| 00:00.30 | mick_linux | CunningPike, although to make comments in that nature is a bit in bad taste for this time |
| 00:01.07 | CunningPike | Also way OT for here |
| 00:01.20 | mick_linux | yes |
| 00:01.49 | wunderkin | blowing up canada |
| 00:01.58 | mick_linux | !? |
| 00:02.55 | mick_linux | though if someone from Canada would like to tell me what their thoughts on Canada are... i'd welcome a /msg -- as I'd like to hear a few things from a Canadian point of view |
| 00:03.40 | nettie | Hello, I'm having some issues with one of my voip carrier. Sometimes I dont receive calls anymore so I check the registrations with sip show registry and all the time I see a Request Sent message. The worse part is that if I forget to check time to time it just hangs there. If I issue a simple sip reload the problem disappear, it finally get registered. I'm also getting some "Sep 11 11:53:38 WARNING[9036] chan_sip.c: Got 200 OK on REGISTER t |
| 00:03.51 | nettie | related to the first problem. |
| 00:04.37 | nettie | Anyone know what could cause the problem please? |
| 00:05.05 | [hC] | Okay, got an image up, now just need to figure out how to refresh it |
| 00:13.47 | *** join/#asterisk Altair256 (n=Altair25@68-171-115-30.atlaga.adelphia.net) |
| 00:16.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:18.03 | *** join/#asterisk janeNarak (n=jane@2001:3c8:c103:a001:1995:2d2f:2503:225e) |
| 00:18.05 | *** part/#asterisk EzWayz (n=ez@c207.134.228-16.clta.globetrotter.net) |
| 00:26.15 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:27.52 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 00:29.15 | *** join/#asterisk janeNarak (n=jane@2001:3c8:c103:a001:1995:2d2f:2503:225e) |
| 00:31.23 | *** join/#asterisk oej (n=oej@12.38.214.2) |
| 00:40.51 | [hC] | Ive got refreshing video working on my 7970 |
| 00:41.05 | [hC] | It crashed once in the middle of a refresh, but so far so good on round 2. |
| 00:44.28 | *** join/#asterisk mtmachen (n=chatzill@adsl-068-209-087-005.sip.bhm.bellsouth.net) |
| 00:45.09 | mtmachen | how can I share one mailbox with two extensions? |
| 00:49.04 | docelmo | make mailbox= the same for both of them |
| 00:49.06 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:50.11 | mtmachen | that's what I thought but the 2nd extension just rings without forwarding to voicemail. does that sound like a different problem? |
| 00:52.06 | *** join/#asterisk N9URK (n=icechat5@cpe-065-184-157-227.ec.res.rr.com) |
| 00:54.50 | N9URK | hi all, I am trying to install *. What command do I need to issue so that it will build all the dependencies? |
| 00:55.54 | *** join/#asterisk jaike (i=jaike@58.69.49.24) |
| 00:55.57 | hacked`` | guys, how come when i go to freepbx, add a trunk, then save it, and then i go and do a sip show registry, it doesnt show the trunk i added |
| 00:57.38 | mtmachen | i double checked and made sure mailbox=1st extension under 2nd extension and the 2nd extension rings indefinitely. any ideas? |
| 00:58.11 | N9URK | what does "termcap support not found" mean? |
| 00:58.41 | mtmachen | that didn't make sense. let me try again. I want two extensions to share the same mailbox i have ext. 100 & 101 in sip.conf |
| 00:58.57 | N9URK | ok, I need libncurses |
| 00:59.27 | mtmachen | Both have mailbox =100. 100 rings to voicemail after a few rings. 101 rings indefinitely |
| 01:02.00 | Qwell | mtmachen: It doesn't work like that |
| 01:02.12 | Qwell | It hits voicemail, because your dialplan tells it to |
| 01:09.50 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 01:12.13 | Qwell | file is currently in the air - thus, you cannot fall on him |
| 01:12.28 | brimstone | i could, if i was ABOVE HIM! (which i am, mostly) |
| 01:12.35 | Qwell | umm....okay |
| 01:13.09 | Qwell | mmm... |
| 01:13.10 | Qwell | chocolate |
| 01:13.17 | brimstone | and lasgana! |
| 01:13.24 | Qwell | chocolate lasagna |
| 01:13.30 | linagee | and plone! mmm... plone |
| 01:14.37 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 01:15.40 | niter3 | Does anyone have an idea how I could get a dial tone working after a user presses 9 so they knwo to dial out? |
| 01:16.09 | Qwell | niter3: tell your phone to give you a second dialtone |
| 01:16.11 | Qwell | What phone is it? |
| 01:16.21 | niter3 | it's a linksys adapter |
| 01:16.31 | Qwell | So, fix the dialplan on the phone |
| 01:17.21 | niter3 | So you have to do it per phone...... |
| 01:17.25 | niter3 | hrm.... |
| 01:18.56 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 01:19.10 | BlepsoaF | anyone have any experience with a 7941 Cisco phone |
| 01:19.11 | ariel_ | niter3, there is a command that you can use in the extensions.conf for just that. |
| 01:19.15 | oej | ~seen russellb |
| 01:19.31 | jbot | russellb <n=russell@asterisk/developer-and-stable-maintainer/drumkilla> was last seen on IRC in channel #asterisk, 3d 7h 49m 12s ago, saying: 'i'll do it for 1 BILLION DOLLARZ !'. |
| 01:19.34 | Qwell | russellb was last seen 3 days, 6 hours, 42 minutes, 12 seconds ago |
| 01:19.42 | niter3 | ariel_: That's what I'm hoping. Do you have any idea waht that commandi s? |
| 01:19.44 | Qwell | wow, I was pretty close |
| 01:19.46 | ariel_ | niter3, look at the /usr/src/asterisk/configs/extensions.conf.sample it's one of the first few items. |
| 01:19.47 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 01:21.22 | ariel_ | niter3, ignorepat => 9 |
| 01:21.53 | ariel_ | niter3, but really why do you need to use a 9 when you can do great stuff without it and doing pattern matching. |
| 01:22.40 | Qwell | ariel_: ignorepat is zap only (mostly) |
| 01:22.57 | niter3 | Becuase people are a custome to listening for a dial tone before proceeding to dial out. |
| 01:23.58 | ariel_ | Qwell, yes but I don't ever have any need to use 9 to dial out any more. and besides most sip phones don't care only zap are the ones that have the 9 issue he is speaking about. |
| 01:24.33 | ariel_ | sip phones wait till you put in all the digits before sending it out to the asterisk box. |
| 01:24.46 | `Tingles` | anyone have any idea... Sep 12 03:19:09 WARNING[22569]: app_voicemail.c:4988 vm_authenticate: Couldn't read username |
| 01:25.02 | `Tingles` | as everything looks fine... and this is for all usernames in the file... |
| 01:25.03 | ariel_ | username is incorrect or mispelled |
| 01:25.06 | `Tingles` | err. mailboxes |
| 01:25.15 | ariel_ | wrong context is setup |
| 01:25.34 | `Tingles` | that is correct aswell.. in teh voicemail.conf file.. |
| 01:25.48 | Qwell | bad dtmfmode |
| 01:25.56 | ariel_ | that is my next guess |
| 01:26.17 | `Tingles` | dtmfmode? sorry please explain |
| 01:26.41 | jaike | 1.2.12 rocks. no crashes nor deadlocks the whole day |
| 01:26.48 | niter3 | ariel_: hrm.... I've added that ignorepat => 9 but i still loose a dial tone after dialing 9. My guess is that it's related to my adapter now |
| 01:26.50 | ariel_ | dtmfmode is what sends the tones for the digits if asterisk can't hear them then it can take the digits your sending |
| 01:27.02 | ariel_ | niter3, what adapter? |
| 01:27.09 | niter3 | pap2 linksys |
| 01:27.13 | `Tingles` | how or what do i do to correct that? |
| 01:27.18 | `Tingles` | or verify it is correct.. |
| 01:27.20 | ariel_ | niter3, the pap2 has it's own dial plan |
| 01:27.33 | Qwell | `Tingles`: set the dtmfmode to be the same as your device is using |
| 01:28.04 | JT | Qwell: are you confusing niter3's problem with `Tingles`'s? |
| 01:28.05 | ariel_ | niter3, with a pap2 just dial the number and then press the # key to send the call out. |
| 01:28.32 | Qwell | JT: of course not |
| 01:28.48 | JT | simultaneous dtmfmode problems eh |
| 01:29.33 | niter3 | ariel_: So this is depenedent on the adapter... uh aok. |
| 01:29.34 | `Tingles` | umm.. not to be the special child in the room but.. i don`t remeber anywhere seeing the dtmfmode or don`t remeber... |
| 01:29.43 | Qwell | sip.conf |
| 01:30.52 | ariel_ | niter3, in your case yes |
| 01:35.29 | *** join/#asterisk savant42 (i=terr0r@ip68-101-149-70.sd.sd.cox.net) |
| 01:35.55 | savant42 | good evening gang |
| 01:36.19 | `Tingles` | looks like voicemail is looking for the username in the wrong context.. how do i fix that... |
| 01:36.19 | `Tingles` | <PROTECTED> |
| 01:36.33 | `Tingles` | context should be something else |
| 01:37.09 | savant42 | I'm having weird issues with Auto-Dial out. My call file *seems* ok, and it works...sometimes. Othertimes I get an error message. http://pastebin.ca/167498 |
| 01:37.52 | ariel_ | `Tingles`, what context did you setup for your voicemail? |
| 01:37.54 | brimstone | `Tingles`, add "@office2" or something to VoiceMailMain() when you call it |
| 01:38.05 | `Tingles` | this is what i have to check my voicemail in my extensions.conf |
| 01:38.06 | `Tingles` | exten => 500,1,VoiceMailMain(3) |
| 01:39.05 | Qwell | brimstone: Or something? :p |
| 01:39.50 | `Tingles` | perfect.. thanks guys...i thought it would pickup the context from the section of the script it is in but i guess not.. |
| 01:39.56 | Qwell | brimstone: office2 is for nubs |
| 01:40.06 | brimstone | i guess "put an @ before the context name and add it to the parameters of the voicemailmain app" would be a better responce |
| 01:40.08 | Qwell | That's why we're in office3 :D |
| 01:40.12 | brimstone | aww |
| 01:40.33 | `Tingles` | lol |
| 01:40.38 | Qwell | moving to office2? |
| 01:40.45 | Qwell | with... |
| 01:40.50 | brimstone | no, out of office2, back to office1 :/ |
| 01:41.00 | brimstone | oh! pick on russellb! |
| 01:41.04 | brimstone | he's not here to defend himself :P |
| 01:41.05 | Qwell | yes |
| 01:41.13 | Qwell | office2 == file+russellb virtual office |
| 01:41.28 | brimstone | and he keeps throwing things at me when i'm on the phone |
| 01:41.39 | *** join/#asterisk Skarmeth (n=Skarmeth@201009027155.user.veloxzone.com.br) |
| 01:41.51 | brimstone | some paper balls here and there, a rubber ball now and there is ok, but a trout is not |
| 01:42.16 | Qwell | brimstone: When I move, we need to setup like...a tube system |
| 01:42.19 | savant42 | hey, trout means "I love you" |
| 01:42.39 | brimstone | so you can get internet where you are Qwell? |
| 01:42.53 | Qwell | I have no idea |
| 01:43.01 | Qwell | brimstone: What's the cable co out here? |
| 01:43.10 | *** join/#asterisk Brijn (n=Bas@S0106004063c0fa1f.vn.shawcable.net) |
| 01:43.18 | Brijn | Hi all |
| 01:43.19 | brimstone | comcast or knology, depends on where you are, could even been charter |
| 01:43.43 | Qwell | cool |
| 01:44.07 | savant42 | So what's the courteous time limit to wait before reasking a question if no response was given? |
| 01:44.08 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 01:44.24 | savant42 | I'm having weird issues with Auto-Dial out. My call file *seems* ok, and it works...sometimes. Othertimes I get an error message. http://pastebin.ca/167498 |
| 01:44.40 | brimstone | savant42, i have no idea what error 0 is |
| 01:44.59 | brimstone | i would think that if it works sometimes, it's something your provider is doing? |
| 01:45.23 | savant42 | yeah, error 0 is delightfully descriptive |
| 01:46.23 | Brijn | Did I read the specs correctly, if I think the IP SoundPoint 650 (wideband) phone is not going to be very useful with * (just G.722 codec?)?? |
| 01:48.05 | savant42 | What's weird about my error is that the call terminates to my handset and as soon as I say "hello" (or wait a second or two) the call borks out and gives me "error 0" |
| 01:49.22 | *** part/#asterisk jaike (i=jaike@58.69.49.24) |
| 01:49.49 | wunderkin | error 42! |
| 01:50.26 | savant42 | wunderkin: exactly. I have the answer, but what is the question? |
| 01:50.43 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:50.53 | wunderkin | umm how does the song go |
| 01:51.07 | savant42 | so long, and thanks for all the phish? |
| 01:52.40 | savant42 | I don't mind the phish, helps us catch the skiddies :) |
| 02:01.20 | *** join/#asterisk Katty (n=Angela@dialup-4.244.180.234.Dial1.StLouis1.Level3.net) |
| 02:02.33 | Katty | evening. |
| 02:04.08 | savant42 | ok, well I'm off to battle the weasels in the phonesystem |
| 02:04.10 | savant42 | goodnight, all! |
| 02:10.27 | *** join/#asterisk tud (n=tud@c-24-118-177-83.hsd1.mn.comcast.net) |
| 02:13.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 02:15.11 | *** join/#asterisk annonimous (n=annonimo@189.136.62.17) |
| 02:15.14 | annonimous | hello |
| 02:15.37 | annonimous | tdm01b and a wildcard can interoperate in the same box? |
| 02:16.11 | mog | <PROTECTED> |
| 02:16.40 | annonimous | mog, how can i do that? |
| 02:17.00 | mog | what do you want info on |
| 02:17.01 | annonimous | cause when i want to configure my second card (the clone) the tdm gets unconfigured status and viceversa ='( |
| 02:17.43 | mog | well one i would reccomend not to use clones ^_^, but anyways if you look at /proc/zap/1 and 2 you will see which one is loaded first |
| 02:17.43 | annonimous | i donw know if i need to put anything into the zaptel or something like that ? |
| 02:17.49 | mog | and load them in order |
| 02:17.56 | mog | the channels stack on top of each other |
| 02:18.01 | annonimous | ah i see |
| 02:18.14 | mog | including empty ones |
| 02:18.28 | mog | so your tdm01b takes up 4 slots |
| 02:18.28 | annonimous | and i need to input something into my zapata.conf? |
| 02:18.35 | mog | and zaptel |
| 02:19.14 | annonimous | ok |
| 02:19.24 | annonimous | let me try |
| 02:21.01 | annonimous | the tdm400p its taken as unconfigured |
| 02:21.16 | annonimous | and the clone its in red alarm? |
| 02:22.07 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:24.03 | JT | anyone aware of a way to increase the volume of tones played with PlayTones()? |
| 02:24.16 | annonimous | lol |
| 02:24.25 | annonimous | was easy =( (blushes) |
| 02:28.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:28.54 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 02:31.31 | `Tingles` | whats the virtual zaptel program..... ztunnel or something liek that..? |
| 02:36.10 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 02:37.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:37.11 | fafnir | zshiznizzleforizzle |
| 02:38.59 | [TK]D-Fender | Hey all, are there any known problems with SIP presence in 1.2.12? |
| 02:39.09 | [TK]D-Fender | gota pile of phones that just don't seemt o be picking it up right |
| 02:47.17 | hmmhesays | what are you using for presence? |
| 02:52.57 | [TK]D-Fender | hmmhesays : http://pastebin.ca/167562 |
| 02:53.41 | [TK]D-Fender | hmmhesays : Look at that... F'n crazy |
| 02:54.00 | [TK]D-Fender | I'm still on that call! |
| 02:54.10 | Damin | Hey.. can everyone go DIGG this for me? http://digg.com/linux_unix/Ohio_Linux_Fest_2006 |
| 02:54.22 | *** join/#asterisk |TJH| (n=tj@66-169-252-218.dhcp.mdfd.or.charter.com) |
| 02:58.18 | *** part/#asterisk |TJH| (n=tj@66-169-252-218.dhcp.mdfd.or.charter.com) |
| 03:08.29 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:15.37 | Damin | Anyone alive? |
| 03:21.43 | CunningPike | Barely |
| 03:32.00 | mishehu | Damin: seems like everywhere on this network died. |
| 03:32.12 | mishehu | well except for the mythtv-users channel |
| 03:32.44 | CunningPike | It'll take more than that to kill me..... |
| 03:33.26 | mishehu | heh |
| 03:33.39 | CunningPike | ~lart mishehu |
| 03:33.46 | mishehu | muuuuuuuu! |
| 03:33.52 | CunningPike | FFS, jbot, is that the best you can do? |
| 03:33.58 | *** join/#asterisk anthonyl (n=rachel@c-67-167-214-149.hsd1.il.comcast.net) |
| 03:33.59 | CunningPike | Mooing? |
| 03:34.19 | mishehu | CunningPike: it must know that I'm within 100 miles of dairy country |
| 03:34.23 | docelmo | mishehu trolling I see |
| 03:34.23 | CunningPike | lolk |
| 03:34.35 | mishehu | docelmo: I learned from the best... you! heh |
| 03:34.42 | docelmo | wtf ever |
| 03:34.45 | docelmo | oops.. strike 1 |
| 03:34.49 | CunningPike | mishehu: Wisconsin? |
| 03:35.11 | mishehu | CunningPike: unless there's another dairy state in the USA that I don't know about... |
| 03:35.14 | docelmo | So which of the Digium guys are at VON? |
| 03:35.22 | mishehu | which is possible, I never claimed to know everything |
| 03:35.28 | file | docelmo: yes. |
| 03:35.30 | file | :D |
| 03:35.41 | CunningPike | mishehu: I don't think Wisconsin has the monopoly..... |
| 03:36.02 | wunderkin | monopoly!! yey! |
| 03:36.06 | mishehu | CunningPike: nah, I'm sure others have some dairy, but as far as I know Wisconsin is the only one that claims to be the dairy state. |
| 03:36.28 | mishehu | hmm... to update my polycom firmware or not... |
| 03:36.41 | mishehu | that is the question at hand... |
| 03:39.40 | Qwell | file: finally there? |
| 03:39.48 | file | Qwell: yes |
| 03:39.56 | Qwell | file: hote?l |
| 03:40.01 | Qwell | umm...that was weird |
| 03:40.02 | file | aye |
| 03:40.06 | Qwell | k |
| 03:40.11 | file | kpfleming is over there |
| 03:40.25 | file | he is deep in thought |
| 03:40.44 | [TK]D-Fender | file : ... |
| 03:40.50 | file | [TK]D-Fender: ! ! ! |
| 03:40.51 | mishehu | a baseball bat usually is more effective |
| 03:40.52 | [TK]D-Fender | file : http://pastebin.ca/167562 |
| 03:40.53 | mishehu | heh. |
| 03:41.13 | file | I just got in here and you want me to look at something already? |
| 03:41.13 | Qwell | mishehu: baseball bat + boss == bad idea :P |
| 03:41.15 | file | 'tsk 'tsk |
| 03:41.31 | file | [TK]D-Fender: crank up core debug |
| 03:41.38 | Qwell | file: good boy! |
| 03:41.43 | mishehu | Qwell: ah, just wear a mask and make sure to hit so hard he sees stars (and thus won't be able to identify you) |
| 03:41.45 | [TK]D-Fender | file : Its lying through its teeth! |
| 03:41.45 | file | it'll be much more verbose about the device state stuff |
| 03:41.46 | mishehu | heh. |
| 03:41.57 | Qwell | mishehu: I'm easy to id :D |
| 03:42.06 | file | Qwell: you need to hurry up and get here |
| 03:42.11 | Qwell | file: yes, I'm working on it |
| 03:42.18 | mishehu | Qwell: well hmmmm.... convince file to be your pinch hitter |
| 03:42.30 | Qwell | mishehu: That sounds a bit disturbing |
| 03:43.27 | file | I should... go to sleep |
| 03:43.32 | mishehu | it's the internet, aren't we all disturbed just by the fact we are in front of computers? |
| 03:44.18 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:48.59 | mishehu | *sigh* I hate having to do reboots on remote servers |
| 03:49.53 | mishehu | Qwell: What Me Version? |
| 03:49.54 | mishehu | heh |
| 03:50.08 | Qwell | huh? |
| 03:50.14 | mishehu | Qwell: rebooting a system that had the bad reiser4 patch on it. |
| 03:50.25 | mishehu | Qwell: oh sorry, that was CunningPike who did that |
| 03:50.33 | mishehu | CunningPike: What Me Version? |
| 03:50.50 | CunningPike | huh? |
| 03:51.07 | mishehu | CunningPike [n=arodgers@S010600095b33697f.vc.shawcable.net] requested VERSION from mishehu |
| 03:51.29 | mishehu | I don't know why you versions me. *grin* |
| 03:51.35 | mishehu | s/versions/versioned |
| 03:51.42 | CunningPike | I'm with the FBI |
| 03:52.09 | mishehu | CunningPike: Former Business Interests? |
| 03:52.12 | matt_ | :) |
| 03:52.30 | mishehu | Qwell: and I'm sure you learned as much as CunningPike did ;-) |
| 03:52.37 | Qwell | That you're away? |
| 03:52.38 | Qwell | yes |
| 03:52.55 | mishehu | you didn't know I'm away already? psssh. if I was here I wouldn't be rambling. |
| 03:54.00 | mishehu | I must be a lot more tired than I actually feel. |
| 03:54.19 | tzanger | [TK]D-Fender: around? |
| 03:54.25 | tzanger | what's a polycom config file error 0x10000 |
| 03:54.56 | mishehu | sounds bad. |
| 03:55.09 | mishehu | maybe I'll wait on updating my firmware until tomorrow. |
| 03:56.21 | [TK]D-Fender | tzanger : Yup |
| 03:56.45 | tzanger | [TK]D-Fender: do you know what a polycom error 0x10000 is on bootup? it just says config file error: 0x10000 |
| 03:56.52 | tzanger | I seem to remember this but can't place where or why |
| 03:57.27 | tzanger | the boot log doesn't give me any errors and no other logs have been uploaded |
| 03:57.31 | tzanger | ls |
| 03:57.45 | [TK]D-Fender | tzanger : I believe you pointed your mac.cfg file to a missing config file. |
| 03:58.03 | tzanger | ahh let me check |
| 03:58.05 | [TK]D-Fender | tzanger : Double-check their presence and authorities |
| 03:58.30 | [TK]D-Fender | I'll lay bets its the phoneXX.cfg file that wasn't right. |
| 03:58.40 | tzanger | I think that's exactly right |
| 03:58.44 | tzanger | I was misisng my alertInfo.cfg |
| 03:59.04 | CunningPike | [TK]D-Fender: How much money have you made so far with that ;) |
| 03:59.50 | [TK]D-Fender | CunningPike : With? |
| 04:00.03 | CunningPike | Betting the file was missing |
| 04:00.35 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 04:00.49 | [TK]D-Fender | CunningPike : I've seen it before as well. |
| 04:01.37 | CunningPike | I know - that's why......... oh, never mind. I was trying to be humorous :/ |
| 04:01.59 | CunningPike | And failing miserably...... |
| 04:02.01 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:03.10 | file | brimstone: !!! |
| 04:03.11 | [TK]D-Fender | File & brimstone! |
| 04:03.29 | brimstone | :o |
| 04:03.35 | brimstone | how was your trip file? |
| 04:04.55 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 04:06.02 | file | brimstone: beautiful |
| 04:06.06 | file | excellent |
| 04:06.07 | file | fabulous |
| 04:06.17 | brimstone | they left you in the fedex box again didn't they? |
| 04:07.17 | file | yes :( |
| 04:08.50 | Qwell | file: You actually arrived...with luggage? |
| 04:08.51 | tzanger | thanks [TK]D-Fender, that was it |
| 04:09.33 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:10.28 | file | Qwell: yup |
| 04:10.31 | [TK]D-Fender | tzanger : ywc |
| 04:10.32 | file | my luggage was fine :D |
| 04:10.43 | Qwell | fine, as in...there? |
| 04:10.54 | file | yes! |
| 04:10.56 | file | crazy eh? |
| 04:11.00 | Qwell | eh |
| 04:13.53 | tzanger | wtf this thing will not register now |
| 04:13.56 | tzanger | asterisk gives back a 404 |
| 04:14.56 | *** join/#asterisk ssokol (n=ssokol@136.sub-75-192-175.myvzw.com) |
| 04:15.18 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:15.47 | docelmo | ssokol DUDE! |
| 04:16.42 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:16.57 | niter3 | hey guys i'm setting up agents queue |
| 04:17.09 | Qwell | niter3: sorry |
| 04:17.27 | niter3 | i've added an agent under [agents] as member => 8888,8888,bob |
| 04:17.35 | niter3 | and i've add it to my queue |
| 04:17.49 | niter3 | member => Agent/8888 |
| 04:17.59 | niter3 | then an extension as exten => AgentLogin() |
| 04:18.10 | niter3 | exten => 5,1,AgentLogin() |
| 04:18.12 | niter3 | sorry |
| 04:18.26 | niter3 | and when I dial my agent # 8888 it keeps complaining login agent incorrect please try again |
| 04:18.29 | niter3 | I'm clueles.. |
| 04:19.42 | file | ssokol: moo |
| 04:20.09 | CunningPike | niter3: Try AgentLogin(8888) |
| 04:21.30 | niter3 | login incorrect right away |
| 04:25.14 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 04:25.17 | SaTLaN32 | hi |
| 04:25.20 | SaTLaN32 | need help |
| 04:25.23 | arcanine | does anyone tried vfx card on asterisk |
| 04:25.30 | niter3 | CunningPike: I've checked these files like 10 times. |
| 04:25.31 | niter3 | still see nothing |
| 04:25.44 | SaTLaN32 | i have installed asterisk trunk and since then my fxo card is not working. |
| 04:25.44 | niter3 | I don't know why it's complaining.. |
| 04:25.50 | SaTLaN32 | i get channel busy |
| 04:26.01 | CunningPike | niter3: Did you reload after adding your agent? |
| 04:26.08 | niter3 | yes of course |
| 04:26.30 | arcanine | dialogic vfx card |
| 04:26.44 | CunningPike | niter3: Just checking ;) |
| 04:27.25 | niter3 | this is just weird.. I don't know why it won't work.. :s |
| 04:28.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 04:28.19 | [TK]D-Fender | niter3 : pastebin EVERYTHING related to your queues |
| 04:28.26 | niter3 | ok sec |
| 04:29.24 | tzafrir_laptop | SaTLaN32, what trunk? fxo? What do you mean by "not working"? |
| 04:30.09 | niter3 | http://pastebin.ca/167623 |
| 04:30.16 | SaTLaN32 | anyone can help? |
| 04:30.29 | SaTLaN32 | hi tzafrir |
| 04:32.14 | tzanger | ugh |
| 04:32.24 | tzanger | I *cannot* get theese fucking phones to work |
| 04:32.32 | tzanger | asterisk seems happy |
| 04:32.34 | *** join/#asterisk kc5cqm (n=krausen@cpe-24-170-62-63.stx.res.rr.com) |
| 04:32.37 | tzanger | but the phones have white-phone soft buttons |
| 04:32.43 | tzanger | and you get the little voicemail warble too |
| 04:33.01 | kc5cqm | what's the timefram for asterisk 1.4? |
| 04:33.06 | SaTLaN32 | i'm here |
| 04:33.24 | kc5cqm | or, a better question is, what's the roadmap of items that'll be included on 1.4 that are currently being worked on? |
| 04:33.45 | kc5cqm | besides jingle... |
| 04:34.03 | tzafrir_laptop | SaTLaN32, so it's basicaly an issue of troubleshooting zaptel. Do all the channels appear in 'zap show channels'? |
| 04:34.14 | niter3 | [TK]D-Fender: See anything i've done wrong? |
| 04:34.29 | [TK]D-Fender | tzanger : Setup a new provisioning folder, set your phone to it, and factory reset it. |
| 04:34.53 | tzanger | yeah I think I'm gonna factory zap it |
| 04:35.03 | *** join/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 04:35.47 | [TK]D-Fender | niter3 : Yes, your agents.conf is wrong. its Agent => not Member => |
| 04:35.58 | [TK]D-Fender | niter3 : You've got to read the big print... |
| 04:36.00 | kc5cqm | btw, anyone here know of a iax softphone for win/32 that supports the iax-encryption that asterisk does? |
| 04:36.14 | kc5cqm | or better...an ata that does |
| 04:36.27 | tzanger | [TK]D-Fender: yeah... ugh it's 12:35 already |
| 04:38.49 | niter3 | thanks |
| 04:38.49 | SaTLaN32 | tzafrir |
| 04:38.51 | niter3 | didn't catch that |
| 04:38.52 | SaTLaN32 | you here? |
| 04:39.40 | tzafrir_laptop | yes |
| 04:46.55 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:47.24 | wunderkin | kc5cqm, http://www.sineapps.com/news.php?rssid=1483 |
| 04:47.55 | wunderkin | i'm playing with trunk now, i have to setup odbc now, since app_sql_postgres is gone |
| 04:50.06 | tzanger | [TK]D-Fender: got a crash course in buddies on the polycoms? I've never done this before |
| 04:50.17 | tzanger | got line 1 for hte line, but how to get the other soft keys for buddy/presence stuff |
| 04:50.38 | kc5cqm | thanks wunderkin |
| 04:51.56 | znoG | i find that sometimes asterisk fails to authenticate against SIP servers |
| 04:52.10 | znoG | it doesn't seem to put the digest auth stuff in the packets |
| 04:52.23 | znoG | when i reboot the machine, all is fine again... is there a way to do this without having to reboot? |
| 04:53.08 | CunningPike | tzanger: You need to enable the presence feature in the phone - look for 'feature' in your sip.cfg |
| 04:53.20 | *** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:53.31 | JT | znoG: i'm assuming reloading and restarting asterisk don't work? |
| 04:53.34 | CunningPike | tzanger: There should be a list with a .enabled attribute for each one |
| 04:54.39 | znoG | JT: nope, of course i tried that :) |
| 04:55.32 | JT | tried stopping and starting it? |
| 04:57.16 | [TK]D-Fender | tzanger : Enable presence in sip.cfg. Add a contact that you have a hint set up for. In the contact info screen scroll down and enable buddy-watch. the End. |
| 04:57.37 | [TK]D-Fender | tzanger : Minus the fact it seems SVN presence is b0rked |
| 04:57.40 | tzanger | that's too simple |
| 04:57.47 | tzanger | CunningPike: thanks |
| 04:57.47 | [TK]D-Fender | tzanger : Isn't it though? |
| 04:59.12 | CunningPike | It's great |
| 04:59.36 | CunningPike | We set up hints for all our phones, so we can set up BLFs really easily on the phone |
| 05:00.29 | *** join/#asterisk ajungem (n=ajunge@201.238.192.238) |
| 05:00.59 | [TK]D-Fender | tzanger : If you're modding the -directory files manually, its the <bw>1</bw> you'll want to set |
| 05:02.29 | ajungem | any body here? |
| 05:02.56 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 05:04.33 | tzanger | hmm |
| 05:04.35 | *** join/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 05:04.56 | tzanger | is it possible to assign all three (6) softkeys to buddies? I need one softkey to register to to the asterisk server do I not? |
| 05:05.49 | [TK]D-Fender | tzanger : Need 1 key for a reg, the rest you can do what you will with |
| 05:08.54 | HaMYaI | hi, how is the order of the cards in /proc/zaptel arranged? |
| 05:09.24 | HaMYaI | it doesn't seem to follow the order in /etc/modprobe.d/zaptel sometimes |
| 05:10.18 | *** part/#asterisk kc5cqm (n=krausen@cpe-24-170-62-63.stx.res.rr.com) |
| 05:10.34 | tzanger | that's what I thought, thanks |
| 05:13.23 | ajungem | HaMYaI: the cards are assigned in the order they are detected by the kernel. lspci will tell you |
| 05:13.43 | [TK]D-Fender | tzanger : Keeping in mind that there are a number of ways of conveying that info. You can script a microbrowser page for 600/601/650 or use the "buddies" button to view them in a window (limited to 8 in any non 600 class), etc. |
| 05:14.10 | tzanger | [TK]D-Fender: indeed. I'm ust starting out on that |
| 05:14.30 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:14.59 | [TK]D-Fender | tzanger : 1. is supposed to come with a programable "device" for using Presence to convey other info as well (BRISTUFF had a patch for this but those with PRI couldn't take advantage of it) |
| 05:15.09 | [TK]D-Fender | 1.4* |
| 05:18.40 | arcanine | wer cn i get stable version of asterisk? |
| 05:19.04 | [TK]D-Fender | ok, I'm baked... later folks. |
| 05:19.06 | HaMYaI | ajungem: lspci ? why did the order change from time to time? |
| 05:19.23 | HaMYaI | ajungem: can we indicate which order we want? |
| 05:19.30 | ajungem | mmm. strange |
| 05:19.39 | ajungem | i don't think so |
| 05:19.52 | tzafrir_laptop | [TK]D-Fender, what do you mean regarding bristuff? |
| 05:20.08 | *** join/#asterisk RoyK (n=roy@ti211210a080-0574.bb.online.no) |
| 05:20.13 | HaMYaI | ajungem: I thought changing the order in /etc/modprobe.d/zaptel will help |
| 05:20.42 | tzafrir_laptop | HaMYaI, /proc/zaptel is ordered y the order the spans were registered to zaptel |
| 05:21.27 | tzafrir_laptop | Usually a span is registered to zaptel immediately when the module loads. And usually the modules load at the order of the list in that fle |
| 05:21.48 | HaMYaI | tzafrir_laptop: usually |
| 05:21.53 | *** part/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 05:22.21 | tzafrir_laptop | What cards do yo uhave? |
| 05:22.35 | HaMYaI | tzafrir_laptop: when the order in /proc/zaltel changes then I had to re-arrange the order in /etc/zaptel.conf as well |
| 05:23.05 | HaMYaI | tzafrir_laptop: TE110P and TDM400 with 4 x fxs |
| 05:23.13 | tzafrir_laptop | That's why I wrote genzaptelconf in the first place (to help us in testing internally) |
| 05:24.00 | HaMYaI | tzafrir_laptop: where do I find that script? |
| 05:24.00 | JT | tzafrir_laptop: so you're the culprit ;) |
| 05:24.34 | tzafrir_laptop | xpp/utils/genzaptelconf . However the real answer is that you need to figure out the order which they will load at boot and configure accordingly |
| 05:24.58 | tzafrir_laptop | Rewriting the configus on every boot generally is too dangerous |
| 05:25.05 | HaMYaI | <PROTECTED> |
| 05:25.38 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.160.146) |
| 05:25.59 | HaMYaI | tzafrir_laptop: this Tiger Jet is at my eth0 and is really annoying because I have my ethernet config at eth0 |
| 05:26.16 | tzafrir_laptop | JT, well, actually the very first script was a little awk hack by Oron Peled. But almost anything there is mine... |
| 05:26.47 | tzafrir_laptop | HaMYaI, what network GUI? What distro? |
| 05:27.12 | JT | tzafrir_laptop: i just found it was setting up my T1 for pri when I wanted it setup for a channelbank so i gave up and wrote it all from hand :) i'm sure it's useful if you've got a pri |
| 05:29.10 | HaMYaI | tzafrir_laptop: the network configuration in x-windows of my FC5 |
| 05:29.22 | ajungem | tzafrir? are yo tzafrir cohen from xorcom? |
| 05:29.26 | tzafrir_laptop | yes |
| 05:29.57 | ajungem | we were discussing a bug i think i have found.. remember? |
| 05:30.04 | ajungem | Bug#386312: asterisk: deadlocks on channels |
| 05:30.11 | tzafrir_laptop | HaMYaI, anyway, the zaptel init.d script is run before the networking one, so it really doesn't matter |
| 05:30.13 | ajungem | on bugs.debian.org |
| 05:30.14 | HaMYaI | tzafrir_laptop: I just ran your "genzaptelconf" and it just generates and detect TE110P as the first span |
| 05:30.45 | tzafrir_laptop | ajungem, ok |
| 05:30.54 | *** join/#asterisk jart (n=user@wsip-24-120-229-225.lv.lv.cox.net) |
| 05:31.27 | tzafrir_laptop | HaMYaI, if you ran it without -d, it will just configure according to currently-loaded modules |
| 05:31.33 | ajungem | i have new info but i'm still stuck with it. |
| 05:31.55 | tzafrir_laptop | If you run it with -dM is will redetect and also rewrire the zaptel sysconfig file |
| 05:32.19 | tzafrir_laptop | ajungem, what do you mean? what info? what problem? |
| 05:32.25 | ajungem | i have a lot of " Write returned -1 (Resource temporarily unavailable) on channel X" in the debug log file. |
| 05:32.44 | ajungem | sorry. a short explanation first. |
| 05:33.12 | ajungem | after a lot of usage i get some "chan_zap.c: Ring requested on channel 0/7 already in use on span 1. Hanging up owner" |
| 05:33.46 | ajungem | so the channel gets locked and it only get unlocked if I restart asterisk. |
| 05:33.59 | ajungem | no incoming calls on that channel. |
| 05:34.25 | ajungem | i think is some kind of mutex lock problem, but i don't know how to find out |
| 05:35.13 | HaMYaI | tzafrir_laptop: I just ran with no parameter and now it replaces the previous configs with a backup |
| 05:38.10 | HaMYaI | the one of the differences is that mine has "span=1,1,0" and the generated one has "span=1,1,1" |
| 05:38.12 | tzafrir_laptop | is this classic? bristuff? |
| 05:38.25 | HaMYaI | tzafrir_laptop: does it matter? |
| 05:38.55 | tzafrir_laptop | Basically the length of the cable. See the sample zaptel.conf . The "1" there is a pure guess |
| 05:39.08 | ajungem | classic and bristuff the same problem. |
| 05:43.50 | HaMYaI | tzafrir_laptop: distance from the modem? |
| 05:45.06 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
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| 05:48.14 | wunderkin | is there a way to disable realtime for voicemail? i just setup odbc and it automatically assumed that i wanted to use it for voicemail, all i want is to use odbc from the dialplan, nothing else, a replacement for app_sql_postgres |
| 05:51.48 | *** join/#asterisk Toadyus (i=Toadyus@S010600121746f9fe.mh.shawcable.net) |
| 05:56.00 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:59.13 | prog | hello to all. i have just compiled asterisk and when i start it, asterisk says: permission denied creating /var/run/asterisk.pid . i know what does it mean, but how can I force asterisk to use another PID file ( in another directory ), thank you! |
| 05:59.53 | X-Rob_ | prog, mkdir /var/run/asterisk, chown asterisk /var/run/asterisk, nano /etc/asterisk/asterisk.conf and change astrundir to be /var/run/asterisk |
| 06:00.47 | prog | X-Rob_, thank you, astrundir is what i should look for. thank you very much! |
| 06:10.51 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
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| 06:20.58 | HaMYaI | there's Zaptel-1.2.9.1 now |
| 06:23.55 | HaMYaI | and compiles ok on my FC5smp |
| 06:24.21 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
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| 06:41.49 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
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| 07:52.06 | _fa_ | ssssaaaa |
| 07:52.06 | _fa_ | a |
| 07:52.07 | _fa_ | aa |
| 07:52.09 | _fa_ | sorry |
| 08:01.19 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 08:01.47 | adelas | hey guys, i got a question about port forwarding, 5060udp+10k-20kudp, is all you need for sip right? and cisco phones? |
| 08:01.58 | adelas | once in a while, i'm getting like 1 way calling |
| 08:02.38 | adelas | well more common now |
| 08:02.41 | adelas | after firewall install |
| 08:02.50 | adelas | is there any other ports? |
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| 08:23.15 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 08:23.28 | faberk64 | Hi to all |
| 08:23.44 | scage_ | hello |
| 08:24.39 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:25.11 | faberk64 | just one stupid/fast question, from the trixbox web interface, is there a way to let entensions to receive outside calls? |
| 08:25.20 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
| 08:25.21 | faberk64 | to be forwarded |
| 08:25.48 | faberk64 | or I have to setup by hand into extensions_custom.conf? |
| 08:26.00 | faberk64 | is my first trixbox setup |
| 08:26.14 | faberk64 | never used before, just * |
| 08:26.21 | faberk64 | any idea? |
| 08:33.06 | *** join/#asterisk angryuser (n=magasin@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:33.22 | angryuser | good day everybody |
| 08:33.56 | angryuser | can someone help me with asterisk please? |
| 08:35.46 | angryuser | i gor 2 asterisk servers on 2 different IP's, first one is a working stable server, second for my tests, the problem is that when i turn on my second server for testing, asterisk on server number one stops working |
| 08:35.52 | angryuser | *i got |
| 08:37.01 | angryuser | any ideas? |
| 08:39.40 | key2 | angryuser: they have the same IP |
| 08:39.40 | key2 | ? |
| 08:40.01 | angryuser | no, 2 different standalone servers |
| 08:43.20 | CtRiX | change the sip port of 1 of them and don't let authenticate to the same accounts |
| 08:43.33 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 08:44.25 | angryuser | il try now, thank you for help |
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| 09:04.46 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 09:07.55 | angryuser | your solution have worked |
| 09:11.21 | angryuser | i have another tiny problem, i have one fxo behind nat, so i routed port 5060 to ip, and of course 5060 to asterisk server, i can call from user, but when i have a pick up, i hear no voice, maybe should i route some others ports, to the server or fxs? to |
| 09:11.32 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 09:12.23 | angryuser | i got fxs--router----internet-----router-----asterisk |
| 09:12.32 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 09:12.34 | Bert- | hello there |
| 09:12.58 | angryuser | hi |
| 09:14.36 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.50 | angryuser | i forgot to notice that asterisk outgoing is configured like a trunk to external phone provider over the internet |
| 09:16.26 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 09:16.35 | Bert- | I have a issue with background command. I had to install asterisk on other computer |
| 09:17.11 | Bert- | I put my working config and now asterisk is unable to play any audio file. got that in asterisk logs : http://pastebin.ca/167747 |
| 09:17.42 | Bert- | unable to find file, but file is same as old server, in the same path ... So I don't understand why |
| 09:22.44 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:23.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:30.26 | Bert- | ... |
| 09:30.43 | Bert- | have to specify FULL path !!! |
| 09:30.52 | Bert- | :( |
| 09:31.03 | *** join/#asterisk devooo (n=al@124.6.183.1) |
| 09:31.12 | Bert- | is a way to specify sound path or things like that ? |
| 09:46.32 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 09:47.11 | *** part/#asterisk Jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 09:49.39 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 09:53.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 09:56.32 | *** join/#asterisk dwmw2 (n=dwmw2@baythorne.infradead.org) |
| 10:02.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 10:03.34 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 10:08.23 | *** join/#asterisk nounoursfr (n=nounours@core1.mesbox.net) |
| 10:08.26 | nounoursfr | hi all |
| 10:08.33 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:08.35 | nounoursfr | have are you today |
| 10:08.53 | puzzled | morning |
| 10:10.55 | *** part/#asterisk scage_ (n=xzen@202.63.226.41) |
| 10:13.32 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 10:18.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:20.45 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 10:33.57 | key2 | someone knows a softphone writen in flash ? |
| 10:36.38 | *** join/#asterisk aLeSD (n=parappap@197.Red-88-4-222.staticIP.rima-tde.net) |
| 10:36.52 | X-Rob_ | key2, flash can't do UDP |
| 10:37.11 | aLeSD | hi all : maybe is not the right place to make this question? But someone know a minimal scalable sip client ? |
| 10:38.56 | key2 | X-Rob_: with a http tunnel |
| 10:39.02 | key2 | or something like that ? |
| 10:39.15 | X-Rob_ | key2, VoIP requires UDP |
| 10:39.20 | key2 | X-Rob_: could flash do audio codec ? |
| 10:39.33 | key2 | X-Rob_: well with a gateway in the middle it would be possible |
| 10:39.41 | X-Rob_ | key2, have fun writing it! |
| 10:39.48 | key2 | X-Rob_: if I make a chan_flash for asterisk |
| 10:39.53 | key2 | X-Rob_: it won't be possible ? |
| 10:40.11 | X-Rob_ | Um. It might be. |
| 10:40.12 | Bert- | hmm |
| 10:40.13 | key2 | X-Rob_: the question is could flash do video + audio codec ? |
| 10:40.20 | Bert- | I have a .wav for IVR |
| 10:40.22 | CtRiX | and also a stream that flash would understand |
| 10:40.26 | X-Rob_ | I'm pretty sure that flash can capture audio |
| 10:40.30 | X-Rob_ | don't know about it capturing video |
| 10:40.38 | Bert- | does someone has a clear howto to convert it to good format for asterisk please ?? |
| 10:41.00 | key2 | X-Rob_: but if I have a H263 video, could flash decode it ? |
| 10:41.08 | key2 | X-Rob_: or G729, could flash decode it ? |
| 10:41.19 | Bert- | tried with sox, GX Trancoder and onine audioconv from asteriskguru |
| 10:41.22 | key2 | X-Rob_: or it uses a preset set of codec |
| 10:41.30 | X-Rob_ | key2, dude, ask someone who knows flash. |
| 10:41.39 | Bert- | quality is really bad with guru conversion |
| 10:41.46 | Bert- | and I can upload .wav :( |
| 10:41.49 | key2 | X-Rob_: u started to answer, I assumed u knew :) |
| 10:41.59 | Bert- | s/can/can't |
| 10:42.20 | X-Rob_ | key2, I answered a technical reason why I knew it couldn't. |
| 10:44.39 | *** join/#asterisk MacoStefX (n=stephane@gw.sortilege.net) |
| 10:44.40 | file | >_< |
| 10:47.30 | phearless | ;_; |
| 10:47.56 | *** join/#asterisk Tili (n=tili@202.133.65.58) |
| 10:48.03 | *** part/#asterisk MacoStefX (n=stephane@gw.sortilege.net) |
| 10:48.34 | file | I'm up far too early |
| 10:53.57 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 10:57.11 | *** join/#asterisk Qb3rt (n=jhgjkgui@58.68.252.216.dsl1.colba.net) |
| 10:58.21 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 11:01.16 | *** join/#asterisk _deg_ (n=deg@201.40.214.155) |
| 11:11.44 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 11:15.16 | *** join/#asterisk _Problem_ (n=lokesh@estrela.nortenet.pt) |
| 11:17.58 | Dr-Linux | file: good afternood |
| 11:19.08 | file | hi |
| 11:25.36 | *** join/#asterisk basty (n=basty@212.218.65.209) |
| 11:25.37 | basty | Hi |
| 11:27.23 | basty | Is it possible to log via asterisk to an external logfile? I have several agents with different status. If the agent selects status "Not Available". The Agent should log off and the log containing a timestamp should log into /etc/asterisk/test.log. |
| 11:27.27 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:28.07 | basty | I want to analyse the agents what kind of status they have had including the date and time into an external logfile. |
| 11:30.21 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 11:44.01 | Sonderblade | i need a really simple system that can bill incoming calls to certain extensions on an Asterisk with different tariffs depending on extension, anyone know of some project that can do this? |
| 11:45.55 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 11:46.04 | mut | omg file noooooooooo |
| 11:46.11 | mut | stop drop and roll! |
| 11:46.14 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 11:46.45 | file | mut: I refuse |
| 11:46.52 | mut | then perish! |
| 11:47.02 | file | I also refuse to do that |
| 11:47.11 | mut | doh |
| 11:47.47 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 11:50.55 | aadilismail | do me a favor to instal asterisk ... can anybody help |
| 11:52.22 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 11:52.26 | Sasch | hi all |
| 11:52.54 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 11:54.54 | Sasch | i have one problem with musiconhold and Grandstream BT120 |
| 11:55.13 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 11:56.02 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
| 11:56.14 | kindor | does the BLF work on siemens optipoint 410s series? |
| 11:56.21 | kindor | (hint function?) |
| 11:56.35 | Sasch | i have a extension 6000 |
| 11:56.36 | Sasch | exten => 6000,1,Answer |
| 11:56.36 | Sasch | exten => 6000,2,MusicOnHold() |
| 11:56.36 | Sasch | exten => 6000,3,Dial(SIP/sascha) |
| 11:56.46 | Sasch | with xlite the music start |
| 11:56.56 | Sasch | with my telephone grandstream don't start |
| 11:57.01 | Sasch | and asterisk return |
| 11:57.26 | Sasch | Sep 12 15:59:15 WARNING[2207]: chan_sip.c:2570 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/64) |
| 11:57.44 | Sasch | why ?? can help me ..... |
| 11:57.53 | CtRiX | codec issue |
| 11:58.23 | file | you're using G723.1 |
| 11:59.37 | Sasch | when i find the codec configuration ?? in sip.conf ?? |
| 11:59.43 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 12:00.14 | Sasch | this is the configuration of the telephone |
| 12:00.15 | Sasch | [assistenza] |
| 12:00.15 | Sasch | type=friend |
| 12:00.15 | Sasch | callerid="Assistenza" <10> ; Full caller ID, to override the phones config |
| 12:00.15 | Sasch | ;nat=yes ; there is not NAT between phone and As$ |
| 12:00.15 | Sasch | disallow=all ; need to disallow=all before we can use allow= |
| 12:00.16 | Sasch | allow=all ; Pass-thru only unless g729 license obtained |
| 12:00.18 | Sasch | secret=password |
| 12:00.20 | Sasch | host=dynamic |
| 12:00.36 | benjk | ~pastebin |
| 12:00.39 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 12:00.48 | CtRiX | disallow=all && allow=all <-- really usefull |
| 12:01.07 | benjk | LOL |
| 12:01.09 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 12:01.10 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 12:01.12 | Sasch | :-P |
| 12:01.24 | Sasch | I'm little in asterisk |
| 12:01.39 | benjk | that one is more like common sense though |
| 12:01.43 | CtRiX | bisogna leggere ... |
| 12:01.48 | E-bola | hmm http://www.snapanumber.com seems down |
| 12:01.51 | benjk | no offense intended ;) |
| 12:01.59 | Sasch | excusme for my language but I'm Italian ... I live near Montalcino |
| 12:02.06 | Sasch | <CtRiX> sei italiano ?? |
| 12:02.21 | benjk | Brunello di Montalcino |
| 12:02.22 | CtRiX | Sash yes but manuals are i english ! |
| 12:02.52 | Sasch | <CtRiX> nooooo sei un grande !!!!!!!!!!!!! Quanti italiani siamo in questo canale ??? |
| 12:03.04 | CtRiX | Sasch, english |
| 12:03.16 | CtRiX | just to be polite to the ones who cannot read |
| 12:03.25 | Sasch | ok |
| 12:04.09 | Sasch | in my sip.conf i must add this allow = G723.1 |
| 12:05.08 | CtRiX | disallow=all |
| 12:05.11 | CtRiX | allow=ulaw |
| 12:05.13 | CtRiX | allow=alaw |
| 12:05.19 | CtRiX | start with tis one |
| 12:05.27 | CtRiX | *this |
| 12:05.30 | *** join/#asterisk foxmjay (n=root@ll81-144-114-192-81.ll81.iam.net.ma) |
| 12:05.38 | Sasch | ok |
| 12:06.23 | Sasch | ok all work thanks |
| 12:06.49 | Sasch | for music hold i use madplay |
| 12:07.05 | Sasch | whit this configuration application=/usr/bin/madplay --mono -R 8000 --output=snd:- |
| 12:08.57 | CtRiX | it's choppy. Isn't it ? |
| 12:11.01 | X-Rob_ | Madplay's pretty good, but format_mp3 is better |
| 12:11.34 | Sasch | there is format_mp3 in debian reposity ?? |
| 12:11.59 | prog | hello, i upgraded from ast 1.0 to 1.2.11 and im getting "SIP 421 Extension required" error . Do anobody know what is the reason ? |
| 12:12.00 | CtRiX | X-Rob_, i think the problem is that MOH is choppy when connecting to VOIP providers using VAD |
| 12:12.01 | X-Rob_ | it's in asterisk-addons |
| 12:12.07 | X-Rob_ | CtRiX, ah. yeah |
| 12:12.10 | CtRiX | ans silence suppression. That's an asterisk timing issue. |
| 12:12.16 | coppice | X-Rob_: did serverpronto ever sort things out? |
| 12:12.18 | CtRiX | openpbx doesn't have. |
| 12:12.26 | X-Rob_ | coppice, yeah, but I'm sticking with this mob |
| 12:12.29 | X-Rob_ | I paid for a year hosting |
| 12:12.46 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.68.17) |
| 12:12.50 | X-Rob_ | Going to make sure everything's off the serverpronto machine and shut it down |
| 12:13.26 | coppice | X-Rob: the industry standard for customer service - none |
| 12:13.29 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:13.38 | X-Rob_ | coppice, yeah. It's pretty sucky. |
| 12:14.05 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 12:14.27 | Sasch | I have one tdm400p with one fxs and one FXO .... |
| 12:16.49 | Godsey | might anyone know of an NPR ogg stream? :) |
| 12:17.17 | Godsey | I have a script working w/ playing ogg streams, but can't seem to figure out how to do it w/ madplay |
| 12:20.08 | Sasch | if i want to add in extension.conf a directive that when a client in not present call another cliente |
| 12:20.12 | Sasch | client excusmi |
| 12:20.17 | Sasch | :-P |
| 12:23.16 | *** join/#asterisk vhatz (n=Vlasis@194.219.121.194) |
| 12:27.54 | Godsey | can I use xmms to feed asterisk? |
| 12:27.54 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 12:27.54 | *** join/#asterisk Frogdude (n=chris@c-24-16-72-159.hsd1.wa.comcast.net) |
| 12:29.54 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 12:30.03 | CtRiX | Sasch, check DIALSTATUS variable after a call to Dial() or use hints and ast manager. |
| 12:30.09 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-56-39.telkomadsl.co.za) |
| 12:30.15 | |oranjia| | hello peeps :) |
| 12:30.32 | aadilismail | helo |
| 12:30.51 | aadilismail | new to astrisk ....can anybody tell how to instal asterisk |
| 12:30.51 | |oranjia| | hey aadilismail |
| 12:31.05 | aadilismail | helo oranjia |
| 12:31.35 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:31.53 | aadilismail | hi |
| 12:31.56 | |oranjia| | aadilismail: there's enough documentation on the interwebs :) |
| 12:31.57 | aadilismail | guys ... |
| 12:32.19 | |oranjia| | http://www.asteriskguru.com/tutorials/asterisk_installation.html |
| 12:32.22 | aadilismail | which one is the best |
| 12:32.47 | aadilismail | ok let me check |
| 12:33.00 | |oranjia| | did you get all the packages |
| 12:33.37 | |oranjia| | install zaptel : make linux26 && make install |
| 12:33.57 | |oranjia| | then the same for libpri : just make and make install |
| 12:34.05 | |oranjia| | then repeat for asterisk :) |
| 12:34.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:34.57 | aadilismail | ok |
| 12:35.02 | angryuser | install is easy...after install fun is better:) |
| 12:38.50 | |oranjia| | has anyone tried to cluster/load balance asterisk servers? |
| 12:39.24 | aadilismail | what about configuration of asterisk |
| 12:39.36 | ziwapandey1980 | anyone using asterisk based predictive dialer |
| 12:40.03 | aadilismail | thnx budies |
| 12:40.08 | aadilismail | leaving |
| 12:43.48 | *** join/#asterisk VonGodric (n=VonGodri@tuli.elion.ee) |
| 12:43.57 | VonGodric | hello |
| 12:44.03 | Sasch | i want to run asterisk with postgresql |
| 12:44.07 | VonGodric | anyone here who could help me a bit? |
| 12:44.09 | Sasch | I read this article http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 12:44.12 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:44.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:44.24 | Sasch | but where I can find the .sql file ?? |
| 12:44.33 | Sasch | the scheme of db of asterisk ?? |
| 12:44.45 | |oranjia| | Sasch: why not mysql? |
| 12:45.04 | Sasch | because postgresql for commercial is free |
| 12:45.08 | VonGodric | what is wrong with this: |
| 12:45.09 | VonGodric | [incoming] |
| 12:45.09 | VonGodric | exten => s,1,Answer( ) |
| 12:45.09 | VonGodric | exten => s,2,Playback(hello-world) |
| 12:45.10 | VonGodric | exten => s,3,Hangup( ) |
| 12:45.16 | VonGodric | the s doesn't seem to work |
| 12:45.22 | Sasch | and mysql must pay a license |
| 12:45.25 | VonGodric | isn't it supposed to applay to any number sent? |
| 12:45.26 | Sasch | for commercial |
| 12:46.14 | VonGodric | anyone? |
| 12:46.28 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.11.155.revip2.asianet.co.th) |
| 12:48.08 | Sasch | ok I find the .sql in the guide :-P |
| 12:49.26 | jamincollins | Sasch: this might be slightly off-topic but I don't see how mysql can require that their GPL'd version not be used for commercial uses unless you pay them, that's an additional restriction |
| 12:50.39 | VonGodric | can someone explain me how 's' extension works? |
| 12:50.43 | VonGodric | [incoming] |
| 12:50.43 | VonGodric | exten => s,1,Answer( ) |
| 12:50.43 | VonGodric | exten => s,2,Playback(hello-world) |
| 12:50.43 | VonGodric | exten => s,3,Hangup( ) |
| 12:50.46 | niter3 | i've setup a sip provider and i'm placing out going calls through the provider. However, whenever I call another PBX over this trunk and try to push #'s for instance 1 to speak with so and so or 0 to speak with operator my #'s are not being processed by the destination pbx... Any idea what i might be missing? |
| 12:51.14 | VonGodric | if I try this -and dial a number. it says not found - wrong number |
| 12:51.49 | VonGodric | if I put explicit number instead of s |
| 12:51.50 | VonGodric | it works |
| 12:51.51 | *** join/#asterisk ellisdee (i=ellisdee@cpe-70-116-118-236.houston.res.rr.com) |
| 12:52.19 | CtRiX | niter3, dtmfmode = XXXX in sip.conf. Change it. |
| 12:53.03 | [TK]D-Fender | VonGodric: "s" doesnt catch whatever you want. "s" is for very specific reasons. |
| 12:53.26 | [TK]D-Fender | VonGodric: Go read up on "asterisk standard extensions" on the WIKI. |
| 12:53.34 | [TK]D-Fender | VonGodric: And/or go read THE BOOK |
| 12:53.37 | [TK]D-Fender | ~book |
| 12:53.39 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:53.49 | VonGodric | asteriskTFOT |
| 12:53.54 | VonGodric | I'm reading it right now |
| 12:53.57 | VonGodric | there I found teh sample |
| 12:54.33 | VonGodric | yea reading it right now |
| 12:55.03 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 12:55.12 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 12:55.39 | Egonis | When I make outgoing calls, or call voicemail, etc.. it takes up to 5 seconds to register and pick up the channel -- has anyone else experienced this? |
| 12:56.35 | [TK]D-Fender | Egonis: Pastebin the CLI output of a call with this delay. |
| 12:56.48 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:58.05 | niter3 | CtRiX: This will not effect my sip clients will it? |
| 12:58.13 | VonGodric | this start 's' extension isn't working for me |
| 12:58.15 | *** join/#asterisk ionix (n=ionix@p1104-ipbfp03miyazaki.miyazaki.ocn.ne.jp) |
| 12:58.36 | ionix | Hi, how do I strip the 1st digit of ${EXTEN} when I process a DIAL? |
| 12:59.12 | dsfr | ionix: ${EXTEN:1} |
| 13:00.18 | ionix | ah, thx, the manual talk about using STRIP MSD and stuff but I knew there was an easier solution. Thx |
| 13:00.21 | [TK]D-Fender | VonGodric: Define "not working". How are you trying to call it? |
| 13:00.38 | VonGodric | I deal a number |
| 13:00.42 | VonGodric | and try it |
| 13:00.45 | VonGodric | ain't working |
| 13:00.56 | VonGodric | if I put a number instead of 's' |
| 13:00.59 | VonGodric | then it works |
| 13:01.01 | [TK]D-Fender | VonGodric: "s" is only for when * DOESN'T know the number dialed. |
| 13:01.08 | niter3 | CtRiX: What dtmfmode should I select? |
| 13:01.13 | VonGodric | ah |
| 13:01.24 | [TK]D-Fender | VonGodric: "s" is NOT a catch-all! _X would capture any NUMBER. |
| 13:01.37 | ionix | use _X |
| 13:01.45 | pablus | morning |
| 13:01.47 | ionix | use _X. actually |
| 13:01.48 | [TK]D-Fender | VonGodric: Which is something you also don't what to do 99.9% of the time. |
| 13:01.49 | ionix | with a dot. |
| 13:01.58 | pablus | morning |
| 13:02.02 | VonGodric | tnx man |
| 13:02.03 | [TK]D-Fender | ionix: Correct, missed the . |
| 13:02.10 | VonGodric | now works :P |
| 13:07.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.13 | niter3 | which dtmfmode do I need to use to pass outgoing #'s to pstn pbx's? |
| 13:09.17 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 13:11.20 | [TK]D-Fender | niter3: HUH? |
| 13:11.44 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 13:12.07 | [TK]D-Fender | niter3: What is a "pstn pbx" and how are you getting to it? |
| 13:12.23 | niter3 | I have sip outbound trunk and I can call out to normal PSTN. However, if I call a company with a pbx I can't pass digits to the pbx. So for instance, 0 to speak with the operator. |
| 13:12.43 | Sasch | if I want to use this module format_mp3 |
| 13:12.50 | Sasch | for play music in musiconhold |
| 13:12.58 | Sasch | i compile asterisk-addons |
| 13:12.58 | grabeez | Anyone still seeing issues with chanspy w/1.2.12 ? |
| 13:13.09 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:13.10 | Sasch | and make ; make install |
| 13:13.25 | Sasch | the file will create in /usr/lib/asterisk/modules |
| 13:13.40 | Sasch | i edit the file /etc/musiconhold.conf |
| 13:13.51 | [TK]D-Fender | niter3: You should be using the DTMF mode your SIP trunk provider tells you to. |
| 13:14.03 | Sasch | and application = /usr/lib/asterisk/modules/format_mp3.so |
| 13:14.38 | [TK]D-Fender | niter3: If you're on ULAW they might request you use INBAND (often not a good idea, but what they'll support), most others will use RFC2833 |
| 13:14.43 | Sasch | is it is right? |
| 13:15.28 | [TK]D-Fender | Sasch: NO. just use "mode=files" like the sample tells you to and * will pick the appropriate decoder based on what files are in your MOH folder. |
| 13:16.08 | phearless | I got a Linksys PAP2T since today and it is kind of weird |
| 13:16.47 | phearless | in the * CLI I can see : |
| 13:16.49 | phearless | Name/username Host Dyn Nat ACL Port Status |
| 13:16.49 | phearless | 204/204 10.2.12.204 D 5060 Unmonitored |
| 13:16.54 | phearless | (sip show peers) |
| 13:17.07 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 13:17.21 | phearless | but I can not do anything with the phone plugged to the PAP2T ! |
| 13:17.51 | Sasch | ok |
| 13:17.56 | Sasch | thanks <[TK]D-Fender> |
| 13:17.59 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:19.04 | Sasch | the music start but asterisk return this error |
| 13:19.05 | Sasch | Sep 12 17:20:37 WARNING[9796]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 13:20.21 | *** join/#asterisk Frogdude (n=chris@c-24-16-72-159.hsd1.wa.comcast.net) |
| 13:21.28 | brimstone | id3 tags |
| 13:23.08 | phearless | so anybody got a linksys/sipura/cisco SIP phone adapter ? |
| 13:23.38 | benjk | Sipura 3000 |
| 13:23.39 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:25.23 | [TK]D-Fender | Sasch: Yes, you must not have any ID3 tags, or VBR |
| 13:26.00 | [TK]D-Fender | phearless: Pastebin the SIP debug of a failed call. |
| 13:29.23 | phearless | okay 1s I am trying with another phone |
| 13:29.55 | HaMYaI | anyone knows what's the rule to add to iptables to allow SIP? |
| 13:29.57 | HaMYaI | iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT |
| 13:30.10 | HaMYaI | is this correct? |
| 13:30.50 | brimstone | HaMYaI, you don't need "-m udp" but that looks like it'll allow ports 5004 through 5082 |
| 13:31.10 | HaMYaI | I thought only 5060 is required but that's from the wiki |
| 13:31.23 | brimstone | 5060 is the signalling port |
| 13:31.40 | brimstone | you need to allow RTP, the audio ports, into and out of your system as well |
| 13:31.48 | benjk | its more like a default port |
| 13:32.26 | HaMYaI | brimstone: yeah I know for RTP I put -p udp -m state --state NEW -m udp --dport 10000:20000 -j ACCEPT |
| 13:32.31 | benjk | some devices have multiple PTSN jacks which are mapped to 5060, 5061, 5062 ... |
| 13:32.46 | HaMYaI | this is from FC5 |
| 13:32.47 | [TK]D-Fender | HaMYaI: Thats nifty... UDP is STATELESS.... |
| 13:32.56 | brimstone | HaMYaI, then just 5060 should work then |
| 13:33.09 | benjk | depends |
| 13:33.36 | HaMYaI | benjk: ok, go on |
| 13:33.38 | benjk | if you are using a device or provider that does signaling on say 5061, then you probably want that too |
| 13:33.51 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 13:34.20 | benjk | for example, if you have a Sipura device with two FXS ports |
| 13:34.28 | HaMYaI | [TK]D-Fender: so state NEW isn't right? |
| 13:34.32 | benjk | they have to be on different ports |
| 13:34.45 | benjk | because the device has only got one ip address |
| 13:34.49 | [TK]D-Fender | HaMYaI: Probably just gets ignored, but sure doesn't sound right now does it? |
| 13:34.55 | HaMYaI | benjk: yeah, that's correct |
| 13:34.58 | benjk | so FXS1 is on 5060 and FXS2 probably on 5061 |
| 13:35.09 | benjk | iof course this is configurable |
| 13:35.18 | *** join/#asterisk Muck- (n=Muck@145.253.170.162) |
| 13:35.25 | benjk | you could use port 50666 if you so desire |
| 13:35.34 | [TK]D-Fender | benjk = entirely correct. |
| 13:35.46 | CtRiX | <[TK]D-Fender> HaMYaI: Thats nifty... UDP is STATELESS.... |
| 13:36.02 | HaMYaI | benjk: you mean I will need SIP to listen on other ports apart from 5060? |
| 13:36.02 | tzanger | morning |
| 13:36.04 | CtRiX | but iptables and many other firewalls treat UDP streams with a state |
| 13:36.23 | [TK]D-Fender | tzanger: Mornin' get the phone reset from last night? |
| 13:36.24 | CtRiX | which has nothing to do to the fact that udp is stateless |
| 13:36.29 | tzanger | [TK]D-Fender: all but one |
| 13:36.37 | tzanger | this last one just won't fucking clear |
| 13:36.41 | HaMYaI | CtRiX: ok, that makes sense |
| 13:36.43 | tzanger | mind you I'm not the one resetting it so they may be missing something |
| 13:37.05 | tzanger | they claim that it doesn't have both "reset local config" and "reset phone config" options |
| 13:37.09 | CtRiX | so -m state with -p udp is the right thing to do |
| 13:37.53 | HaMYaI | CtRiX: I'm reading about that too |
| 13:38.08 | mut | OUCH!! http://tinyurl.com/g68w9 |
| 13:38.21 | Godsey | asterisk 29964 0.0 1.9 32140 9780 pts/5 S 09:37 0:00 mplayer -ss 30 -rtc-device /dev/zap/timer -cache 768 -really-quiet -quiet -shuffle -ao pcm:nowaveheader:file=/tmp/mplayer.29925.fifo -channels 1 -af resample=8000:0:2 http://66.225.205.60:80/ |
| 13:38.31 | Godsey | that produces pretty good quality moh |
| 13:38.40 | jamincollins | phearless: we use a few sipura/linksys FXS devices |
| 13:39.09 | phearless | okay |
| 13:39.12 | phearless | in fact I got a tone |
| 13:39.24 | phearless | in my phone |
| 13:39.35 | phearless | but after I can't do anything else |
| 13:39.41 | phearless | and there is nothing in the logs |
| 13:39.54 | jamincollins | which device? |
| 13:41.00 | phearless | Linksys/Sipura PAP2T adapter |
| 13:41.00 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 13:41.09 | phearless | with a normal BT cordless phone |
| 13:41.33 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.43) |
| 13:41.53 | phearless | asterisk works fine with all the others phones |
| 13:42.45 | jamincollins | under the Admin login and Line1 or Line2 (depending on where the phone is connected) what is the value for "Make Call Without Reg:" |
| 13:43.19 | phearless | "no" |
| 13:43.34 | [TK]D-Fender | mut: Ouch indeed |
| 13:43.55 | jamincollins | alright, then if you're hearing tone, it should be registered with the Asterisk |
| 13:44.11 | phearless | yes I can see it on "sip show peers" |
| 13:44.29 | phearless | but it seems that the adapter does not understand when I dial something with the phone |
| 13:44.43 | phearless | this phone works fine on a phone line |
| 13:44.58 | jamincollins | switch to the advanced view for the same line |
| 13:44.59 | hank | fg 1 |
| 13:45.16 | jamincollins | what is the value for "Dial Plan:" |
| 13:45.33 | phearless | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 13:45.35 | phearless | the default one |
| 13:45.55 | jamincollins | and you've tried dialing an 11 digit number? |
| 13:45.58 | phearless | I do not know why there is a dial plan here, the dial plan is in asterisk |
| 13:46.02 | phearless | no |
| 13:46.05 | Sasch | in sip.conf the instruction context |
| 13:46.23 | phearless | I tried to call the others local exten, 200, 203 etc |
| 13:46.24 | jamincollins | the dial plan on the PAP2T instructs it how to handle numbers |
| 13:46.33 | phearless | ah I see |
| 13:46.35 | jamincollins | specifically when to consider a number complete |
| 13:47.05 | jamincollins | 200 would be seen as incomplete by the above dial plan and thus wait for more digits until digit timeout |
| 13:47.08 | *** join/#asterisk kuto (n=df2d@125.60.241.24) |
| 13:47.17 | phearless | ah okay |
| 13:47.22 | momelod | hey sorry off topic, but does anyone know of a free php script that displays things like uptime, system load, free memory, hard disk usage..? |
| 13:47.32 | phearless | I will try to modify it :) thanks |
| 13:47.39 | phearless | I will let you know it it worked |
| 13:47.42 | jamincollins | I seem to recall that the PAP2T understands the # key as a dialing termination/send key |
| 13:47.52 | jamincollins | try dialing 200# |
| 13:47.54 | phearless | ah |
| 13:48.05 | *** join/#asterisk swytch (n=root@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 13:48.12 | kuto | hi all, anyone here using the aheeva call center suite? |
| 13:48.30 | jamincollins | there is a good deal of documentation on sipura's site about how their dialplan works |
| 13:48.51 | jamincollins | the PAP2T is essentially a rebranded Sipura 2000, iirc |
| 13:49.18 | jamincollins | momelod: phpsysinfo |
| 13:49.24 | [TK]D-Fender | kuto: Woudn't bet on it. Since theirs is a complete packageds solution for which you don't get the real source etc you won't find its users getting involved on a technical level usually. |
| 13:49.47 | momelod | jamincollins: ty |
| 13:49.54 | [TK]D-Fender | jamincollins: That feature is usually programmable. |
| 13:50.08 | kuto | any alternative i can use aside from aheeva? |
| 13:50.22 | jamincollins | [TK]D-Fender: which? the dial termination key? |
| 13:50.27 | [TK]D-Fender | kuto: Depends what you need. Maybe you should clarify that... |
| 13:50.43 | [TK]D-Fender | jamincollins: Yes, as to whether or not your dialplan will even HAVE one. |
| 13:50.51 | kuto | well i need a function that aheeva does |
| 13:51.09 | jamincollins | [TK]D-Fender: yes, but everything else seemed to be close to defaults on the PAP2T configuration... |
| 13:52.09 | [TK]D-Fender | jamincollins: It may still be configurable, but I don't own that specific model personally. Polycom's have "3" to terminate as a default as well. This is something I naturally remove. |
| 13:52.19 | phearless | I can't find the docs for the PAP2T ... |
| 13:52.22 | [TK]D-Fender | kuto: Do continue clarifying.... |
| 13:52.29 | phearless | I modified the Dial plan to : |
| 13:52.31 | *** join/#asterisk Corydon76-home (i=twelve@pdpc/supporter/sustaining/Corydon76-home) |
| 13:52.31 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:52.33 | phearless | (2xx) |
| 13:52.35 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
| 13:53.05 | [TK]D-Fender | phearless: X.T|*.T|#.T <------- all you should ever need. |
| 13:53.47 | phearless | what is T ? |
| 13:53.51 | jamincollins | phearless: for automatic dialing termination, change that to (200S0) |
| 13:53.53 | kuto | i need a solutions that do inbound outbound recording, autodialling call forwarding |
| 13:54.02 | [TK]D-Fender | jamincollins: "3" should have read as "#" |
| 13:54.06 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:54.33 | phearless | I need to put X.T|*.T|#.T or (X.T|*.T|#.T) in the Dial Plan field ? |
| 13:54.40 | jamincollins | [TK]D-Fender: I've gotten used to it, as one of the PBXs we use defaults to it too |
| 13:54.40 | [TK]D-Fender | kuto: The first part is easy, describe what you mean by the second. |
| 13:55.06 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 13:55.13 | jamincollins | [TK]D-Fender: is that a polycom dialplan or a Sipura/PAP that you posted, I don't recognize it as a PAP2T one |
| 13:55.34 | kuto | its outbound, recording |
| 13:55.58 | [TK]D-Fender | jamincollins: I unfortunately limits what you can use as valid chars in an internal system feature though. Something I dislike. For me its "let me dial whatever the hell I want, THEN refuse me if you're not happy. Not "interrupt me the moment you think my style sucks" |
| 13:56.07 | MacoStefX | re |
| 13:56.09 | Sasch | I create a queque in queques.conf called papinicomputer |
| 13:56.18 | Sasch | why asterisk return Sep 12 17:57:40 WARNING[9882]: app_queue.c:3227 queue_exec: Unable to join queue 'papinicomputer' |
| 13:56.22 | [TK]D-Fender | kuto: "autodialling call forwarding" <- explain. |
| 13:57.55 | kuto | i heard that aheeva can be set to dial numbers on its own using outbound, and you just sit till the line is answered |
| 13:58.39 | kuto | if someone calls you..calls is forwarded to certain locals |
| 13:58.46 | [TK]D-Fender | kuto: What you are looking for is a predictive dialer for outbound call-centers then. Look at Vicidial for * then. |
| 13:59.12 | jamincollins | also, read up on your local regulations for predictive/automated dialers |
| 13:59.13 | [TK]D-Fender | kuto: You ARE looking for an outbound call center, correct? |
| 13:59.46 | jamincollins | lots of potential gotchas with predictive dialers |
| 13:59.49 | kuto | [TK]D-Fender: yes..but i'll have inbounds too |
| 14:00.13 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 14:00.18 | [TK]D-Fender | kuto: Outbound is something Aheeva does rather strongly and the only reason I'd consider them. |
| 14:00.38 | [TK]D-Fender | kuto: Keeping in mind their price is so high. |
| 14:01.00 | kuto | oic.. |
| 14:01.15 | kuto | i guess predictive dialling can solve my problem |
| 14:01.22 | jamincollins | anyone know of PRI problems when using a zaptel with bristuffed patches? |
| 14:01.27 | [TK]D-Fender | kuto: Solve your need more like... |
| 14:01.32 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:02.21 | kuto | if vicidial can do predective dialling like aheeva does..then i'll switch to vicidial, am i right? |
| 14:03.09 | [TK]D-Fender | kuto: Vicidial is just an APP to use with *, not a whole solution. |
| 14:03.33 | niter3 | hey guys. off the topic question, but how can you check when a business was registered? |
| 14:03.44 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 14:03.47 | Godsey | niter3: normally secretary of state |
| 14:04.13 | kuto | well, i need it on * only |
| 14:04.59 | *** join/#asterisk malabar (n=mala@62.97.242.6) |
| 14:05.55 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:07.39 | grabeez | Is there anyway to prevent agents from getting queue calls, while making non-queue calls sent to theie phone/voicemail without pausing them and unpausing them on every call |
| 14:07.53 | *** join/#asterisk xxarmiexx (n=armie@arm.enter.net) |
| 14:09.18 | *** join/#asterisk dalekurt (n=DaleKurt@65.183.3.229) |
| 14:09.31 | dalekurt | UDEV + zaptel giving me hell. |
| 14:09.48 | [TK]D-Fender | grabeez: You'd have to use chan_local for your agents and add in-use detection logic to your dialplan. |
| 14:09.50 | mog | whats wrong charly brown |
| 14:10.25 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 14:10.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:10.30 | wunderkin | grabeez, or if you know their agent number, you can use pausequeuemember |
| 14:11.18 | dalekurt | Can someone help me with my zaptel installation with UDEV. |
| 14:11.21 | grabeez | so I need to build dialplan logic to pause them on every non-queue call correct? |
| 14:11.38 | mog | whats wrong dalekurt |
| 14:11.45 | dalekurt | ok I have Debian |
| 14:11.46 | kuto | [TK]D-Fender: does using the internet running 512kbps cir affect my lan with 30 concurrent user running on 10/100mbps? |
| 14:11.57 | dalekurt | and I think when I installed Xen it installed UDEV as well. |
| 14:12.20 | dalekurt | so now i have have upgraded to asterisk 1.2.11 and zaptel x.x9 |
| 14:12.28 | dalekurt | and I have to get it to now work with udev. |
| 14:12.49 | jamincollins | dalekurt where's it giving you grief? |
| 14:12.50 | _deg_ | Asterisk 1.0.... Is this possible to avoid using mpg123? Just using .wav moh music? |
| 14:12.59 | grabeez | [TK]D-Fender, do you have an example of your way? |
| 14:13.00 | dalekurt | it's just now loading... |
| 14:13.11 | dalekurt | not i's just loading the ztdummy module.. |
| 14:13.25 | jamincollins | _deg_: I don't think it was possible in 1.0 but should be in recent 1.2 |
| 14:13.50 | _deg_ | jamincollins, hmmm. |
| 14:13.54 | jamincollins | dalekurt: so, you modprobe ztdummy and what doesn't happen? |
| 14:13.59 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 14:13.59 | wunderkin | _deg_: only thing you can do is change the player, like madplay, not sure if either of those do wav, but you are way behind the times, 1.4 is almost out |
| 14:14.19 | hwt | do sdm-1 cards that are usable with asterisk available? |
| 14:14.19 | *** part/#asterisk ionix (n=ionix@p1104-ipbfp03miyazaki.miyazaki.ocn.ne.jp) |
| 14:14.23 | _deg_ | jamincollins, but using .wav files insted .mp3 will invoke mpg123 anyway? |
| 14:14.31 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 14:14.45 | dalekurt | well this is what happens... when I restart my zaptel with /etc/init.d/zaptel restart |
| 14:14.51 | dalekurt | I get this now Waiting for zap to come online...Error: missing /dev/zap! |
| 14:14.51 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:14.54 | jamincollins | _deg_: you'd have to use a custom command, I think there's documentation in the voip-info.org wiki on it |
| 14:15.35 | kuto | [TK]D-Fender: can vicidial saves all the call in recording format? |
| 14:15.36 | _deg_ | jamincollins, but with custom I will nedd an external app as well. |
| 14:15.40 | jamincollins | dalekurt: what is the output of "ls /dev/zap/" |
| 14:15.46 | jamincollins | _deg_: yes |
| 14:15.52 | _deg_ | jamincollins, damned |
| 14:16.06 | dalekurt | jamincollins: ls: /dev/zap: No such file or directory |
| 14:16.08 | jamincollins | _deg_: for non-external app I think you need 1.2 |
| 14:16.19 | [TK]D-Fender | kuto: Recording should be *'s job, not the dialer's |
| 14:16.34 | jamincollins | dalekurt: and lsmod lists ztdummy as loaded? |
| 14:17.05 | dalekurt | jamincollins: no it does n ot. |
| 14:17.07 | [TK]D-Fender | grabeez: member => Local/123@context . then in your dialplan make that exten check if they are on the phone, etc. |
| 14:17.20 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 14:17.26 | jamincollins | dalekurt: try loading just it (without the init script for now) "modprobe ztdummy" |
| 14:17.36 | backblue | hi, anyone have messed with something related with AOC? |
| 14:17.41 | kuto | [TK]D-Fender: aah..much better |
| 14:17.51 | dalekurt | jamincollins: WARNING: Error inserting zaptel (/lib/modules/2.6.15-1-486/misc/zaptel.ko): Invalid module format |
| 14:18.03 | dalekurt | jamincollins: WARNING: Error inserting zaptel (/lib/modules/2.6.15-1-486/misc/zaptel.ko): Invalid module format |
| 14:18.17 | jamincollins | sounds like the zaptel modules weren't compiled right |
| 14:18.26 | jamincollins | how did you compile them? |
| 14:18.26 | _deg_ | jamincollins, tks bro |
| 14:18.35 | *** join/#asterisk fulgas (n=fulgas@80.172.227.30) |
| 14:18.39 | [TK]D-Fender | dalekurt: You compile * from source or use packages? |
| 14:18.39 | jamincollins | via module-assistant or from tarball |
| 14:18.40 | dalekurt | jamincollins: from source |
| 14:18.41 | jamincollins | _deg_: np |
| 14:18.58 | dalekurt | [TK]D-Fender: from source |
| 14:18.59 | *** join/#asterisk blue2oo3 (n=blue2oo3@p54A8BC06.dip0.t-ipconnect.de) |
| 14:19.01 | jamincollins | dalekurt: and you have the source/headers for your running kernel installed? |
| 14:19.06 | [TK]D-Fender | dalekurt: Sounds like your kernel source doesn't match what was used for building Zaptel |
| 14:19.22 | jamincollins | or possibly a gcc version mis-match |
| 14:19.23 | Sasch | why when I park a client |
| 14:19.34 | Sasch | if I call 701 don't response ?? |
| 14:19.47 | dalekurt | that was a problem, so I compiled with gcc-3.1 |
| 14:19.47 | *** join/#asterisk Splat (n=Splat@220-253-136-216.TAS.netspace.net.au) |
| 14:19.52 | *** part/#asterisk blue2oo3 (n=blue2oo3@p54A8BC06.dip0.t-ipconnect.de) |
| 14:19.59 | dalekurt | I have gcc-3.1 gcc-4.0 and gcc-4.1 |
| 14:20.23 | dalekurt | it was complaining about it when it compiled with gcc-4.1 so I re-compiled with gcc-3.1 |
| 14:20.24 | jamincollins | dalekurt: which version was used for that kernel? I think Debian's recent kernel's use gcc4 |
| 14:20.31 | phearless | jamincollins, [TK]D-Fender etc |
| 14:20.39 | phearless | so, it still not work |
| 14:20.42 | phearless | :( |
| 14:20.43 | dalekurt | jamincollins: 2.6.15-1-486 |
| 14:20.57 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:21.00 | phearless | in fact I can't even call from antoher phone to the PAP2T |
| 14:21.02 | [TK]D-Fender | phearless: And where is the SIP debug of that failed call I asked for? |
| 14:21.15 | phearless | there is no SIP log because there is no calls |
| 14:21.29 | jamincollins | dalekurt: what is the output of "head -n 1 /var/log/dmesg" |
| 14:21.32 | *** join/#asterisk LoneShadow (n=duh@59.92.183.118) |
| 14:21.55 | [TK]D-Fender | phearless: Then you don't even have them talking in the right direction. |
| 14:22.04 | dalekurt | jamincollins: ops, L1 D cache: 16K |
| 14:22.07 | [TK]D-Fender | phearless: Verify your server IP, etc on your PAP2 |
| 14:22.07 | phearless | maybe maybe |
| 14:22.13 | *** part/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 14:22.26 | jamincollins | dalekurt: that's the first line? |
| 14:22.49 | dalekurt | jamincollins: yeah... what do you want me looking for exactly |
| 14:22.51 | jamincollins | very odd, my first line here indicates the gcc version the kernel was compiled with |
| 14:23.22 | dalekurt | I think that would be 4.0... but let me check |
| 14:23.56 | dalekurt | jamincollins: zaptel: version magic '2.6.15-1-486 486 gcc-4.1' should be '2.6.15-1-486 486 gcc-4.0' |
| 14:24.16 | jamincollins | dalekurt: then you have to use a gcc 4.x version to compile zaptel |
| 14:24.25 | jamincollins | phearless: do you have a proxy configured for your line? |
| 14:24.30 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 14:24.31 | phearless | yes ! |
| 14:24.38 | phearless | the IP of the asterisk server |
| 14:24.41 | jamincollins | got it working? |
| 14:24.43 | dalekurt | Should I recompile zaptel # make CC=gcc-4.0 |
| 14:25.07 | jamincollins | dalekurt: I believe that would be CC=gcc-4.0 make |
| 14:25.16 | dalekurt | thanks |
| 14:26.07 | dalekurt | jamincollins: re-compile it now |
| 14:26.27 | phearless | [TK]D-Fender & jamincollins : http://img246.imageshack.us/img246/9408/screenshotyn3.png |
| 14:26.30 | phearless | my config |
| 14:27.00 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.65.Dial1.SanJose1.Level3.net) |
| 14:28.15 | dalekurt | jamincollins: Must I also do a make install-udev |
| 14:28.27 | jamincollins | phearless: and from the asterisk console a sip debug of that peer shows nothing for a dial? |
| 14:28.28 | phearless | I got just X.T|*.T|#.T in my dialplan, no "(" ")" |
| 14:28.43 | jamincollins | dalekurt: no, we need to get the module loading first |
| 14:28.49 | dalekurt | ok |
| 14:28.54 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:28.58 | dalekurt | so just a make install |
| 14:29.13 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 14:29.14 | *** join/#asterisk ToyMan (n=stuq@74-32-59-182.dsl1.mdl.ny.frontiernet.net) |
| 14:29.15 | jamincollins | phearless: I don't know if that dialplan will work for the PAP2T, try (2XXS0) for starters |
| 14:29.15 | grabeez | [TK]D-Fender I doubt this will work but is this idea the thinking http://pastebin.ca/raw/167908 |
| 14:29.58 | jamincollins | phearless: with that, any 200 series 3 digit extension should automatically end the dial input and send an INVITE to the * server |
| 14:30.03 | backblue | root@193.227.239.13 |
| 14:30.06 | dalekurt | jamincollins: Finished re-compling |
| 14:30.09 | backblue | sorry |
| 14:31.18 | wunderkin | any realtime people here yet? *cringe* i installed unixodbc so i can do use func_odbc in trunk, i dont want to use odbc for anything else, how can i disable realtime for voicemail? it seems to be automatically enabled since i installed unixodbc |
| 14:32.20 | [TK]D-Fender | grabeez: Very close except you don't use "s". |
| 14:32.20 | backblue | wunderkin: i'm with, but not with odbc, that it's good for nuts. |
| 14:33.16 | wunderkin | backblue, using a native driver? |
| 14:33.17 | sevard | [TK]D-Fender: Do you know anything about the sip general setting 'progressinband=no'? |
| 14:33.38 | [TK]D-Fender | sevard: SIP progress should be OOB normally. |
| 14:34.02 | phearless | jamincollins: okay |
| 14:34.18 | sevard | [TK]D-Fender: I get a double ring tone on my ATAs but not on some of my IP Phones, when I enable that option I don't seem to get a double ring on my atas |
| 14:34.27 | *** join/#asterisk somegeek (i=debian-t@tor/regular/somegeek) |
| 14:34.30 | sevard | But I do get Sep 12 09:22:53 WARNING[15782]: channel.c:2070 ast_indicate: Unable to handle indication 3 for 'SIP/2183082764-081b4dc0' |
| 14:34.38 | sevard | roffles. |
| 14:34.43 | phearless | [TK]D-Fender & jamincollins : when I dial any phone number, after a few seconds, I got a "fast busy" dial tone, and nothing is the SIP debug |
| 14:35.01 | phearless | [TK]D-Fender & jamincollins : with any dial plan it is the same... |
| 14:35.04 | grabeez | [TK]D-Fender, do I use congestion to reject it? |
| 14:35.24 | [TK]D-Fender | phearless: If you get nothing, then your IP/port is wrong or something is completely borked in your entworking. |
| 14:35.38 | jamincollins | phearless: you got a syslog server handy? |
| 14:35.44 | sevard | [TK]D-Fender: do you think that's a legit way to eliminate double ring? |
| 14:35.57 | jamincollins | or a linux box you're willing to enable remote syslogging on? |
| 14:36.02 | [TK]D-Fender | sevard: OOB is the only way you should be going. |
| 14:36.11 | sevard | OOB? |
| 14:36.17 | [TK]D-Fender | Out Of Band |
| 14:36.25 | jamincollins | if so, we can get detailed logging from the PAP2T on what it's trying to do |
| 14:36.52 | phearless | jamincollins: I got a syslog on the asterisk box.... you mean to read the logs of the PAP2 ? |
| 14:37.01 | phearless | [TK]D-Fender: what is an entworking ? |
| 14:37.16 | *** join/#asterisk Nix (n=Nix@212.65.148.27) |
| 14:37.26 | jamincollins | phearless: yes, the syslog process on the asterisk would need to be instructed to accept log entries from network sources |
| 14:37.28 | sevard | [TK]D-Fender: so you think the ATA is set up with inband signaling? do you know where that would be in the sip 2002? |
| 14:37.32 | *** part/#asterisk Nix (n=Nix@212.65.148.27) |
| 14:37.36 | phearless | [TK]D-Fender: and my "Proxy:" is the IP of asterisk, and "SIP Port:" is 5060 |
| 14:37.55 | jamincollins | phearless: this is done by starting the syslogd process with the -r switch |
| 14:38.23 | [TK]D-Fender | phearless: And what have you done to try and debig sip on your * server? |
| 14:38.33 | phearless | jamincollins: and on the PAP2 it is "Debug Server:" ? |
| 14:38.58 | jamincollins | phearless: both the Syslog and Debug server and up the debug level |
| 14:39.10 | jamincollins | phearless: 3 should be fine |
| 14:39.27 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 14:39.28 | *** join/#asterisk Nix (n=Nix@212.65.148.27) |
| 14:39.35 | *** part/#asterisk Nix (n=Nix@212.65.148.27) |
| 14:39.48 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 14:39.50 | phearless | [TK]D-Fender: "sip debug" in the CLI, and full debug mode in logger.conf to /var/log/asterisk/full |
| 14:39.58 | phearless | jamincollins: I will try this |
| 14:40.13 | jamincollins | phearless: and under the line there is a SIP debug option, set it to FULL |
| 14:40.39 | jamincollins | then restart the PAP2T, it should spew logging information into the syslogs of the * server |
| 14:40.48 | *** join/#asterisk umay (n=chris@71-208-192-243.hlrn.qwest.net) |
| 14:41.36 | sergee | guys, what is the best way to configure a SIP peer which doesn't register itself on my * (insecure=port,invite) and it can use 192.168.10.0/24 (any of 255 ips) to place a call? - i don't want to have a separate entry for each IP address.. is there any smart approach to have this peer in 1 config entry? |
| 14:41.36 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 14:42.03 | phearless | I can't find "SIP debug option" in the PAP2 |
| 14:42.06 | phearless | jamincollins |
| 14:42.21 | sevard | [TK]D-Fender? |
| 14:42.22 | jamincollins | phearless: under the Line for that port |
| 14:42.41 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 14:42.43 | jamincollins | under SIP Settings in the Admin Advanced mode |
| 14:42.51 | [TK]D-Fender | phearless: Then your port/ip/networking is bad. how about firewalls? |
| 14:43.37 | [TK]D-Fender | sev You should never get a double dial-tone. Double indication might be possible if you are calling out of country and get addition regional progress tones etc at worst. |
| 14:43.40 | rpm | sevard: use host=dynamic |
| 14:43.41 | phearless | no FW |
| 14:43.42 | dalekurt | jamincollins: I re-compiled the SVN of zaptel-1.2.9 with make clean && CC=gcc-4.0 make && make install |
| 14:43.43 | jamincollins | something we probably should have asked earlier, but are they on the same lan segment and network range? |
| 14:43.52 | rpm | sevard: i mean sergee |
| 14:43.59 | jamincollins | dalekurt: and now if you "modprobe ztdummy"? |
| 14:44.00 | sevard | [TK]D-Fender: In the sipura 2002 I see lots of settings for ring cadence but none for signaling |
| 14:44.03 | sevard | rpm: heh |
| 14:44.22 | [TK]D-Fender | sergee: Why wouldn't you have it register? |
| 14:44.24 | dalekurt | jamicollins: ;( same thing happens |
| 14:44.36 | jamincollins | and in the dmesg output the same error? |
| 14:44.45 | phearless | <jamincollins> under SIP Settings in the Admin Advanced mode <-- nothing like this in admin mode / SIP |
| 14:44.46 | sergee | [TK]D-Fender: because it is DIDs provider :) |
| 14:45.01 | dalekurt | jamincollins: zaptel: version magic '2.6.15-1-486 486 gcc-4.1' should be '2.6.15-1-486 486 gcc-4.0' |
| 14:45.02 | *** join/#asterisk ajungem (n=ajungem@201.236.160.154) |
| 14:45.17 | jamincollins | phearless: under the Line tab, SIP Settings section on that tab, right hand side |
| 14:45.23 | sevard | [TK]D-Fender: Actually, I get it when I'm dialing local calls through my PRI |
| 14:45.25 | sergee | rpm: and how would i tell my asterisk about IP addresses, because i have other dynamic SIP hosts? |
| 14:45.36 | [TK]D-Fender | sergee: I don't understand those IP's however.... how can a SIP peer have a range of INTERNAL IP's? |
| 14:45.38 | jamincollins | dalekurt: it still thinks that gcc-4.1 was used or it's using the old module |
| 14:46.25 | dalekurt | jamincollins: hmm... any recommendations |
| 14:46.49 | *** join/#asterisk dennisharrison (n=dennisha@71-81-51-131.dhcp.slid.la.charter.com) |
| 14:46.49 | sergee | [TK]D-Fender: those subnet is just for example, real ips are 80.237.199.0/4 (i hope nobody will consider this as an advertising) |
| 14:46.55 | jamincollins | make sure the zaptel modules are removed from the /lib/modules/$kernel-version/misc directory |
| 14:47.03 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:47.09 | sergee | [TK]D-Fender: 80.237.199.0/24 |
| 14:47.14 | jamincollins | and any other sub directories under /lib/modules/$kernel-version that they are in |
| 14:47.27 | [TK]D-Fender | sergee: Also a PEER is for outgoing calls..... |
| 14:47.47 | jamincollins | try recompiling one last time and run a "depmod -a" before trying "modprobe ztdummy" |
| 14:47.54 | [TK]D-Fender | sergee: Maybe you should clarify exactly what services your are being offered and how you access them. |
| 14:48.02 | dalekurt | jamincollins: ok |
| 14:48.18 | sergee | so, is there any way to store info about anonymous (empty username and secret) USER basing on subnet? |
| 14:48.28 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:48.33 | jamincollins | if it still complains about gcc-4.1 vs gcc-4.0, you could try either removing gcc-4.1 or playing with Debian's alternatives settings to get gcc-4.0 as your preferred compiler |
| 14:48.37 | sergee | except creating an entry for each IP from that subnet? |
| 14:48.53 | sergee | [TK]D-Fender: yes sure |
| 14:49.21 | jmls | hey guys. fyi if I have usetls=no in my jabber.conf it has reduced the number of crashes from 4/5 per day down to nothing. Just for information in case someone else is having the problem |
| 14:49.38 | [TK]D-Fender | sergee: Why is it that you have a range of IP's to connect to? |
| 14:49.41 | phearless | jamincollins: http://img149.imageshack.us/img149/5900/screenshot1ph5.png I do not see "SIP debug" |
| 14:50.09 | [TK]D-Fender | sergee: And yes you would definatley have to set up an entry for each possibility because how else would * choose which one to use? |
| 14:50.10 | sergee | [TK]D-Fender: no, i have a range of ips user (DID provider) connects to me |
| 14:50.10 | phearless | jamincollins: but I got the remote syslog |
| 14:50.44 | jamincollins | phearless: that's the SIP tab, not the Line tab |
| 14:51.00 | jamincollins | phearless: try the Line tab, under its SIP settings |
| 14:51.02 | phearless | jamincollins: http://img149.imageshack.us/img149/5900/screenshot1ph5.png this is "SIP" |
| 14:51.06 | sevard | phearless: ewww gnome |
| 14:51.17 | phearless | ah ok |
| 14:51.19 | jamincollins | phearless: yes, and you want the Line tab |
| 14:51.22 | jamincollins | =) |
| 14:51.23 | dalekurt | lol |
| 14:51.24 | phearless | ok ! |
| 14:51.32 | dalekurt | gnome is the best |
| 14:51.35 | tzanger | this is a totally unrelated question, but does anyone know if BlueZ has an equivalent to /etc/hosts? |
| 14:51.38 | sergee | [TK]D-Fender: that is a right question :) how can asterisk selects info? only by exact username and/or exact host? |
| 14:51.41 | jamincollins | gnome has come a LONG way |
| 14:51.42 | phearless | ok I got it |
| 14:51.42 | sevard | dalekurt: xfce :) |
| 14:51.46 | phearless | jamincollins: ok I got it |
| 14:51.52 | sevard | xfce is the shieeeeet. |
| 14:51.53 | dalekurt | sevard: CLI |
| 14:51.58 | sevard | word. |
| 14:51.59 | sevard | TTY |
| 14:52.07 | [TK]D-Fender | sergee: Obviously * isn't psychic. you have to tell it where you are connecting to. So why is it a RANGE? |
| 14:52.07 | dalekurt | straight console. |
| 14:52.41 | sergee | [TK]D-Fender: let me quote myself: sergee[TK]D-Fender: no, i have a range of ips user (DID provider) connects to me |
| 14:52.53 | sevard | TELNET OVER A SERIAL CABLE TO MY 1 LINE DISPLAY |
| 14:52.53 | Nugget | telnet is eeeeeeevil! |
| 14:52.58 | phearless | [TK]D-Fender - jamincollins I got this in the syslog when I try to call from the phone |
| 14:53.01 | phearless | Sep 12 15:52:11 10.2.12.204 [0]Off Hook |
| 14:53.01 | phearless | Sep 12 15:52:32 10.2.12.204 [0]On Hook |
| 14:53.08 | phearless | just the on/off hook !! |
| 14:53.13 | sergee | [TK]D-Fender: i don't connect anywhere... |
| 14:53.40 | jamincollins | hmmm, it's not even trying to place the call |
| 14:53.56 | dennisharrison | hey everybody |
| 14:53.57 | jamincollins | and when the PAP2T is rebooted, anything in the syslog? |
| 14:54.05 | phearless | and when I call from somewhere else TO the phone I got : |
| 14:54.17 | phearless | Sep 12 15:53:04 10.2.12.204 CC:Clean Up |
| 14:54.17 | phearless | Sep 12 15:53:04 10.2.12.204 --- OBJ POOL STAT --- |
| 14:54.17 | phearless | Sep 12 15:53:04 10.2.12.204 OP:RTPRXB = 96 ( 96 192) |
| 14:54.19 | phearless | etc etc |
| 14:54.33 | phearless | some stuff like : |
| 14:54.35 | phearless | Sep 12 15:53:16 10.2.12.204 [0:0]RTP Rx 1st PKT @16384(3) |
| 14:54.35 | phearless | Sep 12 15:53:16 10.2.12.204 [0:0]DEC INIT 0 |
| 14:54.37 | jamincollins | before or after enabling the SIP debug on that line? |
| 14:54.45 | phearless | after enablign sip debug |
| 14:54.57 | dalekurt | !ahhhh... damn you zaptel |
| 14:55.01 | dennisharrison | I am trying to move over from expensive avaya equipment to hopefully less expensive asterisk for a new (small) call center |
| 14:55.01 | dennisharrison | don't have to bring it up for a month yet |
| 14:55.01 | dennisharrison | is this feasible? |
| 14:55.20 | jamincollins | dennisharrison: insufficient information to say |
| 14:55.27 | dalekurt | jamincollins: I'm gonna try the tar.gz from the FTP rather then the SVN |
| 14:55.27 | phearless | <jamincollins> and when the PAP2T is rebooted, anything in the syslog? <-- I will do this |
| 14:55.43 | jamincollins | dalekurt: tarball will faile |
| 14:55.53 | dennisharrison | jamincollins, well I have a lot of experience with telcom equipment and have several certifications from avaya |
| 14:55.54 | dalekurt | new release is out, that should fix the fail |
| 14:55.54 | jamincollins | dalekurt: err fail... it's incomplete |
| 14:56.07 | dalekurt | 1.2.9.1 is out |
| 14:56.08 | jamincollins | dalekurt: ahhh |
| 14:56.10 | dennisharrison | im downloading asterisk now |
| 14:56.25 | jamincollins | dennisharrison: it really depends on what your feature needs are from * |
| 14:56.35 | dalekurt | gonna give it a whirl... |
| 14:56.45 | dennisharrison | connect two together |
| 14:56.45 | dalekurt | Hey has anyone tried VoiceRoute? |
| 14:56.54 | jamincollins | dennisharrison: ie, inbound, outbound, queues, remote agents, skills, predictive dialing etc |
| 14:57.00 | phearless | [TK]D-Fender, jamincollins I got this when I boot the PAP2 : |
| 14:57.00 | phearless | http://paste-bin.com/426 |
| 14:57.00 | dennisharrison | to pick up 6 lines in canada |
| 14:57.06 | phearless | a loooooot of things |
| 14:57.10 | dennisharrison | from an existing call center in new york |
| 14:57.18 | dennisharrison | nothing too fancy |
| 14:57.42 | dennisharrison | just add them as extensions using sip hopefully and share the lines off the t |
| 14:58.46 | dennisharrison | I was at a conf recently and a speaker had gotten an asterisk box to connect to an avaya ip600 and use most of the features |
| 14:58.57 | dennisharrison | and was able to use less expensive sets also |
| 14:59.25 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 14:59.42 | jamincollins | dennisharrison: should be possible |
| 14:59.54 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:00.12 | dennisharrison | any place besides voipinfo I should be looking for information on this jamincollins ? |
| 15:00.17 | jamincollins | phearless: and does that pastebin include a call attempt? |
| 15:00.20 | dennisharrison | I would appreciate a point in the right direction ;p |
| 15:00.35 | phearless | jamincollins: no |
| 15:00.38 | phearless | jamincollins: 1sec.. |
| 15:00.45 | jamincollins | dennisharrison: voip-info.org has been my goto place for the information so far, well, that and this channel |
| 15:01.23 | jamincollins | dennisharrison: well, with the addition of also putting my entire house on an * system so I have to eat my own dog food |
| 15:01.41 | phearless | here is a call attempt , jamincollins |
| 15:01.42 | phearless | Sep 12 16:01:03 10.2.12.204 [0]Off Hook |
| 15:01.43 | phearless | Sep 12 16:01:26 10.2.12.204 [0]On Hook |
| 15:01.49 | phearless | <PROTECTED> |
| 15:01.54 | jamincollins | grr |
| 15:02.04 | jamincollins | and your dial plan is currently? |
| 15:02.07 | dennisharrison | jamincollins, haha! thanks |
| 15:02.20 | phearless | (2XXS0) |
| 15:02.23 | phearless | this one |
| 15:02.52 | *** join/#asterisk Kuto (n=kuto@125.60.241.24) |
| 15:02.55 | [TK]D-Fender | sergee: They connect to you means an incoming call. That means you just need to set "insecure=very", "allowguest=yes", and set a context in [general] in sip.conf |
| 15:02.59 | jamincollins | very odd, when you try to dial a 200 series extension does it automatically play the tone after the 3rd digit? |
| 15:03.19 | phearless | if I play 303 or 203 it is the same |
| 15:03.38 | Kuto | [TK]D-Fender: i got problem...does vicidial adaptable to posix? |
| 15:03.39 | jamincollins | but if you on dial 2 digits it waits? |
| 15:04.04 | [TK]D-Fender | Kuto: No idea, never used it personally. |
| 15:04.05 | jamincollins | s/on/only/ |
| 15:04.26 | Kuto | - Any Unix with Xwindows, Mac OS9/X or Win98/2k/XP operating system |
| 15:04.46 | phearless | jamincollins: same for 2 digits |
| 15:04.54 | Kuto | it never mentioned linux?? |
| 15:05.05 | jamincollins | immediate busy tone? |
| 15:05.10 | phearless | jamincollins: after the last digit, I wait 5sec, and I got the fast busy tone, 5 seconds after I got another tone more noisy |
| 15:05.42 | jamincollins | and when dialing 203 you still have to wait 5 seconds for the tone? |
| 15:05.50 | phearless | yes |
| 15:06.23 | dalekurt | jamincollins: Well that was pointless :( what was the other option.. and I can't remove gcc-4.1 too many deps. |
| 15:07.08 | dalekurt | I don't think this "CC=gcc-4.0 make" works |
| 15:07.15 | jamincollins | dalekurt: gcc --version I assume gives you 4.1? |
| 15:07.41 | dalekurt | yep |
| 15:07.45 | jamincollins | phearless: are you certain we are working with the correct line of the two? |
| 15:08.06 | jamincollins | dalekurt: ok, this is kludgy, but /should/ work |
| 15:08.33 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.252) |
| 15:08.37 | phearless | jamincollins: yes, and it detects the on/off hook |
| 15:08.39 | jamincollins | dalekurt: rm /usr/bin/gcc; ln -s /usr/bin/gcc-4.0 /usr/bin/gcc |
| 15:08.54 | *** join/#asterisk Inkubot (n=inkubot@200.74.182.45) |
| 15:08.57 | Inkubot | hi |
| 15:09.02 | Inkubot | how are you guys ? |
| 15:09.14 | *** join/#asterisk Andr3www (i=andr3www@HSE-Toronto-ppp295639.sympatico.ca) |
| 15:09.17 | Inkubot | i have a problem with a sip device |
| 15:09.18 | jamincollins | dalekurt: on Debian the /usr/bin/gcc is a symlink to one of the various versions |
| 15:09.36 | IronMan2000 | anyone knoe how I can make it hunt to another extenstion if one ext is in use or busy? I am needing to setup a hunt between 3 ext. |
| 15:09.37 | Inkubot | i can't register a device |
| 15:09.56 | Inkubot | i think it is a problem with the format of the authorization header |
| 15:10.04 | phearless | this great PAP2 will finish in the bin |
| 15:10.11 | Inkubot | theres no space after commas |
| 15:10.27 | jamincollins | phearless: It really sounds like it's not honoring the dialplan |
| 15:10.42 | phearless | I can't call TO the phone |
| 15:10.48 | phearless | it should not use the dialplan ! |
| 15:10.49 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:10.57 | phearless | it is more than a dialplan problem |
| 15:11.11 | dalekurt | jamincollins: THen should I recompile zaptel. |
| 15:11.18 | jamincollins | dalekurt: yes |
| 15:11.33 | jamincollins | dalekurt: and gcc --version should give you 4.0 |
| 15:11.37 | CtRiX | IronMan2000 use hints |
| 15:12.02 | dalekurt | jamincollins: it does... |
| 15:12.29 | jamincollins | phearless: you could try using a softphone to call the pap2 |
| 15:12.35 | *** join/#asterisk shodan (n=shodan@ip084.96-113-216.pppoe1.joliette.intermonde.net) |
| 15:12.44 | jamincollins | just to make sure it's answering for the sip id |
| 15:12.48 | dalekurt | jamincollins: YOU ARE THE MAN! |
| 15:12.58 | jamincollins | dalekurt: nah, I'm just a poser |
| 15:12.58 | Dr-Linux | anybody tried beta 1.4? |
| 15:13.09 | phearless | I called TO the PAP2 FROM another phone |
| 15:13.09 | phearless | jamincollins |
| 15:13.09 | dalekurt | :D |
| 15:13.15 | phearless | and it does not work |
| 15:13.31 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:13.34 | jamincollins | phearless: what do you see on the asterisk console for that call request? |
| 15:13.36 | dalekurt | jamincollins: You are the best man.. it working again... |
| 15:14.21 | jamincollins | dalekurt: just remember that gcc symlink is likely to get changed but gcc updates |
| 15:14.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.80.196) |
| 15:14.39 | jamincollins | you can read up on Debian's alternatives system if you want to truly override it |
| 15:14.51 | phearless | jamincollins: http://paste-bin.com/428 here is the log for a call TO the PAP2 |
| 15:15.49 | jamincollins | phearless: not the syslog, but the asterisk console's sip debug output... might also be in your asterisk full log |
| 15:16.02 | phearless | sorry |
| 15:16.22 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 15:17.21 | phearless | jamincollins: http://paste-bin.com/429 |
| 15:17.23 | phearless | here it is |
| 15:17.29 | phearless | 568 lines ! |
| 15:17.39 | IronMan2000 | can someone help me with my extenstions.conf? I have created a new macro, and don't know how to call it from my extentions. |
| 15:18.18 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e0355b8e998dc879) |
| 15:18.18 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:18.32 | brimstone | !? nat/digium/ ? |
| 15:18.34 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 15:18.56 | jamincollins | phearless: this PAP2 is IP 204, right? |
| 15:19.13 | phearless | yes ! |
| 15:19.16 | phearless | I called from the 203 |
| 15:20.00 | jamincollins | k... give me one sec... just making sure I'm looking at it from the right perspective |
| 15:20.35 | IronMan2000 | Can anyone help me setup a hunt group? I have the macro to do it, just don't know how to apply it. |
| 15:20.41 | wunderkin | k, sweet, i disabled odbc support for voicemail, off to the tracks :D |
| 15:21.09 | jamincollins | phearless: lines 184-205 indicate that the pap2t got the request from the asterisk and tried ringing the phone |
| 15:21.36 | _deg_ | Anybody knows how many threads asterisk can start? |
| 15:22.01 | jamincollins | phearless: line 310 indicates that asterisk tried to bridge the two together |
| 15:22.09 | _deg_ | I mean... ulimit give me this limitation, but has asterisk something that do it? |
| 15:23.21 | jamincollins | phearless: got a 2nd handset? so you can have one plugged into both ports on that device? |
| 15:23.53 | phearless | I just tried with an old handset and it has made a super weird sound |
| 15:24.04 | jamincollins | the other possibility is that it is tryign to ring the right port, but the ring style is wrong |
| 15:24.07 | phearless | the phone has make a BIIIPPKRRRRRBIIPP |
| 15:24.15 | jamincollins | for the ring? |
| 15:24.29 | phearless | the phone has ringed, and I heard this weird sound |
| 15:24.48 | phearless | so the good pint is the fact that it has ringed |
| 15:25.04 | phearless | I should find another handset |
| 15:25.48 | jamincollins | I had to change the Ring waveform: on a few of ours |
| 15:26.16 | jamincollins | it's under the Regional tab and I needed to switch it to trapezoid for several of our phones to ring |
| 15:26.37 | Cresl1n | brimstone: !!!!! |
| 15:26.44 | brimstone | omg hi Cresl1n! |
| 15:26.56 | jamincollins | now we just need hellfire |
| 15:27.10 | brimstone | we have file... |
| 15:28.18 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 15:28.55 | dalekurt | Oh I was asking if anyone tried VoiceRoute's Druid |
| 15:28.57 | phearless | damn |
| 15:29.02 | phearless | it worked.. |
| 15:29.03 | MacoStefX | re |
| 15:29.10 | phearless | I just switched to trapezoidal !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 15:29.19 | jamincollins | and it rings now? |
| 15:29.27 | phearless | yes, with the normal phone |
| 15:29.36 | phearless | how did you know this trick ? |
| 15:29.36 | jamincollins | yea, I beat my head on that one for hours |
| 15:29.48 | jamincollins | luckily one of my test sets had a lamp on it |
| 15:29.50 | phearless | and there is no docs at all |
| 15:29.55 | jamincollins | the phone wouldn't ring, but the lamp flashed |
| 15:30.00 | phearless | ah |
| 15:30.10 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 15:30.15 | jamincollins | so I spent a few hours toggling one setting after another |
| 15:30.43 | phearless | ok so no I can be called on this phone |
| 15:30.47 | phearless | but I still can't call |
| 15:31.01 | p1p | anyone else using a quintum tenor gateway? Imhaving some latency issues with it that I cant figure out |
| 15:31.03 | jamincollins | the second part stinks of dial plan |
| 15:31.22 | jamincollins | lets go very, very basic on it |
| 15:31.24 | grabeez | Anyone familiar with dialplan logic and queues want to take a look at this... it will crash asterisk in about 1 minutes using this http://pastebin.ca/167956 |
| 15:31.41 | phearless | I got (2XXS0) |
| 15:32.00 | jamincollins | let's try even simpler and I'm not sure if it's case sensitive or not |
| 15:32.06 | jamincollins | (2xx) |
| 15:32.17 | jamincollins | you'll likely have to wait after the third digit for some time |
| 15:32.18 | *** join/#asterisk eclark (n=eclark@pool-71-116-105-151.snfcca.dsl-w.verizon.net) |
| 15:32.29 | jamincollins | make sure you get dialtone when you pick the hanset up |
| 15:32.32 | Sasch | can help me to configure voicemail with asterisk ... |
| 15:32.37 | jamincollins | and pause between digits |
| 15:32.47 | jamincollins | just momentarily |
| 15:32.53 | phearless | okay |
| 15:33.14 | phearless | the default setting was (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 15:33.47 | *** join/#asterisk frigidzephyr (n=frigidze@c-71-207-216-231.hsd1.al.comcast.net) |
| 15:33.59 | jamincollins | ye, not sure on the *xx, but the next are for frequent x11 numbers, then operator, international operator, local, LD, and other |
| 15:35.25 | *** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) |
| 15:37.51 | frigidzephyr | okay im a noob to asterisk, this is my first install of it. everything appears to have built and installed correctly, but when i try # asterisk -vvvc to test run, i get a command not found |
| 15:38.46 | Egonis | is asterisk in your default path? |
| 15:38.47 | tamp4x | make config |
| 15:38.59 | Egonis | 'which asterisk' |
| 15:39.06 | frigidzephyr | egonis: not sure, im a linux noob too =D |
| 15:39.28 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 15:39.31 | Egonis | frigidzephyr: try typing 'which asterisk' at the prompt, which will tell you where the binary is located |
| 15:40.05 | frigidzephyr | egonis: says no asterisk in /usr/kerberos....... |
| 15:40.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:41.08 | *** join/#asterisk SlayR (n=gomez@AFontenayssB-153-1-60-202.w86-198.abo.wanadoo.fr) |
| 15:41.13 | SlayR | hi all |
| 15:41.39 | frigidzephyr | tamp4x: should i type make config in the /usr/src/asterisk* directory? |
| 15:41.39 | SlayR | i have some problems with call waiting :( |
| 15:42.24 | SlayR | i can't pickup another call when i'am already on line . |
| 15:42.31 | SlayR | So call waiting doesn't work :( |
| 15:42.57 | SlayR | i search on the web some reply but anyway |
| 15:43.07 | SlayR | i don't find any answer |
| 15:43.28 | SlayR | so if u can help me |
| 15:43.49 | frigidzephyr | SlayR: id help but im doing my first install of asterisk so im asking questions also |
| 15:43.50 | tamp4x | yes frigid |
| 15:44.02 | SlayR | lol |
| 15:44.08 | phearless | jamincollins: I am really confused |
| 15:44.20 | SlayR | maybe i can help |
| 15:44.23 | SlayR | frigidzephyr |
| 15:44.28 | jamincollins | phearless: no luck on the dial? |
| 15:44.29 | SlayR | what is your problem ? |
| 15:44.29 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 15:44.42 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:44.45 | frigidzephyr | Slayr: one second let me try what tamp4x suggested |
| 15:44.52 | SlayR | k |
| 15:44.57 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:44.57 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:45.10 | phearless | jamincollins: 303 or 203 are the same |
| 15:45.10 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 15:45.17 | smackus | is there a way to run chanspy or something similar to it from the cli, or manager? |
| 15:45.25 | phearless | jamincollins: I got a sort of busy dial |
| 15:45.27 | phearless | tone* |
| 15:45.46 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 15:45.46 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:45.49 | jamincollins | phearless: with the basic dialplan of '(2xx)' only the 203 should match |
| 15:46.07 | frigidzephyr | tamp4x: it output some stuff, no errors, i tried running asterisk -vvvc again , no luck, and which asterisk still says it doesnt exist |
| 15:46.17 | phearless | but for 203 or 303 I have to wait around 5sec after dialing to get something (the busy tone) |
| 15:46.22 | jamincollins | phearless: when you dial the 203, there is nothing in your asterisk full log for the attempt? |
| 15:46.32 | phearless | I am still using (2xx) |
| 15:46.41 | phearless | I check this. 1s |
| 15:47.13 | tamp4x | make asymlink to the binary from usr/local/bin |
| 15:47.39 | phearless | nothing in any logs when I dial 203 |
| 15:47.40 | frigidzephyr | tamp4x: how do i locate the binary? |
| 15:47.56 | frigidzephyr | tamp4x: would it just be named asterisk ? |
| 15:47.57 | phearless | just on/off hook on the PAP |
| 15:48.30 | jamincollins | phearless: the on/off hook for the PAP should be in the syslog, not the asterisk full log, right? |
| 15:50.02 | phearless | in the PAP2 syslog |
| 15:50.33 | frigidzephyr | locate 'asterisk' |
| 15:50.40 | frigidzephyr | lol wrong window |
| 15:51.57 | wunderkin | frigidzephyr, you need /usr/sbin in your path |
| 15:52.39 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 15:53.10 | jamincollins | phearless: can you screen shot all the settings for that line for me? |
| 15:53.31 | frigidzephyr | wunderkin: how do i do that? or how do i go about finding out how to do that =D |
| 15:53.38 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.43) |
| 15:54.03 | phearless | ok jamincollins |
| 15:54.12 | wunderkin | frigidzephyr, it depends on your shell |
| 15:54.33 | frigidzephyr | wunderkin: bash on fedora core 5 |
| 15:55.16 | wunderkin | frigidzephyr, http://www.troubleshooters.com/linux/prepostpath.htm |
| 15:55.30 | frigidzephyr | wunderkin: reading now =D thx |
| 15:55.36 | phearless | jamincollins: I DCC you all the settings |
| 15:55.48 | eKo1 | export PATH=/usr/sbin:$PATH |
| 15:56.22 | *** join/#asterisk BlackNTan (n=BlackNTa@12.175.120.250) |
| 15:56.26 | frigidzephyr | eKo1: thanks, i'll also read this site so i know what im doing |
| 15:58.03 | niter3 | hrm.. this is goofy.. |
| 15:58.13 | jamincollins | phearless: didn't come through quite right, it appears to be the summary page, not the line settings |
| 15:58.24 | phearless | click on line |
| 15:58.34 | phearless | it should work |
| 15:58.41 | phearless | (but not 100% sure) |
| 15:59.08 | jamincollins | phearless: very interesting |
| 15:59.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:59.38 | niter3 | seems like my on hold music isn't being played. Only one of the songs in that directory are being played but no more. Anyone have any idea why? |
| 15:59.48 | niter3 | I just get blank air after the first song |
| 16:00.00 | niter3 | permissions are right on the file |
| 16:00.02 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:00.37 | jamincollins | phearless: I'm comparing... it'll be a bit |
| 16:01.16 | momelod | can anyone point me to a good cdr web front end? im using mysql for my cdr collection.. |
| 16:01.35 | eKo1 | asterisk-stat is OK |
| 16:01.40 | eKo1 | check the wiki |
| 16:01.47 | frigidzephyr | wunderkin: okay added /usr/sbin to my path, now tried to run asterisk -vvvc still no luck |
| 16:02.07 | frigidzephyr | wunderkin: command not found, |
| 16:02.22 | wunderkin | frigidzephyr, how did you add it |
| 16:02.51 | frigidzephyr | wunderkin: export PATH=/usr/sbin:$PATH |
| 16:02.53 | *** join/#asterisk bmg505 (n=leon@c1-118-16.rndf.isadsl.co.za) |
| 16:02.54 | momelod | thanx |
| 16:03.15 | frigidzephyr | wunderkin: when i run which asterisk it seems to be looking in that path as well as others |
| 16:03.53 | frigidzephyr | wunderkin: but says thereis no asterisk |
| 16:04.07 | eKo1 | Is it there? |
| 16:04.51 | frigidzephyr | say: /usr/bin/which: no asterisk in (/usr/sbin:/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin) |
| 16:05.17 | frigidzephyr | should i try a make install again? |
| 16:05.31 | jamincollins | phearless: I don't see anything to account for the difference |
| 16:05.42 | eKo1 | frigidzephyr: is the asterisk binary in /usr/sbin? |
| 16:05.55 | jamincollins | phearless: has the pap2 been restarted since the dialplan change? |
| 16:05.56 | phearless | jamincollins: okay thank you |
| 16:06.05 | phearless | jamincollins: no |
| 16:06.27 | *** join/#asterisk angom (n=angom@red-corp-200.79.129.196.telnor.net) |
| 16:06.28 | jamincollins | I'm grasping at straws on this one, but it wouldn't hurt to try rebooting it |
| 16:06.34 | frigidzephyr | eKo1: that would be a file named asterisk right? |
| 16:06.48 | Dr-Linux | anybody tried beta 1.4? |
| 16:06.48 | frigidzephyr | eKo1: if so, then i don't see it, in there |
| 16:08.14 | wunderkin | frigidzephyr, then make install did not complete |
| 16:08.22 | mishehu | Dr-Linux: I'd be afraid to. |
| 16:08.41 | mishehu | beta isn't something I usually like to put on a production system |
| 16:08.45 | frigidzephyr | wunderkin: k so i just need to go back to the asterisk directory and do a make install again? |
| 16:09.29 | Dr-Linux | mishehu: i se |
| 16:09.33 | Dr-Linux | i see |
| 16:09.36 | wunderkin | Dr-Linux, i dont think it is in beta yet, but im trying out trunk.. haven't gotten far since i have to convert my stuff to odbc first |
| 16:09.53 | wunderkin | frigidzephyr, yes |
| 16:10.23 | Dr-Linux | wunderkin: i'm asking as maybe some guys tried it from trunk |
| 16:10.25 | wunderkin | i can do a read but no writes yet |
| 16:11.31 | syzygyBSD | I am having trouble transfering an IAX extension between two sip phones Here is the sip debug from the transfering phone. http://pastebin.ca/167982 |
| 16:11.48 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 16:12.08 | syzygyBSD | it gives a 500 internal server error message but I can't figure out the sequence of messages to understand why there is that error |
| 16:12.21 | wunderkin | testing out func_odbc, i don't even see the writes try to hit postgres |
| 16:14.43 | frigidzephyr | wunderkin: ran make install , tried asterisk -vvvc again still command not found =[ |
| 16:15.11 | frigidzephyr | wunderkin: is there something in output from the make install i can lookat to see what the issue is? |
| 16:15.12 | wunderkin | frigidzephyr, well it didnt install properly then, you will have to see why |
| 16:15.25 | wunderkin | make sure it even built first |
| 16:17.04 | *** join/#asterisk daysmen3 (n=primus@host86-143-6-176.range86-143.btcentralplus.com) |
| 16:17.06 | tamp4x | frigid : ln -s /usr/src/asterisk*/asterisk /usr/bin |
| 16:17.09 | tamp4x | =D |
| 16:17.48 | frigidzephyr | tamp4x: what did that do? |
| 16:18.10 | syzygyBSD | is there a way to limit which debugging is on which console, so I can debug one peer on one console, another peer on another, and IAX on a third? |
| 16:18.14 | frigidzephyr | tamp4x: make a link? |
| 16:19.00 | tamp4x | yes |
| 16:20.24 | frigidzephyr | tamp4x: ran it, tried asterisk -vvvc again , still no command |
| 16:21.04 | wunderkin | there is no reason at all to do that |
| 16:21.15 | frigidzephyr | wunderkin: kinda didntmake sense |
| 16:21.39 | wunderkin | frigidzephyr, you need to make sure that make completes at least and go from there |
| 16:22.01 | *** join/#asterisk lukketto (n=lukketto@host179-132.pool8257.interbusiness.it) |
| 16:22.07 | frigidzephyr | wunderkin: k, what can help me identify whether it completed or not, there is a few warning messages in the report about it exporting some files twice |
| 16:22.40 | wunderkin | well the error messages are what we will need to see |
| 16:23.20 | frigidzephyr | wunderkin: where can i paste those so you can look at them? |
| 16:23.30 | wunderkin | ~pb |
| 16:23.31 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 16:25.11 | jamincollins | alright, running * v1.2.12, zaptel v1.2.9, libpri v1.2.3 and it appears that I'm still getting mysterious instances of * dropping both sides of a TDM <-> SIP call that it's in the middle of |
| 16:25.29 | jamincollins | ideas on how to locate that trigger for the disconnect? |
| 16:28.17 | frigidzephyr | wunderkin: http://pastebin.ca/167995 |
| 16:29.36 | wunderkin | frigidzephyr, well those are warnings, and that looks like zaptel not asterisk |
| 16:30.59 | syzygyBSD | would there be any problems with me downgrading from 1.2.9 to 1.2.7? |
| 16:31.21 | frigidzephyr | wunderkin: i did the make install for zaptel and it seemed to go fine, in the /var/log/messages file it didnt give me any errors, detected my card and everything, modprobes went good |
| 16:31.32 | jamincollins | the output on the asterisk console or in the asterisk logs /is/ in the same order the events took place, right? |
| 16:33.10 | wunderkin | frigidzephyr, what you pasted is from zaptel, not asterisk! |
| 16:33.22 | wunderkin | you downloaded zaptel into the asterisk directory |
| 16:33.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:33.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:33.46 | frigidzephyr | wunderkin: lol crap |
| 16:33.54 | frigidzephyr | wunderkin: told ya im a noob |
| 16:34.01 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.12.1, Zaptel 1.2.9.1 released! (September 12, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx -=- http://pastebin.ca/ for showing others large amounts of text |
| 16:36.14 | eKo1 | frigidzephyr: You really need to go read some tutorials. |
| 16:36.26 | frigidzephyr | eKo1: thats what i followed was a tutorial =d |
| 16:36.34 | eclark | does anyone here have any experience setting up asterisk to use sunrocket to make calls? |
| 16:36.51 | frigidzephyr | eKo1: how could i have accidently downloaded zaptel into the asterisk-1.2 directory? |
| 16:36.52 | eKo1 | frigidzephyr: follow another one then |
| 16:37.05 | wunderkin | he is just lexdexic |
| 16:37.09 | frigidzephyr | eKo1: i need to know so i dont make that dumb mistake again lol =D |
| 16:38.19 | FuriousGeorge | anyone ever use a valecom loudspeaker intercom |
| 16:38.28 | FuriousGeorge | with * |
| 16:38.46 | FuriousGeorge | i believe its got an fxs interface of some sort |
| 16:40.28 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.109.236) |
| 16:40.52 | jamincollins | anyone here able to shed some more light on the 'resetinterval' parameter for zapata.conf? |
| 16:42.17 | [TK]D-Fender | jamincollins: That controls how often * may reset all the channels in a PRI IIRC. |
| 16:42.40 | jamincollins | [TK]D-Fender: any reason you know of for not using it? |
| 16:42.49 | wunderkin | yes, it is supposted to only reset channels not in use |
| 16:43.16 | jamincollins | I ask because it seems to be on by default and there is a thread on the mailing list that seems to indicate it /might/ be the cause of period PRI drops |
| 16:43.31 | jamincollins | s/period/periodic/ |
| 16:43.32 | wunderkin | anyone use func_odbc? any way to use a generic write statement? that is my only problem :/ |
| 16:43.39 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 16:44.03 | jamincollins | but I /thought/ it ignored channels that were in use |
| 16:44.26 | [TK]D-Fender | jamincollins: Have you checked with your telco to see if you are getting dropped frames, etc? That can do it as well. |
| 16:44.55 | jamincollins | [TK]D-Fender: the telco is mostly useless... and I've enabled intense PRI debugging... |
| 16:44.58 | wunderkin | maybe something else is wrong, since it is still not hitting the db |
| 16:45.03 | wunderkin | only for writes though |
| 16:45.11 | jamincollins | but it appears that * is initiating the disconnect when this happens |
| 16:45.21 | [TK]D-Fender | jamincollins: They have to be able to tell you about frame slips & flips though... this is basic stuff... |
| 16:45.45 | jamincollins | [TK]D-Fender: they couldn't tell me whether 911 calls were or were not connecting |
| 16:45.54 | [TK]D-Fender | sadjklsdfkhjlasldhfa |
| 16:46.13 | jamincollins | yea... that was about my response on the situation |
| 16:47.12 | wunderkin | hehe |
| 16:47.33 | jamincollins | you and syzygyBSD helped me isolate that one... along with beating me with the cluex4 |
| 16:50.43 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 16:50.45 | wunderkin | got it! yey! bitching on irc does help work out your problems :) who said that yesterday? |
| 16:50.56 | wunderkin | tzanger? :) |
| 16:51.01 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:51.08 | jamincollins | wunderkin: hasn't worked for me, yet... |
| 16:51.23 | tzanger | no |
| 16:51.24 | tzanger | that was royk |
| 16:51.27 | wunderkin | oh |
| 16:51.29 | tzanger | but about me |
| 16:51.30 | tzanger | :_) |
| 16:51.32 | tzanger | er :-) |
| 16:51.34 | wunderkin | :) |
| 16:51.52 | *** join/#asterisk adorah (n=admin@87.68.149.143.cable.012.net.il) |
| 16:51.53 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 16:52.33 | Egonis | When someone calls a SIP Extension which is free, it rings, what about when that extension is in use? can I implement a sound or playback() while dialing it to notify the user that that person is on the phone? |
| 16:52.33 | jamincollins | tzanger: you sure it wasn't royk about me? |
| 16:52.38 | Egonis | prior to vm |
| 16:52.56 | wunderkin | jamincollins, you have not been asterisk-blessed yet |
| 16:53.12 | jamincollins | sounds painful |
| 16:53.41 | tzanger | it is |
| 16:53.46 | tzanger | you know what that asterisk symbol looks like |
| 16:53.49 | tzanger | and you KNOW where it goes |
| 16:54.26 | jamincollins | now, why did I just get a mental image of a kidney stone??? |
| 16:54.35 | backblue | zaptel: disagrees about version of symbol struct_module |
| 16:54.38 | backblue | wtf? |
| 16:55.04 | jamincollins | compile against the wrong kernel-headers? |
| 16:55.12 | backblue | no, i have not |
| 16:55.23 | backblue | i only have one kernel and one kernel-headers |
| 16:55.52 | syzygyBSD | what am i getting blammed for? |
| 16:56.02 | jamincollins | blame? |
| 16:56.13 | jamincollins | oh... liberal use of a cluex4 |
| 16:56.21 | jamincollins | nothing to be worried about |
| 16:56.29 | jamincollins | I deserved it |
| 16:57.24 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:57.50 | CunningPike | Egonis: Use the same logic that you would use for invoking VoiceMail(), only replace the VoiceMail() command with something else, like PlayBack() or BackGround() |
| 16:58.22 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 16:59.49 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 17:01.54 | wunderkin | now i just need to find the clean way to use func_odbc for deletes, i get an error message but it works.. |
| 17:05.06 | [TK]D-Fender | Egonis: You can do just about anything you want with a call.... |
| 17:06.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.80.196) |
| 17:07.26 | Egonis | [TK]D-Fender: How would I implement something like that? |
| 17:08.13 | [TK]D-Fender | Egonis: After your dial command see what teh DIALSTATUS variable was set to and do whatever you want afterwards. |
| 17:09.12 | Egonis | [TK]D-Fender: Cool, I'll try that. thank you! |
| 17:09.58 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:10.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:11.41 | jamincollins | grrrr... resetinterval didn't fix the mysterious drop |
| 17:12.16 | *** join/#asterisk rado1 (n=rado1@xd141.sstar.com) |
| 17:12.19 | jamincollins | [TK]D-Fender: framing slips would show in the intense pri debug, right? |
| 17:12.46 | [TK]D-Fender | jamincollins: No idea. I would like to think so but have never been so directly involved in it before. |
| 17:13.00 | [TK]D-Fender | jamincollins: Is your card sharing interrupts with anything else? |
| 17:13.17 | rado1 | forgive me...brand new to asterisk. would it be fair to say asterisk compatible hardware = Zaptel compatible hardware? |
| 17:13.25 | jamincollins | nope, looked for that early on |
| 17:13.50 | jamincollins | 185: 97956805 IO-APIC-level wcte11xp |
| 17:16.14 | CunningPike | jamincollins: Framing slips will usually show on your console - often as HDLC errors |
| 17:16.49 | jamincollins | that's what I figured, haven't seen any in days and never during production hours |
| 17:17.19 | tzanger | anyone used those linksys wip300s? |
| 17:17.27 | jamincollins | yet * periodically tears down a call going through it for no apparent reason |
| 17:18.28 | jamincollins | intense pri debugging doesn't seem to shed any light on the trigger |
| 17:19.27 | Egonis | tzanger: I have |
| 17:19.34 | Egonis | tzanger: Neat phones, poor SIP implementation |
| 17:20.29 | tzanger | Egonis: yuck |
| 17:20.29 | mut | is there a way to connect a current call to another extension via cli or manager? |
| 17:20.31 | mut | w/o either end interaction |
| 17:20.33 | tzanger | I think you can do it in the mnager |
| 17:20.55 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 17:21.38 | tzanger | I love that... "and nuts... man they're just tree droppings... I mena the tree dunn want it..." |
| 17:21.43 | jamincollins | anyone know digium's support rates? |
| 17:21.51 | [TK]D-Fender | rado1: Thats too sparse a question. * can run on just about any system. What hardware are you looking to support? |
| 17:22.45 | CunningPike | tzanger: Have you noticed that they dropped the "Sap, that's tree sweat" bit? |
| 17:23.06 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-137-244.dyn.embarqhsd.net) |
| 17:23.21 | tzanger | yeah they changed it to "maple? that's just tree sap" |
| 17:23.46 | CunningPike | tzanger: Wonder why |
| 17:23.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:24.09 | tzanger | CunningPike: I haven't got a clue why, but you should tell me because I'm curious |
| 17:24.39 | CunningPike | tzanger: Me too! |
| 17:28.57 | *** part/#asterisk lukketto (n=lukketto@host179-132.pool8257.interbusiness.it) |
| 17:29.56 | harryvv | jamincollins I think its about 175.00 per hour |
| 17:30.26 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 17:32.15 | jamincollins | harryvv: it's included in the maintenance plan, correct? |
| 17:32.28 | *** join/#asterisk Qb3rt (n=jhgjkgui@58.68.252.216.dsl1.colba.net) |
| 17:32.49 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 17:33.05 | DrkShdw | Kerry_G: Not I, sorry |
| 17:33.39 | harryvv | I have no idea you need to give them a call |
| 17:33.42 | Qb3rt | my asterisk look like he is over calculating abandoned calls! my timeout for the queue is 120 seconds... do you guys think is because my timeout is to short? |
| 17:33.45 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 17:33.56 | BlepsoaF | hello all, how do you force a user to setup their voicemail box? |
| 17:34.07 | BlepsoaF | i know theres an option to do so, I just cant remember where |
| 17:34.23 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:34.44 | Druken | anyone know if sangoma has a 2 port pri with an echo cancler? |
| 17:34.58 | angler | BlepsoaF, theres an option in voicemail.conf |
| 17:35.05 | Cresl1n | Druken: digium does |
| 17:35.07 | jamincollins | Druken: afaik, all sangoma cards have echo cancel |
| 17:35.15 | BlepsoaF | angler: do you happen to remember that config? |
| 17:35.26 | Cresl1n | jamincollins: I don' think they all do |
| 17:35.29 | Cresl1n | as far as I know |
| 17:35.30 | mog | nope |
| 17:35.33 | BlepsoaF | cant find it on the wiki |
| 17:35.43 | mog | its a module just like digium cards |
| 17:35.45 | Cresl1n | I think only their 4 port cards |
| 17:35.47 | Druken | Cresl1n: yeah, but i get rapped for customs when buying from digium... hehe |
| 17:36.22 | angler | BlepsoaF, there is 'forcename' and 'forcegreetings' |
| 17:36.33 | tzanger | Druken: it's coming |
| 17:36.41 | tzanger | they're also coming witha single with hw echo can |
| 17:36.48 | bkw_ | Cresl1n: you at VON? |
| 17:36.51 | tzanger | that's right, digium's got the 207 too! |
| 17:36.54 | tzanger | I forgot all about that |
| 17:37.22 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 17:37.36 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 17:37.58 | Druken | tzanger: 207? |
| 17:37.58 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 17:38.14 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 17:38.25 | angler | Druken, Digium TE207P |
| 17:38.30 | jamincollins | hmmm, as others have already noted, I was wrong on the Sangomas... sorry about that |
| 17:39.03 | jamincollins | I'd be quite happy with the TE110 if I could solve this period drop |
| 17:39.23 | jamincollins | s/period/periodic/ |
| 17:39.44 | jamincollins | I keep doing that... grrr, mind thinks the full word, but fingers skip the last part |
| 17:40.18 | CunningPike | jamincollins: I kno wha yo mea |
| 17:41.10 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:41.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:41.11 | Druken | tzanger: do i REALLY need the echo can ? or is it one of those, ya don't know till you've tested the line? |
| 17:41.26 | *** join/#asterisk LoneShadow (n=duh@59.92.161.88) |
| 17:42.10 | CunningPike | Druken: My advice is to try without and add it in if you need |
| 17:42.14 | *** part/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 17:42.23 | tzanger | Druken: depends on the install |
| 17:42.37 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 17:42.38 | tzanger | I have installs with no echo can other than a 256 tap software (MG2) and it's great |
| 17:42.54 | Cresl1n | woohoo :-) |
| 17:42.56 | tzanger | and I hvae other installs with Sangomas with the Octasic and it's taken that level to squash it |
| 17:43.09 | tzanger | I'm getting me TE406 RMAd for a TE407 to try Digium's octasic echo can |
| 17:43.18 | tzanger | the 406 for me was just unworkable |
| 17:43.28 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 17:43.46 | tzanger | and even the sangoma a104d is giving me grief |
| 17:44.09 | tzanger | they had a software bug in their drivers that would cause the echo can to go batshit crazy, but they "fixed" that and now it just goes a little nuts now and again |
| 17:44.23 | tzanger | you can hear the echo can trip out but the other side is still intelligible, and a second or so later it goes away |
| 17:44.28 | tzanger | they're still working with me on that |
| 17:44.35 | *** join/#asterisk ramtha (n=tk@p5088B3AC.dip0.t-ipconnect.de) |
| 17:45.01 | tzanger | brodiem: I also have a hardware (shelf) echo can somehwere |
| 17:45.07 | tzanger | not tellabs but sonmeone else I cannot remember offhand |
| 17:45.10 | tzanger | I haven't hooked it up at all |
| 17:45.43 | brodiem | well just standaard hw echo can in the digium card even |
| 17:46.10 | BlepsoaF | thankS! |
| 17:46.30 | tzanger | brodiem: well their first hw echo can did *not* work well for me. |
| 17:46.30 | Druken | thanks tzanger |
| 17:46.35 | tzanger | their octasic should be great |
| 17:46.41 | tzanger | just waiting on it to find out |
| 17:46.48 | tzanger | Druken: not sure what I did ot help, but you're welcome :-) |
| 17:47.54 | brodiem | I wonder if their cancellation algorithms will make it into the zaptel driver at some point |
| 17:48.02 | brodiem | probably not |
| 17:48.10 | tzanger | brodiem: it's not easy stuff |
| 17:48.19 | tzanger | and there are lots of patented tricks to make them work better |
| 17:48.21 | *** join/#asterisk lsl23 (n=chatzill@141.214.234.28) |
| 17:48.30 | lsl23 | does anyone know if john from voipjet is online? |
| 17:48.33 | tzanger | the software echo cans are experimental, and may improve with time |
| 17:48.54 | Cresl1n | brodiem: and when you use a HW solution such as octasic, you don't use your own algorithm |
| 17:48.55 | tzanger | but you need to understand that the knowledge and experience necessary to really work on these is very hard to come by |
| 17:49.16 | tzanger | and those who have such experience and knowledge are usually pretty time-constrained |
| 17:49.22 | Cresl1n | so there aren't improvements to speak of that you can contribute back to zaptel |
| 17:49.40 | tzanger | and on top of that, it's open source, so the only time it really gets worked on is if one of those extremely bright guys has a particular itch to scratch |
| 17:50.21 | tzanger | I mean we had steve1/2, mark1/2/3, kris came out with KB1 which really REALLY improved mark2, and then I'm not sure who took KB1 and got us to MG2 |
| 17:50.24 | tzanger | and MG2 is the cat's ass |
| 17:50.32 | bkw_ | shhh |
| 17:50.34 | tzanger | at least as far as open-source, in-zaptel software echo cans go |
| 17:50.35 | bkw_ | how can you say bad things? |
| 17:50.56 | tzanger | but it is still a horrible echo canceller compared to what's available |
| 17:51.15 | tzanger | it does, however, work "well enough" that most people in typical situations can live with it |
| 17:51.19 | tzanger | which is no small feat in itself |
| 17:51.23 | *** join/#asterisk paryl (n=chatzill@www.admiralexpress.com) |
| 17:52.11 | Corydon-w | tzanger: if you want to pay for my graduate school, and I'll do a better echo can for my master's thesis |
| 17:52.24 | Cresl1n | heh |
| 17:52.32 | tzanger | Corydon-w: I haven't got money to eat |
| 17:52.32 | tzanger | i's po |
| 17:52.40 | Corydon-w | What about money to burn? ;-) |
| 17:53.29 | Juggie | http://www.google.com/tools/firefox/browsersync/index.html |
| 17:54.07 | paryl | i'm running into a problem where a call comes into the queue, the agent goes to transfer the call, and it disconnects the caller. in the log it shows "-- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.111" |
| 17:54.19 | paryl | but that phone can transfer non-queue calls |
| 17:54.29 | hmmhesays | bah, the opposite sex drives me insane |
| 17:54.47 | Corydon-w | Same sex is always good... ;-) |
| 17:54.49 | paryl | i just updated asterisk to 1.2.12.1, and this just started |
| 17:55.23 | tzanger | hmmhesays: a-fucking-men |
| 17:55.28 | tzanger | no wait, that's bkw_ :-) |
| 17:55.39 | hmmhesays | haha |
| 17:55.57 | tzanger | doh |
| 17:55.59 | tzanger | he didn't even see that |
| 17:56.00 | tzanger | dammit |
| 17:56.09 | *** join/#asterisk tlow (n=tlow@cypher.punk.net) |
| 17:56.55 | *** join/#asterisk Splat (n=Splat@220-253-104-57.TAS.netspace.net.au) |
| 17:58.36 | tzanger | hmm |
| 17:58.45 | tzanger | this polycom seems to not like the presence reply from asterisk |
| 17:59.12 | tzanger | * sends a SIP NOTIFY with a little xml in it |
| 17:59.17 | tzanger | the 501 sends back a 500 |
| 17:59.36 | paryl | tzanger: are you talking to me? |
| 17:59.39 | tzanger | no |
| 17:59.45 | paryl | haha, ok :) |
| 17:59.49 | tzanger | [TK]D-Fender: didn't you say that presence/notification was buggered in trunk? |
| 17:59.57 | sivana | tzanger: have you done line appearances? |
| 18:00.07 | tzanger | sivana: just did for this one guy |
| 18:00.12 | tzanger | it mostly works |
| 18:00.16 | sivana | cool |
| 18:00.20 | tzanger | I like the 601 screen, wow it's not just the same as the 501 |
| 18:00.31 | sivana | I have an implementation requirement coming up for 12-15 phones |
| 18:00.32 | tzanger | the font's a little screwy |
| 18:00.36 | tzanger | comic sans ms-kind |
| 18:00.59 | sivana | the client needs to decide on 501 or 7960G |
| 18:01.04 | sivana | sorry.. 601 |
| 18:01.33 | *** part/#asterisk tlow (n=tlow@cypher.punk.net) |
| 18:01.33 | tzanger | I know the 601's nice, but I"ve never touched a 7960 |
| 18:02.23 | Juggie | whats the fix in 1.2.12.1&1.2.9.1? |
| 18:02.28 | [TK]D-Fender | tzanger: as of Sunday, yes |
| 18:02.39 | tzanger | [TK]D-Fender: ok so I just leave it be then :-) |
| 18:02.44 | [TK]D-Fender | tzanger: 4x the res. |
| 18:03.06 | tzanger | the 601's got 4x the resolution of the 501? daym |
| 18:03.20 | [TK]D-Fender | sivana: What do you want from this phone you are planning on buying? |
| 18:03.31 | sivana | not sure exactly :) |
| 18:03.41 | tzanger | I'm unbelievably happy with polycom |
| 18:03.42 | sivana | line appearances |
| 18:03.55 | tzanger | I wish you could do more with the screen on the 403/501 |
| 18:04.29 | [TK]D-Fender | sivana: Seriously how many do you need? |
| 18:04.29 | sivana | I like Polycom's too... just to give you an idea on the client.. they want 1 x cat3, 3 x cat6, 2 x fiber runs to each workstation |
| 18:04.42 | sivana | 12-15 |
| 18:05.01 | [TK]D-Fender | sivana: 15 APPEARANCES?! |
| 18:05.05 | sivana | no |
| 18:05.08 | sivana | only 6 |
| 18:05.09 | sivana | max |
| 18:05.15 | [TK]D-Fender | sivana: Why so many? |
| 18:05.23 | sivana | cuz he can I guess :) |
| 18:05.27 | grabeez | [TK]D-Fender, can you look at this. It causes asterisk to crash http://pastebin.ca/168094 |
| 18:05.41 | [TK]D-Fender | sivana: I never met a business user that needed more than 2 |
| 18:05.45 | grabeez | It works for a minute or so |
| 18:05.56 | paryl | has anyone ran into the same transfer issue i have? |
| 18:05.59 | sivana | he just wants to have enough for anything he's doing.. marketing company |
| 18:06.16 | sivana | lots of data transfer... ever hear of Apple's fiber channel? |
| 18:06.22 | sivana | Raid fiber channel or something |
| 18:06.23 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:06.43 | paryl | polycom 501 phones, attended transfer, when transfer is hit the second time i get "Got SIP response 500 "Internal Server Error"" |
| 18:06.58 | [TK]D-Fender | sivana: just to make sure... appearances != calls. you know that right? |
| 18:07.03 | sivana | yrd |
| 18:07.04 | sivana | yes |
| 18:07.24 | sivana | they will have 6 pots line |
| 18:07.28 | [TK]D-Fender | sivana: So why so many identities? You'd need attendant modules for that many... |
| 18:07.41 | [TK]D-Fender | sivana: Yeah, but that doesn't necessarily reflect on the phone at all... |
| 18:07.51 | sivana | and I think he's thinking of having line appearances for lines... not extensions |
| 18:08.12 | [TK]D-Fender | sivana: doesn't work that way... this isn't a key system.... |
| 18:08.24 | sivana | right.. just realized now |
| 18:08.47 | [TK]D-Fender | grabeez: Nice start there.... |
| 18:09.27 | [TK]D-Fender | grabeez: I don;t know anything about "groupcount" though.... |
| 18:09.41 | grabeez | It works, as intended... I have in some other contexts it adding to the group, and it worked, but then asterisk would just crash. Someone just told me not to use local because of deadlock issues |
| 18:09.44 | [TK]D-Fender | Besides, wait just a little and the IP650 will be out :D |
| 18:10.12 | [TK]D-Fender | grabeez: OH... well no idea baout that... |
| 18:10.15 | sivana | so, thinking this through, he can't have appearances on lines, right |
| 18:10.25 | [TK]D-Fender | grabeez: But excellent dialplan coding on your part. |
| 18:10.48 | [TK]D-Fender | sivana: He CAN for the sake fo visibility but will have no impact on functionality. |
| 18:10.59 | sivana | right |
| 18:11.08 | [TK]D-Fender | sivana: Tehn again if he wants to know there are tons of ways of doing that. |
| 18:11.10 | grabeez | [TK]D-Fender, thanks! I will play with it more. I noticed this bug http://bugs.digium.com/view.php?id=7887 shooting everywhere so it may have been the culprit |
| 18:11.21 | sivana | all he wants is visual |
| 18:11.31 | grabeez | now fixed in 1.2.12.1 |
| 18:11.48 | [TK]D-Fender | sivana: Make a web script to poll the lines on demand then. |
| 18:11.57 | [TK]D-Fender | sivana: and save on the phones... |
| 18:12.05 | sivana | I could just use flash operator |
| 18:12.18 | sivana | he still needs 12-15 extensions :) |
| 18:12.48 | [TK]D-Fender | sivana: True, so get a bunch of lower model phones then. |
| 18:13.13 | sivana | anyone use the Polycom IP4000? |
| 18:13.39 | [TK]D-Fender | sivana: If they want a budget PoE get the IP 430. if you don't need PoE get the IP 501. If you don't need speakerphon the IP 301 (if you need PoE forget this and jsut get the IP 430) |
| 18:13.47 | [TK]D-Fender | sivana: One of my clients has. |
| 18:13.50 | sivana | we'll need POE |
| 18:14.22 | [TK]D-Fender | sivana: IP 430 is a great general purpose phone then. Then get an IP 650 or 601 with attendant modules for the receptionist if any. |
| 18:14.43 | sivana | thanks for the input... I'll run these scenarios past him |
| 18:15.21 | sivana | is there GB PoE swithces? |
| 18:15.27 | [TK]D-Fender | sivana: Very economical. Saves 100$/seat |
| 18:15.48 | [TK]D-Fender | sivana: I've never heard of one that does both simultaneously. its either/or per-port |
| 18:16.06 | sivana | Gigabit PoE |
| 18:16.47 | [TK]D-Fender | sivana: Could be there are options now.... go look I guess... |
| 18:17.02 | sivana | actually.. that's not needed.. voice is separate lan |
| 18:17.20 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 18:17.27 | [TK]D-Fender | sivana: Screw Cat3 BTW. complete waste |
| 18:17.41 | sivana | I would... but he wants it "just in case" :) |
| 18:17.46 | [TK]D-Fender | sivana: You can plug an rj11 into an RJ45 anyways.... |
| 18:17.52 | [TK]D-Fender | do taht instead. |
| 18:18.02 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 18:18.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:18.18 | [TK]D-Fender | do NOT waste money on cat3 when cat5 is backwards compatible |
| 18:18.31 | sivana | ya, I think he might be thinking of keeping the existing ca3 |
| 18:18.33 | sivana | cat3 |
| 18:18.53 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 18:19.33 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 18:20.09 | [TK]D-Fender | sivana: I got to rip eveything outta this place before we moved in.... |
| 18:20.19 | [TK]D-Fender | sivana: I'm 2 LANS of Cat5E :) |
| 18:20.51 | [TK]D-Fender | IP 650 = creamy goodness...... |
| 18:20.59 | sivana | yea, I think we're just upgrading the existing office and maybe adding a few workstations |
| 18:21.14 | sivana | he's got cat3, cat5 to each workstation now |
| 18:21.23 | sivana | running gigabit over cat5 |
| 18:21.33 | sivana | cat5e actually |
| 18:21.40 | *** join/#asterisk livesNbox (n=chadkous@165.236.120.14) |
| 18:21.57 | livesNbox | has "autopause" been added to queues.conf in the latest asterisk build? It was in SVN a few months ago... |
| 18:22.38 | livesNbox | or (better) how can you tell when certain features are included in a build ? |
| 18:22.41 | sivana | should be fun, I haven't done a site visit/inventory yet.. still just discussions |
| 18:23.14 | sivana | livesNbox: you talking about between trunk and release? |
| 18:23.47 | sivana | tzanger: no word yet? |
| 18:24.02 | wunderkin | livesNbox, the options that are available should be in the /usr/src/asterisk/configs/blah.conf.sample file |
| 18:24.18 | *** part/#asterisk rado1 (n=rado1@xd141.sstar.com) |
| 18:24.53 | *** join/#asterisk pagec (n=pagec@64-252-108-252.adsl.snet.net) |
| 18:24.55 | livesNbox | wunderkin: thanks |
| 18:25.34 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:25.36 | pagec | if i wanted to have say 10 phones pick up, and i don't know if they are going to be on line, how do i do that? using dial(sip/phone1&sip/phone2) fails if phone1 isn't registered |
| 18:25.42 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:26.04 | Druken | pagec use a queue ? |
| 18:27.49 | CunningPike | pagec: It should work anyway - how does it fail? What error messages are you getting? |
| 18:28.37 | pagec | CunningPike: unable to route to sip/sipclient |
| 18:28.46 | pagec | Druken: any good tutorials on queues? |
| 18:29.02 | CunningPike | pagec: Paste your dial statement |
| 18:29.55 | Druken | pagec: wiki ? http://www.voip-info.org/ |
| 18:30.50 | sevard | Does anyone know where to access the do-not-call list? |
| 18:31.20 | wunderkin | sevard, http://telemarketing.donotcall.gov |
| 18:31.31 | wunderkin | dont ask why i know.. *sob* |
| 18:31.51 | sevard | I know where to register but what about in compliance with it |
| 18:32.08 | wunderkin | well, there are links off of the page for faqs |
| 18:32.16 | sevard | ahh, i see. misread the page. |
| 18:32.24 | Druken | hehehe i read that at first as fags |
| 18:32.33 | livesNbox | wunderkin: look at this please... http://www.asterisk.org/doxygen/Config_qu.html |
| 18:32.43 | Druken | :) |
| 18:32.50 | sevard | my boss just decided to put some people on the phones |
| 18:32.52 | livesNbox | there are a lot of options there that aren't yet showing in the latest asterisk build.. |
| 18:32.53 | sevard | to start making calls out |
| 18:32.55 | sevard | and i was like |
| 18:32.56 | sevard | NOOOOOOOOOOOOOOO |
| 18:33.08 | sevard | there are regulations man, there are rules, y ou can't just fucking call people. |
| 18:33.15 | wunderkin | livesNbox, .... where did i tell you to look? |
| 18:33.19 | livesNbox | But that was buidl back in August -- so I'm trying to get my head around when things are available. |
| 18:34.36 | wunderkin | im not sure if doxygen is always up-to-date and that is probably from trunk |
| 18:34.55 | livesNbox | is the latest asterisk an older branch or something ? |
| 18:37.18 | *** join/#asterisk ToyMan (n=stuq@74-32-63-182.dsl1.mdl.ny.frontiernet.net) |
| 18:43.45 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 18:46.10 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 18:46.55 | *** join/#asterisk Maxxed (n=user@mattcomfg.com) |
| 18:47.01 | Maxxed | yo fellas :) |
| 18:47.04 | Maxxed | i gota quickie |
| 18:47.18 | Maxxed | fugin asterisk keeps answering on a channel that i dont want it to answer on |
| 18:47.48 | Maxxed | [channels] |
| 18:47.48 | Maxxed | language=en |
| 18:47.48 | Maxxed | context=default |
| 18:47.48 | Maxxed | signalling=fxs_ls |
| 18:47.48 | Maxxed | usecallerid=yes |
| 18:47.48 | Maxxed | echocancel=yes |
| 18:47.50 | Maxxed | echocancelwhenbridged=yes |
| 18:47.52 | Maxxed | channel => 1 |
| 18:47.54 | Maxxed | language=en |
| 18:47.56 | Maxxed | signalling=fxs_ls |
| 18:47.58 | Maxxed | usecallerid=yes |
| 18:48.00 | Maxxed | echocancel=yes |
| 18:48.02 | Maxxed | echocancelwhenbridged=yes |
| 18:48.04 | Maxxed | channel => 4 |
| 18:48.06 | Maxxed | no context for channel 4, yet it still answers default |
| 18:48.12 | Maxxed | double you tee eph? |
| 18:48.50 | Maxxed | ok to answer on channel 1, defualt call menue, but dont pick up on channel 4 dag'mabit |
| 18:49.03 | Maxxed | where am i screwin this up? |
| 18:49.05 | CunningPike | Maxxed: Pastebin, please |
| 18:49.07 | CunningPike | ~pb |
| 18:49.08 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 18:49.15 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 18:49.29 | Maxxed | rightyo' |
| 18:49.36 | Maxxed | that didnt seem like much a flood to me |
| 18:49.41 | Maxxed | sry folks ;\ |
| 18:50.10 | CunningPike | 14 lines is enough to float Noah's Ark in a busy channel |
| 18:50.38 | Maxxed | um, yeaaaah ok |
| 18:50.41 | Maxxed | heh |
| 18:50.59 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:51.06 | Druken | and to answer your question, put a diffrent context before the channel => 4 |
| 18:51.15 | Maxxed | i tried that |
| 18:51.19 | Maxxed | i set one as dummy |
| 18:51.23 | Maxxed | same mess? |
| 18:51.26 | Maxxed | maybe i forgot to reload.. |
| 18:51.30 | Maxxed | erum, brb ;) |
| 18:51.31 | Druken | probably... |
| 18:51.36 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 18:52.39 | Maxxed | ah |
| 18:52.49 | flujan | hi all. |
| 18:52.52 | Maxxed | i need to put somthing in the dummy context or it failes to defualt |
| 18:53.20 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 18:53.20 | Maxxed | no s,1 do it then goes to default s,1 |
| 18:53.34 | Maxxed | what should i put in there? for the dummy s,1 conext |
| 18:53.36 | Maxxed | hangup? |
| 18:53.58 | Druken | can put whatever you want... could put in a wait(60) |
| 18:54.05 | Druken | just don't answer |
| 18:54.31 | Maxxed | ok |
| 18:54.38 | Maxxed | so im thinkin hangup should be ok? |
| 18:54.44 | Druken | should |
| 18:54.48 | Maxxed | il give it a shot |
| 18:54.49 | *** join/#asterisk watchy (n=watchy@office2.gwhsi.com) |
| 18:54.52 | Druken | why don't you want asterisk to answer? |
| 18:55.06 | MacoStefX | re |
| 18:55.06 | watchy | anyone here ever setup a Linksys WIP300? |
| 18:55.08 | Maxxed | well its a fax line that i want to use for outbound call fowarding |
| 18:55.22 | Maxxed | inbound, the fax needs to pick it up, outbound who cares |
| 18:55.28 | Druken | watchy: yep.. was real fun.. hehe |
| 18:55.40 | watchy | the linksys dials internal extensions fine but wont dial anything else |
| 18:55.44 | watchy | mind if i message ou druken? |
| 18:56.00 | Maxxed | yep that did it |
| 18:56.01 | Druken | it's a dialplan issue then |
| 18:56.02 | Maxxed | sweet |
| 18:56.04 | flujan | http://pastebin.ca/168137 please take a look |
| 18:56.04 | watchy | i can dial 9xxxxxxx just fine on any phone |
| 18:56.09 | watchy | but not my linksys |
| 18:56.10 | Maxxed | Druken: thanks :) |
| 18:56.12 | flujan | I want the following dialplan to work |
| 18:56.17 | flujan | But I am having no sucess.. :( |
| 18:56.20 | watchy | druken: wheres the dialplan for the linksys |
| 18:56.31 | Druken | not on the linksys, in asterisk |
| 18:56.35 | watchy | hmm |
| 18:56.36 | Maxxed | now i think im having an issue with signaling on that channel :p |
| 18:56.42 | Maxxed | yay, trial and error |
| 18:56.43 | watchy | whys it work for every other phone then? |
| 18:57.06 | watchy | exten => _9NXXXXXX,1,Dial,Zap/g1/${EXTEN:1} |
| 18:57.15 | watchy | that should work for the linksys shouldnt it |
| 18:57.36 | niter3 | hey guys I want to set my dtmf for one sip user to avt |
| 18:57.37 | Druken | should.. make sure the linksys is in the proper context.... |
| 18:57.48 | watchy | hmm |
| 18:58.15 | Druken | niter3: so set dtmfmode = in the users block in sip.conf |
| 18:58.16 | niter3 | how can I accomplish this.. i will just add dtmf= to one user |
| 18:58.38 | niter3 | dtmfmode=avt so like this? |
| 18:59.07 | Druken | is avt is a proper mode... |
| 18:59.07 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 18:59.37 | niter3 | ? |
| 18:59.47 | *** join/#asterisk sbma44 (n=tomlee@dsl092-174-002.wdc2.dsl.speakeasy.net) |
| 18:59.55 | watchy | druken: when i dial i just hear silence its strange |
| 19:00.02 | watchy | think i should goto the .7 firmware? |
| 19:00.16 | Maxxed | is there a way to adjust how long asterisk sits on the dialtone before sending the number (dtmf tones) |
| 19:00.23 | Maxxed | this line seems slow to open |
| 19:00.28 | Druken | i upgraded mine... fixed alot of problems |
| 19:00.47 | watchy | i dunno why i get silence when i dial its strange as hell |
| 19:00.50 | Druken | Maxxed: should be in your zapata.conf |
| 19:00.51 | Maxxed | so asterisk dials 123-456-7890 , but only 3-456-7890 get thru |
| 19:00.53 | *** join/#asterisk clive- (n=pirch@dsl-145-56-115.telkomadsl.co.za) |
| 19:00.56 | sbma44 | hi folks. I was in here last friday, looking for help on a problem I was having with inbound calls from broadvoice. the number rings and the dialplan executes in the asterisk console, but I just hear ringing until broadvoice's voicemail system picks up. |
| 19:01.00 | watchy | do you connect it over usb to upgrade it? |
| 19:01.04 | Maxxed | Druken: thanks again :) |
| 19:01.06 | watchy | or you do it over wifi? |
| 19:01.06 | sbma44 | here's the server-side SIP debug output: http://pastebin.ca/168140 |
| 19:01.17 | Druken | watchy: i upgraded over the wirless |
| 19:01.28 | sbma44 | at that time it was suggested that broadvoice wasn't getting a critical SIP ACK from me |
| 19:01.43 | watchy | the phone keeps wigging out on me |
| 19:01.48 | watchy | i wonder if its my AP |
| 19:02.01 | sbma44 | and that I ought to try to get an xlite softphone working with it, which I have |
| 19:02.07 | *** join/#asterisk mishkiz (n=janusmis@zeus.corsidian.com.br) |
| 19:02.10 | sbma44 | here's a successful xlite SIP debug: http://pastebin.ca/168135 |
| 19:02.39 | sbma44 | I'm not sure what the difference is. if anyone could have a look, or let me know what additional info I should supply, I would greatly appreciate the help. |
| 19:03.25 | watchy | Druken: you actually use this phone in production? |
| 19:03.38 | mishkiz | hello all...im here burning my brain thinking what can i do with asterisk manager...im thinking to do a app to monitoring the extensions...like "flash operator pannel".... |
| 19:04.32 | mishkiz | there is any API or OCX to use with VB6 ? |
| 19:04.57 | benjk | OCX? |
| 19:05.03 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 19:05.06 | benjk | whazzat? |
| 19:05.12 | mishkiz | i dont know...rsrsrs |
| 19:05.57 | mishkiz | i just want to developer something to asterisk using asterisk manager |
| 19:06.20 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.252) |
| 19:06.25 | benjk | asterisk manager sucks ballz |
| 19:06.50 | IronMan2000 | ballz benjk? |
| 19:07.07 | benjk | can never seem to handle the load if you use it for anything more heavy than a SOHO setup |
| 19:07.26 | Juggie | all the more reason to use ast man proxy |
| 19:08.01 | IronMan2000 | Does anyone know how I can forward a call to another extention if the ext is busy? Kind of like a small hunt |
| 19:08.15 | benjk | and the protocol is more of these million monkeys on typewriters accidents than design |
| 19:08.37 | niter3 | hrm.. the dtmfmode=avt is still not working on my sip trunk... |
| 19:08.38 | niter3 | urgg |
| 19:08.48 | niter3 | is there something in xten that needs to be adjustd?? |
| 19:09.31 | benjk | IronMan, use a macro |
| 19:10.19 | IronMan2000 | yea, I have a macro and added it to my extension.conf, but don't know how to set the macro in my etc. |
| 19:10.21 | benjk | the macro could dial, then check dialstatus and if busy, it could try an alternative destination |
| 19:10.42 | benjk | or it could first check chanisavail before dialing, same effect |
| 19:12.32 | IronMan2000 | once you have created the macro, how do you have your extensions make use of this macro? |
| 19:13.28 | benjk | exten => foo,1,Macro(MyFooMacro, ${param1}, ${param2}, ... ) |
| 19:14.51 | benjk | btw, macro is actually a misnomer |
| 19:15.06 | benjk | its more like a procedure call |
| 19:15.09 | IronMan2000 | if my exten already has a Macro, would it be something like: exten => foo,1,Macro(MyExistingMacro, MyFooMacro, ${param2}, .. |
| 19:15.33 | benjk | lets say you macro is defined like this ... |
| 19:15.41 | benjk | [macro-FooBar] |
| 19:15.59 | benjk | then you call this "macro" from anywhere like so ... |
| 19:16.27 | IronMan2000 | what if you need to call more than one macro - can you gice me an example. |
| 19:16.28 | benjk | exten => blob, 1, Macro(FooBar, baz) |
| 19:16.47 | benjk | where baz would be some parameter, assumimng the "macro" has a single parameter |
| 19:17.09 | benjk | exten => blob, 1, Macro(FooBar, baz) |
| 19:17.16 | benjk | exten => blob, 2, Macro(FooFoo, bar, baz) |
| 19:17.34 | *** part/#asterisk mishkiz (n=janusmis@zeus.corsidian.com.br) |
| 19:17.45 | benjk | calling first macro Foobar, then calling macro FooFoo |
| 19:18.18 | Maxxed | Druken: that issue with letting the channel open before dialing, i agree with you, it looks like its in the zapata.conf but what exactly i have no idea |
| 19:18.44 | Maxxed | wink? |
| 19:19.51 | watchy | man i cant get this wip300 to uograde son of a bitch |
| 19:19.53 | *** join/#asterisk hyperthread (n=janusmis@zeus.corsidian.com.br) |
| 19:20.29 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-b492ac934da1f587) |
| 19:22.41 | sbma44 | so, no ideas on why inbound sip from my termination vendor fails, despite my being able to register with a sip softphone from outside the network? |
| 19:24.39 | *** part/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:24.40 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.68.17) |
| 19:24.47 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:26.56 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 19:30.51 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 19:31.11 | *** join/#asterisk ambriento (n=ambrient@200.192.160.100) |
| 19:31.48 | watchy | man this wip300 is a piece of shit |
| 19:33.20 | frigidzephyr | okay trying to make install asterisk, the process stops with this error /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 |
| 19:33.37 | *** join/#asterisk skraelings001 (n=skraelin@201.230.111.148) |
| 19:33.49 | frigidzephyr | any idea what all that means? im a noob here |
| 19:34.13 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.68.17) |
| 19:35.08 | myiagy | you need libssl-dev |
| 19:35.33 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.68.17) |
| 19:35.41 | *** join/#asterisk Mugatu_ (n=mugatu@unaffiliated/Mugatu) |
| 19:38.20 | frigidzephyr | ahh |
| 19:40.33 | *** part/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 19:43.24 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 19:48.21 | wunderkin | is dcskinner here? |
| 19:49.05 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 19:49.37 | *** join/#asterisk dsoTm (n=dsoTm@mail.cobalt-it.com) |
| 19:49.59 | watchy | hey druken you there? |
| 19:51.51 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:52.07 | *** part/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:52.38 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:53.13 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:54.50 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 19:55.41 | |Vulture| | Anyone here ever install a PRI and inbound calling works, but outbound gives "Channel 0/1, span 1 got hangup request" upon dialing? |
| 19:55.56 | sivana | check your callerid name |
| 19:56.07 | |Vulture| | sivana: that should not matter should it? |
| 19:56.08 | sivana | I had to blank mine out |
| 19:56.11 | eKo1 | |Vulture|: many many times |
| 19:56.12 | |Vulture| | really... |
| 19:56.18 | frigidzephyr | im installing bison, before i build asterisk, on the downloads page it says (1.0.X only ) does that mean I should not install bison-devel 2.1-1.2.1 ? |
| 19:56.31 | sivana | you can probably send number, but not name |
| 19:56.32 | eKo1 | What does the callerid name have to do with that? That makes no sense. |
| 19:56.39 | |Vulture| | wow I have done 7 PRI installs and never had a system not place a test without CID Name/# |
| 19:56.44 | sivana | my CLEC rejects CID.name |
| 19:56.49 | *** part/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:56.49 | |Vulture| | okay let me try |
| 19:56.56 | |Vulture| | mostly they just null the data if it isn't sent |
| 19:57.00 | sivana | it's just a suggestion, not saying that's the cure |
| 19:57.16 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 19:57.19 | eKo1 | That is one strange suggestion. |
| 19:57.19 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 19:57.27 | sivana | worked for me |
| 19:57.29 | sivana | :P |
| 19:57.40 | rollergrrl | pri debug it |
| 19:57.54 | sivana | |Vulture|: if you have other pri's with them, then it's probably not that |
| 19:58.03 | |Vulture| | sivana: no this is my first with Qwest |
| 19:58.11 | eKo1 | All my calls have a CID name but only _some_ of them suffer that hangup request |
| 19:58.16 | |Vulture| | we have XO/Broadwing/Nuvox/Xspedius |
| 19:58.26 | sivana | eKo1: first off, it depends on the CLEC or PRI provider |
| 19:58.30 | dsoTm | hey all. is there a console command to determine how installed g729 licenses are being used? my math, and *'s math seem to differ |
| 19:58.39 | sivana | mine rejects anything but a blank cid name |
| 19:59.19 | eKo1 | I have a more complicated setup so I can't really say much. |
| 19:59.23 | |Vulture| | well this is my first VoIP PRI... |
| 19:59.38 | |Vulture| | going through a Cisco 1760 |
| 19:59.51 | [TK]D-Fender | VoIP PRI... interesting term.... |
| 19:59.58 | |Vulture| | [TK]D-Fender: no joke |
| 20:00.03 | sivana | |Vulture|: virtual pri? |
| 20:00.15 | |Vulture| | I am guessing so.. wasnt anything about it in the contract |
| 20:00.19 | [TK]D-Fender | 23 SIP channels originating from "who cares" :) |
| 20:00.26 | eKo1 | hehe |
| 20:00.35 | [TK]D-Fender | Its just SIP! |
| 20:00.36 | |Vulture| | but for Daytona, FL a 7 channel PRI with 768 bandwidth for $500... not bad at all |
| 20:01.06 | eKo1 | oh, that is a channelized PRI |
| 20:01.27 | [TK]D-Fender | eKo1: Just mixed mode T!. |
| 20:01.29 | [TK]D-Fender | T1 |
| 20:01.43 | |Vulture| | yea its basically a SIP channel bank |
| 20:02.07 | harryvv | |Vulture| who is the t1 provider |
| 20:02.17 | |Vulture| | setting the CID Name to null now and trying that.. didn't like a value |
| 20:02.21 | |Vulture| | Qwest |
| 20:02.25 | harryvv | okay |
| 20:02.43 | |Vulture| | 1,Set(CALLERID(name)=) |
| 20:02.48 | |Vulture| | that should work to null it right? |
| 20:02.53 | |Vulture| | or 1,Set(CALLERID(name)="") ? |
| 20:03.08 | sivana | exten => s,1,Set(CALLERID(name)="") |
| 20:03.12 | sivana | that's what I have |
| 20:03.27 | |Vulture| | omg... I bet this system requires a 9... |
| 20:03.31 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 20:03.32 | eKo1 | That doesn't null it though. |
| 20:03.37 | |Vulture| | Im gunna kick myself |
| 20:03.39 | sivana | I doubt it's that though now... I thought you might have had a full pri |
| 20:04.34 | |Vulture| | nah we use fracts since we only have like 4-5 employees per office and non-T1 bandwidth is a joke for VoIP between our offices |
| 20:04.39 | [TK]D-Fender | For those who care I've just set up my first Polycom SIP 2.0.1 phone (IP 600) for testing. Will let you know how it goes. |
| 20:04.45 | |Vulture| | tried DSL and it was so sad.. |
| 20:04.57 | sivana | is there no other app_dial.c error? |
| 20:05.10 | sivana | or warning |
| 20:05.13 | harryvv | [TK]D-Fender how much different is it vs the 500? |
| 20:05.15 | [TK]D-Fender | |Vulture|: if you're using a low bandwidth codec and IAX trunking you should be just fine... |
| 20:05.27 | |Vulture| | no just hangs up and then gives a " Everyone is busy/congested at this time (1:0/0/1)" |
| 20:05.42 | |Vulture| | [TK]D-Fender: I was using ilibc for interoffice |
| 20:05.43 | harryvv | |Vulture| sad in what respect? |
| 20:05.45 | watchy | tk: sip 2.0.1 is out? |
| 20:05.49 | [TK]D-Fender | harryvv: You mean comparing the IP 500 vs 600? Or how each platform reacts to the firmware? |
| 20:05.55 | [TK]D-Fender | watchy: :Been a week now. |
| 20:05.59 | |Vulture| | and the DSL on testing was giving me 256k up/down |
| 20:06.05 | watchy | tk: hook me up i feel like living dangeriously |
| 20:06.13 | [TK]D-Fender | watchy: I was going to do my home upgrade last week but got booked up fast. |
| 20:06.18 | harryvv | [TK]D-Fender in regards to configuring the xml files |
| 20:06.29 | watchy | ill upgrade the office phones here at my store |
| 20:06.33 | [TK]D-Fender | harryvv: Same damn files! NO difference. |
| 20:06.34 | |Vulture| | yea none of that worked Im going to go to debug mode |
| 20:06.39 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 20:06.42 | [TK]D-Fender | harryvv: (that you'd care about) |
| 20:06.45 | watchy | i only have like 5 phones here |
| 20:06.54 | watchy | tk: mind putting up a link for the fw? |
| 20:06.59 | [TK]D-Fender | harryvv: Except of course the MicroBrowser is an option of the 600/601/650 |
| 20:07.01 | sivana | |Vulture|: what does pri show span 1 say |
| 20:07.08 | [TK]D-Fender | watchy: extranet.polycom.com |
| 20:07.18 | harryvv | BTW, I never have had my phone setup as a dual line or triple line phone. Since I will hopfully get a second DID then it would be time to change it. |
| 20:07.35 | [TK]D-Fender | harryvv: Most people never use more than 1 line period. |
| 20:07.54 | harryvv | yea that mabey true or the small office. |
| 20:08.15 | [TK]D-Fender | harryvv: Careful on semantics... I loaded that last statement ;) |
| 20:08.17 | |Vulture| | sivana: Provisioned, Up, Active |
| 20:08.24 | |Vulture| | sivana: inbound calls are working fine |
| 20:08.28 | sivana | ok |
| 20:08.45 | harryvv | But in a case my wifes office gets alot of bussy signals on there tdm circuit and I want to see them have a option of a asterisk box with a second line. |
| 20:08.57 | eKo1 | |Vulture|: perhaps your PRI provider hasn't opened the channels yet. |
| 20:09.08 | hmmhesays | so I got my skype to sip shiat working ok |
| 20:09.13 | [TK]D-Fender | harryvv: : no using more than 1 LINE KEY sure! I typically use 3 line keys supporting 1 call each for IP 5XX/6XX, and 2 for 3XX/4XX. |
| 20:09.26 | hmmhesays | [TK]D-Fender: you figure out your presence stuff last night? |
| 20:09.35 | [TK]D-Fender | harryvv: My home IP 501 however uses 3 completely seperate reg's with support for 5 calls EACH. |
| 20:09.48 | watchy | tk: im logged in there but i dont see it i see hte userguides though |
| 20:09.51 | |Vulture| | eKo1: thats what I am thinking |
| 20:09.54 | [TK]D-Fender | hmmhesays: Nope.... then again a new * build came out... |
| 20:09.56 | Ryushin | Is there a default dns address that polycom phones will try and use to find the asterisk server? |
| 20:09.59 | harryvv | [TK the configuration for those lines are in sip.conf or phone.cfg |
| 20:10.01 | harryvv | err |
| 20:10.04 | harryvv | sip.cfg |
| 20:10.09 | hmmhesays | are you using presence on the poly's? |
| 20:10.12 | [TK]D-Fender | watchy: Contact your reseller |
| 20:10.25 | [TK]D-Fender | hmmhesays: yup, of course |
| 20:10.31 | watchy | my reseller is voipsupply i guess |
| 20:11.04 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 20:11.21 | harryvv | I have used atacom |
| 20:11.25 | |Vulture| | http://pastebin.ca/168207 from what I can see it looks like eKo1 is correct, the router is not setup to handle calls out yet |
| 20:11.32 | hmmhesays | I got my 2102 auto config stuff running |
| 20:11.34 | hmmhesays | that rocks |
| 20:11.36 | IronMan2000 | s |
| 20:11.55 | harryvv | 2102? |
| 20:12.01 | hmmhesays | mediatrix 2102 |
| 20:12.11 | harryvv | what is it a sip gateway |
| 20:12.23 | hmmhesays | yeah 2 port fxs gateway |
| 20:14.22 | harryvv | interesting |
| 20:14.31 | harryvv | how does it compare to asterisk |
| 20:14.50 | benjk | its a front end to * |
| 20:14.55 | |Vulture| | it amazes me how many ppl are rebadging * now |
| 20:15.12 | benjk | to hook up POTS phones |
| 20:15.13 | *** join/#asterisk Delta239 (n=blablabl@201.226.130.55) |
| 20:15.20 | CunningPike | If I want to do followme, I need two channels, right? |
| 20:15.32 | Delta239 | where can i find the email address for digium support on a card im having problems with? |
| 20:15.46 | benjk | support@digium.com |
| 20:15.55 | Ryushin | For the polycom phones, in a default config, how do they find the asterisk (sip) server? There is probably a dns entry right? |
| 20:15.57 | benjk | surprisingly |
| 20:15.58 | Delta239 | thanks benjk |
| 20:16.40 | IronMan2000 | Got my extenstion hunt to work with a new Macro. YooooHooooooo |
| 20:16.47 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 20:16.59 | [TK]D-Fender | Ryushin: TFTP-BOOT DHCP parameter |
| 20:17.51 | [TK]D-Fender | Ryushin: uses taht parm for FTP |
| 20:18.29 | Ryushin | Right. That gets the firmware. Is that the default for the asterisk server as well? |
| 20:19.23 | [TK]D-Fender | Ryushin: there is a DHCP param as well but typically it should pick it up from the config |
| 20:21.00 | [TK]D-Fender | Ryushin: FTP/TFTP isn't just for firmware, its for configs and everything. |
| 20:21.16 | harryvv | So somone in here said that companies are rebadging asterisk what does this mean? |
| 20:21.35 | CunningPike | harryvv: Check out fonality |
| 20:22.18 | harryvv | ohh yea |
| 20:22.38 | harryvv | I have seen the site. |
| 20:23.07 | *** part/#asterisk Delta239 (n=blablabl@201.226.130.55) |
| 20:24.07 | harryvv | ahh thay are showing the price differences between them and the compitition. |
| 20:24.14 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3dc07b47411ccead) |
| 20:25.19 | *** join/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 20:26.20 | rue_mohr | ok, our "problem" is that more than one phone can check a voicemail box at the same time, I realize this may be a feature, all fine and good, but its not working for our situation, how can we aviod this? |
| 20:27.06 | CunningPike | rue_mohr: What problem are you experiencing? |
| 20:27.26 | rue_mohr | two people can simotaniosly access a voicemail box at the same time |
| 20:27.29 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 20:27.33 | Assid | heya |
| 20:27.42 | Assid | anyone know any good dsl providers in new york? |
| 20:28.24 | rue_mohr | we have the same depertment in two different offices that they cant organize one person checking, so when both offices check that vm box in the morning, and both reply to the messages, things get a little befuddled |
| 20:29.13 | rue_mohr | the department has no proper business process, quite like the rest of the departments, and wont have anytime soon |
| 20:29.19 | rue_mohr | anyhow |
| 20:29.51 | rue_mohr | a lock so that a vm box can only be accessed by one person at a time would be great |
| 20:32.39 | *** part/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 20:32.41 | *** join/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 20:33.05 | dsoTm | http://pastebin.ca/168220 if anyone has a moment. Details of a problem I'm having using g729 |
| 20:33.34 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 20:33.38 | *** join/#asterisk The_LightSide (n=dialt@wbs-196-2-100-159.wbs.co.za) |
| 20:34.07 | The_LightSide | hi all, does anyone know of a softphone for windows CE? |
| 20:36.24 | IronMan2000 | I read somehwere that Asterisk can detect faxes. If so, how does it process them when it detects a fax? |
| 20:36.38 | _deg_ | IronMan2000, he goes to the fax extensions |
| 20:36.43 | _deg_ | extension |
| 20:37.36 | IronMan2000 | yea, I have a fax ext. But once the fax ext. pick it up, how does it handle it from there. |
| 20:37.53 | _deg_ | IronMan2000, it is up to you ;) |
| 20:38.05 | _deg_ | IronMan2000, you could use a rxx_fax app |
| 20:38.26 | _deg_ | IronMan2000, or maybe redirect to a FXS connected fax machine |
| 20:38.37 | _deg_ | IronMan2000, or iaxfax. |
| 20:38.45 | _deg_ | IronMan2000, and pray to work |
| 20:39.01 | IronMan2000 | hmmm, since faxing is so poor with VoIP, would it be an improvment to recieve faxes this way, and the redirect the, to a fax machine? |
| 20:39.16 | _deg_ | IronMan2000, better to use T38. Ive never seen working(i tried...) |
| 20:39.47 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 20:39.48 | IronMan2000 | Faxing stinks with Voip |
| 20:39.48 | _deg_ | IronMan2000, I dont think that there is someone using the t38 stack on asterisk |
| 20:40.16 | IronMan2000 | I was thinking of setting up AsterFax |
| 20:40.29 | _deg_ | could be. |
| 20:40.31 | _deg_ | pray again. |
| 20:40.46 | *** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 20:40.51 | *** join/#asterisk sbma44 (n=tomlee@dsl092-174-002.wdc2.dsl.speakeasy.net) |
| 20:41.21 | The_LightSide | who could i ask about softphones? |
| 20:41.29 | IronMan2000 | I ay can help |
| 20:41.29 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 20:41.33 | IronMan2000 | what u need |
| 20:41.47 | The_LightSide | im looking for a softfone for windows CE |
| 20:42.29 | harryvv | btw, all the rest stops on I-5 in washington are going wifi. |
| 20:42.48 | harryvv | Wouldnt that be nice if I had a wifi phone and could use it up and down the highway :) |
| 20:42.56 | The_LightSide | IronMan2000, would xlite work? |
| 20:43.02 | IronMan2000 | I use X-Lite , but I don't know if they have a version for CE |
| 20:43.14 | The_LightSide | not listed on the website |
| 20:43.18 | The_LightSide | :( |
| 20:43.23 | IronMan2000 | X-Lite works very well. |
| 20:43.34 | The_LightSide | i use it on my desktops... |
| 20:43.51 | The_LightSide | but got a guy who has a cellphone/pda device |
| 20:44.01 | The_LightSide | wants to use that instead |
| 20:45.11 | CtRiX | i<_deg_> IronMan2000, better to use T38. Ive never seen working(i tried...) |
| 20:45.19 | IronMan2000 | try: http://www.pocketpccity.com/software/pocketpc/SJPhone-2002-2-12-ce-pocketpc.html |
| 20:45.22 | CtRiX | _deg_, 'cause you didn't ty openpbx |
| 20:45.27 | IronMan2000 | Windows CE softphone |
| 20:45.34 | The_LightSide | thanks :) |
| 20:45.44 | CtRiX | we have T38 termination working and txfax/rxfax works |
| 20:46.09 | Ryushin | I changed the default username and password for the polycom phones. I keep getting a error 4020 when it boots. It doesn't seem to pull the sip.cfg or the phone1.cfg files. I wondering if there is a place to add the username and password into the 000000000.cfg file? |
| 20:46.15 | CtRiX | <_deg_> IronMan2000, I dont think that there is someone using the t38 stack on asterisk |
| 20:46.20 | *** part/#asterisk clive- (n=pirch@dsl-145-56-115.telkomadsl.co.za) |
| 20:46.21 | The_LightSide | my firewall does not like that link IronMan2000 |
| 20:46.24 | Assid | damn.. anyone know a good dsl provider.. verizon breaks down for me everymonth |
| 20:46.27 | CtRiX | asterX (tm) does not have a t38 stack |
| 20:46.34 | CtRiX | if it could be named this way |
| 20:46.51 | CunningPike | Ryushin: How will it get that file if it can't login? :D |
| 20:47.16 | IronMan2000 | link works fine from here.. |
| 20:48.09 | The_LightSide | strange, but i found the manufactureres website for the SJphone |
| 20:48.12 | Ryushin | I put the username and password into the phone for the ftps server. It pull it's boot stuff fine. I'm thinking that it uses something different for the application portion of it. |
| 20:48.17 | The_LightSide | thanks very much IronMan2000 :) |
| 20:48.31 | smackus | so is it possible to run chanspy from the CLI? or something that does the same |
| 20:48.33 | _deg_ | CtRiX, ok, so, how many people are using OpenPBX? Is there a stable version? How often is the commits? |
| 20:48.38 | *** join/#asterisk spackle[work] (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 20:48.45 | CtRiX | look at the trac, _deg_ |
| 20:48.58 | CtRiX | or join the channel to have infos |
| 20:49.08 | The_LightSide | niht all! |
| 20:49.10 | spackle[work] | that's a lotta folk |
| 20:49.16 | *** part/#asterisk spackle[work] (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 20:49.26 | _deg_ | CtRiX, I will. |
| 20:49.38 | _deg_ | CtRiX, what about freepbx? |
| 20:49.43 | _deg_ | CtRiX, same people? |
| 20:49.53 | CtRiX | freepbx is a frontend, not a pbx |
| 20:50.05 | CtRiX | openpbx is an asterisk fork |
| 20:50.33 | arcanine | does anyone used dialogic vfx in replace of the rhino r4fxo for analog |
| 20:50.42 | _deg_ | CtRiX, frontend? |
| 20:50.48 | CtRiX | web gui |
| 20:50.54 | CtRiX | call it asyou prefer |
| 20:51.34 | sivana | has anyone done video conferencing with asterisk? |
| 20:51.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:56.36 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:57.08 | IronMan2000 | no, but Grandstream has a new Video Phone out that I here works really well. |
| 20:57.52 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:58.55 | Cyt | Hi! I'm trying to play an announcement with the option A on dial cmd. The problem is asterisk says the file doesnt exist on any format, but the file is there. I even tried to change the file into vm-next.gsm to see waht happend, but still the same. What could be the problem? (the output is here: http://pastebin.ca/168252) |
| 20:59.34 | sivana | I need to link up three separate "board rooms" from the same company, from 3 different locations |
| 21:00.10 | [TK]D-Fender | Cyt : remove the suffix. |
| 21:00.46 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 21:00.49 | mitcheloc | join #asterisk-dev |
| 21:00.51 | hmmhesays | I'll dumb it down... "remove the .gsm" |
| 21:01.04 | eKo1 | Uh oh. Pressing C-? kills *. |
| 21:01.05 | hmmhesays | hmm |
| 21:01.05 | [TK]D-Fender | hmmhesays : No... people are dumb enough already! |
| 21:01.07 | hmmhesays | video phone? |
| 21:01.17 | eKo1 | Never knew that. |
| 21:02.19 | h3x | why did grandstream even bother making a BLF console |
| 21:02.26 | hmmhesays | I'd like to get my hands on a video phone |
| 21:02.34 | h3x | nobody would ever take their phones seriously for 112 extensions |
| 21:02.47 | Cyt | oh guys! thank you VERY much... The docummentation on voip-info says: A(x): Play an announcement (x.gsm) to the called party. |
| 21:02.50 | h3x | Just use x-ten's eyebeam for video |
| 21:02.52 | hmmhesays | h3x: why not? |
| 21:02.55 | Cyt | x.gsm |
| 21:02.59 | hmmhesays | oh , but hardware video |
| 21:03.01 | hmmhesays | c'mon |
| 21:03.18 | h3x | because their engineering is crap |
| 21:03.18 | Cyt | but I got my stupid error! thank you again |
| 21:03.31 | [TK]D-Fender | h3x : They do sell enough to warrant it. Thats not saying they're GOOD, just that they sell. Too many cheap bastards out there supporting shit products sales figures... |
| 21:08.04 | hmmhesays | I've not had many problems with grandstream |
| 21:08.59 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 21:10.57 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 21:12.18 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.43) |
| 21:12.32 | *** join/#asterisk Kerry_G (n=Kerry_G@adsl-64-149-238-161.dsl.irvnca.sbcglobal.net) |
| 21:13.02 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 21:13.05 | Kerry_G | having an odd problem here I havent seen before, using a TDM2400, users call out but DTMF to remote systems does not work |
| 21:14.03 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 21:15.31 | hmmhesays | does asterisk support h.264 now? |
| 21:16.42 | watchy | anyone here bet sports |
| 21:16.54 | hmmhesays | i play some black jack now and again |
| 21:17.02 | watchy | sports homie |
| 21:17.13 | watchy | like ncaa and nfl |
| 21:17.23 | watchy | i made $1200 this week on nfl and ncaa |
| 21:17.48 | niter3 | I'm about ready to kill someone. :) |
| 21:17.50 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:18.03 | niter3 | to dial in to normal businesses pbx on lan line. Which DTMF should I sue? |
| 21:18.04 | niter3 | use? |
| 21:18.05 | watchy | hug me |
| 21:18.43 | niter3 | i'm using a SIP outbound trunk... |
| 21:18.50 | dmz | hey y'all, i am trying to setup my system with 2 DIDs and have it rollover to the 2nd line if the 1st is busy, the SIP provider I'm talking to says it is asterisk that does that, but it doesn't make sense to me if the 1st # is busy then how does the 2nd call get into asterisk to route to the 2nd DID :| |
| 21:18.57 | niter3 | inband seems to work better but it sounds like shit on the other end. |
| 21:19.23 | niter3 | dmz: that makes sense |
| 21:19.41 | dmz | niter3 thought it was them not me smoking crack |
| 21:19.52 | niter3 | hrm... Check into fall through.. |
| 21:19.55 | dmz | now i just need to find a decent voip provider for a small business |
| 21:19.58 | niter3 | I could be wrong.. but check it out |
| 21:20.07 | hmmhesays | anyone asterisk h.264 google is not helping me much on this one |
| 21:20.53 | harryvv | fonality really put alot of work into there site and its offerings. |
| 21:21.22 | niter3 | dmz for your outbound sip provider what dtmf do you use? |
| 21:21.36 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 21:21.38 | niter3 | When i call other companies their PBX don't reconzie when I punch in numbers.. |
| 21:21.41 | dmz | niter3 haven't setup anything yet |
| 21:21.55 | niter3 | huh... |
| 21:22.15 | dmz | niter3 oh just reread that...i don't have a sip provider yet |
| 21:22.18 | Kerry_G | having the same problem but with aa TDM2400 card |
| 21:22.20 | quid246 | Hmmm... anyone knowledgable about IAX Authentication (and yes, I've read teh Wiki on it)? |
| 21:22.45 | niter3 | damnit... |
| 21:22.53 | niter3 | i need a way of having this work.. |
| 21:23.51 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:23.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:25.06 | *** join/#asterisk j0 (i=dan@CABLE-72-53-45-212.cia.com) |
| 21:26.36 | dserban | omfg what a piece of shit ael is... |
| 21:26.41 | dserban | seriously what a fucking joke... |
| 21:26.43 | dserban | ok i'm done now |
| 21:26.45 | dserban | thanks |
| 21:29.51 | anthonyl | ael2 or ael? |
| 21:30.03 | dserban | ael |
| 21:30.14 | dserban | I can't run anything past 1.2.11 since it's deemed unstable. |
| 21:30.16 | dserban | :s |
| 21:31.29 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 21:34.38 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 21:39.42 | niter3 | hrm.. this blows ass |
| 21:41.19 | Zodiacal | any ideas why i get this error when trying to send a .tif fax with a .call file? app_txfax.c: Fax send not successful - result (14) TIFF/F file cannot be opened. |
| 21:41.22 | vader-- | has anyone done an upgrade to 1.2.12.1? |
| 21:42.03 | quid246 | vader: ya |
| 21:42.09 | vader-- | from what version? |
| 21:42.14 | vader-- | did you upgrade zaptel too? |
| 21:42.15 | quid246 | 1.2.11 |
| 21:42.17 | sbma44 | zodiacal: just a guess, but wrong color depth on the tiff? have you tried sending one generated by asterisk? |
| 21:42.18 | quid246 | yup |
| 21:42.24 | sbma44 | (from an incoming call) |
| 21:42.27 | vader-- | im thinking about going from 1.2.7 |
| 21:42.28 | quid246 | but I have no Zap devices, pure SIP/IAX |
| 21:42.33 | vader-- | think ill hit any issues? |
| 21:42.43 | quid246 | probably not |
| 21:42.58 | Zodiacal | sbma44 i tried a sample tiff i found in the mgetty-1.1.31 folder |
| 21:43.14 | Zodiacal | sbma44 how do you get * to generate a .tiff off hand? |
| 21:43.22 | quid246 | just keep a compiled version of 1.2.7 in your /usr/src and if you don't like .12, then clear it out and put your old version back |
| 21:43.41 | sbma44 | z: can't say I've done a lot with fax. I assumed that was the native format for incoming. |
| 21:43.48 | *** join/#asterisk mangaan (n=chatzill@83-217-93-101.adsl.realdsl.be) |
| 21:43.54 | sbma44 | hopefulyl someone more knowledgeable will chime in |
| 21:44.17 | vader-- | quid know of any instructions for upgrading>? |
| 21:44.46 | Zodiacal | sbma44 im trying to send it outgoing |
| 21:45.03 | sbma44 | I know, but if you had one come in it'd presumably be in the right format to send out |
| 21:45.15 | sbma44 | then if that worked you'd have been able to isolate the problem to the specific tiff you were trying to send |
| 21:45.19 | Zodiacal | sbma44 ahh..ya.. |
| 21:45.35 | Zodiacal | sbma44 if i can't figure this out in a few i'll try that.. but incoming isn't configured either :P |
| 21:47.52 | *** join/#asterisk albertito (n=net@host34.201-253-130.telecom.net.ar) |
| 21:48.11 | mangaan | Can somebody help me with a channel busy setup |
| 21:48.37 | *** part/#asterisk albertito (n=net@host34.201-253-130.telecom.net.ar) |
| 21:49.41 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 21:50.42 | *** join/#asterisk Corydon76-home (i=five@pdpc/supporter/sustaining/Corydon76-home) |
| 21:50.42 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 21:50.56 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 21:51.25 | Ryushin | Great, it looks like ftps is broken on the polycom phones. |
| 21:57.55 | quid246 | Hmm.. anyone have IAX authenticaiton working with MySQL? |
| 21:59.43 | sx-wks | I'm looking for a way to obtain the length (in seconds) of a sound file |
| 21:59.57 | niter3 | wow this is such a piss off.. MY sip provier that i'm using as trunk for outgoing calls says it uses AVT so I've set it in asterisk and on my client phone and when I call out to a company it does not reconize any of the digits I punch in. |
| 21:59.58 | niter3 | urg.. |
| 22:00.04 | niter3 | anyone know what I can do to resolve this? |
| 22:00.26 | eKo1 | AVT? |
| 22:00.53 | niter3 | Audio/Visual Transport |
| 22:00.55 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:03.56 | mangaan | Wavepad will do the job |
| 22:15.16 | |Vulture| | wow... I figured out why my PRI was rejecting calls... |
| 22:15.33 | |Vulture| | Anyone ever have a PRI that didn't let you set CID to a non DID assigned to you? |
| 22:15.46 | *** join/#asterisk oej (n=oej@12.38.214.2) |
| 22:15.54 | wunderkin | some do that |
| 22:16.16 | SomeJ | niter3 : did you set dtmfmode=RFC2833 ? |
| 22:16.50 | |Vulture| | wunderkin: do you know if they will open that if they have it enabled? |
| 22:17.07 | wunderkin | i dont know, ask them |
| 22:17.31 | |Vulture| | yea I am going to just got off the phone with the guy cause I couldnt get it to work |
| 22:21.00 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 22:22.02 | pifiu | anyone know what a "no authority found" error means when using IAX to connect 2 * boxes? |
| 22:22.44 | benjk | most likely password doesn't match, username wrong, or wrong context |
| 22:22.49 | dserban | How can I set the callerid name on an incoming queue? I've searched voip-info :s |
| 22:22.53 | niter3 | SomeJ: that doesn't work.. |
| 22:22.54 | dserban | I know I've seen it before. |
| 22:22.56 | niter3 | i have to use inband |
| 22:23.02 | niter3 | only thing that is working really |
| 22:23.26 | SomeJ | niter3 what codec you using? |
| 22:23.27 | pifiu | benjk and how can i find out for sure? the users and passwords seem to match |
| 22:23.33 | niter3 | ulaw |
| 22:23.36 | benjk | iax debug |
| 22:23.39 | niter3 | and i've did a disallow=all |
| 22:23.42 | niter3 | allow=ulaw |
| 22:23.50 | niter3 | but I'm getting and error in my CLI |
| 22:24.07 | niter3 | Inband DTMF is not supported on codec gsm. |
| 22:24.34 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:24.38 | niter3 | allow=ulaw |
| 22:24.38 | niter3 | disallow=all |
| 22:24.38 | niter3 | dtmfmode=inband |
| 22:24.41 | niter3 | that's exactlyw hat i have |
| 22:24.53 | niter3 | under my sip.conf for one context. |
| 22:26.46 | benjk | if you disallow=all you have no codec |
| 22:27.04 | SomeJ | if your phones are set to avt, then in the sip.conf for that phone try setting dtmfmode=RFC2833. Just see what it does |
| 22:27.12 | benjk | you need to reverse the order |
| 22:27.19 | benjk | first disallow=all |
| 22:27.26 | niter3 | ok that's done. |
| 22:27.31 | benjk | and THEN allow=ulaw |
| 22:27.35 | SomeJ | and what benjk says ;) |
| 22:28.00 | niter3 | I have a couple options in my client phone.... Auto, INFO, AVT, Inband |
| 22:28.24 | SomeJ | avt is not inband, so you dont want your sip.conf looking for inband for the clients |
| 22:28.35 | benjk | a better syntax would have been codecs = ( ulaw, alaw, gsm ) |
| 22:28.48 | benjk | but anyway |
| 22:29.10 | niter3 | the listing on my phone is a pull down menu from a web interface. I was just telling you the ones |
| 22:29.33 | benjk | that's no good for trouble shooting though |
| 22:29.44 | X-Rob_ | That's a Sipura |
| 22:29.46 | benjk | you gotta look at the actual configs |
| 22:29.49 | niter3 | pap2 linksys |
| 22:29.51 | X-Rob_ | and you want AVT which is rfc2883 |
| 22:29.57 | benjk | ok, fair enough |
| 22:30.25 | benjk | I didn't recall the sipura has a disallow=all setting |
| 22:30.45 | X-Rob_ | benjk, that's a really good syntax. Someone shoudl write a parser for that 8) |
| 22:30.56 | benjk | also you said "niter3: that's exactlyw hat i have |
| 22:30.57 | benjk | [07:24am] niter3: under my sip.conf for one context." |
| 22:31.24 | benjk | so I didn't think it was meant to be a descriptopn of you sipura settings |
| 22:31.30 | niter3 | yah when I use AVT on the unit and rfc2883 on asterisk. WHen calling my cell phone it doesn't reconize the digits i punch in |
| 22:31.37 | benjk | X-Rob, I did |
| 22:32.00 | benjk | and the lexer is a monster |
| 22:32.09 | benjk | because the grammar is not context free |
| 22:32.24 | X-Rob_ | niter3, common problem with cellphones. Nothing you can really do about it. They don't send DTMF properly |
| 22:32.34 | X-Rob_ | Try a different cell until you find one that works |
| 22:33.26 | niter3 | X-Rob_: inband works |
| 22:33.39 | benjk | which means nothing |
| 22:34.01 | X-Rob_ | My GXP2000 is black |
| 22:34.02 | benjk | it only means that Asterisk's DTMF recogniser works |
| 22:34.08 | benjk | thanks to Steve Underwood |
| 22:34.25 | X-Rob_ | benjk, heh |
| 22:34.31 | benjk | outbound is a different DTMF detector, in the remote end device |
| 22:34.48 | benjk | and Asterisk makes DTMF tones too short anyway |
| 22:35.07 | benjk | some devices can't detect them properly |
| 22:35.44 | niter3 | yah well if Use inband on my client asterisk can't reconzie shit.. |
| 22:35.56 | niter3 | but when I use inband for my outbound trunk it works |
| 22:36.19 | benjk | doesn't mean anything either |
| 22:36.31 | benjk | whats the transit protocol in between? |
| 22:36.36 | benjk | and which codecs? |
| 22:36.53 | benjk | asterisk may not even get the DTMF |
| 22:37.00 | benjk | it may get stuck before that |
| 22:37.05 | niter3 | benjk: The trunk uses AVT |
| 22:37.19 | benjk | SIP? |
| 22:37.24 | niter3 | yes |
| 22:37.39 | niter3 | That's if I didn't get a moron telling me something he didn't know. |
| 22:37.42 | benjk | did you configure both sides to use the same DTMF method? |
| 22:38.55 | niter3 | I told my asterisk system to use AVT and the receiving end trunk I went by what the guy told me. On my client phone I just used AUTO and it didn't work. However, if I use auto on my client phone and inband for the outbound sip trunk back to normal lines it works on my cell phone and other companies pbx's. |
| 22:39.06 | benjk | in general, inband DTMF should only be expected to work with ulaw and alaw |
| 22:39.16 | niter3 | which are the codecs I told it to use |
| 22:39.35 | niter3 | The point is the inband with ulaw works. |
| 22:39.37 | benjk | on both devices |
| 22:39.45 | niter3 | What do you mean on both devices? |
| 22:39.52 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:39.58 | CtRiX | box1 <----> box2 |
| 22:39.59 | benjk | asterisk is talking SIP to some other device |
| 22:40.12 | *** join/#asterisk RoyK (n=roy@ti211110a081-1432.bb.online.no) |
| 22:40.15 | niter3 | yes all devices connected to the pbx internally work fine |
| 22:40.27 | niter3 | I just defined the inband for that specific context in my sip.conf |
| 22:40.31 | niter3 | just for that one. |
| 22:40.34 | benjk | no, on that connection where ou are having trouble with DTMF |
| 22:40.59 | niter3 | for my internal shit |
| 22:40.59 | niter3 | no where |
| 22:41.01 | niter3 | everything works fine |
| 22:41.05 | benjk | is the remote end also configured to use the same DTMF method? |
| 22:41.07 | niter3 | it's just outside PSTN access |
| 22:41.18 | niter3 | benjk: Again all they told me was that they use AVT |
| 22:41.20 | niter3 | that's it |
| 22:41.24 | niter3 | I tried that, it didnt' work |
| 22:41.28 | niter3 | So I chose inband it works |
| 22:41.40 | benjk | four outbound too? |
| 22:41.48 | niter3 | it's only for outbound |
| 22:41.51 | benjk | so then what's the problem |
| 22:42.02 | niter3 | Nothing I fixed it I guess.. |
| 22:42.28 | benjk | fine then :) |
| 22:43.03 | niter3 | I guess it would be another issue when I accept inbound connections |
| 22:43.35 | benjk | btw, Rob-X, somebody is working on a PHP class to write the new configs, fyi |
| 22:44.10 | benjk | the key is to make sure that both ends use the same DTMF method |
| 22:44.32 | benjk | that solves most DTMF problems |
| 22:44.39 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 22:45.29 | benjk | whats left over, if its inband, usually has to do with DTMF detection and like I said, on outbound DTMF, asterisk's dtmf tones are a bit short which some devices cannpt easily recognise |
| 22:45.53 | benjk | bah, X-Rob I meant |
| 22:47.45 | quid246 | Hmm.. anyone have IAX authentication working with MySQL? |
| 22:53.18 | *** join/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 22:53.52 | dhahn | Hello |
| 22:54.14 | dhahn | Any about? |
| 22:54.45 | benjk | == everybody is asleep right now |
| 22:55.03 | dhahn | Sounds like my life... |
| 22:55.21 | benjk | you're asleep? |
| 22:55.28 | dhahn | I was hoping to get some help on dial plan issues... |
| 22:56.26 | benjk | the way this works is that folks just ask a question and if somebody has something to say, they will answer |
| 22:57.01 | dhahn | I'm trying to put together an outbound message to be played when answered |
| 22:57.23 | dhahn | When the call is answered, no message plays |
| 22:57.41 | dhahn | However, if I use the A(tt-weasels) example, it plays when the call is answered |
| 22:57.47 | dhahn | Not sure what I'm doing incorrectly |
| 22:58.24 | benjk | are you sure you have the correct filename? |
| 22:58.33 | dhahn | Yes |
| 22:58.34 | bkruse | pastbin your dialplan, i have a couple minutes to look at it |
| 22:58.39 | dhahn | [test] |
| 22:58.40 | dhahn | exten => s,1,Wait,1 ; Wait a second, just for fun |
| 22:58.40 | dhahn | exten => s,Playback,demo-congrats ; Play a congratulatory message |
| 22:58.40 | dhahn | exten => s,Playback,demo-instruct ; Play some instructions |
| 22:58.40 | dhahn | exten => s,n,Hangup ; Wait for an extension to be dialed. |
| 22:58.52 | dhahn | This doesn't play anything |
| 22:58.55 | bkruse | ....... |
| 22:58.57 | bkruse | voip-info.org |
| 22:59.07 | bkruse | exten => s,Playback(demo-congrats) |
| 22:59.08 | bkruse | try that. |
| 22:59.12 | benjk | first |
| 22:59.14 | dhahn | However, exten => _1XXXXXXXXXX,s,Dial(SIP/${EXTEN}@globotech,90,A(tt-allbusy)) |
| 22:59.16 | bkruse | woah wait |
| 22:59.17 | benjk | use pastebin |
| 22:59.21 | bkruse | yes plz. |
| 22:59.21 | benjk | ~pastebin |
| 22:59.24 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 22:59.28 | bkruse | there. |
| 22:59.36 | dhahn | got it |
| 22:59.44 | bkruse | first of all, change it to look like this, |
| 22:59.47 | benjk | second, you need to answer the channel first before you can play anything |
| 23:00.00 | bkruse | exten => s,1,Wait(1) |
| 23:00.00 | bkruse | exten => s,n,Playback(demo-congrats) |
| 23:00.03 | bkruse | try that. |
| 23:00.15 | bkruse | benjk: good point, sry |
| 23:00.18 | CtRiX | Answer() |
| 23:00.34 | bkruse | s/exten => s,1,Wait(1)/exten => s,1,Answer() |
| 23:00.37 | CtRiX | some apps would answer automatically,other does not |
| 23:00.48 | bkruse | right, for practice sake, go ahead and answer all calls. |
| 23:00.48 | DrukenHME | dhahn: try answering first :) |
| 23:00.51 | CtRiX | that's a good practice toanswer in all circumstances |
| 23:01.05 | dhahn | Got it, I'll give it a quick shot |
| 23:01.24 | benjk | dont flood |
| 23:01.36 | bkruse | ill pastebin it. |
| 23:01.42 | benjk | thx |
| 23:03.01 | bkruse | http://pastebin.ca/168396 |
| 23:03.13 | bkruse | dhahn: click this, and then copy, analyze, learn it |
| 23:03.15 | bkruse | http://pastebin.ca/168396 |
| 23:03.46 | dhahn | k, trying, thx |
| 23:04.23 | bkruse | tell me if it didnt work, i might have made a mistake, i dont think i did though. |
| 23:04.32 | brimstone | eh, first glance looks good to me |
| 23:04.42 | DrukenHME | yeah.... installing odbc is FUN.... |
| 23:04.45 | bkruse | eww. |
| 23:04.57 | niter3 | hey guys... i want to do an after hours contexts... What can I search up for this? |
| 23:04.57 | hmmhesays | a wave with a happy ending? |
| 23:05.01 | brimstone | i thought playback answered before it played anyway |
| 23:05.05 | bkruse | DrukenHME: you hung up on anything? |
| 23:05.09 | bkruse | hmmhesays: absolutly |
| 23:05.27 | DrukenHME | bkruse: hung up on anything? |
| 23:05.39 | DrukenHME | only telemarketers :) |
| 23:05.48 | DrukenHME | oh, and ex girlfriends.... |
| 23:06.28 | dhahn | bkruse: Tried that, still no playback when I answer the call |
| 23:07.02 | DrukenHME | what are you dialing it with? |
| 23:07.03 | bkruse | asterisk -vvvvvgcT |
| 23:07.05 | *** join/#asterisk corresponder (n=correspo@p54AD7695.dip.t-dialin.net) |
| 23:07.08 | bkruse | set verbose 10 |
| 23:07.09 | corresponder | hi there |
| 23:07.13 | bkruse | tell me what your output is, pastebin it |
| 23:07.47 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 23:09.07 | corresponder | does one know where the german language files are - i guess a german university has them... |
| 23:09.19 | quid246 | What's the purpose of the "dbsecret" field in Realtime IAX? |
| 23:09.22 | dhahn | bkruse: http://pastebin.ca/168401 |
| 23:09.37 | CtRiX | quid246, connecting to a database,i suppose |
| 23:09.51 | CtRiX | if db is for database and secret for the password. |
| 23:10.02 | quid246 | yeah I kind of figured that oen out, thanks. |
| 23:10.24 | quid246 | but "which" database is the question. :) |
| 23:10.26 | CtRiX | quid246, don't play with realtime unless you have some experience. That's a hard part that onw. |
| 23:10.31 | DrukenHME | uhg.... using consol as a phone.... |
| 23:10.32 | DrukenHME | pfft |
| 23:10.44 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:10.47 | CtRiX | quid246, that's what i said !! don't play with it. Youprobably don't need it |
| 23:11.17 | DrukenHME | realtime isn't hard.... it's just database entries.... |
| 23:12.06 | CtRiX | make it work ... being clueless on what a database is... add asterisk weird behaviour and you may need a bedin a clinic. |
| 23:12.40 | quid246 | yeah, I'm decen twith MYSQL... just can't figure out why my * won't accept registrations for RealTime IAX.... the Wiki is weak on real definitions.. like "md5secret"... is the value in plaintext, or should a hash appear there |
| 23:13.02 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:13.16 | CtRiX | quid246, "iax debug" |
| 23:13.19 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.43) |
| 23:14.04 | bkruse | dhahn: looking at it now |
| 23:14.30 | dhahn | bkruse: thx |
| 23:14.34 | dhahn | not much there |
| 23:14.46 | bkruse | dhahn: looks like it isnt even reaching the context, your dialplan looks fine, tell me what you are trying to do |
| 23:14.51 | bkruse | but your sip phone is not reaching those instructions. |
| 23:14.54 | DrukenHME | quid246: do yourself a favour... if you can... use odbc instead of straight mysql |
| 23:14.58 | quid246 | ctirx: Yeah, not much use there when it just says "No registration for peer". |
| 23:15.03 | bkruse | pastebin me your extensions.conf and sip.conf if you can :] |
| 23:15.52 | benjk | the asterisk console is a liar anyway |
| 23:16.01 | quid246 | benjk: amen to that |
| 23:16.19 | benjk | the only trustworthy information is the debug output and log files |
| 23:16.21 | dhahn | bkruse: trying to have asterisk initiate an outbound call, play a message, accept some dtfm, confirm and hangup |
| 23:16.31 | dhahn | bkruse: as you can see, it isn't to that point yet |
| 23:16.31 | CtRiX | dhahn, http://lists.digium.com/pipermail/asterisk-dev/2006-June/021215.html |
| 23:16.41 | benjk | anything else you see on the console is bullshit which is probably not true |
| 23:16.43 | CtRiX | sorry |
| 23:16.45 | CtRiX | quid246, http://lists.digium.com/pipermail/asterisk-dev/2006-June/021215.html |
| 23:17.42 | bkruse | k, give me what you have so far, actually send just send me a PM |
| 23:17.50 | bkruse | lets get out of this pub asterisk channel, i got some questions :] |
| 23:18.31 | benjk | you can always come to #openpbx ;) |
| 23:18.39 | dhahn | bkruse: k |
| 23:18.46 | benjk | no censorship there |
| 23:18.49 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 23:19.50 | quid246 | ctrix: Yeah I saw that URL already... if you notice the follow up message, the bug appeared squashed |
| 23:20.04 | quid246 | thanks for the tip though |
| 23:20.20 | CtRiX | that's not always what you should be ! |
| 23:20.49 | hmmhesays | ugh customers suck sometimes |
| 23:20.55 | hmmhesays | always jumping to stupid conclusions |
| 23:21.13 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 23:21.13 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 23:21.13 | DrukenHME | i wish i could get some of my customers to suck.... |
| 23:23.48 | dennisharrison | tell me about it |
| 23:23.48 | dennisharrison | I have one in particular that looks like a prime canidate ;p |
| 23:25.22 | quid246 | urg... I can get Realtime IAX to register when type=friend but not type=user. |
| 23:26.02 | DrukenHME | so whats the problem ? |
| 23:26.07 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-e6ccdc5439333330) |
| 23:28.16 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:29.24 | *** part/#asterisk dhahn (n=dhahn@ip-216-17-139-63.rev.frii.com) |
| 23:29.41 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 23:32.55 | arcanine | does anyone used rhino r1t1? |
| 23:34.17 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:35.48 | quid246 | hmmm |
| 23:36.11 | quid246 | are "users" allowed to register in IAX or are they denied? |
| 23:36.35 | quid246 | seems that I can get users not to register, but they can make authenticated calls. |
| 23:38.27 | DrukenHME | god damn... compiling takes FOREVER.... |
| 23:38.47 | bkruse | quid246: they should be able to i think, but ive been lately having problems with this! |
| 23:38.48 | DrukenHME | i'm reminded of back in the day when compiling a kernel was an all night affair.... |
| 23:39.24 | quid246 | Drunken: haha, I still remember compiling * on my P100 2 years ago |
| 23:39.27 | quid246 | seems like a dog's age |
| 23:39.49 | corresponder | it talks german - omg! ;-) |
| 23:39.52 | corresponder | *fg* |
| 23:39.55 | DrukenHME | a p100 two years ago ? |
| 23:39.57 | DrukenHME | damn..... |
| 23:40.09 | DrukenHME | i've had a 2.4 ghz for like 3 years now... |
| 23:41.18 | corresponder | what engines did you use for asterisk? |
| 23:42.27 | brimstone | corresponder, mine runs on a diseal |
| 23:42.30 | quid246 | I was always privy to V8... but with the price of gas, I prefer a V6 now |
| 23:42.34 | brimstone | desiel |
| 23:42.35 | corresponder | ha ha ha |
| 23:42.36 | brimstone | maybe? |
| 23:42.38 | quid246 | diesel |
| 23:42.40 | corresponder | how funny |
| 23:42.42 | brimstone | yay! |
| 23:42.42 | corresponder | diesel |
| 23:42.55 | corresponder | joh! |
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| 23:43.44 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:46.07 | DrukenHME | anyone know what other carriers there are in ontario? bell, allstream, GT, ??? |
| 23:47.00 | blitzrage | no L(3)? |
| 23:47.36 | DrukenHME | l(3) ? |
| 23:49.24 | blitzrage | level 3 |
| 23:49.39 | DrukenHME | i figured that much, but who are they ? |
| 23:49.46 | blitzrage | ?! |
| 23:50.08 | blitzrage | level 3 is a pretty major carrier |
| 23:51.17 | h3x | its about the oldest one there is besides at&t |
| 23:51.18 | h3x | heh |
| 23:51.26 | DrukenHME | in the us, or in canada? |
| 23:51.40 | h3x | us |
| 23:51.46 | h3x | in canada theres primus |
| 23:51.50 | DrukenHME | well, ontario would be in canada.... |
| 23:52.02 | h3x | mci |
| 23:52.05 | h3x | global crossing |
| 23:52.22 | blitzrage | possibly dci |
| 23:52.29 | h3x | you didnt say you were looking for LECs or IXCs |
| 23:52.41 | h3x | Rigers |
| 23:52.43 | h3x | Rogers i mean |
| 23:53.18 | DrukenHME | i wouldn't trust rogers with my home phone, let alone my pri.... |
| 23:53.46 | quid246 | my parents have rogers for home phone... in the past year they've had a few times where they are down 24 hours |
| 23:54.09 | DrukenHME | rogers homephone? yeah it's voip |
| 23:54.14 | quid246 | no |
| 23:54.37 | quid246 | well I dunno how they carry their signal.. but atleast, my parents use the same old POTS line they did with bell |
| 23:54.54 | blitzrage | Rogers goes over cable, not standard copper |
| 23:55.18 | DrukenHME | well, i belive rogers purchased sprint, so they may have some copper as well |
| 23:55.31 | quid246 | drunken; you are correct sir |
| 23:55.35 | h3x | i thought rogers would be using VoATM like the other cable cos |
| 23:55.53 | h3x | sprint canada was never really sprint to begin with |
| 23:55.56 | h3x | they franchised the name |
| 23:56.10 | h3x | it was some small privately held comapny |
| 23:56.50 | blitzrage | crazyness |
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