| 00:00.32 | *** part/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 00:03.53 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:04.28 | mosty | it looks like asthostid has changed so that it doesn't require ethX interfaces, anyone know what it uses now? |
| 00:05.35 | *** part/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 00:08.41 | *** join/#asterisk ixx (i=foobar@cpe-70-112-149-89.austin.res.rr.com) |
| 00:09.13 | *** join/#asterisk metfan2007 (n=metfan20@201.103.80.238) |
| 00:09.19 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-58-141.red.bezeqint.net) |
| 00:10.03 | metfan2007 | Hi all! Any idea why DISA app sends discontinuous audio in the invite to dial tone? |
| 00:10.43 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 00:11.36 | Mikeonline | what does this mean?: after AGI script launch: Failed to execute '/var/lib/asterisk/agi-bin/fax-in.sh': Exec format error |
| 00:14.59 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:15.53 | *** join/#asterisk FarrisG (n=jrush@pool-71-164-183-233.dllstx.fios.verizon.net) |
| 00:23.18 | Mikeonline | now i understand why stskeeps has so much problem with drunk coding - sleepless coding is the same.. |
| 00:29.15 | drmessano | lol |
| 00:29.28 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:30.02 | Mikeonline | you look at the code and bet that it hasnt been changed |
| 00:30.17 | Mikeonline | but comments have been added... ;-) |
| 00:31.42 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 00:31.59 | Mikeonline | and now its time for backup |
| 00:32.53 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 00:35.07 | *** join/#asterisk ccvp (n=ccvp@user-24-214-126-81.knology.net) |
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| 00:37.45 | flush | vap0rtranz http://pastebin.com/m6adf30f3 |
| 00:37.50 | flush | sorry.. got guests here |
| 00:39.52 | TJNII | Does the (T) next to port in the iax2 show peers readout mean trunk? |
| 00:43.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-047610241a050f15) |
| 00:44.20 | *** join/#asterisk Katty (n=The@adsl-70-253-164-46.dsl.stlsmo.swbell.net) |
| 00:45.25 | *** join/#asterisk Sweet-P (n=me@ip68-111-191-246.sd.sd.cox.net) |
| 00:45.25 | Katty | <PROTECTED> |
| 00:45.29 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:45.34 | Mikeonline | hai fish |
| 00:46.27 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:47.01 | drmessano | hUA? |
| 00:47.58 | Katty | not quite |
| 00:48.22 | drmessano | No? |
| 00:48.55 | Katty | no. |
| 00:49.05 | Katty | pls try again. |
| 00:49.06 | flush | yo |
| 00:49.20 | flush | holy jesus, how do i get asterisk to ring the phone when a call comes in |
| 00:49.32 | flush | i try to call outside world, i get fast busy tone |
| 00:49.37 | flush | what gives ? |
| 00:49.45 | Katty | did you provide it with cookies? |
| 00:50.09 | Katty | and hardware |
| 00:50.20 | flush | me? |
| 00:50.33 | Katty | well no one else is asking me questions. |
| 00:50.35 | flush | cookies? the box wants to bite my hand |
| 00:50.39 | Katty | tho, i suppose i could go back to talking to drmessano |
| 00:50.46 | flush | i dont know what cookies youre talking about sir, but i have tdm400p card |
| 00:50.52 | flush | 1 fxo modules with 3x fxs |
| 00:50.55 | Katty | the chocolate chip kind, of course. |
| 00:51.04 | flush | affirmative |
| 00:51.05 | Katty | and that's ma'am |
| 00:51.13 | flush | i mean i even tried to give it breast |
| 00:51.19 | Katty | ... |
| 00:51.27 | *** join/#asterisk lyroy (n=lyroy@142.137.215.228) |
| 00:51.28 | flush | i figured out with a tutorial to get the echo test thing |
| 00:51.38 | Katty | what does zap show status from the CLI reveal? |
| 00:51.42 | flush | it works.. but im totally unsure on how to configure it to let me place calls |
| 00:51.48 | Katty | oh. |
| 00:51.49 | Katty | i see. |
| 00:51.56 | Katty | jbot: zaptel? |
| 00:51.56 | jbot | i heard zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 00:52.00 | flush | oh do i get the status with cli ? |
| 00:52.00 | Katty | hrmm |
| 00:52.07 | mosty | flush, status of what? |
| 00:52.16 | flush | <Katty> what does zap show status from the CLI reveal? |
| 00:52.31 | flush | i mean zttool shows OK |
| 00:53.03 | flush | if any of you got a good how to when you dont understand much why it doesnt work, that would be great |
| 00:53.10 | Katty | flush: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 00:53.18 | flush | copy |
| 00:53.20 | Katty | flush: go about halfway down the page to where it says Example |
| 00:53.43 | flush | i have this page opened in a tab already hehe |
| 00:53.54 | flush | but now theres something else.. it doesnt have context names |
| 00:54.02 | flush | only like exten => 1234 and stuff like that.. |
| 00:54.03 | Katty | flush: well you're all set! |
| 00:54.07 | flush | im not sure man |
| 00:54.14 | drmessano | yeah man |
| 00:54.38 | Katty | flush: your context name will be in zaptel.conf |
| 00:54.51 | Katty | flush: usually [from-pstn] or [from-zap] or something |
| 00:55.09 | Katty | flush: and then over in extensions.conf you'd have [from-pstn] |
| 00:55.11 | flush | lets say i would just want to be able to place a simple call.. |
| 00:55.18 | Katty | outgoing? |
| 00:55.19 | Katty | k |
| 00:55.24 | flush | yea i got this from the .pdf file i downloaded |
| 00:55.28 | flush | this is understood |
| 00:55.32 | Katty | what context is your SIP phone in? |
| 00:55.41 | flush | but with the Answer,(1|blahbah) and stuff im not sure |
| 00:55.45 | flush | i dont have voip.. |
| 00:55.51 | flush | SIP is voip ? |
| 00:56.10 | Katty | oh boy. |
| 00:56.17 | flush | ok ok sec |
| 00:56.19 | Katty | *hee* |
| 00:56.25 | flush | my phone is placed in [phones] |
| 00:56.32 | flush | include => internal |
| 00:56.34 | flush | something like this |
| 00:56.50 | Katty | i've never actually worked with analog phones. |
| 00:56.59 | Katty | you need a catch all tho |
| 00:57.03 | husimon | how do I delete a whole bunch of messages at once? |
| 00:57.09 | husimon | just go into the spool dir and delete them? |
| 00:57.11 | flush | damn |
| 00:57.14 | Katty | husimon: yes. |
| 00:57.19 | flush | gonna grab a beer bbiab |
| 00:57.44 | Mikeonline | flush: do you understand german? |
| 00:58.19 | Katty | flush: REALLY simple... exten => 1234567,1,Dial(Zap/g1/1234567) |
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| 01:09.14 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.193) |
| 01:09.37 | Katty | flush: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 |
| 01:09.38 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 01:09.55 | Katty | i sure wish i knew how to transfer two legs of a call into a meetme by dialing an extension. |
| 01:16.17 | AndyGraybeal | jmesquita: i used to know an ann mesquita |
| 01:19.55 | riddlebox | does anyone do "a dollar buyout"? |
| 01:20.33 | *** join/#asterisk anthm (n=anthm@70-9-38-28.area4.spcsdns.net) |
| 01:20.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:20.42 | riddlebox | for systems? |
| 01:21.08 | SteveTotaro | use app_bridge katty |
| 01:22.14 | SteveTotaro | http://bugs.digium.com/view.php?id=5841 |
| 01:22.19 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:25.22 | Katty | SteveTotaro: oh? do you have a link? |
| 01:26.22 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:28.56 | anthm | whatcha ponderin' ? |
| 01:29.29 | Katty | anthm: trying to figure out an easy way to let a manager chanspy an extension, as it were... |
| 01:29.43 | Katty | anthm: and another extension to pull them into a meetme conference and manager enters muted |
| 01:30.04 | Katty | anthm: i'm not sure how to get both legs of the first phone call into the meetme conference |
| 01:30.23 | Katty | anthm: channelredirect looks promising |
| 01:31.17 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:32.21 | Katty | anthm: from the managers perspective, they dial extension... 4, which asks for the extension they want to listen to. which then drags both the extension and the zap channel into the meetme....and the manager enters the meetme room muted. dunno if that makes sense. |
| 01:32.33 | anthm | yep |
| 01:33.02 | Katty | anthm: i don't suppose you have any linkery |
| 01:34.08 | anthm | presumably so he could talk to them if he wants ? |
| 01:34.13 | Katty | yes, both |
| 01:34.27 | Katty | instead of just Chanspy(blah/person|w) |
| 01:35.59 | anthm | chanspy is still around lol, my invention actually |
| 01:36.06 | anthm | that was not fun |
| 01:36.15 | anthm | took months |
| 01:36.27 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 01:36.33 | Katty | anthm: kings to you. i love it (= |
| 01:37.01 | Katty | anthm: did you put in a secret option to let us talk to both legs of the convo? :> |
| 01:37.28 | anthm | that came in the next generation ;) |
| 01:37.31 | seanbright-home | i've been told that would be relatively easy to implement in the latest 1.4/trunk |
| 01:37.48 | seanbright-home | a "poor man's conference call" with chanspy |
| 01:37.56 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 01:38.03 | seanbright-home | requires some coding in c, however. |
| 01:38.05 | Katty | [hC]: herro (= |
| 01:38.14 | [hC] | Katty: hola! |
| 01:39.09 | anthm | in my new stuff when you spy on a call you can press 1 to talk to a leg 2 to talk to b leg 3 to talk to both and 0 to talk to none |
| 01:39.30 | mosty | is it possible to get chan_sip to listen on multiple ip addresses on the same subnet? ie if a call comes in to address A, the return packets should be from A, if a call comes on to address B, the return packets should be from B |
| 01:40.00 | Katty | anthm: neat. when you releasing that? |
| 01:40.17 | anthm | that's an undefined operation in SIP cos there is no way to accurately tell which ip to use on the reply |
| 01:40.19 | Katty | anthm: let me guess. 2 years from now ;) |
| 01:40.58 | anthm | well it's already released but it's not for asterisk |
| 01:41.06 | Katty | oh well. |
| 01:41.15 | anthm | if you are using 1.2 you could use my app_changrab |
| 01:41.20 | Katty | nah |
| 01:41.25 | Katty | using 1.4 |
| 01:41.35 | Katty | if you run into a snippet of code that might help me, you know where to find me (= |
| 01:42.12 | mosty | anthm, can't chan_sip use the to address from the original packet for the from on the reply? |
| 01:42.33 | Mikeonline | yeah yeah yeah |
| 01:43.53 | anthm | it's more complicated than that cos you don't have control over which ip the packets will go out on at that level |
| 01:44.03 | anthm | they just pick the best one to make it there in the udp stack |
| 01:44.30 | Mikeonline | good night everyone |
| 01:44.51 | anthm | it's a long standing bug for like 3 years at least but it's more or less impossible to deal with without a separate ua on every ip you want it use |
| 01:44.52 | mosty | anthm, is that a decision that the kernel makes? ie you can't explicitly set a from address at the application level? |
| 01:45.25 | *** part/#asterisk Mikeonline (i=Mike@p57A7EFC7.dip.t-dialin.net) |
| 01:45.29 | Katty | anyone want half a headache? |
| 01:45.30 | anthm | that code has just regular socket API |
| 01:45.47 | anthm | so all they have is a socket and the sendto call |
| 01:46.07 | anthm | it's actually a violation in the sip spec to not have an absolute uri for a ua |
| 01:46.45 | *** join/#asterisk MikeJ (n=MikeJ@d149-67-220-106.try.wideopenwest.com) |
| 01:47.08 | husimon | Katty I'll trade you your half for my whole headache |
| 01:48.29 | Katty | :< |
| 01:48.42 | husimon | no deal? |
| 01:48.57 | Katty | anthm: i'm goingto pester you tomorrow about that channel redirect thingy |
| 01:49.18 | anthm | maybe that's a clone of my changrab |
| 01:49.21 | MikeJ | its not nice to pester :P |
| 01:49.30 | Katty | he likes it. |
| 01:49.34 | MikeJ | true |
| 01:49.42 | anthm | is it like so you can xfer an in-progress call to another ext ? |
| 01:49.50 | Katty | me and anthm have had some fun times! |
| 01:49.53 | anthm | http://www.freeswitch.org/asterisk_stuff/app_changrab.c |
| 01:51.30 | Katty | oh boy! this looks like it will make my head hurt worse! |
| 01:52.12 | Katty | anthm: i'll lookat that tomorrow |
| 01:54.57 | anthm | k |
| 01:56.32 | MikeJ | that changrab is 1.2 .. |
| 01:56.42 | MikeJ | for ast 1.2 |
| 01:57.16 | MikeJ | but doesn't need much to port forward |
| 02:02.56 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
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| 02:04.49 | *** part/#asterisk ctyler (n=chris@global.proximity.on.ca) |
| 02:05.48 | mosty | anthm, does the same issue occur with chan_iax? i assume so, but want to be sure before i go looking for a more complex solution |
| 02:09.44 | Katty | mmm, tylenol |
| 02:09.57 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.124.250) |
| 02:10.44 | MikeJ | good |
| 02:10.55 | MikeJ | so how are things in #asterisk land lately? |
| 02:11.43 | Katty | i suppose okay. |
| 02:11.53 | Katty | i seem to lurk most times :/ |
| 02:15.07 | *** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096725491.dsl.bell.ca) |
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| 02:25.47 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 02:25.52 | znoG | hey all |
| 02:26.00 | znoG | anyone know how to get res_config_ldap into asterisk 1.4? |
| 02:45.24 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 02:47.43 | hieunm_vips | Hi all, Could I ask question about channel SIP 3 |
| 02:48.03 | hieunm_vips | Does it continue now ? |
| 02:49.00 | hieunm_vips | How can I download it's source code ? |
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| 02:57.57 | graphx | Can anyone help find out why my SPA942 SIPphones will not do Attended Call Transfers>? |
| 02:58.26 | graphx | I am able to do Blind Transfers... |
| 02:59.36 | znoG | how stable is 1.6.x? |
| 03:00.58 | Qwell | it's beta - give it a try |
| 03:02.36 | znoG | i'm interested in it due to the res_config_ldap module that it has, but we plan on using it at the office so I'm wondering if it'll be stable enough for everyday use. |
| 03:02.44 | znoG | Not a big setup; 25 extensions. |
| 03:05.52 | Katty | Nugget: herroes. |
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| 03:20.37 | outtolunc | 9500 |
| 03:21.21 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 03:21.44 | obnauticus | err, anyone here good with bluetooth |
| 03:21.51 | obnauticus | i have a weird problem |
| 03:22.47 | Katty | i'm hungry. |
| 03:22.51 | Katty | i should go get a snack. |
| 03:22.53 | obnauticus | >:| |
| 03:23.09 | *** part/#asterisk MikeJ (n=MikeJ@d149-67-220-106.try.wideopenwest.com) |
| 03:23.22 | Katty | Nugget: what would you suggest? |
| 03:26.30 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 03:26.59 | Katty | hi manx |
| 03:28.16 | Nugget | beer and pretzels. |
| 03:28.25 | *** join/#asterisk rvhi (n=chatzill@udp182407uds.hawaiiantel.net) |
| 03:28.54 | rvhi | hi, got this weird problem, in the middle of the call, suddenly there is no rtp traffic from the server |
| 03:29.01 | Katty | hrmm |
| 03:29.03 | Katty | i have neither. |
| 03:34.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:34.14 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 03:45.57 | Katty | Nugget: mmm, chinese. |
| 03:46.06 | Nugget | yum |
| 03:46.18 | Katty | Nugget: quite. |
| 03:46.35 | drmessano | Yum? Invalid options specified |
| 03:46.57 | Nugget | nutty linux user. |
| 03:47.46 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 03:47.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 03:49.28 | Katty | horay for pears! |
| 03:51.59 | jmesquita | Guys, has anyone here used res_config_ldap before? |
| 03:53.08 | Katty | i have not. |
| 03:53.24 | jmesquita | Wondered if it was stable or not |
| 03:53.41 | znoG | jmesquita: im currently working on that |
| 03:53.46 | znoG | jmesquita: stuck with "protocol error" |
| 03:54.29 | jmesquita | znoG: Working on coding or simply making it work? |
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| 03:55.28 | znoG | jmesquita: making it work :) |
| 03:55.57 | jmesquita | znoG: Yeah, that would be perfect for me right now ... |
| 03:56.10 | jmesquita | znoG: Since all company users use ldap |
| 03:56.25 | jmesquita | znoG: SSO rox |
| 03:57.39 | drmessano | YAAAY |
| 03:57.43 | drmessano | Here we go again |
| 03:58.00 | jmesquita | znoG: Which * version? |
| 03:58.07 | jmesquita | znoG: 1.2 or 1.4? |
| 03:58.42 | J4k3 | here I go again on a troll |
| 03:58.54 | ManxPower | znoG: there are different versions of the LDAP protocol. |
| 03:59.13 | ManxPower | I believe there is v1 and v2, with v2 having optional SSL |
| 03:59.29 | drmessano | J4k3: which troll? |
| 03:59.37 | ManxPower | it's been a LONG time since I worked with LDAP, however. |
| 03:59.38 | J4k3 | all the trolls |
| 03:59.39 | jmesquita | ManxPower: It must be tested against OpenLDAP and MS AD |
| 03:59.49 | drmessano | lol |
| 04:00.08 | *** join/#asterisk VoicePulse_ (n=contact@unaffiliated/voicepulse) |
| 04:00.14 | jmesquita | ManxPower: But it is not on branch yet. Not even for 1.4! |
| 04:00.19 | russellb | it was tested against openldap AFAIK |
| 04:00.21 | jmesquita | ManxPower: The res I mean |
| 04:00.27 | russellb | it is only in asterisk 1.6 |
| 04:01.11 | jmesquita | russellb: I guess there won't be too much trouble to port it to 1.2, would it? |
| 04:01.17 | jmesquita | russellb: Not you guys, of course |
| 04:01.17 | russellb | good god yes there would be |
| 04:01.32 | russellb | asterisk has change a crap load since 1.2 |
| 04:01.35 | russellb | changed* |
| 04:01.57 | jmesquita | russellb: Ehehehe, just wondered b4 start looking at the code |
| 04:02.55 | jmesquita | russellb: Havent looked at anything > 1.2 yet |
| 04:03.33 | hieunm_vips | Hi all, Could I ask question about channel SIP 3 |
| 04:03.34 | hieunm_vips | Hi all, from where can I download chan_sip3 source code ? |
| 04:03.47 | hieunm_vips | Is it stopped now ? |
| 04:03.55 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:04.11 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 04:05.36 | znoG | jmesquita: using 1.6.x as the res_config_ldap seems to work on that version only |
| 04:06.08 | Katty | how is 1.6beta anyway? is it buggy? |
| 04:06.36 | drmessano | I'm gonna port 1.6 to 1.2 |
| 04:06.43 | znoG | ManxPower: not sure whats wrong with this. I'm looking at the res_config_ldap.c and if it can't bind, it just throws a 'bind failed: protocol error' but I can't deduce from the source code why it would fail |
| 04:06.49 | drmessano | j/k |
| 04:06.56 | Katty | drmessano: :< |
| 04:07.51 | znoG | ManxPower: i just started slapd with debug on and it showed the following error when starting asterisk: conn=0 op=0 RESULT tag=97 err=2 text=historical protocol version requested, use LDAPv3 instead |
| 04:08.04 | znoG | so that's a bit of a clue .. even though protocol is already set to 3 ... |
| 04:08.05 | drmessano | I had some issues with 1.6.. But I know others are running it without issue |
| 04:08.33 | Katty | much new shiny stuff? |
| 04:08.50 | drmessano | Yeah, there's some good new stuff |
| 04:08.54 | Katty | neat. |
| 04:09.08 | Katty | i might have a look see |
| 04:10.05 | Katty | drmessano: chicago is baning snack size ziplock baggies. |
| 04:10.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:11.05 | Katty | drmessano: http://www.suntimes.com/news/metro/826059,CST-NWS-bagban05.article <- better not take any carrots or trailmix with you. they'll lock you up! |
| 04:11.17 | drmessano | I heard.. totally cuts into my BIZNASS |
| 04:12.44 | Katty | s/BIZNASS/snacktime |
| 04:13.02 | russellb | to see what's new in 1.6, check out the CHANGES file ... *gets link* |
| 04:13.07 | drmessano | No, I sell asterisk to schoolkids |
| 04:13.19 | russellb | http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 04:13.30 | russellb | jbot: asterisk16 is new features listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 04:13.31 | jbot | russellb: okay |
| 04:13.42 | russellb | jbot: forget asterisk16 |
| 04:13.42 | jbot | russellb: i forgot asterisk16 |
| 04:13.53 | russellb | jbot: asterisk16 is <reply> new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 04:13.54 | jbot | russellb: okay |
| 04:15.02 | *** join/#asterisk fduplex (i=irc@CPE00a1b000eeb1-CM0012254495cc.cpe.net.cable.rogers.com) |
| 04:15.48 | drmessano | ~asterisk18 |
| 04:15.49 | jbot | Some people are never happy |
| 04:15.55 | Katty | hahaha |
| 04:15.57 | obnauticus | LMAO |
| 04:16.04 | Katty | who ever added that, kings to you. |
| 04:16.04 | obnauticus | ~asterisk16 |
| 04:16.05 | jbot | new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 04:16.05 | drmessano | Sorry, I had to |
| 04:16.11 | drmessano | I just did lol |
| 04:16.59 | drmessano | Im probably pushing it, but here's another |
| 04:16.59 | znoG | ManxPower: cool, got it to bind properly from Asterisk :) apart from configuring res_ldap.conf (which I've done, correctly by the look of it as it binds now) and extconfig.conf .. is there anything else that needs touching? |
| 04:17.03 | drmessano | ~asterisk20 |
| 04:17.04 | jbot | AWWWW, COME ON! |
| 04:17.15 | Katty | jbot: simmer down now. |
| 04:17.25 | Katty | jbot: much better. |
| 04:17.26 | obnauticus | ~asterisk30 |
| 04:17.34 | znoG | ManxPower: reason I ask is that when I try to login via SIP, nothing comes up in the LDAP log. Looks like Asterisk doesn't even try to query LDAP. |
| 04:17.34 | obnauticus | :\ |
| 04:17.42 | obnauticus | Foreshadowing -- There will not be an asterisk 3.0 |
| 04:17.55 | russellb | o.O |
| 04:18.07 | russellb | version numbering is arbitrary anyway |
| 04:18.14 | obnauticus | i'm just joking |
| 04:18.15 | obnauticus | silly |
| 04:18.16 | obnauticus | <3 |
| 04:18.25 | drmessano | Surely Asterisk will have a new name by the time 3.0 hits anyway.. it will be "Pound" or "Hash" by then |
| 04:18.27 | Qwell | russellb: 1.6.0 > 12.42.97 |
| 04:18.30 | *** join/#asterisk mattman99 (n=matt@b1F48.static.pacific.net.au) |
| 04:19.12 | drmessano | 1.6 is more of a 4.0 than most 4.0's are |
| 04:19.23 | obnauticus | wat? |
| 04:19.32 | drmessano | 1.6 is really 4.0 |
| 04:19.40 | obnauticus | Lies. |
| 04:19.51 | drmessano | If you go by the "Hey I added new stuff, lets call it 2.0" mentality |
| 04:20.09 | obnauticus | i increment my version numbers by the millionths place |
| 04:20.11 | obnauticus | every change i make |
| 04:20.13 | obnauticus | is a new version |
| 04:20.14 | drmessano | It's the 4th big release.. forget uncrements |
| 04:20.20 | drmessano | increments |
| 04:20.36 | Katty | sleepy. |
| 04:20.38 | Katty | i should go nap. |
| 04:20.45 | drmessano | If this was Microsoft, 1.6 would be 12.0 |
| 04:20.57 | obnauticus | No |
| 04:20.57 | Katty | g'later. |
| 04:21.02 | obnauticus | Microsoft doesn't release 12 |
| 04:21.05 | obnauticus | they release 6 |
| 04:21.06 | obnauticus | then give up |
| 04:21.14 | drmessano | Uh..Office is at 12 |
| 04:21.17 | obnauticus | STFU |
| 04:21.36 | TJNII | Pfft... ha! |
| 04:22.12 | drmessano | Microsoft Asterisk Live Telephony Communications Server 2008 |
| 04:22.16 | drmessano | There you go |
| 04:22.21 | drmessano | Version 13.0 |
| 04:24.38 | znoG | has anyone used res_config_ldap ? |
| 04:24.43 | znoG | i'm so close i can feel the love |
| 04:24.45 | znoG | :) |
| 04:24.55 | drmessano | znoG: It's getting old |
| 04:25.16 | znoG | the code for it? |
| 04:25.38 | znoG | or you mean the subject |
| 04:25.41 | drmessano | No, you seem to be the only one outside the development team that has even heard of it |
| 04:26.00 | znoG | "seem" is the right word. Plenty of people seem to be interested in it |
| 04:26.04 | drmessano | We're all skype users wishing for chan_skype to be a core module |
| 04:26.13 | znoG | pfft, skype |
| 04:26.18 | drmessano | So the topic is way over our heads |
| 04:26.30 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:26.30 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 04:26.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:29.00 | drmessano | Anyone else here on Twitter? |
| 04:37.31 | *** join/#asterisk gotnone (n=gotnone@jpinchak.dsl.xmission.com) |
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| 04:48.24 | lka | ls |
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| 04:50.21 | obnauticus | hey drmessano. |
| 04:50.22 | obnauticus | guess what |
| 04:51.28 | Hadi- | hey everyone |
| 04:54.32 | Hadi- | anyone here/. |
| 04:55.12 | obnauticus | No. |
| 04:55.47 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:56.35 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 05:02.47 | drmessano | What? |
| 05:03.18 | obnauticus | WAT |
| 05:03.32 | drmessano | [23:48] <obnauticus> hey drmessano. |
| 05:03.32 | drmessano | [23:48] <obnauticus> guess what |
| 05:03.34 | drmessano | ? |
| 05:03.36 | obnauticus | Nothing. |
| 05:03.42 | drmessano | oh |
| 05:03.48 | obnauticus | hey |
| 05:03.49 | obnauticus | guess what |
| 05:04.22 | Hadi- | i'm having issues with the Cisco 7960 IP phone using g729a codec on asterisk |
| 05:04.27 | Hadi- | anyone here can help? |
| 05:04.28 | Hadi- | :) |
| 05:04.35 | obnauticus | mine works on ulaw just fine |
| 05:05.15 | drmessano | Hadi-: Did you buy G729 licenses? |
| 05:05.29 | obnauticus | drmessano you don't need that |
| 05:05.32 | VJFROMGT | can someone help me solve Unable to support trunking on peer |
| 05:05.33 | obnauticus | there's a site with pre-compiled ones... |
| 05:05.40 | obnauticus | you MAY need a license to use them thoug |
| 05:05.41 | obnauticus | h |
| 05:05.41 | Hadi- | drmessano: yes i did |
| 05:05.42 | VJFROMGT | ViaZywal' without zaptel timing |
| 05:05.43 | obnauticus | i've used it without |
| 05:06.06 | Hadi- | keeps on losing audio |
| 05:06.11 | drmessano | Except it's not legal... and talking about it in here will only get you a black eye :/ |
| 05:06.14 | Hadi- | in the middle of the conversation |
| 05:06.22 | obnauticus | i don't give a shit drmessano. |
| 05:06.24 | obnauticus | the codec should be open |
| 05:06.25 | obnauticus | lol. |
| 05:06.32 | obnauticus | and i pay for licenses for it |
| 05:06.37 | obnauticus | just to put that out there. |
| 05:07.27 | J4k3 | g729 works, vista doesn't, and sadly I got *Stuck* owning a vista license (cuz I bought a modern laptop), but I actually did pay for my g729 licenses |
| 05:07.54 | obnauticus | i do too |
| 05:08.19 | obnauticus | but you can find precompiled modules for like every arcitecutre to encode transcode g729 |
| 05:08.26 | drmessano | TBH, obnauticus, your rant about G729 was irrelevant anyway.. The issue was whether not they had a G729 codec, because oddly enough some expect it to JUST WORK by adding an allow.. |
| 05:08.49 | obnauticus | :| |
| 05:08.51 | obnauticus | you serious? |
| 05:08.55 | obnauticus | I've never seeon someone do that |
| 05:09.20 | drmessano | Yes.. and question two is "If you're using the "free" one, how do you know it's even working" |
| 05:09.29 | drmessano | Because half the time, people download the wrong one |
| 05:09.34 | drmessano | and then blame Asterisk |
| 05:09.36 | drmessano | or the phone |
| 05:09.37 | drmessano | or both |
| 05:09.39 | J4k3 | there should be a special 'you selected g729 and it doesn't work' pre-encoded audio response |
| 05:09.45 | J4k3 | when you don't own/activate licenss |
| 05:09.48 | J4k3 | er licenses |
| 05:09.54 | drmessano | It's a nightmare when someone says "My G729 doesn't work" |
| 05:10.20 | obnauticus | Module Description Use Count |
| 05:10.20 | obnauticus | codec_g729-ast14-gcc4-glibc-pentium4.so g729 Coder/Decoder 0 |
| 05:10.31 | obnauticus | http://asterisk.hosting.lv/ |
| 05:10.42 | drmessano | "How do I know if I have a Pentium4?" |
| 05:10.44 | obnauticus | i stumbled across that |
| 05:10.49 | drmessano | "I think I have a 3" |
| 05:10.52 | J4k3 | err optimized builds |
| 05:10.53 | obnauticus | lol. |
| 05:11.01 | drmessano | "WUTS ICC?" |
| 05:11.07 | drmessano | Etc, etc |
| 05:11.08 | J4k3 | I CAN'T CRAP |
| 05:11.12 | obnauticus | wtf |
| 05:11.17 | obnauticus | drmessano i never see questions like that |
| 05:11.18 | obnauticus | but meuj |
| 05:11.20 | obnauticus | meh* |
| 05:11.26 | drmessano | Good for you |
| 05:11.29 | obnauticus | Is anyone here good at asterisk redundancy? |
| 05:11.29 | drmessano | I am so happy for you |
| 05:11.32 | drmessano | Really, happy |
| 05:11.48 | Hadi- | http://downloads.digium.com/pub/telephony/codec_g729/ |
| 05:11.48 | drmessano | Yay for obnauticus |
| 05:11.54 | Hadi- | i guess 1.2 is no longer supported? |
| 05:12.46 | Hadi- | so to make sure |
| 05:12.51 | *** join/#asterisk tsabi (n=tsabi@catv-50621e76.catv.broadband.hu) |
| 05:12.52 | drmessano | I hear different answers on that |
| 05:12.56 | Hadi- | the asterisk server is a dual xenon |
| 05:13.09 | Hadi- | whats the correct file to download |
| 05:13.14 | Hadi- | (dell 1850 that is..) |
| 05:15.53 | drmessano | P4 |
| 05:16.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:16.45 | obnauticus | drmessano ... |
| 05:16.55 | Hadi- | codec_g729a_v32_pentium4m.tar.gz? |
| 05:17.09 | Hadi- | this one |
| 05:17.56 | drmessano | yeah |
| 05:18.08 | drmessano | What? |
| 05:19.03 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584628.dsl.bell.ca) |
| 05:24.11 | Hadi- | again loss of audio |
| 05:24.16 | Hadi- | in the middle of a call |
| 05:24.21 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-54-169.lns10.syd7.internode.on.net) |
| 05:24.26 | Hadi- | only the person on the cisco phone is losing audio |
| 05:24.34 | Hadi- | the other party is hearing everything fine |
| 05:52.33 | *** join/#asterisk Olobola (i=Olobola@141.sub-75-209-38.myvzw.com) |
| 06:08.49 | Olobola | test |
| 06:09.08 | Olobola | test |
| 06:09.21 | cleone | wich test lol |
| 06:10.41 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 06:11.21 | BeeBuu | drmessano: help,please. |
| 06:12.03 | BeeBuu | i can't setup my 2 FXS port to work |
| 06:14.01 | drmessano | Is it a card or an ATA? |
| 06:14.46 | BeeBuu | yes |
| 06:15.13 | BeeBuu | zapata.conf |
| 06:15.39 | BeeBuu | only fxo port work,why? |
| 06:16.09 | drmessano | yes? |
| 06:16.09 | BeeBuu | i run genzaptelconf |
| 06:16.15 | drmessano | Is it a card or an ATA> |
| 06:16.16 | drmessano | ? |
| 06:16.26 | BeeBuu | a card |
| 06:16.29 | drmessano | ok |
| 06:16.37 | drmessano | Trixbox system? |
| 06:16.42 | BeeBuu | no |
| 06:16.45 | drmessano | ok.. |
| 06:16.47 | BeeBuu | asterisk |
| 06:16.51 | BeeBuu | only |
| 06:17.26 | drmessano | I've done very little with setting up cards.. im not a good person to ask |
| 06:17.43 | fujin | BeeBuu: have you got the signaling on your fxs ports right? |
| 06:17.47 | fujin | what does ztconfig -vv say |
| 06:18.10 | BeeBuu | Channel 03: FXO Loopstart (Default) (Slaves: 03) |
| 06:18.11 | BeeBuu | Channel 04: FXO Loopstart (Default) (Slaves: 04) |
| 06:18.41 | BeeBuu | fujin: maybe you can help me ,thanks. |
| 06:19.28 | fujin | and whats channel 01 02? |
| 06:19.42 | BeeBuu | FXS |
| 06:19.58 | BeeBuu | that are ok |
| 06:20.16 | BeeBuu | only FXS port doesn't work |
| 06:20.55 | BeeBuu | when i "zap show channels" under CLI |
| 06:21.18 | BeeBuu | only get 1 and 2 (FXO ports) show |
| 06:21.53 | fujin | is the signalling on your fxo ports right? |
| 06:22.27 | BeeBuu | signalling=fxo_ks |
| 06:22.35 | BeeBuu | channel=>3,4 |
| 06:22.46 | BeeBuu | is it correct? |
| 06:23.27 | *** part/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 06:23.28 | fujin | your fxo channels should have fxs signalling |
| 06:24.01 | BeeBuu | i did it |
| 06:24.35 | BeeBuu | signalling=fxs_ks |
| 06:24.35 | BeeBuu | channel=1,2 |
| 06:25.08 | BeeBuu | the problem is port 3 & 4 |
| 06:25.16 | fujin | ugh |
| 06:25.16 | fujin | no |
| 06:25.18 | fujin | stop being stupid |
| 06:25.21 | fujin | fxo ports have fxs signalling |
| 06:25.26 | fujin | fxs ports have fxo signalling |
| 06:25.27 | fujin | it's easy |
| 06:25.36 | fujin | configure your fxo ports to use fxs signalling, and viceversa |
| 06:25.47 | BeeBuu | yes,i did that. |
| 06:25.57 | fujin | pastebin your zapata.conf |
| 06:26.02 | JT | which ports aren't working? fxs or fxo? |
| 06:26.05 | BeeBuu | OK,please wait. |
| 06:26.18 | BeeBuu | FXS doesn't work. |
| 06:26.28 | BeeBuu | ~paste |
| 06:26.29 | jbot | well, paste is http://rafb.net/paste/, or see also pb |
| 06:26.44 | JT | BeeBuu: is the molex connector plugged into the card |
| 06:26.45 | JT | ? |
| 06:27.13 | BeeBuu | JT: plugged |
| 06:27.33 | fujin | sorry jt, I can't stand the lack of english, you can have him |
| 06:28.44 | BeeBuu | my zapata.conf http://rafb.net/p/nwV4rU31.html |
| 06:29.06 | drmessano | wtf |
| 06:29.24 | fujin | wow, fuck me |
| 06:29.26 | fujin | no wonder it doesn't work |
| 06:29.30 | drmessano | yeah |
| 06:29.37 | fujin | best fail ever |
| 06:29.38 | fujin | afk |
| 06:29.39 | drmessano | Im a ztdummy and it stands out |
| 06:29.41 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.35) |
| 06:29.43 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 06:29.52 | BeeBuu | fujin: any problem? |
| 06:29.55 | drmessano | lol |
| 06:29.57 | fujin | the whole damn file |
| 06:29.58 | BeeBuu | in my zapata.conf? |
| 06:30.03 | JT | ;channel=>3,4 |
| 06:30.12 | fujin | you haven't even configured port 3 and 4 |
| 06:30.13 | JT | BeeBuu: what is the semcolin for? |
| 06:30.14 | drmessano | Its as screwed up as a soup sandwich |
| 06:30.48 | BeeBuu | fujin: i add the ; |
| 06:31.08 | BeeBuu | just for now |
| 06:31.14 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
| 06:33.14 | BeeBuu | if remove the ; no "zap show channels" command at CLI! |
| 06:34.20 | BeeBuu | it will display: No such command 'zap show' when i type zap show channels |
| 06:34.39 | BeeBuu | asterisk 1.4.10 |
| 06:37.14 | JT | start asterisk like this: asterisk -cvvvvvvvvvvvvvvvvvvvv |
| 06:37.21 | JT | and look for the zaptel error |
| 06:38.06 | BeeBuu | JT:how to look for zaptel's error? |
| 06:38.29 | JT | read |
| 06:38.40 | BeeBuu | which file? |
| 06:38.43 | JT | look for any errors and warnings mentioning zap* |
| 06:38.45 | JT | the screen |
| 06:38.51 | JT | when you type asterisk -cvvvvvvvvvvvvvvvvvvvvvv |
| 06:39.39 | BeeBuu | get this: http://rafb.net/p/JrsdIa94.html |
| 06:42.02 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 06:42.11 | *** join/#asterisk oej (n=olle@p5485E7EC.dip.t-dialin.net) |
| 06:42.24 | BeeBuu | JT: how? |
| 06:43.11 | Olobola | where can I host asterisk? I tried installing on a godaddy virtual dedicated account and was unable to install |
| 06:43.17 | Olobola | I wasted several hours |
| 06:43.34 | Olobola | I need a host I know I can get asterisk up and running on |
| 06:45.00 | JT | Olobola: dedicated server or decent Xen VPS |
| 06:46.07 | obnauticus | err i have a server that has a trunk to a provider that's over g729, then to another peer which i.. want to use ulaw |
| 06:46.14 | obnauticus | for some reason it's still sending it over g729 |
| 06:46.21 | obnauticus | or i think so |
| 06:46.26 | obnauticus | how can i tell what codec the channel is using? |
| 06:46.33 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:46.51 | Olobola | thanks |
| 06:47.04 | BeeBuu | JT: help please. |
| 06:48.06 | obnauticus | Capabilities: us - 0x103 (g723|gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) |
| 06:48.50 | *** join/#asterisk Stochastik (n=irssi@204.246.139.68) |
| 06:51.03 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 06:53.15 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 06:53.48 | tengulre | hi,all |
| 06:56.21 | obnauticus | Capabilities: us - 0x103 (g723|gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) |
| 06:56.22 | obnauticus | oops |
| 06:56.23 | obnauticus | sorry |
| 07:01.02 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.193) |
| 07:01.07 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:03.27 | *** join/#asterisk whymarkwhy (n=koos@196.211.34.2) |
| 07:03.53 | whymarkwhy | anyone please give me the link to download latest asterisk 1.4 book please |
| 07:04.24 | *** join/#asterisk whymarkwhy (n=koos@196.211.34.2) |
| 07:04.29 | whymarkwhy | anyone please give me the link to download latest asterisk 1.4 book please |
| 07:04.50 | whymarkwhy | sorry link droped don't know if you got first requist |
| 07:05.04 | whymarkwhy | is this link up? |
| 07:05.33 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 07:08.07 | whymarkwhy | exit |
| 07:09.09 | *** part/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 07:09.16 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 07:09.20 | obnauticus | JT can you help me real quick |
| 07:09.22 | obnauticus | with something pretty simple. |
| 07:10.46 | obnauticus | I have no idea why it's diong this |
| 07:11.44 | obnauticus | me <--> i have my box <--(want gsm and/or ulaw)--> Provider boxen (i set it up) <--(g729)--> PSTN |
| 07:12.02 | obnauticus | for some reason it's not reencoding into ulaw or gsm |
| 07:12.20 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 07:12.46 | JT | so it has g.729 licences yes? |
| 07:12.52 | obnauticus | The provider boxen does. |
| 07:13.22 | obnauticus | Or at least i think so |
| 07:13.27 | obnauticus | it IS running g729... |
| 07:13.30 | obnauticus | ya it is... don't worry |
| 07:13.54 | JT | hmm |
| 07:14.03 | obnauticus | at least i think so |
| 07:17.52 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.35) |
| 07:19.07 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 07:19.51 | obnauticus | JT well i think the server is offering g729 first |
| 07:19.52 | obnauticus | or something |
| 07:19.58 | obnauticus | and my * box isn't hearing anything |
| 07:20.08 | obnauticus | like, it doesn't say `can't find the codec' or anything, i just get 1 way audio |
| 07:20.26 | obnauticus | which in the model i posted aboe is from the user "me" the caller |
| 07:22.59 | *** join/#asterisk maniac777 (n=maniac@ppp-2-149.pppoe.sistbg.net) |
| 07:29.30 | mattman99 | obnauticas, is your box behind NAT? |
| 07:29.44 | obnauticus | yes, but i set nat=yes, and when ic all in with my cell phone it works fine |
| 07:29.54 | obnauticus | with the same config from a diffeernt host for my DID |
| 07:30.09 | mattman99 | have you set your localnet and externalIP in sip_custom.conf |
| 07:30.13 | obnauticus | yes. |
| 07:30.18 | mattman99 | ok |
| 07:30.19 | obnauticus | all of that is fine |
| 07:30.22 | obnauticus | all of the other stuff works |
| 07:30.24 | obnauticus | it's thecodec's |
| 07:30.25 | obnauticus | the codec's* |
| 07:30.31 | mattman99 | oh |
| 07:31.28 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-171-72.dsl.tul2ok.sbcglobal.net) |
| 07:33.45 | *** join/#asterisk nofish2 (i=port@61-69-2-27.netspeed.com.au) |
| 07:35.22 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 07:36.02 | Lsodi | Greetings! any cisco voip experts here? |
| 07:36.39 | *** join/#asterisk oej (n=olle@p5485E7EC.dip.t-dialin.net) |
| 07:37.48 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:43.51 | *** join/#asterisk Tili (n=tili@83.68.29.184) |
| 07:46.37 | BeeBuu | fujin & JT ,the problem is fixed,thanks. |
| 07:52.09 | *** join/#asterisk bkw_ (n=brian@adsl-75-50-21-183.dsl.tul2ok.sbcglobal.net) |
| 07:54.21 | JT | BeeBuu: how? |
| 07:56.10 | BeeBuu | modprobe wcfxs |
| 08:00.40 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:01.35 | obnauticus | Is there any way to limit the total minutes on a trunk on asterisk? |
| 08:01.47 | obnauticus | like say i get 300 free minutes on sip channel a... and 300 on b... |
| 08:02.02 | obnauticus | say i use up 300minutes on channel a, how can i make it automatically failover to channel b after 300minutes? |
| 08:06.17 | JT | obnauticus: database keeping tabs on time probably |
| 08:10.53 | obnauticus | JT + agi script? |
| 08:11.10 | JT | not necessary |
| 08:11.17 | obnauticus | how would one be able to do that |
| 08:11.49 | obnauticus | during that switch their call would get cut off though |
| 08:12.08 | JT | yes |
| 08:20.16 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 08:27.01 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 08:29.09 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:31.21 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:38.41 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 08:40.01 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 08:40.01 | Lsodi | <--- SIP read from 10.198.1.1:5060 ---> |
| 08:40.01 | Lsodi | SIP/2.0 100 Trying |
| 08:40.07 | ice_croft | hi all |
| 08:40.13 | Lsodi | <--- SIP read from 10.198.1.1:5060 ---> |
| 08:40.13 | Lsodi | SIP/2.0 501 Not Implemented |
| 08:40.22 | ice_croft | i'm connectin two *s via sip |
| 08:40.37 | obnauticus | yes yes!??!?!?! |
| 08:40.37 | ice_croft | http://pastebin.ca/930107 |
| 08:40.56 | Lsodi | what that could mean, that 10.198.1.1 doesnt have any sip trunk configured? |
| 08:41.05 | ice_croft | when im callin from *1 sip client to *2 sip, im gettin this |
| 08:41.28 | obnauticus | it says it on the top |
| 08:41.33 | obnauticus | authentication required. |
| 08:41.57 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:42.07 | ice_croft | emm |
| 08:42.17 | obnauticus | +proxy |
| 08:42.25 | ice_croft | im callin to *2 group |
| 08:42.39 | obnauticus | is it successfully registered? |
| 08:42.43 | ice_croft | and when im callin *2 sip , it's okay |
| 08:42.49 | obnauticus | oh i have no idea then |
| 08:42.49 | ice_croft | yes, succesfully |
| 08:43.07 | obnauticus | drmessano you there? |
| 08:43.33 | ice_croft | damn |
| 08:45.20 | Lsodi | is problem with asterisk or with cisco device http://pastebin.com/d96de83d ? |
| 08:48.01 | *** join/#asterisk bantu (n=Miranda@p54A33EAF.dip0.t-ipconnect.de) |
| 08:52.13 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:06.12 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:06.54 | *** join/#asterisk NirS_ (n=NirS@87.68.17.141.cable.012.net.il) |
| 09:12.27 | ice_croft | still need help |
| 09:13.15 | ice_croft | SIP/2.0 407 Proxy Authentication Required |
| 09:13.24 | ice_croft | where should i auth? |
| 09:13.36 | ice_croft | i authed everywhere |
| 09:13.54 | ice_croft | obnauticus> now there no any calls :( |
| 09:14.03 | ice_croft | obnauticus> regs a good |
| 09:15.40 | ice_croft | ~book |
| 09:15.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 09:19.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:21.20 | ice_croft | what does mean "ACL" col in "sip show peers"? |
| 09:29.04 | *** join/#asterisk Curus (n=Curus@hd5b9080a.c45-01-12.sta.perspektivbredband.net) |
| 09:29.08 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.35) |
| 09:29.56 | endre | access control list |
| 09:31.13 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 09:37.05 | *** join/#asterisk af_ (n=getsmart@88-149-240-207.dynamic.ngi.it) |
| 09:42.36 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 09:44.11 | *** join/#asterisk hi365 (n=hi365@bzq-79-183-140-55.red.bezeqint.net) |
| 09:45.00 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 09:45.19 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 09:46.24 | jeanmi_i_ | hi |
| 09:46.52 | jeanmi_i_ | I am using Kapanga with Asterisk 1.6 as I'd like to test the SIP over TLS and SRTP feature |
| 09:47.11 | jeanmi_i_ | Kapanga can register with asterisk |
| 09:47.33 | jeanmi_i_ | however when I try making a call, asterisk is complaining about a line in the sdp body |
| 09:47.47 | jeanmi_i_ | m=audio 5100 RTP/SAVP 0 8 101 |
| 09:48.09 | jeanmi_i_ | is this line OK or not ? |
| 09:48.15 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:48.25 | jeanmi_i_ | I would just like to know who is wrong (kapanga or asterisk) |
| 09:54.04 | Curus | Is that in a 180 response or a 183 response or in something else? |
| 09:54.33 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de) |
| 09:54.43 | yang | I am wondering about loading TAPI on asterisk |
| 09:54.47 | yang | Ssh into your Asterisk box and "asterisk -gcvvvr" now run your TSP driver by loading whatever TAPI software you are using you should see your manager logon |
| 09:54.48 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 09:55.13 | yang | which command would load TAPI ? |
| 10:02.02 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:02.44 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 10:03.14 | *** join/#asterisk mattman99 (n=matt@ppp121-44-207-170.lns3.mel4.internode.on.net) |
| 10:07.54 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 10:07.59 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 10:18.48 | *** part/#asterisk BeeBuu (n=beebuu@59.38.97.193) |
| 10:19.49 | *** join/#asterisk siya (n=djerk@194.60.207.239) |
| 10:21.11 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:25.51 | ice_croft | hi all |
| 10:26.30 | ice_croft | http://pastebin.ca/930168 |
| 10:26.51 | ice_croft | im tryin to dial a group from remotely connected * |
| 10:27.08 | ice_croft | how should i "route" the call correctly? |
| 10:29.35 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 10:39.37 | *** join/#asterisk Eagleray (n=erayd@osiris.erayd.net) |
| 10:40.16 | Eagleray | How can I create an "exten => ...." line that will execute for every call in the context, regardless of other rules? |
| 10:40.30 | Eagleray | The goal is to run System() against an email log script |
| 10:52.27 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 10:53.51 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-b0d1436b389acf81) |
| 10:54.26 | dandre | Hello, |
| 10:54.58 | dandre | Is there any opensource alternative to the standard voicmail? |
| 10:56.25 | mvanbaak | Eagleray: exten => _X.,1,System() |
| 10:56.39 | mvanbaak | Eagleray: and make all the other rules start at priority 2 |
| 10:57.40 | *** join/#asterisk kamisalami (n=kami@84.244.141.35) |
| 11:02.02 | *** join/#asterisk nighty^ (n=nighty@p2071-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 11:04.49 | tzafrir | dandre, what do you require of such an alternative? |
| 11:05.00 | tzafrir | have you looked at minivm? |
| 11:07.07 | dandre | no I didn't know |
| 11:07.07 | dandre | My main point is that announces and prompt are not smouth enough because they are cut into small parts. |
| 11:07.07 | dandre | And also for internationnal prompts they are not always grammatically correct |
| 11:08.39 | dandre | MiniVM: Mini-Voicemail (new in v1.6) |
| 11:08.45 | dandre | I am still using 1.4 |
| 11:09.45 | *** part/#asterisk jeanmi_i_ (n=besnard@LMontsouris-152-61-20-55.w80-13.abo.wanadoo.fr) |
| 11:10.04 | *** join/#asterisk jeanmi_i_ (n=besnard@LMontsouris-152-61-20-55.w80-13.abo.wanadoo.fr) |
| 11:10.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 11:11.09 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 11:11.50 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:13.18 | *** part/#asterisk rsc (n=robert@fedora/rsc) |
| 11:13.35 | ice_croft | ergh |
| 11:13.40 | ice_croft | new thing |
| 11:17.30 | *** join/#asterisk donadie (n=tfuente@82.108.46.35) |
| 11:18.34 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 11:30.28 | yang | hey tzafrir What is the status of Asttapi does it need to be downloaded and compiled fro sourceforge? |
| 11:31.08 | *** join/#asterisk adjohn (n=adjohn@nttkyo573240.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:31.23 | agx | yang, isn't that great, you can do just outgoing calls |
| 11:31.57 | yang | outgoing? |
| 11:32.19 | *** join/#asterisk a1200 (n=lamer@dns.it-help.cz) |
| 11:32.50 | a1200 | hi |
| 11:32.53 | *** join/#asterisk adjohn (n=adjohn@nttkyo573240.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 11:33.41 | a1200 | i would ask you, how i can set Callerid in AGI? (asterisk 1.4) |
| 11:35.13 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 11:37.32 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:41.01 | *** join/#asterisk simbol76ss (n=simbol@host228-198-dynamic.49-82-r.retail.telecomitalia.it) |
| 11:43.56 | flush | morning |
| 11:44.13 | nixguy | flush: morning for you is after luncthime for me! |
| 11:44.16 | nixguy | but morning |
| 11:44.21 | flush | arrr |
| 11:44.24 | flush | sir |
| 11:44.37 | tsabi | flush: morning :D its 1 pm here: lunch time!! :DD |
| 11:44.37 | flush | did you ever configure asterisk for use with analog phones only |
| 11:44.48 | flush | you lucky im going to courses |
| 11:45.34 | nixguy | flush: how do you mean analog phones only? and why would one want analog phones? |
| 11:46.46 | flush | i dont have voip services |
| 11:47.08 | flush | i just want to set asterisk as a answering machine and music on hold with maybe some other cool tricks you know |
| 11:48.11 | flush | i have alarms "OK" with zttool, i have my fxo module plugged in the wall jack and my phone plugged in my fxs module and when i answer the phone i have a dial tone, but as soon as i hit a key i get the fast busy tone |
| 11:50.58 | ice_croft | i have ast 1.4 on freebsd, and putty as ssh-term |
| 11:51.17 | ice_croft | how can i make color logs in asterisk console? |
| 11:51.20 | ice_croft | help plz |
| 11:52.10 | Dovid | does anyone here use patton ? |
| 11:52.18 | a1200 | try asterisk -rc |
| 11:52.48 | a1200 | or stop asterisk and try start as console - asterisk -gc |
| 11:54.24 | *** join/#asterisk trnzmeta (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 11:54.33 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:54.46 | ice_croft | a1200> ok |
| 11:56.42 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 11:57.11 | ice_croft | a1200> i did |
| 11:57.20 | ice_croft | a1200> start asterisk -g |
| 11:57.28 | ice_croft | a1200> then asterisk -rc |
| 11:57.31 | ice_croft | a1200> and now color |
| 11:57.33 | ice_croft | a1200> and no color |
| 11:57.46 | ice_croft | a1200> then asterisk -gc was good |
| 11:57.54 | ice_croft | a1200> color present |
| 11:58.10 | *** join/#asterisk trnzmeta (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 11:59.03 | *** join/#asterisk trnzmeta (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 12:02.09 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 12:02.37 | marc7 | what the heck could be causing a Cisco 7960 (or any IP phone) to be returning a "SIP/2.0 404 Not Found" on an incoming call? |
| 12:03.29 | *** join/#asterisk Nuttah (n=andrew@amber2.interdart.co.uk) |
| 12:03.44 | Nuttah | hey guys. |
| 12:05.12 | DarKnesS_WolF | marc7: context |
| 12:05.51 | ice_croft | so any advices? |
| 12:06.29 | defswork | anyone know expected release data of new aastra 2.2 firmware ? |
| 12:06.54 | a1200 | ice_croft try asterisk -cgvvvv |
| 12:06.55 | marc7 | wolf: the phone's internal context? the dialplan is fine, it's just that *sometimes* the phone, on an incoming call, returns 404, and asterisk reports the phone as busy/congested. |
| 12:07.01 | a1200 | as more verbose |
| 12:07.15 | a1200 | and avter try start asterisk as daemon |
| 12:07.25 | Nuttah | Doe anyone have an idea why on asterisk 1.2.10 (using freepbx 2.1.1) I get "all-circuits-busy-now" responses periodically on calls that come back with Congestion,Busy and Channel unavailable (probably others too tbh)? |
| 12:07.37 | *** join/#asterisk barhom (n=barhom@h-89-233-195-211.wholesale.rp80.se) |
| 12:07.48 | defswork | Nuttah: thats the default behaviour |
| 12:08.04 | marc7 | wolf: i'm gonna keep an eye on this, i'll keep you guys posted. |
| 12:08.17 | Nuttah | ok. Is there an easy way to get that to give the tonal responses instead? |
| 12:08.43 | defswork | priindication=outofband |
| 12:08.43 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 12:08.57 | RoyK | l |
| 12:08.59 | RoyK | ops |
| 12:09.03 | *** part/#asterisk RoyK (n=roy@box36.fortel.no) |
| 12:09.05 | defswork | Nuttah: that ^^ |
| 12:09.49 | Nuttah | Defswork: oh this is completely over IAX2. I am assumming that goes in zapata.conf? |
| 12:10.00 | ice_croft | a1200> it works well, but i need * as a daemon and to connect to via asterisk -r |
| 12:10.13 | defswork | Nuttah: yeah - but it won't help on IAX though I doubt |
| 12:10.22 | Nuttah | yeah thats what I thought. |
| 12:11.34 | DarKnesS_WolF | ice_croft: go to the source code |
| 12:11.36 | DarKnesS_WolF | and do make config |
| 12:11.48 | DarKnesS_WolF | it should generate the init scripts to start asterisk |
| 12:11.59 | Dovid | is there any guide to getting faxing to work on asterisk |
| 12:12.38 | a1200 | ice_croft i understand |
| 12:13.07 | a1200 | ice_croft try at 1st connect as on console: asterisk -gcvvvv (+4 lvl verbosity) |
| 12:13.17 | a1200 | ice_croft stop * as console |
| 12:13.24 | a1200 | ice_croft and start as daemone |
| 12:13.30 | a1200 | ice_croft daemon |
| 12:14.22 | ice_croft | a1200> why -g? |
| 12:14.49 | ice_croft | a1200> this line "asterisk -gcvvvv" shows the color. asterisk -r -- no. |
| 12:15.28 | ice_croft | a1200> ^( |
| 12:15.32 | a1200 | ice_croft> :( |
| 12:15.45 | ice_croft | a1200> >_< |
| 12:15.55 | ice_croft | okay then |
| 12:15.57 | a1200 | try -n |
| 12:16.05 | a1200 | on starting |
| 12:16.13 | a1200 | no sory |
| 12:16.14 | a1200 | no :) |
| 12:16.16 | ice_croft | :) |
| 12:16.22 | ice_croft | kh |
| 12:16.26 | ice_croft | huh |
| 12:16.49 | a1200 | forget :) |
| 12:16.50 | ice_croft | well |
| 12:17.15 | ice_croft | gonna suicide by hittin a wall |
| 12:17.42 | a1200 | lol |
| 12:18.06 | a1200 | how i can set callerid on manager? |
| 12:18.12 | a1200 | with * 1.4 ? |
| 12:24.36 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:26.07 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:30.39 | ice_croft | does somebody runnin * on freebsd? |
| 12:31.40 | *** join/#asterisk stoffell (n=stoffell@148.192-136-217.adsl-dyn.isp.belgacom.be) |
| 12:32.48 | *** join/#asterisk cy3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 12:32.49 | cy3o3 | werd |
| 12:32.59 | *** join/#asterisk simbol76ss (n=simbol@host228-198-dynamic.49-82-r.retail.telecomitalia.it) |
| 12:33.13 | stoffell | can anyone confirm the patched bristuff (v0.4.0test4 or higher) to support beronet cards? |
| 12:36.05 | *** join/#asterisk pondiboy (n=Pondiboy@122.164.179.62) |
| 12:37.35 | barhom | what ping times are considered "enough" for normal talk. Im trying to do a link between PSTN->asterisk->asterisk in another country->PSTN |
| 12:37.56 | barhom | Im guessing the ping time between asterisk->asterisk must be good for the quality to be "okay" |
| 12:38.33 | ThoMe | the cdr entry is written AFTER the call? |
| 12:38.48 | marc7 | ThoMe: that's correct |
| 12:39.22 | ice_croft | hi all |
| 12:39.26 | marc7 | ThoMe: there are some ${CDR} variables that aren't available until the call's nearing completion |
| 12:39.39 | ice_croft | can i apply #include several times in sip.conf? |
| 12:39.47 | marc7 | ice_croft: you betcha! |
| 12:40.05 | ThoMe | marc7: hm. ok. i have a question |
| 12:40.23 | ice_croft | marc7> same file several times in several places? |
| 12:40.23 | ThoMe | marc7: incomming calls are put in the queue first. An agent always picks up and realizes "not for me, im transfering". Can i see in the CDR database where he refered the call to? |
| 12:40.29 | ThoMe | marc7: i can see where he refered the call to.. but not where have recieve the call |
| 12:41.06 | marc7 | ice_croft: errr, you may not want to include the same file multiple times in sip.conf unless you have a method to your madness |
| 12:41.40 | *** join/#asterisk quigon (n=matias@32.59.64.129) |
| 12:41.47 | marc7 | ThoMe: right, another CDR record is created for that... tricky! |
| 12:42.12 | ThoMe | marc7: tricky? hm? |
| 12:44.45 | marc7 | ThoMe: that's a little out of my league. I could suggest setting a common value to ${CDR(userfield)} of both records, but I'd recommend throwing that question to some of the heavier hitters in the channel when they all wake up |
| 12:46.14 | marc7 | you miiight also be able to gleam the right details from ${CDR(lastapp)} |
| 12:47.07 | marc7 | and lastdata |
| 12:47.44 | nixguy | can i chain multiple audiofiles in playback somehow? |
| 12:47.56 | nixguy | instead of having them one line below the other |
| 12:49.07 | nixguy | like playback(my/name/is/sam) |
| 12:49.09 | nixguy | instead of |
| 12:49.14 | nixguy | playback(my) |
| 12:49.17 | nixguy | playback(name) |
| 12:49.19 | nixguy | playback(is) |
| 12:49.21 | nixguy | playback(sam) |
| 12:50.18 | marc7 | you might be able to accomplish something like that with a macro |
| 12:51.07 | *** join/#asterisk budol (i=bezpu@202.124.138.70) |
| 12:51.21 | marc7 | Macro(chainplayback,my,name,is,sam) |
| 12:51.38 | nixguy | marc7: ok thnx ill look into macros |
| 12:53.48 | znoG | anyone using/used res_agent_ldap? |
| 12:53.54 | marc7 | also, looking into Background(), it appears like you can call it like so: Background(filename1&filename2&filename3) |
| 12:53.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:54.04 | marc7 | nixguy: so you may want to try that syntax with Playback |
| 12:54.52 | nixguy | marc7: ahh even easier i will try it right away |
| 12:54.54 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.117.5) |
| 12:55.06 | nixguy | i was looking into the background for such a feauture but must have missed it? |
| 12:55.20 | marc7 | http://www.voip-info.org/wiki-Asterisk+cmd+Background |
| 12:55.50 | marc7 | that subtle [&filename2] is what suggested it might be an additional option |
| 12:57.03 | marc7 | znoG: what are you using res_agent_ldap for? |
| 12:59.31 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 13:00.32 | znoG | marc7: to query the LDAP backend for sip/iax users |
| 13:01.46 | marc7 | znoG: i'm having trouble finding anything on google. I'm interested in implementing something like that myself, which is why I'm interested |
| 13:02.41 | znoG | marc7: well, i started by downloading the latest beta of asterisk |
| 13:03.09 | marc7 | now that you mention it, I did see it in the 1.6 changelog, so i'm sure it hasn't had enough time to filter down to google |
| 13:03.22 | znoG | marc7: which you can use --with-ldap to compile res_config_ldap |
| 13:03.43 | znoG | marc7: configuration is fiddly, and personally I can't get it to work, but you might have more luck than me :) |
| 13:05.12 | marc7 | znoG: it's cool to see people are experimenting with it. if you can reproduce specific problems, you might want to ask in #asterisk-dev in a few hours when everybody wakes up |
| 13:06.17 | nixguy | actually we have our own asterisk and ldap implementation |
| 13:06.18 | znoG | marc7: i sent a message to the list and got one reply. Unfortunately I'm not so sure many people use LDAP |
| 13:06.23 | nixguy | i could share the code if you want it |
| 13:06.25 | budol | !ping |
| 13:06.32 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:06.38 | nixguy | it uses pythong and ldap retriive and the #include statement in sip.conf |
| 13:06.40 | znoG | nixguy: actually if you implemented it yourself, I just want to find out a couple of things |
| 13:06.49 | nixguy | so its not entierly integrated but works fine for our needs |
| 13:06.58 | nixguy | i didnt write it but my colleauge did |
| 13:07.02 | nixguy | the code i can give you |
| 13:07.10 | znoG | nixguy: so you basically fetch stuff from LDAP and write the config files with it? |
| 13:07.18 | nixguy | yup |
| 13:07.21 | znoG | ohh i see |
| 13:07.27 | nixguy | we use Active Directory |
| 13:07.38 | nixguy | and just fetch username and telephone number basically |
| 13:07.41 | znoG | i'm looking at getting res_config_ldap to work, which is realtime, although your solution isn't bad. |
| 13:07.42 | nixguy | we dont use password auth |
| 13:08.00 | znoG | i mean, it does the job I guess but I prefer to use the realtime module |
| 13:08.09 | nixguy | the script can be run quite intensly it diffs for changes so it doesnet really do anything to asterisk or the config files unless something has changed... |
| 13:08.24 | znoG | yea, sounds like what Asterisk::LDAP does |
| 13:08.31 | znoG | although it doesn't support sip.conf nor iax.conf |
| 13:08.40 | nixguy | znoG: yup i agree with you, this though allowed us to use 1.2 (i want stable) and i dont want to compile stuff from source :) |
| 13:08.51 | znoG | fair enough |
| 13:08.54 | marc7 | 1.2... the horror! |
| 13:08.59 | nixguy | marc7: why? :) |
| 13:09.00 | marc7 | (kidding) |
| 13:09.02 | nixguy | okok |
| 13:09.03 | nixguy | :) |
| 13:09.11 | nixguy | beeing half a noob when it comes to asterisk you scared me |
| 13:09.25 | nixguy | its my first implementation ever of * |
| 13:09.30 | znoG | im only using 1.6.x cause of res_config_ldap, otherwise I'd probably be on 1.4.x |
| 13:09.36 | nixguy | eventhough i'm fairly experienced when it comes to linux |
| 13:09.40 | marc7 | nah, I have two production servers that are still 1.2 and I haven't gotten around to rebuilding them yet |
| 13:09.46 | nixguy | im ok |
| 13:09.46 | marc7 | solid as a rock |
| 13:09.49 | nixguy | with 1.2 |
| 13:09.56 | nixguy | we dont need that many features prefer stability |
| 13:10.03 | nixguy | and the support of using binaries from a distribution |
| 13:10.11 | nixguy | getting oh_323 to work |
| 13:10.16 | nixguy | only took a couple of hours |
| 13:10.37 | nixguy | (bridgint to our old proprietary pbx) |
| 13:10.40 | nixguy | bridging |
| 13:11.34 | znoG | at least your proprietary PBX does h323 |
| 13:11.48 | znoG | i had to integrate an old PBX with asterisk using a TDM2400B card |
| 13:12.01 | nixguy | znoG: unfortunately we could only the entire migrate the public number series in one go |
| 13:12.11 | nixguy | so were not even gonna use the h323 trunk now... |
| 13:13.04 | ThoMe | marc7: when there is an incomming call that gets accepted and transferd, can i somehow log the transfer part and maybe write it in the "userfield"? |
| 13:13.38 | marc7 | znoG: depends how deep you want to get into the whole LDAP thing. I'd encourage sticking with it. tilghman and russell are the two people who seem to be doing most of the commits to res_config_ldap in SVN trunk... so again, keep your eyes peeled for them if you have very specific questions about what isn't working. |
| 13:13.46 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:14.12 | marc7 | again, community effort helps everyone :) |
| 13:14.49 | znoG | marc7: yep, i saw russell making a lotta commits on it. He was around yesterday but he didn't answer so I guess he either wasn't in the mood, or didn't see me :) |
| 13:17.43 | marc7 | ThoMe: my 7970's been giving me problems lately, i'm having trouble transferring a call myself to see exactly how it shows up on the CDR. i know what you're trying to accomplish, but couldn't give you an answer off the top of my head. |
| 13:18.03 | marc7 | and I have to be up in 3 hours... |
| 13:18.47 | ice_croft | somebody explain me |
| 13:18.51 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:18.52 | marc7 | so best of luck! |
| 13:18.52 | ice_croft | two lines |
| 13:19.01 | ice_croft | include -> ext-local |
| 13:19.09 | ice_croft | include => ext-group |
| 13:19.14 | marc7 | the second one |
| 13:19.23 | ice_croft | include -> ext=local |
| 13:19.41 | ice_croft | when this is the order = works fine |
| 13:19.59 | ice_croft | when ext-group is the last - i can't dial it |
| 13:20.04 | ice_croft | help me |
| 13:20.22 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 13:21.48 | marc7 | ice_croft: I don't think -> is valid. => is accurate |
| 13:22.02 | ice_croft | marc7> its a mistypin |
| 13:22.10 | ice_croft | marc7> :) |
| 13:23.09 | ice_croft | please |
| 13:23.15 | marc7 | ice_croft: you'll need to be much more descriptive than "i can't dial it"... can you pastebin the error in the console/log, the relevant lines in extensions.conf, etc. |
| 13:23.26 | ice_croft | okay |
| 13:23.35 | marc7 | with that info, someone else should be able to help. i gotta crash. g'nite. |
| 13:24.30 | stoffell | can anyone confirm the patched bristuff (v0.4.0test4 or higher) to support beronet cards? |
| 13:27.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:29.08 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 13:35.38 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:43.00 | *** join/#asterisk itzbipin (n=bipin@122.167.55.13) |
| 13:43.18 | budol | hello guys |
| 13:43.19 | itzbipin | hello anybody there |
| 13:43.37 | itzbipin | hi |
| 13:43.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:44.06 | itzbipin | hello can u hear me |
| 13:44.10 | budol | anyone know how to configure the quintum to asterisk box? |
| 13:47.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:47.57 | itzbipin | nope |
| 13:48.12 | itzbipin | have u ever done conference in asterisk |
| 13:48.28 | [TK]D-Fender | itzbipin: "core show application meetme" <- |
| 13:48.29 | itzbipin | i have a doubt in it |
| 13:48.38 | itzbipin | yes |
| 13:48.40 | itzbipin | i know it |
| 13:49.01 | itzbipin | who we can invite a user to join in to our conference room |
| 13:49.13 | itzbipin | sory how we can |
| 13:50.30 | itzbipin | suppose i am already in the confrence and i want to invite my friend to join |
| 13:50.44 | budol | [TK]D-Fender : can you help me with my question? |
| 13:50.55 | itzbipin | then what i have to do. |
| 13:51.20 | [TK]D-Fender | itzbipin: get them to call in and call Meetme. |
| 13:51.35 | *** join/#asterisk el_4_jinete (n=gustavo@mail.pulxar.com.co) |
| 13:51.40 | *** join/#asterisk Stian- (n=stian@062016138186.customer.alfanett.no) |
| 13:51.50 | ThoMe | [TK]D-Fender: hello. is it posible to show the values fo ${CDR} if i recieve a call? |
| 13:52.00 | [TK]D-Fender | ThoMe: Huh? |
| 13:52.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:52.17 | ThoMe | i would view alle infos in ${CDR |
| 13:52.23 | itzbipin | nope..i want to make call to them for the invitation |
| 13:52.46 | ThoMe | but not after the call, but if i |
| 13:52.49 | itzbipin | then wat to do |
| 13:52.51 | ThoMe | call |
| 13:52.59 | ThoMe | itzbipin: you mean me? |
| 13:53.06 | ThoMe | no :) |
| 13:53.08 | [TK]D-Fender | itzbipin: then just take your phone, call them, and transfer them to a MeetMe exten. |
| 13:53.11 | ThoMe | [TK]D-Fender: any ideas? :-) |
| 13:53.27 | [TK]D-Fender | ThoMe: "core show function CDR" |
| 13:54.16 | ThoMe | [TK]D-Fender: if i go to the asterisk-console, how i can output the variable? |
| 13:54.16 | itzbipin | transfer?? how can i |
| 13:54.28 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 13:55.11 | stoffell | tzafrir, do you know if the newest bristuff (v0.4.0test4 or higher) really supports beronet cards? (as read in the changelog) |
| 13:55.15 | itzbipin | transfer to my asterisk server |
| 13:56.06 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 13:56.54 | itzbipin | when i just the call will they enter in the conference room |
| 13:57.02 | itzbipin | ?? |
| 13:57.21 | x86 | itzbipin: english, please |
| 13:57.48 | itzbipin | when i just make call to my fnd will they enter in the conference room |
| 13:57.52 | [TK]D-Fender | ThoMe: In console? No idea. |
| 13:58.44 | *** join/#asterisk graphx (n=jeff@rrcs-67-79-144-150.se.biz.rr.com) |
| 13:58.50 | ThoMe | [TK]D-Fender: i mean for debug |
| 13:58.51 | [TK]D-Fender | itzbipin: you take your PHONE which is connected to *, you CALL the other person. After they answer, jsut TRANFER the call like any other call to an exten that calls MeetMe. |
| 13:58.55 | graphx | good morning all |
| 13:58.55 | ThoMe | NOHO or so? |
| 13:59.01 | ThoMe | good morning? |
| 13:59.09 | ThoMe | 15 o'clock |
| 13:59.10 | ThoMe | 3 pm |
| 13:59.11 | ThoMe | ;) |
| 13:59.14 | graphx | :) |
| 13:59.15 | znoG | morning ... 11:53 here |
| 13:59.16 | [TK]D-Fender | ThoMe: Same answer. You can look at a CHANNEL in CLI, but not CDR in any way I'm aware of. |
| 13:59.29 | graphx | it's 9:00am here |
| 13:59.50 | ThoMe | [TK]D-Fender: ok. |
| 13:59.56 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 14:00.11 | ThoMe | [TK]D-Fender: incomming calls are put in the queue first. An agent always picks up and realizes "not for me, im transfering". Can i see in the CDR database where he refered the call to? i can see where he refered the call to.. but not where have recieve the call |
| 14:00.15 | [TK]D-Fender | ThoMe: go look at "show channel [channel]" |
| 14:00.35 | ThoMe | [TK]D-Fender: when there is an incomming call that gets accepted and transferd, can i somehow log the transfer part and maybe write it in the "userfield"? |
| 14:00.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:00.48 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:01.13 | itzbipin | thank you will try ..where r u from |
| 14:01.16 | [TK]D-Fender | ThoMe: You'd have to add detection code into ALL of your extens. This would be crazy |
| 14:01.37 | ThoMe | detection code ? |
| 14:02.00 | ThoMe | what is it? |
| 14:02.54 | [TK]D-Fender | ThoMe: When you transfer a call, its a completely new call. it would start at the exten you transfer to, but will have inherited the vars from the transferring channel. You'd have to add some dialplan onto every exten they could dial that would check to see if it was trasnferred from a Queue call, etc. |
| 14:03.05 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:03.44 | [TK]D-Fender | itzbipin: Quebec, Canada |
| 14:04.04 | ThoMe | lachs, fish ;) |
| 14:04.13 | *** part/#asterisk Stian- (n=stian@062016138186.customer.alfanett.no) |
| 14:04.16 | ThoMe | salmon |
| 14:04.17 | ThoMe | :) |
| 14:05.11 | graphx | ThoMe: I like salmon... good eat'ns when it's smoked with a side of cheese and wine... yummy!!! |
| 14:06.39 | tzafrir | stoffell, I added the patch to support them after someone posted a bug to Debian asking to apply that patch |
| 14:06.57 | graphx | Does anyone here use SPA942 Linksys SIP phones with Asterisk 1.4? |
| 14:07.22 | stoffell | tzafrir, oh, okay, so no code modifications anymore when using beronet cards with bristuff |
| 14:07.39 | tzafrir | It's still there as a separate patch |
| 14:07.59 | tzafrir | http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/patches/qozap/beronet.diff |
| 14:08.03 | tzafrir | http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/patches/cwain/beronet.diff |
| 14:08.06 | [TK]D-Fender | graphx: Plenty of people |
| 14:08.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.33 | stoffell | oh, okay, thanks tzafrir |
| 14:08.34 | *** join/#asterisk Bock (n=bock@dslb-084-057-034-118.pools.arcor-ip.net) |
| 14:09.42 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:09.43 | *** join/#asterisk Pondiboy (n=Pondiboy@122.164.37.14) |
| 14:09.48 | Bock | Hi, I made a "catchall" rule which uses my external sip provider, now I'm having the problem, that if a previous rule matches, asterisks also dials the second rule... is there a way to tell asterisk to stop parsing the extensions.conf.. and if yes, how? |
| 14:10.18 | tzafrir | stoffell, note that I have not tested that patch myself, as I lack that hardware |
| 14:10.36 | tzafrir | I got some sporadic reports that it seems to work, and no bad reports yet |
| 14:10.37 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 14:11.05 | yang | I am looking at the Busy() command, what I would like is that if the extension is Busy it transfers the call to another extensions - where is that explained? |
| 14:11.12 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:11.16 | Katty | good morning!! |
| 14:11.18 | graphx | [TK] D-Fender: Well I am stuck. I have been trying to research for the last 3 days on how to get the transfer to work on the phones. Then manual says that the phones themselves take care of the transfer, I have the option showing up on a button, but it will not allow the transfer, and I am not sure why. |
| 14:11.45 | stoffell | tzafrir, yeah, it's strange because in bristuff's changes there is a note about "patch qozap to support bero.net cards" but it doesn't look like that when i look at qozap.c |
| 14:11.58 | Katty | [TK]D-Fender: you'll never believe what i dreamt last night! zombies! in my HOUSE! that looked just like people i knew. (coworkers) and i had to resort to impaling them on broom handles to kill them. it was wild! |
| 14:12.04 | [TK]D-Fender | yang: "core show application goto" |
| 14:12.17 | *** join/#asterisk ddunavant (n=David@pool-71-178-109-242.washdc.east.verizon.net) |
| 14:12.42 | kamisalami | hi, i neeed to compile zaptel , so i tried 'm-a build zaptel' , but that does not give me the vzaphfc module |
| 14:12.49 | [TK]D-Fender | Katty: Only decapitation will do! You should know that! |
| 14:12.56 | kamisalami | can't find any clue about it in the Makefile either |
| 14:12.59 | Katty | [TK]D-Fender: the broom handles were much more fun tho. )= |
| 14:13.01 | kamisalami | it is in the source though |
| 14:13.12 | tzafrir | stoffell, it's part of the changelog item "integrated patches, docs, build scripts from bristuff-0.4.0-test4-xr5" |
| 14:13.36 | tzafrir | kamisalami, what kernel version? |
| 14:13.41 | tzafrir | What version of Zaptel? |
| 14:13.51 | kamisalami | 2.6.22-14-server |
| 14:14.05 | tzafrir | dpkg -l zaptel-source |
| 14:14.42 | kamisalami | ii zaptel-source 1:1.4.3~dfsg-2ubuntu1 Zapata telephony interface (source code for kernel driver) |
| 14:14.47 | Katty | [TK]D-Fender: is there a variable that will show what a SIP extension is connected to? |
| 14:15.34 | yang | [TK]D-Fender: like this if 5863165 becomes busy it will transfer it to SIP/66 ? http://openpaste.org/en/5433/ |
| 14:15.36 | *** part/#asterisk Pondiboy (n=Pondiboy@122.164.37.14) |
| 14:16.03 | kamisalami | zaptel version: Version: 1:1.4.3~dfsg-2ubuntu1 |
| 14:16.13 | stoffell | tzafrir, uh, yeah, but i can't find the patch in bristuff so it's not correctly imported then :( |
| 14:16.40 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 14:16.42 | [TK]D-Fender | yang: No, like this : http://pastebin.com/m7536bf2e |
| 14:17.03 | yang | thanks |
| 14:17.04 | [TK]D-Fender | yang: and you aren't "transferring" anything at all. If it doesn't answerit jsut dials something ELSE. |
| 14:17.16 | yang | yes |
| 14:17.28 | [TK]D-Fender | yang: You seriously need to sit down and focus on your understanding of the dialplan. |
| 14:17.57 | *** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096724607.dsl.bell.ca) |
| 14:18.21 | yang | [TK]D-Fender: I am reading the book, but it has lots of pages :) |
| 14:20.25 | graphx | yang: Another alternative is to use a GUI based program that will create the dailplan for you... check out the following - http://www.apstel.com/products/vdp/download.php |
| 14:20.52 | mosty | yang, then stop wasting time here and get through as many pages as you can |
| 14:21.28 | znoG | no-one happens to use res_config_ldap here? |
| 14:21.51 | graphx | yang: it is still important to learn the original manual though - i all seriousness... I am in the same boat... learning as I go... |
| 14:22.03 | tzafrir | kamisalami, hmmm.... that version has a known problem. You basically need to unpack /usr/src/zaptel.tar.bz2, fix the makefile /usr/src/modules/zaptel/debian/rules (add a '|| true' on that failing make clean) and repack |
| 14:22.17 | tzafrir | sorry, not in the mood now to write proper instructions |
| 14:22.21 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:22.22 | lmadsen | yet another reason to compile from source and not use a pkg :) |
| 14:22.44 | tzafrir | lmadsen, yet another reason to use packages that were actually *tested* |
| 14:22.51 | lmadsen | or that :) |
| 14:22.56 | x86 | lmadsen: there was an actual reason to use a pkg at some point? :) |
| 14:22.59 | tzafrir | The Ubuntu Universe packages often aren't |
| 14:23.05 | zeeesh | configuring realtime asterisk ..... can i put sip.conf's general settings into the database? |
| 14:23.07 | lmadsen | x86: :D |
| 14:23.22 | lmadsen | zeeesh: yes, but that would be done with static realtime |
| 14:23.30 | yang | [TK]D-Fender: http://pastebin.com/m7536bf2e this configuration probably calls extension 66 after 30 seconds, it doesnt transfer to 66 if 65 is being in a call |
| 14:23.32 | lmadsen | extconfig.conf: sip.conf => ... |
| 14:23.46 | x86 | lmadsen: you forgot to ping me back the other day, by the way ;) |
| 14:23.56 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 14:24.04 | lmadsen | x86: oh ya, that'll certainly happen :) oh ya! My internet went offline for a couple of hours |
| 14:24.09 | zeeesh | <lmadsen> means not supporting the dynamic realtime ? |
| 14:24.19 | lmadsen | seriously... dropped like 2 mins after I said I'd ping you back :) |
| 14:24.27 | lmadsen | zeeesh: means you have to use both |
| 14:24.34 | lmadsen | the general section wouldn't be in the dynamic realtime |
| 14:24.40 | zeeesh | ok |
| 14:24.45 | kamisalami | tzafrir: thanx! |
| 14:26.15 | yang | [TK]D-Fender: I tested it and it didn't work |
| 14:27.14 | yang | [TK]D-Fender: you pointed me to Goto() previously |
| 14:27.15 | kamisalami | tzafrir: like this.... ? -$(MAKE) clean || true |
| 14:27.37 | tzafrir | kamisalami, the one in the kdist target |
| 14:27.47 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 14:27.55 | kamisalami | tzafrir: ok, where did you get this info from? |
| 14:28.07 | Katty | [TK]D-Fender: thoughts on my insanity? http://www.pastebin.ca/930319 |
| 14:28.15 | tzafrir | sorry: in the binary-modules: target |
| 14:29.33 | yang | graphx: thanks for the link, i am also about to test the asterisk GUI package |
| 14:30.59 | kamisalami | tzafrir: ok, i'll try... not sure if it will help though, as it did build, just does not build that one module |
| 14:31.52 | tzafrir | hmmm... I figure you should try rebuilding Zaptel from Hardy |
| 14:32.51 | graphx | yang: no problem man... It's nice to have already been down a road, but when you have to learn from scratch, it's nice to have people back you up so that you can help someone later when they have a question... It's interesting that sometime people mistake asking questions with leeching.. In anycase hope it helps you.. |
| 14:33.08 | kamisalami | tzafir, i'll try that then |
| 14:33.32 | yang | graphx: its nice to learn the extensions, but i have no prevous programming skills, so that makes it a bit difficult for me to understand it |
| 14:33.34 | Katty | maybe... ${BRidgepeer} |
| 14:33.47 | Lsodi | can anyone recommend a book about connecting asterisk and cisco (routers)? |
| 14:34.12 | kamisalami | tzafrir: should i only use zaptel-source from hardy, or also zaptel packages |
| 14:34.15 | graphx | yang: Working on the same side of the road with you... learning it one variable at a time... I understand.. |
| 14:34.16 | kamisalami | -s |
| 14:35.37 | tzafrir | kamisalami, they have there 1.4.8, which is an API change. So you need zaptel as well |
| 14:35.47 | tzafrir | Or grab the source package and rebuuild it |
| 14:36.00 | tzafrir | (debuild -uc -us) |
| 14:36.12 | kamisalami | ok |
| 14:36.30 | x86 | ok, got a very weird issue... I've got a macro for inbound calls that dumps callers into an IVR, then they are connected to an operator by pressing 1... When they press 1, it rings two phones at the same time (one zap, and one sip)... if there is no answer, it dumps the call to a voicemail box... simple enough right? |
| 14:36.50 | x86 | when the caller starts to leave a voicemail, it cuts them off about 4 seconds into the voicemail message |
| 14:37.03 | x86 | and notifies the SIP phone with MWI that there is a new voicemail |
| 14:37.27 | x86 | when the operator checks the voicemail, they get the 4-5 seconds of voicemail left by the caller just fine |
| 14:37.37 | x86 | so it's not silence detection cutting it off, I dont think |
| 14:37.47 | x86 | any ideas what this might be? |
| 14:38.00 | x86 | I've never seen voicemail act like that before... |
| 14:39.35 | [TK]D-Fender | yang : No, it does not check if the first one is busy, it jsut dials the first, and if they don't answer, dial the 2nd phone. |
| 14:39.54 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 14:40.11 | yang | [TK]D-Fender: ah no, i want it like if the first (65) is busy talking then it should dial the second extension (66) |
| 14:40.32 | [TK]D-Fender | yang: "core show application chanisavail" <- Get reading. |
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| 14:43.27 | [TK]D-Fender | Katty: You'll need an AGI to parse out the active channels for those 2 redirect calls. |
| 14:43.50 | x86 | lmadsen: got a sec to assist with a voicemail issue? :) |
| 14:44.04 | [TK]D-Fender | Katty: Hoever I am not sure the entire idea works. If 2 channels are bridged and you hijack 1 leg, I have this feeling that the other is instantly cut off..... |
| 14:44.08 | lmadsen | you can ask, but my morning is starting to get busy, and I haven't even eaten breakfast yet :) |
| 14:44.21 | Katty | [TK]D-Fender: hrmm. you might be right. |
| 14:44.40 | Katty | surely there must be /something/ to do this. |
| 14:44.45 | Katty | i'll keep digging |
| 14:47.43 | znoG | nice! res_config_ldap is half working now |
| 14:48.25 | yang | [TK]D-Fender: http://openpaste.org/en/5435/ probably also wrong |
| 14:48.52 | JunK-Y | dont be jealous lmadsen ..... |
| 14:49.33 | [TK]D-Fender | Yang, you are clearly not even readying the instructions at all and just slapping application and parameters together nearly randomly. go READ. |
| 14:49.57 | errr | is it possible to make it so when someone gets your voicemail they can dial some other extension instead of having to press 0 to get back to the operator and then having the operator pass them along? |
| 14:51.07 | Katty | yang: s'ok. i'm fumbling around too this morning. |
| 14:51.07 | kamisalami | tzafrir: downloaded http://archive.ubuntu.com/ubuntu/pool/universe/z/zaptel/zaptel_1.4.8~dfsg.orig.tar.gz , but it does not include a debian dir? |
| 14:51.07 | [TK]D-Fender | errr: "operater" is whatever you want it to be. |
| 14:51.07 | VJFROMGT | hi, can someone tell me what file to put rtptimeout=60 |
| 14:51.07 | yang | Katty: luckily i got 10 min. to go off work, i will bugger the next morning |
| 14:51.07 | [TK]D-Fender | VJFROMGT: Quick guess : rtp.conf |
| 14:51.12 | Katty | yang: be sure to have a drink (= |
| 14:51.28 | tzafrir | kamisalami, a source package is the .orig.tar.gz, .diff.gz and .dsc |
| 14:51.46 | errr | [TK]D-Fender: yeah but I dont want them in an ivr, I want when you dial me, get my vm you can press 7002 and get the other guy in IT instead of having to hang up and dial him or dial the operator |
| 14:51.52 | tzafrir | then run: dpkg-source -x package_whatever.dsc |
| 14:51.56 | tzafrir | to extract |
| 14:52.01 | tzafrir | cd package-version |
| 14:52.06 | tzafrir | debuild -uc -us |
| 14:52.13 | Katty | [TK]D-Fender: i know how to do it! tell anthm to go rewrite chanspy! ;P |
| 14:52.16 | VJFROMGT | tk, if i put in an individual trunk, would it work? |
| 14:52.17 | kamisalami | ok, tnx, learning :) |
| 14:52.17 | [TK]D-Fender | errr: "0" + ext to call. Whats so hard about that? |
| 14:52.22 | tzafrir | (or: dpkg-buildpackage -rfakeroot -uc -us) |
| 14:53.01 | errr | [TK]D-Fender: I dont want a dirty hack, I want to know if what I want to do is possible or not |
| 14:53.48 | [TK]D-Fender | errr: Sure its possible... you've got the source... |
| 14:54.01 | x86 | lmadsen: When a call comes in and goes to voicemail, it only lets the caller record about 4-5 seconds worth of a message, then says thank you and hangs up on them... it still delivers the message to the mailbox though |
| 14:54.08 | x86 | lmadsen: every seen something like that? |
| 14:54.49 | errr | [TK]D-Fender: there has to be some other way to do that with out hacking the source.. the freepbx guys have added this thing where you can press 0, 1, or 2 from vm and have it dial people based on that.. |
| 14:55.30 | [TK]D-Fender | errr: yes, it dials PEOPLE. but you wnt to punch an exten DIRECT immediately. I'm saying you'll have to hit "0" and then decide what you want to do yourself. |
| 14:55.48 | errr | hm |
| 14:55.54 | [TK]D-Fender | errr: And no FreePBX does not let you immediately dial an ext. |
| 14:56.27 | [TK]D-Fender | errr: Hit ), dump them in an IVR where they can dial the ext#. End of story. |
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| 14:59.26 | [TK]D-Fender | Katty: Yes, the book makes a reference to "ExtraChannel", but no details.... ther might be a way there... |
| 15:00.11 | *** join/#asterisk mcargile (n=mcargile@mail.in-florida.net) |
| 15:00.58 | mcargile | The announcment for 1.6-beta5 is on the asterisk.org page, but non of the links have been updated to point towards it. |
| 15:01.09 | Katty | [TK]D-Fender: i've been search for it... |
| 15:01.11 | Katty | [TK]D-Fender: with no luck :/ |
| 15:01.53 | mcargile | http://downloads.digium.com/pub/asterisk/ |
| 15:01.58 | mcargile | it's in there |
| 15:02.41 | [TK]D-Fender | Katty: I see a sample that shows throwing both to the SAME plac, but not separate places. |
| 15:03.01 | *** part/#asterisk mcargile (n=mcargile@mail.in-florida.net) |
| 15:03.16 | [TK]D-Fender | Katty: http://pastebin.com/m4c6d6fb5 |
| 15:03.19 | *** join/#asterisk airjump (n=zielonka@62.159.95.82) |
| 15:03.27 | Dovid | anyone know where i can get test-apps-asterisk-1.2' from ? |
| 15:03.51 | Dovid | rather: app_rxfax and app_txfax. i am on the wiki but cant find a download for it |
| 15:04.07 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 15:04.11 | Katty | [TK]D-Fender: there's a part about that which i don't understand. |
| 15:04.18 | grandpapadot | Are parked calls context-sensitive or system-wide in 1.2? |
| 15:04.25 | Katty | [TK]D-Fender: i think it's the whole lack of exten => do something, which i don't get. |
| 15:04.30 | jeanmi_i_ | Does asterisk 1.6 support SRTP ? |
| 15:04.40 | *** join/#asterisk jayer (i=jaray@41-127-188-206.rev.knet.ca) |
| 15:04.47 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:04.54 | [TK]D-Fender | Katty: You specify the context,exten,prio in there pretty clearly. |
| 15:05.16 | [TK]D-Fender | grandpapadot: Single context for all of *. |
| 15:05.27 | [TK]D-Fender | grandpapadot: Thats why so many other parking apps were developed. |
| 15:05.27 | grandpapadot | [TK]D-Fender: Thanks. |
| 15:05.33 | Katty | [TK]D-Fender: yeah... this format doesn't look familiar at all. i odn't know what i'm looking at. |
| 15:05.41 | Katty | [TK]D-Fender: or where. |
| 15:06.05 | *** join/#asterisk wmaulik (n=wmaulik@158.59.195.51) |
| 15:06.11 | jayer | we bought an asterisk appliance AA50 and I'm wondering if I am able to auto provision any phones other than Polycom? |
| 15:06.17 | Katty | [TK]D-Fender: does this have something to do wtih API? |
| 15:08.07 | jayer | perhaps tftp? |
| 15:09.35 | kamisalami | tzafrir: got 4 lines of build-errors, should i privmsg them to you? |
| 15:09.37 | awk | lol, anyone seen 175 of the 255 channels used before? we just hit 175 active channels :) |
| 15:09.47 | tzafrir | ~pb |
| 15:09.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:10.52 | tzafrir | awk, 175 running voip channels? that only takes two PCs and an ethernet cable |
| 15:11.08 | kamisalami | pk |
| 15:11.11 | kamisalami | ok |
| 15:12.22 | tzafrir | kamisalami, what's this 255 limitation? |
| 15:12.51 | kamisalami | http://pastebin.com/m71ec6b32 |
| 15:12.52 | airjump | hello a good paste service with many lines http://ubuntuusers.de/paste Example http://ubuntuusers.de/paste/67630/ |
| 15:12.53 | ThoMe | is it correct?: exten => s,n,Set(CDR(userfield)=${CDR(lastapp)}) |
| 15:12.56 | Dovid | anyone know where I can get app_rxfax and app_txfax for 1.2.X ? |
| 15:13.21 | [TK]D-Fender | Katty: Yes, this is an AMI call. |
| 15:13.42 | Poincare | Set(CHANNEL(language)=fr) , Playback(digits/hundred) and I still get the english version.... Any clues why? |
| 15:14.15 | tzafrir | jbot, tell airjump about pb |
| 15:15.33 | [TK]D-Fender | Poincare: Got a file called "digits/fr/hundred"? |
| 15:15.49 | kamisalami | tzafrir: i'll rebuild it in text mode and get you the whole log |
| 15:16.03 | tzafrir | kamisalami, you don't need to |
| 15:16.06 | kamisalami | ok |
| 15:16.15 | tzafrir | check /var/cache/modass/ |
| 15:16.18 | Poincare | [TK]D-Fender: yes |
| 15:16.37 | tzafrir | modass keeps full build logs there |
| 15:16.38 | Poincare | strange thing is that it will play files under the root fr directory |
| 15:16.55 | tzafrir | one nice thing about it. It keeps trace of everything |
| 15:17.01 | kamisalami | my rebuild was already running |
| 15:17.20 | JunK-Y | Poincare: do a sym link in ur digits/ |
| 15:17.24 | [TK]D-Fender | Poincare: Pastebin your "ls" dump of your sounds folder, and CLI output of the failed attempt at verbose 10 |
| 15:17.35 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 15:17.47 | JunK-Y | or simply read: http://forums.amug.ca/viewtopic.php?t=3782 |
| 15:17.52 | [TK]D-Fender | Poincare: make sure we can see the "ls" call from CLI as well |
| 15:18.22 | Poincare | ok... i'll try to track that |
| 15:19.08 | kamisalami | tzafrir: http://pastebin.com/d1a56989f |
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| 15:22.48 | *** part/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 15:23.54 | kamisalami | tzafrir, as log in /var/cache might be better then my redirect console to file, i have uploaded this one as well: http://pastebin.com/d27b38ecb |
| 15:24.09 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 15:24.17 | tzafrir | I'm trying to remember what's that "generated" directory is |
| 15:25.27 | ThoMe | what is wrong? : |
| 15:25.27 | ThoMe | [intern] |
| 15:25.27 | ThoMe | exten => s,1,Set(CDR(userfield)=${CDR(dstchannel)}) |
| 15:25.27 | ThoMe | exten => 7,1,Dial(SIP/7|99|tw) |
| 15:25.30 | ThoMe | hm :-( |
| 15:25.44 | ThoMe | i would like before dial SIP/7 then set CDR... |
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| 15:27.50 | Docfxit | I can't seem to get anything to log. In Logger.conf I have under logfiles console=notice,warning,error and on a second line messages=notice,warning,error I set verbose 999 nothing is going into the log. |
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| 15:29.33 | JunK-Y | type logger reload |
| 15:29.53 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
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| 15:30.44 | ThoMe | is it correct? JunK-Y ? |
| 15:30.45 | ThoMe | exten => 99,n,Set(CDR(userfield)=${CDR(start)}) |
| 15:31.41 | JunK-Y | replace n by 1 if its ur 1st priorityQUELCH Request Message |
| 15:31.42 | VJFROMGT | i have a trunk which has been hanging for 2 hours, ive tried soft hangup, it is still up |
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| 15:31.48 | VJFROMGT | how can i kill the channel? |
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| 15:32.00 | kamisalami | tzafrir: here is what i did to try and build it... http://pastebin.ca/930378 |
| 15:32.05 | ThoMe | JunK-Y: you mean me? |
| 15:32.46 | JunK-Y | ThoMe: yes |
| 15:33.05 | x86 | anyone ever seen voicemail cut someone off in the middle of leaving a message? |
| 15:33.31 | tzafrir | kamisalami, so that worked? |
| 15:33.58 | [TK]D-Fender | x86: Silence detection / max message length. |
| 15:35.23 | x86 | [TK]D-Fender: it's not silence detection, because it delivers the message |
| 15:35.38 | x86 | [TK]D-Fender: max message length is set to 180, which I interpreted to mean 3 minutes |
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| 15:37.23 | x86 | http://pastebin.ca/930383 |
| 15:37.26 | x86 | that's my voicemail.conf |
| 15:37.27 | kamisalami | tzafrir: no |
| 15:37.42 | tzafrir | kamisalami, same problem? |
| 15:38.17 | kamisalami | yeah |
| 15:38.40 | kamisalami | maybe you can figure out what that generated dir is from the rule file: http://pastebin.ca/930384 |
| 15:39.07 | kamisalami | ? |
| 15:41.12 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 15:41.45 | tzafrir | cp -a $(GENERATED_SOURCES) $(TARDIR)/debian/generated/ |
| 15:42.01 | tzafrir | version.h and the bristuff modules sources |
| 15:42.31 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 15:42.34 | keith4 | is telephonydepot still the best place to buy sip phones? |
| 15:42.44 | kamisalami | not sure what to do with that? |
| 15:42.57 | tzafrir | not sure what is wrong either |
| 15:43.13 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:43.29 | flujan | hi guys, I am trying to use realtime asterisk with version 1.4.18. |
| 15:43.55 | flujan | I can place calls to my E1 lines, but I cannot receive or call another sippeer. |
| 15:45.16 | [TK]D-Fender | keith4: Still good service and some of the best pricing. Shop around as always |
| 15:47.23 | flujan | here goes a pastie. |
| 15:47.24 | flujan | http://pastie.caboo.se/162335 |
| 15:47.50 | flujan | It keep saying that the peer is congest or unavailable... But this is the exact peer I am using to place a call |
| 15:47.52 | keith4 | mmm, pasties |
| 15:49.05 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:49.38 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:51.22 | flujan | also, asterisk is not updating the the database table with information about ipaddr, port and stuff... |
| 15:51.23 | flujan | http://pastie.caboo.se/162336 |
| 15:51.47 | kamisalami | tzafrir: wow, google is fast, found my own post on pastebin, done 17 mins ago :) |
| 15:52.50 | flujan | but asterisk is authenticating the peers... |
| 15:52.57 | flujan | very strange... I am really stucked here. |
| 15:52.58 | flujan | :( |
| 15:53.28 | *** join/#asterisk josephp (n=josephp@shell4.sea5.speakeasy.net) |
| 15:53.42 | [TK]D-Fender | flujan: Yup... thats why the cahnnel is listed as an IP only... |
| 15:54.39 | tzafrir | kamisalami, good thing it is not permanent |
| 15:54.55 | *** join/#asterisk JosephP (n=none@shell4.sea5.speakeasy.net) |
| 15:54.58 | JosephP | oops |
| 15:55.23 | flujan | [TK]D-Fender: any ideas? A long time ago I put it working with 1.2.. |
| 15:55.38 | flujan | [TK]D-Fender: jit, how are you doing? |
| 15:55.40 | JosephP | okay so I'm wondering if there is an easy way to put a call on hold and skip the MoH, the reason for this is when I'm participating in a conference bridge I don't want to flood the other participants with my music. |
| 15:56.51 | JenniferAkemi | does anyone here use dialogic cards? |
| 15:56.55 | JenniferAkemi | (with asterisk i mean) |
| 15:57.46 | x86 | [TK]D-Fender: so no ideas on the voicemail issue? |
| 15:58.49 | BobLutz | JenniferAkemi: Dialogic cards are a hack with Asterisk |
| 15:59.06 | Katty | [TK]D-Fender: that would explain why none of this makes sense. i've never done an AMI call (= |
| 15:59.25 | Katty | also, has anyone setup hud server before? |
| 15:59.25 | Katty | i seem to be stuck |
| 16:02.14 | errr | If I have exten => 7001,2,goto(ivr-1,s,1) once I get in that IVR is there anyway for me to know that 7001 is what exten sent the caller to the ivr? |
| 16:02.46 | Katty | you could change the callerid maybe |
| 16:03.39 | BobLutz | errr: you could do 7001,1,Set(RETURN=${CONTEXT}) and access ${RETURN} where needed |
| 16:03.41 | errr | Katty: you should look at iSymphony |
| 16:03.47 | Katty | errr: oh? |
| 16:03.50 | Katty | errr: i'll go have a look. |
| 16:04.10 | errr | BobLutz: ah good idea, thanks |
| 16:04.11 | BobLutz | s/${EXTEN}/${CONTEXT} |
| 16:05.30 | JosephP | sooo does anybody have any thoughts on disabling music on hold on a per-call basis? like another softbutton that would put someone on a silent hold so I can answer another call? |
| 16:05.37 | Katty | errr: does this user a manger.conf interface? |
| 16:06.27 | errr | Katty: I just saw a short demo on it, but they give a free trial license, I havent had a chance to install it to check it all out yet |
| 16:07.09 | errr | if it works like the demo we are buying 5 licenses |
| 16:07.10 | Katty | meh |
| 16:07.23 | errr | this stuff was 100 times better than hud |
| 16:07.32 | errr | plus no irc needed |
| 16:07.45 | Katty | i don't have a problem with IRC. |
| 16:07.46 | errr | and it intergrates with gabber |
| 16:07.58 | Katty | this looks... too complicated for our sales staff |
| 16:08.22 | errr | and hud is not? |
| 16:08.34 | Katty | not particularly, no |
| 16:08.43 | errr | *shurg* |
| 16:09.29 | Katty | i guess there's no reason we can't have both |
| 16:09.34 | Katty | and they can pick the one they like better tho |
| 16:09.46 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 16:10.54 | flujan | GOT IT. |
| 16:11.10 | flujan | I forgot the username column on the users table. :) |
| 16:12.03 | Dovid | anyone here work with rxfax and txfax ? |
| 16:12.29 | dandre | How should I provide some hierarchical menu with return to the caller menu? |
| 16:12.30 | _ShrikE | Dovid: I do, but with callweaver |
| 16:12.54 | *** join/#asterisk badcfe_ (i=christia@nothing.beats.deathporn.com) |
| 16:12.57 | Dovid | _Shrike: I need to stick with Asterisk (the big boss ;) ) |
| 16:13.35 | Dovid | _ShrikE: I installed SpanDSP and now I need to apply the patches from what I understand. there isnt any really good docs out there |
| 16:13.45 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:13.46 | badcfe_ | how does * decide packetization time to use for an rtp codec channel |
| 16:13.59 | badcfe_ | ? |
| 16:14.01 | _ShrikE | Dovid: Yes, just apply the patches and recompile asterisk. |
| 16:14.52 | Dovid | ShrikE: How do I apply them ? I am using http://www.voip-info.org/wiki/view/AGX+Extra+Addons+for+Asterisk i did hte build. there are other files there. not sure which ones to patch |
| 16:14.58 | Katty | errr: the installation directions look much more straight forward than HUD |
| 16:15.59 | Katty | errr: i'll give it a go (= |
| 16:16.56 | Dovid | _shrikE: seems that when I use exten => 123,X, rxfax it breaks there |
| 16:17.22 | _ShrikE | Dovid: is rxfax listed when you do a core show applications? |
| 16:18.33 | Dovid | hehe. nm. i looked at the debug |
| 16:18.34 | Dovid | v |
| 16:18.34 | Dovid | Mar 6 10:56:28] WARNING[7978] channel.c: Unable to find a codec translation path from g729 to slin |
| 16:18.34 | Dovid | [Mar 6 10:56:28] WARNING[7978] /usr/src/agx-ast-addons/app_rxfax.c: Unable to set to linear read mode, giving up |
| 16:18.35 | Dovid | [ |
| 16:21.26 | _ShrikE | Dovid: that error means you dont have g729 licensed |
| 16:22.02 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:22.03 | Dovid | _Shrike: Exactly. Just rebuilt. asterisk. but i should not need it if I just want to use uLaw (which I set in sip.conf and did a reload) |
| 16:22.34 | *** join/#asterisk svenna (n=svenna@p548D082B.dip0.t-ipconnect.de) |
| 16:22.47 | _ShrikE | Dovid: Right. G.729 will not work with faxes |
| 16:23.09 | Dovid | :) |
| 16:23.11 | Dovid | working now |
| 16:24.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:24.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:24.42 | jameswf-home | thia event invite crap is getting a bit old |
| 16:24.48 | jameswf-home | *this |
| 16:24.54 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/0?3/05), Asterisk 1.4.19-rc1 (2008/03/05), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: sharks sighted today |
| 16:24.57 | russellb | errrr |
| 16:24.59 | russellb | ? |
| 16:25.21 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta5 (2008/03/05), Asterisk 1.4.19-rc1 (2008/03/05), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: sharks sighted today |
| 16:25.36 | Dovid | _ShrikE: May I PM ? |
| 16:26.12 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 16:27.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:27.49 | Katty | what's the remote desktop server side thing called? |
| 16:27.53 | Katty | the one that rdesktop connects to |
| 16:27.58 | Katty | my mind is failling me |
| 16:27.59 | _ShrikE | Dovid: sure |
| 16:30.03 | Nugget | apple's screen sharing think in leopard is the best vnc client I've ever used. |
| 16:30.03 | twisted | sharks? |
| 16:30.08 | Nugget | er, thing. |
| 16:30.30 | Katty | vncserver! that was it |
| 16:30.50 | jameswf-home | crossloop is pretty good if your a windoze user its built around tight vnc |
| 16:31.12 | *** join/#asterisk monkeytype (n=Bill@216.207.245.1) |
| 16:31.13 | cmantito | sweet! using bristuff devstate I was able to make an LED that blinks when my calls are forwarded :) |
| 16:31.24 | Jason99 | in sip.conf does call-limit=0 mean unlimited? |
| 16:31.26 | znoG | russellb: i found a couple of things that could be improved in the asterisk.schema (res_config_ldap). Should I open up a bug for this? |
| 16:32.18 | Dovid | _Shrike: Did you recieve my PM ? |
| 16:33.37 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 16:34.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:34.34 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 16:36.10 | mmlj4 | sharks? |
| 16:49.16 | dandre | is there any way to have menu messages from voicemailmain application smoother (closer to a natural voice)? |
| 16:49.50 | ThoMe | twisted: hello? |
| 16:50.50 | *** join/#asterisk adjohn (n=adjohn@p5155-ipad74marunouchi.tokyo.ocn.ne.jp) |
| 16:50.54 | VJFROMGT | i have a call which has been open for over 4 hrs, soft hangup does nto kill it, ,how can i kill it |
| 16:50.57 | ThoMe | is it posible?: Set(CDR(userfield)=${CDR(userfield)+${CDR(dstchannel) ?? |
| 16:51.11 | ThoMe | a combination .. + + + ... ? |
| 16:52.16 | ThoMe | or + == & ? |
| 16:52.33 | BobLutz | ThoMe: I do stuff like Set(Count=${COUNT}+1) |
| 16:52.56 | BobLutz | The question is: Can you set CDR(userfield) ? |
| 16:53.01 | *** join/#asterisk tRSS (n=tRSS@adsl-68-20-214-239.dsl.chcgil.ameritech.net) |
| 16:53.11 | BobLutz | ThoMe: I do stuff like Set(COUNT=${COUNT}+1) |
| 16:53.11 | ThoMe | BobLutz: no. not to add, string together |
| 16:53.13 | BobLutz | is what i meant |
| 16:53.21 | ThoMe | string one + strong two + string 3 |
| 16:53.28 | BobLutz | yea |
| 16:53.40 | ThoMe | its wit h+ ? |
| 16:53.51 | ThoMe | not with &, BobLutz ? |
| 16:53.59 | tRSS | hey folks... is there any tool available for asterisk that would make the caller sound like a robot or add other sounds |
| 16:54.01 | BobLutz | Ive done it with + and it worked |
| 16:54.06 | *** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 16:54.13 | znoG | russellb: wondering if you can help me. I've switched 'md5secret' in res_ldap.conf to point at 'userPassword'. Now Asterisk segfaults when I try to authenticate. Any ideas why? |
| 16:54.14 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 16:54.18 | vitaminmoo | Hello |
| 16:55.01 | dofear | Hi, I have my asterisk setup with two different SIP provider, with the same config I get ringback tone from one provider, but do not get ringback on the other provider ... can anyone suggest where i should look? |
| 16:55.21 | BobLutz | ThoMe: you might have to do Set(CDR(userfield)="${CDR(userfield)}${CDR(dstchannel)}") |
| 16:56.28 | vitaminmoo | I'm I'm getting "Spawn extension (ringgroup-1, s, 2) exited non-zero on 'Zap/8-1'" where ringgroup-1,s,2 is a Dial cmd with a single extension, a timeout, and no options. This extension is definitely timing out, but it's causing the call to be hung up, instead of moving on to the next priority |
| 16:56.43 | vitaminmoo | Am I missing something |
| 16:57.09 | *** join/#asterisk blueandwhiteg3 (n=blueandw@c-24-20-182-238.hsd1.mn.comcast.net) |
| 16:57.16 | BobLutz | vitaminmoo: Do you have "autofallthrough" set in extensions.conf? |
| 16:57.23 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 16:57.29 | vitaminmoo | it does not appear so |
| 16:57.51 | vitaminmoo | the thing is, it works sometimes |
| 16:58.02 | ManxPower | Nothing like being woken up from a deep sleep and being told the company proxy server is down. Ah, the adrenaline rush! |
| 16:58.11 | blueandwhiteg3 | Hello, I am using asterisk for the first time and having a problem with authentication. [Mar 6 11:57:24] NOTICE[30346]: chan_sip.c:15069 handle_request_register: Registration from '<sip:user@domain.tld>' failed for 'myipaddress' - No matching peer found |
| 17:01.01 | dofear | i have tried with 'r' and no 'r' in extsnsions.conf but no luck |
| 17:01.07 | dofear | can anyone shed some light |
| 17:02.26 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 17:04.13 | BobLutz | dofear: What are you trying to accomplish? |
| 17:04.56 | dofear | Hi, I have my asterisk setup with two different SIP provider, with the same config I get ringback tone from one provider, but do not get ringback on the other provider ... can anyone suggest where i should look? |
| 17:05.16 | vitaminmoo | BobLutz: It doesn't look like changing autofallthrough will help me. One note is that the behavior is intermittent. Sometimes the Dial cmd works fine, other times it returns 0 |
| 17:05.29 | [TK]D-Fender | vitaminmoo: APSTEBIN is your friend.... |
| 17:05.39 | dofear | just need to get ringback tone from the second provider when i make an outbound call |
| 17:05.42 | [TK]D-Fender | PASTEBIN* |
| 17:05.42 | mmlj4 | what's this about sharks? |
| 17:05.58 | znoG | russellb: ? |
| 17:06.07 | vitaminmoo | Roger, I'll put the infos together. |
| 17:06.35 | [TK]D-Fender | mmlj4: Just a running joke. Forget it |
| 17:07.02 | mmlj4 | k |
| 17:08.13 | Katty | errr: http://angela.sleekgeek.org/2008/03/06/isymphony/ <- will resume after lunch |
| 17:08.22 | *** join/#asterisk oej (n=olle@p5485C521.dip.t-dialin.net) |
| 17:09.07 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:10.09 | vitaminmoo | [TK]D-Fender: http://pastebin.com/d322b5a40 |
| 17:11.19 | errr | Katty: cool. I will be using the rpm since we run all centos servers |
| 17:11.26 | [TK]D-Fender | vitaminmoo: "dialplan show ringgroup-1" |
| 17:12.41 | vitaminmoo | [TK]D-Fender: http://pastebin.com/m185e1994 |
| 17:13.24 | [TK]D-Fender | vitaminmoo: is taht you doing a test call in? |
| 17:13.25 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 17:13.29 | vitaminmoo | Yes |
| 17:13.47 | [TK]D-Fender | vitaminmoo: And * hangs up on you? |
| 17:14.06 | *** part/#asterisk FarrisG (n=jrush@pool-71-164-183-233.dllstx.fios.verizon.net) |
| 17:14.11 | vitaminmoo | It tries to, but actually ends up ringing more, I think due to a rhino driver problem |
| 17:14.20 | vitaminmoo | It hangs up and then starts ringing instantly |
| 17:14.25 | vitaminmoo | Sometimes working the second time |
| 17:14.30 | [TK]D-Fender | vitaminmoo: Yeah, Im wonding about this being a source channel problem.. |
| 17:14.39 | vitaminmoo | Hmm |
| 17:14.47 | vitaminmoo | I've got a digium card I can replace that with |
| 17:15.12 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:15.18 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:15.25 | Katty | errr: i'm sure the config files and such will be the same tho |
| 17:15.31 | Katty | errr: i intend to document in extreme detail. |
| 17:15.32 | vitaminmoo | It definitely seems to be malfunctioning, if that seems logical to you, I'll do that asap |
| 17:15.49 | vitaminmoo | I mean, Dial() isn't exactly complex |
| 17:16.03 | vitaminmoo | [TK]D-Fender: thanks for looking at it. |
| 17:16.16 | [TK]D-Fender | vitaminmoo: All hints show that it looks like the calling channel jsut dies |
| 17:16.38 | vitaminmoo | [TK]D-Fender: replace it is, thanks. |
| 17:17.39 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:17.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:17.44 | Katty | anthm: hai |
| 17:18.14 | Katty | anthm: i would offer you lunch! but it'd be cold by the time you got here. |
| 17:23.25 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 17:28.11 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 17:28.49 | Dabba | anyone had any joy with cisco 7921G & chan-sccp-b |
| 17:30.53 | *** join/#asterisk noneo (n=artur@82.152.82.149) |
| 17:31.01 | x86 | ok, so anyone have any ideas on why the asterisk voicemail application would cut people off after 4-5 seconds while someone is leaving a voicemail? |
| 17:31.26 | x86 | it still delivers the short voicemail, so I know it's not silence detection (otherwise it would think it was an empty voicemail and discard it) |
| 17:31.38 | x86 | and I know the max message length is 180 seconds, per voicemail.conf |
| 17:31.48 | x86 | so anyone have any other ideas why it might be acting this way? |
| 17:32.14 | x86 | seems sporadic too, sometimes it works fine |
| 17:32.28 | x86 | just the majority of times it cuts them off, says thank you, then hangs up on the caller |
| 17:33.53 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 17:36.47 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 17:37.58 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 17:38.16 | *** join/#asterisk vap0rtranz (n=jpittman@r75-110-17-157.gvllcmtc01.gnvlnc.ab.dh.suddenlink.net) |
| 17:38.44 | vap0rtranz | registrar - hear me roar! |
| 17:38.49 | ManxPower | x86: turn off relaxdtmf if you have it set in zap or sip confs. silence detection and min message length are NOT the same thing. try turning off silence detection |
| 17:38.53 | vap0rtranz | *sqeek* |
| 17:38.59 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:39.09 | ManxPower | and watch the console and see that it says when the problem happens if you can |
| 17:39.51 | nny_1 | does anyone know of a pre existing Alison recording that is used for a pre-cursor to voicemail, giving the user the option to press 1 for voicemail, 2 for cellphone (or some variant of that)? |
| 17:42.38 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 17:42.40 | ManxPower | nny_1: we just tell users to say in their outgoing greeting to press 0 to reach their cell phone, then set up exten => o to route to their cell |
| 17:43.02 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 17:43.22 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:43.52 | *** join/#asterisk actros1840 (n=A@chello087207097013.chello.pl) |
| 17:44.02 | nny_1 | ManxPower: makes sense |
| 17:44.32 | ManxPower | nny_1: you COULD do it the way you said, I just find my way to be easier to set up. |
| 17:45.13 | nny_1 | ManxPower: nah it sounds pretty reasonable. Gives the user the power to "enable" or disable the feature by mentioning it or not too |
| 17:46.07 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 17:46.09 | ManxPower | nny_1: We actually give the user an option of what "0" does, it can route to the operator or send the call to their cell phone. Many users have the feature enabled, but don't mention it in their outgoing greeting, then they can just verbally tell the people they what to know about the feature. |
| 17:46.40 | ManxPower | if a user does not have the feature enabled in our dialplan the default route for the call is the local receptionist |
| 17:48.11 | znoG | System() ? |
| 17:48.20 | ManxPower | actros1840: you do it in an AGI script however you would do it if you were writting a regular script. From the dialplan see "system" and "trysystem" applications |
| 17:48.25 | *** join/#asterisk SparFux (n=raoul@e182026072.adsl.alicedsl.de) |
| 17:49.18 | BobLutz | actros1840: The typical default location of AGI scripts is /var/lib/asterisk/agi-bin/ |
| 17:49.19 | actros1840 | ManxPower : if I run this way a media player will the caller hear the output or not ? |
| 17:49.41 | *** join/#asterisk ryanteem (n=ryanteem@216.24.168.234) |
| 17:49.53 | actros1840 | BobLutz : I have a working AGI script however I want to experiment with some media players |
| 17:49.56 | nny_1 | ManxPower: roger |
| 17:50.16 | BobLutz | actros1840: I dont believe it will pipe its sound through asterisk |
| 17:50.28 | BobLutz | actros1840: I think it will just cause the player to play sound on the server |
| 17:52.13 | actros1840 | BobLutz : I'm afraid so |
| 17:52.29 | BobLutz | actros1840: why not use Playback()? |
| 17:53.01 | actros1840 | BobLutz : because you can't control the gain |
| 17:53.08 | BobLutz | ahhh.... |
| 17:53.56 | ryanteem | Has anyone here setup multiple SIP trunks to the same provider... having a little trouble with username digests :( |
| 17:54.26 | cmantito | I had the same problem, so I gave up and setup one trunk with SIP and one with IAX ;) |
| 17:54.48 | ryanteem | My provider is requiring a trunk for each phone we have |
| 17:54.49 | ManxPower | ryanteem: replace the word "trunk" with peer. There is no such thing as a "SIP trunk" |
| 17:54.55 | ryanteem | ok |
| 17:55.01 | ryanteem | sorry, peer |
| 17:55.01 | ManxPower | ryanteem: then your provider is lying |
| 17:55.10 | ryanteem | No, it's because they aren't charging for minutes... |
| 17:55.17 | ryanteem | They are billing per phone, and thats how they want to track it |
| 17:55.17 | cmantito | trunk is a relative term depending on your use, if you take the technical out of it :P |
| 17:55.22 | ManxPower | ryanteem: They are still lying. |
| 17:55.32 | ryanteem | hmm.. |
| 17:55.49 | ManxPower | There is no way the provider can REALLY tell what phone is calling it, because the phone is NOT calling it, Asterisk is calling it and then patching the two channels togather |
| 17:55.50 | Katty | HAI |
| 17:56.00 | ryanteem | right |
| 17:56.13 | ryanteem | I understand, and they get that... |
| 17:56.15 | ManxPower | ryanteem: Is your Asterisk server behind NAT? |
| 17:56.26 | ryanteem | yes |
| 17:56.30 | ManxPower | paste one of your Dial lines (just one) |
| 17:56.40 | ManxPower | even better, paste the CLI outout when the dial happens |
| 17:56.47 | ryanteem | It dials out fine... |
| 17:56.48 | ManxPower | of just the dial line, of course |
| 17:56.52 | ryanteem | and it dials on each of the trunks fine as well |
| 17:57.01 | ManxPower | ryanteem: let go of my ears, I know what I'm doing. |
| 17:57.23 | ryanteem | exten => s,1,Dial(SIP/trunk_${CALLERID(num):6}/${ARG1}) |
| 17:57.32 | ryanteem | That's how it determines which sip peer to dial out on |
| 17:57.50 | ManxPower | ryanteem: That is clever, and should work |
| 17:57.55 | ryanteem | Yeah, and it does |
| 17:58.02 | ryanteem | But when a call comes in, it has to act like DID |
| 17:58.20 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 17:58.23 | ManxPower | ryanteem: Oh! so you have a problem with INCOMING calls, not OUTGOING calls |
| 17:58.24 | *** join/#asterisk SteveTotaro (n=root@96.234.251.116) |
| 17:58.26 | ryanteem | So each of the incoming peers are named trunk_extension |
| 17:58.31 | ryanteem | Yeah... the call comes in fine |
| 17:58.44 | ManxPower | OK, so what IS your actual problem? |
| 17:58.46 | ryanteem | but when I hang up from inside the office, the BYE gets a permission denied |
| 17:58.53 | ryanteem | and the call doesn't get terminated correctly |
| 17:58.58 | cmantito | actros1840: I have a bad habit of grabbing the ast sources, extracing them, putting them in the 'apps' directory, then ./configure && make, and you'll end up with the compiled files that you can copy to your ast's apps directory |
| 17:59.01 | BobLutz | actros1840: Type "make" |
| 17:59.06 | cmantito | but....there're reasons why that may not work for you ;) |
| 17:59.36 | BobLutz | actros1840: Type "make install" to install the module in /usr/lib/asterisk/modules/ |
| 17:59.48 | ManxPower | ryanteem: that is far, far beyond the kind of help I do here, that's not a config issue with Asterisk, it's something else |
| 17:59.51 | *** join/#asterisk ceegee (n=christia@p57BD591F.dip.t-dialin.net) |
| 17:59.54 | ceegee | hello |
| 18:01.39 | ryanteem | But Manx... I was hoping you'd know how to fix it :( |
| 18:02.26 | ManxPower | ryanteem: I have no idea. Sounds like a SIP protocol issue or a NAT oddity |
| 18:02.42 | ryanteem | Well, can I PM you one sip frame to look at? |
| 18:02.45 | ManxPower | ryanteem: what version of Asterisk are you using? |
| 18:02.47 | ryanteem | just one... |
| 18:02.49 | ryanteem | 1.4.17 |
| 18:03.03 | ryanteem | sorry |
| 18:03.04 | ryanteem | .18 |
| 18:03.06 | ryanteem | 1.4.18 |
| 18:04.06 | ryanteem | The call is matching the wrong sip peer when sending the "bye", it's only matching the last one defined in SIP.conf |
| 18:04.15 | ryanteem | so I'm wondering if a fromuser can fix it, but don't know how that's supposed to look |
| 18:04.20 | *** join/#asterisk supa_disko (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 18:04.34 | actros1840 | I want just to change two words in .c file and then complie it because mp3player runs mpg123 with some arguments that I don't like |
| 18:05.43 | tzafrir | actros1840, why would you need mpg123? |
| 18:06.18 | Katty | weird question in regards to manager.conf |
| 18:06.33 | Katty | i have a software program asking me if the type if 'tenant' 'location' or 'system' |
| 18:06.33 | ceegee | which active isdn card would you guys recommend for use with an actual asterisk? we have actually three telephone lines, but could rise |
| 18:06.45 | Katty | what do those options mean? |
| 18:06.58 | *** join/#asterisk DiDjCodt (n=djc@yop.chewa.net) |
| 18:07.18 | DiDjCodt | hello |
| 18:07.29 | actros1840 | tzafrir : http://forums.digium.com/viewtopic.php?t=20963 |
| 18:07.49 | DiDjCodt | i have some problems with sip accounts when doing an outgoing call from asterisk |
| 18:08.12 | tzafrir | ceegee, why exactly are you after an active card? |
| 18:08.36 | DiDjCodt | asterisk seems to return from answer immediatly and not when the other side is answering |
| 18:08.40 | ceegee | tzafrir: my boss would like to have one |
| 18:09.05 | tzafrir | Why is your boss after one. Apart from the coolness factor? |
| 18:09.13 | DiDjCodt | and another problem is DTMF not going back to asterisk too |
| 18:09.23 | tzafrir | Why do you specifically need an *active* one? |
| 18:09.36 | DiDjCodt | i already tried SIPDtmfMode() with info, rfc.., inband |
| 18:10.09 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:10.34 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
| 18:10.46 | DonAlex | evening all.. |
| 18:10.55 | DiDjCodt | is answer the good function to detect other side answer ? |
| 18:10.59 | ceegee | tzafrir: no other reason, could be passive too in my opinion |
| 18:11.18 | ManxPower | ryanteem: only if you send large amounts of cash. reading SIP debug takes me forever as I have to read the protocol RFCS, etc. |
| 18:11.28 | ManxPower | this is not an easy task for me, who is NOT an expert in the SIP protocol. |
| 18:11.34 | DonAlex | Can someone point me to a resource or give me a quite understanding of how to make an fall back dial plan that uses PSTN channels if the IP link goes down? |
| 18:12.41 | ManxPower | ~book |
| 18:12.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:12.44 | ManxPower | there you go, DonA |
| 18:12.52 | *** join/#asterisk xacatecas (n=jkroon@wbs-41-208-201-177.wbs.co.za) |
| 18:13.09 | DonAlex | manx: ahhh rigthty cheers |
| 18:13.15 | xacatecas | hi guys, google isn't giving me much, how can i ask chan_zaptel to not wait for a dialtone before dialing? |
| 18:13.59 | ManxPower | xacatecas: it's chan_zap and it does NOT wait for dialtone before dialing EXCEPT on the X100P cards |
| 18:14.06 | ManxPower | and nobody in their right mind would use one of those cards. |
| 18:14.39 | xacatecas | lol, ok, well, i get that it sounds like i'm talking to a floating-Z when I call out on those lines. |
| 18:16.51 | DonAlex | Ahh yes.. bit like the Hitchhikers guide :) nice.. Thanks very much Manx |
| 18:16.58 | DonAlex | Later all.. :) |
| 18:19.22 | Kobaz | anyone have any idea what a pulse density violation on a T1 is |
| 18:20.55 | znoG | Kobaz: hey :) I'm the guy you were helping with res_config_ldap |
| 18:21.42 | Kobaz | znoG: heh, i don't know anything about ldap, so that's probably not me |
| 18:21.54 | znoG | Kobaz: oh ok .. his name was Kobaz too I think :) |
| 18:22.22 | tzafrir | ManxPower, chan_zap waits for dialtone on X100P? |
| 18:23.43 | tzafrir | what component identifies the dialtone there? |
| 18:24.40 | Kobaz | i dont think that's possible |
| 18:24.41 | Kobaz | heh |
| 18:24.51 | Kobaz | i've had kobaz registered on this network for like 5 years |
| 18:25.04 | xacatecas | *sigh* ... the gargly noises is because of asterisk it seems. |
| 18:25.23 | xacatecas | as soon as i unplug the line it stops with line dead. |
| 18:26.35 | znoG | Kobaz: i meant on the mailing list |
| 18:27.39 | *** join/#asterisk budol (i=bezpu@202.124.138.70) |
| 18:28.40 | Kobaz | oh |
| 18:28.42 | Kobaz | heh |
| 18:28.47 | Kobaz | those punks, stealing my name |
| 18:29.01 | Kobaz | anyway |
| 18:29.04 | ManxPower | tzafrir: It may only require volatage on the line. |
| 18:29.12 | ManxPower | But if you have voltage you usually have dialtone. |
| 18:29.12 | budol | hi |
| 18:29.13 | Kobaz | pulse density violation? |
| 18:29.33 | ManxPower | tzafrir: NOT supporting that on any other card is one of the stupidest decision I think Digium has ever made. |
| 18:29.42 | budol | anyone know how to configure a quintum into asterisk box |
| 18:30.00 | tzafrir | ManxPower, driver has supported that for quite some time |
| 18:30.12 | budol | quintum tenor AX |
| 18:30.31 | tzafrir | and Digium have been working on fixing that ommition in their driver in recent versions |
| 18:30.48 | ManxPower | tzafrir: They have had 5 years to fix it, it's a little late now. |
| 18:31.00 | Qwell | bug #? |
| 18:31.21 | DiDjCodt | anyone willing to help me? |
| 18:31.28 | ManxPower | The T-1/E-1 card will of course detect red/yellow alarms, but the analog ones don't detect anything. |
| 18:31.35 | ManxPower | ~ask |
| 18:31.36 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:31.37 | tzafrir | well, menelaut has had a long-standing bug on this one |
| 18:31.50 | tzafrir | and when it finally landed, it opened a can of worms |
| 18:31.50 | DiDjCodt | ManxPower: i asked previously ;) |
| 18:32.01 | DiDjCodt | ManxPower: ii have problems on dtmf |
| 18:32.21 | DiDjCodt | ManxPower: when calling to certain sip phones, i don't receveived their dtmf |
| 18:32.43 | ManxPower | DiDjCodt: It sucks to be you. |
| 18:32.50 | DiDjCodt | ? |
| 18:33.20 | DiDjCodt | ManxPower: i tried to put s,n,SIPDtmfMode() with inband, rfc.., info |
| 18:33.33 | tzafrir | DiDjCodt, there is no strict FIFO queue policy here ;-) |
| 18:33.56 | DiDjCodt | ok, i didn't know there was a queue, sorry ;) |
| 18:34.17 | budol | ~ata |
| 18:34.18 | jbot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:34.28 | budol | ~quintum |
| 18:34.35 | budol | hehe |
| 18:34.36 | Kobaz | it seems he meant there is no queue |
| 18:36.54 | *** join/#asterisk xacatecas (n=jkroon@iburst-41-213-38-114.iburst.co.za) |
| 18:37.45 | xacatecas | ok, how can I go about troubleshooting why a line sounds "funky"? essentially when i dial out over zap/1-1 to my cell number my cell never rings and the internal phone just hears a very nasty sound. |
| 18:39.28 | nny_1 | ManxPower: what do you use to define the user's cellphone number per extension? |
| 18:40.02 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 18:40.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:40.43 | nny_1 | ManxPower: basically trying to figure out how to have (CELL${FOO}) capture the specific extensions number when calling the new function |
| 18:42.00 | nny_1 | we are talking about capturing the EXTEN variable as it is and assigning a new variable with the same value, before calling the function |
| 18:42.09 | ManxPower | nny_1: We have a Set(OPER_DEST=5045551212) in each user's extension.conf extension, then my macros handle the rest |
| 18:42.24 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 18:42.26 | ManxPower | we don't use wildcard extensions to route calls to phones |
| 18:42.29 | ice_croft | ~book |
| 18:42.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:42.48 | xacatecas | o.O |
| 18:44.28 | ice_croft | where can i get difference between type=peer and type=friend? |
| 18:44.38 | *** join/#asterisk actros1840 (n=A@chello087207097013.chello.pl) |
| 18:44.43 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 18:45.05 | vap0rtranz | the book doesn't include all these new options! ex: hassip, register = yes/no, etc. |
| 18:45.21 | vap0rtranz | so i wish ppl stop just saying "the book" |
| 18:45.42 | ice_croft | i need just sip.conf file spec |
| 18:45.43 | Nugget | ok. |
| 18:45.47 | Nugget | s/the book/the wiki/ :) |
| 18:45.54 | ice_croft | ~wiki |
| 18:45.57 | vap0rtranz | Nugget: ty |
| 18:45.57 | ice_croft | damn |
| 18:46.16 | ice_croft | vap0rtranz> i need sip.conf specs anyway |
| 18:47.13 | vap0rtranz | ice_croft: ditto |
| 18:47.13 | Corydon76-vcch | There's a spec? |
| 18:47.13 | ice_croft | Corydon76-vcch> i thought it is |
| 18:47.21 | ice_croft | vap0rtranz> excuse me? |
| 18:47.30 | Corydon76-vcch | News to me |
| 18:48.04 | ice_croft | Corydon76-vcch> i just need to get difference between type=friend and type=peer, thats all |
| 18:48.15 | Corydon76-vcch | The nice thing about SIP is that there's so many standards to choose from |
| 18:48.24 | ice_croft | Corydon76-vcch> to create sip trunk |
| 18:48.38 | ice_croft | Corydon76-vcch> i need kinda secure trunk |
| 18:48.44 | ice_croft | bgg |
| 18:49.02 | Corydon76-vcch | ice_croft: when you make a call from the phone to Asterisk, a friend is identified by name. A peer, only by IP. |
| 18:49.41 | znoG | Corydon76-vcch: any ideas what the status of res_config_ldap is? |
| 18:49.47 | ice_croft | Corydon76-vcch> and a type=user? |
| 18:49.57 | jameswf | if you dont know your friends name maybe they need a new friend |
| 18:50.00 | znoG | Corydon76-vcch: i noticed you had assigned a huge bug report about it |
| 18:50.04 | Corydon76-vcch | There used to be a greater difference, but thanks to the aforementioned muddying of the SIP spec (thanks guys), there is otherwise very little difference now |
| 18:50.09 | vap0rtranz | Corydon76-vcch: so a friend sip header would look like [extension]@[provider] but a [ipaddr]@[provider] ?? |
| 18:50.33 | ice_croft | damn |
| 18:50.34 | Corydon76-vcch | vap0rtranz: who said anything about SIP headers? |
| 18:50.56 | vap0rtranz | Corydon76-vcch: oh you're talking about w/in *.conf |
| 18:50.57 | ice_croft | so there's no need to secret= ? |
| 18:51.26 | Kobaz | are there any secrets i should know about when linking up a sangoma t1 to a t1 from an avaya box |
| 18:51.54 | ManxPower | Kobaz: chances are you'll need a T-1 crossover cable (which is not the same as an ethernet crossover |
| 18:52.10 | Kobaz | yeah i have a crossover |
| 18:52.54 | Kobaz | i'm getting a pulse density violation on the line |
| 18:53.16 | ice_croft | bugoga |
| 18:53.43 | ManxPower | Kobaz: sounds more like an E-1 error than a T-1 error |
| 18:53.55 | Kobaz | hmm |
| 18:53.58 | Corydon76-vcch | Kobaz: call your mfg tech support for assistance |
| 18:54.33 | DiDjCodt | how do you know if you're on the queue ? |
| 18:54.53 | Kobaz | queue show |
| 18:56.49 | ice_croft | Corydon76-vcch> trunk with type=peer doesn't work |
| 18:57.00 | ice_croft | Corydon76-vcch> it need password anyway |
| 18:57.09 | ice_croft | Corydon76-vcch> *needs |
| 18:57.52 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:58.06 | *** join/#asterisk stoffell_h (n=stoffell@d51A4DB81.access.telenet.be) |
| 18:59.06 | ice_croft | Corydon76-vcch> so how can i establish sip trunk without type=friend. can u help? plz |
| 18:59.20 | Havokmon | Is there a way to have a queue or huntgroup randomly choose an extension and put the caller in VM? Looks like the defaults are either non-random selection, or no vm. But I'm currently using DIDs for queue agents so the call isn't supervised to get it work work. Messes up call stats though.. |
| 19:01.11 | pigpen | quick question about faxdetect.... |
| 19:01.34 | vap0rtranz | ice_croft: it looks like this changed. in 1.2, you have a separate peer and user; in 1.4 it now looks like everything (including the registration) is in peer. (but i know that's wrong) |
| 19:02.18 | pigpen | if I have a DID going to a device directly, and I am running faxdetect....can faxdetect grab it away from the device and re-route it? |
| 19:02.45 | vap0rtranz | pigpen: wish you had been here yesterday |
| 19:03.04 | pigpen | errrr....why? |
| 19:03.18 | vap0rtranz | pigpen: all the telco guys start screaming NO and even though it sounds nice, systems people suck |
| 19:03.19 | vap0rtranz | lol |
| 19:03.46 | pigpen | Ah... |
| 19:04.25 | pigpen | I am getting the feeling that if faxdetect is enabled, hard coding it to an exten is not a good idea.... |
| 19:04.44 | pigpen | maybe I'll stick it in a nice long wait after answer....to give faxdetect a chance. |
| 19:04.51 | pigpen | thanks anyway. |
| 19:04.56 | ManxPower | you want faxdetect when you CAN'T hardcode the fax number |
| 19:05.16 | pigpen | ManxPower, well, I have a mix of both. |
| 19:05.37 | nny_1 | ManxPower: do you know where you turn on the "feature" (or it's name) for comedian mail to allow press 0? I am looking in features.conf atm |
| 19:05.37 | pigpen | I just setup a macro with a db match for the destination. |
| 19:05.42 | *** join/#asterisk ddunavant (n=David@pool-96-231-80-138.washdc.east.verizon.net) |
| 19:06.09 | ManxPower | nny_1: try looking in voicemail.conf |
| 19:06.16 | nny_1 | ManxPower: ty |
| 19:06.53 | pigpen | ManxPower, anyway, nice talking to you again...but time to be "Pigpen, the fax boy" |
| 19:07.29 | errr | is there a way to run a script from asterisk and get the return values from it? |
| 19:07.30 | [TK]D-Fender | ManxPower: So if I want to change the LBO on my T1 because the moved the server away from the demarc & smartjack..... I should be looking in musiconhold.conf, right? |
| 19:07.59 | [TK]D-Fender | </sarcasm> |
| 19:08.04 | pigpen | heh.. |
| 19:08.05 | [TK]D-Fender | :) |
| 19:08.31 | pigpen | [TK]D-Fender, hey, try #trixbox |
| 19:08.43 | pigpen | they will have a good answer I am sure. |
| 19:09.36 | xacatecas | what would a response of 488 indicate? |
| 19:10.17 | tzafrir | no, no. We have just one config file: users.conf |
| 19:10.18 | xacatecas | SIP/2.0 488 Not acceptable here ... nm. |
| 19:10.29 | [TK]D-Fender | xacatecas: Typically a Codec negotiation failure |
| 19:10.35 | *** join/#asterisk svenna (n=svenna@p548D082B.dip0.t-ipconnect.de) |
| 19:10.43 | xacatecas | argh ... how to fix that? |
| 19:13.12 | [TK]D-Fender | ~users.conf |
| 19:13.13 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 19:13.45 | xacatecas | fixed. |
| 19:14.42 | J4k3 | asterisk = a huge pile of gayass worthless picky-assed .conf files |
| 19:14.56 | vap0rtranz | J4k3: *cheers* |
| 19:14.58 | J4k3 | realtime should be default. |
| 19:15.17 | J4k3 | if your platform can't run an sql server, or the user can't, thats their problem. |
| 19:15.40 | Jason99 | Does anyone here know if in sip.conf, call-limit=0 mean unlimited? |
| 19:16.26 | Jason99 | It would appear that once you apply a limit, if you remove the call-limit entry the previous limit is still in place until you apply a new one |
| 19:17.54 | Corydon76-vcch | Jason99: that is the case for ALL config options |
| 19:17.58 | *** join/#asterisk chigital (n=chigital@p4FE5FE4A.dip.t-dialin.net) |
| 19:18.24 | Corydon76-vcch | Once set, it remains set until you explicitly change it or restart the process |
| 19:18.47 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 19:19.08 | J4k3 | (asterisk seems like a project optimized toward dragging people into support channels then making fun of them) |
| 19:19.10 | *** join/#asterisk cmasternig (n=cmastern@87.249.74.210) |
| 19:19.53 | Corydon76-vcch | J4k3: on the battle between us making it foolproof and the Universe on making a better fool, the Universe is winning |
| 19:19.58 | [TK]D-Fender | J4k3: No, that is jsut part of the bundled solution. |
| 19:20.24 | ryanteem | Does anyone have some resources on the issue of matching only the last peer in the sip.conf based on host IP? |
| 19:20.30 | Havokmon | You could say the same thing about 'Linux'... What are you going to do with just a kernel? |
| 19:20.33 | J4k3 | Corydon76-vcch: yeah, but I'm wondering on which side of the developer/user fence the fools are most highly concentrated. |
| 19:20.36 | J4k3 | :D |
| 19:20.57 | cmasternig | Can anybody tell me how to log CDR to mysql database? Am using asterisk BE i am sure if it comes with mysql suppor |
| 19:21.27 | nny_1 | hehehe |
| 19:21.31 | nny_1 | yay cart and horse |
| 19:21.34 | J4k3 | hehe |
| 19:21.53 | J4k3 | I dunno, I'm hoping asterisk is used as a proof-of-concept, then somebody actually runs and makes something good |
| 19:21.57 | nny_1 | so anyone have a bulletproof way to capture the extension that someone dialed to get to voicemail (including transfers)? |
| 19:22.03 | J4k3 | theres enough 'competing' projects now that I suspect that may be the case. |
| 19:22.24 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 19:22.29 | xacatecas | is there a way to ask asterisk to delay for 2 seconds before dialing out on a zap trunk? |
| 19:22.30 | nny_1 | I was gonna capture $EXTEN before calling a new variable, but i wasn't sure if a transfer would work |
| 19:22.34 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 19:22.35 | *** join/#asterisk xoqa (n=xoqa@unaffiliated/hydan) |
| 19:22.36 | nny_1 | eh i can test nm |
| 19:22.39 | *** join/#asterisk L1Nux3R (n=None@84.36.148.225) |
| 19:22.47 | Havokmon | xacatecas: w I believe |
| 19:22.53 | vap0rtranz | isn't it wrong when the From: and To: are the same when trying to register a sip channel from a provider?! |
| 19:23.02 | xacatecas | Havokmon, yes, where do I add that? |
| 19:23.04 | nny_1 | lol |
| 19:23.12 | nny_1 | (${SIP_HEADER(TO)}) |
| 19:23.12 | BobLutz | Teehee |
| 19:23.18 | xoqa | i'm looking for a software phone, could i use asterisk for this purpose? |
| 19:23.24 | nny_1 | should always have the extension, right? |
| 19:23.31 | BobLutz | xoqa: Absolutely not |
| 19:23.33 | vap0rtranz | nny_1: that's a yes? |
| 19:23.38 | Havokmon | xacatecas: In your FreePBX call prefix section ;) |
| 19:24.00 | nny_1 | vap0rtranz: nah i am trying to get a variable that is *always* the extension being dialed |
| 19:24.13 | xacatecas | Havokmon, n00b here. |
| 19:24.29 | *** join/#asterisk simbol76ss (n=simbol@host228-198-dynamic.49-82-r.retail.telecomitalia.it) |
| 19:24.33 | J4k3 | xoqa: asterisk is a 'soft pbx', you need a phone. |
| 19:25.44 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:28.52 | Havokmon | xacatecas: same here - that's why I have a front-end. w is just a prefix, so wherever yourre passing the digits to the trunk, stick in the w |
| 19:29.13 | xacatecas | Havokmon, trixbox. makes this insanely difficult. |
| 19:29.17 | xacatecas | more so than needed. |
| 19:29.21 | xacatecas | imho. |
| 19:29.34 | Havokmon | xacatecas: same here - there's a prefix section under trunks |
| 19:29.35 | Qwell | ~trixbox |
| 19:29.36 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 19:30.01 | graphx | xoqa: What OS are you going to run the softphone on? |
| 19:30.04 | xacatecas | Qwell, fully agreed, however, they don't always respond :) |
| 19:30.09 | xoqa | graphx: gentoo |
| 19:30.16 | J4k3 | 'it doesn't work and we've got great excuses as to why we can't offer support' |
| 19:30.18 | Qwell | xacatecas: doesn't mean we can/will help you |
| 19:30.26 | xacatecas | fair enough. |
| 19:30.41 | jameswf | jbot: no nowwhat is <reply> So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E\ |
| 19:30.42 | jbot | jameswf: okay |
| 19:30.43 | Qwell | Fonality can pony up the resources to support their distro |
| 19:30.58 | Qwell | obviously people in the community don't want to/can't do it :) |
| 19:31.27 | cmasternig | Please help !!!!!!! i need to log CDR to my sql |
| 19:32.45 | vap0rtranz | Qwell: i ported a trixbox @ 1.2 to asterisknow @ 1.4 and cut the the *.conf lines down significantly slimmer. they were almost illegible |
| 19:32.47 | graphx | xoqa: You have your choice of a couple that come package in most distro's Ekiga, Twinkle, and Kphone |
| 19:33.00 | J4k3 | BobLutz: hey, you suck at being GM's CEO, but you might be qualified to run their pbx ;) |
| 19:33.01 | Qwell | vap0rtranz: s/almost// |
| 19:33.10 | Qwell | s/illegible/intelligible/ |
| 19:33.13 | jameswf | kphone is butt twinkle is ok I like moziax |
| 19:33.17 | xoqa | xoqa: i've used twinkle and i'm considering linphone |
| 19:33.19 | vap0rtranz | Qwell: hehe. hell to troubleshoot |
| 19:33.27 | Qwell | s/to troubleshoot// |
| 19:33.28 | xoqa | graphx** ^ |
| 19:33.48 | graphx | xopa: but there are other out there are others... like jameswf pointed out... |
| 19:34.16 | nny_1 | dude did i just see "butt twinkle"? |
| 19:34.24 | Qwell | nny_1: yes, yes you did. |
| 19:34.26 | graphx | lol |
| 19:34.28 | nny_1 | lol |
| 19:34.39 | tzanger | ... butt twinkle? |
| 19:34.46 | J4k3 | "butt twinkle" sounds like a great way to describe the asterisk conf file system. |
| 19:34.48 | tzanger | I use twinkle but it's got its flaws |
| 19:34.50 | graphx | I don't even want to know what that is....lol |
| 19:34.59 | xoqa | jameswf: interesting, i might try that as well |
| 19:35.01 | jameswf | butt twinkle what happens when you eat chili with golgschlager |
| 19:35.05 | tzanger | they way it opens the audio devices is *fucked* |
| 19:35.16 | tzanger | it opens in blocking mode, goes "oh, I need nonblocking" and closes/reopens |
| 19:35.22 | nny_1 | lol |
| 19:35.23 | tzanger | totally kerfuckers the new bluetooth system |
| 19:35.27 | *** join/#asterisk svenna_ (n=svenna@p548D0D84.dip0.t-ipconnect.de) |
| 19:35.32 | tzanger | cvs head for bluez-utils fixes it |
| 19:35.49 | graphx | no the way it opens the audo device is through butt twinkle.... come on tzanger... you should know this... lol |
| 19:35.56 | tzanger | graphx: heh |
| 19:36.06 | tzanger | the way it opens the audio device is just fucked.. I have no idea why it does it the way it does |
| 19:36.24 | nny_1 | is this variable correct: CELL${SIP_HEADER(TO)} |
| 19:36.32 | nny_1 | trying to get ex: CELL101 |
| 19:37.47 | BobLutz | xoqa: I use Linphone, I recommend it |
| 19:37.54 | BobLutz | J4k3: I drive a Hyundai haha |
| 19:38.12 | graphx | Linphone is not bad.. I have used it... |
| 19:38.50 | xacatecas | linphone had many bugs last time i used it (2005) |
| 19:39.15 | BobLutz | Linphone was the easiest for me to setup I found |
| 19:39.19 | Jason99 | nny_1: I think you'd want to use CELL${EXTEN} ? |
| 19:39.25 | graphx | I would have to think they cleened it up a bit.. it's 2008 |
| 19:39.35 | BobLutz | When I was first trying out softphones, I knew nothing about Asterisk/Asterisk-related things |
| 19:39.36 | ZPertee | I have festival installed and asterisk. what is the easiest quickest way to connect them? |
| 19:40.24 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:40.46 | ZPertee | sorry didn't mean to ask question and then disconnect hopefully I didn't miss the advice |
| 19:41.21 | Havokmon | sounds like a party line |
| 19:41.49 | BobLutz | ZPertee: Have you looked at festival.conf? |
| 19:42.03 | BobLutz | It shouldnt be any more difficult then calling Festival() in the dialplan |
| 19:42.38 | ZPertee | if I try to use festival in the dialplan it won't work. says it can't connect. how do I configure festival.conf? |
| 19:43.05 | ZPertee | is this http://www.voip-info.org/wiki/view/Asterisk+config+festival.conf all I need |
| 19:43.40 | BobLutz | ZPertee: let me check |
| 19:44.01 | nny_1 | Jason99: i would except the EXTEN is "o" at that point |
| 19:44.29 | BobLutz | ZPertee: Yea you should be good with that |
| 19:44.31 | nny_1 | Jason99: I could capture EXTEN and assign it to a new variable, but i am not sure it would work on a transfer |
| 19:44.58 | BobLutz | ZPertee: Double check and make sure that Festival works, that Festival is running on port 1314, and that your box can resolve localhost |
| 19:45.59 | BobLutz | ZPertee: and mine would go in /etc/asterisk/ (with sip.conf, extensions.conf, etc) |
| 19:46.19 | tzanger | I wanted a KDE sip phone.. twinkle works well enough :-) |
| 19:46.23 | ZPertee | BobLutz: ok thanks. I'll try it and hopefully it works. I hadn't to reinstall my OS so I am starting over. I'll try your advice though thanx |
| 19:46.36 | ZPertee | sorry I mean had to reinstall |
| 19:46.51 | BobLutz | ZPertee: what OS..out of curiosity |
| 19:47.00 | ZPertee | Ubuntu |
| 19:47.37 | ZPertee | BobLutz: for an average linux user it is easy for me to maintain and has good repository |
| 19:48.05 | BobLutz | Ubuntu was the first distro I really sank my teeth into, but have since moved on |
| 19:48.14 | ZPertee | BobLutz to what? |
| 19:48.19 | BobLutz | I tell people who want to get into Linux to start there |
| 19:48.22 | BobLutz | Arch |
| 19:48.44 | ZPertee | BobLutz: why did you switch? whats the benefits? |
| 19:48.50 | fujin | oh fuck arch, you've gotta be shitting me |
| 19:48.53 | fujin | might aswell run LFS |
| 19:48.57 | fujin | be a real sadist |
| 19:49.05 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:49.08 | BobLutz | Arch is LFS is GNU/Linux |
| 19:49.16 | fujin | it's shit that's what it is |
| 19:49.23 | BobLutz | lol? |
| 19:49.27 | fujin | completely unmanageable |
| 19:49.27 | tzanger | BobLutz: you are not a very good advocate for linux... new users would get stuck on arch... I am an old slacker but I run kubuntu on desktop now and recommend something like that to newbies |
| 19:49.44 | BobLutz | I tell people who want to get into Linux to start there (referring to Ubuntu) |
| 19:49.51 | tzanger | slackware rox muh sox but I've been using linux long enough to know that it's not a place to start newbies out with |
| 19:50.04 | fujin | tzanger: it's not the place for a LOT of standard linux stuff |
| 19:50.13 | graphx | I agree with BobLutz... Ubuntu is great for newbies |
| 19:50.22 | tzanger | fujin: whyso? nothing standing in your way |
| 19:50.26 | BobLutz | fujin: You run BSD? |
| 19:50.34 | fujin | Fuck BSD too, that's nearly as dumb |
| 19:50.37 | tzanger | fujin: no package manager, no fucked up init wrappers or config managers... it's about as "pure" as you can get |
| 19:50.40 | BobLutz | NT? |
| 19:50.41 | Nugget | OS X on the desktop, FreeBSD in the server room. |
| 19:50.48 | Nugget | Linux for machines where there's no other choice. |
| 19:50.51 | fujin | tzanger: that's true |
| 19:50.55 | fujin | tzanger: no manageability |
| 19:51.04 | fujin | "what packages have security fixes" |
| 19:51.04 | BobLutz | fujin: ALL the manageability |
| 19:51.12 | J4k3 | osx has no place, ever. |
| 19:51.15 | ZPertee | BobLutz: I know an embedded linux consultant who runs ubuntu on all of his laptops without any problem. so maybe it works ok for non newbs as well |
| 19:51.15 | tzanger | although slapt-get and its derivatives are pretty clean |
| 19:51.19 | tzanger | but bsd? why? What' the point? |
| 19:51.28 | Nugget | OS X is the most popular Unix on the planet, so clearly other people disagree. :) |
| 19:51.32 | tzanger | ZPertee: I am an embedded linux consultant :-) |
| 19:51.36 | BobLutz | ZPertee: yea it works great for anyone...I just didnt want to do things the "Ubuntu" way |
| 19:51.51 | J4k3 | BSD works very damned well... linux is a great way to go from release-to-release knocking bugs down. |
| 19:52.04 | tzanger | Nugget: if Apple chose QNX you could claim the same type of thing... just because they don't know what it's running doesn't mean it's popular |
| 19:52.07 | Nugget | Linux is for hackers, FreeBSD is for sysadmins. BSD's management tools and infrastructure make site management a breeze. Way moreseo than any Linux distro I've ever used. |
| 19:52.11 | tzafrir | Nugget, it is not popular as a BSD |
| 19:52.13 | J4k3 | I'll admin microsoft crap first. |
| 19:52.19 | fujin | lol @ freebsd is for sysadmins |
| 19:52.22 | fujin | maybe sadist sysadmins |
| 19:52.25 | Nugget | hardly |
| 19:52.34 | BobLutz | fujin: What distro do you use? |
| 19:52.37 | tzanger | J4k3: weird, I have no problems whatsoever... windows management sucks ass |
| 19:52.39 | fujin | Ubuntu |
| 19:52.42 | Nugget | ha ha |
| 19:52.43 | J4k3 | I've been running FreeBSD on my servers since july 1996 |
| 19:52.53 | J4k3 | and I've had exactly ZERO downtime minutes due to software issues |
| 19:53.01 | J4k3 | (and exactly zero downtime due to hardware, either...) |
| 19:53.05 | graphx | J4k3: You got that right about M$... --CRAP!!! |
| 19:53.14 | tzanger | BSD's development is glacial, but I cannot argue against its "just works"-ness |
| 19:53.23 | ZPertee | BobLutz: yeah makes sense. for my server I use Ubuntu and on I use an apple laptop (allow me to cleanly run Windoze, OS X, and Linux) |
| 19:53.29 | J4k3 | graphx: I'll put microsoft a few steps above the crapshoot development model of Linux and related projects. |
| 19:53.51 | J4k3 | and well... you can't use linux for commercial items, due to the gpl |
| 19:54.02 | tzanger | J4k3: uh, no |
| 19:54.03 | J4k3 | sell an embedded device with linux, regret it later. |
| 19:54.03 | BobLutz | ZPertee: Really I just didnt like how Ubuntu (or a lot of other distro's) comes with software pre-installed |
| 19:54.04 | tzafrir | J4k3, nice trolling |
| 19:54.06 | tzanger | that's totally wrong |
| 19:54.08 | fujin | that was some epic trolling |
| 19:54.10 | graphx | J4k3: You be supprised if you opened up the M$ code to see what you find... |
| 19:54.16 | fujin | any OSS consultant will say "rhel or nothing" |
| 19:54.18 | tzanger | I am at a contract right now where they make embedded linux jukeboxes |
| 19:54.22 | tzafrir | J4k3, come back here when you actually want to talk about Asterisk |
| 19:54.23 | J4k3 | tzafrir: what? so jerkoff kids can break the boxes and return them? |
| 19:54.26 | tzanger | there's 30k of them out in the field, 10k of those connected ot the internet |
| 19:54.29 | tzanger | www.touchtunes.com |
| 19:54.41 | J4k3 | tzanger: high end projects are one thing, where nobody will ever care to hack the unit. |
| 19:54.55 | J4k3 | tzanger: linksys proved that linux-at-the-consumer-embedded-level is a bad idea. |
| 19:55.04 | tzanger | J4k3: your understanding of the GPL and commercial applications of linux aren't even wrong, they're FUD |
| 19:55.11 | tzanger | J4k3: how did linksys prove that? |
| 19:55.18 | tzafrir | J4k3, lynksys were clueless about the license |
| 19:55.20 | J4k3 | tzanger: you have to provide code, which gives children the ability to octuple the quantity of returns. |
| 19:55.30 | J4k3 | ITS THE USERS NOT THE GPL ITSELF |
| 19:55.31 | tzafrir | They would have violated the BSD license as well |
| 19:55.39 | tzanger | J4k3: I also designed an embedded communications board that runs Linux on m68k/coldfire that sits in some of the harshest and most demanding environments you can think of with zero problems |
| 19:55.45 | tzanger | J4k3: again, wrong |
| 19:56.08 | tzanger | J4k3: if you write a device driver and use GPL code, then yes. If you port to a new processor, yes. |
| 19:56.16 | tzanger | you can write a totally closed-source device driver and not have to provide code |
| 19:56.23 | *** join/#asterisk Boz0 (n=jerrk@CBL217-132-247-46.bb.netvision.net.il) |
| 19:56.27 | J4k3 | linksys would have had damn near zero returns on the wrt54g had they never put linux on it.. the hardware is good, the firmware was good enough, but the quantity of people out hacking them, fucking up, and returning them, is what caused linksys to run to vxworks and nearly-totally-worthless hardware. |
| 19:56.27 | Boz0 | hi |
| 19:56.28 | Qwell | J4k3: simple solution; "If you brick it, you've voided your warranty." |
| 19:56.37 | J4k3 | Qwell: but thats not true, legally. |
| 19:56.40 | tzanger | you can also write totally closed-source applications that do not link to GPL libraries or use GPL code and never have to release the osurce |
| 19:56.48 | tzanger | J4k3: you're spreading FUD, which is unacceptable |
| 19:56.53 | J4k3 | Qwell: in the USA cisco has to *prove* you hacked it and broke it |
| 19:57.07 | J4k3 | and it'd cost cisco more to do that for every damned wrt54g return than it would be to just replace the units |
| 19:57.11 | J4k3 | either way its a loss |
| 19:57.13 | graphx | J4k3 not so... releasing code in open fashion has shown to help with fixing bugs, security flaws... on the other hand look at how M$ is doing it...and how long it takes them to fix it and get things right... it's obvious that the open source way works better... |
| 19:57.19 | J4k3 | and either way its an avoidable loss - don't embed linux in consumer products. |
| 19:57.27 | tzanger | J4k3: and how is that any different with people returning bricked Wiis and PS3s due to fucked modchip installs?? |
| 19:57.44 | pat2man | J4k3: even with returns linksys probably makes more money of the wrtgs than any other home router company |
| 19:57.45 | J4k3 | tzanger: very. |
| 19:57.47 | tzafrir | J4k3, funny thing how Linux is so popular in cnsumer products |
| 19:57.51 | tzanger | J4k3: howso? |
| 19:58.01 | *** join/#asterisk actros1840 (n=A@chello087207097013.chello.pl) |
| 19:58.04 | tzanger | J4k3: invalid returns are a problem for *anything* |
| 19:58.06 | J4k3 | pat2man: actually no, cisco cries horribly about it since the average return costs them around $50 |
| 19:58.07 | BobLutz | fujin: See what you started? |
| 19:58.13 | J4k3 | and its not like cisco makes more than about $10/unit on them |
| 19:58.21 | J4k3 | prolly more like $5 |
| 19:58.28 | fujin | BobLutz: I'm right :) |
| 19:58.32 | BobLutz | haha |
| 19:58.34 | graphx | Microsoft is now looking in to being more Open with OpenSource... explain that... |
| 19:58.41 | tzanger | J4k3: then they're driving the RMA system wrong. This is Cisco's first foray into general consumer products as well, so I wouldn't imagine they're doing it right |
| 19:58.42 | BobLutz | !open |
| 19:58.55 | J4k3 | tzanger: it existed long before the buyout |
| 19:59.07 | cpm | graphx, they are bullshitting, they don't have a clue what that even means, easy to say though, get's ' |
| 19:59.10 | J4k3 | cisco bought them out around the same time the v3's releases. |
| 19:59.11 | cpm | em lots of press |
| 19:59.12 | tzafrir | graphx, OpenSolaris is copyleft as well, and hence a problem to MS as well |
| 19:59.13 | J4k3 | err released |
| 19:59.13 | tzanger | J4k3: you accept the RMA and take a credit card. if the RMA was due to legit reaons, you don't charge the card. otherwise you charge it and you make your policy very clear |
| 19:59.37 | J4k3 | tzanger: do you know what an attorney costs? |
| 19:59.44 | tzanger | J4k3: but yeah... your ideas of how GPL works in the commercial world are hideously wrong. |
| 19:59.50 | tzanger | J4k3: where is an attorney needed? |
| 20:00.07 | J4k3 | tzanger: you're going to have people say 'fuck that, lets go to court' and cisco will have *nothing* to respond to that, short of several hundred or possibly up to a couple thousand dollars worth of cost *per RMA* |
| 20:00.22 | tzanger | J4k3: bring on the class action |
| 20:00.29 | tzanger | J4k3: cisco can *prove* the devices were fucked with |
| 20:00.35 | J4k3 | not really |
| 20:00.42 | J4k3 | and, again |
| 20:00.57 | J4k3 | just because you reflashed th firmware doesn't mean you took complete responsibility for the unit |
| 20:01.00 | J4k3 | just the software itself |
| 20:01.13 | J4k3 | so if the hardware fails, running stock firmware or dd-wrt or openwrt or whatever, they're responsible for it |
| 20:01.28 | tzanger | J4k3: if they didn't make a safe reflash that is their problem (now that I think of it, it is too easy to brick a wrt, even with official firmware) |
| 20:01.35 | tzafrir | J4k3, you know, the same applies for any firmware |
| 20:01.44 | J4k3 | http://en.wikipedia.org/wiki/Magnuson-Moss_Warranty_Act |
| 20:01.46 | tzanger | the embedded comm modules I designed cannot be bricked with standard reflash tools |
| 20:01.56 | tzafrir | You never know what software you'll find in there |
| 20:02.13 | Qwell | J4k3: see the part where it says hardware? or the part about frying something else from negligence? |
| 20:02.14 | J4k3 | tzanger: Linksys's CFE always sucked. |
| 20:02.24 | tzanger | mcf5282 has bootloader in its flash that allows you to always recover and try again... also, two kernels and two rom filesystems... main and backup |
| 20:02.26 | Qwell | yeah, those are excluded from magnuson moss |
| 20:02.36 | J4k3 | Qwell: prove it. |
| 20:02.36 | robmac67 | J4K3: I think Linksys changed to VXWorks to reduce the memory required and therefore material costs - you can buy the Linux variants which have more memory installed |
| 20:02.41 | Qwell | read the link you pasted |
| 20:02.48 | tzanger | J4k3: anyway my point is that it isnt' linux that fucked cisco. linksys fucked cisco, and cost-cutting fucked cisco |
| 20:02.48 | J4k3 | see, the thing with magnuson-moss is the WARRANTY HOLDER has to prove it |
| 20:02.57 | J4k3 | so cisco will have to prove, in every case, individually, that the user bricked the unit and not them |
| 20:03.05 | J4k3 | this would |
| 20:03.06 | tzanger | how many phones do you know of that have software mods for them that people can brick? How many run linux? |
| 20:03.17 | J4k3 | A> run every cisco consumer-customer off, since warranty service would turn into a legal case |
| 20:03.20 | *** mode/#asterisk [+b %J4k3!*@*] by Qwell |
| 20:03.21 | tzafrir | J4k3, consider strange cases such as: http://arstechnica.com/news.ars/post/20071204-mpaas-university-toolkit-hit-with-dmca-takedown-notice-after-gpl-violation.html |
| 20:03.27 | Nugget | don't need mods to brick my cisco phones! |
| 20:03.30 | Nugget | cisco is way ahead of the game |
| 20:03.31 | Qwell | You're trolling. Enough. |
| 20:03.32 | tzanger | J4k3: where's the outgoing QA sheet that shows it booted and this was the checksum of the ROM at the time? |
| 20:03.38 | tzanger | Nugget: hahahahahhahaha |
| 20:04.24 | *** mode/#asterisk [-b %J4k3!*@*] by Qwell |
| 20:04.31 | *** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 20:04.32 | tzanger | J4k3: qwell's right, this is hideously offtopic, I'd be happy to continue this with you in another channel |
| 20:05.01 | tzanger | (channel so anyone else could join) -- maybe #ciscolinuxsux |
| 20:05.02 | [TK]D-Fender | tzanger: I think #bile is free... |
| 20:05.06 | tzanger | haha |
| 20:05.13 | Havokmon | MS has never been responsbile for PCs crashing - good luck :) |
| 20:05.38 | *** mode/#asterisk [+b *!*n=jsuter@*.us] by Qwell |
| 20:05.39 | *** kick/#asterisk [J4k3!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 20:05.44 | Qwell | perm |
| 20:05.49 | tzanger | ? |
| 20:06.06 | Qwell | if that gets removed, I'm putting it back :) |
| 20:06.24 | tzanger | did he insult your mother in privmsg or something? |
| 20:06.49 | Qwell | something like that |
| 20:06.49 | *** join/#asterisk mike-ekim (n=mike@apoc.digiport.com) |
| 20:08.14 | mike-ekim | [Mar 6 14:43:00] WARNING[2606]: channel.c:3232 ast_request: No channel type registered for 'Zap' |
| 20:08.15 | cpm | what was that about? well, maybe I don't wanna know. Jake gets in moods sometimes |
| 20:08.17 | mike-ekim | [Mar 6 14:43:00] WARNING[2606]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 20:08.23 | mike-ekim | I am not too good with PRI |
| 20:08.49 | *** join/#asterisk rhett (n=rhett@cpe-72-177-2-151.austin.res.rr.com) |
| 20:08.51 | Greek-Boy | can anybody suggest a good billing solution for asterisk? |
| 20:09.01 | [TK]D-Fender | mike-ekim: You don't seem to have build zaptel FIRST, followed by * |
| 20:09.03 | *** part/#asterisk rhett (n=rhett@cpe-72-177-2-151.austin.res.rr.com) |
| 20:09.09 | [TK]D-Fender | mike-ekim: recompile everything |
| 20:09.25 | mike-ekim | [Tk]D-Fender: this was a working system |
| 20:09.34 | mike-ekim | yesterday this was working perfectly finre |
| 20:09.56 | Qwell | who's angry? |
| 20:10.31 | Qwell | If I was angry, you'd know it :) |
| 20:10.55 | [hC] | Last time i saw qwell angry i came out with brand marks on my back |
| 20:11.05 | [hC] | and he wasnt even angry at me! |
| 20:11.06 | jameswf | ah well atleast you got some goat meat out of it :) |
| 20:11.16 | mike-ekim | is this a known error for any physicall outages as well? |
| 20:11.30 | Qwell | Nugget: does that mean I don't have to upgrade them myself? ;) |
| 20:11.43 | Qwell | I can't see how that would make anybody angry, heh |
| 20:11.58 | [hC] | Qwell: hey, does chan_skinny support 3way calling on the phone for a 7970? |
| 20:12.09 | mike-ekim | ? |
| 20:12.19 | jameswf | he put grandstream firmware on your polycomm... how did that even work |
| 20:12.44 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 20:12.52 | CrashHD | Hey All |
| 20:13.34 | ZPertee | ~bot |
| 20:13.35 | jbot | I ain't no stinkin' bot. I am a finely tuned and hand crafted tool. Oh wait... I guess I am a bot (that you should not abuse). |
| 20:14.06 | marc7 | sorry for the silly question guys, how do you name extensions for the purpose of using Goto? for example, if you've used s,1, then s,n repeatedly, isn't it s,n(special) and then Goto(s,special) |
| 20:14.42 | ZPertee | ~bot sangoma |
| 20:14.54 | [TK]D-Fender | mike-ekim: "channel not implemented measn at the very lease chap_zap.so did not load. Try to load it manually |
| 20:15.04 | xoqa | were you guys ever able to use linphone in the console? |
| 20:15.06 | jameswf | ~book |
| 20:15.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:15.14 | mike-ekim | can anyone help me with zap errors? |
| 20:15.15 | jameswf | marc7: ch 5 |
| 20:15.15 | [TK]D-Fender | mike-ekim: "module load chan_zap.so" |
| 20:15.23 | marc7 | jameswf: thanks |
| 20:15.26 | mike-ekim | already did that |
| 20:15.30 | mike-ekim | unloaded and loaded |
| 20:15.45 | [TK]D-Fender | mike-ekim: and? |
| 20:15.51 | ZPertee | jameswf: how do I get the bot in private message I want to look some stuff up but dont want to flood channel |
| 20:16.01 | mike-ekim | well: on loading it gives errors too |
| 20:16.04 | jameswf | <PROTECTED> |
| 20:16.05 | Qwell | ZPertee: you can just msg him |
| 20:16.08 | [TK]D-Fender | ZPertee: just PM it |
| 20:16.22 | ZPertee | thanx |
| 20:16.30 | mike-ekim | <PROTECTED> |
| 20:16.30 | mike-ekim | [Mar 6 15:16:03] WARNING[2599]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address |
| 20:16.34 | mike-ekim | [Mar 6 15:16:03] ERROR[2599]: chan_zap.c:7160 mkintf: Unable to open channel 1: No such device or address |
| 20:16.37 | mike-ekim | here = 0, tmp->channel = 1, channel = 1 |
| 20:16.39 | mike-ekim | [Mar 6 15:16:03] ERROR[2599]: chan_zap.c:10466 build_channels: Unable to register channel '1-8' |
| 20:16.42 | mike-ekim | [2~ |
| 20:16.46 | mike-ekim | sorry |
| 20:16.55 | [TK]D-Fender | mike-ekim: Please don't do taht again. |
| 20:17.16 | [TK]D-Fender | mike-ekim: pastebin "ztcfg -vvvv" , "cat /proc/interrupts" |
| 20:17.19 | [TK]D-Fender | ~pb |
| 20:17.20 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:17.22 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:18.58 | CrashHD | so how has everyone been |
| 20:19.02 | CrashHD | channel seems packed |
| 20:19.13 | CrashHD | been awhile since I've hung around |
| 20:20.27 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:20.54 | djs26 | Wow, I am getting asterisk running for sip to sip internal calling atm, and am using xlite to test, but when I call something like voicemail, the audio is horrendous, like bad cell phone reception. The box is OBSD 4.2, asterisk 1.2.22, hardware is xeon 2.0, 3GB ram. From what I see of usage, it is using g711u as codec. |
| 20:21.16 | mike-ekim | [TK]D-Fender: ive already tried that i found that on a google doc |
| 20:21.36 | [TK]D-Fender | mike-ekim: I want to SEE IT. |
| 20:22.00 | djs26 | Is this typical of xlite, or am I to blame? |
| 20:22.13 | mike-ekim | ok |
| 20:22.17 | mike-ekim | let me paste results in pastebin |
| 20:22.18 | ManxPower | mike-ekim: I'm sure [TK]D-Fender is flattered that you think he is Google. |
| 20:22.48 | [TK]D-Fender | ManxPower: He didn't paste his stuff anywhere... Google has nothing to FIND. |
| 20:22.54 | jameswf | between me and google I know everything... probably better to ask google first |
| 20:23.04 | [TK]D-Fender | ManxPower: I don't want someone else's recollection of a problem, I want HIS system status |
| 20:23.10 | [TK]D-Fender | ManxPower: and... |
| 20:23.13 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 20:23.14 | [TK]D-Fender | ~[TK]D-Fender |
| 20:23.15 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 20:23.17 | [TK]D-Fender | ;) |
| 20:23.45 | mike-ekim | thats cute. |
| 20:23.49 | ManxPower | [TK]D-Fender: one of the significant diffences between us is that if a user is being stupid you still work with them. |
| 20:23.52 | mike-ekim | ztcfg = http://pastebin.ca/index.php |
| 20:23.57 | mike-ekim | ah |
| 20:24.04 | mike-ekim | http://pastebin.ca/930739 |
| 20:24.18 | CrashHD | lol |
| 20:24.22 | CrashHD | manx no patience? |
| 20:24.24 | CrashHD | :) |
| 20:24.36 | jameswf | I saw a book once called AOL for dummies... I was like heh yeah it is |
| 20:24.38 | mike-ekim | interrupts = http://pastebin.ca/930743 |
| 20:24.43 | lirakis | [TK]D-Fender: i like the google proxy .. thats good |
| 20:26.15 | [TK]D-Fender | mike-ekim: You don't seem to have anything defined in zaptel.conf |
| 20:26.42 | [TK]D-Fender | mike-ekim: And your card is sharing an IRQ with a LOT of other stuff.... |
| 20:26.45 | mike-ekim | [TK]D-Fender: that is weird though because noone has changed configuration, and this just happened |
| 20:26.49 | ManxPower | CrashHD: If a user argues with me about doing what I tell them to do, that' not "no patience". This is not drug rehab, you are not forced into anything. |
| 20:26.51 | mike-ekim | I am assuming that its always been like this |
| 20:26.55 | [TK]D-Fender | mike-ekim: go LOOK |
| 20:27.00 | [TK]D-Fender | ~assume |
| 20:27.01 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 20:27.01 | ManxPower | mike-ekim: did you try to reboot? |
| 20:27.02 | [TK]D-Fender | ^^^^^^^^^^^ |
| 20:27.06 | mike-ekim | yes |
| 20:27.21 | fujin | version control your /etc/asterisk |
| 20:27.23 | mike-ekim | i rebooted asterisk, and the whole system |
| 20:27.36 | *** join/#asterisk jdg (n=jdg@203.185.179.116) |
| 20:27.42 | ManxPower | mike older analog cards sometimes just stop working until a reboot, just wanted to make sure that was not the case here. |
| 20:27.50 | mike-ekim | ooh. |
| 20:28.09 | ManxPower | specifically older TDM400P, I don't have personal experience with any other analog cards |
| 20:28.24 | CrashHD | ManxPower: I agree with you, I have a very similar attitude |
| 20:28.35 | CrashHD | or even worse when they want you to do the leg work for them |
| 20:29.05 | mike-ekim | ManxPower: you sugggest rebooting the whole system? |
| 20:29.10 | mike-ekim | or for the modules? |
| 20:29.25 | ManxPower | mike-ekim: not if you did iy already |
| 20:29.32 | mike-ekim | oh ok |
| 20:29.45 | mike-ekim | could this be a cable that these ****heads might of unplugged?? |
| 20:29.51 | ManxPower | Do you have a TDM400P? |
| 20:30.15 | ManxPower | mike-ekim: power cable going into the card? |
| 20:30.26 | mike-ekim | i think so. not too sure just know theyre using PRI |
| 20:30.37 | mike-ekim | thrers a power cable into the card? |
| 20:30.41 | ManxPower | Then you are NOT using an analog card and so my comments do not apply to you |
| 20:31.05 | mike-ekim | .. |
| 20:31.22 | mike-ekim | I have not seen the physical setup |
| 20:31.25 | mike-ekim | i am familiar with sip |
| 20:31.28 | mike-ekim | not this PRI stuff |
| 20:31.32 | mike-ekim | so please excuse my ignorance |
| 20:31.52 | mike-ekim | these cheap sarcastic comments are pretty much a waste of both of our time dont you think |
| 20:32.03 | mike-ekim | im just trying to get some assistance on a problem. |
| 20:33.21 | *** join/#asterisk RoyK (n=roy@ip-191-24-149-91.dialup.ice.no) |
| 20:33.26 | ManxPower | mike-ekim: I have no idea what might fix your problem. |
| 20:33.47 | CrashHD | wow |
| 20:33.50 | mike-ekim | alright |
| 20:33.50 | CrashHD | I hear attitude |
| 20:33.53 | mike-ekim | i appreciate the time |
| 20:34.56 | mike-ekim | i am just pretty sure they are using the zaptel cards with 4 fxo, fxs, and i am sure theyre using libpri |
| 20:35.11 | ManxPower | mike-ekim: A PRI is not FXO/FXS |
| 20:35.16 | mike-ekim | oh |
| 20:35.23 | ManxPower | I think you need to figure out what card you have before you go any further |
| 20:35.24 | Nugget | thank god for that. |
| 20:35.28 | mike-ekim | like i said excyse my ignorance with this |
| 20:35.44 | mike-ekim | well theyre not using a tdm then |
| 20:36.03 | mike-ekim | the errors is stating from zapata.conf |
| 20:36.07 | mike-ekim | not zaptel |
| 20:42.37 | ManxPower | mike-ekim: the paste http://pastebin.ca/930739 shows you have 0 channels configured |
| 20:43.00 | ManxPower | You need to edit your /etc/zaptel.conf then run ztcfg -vvv again |
| 20:43.19 | mike-ekim | whatever |
| 20:43.40 | ManxPower | Poor guy. |
| 20:44.02 | ManxPower | I guess he'll eventually realize I was right. |
| 20:44.09 | Havokmon | That's what counts |
| 20:44.48 | ManxPower | Havokmon: Me being right is not what counts. Getting it fixed is what counts. |
| 20:45.15 | Havokmon | I know that's what you meant, it sure was a long road to there though. |
| 20:45.18 | ManxPower | But I don't complain when someone has to realize I'm right before they can get it fixed. 8-) |
| 20:45.24 | *** part/#asterisk RoyK (n=roy@ip-191-24-149-91.dialup.ice.no) |
| 20:46.15 | Havokmon | I don't think you guys realize how much you rub people the wrong way. Right or not. |
| 20:46.22 | CrashHD | lol manx |
| 20:46.37 | CrashHD | just nod and laugh |
| 20:46.40 | CrashHD | :) |
| 20:46.48 | jameswf | I am always thrown back when someone asks for my help. Asking for my help implies I know something you dont, then they respond to my advice with " No thats not it" Dude if you know so much why are you asking me |
| 20:46.59 | twisted | soooo |
| 20:47.11 | twisted | who has flipped their stack over the iphone sdk? |
| 20:47.19 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 20:47.28 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 20:47.30 | ManxPower | jameswf: that's when they don't get my help anymore |
| 20:48.25 | Havokmon | uhh yeah.. it's a back and forth thing. The only way anyone is going to get to the end is an actual conversation with suggestion, attempt, rebuttal or acceptance. |
| 20:48.26 | jameswf | 275 asterisk users walk in to a bar.... doesnt sound that funny |
| 20:49.14 | [hC] | Wait, wasnt that last astricon? |
| 20:49.37 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 20:49.44 | jameswf | heh astricon should do a series of flash mobs... |
| 20:49.54 | Qwell | jameswf: we basically do |
| 20:50.09 | Qwell | it's not uncommon to walk into a restaurant with 30 people |
| 20:50.14 | Yourname`` | Hi. I have two IPs on a server, but I want Asterisk to work only on one IP and make calls from that IP. Do I use that bindaddr in sip.conf to make it so? |
| 20:50.22 | jameswf | 200 people in walmart chanting wecon need canadien ba |
| 20:50.41 | jameswf | s/wecon/we/ |
| 20:50.48 | jameswf | bacon |
| 20:50.49 | jameswf | bah |
| 20:51.30 | plik | sooo, is having a whole load of different time based rules that are mostly there just in case of future changes likely to impair performance at all? I notice that each appears on the CLI as it searchees for a match |
| 20:52.02 | hmmhesays | wahoo I segfaulted asterisk |
| 20:52.29 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 20:52.29 | ManxPower | plik: yes it will impact performace, I doubt will impact it enough to make any difference to you |
| 20:52.52 | plik | ManxPower: thanks :) |
| 20:54.09 | plik | thats mon-thu, fri, sat & sun each split in to 2 hour segments to cover all evenualities of diverting phones on different shifts then |
| 20:54.16 | *** join/#asterisk dmaust (n=dmaust@rrcs-64-183-21-194.west.biz.rr.com) |
| 20:55.24 | twisted | dink |
| 20:56.02 | *** mode/#asterisk [-b *!*n=jsuter@*.us] by Qwell |
| 20:56.04 | *** mode/#asterisk [+b *!*n=jsuter@*] by Qwell |
| 20:57.32 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:57.42 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 20:59.08 | CrashHD | any have any experience with asterisk's implementation of SLA in conjuction with Aastra phones? |
| 21:03.15 | [TK]D-Fender | CrashHD: * doesn't support SLA. What we have is a largely unworkable hack pretending to be SLA |
| 21:03.25 | CrashHD | hehe |
| 21:04.07 | CrashHD | so not a solution worth workin through? |
| 21:04.20 | jameswf | by astra does an sla thing it works 2% of the time |
| 21:05.16 | CrashHD | based on aastra's bla line setting? or...? |
| 21:05.52 | jameswf | some xml crap someone got bored one day |
| 21:05.59 | CrashHD | link? |
| 21:06.38 | jameswf | I think it was all trial and error |
| 21:06.43 | CrashHD | ahh |
| 21:06.57 | *** join/#asterisk ether0 (n=afarring@apoc.digiport.com) |
| 21:10.43 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 21:11.36 | tzafrir | tip: don't call that Centos-compatible OS "RHEL" |
| 21:11.40 | tzafrir | http://www.redhatmagazine.com/2008/02/04/tips-and-tricks-rhel-ref/ |
| 21:12.49 | *** join/#asterisk ambriento (n=ambrient@200.190.197.66) |
| 21:13.38 | hmmhesays | I wish func_odbc would return more than one row |
| 21:13.42 | hmmhesays | that is a serious drawback |
| 21:14.07 | *** join/#asterisk metfan2007 (n=metfan20@201.103.80.238) |
| 21:15.54 | ZPertee | asterisk won't play my custom wav files but will play the default MOH files. any ideas? |
| 21:16.47 | *** join/#asterisk oej (n=olle@p5485C521.dip.t-dialin.net) |
| 21:16.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:16.51 | [TK]D-Fender | ZPertee: Tell us something useful. |
| 21:17.45 | ZPertee | it says file doesn't exist in any format. but if I do an "ls" command on it it will show up |
| 21:17.48 | tzafrir | ZPertee, sox to the rescue |
| 21:18.16 | tzafrir | ZPertee, you should not include the .wav suffix |
| 21:18.48 | ZPertee | tzafrir: so instead of /folder/welcome.wav do /folder/welcome? right? |
| 21:18.54 | tzafrir | yes |
| 21:19.01 | ZPertee | ok. whats this about sox |
| 21:19.09 | ZPertee | ~bot sox |
| 21:19.36 | ZPertee | bot don't know |
| 21:19.45 | jameswf | jbot: tell ZPertee about sox |
| 21:20.35 | tzafrir | ~sox |
| 21:20.36 | jbot | extra, extra, read all about it, sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 21:21.40 | ZPertee | figured it was me. bot hasn't let me down yet...as long as I know how to use it |
| 21:21.45 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 21:21.55 | *** join/#asterisk SomethingISODD (n=noc@S010600a0d1757bfb.cg.shawcable.net) |
| 21:22.18 | SomethingISODD | hello all quick question is there anyway to get the callers ip address using phpagi scripting? |
| 21:22.48 | *** join/#asterisk supjigator (n=shanebur@64.147.217.2) |
| 21:24.24 | docelmo | yes |
| 21:24.29 | docelmo | well in a round about way |
| 21:24.46 | hmmhesays | you have to use an offset if you need more than one row, that sucks |
| 21:24.54 | docelmo | $agi->get_variable("SIP_HEADER(From)"); |
| 21:26.11 | docelmo | hmmhesays only in the via portion.. I have never seen more than one from |
| 21:26.52 | SomethingISODD | thanks all |
| 21:26.57 | SomethingISODD | eek thanks docelmo |
| 21:27.13 | hmmhesays | docelmo, sorry I was ranting about func_odbc |
| 21:27.14 | docelmo | ya.. all good.. Just look over one of the phpagi developers.. :) |
| 21:27.32 | Juggie | yell @ qwell thats what i do :) |
| 21:27.50 | docelmo | eh? Am I missing parts of conversations? |
| 21:29.08 | *** join/#asterisk oej_ (n=olle@p5485DC78.dip.t-dialin.net) |
| 21:29.48 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 21:29.53 | SomethingISODD | docelmo mind if i ask you one more question |
| 21:30.18 | docelmo | it will cost ya $50 buks.. being the US economy sucks.. I would rather have 50 euro.. :) |
| 21:30.21 | docelmo | whats up? |
| 21:30.47 | SomethingISODD | I am trying toget the phone number or extension... the person is dialing. am i doing it wrong $agi->get_variable("extension"); |
| 21:31.25 | Sci_05 | anyone got a link for setting up sip between 2 asterisk boxes (instead of using IAX)? |
| 21:31.36 | docelmo | yes.. use $agi->request['agi_extension']; |
| 21:31.48 | ice_croft | Sci_05> man, say ~book |
| 21:31.58 | SomethingISODD | oh let me try that thanks |
| 21:32.00 | [TK]D-Fender | Sci_05: lookup "asterisk dual servers" on the WIKI |
| 21:32.11 | Sci_05 | thanks [TK]D-Fender |
| 21:32.49 | SomethingISODD | docelmo for some reason its still not getting the full extension all its getting is the first number |
| 21:33.05 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 21:39.05 | hmmhesays | hmm can you unset a variable in asterisk dialplan so it no longer exists? |
| 21:39.19 | ZPertee | I know nothing about audio but asterisk is telling me that my wav files are the wrong frequency can I use sox to get my files converted to something useful? |
| 21:39.30 | ZPertee | what is asterisk compatible |
| 21:39.45 | hmmhesays | 8khz |
| 21:40.38 | ZPertee | should I use gsm or wav? |
| 21:45.52 | hmmhesays | wav |
| 21:46.03 | hmmhesays | now I don't see any way to destroy a dialplan variable |
| 21:46.10 | hmmhesays | and that is irritating me |
| 21:46.30 | *** join/#asterisk Eagleray (n=erayd@osiris.erayd.net) |
| 21:46.59 | graphx | Does anyone here have experience on the SPA942 Linsys SIP Phones? I need some help |
| 21:47.18 | graphx | Linksys* |
| 21:47.22 | Eagleray | Hi, how to I create a dialplan rule that will always be executed, regardless of the extension? (And then execute other, extension-specific rules later in the context). |
| 21:47.45 | Eagleray | graphx: No, but I do have a PAP2T here - do you know if they are similar? |
| 21:48.12 | graphx | Eagleray: I will look ... hold on |
| 21:48.33 | SomethingISODD | question in agi scripting i can set a variable can i do it from the extension config file? |
| 21:49.49 | graphx | Eagleray: That's what I thought... That's like a router/pap box... for like vonage or lingo... I am referring to an actual phone... such as http://www.ipphone-warehouse.com/Linksys-SPA942-VoIP-SIP-Phone-SPA-942-p/spa942.htm?gclid=COTIp7a7-ZECFQcygQodFA-ppw |
| 21:49.56 | *** join/#asterisk Tommy3 (n=Tommy2@66.0.46.210) |
| 21:50.18 | Eagleray | graphx: I am aware of this, but if the webconfig is similar I may still be of some use |
| 21:50.27 | Eagleray | graphx: where are you running into problems? |
| 21:50.48 | Eagleray | graphx: the PAP2T is an ATA box (not a router) |
| 21:50.57 | ZPertee | how do I turn up the volume in my audio files? |
| 21:51.23 | graphx | I was running in to problems getting my Attended Call Transfer to work across my asterisk. |
| 21:51.37 | graphx | I can do Blind Transfers |
| 21:52.32 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) |
| 21:52.33 | Eagleray | graphx: Hmmmm - is three-way calling enabled on the linksys? |
| 21:52.53 | graphx | I think it is... |
| 21:52.55 | Eagleray | graphx: If it's the same as mine, you need to tell it that a three-way service is available before it will do an attended transfer |
| 21:52.59 | graphx | let me check |
| 21:53.06 | Eagleray | graphx: the option is in the advanced settings for the line |
| 21:53.24 | tld | Any recommendation for a termination provider in Europe? (origination would be nice, but less important) |
| 21:55.36 | graphx | Eagleray: I have two locations on my phone for items like this. The one for services is "Supplementary Services". |
| 21:56.06 | Eagleray | graphx: Hmm. This means the config is different to mine then - unfortunately I cna't be of any help |
| 21:56.09 | Eagleray | graphx: sorry |
| 21:56.23 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 21:56.44 | graphx | Eagleray: No worries... I appreciate the attempt. |
| 22:09.32 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 22:12.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:14.42 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 22:16.57 | flush | yo |
| 22:17.06 | flush | how do you configure asterisk with analog phones |
| 22:17.47 | jsmith | flush: The short answer is: Configure zaptel.conf and zapata.conf |
| 22:17.53 | seanbright | ~tfot |
| 22:17.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:18.03 | seanbright | written by some guy named j. smith |
| 22:18.16 | jsmith | Who's he? |
| 22:18.21 | flush | i have my fxo module set up and fxs too, i set up the echo thing it works, zttool shows "alarms: OK" and i have dial tone when i pick up the phone. however when a call comes in asterisk wont make the phone ring and when i hit a key it immediately gives me the fast busy tone, how can i place a call then |
| 22:18.22 | seanbright | heh |
| 22:18.29 | flush | jsmith this is already done |
| 22:18.42 | flush | seanbright this book i followed |
| 22:18.43 | jsmith | flush: OK, sounds like your contexts are set wrong. |
| 22:18.47 | flush | for the echo thing, extension 500 |
| 22:18.56 | flush | can you gimme sec ill give url of my configs |
| 22:19.04 | jsmith | flush: Look at your zapata.conf file... what is the context set to for your FXO port? |
| 22:19.52 | flush | http://allpr0h.servebeer.com:8181/ |
| 22:20.11 | flush | context=incoming |
| 22:21.08 | Eagleray | How can I create a dialplan rule that will always be executed, regardless of the extension (and then execute other, extension-specific rules later in the context)? |
| 22:21.16 | jsmith | OK... now from the Asterisk CLI type "dialplan show s@incoming" |
| 22:22.06 | flush | k sec |
| 22:22.24 | flush | There is no existence of 'incoming' context |
| 22:22.32 | flush | hrmm |
| 22:23.31 | flush | jsmith how do i create the context if not in extensions.conf ? |
| 22:26.18 | jsmith | flush: Have you typed "dialplan reload" after changing extensions.conf? |
| 22:29.01 | SomethingISODD | could someone tell me what i am doing wrong $results = $agi->request['agi_extension']; $result = $results['data']; all i am getting is a value of number 1 and not the number dialed |
| 22:30.54 | SomethingISODD | anyone plesae |
| 22:36.41 | Tommy3 | ???? I installed my new zapmicro 400p card with an fxs and fso last nite. Using Elastix, auto config did not find fsx and fso, but did find 400p card. (did not connect the power cable yet...is it required for fxo to work too?) |
| 22:38.08 | jameswf | ~clones |
| 22:38.15 | jameswf | ~clone |
| 22:38.16 | jbot | methinks clone is a clone card - i.e. a worthless, unreliable piece of junk. Is all that extra headache really worth the few dollars you're about to save? |
| 22:38.33 | Tommy3 | thanks |
| 22:39.04 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 22:43.46 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 22:44.01 | mchou | wow, I think I finally managed to make my wrtp54g stable |
| 22:44.08 | mchou | what a PITA |
| 22:44.59 | mchou | too many freaking buttons and knobs on this thing |
| 22:46.24 | Eagleray | How can I create a dialplan rule that will always be executed, regardless of the extension (and then execute other, extension-specific rules later in the context)? |
| 22:49.57 | TJNII | Eagleray: What are you trying to do now? |
| 22:50.30 | Eagleray | TJNII: I'm trying to run a System() call to sendmail for every call run through the context |
| 22:50.39 | Eagleray | TJNII: then pass the call on to the normal rules |
| 22:51.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:52.35 | jsmith | Eagleray: Create yourself a new context, let's call it [bob] |
| 22:52.46 | jsmith | Eagleray: Point your incoming calls at the [bob] extension |
| 22:52.51 | jsmith | [bob] |
| 22:52.57 | jsmith | exten _X.,1,System(something) |
| 22:53.14 | jsmith | exten => _X.,n,Goto(some-other-context,${EXTEN},1) |
| 22:53.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:54.03 | Eagleray | jsmith: thanks |
| 22:54.37 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 22:55.03 | jsmith | Eagleray: No problem... just happy to help when I can |
| 22:55.12 | Eagleray | jsmith: :-) |
| 22:55.54 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 22:58.14 | jameswf | i pointed a few calls at bob he got kinda pissed |
| 22:59.41 | Eagleray | jsmith: This isn't really working too well - I have a ton of contexts, and this is meaning I have to create an extra context for every 'real' one just to do logging |
| 22:59.54 | Eagleray | jsmith: is there another way to handle this from inside the context? |
| 23:00.22 | jameswf | jbot: tell Eagleray about ch6 |
| 23:00.39 | jsmith | Eagleray: Not an easy way, no |
| 23:01.21 | Eagleray | jameswf: cheers, I'll take a loog |
| 23:01.27 | Eagleray | jameswf: *look |
| 23:02.56 | *** join/#asterisk riddlebox (n=cts@75-128-170-26.static.stls.mo.charter.com) |
| 23:03.12 | riddlebox | wohoo, gotta leave soon to go get a contract signed on an asterisk system! |
| 23:04.39 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 23:04.55 | [TK]D-Fender | riddlebox, Dirty deeds.... done dirt cheap.... |
| 23:05.43 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.105.245) |
| 23:07.40 | riddlebox | [TK]D-Fender, hehe its a 4 fxo port system with 8 phones |
| 23:08.14 | Mercestes | ebul |
| 23:09.15 | Eagleray | jsmith: sadly CH6 did not help |
| 23:09.45 | Eagleray | jsmith: the 'bob' method is the only way I can think of that will work :( |
| 23:10.52 | *** join/#asterisk lostdave_98 (n=chatzill@ppp118-208-57-198.lns4.bne1.internode.on.net) |
| 23:11.02 | kamisalami | can someone help me with a zaptel build problem on ubuntu 7.10? here is what i did (install zaptel from source of hardy): http://pastebin.ca/930378 and here is the result: http://pastebin.com/d27b38ecb |
| 23:12.26 | lostdave_98 | Hi All, having a bit of a problem setting up a B410p Card. when asterisk attempts a start, I get stack_init: Cannot add lyer 4 to this port. I have previously used mISDN wiht other cards ithout any problems and I have not sween this error before... |
| 23:13.36 | graphx | kamisalami: Do yourself a favor a switch to Fedora Core... I tried my first install on Ubuntu.. and I ran in to multiple problems.. I switched to Fedora Core and had a better experience... |
| 23:13.53 | kamisalami | graphx: can't, company policy |
| 23:14.28 | graphx | kamisalami: They only allow Ubuntu? |
| 23:14.33 | kamisalami | yep |
| 23:14.37 | graphx | ic |
| 23:14.41 | kamisalami | standardized on it |
| 23:14.41 | *** join/#asterisk cleone (i=cleo@41.251.25.126) |
| 23:14.44 | Qwell | you only allow unreleased versions of ubuntu? |
| 23:14.53 | kamisalami | ? |
| 23:14.57 | kamisalami | 7.10 server ? |
| 23:15.01 | kamisalami | unreleased? |
| 23:15.21 | Eagleray | kamisalami: 7.10 is gutsy |
| 23:15.28 | graphx | Is it LTS? |
| 23:15.29 | Eagleray | kamisalami: hardy is unreleased |
| 23:15.36 | Qwell | Eagleray: right |
| 23:15.39 | Qwell | what he said |
| 23:15.41 | kamisalami | i know, just need the zaptel module from it |
| 23:15.43 | Eagleray | graphx: I think hardy is LTS |
| 23:16.08 | kamisalami | contains patches i need |
| 23:16.17 | Eagleray | graphx: although tbh LTS doesn't mean much, they've done a less than stellar job with dapper (6.06 LTS) |
| 23:16.50 | kamisalami | but, can any of you help me with having a look at the compile log? |
| 23:16.51 | graphx | Well... The "LTS" version of Ubuntu receives long-term support. 3 years for desktop versions and 5 years for server versions. |
| 23:17.26 | graphx | Eagleray: Well... The "LTS" version of Ubuntu receives long-term support. 3 years for desktop versions and 5 years for server versions. |
| 23:17.39 | graphx | Eagleray: Ubuntu 6.06 LTS - Supported to 2011 |
| 23:17.59 | *** join/#asterisk nofish2 (i=port@61-69-2-27.netspeed.com.au) |
| 23:18.09 | kamisalami | oh no, i started a distro war :) |
| 23:18.34 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 23:18.37 | Eagleray | graphx: lol yes I know that's what they claim, I just don't think they have done a terribly good job |
| 23:18.50 | graphx | I use different distro's for what ever I deem fit.... |
| 23:19.06 | Eagleray | graphx: imo if you want long term support, you're far better off with Debian, CentOS, or RHEL |
| 23:19.26 | graphx | Eagleray: Have you have to call support and deal with them yet? Just curious... |
| 23:19.55 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:20.24 | Eagleray | graphx: nah I'm not talking about phone support, I'm talking about security patching & backports |
| 23:20.42 | Eagleray | graphx: I have had dapper break on me several times due to shoddy packages |
| 23:21.22 | graphx | Eagleray: I hate to say it, but Ubuntu have gone lightyears what they have done in the amount of time that they have been in it with Ubuntu... |
| 23:21.23 | Eagleray | graphx: at least redhat & the debian people seem committed to their longterm support |
| 23:21.31 | *** join/#asterisk RoyK (n=roy@ip-183-54-149-91.dialup.ice.no) |
| 23:21.37 | Eagleray | graphx: with ubuntu the LTS thing seems more like an afterthought |
| 23:21.57 | Eagleray | graphx: I'm not saying that the ubuntu people haven't done a good job - they have |
| 23:22.04 | *** part/#asterisk RoyK (n=roy@ip-183-54-149-91.dialup.ice.no) |
| 23:22.09 | Mercestes | Gentoo. |
| 23:22.09 | Eagleray | graphx: I just don't believe that LTS is their strong point |
| 23:22.30 | Eagleray | Mercestes: agreed, but I cna't be assed spending all that time compiling stuff. I ran it for a year, then ditched it |
| 23:22.32 | graphx | Eagleray: I would agree they have room to grow in support... but who doesn't.... |
| 23:22.53 | graphx | Eagleray: Lets put it this way... at least it isn't like VISTA!!! LOL!!!! |
| 23:22.56 | *** join/#asterisk seaq (n=seaq@190.25.251.15) |
| 23:23.03 | Mercestes | Eagleray, lol. Yea, I just downloaded a completed VM image for gentoo....I'm such a slacker. :( |
| 23:23.30 | kamisalami | but ehm.... can someone help me with a zaptel build problem on ubuntu 7.10? here is what i did (install zaptel from source of hardy): http://pastebin.ca/930378 and here is the result: http://pastebin.com/d27b38ecb |
| 23:23.36 | Eagleray | graphx: word |
| 23:23.36 | mchou | does vista provide ANY support?? :) |
| 23:23.39 | Mercestes | I liked Sabayon.....for awhile....I think my last build was Kubuntu. |
| 23:23.48 | graphx | Mercestes: Congrats... and my condolences in too.... :) |
| 23:23.59 | Mercestes | mchou: New, up to date blue screen. |
| 23:24.14 | Mercestes | mchou: And my putty windows into linux have never looked better. |
| 23:24.19 | mchou | Mercestes: lol. that's useful for debugging |
| 23:24.19 | Eagleray | Mercestes: same old white text though :po |
| 23:24.34 | Mercestes | mchou:: And I have this sidebar with all my system stats on it so I can see how much ram and processor my mouse pointer takes up. |
| 23:24.41 | Eagleray | damn windows needs a decent shell though |
| 23:24.48 | mchou | Mercestes: haha |
| 23:25.01 | mchou | Mercestes: in linux or vista? |
| 23:25.05 | Mercestes | Vista. |
| 23:25.22 | mchou | Mercestes: you can have the same junk in linux too :) |
| 23:25.38 | Eagleray | Mercestes: try gkrellm :-) |
| 23:25.44 | Eagleray | Mercestes: very useful |
| 23:25.47 | Mercestes | Yea, for that I use kubuntu or sabayon. I'm too lazy to compile all that into gentoo |
| 23:25.51 | graphx | Since I got my COD4 running in wine.. I don't really use Windblows unless it is in VBox, or VMware on my linux box... |
| 23:26.10 | Eagleray | Mercestes: well if you're not compiling stuff that pretty much defeats the whole point of gentoo.... |
| 23:26.30 | mchou | graphx: how difficuly is that to do? (COD4 in wine) |
| 23:26.35 | graphx | the only reason I use linux is for my customers... or for games I don't get to load in wine... |
| 23:26.40 | mchou | difficult* |
| 23:26.57 | Mercestes | Eagleray, I like gentoo because it doesn't add support for things I don't specify. So for * I use gentoo, because....I don't need beryl on my * box. For my PC, I use something more fun. |
| 23:27.14 | Mercestes | ....something with....I dunno...graphics. |
| 23:27.28 | Eagleray | Mercestes: I use debian, with no packages installed by default - just the bare minimum needed to boot |
| 23:27.33 | Eagleray | Mercestes: then I add what I want to use |
| 23:27.37 | graphx | mchou: Not difficult... just have too remove wine and recompile it... adding in a debug for DX and add a dll to confg.. that;s it... |
| 23:27.37 | tzafrir | kamisalami, hmmm... what is the output of: tar tjf /usr/src/zaptel.tar.bz2 |
| 23:28.20 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 23:29.14 | graphx | mchou: What disto you using? |
| 23:29.37 | mchou | I realized how lousy windows was when I started building my PVR |
| 23:29.56 | mchou | ditched windows and never looked back |
| 23:30.03 | graphx | mchou: What is your Linux Distro? |
| 23:30.25 | mchou | graphx: why do you want to know? :) |
| 23:30.42 | kamisalami | tzafrir: http://pastebin.ca/931114 |
| 23:30.57 | graphx | to see if I can send you a link to compiling it on your distro |
| 23:31.20 | graphx | If you have Ubuntu... here is the one for that distro - http://www.fsckin.com/2008/02/21/how-to-run-call-of-duty-4-cod4-modern-combat-in-linux/ |
| 23:31.22 | mchou | graphx: that's ok. rarely play games anymore. but thx thou |
| 23:31.32 | *** join/#asterisk Igbothom_III (n=Hilton@203.53.147.18) |
| 23:31.52 | mchou | only game I play these days is openttd :) |
| 23:32.04 | mchou | but there are some shortcomings |
| 23:32.43 | Yourname`` | Hello. I was wondering about cascading queues. Say there are 5 agents logged into a queue, the 6call that comes into the queue obviously has to hold till an agent is free. I was wondering if I can make asterisk detect that the 6th call won't have a place on the queue and so it "overflows" the call to another external DID (which is a queue, excetp setup on a different system than asterisk) Is... |
| 23:32.45 | Yourname`` | ...that possible? |
| 23:32.48 | mchou | openttd is not very smart about particular sources and sinks |
| 23:33.17 | mchou | so it's easy to "beat the system" |
| 23:33.46 | tzafrir | kamisalami, sorry, just can't figure it |
| 23:34.22 | phix | mchou: openttd? what is that? |
| 23:34.50 | mchou | phix: http://www.openttd.org/ |
| 23:34.58 | mchou | phix: very addictive |
| 23:35.02 | SomethingISODD | could someone tell me what i am doing wrong $results = $agi->request['agi_extension']; $result = $results['data']; all i am getting is a value of number 1 and not the number dialed |
| 23:35.03 | lostdave_98 | No Clues on what I am even looking for with my B410 Prob? |
| 23:35.05 | Eagleray | Yourname``: You can just use a Dial() as a lower-priority action |
| 23:35.07 | phix | moemoe: The only game I play atm is TF2 :) |
| 23:35.23 | mchou | phix: be prepared to spend weeks :) |
| 23:35.37 | phix | nice |
| 23:35.42 | Yourname`` | Eagleray: Or retrydial? |
| 23:35.43 | phix | I should look for a job though |
| 23:35.45 | phix | oh well :) |
| 23:35.49 | Eagleray | Yourname``: for example s,1,Dial(SIP/phone1&SIP/phone2&SIP/phone3&SIP/phone4&SIP/phone5) |
| 23:36.09 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 23:36.14 | Eagleray | Yourname``: then s,n,Dial(SIP/fallbacktrunk) |
| 23:36.24 | phix | s,1,Dial(SIP/phone1&SIP/phone2&Zap/1) |
| 23:36.32 | Eagleray | if all the phones in the first line are busy, it'll dial the trunk instead |
| 23:36.56 | Yourname`` | Eagleray: Actually, I don't dial all phones, I dial to a queue() |
| 23:37.02 | kamisalami | tzafrir: no problem, neither can i |
| 23:37.03 | phix | Eagleray: I like to use dial status var :) |
| 23:37.13 | kamisalami | tzafrir: thnx for trying |
| 23:37.22 | Eagleray | Yourname``: aah - I'm no help then sorry, I'm pretty new to this myself |
| 23:37.41 | Yourname`` | Eagleray: Thanks, you tried. :) |
| 23:37.51 | Yourname`` | Eagleray: When I was new, I wasn't as good as you ;) |
| 23:38.30 | Eagleray | Yourname``: hahah |
| 23:39.06 | Mercestes | Yourname``, Could set a queue timeout to a really low value and do the dial statement like Eagleray suggests. |
| 23:39.40 | Mercestes | Yourname``, that way it will drop out of the queue and dial a new number. Or you can setup priorities in the queue and setup members. I don't think you can setup external numbers as members though. |
| 23:39.41 | angryuser | offtopic: Is there anyone here who got bluray burner ? |
| 23:40.18 | *** join/#asterisk tales09 (n=talesdem@190.51.178.2) |
| 23:40.22 | Yourname`` | Mercestes: That's the thing. I want to cascade an Asterisk queue with an external DID. |
| 23:40.24 | tales09 | hello |
| 23:41.00 | Mercestes | Yourname``, Well unless there was some fancy way to make Asterisk treat the external link as a member (Member => Zap/g1/blah maybe) then you'll probably have to use a timeout to drop out of the queue. |
| 23:41.18 | Mercestes | but honestly, that goes a little beyond what a queue is designed to do. |
| 23:41.38 | Yourname`` | Mercestes: But then you're "queuefying" an external link. That, for some reason, doesn't look too good, lol |
| 23:41.43 | Mercestes | It really is setup to allow 4 people to handle 6 calls. If you want "no hold time" your not really dealing with a queue anymoer. |
| 23:42.01 | Yourname`` | Mercestes: In that case you're dealing more with risk management ;) |
| 23:42.18 | tales09 | Hello? |
| 23:42.21 | Mercestes | That too. Try a queue timeout then and make your next priority a dial statement. |
| 23:42.26 | Yourname`` | tales09: HI! |
| 23:42.31 | Mercestes | tales09: YOu have wasted 11 keystrokes. |
| 23:42.37 | Mercestes | tales09, and forced someone else to waste 3. |
| 23:42.50 | Yourname`` | tales09: He's right. By now you could have typed a question, and maybe even gotten an answer. |
| 23:42.56 | Mercestes | tales09, 13 including carrige returns. |
| 23:43.21 | Yourname`` | The queue timeout sounds plausible, Mercestes! Thanks! |
| 23:43.25 | Yourname`` | I guess I can try that one out. |
| 23:43.31 | Eagleray | Mercestes: unless you have tab-completion of usernames :p |
| 23:44.20 | tales09 | Hello, i want to know if someone can help me to configure Aterisk 1.4 with BroadVoice? |
| 23:44.42 | *** join/#asterisk SpaceBass (n=SP@pool-72-84-196-212.rcmdva.fios.verizon.net) |
| 23:44.49 | Mercestes | Eagleray, I wasn't counting usernames. ;) |
| 23:44.54 | SpaceBass | hey folks |
| 23:45.24 | SpaceBass | Mercestes, I think irssi has that ! |
| 23:45.29 | tales09 | Hello, i want to know if someone can help me to configure Aterisk 1.4 with BroadVoice? |
| 23:45.34 | Mercestes | tales09, I haven't had any specific experience with broadvoice. Have you googled asterisk broadvoice? Are you running into any particular problems. |
| 23:45.36 | Yourname`` | tales09: http://www.broadvoice.com/support_install_asterisk.html -> First result in google. http://tinyurl.com/2u3vmv |
| 23:45.47 | Yourname`` | SpaceBass: And BitchX has a script for it too! |
| 23:45.57 | SpaceBass | Yourname``, really? cool! |
| 23:46.10 | tales09 | Yes, but i dont know where is Sip.conf file |
| 23:46.12 | SpaceBass | tales09, I'm using BV...but with freepbx |
| 23:46.34 | tales09 | I use asterisk + freepbx |
| 23:46.42 | slima | tales09: find / -name sip.conf |
| 23:46.44 | Mercestes | ~book |
| 23:46.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:46.47 | Mercestes | ~freepbx |
| 23:46.47 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:47.11 | tales09 | I use asterisk with freepbx spacebass |
| 23:47.14 | Mercestes | tales09, must read if you are new to asterisk. |
| 23:47.54 | tales09 | I follow a install guide and some pages into one wiki. Sorry if im lossing your time. |
| 23:47.56 | JT | tales09: freepbx is NOT supported here |
| 23:48.03 | JT | freepbx always uses asterisk |
| 23:48.10 | JT | it is not asterisk to support, though |
| 23:48.29 | SpaceBass | tales09, I can pvt you with my setup...but dont have much time to help otherwise |
| 23:48.34 | graphx | uses is a interesting term to use ... lol |
| 23:49.01 | Mercestes | I would have said "Rapes" but ....uses works too. |
| 23:49.27 | graphx | heheheheh |
| 23:49.39 | tales09 | You dont have some guide or if i follow the guide into broandvoice that going to be sufficient? |
| 23:49.40 | Mercestes | I would like Freepbx alot more if I could get all the addons Freepbx uses to work without Freepbx. Like that HUDlite deal. |
| 23:50.05 | SpaceBass | asterisk@home is what got me started....and forced me to learn the conf syntax .... and for my home use freepbx has evolved nicely, but I'm glad I bothered to learn the conf stuff too |
| 23:50.20 | SpaceBass | Mercestes, thats just Trixbox ... which, IMHO took a wrong turn |
| 23:50.26 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 23:50.39 | Mercestes | Aye |
| 23:51.04 | SpaceBass | tales09, I do not...but if I could remember Kerry's old blog...he had a guide that would still work |
| 23:51.25 | jameswf | dial4free guy on list got owned:)) spammers |
| 23:52.13 | tales09 | Ok if you can remember it going to be great =) |
| 23:52.40 | graphx | Man.. I am going to have to rewrite my extensions.conf..... GRRRRRR!!!! |
| 23:53.44 | Mercestes | Maybe this wil help: http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
| 23:55.50 | Mercestes | So it's all cold outside...and raining....storming really..... |
| 23:56.12 | graphx | What type of Asterisk systems do people have here? |
| 23:56.17 | Mercestes | and there is this young hispanic girl outside crying and she was all sprawled out on the steps adn stuff and the only phrase I could make out was "I don't know where to go." so I went outside to ask if everything was alright... |
| 23:56.21 | graphx | just curious of course... |
| 23:56.21 | Mercestes | and I *still* didn't get laid.... |
| 23:56.23 | Mercestes | I've hit a new low. |
| 23:56.26 | Eagleray | graphx: Asterisk 1.4 on a Debian Lenny VPS |
| 23:56.42 | Eagleray | graphx: SIP trunk provider with POTS termination |
| 23:56.55 | graphx | Eagleray: How many phones? |
| 23:57.10 | graphx | and what type.. |
| 23:57.18 | Mercestes | graphx, I don't do much asterisk anymore outside of contract stuff...and that's a range of asterisk 1.2 to 1.4 on several different distros..mostly Debian now a days. |
| 23:57.21 | Eagleray | graphx: 5 at the mo, but with the intention of adding another trunk and another three phones |
| 23:57.32 | Mercestes | I do video conferencing now. |
| 23:57.40 | flush | jsmith-away No such command 'dialplan' (type 'help' for help) |
| 23:57.45 | Mercestes | ...which reminds me....I need to try out that h323 stuff in asterisk....>.> |
| 23:57.49 | Eagleray | graphx: combination of linksys ATA, grandstream SIP phones & x-lite |
| 23:57.58 | Mercestes | flush try core show dialplan |
| 23:58.18 | Mercestes | ...or core dialplan show.... |
| 23:58.29 | Mercestes | .....or ...core dialplan REVEAL YOURSELF TO ME! |
| 23:58.34 | flush | No such command 'core' (type 'help' for help) |
| 23:58.35 | Mercestes | something like that. |
| 23:58.52 | Mercestes | flush, Try a cat /proc/cpuinfo and see what that says. |
| 23:58.54 | Eagleray | Mercestes: 'dialplan show' |
| 23:58.55 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 23:58.56 | flush | asterisk -r |
| 23:59.01 | Eagleray | Mercestes: dialplan isn't part of core |
| 23:59.05 | graphx | I see the force is strong with this one.... |
| 23:59.11 | Mercestes | graphx, lol |
| 23:59.20 | Mercestes | flush, yea, that was going to be my next suggestion. |
| 23:59.24 | graphx | :) |
| 23:59.37 | Eagleray | Mercestes: ... it's an old jedi mind trick.... |
| 23:59.40 | Mercestes | Eagleray, Oh...yea, I can't figure out the new syntax..I just randomly core everything. |
| 23:59.41 | *** join/#asterisk actros1840 (n=A@chello087207097013.chello.pl) |