| 00:04.25 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 00:21.26 | De_Mon | homebrew for the wii in 2008! |
| 00:22.00 | lmadsen | homebrew? |
| 00:24.19 | De_Mon | created without an offical sdk full DRM circumvention |
| 00:27.33 | lmadsen | huh? |
| 00:28.36 | De_Mon | sorry *run software* created... |
| 00:29.38 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 00:30.01 | lmadsen | gotcha |
| 00:38.22 | Sargun | De_Mon, how much do the Wii controllers cost? |
| 00:38.27 | Sargun | they interface over BT right? |
| 00:38.36 | Sargun | do they have an LCD? |
| 00:41.06 | nhuisman_work | after seeing what that guy did with them |
| 00:41.12 | nhuisman_work | helmet sensor stuff |
| 00:41.17 | nhuisman_work | and poormans lcd overlay board |
| 00:41.23 | nhuisman_work | really neat |
| 00:43.22 | De_Mon | duno, but wii remotes have been usable on computers for a while. You just need the sensor bar for some of the motion sensing |
| 00:43.39 | nhuisman_work | the sensor bar is just a few leds |
| 00:43.39 | De_Mon | errm pointing? -shrug- |
| 00:43.51 | nhuisman_work | just powered leds |
| 00:43.56 | De_Mon | yup, easy to hook into a computer |
| 00:45.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-35c90ed92e6623c8) |
| 00:45.24 | De_Mon | I want a video player that doesn't require streaming over the opera client myself |
| 00:57.27 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:00.56 | *** join/#asterisk cesar_CR (n=cr@celord.ice.co.cr) |
| 01:02.32 | *** part/#asterisk rnovotny22 (n=rnovotny@99-203-56-203.area2.spcsdns.net) |
| 01:03.37 | *** join/#asterisk SOLARIS_s (n=solaris@CPE0000c5c05cdc-CM0013718cb79c.cpe.net.cable.rogers.com) |
| 01:11.00 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 01:11.12 | jameswf-home | ping tzafrir_home |
| 01:11.23 | tzafrir_home | hi |
| 01:11.54 | *** join/#asterisk tengulre (n=tengulre@124.42.50.54) |
| 01:12.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:12.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:12.57 | tengulre | anybody developt rtp here? |
| 01:17.16 | remmo | why |
| 01:19.21 | *** join/#asterisk vetetix (n=vetetix@83.222.34.12) |
| 01:20.28 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 01:34.15 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 01:37.34 | *** join/#asterisk daven (n=daven@145.175.adsl.brightview.com) |
| 01:44.28 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/staff/sargun) |
| 01:51.28 | nhuisman_work | bout to say f u to digiums low support list of distros |
| 01:51.33 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 01:51.35 | nhuisman_work | fc4 and rhel 4 |
| 01:51.37 | nhuisman_work | come on guys |
| 01:52.13 | bmcghee | on my asterisk system (1.4.14) when i change music on hold in the ques it works but when i change it in a extention it stays at default |
| 01:52.28 | jengelh | nhuisman_work: make it your advantage - sell fc8 support yourself |
| 01:54.48 | nhuisman_work | hehe |
| 01:55.03 | nhuisman_work | screw that, how bout debian support |
| 01:55.37 | russellb | don't underestimate how much work it is to add support for a distribution ... |
| 01:55.46 | rob0 | Huh? They did some support work on my Slackware long ago. |
| 01:55.47 | jengelh | nhuisman_work: debian no way, ubuntu perhaps :D |
| 01:55.54 | nhuisman_work | ubuntu is not a production os |
| 01:56.02 | nhuisman_work | oh i understand it's a ton of work |
| 01:56.03 | russellb | and you need to clarify, you're talking about support for users of the commercial version of asterisk |
| 01:56.05 | jengelh | and there is no company behind debian |
| 01:56.09 | nhuisman_work | they are just behind and i'm joining in at a bad time |
| 01:56.12 | russellb | not general support contracts, or support of the hardware |
| 01:56.35 | nhuisman_work | their next release, asterisk be version C will have rhel 5 I assume |
| 01:56.48 | rob0 | The guy ssh'ed in and helped me set up a zaptel card. |
| 01:57.02 | russellb | and besides, we want to encourage users of BE to use our rpath based distribution, anyway |
| 01:57.13 | russellb | rob0: you're talking about hardware support, he's talking about asterisk business edition support |
| 01:57.13 | nhuisman_work | yeah I had problems with rpath earlier today |
| 01:57.20 | rob0 | oh |
| 01:57.35 | nhuisman_work | the repository was all messed up and the support guy couldn't figure it out. |
| 01:57.46 | nhuisman_work | somehow it was looking for debug versions instead of the normal ones |
| 01:57.59 | russellb | ah, i apologize. our main rpath developer is on holiday vacation ... |
| 01:58.05 | nhuisman_work | it's ok. |
| 01:58.09 | nhuisman_work | i may try out rpath again though with the new version |
| 01:58.18 | nhuisman_work | the cd I got was an older version then the latest on the website |
| 01:58.25 | russellb | gotcha ... |
| 01:58.38 | bmcghee | ok this is wierd |
| 01:58.52 | nhuisman_work | russellb, you have any idea how to run a later glibc in a chroot? |
| 01:58.59 | russellb | nhuisman_work: no clue. |
| 01:59.13 | nhuisman_work | yeah, lets try rpath one more time. |
| 01:59.17 | nhuisman_work | *puts in cd* |
| 02:00.44 | bmcghee | DID (set to non for MOH) , QUE (set to "brent" MOH), My EXT (set to "brent" MOH) but it plays no music if calling in to the system, when i call out, put on hold. plays default |
| 02:00.57 | nhuisman_work | russellb, is it very hard to create rpath packages? ie, heartbeat |
| 02:01.14 | russellb | nhuisman_work: no, it's not that bad, actually |
| 02:02.29 | russellb | i haven't messed with it in quite a while, though. |
| 02:02.49 | nhuisman_work | hmm it seems like the iso doesn't have rpath on it |
| 02:04.33 | nhuisman_work | any clue on how to get the latest rpath installer? |
| 02:05.42 | russellb | if you bought BE you should have access to the BE portal on digium.com to download the latest stuff ... |
| 02:05.54 | russellb | i'm not sure exactly how it works, i rarely work on any of the digium commercial things. |
| 02:06.27 | nhuisman_work | yeah I have access |
| 02:06.37 | nhuisman_work | the iso i downloaded in there just has the 3rd party distro packages |
| 02:06.42 | nhuisman_work | no updated rpath |
| 02:10.33 | *** join/#asterisk ariel_ (n=ariel_@c-66-176-41-202.hsd1.fl.comcast.net) |
| 02:10.37 | nhuisman_work | russellb, is there a cutoff for tech support hours? or are they 24/7? |
| 02:10.42 | nhuisman_work | i'm in hawaii so... heh |
| 02:17.53 | russellb | i don't have the answer to that one ... |
| 02:18.00 | russellb | there are 24/7 options available, though. |
| 02:18.06 | Qwell | I think 7-7 |
| 02:18.26 | nhuisman_work | guess i'll wait until tomorrow |
| 02:18.28 | nhuisman_work | err nm |
| 02:18.29 | nhuisman_work | monday |
| 02:18.55 | nhuisman_work | pretty burned out with installing linux over and over for today :P |
| 02:19.06 | nhuisman_work | installed rpath, then rhel 4, then fedora |
| 02:20.13 | nhuisman_work | ooooh |
| 02:20.29 | nhuisman_work | maybe I can install b-2.2 then pop that 2.3.6 cd in and ruh the install script |
| 02:22.36 | nhuisman_work | hmm, guess not, the install.sh doesn't have anything specific in it to detect rpath |
| 02:24.03 | nhuisman_work | man |
| 02:24.13 | nhuisman_work | that asterisk book is layed out pretty terribly |
| 02:24.22 | *** join/#asterisk coppice (n=chatzill@137.192.17.210.dyn.pacific.net.hk) |
| 02:38.05 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 02:44.25 | *** join/#asterisk asdx (n=diego@adsl-154-255.click.com.py) |
| 02:48.48 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:52.31 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 02:52.37 | teknoprep | hey all |
| 02:52.41 | teknoprep | i am on the east coast of the USA |
| 02:52.58 | teknoprep | anyone have a good provider for voip that is reasonable priced and has decent quality |
| 02:55.05 | asdx | teliax? |
| 02:56.31 | teknoprep | let me check it out |
| 02:56.41 | teknoprep | don't really want IAX2 trunks tho |
| 02:56.46 | alrs | teknoprep: I like Garachi |
| 02:56.50 | alrs | er, Gafachi |
| 02:57.02 | coppice | I like smoked salmon |
| 02:57.56 | asdx | teknoprep: they do SIP as well |
| 02:58.08 | teknoprep | hey i have a question |
| 02:58.14 | teknoprep | can you port 1-800 #'S ? |
| 03:02.12 | De_Mon | ~voip-us |
| 03:02.22 | De_Mon | ~voip |
| 03:02.23 | jbot | somebody said voip was Voice over IP |
| 03:02.32 | De_Mon | oh ~itsp-us |
| 03:02.44 | De_Mon | bleh |
| 03:02.44 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:02.45 | De_Mon | ~itsp |
| 03:02.46 | 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 |
| 03:02.53 | De_Mon | ~itsplist-us |
| 03:02.53 | 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 |
| 03:03.58 | coppice | jbot: obscure is choosing tags like "itsplist-us" to get the names of a few ITSPs :-) |
| 03:03.59 | jbot | ...but obscure is already something else... |
| 03:04.31 | coppice | ~obscure |
| 03:04.32 | jbot | ACTION is a store of obscure knowlege, or a bit fuzzy |
| 03:08.40 | teknoprep | how is teliax ? |
| 03:08.45 | teknoprep | they seem decent |
| 03:08.50 | teknoprep | asdx you still there ? |
| 03:09.01 | asdx | teknoprep: yeah |
| 03:09.15 | asdx | teknoprep: they are decent |
| 03:09.22 | teknoprep | asdx, how is setup done? |
| 03:09.34 | teknoprep | asdx, do i have to get someone on the phone before i am able to get my sip information ? |
| 03:09.40 | teknoprep | asdx, or is everything done online ? |
| 03:10.06 | asdx | teknoprep: well, i used iax and they explain you what to put in your sip or iax configuration |
| 03:10.20 | teknoprep | asdx, that is now what i was asking |
| 03:10.23 | asdx | teknoprep: i used it with asterisk |
| 03:10.32 | teknoprep | asdx, i was using voicepulse.. and before you get your information |
| 03:10.39 | teknoprep | asdx, you have to fax something into them |
| 03:10.46 | teknoprep | asdx, and they aren't open on saturday |
| 03:10.46 | asdx | teknoprep: i see |
| 03:10.57 | teknoprep | asdx, do you have to do this with teliax ? |
| 03:10.59 | asdx | teknoprep: i did the setup myself |
| 03:11.15 | teknoprep | asdx, so i order the setup... and then they email the information ? |
| 03:11.17 | asdx | teknoprep: i think they have a support, i didn't use their support myself |
| 03:11.21 | teknoprep | lol |
| 03:11.34 | teknoprep | asdx, do i have to sign anything and fax it in when i sign up for this ? |
| 03:11.57 | asdx | teknoprep: |
| 03:12.06 | teknoprep | asdx, yes ? |
| 03:12.09 | asdx | teknoprep: no, i think you buy it straight away with a credit card |
| 03:12.24 | teknoprep | asdx, ok... how is there voice quality ? |
| 03:12.26 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 03:12.37 | asdx | teknoprep: it's good |
| 03:12.59 | asdx | i used the gsm codec with asterisk |
| 03:13.00 | teknoprep | asdx, i like the unlimited 4 channel plan |
| 03:13.11 | teknoprep | asdx, you don't use asterisk anymore ? |
| 03:13.39 | asdx | teknoprep: nope, but i plan to use it on the future again, this teliax account i bought for a customer |
| 03:13.55 | teknoprep | asdx, ahh |
| 03:14.26 | teknoprep | asdx, i have a box in my basement with some cisco 7940's connected to it.. i also have a polycom 320 |
| 03:14.30 | asdx | asterisk owns |
| 03:14.33 | teknoprep | asdx, yes it does |
| 03:14.40 | *** join/#asterisk jblack (n=jblack@pool-71-181-136-33.sctnpa.east.verizon.net) |
| 03:14.43 | asdx | teknoprep: cool |
| 03:16.38 | jblack | Am I going crazy, or does ekiga like to drop dtmf keys? |
| 03:19.26 | teknoprep | does taliax send you a bill ? |
| 03:19.40 | teknoprep | or does teliax just take right off your CC ? |
| 03:19.51 | *** join/#asterisk exvito (n=exvito@89.181.12.156) |
| 03:21.31 | *** join/#asterisk tobias (n=tobias@pool-71-176-133-75.hag.east.verizon.net) |
| 03:22.54 | asdx | teknoprep: i don't know, i didn't use my credit card, the customer bought it directly |
| 03:23.09 | tobias | hello, is there a way to use asterisk to support basic dial-in internet service for a couple users? |
| 03:26.49 | teknoprep | dial-up ? |
| 03:27.05 | teknoprep | you would probably want to look into PPP |
| 03:28.01 | tobias | ok |
| 03:28.48 | tobias | teknoprep: my asterisk instance is connected to the PSTN only through a voip provider (no hardware). is it still possible? |
| 03:29.10 | coppice | tobias: very unlikely |
| 03:29.51 | tobias | coppice: for reasons similar to those that prevent faxing over voip lines? |
| 03:29.59 | coppice | yep |
| 03:30.53 | coppice | only its even worse than FAX, because there is less opportunity for error recovery, echo cancellation needs to work end-to-end, etc. |
| 03:31.25 | tobias | so attempting something like http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD isn't even worth the effort |
| 03:33.00 | coppice | that page isn't talking about the use of a VoIP path. |
| 03:33.37 | teknoprep | rob0, you want to use dial-up over VoIP ? |
| 03:33.41 | *** join/#asterisk PepOSX (n=pepOSX--@190.78.221.19) |
| 03:33.57 | teknoprep | tobias, you want to use dial-up over VoIP ? |
| 03:34.01 | rob0 | Sign me up. |
| 03:34.18 | tobias | teknoprep: yes |
| 03:35.06 | teknoprep | tobias, thats just crazy.. you would only be able to get reliable speeds at maby 4800 buad |
| 03:35.10 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 03:35.14 | teknoprep | MABY if you're lucky 9600 |
| 03:35.17 | teknoprep | but i doubt it |
| 03:35.26 | tobias | teknoprep: k :) |
| 03:36.07 | tobias | shucks. i was trying to figure a way to use up some of my voip channels |
| 03:36.16 | coppice | it has little to do with the bit rate. if it works for V.22bis, it will probably work for V.34. The snag is it probably won't work for V.22bis |
| 03:36.28 | tobias | and give myself free internet when i'm out in the boonies |
| 03:44.58 | *** join/#asterisk zerohalo (n=zeroHalo@c-71-192-56-252.hsd1.ma.comcast.net) |
| 03:46.04 | WilliamK | tobias whatcha trying to do? sorry I missed the convo |
| 03:47.27 | tobias | WilliamK: hey. I'm just wondering if there's a way that I can use asterisk to get myself on the internet, by dialing in through my cell phone |
| 03:47.53 | tobias | the problem is that i have no hardware - just a voip provider |
| 03:48.07 | tobias | lots of free channels and plenty of bandwidth though :) |
| 03:50.33 | WilliamK | I wouldn't do that even if I could |
| 03:50.39 | WilliamK | cell phone bill would be HIGH |
| 03:51.23 | WilliamK | and if you're in a VZ/Sprint area, you'd be better off getting an aircard |
| 03:51.34 | WilliamK | 49.99 for an unlimited sprint EV-DO aircard |
| 03:51.57 | WilliamK | 500-3Mbps down, 1.8Mbps up (variable of course) |
| 03:56.56 | WilliamK | by the way, that's an unlimited plan |
| 03:57.25 | tobias | WilliamK: yeah, i won't need it much though. and i can confine my usage to N&W |
| 03:57.50 | tobias | but once or twice a year i spend a few days out in the country |
| 03:58.00 | tobias | where there may not even be EV-DO |
| 03:58.14 | tobias | but there is regular cell service |
| 03:58.54 | WilliamK | if you don't have EV-DO, it'll goto RT1xx |
| 03:59.01 | WilliamK | it works off regular cell towers |
| 03:59.43 | WilliamK | you could add a data service as needed to your cell and tether your notebook to your cell |
| 04:00.19 | tobias | yeah. probably easier. |
| 04:00.38 | tobias | i love hacking at things that are impossible though :p |
| 04:00.41 | WilliamK | who's your cell provider? |
| 04:00.47 | tobias | vzw |
| 04:01.41 | WilliamK | you can probably tether, also if you work for a major company see if you get discounts if you haven't checked yet |
| 04:01.49 | WilliamK | most major companies get like 15-17% |
| 04:03.24 | WilliamK | also VZ gives 10% discount if you bundle wireless with landline |
| 04:13.35 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:14.07 | *** join/#asterisk Corydon76-dig (i=grey@pdpc/supporter/bronze/Corydon76-home) |
| 04:14.07 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 04:18.00 | jameswf-home | gprs isnt that like 8kbps |
| 04:20.33 | daven | yeah |
| 04:21.36 | daven | also |
| 04:22.15 | daven | the symbian softphone seems to do quite interesting stuff when it crashes |
| 04:22.22 | daven | as it takes a lot of other stuff out when it falls over |
| 04:26.02 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:44.09 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 04:51.52 | *** join/#asterisk etfonhomey (n=chatzill@74.131.130.161) |
| 04:54.20 | *** join/#asterisk prg3 (n=prg3@chatter.cein.ualberta.ca) |
| 05:09.27 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 05:21.54 | *** join/#asterisk tRSS (n=tRSS@adsl-68-20-182-161.dsl.chcgil.ameritech.net) |
| 05:22.12 | tRSS | how can I place a call from the cli and then forward that call to another number, just from asterisk cli? |
| 05:23.23 | Corydon76-dig | tRSS: you can't |
| 05:23.46 | Corydon76-dig | The CLI is for status and command and control, not for call manipulation |
| 05:24.01 | tRSS | ooh, i thought it was possible.. i was wrong, hmmm... thanks though |
| 05:24.10 | Corydon76-dig | If you want remote call manipulation, use AMI |
| 05:24.30 | tRSS | but I remember doing something like this in the past... it has been while using asterisk again |
| 05:24.51 | jblack | take a look at AGI |
| 05:25.00 | Corydon76-dig | Well, you could use the originate CLI command |
| 05:25.11 | jblack | It's a way to pass control from dialplan to scripts. |
| 05:25.18 | Corydon76-dig | but once the call starts, the only thing you can do to the call is request a hangup |
| 05:25.19 | prg3 | Is there any good starting point to start looking at building a small callcenter on Asterisk? |
| 05:25.36 | Corydon76-dig | prg3: read the book |
| 05:25.39 | Corydon76-dig | ~thebook |
| 05:25.39 | 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 |
| 05:26.10 | tRSS | hmm.. i will check my options.. thanks Corydon76-dig: |
| 05:26.12 | jblack | prg3: The O'reilly book is ok. It has enough information to get started, and makes the example config stuff understandable |
| 05:27.09 | prg3 | jblack: Corydon76-dig I've done some home stuff with asterisk, but some of the call center routing looks tough.. |
| 05:27.29 | Corydon76-dig | Uh, routing? |
| 05:27.30 | prg3 | the client I have wants to have incoming calls forwarded out to remote agents over POTS on both sides.. this posisble? |
| 05:27.41 | Corydon76-dig | You're just talking about queues, right? |
| 05:27.41 | prg3 | ok, routing may be the wrong word.. |
| 05:28.03 | jblack | With the right hardware, sure. |
| 05:28.07 | prg3 | queues sound like a reasonable concept for what I'm thinking :) |
| 05:28.24 | prg3 | They have a Sangoma 8 port FXO board.. |
| 05:28.25 | tRSS | prg3: that shouldnt be hard.. although, it doesn't make sense to forward the call onto POTS again. |
| 05:28.48 | prg3 | tRSS: the remote agents are offsite, with no reliable net.. possibly to cells or something |
| 05:28.57 | jblack | unless he's shipping calls to some place in india with no intar-web but plenty of phone lines. ;D |
| 05:29.03 | Corydon76-dig | Not a great idea, but if you use AddQueueMember with a Local channel, it's doable |
| 05:29.16 | prg3 | and it's going to be ever changing who's actually taking calls.. |
| 05:29.31 | Corydon76-dig | That's true in any call center |
| 05:29.54 | Corydon76-dig | but how exactly is it a call center, if you only have 8 POTS lines and 4 of them will be used by agents? |
| 05:30.00 | prg3 | Corydon76-dig: right, but the off-site with non-PBX lines is odd |
| 05:30.18 | prg3 | 4 inbound, up to 4 outbound to agents, with some local calls.. |
| 05:30.27 | Corydon76-dig | Ah |
| 05:30.31 | prg3 | little call center |
| 05:30.31 | *** join/#asterisk ar3dam (n=fl3pix@189.156.231.173) |
| 05:30.34 | Corydon76-dig | Yeah, tiny call center |
| 05:30.43 | tRSS | very tiny indeed :) |
| 05:30.52 | Corydon76-dig | When you said call center, I was thinking 200 agents |
| 05:31.01 | prg3 | :) |
| 05:31.18 | Corydon76-dig | and if they're all on SIP phones, it's a LOT easier |
| 05:31.39 | prg3 | Oh, I know.. I've got Polycom IP600 and IP500 phones for the local agents.. |
| 05:31.44 | ar3dam | hi there .. how need to do for receive call from my voip provider... any advice? |
| 05:31.52 | Corydon76-dig | The problem with POTS is typically glare |
| 05:32.16 | prg3 | glare? |
| 05:32.16 | Corydon76-dig | and glare on POTS confuses the crap out of Asterisk |
| 05:32.37 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:32.39 | Corydon76-dig | Glare is the term for when an incoming call and an outgoing call collide on the same trunk |
| 05:33.01 | Corydon76-dig | You ever pick up the phone just before it rings, so somebody is already on the line? That's glare. |
| 05:33.07 | prg3 | I think I can avoid that by having dedicated inbound and outbound lines.. |
| 05:34.00 | prg3 | And I'm going to push them to move from POTS to SIP, but I need to get things working before.. and they need a fatter pipe then they have now for net |
| 05:34.55 | De_Mon | POTS or PSTN |
| 05:34.59 | tRSS | prg3: i would suggest setting things up for SIP rather than doing it with POTs and then moving over |
| 05:35.35 | prg3 | tRSS: I agree.. but I don't know if I can get them to make the move that way. |
| 05:36.01 | prg3 | De_Mon: I'm not familiar enough to know what the distinction you are getting at is.. |
| 05:36.08 | tRSS | it would be a smart gamble right now, rather than doing it later |
| 05:36.28 | tRSS | POTS: plain old telephony systems, PSTN: Public Switched Telephone Network |
| 05:36.49 | prg3 | It's POTS over PSTN then |
| 05:37.14 | De_Mon | i thought they were the same thing |
| 05:37.20 | prg3 | what sort of bandwidth do I need to ensure decent quality per call? |
| 05:37.24 | Corydon76-dig | De_Mon: if you read up, you'll see that he was using POTS for agents |
| 05:37.44 | Corydon76-dig | De_Mon: POTS and PSTN are not the same thing, no |
| 05:37.54 | Corydon76-dig | POTS is a method of getting to the PSTN |
| 05:38.01 | De_Mon | POTS and T1s and such all use the PSTN? |
| 05:38.04 | Corydon76-dig | PRI is another method |
| 05:38.07 | prg3 | Corydon76-dig: all of my POTS/FXO lines are PSTN lines.. |
| 05:38.17 | Corydon76-dig | Public Switched Telephone Network |
| 05:38.27 | prg3 | or is that implicit in FXO? |
| 05:38.28 | Corydon76-dig | The whole shebang is the PSTN |
| 05:38.49 | De_Mon | ooh everyone i've known in the "industry" made fun of me when I used the term pots like it was last centry |
| 05:38.56 | De_Mon | century |
| 05:38.58 | coppice | except all the specs call it the GSTN |
| 05:38.58 | Corydon76-dig | prg3: no, you can have FXO ports that do not attach to the PSTN |
| 05:39.11 | prg3 | Corydon76-dig: as in they attach to FXS ports on another system? |
| 05:39.14 | Corydon76-dig | Think of the PSTN as an analogue of the Internet |
| 05:39.36 | coppice | except its mostly digital :-) |
| 05:39.43 | prg3 | POTS = Ethernet, PSTN =Internet |
| 05:39.46 | De_Mon | yay I learned something today, I was getting worried |
| 05:39.52 | Corydon76-dig | Just like you need a network connection to connect to the Internet, but a network connection does not necessarily connect to the Internet |
| 05:40.34 | tengulre | anybody which rtp library is best for developtment? |
| 05:40.54 | tengulre | anybody known wich rtp library is best for developtment? |
| 05:41.00 | De_Mon | shoulda just looked at wikipedia a few months ago when I got confused over the whole thing |
| 05:41.03 | coppice | tenguire. if you want to use C++ there are a couple. if you want to use C, they all suck |
| 05:42.23 | prg3 | Ok, I think I need to digest a bit what I've found out here.. Thanks everyone, I'll come ask more questions when I know enough to ask good ones |
| 05:43.02 | tRSS | prg3: it is always good to ask... asking never hurts! :) |
| 05:44.20 | prg3 | tRSS: It's painful to ask the dumb questions though.. :) |
| 05:44.32 | prg3 | I've got to play with this a bit, and then things will start making more sense.. |
| 05:48.51 | *** join/#asterisk JJLinman (n=linman@dsl-241-136-03.telkomadsl.co.za) |
| 05:50.30 | JJLinman | Ne1 know something about recording calls? |
| 05:51.38 | JJLinman | In the docs it says that i am supposed to use soxmix but it is nowhere to be found |
| 05:51.45 | [TK]D-Fender | JJLinman, "show application monitor" , "show application mixmonitor" |
| 05:52.26 | [TK]D-Fender | JJLinman, Then go install it for your OS. And thats if you want * to mix the ends together. You could record them seperate and leave them that way if you really wanted. |
| 05:53.15 | JJLinman | can you use mixmonitor instead of soxmix? |
| 05:56.32 | [TK]D-Fender | JJLinman, looks like |
| 05:57.39 | JJLinman | thx - I am on the right track now.. |
| 05:58.39 | prg3 | Can you have time dependant dialplans? |
| 05:58.47 | prg3 | as in, behavior changes dependant on time of day? |
| 05:59.25 | De_Mon | prg3 like show application gotoiftime? |
| 06:00.07 | prg3 | De_Mon: something exactly like that :) |
| 06:00.21 | De_Mon | there's an execiftime too |
| 06:00.25 | codefreeze | or contexts with time specs... |
| 06:01.35 | prg3 | and blacklists look pretty easy to implement as well? |
| 06:02.09 | [TK]D-Fender | prg3, "show function DB" , "show application gotoif" , "show application gotoiftime" |
| 06:02.46 | De_Mon | wow I didn't know include could do that! |
| 06:03.22 | De_Mon | codefreeze are you talking about include => or something else? |
| 06:03.48 | codefreeze | yes, sorry, I'm too tired to blurt out the right stuff |
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| 07:42.54 | dennisonicc | Hi |
| 07:43.09 | dennisonicc | I have a small question about asterisk |
| 07:43.13 | *** part/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 07:43.22 | dennisonicc | I mean about voip |
| 07:43.55 | dennisonicc | can I connect my PSTN phone to my voip provider via a normal 56k modem? |
| 07:45.06 | mihinomenest | no. |
| 07:45.36 | dennisonicc | mihinomenest: so I need an adsl modem right? |
| 07:46.36 | mihinomenest | no |
| 07:46.55 | dennisonicc | so how can I connect pstn phones to a computer with asterisk? |
| 07:49.18 | mihinomenest | http://en.wikipedia.org/wiki/Analog_telephony_adapter |
| 07:50.11 | dennisonicc | mihinomenest: thank you very much and sorry for aking stupid n00b questions :) |
| 07:53.50 | mihinomenest | well, you don't know what a modem is, and obviously never read about it, but hey, it's not your fault. |
| 07:53.58 | mihinomenest | you're the victim of a failing education system. |
| 07:54.04 | mihinomenest | hooray! |
| 07:55.09 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 07:57.21 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 08:02.54 | dennisonicc | mihinomenest: I just thought if a modem can recieve calls |
| 08:03.11 | dennisonicc | It can also be a gateway |
| 08:03.16 | dennisonicc | but nevermind |
| 08:05.37 | dennisonicc | mihinomemi: can I achieve the same using fxo to just connect my pstn phone to asterisk? |
| 08:06.54 | dennisonicc | oh sorry |
| 08:06.57 | dennisonicc | its fxs |
| 08:11.50 | *** join/#asterisk flatline5 (n=flatline@modemcable030.131-57-74.mc.videotron.ca) |
| 08:13.05 | flatline5 | hi! if anyone around at this time... i'm new to asterisk and wonder how a call out can be made? (like sip->pstn) |
| 08:17.03 | flatline5 | like i dont want the whole details... i understand you need a did provider that would support sip to get something like pstn->sip... what about the reverse? |
| 08:22.13 | *** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com) |
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| 08:27.45 | dennisonicc | flatline5: most of sip providers provide ability for calling pstn and vice verse |
| 08:28.12 | flatline5 | dennisonicc, yea i've been reading... like the provide would have a sip gateway or sip proxy, is that it? |
| 08:28.24 | flatline5 | s/prodide/provider |
| 08:58.54 | *** join/#asterisk ChannelZ (i=channelz@c-24-8-221-165.hsd1.co.comcast.net) |
| 09:02.05 | ChannelZ | I just hacked my own asterisk and made a new option for the Voicemail app to not play a mailbox greeting at all (I wanted to allow callers to leave a voicemail into a specific user's box after hours but to play an 'after hours' message as part of the dialplan). Any merit to anyone else? |
| 09:11.08 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
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| 09:39.13 | JJLinman | Ne1 know which port(s) must be open on my firewall to make a SIP connection? |
| 09:39.46 | fadey | udp 5060 |
| 09:40.10 | JJLinman | that the only port? |
| 09:41.53 | fadey | mmm... depends. tcp 5060 could also be needed. you'll need to open some RTP ports as well |
| 09:42.01 | fadey | if you are going to make calls |
| 09:42.26 | JJLinman | RTP? |
| 09:43.11 | fadey | real time protocol |
| 09:43.57 | JJLinman | which ports does it run on? |
| 09:44.47 | fadey | for asterisk those usually are udp 10000-20000 |
| 09:44.55 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 09:45.03 | fadey | other programs could be different |
| 09:45.13 | JJLinman | thx |
| 09:47.26 | *** join/#asterisk RoyK (n=roy@ip-154-11-149-91.dialup.ice.no) |
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| 09:59.59 | jengelh | Where would I find the 'capi_request' function? Compilation on channels/chan_capi.c fails because of it. |
| 10:00.16 | *** part/#asterisk ChannelZ (i=channelz@c-24-8-221-165.hsd1.co.comcast.net) |
| 10:00.32 | jengelh | (ast 1.4.16.2) |
| 10:01.46 | jengelh | *sigh* a patch is culprit |
| 10:05.16 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fbf3d87bfe482d53) |
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| 10:08.50 | *** join/#asterisk clusco (i=clusco@203.114.50.232) |
| 10:08.57 | clusco | hello everyone!!!! |
| 10:11.02 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 10:13.12 | clusco | anyone.... |
| 10:13.43 | clusco | between trixbox n centpbx.... which one much easier n simple ????? |
| 10:28.21 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 10:29.45 | markit | hi, I'm desperate :( time to time I get the message "ast_rtcp_read: RTCP Read too short" but seems to make no harm, until I receive a call and I route to VoiceMail... during the message recording, a flow of these errors is shown and the message recording stops there :( no crash, and after the caller hangs up the messages stop |
| 10:30.07 | markit | if I use voicemail from a local extension, it works fine |
| 10:30.44 | markit | also if I speak to the caller instead of record his message |
| 10:33.40 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 10:54.59 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 11:03.10 | tzafrir_home | markit, the same thing happens with a simple Record() ? |
| 11:03.59 | markit | tzafrir_home: yes :( |
| 11:06.51 | markit | tzafrir_home: if I answer the phone, everything is ok, just some of those messages |
| 11:07.12 | markit | if a voicemail or record is started, it's of for 2-4 seconds, then I've that flow of errors |
| 11:07.19 | markit | and recording does not work anymore |
| 11:07.55 | markit | but if I have voicemail or record of a call that comes from a local phone, everything is ok |
| 11:08.07 | markit | maybe some codec translation triggers the problem? |
| 11:09.58 | markit | let's try with wav instead of gsm |
| 11:10.30 | markit | no :( |
| 11:11.41 | markit | tzafrir_home: any clue? |
| 11:11.42 | *** join/#asterisk evilsense (n=santhu@122.167.108.77) |
| 11:12.34 | tzafrir_home | markit, does the audio work well in both directions? |
| 11:12.38 | evilsense | ok , I want to setup a video + audio chat server on my linux box, is asterisk the right software? |
| 11:12.53 | evilsense | I am new here :{ |
| 11:13.01 | markit | tzafrir_home: if I answer the phone, I can talk with the caller without problems |
| 11:13.22 | markit | if I let the call go through voicemail, or if I use record, after some seconds I have that problem |
| 11:14.17 | tzafrir_home | But the problem is unidirectional? Just on the recording side? The remote caller can properly hear prompts? |
| 11:14.33 | evilsense | How much bandwidth will it take? IS video supported? |
| 11:14.45 | tzafrir_home | evilsense, I don't think Asterisk can serve video chats right now |
| 11:15.01 | evilsense | tzafrir_home: oh ok |
| 11:15.10 | markit | tzafrir_home: you can hear prompts, everything is ok until the recording starts, then after some seconds, recording does not work, nor seems able to get # or other from caller |
| 11:15.14 | tzafrir_home | audio chats are generally a well-known and mature feature to set up. You should have no problem with that |
| 11:15.38 | evilsense | tzafrir_home: oh thanks ! |
| 11:16.00 | *** join/#asterisk coppice (n=chatzill@137.192.17.210.dyn.pacific.net.hk) |
| 11:16.11 | tzafrir_home | clusco, simple for what? For initial setup, or maintinance later on? |
| 11:17.07 | markit | tzafrir_home: http://www.pastebin.ca/837161 |
| 11:17.33 | *** part/#asterisk evilsense (n=santhu@122.167.108.77) |
| 11:17.46 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:18.14 | tzafrir_home | markit, what codec is used for the call? |
| 11:18.34 | tzafrir_home | What if you record to a format that uses the same codec? |
| 11:19.32 | markit | I've tried gsm, wav and wav49 without success |
| 11:19.42 | markit | how can I tell what format is used ? |
| 11:20.00 | markit | I mean, what format the caller - asterisk agree upon? |
| 11:23.22 | markit | I've forced ulaw and same result |
| 11:25.31 | mvanbaak | I have seen the same with the latest firmware load in a GXP2000 |
| 11:26.19 | mvanbaak | I fixed it by forcing the phone to use a 20ms payload size |
| 11:27.17 | mvanbaak | when it's set to the defaults (30ms) I get those RTCP Read mesages |
| 11:29.20 | CpuID | hey, any of you guys ever used any call switching hardware? |
| 11:29.49 | CpuID | eg. something which lets you talk to it at a high enough level to give application/business logic to the calls, eg. bridge x channel to y channel...but the calls themselves are routed purely through the hw |
| 11:30.02 | markit | mvanbaak: same rtp problem I have? |
| 11:30.09 | mvanbaak | markit: yes |
| 11:30.22 | markit | mvanbaak: but this is not with sip <--> asterisk, but Voip -> asterisk IF recording |
| 11:30.35 | markit | if I'm not recording, I've only some of those messages |
| 11:30.38 | markit | I'm puzzled |
| 11:30.43 | CpuID | mainly after some kinda hardware to do the job to handle high volumes of voice traffic really |
| 11:30.44 | mvanbaak | I also had this trouble with NEC_Philips dect phones |
| 11:30.55 | CpuID | but be able to throw IVRs into the equation in certain situations |
| 11:31.11 | mvanbaak | markit: correct. phones can handle this, but the recording stuff in asterisk is fixed for 20ms |
| 11:31.16 | CpuID | dont really feel comfortable with my experiences so far pushing large quantities of calls through * |
| 11:31.39 | mvanbaak | markit: as long as the phone is receiving 20ms and sending 30ms and asterisk does not have to record it everything is fine |
| 11:31.40 | markit | mvanbaak: so how can I solve this problem? |
| 11:32.01 | mvanbaak | markit: you see some of those messages when asterisk is transcoding, but you dont hear any chops or anything. |
| 11:32.19 | mvanbaak | markit: force the phone in question to use 20ms payload for RTP |
| 11:32.31 | markit | mvanbaak: when is recording, I see a FLOW of those messages, and no sound is recorded anymore |
| 11:32.39 | mvanbaak | I know |
| 11:32.42 | mvanbaak | I had the same |
| 11:33.05 | markit | mvanbaak: it's ok with my local phones, the problem is when I receive a call from my VoIP provider |
| 11:33.13 | markit | so how can I fix it with them? |
| 11:33.23 | mvanbaak | the FLOW is only generated when asterisk has to record audio, during normal call you get an occacional warning because the transcoding is not going too well |
| 11:33.27 | markit | I mean, is a sort of parameter I can negotiate when registering? |
| 11:33.38 | mvanbaak | hhmm |
| 11:33.44 | mvanbaak | I never had that problem |
| 11:33.49 | mvanbaak | only with local phones |
| 11:34.13 | mvanbaak | SIP or IAX2 ? |
| 11:34.18 | markit | SIP |
| 11:34.31 | markit | btw, I suppose they use an asterisk box too, but they don't provide iax registration so far |
| 11:34.55 | mvanbaak | can you try to enable the jitterbuffer ? |
| 11:35.49 | markit | mvanbaak: forgive my ignorance, is long time I don't play with asterisk, had to "resume" now that I've switched my old activity isdn phone line to Voip |
| 11:35.57 | markit | is something I can set where? sip.conf? |
| 11:37.28 | markit | ok, I see a lot of parameters in sip.conf regarding it |
| 11:38.09 | mvanbaak | ok |
| 11:38.13 | mvanbaak | just enable it |
| 11:38.21 | markit | ok |
| 11:38.25 | tzafrir_home | markit, 'sip show chanels' show the current codec of the SIP channel |
| 11:38.32 | mvanbaak | jbenable = yes |
| 11:38.33 | mvanbaak | jbmaxsize = 200 |
| 11:38.33 | mvanbaak | jbresyncthreshold = 1000 |
| 11:38.33 | mvanbaak | jbimpl = adaptive |
| 11:38.37 | mvanbaak | that's what I have |
| 11:39.02 | mvanbaak | sorry for the paste, should have used pastebin I know |
| 11:40.01 | markit | mvanbaak: ok, I reload now and try |
| 11:40.29 | markit | no, same problem :( |
| 11:40.46 | mvanbaak | it was worth a try at least |
| 11:41.00 | markit | sure, I'm desperate :( |
| 11:42.22 | markit | could it be that I've too recent / too old libraries? I'm compiling asterisk from 1.4.x stable trunk under debian sid |
| 11:42.34 | markit | or some compilation option I've missed? |
| 11:43.19 | markit | in any case, since works locally, should not be the case |
| 11:43.35 | markit | (locally = if I record calls from local extensions) |
| 11:43.45 | mvanbaak | indeed |
| 11:43.48 | markit | mvanbaak: could you call me and try to record a message? |
| 11:43.54 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 11:44.01 | mvanbaak | can I reach you by sip ? |
| 11:44.02 | *** join/#asterisk nirz (n=nir@194.90.229.88) |
| 11:44.18 | markit | mvanbaak: yes, you should, let me find my public IP |
| 11:44.31 | mvanbaak | http://www.whatismyip.com |
| 11:45.00 | mvanbaak | gheh, spammers are now writing the word sex like this: se)( |
| 11:46.21 | markit | mvanbaak: 88.149.177.66 the phone should ring 5 seconds, and then you will enter voicemail without prompts, just hear beep |
| 11:46.30 | markit | at least, this is how is expected to work :) |
| 11:46.39 | mvanbaak | hang on |
| 11:46.57 | mvanbaak | any extension I should add ? |
| 11:47.02 | tzafrir_home | markit, http://rapid.tzafrir.org.il/~tzafrir/sip_net_settings |
| 11:47.02 | markit | none |
| 11:47.19 | *** join/#asterisk qdk_ (n=qdk@195.242.194.41) |
| 11:47.35 | tzafrir_home | I figure it should work for most cases |
| 11:47.58 | markit | tzafrir_home: thanks, I will have a look later, seems interesting tip |
| 11:48.13 | markit | mvanbaak: you are coming |
| 11:48.21 | *** join/#asterisk RedStalker_Mike (n=kvirc@unixway.tversu.ru) |
| 11:48.22 | tzafrir_home | markit, generally, run that, and put the output in the "general" section |
| 11:48.25 | RedStalker_Mike | hi all |
| 11:48.28 | mvanbaak | <PROTECTED> |
| 11:48.30 | markit | but phone did not ring |
| 11:48.33 | tzafrir_home | Saves you finding that information manually |
| 11:49.10 | mvanbaak | nice script there tzafrir_home |
| 11:52.08 | markit | mvanbaak: could you try again? I've put in sip.conf incoming-voip as default context now |
| 11:53.34 | mvanbaak | hang on |
| 11:53.55 | mvanbaak | <PROTECTED> |
| 11:53.57 | markit | han_sip.c:13854 handle_request_invite: Call from '' to extension '88.149.177.66' rejected because extension not found |
| 11:54.04 | markit | mm I've _X. |
| 11:54.13 | mvanbaak | ok |
| 11:54.17 | mvanbaak | hang on |
| 11:54.46 | markit | rings |
| 11:54.52 | markit | ok, recording |
| 11:55.07 | markit | [Dec 29 12:54:52] WARNING[5790]: app.c:602 __ast_play_and_record: No audio available on SIP/82.95.250.75-08232230?? |
| 11:55.13 | mvanbaak | that can be me |
| 11:55.19 | mvanbaak | I need to fix my nat shit ;) |
| 11:55.35 | mvanbaak | --- set_address_from_contact host '192.168.1.252' |
| 11:55.42 | markit | :) ok, let's give up if too complicated for you |
| 11:55.58 | markit | that's the ip of my asterisk box |
| 11:56.02 | mvanbaak | hhmm |
| 11:56.06 | mvanbaak | you behind nat |
| 11:56.10 | markit | yes |
| 11:56.12 | mvanbaak | did you setup nat stuff correctly ? |
| 11:56.22 | markit | I've shorewall, and DNAT rules |
| 11:56.29 | mvanbaak | like extern ip and localnet ? |
| 11:56.41 | markit | well, don't know... |
| 11:56.45 | markit | let me check |
| 11:56.46 | mvanbaak | in sip.conf |
| 11:57.02 | mvanbaak | externip=88.149.177.66 |
| 11:57.18 | mvanbaak | localnet=192.168.1.0/255.255.255.0 |
| 11:57.27 | markit | I've not them set up |
| 11:57.37 | mvanbaak | please do |
| 11:57.46 | markit | thought were important only if you run asterisk in the same box that acts as router/firewall |
| 11:57.48 | markit | sure |
| 11:58.07 | mvanbaak | no, it's important if you run asterisk behind nat |
| 11:58.10 | mvanbaak | ~sipnat |
| 11:58.11 | 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 |
| 11:58.18 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 11:59.53 | markit | do I have to set nat=yes also? as global setting (I have in single peers setup) |
| 12:00.58 | mvanbaak | dont think so |
| 12:01.36 | markit | ok, retry please |
| 12:01.56 | markit | (is nat=yes in the url you provided me with jbot) |
| 12:02.05 | mvanbaak | ah |
| 12:02.10 | mvanbaak | lets try |
| 12:02.16 | mvanbaak | if it wont work you can set nat=yes |
| 12:02.23 | markit | ringing... |
| 12:02.38 | markit | seems recording |
| 12:02.50 | mvanbaak | yeah |
| 12:02.51 | markit | exited without errors |
| 12:02.58 | mvanbaak | and no more internal ip on my console |
| 12:03.14 | markit | mvanbaak: well, in any case thansk a lot for these tips |
| 12:03.15 | mvanbaak | --- set_address_from_contact host '88.149.177.66' |
| 12:03.22 | mvanbaak | no problem dude |
| 12:03.25 | markit | I try again with my voip provider |
| 12:03.32 | markit | maybe it fixed the rtp problem also |
| 12:03.37 | mvanbaak | who knows |
| 12:03.46 | markit | yes, guesswork :) |
| 12:03.57 | markit | yessssssssssssssssssssssssssss |
| 12:04.01 | markit | seems to work now |
| 12:04.12 | mvanbaak | so it was a NAT issue afterall |
| 12:04.21 | markit | INCREDIBLE! |
| 12:04.28 | mvanbaak | gheh |
| 12:04.32 | mvanbaak | we just saved his day |
| 12:04.37 | markit | yes, but is something I could never fix myself |
| 12:04.57 | markit | yes, really I was desperate |
| 12:05.13 | markit | I've switched my isdn to voip and the day I had to make it work, I had this problem |
| 12:05.22 | markit | that would have made almost useless the switch |
| 12:05.33 | markit | and google was of no help at all |
| 12:05.45 | markit | nor the channel, since I'm sure is a really strange error |
| 12:05.55 | markit | mvanbaak: I do thank you a lot!!! |
| 12:06.11 | mvanbaak | no problem. I'm glad it's working now |
| 12:06.16 | markit | tzafrir_home: and thanks to you also :) |
| 12:08.40 | markit | mmm no errors but not sound in the message, let me try again |
| 12:12.22 | markit | WORKS |
| 12:12.25 | markit | ok :)) |
| 12:13.59 | mvanbaak | congrats |
| 12:27.45 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 12:37.07 | tzafrir_home | Bladerunner05, you asked about about busydetect in Italy? |
| 12:37.26 | Bladerunner05 | YES this is my big problem |
| 12:38.43 | tzafrir_home | What are your current settings? |
| 12:39.23 | Bladerunner05 | http://www.pastebin.ca/837194 |
| 12:39.51 | Bladerunner05 | consider I'm using asterisk 1.4.16.2 and zaptel 1.4.7.1 |
| 12:40.41 | tzafrir_home | busydetect=yes ''this cause problem with asterisk 1.4.16.12 |
| 12:40.51 | tzafrir_home | A comment should begin with ';' |
| 12:41.15 | tzafrir_home | Maybe the value "yes ''this cause problem with asterisk 1.4.16.12" simply is not detected as "yes"? |
| 12:42.04 | mvanbaak | and where did you get asterisk 1.4.16.12 ? |
| 12:42.05 | mvanbaak | lol |
| 12:42.20 | tzafrir_home | anyway, it worked before and doesn't work now? In what version it did work? |
| 12:42.26 | Bladerunner05 | I download it from asterisk.org |
| 12:42.42 | tzafrir_home | 1.4.16.[12], I guess |
| 12:43.10 | Bladerunner05 | sorry mistake 1.4.16.2 |
| 12:43.14 | tzafrir_home | yeah. I hope we won't get to .12 . .2 is the current record |
| 12:44.09 | mvanbaak | indeed |
| 12:44.11 | tzafrir_home | anyway, in what version this has worked? |
| 12:44.16 | mvanbaak | 1.0.9 |
| 12:44.17 | mvanbaak | ;) |
| 12:46.03 | *** part/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 12:46.10 | Bladerunner05 | So if I use busydetect=yes it recognize the hang up but asterisk return this error WARNING[2687]: file.c:643 ast_readaudio_callback: Failed to write frame and hang up the line |
| 12:46.38 | Bladerunner05 | If I comment busydetect=yes no error but no hangup ! |
| 12:47.20 | tzafrir_home | well, first-off it is a warning. The call should be hung up and is hung up, right? |
| 12:47.33 | tzafrir_home | ah, it doesn't |
| 12:47.43 | Bladerunner05 | So If busydetect=yes the hang up is arbitrary |
| 12:48.04 | tzafrir_home | what version did work? |
| 12:48.05 | Bladerunner05 | while I press a key or listen to a message it hang up |
| 12:48.26 | Bladerunner05 | I notice that this problem persist from 1.4.x |
| 12:48.29 | tzafrir_home | try setting busycount=5 ; or even 7 |
| 12:48.51 | Bladerunner05 | and leave busydetect=yes ? |
| 12:48.54 | tzafrir_home | So the problem was with previous 1.4 versions as well? |
| 12:49.22 | Bladerunner05 | I don't remeber 'cause before I used capi |
| 12:51.44 | Bladerunner05 | the problem persist also with busycount=5 and busycount=7 |
| 12:52.02 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 12:52.42 | Bladerunner05 | this is zapata.conf now http://www.pastebin.ca/837200 |
| 12:53.51 | tzafrir_home | Bladerunner05, anyway, "busy" is always detected but sometimes you get that warning and the hangup fails? |
| 12:55.37 | Bladerunner05 | •tzafrir_home• this is the asterisk log http://www.pastebin.ca/837203 but the hangup u will see is not caused by the caller |
| 12:58.16 | *** join/#asterisk feqma (n=paul@pool-71-243-150-122.buff.east.verizon.net) |
| 12:59.59 | tzafrir_home | Bladerunner05, in that stage, do you still see the channel in 'show channels'? |
| 13:00.32 | tzafrir_home | Not sure exactly what might cause this. But looks like a bug-reporting-worthy behaviour |
| 13:03.36 | Bladerunner05 | this is the result while a call is in progress http://www.pastebin.ca/837214 and when it do hangup (without any reason) all channels are free |
| 13:36.23 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 13:36.58 | dacs | Good morning |
| 13:37.02 | R1ck | hello, any idea why outbound cid doesnt work, the interface says to use the magic string 'hidden' to hide it, but that doesnt work, also, if I set one of our other numbers, it doesnt work, it only (and always) uses our main number for outbound dialing.. the line is an isdn2 line, and I use a junghanns.net quadbri ISDN card |
| 13:37.20 | R1ck | <PROTECTED> |
| 13:40.30 | DarKnesS_WolF | anyone can send me a fax to test with in egypt ? |
| 13:41.08 | DarKnesS_WolF | R1ck: explain more i didn't get the problem . |
| 13:41.14 | dacs | enta f masr |
| 13:41.20 | DarKnesS_WolF | dacs: ah :-) |
| 13:41.44 | DarKnesS_WolF | dacs: u ? |
| 13:42.02 | dacs | tab 3aweez the fax or if i heard the tone i will hung up |
| 13:43.05 | DarKnesS_WolF | dacs: la2a 3awiz fax :-) 3lshan test ;-) |
| 13:43.18 | R1ck | DarKnesS_WolF: as you can see it sets the CallerID, but its not the same as number I see on my phone |
| 13:44.39 | DarKnesS_WolF | R1ck: then ur provider not passning the CID |
| 13:45.07 | R1ck | but it did work on our old siemens pbx |
| 13:45.15 | DarKnesS_WolF | R1ck: most of providers having local termination on the contry ur calling so when u dial u just get local number |
| 13:46.53 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:47.01 | *** join/#asterisk marcan (i=1337@29.Red-88-9-94.dynamicIP.rima-tde.net) |
| 13:53.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:53.29 | DarKnesS_WolF | ~book |
| 13:53.29 | 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 |
| 14:00.01 | *** join/#asterisk k2nt23 (n=rubsoft@190.40.239.8) |
| 14:12.20 | riddlebox | if person1 conferences in person2 with their phone to a meetme conference can an admin of the conference drop person2 if they want |
| 14:19.06 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 14:27.41 | *** join/#asterisk SOLARIS_s (n=solaris@CPE0000c5c05cdc-CM0013718cb79c.cpe.net.cable.rogers.com) |
| 14:31.46 | *** part/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 14:37.00 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 14:45.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:50.16 | riddlebox | [TK]D-Fender, you alive? |
| 14:52.45 | [TK]D-Fender | nope |
| 14:59.36 | *** join/#asterisk jblack (n=jblack@pool-71-181-136-33.sctnpa.east.verizon.net) |
| 14:59.45 | jblack | Good morning/afternoon/evening. |
| 15:02.28 | jblack | Would I be crazy to think that each sip client that is ported to linux suffers a different key flaw? Ekiga likes to drop dtmf tones, linphone has problems with authentication, x-lite (at least the windows version) won't make sip calls and is proprietary. |
| 15:04.51 | De_Mon | jblack xlite works for me in windows? |
| 15:04.59 | jblack | With sip calls? |
| 15:05.19 | Iamnach0 | works for me too |
| 15:05.40 | De_Mon | yup |
| 15:05.48 | jblack | Oh, then I must have done something wrong. x-lite for windows won't place calls to an ip for me. |
| 15:06.07 | *** join/#asterisk tobias (n=tobias@pool-71-176-133-75.hag.east.verizon.net) |
| 15:06.23 | De_Mon | riddlebox if person1 is conferencing someone else in, that person didn't actually join the conference and no a meetme admin can't kick them without kicking the person who conferenced them in |
| 15:08.58 | riddlebox | De_Mon, thats what I thought |
| 15:09.26 | riddlebox | so person1 could transfer person2 to the conference and let them join it properly though |
| 15:10.16 | De_Mon | yes |
| 15:16.24 | [TK]D-Fender | jblack, what mode are you using for DTMF? |
| 15:16.45 | [TK]D-Fender | jblack, And why are you dialing by IP on a soft-phone? |
| 15:18.47 | tzafrir_home | jblack, considering Ekiga used to be called GnomeMeeting, it's not exactly "ported to Linux" |
| 15:28.07 | DarKnesS_WolF | anyone did iaxmodem + * + hylafax ? |
| 15:28.35 | DarKnesS_WolF | i used NVfaxDetect and asterisk dials the iaxmodem peer but it keeps rining and the other side got NO Answer any idea where i can debug ? |
| 15:29.46 | jblack | [TK]D-fender: Sorry. Had to fix the daughter's computer from a botched upgrade. |
| 15:30.24 | jblack | tzafrir: Yeah. Ported was a bad term. s/ported//. |
| 15:31.27 | jblack | [TK]D-fender: All of them. And I'm dialing by ip to reach the server itself. ;) |
| 15:32.45 | [TK]D-Fender | jblack, You should not be specifying it when you dial in the # entry line |
| 15:33.15 | jblack | I can reach the server fine. The problem happens when I'm already connected, trying to navigate through the dialplan prompts. |
| 15:33.27 | [TK]D-Fender | jblack, And what mode are you using? |
| 15:33.33 | jblack | offhand, it feels as if about 75% of button presses are ignored by ekiga. |
| 15:33.39 | jblack | I tried all of the modes one at a time. |
| 15:33.49 | [TK]D-Fender | jblack, you SHOULD be on rfc2833. If you aren't go fix that now |
| 15:34.01 | [TK]D-Fender | jblack, And make sure it matches in sip.conf entry |
| 15:34.11 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:34.11 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:35.13 | jblack | Yah, ekiga is on rfc2833. I'll set it for the peer in sip.conf now |
| 15:35.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:35.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:37.05 | jblack | Nope. That didn't solve it. |
| 15:37.42 | jblack | Under the [jblack] sip entry, I set dtmfmode=rfc2833 and verified that dtmf was set to rfc2833 as well |
| 15:38.21 | jblack | Ohhh... wait. |
| 15:38.49 | [TK]D-Fender | jblack, and is it USING that entry? |
| 15:39.04 | [TK]D-Fender | jblack, or are we back to your "un-authed" sillyness? |
| 15:39.13 | jblack | Nah, we're authed. |
| 15:39.24 | [TK]D-Fender | jblack, and multiple soft-phone on 1 PC deal? |
| 15:39.40 | jblack | No, just one soft phone, on one pc. :) |
| 15:42.29 | jblack | I'm watching the asterisk console with level 10, sip debug on. Sometimes, asterisk gets dtmf signals from ekiga, sometimes it doesn't. |
| 15:44.36 | jblack | DTMF through the prompt by calling in over the phone (using IPKall) works fine, DTMF with linphone (unauthenticated) works fine as well. |
| 15:46.52 | jblack | googling gave me plenty of dtmf related bugs with older versions of ekiga |
| 15:50.29 | jblack | I found the problem. |
| 15:52.06 | jblack | well, at least a solution. Things work much more reliably when I disable all the codecs but gsm. I suspect that whatever codec I was defaulting to didn't handle dtmf right. |
| 15:56.13 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:58.16 | jblack | Ok. I definitely found it. The problem occurs when, and only when "silence detection" is turned on in ekiga. Turn it off, no problem, regardless of codec. Turn it on, very unreliable dtmf. ;) |
| 15:59.43 | [TK]D-Fender | jblack, * does not support VAD |
| 15:59.49 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 16:04.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:04.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:04.37 | jblack | I'm sorry. I have no idea what you mean |
| 16:04.54 | piper69 | good morning all |
| 16:04.59 | jblack | Good morning! |
| 16:05.04 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.119) |
| 16:05.21 | piper69 | and good ,evening to those who are in evening time now :) |
| 16:05.53 | jblack | awww. You're gonna make people in late-night time zones jealous. |
| 16:06.03 | piper69 | hahahah |
| 16:12.05 | jblack | Ahh. VAD = voice activated detection |
| 16:17.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:17.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:20.56 | *** join/#asterisk feqma (n=paul@pool-71-243-150-122.buff.east.verizon.net) |
| 16:21.19 | russellb | anyone here want to test a new console channel driver? |
| 16:21.43 | russellb | i wrote one that works on my mac, but it should work on linux and other platforms as well |
| 16:22.10 | *** part/#asterisk feqma (n=paul@pool-71-243-150-122.buff.east.verizon.net) |
| 16:27.55 | coppice | VAD is short for DARTH VADER |
| 16:29.47 | *** join/#asterisk ronr (n=ron@82-170-109-196-static.dsl.ip.tiscali.nl) |
| 16:30.13 | PepOSX | russellb, channel driver? |
| 16:30.51 | jameswf-home | chan_vad.c [DEBUG] luke I am your father// |
| 16:31.55 | ronr | hi, I got a Goto statement in my dialplan that should goto to some context,extension,priority, the extension is in a variable, so I use Goto(outgoing,${VAR},1) but asterisk complains that there's no priority labeled outgoing, which is the 1 arg Goto mode, what am I doing wrong? |
| 16:32.55 | jameswf-home | pastebin your dialplan |
| 16:33.01 | jameswf-home | !pb |
| 16:33.06 | jameswf-home | ~pb |
| 16:33.06 | 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 |
| 16:38.05 | ronr | http://pastebin.ca/837381 |
| 16:41.51 | jameswf-home | what context does 200 sit in |
| 16:42.45 | jameswf-home | s/200/30/ |
| 16:42.53 | jameswf-home | screw it |
| 16:43.01 | ronr | 30? doesn't exist |
| 16:43.10 | jameswf-home | add include => outgoing to phones |
| 16:43.16 | ronr | or 300, thats phones |
| 16:43.45 | ronr | sry, last line, including outgoing in phones, wasn't pasted |
| 16:46.45 | ronr | ah, I see it |
| 16:46.53 | [TK]D-Fender | ronr, exten => _[1-8]XX,n,GotoIf($[${OUTGOING} = 1]?outgoing) <-- here you refer to "outgoing" as a label within the CURRENT exten. |
| 16:46.55 | ronr | forgot to add the prioirty label |
| 16:47.03 | ronr | in the goto |
| 16:47.07 | [TK]D-Fender | ronr, you don't HAVE a priority with that label on it |
| 16:47.10 | ronr | *gotoif |
| 16:47.19 | ronr | [TK]D-Fender: I just noticed that |
| 16:47.55 | *** join/#asterisk tobias (n=tobias@pool-71-176-133-75.hag.east.verizon.net) |
| 16:47.56 | ronr | having the goto forwarding me to a context with the same name got me on the wrong track |
| 16:57.09 | *** join/#asterisk joelsolanki (i=joelsola@220.224.114.129) |
| 16:57.50 | SwK | anyone have iraq DIDs? |
| 17:05.21 | SOLARIS_s | not i |
| 17:05.27 | SOLARIS_s | but im looking for toronto DIDs |
| 17:05.46 | Bladerunner05 | I need to see any italian zapata.conf script using tdm400p |
| 17:20.47 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 18:19.40 | lunaphyte | hi |
| 18:26.38 | jameswf-home | the biggest problem with releasing code is someone who shouldnt will get it and blame you because they are stupid |
| 18:27.31 | mvanbaak | lol |
| 18:27.39 | jameswf-home | you shouldnt have to write 5 pages to document a 1 page script... bleh |
| 18:29.24 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
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| 18:31.23 | jameswf-home | I am going to add an --emo switch to my scripts which will make all output sad and depressing :)) |
| 18:31.54 | lunaphyte | i'm hoping some folks here might be able to offer some insight regarding an ata i'm suddenly having trouble with - not specifically asterisk related though - it won't acknowledge an ethernet connection. |
| 18:35.30 | lunaphyte | it's a linksys pap2t, new out of the box as of 2 days ago. i put it on the network, poked around a bit, came back to it a day later, and now i get power and internet leds blinking. |
| 18:35.54 | lunaphyte | i'm wondering if i broke it by changing something and not realizing it. |
| 18:39.32 | mvanbaak | switchport died ? |
| 18:41.21 | jameswf-home | lunaphyte: is it a unlocked pap2 or a vonnage pap2 |
| 18:42.22 | lunaphyte | it's unlocked (err, was, anyway) :) - bought here : http://www.telephonyware.com/telephonyware/tw00306.html |
| 18:42.50 | lunaphyte | mvanbaak: i've tried it on multiple ports, all of which work ok w/ a laptop. |
| 18:43.50 | lunaphyte | i found some suggestions for "resetting" it via touchtones from a connected phone, but it doesn't appear to work in my case. it sort of sounds like you need a dial tone first, and i hadn't even made it to that point yet. |
| 18:45.05 | jameswf-home | if you press **** you should get a voice |
| 18:47.17 | rob0 | ./jameswf-home --emo |
| 18:47.40 | jameswf-home | <PROTECTED> |
| 18:47.42 | jameswf-home | lmao |
| 18:47.50 | mvanbaak | lol |
| 18:47.54 | jameswf-home | cant spell |
| 18:48.01 | jameswf-home | ~emp |
| 18:48.02 | jbot | emp is probably in physics an ElectroMagnetic Pulse, which is essentially a high density wave of beta radiation |
| 18:48.05 | jameswf-home | ~em0 |
| 18:48.08 | jameswf-home | ~emo |
| 18:48.08 | jbot | /wrists |
| 18:48.11 | jameswf-home | damnit |
| 18:58.22 | lunaphyte | jameswf-home: even w/out a dial tone? it doesn't seem to work. |
| 18:59.07 | jameswf-home | dialtone is for humans and has no bearings on anything in the telephone world |
| 18:59.11 | mvanbaak | unplug the ethernet and power, and replug the power |
| 18:59.29 | jameswf-home | there are reset pins on the pap2 you may have to crack the case |
| 19:02.12 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
| 19:12.47 | lunaphyte | jameswf-home: well, maybe not a dialtone, but 48v, right? i have a basic wired handset connected, and can't generate tones. |
| 19:13.01 | lunaphyte | mvanbaak: no luck there. |
| 19:13.48 | lunaphyte | looking at the board, i don't see anything that appears to be reset pins or such. |
| 19:14.25 | lunaphyte | i do see pads that look they're intended for a power switch, but that's about it. |
| 19:17.48 | *** join/#asterisk mechanicus01 (n=none@189.149.76.68) |
| 19:24.56 | [TK]D-Fender | If you have no dialtone its simply not registered |
| 19:26.10 | lunaphyte | yeah, that much i know. :) i'd be happy at this point if it would lease an ip. |
| 19:26.40 | [TK]D-Fender | lunaphyte, it isn't picking up an IP? |
| 19:28.27 | [TK]D-Fender | lunaphyte, I believe **** (pause) 110# will tell you its IP |
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| 19:41.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:42.36 | lunaphyte | [TK]D-Fender: i don't even get a link light on my switch. |
| 19:47.29 | lunaphyte | it seems that this thing may have just simply died. i'm not sure how long it should normally take before hearing a response from the ivr after doing ****, but i can't imagine it's any longer than 60 seconds. i guess i'll give linksys a call on monday. |
| 19:48.12 | jameswf-home | the ivr responds almost immediatel y have you tried a hard reset |
| 19:49.47 | lunaphyte | jameswf-home: it's not clear to me how to do a hard reset (aside from power cycling). there is no reset button on the unit, and no mention of hard reset that i can find. |
| 19:54.31 | jameswf-home | you have to open it up and thre ae two pins |
| 19:57.44 | lunaphyte | looking at the board, i don't see anything that appears to be reset pins or such. |
| 19:57.49 | lunaphyte | i do see pads that look they're intended for a power switch, but that's about it. |
| 19:58.56 | jameswf-home | did centpbx die? |
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| 20:17.40 | mechanicus01 | hi everyone, about how many simultanious calls for 1mg are posible? |
| 20:17.46 | mechanicus01 | *possible |
| 20:18.00 | mechanicus01 | on g729 |
| 20:18.16 | *** join/#asterisk ralfep (n=ralfe@vc-196-207-35-48.3g.vodacom.co.za) |
| 20:19.13 | mihinomenest | you know what the rate for g729 is, right? |
| 20:20.02 | mechanicus01 | looking for it now |
| 20:21.20 | markit | hi, upon loading, in cli I have this message: WARNING[10286]: chan_sip.c:16170 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 861 what could I have wrong in sip.conf? |
| 20:22.02 | dlynes | markit: let's see line 861 |
| 20:22.14 | markit | oh, I thought was the source code line, lol |
| 20:22.14 | mechanicus01 | rate for g729 is 8kbps |
| 20:22.30 | markit | yes, I've a auth=md5, that I don't know what means, I will investigate |
| 20:22.37 | markit | thanks dlynes |
| 20:22.41 | dlynes | markit: 16170 is the line number in the source code |
| 20:22.50 | dlynes | markit: 861 is the line in your sip.conf file |
| 20:22.50 | mihinomenest | mechanicus01: what then, can we infer the call volume to be for 1mbps? |
| 20:22.59 | russellb | that line is not valid for sip.conf |
| 20:23.01 | russellb | just remove it |
| 20:23.31 | mechanicus01 | mihinomenest about 100 calls |
| 20:23.33 | markit | russellb: thanks, I've in in wengophone account section, maybe I found it in some sample online |
| 20:23.35 | dlynes | mihinomenest: under ideal conditions |
| 20:23.44 | mihinomenest | mechanicus01: don't forget overhead. |
| 20:23.54 | dlynes | markit: auth=md5 is for iax.conf, not sip.conf |
| 20:24.29 | markit | I see |
| 20:24.46 | dlynes | markit: generally, you don't want an auth= line in your sip.conf |
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| 20:30.57 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 20:31.12 | markit | I also have some of these messages, time to time: ast_get_srv: SRV lookup for '_sip._udp.sip.squillo.it' mapped to host sip.squillo.it, port 5060 |
| 20:31.42 | markit | are them of any interest / problmematic simpthom? |
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| 20:59.36 | *** join/#asterisk etfonhomey_ (n=chatzill@74.131.130.161) |
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| 21:07.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:12.57 | dlynes | markit: that just means that your dns server is providing an SRV entry for your sip server |
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| 21:37.33 | *** part/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 21:52.57 | etfonhomey_ | Does * use UDP sip or TCP sip? |
| 21:53.52 | etfonhomey_ | ~pb |
| 21:53.53 | 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 |
| 21:53.54 | *** join/#asterisk jedaustin (i=PJirc@63-253-153-93.ip.mcleodusa.net) |
| 21:55.20 | jedaustin | I have asterisk/Freepbx setup with device and user with multiple sip devices pointing to the same user. When you call the user extension all the phones ring as expected. When the first call is answered, additional calls go to VM.. How do I get it to ring the remaining devices instead before timing out and going go voicemail? |
| 21:57.22 | [TK]D-Fender | etfonhomey_, SIP only until 1.6 |
| 21:57.45 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:57.46 | [TK]D-Fender | jedaustin, FreePBX is not supported here, please refer to their support channels. |
| 21:57.54 | [TK]D-Fender | ~freepbx |
| 21:57.55 | 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 |
| 21:58.05 | etfonhomey_ | [TK]D-Fender, is it SIP via tcp or SIP via udp, though? |
| 21:58.37 | [TK]D-Fender | etfonhomey_, Sorry, meant to say : SIP via UDP only. SIP TCP support will come with 1.6 supposedly |
| 22:00.13 | etfonhomey_ | [TK]D-Fender, Thanks. Next question: With * behind a PIX, you're supposed to disable the "fixup protocol sip udp 5060", right? |
| 22:00.42 | [TK]D-Fender | etfonhomey_, last I heard yeah, you want the PIX to keep out of things. |
| 22:03.09 | rob0 | Does a PIX do anything right? SMTP fixup is a/k/a fuxup. |
| 22:03.44 | [TK]D-Fender | rob0, not much... its the worst thing they've made I believe. |
| 22:11.09 | etfonhomey_ | [TK]D-Fender, Can you tell me what would cause the 404 in this sip debug output? http://pastebin.ca/837766 |
| 22:11.31 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 22:11.33 | [TK]D-Fender | etfonhomey_, the WHOLE CALL PLEASE |
| 22:11.38 | etfonhomey_ | [TK]D-Fender, If you look at the sip show peers results, this registration is supposedly OK. |
| 22:12.14 | etfonhomey_ | [TK]D-Fender, this is not a result of a call. |
| 22:12.17 | [TK]D-Fender | etfonhomey_, registration does NOT mean you peer is set up right! |
| 22:12.33 | [TK]D-Fender | *sheesh* |
| 22:12.37 | etfonhomey_ | [TK]D-Fender, just means it registered right... |
| 22:12.45 | [TK]D-Fender | etfonhomey_, .... |
| 22:14.22 | etfonhomey_ | [TK]D-Fender, now, I'll make a call. The symptoms are that if I call my cell phone using this SIP provider, it takes over a minute for the call to ring (on both the cell phone and on the local phone I'm using to call from). |
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| 22:27.07 | etfonhomey_ | [TK]D-Fender, is there a way to redirect the console output to a file? |
| 22:29.23 | *** join/#asterisk d-tech (n=d-dtech@72.245.233.107) |
| 22:30.48 | d-tech | I executed a 'yum install kernel-devel' and still do not have a kernel source ... can someone give me a hint? |
| 22:34.06 | mihinomenest | why don't you just download them as a tarball and extract that? |
| 22:38.22 | mvanbaak | d-tech: try kernel-sources |
| 22:45.38 | etfonhomey_ | [TK]D-Fender, here's most of a call attempt http://pastebin.ca/837805 |
| 22:46.56 | [TK]D-Fender | etfonhomey_, # |
| 22:46.56 | [TK]D-Fender | X-Asterisk-HangupCause: Network out of order <-- |
| 22:47.09 | [TK]D-Fender | etfonhomey_, And they are 404-ing your Quailfy. Disable it. |
| 22:47.26 | etfonhomey_ | aka qualify=no? |
| 22:47.39 | [TK]D-Fender | etfonhomey_, .... |
| 22:50.01 | etfonhomey_ | [TK]D-Fender, the Hangup is happening when I actually hangup the call from the originating phone once it finally starts ringing (after over a minute of silence). |
| 22:50.21 | [TK]D-Fender | etfonhomey_, they list a network error, not you |
| 22:50.29 | [TK]D-Fender | etfonhomey_, looks like their side has issues |
| 22:51.20 | etfonhomey_ | [TK]D-Fender, the network error being "Network out of order"? |
| 22:51.51 | [TK]D-Fender | etfonhomey_, says it out loud, doesn't it? |
| 23:00.16 | d-tech | mvanbaak: no difference |
| 23:01.04 | d-tech | it load the repo lists and then says 'nothing to do' |
| 23:10.50 | tzafrir_home | d-tech, uname -r; rpm -qa | grep kernel |
| 23:11.00 | *** part/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 23:11.19 | tzafrir_home | I suspect you installed kernel-devel of a newer version |
| 23:11.37 | tzafrir_home | With some luck: ./install_prereq install |
| 23:11.44 | tzafrir_home | would get the right version |
| 23:11.53 | mvanbaak | gheh |
| 23:12.02 | mvanbaak | I really like the 'with some luck' |
| 23:30.36 | *** join/#asterisk RoyKa (n=roy@ip-176-23-149-91.dialup.ice.no) |
| 23:30.57 | markit | I've an ag468 ATA, I've upgraded the firmware but now it asks for username and password, and the "admin / voip" combination of the manual does not work. Anyone has that device and knows why? since has telnet interface, any tool that can discover by "brute force"? |
| 23:30.57 | Nugget | telnet is eeeeeeevil! |
| 23:33.21 | dacs | Nugget: ssh is GOD! |
| 23:33.24 | dacs | lol |
| 23:33.32 | mvanbaak | Nugget: wouldn't it be cool if all voip phones and ata's would come with an ssh server ? |
| 23:34.10 | dacs | mvanbaak: that would be like making love to a hot blond! |
| 23:34.35 | mvanbaak | dacs: now calm down. it would be nice, but not _that_ nice |
| 23:34.46 | Nugget | yay ssh |
| 23:35.18 | mvanbaak | making love to a hot blond > ssh |
| 23:39.40 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 23:41.57 | tzafrir_home | You can view star-wars through telnet://towel.blinkenlights.nl . Yet voices in the channel say |
| 23:42.33 | tzafrir_home | (ah, ok. Just testing is that was fully-automated) |
| 23:43.11 | mvanbaak | blinkenlights.nl. gheh |
| 23:43.43 | dacs | anyone know a software/guide that will convert PDF to audio , that way i can load it in my ipod a listen to it. |
| 23:43.47 | dacs | please |
| 23:43.56 | mvanbaak | hahahahahaha |
| 23:44.08 | mvanbaak | sure |
| 23:44.20 | mvanbaak | cat file.pdf | festival |
| 23:44.27 | d-tech | found the problem with 'yum install kernel-devel' ... note CentOS now 'excludes=kernel*' by default ... you'll need to edit the yum.conf if you want a source |
| 23:44.55 | mvanbaak | d-tech: that's just.......weird |
| 23:44.57 | mvanbaak | stupid centos |
| 23:45.38 | d-tech | no-one wants you to have a kernel-tree anymore??? |
| 23:46.09 | mvanbaak | no-one in the centos team |
| 23:46.10 | tzafrir_home | d-tech, they want to prevent you from upgrading the kernel too easily? |
| 23:46.19 | mvanbaak | sane distributions dont do that |
| 23:46.28 | dacs | mvanbaak: and that will output to an ipod format? |
| 23:46.48 | mvanbaak | dacs: no, that will read the source of a pdf |
| 23:46.50 | tzafrir_home | mvanbaak, xpdf-utils has pdftotext |
| 23:46.51 | mvanbaak | but you dont want that |
| 23:47.08 | mvanbaak | tzafrir_home: I know |
| 23:47.17 | mvanbaak | but most PDF files are just one big image |
| 23:47.41 | mvanbaak | so no text can be extracted |
| 23:47.41 | tzafrir_home | That requires an OCR |
| 23:47.51 | mvanbaak | yup |
| 23:48.04 | tzafrir_home | Both tiff and PDF are common inputs to OCR programs |
| 23:48.05 | mvanbaak | ok, dont look weird when I type strange now |
| 23:48.12 | mvanbaak | my cat is walking around on my laptop |
| 23:48.23 | mvanbaak | oooooooooooooooooooooooooooooooooooooooooooooooooooooooaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa |
| 23:48.49 | tzafrir_home | cat >/dev/null #:-( |
| 23:49.25 | dacs | tzafrir_home: what do you mean " xpdf-utils has pdftotext" |
| 23:49.44 | dacs | mvanbaak: why i don't want to do that |
| 23:49.52 | tzafrir_home | The package xpdf-utls (in Debian) has a program called pdftotext |
| 23:50.07 | mvanbaak | dacs: well try it: cat file.pdf |
| 23:50.11 | mvanbaak | and see for yourself |
| 23:50.33 | dacs | tzafrir_home: happen that i do use debian :) |
| 23:57.25 | dacs | so PDFtoTEXT -> festival |