| 00:00.00 | sxntixgo | i can compile pwlib and openh323, but i cant complie asterisk |
| 00:00.35 | ManxPower | sxntixgo: H323 is one of the hardest, most complicated, error prone and just downright nastiest things you can try to do with Asterisk. |
| 00:00.43 | JT | fujin: also, downloaded packages may not have the version or patches people require |
| 00:00.49 | JT | and silly defaults too |
| 00:00.51 | Stridernzl | manpower: yeah they are dodgy and i have watched this channel enough to know to do properly i really need to get a real phone ... but my call forwarding has solved the immediate problem ... no i just have 3 businesses comming to one line which is why i'd like to massge the P.C and say X line just rang .. then a few seconds later the phone will start ringing and i'll know whats happening |
| 00:00.52 | ManxPower | Your best bet is to search the mailing list archives |
| 00:01.25 | ManxPower | JT: or have patches that are not part of the standard distro. |
| 00:01.25 | SwK | anyone know of a good place to get t38 term into canada? |
| 00:01.55 | ManxPower | Now, if package maintainers do a good job, things might be different. |
| 00:02.08 | JT | sxntixgo: what h.323 devices are you connecting to? |
| 00:02.33 | JT | ManxPower: to be fair, the debian asterisk packages are some of the least offensive |
| 00:02.40 | ManxPower | For example the list of text for the asterisk sound files packages for Debian is not in /usr/share/doc/whatever, but in the same directory that the package installs the sound files. |
| 00:02.52 | ManxPower | JT: That is pretty scary. |
| 00:02.59 | sxntixgo | i really dont nokw |
| 00:03.12 | ManxPower | We could not even help the poor bastard find the textual information about the audio in the sound files. |
| 00:03.48 | sxntixgo | did you ever try to do it? is frustating |
| 00:03.50 | JT | ehh, documentation location is a thing that affects all sorts of packages |
| 00:04.06 | JT | if they know where debian keeps it, they should be able to find it |
| 00:04.06 | ManxPower | sxntixgo: NOBODY uses H323 with Asterisk if they have ANY choice. |
| 00:04.25 | JT | sxntixgo: are you using H.323 for a reason? |
| 00:04.29 | sxntixgo | well i have a customer that sends me h323 |
| 00:04.33 | ManxPower | JT: yeah, but that docs file was NOT where it should have been for Debian |
| 00:04.46 | JT | ManxPower: complain to tzafrir ;) |
| 00:04.50 | ManxPower | sounds-core.txt or whatever it is called. |
| 00:05.10 | JT | sxntixgo: just use yate is a H.323 translator and be done with it |
| 00:05.15 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 00:05.20 | sxntixgo | ty!!! |
| 00:05.25 | ManxPower | Asterisk does thru HEAVY development, I feel that such projects should not be packaged, they should be built from source. |
| 00:05.31 | JT | s/is a/as a/ |
| 00:05.52 | ManxPower | More mature projects that do not release a new version every other week -- those are perfect candidates for packages. |
| 00:06.20 | sxntixgo | but the funny thing is that i have complide h323 and asterisk last week on other server |
| 00:06.30 | sxntixgo | and i cant do it on this |
| 00:06.59 | JT | you compiled it, but did it work? |
| 00:07.10 | sxntixgo | yes!! |
| 00:07.45 | JT | lucky |
| 00:07.46 | sxntixgo | also i have used oh323, but h323 works better |
| 00:12.28 | sxntixgo | hello??? |
| 00:14.28 | sxntixgo | why nobody is writting? |
| 00:16.51 | JT | sxntixgo: because perhaps they have nothing to say, quite a possibility indeed. |
| 00:18.24 | ManxPower | I tried to use H323 with Asterisk many years ago. It failed miserably and I switched to SIP. Never looked back. |
| 00:21.22 | sxntixgo | this is diferente, i cant chose |
| 00:21.24 | sxntixgo | i mean |
| 00:21.36 | JT | sxntixgo: just get yate and be done with it |
| 00:21.48 | sxntixgo | thanks JT |
| 00:21.57 | JT | it has a sip to h.323 translation mode |
| 00:22.01 | JT | passes rtp directly |
| 00:22.14 | *** join/#asterisk toot (n=toot@84.19.255.122) |
| 00:22.32 | sxntixgo | an codecs too? |
| 00:26.10 | JT | it passes rtp directly.... |
| 00:26.18 | JT | codecs would be irrelevant |
| 00:27.08 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-65c0d419b8ec1cfd) |
| 00:27.26 | sxntixgo | not for me |
| 00:27.50 | sxntixgo | if it can translate codecs would be perfect |
| 00:28.38 | *** join/#asterisk nighty^ (n=nighty@p2136-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 00:29.41 | JT | sxntixgo: why's that? |
| 00:31.48 | sxntixgo | sorry... i will tell you my problem |
| 00:32.07 | sxntixgo | i need to create many many conference rooms |
| 00:32.13 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:32.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:32.24 | sxntixgo | so i use meetme command |
| 00:32.40 | sxntixgo | but meetme works whit g711 codec |
| 00:33.07 | sxntixgo | so if someone sends me g729 asterisks translates it |
| 00:33.30 | sxntixgo | and this translation uses a lot of cpu resources |
| 00:33.52 | JT | "asterisk" |
| 00:33.54 | JT | sure |
| 00:34.07 | sxntixgo | what about yate? |
| 00:34.27 | JT | the translation uses up a lot of cpu |
| 00:34.28 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 00:34.32 | JT | no matter what you do |
| 00:34.37 | JT | asterisk can still do that |
| 00:34.49 | JT | yate would take care of translating from h.323 |
| 00:35.23 | sxntixgo | the thing is that im handling like 100 calls |
| 00:35.23 | fujin | can I configure a sip peer with two different Ip addresses for SIP and RTP? |
| 00:35.47 | JT | sxntixgo: then you might need a couple of beefy machines |
| 00:36.20 | sxntixgo | right now im using a doble Xeon |
| 00:36.52 | JT | how many cores |
| 00:36.58 | sxntixgo | 2 cores |
| 00:37.19 | sxntixgo | when i do 'top' i can se 4 cpu |
| 00:40.11 | *** join/#asterisk WindBack (n=Administ@host95.190-136-114.telecom.net.ar) |
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| 00:41.23 | [TK]D-Fender | sxntixgo, Then you should go buy the TC100B |
| 00:41.47 | JT | as long as the drivers work properly yet... |
| 00:42.55 | sxntixgo | what is TC100B? |
| 00:43.08 | WindBack | In the Asrterisk Documentation say that if the chanel is busy the Dialplan jump to the n+101 priority. But how I can do, for example with idefisk client to make the chanel busy??? |
| 00:43.38 | [TK]D-Fender | WindBack, * does not use the n+101 method anymore, you should be using the DIALSTATUS vairable. |
| 00:44.18 | [TK]D-Fender | WindBack, And unless idefisk is already completely full of channels you'll never get "busy". You'll get a CW beep and it will count as NOANSWER |
| 00:45.55 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 00:48.02 | WindBack | [TK]D-Fender, n+101 is deprecated?? |
| 00:49.03 | ManxPower | WindBack: What you really should be doing is reading the UPGRADE.txt file for both 1.2 and 1.4 |
| 00:49.43 | ManxPower | WindBack: DO NOT look on the Wiki for Asterisk docs for dialplan applications. "show application dial". Also "show applications" in the Asterisk CLI. |
| 00:50.10 | ManxPower | [TK]D-Fender: can you turn off CW on idefish? |
| 00:50.14 | ManxPower | fisk too |
| 00:50.41 | ManxPower | sxntixgo: TC100B is a HARDWARE transcoding card for Asterisk |
| 00:50.49 | WindBack | ManxPower, [TK]D-Fender, I'm reading AsteriskTFOT. What is the best documentation or haw to??? |
| 00:51.01 | WindBack | *howto |
| 00:51.45 | ManxPower | WindBack: TFOT is a very good resource for concepts, but for actual documentation the sample config files that come with Asterisk and the Asterisk CLI "show application X" is the best place. |
| 00:51.50 | [TK]D-Fender | ManxPower, Maybe.. not sure. |
| 00:52.27 | ManxPower | as well as UPGRADE.txt |
| 00:52.53 | ManxPower | you would find those in /path/to/src/asterisk/docs and /path/to/src/asterisk/configs |
| 00:52.57 | fujin | what does Aug 1 12:52:42 WARNING[4454]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x82dcf68', 10 retries! mean? |
| 00:53.43 | [TK]D-Fender | WindBack, lookup "show application dial" in * CLI and then "asterisk variables" on the WIKI. |
| 00:54.15 | ManxPower | [TK]D-Fender: I think README.variables is better. |
| 00:57.32 | [TK]D-Fender | ManxPower, Good as well. |
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| 01:06.41 | sxntixgo | well people |
| 01:06.49 | sxntixgo | thanks for you help |
| 01:06.51 | sxntixgo | seee ya |
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| 01:14.51 | toot | why did i decide to install this server as a favour... will i ever learn, etc |
| 01:18.08 | *** join/#asterisk robh71 (n=robh71@host-206-54-206-57.entouch.net) |
| 01:19.32 | JT | eh |
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| 01:38.33 | pepesz76 | [TK]D-Fender: I went through the sites you suggested me about sip+nat - thanks. I'm making progres, yet long road ahead me. |
| 01:39.32 | pepesz76 | I make schema of my planed network + some config files: http://130.161.41.170/voip.txt |
| 01:40.05 | pepesz76 | any suggstions how o make all segments in one thing ? |
| 01:40.41 | [TK]D-Fender | pepesz76, cut & paste |
| 01:40.47 | *** join/#asterisk saftsack (n=saftsack@pD9E07BC2.dip.t-dialin.net) |
| 01:40.56 | pepesz76 | be patient with me ... - still learning |
| 01:41.21 | pepesz76 | can't do that - it will be mess |
| 01:41.51 | [TK]D-Fender | pepesz76, no it wont |
| 01:42.34 | pepesz76 | ok: so far phones_1_X can call between themselves. same for phones_3_X |
| 01:42.42 | JT | haha |
| 01:43.08 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
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| 01:44.09 | [TK]D-Fender | pepesz76, not a long road. fust a handful of minutes |
| 01:44.13 | [TK]D-Fender | just* |
| 01:45.51 | pepesz76 | I believe to connect those I have to use 'register' somewhere ...dunno |
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| 01:52.32 | [TK]D-Fender | pepesz76, you might want one between asterisk1/2/3 |
| 01:52.55 | [TK]D-Fender | pepesz76, though with 1 phone on each remote site I don't see what you have * there at all... |
| 01:53.15 | [TK]D-Fender | pepesz76, @2 &3 seem unnecessary |
| 01:54.24 | pepesz76 | @3there are 4 phones (not shown for simplicity) |
| 01:54.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 01:55.42 | pepesz76 | can asterisk #1 talk to #2 directly ? without #3 (proxy) ? |
| 01:56.23 | pepesz76 | both #1 & #3 are behind NAT |
| 01:56.40 | [TK]D-Fender | pepesz76, with only 1 phone behind #2 whats the poin of #2 & 3 at all? |
| 01:57.46 | pepesz76 | #3 will have 4 phones |
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| 02:00.06 | [TK]D-Fender | pepesz76, ok, tahts a decent reason |
| 02:00.15 | pepesz76 | * no 2 is there since it has public IP and I was afraid #1 & #3 will not see each other. If they can see then there is no need for it |
| 02:01.47 | *** join/#asterisk tux2067 (n=jfl@205.205.252.25) |
| 02:02.25 | [TK]D-Fender | pepesz76, no need. he can register directly to one of the other servers |
| 02:02.49 | tux2067 | Hi all! Is there a way I can undefine a variable? I want to undefine MACRO_RESULT to bridge calls after I have set it to CONTINUE. Any idea? |
| 02:07.45 | pepesz76 | So let's make it clear: First: Phone #2 can register to *#1 or *#3 ? Second: *#1 & *#3 can see each other? Third: How to make them talk to each other? |
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| 02:26.17 | stridernzl | [TK]D-Fender: u busy ? |
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| 02:37.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:37.23 | lmadsen | yo |
| 02:38.48 | Qwell | hey |
| 02:40.44 | fujin | hey, anyone have any ideas how to tell asterisk to failover on a dial? |
| 02:40.59 | fujin | like if I dial through my primary peer I'd like it to failover to the second peer |
| 02:42.00 | JT | put another Dial line below the first. |
| 02:42.05 | *** part/#asterisk daveburr (i=Miranda@208.254.183.84) |
| 02:42.58 | fujin | yeah? cool |
| 02:43.06 | fujin | but what about when the first dial finishes |
| 02:43.12 | fujin | or does that jump to another prio |
| 02:43.25 | JT | if it is successful, it does not go to the next priority |
| 02:43.30 | JT | if it fails, it will |
| 02:43.30 | fujin | oh sweet. |
| 02:43.33 | fujin | awesome :) |
| 02:44.20 | fujin | thanks again dude lol |
| 02:44.25 | fujin | you're a wealth of information as usual |
| 02:44.33 | lmadsen | Dial() will not continue in the dialplan if the other end answers |
| 02:44.45 | lmadsen | it only continues on failure |
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| 02:54.26 | Hmmhesays | this is weird I have cdr_odbc inserting " in empty fields |
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| 02:57.28 | riddlebox | is there a way I can see when asterisk tries to email me my voicemail messages, to see if there are any errors? |
| 02:57.38 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 02:58.14 | lmadsen | read the sendmail logs |
| 03:03.47 | *** join/#asterisk WindBack (n=Administ@host95.190-136-114.telecom.net.ar) |
| 03:07.51 | Hmmhesays | so the asterisk dst field takes its data from extension |
| 03:08.02 | WindBack | In Asterisk 1.2 when the chanel is busy the dial plan jump to the priority n+101. Now, in asterisk 1.4, I have to use GOTO extension-${DIALSTATUS}. How I can know if that extension don't overlap other existing extension???? |
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| 03:28.09 | fujin | hrmph |
| 03:28.24 | fujin | going to have to learn how to SER. |
| 03:28.36 | JT | is that right? |
| 03:28.40 | phsdshft | Can anyone advise where the integer values of the int reason inside of the struct for channel is defined? i.e. what each integer value means |
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| 03:28.56 | phsdshft | I'd assume it would be in the comments next to the part where it is defined, but it isn't |
| 03:28.59 | JT | phsdshft: what are you talking about? |
| 03:29.47 | SktyNick | Afternoon all. Can someone just confirm for me that a Noop run from within an AGI script will work the same as if it were in the Dialplan? (Ie. Dump our text back to the CLI?) |
| 03:32.04 | JT | phsdshft: ? reasons for what? |
| 03:32.11 | phsdshft | JT: in asterisk/channel.h the struct for ast_channel is defined |
| 03:32.22 | phsdshft | 'reason' is why it is unsuccessful.. |
| 03:32.26 | phsdshft | for example.. I'm using outbound dialing |
| 03:32.37 | JT | oh, failure reason |
| 03:32.41 | phsdshft | when it dials out, and say it gets a busy signal it returns a reason code of.. 1 or 3 or something |
| 03:32.50 | JT | google.com > isdn cause codes |
| 03:33.02 | JT | unless these are differen't |
| 03:33.04 | phsdshft | they don't seem to coorespond to ISDN cause codes |
| 03:33.09 | JT | s/differen't/different/ |
| 03:33.19 | JT | i'm sure they show where the mapping is |
| 03:33.35 | JT | in chan_zap or something |
| 03:33.38 | phsdshft | I submitted a patch to put them into a variable because they weren't before |
| 03:33.41 | phsdshft | but |
| 03:33.52 | phsdshft | I don't know what all of the cause codes are in human readable format |
| 03:33.59 | phsdshft | I don't see it defined anywhere... |
| 03:34.34 | JT | try chan_zap, or something that deals with isdn |
| 03:34.49 | phsdshft | well, it isn't isdn specific |
| 03:35.01 | phsdshft | I'm using H323 for example.. but I believe the codes are standardized looking at the code |
| 03:35.25 | JT | H.323 is based off the ISDN cause codes |
| 03:35.29 | JT | ITU Q.931 |
| 03:35.55 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:35.55 | JT | if you find the mapping from funny asterisk numbers to isdn cause codes, you get the reasons |
| 03:36.04 | JT | i'm SURE i've seen it in the source before |
| 03:36.10 | JT | grep it |
| 03:36.51 | Swat2 | *Sigh* @ freepbx |
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| 03:41.39 | WindBack | In Asterisk 1.2 when the chanel is busy the dial plan jump to the priority n+101. Now, in asterisk 1.4, I have to use GOTO extension-${DIALSTATUS}. How I can know if that extension don't overlap other existing extension???? |
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| 03:42.18 | JT | ?!??!??!?!?? |
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| 03:57.53 | riddlebox | on a grandstream GXP-2000, can I create a button for 701, so when someone parks a call, the button will light and you can press it to pick it up? |
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| 04:36.19 | Hmmhesays | da da da da hey |
| 04:42.40 | Hmmhesays | anyone else have problems with cdr_odbc inserting " in blank field? |
| 04:43.10 | Corydon76-home | not since I switched to cdr_adaptive_odbc |
| 04:45.03 | Hmmhesays | hrm what is that? |
| 04:46.13 | Corydon76-home | It's a CDR module which adapts CDR variables to the columns you have in your CDR table |
| 04:46.58 | Corydon76-home | So no more cramming 50 bits of information into the userfield |
| 04:47.43 | Hmmhesays | link? |
| 04:48.07 | Corydon76-home | http://svn.digium.com/view/asterisk/trunk/cdr/cdr_adaptive_odbc.c?view=log |
| 04:48.24 | Corydon76-home | Works in 1.4 as-is |
| 04:48.25 | Hmmhesays | and why does cdr_odbc insert a single double quote into blank fields |
| 04:48.46 | Corydon76-home | It shouldn't |
| 04:49.02 | Hmmhesays | yeah its the strangest thing on my box here |
| 04:49.44 | Corydon76-home | Which DB backend? |
| 04:50.01 | Hmmhesays | pgsql |
| 04:50.42 | Corydon76-home | Don't know offhand. But I've seen a fair share of weird behavior out of that driver |
| 04:50.58 | Corydon76-home | which fields, specifically? |
| 04:51.48 | Hmmhesays | any field that is blank |
| 04:52.09 | Corydon76-home | which for example, is? |
| 04:52.34 | Hmmhesays | which can be all of them except date dst, dcontext channel an dchannel |
| 04:53.33 | Corydon76-home | src? |
| 04:54.20 | Corydon76-home | I might could understand clid, but src? |
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| 06:08.39 | mmurdock | Good eventing all. |
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| 06:08.51 | mmurdock | yikes. evening... |
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| 06:09.40 | mmurdock | is there anything special I must do to have hints on parking spots in 1.4? |
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| 06:20.50 | jarod14 | hi there |
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| 06:30.45 | tzafrir | hi |
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| 06:41.38 | kova | hi all! |
| 06:42.10 | kova | I've run into some problems with the latest 1.2 |
| 06:42.20 | kova | SIP doesn't work anymore |
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| 06:44.52 | yonahw-work | is anybody familair with func_odbc? I am getting an "SQL Prepare failed!" error and am not really sure what the SQL prepare is supposed to do exactly in order to figure out where I am going wrong |
| 06:45.59 | JT | prepares your statement generally |
| 06:46.03 | JT | the statement must be bad |
| 06:47.12 | yonahw-work | JT: thanks, I will have to see what could be wrong with the statement |
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| 06:52.21 | yonahw-work | JT: thanks for the help i got it working |
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| 06:55.08 | JT | nice |
| 06:56.12 | sweeper | btw, that startup noise the polycoms used freaked me out the first time |
| 06:57.04 | JT | "self destruct imminent" |
| 06:57.54 | sweeper | ooo, that would be a good prank |
| 06:58.07 | sweeper | replace the wav file on the prov server after the installation... |
| 07:00.53 | JT | "Paper load letter" |
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| 07:02.25 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 07:02.33 | foo | Do I need permission do use MOH / IVR music from the original author? I'm a one man team right now, very small business... but just curious. |
| 07:02.55 | Juggie | you mean the files that are in asterisk? |
| 07:05.01 | foo | Juggie: No, I mean mp3 music I setup for MOH / IVR menu |
| 07:05.32 | JT | foo: music from where? |
| 07:05.37 | Juggie | well it all depends on where you got it |
| 07:05.55 | foo | oh, it was a mix I got from mp3.com. A friend gave it to me, but that's what the id3 tags say |
| 07:06.08 | sweeper | foo: look for CC music if you want to avoid troubles... |
| 07:06.11 | JT | foo: depends, usually you have to pay royalties |
| 07:06.14 | sweeper | it's pretty easy to come by |
| 07:06.31 | JT | foo: with royalty free music, you have to pay once, then no royalties |
| 07:06.41 | JT | creative commons is another option |
| 07:06.55 | JT | otherwise you will probably need to pay royalties on an ongoing basis |
| 07:06.56 | foo | Ok. If I need permission, I'll see if I can get in touch with the artist. |
| 07:07.03 | foo | I see, thanks, appreciate it. |
| 07:07.43 | foo | Hm, another question ... asterisk is returning a local network address (192.168.0.232) in the contacts details... any ideas how to fix this? Because of that, my voip softphone on my treo dies because no audio is able to come/go. |
| 07:08.56 | JT | foo: set externip= and localnet= in sip.conf |
| 07:09.26 | foo | JT: ah, thanks. |
| 07:10.50 | foo | JT: both are IP addresses, right? Or, external IP, and local network? eg. 192.168.0.0/24, I think it is. |
| 07:11.15 | JT | right, check the docs on format of localnet :) |
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| 07:24.28 | elvisthedj | any broadvoice users here? Suddenly I can't register :( |
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| 07:34.04 | yonahw-work | is there a way to change the speed when using SayDigits or SayNumber, so that it says the digits or numbers slower or faster? |
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| 07:46.43 | Aurs | is it possible to use environment variables in asterisk conf files? |
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| 07:48.06 | Aurs | ie in sip.conf something like: domain=$HOSTNAME |
| 07:48.16 | Aurs | that would be neat |
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| 07:56.32 | pepesz76 | <PROTECTED> |
| 07:57.55 | pepesz76 | I have two networks, in both couple of ipphones connected with * |
| 07:58.16 | pepesz76 | phone_1_1 ------------- asterisk1 - router/DSL ---[PUBLIC NETWORK]--- router/DSL - asterisk3 ------------------ phone_3_1 |
| 07:58.16 | pepesz76 | [user11] [conf_1 below] [conf_3 similar] [user31] |
| 07:58.16 | pepesz76 | [secret11] [server] [ to conf_2 ] [secret31] |
| 07:58.16 | pepesz76 | [outbound:asterisk1] | [ client ] outbound:asterisk3] |
| 07:58.18 | pepesz76 | <PROTECTED> |
| 07:58.23 | pepesz76 | phone_1_2 -------------------| |----------------------- phone_3_2 |
| 07:58.26 | pepesz76 | [user12] | |----------------------- phone_3_3 |
| 07:58.29 | pepesz76 | [secret12] | |----------------------- phone_3_4 |
| 07:58.32 | pepesz76 | [outbound:asterisk1] | |
| 07:58.34 | pepesz76 | <PROTECTED> |
| 07:58.38 | pepesz76 | phone_1_3--------------------| |
| 07:59.40 | pepesz76 | Question: asterisk1 shoud be server, asterisk2 client. How it should be configured? |
| 08:00.29 | JT | pepesz76: can you please never do that again? |
| 08:00.51 | pepesz76 | Do I have to register each client from network 3 in *1, or just register *3 in *1 |
| 08:00.56 | pepesz76 | ? |
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| 08:01.46 | JT | pepesz76: you flooded the channel with over 15 lines |
| 08:01.55 | JT | if it's more than 3 lines, pastebin it |
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| 08:02.48 | JT | ~pb |
| 08:02.48 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 08:03.11 | pepesz76 | Lesson learned :) |
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| 08:08.11 | puzzled | hi |
| 08:11.03 | foo | sweeper: You said to look for CC music, what is that? |
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| 08:11.32 | sweeper | foo: creative commons licensed music |
| 08:11.40 | foo | sweeper: ah, I see. Thanks |
| 08:11.46 | sweeper | it'll mostly be electronica and strange indie stuff |
| 08:12.31 | foo | sweeper: ah, what I was wanting in the background was some music from this guy: http://profile.myspace.com/index.cfm?fuseaction=user.viewprofile&friendID=67100018 |
| 08:18.07 | foo | sweeper: eh, I wonder how much of an issue it really is. I get like 4 phone calls a week ... |
| 08:18.36 | sweeper | wait, on your personal home phone system? |
| 08:18.41 | sweeper | that's fair use right there |
| 08:19.08 | foo | oh, no, it is business... just very very small :) I'm a one man team. |
| 08:20.21 | foo | I assume once I say "business" it's no longer fair game |
| 08:26.12 | JT | sweeper: fair use doesn't apply to broadcast i think |
| 08:26.30 | JT | foo: once you broadcast... |
| 08:26.43 | sweeper | JT: mmm, I dunno if it counts as broadcast |
| 08:27.16 | JT | it does |
| 08:27.34 | JT | phone system MoH is royalty fee applicable |
| 08:27.48 | sweeper | for a business! |
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| 08:43.07 | tzafrir | foo, honestly, finding some decent royalty-free MoH music is not *that* difficult. At least one that is good enough |
| 08:47.16 | pepesz76 | JT: This is my configuration with some config done for one of the asterisks. I stucked there. Don't know how to setup *3 and place calls e.g phone_1_1 to phone_3_2 http://pastebin.com/m1e132d64. |
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| 08:58.35 | kv0s | Hi! |
| 09:13.32 | tzafrir | pepesz76, that sounds like a pretty basic dialplan setup. What am I missing here? |
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| 09:22.48 | JT | sweeper: do you have any legislation to backup your assertion that royalties only apply to businesses? |
| 09:23.33 | sweeper | JT: not really! just a general impression~ |
| 09:23.47 | JT | yeah, probably not a good idea to go off that |
| 09:23.55 | sweeper | mmm |
| 09:24.10 | tzafrir | if you're a bisuness they'll clearly come after you. If not: maybe they won't |
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| 09:24.32 | sweeper | I still think having a recording on your home answering machine of copyrighted music isn't legally persuable :3 |
| 09:25.03 | sweeper | I mean |
| 09:25.25 | sweeper | do you have to pay royalties if you use music as a soundtrack for your wedding videos? |
| 09:25.26 | tzafrir | home answering machine? probably not an issue. But Asterisk is hardly a simple answering machine |
| 09:25.49 | sweeper | tzafrir: if it's at his house, being used for personal purposes, it might as well be |
| 09:26.13 | tzafrir | If you start giving away your phone number so people will dial to your line to listen to your cool message on your answering machine, then it is a bit closer ot "broadcast" |
| 09:27.19 | sweeper | even that's pretty thin...especially since it's self-limiting |
| 09:27.51 | tzafrir | sweeper, arguably: yes. |
| 09:27.52 | sweeper | either you're charging, or you're hemoragging money |
| 09:28.05 | pepesz76 | tzafrir: Sorry for the delay: "fire drill in the office". I'm sure it's simple. But I missing part of asterisk#3 config with the "register" and extensions.conf part saying how to dial to other network phones (e.g phone_1_1 to phone_3_2) |
| 09:28.09 | sweeper | gah spelling not good at 5AM |
| 09:28.22 | sweeper | pepesz76: bahahaha fire drills |
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| 09:30.49 | pepesz76 | sweeper: It's no fun runing from 14th floor to groundfloor on staircase ;) |
| 09:40.48 | *** join/#asterisk Paul_UK (i=Paul_UK@78.32.14.83) |
| 09:41.09 | Paul_UK | hey guys, besides g.711a whats the other best codec and not g729a? |
| 09:41.24 | sweeper | speex |
| 09:41.27 | Paul_UK | i need something low on the bandwidth, i dont care about processor intensiveness at the moment |
| 09:42.03 | Paul_UK | i doubt my provider supports speex |
| 09:42.24 | Paul_UK | plus its still in beta, so no thanks, anyone else? |
| 09:43.31 | sweeper | uh |
| 09:43.45 | Paul_UK | ok, its not just for me. i have 100 end points here |
| 09:43.46 | sweeper | you should probably take a look at what your provider supports |
| 09:44.28 | Paul_UK | i know they do the G711, 729 |
| 09:44.30 | *** join/#asterisk s0ck_ (n=m@unaffiliated/s0ck) |
| 09:44.40 | Paul_UK | dammit i had a link that explained it all nicely and i cant find it |
| 09:44.59 | sweeper | I mean, sure you could use mu-law encoding with a fast furier transform nested in a huffman tree. But you're limited to your provider....~ |
| 09:46.58 | s0ck_ | few = 3 or 4 |
| 09:47.08 | s0ck_ | surely can't be a processor issue...? |
| 09:52.51 | zeeesh | i hv been installed mysql-server,client,devel,-shared-compat ...do i need to install dbdmysql for running realtime asterisk? |
| 09:54.38 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 10:07.33 | Chris-NB | hi |
| 10:07.58 | Chris-NB | is it possible to specify the dial time for blind transfer (from features.conf)? |
| 10:08.01 | Chris-NB | is this possible? |
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| 10:09.42 | Tili | does anyone know why we need portaudio 2.0 for iaxclient library and i cant find 2.0 only 1.9 |
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| 10:13.59 | giasai68 | hello |
| 10:14.17 | giasai68 | I have fxs - fxo card |
| 10:14.58 | giasai68 | I'm using asterisk 1.4.9 |
| 10:15.11 | giasai68 | I'm interested to send fax |
| 10:15.16 | giasai68 | any suggest? |
| 10:15.45 | Tili | hylafax and iaxmodem |
| 10:15.49 | Tili | best solution i know |
| 10:15.59 | Tili | and asterisk of course |
| 10:16.17 | Tili | but if u have analog modems then use hylafax and whatever driver is for that modem. no asterisk no iaxmodem |
| 10:17.06 | giasai68 | I need use analog fax |
| 10:17.15 | giasai68 | for send and receive fax |
| 10:17.27 | giasai68 | can to know better solutions? |
| 10:17.29 | giasai68 | thanks |
| 10:17.52 | Tili | i have used hylafax with conexant modems |
| 10:17.55 | Tili | it works great |
| 10:18.25 | JT | giasai68: forget about faxing on an FXO port, waste of time. |
| 10:19.43 | Tili | dude get a proper HW Fax analog modem and use Hylafax |
| 10:20.21 | Tili | if u use asterisk with FXO cards then u need a lot of luck |
| 10:20.45 | Chris-NB | does anybody know if this is possible? Or if this is possible? |
| 10:21.17 | giasai68 | OK... but is it possible or no? |
| 10:22.30 | JT | giasai68: not reliably |
| 10:22.31 | giasai68 | I have installed app_rxfax.so and app_txfax.so also nvfaxdetect |
| 10:23.01 | JT | if you want to use fax to the pstn, use digital ports or real modems and hylafax |
| 10:24.29 | Swat2 | rtp.c: Unknown RTP codec 126 received from '172.16.1.44' <-- Whats that mean? |
| 10:26.11 | giasai68 | JT: can explain, pleas? |
| 10:26.50 | Tili | giasai68: JT is right. use real modems and hylafax if u want a reliable solution |
| 10:28.10 | JT | giasai68: i have already explained, so his Tili |
| 10:28.22 | JT | faxing to the pstn with an FXO port in asterisk sucks |
| 10:28.30 | JT | s/his/has/ |
| 10:29.30 | Tili | giasai68: by real modem JT meant modems that come with HW DSP for FAX. Hardware based signal processing like from Conexxant. they are a bit expensive than Software based DSP but they r the ones for commercial solution. |
| 10:30.21 | *** part/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 10:30.36 | zeeesh | i hv been installed mysql-server,client,devel,-shared-compat ...do i need to install dbdmysql for running realtime asterisk? |
| 10:30.39 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 10:32.27 | Tili | where am I going to get portaudio-2.0 |
| 10:33.10 | giasai68 | JT: ok... pleas, can explain schema ? |
| 10:34.18 | giasai68 | JT: I neet to connect PC with asterisk and modem to PSTN line? |
| 10:35.31 | JT | asterisk isn't actually involved |
| 10:36.04 | giasai68 | JT: any url for suggest? |
| 10:36.19 | JT | google hylafax |
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| 10:43.06 | creativx | morning JT :) |
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| 10:45.32 | tzafrir | JT, faxing from an FXO when that FXO is the only card still "sucks" in your experince? |
| 10:46.48 | *** join/#asterisk friedrich| (n=friedric@e177248027.adsl.alicedsl.de) |
| 10:49.33 | JackEStorm | has anyone seen this? kernel modules used: wcte12xp,wctdm ...when the system cold, or warm boots, the pri goes yellow, when I remove the modules, commit out the wctdm stuff in /etc/zaptel.conf, then load wcte12xp it goes green, then I can uncommit out the wctdm stuff, and reload that. |
| 10:49.53 | JackEStorm | err s/commit/comment/ |
| 10:54.45 | *** join/#asterisk stkn_ (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:11.27 | *** join/#asterisk shinao1 (n=shinao1@41.205.188.44) |
| 11:12.11 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
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| 11:21.30 | *** join/#asterisk vlt (n=dm@port-87-234-95-227.dynamic.qsc.de) |
| 11:22.21 | vlt | Hello. OT: Does anyone know how to reset the settings and password of a Grandstream BT100 w/o knowing the password? |
| 11:22.33 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 11:23.13 | razu | hi |
| 11:23.46 | razu | does anyone run asterisk on xen using SuSE here ? |
| 11:28.17 | vlt | Ok, found the solution in GS's FAQ. |
| 11:48.39 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 11:49.54 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 11:51.09 | yonahw-work | is there a way to get func_odbc to not cache query results or at least to manually clear the cache? |
| 11:54.58 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.52.161) |
| 12:05.11 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
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| 12:13.09 | tzafrir | your problem being? |
| 12:13.19 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:13.23 | tzafrir | (besides the strange color of the nick) |
| 12:14.01 | creativx | what? we have colors?!? |
| 12:14.18 | pepesz76 | You said it should be easy to make that one http://pastebin.com/m1e132d64 |
| 12:15.40 | yonahw-work | tzafrir: who are you talking to? |
| 12:15.56 | tzafrir | -pepesz76- any suggstions to my problem? |
| 12:16.12 | yonahw-work | oh missed that one |
| 12:19.48 | pepesz76 | The question was how to set extensions.conf for calling from one subnet to another... |
| 12:20.43 | *** part/#asterisk LoBoGoL (n=LoBoGoL@200.142.82.130) |
| 12:22.41 | ManxPower | pepesz76: There is nothing to set. It happens automattically. |
| 12:23.13 | ManxPower | In fact Asterisk does not even really know they are on different subnets, unless of course NAT is involved. |
| 12:24.16 | ManxPower | The setup your diagram shows is "Calling between two asterisk servers, each behind a different NAT router. |
| 12:25.10 | pepesz76 | yes - that's it |
| 12:25.47 | ManxPower | that is complex to do |
| 12:27.06 | mbranca | is not complex if you can control at least 1 router |
| 12:28.05 | pepesz76 | mbranca: I can do this. I can also setup an extra * with public IP if necessary |
| 12:28.53 | mbranca | can you do port redirection on the router to * boxes ? if yes just redirect port 5060 and rtp ports |
| 12:29.06 | mbranca | then configure externalip on both * and you're done |
| 12:29.48 | pepesz76 | I did that already - on one router I redirected 5060 and 10000-20000 to * |
| 12:30.02 | mbranca | ok |
| 12:30.14 | mbranca | put nat=yes on the peers definitions |
| 12:30.18 | ManxPower | pepesz76: you do not have localnet= or externip= set |
| 12:30.46 | mbranca | yup |
| 12:31.39 | *** join/#asterisk Tili (n=faizan@124.17.221.87.dynamic.jazztel.es) |
| 12:31.51 | pepesz76 | OK now it is set |
| 12:33.08 | ManxPower | pepesz76: what did you set externip to. |
| 12:33.24 | ManxPower | BTW, you have to set those things on BOTH sides. |
| 12:33.30 | pepesz76 | I can register phone from public IP and call with extensions both ways public <-> local(*1) |
| 12:34.28 | pepesz76 | On the other site the router doesn't support port range forwarding :( |
| 12:34.51 | ManxPower | pepesz76: then you cannot do what you want to do. |
| 12:35.03 | pepesz76 | will change it anyway in couple of days |
| 12:36.11 | *** join/#asterisk meppl (i=mephisto@meppl.net) |
| 12:38.21 | pepesz76 | One more question - so far I'm calling just with two digit extensions. How to setup * to call phone_1_1<->phone_3_1 the same way as phone_1_1<->phone_1_2? Is it possible ? |
| 12:39.04 | ManxPower | You don't. You send the call to the remote server and it handles that. |
| 12:39.31 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 12:39.43 | ManxPower | exten => _3XX,1,Dial(SIP/${EXTEN}@server1) where server1 |
| 12:39.54 | ManxPower | this should be covered in both ATFOT and the Wiki |
| 12:39.57 | ManxPower | ~wiki |
| 12:40.04 | ManxPower | ~book |
| 12:40.05 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:40.30 | FlatFoot | MankPower, ATFOT ? whats that abbreve for ? |
| 12:40.44 | ManxPower | Asterisk: The Future of Telephony |
| 12:40.49 | FlatFoot | sorry i meant ManxPower |
| 12:40.57 | FlatFoot | cant type for toffee |
| 12:42.18 | *** join/#asterisk stkn_ (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:43.00 | pepesz76 | I see. :) I was thinking of 'register' directive which looked promissing. Thanks. |
| 12:44.51 | meppl | has this Anthony Minessale, who developed "res_perl" for asterisk, any homepage or sth, where he wrote some statements, if he wants to go on developing this module or not and so on? |
| 12:45.24 | JT | meppl: freeswitch.org |
| 12:45.34 | *** join/#asterisk leif_madsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:45.34 | *** mode/#asterisk [+o leif_madsen] by ChanServ |
| 12:45.37 | ManxPower | register ONLY informs the remote side what IP is associated with a specific user/password. |
| 12:45.44 | ManxPower | at least one side must by on a static IP. |
| 12:45.55 | ManxPower | If both sides are on static IP, there is no need to register. |
| 12:46.26 | meppl | JT, on freeswitch.org i fond a download-link to the module with this text: "enables you to embed perl into Asterisk® giving you the power of perl for dialplans and applications without the overhead of AGIs." |
| 12:46.44 | JT | ok |
| 12:47.29 | meppl | found |
| 12:48.04 | pepesz76 | ManxPower: both sides have static IP. Thanks again :) |
| 12:52.22 | *** join/#asterisk nighty^ (n=nighty@p2136-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 12:52.37 | *** join/#asterisk el_critter (n=chatzill@190.74.124.49) |
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| 13:14.34 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.147.15) |
| 13:14.38 | xheliox | Anyone in the Florida area seen these BrightHouse commercials? Regarding "Asterisk sightings?"... they're not referring to Asterisk the PBX, but asterisk the puncuation.. but the more I see the commercial and how closely it's tied telecom buzz words, I'm afraid it might be causing some confusion. I know Asterisk is a registered trademark, is Digium aware of these ads? |
| 13:15.23 | xheliox | Tired or you think my concern is lame? :) |
| 13:15.52 | Katty | i just woke up |
| 13:16.02 | xheliox | Slacker. |
| 13:16.09 | Katty | no, i use debian, thanks. |
| 13:16.12 | xheliox | Har har. |
| 13:16.23 | lmadsen | xheliox: if you are concerned, you can email jwebster@digium.com |
| 13:16.29 | lmadsen | and he will look into it |
| 13:17.20 | xheliox | lmadsen: Sure thing.. I've now had two calls from clients thinking they meant Asterisk PBX. Which it's pretty obvious they don't mean that, if you pay close enough attention, but if you're casually listening, you might be 'shit, Asterisk is that thing that guy is trying to sell me on'. :) |
| 13:17.49 | lmadsen | xheliox: for sure! I'm sure they'll appreciate the heads up. |
| 13:18.09 | ManxPower | Somehow I can't be upset about ANYTHING that makes people think about Asterisk(tm) |
| 13:19.22 | ManxPower | Digium should just PATENT the Asterisk name (you can patent anything these days), then wait 10 years and sue the rest of the telecom industry out of existance. 8-) |
| 13:19.54 | *** join/#asterisk NirS_ (i=Nir@84.94.122.205.cable.012.net.il) |
| 13:20.10 | *** join/#asterisk CVirus (n=GoD@82.201.178.12) |
| 13:20.16 | el_critter | I'm used to digium cards but I'm considering sangoma for a new client, anyone with experience with that brand? |
| 13:20.17 | xheliox | www.asteriskhunters.com - they even have a website. |
| 13:20.25 | xheliox | Tell me that's not a little bad? |
| 13:20.33 | tzafrir | ManxPower, hmmm.... next someone will patent "Pound" for a "heavy-weight PBX" |
| 13:20.36 | ManxPower | el_critter: Sangoma is a bitch to set up, but work well. |
| 13:21.14 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:21.54 | JT | ManxPower: i debate the "bitch" bit |
| 13:22.10 | xheliox | el_critter: What ManxPower said, they're more complex (the setup) than Digium cards, but they're pretty much all I use now.. |
| 13:22.27 | JT | it's actually easier for most because it sets up options they don't understand on PRIs for them |
| 13:22.38 | xheliox | el_critter: Good warranty, good support, good people... not to say those things aren't always true about other brands of cards too. |
| 13:22.39 | Katty | speaking of PRIs. |
| 13:22.57 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 13:23.03 | Katty | our ISP wants to charge us twice as much for a PRI than a T1 |
| 13:23.10 | Daejeo1 | anyone can recommend me GSM Cellular Terminal Gateway (1-2 channels) for Asterisk PBX? |
| 13:23.11 | el_critter | ManxPower: my main reason to consider sangoma is that digium uses pci 2.2 and (I don't know why) I can only find servers with pci-express |
| 13:23.15 | Katty | however! i'm a little confused about why a pri is supposedly so much better. |
| 13:23.26 | JT | why is the isp providing telephony? |
| 13:23.30 | Katty | as i understand it 23 channels are available on a pri, and 24 on a t1 |
| 13:23.44 | JT | Katty: use of out of band signalling |
| 13:23.52 | JT | it's far better |
| 13:23.56 | Katty | what does band signalling do for me. |
| 13:24.00 | JT | not sure about double price though |
| 13:24.01 | xheliox | Katty: The D channel handles call control (hang up requests, etc...) It's much cleaner, can have CID name... etc. |
| 13:24.12 | JT | you can easily have hundreds of DIDs |
| 13:24.18 | JT | proper callerid |
| 13:24.19 | Katty | with a pri |
| 13:24.24 | Katty | correct? |
| 13:24.24 | JT | proper call progress signalling |
| 13:24.25 | JT | yes |
| 13:24.26 | Katty | k |
| 13:24.35 | el_critter | xheliox: so you say that if I'm experienced with digium analog cards maybe the setup of sangoma analogs is not going to be so difficult? |
| 13:24.41 | Katty | so, advantage of a pri is more reliable callerid... and the ability to have DIDs |
| 13:24.44 | Katty | is that correct? |
| 13:24.52 | Katty | and smoother calls. |
| 13:24.54 | Katty | obviously. |
| 13:24.57 | JT | and all the call progress is signalled properly |
| 13:24.57 | xheliox | el_critter: Shouldn't be.. |
| 13:25.28 | Katty | none of this pick up the line and you accidentally got a call. |
| 13:25.28 | JT | you can also hangup up calls without answering them |
| 13:25.28 | el_critter | digium should launch pci-express cards too |
| 13:25.28 | Katty | we won't be doing that. |
| 13:25.48 | JT | anyway, that gives you an idea of the flexibility |
| 13:25.59 | Katty | it sounds unreasonable to me that an ISP wants to charge us twice as much. |
| 13:26.04 | JT | again |
| 13:26.10 | JT | why is an isp offering telephony? |
| 13:26.12 | Katty | especially if the only thing that's different is the signalling. |
| 13:26.17 | JT | pris are the domain of telcos |
| 13:26.22 | Katty | the ISP is a reseller of AT&T |
| 13:26.28 | JT | and yes it's pretty unreasonable on their part |
| 13:26.34 | Katty | we have an integrated t1 from them right now. |
| 13:26.46 | JT | can you go direct to ATT? |
| 13:26.46 | Katty | which, unfortunately, gets dumped into a channel bank |
| 13:26.52 | Katty | nope |
| 13:26.59 | Katty | apparently the owner of this company is friends with the other owner |
| 13:27.01 | JT | how come? |
| 13:27.02 | JT | bah |
| 13:27.05 | Katty | or some other ammount of BS that means nothing to me |
| 13:27.10 | JT | not good enough friends |
| 13:27.11 | Katty | but the world to him |
| 13:27.16 | JT | if he can't get good rates |
| 13:27.16 | *** join/#asterisk syco (n=mike@88.197.224.73) |
| 13:27.21 | Katty | he gets good rates. |
| 13:27.25 | Katty | just crappy service |
| 13:27.33 | JT | double isn't good |
| 13:27.35 | Katty | i don't know how many times we've been down for half a day |
| 13:27.45 | Katty | or our lines can't make long distance calls |
| 13:27.45 | Katty | always something stupid |
| 13:27.46 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:27.52 | Katty | and they ALWAYS try to blame it on my server. |
| 13:28.10 | Katty | i guess they think i'm a dumb little girl or something (= |
| 13:29.46 | jcolp | el_critter: PCI-express has already launched |
| 13:29.51 | *** part/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 13:29.58 | el_critter | jcolp: really????? |
| 13:30.02 | jcolp | yes. |
| 13:30.03 | el_critter | let me check |
| 13:30.13 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 13:30.30 | jcolp | oh, analog |
| 13:30.35 | jcolp | no - no analog PCI express |
| 13:30.38 | JT | for like 2 products |
| 13:30.46 | JT | and no integrated HWEC |
| 13:30.48 | JT | heh |
| 13:30.56 | el_critter | jcolp: oh :( |
| 13:31.53 | Katty | analog is sucky |
| 13:31.56 | Katty | and it makes me all sad inside. |
| 13:32.19 | jcolp | Katty: <3 |
| 13:32.27 | JT | Katty: channelised T1 with RBS signalling is almost analogue, signalling wise |
| 13:32.28 | lmadsen | jcolp: <\3 |
| 13:32.36 | jcolp | lmadsen: nub. |
| 13:32.43 | Qwell | jcolp: we miss file |
| 13:32.52 | lmadsen | no we don't |
| 13:32.56 | lmadsen | :D |
| 13:33.09 | lmadsen | s/Qwell/jparker/g |
| 13:33.15 | Qwell | never |
| 13:33.22 | jcolp | Qwell: doooooo it |
| 13:33.30 | Katty | jcolp: yes, we do miss file. |
| 13:33.44 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.49.71) |
| 13:33.48 | Katty | anonymouz666: :> |
| 13:33.56 | anonymouz666 | Katty!!! |
| 13:34.00 | Katty | herro. |
| 13:34.16 | Katty | anonymouz666: can you believe our isp wants to charge us twice as much for a pri than a t1? |
| 13:34.31 | anonymouz666 | I can't believe! |
| 13:34.41 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 13:35.02 | Katty | too bad they won't listen to me. |
| 13:35.04 | MindTheGap | morning all... i've set hints for extensions in a separate [phones] context than included it in every other context in the dialplan. even so, when a user is on the phone, a "core show hints" shows its number as idle, and i still got "Got SUBSCRIBE for extension 308@adm from 10.0.0.11, but there is no hint for that extension". |
| 13:35.14 | anonymouz666 | Katty: We use E1 here :) |
| 13:35.42 | Katty | anonymouz666: well that's just because you're special. |
| 13:35.48 | Katty | anonymouz666: and we can't /all/ be special. |
| 13:35.54 | anonymouz666 | Hehehe |
| 13:36.45 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5ebe91d3e297dc18) |
| 13:36.57 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 13:36.58 | MindTheGap | i thought hints were supposed to give status on extensions, i tont get why sould it be on extensions.conf inside contexts, its mor like a global var. can someone enlighten me? |
| 13:37.07 | anonymouz666 | putnopvut has joined #asterisk - the app_queue master :D |
| 13:37.43 | putnopvut | Uh oh, the title is catching on... |
| 13:38.36 | Katty | i need caffeine |
| 13:38.38 | Katty | MEDIC |
| 13:38.56 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:39.00 | creativx | i can offer you some cancer advancing pepsi max Katty. |
| 13:39.02 | *** join/#asterisk guillote_GNU (n=guillote@host212.201-252-198.telecom.net.ar) |
| 13:39.02 | anonymouz666 | Katty |
| 13:39.04 | anonymouz666 | oops |
| 13:39.05 | anonymouz666 | hehe |
| 13:39.07 | *** part/#asterisk sketc1 (i=ryan@nat/digium/x-563f3ddf20db0a03) |
| 13:39.17 | Katty | creativx: that's sounds swell. |
| 13:39.24 | Katty | anonymouz666: wings :> |
| 13:39.29 | anonymouz666 | lol |
| 13:39.32 | creativx | Katty: atleast it tastes good when cold :) |
| 13:39.58 | waptaxi | hi! how can i cancel outgoing call queue using call-files? is it safe to delete these files from /var/spool/asterisk/outgoing/? |
| 13:41.14 | Sci_05 | lets see what katty gets this time when she pushes a button |
| 13:41.17 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 13:43.33 | *** join/#asterisk NirS (i=Nir@87.68.207.205.adsl.012.net.il) |
| 13:45.23 | Katty | vending machine was out of mt. dew :< |
| 13:45.29 | Katty | so, i had to resort to tea. |
| 13:46.14 | anthm | from the nutrimatic? was it not incredibly unlike tea? |
| 13:46.35 | Sci_05 | Katty: should have tried hitting the diet code button ir probably would have given you diet chocolate fudge soda |
| 13:48.15 | Katty | anthm: this is regular tea, not long island :P |
| 13:48.42 | Katty | non vending machine tea. |
| 13:49.04 | lmadsen | green tea y0 |
| 13:49.17 | Katty | green tea makes me feel funny. |
| 13:49.24 | Katty | might be that unusual "healthy" feeling. |
| 13:49.24 | lmadsen | interesting |
| 13:49.29 | lmadsen | quite possible |
| 13:49.53 | JT | some premade green tea is a LOT sweeter than others |
| 13:50.17 | Katty | the stuff i had was awfully bitter :/ |
| 13:50.28 | lmadsen | I like bitter tohugh |
| 13:50.49 | Katty | then it's perfect for you! |
| 13:51.12 | JT | yeah |
| 13:51.20 | JT | a lot of the stuff in japan is bitter |
| 13:51.31 | Katty | a lot of it's really neat too. |
| 13:51.35 | Katty | like the gothic lolita shops |
| 13:51.40 | JT | and like half of the drinks section in vending machines and convenience store is variants of green tea |
| 13:51.42 | Katty | those're awesome ^_^ |
| 13:51.54 | JT | i thought i was buying an awesome mango drink |
| 13:52.00 | JT | but no, more bitter green tea |
| 13:52.06 | Katty | aww. |
| 13:52.10 | Katty | what about rice milk? |
| 13:52.15 | Katty | rice milk's yummy. |
| 13:52.20 | JT | silly picture of ice cubes in orange liquid looked like mango |
| 13:52.22 | MindTheGap | no answer on "hints" so, here goes another one... :) Server has USB enabled and up and according to google it should be enough to have internal timing enabled on asterisk. well it doesnt, even if i run ztdummy it still spits out: channel.c:2338 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=44). I do have option_internal_timing=1 and internal_timing=yes set on asterisk.conf. What am I missing? ps. it |
| 13:52.22 | MindTheGap | s working just fine, its just flooding me w this messages. |
| 13:52.28 | JT | dunno if i've tried that, sounds ok |
| 13:52.35 | *** join/#asterisk Op3r (n=Op3r@121.97.251.114) |
| 13:52.56 | Katty | JT: it's kinda sweet. sells at your average american grocery store for those lactose intollerant peoples. |
| 13:53.21 | JT | ah ok |
| 13:54.07 | *** join/#asterisk mog (i=mog@nat/digium/x-eba31c7106325ddd) |
| 13:54.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:54.15 | Katty | hi mog. |
| 13:54.58 | mog | hi katty |
| 13:55.13 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:55.19 | Katty | everyone hide! |
| 13:56.21 | [TK]D-Fender | Grawr! |
| 13:57.37 | jcolp | woof |
| 13:58.18 | [TK]D-Fender | jcolp: I DON'T WANT TO BE AT WORK |
| 13:58.56 | karleeto | [TK]D-Fender: at least you can IRC from work |
| 14:00.15 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:01.32 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:01.59 | creativx | [TK]D-Fender: love your work! |
| 14:02.42 | Katty | i irc from everywhere. |
| 14:02.45 | Katty | including my blackberry. |
| 14:02.54 | anonymouz666 | that's beautiful, Katty |
| 14:02.56 | anonymouz666 | lol |
| 14:03.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:03.39 | *** join/#asterisk Gled|work (n=gled@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
| 14:03.55 | anonymouz666 | My nokia E61 does not work with Asterisk :( |
| 14:04.42 | anthm | it does if you don't use auth on it |
| 14:05.01 | Katty | ... |
| 14:05.05 | Katty | my co-worker is so /stupid/ |
| 14:05.11 | Katty | "so how do you make pickled bread?" |
| 14:05.13 | Katty | "dilldough!" |
| 14:05.20 | [TK]D-Fender | jcolp: ... |
| 14:05.29 | jcolp | o.o |
| 14:05.31 | anonymouz666 | anthm: I noticed that. I tested with SER this way. |
| 14:05.37 | [TK]D-Fender | Katty: I'll have to remember that one ;) |
| 14:05.48 | anonymouz666 | anthm: just reply 200 OK to Register without checking anything |
| 14:05.53 | datachomper | What kind of cheese isn't your cheese? |
| 14:05.59 | datachomper | NACHO CHEESE! |
| 14:06.25 | anthm | How do scarecrows get turned on? |
| 14:06.31 | anthm | CORNography! |
| 14:06.39 | [TK]D-Fender | ok, I can't take much more of this PUNishment.... |
| 14:06.46 | datachomper | roflcoptor |
| 14:07.11 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:08.38 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 14:08.47 | generalhan | hey all ! |
| 14:09.00 | Katty | hey han. |
| 14:09.17 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:09.39 | generalhan | its been a while since i have compiled * so i want to be sure that i have this order down correctly ! Libpri, zaptel, asterisk, addons |
| 14:10.02 | generalhan | or zaptel first ? |
| 14:10.26 | [TK]D-Fender | generalhan: 1st order was correct |
| 14:10.29 | Katty | lib, zap, asterisk, addons |
| 14:10.43 | generalhan | awsome ! i actually retained something !! thanks ! |
| 14:10.50 | Katty | i didn't |
| 14:10.53 | Katty | i just wrote mine down :P |
| 14:10.57 | generalhan | lol |
| 14:11.03 | generalhan | maybe i should have done that ! lol |
| 14:11.07 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 14:14.14 | creativx | hm.. if i want to set up an extension pattern like "exten => 556274[1-5],1,noop(${EXTEN})" - is there any obvious reason for the following error: Rejected connect attempt from 0.0.0.0, request '55627451@inbound' does not exist |
| 14:15.12 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 14:15.48 | Katty | hmm. glenn miller or nat king cole? |
| 14:15.50 | Katty | i can't decide. |
| 14:16.00 | Rienzilla | nat |
| 14:16.01 | Rienzilla | :) |
| 14:16.02 | Katty | k |
| 14:16.14 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:16.24 | Katty | (mona lisa)++ |
| 14:17.03 | *** join/#asterisk sakic (n=sakic@adsl-146-134-195.clt.bellsouth.net) |
| 14:17.18 | sakic | anyone have a easy to follow installation instruction link? |
| 14:17.25 | Katty | my blog, perhaps. |
| 14:17.31 | sakic | url? |
| 14:17.34 | Katty | but it depends on what you're trying to install. |
| 14:17.38 | lmadsen | Katty: Glenn Miller |
| 14:17.41 | tzafrir | sakic, what distro? |
| 14:17.48 | Katty | lmadsen: meh, already got nat going. |
| 14:17.54 | lmadsen | then you screwed up :) |
| 14:17.56 | sakic | asterisk 1.4.9 on CentOS |
| 14:18.03 | Katty | then my blog won't be to useful. |
| 14:18.11 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 14:18.12 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:18.18 | Katty | Qwell_: you have a tail :> |
| 14:18.32 | tzafrir | Katty, your blog entry? |
| 14:18.43 | Qwell_ | so I do |
| 14:18.53 | Katty | tzafrir: i blog astersisk because my memory is awful (= |
| 14:19.16 | Katty | tzafrir: it's just a reference for me. |
| 14:19.21 | Katty | tzafrir: nothing fancy., |
| 14:19.40 | kv0s | Any bristuff-Users out there? I've just setup my asterisk with bristuff. Outgoing calls with my sipphone via isdn works well. Also incoming from isdn to my sip ... but not my isdn-phone on my second hfc-s card. i need some hints to configure it ... |
| 14:19.47 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:19.59 | tzafrir | kv0s, there are |
| 14:20.38 | tzafrir | Katty, where is that blog? |
| 14:21.44 | kv0s | tzafrir? Say it! ;-) I've a hfc-s for external in te-mode and one hfc-s in nt-mode. It works, but i don't know how to configure my isdn phone so i can call it internal and make outgoing calls ... |
| 14:23.17 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:24.07 | Katty | tzafrir: sec, i'll get ya the link. |
| 14:24.42 | MindTheGap | morning all... i've set hints for extensions in a separate [phones] context than included it in every other context in the dialplan. even so, when a user is on the phone, a "core show hints" shows its number as idle, and i still got "Got SUBSCRIBE for extension 308@adm from 10.0.0.11, but there is no hint for that extension". |
| 14:24.49 | MindTheGap | i thought hints were supposed to give status on extensions, i dont get why sould it be on extensions.conf inside contexts, its mor like a global var. can someone enlighten me? |
| 14:24.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:25.00 | tzafrir | does the card provide power to the phone? or does the phone have an external source? |
| 14:25.24 | creativx | arggh this makes no sense. pattern matching doesnt want to work in my inbound iax calls |
| 14:25.37 | generalhan | ok finished the compile !! im making a file for chkconfig to start * at boot ... will some one take a look and tell me if this looks ok ! http://generalhan.pastebin.ca/641803 |
| 14:25.43 | creativx | MindTheGap: put the hint in the same context as the extension |
| 14:25.50 | jcolp | MindTheGap: a hint maps an extension to a device, for some setups you may have overlapping extensions (ie: different companies on the same system) so having it be global would be silly |
| 14:26.13 | tzafrir | generalhan, have you looked at the samples provided with asterisk? |
| 14:26.30 | generalhan | tzafrir: no i havent, in make samples you mean ? |
| 14:27.20 | tzafrir | generalhan, that's very bad: you run it at shutdown? |
| 14:27.32 | MindTheGap | i see... can it be included? like include [phones] "this context contains the hints |
| 14:27.41 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:27.45 | Katty | tzafrir: http://42ndgeekstreet.blogspot.com/2007/04/asterisk-install.html |
| 14:27.48 | tzafrir | You run 'modprobe wct4xxp on each reload? |
| 14:28.21 | generalhan | tzafrir: yes ... i need it to be loaded in order to start * at boot right ? |
| 14:28.51 | generalhan | unless the OS loads it automatically every boot |
| 14:28.59 | Katty | tzafrir: i halfway wrote it for whoever takes my place when i leave this company too. |
| 14:29.13 | tzafrir | generalhan, that's for a separate zaptel init.d script, IMHO |
| 14:29.22 | anonymouz666 | Katty: the blog is pink |
| 14:29.24 | creativx | MindTheGap: what version are you running |
| 14:29.24 | anonymouz666 | looool |
| 14:29.30 | Katty | anonymouz666: of course it's pink. |
| 14:29.35 | Katty | anonymouz666: i am a girl after all :P |
| 14:29.38 | creativx | oh pink indeed |
| 14:29.44 | MindTheGap | 1.4x |
| 14:29.51 | creativx | its actually almost the same as my desktop bg color Katty. hehe |
| 14:29.52 | generalhan | tzafrir: why ? i mean what is the bennefit of having it in a seperate file ? |
| 14:29.55 | MindTheGap | creativx, 1.4x |
| 14:29.57 | tzafrir | generalhan, and anyway, you did not check if the command is "start" or "stop" |
| 14:30.00 | Katty | creativx: :> |
| 14:30.02 | creativx | MindTheGap: what is the context name where you extensions are |
| 14:30.36 | MindTheGap | creativx, the hints? or the patterns? |
| 14:30.43 | tzafrir | You also re-run safe_asterisk even if one is running. As safe_asterisk's error handling is less than optimal, you can easily get multiple "safes" runinng that way |
| 14:30.46 | Katty | tzafrir: i'm gonna wipe that blog entry and put the new install i'm doing in there in its place. |
| 14:30.50 | creativx | MindTheGap: the device extensions |
| 14:30.53 | generalhan | tzafrir: but if * doesnt load on boot by default, isnt it safe to assume that it is not "start" after a boot ? |
| 14:30.55 | *** join/#asterisk DaveCanoe (n=Dave@ool-182c60c9.dyn.optonline.net) |
| 14:31.04 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:31.06 | MindTheGap | [interno] |
| 14:31.14 | Katty | i do have one problem tho. |
| 14:31.15 | generalhan | tzafrir: and i dont mean to sound contradictory, i just dont know ! |
| 14:31.20 | creativx | then put your hints into [interno](+) |
| 14:31.25 | creativx | and include that file |
| 14:31.26 | Katty | i tried to put the FOP ./op_server.pl command in my /etc/rc.local |
| 14:31.33 | tzafrir | what if you run that script and forgot that asterisk is already running? |
| 14:31.33 | Katty | after all the other commands i run |
| 14:31.39 | *** join/#asterisk ToyMan (n=Stuart@74-32-40-214.dsl1.mdl.ny.frontiernet.net) |
| 14:31.46 | Katty | but i can't seem to get FOP to come up on its own. |
| 14:31.56 | creativx | i like to play it unsafe too tzafrir |
| 14:31.57 | *** part/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 14:32.12 | tzafrir | Katty, there's already a deb of op-panel |
| 14:32.18 | *** join/#asterisk pepesz76 (n=pepesz76@wolin.et.tudelft.nl) |
| 14:32.58 | Katty | tzafrir: what? |
| 14:33.02 | Katty | tzafrir: that does not parse. |
| 14:33.04 | generalhan | tzafrir: but i would never manually run the script .... only in init.d for boot |
| 14:33.13 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:33.24 | sakic | is there a gui interface for asterisk install? |
| 14:33.26 | tzafrir | Katty, a .deb package of op-panel . An official one, actually |
| 14:33.59 | Katty | of fop, you mean? |
| 14:34.06 | tzafrir | It is generally a good idea to look at what packagers used in their packages |
| 14:34.40 | tzafrir | http://packages.debian.org/unstable/comm/op-panel , http://ftp.debian.org/debian/pool/main/o/op-panel/op-panel_0.27.dfsg-1.diff.gz |
| 14:34.53 | Katty | oh ah. |
| 14:35.00 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:35.09 | *** join/#asterisk karleeto_lap (n=karl@techwifi.franklincomputer.net) |
| 14:35.10 | tzafrir | that diff should have all the relevant ifles if the package misses them |
| 14:35.36 | Katty | i'll do that for this server i'm setting up here shortly. |
| 14:35.42 | Katty | see if it makes a difference (= |
| 14:36.12 | Op3r | can anyone recommend a cheap provider that offers unlimited channels? |
| 14:37.05 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 14:38.33 | Katty | Mercestes: (= |
| 14:38.40 | Mercestes | :) |
| 14:39.22 | *** join/#asterisk NirS_ (n=Nir@84.94.122.20.cable.012.net.il) |
| 14:43.04 | mocker | Crap, people are experiencing pauses when checking voicemail w/ the ODBC setup. |
| 14:43.37 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 14:43.37 | Mercestes | That would be your Dbase locating and extracting the wav files Ibet |
| 14:43.45 | centrex | Katty, You mean the modified free to air boxes? Those things are nifty! |
| 14:43.49 | mocker | Mercestes: I agree. |
| 14:44.01 | *** join/#asterisk joelbryan (n=joelbrya@124.104.13.167) |
| 14:44.15 | mocker | Mercestes: Do you know if there is a way to specify gsm or wav49 instead of wav for the blob? |
| 14:44.46 | Mercestes | not realy |
| 14:45.11 | Mercestes | Maybe if you disable all other available codecs for that connection you can try to 'force" it to use somethign specific |
| 14:45.25 | Mercestes | but, you run the risk of forcing a call offline. |
| 14:45.36 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:45.37 | *** part/#asterisk joelbryan (n=joelbrya@124.104.13.167) |
| 14:46.11 | generalhan | OMG ... something is seriously wrong here ... this is the 4th time i have compiled * in the past 2 weeks, 2x on fedora, 1x gentoo, and now on centOS, and everytime i finish compiling and reboot my eth0 doesnt come up at boot. i have to hit ifup eth0 a few times after bootup for it to finally work. has anyone ever even *heard* of this before ? |
| 14:46.18 | Katty | creativx: google coolsat |
| 14:46.31 | mocker | generalhan: Same hardware? |
| 14:46.38 | generalhan | mocker: yes same server |
| 14:46.38 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 14:46.41 | *** join/#asterisk y7n (n=na@office.intercea.co.uk) |
| 14:46.50 | mocker | generalhan: Don't think it's asterisk. |
| 14:46.58 | mocker | Anything odd in your dmesg? |
| 14:47.03 | generalhan | but i rebooted *SEVERAL* times before * was compiled with no prob |
| 14:47.17 | centrex | Katty, Yeah I use a pansat box, it's a different type of free to air box that a guy writes an OS for to decrypt the channels. |
| 14:47.22 | y7n | When I move a file to /var/spool/asterisk/outgoing does the file only get removed when the call has ended? |
| 14:47.26 | Katty | centrex: it's nice (= |
| 14:47.41 | generalhan | dmesg only says that there is no link to check the cable, i have replaced the cable 5 or 6 times now ... and took them from known working links |
| 14:48.05 | mocker | generalhan: Do you upgrade your kernel when you install asterisk? |
| 14:48.23 | generalhan | mocker: before hand i did ... i think in every instance |
| 14:48.52 | generalhan | but rebooted afterwards, and before compiling * |
| 14:52.15 | creativx | what am I doing wrong here? http://pastebin.ca/641827 |
| 14:53.28 | JackEStorm | has anyone seen this? kernel modules used: wcte12xp,wctdm ...when the system cold, or warm boots, the pri goes yellow, when I remove the modules, comment out the wctdm stuff in /etc/zaptel.conf, then load wcte12xp it goes green, then I can uncomment out the wctdm stuff, and reload that. |
| 14:54.18 | yonahw-work | creativx: I believe you need to put in an underscore before "5562745X" so that it pattern matches |
| 14:54.30 | Mercestes | generalhan, Are you using any zap hardware? |
| 14:54.32 | Katty | stupid debian cd just won't download. |
| 14:54.50 | generalhan | Mercestes: yes |
| 14:55.05 | generalhan | TE210P and there is a TDM40B in there too that im not really using |
| 14:55.08 | Mercestes | Do you have one of those silly riser cards or is your Eth0 on your motherboard? |
| 14:55.37 | generalhan | eth0 is on the motherboard, the digium hardware is on a riser bay |
| 14:55.52 | Mercestes | hrm. |
| 14:56.10 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 14:56.25 | creativx | yonahw-work: i'll be damned :) thanks for pointing that stupid error out |
| 14:56.26 | Mercestes | TE210P? Hrm..... |
| 14:56.35 | generalhan | Mercestes: yea "hrm" indeed ... this is the strangest issue ... and it only happens on reboots AFTER * is installed |
| 14:56.50 | Mercestes | Well, I was driving at a "Timing issue" on your riser card but....I think we're ruling that out. |
| 14:57.31 | Mercestes | what happens if you remove asterisk and leave all it's dependencies? Zaptel, cciritt_crc, etc. etc.? |
| 14:57.32 | *** join/#asterisk daqqal (n=lexiana@64.129.99.70) |
| 14:57.32 | yonahw-work | :) |
| 14:57.39 | daqqal | hello |
| 14:57.50 | generalhan | Mercestes: how would i go about removing asterisk and leaving zap ? |
| 14:57.58 | daqqal | is there a timecard application that works with asterisk? |
| 14:57.59 | tzafrir | cciritt_crc is a standard kernel module |
| 14:58.04 | Mercestes | zaptel is a seperate emerge in gentoo, and a seperate source in everything else. |
| 14:58.16 | Mercestes | tzafrir: It's off by default in gentoo as I recall |
| 14:58.28 | Mercestes | uses something else. i have to menuconfig it back in. |
| 14:58.39 | generalhan | Mercestes: i am on the CentOS distro now ... but i am not too familar with uninstalling in linux to begin with |
| 14:58.44 | Mercestes | generalhan, so Just make clean asterisk and leave everything else. |
| 14:58.48 | generalhan | ok |
| 14:59.10 | Mercestes | should still have zaptel and a zaptel module. |
| 14:59.16 | generalhan | done ... rebooting now ? |
| 14:59.21 | Mercestes | Yea. |
| 15:00.07 | generalhan | it sux cause im SSHing into this machine ... so when it doesnt work, i have to get off my lazy but and go bring Eth0 back up at the machine to come back and administer it ! lol |
| 15:00.38 | Katty | good ol ssh. |
| 15:00.52 | generalhan | Katty: making lazy people lazier ! |
| 15:00.58 | Katty | ^_^ |
| 15:01.01 | Mercestes | Try Raritan |
| 15:01.18 | Katty | i didn't jog around the block this morning :/ |
| 15:01.22 | Mercestes | It accomodates lazy even when the eth0 is offline |
| 15:01.23 | Katty | i feel like a bad person. |
| 15:01.30 | Mercestes | you are a bad person. |
| 15:01.56 | Mercestes | you naughty, naughty girl |
| 15:02.06 | Katty | ... |
| 15:02.06 | generalhan | Mercestes: yea ..,. i wish i could put this asterisk machine inside my bladecenter ... they have a KVM-Like peice built in |
| 15:03.00 | generalhan | Mercestes: Eth0 came up at boot ! i didnt have to get up !! |
| 15:03.25 | Mercestes | Katty, >.> |
| 15:03.28 | generalhan | ok im going to verify this .. im going to reinstall asterisk and reboot again ! |
| 15:03.39 | Mercestes | yea, just run a nice make install in asterisk |
| 15:04.16 | Mercestes | Katty: So I guess punishment is out of the question, huh? |
| 15:04.19 | generalhan | it seems to take A LOT longer than back in the 1.0.9 days when i last compiled * ! lol |
| 15:04.41 | Katty | uhh |
| 15:05.08 | Mercestes | ... |
| 15:05.11 | Mercestes | what if I'm bad? |
| 15:05.35 | Mercestes | generalhan, it's all that new leetness it has to compile in. |
| 15:05.48 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 15:05.53 | Mercestes | I feel so rejected |
| 15:06.03 | generalhan | ok ... done compiling, rebooting now ! |
| 15:06.05 | DirtyD | Hello. |
| 15:06.22 | generalhan | DirtyD .... |
| 15:06.26 | generalhan | Done dirt cheap ! |
| 15:06.27 | generalhan | lol |
| 15:07.06 | generalhan | ok ... not a big Angus Young fan i see ! |
| 15:07.29 | generalhan | Mercestes: i forgot to uninstall the addons and sound ! asterisk kinda freaked out there ! lol |
| 15:07.35 | DirtyD | Ok, in my IVR context, I have two 1 digit extensions. When I press 1, there is a delay before it goes to the next menu.. How do I tell the Backgroun command to only expect 1 digit. |
| 15:07.41 | generalhan | WARNING WARNING WARNING |
| 15:07.55 | Mercestes | Addons are ok |
| 15:08.18 | generalhan | well yea ... it just was warning me to make sure it is comapatible with the version of * i just conpiled |
| 15:09.49 | yonahw-work | can I open and close ODBC connections from the dialplan? |
| 15:11.03 | generalhan | Mercestes: maybe its something else ... cause it worked that time ! came right up at boot. |
| 15:11.30 | generalhan | i think i would have rather it been *, then at least there is *something* i could tie this too ... now its just completely random |
| 15:13.22 | Mercestes | well, not completely random |
| 15:13.43 | Mercestes | We've ruled out asterisk, which made no sense at all. |
| 15:14.01 | Mercestes | But you are still adding zaptel in between reboots, which does make some sense. |
| 15:14.08 | Katty | blah. |
| 15:14.15 | Mercestes | and ti could be bouncing it a few hundred times fixes it |
| 15:14.36 | Mercestes | now, I will say, zaptel doesn't make *alot* of sense but is atleast somewhere in the realm of possibility |
| 15:14.45 | generalhan | im rebooting again right now ... im gonna try rebooting a couple times in a row and see how many times it fails |
| 15:15.01 | Mercestes | it could be entirely unrelated and your cooincidentally exposing another issue though. |
| 15:15.02 | [TK]D-Fender | DirtyD: You don't. Your extens available to your IVR determine that. PASTEBIN is your friend.... |
| 15:15.03 | [TK]D-Fender | ~pb |
| 15:15.04 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:15.12 | *** join/#asterisk my007ms (i=alazif@217.139.224.221) |
| 15:15.41 | generalhan | im using zaptel in my current machine too, and there is no issue ... maybe i should try to use Eth1 and see what happens |
| 15:15.54 | generalhan | then ill have to reconfigure DHCP for that mac |
| 15:18.49 | Katty | is it lunch time yet :< |
| 15:19.30 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 15:19.44 | [TK]D-Fender | Hey, I'm having to do some Mac shopping for the office and I'm looking to validate a new system VS upgrading a laptop. can someone point me to a good Mac benchmark comparison site? |
| 15:20.14 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 15:24.23 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:24.36 | hi365 | anyone using the grandstream gxw-4104? |
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| 15:25.10 | Qwell[] | [TK]D-Fender: pfft. newest == best |
| 15:25.10 | Qwell[] | :p |
| 15:25.29 | [TK]D-Fender | Qwell[]: True, but not necessarily WORTH it :) |
| 15:25.31 | ManxPower | newest != best |
| 15:25.37 | pj_ | newest > best |
| 15:25.40 | Qwell[] | okay, I'll rephrase |
| 15:25.40 | generalhan | Mercestes: 2nd reboot and it failed again :( |
| 15:25.43 | Qwell[] | newest == fastest |
| 15:26.00 | mvanbaak | yeah |
| 15:26.01 | ManxPower | newest is usually = most bug ridden, poorly documented, and pain inducing |
| 15:26.06 | mvanbaak | dont get an ibook G4 |
| 15:26.09 | mvanbaak | it's SLOW |
| 15:26.10 | mvanbaak | :) |
| 15:26.13 | pj_ | don't get a knife |
| 15:26.27 | ManxPower | ~grandstream |
| 15:26.27 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 15:26.28 | pj_ | stay with the good ol' proven technology : cutting stone |
| 15:26.31 | ManxPower | ~gs |
| 15:26.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:26.40 | ManxPower | hi365: those were for you |
| 15:26.46 | creativx | man i suck at gotoifs() |
| 15:26.54 | hi365 | ManxPower: ?? |
| 15:27.03 | Mercestes | generalhan, I don't think it's * |
| 15:27.05 | ManxPower | (10:26:16 AM) jbot: GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:27.08 | Qwell[] | ManxPower: It's Apple. They have no bugs. They need no documentation. |
| 15:27.10 | Mercestes | I think you hav ean eth0 issue. ;) |
| 15:27.11 | generalhan | Mercestes: me either anymore |
| 15:27.12 | Qwell[] | (and yes, I'm trolling) |
| 15:27.16 | [TK]D-Fender | Qwell[]: Heres the scenario : 17" G4 w/1gig for Adbose usage. I'm considering a replacement instead of spending about $300 on 2G ram for it. Yes it'll cost more, but it should perform better and we'd get a laptop back. |
| 15:27.31 | ManxPower | Qwell[]: I stopped hating Apple after they moved to OSX |
| 15:27.33 | generalhan | Mercestes: yea maybe i should run into #centos for some troubleshooting advice ! |
| 15:27.34 | [TK]D-Fender | Qwell[]: the G4 = laptop |
| 15:27.59 | NOT_guru | ~linksys |
| 15:27.59 | jbot | extra, extra, read all about it, linksys is a tool of satan |
| 15:27.59 | generalhan | brb ... im gonna move to a closer station ! |
| 15:28.14 | NOT_guru | ~cisco |
| 15:28.14 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 15:28.17 | *** join/#asterisk soul_rebel (n=andrea@host-84-223-100-232.cust-adsl.tiscali.it) |
| 15:28.25 | anonymouz666 | hahahahaha |
| 15:28.26 | ManxPower | Linksys is the tool of Buddha! |
| 15:28.26 | Qwell[] | [TK]D-Fender: G4 == SLOW |
| 15:28.48 | anonymouz666 | the best quote ever |
| 15:29.04 | centrex | out of curiosity, why is linksys the tool of satan? |
| 15:29.21 | soul_rebel | hello! Can anyone tell me if asterisk has any voice recognition capability? |
| 15:29.22 | [TK]D-Fender | Qwell[]: I know... I want to prove HOW slow. They've quoted me long save times 5-6 mins, and they usually run off our server to which it appears to have a gigabit connection... so its either RAM, or CPU (or RAM + slow laptop HDD) |
| 15:29.26 | Mercestes | well, it's not *the* tool, just...Satan's tool of choice. |
| 15:29.49 | NOT_guru | ~digium |
| 15:29.49 | jbot | digium is probably evil, or reachable here: http://www.digium.com/en/company/contact.php |
| 15:29.55 | [TK]D-Fender | soul_rebel: * no, but other 3rd party apps, yes |
| 15:29.59 | centrex | ~sangoma |
| 15:30.00 | jbot | [sangoma] a company that makes PRI cards |
| 15:30.01 | NOT_guru | ~rhino |
| 15:30.03 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 15:30.14 | [TK]D-Fender | soul_rebel: Go read the WIKI for a list (which includes CMU Sphinx) |
| 15:30.16 | [TK]D-Fender | ~wikis |
| 15:30.17 | jbot | well, wikis is http://www.voip-info.org |
| 15:30.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 15:30.30 | NOT_guru | ~currentweather |
| 15:30.33 | NOT_guru | =D |
| 15:30.36 | centrex | ha. An asterisk channel bot that derides digium as evil and doesn't comment on sangoma =( |
| 15:30.40 | soul_rebel | [TK]D-Fender: any of those 3 party is free software |
| 15:30.56 | [TK]D-Fender | soul_rebel: Sphinx is. Go read up on it |
| 15:31.05 | soul_rebel | ok thx |
| 15:31.12 | Hmmhesays | areski's asterisk-stat-v2 is very pretty |
| 15:31.36 | [TK]D-Fender | Qwell[]: So any useful links for me? |
| 15:31.42 | Qwell[] | google.com? :P |
| 15:31.48 | Qwell[] | no, I don't own a Mac |
| 15:31.56 | [TK]D-Fender | Qwell[]: Surprisingly poor search results... |
| 15:32.08 | ^majik^ | using the database commands for call forwarding in Asterisk 1.4: is it legal to use, ie, CFIMN for the family name, where N is a number? or should I always use CFIM for the family - nothing more, nothing less |
| 15:32.14 | [TK]D-Fender | Qwell[]: I'd just hit Tom's Hardware if it were a PC thing.... |
| 15:32.19 | Qwell[] | Macs don't need benchmarks. They all perform equally. |
| 15:32.36 | Qwell[] | It's not better to have more RAM or a "newer" CPU, it just makes you more popular. |
| 15:32.46 | [TK]D-Fender | ^majik^: your dialplan does what it does, and you can use ASTDB any way you want to... |
| 15:33.01 | [TK]D-Fender | Qwell[]: Thanks.... I think... |
| 15:33.10 | NOT_guru | ~trixbox |
| 15:33.11 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 15:33.34 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 15:33.50 | mvanbaak | I have a G4 with 1GB ram |
| 15:33.59 | Hmmhesays | what do you guys use for CDR parsing? |
| 15:34.06 | centrex | [TK]D-Fender, http://db.xbench.com/ Has a long list of user submitted benchmarks, but it doesn't look like the ibook g4 is on there, unfortunately. |
| 15:34.16 | mvanbaak | it's ok for day to day usage by normal users. surfing, mail, neooffice |
| 15:34.22 | mvanbaak | but for -dev work it sux |
| 15:34.26 | mvanbaak | too slow |
| 15:34.35 | Idle | ok, I'm dialing out with asterisk, but it only dials, then I get dead silence.... as far as I can tell, my zap lines arn't actually dialing either |
| 15:34.36 | ^majik^ | [TK]D-Fender: well, I have a strange problem; my incoming dialplan doesn't realize there's no CFIM key and attempts to dial it anyways, when results in 'Zap/gN/' instead of skipping down and dialing what it needs to. any ideas? |
| 15:34.38 | [TK]D-Fender | centrex: its a PowerBook G$, and appears on there |
| 15:34.50 | mvanbaak | already compiling 1.4 for over 15 minutes now |
| 15:35.09 | centrex | [TK]D-Fender, yeah I saw the powerbook on there. |
| 15:35.13 | ^majik^ | [TK]D-Fender: for example, a jump to s,102 does not happen |
| 15:35.56 | mvanbaak | ^majik^: pastebin the relevant parts of your extensions.conf |
| 15:36.07 | [TK]D-Fender | ^majik^: Priority jumping changed with 1.4 read your apps instructions |
| 15:36.11 | mvanbaak | ~pb |
| 15:36.11 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:36.23 | [TK]D-Fender | ^majik^: "show function DB" |
| 15:37.58 | *** join/#asterisk galeras (n=galeras@201.244.199.31) |
| 15:38.08 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 15:38.45 | ^majik^ | [TK]D-Fender: ah, I think I just found a solution GoToIf and DB_EXISTS |
| 15:38.54 | yonahw-work | noone for opening and closing an ODBC connection directly from the dialplan? |
| 15:39.18 | mvanbaak | hey, did downloads.digium.com got a new connection during today ? |
| 15:39.28 | mvanbaak | this morning it was giving me 30KB/s |
| 15:39.35 | mvanbaak | now it gives me 460KB/s |
| 15:39.57 | russellb | there are multiple different boxes it could be |
| 15:39.59 | Hmmhesays | fun |
| 15:40.01 | [TK]D-Fender | ^majik^: so much easier to simply GotoIf on the RETURNED VALUE. |
| 15:40.26 | mvanbaak | russellb: ah, bet I hit the wrong box earlier today I guess |
| 15:40.32 | russellb | guess so .. |
| 15:40.33 | ^majik^ | [TK]D-Fender: :-) Thanks! |
| 15:41.05 | ManxPower | This is why you should read UPGRADE.txt and why 1.2's UPGRADE.txt should be in 1.4's source tree (in addition to 1.4's UPGRADE.txt) |
| 15:41.36 | ManxPower | There is SO much info out there that is 1.0 specific (without mentioning it is 1.0 specific) that 1.4 users are reading. |
| 15:42.59 | ManxPower | Hell, "make install" should 'less UPGRADE.txt' at the end of the install |
| 15:43.09 | neverblue | anyone VOIP providers in the channel, I am looking for service.... |
| 15:43.37 | ManxPower | neverblue: you have been asking for 2 days. |
| 15:44.08 | neverblue | so? |
| 15:44.24 | ManxPower | It would be faster if you just looked at the archives of the asterisk-biz mailing list. |
| 15:44.42 | *** join/#asterisk ethemc (n=tet@88.248.14.186) |
| 15:44.53 | ethemc | configure: error: C++ preprocessor "/lib/cpp" fails sanity check--> |
| 15:44.57 | ethemc | can anybody help please |
| 15:45.14 | Katty | Hmmhesays: herro (= |
| 15:45.17 | neverblue | ManxPower, i can ask in the channel if I want |
| 15:45.28 | neverblue | doesnt matter what you think would be 'faster' |
| 15:45.53 | ethemc | i want to install asterisk 1.4.9 on a debian Linux pbx 2.6.18-4-686 |
| 15:46.00 | ethemc | but i get thþs error after ./configure |
| 15:46.09 | ethemc | onfigure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 15:46.28 | ethemc | i have reinstalled the preprocessor |
| 15:46.57 | [TK]D-Fender | neverblue : How is they choice so difficult? I mean you know the players already. What are you missing to make up your mind? |
| 15:47.26 | neverblue | maybe you need to not reply unless your a provider :) |
| 15:48.00 | *** join/#asterisk Blumpkin (n=major@216.120.167.82) |
| 15:48.25 | Uatec | you're |
| 15:48.34 | Hmmhesays | asterisk pattern matching is confusing some days |
| 15:48.57 | Mercestes | Hmmhesays, how so? |
| 15:49.07 | Strom_M | neverblue: with an attitude like that, you'll make enemies real fast around here :) |
| 15:49.15 | Blumpkin | Hey all quick question. I'm having a terrible time with audio on my PBX. People here in the office hear no choppyness, but callers (or call recipiants) complain of choppyness. I'm using just IAX. Any ideas? |
| 15:49.25 | Blumpkin | I've tried different codecs and it made no difference. |
| 15:49.41 | Blumpkin | I've got lowbandwidth=on and jitterbuffer=on. |
| 15:49.59 | Strom_M | Blumpkin: so the choppy calls are on calls that go out over your DSL connection, right? |
| 15:50.08 | Blumpkin | well, T1. |
| 15:50.13 | neverblue | Strom_C, and not minding your own business is a 'good attitude' ? |
| 15:50.13 | Blumpkin | or calls coming on on the T1. |
| 15:50.31 | Blumpkin | the choppyness is always for the folks that aren't on the PBX side of things. |
| 15:50.36 | Strom_M | Blumpkin: jitterbuffering only works on packets you receive |
| 15:50.51 | Blumpkin | ah duely noted. |
| 15:50.52 | Mercestes | Strom_M: Neverblue has already made alot of enemies in here with his rather acrid charm. |
| 15:51.02 | Strom_M | you can't jitterbuffer packets you send; that has to be done on the receiving end |
| 15:51.12 | neverblue | everyone in here is 'charming' as well :) |
| 15:51.22 | Mercestes | Strom_M: So much so that I have a script that does a s/neverblue/incessent, unforgivable troll/ just so I can put him in proper perspective every time he speaks. |
| 15:51.36 | Blumpkin | strom: I'm using voicepulse connect for my VOIP needs. Is my only option to try another provider? |
| 15:51.36 | Mercestes | yonahw-work, Worse, normally. |
| 15:51.40 | Strom_M | Blumpkin: are you sharing the T1 connection with anything besides your voip calls? |
| 15:51.58 | Blumpkin | strom: Data. I've prioritized packets on the T1 for voip. |
| 15:52.11 | Blumpkin | nobody's here but one other guy today and it's still happening. |
| 15:52.28 | *** join/#asterisk JoseBravo (n=jbravo@190.9.74.174) |
| 15:52.30 | Mercestes | neverblue: Providers dont' want you. reality check |
| 15:52.36 | neverblue | Mercestes, lol |
| 15:52.44 | neverblue | your ignorant is funny |
| 15:52.51 | neverblue | its* |
| 15:53.03 | Mercestes | neverblue: Yes, please, insult my intelligence using retarded grammar. |
| 15:53.16 | [TK]D-Fender | neverblue : You've almost corrected that last sentence, keep it up! |
| 15:53.23 | neverblue | guess you havent been on irc long Mercestes |
| 15:53.30 | Strom_M | Blumpkin: run an extended ping test to voicepulse's servers and see if there's considerable jitter in the response times |
| 15:53.45 | neverblue | anyways, teaming up on me, how is that solving anything? |
| 15:53.52 | neverblue | you ppl just rant and rant and rant |
| 15:53.55 | neverblue | its boring really |
| 15:54.00 | [TK]D-Fender | Blumpkin: How many simultaneous channels do you use? |
| 15:55.06 | ManxPower | providers tend not to hang out here, as they gain nothing from it. |
| 15:55.18 | Blumpkin | we rarely ever have more than 2 calls going at once. A fella here just made a call and he was the only one and the recipient said he was barely audible. |
| 15:55.31 | ManxPower | Blumpkin: what brand of IP phones? |
| 15:55.34 | Blumpkin | pinging voicepulse I get about 53ms consistently. |
| 15:55.53 | JoseBravo | Im triying to compile wanpipe-2.3.4-12, but Im getting this error: Wanpipe header files were not installed properly in /lib/modules/2.6.18-8.1.8.el5/build/include directory |
| 15:55.57 | JoseBravo | Any idea? |
| 15:56.00 | Blumpkin | manx: Polycom Soundpoint 501 |
| 15:56.11 | [TK]D-Fender | Blumpkin: I've had some customers using them before who said that this problem evaporated immediately upon switching to SIP with them. |
| 15:56.22 | datachomper | woot, I <3 polycom 501's |
| 15:56.25 | [TK]D-Fender | Blumpkin: I would highly suggest you try the switch. |
| 15:56.27 | neverblue | really |
| 15:56.27 | ManxPower | Blumpkin: we have over 200 Polycoms, no issues with them, so it is not the actual phone you are having problems with. |
| 15:56.36 | neverblue | is that why there are atleast 6 in here? |
| 15:56.46 | neverblue | more ignorance... I laugh |
| 15:56.47 | neverblue | lol |
| 15:56.49 | ManxPower | neverblue: if they are here they don't talk much. |
| 15:57.00 | neverblue | im sure they dont |
| 15:57.08 | Blumpkin | well asterisk talks to the phones via SIP. You saying configure * to talk to voicepulse via SIP? |
| 15:57.32 | ManxPower | Blumpkin: also try jitterbuffer=no in iax.conf and see what happens. |
| 15:57.35 | Blumpkin | we have a NAT'd firewall so I always avoided using SIP. |
| 15:57.39 | Blumpkin | manx: Already have it on. |
| 15:57.48 | [TK]D-Fender | Blumpkin: Yes, Several people I've spoken with have noticed a remakable improvement. |
| 15:57.52 | Blumpkin | oh no you mean sorry. |
| 15:58.10 | Blumpkin | isn't SIP a firewall nightmare? I have to route all those ports. Ugh. |
| 15:58.16 | datachomper | How would SIP improve quality of service over IAX? Just curious |
| 15:58.19 | Blumpkin | not to mention a bandwidth hog? |
| 15:58.20 | ManxPower | Blumpkin: you have iax jitter buffer on or off? |
| 15:58.24 | Blumpkin | manx: On. |
| 15:58.31 | [TK]D-Fender | datachomper: I think its something in their service route.... |
| 15:58.35 | ManxPower | Blumpkin: I suggested you turn it OFF. |
| 15:58.40 | Blumpkin | sorry |
| 15:58.59 | [TK]D-Fender | Blumpkin: We're talking about 2 channels here... the overhead is irrelevent to your T1 |
| 15:59.00 | ManxPower | Blumpkin: in 1.2 we had nothing but problems with IAX2 jitter buffers. |
| 15:59.27 | ManxPower | Blumpkin: also if you have trunking turned on, try turning it off, but try turning off the jitterbuffer first and see if that helps. |
| 15:59.31 | Hmmhesays | say I have _00. and _009. in the dialplan _00. will be matched first right? |
| 15:59.43 | *** part/#asterisk ethemc (n=tet@88.248.14.186) |
| 15:59.47 | ManxPower | Hmmhesays: in the same context? |
| 15:59.52 | Hmmhesays | yes in the same context |
| 15:59.58 | Blumpkin | I wonder if SIP will work without firewall modifications? |
| 16:00.06 | [TK]D-Fender | Blumpkin: here : |
| 16:00.08 | [TK]D-Fender | ~sipnat |
| 16:00.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:00.09 | ManxPower | Hmmhesays: no, the most specific always tries to match first if both patterns are in the same context. |
| 16:00.19 | Hmmhesays | I thought I understood the sort order, but this dialplan here has thrown me for a loop |
| 16:00.23 | Blumpkin | should I disable IAX for testing purposes for the time beeing? |
| 16:00.35 | lirakis | Blumpkin: i have used sip in a nat setting before without issue |
| 16:00.38 | [TK]D-Fender | Hmmhesays: there is sort order, and includ order to consider.... its a little whacked... |
| 16:00.44 | ManxPower | Blumpkin: I would not suggest that unless turning the jitterbuffer off doe snot help |
| 16:00.50 | Hmmhesays | [TK]D-Fender: i know |
| 16:01.00 | [TK]D-Fender | Blumpkin: You shoul dbe able to auth either way. So comment out your IAX2, and enable SIP. |
| 16:01.16 | ManxPower | For the most part, patterns in the current context are always "more specific" than patterns in include =>'d contexts |
| 16:01.32 | Hmmhesays | ok. in the same context I have a _00XXX. and a _001701541. and the _001701541. gets matched first |
| 16:01.59 | Hmmhesays | when a call for 001701541XXXX comes in |
| 16:02.43 | Hmmhesays | doesn't that seem backwards? |
| 16:02.56 | ManxPower | It has been so long since I've had a pattern match happen when I didn't want it to, I'm a bit rusty about this. |
| 16:03.11 | Mercestes | That doesn't seem bacwards to me. =/ |
| 16:03.13 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 16:03.16 | ManxPower | Hmmhesays: no, the most specific match should be the one that happens. |
| 16:03.29 | Mercestes | _001701541. seems to be more specific than _00XXX. |
| 16:04.04 | Hmmhesays | it was my understanding that a shorter wildcard match would happen, thats why you used to have to use includes if you wanted a _X. in the same context as a _XXX. |
| 16:04.17 | ManxPower | Um, . makes the wildcard an infinite length |
| 16:04.30 | Hmmhesays | yes I know |
| 16:04.37 | Mercestes | not shorter, more specific. |
| 16:04.49 | Mercestes | and _XXX. is an infitite match so it's not shorter, it's infinite |
| 16:04.50 | ManxPower | so both _00XXX. and a _001701541. the same length |
| 16:04.55 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 16:05.11 | Mercestes | Yea, length(infinite) = length(infinite) |
| 16:05.25 | Hmmhesays | this might seem like a stupid question |
| 16:05.32 | ManxPower | they could also be written as _00XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX. and _001701541XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX. |
| 16:05.50 | Mercestes | lmao |
| 16:05.51 | Hmmhesays | but _00XXX. followed by any number of digits right? |
| 16:05.56 | ManxPower | Hmmhesays: Us in the NANPA have it easy, we don't have variable length dialplans |
| 16:06.51 | Hmmhesays | yeah, this is a goofy ass UK setup |
| 16:07.45 | Blumpkin | crap now that I disabled iax I can't dial out |
| 16:07.46 | ManxPower | Hmmhesays: both of your patterns are the same length, so the more specific one will match. |
| 16:09.07 | Hmmhesays | say I had something goofy like _00. and _001701541NNN. and a call came in for 1701 blah |
| 16:09.22 | gammah | Blumpkin: why'd you disable iax? |
| 16:09.37 | Blumpkin | dealing with a lot of choppyness on the other end. |
| 16:09.40 | Blumpkin | trying just SIP. |
| 16:09.50 | Blumpkin | but my dialplan tries to dial out via IAX. |
| 16:09.58 | Hmmhesays | ot _X. and _001701541NNN. and a call came in for 1701 blah |
| 16:10.23 | [TK]D-Fender | Blumpkin: Well I guess you should change that, now shouldn't you? ;) |
| 16:10.57 | ManxPower | assuming the call came in for something that matched _001701541NNN. then it would be what matched. |
| 16:11.37 | mocker | Ugh, horrible delays w/ asterisk odbc. |
| 16:12.14 | mocker | er, odbc voicemail storage. |
| 16:14.35 | Blumpkin | this is odd, I think SIP is working better |
| 16:14.48 | Blumpkin | I didn't even add ports on the firewall |
| 16:15.43 | lmadsen | shouldn't really need to |
| 16:15.47 | lmadsen | I never do on my linksys devices |
| 16:16.05 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:16.10 | Blumpkin | would SIP with 4 channels on a T1 that is also used for data be pushing it? |
| 16:16.32 | ManxPower | Blumpkin: NO!!!!!!!!!!!! |
| 16:16.43 | ManxPower | And you can tell that by the jitter in the pings |
| 16:17.05 | ManxPower | Blumpkin: what codec are you using? |
| 16:17.27 | Blumpkin | ulaw |
| 16:17.46 | Blumpkin | I tried them all with IAX and it made no difference. Some sounded worse than others |
| 16:20.24 | Blumpkin | can you use RSA authentication with SIP too for added security? Or is SIP in itself already secure? |
| 16:21.29 | ManxPower | ulaw and alaw use 80Kbps it uses the most bandwidth of any of the codecs |
| 16:21.39 | Hmmhesays | but it sounds so damn good |
| 16:21.39 | ManxPower | Blumpkin: if you want secure, use a VPN |
| 16:22.18 | Blumpkin | manx: So at 80Kbps, it would saturate a T1 with 3 concurrent channels, right? |
| 16:22.25 | ManxPower | Um, no. |
| 16:22.30 | mvanbaak | no |
| 16:22.47 | Blumpkin | oh maybe I'm thinking 800Kbps |
| 16:22.50 | ManxPower | A T-1 has 1536Kbps |
| 16:22.52 | mvanbaak | T1 is 24 channels of 64kbps right ? |
| 16:23.06 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-70-107-175-93.ny325.east.verizon.net) |
| 16:23.08 | Sci_05 | right |
| 16:23.35 | pigpen | Ok..here is an odd question. I am asking it here because you all are -SO- knowledgeable. :) |
| 16:23.39 | ManxPower | Do you REALLY have a T-1, or do you have DSL? |
| 16:23.56 | pigpen | I need to setup a VPN from the US to Zhongshan City, GuangDong, China |
| 16:24.12 | pigpen | what is the max encryption I an use? I prefer AES256. |
| 16:24.41 | mvanbaak | depends on the gear and stuff |
| 16:24.56 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 16:24.58 | pigpen | Gear is no issue. I am referring to Laws. |
| 16:25.01 | mvanbaak | smack an openbsd box on both sides and read "man ipsec.conf" |
| 16:25.41 | pigpen | No..I know what what I am doing there...I am referring to the issues with the US and Chineese laws. |
| 16:26.04 | mvanbaak | ah |
| 16:26.14 | mvanbaak | then it's prolly XOR only |
| 16:26.21 | pigpen | I setup vpns for a living, I am just unsure of the encryption level. |
| 16:26.27 | pigpen | nice. |
| 16:26.41 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 16:26.46 | explidous | I cannot select DETECT_DEADLOCKS in the SVN trunk 77858 |
| 16:27.05 | mvanbaak | explidous: it became dependent on some other setting |
| 16:27.05 | explidous | even though I have DEBUG_THREADS |
| 16:27.09 | mvanbaak | ah |
| 16:27.10 | mvanbaak | yeah |
| 16:27.13 | mvanbaak | that's it |
| 16:27.15 | Qwell[] | works here O.o |
| 16:27.29 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:27.31 | mvanbaak | it works here too |
| 16:27.37 | Qwell[] | explidous: try doing a make distclean, then configure and then make menuselect |
| 16:27.41 | Qwell[] | something might've gotten hosed up |
| 16:27.47 | explidous | k |
| 16:28.05 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 16:30.45 | *** join/#asterisk saftsack (n=saftsack@pD9E07A84.dip.t-dialin.net) |
| 16:32.24 | explidous | same error |
| 16:32.28 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 16:32.53 | explidous | The existing menuselect.makeopts file did not specify that 'DETECT_DEADLOCKS' should not be included. However, either some dependencies for this module were not found or a conflict exists. |
| 16:33.10 | explidous | could DETECT_DEADLOCKS or DEBUG_THREAD be conflicting with something? |
| 16:33.54 | ManxPower | explidous: try asking on #asterisk-dev |
| 16:34.03 | putnopvut | explidous: I've seen that with other options. What I do is run make menuselect. |
| 16:34.08 | putnopvut | Then exit with saving. |
| 16:34.15 | putnopvut | Then try make && make install. |
| 16:34.53 | *** join/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 16:34.54 | explidous | right when I go back into menuselect DETECT_DEADLOCKS is no longer set |
| 16:35.03 | explidous | all that does is remove the offender |
| 16:36.00 | obiwanmikenolte | Has anyone had a problem with voicemail where forwarding a message to another mailbox sends a blank message (length 0:00)? |
| 16:36.37 | Blumpkin | what file holds the text emailed with voicemail attachments? |
| 16:37.17 | explidous | voicemail.conf |
| 16:37.51 | Blumpkin | doh! thn |
| 16:37.52 | Blumpkin | x |
| 16:38.05 | explidous | np |
| 16:39.34 | [TK]D-Fender | bkruse: You about? |
| 16:40.44 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:44.46 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:44.48 | MrTelephone | hey |
| 16:45.00 | MrTelephone | anyone know if you can silence the ring per line on the polycom 501's? |
| 16:45.18 | MrTelephone | Bell canada megalink service doesn't offer ANI2 service. They said it was an american thing :-/ |
| 16:45.31 | MrTelephone | And they said they won't block collect calls |
| 16:47.32 | ManxPower | find another provider |
| 16:47.49 | ManxPower | or make all calls get answered with "we will NOT accept collect calls", then process the call normally. |
| 16:48.32 | Blumpkin | Anyone using voicepulse? I wonder what happens when I dial 911. |
| 16:48.40 | Blumpkin | I'd hate to have a heart attack and find out the hard way. |
| 16:49.29 | MrTelephone | when somoene makes a collect call, does the ani=the switchboard number? |
| 16:49.59 | MrTelephone | blumpkin, what does it say on their website? |
| 16:51.40 | Blumpkin | nothing. Maybe I'll call their tech support. |
| 16:51.51 | MrTelephone | i got heat stroke building a fence on the weekend, hot weather sucks |
| 16:51.55 | *** join/#asterisk matdon (n=matdon@eagle.bsd.st) |
| 16:52.01 | ^majik^ | hm.. right now, if a second person calls in, it goes straight to voicemail instead of ringing the other Zap channels in the Dial command. any ideas? I'm using the 'r' option |
| 16:52.29 | MrTelephone | majik, apply group numbers |
| 16:52.32 | obiwanmikenolte | majik: are you using a call group? |
| 16:52.33 | MrTelephone | to the zap channels |
| 16:52.48 | MrTelephone | group=3 |
| 16:52.50 | obiwanmikenolte | Word^ |
| 16:52.54 | ^majik^ | I thought I was. don't I set that up in zapata.conf? |
| 16:52.58 | MrTelephone | yeah |
| 16:53.25 | MrTelephone | group=>1 before your channel=> entry |
| 16:53.34 | MrTelephone | then Dial(ZAP/g1) in your extensions.conf |
| 16:53.54 | ^majik^ | oh, I see |
| 16:54.33 | MrTelephone | either that or you have to make a macro and have a fall through script like macro-stdexten |
| 16:59.15 | Blumpkin | thnx for the help today all! |
| 16:59.23 | Blumpkin | much appreciated. |
| 17:00.19 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 17:03.24 | *** join/#asterisk mattchis (n=mattchis@216.54.143.246) |
| 17:03.57 | mattchis | I have a question.... Can I set gains to a negative value in the zapata.conf |
| 17:06.20 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 17:06.34 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 17:06.41 | [TK]D-Fender | MrTelephone: Not "directly" that I know of, but you can change the ring-tone to a silent one. |
| 17:06.50 | [TK]D-Fender | mattchis: Yes |
| 17:06.58 | masus | asterisk 1.4.9 + debian installation problem after ./configure -->configure: error: C++ preprocessor "/usr/bin/cpp" fails sanity check |
| 17:06.58 | masus | See `config.log' for more details. |
| 17:07.02 | [TK]D-Fender | mattchis: Many people do this for EC reasons |
| 17:07.04 | masus | can anybody help |
| 17:07.51 | nybble | hey guys, i hope i'm in the right room. could anyone give me a quick runthrough on using the asterisk manager api? |
| 17:09.08 | masus | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 17:10.12 | centrex | MrTelephone, out of curiosity, why would you use group => 1 instead of group=1? I've seen both work, and I'm sure I've seen channel=1-10 and channel => 1-10 work, anyone know if there's any real difference? |
| 17:10.44 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:11.08 | MrTelephone | no |
| 17:11.16 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 17:11.25 | MrTelephone | what happens in the code is that they split the key,value pairs using more than one search string |
| 17:11.38 | MrTelephone | the code looks for = or => |
| 17:12.01 | MrTelephone | you know what I mean? |
| 17:12.03 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 17:12.03 | centrex | That's what I thought, I had just always read to use =>, and seen = working, just wondering. |
| 17:12.08 | MrTelephone | me too |
| 17:12.16 | MrTelephone | i noticed both worked tho |
| 17:12.27 | MrTelephone | I remember reading where people had trouble grouping channels |
| 17:12.37 | MrTelephone | channels=10-15,22 |
| 17:12.48 | MrTelephone | and they had to put channels=1 |
| 17:12.50 | MrTelephone | channels=2 |
| 17:12.53 | MrTelephone | wierd |
| 17:13.06 | MrTelephone | maybe it was a config problem on the users side |
| 17:13.21 | centrex | weird |
| 17:13.37 | Yourname` | Hi. I have an inbound box that transfers a call via sip to extension 1012 on the inbound box. But today, it's been giving circuit-busy/congestion. What are the most common causes of it doing that? When calls go to 1012, agents from 1000 to 1010 ring. So it can't REALLY be busy. And at the time this happened, only 3 agents were on call. What am I doing wrong? |
| 17:14.22 | MrTelephone | call-limit on 1012? |
| 17:15.00 | MrTelephone | check for call-limit on the sip account you use to send the calls on |
| 17:15.22 | gammah | I have a question about compiling asterisk... |
| 17:15.29 | MrTelephone | say it don't spray it |
| 17:15.32 | ^majik^ | MrTelephone: I just tried grouping the channels using 'group' and then Zap/gN. it only rings the first channel in the group |
| 17:16.05 | MrTelephone | majik, what happens when the first channel is in use? |
| 17:16.07 | gammah | why, when I specify ./configure --prefix=/some/weird/path/on/my/system , does the binary *still try and read it's configs from /etc/asterisk, instead of /some/weird/path/on/my/system/etc/asterisk ? |
| 17:16.28 | gammah | I know I can specify an alternate config file on cmdline |
| 17:16.29 | ManxPower | As I understand it (this may have been fixed) call-limit does not get reset on a transfer, so if the user transfers a call their phone will still be busy |
| 17:16.42 | gammah | seems strange to me -- even the logs and everything are read from the normal locations |
| 17:17.06 | MrTelephone | logs are always in /var/logs though |
| 17:17.24 | centrex | ^majik^, Yes, that's what grouping does. It only dials out the first available channel with that dial command. |
| 17:17.24 | MrTelephone | the configs should always be in /etc/ |
| 17:17.35 | creativx | screw this, im going home. |
| 17:17.40 | Yourname` | MrTelephone: I just looked, and I don't know what I'm looking at.. :S |
| 17:17.41 | ^majik^ | MrTelephone: ah.. hadn't tried it. all we want is for all phones to ring, even if one phone is in use. (I suppose we could use what we just tried, but we were used to the old setup) |
| 17:18.04 | ^majik^ | centrex: gotcha |
| 17:18.07 | MrTelephone | oh |
| 17:18.08 | ManxPower | ^majik^: Zap/g1 is a HUNT group. |
| 17:18.25 | ManxPower | if you want parallel dial you do Dial(Zap/1&Zap/2&SIP/abdc) |
| 17:18.26 | MrTelephone | to dial all phones you have to Dial(ZAP/1&ZAP/2&ZAP/3) |
| 17:18.29 | centrex | ^majik^, : There might be a better way to do this, but Dial(zap/1&zap/2&zap/3) should work (just obviously add all the channels. |
| 17:18.49 | centrex | ) |
| 17:19.08 | ManxPower | See, when you use the correct terms people can help you. 8-) |
| 17:19.20 | ^majik^ | ManxPower, MrTelephone: that's what were doing before all of this :) I don't know why it would go to voicemail instead of ringing the others |
| 17:19.27 | datachomper | Is there a way to have two extensions connect to each other via CLI? |
| 17:19.35 | ^majik^ | s/were/we were/ |
| 17:19.37 | *** join/#asterisk lineman60 (n=tech@azerial.fastwave.biz) |
| 17:19.39 | ManxPower | datachomper: no |
| 17:20.01 | MrTelephone | majik: you got a problem on your hands, lol |
| 17:20.07 | *** part/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 17:20.07 | ManxPower | In a parallel dial, it will only go to the next priority if ALL destinations are busy. |
| 17:20.26 | MrTelephone | maybe majik found a bug? |
| 17:20.26 | ManxPower | It does not go to voicemail unless you tell it to by checking the value of HANGUPCAUSE or DIALSTATUS |
| 17:20.32 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 17:20.54 | Katty | boss is going to go with the sangoma echo cancelation t1 card. |
| 17:20.59 | MrTelephone | majik, paste your dial line |
| 17:21.06 | Katty | and he's going to argue with the ISP about getting the pri at the same price as a t1 (= |
| 17:21.13 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:21.18 | MrTelephone | katty, pris are usually cheaper than data t1s |
| 17:21.18 | ^majik^ | I was just going by priority, eg: s,102,Dial(Zap/1&Zap/2,15,r)...s,103,Voicemail(...) |
| 17:21.21 | ManxPower | Katty: I think your argue with the telcos about that. |
| 17:21.27 | Katty | ManxPower: no, it's our isp. |
| 17:21.29 | ManxPower | ^majik^: DON'T USE 102 |
| 17:21.33 | Katty | ManxPower: they're an at&t reseller. |
| 17:21.36 | Katty | ManxPower: trust me (= |
| 17:21.46 | Katty | MrTelephone: i agree. but for some reason, they want to charge us twice as much as a t1 for a pri |
| 17:21.46 | ManxPower | Katty: it sucks to be you. |
| 17:21.51 | Katty | MrTelephone: it's absolutely stupid. |
| 17:21.52 | ^majik^ | but I used a GotoIf, so I figure 102 is okay, for how its setup at the moment anyways |
| 17:21.53 | Katty | ManxPower: oh hush :P |
| 17:21.54 | MrTelephone | katty, don't skimp out and get a dual port or something, its nice having an extra port for other t1 appliances |
| 17:22.09 | ManxPower | ^majik^: gotoif w/102 is ok, but still not a good idea as it will confuse people |
| 17:22.22 | ^majik^ | ManxPower: how should it be? |
| 17:22.35 | Yourname` | ~call-limit |
| 17:22.42 | Yourname` | ~calllimit |
| 17:22.47 | ManxPower | ^majik^: look at [macro-std-exten] on extensions.conf.sample |
| 17:22.50 | MrTelephone | yourname, i forgot about you :-/ |
| 17:23.10 | Yourname` | lol it's ok, while I was waiting for your answer I was looking around, but can't find anything. |
| 17:23.18 | MrTelephone | yourname, you added something to sip.conf for your incoming calls via sip? |
| 17:23.25 | ManxPower | Yourname`: try sip.conf.sample in the asterisk source |
| 17:23.38 | ^majik^ | ManxPower: ok, I just hope I still have the sample :) it's almost lunch time though |
| 17:23.55 | Yourname` | k |
| 17:23.58 | ManxPower | ^majik^: it would be in the asterisk source code dir under configs/ |
| 17:24.20 | MrTelephone | yeah he should be testing dialstatus instead of jumping to voicemail on the next priority |
| 17:24.22 | MrTelephone | heh |
| 17:24.37 | Yourname` | MrTelephone: Can I use call-limit for friend? |
| 17:25.19 | MrTelephone | majik.. just use exten=s,1,macro(stdexten,voicemailbox,ZAP/1&ZAP/2&ZAP/3) |
| 17:25.32 | ^majik^ | ManxPower: I'm just using AsteriskNow with the GUI disabled. it was the painless way to get the latest asterisk going with Zaptel |
| 17:25.47 | Yourname` | MrTelephone: But also, I don't really know about the call-limit, because this is the thing. When the first call came in, almost no agent was on call. YET, we got circuit-busy/congestion. |
| 17:25.55 | ManxPower | ^majik^: Then you are out of luck when it comes to me helping you. |
| 17:25.55 | MrTelephone | manx, does sip report busy/conjested on auth failure? |
| 17:26.09 | ManxPower | MrTelephone: CONGESTION I suspect |
| 17:26.16 | Yourname` | Hmm. |
| 17:26.39 | MrTelephone | yourname, ur getting conjestion when noone is even on the line? |
| 17:26.50 | MrTelephone | you may have an authentication problem |
| 17:27.00 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:27.44 | MrTelephone | i was never a fan of GUI because it makes it much harder to see whats happening behind the scenes |
| 17:28.11 | *** join/#asterisk joshua_d (n=admin@stealth.tek-hut.com) |
| 17:28.31 | Yourname` | Umm, I'm not on GUI, lol. |
| 17:28.33 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 17:28.39 | Yourname` | I'm doing it plain asterisk style. |
| 17:28.39 | ManxPower | I suspect Yourname`'s problem is with the way call limits work. |
| 17:28.50 | ManxPower | We never ever use them because of that. |
| 17:29.01 | Yourname` | ManxPower: Not using call-limit at all. |
| 17:29.04 | MrTelephone | how do you limit calls |
| 17:29.05 | MrTelephone | then |
| 17:29.11 | Yourname` | I didn't even know it till MrTelephone mentioned it. |
| 17:29.14 | Yourname` | MrTelephone: We don't. |
| 17:30.30 | MrTelephone | yourname are you making calls between asterisk boxes? |
| 17:30.45 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:31.00 | hi365 | anyone here fro switzerland? |
| 17:31.40 | MrTelephone | the land of chocolate? |
| 17:31.49 | datachomper | I was going to say that land of hot babes, but ok |
| 17:31.55 | Yourname` | MrTelephone: Outbound call places a call, and when connected.. transfers a call to 1012@inboundbox, and when it goes there, it rings all to agents from 1000 to 1010. |
| 17:32.00 | Yourname` | Outbound box* |
| 17:32.07 | generalhan | how bout the land of clocks ? |
| 17:32.16 | ManxPower | Yourname`: Did you not see my comment about transfers and call limit |
| 17:32.36 | zdsl | I am back |
| 17:32.38 | Yourname` | ManxPower: But we don't use it. |
| 17:32.45 | ManxPower | Yourname`: OK. |
| 17:32.50 | Yourname` | ManxPower: You said it could be causing trouble because we use it, but we don't. |
| 17:33.18 | zdsl | Does anyone use Sangoma for there hardware? |
| 17:33.19 | MrTelephone | is asterisk snmp reliable? |
| 17:33.23 | ManxPower | I also seem to recall something about agents in queues transfering calls having the same issue, but don't ask me to cite my source. |
| 17:33.24 | MrTelephone | zdsl yeah |
| 17:33.28 | ManxPower | zdsl: yes |
| 17:33.46 | zdsl | I have heard it is much better than Digium |
| 17:33.52 | MrTelephone | will asterisk report the max calls for the last 30mins, 1 hour, 1 day sort of thing? |
| 17:34.00 | MrTelephone | zdsl, both products are good |
| 17:34.18 | MrTelephone | sangoma is supposed to be a little better on resources but i still had problems with mine |
| 17:34.29 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:35.02 | zdsl | MrTelephone: I was just getting a general concensus I have contacted both vendors |
| 17:35.13 | ManxPower | Sangoma is also supposed to have a better compatibility track record. |
| 17:35.21 | ManxPower | (with different motherboards) |
| 17:35.27 | MrTelephone | yeah |
| 17:35.32 | MrTelephone | they support many types of pci slots |
| 17:35.37 | MrTelephone | for example the pci 64 bit |
| 17:35.43 | kiscokid | Mr. T: what kind of problems did you have with Sangoma cards? |
| 17:36.02 | ManxPower | MrTelephone: Asterisk reports all calls in /var/log/asterisk/cdr-csv |
| 17:36.04 | MrTelephone | it loses clocking after a week or two.. but i can't blame it on the card because it could be the machine its in |
| 17:36.16 | ManxPower | kiscokid: they are a bitch to install the drivers |
| 17:36.23 | MrTelephone | manx, i see, make a script to figure out the times |
| 17:36.26 | [TK]D-Fender | ManxPower: Never in my experience..... |
| 17:36.33 | anonymouz666 | ~cisco |
| 17:36.33 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
| 17:36.39 | anonymouz666 | hahaha |
| 17:36.44 | ManxPower | [TK]D-Fender: In my experience it is. |
| 17:37.04 | [TK]D-Fender | ManxPower: Could be Mandrake's Kernel funkyness..... |
| 17:37.12 | ManxPower | I never could get their drivers installed on a 2.4 kernel. I had to comment out some of the utilities from building on our most recent install |
| 17:37.36 | MrTelephone | i don't know why sangoma asks to install native zaptel support because if you don't nothing works |
| 17:37.37 | kiscokid | Manx: I installed the Sangoma 4 port fxo/fxs card with no driver issues |
| 17:37.42 | ManxPower | [TK]D-Fender: zaptel never has a problem installing |
| 17:37.49 | ManxPower | /building |
| 17:37.54 | [TK]D-Fender | ManxPower: One of my clients is on Slack 2.4 + A200. Wroked 100% fine... |
| 17:38.00 | Katty | [TK]D-Fender: YOU |
| 17:38.04 | Katty | [TK]D-Fender: why i outta... |
| 17:38.05 | Katty | [TK]D-Fender: just.. |
| 17:38.06 | [TK]D-Fender | ManxPower: Agreed on Zaptel for sure... its never really a problem... |
| 17:38.08 | MrTelephone | fxs or fxo? |
| 17:38.10 | Katty | [TK]D-Fender: HUG YOU |
| 17:38.14 | [TK]D-Fender | :D |
| 17:38.21 | kiscokid | fxo |
| 17:38.22 | ManxPower | and I have had to frequently downgrade zaptel to something that is not the most recent for the Sangoma drivers to patch Zaptel |
| 17:38.27 | MrTelephone | the problem is sangoma needs zaptel and its own drivers.. twice the chance of failure |
| 17:38.56 | ManxPower | But ALL new T-1 cards we have been buying for the past year are Sangoma, we are phasing out Digium cards. |
| 17:39.00 | datachomper | Mandrake is still around? |
| 17:39.01 | MrTelephone | spent 1 day trying to figure out why my adit 600 wasn't working.. turned out to be bad RBS bits |
| 17:39.03 | ManxPower | Mostly because of motherboard issues |
| 17:39.04 | [TK]D-Fender | MrTelephone: Well... only a problem if somethat actually goes wrongs, so its merely a POTENTIAL point of failure... |
| 17:39.15 | kiscokid | but I used a 2.6 kernal |
| 17:39.20 | ManxPower | datachomper: they are called Mandriva now, which to me sounds vaguely pornographic |
| 17:39.26 | datachomper | rofl |
| 17:40.10 | ManxPower | Sangoma also did not support DACS RBS until I complained to them about it. They HAVE been very responsive the few times I've had to contact support |
| 17:40.18 | *** join/#asterisk marcan (n=marcanso@10.Red-88-24-250.staticIP.rima-tde.net) |
| 17:40.41 | [TK]D-Fender | ManxPower: Nenad = God. |
| 17:41.21 | ManxPower | ALL our Sangoma cards are A102Ds |
| 17:42.09 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 17:42.37 | [TK]D-Fender | ManxPower: I thought you only ran A101/102 w/ tellabs... |
| 17:42.58 | Katty | wow, this smbclient command i've got running is really annoying |
| 17:43.14 | Katty | phone call -> popup, phone call -> popup, ringring -> annoying annoying, go away |
| 17:43.49 | Hmmhesays | heh |
| 17:44.03 | [TK]D-Fender | Katty: "Mission Accomplished" |
| 17:44.12 | Katty | [TK]D-Fender: oh yes. |
| 17:44.25 | Katty | [TK]D-Fender: it is my daily goal to annoy the living crap out of our recptionist. |
| 17:44.28 | Katty | [TK]D-Fender: this will do nicely. |
| 17:44.42 | [TK]D-Fender | Katty: I could do FAR worse :D |
| 17:45.00 | [TK]D-Fender | Katty: tt-weasels as her ring-tone :D |
| 17:45.05 | datachomper | lol |
| 17:45.05 | Katty | [TK]D-Fender: ah hahahahahha |
| 17:45.10 | Katty | [TK]D-Fender: you're awful. |
| 17:45.13 | Hmmhesays | fun |
| 17:45.13 | Katty | [TK]D-Fender: just plain awful. |
| 17:45.28 | [TK]D-Fender | You know only YESTERDAY did I play tt-weasels & tt-monkeys for the first time! |
| 17:45.33 | Qwell[] | ... |
| 17:45.37 | Qwell[] | [TK]D-Fender: You're fired. |
| 17:45.39 | Katty | whaaaaat?! |
| 17:46.11 | Qwell[] | [TK]D-Fender: occasionally, PQ (who are right outside my office) will play tt-weasels on a loop. |
| 17:46.12 | ManxPower | [TK]D-Fender: It's whatever the one is without EC |
| 17:46.15 | Qwell[] | like, all day |
| 17:46.20 | ManxPower | and we only get the 2-port cards |
| 17:46.24 | Qwell[] | it's pretty much the most awesome thing ever |
| 17:46.26 | J4k3 | don't all pbx programmers start their carreers out as people that strive to annoy end users? :) |
| 17:46.41 | ManxPower | [TK]D-Fender: forward the polycom to it'self |
| 17:46.45 | J4k3 | "press 2 for spanish" |
| 17:46.50 | [TK]D-Fender | Qwell[]: To whom are you referring exactly? |
| 17:46.50 | ManxPower | then she will get a hundred calls and popups |
| 17:47.05 | Qwell[] | to whom what? |
| 17:47.11 | [TK]D-Fender | ManxPower: She wants to annoy her receptionst, not crash her system ;) |
| 17:47.33 | [TK]D-Fender | Qwell[]: about "PQ" |
| 17:47.34 | zdsl | Katty: did you find a solution for telephone conference / video conference solution? |
| 17:47.39 | Qwell[] | our PQ dept? |
| 17:47.42 | ManxPower | [TK]D-Fender: Some idiot forwarded their phone to themselves a week or so ago. |
| 17:47.46 | Qwell[] | I don't understand the question. |
| 17:47.54 | J4k3 | ManxPower: sweet. |
| 17:47.55 | [TK]D-Fender | Qwell[]: Abbreviation of...?" |
| 17:47.56 | J4k3 | haha |
| 17:47.58 | Qwell[] | product quality |
| 17:47.59 | datachomper | ManxPower, awesome, what happened? |
| 17:48.05 | Qwell[] | or, qualification? I don't know |
| 17:48.06 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 17:48.10 | ManxPower | datachomper: Asterisk did not crash - thank Dog |
| 17:48.13 | Katty | zdsl: no, i'm still attempting to download the debian cd :P |
| 17:48.15 | J4k3 | the local CO tech wasn't amused. |
| 17:48.20 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:48.25 | iPod-nano | Do you have to create the asterisk configuration files after a fresh install? |
| 17:48.28 | datachomper | ManxPower, Did asterisk figure out what was happening or what? |
| 17:48.30 | ManxPower | I once crashed the bellsouth corporate e-mail system |
| 17:48.32 | [TK]D-Fender | Qwell[]: Proper form would be "QC" |
| 17:48.33 | ManxPower | accidently |
| 17:48.45 | Qwell[] | q must be qualification then |
| 17:48.47 | ManxPower | datachomper: I think so, things were spewing on the console so fast... |
| 17:49.12 | [TK]D-Fender | Qwell[]: Either of which were official Postal Code abbreviations for QUEBEC which is what I was getting confused on in the fisrt place :) |
| 17:49.39 | Qwell[] | yeah, we have a whole dept dedicated to shipping to Quebec :P |
| 17:49.44 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 17:51.34 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:51.36 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 17:52.18 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 17:54.21 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 17:55.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:57.06 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 17:57.13 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 17:57.32 | Yourname` | <PROTECTED> |
| 18:01.09 | datachomper | Off topic, but how do I view a list of active leases in a dhcpd server? |
| 18:02.38 | Yourname` | I don't quite know this for sure, but I *think* it should be in /var/lib/dhcpd/dhcpd.leases ? |
| 18:03.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:05.21 | generalhan | ok, so i ran 'make config' to get the /etc/init.d/asterisk file, i chkconfig on to make it run ... after reboot i check 'ps' and safe_asterisk is running yet when i do asterisk -r it says its not running ? |
| 18:06.05 | Strom_M | are you root? |
| 18:06.09 | generalhan | yea |
| 18:06.21 | Strom_M | if you kill safe_asterisk and then type asterisk -cvvvvvvg |
| 18:06.25 | Strom_M | does asterisk start cleanly? |
| 18:06.34 | generalhan | lol ... well no |
| 18:06.49 | *** join/#asterisk NirS (n=Nir@87.68.170.190) |
| 18:07.01 | generalhan | cause i dont have my modprobes in a config file yet ... so i get the asterisk ended with error code 1 |
| 18:07.07 | [TK]D-Fender | generalhan: safe_asterisk is just a script that allows asterisk to crash in a loop ;) |
| 18:07.37 | Yourname` | That's probably what's happening with me! |
| 18:07.41 | generalhan | well then what is the point in having this config file made for me so that i can run service asterisk start ? if it really doesnt start ! |
| 18:11.13 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 18:11.19 | datachomper | Yourname` : Thanks bro :O) |
| 18:11.33 | neuwald | what command substitute DBget on asterisk 1.4 ? |
| 18:12.35 | Strom_M | neuwald: the DB() function |
| 18:12.36 | [TK]D-Fender | neuwald: "show function DB" |
| 18:12.42 | Strom_M | read UPGRADE.txt |
| 18:12.44 | Yourname` | You're welcome datachomper |
| 18:14.24 | Yourname` | Hello. -- Got SIP response 482 "Loop Detected" back from 64.34.139.11 -> What am I doing wrong now?! lol.. this happens when all agents are on call. |
| 18:15.44 | neuwald | Ok. I used before: exten => _1.,2,DBget(passkey=auth_user/${CALLERIDNUM}) |
| 18:15.47 | [TK]D-Fender | Yourname`: it means exactly what it says |
| 18:15.49 | neuwald | now, how should I do ? |
| 18:16.02 | [TK]D-Fender | neuwald: you should read "asterisk functions" on the WIKI |
| 18:16.04 | [TK]D-Fender | ~wikis |
| 18:16.04 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 18:16.25 | [TK]D-Fender | neuwald: Because that var is toast too. |
| 18:16.32 | Yourname` | [TK]D-Fender: I know, but why does it happen when all agents are on call? :S |
| 18:17.30 | [TK]D-Fender | Yourname`: Considering I'm on psychic on TUESDAYS and you haven't shown us your configs & CLI output.... not a clue. |
| 18:18.18 | Yourname` | lol one sec. |
| 18:19.07 | Strom_M | PSYCHIC POUTINE |
| 18:19.48 | [TK]D-Fender | Strom_M: bien-oui mon ostie! |
| 18:20.15 | Strom_M | Je ne me souviens compose mauvais numero |
| 18:21.00 | *** join/#asterisk saftsack (n=saftsack@pD9E07A84.dip.t-dialin.net) |
| 18:22.25 | Yourname` | [TK]D-Fender: sip.conf http://pastebin.ca/642073 |
| 18:22.56 | Yourname` | [TK]D-Fender: extensions.conf http://pastebin.ca/642074 |
| 18:22.57 | [TK]D-Fender | Strom_M: thats the problem with translation tools... you can get the spelling right ands till make no sense due to changes in sentence structure ;) |
| 18:23.15 | Strom_M | that's no translation tool |
| 18:23.24 | Strom_M | that's my horrid understanding of French grammar |
| 18:23.42 | Yourname` | [TK]D-Fender: extensions_more.conf http://pastebin.ca/642075 |
| 18:23.46 | [TK]D-Fender | Yourname`: Oh shit.. you're using 1 account to like to 3 other systems |
| 18:24.09 | Yourname` | Basically same configs.. :S |
| 18:24.29 | [TK]D-Fender | Yourname`: splait those up. they should be 3 different setups... |
| 18:24.54 | Yourname` | You mean the registers, right? |
| 18:25.13 | [TK]D-Fender | Yourname`: you also have a singe friend entry to match them.... HORRID! |
| 18:25.19 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:25.25 | [TK]D-Fender | Yourname`: make them all unique |
| 18:25.30 | Yourname` | ok |
| 18:26.00 | *** join/#asterisk pigpen2 (n=pigpen@015-840-551.area5.spcsdns.net) |
| 18:26.06 | [TK]D-Fender | Strom_M: Are you at all "function" in french or any other language than english? |
| 18:26.13 | pigpen2 | hi all. I have a fun one. |
| 18:26.17 | pigpen2 | Please reference: http://pastebin.ca/642077 |
| 18:26.33 | Strom_M | [TK]D-Fender: I'm passably decent in Spanish |
| 18:26.50 | [TK]D-Fender | Strom_M: Already ahead of the curce then.... |
| 18:26.57 | pigpen2 | In short, after increasing the default string length from 256 to 8000, when I all page page past a certian number of phones, asterisk dies. |
| 18:27.04 | pigpen2 | Detail is in the pastebin. |
| 18:27.25 | *** join/#asterisk kempist (n=jh@200.69.198.45) |
| 18:27.38 | [TK]D-Fender | pigpen2: Holy McFuck..... |
| 18:27.40 | pigpen2 | I would -like- the ability to do a all page to all phones, but it dies after around 108...consistantly. |
| 18:27.47 | pigpen2 | [TK]D-Fender, :) |
| 18:27.51 | Yourname` | [TK]D-Fender: So.. is that the reason why I'm getting loop detected? |
| 18:28.01 | [TK]D-Fender | Yourname`: Great odds. |
| 18:28.15 | Yourname` | [TK]D-Fender: What do you mean? :S |
| 18:28.26 | [TK]D-Fender | pigpen2: Thats a nasty load on a MeetMe as well.... you really should consider off-line paging... |
| 18:28.35 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 18:28.55 | [TK]D-Fender | Yourname`: what it sounds like. If I had to bet on what the origin of your problems, is, that'd be it. |
| 18:29.45 | [TK]D-Fender | pigpen2: Try spliiting your page in 2 using 2 local channels. |
| 18:29.50 | Yourname` | Ok, so basically, you say it should not register as the same extension on different boxes. And then I didn't get the friend part.. can you say that again please? |
| 18:30.11 | [TK]D-Fender | Yourname`: The friend matches your register which is what screws stuff up. |
| 18:30.23 | [TK]D-Fender | Yourname`: the accounts need different entries. |
| 18:30.32 | Yourname` | For different boxes, right? |
| 18:30.33 | [TK]D-Fender | Yourname`: jsut start splitting them up ok? |
| 18:30.38 | [TK]D-Fender | Yourname`: yes |
| 18:30.41 | Yourname` | Ok. |
| 18:30.58 | Yourname` | I just wanted to make it easy to "scale" to new boxes. |
| 18:30.58 | generalhan | COOL BEANS !! that config file for asterisk runs both safe_asterisk AND the -c -vvvvvvvv -g as long as i modprobe first !! so NOW i see the point ! |
| 18:31.37 | [TK]D-Fender | generalhan: Yes, starting zaptel makes all the difference, doesn't it? :) |
| 18:31.46 | kempist | Hello, got a problem with teliax where i dont know if they are sending the wrong number for gsm codec or when i updated asterisk the other day something changed: http://pastebin.com/m3fe07bd5 has part of the log, anyone has any clue to where might be the problem, my ast or teliax? |
| 18:32.00 | Yourname` | [TK]D-Fender: Just so I can just copy the config to another box, and bam, it works. |
| 18:32.28 | pigpen2 | sorry, on phone. |
| 18:32.51 | generalhan | lol ... i was looking through the config and didnt see an If zaptel is loaded lines ... so i didnt think it had any bearing on how that config file started asterisk ! but all is well now .. all i have to do is chkconfig a file to modprobe zaptel and wct4xxp before the asterisk file and i should be set ! |
| 18:32.57 | pigpen2 | [TK]D-Fender, would you mind elaborating off-line paging and 2 local channels. |
| 18:33.12 | [TK]D-Fender | pigpen : callfile / ami vs Page |
| 18:33.49 | [TK]D-Fender | pigpen2: and for the 2x Local channel, rather than 1 super-long page, have it split 2 local channels each one doing HALF. |
| 18:33.50 | pigpen2 | Well call files are cool, but don't I need to dump them into a meetme still? |
| 18:34.21 | [TK]D-Fender | pigpen2: tahts the "offline" part. Make a recording, and jsut do a playback. No need for meetme. |
| 18:35.06 | [TK]D-Fender | pigpen2: I know you said you didn't like the fractional offset that can happen, but you want this to WORK, right? |
| 18:35.22 | kempist | http://pastebin.com/d6d89afc1 <- anyone knows if 2048 is the codec number for GSM? |
| 18:35.25 | pigpen2 | well, the issue is with that, is across 150 or so phones, the audio comes across slightly off. So when it finally gets to the end, the first ones have heard the page 15 seconds prior. |
| 18:35.56 | [TK]D-Fender | pigpen2: tahts a pretty decent response time for 911.... you aren't anywhere near as important ;) |
| 18:36.08 | pigpen2 | but yes, I like the idea of pre-recorded pages. |
| 18:36.10 | [TK]D-Fender | pigpen : I'd still suggest the 2xLocal method then. |
| 18:36.29 | neuwald | quit |
| 18:36.39 | pigpen2 | I'll probably just setup both, and let them choose. |
| 18:36.53 | pigpen2 | but nevertheless, is this something I should report as a bug? |
| 18:36.57 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 18:37.10 | neverblue | is Twinkle only capable of making SIP calls? |
| 18:37.15 | pigpen2 | I will also mention that the page app works great....with 108 phones. |
| 18:39.10 | [TK]D-Fender | pigpen2: Considering you had to recode past the string-length limit, I think you've voided your warranty ;) |
| 18:39.34 | pigpen2 | Well, I was told the limit was 8000 |
| 18:40.01 | [TK]D-Fender | neverblue : "Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages." |
| 18:40.14 | [TK]D-Fender | neverblue : Well... it seems pretty clear... |
| 18:40.33 | [TK]D-Fender | pigpen2: Thats the theory.... they flunked the practical test ;) |
| 18:40.43 | pigpen2 | True. |
| 18:40.47 | pigpen2 | True. |
| 18:41.00 | pigpen2 | As always, thanks for your consult. |
| 18:41.20 | [TK]D-Fender | pigpen2: Do try my 50/50 spilt... should work (better) |
| 18:41.26 | [TK]D-Fender | split* |
| 18:41.45 | *** join/#asterisk kiscokid (n=sysadmin@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 18:41.47 | pigpen2 | So basically, do a page, then do a second page. |
| 18:42.05 | pigpen2 | or run it dual into the same meetme? |
| 18:42.08 | neverblue | ah, you have the single best attitude in the channel, thanks for your response |
| 18:42.29 | Yourname` | [TK]D-Fender: That didn't help. |
| 18:42.45 | [TK]D-Fender | pigpen2: 3 x meetme |
| 18:42.53 | pigpen2 | [TK]D-Fender, k. |
| 18:43.12 | [TK]D-Fender | pigpen2: 1 - fork 2 others. |
| 18:43.29 | pigpen2 | so 3 seperate pages. |
| 18:43.32 | pigpen2 | or 2 |
| 18:43.34 | pigpen2 | whatever. |
| 18:43.49 | [TK]D-Fender | Yourname`: you'll need to provide CLI output, etc to match, and did you update all of your systems to match? You may have to kill off your astdb registrations..... |
| 18:43.57 | [TK]D-Fender | pigpen2: First calls 2 more. |
| 18:44.07 | pigpen2 | k. |
| 18:44.13 | Yourname` | [TK]D-Fender: How do I kill them? I completely stopped asterisk too, and they're still there. |
| 18:44.16 | [TK]D-Fender | pigpen2: C -> A <- B |
| 18:44.33 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:44.51 | antimoof | are any of y'all familiar enough with the guts of asterisk to enlighten a noob (such as myself) why a kernel module is required for realtime timing, rather than just using (say) setitimer() or the like? |
| 18:44.54 | [TK]D-Fender | pigpen2: : C & B do all your real work split in half |
| 18:45.01 | [TK]D-Fender | pigpen2: I suspect it MIGHT actually be more efficient.... |
| 18:45.14 | pigpen2 | So create 2 meetme's, pulling them into another meetme. |
| 18:45.34 | pigpen2 | sorry if I am being dense....caffiene is still taking effect. |
| 18:45.38 | [TK]D-Fender | pigpen2: but I'm sure you'd settle for functional ;) |
| 18:45.54 | pigpen2 | Well, I hope the server has enough proc. |
| 18:45.57 | pigpen2 | :) |
| 18:46.02 | antimoof | or perhaps an easier question: what are the bottlenecks (CPU, I/O, whatever) in asterisk for high call volumes? |
| 18:46.39 | pigpen2 | [TK]D-Fender, fyi, that was a joke. It is a quad dual core 64bit xeon, 8gb ram, etc,.etc. |
| 18:46.47 | [TK]D-Fender | pigpen2: 1 Page calling 2 local channesl that will each Page their respective halves of the population. |
| 18:47.01 | [TK]D-Fender | pigpen2: using the Meetme ENGINE. |
| 18:47.13 | [TK]D-Fender | pigpen2: So in dialplan, thats all app_page |
| 18:47.14 | pigpen2 | k. |
| 18:47.32 | pigpen2 | I think I got it. |
| 18:47.48 | Yourname` | [TK]D-Fender: It worked!! |
| 18:47.52 | [TK]D-Fender | antimoof: Define YOUR idea of "high call volumes". |
| 18:47.58 | [TK]D-Fender | Yourname`: You're welcome. |
| 18:48.47 | antimoof | well, the most obvious scenario is "suppose I want to have 12 people on a voice conference" - but just knowing where the hits are in general would be good. |
| 18:48.47 | Yourname` | [TK]D-Fender: That's awesome. I got two calls on xlite, one with my cellphone, and one with my landline. Tried calling with the 3rd line, and it said "No line available" on xlite, and on the CLI is said "Busy here".. how can I make it so that line will wait for me to get free? |
| 18:48.58 | sweeper | in make menuconfig, does XXX mean some of the dependencies are missing? |
| 18:49.16 | [TK]D-Fender | Yourname`: that will take all sorts of dialplan login. |
| 18:49.20 | [TK]D-Fender | logic* |
| 18:49.23 | Qwell[] | sweeper: yes, and it tells you which ones at the bottom |
| 18:49.28 | Qwell[] | or, rather, it tells you what it depends on |
| 18:49.30 | [TK]D-Fender | sweeper: Sounds about right |
| 18:49.36 | Yourname` | [TK]D-Fender: A pointer on what I could read to get there, [TK]D-Fender? |
| 18:49.42 | *** join/#asterisk RSAMan (n=a@dsl-242-44-247.telkomadsl.co.za) |
| 18:49.48 | RSAMan | hi guys |
| 18:49.53 | sweeper | Qwell[]: can I tell make where stuff lives? I've got unixodbc installed, but make menuconfig can't see it |
| 18:49.57 | [TK]D-Fender | antimoof: 12 people in a conference? Thats no big deal... |
| 18:50.18 | antimoof | mostly, I'd like to know when I can parallelize resources (having more servers or whatever), and when I'm hit by atomic limitations. |
| 18:50.21 | Qwell[] | configure --with-odbc=/path/to/odbc |
| 18:50.31 | RSAMan | ... setting up a sip server , from a fresh installation of asterisk 1.4.. Will it be enought to create/edit the sip.conf and extensions.conf files only ? |
| 18:50.43 | [TK]D-Fender | Yourname`: look at DIALSTATUS after your dial attempt, do GotoIf's etc, have some sort of timeout, etc. Maybe use queues... depends on how you want to manage it. |
| 18:51.02 | [TK]D-Fender | RSAMan: Thats most of it, yes. |
| 18:51.20 | kiscokid | RSAMan: yes unless you have Zap cards to configure |
| 18:51.25 | [TK]D-Fender | RSAMan: Assuming you don't want voicemail, etc./ |
| 18:51.32 | Yourname` | [TK]D-Fender: Gotcha. Thanks! Another question, where does it say about what you mentioned to me about using the same extension... so I can read up more on it? |
| 18:51.47 | RSAMan | [TK]D-Fender: kk cause i did the make samples thing by mistake , so was not sure what i needed |
| 18:52.06 | RSAMan | just want a sip server with a few users and a basic dialplan |
| 18:52.14 | [TK]D-Fender | Yourname`: there is no specific reference. jsut think of it as all roads leading to Rome, and your system sending calls in circles because of it. |
| 18:52.45 | [TK]D-Fender | RSAMan: If you're starting from scratch thats a good idea, except rename those 2 files and start your from scratch. |
| 18:52.49 | sweeper | Qwell[]: does E mean exists, and M mean missing, or what? |
| 18:53.04 | sweeper | mm, n/m |
| 18:53.07 | [TK]D-Fender | sweeper: So E&M = MIA? :) |
| 18:53.24 | Katty | Ooo, redbull :> |
| 18:53.34 | sweeper | well, apparently menuconfig won't tell me WHICH dep is missing... |
| 18:54.01 | RSAMan | [TK]D-Fender : starting from scratch means its a good idea to make samples ? |
| 18:54.06 | RSAMan | sorry i am slow |
| 18:54.16 | ManxPower | E&M = Ear and Mouth (yes, that is what it really means) |
| 18:54.16 | datachomper | try -j3 |
| 18:55.04 | [TK]D-Fender | RSAMan: Good idea for all of the OTHER files. |
| 18:55.23 | [TK]D-Fender | ManxPower: that was HUMOUR, not a question ;) |
| 18:55.25 | kiscokid | RSA: Yes making the samples is a good idea but starting the sip and extensions.conf from scratch is much less confusing |
| 18:56.03 | RSAMan | kk |
| 18:56.10 | RSAMan | just want to make 3 users |
| 18:56.13 | RSAMan | in sip.conf |
| 18:56.24 | RSAMan | then give them all extensions |
| 18:56.29 | RSAMan | in extensions.conf |
| 18:56.36 | RSAMan | like 4 lines |
| 18:56.42 | sweeper | Qwell[]: what is configure looking for when it looks for unixodbc? the .so files, or some exec somewher? |
| 18:56.42 | RSAMan | 6 |
| 18:56.53 | RSAMan | so all i need to do is edit those files |
| 18:56.55 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 18:57.16 | kiscokid | RSA: yes and then reload |
| 18:57.28 | RSAMan | kiscokid: got it ... thanks |
| 18:57.52 | RSAMan | off to bed |
| 18:57.56 | RSAMan | thanks for the help guys |
| 18:58.06 | kiscokid | np |
| 18:58.06 | RSAMan | its sleep time here in south africa |
| 18:58.33 | Yourname` | [TK]D-Fender: Thanks! |
| 19:01.39 | *** join/#asterisk centrex (i=centrex@nat/digium/x-a6e33b8013c8f4d3) |
| 19:02.40 | generalhan | aww ... asterisk starting from that script was a fluke ... i cant make it happen again, even though zaptel is loaded before its run. and when i run 'service asterisk restart' it FAILED to shutdown and then starting asterisk was OK ? |
| 19:03.02 | generalhan | i cant even stop it ! lol it just keeps failing |
| 19:03.36 | Strom_M | kill safe_asterisk |
| 19:03.40 | [TK]D-Fender | generalhan: nukes saf_asterisk first, then all *'s spawned by it |
| 19:03.44 | centrex | Use a -9 with that if you have to. |
| 19:03.45 | generalhan | i did |
| 19:03.51 | centrex | generalhan, kill -9 it |
| 19:04.02 | generalhan | ps shows NO asterisk running |
| 19:05.03 | [TK]D-Fender | generalhan: then jsut start it after restarting zaptel. |
| 19:05.29 | generalhan | [TK]D-Fender: i just did that ... and no dice. it ONLY starts the safe_asterisk. |
| 19:05.39 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:05.48 | [TK]D-Fender | generalhan: stop using the script and test it by hand |
| 19:05.50 | generalhan | so i manually do asterisk -cvvvvvg and now im getting errors :( ? |
| 19:06.10 | generalhan | loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 |
| 19:06.39 | generalhan | this is fun ! ok lets see what i can try next ! |
| 19:07.41 | [TK]D-Fender | generalhan: modprobe your devices, ztcfg -vvvv |
| 19:07.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:08.31 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/642118 |
| 19:08.47 | generalhan | looks good to me ! |
| 19:09.08 | [TK]D-Fender | generalhan: pastebin your configs |
| 19:09.10 | sweeper | gar missing libtool |
| 19:09.10 | generalhan | and that is after i rmmod'd each of those modules |
| 19:10.16 | centrex | generalhan, have you recently upgraded versions of asterisk? |
| 19:10.36 | generalhan | [TK]D-Fender http://generalhan.pastebin.ca/642121 |
| 19:10.55 | generalhan | centrex: kinda ... this is a new version for me ... but its on a new server so not really an upgrade |
| 19:11.19 | [TK]D-Fender | generalhan: ok, looks kosher... |
| 19:11.22 | centrex | generalhan, k. Sometimes there can be problems with the old modules still lingering around. |
| 19:11.27 | [TK]D-Fender | generalhan: And when you try starting it now? |
| 19:11.31 | *** join/#asterisk ServerCrash (n=ServerCr@unaffiliated/servercrash) |
| 19:11.33 | generalhan | umm lets see |
| 19:11.41 | [TK]D-Fender | generalhan: by hand of course |
| 19:12.07 | ServerCrash | how to use asterisk for mass automatic notification of customers, for instance: emergency notifications, informing on existing liabilities, advertising-informing events, etc. |
| 19:12.26 | Mercestes | ServerCrash: Do you have a server already prepared? |
| 19:13.06 | [TK]D-Fender | ServerCrash: Define "mass" in there. A number please, jsut for reference... |
| 19:13.24 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 19:13.26 | generalhan | [TK]D-Fender: by hand it seems to work now ! |
| 19:13.34 | ServerCrash | Mercestes, I am using one for my office, and currently planning to set a second box for automatic notification |
| 19:13.39 | [TK]D-Fender | generalhan: stop it, and restart the service |
| 19:13.42 | Mercestes | ServerCrash, First....you get one of those high grade servers capable of handling all those automated call-outs and "play recordings" you want to do. make sure it has a *switchign* power supply capable of handling 220... |
| 19:14.16 | Mercestes | ServerCrash, Go ahead and set it to 220 and plug it in. Make sure you are well grounded....and then pee-in it to punish yourself for considering using asterisk to make automated calls with recordings. >.< |
| 19:14.21 | sweeper | Ok, so configure DETECTS both libtool and ODBC, but menuconfig still lists them as having requirements not met :/ |
| 19:14.42 | generalhan | [TK]D-Fender: and that works too ???? OMG what is happening here ... you know what ... im not running the ztcfg -vvvvv in my modprobe initscript ... maybe if i put that in there it will work again |
| 19:14.52 | generalhan | i just wonder why it worked the first time and then not again |
| 19:15.12 | ServerCrash | [TK]D-Fender, I looking for calling upto 10 calls simualaneously per min. IS that too much ? what kind of hardware would i need ? |
| 19:15.25 | Mercestes | ServerCrash, that's not all that massive at all. |
| 19:15.43 | Mercestes | T1card and some fancy scripting would do the trick. |
| 19:15.49 | generalhan | [TK]D-Fender: so do you think that could really make a difference ? |
| 19:15.50 | Mercestes | after you pee in the high voltage server first, of course. |
| 19:15.55 | [TK]D-Fender | ServerCrash: You'd just need a system to queue up the calls and prevent overrun. |
| 19:16.09 | [TK]D-Fender | generalhan: Yes. |
| 19:16.42 | ServerCrash | What does 220 server means ? |
| 19:16.51 | generalhan | crap ... wonder why it worked that first time ? ok im going to drop that line in there and reboot again |
| 19:17.23 | ServerCrash | I have system with core 2 duo 1.86 Ghz 1033 FSB 2GB DDR2 RAM |
| 19:17.58 | ServerCrash | And two phone lines |
| 19:18.26 | [TK]D-Fender | ServerCrash: How do you expect to do 10 calls SIMULTANEOUSLY using 2 phone lines? |
| 19:19.46 | Strom_M | [TK]D-Fender: with the POWER of LARD! |
| 19:20.02 | ServerCrash | Well this is the current setup that i have, i need to understand what hardware and other resources would be required to achive the given task. And is Asterisk the best solution for this or there are other good options |
| 19:20.33 | [TK]D-Fender | ServerCrash: * is completely capable of delivering messages like that. |
| 19:20.53 | [TK]D-Fender | ServerCrash: You should first begin development o fthe call-out distriubuter. |
| 19:21.15 | [TK]D-Fender | ServerCrash: AFTER of course downloading * and installing it and reading THE BOOK |
| 19:21.16 | [TK]D-Fender | ~book |
| 19:21.17 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:21.27 | [TK]D-Fender | ServerCrash: And them looking up a lot of stuff on the WIKI. |
| 19:21.30 | [TK]D-Fender | ~wikis |
| 19:21.31 | jbot | wikis is, like, http://www.voip-info.org |
| 19:22.13 | ServerCrash | "call-out distributer" ?? |
| 19:22.25 | ServerCrash | ~call-out distributer |
| 19:22.58 | Mercestes | nubcake |
| 19:23.05 | Mercestes | ~providers |
| 19:23.11 | Mercestes | ... |
| 19:23.24 | *** join/#asterisk mtaht4 (n=m@wireless-19-108.media.mit.edu) |
| 19:23.41 | Mercestes | ~provider |
| 19:23.43 | Mercestes | ~ITSP |
| 19:23.44 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 19:23.48 | Mercestes | ah, there it is. |
| 19:23.51 | Mercestes | sorta. |
| 19:24.05 | sweeper | checking sql.h usability... no <-- what do I pass to configure to remedy this? |
| 19:24.27 | [TK]D-Fender | ServerCrash: YOU are going to have to code up an engine to generate the calls. |
| 19:24.27 | ServerCrash | [TK]D-Fender, any other hardware required? |
| 19:24.28 | [TK]D-Fender | ServerCrash: Depends on how you intend on connecting to the PSTN |
| 19:25.49 | ServerCrash | [TK]D-Fender, nothing planned as such, what would you suggest |
| 19:26.31 | [TK]D-Fender | ServerCrash: I suggest you go read the book and come up with a plan. |
| 19:26.47 | Mercestes | or hire a consultant |
| 19:27.02 | sweeper | yes :) |
| 19:27.52 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 19:27.56 | ServerCrash | [TK]D-Fender, we have already implemented a local *, currently using linksys pap2 along with asterisk |
| 19:28.23 | sweeper | does asterisk require some kind of odbc-devel library? configure seems to be looking for sql.h, and that file doesn't exist on my system |
| 19:28.54 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 19:31.26 | ServerCrash | has anyone used Tribox ? |
| 19:31.36 | ServerCrash | tribox/ Trixbox |
| 19:31.47 | Katty | nope. |
| 19:31.50 | Katty | :P |
| 19:31.51 | sweeper | ServerCrash: maybe, but they wouldn't admit to it |
| 19:31.55 | Katty | Mercestes: herro! |
| 19:32.26 | kiscokid | I tried it once but I didn't understand what was going on so I started over with the Asterisk sources |
| 19:32.36 | ServerCrash | sweeper, why, trixbox is based on * i guess |
| 19:33.07 | ServerCrash | sweeper, AsteriskNow Vs Trixbox ? which is easy ? |
| 19:33.09 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:33.11 | sweeper | ServerCrash: yes, but it's also a gigantic piece of crap |
| 19:33.20 | Katty | ServerCrash: really, asterisk is easy. |
| 19:33.28 | Katty | ServerCrash: if you want SUPER easy, look into Fonality. |
| 19:33.37 | Katty | ServerCrash: they're pricey... but it's practically setup for you |
| 19:33.41 | lineman60 | whats involed in swaping out phonecards in a pbx, just open the box and swap them or is it more involved then that? |
| 19:33.50 | Qwell[] | lineman60: same card? |
| 19:33.55 | Qwell[] | brand/model |
| 19:34.05 | ServerCrash | KAtty : I am looking for a solution for mass automatic notification of customers, for instance: emergency notifications, informing on existing liabilities, advertising-informing events, etc. |
| 19:34.05 | lineman60 | 2 of the 3 are |
| 19:34.14 | ServerCrash | Katty, what would you suggest |
| 19:34.14 | Qwell[] | you'll have to reconfigure the 3rd |
| 19:34.22 | Qwell[] | the other two should "Just Work" |
| 19:34.51 | Katty | ServerCrash: email. |
| 19:34.57 | [TK]D-Fender | ServerCrash: your setup has NO use for a GUI and the mere fact you your near obvious reliance on one does not bode well.... |
| 19:34.59 | Katty | ServerCrash: or an auto answer thing |
| 19:35.02 | Katty | ServerCrash: maybe (= |
| 19:35.04 | Katty | ServerCrash: lol. |
| 19:35.23 | Katty | advertising-- |
| 19:35.40 | Katty | misssssster fender. |
| 19:35.42 | ServerCrash | Katty, I am looking for voice based solution here, where by i can record them and then send mass |
| 19:36.04 | Katty | ServerCrash: so execute a system() command |
| 19:36.15 | Katty | ServerCrash: and dump the audio file onto an email |
| 19:36.17 | Katty | ServerCrash: and off it goes. |
| 19:37.04 | Katty | OR |
| 19:37.06 | ServerCrash | I am trying to go via airmail and you are sending me via postmail. |
| 19:37.13 | Katty | you can record the file, and then do an duplicate call at the sam etime |
| 19:37.20 | Katty | to whatever SIP / IAX extensions you have |
| 19:37.59 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 19:38.05 | Katty | irule: :> |
| 19:38.16 | irule | hi there Katty |
| 19:38.19 | Katty | herro. |
| 19:38.26 | irule | how can I play with agi-test.agi? |
| 19:38.38 | Katty | i hear it responds well to rattles. |
| 19:38.52 | Katty | you could also try a rubber ducky from the toddler section. |
| 19:39.18 | datachomper | I would buy it some big wheels. I always wanted some as a kid and never got it :O( |
| 19:39.25 | *** part/#asterisk kiscokid (n=sysadmin@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:39.26 | Katty | :< |
| 19:39.29 | Katty | poor datachomper |
| 19:40.10 | irule | thanks katty! |
| 19:40.15 | Katty | my pleasure, deary |
| 19:40.24 | irule | grin |
| 19:41.50 | *** join/#asterisk mog (i=mog@nat/digium/x-f7d1adc924d2e426) |
| 19:41.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:42.44 | Hmmhesays | heh |
| 19:43.05 | mocker | ServerCrash: You should check out nerdvittles.. |
| 19:43.17 | mocker | ServerCrash: They did an automated broadcast type thing that may meet your needs. |
| 19:44.13 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 19:44.48 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 19:44.56 | [TK]D-Fender | *boing* |
| 19:44.58 | irule | I like the bluetooth proximation one |
| 19:45.12 | irule | [TK]D-Fender hi :D |
| 19:45.18 | Aces1Up | anyone here have much familiarity with app_cepstral or Swift which one is better for using with cepstral? or is there a better way altogether to get cepstral to work with asterisk? |
| 19:46.02 | mocker | [TK]D-Fender: I get the feeling that ODBC support for Asterisk is a total hack add-on in 1.2.x ;) |
| 19:46.19 | [TK]D-Fender | mocker: never touched it personally... |
| 19:46.29 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 19:47.00 | [TK]D-Fender | mocker: http://www.sofaswitch.org/d/ |
| 19:47.32 | waverly360 | Research has already told me that it can't be done, but I figured I'd ask..is it possible to assign a sip or iax trunk to a trunk group like it can be done in zapata.conf? |
| 19:47.52 | [TK]D-Fender | waverly360: No |
| 19:48.31 | [TK]D-Fender | waverly360: Besides dialing may be non-homogeneous and completely non-vialble.. |
| 19:49.02 | waverly360 | [TK]D-Fender: non-homogeneous and completely non-vialble? |
| 19:49.02 | mocker | [TK]D-Fender: awesome. |
| 19:49.24 | [TK]D-Fender | waverly360: uload chan_echo.so |
| 19:49.52 | Mercestes | Katty, herro |
| 19:50.12 | centrex | Aces1Up, I was under the impression that swift was the actual software, and that app_cestral was the dialplan application to call it, but that might be wrong. |
| 19:50.18 | Trevor_b | anyone here dealing with the teliax fallout? |
| 19:50.27 | mocker | What happened to teliax?? |
| 19:50.32 | Mercestes | teliax fallout? |
| 19:50.35 | Mercestes | what teliax fallout? |
| 19:50.53 | mocker | I had an account w/ them for awhile. |
| 19:51.01 | Trevor_b | currently apparently all codecs are failed except ulaw (or so their tech tells me) as certain servers are down, and all outbound calls seem to drop to busy or congested. |
| 19:51.17 | Mercestes | really? |
| 19:51.18 | Trevor_b | i couldnt get ulaw to work either, so just set them low on the outbound route.... |
| 19:51.21 | Mercestes | I should send them my resume. |
| 19:51.21 | explidous | anybody knows where i can get a copy of dundi-intro.pdf |
| 19:51.37 | Trevor_b | explidous: Hmm, i htink i have it, not sure where i got it |
| 19:51.55 | Trevor_b | oops, sorry that is the DUNDI with trixbox. |
| 19:52.08 | explidous | Its most likely not there anymore... the website that had it does not anymore |
| 19:52.23 | Mercestes | I think jbot should auto-kick/ban everytime someone says the "T" word |
| 19:52.49 | mocker | explidous: Sorry, don't have it. |
| 19:53.00 | Trevor_b | Yeah i like teliax, for ITSP they are fairly good. Never really found another one i liked the uptime&support. Although my main other experience is AT&T callvantage, broadvoice, and vonage. |
| 19:53.15 | irule | I need another hug Katty please :D |
| 19:53.53 | explidous | thanks for looking everyone |
| 19:54.20 | Mercestes | I liked Teliax too. |
| 19:54.52 | irule | thanks Mercestes! good to read you! :) |
| 19:54.58 | Mercestes | :) |
| 19:55.14 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:55.14 | Mercestes | good to read you too |
| 19:56.27 | explidous | web.archive.org has it.... thanks! |
| 19:56.36 | Katty | oh |
| 19:56.40 | Katty | i want in on these huggable things. |
| 19:57.20 | Katty | :> |
| 19:57.22 | Katty | yay! |
| 19:58.00 | irule | my doctor told me to make all hugs last 20 seconds each O8) |
| 19:59.32 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
| 20:00.08 | Katty | hugs can fix lots of things, including sickness. |
| 20:01.08 | Katty | wohoo! |
| 20:01.18 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 20:01.25 | neuwald | hi folks |
| 20:01.27 | neuwald | on this: |
| 20:01.28 | neuwald | exten => _1.,2,Set(passkey=${DB(auth_user/{$CALLERIDNUM})}) |
| 20:01.28 | neuwald | exten => _1.,3,NoOp(${passkey}) |
| 20:01.29 | irule | hi |
| 20:01.45 | neuwald | <PROTECTED> |
| 20:02.05 | neuwald | i got this. But I only filter and get only 99, the user extension |
| 20:02.18 | [TK]D-Fender | neuwald: Nice know you paid attention when I told you that CALLERID is toast as well.. |
| 20:02.30 | neuwald | oops :-( |
| 20:02.31 | [TK]D-Fender | neuwald: And you should ahve noticed that on the functions page.. |
| 20:02.37 | Mercestes | yay, threesome with file |
| 20:02.46 | [TK]D-Fender | neuwald: "show function CALLERID" |
| 20:02.49 | irule | neuwald CALLERID(num) |
| 20:02.58 | datachomper | What's with all the hugs on IRC? Whatever happened to slapping people around with trouts. |
| 20:03.03 | [TK]D-Fender | neuwald: ${CALLERID(num)} |
| 20:03.24 | [TK]D-Fender | datachomper: mIrc.... that for Winblows ;) |
| 20:03.43 | Trevor_b | datachomper: its all touchy feely now, feel like im in AOL rooms sometimes ;) |
| 20:03.49 | irule | datachomper everyone gets better answers to their asterisk questions because we are so happy hugging :D |
| 20:03.53 | cpm | it never was a trout |
| 20:04.01 | cpm | I think it was a tuna |
| 20:04.42 | Katty | tuna? *perk* |
| 20:04.51 | centrex | lawl |
| 20:06.26 | Katty | i like tuna :< |
| 20:07.11 | irule | now that we are at it, it is time to eat where I am from, and that you mentioned tuna, here is a great simple dish I like http://www.recetas.com/receta-de-Tostadas-de-atun-2406.html |
| 20:07.40 | Aces1Up | is there a way to use multiple voices from capestral and asterisk? |
| 20:07.50 | irule | jsut a toasted tortilla with beans and cream at the bottom, plus tuna and veggies on top! |
| 20:08.33 | datachomper | I must admit, that sounds terrible. |
| 20:09.01 | Katty | sounds kinda good to me. |
| 20:09.11 | *** join/#asterisk saftsack (n=saftsack@pD9E07A84.dip.t-dialin.net) |
| 20:09.14 | Strom_M | poutine |
| 20:09.41 | Katty | saftsack: (= |
| 20:10.23 | ManxPower | datachomper: I was AFK, that's why. |
| 20:12.23 | kempist | lol, irc huging? gladly the only other channel im in has "It's not our fault you suck" as part of its topic, which pretty much says the style of the channel |
| 20:12.51 | Katty | oh boy, they want me to go to a conference in Seatle. |
| 20:12.57 | Katty | screw those hippies. |
| 20:13.34 | datachomper | Somebody change the topic to "Free hugging pbx hippies" |
| 20:14.32 | explidous | Katty, they have been offering us help to compete with Linux.... my approach: a working OS would be fine... |
| 20:15.12 | *** join/#asterisk guillote_GNU (n=guillote@host212.201-252-198.telecom.net.ar) |
| 20:15.24 | Katty | explidous: may the schwartz be with you (= |
| 20:15.55 | sweeper | ~_~ |
| 20:15.55 | jbot | :/ |
| 20:16.06 | datachomper | Lol, i work with a "Schwartz" and he says that all the time. |
| 20:16.19 | Katty | jbot: :> |
| 20:17.03 | [TK]D-Fender | Katty: "Barf?! Not in here, this is a Mercedes!" |
| 20:17.38 | Strom_M | and now...the short short version! |
| 20:18.26 | Katty | [TK]D-Fender: *hee* |
| 20:19.35 | datachomper | When does then become now? ... soon |
| 20:21.02 | *** join/#asterisk joaop (n=taken@189-11-162-115.bnut3700.dsl.brasiltelecom.net.br) |
| 20:22.48 | Aces1Up | will any module in the usr/lib/asterisk/modules be loaded on asterisk restart? |
| 20:23.35 | Aces1Up | or do i have to specify them somewhere? i am trying to load a new moduel app_swift.so when i do a load command it says it cannot find it but it is where it is supposeed to be in /usr/lib/asterisk/modules. |
| 20:24.43 | ManxPower | Aces1Up: The module should be installed when you do a "make install" in Asterisk |
| 20:25.14 | *** join/#asterisk sakic (n=sakic@adsl-146-134-195.clt.bellsouth.net) |
| 20:25.46 | Aces1Up | hrmm ok i did not do a rebuild for this module, so when i do a make install the load command will be able to find the module? |
| 20:27.03 | ManxPower | Aces1Up: the module must be built for that exact Asterisk |
| 20:27.11 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 20:27.36 | ManxPower | Did the module come with instructions? |
| 20:27.42 | Aces1Up | yes |
| 20:28.24 | ManxPower | And what did those and is the module compatible with the version of Asterisk you are using? |
| 20:29.05 | Aces1Up | it just said to do a make install on the package and then load the module using "load app_swift.so" |
| 20:29.29 | *** join/#asterisk innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 20:29.48 | innatech | FYI, for today, teliax is dead to anything but ulaw--in case anyone's having problems. |
| 20:30.32 | Aces1Up | manx so i need to do a asterisk make install? to have integrated into asterisk? |
| 20:30.43 | ManxPower | I would try copying the app_swift.c into /path/to/src/asterisk/apps and then edit /path/to/src/asterisk/apps/Makefile to add app_swift.so to the list near the top of the file, cd to /path/to/src/asterisk and make clean && make install |
| 20:31.04 | Aces1Up | cool thanks will do. |
| 20:32.11 | Aces1Up | manx what is the normal path to src? |
| 20:32.13 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 20:32.29 | ManxPower | Aces1Up: where ever you put the Asterisk source code. |
| 20:33.44 | sakic | any pro here want to make some $$ setting up my system |
| 20:34.23 | ManxPower | Not me unless it can turn into a long term contract and only then if you have decent hardware and lines. |
| 20:34.30 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 20:34.31 | sakic | lol |
| 20:35.21 | sakic | ok well there is always the indians on elance :P |
| 20:35.30 | ManxPower | i.e. T-1 card, PRI, etc |
| 20:35.33 | ManxPower | Polycoms |
| 20:35.54 | sakic | I have polycoms, I have digium card, and pots |
| 20:36.03 | ManxPower | You poor sod. |
| 20:36.50 | explidous | sakic we have experience with cepstral, what do you need? |
| 20:37.12 | sakic | cepstral? |
| 20:37.21 | ManxPower | I no longer to any consulting for system with analog cards. |
| 20:37.28 | datachomper | http://www.google.com.mx <- How does that work? Is there a "www.com.mx" TLD that forwards subdomains in mexico? or is there some sort of special DNS routing for countries other than America? |
| 20:37.32 | centrex | ManxPower, out of curiosity, why is that? |
| 20:38.05 | ManxPower | centrex: in the situations my clients had them installed in, the modules tended to lock up under even medium call load. |
| 20:38.11 | Aces1Up | manx what is the correct way of adding it to the makefile? i don't see any other includes or anything for any other .so files in the makefile. |
| 20:38.34 | Aces1Up | "include app_swift.so" ??? |
| 20:38.40 | centrex | ManxPower, weird, any specific card that did that? |
| 20:38.45 | innatech | Manx: so you work w/media gateways, pure IP, or what? |
| 20:39.17 | ManxPower | Aces1Up: line 32 at the end of the line ending in app_stack.so |
| 20:39.23 | ManxPower | innatech: PRI |
| 20:39.26 | innatech | ah. |
| 20:39.43 | ManxPower | I also don't route calls over the internet unless all the PRIs fail or are full. |
| 20:39.55 | ManxPower | bbiaw |
| 20:42.01 | Aces1Up | manx I don't have that line in my /src/apps/Makefile |
| 20:43.18 | sakic | what is the default password for a polycom? |
| 20:43.38 | sakic | they don't provide documentation in the box... kinda odd to me |
| 20:43.42 | Strom_M | 456 |
| 20:44.30 | datachomper | It's in the polycom admin guide |
| 20:45.18 | sakic | thanks |
| 20:45.21 | sakic | I think I am in over my head :) |
| 20:45.33 | datachomper | Na, what do you got. Poly 501's ? |
| 20:45.45 | sakic | 120s |
| 20:45.52 | generalhan | ok ... this is getting slightly aggrivating -- i finally got the asterisk config file to do its thing on boot, it looks like its running but its not ??? |
| 20:45.52 | generalhan | Can some one take a look and tell me if they see the same thing that i do: http://generalhan.pastebin.ca/642225 |
| 20:46.03 | sakic | 320 |
| 20:46.33 | datachomper | sakic: A good reference http://www.voip-info.org/wiki-Polycom+Phones |
| 20:47.30 | Aces1Up | if a have a .so module app_swift.so and i want to install it into asterisk 1.4 what is the correct procedure, In the makefiles i do not see anywhere where i specifiy new modules to be loaded on the next compile. |
| 20:48.58 | sakic | screw it, I am outsourcing |
| 20:49.10 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 20:49.18 | Voicemeup | : anyone have DAL (direct acess lines) experience ? i was told you could hook that up to a sangoma liek a T' (for inbound toll frees ) |
| 20:49.32 | datachomper | rofl sakic |
| 20:51.37 | *** join/#asterisk red9012 (n=marc3234@206-248-135-223.dsl.teksavvy.com) |
| 20:51.38 | sakic | I can install a security system, or a card access system |
| 20:52.48 | centrex | generalhan, is /var/run/asterisk.ctl there? |
| 20:52.53 | red9012 | is there a *reliable* method for handling faxing with asterisk? |
| 20:53.01 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 20:53.34 | generalhan | centrex: sorry i am rebooting now ... i will check assoon as it comes back up .... i tried moving my ztcfg -vvvvvv initscript farther down the boot order line |
| 20:53.55 | centrex | generalhan, That shouldn't be related to this error, imo. |
| 20:54.30 | explidous | Aces1Up, is your module 1.4 type or 1.2 type... the 1.2 modules have to be changed to work with 1.4 |
| 20:54.30 | generalhan | centrex: well after what you just saw in that post, i ran ztcfg -vvvvvv; asterisk -cvvvvg; again and it loaded right up |
| 20:54.33 | jcaceres | hello i have a trouble trying to make work two cards one a te120 and a tdm400 |
| 20:54.47 | jcaceres | <PROTECTED> |
| 20:54.49 | centrex | generalhan, that's pretty weird. |
| 20:54.53 | jcaceres | i oly get this |
| 20:55.00 | jcaceres | pci:0000:05:05.0 e159:0001 [wctdm] |
| 20:55.02 | generalhan | centrex: agreed ... everything has been SOO weird on this install |
| 20:55.24 | tzafrir_home | generalhan, just out of curiosity, why put ztcfg -vvvvvvvvv in your init script? put 'ztcfg' . If there are no errors, you have no noise in the logs and all's well |
| 20:55.33 | jcaceres | why is that i am not getting information about the te120 |
| 20:55.37 | explidous | centrex, generalhan : looked at it as well... it is REALLY WEIRD |
| 20:56.23 | jcaceres | Strom_M, are you there? |
| 20:56.43 | Strom_M | yes |
| 20:57.24 | centrex | generalhan, I don't know if this will help, but I would turn on all logging in logger.conf, reboot the system, and watch it and see what's going on. |
| 20:57.40 | generalhan | centrex: where does that output to ? |
| 20:58.43 | centrex | generalhan, wherever /etc/asterisk/asterisk.conf is set to log to. |
| 20:59.23 | generalhan | centrex: so uncomment the "full=>" line to get full logging then check the file that is setup in asterisk.conf after a reboot |
| 20:59.40 | explidous | generalhan, why dont you stop asterisk and start it from the commandline with -cvvvvvvvvdddddddg |
| 21:00.14 | centrex | generalhan, I would, just to see. |
| 21:00.21 | centrex | Might turn up something useful. |
| 21:00.26 | explidous | generalhan, that way you see right away if it loads correctly and see if you can connect to it then... |
| 21:00.41 | generalhan | centrex: i cant stop the service LOL ... i havent been able to since i turned on that config file ! |
| 21:00.42 | jcaceres | Strom_M, i dont know if you remember that i had that a trouble loding those cards some time ago |
| 21:01.03 | generalhan | service asterisk stop [FAILED] and that is before and/or after i kill all asterisk processes |
| 21:01.13 | centrex | wow |
| 21:01.38 | jcaceres | i we found that the order in loading the modules was afecting the cardas behaveior |
| 21:01.47 | jcaceres | and we found that the order in loading the modules was afecting the cardas behaveior |
| 21:02.21 | generalhan | centrex: and here are the errors logged already from the last boot (as to why asterisk wasnt really running): http://generalhan.pastebin.ca/642263 |
| 21:02.25 | jcaceres | well, now i am having the same troubles, but just for loading zap in asterisk |
| 21:02.38 | Aces1Up | i posted my /asterisk/src/apps/Makefile at http://pastebin.ca/642262 can someone help me on where i add app_swift.so to load a module correctly? |
| 21:02.45 | generalhan | and then just by running ztcfg -vvvvvvvv and retrying it fixed that issue and actually started up |
| 21:02.48 | mocker | Weeeird. Using ODBC storage if I call from a phone that's pulling to the database remotely I get huge pauses. If I call from a phone that logs directly into the server I get no pauses. Both phones are right next to each other so really they are both pulling remotely. |
| 21:03.04 | jcaceres | <PROTECTED> |
| 21:03.32 | jcaceres | any idea of the reason for that? |
| 21:04.23 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 21:04.56 | variable_office | i am unable to change the voicemail password from comedian mail; i am running asterisk realtime in odbc/pgsql; is realtime not capable of this? |
| 21:05.01 | centrex | generalhan, oh, okay, looks like the safe asterisk script keeps trying to restart asterisk, but asterisk fails due to the ztcfg. Safe asterisk keeps restarting it so it looks like it's running, but never gets to the point where you can connect. |
| 21:05.14 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 21:06.06 | generalhan | centrex: but it doesnt make any sense cause i have another config file that runs at 1 boot order before the asterisk one that modprobes zaptel and wct4xxp and runs ztcfg -vvvvv , and the modules are loaded at boot so i just assume that its doing the ztcfg also |
| 21:06.10 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 21:06.58 | generalhan | centrex: one sec and i will pastebin something else ... im doing a reboot right now and i will paste what goes on right after ! |
| 21:07.01 | centrex | generalhan, not sure. Do you have to reload the zaptel modules for it to work? If so, it might be that kudzu is autodetecting them as something else (it always does). |
| 21:07.36 | generalhan | centrex: nah ... i dont have to reload the modules ... i ONLY have to do the ztcfg -vvvv again to get asterisk to start |
| 21:08.07 | centrex | generalhan, maybe your init scripts are possible trying to run ztcfg before the modules are loaded? Where is your ztcfg script located at? |
| 21:09.03 | pigpen | anyway to have multiple channels in a call file? |
| 21:09.10 | generalhan | centrex: /etc/init.d/ along with the other asterisk one ... but the asterisk config does chkconfig boot number 40 and my modprobe script is chkconfig 39 |
| 21:09.21 | generalhan | and the ztcfg line is after the 2 modprobe lines |
| 21:10.13 | centrex | generalhan, no idea then. Depending on your distro make config in zaptel might work for you. (should install a zaptel init.d). |
| 21:12.23 | centrex | generalhan, and I would also disable kudzu just in case. |
| 21:14.27 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 21:14.34 | generalhan | centrex: ok i have a killer pastebin for you ! lol: http://generalhan.pastebin.ca/642281 |
| 21:17.18 | *** join/#asterisk Luria (n=abulafia@cpe-68-175-110-250.nyc.res.rr.com) |
| 21:17.18 | Mercestes | generalhan, that looks realtively normal. |
| 21:17.30 | Mercestes | s/realtively/relatively/ |
| 21:17.39 | generalhan | Mercestes: then why is it not starting asterisk at boot ? |
| 21:18.03 | Mercestes | Did you do your modules.auto-load correctly? |
| 21:18.05 | generalhan | and why when i have a script that is running the modprobe and ztcfg, is it waiting for that to actually allow asterisk to start |
| 21:18.21 | generalhan | Mercestes: dunno ... thats a good question |
| 21:18.24 | centrex | Mercestes, he's using an init script to load them instead of the modules configuration files. |
| 21:18.28 | generalhan | though lsmod shows them loaded at boot |
| 21:18.33 | Mercestes | Okies. |
| 21:18.40 | Mercestes | is the init script in rc.conf? |
| 21:19.13 | generalhan | Mercestes: /etc/init.d/modprobe ... and chkconfig on |
| 21:19.33 | Luria | hey a bit of a newbish question... can anyone recommend a card for asterisk on openbsd? just one pots line involved (for home use) |
| 21:19.51 | Mercestes | well, you have a broken safe_asterisk process running in the beginning. |
| 21:20.02 | *** join/#asterisk icel (n=icel@63.78.162.54) |
| 21:20.22 | Mercestes | I think you should comment out the safe_asterisk line, reboot, and the *very* first thing you should do after you log in is run asterisk -cvvvvvvvvvv and see what's broken. |
| 21:20.26 | generalhan | right ... and i cant figure out why ... everything has been so very very strange with this install ... i have NEVER run into issues like this before |
| 21:20.46 | icel | hey, is asterisk supposed to use 100% of your cpu? |
| 21:20.48 | generalhan | Mercestes: i already know what that will do ... ive already done it |
| 21:20.53 | Mercestes | Luria, TE210P |
| 21:21.00 | generalhan | it was in one of my other pastebins from just a bit ago ... lemme find it |
| 21:21.05 | Luria | Mercestes, thanks |
| 21:21.09 | Mercestes | Luria, or something like that.....they (digium) of course, could not come up with sensible naming conventions |
| 21:21.17 | generalhan | Mercestes: http://generalhan.pastebin.ca/642263 |
| 21:21.38 | generalhan | thats what happens with out the safe_asterisk and me trying to manually run it after boot |
| 21:21.48 | Luria | i was just worried about driver support |
| 21:21.50 | centrex | Mercestes, The TE210P probably isn't what she's looking for, that's a t1 card =) |
| 21:21.55 | Mercestes | oh |
| 21:22.04 | Mercestes | ..TDM210P? |
| 21:22.06 | centrex | Luria, tdm400p is a four port analog card, smallest card digium has. |
| 21:22.14 | Mercestes | Yea, what Centrex said |
| 21:22.21 | Luria | yeah, thats not what i need :-) |
| 21:22.21 | Mercestes | I thought they had a 2 port version? |
| 21:22.28 | centrex | Mercestes, We dont. |
| 21:22.40 | generalhan | centrex: waht about TDM20B ? |
| 21:22.41 | centrex | I mean you can get only 2 modules on the 40. |
| 21:22.46 | centrex | 400 |
| 21:22.47 | Mercestes | Yea, true. |
| 21:22.59 | Mercestes | generalhan, your not modprobing zaptel. |
| 21:23.15 | generalhan | Mercestes: i AM ! lol ... its listed in lsmod even ! |
| 21:23.19 | Mercestes | generalhan, no, your not. |
| 21:23.26 | Mercestes | generalhan, your trying to run asterisk and *THEN* modprobing zaptel |
| 21:23.32 | Mercestes | so the first time around, it fails...and dies... |
| 21:23.37 | Mercestes | and then it loads up everything you need. |
| 21:23.42 | Mercestes | and then you manually run it....and it works |
| 21:23.42 | generalhan | Mercestes: and i can fix that error by simply typing ztcfg -vvvvv THEN asterisk -cvvvvvg |
| 21:23.50 | pigpen | can anyone think of why when I call a process via the Manager API, after it has successfully completed, I cannot run it again? |
| 21:24.30 | pigpen | even after restarting asterisk |
| 21:24.34 | Aces1Up | is there a new way of loading modules in asterisk 1.4? or can someone point me to a doc on how to add a module? |
| 21:27.37 | icel | does asterisk use 100% of cpu for everyone else or just me? |
| 21:27.47 | pigpen | Just you. |
| 21:27.52 | icel | sweet |
| 21:28.11 | *** part/#asterisk Luria (n=abulafia@cpe-68-175-110-250.nyc.res.rr.com) |
| 21:28.24 | icel | guess a core2 duo isnt good enough or there is some bug |
| 21:28.49 | red9012 | anyone knows if I can receive faxes using asterisk? |
| 21:28.58 | generalhan | icel: unless youre running 100 T-1 interfaces with 5000 users on the same server LOL ... prolly an issue somewhere ! |
| 21:29.27 | pigpen | red9012, Asterisk cannot receive faxes. |
| 21:29.28 | icel | generalhan: i am running 1 T-1 interface (4 port card, only 1 T-1) and a handful of sip users |
| 21:29.56 | pigpen | red9012, however, asterisk can pass it off to a fax handler such as iaxmodem configured for hylafax. |
| 21:30.02 | generalhan | icel: yea something is not quite right ... if i wasnt knee deep in my own issues i would try to help you out |
| 21:30.05 | pigpen | which works great IMHO |
| 21:30.39 | icel | generalhan: no problem |
| 21:32.19 | mocker | 1/window 5 |
| 21:33.02 | Aces1Up | ummm Hey how do i load a dang module lol yeh wow yeh GUUUURRRRLLLLL |
| 21:33.03 | gammah | red9012: I believe the T38 protocol is also used for faxing |
| 21:33.16 | Aces1Up | asterisk making me go crrraaazzzy, |
| 21:33.58 | pigpen | gammah, sure, but asterisk itself does not receive faxes....it only passes T38 to a T38 compatable device. |
| 21:34.19 | red9012 | pigpen -- hylafax does fax -> email? |
| 21:34.20 | pigpen | asterisk, iaxmodem, hylafax is the most stable option at this point. |
| 21:34.22 | gammah | ah, never knew really how it worked :) Just knew there was such a thing |
| 21:34.24 | pigpen | yes. |
| 21:34.31 | pigpen | via tiff or pdf. |
| 21:34.40 | pigpen | and outbound from the pc |
| 21:34.44 | red9012 | is is 100% reliable? |
| 21:35.12 | pigpen | I have been running it now for a customer (and myself) for about 1 year. I have never had to address it. |
| 21:35.22 | pigpen | short of adds/deletes/changes |
| 21:35.53 | pigpen | For example, many places like it, because rather than handing out a DID and a Fax DID, you can run faxdetect on their normal DID. |
| 21:35.59 | brad_mssw | think openpbx/callweaver is better for fax receiving (it's a fork of asterisk), it merged in spandsp's rx/txfax with T.38 termination |
| 21:36.19 | red9012 | pigpen -- faxdetect on existing numbers? |
| 21:36.51 | pigpen | yeah..like, you already have 100 people who have their own DID. Add in faxdetect, and they can now rec faxes on their did. |
| 21:37.07 | pigpen | it only costs you 1 second on the inbound call. |
| 21:37.55 | pigpen | So far I have deployed this for 4 systems. One of which is mine. |
| 21:38.06 | red9012 | pigpen-- how do you connect hylafax to sip? |
| 21:38.23 | pigpen | One setup has 400 DID's, other has 180'ish, other 85, other 55 |
| 21:38.35 | pigpen | hylafax does not connect to sip. |
| 21:38.48 | pigpen | read about iaxmodem. |
| 21:38.57 | pigpen | iaxmodem is a virtual modem for hylafax. |
| 21:39.34 | pigpen | incoming calls are "detected" as a fax call, then sent to the fax,s,1 context. |
| 21:39.57 | pigpen | then the IAX modem answers, allowing hylafax to receive it. |
| 21:40.22 | red9012 | which app does detection of fax? |
| 21:40.43 | pigpen | You can hard code the DID to directly go to the fax context (such as a dedicated fax), or use the zaptel faxdetect feature. |
| 21:43.32 | brettnem | hey, anyone using MacroExclusive? |
| 21:43.50 | brettnem | I'm curious how use of MacroExclusive to keep global stats on a high load server will effect performance |
| 21:44.42 | Corydon76-work | Shouldn't affect it too badly, as long as you keep the duration down |
| 21:44.57 | brettnem | it's just incrementing a global |
| 21:45.29 | brettnem | that should be really fast, right? |
| 21:46.04 | Corydon76-work | Yes |
| 21:46.45 | brettnem | I don't suppose there is a safe way to increment a global without obvious race conditions.. other than macroexclusive, right? |
| 21:47.23 | red9012 | I wonder if its possible to expose the list of extension in one context, to another context? |
| 21:47.31 | Corydon76-work | That's pretty much the only way |
| 21:47.37 | Corydon76-work | Are you just counting calls? |
| 21:48.13 | Corydon76-work | ${CUT(UNIQUEID,.,2)} will get you the number of calls since the server was last restarted |
| 21:48.26 | brettnem | well, counting calls, and the different call results |
| 21:48.50 | brettnem | like busys, etc.. |
| 21:49.00 | Corydon76-work | Ah |
| 21:49.14 | Corydon76-work | Do you care if those stats are lost on a restart? |
| 21:49.17 | icel | Does anyone have an /etc/init.d/asterisk init script for gentoo they would like to share? |
| 21:49.25 | brettnem | nah, I'd almost prefer it |
| 21:49.45 | red9012 | whats callweaver? |
| 21:49.47 | brettnem | which is why I went with the global |
| 21:49.48 | Corydon76-work | Ah, otherwise I'd suggest running a SQL statement, as those are atomic |
| 21:50.16 | brettnem | that seems like a lot of overhead.. I do a lot of SQL calls via CURL to another server to offload that stuff |
| 21:50.19 | Hmmhesays | ok why is my postgres cdr table not getting updated |
| 21:50.34 | Corydon76-work | Heh, why CURL? |
| 21:50.38 | brettnem | ok, I'll try macroexclusive+global |
| 21:50.42 | brettnem | I believe in libcurl. :) |
| 21:50.56 | brettnem | it's dependable.. and building the backend is really simple |
| 21:50.57 | Corydon76-work | I prefer func_odbc |
| 21:51.18 | brettnem | I've had some.. um.. bad experiences with asterisk included DB routines. :) |
| 21:51.29 | Corydon76-work | Hmm |
| 21:51.48 | Corydon76-work | Should be better now |
| 21:52.00 | brettnem | yeah, that's what I hear. I just need to get over it. :) |
| 21:52.18 | brettnem | but then again.. I like to keep the processing requirement on Asterisk really low |
| 21:53.47 | Hmmhesays | anyone have cdr_pgsql running? |
| 21:54.01 | pigpen | Hmmhesays, yes.... |
| 21:54.09 | pigpen | for several years now. |
| 21:54.42 | Hmmhesays | I'm having trouble getting it to insert records |
| 21:54.48 | pigpen | can anyone think of why when I call a process via the Manager API, after it has successfully completed, I cannot run it again? |
| 21:54.58 | pigpen | Hmmhesays, hmm...sounds like DB permissions. |
| 21:55.11 | pigpen | turn on psql debugging and see if it connects. |
| 21:55.20 | Katty | i... |
| 21:55.24 | Katty | i just found my sister's livejournal. |
| 21:55.37 | pigpen | Katty, well, be nice and share.... |
| 21:55.46 | pigpen | with us. :) |
| 21:55.56 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 21:56.17 | pigpen | blinks = no way. |
| 21:56.22 | brettnem | hmm how do you derefernce a channel variable? :) |
| 21:56.28 | Katty | <PROTECTED> |
| 21:56.48 | pigpen | ah...hopefully it is a small country. |
| 21:56.51 | Katty | i've not seen her in 8 years. |
| 21:56.58 | brettnem | like ${myvar}=stats |
| 21:57.02 | Katty | usa |
| 21:57.04 | brettnem | but then I want ${stats} |
| 21:57.12 | brettnem | I want something like $${myvar} |
| 21:57.19 | pigpen | yeah...kinda a long drive. |
| 21:57.45 | pigpen | shit, it takes me 5 hours just to get out of texas. |
| 21:57.57 | brettnem | pigpen: where are you? |
| 21:58.02 | *** join/#asterisk mmcgrath (n=mmcgrath@74-136-11-144.dhcp.insightbb.com) |
| 21:58.08 | *** part/#asterisk mmcgrath (n=mmcgrath@74-136-11-144.dhcp.insightbb.com) |
| 21:58.18 | pigpen | San Antonio, TX |
| 21:58.27 | Hmmhesays | it gives me a lot of output when I have -d 5 |
| 21:58.29 | brettnem | Dripping Springs here |
| 21:58.32 | pigpen | "Where Men are Men and sheep are scared" |
| 21:58.48 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 21:58.54 | pigpen | Hmmm..Dripping Springs? That is where the "sheep are scared" part comes from. |
| 21:58.56 | pigpen | :) |
| 21:59.06 | brettnem | hey, so anyone know if I can $${myvar} to get the value of the variable in myvar |
| 21:59.15 | brettnem | pfft :P |
| 21:59.34 | pigpen | brettnem, you may want to elaborate. |
| 21:59.39 | antimoof | "yes". |
| 21:59.59 | pigpen | and if I disappear, sorry...Wife is calling. |
| 22:00.00 | Corydon76-work | brettnem: ${${myvar}} |
| 22:00.09 | brettnem | rad |
| 22:00.10 | brettnem | thanks |
| 22:00.11 | Hmmhesays | ok its not even trying to connect |
| 22:00.23 | Hmmhesays | I see my odbc connection, but not the cdr connection |
| 22:01.10 | Corydon76-work | brettnem: actually... is it the NAME that you have stored, or is ${var} inside ${foo} ? |
| 22:01.27 | Corydon76-work | brettnem: if the second, then you need to use ${EVAL(${foo})} |
| 22:01.50 | brettnem | Set(foo=3) |
| 22:01.54 | brettnem | Set(myvar=foo) |
| 22:02.00 | brettnem | I'm using myvar. I want 3 |
| 22:02.17 | Corydon76-work | Okay, then EVAL is not needed |
| 22:02.29 | brettnem | yeah, I'm passing the global variable to increment to the macro |
| 22:02.45 | brettnem | exten => s,1,Set(GLOBAL(${ARG1})=${MATH(${${ARG1}} + 1)}) |
| 22:02.55 | brettnem | almost as ugly as lisp |
| 22:03.07 | Corydon76-work | Uh, why are you using MATH? |
| 22:03.22 | Corydon76-work | $[${ARG1} + 1] is simpler and does the same thing |
| 22:04.05 | brettnem | cool |
| 22:04.15 | brettnem | um |
| 22:04.34 | brettnem | you mean $[${${ARG1}} + 1] |
| 22:04.35 | brettnem | ? |
| 22:04.47 | Corydon76-work | Oh, right |
| 22:04.49 | brettnem | otherwise it's literally foo +1 |
| 22:04.53 | brettnem | ok, great.. thanks for your help |
| 22:05.21 | foo | brettnem: Nope, not literally |
| 22:06.55 | brettnem | hmm.. I wasn't expecting that |
| 22:07.28 | brettnem | now.. how to find out how to keep my globals from being overwritten on a dialplan reload. |
| 22:08.17 | *** part/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 22:08.52 | Corydon76-work | brettnem: clearglobalvars=no in [general] |
| 22:09.36 | brettnem | I have that.. but dialplan reload clears em anyway |
| 22:09.44 | generalhan | anyone here use the 'make config' for zaptel ??? |
| 22:10.32 | brettnem | Corydon76-work: yeah, no question.. clearglobalvars=no is set and extensions reload is showing them all being reset |
| 22:10.51 | Corydon76-work | brettnem: 1.4 or 1.2? |
| 22:10.54 | brettnem | 1.4 |
| 22:10.56 | brettnem | 1.4.9 |
| 22:11.01 | Corydon76-work | hrm |
| 22:11.22 | brettnem | I got an exec in there.. not sure if that messes anything up |
| 22:11.39 | brettnem | Ihave static and writeprotect both yes |
| 22:11.55 | Corydon76-work | Uh, an #exec in [general]? |
| 22:12.13 | Corydon76-work | or in [globals]? |
| 22:13.13 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 22:14.26 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:14.27 | brettnem | bah, let me see |
| 22:15.11 | brettnem | in globals |
| 22:15.11 | puzzled | hi |
| 22:15.58 | Corydon76-work | brettnem: post a bug on that, please |
| 22:17.28 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 22:21.52 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:23.35 | FastFeet | Is there an IRC channel for the Linksys SPA-3102 support? |
| 22:23.46 | *** join/#asterisk nath0099 (i=NLVscrip@77-96-249-156.cable.ubr02.maid.blueyonder.co.uk) |
| 22:24.28 | nath0099 | ok guys need to solve this plan: extension dials 999/911 call goes to pbx and plays messege "this is an emergency number hang up now to cancel or press 1 to continue" by defualt if no number is pressed within ten seconds it continues call. extension presses 1 pbx continues the call, operator answers pbx play messege "emergency call from 'number', 'name', 'address'." pbx connects extension to operator. |
| 22:29.50 | Qwell[] | nath0099: That seems like a very very very very bad idea. |
| 22:29.57 | Qwell[] | Check your local laws/regulations :) |
| 22:30.21 | Strom_C | s/very/very very very very very very very very very very very very very very very very very very very very very very very very very very very very /g |
| 22:30.48 | nath0099 | im in the uk |
| 22:30.54 | Qwell[] | regardless |
| 22:31.00 | Qwell[] | human life == human life |
| 22:31.12 | Qwell[] | it doesn't matter where you are. Denying immediate emergency access is *BAD* |
| 22:31.13 | nath0099 | and all clients go out on one trunk |
| 22:31.39 | nath0099 | ok |
| 22:31.46 | nath0099 | point taken |
| 22:32.10 | nath0099 | but i would still need to be able to give the operator there details |
| 22:32.39 | generalhan | Qwell[]: do you use the make config option with zaptel ??? i cannot get it to load *anything* so as a result asterisk config file fails also |
| 22:33.22 | nath0099 | so minus the "this is an emergency call..... how would i go about the messege with their details |
| 22:36.29 | nath0099 | ext dials 999/911 call connects then plays name address number messege to operator then lets ext talk to operator |
| 22:40.10 | brettnem | nath0099: You're gearing up for a bad lawsuit |
| 22:41.00 | Qwell[] | You'd probably lose too ;/ |
| 22:41.10 | brettnem | yeah |
| 22:41.14 | tzafrir_home | generalhan, which version? |
| 22:41.18 | tzafrir_home | 1.4.4? |
| 22:41.21 | brettnem | the 911 systems typically work pretty well without your intervention. :) |
| 22:41.28 | generalhan | 1.2.23 |
| 22:41.51 | generalhan | i commented out every line except the wct4xxp one |
| 22:42.08 | tzafrir_home | there's no zaptel 1.2.23 |
| 22:42.12 | tzafrir_home | zaptel 1.2.19? |
| 22:42.22 | generalhan | tzafrir: sorry i thought we were talking about asterisk ver |
| 22:43.00 | tzafrir_home | generalhan, are you on centos? |
| 22:43.07 | generalhan | tzafrir_home: yes |
| 22:43.10 | generalhan | 4.5 |
| 22:43.24 | generalhan | wiht zaptel 1.2.19 |
| 22:43.25 | tzafrir_home | you happen to have /etc/default/zaptel but not /etc/sysconfig/zaptel ? |
| 22:43.46 | joshua_d | why do my zap channels go off the hook after about a day of sitting? this morning i came into the office and they were offhook, so i hungup both of the channels. i check about 7 hours later with no real activity and there they are, off hook again. |
| 22:43.48 | generalhan | tzafrir_home: i actually just came across that on digiums site and copied it over ... i am doing the reboot now to see |
| 22:43.54 | nath0099 | ok i know that however all outgoing calls go through one trunk peeps locations very greatly so what is the solution. |
| 22:44.18 | generalhan | tzafrir_home: and that was it !!!!! |
| 22:44.25 | generalhan | loaded them up this time ! |
| 22:44.39 | tzafrir_home | generalhan, actually, if the answer to my q. way "yes", then: mv /etc/default/zaptel /etc/sysconfig/zaptel |
| 22:45.00 | joshua_d | i left the console open with verbosity set high and i see nothing that would cause that, actually there was no activity today because my boss is in the hostiptal having a child. they just randomly go off hook all of the sudden |
| 22:45.08 | tzafrir_home | A known bug "someone" introduced into 1.2.19 and 1.4.4 and fixed in svn |
| 22:45.45 | generalhan | tzafrir_home: i was going off of this: http://bugs.digium.com/view.php?id=10257 ... i followed that and it worked perfectly ... and asterisk started at boot for the first time !! im going to reboot again to check it out cuse it has worked once before and then just stopped |
| 22:46.03 | tzafrir_home | joshua_d, fxs or fxo? what card? |
| 22:46.07 | joshua_d | x100p |
| 22:46.15 | meppl | good night |
| 22:46.17 | joshua_d | 2 of them go offhoook after sitting |
| 22:46.53 | generalhan | tzafrir_home: thanks for the reply, and reconfirmation that, that was what my problem was !! |
| 22:47.07 | tzafrir_home | goes off-hook or starts riinging? |
| 22:47.22 | generalhan | i think ill know after this reboot if its working properlly, it has never worked 2 times in a row before ! |
| 22:47.42 | joshua_d | tzafrir_home: goes offhook after sitting for a bit ^^^ |
| 22:48.06 | joshua_d | ok right now i do i zap show channel 1, it tells me it's offhook |
| 22:48.35 | joshua_d | if i hang it up and go home for the night and come back in the morning it will be offhook again |
| 22:48.38 | *** part/#asterisk nath0099 (i=NLVscrip@77-96-249-156.cable.ubr02.maid.blueyonder.co.uk) |
| 22:48.46 | tzafrir_home | joshua_d, and it's not that it has remained off-hook from a previous call? |
| 22:49.07 | joshua_d | no, there has been no calls placed today |
| 22:50.27 | JT | heh, x100p. |
| 22:50.44 | joshua_d | i will keep the console open another night to see if it is a call that is not getting terminated properly |
| 22:50.48 | joshua_d | JT: small system. |
| 22:51.15 | JT | i wouldn't use the X100p on ANY system |
| 22:52.09 | joshua_d | JT: even if you needed to interface only 1 analog line to your box? what would you recommend then? |
| 22:52.36 | JT | right, i'd probably get a linksys ATA if i had to do it on a budget |
| 22:52.47 | JT | mind you i usually avoid analogue |
| 22:53.05 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 22:54.18 | joshua_d | JT: im looking for an alternative because this is becoming a pain and i know it's because of the crappy cards. what i am doing is interfacing an old panasonic KSU to asterisk so my bosses can have an "office phone" at their house, i got it setup prertty well minus this issue and im only in to it $20 :) |
| 22:54.46 | JT | the ksu isn't at their hous is it? |
| 22:55.24 | joshua_d | JT: no the KSU is at the office, we do a VPN to their house over the internet and they have SIP phones there |
| 22:56.09 | JT | ok |
| 22:56.13 | joshua_d | i got all the rules set up so its virtually the same as sitting in the office, like if they dail 101 they get the receptionist at the office via the pana KSU |
| 22:56.54 | joshua_d | but this little problem is quite annoying, i have to come in and restart * so the stupid cards will hang up |
| 22:59.00 | joshua_d | so what i am looking for is an alternative to the x100p, but something that isn't totally overkill for our situation. i might just break down and get a tdm400p with 2 fxo's but i really dont want to fork out that much money for something that im only $20 in to if you know what i mean :) |
| 22:59.48 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:02.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.148.87) |
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| 23:05.19 | *** join/#asterisk chai_sangeen (n=aaljishi@82-44-68-150.cable.ubr02.nmal.blueyonder.co.uk) |
| 23:05.33 | chai_sangeen | hello everyone |
| 23:06.02 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 23:06.55 | chai_sangeen | i'm having trouble with one way audio... i read most almost all articles but im still facing this problem... the asterisk server is behind a nat but i have all the ports forwarded. |
| 23:07.36 | chai_sangeen | if anyone can help i can provide you with more details about my setup. |
| 23:09.21 | chai_sangeen | i also followed this link http://forums.digium.com/viewtopic.php?t=7854 from digium forums but no luck the client won't even register. |
| 23:14.43 | *** join/#asterisk JoseBravo (n=jbravo@190.9.74.174) |
| 23:14.49 | JoseBravo | Im trying to receive incoming call from external SIP. I added the trunk and the SIP is registered. The asterisk have invalid IP address. When I call from external SIP I can't see anything in the CLI. Any idea? |
| 23:20.55 | *** join/#asterisk saftsack (n=saftsack@pD9E07A84.dip.t-dialin.net) |
| 23:23.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:27.49 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
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| 23:33.44 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 23:33.55 | *** join/#asterisk marcan (i=1337@10.Red-88-24-250.staticIP.rima-tde.net) |
| 23:34.32 | andresmujica | hi, which is the best way to support BRI channels?? at thist moment i'm using bristuff with florz patch, is there any better than this??? |
| 23:35.45 | JoseBravo | Im trying to receive incoming call from external SIP. I added the trunk and the SIP is registered. The asterisk have invalid IP address. When I call from external SIP I can't see anything in the CLI. Any idea? |
| 23:36.17 | JT | andresmujica: no, there isn't |
| 23:36.41 | andresmujica | ok,. TKS! |
| 23:39.42 | fujin | damnit |
| 23:43.11 | fujin | JT: for my callerID thing, should I be using app_dbodbc? or were you referring to something else |
| 23:44.29 | JT | that would probably do the job |
| 23:44.54 | fujin | something better? |
| 23:45.13 | fujin | my Priority based stuff doesn't work. if a callerid doesn't match, it just hangs up ;| |
| 23:45.43 | JT | then make something that matches everything else? |
| 23:46.05 | fujin | can't do that with priorities though.. as far as I could tell |
| 23:47.46 | fujin | I'm sure I'm going about this wrong, just can't find any solid documentation on how to do what I'm trying to achieve |
| 23:49.09 | *** join/#asterisk CVirus (n=GoD@82.201.222.11) |
| 23:53.27 | x86 | hey guys |
| 23:54.04 | x86 | i've only done transfers with SIP phones (namely, Polycom 601's), but I have a need to make a generic analog phone do a transfer |
| 23:54.07 | x86 | how do i do that? |
| 23:55.03 | Mercestes | <PROTECTED> |
| 23:55.08 | Mercestes | Feature code? |
| 23:55.13 | Mercestes | *8 for example? |
| 23:56.37 | x86 | so i hit the flash button, then hit *8 (assuming that's the code in features.conf) ? |
| 23:56.48 | x86 | and it will transfer the active call? |
| 23:57.11 | x86 | just need blind transfer, don't care about attended transfer |
| 23:57.22 | Mercestes | Those were *two* suggestions, not one. |
| 23:58.57 | x86 | so 1. Answer call, 2. flashhook, 3. enter destination extension? |
| 23:59.12 | andresmujica | or use # |
| 23:59.32 | x86 | i like the flashhook option... how do i set that up? |