| 00:20.14 | ManxPower | install the "patch" command |
| 00:20.30 | ManxPower | should be part of your distro |
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| 00:28.11 | tzafrir_laptop | Sadly dieno is no longer part of this channel :-( |
| 00:34.59 | WilliamK | evening tzafrir |
| 00:39.27 | MatBoy | [TK]D-Fender: I fixed it all totally, the AGI + dialplans are very nuce |
| 00:39.34 | MatBoy | *nice |
| 00:43.55 | tzafrir_laptop | WilliamK, err, Night? Morning? |
| 00:46.20 | WilliamK | it's 7:48pm here :) |
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| 01:47.43 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:57.33 | jeev | [TK]D-Fender you there! |
| 01:59.24 | [TK]D-Fender | diejeeyes |
| 01:59.31 | [TK]D-Fender | kjlsdhfkljjsdf |
| 01:59.34 | [TK]D-Fender | jeev: yes |
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| 02:11.01 | wwalker | ////part |
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| 02:12.01 | x86 | nn: what are you doing with asterisk? |
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| 02:13.54 | nn | run it on my wrt54gs to provide iax to sip for my ATA |
| 02:13.58 | nn | and vmail |
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| 02:57.13 | jtoy | im trying to finsih porting over out altigen system to asterisk, does asterisk support Direct Inward Dial even if i have a trunk, how would I set thast up? |
| 02:59.37 | jaytee | ~book |
| 02:59.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:01.24 | jaytee | jtoy, and the answer to your question is yes |
| 03:01.33 | jtoy | thanks |
| 03:01.51 | jtoy | i realize i asked that pretty ambiguosly |
| 03:01.57 | jaytee | how are the calls coming in? POTS? PRI? ITSP SIP account? |
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| 03:11.19 | jeev | [TK]D-Fender aaaaaaaaaaaaaaaaaah |
| 03:11.51 | jeev | pump, pump the jam, pump it up |
| 03:12.09 | jaytee | mutters "must suck to be so popular" |
| 03:12.37 | jeev | jaytee, if you want to be popular, msg me. |
| 03:12.56 | C4colo | I beleive it is Pump-pup the jam |
| 03:13.07 | jaytee | I don't. I cherish my anonymity |
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| 03:24.31 | jeev | gnn |
| 03:24.37 | jeev | i dont care, pump the jam. |
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| 04:10.30 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 04:10.34 | phix | hello |
| 04:12.21 | phix | Asterisk doesn't seem to want to start, I tried with asterisk -vvvdddcf, but no errors are reported (there were but I disabled the modules that were complaining as I don't use them, res_odbc.so, res_jabber.so, app_amd.so, chan_gtalk.so). I am using asterisk version 1.4.20 (dfsg-1, Debian package). |
| 04:12.58 | De_Mon | why is asterisk telling me I have a user without a mailbox since upgrading from .18 to .21? |
| 04:12.59 | phix | oh and chan_vpb.so, I don't use that either to my knowlesge |
| 04:13.13 | phix | so any ideas? |
| 04:13.39 | De_Mon | more to the point, how do I turn off that warning, the user isn't supposed to have a mailbox |
| 04:14.05 | phix | ok I will sit back and have a rest |
| 04:15.14 | De_Mon | nevermind google finally told me |
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| 04:56.02 | Mike8861 | hello all |
| 04:56.10 | Mike8861 | hey [TK]D-Fender, i got a question |
| 04:57.04 | Mike8861 | my asterisk server will restart sometimes, and it halt during the boot process. any clue ?? |
| 04:57.35 | Mike8861 | it restart by itself. |
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| 05:00.51 | kash | hardware issue |
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| 06:20.55 | Mike8861 | kash: thank you, is it a resources issue ? |
| 06:24.49 | gnorbert | Hi, does somebody know, how can be a wav file played in a meetme conference for all the attandants? |
| 06:26.37 | gnorbert | I use linphone as a softphone, so that is also good, if somebody knows, how to play wav files with linphone. (play command isn't a good solution, it just transfers the sound input) |
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| 06:45.27 | j0 | DISA under trixbox 1 doesn't work with IAX trunks. it works on SIP trunks or in the exact same context when dialing locally, but IAX just hangs up or gives me the error |
| 06:45.54 | j0 | number invalid as soon as i start dialing at the DISA dialtone |
| 06:45.58 | j0 | it makes no sense! |
| 06:46.20 | creativx | trix...box |
| 06:47.24 | j0 | creativx: well what would you reccomend? |
| 06:48.56 | creativx | well problem is you wont get much trixbox help here probably, that was my point.. |
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| 06:55.05 | TJNII | ~trixbox |
| 06:55.06 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 06:55.59 | [netman] | where is the patch to compile zaptel on 2.6.26 kernel? |
| 06:56.25 | creativx | tnx TJNII |
| 06:56.26 | creativx | hehe |
| 06:57.18 | TJNII | Behold the power of the bot |
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| 07:25.03 | gnorbert | Does anybody have an idea, how can be played a wav file in a meetme conference? |
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| 07:41.31 | Jerjer[mobile] | gnorbert: drop a call file |
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| 07:48.01 | implicit | kevin mitnick doing uses asterisk lol at hope 2008 callerid unmasking |
| 07:48.07 | implicit | http://www.youtube.com/watch?v=q3S0RjrXhw0 |
| 07:49.24 | creativx | there is no such think as sip trunk |
| 07:50.31 | implicit | http://www.siptrunk.org/whatissiptrunking.php |
| 07:50.35 | implicit | lol |
| 07:50.36 | implicit | ;) |
| 07:50.42 | implicit | it's not a physical trunk of course |
| 07:50.49 | implicit | but that video is pretty cool anyway |
| 07:51.19 | gnorbert | Jerjer[mobile]: http://pastebin.com/d6dea5eb8 |
| 07:53.18 | gnorbert | Jerjer[mobile]: Or if not this, then I don't know, what did you think of with "call file". |
| 07:54.36 | Jerjer[mobile] | ugh - lines 67 thru 70 are not necessary |
| 07:55.36 | Jerjer[mobile] | drop a call file that connects to a meetme and plays a wav, then hangs up - pretty damn simple if you ask me |
| 07:56.14 | Jerjer[mobile] | and WTF is [userA] ?! that's not an extensions.conf entry |
| 07:56.37 | Jerjer[mobile] | if that is a sip.conf entry - then you should NOT use context=default |
| 07:57.08 | Jerjer[mobile] | call it anything else |
| 07:58.23 | gnorbert | Jerjer[mobile]: Calling it doesn't work, because while it calls in, it gets a delay like 1-2 secs. |
| 07:58.43 | Jerjer[mobile] | obviously you haven't even looked how a call file works |
| 07:59.02 | Jerjer[mobile] | you can call a specific application (MeetMe) and give it an argument (the conference number) |
| 07:59.17 | Jerjer[mobile] | no need to run it thru the dialplan whatsoever |
| 08:00.05 | Jerjer[mobile] | good luck - i'm gonna go try to pass out (i'm stuck at the damn airport until 9:30am edt) |
| 08:00.42 | gnorbert | Jerjer[mobile]: Ok, thanks anyway, I'm gonna try it. |
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| 08:10.10 | zydoon | guys, about chan_unistim |
| 08:10.41 | zydoon | am using it with nortel E1120 |
| 08:10.49 | zydoon | 1120E |
| 08:11.05 | zydoon | I have a small issue |
| 08:11.20 | zydoon | when dialling an number, I don't see what I really type |
| 08:11.24 | zydoon | strange isn't ?? |
| 08:11.31 | zydoon | that's the only phone I have |
| 08:12.02 | zydoon | I mean I don't see the numbers I dial |
| 08:12.06 | zydoon | on the screen |
| 08:12.26 | zydoon | any idea ?? similar experience ?? |
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| 09:23.28 | BrokenNoze | Hi, does anyone know how I can get the Member Status (as in the QueueMemberStatus event) to query in my dialplan? |
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| 09:25.13 | tzafrir_laptop | has had enough of the spam from atcom |
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| 09:27.04 | mandh | does any one use "PortaOne's Radius client for Asterisk" with freeradius |
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| 09:41.16 | Zoup | latest Zaptel svn is not compile able under 2.6.26 : http://rafb.net/p/Qgorni37.html |
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| 09:55.57 | tzafrir_laptop | Zoup, update: latest zaptel in svn can compile under 2.6.26. Please update. |
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| 09:58.48 | Zoup | tzafrir_laptop: can you please take a look at : http://rafb.net/p/UmHIyr20.html , problem exist |
| 09:59.14 | Zoup | tzafrir_laptop: same issue : http://bugs.digium.com/print_bug_page.php?bug_id=13088 |
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| 10:00.14 | tzafrir_laptop | Zoup, after an update? |
| 10:00.27 | Zoup | tzafrir_laptop: yes , take a look at that link |
| 10:00.45 | Zoup | its revision 4425 |
| 10:03.47 | joobie | guys, anyone able to help out with this sip issue? I had a debug log at http://pastebin.com/m2b8e3e0 |
| 10:04.13 | tzafrir_laptop | Zoup, sorry, didn't commit it yet. :-( |
| 10:04.21 | joobie | basically * is a sip client to my sip provider.. i have a phone in my lan connected via sip to * .. when i try to dial out to an external number that needs to go to the sip provider, it shoots an invite error |
| 10:04.34 | Zoup | tzafrir_laptop: np , are you going to do soon ? :) |
| 10:05.40 | tzafrir_laptop | did already |
| 10:05.41 | Zoup | looks like you did :) |
| 10:06.15 | Zoup | compiled without any problem :) you might want to close http://bugs.digium.com/print_bug_page.php?bug_id=13088 |
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| 10:15.35 | implicit | kevin mitnick callerid unmasking with asterisk - http://www.reddit.com/new/ |
| 10:15.50 | many | is he allowed to use computers again? |
| 10:15.57 | implicit | ya |
| 10:16.04 | implicit | pretty crazy |
| 10:16.28 | implicit | click up on it |
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| 10:21.46 | implicit | its in firehose too, lets get asterisk slashdoted again |
| 10:21.47 | implicit | http://slashdot.org/firehose.shtml |
| 10:22.41 | implicit | http://slashdot.org/firehose.pl?op=view&id=782945 |
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| 10:26.12 | hi365 | Could it be that dial ignors country indications? |
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| 10:39.33 | ^shark_ | hey just a quick question -- do i have to have a license for the digum 410 card |
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| 10:41.27 | joobie | guys, when i try to dial a number on my sip phone |
| 10:41.31 | joobie | i keep seeing these nslookups going |
| 10:41.54 | joobie | any idea what this is for? like dns queries everytime i try to dial |
| 10:43.01 | many | enum? |
| 10:43.12 | many | whats the DNS queries asking for? |
| 10:43.32 | ^shark_ | do i have to have a license for the digum 410 card?? |
| 10:44.12 | ^shark_ | many: how does your phone get IP/ |
| 10:44.21 | joobie | A? 1234-09995038.localdomain. |
| 10:44.23 | joobie | is does that |
| 10:44.29 | joobie | A? comvergence-099930e0.localdomain. |
| 10:44.30 | joobie | then that |
| 10:44.31 | joobie | etc |
| 10:44.45 | joobie | like it's looking up the from name / to name sor something |
| 10:47.01 | ^shark_ | joobie: what ip phone is that/ |
| 10:47.52 | many | shido6: hardwired |
| 10:48.16 | many | sounds like its looking up its own hostname for some odd reason |
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| 10:48.29 | many | ^shark_: hardwired |
| 10:48.37 | many | i dont use dhcp if thats your question. |
| 10:49.05 | ^shark_ | joobie: manually set its ip, dns settings |
| 10:49.57 | ^shark_ | then manual settings shld do. |
| 10:51.47 | joobie | ^shark_, it's polycom 320 |
| 10:52.03 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 10:52.15 | joobie | ^shark_, its ip is assigned via dhcp |
| 10:52.17 | joobie | it's setup ok though |
| 10:52.22 | joobie | like it has the ip allocated to it.. |
| 10:52.36 | ^shark_ | you need to set it up manually |
| 10:52.54 | Vec | In asterisk 1.4 does one have to Answer the channel before Queue ? |
| 10:54.56 | joobie | why shark? |
| 10:56.26 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:59.49 | hi365 | is Dial() meant to take indications in to account? |
| 11:01.51 | *** join/#asterisk skirmisha (n=asda@92.66.88.17) |
| 11:01.57 | skirmisha | guys |
| 11:02.13 | skirmisha | why how can i set MOH on remote channel |
| 11:02.41 | skirmisha | or why asterisk set MOH on remote channel, thus i have no control over MOH and what calss to play |
| 11:03.02 | skirmisha | i am trying to set MOH for differrent calls |
| 11:03.25 | skirmisha | but setmusiconhold set this on answered channel |
| 11:03.37 | skirmisha | in most cases this is phone registered with asterisk |
| 11:04.13 | skirmisha | when that user put MOH , asterisk set broadcast moh to remote channle |
| 11:04.20 | skirmisha | channel and not to answered one |
| 11:04.48 | skirmisha | so i am stuck now and trying to sort this out |
| 11:05.01 | skirmisha | any ideas why asterisk set moh on remote channel? |
| 11:06.15 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:07.31 | skirmisha | are u sleeping |
| 11:12.17 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 11:18.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:21.21 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:21.43 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 11:24.02 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:25.05 | joobie | guys |
| 11:25.19 | joobie | need a hand with "failed to authenticate on invite" sip error.. anyone decent here with that sorta stuff? |
| 11:25.58 | joobie | SIP/2.0 401 Unauthorized |
| 11:25.58 | joobie | im getting that response |
| 11:26.32 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 11:26.37 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 11:29.00 | Vec | Sorry, asked a question earlier but got disconnected. |
| 11:29.39 | Vec | Is there a way to use the queue application so it only answers the channel when the agent picks up the phone ? |
| 11:31.31 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:32.14 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 11:35.37 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 11:40.22 | *** join/#asterisk spacular (n=coder@static-host119-30-125-79.link.net.pk) |
| 11:42.24 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 11:43.00 | joobie | w00t, found the issue |
| 11:46.47 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 11:47.44 | ^shark_ | joobie: check the ip phone settings to the server for the authentication error you are getting, it seems as though something you entered in the phone settings is wrong |
| 11:48.03 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:49.31 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 11:54.32 | joobie | shark, it was username= instead of user= |
| 11:54.34 | joobie | works ok now |
| 11:54.38 | joobie | dynamic is fine |
| 11:54.39 | joobie | thanks tho |
| 11:54.55 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:58.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:02.32 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 12:04.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:04.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:07.35 | zydoon | guys am testing the nortel 1120E with chan_unistim |
| 12:07.53 | zydoon | it works almost aout of the box ..... the almost is beacause ... |
| 12:08.04 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:08.09 | zydoon | when I dial a number I never see what I type !! |
| 12:08.14 | zydoon | any idea ? |
| 12:09.04 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 12:09.19 | [TK]D-Fender | zydoon: Where do you think you should be seeing something, and what are you dialing from? |
| 12:09.34 | joobie | guys what does transmit_silence_during_record do if enabled? |
| 12:09.47 | joobie | Fender, fixed that issue from yesterday / day before :) |
| 12:09.52 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 12:09.57 | zydoon | when I type 1 or 2 or any number .... I don't see it in the screen ... as simple and dumb as is :) |
| 12:10.00 | joobie | It was username= i had to specify for the username of the sip provider, rather than user= |
| 12:10.16 | angryuser | joobie it's a confort noise maybe |
| 12:11.10 | joobie | angryuser, "transmit_silence_during_record = yes | no ; send SLINEAR silence while channel is being recorded" |
| 12:11.41 | joobie | if it's not enabled, chanspy() wont work supposedly if it's being Record()'ed |
| 12:11.50 | [TK]D-Fender | zydoon: on what "screen" are you expecting to see this? |
| 12:11.51 | joobie | not sure what it means tho.. SLINEAR silence? |
| 12:12.26 | [TK]D-Fender | zydoon: You need to be at very high debug & verbose to see them in CLI. |
| 12:22.19 | joobie | hey guys |
| 12:23.17 | joobie | if i use Monitor() in my dialplan to track calls going outbound from a handset.. i put it just before the Dial() and it ends up recording even when the other end hasn't picked up the call (when it's still ringing, waiting for the remote end to pick up). Is there a way to do this so it only starts recording when the other end picks up? |
| 12:23.18 | *** join/#asterisk ronr (n=ron@82-204-104-172.fttx.bbeyond.nl) |
| 12:23.27 | joobie | i tried putting it after the dial but that's no good |
| 12:24.44 | ronr | I run an asterisk server with several polycom phones, some of there have 6 SIP channels, is it possible to configure five of these six lines as direct lines to asterisk's parking lot (so activating a line is the same as calling the parking lot number)? |
| 12:24.57 | [TK]D-Fender | joobie: Pastebin a call |
| 12:25.14 | [TK]D-Fender | ronr: Nope./ |
| 12:25.16 | joobie | Fender, with Monitor before or after the Dial? |
| 12:25.36 | [TK]D-Fender | joobie: Before, like you're already doing. |
| 12:25.39 | joobie | okie |
| 12:25.45 | [TK]D-Fender | joobie: and it HAS to be before the dial |
| 12:26.06 | ronr | [TK]D-Fender: thx for saving me a lot of time searching for something that can't be done |
| 12:27.15 | [TK]D-Fender | ronr: You can do [transfer] [blind] [Line-key w/ speed-dial] however |
| 12:27.23 | joobie | Fender, http://pastebin.com/mc17d2 |
| 12:27.27 | joobie | ahh |
| 12:27.28 | [TK]D-Fender | ronr: Just won't be one-touch |
| 12:27.37 | joobie | fender check taht out, there's a dump |
| 12:27.59 | [TK]D-Fender | joobie: And where the hell is "monitor"? |
| 12:28.04 | joobie | i noticed it was recording before the other end picks up purely because whilst it was ringing, the filesize of the wav was going up |
| 12:28.17 | joobie | monitor is in extensions.conf |
| 12:28.21 | joobie | before dial() |
| 12:28.28 | [TK]D-Fender | joobie: Well its sure as hell not being called. |
| 12:28.39 | joobie | it is - check line 134 |
| 12:28.44 | ronr | [TK]D-Fender: yeah, but that wouldn't clearly show the user how many people are waiting, and that's basically what this is about. I think I'll just keep calling to the other lines until they're picked up |
| 12:28.45 | [TK]D-Fender | joobie: and remove the SIP debug, we shouldn't need it |
| 12:29.04 | [TK]D-Fender | ronr: Yes, you can attach presence to those lots as well |
| 12:29.07 | joobie | ok |
| 12:29.25 | *** join/#asterisk zydoon1 (n=zydoon@41.225.155.169) |
| 12:29.33 | ronr | [TK]D-Fender: what do you mean by attach presence to those lots? |
| 12:29.49 | zydoon1 | I was diconnected |
| 12:29.58 | [TK]D-Fender | ronr: I presume you want it lit up if someone is parked there... |
| 12:30.12 | ronr | [TK]D-Fender: yes |
| 12:30.20 | zydoon | any idea about nortel 1120E ? |
| 12:30.44 | [TK]D-Fender | ronr: Read up : |
| 12:30.44 | [TK]D-Fender | ~devstate |
| 12:30.45 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 12:30.46 | joobie | Fender, http://pastebin.com/m90d74ab there is a cleaner log |
| 12:30.50 | joobie | it's definitely being called |
| 12:31.00 | [TK]D-Fender | ronr: And review 1.4s Presence to learn how to track a parking lot |
| 12:31.22 | zydoon | it works fine, but it doesn't print digits I dial ! |
| 12:31.36 | ronr | [TK]D-Fender: thx, I'll read up |
| 12:31.49 | [TK]D-Fender | joobie: Yeah, and monitor has NOT started, and you didn't even bother waiting for the call to get answered |
| 12:32.21 | [TK]D-Fender | zydoon: And I asked you TWICE, where the hell are you expecting to see "digits"? |
| 12:32.39 | zydoon | sorry was disconnected ... shame on me |
| 12:32.59 | zydoon | normally when I type a number I expect to see it in the screen |
| 12:32.59 | [TK]D-Fender | zydoon: Just answer the question. |
| 12:33.10 | joobie | [TK]D-Fender, why does the filesize increase then ? |
| 12:33.13 | [TK]D-Fender | zydoon: what, your PHONE'S screen? |
| 12:33.27 | [TK]D-Fender | joobie: monitor tells you when its recording. |
| 12:33.36 | zydoon | the small LCD screen |
| 12:33.43 | joobie | -rw-r--r-- 1 root root 80684 Jul 22 08:30 myfilename-in.wav |
| 12:33.50 | joobie | that file was created, when i didnt answer. |
| 12:33.51 | [TK]D-Fender | joobie: make sure you're at verbose 10, debug 10 |
| 12:33.52 | zydoon | where I see my line label, my speeddials (bookmarks) ... |
| 12:34.22 | [TK]D-Fender | zydoon: Almost NOBODY uses those phones. Go read your manual |
| 12:34.47 | zydoon | http://www.nortel.com/products/01/succession/es/ip_phones/images/ipp1120e_dv-4147_ver3-sm.jpg |
| 12:34.58 | zydoon | yup, nothing about it in the manual :( |
| 12:35.31 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 12:35.38 | lirakis | hey guys |
| 12:36.17 | lirakis | ive just gotten a payphone and am looking for any resources for converting it to a "standard" handset so i can use it with an ATA |
| 12:36.39 | lirakis | this is i know.. not really the place for this question.. but i thought some one here might know of some resources |
| 12:36.48 | joobie | [TK]D-Fender, verbose and debug were set to 10 in that dump |
| 12:37.11 | lirakis | i spoke with some of the telephreak guys b/c they have done this... but info is spotty |
| 12:37.21 | [TK]D-Fender | joobie: Ok, well I've always seen monitor list its start, and you never answered the call. You are showing broken examples |
| 12:39.20 | ^shark_ | do i need a license for my digium 410 card |
| 12:39.41 | [TK]D-Fender | ^shark_: No. |
| 12:39.52 | ^shark_ | gr8 |
| 12:39.53 | joobie | [TK]D-Fender, i didnt answer the call.. I just downloaded the wav files that are created and listened to them.. it 100% has recorded the call before it is answered. |
| 12:39.54 | joobie | 100%. |
| 12:39.58 | ^shark_ | [TK]D-Fender: thanks |
| 12:40.06 | C4colo | joobie, that is how it works |
| 12:40.11 | [TK]D-Fender | joobie: Show complete samples. |
| 12:40.24 | joobie | C4colo, can i amend it so it only records when the call is answered? |
| 12:40.28 | joobie | or use another function to do this? |
| 12:40.45 | joobie | [TK]D-Fender, you don't need a complete example.. there's enough info there to conclude it is recording before the answer |
| 12:41.04 | joobie | i can publish the wav files if you like fender |
| 12:41.15 | C4colo | just pull the billseconds and call duration from the database, then trim the difference off with sox |
| 12:41.30 | C4colo | set a script and put it in the h priority of the extension |
| 12:41.30 | [TK]D-Fender | joobie: "core show application monitor" <- see if something stands out for you. |
| 12:41.44 | joobie | C4colo, i like the way you think :) |
| 12:41.45 | joobie | thanks mate |
| 12:41.48 | C4colo | lol |
| 12:42.13 | [TK]D-Fender | joobie: Yes, you can, now read the instructions. |
| 12:42.37 | joobie | ok Fender, ill re-read |
| 12:43.54 | *** join/#asterisk af_ (n=getsmart@88-149-241-217.dynamic.ngi.it) |
| 12:44.51 | joobie | <PROTECTED> |
| 12:44.57 | joobie | what does that mean in english?:P |
| 12:45.11 | joobie | does that mean don't record unless it's answered? |
| 12:45.14 | C4colo | I'm guessing that is a flag/option |
| 12:45.40 | [TK]D-Fender | joobie: SMRT |
| 12:45.47 | C4colo | yea basically |
| 12:46.00 | joobie | it is c4colo |
| 12:46.03 | joobie | thanks fender |
| 12:46.04 | joobie | you da man |
| 12:46.14 | C4colo | as long as the call is running along the dialplan by itself it won't record |
| 12:46.33 | joobie | what do you mean c4? |
| 12:46.38 | C4colo | once it bridges with any other channel, answered on the other end, or ends up in an application such as app_voicemail it will begin |
| 12:47.15 | joobie | ahhh |
| 12:47.16 | joobie | i see |
| 12:47.17 | C4colo | bridged = connected |
| 12:47.31 | joobie | ahh |
| 12:47.40 | joobie | all this lingo to learn :) |
| 12:47.48 | C4colo | lots of fun stuff out there, read up on the help and show applications in asterisk |
| 12:48.16 | C4colo | yea, just call a format a codec on accident around [TK]D-Fender sometime |
| 12:48.23 | C4colo | he'll set you straight |
| 12:48.45 | [TK]D-Fender | C4colo: :) |
| 12:48.49 | C4colo | ror |
| 12:48.51 | C4colo | oops |
| 12:48.52 | C4colo | lol |
| 12:49.32 | joobie | ahha :) |
| 12:49.41 | joobie | ok this is much better - one more quirk though |
| 12:49.55 | joobie | it now records only if the other end picks up.. however if the other end hangs up.. it keeps recording until i hangup my handset |
| 12:50.02 | C4colo | mixmonitor |
| 12:50.09 | C4colo | oh |
| 12:50.21 | joobie | ie. bridge is broken (other end hangs up) but monitor stil goes on until i put my handset on the hook |
| 12:50.23 | C4colo | I thought you were going to say "why are there two files, one for each end of the conversation?" |
| 12:50.29 | joobie | hehe naa |
| 12:50.37 | joobie | i read about the solution in monitor() for that |
| 12:50.41 | C4colo | ah |
| 12:50.42 | joobie | natively it supports integrating the two files |
| 12:50.58 | *** join/#asterisk grEvenX (n=even@89.105.43.19) |
| 12:51.04 | joobie | damn im surprised how simple asterisk is.. and how damn flexible it is with the dialplan |
| 12:51.23 | C4colo | ah, hard to keep up with all the updates to applications, I usually just do things the same way until they are removed and everything breaks |
| 12:51.28 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:51.31 | joobie | lol |
| 12:51.49 | joobie | just dont update :P |
| 12:51.53 | C4colo | heh |
| 12:52.02 | joobie | lock the box to your sip provider and game over:P |
| 12:52.40 | joobie | neway ne idea how that works - it's still recording when the bridge is brought down |
| 12:52.46 | C4colo | yea, but then they dangle things like t.38 origination/termination in front of your nose and you spend a week building a 1.6 system to find out that the feature didn't make the branch timeframe |
| 12:52.48 | joobie | it waits to stop recording until i hangup my handset |
| 12:53.15 | joobie | C4colo, that went way over my head |
| 12:53.21 | joobie | too hardcore for an amature like me:P |
| 12:53.30 | C4colo | is it hard drive space you are worried about? |
| 12:53.56 | C4colo | lol, someday you will want to fax on asterisk without having to set your fax machine to 2400 baud and you will want t.38 |
| 12:54.04 | joobie | C4colo, ya.. somewhat.. like if a user doesnt hangup the phone right.. it will record for ages before it's picked up, potentially.. |
| 12:54.09 | C4colo | and you will then install callweaver |
| 12:54.14 | C4colo | for your fax server |
| 12:54.28 | C4colo | ... oh I had such high hopes for 1.6, I really did |
| 12:54.34 | joobie | hehe |
| 12:54.38 | joobie | i heard a bit about fax on voip |
| 12:54.46 | joobie | how the codecs dont support all the squeels it makes |
| 12:54.53 | joobie | presuming t38 supports all those squeels? |
| 12:54.57 | C4colo | why can't the asterisk devs just steal the opensource t.38 code from the callweaver guys? |
| 12:55.07 | zydoon | I want to make a specific extension dials outside using a specific zap channel, is it'feasable ? (Freepbx) |
| 12:55.21 | C4colo | it converts those signals into TCP/IP packets and then regenerates them on the other end |
| 12:55.55 | C4colo | instead of using a codec to transfer the audio, it emulates a modem on each end and runs the data between the two as actual data, not an encoded audio stream |
| 12:55.57 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:56.37 | C4colo | zydoon, yes |
| 12:56.45 | joobie | nice |
| 12:56.55 | C4colo | Dial(ZAP/3/3031234567,30) |
| 12:57.01 | C4colo | ... I think |
| 12:57.12 | joobie | that sounds pretty easy to do |
| 12:57.16 | C4colo | haven't done that sort of thing with zap in a while |
| 12:57.18 | [TK]D-Fender | zydoon: FreePBX is NOT supported here. Please use their channels |
| 12:57.19 | joobie | i mean emulate the modem and transfer the data |
| 12:57.26 | joobie | * havent done it? |
| 12:57.26 | zydoon | ok ok ok |
| 12:57.31 | [TK]D-Fender | ~freepbx |
| 12:57.31 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:57.42 | joobie | freepbx is a cop out ! use * form the cli |
| 12:57.47 | joobie | life's short, play hard |
| 12:58.00 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:58.02 | C4colo | where did you get freepbx in what zydoon said? did I miss something? |
| 12:58.05 | joobie | fender, can we add that to the freepbx defintition on jbot?:P |
| 12:58.07 | zydoon | yeah, tell that to your client :) |
| 12:58.28 | joobie | hi jaytee |
| 12:58.31 | jaytee | hi |
| 12:59.58 | joobie | c4, is it possible to make monitor stop when the bridge is terminated, rather than the handset hung up? |
| 13:00.45 | C4colo | I don't know for sure, but why does it matter if there are a few extra seconds of audio? |
| 13:00.57 | C4colo | are you worried about HDD space? |
| 13:01.46 | C4colo | in my opinion I'd want all of the audio, including while they are dialing, on hold, etc |
| 13:01.48 | joobie | i am |
| 13:01.51 | [TK]D-Fender | C4colo: zydoon>I want to make a specific extension dials outside using a specific zap channel, is it'feasable ? (Freepbx) <------- |
| 13:02.01 | joobie | like take the scenario that a user doesnt hang up their phone and leaves it off hook for an hour |
| 13:02.28 | C4colo | oh, haha, that (freepbx) part wrapped... I didn't even see it |
| 13:02.43 | C4colo | even the three times I searched for zydoon and looked at everyhting they said |
| 13:02.44 | [TK]D-Fender | joobie: if the other side hangs up, * terminates the call unless you told it to do otherwise |
| 13:03.22 | C4colo | hard drives are cheap |
| 13:03.24 | joobie | [TK]D-Fender, * is terminating the call.. but until i hangup the phone monitor() keeps recording |
| 13:03.33 | zydoon | c4colo suppose I'll do it in the config files |
| 13:03.33 | C4colo | buy a 750GB drive from newegg for $120 |
| 13:03.41 | joobie | yeaa |
| 13:03.41 | x86 | "Have you seen the latest Japanese camera? Apparently it is so fast it can photograph an American with his mouth shut!" |
| 13:03.46 | joobie | i was looking at HDD's today |
| 13:03.47 | x86 | gotta love fortune ;) |
| 13:03.50 | joobie | 1TB was like 200$ |
| 13:03.51 | zydoon | how can I tell the "FROM" exten ? |
| 13:03.53 | joobie | i was freaking |
| 13:04.06 | joobie | i remember paying 500$ for a 120GB |
| 13:04.07 | zydoon | Dial(ZAP/3/3031234567,30) ?? |
| 13:04.18 | C4colo | I'll answer you in #freepbx |
| 13:04.20 | coppice | I remember paying $1500 for 5MB |
| 13:04.26 | zydoon | I need exten 1111 dial always through ZAP/18 |
| 13:04.29 | joobie | C4colo, traitor! |
| 13:04.30 | joobie | :P |
| 13:04.35 | [TK]D-Fender | joobie: Do a channel dump in addition to full CLI |
| 13:04.40 | zydoon | ha ha ha |
| 13:05.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:05.06 | C4colo | whether you do it in the config files or in the freepbx interface it is still different than on a normal asterisk system |
| 13:05.10 | joobie | Fender, full cli as in debug 10, verbose 10 ? |
| 13:05.29 | [TK]D-Fender | zydoon: that is a FreePBX issue and not a topic for this channel |
| 13:05.31 | [TK]D-Fender | joobie: yes |
| 13:05.31 | C4colo | you would do it in a different file and would have to take into account the various contexts they have set up |
| 13:05.32 | joobie | okie |
| 13:05.39 | [TK]D-Fender | joobie: and a channel dump after what you consider the "end" of the call. |
| 13:05.46 | zydoon | comon fender, I just want to know how it is done on the config file |
| 13:05.53 | [TK]D-Fender | zydoon: IT ISN"T |
| 13:06.02 | creativx | doublequote fiery! |
| 13:06.12 | [TK]D-Fender | zydoon: Wher your phone can dial out is tracked in that stupid AGI of thier |
| 13:06.20 | [TK]D-Fender | theirs |
| 13:06.50 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:06.59 | [TK]D-Fender | zydoon: the channel driver config file for your device sets the context it uses, and that contains FreePBX's BS macros. |
| 13:07.04 | joobie | fender, how do to the channel dump? |
| 13:07.11 | joobie | sip debug ? |
| 13:07.39 | [TK]D-Fender | joobie: "show channels concise" |
| 13:07.42 | af_ | how to create a voicemail ? |
| 13:07.46 | joobie | Cheers |
| 13:08.02 | [TK]D-Fender | af_: How to create a complete question? |
| 13:08.21 | yangvnc | Is it possible to specify the DIAL (out) command to use a secondary line if the first one is being taken ? |
| 13:08.31 | af_ | I mean, files, directory and stiuffs. there is any utiltt to do that? |
| 13:09.13 | [TK]D-Fender | af_: You don't create those. They get created automatically as soon as VoiceMail is called for a box and the folders aren't set up already |
| 13:09.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.54 | [TK]D-Fender | yangvnc: you can tell it to pick from a zaptel group, or a single other channel. Everything else is up to the REST of your dialplan upon failure. |
| 13:09.55 | gnorbert | Could somebody send a link to THE book? |
| 13:10.02 | joobie | [TK]D-Fender, this is interesting .. http://pastebin.com/m6475b9c0 |
| 13:10.16 | joobie | when i hangup the answered phone.. nothing logs on console |
| 13:10.28 | *** join/#asterisk skirmisha (n=asda@92.66.88.17) |
| 13:10.30 | joobie | i just hear a click on my end.. but it's like asterisk keeps the call open |
| 13:10.35 | skirmisha | anyone alive here |
| 13:10.39 | [TK]D-Fender | ~book |
| 13:10.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:10.43 | [TK]D-Fender | gnorbert: ^^^^^^^ |
| 13:10.50 | gnorbert | [TK]D-Fender: Thanks. |
| 13:11.18 | joobie | ahhhhhhhhh |
| 13:11.19 | yangvnc | [TK]D-Fender: I understand the Call received procedure with the (busy) string, where i can forward the call to another extension, I am just wondering if I dial for example over GSM gateway and that one is being busy, can I route the call over a SIP or ZAP trunk? |
| 13:11.33 | joobie | you wouldn't happen to be fmor Australia would you? |
| 13:12.03 | [TK]D-Fender | yangvnc: There is no such thing as "route", "forward", or "transfer". All you do if execute MORE dialplan and do whatever the hell you want. |
| 13:12.12 | skirmisha | guys i have a question about MOH |
| 13:12.23 | yangvnc | [TK]D-Fender: ok |
| 13:12.28 | joobie | I don't know if this is a worldwide thing.. but i just realised this is normal behaviour in AU.. when the caller dials a recipient, if the recipient hangs up, it doesn't disconnect the call.. unless the recipient keeps the phone hung up for about a min or two... |
| 13:12.41 | joobie | so i think it's performing OK.. the other end hangs up and it maintains the call. |
| 13:12.45 | skirmisha | when incoming call comes in and user pick up and then put remote on hold, MOH is not played on answered channel but on incoming channel |
| 13:13.00 | skirmisha | thus i have no control what music to be played |
| 13:13.36 | [TK]D-Fender | skirmisha: PASTEBIN is your friend... |
| 13:14.09 | skirmisha | means? |
| 13:14.16 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:14.49 | skirmisha | to paste the log? |
| 13:14.56 | [TK]D-Fender | skirmisha: Means show us this call's complete CLI output witth high debug so we can SEE the problem. |
| 13:15.20 | skirmisha | ok let me show you |
| 13:15.27 | [TK]D-Fender | ~pb |
| 13:15.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:15.29 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 13:16.04 | [TK]D-Fender | joobie: If your ITSP doesn't tell you that the other end has hung up, yes, its stay "up" until they do, or until your other device hangs up first. |
| 13:16.14 | joobie | yaa |
| 13:16.16 | joobie | thanks fender |
| 13:16.19 | joobie | i think it's an AU thing |
| 13:16.23 | joobie | it's all good tho |
| 13:16.48 | joobie | damn im wrapped.. in one day ive got the * box dialing out, receiving calls and now recording every outbound call to wav |
| 13:16.53 | joobie | so simple with * |
| 13:17.49 | C4colo | what about 'start -> run -> "notepad" -> run -> CTRL+S -> "mycode.txt" -> putty2.exe -> sftp(somewebserver) -> put "mycode.txt" -> /msg #asterisk look at http://somewebserver/mycode.txt ' ? |
| 13:17.54 | C4colo | I didn't see that pastebin option listed |
| 13:18.23 | *** part/#asterisk ecrist (n=ecrist@t111-gw.c7200-1.bdr6.mpls.iphouse.net) |
| 13:18.48 | joobie | hmm only the polycom 320's.. when you dial a number in, you then have to press 'dial' after you have plugged in your number |
| 13:18.56 | joobie | is it possible to set up the phone so it just dials, like a normal phone? |
| 13:19.26 | C4colo | press speaker first |
| 13:19.31 | [TK]D-Fender | joobie: Go read the Admin Guide and learn how to configure its dialplan |
| 13:19.33 | C4colo | and make sure the dialpattern is correct |
| 13:19.51 | C4colo | it won't dial until you press dial if you are pre-dialing |
| 13:20.03 | C4colo | you have to hit the speaker button first, and it has to match a pattern in the dialstring |
| 13:20.17 | joobie | what if i dont want speaker |
| 13:20.21 | joobie | like just pickup the handset |
| 13:20.25 | joobie | i think my dialplan aint matching |
| 13:20.26 | C4colo | then pick up the handset and dial |
| 13:20.32 | joobie | it's like X. in the pattern match |
| 13:20.35 | C4colo | the dial string needs to match |
| 13:20.37 | C4colo | oh |
| 13:20.38 | C4colo | yea |
| 13:20.39 | joobie | kk |
| 13:20.44 | joobie | ill try lock it to X digits |
| 13:20.45 | C4colo | that means "anything with a timeout" |
| 13:20.48 | joobie | yaa |
| 13:20.50 | joobie | sec ill try |
| 13:20.54 | C4colo | er, "one or more digits with a timeout" |
| 13:20.59 | joobie | btw if i mod this extensions.conf |
| 13:21.03 | joobie | is there a shortcut |
| 13:21.06 | joobie | to suck in teh changes? |
| 13:21.11 | joobie | im doing asterisk restart each time |
| 13:21.16 | C4colo | yea |
| 13:21.19 | joobie | my sip provider must be getting upset :P |
| 13:21.20 | C4colo | dialplan reload |
| 13:21.26 | C4colo | nah, they don't watch that |
| 13:21.30 | joobie | ahhh |
| 13:21.33 | joobie | sweet |
| 13:21.33 | joobie | thanks |
| 13:22.24 | C4colo | you should see the CLI on the servers at the CLEC I work for, constant "peer_poke XXX is now reachable" "registration from 123 with no mailbox" "so-and-so is now lagged" etc |
| 13:22.51 | C4colo | hard to see what is going on at peak time with 50-70 calls active |
| 13:23.07 | C4colo | so no, your reloading is not annoying them in the least |
| 13:23.16 | C4colo | in fact, they probably don't even log your registration attempts |
| 13:24.04 | C4colo | if they do, they don't keep them long I would guess |
| 13:24.42 | joobie | heh |
| 13:24.57 | joobie | yea i guess it's sorta like traffic monitoring |
| 13:25.06 | *** join/#asterisk onats (n=onats@unaffiliated/onats) |
| 13:25.31 | onats | hi, can anyone point me to a good tutorial to setup a simple home asterisk pbx? i have an x100p card i want to use also |
| 13:25.31 | joobie | i ended up writing an application at my old work that plotted traffic going through our firewall,because when people complained about the load it was too hard to get a snapshot to see where the data was going |
| 13:25.45 | joobie | ~book |
| 13:25.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:25.53 | joobie | try that onats |
| 13:26.02 | joobie | it's free! |
| 13:26.32 | joobie | C4colo, exten => _XXXXXXXX,n,Dial(SIP/comvergence/${EXTEN}) that is an example of the pattern match im using.. |
| 13:26.38 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) [NETSPLIT VICTIM] |
| 13:26.39 | onats | joobie, thanks... looking into itnow |
| 13:26.49 | joobie | the polycom still doesnt dial when i enter in an 8 digit telephone number |
| 13:26.56 | joobie | despite if i do it via speaker phone / handset.. |
| 13:27.01 | joobie | onats, no worries.. |
| 13:28.28 | skirmisha | here is the log |
| 13:28.32 | skirmisha | -- Executing [s@macro-dial:10] Dial("SIP/999124-b5264e38", "SIP/999123|20|TtrM(auto-blkvm)") in new stack |
| 13:28.32 | skirmisha | <PROTECTED> |
| 13:28.32 | skirmisha | <PROTECTED> |
| 13:28.32 | skirmisha | -- SIP/999123-088a83a0 answered SIP/999124-b5264e38 |
| 13:28.32 | skirmisha | <PROTECTED> |
| 13:28.33 | skirmisha | -- Started music on hold, class 'default', on SIP/999124-b5264e38 |
| 13:28.40 | [TK]D-Fender | joobie: You configure the dialplan... ON THE PHONE ITSELF |
| 13:28.45 | jaytee | use pastebin |
| 13:28.46 | skirmisha | 999124 call 999123 |
| 13:28.52 | [TK]D-Fender | skirmisha: PASTEBIN. Do not spam in here. |
| 13:28.53 | [TK]D-Fender | ~pb |
| 13:28.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:28.54 | jaytee | skirmisha |
| 13:28.55 | skirmisha | 999123 put 999124 on hold |
| 13:28.55 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:28.58 | skirmisha | yes sorry |
| 13:29.41 | [TK]D-Fender | skirmisha: - Executing [s@macro-auto-blkvm:4] SetMusicOnHold("SIP/999123-088a83a0", "old") in new stack <- this is DIALPLAN. why is dialplan being executed for this? Did you try to configure some sort of dynamic feature? |
| 13:29.43 | skirmisha | as u see MOH is set on answered channel |
| 13:29.59 | skirmisha | but moh is played on other cahnnel |
| 13:30.24 | skirmisha | according to manual i have to use M with dialplan |
| 13:30.30 | joobie | [TK]D-Fender, ahhh |
| 13:30.34 | skirmisha | so there is macro used |
| 13:30.40 | joobie | [TK]D-Fender, i thnk that capital that you used made it clearer |
| 13:30.40 | [TK]D-Fender | skirmisha: WTF. undo that macro. |
| 13:30.45 | joobie | thanks ;) |
| 13:30.46 | skirmisha | which set MOH for every call |
| 13:31.20 | C4colo | skirmisha: why don't you set it in sip.conf? |
| 13:31.26 | skirmisha | this won't change anything |
| 13:31.40 | C4colo | mohsuggest and mohinterpret |
| 13:31.40 | C4colo | look those up and see if they do what you want |
| 13:31.45 | [TK]D-Fender | skirmisha: Remove the macro. |
| 13:31.51 | skirmisha | i want when user put on hold to define what music to stream asterisk , means what class |
| 13:32.19 | C4colo | mohsuggest and mohinterpret |
| 13:32.23 | C4colo | look those up and see if they do what you want |
| 13:32.39 | [TK]D-Fender | skirmisha: do that BEFORE you dial. |
| 13:32.47 | skirmisha | if 999123 the dialing user put 999124 on hold then proprer music is played to 999124 |
| 13:32.57 | [TK]D-Fender | skirmisha: And pull out that Macro for your testing. |
| 13:33.28 | skirmisha | ok macro i'll take it out |
| 13:33.34 | C4colo | if joe wants his callers to hear rock and jim wants them to hear classical, add mohsuggest=rock to joe's sip.conf user details and mohsuggest=classical to jim's |
| 13:33.38 | skirmisha | so u suggest to set moh before dialing |
| 13:34.02 | [TK]D-Fender | skirmisha: YES. |
| 13:34.08 | skirmisha | let me try that |
| 13:35.16 | *** join/#asterisk pigpen (n=pigpen@fw.seamans.cc) |
| 13:35.34 | C4colo | maybe I"m missing what is trying to be done here |
| 13:35.35 | yangvnc | [TK]D-Fender: calling out actually switches with the n(busy) string :) |
| 13:36.13 | *** join/#asterisk mwalling (i=mwalling@2001:470:1f0f:81:0:0:0:1) [NETSPLIT VICTIM] |
| 13:36.24 | [TK]D-Fender | yangvnc: Once again this is YOUR dialplan, and you are the one that is making that get called. |
| 13:36.53 | creativx | man i need to learn me some AGI |
| 13:37.51 | joobie | hey guys |
| 13:38.04 | joobie | to process 20 calls concurrently using SIP |
| 13:38.13 | joobie | would a single p4 handle this? |
| 13:38.28 | seanbright | yes |
| 13:38.32 | joobie | just sorta toying with ideas for waht hardware configuration i would need |
| 13:38.39 | joobie | seanbright, im also thinking about recording every call |
| 13:38.42 | joobie | concurrently |
| 13:38.46 | creativx | SSD |
| 13:38.55 | joobie | so 20 callers talking.. 20 recording going at the same time |
| 13:39.02 | seanbright | joobie: that should still be fine |
| 13:39.15 | [TK]D-Fender | joobie: Depends on transcoding. |
| 13:39.25 | seanbright | not really |
| 13:39.27 | seanbright | you'll be fine |
| 13:39.29 | joobie | seanbright, what about the hard disks? I was thinking 3 disks in raid5 |
| 13:39.38 | seanbright | meh |
| 13:39.42 | [TK]D-Fender | joobie: plenty |
| 13:39.43 | seanbright | go raid 10 |
| 13:39.51 | joobie | raid 10 sucks because of redundancy |
| 13:40.03 | joobie | oh raid 10 is mirror and stripe ya? |
| 13:40.07 | joobie | or is it just strip |
| 13:40.08 | joobie | e |
| 13:40.16 | seanbright | raid 1 + raid 0 |
| 13:40.20 | joobie | ahh |
| 13:40.25 | joobie | need 4 disks for that ya? |
| 13:41.11 | seanbright | yeah |
| 13:41.21 | seanbright | so buy another disk |
| 13:41.22 | seanbright | :) |
| 13:41.36 | joobie | heh |
| 13:41.43 | joobie | and this would be just for the recording ya? |
| 13:41.54 | joobie | i mean that is the only io intensive part of it? |
| 13:41.55 | seanbright | that's what i use here |
| 13:41.58 | seanbright | yeah |
| 13:42.04 | joobie | do you have a seperate disk for the OS? |
| 13:42.05 | seanbright | well, disk i/o |
| 13:42.09 | joobie | or a seperate raid1 for the OS |
| 13:42.23 | seanbright | nope, just one big disk |
| 13:42.28 | *** join/#asterisk UngaMan (n=jvannini@dynamic55-30.MASAYA.cablenet.com.ni) |
| 13:42.32 | seanbright | we only use the local disk for temporary storage |
| 13:42.33 | UngaMan | good morning |
| 13:42.39 | joobie | ahh |
| 13:42.41 | seanbright | everything gets moved off after the call is recorded |
| 13:42.51 | joobie | interesting |
| 13:43.00 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:43.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.15 | joobie | one more Q |
| 13:43.18 | joobie | then im going to bed:P |
| 13:43.26 | seanbright | i'm making up all these answers |
| 13:43.29 | seanbright | for the record |
| 13:43.31 | seanbright | heh |
| 13:43.40 | joobie | you lie! |
| 13:43.41 | joobie | :P |
| 13:43.55 | joobie | how can i figure out how much hdd space i need for like a minute of recording? |
| 13:44.00 | joobie | is ther ea forumla? |
| 13:44.08 | seanbright | there is, but i don't know what it is |
| 13:44.14 | seanbright | [TK]D-Fender could probably tell you |
| 13:44.19 | [TK]D-Fender | joobie: Record a call, then take an Primary School math course. |
| 13:44.20 | seanbright | he's super smart |
| 13:44.23 | seanbright | he tells me all the time |
| 13:44.25 | seanbright | heh |
| 13:44.53 | joobie | [TK]D-Fender, is the size the same always if i record for X seconds? or is it depenadnt on what is said in the call? |
| 13:45.00 | joobie | hehe |
| 13:45.10 | joobie | he is super smart - must be the primary school math course |
| 13:45.11 | [TK]D-Fender | joobie: Constant bit-rate |
| 13:45.17 | joobie | cheers fender |
| 13:45.31 | joobie | you learn something everyday. |
| 13:45.36 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 13:45.49 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:46.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:47.42 | skirmisha | [TK]D-Fender that works |
| 13:48.08 | [TK]D-Fender | joobie: Of course you could have figured this out jsut by trying or reading the WIKI, but.... nah that'd never work. |
| 13:48.45 | *** join/#asterisk patrick-- (n=patrick@noc.incoweb.de) |
| 13:48.52 | *** join/#asterisk `Sean (i=Un1x@CPE001d451b875f-CM00111ade88b6.cpe.net.cable.rogers.com) |
| 13:49.07 | joobie | does 16KB/second sound about right? |
| 13:49.13 | joobie | for a rough .. in wav format |
| 13:49.24 | joobie | [TK]D-Fender, that's too involved |
| 13:49.43 | [TK]D-Fender | joobie: You're going to have to get off your ass for this one. |
| 13:49.55 | patrick-- | Hey all! Im having trouble with my X-Lite Softphone. After i Picked up a call from another extension, i cannotperform a call transfer with #. does anyone know why that is so?= |
| 13:50.05 | joobie | hehe |
| 13:50.59 | joobie | ok im out |
| 13:51.16 | joobie | fender, sean, c4 |
| 13:51.23 | joobie | thanks for your help |
| 13:51.31 | joobie | made some HUGE progress |
| 13:51.34 | joobie | much appreciated |
| 13:51.44 | joobie | enjoy the night boys |
| 13:51.50 | joobie | peace |
| 13:52.04 | [TK]D-Fender | patrick--: Because odd are whn you do "pickup", you can't TELL IT to look for DTMF to transfer |
| 13:52.33 | [TK]D-Fender | patrick--: So go find a way to "Dial" into your pickup, or get a real phone. |
| 13:53.12 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:54.43 | keith4 | so, AgentCallbackLogin is deprecated in 1.4? What's the new "correct" way to do this? |
| 13:55.40 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:56.32 | deeperror | i've got a family in astdb that i am unable to remove due to some unknown space in the string. Any clues how to remove it? |
| 13:57.31 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:58.06 | [TK]D-Fender | deeperror: pastebin the AstDB dump, and your attempt |
| 13:58.24 | [TK]D-Fender | keith4: "core show applications like queue" |
| 13:58.26 | deeperror | database show or is there another way of dumping this? |
| 13:59.14 | deeperror | http://pastebin.ca/1078389 |
| 13:59.16 | [TK]D-Fender | deeperror: do taht first |
| 13:59.38 | [TK]D-Fender | deeperror: and now your attempts to kill it off. |
| 13:59.55 | deeperror | database deltree BTTS_199 |
| 14:00.22 | deeperror | and several other \n \t \0 attempts at the end not sure how to represent that in cli to kill it |
| 14:00.32 | [TK]D-Fender | deeperror: do it from the dialplan then. |
| 14:00.36 | keith4 | will it accept regexps? |
| 14:00.54 | deeperror | just use \n \t \0 \x0b stuff in there? |
| 14:01.03 | [TK]D-Fender | deeperror: Considering it seems somewhat clear thats how it got there in the first place. |
| 14:01.19 | [TK]D-Fender | deeperror: do it witht he space, and kill the key directly. |
| 14:01.41 | [TK]D-Fender | deeperror: DBDel(/BTTS_199 /TERMINAL) |
| 14:02.00 | [TK]D-Fender | deeperror: and go hunt down the source of this poorly chosen key |
| 14:02.41 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:02.48 | neurosys | Morning :) |
| 14:03.14 | deeperror | [TK]D-Fender, it actually came from an agi script but i'm on the case now will let you know of the results |
| 14:03.35 | [TK]D-Fender | deeperror: Well go fix the source so it doesn't spread |
| 14:04.17 | Katty | morning |
| 14:04.50 | _ShrikE | morning Katty |
| 14:04.54 | [TK]D-Fender | Katty: Mew. |
| 14:05.18 | deeperror | [TK]D-Fender, yea its all fixed up...removing from dialplan worked thanks |
| 14:05.32 | [TK]D-Fender | deeperror: You're welcome |
| 14:07.32 | neurosys | [TK]D-Fender: filled out a support ticked with my ITSP for my outbound issues. :) |
| 14:10.05 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:10.30 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 14:11.28 | Katty | hugs _ShrikE and [TK]D-Fender |
| 14:11.49 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:12.28 | jaytee | I just got a panicky email from one of our users (losers) referencing the Storm worm and had I.T. heard about it? He's all worried. I replied to his email with, "We're doomed. It's the damned Russkies and there's nuthin we can do about it." |
| 14:16.34 | *** join/#asterisk RoyK (n=roy@ip-146-20-149-91.dialup.ice.no) |
| 14:16.38 | Nugget | heh |
| 14:16.55 | Nugget | http://thebigo.sine.com/airwolf.jpg <-- fight the russkies |
| 14:19.10 | *** join/#asterisk freckle (n=jon@195.74.96.118) |
| 14:19.21 | [TK]D-Fender | Nugget: Blue Thunder FTW! |
| 14:19.35 | [TK]D-Fender | Nugget: Ithink we're gonna need a bigger chopper! |
| 14:19.57 | *** join/#asterisk trinux (n=mimi@unaffiliated/trinux) |
| 14:21.13 | Segnale007 | hello guys |
| 14:21.47 | *** part/#asterisk trinux (n=mimi@unaffiliated/trinux) |
| 14:21.47 | *** join/#asterisk idimmu (n=idimmu@holly.queeg.org) |
| 14:21.56 | Segnale007 | I have a question regarding hw |
| 14:21.57 | Segnale007 | so |
| 14:22.10 | *** part/#asterisk idimmu (n=idimmu@holly.queeg.org) |
| 14:22.41 | Segnale007 | does an 3.3v 64bit pci analog card work on a 3.3v 32bit socket ? |
| 14:23.02 | Segnale007 | I am talking about an digium card tdm410b |
| 14:24.35 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:25.52 | coppice | that card is a 32bit PCI card |
| 14:25.57 | *** join/#asterisk kaiowos (n=kaiowos@200.88.48.140) |
| 14:26.02 | Segnale007 | is 32 bit ? |
| 14:26.06 | Segnale007 | how do you know ? |
| 14:26.16 | Segnale007 | I thought it was 64bit |
| 14:26.18 | Segnale007 | :S |
| 14:26.27 | coppice | take a look at a picture of it |
| 14:26.34 | Segnale007 | ok wait |
| 14:26.46 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:28.03 | Segnale007 | http://www.888voipstore.com/digium-tdm413e-pr-19238.html |
| 14:28.10 | Segnale007 | I am taking a look here |
| 14:28.23 | Segnale007 | and it seems like a 64bit card |
| 14:28.35 | Segnale007 | so I am reading the asterisk book |
| 14:28.48 | Segnale007 | they made the example about the pci devices |
| 14:28.56 | Segnale007 | and it look like an 64bit |
| 14:29.01 | Segnale007 | maybe am I in wrong ? |
| 14:29.04 | Segnale007 | could be .. |
| 14:29.49 | *** join/#asterisk jgoddess (n=monkey@phrank.aus.us.siteprotect.com) |
| 14:30.54 | coppice | yes, you could be wrong. it only has on connector |
| 14:31.03 | coppice | s/on/one |
| 14:31.33 | Segnale007 | ok, better if I am in wrong then ;) |
| 14:31.34 | *** part/#asterisk UngaMan (n=jvannini@dynamic55-30.MASAYA.cablenet.com.ni) |
| 14:31.34 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) |
| 14:31.38 | thing1 | hi buys |
| 14:31.48 | Segnale007 | can you show me a 64bit pci card ? |
| 14:31.52 | Segnale007 | of you can .. |
| 14:32.02 | Segnale007 | *if you can .. |
| 14:32.28 | thing1 | i'm trying to connect to a cisco call manager with sip trunk but it keeps saying it's unreachable, though i can telnet into it with asterisk |
| 14:32.28 | Nugget | telnet is eeeeeeevil! |
| 14:33.26 | coppice | none of the digium cards are 64 bit |
| 14:33.38 | thing1 | is it now |
| 14:33.47 | Segnale007 | ah ok ... |
| 14:34.17 | Segnale007 | I am newbie about asterisk and zaptel compatible hw |
| 14:34.27 | thing1 | has anyone here successfully set up a sip peer to cisco call manager |
| 14:34.42 | jgoddess | hey guys is there a list of backports |
| 14:34.43 | Segnale007 | then I am planing about what should I get for build my first pbx server |
| 14:35.10 | jgoddess | backports, I mean those ones that are available from 1.6->1.4 |
| 14:35.41 | [TK]D-Fender | thing1: Yes, I'm sure there are. |
| 14:35.56 | [TK]D-Fender | jgoddess: There are very few |
| 14:36.06 | *** join/#asterisk NovceGuru (n=NovceGur@rrcs-70-62-198-142.central.biz.rr.com) |
| 14:37.49 | thing1 | well i'm wondering if i could talk though my config with someone |
| 14:38.39 | [TK]D-Fender | thing1: pastebin your CLI output with SIP debug for your failed attempt |
| 14:39.03 | jgoddess | yeah I was just curious I have ran accross them but was curious if there was a page that listed all the ones that were available ;) |
| 14:39.10 | jgoddess | so it would make it easier to find them :) |
| 14:39.40 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:39.46 | [TK]D-Fender | jgoddess: So far as I know the only one is the DevState patch |
| 14:39.59 | jgoddess | which is awesome ;) |
| 14:40.05 | jgoddess | thanks though |
| 14:40.10 | [TK]D-Fender | jgoddess: People don't generally waste their time trying to make something old into something new |
| 14:40.33 | jgoddess | yeah but its hard to justify to the CEO that upgrading to a beta version is smart ;) |
| 14:41.39 | jgoddess | but I need devstate since we have 4 servers and need to be able to have the phone states be the same across all servers |
| 14:41.46 | rwaite | i cant wait until 1.6 is released so debian stable will be *2* versions behind |
| 14:41.55 | jgoddess | hah no kidding ;) |
| 14:43.40 | [TK]D-Fender | rwaite: Any anyone using * from packages like that gets what they deserve |
| 14:44.11 | rwaite | yeah, asterisk is pretty good too about not throwing a bunch of crap everywhere |
| 14:44.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:46.45 | _ShrikE | jgoddess: I think there is an extstate patch also |
| 14:46.55 | thing1 | http://pastebin.com/m722a96 |
| 14:47.02 | jgoddess | yeah there are a couple of more I have run across |
| 14:47.03 | jgoddess | ;) |
| 14:47.18 | jgoddess | so I am searching issues now |
| 14:47.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:47.52 | [TK]D-Fender | thing1: that is NOT a call attempt |
| 14:48.22 | [TK]D-Fender | thing1: And you are spanning subnets. Are you VPN'd between the two of them? |
| 14:48.53 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 14:48.58 | thing1 | no |
| 14:49.28 | jgoddess | there is also a backport for certain codecs as well |
| 14:49.36 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 14:49.38 | [TK]D-Fender | thing1: is your * remote to your cisco? |
| 14:49.40 | raz | ~book |
| 14:49.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:49.48 | [TK]D-Fender | jgoddess: No. |
| 14:50.03 | [TK]D-Fender | jgoddess: Let me guess, you want G.722? |
| 14:50.04 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:50.10 | jgoddess | heh no |
| 14:50.51 | jgoddess | I am just discussing not necessarily needing at this point |
| 14:50.51 | jgoddess | all is well sir |
| 14:50.51 | [TK]D-Fender | jgoddess: I wouldn't waste time fishing if I were you. |
| 14:50.51 | coppice | oh, go on. use G.722. wideband is good |
| 14:50.51 | thing1 | no cisco is remote |
| 14:51.00 | thing1 | http://pastebin.com/m341e1e0e |
| 14:51.13 | thing1 | my network is 192.168.0.x |
| 14:51.19 | thing1 | cisco is 10.2.0.2 |
| 14:51.56 | [TK]D-Fender | thing1: All of your IP's there are private subnets. You ahve not configured * to correctly handle NAT. Go read these guides : |
| 14:52.00 | [TK]D-Fender | ~sipnat |
| 14:52.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:52.02 | jgoddess | fishing for what there are certain things I need that I came across if 1.6 isn't in a stable released version than what is the point of upgrading if I need the features I need the features and if they are available then I might as well only patch and compile once no big deal though ;) |
| 14:52.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 14:52.35 | [TK]D-Fender | jgoddess: Well, good luck with that, there won't be much of anything to find. |
| 14:53.04 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 14:53.51 | bijit | anyone know where I can get the libnewt for slackware? |
| 14:54.24 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 14:54.49 | [TK]D-Fender | bijit: Comes on the CD. |
| 14:55.06 | [TK]D-Fender | bijit: www.slackware.com if you can't grab it from there |
| 14:56.16 | bijit | really hmm just checked machine to see if it had it but did not find. Let me go back and check again. ty [TK]D-Fender |
| 14:56.31 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:58.13 | [TK]D-Fender | bijit: It's there, I used to use Slackware exclusively until just over a year ago |
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| 15:07.03 | jaytee | [TK]D-Fender, I bought several tunes of Eric Mongrain's from his website. He just released his first CD |
| 15:07.15 | [TK]D-Fender | jaytee: Cool... |
| 15:07.25 | jeev | FENDER |
| 15:07.53 | keith4 | sometimes, i wish the wiki were divided into 1.2/1.4/1.6 sections |
| 15:08.12 | bijit | [TK]D-Fender: slack 12 was the last version to bring newt. |
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| 15:12.11 | [TK]D-Fender | bijit: When in doubt : www.linuxpackages.net |
| 15:12.54 | jeev | tsk |
| 15:14.44 | tzafrir_laptop | any asterisk there? |
| 15:15.36 | Qwell | wow, linuxpackages.net still exists? |
| 15:15.42 | Qwell | my buddy runs that site |
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| 15:29.19 | hunmonk | can somebody tell me what the speex dependencies are for asterisk 1.4 on CentOS? i keep getting build errors and can't seem to figure out what i'm missing. i've got speex and speex-devel installed, but still no love |
| 15:29.49 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:30.37 | [TK]D-Fender | hunmonk: You'll have to re-run "./configure" and rebuild *. |
| 15:30.59 | tzafrir_laptop | hunmonk, speex-devel? what version? 1.0 is not good. |
| 15:31.06 | *** join/#asterisk ecrist (n=ecrist@t111-gw.c7200-1.bdr6.mpls.iphouse.net) |
| 15:31.31 | tzafrir_laptop | You probably need something of the sort of speex1.2-devel |
| 15:31.43 | ecrist | can anyone recommend some good voip phones (SIP) that have 10+ 'feature' keys for extension mapping? |
| 15:31.54 | tzafrir_laptop | or something with speexdsp |
| 15:32.52 | hunmonk | tzafrir_laptop: i can try 1.2 i guess. just need to find it. centos default repos have it. |
| 15:32.57 | hunmonk | hunts in trixbox repos |
| 15:33.22 | tzafrir_laptop | trixbox repos != centos default repos, BTW |
| 15:33.30 | patrick-- | is it possible to perform actions within a dialplan if the Dial'ed extension is busy ? |
| 15:33.33 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 15:34.02 | tzafrir_laptop | yes, check ${DIALSTATUS} |
| 15:34.15 | hunmonk | tzafrir_laptop: right, i've added the trixbox ones where necessary |
| 15:34.19 | patrick-- | thanks |
| 15:34.57 | hunmonk | tzafrir_laptop: 1.2beta2-1 <-- that workable for my needs? |
| 15:35.08 | tzafrir_laptop | yes |
| 15:35.19 | tzafrir_laptop | actually anything above 1.1 will do |
| 15:36.02 | jaytee | ecrist, the Polycom 650 with the Backlit Expansion module would give you an extra 14 line appearances or speed dial with BLF keys. |
| 15:37.10 | ecrist | jaytee: thanks |
| 15:37.23 | ecrist | saw that, was hoping for something a bit more compact |
| 15:37.48 | ecrist | something around the size of the 330, with more keys. |
| 15:37.55 | hunmonk | tzafrir_laptop: hrm. 'make menuselect' still shows 'XXX' for the speex codec. it says 'Depends on: speex(E), speex_preprocess(E)' |
| 15:38.15 | tzafrir_laptop | maybe you need to re-run configure |
| 15:38.57 | [TK]D-Fender | ecrist: http://digiumcards.com/snom_370.html |
| 15:39.33 | ecrist | [TK]D-Fender: that's pretty sweet looking. |
| 15:39.41 | ecrist | have you used it? |
| 15:39.50 | gr0mit | ecrist, i have a snom 360 here |
| 15:39.53 | gr0mit | works a treat |
| 15:39.55 | [TK]D-Fender | ecrist: or http://digiumcards.com/linksys_spa962_voip_phone_lvs9000.html + http://www.telephonydepot.com/product_p/105-054-932.htm |
| 15:40.01 | Qwell | [TK]D-Fender: that site is shady |
| 15:40.06 | Qwell | just throwing that out there |
| 15:40.15 | [TK]D-Fender | ecrist: Snom can be dodgy with their firmware, but its a question of meeting your other needs. |
| 15:40.25 | [TK]D-Fender | Qwell: In what way? |
| 15:40.36 | [TK]D-Fender | Qwell: e4 is a pretty big co from what I've heard. |
| 15:40.41 | ecrist | [TK]D-Fender: can you define 'dodgy'? |
| 15:40.49 | [TK]D-Fender | Qwell: though I don't like their taking part of your anme. |
| 15:40.59 | [TK]D-Fender | ecrist: Has had a history of instability. |
| 15:41.02 | [TK]D-Fender | ecrist: YMMV |
| 15:41.08 | ecrist | thanks. |
| 15:41.09 | keith4 | what would be a good argument for using dynamic queue members? |
| 15:41.13 | keith4 | other than hot-desking |
| 15:41.13 | [TK]D-Fender | name* |
| 15:41.50 | [TK]D-Fender | keith4: adding queuemembers that aren't necessarily part of that queue |
| 15:42.05 | keith4 | hmm |
| 15:42.17 | [TK]D-Fender | keith4: Imagine support is missing people, and one sales guy is good enough to help out. He could sing in and be a member of that queue as well |
| 15:42.37 | keith4 | gotcha |
| 15:43.25 | DarKnesS_WolF | tzafrir_laptop: any new news about this issue of DTMF interfearinc in the analog devices? |
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| 15:43.59 | keith4 | I need to implement a virtual office of 5 people. I was thinking of the standard "if you know the extension you want, dial it" IVR crap, and then if the caller does nothing, dump to a queue. If all agents are using SIP phones, can I just make them all members of that queue? I'm concerned about what happens if, say, some idiot presses the DND button on his phone |
| 15:44.04 | tzafrir_laptop | DarKnesS_WolF, already resolved in SVN and in e.g. latest zaptel debs (patched from SVN) |
| 15:44.16 | DarKnesS_WolF | tzafrir_laptop: great ! |
| 15:44.31 | DarKnesS_WolF | tzafrir_laptop: what is e.g ? |
| 15:46.02 | tzafrir_laptop | <PROTECTED> |
| 15:46.41 | [TK]D-Fender | keith4: DND usually comes back as "busy" or "congestion". Should be fine. |
| 15:46.52 | bijit | make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'. Stop. |
| 15:47.05 | bijit | doe that mean I am missing a lib? |
| 15:47.22 | tzafrir_laptop | bijit, have you downloaded the ilbc codec? |
| 15:48.02 | bijit | possible error haven't dll anything related to ilibc |
| 15:48.03 | DarKnesS_WolF | tzafrir_laptop: thx ;-) |
| 15:48.13 | keith4 | [TK]D-Fender: am I correct in thinking that I don't need dynamic members, really? *I* would like people to have to log in / log out when they're actually at their desks or not, but I don't think I trust them to do that |
| 15:48.19 | bijit | tzafrir_laptop: ty |
| 15:48.26 | keith4 | and I'm afraid of what will happen if I turn on auto logoff |
| 15:48.38 | [TK]D-Fender | keith4: I just told you what "Dynamic" was for. |
| 15:50.12 | keith4 | it seems to me that it might be appropriate in the case of: 4 guys out on jobs, only 1 in the "office" today... no reason to have 4 missing agents on the queue, no? |
| 15:52.03 | [TK]D-Fender | keith4: You need to find your brain. It seems to have slipped out of your skull.... Look for the oozing trail |
| 15:52.19 | keith4 | i think it's somewhere between here and the coffee shop, let me check |
| 15:52.45 | keith4 | is the situation that I just proposed not similar to your hypothetical scenario? |
| 15:53.25 | [TK]D-Fender | keith4: You don't seem to know how you wnat it to work let alone how to do it. |
| 15:53.59 | keith4 | I'm definitely not at the "how to do it" stage yet. That part I can look up, though |
| 15:54.23 | keith4 | what I *can't* look up is "best practices" |
| 15:55.09 | keith4 | wonders if there's anything like this in the cookbook |
| 15:58.51 | [TK]D-Fender | keith4: There is no "best practice" for this. This is a CORPORATE decision. |
| 15:59.12 | keith4 | how about "available practices" |
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| 15:59.31 | keith4 | cries |
| 15:59.35 | keith4 | I need someone to tell me what to do! |
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| 16:01.26 | [TK]D-Fender | keith4: www.drphil.com |
| 16:02.04 | keith4 | emails Dr Phil |
| 16:02.11 | keith4 | doubts he'll be able to help |
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| 16:05.41 | keith4 | alright, forget the management angle of this. what difference will the caller notice if there are 4 absent agents assigned to the queue, or if there is only busy agent? |
| 16:05.51 | keith4 | only *one* busy agent |
| 16:06.35 | [TK]D-Fender | keith4: Depends on how you set up your queue |
| 16:06.57 | fogo | keith4: if you have things setup correctly, it shouldn't matter - it should just ring through busy/unavailable agents (again, depending on setup) |
| 16:08.28 | keith4 | hmmm |
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| 16:12.21 | javb | i have an openvox t1 card installed, with the correct zaptel conf, when doing service zaptel start, i get: Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) ; any idea guys ? |
| 16:18.44 | RoyK | just uses sangoma cards |
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| 16:26.27 | tzafrir_laptop | javb, obviously it means that your zaptel.conf does not match the reality :-) |
| 16:26.54 | tzafrir_laptop | Please pastebin /etc/zaptel.conf and the output of cat /proc/zaptel/* |
| 16:27.19 | javb | tzafrir_laptop, its not obviosly, just installed zaptel 1.2, with the same file, and problem solved. |
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| 16:27.43 | javb | =) |
| 16:28.12 | tzafrir_laptop | "obviously" because this is what this error means :-) . From your description I suspect modules failed to load or whatever |
| 16:28.27 | tzafrir_laptop | cat /sys/module/zaptel/version |
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| 17:00.59 | zydoon | j |
| 17:01.19 | SwK | anyone in huntsville have a Gen1 iPhone they wanna part with |
| 17:01.36 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 17:02.33 | Qwell | SwK: awful nice of you to try to find me a phone. ;) |
| 17:02.55 | *** part/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 17:02.55 | SwK | Qwell, trying to find me one :P |
| 17:02.59 | SwK | Qwell, you still in HSV |
| 17:03.03 | Qwell | yeah |
| 17:03.06 | SwK | werd |
| 17:03.07 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 17:03.29 | SwK | Qwell, hows things at D |
| 17:03.35 | Qwell | great |
| 17:10.13 | De_Mon | duuuuu, where are the docs for sip.conf? |
| 17:10.31 | seanbright | sip.conf.sample? |
| 17:10.35 | [TK]D-Fender | De_Mon: in the sample config & the WIKI |
| 17:12.55 | De_Mon | ahh, debian packages the sample configs and the docs separately... and I didn't consider the samples as being that informative |
| 17:14.13 | keith4 | oh, they are |
| 17:23.18 | ecrist | is asterisk RFC 4235 compliant? |
| 17:26.44 | [TK]D-Fender | ecrist: http://svn.digium.com/view/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup&pathrev=63150 |
| 17:30.58 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 17:32.46 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 17:33.28 | ecrist | [TK]D-Fender: can you tell me, on the phone you linked me, is there support for extension status on the handset (on a call, etc)? |
| 17:33.42 | ecrist | the page you just linked states "Supported (server)" |
| 17:34.35 | [TK]D-Fender | ecrist: "busy / disconnected / available" as far as I know, the rest, no idea |
| 17:34.55 | ecrist | [TK]D-Fender: that's what I was looking for. thanks. |
| 17:34.58 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:35.14 | Wayhigh | how do you use System() from the asterisk manager interface? |
| 17:35.33 | [TK]D-Fender | Wayhigh: You don't |
| 17:35.57 | [TK]D-Fender | Wayhigh: AMI is not made for you to execute dialplan commands |
| 17:36.09 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:37.04 | Wayhigh | fender: can you modify the dialplans at least? |
| 17:37.11 | Wayhigh | even temporarily? |
| 17:39.19 | bijit | is it very necessary to install unixobdc? |
| 17:39.24 | De_Mon | Wayhigh you seem to be missing the point |
| 17:39.40 | De_Mon | bijit to run asterisk? no |
| 17:40.07 | Katty | scooby do! |
| 17:40.17 | De_Mon | http://www.engadget.com/2008/07/21/how-to-reveal-blocked-caller-id-info-a-video-guide-to-risky-beh/ |
| 17:40.32 | De_Mon | asterisk hacking just got more fun |
| 17:40.34 | Wayhigh | I guess I'm totally missing the value of the AMI without it having access to all the commands |
| 17:40.48 | [TK]D-Fender | Wayhigh: Yes, temporary, but to what end? |
| 17:40.58 | bijit | De_Mon: ty |
| 17:41.05 | [TK]D-Fender | Wayhigh: You've more likely missed the entire POINT. |
| 17:41.16 | [TK]D-Fender | Wayhigh: AMI is not the way to manae a UNIX system. |
| 17:41.22 | [TK]D-Fender | manage* |
| 17:42.48 | Wayhigh | ok.. well.. thanks |
| 17:43.30 | [TK]D-Fender | Wayhigh: If you want to issue CLI commands (not * CLI), then thats what SSH is for. |
| 17:44.01 | De_Mon | oh, heh Kevin Mitnick is doing that video above |
| 17:44.13 | De_Mon | I didn't think he was alowed to touch computers still... |
| 17:44.27 | Qwell | that expired in like '05 or something |
| 17:45.14 | *** join/#asterisk joe (n=nnnnnnnj@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 17:45.19 | De_Mon | ah |
| 17:45.40 | De_Mon | his vim colors are fuuugly |
| 17:46.52 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 17:52.37 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 17:52.38 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 17:54.33 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 17:54.35 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:00.27 | *** join/#asterisk nicox (n=nicox@213-33-6-188.adsl.highway.telekom.at) |
| 18:02.55 | tzafrir_laptop | :colorscheme delek |
| 18:03.15 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 18:06.17 | waverly360 | Is it possible to setup asterisk queues to distribute different callers to different agents at the same exact time? For example, I have 20 agents, and 10 people waiting in the queue. All 20 agents can be idle and waiting for calls, but the asterisk queue will only ring one person at a time until an agent picks up that call. Now, I know I can use the ring-all strategy, but that's still only getting one person out of the queue at a time. I would li |
| 18:06.30 | implicit | http://www.engadget.com/2008/07/21/how-to-reveal-blocked-caller-id-info-a-video-guide-to-risky-beh/ |
| 18:06.36 | implicit | haha front page of engadget |
| 18:07.31 | [TK]D-Fender | waverly360: "autofill=yes" |
| 18:07.51 | [TK]D-Fender | implicit: [13:40]<De_Mon>http://www.engadget.com/2008/07/21/how-to-reveal-blocked-caller-id-info-a-video-guide-to-risky-beh/ |
| 18:07.53 | [TK]D-Fender | implicit: OLD |
| 18:08.04 | waverly360 | [TK]D-Fender: wow..that easy huh? |
| 18:08.44 | [TK]D-Fender | waverly360: Its amazing what you find when you actually read the samples, isn't it? :) |
| 18:08.49 | raz | hrm. i have a sipgate account and can receive calls via that. now i made a dialplan to forward such calls to my local sip phone, the rule is quite simple: Dial( SIP/2000 ) ... so, the phone rings but then - no voice in either direction. any idea anyone? :\ |
| 18:08.58 | waverly360 | [TK]D-Fender: crap..that's asterisk 1.4 only though... |
| 18:09.15 | [TK]D-Fender | raz: First guess is your system isn't set up to handle NAT properly. |
| 18:09.29 | [TK]D-Fender | waverly360: Time to wake up and smell the toast burning! |
| 18:09.43 | outtolunc | food? |
| 18:09.43 | waverly360 | [TK]D-Fender: if only it were that simple to upgrade :P |
| 18:10.18 | [TK]D-Fender | waverly360: When 1.4 has been out for over 2 years now, you should have long since been preparing yourself. |
| 18:10.43 | raz | [TK]D-Fender, hmm.. how would i check that? both parts of the link work individually ( local SIP phone vs asterisk / remote sip call vs asterisk ). just when i try to have them connect to each other... |
| 18:10.56 | waverly360 | [TK]D-Fender: difficult to do in my situation. It's one of those "Well we really need to do this..." but no one really gives you the time. |
| 18:11.20 | [TK]D-Fender | waverly360: 2 YEARS <- |
| 18:11.31 | [TK]D-Fender | waverly360: Sorry... just not buying it... |
| 18:11.43 | waverly360 | [TK]D-Fender: *shrugs* didn't ask you to. Just telling you how it is. |
| 18:11.44 | [TK]D-Fender | raz: Read up : |
| 18:11.47 | [TK]D-Fender | ~sipnat |
| 18:11.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:11.49 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 18:11.59 | raz | thx |
| 18:11.59 | [TK]D-Fender | waverly360: ok, fine, sure :) |
| 18:12.49 | De_Mon | waverly360 now you have -yet-another-reason- to find time |
| 18:13.02 | De_Mon | hopefully before 1.6 comes out, eh? |
| 18:13.36 | waverly360 | De_Mon: Yes I do have another reason now..I have plenty already, but also have reasons to stay on 1.2 |
| 18:16.12 | waverly360 | From what I understand, there are still plenty of people around sticking to 1.2 because of the instability of 1.4. |
| 18:16.24 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 18:16.29 | doolph | hello |
| 18:16.37 | doolph | can asterisk cli run shell commands? |
| 18:17.15 | M1s3ry | dogmeat, yes |
| 18:17.15 | doolph | ~centos52bug |
| 18:17.16 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 18:17.16 | [TK]D-Fender | doolph: type "help" and see. Then again, whats the point? * CLI si not mean to be used as UNIX CLI |
| 18:17.28 | M1s3ry | append ! before the command |
| 18:17.33 | M1s3ry | er... |
| 18:17.36 | M1s3ry | prepend |
| 18:17.43 | doolph | i tried and it didnt work |
| 18:17.54 | doolph | there's anything that i need to change |
| 18:18.21 | M1s3ry | cli command: help |
| 18:19.07 | keith4 | !foo seems to work fine for me |
| 18:19.25 | doolph | can i do !more zapata.conf |
| 18:19.27 | styelz | he must be using the cli via a console |
| 18:19.40 | doolph | something like that? |
| 18:19.54 | keith4 | sure. works fine for me |
| 18:20.02 | styelz | i mean tty |
| 18:20.48 | doolph | um yeah it works |
| 18:21.38 | styelz | with no console |
| 18:21.42 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 18:22.35 | styelz | ... |
| 18:23.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:23.26 | neurosys | I have a headache. Would anyone care to recommend a good ITSP aside from les.net? |
| 18:23.39 | keith4 | !itsp-list |
| 18:23.47 | keith4 | ~itsp-list |
| 18:23.57 | keith4 | sighs |
| 18:24.04 | raz | [TK]D-Fender, hmm i played around a bit but to no avail. well, thing is, both "parts" of the connection seem to work properly. that is: i can receive a call via the sipgate account to an asterisk voicebox and it will work. and i can call asterisk "internally" with the local sip phone, too. audio works in both directions in these cases. only when i get an inbound call via sipgate and forward that to the local phone (via Dial( SIP/2000 ) ) i get no voice. sigh.. |
| 18:24.09 | [TK]D-Fender | ~itsplist-us |
| 18:24.10 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:24.21 | styelz | man im lagged |
| 18:24.24 | neurosys | [TK]D-Fender: thx again :) |
| 18:24.45 | [TK]D-Fender | raz: Because you have most likely not followed all of the guide. PASTEBIN Your sip.conf masking only passwords. |
| 18:24.56 | [TK]D-Fender | neurosys: For what area? |
| 18:24.58 | raz | [TK]D-Fender, ok sec |
| 18:25.02 | Yourname`` | De_Mon: I think that clid thing would work in 1.4, no? |
| 18:25.03 | [TK]D-Fender | neurosys: (and you're welcome) |
| 18:25.56 | neurosys | [TK]D-Fender: Florida |
| 18:26.34 | [TK]D-Fender | neurosys: Give Vitelity & Teliax a look. |
| 18:26.49 | waverly360 | Anyone have a large subset of asterisk boxes on 1.4 that they can give me their "asterisk crash rate" of? |
| 18:26.58 | raz | [TK]D-Fender, http://rafb.net/p/fXYA1Y21.html |
| 18:27.41 | Yourname`` | waverly360: What do you mean crash rate? |
| 18:27.42 | raz | i have commented out all the nat'ing again because it didnt change anything |
| 18:28.05 | Kobaz | waverly360: i've got a 1.4.14 that's been going down once every two weeks or so... it doesn't completely crash, which is much worse, it gets into an odd state where it just can't handle calls |
| 18:28.17 | [TK]D-Fender | raz: And you've shown me a USELESS config. Go follow the guide I linked you and come back when you've gotten somewhere. |
| 18:28.25 | Kobaz | waverly360: i bumped it up to 1.4.20.1 over the weekend, we'll see what happens |
| 18:28.56 | waverly360 | Yourname``: Almost everyone I've listened to in here talks about how asterisk crashes occasionally. |
| 18:29.00 | Kobaz | waverly360: if it completely crashed out, at least it would have brought itself back up |
| 18:29.12 | waverly360 | Yourname``: For me, it's about twice a year..and safe_asterisk doesn't seem to recover it. |
| 18:29.13 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 18:29.59 | *** join/#asterisk mwalling_ (i=mwalling@you.dontlike.us) |
| 18:30.03 | waverly360 | Yourname``: It looks like I'm going to have to start figuring out what's going to change when I upgrade to 1.4, and I'd like to know which version has proven to be the most stable. |
| 18:30.15 | waverly360 | Yourname``: ...according to everyone here of course. |
| 18:30.51 | Kobaz | waverly360: other than that one box, it's been pretty good... although i've ran into the zombie state some other times on rare occasion |
| 18:31.34 | waverly360 | Kobaz: So you have other 1.4.14 boxes that are pretty stable? |
| 18:32.47 | Kobaz | waverly360: yeah |
| 18:33.13 | Kobaz | waverly360: it tends to happen when you have high cpu load |
| 18:33.19 | Yourname`` | waverly360: I have a scheduled restart at 4am every night, so it's working fine so far. |
| 18:33.47 | Kobaz | waverly360: when the box itself (non asterisk processes) start getting high is usage, asterisk sometimes will start chewing up 100% cpu on its own, and will not handle calls |
| 18:34.05 | waverly360 | Yourname``: I have a few customers who are 24/7 users. Can't really schedule a restart that often. |
| 18:34.32 | *** join/#asterisk angryuser (n=sdfsdf@78.115.236.230) |
| 18:34.54 | Kobaz | waverly360: i've had the same problem with 1.4.17 and 1.4.18 |
| 18:35.04 | Kobaz | waverly360: i'll see how it goes with 1.4.20.1 |
| 18:35.17 | waverly360 | I'd be interested in knowing how it turns out. |
| 18:35.48 | Yourname`` | waverly360: Ah, I haven't really witnessed crashes unlike the once/twice in a long time where only a complete reboot works. |
| 18:36.43 | [netman] | could anybody help me with vicidial please? |
| 18:37.13 | Yourname`` | I think we should have a bounty or something to actually bury vicidial. |
| 18:39.29 | keith4 | is that a commercial product? |
| 18:39.59 | [netman] | why Yourname``? |
| 18:40.06 | Kobaz | oooo |
| 18:40.14 | Kobaz | there's some good fixes in 1.4.21.1 |
| 18:40.19 | Kobaz | r75053-75067 |
| 18:41.14 | Kobaz | When using a TDM400P (and probably other analog cards) there was a chance that you could hang up and pick the phone back up where it has been long enough to be not considered a flash hook, but too soon such that the device reports that it is busy and the person on the phone will only hear silence. This patch makes chan_zap more tolerant of this and gives the device a couple of seconds to succeed so the person on the phone happily gets their dialtone. |
| 18:41.23 | Kobaz | that's always bugged me |
| 18:41.30 | Kobaz | heh, nice that it's fixed |
| 18:42.10 | waverly360 | So if there were a 1.4 release that I should steer away from, or towards...what would you guys suggest? |
| 18:42.38 | keith4 | uh, towards the latest release? |
| 18:42.42 | keith4 | is that a trick question? |
| 18:42.56 | waverly360 | Latest release doesn't always mean the best release |
| 18:43.14 | keith4 | nobody is suggesting that you upgrade to 1.6b9 |
| 18:43.15 | pigpen | waverly360, current latest is good. |
| 18:43.33 | waverly360 | pigpen: Cool, thanks |
| 18:43.37 | keith4 | latest stable is probably a good direction to take, though |
| 18:43.51 | bijit | I tried make config on my slack bos on asterisk and it says slackware is not supported. Do I have to start asteisk manually each time? |
| 18:44.09 | pigpen | we have it running on several Dell 6850's with about 60 - 250 sip extensions. |
| 18:45.24 | *** join/#asterisk trelane (i=trelane@lan.trelane.net) |
| 18:45.37 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:46.05 | trelane | I need a device that speaks sip with asterisk and can handle 25-50 copper pair for analog phone (FX's) |
| 18:46.08 | trelane | FXS even |
| 18:47.04 | trelane | amp connections preferred |
| 18:47.31 | [TK]D-Fender | bijit: Make your own script for ir or shove it in /etc/rc.d/rc.local |
| 18:48.00 | [TK]D-Fender | trelane: Mediatrix 1124 / AudioCodes MP-124 |
| 18:49.49 | keith4 | is there a length limit on voicemail passwords? |
| 18:49.57 | bijit | [TK]D-Fender: yeah i think the best way is to add it to rc.local.. ty |
| 18:50.23 | [TK]D-Fender | bijit: Make sure to init Zaptel first in there |
| 18:50.33 | *** join/#asterisk eskimofo (n=eskimofo@77.119.111.249) |
| 18:51.14 | *** join/#asterisk unafilliate (n=md5@unaffiliated/unafilliate) |
| 18:51.14 | bijit | zaptel is in init.d |
| 18:51.58 | [TK]D-Fender | bijit: OK, I always ran both from rc.local myself. then again, I was never even particularly good with BSD init's |
| 18:52.36 | *** join/#asterisk jonsmith1982 (n=jon@82-47-175-34.cable.ubr01.donc.blueyonder.co.uk) |
| 18:54.40 | *** join/#asterisk mwalling1 (i=mwalling@you.dontlike.us) |
| 18:54.54 | bijit | [TK]D-Fender: I will just try it like this..if it doesn't work then I will just run both from rc.local :) |
| 18:55.22 | [TK]D-Fender | bijit: Ok, Figure you're well on our way now. Let us know |
| 18:55.57 | angryuser | just wanted to say, i hate exchange server, thank you |
| 18:56.26 | Kobaz | haha |
| 18:57.52 | jeev | f3nd3r |
| 18:58.17 | jeev | the 330's are coming this week |
| 18:58.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:00.01 | [TK]D-Fender | jeev: \o/ |
| 19:01.00 | Kobaz | heh |
| 19:02.35 | jeev | what the hell is that supposed to mean! |
| 19:02.41 | jeev | is that your asshole ? |
| 19:03.00 | seanbright | no, it's a backslash, a lowercase 'O' and a slash |
| 19:03.24 | keith4 | he's cheering, with arms in the air |
| 19:04.17 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 19:05.02 | jeev | oh |
| 19:05.02 | jeev | hahah |
| 19:06.20 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 19:09.07 | jaytee | <jeev> what the hell is that supposed to mean! <jeev> is that your asshole ? <<<<< with this kind of content I may just cancel my cable, who needs Comedy Central when I've got #asterisk |
| 19:09.27 | jeev | i thought he was hating on my hemmorhoids. |
| 19:14.16 | styelz | (_!_) |
| 19:22.33 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.2) |
| 19:23.03 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:23.37 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:24.19 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 19:24.21 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 19:26.51 | [TK]D-Fender | jeev: Yes... we all know you're just a pain in the ass ;) |
| 19:27.01 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 19:33.35 | neurosys | [TK]D-Fender: What range of ports need to be avail. for a sip based connection with ITSP? |
| 19:34.07 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-82-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:34.22 | [TK]D-Fender | neurosys: typcailly 5060,10000-20000 all UDP |
| 19:36.55 | *** join/#asterisk sakajawebe (n=chazz@216.207.245.1) |
| 19:40.31 | *** join/#asterisk [netman] (n=netman@68.Red-88-8-164.dynamicIP.rima-tde.net) |
| 19:43.06 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 19:45.08 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:46.43 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 19:50.41 | *** join/#asterisk gardo (n=gardo@121.97.215.164) |
| 19:50.50 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 19:51.23 | GhOnDiE | ~centos52bug |
| 19:51.25 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 19:51.44 | *** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 19:52.58 | Qapf | a question, how do i control in asterisk which files in /etc/asterisk that it will parse? |
| 19:55.40 | Qwell | Qapf: can you rephrase your question? |
| 19:56.27 | Qapf | my install of asterisk lacked codecs.conf, i copied the sample and put it in, but i want to make sure asterisk actually parses it and reads the lines in it, is there a mechanism by which you specify to asterisk which config files it looks at, or does it just read anything in /etc/asterisk |
| 19:58.35 | *** join/#asterisk Fiapo-CE (n=dot@jupiter.hapvida.com.br) |
| 19:58.45 | *** part/#asterisk Fiapo-CE (n=dot@jupiter.hapvida.com.br) |
| 19:59.22 | [TK]D-Fender | Qapf: use modules.conf to control which modules get loaded at startup, and that will disable that module from loading and needing its related config |
| 19:59.39 | Qapf | thanks [TK]D-Fender |
| 20:02.00 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 20:03.07 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 20:04.59 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 20:05.00 | *** mode/#asterisk [+o twisted] by ChanServ |
| 20:09.24 | *** part/#asterisk nicox (n=nicox@213-33-6-188.adsl.highway.telekom.at) |
| 20:09.54 | twisted | shhhh |
| 20:10.13 | *** join/#asterisk derelm (n=derelm@p5B23F283.dip.t-dialin.net) |
| 20:12.55 | outtolunc | checks the mail box for an orange fly swatter |
| 20:14.49 | tompaw | hi guys |
| 20:15.20 | tompaw | I desperately need to know, how to limit the number of connections that one sip peer may perform at one time. |
| 20:15.29 | tompaw | so like to limit it to 20 calls. |
| 20:15.32 | tompaw | (incoming) |
| 20:15.44 | tompaw | is it possible? I can't find much in sip.conf documentation. |
| 20:16.03 | GhOnDiE | hi yes it is |
| 20:16.12 | GhOnDiE | 1 moment will post it up |
| 20:16.23 | GhOnDiE | once i remember what it is |
| 20:16.53 | tompaw | is this the busy-level? |
| 20:17.23 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:17.43 | GhOnDiE | it basically means that 1 sip peer can have a certain ammount of active connections |
| 20:17.50 | GhOnDiE | so yes busy level i guess |
| 20:18.57 | GhOnDiE | what are you trying to do exactly? |
| 20:19.14 | tompaw | to allow 20 incoming connections, no more. |
| 20:19.19 | GhOnDiE | ok |
| 20:19.21 | tompaw | busy-level = 20 should do the trick you think? |
| 20:20.11 | GhOnDiE | no |
| 20:20.21 | GhOnDiE | all that does is make it so that the user is busy |
| 20:20.30 | GhOnDiE | Asterisk sip call-limit = number : Number of simultaneous calls through this user/peer. |
| 20:20.34 | tompaw | how about call-limit = 20 AND busy-leve = 20? |
| 20:20.38 | GhOnDiE | unless thats what your aiming for? |
| 20:20.48 | tompaw | well, I have a peer, right? |
| 20:20.52 | GhOnDiE | ok |
| 20:21.00 | tompaw | he calls my asterisk for GSM termination |
| 20:21.00 | tompaw | now |
| 20:21.10 | tompaw | I want to limit the amount of connections he can perform at the same time |
| 20:21.11 | tompaw | to 20. |
| 20:21.27 | tompaw | if he tries 21, I'd like that 21st connection to hear a busy tone and receive SIP error 34. |
| 20:21.32 | GhOnDiE | in that case you need call-limit |
| 20:21.52 | GhOnDiE | yeah your busy limit 20 and call limit 20 should work fine |
| 20:21.58 | GhOnDiE | give it a try |
| 20:22.48 | GhOnDiE | call-limit is the maximum calls that can be made |
| 20:22.52 | GhOnDiE | not taken and made |
| 20:23.09 | GhOnDiE | so you will need to use busy-limit to limit the incoming calls |
| 20:23.25 | GhOnDiE | thats how i understand the sip.conf document |
| 20:23.28 | tompaw | but this peer doesn't MAKE calls, it just receives them |
| 20:23.40 | tompaw | ok. will set them both and I'll see |
| 20:24.09 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:24.16 | tompaw | thank you very much for your help. |
| 20:24.23 | GhOnDiE | ok well yeah busy-limit definately seems like the ideal one to use |
| 20:24.49 | tompaw | thanks once again. now I just wait for the traffic flow :-) |
| 20:24.52 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 20:25.02 | deeperror | is there a way to take 2 callers in a meetme conf and turn them back into a bridged call? |
| 20:25.04 | *** join/#asterisk gego (n=gego@p5490DDCB.dip.t-dialin.net) |
| 20:26.00 | [TK]D-Fender | deeperror: Probably through AMI. |
| 20:26.10 | *** join/#asterisk devhen|Work (n=devhen@216.194.118.110) |
| 20:26.25 | deeperror | [TK]D-Fender, that is wha t i'm looking at but not sure on which action to use |
| 20:26.38 | [TK]D-Fender | deeperror: Look at Redirect first. |
| 20:27.22 | [TK]D-Fender | deeperror: What is your funcional goal of doing this? |
| 20:29.47 | deeperror | I call up someone and while talking to them I push a button and redirect them into a conference and hangup...(they hear moh)...i call someone else and redirect both of us into the conf making a 3way call...i then hangup this all works...but i would like when I hangup to also end the conf and bring them back into a bridged call since the conf is no longer needed. |
| 20:29.49 | GhOnDiE | i guess if you were in a full conf and then only have 2 left? |
| 20:30.10 | GhOnDiE | cant you just do an attended transfer |
| 20:30.11 | GhOnDiE | ? |
| 20:30.32 | deeperror | has to be a hot transfer |
| 20:30.41 | [TK]D-Fender | deeperror: Why not jsut do a 3-way call on your phone>? |
| 20:31.07 | GhOnDiE | how long are you on the phone for in teh 3 way call? |
| 20:31.21 | GhOnDiE | the person on the transfer would be on hold all the time |
| 20:31.24 | deeperror | because were not using phones...using softphone and a crm that has no asterisk support and i'm writing a web service to give it such support |
| 20:31.32 | GhOnDiE | ahh |
| 20:31.33 | GhOnDiE | ok |
| 20:31.34 | *** join/#asterisk viperdudeuk (n=chatzill@195.74.96.121) |
| 20:32.43 | deeperror | to make things short...everything works 100% but I fear having several conferences open when i could be bridging calls could add a lot of load on the box so was wanting to make it work better and not leave the conf open if we are really just in a standard call at that point |
| 20:33.06 | [TK]D-Fender | eek |
| 20:33.11 | [TK]D-Fender | ok, well I'm out for now |
| 20:33.13 | [TK]D-Fender | later all |
| 20:33.16 | deeperror | l8 |
| 20:34.00 | GhOnDiE | does seem quite a trivial thing |
| 20:34.08 | GhOnDiE | im not realy sure how you could do that tbh |
| 20:35.10 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@24-231-253-65.dhcp.aldl.mi.charter.com) |
| 20:36.41 | *** join/#asterisk anthm (n=anthm@173-7-35-9.area4.spcsdns.net) |
| 20:38.42 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 20:39.38 | *** join/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 20:40.21 | sgtpepper | Hello everyone, I'm having a weird issue with unicall |
| 20:40.25 | sgtpepper | I'm getting a MFCR2_PROTOCOL_FAIL_T3_TIMEOUT |
| 20:40.52 | derelm | i am trying to integrate an external sip-client as a local extension into my asterisk, but when i am calling that extension i don't get to hear any sound and neither does the one on the extension. any hints? |
| 20:41.19 | angryuser | derelm port routing |
| 20:42.07 | derelm | 5060 is routed to the asterisk server but the other client is behind nat while i am in the same net as the asterisk server |
| 20:42.10 | moy | sgtpepper: youre getting that for any call? |
| 20:42.28 | angryuser | derelm udp ports routed ? |
| 20:42.35 | sgtpepper | moy.. for in - calls |
| 20:42.44 | sgtpepper | I can dial out perfectly |
| 20:43.05 | moy | pastebin the debug output for an incoming call |
| 20:43.08 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:43.25 | derelm | angryuser: the "external" client or to my asterisk server? in the latter case, yes udp+tcp 5060-5070 get routed to my asterisk server |
| 20:44.36 | angryuser | derelm external cliend does not any port routed, by defaul asterisk need 5060 tcp and 1000-20000(udp) port routed from internet |
| 20:44.48 | sgtpepper | moy http://pastebin.com/m304250d0 |
| 20:44.49 | angryuser | 10000-20000 |
| 20:45.22 | angryuser | derelm also enable qualify = yes to keep nat ports opened |
| 20:45.33 | moy | sgtpepper: need more details, you need to set loglevel=255 in unicall.conf and all debugging levels in logger.conf for device console |
| 20:45.45 | derelm | angryuser: i'll retry with the latter ports forwarded. what about canreinvite and friends |
| 20:46.17 | angryuser | derelm canreinvite = no for external,, friends ? you can have friends |
| 20:46.44 | angryuser | and girlfriends |
| 20:46.54 | derelm | angryuser: canreinvite & co :) |
| 20:47.23 | angryuser | open ports and test |
| 20:47.29 | derelm | ok thanks, will do |
| 20:49.35 | *** join/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 20:49.43 | sgtpepper | ping moy |
| 20:50.31 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:50.34 | moy | sgtpepper: didn't you see my msg? |
| 20:50.44 | sgtpepper | nope... I fell |
| 20:52.07 | moy | sgtpepper: need more details, you need to set loglevel=255 in unicall.conf and all debugging levels in logger.conf for device console |
| 20:52.30 | sgtpepper | loglevel its 255 in unicall.conf |
| 20:52.42 | moy | then you are just missing all debugging levels in console device |
| 20:52.47 | moy | at logger.conf |
| 20:53.39 | sgtpepper | i enabled debug in logger.conf |
| 20:53.41 | sgtpepper | hold on a sec |
| 20:53.44 | sgtpepper | I'm trying again |
| 20:54.47 | *** join/#asterisk n3hxs (n=HAMming@66.251.14.130) |
| 20:55.09 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:55.51 | *** part/#asterisk gego (n=gego@p5490DDCB.dip.t-dialin.net) |
| 20:55.59 | sgtpepper | moy check http://pastebin.com/m6faadfde |
| 20:56.04 | sgtpepper | i've debugging now |
| 20:57.13 | sgtpepper | /MSG NickServ VERIFY REGISTER sgtpeppe1 geievpdmdkbk |
| 20:57.29 | sgtpepper | I hate pidgin |
| 20:57.45 | *** join/#asterisk icel (n=dan@75.150.16.118) |
| 20:58.07 | twisted | ahhaha |
| 20:58.16 | moy | sgtpepper: are you using Mexico variant? |
| 20:58.34 | sgtpepper | trying mexico variant |
| 20:58.40 | sgtpepper | is connected to a panasonic pabx |
| 20:58.45 | sgtpepper | mx variant seemed to work |
| 20:59.21 | sgtpepper | actually in-going calls are working |
| 20:59.30 | sgtpepper | sorry |
| 20:59.37 | sgtpepper | outgoing calls.. from asterisk point of view |
| 21:00.04 | moy | sgtpepper: in fact I don't see the error you mention, so, wrong log? |
| 21:00.20 | sgtpepper | I don't know whi I have a new error now |
| 21:05.26 | sgtpepper | moy http://pastebin.com/mb4ea178 |
| 21:09.22 | moy | sgtpepper: Unexpected MF6 signal usually means you have the wrong R2 variant |
| 21:09.37 | sgtpepper | which should be for this panasonic system? |
| 21:11.00 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:12.46 | moy | dunno |
| 21:13.03 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:13.09 | moy | you have to try different variants or get someone who gives support on that piece of shit |
| 21:13.19 | moy | or read the manual or whatever |
| 21:13.29 | moy | I have no experience with traditional PBXs |
| 21:13.37 | sgtpepper | moy: I though that |
| 21:17.10 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 21:18.21 | CoffeeIV | ~centos52bug |
| 21:18.23 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 21:21.32 | CoffeeIV | I have an * version 1.2.10 on which I just set up a voicemail using the built-in voicemail. The messages that are emailed as attachements have way to quiet a volume. The calls come in on a VoIP connection, so I don't think the gain argument to Voicemail works, and setting VOLGAIN in voicemail.conf didn't seem to work. |
| 21:23.20 | [TK]D-Fender | CoffeeIV: You have to reload the module or restart * for that to take effect, FYI |
| 21:23.56 | CoffeeIV | [TK]D-Fender: I think I restarted, but let me do that and test again |
| 21:28.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:29.49 | *** join/#asterisk n3hxs (n=HAMming@66.251.14.130) |
| 21:30.35 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 21:30.58 | *** join/#asterisk tloges (n=tiagolog@mail.abyzti.com.br) |
| 21:35.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:36.49 | Ryushin | I was thinking of playing with a pre built asterisk distro. I have something against RedHat based distro's, so Trixbox was out. Is Elastix rpm based as well? |
| 21:37.22 | styelz | yea it runs on centos |
| 21:38.06 | GhOnDiE | i think that most of them run on rpm based distro's |
| 21:38.15 | Ryushin | So much for that then. |
| 21:38.18 | *** join/#asterisk akio (n=akio@181-161.187-72.tampabay.res.rr.com) |
| 21:38.21 | GhOnDiE | majority of them on centos i think |
| 21:38.22 | GhOnDiE | ? |
| 21:38.23 | Ryushin | I'll stay on Debian then. |
| 21:38.24 | styelz | install freepbx on your own os |
| 21:38.53 | GhOnDiE | you get much better control over it then and you also learn more about it |
| 21:38.58 | Ryushin | Oh, is freepbx the gui that they use then. |
| 21:39.07 | styelz | yes |
| 21:39.09 | GhOnDiE | yeah |
| 21:39.13 | Ryushin | Even better then. |
| 21:39.42 | Ryushin | Can freepbx use flat files, or does it need a database? |
| 21:39.51 | GhOnDiE | database only i think |
| 21:39.56 | styelz | it uses mysql |
| 21:40.01 | angryuser | why gui anyway ? |
| 21:40.12 | *** join/#asterisk Psychobilly (n=moi@adsl144-78.kln.forthnet.gr) |
| 21:40.31 | Ryushin | Hmmm.... I'm been cli asterisk for the last 3 years. I wanted to give some web based controls to the users. |
| 21:40.49 | angryuser | ah to the users, so elastix is your choice then |
| 21:41.04 | styelz | elastix is best for multi user |
| 21:41.05 | Ryushin | Well, it isn't, so I won't use redhat. |
| 21:41.24 | styelz | try port it |
| 21:41.29 | styelz | its just php and guff |
| 21:41.43 | styelz | not too hard |
| 21:41.48 | Ryushin | styelz: Then that is probably best. |
| 21:41.51 | angryuser | styelz a lof of php and stuff |
| 21:41.56 | angryuser | lot* |
| 21:41.58 | [TK]D-Fender | Ryushin: there is no "half-way" with FreePBX. It completely owns you once you switch to it |
| 21:42.14 | Ryushin | And I don't like being owned. |
| 21:42.23 | styelz | few days work.. fun |
| 21:42.33 | [TK]D-Fender | Ryushin: Go take a look at AsteriskGUI then. See if it works for you |
| 21:42.35 | akio | I have searched all over the net to try to find a way to fix a problem with an SIP 404 for incoming calls |
| 21:42.45 | [netman] | how could I debug Meetme? |
| 21:42.47 | akio | i don't have asterisk |
| 21:42.48 | Ryushin | It would be nice to find a hi-bred. I want to keep my flat files as it easier to admin. |
| 21:43.14 | Ryushin | [TK]D-Fender: I'll check out AsteriskGUI. |
| 21:43.15 | [TK]D-Fender | akio: 2 things can 404, 1: the peer (can't tell which account entry to auth against) or 2: the incoming #. |
| 21:43.23 | [netman] | I use ztdummy, I typed ztcfg -vvv and I don't see any error, but Meetme doesn't work for me |
| 21:43.34 | [TK]D-Fender | akio: If you don't have *, then you're in the wrong place. |
| 21:43.37 | akio | how do i auth? |
| 21:43.45 | akio | i know im in the wrong place |
| 21:43.55 | akio | but i can't find the right place |
| 21:44.20 | [TK]D-Fender | akio: Not our problem. Go see if there is a channelf or whatever it is you are using. |
| 21:44.27 | [TK]D-Fender | for* |
| 21:44.52 | *** part/#asterisk akio (n=akio@181-161.187-72.tampabay.res.rr.com) |
| 21:44.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:46.23 | Ryushin | [TK]D-Fender: Do you use any kind of gui? Or you just a pure cli guy like myself? |
| 21:47.07 | [TK]D-Fender | Ryushin: My office uses one because it was the selling point that got us off a proprietary hardware platform. I do not use one anywhere else. |
| 21:48.11 | Ryushin | I don't like gui's because they are a crutch. But sometimes it's necessary to see everything in a clearer manner, such as a IPTables frontend. |
| 21:48.17 | angryuser | maybe TK is TeamKiller need to find the answer |
| 21:49.28 | [TK]D-Fender | Ryushin: well * is so flexible you'd have to be running your system like a toaster to use one typically. |
| 21:53.07 | Ryushin | Then I'm going to keep doing it the same way I've been doing since I started. I just wanted a method that users could change things for their extension. |
| 21:53.17 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 21:54.15 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 21:59.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:01.34 | *** join/#asterisk eclark (n=eclark@75-164-248-205.ptld.qwest.net) |
| 22:02.27 | icel | where can i find out what letters (like s,h,t) mean (ex: exten => 1,t,1,Blah() ) |
| 22:03.10 | [TK]D-Fender | icel: Go read about "Asterisk Standard Extensions" on the WIKI and in the BOOK |
| 22:03.12 | [TK]D-Fender | ~wikis |
| 22:03.12 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:03.13 | [TK]D-Fender | ~book |
| 22:03.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:03.17 | icel | thx |
| 22:03.27 | CoffeeIV | from looking at app_voicemail.c, I suspect the "volgain" option in voicemail.conf was put in after asterisk version 1.2.10 |
| 22:03.50 | eclark | on my outgoing sip calls, it seems like the line is being "picked up" as soon as the ringing starts on the other end, which makes my dialplan act funny. is this normal? |
| 22:04.50 | [TK]D-Fender | eclark: Depends what your call is going out over. |
| 22:05.32 | eclark | [TK]D-Fender: so i can possibly fix this with a different carrier? i'm using sipphone/gizmo right now |
| 22:06.02 | [TK]D-Fender | eclark: Your carrier may answer *'s call and THEn try its own outbound leg. |
| 22:06.33 | eclark | okay, that makes sense. |
| 22:06.51 | eclark | thanks! |
| 22:31.10 | twisted | does not like this heat crap |
| 22:32.57 | [TK]D-Fender | <PROTECTED> |
| 22:34.21 | *** join/#asterisk Fiapo-CE (i=Fiapo-CE@201.70.137.40) |
| 22:35.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:37.54 | *** join/#asterisk [hC] (n=hardcore@216.251.157.146) |
| 22:38.47 | jaytee | I thought I truly understood the meaning of pain but I was mistaken, I'd never dealt with setting up grammars and code for an IVR using LumenVox. |
| 22:39.06 | twisted | lol |
| 22:42.13 | Wayhigh | damn it.. I thought my computer had been pwned but it turnedout to be a stale nfs handle that hosed everything up |
| 22:42.49 | twisted | pwnt by nfs |
| 22:43.24 | Wayhigh | I had to haxor into my own box remotely in order to fix it |
| 22:46.38 | _mm_ | jaytee: that bad? |
| 22:47.06 | jaytee | well, just not as easy as configuring the speech rec enabled IVR system we currently have |
| 22:47.37 | _mm_ | why'd you switch if you had one? |
| 22:48.54 | jaytee | we're moving from a Nortel Meridian PBX to an Asterisk VOIP PBX and we will need to replace the Liaison IVR system from Nuance (formerly owned by Locus) |
| 22:49.13 | _mm_ | gotcha |
| 22:49.47 | jeev | FENDER |
| 22:49.52 | jeev | \o/ |
| 22:53.07 | angryuser | <PROTECTED> |
| 22:53.26 | angryuser | ^^^legs |
| 22:54.09 | angryuser | \°/ |
| 22:54.26 | angryuser | '/ \ |
| 22:54.59 | outtolunc | sliced in half with blood drop? |
| 22:55.33 | angryuser | nasty accident |
| 22:55.43 | outtolunc | see kids, this is what happens when you use asterisk near trains <G> |
| 22:59.44 | bbryant | `itsp |
| 22:59.53 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 22:59.58 | bbryant | `help |
| 23:00.05 | bbryant | ~itsp |
| 23:00.06 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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 |
| 23:00.07 | orionr | ~help |
| 23:00.13 | bbryant | ~itsplist-us |
| 23:00.14 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 23:00.21 | bbryant | orionr: --^ |
| 23:01.38 | GhOnDiE | ~itsplist-uk |
| 23:01.38 | jbot | extra, extra, read all about it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 23:03.04 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 23:15.58 | devhen|Work | can anyone enlighten me as to how to do call snooping ? |
| 23:20.15 | [TK]D-Fender | Hrm, UK... will have to clean that one up. |
| 23:28.54 | devhen|Work | anyone have any good links for tutorials on setting up/using call snooping ? |
| 23:29.47 | angryuser | devhen|Work under cli core show application chanspy() |
| 23:30.18 | angryuser | or it was a functions ? whatever |
| 23:31.44 | angryuser | ~enter |
| 23:31.45 | jbot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 23:32.26 | GhOnDiE | lol |
| 23:32.43 | angryuser | it's for other channel so dont worry ;=) |
| 23:33.20 | tzafrir_laptop | angryuser, you can also use a private message to jbot |
| 23:33.21 | GhOnDiE | fair enough |
| 23:33.31 | GhOnDiE | quite funny what the reply was though |
| 23:33.36 | tzafrir_laptop | <PROTECTED> |
| 23:33.42 | GhOnDiE | a sarcastic bot |
| 23:35.01 | tzafrir_laptop | ~bot |
| 23:35.01 | jbot | I ain't no stinkin' bot. I am a finely tuned and hand crafted tool. Oh wait... I guess I am a bot (that you should not abuse). |
| 23:35.13 | tzafrir_laptop | Good night :-) |
| 23:37.43 | GhOnDiE | classic |
| 23:37.45 | GhOnDiE | night all |
| 23:37.58 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:41.15 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 23:42.51 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 23:47.07 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 23:54.47 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 23:58.12 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 23:58.55 | Ritzerisk | im trying to work with IAXmodem and ... have no idea where to even look for like a macro |
| 23:59.16 | kash | how does IAX do video? |
| 23:59.42 | Ritzerisk | call comes in but the Caller From the Fax machine enters the number then PPPP for pauses then a control number then needs to send that off via email |
| 23:59.53 | *** join/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
