00:00.28 | bkw_ | just passed 211th street |
00:00.30 | file | cool I can hit the Send key on this phone and get a dialtone |
00:00.35 | terrapen | bkw: i'm sorry :) |
00:00.39 | bkw_ | haha |
00:00.45 | terrapen | <--- texas boy |
00:00.46 | bkw_ | terrapen, almost out of glenpool now |
00:00.49 | terrapen | are you on the road? |
00:00.53 | bkw_ | CLEAR |
00:00.53 | bkw_ | yep |
00:00.54 | terrapen | what are you using for commo? |
00:01.00 | file | he's using GPRS baby! |
00:01.02 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
00:01.02 | bkw_ | Sapulpa Exit 1 Mile |
00:01.03 | terrapen | ah |
00:01.07 | terrapen | SprintPCS? |
00:01.11 | bkw_ | no |
00:01.13 | bkw_ | T-mobile |
00:01.15 | mikegrb | bkw_: I'm from okc |
00:01.16 | terrapen | ahh |
00:01.20 | mikegrb | bkw_: :D |
00:01.21 | terrapen | GPRS is fun |
00:01.24 | bkw_ | mikegrb, kewl |
00:01.25 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
00:01.25 | mikegrb | bkw_: but in florida now |
00:01.27 | terrapen | please tell me you aren't driving, bkw |
00:01.28 | bkw_ | McAlester, Here |
00:01.30 | terrapen | i mean |
00:01.34 | terrapen | *You* aren't driving |
00:01.35 | bkw_ | no i'm not driving |
00:01.37 | bkw_ | are you MAD |
00:01.39 | terrapen | ok, good :) |
00:01.41 | mikegrb | bkw_: oh really? I never knew that |
00:01.43 | mikegrb | spiff |
00:01.43 | terrapen | well, i've done it before |
00:01.44 | bkw_ | i'm crazy |
00:01.45 | bkw_ | but gee |
00:01.48 | bkw_ | give me a break |
00:01.52 | ozJames79 | hi all anyone having problems with broadvoice and incoming calls ? i fixed up the outgoing but it appears no incoming now |
00:01.53 | bkw_ | :P |
00:01.57 | mikegrb | well, are you from there, or did you live there for 15 to 20? |
00:02.05 | bkw_ | some guys minnow bucket flew out of his boat |
00:02.09 | bkw_ | an almost hit us |
00:02.10 | sivana | ozJames79: what kind of issues, errors? |
00:02.17 | terrapen | you have to dip if you are going to drive through OK |
00:02.18 | bkw_ | terrapen, ewww |
00:02.20 | bkw_ | thats nasty |
00:02.27 | ozJames79 | its not even hitting my CLI .. which is weird |
00:02.28 | terrapen | where are you headed? |
00:02.32 | bkw_ | I was born here.. I hate it all |
00:02.34 | bkw_ | Tulsa |
00:02.38 | terrapen | from? |
00:02.39 | ManxPower | I wonder if I could do BOTH VON Europe AND AstriCon Europe? |
00:02.41 | ozJames79 | when someone calls my BV number it rings busy |
00:02.55 | bkw_ | welcome to BV |
00:02.57 | terrapen | i'm a hick, admittedly |
00:03.05 | mikegrb | ozJames79: then call BV, #asterisk is not #bv-support |
00:03.17 | ozJames79 | i was not asking for support |
00:03.21 | terrapen | bkw, whats in SJC |
00:03.23 | sivana | ozJames79: they have instructions on their site |
00:03.26 | ozJames79 | i was asking if anyone else with bv was having the problem |
00:03.29 | NirS | hey bkw, I heard congrats are in order |
00:03.31 | ozJames79 | i am on the phone to them right now |
00:03.46 | bkw_ | NirS for? |
00:03.47 | sivana | ozJames79: I don't think so, I havent' checked laately |
00:04.09 | sivana | ozJames79: they are my last hope route |
00:04.20 | ozJames79 | thanks sivana :) |
00:04.24 | sivana | :) |
00:04.30 | Goshen | what are these called on the end of a dial string, I am looking for documentation on them... ,60,Ttm) |
00:04.39 | file | parameters...options... |
00:04.48 | terrapen | i've never been around Tulsa |
00:04.52 | file | show application dial to see what options are available |
00:04.53 | terrapen | only flew through on Southwest |
00:04.54 | file | Goshen: that was for you |
00:05.03 | file | domo arrigato Mr. Roboto |
00:05.18 | Goshen | at the command prompt? |
00:05.24 | file | at the asterisk CLI |
00:05.30 | terrapen | take me back to Tulsa, I'm too young to marry, take me back to Tulsa, I'm too young to wed thee..... |
00:05.52 | Goshen | wow sweet! thanks :) didn't know it did that |
00:05.52 | file | show application <application name> Will give you... documentation |
00:05.55 | tzanger | dammit |
00:06.01 | tzanger | how do I get ipv6 DOWN so I can remove the module |
00:06.05 | tzanger | use count is like 96 |
00:06.27 | Goshen | ~docs |
00:06.28 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
00:07.48 | terrapen | we always have a great big time, never do look sour / travel 'round the country, day and by the hour |
00:08.06 | terrapen | (it's Bob Wills and the Texas Playboys) |
00:08.34 | Goshen | jbot: docs is Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org show application <application name> in the Asterisk command line. |
00:08.35 | jbot | ...but docs is already something else... |
00:08.42 | Goshen | :) |
00:09.07 | ckruetze_ | I wish VON wouldn't be so expensive :( |
00:09.09 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
00:10.05 | bkw_ | woops missed the exit |
00:10.15 | file | haha |
00:10.28 | bkw_ | i'm not the one driving |
00:10.31 | bkw_ | remember |
00:11.04 | terrapen | On a clear day in Tulsa, you can see the back of your head. |
00:11.19 | bkw_ | oh really? |
00:11.20 | bkw_ | since when? |
00:11.29 | terrapen | :P |
00:11.43 | terrapen | actually, its supposed to be Lubbock, i just changed the joke |
00:11.54 | terrapen | Lubbock is quite flat, you see... |
00:13.00 | sivana | heh |
00:13.09 | bkw_ | ok we are almost there |
00:13.10 | bkw_ | bbl |
00:14.14 | sivana | what device are they using to be able to chat on IRC mobile? |
00:14.33 | terrapen | you can use a GPRS-enabled phone and a USB cable |
00:14.37 | terrapen | phone acts like a modem |
00:14.39 | sivana | I'm missing out on all the gadgets |
00:14.42 | terrapen | laptop sees it as a modem |
00:14.43 | sivana | GPRS? |
00:15.05 | sivana | I see |
00:15.14 | terrapen | its networking that works over most modern GSM cellular networks |
00:15.26 | tzanger | sivana: I can use my CDMA1xRTT |
00:15.28 | tzanger | but it's slow |
00:15.28 | terrapen | its somewhat pricy |
00:15.29 | tzanger | and expensive |
00:15.40 | terrapen | pricey even |
00:16.40 | sivana | I see |
00:17.05 | sivana | maybe someday Canada will upgrade their tin cans to 3G or whatever is newest |
00:17.07 | tzanger | keep it under the seat of the car with the CDMA1xRTT |
00:17.09 | j0 | i'm having problems connecting to the iaxtel.com gateway.. it will register for a few moments when asterisk first loads.. but otherwise i keep getting "DEBUG[1152]: Raw Hangup 69.73.19.178:4569, src=2, dst=136" |
00:17.24 | sivana | heh |
00:17.32 | j0 | i'm behind nat, i have all data from their ip forwarded to *, as well as the appropriate iax ports |
00:17.48 | sivana | j0: do yourself a favor, forget IAXTel |
00:17.59 | sivana | it's lagged and uselss |
00:18.02 | j0 | sivana: so its not me? :) |
00:18.09 | j0 | i just need something to test with right now |
00:18.11 | ManxPower | tzanger: You can use 1xRTT without a plan and it will use your plan mins. |
00:18.12 | sivana | spend $10 and get a Nufone acct |
00:18.21 | tzanger | ManxPower: not with bell canada |
00:18.23 | tzanger | it's expensive |
00:18.38 | ManxPower | tzanger: You can with Verizon. |
00:18.48 | ManxPower | Has anyone here been to VON? If so, please /msg me. |
00:20.26 | sivana | can we even get Verizon in Canada? |
00:21.06 | sivana | maybe with their unlimited roaming plan? :) |
00:21.35 | ManxPower | sivana: Yes. The "north american unlimited plan" but I think you have to have a USA billing address. |
00:21.43 | sivana | I see |
00:22.36 | sivana | in a bottle?... brilliant |
00:22.56 | tzanger | hahaha |
00:23.53 | sivana | hehe |
00:26.30 | terrapen | so drive across the border and get a UPS Store box |
00:26.46 | jesster | in sip.conf I want to setup SER for incoming and outgoing calls - Should I create a [ser_in] type=user and a [ser_out] type=peer or just a [ser] type=friend ? |
00:26.59 | BrianR___ | tzanger: I'm lucky enough to live in an area with CDMA EVDO.. It's like 2 megabit... |
00:27.19 | BrianR___ | Unlimited access for like $80/mo. |
00:27.47 | tzanger | nice |
00:27.48 | tzanger | very nice |
00:27.54 | tzanger | where's the area that has that |
00:27.56 | BrianR___ | Fails over to 1xRTT in areas without EVDO equipment. |
00:28.08 | BrianR___ | tzanger: A number of big metros in new england... |
00:28.15 | tzanger | ahh |
00:28.24 | BrianR___ | at least 50 miles around boston in my area. |
00:28.41 | terrapen | how is the latency on that |
00:28.43 | BrianR___ | http://www.verizonwireless.com/b2c/mobileoptions/broadband/index.jsp |
00:28.59 | BrianR___ | terrapen: Usually <100ms. |
00:29.10 | BrianR___ | Always much better than GPRS, etc. though. |
00:29.14 | terrapen | not bad |
00:29.44 | terrapen | good lord |
00:29.50 | BrianR___ | latency varies quite a bit based on the number of retransmits required, how busy the cell site is, etc. But it's as good or better than CDMA circuit switched data and 1xRTT. |
00:29.57 | terrapen | they serve Baltimore but they don't serve San Antonio |
00:30.05 | terrapen | 10th largest city in the US, or thereabouts |
00:30.09 | terrapen | #1 most forgotten |
00:30.14 | DaLion | quick question |
00:30.14 | DaLion | i need a perl cgi to talk to asterisk to make it dial.. is that command exec dial ? |
00:30.20 | BrianR___ | terrapen: Heh. It'll get there eventually. |
00:31.04 | DaLion | or i use manager ? |
00:31.28 | DaLion | cant really seem to find examples |
00:31.49 | DaLion | so i guess i got 3 choices.. AGI, manager or dump a file in queue ... rirhgt ? |
00:32.06 | DaLion | or is AGI only * to cgi |
00:33.06 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
00:34.54 | *** join/#asterisk trelane (~trelane@lan.trelane.net) |
00:34.58 | terrapen | <PROTECTED> |
00:35.05 | terrapen | brian, it seems that it IS available in my area |
00:35.11 | terrapen | if you put in my zip code... |
00:35.27 | DaLion | terra ? |
00:35.30 | terrapen | <PROTECTED> |
00:35.34 | terrapen | this is really confusing |
00:35.40 | DaLion | im trying to talk to * from perl.. got AGI:asteirsk etc |
00:35.48 | DaLion | anyidea ? |
00:35.51 | DaLion | i need to dial out |
00:36.29 | terrapen | yeah, we dont have it brian |
00:36.32 | terrapen | just austin |
00:37.51 | trelane | anyone having issues with broadvoice? |
00:38.09 | DaLion | bah i use teliax no issues |
00:38.31 | DaLion | u can make special corp arangements |
00:38.50 | Bruns | !@#$% |
00:38.53 | Bruns | excuse my cursing |
00:38.59 | Bruns | broadvoice changed their setup |
00:39.02 | DaLion | Bruns yeah tell me about it |
00:39.05 | DaLion | lol |
00:39.05 | Bruns | they are sending me details |
00:39.58 | jesster | anyone run into auth problems having phones register with SER and then have ser forward calls to Asterisk? Asterisk is giving me auth errors |
00:41.21 | Bruns | apparently, theres an unusually high amount of asterisk users calling them |
00:41.55 | *** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net) |
00:42.38 | Damin_Mobile | Yo yo yo! |
00:43.11 | trelane | Bruns, umm what? |
00:44.45 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
00:45.12 | shmaltz | andrew? |
00:45.43 | ManxPower | Has anyone here been to VON? If so, please /msg me. |
00:46.08 | Bruns | http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup |
00:46.27 | Bruns | they changed their setup slightly, ever so slightly |
00:48.15 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
00:53.44 | Damin_Mobile | Manx: Been to? Or going to? |
00:57.56 | Bruns | note to self, must kill broadvoice |
00:58.12 | *** join/#asterisk expressfone1 (~expressfo@62-15-97-163.inversas.jazztel.es) |
01:00.17 | tzanger | Nugget needs to use nufone like all the cool people do |
01:00.36 | Nugget | I use them too, but I still have this accursed DID from voicepulse that I hope will actually work some day. |
01:00.42 | tzanger | ahh |
01:05.02 | shmaltz | anybody here using Cisco? |
01:05.09 | shmaltz | I mean cisco phones |
01:05.12 | Nugget | sure |
01:06.36 | Damin_Mobile | I use cisco |
01:06.55 | Damin_Mobile | I love cisco |
01:07.24 | r0d3nt|m | cisco or death. |
01:09.19 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
01:14.58 | JohnnyC | can I forward a call to an extension based on the number called ? |
01:15.25 | JohnnyC | I have a BRI with 10 numbers |
01:15.27 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
01:15.36 | JohnnyC | number X would go to extension 20 directly |
01:15.47 | JohnnyC | or awnsered by music , anything |
01:16.02 | *** join/#asterisk GrimStone (~Pkunkage@203.187.245.49) |
01:16.31 | shmaltz | JohnnyC, why not? |
01:16.40 | JohnnyC | hehe |
01:16.45 | JohnnyC | how ! :) |
01:17.01 | bjohnson_ | quick question. What advantage is there to running * on a wrt54g compared to just using sip phones? |
01:17.01 | shmaltz | do you know how to configure Asterisk? |
01:17.05 | JohnnyC | how do you name this ? for me to look in docs and mailing ? |
01:17.13 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
01:17.22 | JohnnyC | yes sure |
01:17.29 | tzanger | damn all you and your fancy schmanzy domains |
01:17.34 | blitzrage | anyone use the SMS() app? I'm looking for an example, but I'm not in the UK, so I can't test. |
01:17.46 | shmaltz | tzanger, whats wrong? |
01:17.49 | expressfone1 | hi |
01:17.57 | JohnnyC | DDI ? |
01:18.08 | tzanger | 20:24 -!- blitzrage [~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk] has joined #asterisk |
01:18.10 | blitzrage | tzanger: get back to work. |
01:18.17 | tzanger | blitzrage: eat me :-) |
01:18.21 | blitzrage | tzanger: yah I know... I was looking for someone specifically, but he isn't here. |
01:18.30 | j0 | any thoughts on www.simpletelecom.com .. nufone seems expensive compared to them |
01:18.42 | blitzrage | file: ! |
01:18.47 | file | hi hi |
01:18.52 | blitzrage | j0: I use them, work fine (I don't pay them... but I use them :)) |
01:18.54 | tzanger | j0: expense is in the eye of the beer-payer |
01:19.05 | shmaltz | JohnnyC, can you explaing again what you are trying to do? |
01:19.05 | tzanger | j0: not having ANY issues is worth a lot to me |
01:19.08 | blitzrage | but NuFone's network is supposedly pristine. |
01:19.17 | file | how are you? |
01:19.18 | JohnnyC | Direct number to an Extension |
01:19.31 | tzanger | I won't say pristine, but in the past year and a bit I've used it I have neve rhad issue that was their problem |
01:19.36 | blitzrage | file: pretty good, j00? I'm working on DOCS! |
01:19.44 | file | blitzrage: I had surmised you were |
01:19.49 | file | blitzrage: still on track? |
01:19.53 | blitzrage | file: mostly. |
01:19.54 | j0 | blitzrage: do you just use their free service? |
01:20.02 | file | silly mostly |
01:20.07 | blitzrage | j0: I signed up for a beta a long time ago, and never got cut off :) |
01:20.18 | j0 | thats what i'm using now to test.. its my first attempt |
01:20.22 | *** join/#asterisk Rick_Hunter (~rhunter@adsl-69-209-173-100.dsl.sfldmi.ameritech.net) |
01:20.26 | blitzrage | tzanger: SEE! Already sucked in |
01:20.31 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
01:20.40 | blitzrage | file: chat with me on another channel... I'm out of this window. |
01:20.41 | *** part/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
01:20.55 | tzanger | hahaha |
01:23.22 | shmaltz | JohnnyC, you mean did? |
01:23.25 | shmaltz | DID |
01:23.32 | JohnnyC | DDI or DID ? |
01:23.38 | JohnnyC | Direct Dial in ? |
01:24.34 | GrimStone | anyone have problems with broadvoice outgoing ? |
01:25.22 | GrimStone | even after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls |
01:26.07 | GrimStone | i can see voice data coming in from broadvoice on port 5060 .. but asterisk just seems to ignore it |
01:26.38 | JohnnyC | shmaltz: I want to ring directly an extension or voicemail to diferent numbers |
01:27.30 | GrimStone | this was working great until today , when bvoice did thier "upgrades" |
01:27.56 | JohnnyC | have to go |
01:28.21 | shmaltz | JhonnyC, |
01:28.23 | shmaltz | wait |
01:29.16 | shmaltz | JohnnyC, look up DID on the wiki, all you have to do is find out how many digits your carried sends with the bri, in extensions.conf create extensions that match the digits received from you carriedr as DID. |
01:31.07 | GrimStone | so anyone here use Broadvoice ? |
01:31.32 | GrimStone | cause right nows it badly broken |
01:36.53 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
01:37.31 | GrimStone | why would asterisk just ignore the voice data coming in on port 5060 ? |
01:41.48 | *** join/#asterisk mischko (~Scott@p29-25-150.vcr.centurytel.net) |
01:46.51 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
01:48.15 | *** join/#asterisk trimi` (~Pharrell@62.162.232.118) |
01:53.06 | MikeJ[Jayden] | wow, it's quiet... what, is it saturday night or somthin |
01:55.09 | GrimStone | why would asterisk just ignore the voice data coming in on port 5060 from broadvoice ? |
02:00.18 | jesster | I have SIP phones => SER(register) => Asterisk -PSTN. For some reason my phones will only dial through if a certain UserID is set on the phones, otherwise if the UserID is not the correct one, I get a 407 Proxy Auth Required / 403 Forbidden (Im doing auth with AuthID) anyone have suggestions how to fix? |
02:01.21 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
02:05.20 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
02:05.29 | *** join/#asterisk JohnJar (~JohnJar@85-250-71-159.bb.netvision.net.il) |
02:06.15 | JohnJar | Hi, where can i get more info about Asterisk and about PBX's? thanks |
02:06.50 | mikegrb | http://www.google.com/ |
02:07.15 | *** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net) |
02:07.28 | mikegrb | da-manFL: I disagree. |
02:08.20 | da-manFL | why? |
02:09.51 | mikegrb | I am da-man in florida, this state is not big enough for both of us |
02:12.40 | tuxinator_linux | ~docs |
02:12.41 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
02:13.21 | j0 | what was that * project that let u share your phoneline with other people around the world to make ld calls? |
02:13.27 | j0 | was a sort-of p2p system based on credits |
02:14.01 | cbachman | j0 bellster/fwd-out ? |
02:16.11 | *** part/#asterisk mischko (~Scott@p29-25-150.vcr.centurytel.net) |
02:18.15 | trimi` | <j0> |
02:18.22 | trimi` | <j0> it was something with out |
02:18.24 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
02:18.28 | trimi` | the name i dont remember |
02:18.33 | trimi` | like s****oute |
02:18.40 | trimi` | like s****out |
02:18.54 | trimi` | let me see if i still got it in my favorite links |
02:19.34 | trimi` | yeap i found it |
02:19.35 | trimi` | http://www.fwdout.net/web/ToSignup |
02:19.37 | mikegrb | trimi`: the question has been answered, no need for your pollution |
02:20.06 | trimi` | sorry didnt see it |
02:21.46 | trimi` | is there any bootable version of asterisk with calling card platform installet or which include ASTCC or any simmilar aplication ? |
02:23.00 | mikegrb | no |
02:23.34 | trimi` | :( |
02:23.44 | shepherd | jeeze harsh :) |
02:23.56 | *** join/#asterisk file (~file@mctn1-142166194173.nb.aliant.net) |
02:24.08 | mikegrb | shepherd: it's the answer |
02:24.21 | j0 | cbachman: yeah thanks :) |
02:24.26 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
02:26.59 | *** join/#asterisk PhoneBooth (Phonebooth@208-25-55-247.stk.inreach.net) |
02:28.37 | jesster | I have SIP phones => SER(register) => Asterisk -PSTN. For some reason my phones will only dial through if a certain UserID is set on the phones, otherwise if the UserID is not the correct one, I get a 407 Proxy Auth Required / 403 Forbidden (Im doing auth with AuthID) anyone have suggestions how to fix? |
02:28.39 | Chuji | damn broadvoice. Why do I still use them? |
02:28.54 | Chuji | I should just go to pay per use |
02:29.46 | file | jesster: your question makes no sense, if the authid isn't the correct one... shouldn't it send back the 407 or 403? |
02:30.54 | jesster | file - I cannot find where this perticular UserID is being approved, like on Asterisk side. The AuthID is fine.. |
02:32.15 | jesster | file - ie: PhoneA has UserID 325 and another has UserID joe -- the phone that calls through with joe is approved and the call with 325 is given 407 / 403 |
02:32.35 | file | jesster: okay, is joe a valid username in sip.conf? |
02:33.48 | jesster | file - ah crap. I just realized that i had a matching 325 account .. that was it (my sip.conf file is huge and I overlooked it) |
02:34.05 | jesster | file: ie a local 325 account with different credentials |
02:39.08 | MikeJ[Jayden] | wassup all |
02:40.16 | *** part/#asterisk JohnJar (~JohnJar@85-250-71-159.bb.netvision.net.il) |
02:46.39 | GrimStone | why would asterisk just ignore the voice data coming in on port 5060 from broadvoice ? |
02:47.01 | GrimStone | even after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls |
02:59.04 | *** part/#asterisk trelane (~trelane@lan.trelane.net) |
03:04.10 | Moc | is there some disk quota problem again ? |
03:04.36 | *** join/#asterisk obelisque (~samifruit@Ottawa-HSE-ppp4015539.sympatico.ca) |
03:04.48 | obelisque | hello guys! IAX2 or SIP? |
03:04.59 | Nugget | either's great. |
03:05.10 | Nugget | it's not like you have to choose. |
03:05.24 | obelisque | I heard that IAX2 was better |
03:05.32 | obelisque | because it was consuming less bandwith |
03:05.32 | Nugget | they each have their appeal. |
03:05.41 | Nugget | for a single channel the difference is immeasurable. |
03:05.54 | Moc | iax2 support 2 thing, no NAT problems, and trunking make you consume alot less bandwidth on multiple concurent connections(calls) |
03:06.11 | obelisque | sound quality stills the same? |
03:06.18 | Nugget | sure, that's a matter of the codec. |
03:06.18 | Moc | obelisque, IAX aint a codec |
03:06.22 | obelisque | ok ok |
03:06.27 | Moc | just a transport tunnel |
03:06.28 | obelisque | sorry im a little noob |
03:06.33 | obelisque | yeah I understand |
03:07.02 | obelisque | does X-Lite supports IAX2? |
03:07.05 | Moc | the office I setup the other day could do 10 g726 call, once I activated trunking, I could go up to 20 |
03:07.05 | Nugget | no |
03:07.06 | riksta | no |
03:07.32 | obelisque | any free softphone support it? |
03:07.34 | Moc | actually 23 after I resync the DSL ;) |
03:07.35 | Nugget | yes. |
03:07.59 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
03:08.02 | obelisque | ok... |
03:08.18 | obelisque | and... |
03:08.27 | obelisque | what are they? |
03:08.31 | Nugget | google for "iax softphone" |
03:08.37 | obelisque | roger that! |
03:08.39 | Moc | obelisque, check voip-info |
03:09.09 | obelisque | damn...grandstream phones dont do IAX2 |
03:09.57 | Nugget | the differences between IAX and SIP rarely matter when you're looking at hardware phones. |
03:09.59 | Moc | only vaporphone support IAX2 |
03:11.09 | obelisque | cause grandstream phones are cheap... |
03:11.14 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
03:11.23 | Nugget | seriously. you will never care one way or ther other what protocol your phone is using. |
03:11.31 | Nugget | it will have zero impact on your life. |
03:11.55 | Moc | obelisque, get the sipura phone instead, probably better |
03:11.57 | obelisque | I need IAX2 protocol...for consuming less bandwith! |
03:12.08 | Nugget | IAX2 does not consume less bandwidth. |
03:12.23 | harryvv | its in the codec ;) |
03:12.30 | Nugget | you will not be trunking calls between your server and your desk phone |
03:12.46 | obelisque | why not! |
03:12.53 | Nugget | because you only have one mouth. |
03:13.02 | Moc | hehe |
03:13.08 | Ron-Na | I am looking for a softphone for my HP5555 PDA, has anybody experience with (a good) one? |
03:13.14 | Moc | that a good one Nugget ;) |
03:13.17 | obelisque | we are using sjphone |
03:13.21 | Luhiwu | it's a bug in the human design, why two ears and just one mouth? :) |
03:13.26 | *** join/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
03:13.29 | CoderCR | hello all |
03:13.40 | CoderCR | i am trying to get some help on an issue i am having with a channel bank |
03:14.12 | harryvv | obekusque how has it been as a wifi phone? |
03:14.22 | obelisque | what IAX softphone you would recommand me? |
03:14.22 | harryvv | and what model |
03:14.39 | obelisque | im using sjphone on my ipaq 5835 |
03:15.12 | harryvv | how is the clearity and range to a wifi hotspot? |
03:16.02 | CoderCR | I cannot get my channel bank to dial out. I see ztmonitor send data and then the channel bank seems to hang up the line. But if i have a phone on the line off the hook, then channel bank dials out. |
03:16.14 | tuxinator_linux | My cat just laid a really awful poo |
03:16.22 | tuxinator_linux | need fresh air |
03:16.40 | obelisque | around 5 meters |
03:18.03 | harryvv | obelisque, your kidding right? Usually wifi devices are limited to 100 meters and some times alot more. |
03:18.50 | PatrickDK | heh, I have a good 1/4 mile with my wifi phone |
03:19.20 | harryvv | Patrick thats great is that in downtown with alot of buildings or in the open? |
03:19.36 | harryvv | What model do you use and is it built in wifi? |
03:19.40 | riksta | anyone here been using ADM ? |
03:19.43 | PatrickDK | that is at the house |
03:19.48 | PatrickDK | outside is easy |
03:19.52 | PatrickDK | inside the house is alot harder |
03:20.01 | PatrickDK | so much brick/concrete |
03:20.03 | obelisque | well...voip with my pda sucks |
03:20.19 | obelisque | the microphone gets all the ambient sounds |
03:20.41 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
03:20.43 | obelisque | and its seems to take a lot of cpu... |
03:21.03 | obelisque | i suggest you to get a good pda for doing voip |
03:21.16 | harryvv | ob, who cares about the cpu cycles as long as you can have a decent conversation on it. |
03:21.43 | obelisque | yeah but conversation is not descent |
03:21.44 | harryvv | PatrickDK, what model do you use and does it include the wifi or what card does yours have? |
03:21.46 | Moc | I should try my pocket PC + MY bluetooth headset |
03:21.48 | obelisque | it always cut... |
03:21.57 | PatrickDK | harry, wisip |
03:21.58 | obelisque | blue tooth head set? |
03:22.16 | harryvv | patrick, what are you using to do wivi voip |
03:22.23 | obelisque | now thats interesting... |
03:22.35 | obelisque | how much does a bluetooth head set costs? |
03:22.36 | PatrickDK | harry, wisip |
03:22.40 | harryvv | ob, it means you dont have to worry about a coard. |
03:22.42 | Nugget | do not buy a wisip. |
03:22.45 | harryvv | 300 dollars |
03:22.51 | Nugget | the pulver/zyxel wireless phones SUCK. |
03:22.59 | PatrickDK | heh, wisip works fine for me |
03:23.03 | harryvv | I have seen blue tooth earpieces at 300 dollars. |
03:23.07 | PatrickDK | does everything it's suppost to do |
03:23.10 | Nugget | the hitachi one seems be getting favorible reviews. |
03:23.10 | file | LONG LIVE THE HITACHI WIP-5000! |
03:23.22 | Nugget | but DO NOT buy that pulver piece of shit. |
03:23.24 | harryvv | Okay PatrickDK thats fine, What make and model are you using. |
03:23.36 | PatrickDK | nugget, was the hitachi one being sold over a year ago? |
03:23.42 | Nugget | I don't know. |
03:23.45 | Nugget | I don't think so |
03:23.46 | PatrickDK | harry, hmm? they have make and models? |
03:23.51 | obelisque | bluetooth earpieces for 300$ ? |
03:23.51 | Moc | file, SEND ME ONE ;) |
03:24.00 | Damin | Moc: I just got a Samsung I700 PocketPC w/ an 802.11b Wifi card.. |
03:24.15 | Moc | I got the iPAQ 4150 |
03:24.20 | file | Moc: time little grasshopper |
03:24.27 | Moc | hehe |
03:24.28 | obelisque | hey moc...do you have an blue tooth head set with that? |
03:24.36 | harryvv | Patrick.. third time...What make and model of pda are you using to so wifi voip? you never said which one you are using. |
03:24.41 | file | Moc: I can have it shipped direct, I *think* it comes customs free due to the NAFTA paperwork |
03:24.42 | Nugget | the pulver/zyxel wisip can't hop access points, it's got earsplitting ringtones, it's too slow, it's a royal pain to configure, and has flaky firmware. |
03:24.55 | obelisque | 320$ for wip-5000!! |
03:24.57 | Damin | Moc: We need a good IAX2 VoIP client for the PocketPC. |
03:25.00 | PatrickDK | harry, I never said I was using pda :) I was I was using a WISIP PHONE |
03:25.07 | Damin | Moc: Like FireFly PPC version.. :) |
03:25.08 | PatrickDK | wisip phone has built in wifi |
03:25.12 | harryvv | okay thats fine. which one |
03:25.18 | Nugget | the WISIP phone is which one. |
03:25.26 | obelisque | you guys are rich |
03:25.32 | Moc | yea hehe |
03:25.45 | Nugget | the WISIP is the one you don't want to buy. |
03:25.51 | Nugget | unless you want to buy mine. :) |
03:25.59 | Nugget | because lord knows I don't use it. |
03:26.00 | Moc | obelisque, we just dont have girlfriends.. |
03:26.10 | obelisque | LOL |
03:26.20 | harryvv | Patrick, where have you used the Pulver wisip? |
03:26.34 | PatrickDK | where? hmm, at HOME |
03:26.35 | Nugget | harryvv seems determined not to take my advice. :) |
03:26.36 | Moc | let me tell ya when I get one, it bye bye telephoneS!! :'( |
03:27.00 | harryvv | nugget, yea I have heard some bad things about them. I saw its design...to simple. |
03:27.15 | obelisque | hey guys...I need something NOT EXPENSIVE wireless for VOIP (not a PDA)...any suggestion? |
03:27.15 | PatrickDK | I setup my own wireless mesh, for in the house, and have a good 12db omni with 500mw power for outdoor access |
03:27.16 | Nugget | the WISIP phone is quite possibly the worst piece of hardware I've ever owned. |
03:27.19 | Nugget | it's *terrible* |
03:27.45 | PatrickDK | obelisque, use what my wife uses, cordless phone + sipura2000 |
03:27.49 | drumkilla | Nugget: I had my hands on a demo version at one point ... for some reason, they accidently left out the SIP config menu |
03:27.52 | Moc | Nugget worst than the barbietone ? |
03:28.00 | drumkilla | so you couldn't change any of its provisioning ... |
03:28.03 | harryvv | Nugget, possibly its just a lemon ? What is a good sip phone or pda combination that works well? |
03:28.09 | obelisque | sipura2000? |
03:28.18 | obelisque | sounds nice... |
03:28.19 | Nugget | no. the problems are the firmware. and the underpowered hardware. |
03:28.20 | Damin | Fucking Verizon.... |
03:28.24 | obelisque | ill ebay that |
03:28.25 | Damin | "We never stop working for you.." |
03:28.28 | Nugget | I have no reason to believe it isn't behaving exactly as they intend it to. |
03:28.31 | Damin | What a bunch of bullshit.. |
03:28.40 | Damin | They just stopped working for me.. My connection dropped.. |
03:28.44 | *** join/#asterisk erwinism (~pogz@210.213.143.73) |
03:28.45 | file | drumkilla: VON! |
03:28.46 | harryvv | the pulver phone is made in china ? |
03:28.47 | harryvv | :) |
03:29.00 | Nugget | I suggest the hitachi phone, but not from personal experience. Several others in the channel have the hitachi and are happy with it. |
03:29.13 | obelisque | Wow!! sipura rules |
03:29.22 | file | I *love* my Hitachi phone |
03:29.23 | Moc | Nugget, I got one on the way.... Well I hope I do ;) |
03:29.23 | file | it's sexy |
03:29.38 | Moc | and you got the damn extended battery bas.. ;) |
03:29.39 | drumkilla | for the sake of testing! |
03:29.40 | erwinism | hello... i am planning to put my own asterisk server in my office. what phones should i use to connect to the server? |
03:29.43 | harryvv | btw, shaw and rogers cable is going to or alredy has started pushing there voip service. I wonder how thats going to affect smaller voip services. |
03:29.52 | file | Moc: I think it comes standard where I get it from |
03:30.02 | Moc | that extremely cool.. |
03:30.08 | Damin | erwinism: Any phone that you like. |
03:30.11 | harryvv | nugget, that is the 300 dollar phone? Ive seen it on voipsupply |
03:30.18 | Moc | I'll have to get a wifi bridge at the office |
03:30.23 | obelisque | Hey! where can I get cheap sipura-2000 |
03:30.28 | Damin | erwinism: Althought, you might want to stay away from rotary dial phones.. ;) |
03:30.29 | file | I should just take orders |
03:30.38 | harryvv | ob, on ebay. But be leary of the seller. |
03:30.48 | obelisque | uh ok |
03:31.01 | obelisque | is 70$ too much& |
03:31.04 | Damin | Hey.. does anyone know if VON will have public WiFi access points? |
03:31.10 | file | Damin: I was wondering that too |
03:31.14 | harryvv | ob, Ive seen them for under 100 |
03:31.19 | brc__ | wouldn't count on it |
03:31.33 | drumkilla | Damin: at the last VON, the trade show floor was so saturated with wireless that we coudln't get it working with stuff right next to each other |
03:31.33 | erwinism | Damin do you know a diagram to use asterisk? |
03:31.40 | harryvv | Damin would not suprise me if thay do. |
03:31.48 | file | drumkilla: ohhhhh I wonder how my wifi phone will handle it |
03:32.05 | Damin | erwinism: No. |
03:32.07 | obelisque | ebay is great.... |
03:32.09 | harryvv | drumkilla :) not enough bandwith in the area? |
03:32.12 | obelisque | can we sell girls on ebay& |
03:32.36 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
03:32.37 | file | I'm excited about tomorrow! |
03:32.44 | harryvv | file, lucky you |
03:32.46 | erwinism | Damin lets say i already setup my asterisk server at my office. how can i connect all the telephone lines from all department? |
03:32.46 | mitcheloc | file: you going to von? |
03:32.47 | Moc | lol, dont forget to charge it ;) |
03:32.51 | file | yessssssss I'm going! |
03:32.58 | Moc | my damn collegue got to go there .. |
03:33.03 | mitcheloc | file: nice, i was going to go, but i have finals the week after this |
03:33.04 | file | Moc: 'da phone is charging right now |
03:33.08 | Moc | nice hehe |
03:33.10 | mitcheloc | it's so close, come down and say hi =) |
03:33.14 | drumkilla | mitcheloc: I'm missing a week of school :( |
03:33.38 | Damin | erwinism: My first suggestion would be to read the documentation. |
03:33.51 | erwinism | Damin ok thank you |
03:33.56 | file | drumkilla is VERY silly |
03:33.57 | mitcheloc | well i'll extend an invitation to any asterisk people, if you want to drive down to diamond bar (1hr south of LA), I can put you up for a night or two! (PART @ MY HOUSE) =) |
03:34.07 | Damin | You know.. I'm getting an incredible amount of battery life on this Samsung PDA! |
03:34.26 | erwinism | Damin how does it lasts? |
03:34.27 | drumkilla | mitcheloc: haha ... I think we're going to San Fran on Friday |
03:34.30 | harryvv | damin, how long when its on? |
03:34.50 | Damin | It's lasted all day.. |
03:35.06 | erwinism | Damin without playing on it |
03:35.09 | obelisque | Am I invited? |
03:35.18 | *** part/#asterisk CoderCR (~creyna@adsl-67-112-135-29.dsl.sndg02.pacbell.net) |
03:35.22 | erwinism | ok got to read docs |
03:35.34 | Damin | No.. Using it the entire day.. |
03:35.37 | mitcheloc | yea, anyone who knows what asterisk is is invited heh |
03:35.51 | Damin | Hell.. I just got it yesterday, so hell YEAH I am going to use it all day! :) |
03:36.02 | file | ugh I'm hungry |
03:36.08 | drumkilla | I got a PDA a couple months ago ... and it doesn't turn on anymore :( |
03:36.40 | drumkilla | the last thing I did was type "apm -s" |
03:36.40 | Damin | I got a Double Cheese, Double Pepperoni Pizza delivered... |
03:36.40 | erwinism | drumkilla needs a truck battery i think heheh |
03:37.24 | mitcheloc | hey, can anyone answer this for me, i neede a second number (via broadvoice) and they say the way i can tell the difference is via distinctive rings.... |
03:37.34 | obelisque | ciao guys |
03:37.37 | mitcheloc | now can asterisk take out the header in the packet initiating the connection? |
03:37.38 | harryvv | drumkilla, I dont think its ever been discussed but has anyone though of putting a wifi hotspot in a dense residential area and selling wifi ata settop boxes to residents in the area? |
03:37.43 | mitcheloc | i'd like to ring a different phone when that number comes in |
03:37.44 | Damin | Hmmm... |
03:38.17 | mitcheloc | harryvv: the range on a wifi box would make the effort impossible |
03:38.21 | Damin | I just realized that the music for the movie "Backdraft" is also used as the music for the Food Channel series "Iron Chef" |
03:39.23 | shmaltz | does slackware have an AMD 64 arch port? |
03:39.26 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfi1r.dialup.mindspring.com) |
03:39.45 | Ron-Na | obelisque: how is sjphone on the ipaq? I tried Xten, but it makes my ipaq a phone - cannot use for anything else, ... that is nonsense, if I want that I buy a WiFi phone!!! |
03:39.45 | cbachman | Damin, they've discussed that before on the iron chef newsgroup |
03:40.24 | mitcheloc | anyone on that distinctive ring packet? |
03:40.45 | obelisque | sjphone on my ipaq 3850 is unusable |
03:41.00 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
03:41.59 | harryvv | mitch, range is a function of db or wattage out. But with some properly orinated yagis it may work. There has been some discussion on popular science to place a heliostat in the ionospere with wifi repeater placed on it and blanketing a area of the city. My guess is the pm who is putting this together may have the initial wifi power out waived to a higher rating since it will be away from the general population. The barrier thay will |
03:41.59 | harryvv | have to overcome is to fine a way to keep it afloat during night hours. |
03:42.04 | obelisque | its very slow and when I get the stream, discussion is always interrupted, lets say each 1second I loose 500ms of voice... |
03:42.46 | GrimStone | anyone have problems with outbound calls on broadvoice ? |
03:43.03 | mitcheloc | obelisque: are you setting up 600 phones? |
03:43.28 | GrimStone | even after setting the userid , secret, authid etc. like the email on asterisk-users says , asterisk doesn't seem to be able to "hook up" outgoing calls . it just ignores the voice packets on port 5060 |
03:43.36 | obelisque | mmm |
03:43.39 | obelisque | not yet... |
03:43.46 | obelisque | why? |
03:44.04 | harryvv | I figure that I would get some feed back on this. |
03:44.26 | GrimStone | ..... no one here uses broadvoice ? |
03:44.34 | harryvv | no |
03:44.51 | mitcheloc | just wondering what you were talking about earlier? |
03:44.53 | jsolares | i currently use voipjet and nufone |
03:45.01 | MikeJ[Jayden] | I have an account w/ broadvoice, but I rarely use it |
03:45.01 | MikeJ[Jayden] | sorry |
03:45.15 | jsolares | mitcheloc: why not a second account? that would definetely be easier |
03:45.27 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.173) |
03:46.04 | mitcheloc | jsolares: heh cause i only want to pay $2 a month for this...it's actually for an access system so i can get into my house =) |
03:46.06 | jsolares | atleast beacuse i have no ide how to use distinctive rings yet |
03:46.15 | jsolares | idea* |
03:46.18 | mitcheloc | i'm thinking theres a way to read the sip packet properly |
03:46.27 | obelisque | yeah... |
03:46.32 | mitcheloc | just wondering weather the distinctive packet is on the first request |
03:46.37 | MikeJ[Jayden] | alert-info |
03:46.52 | obelisque | Its a project...I have to talk about this to my boss first... Cause the optical fiber will go in soon |
03:46.56 | MikeJ[Jayden] | =distinctive rings... it will be on the wiki |
03:46.59 | riksta | would anyone be interested in helping me in the development of ADM? |
03:47.06 | mitcheloc | mike: looking it up |
03:47.45 | mitcheloc | adm = ? |
03:47.53 | riksta | uh http://adm.hamnett.org |
03:48.50 | brc__ | what is adm |
03:48.54 | mitcheloc | MikeJ[Jayden]: the distinctive ring i'm talking about is slightly different, i want to catch it on the incoming, not the outgoing |
03:48.56 | brc__ | ah |
03:49.18 | drumkilla | riksta: where are you a student? |
03:49.30 | riksta | Manchester University, UK |
03:49.48 | drumkilla | cool |
03:50.14 | MikeJ[Jayden] | sane thing on sip, just getting instead of setting |
03:50.24 | riksta | am looking for some help with the project |
03:51.04 | drumkilla | riksta: have you tried the -dev list? |
03:51.17 | MikeJ[Jayden] | riska- are you writing adm? |
03:51.24 | riksta | yes mikegrb |
03:51.37 | riksta | drumkilla: i posted a while ago, didn't get much of a response, was a bit disappointing |
03:51.47 | riksta | yes MikeJ[Jayden] |
03:51.49 | drumkilla | I totally understand. |
03:52.09 | riksta | didn't want to post again too soon, might piss people off |
03:52.11 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
03:52.31 | riksta | actually, i posted to -users |
03:52.43 | MikeJ[Jayden] | especially that flemming guy.. I hear he needs anger management therapy he is so bad |
03:53.20 | riksta | hehe |
03:55.04 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
03:57.36 | harryvv | mikej, who is this flemming guy |
03:57.48 | MikeJ[Jayden] | kpflemming |
03:57.54 | harryvv | on this channel? |
03:57.57 | MikeJ[Jayden] | you don't read the dev list |
03:58.01 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
03:58.02 | brc__ | heh |
03:58.06 | brc__ | harryvv, what about him? |
03:58.07 | JunK-Y | lo guys |
03:58.09 | MikeJ[Jayden] | yeah |
03:58.11 | harryvv | no i dont. But I would like to meet him in person. |
03:58.20 | brc__ | are you in phx? |
03:58.21 | MikeJ[Jayden] | go to von |
03:58.24 | drumkilla | he'll be at VON :) |
03:58.28 | brc__ | yeah |
03:58.43 | harryvv | I deal with people like that all the time. All the time since the millitary. |
03:58.44 | harryvv | :) |
03:58.59 | harryvv | I will miss von I am up here in vancouver. |
03:59.13 | harryvv | But may give Toronto a shot in april |
03:59.30 | MikeJ[Jayden] | you are closer to san jose than toronto |
03:59.52 | shmaltz | Andrew? |
04:00.01 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
04:00.14 | harryvv | Yea that is probebly true. But then, I have not seen eastern canada :) |
04:01.47 | shmaltz | ~weather ketl |
04:02.04 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
04:03.59 | MikeJ[Jayden] | ~weather ksjc |
04:04.36 | MikeJ[Jayden] | yeah.. all I have to say is I have 6" of snow on the ground and it sucks here... |
04:04.54 | drumkilla | I rarely see any snow |
04:04.58 | drumkilla | :p |
04:05.52 | MikeJ[Jayden] | yeah, forget the south...you guys never have to deal w/ real weather :) |
04:05.52 | shmaltz | MikeJ where you located? |
04:05.54 | drumkilla | I get hurricanes |
04:05.55 | shmaltz | ~location |
04:05.56 | jbot | ACTION is omnipresent |
04:06.02 | shmaltz | ~sex |
04:06.03 | jbot | I'm gay |
04:06.25 | shmaltz | ~pregnant |
04:06.26 | jbot | Yes, shmaltz, and it's your child. |
04:06.26 | MikeJ[Jayden] | hehe |
04:06.27 | MikeJ[Jayden] | detroit |
04:06.34 | shmaltz | ~lol |
04:06.35 | jbot | lol is, like, stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
04:07.14 | shmaltz | ~sex |
04:07.15 | jbot | I'm female |
04:07.19 | shmaltz | ~play |
04:07.21 | jbot | and now, for something completely different |
04:07.30 | shmaltz | ~chat |
04:07.48 | shmaltz | ~wtf |
04:08.09 | drumkilla | shmaltz: you can /msg jbot, too, you know ... :p |
04:08.39 | MikeJ[Jayden] | hey, let him be.. they are having a very nice conversation :) |
04:08.45 | shmaltz | drumkilla, thanks, I know, but if the channel is quiet why not give some ppl a laugh, thanks again |
04:09.02 | shmaltz | ask jbot about me |
04:09.04 | drumkilla | just playin' |
04:09.22 | shmaltz | ~shmaltz |
04:09.22 | jbot | you are, like, annoying the channel by playing with jbot |
04:09.39 | file | drumkilla: staying up all night is bad, mmmk? |
04:09.50 | shmaltz | ~sleep |
04:09.51 | jbot | hmm... sleep is overrated, and a poor substitute for caffeine |
04:09.56 | shmaltz | ~wife |
04:09.56 | jbot | hmm... wife is the Wide Interface File Engine |
04:10.06 | shmaltz | bye gtg |
04:10.08 | shmaltz | ~bye |
04:10.10 | jbot | l8tr |
04:10.14 | MikeJ[Jayden] | ~wtf shmaltz |
04:10.36 | brc__ | ~data |
04:10.37 | jbot | Don't Ask To Ask. Just ASK |
04:10.43 | MikeJ[Jayden] | me either :) |
04:10.45 | shmaltz | ~quit |
04:10.46 | jbot | ACTION what bananas eat |
04:10.54 | shmaltz | oh i meant /quit |
04:11.54 | brc__ | muhaha HAHAHAH! |
04:12.09 | MikeJ[Jayden] | EVIL |
04:12.47 | MikeJ[Jayden] | drunkilla, what year are you in school? |
04:12.53 | drumkilla | Junior |
04:13.08 | drumkilla | undergraduate |
04:13.23 | MikeJ[Jayden] | I miss that... |
04:13.32 | MikeJ[Jayden] | I need to go back to school |
04:13.34 | MikeJ[Jayden] | again |
04:13.43 | drumkilla | it's tough ... I have a lot on my shoulders right now |
04:13.56 | MikeJ[Jayden] | why do you keep it there? |
04:13.57 | drumkilla | I really don't have much time to work on Asterisk while I'm in school full-time :( |
04:14.03 | drumkilla | it makes me sad |
04:14.12 | MikeJ[Jayden] | y |
04:14.42 | MikeJ[Jayden] | just write a grant proposal to pay you for 6 months and then have some real fun |
04:15.02 | drumkilla | not a bad idea! |
04:15.11 | drumkilla | but I want to graduate asap |
04:15.21 | drumkilla | so I can get to working on Asterisk full-time :) |
04:15.44 | harryvv | drum, how long you been coding in c |
04:15.52 | drumkilla | not very long at all |
04:16.03 | MikeJ[Jayden] | I would love too.. need to get my other work done first at the moment |
04:16.31 | MikeJ[Jayden] | sigh.. need to just get enough clients and quit my job is what I need to do |
04:17.58 | harryvv | sigh, how long you been at it |
04:18.25 | MikeJ[Jayden] | huh? |
04:18.42 | harryvv | I would say stay with it at least a year and if the income exceeds your existing one. |
04:18.57 | harryvv | Thats from a common biz experiance. |
04:19.48 | drumkilla | harryvv: would you believe me if I said a year ago, I didn't know any C? |
04:20.20 | MikeJ[Jayden] | y |
04:20.53 | MikeJ[Jayden] | I have some stuff lined up at the moment.. unfortunately I had an office move and a dozen t1 hotcuts this weekend |
04:21.24 | MikeJ[Jayden] | then, after I get the lines up, I have wiring guys going behind me and unhooking things...sigh |
04:21.37 | harryvv | Thats good. But the problem is your still in collage. What customers really want is confidence in the product and the service. Graduate and I am sure your base will increase. I will not to this day deply a asterisk system untill I am absolutly sure it will not crash or at least give me some warnings it may crash. Basicly the asterisk system really needs to be hardened. |
04:21.49 | MikeJ[Jayden] | me? |
04:21.55 | MikeJ[Jayden] | please! |
04:22.03 | drumkilla | harryvv: well, I don't mean consulting |
04:22.04 | MikeJ[Jayden] | I am maried with kid |
04:22.14 | harryvv | drum, you mean dev right? |
04:22.25 | erwinism | hello, just wanna ask.. channel bank is a hardware? |
04:22.27 | drumkilla | yeah |
04:22.33 | harryvv | drum thats cool. |
04:22.34 | MikeJ[Jayden] | yes |
04:22.51 | harryvv | just hope india does not beat you :) |
04:22.55 | drumkilla | ha |
04:23.08 | harryvv | my supervisor at M$ was east indian. |
04:23.09 | drumkilla | well, if I weren't in school right now, I'd be in Huntsville working for Digium |
04:23.23 | harryvv | ick alabama? |
04:23.27 | harryvv | :) |
04:23.30 | drumkilla | m hm |
04:23.37 | harryvv | dont thay allow remote employment? |
04:23.44 | MikeJ[Jayden] | M3718 |
04:23.52 | drumkilla | it's a time issue, really |
04:25.00 | erwinism | if i use asterisk, how much cost would i save from buying any chap pabx hardware on the market? |
04:25.04 | harryvv | I had a friend who moved from Atlanta and gave up everything to move to Bellingham. Everyone there protested and he had a big income drop as a result but he is alot happier. Bellingham has the cleanest air in the country. |
04:25.26 | harryvv | er, depends what you do with it; |
04:25.27 | MikeJ[Jayden] | drumlilla, you wanna post a response to that one.. cuz mine is gunna be, "do you want fries with that" and that probably isn't that helpful |
04:25.40 | Moc | erwinism, it depend on how you want to implement it |
04:25.53 | MikeJ[Jayden] | erwinism? huh? |
04:26.09 | drumkilla | I've got to go to bed ... |
04:26.10 | Moc | you could make it extremely cheap if you got PC and can use softphone with 15$ headset |
04:26.17 | drumkilla | I have an early flight in the morning |
04:26.46 | drumkilla | g'night everyone *waves* |
04:26.48 | erwinism | Moc, im because i want to have a PBX on my office. i want to know the advantages of asterisk and pbx hardware. |
04:26.53 | MikeJ[Jayden] | niht |
04:26.58 | erwinism | Moc, ..because i want to have a PBX on my office. i want to know the advantages of asterisk and pbx hardware. |
04:27.12 | MikeJ[Jayden] | asterisk is a pbx |
04:27.36 | Moc | erwinism, well * can do it |
04:27.45 | Moc | and it will save you alot on liscencing / upgrade |
04:27.48 | MikeJ[Jayden] | and probably cheaper |
04:27.52 | MikeJ[Jayden] | much cheaper |
04:27.56 | Moc | yea |
04:28.14 | MikeJ[Jayden] | for example, you can pay 500 ish for a pri card for asterisk |
04:28.26 | erwinism | ok, i will present that on the board. |
04:28.32 | erwinism | thanks moc and mike |
04:28.44 | MikeJ[Jayden] | or 3-10 times that for one on a propriatary phone systemm |
04:28.45 | Moc | I just installed * the other day to someone who bought a Avaya PBX, for 8k$, but he didnt had the little liscence card, so PBX is useless, it resold it on ebay and got * With Polycom phone .. |
04:29.20 | MikeJ[Jayden] | hey moc, what phones do you like... |
04:29.32 | Moc | erwinism, most important thing to do is, make a list of the feature you want from your PBX |
04:29.44 | MikeJ[Jayden] | for office enviornment, high functionality need |
04:29.45 | Moc | then it will help you check the price of all the PBX on the market |
04:29.47 | *** join/#asterisk Landrocker (~landrocke@port-222-152-54-115.fastadsl.net.nz) |
04:29.53 | Moc | MikeJ[Jayden], Polycom |
04:30.04 | Moc | Polycom IP 500 or better the IP 600 |
04:30.05 | MikeJ[Jayden] | ip500? |
04:30.10 | MikeJ[Jayden] | k |
04:30.26 | erwinism | E100P is 600$ |
04:31.02 | Moc | erwinism, your in the UK ? |
04:31.12 | MikeJ[Jayden] | need to start specing phones for a job and I have only had experience with the cisco phones, and while they are nice, I have not been overwhelmed by them and I think it is time for somthing new |
04:31.47 | mitcheloc | anyone need a t100p? i've got one, make me an offer =) |
04:32.05 | erwinism | moc im in philippines |
04:32.13 | mitcheloc | i need to make my carpayment on my clk320, so be generous! |
04:32.41 | Moc | MikeJ[Jayden], samething for me, they are nice (except of a couple of flaws), but Polycom is just so much flexible .. |
04:32.50 | mitcheloc | actually jk, i wish i had one of those cars...they are nice |
04:32.56 | MikeJ[Jayden] | are they backlit? |
04:33.02 | harryvv | ahh the PI :) |
04:33.04 | Moc | MikeJ[Jayden], no |
04:33.14 | mitcheloc | mm the cisco backlit, that would be awesome |
04:33.23 | MikeJ[Jayden] | maybe I will order one to play with |
04:33.57 | Moc | MikeJ[Jayden], 1 thing you need to understand, get the DEFAULT config, and only change the minimum, then once it work, you can play with all the options ;) |
04:34.13 | MikeJ[Jayden] | k |
04:34.14 | Moc | MikeJ[Jayden], check at the admin manual : http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf |
04:34.27 | Moc | you can change nearly EVERYTHING |
04:35.03 | MikeJ[Jayden] | and change is good :) |
04:35.11 | MikeJ[Jayden] | spread the word |
04:35.19 | MikeJ[Jayden] | you going out west moc/ |
04:35.20 | MikeJ[Jayden] | ? |
04:35.35 | MikeJ[Jayden] | low battery... |
04:35.37 | Moc | why should I go west ? |
04:35.38 | MikeJ[Jayden] | time to sleep |
04:35.42 | MikeJ[Jayden] | von? |
04:35.56 | Moc | ha nope, I nearly go, but was too late to ask for tickets |
04:36.14 | MikeJ[Jayden] | ok.. come to cluecon in august.. |
04:36.22 | Landrocker | anyone know if it's possible to use a keypad button instead off hook-flash to use call parking, etc? (the '#' key for instance) |
04:36.23 | MikeJ[Jayden] | battery dying |
04:36.26 | MikeJ[Jayden] | gotta go.. |
04:36.27 | *** join/#asterisk threeo (~threeofiv@adsl-146-114-25.mia.bellsouth.net) |
04:36.29 | MikeJ[Jayden] | goodnight |
04:37.57 | *** part/#asterisk Bruns (bruns@pool-141-153-151-58.nwrk.east.verizon.net) |
04:37.58 | hardwire | yay |
04:38.04 | Grooby | ?? |
04:38.06 | hardwire | 48 voice channels over 802.11a backbone |
04:38.12 | hardwire | gsm |
04:38.15 | hardwire | iax trunking |
04:38.18 | Grooby | nice |
04:38.31 | hardwire | I did super extension transfering |
04:38.32 | hardwire | heh |
04:38.41 | Grooby | and i have no idea what that means |
04:38.41 | hardwire | dial remote.. dials local exten + 1 |
04:38.41 | Landrocker | hardwire, awesome - I'm setting up something similar over the next month or so |
04:38.42 | Grooby | :-D |
04:38.43 | hardwire | which dials back |
04:38.46 | hardwire | I hope thats adequite |
04:38.52 | hardwire | Landrocker: I bought some wrap boards |
04:38.58 | Landrocker | ...? |
04:39.00 | hardwire | and a wrap outdoor enclosure |
04:39.05 | hardwire | and some 30dbi dishes |
04:39.10 | hardwire | geode 233s |
04:39.13 | Landrocker | ah nice |
04:39.18 | hardwire | w/ madwifi supported mini-pci cards |
04:39.26 | hardwire | it took a while to get it linked at 3 miles |
04:39.31 | Landrocker | we have the network in place - I just need to get VOIP working over the top of it |
04:39.36 | Landrocker | what's the latency like? |
04:39.42 | hardwire | 8 times faster than a t1 so far |
04:39.55 | hardwire | doing a b/w test |
04:39.59 | hardwire | give me a sec |
04:39.59 | *** join/#asterisk Newbie___ (some@218.111.157.90) |
04:40.03 | Landrocker | k |
04:41.23 | hardwire | <PROTECTED> |
04:41.24 | hardwire | sent 36 bytes received 238556657 bytes 875437.41 bytes/sec |
04:41.24 | hardwire | total size is 515579904 speedup is 2.16 |
04:41.29 | hardwire | compressed hard drive image over the backbone |
04:41.36 | hardwire | apparently almost 2/1 compression |
04:41.41 | hardwire | so.. maybe not a good test :) |
04:41.41 | Landrocker | on that note - does anyone have any tips on taking latency down? using the echo test it sounds like I'm getting about 500ms round-trip, but that's just over 100mbit ethernet so it should be faster, right? |
04:41.48 | Landrocker | lol |
04:41.55 | hardwire | Landrocker: dude |
04:42.03 | hardwire | is your machine fast enough? |
04:42.18 | hardwire | I am using 233 mhz machines to pass atleast 4 channels at a time |
04:42.25 | eraser` | Landrocker: how long is the cable? |
04:42.34 | erwinism | moc if i but E100P, what else hardware should i need to setup a PBX on my office? my setup is one pot line to connect to office from outside. and dial 9 to make outside calls. |
04:42.44 | erwinism | but = buy |
04:42.47 | Landrocker | my test was just using my laptop which is a cel-m at 1.4ghz |
04:43.01 | hardwire | 64 bytes from 10.0.9.5: icmp_seq=1 ttl=64 time=1.73 ms |
04:43.01 | hardwire | 64 bytes from 10.0.9.5: icmp_seq=2 ttl=64 time=0.351 ms |
04:43.01 | hardwire | 64 bytes from 10.0.9.5: icmp_seq=3 ttl=64 time=0.316 ms |
04:43.03 | hardwire | 64 bytes from 10.0.9.5: icmp_seq=4 ttl=64 time=1.00 ms |
04:43.06 | hardwire | its going to be just fine |
04:43.13 | hardwire | after going back and forth 48 times.. its noticable |
04:43.13 | harryvv | I wonder how many sip connection I could get with my opteron 244 with gigabit :) |
04:43.14 | hardwire | however.. |
04:43.24 | Landrocker | there's probably about 15m of cable in between the two machines I was using to test |
04:43.29 | mitcheloc | grrr, mirc is beeping me speakers, but i dunno why |
04:43.30 | hardwire | we are going over sat once it goes to the internet |
04:43.30 | hardwire | so |
04:43.30 | hardwire | hah |
04:43.31 | Landrocker | plugged into the same switch |
04:43.35 | hardwire | it doesn't matter how slow my wireless link is |
04:43.41 | hardwire | I am slapping 600ms on top of it |
04:43.48 | hardwire | with a shringing jitter buffer |
04:43.49 | Landrocker | hmm |
04:43.57 | hardwire | that starts at around 700 |
04:44.02 | Moc | harryvv, I got 20 call withg 800kbits sec |
04:44.09 | Moc | with g726 |
04:44.10 | Landrocker | eraser`, any ideas? |
04:44.17 | erwinism | moc |
04:44.31 | Moc | yes ? |
04:44.42 | erwinism | moc if i buy E100P, what else hardware should i need to setup a PBX on my office? my setup is one pot line to connect to office from outside. and dial 9 to make outside calls. |
04:44.55 | Moc | you got a E1 ? |
04:44.56 | eraser` | so to make sure I understand, you're pinging a machine that you're on the same switch with and both are configured as fastethernet and you're getting ~500ms |
04:44.57 | jsolares | lots of sip/iax phones |
04:45.12 | mitcheloc | e1 for one phone? |
04:45.13 | erwinism | Moc. no |
04:45.15 | Landrocker | no, ping shows about 1ms latency |
04:45.21 | Moc | erwinism, no need for the E100P then |
04:45.22 | eraser` | ohh |
04:45.26 | Landrocker | but call latency is definately higher than that |
04:45.31 | mitcheloc | what country are you in earser? |
04:45.35 | erwinism | i Only got One ordinary telephone line |
04:45.40 | eraser` | US |
04:45.45 | jsolares | you need an fxo device erwinism |
04:45.46 | Moc | erwinism, then you only need a FXO card/device |
04:46.15 | Moc | erwinism, E100P is a card for Digital Line called E1, that support up to 23 digital channel |
04:46.22 | Moc | + 1 Data channel |
04:46.24 | eraser` | Landrocker: sorry but I don't know asterisk well |
04:46.30 | erwinism | Moc, what else hardware should i need if i got FXO |
04:46.38 | Landrocker | np, thanks anyway :) |
04:46.44 | jsolares | do you want to stick to analog phones or move to voip phones? |
04:46.50 | jsolares | that's what you should be asking yourself next |
04:46.56 | Moc | erwinism, well if you were in the US/Canada, a X100P, or a TDM400P + FXO module, or SPA-3000 would do the trick |
04:46.57 | eraser` | I'd assume it may have something to do with the other machine, you said it was 200MHz? |
04:47.02 | mitcheloc | eraser: e1 is for europe |
04:47.06 | erwinism | jsandnes yes hhehe |
04:47.10 | Landrocker | nope, they're both above 1ghz |
04:47.13 | Moc | I donno if those device will work where you are located |
04:47.16 | eraser` | ah |
04:47.26 | eraser` | something is seriously defunct |
04:47.31 | Landrocker | hmm |
04:47.33 | *** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net) |
04:47.48 | riksta | can you use an intel voice modem as a cheap type of fxo |
04:47.58 | erwinism | jsolares i want to stick to analog.. |
04:48.00 | Moc | riksta, you can try, but dont ask us !!! |
04:48.10 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
04:48.21 | riksta | Moc: i'm not asking how...i'm just asking if it's poss :) |
04:48.28 | jsolares | erwinism: then you'll need fxs devices, btw sticking to analog phones could be more expensive depending on what voip phone you end up with |
04:48.39 | Inv_arp | riksta: google says it is |
04:48.41 | mitcheloc | riksta: yes you can |
04:48.53 | mitcheloc | riksta: if you want the model number i've got it, but can't get it till monday |
04:48.53 | Moc | riksta, when I say that is, everything is posible.. just might be supported |
04:48.57 | Moc | might NOT |
04:49.05 | jsolares | although i've seen good prices for fxs devices |
04:49.15 | Moc | I havent seen 1 person in over 1 year play with other than x100p card |
04:49.30 | mitcheloc | moc, count me as the first then ;) |
04:49.46 | harryvv | Moc what do you mean by play with other? |
04:49.54 | Moc | ok.. I mean considering getting a standards Voice modem to work with * |
04:50.16 | erwinism | jsolares: i have 20 deparments in the office. so that means i need 20 FXS cards? |
04:50.26 | mitcheloc | which should work! |
04:50.31 | harryvv | okay I see what you are saying. Are there other voice modems other then the x100p that will work with asterisk? |
04:50.32 | Moc | erwinism, not exactly |
04:50.45 | mitcheloc | erwinism: nope, thats not possible unless you have multiple servers |
04:50.46 | Moc | erwinism, you say you want to be analog only ? no IP phone ? |
04:51.02 | mitcheloc | digium has cards with 4 ports each, but choose if you want voip phones first though (probably the better route) |
04:51.02 | erwinism | Moc yes, analog only |
04:51.12 | jsolares | yeah, not exactly there are fxs devices called ata's like digiums iaxy, or sipuras or grandtream, or even linksys pap2 (or something) |
04:51.26 | Moc | erwinism, ish, then you need to get standards Home Analog phone ;) + a channel bank + the T1 Card ;) |
04:51.27 | mitcheloc | so 5 digium cards, or 1 t100p card and a nice little adtran box to break it out to a punch block |
04:51.43 | jsolares | if you're lucky with ebay you could score a t1 channel bank for cheap |
04:51.57 | mitcheloc | jsolares: i got one for free =) |
04:52.04 | jsolares | :p |
04:52.11 | mitcheloc | just go bug someone to install asterisk for them, use a t100p to replace their adtran, then jack the one on the wall |
04:52.34 | Moc | erwinism, why you dont want to go IP ? |
04:52.54 | erwinism | Moc, because my phones here are analog.. |
04:53.11 | mitcheloc | replace them? |
04:53.13 | Moc | so you want to reuse your current analog telephone ? |
04:53.22 | jsolares | you have to take the channel bank + t1 card into account, and see if it wouldnt be better to replace the analog phones |
04:53.30 | Moc | yea |
04:53.33 | tuxinator_linux | I got a dialtone on my TDM400P ! |
04:53.38 | erwinism | Moc how much would cost a digital phone? |
04:53.42 | erwinism | ip phone i mean |
04:53.43 | erwinism | hehehe |
04:53.48 | Moc | erwinism, depend, there is different type |
04:53.49 | jsolares | 80$ and up |
04:54.14 | Moc | the recommanded model cost 180$ |
04:54.14 | erwinism | im making some plans here ") |
04:54.23 | jsolares | a channel bank + t1 card could be 100$ per channel |
04:54.55 | erwinism | Moc what if i will switch to IP phone? what hardware should i need? |
04:54.58 | Moc | or for lower, you drop down to 84$ for a cheap buisness phone |
04:55.12 | Moc | erwinism, ethernet card ;) |
04:55.13 | jsolares | nothing more than the asterisk box with the fxo device |
04:55.19 | jsolares | yeah with an ethernet card hehe |
04:55.27 | riksta | and a switch ;) |
04:55.41 | jsolares | you could also hook it up to a voip provider to have long distance calls for cheap |
04:55.46 | mitcheloc | or $12/channel if you do what i said to get a free channel bank =) |
04:55.48 | Moc | erwinism, you could also use SoftPhone |
04:55.55 | erwinism | Moc, Ethernet card, one FXO for outside calls and one FSX for Imcoming calls? |
04:56.09 | Moc | erwinism, FXO for connecting to your POT line |
04:56.22 | mitcheloc | ** actually i got my t100p for $280 so thats not right, it's around $20/channel |
04:56.23 | jsolares | yeah, it takes care for calling and receiving |
04:56.25 | erwinism | wow thats cheap |
04:56.30 | Moc | no need of FXS (unless you want to plug the fax on it) |
04:56.44 | erwinism | ok thanks i got it |
04:56.49 | erwinism | this is what i need |
04:56.57 | Moc | erwinism, softphone cost make the cost at about 20$ per phone too |
04:57.00 | jsolares | it might also be cheaper to go with ata's that go from network to analog phone like the iaxy |
04:57.10 | Moc | lol softphone is free + headset = 20$ |
04:57.25 | erwinism | hehe |
04:57.31 | jsolares | not cheaper than sofpthone or stolen channel bank tho ;p |
04:57.52 | mitcheloc | stolen channel bank, and 1/2 stolen t100p cards (someone sold me one here for $280) hehe |
04:58.16 | Moc | erwinism, what make * powerfull too is, you can modify it as your need, and you can interconnect all those type of device |
04:58.42 | erwinism | okay |
04:58.45 | erwinism | i got it |
04:58.53 | jsolares | indeed it is, i have 2 voip providers for redundancy to make cheap calls to the us :) |
04:59.06 | Moc | you can have FXO/FXS/ISDN/VoIP(IAX/SIP/MGCP/SKINNY/H323..) |
04:59.11 | jsolares | both are cheaper than making a local call :| |
04:59.33 | jsolares | only if you're a masochist moc |
04:59.40 | jsolares | :p |
04:59.53 | mitcheloc | hey btw THIS IS A PSA!!!! please SPREAD THE KNOWLEDGE >>>>> I learned today that, most phone companyes (99%) leave the 911 service live on the phone lines running into your house...so you can have real 911 service without paying for a phoneline @ your local telco. <<< (yes some of you may know this, but many do not) |
04:59.55 | Moc | erwinism, over here, I got no Analog line, it all done over Internet and VoIP provider, my incoming and outgoing |
05:00.28 | erwinism | wow thats great |
05:00.36 | erwinism | i will plan for this. |
05:00.48 | mitcheloc | so buy an fxo card and hook it up to your house system, even if theres no dial tone, the 911 service might and probably will work |
05:00.52 | jsolares | yeah, depending on your provider it might be cheaper than having an anolog line |
05:01.19 | Moc | mitcheloc, even if no dialtone, that cool |
05:01.27 | Moc | I never tryed that |
05:02.03 | erwinism | moc, FXO will handle the outgoing and incoming calls ? |
05:02.04 | mitcheloc | yep, it's some liability they didn't want to be responsible for (i.e. you did not pay your bill, then it'd be like them saying screw you mr. dying man!) |
05:02.22 | Landrocker | they're required to by law in the states iirc |
05:02.23 | Moc | recently our local provider keep the dialtone, and you can dial, but get a prompt that if we want the service, we need to call the 611 ;) |
05:02.30 | Moc | erwinism, yes |
05:02.36 | erwinism | ok got it.. |
05:02.43 | mitcheloc | but your 911 service will work, i was about to buy a real phone line for this |
05:02.56 | mitcheloc | but now i don't have to =) |
05:04.15 | erwinism | ok jsolares and moc i have to go... reading asterisk manuals |
05:04.20 | jsolares | gluck |
05:04.20 | erwinism | thanks |
05:04.40 | Landrocker | anywho - anyone got any idea on the flash-hook thing? |
05:05.11 | Landrocker | xlite can't flash-hook and my voip hardphone probably won't get here for a week or two |
05:05.44 | Landrocker | alternatively, are there any softphones that do have flash? |
05:07.59 | harryvv | mit, buy a real phone for 911? thats a good idea. BTW, a federal building where I have worked has a seperate phone for calling the fire department. I never asked but is it my assumption that is a strait analog phone to the CO that does bypass the pbx in the event it fails? |
05:08.01 | ManxPower | ~docs |
05:08.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
05:08.02 | riksta | is voipuser.org down? |
05:08.45 | *** part/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net) |
05:08.53 | Inv_arp | hhmm can i set for ex.. #7 to perform an action during a conversation with *? |
05:14.16 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
05:15.25 | Inv_arp | come on i thought that was an ez one |
05:16.10 | brc__ | eh |
05:16.12 | brc__ | you can't |
05:16.29 | brc__ | well |
05:16.30 | brc__ | depends |
05:16.52 | brc__ | you can set a key to start recording while two channels are bridged |
05:17.01 | brc__ | I dunn remember how |
05:17.18 | brc__ | so if you can dive into the code it's possible |
05:17.54 | jsolares | you could use my new noob super app that's the inverse of waitforsilence that waits on sound, and start doing something only when the dsp thinks there's no silence :XD |
05:20.33 | Inv_arp | heh k |
05:21.22 | jsolares | it's great for monitoring extensions on an avaya definity with service observing |
05:24.43 | Inv_arp | heh trying to come up with a agi/php script to handle agents and implement with sqlite... trying to figure out how i can prgram keys to say "agent on break" or "ACW" |
05:24.59 | riksta | Inv_arp: i wanted something like that too |
05:25.20 | riksta | kinda like the avaya phones, where you can press keys to log into lunch or ACW etc |
05:26.12 | Inv_arp | riksta: exactly... hmm i might have to implement a "middle man" app to do something like that for me |
05:26.53 | riksta | i was thinking the same thing, maybe write an app that they have running on the computer that they can press buttons on, that integrates with asterisk manager |
05:27.23 | Inv_arp | riksta: hmm im thinking python/twisted combination |
05:27.33 | jsolares | why not make an agi that's on an extension, and when the agent calls that extensions it executes the agi putting them on break |
05:27.55 | riksta | true |
05:27.57 | ta[i]nted | looks like broadvoice is down for asterisk users |
05:28.04 | riksta | voipuser.org is down too |
05:28.06 | riksta | annoying |
05:28.14 | ta[i]nted | really |
05:28.17 | eraser` | how reliable is broadvoice usually? |
05:28.28 | ta[i]nted | some tech guy just told me its some kind of SIP INVITE auth issue |
05:28.33 | Inv_arp | jsolares: but at work i ca put myself on flashing break during a call, then when call ends im on it |
05:28.33 | loud | really ? down ? let me check |
05:28.44 | jsolares | ah ic |
05:28.45 | ta[i]nted | eraser` i'd say it's got problems.. international calling is ass |
05:28.46 | loud | oh damn |
05:29.14 | Inv_arp | BV is down? incoming werks fine |
05:29.25 | jsolares | well if you have two line phones you can dial the break extension on the other line :p |
05:29.34 | brc__ | O M G! http://www.rasterwerks.com/dev.public/phosphor_alpha_4_248.htm |
05:29.44 | brc__ | check it out! |
05:29.49 | brc__ | play with me |
05:30.06 | ta[i]nted | Inv_arp really everything is down for me |
05:30.15 | loud | same here. |
05:30.18 | *** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com) |
05:30.24 | D1ng0 | anyone alive ? |
05:30.26 | Inv_arp | ta[i]nted: on a call right nno on BV (incoming) |
05:30.31 | Inv_arp | err now |
05:30.38 | D1ng0 | Inv_arp, yeah for me too |
05:30.45 | ta[i]nted | Inv_arp using asterisk or ip phone |
05:30.51 | D1ng0 | asterisk here |
05:30.55 | ta[i]nted | Inv_arp BV tech said it is specific to asterisk |
05:30.56 | Inv_arp | ta[i]nted: asterisk |
05:31.10 | D1ng0 | they are full of it |
05:31.14 | eraser` | any suggestions in terms of a provider then, New York/New England area |
05:31.16 | Inv_arp | ta[i]nted: asterisk -> HT486 |
05:31.21 | D1ng0 | asterisk worked fine until they made changes today |
05:31.34 | Inv_arp | eraser`: outgoing/incoming? |
05:31.38 | loud | you think ? becaose of the invite thing ? |
05:31.41 | ta[i]nted | looks like they changed SIP auth |
05:31.42 | riksta | whats the key combo to reset a 79xx phone |
05:31.43 | D1ng0 | yes |
05:31.44 | eraser` | both |
05:31.45 | loud | bacause rather |
05:31.54 | loud | **# |
05:32.04 | riksta | nop |
05:32.20 | Inv_arp | whats wrong with BV? cant recieve incoming? outgoing? |
05:32.25 | loud | type reset through CLI, ill reset everything :) |
05:32.26 | D1ng0 | outgoing works |
05:32.35 | D1ng0 | you need to add a few lines to asterisk for it |
05:32.42 | D1ng0 | incoming is still broken |
05:32.49 | loud | which likes D1ng0 |
05:32.51 | Inv_arp | i just called myself on BV thru my cell |
05:32.56 | *** join/#asterisk krilloz (majestic@220-253-7-238.VIC.netspace.net.au) |
05:32.58 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
05:33.08 | krilloz | hey |
05:33.13 | D1ng0 | see http://lists.digium.com/pipermail/asterisk-users/2005-March/092953.html |
05:33.19 | Inv_arp | cell->BV->*->HT486 |
05:33.29 | D1ng0 | it fixes outbound, but not inbound |
05:34.59 | Inv_arp | hmm my inbound still works..... knock on wood |
05:35.12 | *** join/#asterisk DyOS (~me@ip68-2-153-157.ph.ph.cox.net) |
05:35.13 | D1ng0 | Hrmmm mines been broekn for hours |
05:35.39 | Inv_arp | D1ng0: lemme show u my sip.conf for BV to compare |
05:35.39 | DyOS | if anyone is interested in making a few bucks msg me i need help setting up a script in asterisk that won't work for me |
05:35.46 | BoRiS | <in an Australlian voice> "The Dingo ate your baby!" |
05:35.46 | DyOS | i will pay via paypal if anyone is interested |
05:35.55 | D1ng0 | Inv_arp, /msg it to me |
05:36.02 | riksta | you know on a cisco 7940, where you have the buttons on the right for each line, how do you set the name that is displayed, can you have alphanumeric chars instead of just the line's number, eg "line 1" rather than 1000 ? |
05:36.11 | D1ng0 | BoRiS, LOL |
05:36.18 | BoRiS | :) |
05:36.52 | krilloz | I was attacked savagely by a Dingo once, in the outback.. |
05:36.59 | D1ng0 | damn i cant even get BV customer support on the phone |
05:37.05 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
05:37.09 | D1ng0 | krilloz, no way, wasnt me :) |
05:37.10 | loud | riksta, yes. line1_shortname: "something" |
05:37.13 | BoRiS | lol |
05:37.14 | riksta | thanks :) |
05:37.21 | riksta | i had line1_displayname :) |
05:37.27 | krilloz | as you can see I'm from .au , so I must be telling the truth! |
05:37.40 | D1ng0 | krilloz, im Aussie also hence the nickname |
05:37.46 | D1ng0 | i just like in the USA |
05:37.51 | krilloz | ahh |
05:37.53 | D1ng0 | ehh live in the USA |
05:38.02 | krilloz | I see |
05:38.15 | krilloz | using asterisk to call back home? |
05:38.20 | D1ng0 | though ive been debating going back home to brisbane |
05:38.39 | krilloz | ahh, brisbane is a bit of a backwater though isnt it? ;) |
05:38.44 | D1ng0 | krilloz, yeah thru BV which is now very broken |
05:38.49 | D1ng0 | heh nahhhhh |
05:41.18 | *** part/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
05:44.10 | loud | well |
05:44.24 | loud | international calls work .. just tried. |
05:44.33 | jsolares | ahh fun to see all of the angry bv costumers in asterisk-users |
05:44.51 | loud | hah, the wife one. |
05:45.03 | loud | "i can't stand my wife when she can not make calls". |
05:45.04 | jsolares | i'm not sure what's worse, bv not notifying customers or the customers acting like spoiled children |
05:45.42 | jsolares | yeah that one was funny |
05:45.59 | D1ng0 | well it would be nice if BV inbound worked, i got outbound working |
05:46.30 | DyOS | I'm having a few problems setting up some things in asterisk for my small business i'm wondering if anyone is intersted in helping me out I will pay 20/hour for help if anyone is interested email me at irc@DynamicOnsiteSolutions.com |
05:47.09 | GrimStone | D1ng0: i added those lines but even outbound doesn't work yet for me |
05:47.41 | GrimStone | like the voice data comes in on port 5060 from BV , but asterisk seems to just ignore it |
05:47.44 | jsolares | are you getting registered at bv server? |
05:48.02 | GrimStone | yeah i'm registered , and it accepts my INVITE password too |
05:48.04 | D1ng0 | yes |
05:48.07 | D1ng0 | me too |
05:48.16 | Shido | boink |
05:48.17 | D1ng0 | my outbound works now, but not inbound |
05:48.25 | GrimStone | can you hear any sound when you do outbound ? |
05:48.40 | GrimStone | cos asterisk just ignores the incoming voice packets for me |
05:48.51 | Shido | voice packets? |
05:48.52 | jsolares | atleast you havent gone mad on us like the users on the list saying how bv screwed your life since incoming/outbound was not working :) |
05:48.53 | Shido | are you nat'd ? |
05:49.24 | GrimStone | nope , nat=no |
05:49.33 | Shido | no, I mean |
05:49.33 | GrimStone | worked fine until yesterday |
05:49.35 | Shido | are you behind a router |
05:49.45 | GrimStone | no . got external IP |
05:49.47 | Shido | ok |
05:49.50 | Shido | where is the * box |
05:49.51 | Shido | ? |
05:50.43 | GrimStone | Shido: well iconnecthere works fine , running on a DSL line |
05:50.49 | jsolares | listen to greg, he's good with the asterisk voodoo, he'll shake his * voodoo doll and have you rsystem working in no time :) |
05:51.25 | Shido | ppl know me real name now , scary |
05:51.41 | GrimStone | when i make a call with BV .. asterisk sends the INVITE , with password and BV accepts it |
05:51.56 | jsolares | it IS on your hostmask, and i CAN call you on your phone ;p with the 31337 extension |
05:52.26 | GrimStone | and then i get a lot of data on port 5060 from BV , but asterisk just never seems to ignore it and act as if it got nothing |
05:52.50 | jsolares | and IT is thanks to you that i've gotten along fast with asterisk :) |
05:53.44 | GrimStone | D1ng0: do you have nat=yes ? |
05:56.09 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
05:56.19 | D1ng0 | yes |
05:56.37 | GrimStone | hmm .. strange cos with nat=no , nothing works , heh |
05:56.48 | Shido | if the box is on a public ip |
05:56.54 | Shido | and your sip phone is on a public ip |
05:56.58 | Shido | you dont need to set nat to anything |
05:57.02 | Shido | leave it alone ;) |
05:57.23 | GrimStone | well would it hurt setting it to no ? |
05:57.44 | jsolares | the way i remember nat working yes it hurts setting it to no |
05:58.43 | GrimStone | what is the default nat= setting then ? |
05:59.19 | D1ng0 | how can i set the refresh rate on a registration ? |
05:59.49 | Shido | set that in your phone D1ng0 |
06:00.02 | D1ng0 | no i mean for a rergistration |
06:00.19 | Shido | registering your phone to asterisk? is set in the phone |
06:00.23 | Shido | what do you mean? |
06:00.32 | riksta | Shido: he means to a voip service i guess |
06:00.38 | riksta | like FWD etc |
06:00.46 | GrimStone | LMAO .. |
06:00.53 | GrimStone | i just set nat=yes and it works now |
06:01.07 | Shido | heh |
06:02.37 | GrimStone | well thanks for the clue .. strange thing is nat=no worked fine until yesterday cos i have external ip with the right ports un-firewalled |
06:02.48 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
06:02.53 | D1ng0 | sip show registry shows sip.broadvoice.com:5060 3219892181@s 2528 Registered |
06:03.09 | D1ng0 | look at the refresh of 2528 |
06:03.19 | GrimStone | until yesterday BV had a 10 second timeout |
06:03.34 | D1ng0 | well my setup worked fine until today |
06:03.41 | D1ng0 | and their dumb changes |
06:03.48 | GrimStone | even if you try to force a 10 sec. timeout it will have 2528 |
06:04.04 | GrimStone | or 2000+ |
06:04.19 | D1ng0 | well outbound works, inbound is still broken |
06:04.52 | GrimStone | hmm .. do you have it defined as a peer or friend ? |
06:05.08 | D1ng0 | what broadvoice ?? |
06:05.13 | GrimStone | yeah |
06:05.17 | Shido | hehe |
06:05.20 | Shido | ppl are listening! |
06:05.26 | Shido | friends are evil |
06:05.43 | D1ng0 | [incoming] |
06:05.43 | D1ng0 | username=3219892181 |
06:05.43 | D1ng0 | type=user |
06:05.43 | D1ng0 | secret=password |
06:05.44 | D1ng0 | host=sip.broadvoice.com |
06:05.44 | GrimStone | and setting sip.broadvoice.com in your /etc/hosts doesn't work well too today |
06:05.44 | D1ng0 | fromuser=3219892181 |
06:05.46 | D1ng0 | fromdomain=sip.broadvoice.com |
06:05.48 | D1ng0 | context=from-pstn |
06:05.49 | jsolares | ack! |
06:05.50 | D1ng0 | canreinvite=no |
06:05.52 | D1ng0 | authuser=3219892181 |
06:05.53 | jsolares | ~pastebin |
06:05.54 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
06:06.00 | D1ng0 | ouch damn sorry i know better |
06:06.13 | mitcheloc | ding: you having bv problems? =) |
06:06.14 | D1ng0 | i went to paste it in another window |
06:06.20 | GrimStone | you have a seperate context for incoming ? |
06:06.24 | D1ng0 | yes |
06:06.38 | mitcheloc | anyone know whats wrong here? |
06:06.39 | mitcheloc | <PROTECTED> |
06:06.43 | mitcheloc | Mar 5 22:03:39 WARNING[5256]: ast_expr.y:483 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: |
06:06.43 | mitcheloc | <PROTECTED> |
06:06.44 | GrimStone | Shido: they're fine as long as you have the right permits |
06:07.24 | GrimStone | D1ng0: that isn't a good idea .. try commenting out the incoming context and put it all in one context |
06:07.37 | D1ng0 | i have an incoming context, and a sip.broadvoice.com context and until today it all WORKED fine |
06:07.47 | GrimStone | and use permit=147.135.0.0/16 |
06:08.16 | D1ng0 | GrimStone, even if it was working ? |
06:09.22 | mitcheloc | how can i escape this to use it in the dialplan? "<http://127.0.0.1/Bellcore-dr3>" |
06:09.57 | Shido | errr |
06:10.06 | Shido | did you put a register line in your sip.conf for FWD? |
06:10.11 | mitcheloc | it seems escaped to me (or doesn't need to be), but asterisk complains about it |
06:11.59 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
06:13.31 | D1ng0 | GrimStone, nope still broken incoming |
06:17.01 | *** join/#asterisk erwinism (~pogz@210.213.143.73) |
06:17.23 | erwinism | hello, what port does asterisk uses? |
06:18.02 | riksta | erwinism: depends what protocols you use |
06:18.10 | Inv_arp | D1ng0 and I have similair setup they only diff is my refresh in sip show registry is 15 and his is 2823 and my incoming BV works |
06:18.18 | erwinism | riksta i use the default |
06:18.26 | riksta | erwinism: what!? |
06:18.27 | Inv_arp | my setup http://pastebin.ca/6916 |
06:18.29 | Sedorox | IAX, IAX2, SIP, MG... something |
06:18.34 | Sedorox | all use different ports |
06:19.00 | riksta | erwinism: SIP is 5060, with RTP ports of default 10000-20000 |
06:19.33 | erwinism | riksta how can i enable those in my firewall? |
06:19.39 | riksta | read the manual |
06:19.58 | D1ng0 | Inv_arp, how did you set refresh |
06:22.16 | Inv_arp | D1ng0: never did think its in chan_sip.c but im no C coder |
06:24.09 | *** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net) |
06:24.31 | erwinism | riksta i got it.. my next question is, do i have to add users to asterisk server? |
06:25.12 | riksta | well..what do you think? |
06:25.47 | Sedorox | erwinism: The Answer you seek.... is 42... |
06:26.20 | GrimStone | hmm. BV works now for outgoing but it doesn't seem to hangup calls right |
06:27.05 | GrimStone | when i stop a call , the CDR says call had NO ANSWER and billsec=0 |
06:27.26 | D1ng0 | my outgoing has been fine, its my incoming thats broken |
06:27.46 | D1ng0 | Inv_arp, so both your inboind and outbound are working ? |
06:27.58 | erwinism | riksta how can i add users to my asterisk server? |
06:28.20 | *** join/#asterisk andrew` (~andrew@adsl-67-119-26-16.dsl.snfc21.pacbell.net) |
06:28.46 | D1ng0 | so is there another good VOIP service that works with asterisk, BV is going to loose me as a customer |
06:28.48 | Inv_arp | D1ng0: i only use BV inbound yes |
06:29.01 | GrimStone | Inv_arp: does CDR work fine for you ? asterisk doesn't seem to hangup calls right now |
06:29.03 | D1ng0 | Inv_arp, well your config didnt work for me at all |
06:29.13 | GrimStone | and it shows the call as NO ANSWER |
06:29.23 | *** join/#asterisk roamer323 (~sing@67.71.60.238) |
06:29.37 | Inv_arp | hmm what version ya use? |
06:32.11 | GrimStone | 2/15 cvs |
06:35.07 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
06:38.04 | erwinism | what is a good softphone to use? |
06:38.27 | riksta | x-lite |
06:39.00 | PTG123 | eyebeam :) |
06:42.05 | GrimStone | is there any page on wiki with a description of the "register => " command ? |
06:44.24 | *** join/#asterisk Damin_Mobile (~pocketirc@213.sub-166-155-119.myvzw.com) |
06:44.48 | MuppetMaster | Although I can not get Eyebeam video to work with Asterisk at the moment. A patch is needed for H.263+ to work, and attempting to sort that out. |
06:45.21 | BoRiS | Look at the latest CVS HEAD. Should have it. |
06:45.46 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
06:46.01 | MuppetMaster | BoRiS: Latest CVS HEAD should have what? |
06:47.31 | Damin_Mobile | Kram just committed a h263+ patch to cvs head. |
06:47.34 | PTG123 | how do i make asterisk play a ring |
06:47.36 | PTG123 | when its calling |
06:48.19 | BoRiS | add r in dial |
06:48.35 | PTG123 | add ,r where? |
06:48.46 | Moc | PTG123, after the timeout |
06:50.23 | *** join/#asterisk Newbie___ (some@60.48.53.154) |
06:50.41 | Newbie___ | AMP rocks ! |
06:50.57 | BoRiS | Moc!!!!! |
06:51.14 | PTG123 | hey what does the ,t do? have a timeout? |
06:51.41 | BoRiS | transfer |
06:51.55 | PTG123 | so i just want DIAL(IAX2/BLAH,30,r) |
06:51.55 | PTG123 | ? |
06:52.03 | BoRiS | Moc: Whats up? |
06:52.11 | Moc | hi, nothing |
06:52.19 | Moc | doing some virtual cleanup |
06:52.33 | BoRiS | oh yeah? |
06:52.50 | Moc | yep |
06:53.08 | Moc | I got like 5 different linux machine just for my personal usage.. |
06:53.17 | Moc | alittle over kill... |
06:53.23 | BoRiS | Uhhhhhhh........ Yeah! |
06:53.25 | BoRiS | hehe |
06:53.41 | *** join/#asterisk The_Ball (~alex@dialup-211.43.194.203.acc02-wick-bne.comindico.com.au) |
06:53.42 | Moc | Im trying to bring the number to 2.. |
06:53.54 | The_Ball | is it possible to use a TDM card as a modem? |
06:54.19 | PTG123 | i now use my notebook running winxp with a vmware freebsd windows.. and hummingbird exceed for my xterm |
06:54.21 | PTG123 | it works great :) |
06:54.29 | PTG123 | my linux and frebsd boxes collect dust now |
06:55.24 | Shido | PTG123, thats right |
06:55.25 | Moc | oh you got exceed ? is it nice ? |
06:55.34 | Shido | if u dont want to use the transfer |
06:55.36 | PTG123 | awesome |
06:55.38 | PTG123 | i have used it for years |
06:55.47 | Moc | I never found it hehe |
06:55.57 | PTG123 | hehe never found it warezed you mean? :) |
06:56.15 | Moc | heu.. yea ; |
06:56.23 | PTG123 | i think the eDonkey network probably has it |
06:56.24 | Moc | well I might just found it hehe |
06:56.39 | Moc | powersuit 10 SP5 should be ok ? |
06:56.55 | PTG123 | let me see |
06:58.16 | PTG123 | i have no idea |
06:58.18 | PTG123 | i use 9 :) |
06:58.21 | PTG123 | just make sure it has exceed in it |
06:58.35 | Moc | Exceed PowerSuite™ 10 |
06:58.51 | Moc | will see sone ennuf |
06:58.53 | PTG123 | ah yah |
06:58.54 | PTG123 | that is it |
06:58.58 | PTG123 | just has nfs with it |
06:59.06 | Moc | ha.. |
06:59.07 | PTG123 | http://connectivity.hummingbird.com/products/nc/exceed/index.html |
06:59.20 | PTG123 | i think 10 got better mutiple monitor support |
06:59.26 | tzafrir | good morning |
06:59.52 | Moc | morning |
07:00.02 | tzafrir | now why would you use an illegal warez exceed? |
07:00.36 | Moc | to try it |
07:00.37 | tzafrir | Take a look at cygwin and tell me what is missing from there |
07:00.43 | Moc | before buyin |
07:00.57 | PTG123 | try before you buy |
07:01.02 | PTG123 | and cygwin is nothing like exceed |
07:01.23 | tzafrir | what do you need? |
07:01.53 | The_Ball | what do i press/dial to en a Record()? |
07:03.34 | tzafrir | The_Ball, you need it in your dialplan |
07:04.10 | tzafrir | An besides, what's this on-topic talk all of a sudden? You're interrupting our conversation! ;-) |
07:04.15 | The_Ball | tzafrir, the docs doesn't say anything about thathttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x958.html |
07:04.36 | The_Ball | hehe |
07:05.17 | tzafrir | The_Ball, you can configure Asterisk to do many things. The dialplan (extensions.conf, basically) tells Asterisk how to react to whatever is dialed. |
07:05.34 | tzafrir | If you put in your dialplan something like: |
07:07.01 | tzafrir | exten => 12345,1,Record('/var/lib/asterisk/my.sound') (or whatever, I don't remember the syntax now) it will Record when dial 12345. Assuming you put it in a sensible context |
07:15.27 | tzafrir | slightly OT: for those of you who think that the most intruders can do is steal calls: |
07:15.38 | tzafrir | http://www.theregister.co.uk/2005/03/03/skype_broadreach_voip_calls/ IP over VoIP |
07:16.07 | tzafrir | skype as a tunneling protocol? |
07:18.12 | PTG123 | anyoen here using g729 on bsd? |
07:20.13 | The_Ball | tzafrir, that's fine, i have asterisk recording, but when i try to end the record by presing hash or star i get the buzy tone and Mar 6 16:52:52 WARNING[9196]: pbx.c:1923 ast_pbx_run: Invalid extension '#', but no rule 'i' in context 'setup' in the console |
07:21.16 | GrimStone | chan_sip.c:7943 handle_request: That's odd... Got a response on a call we dont know about. |
07:21.40 | D1ng0 | Broadvoice SUCKS |
07:21.50 | GrimStone | if i use nat=no with broadvoice i get this and no sound .. with nat=yes i get sound and same message |
07:22.09 | GrimStone | D1ng0: depends .. was great while it worked |
07:22.16 | D1ng0 | they really screwed alot of people today |
07:22.23 | tzafrir | The_Ball, pressing '#' should get you to the "next priority". Put a Hangup there |
07:22.42 | tzafrir | Alternatively, terminate the recording by hanging up |
07:22.52 | hardwire | hmm |
07:22.57 | hardwire | tzafrir: you just gave me an idea |
07:23.01 | GrimStone | it seems to be an issue with asterisk not recognizing that a call has started |
07:23.02 | D1ng0 | whats a good us VOIP DID provider |
07:23.09 | hardwire | D1ng0: where? |
07:23.14 | D1ng0 | florida |
07:23.20 | hardwire | voiceconduits.com |
07:23.40 | D1ng0 | ive gotta dump Broadvoice since they dont know anything about customer notifications |
07:24.01 | PTG123 | teliax.com |
07:24.02 | PTG123 | try them |
07:24.47 | The_Ball | tzafrir, will you look at my dialplan: http://channels.debian.net/paste/316 |
07:25.28 | hardwire | I need to up my prices for broadband in a few rural communities |
07:25.56 | hardwire | I think if I up them and offer 1000 minutes VoIP/LD that will help pay for more dedicated bandwidth |
07:26.00 | hardwire | as well as use it up :) |
07:26.34 | tzafrir | The_Ball, say, do you use the Debian packages? |
07:26.46 | The_Ball | tzafrir, no, this is on a gentoo box |
07:27.26 | tzafrir | anyway, one thing you can do is set verbosity to something high enough (e.g: 3) and look at the CLI while the call is running |
07:27.41 | tzafrir | 'set verbose 3' in the asterisk cli |
07:28.49 | tzafrir | anyway, I odn't think you need Answer if this is not an incoming call |
07:29.30 | The_Ball | tzafrir, http://channels.debian.net/paste/paste |
07:29.41 | hardwire | pastey |
07:29.46 | The_Ball | tzafrir, oups, -> http://channels.debian.net/paste/317 |
07:30.12 | D1ng0 | hrmmm |
07:32.10 | tzafrir | not that it would matter here, but it is generally sensible to use two different sounds |
07:34.09 | tzafrir | It seems that the error is from Playback. Playback returned -1 ? |
07:34.22 | tzafrir | Did you try to remove the Answer ? |
07:35.13 | The_Ball | tzafrir, that didn't work |
07:36.01 | tzafrir | The_Ball, meaning? What happened? |
07:36.24 | The_Ball | i think i have found a error, show dialplan does not show the record() |
07:42.32 | D1ng0 | okay so how do i tell what my teliax phone number is after i sign up ? |
07:43.55 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
07:46.53 | Shido | ok |
07:46.55 | Shido | back |
07:49.05 | D1ng0 | PTG123, okay so how do i tell what my teliax phone number is after i sign up ? |
07:50.14 | Shido | tzafrir whats wrong? |
07:51.16 | tzafrir | Shido, what's right? |
07:52.47 | GrimStone | man .. it sucks that i can get everything to work except the CDR records are screwed |
07:53.22 | Shido | tzafrir you are recording tzafrir_home ? |
07:53.38 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
07:53.51 | GrimStone | with Broadvoice , i can hear the sound with nat=yes , but asterisk keeps saying "got a response on a call i don't know about" |
07:53.52 | tzafrir | Shido,, no I'm recording tzafrir |
07:55.58 | tzafrir | Shido, last time I talked from tzafrir_home I said something related to licensing. Do you ask about that? |
08:01.06 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
08:01.15 | ta[i]nted | is BV still having problems? |
08:01.35 | ta[i]nted | i'm getting a "the device you are using is not registered to place calls on the network.. please contact your administartor" |
08:02.27 | D1ng0 | ta[i]nted, YUPP my outbounad works but no inbound |
08:03.23 | ta[i]nted | how did u get outbound working? |
08:03.32 | ta[i]nted | neither of mine works now |
08:05.10 | Shido | bbl |
08:05.52 | WilliamK | they haven't been able to port #s and keep saying "Soon.." |
08:05.52 | ta[i]nted | WilliamK are u able to place calls? |
08:05.54 | WilliamK | I called em up and said, ya'll said soon was last summer too, so which is it |
08:06.09 | WilliamK | tainted, can't be a cust if I can't get a # ported |
08:06.10 | WilliamK | =) |
08:06.22 | PTG123 | heh |
08:06.34 | PTG123 | williamk: i'll port your # for you :) |
08:06.54 | WilliamK | PTG, I'm hoping we've taken care of that problem =) |
08:07.07 | PTG123 | heh |
08:07.29 | PTG123 | man i need to go to bed |
08:07.30 | WilliamK | do I "DARE" upgrade to the later cvs? |
08:07.37 | PTG123 | i wouldn't :) |
08:08.05 | PTG123 | whats wrong with your 729? |
08:08.09 | WilliamK | the last time I tried, it ate it for lunch |
08:08.11 | PTG123 | i got mine working tonight on freebsd baby |
08:08.27 | WilliamK | bad enough I gotta re-reg it |
08:08.59 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
08:12.25 | jontow | i just did a successful test on an outbound call using Jeff Rizzo's ported zaptel drivers, using an FXO clone card |
08:12.33 | jontow | .. on NetBSD -CURRENT :) |
08:15.44 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
08:17.14 | D1ng0 | WilliamK, i got outbound BV working no inbound |
08:17.46 | GrimStone | D1ng0: do you get messages like "thats odd. got a response on a call ..." ? |
08:18.45 | PTG123 | hey can some people ping sip1.way2fast.com |
08:18.55 | PTG123 | i wanna know the latency and the location of your cablem modem :) |
08:19.21 | riksta | im in the uk, i get rtt min/avg/max/mdev = 169.847/316.353/475.396/90.573 ms |
08:19.25 | jontow | --- sip1.way2fast.com ping statistics --- |
08:19.25 | jontow | 5 packets transmitted, 5 packets received, 0% packet loss |
08:19.25 | jontow | round-trip min/avg/max/stddev = 105.564/108.220/110.406/2.185 ms |
08:19.25 | jontow | [03:13] > i just did a successful test on an outbound call using Jeff Rizzo's ported zaptel drivers, using |
08:19.25 | jontow | +an FXO clone card |
08:19.25 | jontow | [03:13] > .. on NetBSD -CURRENT :) |
08:19.27 | jontow | [03:14] QUIT: BoRiS : "Ping timeout: 360 seconds" |
08:19.29 | jontow | [03:14] QUIT: Luhiwu : Read error: 60 (Operation timed out) |
08:19.31 | *** join/#asterisk denon (denon@synapse.subneural.net) |
08:19.31 | jontow | [03:16] JOIN: dersteer to #asterisk |
08:19.31 | *** mode/#asterisk [+o denon] by ChanServ |
08:19.34 | jontow | [03:18] <D1ng0> WilliamK, i got outbound BV working no inbound |
08:19.35 | jontow | [03:18] <GrimStone> D1ng0: do you get messages like "thats odd. got a response on a call ..." ? |
08:19.37 | jontow | [03:19] <PTG123> hey can some people ping sip1.way2fast.com |
08:19.40 | jontow | eh, woah |
08:19.49 | jontow | fucking table fell apart while i was pasting |
08:19.57 | jontow | never been able to claim that before |
08:19.58 | riksta | LOL |
08:20.11 | jontow | (..shit) |
08:20.23 | jontow | nonetheless, yeah, 108ms avg.. upstate NY |
08:20.40 | PTG123 | jontow: where are you located? |
08:20.48 | jontow | upstate NY :) |
08:20.53 | PTG123 | hmm |
08:20.57 | PTG123 | can you traceroute |
08:20.58 | PTG123 | that seems high |
08:21.15 | jontow | sure.. looking for number of hops or highest-latency node or other? |
08:21.28 | PTG123 | i wanna see where the extra latency comes in |
08:21.39 | PTG123 | we are going across the country so its not horrible |
08:21.41 | PTG123 | but it shoul dbe better |
08:21.47 | jontow | 13 ewr-core-01.inet.qwest.net (205.171.17.125) 29.202 ms 27.723 ms 27.052 ms |
08:21.47 | jontow | 14 tmp-core-02.inet.qwest.net (205.171.205.85) 108.552 ms 108.275 ms 107.821 ms |
08:21.53 | jontow | theres your bad node :) |
08:22.07 | PTG123 | thats probably our cross country link |
08:22.11 | PTG123 | why the 30ms before? |
08:22.19 | PTG123 | paste the whole thing i wanna see how much it bounces around |
08:22.30 | riksta | dont paste it all here |
08:22.58 | PTG123 | riskta: 10 lines of text gonna knock you off or something? :) |
08:23.12 | riksta | yes :) |
08:23.16 | jontow | http://mno.bsd.st/~jontow/tr.txt |
08:23.55 | PTG123 | what do you think tmp stands for? |
08:23.59 | PTG123 | and ewr |
08:24.03 | jontow | tempe, arizona? |
08:24.07 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
08:24.24 | jontow | (or temporary, as in man-we-fucked-up-that-router,-lets-put-this-one-in-instead-and-modify-the-bgp-tables) |
08:24.33 | PTG123 | and ewr? |
08:24.53 | jontow | east-west-route? |
08:24.57 | PTG123 | heh |
08:25.02 | jontow | (blatant guesses) |
08:25.07 | PTG123 | we need one of those visual traceroutes with a map |
08:25.23 | jontow | but you know what |
08:25.28 | jontow | i don't think that its too far off |
08:25.38 | PTG123 | which? |
08:26.07 | jontow | it goes from time warner up here, to their syracuse POP, to NY's Level 3 major hop |
08:26.22 | jontow | then it hits ewr-border, then ewr-core |
08:26.59 | PTG123 | its probably worthwhile for me to colocate some stuff on the east coast |
08:27.05 | dersteer | 64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=1 ttl=48 time=788 ms |
08:27.06 | dersteer | 64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=2 ttl=48 time=756 ms |
08:27.06 | dersteer | 64 bytes from ip-66-235-234-131.sterlingnetwork.net (66.235.234.131): icmp_seq=3 ttl=48 time=831 ms |
08:27.12 | jontow | dersteer; OUCH. |
08:27.24 | PTG123 | ok now thats bad |
08:27.29 | PTG123 | dersteer: where are you? |
08:27.33 | dersteer | michigan |
08:27.39 | PTG123 | which hop is killing us? |
08:27.39 | jontow | (...really ouch.) |
08:28.04 | PTG123 | i need to know so ic an report that |
08:28.28 | jontow | lemme try from a few other places ;) |
08:28.53 | dersteer | sorry PTG123 its me |
08:28.59 | dersteer | I got someone hitting my ftp hard |
08:29.01 | PTG123 | ah ok |
08:29.02 | PTG123 | good |
08:29.03 | PTG123 | heh |
08:29.06 | PTG123 | had me nervous |
08:29.37 | dersteer | first hop was 819.717 ms |
08:29.37 | jontow | --- sip1.way2fast.com ping statistics --- |
08:29.37 | jontow | 5 packets transmitted, 5 packets received, 0% packet loss |
08:29.37 | jontow | round-trip min/avg/max/stddev = 52.975/55.365/62.967/3.816 ms |
08:29.39 | dersteer | :p |
08:29.40 | dersteer | lol |
08:29.47 | jontow | thats from a machine of mine in arizona |
08:29.56 | PTG123 | jontow: thats horrible too |
08:29.59 | PTG123 | paste me the traceroute |
08:30.01 | PTG123 | :) |
08:30.11 | jontow | .. its a 384k business DSL line thats heavily used |
08:30.13 | jontow | thats pretty good :P |
08:30.28 | PTG123 | hah |
08:30.31 | PTG123 | considering i am in arizona |
08:30.34 | PTG123 | we shoul dbe like 20ms |
08:30.36 | jontow | ;) |
08:30.51 | dersteer | PTG123: what kind of connection u got? |
08:31.24 | jontow | well eh.. its gotta hit san jose first |
08:31.31 | jontow | (Level 3 San Jose) |
08:31.43 | PTG123 | multihomed, OC192 with cox (for cable modems who peer with all cable proviers) OC192 to qwest(for business DSL) and an OC48 from some other tier1 provider who has rthe best routes to everywhere else |
08:31.55 | PTG123 | yah i think i may bring in a l3 link |
08:31.58 | PTG123 | although |
08:32.05 | PTG123 | cox is suppose to be direct on l3 backbone |
08:32.42 | jontow | --- sip1.way2fast.com ping statistics --- |
08:32.43 | jontow | 6 packets transmitted, 6 packets received, 0% packet loss |
08:32.43 | jontow | round-trip min/avg/max/stddev = 86.716/88.657/93.570/2.400 ms |
08:32.57 | jontow | <PROTECTED> |
08:32.59 | PTG123 | jontow: where is that from? |
08:33.01 | PTG123 | ah yah |
08:33.05 | PTG123 | see thats awesome for ny |
08:33.09 | jontow | the ISP I work for |
08:33.17 | jontow | its from my desk at work actually :) |
08:33.20 | PTG123 | a voip call on that one would be pretty good |
08:33.28 | jontow | no |
08:33.31 | jontow | .. no it isn't :) |
08:33.35 | jontow | only at 3am ;) |
08:33.51 | jontow | keep in mind, this is the least utilized time for EVERYTHING i'm giving you ;) |
08:33.55 | PTG123 | hah |
08:34.00 | PTG123 | well ping times shouldn't change |
08:34.02 | PTG123 | with traffic |
08:34.06 | PTG123 | unless your network is way oversaturated |
08:34.08 | dersteer | what kinda ping time u need for good voip ? |
08:34.18 | PTG123 | probably under 150ms |
08:34.28 | jontow | yeah, it is way oversaturated |
08:34.37 | jontow | 4 T1s, all 80-90% drained all day |
08:34.39 | jontow | until about 1am ;) |
08:34.54 | PTG123 | wow |
08:34.56 | jontow | better lately though, since they upgraded to a cisco 7200 or whatever :) |
08:34.59 | PTG123 | voip would suck through that connection |
08:35.03 | jontow | yes.. yes it does |
08:35.07 | dersteer | wow jontow |
08:35.11 | jontow | its scratchy as shit and breaks up all the time :( |
08:35.12 | dersteer | what kind of place u work for? |
08:35.40 | PTG123 | a place thats too cheap to buy a decient internet connection |
08:35.41 | PTG123 | heh |
08:35.41 | jontow | they're oversold as far as im concerned.. and working on a 10mbit DANC fiber connection, as well as multi-homing with verizon fiber as well |
08:36.05 | jontow | so its gonna get better; but the problem is that the bandwidth just isn't available to smaller companies in this area |
08:36.18 | jontow | it isn't their fault.. but they haven't exactly been doing enough to correct it, either.. |
08:36.30 | PTG123 | why not get a ds3 |
08:36.48 | jontow | single connection isn't a good idea up here |
08:36.52 | jontow | unreliable routing :) |
08:37.00 | jontow | very rural area |
08:37.07 | PTG123 | you connect to a datacenter |
08:37.10 | PTG123 | then multihome the bw yourself |
08:37.16 | PTG123 | private loop ds3 you control |
08:37.20 | jontow | if you don't have multiple uplinks, you're fucked up here.. |
08:37.40 | jontow | only way you could do that here is with verizon, and they're so unresponsive that its unreal.. |
08:37.41 | jontow | :o |
08:37.59 | PTG123 | makes me appreciate living in phoenix :) |
08:38.16 | *** join/#asterisk coppice (~chatzilla@96.196.17.210.dyn.pacific.net.hk) |
08:38.17 | jontow | we had a cross-county circuit drop on a friday.. a week later, on the next friday, the trouble ticket was finally processed by verizon and they called us back |
08:38.21 | jontow | .. we cancelled the circuit |
08:39.35 | dersteer | I'm hoping to have good luck picking a carrier.... I'm working on starting a wireless ISP here |
08:39.50 | dersteer | I'm in a remote rural area too |
08:40.08 | coppice | who here uses a variety of different ATA boxes? |
08:41.32 | jontow | we're in the foothills of the adirondack mountains, long-haul wireless is a giant pain in the ass with so many obstructions :/ |
08:42.40 | PTG123 | i talked to a guy with a 80 mil wireless network |
08:42.46 | PTG123 | all based on cheap wireless hardware |
08:43.13 | dersteer | thats what I'm working on |
08:43.19 | dersteer | the equipment really is cheap :) |
08:43.23 | PTG123 | well |
08:43.30 | PTG123 | you can even use wrt54gs |
08:43.37 | PTG123 | if you keep them less then 5 miles appart |
08:43.45 | PTG123 | he uses a used laptop and cheap wireless cards |
08:43.49 | PTG123 | and he gets 8mile links |
08:44.01 | jontow | damn yeah :D |
08:44.03 | PTG123 | his antennas costed him $35 |
08:44.11 | PTG123 | i was impressed :) |
08:44.14 | PTG123 | running openwrt |
08:44.33 | jontow | nice. |
08:45.06 | PTG123 | i guess its time for bed for me |
08:45.16 | jontow | hey |
08:45.20 | jontow | http://mno.bsd.st/~jontow/tr.txt |
08:45.22 | jontow | hit that one more time |
08:45.24 | dersteer | where can I find a wrt54gs? |
08:45.27 | jontow | its a paste of all the traceroutes i've done |
08:45.30 | dersteer | one that is the right version? |
08:45.59 | dersteer | I've had a hard time finding the good wrt54g versions |
08:46.47 | dersteer | I'm going to be using devices called SBC |
08:46.57 | dersteer | single board computer |
08:47.06 | dersteer | for the AP |
08:47.12 | jontow | hey |
08:47.15 | jontow | http://www.soekris.com |
08:47.16 | dersteer | don't know what I wanna use for customers |
08:47.17 | jontow | :) |
08:47.32 | dersteer | I want something cheap for the customers |
08:47.46 | jontow | SBCs for about $250 each, with case and power supply ;) |
08:47.58 | dersteer | ouch |
08:48.11 | dersteer | http://www.wisp-router.com/index.php?cPath=39&osCsid=d1e9e3e5daa2c298d98923242c978c71 |
08:48.22 | jontow | no, no.. they're heavy on the features |
08:48.28 | jontow | you can get them down to about $150 each i think |
08:48.46 | jontow | varying features, guaranteed compatible with netbsd and freebsd and linux and such :) |
08:49.14 | dersteer | cool |
08:49.30 | dersteer | I'm more worried about something for the client side |
08:49.34 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
08:49.39 | dersteer | would like to use something like the wrt54g |
08:49.42 | jontow | AP or just router? |
08:50.54 | dersteer | I want it to act as a client... taking in the wireless... then routing it to the customers machines via ethernet |
08:53.57 | [cc]smart | dersteer: afaik, openwrt has a ser router package |
08:54.19 | dersteer | hmmm |
09:12.37 | *** join/#asterisk meshugga (philip@loeblich.linuxteam.at) |
09:12.41 | meshugga | hi |
09:12.54 | cjk | apparently some peopole got * working on openwrt |
09:13.04 | meshugga | yesterday i solved the one-way audio problem with chan_bluetooth |
09:13.17 | meshugga | now i have the problem that asterisk doesnt take the call when my phone is ringing |
09:13.42 | meshugga | anybody here who has clue about how asterisk expects a module to route a call before taking the call? |
09:13.56 | dersteer | really cjk? |
09:14.07 | coppice | meshugga: do tell. what did you do to get the audio to work? |
09:14.21 | meshugga | coppice: hehe, i knew somebody would be interested ;P |
09:14.27 | meshugga | actually it is just a three line patch |
09:14.45 | meshugga | get_buffer always returns 0, so asterisk thinks it gets null frames |
09:15.01 | meshugga | the trick was to teach get_buffer to return the actual len of the frame |
09:15.24 | meshugga | recompile module -> works |
09:16.00 | coppice | OK, so the 3 lines are....... |
09:16.13 | meshugga | but yet i dont know how to post that to the ml, im not getting my subscription |
09:16.34 | coppice | just post it here. its only 3 lines |
09:17.23 | meshugga | get_buffer(char * dst, char * ring, int ring_size, int * head, int to_copy) |
09:17.23 | meshugga | { |
09:17.23 | meshugga | <PROTECTED> |
09:17.23 | meshugga | <PROTECTED> |
09:17.31 | meshugga | <PROTECTED> |
09:17.31 | meshugga | <PROTECTED> |
09:17.32 | meshugga | <PROTECTED> |
09:17.32 | meshugga | <PROTECTED> |
09:17.36 | meshugga | ehm |
09:17.37 | meshugga | fuck |
09:17.40 | meshugga | wait |
09:18.20 | meshugga | http://rafb.net/paste/results/3B1XWl66.html |
09:18.29 | meshugga | this is how the function is supposed to look like |
09:18.33 | meshugga | as said, just three more lines |
09:19.15 | meshugga | you can kick that ast_log of coz, thats quite noisy |
09:19.55 | meshugga | so now anybody here who can point me some documentation about asterisk module coding? |
09:20.09 | meshugga | because |
09:20.22 | meshugga | actually, that chan_bluetooth is a bit messy altogether |
09:20.26 | meshugga | nice functionality though |
09:21.35 | meshugga | coppice: you will tell me if it worked for you? and if your asterisk takes the call? |
09:21.46 | meshugga | i wonder how this is supposed to work anyway |
09:21.49 | *** join/#asterisk ibr (~ibr@dsl-082-083-197-172.arcor-ip.net) |
09:21.53 | coppice | i don't know of any documentation. the usual ercommendation is to take a simple channel and follow it. the channel I wrote still has a couple of quirks because I haven't probed enough to figure out what it should do. |
09:22.14 | ibr | Hi! |
09:22.21 | meshugga | coppice: which chan did you write? |
09:22.47 | coppice | chan_unicall, which interfaces my unicall protocol modules like MFC/R2 to * |
09:23.07 | meshugga | ah, ok, dont know them |
09:27.03 | coppice | if you were in a country that needs MFC/R2 you would :-) |
09:27.21 | meshugga | is that like BRI? |
09:27.32 | coppice | not even remotely :-) |
09:27.37 | meshugga | hehe |
09:28.04 | coppice | its a weird old tone signalling system used over E1s in many countries |
09:28.19 | meshugga | ah |
09:28.23 | meshugga | in line signalling? |
09:28.28 | meshugga | this ccitt-5 thing? |
09:28.33 | coppice | no |
09:28.54 | coppice | its a mix of in band tones and separate line signals |
09:29.52 | meshugga | i c |
09:30.02 | meshugga | i accept that "weird" ;) |
09:30.35 | meshugga | btw, what is that abbreveation "CIND" supposed to mean? |
09:35.41 | coppice | meshugga: Control INDicator |
09:35.44 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
09:35.51 | meshugga | does what? |
09:36.00 | *** join/#asterisk jhiver (~jhiver@ABesancon-102-1-2-7.w80-11.abo.wanadoo.fr) |
09:36.09 | jhiver | hi everybody |
09:36.17 | coppice | it reports thing like signal strength and battery state on GSM phones |
09:36.52 | meshugga | ah, i c |
09:36.54 | meshugga | thanks# |
09:39.51 | cjk | anyone here who has more detials on the disposition fields in cdr records? disposition=4 means what exactly? |
09:40.28 | coppice | i think disposition is "grumpy" :-) |
09:41.04 | cjk | grumpy, well it can be important. i have a special case on my systems. |
09:41.23 | GrimStone | well disposition is a plain text field |
09:41.44 | cjk | GrimStone, yeah, but if you use odbc it just logs an integer |
09:42.07 | GrimStone | open the odbc source and see then |
09:42.22 | GrimStone | using pgsql it logs in plain text |
09:43.33 | *** join/#asterisk skraps (~mike@c-24-0-190-239.client.comcast.net) |
09:44.09 | skraps | anyone here? |
09:44.48 | skraps | wow, 261 in the channel and nobody is here. that must be a new record. |
09:45.13 | cjk | GrimStone, ok whats the disposition you see the most in your records? that must be 4 in mycase |
09:46.03 | *** part/#asterisk skraps (~mike@c-24-0-190-239.client.comcast.net) |
09:48.09 | tzafrir | <PROTECTED> |
09:50.03 | *** join/#asterisk jjg (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
09:50.05 | jjg | hi |
09:50.31 | jjg | has anyone been successful in using SIP to establish video calls through *? |
09:51.09 | ibr | not even for audio yet :) . |
09:51.18 | jjg | heh, hang in there |
09:51.35 | ibr | have you done that part? |
09:51.35 | jjg | ibr / what's the issue? |
09:51.52 | ibr | I defined two users in sip.conf. |
09:52.05 | ibr | They can register on the server. |
09:52.30 | ibr | But when they call each other, the server responds with 404. |
09:52.40 | ibr | I enabled sip debug. |
09:52.44 | jjg | what's 4040, don't remember |
09:52.46 | jjg | 404 |
09:52.54 | ibr | User not found. |
09:53.18 | jjg | have you made a context extension to dial the sip user? |
09:53.24 | jjg | i think that is necessary |
09:53.43 | ibr | I'm not sure. |
09:53.47 | ibr | Everything I did was: |
09:54.14 | GrimStone | is the current CVS in a working state ? |
09:54.26 | ibr | [user1] type=friend host=dynamic |
09:54.32 | ibr | The same for the second one. |
09:54.42 | ibr | How do I define a context extension? |
09:55.08 | jjg | exten => 1000,1,Dial(sip/username,20,tm) |
09:55.26 | ibr | Where does it go? |
09:55.34 | jjg | exten => 1000,2,Hangup |
09:55.43 | jjg | then if yo ucall 1000, it rings username for 20 seconds |
09:56.04 | ibr | extensions.conf? |
09:56.06 | jjg | yah |
09:56.20 | jjg | so you should create an extension for each sip user |
09:56.25 | ibr | Aha. Let me try... |
09:56.52 | jjg | i'm not sure if it's absolutely necessary but that is how i do it...and my server isn't online for me to look right now |
09:57.40 | GrimStone | is the current CVS at 1.1.0 level , or at 1.0.6 ? |
09:58.02 | jhiver | hey guys |
09:58.26 | jhiver | do you know some URL where there is a Asterisk + SER tutorial? |
09:58.51 | ibr | jjg: Hmm, as I understand, 1000 is the "called number". |
09:59.08 | ibr | jjg: But I have two kphone clients, user1 and user2. |
09:59.21 | ibr | jjg: Do I still need extension definitions? |
09:59.26 | GrimStone | jhiver: try the voip-info wiki ? |
10:01.04 | ibr | jjg: Or should I write "exten => user1,1,Dial(sip/user1,20,tm)"? |
10:02.40 | ibr | BTW, how do I complete a user name ("jjg: ...")? I'm using ircII. |
10:03.23 | jjg | ibr : i think extensions have to be numberic |
10:03.26 | jjg | numeric |
10:04.09 | ibr | jjg: Ok, I'll try that first. I should call 1000@myserver, right? |
10:04.29 | Faithful | Anyone the zaphfc thing??? |
10:04.35 | ibr | jjg: Do you also use two software clients, or phones? |
10:06.30 | *** join/#asterisk jeofrey (~jeofrey@202.160.45.29) |
10:06.35 | jeofrey | hi |
10:06.58 | jeofrey | anyon can help me please in MySQL problem in Fedora Core |
10:07.17 | jeofrey | ERROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) |
10:07.33 | jeofrey | how to solve that error |
10:08.24 | Faithful | jeofrey: how are you trying to connect? |
10:08.42 | Faithful | might be that you are not listening on TCP socket |
10:08.54 | jeofrey | so what do i have to do |
10:09.07 | jeofrey | because i am first time user of myqsl |
10:09.10 | jeofrey | mysql |
10:09.24 | jeofrey | i just want to use in asterisk |
10:09.30 | jeofrey | to record our calls |
10:09.59 | Faithful | uncommnet "port = 3306" in my.cnf |
10:10.39 | jeofrey | where i can get the my.cnf |
10:11.10 | Faithful | If you are running linux... where all your cnf type files are |
10:11.19 | jeofrey | ahhh ok |
10:11.21 | ibr | jjg: Yes, this worked! Thank you :) ! |
10:11.22 | jeofrey | i will |
10:11.38 | ibr | jjg: But the client immediately died :( . |
10:11.52 | ibr | See you all! |
10:12.18 | jeofrey | try to locate |
10:12.45 | jeofrey | im not sure also the location..... new user of asterisk |
10:14.51 | ta[i]nted | i have a question |
10:15.06 | ta[i]nted | when a user dials an invalid number using Dial() |
10:15.17 | ta[i]nted | and the provider says 'invalid number blah blah' |
10:15.24 | jeofrey | usr/share/mysql/ |
10:15.30 | jeofrey | i cannot find also inside there |
10:15.32 | ta[i]nted | get_Variable(DIALSTATUS) still returns 'ANSWERED' |
10:15.51 | ta[i]nted | how can i get the true disposition of the call? |
10:16.53 | ta[i]nted | i guess technically the provider 'answered the call'.. but the call shouldn't register as a successful call |
10:17.27 | Faithful | jeofrey: /etc/mysql/my.cnf |
10:17.46 | Faithful | jeofrey: locate my.cnf |
10:18.17 | jeofrey | okkk Faithful |
10:20.01 | coppice | meshugga: hey! chan_bluetooth seems to be working! |
10:20.17 | meshugga | of course |
10:20.26 | meshugga | i wonder why this bug was such a stupid one |
10:21.41 | meshugga | and /now/ i'd like to have incoming calls working |
10:21.46 | meshugga | but i still dont know where to start hehe |
10:22.04 | coppice | I just called my headset and talked both ways |
10:22.18 | meshugga | yeah, that works |
10:22.30 | meshugga | and dialling out on a S55 works too |
10:23.09 | coppice | so what doesn't work? |
10:23.19 | meshugga | but there seems to be a timing problem left, with both, headset and phone, sometimes there is just noise on the line |
10:23.30 | meshugga | coppice: incoming calls on the phone arent taken by asterisk |
10:24.10 | coppice | i don't follow you. you just agreed that calls to the headset work |
10:24.17 | meshugga | and i dont find the point in the module where this is supposed to happen |
10:24.34 | meshugga | coppice: yes, but you can also use your mobilephone with that module |
10:24.39 | meshugga | and thus dial out, right? |
10:24.54 | meshugga | this works, too |
10:25.05 | coppice | er, no. the dialing out thing is phone dependant |
10:25.11 | meshugga | but what doesnt work is, when the phone has an /incoming/ call, from the GSM provider |
10:25.21 | meshugga | coppice: well, i told you, it works with my S55 |
10:25.33 | meshugga | and it also worked with an ericsson t630 |
10:25.43 | coppice | so, it works with that one. it doesn't work generally |
10:25.51 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-236.arcor-ip.net) |
10:26.02 | meshugga | well, it should work generally if the phone has bluetooth headset support |
10:26.04 | meshugga | but anyway |
10:26.15 | meshugga | in my setup, dialling out works |
10:26.24 | Blackvel | do I have to use this line: ;deny=216.207.245.47/255.255.255.255 for iax tel incoming calls? |
10:26.29 | coppice | you mean you expect an incoming GSM call to get routed to *? |
10:26.33 | meshugga | but chan_bluetooth doesnt know what to do with the incoming call |
10:26.37 | meshugga | coppice: yes |
10:27.09 | meshugga | this should work too, the "RING" is coming, also "+CLIP", but chan_bluetooth somehow messes it up to set the line to "ringing" |
10:27.20 | coppice | meshugga: bluetooth doesn't seem to standardise digits from a headset, or to a headset for hat matter |
10:27.42 | meshugga | ? |
10:28.14 | jeofrey | Fatihful i search inside of root/etc but i cannot find the mysql directory inside... |
10:28.38 | jeofrey | is it inside of that directory?? |
10:28.40 | meshugga | coppice: -v? |
10:28.56 | coppice | but you are right that if chan_bluetooth acts as a headset it should be able to answer and handle GSM calls. |
10:29.55 | meshugga | as far as i understood some postings on the mailinglist, there /is/ this functionality already implemented in chan_bluetooth |
10:30.07 | meshugga | and it should route the call to extension "s" in context [bluetooth] |
10:30.17 | meshugga | but woe is me, i cant find that ;) |
10:30.50 | meshugga | AH |
10:30.51 | meshugga | i have |
10:30.55 | meshugga | found it |
10:30.57 | coppice | i thought so, but i never tried. i only tried to use a headset before |
10:31.56 | meshugga | it would rock to be possible to dialin |
10:32.18 | meshugga | then it would be possible to do some nice least-cost-routing setups with the cost of some handsets |
10:32.20 | coppice | I tried another bluetooth headset, and that also works now. |
10:32.28 | jerlique | has anyone got any flash operator panel screen dumps, with lots of extensions?? |
10:32.30 | meshugga | coppice: can you give me the models which work? |
10:32.57 | meshugga | the one which works here is a logitech hs01, jabra bt200 doesnt work though :'/ |
10:33.12 | coppice | A Sony Ericsscon HBH-660 and an Omiz OM8055 |
10:33.17 | meshugga | thanks |
10:33.34 | GrimStone | anyone have "Got a response on a call we dont know about" errors with Broadvoice ? |
10:33.46 | Faithful | meshugga: :-( I have a bt200 |
10:34.05 | The_Ball | is there any demo FWD or IAX number? |
10:34.18 | The_Ball | to test the iaxtel setup |
10:34.33 | Faithful | You can try me |
10:34.47 | Faithful | but that's no guarentee |
10:35.06 | GrimStone | the call starts but asterisk doesn't generate proper CDR's or hangup the call when its done |
10:35.09 | meshugga | Faithful: i will let you know when i made it work |
10:35.13 | coppice | meshugga: a number of people are looking for this fix. can you post it on the mailing list? |
10:35.17 | The_Ball | what's your number |
10:35.29 | meshugga | i would like to have the jabra working too, since it is more comfortable to carry |
10:35.40 | meshugga | coppice: i didnt get the ack from the ml for my subscription |
10:35.42 | Blackvel | anyone knows how to configure sarp.sourceforge.net with asterisk on the same server? feel free to msg me. Do I have to use different SIP/RTP ports? what do I have to do with the router port forwarding? twice? how do I configure asterisk to use SARP? |
10:35.42 | meshugga | :/ |
10:35.59 | coppice | would you like me to post it? |
10:36.22 | Faithful | 17005672555 |
10:36.23 | meshugga | well, yes why not |
10:36.25 | coppice | "meshugga says... " :-) |
10:36.28 | meshugga | let me generate a patch |
10:36.45 | Faithful | meshugga: so are you using a BT dongle on the PC? |
10:38.20 | coppice | Faithful: I am. A CSR based one |
10:38.27 | meshugga | Faithful: USB, yes |
10:38.37 | meshugga | some noname shit |
10:38.49 | *** join/#asterisk ranliv (~ranliv@210.5.85.249) |
10:38.50 | meshugga | coppice: but actually, i'd like to put some docu also in it |
10:39.03 | meshugga | e.g. the s55 needs "sdptool add hf" |
10:39.10 | meshugga | and then there is some timing problem |
10:39.50 | meshugga | so just give me time until tomorrow, then i will try to post it |
10:40.03 | meshugga | people have been waiting months now, they can wait until tomorrow ;P |
10:41.06 | The_Ball | Faithful, anything happening? |
10:42.13 | coppice | meghugga: fixing this bug makes you a good guy. documenting things makes anyone a superhero :-) |
10:42.46 | meshugga | lol |
10:43.45 | Faithful | meshugga: so you have blue tooth headsets working with softphones in linux??? |
10:46.03 | Faithful | Nope |
10:46.20 | The_Ball | Faithful, did you get anything inbound? |
10:48.15 | Blackvel | what is the best service you can get for the moment, for a germany/UK/US flatrate? |
10:48.29 | Faithful | The_Ball: Try again |
10:48.35 | The_Ball | Faithful, i get: -- Executing Dial("Zap/1-1", "IAX2/user:password@iaxtel.com/17005672555@iaxtel") in new stack -- Called user:password@iaxtel.com/17005672555@iaxtel -- Hungup 'IAX2/69.73.19.178:4569/3' == No one is available to answer at this time |
10:49.18 | The_Ball | hmmm, this time i got: Mar 6 20:42:41 WARNING[19867]: chan_iax2.c:5546 socket_read: Call rejected by 69.73.19.178: No authority found |
10:49.45 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
10:50.03 | Faithful | The_Ball CODEC? |
10:50.14 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
10:50.21 | The_Ball | Faithful, allow=all in iax.conf |
10:50.39 | The_Ball | and im using a TDM card and a analog phone |
10:51.58 | ckruetze | good morning |
10:52.01 | meshugga | Faithful: no, with asterisk |
10:52.16 | meshugga | but i guess bt headsets can work with the softphones if you use that alsa-bluez stack |
10:53.00 | meshugga | also, i found some other code which is called "handsfree", this put my softphone successfully on the stereo / my soundcard |
10:53.07 | Faithful | meshugga: so how do you dial... or um? what does it do? |
10:53.14 | coppice | meghugga: as you said, things are still not reliable. sometimes I get good sound in the bluetooth headset, but the other end gets an awful noise. |
10:53.21 | meshugga | Faithful: Dial(BLT/nameoftheheadset) |
10:53.39 | meshugga | coppice: i told you, /sometimes/ there is a timing problem |
10:53.43 | meshugga | but it is not always the case |
10:53.48 | Faithful | oh cool so you can answer |
10:53.55 | meshugga | if you repeat, the noise will go away |
10:54.03 | meshugga | i experienced noice in 1/3 of all cases |
10:54.13 | meshugga | same when using a handset |
10:54.23 | Faithful | how do I specify the codec on outgoing iax2 calls |
10:54.29 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
10:54.58 | coppice | meshugga: it seems to be less for me. I was starting to think I had something reliable :-) , but no. I get the same result as you. is the noise always in the same direction? |
10:55.02 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
10:55.32 | meshugga | coppice: appaerently |
10:55.53 | meshugga | i guess this has to do with the starting of sco_thread |
10:55.59 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
10:56.23 | meshugga | if you find a way to reproduce the problem reliantly, pls msg me or write to philip@linuxteam.at |
10:56.34 | meshugga | if i can reproduce it reliably, i can debug it too ;) |
10:56.36 | Faithful | meshugga: how do you pair the device? |
10:56.41 | coppice | meshugga. i don't think so. I found i can call and get silence. when I start to speak it seems like the very first sound I make gets repeated forever |
10:56.45 | meshugga | Faithful: rtfm ;) |
10:56.57 | Faithful | which asterisk? |
10:57.07 | meshugga | Faithful: this is not ready to deploy yet |
10:57.10 | coppice | Faithful: the documentation on bluez sucks badly, but there is some around |
10:57.22 | meshugga | if you want to play with it, you need to get in the docs on the net |
10:57.49 | Faithful | it would be so cool for here at work |
10:58.05 | Faithful | I neet to buy a dongle at last I guess |
10:58.07 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
10:59.43 | The_Ball | Faithful, still nothing? i get no error now, i fixed the username/password |
10:59.50 | meshugga | im off, going hiking |
11:01.28 | coppice | meshugga: something * locks up completely |
11:02.34 | meshugga | coppice: thats the nature of bluetooth ;P |
11:03.07 | Faithful | The_Ball: do you have a 1700 number? |
11:03.26 | The_Ball | Faithful, yes, but im behind a bad firewall |
11:03.29 | meshugga | well anyway, i am away now, pls query me with anything |
11:04.00 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
11:04.37 | *** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
11:05.36 | The_Ball | Faithful, but i found on voip-info.org Echo test: 17009999613 |
11:08.47 | The_Ball | Faithful, it't probably that damn xp box that is sharing the dial-up connection which is making trouble. when i get "broadband" in a couple of days and use the linux machine it should work |
11:09.18 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
11:09.24 | ranliv | guys why is that when i type zap show channels i get No such command 'zap' (type 'help' for help) |
11:09.46 | ranliv | when i type help zap command cannot be sen |
11:10.27 | djin | are the zap channels running? |
11:10.45 | djin | ztcfg -vvv |
11:10.47 | ranliv | yes doing ztcfg -vv |
11:11.10 | djin | what is the bottom line of that output? |
11:11.13 | ranliv | 4 channels configure |
11:12.11 | djin | did you restart asterisk already |
11:13.22 | Faithful | Am I dreaming??? is it possible to use my Nokia 6310i as a telephone extension? |
11:13.58 | coppice | Faithful: if chan_bluetooth works properly it should be possible |
11:14.01 | ranliv | yes 2x |
11:14.23 | nextime | and if chan_bt don't work you can try also "miax" |
11:14.52 | Faithful | That is just so cool... so I will have to buy one of those 100 meter bt dongles ! |
11:15.01 | ranliv | also reinstalled zaptel many times |
11:15.25 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
11:17.34 | Zeeek | do voicemail boxes have to be numeric? |
11:18.13 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
11:18.32 | djin | Zeeek, I wouldn't know why they should be. |
11:19.45 | Zeeek | only because you couldn't easily dial them from a phone, |
11:20.02 | Zeeek | as in "mailbox" ? |
11:20.06 | Zeeek | prompt |
11:20.06 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
11:20.17 | MuppetMaster | Hello. |
11:20.23 | Zeeek | hi djin and mup |
11:20.34 | MuppetMaster | Does anyone know how to get h.263 working from behind NAT (ie - endpoints behind NAT, Asterisk on public IP)? |
11:22.05 | ranliv | I get this warning doing make in zaptel" *** Warning: "zt_register" [/root/zaptel/ztdynamic.ko] has no CRC!" |
11:22.11 | ranliv | is this just ok |
11:22.24 | ranliv | or will i be concerned |
11:24.02 | coppice | nexttime: miax looks interesting. I hadn't heard of that before |
11:24.42 | *** join/#asterisk in (int@stackhack.net) |
11:25.00 | MuppetMaster | Anyone on h.263? |
11:25.26 | MuppetMaster | I was able to do FWD -> FWD with two endpoints behind the same NAT, but using the FWD nat proxy. |
11:26.00 | MuppetMaster | I have enabled verify=yes and nat=yes in sip.conf, but to no avail. The g729/g711 work fine, but then when I try to launch the video from within Eyebeam nothing happens. |
11:26.47 | file | ranliv: did you recompile asterisk afterwards so it compiles the zaptel stuff? |
11:27.53 | *** join/#asterisk mh- (~mh@202.5.145.13) |
11:28.26 | coppice | i wonder why miax has chosen to use the world's worst DTMF detector ;-) |
11:28.52 | *** join/#asterisk darby_t (mua@dmx189.neoplus.adsl.tpnet.pl) |
11:31.45 | ranliv | no? i do make for pwlib, openh323, asterisk, then make install for asterisk-oh323, asterisk.. after that on zaptel I make radfw.h , make linux26 then make install |
11:32.11 | file | you have to recompile asterisk so it knows to compile the zaptel stuff now |
11:32.32 | ranliv | after compi;ing zaptel do i need to recomplie asterisk? |
11:32.53 | file | yes |
11:35.47 | *** part/#asterisk darby_t (mua@dmx189.neoplus.adsl.tpnet.pl) |
11:47.36 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
11:49.18 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
11:51.22 | *** join/#asterisk hemant (hemant@220.226.24.114) |
11:53.10 | The_Ball | anyone her get gnophone to work with asterisk? |
11:57.35 | The_Ball | hm, do I have to enable something to make asterisk listen for iax connections? Asterisk is not listening to port 5036 at the moment |
11:57.56 | file | IAX2 uses port 4569 |
11:58.25 | file | the old old IAX which you must never use, is 5036 |
11:58.38 | The_Ball | file, but in iax.conf it says port=5036 |
11:58.44 | The_Ball | oh |
11:59.15 | The_Ball | should i make a config file called iax2.conf ? |
11:59.33 | file | no |
11:59.39 | file | you can't change the port on IAX2 |
11:59.47 | file | that entry in iax.conf is for the old IAX |
11:59.57 | The_Ball | so i should just remove that part |
12:00.03 | file | just leave it be... it'll do no harm |
12:01.52 | The_Ball | file, is iax2 backwards compatible? gnophone says it's made for iax not iax2 |
12:02.02 | file | nope |
12:02.06 | file | don't use gnophone, it's old... |
12:02.19 | The_Ball | is there any other good ones? |
12:02.34 | file | Google is your best hope |
12:06.27 | ranliv | guys why do i get this "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
12:06.54 | ranliv | before it was not there but when i rebooted my pc there it was |
12:07.17 | ranliv | is it hardware conflict? |
12:07.21 | file | the devices are not present in /dev? |
12:07.56 | ckruetze | file Have fun at VON, I wish it would be cheaper. |
12:08.57 | ranliv | how do i go about this recompile zaptel? asterisk? |
12:09.40 | file | ranliv: find out if your distribution is using udev, if so you need to read the instructions on it |
12:09.44 | file | ckruetze: thx |
12:09.54 | file | 'nor get dressed |
12:11.23 | ckruetze | file: If you find a cool, new VoIP phone, bring one for me, that is if nobody would be missing it. |
12:11.25 | ranliv | i'm using fedora core 2 |
12:11.38 | file | ckruetze: haha |
12:12.23 | file | ranliv: dunno |
12:12.46 | file | ranliv: Google. |
12:13.05 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
12:13.09 | yaboo | hi all getting this error under asterisk with both soft phones being xlite |
12:13.11 | yaboo | Mar 6 23:12:10 WARNING[6443]: dsp.c:1469 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833 |
12:13.24 | file | yaboo: use rfc2833, not inband |
12:13.35 | yaboo | thanks file |
12:13.57 | file | did you actually read the warning? <- note the word, warning |
12:14.14 | file | I should take a poll |
12:15.55 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
12:19.10 | file | I *must* not freak out |
12:20.47 | file | Peter Packet! |
12:21.13 | yaboo | file sounds sounds so crap thou |
12:21.43 | coppice | as in "Peter Packet picked a peck of pickled pepper"? |
12:22.42 | file | sure! |
12:23.40 | file | Penny, and Byte? |
12:25.20 | Zeeek | Why do I get this message? |
12:25.20 | Zeeek | Apr 1 13:45:10 WARNING[6466]: phones.c:bt100 ast_phone_process: Get a life! Your Grandstream can't do RFC2833! Use INFO, idiot! |
12:25.35 | Zeeek | what does it mean? |
12:25.54 | file | I should force myself to get out of bed |
12:26.05 | coppice | I think its a phoney message :-) |
12:26.06 | Zeeek | too much information |
12:26.22 | Zeeek | coppice when can I fax you with my bad fax? |
12:26.52 | Zeeek | it's a problem during initial negotiation |
12:26.58 | Zeeek | loops in the dialogue |
12:27.02 | coppice | I don't need any bad faxes. I have some since I started T.38 development :-) |
12:27.13 | Zeeek | isn't this an odd problem tho? |
12:27.19 | Zeeek | or is it common? |
12:27.47 | *** join/#asterisk memic (~memic@dsl-084-056-106-237.arcor-ip.net) |
12:28.12 | coppice | many people send and receive large volumes with no trouble. almost everything that goes wrong these days is something specific to the installation |
12:28.23 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.135) |
12:28.58 | Zeeek | well, this fax software has never had a problem sending to any other machine. I thought it would be of interest to see why it loops when calling spandsp |
12:29.35 | coppice | I am. |
12:29.36 | Zeeek | I think I mentioned that j2.com sends me a beautiful zero distortion fax |
12:29.37 | MuppetMaster | Has anyone used XTunnels with Asterisk? |
12:30.04 | Zeeek | so at least that cool! |
12:30.12 | Zeeek | (zero distortion 3 pages of fax) |
12:30.40 | Zeeek | spandsp must work with some fax software, because spam faxes seem to get through no problem |
12:30.48 | *** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com) |
12:31.54 | Zeeek | my Sunday cappucino is ready. thanks for all the phish |
12:32.56 | MuppetMaster | Does anyone know how to use ICE (http://www.voip-info.org/wiki-ICE) in conjunction with Asterisk? |
12:33.43 | file | asterisk doesn't support it |
12:35.13 | mh- | we're going to setup asterisk shortly for a small demo -- implementing it for a call center in 4 months or so; thought this would be a good spot to sit and listen |
12:35.31 | mh- | so far so good :) |
12:37.00 | mh- | does anyone have it running on sles 9? a resident asterisk guy we've hired for some reason had issues with it -- got it up and running on rh advanced server |
12:42.58 | Zeeek | sles9? |
12:44.56 | yaboo | anyone using usb-audio under linux |
12:47.27 | mh- | Zeeek, suse enterprise server 9 |
12:47.37 | Zeeek | ok |
12:48.03 | Zeeek | never heard of anyone using suse here, but there must be somewhere |
12:49.31 | Zeeek | so mh- how's it working? |
12:50.02 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
12:51.56 | mh- | Zeeek we haven't got it up as yet -- put it up for an in-house demo on redhat advanced server with no problems |
12:52.24 | Zeeek | what does "advanced server" mean in this context? |
12:52.57 | RoyK | windows nt advanced server :P |
12:53.09 | Zeeek | ~lart RoyK |
12:53.13 | mh- | yeah, exactly that :) |
12:53.23 | Zeeek | there wasn't any such thing! |
12:53.38 | Zeeek | Windows 200 Advanced server, yes |
12:54.19 | RoyK | Zeeek: there was a 4.0 NTAS |
12:54.32 | RoyK | I was an MCSE at that time |
12:55.17 | Zeeek | but they kicked you out? |
13:10.24 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
13:12.10 | *** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp) |
13:12.23 | Blackvel | do I have to configure asterisk for the sip proxy SaRP? |
13:12.41 | Blackvel | is there any sip module which transparently rewrites the SIP header? |
13:12.56 | Blackvel | I do not want (can't) configure the asterisk server for anything |
13:13.49 | Zeeek | what is the problem you want to solve, Blackvel? |
13:14.34 | Blackvel | NAT |
13:14.42 | Blackvel | asterisk sends out the wrong ips, even with externip |
13:14.55 | Blackvel | I can not upgrade to cvs head where I could use the syntax variable externhost |
13:15.12 | *** join/#asterisk file2 (~jcolp@mctn1-142166194173.nb.aliant.net) |
13:15.15 | Blackvel | so I need some application which rewrites the SIP UDP 5060 packets |
13:15.16 | file2 | okay I'm packed |
13:15.35 | file2 | if anybody wants to hear me at the airport, I'll be on from there in the conference in the URL |
13:15.39 | Zeeek | Blackvel I use asterisk with NAT on both sides |
13:16.46 | RaYmAn-Bx | Blackvel: I've heard of it not sending out the externip ip unless localnet is also setup |
13:16.59 | Blackvel | file2: how do you do this? |
13:17.09 | file2 | there's free wifi at the airport, and I have a wifi phone |
13:17.12 | file2 | plus I have a cellphone |
13:17.22 | Blackvel | RaYmAn-Bx: that is interesting. I have now configured localnet too |
13:17.32 | Blackvel | i mean okay, even FWD works with my asterisk behind NAT setup |
13:17.37 | Blackvel | but german GMX seems not |
13:17.52 | Blackvel | i think I'll try to debug sip again |
13:17.54 | Blackvel | and have a luck |
13:17.59 | Zeeek | Blackvel there's no reason why all those SIP things won't work |
13:18.06 | file2 | I think I have everything... |
13:18.15 | Blackvel | file2: wifi phone thats cool |
13:18.22 | RaYmAn-Bx | NAT can provide some weird errors SIP...I had huge issues getting sipgate to work with my (highly broken) NAT router |
13:18.25 | Blackvel | any good quality? how much does that wifi cost at the airport? |
13:18.37 | Blackvel | sipgate (with port forwarding) works quite well |
13:18.41 | file2 | the wifi at the airport is free |
13:18.44 | file2 | and the phone itself is great |
13:18.58 | Blackvel | sipgate, nikotel, fwd, iaxtel (before) worked great |
13:19.00 | Blackvel | but not gmx |
13:19.02 | Blackvel | dunno why |
13:19.05 | Blackvel | i have one way audio |
13:19.13 | Blackvel | file2: what wifi phone? |
13:19.24 | bpoint | I'm having some weird problems with asterisk's handling of rtp... anyone up for a few questions? :) |
13:19.28 | file2 | Hitachi WIP-5000 |
13:19.42 | Blackvel | IAX2/guest@66.250.68.194/996 |
13:19.45 | Blackvel | you are online at this url? |
13:19.54 | file2 | not right now, I can be though |
13:20.46 | Blackvel | how do you handle all this different extensions with your callphone (or maybe even the WEP/WPA different keys or public hotspot authentication)? |
13:20.51 | Blackvel | wifiphone |
13:21.12 | file2 | it'll automatically associate to the nearest signal |
13:21.22 | file2 | and if it needs WEP it'll tell me ask me to enter the key or whatever |
13:22.46 | file2 | I leave in 2 hours 23 minutes |
13:24.32 | Blackvel | can you configure several fixed extensions and locations? |
13:24.47 | *** join/#asterisk mesi (~raoul@dsl-082-083-143-041.arcor-ip.net) |
13:24.49 | file2 | what do you mean? it just communicates with my asterisk machine... |
13:24.59 | Blackvel | like you if can create multiple extensions (short dailing numbers) and different profiles |
13:25.02 | Blackvel | oh it does |
13:25.04 | Blackvel | how clever |
13:25.10 | file2 | it's a regular SIP phone |
13:25.13 | file2 | except it uses wifi for the signal |
13:25.22 | Blackvel | do you configure every extension in extensions.conf? |
13:25.28 | file2 | yes |
13:25.36 | Blackvel | i am fed up adding all this direct ip calls to extensions.conf :) |
13:25.40 | file[atVON] | bbl |
13:26.28 | file[atVON] | I shall be on the conference when I can be btw |
13:26.37 | mesi | When somebody calls asterisk and immediately hangs up, without a connection being established at all, can asterisk call back the given callerid and offer a menu? |
13:27.16 | Zeeek | mesi if it gets the callerid, ya |
13:27.45 | mesi | Zeeek: It gets. But I have no clue how to make it call back. |
13:27.58 | Zeeek | look up call files on the wiki |
13:28.10 | mesi | Zeeek: Ok, thanks. |
13:28.30 | Zeeek | or just use a wait() and then dial the number back |
13:28.46 | bpoint | anyone have any idea why I might be getting a continuous stream of "NOTICE[7086]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum" notices from asterisk? |
13:29.00 | Zeeek | Dial(Technology/${CALLERIDNUM},time, options) |
13:29.17 | Blackvel | anyone used siproxd with asterisk? |
13:29.25 | Blackvel | how can I configure asterisk for this? |
13:29.48 | Blackvel | is anybody using broadvoice? |
13:30.21 | GrimStone | broadvoice is presently screwed , Blackvel |
13:30.33 | Zeeek | Blackvel would this be of interest? http://lists.digium.com/pipermail/asterisk-users/2004-April/044626.html |
13:30.39 | GrimStone | they did something yesterday which has pretty much broken asterisk |
13:31.40 | mesi | Zeeek: Yes, that might work. If it is like the context is executed even when no connection has been established. |
13:31.49 | Blackvel | oh it is? |
13:31.59 | Blackvel | that 20$ for everything in the world is so interesting |
13:32.08 | Blackvel | ohoh |
13:32.15 | Zeeek | mesi if a call comes in, it drops into the context. You do whatever you want after that |
13:32.18 | GrimStone | yeah i got 1 of those plans |
13:32.21 | Zeeek | inlcudinf NOT answering it |
13:32.26 | GrimStone | worked great until yesterday |
13:32.39 | Blackvel | i talked to someone with broadvoice (twice asterisk) two weeks ago |
13:32.41 | Zeeek | why does everyone want BV ? Are they way cheaper than the rest? |
13:32.52 | Blackvel | well 20$? |
13:33.00 | Blackvel | what other providers do you know? |
13:33.05 | Zeeek | or maybe they have more local DID? |
13:33.18 | Zeeek | where are you, USA? |
13:33.19 | GrimStone | Zeeek: they got real unlimited |
13:33.23 | Blackvel | no |
13:33.24 | Blackvel | me germany |
13:33.32 | Zeeek | you want to call USA? |
13:33.33 | Blackvel | and I am thinking to get a german flatrate |
13:33.48 | Zeeek | are most of your calls to US? |
13:33.49 | Blackvel | but if I would get a flat for the same price, and I could also call UK/USA |
13:33.52 | Blackvel | that would be even better |
13:33.57 | Blackvel | no germany |
13:34.00 | Zeeek | I use several providers |
13:34.07 | Blackvel | but I want to change things |
13:34.21 | Zeeek | here in France we now have a national unmimited SIP for 8eu/mo |
13:34.32 | Zeeek | I guess Germany should too |
13:34.35 | Blackvel | I don't really want to have 3 providers that would be something like 3x20$ :) |
13:34.53 | Zeeek | I only pay 8eu§mo for ten providers |
13:35.03 | Zeeek | all the rest are pay as you go |
13:35.08 | Blackvel | flat? |
13:35.12 | Zeeek | (not unmimited) |
13:35.15 | Blackvel | ah |
13:35.15 | Zeeek | no |
13:35.18 | Blackvel | per mintue |
13:35.23 | Blackvel | minute |
13:35.24 | Zeeek | but are you sure you need unlimited? |
13:35.27 | Zeeek | we don't |
13:35.40 | Blackvel | no i am not |
13:35.52 | Zeeek | like you could use pay as you go for international |
13:35.54 | Blackvel | I just know my german costs for a familay are usaually more than 30 Euros per month |
13:35.57 | Zeeek | it's very cheap |
13:36.00 | Blackvel | arcor german flatrate is 20EUR |
13:36.03 | Blackvel | but without US/UK |
13:36.18 | Blackvel | maybe its enough to use for UK/USA per minute billing, but who knows |
13:36.19 | Zeeek | does sipgate.de do unlimited or very cheap German calls? |
13:36.36 | Zeeek | well you need to know about how many minutes you'll use per month, ya? |
13:36.41 | Blackvel | very cheap is the word |
13:36.45 | Blackvel | you can buy 8,90EUR |
13:36.50 | Blackvel | 1000 minutes |
13:36.50 | Zeeek | and with asterisk, you can count up the seconds |
13:36.51 | ranliv | I got it working, I just transfered the card to a diffrent PCI slot |
13:36.56 | Blackvel | thats 0,89EUR cent |
13:37.10 | Blackvel | Zeeek: i am not too sure |
13:37.21 | Blackvel | I think it was 15 hours or something |
13:37.24 | Blackvel | I would have to check again |
13:37.35 | Blackvel | but my monthly costs are more than 20EUR |
13:37.39 | Zeeek | like I said, for family use in this country we have 8.00eu§month for the whole country |
13:38.05 | The_Ball | what exactly happens on "Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996" ? |
13:38.25 | Zeeek | a bunch of devs talk for three hours and you listen |
13:39.20 | Blackvel | btw |
13:39.32 | Blackvel | i have the urgent request for a voip traffic calculator |
13:39.38 | Blackvel | i am trying to find a good url on voipinfo |
13:39.44 | Blackvel | is tehre anything like this? |
13:39.59 | Zeeek | you mean bandwidth vs calls? |
13:40.19 | file[atVON] | eek we aren't leaving yet |
13:40.23 | Blackvel | he needs to know how far to come with 1 gIG for g711/g726 |
13:40.45 | Blackvel | file let me call you |
13:40.51 | Zeeek | statr a call and measure the bw used on the asterisk box |
13:40.58 | file[atVON] | I don't have the phone on, conserving battery |
13:41.03 | Blackvel | ah ok |
13:41.14 | Zeeek | file is waiting for Mr. Right |
13:41.23 | Blackvel | i dont care :) |
13:41.25 | file[atVON] | file is waiting to go to the airport |
13:41.35 | Zeeek | better than waiting for Godot |
13:41.36 | Blackvel | where are you flying to |
13:41.38 | Blackvel | ? |
13:41.39 | file[atVON] | so he can board his flight to Toronto |
13:41.42 | file[atVON] | then to Chicago |
13:41.44 | file[atVON] | then to San Jose |
13:41.48 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
13:41.50 | file[atVON] | so he may be at VON, http://www.von.com/ |
13:42.33 | Blackvel | ah |
13:42.37 | Blackvel | cool |
13:42.44 | Blackvel | how expensive is that trip then? |
13:43.04 | file[atVON] | for normal people? expensive |
13:43.09 | Zeeek | not to be confused with http://www.vons.com |
13:45.11 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
13:45.26 | coppice | how much for weirdos, then? :-) |
13:45.48 | file[atVON] | and this is my *first* time flying |
13:46.54 | coppice | the important thing to ensure is its not the pilot's |
13:47.08 | Blackvel | file: are you normal or not? what is a normal person btw? :) |
13:47.27 | file[atVON] | a normal person is someone who doesn't work for a VoIP company, and is not an asterisk developer |
13:47.46 | Blackvel | oh |
13:47.49 | Ron-Na | Has anybody SJphone working registered to an Asterisk box? |
13:47.51 | Blackvel | digium is sponsering it |
13:47.52 | Blackvel | great |
13:47.52 | Blackvel | :) |
13:48.06 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
13:48.07 | file[atVON] | no they aren't, they have a pavilion though... the... Asterisk Pavilion! |
13:48.20 | Blackvel | how do you chat for the moment? laptop and wifi? |
13:48.28 | file[atVON] | I'm at home on my workstation |
13:48.34 | file[atVON] | we haven't left to go to the airport yet |
13:48.38 | Mw3 | goddamn i'm a normal people :( |
13:48.39 | Blackvel | ahhhh |
13:48.48 | file[atVON] | I'm trying to think if there is anything I Forgot to pack |
13:48.51 | Blackvel | i need someone "un-normal" |
13:49.04 | Blackvel | to back port externhost into asterisk 1.0.6 |
13:49.22 | GrimStone | Unable to create channel of type 'SIP' (cause 3) <== what does this mean ? |
13:49.22 | file[atVON] | all my power adapters... |
13:49.30 | *** join/#asterisk afe ([snGf3oJZj@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
13:49.35 | Blackvel | 2nd battary pack? |
13:49.47 | Blackvel | pda? |
13:49.57 | Blackvel | your VIP pass ticket? |
13:50.08 | Blackvel | first call ticket? |
13:50.16 | Blackvel | class |
13:50.31 | *** part/#asterisk ranliv (~ranliv@210.5.85.249) |
13:50.33 | Blackvel | who runs openwrt + siproxd + asterisk voip? |
13:50.53 | file[atVON] | well, I have my plane ticket |
13:50.59 | file[atVON] | and my pass for VON is... well, in San Jose |
13:51.29 | Blackvel | where do you live? how long do you have to fly to san jose? |
13:51.41 | file[atVON] | it'll take me 16 hours |
13:51.50 | file[atVON] | cause I live in Canada on the other side of the country |
13:52.11 | Darwin35 | File come home we miss you |
13:52.18 | file[atVON] | ha |
13:52.21 | Darwin35 | Toronto is calling you back |
13:52.21 | file[atVON] | I haven't left yet |
13:52.26 | Darwin35 | hehhe |
13:52.28 | file[atVON] | and I don't live in Toronto :p |
13:52.37 | file[atVON] | thank god I don't live in Montreal |
13:52.42 | file[atVON] | I purposely booked this flight just to avoid Montreal |
13:52.50 | Darwin35 | http://www.von.com/schedule_wifi5.htmlwait your a frenchie right |
13:52.55 | Darwin35 | grrr |
13:52.59 | file[atVON] | nope |
13:53.02 | file[atVON] | I'm english :p |
13:53.06 | file[atVON] | muahahahaha |
13:53.09 | Darwin35 | i hate cut and paste |
13:53.14 | file[atVON] | Moc is french, and Junky is french |
13:53.30 | Darwin35 | <---is scottish Canuck |
13:53.43 | Blackvel | can I configure asterisk to use siproxd? |
13:54.13 | coppice | Darwin35: did you head to canada for warmer weather? :-) |
13:54.27 | file[atVON] | I'm heading to San Jose for warmer weather, that's for sure |
13:54.30 | file[atVON] | I can actually... wear shorts! |
13:54.57 | Darwin35 | I wear shorts and kilts all year round |
13:55.12 | file[atVON] | Darwin35, get pneumonia every year too don't you |
13:55.25 | Darwin35 | nope |
13:55.31 | Darwin35 | healthy as a horse |
13:55.34 | file[atVON] | uh huh a likely story |
13:56.06 | Darwin35 | I have 6 utility kilts 4 family kilts and 7 pairs of shorts |
13:56.11 | coppice | a horse as in the expression "flogging a dead *****" |
13:56.22 | file[atVON] | I must soon go |
13:56.30 | file[atVON] | very very soon |
13:56.35 | file[atVON] | everyone wish me luck |
13:56.55 | Darwin35 | I wanted to go to Von but the company would not give me the time off or pay for it |
13:57.06 | file[atVON] | be back later, maybe from... an airport... or something |
13:57.07 | Darwin35 | have fun for me |
13:57.13 | file[atVON] | maybe cafe... I dunno |
13:57.33 | file[atVON] | oh, and if bkw, twisted, paulc, drumkilla, or kram show up tell them I'm on my way |
13:57.34 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
13:57.54 | ckruetze | Darwin35: you are not the only who can't go to VON :( |
13:57.59 | Darwin35 | you all must be sharing a room |
13:58.14 | \Grooby\ | VON? |
13:58.27 | Darwin35 | lets see if they setup a asterisk box for chat from there |
13:58.35 | Darwin35 | www.von.com |
13:58.38 | coppice | can you really wear shorts in san hose now? its north of here, and i'm bloody cold right now. |
13:58.58 | \Grooby\ | ahhh ok |
13:59.00 | Darwin35 | I wear them all year round |
13:59.05 | Darwin35 | I have thick blood |
13:59.23 | Darwin35 | but I wear flannel boxers to keep the boys warm |
14:00.36 | Darwin35 | we no do regamental in the winter |
14:00.50 | YoYo | ok, so what's new in asterisk land? can I send mp3 attachments for voicemail yet? |
14:01.03 | Darwin35 | and in the summer in the office we have to wear a jock company rules |
14:01.26 | Darwin35 | read the wiki it tells all |
14:02.38 | YoYo | bah... yer no fun |
14:02.57 | YoYo | and neither is that wiki... it's overloaded with disorganized info |
14:03.35 | Darwin35 | then go organize it |
14:03.46 | Darwin35 | put yourself to some use |
14:03.54 | Darwin35 | help klean the place |
14:06.32 | Blackvel | hm |
14:06.44 | Blackvel | siproxd runs without cahgning asterisk configuration? |
14:06.55 | Blackvel | there is simply no sip outgoing proxy gateway option in asterisk |
14:06.57 | Blackvel | only chansip2 |
14:07.02 | Blackvel | not sure how smooth it runs |
14:08.08 | mesi | Is there a Dial command wich continues execution after a connection has been established? |
14:09.07 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
14:09.58 | Blackvel | what do you want to do? |
14:11.25 | mesi | I want to Dial out to somebody and put him into a certain context as if had dialed in to my * box |
14:11.47 | Blackvel | i dont know what you mean |
14:12.27 | mesi | Well, if somebody dials in to my box, he is in a special context. |
14:12.54 | mesi | I on the other hand, want asterisk to call out and if the called party picks up the phone, it should be like he had called in to my asterisk box. |
14:13.34 | mesi | Unfortunately, Dial() will stop executing the current context. So if I use Dial() to dial out then the called party wouldn't have a context executed and I cannot offer a menu or the like. |
14:13.50 | GrimStone | whats wrong if i Dial on a SIP channel and can hear the other side fine , but asterisk keeps saying "Got a response on a call we dont know about." |
14:15.00 | Blackvel | mesi: hm...I have never done this. maybe AGI is the way to go |
14:15.02 | GrimStone | and the CDR records show the call as "NO ANSWER" , even though it got connected |
14:15.46 | roamer323 | mesi - call out, once pickedup - transfer to an extension -> the 'outside guy' will get context of extension (for dialplan etc) |
14:16.20 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
14:16.49 | mesi | Blackvel: I'll check that, thanks. |
14:17.26 | mesi | roamer: Ok, but the outside guy would have to do this transfer, right? |
14:17.33 | mesi | That's not comfortable. ;-) |
14:17.52 | Blackvel | YES YES YES STRIKE STRIKE STRIKE |
14:18.00 | Blackvel | asterisk 1.0.6 and GMX works now |
14:18.13 | Blackvel | even without the externhost but only externip feature |
14:18.40 | mesi | Blackvel: You mean, GMX internet telephony? |
14:19.00 | Blackvel | yeah! |
14:19.08 | Blackvel | there had been a problem in asterisk 1.0.2 |
14:19.18 | Blackvel | dunno what is was, something with sending out NAT ip even it shouldnt |
14:19.38 | mesi | well, I should register with gmx, too. Do you have to pay a fee? |
14:19.47 | Blackvel | no |
14:19.53 | Blackvel | costs are 1ct |
14:20.08 | Blackvel | you have to register with gmx and give your landline number |
14:20.25 | roamer323 | mesi - actually , you can do the transfer without the 'outside guy' knowning any diff - but it is a kludge :-) |
14:20.28 | Blackvel | they will send you a bill each month and pay your bank |
14:22.38 | Blackvel | so what has broadvoice changed? |
14:22.43 | Blackvel | that asterisk is not working anymore? |
14:24.56 | mesi | roamer323: So how does this kludge work? ;-) |
14:25.16 | Blackvel | hm |
14:25.20 | Blackvel | if you call the person |
14:25.27 | Blackvel | is it really clever to show up a IVR menu? |
14:25.30 | Blackvel | an |
14:26.22 | mesi | Blackvel: So I would not be able to register with GMX for free to do internet telephony? |
14:26.46 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
14:26.51 | roamer323 | blackvel - I think mesi is doing a callback - and call through (LD savings from remote part of the world?) the outcalled party can probably blind-dial |
14:26.51 | mesi | roamer: I can use M(mymacro) and call Transfer in this macro? |
14:27.23 | roamer323 | mesi - yes - or the manager interface with an outside monitoring script (big kludge) |
14:27.42 | Blackvel | mesi: you can |
14:27.47 | Blackvel | but pstn calls cost something |
14:27.51 | Blackvel | as other providers do |
14:27.51 | mesi | roamer: Yes, if I use my answering machine, I would make people angry when using their phone ;-) |
14:28.16 | mesi | roamer : so I have to make my asterisk call me back on an outside pstn. |
14:28.40 | mesi | Blackvel: ah, I see. |
14:28.45 | *** join/#asterisk massivexb (~mirc@HSE-Toronto-ppp300289.sympatico.ca) |
14:29.03 | massivexb | hey folks what operating system will be the best to install asterisk onto? |
14:29.45 | mesi | roamer: The macro is better. It would be a one - line - macro. |
14:29.56 | roamer323 | mesi - I agree :-D |
14:30.01 | mesi | massivexb: any free unix or derivate. Linux, FreeBSD or the like. |
14:30.14 | mesi | roamer: Ok :-)) |
14:30.35 | Blackvel | is there some changeliste of asterisk 1.0.6? |
14:30.44 | Blackvel | change list |
14:30.55 | Blackvel | suddenly NAT IPs are working |
14:31.14 | massivexb | cool thanks.. was going to install on freebsd but wasnt sure of compatibility :) |
14:31.26 | *** join/#asterisk sudhir492 (~sudhir@4.7.58.171) |
14:31.34 | sudhir492 | hi all |
14:31.38 | massivexb | does anyone know the digium products well? |
14:31.50 | sudhir492 | Anyone using a quad T1/E1 card here |
14:32.04 | sudhir492 | massivexb: what do you want to know |
14:32.40 | massivexb | looking at gettign a 24 channel pri installed wanted to know what card to get to interface with it |
14:32.56 | Nukemizer | Do any Readhat users know where I can get the kernel-source ? I downloaded and installed FC3 but canno find the kernel-source on the CD's or through googling |
14:32.58 | sudhir492 | for US right |
14:33.03 | massivexb | canada |
14:33.08 | massivexb | probably same as us |
14:33.12 | sudhir492 | yes |
14:33.19 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
14:33.19 | *** mode/#asterisk [+o bkw_] by ChanServ |
14:33.32 | bkw_ | yo yo yo |
14:33.35 | bkw_ | guess file is on his way |
14:33.58 | D1ng0 | bkw_, where ya going |
14:34.02 | Mw3 | von ... |
14:34.07 | bkw_ | VON |
14:34.27 | sudhir492 | I have been using T1 cards for about a year. They work well |
14:34.36 | D1ng0 | hey bkw_ what VOIP provider you using ? |
14:34.51 | sudhir492 | newer card TE110P should work well too. I have not used them yet |
14:34.59 | bkw_ | D1ng0, um |
14:35.01 | bkw_ | i'm a provider |
14:35.01 | massivexb | are you running them on freebsd? |
14:35.04 | bkw_ | asterlink.ccom |
14:35.14 | bkw_ | er asterlink.com |
14:35.15 | bkw_ | haha |
14:35.17 | bkw_ | double c's |
14:35.31 | sudhir492 | bkw_: do you use quad T1/E1 card at all? |
14:35.41 | bkw_ | yes |
14:35.57 | bkw_ | why oh why do you ask? |
14:36.20 | D1ng0 | cause BroadVoice has literally screwed alot on people inbound calls are not working |
14:36.35 | sudhir492 | I need to prepare a system for that which can get pretty busy sometimes |
14:37.11 | sudhir492 | I want to know a good combination of hardware that can take significant load, |
14:37.32 | bkw_ | well for that my friend you'll either have to pay someone |
14:37.35 | bkw_ | or learn like we did |
14:37.37 | bkw_ | :P |
14:37.44 | bkw_ | that information is worth its weight in gold |
14:38.07 | sudhir492 | bkw_: If I had gold, I wouldnt be here :-( |
14:38.27 | bjohnson_ | silver? |
14:38.31 | bjohnson_ | diamonds? |
14:38.43 | sudhir492 | bkw_: how much do a few megabytes weigh anyway? :-) |
14:39.12 | bkw_ | no clue |
14:39.19 | sudhir492 | bkw_: what did you learn? :-) |
14:39.25 | D1ng0 | even platinum |
14:39.31 | bkw_ | what not to do |
14:39.40 | massivexb | gold pressed platinum :) |
14:39.53 | massivexb | what os are you unning bkw |
14:40.46 | GrimStone | whats wrong if i Dial on a SIP channel and can hear the other side fine , but asterisk keeps saying "Got a response on a call we dont know about." |
14:40.54 | bpoint | would anyone have any idea why asterisk would be spewing "NOTICE[7210]: rtp.c:451 ast_rtp_read: RTP: Received packet with bad UDP checksum" messages at me? :/ |
14:41.02 | GrimStone | and the CDR records show the call as "NO ANSWER" , even though it got connected |
14:41.22 | bkw_ | ok guys |
14:41.25 | bkw_ | i'm putting my laptop away |
14:41.28 | bkw_ | and waiting on my flight |
14:41.30 | bkw_ | HAVE FUN |
14:41.32 | bkw_ | see you in denver |
14:41.34 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
14:43.01 | Blackvel | denver? |
14:43.07 | Blackvel | what does bkw do in denver? :) |
14:43.56 | mesi | When executing the Dial() M Macro, I get the error: Could not stop autoservice on calling channel. |
14:47.21 | Nukemizer | Do any Readhat users know where I can get the kernel-source ? I downloaded and installed FC3 but cannot find the kernel-source on the CD's or through googling |
14:48.26 | *** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz) |
14:49.00 | dfunnell | Hi, anyone online who can help regarding CAPI (namely dialing out using CAPI)? |
14:49.51 | Mw3 | ~bluetooth |
14:49.52 | jbot | it has been said that bluetooth is at http://www.handhelds.org/z/wiki/Is%20anyone%20working%20on%20Bluetooth%20for%203870 or at http://dnsv6.iihe.ac.be/iPAQGPRSv6/BtConfig.html .. and you might wanna take a look at http://bluez.sourceforge.net/contrib/HOWTO-Mobile-Phone |
14:50.09 | Mw3 | hm |
14:50.15 | Mw3 | where is the bluetooth channel driver ? |
14:51.19 | Blackvel | Contact: <sip:<mynumber>@213.23.34.236> |
14:51.20 | Blackvel | cool |
14:51.25 | Blackvel | the sip header is fine with 1.0.6 |
14:51.39 | Blackvel | hmm bluetooth |
14:51.40 | Blackvel | for pda? |
14:51.46 | Blackvel | do you try to use voip? |
14:52.40 | Mw3 | no, somebody said here that i can use my mobile phone as a handset with chan bluetooth and an usb dongle |
14:52.46 | Mw3 | i'd like to try |
14:53.23 | cbachman | Nukemizer, I remember reading something to the effect that with one of the FCx releases they started only releasing it as a SRPM |
14:54.20 | Nukemizer | cdbachman: thank you I will get those CD's ( big help thanks much) |
14:54.56 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
14:56.12 | tzafrir | Nukemizer, if it's not in a separate package, get the SRPM of the kernel and only build it up to the "patch" stage (rpmbuild -bp) |
14:57.59 | ckruetze | Mw3: Try http://tribble.crazygreek.co.uk/content/chan_bluetooth |
14:59.02 | Nukemizer | tzafrir, thanks i will try |
14:59.26 | tzafrir | rpm -i kernel-something.srpm |
14:59.48 | tzafrir | rpmbuild -bp ~/RPM/SOURCES/kernel.spec |
15:00.00 | tzafrir | (you should build RPMs as non-root) |
15:00.12 | *** join/#asterisk Tarox (someone@pD9E7B5D4.dip.t-dialin.net) |
15:02.17 | Nukemizer | tzafrir, thank you for guidance :) this helps |
15:05.27 | dfunnell | Hi all - trying to dial-out via CAPI, but keep getting message 'didn't find capi device with outgoing msn = xxx', where 'xxx' is the MSN number I'm trying to use. Have spent two days trying every possible combination of MSN numbers, local telco (Telecom NZ) provided MSN gives same error. Can anyone help? |
15:07.39 | *** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp) |
15:09.49 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
15:10.15 | Mneumonic | hey anyone know why on my sipura 841 it takes like 10 seconds after i dial an internal extension to connect to it? |
15:11.37 | sudhir492 | I have a Polycom phone behind a firewall that registers fine with the Asterisk and make calls too. However, I am not able to reach it from outside. Anyone has a suggestion for that? I do have nat=yes and canreinvite=no in sip.conf |
15:11.49 | bjohnson_ | anyone understand traceroute? trying to track down a slow ping problem and traceroute keeps giving me asterisks instead of data |
15:12.01 | RaYmAn-Bx | Mneumonic: assuming the sipura 841 works anything like Sipura SPA-2000, it means you are lacking a dialplan entry for the numbers you dial.. (i.e. if internal numbers are 6 digits you need a setting that tells it to accept 6 digit numbers..) |
15:12.36 | *** part/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp) |
15:12.49 | sudhir492 | Mneumonic: also try entering '#' after the digits |
15:12.53 | Mneumonic | RaymAn - you mean the dialplan on the phone itself? |
15:12.56 | RaYmAn-Bx | bjohnson_: that means it doesn't get a reply usually.. |
15:13.01 | RaYmAn-Bx | Mneumonic: yes |
15:13.25 | RaYmAn-Bx | and as sudhir492 says it will prolly do it immediately if you type # after the number |
15:13.45 | *** join/#asterisk bpoint (~bpoint@cn220.opt2.point.ne.jp) |
15:13.56 | Mneumonic | yea, it does... |
15:14.02 | Mneumonic | this is the phone's dialplan |
15:14.04 | Mneumonic | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
15:14.14 | Mneumonic | i have 3 digit extension |
15:14.45 | Mneumonic | so the first batch of x's should only be 3? |
15:16.02 | bjohnson_ | does a nat router port need to be forwarded to get traceroute to work? |
15:16.11 | RaYmAn-Bx | only add to it..but yeah, you just need to add an xxx part |
15:17.09 | RaYmAn-Bx | bjohnson_: not afaik, but it requires a few things...Generally traceroute uses UDP, so it relies on getting certain replies about the attempted UDP send...it could either mean that the servers refuse to send these or that they simple doesn't reach you |
15:17.32 | *** join/#asterisk jsolares (~jsolares@200.12.33.64) |
15:17.40 | bpoint | traceroute uses icmp, not udp |
15:17.44 | RaYmAn-Bx | try with traceroute -I (depending on the version of traceroute you have) and reach the manpage for more information |
15:17.44 | sudhir492 | Mneumonic: there is an ambiguity for 3 digit extensions in the dialplan, hence your best bet is to dial a '#' indicating end of dialing. Another approach can be to have 3 digit extensions *xxx that way you will have dial a star in the beginning but it will resolve the ambiguity |
15:18.23 | bpoint | most servers filter icmp packets for security reasons |
15:18.48 | bjohnson_ | -I did the trick |
15:18.56 | RaYmAn-Bx | bpoint: it depends on the implementation. The traceroute on my server uses udp. (unless you tell it to use ICMP with -I) |
15:19.11 | bjohnson_ | what do the three ms times mean? |
15:19.12 | Mneumonic | sudhit - thanks :) |
15:19.22 | RaYmAn-Bx | there are also implemenations using TCP |
15:19.24 | Mneumonic | err sudhir |
15:19.37 | bpoint | rayman: that's interesting... first for me :) |
15:21.38 | RaYmAn-Bx | bpoint: I believe it sends a UDP packet to a port nothing is expected to listen on..it sends it with a ttl so low that each router on the way sends back an ICMP TIME EXCEEDED reply, hence getting the addresses and response times for each router on the way |
15:21.47 | sudhir492 | anyone here an idea why I can only call from a phone not receive calls there? They register with Asterisk fine. |
15:21.50 | RaYmAn-Bx | (as far I can understand..Not sure it's entirely accurate) |
15:23.02 | jsolares | sudhir492: is it behind nat? |
15:23.05 | bpoint | rayman: hmm.. sounds like it would work the same as icmp |
15:23.22 | bpoint | I've just never personally seen a traceroute use udp tho :) |
15:24.43 | RaYmAn-Bx | bpoint: the difference is that routers happily send out ICMP errors :) Sure, they might block ICMP ECHO's (ping), but it would be silly to block ICMP errors |
15:26.53 | *** join/#asterisk Ron-Na (~ronald@203.70.36.126) |
15:27.28 | sudhir492 | jsolares: yes it is behind nat and I have nat=yes and canreinvite=no in sip.conf |
15:27.48 | jsolares | try with qualify=yes as well |
15:28.03 | Ron-Na | Has anybody setup a softphone on a PDA? I tried SJphone, but it does not register, ... any hints? |
15:28.12 | sudhir492 | thx. I will try. what does qualify=yes mean? |
15:28.13 | jsolares | can you atleast receive calls like a second after it registers? (it should) |
15:28.38 | jsolares | sudhir492: that'll it'll poke the device from time to time to keep the connection open and up |
15:30.11 | sudhir492 | jsolares: maybe. Because, once, and only once the phone rang when called. :-( |
15:30.36 | jsolares | hope and pray that your phone isnt one of the ones that die if poked via qualify=yes then |
15:30.55 | jsolares | it will work, that's how i have the phone in my house behind a nat connecting to the * box in the office |
15:34.16 | sudhir492 | jsolares: I set qualify=yes, and on reload I see the message that the extension is no longer reachable |
15:34.39 | bpoint | is there anyone out there that has to disable rtp checksums? |
15:35.03 | sudhir492 | jsolares: It is a Polycom 500, hence I presume that the phone should be stable. |
15:35.38 | sudhir492 | Unfourtunately, I cannot physically verify that as the phone is 40 miles away from my home right, with no one in the office |
15:35.57 | RaYmAn-Bx | bpoint: I ended up disabling it because I got a fair amount of checksum errors, without any problems with sound or anything |
15:36.43 | bpoint | rayman: after ~0.5sec into a call, asterisk is just spewing out udp checksum errors... :/ |
15:37.01 | Zeeek | sudhir492 some phones (and some providers) can't qualify |
15:37.20 | Zeeek | don't ask me why |
15:37.21 | Blackvel | who knows freenet and gmx? |
15:37.35 | bpoint | it only happens when dialing out remotely or when a call comes in. local extensions have no problems (weird) |
15:37.54 | RaYmAn-Bx | bpoint: do you have any sound problems or anything? |
15:38.47 | bpoint | once the udp checksum errors start, the downstream audio gets dropped (internet -> asterisk) |
15:39.19 | bpoint | I don't think this is an asterisk problem, though... |
15:39.25 | RaYmAn-Bx | How is it with rtp checksums off? |
15:39.38 | RaYmAn-Bx | No, It sounds like a bad connection (or a dumb router that breaks things) |
15:39.43 | bpoint | no different :) it still shows the errors too |
15:40.00 | *** join/#asterisk Voip_Help_Me (TheJudge@196.46.66.98) |
15:40.00 | bpoint | it could very well be this crappy router |
15:40.10 | Voip_Help_Me | hello all |
15:40.18 | sudhir492 | jsolares: it seems that qualify=yes did the trick. I had to restart asterisk though. Now when I call this extension, it goes to voicemail after 20 seconds, whereas another extension without qualify=yes goes to voicemail right away ! |
15:40.19 | vaewyn | ok... so who's gonna be in San Jose tomorrow? :} |
15:40.24 | psywar | Is it normal for the SPA-2000 to gen UDP packets with bad checksums? |
15:40.33 | psywar | they're cutting off my calls. |
15:40.33 | bpoint | since it works fine on the inside of the network.... hmm |
15:40.44 | Zeeek | this seems to be a bad checksum day! |
15:40.49 | Voip_Help_Me | does any one know how to priopritse voip packets ? usign sip phones and have cisco infrastructure ? |
15:40.53 | bpoint | psywar: bad checksums for you too? |
15:40.56 | bpoint | :) |
15:40.58 | psywar | yeah |
15:41.08 | bpoint | I've been poking through rtp.c, but there's nothing out of place |
15:41.26 | bpoint | the kernel would have to be returning EAGAIN from the recvfrom() call anyway |
15:41.29 | Zeeek | someone put a backdoor in the checksum code and now they're all sabotaged muhahaha |
15:41.29 | bpoint | *sigh* |
15:41.37 | psywar | I replaced a homemade cat5 cable with a professional one, haven't had another yet. |
15:41.45 | psywar | But I havne't been on it that much |
15:41.49 | bpoint | zeeek: the kernel is doing the checksums, afaik |
15:41.54 | RaYmAn-Bx | rtp checksums shouldn't be the same as udp checksums though? |
15:41.59 | bpoint | yeah |
15:42.12 | bpoint | er? they are, aren't they? :) |
15:42.28 | Zeeek | something just occurred to me: I saw some of those yesterday; I just switched to 1.0.6 - what vers are you guys using? |
15:42.29 | bpoint | rtp is over udp |
15:42.33 | RaYmAn-Bx | What's the point in naming them entirely different then? |
15:42.47 | RaYmAn-Bx | yeah, but isn't it simple udp checksums then? If that's all it is |
15:42.50 | bpoint | zeeek: cvs head as of friday :) |
15:42.59 | Zeeek | I got 1.0.6 stable friday |
15:43.01 | bpoint | rayman: that's what it should be |
15:43.06 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net) |
15:43.28 | psywar | I'm on 1.0.5 |
15:43.41 | bpoint | 1.0.6 didn't compile for me |
15:43.53 | bpoint | so I had to get cvs head... :/ |
15:43.59 | Ron-Na | Is anybody using SJphone to connect to * ? |
15:44.05 | psywar | UDP suggests either the SPA-2000 is messed up or something is mangling packets |
15:44.05 | vaewyn | ouch... bus error on my linux sparc box... on a simple IAX2 auth |
15:44.21 | psywar | ~docs |
15:44.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:44.30 | Voip_Help_Me | anyone know sip priopirty for a cisco router ? |
15:44.32 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net) |
15:44.58 | dfunnell | Hi all - trying to dial-out via CAPI, but keep getting message 'didn't find capi device with outgoing msn = xxx', where 'xxx' is the MSN number I'm trying to use. Have spent two days trying every possible combination of MSN numbers, local telco (Telecom NZ) provided MSN gives same error. Can anyone help? |
15:48.29 | psywar | I just realized I could make a digit on my "local extension" context go to my "dial-in welcome" context, so I can stop calling my own number on the cell now. |
15:48.32 | psywar | for debugging |
15:49.00 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
15:49.23 | psywar | Does anyone have any theories or guides on mapping out your contexts? |
15:49.46 | ManxPower | Keep them simple enough to easily understand. |
15:50.20 | psywar | Mine are like spaghetti code |
15:50.29 | ManxPower | psywar: Then simplify them. |
15:50.35 | bpoint | I gotta get to bed... I'm gonna poke with this rtp stuff later *sigh* |
15:50.49 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@adsl-69-212-48-186.dsl.sfldmi.ameritech.net) |
15:50.50 | sudhir492 | jsolares: qualify=yes makes the phone more reachable. However it is still not foolproof |
15:51.21 | sudhir492 | is there some kind of keepalive that need to be set on the phone? |
15:51.23 | ManxPower | For example all incoming calls from untrusted sources come into the [incoming] context. From that context I include [extensions] and that's about it. |
15:51.28 | ariel_ | Morning all |
15:51.29 | psywar | what are the different valid states? I saw an error recently about not having an "h" state. What is that? Never heard of "h", just "s" and digits. |
15:51.42 | vaewyn | h is hangup |
15:52.04 | RaYmAn-Bx | sudhir492: if it's only a single phone (or a single asterisk) behind the NAT and you have access to it, forwarding the needed ports would prolly help |
15:52.05 | ManxPower | psywar: You must have been running in debug modde. "h" is called when the far(?) end hangs up. |
15:52.08 | Zeeek | psywar /dial |
15:52.12 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
15:52.21 | psywar | cool ty |
15:52.46 | sudhir492 | RaYmAn-Bx: Actually there are multiple phones |
15:54.01 | RaYmAn-Bx | in that case you could set them up to use different ports for the phones (i.e. 5061 for second phone, 5062 for third, etc) and do static mapping to them...It's not exactly a plug and play solution (and it doesn't really work with dynamic ips) |
15:54.08 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr) |
15:55.08 | PoWeRKiLL | Hi |
15:55.33 | PoWeRKiLL | I have a strange sip error the same peer suddenly got a 401 unauthorized |
15:56.03 | sudhir492 | RaYmAn-Bx: Yes, in this case making a static mapping like that creates more problems that solves. |
15:56.12 | sudhir492 | than it solves |
15:56.37 | PoWeRKiLL | after day and day of good connection I trace sip packet and see the 401 error but password are correct on both side |
15:57.24 | Blackvel | anyone has some workaround for this broadvoice / asterisk problem yet? |
15:57.24 | RaYmAn-Bx | sudhir492: in that case try and increase the nat timeout on the router |
15:57.27 | sudhir492 | any idea what causes this kind of behavior? I suspect either the router in the middle or the phone is dropping the NAT connection |
15:57.54 | Blackvel | i mean how can it be that the call goes to voicemail instead of to asterisk? |
15:58.05 | Darwin35 | yes my fbsd-asterisk -cf drive works |
15:58.19 | Darwin35 | 512 meg |
15:58.48 | Darwin35 | just no festival of sphinx |
15:58.51 | sudhir492 | RaYmAn-Bx: thx for confiriming my hunch. Is there a way to set time for qualify in asterisk? |
15:58.56 | Darwin35 | thats next |
15:59.22 | Darwin35 | is kram around |
15:59.31 | sudhir492 | If that is possible, I can tweak that, i.e. make that smaller than routers timeout |
16:00.34 | Zeeek | exit |
16:00.39 | Zeeek | not. |
16:01.11 | ManxPower | It's Caffiene Awareness Month? |
16:01.28 | Zeeek | what about sleling awareness? |
16:01.55 | ManxPower | Organized my anti-caffiene people, I'm sure. Damn tree huggers. |
16:02.01 | tzafrir | ManxPower, you mean "a month in which we really need caffeine to be aware of things"? |
16:02.06 | sudhir492 | RaYmAn-Bx: I found in the wiki. qualify=xxx in milisecs |
16:02.18 | vaewyn | Hey guys... on SIP stuff... are the RTP ports initiated by the server or the phone? ie will UDP session tracking be enough to let them out a firewall or do I have to open up all those ports to any incoming as well? |
16:02.28 | Moc | qualify is being BANNED from all my config file (except IAX peer) |
16:02.30 | ManxPower | vaewyn: both. |
16:03.01 | vaewyn | ManxPower: so i need to leave a big gapping hole in the firewall? |
16:03.10 | ManxPower | vaewyn: Yup! |
16:03.16 | ManxPower | vaewyn: Sucks, doesn't it? |
16:03.21 | Zeeek | what are they gonna do with 1000-10100? |
16:03.24 | vaewyn | ohh geese... and I thought the NAT part was bad enough |
16:03.24 | Moc | there is a big problem in the SIP message managements with qualify |
16:03.45 | ManxPower | vaewyn: BUT if Asterisk is on a public ip, then the SIP client can be behind NAT and things are usually just fine. |
16:03.56 | sudhir492 | Moc: thx for the warning |
16:04.13 | vaewyn | The * is on a public IP... but is also behind a firewall so... |
16:04.18 | Zeeek | vaewyn I open 10000-10100 because we dion't have a lot simul calls |
16:04.24 | Moc | sudhir492, np, I keep having phone get 2500ms+ so it get unrechable, but it local... |
16:04.35 | *** join/#asterisk nctk (~nctk@lsne-catv-dhcp-29-238.urbanet.ch) |
16:04.48 | vaewyn | Zeeek: how many RTP ports does it use per call? |
16:04.51 | Moc | best way to see the problem is set qualify, and check what is the MS with your peer on a local network |
16:04.52 | roamer323 | vaewyn: double-NAT ... you're playing with explosive |
16:05.01 | Zeeek | I would think 2-4 |
16:05.03 | vaewyn | roamer323: is NOT NAT... |
16:05.09 | vaewyn | is justa firewall |
16:05.13 | Moc | NEVER NAT your *.. |
16:05.21 | sudhir492 | Moc: what is the solution? The clients router is probably unconfigurable for me. I dont know the password and nor do they :-( |
16:05.30 | Moc | unless your dealing with IAX |
16:05.52 | *** part/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
16:05.52 | Zeeek | Moc I have double NAT |
16:05.52 | ManxPower | vaewyn is one of those people that realize that NAT != firewall, even though a lot of people (myself included) use them to mean the same thing. |
16:05.56 | Moc | Zeeek, your in for a bunch of problems... |
16:06.04 | Zeeek | none yet after a year |
16:06.26 | Zeeek | my asteriks anniversary is coming up soon |
16:06.30 | Moc | sudhir492 ?? I used to lower the register on the phone to 60 second |
16:06.37 | ManxPower | Zeeek is one of those people that the Universe Smiles Upon and does not seem to have NAT related issues, even in the most macabe configurations. |
16:06.42 | Moc | it get anoying within *, but it working fine |
16:06.49 | sudhir492 | Moc: My asterisk box is on a public IP, no nat at all. its the darn phone |
16:07.04 | roamer323 | ManxPower - NAT==firewall for all home folks ... anyone who says NAT!=firewall is prob talking about equipment at work :-) |
16:07.05 | Moc | sudhir492, what is your problem exactly ? ;) |
16:07.05 | Zeeek | I don't know what I did, but since 80% of the American people believe in the existence of angels... |
16:07.06 | *** join/#asterisk nctk (~nctk@lsne-catv-dhcp-29-238.urbanet.ch) |
16:07.08 | sudhir492 | Moc: Thx. I had completely forgotten about that part . |
16:07.34 | ManxPower | Zeeek: Pretty sad huh? |
16:07.55 | Zeeek | ManxPower, there was one router that wouldn't do astrisk no matter what, but even in bridge mode with no firewall, it still didn't work |
16:08.02 | sudhir492 | Moc: Actually in the wiki, about Polycom 500, I myself had made a note to decrease the register timeout. |
16:08.09 | Zeeek | maybe I have an angel in my asterisk box? |
16:08.18 | coppice | Zeeek: and 100% of CEOs believe in angles :-) |
16:08.27 | Zeeek | Business Angels |
16:08.31 | roamer323 | Zeeek - never run fdisk on that * box!! |
16:08.33 | Moc | Zeeek, hope he doesnt get her wing stuck in the fans |
16:08.42 | [cc]smart | ye, they are called powerpoints in business |
16:08.44 | ManxPower | Zeeek: You were prolly someone like Ghandi in a previous live and you are just getting Karma Carryover. |
16:08.50 | sudhir492 | Moc: Never mind. And thanks for waking me up :-) |
16:08.55 | Zeeek | there was a burnt feather smell down there yesterday |
16:08.59 | Moc | sudhir492, ha Polycom, my favorite phones .. |
16:09.14 | vaewyn | polycom rocks! |
16:09.18 | Moc | oh yea |
16:09.28 | sudhir492 | Moc: Mine too. Though I hate the company for not supporting asterisk! |
16:09.30 | vaewyn | I only like my hitachi cable wireless one better :P |
16:09.33 | Zeeek | This may be a foolish question but can't you just use STUN on most home NAT routers? (I don't!) |
16:09.42 | Zeeek | er phones I mean |
16:09.44 | [cc]smart | karma is under extreme inflation, forget about investing in that |
16:09.52 | Zeeek | the GS has a STUN field |
16:09.53 | roamer323 | Zeeek - that's one karma point per NAT tranversal with no glitches... better top up those karma point soon :-) |
16:09.54 | sudhir492 | Moc: At $170 cannot find a better value ! |
16:10.22 | Moc | sudhir492, same as cisco, but atless polycom doesnt freekout on firmware avability, and thought, polycom have alot of feature that * dont support |
16:10.27 | Zeeek | so if I call myself with FWD I'd need at least 3 points? |
16:10.34 | PoWeRKiLL | what that -- Got SIP response 415 "" back from 81.218.111.79 |
16:10.35 | Moc | polycom will never support * until we do support them |
16:10.52 | sudhir492 | Moc: Yes. which version are you using? |
16:10.56 | Moc | sudhir492, at 170 yes it nice, I get it at 185$ |
16:10.57 | Zeeek | me ->NAT->NAT-->ast-->NAT-->FWD-->NAT-->asterisk-->NAT-->me... |
16:11.00 | Moc | 1.4.1 I think |
16:11.24 | Zeeek | that can't be right because it still works |
16:11.29 | ManxPower | My eyes! My eyes! |
16:11.41 | Zeeek | it is Sunday |
16:11.56 | ManxPower | Zeeek: You realize you're going to hell for twisting a network into that config. |
16:11.58 | Zeeek | I got yer hairpin right here |
16:12.05 | vaewyn | my polycoms and * seem to get along great... only wish they would release the devel info for playing with the menus on the phones |
16:12.28 | sudhir492 | Moc: tritechoa.com gives me at $175, no shipping if I order 5 at a time. (talk to steven or scott) You can tell him my name |
16:12.30 | roamer323 | Zeeek - with your configuration... it looks like you objective is to connect your NATs through your *'s :-) |
16:12.31 | Zeeek | I think you guys have talked me into an ip500 when I go to the US |
16:12.53 | vaewyn | Zeeek: they really do rock |
16:12.59 | Zeeek | (why not buy one here you ask? - 20% VAT) |
16:13.14 | vaewyn | hehehe |
16:13.20 | sudhir492 | Moc: another friend of mine gets it from a wholeseller in MD for $165 a piece and gives that to me at $170. |
16:13.30 | Zeeek | who has the lowest single unit pricing on Polycom? |
16:13.48 | vaewyn | Zeeek: voipsupply.com had it when I looked about 3 weeks ago |
16:13.50 | Zeeek | and how much is that in 2005 dollars? |
16:13.56 | Zeeek | I'll look now* |
16:14.11 | Moc | sudhir492, how much for the IP 600 ? |
16:14.17 | ManxPower | The problem with Polycom is that people that sell polycom supported PBXs get really good discounts on polycom phones. The rest of them have to pay a much higher price. |
16:14.25 | sudhir492 | Moc: I never inquired. |
16:14.37 | sudhir492 | Zeeek: where do you live? How many do you need? |
16:14.38 | Zeeek | so we need to find someone that works at a polycom pbx place but is hip |
16:14.49 | Zeeek | one |
16:15.00 | sudhir492 | Zeeek: where do you live? |
16:15.00 | Zeeek | are you a reseller? |
16:15.04 | Zeeek | France |
16:15.23 | Zeeek | but I'll be in the USSR^H^HA in May |
16:15.27 | roamer323 | they VAT on the declared value? declare it at $30 |
16:15.43 | ManxPower | LOL! |
16:15.57 | Zeeek | let me explain - FedEx bless them, is obliged to open everything now, anti-terror ya know - and they seem to be in bed with customs |
16:16.01 | sudhir492 | no I am not a reseller, but would not mind helping a fellow *user |
16:16.15 | Zeeek | sud where are you? |
16:16.31 | sudhir492 | I do provide solution to businesses though. I am a reseller in that sense. |
16:16.43 | sudhir492 | Near D.C in northern VA |
16:17.13 | Zeeek | ah, well I'll have to see what the cost situation is on the various offers |
16:17.49 | Moc | http://www.voipsupply.com/product_info.php?cPath=94_161&products_id=455 |
16:17.51 | Moc | that cool |
16:17.54 | Moc | but expensive |
16:18.21 | Moc | wish it were 1000$ ;) |
16:18.21 | Zeeek | sughir you have a bunch new as in 'in the unopened box"? |
16:18.26 | *** join/#asterisk jeofrey (~jeofrey@202.160.45.29) |
16:19.01 | jeofrey | RROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) |
16:19.13 | jeofrey | anyon can help me please with this error in mysql |
16:19.14 | ManxPower | Polycoms support PoE and have 2 ethernet ports. There are phones that are much cheaper, but still pretty good for much less. However, they either don't support PoE or only have 1 switch port. |
16:19.20 | vaewyn | Moc: wow... that is cool |
16:19.42 | jeofrey | im using fedora core core..... |
16:20.05 | Zeeek | god there's 20 times more SIP phones <$200 since last time I looked! |
16:20.24 | jeofrey | but i cannot run the mysql .... i want to use it for recording our calls |
16:21.14 | jeofrey | everytime i run mysql alway come out this error RROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) |
16:21.20 | Zeeek | anyone tell me what ground shipping is on the Polycom? About $10? |
16:21.25 | sudhir492 | Zeeek: Yes I do. Usually keep around 10 in stock. When I get an inquiry from a business, I usually like to complete the sale the same day. Hence have to keep some Polycom and PAP2-NA (for cheaper clients) in stock |
16:21.28 | Moc | jeofrey, you need to start mysql server first |
16:21.52 | Moc | how much is the PAP2-NA ? |
16:21.52 | jeofrey | Moc i do it for so many times already |
16:21.56 | sudhir492 | Zeeek: In priority mail, probably a few bucks less |
16:22.25 | Darwin35 | grrr kram where are you |
16:22.40 | Zeeek | sudhir492 msg me your email and I'll get back to you before I leave (late April) |
16:22.40 | sudhir492 | Darwin35: what do you need from kram? |
16:22.52 | Darwin35 | work on the freebsd g729 |
16:22.53 | vaewyn | Darwin35: probably getting ready for San Jose |
16:23.08 | Zeeek | or a URL is you have one |
16:23.29 | jeofrey | Moc after i restart the mysql server this error comeout ERROR 2002: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111) |
16:23.43 | Darwin35 | the freebsd g729 is having issues it is not always loading |
16:23.50 | Moc | joe, check the logs for errors |
16:24.04 | sudhir492 | sudhir@cequip.com |
16:24.23 | jeofrey | Moc where can i get the log..... sorry im new to this..... |
16:24.26 | vaewyn | Anyone else gonna be in San Jose with us this week? |
16:24.50 | Moc | jeofrey, dont recall for mysql, but check in /var/log or /var/lib/mysql |
16:25.01 | Zeeek | ManxPower are all the firmware updates and docs (if exist(docs)) available somewhere for download? Or do yiou have to cisco them for money? |
16:25.12 | Darwin35 | i WISH i WAS BUT WORK WOULD NOT PAY FOR IT |
16:25.15 | Darwin35 | sorry |
16:25.16 | vaewyn | hehehe |
16:25.30 | vaewyn | They are barely paying for me to hit the exhibits only |
16:25.45 | Darwin35 | ahh that bites |
16:25.47 | vaewyn | but it is worth it... got some contacts to hang out with that will be there |
16:25.49 | jeofrey | ok moc |
16:26.13 | Zeeek | sudhir492 ? Not hearing from you |
16:26.15 | Darwin35 | well I wanted to be there because snom stole my mini pbx box |
16:26.26 | Darwin35 | and I wanted to confront them in public |
16:26.37 | vaewyn | Hope jerJer is gonna be there cause I want the telco guy in my group to meet him |
16:26.38 | Darwin35 | thier box looks just like mine |
16:26.46 | vaewyn | Darwin35: hehehe |
16:26.59 | Zeeek | vaewyn, what group, you play in a band? |
16:27.05 | Zeeek | Like Oasis? |
16:27.16 | vaewyn | Zeeek: hahaha... nah... our department group :P |
16:27.32 | vaewyn | nothing quite that fun :P |
16:27.40 | Zeeek | who can answer my Polycom quest: do you have to pay for firmware? |
16:27.45 | Damin | No, I'm the one in Oasis. He played in Journey. |
16:27.53 | jeofrey | Moc here is the log http://www.pastebin.com/250075 |
16:27.54 | vaewyn | Zeeek: nope... is free |
16:28.03 | Zeeek | and I sang the SttarTrek Enterprise theme song |
16:28.16 | Zeeek | "It's been a long unload..." |
16:28.29 | vaewyn | Will be fun... can do some long distance geocaching while I am there also :P |
16:28.31 | Zeeek | "but I think I can dial now..." |
16:28.33 | Darwin35 | StarTrek Enterprise with Scott Beckula is dead |
16:28.38 | Darwin35 | put it to rest |
16:28.48 | Zeeek | I thouhgt he landed in Nazi America? |
16:28.49 | Darwin35 | like all his other shows he never finished |
16:28.52 | vaewyn | Or at least make him leap again :P |
16:29.07 | Zeeek | The last episode was just on here |
16:29.19 | Zeeek | due to alien help, Germany won the war |
16:29.45 | Zeeek | I could have consoled T'pen or whatever her nampe is with the tight purple velvet suit |
16:30.00 | Darwin35 | Tapal |
16:30.02 | Zeeek | she comes on here sometimes with the name Katty |
16:30.08 | Zeeek | be sure to be nice to her |
16:30.20 | Zeeek | T'Pal not Tapal |
16:30.31 | Zeeek | Tapal is a dishwashing detergent on Mars |
16:30.39 | Darwin35 | heheh |
16:30.58 | Zeeek | Registered to '69.73.19.178', who sees us as T'Pal |
16:31.09 | Zeeek | probe me |
16:31.37 | Zeeek | not big enough |
16:31.42 | Zeeek | or long enough |
16:32.07 | vaewyn | qualify=yes |
16:32.08 | vaewyn | :P |
16:32.21 | Zeeek | I think kram should sing that song at the next AstCon: "It's been a looooong road...." |
16:33.38 | Zeeek | shit T'Pal is now UNREACHABLE |
16:34.15 | Moc | Zeeek, remove qualify=yes, it a bug ;) |
16:34.21 | nctk | hi, everybody, after seeking for a while, I still can't find the answer of may question |
16:34.32 | Zeeek | I love qualify - I live for it |
16:34.47 | Zeeek | nctk ask and yee shall be told |
16:35.08 | Moc | joe: install TaoLinux (RH Enterprise recompiled/free version) |
16:35.14 | nctk | is there a way of getting the tax information from the isdn level using zaphfc |
16:35.57 | Darwin35 | so those of use not going will hoold a online lunch gathering and discuss asterisk and other thigns |
16:36.00 | Zeeek | what information do you want? |
16:36.18 | Darwin35 | nsal |
16:36.39 | Darwin35 | n/s/a/l |
16:36.39 | Zeeek | asterisk and other thighs... I like that |
16:36.52 | Darwin35 | things |
16:37.00 | Darwin35 | fingers typing to fast |
16:37.00 | Zeeek | less interesting |
16:37.05 | nctk | at least in switzerland, but in germany to, you get messages on the D channel saying you just paid 10 cents for example |
16:37.05 | Zeeek | thighs |
16:37.05 | Darwin35 | true |
16:37.43 | Zeeek | nctk I don't know if that happens in USA |
16:38.28 | nctk | zeek you said you're in france, no? Isn't there something like that with "FT Numeris"? |
16:39.14 | Blackvel | how do i handle 0911 and 49 in the dailplan? |
16:39.38 | Blackvel | if EXTEN starts with 09 do this, if extension does not start with 0 do this? |
16:41.27 | Zeeek | Numeris is ISDN but I don't use it anymore so I can't answer your question |
16:41.44 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
16:47.09 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfl9t.dialup.mindspring.com) |
16:48.18 | nctk | I know Numeris is ISDN, I continue to dig, hopefully I will find something ;) |
16:48.21 | sudhir492 | Zeeek: I went out for a while. what do you want to hear from me? |
16:48.35 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-141-149-253-154.buff.east.verizon.net) |
16:48.42 | Zeeek | I was saying you could msg me an email or URL for contact later |
16:48.48 | Darwin35 | I need to find more features to add to my dial plan I need dial100 and callrecord |
16:48.55 | Zeeek | I'll be in the uS for only 4 days |
16:49.09 | Darwin35 | zeek where you from |
16:49.25 | Zeeek | Minneapolis,MN,USofA |
16:49.31 | sudhir492 | Zeeek: my email is sudhir@cequip.com |
16:49.38 | Zeeek | ok |
16:50.03 | vaewyn | Zeeek: heh... I'm flying through there tomorrow on way to San jose.... is a nice place |
16:50.35 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
16:50.40 | vaewyn | got 2 friends that live there also |
16:50.41 | Zeeek | no way! |
16:50.44 | PatrickDK | anyone know what size powersupply supira2000 needs? |
16:50.55 | Zeeek | sudhir492 I wrote you with my addr |
16:51.34 | Zeeek | by the way I see mention of 12v power supply but not input voltage. Would it be 100-250 by any chance? |
16:52.28 | PatrickDK | I just need output of the transformer, or the input of the 2000 |
16:52.32 | PatrickDK | voltage and amps |
16:53.07 | Zeeek | Sorry I was talking to sudhir on a similar subject |
16:53.17 | PatrickDK | ah :) |
16:53.33 | Zeeek | doesn't sipura have a site with a datasheet? |
16:53.39 | Blackvel | anyone has some workaround for this broadvoice / asterisk problem yet and why broadvoice sends you to their voicemail? |
16:53.44 | PatrickDK | I'm looking around, but haven't found anything |
16:53.53 | PatrickDK | it doesn't even say on the device, normally they do |
16:53.59 | Darwin35 | there is a fix in 1.0.5 |
16:54.04 | djin | I read about a wifiaxy, does anyone know what that is? |
16:54.15 | Blackvel | no |
16:54.19 | Blackvel | i tried with 1.0.6 today |
16:54.21 | Blackvel | not working |
16:54.25 | shepherd | djin: it's a joke |
16:54.39 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
16:54.44 | Darwin35 | I am building 1.0.6 now i will let you know |
16:54.47 | Blackvel | there is a BV fix in 1.0.4 |
16:54.53 | djin | ok, don't get it though :) |
16:54.58 | shepherd | djin: basically it was a ton of batteries + wifi card + iaxy |
16:55.04 | shepherd | and it only lasted 30 mins |
16:55.05 | Blackvel | ah you mean there is an additional fix for BV in 1.0.5? |
16:55.10 | Blackvel | that would explain it |
16:55.14 | Darwin35 | yes |
16:55.25 | djin | a sheperd, but was it meant as a joke? |
16:55.27 | Blackvel | and it only got commited to 1.0.5? |
16:55.28 | Blackvel | weird |
16:55.32 | PatrickDK | ah, found it |
16:55.35 | Blackvel | shouldnt the lastest be fixed |
16:55.36 | Blackvel | ? |
16:55.50 | Darwin35 | should I am building now |
16:55.59 | Darwin35 | let me check and test |
16:56.00 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
16:56.02 | Blackvel | you have to be very patient with me darwin :) |
16:56.07 | shepherd | djin: yeah.. .they embeded it into an old rotary phone, and were putting it in elevators with no cords :) |
16:56.10 | Darwin35 | its ok |
16:56.13 | Blackvel | my english is sometimes werid |
16:56.15 | Blackvel | weird |
16:56.18 | Zeeek | IAXy needs almost 1A power |
16:56.19 | Blackvel | i dont understand too fast |
16:56.22 | Blackvel | sorry |
16:56.27 | Darwin35 | my human is some times strange |
16:56.31 | djin | cool :0 |
16:56.42 | shepherd | zeek: 1500 mA |
16:56.47 | Blackvel | is it the cvs of 1.0.5 or the .gz files of 1.0.5? |
16:56.56 | Zeeek | actually I think it's 1200ma |
16:56.59 | Zeeek | I have one here |
16:57.14 | Zeeek | but they may have recommended a 1500ma supply |
16:57.14 | shepherd | haha.. you could be right about that |
16:57.15 | Darwin35 | I know its in the tar.gz and should be in the cvs |
16:57.20 | Darwin35 | brb restroom |
16:57.46 | Nukemizer | I am trying to test my TE110P card with Redhat, since I could not get PRI to work correctly with Mandrake. I am trying to successfully test PRI in loopbackmde but I always get alarms. Is there a switch for letting card run in LoopBack mode ? |
16:57.52 | djin | did anyone look into this dCAP certification? |
16:58.48 | djin | or signup for it? |
16:58.51 | Blackvel | Darwin35: you are a killer :) |
16:59.19 | *** join/#asterisk jtar (~john@cpc2-mapp3-4-1-cust214.nott.cable.ntl.com) |
17:01.08 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
17:04.03 | *** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz) |
17:04.17 | Nukemizer | I may have found my problem... To get a Digium PRI to work do you need the hisax module for ISDN ? |
17:04.41 | djin | nope |
17:04.49 | djin | why? |
17:05.13 | Nukemizer | looking for diff between RH and Drake.. to get PRI up.. |
17:05.41 | Nukemizer | I still am trying to find a way to test in Loop Back mde without card complaining |
17:05.58 | djin | Don't know much about Mandrake, but RH shouldn't have that much obstacles. |
17:06.49 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
17:07.14 | Nukemizer | RH still complains in Loop back mode. I amnot connected to PBX since I am testing at home today. Should be able to create a loopback test right ? where card does not complain or go into RED alarm ? |
17:07.37 | djin | ehat card do you use? |
17:07.48 | Nukemizer | TE110P |
17:08.51 | Nukemizer | Thanks to all that tolerte my questions, I am just looking for a predictable testing path for trouble shooting |
17:09.21 | djin | Not sure you can loopback a one port card. Only used it with 4 ports. |
17:09.40 | *** join/#asterisk Tarox (~chris@pD9E7B7CA.dip.t-dialin.net) |
17:09.50 | djin | I assume you want to test an outgoing call being routed back in? |
17:10.29 | Nukemizer | perhaps the very nature of PRI and D channel communication will prevent my from just using a Loop Back plug |
17:10.42 | memic | <PROTECTED> |
17:10.58 | memic | do i have to patch asterisk? |
17:11.17 | memic | im using asterisk from debian/testing |
17:11.20 | Darwin35 | ok back |
17:11.22 | Nukemizer | no, not even to that stage.. calls will go through if card is not in the midle of a choke fest. I want to get card(s) to work just in an idle state without error |
17:11.24 | Darwin35 | sorry |
17:12.00 | Darwin35 | I heard there where some bad cards and a new rev of the card was comming out |
17:12.16 | Darwin35 | any further info call digium |
17:13.03 | Nukemizer | I got digium to send me out a second card but they both behave the same. even in different boxes. so I am not trying RH and Xorcom to see if it really is the OS |
17:13.03 | scrubb | any sip gurus around? |
17:13.11 | Blackvel | http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
17:13.13 | scrubb | I think I found a bizzare bug! |
17:13.19 | Blackvel | is the last descrition useless? |
17:13.27 | Blackvel | it doesn't tell anything about the filename |
17:13.33 | *** join/#asterisk Jayden (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
17:13.46 | tzafrir | Nukemizer, what's the problem? |
17:13.54 | Darwin35 | <PROTECTED> |
17:14.43 | scrubb | two SIP registrations seem to always use the first information for an inbound call, inspite of the SIP invite. |
17:14.53 | Nukemizer | tzafrir, I am looking for a way to loopback test my TE110P to help my locate why my card goes into Red Alarm all the time |
17:15.27 | scrubb | no sip heros in here? |
17:15.28 | Nukemizer | Loopback will work with e&m wink trunk mode but there is really no functionality like ISDN |
17:15.36 | scrubb | ok, will post to BUGS and see what I get. |
17:16.00 | Zeeek | scrubb describe your network and phones |
17:16.02 | tzafrir | scrubb, better ask your question anyway. One of those heros may be away but reading logs |
17:16.14 | scrubb | ok, I have TWO seperate broadvoice accounts. |
17:16.18 | Zeeek | stop |
17:16.18 | scrubb | I register each properly. |
17:16.21 | Nukemizer | I was just hoping that someone might be able to help me find a way to test via loop back mode so I can fight this one step at a time. |
17:16.33 | scrubb | its not broadvoice's problem. |
17:16.36 | scrubb | its * problem. |
17:16.50 | scrubb | broadvoice sends the right INVITE for each account. |
17:17.00 | dfunnell | Hi all, trying to dial out with exten => _1.,1,Dial,CAPI/470:${EXTEN:1} but it tries to dial only second digit of EXTEN (i.e. digit after '1'), as it is dialled. Means it always fails to dial out. Any ideas? |
17:17.10 | Zeeek | everyone seems to be saying BV is broken as of a day or two |
17:17.12 | scrubb | * receives the inbound call as if they are the SAME (first registration) number. |
17:17.17 | scrubb | its NOT BV! |
17:17.24 | JerJer[mobile] | dfunnell: because _1. is invalid |
17:17.24 | scrubb | I packet sniffed. |
17:17.37 | JerJer[mobile] | _1X. is valid |
17:17.39 | scrubb | BV is sending a seperate and proper invite for each call. |
17:17.58 | scrubb | if I do a SIPGetHeader on the To field it is correct. |
17:18.08 | MikeJ[Jayden] | hey JerJer... wassup. |
17:18.12 | mikegrb | JerJer[mobile]: and he has ${EXTEN:1} |
17:18.20 | scrubb | * just is interpretting the call to the second account as if it came to the first one. |
17:18.46 | scrubb | Bv is working fine for me otherwise. |
17:19.24 | scrubb | ok, I'll wrap up my findings and post it. |
17:21.10 | RaYmAn-Bx | scrubb: if you avoid using the /XXXX in the register => statement (i.e. to send it to a specific extension from there) and simple send it to a context, you should be able to seperate by extension and hence process the calls seperately. |
17:21.59 | scrubb | RaYmAn-Bx: hmmm, checking. |
17:22.45 | dfunnell | JerJer[mobile]: Thanks, have changed to _1X. but now dials only first two digits (after initial '1' for outside line). Any way I can get it to dial full number without specifying number of digits (number length will be variable). |
17:22.53 | scrubb | RaYmAn-Bx: my register looks like: register => #####:password@sip.broadvoice.com |
17:23.02 | scrubb | RaYmAn-Bx: there is no /XXXX |
17:23.25 | dfunnell | mikegrb: Is this incorrect? Intention is to strip first '1' from number to prevent it from dialing it externally. |
17:24.03 | mikegrb | dfunnell: that is correct |
17:24.17 | mikegrb | dfunnell: I misread statements above |
17:24.57 | dfunnell | mikegrb: Cool, thanks. |
17:25.48 | RaYmAn-Bx | scrubb: then if both calls are being caught by the same block in sip conf, you should be able to "catch" the seperate extensions in the extension blok it directs to |
17:27.45 | dfunnell | JerJer[mobile]: Still there? |
17:27.56 | *** join/#asterisk Tarox (someone@pD9E7BB98.dip.t-dialin.net) |
17:28.04 | scrubb | RaYmAn-Bx: yeah, except that its not distinguishing between them. |
17:29.35 | Blackvel | Darwin35: i mean, what happens the instructions on http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice when it is not clear what filename it is? :) better to remove the code instructions, I am so confused. |
17:29.45 | scrubb | RaYmAn-Bx: asterisk show the channel name as the wrong one. |
17:30.56 | Blackvel | Darwin35: I mean, what do the instructions help on ... |
17:34.39 | RaYmAn-Bx | scrubb: hmm, weird. I've had it working vaguely ok with two registrations to same server before..I seem to remember having the same problem but I can't remember what I did to fix it..I think I just put specific exten => number,1 extensions in the extension tbh |
17:34.40 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
17:43.31 | Blackvel | hey bjohnson_ |
17:43.43 | Blackvel | feel free to ask me privately if I can be any guide for you :) |
17:43.51 | Blackvel | forget |
17:47.29 | *** join/#asterisk fugitivo (~ajf@201.255.100.59) |
17:48.44 | *** join/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc) |
17:50.21 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
17:50.50 | *** join/#asterisk feklee (feklee@genba.ffii.org) |
17:51.31 | feklee | I always get this message when trying to authenticate to sipgate: Mar 6 18:50:30 NOTICE[4831]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:9779619@sipgate.de>;tag=as6a89cf36' |
17:51.41 | feklee | How do I find what is going wrong? |
17:51.52 | feklee | I already tried different configurations, but to no avail. |
17:52.12 | feklee | If sipgate is hard to set up, what alternatives are there Ggermany)? |
17:52.59 | Blackvel | nikotel |
17:53.00 | Blackvel | gmx |
17:53.03 | Blackvel | freenet |
17:53.10 | Blackvel | but sipgate is not hard to setup |
17:53.19 | Blackvel | who told you that nonsense? :) |
17:53.51 | Blackvel | please make sure you have set insecure=very in sip.conf [sipgate.de] |
17:53.54 | feklee | Blackvel: Do you run a sipgate setup? |
17:53.57 | Blackvel | sure |
17:54.07 | feklee | Blackvel: Already have insecure=... in sip.conf |
17:54.15 | Blackvel | remove auth=md5 |
17:54.24 | feklee | Blackvel: Could you make the setup available to me? |
17:54.31 | *** join/#asterisk IceBerg (iceberg@cpe-24-166-0-83.indy.res.rr.com) |
17:54.38 | Zeeek | I don't have insecure in mine |
17:54.49 | Blackvel | paste your config on pastebin.com |
17:55.00 | Blackvel | without the username + passwort :) |
17:55.28 | *** join/#asterisk memic (skdmwnf@dsl-084-056-106-237.arcor-ip.net) |
17:56.48 | *** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.res.rr.com) |
17:56.52 | feklee | Blackvel: Well, I tried this config: http://www.sipgate.de/faq/index.php?aktion=artikel&rubrik=650&id=359&lang=de&highlight=asterisk |
17:56.55 | feklee | and others |
17:57.16 | Blackvel | try to put on pastebin and I take a look |
17:57.20 | Blackvel | I'll |
17:57.44 | dan2 | drumkilla: ping |
17:57.47 | feklee | OK, I'll justclean up my config a bit before poting. |
17:57.54 | D1ng0 | so has Broadvoice fixed incoming calls yet ??? |
17:58.09 | dan2 | Everybody who is having broadvoice troubles join #broadvoice please |
17:58.18 | Blackvel | 1.0.5 |
17:58.20 | Blackvel | try to get this |
17:58.24 | Zeeek | his problem seems to be on the register |
17:58.26 | Blackvel | darwin has mentioned a fix |
17:58.31 | memic | anybody has a hfc card running with 2.6.9 ? |
17:58.42 | Blackvel | memic: i use 2.4 |
17:58.45 | memic | hm |
17:58.46 | Blackvel | join #asterisk-drinkers |
17:59.05 | D1ng0 | dan2, jeeez a whole two people in there me and you |
17:59.07 | memic | whats that chan 4? |
17:59.21 | scrubb | RaYmAn-Bx: thanks. |
17:59.25 | memic | i cant use 2.4 |
17:59.58 | IceBerg | Hi guys, Im a noob, and Im trying to get basic asterisk up, Im running it on FreeBSD and Im following the setup on voip-info.org. I backed up all the installed configs that the BSD ports installs and put in the sample ones from the source packagea renaming them all to get rid of the .sample. edited the sip.conf so that my xlite phone is listed. But I get this error: retrans_pkt: Maximum retries exceeded on call |
18:00.30 | feklee | Blackvel: http://www.pastebin.com/250099 |
18:00.44 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
18:00.44 | *** mode/#asterisk [+o bkw_] by ChanServ |
18:00.55 | psywar | how come * sometimes requires a # to terminate a single-digit extension and other times not? |
18:01.04 | psywar | is it because it's a prefix of another extension? |
18:01.19 | bkw_ | i'm here in denver |
18:01.22 | bkw_ | with steve and oej |
18:01.30 | bkw_ | they are on the same flight with me to San Jose |
18:01.39 | dan2 | bkw_: could you modify the topic for me |
18:01.48 | psywar | IceBerg: I get that every time I restart *. Ignore it. |
18:02.01 | Blackvel | feklee: where is your [sipgate] section? it is missing |
18:02.04 | dan2 | bkw_: I'm leaving for San Jose tomorrow, but I'd like a note about #broadvoice in the topic |
18:02.25 | *** join/#asterisk Bentley (~rbc@S01060080c8135e6a.cg.shawcable.net) |
18:02.48 | IceBerg | psywar, ok, but also the demo call never goes through, xlite give me the reorder tone and then I get errors about not being ableto start the mp3player, and ues mpg123 is installed and working |
18:03.34 | feklee | Blackvel: That's the config they propose here: http://www.sipgate.de/faq/index.php?aktion=artikel&rubrik=650&id=359&lang=de&highlight=asterisk |
18:03.41 | psywar | I got the mohmp3 errors a bit, make sure you have the right mpg123 version |
18:03.51 | psywar | I had to manually create the mohmp3 dir |
18:03.56 | *** join/#asterisk FarrisG (~farris@c-67-162-181-62.client.comcast.net) |
18:04.19 | psywar | start a console with -vv and see what errors you're getting when you try to make the demo call, whatever that is |
18:04.29 | feklee | Blackvel: Couldn't you just post your config. I'm not so much interested in learning * at the moment. I need it running by tomorrow. |
18:04.38 | IceBerg | psywar, yea I had to create it, and I know the one error I get is due to no mp3's in there but shouldnt the demo call go through no matter what? |
18:05.03 | psywar | yes, I think things things are not the problem though |
18:05.10 | FarrisG | Anyone have the patience to help me get asterisk running? I'm at a new job, in charge of phones/servers... Asterisk had been running smoothly for over a month, and then somehow the machine got restarted, and now it won't start back |
18:05.31 | psywar | /etc/init.d/asterisk start |
18:05.31 | FarrisG | I can post error messages or configs, just wondering where would be the first place I ought to look |
18:05.43 | dan2 | bkw_: ? |
18:05.52 | FarrisG | psywar: I wish. It appears to have been setup differently |
18:06.36 | psywar | have you started a console with -vv? |
18:06.42 | psywar | asterisk -gvvc |
18:06.45 | psywar | see what it says |
18:07.08 | bkw_ | flight delayed |
18:07.20 | bkw_ | i had to go from gate B59 to G16 |
18:07.20 | bkw_ | er B16 |
18:07.30 | bkw_ | its like 2 miles |
18:07.31 | IceBerg | where are the audio files by default for the demo line, are they also in the mp3moh dir? |
18:07.53 | bkw_ | oej is trying to get on GPRS |
18:08.34 | Blackvel | feklee: are you behind NAT? |
18:08.40 | dan2 | bkw_: could you update the topic to make a note for #broadvoice |
18:09.20 | feklee | Blackvel: No, and I even disabled the firewall. |
18:09.37 | Blackvel | you run it on the firewall? |
18:09.49 | shepherd | denver is like the coolest airport though |
18:09.50 | Blackvel | or the router |
18:09.52 | feklee | Blackvel: No I'm connected via dialup. |
18:10.01 | Blackvel | what do you mean? |
18:10.10 | Blackvel | adsl ppp0? |
18:10.22 | Blackvel | linux? windows? |
18:10.23 | feklee | NO, via 56k-modem. |
18:10.25 | feklee | LINUX |
18:10.47 | feklee | (I know that this is too slow) |
18:10.57 | FarrisG | "Loading module chan_zap.so failed" |
18:10.58 | feklee | I want to install it on a different machine later |
18:11.09 | Blackvel | what is not working? outgoing or incoming? |
18:11.09 | IceBerg | psywar, the version of mpg123 I have is also installed from the ports, there is no reason it should not work |
18:11.20 | *** join/#asterisk syncoherent (~synco@c-24-98-180-64.atl.client2.attbi.com) |
18:11.29 | tzafrir | FarrisG, there should be a reason a line or so above |
18:11.36 | feklee | Blackvel: If you could just post your config, that would be great. |
18:11.54 | Blackvel | not really |
18:12.02 | Blackvel | too different |
18:12.07 | FarrisG | "Unable to open D-channel 24 (No such device or address)" |
18:12.28 | Zeeek | does sipgate.de have test numbers? |
18:12.34 | Blackvel | 10000 |
18:12.45 | tzafrir | As there are a number of existing security holes with mpg123: when I use it with a sane Asterisk configuration can I be certain that it will only play the moh, and not anything from remote users? |
18:12.51 | BrianR___ | Hmm.. I wonder if anyone's working on a debian package for asterisk-1.0.6... Perhaps I should do it.. |
18:13.06 | Zeeek | Blackvel for me? |
18:13.11 | Blackvel | yeah its 10000 |
18:13.12 | feklee | Blackvel: The problem is that I've no clue where the problem may be. When I google for the error message, I get no hit. |
18:13.13 | tzafrir | BrianR___, I have a working prototype, or so |
18:13.21 | dfunnell | Hi all, can anyone tell me how to get * to wait until entire number is dialled (before dialling out) without having to specify exact # of digits? Have tried exten => _1X.,1,Dial,CAPI/470:${EXTEN:1}, but this tries to dial out with first two digits after '1'. |
18:13.27 | BrianR___ | tzafrir: Based on the 1.0.5 package? |
18:13.34 | Blackvel | you didnt paste the complete config on pastebin anyways |
18:13.38 | Blackvel | and i have no time now |
18:13.47 | dfunnell | Numbers are of variable length, but I'm not sure how to handle them. |
18:13.48 | tzafrir | BrianR___, http://tzafrir.org.il/rapid/APT.html |
18:13.52 | Blackvel | preparing for a UK conference the next minutes |
18:13.59 | syncoherent | drunnell: you can use Disa() -- that's what i have to do for one of my phones |
18:14.06 | tzafrir | based mostly on the pkg-vopip debs |
18:14.13 | Blackvel | i would remove srvlookup=yes completely |
18:14.14 | shepherd | dfunnel: you can use agi |
18:14.20 | Zeeek | Blackvel that wasn't me |
18:14.24 | tzafrir | Anyway, there are now 1.0.6 packages ready for upload |
18:14.31 | tzafrir | I don't have the URL , though |
18:14.32 | Blackvel | <Zeeek> does sipgate.de have test numbers? |
18:14.39 | Blackvel | yes the number is 10000 |
18:14.42 | Zeeek | yes but the rest about pasting |
18:14.44 | BrianR___ | tzafrir: Aah. Are you on that team? |
18:14.48 | Zeeek | yhx gotcha |
18:14.50 | tzafrir | yes |
18:14.51 | Blackvel | no |
18:14.53 | Blackvel | someone else |
18:14.54 | FarrisG | psywar: what exactly would cause that "Unable to load D-channel" error, or where should I look? |
18:15.09 | BrianR___ | tzafrir: I'm a debian maintainer also - albeit a bit of a slacker these days. |
18:15.16 | Nugget | hooray for slackers. |
18:15.24 | psywar | sorry, dunno FarrisG, I'm a * noob |
18:15.27 | psywar | sounds like an ISDN thing |
18:15.29 | shepherd | slackers + asterisk #1 |
18:15.51 | FarrisG | It occurred right after it parsed zapata.conf |
18:16.41 | BrianR___ | yay.. dev.asteriskdocs.org is working again. . |
18:16.54 | feklee | Blackvel: Here's an update but it's still nto working: http://pastebin.com/250111 |
18:17.19 | Zeeek | sipgate isn't letting me in |
18:17.19 | BrianR___ | Hmm.. Maybe it's not.. |
18:18.39 | dfunnell | syncoherent: Disa() is for dialling in to *, isn't it? Will this help me with dialling out? |
18:18.52 | FarrisG | But there is no mention of D-channel 24 in zapata.conf |
18:19.46 | syncoherent | dfunnell. i use it to dial out as well. |
18:19.47 | dfunnell | shepherd: I was trying to avoid using agi, as I'm not looking forward to buggering up scripts, no other (easy) way? |
18:20.20 | shepherd | i can't think of any |
18:20.25 | shepherd | but there might be |
18:20.25 | dfunnell | syncoherent: Sounds interesting, will give it a try. Any chance of sending me your exten string so I can see how you do it? |
18:20.37 | dfunnell | shepherd: Ok, thanks. |
18:20.45 | shepherd | extensions.conf is it's own programming language :) |
18:20.57 | feklee | Zeeek: I've the same problem. Are you trying to connect to Sipgate in Germany? |
18:21.03 | Zeeek | Blackvel do you have a paid account at sipgate? |
18:21.13 | Blackvel | yes |
18:21.16 | Zeeek | yes I am registered but it doesn't accept a call to 10000 |
18:21.23 | feklee | Zeeek: However, I'm quite certain that this is a configuration problem. It worked with X-lite this morning. |
18:21.24 | Zeeek | but I don't pay - |
18:21.33 | Blackvel | dunno |
18:21.38 | Blackvel | may be becoz of that |
18:21.39 | Blackvel | not sure |
18:21.48 | Zeeek | I think I have an X-Lite setup - I'll try it now |
18:21.54 | IceBerg | I still get the reorder tone when I try to dial the demo line even though the console says it's playing the greeting files |
18:22.38 | Zeeek | WOW sexy Greman voice say ing I'm cool! |
18:22.45 | Zeeek | so there *is* an issue |
18:23.07 | *** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net) |
18:23.15 | feklee | Could anyone have a look http://pastebin.com/250111 ? I still cannot connect to Sipgate and I tried different configs all afternoon (although I absolutely don't have the time for that). |
18:23.18 | *** join/#asterisk rephorm (~rephorm@cpe-24-28-67-25.austin.res.rr.com) |
18:24.39 | Blackvel | try again end of week then :) |
18:24.50 | Blackvel | or pay someone to do it ;) |
18:25.20 | Blackvel | type=peer |
18:25.23 | Blackvel | type=friend |
18:25.25 | Blackvel | change this |
18:25.52 | BrianR___ | damn it. the 1.0.6 branch doesn't have the disconnect dtmf tone configurable in features.conf :( |
18:25.58 | RaYmAn-Bx | feklee: try adding the authuser in the register => line..i.e. number:password:authuser@host/exten |
18:26.03 | Blackvel | exten => _X.,2,Dial(SIP/${EXTEN}@sipgate.de |
18:26.06 | Blackvel | change this to @sipgate |
18:27.38 | Zeeek | authuser is same as user, no? |
18:27.54 | Blackvel | you dont need authuser |
18:29.37 | Nugget | Asterisk CVS-v1-0-03/06/05-12:24:02 built by nugget@suburbia.slacker.com on a i386 running FreeBSD |
18:29.40 | Nugget | yay |
18:29.48 | feklee | Blackvel: Still doesn't work. I'd be happy to pay someone say 10 EUR for helping me. |
18:30.06 | feklee | RaYmAn-Bx: authuser doesn't improve things. |
18:30.32 | shepherd | nugget: what all did you have to do to get that running? |
18:30.34 | RaYmAn-Bx | what is the exact problem again? Failing to register, unable to make calls, unable to receive calls? |
18:31.03 | Zeeek | feklee maybe sipgate changed something because mine used to work a long time ago |
18:31.06 | Nugget | I compiled it. |
18:31.12 | Nugget | then I ran it. |
18:31.13 | feklee | RaYmAn-Bx: I get this error message: NOTICE[5660]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:9779619@sipgate.de>;tag=as1670ba42' |
18:31.27 | feklee | RaYmAn-Bx: And I cannot make calls from outside (that's what I'm interested in) |
18:31.27 | shepherd | heh... ok |
18:31.42 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
18:31.42 | shepherd | i hear it will hardlock |
18:31.49 | shepherd | at times |
18:31.50 | Zeeek | feklee you get that without even making a call? |
18:31.57 | Nugget | it's flaky with zaptel, but just asterisk is just fine. |
18:32.32 | feklee | Zeeek: Yes I get the error message right after strating up * |
18:32.34 | TheBear | hi all, after a long while of not using my * server or my snom200 phones I ant to get back into VoIP. However I can remeber my snom phone password and help ? |
18:33.01 | RaYmAn-Bx | feklee: ignoring the error, do you show as online on the webpage? Can you make/receive calls at all? |
18:33.59 | feklee | RaYmAn-Bx: I cannot receive calls. |
18:34.12 | feklee | RaYmAn-Bx: I'm not interested in making calls, but I'll try. |
18:34.13 | feklee | ... |
18:34.29 | RaYmAn-Bx | feklee: it helps in debugging |
18:34.36 | Zeeek | I reister ok and show as online but I can't call either! |
18:35.02 | RaYmAn-Bx | I can't say I've tried using sipgate for anything but free calls so I dunno whether I can call properly tbh |
18:35.16 | *** join/#asterisk jsolares (~jsolares@200.12.33.64) |
18:35.25 | Zeeek | I used it a long time ago but he's right, I can call with X-Lite and not thru asterisk |
18:35.37 | Zeeek | I am not authenticating properly at INVITE |
18:38.26 | RaYmAn-Bx | doesn't sipgate have a test number of some kind? |
18:38.55 | Zeeek | 10000 |
18:39.14 | Zeeek | I call it and it won't auth me on INVITE but it works with X-Lite |
18:39.48 | RaYmAn-Bx | hmm, it works fine for me |
18:40.18 | Zeeek | Can you pastebin your peer entry (w/o passwd) |
18:40.43 | Zeeek | note that on sipgate page on asterisk they have the same allow= twice - they didn't proofread the page very well ;) |
18:41.19 | *** part/#asterisk rephorm (~rephorm@cpe-24-28-67-25.austin.res.rr.com) |
18:42.24 | RaYmAn-Bx | Zeeek: http://skumler.net/sipgate.txt (Admittedly this is for sipgate.co.uk, but it seems to work 100% identically..I have a .de account as well) |
18:42.47 | Zeeek | thx - i'll take a look |
18:44.43 | Zeeek | that worked! |
18:44.49 | Zeeek | but mail voice this time :( |
18:45.01 | Zeeek | I removed tyhe register statement and restarted |
18:45.35 | Zeeek | they must require one of the new lines (I didn't use before): fromdomain or fromuser |
18:45.46 | Zeeek | feklee you get that without even making a call? |
18:45.57 | Zeeek | feklee oops - you paying attention? |
18:46.09 | RaYmAn-Bx | I seem to remember I had to add the third field to the register statement as well... (i.e. authuser, being the same as username) |
18:46.40 | Zeeek | I don't have that. There is some confusion tho since I never call out, never get calls from mo sipgate number and never register :) |
18:47.14 | feklee | Zeeek: I'm still not successful. But I guess I need to learn some Asterisk basics first. |
18:47.16 | Zeeek | I'll put the register back in now and see |
18:47.27 | RaYmAn-Bx | heh. I only receive calls from PSTN on my sipgate number. And that works fine |
18:47.34 | Zeeek | funny with the register I get the woman! |
18:47.51 | Zeeek | Well Raym somehow one of the two lines did it |
18:47.54 | RaYmAn-Bx | I get a male voice as well |
18:47.59 | Zeeek | feklee Look at his file |
18:48.14 | feklee | Zeeek: What's your config now? (for sip.conf and extensions.conf) |
18:48.15 | RaYmAn-Bx | it makes sense, they are often required |
18:48.17 | Zeeek | I got th ewoman when I registered (or maybe they alternate - politicall correct?) |
18:48.38 | Zeeek | Look at the file Mr. RaYmAn posted |
18:49.15 | FarrisG | Ok, when asterisk tries to start and parses zapata.conf, it errors out with "Unable to load D-channel 24", but there is no 24 in zapata.conf. How do I determine why this is happening and fix the problem? |
18:49.48 | Zeeek | Although I don't really care about sipgate.de, thanks for helping - I HATE when stuff doesn't work! Even free stuff |
18:50.00 | Zeeek | RaYmAn-Bx^^^^^^^^^^^^ |
18:50.43 | RaYmAn-Bx | heh, same here..It just bugs me |
18:50.53 | Zeeek | <PROTECTED> |
18:51.05 | Zeeek | they actually have a decent interface |
18:51.27 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
18:51.29 | RaYmAn-Bx | I love the fact that sipgate uk gives free geographical DID's though :) It rocks. |
18:51.40 | Zeeek | my number is a 207 - that must be expensive to call, ya? |
18:52.00 | Zeeek | like 8p/min |
18:52.07 | RaYmAn-Bx | a uk 207? as in 0207 or? |
18:52.21 | Zeeek | <PROTECTED> |
18:52.30 | RaYmAn-Bx | I can't quite remember how it works, but most 020 numbers are just london numbers... |
18:52.34 | RaYmAn-Bx | and hence at normal cost |
18:52.39 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
18:52.41 | Zeeek | 3p ? |
18:52.54 | RaYmAn-Bx | but I remember something about there being special numbers in that range...not sure though |
18:53.02 | RaYmAn-Bx | same rate as calling any other UK number |
18:53.17 | RaYmAn-Bx | (except for 07,08,09 numbers) |
18:53.23 | Zeeek | Last time I looked itr wasn't but I think we killed this horse once already :) |
18:53.49 | RaYmAn-Bx | actually, it's only 01XX and 02XXX numbers that are definitely cheap (and of course 0800) |
18:53.53 | RaYmAn-Bx | okay |
18:53.53 | feklee | Zeeek: It seems to work now. |
18:53.59 | RaYmAn-Bx | and yeah, we did |
18:54.04 | Zeeek | Great! I'll sleep better tionight |
18:54.22 | Zeeek | well, gotta run maybe I'll check the conf in a while |
18:54.46 | Zeeek | no one there yet |
18:54.51 | Zeeek | later folks |
18:55.02 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
18:55.11 | feklee | Zeeek, RaYmAn-Bx: The reason for the problem was almost too stupid to be true: I used the wrong password (Sipgate password instead of SIP-Passwort) |
18:55.19 | feklee | Typical beginners problem. |
18:55.22 | FarrisG | D-channel should be 11, but it keeps truing to open the default 24. How do I make certain it uses 11 instead of 24? |
18:55.38 | shido6 | set it in zapata.conf FarrisG |
18:55.51 | *** part/#asterisk kodomo (~memyself@emu.net.informatik.tu-muenchen.de) |
18:55.58 | *** join/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com) |
18:56.27 | FarrisG | shido6: Where is zapata.conf? |
18:56.37 | FarrisG | shido6: Sorry, I meant where IN zapata.conf\ |
18:57.34 | shido6 | "/etc/asterisk/zapata.conf" |
19:00.03 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr) |
19:00.21 | PoWeRKiLL | Got to unlock the PAP2 ! Now working on the GP2 :) |
19:00.25 | mitcheloc | FarrisG: i missed the conversation, you can probably just add it "setting=value" anywhere in zapata.conf |
19:01.59 | FarrisG | mitcheloc: I tried that, but (sorry, total n00b here) "dchannel" doesn't seem to be a valid setting. It got ignored when asterisk parsed zapata.conf |
19:02.33 | FarrisG | The odd thing is that this was working fine yesterday, and no conf has been changed. So maybe some module or init didn't get loaded/run when the server mysteriously rebooted last night? |
19:03.42 | *** join/#asterisk Tili (~Tili@202-133-67-103-dialup.sat.net.pk) |
19:04.23 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
19:05.50 | *** join/#asterisk Tr0j4N (b3nz3r@pcp08761169pcs.mtlrel01.nj.comcast.net) |
19:07.08 | Tr0j4N | Hello all, n00b asterisk user here. Just came by to see if anyone's talking about anything interesting. |
19:07.40 | Tr0j4N | btw, if any of the asterisk devs are here, just wanted to give a BIG THANKS for your work! |
19:09.31 | jontow | stick around, you'll find interesting conversation.. this is one of the most active channels im in, overall :) |
19:09.40 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com) |
19:10.05 | jontow | if you wish to follow any of it make sure you have a nice long scrollback buffer in your irc client ;) |
19:10.59 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
19:11.39 | FarrisG | Ok, I think I may know what the problem is now, but still don't know how to fix it |
19:11.43 | mitcheloc | does anyone in here know .net? |
19:11.44 | FarrisG | zaptel isn't being initialized |
19:11.49 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
19:12.05 | mitcheloc | are you modprobing your cards? |
19:12.07 | jontow | farrisg; had that one last night, amongst other things.. did you install zaptel before compiling asterisk? |
19:12.10 | harryvv | I am sure alot of people are having a good time at von. My asterisk hd failed last night so now trying to install it on a new system with fd3 instead of debian. Having a issue finding the nessesary libraries that were not installed on this system. fun fun. |
19:12.46 | jontow | harryvv; if you have a bit of time and patience, i'd give gentoo a go |
19:12.52 | jontow | .. it really works well for me |
19:13.06 | harryvv | jontow, to late fdc3 is installed. |
19:13.26 | jontow | ;) np, just giving a third party opinion |
19:13.33 | harryvv | This is what some of the tutorial authors used So im sticking with what a majority use. |
19:13.36 | jontow | give it a try sometime when you've got a moment |
19:13.54 | harryvv | btw, for the most part really like fdc3 is was a smooth install. |
19:14.13 | jontow | gentoo is much more of a manual process.. but its well documented |
19:14.18 | harryvv | just this system is a workstation/server and deserves raid :) |
19:15.35 | PTG123 | suse :) |
19:15.44 | PTG123 | i tried gentoo, after 8 hours of install process |
19:15.45 | PTG123 | it failed |
19:15.52 | PTG123 | suse works great |
19:15.55 | PTG123 | and is easy to use |
19:15.57 | jontow | wow |
19:16.01 | jontow | i've never had an 8 hour install |
19:16.06 | PTG123 | oh man |
19:16.11 | PTG123 | it ftp'd and compiled everything |
19:16.12 | jontow | try the universal cd image :) |
19:16.13 | PTG123 | was insane |
19:16.21 | jontow | you don't need to start from scratch unless you have particular needs |
19:16.27 | PTG123 | hah |
19:16.31 | PTG123 | well i'll stick with suse |
19:16.32 | RaYmAn-Bx | just do a stage3 install and you're done quite quickly |
19:16.33 | PTG123 | and a mouse click :) |
19:16.39 | jontow | it is INSANE if you start from scratch |
19:16.46 | jontow | i've yet to complete one of those.. :P |
19:16.51 | PTG123 | suse detected all my hardware great, etc |
19:16.52 | RaYmAn-Bx | I did a stage1 install on my server |
19:16.52 | PTG123 | heh |
19:17.05 | PTG123 | maybe no one ever did a stage1 |
19:17.08 | PTG123 | thats why it faile d;) |
19:17.12 | jontow | if you're building all binaries with optimization on purpose to squeeze the last amount of performance |
19:17.20 | jontow | then stage1 is a good idea |
19:17.40 | jontow | but honestly.. the machines i work with aren't the highest-end servers |
19:17.42 | PTG123 | i just wanted my computer to work |
19:17.50 | jontow | and compiling shit for 3 days isn't always my favorite thing |
19:17.56 | PTG123 | i am too use to freebsd, why can't it be that easy |
19:18.10 | Nugget | freebsd is tasty. |
19:18.23 | jontow | freebsd is what i use everyday on so many machines |
19:18.34 | IceBerg | I cant figure out why asterisk's console says it's playing the files for the demo extension, but x-lite gets a reorder signal |
19:18.53 | jontow | which is why im a fan of gentoo :) if im forced to be using linux; then a system as comprehendable as gentoo works for me :) it probably will not for everyone |
19:19.06 | harryvv | iceburg do a asterisk -vvvvvvvvvvgc then if its sip do a sip debug |
19:19.18 | shido6 | IceBerg |
19:19.30 | PTG123 | so then why even use linux :) |
19:19.39 | shido6 | show me what you have for that softphone in sip.conf at pastebin.ca, IceBerg |
19:19.45 | harryvv | its nice to see asterisk show every sigle step on the cli and debug it that way. |
19:20.01 | IceBerg | shido6, ok, one sec |
19:20.27 | *** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com) |
19:21.49 | IceBerg | shido6, http://pastebin.ca/6935 |
19:22.20 | psywar | has anyone else had problems with using "toast" to convert .wav or .au to .gsm? |
19:22.26 | psywar | it makes loud static for me. |
19:22.46 | shido6 | where is your context IceBerg ? |
19:26.27 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
19:26.33 | shmaltz | tzanger you around? |
19:26.39 | shido6 | what context do you want to use for this phone, IceBerg |
19:26.39 | shido6 | ? |
19:27.05 | shmaltz | anybody here using slackware? |
19:27.36 | jontow | because lets face it.. * doesn't run as well on freebsd yet ;) |
19:28.28 | jontow | all of my servers / desktops / laptop run freebsd or netbsd or openbsd .. not linux (excepting a webserver implemented before I got the job and my * servers) |
19:29.50 | *** part/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com) |
19:34.57 | jontow | i do have * on free and netbsd successfully though |
19:38.43 | *** part/#asterisk NatRH (~Nat@dargo.trilug.org) |
19:39.29 | harryvv | does it make a difference what version of gcc is installed when compiling asterisk |
19:40.08 | *** join/#asterisk Tili (~Tili@202-133-65-168-dialup.sat.net.pk) |
19:42.02 | Trepalium | It probably matters for the zaptel kernel modules, since they have to match the kernel's compiler version. Other than that, no idea. |
19:42.21 | harryvv | mmm okay well im getting some compile issues with make |
19:43.02 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
19:44.20 | psywar | what other values does Monitor accept as the 1st arg |
19:44.23 | psywar | instead of wav |
19:46.47 | shido6 | show application Monitor |
19:47.04 | shido6 | harryvv what do you get? |
19:47.08 | shido6 | pastebin.ca them |
19:47.12 | vaewyn | interesting... anyone else run into a SIP device that works just fine... but you can't qualify against it reliably? |
19:48.21 | *** join/#asterisk NatRH (~Nat@dargo.trilug.org) |
19:48.47 | *** join/#asterisk viLeR (1000@ip-33-104.telesat.com.co) |
19:51.30 | Moc | I never got good qualify using SIP |
19:52.02 | *** join/#asterisk letherglov (~robbie@8036aa59.resnet.ucsd.edu) |
19:52.49 | vaewyn | This is just freaky.. i am on the phone and seeing Peer '6103IP5000' is now UNREACHABLE! Last qualify: 1529 |
19:53.06 | vaewyn | So I'm like "the @#$@#$ it isn't!" :} |
19:53.11 | Nugget | heh |
19:53.19 | letherglov | is the phone locking up? |
19:53.22 | vaewyn | nope |
19:53.28 | vaewyn | runing just great |
19:53.32 | letherglov | does it become reachable when you pick up the handset? |
19:53.35 | letherglov | or do you get a fast busy? |
19:53.48 | letherglov | mabe it's just sleepy ;-) |
19:53.48 | vaewyn | is a wireless... so no handset :} |
19:53.58 | letherglov | low-power mode or something |
19:54.11 | letherglov | but that doesn't help you so much, because it'll never take incoming cals if it's unreachable |
19:54.12 | vaewyn | It does it even when i am talking on it... |
19:54.37 | vaewyn | that's the funny part... if i turn qualify off it always takes the calls just fine... with it on * freaks out |
19:55.06 | Moc | just dont use qualify using SIP |
19:55.15 | Moc | qualify using IAX seem ok |
19:55.24 | vaewyn | was seeing if it would help a registration drop problem I am having at one site |
19:55.56 | Moc | vaewyn, exactly, remote qualify, it what I did, no more stupid problem |
19:56.03 | Moc | remote = remove |
19:56.03 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
19:56.16 | Moc | until someone fix qualify in SIP, it pretty useless |
19:56.58 | Moc | just reduce the registration time |
19:57.02 | Moc | your behind a na |
19:57.03 | Moc | t |
19:57.10 | vaewyn | nope... no NAT |
19:57.26 | vaewyn | WAP11 -> Linux router for subnet -> * server |
19:57.58 | vaewyn | I am actually thinking it may be the WAP11 piece of junk |
19:58.05 | Moc | well wap doesnt help |
19:58.24 | Moc | but in normal opperation, what is your average qualify MS when you show sip show peers |
19:58.32 | vaewyn | not WAP protocol... linksys WAP11 Access point |
19:58.39 | vaewyn | 34ms or less |
19:58.46 | vaewyn | if it is there |
19:59.07 | Moc | and what is your real ping with the device ? |
20:00.15 | vaewyn | 9 > 18ms |
20:00.56 | Moc | your qualify should show about the samething as a ping |
20:01.09 | vaewyn | well... it isn't |
20:01.12 | vaewyn | :} |
20:01.18 | Moc | like IAX does... but SIP qualify have a problem and sometime take a while before he see the answer |
20:01.40 | Moc | I get 1MS qualify with a IAX connection, but with the same system using SIP, I get 30 |
20:01.46 | Moc | and sometime 2690 |
20:01.50 | vaewyn | heh |
20:02.06 | Moc | it broken in sip, and until someone fix it, it will stay that way |
20:02.19 | Moc | but maybe you should post a bug with info on the tracker |
20:03.14 | Moc | this problem exist for over a year now hehe |
20:04.14 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
20:11.00 | *** join/#asterisk revjim (~revjim@24.32.18.59) |
20:11.25 | shmaltz | anybody here using slackware? |
20:11.44 | Nugget | I do. |
20:12.09 | Nugget | I sort of felt obligated to. It's not too bad if you install minimally, but it has some really stupid ideas about where things belong if you do a full install. |
20:12.48 | Nugget | you can also expect to get asterisk compile errors if you install slackware without x11. |
20:14.35 | shmaltz | Why is that? |
20:14.39 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
20:15.47 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr) |
20:16.15 | Nugget | because the slackware installer is stupid and the asterisk makefile is optimistically short-sighted. |
20:16.30 | PoWeRKiLL | last stable cvs are broken since I upgrade last week I got one or two time a day Too many open files ! any idea ? |
20:16.37 | Trepalium | Well, at least D.J. Bernstein didn't write slackware. Everything might've ended up in /var or something. |
20:16.37 | tuxinator_linux | optimistically short-sighted, that's funny |
20:16.42 | Nugget | if you install slackware without x11 it will still install the gtk libs. |
20:17.02 | Nugget | and the asterisk makefile uses the presence or absence of the gtk makefiles to decide whether or not to build the gtk gui console stuff. |
20:17.22 | Nugget | so it tries valiently to build the gtk console which fails miserably because x11 isn't present |
20:18.03 | Nugget | Trepalium: funny you should say that. one of my big beefs with slackware is that it installs all of apache to /var/lib/apache. |
20:18.07 | shmaltz | but why only with slackware does it have this problem? |
20:18.31 | Nugget | because no other linux (that I'm aware of) is stupid enough to install gtk libs on a machine that doesn't have x11 |
20:18.38 | Trepalium | Slackware has no dependancy checking, which is how GTK could be installed without X11 libs. |
20:19.14 | shmaltz | so how do I make sure that slackware doens't install the gtk library? |
20:19.15 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-140.dsl.scarlet.be) |
20:19.29 | Nugget | remove it after you're done installing. |
20:19.29 | Nugget | or install the x11 libs. |
20:19.31 | shmaltz | how do I remove it? |
20:19.38 | Nugget | with the slackware package tools. |
20:19.45 | shmaltz | thanks |
20:19.59 | shmaltz | is there any advantage of using 2.6 with asterisk? |
20:20.02 | Trepalium | Nugget: That is a pretty strange place to put Apache. |
20:20.06 | shmaltz | or can I leave it with 2.4 |
20:20.10 | tuxinator_linux | shmaltz: I'm using it |
20:20.20 | Nugget | Trepalium: indeed. I feel like I need a shower whenever I have to type "/var/log/apache/bin/apachectl stop" |
20:20.22 | shmaltz | tuxinator_linux, what are you using? |
20:20.25 | Nugget | s/log/lib/ rather |
20:20.31 | tuxinator_linux | I make a wiki page http://www.voip-info.org/wiki-Asterisk+CentOS-4.0+Zaptel |
20:20.45 | Nugget | fucking linux. I hate linux. |
20:20.56 | shmaltz | tuxinator_linux, we are taling about slackware |
20:21.17 | shmaltz | Nugget, is there any reason to use 2.6 instead of 2.4 for asterisk? |
20:21.31 | Nugget | shmaltz: I have no opinion. I'm waiting until 2.6 is stable. |
20:21.37 | mitcheloc | isn't 2.6, newer, more secure, more stable? |
20:21.40 | mitcheloc | or i guess not |
20:21.44 | Nugget | 2.6 is not stable release yet. |
20:21.51 | Sedorox | I find it stable... |
20:22.02 | mitcheloc | it's stable so long as you don't through mythtv at it heh |
20:22.03 | Trepalium | Maybe never will be with this new 'development model' for 2.6 |
20:22.05 | tuxinator_linux | It has been stable for me so far, but I don't do much outside the norm |
20:22.08 | Nugget | 2.6.8 has some crippling networking bugs. I've never tried 2.6.9 or newer. |
20:22.15 | Nugget | but it's not stable yet. so says linus, at least. |
20:22.42 | tuxinator_linux | and I am not using X |
20:22.55 | Nugget | too much flux and change in 2.6 development to use on production boxes, imho. |
20:23.16 | Sedorox | I'm running 2.6.10-cko3 right now.. with Reiser4... with X, ATI Binary.. etc.. on my personal laptop.. and everything is great |
20:23.30 | Nugget | the plural of anecdote is not "data". |
20:23.46 | Nugget | 2.6 is not stable release yet. any success you are having with 2.6 is both fortunate and accidental. |
20:24.04 | `Sauron | 2.6 isn' |
20:24.08 | `Sauron | isn't stable? |
20:24.08 | tuxinator_linux | 2.6.9 here |
20:24.13 | `Sauron | Hummrh. |
20:24.15 | vaewyn | 2.6 is stable |
20:24.17 | Nugget | no, 2.6 is not stable release. |
20:24.21 | vaewyn | and supported by * |
20:24.24 | Nugget | that's straight from linus. |
20:24.25 | Trepalium | 2.4.x wasn't stable until well past 2.4.14, so I guess I can't complain... much. |
20:24.27 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
20:24.32 | Nugget | 2.6 is a development branch |
20:24.37 | tuxinator_linux | define stable |
20:24.40 | tuxinator_linux | ~stable |
20:24.41 | jbot | extra, extra, read all about it, stable is the status of a Debian release when no packages will be added or changed unless a security fix is needed, or sta-ble adj; uptime in excess of 365days, or where the horses live The current stable version of Debian is woody (3.0). |
20:25.00 | `Sauron | I don't know that linux ever won't be a development branch. |
20:25.19 | *** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net) |
20:25.21 | Nugget | `Sauron: it used to have a stable branch. with 2.6 that all flew out the window. |
20:25.22 | vaewyn | Nugget: 2.6 is both... read back some of the kerneltrap archives |
20:25.38 | `Sauron | Nugget: I know. Sucks, if you ask me. |
20:25.42 | Nugget | I agree |
20:25.56 | `Sauron | Bring back the even/odd numbered version thing. |
20:26.00 | harryvv | typing bison on the command line show result in a responce? I have it installed but getting command not found when typing bison. |
20:26.00 | Nugget | yeah, exactly. |
20:26.06 | Nugget | that system was working well |
20:26.16 | vaewyn | hate to say it... but they are correct... with something that huge getting every part "stable" is a joke |
20:26.30 | tuxinator_linux | harryvv: check path |
20:26.35 | Trepalium | It was supposedly done to reduce the number of patches Red Hat, and similar distributors put in their own kernels, but I think it's encouraging it more than anything else now. |
20:26.49 | tuxinator_linux | try 'find / -name (name of prog)' |
20:27.03 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) |
20:27.10 | harryvv | I install bison rpm on /var/lib/rpms |
20:27.25 | Blackvel | what are the latest features in asterisk 1.0.6? |
20:27.27 | harryvv | but typing bison on the command line echos no command found. |
20:28.37 | tuxinator_linux | harryvv: I'm not familiar with bison, sorry |
20:29.36 | harryvv | odd, doing a find / -name bison echos no such file. |
20:30.07 | Trepalium | How about an 'rpm -qa bison' |
20:31.35 | harryvv | echos no reply |
20:31.52 | Trepalium | Then it's not installed. "rpm -Uvh rpmname.rpm" to install it. |
20:32.33 | harryvv | did, command not found after executing it. |
20:32.47 | `Sauron | Sigh. |
20:32.48 | harryvv | im going to find another copy |
20:32.53 | `Sauron | Hate the new nvidia driver |
20:33.01 | `Sauron | It doesn't clear the splash screen. |
20:33.26 | dfunnell | Hi all - having trouble with Chameleon VM where it doesn't seem to be registering keys pressed by the user (i.e. mailbox and password). Any ideas? |
20:34.57 | harryvv | meridian voicemail df |
20:35.40 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
20:37.33 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-140.dsl.scarlet.be) |
20:37.49 | dfunnell | harryvv: Hi, was that for me? If so can I get you to elaborate a little? D> |
20:37.49 | harryvv | Trepalium, found the issue. The bison i downloaded was one charicter off. |
20:39.08 | harryvv | not really you need to read the wikis. ohh take a look at tx/rx and also if its a ata look at the dialplan if its a sipura ata. |
20:40.05 | shido6 | dfunnell yes |
20:40.09 | shido6 | dtmfmode dfunnell |
20:40.15 | shido6 | u need to set the proper dtmfmode |
20:44.51 | dfunnell | Ok, thanks, looking in to it now. dtmfmode seems to work (with external IVR), just not for VM. |
20:45.01 | harryvv | shido it could also be what i mentioned since i had the same issue. if his tx is so low that the asterisk box cannot hear it then its moot. |
20:45.22 | harryvv | df, what asterisk client are you using |
20:46.01 | harryvv | df, has anyone complained of "you sound far away" when dialing though the asterisk to somone else? |
20:46.35 | Tr0j4N | what are all the v's for in the command " |
20:46.43 | Tr0j4N | asterisk -vvvvvvvvcg |
20:46.46 | Nugget | verbose logging. |
20:46.46 | znoG | verbose |
20:46.50 | Nugget | each v makes it more chatty |
20:46.52 | Tr0j4N | how many do I need? |
20:46.59 | Tr0j4N | oh, ok |
20:47.01 | znoG | all the ones that you want |
20:47.05 | Tr0j4N | thx muchly |
20:47.53 | Tr0j4N | can I just take any old house phone and plug it into an fxo port? |
20:48.25 | dfunnell | shido6: you are a star, works, thanks. Had set it already, but noticed I had a mistake in config. |
20:49.03 | shido6 | :) |
20:49.22 | psywar | how can I convert gsm<=>wav? toast makes garbage files. |
20:49.42 | psywar | wait this is in the FAQ isn't it |
20:49.45 | psywar | nm |
20:50.19 | Nugget | wow. that's the first time that's ever happened! :) |
20:50.57 | harryvv | df, what was the config issue |
20:51.00 | Tr0j4N | nevermind, I got this tdm4000p card but it's only got RJ45 ports on the back and my phones are all RJ11 |
20:51.25 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
20:51.44 | *** join/#asterisk iceberg (iceberg@cpe-24-166-0-83.indy.res.rr.com) |
20:56.41 | BrianR___ | Tr0j4N: You can plug any POTS phone into a FXS port... |
20:57.02 | BrianR___ | Tr0j4N: If you're careful, male RJ-11 connectors will fit into a female rj45 |
20:57.07 | Tr0j4N | hmm, yes I was just reading that |
20:57.35 | Tr0j4N | the card has a 2 FXO modules and FXS modules on it so in theory I should be ok. |
20:58.07 | Tr0j4N | however, I tried plugging the RJ11 into the port numbered 1 and I ran zapcfg but it had an error |
20:58.19 | BrianR___ | the way it's usually done is to connect to the PSTN with a bunch of FXO's, a T1/PRI, or VOIP gateway service. Then you use Analog Terminal Adaptors (ATA's) to provide FXS ports for any legacy POTS devices you might have. |
20:58.37 | BrianR___ | Tr0j4N: One thing that you might find very confusing at first is that FXO ports use FXS signalling and vice versa. |
20:59.04 | Tr0j4N | thx, it looks like I have some reading to do. |
20:59.08 | BrianR___ | Tr0j4N: So if ztcfg bitches about the signalling type being wrong for the card, that's probably why. |
21:00.04 | BrianR___ | Tr0j4N: The TDM400p provides very good FXS jacks for pots gear, but is a bit expensive and you can only fit so many of them into a PC. Past about 4 ports you want to use either a channel bank or ATA's, depending on your wiring situation. |
21:00.17 | Tr0j4N | should I see a green light on the port if something is plugged in right? |
21:00.32 | Tr0j4N | I just bought one for testing |
21:00.33 | BrianR___ | Tr0j4N: I don't know the answer to that particular question. |
21:00.44 | harryvv | troj, what do you have |
21:00.57 | Tr0j4N | a TDM400P digium card |
21:01.05 | iceberg | are there any good wireless IP phones that work with asterisk? |
21:01.08 | harryvv | cant help you there |
21:01.21 | Tr0j4N | np, I will get it. thx for help |
21:01.23 | BrianR___ | iceberg: A few.. The cisco 802.11 / SIP phones work... |
21:01.46 | iceberg | BrianR___ is it B or G? |
21:01.56 | BrianR___ | iceberg: B, I think. |
21:02.11 | iceberg | bummer, wont work on my wifi then |
21:02.15 | Tr0j4N | when I run modprobe wcfxo I get ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
21:02.16 | BrianR___ | You only need to be able to transmit at 7KB/sec worst case, so.. |
21:02.26 | BrianR___ | iceberg: You don't have dual mode AP's? |
21:03.14 | iceberg | I do but I require IPSec and Authentication and a minimum connect of 24kbps |
21:03.18 | psywar | I wonder if GSM compression would affect voice stress analysis like "liar liar" |
21:03.40 | psywar | (open-source VSA) |
21:03.57 | BrianR___ | iceberg: Dunno if the cisco phones do ipsec either... |
21:03.59 | iceberg | psywar yes |
21:04.04 | psywar | hrm |
21:04.09 | psywar | :-( |
21:04.39 | psywar | ipsec + voip is the bomb |
21:04.52 | iceberg | BrianR___ I guess for wireless I'll have to use POTS to Lan converters for now |
21:04.54 | *** join/#asterisk Damin_Mobile (~pocketirc@110.sub-166-155-107.myvzw.com) |
21:05.05 | BrianR___ | wish sips / srtp was more widespread too.. |
21:06.17 | *** join/#asterisk alegh (~ag10@OL217-17.fibertel.com.ar) |
21:06.23 | iceberg | are there any good yet inexspensive hard phones that support confencing? (key here being inexspensive) |
21:06.45 | BrianR___ | Could always use something like the Uniden UIP1868 too. It's essentially a 5.8ghz cordless station with a built in ATA. |
21:07.37 | iceberg | checking that out |
21:08.22 | iceberg | 5.8 would be nice, less interferance with my wifi |
21:08.22 | alegh | Hi, I'm looking for IP phones. Any recomendations about Sipura, Clipcomm or ArtDio? |
21:09.23 | Goshen | alegh: if you are looking for IAX phones I know iaxtalk.com has them but they ship from China |
21:09.43 | Goshen | iceberg: grandstream? |
21:09.54 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
21:10.04 | iceberg | Goshen what do you mean? |
21:10.14 | Goshen | iceberg: http://voipstore.pulver.com/product_info.php?products_id=33&osCsid=c428ece0900381122735332294d7001f |
21:10.36 | Goshen | black grandstream phone...I have one, it works great, and the speakerphone is nice and LOUD(adjustable) |
21:10.40 | *** join/#asterisk DonX (~don@adsl-69-155-217-211.dsl.rcsntx.swbell.net) |
21:10.41 | Goshen | comes in white too |
21:11.11 | iceberg | Goshen , nice thanks |
21:11.26 | BrianR___ | The grandstream gxp-2000 looks like it's going to be sweet. POE, 4 line appearances, and in the ~$120 price range.. |
21:11.30 | Goshen | welcome :) |
21:11.38 | DonX | I'm having strange problems with MOH on xlite. Everything seems to be working fine but music on hold is choppy on the soft clients. My ATA-186's and 7940's are working fine |
21:11.53 | DonX | anyone seen this issue before? |
21:12.11 | iceberg | UIP1868 doesnt exist on Uniden's site |
21:13.42 | *** join/#asterisk fitzel (~flint@p3EE39AF8.dip0.t-ipconnect.de) |
21:13.49 | fitzel | Hello |
21:13.51 | BrianR___ | The ZyXEL 2000W has 802.11 b and g |
21:14.38 | fitzel | I am playing around with iaxclient and speex. xlite with speex sounds nice, but iaxclient sounds really like I have my head in a tincan. |
21:14.45 | iceberg | WOW that baby is exspensive |
21:14.57 | fitzel | Any settings that are recomendable? |
21:15.53 | Goshen | iceberg: I like my grandstream...where there is voicemail waiting the display flashes letting you know |
21:16.18 | Damin_Mobile | Sitting in the airport drunking beer, waiting for my flight. |
21:16.25 | BrianR___ | Goshen: Have you been able to get your hands on a GXP2000 yet? |
21:16.32 | BrianR___ | Everywhere I look they're backordered... |
21:17.28 | vaewyn | Damin_Mobile: going to VON? |
21:17.46 | Damin_Mobile | Yep... |
21:18.24 | vaewyn | I've got the 6 am flight in the morning |
21:18.37 | Goshen | BrianR: I haven't heard of the GXP2000 yet, have a link? |
21:18.40 | *** join/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk) |
21:18.41 | vaewyn | nothing like 6+ hours in the sky to get you bored |
21:19.34 | alegh | Any experience with clipcomm CP-100D? |
21:19.43 | Damin_Mobile | vaewyn: Email me your cell # and we can hook up at San Jose. |
21:19.50 | SirPrize | Is there a way that I can send incoming calls to Voicemail, but in such a way that if *I* dial-in on the same number, that I can enter my PIN to listen to my voicemail instead? |
21:20.11 | Damin_Mobile | damin@nacs.net |
21:20.28 | vaewyn | Damin_Mobile: cool... will do |
21:20.36 | Goshen | SirPrize: I have been woundering that too |
21:21.29 | RaYmAn-Bx | SirPrize: if you always dial in from the same number, then yes, definitely..Otherwise it might be possible...It should be at least |
21:21.30 | SirPrize | My only thought so far was to set up a two-second silence through background, and a timeout of 2 seconds, and on the timeout dialplan, do the "Leave Voicemail" path |
21:21.41 | vaewyn | SirPrize: You can do a "super secret extension" an use background to play the greeting message instead of letting VM do it |
21:22.08 | SirPrize | RaYmAn-Bx, Ideally, I want to dial in from multiple numbers |
21:22.28 | vaewyn | Or use the '0' for operator and dump into a context with the super-secret extension |
21:22.38 | SirPrize | vaewyn: so there's no 'built-in' method at the moment ? could then put an Authenticate on that super-secret extension too |
21:23.00 | RaYmAn-Bx | SirPrize: then you might be able to make it so you can press an extension while playing the "please leave a message" message |
21:23.01 | vaewyn | Only buuilt in is the '0' operator path |
21:23.39 | SirPrize | vaewyn: Could you please explain a bit more about the '0' operator path? I didn't know there was such an extension, or that it had special capabilities |
21:23.46 | Damin_Mobile | Signing off |
21:24.25 | SirPrize | Is that the 's' extension ? |
21:24.53 | BrianR___ | Goshen: http://www.voipsupply.com/product_info.php?cPath=95_111&products_id=331 |
21:26.17 | vaewyn | SirPrize: pressing '0' in most points in the VM will send you to the 'o' extension in the current dialplan context |
21:26.48 | vaewyn | 'o' for operator |
21:27.26 | SirPrize | cool, I'll read up on that |
21:27.41 | SirPrize | Where could I find a ChangeLog for 1.0.6? |
21:28.51 | Sedorox | ont he FTP Server |
21:29.35 | SirPrize | stupid me - should have looked there first. Was wandering around the web site looking for one |
21:31.22 | *** join/#asterisk mtmachen (~matthewma@cable-68-113-71-35.grd.al.charter.com) |
21:31.42 | Sedorox | :-p |
21:33.54 | *** join/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk) |
21:34.35 | SirPrize | Might this be a place to ask about an issue I'm having with a Sipura SPA 3000, or could anyone suggest a better place to ask in ? |
21:35.07 | SirPrize | the issue isn't at all related to Asterisk, so I know it would probably be ot here |
21:35.14 | shido6 | this is the place |
21:35.26 | shido6 | you have a user configured for the spa? |
21:35.26 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
21:35.34 | shido6 | and a user and peer in sip.conf? |
21:35.41 | SirPrize | Well, in my Line1 dialplan, I've got it forwarding all calls to Asterisk |
21:35.50 | SirPrize | Sorry, start again |
21:35.53 | shido6 | hehe |
21:36.04 | SirPrize | Well, in my Line1 dialplan, I've got it forwarding all calls out via PSTN on the PSTN line |
21:36.26 | SirPrize | if I want to make a VoIP call, that has to be accessed by dialling #9 first |
21:36.44 | SirPrize | this works fine - I can make standard PSTN calls, as well as VoIP calls via #9 |
21:37.06 | SirPrize | the problem is with the speed-dials - they don't work in this configuration |
21:37.20 | SirPrize | if my dialplan simply says (xx.|*xx.), then the speed dials work |
21:37.36 | SirPrize | But I don't want the default out being the VoIP line |
21:38.00 | SirPrize | hitting a speed dial button will give a UK BT recording of "That number or option is unavailable" |
21:38.16 | SirPrize | any ideas why the dial-plan is affecting the speed dials? |
21:38.38 | SirPrize | have programmed the speed dials with numbers that CAN be dialled properly from the keypad when my own custom dialplan is installed |
21:40.58 | Goshen | I need a sipura spa 2100 dialplan howto... |
21:42.01 | SirPrize | there's a general Sipura one here: http://voxilla.com/forum-viewtopic-t-619.html |
21:42.29 | shmaltz | tzanger, you around? |
21:42.34 | shmaltz | ~seen tzanger |
21:42.36 | jbot | tzanger is currently on #asterisk. Has said a total of 748 messages. Is idling for 20h 21m 41s |
21:42.43 | SirPrize | Goshen: there's a general Sipura one here: http://voxilla.com/forum-viewtopic-t-619.html |
21:42.57 | dfunnell | Does anyone make use of DDI numbers with BRI ISDN and *? Trying to find best way of running it, but best I can find is DNID and wiki page (http://www.voip-info.org/wiki-DNID) doesn't make it sound very reliable. Any ideas? |
21:42.58 | Goshen | SirPrize: great thank you |
21:43.38 | RoyK | <PROTECTED> |
21:43.46 | RoyK | ~seen zoa |
21:43.47 | jbot | zoa <~zoa@ip-212-239-162-26.dsl.scarlet.be> was last seen on IRC in channel #asterisk, 2d 8m 17s ago, saying: 'brian!!!!'. |
21:48.12 | *** part/#asterisk SirPrize (~roshan@pc016.dcs.kcl.ac.uk) |
21:48.34 | Damin_Mobile | ady for takeoff |
21:49.13 | vaewyn | Bon voyage! |
21:49.51 | vaewyn | hehehe... I wonder if we can get his plane grounded by typign in IRC 'bomb voyage' :P |
21:50.15 | *** join/#asterisk DarkFlib (darkflib@dialup357.ts002.bmt.esat.net) |
21:50.20 | vaewyn | oops... i said the 'b' word |
21:50.39 | *** join/#asterisk NirS (Nir@l192-117-110-178.cable.actcom.net.il) |
21:52.23 | *** join/#asterisk ^HeLL^ (~admin@85.137.127.182) |
21:52.30 | ^HeLL^ | hello all |
21:53.45 | *** part/#asterisk DonX (~don@adsl-69-155-217-211.dsl.rcsntx.swbell.net) |
21:55.11 | Tr0j4N | woohoo got my tdm400p working! (had to run modprobe wctdm instead of modprobe wcfxo) |
21:55.26 | Goshen | congrats :) |
21:56.00 | Tr0j4N | thx for all your help. in a few months I should be able to help some fellow n00bs myself. |
21:59.35 | Moc | ;) |
22:00.25 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
22:00.42 | mikegrb | doubtful |
22:01.55 | TheBear | If I have my * server and standard phone connected to the same phone line. and have a SIP Phone connected on my network. Can I answer an incoming call on a standard phone, and then (a) redirect (b) pick up this call on the SIP Phone ? |
22:02.38 | TheBear | Anyone know ? |
22:02.55 | shido6 | yes |
22:03.09 | shido6 | fxs card in the * box? |
22:03.15 | shido6 | or are you connecting an fxo card |
22:03.17 | shido6 | to the same line |
22:03.20 | shido6 | as the analog phone |
22:03.22 | shido6 | with a splitter |
22:03.22 | shido6 | ? |
22:03.57 | Qwell | probably a splitter |
22:04.01 | Qwell | Thats what it sounds like |
22:04.54 | Qwell | * will probably pick it up right away |
22:05.08 | TheBear | shido6: using a splitter one side goes into the std phone the other into a Digium card |
22:05.09 | Qwell | If you want a normal phone, you should get an FXS port too |
22:05.15 | shido6 | oh lord |
22:05.17 | DarkFlib | depends if you have an answer before you dial the sip phone |
22:05.21 | shido6 | get a fxs card |
22:05.30 | shido6 | or use your softphone to answer or sip phone to answer |
22:05.48 | Qwell | fxs port is probably the best option |
22:05.54 | shido6 | or IAXy |
22:05.56 | shido6 | or Sipura |
22:06.03 | Qwell | which is also an fxs port ;] |
22:06.23 | shido6 | what kind of FXO do you have in the * ? |
22:06.27 | TheBear | SIP phone is a snom 200 |
22:06.33 | shido6 | modular or the x10xp ? |
22:07.35 | *** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net) |
22:08.06 | shido6 | what fxo device do you have for your asterisk system? |
22:08.13 | TheBear | digium card has 1 FXS module on the 4 port TDM card |
22:08.24 | Qwell | fxs port? You mean fxo |
22:08.38 | Qwell | at least...I hope you mean fxo |
22:08.48 | Qwell | otherwise, its not gonna work, no matter what you do. heh |
22:09.05 | TheBear | the mini daughter card says " Digium Ringing FXS module" |
22:09.15 | Qwell | yeah... |
22:09.16 | TheBear | I also have the FXO. |
22:09.31 | Qwell | Then you may wish to restate your question. |
22:09.35 | shido6 | ok both of em on the card? |
22:09.35 | Qwell | Why are you not using the FXS port? |
22:09.37 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
22:09.46 | shido6 | so u have 2 modules on the card? |
22:09.47 | shido6 | one fxo |
22:09.49 | shido6 | one fxs? |
22:09.56 | shido6 | or one fxs on the card and a x100p? |
22:09.59 | TheBear | 1 year back I had the * server connected to a phone line and a std phone connected into the * server. I could then make calls from the std phone via the * server to the phone line |
22:10.04 | Qwell | wall > fxo | fxs > analog phone | ethernet > SPA |
22:10.09 | TheBear | No two cards |
22:10.10 | da-manFL | hi shido6 |
22:10.29 | Qwell | TheBear: You have two TDMs, with one module each? |
22:10.45 | *** join/#asterisk DyOS (~me@ip68-2-153-157.ph.ph.cox.net) |
22:10.55 | *** join/#asterisk lesouvage (~chatzilla@cc341200-a.assen1.dr.home.nl) |
22:11.04 | *** join/#asterisk xlyz (~xl@81-208-36-176.fastres.net) |
22:11.22 | fitzel | Anybody know something about the digi datafire cards and how to make them run with linux and maybe even asterisk? |
22:11.32 | TheBear | trying to find the module now ..... |
22:12.24 | shido6 | sounds like an x100p |
22:12.45 | shido6 | as it has 2 ports, 1 fxo and 1 for adding another phone or whatever you like but they both cant use the line at the same time |
22:13.08 | Qwell | I think he's just confused. |
22:13.14 | shido6 | hehe |
22:13.42 | xlyz | hi! I've a little problem with musiconhold. the sound is awfully distorted. I tried to change mpg123 parameters but no luck. Any idea how can I fix it? |
22:16.33 | xlyz | anyone? |
22:18.30 | shido6 | using xwindows on that box, too? |
22:18.30 | fitzel | xlyz, do you use the 'real' mpg123? |
22:18.36 | xlyz | yep |
22:18.39 | shido6 | why? |
22:18.39 | Qwell | 0.59r? |
22:19.07 | xlyz | 0.59s |
22:19.15 | Qwell | bad |
22:19.19 | Qwell | get 0.59r |
22:19.53 | xlyz | thanks |
22:20.06 | TheBear | ok Card 1 has two ports RJ11 marked Line and Phone Digium S/N BJX116986. The other card has four ports RJ45 with a single daughter card marked "Digium Rining FXS Module". |
22:20.08 | Qwell | it may or may not fix your problem...but... |
22:20.28 | Qwell | rj45, on a pci? |
22:20.32 | TheBear | I know I connected a std. phone RJ11 into the second card before. And had * dial the std phone |
22:20.35 | Qwell | quadspan? |
22:21.26 | Qwell | The first one sounds like an x100p |
22:21.30 | TheBear | Qwell: I guess so it can take four daughter card one for each port I guess |
22:21.47 | Qwell | You're sure its rj45? |
22:22.05 | *** join/#asterisk in (int@2001:5c0:8fff:fffe:0:0:0:1d41) |
22:22.08 | TheBear | it's the same size as rj45 it is not as small as rj11 |
22:22.09 | in | hey |
22:22.18 | Qwell | Is it a TDM? |
22:22.24 | sedorox | TheBear: you have a TDM400P card I believe |
22:22.27 | TheBear | yes |
22:22.28 | sedorox | with one FXS module |
22:22.32 | sedorox | for regular phones |
22:22.32 | Qwell | Is the tdm rj45? |
22:22.36 | sedorox | yes |
22:22.38 | Qwell | oh |
22:22.41 | sedorox | hehe |
22:22.48 | Qwell | I guess that makes sense |
22:23.07 | iceberg | What are the best docs for a noob wanting to get Asterisk working for internal VoIP only? |
22:23.18 | Sedorox | My only guess (This confused Katty too) is that they just keep the RJ45's because of the ability to use the board for the TDM and for the T1 TDM cards... |
22:23.32 | Qwell | ahh |
22:23.34 | TheBear | lookin on the Digium website I would says it's a TDM10B |
22:23.35 | Sedorox | iceberg: voip-info.org has really good stuff on asterisk |
22:23.54 | iceberg | Sedorox, not really I followed the directions there and nothing works |
22:24.08 | Sedorox | yea.. looks right |
22:24.13 | Sedorox | iceberg: for sip.. or? |
22:25.00 | in | iceberg |
22:25.02 | in | hey man |
22:25.11 | Sedorox | I have asterisk setup.. with phones using SIP.. then IAX2 to other * boxes and other providers |
22:25.15 | iceberg | Sedorox, yea, when I try to dial 1000 from x-lite it waits, the console says everything is workign but then x-lite just gives me a reorder and I never get the voice info from the demo box and asterisk never closes the call |
22:25.17 | in | voip-info.org is all i have used really |
22:25.31 | Sedorox | hmmm |
22:25.33 | TheBear | Ok now having est. the cards If I split a phone line to the 1st card and a std phone. Can I answer a call on the std phone and then transfer this call to a SIP phone (snom 200) via the * server ? |
22:25.37 | Sedorox | check codecs |
22:26.02 | Sedorox | TheBear: yes |
22:26.23 | Sedorox | iceberg: I had problems with x-lite and codecs... I always moved gsm so it was first... |
22:26.30 | Sedorox | and make sure you have it enabled in sip.conf too |
22:26.45 | Sedorox | either in general.. or in the section for the login for x-ite |
22:27.03 | TheBear | Sedorox: care to explain a bit how, can I just pick a SIP phone and be connected to the call or what ? |
22:27.09 | dfunnell | Is anyone familiar enough with CAPI to know why when one channel of my BRI is in use Asterisk ignores the other one when it is ringing? Can dial-out using other channel, but just rings and rings when trying to dial in. I have incomingmsn=* in capi.conf and it does answer first channel, just not second. |
22:28.01 | lesouvage | I have .wav files that I want to convert to sox I read the info on the wike but it doesn't work. I got a message: bad input format for file foo.wav: data size 44 is invalid. What am I doing wrong |
22:28.05 | Sedorox | I'm not sure how to make it where you can take over a call once its been answered.. but you just set it up where Asterisk would take the call,, and then you could just ring out the other card for the std phone, and ring out a sip interface for the snom200, and you can either transfer between the two, or use call parking to switch phones |
22:28.35 | Sedorox | lesouvage: are you telling it special options for the input file? |
22:29.49 | xlyz | Qwell: 0.59r is not secure http://www.mpg123.de/ |
22:30.25 | xlyz | any other player / format that can be used in musiconhold? |
22:30.47 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
22:30.54 | lesouvage | sedorox: I just following the instruction from http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk I used: sox foo-in.wav -s -w foo-out.wav (with a name of an existing file) |
22:31.13 | Sedorox | hmmmm |
22:31.38 | TheBear | since it has been at least a year since I last used * and the SIP phones. Can I setup * where it will only handle calls between certain hours eg 8 - 4 otherwise it will ignore all calls ? |
22:32.08 | Sedorox | TheBear: yes.. its in the wiki somewhere.. I've seen it.. just not sure where... |
22:32.21 | Sedorox | lesouvage: I'm not sure.. I just converted mine to pcm using that guide.. and it worked... |
22:32.25 | marlowe | Sedorox: Give sox <in_file.wav> -r 8000 -c 1 <out_file.gsm> a try |
22:32.34 | Sedorox | haven't tried playing .wav's |
22:32.43 | Sedorox | marlowe: its for lesouvage :-p |
22:32.52 | marlowe | woops |
22:32.55 | TheBear | Sedorox: ok a wiki'ing I will go |
22:32.55 | Sedorox | thats the way I did mine |
22:32.57 | Sedorox | and it works |
22:32.58 | TheBear | thanks |
22:33.19 | Sedorox | TheBear: yup... I remember seeing it.. just not sure what page.. I think if you look in the example config parts.. there is one in there that has it.... |
22:33.27 | iceberg | does asterisk require a soundcard? |
22:33.53 | marlowe | iCEBrkr: Read the FAQ... But, no.. It doesn't. |
22:33.58 | Sedorox | iceberg: nope... unless you wanna do calls on the console... or do a PA System type thing... |
22:33.58 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
22:34.00 | marlowe | Ok nick completion sucks today |
22:34.22 | Sedorox | lol |
22:34.26 | Sedorox | mine always seems to suck :/ |
22:35.30 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
22:35.43 | harryvv | anyone ever see this when doing a zaptel compile on fc3 in the /usr/src/zaptel dir sorry, unimplemented: 64-bit mode not compiled in make: *** [gendigits.o] Error 1 |
22:36.10 | harryvv | make linux26 is generating this error. |
22:37.30 | TheBear | Is there somewhere the changes made to * over the last year esp. regarding NAT support and development ? |
22:37.57 | Sedorox | TheBear: changes should be in the changelog on the ftp server |
22:38.11 | harryvv | gendigits.c:1: sorry, unimplemented: 64-bit mode not compiled in |
22:38.12 | harryvv | make |
22:38.19 | harryvv | thats not good. |
22:38.33 | elric | can I have 3 groups within zapata.conf? |
22:38.36 | Sedorox | harryvv: you running a 64bit system? |
22:38.39 | harryvv | yes |
22:38.42 | harryvv | opteron system |
22:38.55 | elric | with three different contexts? |
22:39.01 | harryvv | so something needs to be compiled to run un 64 bit mode. |
22:39.09 | Sedorox | and did you install fc3 as 64bit.. or you running it in 32bit mode? |
22:39.28 | harryvv | yes I downloaded everthing to run as 64 bit. |
22:39.56 | harryvv | sed, but is there a way to know or find out? |
22:39.58 | Sedorox | what version of zaptel you compiling? 1.0.6? |
22:40.05 | Sedorox | I think uname -a will tell you |
22:40.17 | Nugget | I wasn't aware that zaptel would build on 64 bit linux. |
22:40.27 | Sedorox | thats what I'm not sure about... |
22:40.30 | harryvv | 2.6.9-1.667 #1 Tue Nov 2 14:50:10 EST 2004 x86_64 x86_64 x86_64 GNU/Linux |
22:40.33 | Sedorox | seems I wanna say no |
22:40.38 | Nugget | I'd be surprised to learn that it did. |
22:40.40 | Sedorox | and yes.. your running 64bit... |
22:40.51 | harryvv | i know :) its fast |
22:40.53 | Sedorox | hehe |
22:41.01 | Nugget | 64 bit doesn't make things run faster. |
22:41.04 | Sedorox | doesn't seem like you can do zaptel on 64... |
22:41.14 | Nugget | in reality is just means that your cache is only half as useful. |
22:41.16 | Sedorox | Nugget: have you used a 64bit system? |
22:41.17 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
22:41.18 | Nugget | sure. |
22:41.22 | Sedorox | ok.. |
22:41.29 | Sedorox | 'cause I'ce noticed a difference when I used one |
22:41.30 | Nugget | there's no inherent benefit to 64 bit. |
22:41.49 | Nugget | unless you have processes that need to access gobs and gobs of ram |
22:41.56 | Sedorox | eh |
22:42.02 | Nugget | opterons rock, but they rock with 32 bit os just as much |
22:42.05 | lesouvage | I have a set of sounds in Dutch but it's in wave format. Is it possible that they have a kind of protection on the .wav files. |
22:42.14 | Nugget | Ik ben een vliegende koe. |
22:42.27 | harryvv | opterons have the memory controler on the cpu running at cpu speed is one reasons. |
22:42.31 | Nugget | yeah |
22:42.45 | lesouvage | Nugget: hoi |
22:42.49 | harryvv | my bottle neck is the hard drive. |
22:43.03 | Sedorox | go SATA2 :-p |
22:43.08 | Nugget | go 3ware. |
22:43.09 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
22:43.11 | Sedorox | 3gb/s transfer |
22:43.14 | Sedorox | *drools* |
22:43.25 | Sedorox | yea.. 3ware sata2 controller |
22:43.26 | Trepalium | Better yet, go Ultra320 SCSI |
22:43.37 | Sedorox | Trepalium: slower then sata2 |
22:44.19 | Trepalium | Not when you can get 15,000 RPM drives. |
22:44.27 | lesouvage | Is there a list in writing of the sounds that comes with Asterisk |
22:44.27 | harryvv | sed...I lost my asterisk box last night the harddrive kept renitializing with hda: dma timmer_expirery dma status = 0x21 and did a reboot. even fsck -p could not repair it. |
22:44.47 | Sedorox | eh.. looks like they are on par... |
22:44.53 | Sedorox | nice :( |
22:44.55 | harryvv | yea |
22:45.13 | harryvv | I have my configs backed up on a windows box but thay are not current. |
22:45.28 | harryvv | anyway need to get this system up and running and get asterisk working again. |
22:45.33 | Sedorox | yea |
22:45.38 | harryvv | you said something about zaptel version? |
22:45.46 | lesouvage | Sedorox: how can I dial a numer from the asterisk prompt. |
22:46.02 | Sedorox | I wanna say the dial command.. but I'm not sure.. never did dialing from the cli |
22:46.13 | harryvv | ls, you cannot download xlite from xten |
22:47.17 | Sedorox | you can... |
22:47.22 | Sedorox | thats whereI get it from |
22:47.42 | Trepalium | I think there should've been a period in the middle of that statement. |
22:48.34 | Sedorox | hmmm |
22:48.47 | feklee | I'd like to stream an incoming phone call using Ices (interfaces with Icecast). Has anyone ever tried that successfully? |
22:49.14 | feklee | I've the following line in my extensions.conf: exten => 9779619,3,Ices(/home/feklee/asterisk/asterisk-ices.xml) |
22:49.16 | Sedorox | my guess... have it being recorded with monitor using wav.. and use the wav to stream... |
22:49.28 | Sedorox | hmm |
22:49.40 | feklee | But, I always get the error message "Execute of ices failed" when a call arrives |
22:50.03 | lesouvage | Nugget: Do you know about protecton messures for the .wave files from tric? |
22:50.13 | feklee | There's a tutorial for streaming a conference: http://www.voip-info.org/wiki-Asterisk+cmd+ICES |
22:50.15 | Nugget | who is tric? |
22:50.25 | feklee | But I just want to stream one call, not multiple ones. |
22:50.44 | Trepalium | The wav files may be already compressed with some other codec. |
22:51.36 | lesouvage | nugget: a dutch company who offers services around asterisk. They have dutch sounds for asterisk for download but I can't get them working. |
22:52.54 | harryvv | who here has asterisk running on a 64 bit system |
22:53.23 | *** join/#asterisk stifl3r (~nasty@xtreme-28-156.dyn.aci.on.ca) |
22:53.35 | Nugget | no clue, sorry. |
22:53.43 | *** join/#asterisk GaryH (~ghawkins@gromit.garysoft.co.uk) |
22:53.53 | *** join/#asterisk plc5_250 (~chatzilla@pcp01103028pcs.pntiac01.mi.comcast.net) |
22:54.32 | harryvv | well, next time * is installed its going to be on a system with raid. |
22:54.44 | harryvv | or just have two asterisk boxes side by side. |
22:54.52 | Sedorox | lol |
22:54.53 | Nugget | I don't think I'll ever build another machine without raid. it's just too cheap and too easy to not do it. |
22:55.35 | Sedorox | I've been pricing a computer for my parents.. pricing a Raid1 80gig SATA setup.. |
22:55.55 | Nugget | I've never had a drive failure that I wouldn't have happily and instantly paid $300 or whatever to not have had to deal with at the time it happened. |
22:56.07 | Nugget | but you can't buy raid /after/ the drive fails. :) |
22:56.19 | mikegrb | hehe |
22:56.27 | harryvv | nugget well this was installed on one of my systems that was cheap. now i can see that it should have had some redundency built in. |
22:57.15 | Trepalium | Just make sure you either get a high quality RAID controller (not Adaptec), or one so stupid nothing can go wrong with it (ATA fakeraid). |
22:57.46 | Sedorox | Do Not EVER get a Promise controller |
22:57.52 | Nugget | yes. |
22:57.58 | Nugget | promise controllers are awful |
22:58.19 | sivana | what's a good mfg then |
22:58.20 | Qwell | To be fair, they never do promise that it'll work |
22:58.22 | Nugget | I swear by those 3ware things. cheap, rock-solid drivers, and they're supported in any os. |
22:58.30 | Sedorox | hehe |
22:59.03 | Nugget | and there's a nice healthy range of performance, from the $99 two channel cards to the $900 dollar hardware cache zillion channel things |
22:59.19 | Trepalium | Don't get Adaptec ATA RAID controllers either. Those stupid PATA ones never seem to work unless you want to use the highpoint drivers instead of Adaptec's version. |
23:00.05 | plc5_250 | can someone help me with connecting via IAXTEL? |
23:00.24 | Sedorox | if I was building a system that needed the preformance and stability to stay up through naything.. I'll go 3ware.. otherwise.. I'll stick with the onboard stuff... normally VIA |
23:00.27 | PatrickDK | heh, I like the 2120S for my scsi raids, and use 3ware for ide bulkstorage |
23:00.35 | Sedorox | good enough for a faimly computer running raid1 |
23:00.51 | PatrickDK | hmm, they don't make onboard raid |
23:00.59 | elric | When I do a Background(file) it chops off a bit of the initial voice? but when I use PlayBack() it works just fine. |
23:01.01 | PatrickDK | that is software, not hardware |
23:01.07 | Sedorox | for sata they do... |
23:01.19 | Nugget | the onboard stuff is sufficient for saving data, which is certainly the more important point. they tend to fail miserably at keeping a machine up and running though -- for that goal there's 3ware. |
23:01.25 | harryvv | sed, i have a old promise controler that controled my iomega jaz for years. |
23:01.36 | Sedorox | hmmm |
23:01.56 | Nugget | I hate using raid that requires cooperation from the os, though. |
23:01.58 | Sedorox | ok... Promise PATA or Promise SATA controllers :-p I do admit the old scsi ones were nice |
23:02.02 | Sedorox | ditto |
23:02.06 | Nugget | that's the other big upside of 3ware |
23:02.09 | elric | so Playback() is "Welcome to ABC, Press 1 for Sales ..." Background() is " to ABC, Press 1 for Sales .... " |
23:02.11 | xlyz | musiconhold plays with very high distortion even with mpg123 0.59r. any other idea? |
23:02.21 | elric | anyknow what might be causing this? |
23:02.22 | xlyz | Qwell ? |
23:02.26 | Qwell | ? |
23:02.31 | Qwell | dunno |
23:02.36 | Qwell | but you should keep 0.59r |
23:02.44 | Trepalium | The volume of your MP3s might be too high. |
23:02.59 | harryvv | qwell, ever run asterisk on a amd 64 box |
23:02.59 | vaewyn | distortion usally means you have a sucky timebase |
23:03.10 | Qwell | elric: might want to do a Wait() first |
23:03.12 | xlyz | Trepalium: how do I set it? |
23:03.15 | Qwell | harryvv: can't say that I have |
23:03.28 | elric | Qwell, alright Wait(1) should work right? |
23:03.33 | Qwell | elric: probably |
23:03.34 | harryvv | i have googled my make linux26 error and have yet to see anything come up. |
23:03.50 | Qwell | You don't need to do make linux26 anymore, do you? |
23:03.57 | Qwell | I think the makefile does the check on its own now |
23:04.05 | Trepalium | harryvv: You have the x86_64 version of the kernel headers, right? |
23:04.16 | elric | ok testing. |
23:04.19 | elric | thanks Qwell |
23:04.27 | harryvv | well even with make it generates this error. gendigits.c:1: sorry, unimplemented: 64-bit mode not compiled in |
23:04.27 | harryvv | make: *** [gendigits.o] Error 1 |
23:04.28 | ManxPower | xlyz: use quietmp3: in musiconhold.conf |
23:04.51 | xlyz | I did, but no luck |
23:05.12 | harryvv | Trep I suspect I do but what is needed to verify this. Im running fedora core 3 with opteron system. |
23:05.21 | xlyz | now I'm using manual |
23:05.43 | harryvv | darn...company showed up now need to leave. If anyone has a answer message me. |
23:06.18 | plc5_250 | is IAXTEL having problems? It won't register me via * |
23:06.33 | Qwell | plc5_250: What error? |
23:06.38 | Qwell | I'm not getting any. |
23:07.28 | plc5_250 | doing an iax2 show registry takes multiple minutes to get to a "registered state" |
23:07.29 | GaryH | hi all, I have just upgraded to the 2.6.11 kernel from a 2.6.10 kernel and now the capi drivers don't seem to work any more. I don't know whether to blame asterisk,chan_capi or the Fritz CAPI drivers for this. I suspect not Asterisk as the same code (latest stable CVS) works fine on 2.6.10. Anyone else seen this? |
23:08.48 | plc5_250 | generally over 10 minutes. Can't seem to receive calls via IAXTEL either. |
23:10.05 | plc5_250 | Mar 6 19:24:31 WARNING[7703]: pbx.c:1945 ast_pbx_run: Timeout, but no rule 't' in context 'local' |
23:10.22 | plc5_250 | I know about the no "t" rule, but it always times out when trying to connect |
23:10.33 | plc5_250 | er... dial to myself |
23:12.04 | xlyz | anybody knows what value want mpg123 -g to reduce gain? I'm googling around and find nothing |
23:12.51 | *** join/#asterisk Tarox (~chris@pD9E7B7CA.dip.t-dialin.net) |
23:13.01 | ManxPower | 't' means "I didn't get enough digits to find a matching extension" |
23:13.48 | plc5_250 | I thought 't' was the timeout rule |
23:14.15 | ManxPower | plc5_250: It is. |
23:14.21 | ManxPower | 't' means "I didn't get enough digits to find a matching extension so I'm giving up" |
23:14.31 | plc5_250 | interesting... |
23:14.37 | ManxPower | i.e. I didn't get enough digits to make a match before DititTimeout happened. |
23:14.45 | plc5_250 | can someone try connecting to me via iaxtel? 7003432431 |
23:15.00 | ManxPower | plc5_250: IAXtel will let you call yourself. |
23:15.15 | plc5_250 | that is what I am trying and it is not working |
23:15.27 | ManxPower | plc5_250: Then it's not going to work for other people. |
23:19.09 | *** join/#asterisk mrgoby (~mrgoby@141.211.162.97) |
23:19.13 | plc5_250 | is there a test number on IAXTEL that I can use to see if my setup is correct for the outbound dialing? |
23:20.07 | mrgoby | can you use macros recursively and maintain the original macro exten var ? |
23:20.28 | plc5_250 | I am not convinved that my exten string is correct for iaxtel... |
23:20.34 | mrgoby | or jump from one macro to another and get back to your original extension ? |
23:20.49 | mrgoby | via the vars ? |
23:21.45 | *** join/#asterisk zippp (~zip@c66.190.109.98.ts46v-01.rckprt.tx.charter.com) |
23:22.35 | cjk | moh works great from call coming from an external source (zap, remote voip provider) but when i call someone on my network, both moh's are messed up |
23:22.41 | cjk | any one know this problem |
23:23.50 | cjk | moh works great from call coming from an external source (zap, remote voip provider) but when i call someone on my network, both moh's are messed up |
23:23.50 | cjk | any one know this problem |
23:24.07 | zippp | can anyone help w/ compiling iaxcli (iaxclient simpleclient) for arm using crosstool .28rc39 on linux |
23:24.16 | *** part/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
23:25.15 | *** join/#asterisk ixx (foobar@cpe-70-113-47-137.austin.res.rr.com) |
23:25.24 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
23:25.56 | ManxPower | cjk: Are you using 1.0.6? |
23:26.39 | cjk | ManxPower, no. 1.0.4 |
23:26.57 | cjk | ManxPower, the problem is that i define before every number (extension) the setmusiconhold of the user |
23:27.19 | cjk | well if user1 calls user2 i have the command setmusiconhold executed twice |
23:27.29 | cjk | this is not the case when the call comes from zap |
23:27.37 | cjk | at the moment i do not see the solution |
23:28.01 | PatrickDK | sounds like your using dial(,,m) |
23:28.29 | ManxPower | PatrickDK: I don't think setmusiconhold twice would cause problems |
23:28.36 | PatrickDK | na |
23:28.52 | PatrickDK | hmm, you have vad turned on? |
23:29.00 | cjk | vad == ? |
23:29.08 | PatrickDK | guess that is a yes |
23:29.15 | PatrickDK | silence suppression |
23:29.16 | *** join/#asterisk OloBola (~not@h-66-134-67-154.snvacaid.covad.net) |
23:29.24 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
23:29.33 | cjk | no |
23:29.44 | cjk | its not turned on |
23:29.47 | PatrickDK | what phone is user1 and user2? |
23:30.24 | *** join/#asterisk ncjp (~switch@61.206.115.5.user.ad.il24.net) [NETSPLIT VICTIM] |
23:30.40 | OloBola | how can I save voicemail using AGI? |
23:30.41 | cjk | PatrickDK, well does not really matter. but user1 is grandsteram and user2 is fritz box |
23:31.05 | lesouvage | Is there a written list of the sounds with filename and the lines used so I can easily make a tanslation andrecord my own files. The list should be on http://www.ctitec.com/asterisk/ |
23:31.07 | lesouvage | sounds.htm but isn't. |
23:32.29 | ManxPower | you mean like sounds.txt in the astrisk source code, or sounds-extra.txt in the asterisk-sounds root? |
23:32.35 | cjk | PatrickDK, ManxPower. as i sees it muiconhold is defined on a per channel basis and * can only handle one definition per channel not two. am i wrong here? |
23:32.48 | PatrickDK | no |
23:32.52 | *** part/#asterisk xlyz (~xl@81-208-36-176.fastres.net) |
23:32.54 | PatrickDK | the current one wipes out the old one |
23:32.57 | ManxPower | cjk: a "channel" is "one leg of a call" |
23:33.12 | cjk | ManxPower, ok |
23:33.17 | ManxPower | so it would make no sense to have multiple music on hold for the same leg of the same call. |
23:34.04 | cjk | ManxPower, thats true. |
23:34.18 | cjk | but i predd the hold button on user1's phone and it starts onhold of user2 |
23:34.25 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) [NETSPLIT VICTIM] |
23:34.33 | cjk | i guess this is somehow related to my enum lookups..... |
23:34.52 | ManxPower | *grumble* I don't want to go to New Orleans tomorrow. |
23:35.00 | cjk | <PROTECTED> |
23:35.00 | cjk | <PROTECTED> |
23:35.10 | cjk | I guess this part is causing me the problems |
23:35.27 | cjk | the channels are the mixed up |
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23:38.51 | Sedorox | mmm.. wow... |
23:39.21 | Sedorox | anyone on sipphone? |
23:40.13 | OloBola | how can I save voicemail using AGI? |
23:43.08 | shmaltz | why am I getting this: |
23:43.09 | shmaltz | kernel is too big for standalone boot from floppy |
23:43.11 | shmaltz | when running make bzImage in /usr/src/linux on slackware 10? |
23:43.20 | BoRiS | shmaltz: Thats normal |
23:43.31 | shmaltz | Menaing? BoRiS |
23:43.36 | BoRiS | when you compile alot of stuff into the kernel. |
23:43.36 | shmaltz | I should ignore it? |
23:43.47 | BoRiS | yeah |
23:45.20 | Sedorox | shmaltz: thats just saying if you wanna put it on floppy.. its too big.. you use to have to do that.. but its old... |
23:45.27 | Sedorox | like BoRiS said.. safe to ignore... |
23:45.38 | shmaltz | going for reboot will see |
23:47.47 | Sedorox | can anyone dial sipphone from fwd and have it go through? |
23:48.58 | shmaltz | looks good, thanks guys |
23:50.47 | brc__ | woohoo http://www.ssiworld.com/watch/computers.htm |
23:54.51 | brc__ | http://www.ssiworld.com/watch/washing_machine.htm |
23:57.51 | PTG123 | should it pass $EXTEN between contexts? |
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