| 00:01.03 | Yourname`` | If a sip peer is set to context [test], and then in [test] a call is sent to a different context called [sandiego] which sends the call to a queue, can that sip peer use every exten=> made in the [sandiego] context? or does the sip peer get to use extens only in the context it's set to? In this case [test |
| 00:02.00 | *** join/#asterisk propellerhead (n=yogurt2u@host35.190-30-186.telecom.net.ar) |
| 00:02.04 | cmantito | Yourname``: how is it sent? |
| 00:02.18 | jcaceres | cmantito? |
| 00:02.23 | jcaceres | the call? |
| 00:02.31 | Yourname`` | cmantito: Either a sip transfer from other box like 100@IP, or an incoming DID. |
| 00:02.39 | cmantito | jcaceres: yes. |
| 00:02.44 | propellerhead | keith4 I'd rebuild the modules |
| 00:02.54 | jcaceres | Dial(Zap/5/027848800wwwwwwwwwwwwwwwwwwwwwwww,200,R) |
| 00:02.56 | propellerhead | run fine |
| 00:03.09 | jcaceres | that the last way i tried |
| 00:03.13 | kamaji | hmmm |
| 00:03.13 | cmantito | Yourname``: sorry |
| 00:03.16 | cmantito | I need to clairfy my question |
| 00:03.28 | cmantito | once the call comes into the 'test' context, |
| 00:03.34 | cmantito | how is it transferred to the 'sandiego' context? |
| 00:03.40 | propellerhead | thanks keith4 |
| 00:03.42 | kamaji | I can't register with a SIP server: I wireshark'ed it, and my regular SIP client is sending a SUBSCRIBE as well as REGISTER, whereas asterisk is only sending REGISTER |
| 00:04.01 | kamaji | and asterisk just times out |
| 00:04.07 | cmantito | SUBSCRIBE is for subscribing to voicemail notifications, etc. Usually, anyway. |
| 00:04.21 | kamaji | :\ |
| 00:04.25 | Yourname`` | cmantito: Using goto |
| 00:04.37 | kamaji | Why would asterisk not get a reply, then? |
| 00:05.18 | cmantito | Yourname``: then they wouldn't be able to use extens in the sandiego context, unless something allowed them to (ie, sip.conf) |
| 00:05.31 | cmantito | kamaji: not sure, could depend on a lot of factors, wanna pastebin your conf? |
| 00:05.31 | cmantito | ~pb |
| 00:05.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:05.46 | Yourname`` | Ahhh..but they can use it in [test] context, right cmantito? |
| 00:05.59 | cmantito | da |
| 00:06.08 | kamaji | ehh... will require lots of editing |
| 00:06.08 | kamaji | hang on |
| 00:06.13 | Yourname`` | da = ya? lol |
| 00:06.30 | cmantito | yes, sorry |
| 00:06.31 | cmantito | =p |
| 00:06.35 | Yourname`` | haha |
| 00:06.47 | Yourname`` | Ok, cmantito.. this is what I'm essentially looking to do. |
| 00:07.16 | Yourname`` | I have two separate sites who would use queues for their own reasons. I have assigned 2XX to site2, and 1xx to site1. |
| 00:07.39 | jcaceres | hello i have some issue with zaptel, a am intending ti send calls to 12 celulinks using 3 tdm400, but it's not woking properly, i have captured the sounds sent with ztmonitor, i have even putted a telephone in paralel to the line in order to hear the dtmf sound |
| 00:07.59 | jcaceres | i think, my celulink needs more time to start the call |
| 00:08.00 | jameswf-home | ${you} != "clue"?heh:liar |
| 00:08.54 | Yourname`` | On site 2, all their eyeBeams are configured from 200 - 299. But then there are different groups of agents within those that will be reassigned to a different queue and stuff. And these guys use AgentLogin() .. I just want it to be easy for me to segregate these groups of agents on site2 to another context like [feedback] or [sales], etc. And I can't figure out a way to do so. |
| 00:09.40 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 00:09.46 | Yourname`` | That is what I'm looking to do, and it seems a little hard for me, cmantito. |
| 00:10.08 | cmantito | well queues are not my area of smartitude. dialplans/contexts I can help with, but I'm afraid the only queues I can do are the ones to get food from a McDonalds, or to get on to a highway ;p |
| 00:10.21 | Yourname`` | lol |
| 00:11.18 | Yourname`` | Someone should really write a lot about queues. I feel that queues are the one big thing that is least documented about. :( |
| 00:11.25 | cmantito | yeah |
| 00:11.35 | cmantito | I'm going to be learning them shortly, but I figure one part of ast at a time. |
| 00:15.47 | kamaji | cmantito: http://rafb.net/p/AlZ9tC61.html |
| 00:16.10 | eric2 | I have long distance rates loaded into mysql from provider A, what's the best way to grab the dialed country code + city code from the dialed number after 011? |
| 00:16.38 | eric2 | as country codes and city codes vary in length.... |
| 00:17.00 | Katty | hello |
| 00:17.35 | cmantito | kamaji: you may want to try dropping the /user on the register => line |
| 00:17.46 | cmantito | the other thing you may want to try is changing [orbtalk] to [<user>] |
| 00:17.58 | _ShrikE | Katty! |
| 00:18.16 | Katty | hugs _ShrikE |
| 00:18.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:18.28 | kamaji | cmantito: what's the /user for anyway? |
| 00:18.29 | Katty | _ShrikE: how was your sunday? |
| 00:18.47 | _ShrikE | Katty: could have been better, my doggy is sick :( |
| 00:19.03 | Katty | _ShrikE: aww. |
| 00:19.07 | Katty | _ShrikE: nothing serious i hope |
| 00:19.16 | cmantito | that's directing it at a specific context |
| 00:19.38 | _ShrikE | Katty: we just took her in (a stray) and it appears that she has some serious liver issues. |
| 00:20.05 | kamaji | oh right, so that should be /orbtalkin? |
| 00:20.19 | _ShrikE | eric2: You need to take the maximum country+city code length, and drop digits until you find a match. |
| 00:20.33 | cmantito | kamaji: it's not really necessary because you've got the context specified elsewhere |
| 00:20.51 | Katty | _ShrikE: on no )= |
| 00:20.58 | Katty | _ShrikE: the vet got her on meds? |
| 00:21.05 | kamaji | cmantito: ok |
| 00:21.06 | _ShrikE | Katty: Oh yeah. |
| 00:21.12 | Katty | _ShrikE: good. |
| 00:21.46 | Katty | _ShrikE: i got groceries today. i'm exhausted. |
| 00:21.56 | eric2 | _ShrikE would you suggest doing this in the dial plan (extensions.conf) or do it outside? I'm trying to put together the billing part of the system... |
| 00:22.45 | _ShrikE | eric2: there is no reason you cant do it in the dialplan. |
| 00:22.53 | eric2 | ok |
| 00:22.55 | eric2 | tx |
| 00:23.50 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 00:27.01 | jcaceres | hello i have some issue with zaptel, a am intending ti send calls to 12 celulinks using 3 tdm400, but it's not woking properly, i have captured the sounds sent with ztmonitor, i have even putted a telephone in paralel to the line in order to hear the dtmf sound |
| 00:27.06 | jcaceres | i think, my celulink needs more time to start the call |
| 00:27.09 | jcaceres | any idea? |
| 00:27.30 | Katty | analog lines? |
| 00:27.38 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:28.33 | *** join/#asterisk propellerhead (n=yogurt2u@host201.190-138-95.telecom.net.ar) |
| 00:31.36 | ManxPower | For analog Zaptel interfaces a "w" in the dial string will pause for .5 seconds. Example: Dial(Zap/1/ww5551515) would go off hook and delay dialing for 1 second. |
| 00:31.53 | Katty | yes, what ManxPower said |
| 00:31.57 | Katty | works like a charm (= |
| 00:34.10 | jameswf-home | ~badjoke |
| 00:34.10 | jbot | how do you fix a womans watch? .....no we cant answer |
| 00:34.19 | cmantito | o.O |
| 00:37.34 | jcaceres | hello i am capturing the dtmf tone to initiate a call an they sound noisy, aby idea in how can i solve this? |
| 00:37.53 | Katty | jameswf-home: i'm car sick :< |
| 00:38.02 | Katty | jameswf-home: you ever get car sick? while driving. |
| 00:41.39 | *** join/#asterisk ta^3 (n=tacvbo@189.136.26.14) |
| 00:45.39 | jcaceres | ManxPower: i was using this |
| 00:45.43 | jcaceres | <PROTECTED> |
| 00:45.44 | jcaceres | <PROTECTED> |
| 00:45.44 | jcaceres | <PROTECTED> |
| 00:45.54 | jcaceres | and its the same |
| 00:46.22 | jcaceres | i do not know why it sais "answered" if noting happend |
| 00:47.02 | jcaceres | i have tuned each chanel with fxotune -i |
| 00:47.11 | Yourname`` | What if I don't set context= in queues.conf? |
| 00:47.48 | jcaceres | and i made a program in matlab to see if the card are sending the correct tones |
| 00:48.19 | Yourname`` | Nevermind. I think the context in queues is for a different purpose than a context in extensions.conf |
| 00:48.50 | jcaceres | i can see that some channels are noisy but it does not matter they have the same behavior |
| 00:48.54 | jcaceres | any idea plz |
| 00:53.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-830f2540d3003473) |
| 00:57.08 | jcaceres | how can i tell zaptel o zapata, i am not sure, to wait for the ringing tone? |
| 01:02.37 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:02.42 | jcaceres | how can i tell zaptel o zapata, i am not sure, to wait for the ringing tone? |
| 01:06.43 | *** join/#asterisk alrs (i=foobar@216.151.159.21) |
| 01:09.19 | Yourname`` | Do member => Agent/21 in queues.conf need to be defined as a peer in sip.conf like [21]? |
| 01:14.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:14.32 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:15.20 | *** part/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 01:18.48 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:19.19 | jameswf-home | ~repeat |
| 01:19.19 | jbot | it has been said that repeat is probably If nobody answers your question, don't just repeat it. Spamming the channel and getting ignored/banned/silenced isn't going to get a faster answer; spamming is a function of time. More likely, nobody knows the answer (/msg apt ask), or you need to provide more information (/msg apt sicco)), or ask me about "rephrase", or ask smart questions |
| 01:19.44 | Yourname`` | lol |
| 01:20.25 | Yourname`` | I dont know if you know jameswf-home, but I think #asterisk hands out "ops" only to people who work for digium or are some form of celebrated committers. :P |
| 01:20.56 | jameswf-home | wtf is the point of a 12.5sec pause after a dial |
| 01:21.06 | jameswf-home | is to immature to be an op :) |
| 01:21.34 | jcaceres | jameswf-home, is that for me? |
| 01:21.42 | Yourname`` | You don't have to be mature. I knew an 8 yr old channel op on EFnet who later became an oper. |
| 01:22.30 | jameswf-home | talks in generalities if his comment fits in to someones situation and works he takes full credit... if it doesn't work then he meant it for someone else |
| 01:23.45 | Yourname`` | LOL |
| 01:24.15 | jameswf-home | I need to add more random wisdom blips to by site.... |
| 01:35.46 | jcaceres | asterisk answers me but the call is not ringing through the fxo? |
| 01:35.55 | jcaceres | any idea plz |
| 01:38.11 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 01:39.22 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 01:40.31 | boblutz | jcareres, wait! |
| 01:41.54 | *** join/#asterisk WindBack (n=jorge@host82.190-31-201.telecom.net.ar) |
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| 01:52.05 | *** join/#asterisk RoyK (n=roy@ip-22-54-149-91.dialup.ice.no) |
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| 01:55.58 | Katty | mew? |
| 01:56.38 | jameswf-home | ~moo |
| 01:56.39 | jbot | ACTION mooooooooo! I am cow, hear me moo, I weigh twice as much as you. I am cow, eating grass, methane gas comes out my ass |
| 01:56.39 | *** join/#asterisk dcmwai (n=dcmwai@60.54.46.150) |
| 01:57.10 | CCFL_Man2 | lol |
| 01:57.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:58.00 | jameswf-home | I should quit being cheap and pay the $3 a month for hosting... |
| 02:03.01 | CCFL_Man2 | does the T100P support 5v or 3.3v pci? |
| 02:03.08 | CCFL_Man2 | docs don't really say7 |
| 02:03.30 | Katty | tries putting asterisk book on head and absorbing |
| 02:11.12 | jameswf-home | I use mine as a pillow |
| 02:12.34 | CCFL_Man2 | the cisco guys put down asterisk |
| 02:13.06 | jameswf-home | the whole world puts down Cisco I would say that its about even |
| 02:13.28 | CCFL_Man2 | yeah |
| 02:13.48 | CCFL_Man2 | why was the T100P discontinued other than the 5v pci interface? |
| 02:16.00 | *** join/#asterisk Speedy2 (n=John@cpe-66-91-247-165.san.res.rr.com) |
| 02:16.34 | Speedy2 | Hey all. This isn't strictly an Asterisk question, but is there any SIP software that can direct pc-to-pc communication (without a SIP proxy/server, etc). SpeakFreely isn't cutting it for me anymore. |
| 02:17.07 | Speedy2 | I've tried a few like wxCommunicator and Efiga with little luck |
| 02:17.52 | jameswf-home | you could write one |
| 02:19.58 | *** join/#asterisk hohum (n=dcorbe@70.0.200.23) |
| 02:20.42 | *** join/#asterisk IguanaNed (n=me@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 02:20.51 | IguanaNed | hello |
| 02:20.59 | IguanaNed | ? |
| 02:21.14 | IguanaNed | need help with .call files in * 1.4 |
| 02:21.19 | jameswf-home | is hello a question? |
| 02:21.24 | djs26 | throws IguanaNed a big juicy fly |
| 02:21.28 | IguanaNed | could be |
| 02:22.20 | IguanaNed | to generate a cal in the future, if I set the modify datetime using touch |
| 02:22.35 | IguanaNed | should the call be made in the future? |
| 02:22.40 | IguanaNed | doesnt seem to work |
| 02:24.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:24.56 | jameswf-home | call files are processed once they hit the spooler... you could use at command to copy the file at a later time |
| 02:25.14 | jameswf-home | q |
| 02:25.22 | jameswf-home | man at |
| 02:26.00 | IguanaNed | interesting as I see doc that state asterisk will ignore any files with future modification datetime |
| 02:28.07 | jameswf-home | maybe A2 - This most likely is because your TMP directory is on a different physical disk in the system. Make a tmp directory just above the OUTGOING directory in asterisk and use that , so when the MV command is used the date and time of the file won't be changed |
| 02:29.54 | *** join/#asterisk efort (n=efort@74-86-100-202.lx-vs.net) |
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| 02:34.24 | coolblade | Is there a way with AGI to pass a date to saydatetime instead of having it read a current date, if not, any suggestions for how I can say a date to the listener? |
| 02:34.31 | *** part/#asterisk Speedy2 (n=John@cpe-66-91-247-165.san.res.rr.com) |
| 02:35.49 | jameswf-home | coolblade: GooGle: http://www.voip-info.org/wiki/view/say+datetime |
| 02:35.57 | coolblade | i did... |
| 02:36.43 | jameswf-home | google better :) the article above explains |
| 02:37.06 | coolblade | i read that |
| 02:37.32 | coolblade | i will reread |
| 02:44.51 | IguanaNed | solution |
| 02:45.02 | CCFL_Man2 | how well does the sccp channel work with a 7920 wifi phone? |
| 02:45.04 | IguanaNed | using cp |
| 02:45.36 | IguanaNed | using cp instead of mv tead of mv command to puyt call file in outgoing dir |
| 02:45.50 | *** join/#asterisk jcaceres (n=asd@182-98-112.adsl.terra.cl) |
| 02:47.23 | jcaceres | hello i am having some troubles with a server with 3 cards tdm400 with 4 fxo modules each one, |
| 02:47.44 | jcaceres | all the channels are in the same group |
| 02:48.12 | jcaceres | when i send a call the channel dial |
| 02:49.20 | jcaceres | dials, an then answers, but the call is not starting in the other side |
| 02:49.51 | jameswf-home | perhaps your not plugged in |
| 02:50.26 | jcaceres | i am connecting the fxo modules to some celular adapters, an they work fine when i connect them to a analog phone |
| 02:51.42 | jcaceres | [Mar 31 11:49:44] VERBOSE[3046] logger.c: -- Called g0/027848800 |
| 02:51.42 | jcaceres | [Mar 31 11:49:54] DEBUG[3046] chan_zap.c: Engaged echo training on channel 1 |
| 02:51.42 | jcaceres | [Mar 31 11:49:58] DEBUG[3046] chan_zap.c: Echo cancellation already on |
| 02:51.42 | jcaceres | [Mar 31 11:49:58] VERBOSE[3046] logger.c: -- Zap/1-1 answered SIP/200.6.115.35-08d67d58 |
| 02:51.57 | jameswf-home | jcaceres: use pastebin |
| 02:52.01 | jameswf-home | ~pb |
| 02:52.01 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:53.10 | jameswf-home | if your connecting to a gsm adapter you may need to set your signalling to revpol |
| 02:54.08 | jcaceres | http://pastebin.com/d5dede20c |
| 02:54.21 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 02:54.34 | jameswf-home | your card needs to support this of course |
| 02:55.15 | jcaceres | hanguponpolarityswitch=no |
| 02:55.16 | jcaceres | answeronpolarityswitch=no |
| 02:55.28 | jcaceres | i have those parameters set |
| 02:55.31 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 02:55.45 | jcaceres | i am not sure i those are correct |
| 02:56.03 | jcaceres | but before i putted them i had the same result |
| 02:57.07 | jcaceres | escuseme, jameswf-home: how can i know if my cards support revpol |
| 02:57.20 | jameswf-home | its more involved than that you need to get tech support for your cards and gsm equipment on the phone in a conference |
| 02:59.19 | jcaceres | in your experience does a tdm400p has support por revpol? |
| 02:59.34 | jameswf-home | ?me doesnt use digium stuff :) |
| 03:01.34 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
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| 03:10.52 | Yourname`` | Can AMI operate not on a network socket by on a unix socket? |
| 03:18.02 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.02 | jcaceres | jameswf-home: there somthing i couldn't understand from what you said, is my card needes to support to detect revpol? or my card needs to give revpol? |
| 03:26.36 | drmessano | It needs to detect it |
| 03:27.42 | jcaceres | ahh ok thanks, yes it does |
| 03:29.11 | jcaceres | but, it stills answering and the call is not being done by the celular adapter? |
| 03:31.29 | jcaceres | can i modify another parameter to define when the answer is done? |
| 03:38.20 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 03:40.06 | keith4 | with AGI, I can use stream_file to play a sound, and accept DTMF during it. what's the equivalent in extensions.conf? |
| 03:42.16 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 03:48.04 | keith4 | ah, background |
| 03:48.05 | keith4 | of course |
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| 03:56.59 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 03:57.11 | drmessano | EGADS! |
| 03:59.00 | Yourname`` | How can I have a context-neutral sip peer that will assume the context that's set for the agent the peer logs in as? For example peer41 is set to context=default, but peer41 logs in as agent 30 for testq and i want agent30 to use the context thats supposed to be used for [testq] instead of the context set for peer41.. how can I do so? |
| 03:59.38 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 04:04.51 | *** join/#asterisk LakeSolon (n=blake@12-202-198-20.client.mchsi.com) |
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| 04:05.38 | jblack | sighs and drops his head. |
| 04:06.38 | jblack | Building a network with all the modern servers and associated toys is like ice cream. Documenting it in case of EHITBYTRUCK is like spinach |
| 04:06.41 | BeeBuu | anyone know how to convert a word doc file to tif file for fax in asterisk. |
| 04:06.42 | BeeBuu | ? |
| 04:06.59 | jblack | perhaps ghost script can do it. |
| 04:07.24 | jblack | or perhaps even a2ps. |
| 04:07.54 | BeeBuu | any more suggestion? |
| 04:08.04 | drmessano | Did you try those? |
| 04:08.32 | jblack | Yeah. Fix the cups subsystem so that efax works. |
| 04:08.39 | BeeBuu | drmessano: how? |
| 04:09.23 | drmessano | jblack gave you two things to try |
| 04:09.30 | drmessano | Maybe you should run off and try them |
| 04:09.39 | drmessano | I know thats not what you like to do.. you like to ask 100 times |
| 04:09.41 | jblack | beebuu: I'll be more clear. Faxing anything but tifs right now is difficult, error prone, and practically takes a PHD in system administration. |
| 04:09.46 | drmessano | But trying will show you if it works |
| 04:10.00 | BeeBuu | .... |
| 04:10.09 | jblack | BeeBuu: Unless you feel very comfortable with your skillset, I'd drop it into the "it can't be done" basket for now. |
| 04:10.16 | BeeBuu | i just want to make sure that can work. |
| 04:10.28 | drmessano | BeeBuu: So go try those two suggestions |
| 04:11.13 | BeeBuu | search for ghost guide |
| 04:11.28 | jblack | ghostscript, not ghostguide. :) |
| 04:11.36 | drmessano | jblack: good luck |
| 04:11.38 | BeeBuu | GS |
| 04:11.44 | BeeBuu | right? |
| 04:11.52 | jblack | sure. |
| 04:12.11 | jblack | But take my advice. Give up on this one now |
| 04:12.32 | drmessano | http://xkcd.com/178/ |
| 04:12.47 | BeeBuu | another question:fax on asterisk can fax PDF file? |
| 04:12.54 | BeeBuu | or not? |
| 04:13.00 | drmessano | Probably |
| 04:13.20 | drmessano | Asterisk isn't the one faxing, you need to check the fax app itself |
| 04:13.25 | drmessano | This is not the place |
| 04:13.32 | jblack | beebuu: It's almost exactly the same problem as faxing word documents. |
| 04:13.37 | BeeBuu | ...ok,i got you. |
| 04:13.49 | *** join/#asterisk classyhuman (n=classyhu@auh-b1453.alshamil.net.ae) |
| 04:13.56 | BeeBuu | thanks.jbalck & drmessano |
| 04:14.01 | jblack | Actually.. faxing anything other than a tif is almost exactly the same problem as faxing word documents. |
| 04:14.31 | drmessano | jbalck is very halpful |
| 04:14.38 | classyhuman | Hello, Jblack |
| 04:14.43 | drmessano | he haz cheezeburger |
| 04:14.44 | classyhuman | Hello all |
| 04:15.18 | BeeBuu | beer? |
| 04:15.30 | drmessano | Hello WellManicuredHomosapien |
| 04:15.37 | jblack | Dear pppd: Catch up with the times. LDAP has been around for a decade, and you still can't authenticate against it. |
| 04:15.55 | classyhuman | hello drmessano |
| 04:16.07 | *** join/#asterisk mandd (n=dache@dsl-131-46.aei.ca) |
| 04:16.21 | drmessano | Welcome to planet #asterisk |
| 04:16.47 | mandd | whoa, lots of users |
| 04:16.51 | mandd | thats awesome |
| 04:16.57 | classyhuman | Hi mandd |
| 04:17.05 | mandd | hello classyhuman |
| 04:17.27 | classyhuman | Im newbie here.. seeking help in DTMF issue |
| 04:17.57 | mandd | ah, i am new as well |
| 04:18.12 | classyhuman | ok |
| 04:18.42 | classyhuman | Its an experts channel, I guess someone will have a solution |
| 04:20.30 | drmessano | Maybe |
| 04:21.53 | classyhuman | The SIP providers sends different mode of DTMF Sometimes RFC2833, Inband, info. I changed dtmfmode=auto But it cant detect the DTMF |
| 04:22.34 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-166-99-223.bltmmd.east.verizon.net) |
| 04:22.50 | drmessano | Who is the provider? |
| 04:23.13 | jblack | I'd yell at the inbounds to turn on rfc2833 |
| 04:23.36 | classyhuman | teleglobe, singtel etc. Its on failover priority and each one use different mode |
| 04:23.58 | drmessano | and none of them work? |
| 04:24.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:25.05 | classyhuman | some of them work when it match my mode. If I use rfc2833 and when they have similar it works, but when it falls to failover provider they might have info or inband |
| 04:25.33 | drmessano | So why don't you have it configged per peer? |
| 04:26.00 | drmessano | That only makes sense if you have one or a few that are not rfc2833 |
| 04:26.31 | classyhuman | They dont stick to one mode, it changes several times. I guess they also have failover providers who sends them different modes. |
| 04:26.43 | drmessano | Uh |
| 04:27.26 | drmessano | sounds to me like you have something else wrong then |
| 04:27.42 | drmessano | I haven't heard of providers randomly changing dtmf modes all over the place |
| 04:27.56 | jblack | I dunno. I saw that myself the other day with IPKall. |
| 04:28.02 | drmessano | If I had one that did that, I would kill them |
| 04:28.24 | jblack | They use rfc2833 with their 206 areacode, and inbound on 360. |
| 04:28.48 | classyhuman | It happens. My providers change their modes frequently. |
| 04:28.55 | drmessano | There's probably different proxies, jblack |
| 04:29.30 | classyhuman | dtmfmode = auto is not helping |
| 04:29.32 | jblack | Probably. |
| 04:29.48 | jblack | However, proxies being proxies, they all come to me from the same ip. |
| 04:30.05 | drmessano | I'll let someone else tell you that providers don't change up DTMF mode all over the place |
| 04:31.12 | classyhuman | I contacted providers, they said they buy service from different parties and each one of they may use different equipments and DTMF |
| 04:31.47 | jblack | time for new providers, perhaps? |
| 04:32.48 | classyhuman | is there a possible way to detect what mode they r sending and change it accordingly in asterisk |
| 04:32.59 | drmessano | I guess it's just a matter of asterisk doing a shitty job detecting dtmf.. I would submit a bug report |
| 04:33.47 | classyhuman | thats great |
| 04:34.31 | drmessano | Surely it can't be a problem with your providers.. it never is |
| 04:36.50 | classyhuman | That is what I have found finally when DTMF showed up issues |
| 04:38.35 | classyhuman | dtmfmode=auto is that all i have to set? |
| 04:43.14 | mandd | if I do not have a service provider connected/installed yet, but I want to configure my dialplan as if It was receiving incoming calls, how can I link my SIP Lan phone to [incoming] ? |
| 04:44.48 | mandd | exten => 225,1,Goto(incoming) works, but then I get Auto fallthrough |
| 04:48.32 | *** join/#asterisk s0lid (n=s0lid@210.213.198.56) |
| 04:49.46 | [TK]D-Fender | mandd, pastebin your dialplan and the CLI output of your failure at verbose 10 |
| 04:49.47 | [TK]D-Fender | ~pb |
| 04:49.48 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:49.49 | [TK]D-Fender | ^^^^^^^^^^ |
| 04:52.36 | Yourname`` | A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context. -> This is what it says in queues.conf. Does that mean that this queue can have a context, or does it mean that it should have a context only if the context supports dropping out of a queue? |
| 04:53.30 | [TK]D-Fender | Yourname``, it means if you point it to a queue and it matches a digit pressed by the queued call, it will ext to there |
| 04:53.48 | mandd | [TK]D-Fender http://pastebin.com/m70890917 |
| 04:54.17 | mandd | another problem, is that I can call some extensions, like 301, but 303 It fails |
| 04:54.49 | mandd | I am attempting to follow examples from Asterisk - Future of Telephony book. |
| 04:55.34 | [TK]D-Fender | mandd, Yes, 303 fails because you mispelled SIP. exten => 303,1,Dial(STP/EPHERE) |
| 04:56.03 | mandd | wow |
| 04:56.25 | mandd | This is a bit embarrassing , thank you [TK]D-Fender |
| 04:56.48 | [TK]D-Fender | mandd, And for you other problem, you need to set "autofallthrough=no" under [general] |
| 04:56.49 | Yourname`` | [TK]D-Fender: Yeah.. but my confusion is this. A sip peer 41 has context=test, a call comes in on [incoming] and its transferred by Goto to [sandiego] which inturn puts the call in a queue(), and when 41 (who is logged in to a queue(manila) as 41) tries to transfer the call to an extension within [sandiego], asterisk says "No such extension in context "test"" -> obviously because even though... |
| 04:56.50 | Yourname`` | ...41 is logged into a queue called manila it still is in the same context as 41 = test. So I'm wondering if I set the context to manila in queues.conf itself, no matter what device/agent logs into manila queue.. it should do everything in the [manila] queues. |
| 04:57.35 | [TK]D-Fender | Yourname``, WTF? |
| 04:57.46 | Yourname`` | lol i know! |
| 04:58.07 | Yourname`` | Ok, how about in queues.conf what does context= under a queue do? |
| 04:58.09 | [TK]D-Fender | Yourname``, First you're talking about an exit context, now you're transferring calls areound? What are talking about ehre? |
| 04:58.45 | Yourname`` | [TK]D-Fender: It's the biggest clusterfuck, I tell you. But I can't test right now, so I'm not gonna go any further than this but finding out what the purpose of context= is in queues.conf. |
| 04:58.57 | [TK]D-Fender | Yourname``, the queue exit context has NOTHIGN to do with agents or your phones, or any of that. |
| 04:59.01 | Yourname`` | Is it the same as context= in sip.conf? |
| 04:59.54 | [TK]D-Fender | Yourname``, when you set a context for a queue, your caller can dial a 1-digit exten while waiting to QUIT and do somethign PRODUCTIVE rather that sit around waiting for you to ANSWER him. |
| 05:00.04 | mandd | still get Auto fallthrough, channel 'SIP/mandd-081eba58' status is 'UNKNOWN' with autofallthrough=no in [general], |
| 05:00.13 | mandd | after realod in cli. |
| 05:00.19 | mandd | reload* |
| 05:00.30 | Yourname`` | [TK]D-Fender: And you sure that's ALL it needs the context for? |
| 05:00.37 | [TK]D-Fender | Yourname``, So you'd play an announcement like "Please hold and the next available agent will take you call some time before Hell freezes over, or jsut press 1 to leave a ^@%#ing voicemail (if you know whats good for you)" |
| 05:00.48 | Yourname`` | [TK]D-Fender: Does it not server the same purpose as the context= in the sip.conf AND the 1 digit thing? |
| 05:00.50 | [TK]D-Fender | Yourname``, Yes, Please read the BIG print. |
| 05:01.19 | Yourname`` | Holy mac! |
| 05:01.32 | Yourname`` | I only wish you were the best in queues as you are in everything else so I could bombard you with those questions tomorrow :P |
| 05:01.54 | [TK]D-Fender | Yourname``, I've already answered your queue question. |
| 05:01.57 | Yourname`` | Because as always, as soon as I go into details, you go "I'm not well proficient with queues, sorry." |
| 05:02.07 | Yourname`` | This is far more complicated good sir! |
| 05:02.17 | Yourname`` | I'll have to show you and test it out.. |
| 05:02.23 | [TK]D-Fender | Yourname``, You are mixing shit up that doesn't deserve to be in the same sentence. |
| 05:02.28 | Yourname`` | I meant the best in queues part for what's to come :P |
| 05:03.02 | [TK]D-Fender | Yourname``, the "context=blah" you'd put in queues.conf is so your calling can GTFO of the queue when he feels like you are ignoring him. Clear? |
| 05:03.34 | Yourname`` | Clear. But it's a little more than that, which I'm going to poke you with tomorrow! |
| 05:03.41 | [TK]D-Fender | Yourname``, No, its not. |
| 05:03.48 | Yourname`` | No no, I mean I got the answer you gave. |
| 05:04.05 | Yourname`` | And since that's the answer I was kinda thinking of too, it leads me to look for another alternative. |
| 05:04.10 | Yourname`` | Which is what is coming tomorrow. :D |
| 05:05.24 | [TK]D-Fender | Yourname``, You said 41 has context = test. that is his context. He will not transfer a call to [sandiego] |
| 05:05.41 | Yourname`` | Meanwhile, GotoIfTime needs a context to send the call to, is there a simple app which does not need a context to send to? Like if time is this and that, dial this, if not dial that? |
| 05:06.18 | [TK]D-Fender | Yourname``, You don't have to use a context with GotoIfTime. |
| 05:06.39 | [TK]D-Fender | (specify) |
| 05:06.47 | Yourname`` | WHAT?! |
| 05:06.49 | Yourname`` | GotoIfTime(10:30-23:59|mon-sat|||?ck-forwarding,100,1) |
| 05:06.50 | [TK]D-Fender | Yourname``, Or an exten if you don't want to. |
| 05:07.00 | Yourname`` | I'd rather do a retrydial in there. |
| 05:07.05 | Yourname`` | So you're saying I can? |
| 05:07.22 | [TK]D-Fender | Yourname``, it does a Goto, whats so hard to understand? |
| 05:07.34 | Yourname`` | GotoIfTime(10:30-23:59|mon-sat|||?RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/419xxxxxxx@provider1,,tT,) |
| 05:07.39 | [TK]D-Fender | Yourname``, I jsut said you didn't have to specify a CONTEXT or EXTEN if you don't want to. |
| 05:07.56 | [TK]D-Fender | Yourname``, No, I did NOT say you can do whatever the hell you please, its still a damn goto. |
| 05:08.00 | Yourname`` | Can I do that -> GotoIfTime(10:30-23:59|mon-sat|||?RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/419xxxxxxx@provider1,,tT,) |
| 05:08.02 | Yourname`` | :P |
| 05:08.10 | Yourname`` | I know, I know, I got you. I'm just buggin ya. :D |
| 05:08.25 | Yourname`` | Ok, so how can I use RetryDial on an iftime if I wanted to? |
| 05:10.04 | mandd | [TK]D-Fender "autofallthrough=no" under [general] in sip.conf? |
| 05:10.19 | [TK]D-Fender | mandd, Yes |
| 05:10.35 | mandd | still get Auto fallthrough |
| 05:10.36 | [TK]D-Fender | mandd, Sorry, no, extensions.conf |
| 05:10.40 | [TK]D-Fender | oops |
| 05:10.41 | mandd | aha. |
| 05:10.46 | mandd | will try |
| 05:10.48 | mandd | thank you |
| 05:11.01 | [TK]D-Fender | mandd, Thats all dialplan stuff |
| 05:11.12 | [TK]D-Fender | Yourname``, Gotoiftime. |
| 05:11.44 | Yourname`` | Yeah, but I don't want it to do a goto. I want it to run an application depending on the time.. |
| 05:11.45 | [TK]D-Fender | Yourname``, Or ExecIf + IfTime |
| 05:11.54 | Yourname`` | Now that sounds complicated. |
| 05:11.57 | [TK]D-Fender | Yourname``, Same friggen thing |
| 05:12.13 | [TK]D-Fender | Yourname``, You clearly should not be coding your dialplan. Go hire someone. |
| 05:12.51 | mandd | it's working! thanks again [TK]D-Fender |
| 05:13.00 | mandd | been stuck on that one for a while. |
| 05:13.00 | [TK]D-Fender | mandd, You're welcome |
| 05:13.24 | Yourname`` | No. GotoIfTime would mean I'd have to have a context doing things. All I wanna do is a dial if the time is this, and dial that if the time is that. |
| 05:13.24 | Yourname`` | LOL I'm bored.. :( |
| 05:14.10 | [TK]D-Fender | Yourname``, "have a context do things? |
| 05:14.24 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:14.33 | [TK]D-Fender | Yourname``, "Put down the crack pipe" (c) JerJer |
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| 05:19.43 | jblack | hi |
| 05:20.04 | talntid | sup |
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| 05:29.59 | jameswf-home | heh http://linuxgangster.org/ |
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| 05:45.37 | justdave | so I'm having a strange problem where the asterisk server in an office is showing all of the phones in that office as unreachable |
| 05:46.11 | justdave | they're polycom phones... I can browse the web interface of the phones from the shell on the asterisk server with elinks, so physically it can see them on the network |
| 05:46.32 | justdave | the phones think they're logged in, and asterisk thinks they're logged in but unreachable |
| 05:47.09 | justdave | users can place calls, and the calls go through... according to asterisk, except the response packet never makes it back to the phone, so the phone thinks it's still trying until it times out |
| 05:47.11 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 05:47.39 | justdave | if I tell the server to send a notify packet to the phone to tell it to reboot, the phone does reboot... but the server keeps trying to tell it to because it never got the response from the phone saying it got the request |
| 05:48.38 | justdave | iptables is off on the server |
| 05:48.46 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 05:48.50 | justdave | and all of the phones are in the same lan segment with the server |
| 05:49.11 | justdave | it kinda feels like chan_sip is broken, but I've got no idea how |
| 05:50.32 | justdave | yeah, as usual, I just need to tell someone and the solution comes before they answer. :P |
| 05:50.44 | justdave | nat config had the wrong netmask on the local network address space |
| 05:50.56 | justdave | so it was sending them the external IP to respond to instead of the internal one |
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| 06:13.17 | jameswf-home | yay node crash |
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| 06:39.25 | stony | hi |
| 06:40.22 | stony | i'm looking for an eclipse plugin to write the dialplan (syntax highlighting only would be ok) - but the only thing i found is the post in the developer mailinglist from 2005 where someone is working on a plugin |
| 06:40.37 | stony | does anyone know what the state of the plugin is and are there other plugins ? |
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| 06:47.02 | Yosam | hello |
| 06:47.08 | *** join/#asterisk mandd (n=dache@dsl-131-46.aei.ca) |
| 06:47.45 | mandd | getting -- Unregistered SIP 'EPHERE', from all phones |
| 06:48.10 | mandd | any command in CLI I can use? or i have to reset all phone manually? |
| 06:50.24 | stony | mandd: sip reload registry |
| 06:51.04 | mandd | == Parsing '/etc/asterisk/sip.conf': Found |
| 06:51.19 | mandd | and still same thing, in terms of Unregistered SIP |
| 06:51.25 | Yosam | is there open source speech2text and text2speech applications? |
| 06:51.31 | mandd | scrolling really fast too |
| 06:52.52 | mandd | stony anytihng else I can try? |
| 06:55.36 | classyhuman | What is the standard DTMF frequency used in asterisk, can it be changed to tackle DTMF detection issues |
| 06:56.21 | justdave | stony: current version of vim seems to have syntax highlighting for asterisk dialplans |
| 06:56.35 | justdave | don't know anything about eclipse stuff though |
| 06:57.37 | stony | mandd: hmm i'm not sure where this error comes from, but it looks like the pbx can't register an outgoing voip trunk |
| 06:57.47 | stony | justdave: jep, that's what i'm using atm |
| 06:58.16 | mandd | oh |
| 06:59.05 | mandd | it just happened a few times now, cant figure out what's causign it |
| 06:59.13 | mandd | all phone go insane |
| 06:59.17 | mandd | phones* |
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| 07:35.05 | classyhuman | Hi |
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| 08:19.25 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) [NETSPLIT VICTIM] |
| 08:24.15 | *** join/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 08:29.01 | *** join/#asterisk kiko69 (n=keith@udp218844uds.hawaiiantel.net) |
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| 08:35.15 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 08:40.21 | sysadmin-lb22 | hi all I know I can make outbond calls to jingle google talk using astersik..however I do have a jingle client "other than google talk"..can I register it on the asterisk server..and make PSTN calls through the Asterisk server ? |
| 08:46.31 | *** join/#asterisk charleszsz (n=gces47@67.159.178.21) |
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| 09:17.14 | *** join/#asterisk Al_WinKiller (n=winkille@mgm.cornet.am) |
| 09:17.33 | Al_WinKiller | hi guys, can somebody help me with translation rules ? |
| 09:18.35 | Al_WinKiller | ppl ? anybody alive ? |
| 09:19.43 | *** join/#asterisk fatcop (n=223343@ppp121-44-111-12.lns10.syd6.internode.on.net) |
| 09:19.49 | fatcop | hey |
| 09:20.01 | *** join/#asterisk oej (n=olle@bkkb-gw.voop.net) |
| 09:20.19 | Al_WinKiller | hey, can you help me with translation rules ? |
| 09:20.41 | fatcop | here's an original question .... anyone know where I can get a windows installer for latest 1.4 build ?? |
| 09:21.06 | fatcop | or anything in 1.4 :) |
| 09:21.44 | *** join/#asterisk matrix1233 (n=Administ@196.203.192.150) |
| 09:22.03 | *** join/#asterisk atis_work (n=atis_wor@81.198.164.2) |
| 09:22.40 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 09:24.40 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 09:25.18 | matrix1233 | hello |
| 09:25.25 | classyhuman | Hi |
| 09:25.29 | mvanbaak | fatcop: I dont think asterisk has a windows installer |
| 09:25.37 | fatcop | binary ? |
| 09:25.41 | mvanbaak | no |
| 09:25.47 | matrix1233 | am new here :D |
| 09:25.48 | FlatFoot | morning all |
| 09:26.04 | classyhuman | Hello.. any expert in here? |
| 09:26.08 | mvanbaak | Al_WinKiller: what translation rules |
| 09:26.23 | FlatFoot | anyone in who had summit to do with the writing / creation of IAX ? |
| 09:26.29 | fatcop | well i installed something very old then i guess ... AsteriskWin32-0.66b-Setup.exe |
| 09:26.44 | fatcop | lol .. man that must be old as |
| 09:27.24 | *** join/#asterisk dob1 (n=rob122@78.13.166.99) |
| 09:27.38 | matrix1233 | what is the best pbx whre i can found an plug and play to detect all type of card |
| 09:27.39 | matrix1233 | :d |
| 09:27.39 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ff4ae2f72f3d7039) |
| 09:28.19 | mvanbaak | matrix1233: ??? |
| 09:28.24 | JT | trick question? |
| 09:28.43 | matrix1233 | sorry bad in english :):) |
| 09:28.43 | dob1 | hi, just to understand, if i install asterisk what can i do ? i can call the normal phone from pc ? |
| 09:28.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:30.47 | classyhuman | mvanbaak, Hi.. What is the default frequency of DTMF used in asterisk? |
| 09:31.47 | JT | dtmf is a standard |
| 09:31.51 | JT | you can look it uo |
| 09:31.53 | JT | up |
| 09:32.36 | fatcop | so just to clarify .. asterisk is pretty much a linux thing ... not supported on windows (tho may build) .. is that the deal ? |
| 09:33.12 | mvanbaak | fatcop: pretty much yes |
| 09:33.21 | fatcop | and there is a "Windows GUI for Asterisk PBX - GlassConsole Lite" so you can manage it from a window box |
| 09:33.25 | mvanbaak | I think you can compile it under cygwin |
| 09:33.25 | classyhuman | JT, is there a way to detect different mode of DTMF received from Provider automatically |
| 09:34.35 | *** join/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au) |
| 09:34.36 | JT | classyhuman: are you talking about frequencies of voip dtmf transmission methods? |
| 09:35.10 | fatcop | mvanbakk: but the fact no one seems to make a binary package avail seems to indicate .. its not something stable ? |
| 09:35.38 | mvanbaak | fatcop: that, or noone is intersted in doing it |
| 09:35.43 | *** join/#asterisk cfh (n=luca@87.241.50.50) |
| 09:36.26 | cfh | hi all, i have some problem with features.conf and asterisk 1.4.15 |
| 09:36.32 | cfh | it doesnt works |
| 09:36.57 | matrix1233 | it's exist a pbx like switchvox |
| 09:37.01 | matrix1233 | but free ? |
| 09:37.04 | cfh | with asterisk 1.2.x it works |
| 09:37.04 | matrix1233 | opensource |
| 09:37.26 | cfh | with 1.4 version are there something different ? |
| 09:37.54 | *** part/#asterisk Hyphenex (n=Hyphenex@60-241-72-151.tpgi.com.au) |
| 09:44.44 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 09:49.18 | tzafrir | matrix1233, "like switchvox", as in: will limit the number of channels you can use? |
| 09:49.23 | tzafrir | Please clarify |
| 09:49.53 | tzafrir | I'm not aware of one with that feature |
| 09:53.25 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 09:55.03 | matrix1233 | tzafrir, wanna a pbx that don't have a limit of channel |
| 09:55.24 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 09:55.39 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 09:56.36 | matrix1233 | tzafrir,i have tested switchvox, it's good beacause is a plg and ply and detect all card, it's exist anadher one like it... now i have downloaded pbxinAflash is good ? |
| 09:57.50 | JT | matrix1233: you're really looking at the wrong channel |
| 09:59.01 | matrix1233 | JT: perhaps, am just a begginer in asterisk so ... :D |
| 09:59.27 | *** join/#asterisk Dextorion (n=dex@fingerbottom.tekproj.bth.se) |
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| 10:04.53 | classyhuman | hi JT |
| 10:05.07 | classyhuman | any clue on getting right DTMF mode |
| 10:06.53 | *** join/#asterisk gego (n=ubuntuus@b238085.customer.hansenet.de) |
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| 10:18.03 | *** part/#asterisk classyhuman (n=classyhu@auh-b1453.alshamil.net.ae) |
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| 10:25.28 | gego | Hello, I've got the problem to identify who picked up a channel (for group_count) or who actually answered "got" several ringing lines (dial(sip/1&sip2&...) |
| 10:33.43 | *** part/#asterisk airjump (n=zielonka@62.159.95.82) |
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| 10:49.02 | zeeesh | calling by using xlite.. most of time it works fine ... but sometime i got echo problem from xlite end ?how to troubleshoot it ? |
| 10:56.02 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
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| 11:20.55 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 11:20.58 | *** part/#asterisk RoyK (n=roy@box36.fortel.no) |
| 11:24.15 | *** part/#asterisk masus (n=ethemc@88.248.14.186) |
| 11:24.49 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:34.28 | sysadmin-lb22 | hi all I know I can make outbond calls to jingle google talk using astersik..however I do have a jingle client "other than google talk"..can I register it on the asterisk server..and make PSTN calls through the Asterisk server ? |
| 11:36.10 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:38.14 | *** join/#asterisk flynux (i=xuz4daf@cl-8.bru-01.be.sixxs.net) |
| 11:38.51 | Dextorion | asterisk -r gives: unable to connect( does /var/run/asterisk.ctl exist?). Anyone know what to do? |
| 11:39.35 | Mavvie | Dextorion: start asterisk to start with. |
| 11:45.46 | Dextorion | its running |
| 11:46.02 | mvanbaak | check permissions on the asterisk.ctl file |
| 11:46.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:46.47 | Dextorion | ahm.. also i probably should say that im running asterisknow atm. |
| 11:48.19 | Dextorion | running asterisk with the -c flag would start it in some kinda of command line mode, right? |
| 11:48.31 | Dextorion | could that be why i cant run asterisk -r in antother tty? |
| 11:48.42 | mvanbaak | no |
| 11:48.51 | Dextorion | no? hrm. oki mvanbaak |
| 11:49.25 | Dextorion | permissions are srwx for root. and asterisk is running as root |
| 11:49.25 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:55.20 | *** part/#asterisk cfh (n=luca@87.241.50.50) |
| 11:55.56 | yang | What kind of error would be this - auto-congesting http://openpaste.org/en/5877/ |
| 11:57.38 | *** join/#asterisk mpwizard (i=cjs@trinity-32.xnk.nu) |
| 11:59.35 | mpwizard | I'm currently setting up Comedian Voicemail. Is it possible to play a special message when a user checks his voicemail for the first time? E.g. Welcome blaahaha Press 1 to enter your pin code. |
| 12:00.32 | *** part/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:00.34 | *** join/#asterisk MmixX (i=senti@202.58.249.25) |
| 12:00.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 12:01.10 | tzafrir | mpwizard, how exactly do you know that this is the first time? |
| 12:01.35 | tzafrir | One way: save it in the DB |
| 12:02.15 | drmessano | classyhuman? |
| 12:02.18 | tzafrir | In the dialplan - check , and potentially set, this in the dialplan |
| 12:02.38 | drmessano | nope, gone |
| 12:04.38 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-106-67.bstnma.east.verizon.net) |
| 12:06.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:08.20 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 12:10.17 | *** part/#asterisk airjump (n=zielonka@62.159.95.82) |
| 12:14.42 | mpwizard | tzafrir: Hmmm... That should work. |
| 12:15.06 | tzafrir | mpwizard, also take a look at minivm |
| 12:15.16 | tzafrir | and compose your own vm menu |
| 12:15.58 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 12:16.08 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 12:18.12 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 12:19.47 | defswork | needs a new job |
| 12:25.55 | *** join/#asterisk matrix1233 (n=Administ@196.203.192.150) |
| 12:26.37 | *** join/#asterisk nighty^ (n=nighty@p1022-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 12:29.40 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 12:38.14 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:41.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:41.17 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 12:42.42 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 12:43.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:46.18 | *** join/#asterisk ManxPower (n=manxpowe@119.sub-75-201-31.myvzw.com) |
| 12:48.36 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 12:54.42 | *** join/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 12:55.08 | twitchnln | morning |
| 12:58.09 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:58.47 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:59.49 | twitchnln | how do i do a tech prefix on an outbound call in extensions.conf? |
| 12:59.49 | twitchnln | exten => _1800NXXXXXX,1,Dial(SIP/ProviderTrunk/1620${EXTEN}) |
| 13:00.10 | [TK]D-Fender | twitchnln: Looks fine |
| 13:00.13 | twitchnln | or do i need a pause in there before the ${EXTEN} |
| 13:00.44 | [TK]D-Fender | twitchnln: No need for any kind of pause. Its SIP. All digital. Only time you need to "wait" on anything is on some crappier analog situations |
| 13:01.03 | twitchnln | [TK]: cool, thanks |
| 13:02.44 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 13:04.46 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-143-27.red.bezeqint.net) |
| 13:05.08 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 13:05.10 | Dovid | hi is video supporte in 1.4.x only or also in 1.2.x ? |
| 13:07.27 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:07.31 | Katty | hai |
| 13:08.32 | JayTee52 | Dovid, I've used video with Eyebeam softphones on 1.2 |
| 13:09.10 | Dovid | JayTee52: Thanks. goto mess around with the paramaters now ;0 |
| 13:09.27 | JayTee52 | you have to add video support for h.323 in SIP.CONF |
| 13:12.39 | [TK]D-Fender | JayTee52: Because yeah... we always put H.323 settings ins sip.conf.... |
| 13:12.49 | Katty | hi fender. |
| 13:13.12 | [TK]D-Fender | Katty: Mew. |
| 13:13.13 | JayTee52 | in the [General] section of sip.conf add the statement: videosupport=yes and allow=h263 |
| 13:13.20 | Katty | [TK]D-Fender: how're mew? |
| 13:13.32 | JayTee52 | Dovid, sorry that was h263 not 323 |
| 13:13.37 | [TK]D-Fender | Katty: Meow-K |
| 13:14.08 | JayTee52 | [TK]D-Fender, thanks for catching that. Not fully awake yet. |
| 13:16.10 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:16.10 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:17.32 | Dovid | JayTee52: how does it work ? besides for videosupport=yes I need nmjust saw ur post |
| 13:17.44 | Dovid | allow=h263 is for sip |
| 13:17.46 | Dovid | ? |
| 13:17.49 | Katty | lmadsen: GET OUT |
| 13:17.59 | Katty | lmadsen: sorry. i've had too much caffeine this morning. |
| 13:18.10 | Katty | lmadsen: smidgen hyper. |
| 13:18.32 | lmadsen | Katty: lol... you're way too much like file |
| 13:18.44 | Katty | lmadsen: i knows. that's why we get along so great. |
| 13:18.51 | lmadsen | it kinda freaks me out |
| 13:18.59 | Katty | lmadsen: i steal his muffin. he steals my orange juice. we pout. |
| 13:19.01 | Katty | lmadsen: etc. |
| 13:19.11 | lmadsen | weirdos |
| 13:19.12 | Katty | lmadsen: how are mew? |
| 13:19.24 | lmadsen | Katty: I am mewing fine thank you |
| 13:19.36 | Katty | horays! |
| 13:19.39 | lmadsen | mewrself? |
| 13:20.05 | JayTee52 | Dovid, it's been awhile but as I recall we had to use h263 with SIP softphones for the video. We used Eyebeam which is the Pro version of X-Lite softphones. |
| 13:20.18 | Katty | lmadsen: hyper! |
| 13:20.20 | Katty | boingboign |
| 13:20.29 | lmadsen | lol, I need breakfast and caffeine |
| 13:20.48 | *** join/#asterisk wolvienews (n=lkusmir@h-B202-46.resetnet.pl) |
| 13:21.25 | drmessano | I am out sick.. I need a vacuum and some vitamin C |
| 13:21.52 | drmessano | <lmadsen> mewrself? <-- please don't encourage her |
| 13:22.01 | Katty | drmessano: SHUUSH! |
| 13:22.05 | Katty | drmessano: go back to bed, sicko. |
| 13:22.11 | lmadsen | I always encourage Katty |
| 13:22.48 | drmessano | She doesn't need it.. she does fine just by mewrself |
| 13:22.48 | lmadsen | katty katty bo batty, banana fanny fo.... latty... fee fi fo matty... oooooooh katty! |
| 13:22.50 | drmessano | ZOMG NO |
| 13:22.55 | Katty | haha |
| 13:22.58 | Katty | cute. |
| 13:23.02 | Katty | now, GET TO WORK |
| 13:23.09 | Katty | i mean, breakfast. scoot! |
| 13:23.09 | lmadsen | work is for suckers |
| 13:23.12 | ManxPower | .part #asterisk-drinkers |
| 13:23.12 | ManxPower | oops |
| 13:24.01 | Katty | hi manx (= |
| 13:24.03 | Katty | hugs ManxPower |
| 13:25.03 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 13:29.23 | Dovid | JayTee52: didnt work |
| 13:29.23 | Dovid | Mar 31 09:28:28 NOTICE[15344]: rtp.c:579 ast_rtp_read: Unknown RTP codec 126 received |
| 13:29.25 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:29.30 | Dovid | is all I got |
| 13:29.57 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 13:30.26 | grandpapadot | Hi all. I have a bunch of polycom 501's at one of our locations. If I swt DHCP Option 66, will it set the download server's IP address even know the protocol is FTP instead of TFTP? |
| 13:30.54 | [TK]D-Fender | Dovid: http://www.voip-info.org/wiki/view/Asterisk+video |
| 13:30.56 | boblutz | How can one allow non-root users to load zaptel drivers? (2.6.x kernel) |
| 13:31.23 | [TK]D-Fender | grandpapadot: Opt66 will pass the server to contact, the phones base setting chooses the protocol. |
| 13:31.33 | grandpapadot | Ok, great! Thanks! |
| 13:31.42 | Dovid | TK: Thanks |
| 13:31.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:32.20 | tzafrir | boblutz, why do you want to do that? |
| 13:32.35 | tzafrir | allowing non-root users to load modules is not such a grand idea |
| 13:33.08 | boblutz | tzafrir: When I run Asterisk as non-root, the line wont pick up |
| 13:33.24 | tzafrir | the zaptel init script needs to run as root |
| 13:33.25 | boblutz | I think it is because the user "asterisk" doesnt have permission to write to /dev/zap |
| 13:33.58 | grandpapadot | chown -R asterisk:asterisk /dev/zap |
| 13:33.59 | tzafrir | and the /dev/zap/* files need to be owned by a user / group that asterisk is a member of |
| 13:34.12 | tzafrir | that should normally b edone in udev rules |
| 13:34.25 | grandpapadot | also your asterisk user needs to be a member of audio and dialout |
| 13:34.26 | boblutz | hmm.. I saw that in chapter 3 of ~thebook |
| 13:34.36 | tzafrir | in Debian and Gentoo - add asterisk to the group dialout |
| 13:34.54 | boblutz | `usermod -aG dialout asterisk` ? |
| 13:35.04 | tzafrir | (and let the distro's udev rules do the rest) |
| 13:35.14 | tzafrir | adduser asterisk dialout |
| 13:35.27 | tzafrir | that very strange syntax works in Debian |
| 13:37.05 | boblutz | ok sweet |
| 13:37.16 | boblutz | Step 1, `service zaptel start` |
| 13:37.30 | boblutz | Step 2, `chown -R asterisk:asterisk /dev/zap/*` |
| 13:37.37 | boblutz | Step 3, `asterisk -U asterisk` |
| 13:37.43 | boblutz | typical? ^^^ |
| 13:38.23 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:39.03 | Katty | hugs fskrotzki |
| 13:39.13 | madduck | what does this mean? |
| 13:39.17 | madduck | handle_request_invite: Sending fake auth rejection for user "martin f. krafft" ...? |
| 13:40.28 | *** join/#asterisk akafurious (n=akafurio@gw1.pickeringcollege.on.ca) |
| 13:44.31 | *** join/#asterisk hugohagogo (n=cleber@189.23.20.10) |
| 13:46.08 | tzafrir | boblutz, what distro do you use? |
| 13:46.25 | tzafrir | Step 2 should not be required . Use proper udev rules |
| 13:46.31 | Katty | has new puppeh wallpaper! |
| 13:46.42 | boblutz | RHEL 4 |
| 13:46.58 | boblutz | tzafrir: I made the change to the udev rules as mentioned in chapter 3 of ~thebook |
| 13:47.14 | boblutz | However, that was last night and I couldnt get it to work, so I said whatever and commented it |
| 13:47.28 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 13:47.51 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:48.14 | boblutz | Interesting, I just noticed there is a zaptel.rules in /etc/udev/rules.d/ |
| 13:48.27 | Shotygun | Hello. I am using asterisk 1.2 and I have a scenario where I need to support about 1000 sip users. I was wondering if it's possible to create like a context with wildcard name like it's possible for extensions, so I won't have to create 1000 contexts |
| 13:48.48 | tzafrir | boblutz, grep zap /etc/udev/rules.d/* |
| 13:48.56 | tzafrir | the order there is meaningful |
| 13:49.49 | Shotygun | The common thing for the 1000 users is a single LAN subnet |
| 13:50.38 | boblutz | tzafrir: http://pastebin.ca/964699 <--- zaptel.rules looks right |
| 13:50.54 | [TK]D-Fender | Shotygun: No. |
| 13:51.09 | tzafrir | boblutz, the grep was intended to check if there aren't other udev rules that mention zaptel |
| 13:51.19 | tzafrir | boblutz, again, what distri do you use? |
| 13:51.23 | boblutz | RHEL 4 |
| 13:52.39 | tzafrir | hmm.. it might be too old to have zaptel rules. Not sure |
| 13:52.42 | CCFL_Man2 | i just won an auction for a used T100P, did i overpay at $102? |
| 13:53.12 | tzafrir | What do you want to use it for? |
| 13:53.31 | *** join/#asterisk s0lid (n=s0lid@210.213.198.56) |
| 13:53.46 | CCFL_Man2 | to originate a CAS T1 |
| 13:53.53 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 13:54.05 | [TK]D-Fender | CCFL_Man2: Probably fine |
| 13:54.41 | [TK]D-Fender | CCFL_Man2: You'll have SWEC to contend with and greater risk of PCI flakeyness, but might do ok |
| 13:55.04 | CCFL_Man2 | [TK]D-Fender: why was it discontinued? |
| 13:55.26 | [TK]D-Fender | CCFL_Man2: for the fact its only T1 capable (no E1/J1), older PCI reference design, etc. |
| 13:55.36 | CCFL_Man2 | ahh |
| 13:56.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.122.138) |
| 13:56.04 | [TK]D-Fender | CCFL_Man2: The TE110P the TE120P replaced that line in order |
| 13:56.10 | CCFL_Man2 | are there any problms with the T1 part of it? |
| 13:56.19 | CCFL_Man2 | ahh, yeah |
| 13:56.20 | [TK]D-Fender | CCFL_Man2: You should be fine. |
| 13:56.26 | CCFL_Man2 | ahh |
| 13:56.54 | coppice | I think it should do J1, but you probably don't care :-) |
| 13:57.18 | CCFL_Man2 | i'm going to put it in my sun netra T1 200 |
| 13:57.41 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 13:58.16 | CCFL_Man2 | [TK]D-Fender: it has an "*" right on the card, you think it's real? |
| 13:58.48 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 13:59.13 | CCFL_Man2 | not a counterfiet |
| 13:59.21 | [TK]D-Fender | CCFL_Man2: Couldn't say |
| 13:59.53 | CCFL_Man2 | there any real way to tell? |
| 14:00.11 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 14:00.27 | *** join/#asterisk keith4 (n=kbe2@lust.CC.Lehigh.EDU) |
| 14:00.33 | mort_gib | Afternoon |
| 14:00.50 | mort_gib | I need a Snom phone to report busy when in a call |
| 14:00.54 | ManxPower | I think I have a bug in my script 8-) "ktheriot@example.com has not logged in since 1969/12/31 (38 years, 3 months ago)" |
| 14:01.40 | CCFL_Man2 | it looks different from the T100P in the digium datasheet, but who knows |
| 14:01.41 | mort_gib | I have tried to use the incominglimit in sip.conf and although it forces the phone to accept only one call |
| 14:01.59 | CCFL_Man2 | ManxPower: ahh, the good old days |
| 14:02.00 | mort_gib | It also makes the handset report unavail rather than busy |
| 14:02.08 | [TK]D-Fender | mort_gib: "core show application chanisavail" |
| 14:02.30 | ManxPower | CCFL_Man2: I hate programming |
| 14:02.33 | rupa | I have a Linksys 3201. My incoming pstn line does not have callerid. It (the linksys) seems to wait up to 3 rings before it sends a ring message to asterisk. Any thoughts? |
| 14:02.44 | CCFL_Man2 | ManxPower: so do i :P |
| 14:02.51 | boblutz | C programming is fun but mad difficult for a rookie such as myself |
| 14:03.03 | ManxPower | rupa: My thought is that you have to look at the Linksys docs, this has nothig to do with Asterisk |
| 14:03.13 | CCFL_Man2 | most of my vintage phone parts come today |
| 14:03.15 | rupa | true |
| 14:03.16 | mort_gib | Cool |
| 14:03.35 | CCFL_Man2 | i can almost finish restoring this candle stick |
| 14:03.57 | [TK]D-Fender | rupa: And lok for any kind of answer delay in your setup of it, or references to apssing CID on to * |
| 14:04.26 | rupa | [TK]D-Fender, ok, going through the config screen |
| 14:07.37 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:08.58 | *** part/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 14:11.36 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 14:11.52 | asteriskmonkey | is there anyway to set a sip phone to have a european ring when dialing out? |
| 14:14.20 | mort_gib | [TK]D-Fender: How do you make it return busy rather than unavail?? |
| 14:15.19 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 14:15.29 | Lsodi | ~pastebin |
| 14:15.30 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:15.44 | mort_gib | I have a blind girl I need to cater for, she will have to redirect calls to the voicemail when she leases, and take it off in the morning. |
| 14:16.14 | mort_gib | I wanted to use the dnd button, but I need to be able to see if the extension is busy or unavailable |
| 14:16.51 | [TK]D-Fender | mort_gib: you use chanisavail to see if the phone is in-sue and you don't have to limit the phone itself. The phone doesn't return ANYTHING. |
| 14:17.11 | [TK]D-Fender | asteriskmonkey: Tell the phone what indication to use |
| 14:17.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:17.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:18.14 | mort_gib | [TK]D-Fender: Still I need to see the difference between in use and dnd |
| 14:18.17 | [TK]D-Fender | mort_gib: Oh, DND.... NO way to detect that unless the phone allows you to change its response somehow |
| 14:18.24 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 14:18.56 | mort_gib | I don't need to, I need to let the phone redirect to say 7100, which would be the voicemail for reception, on dnd |
| 14:19.25 | mort_gib | I can use dialstatus to see if the phone is on dnd |
| 14:19.39 | [TK]D-Fender | mort_gib: I don't see how... |
| 14:20.06 | mort_gib | If I can detect busy, I can fallover to the rest of the reception staff (4) |
| 14:20.26 | mort_gib | But allow DND to forward ALL incoming calls to Voicemail |
| 14:21.30 | mort_gib | I know this is not perfect, but the point is to make it easy for a disabled person. |
| 14:21.50 | ManxPower | mort_gib: you say you need the phone to do this, but you are asking about Asterisk. |
| 14:21.56 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 14:22.07 | ManxPower | Asterisk can't control how the phone handles specific situations. |
| 14:22.35 | *** join/#asterisk frawd (n=francois@89.130.32.92) |
| 14:23.13 | anonymouz666 | damn, sometimes rtptimeout just doesn't work. |
| 14:23.25 | mort_gib | Well ideally I do it all on Asterisk... I CAN accept to have to do SOME of it on the Phone though. |
| 14:23.52 | ManxPower | mort_gib: How exactly do you plan on doing DND in Asterisk? |
| 14:24.12 | ManxPower | You, of course, won't be able to use the DND button on the phone, if you do DND in Asterisk. |
| 14:24.18 | *** part/#asterisk frawd (n=francois@89.130.32.92) |
| 14:24.29 | *** join/#asterisk frawd (n=francois@89.130.32.92) |
| 14:24.51 | ThatKidKel | Is this possible? I have a AGI script that returns to me between 1 and x number of carriers as variables CARR0, CARR1, CARR2, CARRx.. In my dialplan I have a While that begins with set(i=1) and ends with set(i=$[${i} + 1]) .. i'd like to reference my CARR variables based on the value of ${i}.. I try Set(CARRIER=${LCR_CARR}${i}) but it sets ${CARRIER} to the value of ${i}.. and not the true value of ${CARR1}... |
| 14:24.52 | ManxPower | mort_gib: The answer to most of your questions is "Look in the Polycom Admin Guide" |
| 14:25.28 | mort_gib | Well, I don't using ${DIALSTATUS} is fine, but it does not diffrenciate between busy and unavail |
| 14:25.40 | ManxPower | ThatKidKel: The only reason would be if ${LCR_CARR} is empty |
| 14:25.42 | mort_gib | -And I don't use Polycom |
| 14:25.56 | ManxPower | mort_gib: Well then CHECK THE DOCS FOR YOUR PHONE. |
| 14:26.26 | mort_gib | But I need to see the difference in Asterisk, not on the phone |
| 14:26.44 | ManxPower | mort_gib: There IS NOT DIFFERENCE IN ASTERISK BECAUSE THE PHONE DOES NOT DO ANYTHING DIFFERENT. |
| 14:27.00 | ManxPower | If the phone did something different you would see it in HANGUPCAUSE or DIALSTATUS |
| 14:27.10 | ManxPower | I give up. |
| 14:27.24 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:27.26 | tzanger | mort_gib: give yourself a test... |
| 14:27.35 | tzanger | exten => _X.,1,Dial(SIP/foo,,g) |
| 14:27.38 | mort_gib | Which is exactly my problem, DIALSTATUS returns unavail regardless |
| 14:28.00 | tzanger | exten _X.,n,NoOp(Dialstatus is "${DIALSTATUS}", Hangup Cause is "${HANGUPCAUSE}") |
| 14:28.18 | tzanger | then play with it until you figure out why the phone is not responding correctly. |
| 14:28.22 | tzanger | sip debug would help you as well |
| 14:28.37 | ManxPower | tzanger: He wants DIALSTATUS/HANGUPCAUSE to be different depending on if the phone is set for DND or not. |
| 14:29.00 | tzanger | ManxPower: well, as you said, the phone needs to give different results. sip debug will prove either you or him wrong. |
| 14:29.00 | ManxPower | tzanger: AND he's using incoming-limit |
| 14:29.08 | tzanger | you can't both be right, and my money's on you |
| 14:29.20 | ManxPower | tzanger: Yeah, but he doesn't want to listen to me. |
| 14:29.29 | tzanger | sounds like an all-aorund shitty situation |
| 14:30.02 | ManxPower | tzanger: I think it's more like the psych patient that thinks that if he flaps his arms fast enough he will fly. |
| 14:30.27 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 14:30.33 | mort_gib | Yes, and thank you for such a compelling comparison... |
| 14:30.56 | tzanger | mort_gib: turn on sip debugging for the phone and see if there is a difference between dnd and not |
| 14:31.04 | tzanger | mort_gib: that's the only way to see for sure |
| 14:31.13 | mort_gib | Will do, just strange... |
| 14:31.14 | ManxPower | And when the doctor tells him he can't fly without a machine the patient ignores the answer because he doesn't like the answer |
| 14:31.19 | tzanger | if there *IS* a difference, there is a very high probability that Asterisk can be made to respond differently |
| 14:32.08 | mort_gib | I'm quite confident that there is some way to work it out... If not I just have to do the divert in another way... |
| 14:32.41 | ThatKidKel | Hey ManxPower.. Take a look at http://www.pastebin.ca/964732 |
| 14:32.48 | mort_gib | The one button solution would just be very comfortable... |
| 14:33.48 | mort_gib | And thank you for your time ManxPower, I remember now clearly what having paid support is like: -Sob, sob it those guys over there not me |
| 14:33.49 | drmessano | I show BUSY for both Asterisk-set DND and phone-set DND |
| 14:34.02 | drmessano | took me all of 10 seconds to test |
| 14:34.20 | mort_gib | Well, I'm not in my clients office now... |
| 14:34.39 | tzanger | mort_gib: for sure, the one-button is best, but you need to see what exactly the phone is returning |
| 14:34.49 | drmessano | Well, thats all the more reason to argue an hour on IRC about something "Im not on site to test it" |
| 14:36.04 | mort_gib | Anyway, thanks tzanger, and yes of course I need to know what the phone responds, but I just thought that Asterisk would be able to tell id the extension was in use |
| 14:36.19 | file | a normally configured Polycom returns 486 "Busy Here" |
| 14:36.21 | tzanger | mort_gib: it can do that |
| 14:36.23 | file | if DND is on. |
| 14:36.34 | tzanger | mort_gib: asterisk can do that, I have used it as well |
| 14:36.49 | tzanger | mort_gib: but your particular phone I have not used |
| 14:37.04 | tzanger | asterisk 'hint' extensions can tell the phone state, I've used that as well |
| 14:37.14 | tzanger | your phone may be infomring asterisk in a way that it does not know about yet |
| 14:37.44 | mort_gib | I can't restrict the phone to one line, this works though |
| 14:38.01 | mort_gib | If I do so, they can do the attended transfer they so like |
| 14:38.22 | mort_gib | I'll go to my clients now... |
| 14:38.50 | ManxPower | ThatKidKel: What is that pastebin supposed to help me with? |
| 14:39.09 | ThatKidKel | you said that it was probably nothing in the variable.. and there is.. |
| 14:39.26 | ThatKidKel | my question is, how when i'm in that loop can i increment the number at the end of the variable i want to reference.. |
| 14:39.37 | ThatKidKel | i want to start with LCR_CARR0, then LCR_CARR1, then LCR_CARR2 |
| 14:40.02 | ThatKidKel | apparently ${LCR_CARR}${i} wont' do it.. |
| 14:40.34 | ManxPower | ThatKidKel: no, that is a complex AGI that I would charge to support. What we need is the minimum code required to reproduce the issue. |
| 14:41.04 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-166-99-223.bltmmd.east.verizon.net) |
| 14:41.27 | *** join/#asterisk af_ (n=getsmart@88-149-241-15.dynamic.ngi.it) |
| 14:41.33 | ThatKidKel | the AGI is fine.. I just need Asterisk's dialplan to cycle through the variables based on the current value of ${i} |
| 14:42.24 | *** join/#asterisk af_ (n=getsmart@88-149-241-15.dynamic.ngi.it) |
| 14:42.49 | Lsodi | Hi, for a week I have tryed to figure out why cisco cme cant send dtmf to asterisk through sip trunk, debug shows that it even wont try to use rtp-nte, instead it tryes sip-notify... log and conf can be found here: http://pastebin.ca/964723 any idea or have someone encountered such kind of behivor from cisco cme? |
| 14:43.18 | ManxPower | ThatKidKel: I shall write you an example |
| 14:43.35 | ThatKidKel | If ${i} = 2 , then i want: # |
| 14:43.35 | ThatKidKel | exten => _NXXNXXXXXX,n,Set(CARRIER=${LCR_CARR}${i}) to actually be # |
| 14:43.35 | ThatKidKel | exten => _NXXNXXXXXX,n,Set(CARRIER=${LCR_CARR2}) |
| 14:47.09 | ManxPower | ThatKidKel: Patience, grasshopper. I shall write you an example |
| 14:47.30 | ThatKidKel | :) |
| 14:47.54 | [TK]D-Fender | ThatKidKel: "core show function EVAL" |
| 14:48.50 | Katty | i need a title for my business card. help. |
| 14:48.58 | *** join/#asterisk af_ (n=getsmart@88-149-241-15.dynamic.ngi.it) |
| 14:49.08 | ThatKidKel | a title for a business card?! |
| 14:49.25 | rupa | chief grasshoppers |
| 14:50.02 | drmessano | Katty: How about "Veterinarian" |
| 14:50.07 | Katty | hehe |
| 14:50.09 | Katty | i can't give shots. |
| 14:50.15 | Katty | i was thinking something with /server/ |
| 14:50.59 | coppice | you mean, like "waitress" |
| 14:51.21 | Katty | :< |
| 14:51.31 | drmessano | CAN HAZ CHEEZSERVER? |
| 14:52.50 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:54.12 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.130.247) |
| 14:55.32 | *** join/#asterisk comprookie2000 (n=comprook@mobile-166-214-115-156.mycingular.net) |
| 14:55.54 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:56.01 | ThatKidKel | i foudn it |
| 14:56.19 | ThatKidKel | Set(CARRIER=${EVAL(${LCR_CARR${i}})}) |
| 14:56.42 | ManxPower | ThatKidKel: http://www.pastebin.ca/964757 |
| 14:56.44 | ThatKidKel | thanks [TK]D-Fender.. i needed the eval and putting the ${i} on the inside |
| 14:57.24 | ThatKidKel | thanks Manx |
| 14:57.50 | ManxPower | You're welcome. I write dialplans for people for free all the time. |
| 14:58.58 | ThatKidKel | maybe you shoudl make a business out of it |
| 14:59.16 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:59.44 | ManxPower | ThatKidKel: nobody is willing to pay. |
| 14:59.46 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 14:59.46 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:00.01 | ManxPower | I've gotten about $200 send to my paypal account for this tuff over the past 4 years |
| 15:00.08 | ManxPower | tuff = stuff |
| 15:00.44 | ThatKidKel | hrmm.. maybe you wanna review your marketing.. |
| 15:00.48 | ThatKidKel | j/k |
| 15:00.57 | ManxPower | ~manxpower |
| 15:00.57 | jbot | rumour has it, manxpower is Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. |
| 15:01.20 | drmessano | ..or you could do the opposite and whore yourself out after every reply, like I have seen others do.. |
| 15:01.30 | drmessano | That's my #2 pet peeve |
| 15:01.42 | ManxPower | My paypal address is eric@fnords.org |
| 15:02.04 | drmessano | "hey, no probs.. if you don't mind, drop by my website.. post on my forum, send me some paypal, comment on my blog" |
| 15:02.30 | rupa | Put it at the top/bottom of your pastebin |
| 15:02.37 | boblutz | lol |
| 15:02.37 | drmessano | HA |
| 15:02.42 | drmessano | ..NICE |
| 15:05.48 | drmessano | I have my limits when it comes to free help.. and if I get to the point I am helping someone to where someone else would rightfully charge REAL money to do so, i'll tell them "you know, you really need to pay someone to do this if you're not gonna do it yourself" |
| 15:06.44 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 15:07.17 | drmessano | It's not so much "I should make money if I am gonna help you", it's "You're asking me to help you do something you can't/won't do yourself so you don't have to pay someone else real money to do it" |
| 15:07.21 | drmessano | and to me, theres a line |
| 15:07.38 | *** part/#asterisk matrix1233 (n=Administ@196.203.192.150) |
| 15:08.05 | ManxPower | To me the difference is between "Here's your answer" (free) and "I'm going to spend 30 mins trying to figure out the SIP debug you gave me" (not free) |
| 15:09.06 | drmessano | Yep.. and "Does this look right for an extension" and "can you create the rest of my extensions for me?" |
| 15:11.40 | coppice | "My friend has a problem with his extension. Do you think some guy on the internet diagnose his problem, or should he see a doctor?" |
| 15:11.49 | drmessano | lol |
| 15:12.36 | drmessano | I guess I have gotten so cynical because I have ALWAYS been one to help anyone out.. but when it becomes "Do it for me because I am lazy and cheap", you really start to have enough |
| 15:12.40 | rupa | now to figure out if I can get SLA + callwaiting to work through a linksys ta |
| 15:12.59 | boblutz | OMG can I ask a ???? |
| 15:13.36 | boblutz | sorry, im a smartass by nature |
| 15:13.37 | drmessano | boblutz: You are banned from asking questions, for 2 releases |
| 15:13.44 | drmessano | See you at 2.0 |
| 15:13.45 | coppice | "You must be wrong, because I followed the instructions on voipinfo.,org, which were last updated when the US president had a functional brain" |
| 15:13.47 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:13.48 | boblutz | lol |
| 15:13.54 | rupa | snort |
| 15:13.54 | drmessano | HAHAH |
| 15:14.18 | Katty | hmm. |
| 15:14.27 | boblutz | Well, seeing how I didnt know what Asterisk was a couple of months ago, you cant blame a newb |
| 15:14.38 | boblutz | You google Asterisk, and voip-info always shows up |
| 15:15.12 | drmessano | If someone asks you to SSH into their box |
| 15:15.19 | drmessano | Ask them if pastebin is broken |
| 15:15.25 | drmessano | "Ok" is never the correct answer |
| 15:15.37 | coppice | wikis like voipinfo should date stamp everything. in most cases you have no idea how out of date the info is |
| 15:15.56 | drmessano | "New for 1.2" |
| 15:18.23 | coppice | i get suspicious about how up to date things are when they use "ye" regularly |
| 15:18.50 | drmessano | ROFL |
| 15:19.09 | drmessano | Thou shalt not use thyne canreinvite |
| 15:20.48 | Nugget | icanhasreinvite? |
| 15:22.08 | Katty | hehehe |
| 15:22.10 | Katty | hugs Nugget |
| 15:23.08 | hmmhesays | Katty |
| 15:23.32 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 15:23.56 | Katty | hmmhesays: baroo? |
| 15:24.08 | hmmhesays | Good morning |
| 15:24.14 | Katty | good morning (= |
| 15:25.05 | *** join/#asterisk af_ (n=getsmart@88-149-241-199.dynamic.ngi.it) |
| 15:27.28 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:30.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:30.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:30.43 | Katty | hugs russellb |
| 15:30.48 | russellb | thanks :) |
| 15:30.50 | russellb | how are you? |
| 15:31.00 | drmessano | hugs russellb |
| 15:32.10 | russellb | stares at DrAk0 |
| 15:32.18 | russellb | DrAk0: owned by tab completion, sorry. |
| 15:32.22 | russellb | I blame drmessano |
| 15:32.31 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 15:35.10 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 15:35.26 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:37.58 | boblutz | lol |
| 15:38.10 | drmessano | Everyone blames me |
| 15:38.19 | drmessano | That's old and busted |
| 15:38.25 | drmessano | "WOW" me |
| 15:39.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:39.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:44.03 | Katty | russellb: i'm good. waiting for nails to dry. |
| 15:44.21 | drmessano | You hanging drywall? |
| 15:44.34 | Katty | eww. |
| 15:44.37 | Katty | i don't wanna get dirty. |
| 15:45.10 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:45.41 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:46.49 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 15:47.32 | madduck | if, from a softphone on the inside of my home gateway, i call a sip address which resolves to asterisk on the gateway's public IP, and I go via a proxy server out there, then the INVITEs come in to asterisk with To:<sip:$PROXY_DNS_NAME>;tag=.... |
| 15:47.41 | madduck | is their proxy broken? or am i doing something wrong? |
| 15:47.50 | Katty | howard. |
| 15:47.52 | Katty | the duck. |
| 15:47.59 | drmessano | ZOMG |
| 15:48.03 | madduck | is trying to test receiving SIP calls on the gateway |
| 15:48.05 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:48.09 | Katty | jameswf-home: GET OUT |
| 15:48.31 | jameswf-home | wasn't aware I was in ? |
| 15:48.55 | Katty | jameswf: mew? |
| 15:49.03 | Katty | jameswf: i say that to everyone. don't take it personally. |
| 15:49.52 | ManxPower | catturrets 8-) |
| 15:50.43 | Katty | catturrets does not parse. |
| 15:50.46 | Katty | jbot: catturrets? |
| 15:52.54 | drmessano | http://www.youtube.com/watch?v=4A67n0G6JXk <--- for you, my feline friend |
| 15:53.05 | *** join/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 15:53.34 | Shotygun | FlatFoot: Anybody can recommend on a good SIP Signaling & Media gateway that supports full SRTP/TLS? |
| 15:53.42 | Shotygun | hrm, lame script. |
| 15:53.45 | Shotygun | FlatFoot: Anybody can recommend on a good SIP Signaling & Media gateway that supports full SRTP/TLS? |
| 15:53.53 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:53.54 | jameswf | drmessano: did you get happyclownpbx.info |
| 15:53.56 | Shotygun | Sorry, FlatFoot == "OOT:" |
| 15:54.05 | drmessano | nope |
| 15:54.05 | *** part/#asterisk scoates (n=sean@64.15.79.181) |
| 15:54.25 | RockHound | good day ... in running the risk of making a fool out of myself, but is there an easy way how to copy one compile configuration from an older release to the newest sources? basically like copying linux kernel .config file? |
| 15:54.46 | boblutz | RockHound: makefile.makeopts |
| 15:54.51 | boblutz | right? |
| 15:55.06 | ManxPower | ~cattourettes |
| 15:55.07 | jbot | extra, extra, read all about it, cattourettes is a condition where people randomly make cat noises |
| 15:55.15 | ManxPower | much better |
| 15:57.25 | RockHound | boblutz: that is what I was looking for |
| 15:57.26 | RockHound | thx |
| 15:57.50 | RockHound | called makeopts |
| 15:57.53 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 15:58.46 | ManxPower | RockHound: as long as you are moving between the same major version (1.2.5 -> 1.2.18 or 1.4.6 -> 1.4.18) |
| 15:58.56 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:59.08 | RockHound | yes |
| 15:59.23 | ManxPower | makeopts is a 1.4 specific thing, I believe |
| 15:59.23 | RockHound | thx everyone |
| 15:59.24 | *** part/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 15:59.45 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 16:00.09 | Yourname`` | Hi. Is there a way I can record CDRs when sip peers make calls to a particular number? |
| 16:00.55 | ManxPower | Yourname``: that is the default |
| 16:01.00 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:01.20 | ManxPower | see /var/log/asterisk/cdr-csv |
| 16:01.22 | *** join/#asterisk galeras (n=galeras@190.156.212.43) |
| 16:02.16 | ManxPower | What are you doing to prevent this from happening? |
| 16:02.25 | Yourname`` | enable logging = off |
| 16:02.44 | Yourname`` | What I mean is, I want CDRs to be saved only if I call say, 4192292299. |
| 16:02.46 | ManxPower | Yourname``: in cdr.conf? |
| 16:02.53 | Yourname`` | Yes, cdr.conf |
| 16:04.16 | ManxPower | Yourname``: that does not look like a valid cdr.conf option, but that might be 1.4 specific |
| 16:04.32 | *** part/#asterisk galeras (n=galeras@190.156.212.43) |
| 16:04.39 | Yourname`` | ManxPower: Sorry, enabled=no |
| 16:04.41 | ManxPower | Yourname``: I suggest you turn CDRs back on, for every call except for a call to 4192292299 run NoCDR before anything else. |
| 16:05.07 | Yourname`` | I don't get it, I want CDRs only for that number though. |
| 16:05.33 | ManxPower | Yourname``: if you have CDRs enabled, it will by default generate a CDR for every call. |
| 16:05.50 | ManxPower | I gave you my suggestion. |
| 16:06.10 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 16:06.22 | Yourname`` | I know, but I'm asking if there is a way to get CDRs generated for a specific number, and not generated for everything else. |
| 16:06.43 | ManxPower | Yourname``: And I already said the only way I can think of to do that. |
| 16:06.44 | boblutz | Yourname``: Nothing stops you from running AGI to do your bidding |
| 16:07.28 | Nugget | http://www.bbspot.com/News/2000/9/linux_laid.html <-- heh |
| 16:07.46 | Yourname`` | ManxPower: If I enable CDRs, and set NoCDR for that number, how am I going to get CDR info for that number? |
| 16:08.45 | keith4 | Yourname``: using AGI |
| 16:08.53 | boblutz | Nugget: Is that real? |
| 16:09.03 | ManxPower | Yourname``: No, you set NoCDR for ALL OTHER NUMBERS |
| 16:09.14 | ManxPower | The only number you do not set it for is the one you want to log. |
| 16:09.37 | Yourname`` | Aaahhhh |
| 16:09.43 | Yourname`` | Now, THAT, makes sense. |
| 16:09.53 | Yourname`` | keith4: No AGI experience |
| 16:09.56 | ManxPower | ManxPower: Yourname``: I suggest you turn CDRs back on, for every call except for a call to 4192292299 run NoCDR before anything else. |
| 16:10.08 | ManxPower | notice the , |
| 16:10.25 | Yourname`` | I apologize, ManxPower. I read it too fast. |
| 16:10.26 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 16:10.28 | keith4 | Yourname``: what do you mean? you can write an AGI program in any language. surely you know perl or something |
| 16:10.38 | Yourname`` | keith4: NOpe. |
| 16:10.50 | keith4 | php? |
| 16:10.56 | keith4 | bash? |
| 16:11.13 | ManxPower | keith4: Chances are he's not a system admin |
| 16:11.14 | boblutz | irc? |
| 16:11.19 | keith4 | hell, there's even a nice perl language to do most of the work for you |
| 16:11.30 | keith4 | ManxPower: yah, apparently. |
| 16:12.33 | Yourname`` | Not a sysadmin. |
| 16:13.00 | ManxPower | Yourname``: if you want to keep working with Asterisk you will learn some programming |
| 16:13.18 | Yourname`` | I had some knowledge long ago, lol |
| 16:13.24 | ManxPower | (perl or PHP would be best in my IMNSHO) |
| 16:14.25 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 16:16.12 | *** part/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 16:17.09 | ManxPower | weather.com needs to fix their forecasts. The forcasted high for my area is 68F, but the hour-by-hour forecast says it will get up to 58F |
| 16:17.32 | drmessano | Accuweather is worse |
| 16:17.52 | drmessano | They all overinflate the forecast now to bring in more hits |
| 16:18.37 | *** join/#asterisk dcmwai (n=dcmwai@192.228.184.159) |
| 16:19.27 | *** join/#asterisk R-MAN (n=raficmas@client-86-27-169-69.popl.adsl.virgin.net) |
| 16:19.31 | R-MAN | yo yo |
| 16:19.48 | boblutz | R-MAN: yo sup dawgg? |
| 16:20.01 | R-MAN | not much my man sup with chew? |
| 16:20.05 | Nugget | http://bash.org/?329542 <-- hardcore weather |
| 16:20.12 | boblutz | nuthin...jus chillin wit sum IRC peepz |
| 16:20.24 | R-MAN | yeah im jammin to |
| 16:20.28 | boblutz | lol |
| 16:20.47 | russellb | break dances |
| 16:20.47 | R-MAN | at the same time im about to kick the S***T out of my voip provider lol |
| 16:21.36 | R-MAN | any of you peeps good at working out why I cant make outgoing calls through my provider sipgate? |
| 16:21.47 | grandpapadot | wor |
| 16:21.48 | grandpapadot | d |
| 16:21.56 | Yourname`` | If only people could really go and beat the shit out of providers, sigh |
| 16:22.25 | R-MAN | lol |
| 16:27.52 | *** part/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 16:28.11 | R-MAN | guys |
| 16:28.15 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:28.16 | R-MAN | what does this mean |
| 16:28.36 | R-MAN | configure an extension in your dialplan with the same number |
| 16:28.48 | R-MAN | same number from my provider I got that bit |
| 16:29.25 | Qwell | R-MAN: freepbx? |
| 16:29.41 | R-MAN | dare I say yes |
| 16:29.45 | Qwell | ~freepbx |
| 16:29.46 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:29.48 | Qwell | you daren't |
| 16:29.58 | R-MAN | I know I know sorry im just jammin with the guys |
| 16:29.59 | R-MAN | lol |
| 16:30.10 | Qwell | gives R-MAN a cowbell |
| 16:30.15 | R-MAN | since its a bit quiet I thought I would make conversation lol |
| 16:30.17 | Qwell | jam with that |
| 16:30.42 | Yourname`` | cdr.c:831 ast_cdr_init: CDR already initialized on '**Unknown**' -> Why does this keep happening if cdr are not enabled? |
| 16:30.52 | outtolunc | wonders if that just qualified as cow tippin |
| 16:31.03 | Jumpie | how the hell am i supposed to load other packages on a server tottaly dedicated to asterisk |
| 16:31.04 | Yourname`` | More cowbell. |
| 16:31.07 | Jumpie | with a wierd linux instlal? |
| 16:31.08 | Jumpie | lol |
| 16:31.26 | R-MAN | so ive got to ask...sup with the freepbx.trixbox guys I get the feeling Asterisk people got a love hate relationship |
| 16:31.36 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 16:31.56 | Qwell | R-MAN: read what he bot said |
| 16:34.12 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 16:34.22 | bkruse | R-MAN: Call it what you want :] |
| 16:36.36 | [TK]D-Fender | bkruse: I prefer the term "hate-hate" relationship :) |
| 16:36.43 | grandpapadot | lol |
| 16:37.16 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 16:38.13 | bkruse | [TK]D-Fender: That would be accurate :P |
| 16:43.16 | *** part/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 16:48.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:49.15 | *** join/#asterisk Netgeeks (n=chris@gw-hmb.netgeeks.net) |
| 16:51.08 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:52.00 | rupa | Is it possible to background the SayDigits() command? If I'm reading off (say) the callerid I want the user to be able to press a key to break out of SayDigits() |
| 16:52.23 | *** join/#asterisk hohum (n=dcorbe@68.26.208.9) |
| 16:53.07 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 16:54.21 | *** join/#asterisk DuRaZNo (n=angel@mail.soportelinux.com.pe) |
| 16:54.29 | *** part/#asterisk DuRaZNo (n=angel@mail.soportelinux.com.pe) |
| 16:56.19 | [TK]D-Fender | rupa: No, and from the look of it, I don't think SayDigits should be a dialplan applicationa t all, but rather a function. |
| 16:57.03 | Jumpie | hmm if im still waiting for my ip phone and ptsn card, i can still use asterisk to just do 'in house' calling to extensions on phones based on soft phones right? just to test out? |
| 16:57.06 | Jumpie | whats a good softphone package |
| 16:57.15 | drmessano | X-Lite |
| 16:57.26 | mkillebrew | he said good |
| 16:57.33 | Jumpie | lol |
| 16:57.35 | Jumpie | as in....free |
| 16:57.37 | Qwell | [TK]D-Fender: what, like Background(${DIGITS(1234)}) ? |
| 16:57.39 | drmessano | X-Lite |
| 16:57.45 | mkillebrew | ok yea, x-lite |
| 16:57.47 | Jumpie | great, thanks |
| 16:57.51 | rupa | Hrrrm.. ok, I'll add a way for the user to record a name and I can play that in the background. Waiting for callerid to say the # would be an incentive to actually record a name rather than relying on the # |
| 16:58.10 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:58.14 | [TK]D-Fender | Qwell: not a bad idea |
| 16:58.47 | [TK]D-Fender | Qwell: it would return a composite string like "digits/1&hundred&and&one", etfc |
| 16:58.56 | Qwell | yeah |
| 16:58.59 | Qwell | hmm |
| 16:59.15 | *** join/#asterisk talntid (n=swarm@66.208.251.170) |
| 16:59.15 | [TK]D-Fender | Qwell: Several apps could be removed using this style of thinking |
| 16:59.29 | Qwell | and there should be no "and" between hundred, one.. :p |
| 16:59.40 | drmessano | and and and |
| 16:59.42 | Qwell | "and" is reserved for decimals |
| 16:59.51 | rupa | that would be sweet |
| 16:59.54 | [TK]D-Fender | Qwell: for 101 dollars? |
| 16:59.58 | Qwell | correct |
| 17:00.05 | [TK]D-Fender | Qwell: On Hundred and One <- |
| 17:00.07 | Qwell | no |
| 17:00.17 | Qwell | One hundred, one dollars, and eleven cents. |
| 17:00.26 | drmessano | AH |
| 17:00.29 | Qwell | worked for a bank for 5 years. :p |
| 17:00.29 | drmessano | Thats cool |
| 17:00.44 | [TK]D-Fender | Qwell: Ok, I've always heard the "and" |
| 17:00.51 | Qwell | yes, MANY people say it incorrectly |
| 17:01.00 | [TK]D-Fender | Qwell: Suppose that could be a regional thing. |
| 17:01.01 | Qwell | but, working at a bank, you WILL get corrected - every time. |
| 17:01.11 | drmessano | Where we you last week when I needed to know what an audio credit was |
| 17:01.23 | Qwell | anyhow, sorry, that was a random interjection |
| 17:01.26 | Jumpie | hey qwell |
| 17:01.35 | [TK]D-Fender | Qwell: then again, that would be "American" English, so I'm sure we could entrench ourselves quite far on the idea of "correct" ;) |
| 17:01.43 | Jumpie | lol |
| 17:01.49 | [TK]D-Fender | Qwell: But yeah, you go my point earlier |
| 17:02.03 | Jumpie | any idea why i can only set numbers in caller id in asterisknow? :) |
| 17:02.12 | drmessano | [TK]D-Fender: You're missing the #1 rule of banking. "You're wrong" |
| 17:02.13 | [TK]D-Fender | load chan_semantics.so |
| 17:02.28 | [TK]D-Fender | Jumpie: Because its a dumb GUI. |
| 17:02.31 | Katty | ugah. |
| 17:02.33 | Katty | so stuffed. |
| 17:02.33 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 17:02.35 | Katty | can't... move... |
| 17:02.43 | [TK]D-Fender | Jumpie: Don't like it, you've got the source.... get busy... |
| 17:02.50 | Qwell | ha, en_GB does use and |
| 17:02.52 | drmessano | Katty: Keep your personal life in PM's please |
| 17:03.01 | Jumpie | hehe |
| 17:03.01 | Katty | gosh. |
| 17:03.05 | Qwell | [TK]D-Fender: ^^ |
| 17:03.14 | Katty | drmessano: it's not nice to be rude. |
| 17:03.22 | Katty | drmessano: and i don't have time for mean people. |
| 17:03.23 | mkillebrew | obviously.. |
| 17:03.32 | mkillebrew | pokes Katty in the eye |
| 17:03.35 | drmessano | Katty: I'm not being rude.. but bragging about getting stuffed.. That's TMI |
| 17:03.50 | Jumpie | i think it depends on what kinda stuffed you mean tho DrAk0 |
| 17:03.51 | Jumpie | drmessano |
| 17:03.52 | drmessano | There's children in here |
| 17:03.52 | Jumpie | lol |
| 17:04.02 | Katty | drmessano: you, sir, are redonkulous. |
| 17:04.31 | mkillebrew | redonkulous? |
| 17:04.43 | [TK]D-Fender | Qwell: So, think someone would take up the task of converting SayDigits to "${DIGITS()}"? would apply to "SayNumber" and other similar apps. |
| 17:04.46 | x86 | heh i used to use that word all the time |
| 17:05.14 | Qwell | [TK]D-Fender: I wouldn't say convert. SayDigits et al are quite useful on their own |
| 17:05.20 | drmessano | It took me a few weeks of knowing Australians online to figure out what "Get Stuffed" meant |
| 17:05.36 | drmessano | Shockingly, I was told that quite often |
| 17:05.53 | Kobaz | heh |
| 17:06.01 | [TK]D-Fender | Qwell: But completely replaceable by "Background(${SAYNUMBER(12345)}) |
| 17:06.12 | Qwell | [TK]D-Fender: sure |
| 17:06.30 | mkillebrew | drmessano: eh? |
| 17:06.31 | drmessano | Fair dinkum |
| 17:06.43 | Jumpie | [TK]D-Fender so i can edit/change the configs? |
| 17:06.48 | Jumpie | do i have all the tools to recompile? |
| 17:06.52 | drmessano | Bloody oath |
| 17:07.04 | [TK]D-Fender | Qwell: this way adds flexibility, while making the most mundane application of it only slightly longer for those who would use it that way. |
| 17:07.21 | rupa | would also be nice if the FollowMe app would have an option (like Background) to not answer the channel |
| 17:07.24 | [TK]D-Fender | Jumpie: So you can edit the SOURCE CODE that limits what you can put in. |
| 17:07.30 | madduck | "Found no matching peer or user for '84.75.151.30:5060'" - how do I configure asterisk to accept calls from third parties? |
| 17:07.43 | Jumpie | right |
| 17:07.51 | [TK]D-Fender | Jumpie: I'm not psychic you know... do yoU? |
| 17:07.56 | Jumpie | i wanted to be sure if i had all those tools |
| 17:07.58 | drmessano | Aussie, aussie, aussie |
| 17:08.01 | Jumpie | ugh |
| 17:08.13 | Jumpie | k |
| 17:08.14 | rupa | damn, brb, bluetooth is farked on this pc |
| 17:08.14 | [TK]D-Fender | Jumpie: If you don't even know what you've got, then what you've got is a problem... |
| 17:08.21 | Jumpie | no im just not a coder |
| 17:08.24 | Jumpie | i can hack it |
| 17:08.30 | Jumpie | i just wanted to be sure i had all i needed on the install |
| 17:08.34 | drmessano | You're a hacker? |
| 17:08.36 | drmessano | ZOONOO |
| 17:08.40 | Jumpie | i meant 'i can deal on my own' |
| 17:08.45 | Jumpie | but ya i did security stuff too |
| 17:08.46 | [TK]D-Fender | Jumpie: Then feel free to post a request/bounty for someone to change it to work the way you want it to. |
| 17:08.52 | Jumpie | argh... |
| 17:08.52 | Jumpie | nm |
| 17:09.04 | drmessano | Money changes everything |
| 17:09.25 | [TK]D-Fender | drmessano: Only motivations :) |
| 17:10.15 | drmessano | "We've already established that, now we're negotiating price" <--- Punchline to one of the best jokes ever |
| 17:11.03 | [TK]D-Fender | drmessano: Yup.. I know the one :) |
| 17:11.23 | [TK]D-Fender | drmessano: Tends induce dry-cleaning bills ;) |
| 17:11.33 | drmessano | lol |
| 17:11.54 | CCFL_Man2 | everybody loves money |
| 17:12.43 | CCFL_Man2 | dammit the hell where are those mib files |
| 17:13.39 | *** join/#asterisk Bourrelle (n=Bourrell@132.207.156.100) |
| 17:13.44 | grandpapadot | "Damn it to Hell" |
| 17:14.07 | Bourrelle | Hello, im receiving an ICMP Destination Port Unrecheable |
| 17:14.16 | Bourrelle | Im sending an invite to asterisk with the right port |
| 17:14.18 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 17:14.22 | Bourrelle | but asterisk tries to send it to port : 1 |
| 17:14.38 | Bourrelle | ive read on the internet that it might be a firewall problem |
| 17:14.53 | Bourrelle | I don't think so... any have encounter this before ? |
| 17:14.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:16.08 | madduck | are there any open SIP proxies out there which I can use? I have * on the gateway and obviously, if I call it, the call never goes out to the Internet and back, so I want to go via a proxy |
| 17:16.17 | madduck | is testing/learning/debugging |
| 17:16.19 | *** join/#asterisk fupjack (n=justin@cpe-69-207-186-254.rochester.res.rr.com) |
| 17:16.52 | CCFL_Man2 | yay, they sent me the mib file |
| 17:18.15 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 17:18.20 | ManxPower | madduck: Asterisk acts as a proxy |
| 17:19.15 | ManxPower | It is more of a B2BUA, but you can put the address of the Asterisk server in as the proxy address in your SIP client. |
| 17:19.16 | madduck | yeah, but i need one out there... |
| 17:19.45 | ManxPower | madduck: sounds like you need to fix your NAT problem, not use a proxy as it will do no good whatsoever until you fix your NAT problem anyway |
| 17:20.07 | ManxPower | madduck: sign up for FWD and use their proxy |
| 17:20.10 | *** part/#asterisk R-MAN (n=raficmas@client-86-27-169-69.popl.adsl.virgin.net) |
| 17:20.15 | ManxPower | but using a proxy almost never fixes anything |
| 17:21.05 | ManxPower | madduck: and oddly you are the only person ever to be running Asterisk on a NAT gateway and need to use a proxy -- you are special |
| 17:21.40 | ManxPower | I'll bet you did something silly like setting bindaddr= or something like that. |
| 17:21.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:22.15 | ManxPower | don't use bindaddr=, set canreinvite=no and things should work a bit better. |
| 17:22.36 | ManxPower | madduck: come to think of it, you should be able to use any provider as a proxy |
| 17:23.31 | ManxPower | Bourrelle: the only reason you would get ICMP port unreachable is 1) nothing listening on the port 2) port is blocked by a firewall or 3) you screwed something up. |
| 17:23.45 | tzafrir | madduck, http://rapid.tzafrir.org.il/~tzafrir/sip_net_settings |
| 17:24.17 | ManxPower | tzafrir: that works if Asterisk is running on the NAT machine? |
| 17:24.28 | tzafrir | no |
| 17:24.34 | tzafrir | probably not |
| 17:24.47 | ManxPower | tzafrir: as far as I can tell that's madduck's situation |
| 17:24.55 | ManxPower | madduck: are there any open SIP proxies out there which I can use? I have * on the gateway and obviously, if I call it, the call never goes out to the Internet and back, so I want to go via a proxy |
| 17:25.11 | ManxPower | as you can see above. |
| 17:25.38 | ManxPower | looks like madduck is not really looking for help anyway as he's not trying to help troubleshoot his own issue. |
| 17:25.47 | tzafrir | Shotygun, depends on your definition of popularity |
| 17:25.56 | madduck | ManxPower: ? |
| 17:26.05 | Shotygun | crowded |
| 17:26.14 | ManxPower | madduck: ? |
| 17:26.19 | Shotygun | Crowded is incorrect actually |
| 17:26.23 | Shotygun | but.. common.. that's the right one |
| 17:26.30 | madduck | ManxPower: i am looking for help but i was debugging... |
| 17:26.38 | ManxPower | madduck: do one or the other. |
| 17:26.41 | ManxPower | We do this for free. |
| 17:26.41 | madduck | ManxPower: i run * on the gateway, yes. |
| 17:26.55 | madduck | ManxPower: sorry... I didn't think IRC was *that* real-time. :) |
| 17:26.58 | ManxPower | madduck: look at the issues I've been talking about. |
| 17:27.07 | madduck | i did and i was about to answer... |
| 17:27.23 | madduck | i don't *need* a proxy |
| 17:27.27 | madduck | nor do i want one |
| 17:27.32 | ManxPower | madduck: If you and the helper are AFK all the time nothing is going to get done. |
| 17:27.37 | madduck | but i want a way in which I can call my * from the outside |
| 17:27.55 | madduck | if i call it from the inside, then it just loopbacks |
| 17:27.57 | ManxPower | madduck: Um, sign up for a provider |
| 17:28.04 | ManxPower | madduck: if you call WHAT from the inside?? |
| 17:28.21 | asteriskmonkey | My asterisk 1.4 install seems to stop reloading configs after a while |
| 17:28.24 | madduck | ManxPower: my gateway's public IP - or well, a SIP address that resolves to that. |
| 17:28.28 | asteriskmonkey | anyone seen this behavious before? |
| 17:28.39 | ManxPower | bad: I can't call it. good: I get a 302 redirect in the CLI when I try to call 5551212 from my polycom sip phone |
| 17:28.40 | bkruse | asteriskmonkey: If the config file is unchanged, it will not reload it |
| 17:29.08 | ManxPower | madduck: I assume you want random people on the internet to be able to do a guest SIP call to your server directly? |
| 17:29.16 | madduck | ManxPower: yes. |
| 17:29.25 | madduck | ManxPower: and i want to test and debug it |
| 17:29.26 | asteriskmonkey | bkruse: but it is changed other command no longer work either ie. show channels |
| 17:29.27 | ManxPower | madduck: Um, OK, so what is the problem? |
| 17:29.38 | madduck | ManxPower: well, how would you do it? |
| 17:29.40 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 17:29.43 | bkruse | asteriskmonkey: that is not normal |
| 17:29.44 | ManxPower | what HAPPENS when you tell your SIP phone to dial 2344@yourhostname.com |
| 17:29.50 | madduck | if i just call from the inside, even to the public IP, then it never leaves the machine |
| 17:29.53 | ManxPower | well, your sip phone that is outside your network |
| 17:30.05 | madduck | i don't have a SIP phone outside the network, that's my problem |
| 17:30.11 | ManxPower | don't expect to be able to call your public IP from inside |
| 17:30.13 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:30.27 | madduck | i can call from the inside just fine, but as I said, that's because of Linux loopback |
| 17:30.36 | madduck | but that's not a realistic testing scenario |
| 17:30.42 | drmessano | Have you had someone call you? |
| 17:30.50 | ManxPower | madduck: so you want to set up something that is pretty complex but don't have any kind of environment to do real world testing? |
| 17:30.59 | madduck | drmessano: i don't want to have to ask others all the time while i am learning. |
| 17:31.00 | ManxPower | madduck: define "loopback" |
| 17:31.32 | madduck | ManxPower: it never even see the external network interface card on the gateway because the routing engine determines that it'll be delivered to the gateway anyway. |
| 17:31.34 | [T]ank | so i have a server with a digium TE410P that has been working well for a few years now. I recently took the pris out of it and went to an ip connection to another one of my servers with pris in it. I recently noticed that i have trouble with dtmf oun outbound calls now. Could this be related to removing the pris? |
| 17:31.36 | ManxPower | loopback = 127.0.0.1, loopback = calling same phone on same server as you are calling from, loopback=type of T-1/E-1 cable |
| 17:31.47 | ManxPower | madduck: routing engine? |
| 17:31.55 | ManxPower | you are using all these terms that mean nothing to us. |
| 17:32.05 | madduck | ManxPower: loopback: when you ping your own public IP |
| 17:32.14 | ManxPower | madduck: no, that is called pinging an IP. |
| 17:32.15 | madduck | on linux at least, that doesn't go via OSI or the driver, that goes via loopback |
| 17:32.23 | drmessano | madduck: You have no real way of testing outside without something OUTSIDE calling you |
| 17:32.31 | madduck | ManxPower: i appreciate your trying to help me and all, but please don't assume I am an idiot. |
| 17:32.38 | drmessano | So ask someone for a test, and if it works, dont fuck it up |
| 17:32.40 | madduck | drmessano: which is why i want to go via a proxy. |
| 17:32.41 | ManxPower | madduck: stop acting like one. |
| 17:32.57 | madduck | ManxPower: thanks, i'll see if someone else can help. |
| 17:33.02 | ManxPower | madduck: best of luck. |
| 17:33.10 | tzafrir | [T]ank, hmm... nothing connected to the card? so it does not get timing? |
| 17:33.22 | [T]ank | yeah... thats what I was wondering... would that cause my issue? |
| 17:33.38 | drmessano | madduck: You're arguing on the internet instead of testing.. You're losing karma by the minute here |
| 17:33.43 | [TK]D-Fender | tzafrir :Thats a nifty little script. How is it that it pulls only to "local" subnets for the localnet clause? I suck at bash/sed |
| 17:33.48 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:33.48 | ManxPower | [T]ank: Usually you just stop getting audio if you load the card driver, but don't configure it |
| 17:33.59 | *** join/#asterisk razu__ (n=razu@195.222.7.33) |
| 17:34.10 | [T]ank | well.. the card is configured... it is just missing the pri connections... |
| 17:34.11 | madduck | is it a wrong assumption of me that going via a proxy will make my call appear to original from that proxy? |
| 17:34.13 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:34.14 | [TK]D-Fender | tzafrir : I have 2 interfaces on my home server that could apply, yet it pulls only the internal one, not my WAN. |
| 17:34.27 | grandpapadot | madduck: Yes, go get a les.net account and test away. |
| 17:34.29 | [T]ank | everything is in red alarm on zap show stats |
| 17:34.30 | madduck | s/original/originate/ |
| 17:34.38 | drmessano | madduck: Get over this whole proxy thing.. Just get someone to test with you |
| 17:34.47 | tzafrir | [TK]D-Fender, I opted for the very simple case. Your case will require something smarter... |
| 17:34.54 | drmessano | If you want real world, use real world |
| 17:34.54 | madduck | drmessano: that's hardly helpful. really... |
| 17:35.12 | [TK]D-Fender | tzafrir : What I don't get is the fact it ONLY pulls my internal one. |
| 17:35.16 | drmessano | Well, google is your friend.. go find a proxy.. Good luck |
| 17:35.22 | [TK]D-Fender | tzafrir thats what I don't get.... |
| 17:35.25 | ManxPower | [TK]D-Fender: I assume you have no bindaddrs |
| 17:35.35 | tzafrir | happens to be the first, I guess |
| 17:35.41 | [TK]D-Fender | ManxPower: I'm just talking about se + ifconfig |
| 17:35.42 | tzafrir | (or the last?) |
| 17:35.49 | madduck | grandpapadot: okay, so les.net allows you to make test calls? do they provide the SIP debug output if needed? |
| 17:35.53 | ManxPower | [TK]D-Fender: SELinux, I assume? |
| 17:36.01 | [TK]D-Fender | tzafrirYes, it is the first... I don't see it limiting to just the first however |
| 17:36.08 | [TK]D-Fender | ManxPower: Nope. |
| 17:36.12 | grandpapadot | madduck: You'll pay per minute. |
| 17:36.15 | ManxPower | service engineer? |
| 17:36.43 | coppice | this engineers likes servicing :-) |
| 17:36.53 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 17:38.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:41.11 | ManxPower | drmessano: I suspect madduck either has a dialplan problem or a network routing problem, but he's arguing rather than trying to fix it. |
| 17:41.26 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 17:41.49 | ManxPower | drmessano: we agree on something -- maybe today I should buy a lottery ticket 8-) |
| 17:42.00 | drmessano | lol |
| 17:42.31 | drmessano | That's pet peeve #3, I think |
| 17:42.54 | madduck | ManxPower: dude, maybe you could stop discouraging others from trying to help? I have neither a dialplan problem nor a routing problem. I know exactly what I am trying to do |
| 17:43.09 | drmessano | Not being near the box and arguing, not planning to test and arguing |
| 17:43.11 | madduck | now, unless you can actually answer my question, please just put me on /ignore |
| 17:43.12 | drmessano | "No it's not" |
| 17:43.15 | ManxPower | Not being in an enviroment where you can test what you need to test is my main one. |
| 17:43.16 | drmessano | "yes it is" |
| 17:43.18 | drmessano | "No it's not" |
| 17:43.34 | ManxPower | madduck: you are welcome to put me on /ignore |
| 17:43.56 | ManxPower | madduck: I said "no you do not need a proxy". That was your question, that was my answer. |
| 17:44.08 | madduck | no, my question was "is it a wrong assumption of me that going via a proxy will make my call appear to originate from that proxy?" |
| 17:44.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:44.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:44.14 | madduck | i never asked whether i needed a proxy |
| 17:44.14 | drmessano | madduck: You don't need to worry about someone discouraging anyone from helping you.. if you're not willing to help yourself or follow advice in here, people pick up on that.. |
| 17:44.27 | ManxPower | drmessano: maybe [TK]D-Fender can help him 8-) |
| 17:44.28 | madduck | trolls |
| 17:44.40 | drmessano | Yes. [TK]D-Fender can help you |
| 17:44.52 | madduck | he has been helpful in the past, yes. unlike you two monkeys. |
| 17:44.55 | drmessano | PM him, he loves that |
| 17:45.03 | madduck | *plonk* |
| 17:45.08 | ManxPower | drmessano: what madduck doesn't realize is that asterisk will throw a fit if he tries to hairpin calls. |
| 17:45.33 | madduck | sorry to everyone else in the channel for this. |
| 17:45.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 17:45.42 | ManxPower | well, MAY, rather than WILL |
| 17:46.06 | drmessano | I'm not sure how signing up with an ITSP and having it send calls to his box is going to show him one bit of what an unauthenticated guest call is going to do |
| 17:46.12 | drmessano | But.. you know |
| 17:46.16 | [T]ank | madduck: thanks for your appology... I am leaning towards the side of ManxPower |
| 17:46.30 | grandpapadot | lol wow |
| 17:46.30 | ManxPower | drmessano: can't force them to do the right thing. |
| 17:46.54 | madduck | wtf? i am used to #debian-* channels, but this is nowhere near... |
| 17:47.18 | [T]ank | yeah, this is not debian. |
| 17:47.19 | drmessano | madduck: So how was your 16th birthday party? |
| 17:47.24 | madduck | apparently only those people without a clue and no will to even try to understand are the ones speaking. |
| 17:47.27 | grandpapadot | rofl |
| 17:47.58 | russellb | madduck: the people in here are a bit harsh sometimes, I apologize. |
| 17:48.00 | ManxPower | [T]ank: try asking if you a using a proxy will help you access a web page on your public IP and I suspect you will get a similar response. |
| 17:48.04 | Kobaz | madduck: that's usually the case |
| 17:48.11 | madduck | sigh |
| 17:48.15 | drmessano | ManxPower: .. and even after being told the golden secret to getting help in #asterisk, he's still whining like a 12 yr old |
| 17:48.22 | madduck | so is there anyone who can actually answer my question? is it a wrong assumption of me that going via a proxy will make my call appear to originate from that proxy? |
| 17:48.31 | grandpapadot | The answer to your question is yes, if you use an external proxy to call your SIP server, it will work the way you're describing. |
| 17:48.39 | [TK]D-Fender | grabs some popcorn and sits back to watch the show... |
| 17:48.42 | ManxPower | russellb would be a good one to ask, as he is a Digium developer. |
| 17:48.44 | Qwell | madduck: depends on the proxy and your setup |
| 17:49.02 | ManxPower | just don't /msg him. |
| 17:49.05 | *** join/#asterisk KaiK (n=KaiK@dslb-088-076-061-225.pools.arcor-ip.net) |
| 17:49.10 | [T]ank | ManxPower: yep... I probably would. I don't understand why someone would put so much effort into fighting this and doing so much work around to simulate a much easier test. |
| 17:49.14 | drmessano | I have his home phone number, if you like |
| 17:49.24 | Kobaz | heh |
| 17:49.27 | russellb | drmessano: stalker |
| 17:49.48 | drmessano | russellb: OMG, you said my nickname |
| 17:49.51 | drmessano | faints |
| 17:50.01 | ManxPower | [T]ank: To be fair, loading up a softphone an a SEPARATE internet connection not exactly easy when you don't have one handy |
| 17:50.11 | Jumpie | lol |
| 17:50.17 | russellb | drmessano: drmessano drmessano drmessano drmessano drmessano |
| 17:50.34 | drmessano | takes multiple screenshots |
| 17:50.38 | drmessano | <3 |
| 17:50.46 | [T]ank | ManxPower: that is fair. I think it is more the effort put into this argument that amuses me |
| 17:50.46 | Qwell | drmessano: type /clear, it's awesome |
| 17:50.53 | ManxPower | [T]ank: Unfortunately that's the only way I can think of for him to properly test it. |
| 17:51.05 | drmessano | So is Ctrl-Alt-Shift-F4 |
| 17:51.21 | denon | but I only have 10 fingers! |
| 17:51.29 | ManxPower | [T]ank: why anyone would want to accept guest calls is beyond me, but many people what that. |
| 17:51.30 | drmessano | denon: eBay |
| 17:51.54 | madduck | ManxPower: i could do some tunneling magic, but i figure a proxy is all i need, which has now been confirmed by one other person. |
| 17:52.01 | drmessano | ManxPower: It's possible someone with no friends would pray for guest calls |
| 17:52.05 | [T]ank | off to lunch... bye all |
| 17:52.07 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:52.12 | madduck | ManxPower: now if you don't understand why that works, then you really ought to keep your mouth shut more frequently. |
| 17:52.20 | drmessano | ManxPower: Wishful thinking, perhaps |
| 17:52.27 | drmessano | ouch |
| 17:52.40 | drmessano | What a mad, mad duck |
| 17:52.43 | ManxPower | drmessano: I blame the "use voip and get free calls" crap everyone spews. |
| 17:53.17 | drmessano | ManxPower: As I said the other night.. give out your SIP URI to 1000 people, get -3 calls back |
| 17:53.20 | Jumpie | lol |
| 17:53.52 | [TK]D-Fender | ManxPower: Well... the call is free... up to the ITSP. Now convincing them to accept it and then actually terminate it ... well thats another matter :) |
| 17:54.12 | drmessano | Great idea.. and I do make some SIP calls... to other elitist asterisk-loving nerds like myself |
| 17:54.28 | ManxPower | [TK]D-Fender: I was referring to a non-ITSP situation. |
| 17:54.50 | drmessano | les.net is actually accepting SIP calls |
| 17:54.57 | drmessano | As of late |
| 17:54.58 | madduck | drmessano: you'll have to agree that it's more comfortable to test a setup and experiment with it when you don't have to keep asking others for a ring. |
| 17:55.02 | drmessano | But that's.. rare |
| 17:55.11 | madduck | les.net appears to require a US phone number. |
| 17:55.20 | drmessano | HA |
| 17:55.21 | drmessano | No |
| 17:55.23 | drmessano | Go read more |
| 17:55.27 | grandpapadot | Yea, les will accept anonymous calls to his customers accounts |
| 17:55.34 | Katty | wonders if drmessano is turning into [TK]D-Fender |
| 17:55.35 | madduck | drmessano: i called them up even. |
| 17:55.38 | ManxPower | drmessano: I'm not a fan of VoiceOverIPOverInternet anyway |
| 17:55.54 | drmessano | madduck: THEY are MY ITSP, so FAIL |
| 17:56.32 | madduck | you own the company, work for them or just pay them? |
| 17:56.36 | drmessano | I use them |
| 17:56.39 | drmessano | I know how they work |
| 17:56.44 | drmessano | You obviously do not |
| 17:56.52 | Katty | think drmessano shoudl just simmer down a bit. |
| 17:57.03 | Bourrelle | Anyone experimented with symmetrical RTP session ? |
| 17:57.20 | drmessano | madduck: I suggest you go to another net, make a friend.. don't piss them off for a week, get them to help you test your system |
| 17:57.32 | Katty | drmessano: if you're so convinced it will work, give madduck your contact person. |
| 17:57.37 | madduck | drmessano: you are possibly the biggest idiot i've met on IRC in a long time. |
| 17:57.52 | grandpapadot | madduck: dude, you're not far behind, stop casting stones ... |
| 17:57.53 | drmessano | He doesn't need a contact person.. he can go to the website and see what an ITSP is |
| 17:58.03 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 17:58.03 | drmessano | They don't "require a US phone number" |
| 17:58.05 | ManxPower | drmessano: I suggest he hire a consultant 8-) |
| 17:58.29 | madduck | grandpapadot: true. |
| 17:58.34 | drmessano | You can go to their site and see what they offer |
| 17:58.58 | jjshoe | irc and pissing matches <3 |
| 17:59.10 | drmessano | madduck: I am a far bigger idiot than you will ever be.. but I happen to be right about this. |
| 17:59.17 | Jumpie | haha jjshoe |
| 17:59.27 | jjshoe | Jumpie :) |
| 17:59.48 | drmessano | madduck: Open up Netscape Navigator, go to http://www.les.net, and read |
| 17:59.49 | Jumpie | i've havent seen a 'whos a bigger idiot' match in awhile |
| 18:00.08 | jjshoe | Jumpie you must not be on a lot of channels :D |
| 18:00.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:00.14 | Jumpie | im mainly on efnet |
| 18:00.15 | Jumpie | lol |
| 18:00.26 | jjshoe | Jumpie me too, and that's where it all starts :P |
| 18:00.32 | Jumpie | i suppose lol |
| 18:00.46 | Jumpie | i havent hung out in #teenzone and stuff in o0ver a decade tho :P |
| 18:00.47 | drmessano | My only fault is that I am willing to sink to their level.. it helps get some frustations out.. |
| 18:00.54 | Jumpie | drmessano yea |
| 18:00.58 | Jumpie | sometimes its easy to get sucked in |
| 18:01.26 | drmessano | http://www.xkcd.com/386/ <--- Says it all |
| 18:01.37 | ManxPower | drmessano: especially when you know you are right |
| 18:01.43 | grandpapadot | lol |
| 18:01.49 | drmessano | Yes |
| 18:02.06 | jjshoe | ManxPower the whole problem is both idiots think they are right. |
| 18:02.29 | drmessano | jjshoe: Don't you work for Fonality? |
| 18:03.06 | drmessano | The defense rests |
| 18:03.15 | Jumpie | :P |
| 18:03.21 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:03.31 | jjshoe | of course since irc is life it's worth arguing to the bone :P |
| 18:03.35 | Jumpie | i work for telco :P |
| 18:04.02 | jjshoe | i work for $ |
| 18:04.09 | ManxPower | jjshoe: Usually the one with more experience with Asterisk is right. That would include most Digium devs, Qwell, russelb, [TK]D-Fender, me, and others I don't recall at the moment. |
| 18:04.14 | Jumpie | jjshoe but that could mean your a stripper |
| 18:04.19 | Kobaz | $ works for me |
| 18:04.31 | Kobaz | much more efficient that way |
| 18:05.10 | jjshoe | Juggie I'll strip for money, sure. |
| 18:05.24 | ManxPower | Usually I just wish people that don't listen "the best of luck" and leave it at that |
| 18:05.25 | grandpapadot | hrm.. they usually pay me to keep my clothes on |
| 18:05.35 | jjshoe | Kobaz hrm, really? I earn more a year working then I do on investments, but I would love for that to change! |
| 18:05.50 | mort_gib | Like hell you do ManxPower |
| 18:05.59 | Kobaz | jjshoe: make more investments :P |
| 18:06.00 | jjshoe | ManxPower I don't want to get involved in your scuffle, just stand, point, laugh ;) |
| 18:06.05 | ManxPower | mort_gib: I DO try. |
| 18:06.10 | mort_gib | -Sure |
| 18:06.19 | jjshoe | Kobaz seen the arc on chipotle stock? wish I would have invested in them three years ago |
| 18:06.19 | Jumpie | i wish i had invested in gold in 1999 |
| 18:06.20 | Jumpie | like i almost did |
| 18:06.24 | madduck | ManxPower: see, the thing is that I am actually trying to understand * and so even though there may be more logical things to do in my situation, like a second IP, or a friend that calls me, I also really want to know why going via a proxy would or would not work. |
| 18:06.33 | Kobaz | jjshoe: nope |
| 18:06.39 | madduck | because if it isn't a proxy it shouldn't be called that. |
| 18:06.47 | madduck | and if it is a proxy, then what I am trying to do should just work |
| 18:06.50 | drmessano | I'd rather be a stripper than work for Fonality, IMHO |
| 18:06.58 | madduck | and I should look elsewhere for the problem. |
| 18:07.56 | Jumpie | DrAk0 |
| 18:07.58 | Jumpie | drmessano |
| 18:08.01 | Jumpie | i married a stripper |
| 18:08.02 | Jumpie | turned out nasty |
| 18:08.07 | ManxPower | madduck: best of luck |
| 18:08.12 | jjshoe | Jumpie herpes nasty? |
| 18:08.18 | drmessano | Jumpie: That was my experience with trixbox |
| 18:08.20 | Kobaz | jjshoe: guy i know had invested his life savings in intel in like. 1970, he made out pretty well |
| 18:08.26 | drmessano | Jumpie: So, I can relate |
| 18:08.32 | Jumpie | lol |
| 18:08.35 | Jumpie | jjshoe no more like |
| 18:08.42 | Jumpie | i went to iraq and she was a whore with my whole unit |
| 18:08.44 | Jumpie | kinda nasty... |
| 18:08.56 | jjshoe | Jumpie supporting the troops :) |
| 18:09.03 | jjshoe | one rifle at a time.. |
| 18:09.08 | Jumpie | heh |
| 18:09.08 | drmessano | Yoou do have to appreciate the irony |
| 18:09.11 | Jumpie | glad it happened tho |
| 18:09.15 | Jumpie | im much happier and smtarter now |
| 18:09.42 | drmessano | Someone who takes their clothes off for money asking you to trust them around potential clients |
| 18:09.46 | ZPertee | whats this mean http://pastebin.ca/964968. see it when I try to dial out |
| 18:11.08 | coppice | Kobaz: I bet he was nervous in the early 80s :-) |
| 18:12.09 | ManxPower | ZPertee: Not sure, but it looks like it is when Asterisk is accepting a call, not making a call. |
| 18:12.36 | ManxPower | ZPertee: what phone are you using? |
| 18:14.31 | ManxPower | well back to working in the yard. |
| 18:15.14 | Kobaz | coppice: heh, it's nerve racking with any investment, but yeah, i think he made something like 2 million.... mmm |
| 18:15.27 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 18:15.27 | Kobaz | so it just goes to show, don't bail early on your investments |
| 18:15.45 | jjshoe | Kobaz 2 mill isn't a lot so to speak, enough to live off the interest, but still... |
| 18:16.19 | Kobaz | 1 million, at 5% interest a year, is 50k in interest |
| 18:16.35 | coppice | jjshoe: but not a bad return when your initial investment is only 1.9M :-) |
| 18:17.08 | jjshoe | coppice hahahah |
| 18:17.15 | jjshoe | Kobaz -taxes = 25k a year |
| 18:17.31 | Kobaz | you pay almost that with income taxes anyway |
| 18:17.43 | Kobaz | so your 50k salary is also blotted out |
| 18:17.55 | jjshoe | Kobaz indeed, but I wouldn't want to survive on 50k net |
| 18:18.11 | ZPertee | ManxPower, it is a linksys ata spa8000 |
| 18:18.25 | Kobaz | depends on your lifestyle |
| 18:18.31 | jjshoe | Kobaz indeed. |
| 18:19.21 | jjshoe | hookers and blow adds up ;) |
| 18:19.28 | Kobaz | it does |
| 18:19.41 | x86 | heh where I live, 50k (gross) is enough for my wife to stay at home, make double house payments every month, eat healthy, pay all of our bills, and still have money left over to go out |
| 18:19.56 | x86 | :p |
| 18:20.15 | x86 | including new car payment, insurance, etc too |
| 18:21.14 | jjshoe | x86 the sticks? :P |
| 18:22.09 | Jumpie | um wow |
| 18:22.11 | Jumpie | x86 |
| 18:22.15 | Jumpie | wher do you live oklahoma |
| 18:22.21 | jjshoe | I spend too much money eating out and drinking alcohol, that I know. If I cooked at home and didn't drink I would have quite a bit more, but still, it would be nice to rake in way more then needed :) |
| 18:22.40 | jjshoe | money can't buy happyness, but I'll sure try! |
| 18:23.05 | Jumpie | i make 130k cumulatively |
| 18:23.06 | Jumpie | and its still rough |
| 18:23.14 | jjshoe | gross or net? |
| 18:23.17 | Jumpie | gross |
| 18:23.18 | Jumpie | hehe |
| 18:23.27 | jjshoe | what metropolitan area are you in? :P |
| 18:23.31 | jjshoe | ahh, washington dc? |
| 18:23.37 | Kobaz | jjshoe: heh |
| 18:23.53 | Jumpie | ye[ |
| 18:23.53 | jjshoe | Jumpie what's a house in a non ghetto cost out there? |
| 18:24.07 | Jumpie | i mean i pay ok |
| 18:24.09 | Jumpie | 1450/mo |
| 18:24.12 | Jumpie | includes utilities, condo |
| 18:24.23 | Jumpie | i also had ALOT of bad debt earli |
| 18:24.25 | Kobaz | jjshoe: get the tom stanly book the millionaire mind... look up income statement affluent |
| 18:24.38 | Jumpie | so i make ok money but its payin old stuff pre dievorce |
| 18:25.16 | jjshoe | Jumpie that's not too bad, how many sq ft? |
| 18:25.26 | jjshoe | Kobaz http://www.amazon.com/Millionaire-Mind-Thomas-J-Stanley/dp/0740703579 ? |
| 18:25.27 | x86 | Jumpie, jjshoe: Peoria, IL -- world headquarters of fortune 500 company Caterpillar |
| 18:25.31 | Jumpie | hheh hmm not exactly sure |
| 18:25.40 | Jumpie | 3 bedrooms, good living room, 2 bath, breakfast nook |
| 18:25.43 | Jumpie | probably like 1500 iksh |
| 18:25.53 | Jumpie | ah ok |
| 18:25.55 | Jumpie | i used to live in belleville |
| 18:26.21 | Kobaz | jjshoe: yeap |
| 18:27.07 | Kobaz | it's a good read for working on making more than you do now |
| 18:27.26 | jjshoe | Kobaz the intro looks cool |
| 18:27.54 | jjshoe | x86 I take it that's who you work for ;) I lived in Batavia, IL for a long time. |
| 18:28.10 | Kobaz | jjshoe: i got the audiobook, it worked out nice listening back and fourth to work |
| 18:29.00 | jjshoe | hrm cool, have a flight middle of this week, might try to get a hold of it pre-flight |
| 18:29.12 | Kobaz | "live well below your means" is a theme throughout |
| 18:29.23 | jjshoe | Kobaz of course. |
| 18:30.16 | x86 | wow... my house payment is like $275/mo (before insurance and tax escrow), and I pay at LEAST double, most of the time even triple, house payments every month |
| 18:30.28 | jjshoe | x86 how many years is your mortgage? |
| 18:30.31 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-166-99-223.bltmmd.east.verizon.net) |
| 18:30.36 | x86 | jjshoe: nope, I'm the director of IT for a medium-sized publishing company |
| 18:30.44 | Jumpie | hehe im not owning ahome for awhile |
| 18:31.03 | x86 | jjshoe: I wanted a 15 year.... since I was a first-time homeowner, they screwed me into a 30 year |
| 18:31.15 | jjshoe | the housing market in la is starting to crap out, but it's still up there in the really nice places |
| 18:31.30 | x86 | it'll be paid off in about 8 years at the rate I'm going though ;) |
| 18:31.33 | Kobaz | my favorite quote "we the lenders own it all" |
| 18:31.39 | jjshoe | x86 ahh well, getting a 30 year and paying more is better then even a 5 year and paying the exact amount |
| 18:31.51 | *** join/#asterisk SteveTotaro (n=Administ@pool-71-166-99-223.bltmmd.east.verizon.net) |
| 18:31.59 | x86 | jjshoe: for sure :) |
| 18:32.04 | x86 | jjshoe: less interest :) |
| 18:32.37 | Kobaz | totally... paying tripple the price over 30 years is perfect |
| 18:32.40 | jjshoe | well as far as your credit score goes even |
| 18:32.57 | jjshoe | Kobaz you don't pay triple the price when you pay off a 30 year in 10 years |
| 18:33.06 | jjshoe | unless you signed a retarded mortgage saying no early pay off |
| 18:33.08 | Kobaz | ooooh, i misread it |
| 18:33.34 | Kobaz | i thought you meant getting a 30 year and sticking with paying more for the 30 years |
| 18:33.37 | Kobaz | heh |
| 18:33.45 | jjshoe | it's also arguably better in emergency situations, a smaller house payment won't drain your emergency fund so quickly. |
| 18:33.52 | Kobaz | yeah |
| 18:33.53 | jjshoe | Kobaz F no :P |
| 18:34.03 | jjshoe | I'd never take 30 years to pay for a house. |
| 18:34.06 | Kobaz | heh |
| 18:34.24 | Kobaz | my parents did, and i learned early that it's not a good idea |
| 18:34.26 | x86 | if you take 30 years to pay off your 30 year mortgage, you're living beyond your means |
| 18:34.41 | jjshoe | I like the invention of the interest only loan |
| 18:34.46 | Kobaz | haha |
| 18:34.47 | jjshoe | bankers know how to feed off of stupid people |
| 18:34.57 | x86 | lol |
| 18:35.05 | *** join/#asterisk s0lid (n=s0lid@210.213.198.56) |
| 18:35.16 | Kobaz | if interest only loan doesn't ring alarm bells right off the bat, that person is screwed anyway |
| 18:35.29 | jjshoe | too true |
| 18:35.40 | *** join/#asterisk Olobola (i=Olobola@98.sub-75-208-35.myvzw.com) |
| 18:35.53 | Kobaz | just like a landlord i had, that got nigerian scammed |
| 18:36.09 | *** join/#asterisk ThoMe (n=tm@81.92.168.130) |
| 18:36.10 | ThoMe | hello |
| 18:36.25 | ThoMe | hello, is it normal: http://paste.zoffix.com/1206988556/index.html |
| 18:36.26 | ThoMe | ?? |
| 18:36.30 | ThoMe | Disconnected from Asterisk server |
| 18:36.38 | Kobaz | everything is normal, we are all normal |
| 18:36.43 | ThoMe | Kobaz: :-) |
| 18:36.48 | Jumpie | lol |
| 18:36.49 | ThoMe | Kobaz: please look: http://paste.zoffix.com/1206988556/index.html |
| 18:37.10 | Kobaz | that looks bad |
| 18:37.18 | Kobaz | is asterisk still running after that? |
| 18:37.23 | ThoMe | no |
| 18:37.27 | ThoMe | shuting down |
| 18:37.41 | ThoMe | olymp:/etc/asterisk# tail -n 1 /var/log/asterisk/messages |
| 18:37.41 | ThoMe | [Mar 31 20:30:34] WARNING[8395] channel.c: No path to translate from CAPI/ISDN1#02/01633568286-0(0) to SIP/82-081e7018(2048) |
| 18:37.44 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 18:37.44 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:37.45 | ThoMe | olymp:/etc/asterisk# |
| 18:37.48 | Kobaz | what versions of everything? |
| 18:37.53 | ThoMe | now erros. |
| 18:38.07 | ThoMe | ii asterisk 1.4.18.1~dfsg-1 Open Source Private Branch Exchange (PBX) |
| 18:38.12 | ThoMe | ii asterisk-chan-capi 1.0.2-1 Common ISDN API 2.0 implementation for Aster |
| 18:38.17 | ThoMe | that. |
| 18:38.27 | Kobaz | looks like the idsn side of things may be crashing |
| 18:38.33 | ThoMe | mh. |
| 18:38.49 | ThoMe | its gay :-( |
| 18:38.52 | Jumpie | doh |
| 18:38.56 | ThoMe | hihi |
| 18:39.13 | Kobaz | i would downgrade asterisk and test |
| 18:39.25 | ThoMe | downgrade? its the first installation. |
| 18:39.29 | ThoMe | debian package. |
| 18:39.35 | Kobaz | and check out the change logs for capi modifications |
| 18:39.41 | Kobaz | ThoMe: compile from source |
| 18:39.45 | ThoMe | Kobaz: mhh |
| 18:39.49 | ThoMe | Kobaz: ok. |
| 18:40.24 | ThoMe | olymp:/# dpkg --purge asterisk asterisk-chan-capi asterisk-config asterisk-prompt-de asterisk-sounds-main && rm -rf /var/log/asterisk/ && rm -rf /etc/asterisk/ |
| 18:40.31 | Kobaz | well |
| 18:40.31 | ThoMe | purged |
| 18:40.33 | ThoMe | :-) |
| 18:40.34 | Kobaz | umm |
| 18:40.41 | Kobaz | you didn't want your configs? |
| 18:40.50 | ThoMe | copy before ;) |
| 18:40.55 | Kobaz | heh |
| 18:41.06 | Kobaz | yeah get the tarball and start playing |
| 18:41.20 | ThoMe | which version is stable? |
| 18:41.22 | Kobaz | does capi use zaptel? i'm not familiar with it |
| 18:41.23 | ThoMe | or good? |
| 18:41.29 | ThoMe | hmm. i dont know |
| 18:41.33 | ThoMe | i have only use misdn |
| 18:41.39 | Kobaz | we've been using 1.4.14 for a while |
| 18:41.50 | ThoMe | is it ok ? http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.18.1.tar.gz ? |
| 18:41.51 | Kobaz | we only move on once we've been testing for a long time |
| 18:41.53 | ThoMe | hm |
| 18:42.11 | ThoMe | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.14.tar.gz yeah? |
| 18:42.14 | lmadsen | Kobaz: thank god someone tests before upgrading |
| 18:42.23 | Kobaz | but try different versions of asterisk, that's an easy way to attempt to elimitate sources of problems |
| 18:42.32 | lmadsen | so many people I see lately seem to use their production machines for test beds because they won't setup a sandbox |
| 18:42.33 | Kobaz | lmadsen: hah |
| 18:43.02 | ManxPower | lmadsen: isn't that what Digium wants? |
| 18:43.03 | ThoMe | so. ok. downloading |
| 18:43.05 | Kobaz | lmadsen: i would prefer not to find out code has to be rewritten due to depricated functions on a prodution box, so yeah, we wait |
| 18:43.08 | ThoMe | i must now to cook |
| 18:43.13 | lmadsen | ManxPower: no idea |
| 18:43.23 | ManxPower | Kobaz: you would find out that info in upgrade.txt |
| 18:43.44 | Kobaz | well yeah that too but, i meant like |
| 18:43.49 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 18:43.53 | Kobaz | not blindly just throwing in a new version |
| 18:44.02 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 18:44.39 | Kobaz | ThoMe: if multiple asterisk versions still crash, then perhaps it's the isdn driver |
| 18:45.13 | Kobaz | ThoMe: if .14 doesn't bomb out, then you've found a regression |
| 18:45.17 | ThatKidKel | would anyone know why a system that's timezone is GMT, and asterisk has been configured to output cdrs in GMT would have an that is GMT-5 |
| 18:45.21 | jjshoe | testing++ |
| 18:45.41 | ThatKidKel | .. is outputing cdr's GMT-5 |
| 18:46.42 | ThatKidKel | let me add a bit to this.. cdr/Master.csv is correct.. but my cdr-custom/Master.csv is incorrect |
| 18:46.43 | iamhrh | would anyone know how to maintain the callerid of the original incoming call through a transfer? I'm using poly ip650's, and whenever i try transfering it seems like I lose all that information :-/ |
| 18:48.20 | ManxPower | iamhrh: the answers you seek are in the output of "show application dial" Pay special attention to the "o" option. |
| 18:48.48 | iamhrh | ty very much, seems like no matter how much i read those options i can't keep them all straight! |
| 18:49.03 | ManxPower | you would only need that on SUPERVISED transfers, blind transfers should do that by default, IIRC |
| 18:57.10 | drmessano | Anyone know if there's a law involved in leaving 911 working on deactivated cell phones? |
| 18:57.19 | jer | i don't suppose anyone in here has ever set up sipx in sip.conf to connect to asterisk, or could gently push me in the right direction? (believe it or not, it is seemingly a difficult thing to google for) |
| 18:57.40 | denon | drmessano: yes, they all do .. but remember, the analog network is going away soon |
| 18:57.49 | denon | or has it already? dont recall |
| 18:57.53 | drmessano | The analog network is gone |
| 18:58.05 | denon | yeah, ok, so no 911 needs to work on those phones ;) |
| 18:58.26 | drmessano | That was only for analog? |
| 18:58.31 | denon | no .. |
| 18:58.33 | drmessano | Ok |
| 18:58.38 | drmessano | I read that as a double negative |
| 18:58.40 | denon | Im just saying, if you have some old POS analog phone, and you expect 911 to work |
| 18:58.41 | denon | it wont :) |
| 18:58.48 | drmessano | lol |
| 18:58.56 | drmessano | But it is indeed a law? |
| 18:59.01 | drmessano | So an expectation |
| 18:59.08 | denon | 99% sure it's law in the domestic US, yes |
| 18:59.15 | denon | I can't cite the code offhand .. |
| 18:59.18 | drmessano | I am 99% sure too.. just needed that 1% |
| 18:59.19 | drmessano | lol |
| 18:59.38 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 18:59.38 | denon | I'd stake your life on it .. |
| 18:59.42 | denon | I'm that confident |
| 18:59.42 | drmessano | HA |
| 19:00.07 | drmessano | My wife just got a new phone, and her old phone is sitting here.. I am gonna put another BT dongle on the * box here and let it sit there for 911 |
| 19:00.18 | s34n | Some kind soul has packaged up asterisk in the fedora repos |
| 19:00.44 | s34n | But the zaptel packages don't seem to include ztdummy |
| 19:01.09 | denon | drmessano: well, that's easy to verify .. |
| 19:01.13 | Kobaz | drmessano: it would be nifty to hook up to asterisk via bluetooth so you can set a pattern for 911 to just use that phone |
| 19:01.55 | denon | drmessano: remember though, that your cell phone record won't report exact address .. |
| 19:02.01 | jer | drmessano, i know for 100% certainty it's required in Canada |
| 19:02.04 | denon | you might want to call 911 and see if they can update it to be a fixed location |
| 19:02.09 | jer | and Canada always seems to be behind the USA in these matters |
| 19:02.14 | jjshoe | http://www.fcc.gov/cgb/consumerfacts/wireless911srvc.html |
| 19:02.48 | denon | that page doesn't really say .. |
| 19:02.51 | denon | was looking there a second ago |
| 19:03.10 | jjshoe | Basic 911 rules require wireless service providers to: |
| 19:03.10 | jjshoe | transmit all 911 calls to a Public Safety Answering Point (PSAP), regardless of whether the caller subscribes to the providers service or not. |
| 19:03.37 | drmessano | ok |
| 19:03.50 | drmessano | Kobaz: Thats what I am doing |
| 19:03.54 | Kobaz | drmessano: ooo |
| 19:04.39 | drmessano | If I can at least make a 911 call, then thats better than the nothing I already have |
| 19:04.40 | jjshoe | denon while it doesn't quote an exact fcc bylaw or some bullshit, that's enough to be %100 to me :) |
| 19:04.58 | drmessano | Dialing non emergency is great, but the Queue at the local PSAP is AWFUL |
| 19:05.08 | Yourname`` | What was that CLI filtering command?> |
| 19:05.23 | Bourrelle | On the RTP session objet, when calling the addDestination() fonction, should I give Remote_RTP_PORT or REMOTE_RTCP_Port wich is +1 ? |
| 19:05.40 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
| 19:05.47 | Jumpie | usb wireless sucksballs |
| 19:05.48 | Jumpie | grr |
| 19:06.34 | *** join/#asterisk ZPertee (n=ZPertee@24.106.241.121) |
| 19:06.37 | jjshoe | is chan_bluetooth any good? |
| 19:07.05 | *** join/#asterisk BrokenArrow (n=Lp@wikipedia/BrokenArrow) |
| 19:07.10 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:07.12 | drmessano | chan_mobile |
| 19:07.14 | drmessano | It works well |
| 19:07.15 | s34n | does anybody know anything about the fedora packaging of *? |
| 19:07.16 | bkruse | and yes |
| 19:07.33 | jjshoe | hrm I'll have to give it a go at some point. |
| 19:07.42 | tzafrir | s34n, not me, but try a more specific question |
| 19:08.24 | s34n | does anybody know whether the zaptel packaging in the main fedora repo should include ztdummy? |
| 19:08.45 | tzafrir | s34n, the zaptel packaging includes *only* the userspace utilities |
| 19:09.13 | s34n | tzafrir: where are the kernel modules? |
| 19:09.30 | tzafrir | Not really sure what they expect of the users to do |
| 19:10.47 | tzafrir | well, at least the Fedora Asterisk packages do include Asterisk modules... |
| 19:12.08 | s34n | tzafrir: so the kernel modules aren't packaged? |
| 19:12.39 | tzafrir | Right. But I'm not really familiar with it |
| 19:20.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:22.27 | [TK]D-Fender | s34n: AFAIK Zaptel modules have to be compiled to your specific kernel. If your kernel changes, then so do your Zaptel modules. I wouldn't think that you could package them in that case unless you were running in a controlled environment where you could expect to enforce version matching. |
| 19:22.36 | *** part/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 19:25.51 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 19:26.24 | [T]ank | anyone interested in buying a 4 port t1 card? |
| 19:26.38 | [T]ank | barely used. switched to ip and dont have use for it anymore. |
| 19:27.16 | *** join/#asterisk gutz|work (n=mark@gateway.meteor-web.com) |
| 19:27.19 | gutz|work | helo |
| 19:28.01 | *** join/#asterisk Teeli (n=tili@58.27.173.156.wateen.net) |
| 19:28.09 | gutz|work | does anyone know if the big regarding the out of sync voices with mix-monitor has been fixed in 1.4.19-rc4? |
| 19:29.25 | [TK]D-Fender | gutz|work: go read the changelog. |
| 19:29.55 | gutz|work | im having a hard time finding it |
| 19:30.02 | [T]ank | i know it works in 1.4.18.1. out of curiosity, why are you using 1.4.19-rc4? are you using it on a production server? |
| 19:30.51 | gutz|work | i was using 1.4.18 and the voices in mixmonitor were out of sync |
| 19:31.01 | [T]ank | probably not a bug then. |
| 19:31.35 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 19:32.03 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:32.05 | gutz|work | i was told in here it was a bug... which makes sense because there is no manual way of adjusting timing ousing mixmonitor |
| 19:32.58 | *** join/#asterisk Cle0 (n=cleo@adsl196-90-190-206-196.adsl196-6.iam.net.ma) |
| 19:33.34 | *** join/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 19:34.22 | *** join/#asterisk felix_da_catz (n=felix_da@66.60.231.164) |
| 19:39.26 | drmessano | ~me |
| 19:39.27 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 19:39.31 | [T]ank | i record hundreds of calls in 1.4.18 and have had no issue at all |
| 19:39.47 | drmessano | Might be a goblin |
| 19:40.11 | gutz|work | a gremlin? |
| 19:40.15 | drmessano | user error > goblin > bug > feature |
| 19:40.19 | [T]ank | troll? |
| 19:40.40 | gutz|work | well, im trying to understand what would cause these symptoms |
| 19:40.51 | jameswf | ~troll |
| 19:40.52 | jbot | rumour has it, troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or http://www.catb.org/~esr/jargon/html/entry/troll.html |
| 19:41.29 | gutz|work | ~me |
| 19:41.38 | gutz|work | :-/ nothing |
| 19:41.47 | jameswf | ~me |
| 19:41.47 | jbot | jameswf loves unsolicited technical support, or http://jameswf.info |
| 19:42.03 | gutz|work | ~bug |
| 19:42.04 | jbot | methinks bug is n: A son of a glitch. An error in design or programming in hardware or software. Effects range from cosmetic errors to system crash and loss of data. See also Feature. |
| 19:42.14 | gutz|work | ~feature |
| 19:42.15 | jbot | from memory, feature is A feature is something that a piece of hardware or software is designed to do. Many things that appear to be bugs are actually features. Often, a hardware or software developer will have to make a tradeoff in functionality that causes some undesirable effects. If they are aware of this and accept it, it is not a bug, but a feature. |
| 19:42.20 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 19:42.27 | gutz|work | ~user |
| 19:42.27 | jbot | it has been said that user is currently detached. Talk to this user upon their return. You will now be ignored. [HackFactor Elite 2.0], or a synonym for moron |
| 19:42.39 | gutz|work | ~jbot |
| 19:42.39 | jbot | well, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 19:43.29 | [T]ank | anyhow... if anyone is interested, i have a fairly new sangoma a104d i am looking to unload. anyone interested? |
| 19:43.40 | gutz|work | well, my question then is to ask what could cause mixmonitor to become out of sync? |
| 19:43.44 | [TK]D-Fender | [T]ank: 5$ |
| 19:43.54 | [T]ank | :-D |
| 19:44.18 | [T]ank | its nearly new, asking around $1200 |
| 19:44.20 | outtolunc | bids $5.53 |
| 19:44.34 | Kobaz | 6.02 |
| 19:45.19 | outtolunc | going once... |
| 19:45.25 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 19:45.49 | jameswf | .25 cents |
| 19:46.15 | *** part/#asterisk BrokenArrow (n=Lp@wikipedia/BrokenArrow) |
| 19:46.21 | outtolunc | kicks jameswf off the island |
| 19:46.25 | jameswf | pay .0. cent on the dollar $12.00 |
| 19:46.34 | jameswf | *.01 |
| 19:46.57 | jameswf | there is an island?? is it pen Island |
| 19:56.03 | *** join/#asterisk ManxPower (n=manxpowe@119.sub-75-201-31.myvzw.com) |
| 19:57.49 | Jumpie | lol |
| 19:57.55 | jasonwoot | thoughts on what would be the source of call interference that resembles a sound like a dot matrix printer? Only when someone is speaking... |
| 19:57.56 | Jumpie | jameswf have you been to pen island before? |
| 19:57.57 | Jumpie | :D |
| 20:00.59 | jameswf | lol no |
| 20:01.10 | ManxPower | jasonwoot: using SIPura equipment? |
| 20:01.32 | jasonwoot | I am |
| 20:01.40 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 20:01.47 | ManxPower | set the RTP packetsize to .20 instead of the default .30 |
| 20:01.58 | ManxPower | or ms per audio packet, or something like that. |
| 20:02.05 | ManxPower | I don't recall the exact phrasing |
| 20:02.46 | jasonwoot | gotta wait for this conf call to end, but I'll try that immediately |
| 20:03.57 | ManxPower | jasonwoot: you can check the value while you are on the conference call. |
| 20:04.06 | ManxPower | If it's already set to .20 then you have some other issue. |
| 20:08.49 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 20:15.07 | Jumpie | ok so..here is the question i keep gettin diff answers |
| 20:15.08 | *** join/#asterisk rvhi (n=chatzill@udp186710uds.hawaiiantel.net) |
| 20:15.14 | Jumpie | for low end, like 20 employees or less |
| 20:15.22 | Jumpie | wanting a voip solution and elminate the ass raping verizon charges for pots/pri |
| 20:15.26 | Jumpie | whats a good solution? |
| 20:15.32 | Jumpie | the digium apliance is what i was gearing toward |
| 20:15.43 | rvhi | hi, trixbox claimed that they are more stable, anyone has any real life experience? |
| 20:15.57 | Jumpie | im trying to go after all the 'small guys' the big companies dont wanna bother with, or if they do, they wanna sell them on a $200k cisco call manager thing |
| 20:15.59 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 20:16.31 | drmessano | Jumpie.. You're going with VoIP for all the wrong reasons |
| 20:16.52 | Jumpie | greater flexibility, lower cost, less complicated |
| 20:16.55 | Jumpie | how is that wrong reasons? |
| 20:17.00 | Jumpie | and "im" going for it to make money for my business |
| 20:17.22 | Jumpie | i want to get more experience before i approach big players with a proposal |
| 20:17.51 | drmessano | Going with an ITSP can be a bad move for a business.. you need to make sure you have the bandwidth, QoS, and that you are willing to accept that it's more of a point of failure than a traditional PSTN line |
| 20:18.03 | drmessano | Not just sticking it to the man |
| 20:18.05 | Jumpie | drmessano lets say i have the itsp portion in the bag |
| 20:18.08 | Jumpie | thats a non issue |
| 20:18.14 | Jumpie | all im concerne dwith is my ahrdware |
| 20:18.16 | drmessano | How so? |
| 20:18.23 | drmessano | How is that a non-issue? |
| 20:18.26 | Jumpie | because i know a director of ops for a very successful itsp |
| 20:18.30 | Jumpie | thats givin me a hookup |
| 20:18.41 | Jumpie | the pots gateway portion is all good |
| 20:18.50 | Jumpie | the itsp and all that is branded as my own |
| 20:18.52 | Jumpie | i invoiced the customer |
| 20:19.01 | Jumpie | i charge what i want on a plan thats flexible unlike most providers |
| 20:19.10 | Jumpie | i just want to get a fairly straightforward hardware package |
| 20:19.15 | drmessano | Great.. Fantastic.. Is he gonna QoS your network, guarantee low latency, and provide a backup for your internet connection? |
| 20:19.20 | drmessano | The ITSP is not the issue |
| 20:19.24 | Jumpie | i handle what he cant |
| 20:19.41 | Jumpie | his latency, qos, etc for the actual voice traffic is very good |
| 20:19.53 | Jumpie | i handle the data connection, and thats also taken care of |
| 20:20.07 | drmessano | I don't care about HIS connection |
| 20:20.11 | Jumpie | just because im a voip noob doesnt mean im a networking noob man |
| 20:20.25 | drmessano | You <---?????????????? ZOMG INTERNET ????????????--> Him is the issue |
| 20:20.32 | Jumpie | i know this |
| 20:20.35 | Jumpie | lol |
| 20:20.41 | drmessano | yet youre ignoring it |
| 20:20.43 | Jumpie | im going with zomg internet for my data |
| 20:20.47 | Jumpie | lol jk |
| 20:20.53 | Jumpie | trust me...im not ignoring |
| 20:20.55 | Jumpie | its just not my question |
| 20:21.00 | Jumpie | all im concerned with is the hardware |
| 20:21.03 | errr | anyone looking for a good deal on some sangoma a101 cards we are selling ours on craigs list: http://sanantonio.craigslist.org/sys/625445342.html |
| 20:21.27 | Jumpie | drmessano i have a data and cabling services company |
| 20:21.38 | Jumpie | <PROTECTED> |
| 20:21.45 | Jumpie | i just am tryi nto evaluate diff hardware platforms |
| 20:22.21 | Jumpie | i understand your questoining though, but the other 'peices' are taken care of |
| 20:22.22 | *** join/#asterisk mastaofdisasta (n=david@200.31.124.190) |
| 20:22.48 | drmessano | You do realize this conversation has been had over 1000 times in here |
| 20:22.54 | drmessano | 1. My friend owns an ITSP |
| 20:23.03 | Jumpie | drmessano this is not a ma and pop thing |
| 20:23.06 | drmessano | 2. I am a VoIP newb, but a networking god |
| 20:23.12 | drmessano | or |
| 20:23.17 | drmessano | 2a. I used to work for a telco |
| 20:23.17 | Jumpie | this is a major itsp that provides somethin that only 2 othe rpeople in the whole country offer |
| 20:23.25 | Jumpie | drmessano ok so whast the issue then? |
| 20:23.40 | drmessano | 3. My network is fine, the ITSP is fine, QoS is not an issue.. I just need a recommendation for a server |
| 20:23.56 | Jumpie | i understand you guys know voip in and out but, things arent always as cut and dry as you may think |
| 20:24.09 | Jumpie | well, i guess thats me 1 2 3 |
| 20:24.15 | drmessano | A lot of folks would not recommend and ITSP to replace a hardcore PSTN connection |
| 20:24.19 | drmessano | an* |
| 20:24.45 | Jumpie | thats why you have a couple fallback pots lines |
| 20:24.50 | Jumpie | and a reliable data connection |
| 20:24.55 | Jumpie | and your itsp having a reliable connection |
| 20:25.02 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 20:25.09 | Jumpie | of course nothing is 'perfect' and if customers ralize their voice is contingent upon data being up, thats fine |
| 20:25.11 | drmessano | 4. I guess you have it all figured out then |
| 20:25.29 | Jumpie | eveyr client's need is different |
| 20:25.36 | Jumpie | al im asking was what was a decent package for hardware |
| 20:25.41 | Jumpie | let me handle the 'other peices' |
| 20:25.42 | drmessano | How many lines do you have currently? |
| 20:25.50 | Jumpie | you are asking me like its one thing, its not |
| 20:25.54 | Jumpie | im tryin to prepare a solution |
| 20:25.59 | Jumpie | one particular customer has 22 pots lines |
| 20:26.03 | Jumpie | is payin for $1400 mo |
| 20:26.09 | Jumpie | plus $200 on a long distance provider |
| 20:26.15 | Jumpie | his total usage is less than 2000 minutes a month |
| 20:26.27 | Jumpie | he doesnt even have a pbx |
| 20:26.29 | Jumpie | straight pots to co |
| 20:26.37 | drmessano | So cut back his lines to something sensible for concurrent |
| 20:26.38 | Jumpie | we spent 10 hours doing cross conneccts and cabling to get it |
| 20:26.50 | Jumpie | define cut? as in, less lines? he needs the numbers |
| 20:26.52 | Jumpie | its a law firm |
| 20:27.06 | [TK]D-Fender | Jumpie: Partial PRI |
| 20:27.22 | Jumpie | yea thats a possibility |
| 20:27.24 | talntid | ouch |
| 20:27.29 | drmessano | 22 lines? |
| 20:27.34 | Jumpie | verizon is, rediculous |
| 20:27.35 | talntid | i have 30,000 LD minutes per month, 24 lines |
| 20:27.38 | ManxPower | Jumpie: the problem is that the hardware varies depending on requirements/limitations |
| 20:27.40 | talntid | pay $1400/mo tiotal |
| 20:27.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:27.44 | Jumpie | they are losing money to pots service |
| 20:27.47 | Jumpie | which used to be their cash crop |
| 20:27.52 | Jumpie | its all goign to fios/tls |
| 20:27.56 | talntid | er, $2400/mo total. |
| 20:27.57 | Jumpie | so they rape customers till ignorant enough to by it |
| 20:28.06 | drmessano | Put in an asterisk box and a partial PRI |
| 20:28.06 | ManxPower | for 22 lines a partial PRI with however many DIDs you need |
| 20:28.07 | Jumpie | i get a good bonded t1, 3 mbit, setup the qos right to handle the voip |
| 20:28.16 | Jumpie | get 2 bakcup pots |
| 20:28.24 | Jumpie | he pays me 2 cents a minute and whatever i wanna charge |
| 20:28.42 | drmessano | Oh, so you're gonna be a CLEC as well? |
| 20:28.44 | Jumpie | ManxPower he doesnt have satellite offices |
| 20:28.48 | Jumpie | yes |
| 20:28.51 | ManxPower | Oh! You want to do VoiceOverIpOverGenericInternet? |
| 20:29.11 | Jumpie | you take more advantage of voip when you have many satellite offices to call i'd think, he doesnt |
| 20:29.23 | Jumpie | that partial pri, 22 lines, thats only 2 shy of a full |
| 20:29.25 | ManxPower | Jumpie: nothing in my statement should have implied that I thought he had satellite offices. |
| 20:29.32 | Jumpie | ManxPower no i know |
| 20:29.34 | Jumpie | i was just stating sorry |
| 20:29.49 | drmessano | Now I see what the issue is |
| 20:30.02 | ManxPower | Jumpie: So get a full PRI, but I suspect you will need FEWER channels, rather than more channels. |
| 20:30.03 | Jumpie | bototm line is he doesnt want to PAY for these pri |
| 20:30.26 | ManxPower | Jumpie: He doesn't want to pay for it? Best of luck - you are not doing anything close to what I recommend so I can't help you. |
| 20:30.28 | Jumpie | remember, pri you pay for the 'capability to make calls' plus the calls |
| 20:30.45 | drmessano | Don't you think you need to be a little more familiar with VoIP before you start acting as a CLEC? Maybe put it through some paces. |
| 20:30.53 | ManxPower | Jumpie: If you have the right carrier local calls are free |
| 20:31.07 | Jumpie | ok so a single pri...maybe whats that, $250 |
| 20:31.08 | Jumpie | give or take |
| 20:31.26 | ManxPower | Heck, on our PRI we get 23 channels and something like 30,000 mins of long distance |
| 20:31.54 | drmessano | If you're gonna put in a PBX, you can scrap a TON of those lines |
| 20:31.56 | Jumpie | ill have to look into that, my other partner handles the clec stuff |
| 20:32.03 | drmessano | You already said the usage was low |
| 20:32.05 | Jumpie | yeah |
| 20:32.09 | drmessano | So you're looking at even MORE savings |
| 20:32.09 | Jumpie | but i know they need the 22 numbers |
| 20:32.16 | Jumpie | i realize thats not lines |
| 20:32.17 | drmessano | That's insane |
| 20:32.24 | ManxPower | On a PRI you DO NOT need the same number of channels as you have numbers |
| 20:32.27 | Jumpie | i know |
| 20:32.37 | jameswf | bolocks |
| 20:32.40 | ManxPower | We have 100 numbers on each of our PRIs, even if there's only 8 channels on it. |
| 20:32.50 | jameswf | ~callbsonthat |
| 20:32.50 | jbot | OMFG NFW TSBS ICBSOT BBQ VOTE FOR RON PAUL |
| 20:32.56 | Jumpie | client already said he can deal with 10 channels |
| 20:33.08 | Jumpie | 10 concurrent calls is what he could handle |
| 20:33.31 | drmessano | Ok, so thats even more savings, and less reason to go with VoIPOverGenericZOMGTORRENTTHATInternet |
| 20:33.35 | ManxPower | Jumpie: I suggest the number of channels that can handle calls most of the time, then route to a VoIP carrier when you care close to being out of channels. |
| 20:33.45 | *** join/#asterisk SteveTotaro (n=Administ@96.234.221.143) |
| 20:34.01 | Jumpie | ManxPower yea..i suppose i had considered that |
| 20:34.13 | Jumpie | i was figuring a good connection, with a close hop to my itsp, very reliable both ends |
| 20:34.15 | Jumpie | he was cool with it |
| 20:34.19 | drmessano | You can phase them into a PBX with far less lines and overhead and they will see HUGE savings |
| 20:34.20 | Jumpie | and he would pay around $250 a month total |
| 20:34.21 | ManxPower | Jumpie: Ot |
| 20:34.23 | Jumpie | for all costs |
| 20:34.23 | drmessano | Forget the ITSP |
| 20:34.32 | ManxPower | Jumpie: It will be reliable until you NEED it to be reliable. |
| 20:34.48 | Jumpie | so what is this then? lack of faith in my itsp? |
| 20:34.49 | *** part/#asterisk mastaofdisasta (n=david@200.31.124.190) |
| 20:35.03 | drmessano | i explained it twice |
| 20:35.09 | ManxPower | Jumpie: Do you have a direct connection to your ITSP? |
| 20:35.15 | drmessano | VoIPOverInternet blows |
| 20:35.22 | drmessano | It "works" |
| 20:35.24 | drmessano | Wait |
| 20:35.28 | drmessano | It works* |
| 20:35.31 | drmessano | * see fine print |
| 20:35.34 | Jumpie | no tif you implement it right and know what you're doing |
| 20:35.36 | ManxPower | Jumpie: I have no faith in the internet at all. |
| 20:35.45 | Jumpie | itsp is on same backbone |
| 20:35.50 | Jumpie | i suppose... |
| 20:35.51 | Jumpie | if verizon fails |
| 20:35.53 | ManxPower | Jumpie: that's crap. 90% of internet problems are inter ISP issues. |
| 20:35.53 | Jumpie | in general |
| 20:35.54 | Jumpie | we're screwed |
| 20:36.06 | drmessano | Jumpie: You're making a bad argument, especially with ManxPower, who knows that sucks |
| 20:36.27 | ManxPower | Jumpie: traceroute to the ITSP, each hop is a point of failure |
| 20:36.33 | Jumpie | i realize that |
| 20:36.55 | Jumpie | again, i said though its all on same backbone, a backbone im failiar with, and would take a total failure to really mess up |
| 20:37.02 | Jumpie | thats the ONLY reason i considered this |
| 20:37.06 | Jumpie | i even pitched the PRI suggestion |
| 20:37.08 | Jumpie | he said he didnt want it |
| 20:37.09 | ManxPower | Jumpie: go for it then |
| 20:37.14 | Jumpie | however, i can look at it again and see the cost comparison |
| 20:37.26 | ManxPower | When you have an outage, don't come here asking |
| 20:37.32 | Jumpie | doing data only with voip over internet maybe a good at first, because mayb ehe can SEE if it sucks |
| 20:37.34 | Jumpie | and then upgrade |
| 20:37.55 | Jumpie | ManxPower im not like....trying to disagree, its just ihave to go within the confines of my clients requirements or requests |
| 20:37.56 | drmessano | 5. When it fails, Google for the logs.. This conversation has been assigned ticket number 61043 |
| 20:37.59 | ManxPower | Jumpie: Trust me, it will be your fault if it breaks |
| 20:38.05 | drmessano | Please save that for reference |
| 20:38.34 | ManxPower | Jumpie: At this point I don't accept stupid clients -- I guess I'm lucky in that way |
| 20:38.41 | Jumpie | i cant be responsible for interhop outages |
| 20:38.45 | Jumpie | and he will also sign such an agreement |
| 20:38.48 | Jumpie | tuff shit fo rhim |
| 20:39.25 | Jumpie | if i can get a partial pri for him, can it assign the correct caller id outbound on whatever channel eh's going? |
| 20:39.27 | Jumpie | thats whats so important |
| 20:39.28 | drmessano | Jumpie: Your client expects and trusts you to supply them with a system that meets their requirements, and YOURS.. and it doesn't sound like you're serving them well with the attitude of "Well, thats what he wants" |
| 20:39.57 | Jumpie | he wants, least possible cost |
| 20:40.02 | Jumpie | bottom line |
| 20:40.11 | drmessano | Frankly, it sounds to me like you're seeing $$$ with this CLEC thing.. and you want your share of his calls |
| 20:40.24 | Jumpie | drmessano if i make anything it'll be like 1/10th of a cent |
| 20:40.26 | Jumpie | nothin to jump over |
| 20:40.33 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 20:40.39 | Jumpie | IF i can get him a pri line with free local |
| 20:40.44 | Jumpie | then yes, i would consider that |
| 20:41.03 | Jumpie | he's concerned with his caller id showing right number to his clients |
| 20:41.12 | Jumpie | if they have a zillion numbers and 12 channels for example |
| 20:41.15 | ManxPower | Jumpie: many CLECs give you expanded local. Our CLEC considers any call to Louisiana or Missippi to be "local" |
| 20:41.16 | drmessano | [16:28] <Jumpie> he pays me 2 cents a minute and whatever i wanna charge <-- O RLY |
| 20:41.28 | Jumpie | drmessano i was referring to a small fla tmonthly fee |
| 20:41.32 | Jumpie | not mark up the per minute |
| 20:41.33 | ManxPower | Jumpie: if your carrier lets you set the callerid, it's a non-issue |
| 20:41.59 | Jumpie | ManxPower ok so we go back to initial question |
| 20:42.02 | Jumpie | partial or full pri |
| 20:42.10 | ManxPower | Jumpie: partial |
| 20:42.17 | Jumpie | rioght, im sayin either or |
| 20:42.19 | Jumpie | hardware to take it |
| 20:42.28 | Jumpie | if i have lets say, 12 channels |
| 20:42.37 | ManxPower | especially if you can failover to an ITSP if you run out of channels |
| 20:42.37 | Jumpie | i thought the CID was based on the # assigned to that channel or something |
| 20:42.44 | ManxPower | Jumpie: stop thinking analog |
| 20:42.52 | jjshoe | ManxPower++; |
| 20:43.01 | ManxPower | In PRI NO CHANNEL has a number assigned to it. |
| 20:43.09 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 20:43.32 | Jumpie | hmm, the billing breakdown kinda looks like thats how it is |
| 20:43.36 | Jumpie | confusing as crap |
| 20:43.45 | ManxPower | Jumpie: you have a bill for a PRI? |
| 20:43.45 | jjshoe | Jumpie ignore the billing department, they arn't smart ;) |
| 20:43.50 | ManxPower | Not T-1, PRI. |
| 20:43.54 | Jumpie | yea |
| 20:43.59 | Jumpie | custoflex? |
| 20:44.00 | Jumpie | o osmethin |
| 20:44.06 | ManxPower | A channelized voice T-1 basically acts like a bunch of analog lines |
| 20:44.13 | Jumpie | no..i know |
| 20:44.15 | Jumpie | he was payin for 3 pri |
| 20:44.19 | Jumpie | well anothe rcustomer |
| 20:44.21 | Jumpie | but same concept |
| 20:44.29 | Jumpie | jjshoe heh yeah |
| 20:44.37 | ManxPower | Jumpie: chances are that bill is showing the callerid of the source of the call |
| 20:44.47 | ManxPower | Jumpie: MANY things depend on the CLEC |
| 20:44.52 | Jumpie | ok so itsp for when channels are used up, or maybe for LD? |
| 20:44.57 | Jumpie | all others carry pri for local |
| 20:45.02 | Jumpie | thats what you recommend? |
| 20:45.15 | ManxPower | Jumpie: I would say route to ITSP when 80% of your channels are used. |
| 20:45.27 | ManxPower | regardless of if it's local or not. |
| 20:45.33 | Jumpie | well..yea of course |
| 20:45.39 | Jumpie | but im saying, depending on the range of local |
| 20:45.49 | Jumpie | i can setup what to use itsp on specific calls |
| 20:45.55 | Jumpie | like i fim in california, to call east coast |
| 20:45.56 | Jumpie | obviously |
| 20:46.00 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 20:46.06 | ManxPower | Jumpie: you have to ask the customer when making local calls would you rather have the call fail or work and be billed per min |
| 20:46.38 | jjshoe | which is an obvious question for any serious buisness :P |
| 20:46.38 | Jumpie | so are you saying most clients who make voip calls to other offices |
| 20:46.42 | Jumpie | have dedicated connections? |
| 20:46.51 | drmessano | and your failover to the ITSP is going to be OUTBOUND |
| 20:46.51 | Jumpie | i guess i always assumed it was largely done over the internet |
| 20:46.53 | Jumpie | on a reliable pipe |
| 20:46.59 | Jumpie | right |
| 20:47.16 | Jumpie | but i was thinking like you said 80% channel usage AND interstate LD outbound? |
| 20:47.20 | jjshoe | Jumpie reliable piper over the internet? what? |
| 20:47.21 | ManxPower | Jumpie: Not a single one of our interoffice calls go over the Internet, they frequently go over point to point data T-1s, but that is all on the corporate WAN |
| 20:47.37 | Jumpie | yeah |
| 20:47.41 | *** join/#asterisk adjohn (n=adjohn@68-248-62-131.ded.ameritech.net) |
| 20:47.45 | ManxPower | Jumpie: any company of any size does not run interoffice calls over the internet. |
| 20:47.45 | Jumpie | thats initially expensive :D |
| 20:47.51 | Jumpie | shit |
| 20:48.11 | Jumpie | yea i know voip is alot more sensitive to issues than other |
| 20:48.12 | ManxPower | I would have said "most companies" rather than "any company" |
| 20:48.43 | ManxPower | Jumpie: but I've not heard about other offices for this client, in fact, I thought you said there are no other offices for this client |
| 20:49.19 | Jumpie | for this one...correct |
| 20:49.23 | ManxPower | Jumpie: not expensive, as we already had the lines in place for data |
| 20:49.28 | Jumpie | right |
| 20:49.34 | Jumpie | the client i know of with 3 offices, does not |
| 20:49.38 | drmessano | ManxPower: Surely if you use IAX, this will all work |
| 20:49.44 | drmessano | ducks |
| 20:49.57 | ManxPower | *** drmessano is now on IGNORE List. |
| 20:49.59 | denon | as I understand it, IAX2 works fine with up to 6 hours of jitter |
| 20:50.06 | Jumpie | ManxPower ok so my concern is the cid then |
| 20:50.07 | drmessano | HAHAH!!!!!!!! |
| 20:50.16 | denon | anything over 6 hours .. is touch and go |
| 20:50.26 | ManxPower | Jumpie: and the answer to the question is "ask your carrier" |
| 20:50.27 | drmessano | denon just asterolled me there |
| 20:50.36 | Jumpie | but how much of it is setup on the ip pbx |
| 20:50.37 | denon | snickers |
| 20:50.55 | Jumpie | i also want certian inbound numbers to wrong 2 extensions for example |
| 20:50.57 | Jumpie | stuff like that |
| 20:50.59 | Jumpie | er ring |
| 20:51.07 | jjshoe | Jumpie ask your carrier if they allow you to set caller-id. |
| 20:51.20 | Jumpie | ok |
| 20:51.25 | jjshoe | Jumpie making multiple inbound did's ring the same phone is easy as long as they pass the dialed digits. |
| 20:51.34 | Jumpie | i also know there are ways to force a trunk |
| 20:51.46 | Jumpie | like you can setup dialing 7,1, forces sip |
| 20:51.55 | jjshoe | of course. |
| 20:51.56 | Jumpie | 9, 1 forces pots, 8, 1 forces pots, then sip if pots used, etc |
| 20:52.09 | jjshoe | asterisk 101 |
| 20:52.11 | Jumpie | right |
| 20:52.16 | Jumpie | ok..so back to the beginning an dill shaddup |
| 20:52.20 | Jumpie | if i do the partial pri setup |
| 20:52.21 | drmessano | It sounds to me like you could make your client very happy be reevaluating their needs with a partial PRI and installing a PBX for them.. I would leave it there before you get into something less reliable. Make the sale, make them happy.. |
| 20:52.32 | Jumpie | what do you recommend then? |
| 20:52.34 | Jumpie | for hardware |
| 20:52.41 | Jumpie | would the lil digium appliance suffice? |
| 20:52.56 | ManxPower | Jumpie: Last I heard Digium Appliance did not support T-1 or PRI |
| 20:53.02 | ManxPower | I also heard they were changing it. |
| 20:53.03 | drmessano | Not sure, I can't testify to the effectiveness of one.. |
| 20:53.03 | Jumpie | hmm |
| 20:53.06 | Jumpie | i thought they did |
| 20:53.10 | Jumpie | ok what then do you KNOW does? |
| 20:53.11 | [hC] | They do not currently. |
| 20:53.17 | ManxPower | Jumpie: I can't really say anything about non-Asterisk systems |
| 20:53.18 | Jumpie | is that what the switcxvox platform is about |
| 20:53.35 | [hC] | Switchvox's stuff does support PRI, as does any other asterisk box you build yourself. |
| 20:53.35 | Jumpie | non asterisk? it is asterisk |
| 20:53.38 | jjshoe | Jumpie yes, the switchvox is an ip-pbx appliance. |
| 20:53.39 | ManxPower | Jumpie: Digium or Sangoma T-1/E-1 card is what you will NEED |
| 20:53.44 | Jumpie | ok |
| 20:54.09 | denon | sangoma? c'mon manx, you may not like the guy, but no need to be cruel ... |
| 20:54.11 | denon | :) |
| 20:54.19 | ManxPower | Jumpie: It claims it's Asterisk, but I doubt even one asterisk expert here could tell you how to setup and configure the Digium appliance -- to me that means "it's not asterisk" |
| 20:54.32 | Jumpie | ya..i have the asterisknow setup which i beleive it runs |
| 20:54.33 | ManxPower | denon: be glad I still include Digium in my recommendations. |
| 20:54.39 | Jumpie | i'd rather get used to what the majority uses |
| 20:54.48 | lirakis | is away (leaving..."the internets" are safe ... for now) |
| 20:54.54 | rupa | the majority HERE uses straight asterisk |
| 20:54.55 | Jumpie | ManxPower so what do you recommend? |
| 20:54.58 | Jumpie | i see |
| 20:55.00 | Jumpie | so basically |
| 20:55.02 | Jumpie | nix box |
| 20:55.05 | Jumpie | setup with old school version |
| 20:55.07 | jjshoe | Jumpie there are many many options, you need to research :) |
| 20:55.13 | ManxPower | ManxPower: Jumpie: Digium or Sangoma T-1/E-1 card is what you will NEED |
| 20:55.15 | Jumpie | shit you know, i had just setup an ubuntu box |
| 20:55.17 | Jumpie | for this very reason |
| 20:55.20 | Jumpie | and asterisknow overwrote it all |
| 20:55.33 | ManxPower | Jumpie: All the guis do that |
| 20:55.42 | ManxPower | the guis want THEIR config files. |
| 20:55.46 | Jumpie | yeah |
| 20:55.47 | Jumpie | bleh |
| 20:55.49 | Jumpie | wate of time... |
| 20:55.56 | Jumpie | so , for testing in my house in the meantime |
| 20:56.01 | Jumpie | im gonna put ubuntu back on |
| 20:56.02 | jjshoe | it's a trade off, learn all the config options, or let the guis do it for you *shrug* |
| 20:56.11 | Jumpie | jjshoe i for the former :) |
| 20:56.16 | Jumpie | plus you can customize |
| 20:56.22 | ManxPower | jjshoe: the other tradeoff is not getting any support for GUIs |
| 20:56.25 | jjshoe | Jumpie as long as you can get your company out of a jam in an emergency situation |
| 20:56.35 | Jumpie | jjshoe thi sis why i want to be very familiar with this stuff |
| 20:56.47 | Jumpie | but i was also hoping on having a simple method to train a non IT guy on how to do simple edits/changes |
| 20:56.52 | Jumpie | if its command line/config stuff, thats outta the question |
| 20:56.52 | ManxPower | If the GUIs had decent support why do people keep coming here whining about it? |
| 20:56.58 | Jumpie | that was the only reason why i was wanting to go gui initially |
| 20:57.05 | jjshoe | ManxPower dunno, which guis get whined about? |
| 20:57.13 | ManxPower | Jumpie: use whatever you want |
| 20:57.20 | Jumpie | doesnt even vanilla asterisk have a simple gui for 'non it employees' to make changes? or no |
| 20:57.23 | ManxPower | jjshoe: trixbox, amp, freepbs, digium gui |
| 20:57.25 | Jumpie | i myself, i dont care, i can learn it |
| 20:57.37 | ManxPower | they all say "nobody is helping on #whateverguichanneltheyareon" |
| 20:57.43 | jjshoe | ManxPower see all the major ones get whined about ;) no shock.. |
| 20:57.43 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-192-102.dsl.scarlet.be) |
| 20:57.55 | jjshoe | ManxPower everybody wants something for nothing.. |
| 20:58.06 | Jumpie | so basically i have to work in a suport contract for any edits |
| 20:58.15 | Jumpie | as long as i can remotely do it :) |
| 20:58.23 | ManxPower | Jumpie: Asterisk is not a PBX, it's a toolkit that lets you design your own PBX |
| 20:58.29 | drmessano | Jumpie: I heard trixbox (tm) is nice.. Maybe someone can code a CLEC module for the GUI |
| 20:58.40 | ManxPower | Jumpie: USE A GUI if you need it, but just don't expect us to help you with it. |
| 20:58.44 | denon | maybe someone can code a firewall for it.. |
| 20:58.50 | Jumpie | ManxPower no.i understand that non gui is better |
| 20:58.51 | denon | outbound fw, that is |
| 20:58.58 | Jumpie | i just thought there was a quick changes gui for small fixes |
| 20:59.01 | Jumpie | to supplmement |
| 20:59.04 | ManxPower | Expect people that support that GUI to help you in the correct channels and correct "forums" |
| 20:59.16 | Jumpie | ya #asterisknow has been kinda quiet :) |
| 20:59.24 | denon | ironic |
| 20:59.34 | Jumpie | i heard trixbox requires centralized management |
| 20:59.38 | Jumpie | which i dont want |
| 20:59.41 | ManxPower | Jumpie: I can't REALLY say non-GUI asterisk is better then GUI Asterisk -- just that the SUPPORT is better. |
| 20:59.54 | denon | Jumpie: I wouldn't call it management, it just calls home with all your data |
| 21:00.05 | Jumpie | ManxPower but if it gets to the point i realy learn the configs and want to make customized changes, if i lose that in gui |
| 21:00.07 | Jumpie | then thats definately a nogo |
| 21:00.14 | jjshoe | Jumpie depends on the version of trixbox. |
| 21:00.21 | Jumpie | denon i was told that all of trixbox connects to a central server |
| 21:00.23 | ManxPower | Jumpie: no, you just have to learn how the gui requires you to do your customization |
| 21:00.27 | Jumpie | and you have to connect to that, cant dirctly manage |
| 21:00.30 | drmessano | FreePBX is well supported, but you need to REALLY.. REALLY understand what it WILL and WONT do.. There will be some things in the ~book that you won't be able to do. |
| 21:00.48 | ManxPower | trixbox, for example requires your changes to be in specific file names, not the default file names you would use on a non-gui |
| 21:01.15 | Jumpie | well i have the book qwell told me to get, asterisk the future of telephony |
| 21:01.16 | Jumpie | :D |
| 21:01.18 | ManxPower | like extensions_conf_addtional or something like that |
| 21:01.23 | Jumpie | i want whatever i can follow along with in that book |
| 21:01.24 | ChkDigit | Who's documented a filter to make a coworkers voice sound like a chipmunk for April Fools Day? |
| 21:01.32 | Jumpie | ChkDigit hhe |
| 21:01.35 | jjshoe | ChkDigit dear god that would rock. |
| 21:02.00 | denon | I thought we were going to write all the dialplans with s,1,Hangup() for april fools |
| 21:02.13 | denon | or zapteller or whatever it's called |
| 21:02.14 | Jumpie | ManxPower in your professional opinion |
| 21:02.17 | Jumpie | what would you recmmend |
| 21:02.21 | Jumpie | if i have a ubuntu box ready to go |
| 21:02.24 | ManxPower | denon: that only works for analog fxo! |
| 21:02.27 | drmessano | Oh god |
| 21:02.27 | Jumpie | at least for 'my testing/hobby' at home to learn |
| 21:02.28 | denon | zapateller |
| 21:02.30 | drmessano | You said Ubuntu |
| 21:02.35 | drmessano | I need some red label |
| 21:02.36 | Jumpie | whats wrong with ubuntu |
| 21:02.50 | denon | ManxPower: zapateller you mean? |
| 21:02.56 | ManxPower | Jumpie: my opinion has nothing whatsoever to do with what Trixbox does and does not require you to do for customization |
| 21:03.04 | drmessano | I've said enough for today.. |
| 21:03.06 | ManxPower | denon: exten => s |
| 21:03.09 | jameswf | ~ubuntu |
| 21:03.09 | denon | ManxPower: why wouldn't it work for fxs? |
| 21:03.10 | denon | oh yeah |
| 21:03.21 | ManxPower | denon: well if you had immediate=yes, I guess it woulf |
| 21:03.26 | Jumpie | ManxPower hmm well i just wanna start learning |
| 21:03.27 | Jumpie | vanilla |
| 21:03.34 | Jumpie | i kinda dont know where to start |
| 21:03.44 | Jumpie | everyone throwing different possibilites at me :) |
| 21:04.01 | jameswf | ~ubuntu is <reply> Ubuntu- It is like Debian except it just works... |
| 21:04.02 | jbot | ACTION lovingly explains to is <reply> Ubuntu- It is like Debian except it just works... in a way that causes is <reply> Ubuntu- It is like Debian except it just works... to weep with gratitude that is <reply> Ubuntu- It is like Debian except it just works... must read the fine, friendly manual |
| 21:04.11 | jameswf | bah |
| 21:04.13 | ManxPower | Jumpie: Look straight up. You are looking at the Asterisk learning curve |
| 21:04.29 | Jumpie | ManxPower i can learn what i have to learn, i just need a starting point |
| 21:04.31 | Jumpie | i dont care how hard it is |
| 21:04.44 | [hC] | i presume someone has already done this |
| 21:04.45 | [hC] | ~book |
| 21:04.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:04.45 | jameswf | the book is an excelent starting point |
| 21:04.45 | denon | then open up extensions.conf.sample |
| 21:04.47 | denon | and the wiki |
| 21:04.48 | jameswf | ~buybook |
| 21:04.49 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 21:04.49 | ManxPower | Look straight down, that's where you're going when you die if you use a GUI 8-) |
| 21:04.51 | denon | ..or the books, I guess |
| 21:04.53 | jameswf | bah |
| 21:04.56 | Jumpie | im lookin at that book now :D |
| 21:05.00 | drmessano | Read the BOOK |
| 21:05.03 | Jumpie | im talking about the ACTUAL product to put on the server |
| 21:05.10 | jameswf | ~jinx |
| 21:05.11 | Jumpie | digium based, trixbox, freepbx, what |
| 21:05.13 | denon | check it out of svn then |
| 21:05.16 | bipolar | Jumpie: get Trixbox. figure out how it works in the web gui, then learn whats going on behind the interface. |
| 21:05.18 | drmessano | Then you've not read anything |
| 21:05.34 | jameswf | Jumpie: use Gentoo |
| 21:05.35 | bipolar | Jumpie: that will help smooth out the learning curve a bit |
| 21:05.36 | denon | dont screw with trixbox or other GUIs .. honestly, they just make it harder to get stuff done |
| 21:05.49 | outtolunc | shouldn't #trixbox be doing the pre-sales support also? <G> |
| 21:05.51 | jameswf | ~trixbox |
| 21:05.52 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 21:06.06 | Jumpie | and there you go , more confused |
| 21:06.07 | Jumpie | lol |
| 21:06.18 | drmessano | Jumpie: If you learn asterisk, you're making the ACTUAL PRODUCT |
| 21:06.31 | jjshoe | Jumpie you should 1) take everyones opinion with a grain of salt while 2) you go research what will work best for you. |
| 21:06.31 | drmessano | Thats like saying "I want to learn apache.. So which apache appliance should I get" |
| 21:06.33 | Jumpie | yes but i need a start, a distro |
| 21:06.35 | rupa | wonders how long this conversation will last |
| 21:06.35 | drmessano | No, no, no, FAIL |
| 21:06.42 | jjshoe | rupa yes. |
| 21:06.43 | Jumpie | explodes |
| 21:06.45 | ManxPower | Jumpie: I can tell you defiantly what distro to use with Asterisk. |
| 21:06.49 | denon | Jumpie: distro doesn't matter .. whatever you like. Lots of us like Debian |
| 21:06.56 | Jumpie | ManxPower i guess its not ubuntu :) |
| 21:07.06 | [hC] | why dont you just go download the trixbox bootable ISO? |
| 21:07.08 | jameswf | you should get linux from scratch and build off that... a 15Meg asterisk distro.... then you wil be 1337 |
| 21:07.08 | ManxPower | Jumpie: It is whatever distro you are MOST FAMILIAR WITH. |
| 21:07.10 | bipolar | Jumpie: If you want a distro you can install and run, look at trixbox |
| 21:07.11 | [hC] | thats going to be your easiest onramp |
| 21:07.13 | Jumpie | ManxPower which is ubuntu |
| 21:07.22 | ManxPower | Jumpie: then that is what you should use. |
| 21:07.23 | drmessano | Forget Trixbox |
| 21:07.25 | Jumpie | ok |
| 21:07.26 | denon | so go with ubuntu |
| 21:07.34 | bipolar | Jumpie: it's based on CentOS (Redhat) |
| 21:07.37 | jameswf | ubuntu server not desktop |
| 21:07.37 | denon | now, install SVN |
| 21:07.38 | ManxPower | Jumpie: there is not really significant differences with regards to Asterisk |
| 21:07.39 | Jumpie | but trixbox means no ubuntu |
| 21:07.43 | denon | svn checkout asterisl .. |
| 21:07.53 | drmessano | trixbox means you get a box that phones home and looks like alien vomit |
| 21:07.54 | denon | Jumpie: forget trixbox. pretend it doesn't exist |
| 21:07.56 | jjshoe | watches 50 people try to yell over each other |
| 21:08.15 | Jumpie | ok, and if ihave my distro im most familiar with |
| 21:08.18 | Jumpie | then what do i download? |
| 21:08.21 | rupa | tosses in a tin can + string |
| 21:08.22 | jameswf | Elastix ix cool if you need a premade dealio |
| 21:08.28 | [hC] | you guys are amusing. you're suggesting that someone who asks the question 'what distro do i need' should be doing this from scratch instead of using trixbox? |
| 21:08.29 | denon | Jumpie: asterisk source, from SVN |
| 21:08.36 | [hC] | its no wonder this channel is filled with 98% noob. |
| 21:08.46 | denon | [hC]: he wants to learn asterisk .. you won't learn asterisk by downloading trix |
| 21:08.48 | jameswf | Doesnt the book have an installation section lmadsen |
| 21:08.50 | Jumpie | hc well alot of diff experiences and opinions i guess |
| 21:09.06 | Jumpie | i want to be asterisk g0d |
| 21:09.11 | drmessano | I'm fairly certain this channel is NOT for GUI users.. theres other channels for that |
| 21:09.16 | bipolar | Jumpie: You can even start with the Trixbox Vmware image, and use it with Vmware Player. |
| 21:09.20 | lmadsen | jameswf: yep, a whole chapter |
| 21:09.24 | ManxPower | Jumpie: [tk]Fender and Qwell are both Asterisk gods. |
| 21:09.31 | [hC] | fail. |
| 21:09.31 | Jumpie | yes...i must aspire to that |
| 21:09.36 | ManxPower | Some people might say I am as well, but if so I'd be a vengeful one. |
| 21:09.41 | jameswf | see Jumpie there is a whole install chapter in the book |
| 21:09.43 | bipolar | Jumpie: that should help you get your feet wet. |
| 21:10.05 | jameswf | ~nowwhat |
| 21:10.06 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 21:10.18 | Jumpie | but if the gui is so strict and impossible to customize then how ma i learning? how is it any different than the asterisk now i have now? |
| 21:10.29 | denon | Jumpie: install trixbox, then once you totally understand how tb works, throw that all away, and learn asterisk .. </sarcasm> |
| 21:10.37 | file | oh denon... |
| 21:10.37 | Jumpie | lol yea thats how it seems |
| 21:10.46 | jameswf | guis arent strict if you do it right |
| 21:10.47 | file | you are such the silly |
| 21:10.57 | denon | file, you're such the muffin |
| 21:11.01 | ManxPower | Jumpie: I don't think anyone here meant to imply you can use Trixbox to learn Asterisk |
| 21:11.09 | drmessano | I thought Asterisk was coded to be a backend for Trixbox? |
| 21:11.10 | file | I can't argue with that |
| 21:11.12 | drmessano | ducks |
| 21:11.18 | Jumpie | well i'd rather eat the bullet and go right hte first time |
| 21:11.18 | Nugget | ./configure --with-muffins |
| 21:11.22 | Jumpie | i'd rather just learn it right |
| 21:11.23 | ManxPower | That's like saying you can learn BASIC to learn programming. |
| 21:11.25 | lmadsen | drmessano: lol |
| 21:11.26 | Katty | mmm, muffins |
| 21:11.28 | Jumpie | heh |
| 21:11.47 | Yourname`` | G729 is what, 5bucks per channel? |
| 21:11.48 | file | Trixbox is a group of programs don't forget... the GUI being FreePBX... give the FreePBX folk some credit... |
| 21:11.50 | outtolunc | sneezes... aww you stepped in my GUI |
| 21:11.53 | file | Yourname``: $10 |
| 21:11.55 | file | tickles Katty |
| 21:11.56 | Yourname`` | Per month? |
| 21:12.01 | file | Yourname``: one time fee. |
| 21:12.04 | Katty | asplodes |
| 21:12.05 | Jumpie | maybe im confused as the actual 'package' to get or put on a cd |
| 21:12.15 | jameswf | gave my credit card |
| 21:12.16 | drmessano | Like, one day, Fonality was looking to make a PBX.. and they created this green GUI and said "Shit, we don't have a core", and Mark Spencer was like "I'm not doing anything this weekend..." |
| 21:12.17 | Jumpie | if i wanna stick with my ubuntu box |
| 21:12.30 | file | night stars shining in my eyes! |
| 21:12.45 | bipolar | drmessano: haha |
| 21:12.51 | Yourname`` | file: And per channel whether it's VoIP or not, right? |
| 21:12.57 | Jumpie | ManxPower put it this way, i want the easiest way i suppose, but still flexible that i can take an dapply to customers |
| 21:13.07 | ManxPower | Jumpie: what you want is impossible |
| 21:13.13 | Jumpie | hmm |
| 21:13.34 | file | Yourname``: what? it's $10 per simultaneous encoding and decoding... so if you are recording to a file that is ulaw but your channel is g729... that's one |
| 21:13.43 | ManxPower | Telecom is HARD STUFF. No getting around that. VoIP is even more so, as you need to know telecom AND linux AND networking AND Asterisk |
| 21:13.50 | Yourname`` | file: Any bulk discounts? :) |
| 21:14.02 | Jumpie | ManxPower well i wanna learn the ins and outs of asterisks |
| 21:14.04 | bipolar | Jumpie: to learn it, get trixbox, or some other distro that uses the FreePBX web ui, make something that works, then see what the config files it creates look like. then you can learn to do it by hand. |
| 21:14.08 | Jumpie | and i dont want any crippled wierd gui changing whats the norm |
| 21:14.11 | lmadsen | ManxPower: amen |
| 21:14.15 | Jumpie | bipolar ok |
| 21:14.17 | file | Yourname``: I have no idea. |
| 21:14.18 | drmessano | If you want flexibility, a nice GUI, and something the customer is familiar with, theres always Windows Server 2003 and AsteriskWin32.. But if you go that route, there's a place in hell for you. |
| 21:14.18 | Jumpie | and where do i get those files |
| 21:14.25 | ManxPower | bipolar: you have never done that have you? |
| 21:14.32 | bipolar | ManxPower: yep |
| 21:14.34 | Jumpie | drmessano even i know that, i dont want to become a spawn of satan |
| 21:14.40 | ManxPower | bipolar: your poor poor thing |
| 21:14.44 | bipolar | ManxPower: I've got a trixbox system up now. |
| 21:14.46 | Yourname`` | ulaw uses 85 kilobits, right? |
| 21:15.00 | Yourname`` | Is it bits or bytes? |
| 21:15.01 | Yourname`` | lol |
| 21:15.05 | ManxPower | You didn't learn Asterisk by looking at it's config files. |
| 21:15.33 | bipolar | Jumpie: well, I've given you my advise... take it as it is :) |
| 21:15.42 | drmessano | I have a trixbox in use too.. Every couple weeks I remote in and reboot it because it decides not to reg to my ITSP.. Im SO glad I built it |
| 21:15.44 | Jumpie | well the channel seems rather split down the middle on trixbox or no |
| 21:15.47 | Jumpie | confusing to me |
| 21:15.58 | drmessano | Jumpie: Split down the middle? |
| 21:15.58 | Jumpie | im leaning more towards the not |
| 21:15.58 | lmadsen | Yourname``: bytes |
| 21:16.05 | ManxPower | Jumpie: you ARE going to screw up. royally. several times. There is nothing you can do to stop that. Accept it and move on. |
| 21:16.06 | Katty | tzanger: you around? |
| 21:16.12 | Jumpie | ManxPower i undrstand that |
| 21:16.15 | lmadsen | Yourname``: errr.. bits.. lol |
| 21:16.18 | Jumpie | what is the site i actually go on to get the releases? |
| 21:16.23 | Yourname`` | lol thanks |
| 21:16.24 | Jumpie | www.svn.org? |
| 21:16.35 | drmessano | oh man |
| 21:16.41 | outtolunc | *read ~thebook* |
| 21:16.45 | rupa | ponders |
| 21:16.58 | drmessano | hands denon the shotgun, with the muzzle pointed towards himself |
| 21:17.02 | drmessano | Pull the trigger man |
| 21:17.08 | denon | *click* |
| 21:17.09 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-86-141.w90-54.abo.wanadoo.fr) |
| 21:17.10 | denon | you forgot to load it |
| 21:17.23 | drmessano | Crap.. I CANT WIN AT THE INTARWEB |
| 21:17.44 | ManxPower | Jumpie: Please step away from the keyboard and step to wherever you have The Book |
| 21:17.53 | Jumpie | sigh |
| 21:17.57 | Jumpie | ok |
| 21:18.00 | drmessano | Put the TRIXBOX DOWN son.. Don't do anything stupid |
| 21:18.03 | drmessano | SLOWLY |
| 21:18.05 | Jumpie | i will |
| 21:18.06 | drmessano | That's it.. |
| 21:18.09 | Jumpie | and for starters |
| 21:18.12 | Jumpie | im gonan re install ubuntu |
| 21:18.28 | Jumpie | or...maybe somethin else if the book recomends |
| 21:18.31 | rupa | . o O ( debian ) |
| 21:19.15 | Jumpie | wow amazing ManxPower |
| 21:19.21 | Jumpie | ther eis a line in here that is lamost like what you said |
| 21:19.31 | Jumpie | "th emlutitude of answers generally boils down to the one you like the best" |
| 21:19.32 | Jumpie | hehe |
| 21:19.54 | [hC] | the sooner you realize that all of this, aside from asterisk itself, is all personal preference.. use what distro you want. use what kind of machine you want. use a gui if youwant, or dont |
| 21:20.03 | [hC] | at the end of the day its still going to do the same thing |
| 21:20.18 | Jumpie | right, and it seems whenever someone says they do this, they like that, you get 15 naysayers over here saying 'that sucks! wtf idiot" |
| 21:20.19 | Jumpie | makes it confusing |
| 21:20.33 | [hC] | right |
| 21:20.39 | Jumpie | this is why i was so confused |
| 21:20.42 | file | I will sell you some free will for a low low price |
| 21:20.49 | [hC] | if you want to do it the 'hardcore' way, install a distro, and go download asterisk from www.asterisk.org |
| 21:20.58 | Jumpie | thats what i am going to do |
| 21:21.05 | Jumpie | the very non used friendly way |
| 21:21.05 | denon | Jumpie: sounds like you need a little handholding .. choose whatever route you feel has the best handholding |
| 21:21.06 | [hC] | if you want an easier approach, where you can be up and running with a gui in a few minutes, go download the trixbox iso |
| 21:21.06 | Jumpie | :) |
| 21:21.10 | *** join/#asterisk bmg505 (n=leon@196-209-76-145-tbnb-esr-2.dynamic.isadsl.co.za) |
| 21:21.14 | rupa | but make sure you use a 14" monitor, not a 19" monitor. very important |
| 21:21.19 | Jumpie | lol |
| 21:21.25 | Jumpie | ill be sure to scrounge one up |
| 21:21.32 | outtolunc | thinks hC is offering to support it also as most here will not <G> |
| 21:21.42 | [hC] | hah |
| 21:21.51 | denon | outtolunc: hence my indirect comment :) |
| 21:21.59 | [hC] | why do you think i suggest people who need hand holding start with trixbox to get their feet wet? |
| 21:22.12 | Jumpie | hc i dont really want my hand held |
| 21:22.15 | Jumpie | i just wanted to know where to start |
| 21:22.17 | denon | because you hope to get them to leave? |
| 21:22.21 | Jumpie | and i had 28908276092876 people with different dos and donts |
| 21:22.25 | [hC] | well, wether you want your hand held or not, thats what you're asking for :) |
| 21:22.32 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 21:22.41 | Jumpie | thanks for all your advice :) im going to read this book more |
| 21:22.48 | outtolunc | regardless of which direction you go.. reading the book is NOT optional |
| 21:22.50 | denon | sounds like a plan |
| 21:22.55 | [hC] | but its simple, pick a distro, and install asterisk... and read the book, and the wiki at voip-info.org |
| 21:23.04 | ManxPower | denon: I suggest dinner and wine before the handholding, but that's just me. |
| 21:23.17 | file | the book is also a handy weapon |
| 21:23.18 | Jumpie | ill give you all some beer in a while when im all good at this |
| 21:23.25 | denon | ManxPower: I suggest .. well .. never ever holding your hand :) |
| 21:23.26 | denon | but that's just me |
| 21:23.28 | file | paper cut someone to death. |
| 21:23.30 | denon | ducks |
| 21:23.49 | ManxPower | denon: I think that's the best advice I've seen all day. |
| 21:24.01 | denon | hehe |
| 21:24.20 | drmessano | The book is also written by the second most popular guy on the planet named "Leif" |
| 21:24.22 | denon | you know, he could have had asterisk up and configured by now |
| 21:24.24 | drmessano | you can't go wrong there |
| 21:24.44 | ManxPower | denon: We've had a lot of that here today |
| 21:25.34 | denon | ManxPower: luckily Ive not had much time for irc, I've been swamped with people wanting to get new T1s ordered before end of the month |
| 21:25.44 | denon | we've got some killer qwest promos running .. which are killing me |
| 21:25.49 | ManxPower | Got me so annoyed that I went out and dug out a tree stump I've been putting off |
| 21:26.10 | drmessano | If everyone didn't want to install asterisk for the first time to start a CLEC or build a callcenter, then things could be easier ;) |
| 21:26.26 | ManxPower | denon: I'm more of an Asterisk Whore, so let me know if you need any consulting 8-) |
| 21:26.53 | denon | ManxPower: well, many of these are to existing asterisk shops and ISPs |
| 21:27.11 | denon | but I'll give ya cheap bandwidth, SIP LD and PRIs |
| 21:27.39 | *** join/#asterisk adjohn (n=adjohn@68-248-62-131.ded.ameritech.net) |
| 21:28.20 | ManxPower | denon: I doubt even you could give me cheap bandwidth out of the 256-538-xxxx |
| 21:28.36 | denon | ManxPower: think I worked some stuff up for you some years ago |
| 21:28.40 | denon | but a lot has changed since then |
| 21:30.13 | drmessano | http://blogs.dfw.com/startle_grams/images/leif75_4.jpg <-- Not the author of TFOT |
| 21:35.08 | [hC] | that is SO leif. |
| 21:35.09 | [hC] | :P |
| 21:35.26 | jameswf | leif is dreamy :) |
| 21:36.09 | *** part/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 21:36.29 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 21:37.15 | jameswf | ~lmadsen is dreamy http://blogs.dfw.com/startle_grams/images/leif75_4.jpg |
| 21:37.15 | jbot | jameswf: okay |
| 21:37.19 | jameswf | heh |
| 21:38.34 | drmessano | LOL |
| 21:42.28 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 21:43.39 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:45.07 | d00gster | does anyone know of a device (wifi preferred or possibly bluetooth) that I have use as a mic/speaker connecting to a PC and looks like a sound card on the PC? |
| 21:45.42 | d00gster | a star trek comminucator concept. I recall cisco was show casing a 2way voice wifi pager once |
| 21:46.33 | Katty | tzanger: ping? |
| 21:47.04 | drmessano | Bluetooth + bluetooth headset? |
| 21:49.26 | d00gster | range is a problem then |
| 21:50.13 | d00gster | like if I have my asterisk in the basement and Im in the second floor this maybe a challenge |
| 21:53.29 | drmessano | Could be |
| 21:53.51 | drmessano | Get a 100 foot USB cable and mount your bluetooth on the roof |
| 21:54.15 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-106-67.bstnma.east.verizon.net) |
| 21:54.17 | drmessano | Eh... seal the end first |
| 22:05.26 | *** join/#asterisk RoyK (n=roy@ip-22-54-149-91.dialup.ice.no) |
| 22:07.05 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 22:12.26 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 22:15.41 | *** join/#asterisk Teeli (n=tili@58.27.173.156.wateen.net) |
| 22:17.25 | jjshoe | heh someone used to be advertising a star trek type hting |
| 22:17.27 | jjshoe | years ago |
| 22:17.31 | jjshoe | qualcomm or someone like that |
| 22:19.18 | jjshoe | http://www.vocera.com/products/products.aspx |
| 22:20.37 | jjshoe | got me on protocol though |
| 22:22.08 | Jumpie | hi guys :) |
| 22:22.15 | Jumpie | os install almost complete, good book here man |
| 22:25.39 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 22:26.40 | *** join/#asterisk Docfxit (n=none@cpe-76-95-77-238.socal.res.rr.com) |
| 22:28.09 | *** join/#asterisk RoyK (n=roy@ip-22-54-149-91.dialup.ice.no) |
| 22:32.35 | *** join/#asterisk klin3d (i=Bsd@pc-32-231-86-200.cm.vtr.net) |
| 22:33.33 | zeniffty2002 | Are there any polycom guru's in the house, or can anyone point me to a better channel for polycoms |
| 22:34.04 | mcab | w |
| 22:34.25 | mcab | heh, oops |
| 22:35.27 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:35.54 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:36.23 | grandpapadot | Hey all, is there a way to *safely* purge my queue_log? Can I just delete it and if I do that will Asterisk just recreate it at next write? This is 1.2.2x |
| 22:37.46 | Jumpie | hey guys, if i get asterisk ready to go and compiled, but do not have my pots card yet (its en route) is it relatively simple to add that on? |
| 22:39.34 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 22:39.41 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 22:40.43 | Docfxit | zeniffty2002 » What question do you have? |
| 22:41.08 | [TK]D-Fender | grandpapadot, Kill @ will |
| 22:42.39 | grandpapadot | tnx, TK |
| 22:44.06 | zeniffty2002 | I inherited the polycoms with the system... I am sending custom Sip Notify packets to my 501's in an attempt to get them to think there is a voicemail. And that works.. but I can't get them to clear the flag that says there is no voice mail with a second Notify |
| 22:44.20 | *** part/#asterisk RoyK (n=roy@ip-22-54-149-91.dialup.ice.no) |
| 22:46.05 | zeniffty2002 | so the phone keeps flashing the voicemail LED, when in fact there isn't any voicemail to get |
| 22:49.09 | zeniffty2002 | I can clear the flag manually by going into the Voicemail feature on the phone and hitting "clear" or by rebooting the phone |
| 22:51.08 | Docfxit | zeniffty2002 » So what's the problem? |
| 22:51.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:51.35 | zeniffty2002 | readers digest version or whole enchilada? |
| 22:52.29 | [TK]D-Fender | zeniffty2002, What are you trying to indicate via VMI? |
| 22:53.58 | zeniffty2002 | my boss wants an indication on the phone that there are calls in queue. The only ways I can think of to show on the phone that this is the case is by lighing up the voicemail light, or putting some kind of graphic on the screen. Voicemail seems the easier way to go. |
| 22:54.03 | *** join/#asterisk autojack (n=owen@pdpc/supporter/active/autojack) |
| 22:54.30 | *** join/#asterisk klin3d (i=Bsd@pc-32-231-86-200.cm.vtr.net) |
| 22:54.31 | [TK]D-Fender | zeniffty2002, use presence on a buddy-watched line-key and the devstate patch |
| 22:55.09 | [TK]D-Fender | zeniffty2002, Far easier, and really thoguh the best way is to use the MicroBrowser.... I monitor 4 agents and 2 queues in detail on the idle screen for my CSR's |
| 22:55.36 | zeniffty2002 | i hadn't though of that. |
| 22:55.43 | zeniffty2002 | thanks |
| 22:56.46 | autojack | I'm totally new to asterisk and feeling a bit overwhelmed by the terminology and amount of docs :) I'm hoping to use it to create local-access numbers in the US and Australia, for my gf and I to be able to call each other via our mobile phones. can anyone point me in the right direction for what I should read up on to accomplish this? CAN I accomplish this? :) |
| 22:57.37 | *** part/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 22:58.48 | [TK]D-Fender | autojack, Yes. |
| 23:00.04 | autojack | it looks like maybe what I want is VOIP providers with termination in the US and in Australia, which both route to my Asterisk box... |
| 23:00.31 | [TK]D-Fender | autojack, If you want to use cell phones, odds are you'll pay an ITSP for a DID in the area of your choice and she can call you on that. It would then get processed by your * server and you'd call out another provider (probably cheaper to use the best service at each end at a lower cost) to your cell. |
| 23:01.00 | [TK]D-Fender | autojack, you can mix & match your providers for whoever gives you the deal you like the most. |
| 23:01.08 | autojack | got it. |
| 23:01.36 | autojack | so I can configure both DIDs to be hosted by my * server? |
| 23:01.56 | autojack | when when I call my US one, it can route via some ITSP to her cell in Australia? |
| 23:01.58 | autojack | and vice versa? |
| 23:02.29 | *** join/#asterisk WindBack (n=jorge@host51.190-136-119.telecom.net.ar) |
| 23:02.31 | [TK]D-Fender | autojack, Any which way you would like, yes |
| 23:03.04 | jjshoe | autojack why not skype or use any of the messengers which support voice talk? |
| 23:03.11 | autojack | ok. any suggestions for how to choose an ITSP? I just found this page: http://www.voipcharges.com/providers/australia |
| 23:03.20 | jjshoe | seems overkill to talk to a single person :P |
| 23:03.20 | autojack | jjshoe: I need to be able to do it mobile to mobile. |
| 23:03.25 | jjshoe | autojack ah. |
| 23:03.31 | autojack | the time difference is such that we can only talk when she's on her way to/from work. |
| 23:03.33 | denon | autojack: take a look at pennytel, you can probably do all of this without an asterisk box even, using their ANI callback service |
| 23:03.44 | drmessano | Australian mobile calls = $$$$$$ |
| 23:03.50 | *** join/#asterisk weazahl (n=jeremy@12.53.40.34) |
| 23:03.54 | denon | yes, Im well aware of australia calling :) |
| 23:03.59 | denon | look at pennytel :) |
| 23:04.01 | autojack | my problem is that we use cheapo phone cards now, and the call quality is terrible. |
| 23:04.16 | [TK]D-Fender | autojack, I'd say pick out your ITSP options and compare to calling-card rates. its all just math... I'd wonder if the dual-itsp option could actually end up cheaper... I somehow doubt it. |
| 23:04.16 | autojack | so I was looking for something where we could, I dunno, understand each other consistently :) |
| 23:04.44 | denon | autojack: has anyone mentioned pennytel to you? |
| 23:04.48 | autojack | I'm willing to pay a little more for something reliable and quality. but trying to avoid .35 a minute :) |
| 23:04.48 | [TK]D-Fender | autojack, Well if wuality is a real problem, see about getting a "trial account" with a few minutes on each side to test. |
| 23:04.52 | autojack | denon: :P |
| 23:05.10 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 23:05.16 | drmessano | Hmmm |
| 23:05.19 | autojack | I can get .35/min just calling her directly from my mobile, so if it's not cheaper than that it's not worth it. |
| 23:05.23 | autojack | checks out pennytel |
| 23:05.50 | weazahl | anyone have an idea why i am getting this? [Mar 31 17:53:02] ERROR[3039] chan_zap.c: Signalling requested on channel 28 is ISDN PRI but line is in Unknown signalling 524416 signalling |
| 23:06.02 | jameswf | wonders if cheapbastard.info is availible... |
| 23:06.32 | drmessano | Mobile <-- Free Mobile<>Mobile -- Your Asterisk+Chan_Mobile ---- Her Asterisk+Chan_Mobile -- Free Mobile<>Mobile --> Mobile |
| 23:06.33 | jameswf | weazahl: my guess is you are using an unknown signalling type |
| 23:06.34 | drmessano | Easy |
| 23:06.38 | [TK]D-Fender | jameswf, trying to down-tune $.35/m is not being "cheap" |
| 23:06.55 | weazahl | gee thanks! |
| 23:07.07 | jameswf | [TK]D-Fender: wasnt refering to that sorry bad timing |
| 23:07.33 | jameswf | weazahl: pastebin your zapata.conf |
| 23:07.37 | drmessano | Yay AsteriskCLEC.info is wide open |
| 23:07.59 | jameswf | you cant use aterisk in a domain name violates TM policy |
| 23:08.06 | jameswf | *asterisk |
| 23:08.14 | drmessano | AkeriskCLEC.info |
| 23:08.18 | drmessano | Handled |
| 23:08.31 | jameswf | asstrix.info was open last check |
| 23:08.43 | drmessano | What about asteriskpound.info |
| 23:08.46 | drmessano | Come one, sue me |
| 23:08.50 | drmessano | *on |
| 23:09.17 | jameswf | fivepointedasciistar.info |
| 23:09.18 | weazahl | jameswf: http://pastebin.com/m39918da5 |
| 23:09.24 | drmessano | I'd go with asteriskhash.info, but I don't need the DEA showing up AGAIN |
| 23:10.25 | drmessano | tricksbocks |
| 23:10.27 | jameswf | weazahl: are you sure your NET |
| 23:10.52 | jameswf | I was thinking about getting xobxirt.info |
| 23:10.53 | autojack | denon: hmm, so I can see how pennytel works for my gf in AU - it looks like it charges her .02/min to call a US mobile. but can I use it in the US to call her? |
| 23:10.54 | weazahl | jameswf: yes, i got it. should have stopped at 27 not 28. |
| 23:11.02 | weazahl | now i have pri in asterisk |
| 23:11.56 | jameswf | I could have it run the trix site in mirror mode.... |
| 23:12.15 | jameswf | like elgoog |
| 23:12.16 | [hC] | What do all you guys do when clients want to have multiple line keys on a phone that log in to separate queues? Register one as say extension 100, and another line as 200, etc? |
| 23:12.36 | [TK]D-Fender | [hC], Yes. |
| 23:12.58 | jameswf | http://elgoog.rb-hosting.de/index.cgi <<thats hawt |
| 23:13.03 | jameswf | ~hawt |
| 23:13.07 | jameswf | bah |
| 23:14.01 | [hC] | [TK]D-Fender: is there anything you do to take advantage of the queue login features on the polycom, or do you just assign your own login extensions to dial? I know the phones have ACD login features, but ive never used them. |
| 23:14.28 | [TK]D-Fender | [hC], not feasable. Use std extensions. |
| 23:15.05 | [hC] | [TK]D-Fender: thats what I thought. |
| 23:16.19 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 23:17.05 | weazahl | jameswf: maybe i should be national??? http://pastebin.com/m5c182a8c |
| 23:17.28 | jameswf | weazahl: are you in the US |
| 23:18.07 | weazahl | jameswf: PBX is merlin legend, Asterisk is the NET side. * is CPE |
| 23:18.16 | weazahl | err CSU |
| 23:18.27 | jameswf | neat,,,, probably national.... |
| 23:19.14 | jameswf | merlin flashes me back to my days in the trenches,,, |
| 23:19.17 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 23:19.18 | weazahl | i had it working on a test in the office. had to use net on it. but it was magix not legend. firmware is much more flexable |
| 23:22.08 | *** join/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 23:22.32 | *** part/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 23:31.53 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:33.53 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:39.38 | jameswf | ~spiderpig |
| 23:39.39 | jbot | somebody said spiderpig was http://www.youtube.com/watch?v=9alejPWHboc |
| 23:39.42 | jameswf | look out |
| 23:41.30 | mwalling | hahaha |
| 23:41.58 | weazahl | this means it worked right? -- B-channel 0/20 successfully restarted on span 2 |
| 23:42.38 | lmadsen | jameswf: LOL |
| 23:51.37 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 23:52.17 | Katty | hihi |
| 23:53.36 | jameswf | jbot: tell lmadsen about lmadsen |
| 23:54.36 | drmessano | jbot: tell me a story |
| 23:54.42 | *** join/#asterisk hmm-home (n=Administ@24-119-176-74.cpe.cableone.net) |
| 23:54.49 | Katty | hmm-home: hai (= |
| 23:54.53 | jameswf | jbot: eat poo |
| 23:54.53 | jbot | ACTION slurps up all the poo available |
| 23:54.53 | drmessano | jbot: tell Katty a story |
| 23:54.54 | hmm-home | Hello |
| 23:55.12 | drmessano | jbot: tell Katty to smell my feet |
| 23:55.16 | hmm-home | setting up pidgin again I just realized I have waaaay too many IM accounts |
| 23:55.45 | drmessano | Tell your friends you only use XMPP |
| 23:56.44 | Katty | hmm-home: yeah, i've got 6 on mine (= |
| 23:56.54 | drmessano | I only have 1 |
| 23:56.58 | drmessano | Dumped all the rest |
| 23:57.07 | hmm-home | Katty 9 on this one |
| 23:57.21 | Katty | hmm-home: mister social butterfly |
| 23:58.22 | drmessano | LOL |
| 23:58.28 | drmessano | How do you have 9 accounts? |
| 23:59.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:59.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:59.48 | hmm-home | 3 msn, 2 gtalk, 1 yahoo, 1 aim, 1 myspace 1 irc |
