probably Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
| 03:21.43 | illumiantus_ | cryptnix: it's for a VoIP service provider and they did not provide me with ANY information other than my phone number and their IP address |
| 03:21.46 | SpaceBass | cryptnix take a look at /var/lib/asterisk/agi-bin/weather.agi |
| 03:21.54 | SpaceBass | and the festival-weather-script.pl |
| 03:21.58 | SpaceBass | in that same direcory |
| 03:22.06 | SpaceBass | illumiantus_ what service? |
| 03:22.23 | illumiantus_ | Centric Voice |
| 03:22.38 | cryptnix | SpaceBass: I'm looking. |
| 03:22.40 | SpaceBass | dont know it |
| 03:22.47 | SpaceBass | illumiantus_ see if |
| 03:22.47 | SpaceBass | /var/lib/asterisk/agi-bin |
| 03:22.48 | mrproper_ | When trying to dial from an IAX client to an extension with exten=> 100,Answer it doesnt answer and gives the response: Auto fallthrough, channel 'IAX2/205@205-2' status is 'UNKNOWN' |
| 03:22.48 | SpaceBass | oops |
| 03:23.02 | SpaceBass | illumiantus_ see if http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=26 helps |
| 03:23.13 | SpaceBass | its geared towards broadvoice |
| 03:23.50 | SpaceBass | cryptnix kind of ambigious place to send you, but maybe if you configure the FTP download stuff for your town it may work... duno... blind leading blind at this point |
| 03:24.06 | cryptnix | ah :/ I see |
| 03:24.10 | cryptnix | tis cool |
| 03:24.21 | cryptnix | if its going to eat my CPU ... I won't implement it. |
| 03:24.48 | SpaceBass | cryptnix I'm on a 500mhz with 256mb and 12 extensions... works fine for me |
| 03:24.58 | cryptnix | ah |
| 03:24.58 | SpaceBass | well, it works ok.. takes about 25 seconds somtimes |
| 03:25.01 | cryptnix | omg |
| 03:25.03 | SpaceBass | but if you are on a 2.5 ghz your fine |
| 03:25.28 | SpaceBass | but I at least hear the first "the current weahther...." then it takes about 20 seconds to FTP the file and festival it |
| 03:25.33 | JerJer | 2.5? |
| 03:26.20 | cryptnix | 2500 plus |
| 03:26.24 | cryptnix | amd me |
| 03:26.26 | cryptnix | :) |
| 03:26.39 | SpaceBass | seacrest out |
| 03:27.00 | Chuji | model name : Pentium MMX |
| 03:27.00 | Chuji | stepping : 3 |
| 03:27.00 | Chuji | cpu MHz : 166.590 |
| 03:27.06 | Chuji | That's all you need :) |
| 03:27.10 | SpaceBass | nice! |
| 03:27.15 | Chuji | That's my asterisk box |
| 03:27.28 | SpaceBass | Chuji how many extensions? |
| 03:27.42 | SpaceBass | Chuji your idea bout foward on busy to bv... genious I TELL YOU |
| 03:27.44 | valence | Use cron and curl to prefetch text file of weather, then modify weather script to reference file. |
| 03:28.35 | cryptnix | naw i lied |
| 03:28.35 | SpaceBass | or *cough* launchd |
| 03:28.37 | Chuji | check out my 'show translations for g726' |
| 03:28.40 | Chuji | <PROTECTED> |
| 03:28.41 | cryptnix | 300mhz/98 megs of ram |
| 03:28.48 | cryptnix | with a drive that is dieing |
| 03:28.48 | cryptnix | :) |
| 03:28.58 | SpaceBass | LOL! |
| 03:29.09 | cryptnix | :) |
| 03:29.12 | cryptnix | i'm such a loser. |
| 03:29.22 | SpaceBass | i lied too.. I'm on a 300mhz not 500... but at least I have a new drive |
| 03:29.23 | JerJer | Chuji: i doubt it |
| 03:29.24 | Chuji | SpaceBass : Only use two or thee extensions at one time |
| 03:29.25 | cryptnix | still wierd though why does it just hangup |
| 03:29.36 | cryptnix | oh well google will tell |
| 03:29.37 | cryptnix | ;) |
| 03:29.40 | Chuji | JerJer : Damin was close, but I got him beat out |
| 03:30.15 | SpaceBass | speaking of drives... thinking of implementiing a "guest" mailbox that is cleared every monday, is it as simple as deleting the files from the vm directory with a chron job? |
| 03:31.36 | JerJer | system type : Broadcom BCM947XX |
| 03:31.36 | JerJer | processor : 0 |
| 03:31.36 | JerJer | cpu model : BCM3302 V0.7 |
| 03:31.36 | JerJer | BogoMIPS : 199.47 |
| 03:31.44 | SpaceBass | where are voicemail files stored? |
| 03:31.52 | outtolunc | damn speedy <G> |
| 03:32.26 | SpaceBass | whats the cmd you guys are using? |
| 03:32.39 | Chuji | cat /proc/cpu |
| 03:32.40 | mariogp | is posible with E1 of digium to make works with MFC/R2 |
| 03:32.54 | SpaceBass | been on os x too long |
| 03:33.10 | Chuji | JerJer : that's running *? What does your show translations look like for g726 |
| 03:33.16 | SpaceBass | cupinfo on os x |
| 03:33.17 | cryptnix | /dev/hda2 5.1G 1.9G 3.0G 38% / |
| 03:33.24 | JerJer | Chuji: yes |
| 03:33.25 | cryptnix | heh |
| 03:33.30 | cryptnix | 6 gigger :-/ |
| 03:33.42 | JerJer | <PROTECTED> |
| 03:33.52 | Chuji | much faster than mine |
| 03:33.55 | mariogp | the asterisk pbx works great with macos x for intel |
| 03:33.56 | Chuji | <PROTECTED> |
| 03:34.19 | SpaceBass | mariogp darwin? |
| 03:34.23 | Qwell | wow, those are high |
| 03:34.39 | Chuji | p166mmx |
| 03:34.42 | mariogp | nop macos x 10.4.1 build beta |
| 03:34.58 | mariogp | in this case beta for developer |
| 03:35.02 | Chuji | Been running my home pbx for over a year though. Works fine |
| 03:36.22 | SpaceBass | ok... really going to bed now |
| 03:36.22 | Chuji | Only problem is MySQL is slowwwww |
| 03:36.23 | SpaceBass | mariogp what hardware? |
| 03:47.00 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
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| 03:47.20 | illumiantus_ | :( |
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| 03:50.18 | cryptnix | wow |
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| 03:50.18 | illumiantus_ | does anyone know how to do sip traces? |
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| 03:50.18 | dos000 | howdy |
| 03:50.18 | illumiantus_ | wait wtf |
| 03:50.19 | illumiantus_ | i hink i just got it to work |
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| 03:51.26 | dos000 | anyone can help with this .. i have 2 sip ata and i cant make one extension call the other for the life of me. They both are being registered with asterisk and all seems fine. just cant get them to dial each other. |
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| 03:51.31 | illumiantus_ | okay... i go it to wor |
| 03:51.37 | *** join/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com) |
| 03:51.42 | illumiantus_ | but it only works for outgoing calls not incoming cals |
| 03:51.42 | cryptnix | Well I appreciate it guys ... |
| 03:51.44 | illumiantus_ | calls* |
| 03:51.55 | illumiantus_ | I get a 503 on incoming calls |
| 03:52.22 | cryptnix | going to sleep ... i'll likely bother you all tomorrow sometime ... until then have a good night. |
| 03:56.26 | illumiantus_ | anyone have any ideas? |
| 03:59.00 | *** join/#asterisk kn0x (~super@adsl-66-73-198-246.dsl.chcgil.ameritech.net) |
| 03:59.09 | kn0x | drumkilla- i need your help |
| 03:59.25 | kn0x | do you have a copy of the Callback application? |
| 04:00.09 | kn0x | looks like drumkillas gone for the night |
| 04:01.16 | kn0x | anyone ever implement a callback application? i feel kinda lazy... |
| 04:05.02 | *** join/#asterisk [illuminatus] (~illuminat@cpe-65-185-103-95.woh.res.rr.com) |
| 04:05.18 | [illuminatus] | ok. I have 1/2 of this SIP trunk setup but I really need help with the incoming settings |
| 04:05.33 | timecop | lolz. |
| 04:05.41 | timecop | i think compiling h323 on k6450/128m was a bad idea |
| 04:05.45 | timecop | i hear hdd swapping like crazy. |
| 04:05.49 | timecop | after like 3 hours of compile. |
| 04:09.16 | mxmasster | i am trying to set a new callerid on my outbound calls... i am doing a SetCallerID(xxxxxxxxx) and then a dial |
| 04:09.20 | mxmasster | however it is not changing |
| 04:10.08 | dudes | mxmasster - SetCallerID((XXX) XXX-XXXX) |
| 04:10.24 | mxmasster | what about private numbers? |
| 04:10.41 | dudes | be more specific |
| 04:10.58 | mxmasster | how do i remove the callerid so it says "Private" |
| 04:12.01 | dudes | First off ... does your teleco provider allow you to do that |
| 04:12.01 | [illuminatus] | don't have a CID? |
| 04:12.10 | dudes | if they don't then you can't |
| 04:12.57 | mxmasster | they told me that they do... however, looking at the sip debug, i am definately passing it correctly. they must not allow it |
| 04:13.32 | dudes | do you have it set like I set above |
| 04:13.52 | dudes | because if it's not it probally won't pass ... also are you using a sample call file for testing? |
| 04:14.18 | dudes | i.e callerid: <(XXX) XXX-XXX> in your sample call file |
| 04:14.21 | mxmasster | i'm just calling my cell phone |
| 04:14.43 | dudes | try callerid: Private <(XXX) XXX-XXXX> in a call file to your cell |
| 04:15.01 | mxmasster | what do you mean, call file? |
| 04:15.23 | dudes | I can pm you a sample call file |
| 04:15.27 | [illuminatus] | why is it so difficult to setup a SIP trunk??? I can call out but I can't call in. When I call in I get this message. |
| 04:15.27 | [illuminatus] | <PROTECTED> |
| 04:16.05 | dudes | illuminatus - for starters sip is actually pretty much not cool |
| 04:16.10 | mxmasster | dudes: in my sip debug i have this |
| 04:16.12 | mxmasster | From: "Private" <sip:3103568026 |
| 04:16.19 | mxmasster | but it isn't being set |
| 04:16.29 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 04:16.29 | mxmasster | i'm thinking that it is not letting me control the callerid |
| 04:18.40 | dudes | I can set cid on a clients trunks |
| 04:18.46 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 04:19.10 | dudes | But if I didn't do it exactly like SetCallerID((XXX) XXX-XXXX) it didn't work |
| 04:19.14 | dudes | that's all Im saying |
| 04:19.23 | dudes | it will say it's passing it through but it doesn't |
| 04:19.24 | JerJer | dont' format it |
| 04:19.27 | JerJer | just put the number |
| 04:19.47 | [illuminatus] | dudes: our VoIP provider didn't provide me with *any* information other than their IP address and our phone number. I have the outbound part of the trunk working but I can't get the inbound working |
| 04:20.12 | JerJer | sounds like your VoIP provider needs to be fired |
| 04:20.14 | loud | call them. |
| 04:20.29 | [illuminatus] | they told me to call "asterisk support" |
| 04:20.31 | [illuminatus] | =/ |
| 04:20.38 | JerJer | ?! |
| 04:20.46 | JerJer | i presume this is a sip provider? |
| 04:20.51 | [illuminatus] | yeah... |
| 04:20.55 | mxmasster | dudes: i did it exactly as you said |
| 04:20.58 | mxmasster | it didn't work |
| 04:21.08 | dudes | illuminatus - you have two contexts in sip.conf |
| 04:21.16 | dudes | mxmasster - then you can't pass it yourself |
| 04:21.26 | JerJer | just register to their proxy |
| 04:21.33 | JerJer | and start calling your number with sip debug turned on |
| 04:22.05 | [illuminatus] | dudes: no i only have one context |
| 04:22.27 | dudes | I think you need one for inbound and one for inbound |
| 04:23.27 | JerJer | you are going to need one type=peer and one type=user |
| 04:23.46 | JerJer | unless you are lazy and want to succumb to the evil of type=friend |
| 04:23.49 | dudes | user for inbound and peer for outgoing |
| 04:25.19 | dudes | username, type, secret, qualify, nat, insecure, host, context (incoming) |
| 04:25.47 | [illuminatus] | according to them there is no password |
| 04:26.00 | [illuminatus] | type is set on both in/outbound |
| 04:26.10 | dudes | so you're on a trunk? |
| 04:26.14 | [illuminatus] | context is from-pstn |
| 04:26.15 | dudes | that uses host auth |
| 04:26.18 | [illuminatus] | yeah |
| 04:26.50 | [illuminatus] | and i have outbound working but not inbound |
| 04:26.54 | dudes | I've never setup inbound on a trunk before |
| 04:27.07 | [illuminatus] | to get outbound working i had to setup the registration string |
| 04:27.23 | [illuminatus] | but the string looks like it has 2 parts |
| 04:27.24 | dudes | register => u:p@addy |
| 04:27.45 | [illuminatus] | user@ip:pass:user@ip |
| 04:27.52 | JerJer | !? |
| 04:27.59 | dudes | that's two |
| 04:28.03 | [illuminatus] | where the fisrt user @ ip is inbound and the second is outbound |
| 04:28.04 | [illuminatus] | err |
| 04:28.08 | [illuminatus] | vice sersa |
| 04:28.09 | [illuminatus] | anyway |
| 04:28.09 | JerJer | ?! |
| 04:28.17 | [illuminatus] | my egistration string is <number>@ip |
| 04:28.21 | dudes | then make another context in sip.conf |
| 04:28.25 | JerJer | register => bob:abc123@proxy-1.nufone.net |
| 04:28.30 | [illuminatus] | that is hw i got it working |
| 04:28.30 | JerJer | simple |
| 04:28.37 | dudes | and register your inbound seperate to the outbound |
| 04:28.49 | [illuminatus] | and my kyboard is laging so sorry for the typos |
| 04:29.04 | [illuminatus] | JerJer - there is no password |
| 04:29.07 | JerJer | you don't register to the proxy for inbound calls |
| 04:29.09 | JerJer | then dont put one |
| 04:29.25 | JerJer | er |
| 04:29.32 | JerJer | you don't register to the proxy for outbound calls |
| 04:29.38 | JerJer | proxy authentication takes place on the INVITE |
| 04:29.42 | dudes | according to what you put above one uses a pass |
| 04:30.01 | dudes | user@ip:pass:user@ip |
| 04:30.20 | [illuminatus] | that's how i saw people set it up for broadvoice |
| 04:30.20 | JerJer | that makes no sense |
| 04:30.26 | timecop | heh |
| 04:30.26 | [illuminatus] | but that'snot what iam usin |
| 04:30.28 | timecop | still compiling h323 |
| 04:30.30 | timecop | on k6450 |
| 04:30.33 | timecop | been several hours. |
| 04:30.41 | timecop | like 4+ hours. |
| 04:30.41 | [illuminatus] | 12143294838@209.120.255.14 is my registration string |
| 04:30.45 | dudes | probally is reg user@ip and reg user:pass@ip |
| 04:30.56 | [illuminatus] | I ahd to have that to call outside |
| 04:31.02 | [illuminatus] | but I can't call inside |
| 04:31.08 | JerJer | [illuminatus]: add register => in front of it |
| 04:31.26 | JerJer | and put in the general section of sip.conf |
| 04:31.35 | [illuminatus] | register=12143294838@209.120.255.14 is what is in sip.cof |
| 04:31.44 | JerJer | good |
| 04:31.50 | JerJer | and it doesn't register? |
| 04:31.52 | [illuminatus] | and for some reason i hve to hav the 1 in there |
| 04:31.55 | dudes | you have a user right? are both users the same for inbound and outbound? |
| 04:32.05 | [illuminatus] | when I try to cal it say it's n invalidnumber |
| 04:32.10 | [illuminatus] | but when i call out it ues that number |
| 04:32.32 | JerJer | look to see if you registered successfully |
| 04:32.38 | [illuminatus] | I don't know, the VoIP provider didn't provide me with anything but the IP adress and phone number |
| 04:32.50 | [illuminatus] | how do I lok if I am registerd succesfully? |
| 04:33.10 | [illuminatus] | sorry again for the typos,my comp is laggingon keboard input for some reason |
| 04:33.14 | dudes | like with the sip lines we have we register = user:pass@hostname:5060/Phone_Number |
| 04:33.34 | JerJer | <PROTECTED> |
| 04:33.47 | dudes | we have the phonenumber at the end |
| 04:33.52 | JerJer | ?! |
| 04:33.57 | JerJer | phone number for what? |
| 04:34.06 | dudes | for the sip line |
| 04:34.16 | JerJer | ok why? |
| 04:34.33 | dudes | I don't know ... that's just what commpartner's told us to do |
| 04:34.43 | dudes | so I'm not going to argue because it works great |
| 04:35.29 | [illuminatus] | ok using tat format worked too, but i still cn't make incomi calls |
| 04:35.44 | [illuminatus] | how do i check if I regisered successfully/ |
| 04:35.45 | dudes | I didn't mean to imply t hat'd work for incoming |
| 04:35.46 | [illuminatus] | ? |
| 04:35.50 | dudes | sip show registry |
| 04:36.14 | *** join/#asterisk dos000 (~dos000@ip208-164.tor.istop.com) |
| 04:36.43 | dos000 | is there a simple asterisk online book ? |
| 04:36.58 | dudes | dos000 - voip-info.org |
| 04:37.05 | [illuminatus] | 209.120.255.14:5060 12143294838 120 Request Sent |
| 04:37.22 | dudes | according to that it's been set ... not registered |
| 04:37.38 | dudes | commpartners.us:5060 000010014500 23 Registered |
| 04:37.41 | JerJer | [illuminatus]: ok its not getting there |
| 04:37.44 | JerJer | check firewall |
| 04:37.53 | JerJer | there and back |
| 04:38.10 | dudes | forward port 5036 |
| 04:38.22 | dudes | I think that's for incoming sip |
| 04:38.56 | dudes | then again you don't always have to forward ports if you use a sip proxy |
| 04:39.16 | JerJer | um 5036? |
| 04:39.20 | JerJer | and no you do not port forward |
| 04:39.50 | [illuminatus] | Sip read: |
| 04:39.50 | [illuminatus] | SIP/2.0 403 Forbidden |
| 04:39.50 | [illuminatus] | Via: SIP/2.0/UDP 65.189.246.156:5060;branch=z9hG4bK6a1c39ea |
| 04:39.50 | [illuminatus] | From: <sip:12143294838@209.120.255.14>;tag=as311f31b3 |
| 04:39.50 | [illuminatus] | To: <sip:12143294838@209.120.255.14> |
| 04:39.50 | [illuminatus] | Call-ID: 778b7ce93d7f624e693828657bd884bb@127.0.0.1 |
| 04:39.52 | [illuminatus] | CSeq: 116 REGISTER |
| 04:39.54 | [illuminatus] | Content-Length: 0 |
| 04:40.02 | [illuminatus] | ??? |
| 04:40.07 | Qwell | ~pastebin |
| 04:40.10 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 04:40.10 | JerJer | they are rejecting you |
| 04:40.28 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:40.29 | JerJer | Qwell: add in for more than 4 lines (maybe?) |
| 04:40.43 | dudes | iax is port 5036 ... opps |
| 04:40.56 | [illuminatus] | iax is 4569 |
| 04:41.04 | [illuminatus] | i thought |
| 04:41.08 | JerJer | iax version 2 is 4569 |
| 04:41.13 | JerJer | iax version 1 was 5036 |
| 04:42.00 | [illuminatus] | ah |
| 04:42.03 | [illuminatus] | i see. |
| 04:42.12 | JerJer | asterisk manaager interface is 5038 |
| 04:42.31 | Qwell | JerJer: shido was talking about a form you guys will have saying "I don't want e911". If/when you do that, could you let me know? |
| 04:42.55 | [illuminatus] | ok soI took the registration strnig out bcause it is not ncessary fo outbnd calling |
| 04:42.56 | Qwell | will/might |
| 04:43.44 | [illuminatus] | so is my registration string wrong and tha's why incoming calls aern't woking? |
| 04:43.54 | Qwell | [illuminatus]: or your provider just hates you |
| 04:44.15 | JerJer | [illuminatus]: i would say yes |
| 04:45.01 | [illuminatus] | my provider apparently nver heard of asterisk before ether :( |
| 04:45.11 | Qwell | [illuminatus]: time to find a new provider |
| 04:45.22 | Qwell | might I suggest one who uses asterisk? |
| 04:45.27 | Qwell | nufone perhaps? |
| 04:47.58 | [illuminatus] | that would be a good solution |
| 04:48.13 | [illuminatus] | however, i think my boss already signed the contract to be a prtner with this jokers |
| 04:48.29 | Qwell | So then if they don't get it to work, they broke the contract |
| 04:48.29 | *** join/#asterisk xeet2 (~xeet2@bwi1-br1-gig3-1.jsci.net) |
| 04:48.37 | Qwell | simple as that |
| 04:48.57 | xeet2 | anyone know exactly what iaxcompat does? |
| 04:49.13 | dos000 | anyone : i have an ata that registers to * as 5612000 how would one dial it when a call comes in that matches the number |
| 04:50.09 | [illuminatus] | yeah this guys are all cisco |
| 04:50.11 | heath__ | dos000: exten => _NXXNXXXXXX,1,Dial(sip/yourprovider/${EXTEN}) |
| 04:50.52 | dos000 | heath: there is no provider in this case. the 2 sip ata are both connected to asterisk |
| 04:51.45 | dos000 | unless i am allowed to put asterisk ip address in there |
| 04:54.19 | heath__ | oh, sorry i read you wrong |
| 04:54.57 | dos000 | heath__, i am about to bang my head on the wall here ! |
| 04:55.19 | [illuminatus] | :( i still can't register with these guys |
| 04:55.44 | dos000 | heath__, this must be a one liner i am sure. |
| 04:56.22 | Damin | Hmm.. |
| 04:58.21 | xeet2 | hey greg |
| 04:59.24 | Damin | Howdy.. |
| 05:00.00 | dos000 | iam trying exten => _5NXXXXX,1,Dial(SIP/${EXTEN}); which the docs says but its not ringing ! |
| 05:00.47 | Qwell | dos000: what happens? |
| 05:01.16 | JerJer | the value of ${EXTEN} has to be a valid type=peer in sip.conf |
| 05:01.25 | JerJer | that's a fucking stupid idea |
| 05:01.37 | heath__ | that's cuz you have to call the ata sip/5612000 |
| 05:01.48 | dos000 | Qwell, nothing i get a timeout |
| 05:02.27 | xeet2 | jerjer: why's that? |
| 05:02.28 | dos000 | jerjer: it is a valid number defined in sip.conf |
| 05:02.46 | JerJer | so for every number you dial you need an entry ??? |
| 05:03.33 | xeet2 | if every number you dial is a seperate device anyway, sure why not? then if it doesn't have a peer entry, go somewhere else in your dialplan |
| 05:04.00 | dos000 | jerjer: i know i am a beginner .. but so far i only have 2 phones. in the future there will be more.. pray tell what the best is to get a bunch of them registered |
| 05:04.23 | xeet2 | jerjer: how would you do it? |
| 05:04.35 | JerJer | give meaningful names to sip devices |
| 05:04.43 | Qwell | SIP/bob |
| 05:04.45 | Qwell | etc |
| 05:05.42 | dos000 | jerjer: for now tho all i need is for that phone to ring and i can call it a day ! |
| 05:05.56 | JerJer | Dial,SIP/bob |
| 05:06.00 | JerJer | sip.conf: |
| 05:06.01 | JerJer | [bob] |
| 05:06.03 | JerJer | type=peer |
| 05:06.11 | xeet2 | sometimes a phone number is a meaningful name =) |
| 05:06.12 | JerJer | host=ip.address.or.host |
| 05:07.09 | dos000 | xeet2, exaxtlt |
| 05:07.13 | Qwell | JerJer: exten => 18003911234,1,Dial(SIP/bob)? |
| 05:07.24 | JerJer | sure |
| 05:07.34 | Qwell | is that how you guys are doing it? |
| 05:07.42 | dos000 | ^%#$^%#$^%#$% ... finally ringing ! |
| 05:07.44 | Qwell | I'd love to see some of your configs some time, heh |
| 05:07.59 | Qwell | just to see what a larger provider does |
| 05:08.34 | *** join/#asterisk Strom_C (~strom@66.159.243.60) |
| 05:08.41 | Mavvie | anybody here with a Patton 2977 (aka DigiFire RAS) |
| 05:08.53 | `Sauron | Mavs |
| 05:09.01 | Mavvie | hi `Sauron! you have one? |
| 05:09.02 | `Sauron | digi Rasfire? |
| 05:09.11 | JerJer | exten => _1N.,1,DetermineRoute() |
| 05:09.11 | JerJer | exten => _1N.,2,Dial,IAX2/${ACCOUNTCODE}@${ROUTE}/${EXTEN} |
| 05:09.11 | JerJer | exten => _1N.,n,Goto(${DIALSTATUS},1) |
| 05:09.12 | `Sauron | err |
| 05:09.15 | JerJer | trivial |
| 05:09.19 | `Sauron | Digi RasFire ? |
| 05:09.31 | Qwell | DetermineRoute()? |
| 05:09.33 | JerJer | very trivial softswitch |
| 05:09.38 | Mavvie | DialFire RAS |
| 05:09.44 | JerJer | my own app to determine where to send the call |
| 05:10.05 | `Sauron | hum, nope |
| 05:10.08 | JerJer | then the gateway simply does Dial,Zap/r1/${EXTEN} |
| 05:10.09 | Qwell | JerJer: it sets route, account code, and exten? |
| 05:10.12 | JerJer | very trivial |
| 05:10.14 | Mavvie | `Sauron: hehe |
| 05:10.29 | xeet2 | jerjer: what about using dundi? |
| 05:10.35 | Qwell | seems like it should be easy with a db lookup |
| 05:10.40 | JerJer | i don't need it for outbound |
| 05:10.45 | JerJer | i use it for inbound |
| 05:11.08 | Mavvie | `Sauron: I can't find the right code for setting the callerID. I get it for incoming calls, but setting it for outgoing calls is a no-no right now |
| 05:12.20 | nn | SetCallerID() or friends should work fine |
| 05:12.54 | Qwell | oh, it doesn't even need to set exten...duh |
| 05:13.12 | Mavvie | most likely will become a hack like that. |
| 05:14.21 | dos000 | jerjer: is there a way i can specify sip passwords for a bunch of users ? or i need a separate entry for each ? |
| 05:14.22 | xeet2 | anyone here ever connected * to a merlin pbx with pri? |
| 05:14.42 | Mavvie | xeet2: somebody on -users has it. |
| 05:15.05 | dos000 | i mean a separate entry in sip.conf |
| 05:15.38 | xeet2 | dos000: if you're doing alot of peers and you're running head, you might want to look at using realtime |
| 05:15.52 | JerJer | hell no |
| 05:15.54 | JerJer | realtime is not the answer |
| 05:16.27 | xeet2 | you don't think so? seems to work well so far... |
| 05:16.29 | Strom_C | realtime what? |
| 05:16.53 | xeet2 | realtime is an asterisk addon, config from database |
| 05:17.02 | JerJer | bleh |
| 05:17.26 | dos000 | okay ... |
| 05:17.40 | xeet2 | jerjer: what don't you like about realtime? =) |
| 05:19.50 | *** join/#asterisk michael1234 (~mick@202.22.163.104) |
| 05:20.04 | JerJer | it forces to asterisk to depend on the database |
| 05:20.12 | timecop | holy shit h323 compiled |
| 05:20.19 | timecop | after like 5 hours |
| 05:20.59 | xeet2 | jerjer: yes, but if your database is stable |
| 05:21.11 | JerJer | if |
| 05:21.29 | Qwell | oracle_odbc? |
| 05:21.39 | xeet2 | can always run the database locally on the box and act as a replication slave |
| 05:22.30 | JerJer | and how far will that scale? |
| 05:23.35 | xeet2 | I'd say pretty far as long as your * boxes are slaves only, and your db updates are just user additions/changes and dialplan changes |
| 05:24.32 | JerJer | good luck |
| 05:24.49 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 05:24.51 | xeet2 | hehe |
| 05:25.01 | xeet2 | you don't think that will scale well? |
| 05:25.11 | xeet2 | I mean its better than flat text files imho |
| 05:25.49 | JerJer | then tell me why class 4/5 switches are configured via serial port with a huge text file? |
| 05:26.40 | Mavvie | because you haven't figured out how to do incremental updates? |
| 05:26.50 | xeet2 | just because something is normally done one way doesn't mean thats the best way to do it |
| 05:27.04 | JerJer | who says i haven't? |
| 05:27.05 | xeet2 | and class 4/5 switches are one hell of alot more beefier than a * box |
| 05:27.28 | xeet2 | so a single text file for a single switch is perfectly fine |
| 05:27.58 | Strom_C | xeet: you know that DMS-100 and classic 5ESS are vintage 1979-1983 technology, right? |
| 05:29.50 | *** join/#asterisk blitz_astricon (~blitzrage@54.Red-80-32-211.pooles.rima-tde.net) |
| 05:30.15 | blitz_astricon | morning from Spain! |
| 05:30.17 | xeet2 | yes, and assumed jerjer was referring to more recent versions of switches |
| 05:30.40 | Strom_C | ah ok |
| 05:31.17 | JerJer | why give a solution that used to work with so little, so much to depend on just because technology has improved? |
| 05:31.44 | Strom_C | right...it's the KISS principle |
| 05:32.01 | *** join/#asterisk dos000 (~dos000@ip208-164.tor.istop.com) |
| 05:32.18 | drooth | Anyone recommend a PHP app for call tracking and dispatching techs? |
| 05:33.41 | xeet2 | because its easier to handle many * boxes when your config is in a db, at least in my opinion. yeah the problem then becomes making the db reliable but thats not that hard to accomplish |
| 05:34.24 | JerJer | i have never said that we do not utilize a database for persistent storage |
| 05:34.25 | xeet2 | you could always write some application that distributes configuration files, but why not use something thats already out there and working |
| 05:34.44 | blitz_astricon | its working? :) |
| 05:35.00 | kimo_sabe | xeet2: like applications to distribute and maintain config files? |
| 05:35.36 | xeet2 | blitz: realtime? on all our * boxes it is working quite well, as long as we keep the database locally on the * box |
| 05:36.11 | blitz_astricon | xeet2: interesting. I spent about 30 mins trying to get realtime setup to document, and decided I didn't have the time :) |
| 05:37.31 | xeet2 | it doesn't do too well when the db isn't local, 5 * boxes overwhelmed a hefty db server during peak times, hence the replication |
| 05:37.49 | *** join/#asterisk Robot_ (~robot_@pool-71-113-23-113.sttlwa.dsl-w.verizon.net) |
| 05:38.05 | JerJer | so in reality you have only a small amount of scale |
| 05:38.41 | JerJer | if you can bring a hefty box to its knees with so little |
| 05:39.05 | kimo_sabe | xeet2: are you storing RDP data in the DB? wow, what sort of call volumes are you seeing? |
| 05:39.07 | xeet2 | well lets see, each box has a copy of the db locally, and each box runs pretty well that way. whenever any changes are made, the db replication pushes it out without much effort |
| 05:39.56 | xeet2 | when we ran everything off of the db boxes, without the db being local to each * box, thats when it got overwhelmed |
| 05:40.32 | JerJer | with very little utilization |
| 05:40.49 | JerJer | so it is only going to scale so far in the replicated configuration |
| 05:41.34 | dos000 | anyone know if asterisk at home would work on my debian system ? |
| 05:42.18 | dos000 | i should say would install instead |
| 05:42.32 | xeet2 | jerjer: I'd say it has a lot of room to grow though, I mean its not like there are any updates more than once an hour or so |
| 05:42.33 | JerJer | learn asterisk the right way |
| 05:43.59 | xeet2 | kimo: some of the boxes reach about 2k calls a day, so not too much yet, but seeing how little utilization the box is under at that rate makes me believe there's alot of room to grow |
| 05:44.44 | JerJer | we do 2,000 calls before breakfast |
| 05:45.02 | xeet2 | and you're alot bigger company =) and you have what, 25 servers? |
| 05:46.05 | [illuminatus] | hey how many simultaneous calls does asterisk support? |
| 05:46.10 | JerJer | all of them |
| 05:46.22 | Strom_C | JerJer: out of random curiosity, how many erlangs run through you on a single day? |
| 05:46.37 | Strom_C | [illuminatus]: as much as the hardware you throw at it will handle |
| 05:47.00 | [illuminatus] | i was j/w cause digium is coming out with * business edition which supports only 120 calls |
| 05:47.09 | JerJer | that's a single box |
| 05:47.18 | xeet2 | jerjer: you guys should really be billing cabs =) |
| 05:47.23 | JerJer | and not dedicated to a very specific tasks, like my boxes |
| 05:47.33 | JerJer | billing cabs? |
| 05:47.50 | xeet2 | cabs revenue, inter-clec termination fees |
| 05:48.30 | xeet2 | example - xo customer calls a local verizon customer. even though the call is free to the caller, xo has to pay verizon about .002/min to terminate the call |
| 05:48.40 | JerJer | that is called recip |
| 05:48.43 | JerJer | and we get it |
| 05:48.58 | JerJer | but its trivial amounts of money per call |
| 05:49.18 | xeet2 | is it for you? its pretty high normally out here |
| 05:49.45 | xeet2 | but thats why intra-lata in md is so costly in the first place |
| 05:51.11 | xeet2 | that was the biggest reason people wanted to become a une-p clec, to bill cabs on those inbound calls... its not very trivial when you reach 50k+ customers |
| 05:51.46 | JerJer | it is still trivial amounts of revenue for a major compay like that |
| 05:54.25 | JerJer | the recip game is pretty much dead any more |
| 06:05.25 | *** join/#asterisk newmember (user@S010600d0b76b1f36.cg.shawcable.net) |
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| 07:04.28 | [Outcast] | So anyone else at the keynotes? |
| 07:11.58 | *** join/#asterisk SuperMMan (~sales@d205-206-143-75.abhsia.telus.net) |
| 07:12.12 | SuperMMan | evening all quick question does asterisk not support RFC3389 ? |
| 07:16.15 | SuperMMan | everyone sleeping i take it |
| 07:16.49 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:24.08 | blitz_astricon | SuperMMan: nope - it doesn't |
| 07:26.16 | SuperMMan | oh to bad, i wonder how i am going to get around this problem then, it was told to me by the company i am getting ld service from that i need support for rfc3389, because right now my calls sound like i am on a two way radio |
| 07:26.36 | *** join/#asterisk jansaell (~jan@54.Red-80-32-211.pooles.rima-tde.net) |
| 07:28.43 | SuperMMan | blitz_astricon anyidea how i can resolve this problem? |
| 07:35.59 | *** part/#asterisk jansaell (~jan@54.Red-80-32-211.pooles.rima-tde.net) |
| 07:37.01 | JerJer | not supporting comfort noise would not cause a two-way radio sound |
| 07:37.10 | JerJer | your provider is smokin dope |
| 07:37.37 | Qwell | Is the person you're talking to saying "over" after every sentence perhaps? |
| 07:37.46 | SuperMMan | JerJer well when i connect to his server with my GS soft phone i don`t have the priblem |
| 07:37.57 | SuperMMan | Qwell lol no |
| 07:38.06 | Qwell | What is two-way radio sound exactly? |
| 07:38.54 | SuperMMan | Qwell have you ever used a two way radio? if so you know when the other end lets go of the transmit button that sound you hear |
| 07:39.05 | Qwell | uhh |
| 07:39.13 | SuperMMan | its hard to explain if you have never used a two way radio |
| 07:39.42 | Qwell | You hear it, they hear it, or both do? |
| 07:39.53 | SuperMMan | both |
| 07:40.18 | SuperMMan | only when i am connecting though asterisk to the sip provider, but if i connect my GS to the sip provider its fine |
| 07:41.46 | Qwell | I've said it before, and I'll continue to do so. Thats why you use a provider that uses asterisk. Many others are simply (as JerJer put it) smoking dope. |
| 07:42.38 | SuperMMan | so no recommendations on fixing it? |
| 07:42.47 | JerJer | disable cng |
| 07:42.57 | Qwell | cng? |
| 07:42.58 | SuperMMan | sorry cng? |
| 07:43.08 | JerJer | comfort noise generation |
| 07:43.11 | Qwell | figured |
| 07:43.21 | SuperMMan | jeremywhiting what file is that in? |
| 07:43.27 | SuperMMan | eek sorry i mean JerJer |
| 07:43.35 | Qwell | thats something they have to do, no? |
| 07:43.50 | *** join/#asterisk jansaell (~jan@54.Red-80-32-211.pooles.rima-tde.net) |
| 07:43.57 | Qwell | but see, if both ends are hearing it... |
| 07:44.08 | Qwell | They wouldn't be generating it on incoming packets, would they? |
| 07:44.14 | Qwell | I (sadly) just read the rfc |
| 07:44.48 | JerJer | what SIP UA does this provider run? |
| 07:44.58 | Qwell | don't listen to me, I'm mostly clueless |
| 07:45.06 | SuperMMan | tell you the truth i am not sure |
| 07:46.39 | JerJer | turn on sip debug and find out |
| 07:47.04 | *** join/#asterisk Alexi1 (~Alexis@www.trim.it) |
| 07:47.19 | Alexi1 | hi all |
| 07:47.28 | JerJer | moo |
| 07:47.33 | Alexi1 | is some one using * on fedora 3 ? |
| 07:47.53 | JerJer | sure - asterisk doesn't care about a distro |
| 07:47.58 | Alexi1 | strange thing I cannot see him on top |
| 07:48.01 | skeffling | Alexi1, we are, not tried FC4 yet though, maybe today! |
| 07:48.07 | Alexi1 | :) |
| 07:50.15 | SuperMMan | JerJer still looking but i think voizBridge |
| 07:50.25 | Alexi1 | so have you ever hear about top command problem on fedora 3 or other distrib with asterisk |
| 07:51.09 | Qwell | Alexi1: top only shows the processes that fit in the screen, based on whatever sort order |
| 07:51.39 | JerJer | so some random bullshit device they found |
| 07:51.51 | Alexi1 | why it wouldn't "fit the screen" ? |
| 07:52.01 | Qwell | Alexi1: because screen is too small |
| 07:52.24 | SuperMMan | JerJer i guess. |
| 07:52.45 | Qwell | On an unmaximized window, default sized, you've only got about 15 processes that will fit |
| 07:53.40 | SuperMMan | JerJer any ideas? |
| 07:53.56 | Alexi1 | lol |
| 07:54.12 | Alexi1 | ok |
| 07:54.15 | Alexi1 | lool |
| 07:54.24 | Qwell | Whats so funny? |
| 07:54.25 | Alexi1 | I will look that way |
| 07:54.29 | Qwell | use ps |
| 07:54.40 | Alexi1 | ok |
| 07:54.44 | Alexi1 | thanks |
| 07:56.56 | Qwell | SuperMMan: is another provider not an option? |
| 07:57.08 | Qwell | if they won't help you fix it...ya know? |
| 07:57.26 | SuperMMan | I can`t find a provider that is this cheap for where i call |
| 07:57.35 | Qwell | where do you call, and how much? |
| 07:57.48 | SuperMMan | South Korea for 0.01 |
| 07:57.59 | SuperMMan | with no commit |
| 07:58.03 | Qwell | 0.01 usd? |
| 07:58.08 | SuperMMan | ya |
| 07:58.09 | Qwell | No wonder they suck. :) |
| 07:58.31 | SuperMMan | the call sounds great but that annoying sound |
| 07:58.34 | Qwell | providers have A) Good service. B) Low rates. C) Good customer service. |
| 07:58.39 | Qwell | pick 2 (sometimes 1) |
| 07:59.17 | SuperMMan | well with taking the asterisk box a way i would say a,b having the asterisk box there i would say b |
| 08:01.01 | *** join/#asterisk ManxPower (~eric@54.Red-80-32-211.pooles.rima-tde.net) |
| 08:02.12 | Qwell | ManxPower: afternoon |
| 08:02.32 | SuperMMan | anyway thanx JerJer and Qwell for the help, i am going to take off |
| 08:29.35 | inspired | If a user calls *81*003706920101#, I remove the first four digits with ${EXTEN:4}. how can I also remove the # at the end? |
| 08:44.44 | *** join/#asterisk jerry_hotlinks (~dunno@office.hotlinks.co.uk) |
| 09:01.16 | *** join/#asterisk folsson (~filip@h82n1fls32o985.telia.com) |
| 09:05.17 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:08.47 | blitz_astricon | inspired: if I remember correctly: ${EXTEN:4:-1} |
| 09:09.18 | der[mat] | hi * |
| 09:09.32 | blitz_astricon | inspired: ${EXTEN:x:y} where x removes from the left side and y removes from the right |
| 09:09.37 | der[mat] | whats the difference between dd01 and aaln at the mgcp section? |
| 09:09.44 | inspired | ah, thanks blitz_astricon |
| 09:09.55 | blitz_astricon | inspired: np |
| 09:11.10 | blitz_astricon | inspired: I always forget the syntax for the y value :) |
| 09:11.23 | blitz_astricon | both y and -y do something, but you'll figure it out |
| 09:11.32 | inspired | hehe, ok |
| 09:12.22 | *** join/#asterisk pietro (~pietro@nat.xsec.it) |
| 09:12.25 | pietro | hello |
| 09:12.46 | pietro | where asterisk save the history of CLI commands ? |
| 09:12.57 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 09:14.25 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:14.25 | RoyK | if using round robin queueing with app_queue.. how can i tell asterisk how long it should ring per agent before continuing to the next? |
| 09:18.20 | *** join/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 09:18.27 | SoloFlyer | hey |
| 09:19.30 | *** part/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 09:19.33 | *** join/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 09:19.47 | *** join/#asterisk beavizd (~anders@212.242.87.250) |
| 09:19.49 | blitz_astricon | RoyK: I remember reading that in the queues.conf file |
| 09:20.01 | SoloFlyer | hello |
| 09:20.09 | blitz_astricon | RoyK: although it could be in agents.conf - I forget, but I have seen that option |
| 09:20.16 | blitz_astricon | RoyK: I remember documenting it for the book :) |
| 09:21.45 | blitz_astricon | RoyK: timeout= ? |
| 09:22.04 | RoyK | yes |
| 09:22.06 | RoyK | found it |
| 09:22.13 | blitz_astricon | RoyK: was that it? |
| 09:22.15 | RoyK | timeout=10 sounds reasonable |
| 09:22.19 | RoyK | think so |
| 09:22.32 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 09:22.44 | blitz_astricon | I love people being assholes, then not thanking people when they do help |
| 09:23.20 | SoloFlyer | :) |
| 09:23.47 | SoloFlyer | if you help me ill thankyou :) |
| 09:23.49 | *** join/#asterisk banbanli (~chatzilla@60-248-87-123.HINET-IP.hinet.net) |
| 09:24.29 | blitz_astricon | SoloFlyer: lol, what was the question again? |
| 09:24.56 | SoloFlyer | modprobe-ing wcfxs opermode=AUSTRALIA isnt resulting in the FX0 modules comming up in Australia mode |
| 09:24.59 | SoloFlyer | ... |
| 09:25.46 | SoloFlyer | i am doing something stupid... i just know it |
| 09:25.52 | blitz_astricon | SoloFlyer: try opermode=au |
| 09:26.11 | blitz_astricon | I think its just the two letter country code, not the full country name |
| 09:26.19 | tzafrir | What exactly is opermode Isn't it something that can be set by ztcfg? |
| 09:26.38 | SoloFlyer | its the full country code in wcfsx.c |
| 09:26.57 | blitz_astricon | tzafrir: I don't think it can be set in ztcfg, but I could be very wrong :) |
| 09:27.23 | *** join/#asterisk dan_w (~dan@host86-128-235-147.range86-128.btcentralplus.com) |
| 09:27.35 | tzafrir | What exactly is it? Is it really equired for .au phones? |
| 09:28.16 | *** join/#asterisk Thumann (Thumann@is.a.retard.dk) |
| 09:28.21 | Thumann | :) hi ppl |
| 09:28.24 | SoloFlyer | only if you dont mind the tdm400 not being able to detect ring tones hangup tones and |
| 09:28.34 | SoloFlyer | and causing really bad echo |
| 09:28.38 | SoloFlyer | :) |
| 09:29.01 | dan_w | are questions about impedance allowed in here ? |
| 09:29.17 | tzafrir | SoloFlyer, But why us it something that can only be set at load time? |
| 09:29.30 | blitz_astricon | tzafrir: its used to control line impedence etc... |
| 09:29.30 | SoloFlyer | cause it has to be set on the card |
| 09:29.51 | tzafrir | Questions dan_w , I hope they are |
| 09:29.54 | dan_w | :) |
| 09:30.00 | blitz_astricon | tzafrir: its quite possible it could be setup in zaptel.conf - I forget the exact option name. |
| 09:30.03 | SoloFlyer | i suppose it could be set other than at load but its setup this way |
| 09:30.27 | SoloFlyer | according to every doco i have ever come accross... |
| 09:30.51 | blitz_astricon | I remember documenting that option |
| 09:30.52 | dan_w | well, you know when you've got an FXS interface, and your 2 wire circuit to the FXO is really really short (like 10m) you seem to get very loud sidetone |
| 09:31.06 | dan_w | so I'm guessing this is some kind of impedance mismatch |
| 09:31.32 | dan_w | my FXS hardware is a spa3k, and it has a list of impedance settings for the FXS interface |
| 09:31.33 | SoloFlyer | 10meters ? or |
| 09:31.51 | dan_w | this is kinda relevant to * as well, because I'm having the same troubles on my FXS interface from digium too... |
| 09:32.07 | tzafrir | SoloFlyer, opermode seems like a number, at least in the version of wcfxo I'm looking at |
| 09:32.13 | dan_w | 10 metres or what ? |
| 09:32.41 | SoloFlyer | 10meters doesnt sound short so i was just checking |
| 09:32.43 | tzafrir | try: modinfo wcfxo |
| 09:32.45 | SoloFlyer | :) |
| 09:32.50 | dan_w | its set at 600 Ohms at the moment. I presume just "600" means ohms... |
| 09:33.26 | SoloFlyer | yes iirc |
| 09:33.34 | dan_w | well, it might be even less than 10m, but 10m is pretty short compared to some local loops isn't it ? |
| 09:33.57 | SoloFlyer | suppose... :/ |
| 09:34.42 | tzafrir | 0 is 'FCC' (US). 1 is CTR21 (Austria, Belguim, etc. E.g: europe) |
| 09:35.06 | dan_w | So, anyone got any pointers to info about impedance and FXS circuits ? |
| 09:35.57 | dan_w | I've kinda been trawling google all morning and not come up with much |
| 09:36.07 | SoloFlyer | i know the feeling dan_w |
| 09:36.31 | SoloFlyer | but at least im getting paid for this... ( well up until 1700 i was ) |
| 09:36.43 | dan_w | hehe |
| 09:36.52 | SoloFlyer | its 1900 now!! |
| 09:36.59 | dan_w | ah well |
| 09:37.06 | SoloFlyer | im hungry :( |
| 09:37.08 | dan_w | I've tried one or two random settings |
| 09:37.14 | dan_w | doesn't seem to make much difference |
| 09:37.41 | SoloFlyer | so let me get this right u have the fxs interface connected to a fx0 interface |
| 09:37.46 | dan_w | nah |
| 09:37.59 | dan_w | I mean FXO interface as in "my analogue phone" |
| 09:38.09 | SoloFlyer | yep |
| 09:38.26 | SoloFlyer | and fxs on the 3k |
| 09:38.31 | dan_w | so FXS port on the spa3k connected to my phone |
| 09:38.56 | SoloFlyer | that should just work fine |
| 09:39.10 | dan_w | yeah, but the sidetone is way too loud |
| 09:39.22 | *** join/#asterisk Zgarbi (~my@212.58.125.70) |
| 09:39.22 | dan_w | I'm randomly testing the impedance settings on the spa3k fxs port |
| 09:39.23 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 09:39.30 | dan_w | but I'd rather have a cl00 what I'm doing... |
| 09:39.48 | SoloFlyer | the guy that works with me has a 3k witht eh phone connected about 1 meters away |
| 09:40.08 | *** join/#asterisk martinba (~martinb@dns.mobimedia.de) |
| 09:40.11 | SoloFlyer | no problem |
| 09:40.18 | dan_w | right, so is he slightly hard of hearing or did he change the impedance ? :) |
| 09:40.39 | dan_w | it defaults to 600 Ohm, which *seems* to be the "wrong" setting |
| 09:40.49 | SoloFlyer | he didnt have to play with impedence at all afaik |
| 09:41.01 | Zgarbi | hi. any asterisk developers here? I have problem with SIP register reestablish connection. if fails it didnot tryes to reestablish. anyone could me help? |
| 09:41.40 | |Vulture| | Zgarbi: there are a lot of developers here... |
| 09:42.24 | Zgarbi | and you? |
| 09:42.29 | |Vulture| | and btw a SIP registration does attempt |
| 09:43.06 | Zgarbi | what is btw? |
| 09:43.09 | dan_w | maybe I got my tip and ring the wrong way round |
| 09:43.19 | SoloFlyer | BTW = By The Way |
| 09:43.21 | blitz_astricon | ~btw |
| 09:43.22 | jbot | rumour has it, btw is by the way |
| 09:43.55 | *** join/#asterisk Morex (~blah@host81-157-123-89.range81-157.btcentralplus.com) |
| 09:44.21 | Zgarbi | so what is solution for me? |
| 09:44.26 | SoloFlyer | im gonna recompile with stable instead of cvs |
| 09:44.43 | Morex | Anyone else having trouble with voip-info.org? |
| 09:44.53 | SoloFlyer | nope |
| 09:44.58 | Morex | Hmmm |
| 09:45.01 | blitzrage | nope |
| 09:45.14 | SoloFlyer | wait.... |
| 09:45.15 | |Vulture| | Zgarbi: whats the problem? |
| 09:45.17 | SoloFlyer | yes |
| 09:45.22 | Morex | I can't get through |
| 09:45.26 | SoloFlyer | same |
| 09:45.27 | |Vulture| | Zgarbi: it tried to register and doesn't? |
| 09:45.31 | Morex | Request times out |
| 09:45.38 | Morex | Responds to pings though... |
| 09:45.57 | Zgarbi | if I get lagged it doesnt reregisters |
| 09:46.23 | blitz_astricon | oh yay, grandstream not sending DTMF :) |
| 09:46.28 | Zgarbi | |Vulture| can I PM? |
| 09:46.30 | Morex | Anybody logged in from Astricon? |
| 09:46.44 | blitz_astricon | Morex: I am |
| 09:46.46 | Morex | Probably too busy attending... |
| 09:46.47 | |Vulture| | Zgarbi: sure |
| 09:46.48 | blitz_astricon | Morex: #astricon |
| 09:46.50 | Morex | Cool |
| 09:46.57 | Morex | Wish I was there... |
| 09:47.02 | blitz_astricon | yah, its great :D |
| 09:48.48 | *** join/#asterisk Morex (~blah@host81-157-123-89.range81-157.btcentralplus.com) |
| 09:54.31 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@194.3.191.110) |
| 09:54.34 | PoWeRKiLL | hi |
| 09:54.58 | |Vulture| | sup Poincare |
| 09:55.00 | |Vulture| | urg |
| 09:55.03 | |Vulture| | PoWeRKiLL |
| 09:56.05 | PoWeRKiLL | Any idea why sometime sip show peers don't show any peer |
| 09:56.39 | *** join/#asterisk mithro (~tim@ppp226-233.lns3.adl2.internode.on.net) |
| 09:56.46 | |Vulture| | new one to me |
| 09:56.50 | Morex | PowerKill: Nope. I get the same thing though. |
| 09:57.32 | *** join/#asterisk lters (~lters@mrtcdsl-034.mis.net) |
| 09:59.03 | Morex | Voip-Info's back up. |
| 09:59.28 | SoloFlyer | Thankyou |
| 10:00.06 | Morex | I didn't do it... |
| 10:00.10 | SoloFlyer | lol |
| 10:00.16 | |Vulture| | lol |
| 10:00.19 | SoloFlyer | i take that back then |
| 10:00.22 | Morex | LOL |
| 10:00.25 | SoloFlyer | :) |
| 10:00.57 | Morex | Coolies, 348 people have looked at my new OrderlyQ page since I put it up yesterday evening... |
| 10:01.35 | SoloFlyer | ahhh its not working... |
| 10:01.54 | Morex | They must be working on it |
| 10:02.04 | |Vulture| | yea... |
| 10:02.13 | |Vulture| | :( |
| 10:02.21 | |Vulture| | wiki is my lifeline |
| 10:02.28 | SoloFlyer | lol |
| 10:03.58 | drray | wiki is alive! |
| 10:04.27 | Morex | Gone again... |
| 10:04.40 | Morex | It's up and down more often than my trousers :-) |
| 10:07.05 | *** join/#asterisk meppl (mephisto@p54AADF4F.dip.t-dialin.net) |
| 10:07.27 | SoloFlyer | morex you should get a belt it will fix that problem |
| 10:07.38 | Morex | LOL |
| 10:08.00 | mithro | whats an OrderlyQ? |
| 10:08.38 | Morex | Mithro: Call +44 845 004 5412 and find out :-) |
| 10:08.54 | drray | I'm guessing the wiki needs a mirror |
| 10:09.07 | drray | or does that make me captain obvious |
| 10:09.09 | Morex | They're planning to put one up. Read only. |
| 10:09.26 | SoloFlyer | it has a mirror |
| 10:09.28 | *** join/#asterisk Zgarbi (~my@212.58.125.70) |
| 10:09.51 | SoloFlyer | google cached mode :) |
| 10:10.00 | Morex | Mirror mirror on the wall, why do my SIP calls always stall? |
| 10:10.12 | drray | because sip is the suckiest of them all |
| 10:10.15 | Morex | LOL |
| 10:10.18 | SoloFlyer | lol |
| 10:10.46 | SoloFlyer | yeah tell me that after i marry it |
| 10:10.54 | drray | I married zap |
| 10:11.00 | drray | so I am no one to judge |
| 10:11.04 | SoloFlyer | lol |
| 10:11.11 | drray | er, not |
| 10:11.44 | *** join/#asterisk amir_ (~amir@195.226.9.186) |
| 10:12.03 | Morex | SIP works fine for me... |
| 10:12.14 | drray | yeah |
| 10:12.19 | drray | I have two phones using it |
| 10:12.21 | drray | it's fine |
| 10:12.28 | SoloFlyer | yeah its not sip that im having problems with |
| 10:12.35 | SoloFlyer | its the dam digium cards |
| 10:12.42 | Morex | Though X-Lite has issues with DTMF |
| 10:12.43 | drray | my 7960 sits on the crossover cable |
| 10:12.53 | drray | directly into the pbx |
| 10:12.54 | Morex | And AGI has issues hanging up on SIP softphones |
| 10:13.04 | Morex | But it's workaroundable |
| 10:13.49 | Morex | Ah, crossover. QoS networking with a single cable... |
| 10:14.08 | drray | wiki is aliv |
| 10:14.38 | drray | the budgetone works through the internet just fine |
| 10:14.47 | drray | well as fine as a budgetone can work |
| 10:15.43 | SoloFlyer | hmm |
| 10:15.56 | SoloFlyer | make: *** SUBDIRS=/usr/src/modules/zaptel: No such file or directory. Stop. << anyone know why im getting that |
| 10:16.08 | drray | running as root? |
| 10:16.12 | SoloFlyer | yes |
| 10:16.21 | drray | did you make clean? |
| 10:16.27 | SoloFlyer | yes |
| 10:16.30 | drray | then no |
| 10:16.31 | drray | er |
| 10:16.35 | SoloFlyer | :) |
| 10:16.49 | Morex | Does it exist? |
| 10:17.06 | Morex | Sometimes you need to symlink... |
| 10:17.09 | SoloFlyer | SVRADL50:/usr/src/modules/zaptel# make linux26 |
| 10:17.14 | drray | your secret is safe with us |
| 10:17.21 | Morex | LOL |
| 10:17.38 | SoloFlyer | how the fuck cant it exist !!! im running make from it :P |
| 10:17.56 | SoloFlyer | sorry fsck |
| 10:18.05 | SoloFlyer | :) |
| 10:18.07 | Morex | Sorry |
| 10:18.08 | SoloFlyer | i knwo :) |
| 10:18.40 | SoloFlyer | yes send some of that this way... |
| 10:18.58 | Morex | http://www.mudig.com/satie/Gnoss1.mp3 |
| 10:19.21 | Morex | Or is it http://www.mudig.com/satie/Gnoss2.mp3 ? One of them's soothing... |
| 10:19.42 | SoloFlyer | whats the other one... hardcore urban punk? |
| 10:19.50 | Morex | Hard trance. |
| 10:19.52 | SoloFlyer | lol |
| 10:20.19 | Morex | There's http://www.mudig.com/satie/Gnoss3.mp3 too which is quite soothing. |
| 10:20.20 | PoWeRKiLL | Morex what are you doing when it's happens you just restart and it's goes back right ? |
| 10:21.06 | Morex | Um, I haven't tried it. Show peers hasn't been mission-critical for me. |
| 10:21.15 | Morex | I just noticed my SIP peers weren't showing up. |
| 10:21.55 | Morex | They've been working, just not showing in SHOW PEERS |
| 10:23.12 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 10:23.16 | onkeltimm | helay |
| 10:23.27 | Morex | Ho. |
| 10:23.43 | drray | so if I wanted to write a perl/agi script to check the status of the linus.. how would I know how long they have been connected and who they are connected to? |
| 10:24.01 | Weezey | AGI( |
| 10:24.02 | drray | agi channel status does not appear to give that info |
| 10:24.15 | Morex | Um |
| 10:24.19 | Weezey | AGI('How\'s Linux?'); |
| 10:24.23 | Morex | LOL |
| 10:24.58 | drray | WHERE IS CAPTAIN OBVIOUS when you need him? |
| 10:25.00 | Morex | Drray: Try using the agi_channel variable passed by asterisk |
| 10:25.14 | Morex | Rather than the name of the channel as it appears in sip.conf etc. |
| 10:26.43 | Weezey | I built a PHP script to monitor my extensions with a connection to the monitor |
| 10:27.24 | Weezey | it works pretty well, I use it to update a database record which updates a page for the stupid receptionist who needs to see who is on the phone at all times. |
| 10:28.33 | drray | yeah |
| 10:28.42 | drray | I need that for the cisco 7960 xml stuff |
| 10:28.57 | drray | do you have to compute the time of call yourself? |
| 10:29.05 | Weezey | yeah, you have to use your own solution with cisco and SIP. |
| 10:29.16 | drray | this would be zap |
| 10:29.18 | drray | but yeah |
| 10:29.22 | drray | thanks |
| 10:29.50 | Weezey | drray: date("U") - date("U", strtotime($row["calldate"])); |
| 10:29.57 | Weezey | call time in secs in PHP |
| 10:30.24 | drray | there is an agi_calldate? |
| 10:30.38 | Weezey | i duuno, I use the monitor |
| 10:31.50 | Weezey | Morex: If I had to sit on hold for too long with this mp3, I'd kill myself. |
| 10:32.19 | Morex | Drray: Nope |
| 10:32.38 | drray | so what is the monitor? since I'm asking stupid questions anyway? |
| 10:32.48 | Morex | Drray: AGI's only really useful to handle the call while it's in the AGI script |
| 10:32.49 | Weezey | set up monitor.conf |
| 10:32.56 | Morex | You probably need the Manager API |
| 10:33.20 | drray | monitor is for recording calls |
| 10:33.28 | Weezey | oh, right |
| 10:33.32 | drray | ok |
| 10:33.32 | Weezey | it's too early for me |
| 10:33.33 | drray | managaer |
| 10:33.35 | drray | fine |
| 10:33.38 | Weezey | yeah |
| 10:33.40 | drray | too late for me |
| 10:33.48 | drray | I'm like what the heck? |
| 10:34.13 | Weezey | Leave it to me to use call recording to check the status of calls. |
| 10:34.27 | drray | break out the buttset |
| 10:34.49 | Weezey | "Uhm, yeah, Dave's been on the phone for a good hour now." |
| 10:35.22 | drray | buttset is a lineman's handset, aligator clips |
| 10:35.41 | Morex | This is an American thing, isn't it? |
| 10:35.42 | Weezey | typically orange phonelike thing |
| 10:35.42 | drray | for listening at the telco block or off the wire |
| 10:35.48 | Morex | Oh OK |
| 10:35.49 | Morex | Got it |
| 10:36.04 | Weezey | they cost too much |
| 10:36.19 | drray | my butsett has magnets that pick up the convo without brekaing the wire |
| 10:36.26 | Weezey | you can do a lot of the same functions with a hacked up radio shack phone |
| 10:36.29 | Morex | Fancy.... |
| 10:37.04 | drray | all you need is a pair of aligator clips and a regular phone for 90% of it |
| 10:37.04 | Weezey | new hold music: http://www.simpsoncrazy.com/downloads/music/maxpower.mp3 |
| 10:37.29 | drray | thanks for the push for manager.api that is just what I wanted |
| 10:37.43 | Weezey | sorry for the misdirection |
| 10:37.46 | Morex | More than welcome. |
| 10:37.48 | drray | it's ok |
| 10:41.24 | SoloFlyer | im going home cya |
| 10:41.29 | Morex | CFN |
| 10:41.41 | SoloFlyer | i finished work over 3 and a half hours ago! |
| 10:41.46 | Morex | LOL |
| 10:41.50 | SoloFlyer | cya |
| 10:41.57 | Morex | I'm supposed to have started at least two hours ago... |
| 10:42.00 | SoloFlyer | lol |
| 10:42.34 | jerry_hotlinks | anyone be kind enough to give me an example of the read() function - its suppoosed to write a file perhaps? |
| 10:44.39 | Morex | Jerry: Which language? |
| 10:45.22 | jerry_hotlinks | english or do you mean which programming alng? |
| 10:45.27 | Morex | Programming... |
| 10:45.28 | jerry_hotlinks | *lang |
| 10:45.31 | jerry_hotlinks | ah |
| 10:45.37 | jerry_hotlinks | thats a damn good question |
| 10:45.48 | *** join/#asterisk Abbas (Abbas@203.81.213.27) |
| 10:45.53 | Abbas | hi |
| 10:45.56 | Morex | In java read() on an input stream returns the next character in the file |
| 10:45.58 | jerry_hotlinks | is there no way of just using the read function to store the tones? |
| 10:46.16 | Morex | Or read(buffer) reads lots of characters into a buffer... |
| 10:46.30 | Morex | Ah, and in Asterisk... |
| 10:46.52 | Abbas | who provides toll free and normal DID wholesale in UK, AUS , GERMANY, FRANCE over the IP |
| 10:47.04 | Abbas | ~ toll free |
| 10:47.10 | Abbas | ~toll free |
| 10:47.41 | tzanger | I don't know if you'll find one provider that has DIDs available on three continents |
| 10:48.16 | Abbas | tzanger do know any one provides any of above contries? |
| 10:48.23 | Morex | read(MYVARIABLE|5|skip) |
| 10:48.24 | jerry_hotlinks | yep into asterisk |
| 10:48.34 | jerry_hotlinks | ok |
| 10:48.34 | tzanger | Abbas: plenty do US, not sure about hte others as I'm an untravelled Canadian |
| 10:48.37 | Morex | show application read |
| 10:48.42 | Morex | will tell you all about it. |
| 10:49.06 | Abbas | can u help me getting some name in US and Canada |
| 10:49.10 | jerry_hotlinks | yup i did that - how does it store the tones though? |
| 10:49.29 | Morex | Oh, it'll set the variable to "1234" if you enter 1234 |
| 10:49.36 | jerry_hotlinks | i c |
| 10:49.41 | jerry_hotlinks | thanks for that |
| 10:49.43 | Morex | NP |
| 10:49.45 | tzanger | Abbas: nufone, voctel, sixtel, unlimitel... |
| 10:49.46 | Abbas | tzanger can u help me finding some name in US and canada |
| 10:49.54 | tzanger | Abbas: don't repeat yourself every minute |
| 10:50.09 | tzanger | Abbas: and a modicum of basic research is required or you'll just end up getting ignored |
| 10:50.13 | Abbas | tzanger their websites ? and let me know asmany as u know |
| 10:50.27 | tzanger | Abbas: if you want me ot do your homework, I charge. |
| 10:50.43 | Abbas | how much? |
| 10:50.59 | tzanger | US$95/hr min 1hr |
| 10:51.24 | tzanger | the information you're looking for is very easy to find online |
| 10:51.25 | Abbas | what the hell is this even one website is not being told freee on free asterisk help channel |
| 10:51.32 | Abbas | pethatic |
| 10:51.32 | tzanger | this is a free asterisk help channel |
| 10:51.39 | tzanger | you're not even trying to help yourself |
| 10:51.50 | tzanger | if you insist on being that lazy, I charge. simple as that. |
| 10:52.06 | tzanger | ask pretty much anyone on here, I'm one of the most helpful people you will meet here |
| 10:52.15 | Abbas | i have searched so many on google i just want a second opinion so that can know which are the companies wwell known in US an canada |
| 10:52.17 | tzanger | but I ask that you at least ATTEMPT to help yourself. |
| 10:52.38 | tzanger | Abbas: I gave you a starting list. If you searched the mailing list archives or voip-info.org you'd find many more |
| 10:52.56 | Abbas | thanks tzanger |
| 10:53.26 | tzanger | I'm not trying to be an ass, but I refuse to hand-hold. I prefer to help people who want to learn |
| 10:54.25 | Abbas | tzanger by sitting 10000s of miles away from US how can i know which is reliable by just searching on google and lists |
| 10:54.39 | Abbas | obviously i need an opinion from a reasonable and sincere person |
| 10:54.42 | Abbas | thats y i came here |
| 10:54.43 | tzanger | Abbas: I'm *in* canada and I don't know which are reliable since I don't use all of them. |
| 10:55.00 | tzanger | everyone is going to have their horror stories about specific providers |
| 10:55.09 | Abbas | hmmm |
| 10:55.18 | Abbas | u might be right |
| 10:55.18 | tzanger | personally I have exceedingly good luck with unlimitel for .ca DIDs but they don't do IAX2. |
| 10:55.34 | Abbas | they do SIP? |
| 10:56.06 | tzanger | Sixtel/Voipjet/Broadvoice seem to be "the baddies" but they have DIDs everywhere. |
| 10:56.25 | tzanger | Nufone I've *never* had issue with and I push 5kmin/mo through them, but their DID selection is extremely limited, as is Voctel |
| 10:56.58 | Abbas | thats great knowledge my friend |
| 10:57.09 | Abbas | thanks a lot for helping me |
| 10:57.14 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 10:57.28 | tzanger | My personal opinion is to stay away from sixtel and broadvoice. If you search the lists you will find an unbelievable amount of traffic about them being down or shitty audio or crappy service or DIDs not routing or any number of problems. |
| 10:58.09 | dan_w | ditto sipgate |
| 10:58.11 | tzanger | but then again some people say they are the best. I simply won't deal with them based on what I've seen on the list and what I've seen with my own sixtel did. it always seems to work but my call volume fromthemis so little it's hard to tell if I'm just hitting the right times |
| 10:58.25 | *** join/#asterisk pantanero (~pantanero@bl5-200-230.dsl.telepac.pt) |
| 10:58.55 | dan_w | sipgate didn't acknowledge an outgoing call issue for two days, despite frequent and informative emails from me |
| 10:59.15 | tzanger | I can tell you that I get *constant* "sip registration failed" "sip wrong password "qualify timeout" messages with sixtel. I'm simply waiting for the account balance to hit zero before I terminate. |
| 10:59.19 | dan_w | it must be my config, allegedly, which didn't change before or after the problems started/ended |
| 10:59.19 | Abbas | tzanger u seems to be right |
| 10:59.32 | tzanger | Abbas: of course I'm right. :-) |
| 10:59.37 | Abbas | :) |
| 10:59.41 | Thumann | xD |
| 11:00.08 | tzanger | being a reliable VOIP provider is a lot harder than people think |
| 11:00.14 | Abbas | i have seen many times NUfone downage complaints on channel |
| 11:00.29 | tzanger | I refuse to deal with proviers who don't have their own TDM networks (which is sixtel, voipjet etc) |
| 11:00.52 | tzanger | because all they do is shuttle your traffic on to someone else... so now your problem could be with the guy you're paying or they guy they're paying |
| 11:01.00 | dan_w | tzanger, of course it is, but sipgate cant even get the basics (their billing/auth system) right |
| 11:01.02 | tzanger | yes, nufone gets a lot of complaints about customer service |
| 11:01.35 | tzanger | but you know what -- in the 1.75 years I've used them I've *never* had a technical issue that was their problem. Nufone is VERY MUCH a "not for newbies" termination, but they are taking steps to fix that |
| 11:01.42 | tzanger | and jerjer and shido6 are on here all the time |
| 11:01.44 | *** join/#asterisk [Jedi] (~fdsafasdf@213.162.200.226) |
| 11:02.04 | tzanger | their customer service skills just need some polishing. :-) |
| 11:02.07 | tzanger | I'll tell you what though |
| 11:02.34 | tzanger | when I am able to convince (yes you have to convince them) jerjer that I the problem isn't likely my end... he's on it like a fat kid on a cupcake. |
| 11:03.27 | tzanger | We spent 6 hours together fixing a bug in asterisk. the guy knows asterisk code and is capable of fixing it (as am I for some parts) |
| 11:03.55 | Morex | How do you make an Asterisk feature request? |
| 11:03.57 | tzanger | and he also set up an entire test server for me which terminated to his TDM network so we could test new and exciting bits of asterisk without risking his entire VOIP network |
| 11:04.09 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:04.12 | RoyK | hello world |
| 11:04.13 | Morex | I need to change the way AGI behaves if it can't find the AGI server... |
| 11:04.21 | tzanger | asterlink is good too now that I think of it -- that's bkw_ and anthm, and both those dudes know what the hell they're talking about |
| 11:04.45 | tzanger | Morex: bugtracker is the way to do it but I'd post to -dev first and get some opinions/backup. :-) |
| 11:05.11 | Morex | Tzanger: Many thanks for that. |
| 11:05.12 | Morex | Will do. |
| 11:05.34 | tzanger | dan_w: I've never used sipgate (they're .de, perhaps you know other .de providers for Abbas) |
| 11:06.05 | Abbas | mmm in germany |
| 11:06.33 | [Jedi] | Morex: you're using FastAGI? |
| 11:06.39 | tzanger | voicepulse connect seems to work but I haven't used them in over a year... their iax2 support was always spotty but I hear their sip network is good |
| 11:08.01 | Morex | Jedi: Yup, and Manager |
| 11:08.24 | Morex | Just working on my new application server now... |
| 11:09.38 | tzanger | anyway I gotta get dressed and get the kids ready |
| 11:09.51 | tzanger | that should be a good starting point for north america anyway |
| 11:09.58 | Abbas | tzanger actually i wanted to launch my calling card and wanted to get access numbers for that i have my SIP server in US virginia |
| 11:11.29 | tzanger | I understand -- you want a nationwide provider of DIDs. It's difficult to find a *reliable* provider whos got DIDs in the major US48 and doesn't require a million-minute-a-month minimum to talk to you. All the little guys with US48 DIDs are just getting their DIDs over SIP (i.e. they're not terminating themselves) which is what leads to a lot of problems IME. |
| 11:16.06 | Abbas | tzanger i can start working state by state like initially i can go for NY and then for other |
| 11:16.50 | Abbas | tzanger in this way i might find some good and stable on with no minute commitment |
| 11:21.09 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 11:22.32 | *** join/#asterisk jacks (none@dslam7-21-59-81.dyndsl.versatel.nl) |
| 11:22.35 | jacks | heya |
| 11:22.47 | jacks | when using the DISA application, the following warning comes up: Jun 16 13:21:40 WARNING[28644]: cdr.c:286 ast_cdr_init: CDR already initialized on |
| 11:25.16 | *** join/#asterisk popooya (~popooya@08b8ae1d0a6311b7.session.tor) |
| 11:31.00 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 11:32.05 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 11:33.31 | dan_w | anyone know if the spa3k supports early dial ? |
| 11:35.58 | *** join/#asterisk skiold (~userid@84-121-68-176.onocable.ono.com) |
| 11:40.15 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 11:41.50 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@194.3.191.110) |
| 11:42.35 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 11:44.53 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 11:45.29 | _omer | how to download and install SOX ? |
| 11:45.43 | tzafrir | _omer, apt-get install sox |
| 11:45.56 | tzafrir | or similar methods. It should be part of most distros |
| 11:46.02 | bublbobl | hi all. Is there a way to tell asterisk not to load modem.conf ? (we have an undefined symbol : ast_unregister_modem-driver that prevents * from starting) |
| 11:46.49 | tzafrir | bublbobl, noload=module_name.so |
| 11:47.03 | tzafrir | in /etc/asterisk/modules.conf |
| 11:47.13 | bublbobl | tzafrir> where should I enter it ? :-$ |
| 11:47.20 | bublbobl | tzafrir> thx :-) |
| 11:47.28 | beavizd | bublbobl: noload => chan_modem.so i modules.conf |
| 11:47.32 | beavizd | duh ;) |
| 11:47.35 | _omer | tzafrir: do I only need to type : apt-get install sox ? that's it |
| 11:47.44 | tzafrir | I don't remember the exact module name. Also: it may have dependent modules |
| 11:47.49 | Morex | Gotta love Debian... |
| 11:47.58 | tzafrir | _omer, what distro> |
| 11:48.03 | tzafrir | ? |
| 11:48.07 | _omer | distro? |
| 11:48.28 | tzafrir | On what OS do you run asterisk? |
| 11:48.35 | _omer | RED HAT 9 |
| 11:48.47 | tzafrir | One of those |
| 11:49.52 | tzafrir | _omer, it should be one of the standard RH9 packages. |
| 11:50.16 | tzafrir | check http://fedoralegacy.org/ and/org http://freshrpms.net/ |
| 11:50.27 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
| 11:50.29 | tzafrir | I don't know if that version supports gsm, though |
| 11:51.05 | _omer | u mean RED HAT ? |
| 11:51.07 | _omer | it supports |
| 11:51.34 | _omer | I have tried recording in GSM format....now I need to make 1 file with SOX as I read in wiki |
| 11:52.05 | _omer | but I dont know how to download and install SOX ... |
| 11:53.09 | *** join/#asterisk nemisus (~nemisus@203-206-228-13.dyn.iinet.net.au) |
| 11:58.59 | *** join/#asterisk Romik_ (~romik@212.143.5.146) |
| 11:59.26 | bublbobl | If we disable chan_modem, then we lose SIP and many apps :+( . What is strange is that we have this "undefined symbol: ast_unregister_modem_driver" message since we changed musiconhold.conf. Did anyone meet such a pb ? :-o |
| 12:00.00 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
| 12:02.27 | tzafrir | bublbobl, so maybe try to get rid of the problem instead |
| 12:02.54 | tzafrir | What's the exact error? |
| 12:03.09 | bublbobl | tzafrir> Yes, but i don't know how to troubleshoot. One moment I copy... |
| 12:03.20 | bublbobl | tzafrir> [chan_modem_aopen.so]/usr/lib/asterisk/modules/chan_modem_aopen.so: undefined symbol: ast_unregister_modem_driver |
| 12:04.03 | bublbobl | tzafrir> we didn't change chan_modem_aopen.so or recompile, only modified music on hold |
| 12:04.44 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 12:05.18 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 12:08.46 | bublbobl | tzafrir> it seems to be the same pb as described here : http://lists.digium.com/pipermail/asterisk-users/2004-April/044635.html |
| 12:10.00 | Romik_ | j #gnudialer |
| 12:10.17 | Romik_ | oj |
| 12:10.30 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 12:14.39 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 12:19.27 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 12:19.27 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 12:25.11 | *** join/#asterisk nain (nain@61.5.143.19) |
| 12:26.01 | bublbobl | <PROTECTED> |
| 12:26.19 | beavizd | bublbobl: np :) |
| 12:26.45 | nain | HI Guys |
| 12:29.12 | nain | How i will Dial exten creating in extensions.conf using chan_h323 driver (from h323.conf) ? |
| 12:34.03 | *** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) |
| 12:37.12 | *** join/#asterisk teapot (~tandrews@mail.grok.org.za) |
| 12:37.19 | teapot | hello |
| 12:39.40 | nain | hi |
| 12:40.27 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:40.55 | Ariel_ | hello everyone |
| 12:41.50 | teapot | Can anyone tell me how I can handle a dialplan module returning -1 ? |
| 12:42.07 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 12:45.30 | darkskiez | How can I get asterisk to pay attention to the authorisation username, rather than the sip username? (I want to have multiple phones with the same number as a line appearance for outgoing calls) |
| 12:45.31 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 12:46.18 | *** join/#asterisk coppice (~chatzilla@163.204.17.210.dyn.pacific.net.hk) |
| 12:48.53 | Ariel_ | teapot more info on what you want to do is needed |
| 12:49.24 | Ariel_ | darkskiez, you can have all your accounts normally and just set your outbound caller ID via the dial strings |
| 12:50.02 | Ariel_ | cluecon, morning |
| 12:50.08 | Ariel_ | those were easy ones |
| 12:50.10 | darkskiez | have all of my accounts? |
| 12:50.14 | cluecon | I know. |
| 12:50.28 | inspired | hmm, do the iax and sip protocols support sending images? |
| 12:50.48 | inspired | if so, does the same thing happen as with faxes? do they get screwed up at packet loss or are packets resent? |
| 12:51.00 | inspired | re-sent |
| 12:51.06 | teapot | Ariel_: I'll give you an example.. |
| 12:51.08 | *** join/#asterisk malfi (~malte@dsl-084-059-060-136.arcor-ip.net) |
| 12:51.19 | Ariel_ | inspired, not really there is a codec for sip and h323 to have video |
| 12:51.34 | *** join/#asterisk heka (~heka@82.114.68.126) |
| 12:51.40 | inspired | uhm, ok |
| 12:51.45 | teapot | say I have this in my dialplan: |
| 12:51.48 | teapot | <PROTECTED> |
| 12:51.48 | teapot | <PROTECTED> |
| 12:51.48 | teapot | <PROTECTED> |
| 12:51.50 | teapot | <PROTECTED> |
| 12:51.54 | teapot | <PROTECTED> |
| 12:52.02 | teapot | Now what if Command returns -1 |
| 12:52.03 | Ariel_ | inspired, faxes can only be sent in an uncompress codec ulaw/alaw unless you have a t38 which asterisk does not support at this time. |
| 12:52.03 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 12:52.05 | inspired | Ariel_: so, for sip... will the image still get there if there's a lot of packet loss |
| 12:52.06 | cluecon | darkskiez, before your dial command you need an exten line that says...exten => 1XXNXXXXXX,1,SetCallerID(Your Caller Name<18889995555>|a) |
| 12:52.09 | teapot | How do I test for that ? |
| 12:52.20 | inspired | Ariel_: yes, I know, but I've got some ideas |
| 12:52.28 | inspired | what exactly does the codec for sending images do? |
| 12:53.08 | darkskiez | cluecon, yeh, I have that already, I have each phone set to its ddi, but I want to use the second line appearances to dial out from the main number, I'm not sure of the best way to configure those accounts. |
| 12:53.10 | inspired | let's say I send a pic over sip with that codec and there's huge packet loss on my link. will the full picture get to the receiver eventually? |
| 12:53.10 | malfi | If I call my * via chan_capi and send some dtmfs, asterisk complains about "chan_capi.c:875 capi_read: Fax detected, but no fax extension". softdtmf=0 doesn't help. Any hint? |
| 12:54.18 | Ariel_ | darkskiez, you can do 8 for use of this line or 9 for the normal one in your dialing rules |
| 12:54.37 | darkskiez | Oh, bugger. |
| 12:54.46 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 12:54.58 | darkskiez | so what is the point of multiple line appearances? |
| 12:55.14 | cluecon | darkskiez: I don't think you can use a single phone to have multiple registrations. the purpose of multi-lines is to be able to take more than one call. |
| 12:55.43 | *** part/#asterisk pif (ldm@zenon.apartia.fr) |
| 12:56.14 | Ariel_ | teapot, asterisk adds 101 to the number. like exten => s,1,Dial(Blah) exten => s,2,Congestion exten => s,102,Do something. |
| 12:56.29 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 12:56.34 | teapot | I thought as much Ariel_ |
| 12:56.41 | *** part/#asterisk sigterm (sigterm@devious.info) |
| 12:56.42 | _omer | I just downloaded the SOX. now how to install it ? |
| 12:56.53 | teapot | Ariel_: But in some cases a module won't return +101 |
| 12:56.58 | *** join/#asterisk thieumS (~darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 12:57.30 | Katty | mrow |
| 12:57.36 | Ariel_ | teapot, you then have to fix the dial plan correctly. I use macro's for mine and check for that same thing. |
| 12:57.36 | thieumS | somebody knows about this message : "PRI: received SETUP message for call that is not a new call, wicked!!!" ? |
| 12:57.50 | Ariel_ | Katty, morning |
| 12:57.58 | teapot | Ariel_: In my example above where I use "n" instead of absolute numbers like "2", how do you handle the +101 ? |
| 12:58.00 | _omer | make install doesnt work.. |
| 12:58.05 | darkskiez | cluecon, I have 7960's they do multiple registrations, they have parameters authname, and name, I was going to set authname to be the same for each line, to use the same account, it doesnt register if its the same, and set name to set the callerid. |
| 12:58.18 | darkskiez | cluecon, the problem being, asterisk seems to ignore the authname, is this correct |
| 12:58.36 | Ariel_ | _omer, look to see if there is an install script. you might have to do ./configure the ./install.sh |
| 12:59.43 | darkskiez | you get meat smoothies? |
| 12:59.44 | _omer | thanks Ariel_ .... ./configure works.. |
| 12:59.57 | darkskiez | meatshake! |
| 13:00.56 | *** part/#asterisk heka (~heka@82.114.68.126) |
| 13:00.58 | Ariel_ | darkskiez, asterisk can't use the same account. It will only see the last one that registers |
| 13:01.17 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 13:01.21 | cluecon | darkskiez: try this, including the verification of which code it is running... http://lists.digium.com/pipermail/asterisk-users/2004-February/037709.html |
| 13:01.22 | lehel | hello |
| 13:01.52 | Katty | i want to rtfm on voicemail |
| 13:01.53 | Katty | where is it? |
| 13:02.07 | cluecon | darkskiez: you will need to use different authnames i think. |
| 13:02.17 | cluecon | Katty: what about voicemail do you want to know? |
| 13:02.19 | Katty | seeing how i don't really comprehend voicemail in the first place |
| 13:02.24 | Katty | cluecon: everything |
| 13:02.31 | Katty | cluecon: specifically how it works before i set anything up |
| 13:02.45 | Ariel_ | Katty, wiki has lots of info on it.http://www.voip-info.org/tiki-index.php?page=Asterisk+VoiceMail |
| 13:02.58 | Katty | thanks. |
| 13:03.00 | cluecon | Katty: start here... http://www.voip-info.org/tiki-index.php?page=Asterisk+VoiceMail |
| 13:03.01 | darkskiez | so Asterisk ignores SIP Authorisation headers? |
| 13:03.05 | Katty | i'mma get edjimicated |
| 13:03.30 | Ariel_ | cluecon, your about 2 strokes slow today what's up? No coffee yet? |
| 13:03.47 | Katty | he obviously requires hugging |
| 13:03.57 | cluecon | working from home today...haven't hit the caffiene yet. |
| 13:04.01 | darkskiez | Contact: <sip:5150@10.11.19.255:5060> |
| 13:04.01 | darkskiez | Authorization: Digest username="5176",realm="asterisk",uri="sip:10.3.0.51",response="37f0965dd6a3b04f8486d513d4ab4808",nonce="702741b3",algorithm=md5 |
| 13:04.02 | Ariel_ | darkskiez, asterisk is not a proxy server. It requires proper registrations |
| 13:04.04 | Katty | gah! can't open the cashews :< |
| 13:04.39 | darkskiez | It sees that as username 5150, not 5176 |
| 13:05.14 | Katty | why do they sell things you cannot open without scissors or pocket knife at the gas station? of all places! |
| 13:05.46 | darkskiez | If only I could get SER to accept multiple registrations and pass them on to *. But that looks like brain surgery. |
| 13:05.54 | cluecon | darkskiez: did you try that link? it should give you what you want. |
| 13:05.58 | Ariel_ | darkskiez, ser does |
| 13:06.48 | darkskiez | cluecon, I did thanks, i'm quite familiar with the cisco config already, but I'd forgotton about the weird proxy aspect of sip. |
| 13:07.19 | Katty | i need an explination: |
| 13:07.20 | darkskiez | Ariel_, is there a minimal config somewhere that does that? |
| 13:07.23 | Katty | 210 => 5555,John Smith,jsmith@yourdomain.com |
| 13:07.25 | Katty | what is 210? |
| 13:07.38 | Ariel_ | darkskiez, SER is a bear |
| 13:07.39 | Katty | and what is 5555? |
| 13:07.47 | cluecon | Katty: did you read the page? |
| 13:07.50 | darkskiez | Ariel_, i know :/ |
| 13:07.55 | Ariel_ | Katty, the mailbox number |
| 13:07.55 | Katty | cluecon: yes, there is no explination... |
| 13:08.05 | Katty | Ariel_: yes...but how do you use the mailbox number? |
| 13:08.11 | Ariel_ | 5555 password |
| 13:08.12 | Katty | Ariel_: keep in mind i've never used voicemail before :) |
| 13:08.19 | Katty | so you dial the 210? |
| 13:08.31 | cluecon | Katty: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+voicemail.conf |
| 13:08.44 | martinba | Ariel_, what kind of flyning do you mean? |
| 13:09.08 | cluecon | 210 is the mailbox, 5555 is the password. that link is for configuring voicemail.conf which is only half the puzzle. |
| 13:09.19 | Ariel_ | martinba, In my airplane going over the water and enjoying a open canopy flight. |
| 13:09.37 | Katty | cluecon: gah, that confuses me entirely |
| 13:10.25 | martinba | Ariel_, nice .. all I have is a RC Canopy aircraft ;) |
| 13:10.31 | Ariel_ | Katty, extensions are different you can have an extension as 1000 with mailbox 200. It is all in the way you setup the dialing rules and via sip or iax2 mailbox=200 |
| 13:10.55 | Ariel_ | martinba, I have a Stol airplane called the Koliber 150a great fun plane |
| 13:10.58 | Katty | what is the 210 actually for though? |
| 13:11.06 | Ariel_ | mailbox number |
| 13:11.07 | Katty | you pickup and dial 210 and they go into their mailbox? |
| 13:11.11 | Ariel_ | no |
| 13:11.19 | Ariel_ | and yes |
| 13:11.27 | Katty | way to confuse the crap out of me :< |
| 13:11.31 | Ariel_ | it is dependent on your dialing rule |
| 13:11.57 | Ariel_ | Katty, the two are not sync. You can dial into the voicemail application and put the mail box number |
| 13:12.08 | Katty | oh. |
| 13:12.09 | Ariel_ | some people have them the same as the extension but allot don't |
| 13:12.16 | Katty | hmm. |
| 13:12.23 | Katty | so the voicemail app sits on a single extenison |
| 13:12.30 | Ariel_ | it ca |
| 13:12.31 | Katty | then asks for username (210) and password (5555)? |
| 13:12.33 | Ariel_ | can |
| 13:12.40 | Ariel_ | yes it can |
| 13:12.47 | Ariel_ | or it can also work off a callerID |
| 13:12.55 | Katty | i think i'd rather keep it simple :< |
| 13:13.23 | Katty | if working off callerid is simple to setup (AND understand) then that would be neat |
| 13:13.23 | Ariel_ | Katty, it's very flexable in how you can set the voicemail up. |
| 13:13.35 | Katty | Ariel_: i imagine it is, but when you don't know anything, it's rather difficult :P |
| 13:13.46 | Ariel_ | Katty, yes your correct |
| 13:13.47 | *** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust27.bagu.broadband.ntl.com) |
| 13:13.50 | Katty | i think someone's going to have to hold my hand through this one |
| 13:14.46 | `Sauron | Oh, you're so Katty today. |
| 13:15.22 | cluecon | Katty: i have a voicemail extension (in extensions.conf) that looks like this...exten => 100,1,VoiceMailMain(s${CALLERIDNUM}) |
| 13:15.50 | Katty | :> |
| 13:16.01 | Katty | Ariel_: ((((((= |
| 13:16.03 | Katty | clean++ |
| 13:16.16 | inspired | is the codec or the protocol doing retransmission on packet loss? |
| 13:16.17 | Katty | `Sauron: i'm Katty everyday |
| 13:16.18 | inspired | is it* |
| 13:17.09 | cluecon | Katty: in voicemail.conf, in my [default] section i have 222 => 222,Cluecon,cluecon@cluecon.com |
| 13:17.24 | Katty | 222 is your callerid number? |
| 13:17.31 | Ariel_ | inspired, not really |
| 13:17.38 | Katty | and your password too? |
| 13:17.57 | inspired | Ariel_: hmpf |
| 13:17.58 | Ariel_ | Katty, when you use s option it skips the password. |
| 13:18.18 | cluecon | Katty: in my sip.conf for peer [222] I have a line that says mailbox=222@default |
| 13:18.24 | inspired | Ariel_: if _I_ make my own codec codec_filesend, can I make it retransmit until the whole damn file is received at the other end? |
| 13:18.24 | Ariel_ | inspired, that is why you get loss of works or sounds crappy some times. |
| 13:18.41 | inspired | or doesn't the codec have control of that? is it the iax2/sip protocol, and in that case I have to make my own protocol? |
| 13:18.48 | Ariel_ | inspired, voice is real time does not work well with packet loss |
| 13:19.02 | inspired | I don't want to send no damn voice ;) |
| 13:19.02 | Katty | Ariel_: s option? |
| 13:19.16 | cluecon | Katty: putting it all together, the sip peer 222 has a voicemail box 222 (this could be anything) with a password of 222 (this could be anything). |
| 13:19.41 | Katty | so...it would just say that box 210 has no password? and pull 210 from the callerid? |
| 13:19.43 | inspired | Ariel_: want to send a file through the iax or sip protocol and make sure it gets to the other end eventually, no matter how much packet loss there is |
| 13:20.03 | inspired | is that possible if I make my own codec? |
| 13:20.10 | cluecon | Katty: when they dial extension 100, it reads the calleridnum (which must be 222 or whatever the vm box number is) |
| 13:20.21 | Katty | vm box? |
| 13:20.26 | Katty | oh |
| 13:20.27 | Katty | k |
| 13:20.36 | Katty | where do you set up the callerid number? |
| 13:20.39 | cluecon | Katty: since i use the s in my extension line, it doesn't prompt for the password, without it, it would ask for the password. |
| 13:20.50 | Katty | m'kay |
| 13:21.09 | Katty | first let's get the voice app on the extensions |
| 13:21.09 | cluecon | Katty: i have a line in sip.conf under my [222] peer that says callerid=Cluecon <222> |
| 13:21.17 | Katty | is it already setup, like the echo test? |
| 13:21.34 | cluecon | you will need to add an exten line similar to the one i posted. |
| 13:21.40 | Katty | k, moment |
| 13:21.47 | lehel | people on asterisk: why my CLI> show channeltypes says me that: |
| 13:21.53 | lehel | Zap Zapata Telephony Driver w/PRI no |
| 13:22.04 | lehel | Devistate: no << |
| 13:22.06 | lehel | ? |
| 13:22.50 | *** part/#asterisk Maksim (~max@213.142.207.20) |
| 13:23.04 | cluecon | lehel: because you don't have any zap channels configured? |
| 13:23.14 | lehel | of course i have! cluecon |
| 13:23.17 | lehel | 4 channels |
| 13:23.34 | cpatry | someone already played with AEL ? |
| 13:23.36 | cluecon | pastebin your zapata.conf and your zaptel.conf |
| 13:23.40 | lehel | that's why is dubious |
| 13:23.44 | cluecon | cpatry: AEL? |
| 13:23.58 | cpatry | Asterisk Extension Language |
| 13:24.23 | cpatry | jbot, ael is Asterisk Extension Language |
| 13:24.25 | jbot | okay, cpatry |
| 13:24.31 | cpatry | ~ael |
| 13:24.32 | jbot | methinks ael is Asterisk Extension Language |
| 13:25.11 | teapot | URL for AEL ? |
| 13:25.12 | Katty | ok, i setup exten => 100,1,VoiceMailMain |
| 13:25.38 | Katty | where do i go to setup the individual's callerid number? |
| 13:25.43 | Katty | is that in sip.conf? |
| 13:25.44 | cpatry | teapot: dunno, saw it from cvs. |
| 13:25.50 | Katty | with their username/secret/pickupgroup, etc |
| 13:25.58 | cpatry | teapot: pbx pbx_ael.c |
| 13:26.14 | cpatry | see the readme too. |
| 13:26.23 | cpatry | i'll have to learn about all that. |
| 13:26.33 | Ariel_ | Katty, yes callerID="Bob Somebody"<222> |
| 13:26.41 | Katty | Ariel_: which conf is that in? |
| 13:26.46 | Ariel_ | sip.conf |
| 13:26.49 | Katty | thanks |
| 13:26.52 | cluecon | Katty: in sip.conf you will need 2 lines 1 for the callerid and 1 for the mailbox...callerid=Name<555> where 555 is the number, mailbox=555@default |
| 13:27.25 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 13:27.25 | *** mode/#asterisk [+o twisted] by ChanServ |
| 13:27.39 | inspired | hey twisted |
| 13:28.35 | Katty | Ariel_: i added callerid="Angela @ Copi-Rite"<1000> |
| 13:28.50 | Katty | Ariel_: that character won't throw it off, will it? more importantly, is that useless? |
| 13:29.01 | Katty | Ariel_: will callerid show up at another company or just internally? |
| 13:29.15 | cpatry | teapot: get HEAD |
| 13:30.01 | Ariel_ | Katty, it will show up everywhere unless you change it via the dialing rules. I have never tried that for mine. don't forget the mailbox=1000 in there as well. |
| 13:30.18 | Katty | on, where's mailbox=1000 go? in sip.conf? |
| 13:30.26 | Ariel_ | Katty, yes |
| 13:30.31 | Katty | right below callerid? |
| 13:30.41 | kajtzu | order doesnt matter |
| 13:30.59 | cpatry | AEL is intended to provide an actual |
| 13:30.59 | cpatry | programming language that can be used to write an Asterisk dialplan. |
| 13:31.01 | Ariel_ | Katty, did you read what cluecon posted |
| 13:31.04 | Katty | kajtzu: i'm double checking i have it in the right spot :) |
| 13:31.18 | Katty | Ariel_: of course, but i'm paranoid as hell :) |
| 13:31.42 | Katty | just be patient with me |
| 13:31.45 | teapot | ah, thanks cpatry |
| 13:32.24 | Ariel_ | cpatry, so your going to put in actuall programming language into the dial plan instead of normal scripting... |
| 13:32.38 | thieumS | there's no way to change the ptime value in the SDP ? |
| 13:32.58 | drumkilla | Ariel_: well, extensions.conf is getting out of hand |
| 13:33.05 | cluecon | Ariel: that is what ael is. |
| 13:33.27 | Katty | i put the mailbox=1000@copi-rite |
| 13:33.30 | drumkilla | I have already completely redone my dialplan in ael |
| 13:33.34 | Katty | now i'm going to go setup the context in voicemail.conf |
| 13:33.34 | cpatry | ariel: give a shot to AEL |
| 13:33.43 | cpatry | drumkilla: ya, just saw it, seems really great. |
| 13:33.54 | Katty | do i setup 1000= s here or do i put that in the extensions.conf? |
| 13:34.02 | drumkilla | it is soooo much easier |
| 13:34.05 | cpatry | and performance is equivalent to dialplan? |
| 13:34.12 | drumkilla | cpatry: yes |
| 13:34.14 | Katty | 1000=s,Person,email |
| 13:34.23 | Katty | or would that make 's' the password? |
| 13:34.26 | cluecon | Katty: in voicemail.conf..you need [copi-rite] |
| 13:34.30 | drumkilla | what it does is 'compiles' your extensions.ael into normal dialplan |
| 13:34.33 | Katty | cluecon: i've got that setup :) |
| 13:34.39 | Katty | cluecon: i'm right below that, trying to put my first line in |
| 13:34.57 | Katty | cluecon: i'd rather not have a password...but i don't know how to get it to skip it (don't know where the s goes) |
| 13:35.15 | cpatry | drumkilla: great. |
| 13:35.16 | Katty | 1000 =>,,Angela,email? |
| 13:35.33 | cluecon | then 1000 => 1000,Angela,email |
| 13:35.43 | Katty | they must equal then? |
| 13:35.43 | cluecon | then you'll need to modify that line you put in extensions.conf |
| 13:35.45 | cpatry | drumkilla: but the most amazing thing ive saw this week is the cdr buffer recently added. |
| 13:35.57 | cpatry | when few bugs will be fixed, that's gonna be awesome. |
| 13:35.59 | cluecon | no, they don't have to equal. |
| 13:36.06 | Katty | cluecon: let me get this straight. to use the s option, you have to have your mailbox number and password match? |
| 13:36.09 | Katty | ... |
| 13:36.11 | Katty | did i really just ask that? |
| 13:36.13 | Katty | gah. |
| 13:36.28 | cluecon | repeat: no, they don't have to equal. |
| 13:36.29 | Ariel_ | Katty, no they don't |
| 13:37.32 | Katty | ok, i have 1000 => 1000,Angela,myemailaddress@here.wherever |
| 13:38.03 | Ariel_ | Katty, exten => 100,1,VoiceMailMain(s${CALLERIDNUM}@copi-rite) would be your exten to get the voicemail |
| 13:38.13 | Ariel_ | Katty, that is fine |
| 13:38.26 | Katty | dreamy |
| 13:38.33 | Ariel_ | Katty, note the s in the new exten => |
| 13:39.10 | cluecon | Katty: now go back to extensions.conf and change the line to exten => 100,1,VoiceMailMain(s${CALLERIDNUM}) |
| 13:39.21 | Katty | umm |
| 13:39.28 | Katty | not to say i don't want your help |
| 13:39.29 | cluecon | er...yeah, what Ariel_ said. |
| 13:39.33 | Katty | but....those two lines are different |
| 13:39.37 | Katty | and now i'm all confuzzled! |
| 13:39.42 | Katty | but i shall trust Ariel_ (= |
| 13:40.05 | cluecon | Ariel_ has the right line for your setup. |
| 13:40.34 | Katty | whoohoo! |
| 13:41.02 | Katty | now i can ask if people want passwords. |
| 13:41.12 | twisted | omfg |
| 13:41.18 | twisted | AEL looks so flexible |
| 13:41.47 | cluecon | twisted: where can i read about AEL? |
| 13:41.57 | cpatry | twisted: exactly. |
| 13:42.00 | twisted | cvs lists |
| 13:42.02 | cpatry | cluecon: cvs? |
| 13:42.03 | lehel | WARNING[13722]: chan_skinny.c:3180 reload_config: Unable to get our IP address, Skinny disabled << why is that? |
| 13:42.10 | twisted | or check out latest head |
| 13:42.18 | cpatry | README.ael |
| 13:42.49 | teapot | lehel: prolly the wrong IP address in skinny.conf |
| 13:44.04 | lehel | teapot: ok! now: Jun 16 15:49:18 WARNING[13722]: chan_zap.c:10445 setup_zap: Ignoring signalling |
| 13:44.04 | lehel | <PROTECTED> |
| 13:44.26 | lehel | can't imagine.. why is this happening.. on all four channels |
| 13:44.40 | Ariel_ | wrong signalling |
| 13:44.44 | *** join/#asterisk truz24 (~raydogg@12-220-103-82.client.insightBB.com) |
| 13:44.55 | twisted | no |
| 13:45.08 | twisted | zap on reload will reload the information about a channel, but ignores the signalling |
| 13:45.13 | twisted | notice it says "Reconfigured" |
| 13:45.24 | twisted | you can't change the signalling on a running platform |
| 13:45.55 | lehel | but i don't want to change the signalling |
| 13:46.03 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 13:46.12 | lehel | signalling = fxo_ks |
| 13:46.15 | twisted | i should remove that message, as it's sorta confusing, or change it to a notice |
| 13:46.23 | twisted | lehel, yeah, and it doesn't change it |
| 13:46.53 | cluecon | twisted: how about make it say that signalling can't be changed on a reload and to change the signalling a restart is required. |
| 13:46.55 | Ariel_ | twisted, nice idea |
| 13:47.20 | lehel | Zap Zapata Telephony Driver w/PRI no (Devicestate) ... it is still "no" |
| 13:47.22 | twisted | cluecon, that message actually comes from a completely different area |
| 13:47.44 | twisted | i believe mkintf() spits that out |
| 13:47.54 | *** join/#asterisk xheliox (~jeff@user-0c6se0h.cable.mindspring.com) |
| 13:48.06 | twisted | it's been a long time since i looked at it |
| 13:48.40 | twisted | cluecon, sure, write a patch |
| 13:48.59 | twisted | and a WARNING is *NOT* an error |
| 13:49.05 | twisted | nor is a NOTICE |
| 13:49.18 | twisted | only ERROR is an error ;) |
| 13:49.18 | cpatry | cluecon: excellent idea. |
| 13:49.23 | cluecon | er...ok, i'll watch my language in the future. |
| 13:49.27 | Ariel_ | twisted, yes and how many times will people keep posting them here. |
| 13:49.37 | twisted | Ariel_, doesn't matter |
| 13:49.44 | cpatry | im sure blitz would like create such a document. |
| 13:50.02 | twisted | even if we implimented cpatry's idea, people will still post them here. |
| 13:50.04 | cluecon | Ariel_: then it becomes a matter of saying go to www.getaclue.com and put in that number to find out what the problem is. |
| 13:50.14 | twisted | possibly more so in the beginning |
| 13:50.36 | cpatry | twisted: but we can refer them to that doc. |
| 13:50.50 | twisted | cpatry, as of right now, it prints out what the problem is |
| 13:51.08 | cpatry | i suggest 4 docs: 1) ERROR, 2) WARNING, 3) NOTICE 4) DEBUG (good luck) |
| 13:51.08 | twisted | if you don't think the language is sufficient, make a patch with better explanations |
| 13:51.20 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-104-54-211.buff.east.verizon.net) |
| 13:52.29 | lehel | people i'm trying to connect two asterisk box.. with iax2... tells me: |
| 13:52.32 | wrmem | Grin, VMS style? ZAPTEL-I-SIGCHANGE, Signallng requires a restart |
| 13:52.37 | lehel | Jun 16 15:57:34 NOTICE[13748]: chan_iax2.c:6609 socket_read: Rejected connect attempt from 172.24.2.3, who was trying to reach '1000@' |
| 13:53.26 | twisted | wrmem, but is that all it requires? :P |
| 13:53.32 | cluecon | lehel pastebin your iax.conf files with private info removed. |
| 13:53.35 | [TK]D-Fender | lehel : looks like you forgot to have the other box imply which context to apply that extension to |
| 13:54.43 | *** join/#asterisk \usr\sbin (~squirrelv@202.57.81.146) |
| 13:54.54 | wrmem | If you are faking it, yes. (Not that I actually believe that would be a good solution, but then you can have someone type in the error "name" (SIP-E-NOPEER) and not a number. |
| 13:55.22 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 13:55.42 | thieumS | do you have any information about the ptime value in SIP and g729 ? |
| 13:55.45 | twisted | i believe that if you have half a clue, the current messages are easy to understand, and are in plain english |
| 13:55.46 | martinba | uarg... everytime i try 'ztcfg -vv' my machine stops responding, i have a 2.6 kernel, and my hfc card is not sharing any irq with another device. the zaphfc module loaded but it complains about zt_register. Any hints for me? |
| 13:55.57 | twisted | but like I said |
| 13:56.00 | thieumS | is it set in the code ? |
| 13:56.08 | twisted | if you feel they should be changed, or elaborated upon, write a patch |
| 13:56.33 | cluecon | twisted: the key phrase is 'half a clue'. i have not an opinion one way or the other, just an idea. |
| 13:56.40 | \usr\sbin | what is the proper way of starting * |
| 13:56.52 | twisted | speaking of half a clue. |
| 13:57.00 | Ariel_ | \usr\sbin, you mean like asterisk -vvvvgc |
| 13:57.07 | cpatry | twisted: that document, like cluecon probably thinks too, could explain how to make these LOG_* disapears. |
| 13:57.08 | Ariel_ | or safe_asterisk |
| 13:57.38 | \usr\sbin | what is safe_asterisk |
| 13:58.09 | Ariel_ | \usr\sbin, it's a script that loads asterisk with some fail restart settings |
| 13:58.31 | \usr\sbin | can i put that on the init to start at startup? |
| 13:58.41 | lehel | cluecon: [TK]D-Fender: http://pastebin.ca/14766 |
| 13:58.47 | Ariel_ | \usr\sbin, which version of os are you running |
| 13:58.59 | \usr\sbin | rh9 |
| 13:59.09 | Ariel_ | in the asterisk directory do make config |
| 13:59.26 | Ariel_ | /usr/src/asterisk |
| 13:59.45 | Ariel_ | also if you have zaptel device do the same there /usr/src/zaptel |
| 13:59.55 | Ariel_ | it will make the files for you correctly. |
| 14:00.50 | jontow | can one 'include' from voicemail.conf ? |
| 14:01.18 | Ariel_ | jontow, include the voicemail.conf???? |
| 14:01.24 | jontow | no, include *from* it |
| 14:01.36 | Ariel_ | why? |
| 14:01.39 | jontow | ie. include other files as contexts, much like one can with extensions.conf, but not IN extensions.conf at all :D |
| 14:01.47 | jontow | for external-tool-parsing reasons |
| 14:02.04 | jontow | would just give me less work; so i ask.. thats all :) |
| 14:02.18 | jontow | if it can't be done, then i'll be doing it the harder way |
| 14:02.19 | jontow | :D |
| 14:02.31 | Ariel_ | you mean you want to in the voicemail.conf include other files? |
| 14:02.35 | Ariel_ | the reply is yes |
| 14:02.37 | jontow | yes |
| 14:02.45 | jontow | like, in voicemail.conf: |
| 14:02.52 | jontow | include "othercompanysvoicemailboxes.conf" |
| 14:02.55 | Ariel_ | #include =/path/filename |
| 14:02.58 | jontow | for a 'virtual hosting' effect |
| 14:03.37 | Ariel_ | jontow, yes but you need to only put mail box context in those files. |
| 14:03.39 | Katty | looks like we're going to use passwords :< |
| 14:04.44 | martinba | when building zaphfc i get warnings that zt_register, zt_transmit, zt_receive, zt_ec_chunk and zt_unregister are undefined ... should i ignore that? |
| 14:06.05 | *** part/#asterisk REdOG (~REdOG@REdOG.user.gentoo) |
| 14:06.07 | *** join/#asterisk gatty (~agatward@tomcat.rdg.ac.uk) |
| 14:06.26 | gatty | afternoon all |
| 14:06.30 | lehel | [TK]D-Fender: how did you say?.. what sould i do? |
| 14:07.18 | gatty | do I have to do anything special to get the new "UK-approved" TDM400P to be recognised? modprobe wctdm doesn't see it but it does show up in lspci output. |
| 14:08.37 | *** join/#asterisk gabb0 (~gabb0@indo1.indosoft.unb.ca) |
| 14:08.45 | gabb0 | hello all |
| 14:09.32 | martinba | does anybody here has experience with zaphfc cards? |
| 14:09.32 | *** join/#asterisk clive- (~pirch@rrba-146-83-76.telkomadsl.co.za) |
| 14:10.23 | clive- | does anyone heer have expereince with the eicon 4bri card ? |
| 14:11.13 | tzafrir | speaking of zaphfc: I've finally started adding zaphfc and zaptel/pri in general support to genzaptelconf. |
| 14:12.08 | tzafrir | I have something that should work for zaphfc cards and looking for testers |
| 14:12.38 | martinba | i would be glad if i could test :) |
| 14:13.16 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 14:13.41 | gatty | forget my question - upgrading to latest CVS zaptel fixed it. |
| 14:13.52 | *** join/#asterisk KeX (~KeX-mobil@argos.campus-sbg.at) |
| 14:14.02 | [Jedi] | martinba: I do |
| 14:14.02 | gabb0 | looking for some knowledge on echo and echo cancellation |
| 14:14.19 | *** join/#asterisk davewise (~icechat5@98.agcllc.net) |
| 14:14.42 | gabb0 | I have a strange issue where if echo cancel is on, longdistance sounds like crap but local doesn't and if echo cancel is off then the opposite occurs |
| 14:14.44 | tzafrir | http://tzafrir.org.il/genzaptelconf (version 0.3.0) . Should generally work. Doesn't try to load the module , though |
| 14:14.46 | [Jedi] | martinba: I have it working with a 13EUR single-BRI card :D |
| 14:15.00 | *** join/#asterisk jmacz (~jmacz@63.245.86.153) |
| 14:15.19 | tzafrir | Well, I need to get going. I hope to come back later (as tzafrir_laptop |
| 14:15.20 | tzafrir | ) |
| 14:15.55 | martinba | [Jedi], superb :) is it safe to ignore these zt_register etc. warnings when comiling / loading the module? |
| 14:16.04 | martinba | tzafrir, i will give it a try |
| 14:16.38 | [Jedi] | martinba: I don't get any of these |
| 14:16.53 | [Jedi] | [root@ccard01 sounds]# dmesg |grep zt_register |
| 14:16.54 | [Jedi] | [root@ccard01 sounds]# |
| 14:16.58 | davewise | has anyone experienced a problem on SIP calls where, when you hear your SIP phone ring and answer it, there is silence for a period of time and then all the audio kicks in (like 4 to 20 seconds) |
| 14:17.19 | davewise | This is on a Sipura1000 |
| 14:17.28 | [Jedi] | martinba: CentOS 3.4 with 2.4.21-32.0.1.EL kernel here |
| 14:18.23 | coppice | gabb0: are your long distance calls being carrier across a VoIP LD backbone? if so, echos on them generally won't cancel, and attempts to do so go bananas |
| 14:19.06 | martinba | [Jedi], ive a debian system with 2.6.11.12 kernel .. with 2.4.31 .. the same warning occured |
| 14:19.53 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:21.10 | gabb0 | coppice, I believe ld travels over ds3 in this case. a tech from the ld company has told us they are not doing any echo cancel at all on their end |
| 14:21.52 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:21.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:21.54 | coppice | gabb0: that isn't what I asked? |
| 14:21.54 | gabb0 | coppice, so I'm not sure why this is happening. very strange |
| 14:22.52 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 14:23.07 | MattH | Hi,... what can I use to load test my asterisk box? astertest doesn't seem to be working correctly... |
| 14:23.11 | gabb0 | coppice, sorry. some background I guess, we are using sip <-> asterisk <-> PRI <-> telco |
| 14:23.18 | brimstone | MattH: call files |
| 14:24.11 | MattH | I considered that... but does that real do a realist load test? |
| 14:24.14 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 14:24.15 | MattH | I guess I could call another asterisk server |
| 14:24.20 | coppice | its not strange. your long distance link is *not* over a DS3, though your local access to it may be. Its probably over an OC192 these days. question is, is it carried across that OC192 as VoIP or as circuit switched data |
| 14:24.29 | brimstone | MattH: yeah, or call out and back in |
| 14:24.45 | MattH | but how do I know when it breaks? call in on a voip phone and talk? |
| 14:25.01 | _omer | anybody please tell me what kind of error is it ? |
| 14:25.01 | _omer | Jun 16 07:29:33 NOTICE[7603]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 14:25.06 | _omer | ?? |
| 14:25.36 | *** join/#asterisk fa__ (faceoff@devel.acdbddh.eu.org) |
| 14:25.38 | fa__ | elou |
| 14:26.00 | brimstone | _omer: turn off silence suppression on the phone |
| 14:26.07 | fa__ | make -C SUBDIRS=/usr/src/zaptel modules |
| 14:26.07 | fa__ | make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. |
| 14:26.07 | fa__ | make: *** [linux26] Error 2 |
| 14:26.09 | fa__ | what's bad |
| 14:26.23 | _omer | brimstone: in asterisk or phone? |
| 14:26.31 | brimstone | _omer: on the phone |
| 14:26.58 | _omer | I use X-PRO |
| 14:26.58 | gabb0 | coppice, ok, I'm not sure. so what you are saying is if it is voip then my echo canceling is making things worse? So how to remedy? if I turn echo cancel off then local sounds like crap? |
| 14:27.12 | *** join/#asterisk bonez39 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 14:27.43 | coppice | gabb0: you must seek your own path in life, grasshopper :-) |
| 14:28.14 | _omer | brimstone: thanks..let me check... |
| 14:28.54 | gabb0 | coppice, ha ha. yeah, that is what I was worried about. the "every case is unique" issue. I noticed there was no more advancement on the patch to turn ec on or off in the dial command |
| 14:29.07 | gabb0 | that would likely have been the solution to my problem |
| 14:29.14 | gabb0 | well one possible one |
| 14:30.31 | *** part/#asterisk popooya (~popooya@08b8ae1d0a6311b7.session.tor) |
| 14:30.55 | Katty | hmm. |
| 14:31.01 | Katty | it's not dialing |
| 14:31.10 | Katty | it says it's dialing |
| 14:31.20 | Katty | it says it answered it |
| 14:31.20 | Hmmhesays | it lies... have nothing to do with it |
| 14:31.28 | *** join/#asterisk beto75_ (~beto75@201.133.240.41) |
| 14:31.31 | Katty | but...there's no ringring! ringring! |
| 14:31.39 | beto75_ | hello guys |
| 14:31.45 | jerry_hotlinks | quick basic dumb question - how to exit the cli without shutting * down please? |
| 14:31.49 | Katty | <PROTECTED> |
| 14:31.49 | Katty | <PROTECTED> |
| 14:31.49 | Katty | <PROTECTED> |
| 14:31.50 | Hmmhesays | ! |
| 14:31.51 | Katty | :<<< |
| 14:32.00 | Katty | LIES |
| 14:32.02 | Katty | ALL LIES |
| 14:32.13 | brimstone | jerry_hotlinks: "exit" if it compains, don't start asterisk with the -c option |
| 14:32.14 | *** join/#asterisk skiold (~userid@84-121-68-176.onocable.ono.com) |
| 14:32.16 | Hmmhesays | put a phone on the line and dial it manually |
| 14:32.18 | jerry_hotlinks | k |
| 14:32.22 | Katty | Hmmhesays: it works |
| 14:32.23 | Hmmhesays | jerry_hotlinks: ! |
| 14:32.29 | jerry_hotlinks | ok |
| 14:32.36 | Katty | Hmmhesays: just...sporadically does this to me |
| 14:32.52 | Katty | like...if the moon is exactly some distance to the sun it will work |
| 14:32.55 | Katty | otherwise, DOOM |
| 14:33.05 | Katty | like see, now it works |
| 14:33.09 | Katty | madness! |
| 14:33.18 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 14:34.05 | Hmmhesays | is at always going out the first port? |
| 14:34.13 | Hmmhesays | cause you have all 4 channels in that group |
| 14:34.18 | Hmmhesays | last I checked |
| 14:34.54 | jerry_hotlinks | thanks |
| 14:35.11 | Katty | Hmmhesays: yes, it is |
| 14:35.18 | Katty | Hmmhesays: i have to figure out how to fix that sooner or later |
| 14:35.40 | brookshire | http://www.digium.com/downloads/AstriconEurope2005Tutorial.pdf |
| 14:35.47 | brookshire | kevin's tutorial is up :) |
| 14:36.09 | cpatry | brookshire: thx |
| 14:37.56 | *** join/#asterisk kn0x (node4@adsl-66-73-198-246.dsl.chcgil.ameritech.net) |
| 14:38.27 | *** join/#asterisk marzl (~marcel@p5085C597.dip.t-dialin.net) |
| 14:38.45 | kn0x | drumkilla- you there? |
| 14:40.20 | brookshire | drumkilla the nubbkilla |
| 14:42.19 | kn0x | i was looking for the Callback application.. i cant get it to work here |
| 14:42.40 | kn0x | anyways i still have the problem of getting inband DTMF from FWD to me... |
| 14:42.47 | kn0x | im using ulaw... |
| 14:43.22 | Hmmhesays | ROCK |
| 14:43.39 | [illuminatus] | how come when I make an outgoing call, people can hear me but I can't hear them? |
| 14:44.08 | kn0x | you have your volume to low.... or maybe you need a hearing aid |
| 14:44.12 | kn0x | aahhaha |
| 14:44.13 | *** join/#asterisk togusa (~Togusa@labo-unix.org) |
| 14:44.17 | togusa | hello there |
| 14:44.25 | kn0x | hola |
| 14:44.50 | togusa | is there someone to help me please ? I just have a question about a TE110P card |
| 14:45.02 | [illuminatus] | geee... you're a bunch of help |
| 14:45.08 | togusa | ok |
| 14:45.21 | [illuminatus] | could it be because I can't receive incoming calls? |
| 14:45.27 | togusa | nope |
| 14:45.41 | togusa | in fact I'm from France, I recently ordered this card |
| 14:45.48 | togusa | now I want to buy a 1U server |
| 14:45.50 | teapot | bye all |
| 14:46.02 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 14:46.04 | togusa | Dell has a good one very cheap with a PCI-X slot |
| 14:46.16 | togusa | I just want to know if it is compatible ... |
| 14:46.21 | togusa | can you tell me ? |
| 14:47.00 | togusa | on the Dell website it is written 1 slot PCI-X 64bit/133Mhz |
| 14:47.18 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmpb.dialup.mindspring.com) |
| 14:47.48 | *** join/#asterisk james_ed (~james_ed@wsip-68-110-218-186.ks.ok.cox.net) |
| 14:48.20 | togusa | allo ? |
| 14:49.50 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 14:50.28 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-134.range217-44.btcentralplus.com) |
| 14:50.35 | SpaceBass | i'm tryting to add a caller id preface to one of my zap lines. I have callerid = "MCK" but it doesnt seem to work |
| 14:50.39 | SpaceBass | is my syntax incorrect? |
| 14:52.10 | gabb0 | coppice, |
| 14:52.21 | gabb0 | coppice, thanks for the info |
| 14:52.24 | coppice | gabb0 |
| 14:54.20 | SpaceBass | anyone ever made a cable to use a standard PoE intjector with a cisco phone? as per http://www.voip-info.org/wiki-Cisco+POE |
| 14:55.56 | cluecon | SpaceBass: on the callerid...are you trying to change the callerid when you dial, or making that particular zap always have the same caller id? |
| 14:56.28 | marzl | togusa, we are running a dell poweredge 2850 with te110p without any problems |
| 14:57.18 | SpaceBass | cluecon trying to add a prefix to incoming calls from ZAP/2 |
| 14:57.43 | togusa | ok marzl thank you very much |
| 14:57.50 | togusa | have a nice day ! |
| 14:57.58 | *** part/#asterisk togusa (~Togusa@labo-unix.org) |
| 14:58.01 | SpaceBass | cluecon I'd prefer to add it as a prefix, but if I have to overwrite it completely, then I will |
| 14:58.03 | *** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net) |
| 14:59.40 | Abbas | how can we kill the call when callers balance hits 000 |
| 14:59.43 | JerJer | show application SetCallerID |
| 15:00.28 | SpaceBass | JerJer thanks |
| 15:00.35 | cluecon | SpaceBass, try using SetCallerID |
| 15:00.40 | JerJer | SetCallerID(1${CALLERIDNUM}) |
| 15:00.40 | cluecon | doh. |
| 15:00.43 | JerJer | cluecon: TO SLOW |
| 15:00.43 | *** join/#asterisk asdfblah (~UNIX@pcp04541662pcs.brmngh01.mi.comcast.net) |
| 15:00.45 | JerJer | too |
| 15:01.01 | asdfblah | anyone know is the avaya 4624 will work with asterisk? |
| 15:01.06 | asdfblah | if* |
| 15:01.19 | SpaceBass | but I am determined to make this cisco 7940 work first |
| 15:01.25 | Ariel_ | SpaceBass, do you use amp? |
| 15:01.31 | drumkilla | SpaceBass: get a red bull |
| 15:01.44 | SpaceBass | Ariel_ for some stuff |
| 15:01.51 | drumkilla | we're working on our second tower of red bull in this office ... |
| 15:02.00 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 15:02.10 | cluecon | Abbas: SoftHangup and some AGI Scripting. |
| 15:02.18 | MattH | is there a way to see what a zap channel is doing for outbound calling? ie.. when I do zap show channels it only shows caller-id stuff for inbound |
| 15:02.29 | Ariel_ | due to in amp you can put a pre fix to a ring group. When your zap port call comes in send it to that ring group can be for just one device |
| 15:02.35 | JerJer | SetCallerID |
| 15:02.38 | JerJer | is there an echo in here? |
| 15:02.51 | JerJer | AMP is a joke |
| 15:03.10 | DarkSpectre | argh |
| 15:03.19 | SpaceBass | Ariel_ how do I direct to a ring group? |
| 15:03.21 | DarkSpectre | anyone ever tried intergration of sphinx and asterisk? |
| 15:03.46 | SpaceBass | I have a ring group set up for these phones/lines in question but I had to manually write the context |
| 15:04.07 | SpaceBass | so I'm just using dial(SIP/xxxx&SIP/zzzz)... etc |
| 15:04.42 | DarkSpectre | i found some documentation on it but it was for sphinx2 |
| 15:05.20 | [TK]D-Fender | DarkSpectre : Could you link me on it? even if its old I'd like to see |
| 15:06.27 | [illuminatus] | so, does anyone have any idea why i can make outgoing calls but I can't hear anything I can only send but not receive |
| 15:06.51 | Hmmhesays | JerJer: amp is great if you are trying to impress someone who knows nothing but has the power to give you money to set up asterisk |
| 15:06.56 | SpaceBass | [illuminatus] using a sip provider? |
| 15:07.06 | Hmmhesays | you don't necessarilly have to use it <chuckle> |
| 15:07.19 | [illuminatus] | SpaceBass: yes |
| 15:08.00 | SpaceBass | [illuminatus] sounds like you are blocking rtp packets...or even port 5060 |
| 15:08.22 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 15:08.54 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 15:09.30 | brookshire | yes.. asterisk developers do take donations of redbull. |
| 15:09.39 | [illuminatus] | no, UDP 5060 is forwarded all the way to the * box |
| 15:09.42 | Hmmhesays | and jager I hear |
| 15:09.59 | Abbas | cluecon thanks |
| 15:10.13 | [illuminatus] | SpaceBass: what if I don't register with the provider, would that do it? |
| 15:10.26 | SpaceBass | probably |
| 15:10.32 | *** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) |
| 15:13.19 | [illuminatus] | my VoIP provider gave me nothing but an IP address and my phone number. So I've been trying to figure this out for about 2 days now |
| 15:13.32 | *** join/#asterisk QuaG|NaR (~quag1nar@adsl-146-232-20.mob.bellsouth.net) |
| 15:13.58 | DarkSpectre | [[TK]D-Fender]: sure, one sec |
| 15:13.58 | Hmmhesays | who is that? |
| 15:14.03 | DarkSpectre | http://turnkey-solution.com/asterisk-sphinx.html |
| 15:14.10 | [illuminatus] | Hmmmhesays: the provider? |
| 15:14.14 | Hmmhesays | yeah |
| 15:14.17 | SpaceBass | damn... that crazy PoE cable from http://www.voip-info.org/wiki-Cisco+POE worked! |
| 15:14.33 | [illuminatus] | CentricVoice |
| 15:14.42 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 15:15.10 | Hmmhesays | SpaceBass: awesome I was thinking about doing that for a few 7960's I have here |
| 15:15.41 | SpaceBass | Hmmhesays I tried it yesterday and didnt work... gave up |
| 15:15.51 | SpaceBass | but then I thought... what if the cable was bad... made a new one and it worked! |
| 15:15.51 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 15:16.00 | SpaceBass | now I just need the sip firmware... (ANYONE? ANYONE?) |
| 15:16.23 | _omer | sip firmware of which device? |
| 15:16.23 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 15:16.31 | Hmmhesays | SB: pm me |
| 15:17.10 | obsidian-studios | greetings all, anyone know if there is a way to increase the volume on WAV formatted voicemails? If I used standard wav format volume is great, if I use WAV format volume is extremely low |
| 15:18.00 | obsidian-studios | it's the file's playback volume I am concerned with, normally I have to run through sox's with a volume of 9 which is like it's max and that brings it to a normal volume level? |
| 15:19.10 | [illuminatus] | can anyone help me with this registration string? |
| 15:19.10 | [illuminatus] | http://pastebin.com/pastebin.php?dl=300349 |
| 15:19.33 | jerry_hotlinks | I could use some 1 on 1 help with capturing dtmf tones entered by the caller then sending them to an .asp script, anyone care to step up? |
| 15:20.00 | obsidian-studios | man lots of people needing help, and it's quite in here, very un * like? |
| 15:20.11 | [illuminatus] | this is the info they gave me BTW http://pastebin.com/pastebin.php?dl=300353 |
| 15:20.51 | eKo1 | Well, maybe because everyone is busy at the moment. |
| 15:20.52 | *** join/#asterisk kn0x (node4@adsl-66-73-198-246.dsl.chcgil.ameritech.net) |
| 15:21.11 | obsidian-studios | eKo1: :) np, just found it odd, usually this channel is going off |
| 15:21.52 | cluecon | illuminatus: i didn't see a registration in either of those pastebins? |
| 15:22.28 | cluecon | obsidian: not sure. what is the difference between wav and WAV? |
| 15:22.28 | [illuminatus] | the first one it's trying to register but it getting 501 and 403 messages |
| 15:22.42 | cluecon | illuminatus: who is the provider? |
| 15:22.45 | freat | hmm... "SIP/2.0 501 Not Implemented" |
| 15:22.47 | [illuminatus] | the second one is a sip debug dump they sent me when they tried to call me |
| 15:22.58 | [illuminatus] | CentricVoice is the provider |
| 15:23.22 | _omer | is there any one who could provide me a SCRIPT to use SOX in extensions.conf ..etc |
| 15:23.26 | obsidian-studios | cluecon: file size, WAV's are much smaller and quality is not as good as wav, but volume differences is my main problem |
| 15:23.38 | [illuminatus] | i'm trying to use this as my register string: 2143294838:@209.120.255.14/2143294838 |
| 15:24.07 | freat | I don't think there should be a colon between the number and @ |
| 15:24.32 | [illuminatus] | i tried it without the : too |
| 15:24.35 | [illuminatus] | that didn't work either =/ |
| 15:24.42 | obsidian-studios | cluecon: since all voicemails are emailed, and immediately deleted off the * machine, I elected for smallest file size. But to listen to all volumes on the end PC must be maxed, or use sox to bring up volume before listening? |
| 15:25.53 | *** join/#asterisk stkn (~stkn@stkn-active-pdpc.developer.gentoo) |
| 15:26.02 | JerJer | why don't you unfuck voicemail with imap storage of messages? |
| 15:26.24 | cluecon | obsidian: why not record in wav and re-encode into mp3 or something if you really must email them? |
| 15:26.28 | freat | imap storage? sounds cool |
| 15:26.47 | *** join/#asterisk agave-txlink (phanop@216.81.47.201) |
| 15:26.47 | MikeJ[Laptop] | there are a number of issues, one related to format, one related to a 10 db loss when using zap to record e-mail |
| 15:26.55 | obsidian-studios | JerJer: physicall locations, there is no email server on the network of the * machine |
| 15:27.01 | MikeJ[Laptop] | as of yet, no one has been willing to fix it or pay for it to be fixed |
| 15:27.02 | agave-txlink | okay. is there some kind of trick to get CVS-head to be able to send calls to CVS-stable-1.0.7 via IAX? |
| 15:27.05 | MikeJ[Laptop] | so it be broke |
| 15:27.09 | MikeJ[Laptop] | sorry... |
| 15:27.14 | JerJer | but it has to talk to the 'net to email |
| 15:27.19 | cluecon | illuminatus: sip or iax registration? |
| 15:27.20 | freat | in voicemail.conf you can choose what formats get recorded.... |
| 15:27.23 | MikeJ[Laptop] | put a bounty up |
| 15:27.27 | [illuminatus] | Cluecon: SIP registration |
| 15:27.30 | obsidian-studios | MikeJ[Laptop]: ah so there are underlying issues |
| 15:27.34 | JerJer | so why couldn't it simply talk to an imap server and/or smtp server - not exactly sure of the implemenation |
| 15:27.36 | MikeJ[Laptop] | yes |
| 15:27.58 | obsidian-studios | JerJer: file size to upload, wav is considerably bigger and takes longer |
| 15:27.59 | MikeJ[Laptop] | one related to format (one of the wav formats works better), and one related to zap |
| 15:28.03 | freat | :q |
| 15:28.13 | JerJer | use speex |
| 15:28.22 | JerJer | (requires development) |
| 15:28.40 | MikeJ[Laptop] | one of the wav formats has code that brings up the volume when it saves to file. That seems to do the trick to some extent |
| 15:28.55 | MikeJ[Laptop] | somone could port that code to the other format_*.c?? |
| 15:29.00 | obsidian-studios | MikeJ[Laptop]: I am using a zap channel, but I believe it's more of a conversion/formatting issue than volume on zap channel? As wav and other format, gsm I think? Volumes are fine, but gsm's do not play easily on windows. I am pretty sure I can play on my linux desktop |
| 15:29.00 | cluecon | illuminatus: did centricvoice give you directions on how to register with them? |
| 15:29.27 | MikeJ[Laptop] | ok, so you have issue #2. |
| 15:29.31 | obsidian-studios | MikeJ[Laptop]: ah ha, that must be the standard wav format |
| 15:29.38 | MikeJ[Laptop] | there is a closed bug in mantis where you can find more info |
| 15:29.50 | obsidian-studios | MikeJ[Laptop]: it's the WAV format I am hurting on, but might look into this myself |
| 15:29.58 | MikeJ[Laptop] | like I said, one of the wav formats has code to crank up the volume |
| 15:30.28 | *** join/#asterisk algorithmn (~na@ool-44c29ac5.dyn.optonline.net) |
| 15:30.39 | obsidian-studios | MikeJ[Laptop]: wonder if it's specific to that format? Or if that same code is generic enough to be applied to both wav formats? |
| 15:31.06 | *** join/#asterisk djin (~djin@196.Red-80-37-200.pooles.rima-tde.net) |
| 15:32.15 | Ariel_ | There are 2 wav file types one is wav49 which is compressed and the sound is lower the normal wav is louder but files are very large |
| 15:33.07 | obsidian-studios | Ariel_: exactly and I am looking into if anything can be done to increase the volume on the wav49 ones, but not sure if that goes against the smaller file size or what? |
| 15:33.08 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
| 15:33.09 | calisto | anyone here familiar with debugging app_sms |
| 15:33.17 | obsidian-studios | MikeJ[Laptop]: thanks for the info |
| 15:33.57 | *** join/#asterisk QuaG|NaR (~quag1nar@adsl-146-232-20.mob.bellsouth.net) |
| 15:34.03 | Ariel_ | obsidian-studios, if you find something let me know. |
| 15:34.33 | [illuminatus] | WTF? Now these guys are saying "There is no registration necessary. I will send an invite and Dimitriy just |
| 15:34.33 | [illuminatus] | needs to accept it from 209.120.255.14. |
| 15:35.07 | obsidian-studios | Ariel_: seems like I will have to go into code base to see about changes etc, if you saw what MikeJ[Laptop]: mentioned something was done to one of the wav formats to increase volumes, but there are also issues with the volume of the zap channels |
| 15:35.40 | obsidian-studios | Ariel_: I am hoping what ever was done to the one format to increase volume can be done to the other? I will have to look in code or etc |
| 15:35.55 | freat | turn it to 11 ! |
| 15:37.14 | jontow | lotsa testing today.. hmm |
| 15:38.21 | *** part/#asterisk beavizd (~anders@212.242.87.250) |
| 15:38.55 | *** join/#asterisk BuckRogers (~steve@ool-44c29ac5.dyn.optonline.net) |
| 15:39.44 | MikeJ[Laptop] | http://bugs.digium.com/view.php?id=2023 |
| 15:39.50 | MikeJ[Laptop] | the details are in there |
| 15:41.22 | MikeJ[Laptop] | zap volume issue was specific to vm tho... weird. |
| 15:41.56 | MikeJ[Laptop] | very good details in that bug report... if you can come up with a patch, I am sure it would be appretiated |
| 15:42.03 | *** join/#asterisk AgiNamu (~Michael@200.6.216.203) |
| 15:42.18 | AgiNamu | Am I correct in reading the CVS list? There's a new 'language' for extensions? |
| 15:42.25 | obsidian-studios | MikeJ[Laptop]: ok thanks, not allot of time for this, but at the level it's irritating me, I might escalate it ;) |
| 15:42.56 | BuckRogers | good morning all |
| 15:43.02 | BuckRogers | long time no see |
| 15:44.12 | algorithmn | mr rogers.. im bucking my enthusiasm right now... |
| 15:44.31 | BuckRogers | curve ur enthusiasm |
| 15:44.50 | algorithmn | just like that chick last weekend... |
| 15:45.28 | algorithmn | 'sloppy drunk * on a saturday night' - sublime... |
| 15:46.06 | BuckRogers | yeah i like to complile drunk, |
| 15:46.18 | BuckRogers | then spew bile |
| 15:46.24 | BuckRogers | after walking a mile |
| 15:46.31 | algorithmn | through the ghetto... |
| 15:46.35 | BuckRogers | with my friend lile |
| 15:47.30 | algorithmn | how much is too much.. |
| 15:47.32 | BuckRogers | yeah well you would too when doing none stop deve for a year now |
| 15:47.46 | algorithmn | i drink on the job... |
| 15:47.56 | bublbobl | ;-) (even without this reason :-P ) |
| 15:48.35 | BuckRogers | JerJer you watching the room |
| 15:48.48 | BuckRogers | did nufone start accepting customers agian? |
| 15:49.06 | drumkilla | bublbobl: why do you say that * coders drink too much? |
| 15:49.23 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 15:49.36 | algorithmn | i think it may be my fault for that one drumkilla |
| 15:49.39 | SpaceBass | anyone know where the tftp server logs are by default on *@home? |
| 15:49.47 | tzafrir_laptop | hi |
| 15:50.11 | bublbobl | drumkilla> cause of the docs ! :-D No I'm jokin' of course. |
| 15:50.42 | obsidian-studios | bublbobl: or is it because it works ;) |
| 15:50.48 | bublbobl | I'm afraid my VoiceMailMain doesn't listen to my password when I dial it, I have "incorrect password '' for user '1001' what ever I type. I don't know how to troubleshoot. |
| 15:50.58 | obsidian-studios | not sure I trust the sober :) |
| 15:51.20 | SpaceBass | bublbobl sounds like the wrong setting for DTFM |
| 15:51.27 | SpaceBass | bublbobl try changing it to inband |
| 15:51.38 | bublbobl | SpaceBass> I thought of this but it works for IVR, gonna try anyway |
| 15:51.51 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 15:53.12 | davewise | bublbobl: Have you tried making the password a single digit? |
| 15:53.33 | [illuminatus] | WTF no outbound calling won't work because it's "congested" |
| 15:54.39 | *** join/#asterisk Dom` (~poop@mobileweb02.london.02.net) |
| 15:55.03 | Dom` | Does musiconhold not line variable bit rate mp3's? |
| 15:55.04 | davewise | has anyone experienced whare a sip phone rings (thats registered to an * server) and when you pick it up, the person calling in from the PSTN gets a message that the person is not available now? |
| 15:55.17 | Ariel_ | obsidian-studios, as the bug reports there is no fix yet for the low volume |
| 15:56.00 | obsidian-studios | Ariel_: yep a task for us |
| 15:56.03 | [TK]D-Fender | Dom` : I believe converting them to non-VBR is a good idea, and you must remove the ID3 tag if any |
| 15:56.24 | Dom` | When it plays its as if it only plays the louder noises |
| 15:56.27 | Dom` | its quite odd |
| 15:56.30 | Dom` | as if ajust a symbol is playing |
| 15:56.41 | SpaceBass | Hmmhesays is it usual that the phone would reboot itself several times? |
| 15:56.43 | Dom` | Is there a program which does that? |
| 15:57.48 | Ariel_ | obsidian-studios, I just tested the sound files from an sipura 3000 and it's also very low. But if it's left from a normal sip phone it's loud and normal sounding |
| 15:58.16 | QuaG|NaR | good morning, could somebody help me figure out why i can modprobe my tdm400p and run ztcfg -vvv and see the channels setup but in the asterisk cli if i type zap show channels i only have the pseudo interface listed |
| 15:58.36 | obsidian-studios | Ariel_: hmm, regardless of wav format? When I record stuff internally via a analog -> fxs -> sip connection it's the same volume as on a zap channel? |
| 15:58.45 | davewise | Does anyone know how to set GLARE settings on a PRI in * |
| 15:58.46 | Jas_Williams | QuaG|NaR: Error in zapata.conf |
| 15:59.01 | QuaG|NaR | Jas_Williams, great thanks |
| 15:59.07 | Dom` | [TK]D-Fender? |
| 15:59.10 | obsidian-studios | Ariel_: from what I have seen no matter the source, the volume is low on wav49 as compared to wav. Guess it can get worse than what I have seen |
| 15:59.32 | Ariel_ | obsidian-studios, thats correct. |
| 15:59.40 | cluecon | davewise: what kind of glare settings? |
| 16:00.05 | davewise | QuaG|NaR: wehen you run zttool, what does it show? do you have the channels programmed in zapata.conf? |
| 16:00.05 | obsidian-studios | Ariel_: pretty sure we will have to modify/play with code and etc to make progress on this |
| 16:01.05 | bublbobl | SpaceBass> That was it, thank you :-) |
| 16:01.16 | QuaG|NaR | davewise, it shows 5 channels configured, 1 x100p first channel, and channels 2-5 tdm400p, 5 channels configured |
| 16:01.19 | QuaG|NaR | not exactly like that |
| 16:01.19 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:01.22 | QuaG|NaR | but i think you get |
| 16:01.26 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 16:01.32 | Ariel_ | davewise, your either pick a channel from the top number down or up from lower to top. We use dial,zap/g1 for lower to top and zap/G1 from top to bottom opposit of what the channels come in via the telco |
| 16:01.44 | davewise | cluecon: On a lot of Telco equipment for T1/PRI, there are settings for wether GLARE is handled by the CO or the CPE, the other has to yield GLARE |
| 16:02.23 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfivt.dialup.mindspring.com) |
| 16:03.05 | bublbobl | davewise> sorry for delay for my answer, pb came from my phone settings (used "inaudio for dtmf instead of rtp (rfc2833) |
| 16:07.03 | sivana | what's the best software based fax for use with *? |
| 16:07.48 | JerJer | app_txfax |
| 16:08.01 | eKo1 | My provider is sending me calls through SIP. The URI has the format sip:<DID number>@<provider.com>:5060. I've already made a entry in sip.conf for * won't answer the call. |
| 16:08.10 | sivana | JerJer: is that SpanDSP? |
| 16:08.31 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
| 16:08.32 | eKo1 | s/for/but |
| 16:09.00 | eKo1 | It just get's stuck in SIP 180 Ringing. |
| 16:09.14 | AgiNamu | How many people here provide VoIP services? |
| 16:09.18 | Dom` | Can anyone tell me a program for removing id3 tags/converting from vbr for use with musiconhold? |
| 16:09.50 | *** join/#asterisk tugalone (~tugalone@219.65.136.203) |
| 16:10.52 | eKo1 | I do. |
| 16:11.48 | Micc | Does AT&T Callvantage use asterisk? |
| 16:12.03 | AgiNamu | eKo1, are you capable of setting ANI? |
| 16:12.32 | AgiNamu | Cause I've heard there are lot of providers doing VoIP that don't have the ability to set ANI (like for emergency services) |
| 16:12.47 | eKo1 | Nope. I don't even know if ANI exists here. |
| 16:13.09 | AgiNamu | oh, you're not usa-based... i shoulda asked |
| 16:13.55 | agave-txlink | you can set CPN but not ANI |
| 16:15.20 | *** join/#asterisk sparrow (sparrow@mortar.walled.net) |
| 16:18.07 | *** part/#asterisk thieumS (~darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 16:19.05 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
| 16:19.08 | exonic | What's up folks |
| 16:19.32 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 16:19.44 | exonic | for the last half hour i've been reading on debugging RTP streams, I've got jittery sound and I think it's due to packet loss/latency. |
| 16:20.06 | exonic | My question is does anyone know of some good software to help me debug this information? |
| 16:20.13 | eKo1 | Well, why don't you do a path analysis and find out. |
| 16:20.43 | *** join/#asterisk leandro_pt (~leandro@82.155.114.204) |
| 16:21.20 | Ariel_ | it's lunch time |
| 16:21.29 | eKo1 | software? all you need is ping and mtr. |
| 16:21.54 | exonic | eKo1, it waves in and out, sometimes it's solid, other times it's perfect. I think it's Comcast QoS. |
| 16:22.10 | *** join/#asterisk Exstatica (exstatica@65.119.22.200) |
| 16:22.36 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 16:22.44 | leandro_pt | hi.. any PHP ppl out there know how to access the asterisk berkley Database?? |
| 16:22.46 | eKo1 | OK then. Contact them and tell them to get their assess in gear. |
| 16:23.16 | eKo1 | leandro_pt: well, isn't there a set of functions to do that? |
| 16:23.27 | exonic | eKo1, =). Well that leads me to my first question, if there were some monitoring software where I could show them latency /packet loss.. i'd be well of in convincing 'em. |
| 16:24.05 | exonic | eKo1, i'll see what I can find on the server while it's happening. Thanks a ton |
| 16:24.27 | eKo1 | get smokeping |
| 16:24.35 | eKo1 | I used to monitor latency on my network. |
| 16:24.41 | exonic | ok, cool. |
| 16:24.45 | leandro_pt | eKo1, i wouldn't be asking if I knew.. however I don't want to use system + database get |
| 16:25.27 | eKo1 | what do you mean "system + database get"? |
| 16:25.29 | leandro_pt | knowing with dba (flatfile, cdb?) type to use would be a big help |
| 16:26.05 | leandro_pt | eKo1, my CID names are stored in the regular asterisk database |
| 16:26.05 | [TK]D-Fender | Hey got a quick question : I'm looking to set up a server with 1 TE100P, 1 TDM04B, 1 TDM40B, (4 line analog in/out), Any issues with that many cards (of those mixed types) in a system? Also I need suggeestions for a good motherboard to run those on. |
| 16:26.22 | leandro_pt | if you do a "database show/get/put" in the CLI you can manipulate the DB |
| 16:26.52 | eKo1 | leandro_pt: OK. I understand now. I have no idea where * keeps its local database. |
| 16:27.01 | leandro_pt | i want to access that data through PHP without having to do a system("asterisk -rx 'database get cidname'") |
| 16:27.03 | eKo1 | You'll have to snoop around. |
| 16:27.14 | leandro_pt | /var/lib/asterisk/astdb |
| 16:27.17 | leandro_pt | at least on debian |
| 16:27.19 | exonic | leandro_pt, http://us2.php.net/manual/en/ref.dba.php |
| 16:27.27 | jeremywhiting | hi all |
| 16:27.43 | jeremywhiting | anyone here use polycom phones or know them very well |
| 16:28.00 | SpaceBass | Hmmhesays THANKS! everything is working great! |
| 16:28.12 | [TK]D-Fender | I use them jeremywhiting, and we've chatted before. Whats up? |
| 16:28.15 | jeremywhiting | I'm trying to figure out a way to have a different ring depending on which number was dialed, not based on the callerid of the caller |
| 16:28.21 | leandro_pt | exonic, thanks.. i'm looking at that |
| 16:28.40 | leandro_pt | but what is the right handler for the asterisk DB? |
| 16:28.44 | jeremywhiting | D-Fender: actually, I think I helped get your ntp server going, but that's ok |
| 16:28.47 | leandro_pt | that would be a big help |
| 16:28.53 | [TK]D-Fender | jeremywhiting : you mean that when you place a call from the Poly that the caller hears a different ring sound? |
| 16:29.11 | [TK]D-Fender | jeremywhiting : Yeah thats right, and my clocks thank you endlessly :) |
| 16:29.11 | jeremywhiting | no so that our different lines coming in have a different ring sound |
| 16:29.44 | jeremywhiting | for example if someone dials our 800 number have one ring sound, and if the call is a forward from the other office have a different ring sound |
| 16:29.52 | Dom` | hmmm |
| 16:29.54 | [TK]D-Fender | Hmmm, I think there is an option for ring tone on the interface level but that may only be FXS ZAP channels |
| 16:30.04 | [TK]D-Fender | not sure in SIP how to do that w/o CID |
| 16:30.05 | Dom` | Whats the best way to convert a file from a variable bit rate |
| 16:30.11 | Dom` | i stripped the tags out |
| 16:30.23 | [TK]D-Fender | which you COULD mod to have that effect but you lose that info at the phone level then. |
| 16:31.10 | [TK]D-Fender | Dom` : You may want to pick up an audio editing prog out there for that, not sure which to suggest as to avoid double compressing your MP3 into shit wuality./ |
| 16:31.15 | [TK]D-Fender | quality* |
| 16:31.24 | Dom` | hmm yeh |
| 16:31.29 | Dom` | Could it be that? |
| 16:31.36 | Dom` | It like causes sea noises on the phone |
| 16:31.42 | Dom` | doesnt actually play the track |
| 16:31.47 | [TK]D-Fender | Does it sound like shit on your speakers? ;) |
| 16:31.50 | *** join/#asterisk houcj (~chatzilla@cpe-66-69-132-115.houston.res.rr.com) |
| 16:31.58 | Dom` | The mp3 file? |
| 16:32.00 | jeremywhiting | yeah, that's what I thought, thanks |
| 16:32.01 | [TK]D-Fender | yeah |
| 16:32.02 | Dom` | no its vbr though |
| 16:32.18 | Dom` | What dialline would you suggest? |
| 16:32.20 | Dom` | just |
| 16:32.24 | Dom` | musiconhold() |
| 16:32.25 | Dom` | ? |
| 16:32.29 | *** join/#asterisk Defraz (~t0tal@tim.ibccom.net) |
| 16:32.36 | [TK]D-Fender | Oh, well if it warbles it may not like VBR. Try the "lossy" way of backporting to non-VBR to test it |
| 16:32.53 | [TK]D-Fender | Dom` : I use "MP3Player" for my testing |
| 16:33.17 | Dom` | instead of musiconhold? |
| 16:33.24 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 16:33.28 | eKo1 | non-vbr files are recommended. |
| 16:33.42 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 16:33.42 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 16:34.10 | Dom` | yeh i know, just need to either convert it or put one on that isnt vbr |
| 16:34.23 | Dom` | although i think it did the same when it had asterisk's default in the folder |
| 16:34.53 | jeremywhiting | D-Fender: is there a way to set a different ring tone for a different registration? maybe I could have one ring line1 and the other ring line2 on our IP-300 phones |
| 16:36.00 | [TK]D-Fender | jeremywhiting : For sure on different regs, but thats MESSY :/ |
| 16:36.16 | jeremywhiting | yeah, but it works for what I'm doing I think |
| 16:36.20 | [TK]D-Fender | I only set mine up as 1 reg, 6 line keys assigned |
| 16:36.31 | [TK]D-Fender | Whats the goal? |
| 16:36.35 | jeremywhiting | boss's requirements, so they know how to answer the phone |
| 16:37.11 | jeremywhiting | we've got employees working for both companies at the remote office, so they need to know which line the calls are coming on to know whether to say "Teen Options" or "Tipton Academy" |
| 16:37.20 | jeremywhiting | I know, it's silly, but nevertheless |
| 16:37.25 | [TK]D-Fender | as in to treat an incoming call differently while not mangling the CID? |
| 16:37.37 | [TK]D-Fender | Because thats the EASY way |
| 16:37.38 | jeremywhiting | and the lines are already on their own registrations, I did that from the beginning for some reason |
| 16:37.55 | jeremywhiting | correct |
| 16:38.22 | jeremywhiting | I think I just wanted control over which lines I set asterisk to ring, etc, so I set each registration on each phone with it's own registration, etc |
| 16:38.22 | [TK]D-Fender | jeremywhiting : doing that will give you busy responses instead of ringing on another line key. Makes for poor service levels |
| 16:38.33 | jeremywhiting | oh, I see |
| 16:38.38 | jeremywhiting | didn't think about that |
| 16:38.50 | [TK]D-Fender | Well with eack key with its own reg, you can deficately set a distict ring for them |
| 16:38.58 | jeremywhiting | but most of the time I have the dialplan ring line1&line2&line3 anyway, just in case |
| 16:39.05 | Dom` | does the same thing with non-vbr files to [TK]D-Fender :( |
| 16:39.08 | Dom` | Any suggestions? |
| 16:39.21 | [TK]D-Fender | Using the right ver of MPG123? |
| 16:39.45 | [TK]D-Fender | (as in not using MPG321 which doesn't work) |
| 16:40.39 | Dom` | exten => 10,1,Answer |
| 16:40.40 | [TK]D-Fender | should be 0.59r |
| 16:40.40 | Dom` | exten => 10,2,MusicOnHold() |
| 16:40.43 | Dom` | look correct of testing? |
| 16:40.54 | jeremywhiting | so just to clarify if say, jr1 is in use and a call comes in to dial(jr1&jr2&jr3&...) would it get a busy signal, or just ring the others that aren't busy? |
| 16:41.16 | [TK]D-Fender | more or less. in your console do you see it choosing the MP3? |
| 16:41.21 | Dom` | for* |
| 16:41.25 | Dom` | mpg123 |
| 16:41.25 | Dom` | High Performance MPEG 1.0/2.0 |
| 16:41.25 | Dom` | nope |
| 16:41.28 | [TK]D-Fender | no, not in that case. |
| 16:42.16 | jeremywhiting | [TK]D-Fender: thanks for the help. I think I can figure the rest out from here....hopefully |
| 16:42.20 | [TK]D-Fender | It'll ring the others. Awkward. Is CID really important for your implementation? |
| 16:42.32 | *** part/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 16:42.36 | *** join/#asterisk Tenkawa (~Tenkawa@Tenkawa.user) |
| 16:42.38 | Dom` | [TK]D-Fender whats it supposed to say in console/ |
| 16:42.49 | [TK]D-Fender | I really suggest mangling it :) You could push the old data into a web page for retrieval :) |
| 16:42.55 | Dom` | - Started music on hold, class 'default', on SIP/135.196.5.201 |
| 16:42.56 | jeremywhiting | yeah, because eventually, they want it set up to dial only the last person someone talked to if they call back a second time |
| 16:42.57 | Dom` | thats it |
| 16:43.13 | Tenkawa | Anyone know any voip providers that are fairly inexpensive and would work with software only? |
| 16:43.22 | Tenkawa | I want to build an asterisk box but... |
| 16:43.24 | *** join/#asterisk mxmasster (~maxc@66.113.65.12) |
| 16:43.26 | jeremywhiting | [TK]D-Fender: that seems more messy than just using the different registrations actually |
| 16:43.28 | [TK]D-Fender | jeremywhiting : Hmm... basically like assigning reps to callers.... |
| 16:43.29 | Tenkawa | I dont want any physical phones |
| 16:43.45 | [TK]D-Fender | jeremywhiting : watch out for when the customer calls you back on one of their 20 OTHER lines ;) |
| 16:43.51 | Dom` | [classes] |
| 16:43.52 | Dom` | default => mp3:/var/lib/asterisk/mohmp3 |
| 16:43.55 | jeremywhiting | yeah, after a rep talks to a caller once, they always are supposed to get the call for that one |
| 16:43.58 | Dom` | got that in the config |
| 16:44.02 | jeremywhiting | yeah, I know, it's not perfect, but it works |
| 16:44.37 | Lee__ | how come Asterisk turns my terminal black? |
| 16:44.38 | [TK]D-Fender | jeremywhiting : nifty idea I'd like to discuss further but its lunch time. I'll be around though and think about it (always something I could use here I guess) |
| 16:44.43 | jeremywhiting | and the boss will probably want me to filter out regular calls too, so if the family at home office calls once, and they pick up on the phone in the closet, it won't only ring in the closet from then on |
| 16:44.46 | Lee__ | and how do I make it not do that? |
| 16:45.18 | [TK]D-Fender | jeremywhiting : you can route by a lot of methods. Soo much to do.... |
| 16:45.19 | jeremywhiting | so, needless to say there are some bad side effects |
| 16:45.36 | [TK]D-Fender | Tenkawa : So what do you want to have? Soft phones only? |
| 16:46.02 | jeremywhiting | plus coming from a webmaster/server admin background, having control over the whole phone system as well, is so envigorating |
| 16:46.12 | [TK]D-Fender | jeremywhiting : you can avoid those easily since your solution is all dialplan based like that. "GOTOIF" is your friend. |
| 16:46.26 | [TK]D-Fender | jeremywhiting : I'm in the same boat... I love it. |
| 16:46.39 | jeremywhiting | makes me want to start setting up server-control for the lights and bathrooms or something |
| 16:46.54 | jeremywhiting | more power= greed for even more power somehow |
| 16:47.16 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
| 16:47.16 | jeremywhiting | anyway, go eat lunch |
| 16:47.47 | [TK]D-Fender | jeremywhiting : I *can* help you there, * controls my lighing and makes me coffee already at home :D |
| 16:48.04 | jeremywhiting | sweet |
| 16:48.21 | Dom` | [TK]D-Fender is it possible that its looking in the wrong dir, for the mp3's? |
| 16:48.44 | jeremywhiting | Dom: did you put them in the same directory as the other music on hold files? |
| 16:49.08 | Dom` | There were other mp3's in there |
| 16:49.16 | Dom` | cant say i tried it before i removed them |
| 16:49.23 | jeremywhiting | then that's probably the right place |
| 16:49.42 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 16:49.56 | jeremywhiting | might want to check the permisions on them too, make sure they're the same as the old ones were maybe, although I've never gotten my own mp3's to work either |
| 16:50.03 | SpaceBass | Ariel_ you around? |
| 16:50.06 | jeremywhiting | so take my advice with a grain of salt I guess |
| 16:50.22 | jeremywhiting | only tried once though, didn't go through the effort of removing the tags even |
| 16:51.04 | Dom` | /usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 16:51.11 | *** join/#asterisk minded (~minded@65.211.26.66) |
| 16:51.13 | Dom` | There the custum options it specifies |
| 16:51.15 | minded | hey |
| 16:51.19 | Dom` | but if you run that in the command |
| 16:51.23 | minded | is there a command in asterisk to reboot all hpones |
| 16:51.24 | minded | ? |
| 16:51.27 | Dom` | it comes up as if it doesnt like the options |
| 16:51.34 | Dom` | ie displying the help |
| 16:51.53 | minded | also, why would i be getting the error Maximum retries exceeded on phone .... |
| 16:52.06 | Tenkawa | [TK]D-Fender: yes |
| 16:52.24 | Tenkawa | sorry took me so long to respond.. walked away for a sec |
| 16:52.42 | Lee__ | minded: no |
| 16:52.48 | Lee__ | what kind of phones? |
| 16:52.55 | minded | grandstream budgetone |
| 16:53.04 | [TK]D-Fender | Sounds easy. calls come in/out through a VoIP provider? |
| 16:53.14 | Tenkawa | yup |
| 16:53.21 | Lee__ | some SIP phones can respond to SIP Notify messages but you'll have to do some trial and error. |
| 16:53.22 | Tenkawa | just trying to find one |
| 16:53.28 | minded | hm |
| 16:53.30 | minded | k |
| 16:53.30 | Tenkawa | I don't know who is decent etc etc |
| 16:53.32 | minded | well question |
| 16:53.41 | minded | any idea why im getting the maximum retries exceeded error |
| 16:53.44 | minded | and it wont let me dial through |
| 16:53.47 | [TK]D-Fender | Tenkawa : easy to do. |
| 16:53.51 | minded | but it seems that if i reboot all phones, it lets me dial |
| 16:54.00 | jeremywhiting | Tenkawa: do you need incoming land numbers? aka did's? |
| 16:54.17 | Tenkawa | jeremywhiting: just one. I'll have asterisk handle everything else via extensions |
| 16:54.30 | Tenkawa | ie each computer |
| 16:54.45 | jeremywhiting | I had to find a VoIP provider that actually provided did's in my area code, that was the hard part |
| 16:54.47 | Tenkawa | I may add phones later.. not sure yet though |
| 16:54.50 | *** join/#asterisk pjz (~pj@place.org) |
| 16:54.58 | jeremywhiting | broadvoice, etc didn't |
| 16:54.58 | Dom` | res_musiconhold.c:343 monmp3thread: Only wrote -1 of 1600 bytes to pipe |
| 16:54.59 | Dom` | Is that a problem? |
| 16:55.00 | pjz | if I call system() do I need "'s around the command? |
| 16:55.21 | Tenkawa | whats the name of that one zaptel card to handle incoming pots connection? |
| 16:55.27 | Tenkawa | I cant think of the name of it |
| 16:55.31 | pjz | TDM |
| 16:55.34 | eKo1 | pjz: try it and find out |
| 16:55.42 | Tenkawa | yeah which tdm |
| 16:55.47 | eKo1 | tdm400 |
| 16:55.53 | eKo1 | or the x101p |
| 16:55.59 | Tenkawa | thats 1 in 4 analogs |
| 16:56.01 | Tenkawa | x101p |
| 16:56.03 | Tenkawa | thats it |
| 16:56.04 | Tenkawa | or x100p |
| 16:56.26 | Tenkawa | if I do phones they are going to all be sip phones |
| 16:56.29 | Tenkawa | no analog |
| 16:57.13 | Dom` | [TK]D-Fender what the default options for the mp3player() command |
| 16:57.16 | Dom` | ill see if it works with that |
| 16:57.24 | Tenkawa | I wish asterisk wouldve been around before I got rid of my land line |
| 16:57.33 | Tenkawa | going to be hard to convincemy gf to readd one |
| 16:58.29 | [TK]D-Fender | Dom` : only way I use it is MP3Player(/path/to/my/file.mp3) |
| 16:58.37 | Dom` | rofl |
| 16:58.43 | Dom` | pretty obvious then really |
| 16:58.49 | *** join/#asterisk file[desk] (~jcolp@mctn1-3084.nb.aliant.net) |
| 17:00.53 | Dom` | hmm that works |
| 17:02.54 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 17:02.54 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 17:02.59 | SpaceBass | anyone know how to dial a ring group created in AMP in a maunal dial plan? |
| 17:05.50 | Bile_One | Dial(SIP/g1 <for group etc..>,25,rt). |
| 17:06.02 | Ariel_ | SpaceBass, I am back from lunch |
| 17:06.11 | Ariel_ | in the inbound did route that number there |
| 17:06.28 | SpaceBass | Ariel_ since I'm using zap lines that dont pass DID will that work? |
| 17:06.38 | SpaceBass | Bile_One sonds like that I need |
| 17:06.42 | Ariel_ | SpaceBass, not really |
| 17:07.30 | Ariel_ | Dial(Sip/g1 is not correct |
| 17:07.51 | Bile_One | How's that? |
| 17:07.58 | Dom` | [TK]D-Fender it is as if its playing another music file |
| 17:08.15 | SpaceBass | yeah SIP/g1 didnt work |
| 17:08.29 | Ariel_ | SpaceBass, create a new context in the extensions_custom.conf and then there send the call to your ring group. number |
| 17:09.00 | Ariel_ | Bile_One, you can't group sip devices for dialing to them. |
| 17:09.08 | Bile_One | I assumed you knew to add a context. |
| 17:09.22 | SpaceBass | Ariel_ I have the context in _custom now |
| 17:09.42 | Ariel_ | and what is the dial string there? |
| 17:09.57 | SpaceBass | Ariel_ so how do I send it to the ring group? thats the syntax im missing |
| 17:10.06 | SpaceBass | using sip/xxx&sip/yyy now |
| 17:10.29 | Ariel_ | Yes but you want to add a prefix to the dial caller ID |
| 17:10.41 | sparrow | I was wondering if someone could offer a direction to look or some information about the meetme app. In my dial plan I call to meetme with exten => _8X.,3,MeetMe(${EXTEN}|sM) but if the user dial an incorrect conf number, the conf-invalid.gsm is played to them, and they hung up on without traveling the rest of the way through the dial plan. Is there a way to have the dial plan deal with a non-zero return without hanging up? |
| 17:10.47 | SpaceBass | Ariel_ exactly |
| 17:11.17 | SpaceBass | Ariel_ so I can use setcallerID and overwrite the CID, but I want to use the ring group from AMP so I can add to the caller id |
| 17:11.35 | SpaceBass | and I have the custom context and the ring group, i just dont know the syntax to dial the ringgroup |
| 17:13.09 | Bile_One | include your custom context in the extension.conf. |
| 17:13.18 | SpaceBass | not sure I'm being clear |
| 17:13.31 | Bile_One | use pastebin then. |
| 17:14.18 | [TK]D-Fender | Dom` : Try and use another MP# with MP3Player |
| 17:14.22 | SpaceBass | http://pastebin.ca/14775 |
| 17:14.30 | SpaceBass | it works fine, but its not using the ring group created in AMP |
| 17:14.32 | Bile_One | be right back. |
| 17:14.35 | SpaceBass | not sure how do dial the rin group |
| 17:14.38 | Dom` | another mp3? |
| 17:15.22 | [TK]D-Fender | Ariel_ : got experience with my question an hour earlier about mixing 3 digium cards and suggestions for motherboards? |
| 17:15.52 | [TK]D-Fender | Dom` : yes, just grab another MP3 and test it with MP3Player(/thefile) |
| 17:16.14 | Ariel_ | SpaceBass, let me get you a good syntax in a minute. |
| 17:16.22 | SpaceBass | Ariel_ much appericated! |
| 17:16.27 | Ariel_ | [TK]D-Fender, I was out to lunch. |
| 17:16.41 | *** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
| 17:16.56 | [TK]D-Fender | [12:25] <[TK]D-Fender> Hey got a quick question : I'm looking to set up a server with 1 TE100P, 1 TDM04B, 1 TDM40B, (4 line analog in/out), Any issues with that many cards (of those mixed types) in a system? Also I need suggeestions for a good motherboard to run those on. |
| 17:16.56 | Dom` | What am i doing [TK]D-Fender? |
| 17:17.00 | Ariel_ | [TK]D-Fender, I have used 2 digium TDM400p cards in dells but you have to play with the irq |
| 17:17.14 | Ariel_ | I would not recommend more then 2 boards |
| 17:18.23 | [TK]D-Fender | :/ |
| 17:18.32 | [TK]D-Fender | 2 total, or 2 TDM? |
| 17:18.38 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 17:19.16 | Ariel_ | SpaceBass, all you need to do is set this one line in your context: exten => s,1,Goto(ext-group,220,1) just put the ring group you made in stead of 220 |
| 17:19.49 | Ariel_ | [TK]D-Fender, two total cards not ports... max 8 ports |
| 17:19.57 | SpaceBass | Ariel_ I just came to the same conclusion when I looked at extensions_additional |
| 17:20.04 | SpaceBass | but I had s,1,answer before it |
| 17:20.21 | Ariel_ | ok that works as well just make the 2nd line 2 instead of 1 |
| 17:20.29 | SpaceBass | do i need the answer line? |
| 17:20.36 | Ariel_ | not really |
| 17:20.39 | SpaceBass | k |
| 17:20.40 | SpaceBass | thanks! |
| 17:21.00 | Dom` | Anyone know where it is specified that musiconhold.conf will be used? |
| 17:21.14 | Ariel_ | asterisk.conf |
| 17:21.29 | SpaceBass | Ariel_ thanks, that did the trick! |
| 17:21.30 | [TK]D-Fender | Ariel_ : what I mean is is it safe for 2 x 4-port TDM + 1 T1? |
| 17:21.36 | cpatry | dom: res_musiconhold.c |
| 17:21.38 | jeremywhiting | [TK]D-Fender: hey, do you know where I would be able to set different ring tones for diffferent registrations? I can't find anything in my phone1.cfg, or sip.cfg |
| 17:21.41 | jeremywhiting | any ideas? |
| 17:21.48 | Ariel_ | [TK]D-Fender, once again I would say no |
| 17:21.58 | [TK]D-Fender | Dang.. I need it for redundancy. |
| 17:22.15 | [TK]D-Fender | Maybe get a TDM22B and mix it up. |
| 17:22.17 | SpaceBass | sweet! i can go to lunch now |
| 17:22.20 | cpatry | dom: grep -Rn "musiconhold.conf" * is the solution :) |
| 17:22.29 | SpaceBass | thanks Ariel_ and Hmmhesays! |
| 17:22.30 | Ariel_ | [TK]D-Fender, problem is the tdm400 cards seem to take allot of interupps |
| 17:22.55 | Dom` | I know where it is |
| 17:22.58 | Dom` | Its just not working |
| 17:23.29 | Dom` | plays 'sea' noises for some reason |
| 17:23.32 | Hmmhesays | np SpaceBass |
| 17:23.57 | Ariel_ | SpaceBass, any time glad to help |
| 17:24.06 | *** join/#asterisk escualis (~carlos@201.236.3.36) |
| 17:24.06 | [TK]D-Fender | Dom` : I had that effect one too.. make sure both ID3 tags are gone |
| 17:24.14 | escualis | hello everybody :) |
| 17:24.21 | [TK]D-Fender | (v1 & v2) and definately get some more test MP3's |
| 17:24.38 | Dom` | I used a program to remove the tags |
| 17:24.45 | Dom` | Im looking for the ones that used to be in the folder |
| 17:24.50 | Dom` | cant remember what there called though |
| 17:25.04 | escualis | wich range of ports it's needed for sip protocol? |
| 17:25.18 | escualis | 5060 -> 65000? |
| 17:25.44 | Ariel_ | Ok am going back to finishing some cable runs I am doing....bbl8r |
| 17:25.52 | Dom` | fpm-sunshine.mp |
| 17:25.55 | Dom` | thats it i think |
| 17:25.56 | jeremywhiting | escualis: most things I've read said udp ports 1000-10000 |
| 17:26.06 | jeremywhiting | or some huge range like that |
| 17:26.28 | [TK]D-Fender | 10000-20000 for RTP, 5060 TCP & UDP IIRC..... |
| 17:26.30 | Ariel_ | udp for sound 10,000 to 20,000 works with asterisk best. |
| 17:26.53 | Hmmhesays | Ariel_: why would it matter? |
| 17:27.06 | Dom` | It is as if its playing really quietly in the background |
| 17:27.06 | Hmmhesays | or.. .are you aren't talking quality i'm guessing |
| 17:27.15 | Dom` | and theres a load of rubbish over the top of it |
| 17:27.18 | Tenkawa | can asterisk emulate a voip phone adapter? |
| 17:27.33 | Hmmhesays | Tenkawa: explain |
| 17:27.36 | Ariel_ | Hmmhesays, ??? |
| 17:27.53 | Tenkawa | ie replace needing to use a Cisco ATA |
| 17:27.53 | Dom` | Are there any really obvious known bugs in it? |
| 17:27.55 | Hmmhesays | <Ariel_> udp for sound 10,000 to 20,000 works with asterisk best. <-- at first read I assumed you are talking sound quality |
| 17:28.24 | [TK]D-Fender | Tenkawa : What are you looking to use instead of an ATA186? |
| 17:28.25 | Hmmhesays | Tenkawa: pci cards from digium can have fxs modules in them |
| 17:28.26 | Ariel_ | Hmmhesays, Ahh it's just what asterisk kinda wants. |
| 17:28.40 | Hmmhesays | it's what you configure in asterisk.conf |
| 17:28.50 | Bile_One | SpaceBass http://pastebin.ca/14780. |
| 17:29.21 | Tenkawa | my goal is to take a voip incoming connection route it through asterisk into a tdm400p and hook up analog phones to it |
| 17:29.24 | Dom` | Deffinatly a problem somewhere along the lines [TK]D-Fender and it doesnt look like its anything that can be fixed |
| 17:29.29 | Dom` | theres very few config things you can do |
| 17:29.57 | Hmmhesays | Tenkawa: what's stopping you? |
| 17:30.03 | [TK]D-Fender | Tenkawa : ATA's are cheaper per-port and easier to install / replace / debug for those purposes. |
| 17:30.37 | Tenkawa | hmm |
| 17:30.47 | Lee__ | why does asterisk turn my terminal black? and how do I make it not do that? |
| 17:30.47 | sparrow | Does anyone know how to stop asterisk from hanging up on a non-zero return from an application like meetme? |
| 17:31.02 | SpaceBass | Bile_One thanks for all the input! lots to digets |
| 17:31.06 | [TK]D-Fender | Tenkawa : I have a 2 port FXS / FXO card which works great. I'd be doing what you are asking about myself if they offered DSL on a "dry" line here |
| 17:31.06 | SpaceBass | digest |
| 17:31.20 | SpaceBass | speaking of ATAs will a modem work over them or not? |
| 17:31.20 | Bile_One | Yep, but you'll get it. |
| 17:31.24 | SpaceBass | (thinking about tivo) |
| 17:31.44 | Tenkawa | [TK]D-Fender: well I need 4 independent extensions and the only way I can think of doing that through vopi is route the main number in to asterisk and let it do the rest or am I way off base |
| 17:31.49 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3084.nb.aliant.net) |
| 17:31.52 | SpaceBass | Bile_One what i needed was one line that said s,1,goto(ext-group,2,1) but I need to learn about call groups and pickup groups too |
| 17:31.59 | Bile_One | SpaceBass, hit or miss, some will work and some won't. |
| 17:32.00 | Lee__ | SpaceBass: it should with the fax module and uncompressed audio. |
| 17:32.03 | cluecon | Tenkawa: that would be the way to do it. |
| 17:32.09 | Lee__ | but I haven't tried. |
| 17:32.30 | Tenkawa | having 1 did is fine as long as I can split it internally |
| 17:32.35 | [TK]D-Fender | Tenkawa : Sounds like a workable idea. |
| 17:32.44 | SpaceBass | Bile_One my TiVos and fax machine are the only analog devices i have left |
| 17:32.46 | Bile_One | SpaceBass, my boss hacked his Dishnetowrk DVR to work with it. |
| 17:32.54 | Lee__ | Tenkawa: many origination providers will give you 4 channels per DID. |
| 17:32.55 | SpaceBass | hacked |
| 17:33.03 | Tenkawa | Lee__: really.. interesting |
| 17:33.03 | [TK]D-Fender | Tenkawa : When * answers the call you can give them a menu and do whatever you want from there. All easy stuff... |
| 17:33.08 | Tenkawa | [TK]D-Fender: yup |
| 17:33.29 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 17:33.33 | Tenkawa | I'm trying to design a few system ideas to look into for a small office |
| 17:33.53 | Tenkawa | whats asterisks license ? |
| 17:34.01 | Bile_One | SpaceBass, yes was some software he downloaded off the Inet, to allow him to connect to an * box, but I'm not sure what he did. |
| 17:34.03 | dca[laptop] | morning all |
| 17:34.24 | dca[laptop] | how can i capture all teh debug information from a particular call to a file? (if possible at all) |
| 17:34.31 | SpaceBass | Bile_One might look into that... been thinking of hacking my tivo to display caller id |
| 17:34.32 | cluecon | Tenkawa: there isn't a license. Use it and abuse it all you want. But some of the technologies require a license to use. |
| 17:34.36 | Dom` | [TK]D-Fender, fixed it ;) |
| 17:34.39 | Tenkawa | cluecon: ahh |
| 17:35.06 | [TK]D-Fender | Dom` : Cool, what'd you do? |
| 17:35.17 | Bile_One | SpaceBass, my DirecTv boxes do but my tivo doesn't. |
| 17:35.41 | Dom` | Stop the whole server and started it again |
| 17:35.47 | Dom` | appears reload wasn't doing the job |
| 17:35.55 | Dom` | Which is rarther annoying |
| 17:35.59 | [TK]D-Fender | heh |
| 17:36.20 | Tenkawa | any of you use asterisks voice mail module much? |
| 17:36.24 | Tenkawa | is it decent? |
| 17:36.24 | [TK]D-Fender | Sometimes things become corrupted in run-time and you just need to cut the juice :D |
| 17:36.36 | *** join/#asterisk kimosabe (~natt@216.60.60.103) |
| 17:36.41 | Dom` | rofl |
| 17:36.42 | [TK]D-Fender | Tenkawa : We all do and it works just fine |
| 17:36.47 | jeremywhiting | Tenkawa: sure works great |
| 17:36.56 | jeremywhiting | emails us our voicemail, that's my favorite part |
| 17:36.57 | kimosabe | is any one using h323 on the sipura devices |
| 17:37.04 | Dom` | Cheers ;) |
| 17:37.25 | Tenkawa | I've been looking on the site but I dont see specific documentation for the individual modules |
| 17:38.45 | _omer | anybody who uses SOX ? |
| 17:38.49 | cluecon | Tenkawa: go to voip-info.org |
| 17:38.50 | *** join/#asterisk FuRR_ (~meeps@bko29.chapman.edu) |
| 17:38.57 | cluecon | _omer: we all do. |
| 17:39.06 | jacks | i use SEX |
| 17:39.16 | _omer | jacks: we all do |
| 17:39.18 | Tenkawa | cluecon: thank ya |
| 17:39.19 | _omer | lol |
| 17:39.23 | jacks | i think that *joke* has been made before ;) |
| 17:39.31 | Tenkawa | oooh the wiki is nice too |
| 17:39.35 | [TK]D-Fender | ~sex |
| 17:39.36 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 17:39.44 | [TK]D-Fender | :O |
| 17:39.47 | SpaceBass | DAMN! |
| 17:40.09 | jacks | cluecon: you want me to pm you? |
| 17:40.12 | jacks | :P |
| 17:40.17 | SpaceBass | my directv dish is up on a telephone pole in my alley... been there for almost a year untouched and verizon picks today to install new fiber in my neighborhood |
| 17:40.30 | Hmmhesays | yep don't want that PMS |
| 17:40.30 | _omer | cluecon: Can you please provide me the Script to get some idea about using SOX .... I mean the syntax... |
| 17:40.37 | Hmmhesays | specially someone elses |
| 17:40.48 | cluecon | _omer: what do you want to do with it? |
| 17:40.57 | jacks | man sox |
| 17:41.07 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 17:41.17 | Tenkawa | would a single 2.6 gig athlon with 256 meg of ram be able to support 6-8 sip phones? |
| 17:41.30 | Lee__ | there needs to be a sample of a DJ scratching on every wiki |
| 17:41.31 | Hmmhesays | Tenkawa: unless you fark up the install |
| 17:41.36 | Tenkawa | Hmmhesays: jhaaaahaa |
| 17:42.15 | Hmmhesays | you could even transcode all those calls |
| 17:42.18 | _omer | cluecon: I use "Monitor" to record conversations....but the conversation finishs....I use SOX manually to make ONE file from TWO output files....I want to do it automaticlly when conversation ends.. |
| 17:42.26 | kimosabe | is any one running h323 with sipura 2000 if so where is there a good readme on this ?? |
| 17:42.57 | _omer | cluecon: I dont know the syntax : for example exten = > bla bla bla ...to use SOX |
| 17:43.03 | Tenkawa | duh.. analog because if Ido voip I'm going to chew up 2x the bandwidth easily |
| 17:43.05 | minded | any idea why im getting the maximum retries exceeded error |
| 17:43.07 | minded | and it wont let me dial through |
| 17:43.09 | minded | but it seems that if i reboot all phones, it lets me dial |
| 17:43.36 | [TK]D-Fender | Tenkawa : Unless you really like your analog phones or just really want to save money I'd get SIP phones. SPA-841 should do nice for you. |
| 17:43.58 | Tenkawa | nod.. thats the plan.. thats all lan bandwidth so no biggie |
| 17:44.06 | Tenkawa | but I better use analog for uplink |
| 17:44.33 | Tenkawa | go to regular co |
| 17:44.51 | cluecon | _omer: take a look at the wiki... http://www.voip-info.org/tiki-index.php?page=Monitor+stereo-example |
| 17:45.08 | Tenkawa | time to see how cheap I can get a phone line |
| 17:45.10 | Lee__ | _omer: you'll probably have to write an AGI script which calls sox. |
| 17:46.19 | minded | anyidea? |
| 17:47.37 | _omer | <PROTECTED> |
| 17:47.57 | kimosabe | whos running h323 anyone ?? |
| 17:48.15 | Nuxi | _omer: shell scripts run fine from AGI scripts |
| 17:48.51 | SuperMMan | hey all question for music on hold is there anyway to play a live internet radio station? |
| 17:49.02 | Dom` | hmm |
| 17:49.03 | Lee__ | is there a way to add to Asterisk's search path for sounds? So I don't have to specify a absolute path for Record() and Playback() calls? |
| 17:49.16 | Dom` | If you stick musiconhold() in before a dial() |
| 17:49.27 | Dom` | how can you make it go to the test extension? |
| 17:50.03 | SuperMMan | Dom` was that answer for me? |
| 17:51.43 | Lee__ | SuperMMan: have you tried giving it a URL? |
| 17:52.03 | SuperMMan | Lee__ ya all i get back is a error |
| 17:52.09 | Lee__ | probably can't then. |
| 17:52.15 | Lee__ | what's the error? |
| 17:52.30 | minded | anyidea? |
| 17:52.45 | SuperMMan | one second i will show you my setup and error |
| 17:53.36 | SuperMMan | default => mp3:/var/lib/asterisk/empty,http://www.cowboyculturalsociety.com/ccs_128.m3u error: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player |
| 17:54.59 | minded | id atleast like a "no minded, i dont know |
| 17:55.04 | minded | !!! :) |
| 17:55.15 | Lee__ | SuperMMan: that's because you're not giving it an mp3 stream to play, you're giving it a playlist |
| 17:55.41 | Lee__ | cat out the contents of that playlist and find the URL to the actual stream and try again. |
| 17:56.07 | Lee__ | "no minded, I don't know" |
| 17:56.23 | FuRR_ | can someone help me here with a barge issue. I have 2 sip phones and an analog PSTN circuit, phone 1 dials out and establishes a call, i would like phone 2 to be able to dial *33 then the extension and barge the SIP extension. Is this doable? or would it be better to silently move the sip/pstn connection into a meetme room |
| 17:58.07 | Hmmhesays | haha trying to compile zaptel without kernel source doesn't work so well |
| 17:58.12 | dca[laptop] | how can i capture all teh debug information from a particular call to a file? (if possible at all) |
| 17:59.11 | SuperMMan | Lee__ ok thanx |
| 17:59.15 | SpaceBass | if I have a phone with a message waiting light, can i make it "check" a mailbox that is different than the extension? |
| 17:59.17 | Lee__ | dca[laptop]: /var/log/asterisk/messages and turn debugging on in /etc/asterisk/logger.conf |
| 18:01.33 | *** join/#asterisk ardor (~ardorgof@ip68-224-3-96.lv.lv.cox.net) |
| 18:01.59 | Ariel_ | SpaceBass, edit the sip extension in amp and put the mailbox number you want |
| 18:02.15 | SpaceBass | Ariel_ that simple? thanks! |
| 18:03.21 | Hmmhesays | amp, the great confuser |
| 18:03.53 | Ariel_ | Hmmhesays, some like it some don't it's got it's place |
| 18:04.09 | SpaceBass | well, AMP was great at first, but now I've taken to editing the files manually... but i cannot edit an extension that AMP has created, b/c amp will overwrite it |
| 18:04.28 | SpaceBass | all in all i like amp for adding extension, manaing trunks, on hold music... stuff like that |
| 18:04.40 | SpaceBass | Hmmhesays by the way, got that phone working... thanks! i'm loving it |
| 18:04.48 | Dom` | How do you get music on hold working whilst dialing? |
| 18:05.05 | Ariel_ | add an m |
| 18:05.09 | SpaceBass | Ariel_ no way to assing more than one mailbox to an extension is there? |
| 18:05.21 | Dom` | With no music on hold line Ariel_? |
| 18:05.45 | Dom` | ,Dial(SIP/thegig,30,rt|20|m) |
| 18:05.49 | Dom` | whats up with that then/ |
| 18:05.59 | SpaceBass | although speaking of music on hold and amp, it will not upload any of my mp3s |
| 18:06.00 | SpaceBass | not sure wh |
| 18:06.01 | SpaceBass | y |
| 18:06.11 | Lee__ | SpaceBass: you have lame installed? |
| 18:06.33 | Ariel_ | Dom`, looks right |
| 18:06.37 | SpaceBass | SpaceBass on my * box? |
| 18:06.40 | Dom` | Doesnt work though |
| 18:06.42 | SpaceBass | or on my workstation? |
| 18:06.43 | Dom` | Just dials |
| 18:06.45 | Dom` | no music |
| 18:06.46 | Lee__ | SpaceBass: you should move this over to #amportal |
| 18:06.55 | SpaceBass | done |
| 18:07.09 | Ariel_ | Dom`, what does your cli say? |
| 18:07.11 | Hmmhesays | SpaceBass: learn asterisk before you use amp |
| 18:07.18 | Hmmhesays | problem solved |
| 18:08.02 | minded | does anyone know why im getting the error |
| 18:08.16 | Dom` | Ariel_ |
| 18:08.17 | Hmmhesays | what error? |
| 18:08.17 | Dom` | <PROTECTED> |
| 18:08.18 | Dom` | Jun 16 19:11:46 DEBUG[5722]: app_dial.c:492 dial_exec: DIAL WITH URL=20|m_ |
| 18:08.29 | minded | maximum retries exceeded on clal bleh |
| 18:08.29 | minded | ? |
| 18:08.36 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 18:08.41 | Hmmhesays | cause something is farked up minded |
| 18:08.46 | Hmmhesays | sip debug |
| 18:09.13 | minded | what do i look for in sip debug |
| 18:09.40 | kimosabe | does any one know about running h323 on asterisk and sipura devices |
| 18:09.41 | kimosabe | ?? |
| 18:09.43 | minded | it says unable to create channel type sip as well as the error when i try to place a call |
| 18:09.49 | shido | kimosabe, |
| 18:09.52 | shido | whyyyyyyyyyyyyyyy ? |
| 18:10.19 | Dom` | that shouldn't be adding the bit to url should it Ariel_? |
| 18:10.55 | Hmmhesays | kimosabe: why would you do something like that? |
| 18:11.16 | Hmmhesays | minded: take a closer look at your entry in extensions.conf |
| 18:11.33 | Ariel_ | Dom`, just a sec on phone |
| 18:11.38 | Dom` | kk :) |
| 18:11.47 | *** join/#asterisk Sephen (~Sephen@proxy5.med-web.com) |
| 18:16.02 | *** join/#asterisk smash- (smash@12.108.22.205) |
| 18:16.03 | smash- | sup |
| 18:16.05 | smash- | <PROTECTED> |
| 18:16.08 | smash- | <PROTECTED> |
| 18:18.47 | kimosabe | is any one running g711 ?? |
| 18:20.33 | diablopico | shido , are you there ? |
| 18:21.09 | Ariel_ | Dom`, do show application dial it will tell you all about m |
| 18:21.23 | *** part/#asterisk _omer (dfsdf@202.147.167.213) |
| 18:22.07 | Dom` | 'M(x) -- Executes the macro (x) upon connect of the call |
| 18:22.14 | Dom` | that m Ariel_? |
| 18:22.19 | *** join/#asterisk NSGN (~NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 18:22.24 | NSGN | hello everybody |
| 18:22.27 | Dom` | oh no |
| 18:22.28 | Dom` | <PROTECTED> |
| 18:22.32 | Dom` | Doesnt really say much |
| 18:22.40 | NSGN | n00b here. i'm curious if asterisk is right for me |
| 18:22.44 | Ariel_ | that is all there is for it. |
| 18:23.01 | Ariel_ | I don't use it I like the ringing sound |
| 18:23.17 | NSGN | i have a celeron 700mhz with generic winmodem. will asterisk support this? |
| 18:23.39 | smash- | <PROTECTED> |
| 18:23.41 | smash- | can someone do me a favor? |
| 18:23.46 | smash- | <PROTECTED> |
| 18:23.54 | smash- | <PROTECTED> |
| 18:24.01 | cluecon | smash: you want to know what it resolves to? |
| 18:24.19 | Dom` | <PROTECTED> |
| 18:24.22 | Dom` | Cheers ;) |
| 18:25.29 | NSGN | anybody? |
| 18:25.32 | cluecon | smash: 67.159.11.221 and 66.90.99.226 respectively |
| 18:25.59 | cluecon | NSGN: not likely. |
| 18:26.06 | NSGN | awh, crap |
| 18:26.49 | NSGN | i am looking for a sollution to make calls over my network from a laptop to a pc that has a modem that is making the calls go out to the landline network |
| 18:26.59 | cluecon | NSGN: * may run but the winmodem is probably a no go |
| 18:27.13 | NSGN | x_x |
| 18:27.54 | QuaG|NaR | im having difficulty understanding contexts and how they tie in with the differnt .conf files, can anybody point me to some documentation that may help? |
| 18:32.01 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:32.24 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:33.11 | dca[laptop] | anyone know of a good gui for monitoring queues in asterisk? |
| 18:33.34 | Sephen | Anyone here using meetme2? I can't get the web interface to perform any actions on callers. I can see them in the conference, but kick, talk, listen, etc doesn't do anything. |
| 18:33.35 | *** join/#asterisk smash- (smash@12.108.22.205) |
| 18:34.02 | SpaceBass | is there a way to have music on hold randomly start in the middle of a song? |
| 18:34.13 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 18:34.14 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-1831.nb.aliant.net) |
| 18:34.34 | Lee__ | can someone with a CVS checkout of asterisk-sounds tell me if you have this file: dir-intro-oper.gsm |
| 18:34.37 | *** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 18:34.41 | Lee__ | it's not in my checkout |
| 18:34.41 | minded | WARNING[1925] : chan_sip.c:908 retrans_pkt: maximum retries exceeded on call |
| 18:34.47 | minded | help me get rid of this pleas |
| 18:35.11 | cluecon | Lee__not in mine. |
| 18:35.51 | cluecon | minded: are your sip peers registered when you try to make the call? |
| 18:35.53 | Lee__ | cluecon: thanks |
| 18:35.58 | *** join/#asterisk xarmiex (~armie@arm.enter.net) |
| 18:36.17 | Ariel_ | let I found them in the /usr/src/asterisk/sounds |
| 18:37.04 | minded | yes clue |
| 18:37.06 | minded | everything is registered |
| 18:37.43 | cluecon | pastebin the full cli output before and after that line please. |
| 18:38.06 | Lee__ | Ariel_: thanks |
| 18:38.56 | minded | me? |
| 18:39.06 | cluecon | minded: pastebin the full cli output before and after that line please. |
| 18:39.16 | minded | well.. thats about it, but ok |
| 18:39.17 | Lee__ | Ariel_: it gets stranger... |
| 18:39.18 | Lee__ | nas:/usr/local/src/asterisk/sounds# ls dir* |
| 18:39.18 | Lee__ | dir-instr.gsm dir-intro-fn.gsm dir-intro.gsm dir-nomatch.gsm dir-nomore.gsm |
| 18:39.36 | Ariel_ | yes I see that. |
| 18:40.11 | xarmiex | anyone know if its possible to listen in on calls realtime ? ifso can this be done with both zap and sip channels? |
| 18:41.26 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 18:41.43 | Ariel_ | Lee__, found them /usr/src/AMP/amp_conf/var/lib/asterisk/sounds |
| 18:41.45 | cluecon | minded: paste the FULL cli output, not just that line, please. |
| 18:42.07 | Lee__ | Ariel_: but where did AMP get them from? original recording? |
| 18:42.16 | *** join/#asterisk meppl (mephisto@p54AADF4F.dip.t-dialin.net) |
| 18:42.26 | *** join/#asterisk mxmasster (~maxc@67.109.55.66.ptr.us.xo.net) |
| 18:42.35 | anthm | app_chanspy compliments of Clue Con |
| 18:42.35 | minded | cluecon |
| 18:42.36 | Lee__ | it sounds like the same voice as the other dir-* stuff in the Asterisk tree |
| 18:42.38 | minded | thats ALL itsays :( |
| 18:42.40 | cluecon | xarmiex: ChanSpy |
| 18:42.42 | Ariel_ | Lee__, good question. |
| 18:42.57 | minded | sec |
| 18:43.17 | *** join/#asterisk kimosabe (~natt@216.60.60.103) |
| 18:43.31 | kimosabe | does any one have a good readme on meet me confrencing |
| 18:43.42 | ender | So I'm trying to setup my dialplan so that when a sip phone dials '9' they still get dial tone and then can dial out to an outside world. |
| 18:44.00 | cluecon | xarmiex: http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy |
| 18:44.07 | file[laptop] | ender: it's up to the sip phone, not asterisk |
| 18:44.15 | ender | file[laptop]: oh really? |
| 18:44.17 | ender | hrm. |
| 18:44.18 | file[laptop] | ender: yes |
| 18:44.28 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 18:44.32 | Seyr | heyas people |
| 18:44.49 | james_ed | hello all |
| 18:45.13 | Seyr | if I have an * server using IAX to connect on the backend, and client phones using SIP, is H323 used anywhere in this? |
| 18:45.15 | xarmiex | clue, thx thats looks like what i need |
| 18:45.15 | ender | file[laptop]: any idea what the setting might be then? this is a Sipura phone. |
| 18:45.16 | cluecon | minded: yes, one sec while i look at it. |
| 18:45.31 | file[laptop] | ender: it's in the dialplan part... |
| 18:45.32 | cluecon | Seyr: not unless you force it to. |
| 18:45.40 | file[laptop] | you tell it to keep dialtone... I forget how, I did it a long time ago |
| 18:46.08 | Seyr | cluecon: SonicWall support told me that my problem with my phones behind a SOHO3 is that it only supports version 1 of H323..... |
| 18:46.14 | ender | file[laptop]: oh, well I have that. |
| 18:46.19 | ender | sortof I guess |
| 18:46.25 | james_ed | PSTN question relating to asterisk all?? |
| 18:46.27 | ender | I have that setting in a [outbound] contxt |
| 18:46.38 | ender | then in include => outbound in my [default] context. |
| 18:46.38 | file[laptop] | ender: I meant the dialplan on the Sipura |
| 18:46.59 | cluecon | Seyr: where does our IAX connect to? |
| 18:47.06 | cluecon | s/our/your/g |
| 18:47.14 | *** join/#asterisk tholo (~tholo@ip-64-139-0-132.sjc.megapath.net) |
| 18:47.39 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
| 18:47.42 | james_ed | I have a 2 line analog phone (cheapy). I have tdm400p w/ 3fxo's and 1fxs-to analog phone... |
| 18:47.45 | ender | file[laptop]: ah looking there. |
| 18:47.46 | tzafrir_laptop | james_ed, if you won't ask we won't know ;-) |
| 18:47.46 | Seyr | my Cisco 7960 works fine behind a SonicWall TZ and works fine behind multiple PIX setups and Linksys Firewall, but behind a SonicWall SOHO3, I only get the first inbound call, but cannot get anymore calls unless I turn the phone off, wait a sec and turn back on.. then it gets the first call again, but none after.. unless I boot it again |
| 18:47.55 | Seyr | cluecon: connects to Asterlink |
| 18:48.02 | kimosabe | im trying to configure mmet me conferencing on my server does any one know how |
| 18:48.21 | minded | let me knokw what u think cluecon |
| 18:48.31 | kimosabe | i keep getting not a valid confrence number recording |
| 18:48.34 | james_ed | can i manage multiple calls coming into *box with cheapy anlalog phone..etc. put call on hold...get another call? |
| 18:48.39 | file[laptop] | kimosabe: do you have a zaptel timing device? |
| 18:48.56 | file[laptop] | james_ed: most likely |
| 18:49.16 | kimosabe | filelaptop check this out http://pastebin.com/300511 |
| 18:49.24 | cluecon | minded: what does sip show peers return? |
| 18:49.39 | james_ed | i can't seem to get it working...i can recieve 1 call fine...when i try to make a simultaneous call in ... i get busy |
| 18:49.44 | kimosabe | file lap top i have 8 sipura 2000 models runnin |
| 18:49.56 | minded | it returns all of the peers |
| 18:50.06 | minded | 2 hosts are unspecified (no phone has been assigned to these) |
| 18:50.09 | file[laptop] | kimosabe: I don't care, you asked about meetme I'm answering - do you have a zaptel timing device? if not, you'll need to use ztdummy, zaprtc, or get a zaptel card |
| 18:50.16 | minded | but all of the phones have specified ip's/ports |
| 18:50.23 | file[laptop] | kimosabe: if you don't know what those are, Google - widely discussed stuff |
| 18:50.29 | cluecon | minded: what kind of phones |
| 18:50.44 | minded | grandstream budgetones |
| 18:51.03 | kimosabe | filelaptop i dont run zaptell im just getting not a valid confrence number |
| 18:51.14 | file[laptop] | kimosabe: you NEED a zaptel timing device, or else meetme won't work... |
| 18:51.32 | file[laptop] | kimosabe: it'll give exactly what you're describing I believe |
| 18:51.54 | file[laptop] | and do you have conference room 1234 specified in the config? |
| 18:51.58 | cluecon | minded: if I've followed this conversation, it is the phones and not asterisk. i don't have mine with me at the moment, but mine require a reboot on occassion to force them to re-register. |
| 18:52.05 | kimosabe | oki |
| 18:52.14 | minded | ya like |
| 18:52.19 | minded | when i reboot all the phones it works |
| 18:52.24 | james_ed | any suggestions on how to handle multiple call coming into *box from PSTN? |
| 18:52.28 | minded | but i've done that like 3 times today |
| 18:52.32 | minded | and they have messed up again |
| 18:52.33 | file[laptop] | minded: do you have qualify turned on for them? |
| 18:52.42 | minded | dunno file, how do i do that |
| 18:52.51 | file[laptop] | qualify=yes in the entries in sip.conf |
| 18:53.04 | cluecon | minded: i know there is a way to fix that, but i don't remember what i had to do. |
| 18:53.32 | minded | i dont file, my qualifyyes is commented out |
| 18:53.41 | minded | and itsonly under my main context, not each phones context |
| 18:53.46 | file[laptop] | try and see... |
| 18:54.15 | Seyr | cluecon: so what do you think about the H323 thing? Do you think SonciWall Support is full of it, or is that a reasonable assumption? that since it only supports H323 version 1, my SIP phones wont work behind it |
| 18:54.17 | minded | put it under each phone context? |
| 18:54.23 | file[laptop] | minded: sure |
| 18:55.38 | cluecon | Seyr: verify that you have the right ports open on the firewall. |
| 18:55.44 | minded | when i reloaded after doing that it says |
| 18:55.58 | minded | Phone '... is now UNREACHABLE |
| 18:56.05 | *** join/#asterisk teapot (~tandrews@mail.grok.org.za) |
| 18:56.07 | file[laptop] | okay now get it to register |
| 18:56.12 | minded | how ? |
| 18:56.17 | minded | reboot the phones |
| 18:56.18 | minded | ? |
| 18:56.20 | file[laptop] | yes |
| 18:57.25 | minded | k sec |
| 18:57.27 | minded | rebooting now |
| 18:57.32 | minded | ok i rebooted, now reload? |
| 18:57.40 | file[laptop] | nope |
| 18:57.45 | Seyr | cluecon: for client phone, ports on the firewall shouldnt matter |
| 18:57.48 | minded | try and dial now |
| 18:57.49 | minded | ? |
| 18:57.59 | file[laptop] | just let it go for awhile, and see if it fixes it |
| 18:58.13 | Seyr | the server is at a datacenter with the correct ports open |
| 18:58.18 | minded | so dont mess with it? |
| 18:58.22 | minded | or... try and call? |
| 18:58.28 | file[laptop] | you can try and call if you want |
| 18:58.51 | minded | its ringing |
| 18:58.52 | minded | :D |
| 18:59.03 | minded | wewt |
| 18:59.35 | minded | k another question |
| 18:59.39 | minded | when i hang my phone up |
| 18:59.41 | cluecon | Seyr: if the sip ports aren't open then it won't get thru the firewall at all. |
| 18:59.41 | minded | the LCD reads 487 |
| 18:59.49 | minded | and its like i have to pick it up and hang it up again |
| 18:59.51 | minded | before it closes theline |
| 19:00.18 | file[laptop] | 487 Request Cancelled... |
| 19:00.25 | file[laptop] | am I supposed to know how to operate your phone? :p |
| 19:00.29 | minded | !:) |
| 19:00.50 | minded | lemme ask u something |
| 19:00.52 | minded | what does that qualify do |
| 19:00.53 | minded | ? |
| 19:01.01 | file[laptop] | qualify sends a packet to the phone every minute |
| 19:01.04 | file[laptop] | to make sure it's still there |
| 19:01.13 | file[laptop] | in certain situations, it keeps NAT mappings and firewalls open |
| 19:01.26 | *** join/#asterisk donavan (~donavan@4wx.net) |
| 19:01.33 | file[laptop] | nice to meet you. |
| 19:01.39 | minded | lol |
| 19:01.39 | minded | ;) |
| 19:01.48 | teapot | why does *8# only work on zaptel phones not on an IAXy ? |
| 19:01.55 | Seyr | cluecon: Are you talking about outbound ports or inbound ports? I never have to open ports at client locations to use the phones. |
| 19:02.24 | teapot | lol |
| 19:02.25 | file[laptop] | teapoint: what does that do? |
| 19:02.29 | Seyr | cluecon: I've used the phones behind SonicWall TZ, Cisco PIX and Linksys firewalls and have never touched the firewalls |
| 19:02.40 | teapot | file[laptop]: beep beep beep .... |
| 19:02.57 | file[laptop] | maybe the IAXy doesn't have it implemented? |
| 19:03.14 | file[laptop] | er surprised |
| 19:03.35 | file[laptop] | cute. |
| 19:03.38 | file[laptop] | incompatible encoding! |
| 19:04.26 | file[laptop] | oh call pickup |
| 19:04.34 | file[laptop] | gah I crashed my terminal |
| 19:04.36 | teapot | nothing seems to get logged for *8# from the IAXy |
| 19:04.40 | cluecon | Seyr: if your not using H323 and the SonicWall guy says it's because the firewall doesn't support H323 then i would think it has to do with having the ports open or closed. If the call is simply passing thru the firewall like normal data would then it wouldn't matter if the sonicwall supports H323 or not. |
| 19:04.40 | CoaxD | hah. I just did my ASL fingerspelling alphabet.. I haven't practiced in 2.5 months, and i learned it 3 months ago. but yet, it was right there, and i knew it all.. I *never* learn things that well. Sheesh. |
| 19:04.43 | teapot | ja, pickup |
| 19:04.52 | file[laptop] | maybe it's not implemented like I said! |
| 19:05.01 | teapot | sorry, I mean "yes, call pickup" ;) |
| 19:05.13 | file[laptop] | let me uncrash my terminal and see |
| 19:05.39 | cluecon | ~cluecon |
| 19:05.40 | jbot | hmm... cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 19:05.40 | file[laptop] | nope not implemented |
| 19:05.42 | file[laptop] | please try again. |
| 19:05.45 | outtolunc | anyone know if A@H has gcc on it? |
| 19:05.48 | teapot | erk |
| 19:06.11 | teapot | <dumb question> where would it be implemented if it was ? |
| 19:06.31 | file[laptop] | another channel driver. |
| 19:06.35 | file[laptop] | if you REALLY wanted I could do it |
| 19:06.50 | teapot | :-D |
| 19:07.11 | Ariel_ | outtolunc, yes |
| 19:07.23 | file[laptop] | teapot: what do I get? |
| 19:07.32 | kimosabe | does any one have meet me with out having fxo cards ? |
| 19:08.06 | kimosabe | i currentlly have 8 sipura devices in diffrent areas but i get not a valid confrence number eerrror |
| 19:08.11 | file[laptop] | pfft |
| 19:08.15 | shido | use RTC, kimosabe |
| 19:08.16 | teapot | lol |
| 19:08.30 | kimosabe | shido is there a read me on rtc ? |
| 19:08.42 | shido | file is the readme |
| 19:08.52 | shido | but I think he needs something cute for his new mac |
| 19:09.32 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 19:09.45 | file[laptop] | I'll be right back |
| 19:09.54 | outtolunc | well i would think that if a person had actually used it, then it might be a dumb question, which obviously i have not, and since some are telling you that you are wrong.. hmmm |
| 19:11.00 | kimosabe | shido excuse my ignorance where is the readme |
| 19:11.31 | brimstone | yes, but you have to question who put the mac in the closet to begin with |
| 19:11.34 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 19:12.16 | *** join/#asterisk vooduhal (~christoph@67.19.25.178) |
| 19:12.56 | dros7 | anybody know where I can get an fxo card close to toronto? |
| 19:13.42 | vooduhal | Hey guys. Quick question. I have 2 zaptel cards with 4 fxo modules each, and it takes 2 rings before the card will pick up the incoming call. Does anyone know how to get it to pick up as soon as possible. |
| 19:13.52 | brimstone | dros7: Last one on this page: http://www.digium.com/index.php?menu=order®ion=North%20America |
| 19:13.54 | file[laptop] | teapot: any thoughts? hrm? eh? EH? |
| 19:14.15 | teapot | file[laptop]: where did you look to see that it hadn't been implemented ? |
| 19:14.23 | file[laptop] | chan_iax2 |
| 19:14.28 | [TK]D-Fender | vooduhal : yOU PROBABLY HAVE FAX DETECTION @ CALLER id DEALYS ON |
| 19:14.33 | teapot | thought so |
| 19:14.33 | brimstone | vooduhal: immediate=yes in zapata.conf, but you'll lose caller id |
| 19:14.45 | Ariel_ | vooduhal, yes callerid=no it's waiting for hte caller ID also you can add immediate=yes |
| 19:14.52 | teapot | 9300 lines of module :p |
| 19:14.53 | vooduhal | Thank you. |
| 19:14.55 | file[laptop] | probably never implemented because IAX2 wasn't designed as an ata protocol |
| 19:14.58 | file[laptop] | more server to server |
| 19:14.59 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 19:15.01 | teapot | almost as bad as chan_zap |
| 19:15.12 | teapot | aah! |
| 19:15.16 | pjz | so what do people do for faxes? |
| 19:15.30 | [TK]D-Fender | SpanDSP |
| 19:15.33 | pjz | hrm |
| 19:15.38 | pjz | fax-machine-as-ui is kind of nice though |
| 19:15.54 | Ariel_ | pjz, spandsp and put an actuall fax machine on a fxs port |
| 19:16.00 | file[laptop] | teapot: I can put it in... but the best I could do for testing would be from another server |
| 19:16.11 | dros7 | brimstone: thx :) |
| 19:16.15 | pjz | okay, so is there a standard email-to-spandsp toy someplace? |
| 19:16.24 | pjz | or cgi? |
| 19:16.25 | pjz | or something? |
| 19:16.31 | teapot | file[laptop]: you can toss a patch in my direction - I can take it from there... |
| 19:16.44 | brimstone | dros7: any time |
| 19:17.13 | MikeJ[Laptop] | file[laptop], dont do it.. it's not worth it |
| 19:17.22 | MikeJ[Laptop] | make him paypal you |
| 19:17.29 | file[laptop] | I'll make it then hold it hostage |
| 19:17.31 | teapot | file[laptop]: It's not critical to my life ;) |
| 19:18.06 | teapot | MikeJ[Laptop]: ZA citizens are barred access to paypal :p |
| 19:18.41 | vooduhal | That worked great. Thanks guys. |
| 19:19.24 | *** join/#asterisk smash- (smash@12.108.5.205) |
| 19:20.35 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 19:20.41 | file[laptop] | hrmph |
| 19:21.27 | file[laptop] | MikeJ[Laptop]: what'cha up to? |
| 19:22.02 | MikeJ[Laptop] | ordering norstart phone system parts |
| 19:22.10 | MikeJ[Laptop] | norstar |
| 19:22.15 | file[laptop] | sounds exciting |
| 19:22.25 | MikeJ[Laptop] | yep |
| 19:22.26 | file[laptop] | almost like... selling your soul! |
| 19:22.30 | MikeJ[Laptop] | ummm |
| 19:22.40 | MikeJ[Laptop] | well, that's the system in that office right now |
| 19:22.46 | MikeJ[Laptop] | and they need the callerid |
| 19:23.00 | MikeJ[Laptop] | hey, 4 port fxo cards are only $100 |
| 19:23.06 | MikeJ[Laptop] | so better than digiums |
| 19:23.22 | file[laptop] | but can you make it sing a song? |
| 19:23.50 | MikeJ[Laptop] | ummm |
| 19:23.53 | MikeJ[Laptop] | no |
| 19:24.00 | MikeJ[Laptop] | but digiums cards can't either |
| 19:24.12 | file[laptop] | sure they can |
| 19:25.27 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 19:27.36 | vooduhal | Ok guys. Here is another question. We have some Polycom IP 500s, and some PSTN lines connected to those 4 modular cards. We've been having major problems with echo and after working with digium and doing everything they told us we still had the problem. We figured out a solution, which is along with echotraining and echocancel to add a couple of w's when dialing the number. Does anyone know if this is a good solution, or if there is something else |
| 19:28.16 | r0d3nt | I could never completely get rid of echo with polycom phones. |
| 19:28.19 | jeremywhiting | that ought to work, just puts a brief pause before dialing |
| 19:28.29 | jeremywhiting | [TK]D-Fender: ping |
| 19:28.46 | r0d3nt | Polycom's engineers told me their phones have no echo cancellation on the handset, but do on the speakerphone.... |
| 19:29.52 | MikeJ[Laptop] | ;P |
| 19:29.57 | r0d3nt | this was with a T1 or a PRI |
| 19:30.06 | r0d3nt | the only way to get rid of echo with polycom was not to use polycom |
| 19:30.16 | r0d3nt | I've experienced this on 3 different PBX's.... |
| 19:30.56 | r0d3nt | basically the polycoms are cheap for a reason... |
| 19:31.00 | *** join/#asterisk alerios (~alerios@63.245.86.169) |
| 19:31.46 | r0d3nt | they are designed to work with Polycom's PBX which has it's own crappy propriatary echo cancellation.... |
| 19:32.43 | truz24 | what do you use rod3nt ? |
| 19:32.44 | Ariel_ | r0d3nt, strange I like the polycom's and have been using them with those issues. |
| 19:33.38 | Ariel_ | with/without |
| 19:33.50 | truz24 | what are the polycoms for ? |
| 19:33.55 | truz24 | the agents ? |
| 19:33.57 | smash- | hey |
| 19:34.11 | *** join/#asterisk Apple (~appleboy@Appleboy.wikipedia) |
| 19:34.24 | smash- | what signaling method do i use for zapata.conf? |
| 19:34.26 | jeremywhiting | any of you with polycoms know how to set a ring tones on a per-registration basis. e.g. different rings for each line button? |
| 19:34.43 | Apple | anybody know of a way for when people are in a conference room for them to press something like *6 and it will mute their line, but they can still hear, and then *7 to unmute? |
| 19:34.48 | smash- | ; Signalling method (default is fxs). Valid values: |
| 19:35.15 | JerJer | Apple: sure its called app_meetme |
| 19:35.36 | brimstone | Apple: i believe the default is *1 to mute |
| 19:35.41 | Apple | nice |
| 19:35.54 | JerJer | show application meetme |
| 19:36.26 | alerios | <PROTECTED> |
| 19:36.37 | Apple | bbiam |
| 19:36.40 | Apple | err, whoops |
| 19:40.03 | *** join/#asterisk jo3sm1th (~email@68.252.65.130) |
| 19:40.09 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 19:40.36 | jo3sm1th | Once you have registered Vp in iax.conf how do you set extensions.conf to ring your Firefly phone when someone cals your Voicepulse DID |
| 19:42.23 | *** join/#asterisk madchimp (~joeblow@pcp05189678pcs.sanarb01.mi.comcast.net) |
| 19:42.30 | anthm | context=rtfm ... [rtfm] exten => 1000,1,Dial(SIP/rtfm) |
| 19:42.41 | pjz | ~meetme |
| 19:42.45 | Godsey | something like Dial(IAX2/Firefly) if you have [Firefly] in iax.conf |
| 19:42.48 | pjz | where are full meetme docs? |
| 19:42.51 | Marlow | anthm: eh .. load and clear :) |
| 19:43.19 | Godsey | LOAD "*",8,1 |
| 19:43.42 | [TK]D-Fender | I didn't know Polycom had their own PBX, every rep I've spoken to says they only authorize other provider's solutions with their phones. |
| 19:43.43 | cluecon | pjz: start here http://www.voip-info.org/wiki-Asterisk+cmd+Meetme |
| 19:43.45 | Marlow | Godsey: RUN |
| 19:43.50 | file[laptop] | hrm |
| 19:44.00 | cluecon | Polycom has a PBX? |
| 19:44.13 | anthm | meetme .... in hell mwa ha ha ha |
| 19:44.14 | Godsey | [TK]D-Fender: polycom doesn't allow people to just buy polycom phones. |
| 19:44.24 | [TK]D-Fender | That's what r0d3nt alluded to |
| 19:44.37 | Godsey | people buying them from places like ingram micro are not protected under waranty from polycom |
| 19:44.42 | [TK]D-Fender | Godsey : Not far from given the # of sources I can get them from. |
| 19:44.49 | pjz | cluecon: oh, I've got it set up, I was looking for more of a User's Guide for ... my users. |
| 19:45.14 | ctooley | Godsey, yes the are |
| 19:45.14 | [TK]D-Fender | Godsey : I'm looking to get mine through CCP (a fully auth'd reseller in Canada) |
| 19:45.18 | Marlow | pjz: write one :) |
| 19:45.19 | Godsey | polycom makes their money off software, not the hardware |
| 19:46.14 | Godsey | unless you buy the phone from a polycom partner as part of a bundle incluing the software you are not covered w/ polycom |
| 19:46.15 | ctooley | A new phone purchased through Ingram Micro, Techdata, T2 Supply, or other wholesale distributors are fully covered under the warranty unless the wholesaler sells it to a reseller that isn't authorized |
| 19:46.48 | Godsey | your only recource is to try and RMA w/ who ever you purchased it from |
| 19:46.53 | ctooley | Godsey, I sold hundreds of phones without software, using Asterisk, with Polycom fully aware that I was using Asterisk and seveal got replaced |
| 19:47.35 | Godsey | I get mine replaced to, but I have to lie and tell them I'm using one of their solution providers platforms |
| 19:47.44 | jo3sm1th | GOdsey what if you dont have Firefly in iax.conf |
| 19:47.52 | file[laptop] | cool it worked... |
| 19:47.54 | Godsey | a nice engineer guided me into that course of action to get phones replaced |
| 19:47.59 | ctooley | And the reseller doesn't even have to do the RMA, several customers just got the RMA form from Polycom and got phones fixed. |
| 19:48.04 | Godsey | jo3sm1th: you have it somewhere |
| 19:48.13 | Godsey | maybe sip.conf I don't use softphones |
| 19:48.38 | jo3sm1th | Can you just do exten => 3102221111,1,Dial(IAX2/softphonename) |
| 19:49.56 | *** join/#asterisk sepski (~sep@217.17.211.51) |
| 19:50.03 | Godsey | .. thisis ... *.conf [thisis] |
| 19:50.14 | Godsey | oops Dial(*/thisis)... |
| 19:50.18 | Ariel_ | strange I have a polycom that is hissing right now. I called polycom yesterday and gave them the s/n and they just issue me an RMA for replacement. |
| 19:50.50 | Godsey | I had all kinds of problems last year trying to get some DOA IP300 and IP600 phones replaced |
| 19:50.54 | Ariel_ | they did not want to know who or where I got it from. |
| 19:51.20 | Ariel_ | Godsey, maybe they have changed there ways |
| 19:51.21 | jo3sm1th | Godesey is that for me? |
| 19:51.28 | Godsey | jo3sm1th: yes |
| 19:51.37 | jo3sm1th | I don't understand how is exten => 3102221111,1,Dial(IAX2/softphonename) wrong |
| 19:51.42 | *** join/#asterisk mistral (mistral@jstevenson.plus.com) |
| 19:51.48 | Godsey | pay for a licence for firefly for me and I'll help more :P |
| 19:52.16 | SpaceBass | jo3sm1th i think you need a timeout at least |
| 19:52.17 | sepski | would this be the right channel to ask questions about the possibilities of asterisk, in order to find out if it's something we could use in our local county administration ? |
| 19:52.19 | Godsey | it does video right? |
| 19:52.31 | SpaceBass | sepski as good as any :) |
| 19:52.40 | Godsey | SpaceBass: I don't think ,60 is required |
| 19:52.40 | Lee__ | sepski: you should probably consult the info on the wiki first http://www.voip-info.org |
| 19:52.47 | sepski | been there |
| 19:52.51 | Lee__ | there's copious amounts there. |
| 19:52.52 | cluecon | jo3sm1th: your dial command should be Dial(IAX2/iaxusername) |
| 19:53.12 | Godsey | I think I was thinking of other softphone software |
| 19:53.22 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 19:53.23 | cluecon | sepski: what's the question? |
| 19:53.26 | Godsey | eyeBeam :) |
| 19:53.32 | Lee__ | sepski: also you can search the list archives vial google "site:lists.digium.com keyword" |
| 19:53.42 | Lee__ | s/vial/via/ |
| 19:53.52 | Godsey | how does eyeBeam work in a meetme? Heh |
| 19:54.11 | jo3sm1th | Thats what I have |
| 19:54.16 | jo3sm1th | and it is registered in iax.conf |
| 19:54.36 | jo3sm1th | And it makes calls from firefly just doesnt receive any |
| 19:54.42 | Godsey | and? |
| 19:54.45 | Godsey | I keep telling you |
| 19:54.53 | Godsey | IAX2/WHATyouNAMEDit |
| 19:55.02 | dros7 | anybody know where I can buy cheaper knockoff FXO cards in the toronto area? |
| 19:55.29 | SpaceBass | dros7 always some on ebay |
| 19:55.31 | [TK]D-Fender | Ariel_ : I'd sooner believe that they didn't care who you got it through or that they traces the serial # to the reseller and didn't care |
| 19:55.31 | Ariel_ | jo3sm1th, you have to use the users context name in the iax.conf [user1] |
| 19:55.31 | cluecon | jo3sm1th: what shows on the CLI when it doesn't receive calls? |
| 19:55.43 | jo3sm1th | Nothing at all |
| 19:55.55 | SpaceBass | where is voice mail stored? |
| 19:56.02 | Ariel_ | [TK]D-Fender, could be. either way it's on it's way to them. |
| 19:56.06 | jo3sm1th | It just says "my call did not go through" |
| 19:56.27 | cluecon | nothing? increase your vvvvvvvv level and pastebin the cli output. |
| 19:56.45 | dros7 | Spacebass: thx, I should have thought of that... |
| 19:56.56 | [TK]D-Fender | Ariel_ : Yup :) |
| 19:57.09 | SpaceBass | dros7 I got mine for like $7 each on ebay |
| 19:57.21 | Ariel_ | SpaceBass, /var/spool/asterisk/voicemail/default |
| 19:57.29 | sepski | we have a rather large pbx in the local administration local office, all small official buildings are connected with broadband, and all have small time pbx's paying the telco bloodmoney to connec to the central pbx, idea was to replace all the small pbx's out there with a asterisk server centraly and ip phones remotly, (i assumed propritary pbx phones dont work with asterisk), question was if it's possible to interoperate with a local pbx, using 4 digit inte |
| 19:57.29 | sepski | rnal numbers, and also get citylines thru the large pbx, dialing 0+ 8digits. |
| 19:57.43 | jo3sm1th | There is NOTHING, no response from CLI like it never reaches my pbx at all |
| 19:58.21 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:58.38 | jo3sm1th | Oh |
| 19:58.41 | jo3sm1th | I dont have "answer" |
| 19:58.58 | dros7 | Spacebass: did you buy them from a canadian? |
| 19:59.16 | SpaceBass | dros7 england I think... but payed us dollars |
| 19:59.17 | Ariel_ | sepski, you might have to get a consultant in your area to help out. But it can be done with asterisk. |
| 19:59.18 | Lee__ | sepski: yes, although you might be limited by the outbound bandwidth on the small official building's broadband connections |
| 19:59.20 | cluecon | sepski: should be able to do that. you'll need to be able to connect between * and the central pbx. |
| 19:59.39 | gambolputty | Is there any way to use the manager originate command to immediately ring the caller as the callee is ringing? |
| 19:59.40 | sepski | Lee__, 100mbit CoS fiber network all around |
| 20:00.02 | gambolputty | The caller doesn't get rung unless the callee picks up. |
| 20:00.02 | sepski | Lee__, and they dont have more then 4 lines today anyways, it's realy small officews |
| 20:00.07 | Lee__ | sepski: the company where I work has a "business class" cable line and it drops out every so often. |
| 20:00.09 | dros7 | spacebass: ah, ok. I want to go pick it up. Need it fast, so shipping from states isn't really an option right now |
| 20:00.26 | SpaceBass | I have a business class cable line at my home |
| 20:00.27 | cluecon | gambolputty: isn't that how it works now? |
| 20:00.28 | Lee__ | sepski: if you have enough outbound for as many calls as you have phones, you're good to go. |
| 20:00.33 | SpaceBass | 8mbs down / 1 mbs up |
| 20:00.36 | gambolputty | no |
| 20:00.39 | gambolputty | but I want it do |
| 20:00.41 | sepski | Lee__, our network have hardware to give voip packets priority |
| 20:00.41 | gambolputty | to |
| 20:00.50 | Lee__ | you sound quite well hooked up. |
| 20:00.52 | gambolputty | I want the caller and callee to ring at the same time |
| 20:01.00 | mistral | sepski: QOS is not a hardware its more a software thing |
| 20:01.01 | SpaceBass | just saying never had dropouts (yet) |
| 20:01.08 | jo3sm1th | Its got to have 2 lines exten => XXX,1,Answer |
| 20:01.09 | gambolputty | so that the caller can hear the callee's phone ringing |
| 20:01.11 | Lee__ | mistral: cisco switches have QoS |
| 20:01.14 | sepski | qos is software in our switches (hardware) |
| 20:01.15 | jo3sm1th | Doesnt it? |
| 20:01.18 | FuRR_ | sepski, QOS only works as far as your edge ;-) |
| 20:01.19 | SpaceBass | yeah but a traffic shaper works on level 3 |
| 20:01.30 | cluecon | gambolputty: switch your caller and callee then. |
| 20:01.35 | sepski | FuRR_, i know, but this is all in our own network |
| 20:01.42 | mistral | SpaceBass: QOS can work on level 2 - 4 |
| 20:01.47 | mistral | an beyond :> |
| 20:01.48 | SpaceBass | oh |
| 20:02.01 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:02.29 | sepski | between * and pbx, could i use multiple isdn 64K's ? how many can fit in * ? |
| 20:02.51 | Uther_P | are there any plans for a dynamic, db served dial plan? |
| 20:03.02 | Lee__ | Uther_P: search the wiki for Asterisk Realtime |
| 20:03.04 | sepski | i assume i must maintain a list on * telling it what 4 digit numbers are on voip and what are on the pbx ? |
| 20:03.04 | gambolputty | I don't think that will work |
| 20:03.09 | Uther_P | ah, thanks |
| 20:03.15 | gambolputty | both should ring at the same time |
| 20:03.20 | gambolputty | not one at a time |
| 20:03.30 | Lee__ | sepski: yeah, that's the dialplan. you'll do all your routing in Asterisk |
| 20:03.59 | cluecon | sepski: not sure about the 64k isdn lines. can you do t1 interfaces to the pbx? |
| 20:04.24 | sepski | cluecon, with the size of that thing, i'd be suprised if it couldt have such interfaces added |
| 20:04.55 | sepski | e1 in my case |
| 20:05.24 | cluecon | sepski: if it can do t1/e1 lines to the * side, you could pickup digium cards (4 t1/e1) that would give you access to asterisk from the pbx and from asterisk to the pbx. |
| 20:06.25 | sepski | cluecon, very nice, i was a bit worried if i could get enoughf lines between them, since most conversations is between small office and central |
| 20:06.42 | sepski | but an E1 should be plenty. |
| 20:07.09 | *** join/#asterisk Mike_TK (~Mike@bell.yes.net.ua) |
| 20:08.11 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 20:08.11 | *** topic/#asterisk is Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 || ResetCDR = broken bug 4531 has the FIX |
| 20:08.34 | cluecon | sepski: you can also get a single e1 interface card that will work with *. |
| 20:09.30 | sepski | cluecon, i'v read that on the web, what i was most worried about was 4 digits vs 8 digits etc |
| 20:09.57 | cluecon | sepski: extensions are very flexible in *. |
| 20:10.18 | cluecon | 4,8,16,32, even 1 digit extensions are possible. |
| 20:10.38 | cluecon | although i suspect someone would want to kill you if you used 32 digit extensions. |
| 20:10.55 | SpaceBass | anyone have expirence with distinctive rings on a zap channel? |
| 20:11.04 | sepski | theoreticaly if one wanted to reuse the propritary phones and pbxs on the small offices, could you set up * at all locations skipping the telco that way ? |
| 20:11.22 | cluecon | sepski: yes. |
| 20:11.38 | Lee__ | cluecon: that sounds like Windows programming for COM objects. |
| 20:11.57 | file[laptop] | MikeJ[Laptop]: poke |
| 20:12.28 | sepski | cluecon, and * could be 'transparent for the pbx's ? seams there are several ways to attach this. |
| 20:12.44 | cluecon | sepski: exactly |
| 20:13.15 | sepski | very flexible. but it would take a more machines, and more management then just replaceing i think. |
| 20:14.12 | sepski | perhaps a small embedded computer would suffice, like an itx board or similar at the small offices |
| 20:14.32 | Lee__ | sepski: I got Asterisk running on a Soekris 4801 with a 128meg CF card |
| 20:14.47 | Thumann | xD |
| 20:14.48 | Lee__ | that's too small for what you'd be doing but just as a point of reference... |
| 20:14.52 | Thumann | i'm so drunkj' |
| 20:14.53 | teapot | past bedtime |
| 20:15.16 | sepski | Lee__, as i said very small officces |
| 20:15.27 | Thumann | i wanna be a firetruck |
| 20:15.31 | Lee__ | that board is CPU bound. 233Mhz IIRC |
| 20:15.49 | Lee__ | transcoding is out of the question |
| 20:16.13 | sepski | isee |
| 20:16.30 | sepski | thanks for all your information, wife wants me of the comp now :P |
| 20:16.44 | cluecon | sepski: yvw. |
| 20:16.47 | sepski | test install next week i hope :) |
| 20:17.02 | sepski | cya |
| 20:17.22 | r0d3nt | [TK]D-Fender, what did I allude to ??? |
| 20:17.42 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 20:19.13 | Nethab | anyone here use the [authenticate] sectioin in sip.conf |
| 20:21.02 | smash- | hey |
| 20:21.17 | smash- | can someone help me set signalling on zapata.conf |
| 20:21.35 | smash- | is it cpe or pri-net |
| 20:21.36 | smash- | <PROTECTED> |
| 20:21.37 | smash- | <PROTECTED> |
| 20:21.38 | Nethab | yes, you can use it to specify all your outbound authentication |
| 20:21.38 | [TK]D-Fender | cluecon : My home ext's are organized as ABCD where A=2 (company#), B = 0/1 (hard/soft), C=Floor #, D=room# |
| 20:21.52 | Nethab | so you don't keep multiple copies in your register and peer defs |
| 20:22.23 | [TK]D-Fender | [15:31] <r0d3nt> they are designed to work with Polycom's PBX which has it's own crappy propriatary echo cancellation.... |
| 20:22.53 | [TK]D-Fender | Noone has confrmed there is a Polycom produced PBX. They only mention auth'd solution providers |
| 20:24.23 | *** join/#asterisk funxion (~funxion@mtnuser.icgws.com) |
| 20:25.04 | agave-txlink | ' |
| 20:25.06 | agave-txlink | ; |
| 20:25.25 | funxion | {<o>..<o>} |
| 20:26.11 | Ariel_ | Nethab, I would like to see more on this. but I did not see this in stable any where in there samples. |
| 20:26.37 | funxion | is there a way to reload * from the command prompt? |
| 20:26.40 | Nethab | it's in the sip.conf.sample provided in HEAD |
| 20:26.50 | funxion | let me correct that from shell not CLI |
| 20:26.56 | Uther_P | funxion: you could pipe the comamnds to the cli |
| 20:26.56 | Ariel_ | Nethab, ahh it's new then. I only use stable right now. |
| 20:27.19 | Nethab | i was asking cause there is no documentation other than the config file and it's 2 line sample |
| 20:27.23 | Uther_P | funxion: like, printf "reload\nquit\n" | asterisk -r |
| 20:27.28 | Uther_P | funxion: that might work |
| 20:27.46 | funxion | wouldn't that still start a remote session |
| 20:28.19 | funxion | I was looking for somehting like asterisk -rx without actually restarting |
| 20:28.24 | vooduhal | Can anyone suggest someone who offers a course in general VoIP technologies? |
| 20:28.28 | Uther_P | funxion: yes, it opens the cli, then reloads, then quits.... if you don't wanna see it... redirect it to null.... printf "reload\nquit\n" | asterisk -r > /dev/null |
| 20:28.46 | funxion | hmm |
| 20:28.48 | funxion | nice |
| 20:28.57 | funxion | thnx a lot Uther_P |
| 20:28.58 | r0d3nt | [TK]D-Fender, Interesting... when I worked with Polycom to try and resolve the echo issues.. in the end, they took back all the phones.... |
| 20:29.01 | Uther_P | no problem |
| 20:29.29 | vooduhal | I think we have our polycom echo issues resolved finally. |
| 20:29.50 | [TK]D-Fender | r0d3nt : my only comment was about Poly not making a PBX like you seemed to imply |
| 20:29.57 | Ariel_ | vooduhal, you added ww to the dial out string. |
| 20:30.32 | funxion | Uther_P ti doesn't werk |
| 20:30.38 | funxion | nice try though |
| 20:30.41 | Uther_P | hmm |
| 20:30.45 | vooduhal | Well, the combination of: rxgain=4.0,txgain=4.0, echocancel=64,echotraining=800, and the w to dialing out seemed to fix it both directions. |
| 20:30.54 | vooduhal | That and resolving the IRQ conflicts. |
| 20:31.03 | r0d3nt | [TK]D-Fender, I was under the impression that they do make a PBX.. |
| 20:32.16 | Uther_P | funxion: in the asterisk manpage... -x command |
| 20:32.33 | Uther_P | funxion: it says, conenct to a running astersisk process and execute a command on a comman line |
| 20:32.50 | funxion | thnx |
| 20:32.51 | Uther_P | funxion: sooo... asterisk -x reload |
| 20:33.11 | Uther_P | funxion: fyi, that pipes any cli output to the stdout |
| 20:33.38 | Uther_P | funxion: so it you want it silent, do asterisk -x reload > /dev/null <-- or maybe to a log file if you are scripting it |
| 20:34.19 | vooduhal | I thought it was asterisk -rx 'command' |
| 20:34.21 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 20:34.29 | shmaltz | hi evry1 |
| 20:34.32 | Uther_P | vooduhal: -x implies -r |
| 20:34.45 | vooduhal | Oh. |
| 20:34.56 | Uther_P | when -R is not explicity supplied |
| 20:35.22 | Uther_P | the example in the manpage shows -rx but not needed |
| 20:35.31 | vooduhal | So anyone have any suggestions for training? |
| 20:36.14 | Uther_P | training? |
| 20:36.21 | shmaltz | I had asterisk 1.0.7 crash on me, I have no clue why, I'm assuming that the reason was that my PRI went down and asterisk didn't handle it nicely, has any1 seen this before? |
| 20:36.29 | Uther_P | umm... rtfwiki :P |
| 20:36.58 | drumkilla | shmaltz: no ... did you start asterisk with -g? |
| 20:37.06 | shmaltz | nope |
| 20:37.10 | drumkilla | ok ... |
| 20:37.16 | shmaltz | so shoud I? |
| 20:37.17 | drumkilla | well, if this turns out to be something you can recreate |
| 20:37.23 | drumkilla | yes |
| 20:37.26 | Uther_P | shmaltz: asterisk used to crash on me all the time when I was using a digium quad fxo card.... once I got rid of it and went all voip, I didn't have that problem |
| 20:37.36 | drumkilla | also, build asterisk with 'make valgrind' |
| 20:37.46 | drumkilla | so we can get a clean backtrace |
| 20:37.48 | shmaltz | Uther_P, well I'm not using TDM or fxo |
| 20:37.53 | drumkilla | Uther_P: did you contact support about it? |
| 20:37.57 | funxion | Uther_P wooduhal was right |
| 20:38.08 | funxion | it doesn't work withour r |
| 20:38.12 | shmaltz | thanks drumkilla, I also had an issue with HEAD |
| 20:38.23 | Uther_P | funxion: hrm.. heeh the man page lied |
| 20:38.36 | Uther_P | drumkilla: no, the digium card fried on me, and we were migrating to VoIP anyway |
| 20:39.08 | drumkilla | Uther_P: did you RMA the card? |
| 20:39.11 | shmaltz | drumkilla, is there any disadvantage of using -g? |
| 20:39.12 | Uther_P | funxion: heh, the manpage says that -x implies -r unless you specify -R, guess that was a typo |
| 20:39.16 | drumkilla | shmaltz: no |
| 20:39.21 | Uther_P | drumkilla: nope, still got it around here somewhere |
| 20:39.28 | drumkilla | it just tells asterisk to do a core dump if it crashes |
| 20:39.35 | drumkilla | Uther_P: well, if it's messed up, you can get a new one |
| 20:39.51 | Uther_P | but... I don't want a new one... that card was just problems man |
| 20:40.05 | drumkilla | you probably have a really old rev of the card |
| 20:40.07 | Uther_P | well... probably the drivers more than the card, but |
| 20:40.17 | drumkilla | anyway, just making sure you are aware of your options ..... |
| 20:41.37 | Uther_P | for example... if for some reason the module was removed before asterisk was stopped, the system would hang completely, and it would put the card in a state where it wouldn't mod probe again unless you completely powered off the server and powered it back up again |
| 20:41.42 | Uther_P | a simple reboot didn't fix it |
| 20:43.21 | shmaltz | When using HEAD when ever I would get an Unable to forward frame error on the PRI, I would get dissconnected, but since downgrading to 1.0.7 it just displays the warning and keeps going |
| 20:43.24 | *** join/#asterisk X-Rob (~rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 20:43.38 | Sephen | Anyone here using meetme2? The php web app shows the users in the conference, but it doesn't do anything when I click on kick, talk, listen, etc. |
| 20:45.12 | X-Rob | I go away for a night, and 'AEL' is invented. |
| 20:45.14 | X-Rob | Geez. |
| 20:45.16 | Sephen | shmaltz: I just pulled down HEAD two days ago, and I see only the warnings. |
| 20:45.43 | shmaltz | Ok, I'm talking about around March |
| 20:47.47 | *** join/#asterisk B2382F29 (~daniel@dsl-084-058-144-023.arcor-ip.net) |
| 20:48.15 | [TK]D-Fender | Hey shmaltz |
| 20:48.22 | [TK]D-Fender | Going well with Poly * plan |
| 20:49.23 | shmaltz | D-Fender, not bad, having some issues with NAT |
| 20:49.28 | Sephen | TKD-Fender: Our Poly is going great too. |
| 20:49.38 | *** join/#asterisk tux_rulez (~sfeil@m090e36d0.tmodns.net) |
| 20:50.40 | *** join/#asterisk map71 (~map@adsl-195-213-fixip.tiscali.ch) |
| 20:50.42 | *** join/#asterisk Marlow (~marlow@159-134-145-182.as1.mvw.galway.eircom.net) |
| 20:50.54 | Corydon-w | Yeah, I agree... tux rules... http://drunkcoder.com/tattoo.jpg |
| 20:51.20 | *** join/#asterisk predictive (~jeff@adsl-4-86-9.cae.bellsouth.net) |
| 20:51.28 | [TK]D-Fender | NAT = PITA |
| 20:51.33 | predictive | has anyone seen polycom 501s register on a bizarre port |
| 20:51.42 | Uther_P | haha who owuld do that |
| 20:51.43 | [TK]D-Fender | SO glad its not an iddue for me so far |
| 20:52.12 | file[laptop] | predictive: NAT. |
| 20:52.22 | Uther_P | now he needs to get the bsd daemon stabbing his pitch fork into his skull |
| 20:52.24 | tux_rulez | I was wondering, has anyone written an asterisk module to screen callers for a radio call-in show? |
| 20:52.44 | predictive | file[laptop]: yea, but I have the NAT setting correct in the polycom, so I can't figure out why it's using this weird port |
| 20:52.46 | X-Rob | how would you do that? |
| 20:52.46 | Uther_P | heh, wow... thats rather specific |
| 20:52.51 | file[laptop] | predictive: the way your NAT works |
| 20:52.59 | B2382F29 | hi, i have a problem with call forwarding. i'm using chan_capi... and _X.,1,Answer _X.,2,Dial,CAPI/3127313:bYYYYYYYYY|30|tr in the context for incoming calls (YYYYY is a cellphone number). When i call asterisk with cellphone 1, the second cellphone gets called as intended, but i can only hear the sound from 1 to 2 ... from 2 to 1 i get no sound.... any ideas ? |
| 20:53.03 | X-Rob | Last time I looked, there's no way to check for IQ<80 from caller ID? |
| 20:53.21 | Sephen | Can Asterisk do PRI/B2 transfers? |
| 20:53.35 | Uther_P | B2382F29: sounds like a firewall/nat issue |
| 20:53.43 | B2382F29 | Uther_P, with CAPI? |
| 20:54.03 | X-Rob | Or maybe you could have an IQ test when they call in. 'How many sides does a triangle have? Push 1 for 'not three', push 2 for 'three'." If they push 2 they get put on hold |
| 20:54.09 | Uther_P | B2382F29: err... I'm not familiar with capi... so maybe I cannot help you |
| 20:54.12 | X-Rob | (forever( |
| 20:54.16 | X-Rob | otherwise they get put through |
| 20:54.19 | B2382F29 | Uther_P, ISDN |
| 20:54.59 | Uther_P | B2382F29: one phone dialing into the server, then it dialing back out to another cell phone? |
| 20:55.07 | B2382F29 | Uther_P, yes |
| 20:55.31 | Uther_P | hrm, odd... can't help ya there man |
| 20:55.39 | Uther_P | haven't jacked with isdn |
| 20:57.00 | B2382F29 | Uther_P, is it correct to first answer the call and then dial the second cellphone? so the extensions.conf could not be the problem? |
| 20:57.48 | Uther_P | B2382F29: no, that should be fine... the cli says "attempting to bridge call blah blah blah" |
| 20:58.07 | B2382F29 | Uther_P, CAPI[contr1/3127313]/1 answered CAPI[contr1/3127313]/0 |
| 20:58.26 | B2382F29 | Uther_P, Private bridge between CAPI[contr1/3127313]/0 and CAPI[contr1/3127313]/1 failed |
| 20:58.50 | Uther_P | hrm... |
| 20:59.01 | B2382F29 | Attempting native bridge.... is before that |
| 20:59.05 | Uther_P | have you searched for that problem |
| 21:00.57 | B2382F29 | Uther_P, didn't find anything... could it be a problem that the MSN is the same for the incoming and outgoing call? |
| 21:01.19 | Uther_P | like I said man... I don't know too much about isdn |
| 21:01.35 | Uther_P | does your server have a dsp? |
| 21:01.44 | Uther_P | can you dial from the cli out and get sound both ways? |
| 21:02.14 | B2382F29 | Uther_P, i can dial the voicemail, i hear it and my talking is recorded |
| 21:02.26 | Uther_P | just bridged than |
| 21:03.03 | B2382F29 | Uther_P, from the server i didn't test ... |
| 21:04.58 | Uther_P | B2382F29: did you read this? http://www.voip-info.org/tiki-index.php?page=Asterisk+config+capi.conf |
| 21:07.48 | ardor | whats capi? |
| 21:07.50 | predictive | man linksys must actively be trying to break sip |
| 21:07.57 | Uther_P | ardor: module for isdn |
| 21:08.53 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
| 21:08.56 | B2382F29 | Uther_P, well, my capi.conf looks like that and the connections to the voicemail are all working |
| 21:09.10 | Uther_P | B2382F29: you have 2 cards? |
| 21:09.16 | B2382F29 | Uther_P, no, 2 |
| 21:09.19 | B2382F29 | aeh 1 |
| 21:09.28 | Uther_P | B2382F29 which card are you using |
| 21:09.47 | Marlow | lol: http://www.chromance.de/wtf/lol.htm |
| 21:09.48 | B2382F29 | AVM Fritz! with fcpci |
| 21:10.50 | Uther_P | B2382F29: http://www.voip-info.org/tiki-index.php?page=Asterisk+CAPI+channels |
| 21:11.23 | Uther_P | "Note on DID, AVM Fritz A1 and fcpci, You cannot use the AVM Fritz A1 card in P2P mode where you can use DID. " |
| 21:12.18 | B2382F29 | Uther_P, yes.i can not use direct dialing... but i don't dial, i just forward the call |
| 21:12.39 | Uther_P | looked like it pertained... wasn't sure what it ment however |
| 21:13.07 | *** join/#asterisk cianhughes (~cian@cian.ws) |
| 21:15.25 | B2382F29 | creating pipe for PLCI=0x101 msn = 3011883 .................. creating pipe for PLCI=-1 |
| 21:15.38 | B2382F29 | the second looks suspicious |
| 21:15.45 | smash- | <PROTECTED> |
| 21:15.57 | smash- | <PROTECTED> |
| 21:16.09 | Sephen | If I use rtcachefriends=yes in the sip.conf, how often does Asterisk re-read the list? For instance, if I make a change to someone |
| 21:16.15 | Sephen | err. |
| 21:16.44 | Sephen | If I make a change to someone's config, will asterisk eventually re-read it by itself (a timeout feature to cache), or do I manaully have to force a sip reload? |
| 21:16.52 | anthm | if it re-registers ot always overwrites it |
| 21:17.24 | Lee__ | is there a directory option that reads back the extension the call is getting transfered to before it rings that phone? |
| 21:17.50 | Lee__ | I have this buggy directory AGI that does it but it won't exit when it's done. |
| 21:18.08 | smash- | ~docs |
| 21:18.12 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:18.12 | Sephen | anthm: Only during a re-reg though? It doesn't periodically pull an update? |
| 21:18.20 | anthm | unless you set rtautoclear the peer will stay in there forever but it's still subject to modification all the while |
| 21:18.39 | Sephen | smash- I'm using CPE, and it works fine. |
| 21:18.50 | sivana | does every call to ${UNIQUEID} in the same context yield a different number? |
| 21:19.11 | sivana | or is it the same per context? |
| 21:19.20 | smash- | <PROTECTED> |
| 21:19.30 | Sephen | anthm: Do you have a website which documents all of the realtime directives somewhere? Documentation seems sparse. |
| 21:19.39 | anthm | rtautoclear can be yes to match the expire time exactly |
| 21:19.48 | anthm | or a number for that many seconds |
| 21:20.09 | anthm | the configs/sip|iax.conf.sample is all the docs |
| 21:21.26 | Sephen | Thanks |
| 21:22.14 | *** join/#asterisk brian13 (~user@c-24-98-71-208.hsd1.ga.comcast.net) |
| 21:23.25 | *** join/#asterisk m-00kie (3704558@host-14-204-9-69.midco.net) |
| 21:24.19 | m-00kie | if anyone has any suggestions as to how to set up a videophone system, id appreciate any pointers :) |
| 21:25.25 | *** join/#asterisk map71 (~map@adsl-195-213-fixip.tiscali.ch) |
| 21:27.47 | sivana | d |
| 21:30.28 | shido | UC channel 1 protocol error. Cause 32773 |
| 21:30.29 | shido | ? |
| 21:31.26 | smash- | hey |
| 21:31.29 | smash- | in extensions.conf |
| 21:31.34 | smash- | do i have to monfiy TRUNK=Zap/g2 ; Trunk interface |
| 21:31.34 | smash- | TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) |
| 21:31.37 | smash- | <PROTECTED> |
| 21:31.51 | smash- | <PROTECTED> |
| 21:32.12 | smash- | ~docs |
| 21:32.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:36.35 | map71 | smash: I would like to know that also. On my (working) server I have no such TRUNK variable. Where is it used? |
| 21:37.00 | smash- | <PROTECTED> |
| 21:37.07 | smash- | <PROTECTED> |
| 21:37.14 | smash- | <PROTECTED> |
| 21:37.18 | smash- | <PROTECTED> |
| 21:37.21 | smash- | <PROTECTED> |
| 21:37.28 | smash- | <PROTECTED> |
| 21:37.35 | smash- | <PROTECTED> |
| 21:37.36 | sivana | anyone using spandsp here? |
| 21:37.50 | smash- | <PROTECTED> |
| 21:38.31 | smash- | <PROTECTED> |
| 21:38.34 | wrmem | TRUNK is just a variable, often used to make dialout strings shorter. |
| 21:38.34 | smash- | i just dont wanna stay late. |
| 21:38.38 | smash- | <PROTECTED> |
| 21:38.40 | smash- | .. |
| 21:38.53 | smash- | wrmem i setup zatel.conf and zapata.conf up |
| 21:39.01 | smash- | <PROTECTED> |
| 21:39.01 | map71 | smash: now I saw it. This TRUNK stuff you CAN use in the dialplan in a Dial command: Dial (${TRUNK}/...) ... |
| 21:39.02 | smash- | <PROTECTED> |
| 21:39.04 | smash- | <PROTECTED> |
| 21:39.47 | map71 | hi |
| 21:40.13 | map71 | has anybody a working a asterisk stable server with a working format_mp3? |
| 21:40.22 | map71 | it just does not work for me ... |
| 21:40.34 | wrmem | if zaptel & zapata config files are correct, and you have a context set up in extensions.conf to get incoming calls, you are set. But first, just concentrate on getting out of RED and YELLOW alarms tomorrow. Don't try to build beyond that until the alarms are gone. |
| 21:41.16 | smash- | <PROTECTED> |
| 21:41.20 | smash- | were do these alarms show |
| 21:41.22 | smash- | <PROTECTED> |
| 21:41.32 | smash- | kuz i route alot of traffic out |
| 21:41.34 | smash- | <PROTECTED> |
| 21:41.40 | smash- | <PROTECTED> |
| 21:41.47 | smash- | <PROTECTED> |
| 21:41.53 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 21:42.04 | |Vulture| | Anyone here a customer of Broadwing or Focal? |
| 21:42.06 | smash- | <PROTECTED> |
| 21:42.14 | smash- | <PROTECTED> |
| 21:42.15 | wrmem | smash-: read the docs, you'll find it |
| 21:42.18 | smash- | <PROTECTED> |
| 21:42.22 | smash- | <PROTECTED> |
| 21:42.25 | smash- | <PROTECTED> |
| 21:42.26 | smash- | lol |
| 21:42.27 | smash- | =) |
| 21:42.33 | *** join/#asterisk uuuppz (~uuuppz@oscar.esptl.com) |
| 21:42.39 | smash- | i got it |
| 21:42.47 | |Vulture| | god nothing but bad customer service on the account managment side... want to see if anyone else uses them to see if their rep is worth anything |
| 21:42.52 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 21:42.56 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 21:43.07 | |Vulture| | their tech guys are good... but their CS people.... wow |
| 21:43.10 | uuuppz | hi |
| 21:43.35 | Katty | exten => _xxxx,1,Dial(SIP/${EXTEN}|30) |
| 21:43.35 | Katty | exten => _xxxx,2,Voicemail(${EXTEN}) |
| 21:43.36 | Katty | exten => _xxxx,3,Hangup |
| 21:43.40 | Katty | ^- did i oops? |
| 21:43.55 | uuuppz | I'm trying to set up my voipbuster account as a trunk |
| 21:43.59 | uuuppz | nd am having problems |
| 21:44.08 | Katty | it's supposed to go to voicemail in 30 seconds (5 rings) but i think it's insaned |
| 21:44.16 | uuuppz | if anyone here has got this working, I'd appreciate a hand |
| 21:44.46 | Corydon-w | Creative Labs VOIPBlaster? |
| 21:44.52 | uuuppz | sorry |
| 21:44.55 | uuuppz | no |
| 21:44.56 | smash- | <PROTECTED> |
| 21:45.03 | uuuppz | thought I'd mistyped then |
| 21:45.05 | smash- | <PROTECTED> |
| 21:45.06 | uuuppz | no voipbuster |
| 21:45.50 | *** join/#asterisk concept10 (~concept10@c-67-166-167-125.hsd1.tx.comcast.net) |
| 21:46.42 | *** part/#asterisk concept10 (~concept10@c-67-166-167-125.hsd1.tx.comcast.net) |
| 21:48.28 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 21:50.02 | X-Rob | uuuppz, asterisk@home? |
| 21:50.23 | Ariel_ | Katty, why don't you use a macro for that |
| 21:50.33 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.189) |
| 21:50.40 | Hmmhesays | macro's are cool |
| 21:50.58 | Hmmhesays | keep you from having to type out an annoyingly long dialplan |
| 21:51.18 | Ariel_ | Katty, there should be a macro-standard if you setup off the sample. |
| 21:51.30 | uuuppz | X-Rob: nah installed from source on sarge, but I am using AMP |
| 21:51.34 | brimstone | macro-stdexten |
| 21:51.35 | Ariel_ | Hmmhesays, yes your correct |
| 21:52.05 | *** join/#asterisk mxmasster (~maxc@66.113.65.12) |
| 21:52.17 | X-Rob | uuuppz - the AMP channel is #amportal |
| 21:52.36 | *** join/#asterisk mariosit (~sitelco@201.138.189.253) |
| 21:53.09 | uuuppz | X-Rob: ah thanks, got it working anyway ;) |
| 21:53.19 | *** join/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 21:53.37 | mariosit | is posible use mfc/r2 with digium e1 card |
| 21:54.02 | Hmmhesays | 8 minutes then some fun in the sun |
| 21:54.30 | Nuxi | hmmm. I thought it was 8 light seconds to the sun. |
| 21:54.41 | Hmmhesays | I'm not going to the sun |
| 21:54.45 | Hmmhesays | i'm going for a motorcycle ride |
| 21:54.48 | mariosit | ha ha ha |
| 21:54.54 | *** part/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 21:54.58 | Ariek | Is asterisk able to do a hold call (with the callmanager) I can't find any reference to it |
| 21:54.58 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 21:55.47 | Hmmhesays | hopefully I won't see so many squids riding around without shirts this time |
| 21:56.03 | mariosit | what kind of mothebord is recomend to use with asterisk |
| 21:56.05 | mariosit | ? |
| 21:56.24 | shido | mariosit, yes |
| 21:56.37 | shido | im setting up r2 in saudi arabia now for a student |
| 21:56.43 | sivana | anyone using spandsp here? |
| 21:56.50 | Hmmhesays | r2 can be tricky |
| 21:57.10 | Nuxi | nope, Hmmhesays was right. according to http://www.google.com/search?q=1+AU+in+light+minutes&sourceid=opera&num=0&ie=utf-8&oe=utf-8, it's 8.31675359 minutes to the sun. |
| 21:57.17 | Hmmhesays | especially when they tell you "<insert country here> style r2" |
| 21:57.18 | *** join/#asterisk fugitivo (~ajf@201.255.99.90) |
| 21:57.23 | Hmmhesays | Nuxi: nice |
| 21:57.23 | fugitivo | hello |
| 21:58.05 | Hmmhesays | i was off by 33.33_ seconds |
| 21:58.16 | Hmmhesays | er.. off again, damn |
| 21:58.31 | mariosit | why r2 can be tricky i'm live in mexico |
| 21:58.59 | Hmmhesays | sometimes it's hard to get a straight answer from the telco's what the spec's are for the line they have |
| 21:59.18 | `Sauron | and sometimes you have to dictate to the telco what specs you want |
| 22:00.28 | Hmmhesays | sometimes the telco can't won't change the specs on the line and can't tell you what they are |
| 22:01.03 | `Sauron | sometimes your telco gets a notice that you'll switch 100+ circuits to a different carrier if they don't tell you what you want |
| 22:01.23 | `Sauron | ;) |
| 22:01.25 | Hmmhesays | LOL, good point there |
| 22:02.10 | `Sauron | They jumped in line quickly when we pulled one of those. |
| 22:02.13 | eKo1 | Too bad that can't work here...there is only one god...erm telco |
| 22:02.21 | `Sauron | Considering the fact that we're one of the larger cash cows they have |
| 22:02.26 | `Sauron | if not THE cash cow |
| 22:02.50 | `Sauron | and there's plenty of competitors who'd love to take on our account |
| 22:03.23 | `Sauron | 75-100 T's, ~15 GigE and an unknown number of other crap |
| 22:03.27 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 22:09.45 | *** part/#asterisk tux_rulez (~sfeil@m090e36d0.tmodns.net) |
| 22:10.57 | *** join/#asterisk asteriskforuk (~vircuser@i-195-137-59-254.freedom2surf.net) |
| 22:11.09 | asteriskforuk | Hello everyone.. |
| 22:11.17 | *** part/#asterisk Ariek (~Ariek@famklooster.demon.nl) |
| 22:12.04 | smash- | hey will make samples over right conf files? |
| 22:12.08 | smash- | <PROTECTED> |
| 22:12.14 | asteriskforuk | ne1 can help with connecting remote 7960... tried the FAQs error msg Registration from '<sip:786@192.168.0.200;user=phone>' failed for '195.137.59.254' |
| 22:12.18 | smash- | <PROTECTED> |
| 22:12.25 | `Sauron | yes |
| 22:12.30 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 22:12.54 | smash- | how do i check to see if PRI conf properly |
| 22:13.04 | smash- | <PROTECTED> |
| 22:13.38 | smash- | <PROTECTED> |
| 22:13.40 | smash- | ? |
| 22:13.46 | `Sauron | yea |
| 22:13.50 | mariosit | asterisk -r |
| 22:14.03 | twisted[work] | or zttool ;) |
| 22:14.42 | sivana | am I missing something... rxfax is hanging up before my fax is done being sent |
| 22:14.56 | mariosit | let me stand for exaple if i want to connect asterisk pbx to another asterisk pbx i need to use the iax protocol? |
| 22:15.07 | sivana | mariosit: I would |
| 22:15.50 | twisted[work] | mariosit, that's a good way to do it, yeah. other ways are via sip or hardware, but iax2 would be a good choice ;) |
| 22:16.02 | mariosit | cool |
| 22:16.18 | mariosit | ok i need to study how work iax well |
| 22:16.25 | mariosit | thanks for the advice |
| 22:17.11 | sivana | can someone send me a fax? :) |
| 22:17.38 | smash- | hrm docs on asterisk say http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x291.html#AEN457 |
| 22:17.39 | Lee__ | I'm having a hell of a time with an AGI script that doesn't exit when it's done. It's written in PHP. Could someone recommend instructions to force it to exit cleanly? |
| 22:17.44 | smash- | <PROTECTED> |
| 22:17.50 | smash- | <PROTECTED> |
| 22:18.27 | smash- | <PROTECTED> |
| 22:18.34 | smash- | <PROTECTED> |
| 22:19.00 | hardwire | reboot the machine |
| 22:19.02 | hardwire | its the only way |
| 22:19.17 | smash- | *CLI> zap show status |
| 22:19.17 | smash- | Description Alarms IRQ bpviol CRC4 |
| 22:19.17 | smash- | wanpipe1 card 0 OK 0 0 0 |
| 22:19.17 | smash- | *CLI> |
| 22:19.21 | smash- | <PROTECTED> |
| 22:19.39 | hardwire | I want a wanpipe :( |
| 22:19.44 | smash- | <PROTECTED> |
| 22:19.50 | smash- | <PROTECTED> |
| 22:19.50 | smash- | <PROTECTED> |
| 22:19.53 | smash- | <PROTECTED> |
| 22:20.10 | smash- | <PROTECTED> |
| 22:20.13 | smash- | <PROTECTED> |
| 22:20.36 | dalabera | smash RTFM !! http://www.voip-info.org/tiki-index.php?page=Asterisk+Starting+and+Stopping |
| 22:20.56 | eKo1 | Lee__: exit(); |
| 22:23.35 | *** join/#asterisk Navman (~par_edlun@62.108.206.77) |
| 22:24.23 | dalabera | sivana where you able to set the fax on asterisk? |
| 22:25.20 | sivana | partly |
| 22:25.27 | kn0x | anyone been able to get inband dtmf to work through FWD? |
| 22:25.29 | smash- | <PROTECTED> |
| 22:25.33 | dalabera | let me help you, what you need? |
| 22:25.34 | smash- | <PROTECTED> |
| 22:25.34 | sivana | having an issue with it hanging up before it's received |
| 22:26.10 | dalabera | what card and linux os are you? |
| 22:26.54 | sivana | slackware, te405p |
| 22:27.05 | kn0x | drumkilla- have you ever been able to get the Callback() application to work? |
| 22:27.30 | sivana | dalabera: I'm not using the fax ext.. sending a DID directly to rxfax() |
| 22:27.41 | *** join/#asterisk NSGN (~NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 22:27.48 | NSGN | hello |
| 22:27.53 | twisted[work] | heh. rxfax works great |
| 22:27.58 | NSGN | anybody here have experience with teravoice server? |
| 22:27.58 | sivana | dalabera: faxdetect=incoming |
| 22:28.03 | sivana | ack.. |
| 22:28.15 | twisted[work] | all of our voice did's here at the office double as our fax numbers ;) |
| 22:28.17 | sivana | twisted[work]: not so far :) |
| 22:28.24 | twisted[work] | yes, works beautifully |
| 22:28.40 | sivana | I'm doing this: exten => 7052232145,1,Macro(faxreceive,${EXTEN},richard@aspworld.com); FAX Receive TEST |
| 22:28.55 | sivana | hangs up before it captuers |
| 22:28.58 | sivana | captures |
| 22:28.58 | twisted[work] | uhm, that's a macro pointer. |
| 22:29.27 | twisted[work] | of course that won't work. |
| 22:29.30 | sivana | http://pastebin.ca/14810 |
| 22:29.38 | sivana | goes to that macro :) |
| 22:30.18 | twisted[work] | well, if it's hanging up before it's done, check the directory permissions |
| 22:30.25 | NSGN | :-/ |
| 22:30.34 | dalabera | good point... |
| 22:30.40 | twisted[work] | rxfax will handshake and all that jazz |
| 22:30.55 | twisted[work] | but then will disconnect as it begins to recieve fax data |
| 22:31.00 | sivana | I think I need the ARG1 to be existing in the dir first |
| 22:31.00 | twisted[work] | because it can't write it |
| 22:31.25 | sivana | by golly.. I think you're right |
| 22:31.38 | twisted[work] | i know i'm right |
| 22:31.41 | sivana | :) |
| 22:31.43 | twisted[work] | hehehe |
| 22:31.56 | sivana | bingo... thank you |
| 22:31.58 | twisted[work] | np |
| 22:32.29 | sivana | was missing the directory representing arg1 |
| 22:33.05 | tzanger | you dumbass :-) |
| 22:33.17 | shmaltz | who is this (it showed up on my CC statement): |
| 22:33.18 | shmaltz | TOLL FREE COMMUNIC |
| 22:33.20 | shmaltz | is it asterlink? |
| 22:33.23 | sivana | hehe |
| 22:33.29 | twisted[work] | shmaltz, why don't you ask them? |
| 22:33.37 | *** part/#asterisk B2382F29 (~daniel@dsl-084-058-144-023.arcor-ip.net) |
| 22:33.44 | cluecon | shmaltz: that is most likely asterlink. |
| 22:34.09 | shmaltz | cluecon, thanks |
| 22:34.15 | twisted[work] | hah! |
| 22:34.18 | twisted[work] | good luck |
| 22:37.27 | Marlow | twisted[work] : wouldn't you need to be cable bound to get unplugged ? :o) |
| 22:37.50 | twisted[work] | Marlow, ? |
| 22:38.07 | twisted[work] | i'm not on wireless here if that's what you're thinking |
| 22:38.31 | Marlow | twisted[work] : eh .. thought so, since you said good look .. |
| 22:38.42 | twisted[work] | good luck ;) |
| 22:38.56 | twisted[work] | you'd have to climb through a maze of other cables and desks to unplug my shit |
| 22:39.08 | Marlow | ehehe ......... and get stuck |
| 22:39.17 | twisted[work] | or at least injured |
| 22:41.00 | tzanger | use ser with sphinx! everyone should be very good at using their sphinxser! |
| 22:41.34 | tzanger | oh COME ON |
| 22:41.36 | tzanger | that was FUNNY |
| 22:41.52 | Marlow | tzanger : not really |
| 22:41.58 | Nuxi | using sphinx, yup, that's funny. |
| 22:42.13 | tzanger | bah |
| 22:42.16 | sivana | heh |
| 22:42.16 | tzanger | everyone's a critic |
| 22:43.06 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:43.08 | *** part/#asterisk NSGN (~NSGN@cpe-66-69-197-25.austin.res.rr.com) |
| 22:43.54 | *** join/#asterisk LeeColleton (~lc@dsl092-030-021.sea2.dsl.speakeasy.net) |
| 22:44.46 | eKo1 | So the moral of the story is: don't try sphinxser jokes in #asterisk and expect them to be funny |
| 22:45.09 | twisted[work] | one more mention of sphinxser and I'll put a boot in your sphincter. |
| 22:46.35 | Nuxi | ()*() |
| 22:47.02 | Nuxi | () sphinxser() |
| 22:47.16 | Nuxi | gotta hurt |
| 22:50.32 | Uther_P | someone know if tar has a switch for only extracting new files, and not to overwrite existing files? (i'm looking in the man page but not seeing anything) |
| 22:50.41 | LeeColleton | Hello! I'm trying to get my iaxcomm softphone to connect to asterisk but it keeps saying "Registration rejected". I'm using ACTOS to configure the system but I don't know if I should be setting up peers or users or friends or what. Is there a good HOWTO for this setup? |
| 22:51.08 | eKo1 | yes |
| 22:51.12 | eKo1 | ~doc |
| 22:51.13 | jbot | extra, extra, read all about it, doc is The command is "~docs", moron! |
| 22:51.22 | eKo1 | ~docs |
| 22:51.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:52.17 | LeeColleton | thanks eKo1! jbot is saucy. |
| 22:52.36 | idnar | *snigger* |
| 22:54.11 | Uther_P | snigger? heh |
| 22:59.36 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 23:00.26 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 23:00.56 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 23:00.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:04.03 | Nethab | no not really |
| 23:04.23 | Nethab | anyone used the [authenticate] section in sip.conf |
| 23:04.44 | *** join/#asterisk anthm[tablet] (~anthm@h4608178d.area4.spcsdns.net) |
| 23:09.54 | *** join/#asterisk lyroy (~sebastien@modemcable117.123-202-24.mc.videotron.ca) |
| 23:11.58 | JunK-Y | http://www.chromance.de/wtf/lol.htm |
| 23:13.04 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:13.24 | harryvv | anyone here experaincing a backorder of ip500s from atacomm or other mfg? |
| 23:14.14 | *** join/#asterisk DaLion (~DaLion@69.156.67.70) |
| 23:14.28 | cluecon | harryvv: the 500s are being discontinued if i remember correctly. |
| 23:14.33 | DaLion | hey we all know 7960 works with asterisk.. does the 7914 extention module also work ? |
| 23:15.22 | harryvv | i know, thay are being replace by the 501 for a security upgrade. |
| 23:15.43 | harryvv | seems to be a very popular phone. |
| 23:16.12 | cluecon | yes, they are. |
| 23:16.34 | *** join/#asterisk tPO (~tPO@195.82.106.196) |
| 23:17.13 | *** join/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 23:18.09 | harryvv | and atacomm cannot hold onto them at all. once in the door thay are all shipped. |
| 23:18.09 | Marlow | DaLion : sccp only |
| 23:18.15 | opus__ | yo |
| 23:18.16 | DaLion | oh |
| 23:18.20 | Marlow | DaLion : the expansion module can't talk anything else .. |
| 23:18.23 | DaLion | needs cisco call manag ? |
| 23:18.29 | DaLion | dsrn |
| 23:18.37 | opus__ | junk-y -- yeah i saw that wtf |
| 23:18.39 | harryvv | But who else carries the ip500 and is located in the pacific region? |
| 23:18.42 | Marlow | DaLion : you can use sccp with asterisk .. |
| 23:18.58 | JunK-Y | opus: pretty fucked up huh? |
| 23:19.00 | opus__ | anyone have a realtime config tar ball I could try? |
| 23:19.02 | Marlow | DaLion : but i'm not sure, how much chan_skinny or chan_sccp supports that exp. module |
| 23:19.04 | shmaltz | anthm, you around? |
| 23:19.22 | opus__ | junk-y the site that i first saw that retracted it mysterously |
| 23:19.28 | opus__ | where did you read it? |
| 23:20.12 | JunK-Y | opus__: my roomate told me |
| 23:20.52 | Marlow | harryvv : why not just order from voipsupply ? |
| 23:22.33 | Marlow | harryvv : they are usually fast and unproblematic, no matter where you are .. |
| 23:22.51 | Marlow | harryvv : even easier for me to order from them instead of here inside Europe |
| 23:22.52 | opus__ | thats cool |
| 23:22.56 | opus__ | they must make a lot of money |
| 23:23.13 | opus__ | has anyone tried the auto-answer hack for the ip500? |
| 23:23.18 | opus__ | how do I set the ring type in asterisk? |
| 23:23.30 | opus__ | btw AEL looks cool |
| 23:27.46 | shmaltz | opus_, it's on the wiki |
| 23:27.50 | Marlow | tzanger : uhh .. that's long ago .. |
| 23:28.03 | tzanger | yup |
| 23:28.51 | tzanger | who you gonna call? GHOSTBUSTERS |
| 23:29.47 | agave-txlink | what is different between port 4539 and 5036 for iax? |
| 23:29.48 | JunK-Y | ~ael |
| 23:29.49 | jbot | somebody said ael was Asterisk Extension Language |
| 23:30.07 | tzanger | yeah that looks kind of strange (ael) |
| 23:30.24 | tzanger | I wonder why they went with their own custom language |
| 23:31.22 | DaLion | no tit s actualy ~~~~GOST ~~ BUSTERS ta nana na nan ~~~ |
| 23:32.11 | tzanger | hahahaha |
| 23:33.24 | opus__ | <PROTECTED> |
| 23:33.28 | SwK | anyone have a dell 600m laptop? |
| 23:33.29 | opus__ | does that mean realtime worked? |
| 23:34.08 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
| 23:35.16 | *** join/#asterisk Tili (~Tili@202-133-67-116-dialup.sat.net.pk) |
| 23:36.05 | Marlow | SwK : ehh .. you wouldn't want to own that :) |
| 23:36.49 | Marlow | SwK : dell d600 is listing --> http://www.chromance.de/wtf/lol.htm |
| 23:37.00 | Marlow | SwK : i mean 600m |
| 23:37.06 | SwK | yeah |
| 23:37.17 | SwK | thats exactly what I was ref'ing |
| 23:37.32 | SwK | was looking for independent confirmation of that |
| 23:37.48 | Marlow | SwK : but since the 600m is a model, that is U.S. specific, not sold outside the states .. |
| 23:37.58 | Marlow | SwK : most people on this side of the pond don't care :) |
| 23:38.18 | SwK | yeah |
| 23:38.35 | Marlow | SwK : and it might be the same for the 700m .. again .. U.S. only model |
| 23:38.39 | SwK | yea |
| 23:38.45 | SwK | who the hell knows tho |
| 23:39.03 | Marlow | SwK : bb is watching ya :) .. |
| 23:39.07 | SwK | would be interesting to get secondary conf on the keylogger then see what the /. effect does to dell hah |
| 23:39.12 | SwK | F@BB |
| 23:39.32 | SwK | and I have some words for echelon to... jihad, george bush bomb whitehouse |
| 23:40.28 | opus__ | the dark side |
| 23:41.42 | Marlow | SwK : the original site, that had the article is allready taken down .. |
| 23:41.57 | Nethab | port 5035 is the asterisk management port |
| 23:41.59 | Marlow | SwK : the one in germany is a copy .. i have another copy of that page .. |
| 23:42.03 | Nethab | i mean 5036 |
| 23:42.18 | Nethab | what's this security upgrade for the Polycom IP500 |
| 23:42.28 | Marlow | SwK : we don't want people to forget ... |
| 23:47.32 | pifiu | i have a question |
| 23:47.48 | pifiu | what is asterisk most demanding of in a system, as far as specs go? |
| 23:47.55 | pifiu | to hold maybe 5-10 calls at a time |
| 23:48.05 | pifiu | memory? cpu? cpu cache? |
| 23:51.07 | *** join/#asterisk malfi (~malte@dsl-084-059-037-191.arcor-ip.net) |
| 23:51.18 | Marlow | pifiu : depends, if you want to transcode or not . |
| 23:51.31 | Marlow | pifiu : if you want to use voicemail, conferencing etc. |
| 23:51.37 | Marlow | pifiu : many factors .. |
| 23:52.00 | *** join/#asterisk Robot_ (~robot_@pool-71-113-23-113.sttlwa.dsl-w.verizon.net) |
| 23:52.11 | Robot_ | hey there |
| 23:52.59 | Robot_ | can anybody tell me what does " WARNING[12895]: chan_sip.c:908 retrans_pkt: Maximum retries exceeded on call xxx@xxx for seqno 102 (Non-critical Request)" means ? |
| 23:55.08 | Marlow | Robot_ : what it says, it's a non-critical warning .. ignore it .. bad sip behavior either on asterisk or the other side :) |
| 23:55.34 | Robot_ | can i turn those warnings off ? |
| 23:56.08 | Marlow | Robot_ : sure .. don't log at your logs or the console :) |
| 23:56.40 | Marlow | *pling* ... warnings gone .. |
| 23:56.41 | Robot_ | if i lower the debug level ? |
| 23:58.19 | opus__ | hmmm, realtime not working .. how can I check if its loaded, connected, etc? |
| 23:58.49 | *** part/#asterisk tPO (~tPO@195.82.106.196) |
| 23:59.17 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:59.27 | Ariel_ | hello everyone |
| 23:59.36 | *** join/#asterisk cryptnix (~cryptnix@66.103.243.10) |
| 23:59.48 | cryptnix | hmm, guys ... per trunk can I have a different "greeting" ??? |
