| 00:03.34 | *** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl) |
| 00:03.50 | pooh_ | Hi all |
| 00:04.30 | pooh_ | Had a bad experience today with a SIP device, no sound.... although UDP 5060 and UDP 10000-2000 where open |
| 00:04.50 | pooh_ | Do I forget something here pls? |
| 00:05.25 | pooh_ | device -> NAT -> Internet -> NAT -> Asterisk |
| 00:05.37 | pooh_ | Yeah , I know the worst scenario..... |
| 00:05.51 | pooh_ | On both NAT's all ports where opened up |
| 00:05.57 | pooh_ | but no luck |
| 00:06.01 | pooh_ | pls advice |
| 00:07.53 | Ariel_ | yes we are alive |
| 00:08.10 | pooh_ | :-) luckily |
| 00:08.42 | pooh_ | I know NAT sound and SIP is a weird bunch |
| 00:09.42 | Ariel_ | you have double nat |
| 00:10.10 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 00:10.50 | FarrisG | can someone please suggest an asterisk-compatible hardware sip phone with a good noise-cancelling speakerphone that costs less than $400? |
| 00:11.00 | pooh_ | Ariel_: yes |
| 00:11.07 | pooh_ | the worst |
| 00:12.27 | pooh_ | Signaling goes just fine and I can see the dialplan scroll by, but no sound |
| 00:12.40 | pooh_ | no sound both ways |
| 00:13.42 | Ariel_ | ok lets do this. |
| 00:14.05 | Ariel_ | on the asterisk box do sip show peer then the sip device number. |
| 00:14.35 | pooh_ | show peer shows the INTERNAL IP number of the SIP device within the NATTED network |
| 00:14.36 | Ariel_ | see what it says |
| 00:15.18 | Ariel_ | the natted network for the sip phone not the asterisk box group. |
| 00:15.38 | Ariel_ | in that case you need to setup external ip address setting on the phone. |
| 00:15.51 | pooh_ | yes, I am talking about the device network, the internal IP is shown with show peers |
| 00:16.11 | pooh_ | so, 192.168.226.112 |
| 00:16.18 | pooh_ | not the external IP |
| 00:16.41 | pooh_ | hold on |
| 00:17.03 | pooh_ | correction I mixed them up, show peers shows the EXTERNAL IP, Show channels shows the internal IP |
| 00:17.58 | pooh_ | Show channels while in a call (with no sound) ofcourse |
| 00:18.12 | gambolputty | anyone have odbc working? |
| 00:18.51 | Ariel_ | then do sip debug and see what address it's using when the calls comes in and goes out. |
| 00:19.18 | pooh_ | ok, I will do that 2morrow, please tell me what SHOULD be in the header |
| 00:20.00 | Ariel_ | it should tell you the address it's coming from. it should be the extern one |
| 00:20.03 | pooh_ | lokking = looking |
| 00:20.08 | pooh_ | ok |
| 00:20.33 | pooh_ | and if it is the INTERNAL one, what goes wrong do you think ? |
| 00:21.29 | *** join/#asterisk menger (~menger@dsl-179.77.240.220.rns02-dryb-mel.dsl.comindico.com.au) |
| 00:27.39 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-208-125-78.dsl.sfldmi.ameritech.net) |
| 00:33.29 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 00:42.39 | *** part/#asterisk hyperion (~ircII@2002:5063:d67d:0:0:0:0:0) |
| 00:42.52 | *** join/#asterisk DJ-Studd (~djstudd@adsl-23-162.swiftdsl.com.au) |
| 00:43.00 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
| 00:43.20 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 00:43.42 | [Latre] | someone has configured parking calls.....i have a problem... |
| 00:44.41 | DJ-Studd | Hi Guys. A quick question... our current phone system allows us to pickup one of 5 outgoing lines. If someone is on hold, they can be picked up from another phone. I'm currently testing a Grandstream phone with Asterisk@Home. How would I set it up so it can be used like we use our current phone system. IE, allow someone on a phone to "Pickup Joe on line 3" |
| 00:45.24 | Sed[Mac] | Bt100? |
| 00:47.52 | JerJer | first of, by not using asterisk@home |
| 00:48.18 | MikeJ[Laptop] | hey jer jer.. |
| 00:48.28 | JerJer | moo |
| 00:48.59 | MikeJ[Laptop] | M4101? |
| 00:49.11 | MikeJ[Laptop] | still an issue? |
| 00:49.48 | JerJer | vechers Just to throw some recap and info into the mix: this problem was fixed in 2519 as of HEAD 04/21. JerJer reported the problem again on 04/27. As of current CVS the problem is still there. |
| 00:50.15 | Ariel_ | DJ-Studd, go to the #amportal they can help you there. |
| 00:50.51 | Ariel_ | [Latre], state your problem it's the only way someone can help. |
| 00:50.54 | MikeJ[Laptop] | yes, and at least kp had other questions and such... |
| 00:56.55 | DJ-Studd | JerJer: Yup, I'm going to be spending some time learning the real Asterisk. A@H is just for testing in the mean time |
| 00:57.42 | Ariel_ | DJ-Studd, it's called pickup groups all so parking. |
| 00:57.47 | Ariel_ | and a@h has them. |
| 00:58.11 | DJ-Studd | Thanks Ariel_, I'll have a read into those |
| 00:59.24 | *** join/#asterisk postel (~armaos@postel.user) |
| 00:59.27 | Ariel_ | DJ-Studd, the #amportal can help you out allot with that. It's here on freenode.net |
| 01:01.22 | DJ-Studd | Yup, thanks ariel |
| 01:02.56 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 01:04.53 | JerJer | MikeJ[Laptop]: i am not in development mode right now - sorry |
| 01:08.12 | [Latre] | Ariel_: my problem is when i recive a call in sip hpne(budgetone100) all its ok, but i want transfer this call to another ext and I dial #700 the other person listen music but i cant transfer, if a hangup the call is cut |
| 01:08.57 | [Latre] | Ariel_: this phones have a button (transfer) i dial that button too but the other person only listen music and not trasfer to the other ext |
| 01:09.18 | *** join/#asterisk key2 (~tree@gob75-2-81-56-64-17.fbx.proxad.net) |
| 01:09.20 | key2 | re |
| 01:09.42 | key2 | how and where am I supposed to tell asterisk to go find the sounds in /var/lib/asterisk/sounds ? |
| 01:12.29 | Ariel_ | [Latre], I have not used any barbie tone phones. But if you defined parking correctly it will play for you which location the phone is at. |
| 01:12.31 | shido6 | game 7 |
| 01:12.34 | shido6 | come on ppl |
| 01:12.39 | shido6 | where's your nba cheer |
| 01:12.56 | Ariel_ | shido6, baby has my tv she is watching pooh bear. |
| 01:13.04 | shido6 | well |
| 01:13.07 | shido6 | damn |
| 01:13.07 | Ariel_ | status |
| 01:13.19 | shido6 | launch the PVR |
| 01:13.28 | shido6 | via * |
| 01:13.31 | shido6 | and record it |
| 01:13.36 | shido6 | wanna use my app? :) |
| 01:13.39 | shido6 | phone in record a channel |
| 01:14.04 | Ariel_ | pvr works for the dishnetwork one? |
| 01:15.54 | shido6 | is your dishnetwork "tweaked" |
| 01:15.55 | key2 | so no one could tell me how to tell asterisk to go find the sounds in /var/lib/asterisk/sounds ? |
| 01:16.05 | shido6 | key2 |
| 01:16.06 | shido6 | are you serious |
| 01:16.10 | shido6 | playback , background |
| 01:16.15 | shido6 | they all look for your sounds in that directory |
| 01:16.30 | key2 | shido6: well no aparently |
| 01:16.43 | shido6 | you dont need to "look" |
| 01:16.51 | shido6 | Playback,filename |
| 01:16.54 | shido6 | without the extension |
| 01:16.57 | shido6 | ok |
| 01:16.58 | shido6 | back to the game |
| 01:17.00 | shido6 | bye |
| 01:17.26 | key2 | shido6: honestly, when I do a Playback(welcome) asterisk tells me It can't fine the file, i have to do a Playback(/var/lib/asterisk/sounds/welcome) untill it plays it |
| 01:18.02 | Ariel_ | key see it welcom is there |
| 01:18.16 | key2 | welcome.gsm is there yes |
| 01:18.27 | key2 | but I have to write the path untill it plays it |
| 01:18.36 | key2 | Playback(welcome) doesnt work |
| 01:18.59 | `Sauron | configure your sound path properly |
| 01:19.06 | *** join/#asterisk mag_um (~mag_um@219.95.244.1) |
| 01:19.13 | newl | Playback(nobody-but-chickens) |
| 01:19.19 | key2 | <PROTECTED> |
| 01:19.19 | key2 | Jun 23 21:18:59 WARNING[8163]: file.c:475 ast_openstream: File welcome does not exist in any format |
| 01:19.19 | key2 | Jun 23 21:18:59 WARNING[8163]: file.c:779 ast_streamfile: Unable to open welcome (format gsm): No such file or directory |
| 01:19.30 | Ariel_ | look in /etc/asterisk/asterisk.conf for locations |
| 01:19.54 | key2 | Ariel_: and there is nothing saying sounds => /var/lib/asterisk/sounds |
| 01:20.16 | key2 | and I dunno what to add, it doesnt seam to be "astsounds" |
| 01:21.05 | Ariel_ | key2, this is what controls it. astvarlibdir => /var/lib/asterisk |
| 01:21.36 | key2 | Ariel_: that's what it's set at |
| 01:21.40 | Ariel_ | also do you have asterisk with the correct user rights. |
| 01:21.48 | key2 | what you mean ? |
| 01:21.52 | *** join/#asterisk Sed[Mac] (~brandon@sedorox.staff.smartserv) |
| 01:22.22 | key2 | I can read it with Playback(/var/lib/asterisk/sounds/welcome) but not with Playback(welcome) |
| 01:22.28 | key2 | so I guess it's not a right probleme |
| 01:23.28 | Ariel_ | key2, ok next how did you install the system which distro? which version of asterisk? |
| 01:23.48 | key2 | Ariel_: I installed with debian, doing a apt-get install |
| 01:24.05 | shido6 | oooh |
| 01:24.07 | shido6 | ubuntu? |
| 01:26.09 | *** join/#asterisk nifter (~nifter@190-141.SPEEDe.golden.net) |
| 01:28.58 | Ariel_ | key2, I know that debianputs it in different locations last I use it. |
| 01:31.42 | *** join/#asterisk hypa7ia (~leigh@69.158.180.106) |
| 01:32.01 | robl^ | debian puts it in /usr/lib/asterisk/sounds |
| 01:32.30 | key2 | robl^: but you don't know where to say it in asterisk ? |
| 01:33.26 | robl^ | its done at compile time |
| 01:35.02 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 01:36.53 | shido6 | 12 to 13 |
| 01:36.56 | shido6 | sa |
| 01:37.22 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 01:38.45 | *** join/#asterisk eric`` (~e@adsl-8-101-134.mia.bellsouth.net) |
| 01:40.40 | `Sauron | Nugget, you around? |
| 01:45.25 | robl^ | key2, the sound file location is based on the defintion of $ASTVARLIBDIR, which by default is /var/lib/asterisk/. sounds go in $(ASTVARLIBDIR)/sounds |
| 01:45.51 | robl^ | key2, this is in the Makefile used to compile/build asterisk from source code |
| 01:46.15 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
| 01:47.42 | robl^ | key2, if you installed Asterisk by using "aapt-get install" then you are stuck with the location Debian thinks is best. |
| 01:48.50 | *** join/#asterisk cryptnix (~NNSCRIPT@66-227-239-161.dhcp.bycy.mi.charter.com) |
| 01:49.08 | cryptnix | well this blows my IP changed and now my shit with broadvoice just took a poop :( |
| 01:49.34 | file[laptop] | well |
| 01:49.36 | file[laptop] | I graduated |
| 01:49.41 | file[laptop] | I'm no longer in high school |
| 01:50.07 | hardwire | grr |
| 01:50.12 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 01:50.15 | robl^ | file[laptop], highschool?!!? dang, yer a youngun'! (CONGRATS!) |
| 01:50.17 | hardwire | why does digium need to make 3.3 volt only supplies? |
| 01:50.21 | hardwire | whats the point? |
| 01:50.48 | file[laptop] | the chipset specifically states not to use it with 5V |
| 01:50.55 | *** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 01:50.56 | hardwire | well you can't |
| 01:51.00 | file[laptop] | thus why there is two... |
| 01:51.01 | hardwire | its keyed |
| 01:51.10 | Chuji | evening kids |
| 01:51.11 | file[laptop] | well, you can... |
| 01:51.15 | hardwire | but can't you do voltage regulation? |
| 01:51.21 | file[laptop] | they just don't recommend it |
| 01:51.28 | hardwire | file[desk]: its keyed. |
| 01:51.39 | hardwire | you have to use the backwards PCI ports |
| 01:52.52 | hardwire | which means using 2u and 1u servers is out of the question |
| 01:53.05 | file[laptop] | ugh my head hurts |
| 01:53.13 | hardwire | I need to pee |
| 01:54.26 | robl^ | hardwire, remeber to wash your hands when you finish! |
| 01:56.05 | hardwire | code. |
| 01:57.49 | JerJer[drunk] | hardwire: try it |
| 01:57.58 | hardwire | I thought this was on the wiki |
| 01:58.03 | hardwire | but I am having a hard time finding it |
| 01:58.10 | JerJer[drunk] | just do it |
| 01:58.28 | JerJer[drunk] | post a DID that lands on that T-1 into some chat channel |
| 01:58.51 | JerJer[drunk] | send the calls out to some iax or SIP endpoint somewhere |
| 02:00.46 | hardwire | heh |
| 02:00.48 | hardwire | this is going to suck |
| 02:01.17 | hardwire | these guys are going to need a sync DSL at 2Mbps |
| 02:01.26 | hardwire | and they don't even think they can get that |
| 02:01.39 | MikeJ[Laptop] | file[laptop] !!!!! |
| 02:01.49 | MikeJ[Laptop] | scorecheck anyone? |
| 02:01.52 | hardwire | hey you guys match |
| 02:02.05 | MikeJ[Laptop] | naw, his is a powerbook... |
| 02:02.46 | *** join/#asterisk mrplum (~m@24-52-166-190.lndnnh.adelphia.net) |
| 02:03.33 | MikeJ[Laptop] | ms. peacock, in the library, with the candlestick |
| 02:04.22 | JerJer[drunk] | hickory, dickery, doc |
| 02:04.47 | JerJer[drunk] | jack and jill went up the hell with $1.25. |
| 02:04.57 | shido6 | LOL! |
| 02:05.18 | JerJer[drunk] | jill came back with two fifty |
| 02:05.31 | MikeJ[Laptop] | what's the score? |
| 02:05.37 | JerJer[drunk] | 29 |
| 02:05.37 | shido6 | 29 |
| 02:05.37 | JerJer[drunk] | 28 |
| 02:05.38 | shido6 | 28 |
| 02:05.42 | shido6 | SA |
| 02:05.50 | JerJer[drunk] | det up |
| 02:05.59 | shido6 | oh shit I missed one |
| 02:06.03 | shido6 | NOOOOOOOOOOO |
| 02:06.05 | shido6 | abc is screwing up |
| 02:06.09 | JerJer[drunk] | and i'm the drunk one |
| 02:06.15 | shido6 | ffDf |
| 02:06.29 | JerJer[drunk] | misss |
| 02:06.31 | JerJer[drunk] | damnit |
| 02:07.42 | MikeJ[Laptop] | so it is a game that I can watch the last 2 min of is what you are saying |
| 02:09.48 | JerJer[drunk] | i foresee OT |
| 02:09.55 | shido6 | but of course |
| 02:09.56 | JerJer[drunk] | they both are playing hard |
| 02:09.57 | shido6 | billups |
| 02:09.59 | shido6 | will score a 3 |
| 02:10.04 | shido6 | with .3 on the clock |
| 02:10.08 | shido6 | liek usual |
| 02:11.03 | JerJer[drunk] | ! |
| 02:11.39 | *** join/#asterisk cryptnix (~NNSCRIPT@66-227-239-161.dhcp.bycy.mi.charter.com) |
| 02:11.40 | hardwire | P4.. 3GHZ+HT.. 2Gig.. I should be able to handle 60-80 Zap->GSM |
| 02:11.59 | MikeJ[Laptop] | hardwire, if you say so |
| 02:12.08 | hardwire | http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office+100 |
| 02:12.13 | hardwire | going off of that |
| 02:12.24 | shido6 | i need a pvr card |
| 02:12.24 | MikeJ[Laptop] | I think that is the best bet... |
| 02:12.27 | shido6 | for HD TV |
| 02:12.41 | shido6 | for dishnet |
| 02:12.48 | MikeJ[Laptop] | testing in your environment is a waste anyway... why bother |
| 02:12.53 | MikeJ[Laptop] | ;) |
| 02:12.54 | cryptnix | I just setup a m0n0wall firewall have done alittle reasearch cause i'm getting the issues with my voice not being heard by the people i'm calling to test it ... is there any other suggetsions other then the [general] mod's for nat use to do like ... port forwards to it or something?! |
| 02:13.45 | MikeJ[Laptop] | cryptnix, start by stoping using 0's for o's... you will look less l337 that way. |
| 02:13.54 | JerJer[drunk] | kid rock! |
| 02:14.00 | MikeJ[Laptop] | kiiiiiiiiddd |
| 02:14.04 | MikeJ[Laptop] | what about him |
| 02:14.12 | JerJer[drunk] | at the game |
| 02:14.17 | JerJer[drunk] | like 3rd row back |
| 02:14.18 | cryptnix | MikeJ[Laptop]: apparently you haven't used it. |
| 02:14.22 | hardwire | MikeJ[Laptop]: well I did 4 Zap->GSM calls today on a 1ghz Via C3 and it did nothing to the load |
| 02:14.29 | cryptnix | its the actual name ... http://m0n0.ch/wall |
| 02:14.34 | hardwire | so I am wondering how 24 would change much |
| 02:14.48 | hardwire | oh well.. just have to test it |
| 02:14.50 | MikeJ[Laptop] | even more moronic |
| 02:15.14 | cryptnix | let me remind you ... your commented on it. |
| 02:15.18 | cryptnix | *you even |
| 02:15.33 | MikeJ[Laptop] | what the hell is wrong with people |
| 02:15.49 | MikeJ[Laptop] | lame lame lame lame |
| 02:15.53 | MikeJ[Laptop] | score |
| 02:16.22 | MikeJ[Laptop] | hardwire, ahhh.. real test results.. |
| 02:16.30 | meynax | somebody has experience blank calls using asterisk+amp+t1??? |
| 02:16.33 | MikeJ[Laptop] | score? |
| 02:16.45 | MikeJ[Laptop] | blank calls? |
| 02:17.10 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 02:17.36 | meynax | I just put a T1 card in the asterisk and when I made calls outside, all sounds god, dial tone, etc and when the other part answer |
| 02:17.39 | meynax | nothing |
| 02:17.40 | Juxt | evening |
| 02:17.43 | meynax | just silence |
| 02:18.08 | Juxt | does anyone know how much cpu is requered to pass thru say 1 g729 channel |
| 02:18.28 | MikeJ[Laptop] | cpu for passthrough? |
| 02:18.34 | MikeJ[Laptop] | are you being serious? |
| 02:18.38 | Juxt | yes |
| 02:18.42 | Juxt | here's why |
| 02:18.58 | Juxt | i have a celeron 300 box that maxes out at about 6 calls, no transcoding at all |
| 02:19.07 | MikeJ[Laptop] | passthrough is not cpu intensive at all |
| 02:19.08 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 02:19.10 | meynax | I test the internal extension and all is working fine, this issue is only when we make outside calls to the PSTN using the digium T1 card |
| 02:19.15 | postel | Juxt passthru does no transcoding, no transcoding no cpu |
| 02:19.38 | MikeJ[Laptop] | well, not no cpu, but no cpu for transcoding |
| 02:19.47 | postel | right |
| 02:19.59 | MikeJ[Laptop] | Juxt, maxes out how, what happens? |
| 02:20.03 | meynax | someone has an idea about the silence calls? |
| 02:20.05 | Juxt | load=1 |
| 02:20.40 | MikeJ[Laptop] | asterisk is doing that with 6 passthrough 729 calls? |
| 02:20.45 | Juxt | yep |
| 02:20.53 | MikeJ[Laptop] | are you sure it is the asterisk process? |
| 02:20.59 | Juxt | let me try agian |
| 02:21.06 | MikeJ[Laptop] | what else is on the box/ |
| 02:21.13 | Juxt | postfix |
| 02:21.16 | JerJer[drunk] | framebuffer? |
| 02:21.21 | JerJer[drunk] | irq sharing? |
| 02:21.33 | Juxt | oh irq sharing is a good question |
| 02:21.36 | shido6 | commercials |
| 02:21.42 | shido6 | FASTFWD the commercials JJ |
| 02:21.47 | Juxt | nope |
| 02:21.51 | Juxt | no irq sharing |
| 02:22.06 | MikeJ[Laptop] | Juxt, just sip in and out.. are you sure no transcoding? |
| 02:22.13 | DJ-Studd | Urg @ "All Circuits are Busy now message" |
| 02:22.18 | Juxt | it's iax, in and out |
| 02:22.20 | Juxt | no transcoding |
| 02:22.56 | shido6 | celeron |
| 02:22.58 | shido6 | wtf |
| 02:23.04 | shido6 | 8k cache celeron? |
| 02:23.10 | Juxt | maybe, def. very old |
| 02:23.10 | shido6 | the super cheap one? |
| 02:23.31 | Juxt | no idea, what happened my main box took a massive crap and i had to put something together out of spare parts |
| 02:23.35 | hardwire | super cheap.. super cheap.. |
| 02:23.45 | shido6 | dont go cheap on the proc - manG |
| 02:23.51 | shido6 | esp not a celeron |
| 02:24.02 | shido6 | sell a spare tire |
| 02:24.02 | Juxt | i am shoppoing for a new box |
| 02:24.11 | Juxt | this time something that can transcode |
| 02:24.11 | shido6 | stay in for a couple weeks |
| 02:24.14 | shido6 | ave some gas money |
| 02:24.21 | shido6 | ge thte no name ice cream |
| 02:24.26 | shido6 | and get you a p4 |
| 02:24.30 | shido6 | eat lots of ramen |
| 02:24.38 | shido6 | less asterisk problems |
| 02:24.54 | mrplum | heh |
| 02:25.02 | shido6 | dont go to the cheap chinese joint tho - less u have a constipation problems |
| 02:25.27 | shido6 | so you get a p4 and your sodium intake increases |
| 02:25.27 | Juxt | ahaha |
| 02:25.35 | shido6 | high blood pressure for a fast proc |
| 02:25.38 | shido6 | decisions |
| 02:25.59 | shido6 | not sure if u can still get the government cheese these days |
| 02:26.07 | shido6 | cash in some food stamps |
| 02:26.27 | shido6 | now these ppl are gonna wrap on tv for 13 minutes |
| 02:26.47 | shido6 | I hate bill walton |
| 02:26.53 | shido6 | he can suck a duck and die |
| 02:27.14 | shido6 | he changes his "opinion" faster than you can smell the shit he talks |
| 02:27.46 | Juxt | ok if i set canreinvite=no on iax channel wouldn't it prevent it from "trying to establish a native bridge" ? |
| 02:28.21 | shido6 | Juxt |
| 02:28.30 | shido6 | notransfer=yes |
| 02:28.34 | shido6 | should do it |
| 02:28.34 | shido6 | or |
| 02:28.36 | shido6 | change codecs |
| 02:28.38 | shido6 | in one |
| 02:28.39 | shido6 | out another |
| 02:28.42 | shido6 | no native bridging |
| 02:28.49 | shido6 | if u STILL see native bridging |
| 02:28.52 | shido6 | change the iax code |
| 02:29.01 | mrplum | I thought native bridge has to do with codec conversion, and transfering had to do with handing off the connection to the destination isntead of being a middleman |
| 02:29.08 | shido6 | and tell it to screw off |
| 02:29.19 | Juxt | ok so i was wrong, thanks for correcting me |
| 02:29.19 | shido6 | mmm mocha |
| 02:29.21 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 02:29.21 | shido6 | LOL |
| 02:29.50 | shido6 | can a ISP or voice service provider |
| 02:29.55 | shido6 | get a JD power and associates rank |
| 02:29.56 | shido6 | ? |
| 02:30.52 | MikeJ[Laptop] | no |
| 02:31.03 | Juxt | hmm ok now my cpu is fine |
| 02:31.04 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 02:31.16 | shido6 | turned the bitorrent porn fileshare off ? |
| 02:31.19 | shido6 | that'll do it |
| 02:31.23 | isamar | hi folks! |
| 02:31.23 | Juxt | but i see the issue - it's my greylist getting beat by spammers |
| 02:31.28 | Juxt | shido6: nice one! |
| 02:31.36 | shido6 | :) |
| 02:31.36 | isamar | which mask in extensions.conf I use for: |
| 02:31.40 | robl^ | nice porn? |
| 02:31.49 | isamar | 55xx8 ? |
| 02:32.00 | isamar | it would be 55[1-9]8 ? |
| 02:32.09 | robl^ | _5XX8 |
| 02:32.18 | robl^ | er.. _55xx8 |
| 02:33.10 | isamar | ok |
| 02:33.13 | isamar | thanks dude |
| 02:34.13 | *** join/#asterisk blindmice (~andrewb@202.83.84.15) |
| 02:34.58 | robl^ | ismar: http://voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Patterns <-- check that out. |
| 02:35.29 | robl^ | isamar, X = 0-9, Z = 1-9, N = 2-9 |
| 02:35.44 | isamar | woow.. cool man |
| 02:35.54 | isamar | bookmarked |
| 02:36.24 | Juxt | ok the cpu issue has gone away |
| 02:36.28 | Juxt | thank you guys |
| 02:36.40 | Juxt | maybe this celeron can live a little longer LOL |
| 02:36.50 | shido6 | heheh |
| 02:36.52 | shido6 | hack away |
| 02:36.54 | shido6 | the unessentials |
| 02:36.57 | shido6 | bruce lee style |
| 02:37.08 | shido6 | if its running ins ps -aux |
| 02:37.09 | meynax | someone has an idea why I only get silence in the calls made from to a T1 PRI? |
| 02:37.11 | shido6 | why do u need it |
| 02:37.11 | jr99 | how well does the ztdummy module work now a days? Looking to maybe get rid of my two TD400P cards and go with 2 supura devices to handle my FXS lines in the house... |
| 02:37.13 | shido6 | KILL IT |
| 02:37.21 | shido6 | ztdummy |
| 02:37.24 | shido6 | use RTC |
| 02:37.26 | shido6 | not USB |
| 02:37.31 | jr99 | yea, 2.6 |
| 02:37.35 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 02:37.45 | jr99 | 2.6 uses rtc, correct? |
| 02:37.46 | Juxt | btw is it me or livevoip has gone to total crap? |
| 02:38.08 | shido6 | um |
| 02:38.14 | shido6 | werent they always total crap anwyay |
| 02:38.43 | Juxt | well before they had moderately reliable termination |
| 02:38.47 | blindmice | hi all, i have a tdm400p-04b, was working yesterday (fedora core2 2.6.10), but today I rebooted and now modprobe wctdm errors - no such device? |
| 02:39.03 | shido6 | hehe |
| 02:39.05 | shido6 | fedora |
| 02:39.14 | shido6 | ppl just want to learn the hard way |
| 02:39.37 | jr99 | here we go.... |
| 02:40.12 | mrplum | Fedora is such a cool name though |
| 02:40.18 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:40.21 | Juxt | any carrier suggestion that can take about 40K min per day? |
| 02:40.38 | Juxt | looking for around 1.2 cents per min |
| 02:41.05 | blindmice | hehe, well fedora wasn't exactly my first choice it's just that digium said it was supported and since I had FC2 already I went with it.. |
| 02:41.34 | jr99 | ... am I crazy to get rid of the tdm400p and go with a few sipura devices? |
| 02:41.35 | mrplum | blindmice, accidently bump the card out? |
| 02:41.49 | jr99 | blindmice: is the card plugged in with power? |
| 02:42.03 | blindmice | yes has power, have two machines doing the same.. |
| 02:42.12 | jr99 | sun spots? |
| 02:42.48 | mrplum | I don't even know what hardware to get :-/ have an fxo card, considered constructing a line simulator, maybe by an FXS, or an ip phone.... any suggestions? |
| 02:43.07 | MikeJ[Laptop] | jr99, you may want to uncomment the USE_RTC define in ztdummy.c for possible better performance.. |
| 02:43.17 | blindmice | lspci and /proc/pci show the card |
| 02:44.15 | robl^ | isn't there a manager API proxy that uses XML. I swore I saw a link to one before |
| 02:44.20 | blindmice | also - kudzu wants to remove then add the card back in on every 2nd reboot |
| 02:44.23 | mrplum | ah maybe phone to sound card... heh i'm so cheap |
| 02:44.31 | jr99 | MikeJ[Laptop]: ok.. I'm still trying to decide what I am going to do.. If I keep the TDM400P cards I don't need it.. If I scrap them (have 2) and go with sipura devices for 1 FXO and 4-5 FXS I will need something to generate timing.. the docs say that ztdummy is not for production... |
| 02:45.39 | robl^ | blindmice, fedora take lots of "special care" to use with Asterisk. kudzu misidentifies the cards so it tries to reconfigure them when it shouldn't |
| 02:45.57 | *** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc) |
| 02:46.36 | blindmice | robl^ - what linux dist would you recommend for asterisk? |
| 02:46.45 | blindmice | and digium hardware.. |
| 02:47.21 | robl^ | blindmice, Debian is good. some people like gentoo. red hat enterprise 3 is also popular |
| 02:48.00 | shido6 | dont |
| 02:48.01 | shido6 | use |
| 02:48.03 | shido6 | Fedora |
| 02:48.06 | shido6 | and ur done |
| 02:48.07 | shido6 | next |
| 02:48.15 | robl^ | personally, I use Debian. I used to use RHEL3 |
| 02:48.23 | Juxt | ok my * isn't writing userfiled into pgsql cdr |
| 02:48.25 | Juxt | dammit |
| 02:48.50 | shido6 | robl^ I use ubuntu for my desktop when windows pisses me off |
| 02:48.59 | shido6 | when I find a damn video card |
| 02:49.02 | shido6 | thats linux friendly |
| 02:49.04 | shido6 | I |
| 02:49.07 | shido6 | ll go all the wya |
| 02:49.10 | robl^ | shido6, so.. translates to using ubuntu everyday? :) |
| 02:49.19 | shido6 | not until I rid mysel - |
| 02:49.22 | shido6 | thankS FOR THE IDEA |
| 02:49.26 | shido6 | Im puttin ghe radeon on ebay |
| 02:50.17 | robl^ | ubuntu is a nice desktop distro. I used it but switched back to SuSE for certain needs that ubuntu doesn't handle well yet |
| 02:50.30 | shido6 | oh wow |
| 02:50.33 | shido6 | thats disconcerning |
| 02:50.39 | Nugget | what the hell is the deal with ubuntu, exactly? it didn't exist at all and now I can't go twelve minutes without some sweaty-toothed OSShole raving at me about how ubuntu is the only linux for anyone |
| 02:50.39 | shido6 | radeon all in wonder 9600 is 30 bucsk |
| 02:50.46 | shido6 | i have the 9800 |
| 02:50.50 | shido6 | 128 all in wonder |
| 02:50.52 | robl^ | $35! |
| 02:50.55 | shido6 | 35 bucks |
| 02:50.56 | Nugget | what's it do that's so special and who's in charge of the ubuntu flashmob viral marketing? |
| 02:50.58 | shido6 | crap |
| 02:51.04 | shido6 | I cant buy a full tank of gas in my onda |
| 02:51.06 | shido6 | for that price |
| 02:51.38 | Nugget | ubuntu has overnight replaced gentoo in the "most vocally annoying users" department |
| 02:51.56 | robl^ | Nugget, ubuntu isn't special. its just Debian with some of the rough edges smoothed out and more regular releases. |
| 02:52.03 | mrplum | hehe |
| 02:52.13 | pjz | Nugget: what robl^ said. |
| 02:52.25 | pjz | Nugget: makes a nice desktop distro tho, I understand. |
| 02:52.28 | `Sauron | Hum, if I start irssi, and connect to one server |
| 02:52.38 | mrplum | I'd love to see a Debian/Ubuntu comparison and see how much difference you get. |
| 02:52.44 | mrplum | I doubt it's that much better atuned |
| 02:53.21 | jr99 | kinda of OT.. but anyone have any experience with structured cabling? like OnQ, Leviton, etc? |
| 02:53.24 | mrplum | Tried Kubuntu myself and it was horrible, but I hear if you stick within the main respository, Ubuntu is supposed to be compiled strictly for GNOME... |
| 02:53.27 | mrplum | no sound problems... |
| 02:53.30 | mrplum | wee |
| 02:53.40 | pjz | mrplum: oh, not tuned up so much as smoothed out integration wise, I think. |
| 02:53.46 | robl^ | ubuntu was tweaked and supported suspend/sleep on my laptop without having to hack a custom kernel like I have to with Debian |
| 02:54.16 | *** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 02:54.19 | pjz | yeah, lots of stuff that you have to work at to get working under debian works out of the box with ubuntu. |
| 02:54.38 | mrplum | heh "lots" |
| 02:54.45 | mrplum | oh well, linux rocks any way you get it |
| 02:55.21 | robl^ | distros are a preference. you find one that works the way you do. what works for me may not work for tohers. |
| 02:55.34 | shido6 | yeah |
| 02:55.35 | Nugget | linux is poo. |
| 02:55.54 | robl^ | then again, I'm a distro whore. I switch distros every other month :) |
| 02:56.00 | shido6 | hehe |
| 02:56.05 | shido6 | I ran redhat for a minute |
| 02:56.08 | shido6 | then xandros |
| 02:56.10 | shido6 | mandrake |
| 02:56.18 | shido6 | blah blah |
| 02:56.21 | shido6 | i hear you robl^ |
| 02:56.30 | shido6 | I run macos x in pearpc |
| 02:56.32 | shido6 | on both linus |
| 02:56.34 | shido6 | linux |
| 02:56.35 | Nugget | that stops being fun as soon as you have real work to do. :) |
| 02:56.36 | shido6 | and windows |
| 02:56.46 | shido6 | and am waiting on mac os x 86 to be released on te torrents |
| 02:56.48 | shido6 | since it was leaked |
| 02:56.54 | Nugget | if you've got time to reinstall your os every other month you need more interesting things in your world. |
| 02:57.26 | shido6 | yeah |
| 02:57.30 | shido6 | you're right Nugget |
| 02:57.42 | shido6 | wanna go in with me on buying an island in the phillipines? |
| 02:57.44 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 02:58.51 | Nugget | that's the biggest problem I had with linux. |
| 02:58.57 | Nugget | linux isn't about doing things WITH your computer. |
| 02:59.03 | Nugget | linux is about doing things TO your computer. |
| 02:59.12 | mrplum | like a flamethrower does things? |
| 02:59.22 | Nugget | huh? |
| 02:59.36 | shido6 | Ive heard that - but I can flush my toilet , answer the door, check my email, and buy stock with my cisco 7960 with linux |
| 02:59.53 | mrplum | you can flush your toilet |
| 02:59.58 | mrplum | why would you want to? |
| 03:00.03 | mrplum | such a waste of watter |
| 03:00.08 | shido6 | ehehhe |
| 03:00.19 | Nugget | I'd feel safe betting, though, that despite the technical truth that all those things are possible -- you've never done any of them and never will. |
| 03:00.21 | mrplum | I flush like once a week |
| 03:00.28 | mrplum | <- environmentally concious |
| 03:00.38 | shido6 | oh |
| 03:00.38 | Nugget | well, except the email one |
| 03:00.40 | shido6 | lord |
| 03:00.47 | shido6 | no woman in your life eh? |
| 03:00.53 | Nugget | heh |
| 03:02.06 | *** join/#asterisk vuvie (~S@bb219-74-45-231.singnet.com.sg) |
| 03:05.13 | gambolputty | I have * and odbc connected for realtime but phones don't register. |
| 03:05.21 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 03:05.29 | brettnem | hello all |
| 03:05.51 | brettnem | busy night tonight eh? |
| 03:06.39 | brettnem | hey nugget, what's up? |
| 03:07.01 | brettnem | you still hang with Justin? |
| 03:07.04 | JerJer[drunk] | lol woman are overrated |
| 03:07.20 | file[laptop] | no more school for me! |
| 03:07.27 | file[laptop] | now I can get back to actually doing stuff |
| 03:07.56 | brettnem | anyone have any thoughts on using .call files for CTD? |
| 03:08.07 | brettnem | am I insane? |
| 03:08.07 | Nugget | brettnem: yep, he's around. |
| 03:08.08 | blindmice | thanks all, i pulled card out, booted and removed config, install card again - now it works.. i'm going to try debian now :) |
| 03:08.15 | Nugget | we still do our monthly taco nights |
| 03:08.21 | brettnem | tell him I said hello |
| 03:08.24 | Nugget | you bet |
| 03:08.33 | brettnem | oh hey goes to those? |
| 03:08.38 | brettnem | -y |
| 03:08.51 | Nugget | yup |
| 03:09.06 | brettnem | ah |
| 03:09.12 | brettnem | what did you guys call that?? |
| 03:09.45 | brettnem | so has anyone done ctd here?? |
| 03:10.34 | *** part/#asterisk blindmice (~andrewb@202.83.84.15) |
| 03:10.51 | shido6 | what pvr card should i get jerjer for linux pvr with remote |
| 03:10.59 | nifter | Hey guys, anyone have experience with the GXP-2000? |
| 03:11.50 | brettnem | nifter: I have enough experience with grandstream to know I am skeptical |
| 03:11.59 | JerJer[drunk] | shido6: nexus-s |
| 03:12.12 | nifter | <brettnem> haha, i'm getting there too |
| 03:13.13 | nifter | I'm having an issue with 2 of 3 phones. When the phones are sharing their connection with an attached PC...the phones lose their IP within 10min |
| 03:13.36 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:14.37 | postel | nifter: they get ip with static addressing or DHCP? maybe your lease expires |
| 03:15.07 | nifter | <brettnem> dhcp...the lease is set for a few days |
| 03:15.49 | brettnem | I bet it's a bug |
| 03:15.57 | brettnem | they make crrrrrrap |
| 03:16.04 | brettnem | so CTD anyone anyone? |
| 03:16.13 | nifter | <brettnem> can you recommend a |
| 03:16.18 | nifter | "better" phone |
| 03:16.59 | brettnem | cisco 7940 |
| 03:17.04 | brettnem | :) |
| 03:17.19 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 03:17.36 | postel | or 7960, pricing has gone down, and you get 4 more lines than the 2 of the 7940 |
| 03:17.36 | nifter | <brettnem> i need to have stun though |
| 03:17.44 | *** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com) |
| 03:18.03 | iheartcanada | hi, are there any good simple answering machine/faxing software for windows, whether free as in license or free as in beer |
| 03:18.05 | brettnem | whyd o you need stun? |
| 03:18.40 | nifter | <brettnem> the phone are being used by remote users that often stay in hotels and such... |
| 03:20.21 | santiago | Hi, where I can find a patch against asterisk 1.0.7 to solve the stack overflow that is fixed in 1.0.8?, I was unable to find it on bugs.digium.com. |
| 03:21.43 | MikeJ[Laptop] | ummm |
| 03:21.57 | MikeJ[Laptop] | can you give more info? |
| 03:22.21 | santiago | the patch to solve this http://lists.digium.com/pipermail/asterisk-security/2005-June/000032.html |
| 03:22.27 | stormfr | hello |
| 03:22.33 | stormfr | i have a problem with iax trunk |
| 03:22.53 | stormfr | asterisk hang on iaxtrunk |
| 03:23.03 | MikeJ[Laptop] | santiago, I will find the commit, |
| 03:23.05 | stormfr | iax is still alaive |
| 03:23.19 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 03:23.22 | stormfr | (current cvs head from yesterday) |
| 03:23.39 | Nugget | A CS student in his first semester may think a kilobyte contains 1000 bytes. In his last semester he will think that a kilometer contains 1024 meters. |
| 03:23.56 | MikeJ[Laptop] | stormfr, there is a iax codec pref issue causing some issues that was just fixed 20 min ago in cvs |
| 03:24.43 | *** join/#asterisk `Sauron (sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:25.34 | stormfr | i use only these CODEC_PREFS : (ulaw|alaw) |
| 03:25.38 | stormfr | on both part |
| 03:25.43 | stormfr | i haven't check today cvs |
| 03:25.51 | *** join/#asterisk mag_um (~mag_um@219.95.244.1) |
| 03:26.12 | mag_um | hi |
| 03:26.16 | stormfr | thanks i will try and see if it's the problem |
| 03:27.12 | mag_um | I have problem... |
| 03:27.25 | mag_um | I cant forward call using IAX2 |
| 03:27.38 | mag_um | Any one can help me... |
| 03:28.31 | `Sauron | Hum. |
| 03:29.05 | Nugget | your question can be rephrased: "please tell me what to type" |
| 03:29.17 | Nugget | try to figure it out yourself and come back when you have a more specific question |
| 03:30.15 | mag_um | I try it in many ways... |
| 03:30.21 | mag_um | but i could find out |
| 03:30.37 | mag_um | I already reinstall asterisk |
| 03:30.50 | mag_um | but the problem never been solved |
| 03:31.15 | mag_um | My fren has same configuration' |
| 03:31.25 | mag_um | but he can tranfer cll |
| 03:31.33 | mag_um | transfer call |
| 03:31.46 | mag_um | Is that could be network problem |
| 03:31.48 | *** join/#asterisk infinity1 (brendon@solara.netcal.com) |
| 03:31.55 | mag_um | Is that could be network problem..... |
| 03:32.48 | infinity1 | whats a good place to start for an asterisk noob? i had a hard time getting just a softphone get connected to the pbx |
| 03:33.00 | jontow | www.voip-info.org |
| 03:35.30 | infinity1 | whats a good voip provider that is aasterisk friendly? i've google'd and seen so many i'm overwhelmed? |
| 03:36.12 | drumkilla | anybody that has iax support |
| 03:36.30 | drumkilla | from there, decide which payment plan fits your needs best |
| 03:36.33 | drumkilla | some do prepaid ... |
| 03:36.36 | infinity1 | any recommendations for california, usa? |
| 03:36.37 | drumkilla | some a flat fee ... |
| 03:36.51 | drumkilla | some will give you a toll-free number ... some local ... |
| 03:37.06 | infinity1 | do i have to worry about LNP? |
| 03:37.53 | stormfr | just upgrade last zaptel/libpri and now i have this message filling my log "PRI got event: HDLC Abort (6) on Primary D-channel of span 3". span work fine anyway |
| 03:37.55 | stormfr | any idea ? |
| 03:38.22 | JerJer[drunk] | Abort! |
| 03:38.27 | santiago | MikeJ[Laptop], thanks |
| 03:39.45 | stormfr | JerJer[drunk] : yes but why 15 message per second ? a new debug output ? or telco sending abort since pri is maybe not well sync or something similar ? |
| 03:40.07 | JerJer[drunk] | flooding is not good |
| 03:40.14 | JerJer[drunk] | i get them once in a very great while |
| 03:40.23 | JerJer[drunk] | make sure your span line(s) are correct |
| 03:40.31 | JerJer[drunk] | show me them |
| 03:41.12 | JunK-Y | JerJer[drunk]: what did u drink exactly? |
| 03:41.13 | jontow | stormfr; i had that problem last i tried zaptel on freebsd.. 0.9 i think was what i used, from ports |
| 03:41.20 | JerJer[drunk] | vodka |
| 03:41.30 | JunK-Y | u just remember ive few beers + meaghers :) |
| 03:41.31 | JerJer[drunk] | over a 1/5th of absolute vodka |
| 03:41.34 | JerJer[drunk] | nothing less |
| 03:42.08 | JunK-Y | like its day-off tomorrow. |
| 03:45.54 | *** part/#asterisk ilan (ilan@69.60.110.251) |
| 03:47.48 | Nugget | I've already finished the work I'd hoped to accomplish for tomorrow. |
| 03:48.04 | Nugget | so I plan to sit on the patio in the sun, drink beer, and troll on irc all day |
| 03:58.38 | JerJer[drunk] | damnit |
| 04:02.41 | santiago | MikeJ[Laptop], could you find it? |
| 04:03.22 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
| 04:03.36 | SarahEmm | hihi |
| 04:03.40 | MikeJ[Laptop] | sorry, I got distracted by the basketball |
| 04:03.47 | MikeJ[Laptop] | pass me that link again? |
| 04:04.33 | JunK-Y | MikeJ[Laptop]: san antonio won? |
| 04:04.37 | MikeJ[Laptop] | y |
| 04:04.46 | MikeJ[Laptop] | tied at 3rd qtr |
| 04:04.56 | MikeJ[Laptop] | detroit lost by 6 |
| 04:05.15 | JunK-Y | that would be great if detroit has won, u could see them in street :) |
| 04:06.18 | pigpen | nice...I hear horns honking all over the place... |
| 04:06.23 | pigpen | Yes..I live near SA |
| 04:07.44 | Qorky | anyone here know there stuff with capi? i've got the card working now and I can make calls. i just hear no audio. any ideas please ? |
| 04:09.58 | JerJer[drunk] | que-tip |
| 04:10.16 | santiago | MikeJ[Laptop], jeje, np: http://lists.digium.com/pipermail/asterisk-security/2005-June/000032.html |
| 04:15.51 | JerJer[drunk] | already running yesterdays cvs- head so its a non issue |
| 04:16.17 | JerJer[drunk] | plus i dont' enable manager on public interfaces |
| 04:16.41 | santiago | yes, but I want a patch against 1.0.7 |
| 04:16.59 | NewSole | so thats only if manager is running |
| 04:17.44 | *** join/#asterisk lotza (~lotza@cuscon11797.tstt.net.tt) |
| 04:17.44 | JunK-Y | santiago: why not upgrade to .8? |
| 04:17.50 | SarahEmm | .8 is out? |
| 04:17.55 | SarahEmm | duh. see topic, sarah. |
| 04:18.25 | santiago | I want to build the debian package for sarge |
| 04:18.50 | NewSole | that bug.... thats only if manager is running |
| 04:19.36 | JunK-Y | santiago: there's a lot of patches which has been applied to .7 to form .8 |
| 04:20.21 | vuvie | make install asterisk |
| 04:20.41 | vuvie | error: /usr/bin/ld, app_curl.so Error one |
| 04:20.46 | vuvie | cannot find -lidn |
| 04:20.55 | vuvie | what does that mea? |
| 04:20.59 | santiago | NewSole, yes, but it's still a bug |
| 04:21.02 | vuvie | mean |
| 04:21.11 | JunK-Y | vuvie: it cant find ur lib. |
| 04:21.18 | SarahEmm | looks like you need to install libidn |
| 04:21.25 | santiago | JunK-Y, do you known which one fix the bug? |
| 04:21.44 | JunK-Y | santiago: dunno by heart, take at look at asterisk-cvs |
| 04:22.15 | santiago | I can't find it on bugs.digium.com |
| 04:22.18 | santiago | :S |
| 04:22.33 | JunK-Y | santiago: i said asterisk-cvs, not the bug tracker. |
| 04:22.45 | santiago | let me see |
| 04:22.54 | JunK-Y | all changes on bugs.digium.com appear in asterisk-cvs |
| 04:23.35 | santiago | JunK-Y, the asterisk-cvs mailing list? |
| 04:23.54 | JunK-Y | yes |
| 04:23.58 | JunK-Y | its the cvs log. |
| 04:24.04 | vuvie | Junk-Y, checked. libidn installed |
| 04:24.24 | santiago | ok |
| 04:24.40 | SarahEmm | vuvie: do you have libidn.so in one of your library directories? |
| 04:25.48 | vuvie | i have libidn.so.11 and libidn.so.11.4.6 |
| 04:27.20 | vuvie | libidn.so.11 symplics to .4.6 |
| 04:27.28 | vuvie | symlinks |
| 04:27.30 | SarahEmm | okay |
| 04:27.37 | SarahEmm | you need *something* to symlink to libidn.so |
| 04:27.43 | SarahEmm | then re-run ldconfig |
| 04:27.49 | vuvie | ok alright |
| 04:28.46 | MikeJ[Laptop] | santiago, I don't see anything noted as that |
| 04:29.29 | *** join/#asterisk cyberbuddy (~cyberbudd@24-205-143-165.pas-mres.charterpipeline.net) |
| 04:30.31 | cyberbuddy | hey, anyone here? i got a question.. |
| 04:30.56 | JunK-Y | ask |
| 04:31.13 | cyberbuddy | k |
| 04:31.29 | vuvie | ok did |
| 04:31.44 | SarahEmm | vuvie: and? |
| 04:32.15 | vuvie | make ing... |
| 04:32.51 | cyberbuddy | well, i am seeing a lot of places that talk about IAX providers, the most common one being "NuFone". i want an asterisk PBX at my house but i was wondering if i can become my own provider if i purchase the proper equipment and have my own gateway. meaning i dont have to pay monthly. i have a regular PSTN line here at my house. will i need anything else? |
| 04:34.03 | JerJer[drunk] | lol |
| 04:34.20 | JunK-Y | JerJer[drunk]: continue to drink, im following you buddy! |
| 04:34.34 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 04:34.40 | cyberbuddy | heh, sorry but i may sound like a noob to all this VoIP stuff |
| 04:34.54 | cyberbuddy | but if anyone can answer my question without flaming me, i would appreciate it. |
| 04:35.14 | *** join/#asterisk cluecon (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 04:35.36 | SarahEmm | cyberbuddy: you want one external PSTN line coming in.. what inside the house? POTS phones plugged into the * box, VoIP phones, both? |
| 04:36.02 | MikeJ[Laptop] | what, no one has pri's coming into their house? |
| 04:36.08 | SarahEmm | i wish hehe |
| 04:36.17 | cyberbuddy | i have regular phones inside the house connected to my PSTN provider. |
| 04:36.30 | cluecon | MikeJ: you pay the bill and I'll get one. |
| 04:36.34 | MikeJ[Laptop] | I keep my PRI's at the office... |
| 04:36.43 | cyberbuddy | i can get access to a PRI, but it would be outside my house. i need one INSIDE my house |
| 04:36.46 | MikeJ[Laptop] | 100's of did's at my disposal.... |
| 04:36.47 | SarahEmm | cyberbuddy: right.... and that will run into your Asterisk PBX, what kind of phones will you have in the house? |
| 04:36.52 | SarahEmm | cyberbuddy: POTS? SIP? |
| 04:36.56 | cyberbuddy | i want SIP |
| 04:36.59 | MikeJ[Laptop] | POT? |
| 04:37.00 | SarahEmm | okay.. |
| 04:37.08 | SarahEmm | so all you need is 1 FXO interface then |
| 04:37.08 | cyberbuddy | i am willing to purchase one VoIP phone |
| 04:37.11 | cyberbuddy | or make my computer one |
| 04:37.16 | SarahEmm | either an x100p, or a TDM400p with one FXO module |
| 04:37.19 | santiago | MikeJ[Laptop], thanks anyway |
| 04:37.25 | cyberbuddy | so i only need one card? |
| 04:37.42 | SarahEmm | cyberbuddy: err, you only want one phoneline coming in and no other POTS stuff, right? |
| 04:37.54 | santiago | JunK-Y, do you know if there is a cvs viewer or something similar for asterisk? |
| 04:37.56 | cyberbuddy | whats POTS? |
| 04:38.09 | SarahEmm | Plain Old Telephone Service |
| 04:38.13 | SarahEmm | plain analog phone stuff |
| 04:38.16 | cyberbuddy | ohh |
| 04:38.26 | cyberbuddy | i dont get your question.. |
| 04:38.43 | SarahEmm | err... |
| 04:38.51 | cyberbuddy | i want to have VoIP in my house with asterisk WITHOUT having to pay a VoIP provider somehow |
| 04:38.54 | SarahEmm | why would you need more than one card? |
| 04:38.55 | SarahEmm | okay |
| 04:39.01 | SarahEmm | then you plug your PSTN line into your asterisk machine, and that's it |
| 04:39.03 | SarahEmm | one card |
| 04:39.10 | SarahEmm | lots of people do this |
| 04:39.20 | MikeJ[Laptop] | everyones doing it.... |
| 04:39.21 | cyberbuddy | through a regular modem? |
| 04:39.32 | MikeJ[Laptop] | cyberbuddy, start here: |
| 04:39.33 | MikeJ[Laptop] | ~docs |
| 04:39.33 | 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 |
| 04:39.38 | MikeJ[Laptop] | read all of that. |
| 04:39.43 | cyberbuddy | lol k |
| 04:39.44 | MikeJ[Laptop] | you will be MUCH more informed. |
| 04:39.45 | SarahEmm | cyberbuddy: no..... |
| 04:39.47 | MikeJ[Laptop] | no really |
| 04:40.01 | SarahEmm | cyberbuddy: i gave you two models of cards that will do what you need. but RTFM/RTFW :) |
| 04:40.18 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 04:40.27 | cyberbuddy | and one more question. will i able to spoof my caller id? |
| 04:40.32 | lehel | hi |
| 04:40.35 | shido | jeez |
| 04:40.45 | shido | spoof your callerid if you want |
| 04:40.50 | vuvie | Sarah: symlink libidn.xxxxx to libidn.so |
| 04:40.58 | SarahEmm | shido: over a POTS connection? i don't think so |
| 04:41.02 | shido | but your provider will find you |
| 04:41.02 | vuvie | it make install pretty! :) |
| 04:41.06 | SarahEmm | cyberbuddy: not afaik |
| 04:41.19 | SarahEmm | vuvie: err, that's what i said isn't it? |
| 04:41.30 | vuvie | yep you did! |
| 04:41.35 | SarahEmm | oh, you were just saying it worked? |
| 04:41.35 | cyberbuddy | will i be able to spoof on the PRI line i have access to? |
| 04:41.37 | MikeJ[Laptop] | cyberbuddy, no |
| 04:41.39 | shido | SarahEmm, POTS? gn000 PRI , maybe, voip provider, maybe |
| 04:41.50 | SarahEmm | cyberbuddy: no.. you're limited to what you can do over the phone line, which is no spoofing of any kind |
| 04:41.52 | MikeJ[Laptop] | w/ pri, yeah, probably |
| 04:41.55 | vuvie | yes |
| 04:41.58 | MikeJ[Laptop] | no |
| 04:42.00 | MikeJ[Laptop] | maybe |
| 04:42.01 | shido | if your provider elts you |
| 04:42.02 | MikeJ[Laptop] | depends |
| 04:42.04 | shido | you can |
| 04:42.05 | SarahEmm | you can only do what the line is capable of, cyberbuddy. that's basically dialing out and receiving calls with nothing special |
| 04:42.08 | MikeJ[Laptop] | possibly |
| 04:42.14 | cyberbuddy | ohh |
| 04:42.14 | lehel | what could be wrong with my mpg123?.. when chan_capi is loading asterisk doesn't starts |
| 04:42.18 | MikeJ[Laptop] | not on pot |
| 04:42.22 | SarahEmm | vuvie: oh! woo :o) |
| 04:42.22 | MikeJ[Laptop] | say no to drugs |
| 04:42.28 | shido | a lot of things you cant do on pot |
| 04:42.35 | cyberbuddy | but then how can companies like NuFone, camophone, and star38 do it? |
| 04:42.39 | MikeJ[Laptop] | make babies? |
| 04:42.47 | SarahEmm | lol shido |
| 04:42.55 | SarahEmm | cyberbuddy: that's not on pots. |
| 04:42.58 | shido | like stop talking when people question you about a pimple on on your geni... err jello |
| 04:42.59 | SarahEmm | cyberbuddy: they have PRIs coming in. |
| 04:43.10 | cyberbuddy | ok, i am saying i have access to a PRI |
| 04:43.16 | cyberbuddy | but its at an office |
| 04:43.18 | SarahEmm | but not at your home. |
| 04:43.19 | SarahEmm | oh.... |
| 04:43.35 | cyberbuddy | i will be able to put my box at the office if i want |
| 04:43.37 | SarahEmm | well, you could install * at the office, plug the PRI into your Asterisk machine, and use IAX to link it to your home |
| 04:43.43 | SarahEmm | and the PRI is all yours to use? |
| 04:43.48 | cyberbuddy | nope |
| 04:43.51 | SarahEmm | err |
| 04:43.52 | lehel | anybody ? .. Ouch ... error while writing audio data: : Broken pipe |
| 04:43.52 | cyberbuddy | it belongs to a company |
| 04:44.03 | SarahEmm | okay, but what part can you use? the whole line? one channel? |
| 04:44.13 | cyberbuddy | one channel |
| 04:44.26 | cyberbuddy | actually i will have access to three channels |
| 04:44.28 | SarahEmm | oh... |
| 04:44.40 | SarahEmm | i have no idea how to set that up then :) normally you'd just plug the PRI into the * box.... |
| 04:44.50 | cyberbuddy | oh, its ok |
| 04:44.52 | lehel | <PROTECTED> |
| 04:44.58 | SarahEmm | but not if you only have access to a few channels. with a data T1 i'd use an add/drop CSU/DSU, but i dunno how to do it with that |
| 04:45.15 | cyberbuddy | so if i want to spoof from my house, i need to get a VoIP provider or something like NuFone |
| 04:45.40 | cluecon | cyberbuddy: keep in mind that caller id spoofing may be illegal. |
| 04:46.10 | cyberbuddy | yes i know that, i will not be using it for malicious things |
| 04:46.25 | SarahEmm | cyberbuddy: not if you're connecting to the PRI. |
| 04:46.30 | SarahEmm | that was only if you're just using your home PSTN line |
| 04:46.39 | cyberbuddy | oh ok |
| 04:46.45 | cluecon | cyberbuddy: malicious or not, it may still be illegal. |
| 04:46.49 | cyberbuddy | i need to read up on this stuff.. |
| 04:46.53 | SarahEmm | why are you spoofing, if i may ask? |
| 04:47.12 | cyberbuddy | prank calls and not wanting to reveal my number |
| 04:47.17 | MikeJ[Laptop] | call his ex-girlfriend |
| 04:47.23 | cyberbuddy | and not wanting to block my number completely |
| 04:47.27 | MikeJ[Laptop] | cuz she won't answer his call |
| 04:47.29 | JunK-Y | MikeJ[Laptop]: no, call UR gf :P |
| 04:47.44 | MikeJ[Laptop] | shhh... don't tell my wife |
| 04:47.48 | *** join/#asterisk outtolunc (~me@adsl-69-110-13-227.dsl.pltn13.pacbell.net) |
| 04:47.55 | cyberbuddy | i also read somewhere that you can unmask blocked caller id's on VoIP |
| 04:48.23 | *** join/#asterisk niZon (~ilt@24.78.225.185) |
| 04:49.03 | MikeJ[Laptop] | cyberbuddy, voip has nothing to do with that.. and it is not really unmasking |
| 04:49.22 | SarahEmm | cyberbuddy: err, not wanting to reveal your number? that |
| 04:49.28 | SarahEmm | 's easy to do on any kind of line usually. |
| 04:49.34 | MikeJ[Laptop] | if you have a pri that delivers ani, you get some numbers through ani that are blocked via caller id.. |
| 04:50.02 | cyberbuddy | yes i know |
| 04:50.59 | cyberbuddy | but i read somewhere that when someone blocks there caller, it is still sent to the call receiver but a privacy flag is set to not reveal it on the receiver's equipment. maybe if somehow asterisk can ignore that privacy flag and show the number. |
| 04:51.11 | MikeJ[Laptop] | well you do now, cuz I just told you.. but that is not what you said |
| 04:51.29 | MikeJ[Laptop] | sigh... |
| 04:51.44 | cyberbuddy | i hate being a newbie |
| 04:51.50 | cyberbuddy | my lazy ass has to RTFM |
| 04:52.05 | MikeJ[Laptop] | yes it does |
| 04:52.05 | cyberbuddy | or i can just ask my dad, he went on a 5-day VoIP course |
| 04:52.20 | MikeJ[Laptop] | cyberbuddy, how old are you? |
| 04:52.24 | cyberbuddy | 14 |
| 04:52.29 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 04:52.34 | cluecon | cyberbuddy: it's RTFW not RTFM |
| 04:52.36 | MikeJ[Laptop] | what timezone? |
| 04:52.38 | Juggie | do we have a minimum age to use * |
| 04:52.40 | cyberbuddy | pacific |
| 04:52.41 | Juggie | i think we need one |
| 04:53.01 | JunK-Y | Juggie: how old? 15 :P |
| 04:53.02 | MikeJ[Laptop] | it's past your irc curfew ;) |
| 04:53.16 | cyberbuddy | lol nah |
| 04:53.20 | cyberbuddy | 9:53 here |
| 04:53.23 | cyberbuddy | i sleep around 2am |
| 04:53.30 | Juggie | ummm |
| 04:53.32 | Juggie | 19 |
| 04:53.33 | MikeJ[Laptop] | 7 min then |
| 04:53.34 | JunK-Y | cyberbuddy: WHAT? where's ur parents? |
| 04:53.41 | cyberbuddy | huh? |
| 04:53.47 | cyberbuddy | they're home.. |
| 04:53.59 | JunK-Y | at 15, i was getting in bed at 8:30 |
| 04:54.06 | SarahEmm | cyberbuddy: you're 14 and have a PRI to use at work? |
| 04:54.12 | MikeJ[Laptop] | do they know you are on the computer with a bunch of perverts |
| 04:54.12 | cyberbuddy | yes |
| 04:54.16 | MikeJ[Laptop] | ok then |
| 04:54.42 | cyberbuddy | i am 21 at heart |
| 04:54.43 | cyberbuddy | =] |
| 04:55.13 | *** join/#asterisk cryptnix (~NNSCRIPT@66.103.242.66) |
| 04:56.10 | cyberbuddy | will i be able to do anything if i connect my PSTN line to my asterisk box? |
| 04:56.23 | SarahEmm | cyberbuddy: dial out and receive calls, yes. |
| 04:56.33 | JunK-Y | outtolunc: my bar's open, whatcha want man? blue? |
| 04:56.35 | cyberbuddy | can i use the PBX? |
| 04:56.35 | JunK-Y | bud? |
| 04:56.42 | cyberbuddy | like have a caller menu and stuff |
| 04:56.47 | SarahEmm | cyberbuddy: yes. |
| 04:56.53 | cyberbuddy | crazy |
| 04:56.54 | SarahEmm | cyberbuddy: as we all keep on saying... |
| 04:56.56 | SarahEmm | RTFM! |
| 04:57.06 | cyberbuddy | which one? |
| 04:57.10 | SarahEmm | ~docs |
| 04:57.10 | 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 |
| 04:57.12 | cyberbuddy | the bot sent me like 4 |
| 04:57.12 | SarahEmm | all of those. |
| 04:57.12 | JunK-Y | outtolunc: im drinking a boréale. |
| 04:57.16 | SarahEmm | read them all, cyberbuddy |
| 04:57.19 | SarahEmm | asterisk is not simple. :) |
| 04:57.23 | outtolunc | all of the docs (source code included) |
| 04:57.31 | cyberbuddy | not only asterisk VoIP in general |
| 04:57.36 | SarahEmm | cyberbuddy: you'll need either an x100p or TDM400p with an FXO module. you don't want to use a modem. |
| 04:57.44 | cyberbuddy | why not use a modem? |
| 04:57.51 | cyberbuddy | what problems would a modem cause? |
| 04:57.55 | gambolputty | I have * and odbc connected for realtime but phones don't register. |
| 04:58.17 | MikeJ[Laptop] | cyberbuddy.. dude.. chill... read |
| 04:58.32 | cyberbuddy | ahh i hate acting 14 |
| 04:58.35 | Faithful | USING ISDN is there a way to get asterisk to redirect a particular incoming number to and external number? |
| 04:59.02 | MikeJ[Laptop] | Faithful, HUH? |
| 04:59.14 | SarahEmm | cyberbuddy: modems aren't designed for this use, most won't work with * at all, the ones that will you likely still don't want to use. |
| 04:59.16 | cryptnix | I am having some random issues with the voices not being heard on the Asterisk network side not being able to be heard when I place calls after setting up a pfsense/m0n0wall type setup. Not sure if people are aware that those based off of BSD |
| 04:59.18 | SarahEmm | read the docs. |
| 04:59.31 | Faithful | MikeJ[Laptop]: to an external number |
| 04:59.41 | cyberbuddy | what do you mean when you say * |
| 04:59.45 | MikeJ[Laptop] | cryptnix, head or stable, what version\date |
| 04:59.49 | SarahEmm | cyberbuddy: * is Asterisk. |
| 04:59.54 | SarahEmm | see? shift-8 makes an asterisk. :) |
| 04:59.56 | JunK-Y | ~* |
| 04:59.58 | cyberbuddy | oh haha obviously sorry |
| 05:00.03 | JunK-Y | jbot, * is asterisk |
| 05:00.03 | jbot | okay, JunK-Y |
| 05:00.04 | SarahEmm | now.. go read :P |
| 05:00.09 | cyberbuddy | im about to |
| 05:00.10 | JunK-Y | ~* |
| 05:00.10 | jbot | methinks * is asterisk |
| 05:00.18 | cyberbuddy | thank you everyone for your help |
| 05:00.21 | SarahEmm | then if you still have questions, ask :) |
| 05:00.22 | SarahEmm | np. |
| 05:00.28 | cyberbuddy | thanks for putting up with my noob self. |
| 05:00.40 | cyberbuddy | k, if i have any more questions, ill be sure to ask |
| 05:01.25 | Faithful | external # ---> * ---> external # bearing in mind that ISDN is terminated at the NT device on the clients premise not at the exchange like PSTN is |
| 05:02.08 | MikeJ[Laptop] | your trying to do it without using to channels on the ISDN? |
| 05:02.23 | outtolunc | so faithful, where does the internal part come into play <G> |
| 05:02.54 | lehel | ppl: why my asterisk stops if i run as U&G asterisk and load the chan_capi? |
| 05:02.54 | MikeJ[Laptop] | without using 2 channels that is |
| 05:03.48 | SarahEmm | lehel: stops? explain more? |
| 05:04.13 | cryptnix | What ports does Asterisk use via SIP to broadvoice for example? |
| 05:04.24 | lehel | stops like this: |
| 05:04.26 | lehel | <PROTECTED> |
| 05:04.26 | lehel | pbx02:~# Warning, flexibel rate not heavily tested! |
| 05:04.26 | lehel | Ouch ... error while writing audio data: : Broken pipe |
| 05:04.32 | cryptnix | 5060? ... ok, so what other ports actively does it use |
| 05:04.40 | SarahEmm | ahh... i dunno, sowwy =-/ |
| 05:04.44 | cluecon | cryptnix: 5060 and RTP 10000 to 20000 |
| 05:05.10 | cryptnix | ok, cool. thank you. |
| 05:05.31 | lehel | SarahEmm: i think it is mpg problem..but howto resolv.. have you heard about madplay? |
| 05:05.35 | outtolunc | lehel, are you starting asterisk in verbose mode? asterisk -vvvvvgc and if so, what does the debug log say? |
| 05:06.12 | lehel | outtolunc: if i start asterisk without -U asterisk -G asterisk< no problem! |
| 05:06.34 | lehel | and with the CAPI module |
| 05:06.52 | outtolunc | so your answer is you aren't starting in verbose mode.. that might be a hint |
| 05:07.05 | santiago | long life to colombian coffee |
| 05:07.33 | lehel | outtolunc: this is my command: asterisk -U asterisk -G asterisk -vvvvvvvgc |
| 05:07.53 | outtolunc | and the debug logs say? |
| 05:07.58 | outtolunc | er says |
| 05:08.32 | lehel | it is pasted!.. upthere ~15 lines |
| 05:08.46 | lehel | broken pipe.. |
| 05:08.51 | outtolunc | that was console output |
| 05:10.29 | outtolunc | if you look in /var/log/asterisk/debug you 'should' see something about you not having any zaptel devices <G> |
| 05:10.45 | outtolunc | or some misconfiguration issue |
| 05:11.54 | truz24 | Does zaptel not work with 2.6.* ? |
| 05:12.07 | SarahEmm | zaptel works fine in 2.6 |
| 05:12.09 | SarahEmm | what's the issue? |
| 05:12.18 | outtolunc | yes zaptel devices work with 2.6 kernel's |
| 05:12.21 | truz24 | tryin to compile the zaptel module in ubuntu |
| 05:12.28 | SarahEmm | what error? |
| 05:12.31 | *** part/#asterisk Delemas (~rhardy@CPE000fea6e2535-CM00111ae61f20.cpe.net.cable.rogers.com) |
| 05:13.22 | truz24 | the first one is: /usr/include/asm/mpspec.h:6:25: mach_mpspec.h: No such file or directory |
| 05:13.41 | lehel | outtolunc: i have no "debug" log, but "full".. i have zaptel ! |
| 05:13.49 | santiago | truz24, how are you compiling it? |
| 05:13.56 | lehel | there is nothing in the logs.. stange |
| 05:13.58 | truz24 | make wcfxo |
| 05:14.06 | truz24 | tryin to compile the module for a x100p card |
| 05:14.22 | SarahEmm | plain 'make' work? |
| 05:14.25 | santiago | truz24, from the zaptel-source package? |
| 05:14.26 | truz24 | no |
| 05:14.29 | truz24 | it compiled some stuff |
| 05:14.31 | truz24 | but failed |
| 05:14.38 | truz24 | yeah, i got it from ubuntu repos |
| 05:14.54 | SarahEmm | you've built a kernel from the sources on that box? |
| 05:15.02 | truz24 | no |
| 05:15.03 | santiago | truz24, do you have linux-headers- or linux-source? |
| 05:15.04 | SarahEmm | err |
| 05:15.12 | SarahEmm | you may not have proper symlinks and stuff then |
| 05:15.26 | SarahEmm | err, do you even have kernel source installed, yeah... |
| 05:15.40 | truz24 | Yeah, i do |
| 05:15.47 | truz24 | but you might be right about the symlinks |
| 05:15.53 | truz24 | santiago, both |
| 05:16.02 | *** join/#asterisk cowboykitty (~Cowboy@adsl-66-142-172-180.dsl.rcsntx.swbell.net) |
| 05:16.05 | truz24 | santiago, actually, it might just be linux-headers |
| 05:16.12 | truz24 | cuz i had to compile a vmware module |
| 05:16.17 | truz24 | which worked fine |
| 05:16.54 | santiago | truz24, you could use module-assistant to build the modules |
| 05:17.03 | outtolunc | lehel: then add a 'd' to the -vvvvvvgcd (or edit your logger.conf and enable debug' |
| 05:17.04 | outtolunc | ) |
| 05:17.40 | outtolunc | don't do both tho, not sure if anyone ever fixed that bug <G> |
| 05:18.09 | truz24 | ok, i have module-assistant installed now... what commands would I issue ? |
| 05:18.13 | santiago | truz24, cd /usr/src ; module-assistant auto-install zaptel |
| 05:19.02 | truz24 | says bulid starting, but is sitting at 0% |
| 05:19.04 | truz24 | build* |
| 05:19.40 | lehel | outtolunc: DEBUG[7237]: chan_zap.c:6310 do_monitor: Monitor doohicky got event Event 160 on channel 4 << wow;) never see such thing |
| 05:20.10 | *** join/#asterisk SplasPood (jwb@schizophrenia.paravolve.net) |
| 05:20.13 | santiago | truz24, are you sure that /usr/src/linux heads to the linux headers? or source? |
| 05:20.15 | truz24 | ok, it completed |
| 05:20.25 | truz24 | modprobe wcfxo doesn't show it tho |
| 05:20.27 | santiago | truz24, well |
| 05:20.35 | santiago | try with insmod |
| 05:20.50 | truz24 | i did, "can't read wcfxo" |
| 05:21.09 | lehel | outtolunc: did you? |
| 05:21.16 | outtolunc | lehel: cute message, this is on 'asterisk startup' or did you attempt to monitor a zap channel? |
| 05:21.24 | santiago | truz24, insmod path_to_wcfxo.ko |
| 05:21.39 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 05:21.45 | santiago | truz24, try with zaptel, first |
| 05:21.54 | lehel | outtolunc: i didn't try so far monitoring |
| 05:22.10 | truz24 | Try with zaptel ? |
| 05:22.29 | outtolunc | lehel: does that mean that this 'was on asterisk startup'? |
| 05:22.33 | santiago | insmod path_to_zaptel.ko |
| 05:22.42 | lehel | outtolunc: mention: this is happening just in case if i start asterisk as asterisk User & Group |
| 05:22.45 | truz24 | k |
| 05:23.02 | outtolunc | then it's probably a perms issue |
| 05:23.18 | santiago | truz24, see where are your modules with dpkg --contents /usr/src/zaptel-modules*.deb |
| 05:23.27 | lehel | outtolunc: where? |
| 05:23.35 | outtolunc | the wiki has a step by step 'running asterisk as NON-root' |
| 05:23.45 | truz24 | santiago, yeah, thats what i was looking for, i found them, they are in /lib/modules/2.6.10-5.../zaptel |
| 05:24.09 | truz24 | lol figures: insmod: error inserting 'wcfxo.ko': -1 Unknown symbol in module |
| 05:24.32 | outtolunc | http://www.voip-info.org/tiki-index.php?page=Asterisk%20non-root |
| 05:24.50 | santiago | truz24, :S |
| 05:25.05 | santiago | i've not tested them in ubuntu, only in debian |
| 05:25.06 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 05:25.11 | truz24 | santiago, it let me ins the zaptel tho |
| 05:25.14 | truz24 | zaptel.ko |
| 05:25.24 | truz24 | Will taht let the x100p work ? |
| 05:25.33 | redder86 | anyone from NuFone around tonight? |
| 05:25.51 | santiago | truz24, it works for me, on debian |
| 05:25.59 | truz24 | zaptel with a x100p ? |
| 05:26.11 | santiago | truz24, yes |
| 05:26.30 | santiago | truz24, what's the complete error? |
| 05:26.37 | truz24 | insmod zaptel.ko worked |
| 05:26.41 | truz24 | no error |
| 05:26.43 | lehel | outtolunc: ok... i'll try that; thanks |
| 05:26.50 | outtolunc | np |
| 05:27.08 | lehel | outtolunc: but! if i recompile my asterisk could happen something wrong?? |
| 05:27.23 | lehel | do i need to recomple? |
| 05:27.54 | outtolunc | after you step by step through that doc, you 'should' recomile as root |
| 05:28.00 | outtolunc | er recompile |
| 05:28.04 | santiago | truz24, and with wcfxo? |
| 05:28.18 | truz24 | Do you have the zaptel and wcfxo module loaded ? |
| 05:28.25 | outtolunc | then the user:groups and file placements should be correct |
| 05:28.38 | truz24 | mmm, insmod wcfxo worked too |
| 05:28.42 | santiago | truz24, not in this pc |
| 05:28.49 | redder86 | JerJer: around? |
| 05:29.09 | santiago | truz24, i though that it doesn't |
| 05:29.17 | truz24 | it didn't before, said invalid symbol |
| 05:29.19 | truz24 | but now it worked |
| 05:29.30 | truz24 | bash_history shows same command |
| 05:29.45 | lehel | outtolunc: but i'm afraid if i recompile asterisk, my AMP "will get crazy" |
| 05:29.51 | truz24 | lsmod shows it loaded |
| 05:29.54 | truz24 | it and zaptel |
| 05:29.57 | santiago | :) |
| 05:30.03 | truz24 | Thanks a lot santiago |
| 05:30.11 | outtolunc | it probably will, but if you can't run asterisk, what good is AMP <G> |
| 05:30.14 | santiago | truz24, np |
| 05:30.20 | truz24 | Do you know of a quick way to check that the x100p works |
| 05:31.01 | JunK-Y | truz24: u have cvs-head? zap show status |
| 05:31.18 | lehel | outtolunc: i can run asterisk, but;( without the CAPI modules.. anyway let's see what happens |
| 05:31.25 | truz24 | JunK-Y, no, i have the version on the ubuntu repository |
| 05:31.34 | truz24 | zap show status command not found |
| 05:32.01 | lehel | truz24: ztcfg -vv |
| 05:32.23 | santiago | truz24, you don't have asterisk with the chap_zap.so module loaded |
| 05:32.38 | lehel | 'sorry then;) |
| 05:32.41 | JunK-Y | truz24: its an * command... |
| 05:32.55 | truz24 | santiago, how do you know that ? |
| 05:34.12 | truz24 | ztcfg -vv : 0 channels configured |
| 05:35.04 | truz24 | i ran zttest, it said opening pseudo zap interface... can i run a test with the x100p ? |
| 05:35.38 | lehel | truz24: you have to edit ur /etc/zaptel.conf < to have the channels configured |
| 05:36.10 | truz24 | lehel, ok, thanks. |
| 05:36.53 | SarahEmm | nini all |
| 05:40.16 | cyberbuddy | err, i am frustrated with all this VoIP docs |
| 05:40.34 | cyberbuddy | i dont get why you can spoof on a PRI and not a traditional PSTN POTS line |
| 05:40.51 | JunK-Y | cyberbuddy: continue to read :) |
| 05:40.52 | drumkilla | because you don't send callerid on an analog line |
| 05:41.13 | cyberbuddy | PRI = digital line? |
| 05:41.29 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 05:41.35 | drumkilla | yes |
| 05:42.05 | cyberbuddy | ohh so on an analog line, the POTS provider sends my cid |
| 05:42.19 | drumkilla | indeed :) |
| 05:42.31 | cyberbuddy | what if i get something like vonage |
| 05:43.14 | drumkilla | VoIP providers may or may not let you set your own callerid. |
| 05:43.34 | cyberbuddy | oh, im pretty sure vonage doesnt |
| 05:44.24 | cyberbuddy | so companies like NuFone just pay for a PRI line and allow people to pay and use it? |
| 05:46.25 | *** join/#asterisk _omer (dfsdf@203.215.180.254) |
| 05:46.56 | redder86 | cyberbuddy: "just" doesn't quite do it justice |
| 05:47.38 | cyberbuddy | lol true |
| 05:48.12 | _omer | http://pastebin.ca/15734 <----- anybody....plz :) |
| 05:52.09 | JunK-Y | _omer: sip show peers output? |
| 05:52.30 | _omer | ok |
| 05:53.23 | _omer | to avoid the flooding...I have pasted it in your pvt window. |
| 06:00.13 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
| 06:05.47 | infinity1 | dman. there is a lot to leanr for this. |
| 06:07.33 | *** join/#asterisk X-Rob (~rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 06:10.05 | santiago | time for the bed... |
| 06:10.18 | X-Rob | wuss |
| 06:10.27 | X-Rob | it's not even beer-o-clock yet |
| 06:10.28 | X-Rob | ! |
| 06:10.28 | santiago | good night |
| 06:10.54 | santiago | X-Rob, you'r right :P 1:11 am |
| 06:11.06 | X-Rob | heh |
| 06:11.42 | *** part/#asterisk santiago (~santiago@63.245.86.198) |
| 06:14.07 | infinity1 | <PROTECTED> |
| 06:15.34 | shido | yes |
| 06:15.38 | shido | www.nufone.net |
| 06:15.43 | shido | ok all done |
| 06:15.58 | *** join/#asterisk Tangent (~Arc_Tange@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk) |
| 06:18.59 | infinity1 | quality service. |
| 06:19.00 | infinity1 | Please contact the server administrator, jj@nufone.net and inform them of the time the error occurred, and anything you might have done that may have caused the error. |
| 06:19.47 | tzafrir_laptop | anybody using areskicc? |
| 06:19.48 | *** join/#asterisk Broesmeli (~broesme@195.65.2.68) |
| 06:20.21 | tzafrir_laptop | The PHP code seems to assume register_globals (yuck) |
| 06:20.48 | *** join/#asterisk |Fender| (haha@fctnnbsch17-156034025180.nb.aliant.net) |
| 06:20.48 | tzafrir_laptop | Though I could not find any traces of it in the installation instructions |
| 06:21.49 | infinity1 | i'm looking for an iax privoder with local 408 area code #'s |
| 06:24.17 | Qorky | X-Rob you around ? |
| 06:24.29 | X-Rob | Yeh? |
| 06:24.39 | Qorky | i need some more help please :) got the cards working. need help with my extensions.conf file now. |
| 06:25.09 | Qorky | i can call into my server and it works now. so thats sweet. but when I call out i cant get any audio. its does actually call out though. |
| 06:25.14 | |Fender| | Hi, have people already try AreskiCC ? |
| 06:25.28 | X-Rob | call out through what - PSTN? and what cards do you have? |
| 06:25.45 | Qorky | isdn. fritz. |
| 06:26.44 | X-Rob | Ah... Did I mention I don't know anything about CAPI? |
| 06:26.59 | Qorky | hehe no. |
| 06:27.07 | Qorky | i thought you knew that stuff. no worries :) |
| 06:27.15 | X-Rob | *however* |
| 06:27.22 | X-Rob | if it's dialling, that means that the D channel is working |
| 06:27.51 | X-Rob | what, exactly, is a dial string you're using? |
| 06:30.55 | Qorky | erm. let me see |
| 06:31.04 | Qorky | nah im using the B channels. |
| 06:31.12 | Qorky | i just wonna look at voice at the moment. |
| 06:31.23 | X-Rob | D channel is signalling, eg 'dial this number' |
| 06:31.25 | infinity1 | what serivces are you guys using, i'm having a hard time finding a voip to pstn service providers. |
| 06:31.27 | Qorky | AH OK |
| 06:31.29 | lehel | |Fender|: i'm trying now.. you? |
| 06:31.33 | Qorky | whoops caps |
| 06:31.47 | Qorky | exten => _X.,1,dial(CAPI/93330000:b${EXTEN},60,T) |
| 06:31.55 | Qorky | hmm theres my number world :) |
| 06:31.56 | |Fender| | lehel u install it and work? |
| 06:31.58 | Robot_ | if i equip two boxes with TE110P, and connect them together with some crossover cable will it work ? |
| 06:32.34 | X-Rob | IT's _actually_ +61-8-93330000 for anyone who really wants to waste money 8) |
| 06:32.48 | Qorky | yeah thanks :) |
| 06:32.48 | X-Rob | Mine's on my aussievoip, go wild 8) |
| 06:33.21 | X-Rob | I'm surprised you got a 0000 on an onramp 2 |
| 06:33.42 | Qorky | well my boss is a fool. we got 3 x onramp 2's |
| 06:33.51 | Qorky | and 100 #'s on that. |
| 06:34.10 | X-Rob | Geez. |
| 06:34.22 | X-Rob | Port 'em to an onramp 10 |
| 06:34.31 | X-Rob | will be cheaper, and you won't need three Fritz cards |
| 06:35.55 | Qorky | i know.. |
| 06:36.00 | Qorky | i know this now.. |
| 06:36.05 | _omer | http://pastebin.ca/15734 <----anyone ......... plz ? |
| 06:36.09 | Qorky | i came on board after that decision. |
| 06:36.48 | Pete_Largo | what's an onramp (2, 3, 10)? |
| 06:36.52 | X-Rob | omer, why the '222' on the end of the register string? |
| 06:37.18 | X-Rob | PEte, Telstra, our incumbent telco, only provide ISDN in 2 as a BRI, or as a PRI with 10, 20 or 30 lines activated |
| 06:37.23 | _omer | to pass the incoming calls at 222 extension... |
| 06:37.27 | X-Rob | They're known as OnRamps |
| 06:37.32 | Pete_Largo | ah I see |
| 06:37.52 | Pete_Largo | thanks |
| 06:38.13 | X-Rob | omer, who is that 203.128.7.14? |
| 06:38.17 | X-Rob | broadvoice or something? |
| 06:38.37 | _omer | a Local/domestic Service provider :) |
| 06:39.14 | X-Rob | chat.brain.net.pk |
| 06:39.17 | _omer | yup |
| 06:39.22 | X-Rob | OK |
| 06:39.42 | X-Rob | I'm not sure what that error is, but check your password is right. do a 'sip show registry' |
| 06:39.46 | X-Rob | see if it says registered |
| 06:39.46 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 06:39.54 | _omer | yes....it is registered... |
| 06:40.46 | X-Rob | what's your 'type=' for that in sip.conf? |
| 06:40.49 | _omer | when I just change register line to FWD ...it works.....but doesnt when I change it to chat.brain.net.pk |
| 06:40.57 | Faithful | I am using BRISTUFF... is there a way to get asterisk to redirect a CID to another number without answering the call? I imagine it should be possible seeing with ISDN the NT actually handles the termination of the call. |
| 06:41.26 | _omer | peer |
| 06:41.43 | _omer | type=peer |
| 06:41.59 | X-Rob | I'd be talkign to them and asking them what you should have |
| 06:42.05 | X-Rob | there's obiviously some misconfiguration there |
| 06:42.33 | X-Rob | you've got a 'username' and 'secret' in the [brainnet] or whatever you've called 'em? |
| 06:43.12 | _omer | yes...that is why it is registered when I do sip show registry |
| 06:43.19 | X-Rob | Faithful - Not easily, no. It was discussed on the users list recently, and there's a three letter acronym to describe it, but I'm not sure if it's actually implemented |
| 06:43.28 | X-Rob | omer, no, that's the regiser => line |
| 06:43.33 | X-Rob | register => even |
| 06:43.47 | X-Rob | the 'register =>' is for recieving calls |
| 06:43.58 | _omer | yes....correct username and password... |
| 06:44.22 | _omer | register => should pass the calls at 222 extension..... |
| 06:46.12 | X-Rob | omer, you're trying to dial out, right? |
| 06:46.16 | X-Rob | or are you trying to receive calls? |
| 06:46.27 | _omer | receive calls.. |
| 06:46.29 | X-Rob | ok |
| 06:46.32 | X-Rob | You didn't say that |
| 06:46.36 | X-Rob | you were talking about dialling out |
| 06:46.46 | X-Rob | (or, well, I possibly incorrectly inferred that) |
| 06:46.58 | X-Rob | Try 'insecure=very' |
| 06:47.03 | _omer | ok If I didnt then sorry...but I said..incoming calls :) |
| 06:47.19 | _omer | in peers ? |
| 06:47.25 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 06:47.34 | X-Rob | in that peer, yes |
| 06:47.55 | _omer | I am not using any peer....to call at my asterisk... |
| 06:48.15 | _omer | I use another line from the same SIP provider that I use in X-Pro... |
| 06:48.44 | X-Rob | Sorry, brainfart, insecure=very goes under [general] in sip.conf |
| 06:48.46 | _omer | I call from another line from the same sip provider...from X-Pro ...and the second line that I am trying to use in Asterisk. |
| 06:48.54 | _omer | alright :) |
| 06:48.55 | X-Rob | um no |
| 06:48.56 | X-Rob | no it doesn't |
| 06:49.04 | X-Rob | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 06:49.19 | X-Rob | try 'autocreatepeer=yes' in [general] |
| 06:49.26 | _omer | ok. |
| 06:49.51 | X-Rob | Note: very very insecure. |
| 06:50.58 | _omer | np...atleast it should work . |
| 06:51.06 | _omer | ok done....sip reload ..done |
| 06:51.08 | X-Rob | you'll want to put the peer in there. |
| 06:51.09 | _omer | going to check.. |
| 06:51.16 | X-Rob | and then put insecure=very in that |
| 06:51.45 | _omer | peer for what? |
| 06:51.55 | X-Rob | the peer that you're receiving calls from |
| 06:52.35 | _omer | that is an outsider ...... for example: calling 1 FWD (Asterisk) number from another FWD Number (X-Pro) |
| 06:53.17 | timecop | uh |
| 06:53.20 | timecop | does regexten actually fuckign work |
| 06:53.29 | timecop | because it doesnt for me |
| 06:53.31 | X-Rob | Never tried it |
| 06:53.37 | lehel | ppl: ERROR[8232]: cdr_addon_mysql.c:399 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. |
| 06:53.47 | _omer | timecop ???? |
| 06:54.03 | _omer | regexten => ???? instead of register => ? |
| 06:54.07 | timecop | no |
| 06:54.11 | _omer | then? |
| 06:54.12 | timecop | regexten for a sip peer. |
| 06:55.16 | _omer | I am not using any peer to call at my asterisk box... |
| 06:55.27 | _omer | I use that line in X-Pro.... |
| 06:55.49 | _omer | which is a standalone line......for example: calling 1 FWD (Asterisk) number from another FWD Number (X-Pro) |
| 06:55.51 | _omer | got it ? |
| 06:56.07 | timecop | what the FUCKING SHIT regexten is TOTALLY FUCKING USELESS. |
| 06:56.13 | timecop | im jsut trying ot make a new fuckign extension for each shit that registers |
| 06:56.17 | timecop | so I dotn have to make 234562145 extensions |
| 06:56.29 | timecop | http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+regcontext |
| 06:56.34 | timecop | but the fucking docs are worthless |
| 06:56.39 | timecop | does nayone use this shit |
| 06:56.57 | timecop | OBVIOUSLY NOT |
| 06:57.01 | timecop | FUCKIGN GAY. |
| 06:57.04 | _omer | :) |
| 07:00.28 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 07:01.17 | X-Rob | I have to agree, that's pretty crappy docco. |
| 07:01.23 | X-Rob | it doesn't actually _explain_ anything. |
| 07:02.08 | _omer | Agreed!!! |
| 07:03.03 | X-Rob | *blink* |
| 07:03.25 | X-Rob | omer, create a peer in sip.conf, or, try autocreatepeer=yes. Until then, stop bitching. |
| 07:06.38 | _omer | okei.... |
| 07:20.23 | *** join/#asterisk af_ (~af@ip-130-170.sn2.eutelia.it) |
| 07:21.31 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:24.21 | Qorky | anyone know why I would be able to recieve calls on my fritz isdn card and get audio fine. but when I make calls I hear nothing? please |
| 07:29.00 | X-Rob | http://lists.digium.com/pipermail/asterisk-users/2004-May/046819.html |
| 07:29.08 | X-Rob | Are you running HEAD? |
| 07:29.21 | *** join/#asterisk der[mat] (~mat@gate-nue0.bintec.de) |
| 07:29.25 | der[mat] | hi guys |
| 07:30.55 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr) |
| 07:34.06 | Qorky | im running 1.0.8 |
| 07:34.29 | Qorky | hmm now if answer the call befor calling out. i get outgoing audio. |
| 07:34.30 | Qorky | exten => _X.,1,Answer |
| 07:34.30 | Qorky | exten => _X.,2,Dial,CAPI/93330000:b${EXTEN} |
| 07:34.30 | Qorky | exten => _X.,3,congestion |
| 07:34.30 | Qorky | exten => _X.,4,hangup |
| 07:34.36 | Qorky | but not incomming. |
| 07:35.00 | X-Rob | I'd try HEAD before pulling any more hair out |
| 07:36.17 | Qorky | ok how i grab that ? export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 07:36.21 | Qorky | different directory ? |
| 07:36.49 | Qorky | . /usr/cvshead or something ? |
| 07:37.04 | X-Rob | um. |
| 07:37.08 | X-Rob | mkdir /usr/src/old |
| 07:37.12 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 07:37.18 | X-Rob | move all the asterisk stuff to old |
| 07:39.38 | Qorky | done |
| 07:39.54 | Qorky | should I smoke all the old modules etc too ? |
| 07:40.09 | X-Rob | http://www.aussievoip.com.au/wiki-Asterisk-HEAD |
| 07:40.45 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr) |
| 07:41.04 | Qorky | yar grabbing it now |
| 07:43.59 | *** join/#asterisk znoG (~gs@200.115.216.109) |
| 07:44.46 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
| 07:49.35 | *** join/#asterisk Inv_arp (junya@adsl-3-255-241.mia.bellsouth.net) |
| 07:52.04 | *** join/#asterisk Abbas_ (Abbas@203.81.194.242) |
| 07:53.35 | lehel | WARNING[9688]: cdr.c:434 ast_cdr_free: CDR on channel 'xxxx' not posted < why? |
| 07:57.41 | fish_ | re |
| 07:59.07 | *** join/#asterisk Thus0 (~Thus0@190.111.102-84.rev.gaoland.net) |
| 08:00.20 | Qorky | hmm now chan_capi wont make |
| 08:00.30 | Qorky | with this version |
| 08:00.34 | Qorky | the head version |
| 08:01.46 | lehel | Qorky: you have to patch it |
| 08:02.09 | Qorky | hmm ok. i need the patch then :) |
| 08:03.03 | Qorky | ahh http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 |
| 08:03.05 | Qorky | looks good |
| 08:04.55 | Qorky | yup compiles now |
| 08:05.30 | Qorky | hmm now i get: |
| 08:05.31 | Qorky | [pbx_wilcalu.so]Jun 25 18:03:07 WARNING[30440]: loader.c:313 __load_resource: /usr/lib/asterisk/modules/pbx_ wilcalu.so: undefined symbol: ast_pthread_create |
| 08:05.37 | Qorky | whats wilcalu ? |
| 08:06.16 | Qorky | i'll noload it. |
| 08:07.56 | Qorky | hmm when i call out of my fritz card i can never hear anything. yet the other end can hear me. |
| 08:08.05 | Qorky | didnt make any difference :( |
| 08:09.56 | lehel | what fritz do you have Qorky? |
| 08:13.16 | Abbas_ | can any body tell me Cisco AS5300 configuration with Quintum tenor |
| 08:16.04 | fish_ | sorry for asking this again, but i don't find anything about this on asterisk.org: does the _cvs_ version of asterisk support somekind of encryption and/or tcp support for SIP? or is there a third-party 'addon' which could make asterisk/sip more secure? i think it should be enough if i could add tcp somehow, than it should be easy to use stunnel with that.. |
| 08:17.28 | Ahrimanes | fish_: no encryption for SIP in cvs, not tcp either as far as i know... the way to go is to use ipsec/vpn, you can tunnel UDP within this as well |
| 08:18.52 | fish_ | yes.. i already thought about that. but in my scenario the SIP users have no permissions to use a ipsec tunnel.. |
| 08:18.56 | Qorky | lehel: fritz!Card PCI |
| 08:19.34 | fish_ | afaik ipsec is only possible if you have root (unix)/admin (windows) privileges.. |
| 08:19.58 | fish_ | somekind of userspace tunnel would work.. |
| 08:20.18 | fish_ | but ssh portforwarding won't work with udp eighter.. |
| 08:21.36 | fish_ | if found something which is called 'utunnel' it seams so be something like a tcp over udp wrapper.. but thats not The Right Thing i think... |
| 08:23.50 | |Fender| | Hi, have people already try AreskiCC ? |
| 08:25.14 | Qorky | its a pasive card lehel |
| 08:27.07 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:28.03 | *** join/#asterisk my007ms (~my007ms@217.139.236.110) |
| 08:28.14 | my007ms | hello all |
| 08:29.51 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
| 08:29.56 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:33.05 | my007ms | i get this error ove and over |
| 08:33.06 | my007ms | <PROTECTED> |
| 08:33.07 | my007ms | Ouch ... error while writing audio data: : Broken pipe |
| 08:33.07 | my007ms | [root@localhost asterisk]# |
| 08:33.18 | my007ms | can some one help pleas |
| 08:33.38 | file[laptop] | your zaptel configuration isn't right |
| 08:33.44 | Zeeek | check zapata.conf and /etc/zaptel.conf? |
| 08:33.53 | Zeeek | Great Minds run on like channels |
| 08:33.58 | my007ms | i do as in doc |
| 08:34.11 | my007ms | i have TDM31B |
| 08:34.15 | Zeeek | maybe doc isn't what hardware you have? |
| 08:34.22 | my007ms | http://www.digium.com/index.php?menu=configuration#TDM31B |
| 08:34.37 | Zeeek | what does this tell you? ztcfg -vvv |
| 08:35.01 | my007ms | how to make sure that Zaptel Configuration |
| 08:35.03 | my007ms | ====================== |
| 08:35.03 | my007ms | Channel map: |
| 08:35.03 | my007ms | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 08:35.03 | my007ms | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 08:35.03 | my007ms | Channel 03: FXO Kewlstart (Default) (Slaves: 03) |
| 08:35.05 | my007ms | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 08:35.06 | my007ms | 4 channels configured. |
| 08:35.08 | tzafrir_laptop | ~pastebin |
| 08:35.08 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 08:35.09 | my007ms | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 08:35.35 | Zeeek | that's good |
| 08:35.37 | my007ms | i don't need to be kick |
| 08:35.46 | Zeeek | find http://pastebin.ca |
| 08:35.55 | Zeeek | and put your configs |
| 08:37.08 | my007ms | how can i make sure that i modprobe the right thing |
| 08:37.52 | tzafrir_laptop | my007ms, if ztcfg worked, then modprobe ran fine, right? |
| 08:38.35 | my007ms | so i don't need to do it myself |
| 08:38.38 | my007ms | ? |
| 08:38.44 | tzafrir_laptop | my007ms, lsmod | grep zaptel# will show a list of loaded zaptel modules |
| 08:39.52 | my007ms | i see |
| 08:39.55 | my007ms | zaptel 210052 2 wct1xxp,wcfxs |
| 08:39.56 | my007ms | crc_ccitt 2113 2 zaptel,hisax |
| 08:40.01 | *** join/#asterisk tessier (~treed@222.253.83.42) |
| 08:40.04 | tessier | Hello all! |
| 08:40.10 | file[laptop] | hi |
| 08:40.10 | Abbas_ | hello all |
| 08:40.12 | tessier | Is there realtime support for musiconhold.conf? |
| 08:40.42 | Abbas_ | file[laptop] do u know configuration of cisco AS5300 with Quintum analog on SIP |
| 08:40.49 | file[laptop] | Abbas_: no |
| 08:40.51 | Abbas_ | i have tried some but no voice |
| 08:40.51 | my007ms | what is the next when i remove all in /etc/asterisk/zaptel.conf it work fine |
| 08:41.09 | *** join/#asterisk Guest^DJ (~me@219.95.221.60) |
| 08:41.20 | Abbas_ | anbody else? |
| 08:41.39 | Guest^DJ | hi all, is there a way to turn on asterisk Silence suppression |
| 08:41.51 | file[laptop] | Asterisk doesn't support silence suppression |
| 08:42.31 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 08:42.40 | Guest^DJ | file[laptop]: so some reason, vendor does not want to turn off silence supression. any way to go around it ? |
| 08:42.48 | file[laptop] | nope |
| 08:42.59 | Guest^DJ | darn |
| 08:43.05 | Guest^DJ | was afraid u say that |
| 08:43.39 | Guest^DJ | how about some equipment to sit in between * and the vendor , any ideas? |
| 08:44.24 | Guest^DJ | turn off on * end, and turn ON on vendors end |
| 08:45.17 | my007ms | can some tell me how to know wich fxsks and what fxsks in /etc/zaptel.conf |
| 08:49.38 | *** join/#asterisk skiold (~userid@84-121-68-212.onocable.ono.com) |
| 08:51.09 | tessier | crap...no realtime for music on hold. |
| 08:51.29 | tessier | We'll have to kludge it by using a script to append to the musiconhold.conf file and reload asterisk. :( |
| 08:51.42 | tzafrir_laptop | anybody using areskicc? if so: could you please contact me privately ? |
| 08:52.48 | tzafrir_laptop | I see there something that looks like an obvious hole, but I'd others to test it on their systems becasue my install may be broken |
| 08:53.30 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:53.31 | RoyK | anyone that knows where i can find ES 300 738? i couldn't see it on ETSIs pages |
| 08:55.25 | tzafrir_laptop | ~genzaptelconf |
| 08:55.25 | jbot | [genzaptelconf] a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 08:56.55 | tzafrir_laptop | my007ms, chances are genzaptelconf will generate a working zaptel.conf and zapata.conf for you. |
| 08:57.06 | tzafrir_laptop | If it doesn't, I'd like to know. |
| 08:58.00 | my007ms | i am sorry can some tell me after i edit /etc/asterisk/zaptel.conf asterisk work |
| 08:58.16 | my007ms | but the zaptel card not light as it was y this |
| 08:58.18 | my007ms | ?? |
| 08:58.34 | |Fender| | Hi, have people already try AreskiCC ? |
| 08:58.35 | tzafrir_laptop | my007ms, certainly not. /etc/zaptel.conf or /etc/asterisk/zapata.conf |
| 08:59.09 | *** join/#asterisk daXas (~on@85.96.199.40) |
| 08:59.33 | my007ms | so what |
| 08:59.41 | my007ms | where is the wrong thing |
| 09:00.16 | daXas | hi all, what is wrong with libpri on cvs ? anyone successed compile ? |
| 09:00.48 | daXas | i got this msg: copy_string.c:38: warning: implicit declaration of function `__builtin_expect' |
| 09:02.20 | daXas | when i comment that line , it works ... but i think it is not a good way... anyone has idea? |
| 09:05.21 | tzafrir_laptop | daXas, "implicit declaration" usually comes after a warning about a missing header (.h) file |
| 09:06.31 | *** join/#asterisk phpboy (~sj@tbnb-165-220-50.telkomadsl.co.za) |
| 09:06.44 | phpboy | asterisk over VPN... what are the options regarding that? |
| 09:07.10 | tzafrir_laptop | phpboy, a vpn can add latency. You can easily ttest the latency using ping |
| 09:07.22 | phpboy | yeah, I've seen this |
| 09:07.51 | phpboy | but I need a vpn type solution, because unfortunately the way the internet works in my country... I have no option :< |
| 09:07.53 | srt | phpboy: i have connected several asterisk servers using openvpn - works pretty well |
| 09:08.13 | phpboy | srt: do u mind sending me your configs (server and client) ? |
| 09:08.15 | phpboy | please |
| 09:08.24 | tzafrir_laptop | phpboy, simply using IAX? |
| 09:08.39 | daXas | tzafrir_laptop, umm string.h unistd.h,stdlib.h, libpri.h, pri_internal.h .. these are the included ... you know where is __builtin_expect? |
| 09:08.57 | phpboy | tzafrir_laptop: the QOS rules on our ADSL network does not allow for that :/ |
| 09:09.04 | phpboy | our meaning my country |
| 09:09.18 | tzafrir_laptop | phpboy, what do you mean? |
| 09:09.40 | tzafrir_laptop | phpboy, both iax and openvpn require that you expose basically the same thing |
| 09:09.53 | phpboy | in South Africa |
| 09:10.30 | tzafrir_laptop | phpboy, what QOS rules? |
| 09:10.47 | phpboy | we have one telecom's company and they also happen to own the ADSL network... they have limited anything VoIP related to crap speeds so that people don't use VoIP instead of their telecom's lines... hence the need for VPN :/ |
| 09:11.09 | tzafrir_laptop | phpboy, how do they define voip? |
| 09:11.44 | tzafrir_laptop | phpboy, if you're luck, merely changing port numbers may help |
| 09:11.46 | phpboy | they have special rules on there firewalls for voip traffic :/ |
| 09:11.54 | phpboy | nope |
| 09:12.10 | phpboy | they limit it according to the traffic type and not the port :< |
| 09:13.36 | tzafrir_laptop | It should still be less overhead than an encypting wrapper protocol |
| 09:13.37 | Qorky | hmm when i call out of my fritz card i can never hear anything. yet the other end can hear me. anyone know why this would be ? |
| 09:13.38 | Qorky | please |
| 09:13.42 | X-Rob | phpboy, the traffic type is 'udp' |
| 09:13.55 | X-Rob | I find it very unlikely that they've throttled udp traffic |
| 09:14.00 | phpboy | X-Rob: I know... but that's not the problem :/ |
| 09:14.01 | X-Rob | because that would clag people playing online games. |
| 09:14.52 | tzafrir_laptop | X-Rob, you can classify packets. There's even a nice patch to ipfilter for "7-th layer filtering" or something like that |
| 09:15.31 | X-Rob | yes, but that's based on port and TOS |
| 09:15.39 | X-Rob | eg, stuff that can be easily changed. |
| 09:16.17 | phpboy | it's this is depressing |
| 09:16.18 | phpboy | :/ |
| 09:18.24 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 09:22.59 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
| 09:26.43 | tzafrir_laptop | so, anybody here using areskicc? |
| 09:27.28 | Ahrimanes | i tried it, but now using astcc, why? |
| 09:28.57 | tzafrir_laptop | I'm now batteling with it. I ran into an "interesting" issue which should interest others. I'd really like to have someone else test this on a working system first, though |
| 09:29.10 | tzafrir_laptop | How's astcc? |
| 09:30.34 | tzafrir_laptop | Qorky, what do you listen with? |
| 09:31.39 | tzafrir_laptop | X-Rob, I was talking about the contents of packets |
| 09:32.52 | RoyK | anyone that knows where i can find ES 300 738? i couldn't see it on ETSIs pages |
| 09:35.41 | Ahrimanes | tzafrir_laptop: astcc works well |
| 09:36.42 | X-Rob | quorky, what sort of phone are you using? |
| 09:41.22 | tzafrir_laptop | Ahrimanes, how do you compare the two feature-wise? |
| 09:41.54 | Ahrimanes | tzafrir_laptop: uhm, havent tested areskicc that much.. astcc fits my bill completely for now.. so dont have that good base to compare on |
| 09:42.10 | Ahrimanes | tzafrir_laptop: but i will probably start something new as AGI wont scale well in my solution |
| 09:43.43 | Qorky | tzafrir_laptop: my gxp2000 phone. that what you mean ? |
| 09:43.48 | Qorky | or my cisco phone. |
| 09:57.31 | phpboy | :/ |
| 09:57.47 | vuvie | hi am following the instructions here http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 |
| 09:59.28 | vuvie | ztcfg |
| 09:59.55 | vuvie | unable to open master device /dev/zap/ctl |
| 10:01.28 | X-Rob | tzafrir - if a tier 1 ISP is doing packet inspection, no wonder they're damn slow. |
| 10:03.03 | phpboy | I'm going to need to find a solution to my problem :/ |
| 10:04.35 | RoyK | ~seen zoa |
| 10:04.35 | jbot | zoa <zoa@82.103.76.147> was last seen on IRC in channel #asterisk, 16d 10h 29m 28s ago, saying: 'ciao twisted'. |
| 10:04.40 | RoyK | shit |
| 10:07.27 | Qorky | X-Rob: sorry i ment to reply to you. about type of phones. |
| 10:08.59 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:14.17 | X-Rob | qorky, um, well, fiik. You've tried different phones? |
| 10:14.39 | X-Rob | also try things like putting the call on hold and then picking it back up |
| 10:14.47 | X-Rob | make asterisk transcode it from alaw? |
| 10:15.03 | tzafrir_laptop | vuvie, ls -l /dev/zap/ctl |
| 10:15.05 | X-Rob | eg, set your phone to GSM |
| 10:15.47 | tzafrir_laptop | vuvie, I figure you use udev, though. READEME.ucev may help |
| 10:16.18 | tzafrir_laptop | README.udev , that is |
| 10:19.02 | *** join/#asterisk _omer (dfsdf@203.215.180.254) |
| 10:20.59 | Qorky | yes x-rob. and now i've changed stuff around i dont get any audio in either direction when i call out. |
| 10:21.08 | Qorky | yet when i call in, its perfect. |
| 10:21.18 | Qorky | i read somewhere about timeings on rdp ? |
| 10:21.24 | Qorky | could that be it ? |
| 10:21.30 | X-Rob | Yeah, I posted that link |
| 10:21.35 | X-Rob | how did you go with HEAD? |
| 10:21.56 | Qorky | yeah im using head now. |
| 10:21.59 | Qorky | no difference. |
| 10:22.03 | X-Rob | bugger. |
| 10:22.50 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 10:25.09 | |Fender| | have people try astcc ? |
| 10:26.02 | RoyK | no. noone has ever tried it. it's just written and never tested, never used |
| 10:26.20 | |Fender| | u funny.... |
| 10:27.08 | RoyK | :) |
| 10:27.26 | |Fender| | serious u try it ? is posible have multi call with astcc? |
| 10:28.05 | *** join/#asterisk Q0rky (~Pooa@dip-202-72-131-243.wa.westnet.com.au) |
| 10:33.51 | vuvie | thanks tzafrir |
| 10:42.39 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:43.35 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 10:43.43 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 10:43.57 | gambolputty | hi |
| 10:44.07 | gambolputty | I have * and odbc working but my phones don't register. |
| 10:45.41 | gambolputty | anyone have odbc fully working with *? |
| 10:46.14 | dros7 | I have mysql working very well with it |
| 10:46.59 | gambolputty | your phones register ok? |
| 10:47.43 | bublbobl | do you know how flash panel queries asterisk about SIP phones status (calling, idle, ... ) ? |
| 10:47.44 | dros7 | whups, didn't read your question completely. I'm just running a sip only setup right now |
| 10:48.19 | gambolputty | I have sip running in general too |
| 10:48.36 | dros7 | my sip phones show up fine |
| 10:48.43 | gambolputty | realtime and direct mysql calls work fine |
| 10:48.55 | gambolputty | but not when I make the database calls using odbc |
| 10:49.15 | dros7 | sorry, maybe someone else can help |
| 10:49.25 | RoyK | gambolputty: why don't you use direct mysql if it works? |
| 10:49.44 | gambolputty | I have to try something else for this box I am working on |
| 10:49.56 | *** join/#asterisk kaitseb (~sadie@jutrzenka.firma.o2.pl) |
| 10:51.23 | gambolputty | http://pastebin.com/303374 |
| 10:51.29 | gambolputty | could someone take a look? |
| 10:55.21 | gambolputty | amendment posted to same url |
| 11:00.11 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 11:01.47 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:03.44 | key2 | hey |
| 11:04.01 | key2 | how can I tell asterisk to send the zap call to an other asterisk ? |
| 11:04.28 | Ahrimanes | extensions.conf |
| 11:05.10 | key2 | Ahrimanes: not as a dialplan |
| 11:06.30 | Ahrimanes | key2: dialplan decides where the call goes |
| 11:06.53 | key2 | Ahrimanes: and there is no way to tell asterisk to just act like a gateway ? |
| 11:07.09 | key2 | to just do a ZAP <--> SIP |
| 11:10.39 | Ahrimanes | key2: yes, install sip and zap channels, configure them and use the dialplan to route the calls |
| 11:11.11 | *** join/#asterisk Corydon76-home (one@pcp08665860pcs.500ash01.tn.comcast.net) |
| 11:11.52 | key2 | Ahrimanes: if I have a quad E1 which means 120 concurrent calls, can I make them all go through the same SIPP channel ? |
| 11:12.04 | key2 | s/SIPP/SIP |
| 11:12.16 | Ahrimanes | key2: yes, that's rather standard setup |
| 11:18.41 | *** join/#asterisk KrzyszTof (~christof@194.32.39-62.rev.gaoland.net) |
| 11:20.12 | *** join/#asterisk RedRibbon (~lesha@195.14.41.242) |
| 11:20.15 | RedRibbon | hi, ppl |
| 11:20.35 | RedRibbon | I have a weird problem |
| 11:20.47 | RedRibbon | when I have pusedial=yes in zapata.conf |
| 11:20.57 | RedRibbon | (my telco supports only pulse) |
| 11:21.09 | RedRibbon | asterisk ignores "w" in Dial command |
| 11:21.23 | RedRibbon | so I can tell it to wait after dialing first digit |
| 11:21.32 | RedRibbon | which is "outside" line |
| 11:23.07 | RoyK | wait()? |
| 11:25.36 | Qorky | anyone know why on my fritz card pci i would only get outgoing audio not incomming when calling out my isdn ? |
| 11:29.04 | Romik_ | anybody know how i calculate MOS from .gsm file to compare providers quality on large volume on calls? or possible to calculate it in real time? |
| 11:32.52 | RoyK | MOS? |
| 11:33.18 | *** join/#asterisk flotox (~jovan@host211-204.pool8541.interbusiness.it) |
| 11:35.05 | X-Rob | OK I'm back |
| 11:36.07 | *** part/#asterisk KrzyszTof (~christof@194.32.39-62.rev.gaoland.net) |
| 11:38.23 | Romik_ | RoyK: the quality of voice call |
| 11:39.16 | X-Rob | I doubt you'd be able to do it from a gsm file |
| 11:39.22 | X-Rob | you'd need an uncompressed wav |
| 11:39.37 | X-Rob | and then some funky code to measure audio differences |
| 11:39.48 | X-Rob | note, telco's pay significant chunks of money for hardware that does that. |
| 11:41.29 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 11:42.20 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 11:42.41 | shaZwaz | hi ppl |
| 11:42.42 | *** join/#asterisk |HelioS| (ts07@ozashiki.com) |
| 11:43.09 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
| 11:43.11 | RedRibbon | RoyK: is it possible to combone several Dial's in one? and use Wait() between them? |
| 11:43.30 | shaZwaz | hi RoyK |
| 11:43.52 | shaZwaz | ~seen wasim |
| 11:43.52 | jbot | wasim <~wasim@203.81.220.100> was last seen on IRC in channel #asterisk, 17d 4h 24m 1s ago, saying: 'bonjour monsieur zeek'. |
| 11:44.39 | *** join/#asterisk tkapower (~tord@213.160.242.116) |
| 11:45.33 | tkapower | hi everybody |
| 11:45.55 | *** join/#asterisk meppl (mephisto@p54AAECB8.dip.t-dialin.net) |
| 11:47.01 | RoyK | RedRibbon: i can't see why not |
| 11:47.05 | RoyK | RedRibbon: perhaps use a macro |
| 11:47.14 | RoyK | hi, shaZwaz |
| 11:47.51 | X-Rob | http://maps.google.com/maps?q=northern+territory&ll=-23.705578,133.872871&spn=0.080509,0.124111&t=k&hl=en |
| 11:47.53 | vuvie | hi i've just make asterisk and zaptel and changed the extensions.conf, zaptel.conf and zapata.conf |
| 11:47.56 | X-Rob | __That__ is Alice Springs. |
| 11:48.06 | vuvie | ztcfg loads nicely |
| 11:48.29 | vuvie | problem : asterisk -vvvvvvvvvvcg |
| 11:48.35 | tkapower | screen |
| 11:49.00 | vuvie | shows error mkintf signalling requested on channel 1 is FXO but line is FXS |
| 11:49.09 | vuvie | inable to register chan_zap.c |
| 11:50.43 | vuvie | put my phone in no dial tone but pressing the buttons has pulse |
| 11:52.10 | shaZwaz | how come I can't find anything Release Notes for 1.0.8 |
| 11:52.41 | RedRibbon | RoyK: after first Dial() it connects SIP to ZAP/1 thus it is not getting further... |
| 11:53.08 | RoyK | sorry. don't know then |
| 11:53.12 | RoyK | i don't do analog stuff |
| 11:53.24 | shaZwaz | whats with the analog ? |
| 11:53.25 | RedRibbon | RoyK: I hope some day I will not have to do it too ;) |
| 11:54.22 | shaZwaz | RoyK: whats a 401 SIP error ? |
| 11:54.32 | X-Rob | extension not found |
| 11:54.38 | X-Rob | eg, you've dialled a number that doesn't exist |
| 11:54.41 | shaZwaz | my authorization seems to be ok |
| 11:55.15 | shaZwaz | No X-Rob its 404 |
| 11:55.21 | shaZwaz | not 401 |
| 11:56.02 | X-Rob | Oh yeah |
| 11:56.04 | X-Rob | don't mind me |
| 11:57.00 | RoyK | shaZwaz: unauthorized? |
| 11:57.15 | RoyK | rtfrfc |
| 11:57.22 | RoyK | ~rtfrfc |
| 11:57.30 | shaZwaz | thats what I have found in the description but eveything seems fine |
| 11:58.00 | RoyK | ~rtfrfc |
| 11:58.00 | jbot | rtfrfc is, like, Read that fscking RFC |
| 11:58.25 | RoyK | ~lart tkapower |
| 11:58.25 | shaZwaz | except for one thing I have seen in the SIP REGISTER message Contact field has a * |
| 11:59.31 | shaZwaz | is that preventing my SIP phone from registering my * server |
| 12:01.44 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 12:05.32 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:18.54 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
| 12:22.25 | Romik_ | Royk:any idea how to metter MOS? |
| 12:22.43 | Ahrimanes | anyone done call-waiting with * |
| 12:22.56 | RedRibbon | RoyK: tried Dial, Wait, Dial macro. Zap connects to SIP after first Dial, if I will set timeout "2" for first Dial, then Zap will hangup the line |
| 12:28.48 | MikeJ[Laptop] | lather, rinse, repeat |
| 12:30.20 | RoyK | what about dial(1.......92384759 |
| 12:30.27 | RoyK | iirc . used to be pause |
| 12:30.28 | *** join/#asterisk mistral (mistral@jstevenson.plus.com) |
| 12:30.49 | *** part/#asterisk vuvie (~S@bb219-74-45-231.singnet.com.sg) |
| 12:32.19 | MikeJ[Laptop] | you want to dial, then send digits after the answer? |
| 12:32.52 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:33.05 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 12:33.16 | X-Rob | redribbon, you want to send DTMF tones |
| 12:33.17 | X-Rob | ? |
| 12:33.50 | X-Rob | Dial(Zap/g0/123,D(1234)) will dial 123, wait until it's answered, and then send 1234 via DTMF |
| 12:33.53 | X-Rob | read the fucking wiki. |
| 12:34.50 | MikeJ[Laptop] | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial <-----THAT ONE |
| 12:34.52 | MikeJ[Laptop] | :) |
| 12:35.03 | X-Rob | Thanks mike. |
| 12:35.09 | X-Rob | Happy friday! |
| 12:35.10 | *** join/#asterisk cchasg (~cchasg@24.96.150.105) |
| 12:35.58 | X-Rob | For anyone who cares |
| 12:35.58 | X-Rob | http://maps.google.com/maps?q=northern+territory&ll=-28.451157,132.145443&spn=0.040255,0.062056&t=k&hl=en |
| 12:36.08 | X-Rob | That is the site of the first atomic explosion in Australia |
| 12:36.11 | X-Rob | back in 1964. |
| 12:36.35 | MikeJ[Laptop] | if you need a wait after answer, there is a char for pause, I think it's 'w' off the top of my head..but don't quote me like Dial(zap/g0/5431,,D(wwwww${EXTEN})) |
| 12:37.30 | MikeJ[Laptop] | how do you get google maps to do sat pictures? |
| 12:38.12 | X-Rob | There's a link, top right. 'Satellite' |
| 12:38.23 | X-Rob | Note how you can click on 'Map' |
| 12:38.56 | X-Rob | oh, and if you zoom out, the 'blue' is totally inaccurate. |
| 12:39.01 | X-Rob | it hasn't rained there in about 15 years. |
| 12:39.59 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:40.18 | MikeJ[Laptop] | thats new... |
| 12:40.31 | X-Rob | Yup. They turned it on on monday |
| 12:40.37 | MikeJ[Laptop] | very cool |
| 12:40.42 | X-Rob | ooh |
| 12:40.48 | X-Rob | You can see my place 8) |
| 12:40.56 | MikeJ[Laptop] | so you can look up your address on the regular map, then switch to sat |
| 12:41.02 | MikeJ[Laptop] | let me see |
| 12:41.06 | X-Rob | Not for oz |
| 12:41.11 | X-Rob | we don't have street directions |
| 12:42.07 | MikeJ[Laptop] | :( |
| 12:42.09 | X-Rob | http://maps.google.com/maps?q=northern+territory&ll=-23.852381,151.246816&spn=0.005032,0.007757&t=k&hl=en |
| 12:42.11 | file[laptop] | hi |
| 12:42.18 | MikeJ[Laptop] | I live near the zoo: http://maps.google.com/maps?q=huntington+woods,+mi&ll=42.479925,-83.166933&spn=0.039525,0.032401&t=k&hl=en |
| 12:42.31 | MikeJ[Laptop] | to the right of the golf course |
| 12:42.38 | X-Rob | my place is the one on the corner |
| 12:42.47 | X-Rob | can't zoom in any further |
| 12:42.59 | X-Rob | but you can see the long driveway |
| 12:43.10 | X-Rob | house is || way |
| 12:43.35 | X-Rob | the guy behind me is a professional guituarist, and damn good |
| 12:43.44 | X-Rob | and the empty block to the left I play on my motorbike! |
| 12:44.04 | X-Rob | if you move a little bit to the left, you can see a big empty area at the bottom of the street |
| 12:44.15 | X-Rob | all the young kids go there and practice on their trailbikes |
| 12:44.24 | X-Rob | you can see some tracks in the scrub there |
| 12:44.33 | *** join/#asterisk MacDeath (davidn@196.202.248.34) |
| 12:44.38 | MacDeath | Hi there |
| 12:44.51 | MacDeath | this is an old question |
| 12:45.05 | MacDeath | which was solved for me once before |
| 12:45.15 | MacDeath | I need to debug the startup of Asterisk |
| 12:45.17 | MacDeath | Asterisk Ready. |
| 12:45.18 | MacDeath | *CLI> Segmentation fault |
| 12:45.30 | MacDeath | how can i see what is causing that |
| 12:45.44 | X-Rob | odds are it's a module |
| 12:45.50 | X-Rob | turn off autoload=yes in modules.conf |
| 12:45.57 | X-Rob | then explicitly turn them on |
| 12:46.18 | MacDeath | which modules should be loaded? |
| 12:46.18 | X-Rob | Mike, that looks like a golf course? |
| 12:46.21 | MacDeath | sip.conf? |
| 12:46.57 | MikeJ[Laptop] | in the middle, then too the right is the zoo.. |
| 12:47.09 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:47.21 | MikeJ[Laptop] | I live 4 blocks north and 2 blocks west of the nw corner of the golf course |
| 12:47.54 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:48.02 | MacDeath | isnt there a debug command which would show which modules are causing the problem? |
| 12:48.18 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:48.29 | MikeJ[Laptop] | MacDeath, asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 12:48.54 | Pete_Largo | Mike, isn't it 10 'v' and a 'd'? |
| 12:49.03 | RoyK | MacDeath: asterisk -gvvvvvvc |
| 12:49.07 | RoyK | with -g it dumps the core |
| 12:49.20 | RoyK | and then you can gdb asterisk core.xxx and 'bt' to create a backtrace |
| 12:49.29 | X-Rob | Aaaaah |
| 12:49.32 | X-Rob | indeed it is a zoo-like thing. |
| 12:49.36 | tzafrir_laptop | asterisk -c with no -U? running asterisk as root? |
| 12:49.50 | RoyK | why not..... |
| 12:49.54 | RoyK | for testing |
| 12:50.34 | tzafrir_laptop | does the manager api's "command" permission allows you to use the CLI command '!'? |
| 12:50.54 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:52.12 | MacDeath | RoyK : thanks |
| 12:52.16 | MacDeath | thats what I was looking for |
| 12:52.37 | JunK-Y | tzafrir: sure u can execute CLI commands via manager. |
| 12:53.39 | *** join/#asterisk NoNoUrS (~Guillaume@stardust.noc.frontier.fr) |
| 12:53.52 | NoNoUrS | hello all |
| 12:54.29 | NoNoUrS | some on test http://www.areski.net/asterisk-meetme/about.php in asterisk ? |
| 12:54.49 | NoNoUrS | please explain me the extension.conf |
| 12:56.28 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
| 12:57.07 | *** join/#asterisk iCEBrkr (icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 12:57.12 | MikeJ[Laptop] | NoNoUrS, documentation on extension.conf can be found on the wiki . |
| 12:57.16 | MacDeath | Royce : the debug is not helping me today :( |
| 12:57.31 | MikeJ[Laptop] | MacDeath, bt full? |
| 12:57.43 | MacDeath | let me try that one |
| 12:57.55 | MikeJ[Laptop] | you got attached to the core file? |
| 12:58.22 | MikeJ[Laptop] | you also may want to make valgrind, so you get symbols in your core |
| 12:58.38 | RoyK | just by |
| 12:58.41 | RoyK | and pastebin it |
| 12:58.56 | MikeJ[Laptop] | jusy by what? |
| 12:59.09 | MikeJ[Laptop] | ;) |
| 12:59.35 | MikeJ[Laptop] | file[laptop], pictures? |
| 12:59.39 | NoNoUrS | the wiki do no contain the specification for meetme2 in extension.conf |
| 12:59.53 | MikeJ[Laptop] | NoNoUrS, try again |
| 13:00.48 | MikeJ[Laptop] | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe |
| 13:01.03 | MikeJ[Laptop] | Oh look, it's a search function, how nice. |
| 13:01.12 | MikeJ[Laptop] | file[laptop], pictures? |
| 13:01.39 | MikeJ[Laptop] | echocancel=no |
| 13:03.24 | file[laptop] | unavailable |
| 13:03.30 | MikeJ[Laptop] | :( |
| 13:03.34 | MikeJ[Laptop] | damn you |
| 13:03.37 | file[laptop] | :p |
| 13:03.43 | MikeJ[Laptop] | you recover yet? |
| 13:04.30 | file[laptop] | I got no good sleep lastnight |
| 13:04.37 | MikeJ[Laptop] | good |
| 13:04.41 | MikeJ[Laptop] | aussie |
| 13:04.43 | Rob[laptop] | Yeah! |
| 13:04.52 | file[laptop] | yay laptop |
| 13:04.55 | Rob[laptop] | Peer pressure |
| 13:05.09 | MikeJ[Laptop] | <PROTECTED> |
| 13:05.26 | Rob[laptop] | heh |
| 13:05.31 | MikeJ[Laptop] | :) |
| 13:05.39 | MikeJ[Laptop] | my battery is going... |
| 13:05.40 | RoyK | /kick MikeJ[Laptop] idiot :P |
| 13:05.44 | tzafrir_laptop | MikeJ[Laptop], is that PHP syntax in your nick? |
| 13:05.46 | MikeJ[Laptop] | ;P |
| 13:06.00 | tzafrir_laptop | I stick with plain shell |
| 13:06.06 | *** part/#asterisk RedRibbon (~lesha@195.14.41.242) |
| 13:06.28 | *** join/#asterisk pjo (~chatzilla@mail.trueafrican.com) |
| 13:06.30 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:06.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:07.49 | pjo | hi all, using Asterisk::AGI->get_data to capture digits during a call doesn't work with _MOST_ software sipphones. any ideas/recomendations? |
| 13:08.26 | Ariel_ | hello everyone |
| 13:08.35 | X-Rob | pjo, use the RFC option, rather than 'inline' |
| 13:08.37 | X-Rob | in the phone |
| 13:08.42 | X-Rob | 'Send DTMF via:' |
| 13:08.50 | MikeJ[Laptop] | pjo, set up your dtmf options right |
| 13:09.11 | file[laptop] | RFC2833. |
| 13:09.18 | pjo | I've tried that...most phones simply don't send dtmf to my scripts |
| 13:09.39 | MikeJ[Laptop] | did you set up asterisk with the same type as you set the phones? |
| 13:09.46 | pjo | also, with wrong dtmf u can't make a call in the first place can u? |
| 13:09.52 | file[laptop] | sure you can |
| 13:10.03 | pjo | hmmm.. |
| 13:10.06 | MikeJ[Laptop] | ususally somone says that and they set it up in one place and not the other. |
| 13:10.22 | pjo | ok, well now I'm using firefly, so let me set it in *both* |
| 13:10.23 | MikeJ[Laptop] | pjo, call not made w/ dtmf, sip makes a call w/ an invite message |
| 13:10.40 | MikeJ[Laptop] | and don't set it as inband |
| 13:10.42 | MacDeath | does this mean anything |
| 13:10.45 | MacDeath | #0 0x40027b22 in pthread_mutex_lock () from /lib/libpthread.so.0 |
| 13:10.50 | file[laptop] | deadlock! |
| 13:10.51 | pjo | ok. |
| 13:11.18 | pjo | MikeJ[Laptop]: ok - that clears up some confusion I've had for weeks. |
| 13:11.35 | file[laptop] | SIPtastic! |
| 13:11.48 | shaZwaz | does anyone know what a * means in sip contact field ? |
| 13:12.46 | NoNoUrS | for the meetme my configuration is a realtime and please explain me how to connect meetme2 and db mysql |
| 13:13.14 | file[laptop] | meetme2 is a completely different module, it can't use realtime... and you even have to modify it to get it to compile |
| 13:13.28 | file[laptop] | anthm: just get the filipinos to do it |
| 13:15.21 | file[laptop] | yay my parents leave tomorrow |
| 13:15.25 | *** join/#asterisk rnovotny22 (~rnovotny2@207.195.206.201) |
| 13:16.31 | Hmmhesays | yay it's the weekend in 8 hours |
| 13:17.33 | |HelioS| | weekend in 2 hours :) |
| 13:17.56 | NoNoUrS | that ok file [laptop] then connect to meetme1 to realtime because in the wiki do not explain the extension.conf |
| 13:20.00 | rnovotny22 | Anyone online that has sugestions on setting up Polycom 501 units to stay connected to *? |
| 13:21.22 | Ariel_ | rnovotny22, what's the problem mine stay connect without issue's. |
| 13:22.04 | rnovotny22 | After about an hour they are no longer registered with *. Host is showing as Unspecified. |
| 13:22.38 | rnovotny22 | The IP501's are set up as Dynamic IP. |
| 13:23.27 | SwK | Ariel_: you have some 501s? |
| 13:23.41 | rnovotny22 | I have 1 500 and 5 501/s |
| 13:23.58 | Ariel_ | SwK, I just setup at a customer site 13 of them. |
| 13:24.10 | SwK | Ariel_: how are the 501s compared to the 500s? |
| 13:24.27 | Ariel_ | look and feel same |
| 13:24.31 | X-Rob | 501's have button fly's. |
| 13:24.36 | SwK | same stuff but shit with the 1.5.2 firmware? heh |
| 13:24.53 | SwK | <PROTECTED> |
| 13:25.20 | Ariel_ | rnovotny22, if your phones loose registration then you need to look at the network. I have no problem with them. |
| 13:25.39 | Ariel_ | SwK, actually they shipped with 1.4.1 |
| 13:25.44 | SwK | really... |
| 13:25.53 | Ariel_ | yep all of them |
| 13:26.11 | SwK | i'm not to impressed w/ 1.5.2... cant seem to figure out how to turn blind transfers back on |
| 13:26.17 | Ariel_ | and they took about 3 reboots to upgrade them and about 3 minutes for the first start. |
| 13:26.37 | Ariel_ | SwK, I love the blind transfer part of it. |
| 13:26.44 | Katty | dooby do do do |
| 13:27.06 | SwK | Ariel_: you know where the setting for blind transfer on 1.5.2 happens to hide? |
| 13:27.09 | Ariel_ | when you press transfer and start putting the exten you want to send them the blind buttom comes up on the distpaly. |
| 13:27.14 | Ariel_ | Katty, hello |
| 13:27.27 | SwK | on 1.4.1 or 1.5.2? |
| 13:27.52 | Ariel_ | SwK, reset the phones to default and my setup are really trimed down ones. |
| 13:28.13 | Ariel_ | I am running them on 1.5.2 with the older boot rom 2.6 not 3.x |
| 13:28.32 | SwK | yeah my setup are all running a very trim config w/ 1.5.2 and the 2.6 bootrom |
| 13:28.32 | Katty | :> |
| 13:28.46 | SwK | tftp booting them all |
| 13:28.50 | Ariel_ | ftp |
| 13:29.21 | SwK | i been meaning to check out the 3.x bootrom and http booting |
| 13:29.38 | SwK | thinking about generating configs on the fly from a DB |
| 13:30.18 | Ariel_ | SwK, I know that someone on the amportal is working on a deployment web for them. |
| 13:30.24 | MacDeath | i think i am being an idiot now |
| 13:30.32 | MacDeath | my asterisk was working yesterday |
| 13:30.34 | SwK | i already have a deployment web |
| 13:30.36 | MacDeath | i rebooted the machine |
| 13:30.48 | Ariel_ | why reboot |
| 13:30.57 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 13:31.05 | MacDeath | well, there was a prolonged power failure |
| 13:31.06 | SwK | Ariel_: windows taught him reboots fix everything |
| 13:31.15 | MacDeath | naa, was power problems |
| 13:31.21 | MacDeath | now I get segmentation fault |
| 13:31.36 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 13:31.39 | SwK | sounds like something got corrupted... |
| 13:31.39 | MacDeath | is there something besides insmod zaptel that I am forgetting |
| 13:31.44 | Ariel_ | MacDeath, what did you upgrade or change in the pass weeks. |
| 13:31.45 | MacDeath | i have done zaptel |
| 13:31.47 | SwK | modprobe zaptel |
| 13:31.50 | MacDeath | and ztcfg |
| 13:32.00 | MacDeath | Ariel_ : its a really old installation.... |
| 13:32.01 | Ariel_ | service zaptel restart |
| 13:32.07 | SwK | and you might want to load the driver what whatever hardware card you have |
| 13:32.08 | MacDeath | CVS of december 28 |
| 13:32.25 | Ariel_ | dec 28 oh boy your behind... |
| 13:32.30 | MacDeath | there isnt any hardware :( |
| 13:32.36 | MacDeath | just sip phones |
| 13:32.51 | MacDeath | maybe you could tell me what modules i should be loading |
| 13:32.57 | MacDeath | and i will load each one manually |
| 13:33.05 | MacDeath | and see where the error is |
| 13:33.06 | SwK | then you dont even have to load zaptel uness you are loading ztdummy for MOH and meetme timing |
| 13:33.21 | MacDeath | I was using ztdummy |
| 13:33.28 | fish_ | i need the oh323 plugin for asterisk if i want h323? |
| 13:33.40 | Ariel_ | did you upgrade or patch your kernel in the pass few months? |
| 13:33.55 | MacDeath | didnt change anything :( |
| 13:33.59 | MacDeath | just IP addresses |
| 13:34.04 | Ariel_ | fish_, there are few h323 type setups for asterisk out there. |
| 13:34.05 | MacDeath | but have rebooted since then |
| 13:34.44 | fish_ | k |
| 13:36.14 | Hmmhesays | chan_h323 or chan_Oh323 |
| 13:36.28 | der[mat] | re |
| 13:36.48 | Ariel_ | how about the new one from pbxfreesomethign... |
| 13:37.00 | file[laptop] | or chan_ooh323 or chan_woomera |
| 13:37.05 | der[mat] | one mgcp question again...... |
| 13:37.15 | Ariel_ | MacDeath, if you load the asterisk -vvvvgc it should stop at the last item that is failing. |
| 13:37.18 | der[mat] | the difference between d001 and aaln? |
| 13:37.28 | Ariel_ | woomera |
| 13:37.31 | Ariel_ | that's it |
| 13:37.33 | MacDeath | Ariel_ : libc |
| 13:37.52 | MacDeath | #0 0x40027b22 in pthread_mutex_lock () from /lib/libpthread.so.0 |
| 13:38.20 | Ariel_ | MacDeath, copy your config files and upgrade |
| 13:38.29 | *** join/#asterisk Tili (~Tili@202-133-67-102-dialup.sat.net.pk) |
| 13:40.36 | shaZwaz | anyone has any idea why my GS is sendin * in SIP message Contact field ? |
| 13:40.44 | fish_ | does the h323 gateway support authentification somehow? |
| 13:40.59 | anthm | Ariel_, pbxfreesomethign... ? |
| 13:41.01 | Tili | has anybody used radius with asterisk-b2bua. What service type should I set for radius access |
| 13:41.07 | fish_ | well, the oh323 channel gateway |
| 13:41.33 | Ariel_ | anthm, sorry I am doing 3 things right now. |
| 13:43.17 | anthm | you want to play with h323, got betatest the chan_woomera |
| 13:43.39 | MacDeath | Ariel_ : sad thing is |
| 13:43.52 | MacDeath | that machine doesnt have internet access |
| 13:43.53 | MacDeath | lol |
| 13:43.54 | Katty | oh |
| 13:43.55 | *** join/#asterisk rittwage[work] (~peter@intsys.net) |
| 13:44.02 | Katty | anthm: sorry ;) |
| 13:44.07 | anthm | *grin* |
| 13:44.22 | Katty | would you rather hear me ramble to my co-workers :P |
| 13:44.43 | Ariel_ | MacDeath, sad |
| 13:44.50 | Hmmhesays | Katty: |
| 13:44.54 | MacDeath | i know |
| 13:44.58 | MacDeath | but it was all working |
| 13:44.58 | anthm | actually music is going well with my impatient frequent reload of slashdot |
| 13:45.12 | Katty | calm down :) |
| 13:45.27 | Katty | either neal posts it or he doesn't |
| 13:45.34 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 13:45.35 | Hmmhesays | should I poke you with a pen for pampling me? |
| 13:45.48 | Katty | i asked him to...it's up to him now :) |
| 13:45.52 | Katty | Hmmhesays: :< |
| 13:46.12 | Hmmhesays | I dunno... that's what I got for a hugging you yesterday |
| 13:46.19 | Ariel_ | MacDeath, do that MS says turn the server off leave it off for 5 minutes then restart (hehehehe). |
| 13:46.20 | Katty | oh |
| 13:46.22 | Katty | k'then |
| 13:46.28 | Hmmhesays | lol |
| 13:47.03 | Hmmhesays | awww :) |
| 13:47.11 | Katty | i'm such a hugger it's almost sick. |
| 13:47.38 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 13:48.21 | Hmmhesays | i could really use a recliner at work |
| 13:48.36 | file[laptop] | yay drugs |
| 13:49.04 | file[laptop] | how long does it take advil to kick in? |
| 13:49.13 | Katty | file[laptop]: twice as long as half |
| 13:49.24 | file[laptop] | really? how nice |
| 13:49.27 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 13:49.41 | Katty | i know |
| 13:50.05 | Hmmhesays | depends on how many you took, and how empty your stomach is |
| 13:50.29 | file[laptop] | I haven't eaten in like, a day... and I only took one for now |
| 13:50.36 | Hmmhesays | body weight, metabolism, etc... |
| 13:50.47 | file[laptop] | meh I'll just wait till it kicks in |
| 13:51.05 | Ariel_ | file[laptop], in other words no one really knows. |
| 13:51.18 | Hmmhesays | i usually take 3 |
| 13:51.23 | Hmmhesays | if I take them at all |
| 13:52.03 | Hmmhesays | depends on how bad the hangover is |
| 13:52.31 | *** join/#asterisk jaike (~a@203.131.137.76) |
| 13:52.36 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
| 13:52.43 | *** join/#asterisk darwin35 (~darwin35@ip70-186-117-198.ma.dl.cox.net) |
| 13:52.45 | *** join/#asterisk DarthClue (cluecon@wsip-68-99-73-32.tu.ok.cox.net) |
| 13:52.48 | Ariel_ | hangovers raw eggs and another drink. |
| 13:53.02 | Juggie | mcdonalds is the hangover cure |
| 13:53.20 | Ariel_ | Hmmhesays, you have not been in the military..... |
| 13:53.30 | Hmmhesays | Ariel_ this is true |
| 13:55.12 | MikeJ[Laptop] | pbxfreeware.org? |
| 13:55.21 | MacDeath | Ariel_ : what is the minimum modules I need to load sip.conf |
| 13:55.33 | MacDeath | Ariel_ : what is the minimum modules I need to load chan_sip.so |
| 13:55.40 | Hmmhesays | I'm listening to Tesla right now |
| 13:55.48 | Hmmhesays | lazy days, crazy nights |
| 13:56.03 | MikeJ[Laptop] | MacDeath, there is stuff on slimming on the wiki, start there |
| 13:56.24 | Ariel_ | MacDeath, well that really depends on what your setup has. But I would look at all the .so and noload most lime modem, and work from that. |
| 13:56.38 | Ariel_ | lime/like |
| 13:57.04 | *** part/#asterisk der[mat] (~mat@gate-nue0.bintec.de) |
| 13:57.22 | MacDeath | im busy loading them all manually |
| 13:57.40 | MacDeath | Jun 24 15:53:04 WARNING[2054]: loader.c:309 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_park_call |
| 13:57.47 | MacDeath | but i need to do something in the correct order |
| 13:57.58 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:58.19 | robl^ | Ariel_, my neighbor complained my TV was too loud while watching Teletubbies |
| 13:58.34 | *** join/#asterisk Tangent (~Arc_Tange@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk) |
| 13:58.37 | tzafrir_laptop | MacDeath, it's from libpri, IIRC |
| 13:58.40 | Ariel_ | teletbbies are not allowed in my house. |
| 13:59.22 | file[laptop] | ast_park_call is in res_features/res_parking... depending on your version of asterisk |
| 13:59.41 | Zeeek | what is the advantage of valet_parking? |
| 14:00.00 | file[laptop] | Zeeek: more control, and it's good for use in a hosted PBX environment where you need separate parking spots |
| 14:00.04 | Zeeek | ok |
| 14:00.14 | robl^ | is valet parking even supoorted?? I thought the latest was Super Valet Parking? |
| 14:00.22 | file[laptop] | meh whatever |
| 14:00.24 | Zeeek | I was curious why parking didn't just tell the available slot instead of having to enter 700 ? |
| 14:00.33 | *** join/#asterisk cadal (~darky@103.Red-80-32-7.pooles.rima-tde.net) |
| 14:00.34 | MacDeath | tzafrir : thz |
| 14:00.35 | MacDeath | tzafrir : thx |
| 14:00.35 | shaZwaz | hi Zeeek |
| 14:00.41 | Zeeek | hi shaZwaz |
| 14:00.44 | MacDeath | i think that might explain the problems im having |
| 14:00.44 | cadal | hi all |
| 14:00.58 | shaZwaz | <PROTECTED> |
| 14:01.02 | Zeeek | my users are hard pressed to see the poujnt of having to dial 700 every time |
| 14:01.10 | cadal | anyone here has a Linksys Pap2? |
| 14:01.16 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 14:01.26 | Zeeek | seems to me it could have been # "transfer" # "seven oh one" |
| 14:01.54 | Ahrimanes | ey Zeeek :D |
| 14:02.01 | Zeeek | Beer? |
| 14:02.08 | Zeeek | haven't had any since Madrid |
| 14:02.12 | Ahrimanes | Zeeek: omg |
| 14:02.19 | Zeeek | no wait not true |
| 14:02.25 | robl^ | Zeek: the idea was that if you hjave a larger office, you may not know if a parking slot is already in use. 700 chooses the first empty one and tells you which it was |
| 14:02.29 | Ahrimanes | Zeeek: i'll be in narbonne from 1/7-05 |
| 14:02.29 | Zeeek | I had some just before the big wine dinner monday nite |
| 14:02.32 | tzafrir_laptop | cadal, me |
| 14:02.46 | Zeeek | Ahrimanes go to the beach every day |
| 14:02.50 | Katty | DarthClue: :> |
| 14:02.57 | Ahrimanes | Zeeek: hehe will do, is there ip in that area? |
| 14:03.04 | Zeeek | ip what? |
| 14:04.32 | Ahrimanes | Zeeek: connectivity |
| 14:04.36 | *** join/#asterisk MichaelSaunders (~mick@202.22.163.104) |
| 14:05.28 | [Latre] | i need help with parking calls.....i configured features.conf, extensssions.conf like said in voip.info.org but not works.....someone has configured parking calls??? |
| 14:05.57 | Katty | oh, what? |
| 14:06.06 | Katty | i wasn't paying attention to channel |
| 14:06.16 | Zeeek | France is prolly better connected all over than most countries outside Scandinavia |
| 14:06.24 | Zeeek | Ahrimanes ^^^ |
| 14:06.43 | Zeeek | Your gramma has 2048/256 by now |
| 14:06.52 | Ahrimanes | Zeeek: ok will bring a videophone then :P |
| 14:06.58 | Zeeek | IAX only |
| 14:07.04 | Ahrimanes | haha my mom cant even get isdn i think |
| 14:07.07 | Ahrimanes | why only iax? |
| 14:07.19 | Zeeek | cause it's politically correct on this channel to sat that |
| 14:07.22 | Ahrimanes | hha |
| 14:08.42 | *** join/#asterisk dial_911_now (~niggerplz@66-55-197-254.gwi.net) |
| 14:11.26 | shaZwaz | Zeeek: have any idea why my GS is sendin * in SIP message Contact field ? |
| 14:11.51 | cadal | anyone knows where to get more info about the linksys pap2 ? |
| 14:12.06 | Zeeek | from linksys? from a user group? |
| 14:12.17 | cadal | linksys documentation is really weak |
| 14:12.25 | cadal | this device has a ton of features |
| 14:12.30 | cadal | and none is documented |
| 14:12.30 | Zeeek | I invented the web for that reason :) |
| 14:12.46 | Ahrimanes | haha |
| 14:12.47 | *** join/#asterisk bernarde (~bernarde@bernarde.user) |
| 14:12.51 | Zeeek | there are users of all that stuff that have already taken it apart and know more than linksys about it |
| 14:12.55 | Ahrimanes | i wrote google for that purpose.. or wait |
| 14:13.00 | cadal | lol |
| 14:13.13 | Ahrimanes | i was bored for a weekend, what can i say |
| 14:13.20 | Zeeek | Wehn Al Gore came to me with that silly tin can shit, I set him straight |
| 14:13.32 | cadal | yeah, it was Al who invented all the internet |
| 14:13.37 | cadal | even voip |
| 14:13.37 | Zeeek | shaZwaz more info about your problem |
| 14:13.47 | Zeeek | no voip was me again |
| 14:13.57 | shaZwaz | I am getting a 401 rejection by * |
| 14:13.57 | Zeeek | and it's voy-p |
| 14:14.09 | cadal | lol |
| 14:14.17 | shaZwaz | checked user/pass |
| 14:14.35 | Zeeek | rebooted phone? |
| 14:14.38 | shaZwaz | sip debug shows contact: * sent by GS |
| 14:14.52 | cadal | the world is changing |
| 14:14.56 | cadal | you have to reboot phones |
| 14:14.57 | cadal | xD |
| 14:15.13 | Ariel_ | [Latre], what error is displayed when you transfer to exten 700 |
| 14:15.26 | shaZwaz | well it kind of behaving randomly |
| 14:15.36 | Zeeek | oh, it is a real GS, then? |
| 14:15.43 | Zeeek | not a GS clone? |
| 14:15.44 | shaZwaz | sometimes it registers ok sometimes not |
| 14:15.50 | Zeeek | yep, that's them |
| 14:15.58 | shaZwaz | them ? |
| 14:16.08 | Zeeek | Firmware version: ? |
| 14:16.18 | shaZwaz | just upgraded to the latest |
| 14:16.23 | Zeeek | if (user says I don't know) end; |
| 14:16.28 | shaZwaz | 1.0.6.7 |
| 14:16.38 | Zeeek | ahhhhhhh that's why! |
| 14:16.46 | Zeeek | and you can NEVER GO BACK!!!! muhahahahaha |
| 14:16.56 | shaZwaz | tot may be it was a firmware prob but no success |
| 14:17.10 | shaZwaz | it same as before |
| 14:17.15 | Zeeek | did this same fone work fine for a while though? |
| 14:17.21 | shaZwaz | yup |
| 14:17.29 | Zeeek | I hate when that happens |
| 14:17.41 | shaZwaz | wonder why its behaving randomly |
| 14:17.48 | Zeeek | for dessert, some ice cream I think |
| 14:17.52 | shaZwaz | is it the contact field problem ? |
| 14:18.11 | Zeeek | has the asterisk changed? |
| 14:18.21 | shaZwaz | I think so, coz thats where it should be contacted |
| 14:18.34 | shaZwaz | no same as before |
| 14:18.54 | shaZwaz | but the same phone works ok on LAN |
| 14:19.04 | shaZwaz | only not on this ADSL connection |
| 14:19.10 | shaZwaz | with DHCP |
| 14:20.28 | [Latre] | Ariel_: only the person that call listen music but not transfer...i dont see any error |
| 14:20.29 | dros7 | is it behind a NAT? |
| 14:20.51 | Ariel_ | [Latre], are all your phones doing this? |
| 14:20.54 | shaZwaz | not sure ,but may be |
| 14:21.10 | [Latre] | Ariel_: i have 2 SIP phones budgetone100 |
| 14:21.28 | shaZwaz | well tried replacing it and the problem seems to be still there |
| 14:21.53 | shaZwaz | is the ADSL DHCP doing something ? |
| 14:22.28 | Ariel_ | [Latre], add the tT option to the dial string to the phones to see if the # key will transfer. I think that BS transfer key is the problem. (check your firmware as well). |
| 14:22.34 | jr99 | argh.. every morning I must do a sip reload to get my sip line working again? Is this a cvs head issue? |
| 14:22.55 | Zeeek | BS transfer key has never worked for me |
| 14:23.10 | [Latre] | the option tT in users in sip.conf? |
| 14:23.11 | shaZwaz | jr99: same prob |
| 14:23.22 | Ariel_ | [Latre], did you see what Zeeek said. |
| 14:23.42 | Zeeek | reams of complaints on the mailing list about transfer button |
| 14:24.00 | Zeeek | supposedly fixed in firmware 1.?.? |
| 14:24.21 | shaZwaz | Zeeek: you think its the phone ? |
| 14:24.30 | shaZwaz | or anything else |
| 14:24.53 | Zeeek | shaZwaz I was talking about the other thing, but no I don't think your prob is the phone, I think the change in network is your problem |
| 14:25.09 | [Latre] | mmm...Program--1.0.5.11 Bootloader--1.0.0.18 |
| 14:25.19 | shaZwaz | what you suggest ? |
| 14:25.30 | Zeeek | [Latre] that's very old |
| 14:25.49 | mishehu | I need to extend a PRI line from the telecom closet in a building back to my space. I can't remember if it's 2 pair or 4 pair for a pri, and what type of jack and pinout I need. anybody know of a link or something with good info? (I'm in the USA, btw) |
| 14:25.53 | Zeeek | [Latre] I'd recommend (with no risk to me, but free advice to you) 5.18 |
| 14:26.09 | *** join/#asterisk Thus0 (~Thus0@190.111.102-84.rev.gaoland.net) |
| 14:26.11 | Zeeek | because that was a very stable version and I think transfer may even work |
| 14:26.27 | [Latre] | Zeeek: but that is firmware of budgetone 100 |
| 14:26.28 | Zeeek | I have no experience with the newer versions, I'm at .12 myself |
| 14:26.49 | Zeeek | [Latre] yes, and you put any version you wish in |
| 14:27.06 | *** join/#asterisk ManxPower (~eric@stirprop-S4-0-0-21.ndcr2.datasync.net) |
| 14:27.19 | Ariel_ | mishehu, it's 2 pair but use a cat5 4 pair will work. |
| 14:29.04 | mishehu | Ariel_: datasheet here for mine says I need an RJ48X jack... |
| 14:29.15 | mishehu | I'm not familiar with that spec. |
| 14:29.23 | [Latre] | Zeeek: YEAAA |
| 14:29.25 | [Latre] | works |
| 14:29.30 | [Latre] | weel.... |
| 14:29.36 | [Latre] | i did other thing |
| 14:29.36 | Ariel_ | mishehu, all the ones I have seen here work with a rj45 smart jack. |
| 14:29.58 | [Latre] | in setup of budgetone i change the DTMF |
| 14:29.59 | Ariel_ | [Latre], what other thing did you do? |
| 14:30.07 | Ariel_ | dtmf = info |
| 14:30.14 | [Latre] | =rfc2833 |
| 14:30.38 | Ariel_ | hummmm ok |
| 14:30.42 | Zeeek | [Latre] once one thing works with DTMF, be sure and try to check voicemail |
| 14:30.53 | mishehu | Ariel_: which reminds me, wwhat's diff about a smartjack from a regular rj45? |
| 14:30.54 | Zeeek | because sometimes it does't work |
| 14:31.32 | Ariel_ | mishehu, smartjack can detect if it's crossover or standard. |
| 14:33.10 | *** join/#asterisk Jearil (~Jearil@67.151.27.214) |
| 14:33.28 | file[laptop] | dtmfmode! |
| 14:33.34 | file[laptop] | yay RFC2833 |
| 14:33.43 | eKo1 | I wish all rj45 jacks were smart. |
| 14:34.07 | mishehu | Ariel_: heh. ok. wwell I'll clarify with them why they wrote rj48x instead of rj45 smart |
| 14:35.11 | Ariel_ | mishehu, http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm it's a smartjack. |
| 14:35.14 | [Latre] | Zeeek: works the transfer with only dial # + ext_num |
| 14:35.23 | InfraRed | it's a jackass |
| 14:35.29 | [Latre] | if i want listen voicemail i dial 1001 |
| 14:35.38 | *** join/#asterisk ecto (~ectospasm@70.150.230.2) |
| 14:35.56 | Zeeek | ok does it work? can you enter your pin? |
| 14:36.10 | ecto | I'm having no audio problems with a SIP phone behind a NAT. Asterisk is not behind a NAT. |
| 14:36.22 | Zeeek | changing DTMF sometimes has as a result voicemail doesn't hear tones |
| 14:36.36 | [Latre] | Zeeek: :S not work... |
| 14:36.51 | Zeeek | drums stop. Not good. |
| 14:37.25 | [Latre] | now works |
| 14:37.48 | [Latre] | if the user have dtmfmode=rcf.... and the sip phone too works!! |
| 14:37.52 | [Latre] | :) |
| 14:38.40 | Zeeek | [Latre] you may find that codec dependent |
| 14:38.57 | Zeeek | so many variables, GS firmware, codec... |
| 14:40.47 | [Latre] | rfc2833 did that my phone reconogized # and the numbers for voicemail.....you said that other thing maybe not works? like what? |
| 14:42.09 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 14:42.55 | *** part/#asterisk |HelioS| (ts07@ozashiki.com) |
| 14:46.02 | Zeeek | we'll see I gues |
| 14:49.50 | ecto | I've narrowed my problem down a bit... whenever I execute 'sip show peer <x>', it shows the external IP that needs to be used, but when I execute 'sip show channels' during a call it shows the internal IP address on the local LAN. The Asterisk server is on a different, remote lan, and one of its NICs is unfortunately set to the same network (192.168.0.x) |
| 14:50.11 | ecto | as the local LAN |
| 14:50.29 | *** part/#asterisk bernarde (~bernarde@bernarde.user) |
| 14:53.04 | shaZwaz | see ya later guys |
| 14:53.08 | Zeeek | bye |
| 14:53.13 | Ariel_ | ecto, double nat is always a problem. In the asterisk sip.conf you have extenip= and localnet= setup? |
| 14:53.15 | shaZwaz | bye Zeeek |
| 14:54.34 | *** join/#asterisk jmacz (~jmacz@63.245.86.234) |
| 14:55.01 | Ariel_ | ecto, externip = 68.142.xxx.xxx |
| 14:55.01 | Ariel_ | localnet = 192.168.10.0/255.255.255.0 |
| 14:55.05 | ecto | This is not double NAT. The Asterisk box is not behind a NAT, it has a real IP address. It's just connected to a NAT'd LAN as well as the wild Internet... |
| 14:55.26 | Ariel_ | what sip device is it? |
| 14:55.37 | ecto | This is a Sipura 2002 |
| 14:55.55 | Ariel_ | ok there is a setting in the sip section for external ip did you set this up? |
| 14:56.31 | Ariel_ | also make sure the sip.conf setting for the device has nat=yes |
| 14:57.02 | ecto | They're both set. I couldn't get a dial tone without nat=yes in sip.conf, but now there's no audio going either way. |
| 14:57.32 | ecto | But I can hear a dial tone and ringing if I call an internal asterisk extension |
| 14:58.04 | ecto | It will also ring external phones, but no audio is transferred then, either |
| 14:58.43 | ecto | my next try would be to set up a STUN server, but I don't like that option |
| 15:00.13 | ecto | I'm hoping that it might be fixed with a restart, since all I've done is reload sip... |
| 15:00.30 | rnovotny22 | Anyone know why Polycom ip500 series phones seem to disconnect from * ? |
| 15:01.59 | jontow | damned sudo vulnerabilities |
| 15:03.20 | Hmmhesays | anyone got a link to a decent boot cd that has ms chkdisk |
| 15:03.22 | *** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 15:06.14 | *** join/#asterisk trakk (~trakkasur@adsl-8-246-42.mia.bellsouth.net) |
| 15:06.28 | *** join/#asterisk vooduhal (~christoph@67.19.25.178) |
| 15:07.57 | *** join/#asterisk Caede (~Caede@204.94.248.81) |
| 15:08.05 | Ariel_ | rnovotny22, your the 2nd person that is asking are your running CVS head? |
| 15:08.08 | vooduhal | Can anyone suggest another card equivalent to the TDM400P that is compatible with asterisk? |
| 15:08.19 | Ariel_ | Hmmhesays, use the ultimate CD |
| 15:08.43 | Hmmhesays | where might this be? |
| 15:09.30 | Ariel_ | Hmmhesays, http://www.ultimatebootcd.com/ |
| 15:10.31 | rnovotny22 | Ariel: Just getting my first tast of * and having problems with it. CVS head? Running Ver 1.0.4 |
| 15:10.46 | Katty | hi. |
| 15:11.08 | Hmmhesays | ariel thanks |
| 15:11.18 | Caede | Okay, I have two Asterisk PBXes at two different locations, each connected via Digium T1 card to a legacy PBX and trunking calls in/out. Both Asterisk PBX's are behind identical Cisco PIX 515 Firewalls. Whenever a call comes in from outside the firewall via SIP and goes into one of the ZAP channels, almost exactly 10% of the calls get no audio. (90% are fine). Any thoughts on what might... |
| 15:11.20 | Caede | ...cause this? This happens with the same frequency at both sites. |
| 15:11.34 | Ariel_ | rnovotny22, there have been some posting about head in the pass few weeks. But 1.0.4??? sounds like an older stable? have you mixed them? |
| 15:11.57 | *** join/#asterisk Corydon-w (grey@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 15:12.48 | rnovotny22 | Ariel: I received the version on CD, (stable). I'm sure it's not the latest. How would you be able to mix them? |
| 15:13.23 | Caede | (also, I'm about 99% certain the ZAP channels aren't where the audio is being lost. It's getting lost on the SIP side, so I guess the ZAP stuff is irrelevant.) |
| 15:14.07 | Ariel_ | Sorry read your post incorrectly. But I would upgrade to the newer stable 1.0.8 |
| 15:14.16 | rnovotny22 | Between that and not being able to retreve VM/s I would be able to put the system in production. |
| 15:15.05 | vooduhal | Has anyone had any problems with the TDM400P Rev I cards? |
| 15:15.17 | vooduhal | We have 3 and all of them are having problems. |
| 15:15.40 | Ariel_ | vooduhal, I read something on the list. Digium did a patch to the persons zaptel drivers. |
| 15:15.51 | vooduhal | Dev list or user list? |
| 15:15.54 | Ariel_ | 3 in the same server? |
| 15:15.56 | rnovotny22 | Ariel: Where would I be able to get that version. I think I tried CVS about a month ago and was not able to get the files |
| 15:15.59 | DarthClue | vooduhal: what kind of problems? |
| 15:16.00 | Ariel_ | user list |
| 15:16.18 | Ariel_ | cvs co -r v1.0 |
| 15:16.38 | vooduhal | DarthClue: Serious echo and popping. We've got two Rev H cards that were working just fine in the exact same setup. |
| 15:17.06 | DarthClue | Check the list, if that fails, call support. |
| 15:17.53 | vooduhal | Stupid question. Is there a searchable web interface for the lists? |
| 15:18.33 | Ariel_ | vooduhal, I just did a lookup on my 3 boxes here and there all running TDM400 E/F not I |
| 15:18.51 | vooduhal | K. |
| 15:18.54 | Ariel_ | vooduhal, google |
| 15:19.29 | vooduhal | That works. |
| 15:26.03 | *** join/#asterisk vooduhal (~christoph@67.19.25.178) |
| 15:26.56 | *** join/#asterisk cmk (~cmk_@p54A3DA9F.dip.t-dialin.net) |
| 15:29.44 | *** join/#asterisk cmk (~cmk_@p54A3DA9F.dip.t-dialin.net) |
| 15:33.10 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
| 15:33.28 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:36.47 | *** join/#asterisk Thus0 (~Thus0@59.111.102-84.rev.gaoland.net) |
| 15:38.01 | drumkilla | anyone use one of the pa168 phones? |
| 15:38.27 | drumkilla | I have one sitting on my desk, but it does not appear to have a place to enter server/auth info |
| 15:39.02 | Katty | maybe it's really just a paperweight in disguise |
| 15:39.18 | drumkilla | that's what I'm thinking |
| 15:39.53 | Cresl1n | hrm.... |
| 15:40.32 | robl^ | well the GS BT101 isn't even a paperweight. I turned ont ehe ceiling fan and that phone went flying |
| 15:41.24 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 15:41.37 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 15:41.50 | drumkilla | what a pos ... |
| 15:42.13 | Cresl1n | lol, I feel stupid |
| 15:42.21 | outtolunc | its friday.. you can say it <G> |
| 15:43.09 | Faithful | anyone know of a nice server management interface other than webmin? |
| 15:43.15 | vooduhal | Does anyone have any experience with the different echo cancellation algorithms offered? ie STEVE, STEVE2, MARK, MARK2, MARK3? |
| 15:43.34 | *** join/#asterisk stkn (~stkn@stkn-active-pdpc.developer.gentoo) |
| 15:43.38 | vooduhal | Or more accurately which algorithm is best for which situations? |
| 15:45.06 | outtolunc | faithful, wiki for 'asterisk gui' |
| 15:45.44 | Ariel_ | vooduhal, call digium. |
| 15:45.47 | outtolunc | unless you meant in 'general' |
| 15:48.27 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
| 15:49.24 | *** part/#asterisk Caede (~Caede@204.94.248.81) |
| 15:49.27 | *** join/#asterisk my007ms (~my007ms@217.139.236.110) |
| 15:49.36 | my007ms | hello all |
| 15:50.00 | my007ms | i have problem with TDM31B can some one help |
| 15:50.11 | drumkilla | file[laptop]: !@)(@!#!!!!!!!!!! |
| 15:50.29 | file[laptop] | what'cha doing |
| 15:50.56 | my007ms | i try every thing to config this card but i can not |
| 15:50.56 | file[laptop] | haha |
| 15:51.06 | DarthClue | Now get to work boi! |
| 15:51.26 | my007ms | it's light for some time and soon off |
| 15:51.37 | *** join/#asterisk vuvie (~S@ad202.166.2.3.magix.com.sg) |
| 15:51.38 | file[laptop] | nooooooooo |
| 15:52.14 | drumkilla | file[laptop]: what are you going to do with your life?!!? |
| 15:52.20 | drumkilla | step 1) move to huntsville |
| 15:52.26 | file[laptop] | ooh |
| 15:53.16 | file[laptop] | that's a great idea |
| 15:54.07 | eKo1 | step 2) ?? |
| 15:54.12 | eKo1 | step 3) Profit |
| 15:54.30 | JerJer | no its 1) sell T-1 boards 2) ??? 3) profit |
| 15:55.04 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 15:58.26 | mogorman | but you have to do that in huntsville |
| 15:58.34 | my007ms | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 15:58.55 | *** part/#asterisk vuvie (~S@ad202.166.2.3.magix.com.sg) |
| 15:59.02 | my007ms | can someone help me to solve this i get this error all over |
| 15:59.10 | my007ms | even i try asterisk@home |
| 15:59.33 | *** join/#asterisk _theend_ (~DrEaM@80.18.184.226) |
| 16:00.12 | rnovotny22 | Anyone know what the commad to retreive VM is? |
| 16:00.25 | file[laptop] | voicemailmain? |
| 16:00.42 | Katty | assign an extension to voicemail? (= |
| 16:00.54 | rnovotny22 | voicemainmain lets you leave vm not listen to. |
| 16:01.10 | file[laptop] | wrong |
| 16:01.21 | file[laptop] | voicemail is for leaving messages, voicemailmain is for retrieval |
| 16:01.43 | Katty | exten => 9999,1,VoiceMailMain(${CALLERIDNUM}@copi-rite) <- that's mine |
| 16:01.57 | Katty | then it's all PASSWORD!!111 |
| 16:02.05 | Katty | etc |
| 16:02.09 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 16:03.03 | rnovotny22 | I have 300,1,voicemainmain(${EXTEN}) says "Please leave a message." |
| 16:03.36 | *** join/#asterisk scud (~scud@12-214-190-139.client.mchsi.com) |
| 16:03.52 | file[laptop] | no you don't, because that command doesn't even exist |
| 16:04.05 | file[laptop] | voicemailmain |
| 16:04.11 | file[laptop] | just use that without any arguments |
| 16:04.16 | file[laptop] | ie: exten => *99,1,Voicemailmain |
| 16:04.21 | file[laptop] | and see if it works :P |
| 16:04.50 | rnovotny22 | Tried that . . .. same thing |
| 16:06.15 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 16:06.25 | file[laptop] | then you're doing something wrong, because voicemailmain is the application that is used to access voicemail for pickup |
| 16:07.08 | my007ms | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 16:07.15 | file[laptop] | type "show application voicemailmain" on the asterisk CLI if you don't believe me :p |
| 16:07.16 | my007ms | Help |
| 16:07.19 | *** part/#asterisk scud (~scud@12-214-190-139.client.mchsi.com) |
| 16:07.25 | rnovotny22 | OK, looking on the console. * is calling VoiceMail not VoiceMailMain. |
| 16:07.35 | file[laptop] | rnovotny22: there you go, configuration issue |
| 16:07.59 | wizhippo | No such device or address... are you using a zap channel? |
| 16:08.56 | my007ms | how can i solve this |
| 16:09.06 | dalabera | hello guys, quick question I have a T400P, 1 span -> ChannelBank, 2-4-> T1's, Bellsouth tells me that they supply me syncronization, does that means that the field value should be zero on all my spans? |
| 16:09.43 | my007ms | wizhippo how can i sove this probelm |
| 16:10.10 | my007ms | it's work once and after reboot it's come agian and i have to start all over |
| 16:10.45 | my007ms | now it's become ZT_CHANCONFIG failed on channel 1: Invalid argument (22)Did you forget that FXS interfaces are configured with FXO signallingand that FXO interfaces use FXS signalling? |
| 16:10.47 | wizhippo | let us know more, do you have zaptel hardware? Are you kernel modules loading? etc... |
| 16:10.58 | my007ms | TDM31B |
| 16:11.29 | my007ms | zaptel 178560 4 [ztdummy] |
| 16:11.29 | rnovotny22 | file[laptop]: Thanks 300 was set twice in extensions !! |
| 16:12.06 | wizhippo | do you have wctdm loaded? |
| 16:12.37 | my007ms | no |
| 16:13.14 | wizhippo | if i recall that's the drive for the tdm card |
| 16:13.54 | my007ms | when i try it's give me |
| 16:14.10 | *** part/#asterisk santiago (~santiago@63.245.86.198) |
| 16:14.21 | my007ms | error |
| 16:15.41 | my007ms | "" /lib/modules/2.4.21-27.0.1.EL/misc/wcfxs.o: init_module: No such deviceHint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg/lib/modules/2.4.21-27.0.1.EL/misc/wcfxs.o: insmod /lib/modules/2.4.21-27.0.1.EL/misc/wcfxs.o failed/lib/modules/2.4.21-27.0.1.EL/misc/wcfxs.o: insmod wctdm failed "" |
| 16:16.39 | my007ms | did i have to change place of PCI |
| 16:17.06 | *** join/#asterisk jake1932 (~jpersofsk@pool-70-16-129-221.phil.east.verizon.net) |
| 16:17.41 | wizhippo | it looks like it's not finding your card. i would double check to make sure it is seated properly |
| 16:19.00 | my007ms | in my cat /proc/pci what i have to see to be sure it's work ??? |
| 16:20.19 | wizhippo | Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (rev 0). is what show on mone |
| 16:20.58 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
| 16:21.18 | my007ms | i get msg in first say that this card is remmove |
| 16:21.23 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 16:21.57 | my007ms | if i change place of card do i have to reinstall asterisk |
| 16:22.04 | wrmem | a |
| 16:22.09 | wizhippo | no |
| 16:22.29 | my007ms | do i have to change any thing |
| 16:22.55 | wizhippo | i had that problem with fedora, every time i reboot the machine is would show device remove than added. I switched to gentoo |
| 16:23.15 | my007ms | and it's work |
| 16:23.24 | my007ms | i used FC4 |
| 16:23.25 | wizhippo | every reboot |
| 16:23.55 | tzanger | ;exten => 2915112,1,GotoIfTime(11:30-17:00|*|24|June?callcell,s,1) |
| 16:24.05 | tzanger | what the hell is wrong with that line?? it says "June" is an invalid month |
| 16:24.13 | tzanger | the voip-info page is specifiying "May" so I figured that was right |
| 16:24.17 | tzanger | "6" is also incorrect |
| 16:24.24 | wizhippo | try jun |
| 16:24.26 | tzanger | afternoon katty |
| 16:24.30 | Katty | hi (= |
| 16:24.40 | ecto | d'oh, my problem was simple to fix. externip in the [general] section of sip.conf was set to the wrong IP address...... |
| 16:24.44 | my007ms | so for now i need to change card place do i have change some thing |
| 16:24.53 | Hmmhesays | sumnab1tch |
| 16:25.30 | wizhippo | no, just make sure on reboot it detects the hardware, kudzu or what ever it's called should show up saying there is a change |
| 16:25.32 | Hmmhesays | I wish customers would shut up and let me do my job |
| 16:25.50 | tzanger | wizhippo: that worked, how dumb is that.. thanks :-) |
| 16:26.45 | af_ | I am having a strange situation with a quadbri, and direct inward thing |
| 16:26.50 | af_ | basically I hav 1000 dids available, but some call dial 333444111, but in to * results 333444 only (the root) |
| 16:26.56 | af_ | this seems to be ramdom.... |
| 16:27.01 | af_ | any idea? |
| 16:29.47 | *** join/#asterisk ^HeLL^ (~admin@85.137.127.143) |
| 16:29.56 | *** join/#asterisk Exstatica (exstatica@65.119.22.200) |
| 16:29.59 | ^HeLL^ | hello all |
| 16:31.49 | wizhippo | Hell-o all that's funny. |
| 16:31.50 | sivana | has anyone worked with TDMoE? |
| 16:33.52 | my007ms | i try |
| 16:35.39 | *** join/#asterisk cmk (~cmk_@p54A3DA9F.dip.t-dialin.net) |
| 16:37.04 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 16:37.41 | my007ms | y it's not light like it was ??? |
| 16:37.55 | my007ms | what light mean ? |
| 16:38.15 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 16:38.23 | ManxPower | I can't think of a reason I would actually need TDMoE |
| 16:38.34 | my007ms | i was see 4 port lighting now no port |
| 16:39.01 | my007ms | pleas some one help for all this |
| 16:39.39 | sivana | ManxPower: I have a second * box for prepaid calling cards that I would like to use the PRI from the 1st box |
| 16:40.02 | DarthClue | my007ms: are your modules loaded? (zaptel and wct4xxp) |
| 16:41.21 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 16:42.38 | my007ms | no one of them |
| 16:42.39 | sivana | I guess I could just use a T100P :) |
| 16:42.47 | my007ms | zaptel |
| 16:42.49 | sivana | between the two |
| 16:43.39 | my007ms | so i have to lode wct4xxp |
| 16:43.49 | my007ms | i was think i lode wctdm |
| 16:44.13 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 16:44.15 | my007ms | zaptel 178560 4 [ztdummy] |
| 16:45.07 | my007ms | y i get ZT_CHANCONFIG failed on channel 1: No such device or address (6) all the time |
| 16:45.26 | shido | yeah |
| 16:45.28 | shido | my007ms, |
| 16:45.31 | shido | that makes sense |
| 16:45.34 | shido | take out spans |
| 16:45.35 | shido | or fxs |
| 16:45.39 | shido | out of zapte.conf |
| 16:45.42 | shido | zaptel.conf |
| 16:45.47 | shido | then run ztcfg -vv |
| 16:45.47 | dalabera | hello guys, quick question I have a T400P, 1 span -> ChannelBank, 2-4-> T1's, Bellsouth tells me that they supply me syncronization, does that means that the field value should be zero on all my spans? |
| 16:45.49 | shido | so you have 0 channels |
| 16:45.56 | shido | then you're good to go my007ms |
| 16:46.26 | my007ms | Zaptel Configuration======================Channel map:Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXO Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)4 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 16:47.00 | my007ms | and i just have fxoks=1-3 fxsks=4 |
| 16:47.06 | my007ms | like doc say |
| 16:47.08 | shido | errrrj |
| 16:47.19 | shido | so you DO have fxo cards? |
| 16:47.34 | my007ms | TDM31B |
| 16:47.39 | shido | then |
| 16:47.40 | shido | why |
| 16:47.41 | shido | on earth |
| 16:47.44 | shido | do you need ztdummy? |
| 16:48.12 | my007ms | i don't need it's this is out my lsmod |
| 16:48.27 | shido | then rmmod ztdummy you dont need it |
| 16:48.27 | *** join/#asterisk Morex (~blah@host81-157-226-21.range81-157.btcentralplus.com) |
| 16:48.32 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 16:48.37 | Morex | Evening all. |
| 16:49.21 | my007ms | how do that? |
| 16:49.33 | shido | rmmod ztdummy |
| 16:50.21 | my007ms | then |
| 16:51.02 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 16:51.02 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:54.50 | my007ms | any one how can i uninstall asterisk and zaptel and all this i need to start agian |
| 16:55.12 | eKo1 | rm -rf / |
| 16:55.32 | crash3m | make cluebat |
| 16:55.32 | shido | hehehe |
| 16:56.05 | my007ms | :D |
| 16:56.23 | my007ms | is there is uninstall script some where |
| 16:56.30 | shido | right... |
| 16:56.41 | eKo1 | Cluesaber... |
| 16:56.54 | eKo1 | my007ms: no there isn't. |
| 16:57.01 | my007ms | :( it's give me line 151: Unable to open master device '/dev/zap/ctl' |
| 16:57.18 | DarthClue | my007ms: what OS are you using? |
| 16:57.39 | my007ms | FC4 |
| 16:58.40 | my007ms | do u know one worke well nd u try it |
| 16:58.52 | *** part/#asterisk jaike (~a@203.131.137.76) |
| 17:01.23 | MichaelSaunders | where does the latest cvs aterisk look for its sound file |
| 17:04.32 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 17:06.46 | sivana | MichaelSaunders: /usr/lib/asterisk/sounds? |
| 17:07.11 | sivana | MichaelSaunders: oops /var/lib/asterisk/sounds? |
| 17:07.48 | robl^ | MichaelSaunders, depends where you define $ASTVARLIBDIR in Makefile. default is /var/lib/asterisk/sounds |
| 17:07.59 | dalabera | my007ms read /usr/src/zaptel/REAMD.udev to solve your problem |
| 17:08.07 | *** join/#asterisk brettnem (~brettnem@207.90.232.34) |
| 17:09.52 | MichaelSaunders | robl^: I think my problem might be that chan_h323 does not support codec conversion |
| 17:10.37 | robl^ | MichaelSaunders, I am not sure. I have never used H323. I can't speak on that |
| 17:12.55 | *** join/#asterisk wolfson (~hehe@68-187-187-034.dhcp.mant.nc.charter.com) |
| 17:14.46 | roamer323 | did sipphone just died? no proxy, no forums, no support link working :-( |
| 17:15.25 | Nivex | roamer323: I was using it last night. |
| 17:15.45 | *** join/#asterisk robbins (~robbins@samson1.500.megarouting.com) |
| 17:15.53 | Nivex | hmm, so it would seem. They seem to do that once in awhile. |
| 17:16.21 | robbins | hey, is there a way to *push* a message to a 7940/60 set via SIP or alternate method? |
| 17:16.36 | *** join/#asterisk miguel3239 (~myoung@h-68-167-124-170.cmbrmaor.covad.net) |
| 17:17.11 | robbins | don't everyone answer at once |
| 17:17.26 | jake1932 | only gave em a minute |
| 17:19.01 | miguel3239 | Hi... I have two te110p cards right now. We are going to be purchasing a server with the hopes of eventually purchasing the TE411p card. I know that the TE411p needs 3.3v and that the TE405p needs 5v. My question is, does it matter which PCI slot the TE110p is put into? Will it work on 3.3v pci slots? |
| 17:20.01 | jake1932 | http://lists.digium.com/pipermail/asterisk-users/2005-April/103197.html |
| 17:20.52 | jake1932 | miguel - that was for you |
| 17:20.54 | robbins | second question, when passing a call via IAX, the destination servers gets a busy, but instead of returning busy, it goes to the next rule in the dialplan.. is there an error to capture and send back congestion? |
| 17:21.41 | miguel3239 | jake1932: thank you very much... That was quick! |
| 17:21.45 | robbins | so the local server doesn't attempt a secondary IAX call to another server, when the first server returns a busy |
| 17:21.47 | jake1932 | there's a DIALSTATUS - i believe |
| 17:22.04 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:22.27 | jake1932 | robbins - take a look here http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 17:24.45 | jake1932 | that in combination with GotoIf should work - but haven't used DIALSTATUS yet - so don't know if it works |
| 17:25.30 | shido | DIALSTATUS works |
| 17:25.46 | jake1932 | can anyone recommend a single port ISDN card? |
| 17:25.51 | jake1932 | for voice? |
| 17:27.31 | pawalls | How would you guys recommend implementing "observing" where someone can pick a line to "observe" and not be heard? |
| 17:27.53 | pawalls | We use this for training purposes, where a new technical support person will observe an experienced employee to see how things work. |
| 17:28.05 | tzafrir_laptop | pawalls, monitoring? |
| 17:28.06 | jake1932 | i think there's a zapmonitor cmd |
| 17:28.08 | shido | as long as you mention that when ppl call in |
| 17:28.34 | shido | unless your a service provider |
| 17:28.38 | jake1932 | ZapBarge |
| 17:28.41 | shido | and you have a court order |
| 17:28.48 | pawalls | tzafrir_laptop, Yeah.. but on a temporary basis.. like I want to be like.. "Okay, I'm going to observe the person using extension 430" |
| 17:28.59 | pawalls | And any time 430 initiates a call, I hear what they are saying/doing. |
| 17:29.12 | shido | wife monitor works |
| 17:29.18 | shido | you can record calls |
| 17:29.19 | shido | as well |
| 17:29.36 | jake1932 | you should probably have an automated recording saying "Calls may be recorded.. blah blah |
| 17:30.36 | jake1932 | or ChanSpy |
| 17:31.13 | jontow | question: is that law in effect if you don't *record* the call you're listening to? |
| 17:31.28 | *** part/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
| 17:31.39 | shido | yes |
| 17:31.53 | jake1932 | pretty sure all you need to do is inform the person you're speaking with that their call may be recorded or monitored |
| 17:32.13 | jake1932 | quite common in callcenters |
| 17:32.50 | brettnem | actually |
| 17:32.58 | jontow | aha. |
| 17:33.02 | brettnem | it depends on where you are making the call and where you are making the call to |
| 17:33.04 | jontow | yeah, this callcenter has a notice up |
| 17:33.41 | brettnem | here in Texas.. only one party has to know about the recording.. and it's perfectly valid if YOU (the recorder) is the only one who knows.. in other words, you can't record a call where no one knows on either end. |
| 17:34.11 | jontow | aha |
| 17:34.15 | jontow | thats kinda cool ;) |
| 17:34.18 | jontow | in NY, i have no idea |
| 17:34.26 | shido | http://www.askcalea.net/calea.html |
| 17:34.28 | jake1932 | right - but there's no downside to letting them know - uless your a PI |
| 17:34.29 | jontow | but i do remember all over the place hearing "this call may be monitored for quality assurance purposes" |
| 17:34.35 | jontow | meaning "to cover our ass, we have you on tape." |
| 17:34.49 | brettnem | but in general, just remember to be smart and do what's right.. don't record calls you don't have any business recording.. |
| 17:35.07 | jontow | i don't want to record any.. just wanted to get an opinion or two ;) |
| 17:35.26 | brettnem | jontow: who knows what that's for.. remember, calls that CROSS STATE boundries are under FCC jurisdiction and not local PUC. |
| 17:36.01 | brettnem | However, I think by FCC definition, VoIP calls are local in nature. |
| 17:36.18 | brettnem | lots of funny games.. no one really knows what the heck is going on. |
| 17:36.40 | brettnem | shido: CALEA is a little bit different.. the law applies differently to law enforcement. |
| 17:36.45 | shido | you will when you get subpoena;s |
| 17:36.47 | shido | =-d |
| 17:36.59 | brettnem | right |
| 17:37.09 | pawalls | jake1932, We have a recording like that already. |
| 17:37.28 | brettnem | btw, just because you say the recording doesn't mean you legally can. ;) |
| 17:37.33 | pawalls | I just need to know how to do it with asterisk. |
| 17:37.44 | brettnem | I'd start with the playback command. :) |
| 17:37.59 | pawalls | Well, we aren't recording, we just have an extra person on the line. |
| 17:38.15 | pawalls | For training. |
| 17:38.18 | InfraRed | what a waste |
| 17:38.19 | brettnem | ah |
| 17:38.25 | InfraRed | :) |
| 17:38.31 | jake1932 | pawalls - the cmds i gave you don't work? |
| 17:38.45 | pawalls | jake1932, zapmonitor ? |
| 17:38.52 | pawalls | I need to be able to do it through the voip phones. |
| 17:38.52 | jake1932 | chanspy |
| 17:39.40 | jake1932 | ChanSpy - This adds the ability to spy on any bridged call, this includes VoIP only calls |
| 17:40.10 | Ariel_ | I have a customer that did the same thing via meetme rooms. They would send the calls to room and the trainer would go into the room via his setting as muted. |
| 17:41.01 | pawalls | jake1932, I'll look at that.. is it in the wiki? |
| 17:41.07 | jake1932 | yep |
| 17:41.10 | pawalls | Ariel_, I considered that as an option if I couldn't think of anything else. |
| 17:45.39 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
| 17:49.36 | pawalls | jake1932, Ah.. it's CVS. |
| 17:49.47 | Katty | hmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm |
| 17:49.57 | Katty | there's a way to set a particular ringtone for a particular contact on a polycom 500 |
| 17:50.05 | Katty | but..i'm not sure how to specify |
| 17:50.10 | Katty | there's not a list to pick from |
| 17:50.13 | Katty | just a thing to type in |
| 17:50.38 | Katty | :< |
| 17:50.49 | Katty | i know! |
| 17:50.53 | Katty | i'll rtfm!! |
| 17:51.50 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 17:53.27 | *** join/#asterisk x86 (~x86@x86.developer.poipu) |
| 17:54.08 | x86 | hey guys, is it possible to make my existing asterisk box also be an H.323 gatekeeper? |
| 17:54.27 | x86 | is that something asterisk is capable of supporting? |
| 17:54.37 | shido | not asterisk |
| 17:54.42 | shido | asterisk can be a gateway |
| 17:54.47 | shido | gnugk is a gatekeeper |
| 17:54.48 | shido | you can run |
| 17:54.49 | x86 | right i knew that... |
| 17:54.52 | x86 | ok |
| 17:55.17 | brettnem | hey, anyone want to help me with some SER stuff?? :) |
| 17:55.21 | x86 | i knew there was stuff out there that could do it, but if at all possible i was hoping asterisk could do it, so it'd be an all-inclusive service |
| 17:55.32 | shido | it still can |
| 17:55.40 | shido | your customers doesnt need to know the parts |
| 17:55.42 | shido | that make the whole |
| 17:55.46 | x86 | well sure |
| 17:55.58 | shido | that came out wrong |
| 17:56.02 | x86 | im talking from a serviceability standpoint |
| 17:56.06 | shido | your customers do not need to know the parts that make the whole |
| 17:56.28 | x86 | i'd like it to all be running from a single service you know? |
| 17:56.48 | x86 | but if not possible, that's totally fine, i have no problem using gnugk or some other gatekeeper service |
| 17:56.48 | shido | well thats not going to happen today |
| 17:56.54 | x86 | ok thats fine |
| 17:56.58 | shido | :) |
| 17:57.09 | shido | brettnem, whats up? |
| 17:57.20 | shido | u liek da SER veddy much , yes? |
| 17:57.24 | x86 | does gnugk need to / able to talk dirty to asterisk? |
| 17:57.27 | brettnem | shido: yess.. yes I do |
| 17:57.40 | shido | it can talk dirty to chan_h323.so the nufone h323 channel driver |
| 17:57.56 | shido | I have a biased opinion tho |
| 17:58.18 | brettnem | shido: I'm having some trouble getting BYEs to route properly.. I'm using stateful transactions and exec_dset to route my calls.. right now I have a perl script in the exec_dset and it runs in INVITES AND BYEs which Idon't think it should.. |
| 17:58.25 | shido | my bills get paid every time I mention NuFone |
| 17:58.39 | DarthClue | ~cluecon |
| 17:58.39 | jbot | somebody said cluecon was http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 17:59.25 | Ariel_ | I see our multi-daily call for cluecon is back... |
| 17:59.33 | x86 | shido: hehe |
| 17:59.34 | brettnem | shido: If I don't run my routing script on BYEs, they just die when they hit my ser server.. why is that?? any thoughts? |
| 17:59.38 | x86 | shido: what is nufone? |
| 18:00.03 | shido | NuFone is one of the first IAX providers available for Asterisk users |
| 18:00.10 | Ariel_ | ~lark DarthClue |
| 18:00.12 | shido | Up and running for the past 3 yrs |
| 18:00.16 | shido | www.nufone.net |
| 18:00.20 | shido | ~nufone |
| 18:00.20 | jbot | nufone is, like, Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 18:00.36 | x86 | shido: doesnt asterisk do IAX (IAX2 even) internally already? |
| 18:00.45 | shido | yes it does |
| 18:00.55 | ender | x86: think upstream provider |
| 18:00.56 | brettnem | shido: any ideas? |
| 18:01.00 | shido | NuFone allows you to make calls to the PSTN |
| 18:01.01 | x86 | oh! you mean like a PTSN provider? |
| 18:01.08 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 18:01.08 | shido | or receive calls from the PSTN |
| 18:01.11 | x86 | yeah, thats what i meant ;) |
| 18:01.14 | x86 | very cool |
| 18:01.14 | shido | 800's , Michgian numbers |
| 18:01.17 | shido | etc |
| 18:01.22 | x86 | right |
| 18:01.24 | x86 | cool cool :) |
| 18:01.28 | x86 | i'll check it out |
| 18:02.25 | x86 | the webpage is super slow :( |
| 18:02.45 | x86 | oh duh user error ;) |
| 18:02.51 | x86 | was doing nufone.com not .net haha |
| 18:03.41 | pawalls | Hmm.. ChanSpy can't "spy" on an extension instead of an agent? |
| 18:07.35 | JunK-Y | pawalls: sure u can spy on exten |
| 18:11.37 | pawalls | JunK-Y, Syntax is the same? ChanSpy(exten) ? |
| 18:13.01 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 18:15.26 | *** join/#asterisk Guntanis (~lalal@200.91.95.60) |
| 18:16.59 | Guntanis | hey hey, does anyone knows about a good predictive dialer that works with asterisk@home |
| 18:17.37 | Romik_ | Guntanis:it only work with asterisk@office |
| 18:18.14 | Guntanis | LOL |
| 18:19.05 | Ariel_ | Guntanis, asterisk @ Home is asterisk plus AMP and other things built in. But it's a normal stable asterisk and you can upgrade and change it any way you want. |
| 18:19.36 | *** join/#asterisk CyBerRaT (~nn@64.235.216.34) |
| 18:19.41 | Guntanis | Ariel_: thanks... do you have any predictive dialer suggestion? |
| 18:20.11 | Hmmhesays | telemarketer. |
| 18:20.26 | Ariel_ | Sorry don't work in that area....actually don't support telemarkers.... |
| 18:20.49 | Hmmhesays | predictive dialer is a stupid description too |
| 18:20.59 | CyBerRaT | it's possible to setup asterix with standard serial modem ? |
| 18:21.06 | file[laptop] | CyBerRaT: no. |
| 18:21.08 | Hmmhesays | it makes the thing sound like more than it is |
| 18:21.17 | Hmmhesays | my telephone plus my finger is a predictive dialer |
| 18:21.23 | Guntanis | LOL |
| 18:21.49 | Guntanis | yeah... it should be auto dialer or something like that |
| 18:21.50 | Hmmhesays | call it what it is.... an telephony instrument of dinner time torment |
| 18:22.28 | Hmmhesays | or TIODTT |
| 18:30.35 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 18:33.59 | dros7 | does anyone know if it is possible to use Queue() with confirmation/acknowlegdement (similar to AgentCallBackLogin)? |
| 18:36.26 | *** join/#asterisk alvis2k5 (Alvis@200.105.128.59) |
| 18:36.33 | alvis2k5 | Hello |
| 18:36.52 | alvis2k5 | somebody can help me, with H.323 configuration |
| 18:37.26 | shido | fine |
| 18:37.31 | shido | whats up alvis2k5 |
| 18:37.54 | shido | confirmation/ack , dros7 ? |
| 18:38.50 | *** join/#asterisk khemir (user119@201.133.129.105) |
| 18:38.57 | alvis2k5 | shido: when a call get in, the parameter " Calling party name" doesmt match because is [] |
| 18:39.13 | shido | what are you using? |
| 18:39.22 | alvis2k5 | and the calls dont complete |
| 18:39.28 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
| 18:39.50 | alvis2k5 | (The NuFone Network's Open H.323 Channel Driver |
| 18:40.20 | shido | sweet |
| 18:40.25 | sangee | please help me, my T1 line take inbound call only, I can not make outbound call, do i need to change anything in zaptel.conf? |
| 18:40.41 | *** join/#asterisk funxion (~funxion@mtnuser.icgws.com) |
| 18:41.08 | Ariel_ | sangee, how did you configure your channels and your dialing rules? |
| 18:41.26 | Ariel_ | sangee, what is the error your getting when you dial out. |
| 18:41.44 | sangee | hold on i will give the error |
| 18:42.35 | shido | use pastebin.ca for your .confs |
| 18:43.51 | sangee | i got this error "dial_exec: Unable to create channel of type 'Zap' (cause 0) |
| 18:43.52 | sangee | <PROTECTED> |
| 18:43.59 | *** join/#asterisk ArkyLady (ArkyLady@adsl-66-142-125-19.dsl.ltrkar.swbell.net) |
| 18:44.08 | Exstatica | is there a way to connect to a sip peer with no username only with a password? |
| 18:44.29 | tzafrir_laptop | Is there a simple way to get (at run time) the list of countries with zone data in libtonezone? |
| 18:44.52 | Ariel_ | sangee, use pastebin.ca and put your zaptel.conf, zapata.conf and your dial rule to dial out. |
| 18:45.11 | tzafrir_laptop | Exstatica, the "username" is the name of the peer |
| 18:45.21 | tzafrir_laptop | why do you need that? a gues user? |
| 18:45.29 | sangee | what is pastebin.ca? |
| 18:45.32 | *** join/#asterisk CoolWhip (~CoolWhip@66.236.68.169.ptr.us.xo.net) |
| 18:45.42 | tzafrir_laptop | ~pastebin |
| 18:45.42 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 18:46.19 | jake1932 | just got a TDM01B - using exten => _NXXXXXXXXX,1,Dial(Zap/1/1${EXTEN}) and get this message: Jun 24 14:38:36 NOTICE[1345]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
| 18:46.32 | jake1932 | http://pastebin.com/303544 |
| 18:46.41 | sangee | i will try, thx |
| 18:46.42 | tzafrir_laptop | ${EXTEN:1} |
| 18:46.49 | tzafrir_laptop | ? |
| 18:47.10 | jake1932 | i'll try |
| 18:47.21 | CoolWhip | Sorry for lame Q but the digium site is a bit unclear and I don't use CVS.... |
| 18:47.26 | tzafrir_laptop | jake1932, but it shouldn't be relevant |
| 18:47.33 | CyBerRaT | I looking for the best Gui web to manage asterisk |
| 18:47.38 | CoolWhip | will cvs checkout -r v1-0 zaptel asterisk update me to 1.0.8? |
| 18:47.46 | jake1932 | same thing |
| 18:48.02 | tzafrir_laptop | CoolWhip, or download the tarball from their ftp |
| 18:48.05 | shido | missing a / |
| 18:48.08 | CoolWhip | I already have 1.0.7 and need to update it |
| 18:48.08 | shido | jake1932, |
| 18:48.13 | shido | jake1932, you are missing a "/" |
| 18:48.29 | tzafrir_laptop | Working on 1.0.8 debs as we speak, BTW |
| 18:48.56 | CoolWhip | tzafrir_laptop: Is that the best way to go? |
| 18:49.17 | tzafrir_laptop | If you use debian sarge: definetly debs, IMO |
| 18:49.20 | x86 | who'se working on the gentoo port of 1.0.8? :P |
| 18:51.12 | *** join/#asterisk Tili (~Tili@202-133-67-105-dialup.sat.net.pk) |
| 18:53.28 | Micc | Who wrote the new voicemail system? I'm wonder how someones passcode could get changed to space |
| 18:54.31 | Ariel_ | jake1932, you have an fxo board and your using fxoks you should be using fxsks instead. |
| 18:55.30 | jake1932 | tnx ariel - i'll try that |
| 18:56.38 | CyBerRaT | I looking for the best Gui web to manage asterisk |
| 18:57.17 | pawalls | Ahhh! |
| 18:57.47 | pawalls | ChanSpy(SIP) -> 400# will spy on SIP/400... gotcha. |
| 18:58.21 | Ariel_ | CyBerRaT, that is a matter of choice and look and use. I am getting used to AMP. |
| 18:59.27 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
| 19:00.58 | *** join/#asterisk samwong (~keung@cm61-18-216-102.hkcable.com.hk) |
| 19:01.43 | sangee | One more question, If i change the zaptel.conf, to affect the changes, do i need to reload the zatel driver? or reload the asterisk on terminal mode? |
| 19:01.49 | pifiu | got my cisco 7940g and my polycom 501! |
| 19:02.04 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 19:03.13 | Ariel_ | pifiu, great to hear so which do you like better. |
| 19:07.01 | tzafrir_laptop | sangee, run ztcfg |
| 19:07.03 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 19:07.14 | sangee | thx |
| 19:07.23 | tzafrir_laptop | this is the program using this module. A kernel module can't read a file, normally |
| 19:07.35 | tzafrir_laptop | ~genzaptelconf |
| 19:07.35 | jbot | somebody said genzaptelconf was a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 19:08.34 | *** join/#asterisk cmk (~cmk_@p54A3DA9F.dip.t-dialin.net) |
| 19:09.13 | Tili | tzarfir_laptop: isn't ztcfg for verifying configuration. I dont think it has anything to do with asterisk's loading of zaptel chan |
| 19:10.04 | pifiu | theres actually web based control panels for asterisk, other than a@h? |
| 19:10.11 | shido | Tili, that is incorrect |
| 19:10.22 | shido | ztcfg -vv refreshes the zaptel drivers if you change zaptel.conf |
| 19:10.24 | Tili | oh |
| 19:10.31 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 19:10.32 | shido | try it, you'll see |
| 19:10.34 | shido | :) |
| 19:10.36 | Tili | ok |
| 19:10.53 | Lee__ | pifiu: yes, try AMP. |
| 19:11.15 | jake1932 | AMP comes with *@home |
| 19:12.11 | *** join/#asterisk fugitivo (~ajf@201.255.104.124) |
| 19:12.13 | fugitivo | hello |
| 19:12.21 | pifiu | what if i want to try the normal asterisk but with a web based interface? |
| 19:12.38 | jake1932 | then load asterisk then load AMP |
| 19:13.00 | ender | where is AMP located? |
| 19:13.56 | Lee__ | amportal.sf.net |
| 19:13.56 | jake1932 | ender - http://sourceforge.net/projects/amportal/ |
| 19:13.57 | wunderkin | i dont know if i'd call it suffer |
| 19:13.58 | Ariel_ | ender, #amportal |
| 19:14.34 | Lee__ | *@home is cheezy anyway |
| 19:14.43 | ender | wow. And I thought I was a dork... |
| 19:15.12 | fugitivo | why do people use web interfaces and that userfriendly distributions for asterisk? |
| 19:15.20 | fugitivo | asterisk conf files are easy to setup |
| 19:15.41 | ender | fugitivo: PHBs here want to see something pointy clicky. *shrug* |
| 19:16.01 | fugitivo | where is the fun then? |
| 19:16.10 | ender | oh we're not going to use it. |
| 19:16.16 | ender | we're just going to show them it. (: |
| 19:16.23 | Lee__ | I don't know why people use these new fangled "email clients" when mail works perfectly!?! |
| 19:16.30 | Lee__ | they must all suck. |
| 19:17.27 | PBXtech | can digium put this 2nd gen firmware on our existing cards? |
| 19:17.29 | *** join/#asterisk Thus0 (~Thus0@59.111.102-84.rev.gaoland.net) |
| 19:17.50 | JerJer | PBXtech: no, there will be some sort of upgrade program |
| 19:17.56 | DarthClue | PBXtech: what is the problem? |
| 19:17.56 | *** join/#asterisk szi (szi@switch.5ess.net) |
| 19:17.58 | PBXtech | ok |
| 19:18.00 | szi | greetings |
| 19:18.08 | PBXtech | no prob |
| 19:18.22 | PBXtech | jerjer you know how well the echo cancelation module is? |
| 19:18.33 | szi | i was wondering if someone could answer a hardware question for me? |
| 19:18.43 | JerJer | i presume there is an additional expense for the new gen cards |
| 19:18.47 | Lee__ | I have echo problems with a my blue IAXy. |
| 19:19.10 | PBXtech | yea im just wondering if the echo cancel mod is better code than what comes with * |
| 19:19.45 | szi | does anyone know if the dialogic DTI series cards are supported by *? |
| 19:19.57 | szi | I dont' see them on the list but i believe they use the same API's |
| 19:20.15 | JerJer | don't bother |
| 19:20.43 | jeffgus | firmware upgrade program? |
| 19:20.48 | jeffgus | how will that work? |
| 19:20.53 | bjohnson | wow .. http://www.tshirthell.com/images/contestpics/a42_007.jpg |
| 19:20.56 | jeffgus | i just recently bought a t1 card for a client |
| 19:21.05 | PBXtech | "This is a 67% increase over previous benchmarks." wow |
| 19:21.16 | PBXtech | tshirthell.com is cool |
| 19:22.16 | jeffgus | so it sounds like the firmware on the digium cards is not easy to upgrade |
| 19:22.32 | PBXtech | you cant upgrade the firmware :) |
| 19:23.28 | PBXtech | im dying to try out that ds3 card and the echo cancelation module |
| 19:24.03 | jeffgus | hmm |
| 19:24.12 | jeffgus | so it has to be replaced |
| 19:24.13 | jeffgus | bummer |
| 19:24.26 | jeffgus | well i guess putting a flash chip on the things would increase the price a bit |
| 19:24.31 | PBXtech | has anyone played with the juicebox PoE box? |
| 19:25.53 | szi | k |
| 19:27.08 | Lee__ | PBXtech: yes. it's alright. it's an injector, not a switch. |
| 19:27.20 | Lee__ | so you need a whole lot of cables. |
| 19:27.21 | PBXtech | right, do they power cisco 7960? |
| 19:27.51 | Lee__ | don't know. they have a compatibility chart on their web site. Cisco is notorious so you should definitely read it. |
| 19:28.08 | PBXtech | oh ok didnt see that yet |
| 19:29.25 | PBXtech | you recomend the juicebox? |
| 19:30.32 | PBXtech | looks like it needs an adapter |
| 19:31.08 | fugitivo | what is that? |
| 19:32.02 | PBXtech | what is waht |
| 19:32.09 | fugitivo | the juicebox |
| 19:32.14 | PBXtech | PoE |
| 19:32.49 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 19:33.37 | rephorm | anyone know if its possible to configure a te110p to do dynamic channel allocation on a PRI? ('borrowing' unused voice channels for data) |
| 19:33.50 | CoolWhip | jeffgus: Field upgradable firmware for *future* updates. |
| 19:34.03 | CoolWhip | seems like it is not an option now but will be later |
| 19:34.17 | CoolWhip | that much is good - I guess. |
| 19:38.09 | PBXtech | i want to try out one of those Soekris boards also |
| 19:38.31 | fugitivo | any tool for linux to edit pdf? |
| 19:38.41 | fugitivo | (sorry, offtopic) :) |
| 19:38.46 | Lee__ | PBXtech: they are fun. I got Debian + Asterisk running on a 4801. |
| 19:39.00 | PBXtech | with the astlinux? |
| 19:39.12 | Lee__ | no, with Debian. |
| 19:41.27 | PBXtech | on the flash cards? |
| 19:41.45 | PBXtech | how many phone you got running off it? |
| 19:43.13 | PBXtech | hp sells a nice thin client with a PCI expansion that would be fun to try as well |
| 19:43.23 | *** join/#asterisk DaLion (~DaLion@Quebec-HSE-ppp214288.qc.sympatico.ca) |
| 19:43.56 | PBXtech | 1Ghz vs 266Mhz |
| 19:44.24 | DaLion | anyone know if we have DTMF ssues on latest cvs'es ? |
| 19:48.56 | jake1932 | i'm getting this when I start asterisk: Jun 24 15:48:32 WARNING[2424]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device |
| 19:48.56 | jake1932 | J |
| 19:49.13 | ender | did you mix up fxo vs fxs ? |
| 19:49.26 | jake1932 | did befor - but i fixed it |
| 19:49.34 | jake1932 | i'll pastebin |
| 19:49.37 | eKo1 | you don't want do that. you can blow up your fxs that way |
| 19:50.20 | pifiu | what causes IRQ conflicts? |
| 19:50.40 | DaLion | ? |
| 19:50.42 | DaLion | guys |
| 19:50.46 | jake1932 | http://pastebin.ca/15791 |
| 19:50.58 | DaLion | im getting clipped dtmf ... im RFC and client has xten sip as inband |
| 19:51.49 | *** join/#asterisk Primer (~primus@sh.nu) |
| 19:52.41 | wizhippo | pifiu: irq sharing among pci boards? You need a mother board with apic or that ca assign irq in the bios to specific pci slots |
| 19:52.43 | Primer | Anyone here use teliax? Know how to dial an international number? Is it just country code + area code + number or must something be prefixed to it? I can't seem to find the right combination |
| 19:53.22 | eKo1 | Well, if it's US based, then follow the prefixing rules in the US: |
| 19:53.29 | eKo1 | Otherwise, follow the ENUM rules. |
| 19:53.45 | Primer | I'm in the US. I tried 011 + cc + ac + number |
| 19:53.46 | rnovotny22 | Anyone know why ip500 would stop making calls after about an hour. |
| 19:54.10 | Primer | care to enlighten me with the enum rule? |
| 19:54.14 | Primer | please |
| 19:55.49 | eKo1 | yeah, just dial cc + number |
| 19:56.15 | Primer | heh |
| 19:57.00 | Primer | thanks |
| 19:59.38 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 20:00.10 | *** part/#asterisk Primer (~primus@sh.nu) |
| 20:00.55 | rephorm | rnovotny22: does it still show up in sip show peers? |
| 20:02.19 | rnovotny22 | Yes it does. |
| 20:02.47 | wizhippo | jake1932: make sur eyou kernel modules are loaded |
| 20:05.46 | wizhippo | jake1932: and why are you using fxo_ks in zapata.conf and fxs in zaptel.conf |
| 20:07.43 | vooduhal | With FXO cards in the US, which is best to use, ks, ls, or gs? |
| 20:08.43 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
| 20:09.04 | `Sauron | ks |
| 20:09.12 | `Sauron | most CO's probably are KS |
| 20:14.31 | vooduhal | Stupid question, but is ks, ls, and gs something that my CO guy would understand or is there another name for them? |
| 20:15.55 | eKo1 | I think ks is something the zap guys made up right? |
| 20:16.31 | *** join/#asterisk Strom_C (~strom@66.159.243.60) |
| 20:16.55 | Strom_C | hello |
| 20:17.57 | file[laptop] | Strom_C: STROM!!!! |
| 20:18.26 | Strom_C | file[laptop]: FILE!!!!!!!!! |
| 20:19.09 | Strom_C | whats cookin |
| 20:19.34 | file[laptop] | nothin' |
| 20:19.35 | file[laptop] | u? |
| 20:20.23 | DaLion | Primer 011XXXXXXXXX |
| 20:21.23 | eKo1 | If a SIP channel is busy, does DIALSTATUS get set to UNAVAIL? |
| 20:21.27 | Strom_C | not too much...working on my asterisk presentation |
| 20:23.03 | Ariel_ | vooduhal, ks- kwel start, ls- loop start, gs - ground start, There types of signaling to the pots lines or some call analog ports. |
| 20:24.07 | vooduhal | Yes, I know that, but are kewlstart, loopstart, and groundstart common phone terms? |
| 20:24.49 | Ariel_ | vooduhal, in my view yes. But for the engineer's not your normal telco sales person. |
| 20:26.02 | vooduhal | Ah, ok. The reason being, we are still fighting serious echo problems between our asterisk boxes and the CO (DMS100) and didn't know if I asked them for help if they would know what those options are. |
| 20:27.42 | `Sauron | I doubt those would affect echo |
| 20:28.00 | `Sauron | but they are regular telco terms |
| 20:28.06 | `Sauron | technical, but still regular |
| 20:29.03 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 20:35.06 | Strom_C | vooduhal: loopstart gives dialtone upon closure of the loop. groundstart gives dialtone upon closure of the loop and momentary connection to ground. kewlstart is loopstart with far-end disconnect supervision via momentary battery drop |
| 20:35.11 | Strom_C | none of them have to do with echo |
| 20:36.24 | vooduhal | I realize that, but at this point we are just trying to cover all of our bases. |
| 20:37.04 | Strom_C | vooduhal, why are you connecting via POTS lines? |
| 20:37.58 | rnovotny22 | Strom_C: Is there a better way. Thats how we're doing it here. |
| 20:37.59 | vooduhal | Because the banks we are installing these systems in will not allow for an internet connection and only have 5 POTS lines and won't pay for a T1. |
| 20:38.32 | Strom_C | vooduhal: what type of FXO card are you using and how long is the copper loop |
| 20:38.35 | vooduhal | We're installing 20 Polycom IP 500s and * in each location. |
| 20:38.45 | Strom_C | rnovotny22: ISDN or VOIP |
| 20:38.53 | *** join/#asterisk Cherebrum (jgarland@72.36.136.226) |
| 20:39.10 | Cherebrum | Anyone here using astlinux on the soekris? |
| 20:39.25 | vooduhal | TDM400P with 4 FXO modules. We are at the CO so the loop length is very short. |
| 20:39.56 | rnovotny22 | Strom_C: We only have 4 pots lines now. Any idea on the price difference for ISDN or VOIP? |
| 20:40.02 | vooduhal | We have to have the echo problem solved before we can deploy them so we have a test environment setup here. |
| 20:45.55 | alvis2k5 | Can somebody help me, i want to register the ASTERISK with a CISCO Gatekeeper |
| 20:45.58 | rephorm | vooduhal: do you get echo on local (sip to sip) calls also? and, do you get it on ALL outside calls, or just to specific numbers? |
| 20:46.17 | vooduhal | rephorm: Nope, just SIP -> PSTN. |
| 20:47.08 | rephorm | vooduhal: what about the second part of that question? :) |
| 20:47.31 | alvis2k5 | Can somebody help me, i want to register the ASTERISK with a CISCO Gatekeeper |
| 20:48.26 | eKo1 | eh, what do you need help with exactly. |
| 20:48.42 | vooduhal | rephorm: Yes all outside calls, both inbound and outbound. |
| 20:49.32 | rephorm | vooduhal: ok. so the problem is definitely on your end :) (just wanted to make sure. every time we've noticed echo, it was from the other end) |
| 20:50.13 | vooduhal | rephorm: This is what sucks, we had it resolved and working great and then the next day it was worse than ever. |
| 20:51.13 | rephorm | vooduhal: have you tried plugging a line directly into an analog phone? (bypassing ast) |
| 20:55.07 | *** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 20:57.11 | wwalker | I've read up on EAGI in a couple of places and it's still unclear how it differs from AGI. Some things sound like the AGI can provide audio on fd 3, some like audio comes from fd 3, others like commands go on fd 3 and audio comes out on fd 4. Please, help... |
| 20:57.55 | Ariel_ | vooduhal, echo can come into the picture due to your phone lines as well. If you get it fixed at your test location you may have to do other options at the next location. |
| 20:58.00 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 20:58.03 | rnovotny22 | vooduhal: Have you had any problems with the ip500's not being able to make calls after 30 minutes? |
| 20:58.49 | *** join/#asterisk CoolWhip (~CoolWhip@66.236.68.169.ptr.us.xo.net) |
| 20:59.03 | Ariel_ | rnovotny22, we are using here 7 IP-500 and they all work just fine. |
| 21:00.22 | rnovotny22 | I wonder if I have a setting wrong on the phone configuration. * is able to see them fine. |
| 21:01.27 | rephorm | rnovotny22: can you still dial them? |
| 21:01.55 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:02.01 | rnovotny22 | Only from a zap line, not sip to sip. |
| 21:02.09 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:03.11 | rnovotny22 | If I reload SIP from * then they work fine for another 30 minutes. |
| 21:04.12 | rephorm | rnovotny22: no clue. sorry :( |
| 21:04.51 | *** join/#asterisk neonet_2006 (~icechat5@iphost-64-56-140-28.wpg.wiband.net) |
| 21:04.58 | neonet_2006 | hi everybody |
| 21:05.04 | neonet_2006 | quick question on DISA app |
| 21:05.15 | neonet_2006 | I run it ...everything worked fine |
| 21:05.27 | neonet_2006 | but the cdr on mysql ...only shows one leg |
| 21:05.34 | neonet_2006 | inbound |
| 21:05.40 | neonet_2006 | how about the other one... |
| 21:05.42 | neonet_2006 | ?? |
| 21:06.19 | *** join/#asterisk CoolWhip (~CoolWhip@66.236.68.169.ptr.us.xo.net) |
| 21:07.00 | neonet_2006 | any help will be apreciate it. |
| 21:07.01 | *** join/#asterisk Maxxed (~max@cpe-70-114-238-9.houston.res.rr.com) |
| 21:07.03 | CoolWhip | I having problem with Call progress detection and a TDM400 card... |
| 21:07.06 | khemir | one leg? |
| 21:07.17 | neonet_2006 | calling to the DISA app.... |
| 21:07.21 | CoolWhip | The phone company (ELI) has been great to work with... |
| 21:07.26 | ManxPower | CoolWhip, callprogress does not work. don't use it. |
| 21:07.33 | file[laptop] | ManxPower: darn, you beat me |
| 21:07.45 | neonet_2006 | but when getting the 2nd dial tone...and dialing...it does not show that call |
| 21:07.53 | neonet_2006 | in the cdr |
| 21:07.56 | CoolWhip | they have changed the signal to all diff types... |
| 21:08.00 | ManxPower | What part of this do you not understand? |
| 21:08.02 | ManxPower | ; This feature can also easily detect false hangups. The symptoms of this |
| 21:08.03 | ManxPower | ; is being disconnected in the middle of a call for no reason. |
| 21:08.21 | CoolWhip | In a last ditch effort they asked "How is you box grounded?" |
| 21:08.59 | CoolWhip | has any one had grounding issues that changed how callprogress worked? |
| 21:09.07 | ManxPower | CoolWhip, Did you tell them "It meditates every day. It's well grounded."? |
| 21:09.43 | CoolWhip | ManxPower: without callprogress the server is totally worthless. |
| 21:10.04 | file[laptop] | analog doesn't have reliable call progress stuff, it's a fact |
| 21:10.47 | CoolWhip | But other hardware gets it right - as bad as it may be. |
| 21:11.09 | file[laptop] | like? |
| 21:11.28 | CoolWhip | What hardware? |
| 21:11.32 | file[laptop] | yes. |
| 21:13.59 | CoolWhip | Well, my friend has one that uses pots lines I *think* it's says Toshiba on it. I don't know what hardware they have in it but he never complains... |
| 21:16.05 | CoolWhip | Still, if it didn't work and it is needed to have a half decent phone server - why sell hardware for it. |
| 21:17.02 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 21:17.02 | CoolWhip | It's like "Here, this will almost do what you need but you'll never quite be happy with it." |
| 21:17.37 | drumkilla | CoolWhip: support@digium.com |
| 21:18.52 | CoolWhip | yeah - I have been working with them and they have been great. |
| 21:20.16 | drumkilla | cool |
| 21:21.25 | CoolWhip | I'm just saying the idea the you shouldn't use or expect to use callprogress is insane. You really need it if you want anything close to working properly. |
| 21:22.12 | CoolWhip | It might be a PITB but I still expect it to work :) |
| 21:22.24 | pifiu | anyone ever used soyo ip phones? |
| 21:25.09 | Damin | drumkilla: So 1.0.8 release then? |
| 21:30.35 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 21:30.58 | MikeJ[Laptop] | Damin, was it you I had discussed community wireless stuff with? |
| 21:41.23 | Damin | MikeJ: POerhaps.. |
| 21:41.33 | Damin | MikeJ: How is that going for you? |
| 21:42.14 | pjz | what *is* 'call progress' ? |
| 21:42.23 | eKo1 | What's wrong with the following: exten => s,2,gotoif($["${DIALSTATUS} = "CONGESTION"]?6) |
| 21:42.34 | eKo1 | oh wait, i see why |
| 21:42.37 | eKo1 | duh |
| 21:46.53 | *** join/#asterisk key2 (~tree@gob75-2-81-56-64-17.fbx.proxad.net) |
| 21:49.46 | key2 | if i wanna forward lot of calls from one * to an other one, should I set up a trunk ? |
| 21:50.50 | eKo1 | eh, do it via sip or iax |
| 21:51.00 | eKo1 | i recommend iax |
| 21:51.00 | *** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com) |
| 21:53.08 | x86 | iax2 is schweet |
| 21:54.31 | CoolWhip | pjz: On zap it lets you know when a hangup happens at the far end. |
| 21:55.22 | CoolWhip | Without it your lines stay open. There are hacks that you can to to less how bad that sucks but in the end - they are hacks and it still sucks. |
| 21:55.45 | eKo1 | call progress on analog lines stinks. this is a fact of life because analog sucks. |
| 21:56.21 | key2 | eKo1 so I should set up a trunk in IAX2 between my two asterisks ? |
| 21:56.40 | eKo1 | yeah. |
| 21:57.01 | eKo1 | use switch => ... |
| 21:57.51 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 21:58.08 | file[laptop] | or DUNDi |
| 21:59.31 | eKo1 | i haven't heard about dundi in a while... |
| 22:00.05 | file[laptop] | it's still out there... |
| 22:00.26 | sivana | eKo1: any benefit to switch over IAX peer between two * boxes? |
| 22:00.53 | eKo1 | and anyways, it's overkill for just switching calls |
| 22:01.40 | eKo1 | sivana: well that depends on your application. |
| 22:01.41 | file[laptop] | <PROTECTED> |
| 22:01.45 | *** join/#asterisk viLeR (~miv@gw.mutualasterisk.com) |
| 22:02.16 | sivana | do you have an example use for switch => ? |
| 22:02.25 | eKo1 | check the wiki |
| 22:02.28 | sivana | :) |
| 22:02.44 | sivana | I knew that was coming |
| 22:03.09 | eKo1 | well yeah, where do you think I got examples from |
| 22:03.33 | CoolWhip | thin air |
| 22:03.43 | CoolWhip | and everyone has access to that |
| 22:03.52 | *** join/#asterisk chrishodgetts (~chris@topanga.archnetnz.com) |
| 22:03.56 | chrishodgetts | hello all |
| 22:04.00 | eKo1 | hola |
| 22:04.33 | chrishodgetts | I have done some google searching for this, and there is not a lot on there, letalone what it means: |
| 22:04.36 | chrishodgetts | Jun 25 10:00:37 WARNING[3314]: chan_zap.c:3429 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 22:04.44 | chrishodgetts | does anyone know if it's a bug or a feature :) |
| 22:05.00 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 22:05.31 | sivana | chrishodgetts: I get that notice when someone comes in on the dialup pool |
| 22:05.52 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 22:06.03 | chrishodgetts | yeah, generally just after Asterisk has answered the call |
| 22:06.28 | sivana | are you getting it on dialup as well? |
| 22:06.52 | chrishodgetts | yeha, like when a call comes in and gets picked up by asterisk that error appears |
| 22:06.58 | sivana | I believe it's hearing the modem chatter |
| 22:07.00 | gtigene | Is there a chat room where they talk about using Polycom phones with Asterisk? |
| 22:07.03 | Katty | hi geeks. |
| 22:07.07 | sivana | and reporting the "strange" state |
| 22:07.19 | sivana | gtigene: you can here I would think |
| 22:07.27 | chrishodgetts | arr ok, so it's nothing to worry about then? |
| 22:07.27 | sivana | gtigene: I use Polycom |
| 22:07.34 | sivana | chrishodgetts: I've seen no ill effects |
| 22:07.38 | chrishodgetts | sweet |
| 22:07.43 | chrishodgetts | also :) another question hehe |
| 22:07.45 | chrishodgetts | exten => s,1,Answer |
| 22:07.46 | chrishodgetts | exten => s,2,Playback(locatesomeone) |
| 22:07.46 | chrishodgetts | exten => s,3,Goto(users,4700,1) |
| 22:07.51 | chrishodgetts | is in my extentions.conf |
| 22:08.00 | gtigene | My new Polycom 500 phone does not try to register itself with Asterisk. My Polycom 300 phones work fine. |
| 22:08.11 | chrishodgetts | basically I have it so that after x ammount of rings it plays a trying to locate user for you message |
| 22:08.16 | chrishodgetts | and gotos their extention |
| 22:08.22 | gtigene | The vendor says I need to set up my own boot sserver. |
| 22:08.28 | chrishodgetts | but I was wondering how I can make hold music play on that goto command |
| 22:08.41 | Katty | gtigene: did you configure the IP interface? |
| 22:09.23 | gtigene | Katty: The phone is using DHCP. Is that what you are referring to? And the network statisstics say it is sending and receiving some bytes. |
| 22:09.23 | chrishodgetts | or if there is another way to enable hold music to play untill it reaches voicemail |
| 22:09.57 | Katty | gtigene: not quite. the phone will automatically use dhcp out of the box, but it still needs to know the login information so it can register with asterisk. it doesn't know how to pull that out of a hat. |
| 22:10.17 | alvis2k5 | somebody can help me, i have problems to register the Asterisk in a CISCO Gateway |
| 22:10.22 | chrishodgetts | so you need to get an tFTP server set up to send the configs to the phone? |
| 22:10.35 | gtigene | Katty: If the login info was wrong would I not get a log message saying "Registration failed....." ? |
| 22:10.37 | Katty | gtigene: asterisk will show a whole bunch of failed registrations if you put in the wrong info, but before that...they're basically only able to call direct ips. |
| 22:11.06 | Katty | gtigene: yes, you would. but if you have absolutely NO info in the phone at all for registration, it's basically a big dumb expensive intercom that doesn't use asterisk |
| 22:11.21 | *** join/#asterisk Tangent (~Arc_Tange@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk) |
| 22:11.33 | alvis2k5 | somebody can help me, i have problems to register the Asterisk in a CISCO Gateway |
| 22:11.50 | Katty | gtigene: you can pull up the phone's ip address and then use Polycom and 456 to get into the Registration bit of the configuration |
| 22:11.56 | gtigene | Katty: I set up the login info the same on the 500 as on the 300. The 300's work but the 500 does not. I just entered username and password. And server IP address. |
| 22:12.21 | Katty | gtigene: on the 500 phones, is the little phone blacked in at the top right? |
| 22:12.31 | Katty | gtigene: or is the phone...whiteish, as in not filled in |
| 22:12.41 | Katty | errr, top left |
| 22:13.06 | gtigene | Katty: phone is whitish, I know what you mean, its not filled in. |
| 22:13.16 | Katty | can you ping the phone? |
| 22:13.26 | gtigene | Katty: just a sec I will try. |
| 22:13.30 | Katty | k |
| 22:14.06 | chrishodgetts | If I use exten => s,3,Dial(SIP/4700|1|m(default)) |
| 22:14.12 | chrishodgetts | it hangs up on the first ring |
| 22:14.23 | gtigene | Katty: it replies fine to a ping. |
| 22:14.30 | chrishodgetts | even if I make that dial 30 it still hangs up on the first ring |
| 22:14.42 | *** join/#asterisk DannyF (~dannyf@h74n1fls32o865.telia.com) |
| 22:14.51 | Katty | gtigene: can you make a screenshot of your registration html page and post it somewhere for me? |
| 22:15.32 | *** join/#asterisk tPO (~tPO@195.82.106.196) |
| 22:15.39 | DannyF | happy midsummer to everyone from the woods of Sweden ;) |
| 22:15.52 | Katty | DannyF: thx |
| 22:15.56 | chrishodgetts | happy cold winter from the city of Wellington New ZEaland |
| 22:16.00 | DannyF | :) |
| 22:16.10 | Katty | BYE ARIEL |
| 22:16.10 | gtigene | Katty: can I email you a .bmp file? |
| 22:16.15 | Katty | eww |
| 22:16.23 | chrishodgetts | .bmp :( |
| 22:16.27 | Katty | i'd rather you post it, gtigene |
| 22:16.32 | DannyF | this weekend ensures our population growth ;) |
| 22:16.40 | gtigene | Katty: I don't know where to post it. |
| 22:16.44 | chrishodgetts | hehehe |
| 22:16.47 | Katty | gtigene: i see |
| 22:16.57 | Katty | gtigene: alright, try this. |
| 22:17.07 | Katty | gtigene: get the ip from one phone...take it to another 500 phone |
| 22:17.09 | chrishodgetts | I guess no one can help on my music on transfer question then :) |
| 22:17.20 | Katty | gtigene: i want you to call phone A from phone B, using the ip address. |
| 22:17.26 | DannyF | chrishodgetts, whats up with it? |
| 22:17.32 | chrishodgetts | I am not sure what I need to do :) |
| 22:17.41 | key2 | someone knows if All the SIP phone support ulaw/alaw ? or just some of them |
| 22:17.46 | DannyF | is MOH orking every where else? |
| 22:17.46 | gtigene | Katty: OK |
| 22:17.49 | Katty | gtigene: you can call IP address, just remember to choose Name when you hit new call...see if you can call the other phone with the ip |
| 22:17.50 | chrishodgetts | yeah |
| 22:18.02 | chrishodgetts | DannyF A little background :) |
| 22:18.21 | DannyF | uhm... sure does MOH for me when i'm transfering someone |
| 22:18.35 | gtigene | Katty: should phone A be a working model 300 or a nonworking model 500? |
| 22:18.35 | chrishodgetts | DannyF I have an inbound FXO call, it gets answered after 30 seconds, and plays a voice announcment saying I will locate someone for you |
| 22:18.40 | chrishodgetts | I have a goto command at the moment |
| 22:18.46 | Katty | gtigene: a and b should both be 500s right now. |
| 22:18.46 | DannyF | k |
| 22:18.52 | gtigene | Katty: OK |
| 22:19.00 | chrishodgetts | and when I try and put a dial command it, it rings once and then hangs up |
| 22:19.09 | DannyF | urgh |
| 22:19.17 | DannyF | thats odd |
| 22:19.18 | Katty | chrishodgetts: is your timeout set to 1 second? |
| 22:19.31 | gtigene | Katty: I got a fast busy |
| 22:19.36 | DannyF | was gonna ask that, timeout? |
| 22:19.36 | chrishodgetts | exten => s,1,Answer |
| 22:19.36 | chrishodgetts | exten => s,2,Playback(locatesomeone) |
| 22:19.36 | chrishodgetts | exten => s,3,Goto(users,4700,30) |
| 22:19.39 | Katty | gtigene: rut roh! |
| 22:19.40 | gtigene | Right away |
| 22:19.47 | chrishodgetts | is what I have |
| 22:19.48 | Katty | gtigene: you, sir, have a routing isue |
| 22:19.54 | Katty | gtigene: i mean routing. |
| 22:20.04 | gtigene | Katty: thank you |
| 22:20.11 | DannyF | and 4700 look like? |
| 22:20.27 | Katty | gtigene: check your numbers and make sure you didn't forget a primary dns or a gateway somewhere... |
| 22:20.29 | chrishodgetts | so I chnage that to exten => s,3,Dial(SIP/4700|30|m(default)) and it still hangs up after one ring |
| 22:20.43 | Katty | gtigene: those phones can make direct phone calls to each other using IP address...not even needing asterisk... |
| 22:21.08 | chrishodgetts | exten => 4700,1,Dial(SIP/4700,20,rt) |
| 22:21.08 | chrishodgetts | exten => 4700,2,VoiceMail,u4700 |
| 22:21.08 | chrishodgetts | exten => 4700,102,VoiceMail,b4700 |
| 22:21.19 | gtigene | Katty: that very interesting. I will get to work on this right away. I was so frustrated not knowing where to start. |
| 22:21.24 | DannyF | and IP/4700 is avail? |
| 22:21.27 | chrishodgetts | yep |
| 22:21.36 | rnovotny22 | Katty: I have a little tougher problem for you if you are interested. |
| 22:21.37 | gtigene | Katty: :) |
| 22:21.39 | DannyF | odd |
| 22:21.44 | znoG | OT question: anyone use sylpheed? |
| 22:21.53 | DannyF | HEAD pr 1.0.x? |
| 22:21.56 | Katty | rnovotny22: shoot, but i might have to run off when the hair dye is ready to come out |
| 22:21.56 | DannyF | or* |
| 22:21.58 | chrishodgetts | although that Dial command will not send to voice mail or anything |
| 22:22.03 | chrishodgetts | so I should use a transfer command? |
| 22:22.10 | chrishodgetts | it will only ring the phone |
| 22:22.28 | chrishodgetts | in reality I want it to goto 4700 and go through the setps that 4700 ha |
| 22:22.29 | chrishodgetts | s |
| 22:22.32 | Katty | gtigene: (= |
| 22:22.36 | DannyF | any reason u dont use macro std-exten? |
| 22:22.42 | chrishodgetts | dont know how to :) |
| 22:22.48 | DannyF | ah :) |
| 22:22.52 | chrishodgetts | I have only been using asterisk for about 2 months |
| 22:22.55 | rnovotny22 | Also using ip500 units. The phones seem to "deregister" with * after30 minutes . |
| 22:23.00 | chrishodgetts | and only just got my first Physical phone yesterday |
| 22:23.14 | *** join/#asterisk daork (~daork@don.bhnb.net.nz) |
| 22:23.14 | DannyF | hell i'we been uing it for a year and still dont get half of it.. so dont feel bad ;) |
| 22:23.20 | Katty | rnovotny22: does the cli say anything? |
| 22:23.22 | chrishodgetts | ;) I dont :) |
| 22:23.31 | daork | has anyone put much thought in to doing call routing with LDAP? |
| 22:23.33 | Katty | rnovotny22: any errors...warnings... |
| 22:23.48 | chrishodgetts | I will go and read up about std-exten |
| 22:23.54 | daork | or, some other database that is common to lots of * boxes |
| 22:23.55 | DannyF | it's a good idea |
| 22:23.59 | opus__ | yo |
| 22:24.04 | rnovotny22 | cli says nothing once phones have lost it. |
| 22:24.11 | rnovotny22 | No error or warnings. |
| 22:24.13 | DannyF | cuz then u can NoOp(${DIALSTATUS}) |
| 22:24.17 | eKo1 | daork: no |
| 22:24.19 | Katty | rnovotny22: what does "sip show users" say? |
| 22:24.21 | DannyF | and it will tell u a bit more whats going on |
| 22:24.31 | daork | I'm just trying to figure out the best way to do call routing when i have many * boxes around a country |
| 22:24.31 | Katty | rnovotny22: at the cli...anything in there? |
| 22:24.50 | DannyF | i am sorry if im not making much sence cuz i'we have had a few to many vodkas |
| 22:25.14 | rnovotny22 | show users shows all listed units ACL = no, NAT = no |
| 22:25.21 | daork | DannyF: go out! find a bar! |
| 22:25.22 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 22:25.49 | DannyF | daork, when i said I sent greeting from the WOODs of sweden i meant it ;)) |
| 22:26.04 | daork | DannyF: haha |
| 22:26.06 | Katty | rnovotny22: look through the ip configuration part...see if you have anything that says unregister on reboot |
| 22:26.22 | *** join/#asterisk rizzo2 (~luigi@adsl-130-67.37-151.net24.it) |
| 22:26.42 | rnovotny22 | Katty: If I send the dial ext from CLI, the phone then re-registers with * and is able to call out. |
| 22:26.49 | Katty | oh |
| 22:26.52 | Katty | hmm. |
| 22:27.01 | DannyF | chrishodgetts, you can do many neat things using macros like failovers etc dont stare blindly at the samples but try to think outside the box and you'll find heaps of neat things todo |
| 22:27.03 | Katty | i know what it's doing |
| 22:27.09 | Katty | it doesn't know to reregister |
| 22:27.10 | rnovotny22 | great! |
| 22:27.23 | Katty | the phones are told to resync with asterisk every so often |
| 22:27.30 | Katty | just so that the server knows who's available and who's not |
| 22:27.42 | key2 | someone knows if All the SIP phone support ulaw/alaw ? or just some of them |
| 22:27.46 | Katty | unfortunately, i've never had to play with that before :< |
| 22:28.02 | Katty | i'm guessing it'd be in the ip configuration section though |
| 22:28.08 | rnovotny22 | Makes sense. Just ned to know how to set that |
| 22:28.09 | Katty | somewhere |
| 22:28.12 | Katty | timeout settings or something |
| 22:28.31 | rnovotny22 | That would be in the phone, right? |
| 22:28.40 | Katty | to be honest, i don't know |
| 22:28.53 | Katty | worth a look before you go rooting through asterisk though |
| 22:29.56 | Katty | rnovotny22: might be3 something in phone.conf too |
| 22:30.07 | rnovotny22 | I have been using the web configuration for the phones and did not see any timeout settings |
| 22:30.24 | Katty | k |
| 22:30.30 | Katty | then it's in one of the conf files, but i don't know where |
| 22:30.36 | Katty | probably sip.conf |
| 22:30.41 | Katty | dunno |
| 22:30.54 | rnovotny22 | Thanks Katty! |
| 22:31.16 | ManxPower | sounds like you need qualify=yes to prevent the NAT translations from timing out. |
| 22:31.41 | ManxPower | either that or tell your phone to reregister every 60 seconds |
| 22:31.57 | rnovotny22 | Everything is behind the NAT. Not going outside the local network as of yet. |
| 22:33.09 | ManxPower | then the only explaination I can come up with it "God hates you." |
| 22:33.12 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
| 22:33.48 | rnovotny22 | I don't know about that. Just new to Linux, Asterisk, and Polycom! |
| 22:35.06 | rittwage | 1.0.8: any caveats from 1.0.7? |
| 22:35.48 | DannyF | Katty, btw you running HEAD? |
| 22:37.02 | *** join/#asterisk doctor_za_ljubav (bkwyg@195.252.89.108) |
| 22:37.23 | *** join/#asterisk rizzo2 (~luigi@adsl-130-67.37-151.net24.it) |
| 22:37.51 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 22:39.19 | Katty | DannyF: no clue |
| 22:39.33 | ManxPower | rittwage, any important issues should be in the Changelog in the 1.0.8 source |
| 22:39.36 | Katty | rnovotny22: yeah, everything over here is behind a firewall too...we have nat=yes |
| 22:39.43 | DannyF | cvs head or 10.7 or 1.0.8 ;) |
| 22:39.48 | DannyF | 1.0.7* |
| 22:39.56 | Katty | rnovotny22: it's a good idea to set it :) |
| 22:40.06 | DannyF | they behave a tad differently wit registrations |
| 22:40.11 | DannyF | with* |
| 22:40.20 | Katty | you sure typo a lot DannyF |
| 22:40.37 | *** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 22:40.53 | DannyF | maybee i should just shut the heck up ;) |
| 22:41.20 | ender | ah. Nothin glike rinking beer at your desk. |
| 22:41.27 | doctor_za_ljubav | is there anyone that made bluetooth cell working with asterisk |
| 22:41.28 | Nuxi | rittwage, seems 1.0.8 is just a few bug fixes. I've noticed no difference. |
| 22:41.35 | DannyF | i had terrible problems getting registrations to stick afew days ago |
| 22:41.42 | Wonka | ender: beer's no good... |
| 22:41.44 | Wonka | Drink More Water! |
| 22:41.51 | Katty | willy wonka |
| 22:42.00 | Katty | and your water factory |
| 22:42.01 | Ahrimanes | 1.0.8 is a bugfix/security update and very minor feature update |
| 22:42.11 | eKo1 | chocolate water that is |
| 22:42.12 | Wonka | http://wiki.whatthehack.org/WTW/index.php/Hydration |
| 22:42.43 | rnovotny22 | DannyF: So what cleared it up? |
| 22:43.00 | DannyF | make update (head) worked fine |
| 22:43.20 | rnovotny22 | ? |
| 22:43.33 | DannyF | was something with digests comparisons being case sensitive |
| 22:43.47 | DannyF | was fixed last week |
| 22:43.49 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
| 22:44.11 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:44.56 | rnovotny22 | DannyF: Sorry I'm new to Linux. Still have not been able to get CVS to work. |
| 22:45.45 | DannyF | oh k |
| 22:46.46 | DannyF | if you want latest "stable" cd /usr/src and do |
| 22:46.54 | Katty | rnovotny22 needs hand holding |
| 22:47.01 | DannyF | cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot login |
| 22:47.03 | DannyF | cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 22:47.24 | DannyF | if you want the latest dev verision (HEAD) |
| 22:47.29 | DannyF | cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot login |
| 22:47.37 | DannyF | cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot checkout zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 22:47.37 | Katty | Ahrimanes: pffft :P |
| 22:47.55 | Katty | Ahrimanes: get off you infernal male of DOOM :P |
| 22:48.00 | *** join/#asterisk DaLion (~DaLion@Quebec-HSE-ppp213371.qc.sympatico.ca) |
| 22:48.01 | rnovotny22 | Katty: Thought I was learning fast! Installed linux. Was able to load and compile * and Zaptel and even get it almost working!! |
| 22:48.11 | Katty | rnovotny22: better than me... |
| 22:48.11 | DaLion | 17:06] ManxPower: ; This feature can also easily detect false hangups. The symptoms of thiss being disconnected in the middle of a call for no reason. |
| 22:48.17 | DaLion | what feature isi that ? |
| 22:48.24 | DannyF | then go into zaptel and make; make install then cd ../libpri make; make install etc |
| 22:48.25 | Ahrimanes | Katty: :P |
| 22:48.35 | DannyF | hope that helps |
| 22:49.02 | rnovotny22 | DannyF: I was under the impression you did not need libpri. |
| 22:49.25 | DannyF | well always good to build just in casae (I do ;)) |
| 22:50.00 | DaLion | MAnx ? 17:06] ManxPower: ; This feature can also easily detect false hangups. The symptoms of thiss being disconnected in the middle of a call for no reason. |
| 22:50.01 | DaLion | what feature you talking about |
| 22:52.09 | DaLion | or dumkilla or coolwhip ? |
| 22:52.13 | DaLion | or anyone ? |
| 22:52.41 | chrishodgetts | arrgg rebooted asterisk server, and it didnt come up |
| 22:52.45 | chrishodgetts | and I am in a remote location :( |
| 22:53.11 | Ahrimanes | chrishodgetts: server didnt come up or asterisk didnt? |
| 22:53.16 | chrishodgetts | server |
| 22:53.47 | Ahrimanes | ah crap |
| 22:53.52 | chrishodgetts | hmm 1 more hour before lunch |
| 22:53.55 | Ahrimanes | remote serial console is your friend |
| 22:53.56 | chrishodgetts | I will go home and do it then :) |
| 22:54.21 | rnovotny22 | DannyF: I'll try that and get back to you. |
| 22:54.25 | DannyF | chrishodgetts, no worries |
| 22:54.46 | DannyF | what does tail /var/log/messages say |
| 22:55.21 | DannyF | or did the machine die? |
| 22:55.42 | Ahrimanes | DannyF: machine didnt come up |
| 22:55.47 | Ahrimanes | kinda hard to tail any logs then |
| 22:55.52 | DannyF | ah the wonderz of technology :) |
| 22:55.56 | Ahrimanes | :D |
| 22:56.01 | DannyF | hehe |
| 22:56.28 | Ahrimanes | i have remote serial console on all servers, life saver when shit happens and you're away from the server room |
| 22:56.45 | DannyF | Ahrimanes, indeed |
| 22:57.11 | DannyF | i'we also noticed that APC si our firend |
| 22:57.14 | DannyF | is* |
| 22:57.19 | Ahrimanes | ups? |
| 22:57.23 | DannyF | ya ;) |
| 22:57.26 | DaLion | you got lots of friends |
| 22:57.29 | DaLion | ;) |
| 22:57.36 | DannyF | indeed ;) |
| 22:58.04 | DaLion | someone yelled ; This feature can also easily detect false hangups. The symptoms of thiss being disconnected in the middle of a call for no reason. |
| 22:58.14 | DaLion | wich one ? |
| 22:58.17 | DannyF | external power management sure helps when u need tokick a server in the behind |
| 22:58.22 | DaLion | one knows ? |
| 22:58.23 | Ahrimanes | well, our entire hosting facility jhas emergency power with diesel generators :D |
| 22:58.23 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 22:58.46 | eKo1 | how many seconds before the generator kicks in? |
| 22:58.52 | Ahrimanes | ah, well hp/compaq ilo server management can do hard reboot via ssh or webinterface |
| 22:59.04 | DannyF | Ahrimanes, yes thats allö good but a little over kill to reboot 28000 servers instead of just ine *grin* |
| 22:59.10 | Ahrimanes | uhm, battery power kick in right waway, diesel after 10 mins |
| 22:59.26 | Ahrimanes | DannyF: see what i wrote about my servers, hehe |
| 23:00.51 | Ahrimanes | haha |
| 23:01.47 | rnovotny22 | DannyF: Looks like is did a bunch of updating. If I run Make and Make Install will it overright my config files?? |
| 23:07.58 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 23:12.47 | DannyF | rnovotny22, no onlu if you make samples |
| 23:12.59 | rnovotny22 | Thanks |
| 23:12.59 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 23:18.37 | *** join/#asterisk CoolWhip (~CoolWhip@66.236.68.169.ptr.us.xo.net) |
| 23:21.19 | CoolWhip | Anyone have a good link on how to set up the ppp or Frame Relay with a TE110P. The zap stuff looks easy enough but I haven't done anything like ppp with one of these. |
| 23:21.53 | Ahrimanes | what do you need ppp for? |
| 23:22.17 | CoolWhip | well it can have a few channels of data - right? |
| 23:22.28 | *** part/#asterisk tPO (~tPO@195.82.106.196) |
| 23:22.41 | Ahrimanes | ah |
| 23:22.50 | Ahrimanes | well havent tried that out |
| 23:22.53 | CoolWhip | I was thinking 1 T1. We get a little data and a little voice. |
| 23:23.12 | CoolWhip | As you can see neither have I |
| 23:23.25 | Ahrimanes | yep, hehe |
| 23:24.19 | CoolWhip | I guessing there is a kernel module that you load that will give you a device but I have no idea if that module is part of the standard kernel or how it might be configured... |
| 23:24.31 | key2 | someone knows if All the SIP phone support ulaw/alaw ? or just some of them |
| 23:24.50 | Ahrimanes | key2: most support a/ulaw, a few only speex |
| 23:25.33 | Ahrimanes | CoolWhip: that's the zaptel driver |
| 23:25.57 | *** join/#asterisk CoderCR (~creyna@209.242.148.60) |
| 23:26.00 | CoolWhip | so zap will setup a ppp device for me? |
| 23:26.01 | CoderCR | hello all |
| 23:26.09 | CoolWhip | helo |
| 23:26.32 | Ahrimanes | CoolWhip: no, zap will give you access to the card.. then what you do from there i dont know |
| 23:27.36 | Ahrimanes | CoolWhip: but install zap driver and look in /dev/zap - there might be multiple devices that you can access |
| 23:27.54 | Ahrimanes | so assigning some for * and some for ppp |
| 23:28.22 | CoolWhip | so wcte11xp does the trick |
| 23:28.45 | CoolWhip | I'll have to check it out - After I get hardware. |
| 23:28.49 | Ahrimanes | yep |
| 23:29.22 | drumkilla | there should be info on the digium web site .... |
| 23:29.38 | Ahrimanes | :) |
| 23:29.42 | DaLion | rnovotny22 no.. only make sampls will overwrite config |
| 23:29.55 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 23:29.55 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:30.00 | Ahrimanes | drumkilla: would you mind finishing up 1.2 tonight? ;) |
| 23:30.27 | DaLion | and 1.3 ? |
| 23:30.28 | DaLion | \lol |
| 23:30.34 | Ahrimanes | hehe |
| 23:30.42 | Ahrimanes | nah 1.2 will do.. stable release with realtime plz |
| 23:31.42 | *** part/#asterisk rnovotny22 (~rnovotny2@207.195.206.201) |
| 23:32.00 | brookshire | asterisk business edition comes with real time |
| 23:32.08 | brookshire | it's stable |
| 23:32.22 | Ahrimanes | yeah but my boss wont pay for it.. already checked |
| 23:32.31 | alvis2k5 | HELP PLZ !!! I have to register ASTERISK with a CISCO Gatekeeper (H.323), can somebody help me???? |
| 23:32.57 | CoolWhip | Sangoma A101 |
| 23:33.10 | CoolWhip | why would I buy one over digium? |
| 23:33.11 | Ahrimanes | eek h323 |
| 23:33.21 | Ahrimanes | CoolWhip: hm maybe price? |
| 23:33.58 | CoolWhip | seem pretty close. |
| 23:34.50 | Ahrimanes | ok |
| 23:35.03 | Ahrimanes | i'd go for digium then.. support the * people :) |
| 23:35.43 | *** join/#asterisk outtolunc (~me@adsl-69-110-13-227.dsl.pltn13.pacbell.net) |
| 23:36.06 | CoolWhip | even see cheaper digium at $480 vs $520 |
| 23:36.20 | Ahrimanes | oh nice |
| 23:36.27 | CoolWhip | for the sangoma |
| 23:36.51 | DannyF | alvis2k5, did you check out the woomera? |
| 23:36.55 | CoolWhip | slow typer - baby in one hand |
| 23:37.29 | DannyF | chan_woomera is another alternative h323 implementation. |
| 23:37.29 | DannyF | visit www.pbxfreeware.org for more information. |
| 23:38.03 | key2 | If I want to answer a call from an SIP phone without pressing any button, should I configure that in Asterisk or on the SIP phone ? |
| 23:38.15 | DarthClue | Chan_woomera is the new way to talk. |
| 23:38.41 | mtgh | key2: the phone |
| 23:39.24 | CoderCR | brb |
| 23:39.26 | *** part/#asterisk CoderCR (~creyna@209.242.148.60) |
| 23:39.28 | Ahrimanes | key2: usually called monitor or autoanswer |
| 23:39.45 | key2 | Ahrimanes: so it's on the phone I have to setup that ? |
| 23:39.51 | Ahrimanes | key2: yup |
| 23:40.05 | key2 | Ahrimanes: and I guess it's a feature that not all SIP phone have |
| 23:40.30 | Ahrimanes | key2: hm yes.. the videophones i use tend to have it so they can be used for surveillance |
| 23:41.17 | *** join/#asterisk pv2b (~pvz@1-1-11-10b.o.sth.bostream.se) |
| 23:41.31 | key2 | Ahrimanes: so if I wanna buy sip phones for a call center and I don't want the agent to press any button between two calls, I rather buy a phone that has a autoanswer right ? |
| 23:42.12 | Ahrimanes | key2: that would make sense yes, but autoanswer is usually only done after 10-30 seconds |
| 23:42.56 | DarthClue | key2: just write an on-hook agent handler. |
| 23:43.59 | Ahrimanes | DannyF: you suggested it, now you get the support calls ;) |
| 23:46.17 | key2 | Ahrimanes: and there is no way to say that the phone doesn't have to ring at all but just answer the call ? |
| 23:49.18 | Ahrimanes | key2: depends on the phone |
| 23:49.46 | Ahrimanes | key2: i'd find a phone that fits my bill good, and then call up the manufacturer and request it as a feature.. |
| 23:51.02 | DannyF | Ahrimanes, ;) |
| 23:51.43 | DannyF | if u have any questions about Woomera you might want to pester Brian |
| 23:51.55 | DannyF | and ffs goto cluecon if you can! |
| 23:52.05 | DannyF | it's the real deal in * dev |
| 23:52.22 | Ahrimanes | no astricon is , hehe |
| 23:52.31 | DannyF | neh |
| 23:52.59 | DannyF | clueco will be loads of fun, im in pain since i'm on the wrong side of the puddle for this one ;/ |
| 23:53.49 | DannyF | also if you make brian happy... he'll continue to release such neat things as res_js (try it if you havent) |
| 23:53.50 | twisted[work] | key2, why not just have always on agents? they hear a beep in their ear and are connected to the customer |
| 23:53.58 | key2 | anyone knows an SIP provider that has cheap rates for middle east ? |
| 23:54.13 | key2 | twisted: yes!! what's that ? |
| 23:54.18 | DannyF | key2, did you check out voipjet? |
| 23:54.26 | twisted[work] | key2, uhm, a feature of the queues |
| 23:54.30 | twisted[work] | in asterisk |
| 23:54.33 | twisted[work] | ~rtfw |
| 23:54.33 | jbot | it has been said that rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 23:54.46 | key2 | twisted[work]: it's called "on agent" |
| 23:54.46 | key2 | ? |
| 23:54.57 | twisted[work] | no |
| 23:55.05 | twisted[work] | *I* call them "always on agents" |
| 23:55.10 | DannyF | twisted, be patint ;) |
| 23:55.11 | twisted[work] | but i'm not sure what the real term is |
| 23:55.20 | twisted[work] | DannyF, pardon? |
| 23:55.22 | *** join/#asterisk TechDawg (sumbody@ppp-70-244-249-209.dsl.snantx.swbell.net) |
| 23:55.24 | DannyF | ;) |
| 23:55.48 | DannyF | patient* (/&%/& cant spell anymore |
| 23:56.03 | twisted[work] | i don't have to be patient |
| 23:56.11 | twisted[work] | if you want patience from me you can pay me ;) |
| 23:56.18 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 23:56.27 | *** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-198.midsouth.biz.rr.com) |
| 23:56.29 | DannyF | hell yeah |
| 23:56.50 | DannyF | :) |
| 23:57.03 | key2 | twisted[work]: so what manual should I read then ? |
| 23:57.10 | DannyF | that'll give u 0.001ms of patience ;) |
| 23:57.23 | twisted[work] | key2, the wiki |
| 23:57.26 | Ahrimanes | twisted[work]: haha, good one |
| 23:57.36 | TechDawg | Uhm, a simple question. Can anyone recommend a cheap SIP device that can be setup to dial another SIP on IP and TCP/UDP port designation (ie public 63.160.241.161:3000)? |
| 23:58.10 | DannyF | erm... |
| 23:58.11 | twisted[work] | TechDawg, if by cheap, you mean /cheap/ then grandstream can do IP dialing |
| 23:58.31 | redder86 | can someone give me an example of a voicemail.conf entry where a per-user delete=yes is used? 1234 => 0000,Some Name,some@email.com,delete=yes ?? |
| 23:58.44 | twisted[work] | redder86, you got it |
| 23:58.49 | redder86 | thanks |
| 23:59.00 | TechDawg | That's a start, now does grandstream allow me to specify the TCP or UDP port? |
| 23:59.06 | twisted[work] | hmm, good question |
| 23:59.24 | twisted[work] | IIRC, it doesn't do tcp, but not sure about the port part |
| 23:59.52 | Ahrimanes | do grandstreams have webinterfaces for config? |
