| 00:02.58 | *** join/#asterisk shido6 (n=shido6@74-132-115-11.dhcp.insightbb.com) |
| 00:05.05 | *** join/#asterisk Levonk (n=lk@adsl-76-243-66-74.dsl.lsan03.sbcglobal.net) |
| 00:12.48 | mchou | gah! need help. I'm using sipbroker (and ekiga.net) for testing purposes. I can dial certain sipbroker #s (like *1800-NXX-NXX-XXXX) and get great connections. However if I try dial *673-500 (ekiga's echo test via sipbroker) I get chan_sip.c:12259 handle_response_invite: Failed to authenticate on INVITE to <username>. |
| 00:13.55 | mchou | I also get "SIP/2.0 407 Proxy Authentication Required" |
| 00:14.28 | TJNII | I never got * working as a client to Ekiga's servers with that very error. |
| 00:14.37 | mchou | and "SIP/2.0 403 Forbidden" |
| 00:14.59 | mchou | TJNII: you man via sipbroker or direct to ekiga? |
| 00:15.30 | TJNII | It's been a year, I don't remember anymore. |
| 00:15.41 | mchou | cause my direct to ekiga (w/o going thru sipbroker) works just fine |
| 00:21.16 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 00:24.05 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 00:25.50 | *** join/#asterisk jmacz (n=jmacz@mail.alianza.com.co) |
| 00:28.09 | *** part/#asterisk otavio (n=otavio@debian/developer/otavio) |
| 00:35.04 | *** join/#asterisk hatoon (n=musis@189.71.123.101) |
| 00:36.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-45b13b3a10d326b5) |
| 00:36.50 | *** join/#asterisk adrian (i=xpl@84.126.212.2.dyn.user.ono.com) |
| 00:37.15 | adrian | who can i make outbund call with vonage with unknow call ? |
| 00:37.22 | adrian | she can be do that ? |
| 00:37.34 | mchou | wha?? |
| 00:37.54 | adrian | i have a vonage sip |
| 00:38.07 | mchou | try to speak comprensible english pls |
| 00:38.09 | adrian | and i want to make calls with unknow phone number |
| 00:38.12 | adrian | :P |
| 00:38.13 | adrian | sorry :) |
| 00:38.23 | Alton2 | maybe he means he wants to hide his caller id |
| 00:38.25 | adrian | with vonage and asterisk |
| 00:38.28 | jaytee | you mean you want to block callerID? |
| 00:38.31 | adrian | yes Alton2 |
| 00:38.33 | mchou | what's an unknow phone number? |
| 00:38.47 | adrian | i want to make calls with hide caller id |
| 00:39.09 | adrian | sorry people for my english is a little bad..:( |
| 00:39.32 | adrian | Alton2 : yes i want to hide the caller id |
| 00:39.40 | mchou | try *67 |
| 00:39.48 | mchou | should work |
| 00:39.56 | mchou | at least in US. |
| 00:40.05 | mchou | dunno about elsewhere |
| 00:40.15 | adrian | who can i do that ? |
| 00:40.35 | jaytee | who can? you can! |
| 00:40.39 | mchou | lol |
| 00:40.41 | mchou | wtf |
| 00:40.48 | adrian | lol :P |
| 00:40.48 | adrian | ok |
| 00:40.50 | mchou | who's on first |
| 00:41.34 | jaytee | adrian, if you want to dial 555-1456 but want callerID blocked you'd dial *675551456 |
| 00:41.39 | drmessano | oh |
| 00:41.40 | drmessano | my god |
| 00:41.42 | drmessano | short |
| 00:41.45 | drmessano | sentences |
| 00:41.46 | jaytee | in the US |
| 00:41.49 | drmessano | really annoy |
| 00:42.20 | drmessano | [20:38] <mchou> try to speak comprensible english pls <-- What does that even MEAN? |
| 00:42.26 | *** join/#asterisk onats (n=Juanito@unaffiliated/onats) |
| 00:42.46 | mchou | drmessano: so bust my spelling |
| 00:42.56 | jaytee | comprehensible |
| 00:42.57 | mchou | comprehensible* |
| 00:43.08 | mchou | drmessano: you happy? |
| 00:43.27 | adrian | mchou: thanks |
| 00:43.42 | jaytee | I can deal with a little spelling error here or there. it's when the grammar gets raped and mutilated that I start getting irritated. |
| 00:44.39 | drmessano | What is comprehsible english? |
| 00:44.44 | drmessano | errr yeah |
| 00:44.48 | drmessano | What is comprehensible english? |
| 00:44.56 | mchou | lol |
| 00:45.04 | drmessano | Comprehensive? |
| 00:45.08 | drmessano | Comprehensible? |
| 00:45.09 | mchou | drmessano: speak for yourself, my man |
| 00:45.13 | jaytee | it's more clearly understandable than "brokun engrish" |
| 00:45.27 | drmessano | What the fuck are you even trying to say? |
| 00:45.44 | drmessano | There is no such thing as "comprehensible english"" |
| 00:45.45 | jaytee | me? |
| 00:45.49 | drmessano | No |
| 00:45.51 | drmessano | Mchou |
| 00:46.03 | drmessano | He wants to joke about my mispelling his made-up word |
| 00:46.13 | drmessano | But he can't even tell me what it's supposed to mean |
| 00:46.26 | jaytee | if there's no such thing as comprehensible english then how am I able to comprehend anything you type? |
| 00:46.27 | drmessano | Comprehendable? |
| 00:46.40 | drmessano | comprehensible doesn't exist |
| 00:46.42 | mchou | drmessano: http://www.merriam-webster.com/cgi-bin/dictionary?book=Dictionary&va=comprehensible |
| 00:46.59 | mchou | drmessano: look it up my man |
| 00:47.09 | drmessano | http://www.merriam-webster.com/dictionary/comprehendible |
| 00:47.15 | drmessano | No, you look it up "my man" |
| 00:48.05 | mchou | drmessano: when's the last time you made a meaning contribution to this channel instead of peanut gallery bullshit you always engage in? |
| 00:48.22 | mchou | meaningful* |
| 00:48.30 | jaytee | look, if you two don't stop fighting over adjectives I'm going to have to whip out a deadly adverb and beat you both over the head with it. |
| 00:48.32 | drmessano | When was the last time you stopped acting like a flaming douchebag? |
| 00:48.55 | mchou | drmessano: [17:43] <adrian> mchou: thanks |
| 00:49.14 | mchou | drmessano: I assume adrian wasnt speaking sarcastically |
| 00:49.19 | drmessano | [11:37] <Founderisk> mchou: You're a dick |
| 00:49.24 | drmessano | Point? |
| 00:49.39 | drmessano | I help people all the time.. not like I need to justify it to you |
| 00:49.44 | drmessano | So go back to your hole |
| 00:50.27 | mchou | drmessano: nor do I have to justify my comments here which werent even direct to you in the first place |
| 00:50.32 | drmessano | Besides, all you did was tell him how to dial *67.. you consider that meaningful? |
| 00:50.36 | drmessano | My cat can do that |
| 00:50.44 | drmessano | mchou: But you DID |
| 00:50.48 | rob0 | Catserisk |
| 00:50.58 | drmessano | ~asteriskcat |
| 00:50.58 | jbot | somebody said asteriskcat was not amused |
| 00:50.59 | mchou | drmessano: dude, someone asked. Did your cat answer? |
| 00:51.09 | drmessano | mchou: No, another pussy did |
| 00:51.13 | mchou | right |
| 00:51.14 | drmessano | mchou: Same difference.. |
| 00:51.29 | mchou | more peanut gallery bullshit |
| 00:52.06 | drmessano | Remind me to ask you sometime how to fix a dialplan.. I'd rather just get a skype phone.. |
| 00:52.14 | mchou | try to be an adult instead of trolling in your mom's basement |
| 00:52.27 | onats | what's a good SIP phone for home? |
| 00:52.29 | drmessano | *attic* |
| 00:52.34 | drmessano | Not basement |
| 00:53.02 | drmessano | onats: Linksys SPA921 |
| 00:54.10 | mchou | drmessano: yeah, my cat knows that too |
| 00:54.23 | drmessano | Sorry, you don't have permission to speak to me |
| 00:55.02 | drmessano | Move along little n00b |
| 00:55.02 | mchou | haha |
| 00:55.02 | shido6 | get a pap and a uniden cordless :) |
| 00:55.02 | mchou | drmessano: legend in your own mind, eh? |
| 00:55.21 | drmessano | mchou: No, unlike you, I actually know something about asterisk.. So my legend extends beyond my attic |
| 00:55.28 | drmessano | mchou: But thanx for playing |
| 00:55.45 | onats | drmessano, ok, how much is that from where you are? |
| 00:56.00 | drmessano | onats: Amazon.com has the best price, believe it or not |
| 00:56.20 | drmessano | onats: I found all the Linksys stuff on Amazon cheaper than most VoIP stores |
| 00:56.32 | mchou | drmessano: yeah right, u dont know jack about asterisk jusging by the fact you havent made any contributions to this channel lately |
| 00:56.48 | drmessano | mchou: You're so smart :) |
| 00:56.56 | drmessano | mchou: Keep trying :) |
| 00:57.03 | mchou | drmessano: you've far outlived any possible usefulness you might once have had |
| 00:57.28 | drmessano | mchou: Don't be so hard on yourself :) |
| 00:57.46 | drmessano | mchou: Maybe there is a nice asterisk book you can read :) |
| 00:57.57 | mchou | drmessano: you're a living fossil even no paleotonolgist wouldnt give a shit about |
| 00:58.12 | drmessano | onats: The SPA941 may be a better value for it's money though.. $88 I think |
| 00:58.20 | mchou | drmessano: total waste of protoplasm |
| 00:58.38 | drmessano | mchou: Flattery will get you nowhere with me :) |
| 00:58.51 | drmessano | mchou: You did spell protoplasm correct.. well done |
| 00:59.42 | mchou | drmessano: yeah, you add real value as my spell checker |
| 01:00.23 | mchou | drmessano: maybe I was wrong about your meaningful contributions on this channel after all. ... NOT |
| 01:00.54 | drmessano | mchou: Oh, how cute.. using "NOT" at the end of a sentence.. that's so 1996. :) |
| 01:01.24 | drmessano | mchou: You may not know anything about asterisk, but your tries are adorable :) |
| 01:01.41 | mchou | drmessano: yeah, apparently you're a real armchair historian too |
| 01:01.50 | rob0 | sighs |
| 01:02.03 | drmessano | lol |
| 01:03.10 | drmessano | wow.. How do I counter that.. well, mchou, you're a real circus acrobat too?!? |
| 01:03.30 | jaytee | when i grow up i wanna be a fireman |
| 01:03.38 | drmessano | I'm a historian! |
| 01:03.44 | drmessano | Look at me.. ! Weeee! |
| 01:03.54 | rob0 | turns a high-pressure hose on #asterisk |
| 01:03.55 | drmessano | jaytee: You can be a picnic basket |
| 01:03.56 | mchou | drmessano: heh, you're creativity running out of colorful invective? That's so unlike you |
| 01:03.59 | rob0 | psssssssssssssssssssssssss |
| 01:04.16 | rob0 | jaytee, join in, it's fun! |
| 01:04.28 | drmessano | mchou: Oh, you called me an armchair historian.. How can I possibly combat such an effective dig |
| 01:04.50 | drmessano | You're a deputy coroner! |
| 01:05.13 | mchou | drmessano: clearly trhe subtlety was too much for your cromagnon asterisk head to grasp |
| 01:05.19 | mchou | the* |
| 01:05.23 | heedly | haha |
| 01:05.31 | heedly | this is my new favorite channel |
| 01:05.36 | jaytee | i would think studying the history of armchairs would be a bit dull |
| 01:05.42 | drmessano | You need to retype that.. I have no idea where the "the" was supposed to go |
| 01:05.51 | drmessano | I'll wait.. |
| 01:06.00 | mchou | remember it was in the context of you making meaningful contributions in this channel |
| 01:06.12 | drmessano | mchou: Still lost me.. retype please |
| 01:06.24 | drmessano | You're slowing the momentum |
| 01:06.47 | drmessano | drmessano: clearly trhe subtlety was too much for your cromagnon asterisk head to grasp the <--- At the end? |
| 01:06.51 | mchou | drmessano: I knew you were lost long time ago |
| 01:07.32 | drmessano | mchou: this is the part where you go all emo and stomp off, your cleverness wore off about 30 lines ago.. |
| 01:07.37 | drmessano | Now run along little one |
| 01:07.39 | mchou | lol |
| 01:07.49 | jblack | what's an armchair historian? |
| 01:07.49 | mchou | the only emo here is you, drmessano |
| 01:08.02 | drmessano | jblack: mchou won't tell me |
| 01:08.10 | mchou | nobody is about to stomp off here. |
| 01:08.11 | drmessano | jblack: I'd rather be a fire truck |
| 01:08.22 | drmessano | jblack: or.... a movie star! |
| 01:08.54 | drmessano | mchou: Stop being such an occupational theraptist |
| 01:09.01 | rob0 | I like my armchair, but it doesn't have a very long history. |
| 01:09.18 | jblack | Is that like a 'Dr Phil' / |
| 01:09.27 | drmessano | rob0: I hear I am qualified to study it.. may I have a look? |
| 01:09.41 | rob0 | sure |
| 01:10.14 | jblack | mchou: Can I be qualified in something? I'm jealous that drmessano is an expert in armchair history. |
| 01:10.31 | drmessano | jblack: you can be captain america! |
| 01:10.35 | jblack | mumbles something about the wizard of oz. |
| 01:11.16 | jblack | awww. he stomped off in an emo rage. |
| 01:11.33 | drmessano | jblack: It's not emo, it's "new wave" |
| 01:11.45 | mchou | jblack: oh, you referring to drmessano? |
| 01:11.51 | jblack | death metal, goth, emo, new wave, what's the diff? |
| 01:12.02 | jblack | mchou: I want to have a qualification bestowed upon me! |
| 01:12.08 | drmessano | <mchou> jblack: oh, you referring to drmessano? <-- Hey, hey.. THAT was snappy! |
| 01:12.29 | drmessano | Here I was thinking mchou wasn't witty.. I FAIL'ed |
| 01:12.44 | drmessano | jblack: I know YOU are.. but what am I? |
| 01:12.57 | jblack | I don't know! i'm not qualified to say! |
| 01:13.06 | drmessano | jblack: :( |
| 01:13.07 | mchou | jblack: you dont need no stinking qualifications. That belongs exclusively to drmessano |
| 01:13.13 | rob0 | Someone said I was a wit, but he was only half right. |
| 01:13.24 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 01:13.24 | drmessano | "no stinking".. ohh.. I have goosebumps |
| 01:13.50 | jblack | Now I feel depressed. I should steal my kid's black eyeliner |
| 01:14.01 | *** join/#asterisk cryptnix (n=andrew@65.183.179.26) |
| 01:14.01 | mchou | jblack: drmessano clearly thinks he is an authority on all things now that he has mastered asterisk |
| 01:14.03 | jblack | make some pop music about feelings. |
| 01:14.13 | jblack | mchou: He is. |
| 01:14.31 | rob0 | Masterisk! |
| 01:14.42 | drmessano | rob0: Thats completely disgusting |
| 01:14.58 | drmessano | oh wait, I see what you did there |
| 01:15.03 | mchou | jblack: that's why drmessano gets the qualifications |
| 01:15.19 | jblack | You should bow to him. |
| 01:15.24 | drmessano | mchou: I wish I had as many cerifications as you, but the last one requires 3 signatures :) |
| 01:15.30 | mchou | jblack: at least the so called qualifications anyways |
| 01:15.44 | mchou | drmessano: lol |
| 01:16.23 | mchou | drmessano: I have zip certifications, but I do have degrees. And they dont come from DeVry :) |
| 01:16.32 | [TK]D-Fender | drmessano: Yes and as soon as 3 of your relatives gain enough literacy sign their names with somthing OTHER than a big "X" your days are numbered! |
| 01:16.34 | jblack | so, in all seriousness, is there any particiular reason you're acting like a dork, mchou? |
| 01:16.38 | *** join/#asterisk Kira[work] (n=chatzill@210.176.243.218) |
| 01:16.49 | [TK]D-Fender | zing! |
| 01:16.51 | drmessano | [TK]D-Fender: Shhhh.. Don't give anyone any ideas! |
| 01:16.58 | rob0 | I showed up at a restaurant tonight. The Maitre d' asked if I had reservations. I said, "Sure, but I stopped here anyway!" |
| 01:17.08 | jblack | nice |
| 01:17.09 | drmessano | rob0: Ba-dump, ching! |
| 01:17.11 | [TK]D-Fender | rob0: Good one |
| 01:17.19 | Kira[work] | Good day. |
| 01:17.33 | mchou | jblack: I'm acting like a dork? why dont you scroll up in the channel log and see when drmessano jumped in. |
| 01:17.44 | drmessano | That's mature |
| 01:17.47 | jblack | I would, but I'm not a historian. |
| 01:17.51 | drmessano | "Because HE is..." |
| 01:17.54 | jblack | The paste is opaque to me. |
| 01:17.57 | rob0 | Thank you, I'll be performing here all week (or until the next netsplit kicks me out.) |
| 01:19.25 | WimpMan | . o O ( Wouldn't it be great if people who think their pets are cleverer than other people would leave their keyboards to their pets? ) |
| 01:19.45 | drmessano | "more clever" |
| 01:20.12 | [TK]D-Fender | WimpMan: CAN HAZ KEE-BORED PLZ?! |
| 01:20.12 | heedly | "cleverer" |
| 01:20.20 | drmessano | "clevered" |
| 01:20.20 | heedly | you misquoted.. |
| 01:20.26 | heedly | again! |
| 01:20.31 | drmessano | heedly: Sorry :( |
| 01:20.34 | heedly | it's ok. |
| 01:20.52 | drmessano | heedly: I was distracted... Busy Masterisking over here |
| 01:21.09 | heedly | it's ok. |
| 01:21.12 | drmessano | I won't be much longer.. I have a short dialplan |
| 01:21.32 | heedly | pat pats drmessano |
| 01:21.33 | heedly | it's ok. |
| 01:21.37 | drmessano | :( |
| 01:21.41 | heedly | :) |
| 01:21.44 | rob0 | My wife says the length doesn't matter. Go figure. |
| 01:22.14 | drmessano | rob0: Yoru length doesn't matter.. but your neighbor is huge |
| 01:22.14 | heedly | they most delicate parts are on the outside.. |
| 01:22.26 | heedly | I never realized how people don't get that. |
| 01:22.26 | drmessano | Your* |
| 01:22.40 | rob0 | Yeah, you should see the way she smiles at him. |
| 01:23.18 | drmessano | My ex-wife told me "it's not the size of the wave, it's the motion of the ocean".. She left me for an oceanographer.. :( |
| 01:24.19 | [TK]D-Fender | rob0: drmessano hey, whatever floats her boat.. |
| 01:24.30 | drmessano | sobs |
| 01:24.44 | Kira[work] | Strange |
| 01:25.32 | Kira[work] | one of my colleague's SIP used to work, but her SIP won't register with the server anymore ever since I swapped the office firewall for a new one yesterday. |
| 01:25.39 | Kira[work] | My own SIP still works though. :S |
| 01:30.58 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:31.53 | trnzmeta | guys: my input and output streams are split into two in my /var/spools/monitor directory |
| 01:32.21 | trnzmeta | for my voice recordings, what do I need to do to enable them as one file |
| 01:33.33 | [TK]D-Fender | trnzmeta: "core show application monitor" |
| 01:34.17 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 01:37.13 | trnzmeta | ta :0 |
| 01:39.22 | *** join/#asterisk seanmh (n=johndoe@c-68-35-21-64.hsd1.nm.comcast.net) |
| 01:39.30 | *** join/#asterisk chendy (n=chatzill@58.61.196.45) |
| 01:40.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:42.51 | *** join/#asterisk _ShrikE (n=_ShrikE-@64.128.34.118) |
| 01:55.48 | *** join/#asterisk seanmh (n=johndoe@c-68-35-21-64.hsd1.nm.comcast.net) |
| 01:58.25 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 02:04.20 | *** join/#asterisk logicwrath (n=no@68.41.24.98) |
| 02:04.43 | logicwrath | can someone tell me which file has the register statement in AsteriskNOW |
| 02:06.12 | [TK]D-Fender | logicwrath: Probably users.conf |
| 02:06.33 | logicwrath | i reviewed that conf and there is no register statement and I dont see it in sip.conf either |
| 02:06.37 | [TK]D-Fender | logicwrath: And it isn't a clear statement. It runs off of the peer auth + "registersip=yes", etc |
| 02:06.56 | [TK]D-Fender | logicwrath: Hybridized BS |
| 02:08.46 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 02:08.54 | logicwrath | i thought i had to manually enter a register line.. ill review more |
| 02:09.38 | logicwrath | i thought there would be a register => line somewhere |
| 02:09.53 | Qwell | nope, the register is built from other items in the config |
| 02:10.04 | Qwell | (if you have registersip=yes, or registeriax=yes, or whatever tech) |
| 02:10.20 | *** join/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 02:10.27 | logicwrath | does it pass in this format: phonenumber>@sip.broadvoice.com:<password> |
| 02:10.38 | logicwrath | with :password |
| 02:11.19 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 02:19.06 | [TK]D-Fender | logicwrath: user:pass@host <- |
| 02:19.39 | Qwell | user:pass@host</something that starts with a c...what option is that?> |
| 02:19.47 | Qwell | erm, [], not <> |
| 02:19.53 | Qwell | contact? |
| 02:20.09 | Qwell | user:pass@host[/contact] |
| 02:25.53 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 02:27.03 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 02:43.33 | Kira[work] | Something is really strange about our asterisk configuration. The asterisk server is behind a NAT firewall, yet the configuration files do not show much evidence of NAT-oriented settings, aside from externip. |
| 02:44.02 | Kira[work] | I thought other settings were supposed to be used in conjunction with externip. :S |
| 02:45.51 | [TK]D-Fender | ~sipnat |
| 02:45.51 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:45.53 | [TK]D-Fender | ^^^^^^^^^^^ |
| 02:45.54 | *** join/#asterisk newbie___ (n=Some_guy@60.49.23.190) |
| 02:54.57 | jeev | Qwell |
| 02:55.00 | jeev | how's the redbull supply |
| 02:58.25 | oilinki | good morning gentlemen |
| 02:58.58 | Qwell | damn, I just had the "all your base" theme song roll through my head... |
| 02:59.22 | jaytee | it had a theme song? |
| 02:59.27 | Qwell | the video did |
| 02:59.34 | Qwell | maybe it wasn't a theme song. whatever |
| 02:59.47 | Qwell | jeev: don't know |
| 03:00.47 | *** join/#asterisk luxxx77 (n=luxxx77@g228071058.adsl.alicedsl.de) |
| 03:01.34 | *** join/#asterisk hatoon (n=musis@189.71.123.101) |
| 03:08.20 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 03:08.26 | jeev | ah |
| 03:08.47 | *** join/#asterisk logictelecom (n=TieFight@201.36.50.5) |
| 03:11.17 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 03:12.29 | *** join/#asterisk Logictelecom (n=Logictel@201.36.50.5) |
| 03:13.55 | cryptnix | all your base are belong to us |
| 03:33.21 | Nugget | what you say?!?! |
| 03:34.21 | [TK]D-Fender | You have no chance to survive. Make your time |
| 03:37.37 | trnzmeta | guys: ever heard of asterisk automatically cutting over to say IVR queue on long call |
| 03:37.58 | trnzmeta | talking talking talking then cuts over to call queue |
| 03:40.38 | [TK]D-Fender | trnzmeta: huh? |
| 03:41.02 | trnzmeta | not sure how to explain it |
| 03:41.12 | trnzmeta | I get transfered a call, pick it up |
| 03:41.21 | trnzmeta | talk for a while, then out of the blue |
| 03:41.47 | trnzmeta | the call will cut off and goto a queue |
| 03:42.34 | [TK]D-Fender | trnzmeta: makes no sense. pastebin CLI output with debug to back it |
| 03:43.21 | *** part/#asterisk xeet2 (n=no@dsl092-052-249.phl1.dsl.speakeasy.net) |
| 03:44.02 | *** join/#asterisk Roge152 (n=roger@stjhnf0124w-142162082245.pppoe-dynamic.nl.aliant.net) |
| 03:47.47 | logicwrath | should my extension show up in this error and not be '' ...ie: Call from '' to extension '########' rejected because extension not found. |
| 03:48.09 | logicwrath | ive been trying to get outgoing calls to work for a while now |
| 03:48.48 | logicwrath | i feel like its ignoring my dialing rules |
| 03:49.22 | [TK]D-Fender | logicwrath: You clearly don't have an extension to match in the context the call falls under. |
| 03:51.05 | logicwrath | exten=_8XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid}) |
| 03:51.05 | logicwrath | comment=_8XXXXXXX!,1,outgoing,standard |
| 03:51.15 | logicwrath | that was autogenerated |
| 03:51.22 | logicwrath | by the gui |
| 03:51.31 | logicwrath | the 8 isnt getting matched? |
| 03:51.36 | [TK]D-Fender | logicwrath: PASTEBIN <---- |
| 03:51.48 | [TK]D-Fender | logicAnd that doesn't prove what was dialed or in what context. |
| 03:51.50 | logicwrath | i thought pastebin was for more than 2 lines |
| 03:52.32 | [TK]D-Fender | logicwrath: Show us the FULL CLI output including SIP debug if you call originates from a SIP device and your COMPLETE dialplan. |
| 03:52.53 | logicwrath | ok, give me a min thanks |
| 03:59.22 | logicwrath | http://pastebin.com/d64716251 |
| 03:59.36 | logicwrath | if you need more CLI let me know, i have core debug set to 10 and i cannot use clipboard |
| 04:01.05 | logicwrath | [numberplan-custom-1] is on the user extension |
| 04:02.19 | [TK]D-Fender | logicwrath: I said SIP DEBUG |
| 04:03.10 | [TK]D-Fender | logicwrath: redo your pastebin and do not mask ANY of it. |
| 04:04.31 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:06.50 | logicwrath | is the CLI logged to a file i can open? |
| 04:10.29 | [TK]D-Fender | logicwrath: No, thats what SCROLL-BACK is for |
| 04:20.10 | logicwrath | ok i figured out how to connect with putty and paisted sip debug here http://pastebin.com/d38341300 |
| 04:22.25 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 04:23.56 | *** join/#asterisk osiris (n=osiris@c-71-205-9-42.hsd1.mi.comcast.net) |
| 04:25.34 | [TK]D-Fender | logicwrath: Found no matching peer or user for '10.0.0.63:50348' <- #1 it isn't IDing your phone. Because of that its looking is [default] <- Looking for 815868992959 in default (domain 10.10.0.14) |
| 04:26.01 | [TK]D-Fender | logicwrath: and clearly there is no exten in [default] to match |
| 04:30.57 | logicwrath | is my phone not authenticating properly with Asterisk? I can dial internally |
| 04:31.07 | *** part/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 04:31.17 | logicwrath | should I be looking at the extension setup or the phone config |
| 04:32.14 | Kira[work] | Is it possible to place asterisk behind a NAT firewall while catering for clients who needs to frequently _move_ among the same NAT network, the public Internet without NAT, and another NAT network? |
| 04:34.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 04:35.02 | Kira[work] | (I also don't want the user to have to change the SIP server address setting of his SIP client software as he moves between the local network and outside.) |
| 04:35.23 | [TK]D-Fender | logicwrath: No, it is not authenticating properly. |
| 04:36.12 | [TK]D-Fender | Kira[work]: his target shouldn't be moving (your *) |
| 04:36.26 | *** join/#asterisk dignome (n=dignome@216-82-199-27.dyn.grandenetworks.net) |
| 04:36.32 | [TK]D-Fender | logicwrath: And that'd be your phone setup. |
| 04:36.40 | [TK]D-Fender | logicwrath: extensions = dialplan. |
| 04:37.03 | Kira[work] | [TK]D-Fender: exactly, I want his client software to pretend to connect to the public IP of the SIP server even when it is actually behind the same NAT firewall. |
| 04:37.19 | [TK]D-Fender | Kira[work]: Should be doable. |
| 04:38.04 | dignome | sigh. i finally figured out my pri problem - i don't have a pri... i have a channel bank. |
| 04:38.42 | Kira[work] | [TK]D-Fender: having "nat=yes" in the user's dialplan at all times should work regardless of where he is right? |
| 04:42.40 | [TK]D-Fender | Kira[work]: that is not a DIALPLAN option. And if the option is set on the peer * will respond to the IP the the call came in under. |
| 04:42.49 | [TK]D-Fender | dignome: SMRT |
| 04:43.33 | dignome | lol. i'm such a noob. |
| 04:44.42 | drmessano | ZOMG |
| 04:45.03 | drmessano | What a coincidence |
| 04:45.10 | drmessano | I figured out my asterisk problem |
| 04:45.22 | drmessano | I'm not even running ASTERISK.. I have SKYPE.. crap |
| 04:48.33 | [TK]D-Fender | ~cluebat drmessano |
| 04:48.37 | jbot | ACTION pulls out a ClueBat (tm) and thwaps drmessano. |
| 04:55.43 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
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| 05:02.57 | *** join/#asterisk nix8n82 (n=nate@63.162.28.92) |
| 05:06.24 | hi365_m | sees drmessano is till at it |
| 05:11.21 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 05:12.17 | drmessano | nope |
| 05:14.02 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 05:14.39 | [TK]D-Fender | ok, bed time. Later all. |
| 05:14.57 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
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| 06:58.00 | *** join/#asterisk beernuts (n=mattfox7@123.211.167.79) |
| 07:00.09 | beernuts | hello. Need some help here, I have 2 boxes connected using an IAX trunk, I am routing incoming Zap calls through the IAX trunk to the other box but I am getting lots of choppiness. The boxes have 1 gb nics on a LAN with no errors. The codec over the IAX is gsm and alaw is even worse even though the zap (E1) is alaw. The load average is around 3. Does anyone have any ideas? |
| 07:01.07 | beernuts | bu the way on the 2nd box I am routing out Zap to agents thus ZAP (alaw) > IAX (gsm) > ZAP (alaw) |
| 07:01.53 | heedly | how many processors/cores? |
| 07:02.02 | beernuts | dual core |
| 07:02.10 | heedly | I don't know IAX, any way to not do the conversion. |
| 07:02.15 | heedly | well then 3 is pretty high. |
| 07:02.49 | beernuts | when I did alaw over the iax it was even worse |
| 07:03.09 | heedly | oh, ic. |
| 07:03.29 | heedly | how is the ping latency? |
| 07:03.58 | heedly | The only other thing I can think of is increased frameing size. |
| 07:04.38 | heedly | When the box load is under 2 is it better? |
| 07:05.22 | beernuts | like .05 ms |
| 07:05.24 | *** join/#asterisk onats (n=Juanito@unaffiliated/onats) |
| 07:05.38 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-3a56d7cf6a0f9af1) |
| 07:05.59 | beernuts | its down to 1 ill just check |
| 07:06.14 | beernuts | yeah there are less pops and chop def |
| 07:06.46 | beernuts | I guess I can pretty much say it is cpu |
| 07:07.21 | beernuts | I wonder why it would be worse when using the same codec as the zap channel though |
| 07:08.25 | beernuts | its just odd cause I average around 40 iax with 20 zap which isn't much at all |
| 07:14.06 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
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| 07:17.01 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:20.26 | BugKhaM | why isn't autodomain=yes working in sip.conf? |
| 07:22.45 | angryuser | BugKhaM: not sure, maybe broken or deprecated ? |
| 07:25.50 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
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| 07:29.21 | *** join/#asterisk hyphenex (n=scott@nickstallman.net) |
| 07:29.54 | hyphenex | G'Day. Is there a protocol I could use to make phone calls on my asterisk box through a HTTP proxy server (while I'm living on campus at uni and my asterisk box is at home) |
| 07:30.59 | hyphenex | or a socks proxy I guess? |
| 07:31.01 | tzafrir_laptop | hyphenex, yes: http://openvpn.org/ |
| 07:31.57 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-50.dsl.lsan03.sbcglobal.net) |
| 07:34.40 | mandh | tzafrir_laptop, by the way can we pass openvpn through proxies "http" ?1 |
| 07:35.01 | tzafrir_laptop | mandh, IIRC it does in some way |
| 07:35.19 | mandh | openvpn? |
| 07:35.20 | tzafrir_laptop | But generally there are ways to create tunnels over http |
| 07:35.28 | mandh | yeah |
| 07:35.42 | tzafrir_laptop | I'm not sure what will be the resulting latency and jitter, though |
| 07:36.16 | hyphenex | it's only for a few phone calls... I know Skype can do it rather well, so I was hoping Asterisk would have a protocol I could use that does a similar job? |
| 07:37.31 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-b996d31acbc72ba2) |
| 07:40.16 | hyphenex | Is there a protocol that uses TCP? That could be an easy one to forward |
| 07:40.39 | hyphenex | because I know I can tunnel TCP through HTTP proxies |
| 07:44.20 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:47.05 | jblack | There is a tool that tunnels through http proxyies. |
| 07:47.11 | jblack | It's found with a simple google. |
| 07:47.52 | jblack | http://www.nocrew.org/software/httptunnel.html |
| 07:47.55 | hyphenex | jblack: I know about desproxy that tunnels TCP connections through http proxies |
| 07:48.06 | hyphenex | but IAX and SIP are UDP, does not help me |
| 07:48.29 | jblack | You do httptunnel, then pptpd over httpdtunnel. |
| 07:48.48 | hyphenex | pptpd over httpdtunnel? |
| 07:49.00 | jblack | Yeah. Typical weekend project. |
| 07:49.15 | hyphenex | Ahhk... I don't think I'll be doing that any time soon then |
| 07:49.27 | hyphenex | contemplates just getting Skype :P |
| 07:49.36 | jblack | Ok. Get skype. |
| 07:50.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:50.25 | hyphenex | there's a waste of $16 credit... I wish you could just use like $2 or something |
| 07:50.49 | jblack | You can be cheap, or you can be lazy. Not both. |
| 07:51.26 | hyphenex | hey, there are such things as UDP to TCP bridges... |
| 07:51.31 | hyphenex | I could probably just use this |
| 07:52.25 | jblack | Whatever you want. You probably do want to use pptpd though. |
| 07:52.41 | hyphenex | wouldn't just a TCP->UDP bridge be faster? |
| 07:52.47 | jblack | that'll give you full vpn over the http proxy. |
| 07:53.22 | jblack | You dont' understand what httptunnel does yet. |
| 07:54.35 | jblack | never mind. That might work |
| 07:57.59 | bkw_ | what are you trying to do? |
| 07:58.05 | *** join/#asterisk xenonex (n=xenonex@89.218.236.76) |
| 07:58.45 | hyphenex | just make a phone call through my asterisk box at home (or bypass that and go directly to a SIP server on the net) through a HTTP proxy here at uni |
| 07:58.54 | hyphenex | I live on campus now |
| 08:00.09 | jblack | pptp would be perfect for you. Why are you scared of it? |
| 08:00.48 | hyphenex | because it's a weekend project :P |
| 08:00.57 | hyphenex | ie. take more then an hour to set up? :P |
| 08:01.20 | jblack | so you spend one weekend, and you solve the problem for 4 years? Sounds like a good deal to me |
| 08:02.32 | bkw_ | hyphenex: if it takes you longer than an hour to setup.. you have problems :P |
| 08:02.38 | bkw_ | hyphenex: what is your goal btw? |
| 08:03.06 | jblack | wait.You're tell him that longer than an hour means he has probems, but you don't know the what? |
| 08:03.14 | hyphenex | will it work with an OSX client? |
| 08:03.25 | bkw_ | setting up pptpd :P |
| 08:03.35 | bkw_ | pptpd works with OS X |
| 08:03.38 | jblack | pptp works with linux and windows. I can't afford macs. |
| 08:03.57 | hyphenex | wait, I could just use hamachi for this and save myself half the time, right? |
| 08:04.04 | bkw_ | you could yes |
| 08:04.12 | bkw_ | hamachi on OS X blows |
| 08:04.15 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:04.45 | hyphenex | awesome :) |
| 08:05.04 | jblack | guess he likes blow jobs |
| 08:05.06 | hyphenex | then I should just be able to connect right up to my server, and that would be awesome (and take like two seconds to get working) |
| 08:05.14 | hyphenex | slippin sweet :) |
| 08:05.18 | hyphenex | flippin! |
| 08:06.24 | jblack | I don't feel good. |
| 08:07.09 | hyphenex | but to get free-proxy internet, the traffic would have to be routed through a SSH tunnel, through HTTP proxy in the US, through hamachi back to my box at home, then out to the SIP call servers |
| 08:07.18 | hyphenex | me thinks that could be a lot of latency |
| 08:08.00 | *** join/#asterisk kraypius (n=kraypius@c-67-167-121-213.hsd1.il.comcast.net) |
| 08:08.42 | kraypius | hello, I am having a problem. i try to start asterisk. it doesnt give any errors but when I look at running processes there is no asterisk.. here is output: http://pastebin.ca/1191257 |
| 08:08.54 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 08:09.30 | jblack | kraypius: Hmm |
| 08:09.37 | jblack | You're the second one today. |
| 08:09.48 | kraypius | heh |
| 08:10.06 | jblack | Try adding -f |
| 08:10.43 | bkw_ | hyphenex: so you're wanting to route voip over a tcp connection? |
| 08:11.06 | hyphenex | bkw_: basically (in order to pipe VoIP through a HTTP Proxy) |
| 08:11.14 | kraypius | jblack, still no asterisk |
| 08:11.56 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 08:12.00 | jblack | If you're running asterisk -f, it should stay running on the console. |
| 08:12.14 | jblack | If it's dying, there should be some indication, even if it's a segfault. |
| 08:12.20 | bkw_ | hyphenex: voip over something like that while it might work sometimes... if a packet is lost it will go to shit in short order |
| 08:12.41 | kraypius | jblack, this will show me what I need? asterisk -vvvgcf |
| 08:12.45 | jblack | I thought iax2 had a tcp option, but I can't find it. |
| 08:12.57 | jblack | kraypius: It may |
| 08:13.00 | bkw_ | jblack: not that I ever seen |
| 08:13.25 | hyphenex | iax2 TCP would be ideal, but I couldn't find that option either |
| 08:13.39 | bkw_ | hyphenex: do you know if DNS traffic gets out? |
| 08:13.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:13.51 | bkw_ | or do they hijack that to? |
| 08:13.59 | bkw_ | I have run OpenVPN on port 53 before |
| 08:14.17 | bkw_ | hell even running IAX on port 53 works too |
| 08:14.18 | jblack | tunneled over dns a couple times. It's not fun. |
| 08:14.22 | *** join/#asterisk Levonk (n=lk@adsl-75-62-138-221.dsl.lsan03.sbcglobal.net) |
| 08:14.48 | hyphenex | bkw_: I think they might block that too... although, I can ping things such as google.com.au :) |
| 08:14.51 | bkw_ | jblack: it shouldn't differ one bit from running it on any other port |
| 08:14.51 | hyphenex | that's a good sign, no? |
| 08:15.02 | kraypius | jblack, http://pastebin.ca/1191263 |
| 08:15.14 | bkw_ | hyphenex: it might be try running ssh on port 53 on the far side |
| 08:15.17 | jblack | nstx does it's magic by doing _dns requests_. Of course it's much slower. |
| 08:15.37 | jblack | bkw_: Are you big on making statements tonight that you're not sure are right? That's not like you. |
| 08:15.47 | bkw_ | jblack: what? |
| 08:16.21 | bkw_ | running IAX or OpenVPN on port 53 if its not filter will work. |
| 08:16.22 | jblack | "it shouldn't differ one bit from running it on any port". Of course running something over nstx would be slower. |
| 08:16.30 | bkw_ | nstx? |
| 08:16.33 | jblack | Oh. Sorry. THought you were replying to what I said. |
| 08:16.38 | jblack | nstx is a tunnel over dns. |
| 08:16.50 | bkw_ | no I didn't mean that.. if you're lucky they don't filter port 53 |
| 08:16.52 | kraypius | this sux |
| 08:17.00 | jblack | If you can do dns requests, you can tunnel. =) |
| 08:17.07 | bkw_ | depends |
| 08:17.14 | bkw_ | jblack: some evilness can happen if they hijack it |
| 08:17.25 | bkw_ | and force all dns traffic to their nameservers locally |
| 08:17.38 | bkw_ | but its a crap shoot if they do or don't |
| 08:17.53 | jblack | kraypius: Ok. so, your next step is to strace asterisk -f . Then, valgrind asterisk -f. Then, build asterisk with debugging symbols, and run it through gdb. |
| 08:18.01 | hyphenex | would it be quicker tunneling over DNS then it would be tunneling over hamachi then? |
| 08:18.06 | jblack | That's the thing. It works with _their_ dns servers. They're valid dns requests. |
| 08:19.02 | jblack | It's the funkiest thing I've ever seen, but it works. |
| 08:19.34 | jblack | http://thomer.com/howtos/nstx.html |
| 08:20.23 | jblack | Oh, I see there is now an icmptx too. |
| 08:22.34 | BugKhaM | I enabled the autodomain=yes to allow local IPs to be added to the domain list in sip.conf, but that just doesn't work |
| 08:23.23 | BugKhaM | I even added domain=192.168.1.10 manually (the local ip is 192.168.1.10) but it just didn't work either |
| 08:23.50 | BugKhaM | I am using Asterisk 1.4.21.2 |
| 08:24.05 | hyphenex | so this method would be faster, and work better with VoIP then the hamachi method would? |
| 08:24.23 | jblack | I don't know what hamachi does. |
| 08:24.56 | jblack | I also don't know about os x to evaluate |
| 08:25.40 | jblack | what college blocks all tcp/ip, excepting proxied http? |
| 08:25.41 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-70d35b1c521a7ca6) |
| 08:26.18 | hyphenex | jblack: it's a method where we only get 400MB a semester quota |
| 08:27.45 | jblack | You're willing to risk getting expelled for theft, to get phone calls? |
| 08:28.03 | hyphenex | heh |
| 08:28.24 | jblack | by the way, linode is cheaper than slicehost. |
| 08:28.38 | jblack | almost 1/2 the cost, for a xen virtual machine. |
| 08:28.59 | jblack | less ram, no backups |
| 08:29.18 | hyphenex | jblack: yeah, I'm already routing my traffic through a VPS in order to give me a tad more then 400MB ;) |
| 08:31.43 | jblack | by the way, pptp has compression |
| 08:33.13 | hyphenex | so pptp would be faster then the DNS method? |
| 08:33.29 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 08:33.44 | jblack | You can find the answer if you scroll back |
| 08:33.53 | jblack | wanders off. |
| 08:37.24 | *** join/#asterisk l8router (n=me@d122-109-92-165.sbr12.nsw.optusnet.com.au) |
| 08:41.35 | *** join/#asterisk kalel008 (n=tamato@209.203.41.26) |
| 08:41.42 | kalel008 | hi there |
| 08:42.03 | kalel008 | im busy reading the book and want to know something |
| 08:42.20 | jblack | That sounds like a great start. What's up? |
| 08:42.40 | kalel008 | the extensions doesnt nececary mean its the tel numbers right |
| 08:43.04 | jblack | correct. |
| 08:43.18 | jblack | half correct. sometimes they're like line numbers, other times, they're like phone numbers. |
| 08:43.21 | kalel008 | so if i want to define the incomming DID's do i do that in extensions.conf |
| 08:43.45 | jblack | Well, those incoming dids are going to come in either sip or iax, via account contexts. |
| 08:44.19 | jblack | In those accounts, you'll specify what context to send them too. What "extension" they end up depends upon the setup. |
| 08:44.34 | jblack | Wit sip, they tend to start on "s,1". With iax, they tend to show up on DIDNUMBER,1 |
| 08:44.51 | kalel008 | ooooh now it makes a bit of sence |
| 08:45.07 | jblack | Yeah. So, those extensions can, but don't need to line up with actual extensions. |
| 08:45.30 | jblack | FOr example, extension one on my main welcoming context Dials my sip line. So does 1000. |
| 08:46.13 | kalel008 | so if i have a iax trunk to my itsp i must define the DID'd in the iax.conf right |
| 08:46.20 | jblack | But both of those extensions use the Dial() extension, that dial the phones attached to various sip accounts. |
| 08:46.37 | kalel008 | and setup the dialplans in extensions.conf |
| 08:46.50 | jblack | Well, the config varies according to the way they like things, so they should give you an example configuration for iax.conf |
| 08:47.30 | jblack | You'll also likely have a register statement in your general section, to tell their server where your server is. |
| 08:47.47 | kalel008 | they only gave me an example to setup the trunk |
| 08:48.01 | dandre | does anyone have feedback concerning dect gateways from aastra(detewe) compared to polycom(kirk)? |
| 08:48.09 | jblack | It would be something like [iaxtrunk], or [bigcompany] or something. |
| 08:48.16 | jblack | Which provider? |
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| 08:49.13 | kalel008 | voipconnection |
| 08:49.23 | jblack | Ok. I'm not familiar with them specifically. |
| 08:50.07 | kalel008 | its in south africa |
| 08:50.36 | jblack | Here are three examples. http://pastebin.ca/1191283 |
| 08:51.10 | jblack | Notice that all three require slightly different things. It depends upon the provider. |
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| 08:53.26 | kalel008 | it doesnt open |
| 08:55.52 | kalel008 | this is what they sent me |
| 08:55.54 | kalel008 | 1. Go to your "Trunk" settings and find the "voipconnect" trunk. |
| 08:55.55 | kalel008 | 2. wherever you see "196.40.106.10", replace it with "iax.voipconnect.connection-telecom.com" |
| 08:55.55 | kalel008 | (look especially in the "PEER Details", and the "Register String") |
| 08:55.55 | kalel008 | 3. Click Submit Changes and then the "Apply Configuration Changes" bar at the top |
| 08:56.58 | kalel008 | thati put in iax.conf |
| 08:57.15 | kalel008 | and then the register and my username and passwrds |
| 08:57.18 | kalel008 | right |
| 08:57.34 | jblack | I've given you the help that I can. |
| 08:58.18 | kalel008 | just not sure if i must put my DID in iax.conf or extensions.conf |
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| 09:07.28 | angryuser | kalel008: you got hte link on the pastebin, look at that config, also read how to configure register in iax.conf at the wiki |
| 09:07.34 | angryuser | ~wiki |
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| 09:07.44 | angryuser | ~wikis |
| 09:07.45 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
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| 09:09.41 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 09:10.27 | Superbartt | Hi, I'm trying to build an Interactive Voice menu, now that isn't quite shocking, But I want it to wait for input untill an # is inputted. So I can make it fill in an complete number and than wait for the # |
| 09:10.59 | Superbartt | Or use an high timeout and make the # key end that timeout |
| 09:13.13 | jblack | sure. Look at Read() |
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| 09:15.36 | Superbartt | hmm, so in that case i need to pass it to a variable manually and than (in my case) forward it to the right extension using Goto? |
| 09:15.45 | Superbartt | think i can work on that :) thanks for the tip jblack |
| 09:16.05 | jblack | Well, you can Goto(${VARIABLE},1) |
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| 09:16.24 | kalel008 | thats exactly what i want to do |
| 09:16.26 | jblack | but if you just want to read the extension and goto it, then look at WaitExten() |
| 09:16.59 | Superbartt | naah, as i think of it it would be better to be able to have it in a own variabel, because i sometimes need to do different things with it |
| 09:17.15 | jblack | by the way. Infinite timeouts will be regretted. If a line gets stuck, you could spend a lot more on phone service than you intended. |
| 09:18.03 | Superbartt | naah it's for an internal service, and well not an infinite timeout, just something "big" |
| 09:18.26 | jblack | I once watched my ex-wife's cell phone loop through my IVR for 8 hours. |
| 09:18.36 | Superbartt | oops :D |
| 09:18.44 | jblack | it couldn't have been that long, but it seemed like it. |
| 09:19.15 | jblack | WaitExten is what you probably want. |
| 09:22.26 | jblack | wonders what to do with 6 microsd adapters |
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| 09:28.25 | newbie___ | hi all, my sip provider only allows me to make 1 call per sip account, if i were to suscribe 5 accounts. how do i make * automatically calls the second account when the first one is in use |
| 09:28.47 | jblack | You just set up multiple dials. |
| 09:29.09 | jblack | out,1,Dial(SIP1/#) |
| 09:29.20 | newbie___ | jblack: can i be like zap channels where i define them as g0 |
| 09:29.27 | jblack | out,2,Dial(SIP2/#) |
| 09:30.46 | newbie___ | i also tried using 1+101, but i think there is something worng with call-limit in sip.conf. am using asterisk 1.4.5 |
| 09:31.07 | newbie___ | asterisk just dont see that call |
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| 09:54.42 | yang | jblack: put them on ebay |
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| 10:02.22 | Superbartt | hmmfg, i seem to be having fax trouble :x A fax is passed over SIP to an linksys ata with the fax connected to it... but the only thing the fax spists out are errors |
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| 10:07.45 | kaldemar | newbie___: check out functions GROUP and GROUP_COUNT. |
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| 10:13.02 | newbie___ | kaldemar: thanks |
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| 10:47.15 | phpboy | how do I go about replacing the default asterisk voice recordings? |
| 10:47.21 | phpboy | do I just overwrite the files? |
| 10:48.05 | kaldemar | yes |
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| 11:30.58 | ric0h | got a question: TC400B specs seem inconsistent, is it 96 g729 channels or 120? |
| 11:35.01 | mocker | Does anyone know the option that makes MeetMe prompt for a person's name? |
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| 11:40.59 | kaldemar | mocker: there doesn't seem to be such an option. "core show application meetme" |
| 11:42.19 | kaldemar | ric0h: "The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations." - http://www.digium.com/en/products/voice/tc400b.php |
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| 11:44.04 | kaldemar | the datasheet states the same. where's the inconsistency? |
| 11:49.14 | ric0h | kaldemar: in the manual, next link from the datasheet.pdf |
| 11:50.48 | ric0h | kaldemar: https://www.digium.com/en/docs/TC400B/TC400B-user-manual.pdf pages 11,12,21 |
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| 11:54.33 | DarnoQ | anyone knows if there is an API link for bridging two numbers which can be used with betamax clones? |
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| 12:10.14 | DarnoQ | <PROTECTED> |
| 12:12.00 | kaldemar | patch it, recompile, replace the old one in astmoddir that is defined in asterisk.conf. |
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| 12:32.46 | kamanashisroy | does anyone have any problem ? |
| 12:33.19 | pawpro | Hey all. I'm having a problem with PHP_AGI 2.14. When I run get_data('filename',5000,20) for the first time it returns immedietly code=200,result=1,data=''. If i run it second time, it works just fine waiting for the user to dial dtmf. How do I approach that? |
| 12:34.46 | pawpro | So basically first time it doesn't even play the prompt and returns straight away. Any ideas? |
| 12:35.37 | [TK]D-Fender | kamanashisroy: Yes, but nothing a little penicillin won't fix |
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| 12:37.28 | kamanashisroy | [TK]D-Fender: do you know any way to get back interest to my job ?? :-P .. So I am waiting for a good/interesting problem ;) |
| 12:37.53 | [TK]D-Fender | kamanashisroy: And what job is that? |
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| 12:38.37 | kamanashisroy | [TK]D-Fender: 2 asterisk projects .. 2 j2me projects .. I am totally ruined with my interest .. pathetically messed with women :(( |
| 12:39.35 | [TK]D-Fender | kamanashisroy: want a quick thing or two to do? first, make functions be NON case-sensitive as well as the CLI help for them. |
| 12:40.23 | kamanashisroy | [TK]D-Fender: yes .. I hate that it is case sensitive .. |
| 12:40.41 | kamanashisroy | [TK]D-Fender: core show function diAl -> no application like that :-P |
| 12:40.51 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:41.00 | kamanashisroy | [TK]D-Fender: oops .. s/function/application/g |
| 12:42.06 | [TK]D-Fender | kamanashisroy: Applications are not case sensitive, functions are. |
| 12:42.08 | kamanashisroy | [TK]D-Fender: it is very simple task .. why did not people already solve this ?? |
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| 12:42.30 | kamanashisroy | [TK]D-Fender: I see .. I did not investigate the case so deeply .. |
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| 12:42.52 | [TK]D-Fender | kamanashisroy: Get to it, I'm sure this is probably amongst the easiest things to do. |
| 12:43.12 | kamanashisroy | so should I get the source from svn ?? |
| 12:43.24 | [TK]D-Fender | kamanashisroy: You should know better than I do. |
| 12:44.08 | kamanashisroy | [TK]D-Fender: thanks for talk ;) .. I am getting interested to my projects .. LOLz .. returning to my J2Me jobs first :) |
| 12:44.41 | kamanashisroy | [TK]D-Fender: thanks again .. |
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| 12:45.11 | [TK]D-Fender | kamanashisroy: np, and its not like I'm asking this for myself so much as confused newbs |
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| 13:19.12 | qp | hi all. anyone know why an internal call isnt playing a ringing tone to the person thats making the call? (IAX2 softphones) the Dial app has "tr" as the option so not sure why it isnt |
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| 13:29.07 | Superbartt | can anyone collaborate on app_fax[tx|rx] and asterisk bristuff? I seem to be getting alot of compile errors but can't find any solution online ( http://paste.barthezz.name/?show=317 errors ) |
| 13:31.00 | [TK]D-Fender | qp: You shouldn't normally use "r" (ever). You might want to make sure you have a proper indications.conf file. |
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| 13:38.24 | qp | [TK]D-Fender: thanks, I will investigate that conf file |
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| 13:39.45 | qp | any hints as to what to look at in there? I am UK btw |
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| 13:52.43 | RypPn | qp: under general make the country=uk , thats it |
| 13:53.22 | qp | RypPn: ok, thanks. it is set to that |
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| 13:54.12 | qp | having another issue in that dialing an international number seems to kill off sound, ie we cant hear them but they can hear us. however, its an international * box, calling their mobiles is fine |
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| 13:55.58 | zarnick | guys...I have a issue about pbx....general...couldn't find an irc channel about this...so I'm asking here....I think it's hardware issue, but I have one connection on an old pbx that is not making the telefone ring...any telefone |
| 13:56.07 | zarnick | guys...I have a issue about pbx....general...couldn't find an irc channel about this...so I'm asking here....I think it's hardware issue, but I have one connection on an old pbx that is not making the telefone ring...any telefone |
| 13:56.21 | zarnick | sorry about the double post....crapy keyboard |
| 13:56.38 | [TK]D-Fender | zarnick: Do you think you could give us LESS details? |
| 13:56.57 | zarnick | [TK]D-Fender, I don't even know how many details I can give |
| 13:57.02 | zarnick | it's an old Panasonic box |
| 13:57.24 | zarnick | we have two telefones...one was working fine and the other IS still working fine |
| 13:57.34 | [TK]D-Fender | zarnick: If you don't even know what you know then you know jsut about nothing and there is no chance we can help you |
| 13:57.46 | [TK]D-Fender | zarnick: and this is NOT #panasonic |
| 13:57.51 | zarnick | I tried to switch the telefones...and this showed to me that it's not the telefone, but the connection |
| 13:58.16 | zarnick | I'm trying to explain [TK]D-Fender |
| 13:58.37 | [TK]D-Fender | zarnick: No matter, this is not a place for Panasonic PBX support |
| 13:58.42 | zarnick | ok |
| 13:58.45 | zarnick | sorry to bother you |
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| 14:00.56 | zarnick | if anyone is willing to help about a generic pbx problem....please send me a message....I'm desperate |
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| 14:01.08 | [TK]D-Fender | zarnick: Call Panasonic |
| 14:01.20 | [TK]D-Fender | zarnick: You are in the wrong place. |
| 14:01.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 14:01.42 | zarnick | it was the first place for this problem that came in my mind...it's not a panasonic issue |
| 14:02.03 | [TK]D-Fender | zarnick: How is this an * issue? |
| 14:02.09 | linuxstb | suggests replacing the Panasonic pbx with an Asterisk box |
| 14:02.31 | zarnick | not * issue |
| 14:02.33 | zarnick | but pbx |
| 14:02.35 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 14:02.40 | zarnick | * is about building pbx |
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| 14:02.49 | zarnick | so I tough of coming in here |
| 14:02.57 | NoxIn- | zarnick: it can also be a cabling issue |
| 14:03.04 | zarnick | this is what I think |
| 14:03.28 | zarnick | NoxIn-, but I don't really now what each color means so I can re arrange the cable |
| 14:03.39 | [TK]D-Fender | zarnick: This is not the place for you. Go call a local telecom consultant. |
| 14:03.40 | zarnick | and I'm looking into google for this as well...but no luck so far |
| 14:03.42 | hi365 | Daviey: ping |
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| 14:21.44 | dr_gogeta86 | hi to all |
| 14:21.54 | dr_gogeta86 | can any one here help |
| 14:22.11 | dr_gogeta86 | i need to set acl on a dial function |
| 14:22.49 | dr_gogeta86 | this is the function |
| 14:22.53 | dr_gogeta86 | thiuexten => _0.,1,Dial(mISDN/1/${EXTEN:1}) |
| 14:23.01 | dr_gogeta86 | i wa |
| 14:23.18 | dr_gogeta86 | *want to deny calls from 9xxx |
| 14:23.45 | [TK]D-Fender | dr_gogeta86: "core show application gotoif" |
| 14:24.13 | dr_gogeta86 | how can i use it? |
| 14:24.22 | [TK]D-Fender | dr_go read the instructions. |
| 14:25.16 | dr_gogeta86 | I'm reading |
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| 14:26.22 | aliver | Is there any way to alter incoming caller ID? For example, I'd like to swap the position of the name and phone number. I'm not talking about calls on my internal * setup, I'm talking about calls coming from the PSTN through my SIP-based ITSP. |
| 14:27.27 | [TK]D-Fender | aliver: "core show application set" , "core show function CALLERID" |
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| 14:33.12 | Dr-Linux|home | anybody configured Wateen SIP user with Asterisk box? |
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| 14:35.08 | lowtek | I don't know anything about Wateen but I question their competence already: http://ipcommunications.tmcnet.com/pressrelease/WATEEN-TELECOM-SELECTS-GRANDSTREAM.aspx (via google when I went to see what Wateen was) |
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| 14:37.08 | Dr-Linux|home | lowlevel: Wateen is Wireless Internet and SIP provider |
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| 14:41.27 | LemensTS | im wanting to test a voip line, anyone know of a good place to call that will not kick me off? (i usually try 800-walmart for short test calls) |
| 14:41.51 | LemensTS | just wanting to hear call quality, and see if it drops my call |
| 14:41.59 | phpboy | how would I go about replacing the asterisk recordings with my own? |
| 14:42.22 | lowtek | ~thebook |
| 14:42.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 14:42.27 | lowtek | phpboy: ^^^ |
| 14:42.45 | LemensTS | phpboy: you can make a record program in your dialplan... |
| 14:43.13 | LemensTS | then ssh in and move them to the proper directories |
| 14:44.26 | Dr-Linux|home | ~centos52bug |
| 14:44.26 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 14:46.19 | *** join/#asterisk Levonk (n=lk@adsl-76-230-109-20.dsl.lsan03.sbcglobal.net) |
| 14:46.38 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-7e180811bfdd544b) |
| 14:46.38 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:47.12 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:51.15 | *** join/#asterisk af_ (n=getsmart@88-149-240-50.dynamic.ngi.it) |
| 14:56.32 | x86 | I've got an inbound hunt group on channels zap/51, zap/52, zap/53, and zap/54, going to a dedicated context (http://pastebin.ca/1191556), but the timeout is never executed and the channel is never hung up |
| 14:57.22 | x86 | I had a test call that I placed in, where it played the message "tyfc.ulaw", and did not timeout even after 30 minutes |
| 14:57.46 | x86 | any idea how to make timeout work? |
| 14:59.29 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.32 | x86 | the relevant part of my zapata.conf is here: http://pastebin.ca/1191561 |
| 15:00.35 | [TK]D-Fender | x86: set your timeouts properly. |
| 15:00.45 | x86 | [TK]D-Fender: url me? |
| 15:00.52 | [TK]D-Fender | x86: Then when you've done that pastebin the new dialplan and CLI output |
| 15:01.01 | [TK]D-Fender | x86: "core show function TIMEOUT" |
| 15:02.54 | x86 | [TK]D-Fender: that says that both digit and response are set by default to 5 and 10, respectively, and my WaitExten's timeout is set to 5, so that shouldn't be the issue at all |
| 15:03.52 | [TK]D-Fender | x86: Do it the full way and include the CLI output |
| 15:04.26 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-cfc9345a145d061b) |
| 15:04.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:05.38 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 15:07.34 | x86 | seems to work now with the TIMEOUTs set manually... odd |
| 15:11.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:12.37 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) |
| 15:13.37 | thing1 | hi i'm trying to get a pri registered to my asterisk , there is a cisco gatekeeper at the other end and when i try and place a call into my * i get a failed to authenticate and the nember i'm trying to call from |
| 15:15.12 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:23.39 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:24.30 | Deeewayne | thing1: pri doesn't register |
| 15:25.16 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 15:26.31 | [TK]D-Fender | Deeewayne: And he gave enough that you should know the Cisco is converting to VoIP |
| 15:26.43 | [TK]D-Fender | Deeewayne: And he simply hasn't set up auth for it properly |
| 15:26.44 | Qwell | "get a pri register to my asterisk".. |
| 15:27.00 | [TK]D-Fender | Qwell: Yeah, over consolidated. |
| 15:27.01 | Qwell | I had to look at that several times to figure it out |
| 15:38.13 | *** join/#asterisk ManxPower (n=manxpowe@4.sub-75-249-90.myvzw.com) |
| 15:42.32 | x86 | I figured it out the first time seeing it |
| 15:42.41 | x86 | my context clues are rad :p |
| 15:45.58 | ajohnson | ~ce |
| 15:45.59 | jbot | [ce] still on it |
| 15:46.05 | ajohnson | okie dokie |
| 15:47.02 | ajohnson | Anyone know if it is possible to execute an arbitrary dialplan command on a channel through the manager interface? |
| 15:47.19 | ajohnson | Specifically, I want to issue ChanSpy commands through the manager interface on a channel |
| 15:52.06 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 15:53.53 | [TK]D-Fender | ajohnson: Who'll be doing the spying? You have to actually be listening, so it can't be "arbitrary" |
| 15:54.08 | [TK]D-Fender | ajohnson: The audio isn't going into thin air you know... |
| 15:54.38 | ajohnson | hehe |
| 15:55.46 | ajohnson | Urgh |
| 15:56.35 | ajohnson | I want to be able to change which channel is being spied upon through a web page |
| 15:56.53 | ajohnson | Without having to originate a new channel each time |
| 15:57.14 | ajohnson | [TK]D-Fender: Sorry, perhaps I should pose the actual problem, as opposed to what I think the general solution would be... |
| 15:57.26 | [TK]D-Fender | ajohnson: You'd use a manager Redirect then on the ALREADY spying channel. |
| 15:57.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:57.35 | ajohnson | muahahha |
| 15:57.40 | ajohnson | so obvious |
| 15:58.10 | *** join/#asterisk MrNaz (n=naz@ppp118-208-176-116.lns10.mel4.internode.on.net) |
| 15:58.12 | ajohnson | [TK]D-Fender: Thanks for getting me over that hurdle... |
| 16:01.36 | nn | :( |
| 16:02.04 | *** join/#asterisk Knightfal (n=jjj@66.178.134.235) |
| 16:02.45 | Knightfal | TK: What was the Sip Phones you recommended to me on friday? |
| 16:03.20 | Knightfal | Polycom 330? |
| 16:08.29 | *** join/#asterisk newbie___ (n=newbie__@156.90.48.60.kmr02-home.tm.net.my) |
| 16:08.57 | nn | anyone know about unlocking linksys rt31p2-na's? got one off ebay (the -na is supposed to be always unlocked) and it's locked to voip.com (who is aweful provider) |
| 16:09.52 | thing1 | hi why would the phone companies number not in service come on when i dial into my * via a sip trunk |
| 16:11.01 | newbie___ | hi all, i have a sip provider that only allows 1 call at a time. i plan to suscribe 4 more sip account. how do i make all 5 sip accounts to behave like a normal zap trunk ? ie exten => s,1,Dial(SIP/g0). i tired to use call-limit=1 in asterisk 1.4 but asterisk seems to ignored it |
| 16:12.24 | ManxPower | you can't use groups with SIP. |
| 16:12.47 | ManxPower | as for call-limit, you should re-read the upgrade.txt file to see what changes have been for call-limit |
| 16:13.10 | kaldemar | newbie___: did you try GROUP and GROUP_COUNT? |
| 16:13.19 | ManxPower | thing1: That happens if the call is rejected for some reason |
| 16:13.27 | newbie___ | ManxPower: any way around it ? when i attempt to make a secnd call, * always look for SIP/01 |
| 16:13.56 | ManxPower | newbie___: you can check the value of DIALSTATUS after each dial attempt, then determine if the call has to be redialed. |
| 16:14.36 | newbie___ | ManxPower: DIALSTATUS as in ChanISAvailble ? |
| 16:15.13 | newbie___ | kaldemar: couldnt figure out how to use GROUP and GROUP_COUNT |
| 16:15.22 | ManxPower | newbie___: DIALSTATUS as in the variable Dial sets. Have you not read any docs? "core show application dial" /path/to/src/asterisk/doc/channelvariables.txt |
| 16:16.01 | ManxPower | newbie___: this is fairly advanced call routing stuff. Maybe you should go learn more (maybe read ATFOT2) first. |
| 16:16.57 | oilinki | question: now SIP seems to be the only protocoll supported by voip-to-pstn operators. is there other alternatives? |
| 16:17.20 | ManxPower | oilinki: IAX2, MGCP, H323. |
| 16:17.30 | newbie___ | shit, everythins is so alien |
| 16:17.31 | ManxPower | But SIP and MAYBE IAX2 are the only ones you should care about. |
| 16:17.40 | oilinki | ManxPower: but does the operators support those as well? |
| 16:17.44 | kaldemar | newbie___: a little hint, you can add channels to a group with GROUP and count channels in the group with GROUP_COUNT. then use if's and goto's to make the channel selection logic in the dialplan. not that hard when you know how those applications and functions work. |
| 16:17.47 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-123-094.pools.arcor-ip.net) |
| 16:18.35 | Blackvel | hi. anyone using patton 4634/4638 or 4554 isdn bri gateway with firmware R5.2? trying to set it up with asterisk |
| 16:18.40 | linuxstb | oilinki: There are plenty of providers who support IAX2 at least. |
| 16:18.44 | oilinki | ManxPower: ok. the information I have of SIP is that 'SIP sucks' :) |
| 16:18.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:19.08 | newbie___ | kaldemar: thanks for the tip |
| 16:19.14 | oilinki | and that was a quote from a man who knows what he is talking about |
| 16:19.20 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 16:19.31 | ManxPower | oilinki: the information you have is WRONG. |
| 16:20.08 | ManxPower | oilinki: What SPECIFIC issues are supposed to "suck" in SIP? |
| 16:20.23 | oilinki | ManxPower: nat problems mainly |
| 16:20.48 | ManxPower | Yes, SIP is not NAT friendly -- unless of course you are using SIP support from like 5 years ago. SIP+NAT has worked fine for YEARS for most people. |
| 16:20.57 | oilinki | then also problems with different operators, servers and end user terminals |
| 16:21.23 | ManxPower | oilinki: Odd then that virtually every provider out there supports SIP, don't you think? |
| 16:21.42 | ManxPower | What is this supposed guru suggesting you use right now in the real world for VoIP if not SIP? |
| 16:21.45 | oilinki | ManxPower: it's the trend today. |
| 16:22.15 | [TK]D-Fender | oilinki: Got some more heresay for us? I haven't quite hit my quota... |
| 16:22.16 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:22.38 | [TK]D-Fender | oilinki: Do you have an actual issue to share with us? |
| 16:22.57 | oilinki | ManxPower: nope, as far as I know there is not an alternative at the moment. but then again. I'd like to know if there is. |
| 16:23.04 | kaldemar | yes, messing things up is pretty easy. avoid using poorly chosen operators, servers and user terminals. |
| 16:23.07 | ManxPower | oilinki: Are you saying that SIP is terrible but everyone uses it anyway? Sounds like MS Windows to me. |
| 16:23.38 | jaytee | I'm thinking * compiled to run on Netware 3.1 using IPX/SPX will kill the competition. :-) |
| 16:23.42 | coppice | SIP was designed to make H.323 look good |
| 16:23.44 | ManxPower | oilinki: ALL current standards based VOIP protocols have NAT issues (they ALL use RTP for audio and that is the real NAT issue), except IAX2. |
| 16:24.39 | oilinki | ManxPower: so iax2 would be an better solution. ok. |
| 16:24.40 | coppice | IAX2 has pretty much the same NAT issues as anything else |
| 16:25.09 | [TK]D-Fender | oilinki: Again, do you have an actual problem or are you just trolling? |
| 16:25.11 | ManxPower | oilinki: Oh, every server out there, even HTTP web servers will have problems with NAT. |
| 16:25.24 | oilinki | [TK]D-Fender: I'm not trolling. |
| 16:25.40 | [TK]D-Fender | ManxPower: Much harder to conceive on HTTP... at least thats all TCP and trackable ::) |
| 16:26.03 | [TK]D-Fender | oilinki: So whats the problem? We've all been using *, SIP, and NAT for years just fine. |
| 16:26.03 | ManxPower | [TK]D-Fender: same issue, if you want to run a web server behind NAT you need to forward at least one port. |
| 16:26.30 | [TK]D-Fender | ManxPower: true, but mapping timeouts isn't an issue |
| 16:26.31 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:26.42 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:27.12 | oilinki | [TK]D-Fender: I have also been playing with sip for year. alltought not an guru. I also have been trying to resolve some of the problems with the terminals and the servers. |
| 16:27.28 | oilinki | s/year/years/ |
| 16:27.33 | [TK]D-Fender | oilinki: Not quite vague enough. Could you try a little harder? |
| 16:29.35 | oilinki | [TK]D-Fender: there is different implementations of servers. one will support one function the other not. then we'll go to look at the terminals which are all different. and go to for example home routers, which will do funny things to the sip messages by trying to be to smart. |
| 16:30.12 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:30.15 | [TK]D-Fender | oilinki: Wow, that cleared up so much for me. |
| 16:30.24 | [TK]D-Fender | goes off to do something productive. |
| 16:30.49 | oilinki | the thing is that this is not clear enought. I have my own asterisk server runninning now. seems to be quite ok at the moment, but I know that to get that one to the production I have to do a lots of testing. |
| 16:30.57 | ManxPower | oilinki: odd since hundreds of people run SIP with NAT and Asterisk. If the Asterisk server is on a public IP (like all servers should be) there are no nat issues except for reinvites |
| 16:31.19 | [TK]D-Fender | And anyone with a brain would simply refuse them anyways |
| 16:32.02 | ManxPower | [TK]D-Fender: oilinki has not provided us with even one specific issue or features so I'll write him off too. |
| 16:32.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:32.47 | ManxPower | now if he said something like "The SIP refer method has problems with Linksys NAT, because of a bug in the Linksys routers) |
| 16:32.56 | [TK]D-Fender | ManxPower: My stuff doesn't work so well with things, and I have been working on it for a long time and talk to people who know what they are talking about! |
| 16:33.07 | ManxPower | Actually SIP used to make older linksys routers reboot. Was festive to discover that. |
| 16:33.28 | ManxPower | that is a router issue, not a SIP issue. |
| 16:34.09 | oilinki | ManxPower: well some of the linksys adsl-routers do not let multiple registrations to happen. and this is alg issue. |
| 16:34.24 | ManxPower | oilinki: cite your source. |
| 16:34.26 | oilinki | with openwrt software everyting is fine naturaly. |
| 16:34.49 | oilinki | ManxPower: momen, I'll go to check the router I have upstairs |
| 16:35.07 | ManxPower | oilinki: no, cite your source of at least one other person in the world experiencing your problem. |
| 16:35.35 | ManxPower | I can think of at least 2 ways to work around that issue, if that is infact the issue. |
| 16:35.53 | ManxPower | oilinki: Just because YOU have a problem does not mean everyone else does. |
| 16:36.11 | oilinki | ManxPower: linksys adsl router wag54g v3. |
| 16:36.20 | ManxPower | <PROTECTED> |
| 16:36.44 | fogo | ManxPower: well, if you're including mental problems, I am too ;) |
| 16:36.47 | oilinki | ManxPower: well, then you have to do google for it. I did my tests. |
| 16:36.59 | ManxPower | Even if the Linksys has brain dead NAT support (and it would not suprize me), simply changing the source port in one of the phones will solve the issue |
| 16:37.24 | ManxPower | oilinki: so your entire claim is based on one box at your house? |
| 16:37.50 | ManxPower | Anyway, BEST of luck in your projects. |
| 16:38.08 | ManxPower | I'm here to help people, not defend SIP against slander. |
| 16:38.18 | oilinki | ManxPower: well, yes. you might get pass the issue with changing the source port. but that will call fail if you wish to get your service for public? |
| 16:39.27 | [TK]D-Fender | oilinki: The public does not put multiple SIP phones behind NAT |
| 16:39.39 | [TK]D-Fender | oilinki: Companies do that. |
| 16:40.01 | ManxPower | If you REALLY understand SIP and RTP and NAT, the solution of a SIP NAT issue is pretty obvious. |
| 16:40.04 | [TK]D-Fender | ManxPower: Yup, another one of those days |
| 16:40.07 | thing1 | can i connect asterisk to a call manager via skinny |
| 16:40.09 | oilinki | these are real issues for real people. to get an approval for some service, these things need to be tested.. and recommended if possible |
| 16:40.35 | [TK]D-Fender | thing1: chan_skinny can only act like a SERVER, not like a PHONE |
| 16:40.37 | ManxPower | oilinki: sad then that all these VOIP companies, hundreds of them in the world, have a service that does not work. |
| 16:40.49 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 16:41.01 | ManxPower | But as I said, best of luck. I cannot "help" your further. |
| 16:41.12 | oilinki | ManxPower: no they have services which work pretty well. |
| 16:41.16 | thing1 | ok, is there any other way to connect to call manager that sip trunk? |
| 16:41.24 | *** join/#asterisk seanmh (i=seanmh@216.31.101.85) |
| 16:41.28 | [TK]D-Fender | oilinki: Yup I guess Broadvoice, Vonage and all the other massive ITSPs jsut don't "work" |
| 16:41.33 | ManxPower | thing1: have you tried SIP? |
| 16:41.44 | [TK]D-Fender | thing1: T1, E1, etc |
| 16:41.46 | oilinki | but if their customer is using routers which missbehave, there is a problem for provider as well |
| 16:41.53 | thing1 | the guy on the other end won't let me |
| 16:42.07 | thing1 | i have to use h323 or skinny |
| 16:42.07 | ManxPower | thing1: then it really sucks to be you. |
| 16:42.08 | [TK]D-Fender | thing1: Then you're screwed. |
| 16:42.22 | thing1 | can't i somehow do it with h323 |
| 16:42.23 | [TK]D-Fender | thing1: You could try H.323, but it isn't terribly stable |
| 16:42.36 | thing1 | well then i gues i have to use an ata |
| 16:42.48 | ManxPower | thing1: sure you can, but almost nobody will be able to help you. Only desperate people with no other options use H323 with Asterisk |
| 16:43.12 | ManxPower | thing1: Hope you are using a Cisco ATA, since that's the only one that I know of that support Skinny |
| 16:43.22 | oilinki | naturally 'get a new router' is an answer.. it's just might not be the best option for a customer who wish to use voip. |
| 16:43.40 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 16:44.10 | ManxPower | [TK]D-Fender: looks like the new IT manager at my biggest client screwed up activating the company's backup NOC. |
| 16:44.35 | ManxPower | Not that it would have done much good as the backup NOC is located in the path that most hurricanes take when they hit NOLA. |
| 16:46.30 | thing1 | ok, back to sip trunk, when i dial the number forwarded into my *, i get the telcompany saying the number is not in service after one ring |
| 16:46.34 | thing1 | what might be the problem |
| 16:46.44 | ManxPower | thing1: something is REJECTING the call. |
| 16:46.48 | seanbright | ~siptrunk |
| 16:46.49 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 16:46.59 | ManxPower | good grief! |
| 16:47.05 | ManxPower | ~trunk |
| 16:47.05 | jbot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 16:47.14 | seanbright | ah yes |
| 16:47.15 | seanbright | there we go |
| 16:47.28 | ManxPower | thing1: either your ITSP is rejecting the call or your server is rejecting the call. |
| 16:47.32 | thing1 | Really destroying SIP dialog '6b71a0585d80e27c2af29cf975bae218@x.x.x.x' Method: OPTIONS |
| 16:47.44 | [TK]D-Fender | thing1: And you have no SIP debug info here for us to help you with. USELESS |
| 16:47.51 | ManxPower | thing1: standard message, you didn't, of course, post the WHOLE message, did you? |
| 16:47.56 | [TK]D-Fender | thing1: look at the debug for the entire ATTEMPT |
| 16:49.07 | ManxPower | "sip debug" + try a call |
| 16:49.22 | ManxPower | or sip debug peer X or sip debug ip X |
| 16:50.01 | thing1 | http://pastebin.com/m700ac74c |
| 16:50.03 | [TK]D-Fender | or just "sip debug" |
| 16:50.21 | [TK]D-Fender | thing1: that is not a CALL |
| 16:50.47 | ManxPower | nope, that is not a call. that is a qualify= thing. |
| 16:50.50 | [TK]D-Fender | thing1: that is just a useless "qualify" packet answer |
| 16:50.53 | ManxPower | you need INVITES for a call. |
| 16:51.04 | Blackvel | what voip provider services do you guys use in the US? is there a good fwd replacement? |
| 16:51.11 | ManxPower | ~itsp-us |
| 16:51.18 | ManxPower | ~itso |
| 16:51.24 | ManxPower | ~itsp |
| 16:51.24 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:51.25 | [TK]D-Fender | ~itsplist-us |
| 16:51.26 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:51.45 | *** join/#asterisk esdrasbeleza (n=esdras@201.49.22.87) |
| 16:51.49 | thing1 | so i show sip set debug and then place a call into 8 |
| 16:51.54 | thing1 | so i show sip set debug and then place a call into * |
| 16:52.05 | ManxPower | thing1: looks like your provider is BEHIND NAT! |
| 16:52.15 | x86 | hahahahahaha |
| 16:52.17 | thing1 | nope |
| 16:52.23 | Blackvel | wasn't it the fact that most of them are pay per use and used for real pstn termination? |
| 16:52.34 | ManxPower | then where are From: "asterisk" <sip:asterisk@10.4.30.15>;tag=as699d5f4d To: <sip:10.2.0.2>;tag=A54FEAE4-37E |
| 16:52.47 | ManxPower | 10.4.x.x and 10.2.x.x |
| 16:53.01 | [TK]D-Fender | ManxPower: All private.. |
| 16:53.04 | thing1 | cisco switch routing vlans |
| 16:53.06 | ManxPower | [TK]D-Fender: exactly |
| 16:53.08 | [TK]D-Fender | ManxPower: perfectly normal |
| 16:53.11 | thing1 | it's a wide area network |
| 16:53.17 | thing1 | private |
| 16:53.27 | legis | how do I dial from the asterisk box with a softphone? |
| 16:53.28 | ManxPower | thing1: maybe you should disable all the other sip stuff until you can get what you want working. |
| 16:53.40 | ManxPower | legis: you don't You dial the softphone. |
| 16:53.44 | [TK]D-Fender | legis: go into your softphone and DIAL SOMETHING |
| 16:54.04 | thing1 | ManxPower i have nothing enabled |
| 16:54.11 | thing1 | all thats in my sip conf is this |
| 16:54.13 | [TK]D-Fender | thing1: stop wasteing time and pastebin a damn call. |
| 16:54.15 | ManxPower | thing1: you have something enabled for those IP addresses. |
| 16:54.25 | ManxPower | [TK]D-Fender: I think his registration is a problem. |
| 16:54.44 | ManxPower | thing1: what is the output FOR THAT PROVIDER in "sip show registry" |
| 16:54.47 | [TK]D-Fender | ManxPower: I suspect nothing except a failure to listen and think |
| 16:54.51 | thing1 | my first thing is to get incoming call answered |
| 16:55.01 | [TK]D-Fender | ManxPower: Registration is not yet relevent |
| 16:55.03 | ManxPower | [TK]D-Fender: that was obvious from the start. |
| 16:55.13 | ManxPower | [TK]D-Fender: if he's trying to receive a call from an ITSP it it. |
| 16:55.15 | ManxPower | is. |
| 16:55.18 | [TK]D-Fender | ManxPower: So far that's all I'll bank on. |
| 16:56.12 | ManxPower | My bet is a bad register => line. |
| 16:56.12 | thing1 | there is nothing in sip show registry :D |
| 16:56.12 | [TK]D-Fender | thing1: stop wasteing time and pastebin a damn call. <--------- |
| 16:56.12 | ManxPower | thing1: then you are not registering to your provider. |
| 16:56.12 | esdrasbeleza | hi. I need to program a configuration interface for Asterisk. can I config Asterisk only via sockets or databases, or there is some configuration I only can do editing files? |
| 16:56.12 | ManxPower | If that is the case you will never ever receive any output from a sip debug for a call. |
| 16:56.17 | thing1 | sip show peers shows the host and OK 37 ms |
| 16:56.17 | [TK]D-Fender | ManxPower: lay of the registration its a LOCAL DEVICE. |
| 16:56.30 | ManxPower | [TK]D-Fender: if you say so,. |
| 16:56.42 | [TK]D-Fender | ManxPower: the IP sure seems to. |
| 16:56.51 | ManxPower | esdrasbeleza: there are at least 4 different GUIS that edit Asterisk config files. find out how they do it. |
| 16:57.26 | esdrasbeleza | ManxPower, thanks. can you tell me their names? |
| 16:57.42 | ManxPower | thing1: AFTER YOU FINISH PASTEBINING A FAILED CALL, diagram your setup like PSTN -> ITSP ->(SIP)/Internet-> NAT ROUTER -> Asterisk or whatver is correct. |
| 16:57.46 | ManxPower | esdrasbeleza: look in the /topic |
| 16:58.14 | ManxPower | [TK]D-Fender: I don't believe that pastebin had anything to do with his problem |
| 16:58.18 | [TK]D-Fender | esdrasbeleza: freepbx, thirdlane, switchvox, scopserv, etc |
| 16:58.22 | oilinki | esdrasbeleza: you could check realtime config for the start |
| 16:58.59 | [TK]D-Fender | ManxPower: the "qualify" one? It is at least referencing the device he's having problems with. Does if have ANYTHING to do with the actualy problem? hell no... |
| 17:00.12 | oilinki | esdrasbeleza: for mysql user accounts and dialplan http://www.voip-info.org/wiki-Asterisk+RealTime |
| 17:00.40 | [TK]D-Fender | esdrasbeleza: SSH uses a socket, have fun. |
| 17:00.44 | ManxPower | [TK]D-Fender: you are assuming the call gets to the server. I don't think it does. |
| 17:01.04 | esdrasbeleza | oilinki, hm, good. |
| 17:01.06 | [TK]D-Fender | ManxPower: qualify makes it back on a local LAN. So far nothing to say period. |
| 17:01.42 | ManxPower | I think I will go enjoy my day off. None of my customers are back from evacuation yet. |
| 17:02.04 | ManxPower | [TK]D-Fender: it would indicate he lied when he said all the other sip devices are disabled. |
| 17:02.08 | oilinki | esdrasbeleza: I got the users, dialplans (extensions) and cdr's working. have not got the registrars and other stuff in the sip.conf working. |
| 17:02.16 | ManxPower | (the one and only pastebin we get from this guy) |
| 17:02.20 | ManxPower | ) |
| 17:02.27 | [TK]D-Fender | ManxPower: I'd simply stop advising until he shows us something of value. |
| 17:02.35 | thing1 | thanks guys |
| 17:02.55 | ManxPower | [TK]D-Fender: I won't fight over him. If you want him you can have him. |
| 17:02.57 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 17:03.13 | [TK]D-Fender | ManxPower: I'm near the end myself. |
| 17:03.22 | esdrasbeleza | [TK]D-Fender, I don't know so much about Asterisk, my boss told me that there were some configuration that can be done using sockets, I have no idea about what can it do or not, lol |
| 17:03.33 | ManxPower | esdrasbeleza: we are not here to tutor you in Asterisk. |
| 17:03.46 | [TK]D-Fender | esdrasbeleza: care to elaborate on "sockets"? |
| 17:03.51 | ManxPower | If you want to learn Asterisk start by reading The Book |
| 17:03.53 | ManxPower | ~book |
| 17:03.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:03.53 | thing1 | ok, this is what i've done, my sip conf have one entry that is the sip trunk, i have nothing else connected to the box, all i am trying to do is get it to answer the call and forward it to a degital receptionist |
| 17:04.03 | thing1 | i don't what it to do anything else |
| 17:04.14 | thing1 | now how should i show u a failed call? |
| 17:04.19 | [TK]D-Fender | thing where is the call debug I asked you for REPEATEDLY? |
| 17:04.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:04.29 | esdrasbeleza | ManxPower, and I don't want that. :) I just want a good starting point about these topics. |
| 17:04.33 | ManxPower | thing1: turn on sip debug, try calling your number from a cell phone. |
| 17:04.39 | ManxPower | It's not rocket science |
| 17:04.47 | thing1 | that is exactly what i did |
| 17:04.51 | ManxPower | esdrasbeleza: there is NO good starting point for what you want to do. |
| 17:04.56 | thing1 | all i get is what i posted |
| 17:05.30 | [TK]D-Fender | thing1: If you get NO SIP traffic for a call then your cisco isn't sending the call to *. |
| 17:05.42 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 17:05.45 | oilinki | esdrasbeleza: don't worry about some of the people in the channel. they can be good what they are doing, but not the helpfull hand you need sometimes :) |
| 17:06.27 | thing1 | before i got an authentication error, then i added insecure=port,invite into sip conf and now i get this |
| 17:06.28 | ajohnson | esdrasbeleza: What you're trying to do can be a little complex and sometimes difficult to debug if you know what you're doing |
| 17:06.38 | ajohnson | If you don't know what you're doing, it is damn near impossible |
| 17:07.07 | ajohnson | So I suggest getting a decent grasp of what's going on behind the scenes, TFOT is excellent for that |
| 17:07.24 | esdrasbeleza | oilinki, yeah, I realised it :) |
| 17:07.42 | [TK]D-Fender | esdrasbeleza: And I asked you for you meant by "sockets". |
| 17:07.49 | ManxPower | AT&T keeps calling me to tell me T-1s are down. Of course they are down, the site was hit by a hurricane. |
| 17:08.06 | ajohnson | TFOT even has a little information on how to do Asterisk Realtime, which may be a good solution for what you're looking for |
| 17:08.33 | esdrasbeleza | I have an Asterisk running at a machine. I need to do a graphical interface, running at a web server, that could handle Asterisk configuration. |
| 17:08.53 | oilinki | ajohnson: what is TFOT btw? |
| 17:09.02 | [TK]D-Fender | oilinki: BOOK <---- |
| 17:09.11 | [TK]D-Fender | jbot>Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:09.22 | [TK]D-Fender | "The Future of Telephony" |
| 17:09.24 | ajohnson | esdrasbeleza: http://www.voip-info.org/wiki-Asterisk+GUI |
| 17:09.26 | bkw_ | oilinki: the default answer for anyone who is new. |
| 17:09.27 | oilinki | esdrasbeleza: asteriskNOW is good for gui. |
| 17:10.07 | ajohnson | I suggest either AsteriskNOW or elastix for free quis |
| 17:10.21 | ajohnson | esdrasbeleza: or SwitchVox for a more commercial solution |
| 17:10.24 | oilinki | ok. thanks. I liked this version. http://asterisk.name/ |
| 17:10.27 | [TK]D-Fender | elastix is not a GUI |
| 17:10.31 | kaldemar | suggest the GUI itself for a GUI, not the whole AsteriskNOW. |
| 17:10.42 | ajohnson | doh |
| 17:11.19 | esdrasbeleza | oilinki, I've read something about it yesterday, unfortunately it appears that someone at my job wants to reinvent the wheel to make a more specific commercial package with some features to our clients |
| 17:11.38 | ajohnson | [TK]D-Fender: Elastix has a GUI... sorry for the distinction |
| 17:12.11 | ManxPower | esdrasbeleza: maybe your employer should learn a little about the GPL before trying to release a commercial version of Astersk. |
| 17:12.58 | oilinki | esdrasbeleza: there might be sense in that. I tried the gui, but I could not figure out how to do some things. therefore asterisk with realtime and own gui's was better solution for me. might be the same case if your company thinks the same. |
| 17:13.26 | [TK]D-Fender | esdrasbeleza: And you restrict your implementation to remotely configuring * 100% via a "socket" or "database" Sorry, but * is not 100% configurable via DB, and you have not clarified your definition of "socket" |
| 17:13.58 | ajohnson | Does "socket" include SSH? |
| 17:14.10 | [TK]D-Fender | ajohnson: I already alluded to that option :) |
| 17:14.13 | ManxPower | ajohnson: maybe a UDP socket? |
| 17:14.15 | [TK]D-Fender | ajohnson: Take a number... |
| 17:14.22 | ajohnson | hehe |
| 17:14.22 | [TK]D-Fender | ManxPower: ooohh! |
| 17:14.23 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:14.38 | ManxPower | I know! it's a GRE socket? |
| 17:14.44 | ManxPower | (goes GRE even use sockets?) |
| 17:14.49 | [TK]D-Fender | ManxPower: RJ11? |
| 17:14.55 | [TK]D-Fender | ManxPower: thats a socket... |
| 17:15.14 | kaldemar | ssh asteriskbox "echo -e ... > extensions.conf" <-- teh way via socket |
| 17:15.15 | esdrasbeleza | [TK]D-Fender, My boss told me that there was some way to connect directly to Asterisk - using sockets - and configure it using some commands. that's what I mean for sockets |
| 17:15.17 | [TK]D-Fender | 110v? |
| 17:15.21 | ajohnson | [TK]D-Fender: Through the power plug |
| 17:15.22 | ajohnson | doh |
| 17:15.29 | *** join/#asterisk logicwrath (n=no@68.41.24.98) |
| 17:15.31 | [TK]D-Fender | esdrasbeleza: WHAT KIND OF SOCKET!? |
| 17:15.43 | [TK]D-Fender | ajohnson: pwned again. |
| 17:15.43 | ajohnson | [TK]D-Fender: I really think he means SSH |
| 17:15.52 | [TK]D-Fender | ajohnson: Stop thinking, he clearly isn't |
| 17:16.09 | oilinki | esdrasbeleza: most likely connected with mysql? |
| 17:16.21 | ajohnson | esdrasbeleza: That's like saying "My boss said you can build a car with a tool" |
| 17:16.27 | ManxPower | esdrasbeleza: You understand that Asterisk isn't really a PBX, right? Asterisk is a TOOLKIT that lets you BUILD a PBX. |
| 17:17.06 | [TK]D-Fender | Look at all you people wasting time with conjecture for a guy too dumb to say WHAT KIND OF FUCKING SOCKET he means! |
| 17:17.10 | [TK]D-Fender | I want to do STUFF! |
| 17:17.16 | [TK]D-Fender | With THINGS! |
| 17:17.19 | esdrasbeleza | [TK]D-Fender. Like this: |
| 17:17.34 | kaldemar | play with cluebat, how's that for a thing? |
| 17:17.39 | esdrasbeleza | http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+C+Sharp |
| 17:17.43 | [TK]D-Fender | <PROTECTED> |
| 17:17.50 | ManxPower | esdrasbeleza: the problem is that you don't know enough of Asterisk to even ask good questions |
| 17:18.04 | ManxPower | esdrasbeleza: Manager is NOT for configuration. |
| 17:18.07 | ajohnson | esdrasbeleza: That is a good way to manipulate asterisk |
| 17:18.09 | ajohnson | Not configure |
| 17:18.18 | ManxPower | See what I mean? |
| 17:18.37 | esdrasbeleza | ManxPower, I have to agree with you. |
| 17:19.06 | lowtek | esdrasbeleza: is this a new asterisk install or one that you're having to manage? |
| 17:19.22 | [TK]D-Fender | esdrasbeleza: Go read the book. No more shortcut. You are shooting bullet randomly in the air with no clue about how any of * works and we are not here to teach you everything from the ground up. |
| 17:19.41 | esdrasbeleza | ManxPower, but they told me to do that and didn't even ask me if I know something about Asterisk. Do you know how employers are... |
| 17:19.51 | ajohnson | esdrasbeleza: To put it plainly... What you are trying to do is impossible with knowledge of 1) Linux, 2)SSH, 3) Asterisk basics, 4) Probably a few other things |
| 17:19.58 | ManxPower | esdrasbeleza: How exactly is that OUR problem? |
| 17:19.58 | [TK]D-Fender | esdrasbeleza: Go read the book and LEARN |
| 17:20.12 | ajohnson | esdrasbeleza: If my employer gave me such a task, I would give my 2 weeks notice because they are too fucking stupid to run a business |
| 17:20.17 | ManxPower | ajohnson: NAT, UDP, SIP, RTP too. |
| 17:20.24 | ajohnson | right... |
| 17:20.49 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 17:21.03 | ajohnson | esdrasbeleza: I feel for you, but you are looking at at LEAST a 2-6 month lead time just to learn what you need to know to plan out how to accomplish what you want to do |
| 17:21.08 | oilinki | so, give the guy a break. he is trying to understand more and all you do is try to put him down. |
| 17:21.34 | ajohnson | Thats just the time required to learn how much you need to learn to accomplish your goals... |
| 17:21.57 | ajohnson | oilinki: No, that's not true at all |
| 17:21.57 | [TK]D-Fender | oilinki: He has restrictions he can't explain, and understands nothing about *. Even the answers are beyond his understanding. This is a tragic waste of time at this phase. |
| 17:22.13 | esdrasbeleza | you're right. sorry. |
| 17:22.32 | SamuraiDio | when i configure a sip user to accept every calls without auth (by bindind an ip adress to the sip user), how does asterisk checks if the ip address is valid? does is send a feedback authenticaton to that ip? |
| 17:22.35 | oilinki | [TK]D-Fender: surely it seems to be diffucult task at the moment, but not after a week. |
| 17:22.51 | [TK]D-Fender | esdrasbeleza: In the book you'll see what is configurable by DB, what isn't, how AMI works, etc. |
| 17:23.01 | oilinki | esdrasbeleza: go ahead. keep on asking questions. we all will undestand more if you do. |
| 17:23.03 | [TK]D-Fender | oilinki: Yes, well he's here NOW <- |
| 17:24.19 | esdrasbeleza | [TK]D-Fender, good, that's what I want to know: the ways to configure Asterisk, what can I do by DB and what not. |
| 17:24.31 | oilinki | esdrasbeleza: to configure asterisk is pretty simple when you get use to it. some of the information is a bit blurr in the documentation but asking questions and doing a loads of googling will get you forward. |
| 17:24.34 | [TK]D-Fender | esdrasbeleza: then go read the BOOK |
| 17:25.07 | lowtek | The book does go a long way to providing smooth pavement on a relatively rocky road and it's an easy read. |
| 17:25.51 | oilinki | lowtek is right. it's a very good way to start. |
| 17:25.54 | lowtek | We use it for Toilet Training, i.e., take it home and put it next to the dumper, ever time you do your business, read a page or two. Before you know it you got it. |
| 17:26.44 | lowtek | Frankly, our guys have a harder time with vi(m) than with asterisk. |
| 17:29.32 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 17:29.49 | oilinki | esdrasbeleza: for me the most difficult task was to clear out sip.conf and extensions.conf in the way that I could undestand it. |
| 17:34.43 | ManxPower | lowtek: if you understand Linux, networking, NAT, SIP, and telecom, learning Asterisk is not that hard. It's the people that have to learn the basics that have the most problems with Asterisk. |
| 17:34.58 | esdrasbeleza | oilinki, thanks, I'll try to read a bit more. I'm just waiting for a machine that was promised to me to study Asterisk and was trying to get some orientations, thanks |
| 17:35.56 | WimpMan | ManxPower: True, but the way * is configured still seems a little special. |
| 17:36.02 | WhiteWolf | ManxPower: i agree, i picked up the basics of asterisk from the demo file(s) |
| 17:36.13 | WhiteWolf | hm, it's not unique in nature |
| 17:36.42 | WhiteWolf | it's essentially (for ext.conf) a procedural language with jump points |
| 17:36.52 | oilinki | esdrasbeleza: happy to help. you can also find some old laptop and turn it in to linux server for study. if you need to have help in there, I would be happy to come to brasil as I have not visited the place yet ;) |
| 17:38.01 | WimpMan | I didn't say it's inapropriate or something, jusst a little special. |
| 17:38.19 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 17:38.23 | WhiteWolf | that's not what i said either... i said it's not really special =P |
| 17:39.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:40.01 | WimpMan | Thinking again the appropriate bit might only be true as long as I stick to my optionion that * is great for ivr, a goot voip gateway, but no good as a pbx. |
| 17:40.54 | WhiteWolf | i'm not going there today |
| 17:41.00 | oilinki | WimpMan: why so? |
| 17:41.27 | *** join/#asterisk brentc (n=bc316@slim.netnet.net) |
| 17:42.32 | WimpMan | PBXes stand and fall with their support for supplementary services, wich are round about non existant for *. |
| 17:43.21 | brentc | Anyone know where I can find some info on how to size a server for running asterisk as a softswitch? |
| 17:43.28 | ManxPower | WimpMan: odd that you mention that, since Asterisk isn't really a PBX -- It's a PBX toolkit that lets you build a PBX. |
| 17:43.54 | WimpMan | All those little bits and pieces every small off the shelf black box will happily do to make your every day telefone usage a nice experience. |
| 17:44.00 | oilinki | WimpMan: ok I understand that. but I think most of the pbx functionalities are not used by anyone. |
| 17:44.17 | [TK]D-Fender | Asterisk is a POTS interface for me to make coffee with. |
| 17:44.42 | styelz | heh |
| 17:44.47 | WimpMan | ManxPower: That's where I differ. I think it lacks some essential parts to be able to build a good pbx. |
| 17:44.55 | WhiteWolf | such as? |
| 17:44.57 | WhiteWolf | (i lied) |
| 17:45.03 | WimpMan | [TK]D-Fender: Yes, that is a good usage :-) |
| 17:45.06 | esdrasbeleza | oilinki, my employers will give me some machine to test Asterisk, I'll try to study. And I'd appreciate your help, but I guess my boss wouldn't like the idea of paying your airplane tickets, lol |
| 17:45.39 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-1213.yubc.net) |
| 17:45.45 | oilinki | WimpMan: we need call on hold, queuing, call transfer. but not too much more. |
| 17:46.11 | WimpMan | e.g. all sorts of forwarding, CCBS, CCNR, displaying all numbers on forwarded calls. |
| 17:46.43 | oilinki | esdrasbeleza: heh :) we'll I have to come there some day by own cost :) |
| 17:47.11 | WimpMan | BFL and pickup are good already, but that's about it. |
| 17:47.31 | oilinki | BFL? |
| 17:47.54 | WimpMan | Busy lamp function (or was it field?) |
| 17:48.22 | oilinki | ok. thanks. |
| 17:48.30 | Qwell | field |
| 17:48.32 | WimpMan | However with picup we are back to good signalling again, the caller isn't notified that he reached someone else. |
| 17:48.49 | dushantch | Hi. I need some help with extensions.conf. I need to make this exten => 1,1,Dial(SIP/1,40,Ttr) but for 2,3,4,5... like exten => 5,1,Dial(SIP/5,40,Ttr), but universaly, is there a way? |
| 17:49.05 | lowtek | WimpMan: Asterisk is the embodiment of a (P)rivate (B)ranch E(x)change, your points have no merit. |
| 17:49.11 | jaytee | dushantch, read the book |
| 17:49.14 | jaytee | ~book |
| 17:49.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:49.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:49.41 | [TK]D-Fender | dushantch: exten => _[1-5],1,Dial(SIP/${EXTEN},40,Ttr) |
| 17:49.56 | [TK]D-Fender | dushantch: And go read up on dial PATTERNS. This is 101 material. |
| 17:50.06 | dushantch | [TK]D-Fender: thanks |
| 17:52.48 | *** join/#asterisk jackson__ (n=jackson@96-42-218-105.dhcp.roch.mn.charter.com) |
| 17:54.20 | WimpMan | still thinks about the pbx stiff. Does any standsard way exist for SIP to notify users about an activated call forwarding, like e.g. VM notifications? |
| 17:54.37 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:55.08 | esdrasbeleza | oilinki, Asterisk RealTime seems to be what I'm looking for |
| 17:56.14 | [TK]D-Fender | WimpMan: No, there is no standard. "call forwarding"? WHOSE call-forwarding? Indecated HOW? |
| 17:56.49 | oilinki | esdrasbeleza: good to hear. let us know how you get along with it. |
| 17:56.53 | [TK]D-Fender | WimpMan: phone-based CF is not possible. |
| 17:57.25 | WimpMan | Usually if someone activates some sort of forwarding, you'd expect all Telephones assocoated with the number that's being forwarded to display some sort of reminder. |
| 17:57.33 | [TK]D-Fender | WimpMan: Server-based you can indicate with a Custom DevState |
| 17:57.39 | lowtek | WimpMan: Polycom's use just that. |
| 17:57.58 | lowtek | WimpMan: Polycom's do just that with their forwarding feature. |
| 17:58.11 | WimpMan | Ok |
| 17:58.17 | lowtek | WimpMan: A nice little icon that indicates the phone is forwarded shows up on the display. |
| 17:59.07 | dushantch | thanks again [TK]D-Fender , I've set up this a year or so ago so I've forgot, I've got it to work now :) |
| 17:59.12 | [TK]D-Fender | lowtek: I would think he's referring to informing OTHER PHONES that a phone they are watching is "forwarded", for which it will not do by itself |
| 17:59.20 | WimpMan | I think I need to find out where they are available. |
| 17:59.25 | lowtek | Ahh ... |
| 18:00.27 | WimpMan | No phones being watchd, other phones that would otherwise also being rung for calls on the number being forwarded. |
| 18:00.34 | [TK]D-Fender | WimpMan: And you haven't clarified your needs. |
| 18:00.54 | lowtek | Well, I have no clue. |
| 18:01.22 | [TK]D-Fender | WimpMan>No phones being watchd, other phones that would otherwise also being rung for calls on the number being forwarded. <- HUH!?! |
| 18:02.25 | WimpMan | If you have some number that is routed to several phones. |
| 18:02.51 | WimpMan | Shouldn't be that special. |
| 18:03.02 | oilinki | is there a way to 'push' the callerid to a sip-pstn provider so that the original callerid would be shown to the receiving party? in case: sip-call-> voip-provider -> landline/mobile terminal? |
| 18:03.03 | [TK]D-Fender | WimpMan: Please try to phrase that is a COMPLETE sentence fully indicating who is doing what exactly. |
| 18:03.19 | lowtek | lol, or put down the bottle for a few minutes |
| 18:03.22 | oilinki | or some ways to try out if it works. |
| 18:04.18 | WimpMan | You have an exten => 1234,1,Dial(sip/phone1&sip/phone2&sip/phone3) |
| 18:04.25 | *** join/#asterisk metfan2007 (n=jc@189.135.203.16) |
| 18:04.31 | metfan2007 | Hi all!!! |
| 18:04.56 | WimpMan | Ok, bad example as it dosn't allow CF, but you get the idea. |
| 18:04.56 | metfan2007 | I'm receiving the following message in my Asterisk CLI: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 18:05.06 | metfan2007 | what does that means? |
| 18:05.18 | [TK]D-Fender | WimpMan: You're right that is a bad example an you aren't clearing anything up any better. |
| 18:05.55 | [TK]D-Fender | metfan2007: it means you should enable channel debug and show us what you're spooling as well. |
| 18:06.01 | lowtek | WimpMan: In that example, if a phone was forwarded, it would in fact forward the call and the other phones may or may not ring depending on how fast the sip response would be returned from the forwarded phone ... |
| 18:06.03 | WimpMan | Now guy at phone1 decides to tell * to forward exten 1234 to whereever. Now user2 and user3 should be able to see what has happened. |
| 18:06.20 | [TK]D-Fender | WimpMan: Not possible <- |
| 18:06.29 | *** join/#asterisk hardwire (n=hardwire@rdbk-12639.mtaonline.net) |
| 18:06.39 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-176.dsl.sfldmi.sbcglobal.net) |
| 18:07.04 | SamuraiDio | can someone help me with a basic task: http://pastebin.com/d104e0215 : is the 503 error send by the SERVER or CLIENT? |
| 18:07.24 | deeperror | anyone work for digium know why astricon costs so much? |
| 18:07.48 | [TK]D-Fender | SamuraiDio: <-- SIP read from CLIENT_IP_ADDRESS:5060: <-- * is READING a SIP message. What do YOU think? |
| 18:08.56 | SamuraiDio | [TK]D-Fender, so, the problem is that the client is blocking the asterisk to check it's identify (or something like that)? |
| 18:09.35 | [TK]D-Fender | SamuraiDio: That Cisco gateway is sayying "I don't like qualify packets so stop &^$#ing sending them" |
| 18:09.39 | deeperror | I also find it a joke that a week after spending 600+ dollars to goto the con they send me to the same e-mail address I used to sign up a 25% discount on costs....?!? |
| 18:09.49 | SamuraiDio | [TK]D-Fender, thanks |
| 18:10.16 | fogo | Where can I start troubleshooting poor audio on zaptel calls? |
| 18:10.33 | Qwell | deeperror: many conferences cost far more than that.. and as far as the discount, I don't know - though I think you can pass that discount to a friend |
| 18:10.50 | *** join/#asterisk matt_keys (i=smatt@usaregs.com) |
| 18:10.56 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:11.17 | deeperror | Qwell, i could if i have friends... just stinks to pay 600 bucks then 3 days later find out if I had waited I could have saved 150 bucks |
| 18:11.51 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 18:11.52 | deeperror | esp when the entire purpose of the expo is to make money from sponsors and advertisers. I have to pay to get spam for the next year |
| 18:12.38 | [TK]D-Fender | deeperror: I get my spam for free... |
| 18:12.46 | Qwell | deeperror: if it's any condolence, the flight I booked on Friday is $75 cheaper today |
| 18:12.55 | deeperror | haha |
| 18:12.57 | Qwell | it happens.. |
| 18:13.11 | deeperror | yea i'm not paying anyways but still...principal |
| 18:13.56 | Qwell | deeperror: you'll just have to go to all the vendors and get a bunch of schwag to make up for it :) |
| 18:14.08 | deeperror | haha believe |
| 18:14.26 | deeperror | i'll have at least 100 bucks worth of stress balls |
| 18:14.53 | *** join/#asterisk shine (n=stroll@lamantin.achamo.net) |
| 18:15.03 | Qwell | if you don't mind wearing random conference t-shirts, you don't have to buy clothes all year :p |
| 18:15.42 | fogo | I'm getting some really bad results from zttest... does this reflect on the card, the hardware, or both? |
| 18:16.05 | Qwell | fogo: what is the card, and what are the results? |
| 18:16.10 | deeperror | unless you're a big guy...those vacuum sealed bricks that turn into shirts are always too small |
| 18:16.16 | *** part/#asterisk carlf (i=carlf@photocarl.org) |
| 18:16.39 | fogo | Qwell: TE410P; Best: 100.000 -- Worst: 87.108 -- Average: 98.752113 (244 passes) |
| 18:17.04 | Qwell | fogo: I'd suggest calling Digium support |
| 18:17.36 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:18.56 | WimpMan | fogo: Try it in another slot. |
| 18:19.19 | oilinki | Qwell: is it possible to call digium with voip? :) |
| 18:19.28 | Qwell | it is |
| 18:19.38 | Qwell | oilinki: dial 500 from the sample extensions.conf |
| 18:19.49 | deeperror | is astricon more like a defcon or is it still like a von? |
| 18:19.59 | Qwell | VON is all businessy |
| 18:20.03 | oilinki | Qwell: that's cool |
| 18:20.05 | Qwell | (was) |
| 18:20.16 | Qwell | astricon has technical talks and such |
| 18:20.50 | deeperror | Qwell, haha yea was....i went there packing light dressed like a hacker and felt a bit out of place...but then maybe everyone else was out of place? |
| 18:21.09 | deeperror | never been to astricon so this will be a first |
| 18:21.19 | Qwell | astricon is far far funner than VON |
| 18:21.24 | deeperror | very good |
| 18:21.32 | fogo | WimpMan: can't - only 1 PCI slot in this server - other 2 are PCI-E |
| 18:21.33 | Qwell | learn a lot more too.. though, I've never actually been to any talks at VON |
| 18:21.38 | outtolunc | i went to the San Jose VON, i was one of the *very* few with a backpack |
| 18:21.52 | deeperror | that is what i was hoping to hear....i had the choice of one or the other last year and I chose poorly |
| 18:21.58 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 18:22.26 | deeperror | i was at boston von...it was going on after the world series win so that added some excitement haha |
| 18:22.37 | WimpMan | fogo: Hmm. Might be you want to put it into another machine then. |
| 18:22.51 | Qwell | deeperror: what year? |
| 18:22.57 | deeperror | last year? |
| 18:23.02 | deeperror | maybe fall von |
| 18:23.19 | Qwell | I forget which one I went to now |
| 18:23.48 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 18:23.54 | deeperror | for a while i thought it was earlier this year but turns out was like october last year....time has no reference |
| 18:25.06 | oilinki | what do I need if I wish to push the callerid to the pstn network? |
| 18:25.17 | [TK]D-Fender | oilinki: a provider that will allow you to. |
| 18:25.58 | oilinki | [TK]D-Fender: how about the asterisk configuration and what is send to the provider? |
| 18:26.22 | [TK]D-Fender | oilinki: None. As long as you aren't forcing it and they accept it, all is good. |
| 18:27.17 | oilinki | [TK]D-Fender: currently I only get to show my call-out number |
| 18:27.42 | [TK]D-Fender | oilinki: And like usual you have confirmed nothing of value. |
| 18:29.29 | oilinki | [TK]D-Fender: I saw some of the config items in sip.conf but I tought that it's just for sip-callerid |
| 18:29.44 | [TK]D-Fender | *yawn* |
| 18:30.53 | *** join/#asterisk juniowww (n=juniowww@189.58.30.73.adsl.gvt.net.br) |
| 18:31.14 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 18:31.57 | brentc | How many calls/second can asterisk handle? |
| 18:32.30 | Juggie | how far is earth from the sun |
| 18:32.51 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:33.07 | oilinki | tranfering the callerid to a sip phone is not an issue, but tranfering it to via sip-provider to gsm network does not work |
| 18:33.20 | bkruse | Juggie: not 9 miles a second, I know that :P |
| 18:33.34 | brentc | haha. |
| 18:33.37 | bkruse | (stabs at not valid "performance" tests) |
| 18:33.54 | Qwell | brentc: there are far too many variables to answer a question like that. |
| 18:33.59 | bkruse | brentc: are you just doing the signalling? Or do you mean asterisk handling the call (rtp) as well |
| 18:34.12 | [TK]D-Fender | bkruse: * can make the kessel run in less than 12 parsecs.... |
| 18:34.16 | brentc | yes just signalling |
| 18:35.01 | bkruse | [TK]D-Fender: heh, exactly, and that's _with_ transcoding! |
| 18:35.24 | brentc | not having much success in finding it documented anywhere. |
| 18:35.29 | bkruse | brentc: Just the signalling, and either way really, it can handle a _lot_, however, if you are just doing signalling, and having Asterisk behind it (we are talking thousands of calls, in general) look at openser |
| 18:35.52 | bkruse | Then for the actual pbx, you use Asterisk of course. I would try with Asterisk first, however, as the load I am thinking of may be higher than what you are thinking of |
| 18:36.01 | bkruse | brentc: Because it's a hard thing to document |
| 18:36.12 | lowtek | brentc: What are you trying to do? |
| 18:37.23 | brentc | User asterisk as a softswitch for 10k lines |
| 18:37.43 | brentc | and I need to size the servers accordingly |
| 18:37.55 | lowtek | Are you transcoding? |
| 18:38.39 | [TK]D-Fender | brentc: You should be using SER for something like that, and * as a back-end app server |
| 18:39.13 | DarylVOIP | Yeah...you really don't want to tru registering that many phones to *, or doing the signalling. |
| 18:39.16 | brentc | I had a feeling I was going to have to do that. |
| 18:39.28 | DarylVOIP | You can easily do that with a commercial SBC or OpenSER if you're willing to work for it. |
| 18:40.09 | DarylVOIP | OpenSER will pretty much just work, but redundancy will require some "creativity" as opposed to a cluster of commercial SPBc (like a Sansay SPX or a Nextone) |
| 18:40.10 | brentc | there were trying to do this on the cheap. So it is looking OpenSER |
| 18:40.11 | brentc | damn.. |
| 18:40.12 | Blackvel | when connecting a pattern gateway to * could the sip.conf section look like [pattongw] type=peer username=patton123 fromuser=pstnnumber host=dynamic? MUST [pattongw] be the same like username=pattongw? |
| 18:40.18 | DarylVOIP | Yes...it can definitely be done. |
| 18:40.48 | DarylVOIP | depending on how redundant you need to be, you'll need up to 6 boxes. |
| 18:41.07 | DarylVOIP | (2 DB, 2 OpenSER, 2 load balancers) |
| 18:41.18 | *** join/#asterisk brettnem (n=brettnem@user-387oe35.cable.mindspring.com) |
| 18:41.20 | DarylVOIP | But can likely do it with 4 and be pretty much OK. |
| 18:41.41 | DarylVOIP | Bonus if you already have a DB cluster you can toss acouple of light tables on. |
| 18:42.17 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 18:42.32 | brentc | decisions decisions |
| 18:43.03 | DarylVOIP | OpenSER seems horribly obscure at first, but once you get into it, it makes a lot of sense. |
| 18:43.24 | bkruse | brentc: It's not as hard as you think, and it's free.... |
| 18:43.34 | *** join/#asterisk profx (n=fook@unaffiliated/wolferine) |
| 18:44.11 | profx | when using X-lite, i have discovered that the Xfer button doesn't work... is there a reason for this? or does no one use X-lite anymore? |
| 18:44.18 | DarylVOIP | Definitely. And the best starting point is probably http://www.sipwise.com/wizard/ |
| 18:44.28 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:44.28 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:45.00 | brentc | Would you run it on linux or solaris? |
| 18:45.03 | lmadsen | Corydon76-dig: I'm curious if there is a good way to force res_odbc to re-read the odbc.ini file? It seems to do it when a database fails, but when I want to fail back over to my remote database, now I have both the local and remote DBs running, so there is no 'failure' to cause res_odbc to re-read the odbc.ini file |
| 18:45.24 | lmadsen | doing a 'module reload res_odbc.so' doesn't seem to cause it to re-read that odbc.ini file |
| 18:45.35 | DarylVOIP | linux unless you want extra challenges :) |
| 18:45.56 | DarylVOIP | Unless you happen to have a good sparc box around already. |
| 18:46.04 | DarylVOIP | but on Intel hardware, I'd just use Linux |
| 18:46.26 | DarylVOIP | Of course, if you're in a Solaris shop, it might make sense to keep it that way. |
| 18:46.28 | brentc | plenty of sun boxes here. |
| 18:46.53 | Corydon76-dig | lmadsen: that's not really the layer that does that |
| 18:47.00 | brentc | But we do have quite a bit of linux as well |
| 18:47.13 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 18:47.17 | lmadsen | Corydon76-dig: oh ya? ah, that makes sense then why the reload doesn't do anything |
| 18:47.58 | DarylVOIP | OpenSER works fine, but I had some issues with RTPProxy compiling, as well as some RADIUS stupidity on solaris. It worked easier on Linux, but I suppose that all depends on what you're most familiar with. |
| 18:48.10 | DarylVOIP | And you may not even need wither of those. |
| 18:48.51 | Juggie | google chrome is up. |
| 18:51.20 | brentc | oh ok. great the boss just stopped by now they want it built by Friday... |
| 18:51.35 | brentc | well I know what I'm doing for the next few days |
| 18:52.09 | oilinki | [TK]D-Fender: I don't understand how to do this. I get an call from psnt-> voi provider -> asterisk -> sip terminal. I get the callerid. |
| 18:52.15 | DarylVOIP | brentc - go to the sip:wise link I sent you and /join openser now :) |
| 18:53.16 | oilinki | but when calling pstn-> voip provider -> asterisk -> voip provider2 -> gsm-phone I get the voip-provider2 call-out number. |
| 18:53.19 | [TK]D-Fender | oilinki: And you have not learned the concept of "details" yet. |
| 18:53.27 | brentc | didn't get the link |
| 18:53.44 | DarylVOIP | http://www.sipwise.com/wizard/ |
| 18:53.49 | DarylVOIP | Good strating point for a config |
| 18:54.13 | brentc | thanks for all the info! |
| 18:54.16 | oilinki | how can I pass the A-party number to the voip-provider in the way that it would be passed to the gsm-number? |
| 18:54.47 | lmadsen | oilinki: typically that is a function of the voip-provider allowing you to do that -- and that would be done with 'sendrpid=yes' in sip.conf |
| 18:55.54 | oilinki | lmadsen: ok thanks. that's already set. therefore I suppose the voip-provider2 will not allow to do it. |
| 18:57.53 | oilinki | I suppose I need to talk with them, but most likely they are not going to change the policy. |
| 18:58.48 | lmadsen | most times they won't allow you to change the CID because that opens them up to you possibly performing a CID spoofing attack |
| 19:00.23 | oilinki | lmadsen: yes. that's undertandable. |
| 19:00.35 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:00.44 | afrosheen | btw anyone using Redhat enterprise 4 and sangoma, the latest drivers won't build |
| 19:03.15 | afrosheen | sangoma is working on the problem, should have it fixed and a new driver release soon |
| 19:04.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:05.25 | *** join/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 19:05.52 | [TK]D-Fender | afrosheen: Keep off the bleeding edge |
| 19:06.05 | afrosheen | uh |
| 19:06.09 | afrosheen | redhat 4 isn't bleeding |
| 19:06.15 | afrosheen | 5.2+ is |
| 19:06.28 | [TK]D-Fender | afrosheen: Who said I was talking about your OS? |
| 19:06.43 | [TK]D-Fender | ~assume |
| 19:06.43 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 19:06.56 | afrosheen | ok, you wanna flip it that way, they have a stable branch and beta drivers, these were their stables |
| 19:07.09 | afrosheen | anyway sangoma is working on it, will have fixed drivers this week I imagine |
| 19:07.19 | [TK]D-Fender | afrosheen: I wasn't flipping, you just assumed the target :) |
| 19:07.40 | afrosheen | beta would be bleeding :) |
| 19:07.47 | Juggie | thats a lame assumption quote |
| 19:07.55 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 19:08.13 | [TK]D-Fender | Juggie: SLJ is the shiznit y0 |
| 19:08.17 | afrosheen | sangoma support said they've heard from 3 other people with the same problem |
| 19:08.19 | *** part/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 19:08.25 | afrosheen | go redhat, way to break compat :) |
| 19:08.33 | Sargun | redhat is a bitch. |
| 19:08.45 | [TK]D-Fender | afrosheen: Yeah, pretty soon they'll have pissed half a dozen people off! |
| 19:08.48 | afrosheen | this is the first problem I've had with redhat enterprise |
| 19:09.05 | Sargun | afrosheen, how much/how long have you been using it? |
| 19:09.05 | Juggie | [TK]D-Fender, http://www.youtube.com/watch?v=wg4trPZFUwc this would be better :) |
| 19:09.25 | afrosheen | it's been on our production asterisk server for about 2.5 years |
| 19:09.29 | afrosheen | give or take |
| 19:09.43 | n3hxs | Redhat? leaves red marks on my forehead. |
| 19:10.04 | afrosheen | cue distro trolls |
| 19:10.12 | [TK]D-Fender | Juggie: Under Seige? Bleh -> http://geekadelphia.com/wp-content/uploads/2008/04/steven_segal_emotion_chart.jpg |
| 19:10.35 | [TK]D-Fender | (2) |
| 19:11.28 | *** join/#asterisk kalel008 (n=tamato@41-195-84-21.access.uunet.co.za) |
| 19:11.44 | afrosheen | it just worries me, maybe I should report the problem to redhat also...if they broke something in the kernel, in the future other stuff may fail to build |
| 19:11.51 | kalel008 | hi there |
| 19:12.03 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 19:12.13 | kalel008 | i want to ask something |
| 19:12.17 | [TK]D-Fender | ~ask |
| 19:12.18 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:12.37 | afrosheen | ask is probably? you fail english jbot |
| 19:12.49 | kalel008 | does it matter were i put my incomming DID's in extensions.conf can i put them in any context |
| 19:12.55 | *** join/#asterisk km- (n=pgrace@2001:4830:1215:2:20c:29ff:fee9:9689) |
| 19:12.55 | Sargun | afrosheen, once it works, it works... but then you never touch it again. |
| 19:13.28 | afrosheen | sargun: yeah I had held off upgrading the kernel for a very long time, fearing a problem like this would spring up |
| 19:13.42 | afrosheen | then to attempt it on labor day weekend when NOBODY was working |
| 19:13.43 | afrosheen | :( |
| 19:13.48 | [TK]D-Fender | kalel008: Don't call it a "DID". In extensions.conf its just an EXTENSIONS : a number dialed. And you can put it wherever you want so long as its in the place that your inbound channel is LOOKING for it. |
| 19:13.51 | Sargun | afrosheen, Uhm, yeah, bad idea |
| 19:13.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:14.01 | Sargun | I'm running asterisk on 2.6.15 |
| 19:14.11 | afrosheen | sargun: what distro |
| 19:14.46 | kalel008 | oh ok thanks the book is a major help i had to read it 3 times to understand a bit :) |
| 19:15.33 | Sargun | afrosheen, Gentoo. |
| 19:15.48 | Sargun | Yes, yes, I know its not a "production distro" |
| 19:15.49 | afrosheen | Sargun, is that a production box or a hobby box |
| 19:15.52 | afrosheen | ;) |
| 19:15.58 | Sargun | but its a production box. |
| 19:16.17 | Sargun | Its the most vanilla distro I could find. |
| 19:16.37 | afrosheen | eh, personally I could just as easily use centos but the boss said "get something with a contract and a phone number" |
| 19:16.51 | afrosheen | public company etc. |
| 19:20.03 | km- | have you guys ever heard of someone configuring a T1 with fxsks= instead of e&m= ? |
| 19:20.19 | km- | this guy is telling me up and down that fxsks= works for a T1 and I'm needing a fact check |
| 19:20.25 | km- | fairly certain fxs signalling is not valid on a T1 |
| 19:20.45 | km- | I'm assuming the channel driver module is ignoring the fxsks= and making the line e&m or something. |
| 19:21.46 | Sargun | hehe |
| 19:23.46 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 19:24.16 | hardwire | does asterisk show a significant performance gain when running on top of a 64 bit kernel? |
| 19:24.25 | hardwire | and compiled 64 bit.. :) |
| 19:24.39 | hardwire | I'm wondering how imperative it may be |
| 19:24.46 | hardwire | and I don't have the means to test it :( |
| 19:24.52 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 19:25.06 | aliver | http://pastebin.com/d2479b208 <-- my macro is screwy |
| 19:25.16 | aliver | I'm thinking somehow I'm using MACRO_EXTEN wrong. |
| 19:26.00 | aliver | I thought that was supposed to give me the number that was called going into the extension ie.. ${EXTEN} before the macro gets kicked off. |
| 19:26.13 | aliver | This used to work with 1.2, but I've upgraded to 1.4. |
| 19:26.22 | aliver | I can't find anything that says that variable was deprecated, though. |
| 19:26.41 | WhiteWolf | you're missing a $ |
| 19:26.51 | WhiteWolf | line 12 of the pastebin |
| 19:26.56 | aliver | Doh! |
| 19:27.11 | aliver | Good catch. |
| 19:27.14 | *** join/#asterisk ant (n=aburton@209.11.127.13) |
| 19:27.17 | aliver | Gah. I'm blind! |
| 19:27.19 | WhiteWolf | # |
| 19:27.19 | WhiteWolf | <PROTECTED> |
| 19:27.22 | WhiteWolf | it told you the problem |
| 19:28.24 | [TK]D-Fender | yup. |
| 19:28.39 | WhiteWolf | as asterisk often does |
| 19:28.45 | aliver | my bad |
| 19:28.55 | aliver | thanks for wet-nursing me, though. |
| 19:29.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:29.07 | aliver | second set of eyes and all that. |
| 19:29.08 | WhiteWolf | it's all good |
| 19:29.53 | jaytee | hardwire, I'm running 64 bit * with 64 bit RHEL 5 and it's very robust but not significantly faster than 32 bit on similar hardware. |
| 19:30.20 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-6e0df723c564404e) |
| 19:30.21 | hardwire | danke |
| 19:31.26 | kensuke_ | Hi, i have a sangoma 104...i can call for port 1(pri_net) an received for port 2(pri_cpe) link this ports for a e1 cable? |
| 19:31.33 | kensuke_ | is possible o i'm crazy? |
| 19:32.12 | jaytee | oh! you engrish velly bad! |
| 19:33.13 | seanbright | racism... it's whats for dinner. |
| 19:34.02 | [TK]D-Fender | seanbright: The grammatically-challenged are not a "race" |
| 19:34.16 | seanbright | sigh |
| 19:34.17 | jaytee | seanbright, no, that's "Nothing - it's what's for dinner" with the picture of the starving Somali kids holding empty plates. |
| 19:34.53 | km- | d-fender: hey, do you have strong experience with T1/E1? My question above. I'm almost 100% sure I'm right |
| 19:35.37 | km- | d-fender: in a nutshell, I posit "fxsks=1-24" is invalid on a T1 span, it should be e&m or bchan/dchan, or some combination otherwise, but definitely not fxsks or fxoks. True? |
| 19:36.20 | jaytee | true if it's PRI but false if it's just T1 to channel bank on both ends |
| 19:36.57 | [TK]D-Fender | km-: signalling is perfectly normal for a channel-bank / "analog" T1 |
| 19:37.38 | km- | I thought that in that case it's e&m or e&m wink |
| 19:37.41 | km- | not fxs/fxo |
| 19:38.42 | [TK]D-Fender | km-: Then you are mistaken. |
| 19:38.45 | jaytee | I've run T1's with digital and analog lines from a channel bank to a Nortel and used both loop start for analog and e&m wink for the digital phones. |
| 19:39.16 | km- | are fxsks and e&m functionally different signalling? |
| 19:39.44 | km- | I vaguely remember mark writing an e-mail about this to asterisk-dev years ago |
| 19:39.48 | km- | off to google I go |
| 19:39.54 | jaytee | similar but not functionally compatible |
| 19:41.27 | km- | this guy's config reads, fxsks=1-15, dchan=16, fxsks=17-31 for the E1 |
| 19:41.28 | x86 | e&m has better disconnect supervision and what not, I hear |
| 19:41.29 | km- | is that even valid? |
| 19:41.55 | x86 | km-: usually you use fxsks=1-15,17-31, dchan=16 |
| 19:42.10 | km- | wouldn't it be bchan=1-15,17-31 dchan=16? |
| 19:42.28 | km- | otherwise what is the dchan really marshalling? |
| 19:42.34 | km- | it's not like it's controlling the fxs lines? |
| 19:44.30 | kaldemar | it's the 16th channel of a span for e1. it doesn't matter to how many lines you put it. |
| 19:44.58 | x86 | kaldemar: in most cases it's the 16th channel, doesn't have to be |
| 19:45.12 | km- | no, I mean |
| 19:45.12 | x86 | km-: yeah it's bchan on a PRI, you're right |
| 19:45.13 | kaldemar | is asterisk's case it it the 16th channel. |
| 19:45.23 | kaldemar | it is |
| 19:45.23 | x86 | kaldemar: no, not always |
| 19:45.23 | km- | yeah, x86 sees where I'm going |
| 19:45.36 | km- | fxsks= shouldn't be valid for PRI B channels |
| 19:45.55 | x86 | shrugs, haven't ever used a PRI with asterisk yet |
| 19:45.58 | [TK]D-Fender | km->this guy's config reads, fxsks=1-15, dchan=16, fxsks=17-31 for the E1 <- nuts |
| 19:46.05 | x86 | just CAS T1's and stuff :p |
| 19:46.12 | [TK]D-Fender | km-: analot T1/E1 doesn't HAVE a D-Chan |
| 19:46.18 | km- | d-fender: exactly. |
| 19:46.26 | x86 | right |
| 19:46.38 | [TK]D-Fender | km-: Put. Down. The. Crack. Pipe. (c) JerJer |
| 19:46.39 | x86 | so is he using an analog T1 or PRI? :) |
| 19:46.42 | km- | haha |
| 19:46.44 | km- | jerjer++ |
| 19:46.49 | km- | see, thats where this gets interesting |
| 19:47.10 | km- | I can't seem to get a straight answer on that, when I ask him, "what is the signalling on the line" he says "this is the config that worked" |
| 19:47.23 | km- | and it has the span defined with cas/hdb3 |
| 19:47.31 | km- | and then shows me this fxsks/dchan/fxsks config |
| 19:47.46 | km- | and I'm wondering whether the whole E1 is just one big analog trunk and he's mis-signalling channel 16 |
| 19:48.00 | [TK]D-Fender | km-: "Where were you tuesday night at 8pm?" -> "THE SKY IS BLUE! THE SKY IS BLUE!" |
| 19:48.01 | WimpMan | km-: Then ask the guy what it's connected to. |
| 19:48.11 | km- | an avaya switch on the other side |
| 19:48.18 | [TK]D-Fender | km-: http://208.101.59.66/bunny-pancake.jpg |
| 19:48.39 | WimpMan | km-: Looks like you want it's config as well then. |
| 19:49.05 | *** join/#asterisk existenz (n=Ameen@unaffiliated/existenz) |
| 19:49.26 | existenz | Hello, what does this error mean: [Sep 2 22:48:10] WARNING[2259]: chan_sip.c:8385 check_auth: username mismatch, have <line1>, digest has <pstn> ? |
| 19:50.16 | km- | d-fender: hahaha. |
| 19:52.06 | jaytee | hahahaha, that's precious!! got any with a waffle? |
| 19:53.46 | [TK]D-Fender | jaytee: waffling is post-digestion |
| 19:54.05 | tzafrir_laptop | hmm... E1 CAS does have channel 16 as a "signalling channel" |
| 19:54.40 | km- | tzafrir: err? |
| 19:54.40 | ManxPower | tzafrir_laptop: E-1 CAS is not all that common it seems. |
| 19:54.58 | ManxPower | And it does not apply to PRI or BRI |
| 19:55.02 | tzafrir_laptop | ManxPower, depends where. E1 CAS is used for MFC/R2 |
| 19:55.18 | *** join/#asterisk _ShrikE (n=_ShrikE@64.128.34.118) |
| 19:55.20 | km- | tzafrir/manx: so is the dchan= line valid in this case? |
| 19:55.27 | ManxPower | tzafrir_laptop: then they are E-1 MRC/R2 aren't they? |
| 19:55.38 | ManxPower | km-: what kind of line do you have? |
| 19:55.49 | _ShrikE | got his power back! |
| 19:55.59 | ManxPower | Do you have PRI or do you have CAS E-1/MFC R2 |
| 19:56.08 | km- | just a sec, talking to the telephony team |
| 19:56.22 | ManxPower | km-: you cannot proceed until you know. |
| 19:56.29 | km- | agreed |
| 20:04.10 | km- | interesting |
| 20:04.38 | km- | the avaya guys tell me that regardless of it being CAS, Avaya switches automatically assume channel 16 on every E1 is for signalling |
| 20:04.40 | WimpMan | km-: It's not an E1 at all? :-) |
| 20:04.58 | km- | so that's why they did it that way. |
| 20:05.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:05.14 | ManxPower | km-: too bad Asterisk does not support channel 16 for signaling on CAS T-1. |
| 20:05.37 | ManxPower | you understand that CAS = Channel Associated Signaling, right? As in the signaling is part of the channel, not on some other channel. |
| 20:05.42 | km- | yeah. |
| 20:05.45 | kaldemar | too bad asterisk doesn't seem to support other timeslots for signalling. |
| 20:05.49 | ManxPower | By definition if you have a plain CAS T-1 or E-1 you don't have a signalling channel |
| 20:05.56 | km- | agreed |
| 20:06.00 | km- | it sounds like avaya is stupid. |
| 20:06.04 | km- | or our avaya guys are stupid. |
| 20:06.12 | ManxPower | km-: no the techs you are talking to are stupid. |
| 20:06.22 | ManxPower | Why not just turn it into a PRI and be DONE with it? |
| 20:06.33 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:08.06 | km- | Much deeper down the rabbit hole, their response is "they're station cards" |
| 20:08.09 | km- | whatever that means. |
| 20:09.07 | ManxPower | km-: welcome to traditional PBXs, the answer to most questions will be "You can't do that, it's not supported, or you'll have to buy the software code to unlock that feature for $2,000" |
| 20:09.47 | *** join/#asterisk _ShrikE (n=_ShrikE@64.128.34.118) |
| 20:11.12 | *** join/#asterisk jmacz (n=jmacz@190.158.236.60) |
| 20:13.32 | lowtek | Any of you guys ever seen a Divinity (sp?) phone system? |
| 20:14.19 | km- | I think that's Definity? |
| 20:14.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:14.40 | km- | it's a type of avaya |
| 20:14.42 | Qwell | Avaya? |
| 20:15.03 | *** part/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-6e0df723c564404e) |
| 20:15.05 | jaytee | Avaya, formerly Lucent |
| 20:15.06 | MikeJ | The last nortel option 11 I had was previously owned by a church... does that count? |
| 20:15.16 | x86 | Definity is made by Lucent |
| 20:15.16 | ManxPower | Definity was an ATT product, and that part of ATT is now known as Lucent. Someone correct me if I'm wrong. |
| 20:15.34 | lowtek | lol, that's it. Well, these things take up like a small room. We just replaced one with two 1U asterisk servers. I'm trying to get permissions to post before/after pics, it's hilarious. Ever seen that IBM mainframe to micro-server commercial? |
| 20:15.45 | x86 | ManxPower: that's right, but i think they bough (or got bought by) avaya, and assumed avaya's name |
| 20:16.00 | jaytee | ManxPower, they were Lucent then the pbx division split off from Lucent to become Avaya |
| 20:16.17 | ManxPower | x86: that is very possible. I really stopped following it after it became Lucent |
| 20:16.19 | km- | lowtek: haha, nice. |
| 20:16.26 | x86 | lowtek: we used a definity the size of a full rack to power phones for a whole 7-floor building once |
| 20:16.47 | jaytee | as for 2000 bucks to unlock a feature I think it's now gone up to 2500 |
| 20:17.37 | x86 | jaytee: if you're lucky ;) |
| 20:17.47 | ManxPower | jaytee: I think it would have been $4,000 to get the unlock code to enable PRI support on a Nortel MICS |
| 20:17.48 | jaytee | but hey, spending 5K to unlock another 10GB of voicemail storage on a 160GB drive you've already paid for is the American way ain't it? |
| 20:17.58 | x86 | I remember we paid $7500 just for CDR access via SMDR |
| 20:18.01 | ManxPower | So we stuck with E&M/Wink and it works well enough. |
| 20:18.22 | ManxPower | x86: and people think $1,500 for 4 ports is expensive. |
| 20:18.27 | x86 | right ;) |
| 20:18.56 | x86 | ManxPower: we have a T1 card in our Toshiba system here, but you need a whole new card for a PRI, which costs ~$4500 |
| 20:19.02 | lowtek | Yea, those days are over. |
| 20:19.24 | x86 | ManxPower: I guess Mitel is about the same, charging extra / requiring a whole new T1 card just to move to PRI from an analog T1 |
| 20:19.48 | x86 | legacy PBX vendors are crazy... |
| 20:19.52 | ManxPower | x86: Oh I found the new T-1 cards on eBay for like $500 so that was not the problem. |
| 20:20.15 | jaytee | I'd have to go over old prices but I'm pretty sure it's about a factor of 10 cheaper using Digium or Sangoma T1 cards and * than it is using Nortel T1 cards and license codes. |
| 20:20.28 | x86 | Avaya used to (dunno if they still do) charge licensing fees per concurrent channel on G.711u SIP calls |
| 20:20.42 | jaytee | x86, I'd believe it |
| 20:21.01 | x86 | if you wanted G.729, it was way more too heh |
| 20:21.26 | jaytee | Nortel charges both for TN (terminal number licenses) and DN (directory number licenses) as well as ACD-DN's and their AST licenses. |
| 20:21.30 | x86 | ManxPower: oh you mean you have to buy the cards, and then also buy more software licenses? |
| 20:21.37 | lowtek | G.729a is worth the $10/channel, imho, it's a great codec for Internet communications. |
| 20:21.53 | lowtek | A lot wouldn't be possible without it. |
| 20:21.59 | x86 | lowtek: I prefer G.723 over the internet |
| 20:22.10 | x86 | 723 is more latency-tolerable, in my experiences |
| 20:22.18 | ManxPower | x86: sometimes |
| 20:22.26 | lowtek | x86: Do any end-points support it? |
| 20:22.30 | x86 | and less bandwidth than g729 |
| 20:22.34 | ManxPower | Nortel, for example have different cards for PRI and you still have to buy the software unlock. |
| 20:22.47 | x86 | ManxPower: ah |
| 20:23.02 | x86 | ManxPower: it's all a racket heh |
| 20:23.42 | x86 | lowtek: no man, avaya was wanting to charge like $100/channel for g.711u SIP trunking, and a lot more if it was g.729 ;) |
| 20:24.13 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 20:24.20 | lowtek | x86: Wow. Not surprising, though. |
| 20:25.55 | x86 | lowtek: that big 7-story building project I was talking about -- it cost about $10,000 per floor just for analog ports, I forget how much it cost on top of that for digital ports.... and that's not including thousands of feet of 25-pair cables and what not, nor installation |
| 20:26.13 | x86 | $70,000 just for analog ports |
| 20:26.38 | x86 | the whole system was about $220,000, but I don't remember how it all was figured... |
| 20:26.51 | x86 | (and of course, that was before installation) |
| 20:27.05 | x86 | we had DS3 cards in it and everything |
| 20:28.22 | *** join/#asterisk LND (n=Lee@89.192.217.45) |
| 20:29.59 | *** join/#asterisk makkksimal (n=makkksim@f054017179.adsl.alicedsl.de) |
| 20:31.00 | km- | telephony shouldn't cost so much., |
| 20:36.38 | *** join/#asterisk Esculapio__ (n=ESCulapi@adsl-224-238.tricom.net) |
| 20:38.46 | *** join/#asterisk morex (n=m@5ac648f4.bb.sky.com) |
| 20:38.52 | morex | Hi there |
| 20:39.08 | *** part/#asterisk profx (n=fook@unaffiliated/wolferine) |
| 20:39.10 | morex | Ooh I want dev.. |
| 20:39.21 | *** part/#asterisk morex (n=m@5ac648f4.bb.sky.com) |
| 20:42.48 | oilinki | x86: what year was your project? |
| 20:44.31 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 20:44.56 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:45.09 | bkruse | km-: Talk to a hospital or government office about the costs of their telephone system |
| 20:45.12 | bkruse | You'd be surprised... |
| 20:45.24 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 20:45.38 | Qwell | bkruse: or a bank |
| 20:46.15 | bkruse | Qwell: oh ya, good one, it's ridiculous. Huntsville hospitals PBX was well over 2-3 million I believe |
| 20:46.59 | Qwell | if we had problems, we had an Avaya rep out within about an hour |
| 20:47.19 | bkruse | Qwell: True, not to mention they had an office above us didn't they? :P |
| 20:47.29 | bkruse | Err, was that cisco |
| 20:47.31 | Qwell | oh, no, I mean when I was working for the bank |
| 20:47.43 | oilinki | bkruse: how redundant that kind of system need to be? |
| 20:48.04 | bkruse | oilinki: It's pretty crazy, no single point of failure, but it still wasn't 2-3 million dollars worth of "impressiveness" |
| 20:49.41 | oilinki | bkruse: you mean 10k for the system, 10k for the connectivity and the rest for 'great sell talk'? :) |
| 20:50.13 | bkruse | oilinki: Literally something like that. Cisco systems. Call them up for a quote, after dealing with Asterisk systems, you will laugh |
| 20:51.14 | oilinki | bkruse: I used to work for nokia network and the systems/prices were.. well quite different :) |
| 20:51.40 | bkruse | oilinki: That's close to the same. My school of <40 phones with 2 T1's was quoted to be ...... 150k with phones/setup |
| 20:53.14 | oilinki | but there is a point. asterisk is good for small scale system, but the big vendors can sell support, name and the redundancy. |
| 20:56.41 | oilinki | to compete with that needs some adjusting. great support for the system. real redundancy. and with no name, great price :) |
| 20:57.21 | *** join/#asterisk logicwrath (n=no@68.41.24.98) |
| 20:58.35 | oilinki | for smaller companies would be great to have a 'ring of trusted administrators' - people who would be able to support the system in the case the small company would not be able to do it. |
| 20:58.41 | x86 | you can do nice redundancy with Asterisk also |
| 20:59.02 | x86 | automatic T1 failover switches and the whole nine yards |
| 21:00.00 | existenz | Qwell, Would asterisk run on a 400Mhz computer with 128Mhz ram? |
| 21:00.03 | x86 | bkruse: why would you need 2 T1's for less than 40 phones? |
| 21:00.17 | Qwell | existenz: is it x86? |
| 21:00.20 | x86 | existenz: dude Asterisk can run on a WRT54G :p |
| 21:00.28 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 21:00.29 | logicwrath | Is there preferred Linux flavor to run Asterisk in VMware server? |
| 21:00.30 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:00.32 | existenz | yeah, but I still have problems with the playback |
| 21:00.43 | existenz | Qwell, Mobile Pentium II |
| 21:00.47 | x86 | logicwrath: you DONT want to run Asterisk virtualized, it'll be a horrible experience |
| 21:00.50 | bkruse | x86: no idea, _especially_ since it was a max of 40 phones |
| 21:00.58 | bkruse | x86: I bet you there are no more than 3 active lines at a time...._MAYBE_ |
| 21:01.02 | logicwrath | ive done it in the past im aware of the timing issues |
| 21:01.07 | logicwrath | but they seem less appearent |
| 21:01.13 | x86 | bkruse: oh, you didn't engineer that? |
| 21:01.14 | logicwrath | in the recent builds |
| 21:01.15 | bkruse | but they have to have a whole bunch of channels for say.....when a tornado comes, and all the parents are calling in |
| 21:01.29 | existenz | Qwell, the playback sounds very slow =/ |
| 21:01.32 | bkruse | x86: Nah, that would kind of be....conflict of interest I guess |
| 21:01.35 | existenz | I am not sure whether it is because of the cpu |
| 21:01.37 | baliktad | I run asterisk on Virtual Server, I just used Ubuntu server |
| 21:01.45 | existenz | Qwell, I used to run winxp on that computer anyways :p |
| 21:01.48 | clever | bkruse: you can never have enough channels for that, enless you have 2 channels for every student;P |
| 21:01.53 | x86 | bkruse: that's nice, but the parents wont be able to talk to anyone anyway, so why not just give them a busy tone instead? :P |
| 21:02.06 | oilinki | I suppose the server needs to be powerfull only if it needs to do transcoding? |
| 21:02.07 | bkruse | clever: you are _exactly_ right heh, when we had a tornado one day it was CRAZY! |
| 21:02.32 | clever | bkruse: and even then, the phone companys trunk lines might be overloaded:P |
| 21:02.33 | logicwrath | The worst virtual asterisk server experience I had was running FreeBSD in VMWare which had its own timing issues besides teh asterisk playback problems |
| 21:02.37 | bkruse | x86: I was thinking automated message, "We know there is a tornado coming, we do care about your children and business" |
| 21:02.46 | bkruse | (private school stab) |
| 21:03.03 | x86 | bkruse: or switch the main inbound hunt group to go to a pre-recorded message, for say 30 seconds or so, allow the caller to leave a voicemail, and hang up on them |
| 21:03.29 | x86 | that will cut the line usage down considerably, and allow your school administrators to properly prepare for the impending disaster |
| 21:03.30 | kaldemar | existenz: i run asterisk on a crappy 400 MHz PII every now and then. it even runs 6 channels in a meetme over IAX pretty nicely without transcoding. |
| 21:03.45 | existenz | kaldemar, what might be the problem with the playback then =/ |
| 21:03.47 | bkruse | x86: That is genius! I have done some fun stuff at that school, however. The network admin was just a windows certified HS dropout, so he was _amazed_ at linux |
| 21:03.52 | existenz | kaldemar, it sounds very slow |
| 21:04.07 | kaldemar | existenz: no problems with playback either, be that mp3, wav, gsm... |
| 21:04.13 | existenz | weird |
| 21:04.20 | clever | bkruse: my school tech guy i think was a hs dropout and only the school would hire him:P |
| 21:04.29 | Qwell | irony |
| 21:04.36 | clever | i knew more about computers then he did |
| 21:04.41 | clever | and i was a noob back then:P |
| 21:04.43 | bkruse | clever: hehe, not that I am stabbing HS dropouts, it was just funny because he thought he was all that and a bag of chips |
| 21:05.05 | bkruse | I still have root on the webserver too.... |
| 21:05.05 | existenz | kaldemar, can I pm you the number to hear how slow it is? |
| 21:05.09 | clever | bkruse: i have since been kicked out of HS(in grade 11) and self learned many more programing languages |
| 21:05.12 | oilinki | hmm. what is hs dropout? :) |
| 21:05.13 | existenz | kaldemar, if you can, of course. |
| 21:05.13 | logicwrath | i wish i would have dropped out of HS |
| 21:05.18 | bkruse | clever: nice nice |
| 21:05.20 | kaldemar | existenz: no thanks. |
| 21:05.27 | oilinki | history science? |
| 21:05.37 | bkruse | I pretty much was, I only went to school 3 hours a day since I was hired at Digium |
| 21:05.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:06.14 | clever | ive sat at the computer for ~20 hours in a row before |
| 21:06.22 | clever | something you cant exactly do at school:P |
| 21:06.32 | oilinki | ah.. high school. (I did not know it's possible to drop out from there) |
| 21:06.49 | existenz | kaldemar, would it be better to install asterisk not through asterisknow? |
| 21:06.54 | existenz | kaldemar, can it solve the problem |
| 21:06.55 | x86 | <-- highschool dropout, college dropout ;) |
| 21:07.01 | clever | oilinki: you can allways stop going, though your parrents may force you back if its 3rd grade:P |
| 21:07.15 | bkruse | I finally finished high school, as I was coming up on my second year @ digium, hurray |
| 21:07.17 | x86 | I've got about 8 certs now, and make real decent money (for the area) |
| 21:07.31 | x86 | bkruse: how old are you? |
| 21:07.36 | bkruse | x86: 18 |
| 21:07.36 | kaldemar | existenz: i've never used asterisknow, but some say that other than compiling from source is the source of all evil. |
| 21:07.40 | clever | x86: i have no certs and no 'grade 12' and sit at home all day and night |
| 21:07.46 | clever | I am 21.452899 years old. (I'll be 22 in 28wks 4days 7hrs 50mins 11secs.) |
| 21:07.51 | existenz | kaldemar, I see, which distro do you recommend for a slow computer? |
| 21:07.52 | bkruse | been at digium since I was.........maybe I was 15 |
| 21:07.54 | deeperror | i see you talking about HS...arrg...i had a case thrown at me after i left for all kinds of tampering, invasion, and misuse of state funded equipment after I graduated...that was the thanks I received for setting up both 250 station computer labs, website, network, and managed it my junior and senior year for them for free ha what a joke |
| 21:08.00 | bkruse | No, I was 16 |
| 21:08.00 | x86 | clever: get your generic education diploma at least |
| 21:08.03 | logicwrath | i agree with kaldemar, my first * was from source and then *NOW and I really dont like *NOW |
| 21:08.17 | clever | x86: i was never good at english class, thats where i would have the most problem |
| 21:08.17 | x86 | bkruse: you know Mancini? |
| 21:08.26 | logicwrath | I actually had a better experience with *NOW beta 3-6 |
| 21:08.34 | x86 | clever: are you in .us? |
| 21:08.39 | kaldemar | existenz: i guess about any distro will do, as long as you don't run anything resource consuming on it. |
| 21:08.41 | clever | x86: canada |
| 21:08.44 | x86 | ah |
| 21:08.49 | x86 | dunno how it works up there |
| 21:08.54 | bkruse | x86: I believe so |
| 21:08.59 | x86 | everything's backwards and costs double in canada ;) |
| 21:09.03 | clever | we still haev a GED to get if you didnt finish HS |
| 21:09.09 | bkruse | OH, x86, yes I do |
| 21:09.14 | x86 | bkruse: :) |
| 21:09.25 | existenz | kaldemar, xubuntu? |
| 21:09.43 | x86 | bkruse: I've known Mark Spencer for a while, but just spoke with Mancini a couple of times on the phone within the last month |
| 21:10.02 | bkruse | x86: Oh ya? sales? |
| 21:10.16 | bkruse | GED = good enough diploma |
| 21:10.21 | clever | lol |
| 21:10.38 | clever | i know linux inside and out |
| 21:10.38 | deeperror | really all you need |
| 21:10.42 | kaldemar | existenz: for example, but don't waste your resources running a GUI. |
| 21:10.42 | clever | and c/c++/php |
| 21:10.48 | x86 | bkruse: yep |
| 21:10.49 | clever | but dont have a cert to proove it:P |
| 21:10.52 | existenz | kaldemar, without gui |
| 21:11.01 | x86 | bkruse: generica education diploma is what I always called it |
| 21:11.09 | kaldemar | existenz: GUI on the linux, i mean. |
| 21:11.14 | x86 | bkruse: -a |
| 21:11.25 | bkruse | x86: What certs do you have? Also, if you are ever in HSV, let me know :) |
| 21:12.01 | *** join/#asterisk Bagualas (n=pedro@189.35.217.156) |
| 21:12.22 | Bagualas | is there anyone that uses "vono" from brazil? |
| 21:12.30 | x86 | bkruse: LPIC-1, CompTIA Linux+, Brainbench RDBMS, Brainbench Linux Server Administrator, Brainbench Cisco Routing Fundamentals.... I know I have more, but those are all I can think of off the top of my head |
| 21:12.31 | clever | goes behind the desk to rewire the KVM |
| 21:12.54 | clever | x86: yeah, listing the programing languages ive used is like listing off every word in the dictionary |
| 21:13.47 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 21:13.52 | bkruse | x86: oh oh! I want to get my LPIC's. I have CompTIA stuff (since it last forever), a+, net+, sec+, linux+, CCNA. My old job paid for them, so it was worth just buying a book, little studying, and having the comptia stuff forever |
| 21:14.36 | x86 | linux+ does not last forever anymore |
| 21:14.58 | clever | ive done some linux kernel c |
| 21:15.04 | x86 | and CCNA is not a comptia cert ;) |
| 21:15.13 | clever | made a basic driver that just did x++ out the lpt port |
| 21:15.16 | *** join/#asterisk Linx (n=linx@60-234-162-14.bitstream.orcon.net.nz) |
| 21:15.16 | x86 | cisco certified network associate -- and it lasts for 3 years |
| 21:15.25 | clever | hook up 8 led's and you got the computer counting on a binary display |
| 21:15.45 | clever | x86: i got problems with my cisco switch you could fix:P |
| 21:15.56 | x86 | clever: probably |
| 21:16.02 | clever | the java is fussy |
| 21:16.06 | x86 | clever: what's the problems / how much does it pay? |
| 21:16.11 | clever | it keeps loading CW.jar over and over and over |
| 21:16.15 | clever | and never ends |
| 21:16.19 | x86 | coughs his Coke and spews it out of his nose |
| 21:16.30 | clever | though under a diff java engine, it loads it 200 times and finaly works |
| 21:16.31 | x86 | DONT USE THE GUI TO CONFIGURE CISCO THINGS.... EVAR! |
| 21:16.41 | clever | but i dont know the cli interface:P |
| 21:16.53 | clever | its like when i started linux |
| 21:17.02 | clever | the cli was confusing so i just used the shiny point and click ui:P |
| 21:17.11 | x86 | start with the CLI, then if you so desire, play with the GUI |
| 21:17.32 | x86 | the fix to your issue is to learn the GUI |
| 21:17.49 | clever | i had to use a older/diff java engine to 'fix' it |
| 21:17.54 | clever | then it actualy spewed usefull errors |
| 21:18.02 | clever | no last-mod stamp on the cw.jar so it couldnt be cached |
| 21:18.19 | clever | so every time it was referenced the engine had to pull it from the switch again |
| 21:18.32 | x86 | what model of switch is it? |
| 21:18.35 | clever | and the newer engine couldnt handle that and just froze |
| 21:18.37 | x86 | I'm guessing 2900 series? |
| 21:19.14 | clever | WS-C2912-XL |
| 21:19.33 | bkruse | x86: woah woah, linux+ _does_ last forever (at least when I took it 2 years ago) and I said I have CCNA, that only last for _two_ years (since also, when I took it) |
| 21:19.39 | bkruse | when I go to retake it, it will then last 3 years |
| 21:20.15 | bkruse | ccna was a whole different beast than the comptia tests |
| 21:20.49 | fogo | bkruse: isn't CCNA 3 years? |
| 21:21.03 | bkruse | fogo: it is now, but when I took it (almost 2 years ago) , it was only for 2 years |
| 21:21.11 | bkruse | not to mention they changed how all their cert stuff works now |
| 21:21.21 | clever | x86: System image file is "flash:c2900XL-c3h2s-mz-120.5.2-XU.bin" |
| 21:21.25 | bkruse | including CCNA (putting in a cert below CCNA). CCNA used to be the entry level, but is no longer |
| 21:21.46 | fogo | bkruse: you're referring to just tests, correct? |
| 21:21.53 | bkruse | fogo: Correct |
| 21:21.53 | *** join/#asterisk sapere (n=sapere@209-249-12-72.ip.openhosting.com) |
| 21:21.57 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 21:22.31 | fogo | bkruse: ok.. just wanted to make sure. I got my CCNA just before they were changing the tests/etc - wondering if they'd changed anything else |
| 21:22.36 | *** join/#asterisk eXistenZ (n=Ameen@unaffiliated/existenz) |
| 21:23.03 | *** join/#asterisk iqson716 (n=iqson716@unaffiliated/iqson716) |
| 21:23.06 | bkruse | fogo: You are lucky, because you now have a non-entry level cert (as they made a test below CCNA that is easier, and CCNA is now a little harder) |
| 21:23.09 | sapere | afternoon, everyone. pardon the complete newb question: what is the absolute minimum i need to purchase, in terms of a number, line, or whatever components the phone company owns, to get a working asterisk implementation? |
| 21:23.13 | bkruse | but you still have to renew it |
| 21:24.00 | fogo | bkruse: yeah, I remember seeing the new test stuff - covers a *lot* more than what I had to |
| 21:24.04 | *** part/#asterisk iqson716 (n=iqson716@unaffiliated/iqson716) |
| 21:24.15 | bkruse | Exactly |
| 21:24.24 | clever | x86: kill -conf `pidof x86` |
| 21:24.30 | clever | -cont! |
| 21:24.37 | brodiem | ugh 2 deadlocks in one day on 2 different systems, one caused by app_queue and the other after doing a sip reload from changing codec order =/ |
| 21:25.21 | sapere | I guess my real question is do I need to actually purchase a phone line at all? Or can I get an Asterisk implementation up using VoIP alone? |
| 21:26.37 | [TK]D-Fender | sapere: Yes you can use VoIP alone. |
| 21:27.19 | sapere | [TK]D-Fender: thanks, do you know does that choice affect which kinds of hardware I must purchase? What is the bare minimum I must purchase going that route? |
| 21:27.32 | [TK]D-Fender | sapere: To interface with lines you have requires hardware. To obtain service from an ITSP requires an internet connection with sufficien bandwidth. Everything else is pretty much optional |
| 21:27.47 | [TK]D-Fender | sapere: How do you WANT to operate? |
| 21:28.00 | sapere | [TK]D-Fender: I'd prefer to operate without a monthly phone line payment |
| 21:28.34 | [TK]D-Fender | sapere: That tends to work via "by-the-minute" providers |
| 21:28.50 | [TK]D-Fender | sapere: So that handles HOW you get to the PSTN. Now what do you want to talk INTO? |
| 21:29.15 | sapere | [TK]D-Fender: ITSP providers are the entities that lease me phone connectivity over the network? |
| 21:31.08 | sapere | [TK]D-Fender: let me rephrase -- what is the entity that I must pay a monthly fee to in order to get a number that can be reached by anyone with a telephone? |
| 21:31.50 | [TK]D-Fender | ~itsp |
| 21:31.51 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:32.05 | seanbright | what can i use to post-process .WAV files (capitalization intended) into .mp3? |
| 21:32.11 | seanbright | lame? |
| 21:32.12 | [TK]D-Fender | sapere: You can get fixed rate "lines", or pay by the minute |
| 21:32.13 | seanbright | sox? |
| 21:32.23 | [TK]D-Fender | sapere: Everything depends on the kind of pacage you choose |
| 21:32.23 | sapere | ah, perfect ... thanks! |
| 21:32.31 | heedly | seanbright: lame |
| 21:33.08 | seanbright | heedly: danke. |
| 21:33.09 | [TK]D-Fender | sapere: So that handles HOW you get to the PSTN. Now what do you want to talk INTO? <--- |
| 21:33.15 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 21:33.30 | Wayhigh | is there a way to shorten up the post-dial delay on tdm400's? |
| 21:33.39 | Wayhigh | right now it seems to be like 6-10 seconds |
| 21:33.41 | sapere | [TK]D-Fender: this will be for a relative's small business, so it would be some kind of phone-like device |
| 21:34.06 | [TK]D-Fender | sapere: How many phones? What do you have now? |
| 21:34.34 | [TK]D-Fender | Wayhigh: disable call-progress |
| 21:34.43 | sapere | [TK]D-Fender: they will need 20 phone extensions, so I guess 20 phones also. I don't know exactly what they have now, assume nothing I suppose |
| 21:36.01 | Wayhigh | fender: for real? |
| 21:36.07 | [TK]D-Fender | Wayhigh: You asked... |
| 21:36.23 | sapere | [TK]D-Fender: but they will only need one incoming phone number, but I suppose you already assumed that |
| 21:36.24 | eXistenZ | zaptel comes within asterisk's rar? |
| 21:36.25 | [TK]D-Fender | sapere: I would suggest Polycom IP 320/330's. www.telephonydepot.com |
| 21:36.32 | eXistenZ | *tar |
| 21:36.40 | x86 | eXistenZ: no |
| 21:36.42 | sapere | [TK]D-Fender: awesome, thanks again! |
| 21:36.45 | [TK]D-Fender | eXistenZ: No, it is a separate package |
| 21:37.05 | x86 | sapere: I would recommend the Polycom IP 501's over the 320/330... |
| 21:37.06 | eXistenZ | ah, it is in the downloads section |
| 21:37.11 | [TK]D-Fender | sapere: Run those off a D-Link DES1228P PoE switch and you're set |
| 21:37.14 | x86 | sapere: or even the 430 model |
| 21:37.41 | x86 | sapere: and HP swtiches with PoE are cheap, and come with a lifetime warranty |
| 21:39.20 | x86 | sapere: http://www.costcentral.com/proddetail/HP_ProCurve_Switch_2610_24_PWR/J9087AABA/Q37784/froogle/ |
| 21:39.33 | Wayhigh | fender: unfortunately, that's already disabled |
| 21:39.35 | sapere | x86: thanks |
| 21:40.04 | sapere | x86: wait, each phone requires a physical cat5 connection to a switch? |
| 21:40.05 | [TK]D-Fender | x86: And only 3 times the cost of mine! :) |
| 21:40.18 | x86 | d-link, 3com, netgear, and linksys should never be used in a commercial environment |
| 21:40.22 | [TK]D-Fender | sapere: they are NETWORK phones you know... |
| 21:40.38 | [TK]D-Fender | x86: So says the guy burned consistently with channel-banks :p |
| 21:40.42 | x86 | they are the epitome of the consumer market ;) |
| 21:40.44 | sapere | [TK]D-Fender: it makes perfect sense, I'm just so new to it all |
| 21:40.55 | Wayhigh | bbl |
| 21:40.56 | x86 | [TK]D-Fender: not burned at all, just once by Rhino |
| 21:41.12 | x86 | [TK]D-Fender: now I've got nice CAC Adit 600's and everything's perfectly fine |
| 21:41.55 | x86 | [TK]D-Fender: and with a lifetime warranty and free software upgrades for life, I think $1000 is a good price for a 24-port PoE switch with 2 10/100/1000 and 2 SFP ports |
| 21:42.08 | MooingLemur | I'd take a Linksys SPA-942 over a 7960 any day. |
| 21:42.34 | x86 | MooingLemur: I'd take Polycom over either of those ;) |
| 21:42.59 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:43.35 | x86 | sapere: you can get the 2810-24-PWR, which is 24port 10/100/1000, with (2) 10/100/1000 copper and (2) 10/100/1000/10000 SFP uplink ports ;) |
| 21:43.46 | x86 | sapere: if you have the need for 10gig uplinks, that is |
| 21:44.08 | sapere | Having not actually seen the current cost of traditional phone service for my uncle's business, how long would guys guess it would take before a $1000 switch + $1700 in phones + monthly ITSP service would pay for itself? |
| 21:44.49 | [TK]D-Fender | sapere: You do the math :) |
| 21:45.15 | sapere | well I actually have no idea what his current costs are, but I imagine others here might've gone thru something similar and might know |
| 21:45.25 | *** part/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 21:45.51 | x86 | sapere: I wouldn't use an ITSP out to the world in a production environment, unless the ITSP is running you a dedicated circuit |
| 21:46.05 | x86 | sapere: don't route calls over the internet and expect reliable communication |
| 21:46.38 | sapere | x86: so you'd recommend a regular phone line or something via Comcast or whatever? |
| 21:46.44 | x86 | sapere: the main cost savings with VoIP are between buildings / different geographical campuses for large companies with dedicated circuits between each office |
| 21:46.54 | x86 | comcast runs phone lines now? |
| 21:47.05 | x86 | last I knew, they were an ITSP :p |
| 21:47.06 | sapere | x86: whoops, at&t even |
| 21:47.11 | mchou | lol |
| 21:47.22 | x86 | yeah you can use at&t, although they'd be a bit pricey |
| 21:47.23 | mchou | comcast has voip over cable and fiber :) |
| 21:47.35 | x86 | how many concurrent calls out and in do you expect at any one time? |
| 21:47.40 | ManxPower | comcast, ATT, sprint, they are all the same Walmar-Utami |
| 21:47.54 | sapere | x86: if Comcast has business-class VoIP service, would that use ITSP service? |
| 21:47.58 | heedly | I have no problem using an ITSP with a colo'ed box. |
| 21:48.01 | x86 | mchou: and niether of those are "regular phone lines" heh |
| 21:48.09 | heedly | not dedicated but very low ping from server to ITSP. |
| 21:48.28 | x86 | sapere: well that's essentially a dedicated circuit, well close enough anyway |
| 21:48.32 | x86 | right |
| 21:48.45 | sapere | x86: so that's not good enough? |
| 21:48.49 | x86 | but they wont hand it off to you in a way that's easily interfaceable with Asterisk |
| 21:49.11 | x86 | how many concurrent outbound and inbound channels will you use at a time in peak conditions? |
| 21:49.28 | oilinki | what do you guys think that the roundtrip time should be for voip? |
| 21:49.40 | clever | x86: you went silent? |
| 21:49.40 | clever | x86: System image file is "flash:c2900XL-c3h2s-mz-120.5.2-XU.bin" |
| 21:49.42 | sapere | x86: are you implying (possibly justifiably so, i have no idea) that comcast business-class voip will inevitably have lag issues due to being serviced via ITSP? |
| 21:49.47 | mchou | oilinki: it better be less than 400ms :) |
| 21:50.03 | x86 | clever: wow, I guessed correctly on the 2900 :P |
| 21:50.12 | clever | yep |
| 21:50.12 | x86 | clever: you should throw that old thing in the trash ;) |
| 21:50.26 | clever | x86: im using all 12 ports on it... |
| 21:50.35 | x86 | sapere: no, I'm implying that they wont allow you to interface it with Asterisk |
| 21:50.37 | oilinki | mchou: I have read that it should be less than 200ms.. but then again I'm dealing with 400-500ms delays. |
| 21:50.37 | clever | without that i wouldnt be able to connect things up |
| 21:50.59 | x86 | clever: it doesn't even have the backplane capable of using 12 ports to full capacity ;) |
| 21:51.02 | lowtek | A 400-500ms delay simply means that there will be 1/2 second lag in audio at the very least. |
| 21:51.12 | oilinki | mchou: actually that's just the ping time.. |
| 21:51.18 | mchou | oilinki: if it's over 400ms you might as well be using old school satellite :) |
| 21:51.24 | sapere | x86: but previously you said you wouldn't use an ITSP out to the world in a production environment, and here is an example of a commercial solution that offers exactly that, correct? |
| 21:51.31 | clever | x86: allmost all of my massive trafic is aimed at a single port, so im limited to 100mbit for that trafic(total) |
| 21:51.51 | clever | so being able to do 100mbit on every port isnt a need |
| 21:51.53 | oilinki | lowtek: that's true. but still the conversations are quite ok. |
| 21:51.54 | lowtek | There's nothing wrong with an ITSP as long as you understand what you're dealing with. |
| 21:51.55 | x86 | sapere: I said EXCEPT for a "dedicated" circuit -- i.e. your ISP is your ITSP |
| 21:52.13 | sapere | x86: ah, so it's dedicated in that case |
| 21:52.16 | x86 | sapere: I said VoIP across the Internet is very bad and should not be used for production purposes |
| 21:52.16 | sapere | gotcha |
| 21:52.20 | ManxPower | x86: the ITSP does not have to have anything to do with your ISP. |
| 21:52.23 | x86 | sapere: close enough ;) |
| 21:52.30 | sapere | x86: thanks, now I understand |
| 21:52.34 | oilinki | mchou: true :) it's about 100ms to geo-orbit and the same back. |
| 21:52.46 | x86 | ManxPower: no, but it cuts the Internet out of the equation if your ISP is also your ITSP |
| 21:52.56 | lowtek | I disagree with x86 here, I think internet based voice has it's place in production. We have over 2600 happy customers that enjoy it. |
| 21:53.05 | ManxPower | x86: "point-to-point T-1" |
| 21:53.13 | x86 | ManxPower: right :) |
| 21:53.30 | ManxPower | From you to your ITSP, your data never even touches another network. |
| 21:53.44 | x86 | lowtek: you try telling that to a telemarketing company with 200 salespeople and (2) T1's for internet that already have 60% utilization |
| 21:53.50 | sapere | yeah, but that's not a cost-effective solution for a small business |
| 21:53.51 | x86 | lowtek: see how reliable it is at that point ;) |
| 21:54.18 | mchou | screw telemarketers |
| 21:54.19 | ManxPower | sapere: there is no cost effective PSTN access for a small business unless you want to sacrifice something. |
| 21:54.27 | lowtek | x86: I said it has it's place in production, that clearly isn't it's place. It's place is the 5-25 phone broadband deployment. |
| 21:54.31 | x86 | sapere: you never answered my question... how many concurrent calls? a few POTS lines should be all you need for a small operation |
| 21:54.48 | x86 | lowtek: 25 phones on broadband... good luck with that :) |
| 21:54.50 | sapere | ManxPower: so you're better off with a traditional phone service |
| 21:54.58 | x86 | ManxPower: thank you! |
| 21:55.00 | lowtek | x86: Works quiet well with G.729a |
| 21:55.23 | x86 | lowtek: i would never recommend VoIP to the outside world to a business, and I used to own an ITSP :) |
| 21:55.23 | ManxPower | sapere: there's no cost effective solution using traditional phone service unless you are willing to sacrifice something. |
| 21:55.35 | sapere | x86: ah, sorry ... concurrent calls? I'd assume about 5 ... not a high-volume thing or anything |
| 21:56.08 | ManxPower | If you're cheap, go with an ITSP, just know it won't be as reliable as standard POTS service. |
| 21:56.16 | lowtek | x86: I see your point but still hubmly disagree, we have a lot of happy customers saving a small fortune and adding a lot of flexibility. |
| 21:56.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 21:56.26 | *** join/#asterisk MazeMiami (n=MazeMiam@c-98-242-196-48.hsd1.fl.comcast.net) |
| 21:56.42 | MazeMiami | Hello world |
| 21:57.00 | lowtek | x86: We're a hosted pbx provider so in essense we're an ITSP although we terminate to PRI, our customers still have to terminate with us. |
| 21:57.03 | MazeMiami | does asterisk support incoming text messages ? |
| 21:57.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:57.23 | ManxPower | My BF gets about %90 of the time he can make a call when he picks up the phone. Going from $60/month phone bill to $7/month phone bill more than makes up for the times it does not work. |
| 21:57.46 | sapere | I assumed Asterisk was primarily aimed at small businesses that wanted to reduce costs and have more features than they could normally afford, but it sounds more like it's something for large enterprises to avoid proprietary phone systems. Is that correct? If that's so, that dramatically alters my options. |
| 21:57.52 | ManxPower | If that is acceptable to you then by all means use VoIPoInternet |
| 21:58.16 | ManxPower | sapere: Asterisk is WEAKEST in the small business area. |
| 21:58.38 | [hC] | ManxPower: wow, talk about the stupidity that comes with assumption. you're female? (or gay?) |
| 21:58.41 | sapere | ManxPower: thanks, that appears to be the same conclusion I was reaching |
| 21:58.43 | ManxPower | It has it's place, it's just it's weakest area. |
| 21:59.00 | ManxPower | [hC]: queer as a two dollar bill. |
| 21:59.31 | [hC] | ManxPower: haha. :) |
| 21:59.32 | oilinki | ManxPower: why asterisk is weakest on small business? |
| 21:59.33 | MazeMiami | can Asterisk suppot reciving text messages ? |
| 21:59.35 | MooingLemur | MazeMiami: US ISDN networks don't support SMS unfortunately. Not sure if it can receive them via other messages. |
| 21:59.50 | ManxPower | sapere: How much use Asterisk is to you is really directly proportional to how good your data network is. |
| 21:59.50 | MooingLemur | via other means, I mean |
| 21:59.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:00.38 | MazeMiami | any one know of any services i can buy a number that will beable to recive text messages? |
| 22:00.47 | sapere | ManxPower: we'd be looking for something that is reliable for the ~15-20 people he employs, easy to maintain once installed, and cheaper than regular business-class phone service |
| 22:00.50 | ManxPower | MooingLemur: e-mail/SMS gateways exist for all providers as well as Web/SMS gateways |
| 22:01.06 | ManxPower | MazeMiami: that feature is not available in USA/Canada |
| 22:01.32 | MazeMiami | i know asterisk allows to send |
| 22:01.37 | MazeMiami | not sure why you canr recive |
| 22:01.40 | ManxPower | sapere: I can tell you what works. PRI + Zaptel T-1 card + Asterisk + LAN + Polycom Phones |
| 22:02.07 | sapere | ManxPower: thanks, I suppose that's about all I could ask. I appreciate it |
| 22:02.17 | ManxPower | MazeMiami: no, asterisk can neither send nor receive SMS in the USA. Only places that implement an SMSC, and no carrier in the USA operates an SMSC. |
| 22:02.29 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 22:02.41 | ManxPower | all this is in the sms docs. /path/to/src/asterisk/doc |
| 22:02.43 | MazeMiami | my asterisk was able to send text |
| 22:02.48 | MazeMiami | using eyebeam |
| 22:02.59 | ManxPower | MazeMiami: not across the PSTN it didn't. |
| 22:03.15 | MazeMiami | not sure what that means i have freepbx |
| 22:03.25 | ManxPower | Eyemean of course supports the SIP messages, but not ISDN messages. |
| 22:03.37 | ManxPower | MazeMiami: then perhaps you should be asking on the #FreePBX channel. |
| 22:04.12 | ManxPower | But in any case, I don't help GUI users. Best of luck with your project! |
| 22:04.23 | *** part/#asterisk MazeMiami (n=MazeMiam@c-98-242-196-48.hsd1.fl.comcast.net) |
| 22:04.35 | oilinki | MazeMiami: openSER had some SMSGW features listed. That might help you out |
| 22:04.46 | *** join/#asterisk MazeMiami (n=MazeMiam@c-98-242-196-48.hsd1.fl.comcast.net) |
| 22:04.58 | mchou | wtf?? "I dont help GUI users?" |
| 22:05.16 | MazeMiami | free pbx is a GUI |
| 22:05.18 | MazeMiami | pretty lame |
| 22:05.26 | ManxPower | mchou: Correct. I am not paid to be here, I volunteer here. If I don't want to help a GUI user, exactly what do you suppose you are going to do? |
| 22:05.27 | mchou | ManxPower: the dude's question regarding SMS had nothing to do with GUI |
| 22:05.31 | bkruse | mchou: freepbx is a gui, #freepbx, or if you are using the Digium gui, #asterisk-gui |
| 22:05.46 | ManxPower | mchou: no it does not, but I won't encourage GUI users in anyway. |
| 22:05.51 | ManxPower | ~zeeek |
| 22:05.51 | jbot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 22:05.59 | tzanger | hahahaha |
| 22:06.03 | tzanger | awesome ManxPower |
| 22:06.11 | ManxPower | mchou: this guy doesn't even understand what SMS is. |
| 22:06.16 | bkruse | tzanger: That's an old one, where have you been :P |
| 22:06.29 | mchou | ManxPower: so? That's more the reason to give him help |
| 22:06.33 | tzanger | ManxPower: know of any way to get a DID that will accept SMSes from mobiles and allow me to get at them? |
| 22:06.41 | ManxPower | mchou: I already said that I don't. |
| 22:06.43 | tzanger | bkruse: I haven't seen that one come up |
| 22:06.59 | bkruse | tzanger: wait for someone to say "help mai gui plz" |
| 22:07.02 | bkruse | Nah, ManxPower does what he wants, if he doesn't want to help you, you can ask elsewhere |
| 22:07.03 | mchou | I mean I wont help a FreePBX guy if his question had to go with config, but SMS? Come on. |
| 22:07.17 | bkruse | #freepbx or #asterisk-gui (if it's a GUI question), then there is always google |
| 22:07.24 | tzanger | bkruse: wow I *so* have to get a lolcats gateway for asterisk |
| 22:07.37 | bkruse | tzanger: Oh we have like 10, in stock! |
| 22:07.38 | mchou | bkruse: the guy didnt ask a GUI question is my point |
| 22:07.40 | ManxPower | tzanger: not really. USA/Canada don't have the SMSCs that are open to the public, unlike Europe which does. Technically SMS only happens on GSM networks, but most people generically refer all text messaging as "SMS" |
| 22:07.56 | oilinki | then again. we others can help people in need. |
| 22:08.05 | bkruse | mchou: I realize that, I am just saying that ManxPower, though sometimes does hate on GUI people, helps other users, in his own time, a lot |
| 22:08.05 | ManxPower | mchou: Exactly how to you suggest he fix it when he can't even edit his own config files? |
| 22:08.07 | tzanger | ManxPower: true enough; I was wondering if there were any NA SMSCs... something that interfaced with gsmsendsms or sommat |
| 22:08.19 | MazeMiami | also my vonage can recive a text |
| 22:08.28 | tzanger | the only thing I can think of is a cellular phone that call-forwards to a regular DID, and is attached to a computer running the sms daemon |
| 22:08.34 | MazeMiami | how is that possiable |
| 22:08.48 | mchou | ManxPower: dude, he never mentioned config files. He was just looking for info on SMS |
| 22:08.55 | ManxPower | tzanger: not that I'm aware of. There are companies that act like an SMSC, but they are really just text messaging brokers. The Wiki should have a list of them. |
| 22:09.13 | tzanger | ManxPower: I've only seen it for sending them, never for rx |
| 22:09.21 | ManxPower | mchou: and I already gave him the answer before he told me he is a GUI user. |
| 22:09.27 | bkruse | http://www.voip-info.org/wiki/view/SMS there, done, no more talking about GUI/sms/ ssshhhhhhh |
| 22:09.38 | bkruse | :) |
| 22:09.40 | ManxPower | tzanger: *nod* Not popular in the USA. |
| 22:09.49 | tzanger | bah humbug |
| 22:09.59 | ManxPower | bkruse: there is also the most current SMS docs in /path/to/src/asterisk/doc |
| 22:10.00 | tzanger | if I had a way of getting SMSes to a DID I'd never have to give out my cell # |
| 22:10.01 | bkruse | ManxPower: not popular because no one allows it :/ |
| 22:10.12 | bkruse | ManxPower: Ahh yes, I just saw that |
| 22:10.26 | ManxPower | The Wiki is the LAST place you should look for docs. |
| 22:10.39 | bkruse | Yes, but general information is the first place I look |
| 22:10.43 | ManxPower | try the actual docs included with Asterisk as well as the "show application" stuff in the CLI. |
| 22:10.47 | bkruse | Eg "What is an sms" |
| 22:10.59 | MazeMiami | Short Message Service |
| 22:11.03 | MazeMiami | aka text message |
| 22:11.06 | mchou | ManxPower: I really dont get it. You think it's OK for me (or you) to hate on gays or gui users? |
| 22:11.10 | ManxPower | bkruse: yes, but YOU know when something on the Wiki only applies to specific versions -- the public does not know that. |
| 22:11.17 | lowtek | Aren't they the same thing? |
| 22:11.18 | bkruse | ManxPower: This is true :) |
| 22:11.24 | lowtek | lol, just kidding, I hate neither. |
| 22:11.26 | ManxPower | mchou: I think you have no obligation to help me at all. |
| 22:11.46 | ManxPower | Just as I have no obligation to help you. |
| 22:11.49 | bkruse | mchou: _his_ point is that he helps here for free, so he does not owe you anything, which is true. |
| 22:11.57 | mchou | ManxPower: That's like saying "I wouldnt stop and piss on a gui user (gay guy) if he was on fire" |
| 22:12.24 | bkruse | mchou: That's not true either. If you were around on other cases, ManxPower usually refers them to #freepbx, or over to me in #asterisk-gui |
| 22:12.28 | bkruse | He is my mediator |
| 22:12.32 | bkruse | :) |
| 22:12.36 | ManxPower | no sweetie, it's like saying "I wouldnt stop and piss on a gui user (verizon cell user) if he was on fire." |
| 22:12.53 | lowtek | mchou: I doubt he actually *hates* anyone, he's expressing that his vast knowlege is wasted on GUI users since ultimately a GUI limits options and flexibility. |
| 22:13.05 | ManxPower | ~freepbx |
| 22:13.06 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:13.12 | bkruse | See, good ManxPower :) |
| 22:13.16 | ManxPower | those are pretty much the reason for my policy ^^^^^^^^^^^^^^^^^ |
| 22:13.21 | outtolunc | charges double for popcorn |
| 22:13.25 | bkruse | ManxPower: We talked about this what like, 2-3 months ago already? |
| 22:13.33 | mchou | bkruse: my issue is not the fact that ManxPower volunteers his time. (he can choose whoever he helps out). But there is no need to be militant about it since the guy didnt even ask a GUI question |
| 22:13.56 | sapere | thanks alot for all the help, later all! |
| 22:14.00 | bkruse | mchou: I understand, I see both sides here. |
| 22:14.08 | oilinki | hiding behind hate and ~urls might not be the best way to show _real_ knowledge |
| 22:14.24 | ManxPower | mchou: just put me on /ignore and be done with it! |
| 22:14.29 | mchou | and quite frankly, since ManxPower just revealed he was gay, I'd think he'd understand "discrimination" |
| 22:14.59 | bkruse | mchou: Ok, you can chill now. |
| 22:16.21 | outtolunc | watches as the cost of popcorn plumets |
| 22:16.28 | ManxPower | I get almost as upset when users come in here expecting us to tutor them in Linux. This is simply not the correct place for that sort of thing. |
| 22:17.37 | ManxPower | If the community support for their GUI is so horrible they have to come here for help -- well then maybe they should reconsider the GUI they are using. |
| 22:17.50 | ManxPower | I hear Asterik-GUI has decent community support. |
| 22:17.58 | oilinki | ManxPower: there is a way to help people as well. by just being rude and giving no information is not a good way. |
| 22:18.00 | ManxPower | 8-) |
| 22:18.23 | oilinki | much better for the whole asterisk community would be just to be quiet, rather than rude |
| 22:18.55 | oilinki | let other people to communicate and help the people who are strugling with their experiences with asterisk |
| 22:19.37 | mchou | there's no need for frigging elistism, especially when it's not called for |
| 22:19.44 | lowtek | In other news, Google released a new web browser today, based on webkit (not gecko) |
| 22:19.55 | oilinki | I have been there and I know that's at the start is damn difficult to find the right questions. |
| 22:20.00 | ManxPower | Go to #linux and start asking Redhat questions sometime. |
| 22:20.04 | mchou | lowtek: nice attempt to change the subject |
| 22:21.18 | oilinki | basically this rudeness what can be found here just keeps people of from the asterisk. |
| 22:21.51 | jaytee | well, there's always barber college |
| 22:21.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:23.02 | bkruse | ManxPower: lol, I recommend not doing that. Or go into #c and ask what parameter to pass for snprintf(var, size?, "text"); do I put in 100? 200? |
| 22:23.39 | bkruse | Honestly though, #asterisk-gui is the place for those gui questions, and ManxPower redirects them accordingly, and I answer _every_ message on the mailing list, because I am not always in the room |
| 22:23.50 | ManxPower | bkruse: I don't know from personal experience, but I imagine doing that would make me look cute and cuddly |
| 22:24.05 | bkruse | ManxPower: umm....exactly! |
| 22:24.48 | mchou | bkruse: I have no issues if ManxPower chooses to redirect. But a statement like "I don't help GUI users" is just plain rude |
| 22:24.55 | lesouvage | Or ask a question about getting a x100p clone up and running (years ago but still remember the response) |
| 22:24.59 | bkruse | I understand |
| 22:25.03 | heedly | lol |
| 22:25.06 | heedly | ManxPower: don't be rude! |
| 22:25.15 | heedly | or no more IRC for you?? |
| 22:25.16 | bkruse | heh, lesouvage, I recommend not doing that |
| 22:25.43 | bkruse | Well, the thing is, I have never seen _any_ of your names answering questions, but I know ManxPower very well from him actually helping people, so....I side with him |
| 22:26.23 | ManxPower | bkruse: It seems to be the classic newbie in a community problems. |
| 22:27.22 | angryuser | stop that useless chit chat and answer real questions, here is mine http://pastebin.com/md102c it is cdr mysql related, i have some problems with dst field, all detail in pastebin, thx for you time |
| 22:27.29 | mchou | bkruse: I've only been hanging out here for 1 month. And if this channel were logged you'd see plenty of good answers from me |
| 22:27.48 | bkruse | mchou: this channel is logged, and I have never seen you |
| 22:28.02 | mchou | bkruse: then grep the log |
| 22:28.04 | lesouvage | mchou: this channel is logged |
| 22:28.10 | ManxPower | angryuser: the destination for those calls is "s". Somewhere in your dialplan you are sending calls to "s" extension. Maybe a Macro? |
| 22:28.23 | bkruse | angryuser: I do not see an s |
| 22:28.26 | mchou | bkruse: why would I bs about crap like this? |
| 22:28.28 | bkruse | ahh, ManxPower already beat me, bah |
| 22:28.42 | bkruse | mchou: I have no idea, I am not calling you a liar, I am just saying ManxPower has some _years_ on you |
| 22:29.06 | angryuser | ManxPower: read it closer, you got it at second example |
| 22:29.19 | mchou | bkruse: so his years gives him the right to be abusive and discriminatory? |
| 22:29.49 | ManxPower | angryuser: Um, your dialplan is one of the most fucked up dialplans I have ever seen in the 6 or 7 years I've been using Asterisk. All your extensions are "s" and you have some really bizarre prioritys |
| 22:30.01 | mchou | bkruse: I think you have misplaced priorities if you answer affirmatively to that |
| 22:30.22 | ManxPower | bkruse: LOOK at his dialplan. |
| 22:30.26 | bkruse | ManxPower: lol |
| 22:30.30 | bkruse | ManxPower: I am |
| 22:30.45 | bkruse | mchou: I do not at all, my answer is simply that he does a lot of helping, and most of the time he will just redirect people |
| 22:30.46 | angryuser | ManxPower: yep i agree to that statement, i had in under it's not mine and i dont care much |
| 22:31.09 | bkruse | there are 259 _other_ people in here that could have helped after he said he wouldn't |
| 22:31.22 | bkruse | Trying asking about pidgin in #x11-devel |
| 22:31.30 | ManxPower | angryuser: I'll tell you how I work around Macros and "s". The first line of my macro is exten => s,1,Goto(${MACRO_EXTEN},1) and then have a wildcard pattern match of something like _X.,1,Whatever |
| 22:32.03 | ManxPower | angryuser: but until you design a working dialplan, nothing is going to work right, |
| 22:32.34 | angryuser | ManxPower: it is working stable for half a year |
| 22:32.57 | ManxPower | angryuser: I don't see how. |
| 22:32.58 | bkruse | ManxPower: You and him have different definitions for working :) |
| 22:33.18 | ManxPower | bkruse: obviously. 8-| |
| 22:33.44 | ManxPower | angryuser: at least you know how to fix it. |
| 22:33.53 | ManxPower | well, the "s" in the CDRs, at least. |
| 22:34.01 | angryuser | ManxPower: yes thx |
| 22:34.27 | ManxPower | angryuser: Major CDR changes went into 1.6, maybe later 1.4 releases that changes the way CDRs are handled to make them more correct. |
| 22:36.05 | ManxPower | bkruse: put in a good word for me with Jared Smith. I'm aka Eric |
| 22:36.18 | bkruse | ManxPower: Ok, you applying or something? |
| 22:36.21 | angryuser | ManxPower: 1.4 has all features i need |
| 22:36.33 | ManxPower | bkruse: yup |
| 22:36.34 | bkruse | I will be talking to jsmith soon actually |
| 22:37.05 | bkruse | ManxPower: done |
| 22:37.14 | ManxPower | bkruse: thanks! |
| 22:37.30 | bkruse | ManxPower: npnp, I would do it again in a heartbeat |
| 22:37.34 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net) |
| 22:37.41 | ManxPower | bkruse: try to sugar coat the bad parts 8-) |
| 22:37.53 | bkruse | hehe |
| 22:40.49 | angryuser | ManxPower: btw i dont see why s should be the problem in my macro and 'bizzarre' priorytis is my business ;) |
| 22:43.53 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:44.28 | bkruse | ManxPower: I think I will be seeing you soon.... :] |
| 22:48.14 | *** join/#asterisk waverly360 (n=waverly@c-98-211-5-255.hsd1.tn.comcast.net) |
| 22:55.49 | Blackvel | COOOOL. my two port patton isdn bri gateway is finally working with Asterisk + internal PBX :) |
| 22:55.58 | Blackvel | jipppiie |
| 22:58.02 | Blackvel | g' night :) |
| 23:04.42 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 23:04.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:09.06 | *** join/#asterisk Firass-z0r (n=asadf@ead224-222.housing.wwu.edu) |
| 23:09.10 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 23:15.32 | *** join/#asterisk legis (i=estar@unaffiliated/legis) |
| 23:18.41 | legis | "Call from '' to extension '3199701' rejected because extension not found", that DID is pointed to my asterisk IP, in the incoming context I have exten => _X.,n,Dial(SIP/xxx_peer/${EXTEN}) |
| 23:18.51 | legis | any ideas why I get that error? |
| 23:19.16 | outtolunc | because '' is less than ' ' |
| 23:19.23 | outtolunc | giggles |
| 23:19.39 | WimpMan | and where is priority 1? |
| 23:19.43 | legis | oops, that was the outgoing context, the incoming context is exten => _X.,1,Dial(SIP/1002) |
| 23:20.15 | WimpMan | Still _X won't match nothing. Use s. |
| 23:20.21 | *** join/#asterisk existenz (n=Ameen@unaffiliated/existenz) |
| 23:20.31 | existenz | is it possible to edit the sip.conf from the cli? |
| 23:20.41 | legis | existenz: sure |
| 23:20.59 | existenz | legis, how? |
| 23:21.16 | legis | existenz: "vi /etc/asterisk/sip.conf" |
| 23:21.36 | legis | WimpMan: doesn't _X. matches everything? |
| 23:22.24 | WimpMan | legis: Everything at least one character long. |
| 23:23.45 | TJNII | legis: What technology? |
| 23:24.11 | legis | TJNII: sip |
| 23:24.49 | joat | existenz: you'll need to know a few vi commands.... |
| 23:25.02 | joat | to use legis's command |
| 23:27.07 | knarfly | existenz, nano is easier than vi |
| 23:27.10 | legis | WimpMan: didn't work with 's' either :/ |
| 23:27.17 | existenz | knarfly, I used both :) |
| 23:27.25 | legis | I'm starting to think is my ITSP |
| 23:29.04 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:29.43 | legis | the weird thing is, when i dial the Did from the PSTN i see the error message in the asterisk box |
| 23:30.36 | TJNII | legis: Are you sure it is looking in the right context? |
| 23:31.06 | TJNII | And coming in the right account? |
| 23:34.33 | legis | TJNII: yeah, is a pretty simple config, just a test config |
| 23:35.47 | TJNII | Simple can still be wrong. Based on that error it is highly unlikely it is the ITSP. |
| 23:36.53 | mchou | holy shit. The federal "Do not call" list does actually cut down on telemarketing calls |
| 23:37.05 | mchou | unbelievable |
| 23:37.15 | TJNII | No wai! |
| 23:37.20 | legis | TJNII: I'll paste the config |
| 23:38.16 | TJNII | legis: Pastebin your sip.conf, dialpla, and cli output |
| 23:38.29 | mchou | from the average 1/hr of telemarketer calls I'm now down to 1/day |
| 23:38.46 | mchou | I still think that's 1 too many though |
| 23:39.22 | legis | TJNII: k |
| 23:47.18 | legis | TJNII: http://rafb.net/p/BQoRw075.html |
| 23:47.59 | legis | I tried with 's' also in [une-incoming] but same error |
| 23:50.35 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:51.01 | WimpMan | legis: Forgot to reload perhaps? s in une-incoming would be the right place. |
| 23:51.29 | WimpMan | You could also use _. instead of _X., but you don't want to. |
| 23:53.33 | *** join/#asterisk _ShrikE (n=_ShrikE@64.128.34.118) |
| 23:56.08 | legis | WimpMan: yes, I reloaded, tried with '_X.' , 's', and '_.' same error |
| 23:58.56 | ManxPower | The type=friend is NOT being matched when the call comes in, therefore it will use the context definwed in [general] |
| 23:59.44 | ManxPower | Maybe your ITSP is not sending as user "une_peer" |
| 23:59.57 | ManxPower | because whatever is inside the [] is the incoming username. |
