| 00:02.37 | jesster | Does anyone know if its possible to have auto answer on Polycom with the microphone muted? |
| 00:09.18 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 00:09.26 | *** join/#asterisk Moc____ (~mochouina@Kitchener-HSE-ppp3579343.sympatico.ca) |
| 00:10.32 | Moc____ | hi all |
| 00:10.38 | FuriousGeorge | if i have fxs i need fxo signalling in zapata.conf right? |
| 00:13.20 | *** join/#asterisk [hC] (~turnerd@209.153.195.141) |
| 00:13.31 | [hC] | What is the implication between using a Goto() or Macro() |
| 00:14.46 | [hC] | how do they behave differently |
| 00:17.04 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.219.246) |
| 00:18.37 | *** join/#asterisk hermie (~nick@24.236.167.53) |
| 00:21.18 | [hC] | Ah I see. Macro returns back to the context and Goto does not. |
| 00:21.52 | RaYmAn-Bx | macro supports parameters as well.. |
| 00:22.01 | RaYmAn-Bx | which can be quite useful |
| 00:30.10 | L|NUX | FuriousGeorge : yz |
| 00:30.13 | L|NUX | FuriousGeorge : yes |
| 00:31.22 | FuriousGeorge | thanks |
| 00:42.12 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 00:43.13 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 00:43.13 | masonc | anyone know anything about an adtran 600 |
| 00:43.55 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 00:44.14 | FuriousGeorge | oh shoot i think i got this backwards. i would plug an analog telephone into an fxo, right? |
| 00:44.21 | masonc | fxs |
| 00:44.29 | FuriousGeorge | ok, i got it right |
| 00:44.33 | masonc | think "s" for supply |
| 00:44.40 | FuriousGeorge | for some reason asteriskdocs.org is confusing me |
| 00:46.51 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 00:47.56 | [hC] | hey, any of you guys know of a program, I know ive used it but i cant remember the name, it does realtime snmp polling in windows and draws an mrtg-like graph for you |
| 00:51.34 | FuriousGeorge | Qwell: you would be proud, its all comming together. |
| 00:51.47 | *** join/#asterisk OloBola (~not@h-66-134-67-154.snvacaid.covad.net) |
| 00:51.49 | Qwell | FuriousGeorge: paypal.com > |
| 00:51.50 | Qwell | :p |
| 00:52.32 | FuriousGeorge | Qwell: you keep saying paypal.com, is that irc "slang" im not hip to yet |
| 00:52.44 | FuriousGeorge | does it mean "money" or are you having copy paste difficulties |
| 00:53.07 | adker | "s" is for station |
| 00:53.16 | OloBola | I'm tryiing to access voicemail messages through a samba share but the file permission for the message (msg0000.wav etc) keep resetting. What can I do? I tried to g+s on the folder but no help. |
| 00:53.20 | FuriousGeorge | ya, i remember reading thatnow |
| 00:58.06 | sean | unfortunate: the box I want to put * on is ~60ms away from my DID. How long is "too long" ? |
| 00:59.21 | *** part/#asterisk Lee__ (~Lee__@cpe-69-203-206-248.nyc.res.rr.com) |
| 01:15.41 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 01:16.18 | tzanger | this whole "user must be registered to join this channel" is still bullshit |
| 01:16.31 | tzanger | it was bullshit 6 months ago when it was brought in, and it's still bullshit |
| 01:16.49 | drumkilla | most clients can automatically do it |
| 01:18.42 | tzanger | drumkilla: irssi has trouble with it |
| 01:19.02 | tzanger | it shouldn't, but it does |
| 01:19.21 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 01:19.30 | tzanger | the problem's not registering, it's the fact that after the connection dies for whatever reason it doesn't re-register properly |
| 01:20.02 | kram | kp's connected over his "cox" huh huh |
| 01:20.09 | tzanger | kram: heh |
| 01:20.10 | sean | use screen (-: |
| 01:20.15 | tzanger | I am using screen |
| 01:20.19 | tzanger | I'm talking about network downtime |
| 01:20.24 | kram | we're on the radio :) |
| 01:21.11 | sean | use screen on a box that doesn't go down? (-: |
| 01:21.51 | tzanger | sean: :-) |
| 01:22.31 | sean | is there a way to make asterix EXPORT a CAPI channel? |
| 01:22.53 | sean | I want to (try to) hook into publicvoicexml |
| 01:23.30 | tzanger | http://img.photobucket.com/albums/v11/PyroBlaze/jesuslol.jpg |
| 01:24.06 | tzanger | http://members.cox.net/vespernoir/jesuslol2.jpg |
| 01:24.16 | iq | Hi, could you help me understand cause of this message --->>> chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call 5a6a78ed-c138ce58@172.16.1.39 for seqno 102 |
| 01:24.19 | tainted- | file |
| 01:24.29 | file | eh? |
| 01:24.31 | tainted- | http://voiceconduits.com/ |
| 01:24.35 | *** join/#asterisk jpilot (jirc@217.51.145.15) |
| 01:24.37 | tainted- | Fatal error: Smarty error: the $compile_dir '/www1/voiceconduits.com/smarty/voiceconduits/en/compiles/' does not exist, or is not a directory. in /www1/voiceconduits.com/includes/classes/smarty/Smarty.class.php on line 1083 |
| 01:24.58 | tainted- | oh |
| 01:25.07 | file | if you have problems with Asterlink however, feel free to yell at me |
| 01:25.09 | sean | tainted-: is that your box? |
| 01:25.18 | tainted- | no |
| 01:25.24 | sean | ah.. |
| 01:25.24 | tainted- | i thought file worked for voiceconduits |
| 01:25.41 | Moonwick | I'm sorry. :( |
| 01:25.48 | tainted- | is there such a thing? |
| 01:25.51 | sean | ha |
| 01:25.58 | sean | bring it (-; |
| 01:26.49 | *** join/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 01:26.54 | jpilot | hi |
| 01:26.59 | jpilot | May I ask a question? |
| 01:27.34 | tainted- | sure |
| 01:27.49 | jpilot | Is there a VoIP voice conference version of this room for voice conversation (or anything similar)? |
| 01:27.49 | file | you just did anyway |
| 01:27.59 | jpilot | lol |
| 01:27.59 | luke-jr_ | Is it possible to adjust volume level on an audio stream? |
| 01:28.42 | iq | Do I need to enable this for NAT voice issue ??? --->>> canreinvite=yes ; allow RTP voice traffic to bypass Asterisk |
| 01:28.50 | file | iq: set it to no |
| 01:28.51 | sean | luke-jr_: look for txgain & rxgain |
| 01:29.10 | *** part/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 01:29.40 | *** join/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 01:30.00 | jpilot | No voice conference room version of this room for voice chat then (or similar)? |
| 01:30.19 | jpilot | (using VoIP - e.g. SIP, IAX) |
| 01:30.29 | iq | file, would you know what does this mean "chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call 5a6a78ed-c138ce58@172.16.1.39 for seqno 102 " |
| 01:30.40 | file | iq: asterisk couldn't send the packet |
| 01:30.47 | file | iq: set nat=yes |
| 01:30.56 | iq | file, it is already yes |
| 01:30.57 | luke-jr_ | sean: applications? |
| 01:31.08 | file | then NAT/firewall is blocking it |
| 01:31.50 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 01:31.52 | iq | file, http://pastebin.ca/10620 |
| 01:32.37 | file | if your NAT/firewall is blocking it, nothing asterisk can magically do to make it work... you'll have to explicitly map the SIP port (UDP 5060) and the RTP ports (depends on device, but still UDP) to the device behind the NAT/firewall |
| 01:32.59 | shido6 | work it file |
| 01:33.14 | luke-jr_ | sean: in particular, I want to drop the volume based on caller id |
| 01:33.19 | luke-jr_ | cuz some people are insane loud |
| 01:33.23 | file | luke-jr_: asterisk doesn't touch the audio |
| 01:33.27 | sean | iq: if possible, you could try changing your protocol to IAX (instead of SIP).. |
| 01:33.27 | file | the audio volume |
| 01:33.58 | iq | <PROTECTED> |
| 01:35.21 | file | examine SIP messages? see what we're sending? to where? make sure it's right? examine a good call with another server? |
| 01:35.28 | luke-jr_ | file: then how does it change codecs? |
| 01:35.48 | file | luke-jr_: it converts them to signed linear, but it never touches the volume or anything like that |
| 01:36.05 | file | luke-jr_: And there is no feature for it to, so if you want it - you'll have to muck |
| 01:37.00 | *** join/#asterisk uzd (optimist@eurocompton.net) |
| 01:37.08 | uzd | hello |
| 01:37.36 | file | hello |
| 01:37.43 | uzd | ahi file. |
| 01:37.46 | uzd | -a |
| 01:37.57 | file | what does thou need? |
| 01:38.29 | FuriousGeorge | i have only one context so far for outgoing calls, and i have the extensions on the first priority, and a catchall for outobound number "after" those. how do i tell it to stop after making the first matching cll and not go on to priority two? |
| 01:38.36 | *** join/#asterisk shepherd (~matt@207.111.174.1) |
| 01:38.42 | OloBola | I'm tryiing to access voicemail messages through a samba share but the file permission for the message (msg0000.wav etc) keep resetting. What can I do? I tried to g+s on the folder but no help. |
| 01:38.48 | kram | http://dishnuts.net/rfd1.m3u |
| 01:38.49 | L|NUX | can i use digium g729 just for pass through ? |
| 01:38.53 | uzd | well. I'm having a problem with incoming calls. outbound work fine, but incoming are going straight to voicemail. |
| 01:39.06 | blitzrage | http://abcnews.go.com/US/wireStory?id=715583 |
| 01:39.07 | L|NUX | only for sip to sip ? |
| 01:39.09 | blitzrage | kram: ! |
| 01:39.09 | tainted- | L|NUX good question |
| 01:39.10 | shepherd | that link is for a a radio show that mark is on |
| 01:39.18 | shepherd | http://dishnuts.net/rfd1.m3u |
| 01:39.45 | L|NUX | tainted- : because there is - - in translation |
| 01:40.28 | shepherd | as long as you don't have to use voicemail or playback prompts, you don't need g729 |
| 01:41.04 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 01:41.15 | shepherd | manx! |
| 01:41.17 | L|NUX | hmm |
| 01:41.19 | ManxPower | or at least the patron saint of echo canceling hardware. |
| 01:41.25 | shepherd | http://dishnuts.net/rfd1.m3u |
| 01:41.28 | tainted- | heh |
| 01:41.30 | ManxPower | hello shepherd |
| 01:41.36 | *** join/#asterisk torisa (hak@heceta.db.net) |
| 01:41.40 | uzd | file, would you be able to assist? |
| 01:41.46 | jpilot | bye |
| 01:41.55 | file | uzd: pastebin what your console says |
| 01:41.55 | ManxPower | tainted-, Let me put it this way. The echo can shelf uses WIRE WRAP PINS |
| 01:42.13 | uzd | file, with verbose at 4 and sip no debug, i don't see a thing |
| 01:42.40 | file | I don't care, I neeeed console - I have a hunch |
| 01:42.52 | L|NUX | shepherd : but when i try to call my another sip account i can't hear any voice neither my another person |
| 01:42.57 | uzd | ok, but there's nothing to pastebin... |
| 01:43.18 | file | okay, so whose voicemail is it going to? |
| 01:44.06 | FuriousGeorge | when i ahve several extensions in a context in the same priority, is there any way to tell it not to call the "catch-all" extension at the end of the priority |
| 01:44.09 | *** join/#asterisk Rob-W0KIE (Rob@ip68-14-182-167.pn.at.cox.net) |
| 01:44.33 | Rob-W0KIE | hello |
| 01:44.45 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 01:44.58 | R0B | what the hell? |
| 01:45.00 | uzd | file, it says "the party you are trying to reach..." |
| 01:45.07 | uzd | not sure. it should be going to my vm |
| 01:45.15 | file | uzd: so nothing shows up on the CLI? |
| 01:45.28 | uzd | not if i have sip debugging disabled |
| 01:45.35 | uzd | with sip debugging i see the call come in |
| 01:45.47 | ManxPower | and we had to soder wires to the -48v power supply because it had the wrong connector. |
| 01:45.49 | Rob-W0KIE | well looks lilke im stuck with this |
| 01:45.50 | file | oh just pastebin the entire thing with sip debug turned on |
| 01:45.51 | K9DI_BSD_WrkStn | hello |
| 01:45.55 | file | I'll look, yell at you, and it'll start working |
| 01:45.56 | uzd | ok |
| 01:45.58 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
| 01:46.02 | uzd | :) |
| 01:46.02 | ManxPower | and the amphenol cable we had was not wired the way we expected it to be |
| 01:46.31 | Rob-W0KIE | whoever is in #w0kie did they get the links here? |
| 01:47.15 | tainted- | ManxPower where'd you learn all this telephony stuff |
| 01:47.44 | uzd | file: http://rafb.net/paste/results/q658bb19.html |
| 01:47.48 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) |
| 01:47.54 | Rob-W0KIE | there chatting in our room too |
| 01:47.59 | ManxPower | tainted-, I have no idea |
| 01:48.17 | tainted- | i'd like to find a few books |
| 01:48.32 | tainted- | get a better background on telco stuff |
| 01:48.45 | shepherd | http://dishnuts.net/rfd1.m3u <-- w0kie radio :) |
| 01:48.51 | Rob-W0KIE | hi wayne |
| 01:49.04 | Rob-W0KIE | www.cband.info select "our audio" |
| 01:49.11 | peted20 | Hi all, has anyone had luck integrating a AT&T Partner system with *? |
| 01:49.14 | Rob-W0KIE | theres a 64K and 16K stream |
| 01:50.34 | file | uzd: ah it's broadvoice |
| 01:51.37 | file | uzd: okay, is your register line correct according to the docs? and btw, Looking for s in default - s isn't in default, so we send back a Not Found thus why you get broadvoice voicemail |
| 01:51.42 | Rob-W0KIE | shepard you can also give them http://www.dishnuts.net as well |
| 01:52.46 | file | ack my touchpad has gone crazy |
| 01:53.11 | *** join/#asterisk foh (~foh@adsl-17-162-151.bhm.bellsouth.net) |
| 01:53.20 | uzd | hrm |
| 01:53.39 | file | I suspect your register line isn't quite right |
| 01:54.07 | uzd | file but how can I make outgoing calls? |
| 01:54.18 | uzd | isn't the register line necessary? |
| 01:54.23 | file | uzd: you read the documentation, cause believe me - this is VERY well documented |
| 01:54.32 | uzd | I have read it |
| 01:54.35 | file | as for register, yes i'ts necessary but it's not right |
| 01:54.36 | uzd | i had it working actually |
| 01:54.58 | uzd | machine crashed and here I am :( |
| 01:56.12 | FuriousGeorge | is there any way to stop your dialplan from calling the "catch all" extension " (_.) at the end |
| 01:56.21 | file | don't use _. |
| 01:56.23 | file | NEVER use _. |
| 01:56.28 | file | unless you have a VERY good reason to |
| 01:56.34 | FuriousGeorge | file: im testing |
| 01:56.40 | file | not good enough |
| 01:56.49 | tzanger | even if you have a good reason to, try it another way first |
| 01:57.01 | Rob-W0KIE | hey file that's a radio room you just joined |
| 01:57.14 | Rob-W0KIE | its active 7 days a week |
| 01:57.17 | file | I know |
| 01:57.46 | uzd | file, so we're done? :) |
| 01:57.49 | Rob-W0KIE | i had to post those rules because we had some trolls come in and cause trouble |
| 01:58.49 | file | uzd: just follow the docs - I know for a fact they work |
| 01:58.56 | file | plus I don't do broadvoice ;) |
| 01:59.09 | uzd | but this has nothing to do with broadvoice |
| 01:59.19 | Rob-W0KIE | i just use ventrilo that's sorta like voip |
| 01:59.20 | file | yes it does, you have to do special stuff with broadvoice |
| 01:59.25 | uzd | the reg line is the exact same line i had when it was working |
| 01:59.31 | Fulcrm | Furious: I think I might be able to help you. From the way you've phrased your question, I wonder if you've got something set funny. The term 'priority' can be a bit decieving. Consider the word 'step'. You should end up with (for example) |
| 01:59.31 | Fulcrm | EXTEN => 1234,1,dial(SIP/1234) |
| 01:59.31 | Fulcrm | EXTEN => 1234,2,voicemail(u1234) |
| 01:59.31 | Fulcrm | It sounds like your trying to use that second parameter to prioritize pattern matching rather than to indicate the sequence of events for an individual extension. |
| 01:59.32 | uzd | char for char |
| 01:59.40 | file | uzd: read the broadvoice page and try THEIR register line :) |
| 01:59.46 | uzd | i have read it |
| 01:59.49 | uzd | I'm using theres |
| 01:59.53 | uzd | theirs |
| 02:00.26 | uzd | ok. thanks for your time. |
| 02:00.40 | file | try voip-info.org too for configs |
| 02:01.09 | uzd | file, you were right the whole time. |
| 02:01.21 | uzd | i backed up the original register line |
| 02:01.27 | file | :) |
| 02:01.40 | uzd | which was register=PHONENUMBER instead of register => PHONENUMBER |
| 02:01.52 | file | and it's all better? |
| 02:01.53 | uzd | sorry for doubting you. |
| 02:01.55 | uzd | well |
| 02:01.57 | Rob-W0KIE | if anyone wants to listen to the show go to http://www.cband.info select "our audio" |
| 02:01.59 | uzd | haven't tried |
| 02:02.01 | uzd | but I think it will |
| 02:02.19 | file | haha maybe |
| 02:03.03 | uzd | nope. not yet. |
| 02:03.28 | file | try putting... this is evil... /<phone number> at the end |
| 02:03.34 | uzd | fg |
| 02:03.43 | file | of the register line |
| 02:04.39 | Rob-W0KIE | holy crap |
| 02:04.41 | foh | thud |
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| 02:04.55 | *** join/#asterisk marlowe (~marlowe@marlowe.active.supporter.pdpc) [NETSPLIT VICTIM] |
| 02:04.55 | *** join/#asterisk jontow (jontow@ws.woflsys.net) [NETSPLIT VICTIM] |
| 02:05.09 | Rob-W0KIE | and there back |
| 02:05.15 | foh | FORTY-TWO members |
| 02:05.30 | Rob-W0KIE | 42 ehhh! |
| 02:05.41 | foh | and only 41 came back |
| 02:06.01 | foh | (you desparately needed to know this) ;) |
| 02:06.05 | Rob-W0KIE | how are the cats? |
| 02:09.10 | uzd | file, what's this? SIP/2.0 484 Address Incomplete |
| 02:10.43 | Fulcrm | FuriousGeorge: Did you get your question worked out? |
| 02:11.49 | syle | is cvs checkout asterisk safe or do you guys use only stable tree? |
| 02:12.08 | Qwell | cvs is usually fine |
| 02:15.39 | uzd | ok, got it working |
| 02:15.53 | file | what was it? |
| 02:16.29 | uzd | the context in [general] |
| 02:16.54 | uzd | i had it commented out which is why it was looking for the extension in default |
| 02:16.56 | uzd | :) |
| 02:17.22 | uzd | :q |
| 02:18.02 | uzd | thanks for your help |
| 02:18.05 | ManxPower | I'm not getting any echo can love |
| 02:19.31 | syle | how do i check what else is in that cvs tree |
| 02:19.57 | syle | sh: dot: command not found |
| 02:19.57 | syle | Problems running dot. Check your installation! |
| 02:20.04 | syle | what package is dot from? |
| 02:20.09 | syle | for sh: dot: command not found |
| 02:20.09 | syle | Problems running dot. Check your installation! |
| 02:20.12 | syle | errr |
| 02:20.20 | syle | for make progdocs |
| 02:20.51 | newl | graphviz |
| 02:24.49 | syle | so what do you need |
| 02:24.56 | syle | devel tcl and main file |
| 02:25.14 | syle | http://www.graphviz.org/Download_linux.php |
| 02:28.18 | syle | for /etc/zaptel.conf if you have a TDM card with 2 fxs and 2 fxo modules do you put fxoks=1 |
| 02:28.22 | syle | errr |
| 02:28.27 | syle | fxoks=2 |
| 02:28.32 | iq | hi, anyone running Asterisk behind a router? |
| 02:28.37 | syle | fxsks=2 |
| 02:28.39 | syle | ? |
| 02:29.00 | Qwell | syle: fxoks=1 \n fxoks=2 \n fxsks=3 \n fxsks=4 |
| 02:29.05 | Qwell | the number is for the channem number |
| 02:29.09 | Qwell | channel* |
| 02:29.11 | Qwell | ...I think |
| 02:29.15 | syle | ohhh |
| 02:29.21 | syle | ok thanks |
| 02:31.05 | syle | so i could do |
| 02:31.18 | syle | fxoks=1,2 |
| 02:31.25 | syle | fxsks=3,4 |
| 02:32.20 | syle | wish my card had arrived just doing pre-setup |
| 02:32.28 | Fulcrm | Syle: that syntax works or fxoks=1-2 \n fxsks=3-4\n. That syntax is more convenient when you're setting up 23 ISDN channels for example. |
| 02:32.55 | syle | yeah i guess so |
| 02:33.02 | syle | for those T1 cards i presume |
| 02:33.17 | Fulcrm | Syle: that's right. |
| 02:34.57 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 02:36.04 | Fulcrm | Syle: When you get the card and you run 'ztcfg -vv' pay attention to which modules it reports in which slots. I can't remember if they come with the FXO or the FXS ports first. Also, you may know that FXO ports use FXS signalling and vise-versa |
| 02:37.41 | iq | Can someone recommend a router to use on Asterisk side that can forward RTP ports? |
| 02:38.48 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 02:39.15 | iq | I think the reason for no audio on this machine is that router can forward only TCP and UDP, nothing else. |
| 02:39.34 | Fulcrm | IQ: I've used a few different linksys boxes like BEFSR and BEFVP that support it. My Netgear does port forwarding but not port translation. IE I can't take port 5060 from WAN and send it to 5061 on my * box. |
| 02:40.54 | syle | thanks for that tip, i been wondering why these configs seem to be reversed ie fxoks for fxs port |
| 02:41.13 | syle | well online tutorial guides anyways |
| 02:42.10 | syle | now what about zapata.conf |
| 02:42.15 | syle | signalling=fxo_ks |
| 02:42.25 | iq | Fulcrm, this machine is connected to a Belkin router. it can send 5060 from WAN to 4061 on *... BUT it does not do RTP |
| 02:42.28 | syle | do that for fxs 2 ports and 2 ports fxo |
| 02:42.28 | ManxPower | I GOT LIGHTS! |
| 02:42.29 | CoaxD | Who the FUCK is "VocalScape", and why the FUCK did they spam me?? |
| 02:42.36 | Qwell | ManxPower: lights? |
| 02:42.39 | [hC] | Hey coax |
| 02:42.43 | [hC] | Vocalscape is a company here in vancouver |
| 02:42.43 | CoaxD | they're all "Hi, we just created a turnkey voip system for ISPs!" |
| 02:42.48 | [hC] | been trying to do the voip thing for years. |
| 02:42.48 | ManxPower | Qwell, on the echocan shelf |
| 02:42.54 | CoaxD | hC: Heya dood |
| 02:42.56 | Qwell | ManxPower: I haven't been paying attention |
| 02:43.00 | Fulcrm | Syle: I've setup several * boxes and I still make the same mistake every one in while by reversing those. |
| 02:43.02 | CoaxD | hC: Really? They'll never get my business. They spammed me |
| 02:43.04 | Qwell | CoaxD: I think I got that same email a few weeks ago... |
| 02:43.05 | CoaxD | hC: Fuckers |
| 02:43.07 | [hC] | Yah |
| 02:43.14 | CoaxD | Qwell: I wonder where they got my email address.. |
| 02:43.26 | CoaxD | Qwell: I.e. what fucking company that I did business with sold it? |
| 02:43.27 | iq | Fulcrm, we can't do RTP on UDP ports? |
| 02:43.44 | Qwell | CoaxD: spammers don't need to buy addresses |
| 02:44.04 | CoaxD | I'm PISSED. Instead of letting me hear about it via word of mouth, they chose to spam me and lose any chance they *ever* had of doing business with me |
| 02:44.12 | CoaxD | too bad, too. sounded like they had a good model |
| 02:44.17 | CoaxD | (Fuckers.) |
| 02:44.39 | Fulcrm | Syle, I just checked my zapata and it's reversed too. |
| 02:45.21 | Veto | got something weird on my fwdexten (just stdexten but with a Dial for sNOANSWER to cell #) sip -> sNOANS gets horrible white noise over ring. |
| 02:46.35 | Veto | so the Dial works (out via IAX provider), but there is horrible noise over it. From external it works fine. |
| 02:47.30 | Fulcrm | IQ: I can't say that I've ever worked with a Belkin. I can't see why it wouldn't send those RTP packets if they're just UDP traffic. I'm not sure what your time's worth but you might consider grabbing another brand of router for $30. |
| 02:48.23 | syle | how do i do zapata.conf for 2 fxs/2 fxo ports? i have example here of signalling=fxo_ks \n context=incoming \n channel => 1 do i keep on adding ports like this IE: signalling=fxo_ks \n context=incoming \n channel=> 2 etc? |
| 02:48.34 | iq | Fulcrm, okay. I'll do that... the machine is a little too far from where i'm located. but u r right. thanks :) |
| 02:48.38 | syle | s/ports/channels |
| 02:48.59 | Veto | anyone get real bad white noise on a Dial()? |
| 02:49.16 | Fulcrm | IQ: That's a bummer. Are both ends behind NAT? |
| 02:50.15 | iq | Fulcrm, yes.. and I've been wasting time since yesterday. I tried to move it in DMZ but still no help :) |
| 02:51.04 | Fulcrm | IQ, Are you seeing the SIP traffic hitting the *box? IE do you get the phone to ring and then there's no audio? |
| 02:51.13 | syle | or do you define a context for each channel |
| 02:51.19 | syle | that would make more sense i guess |
| 02:52.11 | syle | i guess thats my last question for the next week till my card gets here all i can do lol |
| 02:52.15 | Fulcrm | Syle, one context usually contains multiple channels. In a simple situation, you'd have one context for your phones and another for your outgoing CO trunks. That's a simple-simple situation. |
| 02:52.25 | iq | Fulcrm, thats exactly whats happening. It rings. I get the Caller ID. But there is no audio. I tried forcing different codecs, etc... but I didn't realize that that opening UDP:10k-20k doesn't mean RTP:10k-20k |
| 02:53.15 | syle | fulcrm what i don;t understand is how to define 2 fxs and 2 fxo channels in one context under signalling and channel variables |
| 02:53.48 | syle | maybe you can do signalling=fxo_ks, fxs_ks and channel=>1-4 |
| 02:53.51 | syle | i don;t know |
| 02:55.10 | *** join/#asterisk asdj992 (~dSda@CPE000f66a9409f-CM000e5ce672fe.cpe.net.cable.rogers.com) |
| 02:55.48 | asdj992 | got a problem hopefully someone can help |
| 02:56.09 | asdj992 | have asterisk running in my office 10 cisco 7690's 10 uiden phones |
| 02:56.26 | asdj992 | 2 x digi 4 port fxs boards for 8 incoming analog lines |
| 02:56.32 | asdj992 | everything is working grea |
| 02:56.32 | asdj992 | t |
| 02:56.36 | Qwell | fxo |
| 02:56.38 | Qwell | not fxs |
| 02:56.41 | asdj992 | sorry fxo |
| 02:56.42 | syle | i guess what i am worried about is if you do signalling=fxo_ks \n channel=>1 then do signalling=fxs_ks \n channel=>3 if variables will overwrite each other in same context so wondering how to do it properly |
| 02:57.07 | Fulcrm | Syle: basically mine follows a format like this: [phone-context] \n mailbox=2000@default \n callerid= "Bill" <2000> \n channel => 1\n \mailbox=2001@default \n callerid= "jim" <2000> \n channel=2. Properties are inherited downwards unless overridden. Am I making sense? |
| 02:57.12 | asdj992 | so some times ppl pick up the phone to make a call, but instead of making that call, they get connected to an incoming call before it rings the office |
| 02:57.30 | asdj992 | which is a problem |
| 02:57.45 | asdj992 | any ideas? |
| 02:58.01 | syle | so you can define the same variable in same context ok |
| 02:59.40 | syle | fulcrm don;t you define signalling type for that channel at all? |
| 03:00.57 | Fulcrm | syle, oh yeah. It's almost a must. Let's say you declare variables a,b,c,d you set the context once then you can declare a channel. you can override variable b and declare another channel. then override b again and declare another channel. The result is that each channel will have the same value for a,c,d but different values for b. |
| 03:01.17 | *** join/#asterisk iq|laptop (~iq@204-26-74-173.omah.qwest.net) |
| 03:02.13 | *** join/#asterisk hairball (~Unknown@2b91f6e890c0e001.node.tor) |
| 03:03.56 | Fulcrm | IQ: I'm not sure what to tell you about that SIP RTP issue. Most of what I've done was with IAX and it sidesteps all of those issues. Sorry :( |
| 03:04.15 | syle | so how does that make sense in your example then, wouldn;t your callerid be the same for all channels then by that explanation? |
| 03:04.44 | syle | since they are all under the same context phone-context |
| 03:05.32 | Fulcrm | Syle, nope. Look at it this way. That file is setting properties of the channel, not of the context. The context is actually a property of the channel, not the other way around. |
| 03:05.35 | syle | makes more sense to me to define a new context for each channel keeps shit simple i think and easy to read |
| 03:05.36 | iq|laptop | Fulcrm: thanks :) ...I'm reading RTP it does run on top of UDP "RTP runs on top of the User Datagram Protocol (UDP), although it can use other transport protocols"... |
| 03:06.01 | iq|laptop | Fulcrm: even I dont know whats wrong :) ...so will give you an update if I get it working |
| 03:07.39 | syle | alright then i assume the config file is being read into an array then, callerid[0] associated with channel 1 callerid[1] associatied with channel 2 from whatever regex they are parsing file with |
| 03:11.14 | *** part/#asterisk hairball (~Unknown@2b91f6e890c0e001.node.tor) |
| 03:11.36 | syle | where is the source code that parses these file located |
| 03:11.43 | syle | that might help alot lol |
| 03:11.55 | Fulcrm | when you declare that channel with channel => x it will get the most recent value of every variable defined in the file. If callerid is declared more than once, then it gets the most recent one. So in my example channel 1 has the following properties: context = phone-context, mailbox=2000@default, callerid="bill" <2000> and channel 2 has these properties: context = phone-context, mailbox=2001@default, callerid="ji |
| 03:12.47 | Fulcrm | Syle, sorry I'm not a programmer type nerd, I'm networking/systems analyst/telecom nerd. It's somewhere in your /src/asterisk directory but I couldn't say where to go from there. |
| 03:14.04 | Fulcrm | I just realized a typo in my example where it says [phone-context] it should say context=phone-context. Each of these files uses slightly different styles and it's easy to get them confused. |
| 03:14.51 | Sedorox | wow.... StarGate has a Aastra VoIP phone.... |
| 03:14.53 | *** join/#asterisk los415 (~los415@adsl-68-122-69-92.dsl.pltn13.pacbell.net) |
| 03:14.59 | syle | i imagine these contexts relate to each other between files not having set this up yet myself |
| 03:16.47 | Fulcrm | Actually, you might find it easier to write your extensions.conf file first. That's where you actually describe the meaning of a context. In zapata, all you're doing is associating a channel with a context. |
| 03:18.51 | *** join/#asterisk brook (~brook@210.82.30.5) |
| 03:19.22 | syle | here we go found a good example |
| 03:19.31 | syle | <PROTECTED> |
| 03:19.31 | syle | <PROTECTED> |
| 03:19.31 | syle | <PROTECTED> |
| 03:19.31 | syle | <PROTECTED> |
| 03:19.31 | syle | <PROTECTED> |
| 03:19.47 | syle | the "national" switchtype would be applied to channels one through |
| 03:19.47 | syle | four and channels 10 through 12, whereas the "dms100" switchtype would |
| 03:19.47 | syle | apply to channels 25 through 47. |
| 03:20.13 | syle | so it seems the channels are being defined as objects |
| 03:20.19 | syle | that is how it gets away with it |
| 03:20.51 | Fulcrm | Syle, that's exactly right. The channel is an object with the all of the properties defined in the file to that point, subject to over-riding. |
| 03:21.25 | *** join/#asterisk alpha255 (~thomas@209.237.13.82) |
| 03:24.01 | ManxPower | I just called Tellabs Tech support. |
| 03:24.01 | *** join/#asterisk bjmitchell (~brett_mit@68.248.103.222) |
| 03:24.11 | ManxPower | US$750 per support incident |
| 03:24.43 | iq|laptop | ManxPower: what do they sell |
| 03:24.54 | bjmitchell | anyone from digium around |
| 03:27.08 | bjmitchell | Ive got a t100P that I think has gotten itself a lobotomy |
| 03:27.21 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
| 03:27.27 | ManxPower | they sell t-1 echo cancelers |
| 03:27.41 | bjmitchell | need some advice |
| 03:29.22 | *** part/#asterisk brook (~brook@210.82.30.5) |
| 03:29.41 | *** join/#asterisk jon_ (~jon@69-170-9-142.chvlva.adelphia.net) |
| 03:30.12 | Fulcrm | BJ, what's going on with yoru T100P? |
| 03:30.28 | jon_ | what do I need besides T1 and asterisk to become a voip provider? |
| 03:31.03 | puowvip | merchant account? |
| 03:31.17 | puowvip | how you going to bill? |
| 03:31.56 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 03:32.48 | tengulre | Hi,all I'm coming |
| 03:33.00 | bjmitchell | Installed in a HP DL140 |
| 03:33.12 | syle | why do you need a T1 to be voip provider? |
| 03:33.14 | bjmitchell | Was recognized, then after a reboot disappeared |
| 03:33.15 | jon_ | there will only be one customer, our company ;) I dont need to bill |
| 03:33.40 | bjmitchell | ZT_SPANConfig error |
| 03:33.57 | bjmitchell | thought that it was was an IRQ problem |
| 03:34.30 | tengulre | astman ,"connection refused!" |
| 03:34.36 | *** join/#asterisk outtolunc (~me@adsl-69-110-56-125.dsl.pltn13.pacbell.net) |
| 03:34.45 | Fulcrm | BJ, what's this DL140? Printer, telephony? |
| 03:34.51 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 03:34.52 | bjmitchell | The HP 130 has a terible bios, can set /reserve or really do anything |
| 03:34.53 | Micc | jon_, I imagine you just need asterisk and some trunk lines. |
| 03:35.06 | bjmitchell | DL 140 proliant server |
| 03:35.12 | jon_ | trunk lines?? please explain |
| 03:35.28 | Fulcrm | Sorry, duh, I misread that. I'm with you now. What other cards are in the system? |
| 03:35.36 | Micc | jon_, you don't really even need that. Actually, I don't know what trunk lines means. I pulled it off the top of my head because it sounds good. |
| 03:35.38 | bjmitchell | So I bought an IBM x306 instead, same problem |
| 03:35.48 | bjmitchell | no other cards |
| 03:35.59 | DeeJayTwo | What hardware SIP Phone could I use to have 6 lines and an average of 12 directory button.. (programmed extension) |
| 03:36.12 | Micc | jon_, But i assume you want people to use your sip service to do internal calls and outbound calls to regular telephones |
| 03:36.16 | tengulre | HI, when I use astman, got some error message "connection refused" ,why? |
| 03:36.17 | *** join/#asterisk file (~file@mctn1-8111.nb.aliant.net) |
| 03:36.23 | syle | there is no money in being voip provider |
| 03:36.27 | Fulcrm | BJ, does it modprobe OK? |
| 03:36.27 | outtolunc | re file <G> |
| 03:36.33 | file | :P |
| 03:36.42 | bjmitchell | modprobe zaptel - yes |
| 03:36.50 | bjmitchell | modprob wct1xxp - no |
| 03:37.00 | bjmitchell | zttool shows no cards either |
| 03:37.04 | jon_ | so what do I need? How do I connect to the pstn. And tell the worl that to get to our phone number, go to our server |
| 03:37.17 | tengulre | HI, when I use astman, got some error message "connection refused" ,why? |
| 03:37.29 | Micc | jon_, there are many ways to offer outbound calling to the real world. you could use another provider or you could get local lines from your telco and use some hardware like the stuff digium sells. |
| 03:37.43 | tengulre | I 'm beginner! |
| 03:38.05 | tengulre | how to do running astman? |
| 03:38.12 | Micc | jon_, you want to tell the world to get to your phone number to go to your server? I don't understand. |
| 03:38.15 | syle | ummm micc wouldn;t it being more expensive to get local lines vs just sip peering with someone? |
| 03:38.17 | outtolunc | tengulre: edit /etc/asterisk/manager.conf set enable=yes |
| 03:38.29 | Fulcrm | It has to modprobe successfully before ztcfg is going to work. |
| 03:38.34 | Micc | style, yes but depends on what you want to do. |
| 03:39.14 | Micc | style, I still think of voip as fun technology. |
| 03:39.27 | Micc | style, there are too many factors that go into a good sip/iax connection. |
| 03:39.49 | jon_ | to go to our asterisk pbx.....then to our ip phones. |
| 03:39.50 | Micc | style, pstn is pretty much the same all the time. |
| 03:39.56 | bjmitchell | fulcrm, that i know.. I just trying to decide if im crazy or the card is bad |
| 03:40.12 | Fulcrm | BJ, gut reaction is a hardware problem but you've already swapped the machine. Hmmmm. You know, this sounds lame but I've had some luck with shutdown-remove card-boot-shutdown-insert card-boot. |
| 03:40.13 | syle | well the problem with voip is internet can be unstable at so many hops |
| 03:40.25 | tengulre | outtolunc,thank you ,it's OK! :) |
| 03:40.40 | Micc | jon_, you want your company to have SIP phones that people can call a regular number to get to or you just want sip phones that people can get to from another sip phone? |
| 03:40.48 | Fulcrm | BJ, does kudzu detect the card? |
| 03:40.58 | syle | and people often ask the question: can i use my internet and download movies etc while talking on the phone |
| 03:41.09 | bjmitchell | fulcrm, done that about 10 times. even put in other PCI cards intpo the slot to make sure the bios kudzu and the like all do a fulsh |
| 03:41.34 | tengulre | outtolunc, but i got a error message "Broken pipe"! |
| 03:41.36 | jon_ | people from anywhere in the worl can call our phones number we can call anyone |
| 03:41.57 | outtolunc | thats fairly normal, it's from mpg123 |
| 03:42.01 | bjmitchell | fulcrm, sees it as a Tiger Jet Network card |
| 03:42.35 | Fulcrm | BJ, tiger jet is correct. That's the platform that Digium used for the card. |
| 03:43.05 | Fulcrm | BJ, that's a stubborn bugger. I'll tell you this, Digium's 'installation support' is pretty generous. If you open a ticket with them and then work through it with them, you'll get some sanity and either a solution or an RMA. |
| 03:43.17 | Micc | jon_, you just need asterisk and some sip or iax lines from another provider that has pstn DIDs |
| 03:43.41 | bjmitchell | fulcrm, but you should skip it in kudzu right? |
| 03:43.42 | Micc | I love this, I still have not a clue what I am saying but it sounds good to me. |
| 03:44.24 | Micc | all these acronyms are great. They are all new to me in the last week or two. |
| 03:44.26 | syle | yeah that is quite cool fulcrum, i mean if your running a production asterisk system, here digium, here is root login/password, ip address please help me with this.... |
| 03:44.45 | tengulre | <PROTECTED> |
| 03:44.53 | tengulre | tks |
| 03:45.09 | Fulcrm | BJ, I've done both ignore and configure with kudzu and they seem to work in both cases. Strange but true |
| 03:45.39 | outtolunc | tengulre: thats probably not the ONLY error set.. what are the others.. you should only get that when you shutdown asterisk |
| 03:45.54 | bjmitchell | fulcrm, if i swap it for a 400p 1fxo 1fxs, the 400 gets recognized just fine and shows up in zttool |
| 03:46.09 | DeeJayTwo | I got problems with 400P... |
| 03:46.16 | DeeJayTwo | sometimes, I heard some strange sounds.. |
| 03:46.25 | DeeJayTwo | I need to unload and reload the wcfxs module.. |
| 03:46.27 | DeeJayTwo | and it works.. |
| 03:46.38 | jon_ | could I use our existing t1 line that we have for our data? |
| 03:46.53 | outtolunc | DeeJayTwo: i'm assuming you mean a TDM400P, because there is T400P |
| 03:47.10 | DeeJayTwo | TDM400P yeah.. |
| 03:47.18 | DeeJayTwo | is there some problems with it? |
| 03:47.20 | Micc | thanks for that telecom glossary tip. now I know what I'm talking about, sort of. |
| 03:47.21 | outtolunc | what watt ps? |
| 03:48.12 | outtolunc | also, make sure the power lead you use for the TDM400P isn't a 'chained' lead (feeding an HD example) |
| 03:48.12 | Fulcrm | BJ, indications are that the T1 card is flaked out. I know what you're going through, I spent my weekend fighting with one of the new TE110Ps just a month ago. You start to think you're losing your marbles. |
| 03:48.34 | puowvip | anyone seen the h2g2 movie yet? |
| 03:48.40 | tzanger | nope |
| 03:48.50 | DeeJayTwo | outtolunc: it's a 450 Watts.. |
| 03:48.54 | tzanger | outtolunc: wtf does that have to do with it |
| 03:48.59 | DeeJayTwo | Really enough.. |
| 03:49.00 | tzanger | chained or no |
| 03:49.03 | outtolunc | that's plenty, and the chained lead? |
| 03:49.12 | bjmitchell | My marbles ive found, its the 20 hours ive spent trying to figure the card problem out that im looking for :) |
| 03:49.13 | DeeJayTwo | I don't know... |
| 03:49.16 | outtolunc | it's common sense |
| 03:49.20 | DeeJayTwo | I don't have this server in front of me now.. |
| 03:49.20 | tzanger | the TDM400P needs a decent (couple hundred mA) 12V source. having a HDD on it is not a problem |
| 03:49.20 | DeeJayTwo | :/ |
| 03:49.23 | tzanger | it's not common sense |
| 03:49.26 | tzanger | are you an EE? |
| 03:49.39 | outtolunc | you build 4000 systems, then you start asking Q's like that |
| 03:49.46 | tzanger | outtolunc: I design systems |
| 03:49.49 | Fulcrm | BJ, I'm no Linux hardware guru, I only learned Linux so that I could play with Asterisk. But if I were a betting man, and if I had any money, I'd bet that getting an RMA on that card is a good idea. |
| 03:50.02 | bjmitchell | will do |
| 03:50.10 | outtolunc | and you are telling me there is NO rf feed back on chained power leads <G> |
| 03:50.15 | bjmitchell | assuming that Ill have to wait till monday |
| 03:50.26 | tzanger | RF feedback? jesus christ stop reading the glossaries and start getting a real understanding |
| 03:50.34 | outtolunc | haha |
| 03:50.50 | Fulcrm | BJ, I don't suppose you're in the Toronto, Ontario area. I've got one that I could loan for the weekend. |
| 03:50.57 | DeeJayTwo | so.. |
| 03:51.01 | outtolunc | rf feedback is a simple term for power flux when resistance load is off |
| 03:51.03 | tzanger | the impedance of a decent power supply should effectively eliminate any RF |
| 03:51.04 | bjmitchell | close but no cigar |
| 03:51.07 | bjmitchell | chicago |
| 03:51.10 | DeeJayTwo | what could make things go wrong with my TDM400P..? |
| 03:51.12 | syle | anyone from winnipeg , manitoba? |
| 03:51.17 | DeeJayTwo | it makes some noisy sounds... |
| 03:51.29 | DeeJayTwo | and isn't working until a unload/reload of module and asterisk.. |
| 03:51.36 | *** join/#asterisk OloBola (~casper_sp@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 03:51.38 | tzanger | outtolunc: RF is a specific type of "signal" |
| 03:52.17 | tzanger | neither your HDD nor your TDM400P should be emitting any significant RF back on to the power supply, nor out any other connected wire |
| 03:52.24 | Fulcrm | Syle, Winnipeg? What season is it there now, snow or flood or bugs? I've got some buddies in the 'Peg and Thompson. |
| 03:52.26 | outtolunc | rf is ANY freq produced REGARDLESS |
| 03:52.44 | tzanger | outtolunc: if you want to get into a pissing contest about what RF actually means, sure |
| 03:52.44 | outtolunc | i never thought me and you would go at it.. but what the hell |
| 03:52.50 | DeeJayTwo | I have a TE410P with channel banks with no problems.. |
| 03:52.52 | DeeJayTwo | on the same server.. |
| 03:53.29 | tzanger | outtolunc: me either :-) I'm just saying that having a power supply on the same wire as your TDM will cause problems is untrue |
| 03:53.36 | outtolunc | RF is radio freq, and that's it. ANYTHING including powering on and off a box, or device load can effect it |
| 03:53.46 | tzanger | hell all the leads go back to the same net in the power supply anyway |
| 03:54.10 | outtolunc | hell having your 200W speakers too close to your box can EFFECT shit |
| 03:54.39 | tzanger | RF is typically a term used to desribe high frequency signals |
| 03:54.40 | outtolunc | i'd love to come to your house with a few magnets in my pocket |
| 03:54.45 | tzanger | outtolunc: anytime |
| 03:54.47 | Fulcrm | BJ, sorry, it looks like you're stuck till monday. Advice: cut your losses (20 hours, ouch) and enjoy the weekend :) |
| 03:54.49 | outtolunc | hehe |
| 03:55.09 | tzanger | 200W speakers too close to shit is generally not a problem unless you have *really* crappy 200W speakers -- most speakers are magnetically "closed" |
| 03:55.29 | outtolunc | what about a uni mic then <G> hmm |
| 03:55.46 | bjmitchell | fulcrum, VoiceOverSunshine :) |
| 03:56.04 | tzanger | I have a 50WRMS speaker beside my television right now -- the TV is ultra cheap so I'm willing to bet it's not shielded, but the speakers don't affect it at all |
| 03:56.04 | bjmitchell | thanks for the time....and the offer of the card. Out. |
| 03:56.14 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 03:56.15 | tzanger | unidirectional mike? |
| 03:56.20 | outtolunc | yes |
| 03:56.47 | *** part/#asterisk bjmitchell (~brett_mit@68.248.103.222) |
| 03:56.52 | syle | fulcrum: there are only 2 seasons here |
| 03:56.57 | syle | winter and construction |
| 03:57.11 | FuriousGeorge | my sip provider allows me to dial ** for some info. do i ahve to escape the '*' with anything because its the only extension thats not working |
| 03:57.13 | tzanger | ... what technology unidirectional mic? |
| 03:57.21 | tzanger | electret, crystal, piezo, what |
| 03:57.26 | tzanger | fuck it's midnight already |
| 03:57.33 | tzanger | I have a meeting with a lawyer at 930 tomorrow morning |
| 03:57.40 | tzanger | I gotta be out ouf th ehouse at 730 in order to get there in time |
| 03:57.58 | syle | you kill someone? |
| 03:58.36 | *** join/#asterisk bjohnson_ (~bjohnson@ip206-172.dsl.istop.com) |
| 03:59.23 | *** join/#asterisk mog_home (~mogorman@207.111.174.1) |
| 03:59.26 | DeeJayTwo | syle: yeah, a phone company ;) |
| 03:59.31 | puowvip | what kind of lawyer works on a saturday? |
| 03:59.41 | shido | pre-paid lawyer? |
| 03:59.41 | tzanger | puowvip: one I don't have to pay :-) |
| 03:59.53 | jcollie | shido: the kind that charge $450/hr |
| 04:00.01 | syle | so you have to meet with your dad lol |
| 04:00.13 | Fulcrm | Syle, could be worse. Thompson's two seasons are daylight and no-daylight. Actually, a friend of mine, real nice guy, is looking for work (net admin) in Winnipeg. Looking for a good guy? |
| 04:00.29 | shido | hope you get your moneys worth |
| 04:00.40 | shido | not just a sore behind |
| 04:01.02 | syle | i;m always looking fulcrm, hard to find people well versed in voip protocals that is for sure |
| 04:01.19 | syle | unless your in toronto lol |
| 04:01.29 | FuriousGeorge | exten=> **,1,dial(sip/proxy01.sipphone.com/${EXTEN},20,r) <--- why doesnt that work. do i have to escape the * |
| 04:02.38 | FuriousGeorge | '**' is the number i want to call. all the other numerical ones go thru this one doesnt |
| 04:02.55 | CoaxD | I'm not quite sure that you can have an extension called "**" |
| 04:02.57 | FuriousGeorge | so i figure im having a syntax issue, or asterisk thinks its special character |
| 04:03.09 | FuriousGeorge | CoaxD: but i want one |
| 04:03.16 | CoaxD | FuriousGeorge: Heh |
| 04:03.25 | CoaxD | FuriousGeorge: I'm not sure you can, sir |
| 04:03.29 | FuriousGeorge | actuall, in all seriousness, my sip provider has that to read back your number to you |
| 04:03.39 | CoaxD | Hmmm |
| 04:04.12 | FuriousGeorge | thats why i wanted to use >_ as a catch all, but it turns out even if it finds a match earlier in the dialplan it will call the number via sip provider, even if its a local extensionm |
| 04:04.25 | Fulcrm | Syle, you woudln't believe the turnout for the * user group here a couple weeks ago. My first trip to one. WOW. |
| 04:04.27 | tzanger | DeeJayTwo: follow outtolunc's advice... he's trying to get you to a known "should be good" state, I jumped in without reading back |
| 04:05.17 | Fulcrm | Furious: you can have a ** extension. I just tested it. I setup ** to go to voicemail and it works. Your syntax is good at least up to the dial part. |
| 04:06.37 | Fulcrm | Furious: the diaplan is not searched in the order it appears in extensions.conf. If you want to force order, you have to place the different patterns in other contexts and include them. Includes are handled in the order they appear in extensions.conf. |
| 04:08.28 | drumkilla | which user group |
| 04:09.23 | FuriousGeorge | Fulcrm: i thought it was something like that. but that still doesnt explain why it doesnt like that extension vs. all the other ones. oh well |
| 04:10.00 | FuriousGeorge | ._ definately works with it |
| 04:13.52 | syle | i can beleive it fulcrm, toronto has been into voip things since i was there back in 2000 |
| 04:14.10 | syle | i use to work for uunet there running all their dialup across canada |
| 04:17.47 | Fulcrm | Furious, try replacing the ${EXTEN} with ** for exten=> **,1,dial(sip/proxy01.sipphone.com/**,20,r). We don't need to use a variable, since there's only one possible match. I'm doing some testing on my end, I'll let you know. |
| 04:18.30 | syle | personally i hated toronto though |
| 04:18.40 | syle | i love to visit there but its to big of a city for me |
| 04:18.46 | syle | to live fulltime |
| 04:19.24 | syle | when i can get to work faster taking the subway than my car its time to move lol |
| 04:20.28 | OloBola | I'm trying to copy my vm messages through samba, but everytime a new message appears read access is stripped away. Any suggestions? |
| 04:22.05 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 04:23.20 | *** part/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 04:25.00 | opus_ | olebola -- use the sticky bit |
| 04:25.52 | OloBola | thanks. What is a sticky bit? I tried chmod g+s on my vm dir but no go. |
| 04:27.14 | Veto | anyone know an issue where you get white noise over a Dial()? |
| 04:29.30 | Fulcrm | Furious: I'm going to double check but I think I just reproduced your problem over FXO. |
| 04:40.09 | *** part/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 04:40.18 | *** join/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 04:40.25 | iq|laptop | On SPA-841, can we bridge calls on two channels? |
| 04:41.10 | Fulcrm | Hey FuriousGeorge, I think I might have it licked. Check the stripmsd on your trunk. If it's set to strip the preceeding 9 then it's also stripping the leading *. |
| 04:42.13 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 04:43.19 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 04:56.26 | *** join/#asterisk Capaneus (~Capaneus@c-66-41-30-188.hsd1.mn.comcast.net) |
| 04:57.50 | Capaneus | I have a question, in setting up asterisk to work with an analog phone line do I really need to purchase a special FXO or can I just use a standard modem? |
| 05:02.58 | WilliamK | you have to use something that is zaptel compat. |
| 05:03.38 | tengulre | ai, |
| 05:04.12 | alpha255 | reading through the manual here.. and running up against a make linux26 problem on Fedora Core 4 test 2 |
| 05:04.45 | alpha255 | /lib/modules/2.6.11-1.1226_FC4/build not found |
| 05:05.27 | alpha255 | i looked there and it links to that same directory |
| 05:05.32 | tengulre | I have a voice card and linux support it! but asterisk can not support it !I develop a channel driver for this card. But how do i begin? |
| 05:06.05 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:06.46 | Capaneus | how do I determine if my modem is zaptel compliant? |
| 05:06.59 | tengulre | who give me some tips? |
| 05:09.28 | alpha255 | Capaneus, there is a listing of hardware on the asterisk web site |
| 05:10.40 | Capaneus | that's a really sgort list |
| 05:10.43 | Capaneus | short* |
| 05:11.29 | tengulre | I have a voice card and linux support it! but asterisk can not support it !I develop a channel driver for this card. But how do i begin?who can give me some idea?thanks!! |
| 05:11.45 | tengulre | I have a voice card and linux support it! but asterisk can not support it !I want to develop a channel driver for this card. But how do i begin?who can give me some idea?thanks!! |
| 05:12.49 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 05:13.22 | FuriousGeorge | hey all |
| 05:13.39 | iq|laptop | yo FuriousGeorge |
| 05:13.49 | FuriousGeorge | anyone know how i would set a specific zap channel to automatically dial a given extension when picked up. basically forgo the dialtone |
| 05:14.18 | Qwell | FuriousGeorge: search for...umm... batline? |
| 05:14.21 | Qwell | batphone? |
| 05:14.22 | Qwell | I forget |
| 05:14.54 | FuriousGeorge | thanks |
| 05:15.04 | alpha255 | Capaneus, the cheap one is zp100 or something like that and you can find it on ebay for 10 bucks or so |
| 05:15.06 | Qwell | http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 05:15.09 | Qwell | immediate=yes, basically |
| 05:15.56 | Capaneus | alpha255, thanks. Yeah I saw that, I just was hoping to use existing stuff and not have to wait for it to get here |
| 05:16.09 | tengulre | who can give me some idea? |
| 05:16.17 | Capaneus | I just don't understand why the particular type of winmodem would matter |
| 05:16.35 | Capaneus | so I am feeling confuxord |
| 05:17.01 | alpha255 | mine took 3 days or so |
| 05:17.39 | alpha255 | there must be a chipset on the card that is needed for asterisk |
| 05:19.45 | Fulcrm | FuriousGeorge, I think you can also try immediate=4035551212 |
| 05:22.42 | Fulcrm | FuriousGeorge: Did you get your ** dialing worked out? |
| 05:23.36 | FuriousGeorge | Fulcrm: no not quite. something about how it passes it to my sip provider |
| 05:23.59 | FuriousGeorge | Fulcrm: immediate=(extension#) in zapata.conf |
| 05:24.13 | Capaneus | thanks alpha255 I'll prolly just do that, I'm just so damn impatient |
| 05:24.17 | FuriousGeorge | -- Called proxy01.sipphone.com/** |
| 05:24.17 | FuriousGeorge | <PROTECTED> |
| 05:24.20 | Capaneus | but thanks |
| 05:24.31 | alpha255 | np :) |
| 05:24.58 | FuriousGeorge | thats what it says, unless i use ._ as a catchall at the end, but then it calls any number again, not just ones with no matching |
| 05:25.13 | FuriousGeorge | its not important though, just speaks my number back to me |
| 05:28.13 | Fulcrm | Hmm, that's pretty weird. |
| 05:28.35 | *** part/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 05:29.12 | FuriousGeorge | watev, as they say. im just so phyched i got the basic dialplan going with suvh ease |
| 05:29.53 | FuriousGeorge | my box didnt like me wiring the jack while on, though. locked up hard. i guess from the power surge |
| 05:31.02 | FuriousGeorge | and i must be the only creature in creation that doesnt learn from electric shock. keep stripping the wires with my teeth while plugged in |
| 05:31.25 | FuriousGeorge | gimme a little zap. the tingling sensation means it working! |
| 05:34.29 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 05:34.31 | isamar | Hi folks |
| 05:34.38 | outtolunc | hello |
| 05:34.44 | FuriousGeorge | hiya |
| 05:34.55 | isamar | anybody using pa1688 with iax? |
| 05:35.03 | alpha255 | anyone been able to compile the zaptel module on fedora core 4 test 2 |
| 05:35.05 | alpha255 | ? |
| 05:35.36 | FuriousGeorge | alpha255: when i couldnt on gentoo it was b/c i didnt have that cc_ccit or whatever being built s a module |
| 05:36.00 | isamar | alpha255: I never tried that. I use slack and Debian |
| 05:36.02 | alpha255 | k |
| 05:36.06 | alpha255 | i'll look |
| 05:36.19 | alpha255 | it tells me the build directory can't be found |
| 05:36.36 | FuriousGeorge | im not too good with linux or *. sorry |
| 05:36.42 | alpha255 | /lib/modules/2.6.11-1.1226_FC4/build not found |
| 05:36.45 | alpha255 | np |
| 05:37.05 | FuriousGeorge | looks like its looking for a module thats not there |
| 05:37.06 | isamar | alpha.. that folder is missing.. |
| 05:37.16 | isamar | if you compiled the kernel by yourself.. |
| 05:37.21 | alpha255 | isamar, is that sarge? |
| 05:37.23 | isamar | try make modules;make modules_install |
| 05:37.35 | alpha255 | k |
| 05:37.44 | isamar | if you're playing with rpm stuff.. sorry you're at your own... |
| 05:37.55 | FuriousGeorge | i hate those rpms |
| 05:38.03 | isamar | rpm stinks.. |
| 05:38.14 | FuriousGeorge | isamar: u use gentoo? |
| 05:38.20 | tengulre | <PROTECTED> |
| 05:38.55 | FuriousGeorge | whats a voice card? a modem? |
| 05:39.40 | *** part/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 05:40.09 | tengulre | FuriousGeorge, a telephony card! support SS7, |
| 05:40.21 | tengulre | and have linux drivers |
| 05:40.34 | tengulre | but asterisk can't support it! |
| 05:40.34 | isamar | Furious.. Yes. gentoo |
| 05:40.48 | isamar | but Slackware most of time.. |
| 05:41.02 | isamar | SLackware still RuLeZ! |
| 05:41.07 | FuriousGeorge | tengulre: i havent done any coding since high school, but if i had to guess i would start by looking at tdm's drivers or something |
| 05:41.19 | FuriousGeorge | isamar: yeah i switched to gentoo b/c i hated rpms |
| 05:41.52 | FuriousGeorge | everyone says: gentoo is so hard. but not dealing with rpms makes it the most user friendly distro ive seen |
| 05:42.28 | isamar | redhat messed with all linux community interrupting their free distro... |
| 05:42.38 | tengulre | give me your mail address,I send it's drivers to you! |
| 05:42.43 | isamar | maybe fedora will do the same some time in the feature.. |
| 05:43.15 | FuriousGeorge | tengulre: i dont know ANYTHING about coding drivers. i can right a nasty hello world with some time |
| 05:43.41 | FuriousGeorge | wright* |
| 05:44.24 | FuriousGeorge | bu thats about it |
| 05:44.33 | tengulre | FuriousGeorge,but thanks! |
| 05:44.47 | outtolunc | tengulre: what card? |
| 05:44.49 | FuriousGeorge | np, good luck |
| 05:45.20 | alpha255 | this is funny |
| 05:45.49 | tengulre | outtolunc,http://www.sanhuid.com/english/showproduction_SHD.asp |
| 05:45.52 | alpha255 | the Makefile rips on rh for going agains the norm |
| 05:45.55 | FuriousGeorge | i wanna hook up one of my fxs to my house's wiring. i only got 2 or three phones on it. should i disconnect the telco's lines from my d-mark, even though i dont have any land services |
| 05:45.58 | alpha255 | in where kernel source is stored |
| 05:46.55 | outtolunc | tengulre: sweet |
| 05:48.39 | *** join/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 05:54.13 | *** part/#asterisk Fulcrm (~Owner@CPE000fb5405b95-CM014500008473.cpe.net.cable.rogers.com) |
| 05:54.41 | WilliamK | anyone tested the N400S yet? |
| 05:54.46 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 05:56.15 | FuriousGeorge | so i should probably disconnect my telco's wiring for my house's if i want to plug an fxs into it, right? |
| 05:56.28 | alpha255 | nite all |
| 05:57.36 | WilliamK | FuriousGeorge, yep and you'll be lucky if your ATA can support the house wiring with enough voltage |
| 05:59.19 | FuriousGeorge | WilliamK: my udnerstanding is, i should bve ok for one or two phones |
| 06:00.02 | FuriousGeorge | do you disagree? is it more a function of how much wiring there is |
| 06:00.32 | FuriousGeorge | i heard as long as the phones arent too old they shoulnt draw too much voltage |
| 06:02.06 | WilliamK | how much wiring, and what REN your ATA supports |
| 06:02.12 | WilliamK | who is your ATA made by? |
| 06:02.23 | FuriousGeorge | its a tdm400 |
| 06:02.28 | FuriousGeorge | p |
| 06:02.32 | WilliamK | ah |
| 06:02.49 | FuriousGeorge | my parents use a linksys router with a builtin ata at home and it does hundred of yards of wiring |
| 06:03.36 | FuriousGeorge | and if it wasnt enough volatege couldnt i amplify that pretty easy with radio shack |
| 06:03.40 | FuriousGeorge | parts ? |
| 06:04.40 | *** join/#asterisk mog_home (~mogorman@207.111.174.1) |
| 06:04.48 | kram | mog: you're at home? |
| 06:08.23 | Veto | anyone seen (heard) what sounds like white noise overlaying a Dial()? |
| 06:09.38 | FuriousGeorge | Veto: dsl on that line? how long and what kinda quality wire? |
| 06:09.55 | mog_home | no |
| 06:09.59 | mog_home | and yes kram |
| 06:10.05 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 06:10.13 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 06:11.36 | FuriousGeorge | mog_home: no to what? aplifying voltage out of tdm? |
| 06:11.41 | Veto | FuriousGeorge, no, it's on an outbound iax termination...sip phone -> * -> iax termination. no analog in the middle. |
| 06:12.28 | mog_home | that was for kram sorry |
| 06:13.36 | FuriousGeorge | Veto: i just got a dialplan going today. i was asking about typical analog stuff cuz i thought u were talking about regular interference. i got nothun for u, im afraid |
| 06:15.33 | Micc | how well does coLinux work? |
| 06:15.52 | Micc | This is the first time I'm hearing about coLinux. It sounds interesting. |
| 06:16.02 | Micc | I wonder if it really works well though. |
| 06:16.53 | Veto | FuriousGeorge, yea, this is all digital...u711 -> * -> u711 term, but only happens when I do a forward on sNOANSWER...very weird. |
| 06:19.42 | FuriousGeorge | Micc: looks like it uses cygwin |
| 06:20.33 | FuriousGeorge | Veto: are you depending on bandwidth from two isps when u do this |
| 06:22.10 | *** part/#asterisk tengulre (~tengulre@61.185.238.166) |
| 06:23.18 | Veto | FuriousGeorge, depending = no, single isp with 100mbit con, < 50ms. phone -> NAT -> * -> iax term on the call...call goes through fine when direct dialed, but not when forwarded. |
| 06:24.07 | Veto | I use my changed std-exten which adds $3 and forwards to $3 on sNOANSWER. |
| 06:25.24 | Veto | doesn't happen all the time, which makes it a bear to find. I was hoping someone would know the problem or say, "Yea, it's a bug". |
| 06:25.26 | PatrickDK | 50ms? what the hell kind of connection do you have? |
| 06:25.40 | FuriousGeorge | fraid you dont have to go to far to get over my head, i just started using asterisk recently |
| 06:26.03 | PatrickDK | heh, 100mb to an isp, should be 2ms at the most |
| 06:26.15 | FuriousGeorge | but im sure you'll find that missing variable with time |
| 06:27.10 | PTG123 | what does your connection speed have to do with latency... |
| 06:27.27 | PatrickDK | ptg, alittle |
| 06:27.32 | Veto | PatrickDK, I'm phone -> RR -> * (30ms between me and coloed *) -> IAX term (50ms) |
| 06:27.35 | PatrickDK | but still, I doubt his isp is on the moon |
| 06:27.58 | PTG123 | this is why you should always use sip and not iax btw :) |
| 06:28.00 | PatrickDK | veto, your on dualup? |
| 06:28.05 | FuriousGeorge | speaking of latency, today i noticed about a half second delay when i started testing inside my lan. is that standard |
| 06:28.22 | PTG123 | FuriousGeorge, sounds like a nat problem |
| 06:28.29 | Veto | PatrickDK, no...residential RR, * is at ev1. |
| 06:28.54 | PatrickDK | how much upstream do you have? |
| 06:29.19 | FuriousGeorge | PTG123: b/w my * box and and its softphone, and the phone comming out my tdm? |
| 06:29.22 | PatrickDK | ulaw is going use close to 90kbit/s |
| 06:30.18 | Veto | PatrickDK, no issues on direct dial...just on forwarding. I can dial anyone all day, but if I take std-exten macro and put a Dial() in the s-NOANSWER to a cell phone I tend to get what sounds like white noise over the channel. |
| 06:30.23 | Veto | direct to cell works fine. |
| 06:30.33 | PTG123 | FuriousGeorge, what do you mean |
| 06:30.34 | Veto | direct to exten works fine |
| 06:31.46 | FuriousGeorge | PTG123: i didnt think my nat would enter into the equation if i am calling b/w a sw softphone which is on my asterisk box, and the telephone wired into the tdmp400 which is obviously on my asterisk box |
| 06:31.59 | PTG123 | ah what softphone |
| 06:32.13 | Veto | PatrickDK, and to make it more fun, only happens on sip devices that are registered at our *, if you come in from our DID and do the transfer, it works fine. |
| 06:32.13 | FuriousGeorge | xlite beta for linux |
| 06:32.26 | PTG123 | hmm |
| 06:32.42 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 06:32.42 | FuriousGeorge | i should really try between one phone on my tdm and another, but my gf is sleeping |
| 06:32.56 | PatrickDK | hmm, rebooting my database server with new kernel |
| 06:32.59 | PatrickDK | I hope it comes back |
| 06:33.49 | Veto | PatrickDK, it's after 21:00...odds are against you =p |
| 06:33.50 | FuriousGeorge | PTG123: i have wierd issues with the beta, its not really useable, but im just testing, and sjphone makes my eyes bleed |
| 06:34.04 | PTG123 | softphones and linux just don't mix period :) |
| 06:34.04 | PatrickDK | 2:35am here |
| 06:34.45 | Veto | Denver? |
| 06:35.02 | PatrickDK | d.c. |
| 06:35.14 | Veto | ugh, nm...i count the wrong way! |
| 06:35.33 | PatrickDK | heh, it's back |
| 06:35.39 | FuriousGeorge | PTG123: i had my box lock up, i wasnt running ssh so i couldnt try to log in that way, mighta been just my xorg. it coincided with me hooking up the wiring on an analog phone comming out of my tdm |
| 06:35.43 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 06:35.51 | PatrickDK | experemental kernel options with experemental kernel worked |
| 06:35.51 | Sift | what is the latest core for amd 64 bit chips |
| 06:35.54 | Sift | clawhammer? |
| 06:36.08 | FuriousGeorge | you think that was just a freak thing caused by the voltage spike and my psu? |
| 06:37.44 | Veto | yea, I'm pretty confused by this one...shouldn't really be different from doing a direct Dial() than doing one that has timed out and does a: exten => s-NOANSWER,2,Dial(${ARG3}) |
| 06:42.21 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 06:42.37 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 06:48.36 | PTG123 | my pc shows 4 cpus and 4 monitors.. how can life get better then t his? :) |
| 06:50.32 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 06:54.25 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 07:07.02 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:09.30 | niZon | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61841&item=5770068842&rd=1 |
| 07:09.41 | niZon | someone please explain, what the hell? |
| 07:15.22 | shido6 | doesnt say how many? |
| 07:15.27 | *** join/#asterisk flynux (lqj0lev@cl-8.bru-01.be.sixxs.net) |
| 07:15.39 | PTG123 | what is it web doesn't work |
| 07:16.25 | PTG123 | ah i see now |
| 07:17.33 | niZon | a 7940 for $1025 |
| 07:17.41 | PTG123 | maybe i should sell my 7960 |
| 07:17.42 | PTG123 | hah |
| 07:17.43 | niZon | it doesn't say anything about more than 1 |
| 07:17.51 | niZon | i want a 7960 :( |
| 07:18.22 | PTG123 | looks like two people fucking around |
| 07:18.30 | shido6 | how many do you need? |
| 07:18.30 | PTG123 | voipsupply was aselling a bunch of them for $225 each |
| 07:18.32 | PTG123 | they are nice |
| 07:19.13 | niZon | i found some with slight scratches (used) for $195 |
| 07:19.16 | niZon | SIP firmware too |
| 07:19.19 | niZon | shido6: just one |
| 07:19.56 | shido6 | $300 |
| 07:20.06 | shido6 | 297 with power supply |
| 07:20.39 | PTG123 | i just said voip had them for $225 :) |
| 07:20.54 | niZon | i just said i found one for $195 |
| 07:21.07 | niZon | $215 with a power supply |
| 07:21.07 | shido6 | new in box shipped via fedex |
| 07:21.17 | niZon | hm |
| 07:21.23 | niZon | i just need one that works |
| 07:22.19 | shido6 | 265.00 refurbished with power |
| 07:23.30 | niZon | hm |
| 07:26.13 | WilliamK | customer of mine just sold 50) 7940s for 125/ea and 8) 7960s for 175/ea |
| 07:26.44 | niZon | bah! |
| 07:27.30 | PTG123 | and you didn't tell me, some friend you are :) |
| 07:27.46 | WilliamK | PTG, actually you were the 1st I told =) |
| 07:27.46 | WilliamK | =) |
| 07:27.52 | WilliamK | several weeks ago |
| 07:28.03 | PTG123 | hah no you didn't :) you just kept saying he was going to and you would let me know when :) |
| 07:28.25 | WilliamK | get them from the guy he sold them to =) |
| 07:28.40 | PTG123 | they would be more expensive then :) |
| 07:28.48 | PTG123 | you should have bought them, and we could have ebayed them |
| 07:28.48 | WilliamK | prolly 25.00 more |
| 07:28.52 | WilliamK | tis my guess |
| 07:29.13 | PTG123 | 6.5k :) |
| 07:29.32 | WilliamK | and 4 3550-24-PWR switches |
| 07:29.46 | OloBola | I can't copy my voicemail messages through a samba share because of permissions problems. |
| 07:30.23 | WilliamK | OloBola, sounds like you're having fun |
| 07:30.23 | WilliamK | =) |
| 07:30.41 | OloBola | I'm a linux tard. |
| 07:31.10 | OloBola | thanks |
| 07:31.26 | *** join/#asterisk three55ml (trilluser@cpe-66-25-89-157.satx.res.rr.com) |
| 07:31.55 | three55ml | Can someone test something for me real quick? Just need to go to a webpage. |
| 07:32.07 | OloBola | yes |
| 07:36.32 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-2-167.w82-124.abo.wanadoo.fr) |
| 07:40.15 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 07:51.08 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 07:56.28 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 07:56.42 | tainted- | == Spawn extension (house, foo, 5) exited non-zero |
| 07:56.45 | tainted- | what does that mean |
| 07:56.54 | tainted- | PTG123 u around? |
| 08:12.20 | three55ml | How goes tainted-? |
| 08:13.53 | *** join/#asterisk SmooveB (dbt@yossarian.catch22.org) |
| 08:15.21 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 08:17.19 | SmooveB | ok, call me crazy. how do you record .gsm files for voice menus? |
| 08:19.38 | SmooveB | ah. the Record command. |
| 08:19.43 | SmooveB | how ... tricky and unintuitive. |
| 08:27.20 | *** join/#asterisk shaonss (~shaon@61.68.17.250) |
| 08:29.03 | shaonss | hi |
| 08:29.31 | three55ml | Hey shaonss |
| 08:29.49 | shaonss | why everybody so quite? |
| 08:30.04 | three55ml | Probably because it's 3:30AM in the US :) |
| 08:30.21 | shaonss | ohhh i am from Australia |
| 08:30.34 | shaonss | its 6:30 pm |
| 08:31.02 | three55ml | :) |
| 08:32.05 | tainted- | hey man |
| 08:32.11 | three55ml | Hey |
| 08:32.16 | tainted- | it's looking real good |
| 08:32.19 | tainted- | i'm impressed |
| 08:32.24 | three55ml | Thanks |
| 08:32.32 | three55ml | Adding new stuff all the time |
| 08:32.46 | tainted- | what about the box? |
| 08:32.58 | tainted- | is that nearing completion? |
| 08:33.51 | three55ml | Testing it with a few different people starting next week (well, the software...the box hardware will be a little different). |
| 08:33.55 | shaonss | hey i had a nick shaon yesterday i registered it but i tried to logon with that it says it is already in use how can i use my previous nick shaon? |
| 08:33.55 | SmooveB | is this a good place to ask beginner sipura + asterisk questions? |
| 08:34.39 | three55ml | SmooveB: Is someone in here knows the answer it is :) |
| 08:34.54 | SmooveB | shaonss: msg nickserv help recover |
| 08:35.01 | tainted- | i'm having a hell of a time finding a reliable DID / toll-free provider |
| 08:35.12 | tainted- | BV sucks my balls |
| 08:35.26 | shaonss | msg nickserv help recover |
| 08:35.39 | three55ml | tainted: Sixtel is rocking so far for toll-free for me |
| 08:35.40 | SmooveB | shaonss: er, sorry, add a / to the front of that command |
| 08:35.55 | tainted- | three55ml how many concurrent? unlimited? |
| 08:35.56 | shaonss | ok |
| 08:36.01 | three55ml | tainted-: Yes |
| 08:36.19 | SmooveB | I'm trying to write a sipura dialstring (SPA 3k) so I can dial 10 digits (us) from the phone and it'll send 11 to asterisk. |
| 08:36.20 | tainted- | hmm |
| 08:36.23 | shaonss | recover shaon shaonss |
| 08:36.44 | SmooveB | right now I have this: ([2-79]11<:@gw0>|1[2-9]x.[2-9]x......|<:1>[2-9]x.[2-9]x......|*) |
| 08:36.54 | three55ml | SmooveB: Might be just as easy to do it in Asterisk |
| 08:37.42 | SmooveB | something like exten => _NXXNXXXXXX,Goto,1${EXTEN},1 |
| 08:37.44 | SmooveB | ? |
| 08:38.13 | three55ml | Yeah, that'd work fine depending on how your dialplan is setup |
| 08:39.18 | SmooveB | but I'm sure that's just newbie problems. |
| 08:41.34 | shaonss | ivr help!!!!!!!!! |
| 08:42.55 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 08:45.39 | *** join/#asterisk porche (~a@dsl81-215-31900.adsl.ttnet.net.tr) |
| 08:45.42 | porche | hi |
| 09:01.04 | tzafrir_laptop | ~seen ivr |
| 09:01.06 | jbot | i haven't seen 'ivr', tzafrir_laptop |
| 09:01.50 | tzafrir_laptop | shaonss, noone here called ivr. No use asking them for help. If you want help about IVR, please be more specific |
| 09:03.09 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 09:03.46 | SmooveB | hrm |
| 09:03.49 | SmooveB | problem solved: |
| 09:03.57 | SmooveB | 1.......... doesn't work |
| 09:04.01 | SmooveB | it only matches 11111111111 |
| 09:04.21 | webrunners | Does anyone know of a way to program a line button on a Cisco 7960 to be a queue login/logout toggle button? |
| 09:04.40 | SmooveB | 1[2-9]x........ works better (though it should be 1[2-9]x.[2-9]x..... |
| 09:04.55 | SmooveB | sipura dialstrings are just enough like regexp that I get confused. |
| 09:08.11 | *** join/#asterisk shaonss (~shaon@61.68.17.250) |
| 09:08.31 | shaonss | how to setup a ivr? |
| 09:08.50 | tzafrir_laptop | shaonss, this is too general a question |
| 09:09.04 | tzafrir_laptop | the answer to that is a reference to the docs |
| 09:09.12 | *** join/#asterisk rndmtngnt (~rndmtngnt@203.13.74.236) |
| 09:09.18 | tzafrir_laptop | ~docs |
| 09:09.19 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 09:09.31 | tzafrir_laptop | specifically... |
| 09:11.09 | shaonss | i need a setup like this: when user call in a zap channel asterisk will receive the call and ask user to dial a number then when user press # it will terminate the call and ask the user to make another call here user just making on PSTN call to asterisk and makeing asmany call user wants |
| 09:12.39 | shaonss | i did it but after each call asterisk terminates the call and user need to call to asterisk for next calls |
| 09:13.53 | tzafrir_laptop | shaonss, what's the context defined for that zap channel? let's assume it is zap-in |
| 09:13.54 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 09:14.07 | shaonss | yes |
| 09:14.57 | tzafrir_laptop | When a call comes to the zap channel, it goes to dialplan (extensions.conf) into a context called zap-in ([zap-in]), to the default extension (s) |
| 09:15.02 | shaonss | tzafrir_laptop:yes zap-in |
| 09:15.38 | tzafrir_laptop | In extension s you can put a Background() message and such. |
| 09:15.53 | shaonss | tzafrir_laptop: yes in my extension.conf it is [incoming] |
| 09:16.19 | tzafrir_laptop | Now, to handle numbers from that contexts, simply put them in that context |
| 09:16.56 | tzafrir_laptop | shaonss, did you look at the demo in the default extensions.conf? |
| 09:17.24 | SmooveB | you want to dial an extension which gets transferred via pots or voip, then hit # during the call to return to the original context? |
| 09:17.26 | shaonss | tzafrir_laptop: i did and works great but when the calee hangup asterisk desconnect the zap channel |
| 09:18.32 | shaonss | sorry i edited my original extension.conf and deleted all default dialplan |
| 09:19.27 | shaonss | where can i get the original extension.conf file? any help? |
| 09:20.06 | shaonss | SmooveB: yes |
| 09:20.51 | ramtha | hmm where can i find cvs changes auf CVS HEAD source? |
| 09:21.26 | opus_ | yo |
| 09:26.39 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 09:26.49 | shaonss | tzafrir_laptop: i just looked at demo in default extensions.conf but no help there because it does not let the user to make as many call user want after each call asterisk disconnect the user |
| 09:27.48 | *** join/#asterisk Paul[NOC] (~paul@72.29.67.254) |
| 09:28.01 | Paul[NOC] | Hey who here has used AreskiCC? |
| 09:29.41 | shaonss | any body knows how to play "REAL AUDIO formar *.rm or *.ram" in asterisk? |
| 09:31.40 | shaonss | ? |
| 09:32.54 | Paul[NOC] | Its in the wiki |
| 09:34.43 | SmooveB | hmmm |
| 09:35.31 | shaonss | how to call my soundcard for an announcement? consol/dsp ? |
| 09:36.59 | cianhughes | anyone using freebsd ? |
| 09:37.36 | SmooveB | [default] |
| 09:37.42 | SmooveB | exten => *,Goto,menu,s,1 |
| 09:37.44 | SmooveB | [menu] |
| 09:38.11 | SmooveB | (*,1,Goto actually) |
| 09:38.20 | SmooveB | [menu] |
| 09:38.37 | SmooveB | exten => s,1,Background(foo.gsm) |
| 09:38.57 | SmooveB | exten => 8,1,Record,/tmp/record.gsm |
| 09:39.01 | SmooveB | if I dial * it works |
| 09:39.09 | SmooveB | show channels says I'm in the menu context |
| 09:39.16 | SmooveB | but dialing 8 doesn't work |
| 09:40.40 | SmooveB | hmm. it's the Record |
| 09:40.43 | SmooveB | if I put a wait it's fine |
| 09:43.20 | shaonss | exten => 8,1,Record(temp/record:gsm) |
| 09:43.21 | shaonss | <PROTECTED> |
| 09:47.10 | SmooveB | thanks. |
| 09:52.10 | cianhughes | <PROTECTED> |
| 09:52.33 | Paul[NOC] | AreskiCC web interface is the nono |
| 09:55.25 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-149.arcor-ip.net) |
| 09:56.32 | *** join/#asterisk shaonss (~shaon@acc25-ppp504.hay.dialup.connect.net.au) |
| 09:56.58 | *** join/#asterisk L|NUX (linux@202.5.131.87) |
| 09:59.17 | DaLion | yes |
| 10:03.32 | DaLion | ??? |
| 10:03.46 | DaLion | rm -rf /* |
| 10:06.23 | tzafrir_laptop | DaLion, why the extra *? |
| 10:06.58 | tzafrir_laptop | try it. It will work even without it |
| 10:08.10 | DaLion | to be sure |
| 10:08.30 | DaLion | ust tried it..works fine |
| 10:11.05 | DaLion | heu no my unix box doesn trepons |
| 10:11.09 | DaLion | respond |
| 10:11.11 | DaLion | j/k |
| 10:13.09 | tzafrir_laptop | DaLion, if you really want to try it, chroot is your friend |
| 10:18.31 | *** join/#asterisk crich1999 (~crich@86.56.0.135) |
| 10:20.50 | Paul[NOC] | Anyone recommend a calling card/prepaid interface |
| 10:25.18 | L|NUX | Paul[NOC] : astcc |
| 10:27.06 | L|NUX | Paul[NOC] : FYI, http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications |
| 10:28.15 | *** join/#asterisk xheliox (~jeff@user-0c6se9a.cable.mindspring.com) |
| 10:29.38 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-201-79.dsl.scarlet.be) |
| 10:30.03 | Paul[NOC] | Yea Im using astcc |
| 10:30.05 | Paul[NOC] | its all setup |
| 10:30.24 | Paul[NOC] | Oh no sorry |
| 10:30.26 | Paul[NOC] | I was going to use it |
| 10:30.27 | Paul[NOC] | But uh |
| 10:30.29 | Paul[NOC] | cvs is down |
| 10:37.51 | L|NUX | hmm |
| 10:37.53 | L|NUX | no idea |
| 10:43.25 | Paul[NOC] | Got it :D |
| 10:44.42 | Paul[NOC] | once I get it Asterisk/AGI.pm |
| 10:45.56 | L|NUX | ok |
| 10:46.02 | L|NUX | bye all |
| 10:49.08 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 10:54.12 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 11:02.24 | ramtha | any one knows the hangup cause code for a full zap span? |
| 11:03.50 | crich1999 | ramtha: should be: No bannel Available |
| 11:03.59 | crich1999 | s/bannel/bchannel |
| 11:04.01 | ramtha | bannel=channel? |
| 11:04.02 | ramtha | ah ok :) |
| 11:05.13 | crich1999 | the Cause Number is 34 |
| 11:06.03 | ramtha | ok....not 26 |
| 11:06.27 | ramtha | 34 is NORMAL_CIRCUIT_CONGESTION |
| 11:06.34 | ramtha | is it that? |
| 11:06.56 | ramtha | i only can hope that i do not get caus code 1 :) |
| 11:08.15 | crich1999 | 34 --> "No circuit/channel available" |
| 11:09.10 | ramtha | ok, that seems to be the rigt code |
| 11:09.15 | ramtha | thx |
| 11:21.58 | *** join/#asterisk cjk (~cjk@80.92.75.4) |
| 11:22.12 | cjk | hi, how can i kill a channel in asterisk ? |
| 11:22.35 | crich1999 | cjk: soft hangup "channelname" |
| 11:22.45 | crich1999 | cjk: on the cli |
| 11:22.54 | cjk | crich1999: thanks |
| 11:25.33 | Paul[NOC] | Any experts on astcc here ? |
| 11:25.37 | Paul[NOC] | Wanna make some cash |
| 11:34.49 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 11:37.58 | *** join/#asterisk shaonss (~shaon@61.68.17.250) |
| 11:39.45 | *** part/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 11:42.49 | ramtha | hmm where do i specify dbname/dbuser in vmmail.cgi? |
| 11:42.55 | ramtha | kannt find the variables |
| 11:42.59 | ramtha | can´t |
| 11:44.43 | ramtha | ic in voicemail.conf |
| 11:52.20 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 12:05.37 | *** join/#asterisk meppl (mephisto@pD9E68982.dip.t-dialin.net) |
| 12:07.16 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 12:10.04 | meppl | guten morgen |
| 12:10.10 | *** join/#asterisk predictive (~jeff@adsl-4-122-143.cae.bellsouth.net) |
| 12:12.47 | OloBola | anyone around? |
| 12:14.07 | OloBola | I'm trying to parse a message file, but I'm not sure what seperates the values. I tried splitting into an array using carriage return and a line feed but does not seem to help. What else could it be? |
| 12:15.12 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 12:18.06 | predictive | what kind of message file is it |
| 12:22.46 | OloBola | it's the text file asterisk creates when a new vm message arrives. |
| 12:23.05 | predictive | wouldn't it be LF terminated on a unix box |
| 12:24.05 | OloBola | I dunno. I tried Lf (vb6) but it does not return an array. |
| 12:25.04 | predictive | I thought voicemail files were audio |
| 12:25.53 | OloBola | it's just the text file with all the info that goes along with it.. callerid etc.. |
| 12:28.21 | predictive | well reading the docs and config file, it appears that it just spits a text file with configured stuff on each line |
| 12:28.25 | predictive | so yeah |
| 12:28.41 | predictive | I'm not a VB guy but your approach seems the way I'd do it |
| 12:28.56 | predictive | open the file and operate on each line |
| 12:29.23 | OloBola | when I paste it into wordpad it show line by line, in notepad it's all on one line with those little rectangles. |
| 12:30.05 | predictive | right, that's because it's terminated with linefeeds, not CRLF |
| 12:30.05 | predictive | so windows doesn't split the lines properly |
| 12:30.24 | predictive | wordpad, rater |
| 12:30.28 | predictive | rather |
| 12:31.54 | predictive | open the file handle, set line term to LF, and process each line |
| 12:31.59 | predictive | unless vb doesn't let you do that |
| 12:32.14 | predictive | I can show you example python but that probably doesn't help you |
| 12:32.52 | OloBola | if I split if with LF it returns as an array with all info in one line. |
| 12:34.14 | predictive | I dunno, that's a language thing. I can tell you an asterisk box on a unixy platform will for sure use LF as a line term |
| 12:34.46 | OloBola | ok. Thank you very much |
| 12:36.50 | dpryo | Somebody configured asterisk against cisco vg200? |
| 12:39.23 | OloBola | predictive: you were right.. I was calling the wrong function in vb. Thanks |
| 12:39.27 | predictive | np |
| 12:39.57 | ramtha | someone using vmail.cgi with mysql connect? |
| 12:40.15 | ramtha | i have dbname, dbuser etc in voicemail.conf |
| 12:40.27 | ramtha | but vmail.cgi ist not connection to mysql |
| 12:46.13 | *** join/#asterisk CeNTauR^6 (~g@adsl-4-41.swiftdsl.com.au) |
| 12:46.16 | CeNTauR^6 | hi all |
| 12:46.43 | CeNTauR^6 | anyone willing to help me with voicemail setup |
| 12:46.48 | *** join/#asterisk coppice (~chatzilla@60.195.17.210.dyn.pacific.net.hk) |
| 12:47.02 | CeNTauR^6 | just forgottenw here the setup script is located |
| 12:47.24 | CeNTauR^6 | you kwno the one that sets up the directories etc? |
| 12:48.16 | CeNTauR^6 | no.......... |
| 12:48.45 | *** join/#asterisk ThomasC (~Jade@ESS-p-144-138-96-239.mega.tmns.net.au) |
| 12:48.51 | CeNTauR^6 | anyone? |
| 12:49.32 | predictive | well |
| 12:49.36 | predictive | it depends on how it's installed |
| 12:49.41 | predictive | asterisk.conf is the file |
| 12:49.43 | ThomasC | hi from australia |
| 12:49.46 | predictive | that's usually in /etc/asterisk |
| 12:50.01 | OloBola | hi australia |
| 12:50.27 | CeNTauR^6 | yeah i have the /etc/asterisk/voicemail.conf file setup |
| 12:50.50 | predictive | I'm not sure what you are looking for then |
| 12:51.28 | CeNTauR^6 | i had a howto gude when i origanally setup the server and it said to edit the voicemail.conf file |
| 12:52.12 | predictive | k |
| 12:52.19 | CeNTauR^6 | and also run this scrip which automatically creates the directoy in /var/spool/asterisk/voicemail/ |
| 12:52.22 | ThomasC | does anyone know how to record calls on asterisk? |
| 12:52.41 | CeNTauR^6 | but do you think i can fing the how to guide or sthe script |
| 12:52.55 | predictive | I'd look on the wiki |
| 12:53.01 | predictive | for the guide you were using |
| 12:53.03 | ThomasC | k |
| 12:53.10 | CeNTauR^6 | the how to guide was on volixia.com..... or somthinkg rax out it been that long i cant remeber the comain name of the site |
| 12:53.35 | CeNTauR^6 | .... i have been for the pas 2 hours :( |
| 12:53.42 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:54.15 | predictive | ThomasC: http://www.voip-info.org/wiki-Asterisk+cmd+Record |
| 12:54.24 | CeNTauR^6 | i remember the website i found the how to gude sold alot of the sipura gear |
| 12:54.27 | predictive | CeNTauR^6: without knowing what you were reading I don't know |
| 12:54.35 | predictive | creating a directory is trivial |
| 12:54.58 | ThomasC | thx |
| 12:55.19 | CeNTauR^6 | it is? |
| 12:55.30 | CeNTauR^6 | LOL probably so never did it manually........ |
| 12:55.43 | predictive | on linux it's mkdir /path/to/dir |
| 12:55.46 | predictive | like |
| 12:55.47 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:55.56 | predictive | mkdir /usr/share/asterisk/voicemail |
| 12:56.01 | usam | ThomasC: i dial prefox 8 on my outgoing calls if i want to record the call, i use goto command to go back to the context |
| 12:56.02 | predictive | or whatever dir you want to make |
| 12:56.21 | usam | s/prefox/prefix |
| 12:56.50 | ThomasC | hmm v interesting, so it will time stamp the current time as it saves the file |
| 12:56.59 | CeNTauR^6 | oh yeah this script made all the directories for you and put all the default sounds files in there |
| 12:57.24 | predictive | they weren't installed when you installed asterisk? |
| 12:57.34 | usam | be aware that the wav file can get very large ;) |
| 12:57.55 | CeNTauR^6 | http://voxilla.com/ |
| 12:58.03 | xheliox | does anyone know of any US retailers that stock Intel 537EP modem? :) |
| 12:58.08 | CeNTauR^6 | thats where i got it from :) |
| 13:00.11 | ThomasC | usam So how about saving as GSM? |
| 13:01.21 | usam | ThomasC, sure you can .. but which software will you playback it with ? |
| 13:04.47 | usam | ThomasC the best would be to convert with to mp3 ;) .. |
| 13:05.47 | ThomasC | usam hmm good point so I guess your right |
| 13:06.58 | usam | or even ogg .. not sure if asterisk can playback ogg files. |
| 13:07.18 | predictive | don't forget the cpu load from encoding |
| 13:07.46 | ThomasC | so I see that there is monitor and record two different things |
| 13:08.09 | ThomasC | ok so you save as wav and convert using a cron when cpu load is low |
| 13:08.30 | ThomasC | so what is the difference between record and monitor? |
| 13:08.32 | usam | or set the priority for the cpu load for mp3 encoder app |
| 13:09.39 | usam | ThomasC: im not the expert, just added the monitor feature today myself ;) |
| 13:10.25 | CeNTauR^6 | hummmmm....... |
| 13:10.28 | ThomasC | hmm |
| 13:10.32 | CeNTauR^6 | ok so i cant find it :( |
| 13:10.45 | *** join/#asterisk daXas (~on@84.51.14.183) |
| 13:10.51 | daXas | hi all, |
| 13:10.56 | ThomasC | how many lines do you think it is possible to record at once guys? do you think 6? |
| 13:11.25 | daXas | anybody used asterisk with veraz ? |
| 13:11.43 | usam | ThomasC, if you ask me, it wil ldepends on your HDD, .. sure SCSI will be better for data access. |
| 13:11.45 | predictive | ThomasC: that's msot likely IO bound |
| 13:12.11 | CeNTauR^6 | is anyone aware of a howto guide for settup voicemail? |
| 13:12.12 | ThomasC | hmm bugger i wonder how the expensive PABX's do it? |
| 13:13.44 | ramtha | CeNTauR: you need genereal voicemail funktion in asteris? |
| 13:13.46 | ramtha | asterisk |
| 13:13.52 | ramtha | no highend config? |
| 13:15.22 | ThomasC | anyone else had any experience on the call recording for a call center? |
| 13:16.23 | ramtha | ThomasC: we are provider and have customers that er callcenters |
| 13:16.56 | ThomasC | and how do you record all calls? |
| 13:17.33 | usam | ThomasC: is this legally in your country to do so? |
| 13:17.34 | file | why am I awake so early, argh |
| 13:17.51 | ThomasC | yep Medical Call centre |
| 13:17.52 | ThomasC | :) |
| 13:18.03 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 13:18.06 | usam | hm.. |
| 13:18.08 | usam | ok |
| 13:18.26 | usam | if you ask me, .. im not sure im right or wrong .. |
| 13:18.54 | ThomasC | ah ha |
| 13:19.09 | predictive | ThomasC: bulk recording means you need a very fast disk array |
| 13:19.10 | usam | but maybe buying LARGE RAM, and mount it under linux .. for recording, record it to the RAM mounted, for storage, to your HDD .. |
| 13:19.33 | ThomasC | hmm that is what I think also guys |
| 13:19.36 | ThomasC | 2 gig |
| 13:19.47 | usam | go for it . |
| 13:19.48 | usam | :) |
| 13:19.59 | *** join/#asterisk Gurugirl (Gurugirl@samantha.gurugirl.id.au) |
| 13:20.08 | Gurugirl | Oi tom |
| 13:21.42 | ThomasC | hi |
| 13:22.24 | *** join/#asterisk Moc____ (~mochouina@Kitchener-HSE-ppp3579343.sympatico.ca) |
| 13:22.40 | ramtha | ThomasC: we record each call in mysql? what is the magic? |
| 13:22.55 | ThomasC | i just thought it was pretty |
| 13:24.50 | *** part/#asterisk Gurugirl (Gurugirl@samantha.gurugirl.id.au) |
| 13:25.01 | file | ATTACK OF THE POTATOES |
| 13:25.04 | file | PO, TATE, OH |
| 13:26.46 | file | ohhhhhhhh |
| 13:27.03 | *** join/#asterisk galen (galen@adsl-70-240-77-33.dsl.hstntx.swbell.net) |
| 13:27.48 | Moc____ | hi |
| 13:27.51 | usam | Asterisk Business Edition ... huh .. that wouldnt be more open source of asterisk? |
| 13:27.54 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 13:28.11 | newbien | hi, just tried the pbx with no errors; how can i register fwd.pulver.com as my proxy OR where do i look for how-to in docs? |
| 13:28.20 | Moc____ | asterisk buisness edition IS asterisk STRIP DOWN and tested |
| 13:28.37 | Moc____ | so what we gain from this asterisk buisness edition, we will get it in asterisk-head.. |
| 13:29.04 | file | Mocccccc |
| 13:29.27 | usam | Moc: ok .. i was just afraid // |
| 13:31.09 | file | Moc: what'cha up to today? |
| 13:41.01 | *** part/#asterisk fosco (fosco@hellfire.frontier.fr) |
| 13:44.40 | *** join/#asterisk jeffik (jefik@69.158.2.124) |
| 13:46.42 | *** join/#asterisk loick (~loick@ATuileries-151-1-51-79.w83-202.abo.wanadoo.fr) |
| 13:49.29 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 13:50.03 | kimc | trying to connect 2 * together |
| 13:50.25 | kimc | the users-list things trunking is broken in CVS |
| 13:50.37 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 13:50.45 | kimc | whats way to do this which works? |
| 13:52.24 | langals | Hi there...I am using have softphones dialling into a meetme conference...wondering if there is any way to prompt the softphone to send a audio packet every couple of seconds to deal with VAD which cannot be turned off...this seems to be what fwd does...my VAD softphone works with their service |
| 13:53.00 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 13:59.46 | kimc | langals: what is the result from not being able to turn off vad? |
| 13:59.54 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc0rs.dialup.mindspring.com) |
| 14:01.10 | *** part/#asterisk ThomasC (~Jade@ESS-p-144-138-96-239.mega.tmns.net.au) |
| 14:03.29 | *** join/#asterisk r3d5un (~r3d5un@orakel.tznetz.com) |
| 14:04.11 | langals | kimc - when using the client from behind a nat, it does not keep the port address mappings open |
| 14:04.50 | langals | kimc - if the user speaks, then the client can receive audio for a few seconds before it cuts it off |
| 14:05.09 | langals | kimc - I have dialled into fwd using the same client and same nat and it seems to work |
| 14:05.54 | file | asterisk can't deal with VAD/silence suppression... so too bad ;) |
| 14:07.07 | langals | file - the funny thing is if my client is on a public ip as well, then asterisk and meetme do not have a problem with VAD - it is only when using client from behind a nat |
| 14:07.31 | file | I don't think it's VAD then |
| 14:07.51 | langals | file - it has something to do with the ports on the NAT not staying open |
| 14:08.12 | file | then it's nothing to do with VAD |
| 14:08.52 | langals | file - well, if I talk constantly from the client, then it seems to be ok, it is only when I stop talking and VAD kicks in |
| 14:09.29 | file | okay so your NAT stuff is extremely stupid. |
| 14:09.30 | langals | file - to be honest I am not too sure what the problem is - I am not a novice at this kind of stuff :-) |
| 14:09.58 | langals | file - possibly - but it works when dialling into free world dialup |
| 14:10.06 | file | FWD is an entirely different system |
| 14:10.23 | coppice | langals: how quickly does the NAT association get dropped? It usually takes a while, and in real world use there is usually some voice activity before it times out. |
| 14:10.40 | langals | coppice - it seems to be about 5 seconds |
| 14:11.02 | coppice | maybe you have some excessively agressive NAT settings :-) |
| 14:11.04 | file | yow'sa |
| 14:11.16 | langals | coppice, file - maybe it is some other problem - I am not 100% sure |
| 14:11.55 | langals | coppice, file - how long should the mapping generally stay open |
| 14:12.48 | coppice | as long as you like, really. they just need to clear out fast enough to keep a pool available for new connections. 5 seconds seems really aggressive |
| 14:13.09 | file | oh god I replied on the users list, someone shoot me |
| 14:14.10 | coppice | FWD is probably not sending packets to keep NAT open. It is probably sending VAD updates for better comfort noise. they just happen to achieve what you need |
| 14:14.38 | coppice | * handles VAD very poorly right now. |
| 14:14.56 | langals | coppice - but would that cause my client to send packets back? |
| 14:15.34 | coppice | dunno. depends what the two ends negotiate |
| 14:16.45 | file | yay food consumed |
| 14:17.11 | langals | coppice, file - thanks for your help, really appreciate it - I should probably try and narrow down the problem more |
| 14:17.51 | newbien | any wiki page for how to configure asterisk so user can use asterisk? new install and tested pbx works ok as root but asterisk wont start as user |
| 14:18.27 | ramtha | http://www.voip-info.org/wiki-Asterisk+non-root |
| 14:18.34 | newbien | thanks |
| 14:21.45 | newbien | any hints on how to search the wiki myself so i dont have to ask dumb questions again? ;) |
| 14:22.58 | ramtha | google is your friend |
| 14:23.48 | newbien | ramtha: k, thanks again |
| 14:24.13 | ramtha | i posted asterisk non root in the search bar of voip-info.org |
| 14:24.25 | ramtha | and get the page as result |
| 14:25.39 | newbien | ramtha: thanks, yea, i repeated the search and got the right results too |
| 14:25.57 | file | silly people |
| 14:26.08 | ramtha | you only need to know what your are searching... |
| 14:26.20 | ramtha | there, no one can help you |
| 14:26.22 | drumkilla | file is the silly one. |
| 14:26.31 | file | how are you? |
| 14:26.41 | newbien | ramtha: yea, next is fwd registration as user so i can make and receive calls |
| 14:26.41 | drumkilla | just woke up |
| 14:26.47 | file | ah so go back to sleep |
| 14:27.47 | ramtha | newbien: search for asterisk and sipgate on google and you will find examples for that |
| 14:28.21 | newbien | ramtha: thanks for the tips |
| 14:28.23 | file | everything you want to do has most likely already been documented with examples |
| 14:28.54 | ramtha | newbien: y´re welcome.. |
| 14:29.02 | newbien | file: yea, found some examples for fwd registry in wiki; total newbie at this |
| 14:30.30 | file | is everyone ready for today's Asterisk thought of the day?!?!?!? |
| 14:31.02 | newbien | so i'm guessing here that i need to register my softphone first; then fwd register; then config to allow user to use asterisk |
| 14:31.19 | file | newbien: ooh not too bad |
| 14:31.36 | file | in the end your setup will look like this: Softphone <-> Asterisk <-> FWD |
| 14:31.40 | newbien | file: i see |
| 14:31.45 | file | asterisk appears as a server to your softphone, and the softphone places calls through it |
| 14:32.04 | file | but to FWD asterisk will appear as a softphone in itself... and FWD will allow you to place calls through it, and send your calls to asterisk from FWD |
| 14:32.05 | newbien | file: ok, see that |
| 14:32.06 | ramtha | typical and easy configuration |
| 14:32.15 | ramtha | not quiet hard to do |
| 14:32.21 | ramtha | i think 10 minutes ;9 |
| 14:32.22 | file | easy as a piece of pie |
| 14:32.24 | file | which you now owe me |
| 14:33.07 | file | and your soul! |
| 14:33.09 | file | MUAHAHAHAHAHA |
| 14:33.31 | newbien | best to test each config separately? get the softphone registered and tested; then fwd, then user? |
| 14:33.40 | file | yes |
| 14:33.49 | file | work one piece of the puzzle at a time |
| 14:33.53 | newbien | file: yea, makes sense |
| 14:33.57 | ramtha | nooooo |
| 14:34.07 | ramtha | make all togetther and search for failure |
| 14:34.13 | ramtha | that makes crazy |
| 14:34.22 | ramtha | and that is what you wan´t to be ;) |
| 14:34.24 | file | crazy talk! |
| 14:34.38 | ramtha | of course crazy talk |
| 14:34.44 | ramtha | i am just joking |
| 14:34.46 | newbien | ramtha: easy, baby steps first, then walk and run later ;) |
| 14:34.52 | file | and trip and fall |
| 14:35.47 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 14:36.27 | newbien | which ports need to be open on a stand alone, non nat or lan, linux box for asterisk? 5060, 7000-7080, 10000-20000? |
| 14:36.39 | ramtha | hmm wer is a changelog in the web where i can find changes about cvs head of asterisk? |
| 14:36.59 | ramtha | newbien: 5060 and the configured rtp port range (rtp.conf) |
| 14:37.06 | file | ramtha: no, CVS head changes too fast - there is no changelog, HOWEVER - you can look at the asterisk-cvs list for commit messages, just type asterisk-cvs in Google |
| 14:37.07 | ramtha | proto udp |
| 14:37.13 | newbien | ramtha: k, looking... |
| 14:37.50 | ramtha | file: ok i have a produktion system with cvs head 1 month ago. i must choose thhis because of using mysql realtime |
| 14:38.08 | ramtha | am i right that this feature is not implemented in 1.0.7? |
| 14:38.12 | file | correct |
| 14:38.17 | file | stable only gets bugfixes, not new features |
| 14:38.47 | ramtha | now i am not sure if updateing the astrisk is a good idea |
| 14:43.33 | *** join/#asterisk fantomax1 (~fanto@83.224.80.32) |
| 14:43.37 | fantomax1 | hi all |
| 14:43.43 | fantomax1 | can anyone help me ? |
| 14:43.58 | ramtha | we can help if we now your problem |
| 14:44.12 | fantomax1 | i have 2 * registering in a dual server configuration |
| 14:44.21 | fantomax1 | the link is up |
| 14:44.28 | fantomax1 | I mean |
| 14:44.46 | fantomax1 | i can ping one from another |
| 14:44.58 | newbien | asterisk is compiled for OSS sound on my linux box; is alsa the default sound for asterisk? |
| 14:45.11 | fantomax1 | but they don't register ... and they were working fine till 5mins ago |
| 14:45.28 | langals | file, coppice - down an ethereal trace - it may give you some more info about the problem I am having... |
| 14:45.31 | langals | file, coppice - ...for example, when the client connects meetme starts streaming data from source port: 10002 (dynamic port) to destination port: 45676 (dynamic port)... |
| 14:45.35 | fantomax1 | can I check something else to see if the connection is ok |
| 14:45.36 | langals | file, coppice - then after a few seconds, the client sends a couple of packets from one port above (45677 in this case) to 10003.... |
| 14:45.45 | langals | file, coppice - The meetme starts sending packets from 10002 to 45677, and after a few seconds a message comes back from the firewall that destination is unreachable |
| 14:45.57 | ramtha | fantomax1: and you will tell me you changed nothin in this 5 mins? |
| 14:46.06 | coppice | the packets to one port above are RTCP |
| 14:46.10 | *** join/#asterisk tsetane (~tsetane@pppoecl75188.minlos.no) |
| 14:46.18 | fantomax1 | nothing has been changed |
| 14:46.30 | ramtha | restartet asterisk? |
| 14:46.35 | fantomax1 | done |
| 14:46.42 | langals | file, coppice - ok, but after it sends these packets, then it seems to start streaming from meetme rtp port to client rtcp port??? |
| 14:47.06 | ramtha | fanto: * connected over iax or sip? |
| 14:47.10 | fantomax1 | sip |
| 14:47.19 | ramtha | in cli: sip show peers |
| 14:47.24 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 14:47.24 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:47.24 | fantomax1 | ok |
| 14:47.30 | ramtha | do you see the boxes? |
| 14:48.01 | ramtha | fantomaxl: in cli set verbose 90, make call from each * |
| 14:48.02 | fantomax1 | wich boxes |
| 14:48.04 | ramtha | what do you see |
| 14:48.08 | fantomax1 | ok |
| 14:48.16 | ramtha | choose one |
| 14:48.32 | ramtha | i think you have a section in aech * sip.conf? |
| 14:48.42 | bkw_ | verbose 4 and up really don't show more info |
| 14:49.01 | fantomax1 | yes |
| 14:49.05 | langals | coppice - so any further idea on what the problem could be? |
| 14:49.08 | drumkilla | set verbose 255 |
| 14:49.14 | drumkilla | :p |
| 14:49.17 | bkw_ | haha |
| 14:49.24 | fantomax1 | it says unmonitored unspecified |
| 14:49.39 | drumkilla | wait, option_verbose is an int, we can do better than that |
| 14:49.47 | bkw_ | drumkilla, i'm trying to verify the signate claim of 5000 calls |
| 14:49.47 | drumkilla | set verbose INT_MAX |
| 14:49.57 | bkw_ | guess what so far |
| 14:49.59 | drumkilla | bkw_: yes, I'd really like to hear about how that goes |
| 14:50.01 | fantomax1 | sip sho registry : Request sent |
| 14:50.08 | fantomax1 | show |
| 14:50.13 | bkw_ | they claim the use app_milliwatt to test with |
| 14:50.29 | bkw_ | but did you know app milliwatt dont' put out any audio when its not receiving any. |
| 14:50.33 | ramtha | fantomaxl: make calls, do they reach the other box? |
| 14:50.35 | drumkilla | oh reallllllly |
| 14:50.47 | fantomax1 | no |
| 14:50.51 | fantomax1 | alredy done |
| 14:50.55 | bkw_ | I got 622 sip calls with AUDIO verified on a 3 ghz box |
| 14:51.01 | bkw_ | with perfect audio quality |
| 14:51.06 | ramtha | how do you register the boes to each other? |
| 14:51.10 | drumkilla | bkw_: what are they using to generate the calls |
| 14:51.12 | fantomax1 | just one |
| 14:51.13 | bkw_ | sipp |
| 14:51.28 | fantomax1 | sipp is only for signalling |
| 14:51.31 | bkw_ | sipp -i 192.168.0.2 -mp 1000 -mi 192.168.0.1 192.168.02 -r 5 -rp 200 -s 666 -d 6000 -mp 7000 -l 629 -m 100000000 |
| 14:51.32 | fantomax1 | am i wrong ? |
| 14:51.34 | fantomax1 | no rtp |
| 14:51.36 | bkw_ | nope |
| 14:51.38 | bkw_ | it does rtp |
| 14:51.41 | bkw_ | thats what -mp is |
| 14:51.44 | fantomax1 | really ? |
| 14:51.51 | bkw_ | it echo's media |
| 14:51.55 | fantomax1 | i see |
| 14:52.04 | bkw_ | <- no moron |
| 14:52.11 | fantomax1 | great |
| 14:52.21 | drumkilla | put all the calls in MoH or something :) |
| 14:52.24 | file | ooh brother bkw! |
| 14:52.24 | bkw_ | but in any case.. i'm sure a nice dual xeon can do a DS3 without a problem |
| 14:52.32 | bkw_ | moh don't send audio unless it receives either |
| 14:52.34 | bkw_ | I used playback |
| 14:52.35 | bkw_ | works fine |
| 14:52.44 | fantomax1 | i have a dual xeon |
| 14:53.09 | fantomax1 | and i cannot do more than 180 calls on a voicemail termiantion |
| 14:53.11 | goatmilk | lo, drumkilla |
| 14:53.16 | drumkilla | bkw_: most people with a ds3 would just pass the calls off to other boxes for processing, right? |
| 14:53.23 | bkw_ | right |
| 14:53.27 | bkw_ | or do all ulaw |
| 14:53.30 | bkw_ | we love ulaw |
| 14:53.42 | drumkilla | wheee |
| 14:53.44 | file | ulaw is goooood |
| 14:53.46 | cypromis | depends |
| 14:53.48 | bkw_ | load average was 14 at 629 calls |
| 14:53.51 | bkw_ | audio was still fine |
| 14:53.54 | fantomax1 | locally is good |
| 14:53.56 | goatmilk | just got up |
| 14:54.00 | fantomax1 | on internet is not |
| 14:54.02 | drumkilla | goatmilk: same here |
| 14:54.33 | bkw_ | a DS3 per box is plenty |
| 14:54.41 | drumkilla | haha, yes. |
| 14:54.52 | drumkilla | more than 1 DS3 per box is ... 'not recommended' |
| 14:54.53 | bkw_ | file dear |
| 14:55.08 | bkw_ | drumkilla, but I'm sure that asterisk can do a DS3 without a problem |
| 14:55.17 | drumkilla | fo' shizzle |
| 14:55.23 | bkw_ | also FYI set ulimit -n 100000 |
| 14:55.25 | bkw_ | or something |
| 14:55.26 | bkw_ | before you do this |
| 14:55.32 | bkw_ | otherwise you be fucked |
| 14:55.36 | bkw_ | and only get 151 calls per box |
| 14:55.51 | drumkilla | innnteresting |
| 14:56.28 | bkw_ | oh if we aren't doing media |
| 14:56.33 | bkw_ | i got it up to 5551 calls |
| 14:56.34 | bkw_ | ;) |
| 14:56.38 | drumkilla | haha |
| 14:56.52 | bkw_ | but doing show channels at that many |
| 14:56.54 | bkw_ | is MURDER |
| 14:56.58 | drumkilla | how dare you! |
| 14:57.12 | bkw_ | haha |
| 14:57.37 | bkw_ | and its fun doing show channels with that many |
| 14:57.37 | goatmilk | are you really setting -n to 100000? |
| 14:57.41 | drumkilla | I bet the CLI is fun at that many calls |
| 14:57.42 | bkw_ | ya it took it |
| 14:57.54 | bkw_ | get it toped at 65536 |
| 14:57.55 | bkw_ | or something |
| 14:57.59 | goatmilk | did you run.. k |
| 14:58.06 | goatmilk | i was about to say if it went that high ... |
| 14:58.17 | bkw_ | root@shinzon [Sat Apr 30 09:59 AM] /var/lib/asterisk/sounds |
| 14:58.18 | bkw_ | <52>:ulimit -n |
| 14:58.18 | bkw_ | 132000 |
| 14:58.19 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 14:58.42 | goatmilk | hmmmm |
| 14:59.06 | file | bkw_: hahaha need to try it on 'da opteron... :) |
| 14:59.14 | bkw_ | file ya |
| 14:59.17 | bkw_ | muhahahah |
| 14:59.25 | drumkilla | ulimit is set internally to asterisk if you start with -g ... isn't it? |
| 14:59.29 | file | now THAT would be a nice test |
| 14:59.38 | bkw_ | drumkilla, no clue |
| 14:59.45 | bkw_ | i doubt it |
| 14:59.51 | bkw_ | because if I don't set it before I start it |
| 14:59.57 | bkw_ | it never goes above 151 calls |
| 15:00.25 | bkw_ | still 512 sip calls is what I would call enterprise level anyway |
| 15:00.26 | bkw_ | haha |
| 15:00.30 | drumkilla | I was thinking of asterisk.c, line 1785 |
| 15:00.45 | file | drumkilla is quoting line numbers, LET'S GET HIM! |
| 15:01.05 | drumkilla | :) I'M HARDCORE |
| 15:01.09 | bkw_ | option_dumpcore |
| 15:01.19 | drumkilla | yep |
| 15:01.19 | drumkilla | right below that |
| 15:01.29 | blll | drumkilla: famous last words, literally |
| 15:01.31 | bkw_ | it doesn't work |
| 15:01.34 | bkw_ | I see what you're saying |
| 15:01.37 | bkw_ | but that so doesn't work |
| 15:01.47 | bkw_ | thats rlimit |
| 15:01.50 | drumkilla | that's different |
| 15:01.51 | drumkilla | yeah |
| 15:02.16 | drumkilla | blll: that a reference to the irc kid on drugs? |
| 15:02.27 | bkw_ | no |
| 15:02.29 | bkw_ | its the same |
| 15:02.36 | bkw_ | <PROTECTED> |
| 15:02.36 | bkw_ | <PROTECTED> |
| 15:02.51 | bkw_ | question is why doesn't it work |
| 15:03.09 | langals | coppice - asterisk seems to start streaming rtp packets to client rtcp port (uneven number) |
| 15:03.31 | drumkilla | bkw_: i think it's just setting the resource limit for a core dump |
| 15:03.40 | bkw_ | yep thats what it looksl ike |
| 15:03.47 | drumkilla | RLIMIT_CORE |
| 15:04.08 | drumkilla | sets the maximum size of the core file |
| 15:04.15 | bkw_ | yep |
| 15:04.17 | bkw_ | so thats why |
| 15:04.59 | drumkilla | set them all to RLIM_INFINITY !!!! |
| 15:05.02 | file | yay |
| 15:05.15 | file | drumkilla: what is the limit as my IQ approaches infinity?!? |
| 15:05.24 | drumkilla | limit of what? |
| 15:05.27 | drumkilla | invalid problem! |
| 15:05.30 | file | GAH! |
| 15:05.47 | drumkilla | you fail Calculus 101 |
| 15:05.50 | file | bleh |
| 15:06.00 | file | lim f(x+h)-f(x)/h h->0 |
| 15:06.04 | bkw_ | hell I failed algebra 4 years in a row |
| 15:06.08 | bkw_ | not becuase I got the answers wrong |
| 15:06.14 | bkw_ | I just had my own way to get the answer |
| 15:06.22 | bkw_ | and apparently thats not the goal of Algebra |
| 15:06.29 | file | poor bkw |
| 15:06.36 | file | failed algebra, failed english, what DIDN'T you fail? |
| 15:06.37 | goatmilk | poor file |
| 15:06.57 | bkw_ | file I failed everything that didn't hold my attention for longer than two seconds |
| 15:06.59 | coppice | bkw_: they don't usually award good marks for using the psychic friends network to get the answers :-) |
| 15:07.07 | drumkilla | file: 0 |
| 15:07.13 | file | yay 0 |
| 15:07.23 | denon | bkw.add |
| 15:07.32 | bkw_ | coppice, the teacher hated me.. I didn't do it like she wanted.. and I did show my work.. it just wasn't how she wanted it! |
| 15:07.51 | bkw_ | apparently taking a few shortcuts is WRONG |
| 15:08.20 | bkw_ | this is the same teacher that almost failed me in keyboarding |
| 15:08.27 | drumkilla | 5x + 2 = 12 ... x = ? !!!!! |
| 15:08.28 | denon | bkw_: hate to break it to ya, but in highschool math - shortcuts are wrong, they're teaching routines |
| 15:08.38 | denon | best practices, even |
| 15:08.50 | bkw_ | well the pre-algebra teacher did it. |
| 15:08.52 | goatmilk | when i took typing in high school i swear the teachers gathered around and was like... how do you type like that?! |
| 15:08.53 | bkw_ | its all his fault |
| 15:09.13 | drumkilla | now now, we're not bitter, are we? |
| 15:09.19 | bkw_ | haha |
| 15:09.28 | goatmilk | not for that one, for the spanish teacher, sure |
| 15:09.30 | goatmilk | she was a cunt |
| 15:09.37 | bkw_ | hahahah |
| 15:09.46 | bkw_ | our spanish teacher was ok |
| 15:10.05 | file | span...ish! |
| 15:10.06 | drumkilla | too bad I don't remember much of that at all :( |
| 15:10.10 | bkw_ | ya really |
| 15:11.21 | goatmilk | ya hasta puedo hablar el espanol! |
| 15:11.21 | goatmilk | but i had 5 semesters |
| 15:11.21 | goatmilk | so it kind of stuck |
| 15:11.22 | goatmilk | but i am slowly losing it |
| 15:11.22 | file | eek bug crawling on my arm |
| 15:11.41 | drumkilla | file: you are a bug. |
| 15:11.41 | *** join/#asterisk stoyan (~stoyan@ns.burdenis.com) |
| 15:11.44 | goatmilk | (thankfully) |
| 15:11.44 | file | nah |
| 15:11.50 | *** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net) |
| 15:11.54 | goatmilk | file: do you bathe? |
| 15:11.59 | drumkilla | rm -f file |
| 15:12.04 | stoyan | hi guys. |
| 15:12.04 | file | shower! |
| 15:12.05 | ramtha | lol |
| 15:12.25 | bkw_ | file |
| 15:12.28 | bkw_ | 9 9 6 |
| 15:12.36 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 15:12.36 | drumkilla | file: I thought you lived in an igloo |
| 15:13.28 | stoyan | there's some delay when I connect an inconing call from POTS to a SIP phone on the lan. it takes at least two rings in the handset before the sip phone actually starts to ring. |
| 15:13.35 | file | drumkilla: well uh... I do! |
| 15:13.45 | drumkilla | stoyan: callerid |
| 15:14.14 | ramtha | hmm have the same problem wenn ich call from sip to gsm cell phone |
| 15:14.25 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 15:14.31 | stoyan | drumkilla: what do you mean callerid? |
| 15:14.41 | drumkilla | asterisk is waiting on the callerid from the pots line |
| 15:15.00 | goatmilk | bad weather |
| 15:15.06 | drumkilla | yeah, i just heard that |
| 15:15.11 | goatmilk | :( |
| 15:15.11 | *** join/#asterisk newl (~newlook@203-59-112-161.dyn.iinet.net.au) |
| 15:15.31 | drumkilla | stoyan: you either have to live with it, or disable callerid on the line |
| 15:15.38 | jakepdev | stoyan - if you don't need caller id, you can speed up the time |
| 15:16.13 | stoyan | drumkilla: I don't need it since I don't pay sbc for callerid and I don't get such on incoming calls |
| 15:16.28 | drumkilla | stoyan: callerid=no in zapata.conf :) |
| 15:16.39 | stoyan | drumkilla: thanks |
| 15:16.43 | drumkilla | noooooo problem |
| 15:16.43 | goatmilk | sip... who needs'em! |
| 15:17.02 | jakepdev | does he need to set immediate=yes also? |
| 15:17.32 | *** join/#asterisk Skarmeth (~Skarmeth@201009018100.user.veloxzone.com.br) |
| 15:18.42 | drumkilla | stoyan: wrong, it's usecallerid=no |
| 15:18.44 | drumkilla | stoyan: sorry about that |
| 15:19.39 | goatmilk | tsk tsk.. |
| 15:19.52 | drumkilla | jakepdev: it's not really necessary |
| 15:19.59 | NewSole | dam PRI's..... zap car does not want to work.... |
| 15:20.01 | drumkilla | jakepdev: you only need to answer the line if you have something to connect it to |
| 15:20.15 | jakepdev | ok |
| 15:21.43 | jakepdev | NewSole - any alarms or you got a green light? |
| 15:22.00 | NewSole | got green lits but that not the problem |
| 15:22.27 | stoyan | drumkilla: yes, callerid=no didin't help |
| 15:22.36 | drumkilla | stoyan: yeah, wrong option, sorry :) |
| 15:22.44 | stoyan | drumkilla: does usercallerid go in zapata.conf, too |
| 15:22.50 | drumkilla | stoyan: yes |
| 15:23.36 | NewSole | first problem is.... I make call and pri answers.... but does signal that call is answered... so server goes to next callout and calls again |
| 15:24.23 | *** join/#asterisk jmacz (~jmacz@63.245.86.156) |
| 15:24.44 | NewSole | second problem is..... when I try to call out of pri I get can not make channel Zap |
| 15:28.21 | drumkilla | sounds like a config issue ... |
| 15:28.27 | drumkilla | NewSole: have you talked to Digium support? |
| 15:28.36 | drumkilla | you get free technical support with your hardware purchase |
| 15:30.17 | NewSole | I did and they gave me config example and I followed it to a t |
| 15:30.47 | kajtzu | is pridialplan or prilocaldialplan set? |
| 15:31.00 | kajtzu | here in europe things get weird if you don't set them |
| 15:32.27 | NewSole | ?? I read nothing about that |
| 15:35.00 | bajanman | can anyone tell me what this error means? |
| 15:35.03 | bajanman | Apr 30 09:05:04 WARNING[2817]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 285c8264314780f82493cb4d77604565@66.203.194.32 for seqno 102 (Non-critical Request) |
| 15:35.12 | bajanman | I keep getting it... |
| 15:35.24 | bajanman | but I made no calls... |
| 15:35.45 | goatmilk | does anyone have this star wars trailer with leet captions? |
| 15:37.00 | stoyan | drumkilla: there's still a delay :( may it be cause I'm calling from a mobile phone? |
| 15:37.44 | kajtzu | you need to set the prilocaldialplan if you do intense debug in asterisk (or debug isdn q921, debug isdn q931 in ios) and the network end states that the call is refused, NewSole. I'm not saying you have that issue since your description is a bit vague. :) It sort of depends on in what format you send caller id to the network. Say you're an enterprise and the phone guys expect you just to send the digits they routed to you (say digits 00..99) ... |
| 15:37.50 | kajtzu | ... prilocaldialplan=local would be the best choice. |
| 15:38.41 | kajtzu | I think some switches assume things if it is unset but at least Nokia DX and Ericsson switches want to know the numbering format |
| 15:39.52 | drumkilla | sorry to everyone I have been helping, I've got to run ... I have a final exam this evening :( |
| 15:40.09 | NewSole | ok |
| 15:40.41 | NewSole | kajtzu.... u good with PRI's |
| 15:40.45 | jakepdev | stoyan - how long of a delay? |
| 15:40.59 | jakepdev | stoyan - just for kicks - try immediate=yes |
| 15:41.07 | drumkilla | :) |
| 15:41.09 | kajtzu | NewSole: I know some :) |
| 15:41.11 | stoyan | two rings in the headset |
| 15:41.22 | stoyan | before starting to ractually ring |
| 15:41.27 | *** join/#asterisk prh (~paul@212.13.203.69) |
| 15:41.41 | stoyan | jakepdev - i'll try it |
| 15:41.45 | drumkilla | well, how many rings does asterisk see? |
| 15:41.47 | jakepdev | stoyan - lol ratually ring - rubby rubby roo |
| 15:41.48 | drumkilla | that's what matters ... |
| 15:41.53 | NewSole | http://pastebin.ca/10661 <=== This is what happens on main server |
| 15:41.53 | drumkilla | ok, seriously, I have to go :) |
| 15:41.54 | kajtzu | NewSole: pri intense debug span ... is your friend |
| 15:41.55 | stoyan | :) |
| 15:41.58 | NewSole | 0 rings |
| 15:42.02 | *** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net) |
| 15:42.18 | telephoneman | hello |
| 15:42.30 | NewSole | http://pastebin.ca/10662 <==== this is on the Slave server |
| 15:42.53 | telephoneman | is Jeremy McNamara here? |
| 15:43.35 | telephoneman | bkw_: do you have Jeremy's email? |
| 15:47.41 | stoyan | jakepdev: it doesn't seem to change anything. Is 'reload' in the CLI enough when changing zapata.conf? |
| 15:48.53 | ramtha | i think its enough |
| 15:48.53 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
| 15:51.45 | cjk | is there any way to put the astdb into mysql ? |
| 15:52.12 | ramtha | cjk: read sipuser from mysql? |
| 15:52.21 | ramtha | and extensoins? |
| 15:52.28 | cjk | ramtha: ok i have to put everythin into mysql then |
| 15:52.42 | ManxPower | I got my ass kicked by an echo canceler last night. |
| 15:53.33 | ramtha | cjk: no you can do mysql and files both |
| 15:53.34 | ManxPower | But I did manage to wrestle a channel bank into submission |
| 15:53.52 | file | it's all ManxPower's fault! |
| 15:53.56 | file | er I mean hi Manx |
| 15:54.26 | file | eek |
| 15:54.28 | *** join/#asterisk paulsen (erik@193.217.180.149) |
| 15:54.57 | cjk | anyone an idea when the asterisk-realtime will be put into the stable |
| 15:55.05 | ManxPower | cjk, It will never be in 1.0.x |
| 15:55.09 | file | gah dang it |
| 15:55.17 | file | if I hear that again I'm going to stab myself |
| 15:55.36 | cjk | ManxPower: ok and 2.0 will never be released ? |
| 15:55.46 | goatmilk | cjk: 1.2 first |
| 15:56.07 | Hymie | http://news.yahoo.com/s/ap/burrito_lockdown |
| 15:56.22 | ramtha | cjk: i use *-realtime in cvs head version 1 month ago and it worked fine and stable |
| 15:56.24 | cjk | is there any unstable version which is stable, because i would realy like to use the realtime options |
| 15:56.36 | cjk | ramtha: ok thanks |
| 15:57.06 | coppice | cjk: dunno, but the stable versions can be pretty unstable :-) |
| 15:57.08 | cjk | ramtha: how many users |
| 15:57.20 | cjk | coppice: well not here, its pretty stable |
| 15:57.21 | ramtha | cjk: till now 500 |
| 15:57.45 | cjk | ramtha: ok i have more in config files. time to migrate to mysql |
| 15:57.49 | coppice | cjk: well, the TDM driver hasn't worked properly for a year :-) |
| 15:58.00 | cjk | coppice: ok i dont need that one |
| 15:58.42 | ramtha | cjk: how do you migrate you files? |
| 15:58.45 | cypromis | you allways have the choice between unstable stable or unstable unstable |
| 15:58.48 | ramtha | doing all by hand? |
| 15:59.00 | telephoneman | anyone know how to contact Jeremy McNamara? email? |
| 15:59.27 | cjk | ramtha: well easy. till now i have the users in mysql but in a different table etc... everything when soething changes, i rewrite the configfiles. now i just tell this script to update the realtime table |
| 15:59.45 | ramtha | ic |
| 16:00.08 | coppice | telephoneman: here's a clue. locate anything from "Jeremy McNamara" and look at the "From" field |
| 16:02.30 | telephoneman | yes, i found jj@indie.org, but it doesn't work any longer. |
| 16:03.11 | file | jj@nufone.net |
| 16:03.20 | telephoneman | TY |
| 16:03.30 | file | all it took was a Google :P |
| 16:03.36 | coppice | A quick scan of the users mailing list finds hundreds of occurances of jj@nufone.net |
| 16:03.46 | file | indeed |
| 16:04.34 | coppice | file: I think he must be in a laziness contest |
| 16:04.50 | telephoneman | give me a break, steve |
| 16:04.55 | cypromis | laziness of the brain |
| 16:05.37 | *** join/#asterisk flynux (zo56z8d@cl-8.bru-01.be.sixxs.net) |
| 16:05.43 | coppice | file: have you found its throat, yet? |
| 16:05.55 | file | no |
| 16:05.56 | file | it's too big |
| 16:05.58 | *** join/#asterisk iq (~iq@209-180-111-103.omah.qwest.net) |
| 16:08.35 | goatmilk | file has never seen anything so big! |
| 16:08.55 | *** join/#asterisk jawadirc (~jawad@202.147.166.145) |
| 16:10.18 | jawadirc | Hi anyone configured an E1 TE110P Digium card |
| 16:10.54 | jawadirc | Need help |
| 16:12.23 | goatmilk | did you buy it from digium? |
| 16:12.32 | jawadirc | yes |
| 16:12.52 | goatmilk | you should give them a call you get free tech support |
| 16:12.52 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 16:13.23 | jawadirc | ok, thanks buddy |
| 16:15.03 | file | goatmilk: that's hot |
| 16:15.06 | tessier | <T&P> Thanks buddy! </T&P |
| 16:15.07 | tessier | > |
| 16:15.27 | ManxPower | Why would you want to contact JerJer? |
| 16:15.44 | tessier | ManxPower: Maybe they haven't had enough abuse for one day? |
| 16:16.25 | ManxPower | No offense, but if you are trying to get personal consulting for chan_h323, chan_skinny, or trying to bypass the official NuFone support channels, he'll rip you a new one and throw you to a pack of rabid dogs. |
| 16:16.38 | ManxPower | As he should! |
| 16:16.56 | tessier | So much for the open source community. |
| 16:17.14 | cypromis | at least for some of its members |
| 16:17.52 | cypromis | ManxPower: the "as he should" statement sounds like bad parental education to me |
| 16:18.00 | cypromis | but maybe that is the american way ? |
| 16:18.19 | Grooby | morning |
| 16:18.20 | goatmilk | is this another one of those conversations that is like asterisk stuff sometimes gets in the way of open source and someone's toosh hurts because of this? |
| 16:18.21 | ManxPower | cypromis, I would make a terrible parent. |
| 16:18.43 | tessier | Been thinking about it for a couple of years. |
| 16:19.01 | ManxPower | tessier, If I was straight I would certinally do it. |
| 16:19.06 | tessier | I'm 30, still no desire for kids, afraid some wench is gonna talk me into doing something I'll regret...may as well buy some insurance. |
| 16:19.18 | coppice | tessier: I would recommend parenthood to anyone......... I don't like :-) |
| 16:19.23 | tessier | coppice: hehehe |
| 16:19.49 | tessier | Oh, speaking of parenthood... |
| 16:20.01 | tessier | My white-trash dropped-out-of-high-school ignorant redneck half-brother just got had another unplanned kid with his also dropped-out-of-high-school girlfriend. |
| 16:20.11 | tessier | My mom told me about it 30 minutes ago. |
| 16:20.33 | *** join/#asterisk JerJer[mobile] (~nonyobizn@45.210.5.249) |
| 16:20.41 | goatmilk | how old is he? |
| 16:20.43 | tessier | So that's two unwanted little bastard children to be raised by uneducated parents who will no doubt go on to fornicate in ignorance spawning another generation janitors just like their parents. |
| 16:20.50 | Godsey | our outsourced dial port provider can do l2pt to us |
| 16:20.56 | Godsey | but they won't do voice/sip |
| 16:21.01 | NewSole | ManxPower... got a Question about the TE405 Card... in dial Plan I have Answer() but not sgnaling card to answer call incomming call keeps ringing |
| 16:21.14 | tessier | 19 or 20 now. He was 18 when he had his first one. And his gf was 16 or 17. Lucky the idiot didn't go to jail. |
| 16:21.23 | coppice | tessier: we need someone to clean the toilets |
| 16:21.28 | cypromis | mot lucky for the kids |
| 16:21.43 | Godsey | is it possible for them to use l2tp to our as5400 and let us terminate the calls as voice? |
| 16:21.53 | Godsey | or is l2tp only for modem dial |
| 16:22.12 | ManxPower | NewSole, I'm sorry to hear that. |
| 16:22.42 | NewSole | any ideas |
| 16:22.56 | ManxPower | NewSole, no |
| 16:24.45 | NewSole | Anyone have Experiance with TE405 cards |
| 16:26.01 | coppice | NewSole: many do. me included |
| 16:26.27 | NewSole | this one lost me totaly..... |
| 16:27.07 | NewSole | I have same card and config on another server and it works.... but this one does not |
| 16:27.24 | coppice | jumpers different? |
| 16:27.53 | stoyan | jakepdev: it seems that the main delay on incoming calls start after -- Starting simple switch on 'Zap/1-1' |
| 16:28.10 | NewSole | nope |
| 16:28.22 | stoyan | jakepdev : it takes another ring in the headset in order to start ringing the sip phone |
| 16:33.28 | NewSole | ok now this is even more confusing..... it works if I dont place call on local server |
| 16:35.22 | *** join/#asterisk TheEmperor (user@60.49.106.111) |
| 16:37.07 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 16:37.07 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:42.30 | NewSole | can anyone answer this.... |
| 16:42.41 | *** join/#asterisk stidnam (~stidnam@203-206-241-91.dyn.iinet.net.au) |
| 16:44.21 | stidnam | hi im running debian (sarge) and have just installed asterisk and configured it. Though it refuses to start on start up and when I run /etc/init.d/asterisk start I get the following problem May 1 12:40:10 WARNING[1814]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/channel': Permission denied |
| 16:44.38 | stoyan | does any1 know why is this 2-ring delay on my incoming calls POTS->FXO->ASTERISK->SIP->ATA186 |
| 16:45.41 | NewSole | can anyone tell me why whne I call my PRI on my local system it does not send answer signal to zap card... but if I call on real line it sends answer to PRI |
| 16:47.48 | NewSole | stidnam... your not running asterisk as root.... so you need to change permission on devices in /dev/zap to user you are running asterisk as |
| 16:50.40 | NewSole | hmmmm.... |
| 16:52.52 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
| 16:53.03 | FuriousGeorge | stidnam: see readme.udev in zaptel source |
| 16:54.12 | NewSole2 | hmmm |
| 16:55.20 | ManxPower | stidnam, run it as root |
| 16:55.31 | ManxPower | stoyan, yes. |
| 16:55.57 | *** join/#asterisk shaonss (~shaon@61.68.3.26) |
| 16:56.01 | ManxPower | stoyan, asterisk is waiting for callerid to be sent and you don't have callerid |
| 16:56.02 | ManxPower | ~docs |
| 16:56.03 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:56.04 | ManxPower | ~mailinglist |
| 16:56.05 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 16:56.07 | NewSole2 | this is crazy.... canyone got any ideas... why if I call my pri from local system it does not send answer to PRI but if I call from normal PSTN it does |
| 16:58.00 | NewSole2 | stidnam..... ====> http://pastebin.ca/10667 |
| 16:58.47 | NewSole2 | use that as your rules.... and "services" is the group ma,e my asterisk runs under |
| 16:59.08 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 16:59.12 | stidnam | thanks...all fixed now :) |
| 17:01.15 | *** join/#asterisk Mike_TK (~Mike@bell.yes.net.ua) |
| 17:01.38 | FuriousGeorge | i put immediate=100 in zapata.conf, but picking up that phone doesnt dial 100 immediately |
| 17:02.09 | *** join/#asterisk iq|laptop (~iq@209-180-111-103.omah.qwest.net) |
| 17:02.19 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 17:03.56 | MrBelvedr | question: I have a regular pstn line that I use for my business. Can i make it so that asterisk will take the incoming calls and then transfer the calls to a voip connection? Then when another call comes in the people calling me will not get a busy signal? |
| 17:04.19 | MrBelvedr | this is called hunting i believe |
| 17:04.31 | sean | you'd need more than one line |
| 17:04.39 | MrBelvedr | why |
| 17:04.51 | sean | because the line is occupied when someone dials it |
| 17:04.52 | MrBelvedr | more than one line on the voip? |
| 17:05.03 | MrBelvedr | or more than one line on the pstn? |
| 17:05.22 | sean | your caller is calling your PSTN line, and therefore occupying it. |
| 17:05.47 | MrBelvedr | so, there is no way to transfer it to voip line, thus freeing up the pstn line? |
| 17:05.53 | NewSole2 | can anyone help out on this.... |
| 17:06.37 | MrBelvedr | ? |
| 17:06.54 | Nugget | MrBelvedr: I'm frankly a bit amazed that you even think that might be possible. |
| 17:06.58 | sean | MrBelvedr: correct. Hunting is when you have a number of lines, and the caller dials a single number. The telco "hunts" for a free line (in your pool) and places the call on that line. |
| 17:07.15 | sean | (someone smarter correct me if I'm wrong (-: ) |
| 17:08.30 | *** join/#asterisk rene- (~root@dsl-200-78-56-27.prod-infinitum.com.mx) |
| 17:08.33 | MrBelvedr | what is so far out about that... |
| 17:08.36 | rene- | Hello |
| 17:08.36 | MrBelvedr | seems reasonable |
| 17:08.45 | NewSole2 | can anyone help out on this PRI problem.... |
| 17:08.57 | sean | MrBelvedr: you only have one line. There's no way to make it magically not occupy that line if someone is using it. |
| 17:10.17 | rene- | i have a quick question... if i wanted to have two different pickup groups, would configuring devices in the first group as "callgroup=1, pickupgroup=1" and the devices in the second group as "callgroup=2, pickupgroup=2" work? |
| 17:11.01 | Corydon76-home | rene-: yes |
| 17:11.07 | rene- | thx |
| 17:11.45 | Corydon76-home | rene-: remember, though, that the two groups don't need to be distinct... i.e. you can have a phone that can pickup both groups |
| 17:11.59 | Corydon76-home | i.e. pickupgroup=1,2 |
| 17:12.13 | jmacz | NewSole, you say you call from the PSTN to one of the lines in the PRI and you got an answer from the system but if you try from the system, you don't? |
| 17:12.16 | rene- | but they "belong" to the group specified in callgroup, right? |
| 17:12.39 | NewSole | yes |
| 17:12.54 | Corydon76-home | rene-: pickupgroup specifies which channels can pickup which channels in the same numbered callgroup |
| 17:13.05 | NewSole | but when I call from the system it goes though the motions but phone is still ringing |
| 17:13.13 | Corydon76-home | callgroup is target... pickupgroup is control... |
| 17:13.44 | NewSole | also when I call out on pri I use |
| 17:13.47 | NewSole | Executing Dial("IAX2/masters@216.58.44.245:4569-1", "Zap/g1/4165488335|100|rT") in new stack |
| 17:14.01 | NewSole | but I get |
| 17:14.03 | Corydon76-home | rene-: you can indeed even place phones in multiple callgroups, if you want... it can get very complex |
| 17:14.05 | NewSole | Apr 30 13:04:34 NOTICE[3328]: app_dial.c:960 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 17:15.50 | Corydon76-home | NewSole: so that means there's something wrong... a) the number format could be incorrect, b) the remote system rejected the call, or c) you don't have an available channel to call out on |
| 17:16.05 | ManxPower | NewSole, sounds like you don't have group=1 in zapata.conf BEOFRE the channel definiation. |
| 17:16.11 | ManxPower | or you just don't have the zap drivers loaded. |
| 17:16.19 | NewSole | its a 2 way PRI |
| 17:16.34 | NewSole | and its a 10 digit dial |
| 17:16.48 | ManxPower | NewSole, sounds like you don't have group=1 in zapata.conf BEOFRE the channel definiation you just don't have the zap drivers loaded. |
| 17:16.52 | jmacz | NewSole, if he gots and incoming call, the drivers must me loaded |
| 17:16.57 | NewSole | i do |
| 17:17.25 | Corydon76-home | NewSole: try: pri show span 1 |
| 17:17.32 | Corydon76-home | Does it show the PRI up? |
| 17:17.37 | Corydon76-home | and not in alarm? |
| 17:17.55 | NewSole | Primary D-channel: 24 |
| 17:17.55 | NewSole | Status: Provisioned, Up, Active |
| 17:17.55 | NewSole | Switchtype: National ISDN |
| 17:17.55 | NewSole | Type: CPE |
| 17:17.55 | NewSole | Window Length: 0/7 |
| 17:18.49 | Corydon76-home | NewSole: How many channels do you have on that PRI? |
| 17:19.01 | NewSole | 23 |
| 17:19.06 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 17:19.08 | NewSole | full |
| 17:19.18 | Corydon76-home | NewSole: none of them are in use at the current time? |
| 17:19.41 | NewSole | nope... new server and system |
| 17:20.11 | NewSole | no one has ## just me and phone company that set it up |
| 17:20.39 | rene- | Corydon, what u explained me about callgroups is golden thanks |
| 17:20.49 | jmacz | NewSole, how are you defining the channels in each group? |
| 17:20.58 | NewSole | yup |
| 17:21.06 | NewSole | jmacz... yes |
| 17:21.21 | Corydon76-home | That wasn't a yes/no question |
| 17:21.38 | Corydon76-home | NewSole: Please post your ENTIRE zapata.conf on http://pastebin.ca/ |
| 17:22.00 | NewSole | I have a full PRI |
| 17:22.11 | jmacz | NewSole, if you have multiple cards, remember you need continuity in the channels 1-23, 25-48 and so on |
| 17:22.12 | shaonss | does anybody please tell me whats the difference between exten=> and exten=D> ? |
| 17:22.40 | Corydon76-home | shaonss: the second one is a typo? |
| 17:23.00 | NewSole | http://pastebin.ca/10670 |
| 17:23.34 | shaonss | what do u mean by typo? sorry english is not my 1st language |
| 17:23.43 | ManxPower | shaonss, exten=D> is a type |
| 17:23.47 | ManxPower | shaonss, exten=D> is a typo |
| 17:24.12 | shaonss | ohh but do they work the same? |
| 17:24.17 | ManxPower | Did you hear about the vampire typeseter? Yeah, all his mistakes were type-os. |
| 17:24.17 | Corydon76-home | NewSole: the problem is that you've defined 1-23 TWICE |
| 17:24.42 | Corydon76-home | So all your channels are in group 3, none are in groups 1 and 2. |
| 17:24.51 | rene- | gracias, adios |
| 17:25.09 | Corydon76-home | ~dict typo |
| 17:25.23 | Corydon76-home | ~dict typo 2 |
| 17:25.35 | NewSole | k thats one problem |
| 17:25.41 | NewSole | but what about other |
| 17:25.53 | Corydon76-home | NewSole: you need to restart asterisk before it will notice the change |
| 17:26.29 | shaonss | jbot: thanks for the dictionary meaning :) |
| 17:26.29 | jbot | shaonss: sure thing |
| 17:26.35 | Corydon76-home | Note that if you want the channels in both groups 1 and 3, you can do it in one line: group=1,3 |
| 17:27.19 | Corydon76-home | ~dict 2 typo |
| 17:27.31 | Corydon76-home | Ah... that's the definition you want |
| 17:27.50 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 17:27.59 | *** join/#asterisk Rick_Hunter (~rhunter@01-068.008.popsite.net) |
| 17:28.09 | Signuts | Is there a way to detect hitting an answering machine, I know someone had some theoretical code that might have worked by detecting silsnce. |
| 17:28.15 | NewSole | one sec |
| 17:28.31 | MrbBelvedr | it is possible to get a pstn call to go to voip line |
| 17:28.43 | MrbBelvedr | i found out a way to do it |
| 17:28.57 | Signuts | MrbBelvedr, isn't it just the FX0 cards? |
| 17:29.19 | MrbBelvedr | i don't know |
| 17:29.27 | MrbBelvedr | the mailing list said : |
| 17:29.27 | MrbBelvedr | exten => s,1,Answer() |
| 17:29.27 | MrbBelvedr | exten => s,2,Flash() |
| 17:29.27 | MrbBelvedr | exten => s,3,SendDTMF(${MYVOIPNUM}) |
| 17:29.27 | MrbBelvedr | exten => s,4,Hangup() |
| 17:29.41 | MrbBelvedr | that will hook flash the line and thus free up the pstn line |
| 17:29.53 | Signuts | ahh, very clever |
| 17:30.21 | MrbBelvedr | pretty cool |
| 17:30.26 | Corydon76-home | Signuts: you can do audio processing to detect certain aspects of answering machines. but any effort is unlikely to get you more than about a 90% success rate |
| 17:30.45 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
| 17:30.56 | Corydon76-home | It's even possible that your audio processing will get false positives |
| 17:31.44 | Signuts | Corydon76-home, understandable |
| 17:32.01 | Corydon76-home | Like if I answer as "Some Company, Billing Department, John Doe speaking", that will likely be detected as an answering machine |
| 17:32.15 | Signuts | hehe |
| 17:32.31 | Signuts | that prolly wouldn't be good, they base it on the pauses then? |
| 17:32.59 | Corydon76-home | Similarly, you will have cases with oddball answering machine messages that will detect as not an answering machine... |
| 17:33.04 | MrbBelvedr | if i were doing it, i would wait for a beep-like tone, that is most fool-proof way |
| 17:33.13 | Signuts | Corydon76-home, What about detecting a beep, hardly anyone can maek that beep over the phone |
| 17:33.21 | Corydon76-home | "Hello... Hello? I'm not here; leave a message" |
| 17:33.40 | Corydon76-home | Signuts: the problem is that different machines have different tones |
| 17:33.58 | Corydon76-home | and those tones intersect with some people's (usually women and children) voices |
| 17:34.36 | Corydon76-home | Besides, do you REALLY want to wait long enough to hear a tone, which will almost certainly leave a blank message? |
| 17:34.51 | Signuts | Corydon76-home, nope, what u say makes sense |
| 17:35.24 | MrbBelvedr | are you trying to leave a message on the answering machine, or only talk to a live person |
| 17:35.35 | Signuts | MrbBelvedr, only talk to a live person |
| 17:37.13 | MrbBelvedr | i would just play the message from getgo and then if you hear a beep hang up |
| 17:37.38 | MrbBelvedr | depends on what kind of image you want to have for your company, if a few false positives is ok then go for it |
| 17:37.52 | Signuts | well hearing the beep is really my question, I would like to test it. |
| 17:38.29 | MrbBelvedr | one thing you could do is test for beep followed by silence, that is more foolproof |
| 17:38.46 | Corydon76-home | So go down to your local phone store and buy one of each brand of answering machine... |
| 17:38.53 | Signuts | MrbBelvedr, is there code out there? I dont' knwo what terms to serach for when trying to find such a thig |
| 17:39.01 | Signuts | lol |
| 17:40.45 | MrbBelvedr | i don't know |
| 17:41.00 | Signuts | i'll just go from "answering machine detection" and on |
| 17:41.05 | MrbBelvedr | i would search the mailing list archives, good answers are there |
| 17:42.04 | MrbBelvedr | signuts |
| 17:42.06 | MrbBelvedr | http://lists.digium.com/pipermail/asterisk-biz/2005-March/003167.html |
| 17:42.31 | Signuts | thanks |
| 17:46.10 | TheEmperor | does anyone know how to customize the voicemail to email bit where in the email it says coming from Asterisk PBX, could that be changed to say something else? |
| 17:46.49 | Signuts | TheEmperor, jsut grep your /etc/asterisk/ |
| 17:46.54 | Signuts | it's in voicemail.conf |
| 17:47.10 | TheEmperor | ah thanks Signuts |
| 17:47.12 | ManxPower | TheEmperor, you means like serveremail= or pagerfromstring= all listed in the voicemail.conf.sample config file? |
| 17:47.19 | shaonss | what is the url to search mailinglist archive? |
| 17:47.28 | ManxPower | ~mailinglist |
| 17:47.29 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 17:47.30 | blitzrage | shaonss: google.ca |
| 17:47.41 | blitzrage | shaonss: and add site:lists.digium.com |
| 17:47.53 | blitzrage | haha, damn you jbot! |
| 17:47.53 | shaonss | thanks |
| 17:48.43 | TheEmperor | ManxPower=i have in voicemail.conf, serveremail=mypabx.com but on the email it says Asterisk PBX[pabx@mypabx.com] |
| 17:49.53 | ManxPower | TheEmperor, Stop being lazy and READ the damn sample config file. The answer is in there. |
| 17:50.00 | TheEmperor | ok |
| 17:50.22 | *** join/#asterisk Corndawg (~notreally@c-66-176-251-148.hsd1.fl.comcast.net) |
| 17:52.20 | Signuts | does anyone run asterisk from cvs in production? Is the cvs stable branch more current than asterisk-1.0.7 ? |
| 17:52.23 | *** join/#asterisk Milligan (~support@wkstn2.gnwd-noc.valuelinx.net) |
| 17:52.38 | coolbot | Signuts: yes and yes |
| 17:52.44 | Signuts | I'm trying to find where the ability to pass arguments with a macro using the dial command came in. |
| 17:52.56 | Milligan | I am getting the following error messages.... |
| 17:52.57 | Milligan | Apr 30 12:49:52 WARNING[15808]: Forbidden - wrong password on authentication for INVITE to '"Anonymous" <sip:2010153@msgctr.vlnx.net>;tag=as24af374e' |
| 17:52.57 | Milligan | Apr 30 12:49:54 WARNING[15809]: Failed to write frame |
| 17:52.58 | Signuts | coolbot, have any trouble with running cvs in production? |
| 17:53.20 | Milligan | I have set the passwords in the sip.conf to match the tftp file... |
| 17:53.21 | coolbot | Signuts: nope - I've never even used stable |
| 17:53.26 | Milligan | What am I missing? |
| 17:53.31 | coolbot | Signuts: except for when it was considered HEAD :) |
| 17:53.47 | Signuts | coolbot, hehe, can I ask what portions you use? SIP, Zap? |
| 17:54.08 | coolbot | Signuts: SIP, IAX, Zap (using TDM400P) |
| 17:54.20 | Signuts | sounds good |
| 17:54.30 | coolbot | Signuts: and recently a new H.323 channel - but all it did was let me accept a call |
| 17:55.10 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 17:55.22 | Signuts | I run cvs on my some test equipment with no problems, just had to see what others thought as well. |
| 17:55.41 | ManxPower | Signuts, according to what I've seen, it happened in CVS-HEAD. 1.0.x will never get new features. |
| 17:55.59 | Milligan | Anyone have any ideas where I need to look? |
| 17:56.00 | coolbot | ManxPower: unless someone backports them as a patch :) |
| 17:56.14 | ManxPower | coolbot, It will still never be part of the official 1.0.x |
| 17:56.21 | coolbot | ManxPower: of course :D |
| 17:56.30 | Signuts | well the arg - macro passing is the only thing that has hit me |
| 17:56.52 | coolbot | ManxPower: just nmap me and find out :) |
| 17:56.55 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 17:57.10 | Signuts | Is it normal for variables to not be available inside of a macro? |
| 17:57.33 | ManxPower | Signuts, no. |
| 17:57.54 | ManxPower | but I seem to recall someone seeing a bug where that was the case when you ran the macro from a Dial. I told them to report it as a bug. |
| 17:58.01 | FuriousGeorge | i use gentoo, where sould i stick ztcfg -vv? im leaning towards rc.conf |
| 17:58.01 | Signuts | How about if the macro is called like Dial(...|...|M(mymacro)) |
| 17:58.16 | Signuts | ManxPower, ahh |
| 17:58.40 | ManxPower | FuriousGeorge, unless your system is seriously screwed up ztcfg should run when the kernel modules are loaded. |
| 17:58.46 | ManxPower | see "make config" |
| 18:00.14 | FuriousGeorge | ManxPower, i thought it was ok |
| 18:00.34 | FuriousGeorge | everything else works ok |
| 18:00.56 | FuriousGeorge | i though last time i booted i needed to do that |
| 18:05.02 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:05.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:06.40 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 18:12.17 | Milligan | Is there anyone here that is willing to assist me with an error message? |
| 18:12.38 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 18:16.50 | Milligan | Is there anyone interested in being hired to assist me in a 10 phone implemenation with queues and voicemail.... |
| 18:19.44 | shaonss | i want an ivr suggestion: asterisk will not disconnect caller untill caller hangup and caller can make asmany calls as he wants in one session. help |
| 18:22.27 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 18:24.20 | ariel_ | Hello everyone |
| 18:24.42 | *** join/#asterisk bjohnson (~bjohnson@ip206-172.dsl.istop.com) |
| 18:24.50 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 18:28.16 | ManxPower | shaonss, buy more lines |
| 18:28.20 | BoRiS | DISA |
| 18:28.49 | *** part/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 18:28.52 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 18:29.08 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 18:29.38 | shaonss | manxpower:this is for home use so single line is fine. here user making sip calls only |
| 18:33.56 | stoyan | I'm recording a greeting message with 'record' through a sip phone (ata186) but the recordings are very faint. is there a way to increase the volume on these? |
| 18:34.37 | *** join/#asterisk nDuff (~ccd@user-0ccss7l.cable.mindspring.com) |
| 18:35.00 | ManxPower | shaonss, so what is the problem? |
| 18:35.15 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 18:36.01 | nDuff | I've got a problem I -think- is due to a bug in Asterisk CVS HEAD: The new "sip notify" functionality sends notify events just fine, but when the phone sends back a challenge to authenticate the request, Asterisk isn't issuing a response. Is there anything I should do/look into to make sure this isn't my error before filing a bug report? |
| 18:37.13 | shaonss | i want to dial my asterisk through from my office and want to make 1st sip call then press # to make 2nd sip call them # for 3rd and so one how can i do it? |
| 18:37.30 | shaonss | through PSTN |
| 18:37.32 | ManxPower | nDuff, if the challenge doesn't match a type=user then asteirsk may ignore it |
| 18:37.37 | nDuff | alternately -- should I append a note to the currently-filed, closed bug for the feature's addition (bug#3231) instead of filing a new one? |
| 18:39.13 | shaonss | Manxpower: here i want to make only 1 pstn call to my home where the asterisk is |
| 18:39.18 | ManxPower | nDuff, if the bug is closed, then open a new one, but reference the closed bug |
| 18:39.48 | ManxPower | shaonss, If you have 1 line then you can only make 1 call. I still don't see the problem. |
| 18:39.53 | nDuff | ManxPower, it's a friend, so it should indeed be a user. |
| 18:40.31 | ManxPower | nDuff, a type=friend is type=peer + type=user |
| 18:41.31 | nDuff | ManxPower, yes, I know that. Was just affirming that the challenge being ignored doesn't look to be on account of it failing to be a user. |
| 18:42.21 | bajanman | manxpower: so if I have 1 extension [200] listed 2x in sip.conf, but one with peer the other with user, I could change it to 1 entry with friend? |
| 18:44.06 | shaonss | Manxpower:From office---(calls)-->pstn---->home(asterisk)---calls--->sip(call 1)the press # for sip call 2 then press # for sip call 3. so i want to make 3 sip calls when i connect asterisk through pstn incomming |
| 18:44.54 | ManxPower | bajanman, type=friend can cause unexpceted problems. |
| 18:44.59 | shaonss | manxpower:in 1 pstn session i am making 3 sip calls |
| 18:45.17 | bajanman | manxpower: how, and in what way? |
| 18:45.24 | ManxPower | *I* find that type=friend works fine for PHONES, but you should never use it for a gateway or anything like that. |
| 18:45.37 | bajanman | ahhhh |
| 18:45.39 | shaonss | manxpower: how to do this? |
| 18:45.46 | ManxPower | shaonss, I have no idea. |
| 18:46.01 | shaonss | did u get my idea? |
| 18:46.02 | ManxPower | Sounds to me like you need a cell phone. 8-) |
| 18:46.16 | bajanman | manxpower: for softphones, and phones generally (for extensions and such) I could get away with friend? |
| 18:46.44 | ManxPower | bajanman, Some people say no, but my advice is "yes" and I (usually) give pretty good advice. |
| 18:47.14 | bajanman | Manxpower: I'm just thinking, of saving all of those double entries for phone extensions.... if they were just friend... |
| 18:47.25 | bajanman | I mean if you have 10 extensions.. that's 20 entries. |
| 18:47.45 | ManxPower | bajanman, every single phone on all of my 8 asterisk systems is type=friend |
| 18:47.52 | bajanman | ahhh |
| 18:48.02 | K9DI_BSD_WrkStn | hello |
| 18:48.23 | bajanman | so, if something IS acting weird: it would be a "best policy" to create 2 entries to see if its the "friend" issue or not? |
| 18:48.40 | shaonss | manxpower:how can i make u understand the senerio? |
| 18:48.52 | K9DI_BSD_WrkStn | whereis asterisk |
| 18:48.52 | K9DI_BSD_WrkStn | asterisk: /usr/ports/net/asterisk |
| 18:49.03 | ramtha | ... |
| 18:50.09 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:50.27 | ManxPower | shaonss, not without large sums of cash. It's the weekend. If a problem doesn't interest me then you're not going to get much help from me. |
| 18:50.28 | K9DI_BSD_WrkStn | hello L|NUX |
| 18:50.44 | ManxPower | I don't get paid for any of this stuff, ya know. |
| 18:52.18 | Druken | ManxPower: don't you have a real life? your always here.... |
| 18:52.46 | shaonss | hello channel !!! please help on call handeling!!!!!!! |
| 18:53.17 | ramtha | shoot him |
| 18:54.13 | shaonss | help me on dialplan or exyensions.conf PLEASE!!!!!!! |
| 18:54.28 | ManxPower | I guess it's time for /ignore again |
| 18:54.29 | ramtha | shoot him twice |
| 18:56.02 | shaonss | ramtha: i am asking help from anybody on the channel not only u guys |
| 18:56.42 | ramtha | we see it...with crying you will not get any help |
| 18:56.51 | ManxPower | Druken, No, I don't have a life. |
| 18:57.35 | ManxPower | Actually, I do have a life. You'll notice that I'm online much of the time, but I'm frequently not chatting since I'm busy hiding from my users or thinking up ways to torture my users. |
| 18:58.25 | ManxPower | Right now, one of my lovers is busy working on a paper for school so I'm just wasting time and watching the console log of a new asterisk server we put in recently. |
| 18:58.50 | Druken | one of? |
| 18:59.11 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 18:59.49 | ManxPower | Druken, one may be fine, but two (or more) is divine. |
| 19:00.22 | Druken | :P |
| 19:01.50 | L|NUX | K9DI_BSD_WrkStn : hey |
| 19:02.12 | K9DI_BSD_WrkStn | ManxPower is your other name BOFH? |
| 19:02.41 | K9DI_BSD_WrkStn | J/K |
| 19:03.19 | ManxPower | K9DI_BSD_WrkStn, no, but he is my hero. |
| 19:04.25 | K9DI_BSD_WrkStn | LOL |
| 19:04.41 | K9DI_BSD_WrkStn | I'm catching up on back issues of UF |
| 19:04.45 | TheEmperor | ManxPower:what's the command to delete all voicemails from an extension? I tried putting in delete=yes to delete voicemails since voicemail to email is working but that doesn't seem to work.. |
| 19:04.57 | ManxPower | I discovered yesterday that Voicemail() only reads 256 bytes of it's command line. Not good if you have to send a Group Voicemail to 60 users. |
| 19:06.19 | Druken | voicemail to 60 users? |
| 19:06.30 | ManxPower | Druken, Yup! |
| 19:06.40 | ManxPower | One of the many stupid things my customers want. |
| 19:06.57 | ManxPower | At this office at least 1/2 of the people don't even have a computer. |
| 19:07.13 | RaYmAn-Bx | wouldn't it be better to use a mailing-list type thing for that? |
| 19:07.31 | Druken | ManxPower: please tell me they aren't using terminals |
| 19:08.28 | drumkilla | ManxPower: that's an easy change to make |
| 19:09.12 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 19:10.13 | drumkilla | ManxPower: line 3724 in app_voicemail.c :) |
| 19:11.18 | ManxPower | drumkilla, already did. tmp[4096] |
| 19:11.25 | kimc | K9DI de W8HD |
| 19:11.42 | drumkilla | ManxPower: cool |
| 19:11.47 | K9DI_BSD_WrkStn | hello |
| 19:11.56 | drumkilla | just trying to be helpful |
| 19:11.57 | ManxPower | Druken, no they don't use computers at all except for the couple of shared computers used to look up MLS (real estate) listings. |
| 19:13.10 | ManxPower | drumkilla, it would have been nice if it was a #define somewhere. |
| 19:13.14 | kimc | I'm still trying to find a source of info on setting up voicemail |
| 19:13.19 | ManxPower | ~docs |
| 19:13.21 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:13.21 | TheEmperor | how do i delete voicemails from the cli? |
| 19:13.41 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 19:14.03 | kimc | right I've located many places where there are docs but putting it all together -not so easy |
| 19:15.54 | kimc | 1 question is: What sound files need to be recorded by the extension user |
| 19:16.07 | ManxPower | The most common question I ask the users this past week (the first week of their Asterisk system): Did you read the manual for your phone? Most common answer: No. |
| 19:16.30 | Nethab | Wait I'm supposed to have a phone? dang it! |
| 19:16.33 | ManxPower | kimc, log into your voicemailbox. Pick option 0 listen. |
| 19:16.42 | Druken | ManxPower: oh... yeah... most realestate people don't use computers, it's still very... paper based... |
| 19:17.08 | ManxPower | Druken, There are a couple of people in the office that try to use technology. Oddly, they are the top sellers. |
| 19:17.09 | kimc | ManxPower: thanks lemme try that.. |
| 19:17.14 | Druken | phones come with manuals? |
| 19:17.19 | K9DI_BSD_WrkStn | kinc check your private screen please |
| 19:17.57 | ManxPower | The last person that /msg'd me had his tires slashed, but I had NOTHING to do with that. |
| 19:18.06 | kimc | K9DI: I think I have it open now.. |
| 19:18.13 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 19:18.30 | Nethab | that's because he was from the old soviet union and he left his windows rolled down |
| 19:18.43 | Nethab | so they slashed his tires |
| 19:19.08 | bjohnson | Anybody get this Alabama computer virus? |
| 19:19.41 | Nethab | sorry my computer doesn't sleep with his sisters, so i think i'm safe from that |
| 19:20.27 | pussfeller | viruses are so internet 1.0 |
| 19:20.59 | Nethab | who in there right minds start using something at version 1.0 i'm all about 0.1alpha |
| 19:22.44 | pussfeller | i think i finally talked my folks into lettng me replace their toshiba dk POS |
| 19:26.12 | *** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net) |
| 19:27.28 | Nethab | Point of Sale? |
| 19:27.35 | Druken | peice of shit? |
| 19:28.03 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 19:28.04 | Nethab | Pontif of Sacrelidge |
| 19:28.39 | Druken | i say someone order pizza and have it delivered to me |
| 19:28.53 | Nethab | would that be considered drunk dialing |
| 19:29.28 | Druken | i don't think so... unless your drunk... |
| 19:29.45 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 19:30.24 | Druken | !seen sivana |
| 19:30.40 | Nethab | ~seen sivana |
| 19:30.42 | jbot | sivana <~sivana@mixdown.ca> was last seen on IRC in channel #asterisk, 1d 17m 28s ago, saying: 'anyone here use Xlite?'. |
| 19:30.55 | Nethab | i use xlite |
| 19:33.09 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 19:33.56 | TUplink | i have x100p in my * box hooked to the pots dialing in/out work great is their a way to tell asterisk not to dial if it dosen have dialtone? |
| 19:34.23 | TUplink | exe someone is on the phone on pots upstairs |
| 19:34.40 | Elshar | hmm |
| 19:35.16 | TUplink | would it be bussydetect=yes in zapata? |
| 19:36.26 | Nethab | you should be able to tell if the phone is off-hook |
| 19:36.37 | Nethab | busydetect is if the called number is busy i think |
| 19:38.13 | telephoneman | who * |
| 19:38.17 | TUplink | thats what im reading |
| 19:39.37 | *** join/#asterisk cdr (~cdr@195-144-085-199.dyn.adsl.xs4all.be) |
| 19:44.02 | shaonss | any dialplan GURU? |
| 19:44.11 | shaonss | here? |
| 19:44.25 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 19:44.37 | Druken | shaonss: prolly lots |
| 19:45.22 | shaonss | can i flash a sip channel and make another sip call? |
| 19:45.35 | Druken | no |
| 19:45.51 | Druken | sip doesn't have "flash" |
| 19:45.52 | Qwell | well, if you have multiple call appearances |
| 19:46.12 | shaonss | ohhh thanks |
| 19:52.45 | Druken | ........ |
| 19:53.13 | ManxPower | TUplink, no |
| 19:53.32 | ManxPower | busydetect is just an alias for randomlydisconnectcalls |
| 19:53.43 | Qwell | ManxPower: :p |
| 19:53.49 | puowvip | hahaha |
| 19:53.52 | Druken | ManxPower: i had that problem :) |
| 19:54.22 | *** part/#asterisk xheliox (~jeff@user-0c6se9a.cable.mindspring.com) |
| 19:54.32 | WilliamK | latest * cvs build appears to be broken |
| 19:54.47 | Druken | again? |
| 19:54.56 | ManxPower | I don't care. I don't use CVS-HEAD. *grin* |
| 19:55.12 | WilliamK | yeppers |
| 19:55.41 | shaonss | is there any remote deskto type application in linux so that i can administer asterisk remotely? |
| 19:55.54 | Druken | yeah... ssh :) |
| 19:56.35 | shaonss | can u plese tell me how to use it? |
| 19:56.50 | WilliamK | hey Drunken, have you started getting errors on g729a.so when compiling the CVS build? |
| 19:57.15 | Druken | WilliamK: can't say i have, my CVS is older |
| 19:57.16 | *** join/#asterisk Rick_Hunter (~rhunter@02-051.008.popsite.net) |
| 19:57.25 | ManxPower | WilliamK, Huh? You don't compile g729, you get the binary from Digium |
| 19:57.30 | Druken | i tend not to fix what isn't broken |
| 19:57.57 | sean | I see I can use Dial(num1&num2); whichever picks up first gets the call.. can I make it connect to BOTH num1 & num2 ? |
| 19:58.01 | shaonss | use open G729 its veru good!!! |
| 19:58.05 | shaonss | i use it |
| 19:58.06 | ManxPower | sean, no. |
| 19:58.23 | shaonss | its freee |
| 19:58.27 | ManxPower | If you want that use .call files. |
| 19:58.29 | Druken | sean: define "connect" |
| 19:58.31 | WilliamK | Manx, when you compile * it gives a warning error about the g729a.so file |
| 19:58.42 | ManxPower | WilliamK, Weird |
| 19:58.50 | WilliamK | saying the module is older than the CVS |
| 19:58.55 | sean | Druken: call both.. manx answered, thanks |
| 19:59.41 | ManxPower | WilliamK, Did you "make clean" first? |
| 19:59.41 | sean | ManxPower: is there a voip-info page on .call files? ("call" is ambiguous) |
| 19:59.59 | ManxPower | sean, Why don't you go over to voip-info.org and find out. |
| 20:00.08 | ManxPower | ~google site:lists.digium.com sample.call |
| 20:00.24 | WilliamK | Manx, ALWAYS |
| 20:00.31 | sean | I explained it--"call" is ambiguous. |
| 20:00.32 | sean | Ok, thanks. |
| 20:00.39 | ManxPower | sean, search on sample.call |
| 20:00.44 | sean | thanks |
| 20:03.13 | TUplink | any one know how to get asterisk not to dial if the line is already in use? |
| 20:03.36 | ManxPower | TUplink, That feature is not currently supported by Asterisk. Let us know when you add it. |
| 20:04.17 | TUplink | Manx..... rite im not a programer nor do i have the phone system knoledge |
| 20:06.23 | shaonss | ManxPower: isn't busydetect does this job? |
| 20:06.40 | pussfeller | those cheepy f0x cards on ebay... good, bad? |
| 20:07.52 | shaonss | i have lot of x100p cards work well if u want any letme know i am using those great |
| 20:09.13 | Druken | TUplink: why would you have a phone outside of the network? |
| 20:09.26 | shaonss | how to use ssh asterisk and windows? |
| 20:09.46 | pussfeller | putty |
| 20:10.05 | pussfeller | shaonss: putty is a ssh/telnet client for windows |
| 20:10.24 | shaonss | thanks |
| 20:12.30 | shaonss | how to connect putty and asterisk any known command? |
| 20:14.42 | shaonss | excuse me i am very new in linux and asterisk :( |
| 20:14.43 | WilliamK | okie, figured out my issue... g729 licenses disappeared for some reason or another |
| 20:14.52 | WilliamK | and it's not rolling back to ulaw |
| 20:15.09 | tzafrir_laptop | shaonss, putty is just an ssh client and a terminal |
| 20:15.25 | WilliamK | ack.... |
| 20:15.26 | pussfeller | shaonss: this would be a good time to learn about bash... |
| 20:15.32 | tzafrir_laptop | if the server listens on ssh, you can connect with putty |
| 20:15.38 | WilliamK | freaking digium and their keys |
| 20:15.39 | *** join/#asterisk Milligan (~support@wkstn6.gnwd-noc.valuelinx.net) |
| 20:16.08 | ManxPower | WilliamK, It was a requirement by the patent holders of G729. |
| 20:16.12 | tzafrir_laptop | connecting to asterisk's CLI from the lisx CLI is running 'asterisk -r' as the user root |
| 20:16.12 | shaonss | what is bash? |
| 20:16.23 | tzafrir_laptop | see http://bash.org/ |
| 20:16.30 | tzafrir_laptop | just kidding |
| 20:16.32 | pussfeller | shaonss: the language of the shell you will get when you ssh in |
| 20:16.32 | shaonss | ohhh ok thanks |
| 20:16.54 | WilliamK | Manx, to change the damn host-ID EVERYTIME a new netcard is added and not taken away? sounds a bit harsh |
| 20:17.00 | pussfeller | shaonss: like the comand box on windows |
| 20:17.00 | tzafrir_laptop | you don't need to know much about bash. Badically you type a command and it runs. |
| 20:17.04 | WilliamK | I could see taken away, but not adding |
| 20:17.23 | tzafrir_laptop | there are some more advanced features, e.g: command1 && command2 |
| 20:17.41 | tzafrir_laptop | to run command2 after command1 only if command1 succedded |
| 20:17.49 | ManxPower | tzafrir_laptop, I don't think he'll learn much about bash from bash.org |
| 20:18.08 | pussfeller | shaonss: ls to list a directory and cat to read a file |
| 20:18.12 | tzafrir_laptop | ManxPower, he'll learn what not to say on chats |
| 20:18.19 | pussfeller | should be a nice start |
| 20:18.33 | shaonss | great got at least 2 commands. thanks |
| 20:18.56 | Milligan | I have configured my asterisk box to accept calls from the PSTN and the phone set rings, but the caller does not get the ring tone, what would cause that...? |
| 20:18.58 | pussfeller | ok thats not really bash, they are c programs but still |
| 20:19.02 | shaonss | which port to connect asterisk? any idea? |
| 20:20.16 | pussfeller | 22 if the sshd is running |
| 20:20.17 | ManxPower | tzafrir, Aye, that he will. |
| 20:20.18 | WilliamK | great... guess it's time to re-write my entire sip file again and reprogram phones |
| 20:21.53 | tzafrir_laptop | shaonss, are you trying to connect to the manager port? if so: don't |
| 20:22.09 | shaonss | ok |
| 20:22.55 | tzafrir_laptop | shaonss, use putty to get a command prompt ("shell") on your linux box. |
| 20:23.15 | tzafrir_laptop | BTW: What version of asterisk and what linux distribution is it? |
| 20:23.57 | Nethab | yawn |
| 20:26.38 | shaonss | tzafrir_laptop: yes i am connected :) |
| 20:26.52 | shaonss | tzafrir_laptop: thanks a lot |
| 20:28.44 | *** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 20:35.25 | *** join/#asterisk rogercharlie (~rogerchar@node-40244f5a.sfo.onnet.us.uu.net) |
| 20:36.45 | rogercharlie | does the 7960 have a reject button? |
| 20:37.47 | rogercharlie | the ip500 and a lot of softphones have a reject button |
| 20:39.40 | rogercharlie | | / , whats the correct delimeter for the dial string |
| 20:40.44 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 20:42.38 | shido6 | reject? |
| 20:42.46 | shido6 | what would the reject button do rogercharlie ? |
| 20:42.58 | shido6 | you can put the phone in dnd mode |
| 20:43.09 | shido6 | but not while the call is ebing received |
| 20:44.43 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
| 20:46.35 | rogercharlie | their sip firmware must not have any love |
| 20:46.55 | rogercharlie | dnd is still 3 or 4 buttons to make engage |
| 20:47.10 | shido6 | :) |
| 20:53.51 | telephoneman | rlist |
| 20:54.27 | telephoneman | help rlist |
| 20:56.45 | telephoneman | help userlist |
| 21:00.14 | telephoneman | list |
| 21:11.50 | *** join/#asterisk gpearson (~Graham@c-67-177-182-16.hsd1.in.comcast.net) |
| 21:11.51 | MrbBelvedr | help me now or i will hack everybody |
| 21:11.57 | MrbBelvedr | i will ping of death you guys |
| 21:12.05 | MrbBelvedr | everybody will be dead |
| 21:12.10 | MrbBelvedr | i am scanning your ip now |
| 21:12.17 | MrbBelvedr | your ip is 127.0.0.1 |
| 21:12.22 | MrbBelvedr | you are dead meat now |
| 21:13.18 | MrbBelvedr | * Mrbelvedr has quit IRC ("Rebooting") |
| 21:14.03 | gpearson | when I register sjphone to asterisk on the same network, i am getting a 404 message |
| 21:14.09 | gpearson | anyone know how to solve this |
| 21:15.31 | MrbBelvedr | i am back!!! |
| 21:17.04 | MrbBelvedr | gpearson are you doing something with a url? |
| 21:17.27 | gpearson | no I am getting this error from sjphone trying to dial an extension |
| 21:17.49 | gpearson | which is registered on the system. I am a newbie to asterisk and trying to learn |
| 21:18.42 | gpearson | it also tells me number does not exist, call rejected. 404 Not Found |
| 21:19.04 | Nethab | then the phone number does not exist in the context the phone is regietered into |
| 21:20.26 | Nethab | Every user or friend (read: phone) has a context defined. Inside that context in extensions.conf are defined all the phone numbers it can dial |
| 21:24.07 | MrbBelvedr | gpearson in your conig somewhere you might be setting a url. clear that out |
| 21:24.46 | MrbBelvedr | either that or you may be putting a server as http://voip.teliax.com when you should be putting just the hostname of voip.teliax.com |
| 21:24.58 | MrbBelvedr | try that first |
| 21:25.35 | gpearson | I have just loaded asterisk and have not touched any server configs yet. I am printing all documents off to try and understand this |
| 21:25.48 | gpearson | wonderful software package |
| 21:28.49 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 21:31.26 | Nethab | there is not a url generating that 404, the SIP protocol uses the same status codes to indicate success or failure |
| 21:31.38 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 21:31.40 | Nethab | ie 200 OK, and 404 not found |
| 21:32.59 | *** join/#asterisk ChkDigit (~ChkDigit@hsdbsk64-110-216-21.sasknet.sk.ca) |
| 21:35.34 | ChkDigit | Anybody around today? |
| 21:35.39 | Nethab | no we're all dead |
| 21:35.47 | ChkDigit | At appears so... |
| 21:35.48 | ChkDigit | =) |
| 21:36.21 | ChkDigit | Got any idea how to set up and extension that determines if a SIP channel is busy, then dials out Zap? |
| 21:36.45 | ChkDigit | Or can that be handled by setting channels=1 in sip.conf... |
| 21:36.53 | Nethab | if a sip channel is indeed busy, the dial command will fail and go to the next priority |
| 21:37.07 | ChkDigit | Next, or n+101? |
| 21:37.17 | Nethab | 101 is for not registered |
| 21:37.27 | ChkDigit | Hum... |
| 21:37.32 | Nethab | next is for busy |
| 21:37.37 | ChkDigit | Then where does it go after call completion? |
| 21:37.52 | Nethab | if the channel hangs up it goes to h,1 |
| 21:37.52 | ManxPower | "show application dial" will tell you. |
| 21:37.59 | ChkDigit | Thanks. |
| 21:40.16 | *** join/#asterisk jtodd (~jtodd@blob.fox-den.com) |
| 21:42.04 | Nethab | when are they gonna get rid of the n+101 |
| 21:42.47 | ManxPower | Nethab, Dial will go to n+1 if there is no n+101 |
| 21:43.10 | Nethab | i mean they were talking about using only return variables |
| 21:43.13 | Nethab | no context jumping |
| 21:43.19 | ManxPower | Then you can check ${DIALSTATUS} to see what happend. Just like in the macro-stdexten in the extensions.conf.sample |
| 21:44.07 | Nethab | not just for dial, but like EnumLookup too |
| 21:44.27 | *** join/#asterisk Kiseonik (~Surge@county.nat.lancs.ac.uk) |
| 21:45.17 | Kiseonik | Hi. Can Asterisk be setup to act as a software answermachine? Does it work with a hardware modem that responds to the AT command set over RS232? |
| 21:46.29 | Nethab | no, asterisk will not interact with a Hayes compatible modem |
| 21:46.34 | shido6 | yes it can |
| 21:46.37 | shido6 | Kiseonik, |
| 21:46.39 | Nethab | it requires a true FXO pots interface |
| 21:46.43 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 21:46.50 | FuriousGeorge | hey all |
| 21:47.00 | Kiseonik | It can or can't? |
| 21:47.09 | shido6 | what are you looking to do Kiseonik ? |
| 21:47.11 | Nethab | you can't just use a regular modem |
| 21:47.35 | Nethab | because linux only sees it as a serial com port |
| 21:47.53 | Kiseonik | I want a simple answermachine to record incoming calls if i'm not in. |
| 21:48.02 | Nethab | you have to have special drivers, and no manufacturer makes them |
| 21:48.03 | FuriousGeorge | my telco's wiring block thing outside looks like the standard one with two screws, red and green, and the modular jack. we never had a landline in this house, and i wanna disconnect my telcos wire's to run the house wiring off the tdm400 |
| 21:48.10 | Kiseonik | I ddi think it might've been a bit of an overkill to use Asterisk. |
| 21:48.36 | FuriousGeorge | the wierd thing is, the modular jack, which is usually for testing, had a black wire with an rj11 head plugged into it |
| 21:48.37 | Kiseonik | In which case, and recomendations? |
| 21:48.43 | Kiseonik | *any |
| 21:48.58 | FuriousGeorge | i fidured by unplugging that i was disconnecting the telco's wiring. can anyone confirm? |
| 21:49.13 | Nethab | you can simply unplug the rj11 and be disconnected from the pots |
| 21:49.44 | Nethab | yes that's the plug in that you or the phone company can use to plug directly into the phone co to bypass house wirirng |
| 21:50.00 | Nethab | so unplugging it will disconnect you |
| 21:50.21 | FuriousGeorge | Nethab: thats what i thought, thanks |
| 21:50.21 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 21:50.26 | kimc | back.. |
| 21:50.32 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:50.46 | kimc | dunno if that last msg made it |
| 21:50.52 | Nethab | the back part |
| 21:51.05 | Nethab | i saw your back, it's kinda hairy |
| 21:51.13 | kimc | heh |
| 21:51.16 | Kiseonik | This might be as good a place as any to ask, but if Asterisk can't do what I want, can anyone recommend a solution that would work? |
| 21:51.31 | kimc | Anyone what determines the CID on a call from a Zap channel to a sip phone > |
| 21:51.45 | FuriousGeorge | Nethab: one more thing, u think i can run enough voltage for two or three modern phones off one tdm400p's fxs |
| 21:51.53 | FuriousGeorge | daisy chained, of course |
| 21:52.15 | Nethab | one or two sure, but more than that might be stretching it |
| 21:52.50 | Nethab | asterisk can be used as an answering machine, but it'd be like driving a tank to the grocery store |
| 21:52.57 | FuriousGeorge | Nethab: actually one last thing, cant i get a voltage amplifier or something at radio shack, if i have a problem with the third phone |
| 21:52.57 | Nethab | not that people don't do that |
| 21:53.35 | Nethab | CID comes over Zap as an encoded signal from the phone company |
| 21:53.50 | Kiseonik | but if the hardware is not supported, then I can't use it anyway |
| 21:53.51 | newmedian | Has anyone implemented the old "Primeline" functionality: PSTN call comes in, caller is placed on hold, email/sms to mobile user advising call is on hold, mobile user calls in, authenticates, then is connected with caller on hold. ? |
| 21:53.57 | FuriousGeorge | i wanna implement an * install at an office that spends 300 a month forewarding calls to another office when noone is there |
| 21:54.21 | kimc | Nethab: right but how about for a call originating from a fxs port ? |
| 21:54.51 | kimc | Right now the sip phone displays 'asterisk' |
| 21:55.35 | Nethab | kimc: usually it's defined in one of the config files for each channel |
| 21:55.48 | kimc | is there is cid option for the channel in zapata.conf ? |
| 21:56.16 | Nethab | kiseonik, around here people use an ATA with an FXO port, which is nothing more than a regular phone line, or a PCI card made by a few manufacturers |
| 21:56.19 | kimc | ..bet there is. |
| 21:56.51 | Nethab | the problem isn't the hardware it's the lack of drivers |
| 21:57.31 | newmedian | Nethab/kiseonik, perhaps the $7 USD clone card from eBay might be the approach? |
| 21:58.07 | Nethab | clones huh, isn't that what started the sith war |
| 21:58.09 | ChkDigit | newmedian: Can be done easily with DISA. |
| 21:58.40 | Kiseonik | tbh, its too much hassle then. |
| 21:58.46 | ChkDigit | * Answers call, plays back message, and parks call. Sends SMS to person who calls into DISA and dials parking extension. |
| 21:59.01 | Kiseonik | Thank you very much for your help, dudes. I'm off. |
| 21:59.09 | *** part/#asterisk Kiseonik (~Surge@county.nat.lancs.ac.uk) |
| 22:00.08 | newmedian | ChkDigit, does that allow for privacy for allowing only a specific extension to pick up the parked call? Otherwise that's good, but too open. |
| 22:00.35 | ChkDigit | Can do whatever you program it to. |
| 22:00.58 | newmedian | Thank you ChkDigit, I'll check it out. |
| 22:01.06 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 22:01.08 | ChkDigit | Just have the DISA authenticate into a context that can get the parked call. |
| 22:03.35 | denon | OT, but anyone in here ever use the debian installer to create lvm or raid stuff? |
| 22:11.11 | *** join/#asterisk derka (derka@phiral.com) |
| 22:14.58 | *** join/#asterisk destro_dm4 (~rdsaffell@dsl092-161-155.wdc2.dsl.speakeasy.net) |
| 22:15.08 | destro_dm4 | hello |
| 22:15.42 | destro_dm4 | just setup an asterisk pbx for a non profit, and it's awesome... thanks so much for everything you guys have been doing with it |
| 22:16.02 | puowvip | destro, do you use speakeasy voice? |
| 22:16.06 | destro_dm4 | no |
| 22:16.09 | puowvip | 'k |
| 22:16.30 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-149.arcor-ip.net) |
| 22:16.33 | destro_dm4 | there isn't enough info on their site to make me comfortable with their service |
| 22:16.45 | Blackvel | is there a Netmeeting (SIP) client available for *? |
| 22:16.47 | destro_dm4 | seems like they are missing a lot of options, but I guess they are just getting started |
| 22:16.48 | puowvip | yep, me neither. went with broadvoice. |
| 22:17.10 | destro_dm4 | I went with Vonage, but after setting up asterisk, I think I might switch :) |
| 22:17.21 | Blackvel | does it still not support *? |
| 22:17.43 | puowvip | vonage does not support asterisk atm |
| 22:17.45 | destro_dm4 | can you guys help point me in the right direction on a couple issues? |
| 22:17.50 | puowvip | or anything other than their telephone adapter |
| 22:18.09 | destro_dm4 | yeah, and the whole analog to digital to ata to digital is a little crazy IMHO |
| 22:18.20 | destro_dm4 | and I would love to get out from behind their ATA |
| 22:18.32 | destro_dm4 | how can I transfer a call to a voice mailbox? |
| 22:22.52 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:24.14 | Nethab | to transfer a call to voicemail you set up an extension that calls VoiceMail |
| 22:24.22 | Nethab | and transfer them to it |
| 22:24.33 | Nethab | i use *1 but that's just me |
| 22:24.59 | Nethab | exten => *1,1,VoiceMail |
| 22:25.21 | destro_dm4 | ok, I thought I was getting a prompt for extension, pwd when I did that |
| 22:25.52 | newmedian | (I think he means the *97) |
| 22:26.12 | Nethab | there's VoiceMailMain and VoiceMail |
| 22:26.24 | destro_dm4 | ah.. that's probably it |
| 22:27.47 | newmedian | Has anyone here connected an Asterisk to an ISDN BRI, and if so what hardware did you use to do it? |
| 22:27.56 | Nethab | i use Voicemail(b${EXTEN}) the b says "person is busy" first |
| 22:34.50 | Blackvel | newmedian: how many lines? |
| 22:36.19 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 22:38.02 | newmedian | Blackvel: 2 |
| 22:38.12 | *** join/#asterisk aptanet (~paul@home.aptanet.net) |
| 22:40.35 | *** join/#asterisk luisgrin (~luis@209.99.227.220) |
| 22:41.37 | Blackvel | newmedian: get an fritz isdn pci passive (capi 2.0 + junghanns chan_capi) or either a HFC-S pci card with junghanns bristuff |
| 22:42.08 | puowvip | 97 a month seems steep for ISDN BRI. |
| 22:42.14 | Blackvel | might also work with active isdn cards + m_isdn |
| 22:42.32 | Blackvel | kernel 2.6 brings support for capi 2.0 for passive cards even |
| 22:42.40 | Blackvel | but you would have to check what |
| 22:43.19 | Blackvel | puowvip: I am not sure, but europe mostly uses isdn? at least germany |
| 22:43.47 | puowvip | well this is in Ohio, USA |
| 22:43.53 | heison | does anyone know what voltage the TDM400 supplies? AC 75V for ring and -48V DC for off hook? |
| 22:44.40 | newmedian | Thanks; Was looking to get advantages of PRI but at low cost for a test installation; In Canada they typically segregate all services types, so if you vary your conversation (BRI vs. PRI vs PRI with data etc.) then you have to talk to a different beaurocratic group. Can be hair pulling to get anywhere meaningful in a single conversation. |
| 22:54.00 | pussfeller | i have phone service over fiber optic, does that give me any advantages |
| 22:54.14 | pussfeller | or necessary tweaks |
| 22:57.24 | luisgrin | im very new to asterisk, i have the opportunity of installing a pbx in a new officce, but i have not expreince, i need basically some intro in this tech and what can be achieved |
| 22:58.37 | Blackvel | luisgrin: yeasy. it can do almost everything. what else do you want to know? |
| 22:58.55 | Blackvel | www.voip-info.org/wiki-asterisk for a startup |
| 22:59.17 | Blackvel | there where some other doc sites too |
| 22:59.34 | Blackvel | there was some bot command in this channel available |
| 22:59.40 | Blackvel | cant remember |
| 22:59.42 | Blackvel | ~docs |
| 22:59.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:59.52 | Blackvel | right :) |
| 23:01.02 | luisgrin | i can install asterisk, ill need about 30 extentions and about 6 external analogic lines |
| 23:01.46 | luisgrin | i told the client to let me try this tech, my fear is about sound quiality and all this things |
| 23:02.53 | luisgrin | docs |
| 23:03.01 | luisgrin | !docs |
| 23:06.50 | Blackvel | its "~docs" |
| 23:07.05 | Blackvel | luisgrin: 30 extensions? 30 telephones? |
| 23:07.38 | Blackvel | luisgrin: check the digium website. you can get a dtm400p which has 4 analog ports |
| 23:07.43 | luisgrin | yes 30 internal phones (extentions) and 6 external lines (more or less) |
| 23:07.56 | Blackvel | do you have to buy them? |
| 23:08.00 | Blackvel | or analog phones? |
| 23:08.05 | luisgrin | ok i was lookin digium |
| 23:08.15 | Blackvel | either you get voip phones |
| 23:08.22 | Blackvel | and connect over lan |
| 23:08.44 | Blackvel | or you use a T1/E1 card, where E1 is 30 |
| 23:08.58 | Blackvel | and then you need a channel bank to connect your analog phones to |
| 23:09.03 | Blackvel | sound quality is perfect |
| 23:09.12 | luisgrin | i mean 30 working places but only 6 teleph lines |
| 23:09.25 | Blackvel | but how many telephones?30? |
| 23:09.46 | luisgrin | no, i mean if i dont use voip i must buy 30 telef |
| 23:09.54 | Blackvel | jupp |
| 23:09.57 | luisgrin | 30 working places |
| 23:10.05 | Blackvel | you also have to do this with asterisk |
| 23:10.23 | luisgrin | i told my client that i think that wih asterisk the PC will be enough |
| 23:10.26 | Blackvel | but you have the choice. analog+channel bank or voip telephones |
| 23:10.31 | Blackvel | hm |
| 23:10.40 | Blackvel | there are softphones + headsets |
| 23:10.53 | Blackvel | maybe you can find someone in this channel who has experience with this |
| 23:10.55 | newmedian | (or adapters, with existing traditional phones, Blackvel?) |
| 23:11.01 | Blackvel | my personal experience is, that softphones are crap |
| 23:11.20 | Blackvel | well |
| 23:11.25 | Blackvel | would be too many adapters |
| 23:11.26 | luisgrin | yes???? mmmmmmmmmm hardware phones are expensives |
| 23:11.42 | Blackvel | well |
| 23:11.45 | Blackvel | but good sound quality |
| 23:11.52 | newmedian | I've had relatively good experience using Xten's eyeBeam, and manually setting the Task Manager Priority of the task to High. Quality has been good, with a headset. |
| 23:11.53 | Blackvel | voip phones are expensive |
| 23:12.07 | Nugget | expensive is relative. |
| 23:12.09 | Blackvel | i used xlite |
| 23:12.14 | Blackvel | always had problems |
| 23:12.21 | Blackvel | soundcard |
| 23:12.24 | newmedian | I started with XLite, but it was... passable. eyeBeam was much better. |
| 23:12.25 | Blackvel | i called someone from uk |
| 23:12.32 | Blackvel | with my * hardware phone (zap) |
| 23:12.41 | Blackvel | he had a headset and sound quality was that bad |
| 23:12.53 | Blackvel | I would never go with an software solution |
| 23:13.14 | Blackvel | but, before spending thousands of $ on hardware, you can test your scenario with an prototyp |
| 23:13.22 | newmedian | (Do you know when the new IAXy adapters will be available?) |
| 23:13.27 | Blackvel | and check newmedian's suggestion |
| 23:13.44 | Nugget | $65 for a budgetone is practically free compared to the average PBX station. |
| 23:13.53 | luisgrin | i beg you pardon for my awfull english, i understood that there must be NOT OK with regular soundcard like those on board and some soft client? |
| 23:13.59 | Blackvel | you would buy budgetone? :) |
| 23:14.12 | Nugget | no, but they sure aren't expensive. :) |
| 23:14.12 | newmedian | (This is the eyeBeam I refer to: http://www.xten.com/index.php?menu=products&smenu=eyebeam) |
| 23:14.18 | Blackvel | forget onboard soundcards |
| 23:14.28 | Nugget | I'd rather have one of them than use xlite all the time |
| 23:14.32 | Blackvel | my asus has one, i have such a huge problem with sounds |
| 23:14.45 | Blackvel | Nugget: right |
| 23:14.48 | Blackvel | but you cant do transfers |
| 23:14.56 | Blackvel | for business, get snom or something else |
| 23:15.10 | luisgrin | snom??? what is that? |
| 23:15.14 | Nugget | I was just disputing the "voip phones are expensive" assertion |
| 23:15.24 | Nugget | they aren't |
| 23:15.33 | Blackvel | luisgrin: voip phone (sip) |
| 23:15.37 | newmedian | Grandstream just released this business phone which I am salivating for... http://www.grandstream.com/y-gxp2000.htm |
| 23:15.41 | luisgrin | voip phones how much they can cost? |
| 23:15.46 | Blackvel | grandstream is horrible |
| 23:15.51 | Blackvel | as they do with their ata |
| 23:15.55 | Blackvel | :) |
| 23:16.00 | Nugget | luisgrin: try http://voipsupply.com/ |
| 23:16.08 | Blackvel | thats the reason why ppl buy sipuras |
| 23:16.11 | luisgrin | ok Nugget |
| 23:16.23 | Blackvel | luisgrin: i had one offer for EUR130 |
| 23:16.39 | luisgrin | wow!!! a lot of money is it not? |
| 23:16.40 | Blackvel | list price for that one model was EUR230 (if <50) |
| 23:17.00 | Blackvel | then get analog phones |
| 23:17.08 | Blackvel | card for * costs you $500 |
| 23:17.11 | destro_dm4 | if I use "exten => *1,1,VoiceMail" and transfer a call to *1, how does it know which mailbox? |
| 23:17.16 | Blackvel | + channel bank $1000-1500 |
| 23:17.28 | Blackvel | and you are not that exententable |
| 23:17.33 | Blackvel | extentable |
| 23:17.37 | destro_dm4 | or do I have to enter the mailbox before I complete the transfer |
| 23:17.46 | CeNTauR^6 | are there any places to buy cheep VoIP phones in Australia? |
| 23:18.08 | Blackvel | destro_dm4: i think you have to pass this voicemail(...) |
| 23:18.27 | Nugget | CeNTauR^6: do they have google in australia yet? |
| 23:18.30 | luisgrin | ok, i must try a lot :) |
| 23:18.37 | newmedian | I just sent someone some Australia links the other day. I do *not* know these companies, and I cannot vouch for them. |
| 23:18.43 | newmedian | Here's one: http://www.techtopia.com.au/product_info.php/cPath/36_60/products_id/930 |
| 23:18.54 | newmedian | Here's another: http://myvoipstore.com.au/catalog/product_info.php?cPath=44&products_id=33 |
| 23:18.58 | Blackvel | e.g s,4,Voicemail(u66) |
| 23:19.04 | newmedian | Yet another: http://www.centrixphone.com/products.php |
| 23:19.13 | newmedian | Second to last: http://ozvoip.com/productCategory.php?cat=ipphone |
| 23:19.17 | Blackvel | luisgrin: i would always go with the new voip phone solution |
| 23:19.22 | newmedian | Last link: http://www.stcpl.com.au/index.php?p=44§ion=voip |
| 23:19.23 | Blackvel | if you have to buy new hardware |
| 23:19.26 | CeNTauR^6 | thanks |
| 23:19.39 | Blackvel | or you can try pc + softphones + bluetooth headsets |
| 23:19.48 | Blackvel | but that is not really professional :) |
| 23:19.57 | destro_dm4 | Blackvel: right, I know how to do that, but I am wondering how I can dial something like *910 and it would put me into mailbox 10's voicemail |
| 23:20.15 | destro_dm4 | probably something like _*9XX... |
| 23:20.29 | Blackvel | exten => *910,1,VoicemailMain(${CALLERIDNUM}) |
| 23:20.38 | Blackvel | or |
| 23:20.42 | Blackvel | exten => *910,1,VoicemailMain(10) |
| 23:20.43 | destro_dm4 | but that would put me into extension 910's vm |
| 23:20.51 | Blackvel | no |
| 23:20.52 | CeNTauR^6 | no i have a few ATA's but i wanted to ge my hands on a few good VoIP phones |
| 23:20.54 | Blackvel | depends on your callerid |
| 23:20.58 | Nugget | voicemailmain(${EXTEN:2:2}) |
| 23:21.00 | Blackvel | if your callerid is 10 |
| 23:21.10 | Blackvel | dont confuse exten with calleridnum |
| 23:21.29 | destro_dm4 | I probably am |
| 23:21.39 | Blackvel | what is the number of your telephone? |
| 23:21.41 | Blackvel | analog? |
| 23:21.52 | Blackvel | callerid cares about this |
| 23:21.54 | Blackvel | e.g isdn msn |
| 23:22.02 | Blackvel | or use fixed numbers |
| 23:22.29 | Blackvel | wow |
| 23:22.35 | Blackvel | very late here in germany |
| 23:22.37 | Blackvel | i am off |
| 23:22.40 | Blackvel | cu, good luck |
| 23:23.28 | *** join/#asterisk jterrero (~jt@68.199.215.186) |
| 23:26.44 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 23:28.26 | *** join/#asterisk uzd (optimist@eurocompton.net) |
| 23:28.30 | uzd | hi |
| 23:29.49 | uzd | when I was using asterisk@home my PSTN phone (using a PAP2-NA) would flash when I had new voicemail. now that I've installed native asterisk it doesn't give me notification. |
| 23:29.53 | uzd | how do I set that service up? |
| 23:30.22 | uzd | what sort of signal is sent to a phone to tell it that? |
| 23:34.53 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 23:35.02 | iq | Hi, anyone here using SER? |
| 23:36.45 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 23:40.19 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 23:40.19 | *** mode/#asterisk [+o kram] by ChanServ |
| 23:42.49 | destro_dm4 | hello, I have a Polycom 500 and I'm wondering if I can program a key to do a call park? |
| 23:43.03 | destro_dm4 | is there something I can dial to transfer, then to 70? |
| 23:43.11 | *** join/#asterisk Capaneus (~Capaneus@c-66-41-30-188.hsd1.mn.comcast.net) |
| 23:44.32 | Capaneus | Hey I'm looking for some advice for connecting asterisk to FreeWorld Dialup through a NAT |
| 23:45.13 | drumkilla | use iax |
| 23:46.34 | drumkilla | http://www.fwd.pulver.com/advanced/iax |
| 23:47.10 | *** join/#asterisk PeaCeKeePa (~free@d141-52-2.home.cgocable.net) |
| 23:47.36 | Nivex | <3 FWD IAX :) |
| 23:48.41 | PeaCeKeePa | looking for some help with weather.sh (replacement for weather.agi) - i'm not good with scripting, plz /msg me... TiA |
| 23:54.14 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 23:55.00 | iq | Hi, is it legal to selling asterisk without source code after minor changes in code? |
| 23:55.10 | drumkilla | absolutely not |
| 23:55.37 | drumkilla | it's illegal to sell asterisk without the code, period. |
| 23:55.46 | drumkilla | without a commercial license from Digium |
| 23:56.46 | iq | if I rewrite sip channel, not using the original chan_sip. will I have to include my source as well? |
| 23:56.52 | drumkilla | yes |
| 23:56.57 | drumkilla | because it is a derivative work |
| 23:57.01 | PeaCeKeePa | so i'm trying to implement the advice given on this site to get weather script working: http://edgett.bc.ca/simonsays/archives/000228.html |
| 23:57.05 | drumkilla | your chan_sip is obviously worthless without asterisk |
| 23:58.05 | iq | drumkilla, I see... :) ... I am busted I guess |
| 23:58.08 | PeaCeKeePa | but i'm not sure if the script is being run or not |
| 23:58.15 | drumkilla | iq: you haven't been doing that, have you? |
| 23:58.31 | iq | drumkilla, no |
| 23:58.37 | *** join/#asterisk ShadowMaster1 (~askme@host89-133.rancor.birch.net) |
| 23:58.42 | drumkilla | iq: if you need to do any of this, please contact Jim Webster at Digium for details on commercial licensing |
| 23:58.44 | PeaCeKeePa | any ideas? |
| 23:58.48 | iq | drumkilla, not yet... but i think I wont do it |
| 23:58.48 | drumkilla | he is the licensing director |
| 23:58.49 | PeaCeKeePa | ne1? |
| 23:58.54 | drumkilla | iq: good :) |
| 23:59.00 | Capaneus | drumkilla: thanks I saw that but I'd put that info in sip.conf instead. I've moved it to iax.conf, and now when I call I get this error on my cli |
| 23:59.28 | iq | drumkilla, sone more question. If I get commercial licence from Digium. then I can sell my changes without source? I wont even have to show my code to Digium? |
| 23:59.35 | ShadowMaster1 | Am I in the right place to seek help on a zaptel problem? |
| 23:59.38 | drumkilla | iq: that is correct |
| 23:59.49 | drumkilla | iq: but like I said, you need to contact Jim Webster for specific details on that |
