| 00:00.07 | cryzeck | no changes.. :-( |
| 00:00.07 | Uther_P | IF your provider is sending inband dtmf.. that only works with g711 codec (ulaw and alaw), so if you were defaulting to using a different codec, then the dtmf wouldn't work even if you had it configured as inband |
| 00:00.11 | wankel | yay. i managed to make nickserv happy so i could get in here. |
| 00:00.18 | *** join/#asterisk D1ng0 (~dingo@130.205.8.67.cfl.rr.com) |
| 00:00.22 | Uther_P | ah hrm... bitch at your provider man |
| 00:00.38 | cryzeck | wankel: congrats |
| 00:00.42 | cryzeck | Uther_P: will do. :) |
| 00:00.43 | Uther_P | cryzeck: call them and ask what dtmf they use |
| 00:00.54 | Uther_P | cryzeck: and what codec's they recomment |
| 00:00.54 | wankel | cryzeck: it's the small accomplishments in life that are the most fulfilling :) |
| 00:02.18 | trymlap | Im just trying to accept a call: http://pastebot.nd.edu/86 |
| 00:02.37 | trymlap | any ideas whats wrong? |
| 00:02.44 | Uther_P | lemmie look |
| 00:03.14 | trymlap | let me know if you need some pastes from config files |
| 00:04.29 | Uther_P | you have no common codecs |
| 00:04.48 | Uther_P | wait, nevermind |
| 00:04.50 | Uther_P | I read wrong |
| 00:05.21 | trymlap | the authentication thingie seems to be the "Problem" |
| 00:05.25 | sandboy007 | could anyone help me with sip/nat on asterisk here ? |
| 00:05.52 | *** part/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 00:06.17 | sandboy007 | I have a few questions |
| 00:07.53 | wankel | anyone running the latest cvs? i want to play with realtime. nothing much in the bug database, so i assume it's fairly stable. |
| 00:08.15 | trymlap | Uther_P: any ideas? |
| 00:09.42 | wankel | wow, that was fast. my patch got put into cvs in... under 15 minutes. :) |
| 00:11.15 | sandboy007 | could anyone give me some pointers on asterisk behind a nat connecting to a sip server |
| 00:11.16 | sandboy007 | please |
| 00:11.36 | Silik0n | does asterisk support BGMGCP? |
| 00:11.42 | Silik0n | MGCP rather |
| 00:11.51 | kpfleming | chan_mgcp |
| 00:12.06 | twisted | what about BGP? |
| 00:12.18 | Silik0n | yeah |
| 00:12.26 | Silik0n | someone right chang+bgp |
| 00:12.29 | trymlap | nevermind |
| 00:12.30 | trymlap | I fouhnd out |
| 00:12.33 | Silik0n | errr chan_bgp~ |
| 00:12.40 | bkw_ | haha |
| 00:12.46 | twisted | wouldn't it be a res_bgp? |
| 00:12.50 | wankel | sandboy: set externip and localnet/localmask in sip.conf? |
| 00:13.12 | sandboy007 | wankel, do I need to setup anything about stun ? |
| 00:13.42 | wankel | i've never used stun. i just fix the nat instead of proxying, usually. |
| 00:13.55 | twisted | chan_fufme will allow use of http://www.fu-fme.com/ with asterisk |
| 00:13.59 | kpfleming | or use phones that support RFC3581, then you don't need STUN at all |
| 00:14.29 | sandboy007 | I jsut have a home network behind NAT, and the outbound I jsut want to register with sipphone.com |
| 00:14.45 | twisted | you can dial based on the number of shakes |
| 00:15.00 | wankel | you'll need to be able to port-forward on your nat gateway as well, so if you don't have any control over it you'll have to tunnel or talk to a proxy on the other side that handles rewriting and symmetric RTP |
| 00:15.36 | Silik0n | twisted is that anything like http://www.vibrators-faq.com/hightech/internet.html |
| 00:15.49 | Zorix | vibrators? |
| 00:15.53 | twisted | Silik0n, sorta |
| 00:15.53 | sandboy007 | wankel, what ports then ? |
| 00:16.07 | twisted | i could probally add a usb driver to chan_fufme |
| 00:16.10 | sandboy007 | ok, I'll read the doco |
| 00:16.18 | wankel | if you've got a linksys or something like that you can just forward whatever ports you want to use for Sip to the * box and then have * present itself as the external ip. take a look at http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example |
| 00:16.20 | twisted | oops |
| 00:16.34 | Zorix | lmao http://opendildonics.org/ |
| 00:16.56 | *** join/#asterisk nullogic (~tcarter@c-24-98-72-110.atl.client2.attbi.com) |
| 00:16.58 | Silik0n | make sure chan_fufme doesnt get screwed when both ends are behind nat |
| 00:18.07 | ardor | Zorix: hook it up to * to work with dtfm tones |
| 00:18.08 | *** join/#asterisk chuckster (~chatzilla@adsl-69-225-161-234.dsl.pltn13.pacbell.net) |
| 00:18.17 | Zorix | lmao |
| 00:18.22 | ardor | new meaning of phone sex. |
| 00:18.24 | twisted | Silik0n, good point |
| 00:18.33 | twisted | i think i'll make it work over IAX2's unused IE's |
| 00:18.41 | ardor | 'oh i love it when you hold down the 8 digit' |
| 00:19.00 | Zorix | project for today: figure out voicemail |
| 00:19.03 | Zorix | heheh |
| 00:19.46 | *** join/#asterisk Xenesis (~ghost@233-102-118-80.kaptech.net) |
| 00:20.12 | trymlap | "Jan 2 01:20:27 NOTICE[1146719152]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
| 00:20.14 | trymlap | what does that mean? |
| 00:20.22 | Xenesis | hello all |
| 00:20.23 | PTG123 | try: use gsm it will fix that message :) |
| 00:20.29 | PTG123 | means your codec doesn't support silence supression |
| 00:20.32 | PTG123 | can we put that in the ~ |
| 00:20.33 | *** join/#asterisk bratner (~bman@bzq-218-152-167.cablep.bezeqint.net) |
| 00:20.34 | PTG123 | ? :) |
| 00:20.44 | PTG123 | or turn of silence supression |
| 00:21.42 | trymlap | ok.. so which codecs should I disable? |
| 00:22.17 | bkw_ | its not a codec |
| 00:22.18 | PTG123 | well what are you using for your phone? |
| 00:22.22 | bkw_ | turn off silence supression |
| 00:22.26 | bkw_ | ie transmit silence = yes |
| 00:22.34 | PTG123 | thats waht i said bkw :) or you can use a codec that supports it, like gsm |
| 00:22.35 | Silik0n | disable all but G729 |
| 00:22.48 | bkw_ | hahahhaha |
| 00:22.57 | bratner | i'm using kphone to connect to * (SIP) on localhost. how can i improve sound quality (bandwidth is not an issue here)? |
| 00:23.02 | twisted | *sigh* |
| 00:23.03 | Strom_TM | bratner: ulaw |
| 00:23.06 | Strom_TM | ulaw ulaw ulaw |
| 00:23.13 | bratner | Storm_tm: thanks |
| 00:23.14 | Strom_TM | and let's not forget ulaw |
| 00:23.30 | PTG123 | http://www.ietf.org/rfc/rfc3389.txt |
| 00:23.40 | PTG123 | bkw_: http://www.ietf.org/rfc/rfc3389.txt |
| 00:23.54 | Strom_TM | why do people call me "storm"? |
| 00:24.01 | Strom_TM | i'm not a weather pattern |
| 00:24.04 | bkw_ | yes dear |
| 00:24.06 | bkw_ | I know about it |
| 00:24.26 | bkw_ | :) |
| 00:24.31 | bratner | I'm partially dislectic and english isn't my native language |
| 00:24.38 | PTG123 | bkw_: fine fine |
| 00:24.38 | PTG123 | heh |
| 00:24.52 | trymlap | cant seem to get my Dial() through SIP to work.. hmmf |
| 00:24.55 | Strom_TM | alright :) |
| 00:25.03 | trymlap | any good urls for help on that issue? |
| 00:25.19 | PTG123 | try: like what issue? |
| 00:25.48 | chuckster | trymlap: what's the asterisk term displaying? |
| 00:26.25 | chuckster | http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20sjphone |
| 00:26.34 | trymlap | just a sec |
| 00:26.43 | chuckster | that's a good URL for softphone connecetion. |
| 00:26.45 | bratner | trymlap: i have the same problem - app_dial.c:743 dial_exec: Unable to create channel of type 'Console' |
| 00:27.02 | PTG123 | trymlap: like i said you could just use GSM, since it uses alot less bandwidth anyhow and supports it |
| 00:27.32 | PTG123 | bkw_: can we just have that added to the bot, that question comes up a couple of times a day :) |
| 00:27.53 | trymlap | PTG123: using gsm now |
| 00:28.22 | PTG123 | try: you sure? B/c that codec supports it? What phone device are you using? |
| 00:28.45 | trymlap | its not that |
| 00:28.59 | trymlap | cant a sip tunnel have both an incoming and an outgoing call at once ? |
| 00:28.59 | PTG123 | yes b/c the codec doesn't support comfort noise, it trys to use that |
| 00:29.17 | PTG123 | kram: was that what you needed? that code? |
| 00:29.23 | trymlap | SIP/2.0 486 Busy Here |
| 00:29.50 | file | trymlap: it's up to your device |
| 00:30.00 | trymlap | hmm ok |
| 00:30.12 | kram | ptg: yah i think so, i will test it out when i get back |
| 00:30.23 | PTG123 | kram: ok great |
| 00:30.47 | trymlap | lets say I have a "friend" defined in sip.conf.. ip24 |
| 00:30.56 | trymlap | Dial(SIP/ip24) <-- where in this syntax would I put the number I want to call? |
| 00:31.08 | Strom_TM | no no no |
| 00:31.09 | *** join/#asterisk RoyK (~roy@110.80-203-29.nextgentel.com) |
| 00:31.11 | RoyK | hie |
| 00:31.14 | Strom_TM | thats to make a call _to_ the sip channel |
| 00:31.15 | RoyK | ehi |
| 00:31.16 | RoyK | hei |
| 00:31.17 | RoyK | hi |
| 00:31.20 | RoyK | something |
| 00:31.28 | trymlap | Strom_C: hmm.. ok.. how do I make a call through it then ? |
| 00:31.50 | file | SIP/${EXTEN}@ip24 call whatever you dialed through ip24 |
| 00:32.01 | file | it's up to ip24 how to handle it... |
| 00:32.16 | trymlap | h |
| 00:32.17 | trymlap | let me try |
| 00:33.05 | Zorix | guys someone recommend me a good linux softphone for a friend |
| 00:33.33 | chuckster | i've had good luck w/sjphone |
| 00:33.39 | RoyK | hei, trymlap |
| 00:33.46 | RoyK | mye nordmenn her for tida :D |
| 00:33.49 | Zorix | sjphone has linux binaries? |
| 00:34.11 | trymlap | just a quick question |
| 00:34.15 | trymlap | RoyK: jepp :) |
| 00:34.35 | RoyK | just ask |
| 00:34.48 | trymlap | if I define a register in the general area of the sip.conf - is the [ip24] just to match that connection.. or do I need to provide the username and password again to call through it ? |
| 00:35.05 | chuckster | y, http://www.sjlabs.com/sjp.html (not open source though) |
| 00:35.44 | RoyK | trymlap: [ip24] is to define the stuff to call from Dial(). username and password are needed anyway |
| 00:36.44 | trymlap | ok.. but do I need to have seperate in and out defined for ip24 ? |
| 00:37.18 | RoyK | you shouldn't |
| 00:37.28 | RoyK | define it as friend |
| 00:37.31 | RoyK | type=friend |
| 00:37.34 | RoyK | meaning in and out |
| 00:37.40 | file | eep friend |
| 00:37.41 | RoyK | they run SIP, right? |
| 00:37.50 | *** join/#asterisk bill_c_ (~bill@bill-c.active.supporter.pdpc) |
| 00:37.53 | trymlap | yep |
| 00:37.53 | RoyK | just remember to register => |
| 00:38.01 | trymlap | inside [ip24] ? |
| 00:38.05 | RoyK | trymlap: do you know what they're running? |
| 00:38.09 | trymlap | SIP |
| 00:38.12 | RoyK | trymlap: no - register should be in global |
| 00:38.17 | RoyK | I meant what sort of software.... |
| 00:38.17 | trymlap | yeah.. I have register |
| 00:38.20 | trymlap | and I see the incoming call |
| 00:38.22 | bill_c_ | wow g729 is pretty cool ;) |
| 00:38.23 | RoyK | ok |
| 00:38.24 | trymlap | cisco probably |
| 00:38.29 | RoyK | bill_c_: and expensive |
| 00:38.32 | RoyK | trymlap: ok |
| 00:38.37 | RoyK | vi kjører asterisk hos briiz |
| 00:38.38 | trymlap | problem is just dialing out |
| 00:38.44 | RoyK | what happens? |
| 00:38.47 | bill_c_ | RoyK, $10 a channel |
| 00:38.55 | RoyK | bill_c_: I know. we just bought a bunch |
| 00:39.26 | trymlap | well |
| 00:39.29 | trymlap | I get this thing: |
| 00:40.04 | RoyK | heh |
| 00:40.14 | implicit | bill_c_: why is it so cool |
| 00:40.24 | implicit | bill_c_: mu-law is cooler |
| 00:40.25 | RoyK | 8kbps |
| 00:41.01 | RoyK | implicit: alaw is the european stuff. anything ulaw-like should be burned |
| 00:41.02 | RoyK | bah |
| 00:41.03 | RoyK | :) |
| 00:41.22 | implicit | mu-law vs a-law is only changes endianness |
| 00:41.33 | RoyK | not endian |
| 00:41.38 | RoyK | signed vs unsigned |
| 00:41.41 | RoyK | iirc |
| 00:41.48 | implicit | yep, misthink |
| 00:41.49 | implicit | heh |
| 00:42.00 | implicit | anyway, doesn't matter at all |
| 00:42.19 | RoyK | implicit: also, if you don't want to waste 80kbps+ per call, g.729 gives you less than 20kbps including overhead with 20ms slicing |
| 00:42.35 | implicit | why not use gsm then? |
| 00:42.36 | RoyK | implicit: it still requires some cpu to transcode the shite if running isdn in europe |
| 00:42.44 | *** join/#asterisk Animas_Mexico (~alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 00:42.56 | RoyK | because gsm is hardly supported in any hardware, like ATAs |
| 00:42.59 | implicit | transcoding from mulaw to alaw is VERY low cpu |
| 00:43.50 | implicit | RoyK: mulaw sounds noticably better than g.729 though ,fi yo uhave the bw go for it |
| 00:43.52 | RoyK | it's still there |
| 00:44.16 | RoyK | also, what's the fucking point of doing transcoding if you've got the bandwidth in the first place? |
| 00:44.40 | RoyK | implicit: that's the point. there's no need to run g729 or any other complex codec if you've got bandwidth |
| 00:45.00 | implicit | RoyK: there is none |
| 00:45.02 | RoyK | implicit: why run ulaw if you need to transcode it to alaw? |
| 00:45.23 | implicit | RoyK: sometimes it is necessary, coming from europe to USA |
| 00:45.37 | RoyK | eh. what? |
| 00:45.53 | RoyK | all sorts of terminal equipment supporting g.711 supports both a and u |
| 00:46.13 | RoyK | and as long as you're not a telco terminating both in eu and us, you never need to worry |
| 00:46.13 | implicit | from a PRI in the USA to a PRI in europe |
| 00:46.20 | RoyK | right |
| 00:46.24 | implicit | yep |
| 00:46.25 | RoyK | but you haven't got that |
| 00:46.53 | implicit | if you are calling 3 waying one person in usa and one in europe |
| 00:47.02 | implicit | over 2 carriers but doing g711 |
| 00:47.35 | implicit | you would have to transcode one of them somewhere along the line |
| 00:47.37 | RoyK | yeah |
| 00:47.42 | RoyK | but you're missing the point |
| 00:47.52 | RoyK | all your PRI termination is in europe |
| 00:48.01 | RoyK | so using ulaw is stupid |
| 00:48.08 | *** part/#asterisk Animas_Mexico (~alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 00:48.09 | implicit | no all my PRI termination is not in europe |
| 00:48.16 | RoyK | no? |
| 00:48.27 | RoyK | I thought you were just doing latvia |
| 00:48.31 | RoyK | or was it litauen? |
| 00:48.39 | implicit | nope never in latvia |
| 00:48.49 | implicit | hehe |
| 00:48.58 | implicit | mixing me up with someone else maybe |
| 00:49.03 | implicit | :-D |
| 00:49.18 | RoyK | er |
| 00:49.19 | RoyK | sori |
| 00:49.20 | RoyK | :) |
| 00:50.08 | implicit | ;) |
| 00:50.34 | bill_c_ | implicit, I need the reduced bandwidth g729 gives, have many employees in bandwidth tight areas |
| 00:51.04 | Silik0n | why not just play like microsoft and just get more bandwidth? |
| 00:51.34 | bill_c_ | bandwidth is $$ in belarus |
| 00:52.10 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 00:52.27 | RoyK | where's belarus? |
| 00:52.49 | bill_c_ | old soviet republic |
| 00:53.21 | DannyF | mmm |
| 00:53.32 | DannyF | anyone played with eyebeam? |
| 00:54.34 | bill_c_ | RoyK, between poland and russia |
| 00:56.23 | implicit | bill_c_: yeah i knoew bandwidth is extremely expsnsive |
| 00:56.54 | Zorix | ok guys any voicemail experts want to point me to a good document |
| 00:58.12 | Zorix | im finding rather bad docs in that area |
| 00:58.51 | bratner | Can "background" sound be other then .gsm format? |
| 01:07.06 | |Fender| | Hi, |
| 01:07.15 | |Fender| | i try register my codec g729 and i got this error |
| 01:07.15 | |Fender| | Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! |
| 01:07.19 | |Fender| | server down? |
| 01:08.28 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 01:08.32 | slePP | sorry. pastebin's back up :P |
| 01:08.35 | slePP | phpa blew up |
| 01:09.39 | *** join/#asterisk jgenender (~jgenender@serifos.eecs.harvard.edu) |
| 01:09.54 | *** join/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
| 01:10.04 | jgenender | Good evening...quick question for anyone ... |
| 01:10.21 | jgenender | I am considering going from my Windows based PBX to asterisk |
| 01:11.02 | jgenender | However, I have a certain need and I would like to know if/how asterisk can handle this |
| 01:11.02 | jgenender | I have a centrex line... |
| 01:11.02 | jgenender | So I can transfer out calls (i.e. to my cell phone) |
| 01:11.18 | jgenender | Without the need to tie up the line...the line is released upon transfer |
| 01:11.24 | jgenender | Can asterisk handle this? |
| 01:11.40 | jgenender | An dif so, any info on where I can find this configuration? |
| 01:13.00 | jgenender | anyone? |
| 01:15.28 | mishehu | centrex is CO based |
| 01:16.38 | jgenender | Yes |
| 01:16.53 | mlh407 | if you put asterisk at the CO then yes you could but otherwise are you are doing a connect between 2 lines |
| 01:17.21 | jgenender | So it cannot handle the transfer out function and drop the line upon transfer? |
| 01:17.40 | mlh407 | if it drops the line, then it will kill the call |
| 01:17.48 | jgenender | My current Windows based PBX can handle this, so i assumed its standard |
| 01:18.10 | mlh407 | humm ok? I guess I am not getting it then What does it do? |
| 01:18.35 | jgenender | Ok... |
| 01:18.47 | jgenender | On my Windows based PBX... |
| 01:18.51 | mlh407 | How do you tell the CO to transfer the call? |
| 01:18.59 | jgenender | I can have a "follow me" kind of function. |
| 01:19.11 | jgenender | It requires a centrex type line for call forwarding |
| 01:19.22 | jgenender | So I pay an extra 5.00 / month for it |
| 01:19.23 | mlh407 | if you don't have a pbx in there, what would be the key seqance |
| 01:19.49 | jgenender | Its not one...I think it does some kind of flash behind the scenes |
| 01:19.57 | jgenender | What it does is... |
| 01:20.03 | jgenender | It call my extension... |
| 01:20.29 | jgenender | Then my extension "bounces" the call by dialing the number to follow me (i.e. my cell) |
| 01:20.37 | jgenender | Then drops the call... |
| 01:20.54 | jgenender | I *think* it does this through a flash type function behind the scenes... |
| 01:20.54 | mlh407 | sorry, I don't know enough about centrax ... I am sure it can be done ... but I could not tell you how unless you can tell me what is going on from a DTMF stance |
| 01:20.59 | jgenender | Not dialed |
| 01:21.41 | jgenender | I will see if I can get seom info on this |
| 01:21.45 | mlh407 | flash is dtmf .... ok |
| 01:22.24 | jgenender | Please excuse me here...I am very new to this, so my knowledge of some of the buzz words may not be up to par :( |
| 01:22.28 | jgenender | But.. |
| 01:22.33 | *** join/#asterisk enyc (~enyc@furrymonster.enyc.org.uk) |
| 01:22.35 | enyc | mee de boop |
| 01:22.54 | jgenender | I think it does a flash...like when you do a thrid party line... |
| 01:23.02 | jgenender | Dials the number... |
| 01:23.05 | jgenender | Reflashes... |
| 01:23.11 | jgenender | Then drops. the line |
| 01:23.28 | jgenender | The line is freed up... |
| 01:23.45 | jgenender | and my cell and the originating caller are connected |
| 01:23.48 | mlh407 | That can be done in asterisk |
| 01:24.22 | jgenender | Excellent. Do you have a link for which I can see a configuration for something like this? |
| 01:24.28 | jgenender | Thanks in advance BTW. |
| 01:25.32 | enyc | [all]: I would like to know if ''ordinary rockwell style serial modems can be used in 'voice interface mode' to connect to a PABX line (e.g. to provice a VOIP gate that can be 'phoned up' over local-pabx) |
| 01:26.05 | enyc | I know some/?many? rockwell serial modems will indeed enter 'voice' mode over serial.... |
| 01:27.13 | ManxPower | enyc: Then write a driver. Nobody else will. |
| 01:27.30 | enyc | well I want to know if this is already supported in cuirrent configartion ;-) |
| 01:27.37 | ManxPower | enyc: Asterisk does not support voice modems. |
| 01:27.58 | enyc | damn.. what sort of hardware-interfaces does it support for non-IP transports? |
| 01:28.01 | ManxPower | There are some "voice modem" drivers included with Asterisk, but nobody uses them and they have not been updated in ages. |
| 01:28.12 | ManxPower | enyc: Digium hardware. |
| 01:28.22 | *** join/#asterisk CpuID2 (~none@CPE-203-45-152-22.qld.bigpond.net.au) |
| 01:28.29 | enyc | hrrrm, damn, I was hoping to find something that works with the current stuff I have got etc. |
| 01:28.49 | ManxPower | enyc: Good luck. |
| 01:28.58 | jgenender | mlh407: I just confirmed the DTMF sequence... |
| 01:28.58 | jsmith | enyc: The voice modem drivers are only half-duplex, and are nearly impossible to get working. |
| 01:29.04 | ManxPower | enyc: There's mgetty-voice, but that's for voicemail stuff. |
| 01:29.18 | jgenender | flash...dial number...hangup |
| 01:29.39 | CpuID2 | hey ppls, if ive got exten => _7011N.,1,Dial(IAX2/moo@blah/${EXTEN:1}) in one of my extensions.conf contexts, the :1, is that gonna mean itll use the extension minus the first prefix digit? or is it going to use everything after the first instance of the digit 1 in the number? |
| 01:29.49 | enyc | I *have* a real pabx with control phoen etc... and many serial modems .. and workoing multi port serial card and I can even go right up to 230400 if I want etc. |
| 01:30.17 | ManxPower | enyc: Unless you want to do driver hacking, give up. |
| 01:30.22 | enyc | damn |
| 01:30.23 | enyc | ;- |
| 01:30.29 | jsmith | jgenender: Then use the Flash() application to flash the Zap channel, SendDTMF() to send the digits, and Hangup() to hang up the line. |
| 01:30.36 | enyc | Im not familiar with this archtecture to prorgram in .e tc. |
| 01:30.47 | jgenender | jsmith: Excellent...thanks alot. |
| 01:30.50 | jsmith | CpuID2: the :1 strips off the first digit. |
| 01:30.53 | *** join/#asterisk cybyc (~CYB@Ottawa-HSE-ppp258451.sympatico.ca) |
| 01:30.57 | CpuID2 | ah k, yea had a feeling it was that |
| 01:30.58 | CpuID2 | thx |
| 01:30.59 | CpuID2 | ;0 |
| 01:31.11 | CpuID2 | damn whack keyboard, go to type :) and you get ;0 |
| 01:31.15 | CpuID2 | ah :) |
| 01:31.29 | CpuID2 | you guys seen those foldable keyboards on thinkgeek? |
| 01:31.41 | jsmith | Nope. |
| 01:31.46 | CpuID2 | good keyboards, almost silent, yet can be annoying if you dont push the keys hard enough :) |
| 01:31.48 | CpuID2 | their soft rubber |
| 01:31.56 | CpuID2 | so the whole keyboard can be rolled up basically |
| 01:32.08 | CpuID2 | similar to some of the dodgy aftermarket cellphone keypads |
| 01:32.26 | CpuID2 | sec gettin url |
| 01:32.39 | CpuID2 | http://www.thinkgeek.com/computing/input/keyboards/5a7f/ |
| 01:32.46 | CpuID2 | check it L( |
| 01:32.47 | CpuID2 | :)* |
| 01:32.57 | chuckster | greets. anyone got any ideas on how to boost vMail's volume? using TDM4000 w/gsm |
| 01:33.17 | jsmith | chuckster: Just voicemail? |
| 01:33.31 | chuckster | well, |
| 01:33.35 | chuckster | vmail & record |
| 01:33.50 | jsmith | Adjust the txgain and rxgain in zapata.conf? |
| 01:33.57 | jsmith | (Assuming you are using Zap channels) |
| 01:35.14 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 01:37.02 | jgenender | jsmith: Can I configure asterisk to do the transfer, but detect a busy signal before hanging up? |
| 01:38.03 | jsmith | Hmmmn... that might be a little tricky... |
| 01:38.14 | jsmith | I'm not sure how you'd do that... |
| 01:38.59 | jgenender | ok |
| 01:40.07 | jgenender | This is an amazing piece of software...I played with it today...its power blows my $2000 Windows PBX away |
| 01:40.08 | cybyc | Hi! Any secrets in using the register exec for the g.729 codec? I keep getting "Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)!"... |
| 01:40.25 | |Fender| | cybyc i have same problem server is dead :p |
| 01:40.26 | kram | shoot, we had a power outage |
| 01:40.33 | kram | let me try to get someone to get john |
| 01:40.41 | |Fender| | kram :( |
| 01:41.07 | cybyc | kram - nothing urgent (from my perspective)... just wanted to make sure I wasn't doing anything wrong. |
| 01:41.26 | kram | i'm having someone call it in now, thanks for mentioning it |
| 01:41.33 | kram | i hate atx power supplies |
| 01:42.29 | mishehu | kram: you prefer the old at power supplies? |
| 01:42.42 | *** part/#asterisk jgenender (~jgenender@serifos.eecs.harvard.edu) |
| 01:47.13 | dan2 | kram: they aren't that bad |
| 01:47.47 | enyc | hrrrrrm, can I use a (linux supported) ISDN card as an itnerface on asterisk? |
| 01:47.56 | dan2 | enyc: does it support capi? |
| 01:48.31 | enyc | dunno, never tried ;-) |
| 01:48.55 | enyc | and I dont know if I can get this ISDN card to work with PABX's ISDN interface facilities ;-) |
| 01:49.07 | enyc | anyway, I can get hold of some AT-bus linejack cards ;-)) |
| 01:49.14 | *** join/#asterisk blitzrage (~blitzrage@d141-234-145.home.cgocable.net) |
| 01:49.28 | enyc | and got plenty of AT-slots in certain machine... wonder if cards need a /IRQ each.... |
| 01:50.04 | lters | Whats the conf bridge listed for? |
| 01:50.12 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 01:50.37 | zigman | enyc http://www.junghanns.net/asterisk/downloads/?C=M;O=D |
| 01:50.49 | zigman | download bristuff-0.2.0-rc3 |
| 01:50.52 | zigman | and read |
| 01:50.53 | enyc | I could always do a wonderful cheat involving a 8259 and lots of wires ;-) |
| 01:54.16 | enyc | erm Im not sure what signifance-to-me of bristiff-0.2.0-rc3 is, what file inside that tar.gz do you think I shuodl be reading? what shoudl I be leanirning abuot? |
| 01:55.48 | Poincare | enyc: i guess the HFC stuff |
| 01:56.24 | zigman | yeah |
| 01:56.30 | zigman | its isdn support for * |
| 01:56.42 | zigman | where are you from ? |
| 01:56.44 | Zorix | guys anyone know a good document for voice mail |
| 01:56.48 | zigman | us ? or .eu ? |
| 01:57.20 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 01:57.36 | Zorix | .eww :P |
| 01:58.11 | Poincare | .eww? |
| 01:58.17 | Zorix | yea |
| 01:58.24 | Zorix | thats what it sounds like to me |
| 01:58.29 | Poincare | :-) |
| 02:09.02 | *** join/#asterisk pmVee3e (~asdf@d226-73-52.home.cgocable.net) |
| 02:10.28 | Zorix | anyone done voicemail before |
| 02:11.24 | modulus_ | i've left voicemail before |
| 02:11.25 | modulus_ | me me! |
| 02:11.59 | jsmith | Zorix: What's your question? |
| 02:12.16 | *** join/#asterisk mitcheloc (~konversat@67.153.163.202) |
| 02:12.19 | Zorix | well i want to try it out but i havent found any complete docs on it |
| 02:12.32 | Zorix | not sure of the commands and where to begin really |
| 02:12.34 | jsmith | There are no complete docs :-) |
| 02:12.48 | Zorix | noticed |
| 02:12.53 | Zorix | i dont know where to begin either |
| 02:12.57 | jsmith | Check out the wiki at voip-info.org, what's been written so far at asteriskdocs.org, and the handbook from Digium. |
| 02:13.12 | jsmith | First, set up a mailbox in voicemail.conf. |
| 02:13.15 | Zorix | ive been using the wiki and i read all the asterisk docs |
| 02:13.19 | Zorix | ok |
| 02:13.20 | jsmith | You should be able to copy and paste. |
| 02:13.40 | jsmith | We'll assume the voice mailbox is number 1234 |
| 02:13.55 | jsmith | Then, from within your dialplan, you can send callers to your voicemail like this: |
| 02:14.09 | jsmith | exten => 1234,1,VoiceMail(b1234) |
| 02:14.13 | jsmith | (where b stands for busy) |
| 02:14.17 | Zorix | would it be 2 tho |
| 02:14.22 | Zorix | since its 2nd priority |
| 02:14.34 | jsmith | or exten => 1234,1,VoiceMail(u1234), where you're unavailable (didn't answer the call) |
| 02:14.41 | jsmith | Yes, adjust the priorities to match your dialplan. |
| 02:15.00 | Zorix | can i have bu1234 |
| 02:15.06 | Zorix | for both |
| 02:15.37 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 02:15.41 | jsmith | No... the b or u signifies which greeting to play |
| 02:15.47 | Zorix | oh |
| 02:15.50 | jsmith | So you can have a "I'm on the other line, leave me a message" |
| 02:15.52 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 02:15.57 | NTJOCK | meow |
| 02:15.59 | jsmith | or a "I'm not available right now" message |
| 02:16.31 | Zorix | ok now the main question is.. the voice mailbox number must be different from the main extension right |
| 02:16.40 | *** join/#asterisk arrgh (~jhetrick@samwise.frogstar.org) |
| 02:17.00 | Zorix | because im thinkin like at work all i have to do is dial some extension to get my mail but i dont have a voicemail extension it just goes there when im not there or busy |
| 02:17.04 | NTJOCK | hey gang, I notice alot of people run Grandstream budgetone phones.... how is it? what sux? what worx? Any guesses on how it will compare to Sipura's 841? |
| 02:17.47 | blitzrage | twisted[work]: what line did you say to start reading in zaptel source code for the *0 option again? |
| 02:17.54 | syslod | NTJOCK: I can't seem to keep them registered. |
| 02:17.59 | NTJOCK | the BT? |
| 02:18.54 | jsmith | Zorix: It can be, but doesn't have to be. It's completely up to you. |
| 02:19.03 | Zorix | exten => _*1XX,1,Voicemail(u${EXTEN:1}) |
| 02:19.03 | Zorix | exten => _*1XX,2,Hangup |
| 02:19.18 | Zorix | what is the :1 after extension |
| 02:19.37 | jsmith | It strips the first digit off the front of the dialed extension |
| 02:19.43 | jsmith | In your case, it strips off the * |
| 02:19.52 | Zorix | ah |
| 02:19.55 | Zorix | and the _ means? |
| 02:20.16 | NTJOCK | has anyone read Sipura's data sheet onthe 841? It talks about all this PBX functionality as if it were in the phone. |
| 02:20.24 | NTJOCK | http://www.sipura.com/Documents/SPA-841.pdf |
| 02:20.58 | jsmith | Zorix: The _ means it's a pattern, and not just an extension named *1XX |
| 02:21.01 | NTJOCK | I mean pardon me but music on hold is a PBX function, the phone is just a handset/brick |
| 02:21.14 | NTJOCK | it's a device to connect to the pbx and hear the channel. |
| 02:21.17 | Zorix | ahh alright thanks i think i can use that |
| 02:24.17 | syslod | Anyone interested in LCS feature? |
| 02:24.53 | jsmith | LCS? |
| 02:25.04 | jsmith | Least Cost S.....? |
| 02:25.25 | syslod | Live call screening |
| 02:25.43 | NTJOCK | isn't that what a receptionist is for? |
| 02:25.48 | jsmith | syslod: Check out zapbarge |
| 02:25.56 | mitcheloc | lol |
| 02:27.34 | syslod | Yea I know you can listen to channels but its a bit more complicated than that to impliment LCS. |
| 02:28.42 | jsmith | I see... |
| 02:29.25 | blitzrage | jsmith: plus doesn't work on non Zap channels... :( |
| 02:29.29 | syslod | LCS works like at home. Call comes in rings phone goes to voicemail then automatically lets you listen and intercept the call. |
| 02:29.42 | jsmith | I see... |
| 02:29.52 | syslod | Hmm. SIP seems to be working better now. |
| 02:30.06 | syslod | Most PBX's have it. |
| 02:30.36 | jsmith | Hmmmn... I've never seen one with it, or at least with it turned on. |
| 02:30.49 | syslod | Be nice to have two way record and two way transfer too but it'll require some SIP phone interface. |
| 02:31.00 | syslod | :) Most customers love it. |
| 02:31.14 | syslod | Weed out all the sales calls. |
| 02:31.19 | Zorix | why is email needed in voicemail.conf |
| 02:31.31 | Zorix | i mean is it required |
| 02:31.38 | syslod | If you want voicemail to be delivered by email. |
| 02:31.42 | syslod | Its not required. |
| 02:32.00 | Zorix | i want voicemail to be received by dialing the voice mail extension |
| 02:32.00 | Zorix | heh |
| 02:32.16 | syslod | It will happen that way by default. |
| 02:32.22 | Zorix | ok |
| 02:33.32 | Zorix | 1234 => 1234,Some User,email@address.com,pager@address.com,saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes |
| 02:33.42 | Zorix | whats the difference between the 1st and 2nd 1234 |
| 02:34.07 | syslod | VM BOX and then passwd look in the sample file. |
| 02:34.15 | syslod | Or WIKI |
| 02:34.19 | Zorix | i am lookin in wiki now |
| 02:34.28 | Zorix | vm box = extension number right |
| 02:34.59 | syslod | http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 02:35.06 | Zorix | yes i said im there |
| 02:35.10 | Zorix | thats where i pulled that line from |
| 02:35.13 | syslod | vm box could = ext number |
| 02:35.24 | syslod | Usually it does |
| 02:35.27 | Zorix | ok |
| 02:35.29 | syslod | Doesn't have to. |
| 02:35.30 | Zorix | thanks |
| 02:35.33 | syslod | np |
| 02:35.34 | Zorix | alright |
| 02:37.02 | Brixius | vmbox => password,email@address,options |
| 02:37.15 | Brixius | vmbox => password,user name,email@address,options |
| 02:39.12 | syslod | Anyone using Poly microbrowser? |
| 02:39.17 | Zorix | ok thanks |
| 02:41.10 | Brixius | Hello everyone btw |
| 02:47.07 | *** join/#asterisk Slothbag_ (nerf@203-206-248-70.dyn.iinet.net.au) |
| 02:47.37 | NTJOCK | hey gang |
| 02:47.46 | NTJOCK | I'm learning to config my extensions.conf file |
| 02:48.00 | NTJOCK | I've got a TDM400 setup as the dev kit.. 1fxo 1 fxs |
| 02:48.21 | NTJOCK | Is the right way to refer to my FXO port Zap/4? |
| 02:48.53 | Zorix | what was the command to time out a dial after so many seconds so it could go to priority 2 |
| 02:49.32 | Zorix | nm i think its ,10 for 10 seconds |
| 02:50.04 | blitzrage | anyone know if yep |
| 02:50.11 | blitzrage | or rather... just yep |
| 02:50.23 | Slothbag_ | is it possible to get incoming anonymous SIP calls to not need authentication? |
| 02:50.33 | Chuji | NTJOCK : That just depends on how you have zapata.conf setup |
| 02:50.42 | blitzrage | Slothbag_: create a guest account |
| 02:50.55 | Chuji | NTJOCK : but if it's indeed Zap/4, then yes. |
| 02:51.08 | Slothbag_ | as in [guest] |
| 02:51.19 | blitzrage | yah |
| 02:51.29 | Slothbag_ | ahh.. ok, i'll give it a go |
| 02:51.48 | blitzrage | just make it a type=user |
| 02:54.02 | Slothbag_ | nope.. im calling via direct IP using XLite.. xlite forces you to set a displayname which i set to test.. asterisk says Cannon authenticate user test |
| 02:54.21 | Slothbag_ | Cannon = Cannot |
| 02:56.48 | Zorix | i noticed it doesnt ask for a password |
| 02:56.55 | Zorix | for voice mail |
| 02:57.21 | Zorix | exten => 1999,1,VoicemailMain(s${CALLERIDNUM}) |
| 02:57.24 | blitzrage | set it to guest? |
| 02:57.25 | Zorix | i have it set up like that |
| 02:57.57 | blitzrage | you can't just have it not send any authorization... wouldn't make sense... |
| 02:57.58 | Nugget | so you've told it not to ask for a password and you notice it doesn't ask for a password. |
| 02:58.01 | Nugget | makes sense. |
| 02:58.15 | Zorix | is that what the s means |
| 02:58.18 | Brixius | yep |
| 02:58.23 | Nugget | what does the documentation say? |
| 02:58.25 | Zorix | hehe cool i wondered |
| 02:58.34 | Slothbag_ | but it has to handle incoming calls from anyone.. including people not using "test" |
| 02:58.38 | Zorix | i dunno i didnt find any docs on what that meant it was just an example |
| 02:58.44 | Nugget | go to voip-info.org |
| 02:58.52 | Zorix | dude i am there right now |
| 02:58.54 | Zorix | everyone says that |
| 02:58.57 | Zorix | thats where i get my examples |
| 02:59.16 | Brixius | http://www.voip-info.org is your friend. or the show application command is too. |
| 02:59.19 | Zorix | when i make a new password in the voicemail system it updates voicemail.conf or what |
| 02:59.23 | Nugget | I guarantee that "s" is documented on that site. |
| 02:59.34 | Slothbag_ | i read the doco that asterisk checks [users], then [peers] then defaults to [general].. but mine seems to fail the first step and give up |
| 02:59.46 | blitzrage | show application voicemail |
| 02:59.48 | blitzrage | * 's' then instructions for leaving the message will be skipped. |
| 03:00.04 | Nugget | that's VoiceMail() not VoiceMailMain() |
| 03:00.07 | *** join/#asterisk m00use (~69@82.102.55.57) |
| 03:00.10 | blitzrage | oh... oops :) |
| 03:00.11 | blitzrage | haha |
| 03:00.53 | blitzrage | If the mailbox is preceded by 's' then the password check will be skipped. |
| 03:00.58 | *** join/#asterisk Styxfan (~irc@c-24-16-153-121.client.comcast.net) |
| 03:01.01 | Zorix | ok anyone else notice that you cant press any buttons until the entire "menu" has been said |
| 03:01.02 | blitzrage | better :) |
| 03:01.05 | Zorix | ok |
| 03:01.12 | blitzrage | Zorix: you using Playback() |
| 03:01.19 | Zorix | no in the voice mail |
| 03:01.19 | *** part/#asterisk Styxfan (~irc@c-24-16-153-121.client.comcast.net) |
| 03:01.38 | Zorix | like if i hit a numer 2 or 3 it doesnt accept it until the menu has been said |
| 03:01.44 | blitzrage | damnit... I'm obviously not paying enough attention... I'll stop answering questions, lol |
| 03:01.52 | Zorix | :P |
| 03:01.56 | Nugget | weird, I've never seen it behave that way |
| 03:02.02 | Zorix | might just be x-lite |
| 03:02.11 | NTJOCK | I keep getting "zt_set_hook: zt hook failed: device or resource busy" when trying to dial out... if I retry multiple times it will sometimes give me the second dialtone |
| 03:02.13 | NTJOCK | i.e. 9, |
| 03:02.19 | NTJOCK | dialtone, 9, dialtone |
| 03:02.27 | NTJOCK | any ideas on what might cause that? |
| 03:02.36 | Nugget | why do you want a second dialtone? |
| 03:02.44 | NTJOCK | when dialing out of the PBX. |
| 03:02.45 | Brixius | zorix: it works for me, I can press buttons while the voice is playing back. |
| 03:02.50 | Zorix | NTJOCK ah yes i know that one |
| 03:02.54 | NTJOCK | i.e. dial pizza hut at 713-956-pizza |
| 03:02.54 | Nugget | why do you want a second dialtone when dialing out of the pbx? |
| 03:02.57 | Zorix | u have to give it an ignorepattern or something |
| 03:03.13 | Zorix | Brixius x-lite? |
| 03:03.18 | NTJOCK | k |
| 03:03.21 | NTJOCK | extensions.conf? |
| 03:03.40 | Zorix | ignorepat => 9 |
| 03:03.43 | NTJOCK | I have that |
| 03:03.50 | Zorix | hmm |
| 03:03.53 | Brixius | I don't have x-lite installed on this pc, I was using a cisco 7960 and iaxy. |
| 03:03.58 | NTJOCK | what's happening is it appears that the bell interface (zap/4) isn't picking up fast enough |
| 03:03.59 | Zorix | i havent used it but it says in the docs thats what its for |
| 03:04.01 | NTJOCK | and it's busing out |
| 03:04.09 | *** join/#asterisk justinnnnnn (justin@c211-28-205-205.eburwd1.vic.optusnet.com.au) |
| 03:04.09 | NTJOCK | I do get the second dialtone sometimes |
| 03:04.13 | Zorix | Brixius ok then its probably my x-lite thanks for confirming |
| 03:04.20 | Nugget | why do you want a second dialtone when dialing out of the pbx? |
| 03:04.21 | Zorix | hm |
| 03:04.22 | NTJOCK | i.e. try, busy, hang up, try again, get dial, hang up, try again dial. |
| 03:04.30 | NTJOCK | nugget- because that is how the rest of hte world does it |
| 03:04.32 | Nugget | is your zap channel hooked up to a pbx and not a pstn line? |
| 03:04.41 | NTJOCK | dial 9 for an outside line and then dial for amusement |
| 03:04.47 | Nugget | or are you trying to make it give you a secnd dial tone within asterisk? |
| 03:04.48 | NTJOCK | no, straight to PSTN |
| 03:04.57 | Nugget | because that's not how it's "supposed" to work in asterisk. |
| 03:04.59 | NTJOCK | Honestly would be nice to have it just dial, like our nortel does |
| 03:05.01 | Nugget | nobody does it that way |
| 03:05.06 | haha | Hai. |
| 03:05.07 | Nugget | *that's* how it works. |
| 03:05.15 | NTJOCK | ok |
| 03:05.21 | NTJOCK | so should I turn off the ignorepat stuff? |
| 03:05.30 | NTJOCK | it's all over the extensions.conf file in the default |
| 03:05.37 | Nugget | do you want to dial 9 to place outbound calls, or are you just under the impression that you have to? |
| 03:05.43 | NTJOCK | under impression I had to |
| 03:05.50 | NTJOCK | I prefer that you jsut pick up the phone and dial |
| 03:05.53 | Chuji | My sister-in-law just got a job answering Pizza hut delivery calls from home. They supplied her with a sipura and she take pizza delivery orders from all over the country |
| 03:05.54 | syslod | Acutally most PBX using digital have a distinctive tone or if ACR is on it has no second tone. |
| 03:05.55 | haha | I want to dial into an asterisk using FWD, then be able to make outgoing calls on my PSTN, is it worth my time and effort, or should I just go with vonage or a similar carrier? |
| 03:05.59 | NTJOCK | and that the PBX thinks about how to get there and does the dirty work |
| 03:06.05 | Nugget | because it can go either way. dialing 9 for an outside call is possible, if you don't want to retrain users, but it's not necessary at all |
| 03:06.16 | NTJOCK | No, nortel is direct dial |
| 03:06.20 | NTJOCK | we currently have a nortel brick-set |
| 03:06.22 | Nugget | I dont' care about nortel. |
| 03:06.26 | Nugget | I'm talking about asterisk. |
| 03:06.27 | Chuji | haha: It's worth the effort, as long as you have a month to spare |
| 03:06.44 | NTJOCK | yes, and you said retrain... thus nortel is important because they currently are trained to just dial... without the 9 |
| 03:06.50 | haha | Chuji: a month? |
| 03:06.53 | NTJOCK | prefer to just dial |
| 03:06.57 | Nugget | so configure asterisk to behave that way |
| 03:06.58 | NTJOCK | and not worry about dialing 9 for outside |
| 03:07.00 | NTJOCK | k |
| 03:07.05 | Chuji | haha: Asterisk is addicting |
| 03:07.14 | NTJOCK | so, should I remove the 9 from all the exten=> statements |
| 03:07.18 | Nugget | yes |
| 03:07.20 | Chuji | haha: You'll be spending the next month working on it. I promise |
| 03:07.21 | NTJOCK | k |
| 03:07.21 | NTJOCK | thanks |
| 03:07.26 | NTJOCK | chu- who me? |
| 03:07.29 | Brixius | Chuji: I'd have to agree with you on the addicting part. |
| 03:07.33 | Nugget | exten => NXXNXXXXXX,1,Dial(ZAP/4/${EXTEN}) |
| 03:07.36 | Nugget | that should be all you need. |
| 03:07.47 | Nugget | replacing Zap/4 with whatever channel is appropriate |
| 03:07.48 | NTJOCK | k |
| 03:07.56 | NTJOCK | I was going to leave the trunk thing in |
| 03:08.03 | NTJOCK | we have 4 pots lines for voice and 1 for fax |
| 03:08.06 | Nugget | yeah, that's a cleaner way to do it. |
| 03:08.07 | NTJOCK | and under * we will have 5 pots lines |
| 03:08.17 | NTJOCK | right now I'm playing with * on our fax line. |
| 03:08.17 | NTJOCK | :) |
| 03:08.47 | NTJOCK | when I have it behaving and handling the fax brick, 2 fax modems, and credit card creature (terminal) I'll move to testing a couple of SIP phones with it |
| 03:09.05 | NTJOCK | and hopefully by then Sipura will have 841's out |
| 03:09.16 | NTJOCK | and we can forklift the Nortel into oblivion (ebay) |
| 03:09.28 | justinnnnnn | OK PPLS |
| 03:09.30 | justinnnnnn | I NEED MONEY |
| 03:09.36 | justinnnnnn | THIS IS NO LONGER A JOKE |
| 03:09.52 | Brixius | justinnnnnnnnn: so do the rest of us. take a # |
| 03:09.53 | Chuji | justinnnnnn : To get a new keyboards? |
| 03:09.59 | Nugget | haha |
| 03:10.03 | Chuji | justinnnnnn : One with a working capslock key? |
| 03:10.05 | justinnnnnn | i will kill and or perform sexual favours (conditions apply) |
| 03:10.30 | syslod | Why not just get a job? |
| 03:10.30 | NTJOCK | justinnnn try #gaysex |
| 03:10.34 | NTJOCK | this isn't the pimp chanenl |
| 03:10.35 | NTJOCK | channel |
| 03:10.35 | justinnnnnn | i have a job |
| 03:10.36 | NTJOCK | :) |
| 03:10.39 | justinnnnnn | but payday is not till thursday :( |
| 03:10.47 | syslod | get a better one |
| 03:10.47 | Nugget | Chuji: his keyboard seems to also be dropping random characters, like in PEOPLE. :) |
| 03:10.51 | justinnnnnn | nt: since when ? its under new management ?? |
| 03:11.06 | NTJOCK | this channel? or the one I referred you to for sexual favors and other amusements? |
| 03:11.22 | NTJOCK | brb, gonna go to test dials |
| 03:11.26 | NTJOCK | no analog at my desk |
| 03:11.27 | NTJOCK | :( |
| 03:12.44 | Brixius | ok, old joke, but whatever. |
| 03:13.10 | Chuji | Brixius : I got it |
| 03:13.17 | Chuji | Brixius : droids reference |
| 03:13.20 | Chuji | :) |
| 03:13.24 | *** join/#asterisk Section (Lazy@cpc1-darl1-5-0-cust69.midd.cable.ntl.com) |
| 03:13.42 | Section | happy new year guys :) |
| 03:15.03 | kentster | happy new year |
| 03:15.08 | Section | :) |
| 03:15.18 | Chuji | Merry new year |
| 03:15.39 | Section | lol always one different |
| 03:15.45 | NTJOCK | what does trunk digits to strip mean in extensions.conf? |
| 03:16.01 | Section | I dont suppose I could trouble someone for a lil help with an asterisk install? |
| 03:16.01 | NTJOCK | I think it means that it yanks out that digit |
| 03:16.22 | Chuji | NTJOCK : It's the number of digits at the front of EXTEN that it strips |
| 03:16.30 | NTJOCK | ok |
| 03:16.39 | Chuji | i.e. if you were using dial '9' |
| 03:16.41 | NTJOCK | so if I want to just dial, then it should be set to 0 to strip no digits |
| 03:16.43 | NTJOCK | got it |
| 03:16.45 | NTJOCK | thanks |
| 03:17.05 | Chuji | NTJOCK : Yeah, in the example it's just setting a variable |
| 03:17.30 | Section | If I want asterisk to route calls via sip, how would I go about doing it? so far the most I can get it to do is accept sip calls, but it tries to route it to console, and I daren't touch the default config for fear I break it |
| 03:17.47 | Section | and the docs have baffled me |
| 03:17.58 | Chuji | Section : you need to cuddle up to the wiki |
| 03:18.06 | Chuji | Section : and get some balls |
| 03:18.24 | Chuji | Section : cp extension.conf extension.ohshit |
| 03:18.26 | NTJOCK | hmmm.... I'm having trouble dialing out |
| 03:18.27 | *** join/#asterisk Darwin_35 (~darwin35@c-67-171-99-180.client.comcast.net) |
| 03:18.32 | Section | lol.. I'll need IV caffiene |
| 03:18.36 | NTJOCK | keep getting a fast busy related to zt_set_hook |
| 03:18.42 | NTJOCK | for some reason it's taking a second to get a bell dialtone |
| 03:18.50 | NTJOCK | is there a way I can make automatically wait that second ? |
| 03:18.57 | Chuji | NTJOCK : throw a wait |
| 03:19.06 | NTJOCK | is that a channel config? |
| 03:19.20 | Chuji | show application wait |
| 03:19.24 | Chuji | at the cli |
| 03:19.37 | Chuji | ~docs |
| 03:19.38 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 03:19.54 | Chuji | section ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 03:20.22 | Section | ty Chuji, I'll try the wiki.. I tried asterisk doc but that baffled me completely |
| 03:20.24 | NTJOCK | how do I find out what it currently is? |
| 03:20.38 | NTJOCK | asteriskdoc examples helped me trash my system this morning |
| 03:20.41 | Chuji | Section : patience grasshopper |
| 03:20.59 | NTJOCK | fortunately whomever packaged the deb packages did a nice job. |
| 03:21.03 | NTJOCK | and i did a purge and replace |
| 03:21.09 | NTJOCK | kudos to someone. |
| 03:21.41 | Section | ty Chuji, the wiki seems alot better, I'll have a bat with that |
| 03:23.02 | *** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 03:24.16 | jsmith | NTJOCK: Mind if I ask which particular piece of the asteriskdocs examples trashed your system? |
| 03:24.27 | NTJOCK | the config examples |
| 03:24.51 | *** part/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 03:25.10 | Zorix | anyone seen these digital phones called rolm i think |
| 03:25.15 | Zorix | what are those |
| 03:25.33 | Chuji | rolm is a big pbx |
| 03:25.33 | NTJOCK | expensive |
| 03:25.36 | NTJOCK | proprietary |
| 03:25.38 | Chuji | I've worked on them before |
| 03:25.50 | Zorix | well at work our pbx is siemens |
| 03:25.54 | Zorix | but its got rolm phones |
| 03:26.02 | NTJOCK | not very good |
| 03:26.09 | NTJOCK | very sensitive mag switch in hang up sensor |
| 03:26.17 | Zorix | ? |
| 03:26.38 | Chuji | X10 must employ a lot of web developers |
| 03:26.43 | Chuji | their site is different every day |
| 03:26.59 | Chuji | It's always a gd mess too |
| 03:27.54 | jsmith | NTJOCK: Oh yeah? I'm sorry to hear that. |
| 03:28.00 | NTJOCK | and their stuff is always cheaper on Ebay |
| 03:28.01 | NTJOCK | :) |
| 03:28.04 | NTJOCK | np |
| 03:28.08 | NTJOCK | it's better then nothing |
| 03:28.12 | NTJOCK | and it was very educational |
| 03:28.20 | jsmith | Well, hopefully! That's the point... |
| 03:28.22 | NTJOCK | it's just that * quit working after I used the .conf files |
| 03:28.26 | NTJOCK | and I couldn't figure out why |
| 03:28.31 | jsmith | Hmmmn. |
| 03:28.37 | NTJOCK | so I did a purge and replace because it was easier then figuring out what I had messed up |
| 03:28.44 | NTJOCK | shame on me honestly |
| 03:28.46 | jsmith | I understand. |
| 03:28.49 | NTJOCK | tinker tinker tinker, test |
| 03:28.53 | NTJOCK | recipe for disaster |
| 03:28.54 | NTJOCK | :) |
| 03:28.55 | jsmith | Oh well, back to writing more docs... |
| 03:29.02 | NTJOCK | it's a nice start |
| 03:29.05 | NTJOCK | and I did appreciate it |
| 03:29.10 | NTJOCK | So is the "getting started" |
| 03:29.20 | NTJOCK | The biggest thing I had trouble with is it covered too much ground at once |
| 03:29.28 | jsmith | Well, hopefully before too long the docs will get *much* better. |
| 03:29.32 | jsmith | It's just going to take some time. |
| 03:29.42 | jsmith | And yeah, I tend to cover too much ground too quickly... I'm working on that. |
| 03:29.44 | NTJOCK | i.e. it doesn't do a step a, why step a, how step A, test step A success. Step b |
| 03:30.19 | jsmith | Yup. I know. I'm guilty of trying to get you to drink from a fire hose. |
| 03:30.32 | NTJOCK | well newbies must sip before the gulp.... pun intended |
| 03:30.48 | blitzrage | Chuji: good idea :) |
| 03:31.03 | Chuji | I'm going to start doing that for all of our * boxes |
| 03:31.15 | Nugget | It's a good approach for nearly anything. |
| 03:31.58 | Nugget | we go a step further for shared-admin boxes with distributed.net. nobody has the ability to directly edit config files, they have to commit the change and then "cvs up" on the box. |
| 03:33.09 | Chuji | Jameno123: Connected to pstn? |
| 03:33.27 | Chuji | Jameno123: flop the polarity and see if it helps |
| 03:33.31 | Jameno123 | Chuji, ya, FXO cards, TDM40B i think. :( |
| 03:33.40 | Jameno123 | 4xFXO's |
| 03:34.12 | Chuji | Jameno123: I would tell you to use supervision, but I got a lot of random hangups on that |
| 03:34.21 | Jameno123 | you mean reverse the red/green? just to get it to hang up? or? maybe its wired wrong? |
| 03:35.03 | Chuji | Jameno123 : Maybe it's wired wrong. Flip red/green. |
| 03:35.09 | Jameno123 | will try ;) |
| 03:35.30 | Jameno123 | heh, it hangs up eventually, after the timeout, but thats way too long for my environment (Callcenter) too many calls coming in to let it take 10 seconds to timeout. |
| 03:35.42 | Jameno123 | let me try to flip the cord. |
| 03:39.19 | Chuji | Jameno123 : If you have a call center enviornment, you should get away from pots anyway |
| 03:39.32 | Chuji | Jameno123: You really need to run a T1 or PRI |
| 03:39.45 | Chuji | Jameno123 : or at very least, an adit 600 |
| 03:40.03 | syslod | Not many callcenters around thats aren't digital |
| 03:40.27 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 03:40.43 | WilliamK | just had to punch down 24 lines the other day |
| 03:40.44 | jsmith | Jameno123: You may want to test your lines and make sure they have remote hangup detection. |
| 03:40.52 | syslod | I know a few as well but they don't seem to function very well when compared to digital |
| 03:41.25 | WilliamK | true... this one is using it as a lifeline when Unity goes down and I can't bring it back up immediately |
| 03:41.37 | WilliamK | Unity has this mysterious crash syndrome |
| 03:41.39 | syslod | We did one just last week that had over 40 analog trunks at $120 per trunk per month. Converted to predictive dialer and PRI. |
| 03:42.15 | syslod | Most we deal with there is no lifeline option. ONce they go with us its fiber so the eq can't go down anyways. |
| 03:42.58 | WilliamK | this one is waiting on SBC to drop fiber in, and they're going back to regular PBX and getting off the IP phones |
| 03:43.07 | WilliamK | ip phones have been a nightmare to them (Cisco) |
| 03:43.22 | jsmith | WilliamK: Snag a few of the phones and sell them to me cheap :-) |
| 03:43.23 | syslod | Yea its not quite matured yet. |
| 03:43.50 | syslod | There are some good IP systems out there. |
| 03:43.50 | WilliamK | jsmith, I can probably get them ~200'ish if you'd like |
| 03:43.55 | Brixius | WilliamK: did they do all the QOS infrastructure for the Cisco Phones, or just plop them on the network and hope for the best? |
| 03:44.00 | WilliamK | mostly 7960s and 40s |
| 03:44.14 | mishehu | WilliamK: sip firmware on them or skinny? |
| 03:44.18 | jsmith | WilliamK: I might take one. I'd love to have another 7960. |
| 03:44.46 | WilliamK | Brixius, I came in after the fact and redid the entire setup which made everything work 99% of the time, but they're doing some special stuff and Cisco won't support their own crap knowledgeably |
| 03:44.46 | implicit | !valgrind |
| 03:44.49 | implicit | ~valgrind |
| 03:44.50 | jbot | somebody said valgrind was http://valgrind.kde.org : a GPL'd system for debugging and profiling x86-Linux programs |
| 03:44.58 | implicit | i thought so |
| 03:45.06 | Zorix | lol somebody said |
| 03:45.08 | WilliamK | I can get either firmware on them, right now it's skinny but I have the software |
| 03:45.26 | mishehu | WilliamK: got the power bricks? |
| 03:45.42 | WilliamK | mishehu, they have quite a few of the bricks |
| 03:46.13 | WilliamK | they're primarily using PoE |
| 03:46.38 | mishehu | WilliamK: what type of PoE injectors/switches are they using? |
| 03:46.57 | WilliamK | 3550 catalyst |
| 03:47.05 | WilliamK | 24 port |
| 03:47.19 | WilliamK | 2 are brand new, and one is refurb |
| 03:47.45 | jsmith | How many phones? |
| 03:47.47 | mishehu | I'd imagine that's probably a rather expensive system, as it's cisco catalysts |
| 03:47.57 | WilliamK | got at least 50 phones total |
| 03:48.03 | jsmith | Cool... |
| 03:48.22 | WilliamK | over 100,000 was the total investment including labor |
| 03:48.37 | Brixius | WilliamK: They running cisco call manager? |
| 03:48.43 | WilliamK | yeah |
| 03:49.11 | jsmith | CallMangler! |
| 03:49.33 | mishehu | I'd say that whenever I run into problems with asterisk, it's 99 of 100 times an error on my behalf. ;-) |
| 03:49.51 | implicit | more like 95/100 |
| 03:49.54 | Brixius | I've seen it, a friend just got certified in it, then I showed him asterisk and he thinks asterisk is easier too. |
| 03:50.09 | WilliamK | I've got a big prob with * or the telco, haven't figured out yet but I'm leaning towards telco |
| 03:50.12 | mishehu | implicit: I make typos a little too often sometimes |
| 03:50.31 | WilliamK | <PROTECTED> |
| 03:50.33 | implicit | mishehu: unless you are considering not auditing the code thoroughly your problem :) |
| 03:50.48 | syslod | telcos are all knowing being incapable of making mistakes. |
| 03:50.57 | mishehu | implicit: well, more so typos in the configs. |
| 03:51.14 | WilliamK | this one can't figure out the problem yet |
| 03:51.38 | syslod | :) Buy your own test EQ. Gotta have it in this business. |
| 03:51.48 | Brixius | syslod: that's like ISP's calling you back and asking if the link is working now, cause "We didn't change anything" |
| 03:51.52 | mishehu | test eq? |
| 03:52.14 | syslod | Hey usually the ISP does do something they call the telco and repeat what they said. |
| 03:52.19 | syslod | Testing equipment. |
| 03:52.46 | syslod | TBERD, PRI anayliser, Span tester, etc etc. |
| 03:52.53 | Jameno123 | Chuji, heh, its not that big of a callcenter :) just 3 reps, 4 lines... |
| 03:53.16 | Jameno123 | not cost-effective to pull in a full T1, not until i get 10+ lines |
| 03:53.29 | mishehu | syslod: PCMCIA... people can't memorize computer industry acronyms... |
| 03:53.30 | implicit | Jameno123: what about doing all voip? |
| 03:53.58 | Jameno123 | implicit, :) getting ther |
| 03:54.03 | Jameno123 | there* |
| 03:54.46 | *** join/#asterisk Nivex (kjotte@user-0ce2jqe.cable.mindspring.com) |
| 03:57.00 | Jameno123 | Chuji, reversed, didnt help :( |
| 03:57.24 | *** join/#asterisk petrus_79 (~kvirc@p50813E30.dip0.t-ipconnect.de) |
| 03:57.44 | jsmith | Jameno123: Do your lines have remote hangup detection? |
| 03:57.53 | syslod | Jameno123: What is your line doing again? |
| 03:58.19 | Nivex | aww nuts "You are currently the only person in this conference." |