| 00:01.31 | *** join/#asterisk fmueller (~user@p5DE8B7B8.dip.t-dialin.net) |
| 00:02.52 | *** part/#asterisk fmueller (~user@p5DE8B7B8.dip.t-dialin.net) |
| 00:06.57 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:06.57 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 00:10.44 | *** join/#asterisk thenk_ (3a07d043@gateway/web/freenode/ip.58.7.208.67) |
| 00:11.13 | thenk_ | hi, could anyone point me in the right direction of getting app_rtsp to work? |
| 00:39.57 | *** join/#asterisk coppice (~chatzilla@m121-202-91-54.smartone-vodafone.com) |
| 00:47.58 | JerJer | thenk: i'll eyeball it once i finish up with my current consulting task |
| 00:55.52 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 00:55.57 | phix | ~pb |
| 00:55.57 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:58.54 | phix | ok I am ringing from my mobile phone to my other sip phone number --> http://asterisk.pastey.net/146921 |
| 00:59.37 | phix | It is trying to connect to s however the correct number it should be going to is the one in the SIP To: header thingy |
| 00:59.56 | phix | Any way to use that one instead of trying to goto s? |
| 01:00.05 | phix | thnx |
| 01:01.23 | *** join/#asterisk kerx (~kerx@209.208.63.169) |
| 01:03.44 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:05.59 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 01:09.14 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 01:10.23 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 01:15.41 | thenk | hey, i suppose i can try again: anyone able to point me in the direction of getting app_rtsp to work? |
| 01:15.59 | phix | I don't even know what that is |
| 01:16.32 | phix | Yeah I am trying to get some help here too, I thikn everyone is asleep though, or been murdered by some one impatient :) |
| 01:16.42 | thenk | hehe :) |
| 01:17.27 | thenk | JerJer: thanks mate |
| 01:17.57 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 01:19.19 | phix | Perhaps you can help me thenk |
| 01:19.49 | thenk | Yeah just looking at it now |
| 01:19.52 | phix | I have two accounts with the same SIP provider, I want to handle calls coming from each account differently |
| 01:20.12 | phix | It seems to default to the first account so I cant change the context |
| 01:20.27 | phix | nor does it pay any attention to the SIP To: header |
| 01:20.54 | phix | which is the correct number / extension it should be going to |
| 01:20.56 | WIMPy | Did you set an extension in the registration? |
| 01:21.01 | thenk | there's an example dialplan in freepbx which can help you snip the To address out of there |
| 01:21.39 | thenk | phix: "Call from 'myMainSIPAccountPhoneNumber}'" |
| 01:21.52 | phix | thenk: yeah I changed some stuff |
| 01:21.55 | thenk | Is that meant to be {$myMainSIPAccountPhoneNumber} ? |
| 01:22.03 | phix | nah get with the the }, typo |
| 01:22.11 | thenk | ah, okay |
| 01:22.20 | phix | I just changed the numbers and ip addresses, but everything else is untouched |
| 01:22.40 | thenk | what was the original INVITE? I only see the ACK |
| 01:22.56 | phix | ummm, I will have a look |
| 01:23.38 | thenk | also, make sure your registration specifies the extension to contact - if you register like this "username:password@host" it will send it to s, if you register like "username:password@host/extension" it will send to extension@host |
| 01:23.51 | thenk | your host, i mean |
| 01:24.27 | thenk | some providers will even accept username:password/extension@context |
| 01:26.26 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 01:32.14 | phix | ok here we go --> http://asterisk.pastey.net/146927 |
| 01:32.46 | phix | ok thnx thenk I think that is my problem |
| 01:34.35 | thenk | To: <sip:myOtherSIPPhoneNumber@mySIPProvider.nx> |
| 01:34.51 | thenk | hmm |
| 01:41.20 | phix | thenk: sup? |
| 01:41.34 | phix | thenk: ok your suggestion worked! much respect |
| 01:41.50 | thenk | :P) |
| 01:41.52 | thenk | :) |
| 01:42.24 | thenk | now if only I could get my video playback working.. :P |
| 01:42.25 | phix | Is it bad to use Goto in extensions.conf the same reason why it is back to use it in programming code? |
| 01:42.28 | phix | <PROTECTED> |
| 01:42.35 | phix | thenk: If only I could help you with that |
| 01:42.40 | phix | I have never done video phones |
| 01:44.29 | phix | I am guessing they are expensive |
| 01:44.36 | thenk | phix: subroutines arent bad, you just should use them for what they're good for |
| 01:46.57 | phix | Well I have two accounts which are handled differently, I am using Goto to redirect calls from accounts one that should be coming in to account two (I do this based on the callerid) |
| 01:53.05 | phix | hmmm now to change voicemail greeting message |
| 01:53.31 | phix | "Person at extension foobar is unavailable,..." |
| 01:54.02 | phix | yay found it |
| 02:01.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:02.01 | phix | ok, now to de-americanize the voicemail |
| 02:02.57 | thenk | there's a number of packages out there |
| 02:03.18 | thenk | australian accented replacements for all asterisk prompts |
| 02:03.22 | thenk | and phonetics |
| 02:04.00 | thenk | http://www.openvoice.com.au/free/ |
| 02:04.29 | thenk | the free set covers most asterisk prompts |
| 02:04.40 | thenk | all voicemail and basic functionality |
| 02:04.42 | thenk | i think you have to pay for the phonetics |
| 02:06.30 | phix | thenk: w00t! |
| 02:06.36 | *** join/#asterisk KNERD (~KNERD@99.96.118.71) |
| 02:07.08 | phix | How much? and what is the licence like? pay for each asterisk box you want to stick it on? or one fee? |
| 02:07.29 | thenk | it's one fee for the premium package |
| 02:07.39 | thenk | but the free package has all the voicemail prompts and most of the stuff you'd usually encounter |
| 02:09.20 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 02:11.10 | phix | nice |
| 02:11.13 | phix | thnx again <3 |
| 02:12.35 | thenk | :) |
| 02:21.46 | *** join/#asterisk OldMonk (~raju@122.176.230.39) |
| 02:22.36 | OldMonk | using sip extensions of the form ab_12345; given that, is there any way to select an extension for spying using the "....#" form in ChanSpy? |
| 02:22.47 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:23.24 | OldMonk | asterisk 1.6.2.9, debian testing |
| 02:24.47 | WIMPy | There's still no such thing as 'sip extensions', but what form are you referring to? |
| 02:24.59 | OldMonk | sip channels, then |
| 02:25.15 | WIMPy | You should pass the peer name +"-" to spy on a given peer. |
| 02:25.52 | OldMonk | if my peer name is "SIP/ab_12345", exactly what string do i type in to spy on that? |
| 02:26.29 | WIMPy | sip/ab_12345- |
| 02:26.58 | OldMonk | while spying on another channel? that's cool! |
| 02:27.06 | WIMPy | the trailing - is only necessary in case you have overlapping names, but it won't hurt. |
| 02:27.10 | OldMonk | is that documented somewhere? |
| 02:27.20 | phix | OldMonk: no, debian stable now :) |
| 02:27.21 | WIMPy | While spying? |
| 02:27.35 | WIMPy | You need to pass that as parameter to ChanSpy. |
| 02:27.35 | OldMonk | WIMPy: yes, you can switch channels while spying |
| 02:27.57 | OldMonk | nono, not the initial call -- i'm talking about switching to a different channel while spying on one |
| 02:28.12 | OldMonk | s/call/invocation/ |
| 02:28.23 | OldMonk | oh geez |
| 02:28.34 | phix | hmmm ok so I copied en-au directory into /usr/share/asterisk/sounds/digits and set the language but it is using the default asterisk voice :/ |
| 02:28.47 | WIMPy | I don't think there's any way to specify where you want to switch to. |
| 02:29.05 | phix | but it is using the en-au sounds for the voicemail prompts and that except for digits |
| 02:29.35 | OldMonk | WIMPy: sure, chanspy has the facility to type in NNNN# to switch to a different channel while spying on one |
| 02:30.54 | WIMPy | It says it's appended to the initial parameter, so you should be able to split it any way you want. |
| 02:30.58 | OldMonk | From http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy : Dialing a series of digits followed by # builds a channel name to append to <chanprefix> (e.g. run ChanSpy(Agent) and dial 1234# while spying to jump to channel Agent/1234) |
| 02:31.42 | WIMPy | The only trouble is that you will be limited to the 16 DTMF digits. |
| 02:32.02 | OldMonk | hmm, i don't know -- in the example, the "/" is added automatically |
| 02:32.25 | WIMPy | But if you put the ab_ in the call to chanspy , you should be able to interactively input the rest. |
| 02:32.55 | OldMonk | so if i invoke ChanSpy with ("SIP/ab_") and type in "12345#" won't it switch to "SIP/ab_/12345"? |
| 02:33.00 | phix | the weird thing is there was no /usr/share/asterisk/digits directory |
| 02:33.44 | WIMPy | I don't read anything about a / being inserted. Maybe you should just try it. |
| 02:34.09 | OldMonk | WIMPy: look at the example |
| 02:34.21 | phix | how so I find out where the digits directory is? |
| 02:34.27 | WIMPy | Which one? |
| 02:34.40 | OldMonk | WIMPy: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy (the one i posted above) |
| 02:34.52 | phix | so = do |
| 02:36.08 | phix | ok it is in en_US_f_Allison :/ |
| 02:36.30 | WIMPy | The example doesn't seem to make sense and otherwiese the documentation there seems to say the same. |
| 02:37.40 | OldMonk | ok, will try that |
| 02:38.18 | WIMPy | Ah, the one above usage. That's contradictory indeed. |
| 02:39.34 | phix | ok thanx for the non-help, I figured it out |
| 02:40.54 | OldMonk | phix: congrats :) |
| 02:51.28 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 02:56.52 | phix | OldMonk: &%$^@! |
| 02:58.06 | OldMonk | phix: sometimes you can solve your problems by just articulating them -- makes them clearer to you |
| 03:11.35 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 03:15.33 | phix | ok I have en-au language set everywhere and it is working sweet :D |
| 03:15.36 | OldMonk | later |
| 03:23.30 | phix | ok, now to find some creative commons music on hold |
| 03:24.06 | *** join/#asterisk shapr (~shapr@nat/digium/x-jkwygozwlfqxsozv) |
| 03:39.49 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 03:40.11 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:43.01 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 03:51.26 | IsUp | phix: 'secret garden - nocturne' if you need some drama |
| 03:52.07 | IsUp | phix: 'booka shade - body language' if your company says "we have young and dynamic staff" |
| 03:52.31 | IsUp | phix: and if your company looks so serious, play Bach or Mozart |
| 03:52.38 | shapr | "Digium - Music on Hold" if you like the classics. |
| 03:52.43 | CRCinAU_ | lol |
| 03:52.47 | IsUp | haha |
| 03:52.49 | CRCinAU_ | ahhh vomit material |
| 03:53.06 | CRCinAU_ | I found a folder of MP3s that I've used forever |
| 03:53.15 | CRCinAU_ | no idea what CD it was, but it's quite good. |
| 03:53.21 | shapr | sics the MPAA on CRCinAU_ |
| 03:53.38 | CRCinAU_ | heh |
| 03:53.52 | shapr | But really, there's some good stuff on jamendo.com that's specifically licensed for commercial use. |
| 03:54.51 | CRCinAU_ | I don't even know the names of the songs... |
| 03:54.57 | CRCinAU_ | its just 01.mp3, 02.mp3 etc |
| 03:55.07 | CRCinAU_ | I think I first started using them about 6 years ago |
| 03:55.24 | CRCinAU_ | ever since using mpg123 for hold music was supported :D |
| 03:56.53 | CRCinAU_ | that would have been around 2005/6 or so... version 1.2 iirc |
| 03:57.24 | IsUp | why no one running 'playing now' thing? it was fun |
| 03:57.35 | CRCinAU_ | ? |
| 03:57.45 | IsUp | now listening andru donalds - mishale |
| 03:57.51 | CRCinAU_ | oh LOL |
| 03:57.58 | CRCinAU_ | I don't think anyone cares anymore ;) |
| 03:58.07 | CRCinAU_ | but I did have that kinda thing with mIRC back in the day ;) |
| 03:58.26 | CRCinAU_ | I even wrote my own MP3 player that talked DDE to mIRC :P |
| 03:59.10 | IsUp | yeah old days |
| 04:00.52 | Freeaqingme | CRCinAU_, in most channels these automatic messages aren't allowed |
| 04:00.56 | Freeaqingme | suppose 10% of ubuntu did that |
| 04:01.03 | Freeaqingme | * #ubuntu |
| 04:01.14 | CRCinAU_ | yeah - but people don't care these days |
| 04:01.19 | Freeaqingme | then you have every second a status update of what someone is listening |
| 04:01.20 | CRCinAU_ | back in the day, it was all the rage |
| 04:01.37 | CRCinAU_ | you know, when everything was .mp2 |
| 04:01.44 | CRCinAU_ | and .s3m and .xm had real value |
| 04:02.30 | Freeaqingme | My .wav files were just fine |
| 04:02.45 | CRCinAU_ | then this new MP3 format came along, and people shunned it for a little while before it was embraced like a child embracing a puppy |
| 04:03.02 | Freeaqingme | yeah, that was right before napster I guess? |
| 04:03.04 | CRCinAU_ | .wav in ADPCM? ;) |
| 04:03.20 | CRCinAU_ | where you'd get 4:1 compression! :D |
| 04:03.40 | IsUp | just talk without technical terms, just for 1 minute. can you? |
| 04:03.43 | IsUp | no |
| 04:04.04 | CRCinAU_ | shut up and get off my lawn ;) |
| 04:04.05 | CRCinAU_ | lol |
| 04:06.12 | CRCinAU_ | I'd like to find some nice tunes with overtones of Pachelbels canon.... but my search so far has failed. |
| 04:06.43 | Freeaqingme | you cant make your own? |
| 04:07.01 | CRCinAU_ | I'm not that good ;) |
| 04:07.12 | Freeaqingme | apparently nephews exist who "can do everything with computers" |
| 04:07.32 | CRCinAU_ | hahaha |
| 04:07.43 | CRCinAU_ | see, this isn't too bad: http://www.jamendo.com/en/album/27602 |
| 04:08.43 | Freeaqingme | heh |
| 04:08.51 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:09.43 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-artsfiatbtqrlbct) |
| 04:25.24 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 04:29.28 | thenk | phix: why use creative commons moh, it's horrible |
| 04:31.13 | shapr | Whaat? There's some AWESOME cc music! |
| 04:31.31 | WIMPy | What was it I heard earlier? "Don't buy music, make music!" |
| 04:32.26 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:32.34 | thenk | i must be bias, i work for a moh provider :) |
| 04:33.23 | thenk | this last link CRCinAU[a] is pretty good. |
| 04:33.39 | thenk | telstra should use this instead :P |
| 04:33.39 | shapr | thenk: Oh? Which provider? |
| 04:33.53 | thenk | captivate global / online on hold |
| 04:33.56 | thenk | in australia |
| 04:34.25 | thenk | we just recently changed name to captivate global |
| 04:34.38 | shapr | Cool, I work for Digium, but I don't think we're a competitor. |
| 04:35.12 | thenk | i must admit, i haven't come up against digium yet =D |
| 04:35.36 | thenk | we do have a service which runs on asterisk quite well though |
| 04:47.26 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 04:49.27 | *** join/#asterisk sourcode (~code@ppp-61-90-41-190.revip.asianet.co.th) |
| 04:59.00 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 05:01.24 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 05:04.45 | *** join/#asterisk sourcode (~code@ppp-61-90-13-158.revip.asianet.co.th) |
| 05:07.41 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 05:20.09 | *** join/#asterisk cyborg-one (1000@188.115.189.90) |
| 05:48.21 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 05:50.53 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 05:52.41 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 06:00.33 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 06:01.46 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 06:04.36 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:07.52 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 06:09.40 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:22.15 | *** join/#asterisk jkroon (~jkroon@dsl-241-236-235.telkomadsl.co.za) |
| 06:24.44 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 06:26.04 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:33.51 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 06:39.39 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 06:40.35 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 06:49.50 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 07:02.21 | *** join/#asterisk jblack (~jblack@pool-71-181-219-109.sctnpa.east.verizon.net) |
| 07:15.46 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:15.49 | *** join/#asterisk drdru (~Adium@76.77.182.145) |
| 07:16.04 | drdru | where's a good place to hire an asterisk consultant? |
| 07:29.26 | *** join/#asterisk thenk (cb3b1a74@gateway/web/freenode/ip.203.59.26.116) |
| 07:30.10 | shapr | drdru: Asterisk forums maybe? |
| 07:32.19 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 07:33.09 | thenk | anyone have any luck with app_rtsp? |
| 07:35.41 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 07:36.37 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:37.07 | thenk | ast_log(LOG_DEBUG,"-SETUP VIDEO [%s]\n",url); |
| 07:37.21 | thenk | this doesn't show in my console with 'core set debug 10' - whats wrong? |
| 07:37.24 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 07:41.27 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 07:43.22 | drdru | what's the procedure for integrating CMU Sphinx with Asterisk? |
| 07:50.15 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:01.40 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:02.15 | schmidts | good morning |
| 08:03.23 | shapr | gutentag |
| 08:03.28 | shapr | wie geht es? |
| 08:04.02 | KNERD | sehr gut |
| 08:05.56 | thenk | aye mate |
| 08:06.10 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 08:08.16 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:14.21 | schmidts | damn i had to look twice if iam in asterisk.de or only asterisk :D please dont try to make me confuse early in the morning shapr and KNERD :D |
| 08:14.38 | shapr | haha |
| 08:14.51 | shapr | joins #asterisk.se |
| 08:14.55 | KNERD | morning? it's middle of night ;-) |
| 08:15.19 | kleszcz | hi |
| 08:15.39 | KNERD | guten morgen |
| 08:15.52 | shapr | god morgon, hur mår du idag? |
| 08:16.06 | schmidts | KNERD its 9 am here, but you are right it feels like middle of the night |
| 08:16.19 | KNERD | vas ist los |
| 08:16.25 | KNERD | 2 AM here |
| 08:16.32 | shapr | also 2am here |
| 08:16.41 | SiNGLer | 10am here |
| 08:17.04 | thenk | 4:14PM here. |
| 08:17.19 | thenk | waves from west coast of aus |
| 08:17.25 | shapr | The sun never sets on Asterisk installs. |
| 08:18.45 | *** join/#asterisk Tim_Toady (~moi@178.128.4.246) |
| 08:18.48 | KNERD | en realmente, aqui es asterisk.es |
| 08:18.54 | thenk | unless it's an asterisk shaped sun |
| 08:18.55 | schmidts | shapr thats for sure ;) |
| 08:19.06 | thenk | ..then it appears after successful compile:) |
| 08:20.26 | schmidts | thenk and goes down with a segfault :D |
| 08:21.10 | KNERD | Tshuss! |
| 08:21.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:21.41 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:23.23 | *** join/#asterisk SeTTleR (~bernd@p5DDEEA8D.dip.t-dialin.net) |
| 08:25.09 | thenk | sigh, why does music on hold files sound so much better as 16-bit 8khz PCM source |
| 08:25.29 | thenk | I did 8-bit 8khz, and sounded horrible with lots of sibilance on the phone |
| 08:25.37 | thenk | but 16-bit 8khz wav works great |
| 08:25.50 | thenk | maybe there was some problem with the transcode for 8-bit... |
| 08:25.56 | thenk | answers his own question |
| 08:28.48 | schmidts | thenk cause asterisk uses 16bit native without transcoding and upscaling transcodign allways sounds bad ;) |
| 08:30.35 | thenk | ahh! |
| 08:30.52 | thenk | okay i was about to pull my hair out wondering why my ulaw source file sounded bad when using ulaw on the phone |
| 08:31.21 | thenk | so it's 16-bit native in asterisk, converts to ulaw for each SIP channel? |
| 08:31.39 | thenk | perhaps running the mpg123 player using a 16-bit output would sound better then! |
| 08:33.13 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qpvheblsgtcfkqoz) |
| 08:37.21 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:cc7a:9aaf:d468:e722) |
| 08:38.48 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:39.07 | shapr | I speak native asterisk. |
| 08:40.20 | thenk | ** *** ? |
| 08:40.24 | shapr | snickers |
| 08:41.00 | *** join/#asterisk karthik (~chatzilla@122.183.81.194) |
| 08:49.55 | *** part/#asterisk drdru (~Adium@76.77.182.145) |
| 08:52.07 | schmidts | thenk its native for alaw or ulaw, asterisk can use it for these codecs without any transcoding |
| 09:09.11 | *** join/#asterisk Sheeplet (~multivac@41-133-149-75.dsl.mweb.co.za) |
| 09:09.18 | Sheeplet | lo all |
| 09:27.28 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 09:42.10 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 09:46.18 | coppice | 8-bit 8kHz recordings sound OK if the volume is fairly constant. when the volume changes a lot they sound terrible |
| 09:50.09 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:15.24 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:21.51 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 10:23.37 | *** join/#asterisk defswork (~andy@mx1.3gcomms.co.uk) |
| 10:31.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:31.54 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 10:58.41 | *** join/#asterisk coppice (~chatzilla@m121-203-229-211.smartone-vodafone.com) |
| 11:20.18 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 11:22.06 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 11:39.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:48.12 | *** join/#asterisk jkroon (~jkroon@dsl-241-236-235.telkomadsl.co.za) |
| 11:49.24 | jkroon | hi all, I've got an FXO port (fxsks signalling), as soon as the phone is picked up and the first digit is pressed asterisk immediately enters the dialplan without waiting for more digits - what setting do I need to change to make it wait for more digits before going into the dialplan? |
| 11:50.44 | jkroon | signalling is FXO Kewlstart |
| 11:51.30 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 11:53.56 | shapr | jkroon: SIP phone talking to Asterisk? |
| 11:55.27 | jkroon | shapr, no, TDM400 card with two FXO modules on it, FXSKS signalling on the wire. |
| 11:55.55 | jkroon | shapr, otherwise i'd just annoy the manufacturer of said phone :) |
| 11:56.12 | shapr | I'm confused... asterisk can dial out to the rest of the PSTN from FXO ports... |
| 11:56.12 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 11:56.26 | shapr | but what phone is being picked up and dialed that hits the dialplan? |
| 11:56.47 | jkroon | sorry, FXS port ... |
| 11:56.53 | jkroon | FXOKS signalling. |
| 11:57.09 | shapr | You have two FXS ports? green daughterboards on your TDM400? |
| 11:57.19 | jkroon | yes. two red ones too. |
| 11:57.26 | shapr | Jag är lite förvirrad... |
| 11:57.36 | jkroon | just a little :p |
| 11:57.42 | shapr | Heh, talar du svenska? |
| 11:57.48 | jkroon | not that good no. |
| 11:58.01 | shapr | It's a bit like Afrikaans? |
| 11:58.21 | jkroon | ever so slightly yes. |
| 11:58.27 | shapr | Er, were this a SIP phone, I'd say it's your digit map... |
| 11:58.43 | jkroon | some similarities between most of the germanian derived languages. |
| 11:58.58 | shapr | Yah, German is much easier to understand after learning Swedish and a bit of Dutch. |
| 11:59.33 | shapr | I'd like to visit Zuid Afrika someday... |
| 11:59.39 | shapr | but anyways |
| 12:01.01 | shapr | jkroon: dialplan show ? |
| 12:01.12 | shapr | shakes brain |
| 12:01.17 | adeel | anyone else experience this problem with * 1.4 & zapata/dahdi, where after a long period of time, there'll randomly be noise on a line, but if i unload the zaptel modules and restart asterisk, the noise seems to be gone...and i'm checking this via ztmonitor |
| 12:01.48 | shapr | jkroon: "dialplan show 2@default" or whatever is your default context? |
| 12:02.05 | shapr | adeel: What hardware? |
| 12:02.36 | shapr | jkroon: brain tired guess, maybe you have a match that consumes single digits? |
| 12:02.47 | adeel | shapr, iirc, it's an OpenVox card; not the best i know; but i wasn't around when it was built, and they don't have the budget for a better one |
| 12:04.11 | shapr | Oh, I know nothing about OpenVox cards. |
| 12:04.33 | shapr | adeel: fxotune? |
| 12:06.14 | *** join/#asterisk zooz (~zooz@host86-164-222-248.range86-164.btcentralplus.com) |
| 12:06.21 | zooz | hi people |
| 12:06.34 | zooz | I am trying to set up a MeetMe() application |
| 12:07.01 | zooz | but I get the following error: |
| 12:07.04 | zooz | [Mar 9 12:03:05] WARNING[11687]: app_meetme.c:1123 build_conf: Unable to open DAHDI pseudo device |
| 12:08.09 | zooz | what is dahdi and how is it related to asterisk? |
| 12:09.17 | shapr | It's the hardware drivers, but in this case, I suspect it's needed for timing. |
| 12:09.59 | shapr | iirc, MeetMe before Asterisk 1.8 or so requires external timing so it can mix the audio streams |
| 12:10.24 | zooz | right.. is there a workaround this? |
| 12:10.29 | shapr | install dahdi? |
| 12:10.47 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 12:10.50 | zooz | I did |
| 12:11.01 | shapr | er, did you build asterisk after installing dahdi? |
| 12:11.13 | zooz | I installed asterisk from sources |
| 12:11.38 | shapr | As long as you built asterisk from source after building and installing dahdi from source, asterisk will have access to the install dahdi |
| 12:11.48 | shapr | at that point, you need to start dahdi so the kernel module is present |
| 12:12.00 | shapr | as in: /etc/init.d/dahdi start |
| 12:12.04 | shapr | and then you can start asterisk |
| 12:12.46 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 12:13.04 | shapr | and "dahdi show channels" in asterisk should show the pseudo channel |
| 12:13.20 | zooz | http://p.defau.lt/?wXFvAUcMgQCckmJ6o9xjZg |
| 12:13.30 | shapr | zooz: dahdi not installed |
| 12:13.31 | zooz | that's what I get when I do: service dahdi start |
| 12:13.44 | shapr | perhaps you've done make, but not make install? |
| 12:14.06 | zooz | I installed everything from my distro package system |
| 12:14.17 | shapr | What distro? |
| 12:14.24 | zooz | http://p.defau.lt/?PGiR5_gq99bJn7AqYVf89A |
| 12:14.28 | zooz | fedora14 |
| 12:14.40 | shapr | zooz: look for dahdi-kmod or something? |
| 12:14.49 | shapr | I'm a debian/ubuntu user myself. |
| 12:15.01 | shapr | so I don't remember the name of the dahdi kernel modules. |
| 12:15.18 | shapr | zooz: in any case, I know the most recent version of asterisk does not require DAHDI to get timing from the kernel. |
| 12:15.28 | shapr | Though I don't remember the details off the top of my head. |
| 12:15.30 | jkroon | shapr, i'll pastebin the snippet that I get from the CLI: http://pastebin.com/5CXFb7Kz |
| 12:15.48 | jkroon | basically immediately when the user hits the zero it goes there - it doesn't wait at all. |
| 12:15.52 | zooz | shapr, right, that is why I do not have any issues with asterisk 1.8 |
| 12:16.46 | shapr | jkroon: once again, my brain-tired guess is that you have a dialplan match that's too eager, and is matching too few numbers |
| 12:17.05 | shapr | zooz: So, meetme works on asterisk 1.8? |
| 12:17.13 | zooz | shapr, yeah it does |
| 12:17.25 | shapr | Oh, so you just wanted to know what DAHDI does for you? |
| 12:18.21 | zooz | well, I have asterisk 1.6 somewhere else |
| 12:18.36 | zooz | tried to setup meetme |
| 12:18.40 | jkroon | shapr, yea, but this time it's part of the ast config - not on the phone. why does it immediately try to enter the dialplan when a single digit is pressed on the phone? shouldn't it wait for a 3 second timeout or something? |
| 12:19.09 | shapr | jkroon: I don't remember :-/ |
| 12:19.31 | jkroon | kk, google isn't helping either. |
| 12:20.34 | *** join/#asterisk cerberus_za (~coert@dsl-185-103-86.dynamic.wa.co.za) |
| 12:21.38 | zooz | shapr, looks like I am going to have to build dahdi modules myself |
| 12:22.38 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 12:23.29 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 12:31.50 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 12:38.28 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 12:48.53 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 12:58.36 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 13:07.24 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:10.28 | *** join/#asterisk lathama (c818e064@gateway/web/freenode/ip.200.24.224.100) |
| 13:18.37 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 13:20.54 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-mbsbgyfulfeplszx) |
| 13:21.00 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:26.45 | *** join/#asterisk leafartn (~chatzilla@190.145.253.51) |
| 13:30.49 | leafartn | good morning |
| 13:33.22 | leafartn | Yesterday I asked about a parameter in queues: Service Level ...in queues.conf I can change this parameter, but where can I see the statistics of my queue? Thanks! |
| 13:50.29 | *** join/#asterisk Zathraz (~Zzz@a83-163-198-186.adsl.xs4all.nl) |
| 13:51.02 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 13:51.22 | *** join/#asterisk sourcode (~code@ppp-61-90-13-158.revip.asianet.co.th) |
| 13:52.30 | jkroon | leafartn, queues show |
| 13:53.16 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:54.37 | plundra | Anyone doing provisioning with SPAxxx-phones? Is 0 as Resync_Periodic a valid value for "never"? Or what effect will it have? |
| 13:55.47 | plundra | Oh, it is. Pardon me for not remembering I have docs for it :-) |
| 14:00.14 | *** join/#asterisk serafie (~erin@nat/digium/x-urwpexhzbavqyqkz) |
| 14:13.19 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 14:13.40 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 14:13.51 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 14:18.00 | *** join/#asterisk skrusty (~ben@83.166.169.221) |
| 14:18.15 | skrusty | afternoon all |
| 14:18.51 | *** join/#asterisk Gent (~gent@126.225-93-216-fuji-dsl.dhcp.surewest.net) |
| 14:19.02 | Gent | Is it possible to route my google voice number to my asterisk server? |
| 14:19.43 | leifmadsen | Gent: yes |
| 14:20.09 | skrusty | spotted on engadget today you can now make SIP calls through googlevoice |
| 14:20.47 | Gent | leifmadsen: do I have to fake it being "gizmo" ? |
| 14:20.50 | *** join/#asterisk Zathraz (~Zzz@a83-163-198-186.adsl.xs4all.nl) |
| 14:21.09 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 14:22.29 | Gent | or perhaps I'm missing something |
| 14:22.55 | skrusty | ah /. even, not engadget: http://tech.slashdot.org/story/11/03/08/1955200/Google-Voice-Discovered-Allowing-Pure-VoIP-Calls |
| 14:23.10 | lathama | skrusty: you can make sip calls to your +countrycodeandDID@sip.voice.google.com |
| 14:23.31 | skrusty | ah ok, i've never used it, so i guess i was just echoing what i had seen :) |
| 14:24.05 | leifmadsen | skrusty: pretty sure that was already turned off |
| 14:24.12 | Gent | hrm... I don't want to make sip calls to it... but people who call it to SIP to my asterisk server |
| 14:24.26 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-gadfodrqigwmryox) |
| 14:24.54 | skrusty | yeah, i didn't read /. correctly this morning :) |
| 14:25.09 | skrusty | i thought it was saying you could connect to a sip server at their end now - ho-hum :) |
| 14:25.15 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:30.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:30.55 | skrusty | does anyone know much about call segments in asterisk? |
| 14:31.53 | skrusty | i.e. tracking segments of a cal |
| 14:31.54 | skrusty | l |
| 14:36.29 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 14:36.40 | leifmadsen | skrusty: sounds like CEL |
| 14:36.47 | leifmadsen | ~cel |
| 14:36.47 | infobot | i guess cel is Channel Event Logging |
| 14:37.08 | skrusty | hmm, yeah, i am using the AMI to build call models |
| 14:37.23 | skrusty | and just trying to find different ways of modelling the data |
| 14:37.24 | leifmadsen | infobot: cel is also http://ofps.oreilly.com/titles/9780596517342/ch24.html#Monitoring_id246945 |
| 14:37.24 | infobot | okay, leifmadsen |
| 14:37.25 | Lord_Rahl | ? does AsteriskNow have the mysql driver? or do i need to recompile it |
| 14:37.31 | leifmadsen | ~cel |
| 14:37.31 | infobot | hmm... cel is Channel Event Logging, or http://ofps.oreilly.com/titles/9780596517342/ch24.html#Monitoring_id246945 |
| 14:37.39 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 14:37.56 | leifmadsen | skrusty: or just write data to a DB using func_odbc, or any other method to write data from the dialplan at various steps |
| 14:38.59 | skrusty | hmmm |
| 14:39.59 | skrusty | well damn me... i've never seen that before |
| 14:45.02 | skrusty | what version of * is cel in? or is it an addon? |
| 14:46.36 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 14:46.42 | Amnesia | ugh |
| 14:46.43 | Amnesia | I keep getting peer not found |
| 14:46.47 | Amnesia | [Mar 9 15:44:44] NOTICE[7045]: chan_sip.c:21549 handle_request_register: Registration from '126 <sip:126@194.1.100.214>' failed for '194.1.100.247' - No matching peer found |
| 14:47.08 | skrusty | host=dynamic ? |
| 14:47.20 | Amnesia | I do have: http://pastebin.com/P6gVrEXF though |
| 14:47.30 | Amnesia | skrusty: yes |
| 14:47.50 | Amnesia | ah wt heck |
| 14:47.57 | Amnesia | it does work now I've set the statis ip |
| 14:48.01 | Amnesia | why doesnt dynamic work? |
| 14:49.03 | skrusty | leifmadsen: ah, it's only in 1.8 i take it then? although that's exactly what i want, i need compatability with 1.4 and 1.6; cheers though, good to know about! |
| 14:49.10 | Amnesia | ah it still isnt working |
| 14:49.34 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 14:49.49 | Amnesia | the sip node isnt in the: sip show peers list |
| 14:51.18 | Amnesia | anyone:/? |
| 14:51.32 | Amnesia | http://pastebin.com/4CQsHNSW is my sip.conf |
| 14:53.05 | *** join/#asterisk mocker (~user@206.55.118.83) |
| 14:55.30 | *** join/#asterisk datalay (~datalay@unaffiliated/datalay) |
| 14:57.17 | *** join/#asterisk mihai__ (~mihai__@ool-4352652d.dyn.optonline.net) |
| 14:59.35 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 15:01.25 | *** join/#asterisk bongfrog (~winston@c-67-165-236-171.hsd1.co.comcast.net) |
| 15:02.09 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 15:05.17 | Amnesia | I keep getting: No matching peer found |
| 15:05.17 | Amnesia | :/ |
| 15:07.05 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 15:11.16 | *** part/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 15:14.17 | kaldemar | Amnesia: set host=dynamic and type=friend |
| 15:14.29 | Amnesia | already tried that |
| 15:14.51 | kaldemar | what phone are you using? |
| 15:15.47 | Amnesia | a intercom on a pap2t |
| 15:15.53 | Amnesia | and why does it have to be a "friend? |
| 15:20.22 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:20.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.49 | *** join/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 15:28.40 | *** join/#asterisk v0lksman (~volksman@ottawa-hs-64-26-169-151.s-ip.magma.ca) |
| 15:29.42 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:29.49 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 15:32.09 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:36.29 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 15:36.40 | Romeo- | helllo |
| 15:43.13 | *** join/#asterisk mintee (1000@static-64-115-220-213.isp.broadviewnet.net) |
| 15:43.26 | tzafrir_laptop | Amnesia, do you want to make the intercom your enemy? |
| 15:43.41 | mintee | is there a way to default to a specific extension if the extension in a Goto() statement doesn't actually exist? |
| 15:43.50 | Amnesia | tzafrir_laptop: no:P... |
| 15:44.02 | Amnesia | tzafrir_laptop: is to be able to receive calls only right? |
| 15:44.36 | Romeo- | what about trixbox? |
| 15:45.56 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:49.18 | kaldemar | mintee: functions DIALPLAN_EXISTS and IF will do that for you. |
| 15:51.52 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 15:53.30 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 15:53.55 | *** join/#asterisk gpled (~gpled@69.1.105.42) |
| 15:55.23 | gpled | anyone seen a website where i can find 800 numbers that are not in use |
| 15:56.55 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:58.34 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 15:59.10 | mintee | kaldemar: great, thanks |
| 16:01.04 | mintee | kaldemar: ugh, that's only for <=1.6 |
| 16:01.08 | mintee | i'm on 1.4.27 |
| 16:01.16 | mintee | s/27/37/ |
| 16:03.24 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:03.40 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 16:08.11 | Lord_Rahl | <PROTECTED> |
| 16:10.50 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 16:13.52 | Lord_Rahl | no one knows?? |
| 16:16.02 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 16:20.59 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 16:25.59 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 16:32.49 | *** join/#asterisk jkroon (~jkroon@dsl-241-236-235.telkomadsl.co.za) |
| 16:38.43 | *** part/#asterisk lathama (c818e064@gateway/web/freenode/ip.200.24.224.100) |
| 16:45.15 | *** join/#asterisk mawhii (~mawhii@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 16:51.19 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 17:07.22 | *** join/#asterisk bmg505 (~leon@196-209-44-64.dynamic.isadsl.co.za) |
| 17:08.48 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:15.36 | *** join/#asterisk wesphillips (~wphillips@64.19.240.58) |
| 17:17.26 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 17:18.47 | *** join/#asterisk NoobSaibot (~Adium@24-213-19-74.static.mrqt.mi.charter.com) |
| 17:19.42 | *** join/#asterisk b_d (~brian@mail.bsg.bm) |
| 17:19.58 | b_d | hello all |
| 17:20.08 | NoobSaibot | hi b_d |
| 17:20.26 | b_d | I want to know if it is possible to detect the callee initiating a 3-way call, and act upon it |
| 17:20.27 | b_d | ? |
| 17:21.02 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:21.02 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:21.23 | WIMPy | At least I'm pretty certain there's nothing that would be available to the dialplan. |
| 17:22.59 | b_d | WIMPy, is that directed at me? |
| 17:24.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:27.15 | WIMPy | yes |
| 17:27.25 | Qwell | Somebody buy me a new car. Thanks. |
| 17:28.34 | WIMPy | For the desk or for the sand box? |
| 17:30.22 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 17:30.25 | b_d | WIMPy: have you heard of any methods? |
| 17:30.40 | tzanger | weird, I was logged into the channel but nothing was happenning for a few days |
| 17:30.42 | tzanger | just noticed it now |
| 17:31.08 | n3hxs | Operating in stealth mode. |
| 17:31.10 | WIMPy | b_d: AFAIK the only way to get the information in the first place is via ISDN. |
| 17:33.04 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 17:33.23 | WIMPy | Or H.323, probably. |
| 17:34.24 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 17:47.16 | b_d | WIMPy: ok, currently there is only a SIP trunk terminating on the box |
| 17:48.15 | *** part/#asterisk gpled (~gpled@69.1.105.42) |
| 17:52.13 | Lord_Rahl | anyone know a good how for install mysql cdr into asteriskNow 1.6 |
| 17:52.17 | Lord_Rahl | y |
| 17:52.32 | *** join/#asterisk citrus2 (~chatzilla@sb1.serviceobjects.com) |
| 17:52.54 | citrus2 | hey all, is there any hope for me to get a BRI card working in the USA with Asterisk? |
| 17:53.50 | WIMPy | citrus2: If you can find such a line there, the rest should be relatively easy. :-) |
| 17:54.12 | citrus2 | i can, however from what i am finding is that BRI work diferently in the US |
| 17:54.59 | WIMPy | (almost) everthing is different in North America, but dahdi has support for american protocols. |
| 17:56.14 | citrus2 | <PROTECTED> |
| 17:56.58 | WIMPy | I think I need to take a look at Canada :-) |
| 17:57.04 | *** join/#asterisk fauxalliance (~fauxallia@142.163.152.236) |
| 17:57.32 | WIMPy | You mean the physical connection? |
| 17:58.20 | citrus2 | no |
| 17:58.35 | citrus2 | i assume it still some kinda cat5 with rj45 end |
| 17:58.39 | citrus2 | its the signaling |
| 17:59.29 | WIMPy | Take a look at dahdi. It supports NI-1, NI-2, 4ESS, 5ESS and DMS-100 from what I gathered. |
| 18:00.14 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 18:00.39 | WIMPy | TFor the physical part, I heard that you usually only the a U interface in the US, whereas the cards have S/T interfaces. |
| 18:00.58 | WIMPy | +get |
| 18:00.59 | citrus2 | yeah u or uloop |
| 18:01.02 | citrus2 | or something liket |
| 18:01.07 | citrus2 | thats the problem it seems |
| 18:01.12 | citrus2 | with the hardware manufactuers |
| 18:01.14 | citrus2 | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg218010.html |
| 18:01.22 | WIMPy | You need an NT for that. |
| 18:01.39 | citrus2 | it seems like no one had made a PC BRI card that works with US |
| 18:02.34 | WIMPy | You just need to find an NT to convert fom U to S/T. |
| 18:03.31 | WIMPy | Maybe you should call Digium and ask them. |
| 18:04.05 | WIMPy | It would seem logical that they tested their cards on a local network. |
| 18:05.17 | citrus2 | yeah i can give them a call |
| 18:05.20 | citrus2 | not a bad idea |
| 18:06.01 | tzafrir_laptop | citrus2, what's the difference? |
| 18:06.27 | tzafrir_laptop | oh |
| 18:06.52 | WIMPy | Difference? |
| 18:07.12 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 18:07.13 | tzafrir_laptop | I mean: what's stopping you from using any standard BRI adapetr |
| 18:07.14 | tzafrir_laptop | ? |
| 18:07.34 | WIMPy | The missing NT. |
| 18:07.50 | WIMPy | But that must be available somewhere. |
| 18:08.10 | *** join/#asterisk PsuedoICE (~PsuedoICE@flanderscorp.com) |
| 18:08.31 | citrus2 | just spoke with digium |
| 18:08.38 | citrus2 | said "Not gunna happen" |
| 18:08.49 | citrus2 | holy maybe if i can get a bri to sip gateway device |
| 18:08.49 | LemensTS | On incoming sip DID's that do not use a user/pword auth, is it possible to have a register string so I can monitor the SIP voip provider? |
| 18:09.08 | WIMPy | Woot? |
| 18:10.19 | PsuedoICE | I have a few questions about Asterisk. I am looking at purchasing a Switchvox appliance, but I do not understand how the viop service integrates? What should I purchase from the SIP provider? Which SIP provider to use? I want to have about 10 channels and 6 did's. |
| 18:10.29 | WIMPy | citrus2: For what reason? |
| 18:10.31 | citrus2 | tzafrir_laptop: bri is cabeling or signaling is different here, U-loop instead of S/T only country thats different and there are VERY few bri providers. so no one wants to make a card that works |
| 18:10.44 | Qwell | ~itsp |
| 18:10.44 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:10.47 | Qwell | ~itsplist-us |
| 18:10.47 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:10.50 | Qwell | PsuedoICE: ^^ |
| 18:11.22 | WIMPy | citrus2: Cisco does :-) |
| 18:11.23 | tzanger | ~itsplist-ca |
| 18:11.23 | infobot | Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
| 18:11.54 | citrus2 | they make a BRI card that works in the US with asterisk? |
| 18:12.03 | WIMPy | citrus2: Anyway: The physical part must be managable. Maybe you ask your telco for a source for an NT? |
| 18:12.09 | *** join/#asterisk gpled (~gpled@69.1.105.42) |
| 18:12.29 | WIMPy | citrus2: No, they work in their equipment. |
| 18:12.44 | citrus2 | ? |
| 18:13.01 | gpled | hey |
| 18:13.14 | gpled | aidinb: hello |
| 18:13.15 | WIMPy | You can get BRI U interfaces for Routers.# |
| 18:13.50 | n3hxs | Thanks Qwell, I was trying to remember the name of flowroute yesterday. Figured it would pop up sooner or later. |
| 18:14.31 | tzafrir_laptop | citrus2, I have very few confirmed success stories in this case, |
| 18:14.54 | tzafrir_laptop | but all of them seem to involve a Cologne HFC-4S-based card |
| 18:15.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:15.34 | WIMPy | The type of cards shouldn't matter much, unless it needs CAPI. |
| 18:15.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:16.24 | WIMPy | But you need to use dahdi or if it fits chan_capi. The others are DSS-1 only. |
| 18:16.29 | tzafrir_laptop | It means that practically all cards should |
| 18:16.48 | WIMPy | Als the usual ones, yes. |
| 18:17.15 | WIMPy | All |
| 18:17.25 | tzafrir_laptop | (Our device and the new module of the Digium card use XFHC, which is very similar. Lower end single-port cards use a simple HFC-S) |
| 18:19.49 | WIMPy | tzafrir_laptop: So do you have any pointers for getting an NT? |
| 18:21.41 | tzafrir_laptop | Well, ahem, it should be trivial, I guess... |
| 18:22.00 | tzafrir_laptop | is a software guy. Wiring is a hardware issue |
| 18:22.09 | WIMPy | Looks like 2B1Q might be used in Europe as well somewhere. |
| 18:22.33 | WIMPy | We only use 4B3T here. |
| 18:23.38 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:25.54 | WIMPy | citrus2: http://www.1pcn.com/alpha/NT1.htm |
| 18:26.07 | WIMPy | Something like that. |
| 18:26.24 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 18:29.58 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.221.20.107) |
| 18:30.02 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.221.20.107) |
| 18:33.27 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:36.41 | WIMPy | And Canada does seem to be just like the US and not the rest of the world. |
| 18:39.29 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 18:41.03 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:52.19 | *** join/#asterisk wesphillips (~wphillips@64.19.240.58) |
| 18:52.51 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 18:58.13 | Lantizia | Lo I an old 90's PC with only a dial up modem... so I'm thinking it can dial via an FXS SIP adapter (like a Linksys SPA3102) and have something register as an asterisk peer that can be the ISP (i.e. hand out an IP etc) sound plausable? |
| 18:59.24 | Lantizia | actually I mean a FXO adapter like a PAP2T |
| 19:01.33 | Lantizia | no I mean FXS on a PAP2T... damn I never get my FXS vs FXO right |
| 19:02.03 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 19:04.49 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 19:06.03 | _Corey_ | Lantizia: It's not clear, but if you mean to ask whether you will be able to use a data modem to dial via a PAP2, the answer is probably no |
| 19:06.43 | _Corey_ | Unless you're going to be OK with ~9600 baud |
| 19:17.09 | *** join/#asterisk awclin (~alinford@85.210.33.211) |
| 19:22.23 | *** join/#asterisk sahafeez (~sahafeez@4.53.128.211) |
| 19:24.59 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 19:26.16 | *** join/#asterisk distributed (~root@66.241.104.83) |
| 19:27.02 | *** join/#asterisk DennisG (~DennisG@541E88D0.cm-5-7c.dynamic.ziggo.nl) |
| 19:27.29 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:34.20 | *** join/#asterisk awclin (~alinford@85.210.33.211) |
| 19:36.53 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 19:40.11 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:40.30 | *** join/#asterisk shlm (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 19:40.35 | shlm | hello there |
| 19:41.36 | shlm | what is a dial plan, dial pattern really can't understand what is it ? |
| 19:42.25 | pabelanger | ~newbook |
| 19:42.25 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 19:42.43 | pabelanger | shlm: ^ Should give you a better understanding |
| 19:44.01 | *** join/#asterisk _zoom_ (~zero@41.218.60.19) |
| 19:44.02 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 19:44.05 | _zoom_ | hello |
| 19:44.19 | _zoom_ | any idea how to overcome pdd problem? |
| 19:44.46 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 19:47.19 | shlm | well i got asterisk |
| 19:47.34 | shlm | with one trunk |
| 19:48.12 | *** part/#asterisk NoobSaibot (~Adium@24-213-19-74.static.mrqt.mi.charter.com) |
| 19:49.47 | shlm | i m in fr, to make an national call from my pbx i have to dial 00331XYZ..... instead of dialling 01XYZ |
| 19:50.59 | tzafrir_laptop | _zoom_, what do you mean? |
| 19:51.13 | shlm | what i have to do on my system to able to dial directly 01XYZ instead of dialling 00331XYZ |
| 19:51.43 | tzafrir_laptop | shlm, hmm... you can always manipulate the called number in the dialplan |
| 19:51.46 | shlm | and to use the international dialling code for the rest of the coutry ? |
| 19:52.03 | shlm | which mean ? |
| 19:52.20 | _zoom_ | tzafrir_laptop: post dial delaay |
| 19:53.23 | shlm | i m really sorry don't get your answer tzafrir_laptop |
| 19:53.27 | _zoom_ | Can I play ringback while dial cmd trying to get line connected |
| 19:53.28 | _zoom_ | ? |
| 19:56.41 | shlm | anyone there ? |
| 19:59.07 | DennisG | yes |
| 19:59.21 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
| 19:59.52 | DennisG | just use 01XYZ,1,Dial(SIP/trunk/00331XYZ) for national calls |
| 20:00.28 | DennisG | and 00331XYZ,1,Dial(SIP/trunk/00331XYZ) for international calls |
| 20:00.44 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 20:00.48 | DennisG | or just like: 00XXXYZ,1,Dial(SIP/trunk/00XXXYZ) for national calls |
| 20:01.02 | DennisG | or internal calls.. just what you want :) |
| 20:01.14 | DennisG | you have to split the national and international things ;) |
| 20:02.00 | shlm | is that an issue related with dialplan or dial pattern ? |
| 20:02.30 | tzafrir_laptop | shlm, why would you need 00 for national calls? |
| 20:02.40 | DennisG | it's your answer for the following question: what i have to do on my system to able to dial directly 01XYZ instead of dialling 00331XYZ |
| 20:02.42 | shlm | i never said that |
| 20:03.02 | shlm | my question is : how to avoid dialling 00331XYZ for national |
| 20:03.05 | shlm | call |
| 20:03.19 | shlm | that 's all simply ? |
| 20:03.55 | shlm | to able make directly 01XYZ or 02XYZ or 03XYZ etc .... |
| 20:04.36 | DennisG | that's what i said |
| 20:04.45 | tzanger | hmm |
| 20:04.55 | tzanger | got a question about asterisk realtime (if there is still such a thing) |
| 20:05.06 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 20:05.08 | tzanger | does anyone use a realtime (database-driven) sip registry these days? |
| 20:05.26 | DennisG | shlm: 0XXX.,1,Dial(SIP/trunk/0${EXTEN}) |
| 20:05.38 | _Corey_ | tzanger: Yeah, it still works... :) |
| 20:05.40 | DennisG | yeah me tzanger haha |
| 20:06.04 | shlm | where i have to put that code ? DennisG |
| 20:06.22 | tzafrir_laptop | shlm, what do you have in your chan_dahdi.conf ? Specifically: 'pridialplan' and the variaous '*prefix' (nationalprefix and such) |
| 20:06.23 | tzanger | how does the backend db support work? is it native mysql/postgres/whatever or is it odbc? |
| 20:06.53 | shlm | i just only sip trunk |
| 20:07.02 | _Corey_ | tzanger: Well, MySQL support is direct if you use add-ons. Not sure about postgres but everything else works with ODBC |
| 20:07.35 | tzanger | _Corey_: ok, I see, thank you |
| 20:07.50 | shlm | i m just newbie where is the dialplan file ? |
| 20:08.01 | DennisG | postgres works great with the odbc function :) |
| 20:08.07 | tzanger | what about registry "injectors" (processes to create realtime entries int the db) ? are there skeleton apps to do that or are they generally completely custom? |
| 20:09.17 | *** join/#asterisk edersohe (~edersohe@201.161.16.236) |
| 20:09.24 | DennisG | i think they are custom because the configuration and function depends on the processes of the company |
| 20:10.03 | DennisG | but you can take a look at some billing platforms, some use the realtime feature |
| 20:11.51 | *** join/#asterisk JonathanRose (~JonathanR@nat/digium/x-rqzurhyfywqumhwr) |
| 20:11.54 | *** part/#asterisk edersohe (~edersohe@201.161.16.236) |
| 20:12.05 | *** join/#asterisk edersohe (~edersohe@201.161.16.236) |
| 20:12.08 | _Corey_ | tzanger: Not sure what you mean... Asterisk handles all the interaction with the DB when devicxes register |
| 20:12.37 | tzanger | _Corey_: I mean something has to put devices into the DB so that when they try to place a call asterisk knows they're legit |
| 20:13.15 | shlm | wat's up ? |
| 20:13.29 | shlm | <PROTECTED> |
| 20:13.58 | tzanger | hmm, I wonder if sqlite3 has an odbc connector. :-) |
| 20:14.22 | tzanger | (looking at a fixed embedded application that has a need for a realtime sip registry) |
| 20:14.37 | tzanger | sqlite is not a fan of concurrent accesses though |
| 20:15.12 | _Corey_ | tzanger: Well, what I mean to say is the devices register to Asterisk and it automatically puts their entries into the appropriate table, i.e. 'sip_users' |
| 20:16.31 | tzanger | asterisk wouldn't allow the registry to succeed if they aren't already in the table |
| 20:16.51 | tzanger | asterisk isn't adding entries to the db; asterisk would be checking against the db to allow a registration, would it not? |
| 20:16.56 | wdoekes2 | _Corey_: it puts the ip and the port there yes (and fullcontact, if you use rtcache) |
| 20:17.02 | wdoekes2 | indeed tzanger |
| 20:17.28 | *** join/#asterisk a_m_y (~IceChat7@121.96.33.130) |
| 20:17.32 | tzanger | hmm... I might just be crazy enough to try sqlite3 sip realtime... |
| 20:17.32 | _Corey_ | tzanger: Ok, I see what you're getting at... adding the accounts in the first place :) |
| 20:17.46 | _Corey_ | Sorry, I got hung up on "register", as it normally means a SIP REGISTER |
| 20:18.31 | wdoekes2 | can't you add static peers through ami? |
| 20:18.40 | wdoekes2 | (don't know.. just throwing it out there) |
| 20:18.47 | _Corey_ | Not that I know of |
| 20:18.58 | _Corey_ | You're better off inserting them into MySQL externally |
| 20:19.10 | wdoekes2 | yes.. but tzanger doesn't want big-sql |
| 20:19.16 | a_m_y | do queues.conf setting have ring one or twice for using auto answer or auto accept in softphone? |
| 20:19.47 | tzanger | wdoekes2: that's certainly an option as well. I'm still in the planning stages. realtime seems a bit overkill for sure |
| 20:20.10 | tzanger | I need to get some information about just how "busy" this application is |
| 20:20.13 | _Corey_ | Most people using realtime are doing so because they have LOTS of devices and BIG dial plans |
| 20:20.39 | tzanger | I don't think it will be very busy at all... a little bursty maybe (a couple dozen new device registrations all at once, then one or two every few minutes thereafter) |
| 20:21.08 | tzanger | the dialplan will be fairly large and will have lots of db queries (dynamic routing of calls) but the actual sip registry I expect to be fairly static |
| 20:21.09 | _Corey_ | By "registrations" you mean new device creations? |
| 20:21.14 | tzanger | _Corey_: correct |
| 20:21.28 | JerJer | i really see no reason to use realtime, but that's me |
| 20:21.31 | tzanger | imagine a train with voip service |
| 20:21.36 | _Corey_ | Yeah, devices also "register" periodically with Asterisk, so I would avoid that term :) |
| 20:21.42 | tzanger | JerJer: I know, I'm not a fan either |
| 20:22.03 | *** join/#asterisk awclin (~alinford@85.210.33.211) |
| 20:22.12 | _Corey_ | It sounds like realtime would be a fit if you're adding that many devices |
| 20:22.13 | tzanger | _Corey_: yeah :-) |
| 20:22.45 | _Corey_ | It's just one INSERT statement per device creation, so you could build that into any lightweight script |
| 20:23.13 | tzanger | _Corey_: it's bursty though... you get on the train and a bunch of devices get on the wifi and become "active" -- the sip registry would be busy for the first 5 minutes, and then very sporatically busy as random people pick up the service |
| 20:23.29 | tzanger | static SIP via AMI shoudl be able to handle that fairly well |
| 20:24.23 | _Corey_ | True enough, though applications like MySQL are engineered to handle hundreds of transactions per second |
| 20:24.29 | _Corey_ | I think AMI will be a bottleneck |
| 20:24.55 | JerJer | why add devices at all to asterisk? use kamailio for the registrar |
| 20:24.56 | _Corey_ | You're putting no load on Asterisk with the ODBC/MySQL part of things |
| 20:26.36 | tzanger | JerJer: kamailio is overkill for this application |
| 20:26.45 | tzanger | severe overkill, which realtime may also be |
| 20:29.44 | *** part/#asterisk edersohe (~edersohe@201.161.16.236) |
| 20:31.53 | *** join/#asterisk rene- (~rene@187.136.62.232) |
| 20:32.11 | rene- | ~seen |
| 20:32.40 | JerJer | kamailio can be made very tiny |
| 20:32.49 | JerJer | more specific sip router |
| 20:33.13 | JerJer | +ly |
| 20:34.11 | JerJer | use just the state less, transaction and registrar modules - store everything in memcached |
| 20:40.15 | *** join/#asterisk blackxored (~user@ubuntu/member/blackxored) |
| 20:40.45 | blackxored | hey, hello everyone, can someone tell me where are the defaults asterisk extensions I can use for testing my setup???? Like an echo extension or a voicemail one??? |
| 20:42.44 | a_m_y | do queues.conf setting have ring one or twice for using auto answer or auto accept in softphone? |
| 20:43.19 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 20:44.19 | cobra2 | blackxored: in the asterisk console try 'dialplan show' |
| 20:44.43 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:46.33 | tzanger | JerJer: I find your ideas intriguing and I would like to subscribe to your newsletter. |
| 20:47.05 | leifmadsen | blackxored: no such thing exists -- you have to build your dialplan. You can do "make samples" from the asterisk source to get a set of files which are better as documentation than they are for doing much of anything. |
| 21:00.38 | *** join/#asterisk three60 (~RavishinR@c-68-80-110-182.hsd1.pa.comcast.net) |
| 21:01.37 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-037.mobile.uci.edu) |
| 21:08.48 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:20.20 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 21:25.03 | *** part/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 21:25.59 | *** join/#asterisk [1]bond (~bond@151.68.46.170) |
| 21:26.28 | [1]bond | Hi! |
| 21:27.12 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:31.11 | *** part/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 21:33.25 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 21:34.18 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 21:42.46 | three60 | Hi all. My setup is : astersisk server and SIP/Sam at location A, SIP/Owen at location B. Both phones can make and receive calls, but SIP/Owen has no audio. I don't see any errors from CLI. Here's the last few lines from the cli output and sip.conf (http://pastie.org/1653170). Very new to this, so not up on all the terms and the like, so please try to bear with my thickness. |
| 21:44.15 | *** join/#asterisk remyuk (5c18d961@gateway/web/freenode/ip.92.24.217.97) |
| 21:48.14 | remyuk | hello there ! What causes this problem? "Your call cannot be completed as dialed". thanks. |
| 21:49.59 | three60 | dialed the wrong number maybe |
| 21:50.19 | three60 | couldnt help it, sorry |
| 21:50.41 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:52.31 | remyuk | due to dial rules, maybe ? |
| 21:58.50 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 22:06.58 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:07.48 | *** part/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 22:39.36 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 22:45.17 | *** join/#asterisk lathama (c818e064@gateway/web/freenode/ip.200.24.224.100) |
| 22:46.44 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 22:50.45 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:58.33 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 22:59.13 | pabelanger | ~sipnat |
| 22:59.13 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 23:05.39 | *** join/#asterisk jblack (~jblack@pool-71-181-219-109.sctnpa.east.verizon.net) |
| 23:06.53 | *** part/#asterisk lathama (c818e064@gateway/web/freenode/ip.200.24.224.100) |
| 23:07.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 23:13.01 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 23:21.03 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 23:24.38 | *** join/#asterisk LemensTS (~matthew@adsl-70-130-149-3.dsl.stlsmo.swbell.net) |
| 23:25.48 | LemensTS | If I took a 40-50' Cat5e cable, and used 2 pairs of it for a computer, and 2 pairs of it for an IP phone, should their be any problems with interference in the cable? |
| 23:25.53 | *** join/#asterisk edersohe (~edersohe@201.161.16.236) |
| 23:26.04 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 23:26.15 | edersohe | hi everybody |
| 23:29.35 | a_m_y | ~book |
| 23:29.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 23:32.42 | *** part/#asterisk LemensTS (~matthew@adsl-70-130-149-3.dsl.stlsmo.swbell.net) |
| 23:35.08 | edersohe | im new in the community and i'm want to contribute |
| 23:35.50 | edersohe | the question is |
| 23:36.01 | edersohe | how to sign the license agreement |
| 23:36.14 | edersohe | ?? |
| 23:36.44 | WIMPy | What's the issue with that? |
| 23:38.57 | tzanger | you sign it and fax it in.. |
| 23:41.18 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:41.50 | a_m_y | ~queues |
| 23:48.09 | *** join/#asterisk WWGD (~WWGD@c-98-246-81-120.hsd1.wa.comcast.net) |
| 23:48.39 | *** part/#asterisk a_m_y (~IceChat7@121.96.33.130) |
| 23:48.43 | *** part/#asterisk edersohe (~edersohe@201.161.16.236) |
| 23:55.52 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 23:56.59 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 23:57.49 | *** join/#asterisk CRCinAU_ (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 23:58.38 | CRCinAU_ | russellb: when you get this, can you please review the ban on the nick CRCinAU? kthxbai |