| 00:00.59 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 00:17.17 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:24.03 | robin_sz | gah .. useless snom configuration bollocks |
| 00:27.53 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 00:35.30 | Opperior | robin_sz: problem? |
| 00:36.06 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:51.46 | robin_sz | Opperior, sorta .. .used settign files for configuring Snoms much? |
| 00:52.01 | Opperior | I have, yes |
| 00:54.38 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 01:05.34 | *** join/#asterisk ronfox (n=rfoxirc@rrcs-66-91-128-34.west.biz.rr.com) |
| 01:12.29 | robin_sz | Opperior, SO, I copy the settigns out of the setttttigns page of one phoen ... remove all the ip related stuff as ts DHCP, |
| 01:12.53 | robin_sz | remove various other useless crap in there ... and then load it into another phone |
| 01:12.57 | robin_sz | so far so good right? |
| 01:13.26 | robin_sz | then .. the second phoen instead of just booting, loading the settitngs and getting on with its life .. |
| 01:14.11 | robin_sz | comes up, dhcps, loads settings ... "select language" .. ok, DHCP? err, we already did that, but select yes ... phoen reboots |
| 01:14.15 | robin_sz | repeat until bored |
| 01:14.49 | Opperior | it's generally not recommended that you copy the settings page. Better to go from scratch. A good configuration only needs about 20 lines |
| 01:14.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:14.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:14.57 | robin_sz | errr |
| 01:15.00 | robin_sz | whatever |
| 01:15.34 | robin_sz | I only needed it to clear the phone of some internal craziness |
| 01:15.37 | Opperior | can you pastebin your config file? |
| 01:16.11 | robin_sz | two elmeg 290s,. same software .. same config according to the http interface .. one woudl auth, the other wouldnt |
| 01:16.53 | robin_sz | but various differences in the "settings" page ... that didnt seem to show up anywher ein the http interfaces |
| 01:17.42 | robin_sz | eventually, I just copied and pasted it over ... it blew the crap out of the memory of the non-working one and its off and working now ... |
| 01:18.13 | robin_sz | pastebin it/ what? all of it? |
| 01:18.17 | robin_sz | its huge ... |
| 01:18.32 | Opperior | hmm... |
| 01:19.24 | Opperior | can I msg you? |
| 01:19.27 | robin_sz | it basically as it comes off the phone settigns page |
| 01:19.48 | robin_sz | if you like and you feel it cant be said in public ... |
| 01:26.57 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:34.47 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 01:47.32 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 01:52.27 | *** join/#asterisk y3hsDotBiz (n=w00t@124.6.153.226) |
| 01:52.50 | y3hsDotBiz | hi people |
| 01:52.51 | y3hsDotBiz | good day to all |
| 01:52.58 | ManxPower | hello y3hsDotBiz |
| 01:53.55 | y3hsDotBiz | i would like to ask how many concurrent connections a 2 MB line can have if i use g729 codec |
| 01:55.29 | ManxPower | y3hsDotBiz: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 01:56.41 | *** join/#asterisk DocHolliday (i=tabmeist@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:57.38 | y3hsDotBiz | ManxPower , 2000kbps is 2 MEG right? |
| 01:58.19 | ManxPower | 2000kbps would be close enough to the actual number. |
| 01:58.27 | ManxPower | I assume you are on a E-1 line. |
| 01:58.50 | DocHolliday | 2000kbps = 2Mbps ~ |
| 01:59.05 | y3hsDotBiz | ok.. tnx |
| 02:01.27 | DocHolliday | np |
| 02:12.50 | JT | an E1 is 2.048MBit/s |
| 02:13.08 | JT | Mbit/s, i should say |
| 02:13.25 | J4k3 | synchronous |
| 02:13.36 | J4k3 | (4.096MBit/sec total) |
| 02:13.40 | J4k3 | ;) |
| 02:13.44 | JT | umm |
| 02:14.06 | suma | is there is any PHP API class to interface with Asterisk ? |
| 02:14.12 | JT | an E1 is plesiochronous actually |
| 02:14.25 | JT | not synchronous |
| 02:14.35 | JT | maybe you mean symmetrical |
| 02:14.38 | J4k3 | no |
| 02:14.40 | J4k3 | synchronous |
| 02:14.55 | JT | sorry, an E1 is plesiochronous not synchronous |
| 02:14.56 | J4k3 | ie - you can send and recieve data at the same time (based on the same timing) |
| 02:15.14 | JT | J4k3: http://en.wikipedia.org/wiki/PDH |
| 02:17.16 | *** part/#asterisk [1]J (n=new@adsl-065-006-173-139.sip.mia.bellsouth.net) |
| 02:17.43 | J4k3 | this article lacks... |
| 02:17.51 | tzanger | plesiochronous? sounds like a dinosaur |
| 02:17.55 | J4k3 | basically its spending its entire content explaining basic multiplexing. |
| 02:18.00 | JT | synchronous relates to timing |
| 02:18.06 | J4k3 | yes, this is correct |
| 02:18.07 | JT | as in clocking |
| 02:18.16 | JT | not whether you can transmit and receive at the same time |
| 02:18.22 | JT | in telecommunications anyway |
| 02:19.08 | JT | tzanger: i guess it is a bit of a dinosaur ;) |
| 02:19.14 | tzanger | :-) |
| 02:19.52 | *** join/#asterisk [1]J (n=new@adsl-065-006-173-139.sip.mia.bellsouth.net) |
| 02:22.10 | [1]J | New to AsteriskNOW, anyone available to answer a few questions? |
| 02:24.15 | JT | there's actually a specific channel for it i believe |
| 02:25.01 | [1]J | Do you know the name? |
| 02:26.17 | JT | #asterisk-gui |
| 02:26.33 | [1]J | THX JT |
| 02:30.03 | *** join/#asterisk [shodan] (n=shodan@ip117.96-113-216.pppoe1.joliette.intermonde.net) |
| 02:35.26 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 02:39.14 | DocHolliday | what are the advantages of asterisknow versus asterisk? |
| 02:39.20 | DocHolliday | (besides a GUI) |
| 02:39.25 | JT | none |
| 02:40.07 | DocHolliday | JT, i can honestly say thats what i was hoping for :P |
| 02:40.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 02:40.30 | JT | the gui is pretty much the point |
| 02:40.59 | DocHolliday | does the GUI make it nearly impossible to actually review the config files? |
| 02:41.15 | ManxPower | DocHolliday: The FreePBX one seems to |
| 02:41.47 | DocHolliday | yeah, thats not good.. i like doing everything manually |
| 02:43.33 | LoRez | how on earth do you play the voicemail wave files? |
| 02:52.55 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 02:54.04 | flenders | JT: just had a chat with the optus sales rep, and they've given us no installation fees on their ISDN |
| 02:54.25 | JT | not even a credit? |
| 02:54.29 | JT | just flat out nothing? |
| 02:54.36 | flenders | JT: yeah! |
| 02:54.40 | JT | sweet |
| 02:54.48 | flenders | 10 lines for a start |
| 02:54.51 | JT | i wonder if they will provision it over copper or fibre |
| 02:55.22 | flenders | one of their techies is gonna ring me soon, I'll ask him |
| 02:55.48 | JT | cool |
| 02:56.00 | JT | take photos of the finished install, i want to see |
| 02:56.02 | JT | :) |
| 02:56.14 | flenders | are you serious? |
| 02:56.15 | flenders | :D |
| 02:57.43 | JT | yeah, i want to see what optus does |
| 02:57.47 | JT | cable installs and stuff |
| 02:57.50 | JT | silly i know |
| 02:58.03 | flenders | JT: ok, I'll take photos |
| 02:58.16 | flenders | you want to know if theyre as messy as telstra? |
| 02:58.25 | ManxPower | flenders: where are you located? |
| 02:58.47 | JT | flenders: yeah, and what hardware they use |
| 02:58.48 | flenders | ManxPower: those lines will be in brookvale |
| 02:58.57 | JT | flenders: think more global :P |
| 02:59.00 | ManxPower | flenders: uh, what country? |
| 02:59.08 | flenders | ManxPower: .au |
| 02:59.10 | ManxPower | Ah. |
| 02:59.19 | *** part/#asterisk [1]J (n=new@adsl-065-006-173-139.sip.mia.bellsouth.net) |
| 02:59.33 | ManxPower | Why not get a PRI if you are getting 20 channels on 10 BRIs? |
| 02:59.39 | ManxPower | And how are you going to interface them to asterisk? |
| 02:59.41 | JT | he is getting a pri |
| 02:59.45 | JT | 10ch fractional e1 |
| 02:59.46 | ManxPower | oh. nevermind |
| 02:59.54 | flenders | :o) |
| 03:00.04 | ManxPower | Perhaps I am lacking sleep. |
| 03:00.14 | JT | sleep/background story :) |
| 03:00.59 | DocHolliday | the cisco people aren't very happy when you tell them you have made a 28xx series voice box route calls for asterisk *laugh* |
| 03:01.12 | JT | lol |
| 03:01.21 | JT | is that like a low-mid range router? |
| 03:01.59 | DocHolliday | its midrange i would guess, you used for FXO/FXS & PRI.. and data obviously.. |
| 03:02.08 | JT | hmm |
| 03:02.08 | DocHolliday | -you |
| 03:04.50 | JT | does it talk sip? |
| 03:05.07 | ManxPower | JT: It can |
| 03:05.45 | ManxPower | I used a 1750 box with FXOs when I first started using asteirsk. Biggest waste of money I've ever done. (the FXOs in the router, not the router itself) |
| 03:06.43 | JT | hmm |
| 03:06.53 | JT | due to functionality or sheer cost? |
| 03:07.27 | flenders | JT: to run a fax on one of those channels (10ch PRI), do I need a FXS module on a TDM400? |
| 03:07.49 | JT | uhoh |
| 03:08.00 | JT | why not buy an additional analogue line |
| 03:08.04 | JT | it will be so much easier |
| 03:08.06 | JT | for $20 |
| 03:09.24 | ManxPower | I've had nothing but trouble running Fax thru Asterisk |
| 03:09.29 | ManxPower | Others have had no problems |
| 03:10.02 | DocHolliday | ManxPower, even with a T.38 ATA + asterisk pass thru and a T.38 provider? |
| 03:10.42 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:10.54 | ManxPower | DocHolliday: 1.4 is the first version to do T.38 pass thru and 1.4 is not stable enough for my requirements. |
| 03:11.13 | ManxPower | Also, it seems like a lot more work than just getting an analog line for fax |
| 03:12.21 | [TK]D-Fender | DocHolliday : Still shopping for pain I see.... |
| 03:12.27 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 03:12.42 | DocHolliday | [TK]D-Fender, as much as i can get |
| 03:12.51 | [TK]D-Fender | DocHolliday : Welcome to the buffet |
| 03:13.15 | DocHolliday | can i have fries with that? |
| 03:13.26 | [TK]D-Fender | DocHolliday : All you can eat. |
| 03:13.40 | DocHolliday | haha |
| 03:14.06 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 03:14.15 | flenders | JT: a card would cost me 250 max, and only on l ine rental, I would spent 240 a year |
| 03:14.47 | JT | flenders: again, you need to weigh up the pain in the arse factor |
| 03:14.55 | ManxPower | flenders: if you ever have fax problems you don't have to worry about asterisk if you get an analog line |
| 03:15.04 | JT | analogue line should just work |
| 03:15.13 | JT | i assume your time to fix problems isn't free |
| 03:15.32 | flenders | JT: that's true |
| 03:15.35 | ManxPower | and if it does not, everyone does understand that it should work and can't blame it on "that free communist PBX" |
| 03:16.00 | ManxPower | And if your company is anything like my clients, one thing that MUST work is fax. |
| 03:16.39 | ManxPower | When fax is broken it is hard to fix because the villagers with flaming torches are chasing you chanting "Burn the geek! Burn the geek!" |
| 03:16.55 | flenders | hahahaha |
| 03:18.05 | [TK]D-Fender | My company almost roasted my ass when our failure rate passed 50% |
| 03:18.12 | [TK]D-Fender | this is NOT a joke... |
| 03:18.16 | ManxPower | [TK]D-Fender: same here. |
| 03:18.29 | [TK]D-Fender | Pay for a friggen 1FL and be done with it. Your sanity is worth more... |
| 03:18.50 | ManxPower | Personally I believe that the reason the faxes were failing is because I had to screw with the audio gains so much to get rid of the echo |
| 03:19.02 | ManxPower | [TK]D-Fender: 1FB I think |
| 03:19.25 | ManxPower | You should not have to keep fixing the PBX. It should Just Work |
| 03:20.12 | flenders | ManxPower: was that echo on ISDN? |
| 03:20.17 | ManxPower | It should process and route calls and the only time you should have to mess with it is for adds/moves/changes |
| 03:20.28 | ManxPower | flenders: yes. caused by the far end analog lines |
| 03:20.41 | ManxPower | ISDN just means you can't have NEAR end echo. |
| 03:21.00 | ManxPower | But since virtually all echo is FAR end echo, ISDN doesn't fix it. |
| 03:21.01 | [TK]D-Fender | ManxPower : Even after I switched to my Sangoma for which I DIDN'T have to play with gains, it was still just kinda "off". |
| 03:21.05 | DocHolliday | [TK]D-Fender, the problem is as i explained im in executive suites.. they want close to $500 to provision POTS |
| 03:21.06 | flenders | ManxPower: any tips on tweaking echo? |
| 03:21.16 | ManxPower | flenders: buy a good echo canceler |
| 03:21.25 | [TK]D-Fender | DocHolliday : Sucks to be you then. |
| 03:21.35 | ManxPower | we use tellabs EC cards from ebay. dirt cheap and it is what the telcos use for echo canceling |
| 03:22.12 | ManxPower | they are a miserable hell to set up, but once you know what you need to know they are simple and Just Work |
| 03:22.30 | ManxPower | My biggest issue with them is finding -48V power supplies |
| 03:22.49 | ManxPower | well, -48V power supplies that do not cost a fortune |
| 03:23.32 | JT | ManxPower: can you get them in E1 versions for cheap though? :) |
| 03:23.43 | JT | heh i have a few -48VDC supplies sitting around at home |
| 03:23.51 | JT | but yeah they're not cheap new |
| 03:23.54 | ManxPower | I have alot more confidence in the Tellabs then I do in the Digium open source EC, the Digium commercial EC, or the Digium hardware EC. |
| 03:24.41 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 03:25.04 | JT | ManxPower: so what's the E1 availability like? flenders is getting E1 not T1 |
| 03:25.14 | ManxPower | At least in Louisiana and Mississippi Bellsouth seem to run toll calls (Inter-LATA) via an EC, so we only had problems with local calls to POTS lines |
| 03:25.26 | ManxPower | JT: I doubt he'll find E-1 tellabs. |
| 03:25.35 | ManxPower | but there are other hardware ECs out there. |
| 03:28.25 | *** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 03:28.32 | IguanaNed | hello all |
| 03:28.57 | ManxPower | He could always use a 4-port T-1/E-1 card for asterisk and Telco E-1 -> Port 1 Asterisk port 2 T-1 -> Tellabs EC -> T-1 Asterisk Port 3 |
| 03:29.31 | IguanaNed | quick question regarding meetme.. I did not install Zaptel prior to installing asterisk. do I need to recompile asterisk after Zaptel? |
| 03:29.39 | ManxPower | IguanaNed: yes |
| 03:29.49 | JT | ManxPower: that'd be a little crazy, but sure |
| 03:30.01 | ManxPower | JT: It would be horribly crazy |
| 03:30.04 | JT | it'd restrict the number of channels you use too |
| 03:30.17 | IguanaNed | will I lose all my conf file if I reinstall asterisk? |
| 03:30.26 | ManxPower | IguanaNed: only if you do a "make samples" |
| 03:30.31 | ManxPower | back them up just in case |
| 03:30.37 | IguanaNed | thx |
| 03:30.48 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 03:31.24 | IguanaNed | do I need Libpri too for meetme? |
| 03:31.58 | flenders | with the echo, we have a couple of TDM400Ps with 8 FXO modules... we can notice echo in the very beginning of the call, but it's gone after a couple of seconds... wouldn't it be the same case with E1? |
| 03:32.19 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 03:32.47 | ManxPower | flenders: it dends on many things |
| 03:33.12 | ManxPower | JT: I experienced echo on an ALL ZAPTEL config the other day. very weird |
| 03:34.06 | JT | all zaptel? |
| 03:35.43 | flenders | ManxPower: so even with the echo training on, I could experience echo during the whole call? |
| 03:36.47 | ManxPower | JT: Yup. |
| 03:37.06 | ManxPower | Analog -> Adtran -> Asterisk -> same adtran -> analog phone |
| 03:37.14 | ManxPower | the first analog is POTs |
| 03:37.45 | ManxPower | I THINK the issue was gains. The echo sounded "weird" to me. |
| 03:44.27 | JT | ah ok |
| 03:45.06 | ManxPower | I have a line test device I'll put on the system at some point. |
| 03:45.15 | IguanaNed | <PROTECTED> |
| 03:45.20 | IguanaNed | argghhh |
| 03:47.00 | ManxPower | IguanaNed: try a make clean before make install |
| 03:48.09 | IguanaNed | thx Manx will try |
| 03:51.59 | *** join/#asterisk bmg505 (n=leon@c1-250-9.rndf.isadsl.co.za) |
| 03:53.41 | IguanaNed | crap no luck.. still cant find meetme app |
| 03:53.56 | IguanaNed | should I be able to locate a meetme.so file? |
| 03:54.34 | IguanaNed | found /usr/src/asterisk/asterisk/apps/app_meetme.c |
| 03:54.38 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:54.45 | [TK]D-Fender | IguanaNed : I'm assuming you don't have a zaptel card. that in mind did you follow the instructions on how to enable ZTDUMMY? |
| 03:55.02 | IguanaNed | I must have skipped that part |
| 03:55.04 | [TK]D-Fender | IguanaNed : if not, there's your next probelm |
| 03:55.14 | IguanaNed | htx will look into it |
| 03:55.30 | IguanaNed | i have no zpatel card .. |
| 03:55.37 | IguanaNed | this is purely IP pased box |
| 03:55.41 | IguanaNed | er based |
| 03:55.57 | JT | well yeah |
| 03:56.01 | JT | meetme need zap timing |
| 03:57.33 | [TK]D-Fender | IguanaNed : Go recompile ith ZTDUMMY support |
| 03:58.59 | flenders | [TK]D-Fender: someone mentioned the other day that the linksys SPA941 was in forth place on your list... what list is this? |
| 03:59.42 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:07.47 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:10.05 | IguanaNed | is this the line that is suppose to be commented : |
| 04:10.06 | IguanaNed | ztdummy.o: ztdummy.h |
| 04:10.09 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:10.36 | IguanaNed | cuz it was already un-commented |
| 04:12.35 | IguanaNed | My Makefile has : |
| 04:12.36 | IguanaNed | # build ztdummy by default for 2.6 kernels |
| 04:12.37 | IguanaNed | ifeq ($(BUILDVER),linux26) |
| 04:12.37 | IguanaNed | MODULES+=ztdummy |
| 04:12.37 | IguanaNed | endif |
| 04:13.26 | [TK]D-Fender | flenders : on my list of suggested phones for * |
| 04:13.53 | [TK]D-Fender | IguanaNed : My need to modprobe it, and zaptel as well |
| 04:14.26 | IguanaNed | before installing asterisk? |
| 04:16.36 | IguanaNed | modprobe returned no error messages |
| 04:17.44 | [TK]D-Fender | IguanaNed : try "ztcfg -vvvv" and the start * |
| 04:17.47 | IguanaNed | Feb 18 23:06:39 localhost kernel: Zapata Telephony Interface Registered on major 196 |
| 04:17.47 | IguanaNed | Feb 18 23:06:39 localhost kernel: Zaptel Version: 1.2.11 Echo Canceller: KB1 |
| 04:17.47 | IguanaNed | Feb 18 23:06:40 localhost kernel: Registered tone zone 0 (United States / North America) |
| 04:17.52 | IguanaNed | oops |
| 04:17.55 | IguanaNed | will try |
| 04:18.28 | IguanaNed | Fender |
| 04:18.37 | IguanaNed | got Zaptel COnfiguration |
| 04:18.39 | IguanaNed | Channel Map |
| 04:18.45 | IguanaNed | 0 Channels configured |
| 04:18.45 | flenders | [TK]D-Fender: where can I find that list? |
| 04:18.47 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:18.53 | [TK]D-Fender | flenders : In my head :) |
| 04:19.14 | flenders | what's number 1? |
| 04:19.27 | flenders | actually, 1,2 and 3 |
| 04:19.52 | flenders | I'm pretty happy with the SPA-9xx, but want to see what others think |
| 04:20.16 | [TK]D-Fender | flenders : Polycom (any), Aastra 480i, Cisco 7940+, Linksys, Snom |
| 04:20.48 | IguanaNed | Fender: What wa I suppose to see after issuing the ztcfg command? |
| 04:21.12 | bkruse_home | [TK]D-Fender++ |
| 04:21.21 | [TK]D-Fender | IguanaNed : Not a question of what you see, its jsut to help enable everything. try starting *. Then try loading app_meetme.so |
| 04:21.28 | [TK]D-Fender | bkruse_home : y0 |
| 04:21.31 | bkruse_home | IguanaNed: if you dont see anything, your good to go |
| 04:21.36 | bkruse_home | [TK]D-Fender: wuts happ-o-ning |
| 04:21.45 | bkruse_home | IguanaNed: lsmod | grep zap |
| 04:21.46 | [TK]D-Fender | bkruse_home : Just killing time like usual. |
| 04:21.52 | bkruse_home | nice nice nice |
| 04:22.06 | [TK]D-Fender | bkruse_home : We're just walking him through getting ZTDUMMY up and running for MeetMe. |
| 04:22.15 | bkruse_home | gotcha |
| 04:22.19 | bkruse_home | always fun :P |
| 04:22.22 | bkruse_home | <3 ztdummy |
| 04:22.29 | IguanaNed | zaptel 191584 1 ztdummy |
| 04:22.29 | IguanaNed | crc_ccitt 2369 1 zaptel |
| 04:22.29 | IguanaNed | zaptel 191584 1 ztdummy |
| 04:22.30 | IguanaNed | crc_ccitt 2369 1 zaptel |
| 04:22.30 | IguanaNed | thanks guys |
| 04:23.04 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:23.47 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 04:23.53 | [TK]D-Fender | IguanaNed : All good? |
| 04:24.26 | IguanaNed | hmm cant load app_meetme |
| 04:24.45 | IguanaNed | should I be able to locate app_meetme.so? |
| 04:25.08 | [TK]D-Fender | IguanaNed : typically in /usr/lib/asterisk/modules IIRC |
| 04:25.25 | wunderkin | was asterisk recompiled after zaptel |
| 04:25.55 | IguanaNed | ahh |
| 04:26.00 | IguanaNed | I did install libpri |
| 04:26.14 | IguanaNed | that would do it eh? |
| 04:26.33 | IguanaNed | er I mean I did NOT install libpri |
| 04:26.36 | wunderkin | maybe this question would be better for tomorrow afternoon but i saw on some off-site that echotraining is not used for pri, is that right? the echo is probably from the phones but i'm not sure yet, i have a pri and using sip phones |
| 04:26.43 | wunderkin | no |
| 04:27.14 | IguanaNed | I will recompile both |
| 04:27.25 | IguanaNed | but to be clear .. do I need Libpri or not? |
| 04:27.49 | wunderkin | only if you have a pri but if you are using ztdummy then no |
| 04:28.10 | IguanaNed | sorry ,,, I dont know what priu means |
| 04:28.14 | wunderkin | no |
| 04:28.14 | IguanaNed | so I guess not |
| 04:28.29 | IguanaNed | ok I will clean and make both .. back in a bit with results |
| 04:28.38 | wunderkin | just redo asterisk |
| 04:34.37 | JT | wunderkin: you can get echo on pri lines, however it is far end echo from analogue lines |
| 04:35.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:37.11 | wunderkin | yeah, but we do apparantly have some bad phones, they also have reported at least once having echo between 2 sip phones, i don't know if it was due to a speakerphone or not though, hopefully this weekend we will have the new ones in, they used to report echo on almost all of their calls i believe, but now i think it is more intermittant, i've been trying the different echo cancellers and settings |
| 04:38.04 | JT | hmm |
| 04:38.23 | JT | going hardware EC in the firstplace usually heads off a lot of these problems |
| 04:39.10 | wunderkin | there was also a problem with the sidetone setting on polycom sip 2.0.3 i think, but this has gone on for awhile... |
| 04:40.19 | wunderkin | maybe its getting better now i dont know, they have just stopped reporting most of the things i think |
| 04:41.06 | wunderkin | i've never had any echo but it doesn't help that most of my calls are to cell phones |
| 04:42.25 | bkruse_home | anyone messed with the svn api?! |
| 04:44.26 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:51.15 | IguanaNed | No application 'meetme' for extension (conferences, 5101, 1) |
| 04:53.15 | [TK]D-Fender | IguanaNed : Go verify that the module is in your modules folder |
| 04:54.25 | IguanaNed | Fender what shold the module be named? |
| 04:55.02 | [TK]D-Fender | app_meetme.so |
| 04:57.10 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 04:58.17 | IguanaNed | Fender: Is in my "/usr/lib/asterisk/modules/app_meetme.so |
| 04:58.17 | IguanaNed | " |
| 05:00.56 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:01.39 | [TK]D-Fender | IguanaNed : and when you try to load it manually? |
| 05:02.01 | [TK]D-Fender | IguanaNed : "load app_meetme.so" <- |
| 05:03.33 | IguanaNed | hmm it registered! |
| 05:04.19 | bkruse_home | vi /etc/asterisk/modules.conf possibly |
| 05:04.21 | bkruse_home | autoload=yes |
| 05:05.09 | IguanaNed | Woohoo! I have conference! |
| 05:09.19 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:09.39 | [TK]D-Fender | there you go... |
| 05:10.29 | *** join/#asterisk Roey (n=abc@pdpc/supporter/sustaining/Roey) |
| 05:10.30 | Roey | hi all |
| 05:10.55 | Roey | anyone have Verizon? How can I download all my messages from my phone to my email, if I can't connect the phone with bluetooth to my computer? |
| 05:11.43 | IguanaNed | bkruse: autoload is set to yes |
| 05:12.12 | bkruse_home | and you dont have a noload anywhere |
| 05:12.13 | bkruse_home | there u go |
| 05:12.13 | IguanaNed | but I see no reference to app_meetme.so... |
| 05:12.23 | bkruse_home | <3 subversion |
| 05:12.25 | bkruse_home | k |
| 05:12.26 | IguanaNed | there are a couple noloads |
| 05:12.55 | [TK]D-Fender | IguanaNed : maybe it'll all be fine from the get-go on next restart... |
| 05:12.58 | IguanaNed | should I add something like load => app_meetme.so? |
| 05:13.21 | IguanaNed | if I do a 'reload will that tell me ? |
| 05:14.08 | bkruse_home | core show application meetme |
| 05:14.15 | bkruse_home | module load (tab)(tab) |
| 05:15.14 | IguanaNed | show application meetme worked |
| 05:15.31 | IguanaNed | not sure what you mean by module load (tab) (tab) |
| 05:15.55 | bkruse_home | well |
| 05:16.05 | bkruse_home | if its not loaded, you can load it with module load app_meetme.so |
| 05:16.09 | bkruse_home | tab for tab completion :] |
| 05:16.24 | IguanaNed | oh |
| 05:16.26 | IguanaNed | right |
| 05:16.48 | IguanaNed | I wa just wondering if it will restart... say if my server is rebooted for whatever reason |
| 05:17.04 | [TK]D-Fender | IguanaNed : Try it right NOW and see |
| 05:19.55 | IguanaNed | ahh rather not reboot yet.. but will try on next convenient time |
| 05:27.47 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:28.15 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:28.16 | zeeesh | hi |
| 05:28.38 | IguanaNed | anyone know a good source for meetme how-to? configuring confernces? |
| 05:28.55 | IguanaNed | I want to set only one person to be the "speaker" |
| 05:29.12 | bkruse_home | just let him dial in with one option, and the others dial in as muted |
| 05:30.00 | [TK]D-Fender | IguanaNed : "show application meetme" |
| 05:30.00 | IguanaNed | fender am in there now |
| 05:31.05 | IguanaNed | If i am correct those options are only set in the meetme.conf? |
| 05:31.23 | [TK]D-Fender | IguanaNed : sever work on the dialplan line that calls it. |
| 05:31.29 | [TK]D-Fender | several* |
| 05:31.50 | IguanaNed | oh right I see |
| 05:34.13 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 05:35.01 | IguanaNed | ok |
| 05:35.38 | IguanaNed | one more ? .. I want the ability to dynamically add users to my iax.conf file without dropping a conference |
| 05:36.23 | IguanaNed | hey bkruse... BY any chance is your last name kruse? |
| 05:36.35 | Qwell | IguanaNed: what gave you that idea? :p |
| 05:36.59 | IguanaNed | just curious as I know someone else with the lastname kruse |
| 05:37.14 | IguanaNed | wasn;t sure how common it was |
| 05:40.33 | bkruse_home | first name? |
| 05:40.36 | bkruse_home | its not TOO common |
| 05:43.02 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 05:43.25 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 05:44.51 | IguanaNed | bkruse: first name of the person I know is David |
| 05:45.34 | IguanaNed | any relatives named David? |
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| 05:48.48 | bkruse_home | not that i can think of |
| 05:48.50 | bkruse_home | where in the US? |
| 05:49.55 | IguanaNed | nope Canada |
| 05:50.03 | IguanaNed | should have asked that first I guess |
| 05:52.25 | bkruse_home | nvmm |
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| 06:02.34 | flenders | JT: what handsets do you use? |
| 06:02.44 | JT | isdn pabx :P |
| 06:04.32 | tydelCA | asterisk can't be used on a nat gateway? |
| 06:04.37 | IguanaNed | anyone know a good source , or etailed guide for "meetme"? |
| 06:04.44 | tydelCA | like, it can't bind to the inside and outside interface? |
| 06:04.58 | [TK]D-Fender | tydelCA : By default it can bind to ALL interfaces |
| 06:05.05 | [TK]D-Fender | tydelCA : And the answer is YES |
| 06:05.10 | tydelCA | so all or one |
| 06:05.15 | [TK]D-Fender | IguanaNed : www.voi-info.org |
| 06:05.16 | tydelCA | not multiple specific |
| 06:05.24 | [TK]D-Fender | tydelCA : so far, yeah. |
| 06:05.26 | tydelCA | ok |
| 06:05.27 | tydelCA | thanks |
| 06:05.37 | [TK]D-Fender | tydelCA : you can always use iptables to filter off the others though. |
| 06:06.09 | flenders | JT: dunno why I thought you had IP phones |
| 06:07.33 | flenders | JT: you know ingrammicro? |
| 06:08.06 | flenders | JT: they have cisco 7940g phones for 312 AUD inc GST |
| 06:08.27 | tydelCA | they have ingram micro in .au? |
| 06:08.29 | tydelCA | wow |
| 06:08.45 | tydelCA | I didn't know they were so widespread |
| 06:09.54 | JT | ingram micro are ripoffs |
| 06:10.10 | J4k3^ | ingram micro is a ripoff unless you move a few million/month through them |
| 06:10.18 | J4k3^ | then they'll hump your leg (see newegg) |
| 06:11.36 | Qwell | what's wrong with newegg? |
| 06:12.01 | J4k3^ | nothing |
| 06:12.19 | J4k3^ | other than its hard to find a human if something goes wrong, but its rare (never happened to me) for something to go wrong |
| 06:12.27 | Qwell | no it's not... |
| 06:12.39 | Qwell | it takes 2 seconds to get somebody |
| 06:12.44 | J4k3^ | but that was also a few years ago |
| 06:12.50 | J4k3^ | they also don't do order pickups |
| 06:12.59 | *** join/#asterisk |ryan| (n=foo@c-67-188-22-216.hsd1.ca.comcast.net) |
| 06:13.00 | J4k3^ | which is kinda silly IMHO |
| 06:13.13 | tydelCA | they're an online company |
| 06:13.14 | Qwell | why would they? |
| 06:13.17 | Qwell | it only raises costs |
| 06:13.19 | J4k3^ | why wouldn't they? |
| 06:13.24 | tydelCA | do they have a storefront? |
| 06:13.25 | tydelCA | heh |
| 06:13.28 | J4k3^ | tydelCA: no. |
| 06:13.39 | Qwell | if you're close enough to pick something up, you can have it the next day (with standard shipping) |
| 06:13.52 | bkruse_home | they have a wearhouse close to us, kinda |
| 06:13.54 | J4k3^ | Qwell: well, I wish newegg (or somebody with similar clue) would come through and knock the bestbuy/circuitcity/frys/etc out of the way |
| 06:14.09 | Qwell | bkruse_home: Nashville.. |
| 06:14.11 | J4k3^ | anybody with half a bit of sense (and deep pockets and good connections) could. |
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| 06:14.56 | J4k3^ | yeah, I've learned that if I need something *today* to order from directron... its a 210 mile trip one way though |
| 06:15.10 | J4k3^ | these days I don't have those kinds of situations. keep spares onhand. |
| 06:15.38 | *** join/#asterisk dj-fu (n=deejay@203-211-106-190.ue.woosh.co.nz) |
| 06:16.01 | J4k3^ | frys is 150 miles, but damn they suck |
| 06:16.14 | Qwell | frys rocks too :p |
| 06:16.19 | Qwell | I *wish* we had frys here |
| 06:16.48 | J4k3^ | generally I stop by frys, see that they don't have what I want, or they want 3x more than they should for it... then I hop on my cellular connection and make my directron order |
| 06:16.51 | J4k3^ | then drive over and get it. |
| 06:17.04 | J4k3^ | since dropping by frys is maybe 2 miles out of my way for that entire trip |
| 06:17.25 | |ryan| | Is there anything special I have to enable in asterisk to get it to recognize DTMF? I'm dialing a agi app and it does not seem to be recognizing DTMF digits. |
| 06:17.36 | J4k3^ | frys in north houston is fairly ghetto. Its in a crappy neighborhood, they treat everyone line a 3rd rate illegal alien criminal. |
| 06:18.08 | J4k3^ | never went to the one on the southwest side of houston... the neighborhood isn't much better, and if I'm over there I might as well save myself the bother and just drive directly to directron. |
| 06:18.09 | |ryan| | I'm using a sip phone on my lan if that matters. |
| 06:18.09 | J4k3^ | hehe |
| 06:18.52 | bkruse_home | |ryan|: change your dtmf mode in sipconf |
| 06:19.54 | |ryan| | I have it set to inband with the G711u codec |
| 06:20.31 | Qwell | yuck, don't use inband dtmf if you have a choice |
| 06:20.46 | |ryan| | what should I use? |
| 06:20.52 | Qwell | rfc2833 |
| 06:21.14 | |ryan| | My options are InBand, AVT, INFO, and Auto. |
| 06:21.28 | Qwell | bkruse_home: I don't know if it's just here in the south, but wendys makes their food look awful in commercials |
| 06:21.36 | Qwell | </random thought> |
| 06:21.51 | J4k3^ | theres something generally unappetizing about meat thats square. |
| 06:21.54 | tydelCA | rfc2833 = avt |
| 06:22.03 | J4k3^ | it looks... manufactured |
| 06:22.10 | |ryan| | k |
| 06:22.20 | Qwell | J4k3^: well...it is |
| 06:22.53 | J4k3^ | I prefer my burgers from whataburger (texas only) or jack in the box... |
| 06:22.59 | JT | tydelCA: avt? |
| 06:23.06 | tydelCA | dtmf |
| 06:23.08 | J4k3^ | whataburger is almost a real burger, jack in the box is the exact opposite of a real burger, but I love it anyways. |
| 06:23.18 | Qwell | <3 jack in the box |
| 06:23.20 | bkruse_home | Qwell: dang, i totally mentioned we should start a franchise |
| 06:23.22 | J4k3^ | plus |
| 06:23.26 | Qwell | closest one is in nashville :( |
| 06:23.27 | J4k3^ | everybody loves jitb tacos! |
| 06:23.33 | bkruse_home | Qwell: there is one in decatur!!!! |
| 06:23.34 | Qwell | jitb tacos FTW! |
| 06:23.36 | J4k3^ | the closest one here is 28 miles one way |
| 06:23.37 | Qwell | bkruse_home: what?! |
| 06:23.40 | Qwell | are you kidding? |
| 06:23.45 | bkruse_home | no! |
| 06:23.49 | Qwell | a jack in the box? |
| 06:23.52 | Qwell | seriously? |
| 06:23.57 | bkruse_home | unless it closed within the last 2 years, its still there |
| 06:23.57 | bkruse_home | yes@! |
| 06:24.00 | Qwell | I would totally drive to decatur |
| 06:24.10 | bkruse_home | let me look it up |
| 06:24.16 | J4k3^ | http://yp.yahoo.com |
| 06:24.17 | J4k3^ | it'll search |
| 06:24.20 | J4k3^ | for like a 100 mile radius |
| 06:24.22 | bkruse_home | Qwell: we should take a mini roadtrip |
| 06:24.31 | bkruse_home | steak and shake is up there also |
| 06:25.07 | |ryan| | AVT works, thank you. |
| 06:28.16 | [TK]D-Fender | Ok, late & tired.. the deadly duo wins again. Later all..... |
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| 06:34.49 | *** join/#asterisk HeinrichSA (n=hh@196.25.68.131) |
| 06:34.56 | HeinrichSA | Hi all |
| 06:35.13 | HeinrichSA | I am having a problem with h323 and g729... |
| 06:35.34 | HeinrichSA | when i use sip g729 works perfectly |
| 06:36.31 | J4k3^ | free m2m = teh win |
| 06:36.37 | J4k3^ | downsides = additional call setup time |
| 06:36.45 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:38.57 | HeinrichSA | free m2m? |
| 06:39.03 | J4k3^ | mobile-to-mobile |
| 06:39.37 | HeinrichSA | ok ? is that an asterisk module? |
| 06:39.50 | HeinrichSA | will it enable h323 with g729? |
| 06:40.03 | J4k3^ | you'll need licenses for g729 |
| 06:40.18 | J4k3^ | or else the codec police will come arrest you |
| 06:40.32 | HeinrichSA | i already bought one |
| 06:40.36 | HeinrichSA | $10 |
| 06:40.40 | J4k3^ | yeah |
| 06:40.41 | HeinrichSA | so im set |
| 06:40.52 | J4k3^ | you just need to do what the email instructions say |
| 06:40.55 | J4k3^ | and g729 will work |
| 06:41.14 | coppice | lucky fellow |
| 06:41.18 | JT | clearly what J4k3^ was talking about had nothing to do with your problem HeinrichSA... |
| 06:41.19 | bkruse_home | :P |
| 06:44.01 | *** join/#asterisk kb1_kanobe (n=jsmith@bdr2.fieldrd.scrd.ca) |
| 06:44.09 | kb1_kanobe | g'day all |
| 06:44.58 | kb1_kanobe | trying to work around an issue with libpri - how can I match a 'null' extension in the dialplan? Ie., when I get: -- Extension '' in context 'in-cs1000' from '8856800' does not exist. Rejecting call on channel 0/5, span 1 |
| 06:45.49 | JT | s ? |
| 06:46.44 | kb1_kanobe | Duh! I suppose that's it, isn't it. Let me try... |
| 06:48.54 | HeinrichSA | Has anyone had a problem with h323 before? |
| 06:49.54 | kb1_kanobe | JT: no, did't do it. Still get a -- Extension '' in context 'in-cs1000' from '8856800' does not exist. |
| 06:51.11 | HeinrichSA | when i RTP debug. I dont see anything being sent to the gateway... |
| 06:52.03 | HeinrichSA | call attempt X-lite (sip) -> asterisk -> h323 gateway (g729) |
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| 06:57.33 | bkruse_home | omg |
| 06:57.52 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
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| 07:36.51 | J4k3^ | hrm... does anyone have any experience with chan_cellphone with cdma handsets? |
| 07:42.54 | IguanaNed | anyone here familiar with iaxclient? |
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| 08:29.28 | Mavvie | intersting #warning: "xpp_timer must be sampled EXACTLY 1000/per second" |
| 08:29.37 | Mavvie | but who is setting the HZ define? |
| 08:31.22 | *** join/#asterisk Aurs (n=Aurs@81.191.112.104) |
| 08:33.23 | Mavvie | /usr/include/asm/param.h:#define HZ sysconf(_SC_CLK_TCK) |
| 08:37.23 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 08:37.43 | phpboy | hey all, the junghanns phones work fine with bristuff, right? |
| 08:37.49 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
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| 08:40.23 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 08:40.26 | kippi | hey |
| 08:40.42 | phpboy | hi |
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| 08:41.18 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 08:41.22 | kippi | what is the best way to turn up the volume on calls, the handsets are up full, i am sure there is a opstion on asterisk |
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| 08:44.01 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
| 08:48.19 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:48.27 | angryuser | good day |
| 08:48.46 | phpboy | hi |
| 08:50.08 | angryuser | i have a little proble with my asterisk sip peers, sometimes they become Unreachable, and stays unreachable, but when i try to register with softphone with my provider, alls is fine and account is working, asterisk 1.4 svn latest, server ports routed, no firewall on the machine |
| 08:50.47 | angryuser | and when i reboot * peer become reachable |
| 08:52.08 | angryuser | all i need to know, where if the parameter of "reregister timeout" exist at all? |
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| 09:06.58 | corruptor | angryuser: it exists |
| 09:07.16 | kanaeda | :] |
| 09:09.08 | angryuser | corruptor::] * asterisk somehow stuck with the registration.... |
| 09:10.34 | corruptor | such problem can be dns related... |
| 09:10.39 | *** join/#asterisk apardo (n=apardo@87.217.145.116) |
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| 09:11.26 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 09:12.08 | corruptor | there are options registerattempts and registertimeout in sip.conf |
| 09:14.16 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 09:14.30 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 09:17.17 | angryuser | corruptor: i tuned them a bit, will see;) |
| 09:19.44 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
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| 10:18.53 | zeeesh | like vodafone or orange cellular user can block his caller id ... is it possible to block caller id by using asterisk ????? |
| 10:19.38 | angryuser | zeeesh: yes |
| 10:20.18 | zeeesh | will u pls guide how ... ? |
| 10:20.56 | zeeesh | which .. conf file take a part of this feature |
| 10:22.20 | angryuser | zeeesh: you just need to check the variable and use gotoif |
| 10:23.32 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:24.59 | angryuser | gotoif($[${CALLERID(num)} = youridhere]?1:2) |
| 10:25.07 | JT | ~thewiki |
| 10:25.20 | jbot | [thewiki] at http://www.voip-info.org/wiki-Asterisk |
| 10:25.20 | zeeesh | ok |
| 10:25.49 | angryuser | http://www.voip-info.org/wiki-Asterisk+variables al vars used in asterisk |
| 10:28.48 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 10:31.08 | *** join/#asterisk Bazy (n=bazy@eclipse.upcnet.ro) |
| 10:36.56 | Bazy | hi, i need to capture packets for someone to analyse, voice is now working well, jitter, loss. I need to do that with tethereal, can anyone help me with this? |
| 10:40.38 | Mavvie | Bazy: try argus (http://www.qosient.com/argus/) |
| 10:41.52 | *** join/#asterisk dseeb_ (n=dcb@124.177.40.105) |
| 10:43.20 | *** join/#asterisk _omer (i=_omer@9-237-154-202.wol.net.pk) |
| 10:43.26 | _omer | what is best for speech recognition ? |
| 10:43.40 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 10:43.48 | FreezeS | hey guys |
| 10:43.53 | Bazy | Mavvie i need the capture to be in pcap format... 10x for your help, i'll read argus's page later |
| 10:44.06 | FreezeS | anyone knows if it's possible to have zaphfc without bristuff ? |
| 10:44.32 | FreezeS | the problem is they don't have bristuff witn 1.4 at the moment |
| 10:44.38 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
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| 10:50.59 | mafkees | FreezeS: use misdn |
| 10:51.42 | FreezeS | thanks :) |
| 10:55.11 | *** join/#asterisk martineyles_ (n=martiney@adsl-w-234.as15758.net) |
| 10:55.18 | martineyles_ | Hi |
| 10:55.59 | martineyles_ | I'm trying to do hangup on dial-tone in an incoming call |
| 10:56.43 | martineyles_ | (My phone line give me this instead of the usual call progress or polarity change or on/off busy signal) |
| 10:57.09 | martineyles_ | any ideas? |
| 10:58.59 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 10:59.33 | *** join/#asterisk shinux__ (n=shinux@196.220.28.144) |
| 10:59.43 | phearless | hi guys!! |
| 11:00.31 | phearless | when I call from a SIP phone to another local SIP phone, I got this : http://paste.lisp.org/display/37065 |
| 11:00.45 | phearless | 414 is calling 404 |
| 11:00.58 | phearless | how can I debug this? because 404 do not receive the call |
| 11:02.01 | angryuser | phearless: you ment Sip dialing zap? |
| 11:02.16 | phearless | no |
| 11:02.25 | phearless | ahhh i see |
| 11:02.59 | phearless | it may be an extensions.conf error |
| 11:03.37 | phearless | if I do : |
| 11:03.39 | phearless | dialplan show 404@default |
| 11:03.42 | phearless | I got : |
| 11:03.51 | phearless | <PROTECTED> |
| 11:03.55 | phearless | it's good , right? |
| 11:04.18 | angryuser | phearless:and what about registration status? |
| 11:04.28 | phearless | let's have a look... |
| 11:04.40 | angryuser | "sip show peers" |
| 11:06.03 | phearless | ok |
| 11:06.30 | phearless | in fact it was a mistake in sip.conf |
| 11:06.38 | phearless | I did put a wrong context |
| 11:07.41 | phearless | thanks you angryuser |
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| 11:10.02 | *** part/#asterisk s1gny|wrk (n=s1gny@p549173B3.dip.t-dialin.net) |
| 11:11.09 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 11:11.25 | angryuser | phearless: you found it all by yourself |
| 11:11.41 | phearless | yes but I appreciate when people try to help :) |
| 11:12.40 | angryuser | phearless: i am trying to help coz a lot of people helped me |
| 11:13.19 | phearless | excellent :) |
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| 11:20.25 | kippi | can someone help |
| 11:20.31 | kippi | I keep on getting this error |
| 11:20.32 | kippi | SIP/1153-006f7250 is ringing |
| 11:20.32 | kippi | <PROTECTED> |
| 11:20.50 | kippi | keeps on hanging up the calls, have to reboot every 30 mins |
| 11:21.25 | kippi | out going calls are fine |
| 11:23.42 | kippi | and its not reloading probley |
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| 11:42.37 | *** mode/#asterisk [+o denon] by ChanServ |
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| 12:13.05 | jserve | Hi |
| 12:13.10 | creativx | 2 |
| 12:13.10 | creativx | u |
| 12:21.28 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 12:25.46 | Ahrimanes | is it possible to configure outgoing sip registrations (register => ...) in realtime? |
| 12:28.23 | creativx | is there any other way of getting a SIP user's CallerID field other than AMI-> Command: SIPshowpeer, Peer: <user> |
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| 12:39.31 | *** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 12:40.05 | thekidrio | Anyone have a good guide for distro selection for asterisk? |
| 12:40.33 | thekidrio | I am currently leaning towards CentOS/RHEL |
| 12:42.54 | angryuser | thekidrio: i am happy with debian sarge |
| 12:43.26 | tzafrir | thekidrio, you need a guide for that? |
| 12:43.56 | kippi | I keep on getting this error |
| 12:43.58 | kippi | <PROTECTED> |
| 12:44.02 | tzafrir | Debian and CentOS seem to be the most popular. Naturally if you have your favorite distro, you it |
| 12:44.08 | kippi | SIP/1153-006f7250 is ringing |
| 12:46.28 | thekidrio | i like reading opinions yes tzafrir |
| 12:46.51 | thekidrio | server wise i don |
| 12:46.59 | thekidrio | 't have much pref |
| 12:47.14 | thekidrio | most distros look the same in a console really hehe |
| 12:47.45 | thekidrio | looking for the least headache causing route honestly |
| 12:48.57 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 12:51.52 | angryuser | thekidrio: i have used suse 10.1 entreprise also, kernel compatible outbox with misdn (just need to change headers) no pb at all at installation of misdn/astribank/zaptel/libpri/asterisk |
| 12:56.07 | thekidrio | seems then that its fairly compat across the board then with the major distros |
| 12:59.25 | *** join/#asterisk Druken (i=Druken@67.69.139.251) |
| 12:59.49 | Druken | morning everyone, anyone happen to have the pri crossover cable pinouts handy? |
| 13:00.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:04.20 | yansolo90 | hey, anybody knows what login/password are for ssh Cisco 79XX (other than "debug/debug" or "log/log") ? |
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| 13:21.36 | *** join/#asterisk VeNoMouS_ (n=venom@ip-58-28-158-118.ubs-dsl.xnet.co.nz) |
| 13:22.28 | VeNoMouS_ | hay does anyone know when record route is going to be fixed in 302? |
| 13:22.44 | Druken | anyone here done work with a channelbank ? |
| 13:23.52 | tzanger | Druken: yep |
| 13:23.55 | tzanger | all kinds of it |
| 13:23.59 | tzanger | what can I help you with? |
| 13:24.10 | tzanger | Druken: T1 crossover is pin 1->4 and 2->5 |
| 13:24.28 | Druken | well, i have a channelbank... the pri set to NET, i've made the crossover cable, and hooked it all up |
| 13:24.32 | tzanger | uh |
| 13:24.36 | tzanger | channel banks are CAS, not CCS |
| 13:24.57 | Druken | Feb 19 03:21:44 NOTICE[6464]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 13:24.57 | Druken | Feb 19 03:21:45 NOTICE[6464]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 13:25.02 | tzanger | (i.e. signaling=fxs_ks or fxo_ks, not pri_net or pri_cpe) |
| 13:25.04 | Druken | would that explain why i'm, getting those? |
| 13:25.22 | tzanger | I'm not aware of any channel bank which emulates a switch |
| 13:25.54 | Druken | uh... so i need to change the signalling then? |
| 13:26.59 | VeNoMouS_ | so... anyone 302.. route not there ..... |
| 13:28.54 | Druken | tzanger: Signalling requested on channel 1 is FXS Kewlstart but line is in PRI Signalling signalling |
| 13:30.29 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:31.01 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:31.15 | tzanger | Druken: what channel bank is this?? |
| 13:33.14 | Druken | cac |
| 13:33.29 | tzanger | Druken: keep going |
| 13:33.41 | VeNoMouS_ | well if i cant sort this issue im switching to ser |
| 13:33.42 | ManxPower | Druken: that means "You're screwed because data is being lost from the PRI" |
| 13:33.45 | tzanger | I have used Carrier Access Access Bank I and II and Adit 600s... all CCS not CAS |
| 13:34.00 | tzanger | ManxPower: he's using a channel bank, I have never seen a channel bank with PRI signaling |
| 13:34.27 | ManxPower | tzanger: oh, then that message means "You're an idiot, don't set the line as PRI for channel banks" |
| 13:34.37 | Druken | i belive it's a cac1 |
| 13:34.54 | Druken | ManxPower: thanks for those words of encourangement :) |
| 13:35.01 | tzanger | Druken: ok, and use a regular PRI straight-through cable |
| 13:35.21 | tzanger | Druken: make sure zaptel.conf is set for fxs or fxo, whichever is correct for your channel bank (what is in it?) |
| 13:35.25 | tzanger | Druken: and make zapata.conf match |
| 13:35.26 | Druken | so how do i set the signalling properly then ? |
| 13:35.29 | tzanger | don't forget ot re-run ztcfg |
| 13:35.35 | tzanger | (I'm guessing that's your problem right there) |
| 13:35.43 | ManxPower | Druken: do NOT use any switchtype= setting |
| 13:35.46 | ManxPower | that is for PRI |
| 13:36.10 | ManxPower | Druken: put the non-comment lines for your zaptel.conf and zapata.conf on pastebin. |
| 13:36.54 | *** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com) |
| 13:36.55 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 13:36.57 | Telemac | Hello |
| 13:37.23 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:39.03 | ManxPower | hello, Telemac |
| 13:39.25 | Telemac | Has anyone ever tried using cisco 7912 with asterisk ? I'm trying to use it with skinny at least, or with SIP but I didn't succeeded in upgrading its firmware... |
| 13:40.42 | ManxPower | Telemac: many more people use SIP because Asterisk's Skinny/SCCP support is not as stable or full featured. |
| 13:41.33 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 13:41.37 | Telemac | ManxPower: That's what I've read, but the firmware is not longer on cisco website as far as I see, and upgrade procedure is not really clear |
| 13:41.58 | Telemac | ManxPower: I really like to have some docs about this upgrade |
| 13:42.07 | ManxPower | Telemac: Cisco charges for SIP firmware. That is why we do not use them at my clients |
| 13:42.25 | ManxPower | Telemac: SCCP/Skinny works, but you won't find many people that use it. |
| 13:42.28 | Zaw | Cisco charges for everything |
| 13:42.38 | VeNoMouS_ | <ManxPower> Telemac: Cisco charges for SIP firmware. That is why we do not use them at my clients |
| 13:42.41 | VeNoMouS_ | ^^ wtf |
| 13:42.47 | VeNoMouS_ | u just d/l the firmware |
| 13:42.48 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:42.51 | ManxPower | Telemac: later in the day there should be more SCCP/Skinny using people on the channel. |
| 13:42.52 | VeNoMouS_ | and flash the fone |
| 13:43.02 | VeNoMouS_ | i use sccp |
| 13:43.09 | VeNoMouS_ | with 7940's and 41's |
| 13:43.19 | VeNoMouS_ | and 12's |
| 13:43.22 | VeNoMouS_ | had a 60 |
| 13:43.40 | ManxPower | VeNoMouS_: If you have a support contract (most any type of support contract) you CAN download the SIP firmware, but you are violating the Cisco licence and copyright. |
| 13:43.44 | Telemac | VeNoMouS_: The one I try to configure is a 7912 |
| 13:43.57 | HarryR | wow, they charge for sip firmware |
| 13:44.11 | HarryR | I'd hate to think how much it costs per seat just for an average installation |
| 13:44.13 | VeNoMouS_ | technically no |
| 13:44.17 | VeNoMouS_ | they charge for the license |
| 13:44.25 | VeNoMouS_ | ManxPower so whats the problem? |
| 13:44.26 | ManxPower | And individual may not care about not being legally allowed to use the downloaded SIP firmware, but any kind of production enviroment will care. |
| 13:44.33 | VeNoMouS_ | brb |
| 13:44.47 | ManxPower | VeNoMouS_: It increases the cost for 1 thing. |
| 13:45.25 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 13:45.42 | HarryR | wouldn't it be possible just to use snoms or aastras and hack up a sip<->sccp/skinny bridge if you're determined to use CCM? |
| 13:45.58 | Telemac | We have a contract but when we ask for cisco to obtain firmware they didn't give us more information |
| 13:46.10 | ManxPower | HarryR: I believe that CCM supports SIP as well. |
| 13:46.47 | HarryR | ooh |
| 13:47.03 | tzanger | Druken: get it working? |
| 13:47.05 | Telemac | I've d/l firmware from http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx "LATEST FIRMWARE VERSION" but the upgrade is not ok |
| 13:47.32 | ManxPower | Telemac: log into Cisco using your support contract userid/password. Then download the software |
| 13:47.37 | VeNoMouS_ | <ManxPower> Telemac: Cisco charges for SIP firmware. That is why we do not use them at my clients |
| 13:47.58 | Telemac | ManxPower: which software ? |
| 13:48.06 | tzanger | ManxPower: that's a hell of a thread on DNIS on -users... wow |
| 13:48.12 | ManxPower | Telemac: the one for your phone. |
| 13:48.21 | ManxPower | tzanger: yeah, I'm getting sick of it. |
| 13:48.35 | Telemac | ManxPower: ? firmware or software ? |
| 13:51.13 | ManxPower | Telemac: http://cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 |
| 13:51.24 | ManxPower | the 7905 and 7912 use the same firmware |
| 13:52.49 | ManxPower | sorry, they do not use the same firmware, but both phones are on the same firmware page. In fact, here is the SIP firmware for 7912 for non-CCM applications. http://cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/ip-phone/7905/CP7912080001SIP060412A.ZIP&app=Tablebuild&status=showC2A |
| 13:53.41 | ManxPower | You will, of course, need a calid CCO login and password authorized for software downloads. |
| 13:53.48 | ManxPower | calid-valid |
| 13:54.18 | Telemac | ManxPower: And then I just but that firmware on my tftp ? |
| 13:54.28 | ManxPower | Telemac: I don't know. |
| 13:55.45 | ManxPower | Telemac: the readme in the zip file will include information with a link to upgrade instructions |
| 13:55.45 | Telemac | ManxPower: I hope :( |
| 13:55.45 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
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| 13:59.32 | ManxPower | tzanger: the guy has a signalling timing problem. until he gets that fixed nothing is going to work |
| 14:04.57 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 14:05.58 | Druken | tzanger and ManxPower: thanks for the help, got it working |
| 14:06.07 | tzanger | Druken: awesome |
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| 14:09.33 | *** mode/#asterisk [+o mog] by ChanServ |
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| 14:24.28 | JT | what the hell is your problem VeNoMouS_ ? |
| 14:26.42 | JoNate | My damn music on hold still isn't working! |
| 14:27.16 | *** join/#asterisk Underhand (n=gavan@frog.coolfactor.org) |
| 14:27.39 | *** part/#asterisk Roey (n=abc@pdpc/supporter/sustaining/Roey) |
| 14:28.29 | *** join/#asterisk Broom (n=none@12.174.245.7) |
| 14:28.30 | Broom | hey all |
| 14:28.36 | Underhand | i'm having a problem with app_sms.. i'm trying to send sms to * from an SMS-capable DECT handset, via an SPA-3000 |
| 14:28.49 | Broom | can anyone tell me how to stop asterisk loggin either to the file or to the console |
| 14:28.49 | Broom | ? |
| 14:28.50 | Underhand | * is able to receive the message (it appears in the spool file) but the phone reports Message Failed |
| 14:29.04 | Underhand | Broom, logger.conf? |
| 14:29.45 | Underhand | setting verbose shows that after the phone has sent the message, * sends an ACK, and the phone responds with an ERROR of type Wrong message length |
| 14:29.54 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 14:29.58 | Underhand | as far as i can see, the message length is correct (2) |
| 14:30.05 | Underhand | has anybody else seen anything similar? |
| 14:30.10 | Broom | i edited that file |
| 14:30.23 | Broom | and commented out every line |
| 14:30.23 | Broom | and still |
| 14:30.25 | creativx | ~pastebin |
| 14:30.27 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/ |
| 14:32.34 | Underhand | broom: have you tried leaving in a line for console, but with nothing on the right hand side? |
| 14:32.50 | Broom | lets see, wait a sec |
| 14:33.37 | Broom | this is what i get: Logfile Warning: Unknown keyword '' at line 29 of logger.conf |
| 14:34.29 | Underhand | ok, that was just a guess.. |
| 14:34.53 | Broom | underhand: it gave me an error |
| 14:34.56 | Broom | but it worke |
| 14:34.57 | Broom | d |
| 14:35.00 | Broom | it stopped loggin |
| 14:35.05 | Broom | which is what I wanted! thanks |
| 14:37.25 | Underhand | so, anyone with SMS experience? :) |
| 14:38.53 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:42.56 | coppice | Ys, iv snt sms b4 |
| 14:43.06 | tzanger | coppice: :-) |
| 14:43.18 | Ahrimanes | eek |
| 14:43.19 | tzanger | I wish I could find an SMSC for Telus Mobility |
| 14:43.43 | tzanger | I want to send MWI SMS and the email gateways all prepend a space to the msg so I can't send "fun" SMS messages |
| 14:43.53 | coppice | Telus where your SMSC is? |
| 14:44.00 | tzanger | coppice: *groans* |
| 14:44.14 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 14:44.42 | tzanger | coppice: how does your wife put up with you? |
| 14:44.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:45.13 | coppice | as a second language speaker of english, most of it passes over her |
| 14:45.33 | tzanger | coppice: she is fortunate |
| 14:46.25 | Underhand | SMS RX 92 01 02 6B <--- I did understand correctly that that means error, Wrong message length, right? |
| 14:48.27 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:48.51 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:48.58 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 14:49.34 | ManxPower | It sure would be nice if USA carriers like Verizon would have "SMCS" |
| 14:49.46 | ManxPower | or even SMSC |
| 14:50.12 | coppice | they must have. how else would you send an SMS |
| 14:50.12 | ManxPower | ARGH! It is president's day today! |
| 14:50.28 | ManxPower | coppice: you don't. you send it from your handset or via SMTP |
| 14:50.50 | tzanger | ManxPower: yes but the SMTP gateways all talk to SMSCs |
| 14:50.53 | coppice | I bet that's not true for everyone. |
| 14:51.15 | tzanger | I called my Telus rep (a pretty bright guy) and asked for the SMSC phone number or IP address... His precise response was "what's an SMSC?" |
| 14:51.16 | ManxPower | tzanger: yes, but they do not have a real SMSC telephone number to send SMS thru |
| 14:51.41 | tzanger | I've found a few SMS gateways on the intenet for Telus/Bell Mobility but they were all pretty pricey |
| 14:54.16 | coppice | most direct access to an SMSC is through contracted links |
| 14:55.17 | tzanger | coppice: yes, and there are a few SMS gateway providers online which will charge you about US$0.17/SMS to use them |
| 14:56.51 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 14:58.47 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:59.02 | VeNoMouS_ | back |
| 14:59.03 | wwalker | anyone running asterisk in a xen domain? |
| 14:59.21 | JT | VeNoMouS_: so what's your problem? |
| 14:59.23 | JT | seriously |
| 14:59.39 | coppice | you mean like in a Buddist monastery? |
| 14:59.59 | wwalker | coppice: Xen, not Zen! :) |
| 15:00.11 | VeNoMouS_ | JT: that when u transfer you it dont add route in, so u get 0 sip data back when the call ends |
| 15:00.16 | JT | wwalker: no but i plan to |
| 15:00.18 | coppice | its spelt both ways |
| 15:00.27 | JT | VeNoMouS_: oh, i was refering to you flooding the channel |
| 15:00.36 | VeNoMouS_ | and when was i flooding? |
| 15:00.44 | VeNoMouS_ | oh i see back up |
| 15:00.45 | VeNoMouS_ | lol |
| 15:00.51 | VeNoMouS_ | mutsa been cat walking on laptop |
| 15:00.51 | VeNoMouS_ | lol |
| 15:00.56 | JT | i see |
| 15:01.00 | VeNoMouS_ | i been lookin after the kids |
| 15:01.03 | ManxPower | I assumed it was a cat |
| 15:01.06 | JT | fair enough |
| 15:01.33 | VeNoMouS_ | ManxPower so whats your sccp issue now that im back |
| 15:01.57 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 15:02.03 | VeNoMouS_ | so jt is any 1 working on that issue in chan_sip v2? |
| 15:02.17 | JT | i'm not sure, check mantis |
| 15:02.25 | VeNoMouS_ | http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267 |
| 15:02.33 | VeNoMouS_ | thats the only reference i could really find bout it |
| 15:02.41 | VeNoMouS_ | but thats dated jan 2006 :\ |
| 15:03.06 | JT | bugs.digium.com |
| 15:03.15 | wwalker | JT: thx. I've tried running under VMware before but the clock skew/inconsistencies of a VMware guest (even with fixes in place) were just too high. |
| 15:03.16 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 15:03.34 | JT | wwalker: vmware is just... yeah... no good |
| 15:03.42 | JT | massive performance hit |
| 15:03.46 | JT | xen is so low overhead |
| 15:03.49 | JT | efficient |
| 15:04.09 | Makenshi | apples and oranges |
| 15:04.25 | VeNoMouS_ | heh moh sucks on vmware |
| 15:04.25 | JT | sure you can run more OSes in vmware |
| 15:04.30 | VeNoMouS_ | it cant handle it |
| 15:04.31 | elriah | Is skinny NAT friendlY? |
| 15:04.36 | JT | but who cares if you want to do virtualisation |
| 15:04.39 | wwalker | I did some testing with Xen yesterday and it _appears_ that the domains don't actually keep their own time. I think that time calls go all the way to the hypervisor. |
| 15:04.43 | VeNoMouS_ | skinnny isnt friendly to anyone |
| 15:04.44 | Makenshi | vmware is not so good for things like asterisk, but it's great for server consolidation |
| 15:04.46 | VeNoMouS_ | but yes it is |
| 15:04.57 | JT | Makenshi: server consolidation? |
| 15:05.30 | Makenshi | JT, yes, since with Windows VMware can share memory pages between machines, and DR is much much easier |
| 15:05.36 | *** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 15:05.39 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:05.42 | Makenshi | Windows guests, that is |
| 15:05.43 | AlfaScorpii | Hi |
| 15:05.44 | wwalker | Xen is just _so_painful_ to setup new machines |
| 15:05.56 | VeNoMouS_ | lol |
| 15:06.00 | Makenshi | Also, there are plenty of well-supported third-party management tools |
| 15:06.02 | JT | Makenshi: why on earth would you want to do that? i assume the guests are fairly low load |
| 15:06.03 | VeNoMouS_ | u know the answer to that wwalker |
| 15:06.08 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 15:06.11 | tzafrir | anybody tried qemu/kqemu or kmu/kvm ? |
| 15:06.19 | tzafrir | qemu/kvm, that is |
| 15:06.26 | VeNoMouS_ | stop being a cheap bastard and quit using vm's! |
| 15:06.27 | VeNoMouS_ | :P |
| 15:06.32 | JT | ... |
| 15:06.34 | Makenshi | Makenshi, there are many reasons for it |
| 15:06.36 | Makenshi | er JT even |
| 15:06.49 | JT | vms are great when implemented right |
| 15:06.55 | JT | nothing to do with being cheap |
| 15:07.03 | AlfaScorpii | need help with Micronet SP5050 and Asterisk |
| 15:07.09 | AlfaScorpii | please |
| 15:07.18 | Ahrimanes | what's a micronet sp5050 ? |
| 15:07.26 | VeNoMouS_ | jt the only ppl who use vm's are cheap bastards |
| 15:07.32 | JT | VeNoMouS_: wrong. |
| 15:07.33 | VeNoMouS_ | hell even if u have like a sunfire or something |
| 15:07.38 | VeNoMouS_ | ure still being cheap |
| 15:07.48 | JT | you're a f*cking idiot, to put it nicely |
| 15:07.51 | VeNoMouS_ | we did heaps of dev work with xen couple yrs back |
| 15:07.58 | JT | there's these amazing new things |
| 15:08.00 | AlfaScorpii | micronet SP5050 is a voip gateway, im using one as FXO for PSTN interface |
| 15:08.03 | VeNoMouS_ | its shit |
| 15:08.03 | JT | called multi core cpus |
| 15:08.11 | VeNoMouS_ | wtf u think a sunfire is newb? |
| 15:08.12 | tzafrir | I'm checking vserver now. Looks nice if you just want to separate userspace daemons |
| 15:08.15 | JT | that benefit from parallelisation |
| 15:08.22 | VeNoMouS_ | sunfire has more then 8 cpu's lol |
| 15:08.23 | JT | VeNoMouS_: yep, leet speak will get you far |
| 15:08.36 | JT | VeNoMouS_: no shit, and you need the applications to utilise them |
| 15:08.40 | Ahrimanes | tzafrir: we do hosted pbx based on vserver atm, but changing to openvz to get layer 2 networking |
| 15:09.12 | VeNoMouS_ | the fact is , xen is still virtual, it still pushes everything through its own software back plane |
| 15:09.20 | JT | blah blah |
| 15:09.22 | JT | handwaving |
| 15:09.30 | JT | "software back plane" rofl! |
| 15:09.46 | tzafrir | well, debian comes with a vserver kernel. I didn't really bother |
| 15:10.05 | VeNoMouS_ | tzafrir err it does? |
| 15:10.16 | Ahrimanes | tzafrir: yeah, we run debian :) |
| 15:10.43 | infernix | openvz should be ideal if its the same OS for all guests |
| 15:11.22 | infernix | anyway saying that VMs are for cheap bastards is ridiculous |
| 15:11.29 | JT | infernix: indeed |
| 15:11.45 | JT | not everyone using VMs is in the virtual hosting business |
| 15:11.52 | Ahrimanes | we can run about 50 vm's on a dual cpu machine.. |
| 15:11.54 | JT | [TK]D-Fender: uhh, ok... |
| 15:11.59 | coppice | VMs are really for people who's like isn't hard enough right now |
| 15:12.10 | VeNoMouS_ | Ahrimanes and u run that in a production envoriment too? |
| 15:12.14 | coppice | s/like/life |
| 15:12.26 | Ahrimanes | VeNoMouS_: yes, with customers paying happily |
| 15:12.29 | Makenshi | coppice, untrue :p |
| 15:12.36 | [TK]D-Fender | Ok people, feel free to agree to disagreee any time now... |
| 15:12.42 | AlfaScorpii | i have problems routing incoming calls from PSTN-VOIP GATEWAY-ASTERISK |
| 15:12.45 | AlfaScorpii | :( |
| 15:12.54 | Makenshi | for instance, vmware ha will handle the failure of physical resources and reallocate the guests when necessary, reducing downtime |
| 15:12.56 | AlfaScorpii | i can make calls to the ouside |
| 15:13.06 | AlfaScorpii | but cant recibe calls from the outside |
| 15:13.15 | VeNoMouS_ | so basicly your saying is u give your customers bout 120mhz each |
| 15:13.38 | VeNoMouS_ | if they were to max out all there cpu policy |
| 15:13.53 | VeNoMouS_ | or was 50 just some number u pulled out of the air |
| 15:13.57 | *** join/#asterisk ryant (n=ryant@4.17.197.118) |
| 15:14.27 | *** join/#asterisk hal23456 (n=chatzill@host86-149-56-223.range86-149.btcentralplus.com) |
| 15:14.27 | infernix | VeNoMouS_: your logic is flawed |
| 15:14.31 | JT | the applications i'm thinking of a more like a 1 VM to 1 core mapping |
| 15:14.39 | JT | say 8VMs to 8 cores |
| 15:14.43 | Ahrimanes | VeNoMouS_: 50 is our max for any piece of hardware.. cpu and memory usage is nowhere near maxed out with 50, but we dont want any more customers to be affected by hardware failure |
| 15:14.44 | ManxPower | I'll stick to real hardware, thakyouverymuch. No drama. |
| 15:14.46 | JT | performance should be ok :) |
| 15:15.15 | elriah | I missed the first part of this, what's the goal of your VM scenerio? |
| 15:15.17 | VeNoMouS_ | ManxPower word |
| 15:15.18 | ManxPower | For a hosting company I can see using virtualization |
| 15:15.19 | hal23456 | hi all! Does anyone know why some analogue phones connected to the TDM400P don't ring for an incoming call, but others are fine? |
| 15:15.22 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:15.31 | ManxPower | hal23456: usually a config error |
| 15:15.35 | infernix | even with real hardware there's much to say about using VM - even just with one guest - such as VM suspend/resume, live migration etc |
| 15:15.45 | VeNoMouS_ | hal23456 : could be a voltage thing |
| 15:16.03 | JT | ManxPower: the latest server processors benefit a lot from virtualisation, in that most conventional applications don't take fully advantage of so many cores |
| 15:16.08 | hal23456 | Manx - but if I change the phone for one that I know works, it works fine |
| 15:16.35 | JT | yeah vm suspend/resume and migration is quite sweet |
| 15:16.48 | hal23456 | Venomons - yes, I believe this is most likely to be true - is there any way to amend the config so it supports the voltage of the phone? |
| 15:17.04 | Ahrimanes | an average hosted customer in our setup eats around 24 megs of physical ram... |
| 15:17.11 | VeNoMouS_ | hal23456 *shrug* |
| 15:17.22 | elriah | Ahrimanes: What are you hosting? |
| 15:17.39 | VeNoMouS_ | some fones are wanky about the voltage they get to generate a ring tone |
| 15:17.48 | hal23456 | also, how can you tel whether a phone will be compatible? |
| 15:17.53 | VeNoMouS_ | we had a couple cisco ata's that sorted sending enuff voltage |
| 15:18.02 | [TK]D-Fender | hal23456: So you're saying that using the same port, if you swap the phone attached to it, the 2nd one rings properly as opposed to the 1st? |
| 15:18.03 | VeNoMouS_ | so fones wouldnt flash or hang up properly |
| 15:18.17 | Ahrimanes | elriah: asterisk pbx's for customers |
| 15:18.36 | hal23456 | it would be a very expensive task to test lots of different phones, just to see if they are compatible with asterisk |
| 15:18.50 | hal23456 | surely they shold follow some standard? |
| 15:19.17 | elriah | Ahrimanes: How much per month? |
| 15:19.48 | hal23456 | yes, that is right, fender |
| 15:19.54 | Ahrimanes | elriah: around $12/active extension/month |
| 15:20.13 | hal23456 | bare in mind that these are analogue not voip phones |
| 15:20.16 | elriah | Does that include the trunks? |
| 15:20.26 | [TK]D-Fender | hal23456: Just a silly question, but have you physically verified that the ringer wasn't disabled on those phones? |
| 15:20.41 | hal23456 | yes, I have fender |
| 15:20.48 | Ahrimanes | elriah: no, that unfortunately depends alot on the internet providers here in denmark |
| 15:21.02 | *** part/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 15:21.25 | *** join/#asterisk mavior (n=Miranda@88-149-162-164.f5.ngi.it) |
| 15:21.29 | [TK]D-Fender | hal23456: Do you see any pattern in the ones that aren't acting right? (different number of attached devices/port, all the same model, etc) |
| 15:21.42 | mavior | hello dudess |
| 15:22.14 | hal23456 | if nothing springs to mind, I will go and confirm again my setup and report back... |
| 15:22.33 | mavior | somebody knows how to re-set my voicemail unavaible message to the asterisk default one? |
| 15:22.50 | [TK]D-Fender | mavior: Delete the one you recorded |
| 15:22.59 | mavior | how? :P |
| 15:23.05 | VeNoMouS_ | mavior cp the wav |
| 15:23.19 | mavior | there are no voicemailMain() option to do so |
| 15:23.20 | hal23456 | fender - wel, I only have a couple of phones, so it is difficult to determine a general rule of thumb, however, I think venomous is right that it is about voltage |
| 15:23.43 | [TK]D-Fender | mavior: "rm /var/spool/asterisk/voicemail/[context]/[box]/unavail* |
| 15:24.05 | mavior | oh ok...thank you |
| 15:24.34 | mavior | and are there such options to play some combinations of default ast messages instead of the default one? |
| 15:24.46 | [TK]D-Fender | hal23456: Do you have a lto fo phones loaded on the flakey ports? |
| 15:24.55 | [TK]D-Fender | "lot of"* |
| 15:27.14 | elriah | Anyone use chan_skinny for cisco phones? If so, can you point me to a sample XML config for the phone 79x1's? |
| 15:28.39 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 15:29.17 | ManxPower | mavior: either record a new message, or remove the greeting files from the /var/spool/asterisk/voicemail/whatever directory outside of Asterisk |
| 15:29.43 | Telemac | elriah: I too trying to use asterisk with cisco 79xx, SIP seems better but firmware upgrade is a pain |
| 15:29.52 | [TK]D-Fender | elriah: Boy you really are completely up a creek with these phones. I don't think there's been one aspect you haven't had to come crawling back in here for help on.... |
| 15:30.06 | mavior | somebody knows? I mean "combinations of default ast messages" --> "combinations of default ast sounds" instead of the dafult voicemail one |
| 15:30.36 | [TK]D-Fender | ManxPower: We really ought to fix all these odd bits though. Stuff you should be able to do from VoiceMailMain..... |
| 15:30.48 | ManxPower | mavior: there are no defauilt asterisk sounds for voicemail other than the default voicemail sounds |
| 15:30.59 | Ahrimanes | [TK]D-Fender: yeah, like disabling the user menu :) |
| 15:31.15 | [TK]D-Fender | mavior: You get the "glued together by Allison" message when you have no recorded one. |
| 15:31.17 | elriah | [TK]D-Fender: Yea, and I knew it going in. They work great as long as NAT isn't involved. We're going to try chan_skinny and see if it will solve our NAT issues. |
| 15:31.40 | ManxPower | elriah: last I heard Skinny has NO support for ANY NAT |
| 15:31.56 | ManxPower | since skinny has no authentication I assume it was meant for only LAN usage |
| 15:32.47 | ManxPower | mavior: you want the default greeting files played like before you recorded your voicemail greeting, correct? |
| 15:32.55 | mavior | Manx and Fender...OK but if i want |
| 15:32.56 | mavior | no man... |
| 15:33.21 | ManxPower | mavior: you can either have your own custom recorded greetings or the default greetings. You have no other choices. |
| 15:33.30 | mavior | I got it....i understand how to revert to the default one(just deleting my own custom message) |
| 15:34.47 | ManxPower | Actually you do have one other option. Use Playback before running Voicemail to play whatever sound file you want, then run Voicemail with the "s" option. |
| 15:34.56 | VeNoMouS_ | <PROTECTED> |
| 15:35.21 | VeNoMouS_ | anyway |
| 15:35.23 | VeNoMouS_ | im going sleep |
| 15:35.25 | VeNoMouS_ | l8rs |
| 15:35.27 | ManxPower | VeNoMouS_: is this a CCM feature or a phone feature? |
| 15:35.29 | VeNoMouS_ | Tue Feb 20 04:34:41 NZDT 2007 |
| 15:35.55 | Telemac | I've setup a tftp server on my linux box, when my 7219 start I can see with tcpdump that it access tftpserver but on status messages on the phone I get error about timeout to get file (XMLDefault.cnf.xml and SEPXXXXXX.cnf.xml). What can cause that ? |
| 15:35.59 | VeNoMouS_ | ManxPower im pretty sure it supports nat |
| 15:36.00 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:36.04 | Ahrimanes | why do we need to see those timestamps? |
| 15:36.25 | VeNoMouS_ | Telemac data chunk size on your tftp |
| 15:36.27 | VeNoMouS_ | d |
| 15:36.39 | VeNoMouS_ | linux tftpd or wintendo? |
| 15:36.50 | VeNoMouS_ | if using aftpd change the data size |
| 15:36.59 | VeNoMouS_ | atftpd even |
| 15:37.13 | VeNoMouS_ | its a flag btw |
| 15:37.14 | VeNoMouS_ | good nite |
| 15:37.27 | Telemac | VeNoMouS_: tftp server is on linux |
| 15:37.37 | mavior | Oh ok ManxPower, that's what i want.....because i want to achieve this behaviour: have the default asterisk voice say instead of this "The person at extension ... 1234 ... is unavailable" --> "John ... is unavailable" Please leave a message after the tone |
| 15:37.53 | mavior | or better "Marco ... is unavailable Please leave a message after the tone" |
| 15:38.24 | ManxPower | mavior: Why can't you use the custom greeting you record in voicemail for that? |
| 15:38.47 | Ahrimanes | ManxPower: i guess he wants it like that for all users |
| 15:39.13 | ManxPower | Ahrimanes: I guess he wants lots of work to do. |
| 15:39.27 | Ahrimanes | ManxPower: true |
| 15:39.27 | mavior | cause i don't want my voice to be played....but the default one |
| 15:39.46 | ManxPower | mavior: you are going to have someone record everyone's name? |
| 15:40.47 | mavior | probably...yes...or not.....ehm....by the way...i'm new to voicemails..im only in the very alpha-testing of the features right now :P |
| 15:41.26 | mavior | just a bit of curiosity and nerdesses from my side |
| 15:41.29 | mavior | :) |
| 15:41.32 | mavior | just a bit |
| 15:41.51 | ManxPower | mavior: you have two options. You can fight Asterisk oddities and live a miserable life and hate Asterisk or you can accept Asterisk's oddities and live a happy life and love Asterisk. |
| 15:42.01 | Telemac | What's the block size for tftp so that cisco phone can work with ? |
| 15:42.01 | [TK]D-Fender | mavior: Then have them record their name for the directory. It should play the name instead of the box # |
| 15:42.10 | ManxPower | Too many people hate the first option |
| 15:42.27 | ManxPower | ..er... Too many people pick the first option. |
| 15:42.49 | mavior | yes Fender.....is it possible? |
| 15:43.48 | ryant | anyone have any cool asterisk or digium wallpapers? |
| 15:43.50 | [TK]D-Fender | mavior: When I just finish telling you what to do, how about you go TRY it? :) |
| 15:44.02 | mavior | manx,i think that this is true indeed for almost the 90% software around |
| 15:45.15 | mavior | Fender ? (i'm italian..sometimes probably i miss some sarcasm) |
| 15:48.17 | hal23456 | ok, I have fully tested the analogue phones, and the ones that don't work definitely don't work, and the ones that do...err..do. I have tested the cables, which are also ok. I only have one phone connected per module on the TDM400P |
| 15:49.23 | hal23456 | I have found that I can't hear a dial tone when the "non-working" phones are connected, but if I connect them directly to the analogue phone line, they work correctly |
| 15:49.24 | [TK]D-Fender | mavior: First you described what you needed. Then I told you how to do it. Then you ask me if its possible. There is something inherently wrong with that... |
| 15:50.49 | mavior | oh ok...I don't understand how to play the name instead of extension name. |
| 15:53.15 | hal23456 | my questions are, 1) is there any way to modify asterisk/zap drivers to support these (currently) non-working analogue phones and 2) is there any way to tell (before buying them) which phones are no likely to be compatible with asterisk/zap and 3) has anyone else had problems getting some analogue phones to work (or is it just me!!) ? |
| 15:54.44 | ManxPower | hal23456: every single analog phone I've put on a zaptel card worked. |
| 15:55.09 | ManxPower | hal23456: if you have a long run of wire, or too high of an REN you may want the boostringer option to the kernel driver |
| 15:55.21 | *** join/#asterisk robsdesk (n=bab610c5@adsl.ntsols.com) |
| 15:55.43 | tzafrir | hal23456, what do you mean by "non-working"? You don't get a dialtone from Asterisk? |
| 15:55.56 | robsdesk | hi is there a problem with the digium cvs service? |
| 15:55.57 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9d221db515a08342) |
| 15:56.05 | hal23456 | yes, that is right, trafrir |
| 15:56.06 | tzafrir | You only get a dialtone if asterisk is running and a proper channel is defined in zapata.conf |
| 15:56.53 | ManxPower | hal23456: I have never, ever heard of an analog phone not getting dialtone from a correctly configured and wired asterisk box |
| 15:56.55 | hal23456 | I have configured the channel etc, and it works fine with another analogue phone, but not 2 phones I have discovered |
| 15:57.11 | tzafrir | hal23456, does any other phone work with your TDM400P card? |
| 15:57.12 | ManxPower | I've only heard of issues of ringing |
| 15:57.20 | tzafrir | ok |
| 15:57.22 | hal23456 | yes, it does trafrir |
| 15:57.26 | elriah | Does anyone have a working SEP<mac>.cnf.xml for chan_skinny and Cisco 7941/7961s? |
| 15:57.30 | hal23456 | but not 2 (different models) of phones |
| 15:57.52 | mavior | Fender:I don't understand how to play the name instead of extension name.It was a question. :) |
| 15:57.56 | hal23456 | really, manx? I cannot explain why I have the problem |
| 15:58.11 | *** join/#asterisk Feroxis (i=Feroxis@186.84-49-72.nextgentel.com) |
| 15:58.28 | ManxPower | hal23456: You are plugging the non-working phones into the SAME PORT as the working phone for testing? |
| 15:58.42 | hal23456 | yes, that is right, manx |
| 15:58.58 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 15:59.07 | hal23456 | It has to be something to do with voltage, I think, as venomous said |
| 15:59.13 | [TK]D-Fender | mavior: I just told you. Delete the unavailable message, and have them record their name for the directory. |
| 15:59.46 | hal23456 | I hoped that someone may know a workaround, or how to determine whether a phone is likely not to be compatible before purchasing it |
| 16:00.05 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:00.11 | ManxPower | hal23456: ALL standard analog phone should work. |
| 16:00.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:00.27 | Ahrimanes | hey russellb my devstate hero ;) |
| 16:00.28 | ManxPower | hal23456: what is the make/model of one of the non-working phones. |
| 16:00.40 | ManxPower | hal23456: voltage is only an issue for ringing. |
| 16:00.44 | kippi | which file do i need to config for pickup groups? |
| 16:00.47 | ManxPower | not for getting a dialtone |
| 16:01.09 | ManxPower | kippi: zapata.conf, sip.conf, iax.conf, sccp.conf, skinny.conf, h323.conf |
| 16:01.36 | kippi | ok thanks |
| 16:01.47 | ManxPower | kippi: oh, and extensions.conf of course |
| 16:01.59 | hal23456 | Unfortunately I do not have one non-working phone with me, and I am pretty sure I could get a dialtone, but it would not ring. |
| 16:02.42 | hal23456 | the current one I can't get either a dialtone or a incoming ring |
| 16:03.03 | phearless | I got a strange problem, when I call with a mobile my asterisk system, I pick up the VoIP phone (phone 408), then I transfer the call (xfer key on a Linksys/Sipura SPA942), then I transfer the call to the phone 404, then the problem : the guy on the mobile do NOT hear the 404 guy, and the 404 guy DO hear the guy on the mobile. How can I debug this? |
| 16:03.33 | hal23456 | I think, actually, that there is a relationship between the two phones - they are BT (British Telecom) phones |
| 16:03.38 | [TK]D-Fender | phearless: Sounds like a NAT problem. |
| 16:03.43 | hal23456 | has anyone tried these, to their recollection? |
| 16:03.45 | ManxPower | phearless: sounds like a NAT problem from the very confuzing description you just gave. |
| 16:03.58 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:04.04 | ManxPower | hal23456: is it a 2-wire phone or a 3-wire phone? |
| 16:04.18 | [TK]D-Fender | phearless: And never ever give someone extension 404.... its just bad karma... |
| 16:04.20 | phearless | [TK]D-Fender and ManxPower ok I will investigate this.... by the way I use a PRI ISDN line |
| 16:04.26 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 16:04.29 | hal23456 | oooh, I don't know Manx - how do I tell? |
| 16:04.29 | ManxPower | Asterisk only supports 2-wire phones. You can buy a 3-wire to 2-wire adaper |
| 16:04.49 | ManxPower | hal23456: I have no idea. |
| 16:05.00 | hal23456 | that is very interesting information, manx |
| 16:05.23 | ManxPower | hal23456: As I understand it the original UK phone lines used 3-wires. I don't know when 2-wire lines started to be used. |
| 16:05.28 | hal23456 | do you have any suggestions where I could obtain a 3wire to 2wire adatper? |
| 16:05.50 | phearless | NAT problems are mostly when people use VoIP over internet? right? |
| 16:05.53 | ManxPower | hal23456: your local retailer might have them. search the mailing list archives for more information |
| 16:05.56 | ManxPower | ~mailinglist |
| 16:05.57 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 16:07.40 | [TK]D-Fender | phearless: Where are each of your phones located relative to your server? |
| 16:07.54 | kippi | for pickup groups I just need to add pickgroup=1 to the sip.conf to the two extenstions i want to pickup? |
| 16:08.11 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:08.53 | hal23456 | thank you, Manx, and everyone for your valuable help. I appreciate it |
| 16:08.56 | phearless | 10.2.12.104 is 404, 10.2.12.214 is 414, and asterisk is 10.2.8.1 |
| 16:09.00 | phearless | [TK]D-Fender |
| 16:09.46 | phearless | 255.255.192.0 is the netmask |
| 16:10.20 | [TK]D-Fender | phearless: Does that mask imply they are on the same local LAN> |
| 16:10.25 | phearless | yes |
| 16:10.35 | kippi | working |
| 16:10.43 | kippi | anyone know what this error means? |
| 16:10.43 | kippi | Feb 19 17:11:39 WARNING[5633]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) |
| 16:10.44 | phearless | the phones are connected to asterisk, because they can call each others |
| 16:10.49 | [TK]D-Fender | phearless: Ok, NAT shouldn't be an issue... |
| 16:11.08 | Telemac | Why cisco 7912 phone could experience trouble with a ftp server under linux ? I see the phone access the server but I still get timeout error ... |
| 16:12.30 | *** join/#asterisk w0ls0n (n=Me@43-141-135-64.dsl.sacoriver.net) |
| 16:12.45 | phearless | ok [TK]D-Fender |
| 16:12.57 | phearless | it is a quite weird problem |
| 16:13.01 | ManxPower | kippi: "show codecs" will tell you the number / codec names |
| 16:13.07 | w0ls0n | Hi all. I have in my sip.conf my dialing provider but I cannot seem to make a call. I get a dialtone but how do I make outgoing calls? |
| 16:13.35 | ManxPower | w0ls0n: you need to configure you phone to allow that |
| 16:13.48 | ManxPower | w0ls0n: SIP phones provide dialtone, not Asterisk. |
| 16:13.50 | w0ls0n | I have a softfone called x-lite |
| 16:14.02 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 16:14.22 | kippi | so ManxPower: I have a problem with my codecs? |
| 16:14.29 | Telemac | 7912 phone has a tmout error in status message about XMLDefault.cnf.xml but when I try to get the same file by hand against the same server (with a CLI tftp client), everything is ok |
| 16:14.58 | w0ls0n | I have my softfone as ext 101 so how do I make the dialing work |
| 16:17.29 | ManxPower | ~codec |
| 16:17.30 | jbot | rumour has it, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works |
| 16:17.37 | [TK]D-Fender | w0ls0n: Go lern how to create your dialplan.... |
| 16:17.51 | [TK]D-Fender | w0ls0n: That would be "extensions.conf" in case you were wondering.... |
| 16:17.56 | [TK]D-Fender | ~book |
| 16:17.57 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:18.03 | ManxPower | ~codec |
| 16:18.08 | [TK]D-Fender | w0ls0n: Go download and read THE BOOK. |
| 16:18.28 | w0ls0n | I actually have the book downloaded already, just needed something to start with |
| 16:20.16 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:20.40 | [TK]D-Fender | w0ls0n: Good. Start with.... THE BOOK :) |
| 16:20.47 | w0ls0n | really |
| 16:22.33 | badcfe | i now see that i have "writeprotect=no" in my extensions.conf isnt this dangerous? |
| 16:23.47 | Corydon-w | not unless you type 'save extensions' |
| 16:23.51 | ManxPower | ~codec |
| 16:24.03 | badcfe | good lord. im glad i havent. |
| 16:25.58 | ManxPower | ~codecs |
| 16:25.59 | jbot | from memory, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 16:26.18 | ManxPower | I think that should read "If you want to |
| 16:26.50 | ManxPower | I think that should read "If you want to CAUSE codec problems first try disallow=all and allow=all" |
| 16:27.38 | *** join/#asterisk darken_darken (n=marco@248.189.76.83.cust.bluewin.ch) |
| 16:27.43 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 16:27.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:27.47 | kippi | when I transfer calls, i get one way traffic, i can hear them but not the other way, anyideas? |
| 16:28.00 | w0ls0n | Feb 19 11:26:53 NOTICE[717]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 16:28.09 | w0ls0n | ohhh wait a sec |
| 16:28.49 | ManxPower | kippi: Sounds like a NAT problem. |
| 16:29.39 | jserve | *hmms* I have question to the G.729 Licenses, when I have one Softphone what supports G.729. Do I need on the Server 2 G.729 Licenses (one for encoding/one for decoding) when the server isn't working in passthru? |
| 16:29.45 | De_Mon | the last call event for this extension is to enter a Meetme room with some arguments. The lastdata CDR field is empty though. |
| 16:30.10 | De_Mon | after the meetme it goes to the hangup extension... I bet that's screwing up the cdr |
| 16:30.26 | kippi | ManxPower: this is over a local network |
| 16:31.32 | De_Mon | damnit that's exactly whats going on. |
| 16:31.42 | De_Mon | hrm... how to track the conference number |
| 16:32.30 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:35.57 | *** join/#asterisk J4k3 (i=jsuter@226.sub-70-216-111.myvzw.com) |
| 16:41.58 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 16:43.27 | ManxPower | kippi: then look at your codecs. disallow=all and allow=ulaw |
| 16:44.29 | *** join/#asterisk J4k3^ (i=jsuter@181.sub-70-216-114.myvzw.com) |
| 16:45.59 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 16:48.03 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:48.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:49.13 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
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| 16:50.07 | dansmith | anyone have anything good or bad to say about broadvoice? |
| 16:50.25 | MarkWD | We have set up a test server and all is well. We are wondering on a single server how many connections can be made before the quality goes to hell on us? |
| 16:50.43 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 16:51.34 | [TK]D-Fender | MarkWD: You question has a giant "depends" looming over it... |
| 16:52.46 | [TK]D-Fender | dansmith: Somewhat flakey. Occasional DNS issues with their proxies winking out, audio quality somewhat infrequently, idiotic support staff. |
| 16:52.59 | dansmith | heh |
| 16:53.01 | MarkWD | of course but if we fill up one pri |
| 16:53.14 | dansmith | any suggestions for a less-flaky one? |
| 16:53.37 | De_Mon | broadvoice's call quality wasn't as good as we wanted. I'm using bandwidth.com |
| 16:53.54 | [TK]D-Fender | dansmith: Teliax has been recommended by some. I have clients on VoicePulce Connect who've not ill to speak of them. |
| 16:53.56 | De_Mon | more expensive but also crystal clear |
| 16:54.10 | ManxPower | dansmith: All ITSPs suck. Teliax seems to suck less than most. Since your call is going over the internet, there is nothing you can do about network related call quality issues. |
| 16:54.24 | kippi | could this error be why I am getting one way voice: http://pastebin.ca/363692 |
| 16:54.28 | dansmith | ok, well, it's not going to be worth it to me unless it's significantly cheaper than my analog line, of course.. just thought i'd investigate |
| 16:54.29 | De_Mon | ManxPower get a ITSP closer to your server |
| 16:54.46 | ManxPower | De_Mon: That does not change my stance. |
| 16:54.59 | dansmith | playing with freecall gave me surprisingly good quality, aside from the delay (although I was calling from the US to Australia) |
| 16:55.05 | ManxPower | kippi: that is all normal messages and does not indicate a problem |
| 16:55.07 | MarkWD | with a server 1g local network 8g of ram and dual processors on RH AS4 to a pri |
| 16:55.10 | De_Mon | ManxPower if the network is giving you problems reduce the network path. |
| 16:55.19 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 16:55.23 | kippi | how can I trace this problem down? |
| 16:55.29 | ManxPower | De_Mon: no, if the network is giving you problems, add QoS |
| 16:55.53 | De_Mon | ya, good luck getting that implimented across your 30 hop route... |
| 16:56.16 | ManxPower | Obviously the fewer number of hops the less issues you will have, but that does not change the fact that you have non-qos service going thru routers you do not manage |
| 16:56.29 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 16:56.31 | tzanger | most times QoS is a problem at the last hop of each side |
| 16:56.45 | tzanger | the middles are *usually* not the problem |
| 16:56.47 | ManxPower | and unless you use the same ISP as your ITSP company, your data is transiting networks of which you are NOT A CUSTOMER |
| 16:57.02 | tzanger | if you're using cheapass internet though you could be using a provider who's grossly overcommitted his bandwidth |
| 16:57.46 | dansmith | teliax's prices seem reasonable |
| 16:59.26 | ManxPower | kippi: Sorry, but I'm just burnt out on newbie questions for today. |
| 17:01.26 | kippi | ManxPower: where can I read up on things like this? really need to get this fix, its a live system |
| 17:03.24 | *** join/#asterisk J4k3- (i=jsuter@211.sub-70-216-29.myvzw.com) |
| 17:03.27 | [TK]D-Fender | kippi: pastebin at leat an ENTIRE call, from the initial answer, through the transfer, to the very end. Not just a little snippet in between. |
| 17:03.48 | kippi | ok |
| 17:05.12 | kippi | http://www.pastebin.ca/363709 |
| 17:05.31 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 17:06.59 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:08.54 | [TK]D-Fender | kippi: Now do the same with SIP debug enabled |
| 17:09.23 | *** join/#asterisk BPJ (n=jannie@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:11.07 | *** part/#asterisk BPJ (n=jannie@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:11.40 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 17:11.48 | *** join/#asterisk J4k3 (n=jsuter@12.45.185.225) |
| 17:12.12 | kippi | http://www.pastebin.ca/363725 |
| 17:12.37 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 17:14.44 | *** join/#asterisk BPJ (n=jan@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:15.22 | *** part/#asterisk BPJ (n=jan@rrcs-67-78-88-186.sw.biz.rr.com) |
| 17:16.21 | [TK]D-Fender | kippi: I might suspect that the user isn't waiting for the transfer to complete before haning up and killing the call... |
| 17:16.25 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
| 17:17.40 | kippi | where would this be set? |
| 17:19.01 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 17:19.30 | De_Mon | OIY, Why doesn't asterisk raise a WARNING when an 'exten => foo,n,' has no preceeding priority |
| 17:19.55 | Qwell[] | De_Mon: the operator should be smarter than that :) |
| 17:20.01 | Qwell[] | De_Mon: and there are very valid reasons to NOT have one |
| 17:20.12 | De_Mon | like? |
| 17:20.22 | [TK]D-Fender | kippi: This isn't a "setting" this is a user not doing a transfer properly on their phone and hanging up before its properly completed |
| 17:20.23 | Qwell[] | like regcontext in sip.conf or whatever |
| 17:20.24 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 17:20.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:20.31 | Qwell[] | or having _.,1,NoOp(something) |
| 17:20.39 | De_Mon | ahh |
| 17:20.50 | kippi | ah ok |
| 17:20.52 | Qwell[] | (I'm not suggesting anybody use it, but there are very valid reasons for doing so - which I won't explain..) |
| 17:21.03 | CJLinst | I'm trying to change the identity of an SNOM 360 and keep getting this: "chan_sip.c:8065 check_auth: username mismatch, have <226>, digest has <221>" - phone can receive calls but cannot make them. |
| 17:21.10 | CJLinst | Any suggestions? |
| 17:21.27 | Qwell[] | De_Mon: or say you have a setup where you only want an extension to be "live", if you add the 1 priority from the CLI |
| 17:21.33 | CJLinst | 226 is old ID, 221 new. |
| 17:21.38 | [TK]D-Fender | CJLinst: Yeah. Fix your phone. |
| 17:21.55 | De_Mon | Well after spending 30min trying to figure out why my extension didn't exist that little jem finaly got my attention |
| 17:22.20 | De_Mon | I doubt it'll take me this long to check the priority next time hehehe |
| 17:22.28 | [TK]D-Fender | De_Mon: you = silly |
| 17:23.12 | ManxPower | kippi: read the manual for the phone to make sure the user is correctly doing the transfer |
| 17:23.14 | [TK]D-Fender | De_Mon: Always look for your first priority, double check the context and pattern match (forgot a "_" where needed? Oh noes!) |
| 17:23.19 | De_Mon | unexperienced |
| 17:23.54 | De_Mon | [TK]D-Fender make sure you're not using a reserved character [xn .. some others] |
| 17:24.20 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:24.21 | [TK]D-Fender | De_Mon: yup. |
| 17:25.05 | kippi | The phone is saying transfer ok |
| 17:25.50 | ManxPower | kippi: Since it is a Grandstream I would immediatly suspect a firmware bug |
| 17:25.55 | kippi | ah lol |
| 17:25.57 | kippi | ok |
| 17:26.05 | kippi | I'll have a look for a upgrade |
| 17:26.40 | ManxPower | kippi: Grandstream has the worst history of releasing stable firmware of any IP phone company I am aware of. |
| 17:26.42 | mafkees | it's a phone issue |
| 17:26.59 | mafkees | those grandstreams never come with correct firmware |
| 17:27.10 | ManxPower | What most people do is just keep trying different versions of the firmware until they find one that does not have serious bugs for the features they require |
| 17:27.25 | ManxPower | Sometimes the process takes weeks |
| 17:27.31 | kippi | great |
| 17:27.43 | [TK]D-Fender | GrandSuck should be avoided with extreme prujdice. Period |
| 17:28.11 | [TK]D-Fender | prejudice* |
| 17:28.12 | [TK]D-Fender | kdsfahkjsdfalhkjasdhf |
| 17:28.54 | ManxPower | I can't imagine anyone using grandstream products in a production enviroment unless they are trying to get fired. |
| 17:31.32 | [TK]D-Fender | ManxPower: No, there are the "terminally frugal" who will ignore advice of those who know better feeling assuered that because they bough one for home and can't complain that its can't be as bad as they were warned. People assigning insufficient budgets is a key factor. |
| 17:32.17 | Nugget | What ManxPower isn't telling you is that he uses all Grandstream phones clone X100P boards at work. In eMachines servers that he bought from Costco. |
| 17:35.58 | *** join/#asterisk J4k3^ (n=jsuter@12.45.185.225) |
| 17:36.32 | phearless | second try : |
| 17:36.33 | phearless | I got a strange problem : when I call from a mobile to asterisk system, I pick up the VoIP phone (phone 408), then I transfer the call (xfer key on a Linksys/Sipura SPA942), then I transfer the call to the phone 404, then the problem : the guy on the mobile do NOT hear the 404 guy, and the 404 guy DO hear the guy on the mobile. How can I debug this? |
| 17:36.42 | phearless | I still have not fixed that... |
| 17:37.48 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 17:38.09 | [TK]D-Fender | phearless: Try adding "canreinvite=no" to each of your phones entries, and in [general] |
| 17:38.11 | ManxPower | phearless: you want a 3-way call, not a transfer |
| 17:38.32 | phearless | I do not want the 404 guy to be still on the phone |
| 17:38.48 | *** join/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 17:39.11 | *** part/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 17:39.17 | ManxPower | phearless: check the manual for the phone, sounds like the transfer is not completed. most phones require another button to be pressed (transfer?) to complete the transfer |
| 17:39.25 | phearless | ok |
| 17:39.31 | phearless | I will retry ! |
| 17:40.19 | [TK]D-Fender | ManxPower: Um... he says the audio path is ther, just 1-way. thats not a 3-way issue (and he specified the soft-key being used even). |
| 17:40.47 | ManxPower | [TK]D-Fender: What is it with all these non-NAT 1-way audio problems today? |
| 17:41.02 | [TK]D-Fender | ManxPower: CRAZYNESS I say! |
| 17:41.34 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 17:41.40 | elriah | Oh, dear god of Cisco firmware, please deliver to us a NAT capable SIP firmware, for we have invested in your infinite wisdom, for we have decision makers that make buying decisions based on their stock portfolio, for we cannot return to the likes of Polycom, in your name we pray. |
| 17:41.57 | Qwell[] | elriah: use sccp :p |
| 17:42.00 | [TK]D-Fender | ManxPower: I for one believe that its having a non-standard net-mask in on of the cases shown today. Many devices don't like this even if it is "legal". |
| 17:42.03 | Qwell[] | or, skinny rather |
| 17:42.30 | elriah | Qwell[]: We're experimenting with channel_skinny today with 7941's. Have you had luck with 7941's and Asterisk |
| 17:42.31 | elriah | ? |
| 17:42.32 | [TK]D-Fender | elriah: s/invested/"wasted scads of money having been warned against"/ |
| 17:42.38 | Qwell[] | nope, don't have any 7941s |
| 17:42.44 | elriah | [TK]D-Fender: I know, I know. lol |
| 17:43.19 | [TK]D-Fender | elriah: Just adding a litle salt, pepper, some lemon juice, and a sprig of parsley to your wound :) |
| 17:43.42 | elriah | Qwell[]: Well, we got the phones 'trying' to register with skinny, but asterisk reports trying to send a 12SP template to the phone and a bunch of "RECEIVED UNKNOWN MESSAGE TYPE" messages. Any suggestions? |
| 17:43.45 | [TK]D-Fender | (parsley for color only. No noticable effect on nutrient value) |
| 17:43.46 | *** join/#asterisk J4k3 (n=jsuter@12.45.185.225) |
| 17:43.53 | Qwell[] | eh? |
| 17:43.57 | elriah | [TK]D-Fender: lol |
| 17:44.15 | Juggie | elriah, send Qwell a phone. |
| 17:44.38 | elriah | Qwell[]: 7941's |
| 17:44.38 | Qwell[] | well, the 7941g has its own device ID |
| 17:44.40 | Qwell[] | so unless it's doing something blatantly wrong... |
| 17:45.07 | elriah | Qwell[]: Could it be my SEP<mac>.cnf.xml config? |
| 17:45.18 | Qwell[] | 3 != 115/309 |
| 17:45.25 | Qwell[] | no |
| 17:45.35 | Qwell[] | is it a g or g-ge? |
| 17:45.49 | elriah | I'm using the standard chan_skinny, are you using sccp or sccp2? |
| 17:45.50 | Qwell[] | not that it would make sense either way |
| 17:45.54 | Qwell[] | chan_skinny |
| 17:46.01 | Qwell[] | you *ARE* using 1.4, right? |
| 17:46.19 | elriah | Qwell[]: Not in production, 1.2.x |
| 17:46.26 | Qwell[] | then no, it won't work |
| 17:46.37 | elriah | Qwell[]: Will plugging in the updated chan_skinny work? |
| 17:46.41 | Qwell[] | no |
| 17:47.05 | elriah | Qwell[]: one last question, to make sure this battle is worth fighting, does the 1.4 chan_skinny support all the class 5 features? |
| 17:47.19 | Qwell[] | I don't know what that is |
| 17:47.21 | Qwell[] | but no |
| 17:47.33 | Juggie | elriah, i woudnt upgrade to 1.4 in production unless you intend to upgrade to 1.4svn |
| 17:47.50 | Juggie | if you wont do that then you should wait for 1.4.1 |
| 17:48.59 | [TK]D-Fender | Juggie: ETA on 1.4.1? |
| 17:49.11 | Qwell[] | [TK]D-Fender: ...when it's ready |
| 17:49.13 | Qwell[] | what are you, new? |
| 17:49.27 | Juggie | [TK]D-Fender, like i would know! |
| 17:49.36 | [TK]D-Fender | Qwell : I just haven't heard it in a while... was feeling nostalgic ;) |
| 17:49.36 | elriah | Qwell[]: i.e., transfer, dnd, all the basic stuff ... |
| 17:49.43 | Qwell[] | dnd works. :P |
| 17:50.04 | phearless | phearless: check the manual for the phone, sounds like the transfer is not completed. most phones require another button to be pressed (transfer?) to complete the transfer ---> yes the guy 408 has to press xfer another time |
| 17:50.06 | elriah | Juggie: Thanks. I take it 1.4 is buggy? |
| 17:50.14 | Juggie | 1.4.0 has its share of problems yes |
| 17:50.22 | Juggie | most of which have been fixed in the 1.4svn |
| 17:50.38 | elriah | Qwell[]: Here's what I'm trying to ask -> Are there any features on the phone that you're missing with Asterisk 1.4 and chan_skinny with your Cisco phones? |
| 17:50.40 | phearless | phearless: you want a 3-way call, not a transfer --> 3way is with 3 people on the same call ? |
| 17:50.44 | Qwell[] | yes |
| 17:50.45 | Qwell[] | many |
| 17:50.49 | phearless | phearless: Try adding "canreinvite=no" to each of your phones entries, and in [general] ---> ok I will try this ... |
| 17:50.54 | Qwell[] | redial, transfer, speeddials |
| 17:50.56 | elriah | Ok, that's out.. lol GOD HELP ME |
| 17:51.03 | Qwell[] | they're all "unimplemented" - they're quite buggy |
| 17:51.36 | Qwell[] | $6.50/phone |
| 17:51.40 | elriah | (@#$&*@$#*& |
| 17:51.49 | denon | $7, you ship |
| 17:51.49 | Juggie | good luck unloading cisco |
| 17:51.56 | Qwell[] | $8, I'll pay for shipping |
| 17:51.57 | kippi | thanks you guys that fixed it! |
| 17:52.10 | elriah | [TK]D-Fender is dying to chime in here, go ahead, let me have it. |
| 17:52.13 | denon | $8.01, but I get to use my amex for miles |
| 17:52.15 | [TK]D-Fender | kippi: You're welcome. |
| 17:52.25 | Qwell[] | $9, and I'll use denon's amex |
| 17:52.30 | [TK]D-Fender | elriah: Sorry... I capped out at 5$ ;) |
| 17:52.42 | denon | hmm |
| 17:52.53 | denon | Qwell gets stuck with crappy phones, I get the miles .. |
| 17:52.54 | file | $9.50! |
| 17:52.55 | denon | sounds ok |
| 17:53.02 | elriah | lol, well it was a good fight... Guess our remote users are going to be using Polycom's. |
| 17:53.03 | Qwell[] | denon: and a bill for the phones |
| 17:53.14 | denon | Qwell: that's fine, I'll just do a chargeback |
| 17:53.17 | Qwell[] | :P |
| 17:53.18 | Juggie | elriah, you shuld of known better. |
| 17:53.54 | [TK]D-Fender | Juggie: He was warned, but his boss said "we're going Cisco, TFB", and then was left to pray.... |
| 17:54.06 | elriah | Juggie: I did, I did (and I tried, I tried) but the decision makers bought a name here. |
| 17:54.22 | [TK]D-Fender | elriah: You should have kept a few extra virgins around to keep the Cisco God's happy... |
| 17:54.27 | elriah | Hell, I recommended against it just based on issues with TFTP, let alone anything with the phone itself. |
| 17:54.31 | Juggie | i coudnt even get my 7960's to do TFTP across a subnet |
| 17:55.51 | Juggie | eg tftp was on say 192.168.0.x and phones were 192.168.45.x |
| 17:55.51 | Juggie | totally routeable |
| 17:55.51 | Juggie | but the phone just flat out refused |
| 17:55.51 | [TK]D-Fender | Juggie: I like my phones being treated like MORE than trivial, thank you ;) |
| 17:55.51 | Qwell[] | user error :P |
| 17:55.53 | Juggie | Qwell, umm no, if i plugged them in on the same subnet they worked, or if i changed the tftp to the 45 subnet they worked. |
| 17:56.03 | Qwell[] | user error! :P |
| 17:56.08 | mafkees | indeed |
| 17:56.09 | elriah | Qwell[]: Don't you work for Digium? |
| 17:56.09 | Juggie | just not across subnets, the sip firmware is teh suck |
| 17:56.16 | Qwell[] | elriah: yeah, why? |
| 17:56.22 | Qwell[] | oh, sip, yeah, there's your problem |
| 17:56.35 | elriah | Just curious, thought I read that in the past. |
| 17:56.37 | Juggie | Qwell, at the time i did this, i think 1.2 wasnt even out |
| 17:56.43 | Juggie | so chan_skinny was still a dream. |
| 17:56.47 | Qwell[] | Juggie: I'm trolling :) |
| 17:56.52 | file | hehe... Qwell works... |
| 17:56.59 | Qwell[] | file: :( |
| 17:57.02 | Juggie | working's for suckers ;) |
| 17:57.08 | file | Qwell[]: I kid! <3 |
| 17:57.10 | Juggie | Qwell, did you hear those mp3's after? |
| 17:57.21 | Qwell[] | Juggie: yeah, Russell let me borrow his headphones :) |
| 17:57.48 | Juggie | maybe you shuld umm, get some! :P |
| 17:58.09 | Qwell[] | planning on getting speakers eventually |
| 17:58.11 | Juggie | :( i just found more code i have to fix for the daylight savings time change. |
| 17:58.21 | Juggie | this time a SQL stored procedure, fun fun. |
| 17:58.22 | Qwell[] | file: ^^ That's what I thought you meant |
| 17:58.38 | file | Qwell[]: now I'm confused :( |
| 17:58.38 | ManxPower | Juggie: My plan is to ignore the users for 2 weeks until the problem fixes itself. |
| 17:58.42 | Qwell[] | nevermind |
| 17:58.50 | Qwell[] | file: "time change thingie" |
| 17:58.57 | file | ah |
| 17:59.16 | file | 'fraid not |
| 17:59.19 | Juggie | ManxPower :) |
| 18:00.15 | Juggie | and of course, MSSQL doesnt provide anyway to get the timezone |
| 18:00.27 | Juggie | so that means i have to code all the dates in. |
| 18:01.29 | elriah | ManxPower: lol, ignoring users |
| 18:01.36 | Juggie | and i'll admit, this code kind of scares me. |
| 18:02.06 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:02.08 | phearless | [TK]D-Fender: your canreinvite trick worked ! |
| 18:02.17 | phearless | amazing, thanks |
| 18:02.19 | [TK]D-Fender | phearless: You're welcome too.... |
| 18:02.49 | [TK]D-Fender | NEXT!@!@@! (c) BKW |
| 18:03.09 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 18:04.15 | [TK]D-Fender | Juggie: Because some phones are smarter than others (few), and his is actually a local-lan issue. I was just betting its too dumb to know how to pass off RTP if a SASE was sent to it ;) |
| 18:04.51 | *** join/#asterisk friedrich| (n=friedric@e177252182.adsl.alicedsl.de) |
| 18:07.01 | dansmith | I get a lot of echo dialing into my company's conference call numbers.. echo that I don't get when I just call someone directly |
| 18:07.15 | dansmith | both with an SPA-3102 and with a generic X100P card |
| 18:07.39 | dansmith | I've got both tuned so I don't get echo with normal callers... is there anything else I can do? |
| 18:08.26 | [TK]D-Fender | dansmith: Yeah... try realizing that for a real PBX you'll need to invest in REAL hardware.... |
| 18:08.46 | dansmith | I do realize that :) |
| 18:08.53 | mafkees | gheh |
| 18:09.00 | mafkees | in .nl the DST stuff stays the same |
| 18:09.29 | mafkees | no changes needed |
| 18:10.19 | *** join/#asterisk aaqq (n=yytrttry@201-93-242-129.dsl.telesp.net.br) |
| 18:11.01 | Qwell[] | mafkees: heads up |
| 18:12.06 | aaqq | hi! |
| 18:12.10 | aaqq | i'm going to try to make a hack in chan_sip.c, to make it wait some seconds before sending a reinvite. has someone did something like this. any idea the place where to do it? |
| 18:12.14 | mafkees | cool |
| 18:12.23 | Qwell[] | mafkees: if you make a speeddial like the following, it'll magically become a hint |
| 18:12.39 | Qwell[] | speeddial => 1234@hints,Bob Fakeperson |
| 18:12.44 | mafkees | oeh |
| 18:12.51 | Qwell[] | then, of course, in the hints context, you add your hint like normal |
| 18:13.00 | mafkees | ok |
| 18:13.14 | Qwell[] | tested on a 7960, 12SP+, and 30VIP |
| 18:13.21 | Qwell[] | seemed to work pretty well |
| 18:13.26 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 18:15.23 | mafkees | can I mix ael and extensions.conf stuff ? |
| 18:15.54 | *** join/#asterisk fr33bi3 (n=fr33bi3@59.144.5.107) |
| 18:16.19 | mafkees | chan_skinny.c: In function ‘handle_message’: |
| 18:16.19 | mafkees | chan_skinny.c:3538: warning: ‘sd’ may be used uninitialized in this function |
| 18:16.30 | [TK]D-Fender | mafkees : Well... you can jump to contexts & macro's between either, but if you try implement the code-style of one in the other you're in for some upset... |
| 18:16.37 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 18:16.40 | Qwell[] | mafkees: You can pretty safely ignore any warnings in that code, heh |
| 18:17.13 | mafkees | [TK]D-Fender: I was referring to this [hints] thing |
| 18:17.23 | mafkees | I have converted my dialplan to ael2 |
| 18:17.36 | Qwell[] | mafkees: yeah, it should be fine |
| 18:17.38 | mafkees | but I still have to figure out how this hint stuff works in ael |
| 18:17.41 | [TK]D-Fender | mafkees: BRILLIANT! WGLWAT ;) |
| 18:17.43 | Qwell[] | just make it a different context |
| 18:17.57 | Qwell[] | codefreeze: do hints work in ael? |
| 18:17.59 | Qwell[] | ael2 that is |
| 18:18.07 | mafkees | it should work |
| 18:18.15 | mafkees | but I have no idea how ;) |
| 18:18.24 | Qwell[] | mafkees: you can see if it worked by doing a `core show hints` |
| 18:18.28 | [TK]D-Fender | AEL(1/2) .... what a waste... |
| 18:18.42 | Qwell[] | [TK]D-Fender: nah, ael makes dialplan a lot cleaner and more straight-forward |
| 18:18.43 | [TK]D-Fender | Gimme a new chan_sip! |
| 18:18.50 | [TK]D-Fender | brookshire: So.... hows SLA coming along? ;) |
| 18:19.07 | *** join/#asterisk gatuno (n=gatuno@145.red-82-158-215.user.auna.net) |
| 18:19.09 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 18:19.46 | [TK]D-Fender | Qwell : Straight to hell. Structured programming is often a way to combine everything on one line so that its even MORE unreadable because you get to apply your personal "style" to it. |
| 18:20.44 | Qwell[] | [TK]D-Fender: ask codefreeze for some ael2 examples. His dialplan is beautiful :P |
| 18:20.45 | [TK]D-Fender | Qwell : std EL works jsut great, is very linear and easy to interpret. And AEL adds nothing you can't do with std as it is (being that its parsed back to it anyways). |
| 18:20.51 | codefreeze | Qwell: hints work in AEL fine. I'm using them with success on the status lights on the SNOM360. |
| 18:20.56 | Qwell[] | cool |
| 18:21.16 | [TK]D-Fender | codefreeze: Waitasec... you wrote it didn't you? |
| 18:21.43 | codefreeze | [TK]D-Fender: I didn't invent it. I rewrote it for better parsing behavior. |
| 18:21.55 | [TK]D-Fender | GUILTY! |
| 18:22.07 | mafkees | <PROTECTED> |
| 18:22.10 | mafkees | hhmm |
| 18:22.11 | mafkees | so that works |
| 18:22.24 | mafkees | but no watchers |
| 18:22.30 | Qwell[] | mafkees: and if you do the speeddial right, the watchers should go up |
| 18:22.49 | mafkees | speeddial => 6002@hints,Livingroom |
| 18:22.56 | Qwell[] | yeah, that's right |
| 18:22.58 | mafkees | I have that in my skinny.conf |
| 18:23.12 | codefreeze | mafkees: The watchers have to ask for events... you might reboot your phones or whatever. |
| 18:23.13 | Qwell[] | under the line, in the device? |
| 18:23.16 | *** join/#asterisk fr33bi3 (n=fr33bi3@59.144.5.107) |
| 18:23.20 | mafkees | yeah |
| 18:23.39 | Qwell[] | mafkees: turn on skinny debug, see if it says adding hint |
| 18:24.10 | mafkees | resetting phone |
| 18:25.02 | mafkees | http://lunteren.vanbaak.info/tmp/skinny.conf |
| 18:25.06 | mafkees | that's my skinny.conf |
| 18:25.30 | Qwell[] | looks right |
| 18:25.36 | mafkees | yeah |
| 18:25.44 | Qwell[] | does "Livingroom" show up on your phone? |
| 18:26.47 | mafkees | <PROTECTED> |
| 18:26.47 | mafkees | Received Alarm Message: 22: Name=SEP0015626A4B99 Load=8.0(3.0) Last=Reset-Reset |
| 18:26.50 | mafkees | Device SEP0015626A4B99 is attempting to register |
| 18:26.53 | mafkees | <PROTECTED> |
| 18:26.56 | mafkees | <PROTECTED> |
| 18:27.00 | Qwell[] | -1...weird |
| 18:27.06 | Qwell[] | it didn't get the context properly |
| 18:27.06 | codefreeze | [TK]D-Fender: on AEL; you are technically correct; AEL does compile down into EL. There is nothing you can do in AEL that you can't do in EL. Just as in the fact that there is nothing that cannot be programmed in Assembler language. There is nothing in C, C++, Java, etc. that can't be done in assembly. So why use 'em? |
| 18:27.11 | mafkees | it does show up indeed |
| 18:27.16 | mafkees | hhmm |
| 18:27.39 | mafkees | maybe because I put the [hints] in extensions.conf and my phones context is in extensions.ael |
| 18:27.42 | mafkees | ? |
| 18:27.46 | Qwell[] | shouldn't matter |
| 18:27.49 | Qwell[] | it just isn't parsing it right |
| 18:28.05 | Qwell[] | sure the patch applied properly? |
| 18:28.09 | mafkees | yeah |
| 18:28.16 | mafkees | offset 25 lines |
| 18:28.17 | mafkees | :) |
| 18:28.30 | mafkees | I can revert to a clean svn version |
| 18:28.30 | codefreeze | mafkees: Uh... the hints have to be in the same context... |
| 18:28.37 | [TK]D-Fender | codefreeze: Lets just say I shudder to think the kind of dialplan you could have that would warrant such a structure. as of a certain point, AGI becomes far more useful. Oh, and can AGI send you around your dialplan created in AEL as easily? |
| 18:28.38 | ManxPower | Happy Fscking Lundi Gras |
| 18:29.06 | [TK]D-Fender | ManxPower: Ummm.... thats supposed to be Mardi Gras, no? |
| 18:29.19 | mafkees | codefreeze: so I have to add this hint in [internal] ? |
| 18:29.24 | mafkees | how to do that in ael ? |
| 18:29.26 | [TK]D-Fender | (although, yes it is technically Monday today) |
| 18:29.49 | Qwell[] | mafkees: I'm fairly certain chan_skinny isn't parsing the context properly |
| 18:30.37 | codefreeze | mafkees: It is not good to have two contexts of the same name; one in AEL, one in extensions. You may find that one wins and the other is ignored. |
| 18:31.12 | codefreeze | mafkees: Hopefully, some kind of error results. |
| 18:31.49 | *** join/#asterisk sjobeck (n=sjobeck@64.122.245.35) |
| 18:31.56 | mafkees | codefreeze: I have nothing in extensions.conf, only: [hints]\nexten => 6002,hint,Skinny/6002@livingroom |
| 18:32.05 | mafkees | all the other stuff is in extensions.ael |
| 18:32.07 | danp | i usually have a 'hints' context in extensions.conf and include that in my internal/whatever context in extensions.ael |
| 18:32.17 | mafkees | and there's no context named 'hints' in my extensions.ael |
| 18:32.29 | danp | perhaps that's what's being discussed :P |
| 18:32.31 | mafkees | Qwell[]: I did a svn -R revert * && make clean |
| 18:32.34 | mafkees | patched again |
| 18:32.38 | ManxPower | [TK]D-Fender: Lundi Gras is the day before Mardi Gras |
| 18:32.38 | mafkees | make is running now |
| 18:32.55 | codefreeze | mafkees: OK, that'll work. |
| 18:33.10 | [TK]D-Fender | ManxPower: Too damned many fat people ;) |
| 18:33.24 | tzanger | dammit |
| 18:33.30 | tzanger | I used a part that doesn't exist |
| 18:33.31 | ManxPower | [TK]D-Fender: I have come to hate Mardi Gras season |
| 18:33.42 | tzanger | I need a 1% resistor array, but 5% only exists |
| 18:33.46 | tzanger | close enough for the protos though :-) |
| 18:33.51 | mafkees | codefreeze: how can I do them hints in ael ? |
| 18:34.23 | [TK]D-Fender | ManxPower: Start blaring something by "Tragically Hip" on a PA system ;) |
| 18:34.36 | codefreeze | [TK]D-Fender: There are issues with AGI with interprocess comm. times, and process set up and tear down times. As in all programming tasks, the goal is to use the languages and tools most appropriate for the tasks at hand. |
| 18:34.52 | n|cotine | If I wanted to write a PHP script that would be called via crontab, that would execute commands on Asterisk (IE, a PHP script that parses the CDR DB, and issues a RemoveQueueMember() for extensions that missed a queue call), should I be looking at PHPAGI to pull this off? Or should I interface with Asterisk some other way? |
| 18:35.05 | [TK]D-Fender | codefreeze: True, but how well does AEL mesh with AGI, vs STD? |
| 18:35.12 | mafkees | Qwell[]: my patch was in the way |
| 18:35.20 | Qwell[] | kinda what I figured |
| 18:35.30 | mafkees | Ifrid*CLI> |
| 18:35.30 | mafkees | <PROTECTED> |
| 18:35.30 | mafkees | <PROTECTED> |
| 18:35.35 | Qwell[] | excellent |
| 18:35.43 | ManxPower | [TK]D-Fender: I'm 300 miles from New Orleans right now, but all my clients are in the New Orleans area |
| 18:35.55 | [TK]D-Fender | ManxPower: Like I said... BLARING ;) |
| 18:35.57 | mafkees | now if my 7905 would boot ;) |
| 18:35.58 | mafkees | lol |
| 18:36.24 | [TK]D-Fender | ManxPower: You ro P2P Microwave, so I presume you hav access to a high antenna array.... get mounting! ;) |
| 18:36.27 | codefreeze | mafkees: lets see: context somename { |
| 18:36.27 | [TK]D-Fender | do* |
| 18:36.27 | codefreeze | <PROTECTED> |
| 18:36.27 | codefreeze | } |
| 18:36.28 | ManxPower | [TK]D-Fender: but I basically get tomorrow off, and I doubt I will get many support calls the rest of today. |
| 18:37.27 | ManxPower | [TK]D-Fender: I don't do P2P microwave |
| 18:37.29 | codefreeze | mafkees: put your hint() in front of the extension name. |
| 18:37.34 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 18:37.38 | mafkees | ah |
| 18:37.42 | mafkees | hhmm |
| 18:37.52 | mafkees | _6XXX => { |
| 18:37.59 | mafkees | switch ($EXTEN).... |
| 18:38.00 | [TK]D-Fender | ManxPower: Could have sworn that you dealt with those..... |
| 18:38.03 | mafkees | something like that |
| 18:38.14 | [TK]D-Fender | ManxPower: "Point 2 Point,", not "peer 2 peer" |
| 18:38.15 | mafkees | how will it know what device to use for the hints ? |
| 18:38.28 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:38.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:38.32 | wunderkin | n|cotine, well, i believe you can automatically remove members from the queue if they do not answer a call... check agents.conf, autologoff |
| 18:38.39 | mafkees | that's where I'm lost |
| 18:38.46 | n|cotine | wunderkin: Does not work for agents added via AddQueueMember |
| 18:38.48 | [TK]D-Fender | ManxPower: Could be it was one of your customers, or maybe I'm just mixing people up entirely. last mention of which was MANY months ago |
| 18:39.10 | *** join/#asterisk bmd (n=bmd@64.50.19.206) |
| 18:39.16 | elriah | In 1.2.x, how well does SIP presence work with Polycom phones? |
| 18:39.25 | wunderkin | oh, thats only chan_agent, damn.. so just check the dialstatus after the dial for a noanswer, and log them off that way? |
| 18:39.32 | mafkees | Qwell[]: it works |
| 18:39.40 | Qwell[] | excellent |
| 18:39.46 | mafkees | the icon changes when I lift the handset of mi 7905 |
| 18:39.47 | Qwell[] | mafkees: try various states, like busy, hold, etc |
| 18:40.01 | ManxPower | [TK]D-Fender: nope. My main experience with point to point microwave was at Tulane Univ and that one went out every time it rained hard |
| 18:40.04 | mafkees | and it gets back to a normal 'phone' icon when it's onhook |
| 18:40.25 | Qwell[] | mafkees: the 7960 doesn't have lamps, but it does update those also |
| 18:40.38 | n|cotine | wunderkin: I am attempting to wedge this in without changing the dialplan, as the dialplan is dynamically generated. |
| 18:40.44 | n|cotine | Hence the hacky approach |
| 18:40.46 | wunderkin | sucky |
| 18:41.03 | wunderkin | add that as part of the autogenerated template? |
| 18:41.10 | mafkees | wow, the hold is weird |
| 18:41.26 | Qwell[] | mafkees: that block icon is sweet :D |
| 18:41.34 | mafkees | yeah |
| 18:41.35 | Qwell[] | It's like "NO!" |
| 18:41.42 | [TK]D-Fender | elriah: Works just great |
| 18:41.50 | mafkees | but my 7960 shows as if it has a call |
| 18:42.03 | Qwell[] | mafkees: yeah, no other way to do it on those phones. :( |
| 18:42.12 | Qwell[] | 7941/7961 can, but I don't have one, so I have no idea how it works |
| 18:42.19 | Qwell[] | but you can set the lamps to different colors |
| 18:42.20 | mafkees | ah |
| 18:42.24 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 18:42.33 | elriah | [TK]D-Fender: Thanks. Any tips or is it just as simple as specifying a hint to the peer? |
| 18:42.35 | mafkees | ok |
| 18:42.44 | mafkees | well, it's working :) |
| 18:43.06 | mafkees | there's no difference between Idle and Unavail tho |
| 18:43.06 | [TK]D-Fender | elriah: * side is as advertised. For the Polycom side you need only enable presence in your provisioing. |
| 18:43.13 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 18:43.22 | elriah | [TK]D-Fender: Ok, cool. Thanks again. |
| 18:43.37 | [TK]D-Fender | elriah: How many remote phones do you have? |
| 18:45.16 | elriah | Potentially 15-20. |
| 18:45.27 | [TK]D-Fender | ouch. |
| 18:45.37 | elriah | Is that a lot? |
| 18:45.47 | [TK]D-Fender | elriah: You going to be able to return those phones to your vendor? |
| 18:45.57 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 18:46.23 | elriah | Oh! No, the Cisco phones are for internal use. We're ok there, we were just trying to stick to one model. We have a bunch of Polycoms internal and external. |
| 18:46.59 | [TK]D-Fender | elriah: Ah, so you actually have enough of the right kind to shuffle around? |
| 18:47.40 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 18:48.17 | elriah | [TK]D-Fender: Here's the history: We were a Polycom shop but the "powers that be" hated the speaker phone. One of them read somewhere (who knows where) about the quality of the Cisco phones. So, when we ordered a bunch of new phones we ordered Ciscos. You know the rest - so we're back to being a Polycom shop wtih a bunch of Cisco phones, lol |
| 18:48.53 | Nivex | eBay the Ciscos :) |
| 18:48.56 | [TK]D-Fender | elriah: Ok. How much did you get your 7941's for BTW? |
| 18:48.58 | elriah | It kills me though they blame their IT shop for "not being able to get the Cisco phones to work properly for remote users" |
| 18:49.31 | elriah | [TK]D-Fender: Hrm.. I think they ordered them from voipsupply.com.. It was less than from a big distrubutor like Insight but I don't recall the exact amount. |
| 18:49.34 | [TK]D-Fender | elriah: And hating Polycom's speakerphone is almost isane... |
| 18:49.50 | [TK]D-Fender | elriah: Must have gotten a big inside deal. |
| 18:49.52 | ChicagoBud | does anyone have a good script for handling dialing out thorugh an (or multiple) itsp and handing all the various conditions like busy, congestion, etc? |
| 18:50.08 | elriah | [TK]D-Fender: Man, I just process 1's and 0's all day, I pick my battles. |
| 18:50.14 | elriah | .. carefully |
| 18:51.22 | *** join/#asterisk mega (n=mega@217.201.175.28) |
| 18:53.39 | mafkees | still I cannot see why it didnt work with my patch applied |
| 18:53.46 | mafkees | but that has to wait till tomorrow |
| 18:53.47 | mafkees | :) |
| 18:54.02 | mafkees | I'm breaking more then fixing today |
| 18:54.11 | mafkees | so it's time to do something else ;0 |
| 18:56.25 | mafkees | I'll be back tomorrow |
| 18:56.30 | mafkees | thnx for the patch Qwell[] |
| 18:57.38 | *** join/#asterisk soyuz (n=soyuz@50-126-26.leased.cust.tie.cl) |
| 18:58.06 | *** join/#asterisk apardo (n=apardo@87.217.145.219) |
| 19:00.42 | soyuz | good afternoon people. |
| 19:01.12 | elriah | Greets, soyuz. |
| 19:01.21 | soyuz | can anyone point me to some half-decent wholesale carriers? |
| 19:03.29 | elriah | Oh, dear god of Cisco firmware, please deliver to us a NAT capable SIP firmware, for we have invested in your infinite wisdom, for we have decision makers that make buying decisions based on their stock portfolio, for we cannot return to the likes of Polycom, in your name we pray. |
| 19:04.45 | ManxPower | elriah: Um, Asterisk's nat=yes removes the need for NAT aware firmware and in fact if you enable NAT features on the phone it will break Asterisk's NAT magic |
| 19:05.29 | ManxPower | elriah: so unless you have "special needs" like reinvites thru NAT, stop praying and start working on getting your config correct. |
| 19:05.53 | Hmmhesays | anyone in here ever do any cross compiling |
| 19:05.53 | Hmmhesays | ? |
| 19:06.00 | elriah | ManxPower: Eh? It don't work with 8.2.x.. I don't care who you are, the phone drops packets that don't come back on port 5060. |
| 19:06.14 | ManxPower | when we did Cisco testing all the SIP firmware versions we used worked just fine with NAT |
| 19:06.17 | elriah | ManxPower: 79x1's, mind you |
| 19:06.31 | elriah | ManxPower: Our 7940's seem to work well with 8.0.2 |
| 19:06.38 | ManxPower | elriah: do you have multiple phones behind the same nat without asterisk on the local lan? |
| 19:06.50 | elriah | Public Asterisk box, nat'ted 7941's... |
| 19:07.31 | ManxPower | elriah: If the packets go out from 5060 they should return to 5060, that is what NAT routers do. you have some other problem |
| 19:07.32 | elriah | ManxPower: Do you have any working 7941's or 7961's running NAT? If so, beer on me and what did you do? |
| 19:07.50 | *** part/#asterisk mega (n=mega@217.201.175.28) |
| 19:07.53 | ManxPower | elriah: gads, I would not use cisco, but because of their licensing policies. I still think you have a non-cisco issue. |
| 19:08.07 | aydiosmio | how would I send a call back to particular context from an AGI using Asterisk::AGI? |
| 19:08.32 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 19:08.43 | ManxPower | aydiosmio: Goto(context,extension,priroity) does not work? |
| 19:09.05 | ManxPower | also I seem to recall if you SET the context, extnesion, proiroty, etc it will go there when the AGI exits |
| 19:09.21 | aydiosmio | ManxPower: nah, on exit; my calls are dropped |
| 19:09.26 | aydiosmio | need them to go back to my IVR |
| 19:09.43 | aydiosmio | I'll just us ea goto |
| 19:09.54 | elriah | ManxPower: I've tried everything -- it's not a NAT issue as I have other phones working, it's just the 7941's (25 of them). When a sip registration packet is sent to asterisk, it returns the response on a high port (say 39000) then NAT throws that back to the phone. This is also how the Polycoms work and 7940's work. But on the 7941's, the packets are just ignored and asterisk resends until it gives up. |
| 19:10.08 | klasstek | http://www.pastebin.ca/363851 |
| 19:10.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-6-99.red.bezeqint.net) |
| 19:11.02 | elriah | ManxPower: I've also tried various combinations of Cisco firmware. Unfortunately, the 79x1's have a "new generation" of firmware that isn't cross compatible with all the docs on the 79x0's. |
| 19:11.06 | ManxPower | elriah: The response from asterisk should go out the same port it came in on. |
| 19:11.09 | ManxPower | is that not the case? |
| 19:11.12 | *** join/#asterisk s1gny|wrk (n=s1gny@p549173B3.dip.t-dialin.net) |
| 19:11.20 | *** part/#asterisk s1gny|wrk (n=s1gny@p549173B3.dip.t-dialin.net) |
| 19:13.21 | ManxPower | Phone sends registration packet from local port 5060 to remote port 5060, nat router translates the source port to some high port like 16893, keeps the destination port and sends it to asterisk on the original destination port of 5060, asterisk responds from port 5060 to the translated port (16893), the nat router then translates the destination port of the response packet (16983) back to the source port of 5060 and passes it on to the |
| 19:13.33 | ManxPower | This is BASIC nat theory. |
| 19:14.05 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-120-7.dsl.tulsok.swbell.net) |
| 19:14.06 | elriah | ManxPower: Right, totally agree and understand that. Something about the packet is different that makes the 79x1's drop them. Again, other phones on the same LAN work great. |
| 19:14.12 | ManxPower | Asterisk's NAT=yes, localnet, and externip should then handle all the weird stuff INSIDE the DATA part of the SIP packet to have the correct port information. |
| 19:14.15 | elriah | the 'return' packet |
| 19:14.49 | elriah | Asterisk in this case is on a public IP, so no issue there. |
| 19:14.49 | ManxPower | elriah: I assume you removed all nat features from the phones. |
| 19:14.56 | ManxPower | and disabled any SIP nat features of your NAT router? |
| 19:15.00 | elriah | ManxPower: There is a true/false setting, I've tried it with both and there's also an externip type setting that I've tried. |
| 19:15.14 | elriah | tried nat=yes,nat=no,nat=DEARGODPLEASEWORK and none seem to help on the asterisk side... |
| 19:15.21 | ManxPower | elriah: setting NAT stuff on the phone will pretty much guarntee nat failure |
| 19:15.23 | elriah | No sip natting features. |
| 19:15.46 | ManxPower | elriah: ask on the mailing list. |
| 19:15.51 | elriah | Ok, so nat=no in asterisk, nat disabled on phones, yep tried that. |
| 19:15.56 | ManxPower | I'll bet you get zillions of resonses saying it works for me. |
| 19:16.12 | elriah | ManxPower: Will do, I haven't tried that yet. Which list? |
| 19:16.18 | ManxPower | elriah: you want nat=yes in the [device] section for each device. that is the only nat setting you want anywhere. |
| 19:16.22 | ManxPower | asterisk-users |
| 19:16.25 | elriah | Thanks. |
| 19:16.26 | elriah | :) |
| 19:18.38 | JoNate | BAH! |
| 19:19.38 | aydiosmio | == Spawn extension (custom-accountbalance, s, 3) exited non-zero on ... |
| 19:19.53 | aydiosmio | anyone know why this is non-zero and how it affects call flow? |
| 19:20.08 | aydiosmio | my AGI exits returning 0 |
| 19:25.55 | [TK]D-Fender | Ok, I'm feeling dumb again, and have 2 book open in front of me, and the MySQL reg guide CHM. Dangit, whats the command to actually choose a database to use? :) |
| 19:26.35 | Qwell[] | use database? |
| 19:26.46 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 19:27.04 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 19:27.38 | aydiosmio | he feels dumber now. |
| 19:28.31 | JoNate | ok...so remembering that I'm a complete noob with asterisk...any ideas as to why my MoH won't start? |
| 19:29.13 | elriah | JoNate: Which version of asterisk? |
| 19:29.21 | JoNate | 1.2 |
| 19:30.10 | JoNate | I was trying to use just the default...should I use the native? |
| 19:30.31 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 19:30.48 | elriah | JoNate: I've had best success with native... |
| 19:31.04 | elriah | JoNate: Post your config in a pastebin and I'll give you a hand... |
| 19:31.07 | JoNate | ok...let me try that |
| 19:31.53 | *** join/#asterisk giasai68 (i=giasai68@ip-240-139.sn2.eutelia.it) |
| 19:31.56 | giasai68 | hello |
| 19:32.02 | giasai68 | I have 2 PRI |
| 19:32.08 | elriah | Greets, giasai68! I have none... |
| 19:32.16 | JoNate | elriah: I think I might have found it |
| 19:32.25 | giasai68 | when I connect both appear this warning: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too |
| 19:32.27 | JoNate | elriah: When I fail miserably, I'll let you know! |
| 19:32.31 | elriah | JoNate: Cool. |
| 19:32.36 | giasai68 | any help to fix it? |
| 19:32.52 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 19:33.20 | [TK]D-Fender | aydiosmio: No.... not dumber... I know it would be a 1-2 keyword answer, just that none of my references are layed out with any sanity.... |
| 19:33.30 | [TK]D-Fender | Qwell[]: thanks. |
| 19:37.09 | giasai68 | when I connect both appear this warning: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too |
| 19:38.22 | CunningPike | giasai68: Change your PRI to pri_net |
| 19:41.14 | Kritter | s/ 2 |
| 19:41.20 | *** join/#asterisk apardo (n=apardo@87.217.145.219) |
| 19:44.16 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:45.13 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 19:45.29 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 19:51.29 | giasai68 | CunninPike: done... but same warning |
| 19:53.25 | [TK]D-Fender | giasai68: And did you completely restart *? |
| 19:55.12 | klasstek | can anyone test this patch for meetme please. |
| 19:55.18 | klasstek | http://bugs.digium.com/view.php?id=9106 |
| 19:55.31 | giasai68 | yes, but I have noticed layer 2 not syncronized |
| 19:56.32 | ManxPower | giasai68: put all your non-comment lines on pastebin.ca for zapata.conf |
| 19:56.36 | ChicagoBud | does anyone know if Playtones works in 1.4? |
| 19:56.43 | [TK]D-Fender | giasai68: pastebin your zapata.conf and zaptel.conf |
| 19:56.52 | ManxPower | ChicagoBud: it should, but you need /etc/asterisk/indications.conf |
| 19:57.48 | ChicagoBud | ManxPower, I seem to have that. I am doing a "exten => s,1,Playtones(congestion) |
| 19:57.49 | ChicagoBud | " but it just drops to the next priority |
| 19:58.09 | ChicagoBud | I see it on the console |
| 19:58.17 | ManxPower | ChicagoBud: yes, it does that. it makes the line play the tone and then continues the dialplan, that is the way it works |
| 19:58.29 | ManxPower | if you don't want it to do that then use the Congestion() app |
| 19:58.52 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 19:59.41 | ChicagoBud | ManxPower, That's what started this. I want to use Busy() in one case and Congestion() in another but they both seem to just play a standard busy |
| 19:59.54 | ManxPower | Or you could add a Wait(30) (or whatever) then a Hangup. |
| 19:59.55 | ChicagoBud | ManxPower, I expect the Congetion(0 to play a fast busy |
| 20:00.00 | ManxPower | ChicagoBud: on what devices? |
| 20:00.15 | ChicagoBud | hard sip phone - mitel in this case |
| 20:00.22 | *** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net) |
| 20:00.31 | ManxPower | ChicagoBud: of the call has not been answered then congestion and busy just send a message to the device to play the congestion or busy tone as locally defined on the device |
| 20:00.43 | ManxPower | they may do so even if the call HAS been answered |
| 20:00.59 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 20:00.59 | ChicagoBud | ManxPower, is it the device that is "playing" the busy? |
| 20:01.04 | funkmaster | hi guys i was wondering, how many extensions can asterisk handle, is there a limit? |
| 20:01.13 | ManxPower | ChicagoBud: most of the time it is the SIP device playing the tone |
| 20:01.15 | ChicagoBud | ManxPower, when I call Congestion() |
| 20:01.34 | ManxPower | funkmaster: I vaguely recall 65,000 might be a hardlimit |
| 20:01.38 | ChicagoBud | ManxPower, OK. |
| 20:01.54 | *** join/#asterisk khan_gee (i=geek@202.83.165.127) |
| 20:01.54 | ManxPower | ChicagoBud: sip debug would tell you |
| 20:01.56 | elriah | Damn, that's all? pfft, asterisk sucks. |
| 20:02.04 | ManxPower | but don't expect me to sift thru all the debug info for you. |
| 20:02.16 | funkmaster | ManxPower: thx |
| 20:02.56 | ManxPower | funkmaster: the extensions list might be a linked list with no hard limit, but I would not put more then 10,000 extensons on 1 box |
| 20:03.16 | giasai68 | here is zapata.conf: http://rafb.net/p/EweA4j14.html |
| 20:04.07 | ManxPower | giasai68: we TOLD YOU TO SET IT TO PRI_NET! |
| 20:04.27 | ManxPower | And yet it is still signalling=pri_cpe |
| 20:05.02 | giasai68 | here zaptel.conf: http://rafb.net/p/bhvaph43.html |
| 20:05.03 | ManxPower | giasai68: start out by cleaning up your config file. |
| 20:07.16 | tzanger | ManxPower: you know of any magic to get physical fax machines to play nice through a rhino channel bank? |
| 20:07.31 | tzanger | I have them working just fine though my adit600 and had it working though my old access bank 1 |
| 20:07.32 | giasai68 | I have did |
| 20:07.54 | ManxPower | tzanger: my solution is to not run fax thru anything. |
| 20:08.05 | Nivex | eeeeew fax |
| 20:08.10 | giasai68 | now i have 2 different gateway connected to my pri card 1 work fine but other not waork |
| 20:08.12 | elriah | tzanger: I agree with ManxPower on that one. POTS all the way for fax. |
| 20:08.16 | tzanger | ManxPower: gee thanks :-) |
| 20:08.19 | Nivex | emailing PDFs for great justice |
| 20:08.21 | tzanger | elriah: I do use pots :-) |
| 20:08.31 | tzanger | pri - TE407 - adit600 - fax |
| 20:08.36 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:08.42 | giasai68 | i try to sett cpe mode on 2th pri but i have the same problem on layer 2 on gateway's pri |
| 20:08.48 | ManxPower | giasai68: See line 10 and line 37 |
| 20:09.00 | ManxPower | set them to pri_net and see what happens |
| 20:09.13 | giasai68 | I see |
| 20:09.23 | giasai68 | ok |
| 20:09.44 | ManxPower | giasai68: if you now get a message about both sides thinking they are PRI_NET then the remote side is looping the line so anything asteris sends out the line is being echoed back to asterisk |
| 20:09.55 | ManxPower | remove the loop |
| 20:12.05 | giasai68 | if i connect the 2th gateway to a cisco machine the gateway's pri work fine |
| 20:12.08 | [TK]D-Fender | Nivex: All Your PDF Are Belong To Us |
| 20:12.40 | giasai68 | tthis think is strange because on a gateay work fine with the same settings |
| 20:13.29 | *** join/#asterisk ping2921 (n=marc3234@206-248-160-34.dsl.teksavvy.com) |
| 20:14.59 | giasai68 | thanks |
| 20:15.00 | giasai68 | bye |
| 20:17.51 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 20:18.12 | khan_gee | Hi all |
| 20:18.38 | khan_gee | is this possible to use Asterisk as Soft switch to interconnect h323 and sip both |
| 20:19.48 | [TK]D-Fender | khan_gee: * is a B@BUA (back-to-back-user-agent). It is not a proxy or soft-switch, but you can take calls in on each kind of interface, and by your design place calls back out using another protocol and bridge them. |
| 20:20.03 | ManxPower | khan_gee: yes, but I believe that doing so will cause a hole in time-space and destroy us all |
| 20:20.45 | khan_gee | hmmm ok |
| 20:20.52 | khan_gee | thanks |
| 20:21.29 | ManxPower | khan_gee: H323 is very difficult to get working with Asterisk |
| 20:21.41 | ChicagoBud | ManxPower, you were right. it is the phone itself that is generating the tones |
| 20:21.56 | khan_gee | well for h323 i can use Mera as brige |
| 20:22.01 | khan_gee | its not an issue |
| 20:22.23 | ManxPower | ChicagoBud: that is the way SIP works. |
| 20:22.32 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:22.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:22.35 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-5aca69face17e603) |
| 20:22.47 | ManxPower | ChicagoBud: you may be able to force inband tones by issuing an answer first, try it and see. |
| 20:24.59 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 20:28.24 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 20:29.13 | elriah | Is there any way to further 'buffer' to attempt to clear up forward calls? i.e., telco -> sip trunk provider -> asterisk -> sip trunk provider -> telco? |
| 20:29.32 | jserve | *hmms* Perhaps someone help me here faster. I have a problem with implementing the G.729 Codec. When I dial out I got as response http://pastebin.ca/363938. When I use MusicOnHold I not get the problem, so I assume it's a indication problem. Any ideas how I could fix it? |
| 20:29.32 | ManxPower | elriah: buffer? |
| 20:29.40 | Bobthehunter | is TDM400 4FOX can work with analog BELL TELCO lines ? |
| 20:29.45 | Bobthehunter | and does it have echo |
| 20:30.04 | ManxPower | Bobthehunter: all analog lines have echo, usually cause by the FAR end analog lines |
| 20:30.14 | ManxPower | yes, that card works with Bell. |
| 20:30.17 | Bobthehunter | echo canel i meant |
| 20:30.28 | elriah | Bobthehunter: I have one for sale with 4 FXO ports if you want it, $225.00+shipping. |
| 20:30.34 | ManxPower | Bobthehunter: yes |
| 20:30.57 | elriah | ManxPower: I was just looking for a way to clean up the call due to apparent latency. |
| 20:30.57 | |ryan| | Anyone here set up incoming IAX connections with callwithus? |
| 20:31.25 | ManxPower | jserve: what does " show g729" say? |
| 20:31.25 | Bobthehunter | hmm got it at less ;) |
| 20:31.28 | Bobthehunter | new ? |
| 20:31.48 | elriah | Bobthehunter: You got one with 4 FXO cards for less? Where? |
| 20:31.59 | ManxPower | elriah: remember Asterisk's SIP implimentation does not have a jitter buffer. (has this changed in 1.4? I don't think so) |
| 20:32.05 | jserve | ManxPower: 0/0 encoders/decoders of 1 licensed channels are currently in use |
| 20:32.17 | Bobthehunter | MTL |
| 20:32.19 | jserve | ManxPower: Do I need 2 licenses for that? |
| 20:32.20 | elriah | ManxPower: Ahh. Oh well, maybe g729 will help here. |
| 20:32.29 | ManxPower | elriah: it won't |
| 20:32.38 | elriah | Compressed calls = smaller packets? |
| 20:32.39 | ManxPower | jserve: I don't know. I never buy less than 10 licenses |
| 20:32.59 | ManxPower | elriah: jitter is the VARIENCE in the transit time for the packet |
| 20:33.03 | *** join/#asterisk backblue (n=moo@89-180-132-145.net.novis.pt) |
| 20:33.03 | jserve | *hmms* I just thought to test it out with one license... |
| 20:33.08 | ManxPower | not the acutal transit time |
| 20:33.20 | elriah | Right, but if the packets are smaller, they should get there quicker? no? |
| 20:33.27 | elriah | Oh, got ya. |
| 20:33.30 | elriah | I see.. |
| 20:34.27 | jserve | ManxPower: Because, when I use dial() with the MoH stuff it seems to work. Just when I use dial() without the MoH feature (so that I can hear the ringtones from my carrier) I get the problems. |
| 20:35.01 | putzz | 0/clear |
| 20:35.02 | ManxPower | jserve: what specific problem? |
| 20:35.04 | putzz | oops |
| 20:35.34 | *** part/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 20:35.38 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 20:37.28 | Bobthehunter | ok so no echo canel on the tdm 400 ? whats the best for me for echo can 4 FXO lines hardware |
| 20:37.33 | jserve | ManxPower: My problem is that I get the response http://pastebin.ca/363938 when I just do a dial() without the MoH |
| 20:37.41 | ManxPower | Bobthehunter: hardware or software? |
| 20:37.53 | Bobthehunter | hardware |
| 20:37.54 | Bobthehunter | lol |
| 20:38.06 | ManxPower | jserve: yeah, looks like you are out of licenses |
| 20:38.14 | ManxPower | Bobthehunter: no idea. |
| 20:38.27 | jserve | ManxPower: Ok, I just want to have verified it. |
| 20:38.46 | jserve | Not that this is something in the configuration. But so I can buy more licenses and see what happens :) |
| 20:38.50 | Bobthehunter | hmmm ok |
| 20:38.51 | Bobthehunter | lol |
| 20:40.19 | Bobthehunter | Product Code: TC400B Price: $1675.00 lol |
| 20:40.21 | elriah | ManxPower: What would be the best codec in a situation where latency is apparently an issue? |
| 20:40.54 | ManxPower | elriah: not use SIP in 1.2 or upgrade to 1.4 |
| 20:41.18 | ManxPower | Bobthehunter: Hardware EC will add about $1,000 to the price of a card |
| 20:41.22 | elriah | Ahh, lol |
| 20:42.01 | ManxPower | We don't use SIP except for QoS'd links or LAN |
| 20:42.06 | ManxPower | for exactly this reason |
| 20:42.13 | *** join/#asterisk GTX (i=charlie@pdpc/supporter/monthlybronze/GTX) |
| 20:42.22 | GTX | Is there anyway to encrypt convo's going over asterisk? |
| 20:42.30 | ManxPower | GTX: Yes, a VPN |
| 20:42.46 | elriah | Would IAX have better quality in this condition with 1.2.x? |
| 20:42.59 | GTX | What about the User to the server ManxPower ? |
| 20:43.48 | ManxPower | GTX: Use a VPN |
| 20:43.53 | GTX | lol |
| 20:44.05 | ManxPower | elriah: in theory tes. |
| 20:44.22 | ManxPower | GTX: do you know of any IP hardphones with encryption support? |
| 20:44.32 | GTX | ;p |
| 20:44.46 | Kritter | he's saying there is no encryption support, use a VPN. |
| 20:44.56 | GTX | Indee |
| 20:44.58 | GTX | Indeed* Thanks |
| 20:44.58 | elriah | Hrm.. I wonder if there is a way to transfer the call back to my sip trunk provider therby eliminating our asterisk server once the call is transferred/bridged... |
| 20:45.22 | Kritter | so a hardphone with encryption support is really a phone with a VPN, and that is way outside scope of what a phone or asterisk at it's core should do. |
| 20:46.04 | Kritter | There is nothing to stop you from using a * to terminate an OSS VPN tunnel, but I'm not sure I want to see that under load. |
| 20:46.14 | ManxPower | elriah: is there NAT involved? |
| 20:46.34 | elriah | ManxPower: Nope.. Got an idea? |
| 20:46.43 | ManxPower | elriah: it is called canreinvite=yes |
| 20:47.08 | [TK]D-Fender | Bobthehunter: http://www.canadianvoipstore.com/product_info.php?products_id=1339 |
| 20:47.33 | elriah | On my side of the sip.conf definition? Ahh.. let me look this up.. I think I smell what you're stepping in! (thanks) |
| 20:47.35 | ManxPower | SIGNALING will still go thru asterisk but the AUDIO (RTP) should not unles you are doingsomething on Asterisk that requires asterisk to stay in the audio path like t/T/w/W on the Dial line, or many other dial opts or transcoding or protocol translation |
| 20:48.08 | Bobthehunter | so http://www.canadianvoipstore.com/product_info.php?products_id=1339 is for waht ? |
| 20:49.02 | [TK]D-Fender | Bobthehunter: For your request for options and pricing on a 4 port FXO card w/ HWEC |
| 20:49.45 | [TK]D-Fender | Bobthehunter: so $1600 is BS |
| 20:49.45 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 20:49.45 | Bobthehunter | no |
| 20:49.50 | Bobthehunter | 1600 wast the new pci card from digium to transcode only lol |
| 20:50.26 | [TK]D-Fender | Bobthehunter: Well either way thats the price for the kind of card you were looking for. |
| 20:50.59 | ManxPower | Bobthehunter: see line 2 of "technical specs" on that page |
| 20:53.54 | Bobthehunter | calling sangoma |
| 20:54.02 | Bobthehunter | ;) just dont know diff in A200X and TDM |
| 20:54.11 | elriah | ManxPower: Ok, I understand the canreinvite setting. Thanks for that, if It works it will certainly help as it will take about 15 hops out of the RTP stream. My questions are: How will I know if the call was successfully 'reinvited' back to the upstream peer? And does the upstream peer have to have a sip.conf setting change? (it's asterisk as well) |
| 20:54.21 | tzanger | hmm, is Digiums TDM800 not available anywhere yet? |
| 20:54.36 | ManxPower | elriah: you would need to use a packet sniffer on the asterisk box |
| 20:54.41 | [TK]D-Fender | TC400B = 1675$ (digium.com) = 96 channels = 17.45/channel. Standard codec = 10$/channel.... hmmmm |
| 20:54.50 | tzanger | I'm just curious as to whether it's possible to have 6FXO/2FXS on that card or not, and if hardware echo cancelleation is available |
| 20:54.59 | elriah | ManxPower: So the console/logs will still report the call as "bridged" ? |
| 20:55.03 | [TK]D-Fender | tzanger: Sure |
| 20:55.10 | tzanger | [TK]D-Fender: yes, but what CPU :-) |
| 20:55.10 | ManxPower | elriah: I believe so |
| 20:55.11 | file | [TK]D-Fender: you get 723 as well too |
| 20:55.20 | elriah | ManxPower: Even if it's been 'reinvited' upstream, i.e., the RTP terminates at an upstream peer? |
| 20:55.21 | file | [TK]D-Fender: and can handle more per system |
| 20:55.24 | tzanger | [TK]D-Fender: sure, as in the ports are in groups of 2, not 4 like the TDM2400? |
| 20:55.31 | [TK]D-Fender | file: Yeah... if you needed it, I guess it'd factor in. |
| 20:55.36 | elriah | ManxPower: thanks again. |
| 20:56.06 | elriah | Ahh.. I see why NAT breaks canreinvite... |
| 20:56.16 | [TK]D-Fender | tzanger: Yes, A200 series is 2 ports/module, intermixable, and expandable w/ Remora backplanes |
| 20:56.32 | tzanger | [TK]D-Fender: that's A200 (sangoma), not TDM800 (digium) |
| 20:56.46 | [TK]D-Fender | tzanger: And what IS the realistic transcoding power of say a 3.0ghz P4? |
| 20:56.48 | tzanger | I know the A200 is groups of 2 and has octasic option |
| 20:57.03 | [TK]D-Fender | tzanger: Oh ... Digiums? not a clue :) |
| 20:57.05 | Bobthehunter | yeah hehhe jsut got reminded that |
| 20:57.44 | Bobthehunter | k thanks |
| 20:59.00 | *** join/#asterisk J4k3 (i=jsuter@16.sub-70-216-114.myvzw.com) |
| 20:59.15 | elriah | ManxPower: Wow! That makes a HUGE difference in call quality (canreinvite=yes), thanks a bunch. Where do I send the beer? |
| 21:00.36 | ManxPower | elriah: cash to paypal eric@fnords.org is a good place. Imagine how much money my advice has saved you, then send 1/2 of that amount to the paypal address |
| 21:02.04 | elriah | How about a Cisco 7941 instead? I have a pile of them stacking up beside my desk, lol |
| 21:02.23 | ManxPower | uh, no thanks |
| 21:03.31 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:04.28 | elriah | ManxPower: What is an Fnord? |
| 21:04.59 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 21:05.02 | ManxPower | elriah: if I told you I would have to kill you, and nobody wants that. |
| 21:05.07 | *** join/#asterisk sasch (n=sasch@host207-68.pool8251.interbusiness.it) |
| 21:05.18 | elriah | ManxPower: "nobody wants that" <- You'd be surprised... |
| 21:05.47 | ManxPower | Ok, "most people don't want that" |
| 21:07.33 | jserve | ManxPower: Just received my other licenses... And I can acknowledge the problem. |
| 21:07.55 | jserve | My problem stated happens when the ringtones of my carrier can't be converted back to G.729 because of missing licenses. |
| 21:08.10 | *** join/#asterisk JackEStorm (n=no@ip68-225-72-125.no.no.cox.net) |
| 21:08.22 | jserve | (I assume, one license is used for the conversation from the software -> carrier, and the other from carrier -> softphone) |
| 21:09.32 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:10.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:10.24 | *** join/#asterisk J4k3- (i=jsuter@102.sub-70-216-231.myvzw.com) |
| 21:15.07 | [TK]D-Fender | tzanger: So... about that CPU power requirement question.... ? |
| 21:15.28 | tzanger | [TK]D-Fender: no idea, I've never benchmarked it |
| 21:15.53 | [TK]D-Fender | ManxPower: Got any relevent #'s on transcoding power ? |
| 21:15.57 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 21:16.27 | ManxPower | [TK]D-Fender: show translation recalc 30 |
| 21:16.50 | ManxPower | [TK]D-Fender: I've never really worried about it. My servers are not heavily loaded. |
| 21:17.08 | ManxPower | and my servers don't do much transcoding anyway |
| 21:17.12 | Bobthehunter | http://www.canadianvoipstore.com/product_info.php?products_id=1339 |
| 21:17.15 | Bobthehunter | darn |
| 21:18.00 | Bobthehunter | aint no more a asterisk-biz channel ? |
| 21:18.04 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 21:18.25 | elriah | Bobthehunter: $225, Digium TDM400P w/4FXO ports... |
| 21:19.24 | Druken | tzanger: did ya see my thank you this morning? got it working... |
| 21:20.47 | Bobthehunter | lol |
| 21:20.58 | Bobthehunter | need GSM gateway to sip etc |
| 21:25.24 | tzanger | Druken: I did, no worries and you're welcome |
| 21:27.33 | *** join/#asterisk J4k3 (i=jsuter@100.sub-70-216-154.myvzw.com) |
| 21:27.40 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 21:28.04 | Dovid | can anyone help me with this ? |
| 21:28.04 | Dovid | http://pastebin.ca/364015 |
| 21:28.35 | Dovid | what dependancy am i missing ? |
| 21:29.37 | jserve | *hmms* |
| 21:29.49 | jserve | linux/config.h? That looks like Kernel Sources= |
| 21:29.50 | jserve | ? |
| 21:30.10 | Dovid | hmm |
| 21:30.12 | Dovid | i installed me |
| 21:30.45 | elriah | Do you have a link called 'linux' in your /usr/src dir? |
| 21:31.02 | Dovid | checking |
| 21:31.03 | elriah | What distro? |
| 21:31.07 | Dovid | FC5 |
| 21:31.22 | *** join/#asterisk Witwolf (n=carel@c2-354-1.eno.dial.mweb.co.za) |
| 21:31.33 | Dovid | no folder linux in usr/src |
| 21:31.44 | elriah | Assuming you have the kernel-headers or kernel-source installed, you may still need a link called /usr/src/linux. Dunno about FC, but I used to need it in debian. |
| 21:32.03 | elriah | ln -s /usr/src/linux /usr/src/whateveryourdiryourheadersarein |
| 21:32.25 | elriah | Would be a semi-educated guess. |
| 21:32.26 | Witwolf | Hi, what is the rules and what questions may be asked. Have never been here before! |
| 21:32.44 | elriah | Witwolf: These guys are really helpful, just ask away. |
| 21:32.45 | thekidrio | i think you can just ask away :) |
| 21:32.54 | Dovid | have a look at this |
| 21:32.54 | Dovid | http://pastebin.ca/364027 |
| 21:33.10 | Dovid | elriah: what should i point it to ? |
| 21:33.14 | Dovid | what folder is it looking for ? |
| 21:33.23 | elriah | What's in /usr/src now? |
| 21:33.28 | Dovid | checking |
| 21:33.40 | Dovid | kernels |
| 21:33.46 | elriah | and under kernels? |
| 21:34.07 | Dovid | 2.6.17-1.2174_FC5-i686 2.6.17-1.2174_FC5smp-i686 2.6.19-1.2288.fc5-smp-i686 |
| 21:34.07 | Dovid | 2.6.17-1.2174_FC5-smp-i686 2.6.19-1.2288.fc5-i686 2.6.19-1.2288.fc5smp-i686 |
| 21:34.27 | elriah | Ok, now of those kernels, which one are you actually running? Sometimes a uname -r will show you. |
| 21:34.47 | Dovid | 2.6.19-1.2288.fc5smp #1 SMP |
| 21:35.09 | Bobthehunter | ~pb |
| 21:35.16 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:35.16 | Bobthehunter | !pb |
| 21:35.19 | Witwolf | Hi, just finished my trixbox setup and working perfectly. I just need a bit of info which I am struggling to find one the internet. Maybe my own stupidity? I want to know if it is possible to answer a phone call on another extension, even if my phone is not ringing? |
| 21:35.25 | elriah | Ok, link to that one and see if it helps. |
| 21:35.26 | Dovid | ln -s /usr/src/linux /usr/src/kernels/2.6.19-1.2288.fc5smp ? |
| 21:35.30 | Dovid | ? |
| 21:35.33 | elriah | Yep, try it. |
| 21:36.13 | elriah | Make sure you get your casing right, if it's looking for "Linux" make sure you call the link "Linux", etc. |
| 21:36.14 | Dovid | hmph |
| 21:36.37 | elriah | 'make clean' in between failed attempts, fyi |
| 21:37.14 | elriah | Is it compiling? |
| 21:37.26 | [TK]D-Fender | Witwolf: ... |
| 21:37.30 | [TK]D-Fender | ~trixbox |
| 21:37.31 | jbot | methinks trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 21:38.16 | Dovid | not working |
| 21:38.19 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 21:38.30 | Witwolf | Ok, do you know if there is such a feature on Asterisk, and what its called D-Fender? |
| 21:38.32 | elriah | Ok, post the results again ... |
| 21:38.40 | Dovid | ln is bad |
| 21:38.55 | Kritter | suspect you want to call steal |
| 21:38.57 | elriah | You're getting a message that says 'ln is bad' |
| 21:38.59 | elriah | ? |
| 21:39.05 | Witwolf | I am planning to move to just Asterisk on our debian server. |
| 21:39.23 | elriah | Oops, it's ln -s <target> <link> |
| 21:39.27 | elriah | I think we got it backwards. |
| 21:39.39 | elriah | So ln -s /usr/src/kernels/whatever /usr/src/linux |
| 21:40.12 | Dovid | ok |
| 21:40.14 | Dovid | will try agian |
| 21:40.57 | Bobthehunter | ever heard of a pcmcia fxo card ? |
| 21:41.07 | [TK]D-Fender | Witwolf: * can do just about anything. FreePBX's dialplan lets you do a FEW cook-ie-cutter things. |
| 21:41.07 | elriah | Bobthehunter: A modem? |
| 21:41.19 | Bobthehunter | lol |
| 21:41.24 | Bobthehunter | well comaptible to asterisk |
| 21:41.48 | Dovid | still not working |
| 21:41.51 | [TK]D-Fender | Witwolf: Yes its possible if you do it yourself. |
| 21:41.53 | Witwolf | So if I go and look in the dailplan.conf I would find it or even write it myself? |
| 21:41.59 | elriah | Ok, post the results again... |
| 21:42.08 | Kritter | did you look at the forum link I gave you Wit? |
| 21:42.36 | [TK]D-Fender | Witwolf: There is no dialplan.conf and forget about trying to implement it in FreePBX. |
| 21:42.46 | *** join/#asterisk quidpro (n=quidpro@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 21:43.00 | Witwolf | About to. On 56K Modem at home! Very slow. Cant wait until our exchange gets DSL! |
| 21:43.23 | Kritter | it addresses your issue on your choice of implementations. |
| 21:44.28 | quidpro | Hmmm, did shared line appearance ever show up in Asterisk? |
| 21:44.51 | Witwolf | Thanks Fender, Did not get that page before. Will help me lots. New to *. |
| 21:45.48 | [TK]D-Fender | Witwolf: Which page? |
| 21:46.02 | Kritter | I suspect the one I sent him. |
| 21:46.03 | Witwolf | Forums. |
| 21:46.21 | [TK]D-Fender | Witwolf: Actually, that won't help you with * at all.... just FreePBX. |
| 21:46.41 | Witwolf | Sorry! |
| 21:46.52 | Witwolf | Too much info! |
| 21:47.12 | Bobthehunter | let me rephrase for some... a pcmcia card zapable for asterisk |
| 21:47.29 | Witwolf | Do anyone know of a sip of IAX gateway to test you internet connection? |
| 21:47.45 | Witwolf | Without subscribing to anyone? |
| 21:48.11 | [TK]D-Fender | Witwolf: I believe FreeWordDialup has an IAX2 connector. |
| 21:48.14 | quidpro | Witwolf: www.freeworlddialup.com |
| 21:48.18 | quidpro | Ayy beat me to it |
| 21:48.29 | Witwolf | LOL |
| 21:48.49 | cervi | Witwolf: Try sip:enum-test@sip.nemox.net |
| 21:48.59 | Witwolf | Internet = South Africa = Sucks! |
| 21:49.12 | cervi | Witwolf: Or with ENUM +43 720 0101011 |
| 21:49.16 | [TK]D-Fender | Witwolf: "too much info"? What... that the FreePBX forums won't help you learn about * at all? How is that little tidbit of info "too much"? |
| 21:49.44 | [TK]D-Fender | cervi: He just specifically asked for *IAX2* |
| 21:49.46 | Bobthehunter | or do you mean any modem car d works |
| 21:50.05 | Witwolf | LOL, sorry, I mean everything I am dealing with at the moment. Not Trixbox. |
| 21:50.06 | elriah | Hey guys, with a TDP400P w/4FXO ports and asterisk 1.2.12.1, what types of conditions would cause calls to just randomly drop? I have an install that, for no apparent reason and no apparent pattern, drops calls here and there. |
| 21:50.15 | [TK]D-Fender | Bobthehunter: *NO* |
| 21:50.28 | [TK]D-Fender | elriah: "callprogress=yes" |
| 21:50.38 | ManxPower | elriah: busydetect=yes or callprogress=yes. Both options should be renamed randomlydisconnectcalls=yes |
| 21:50.38 | cervi | D-Fender: Oh, you're right |
| 21:51.08 | Bobthehunter | so is there a solution ? for laptop to pstn ? |
| 21:51.11 | elriah | [TK]D-Fender / ManxPower: yea, I've been down that path. Thanks,I'll see if this install has those options enabled. |
| 21:51.13 | Bobthehunter | like usb adaptor or something |
| 21:51.20 | Bobthehunter | not ATA please ;) |
| 21:51.37 | [TK]D-Fender | Bobthehunter: SIP gateway or AstriBank |
| 21:51.59 | Bobthehunter | just need 1 channel |
| 21:52.01 | Witwolf | cervi Do I add it in the * config somewhere? Sorry very new to this. |
| 21:52.05 | Bobthehunter | like a X100P for laptops |
| 21:52.07 | [TK]D-Fender | Bobthehunter: TFB <- |
| 21:52.25 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:52.33 | [TK]D-Fender | Bobthehunter: You are what's classed as an "insignificant demographic" |
| 21:52.36 | ManxPower | Bobthehunter: there are no PCMCIA Zaptel cards that I am aware of. |
| 21:52.40 | Bobthehunter | yes i know |
| 21:52.41 | Bobthehunter | ;) |
| 21:52.51 | *** join/#asterisk mega (n=mega@217.200.37.65) |
| 21:53.57 | [TK]D-Fender | Bobthehunter: So see above ^ |
| 21:53.57 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 21:54.02 | *** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar) |
| 21:54.10 | Bobthehunter | hmm sais astribank is fxs |
| 21:54.30 | aydiosmio | is there a "session" collection associated with a call? can I, for example, store a value someone dials that will persist through context changes or can be passed to AGI? |
| 21:54.39 | Bobthehunter | ah darn |
| 21:54.43 | Bobthehunter | 600$ lol |
| 21:54.48 | *** part/#asterisk mega (n=mega@217.200.37.65) |
| 21:54.55 | Bobthehunter | ill try something else.. maybe a pci to pcmcia adaptor |
| 21:55.15 | [TK]D-Fender | Bobthehunter: Has FXO as well IIRC, but thats before you got too picky on PRICE :) |
| 21:55.29 | Bobthehunter | lol |
| 21:55.52 | Bobthehunter | <PROTECTED> |
| 21:55.56 | Bobthehunter | ah |
| 21:56.18 | [TK]D-Fender | Bobthehunter: Not sure, but I think I HAVE seen PCMCIA>PCI converters around, but there is a severe risk concerning IRQ requirements, and the load order to have zaptel start properly. You are on a severly bad path. |
| 21:56.39 | ManxPower | Bobthehunter: it appears that you are setting your requirements to ensure failure. |
| 21:56.55 | Bobthehunter | yep |
| 21:57.00 | *** join/#asterisk edguy3 (n=edguy3@host-24-225-134-24.patmedia.net) |
| 21:57.01 | Bobthehunter | well.. anysolution is good.. |
| 21:57.29 | Bobthehunter | got a x100p layin arounf.. a laptop .. and .. no idea how to make both work ;) unless the x100 could run in windows of course |
| 22:01.33 | cervi | Witwolf: you are looking for IAX, so it would'nt help you |
| 22:02.10 | Witwolf | No problem. Think I am off to bed! Thanks for the help |
| 22:03.06 | [TK]D-Fender | Bobthehunter: Linksys SPA-3102 is still your best bet |
| 22:05.46 | quidpro | Hmm, what's people opinions about the Linksys or Aastra phones... looking at the 942 and the Aastra 9133i |
| 22:07.50 | [TK]D-Fender | quidpro: You're located where? |
| 22:08.06 | ManxPower | Bobthehunter: You can fight Asterisk's oddities and have a miserable life and hate Asterisk, or you can accept Asterisk's oddities, live a happy life, and love Asterisk. It is your choice. |
| 22:08.30 | [TK]D-Fender | quidpro: I'm Guessing Ontario.... |
| 22:12.06 | quidpro | TK: Sorry, Toronto. |
| 22:12.57 | Juggie | [TK]D-Fender, rogers is more then just ontario. |
| 22:13.02 | Juggie | he is in the center of the universe. |
| 22:13.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:13.29 | quidpro | Manx: Haha.. don't blame Digium for not having a PCMCIA Zaptel card.. blame the laptop manufacturers who don't have PCI slots. :) |
| 22:13.37 | *** join/#asterisk apardo (n=apardo@87.217.145.219) |
| 22:13.54 | Bobthehunter | hmm you |
| 22:14.22 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:14.43 | [TK]D-Fender | Juggie: You're jsut jealous because I'm right ;) |
| 22:14.50 | quidpro | hehe |
| 22:14.55 | [TK]D-Fender | quidpro: Forget BOTH of those phones. |
| 22:15.14 | Bobthehunter | bon cop bad cop |
| 22:15.22 | Bobthehunter | that my 2 cents on soem regions |
| 22:15.59 | quidpro | TK: Only reason I am particular to the Aastra is that I am replacing a legacy Norstar 3x8, and the employees at my relative's business aren't exactly the brightest people in the world. |
| 22:16.05 | [TK]D-Fender | quidpro: In order here's the pecking order : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-94X. |
| 22:16.13 | [hC] | Do you guys have any idea how i can limit users on polycom 501's to only be able to take one incoming call at a time (no call waiting, a second call goes to VM) but things like 3 way calls, transfers, etc, still work? (taking up two channels) ... the call-limit feature didnt work for this, because it broke 3 way calls etc |
| 22:16.40 | [hC] | If i use the calls per line key on the polycom, will 2 simultaneous originated calls from the phone (ie 3way call) still work? |
| 22:16.44 | [hC] | off of a single line |
| 22:16.55 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 22:17.01 | [TK]D-Fender | [hC]: linekeys="1" callsperlinekey="1" |
| 22:17.19 | [hC] | [TK]D-Fender: yeah... and a 3 way call still works in that scenario? |
| 22:17.24 | [TK]D-Fender | [hC]: Off that, yes, you can originate 3-way / transfer |
| 22:17.32 | [hC] | Perfect, all i needed to know |
| 22:17.35 | quidpro | TK: Who gives the best deals on phones in Canada? |
| 22:17.53 | Bobthehunter | waht do you need ? |
| 22:18.07 | [TK]D-Fender | quidpro: Depends... I don't know them all, but call up CCP (www.ccpin.com) for Polycom pricing. and check out : |
| 22:18.27 | [TK]D-Fender | http://www.canadianvoipstore.com/home.php |
| 22:18.44 | [TK]D-Fender | http://www.voipdepot.ca/index.php?main_page=index&cPath=1 |
| 22:18.46 | [TK]D-Fender | and |
| 22:19.00 | Bobthehunter | ioh phones n/m |
| 22:19.02 | [TK]D-Fender | http://www.voipware.ca/ |
| 22:19.05 | Bobthehunter | we dont ship to ontario |
| 22:19.06 | Bobthehunter | ;) |
| 22:19.20 | Bobthehunter | .voipware.ca sounds like vaporware lol bad biz decision |
| 22:19.32 | quidpro | I was looking at Voipware... prices are much better than Canadian Voip Store.. and no border hassle. |
| 22:19.54 | quidpro | I wish there was a B&M place I actually go to try these phones out. |
| 22:20.04 | [TK]D-Fender | Bobthehunter: And you're looking to hack a X100P into a PCMCIA>PCI adaptor for processing calls. Talk about the pot & the kettle.... |
| 22:20.16 | Bobthehunter | hehehe |
| 22:20.22 | Bobthehunter | for dev purposes |
| 22:20.24 | Bobthehunter | and fun |
| 22:20.33 | [TK]D-Fender | Bobthehunter: Oh... you mean MASOCHISM |
| 22:20.35 | Bobthehunter | nothing serious about it other then you taking it seriousely ;) |
| 22:21.16 | Bobthehunter | just testing soem thing im writing ATM.. and need that.. but i amnaged something else.. |
| 22:21.49 | quidpro | The Polycom 301 falls into the price bracket I am looking... but no full-duplex speakerphone |
| 22:22.05 | ManxPower | quidpro: what about the Polycom 430? |
| 22:22.34 | quidpro | Hmm, let me look that one up |
| 22:22.56 | [TK]D-Fender | quidpro: Unless you're planning for PoE, get the IP 501 instead. +/- $200 is you shop around |
| 22:23.10 | ManxPower | [TK]D-Fender: I assume you have some 430s? |
| 22:23.22 | [TK]D-Fender | ManxPower: I have everything except the 4000 |
| 22:23.43 | [TK]D-Fender | ManxPower: and I HAVE configured one for a client :) |
| 22:23.57 | quidpro | TK: 430 is ~ 210CAD...... |
| 22:24.06 | ManxPower | [TK]D-Fender: we put about 8 of them in service last week |
| 22:24.13 | quidpro | 501 is $240 CAD |
| 22:24.21 | quidpro | That's from voipdepot |
| 22:24.55 | [TK]D-Fender | ManxPower: Don't get me wrong, the IP 430 is PLENTY enough for most uses, but the 501 is worth the extra 20$ |
| 22:25.00 | [TK]D-Fender | quidpro: Call CCP. |
| 22:26.02 | quidpro | TK: Thanks.. will give them a shout tomorrow... going wrap up here in a lil' bit |
| 22:26.09 | [TK]D-Fender | ManxPower: Unless you're factoring in the included PoE, which is where it swings in the IP430's favour. But the bigger screen, stronger base of the IP 501 plus considerably larger screen ( matters with MicroBrowser a lot, as well as basic call handling) |
| 22:26.55 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 22:27.12 | [TK]D-Fender | ManxPower: Depends how you want to value things of course. |
| 22:27.33 | quidpro | How are the Polycoms though in terms of user friendliness... in other words, are they almost idiot proof? |
| 22:28.51 | [TK]D-Fender | quidpro: Nothing is idiot-proof, because we all know how gosh-darned clever idiots can be.... |
| 22:29.19 | [TK]D-Fender | quidpro: Lets just say the screen and buttons are very clear about what they are doing, the printed documentation is nice. |
| 22:29.24 | quidpro | That's the truth. |
| 22:29.53 | [TK]D-Fender | quidpro: Or worse... you find out you were actually dealing with MORONS, which is a whole other ball-game |
| 22:30.01 | quidpro | Yeah, maybe I should pick one up and see what the "users" think of it... worst comes to worse, I've got myself a nice new deskphone |
| 22:34.42 | *** join/#asterisk jero (n=jerome@modemcable046.23-81-70.mc.videotron.ca) |
| 22:35.27 | jero | hi |
| 22:35.55 | quidpro | TK: Ouch, called CCPIN for the heck of it (Vancouver office)... ~ $360 she quoted me. |
| 22:36.00 | quidpro | For IP-501 |
| 22:36.13 | [TK]D-Fender | quidpro: BS. she's an idiot |
| 22:36.24 | [TK]D-Fender | quidpro: I get my IP 601's for less than that |
| 22:36.45 | [TK]D-Fender | quidpro: As someone else. |
| 22:36.47 | [TK]D-Fender | ask* |
| 22:36.48 | jero | is there a way to trap the end of an agent (from a queue) at the end of the communication, in order to do some action at the end of the call ? |
| 22:37.17 | quidpro | TK: Yeah, maybe I made the mistake of saying "yes" to the "are you an end user" question |
| 22:40.57 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-80-108.vic.bigpond.net.au) |
| 22:40.58 | *** join/#asterisk Sputtering (n=Keelan@192.197.213.245) |
| 22:41.13 | Sputtering | does anyone here have any experience with fxotune? |
| 22:42.06 | Sputtering | more specifically, the effect of rxgain and txgain settings on it |
| 22:48.14 | [TK]D-Fender | quidpro: Winston Zeddemore: Ray, when someone asks you if you're a *God*, you say "YES"! |
| 22:50.10 | quidpro | TK: Winston Zeddemore? |
| 22:51.00 | quidpro | Oh.. ghostbusters |
| 22:51.08 | quidpro | hehe |
| 22:52.15 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:55.19 | quidpro | Hmm, anybody worked with a Valcom paging unit? I am thinking I am going to need an FXO card for it.. or is it FXS? |
| 23:03.28 | Qwell[] | quidpro: where do you normally plug it in? |
| 23:03.51 | [TK]D-Fender | quidpro: Won't need valcom. Using Polycom's you can just page through the phones. |
| 23:04.18 | [TK]D-Fender | quidpro: And you can get differnt models for Valcom, being FXO or FXS depends on what your system supports |
| 23:06.09 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 23:06.39 | errr | how can I simulate a call coming in from the outside to test if some time conditions are working or not? |
| 23:07.48 | osas | hi all :-) |
| 23:07.49 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-6.central.biz.rr.com) |
| 23:08.13 | osas | Is there a way to answer a call with 183 in SIP? (early media) |
| 23:10.30 | *** join/#asterisk luke-jr (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d) |
| 23:11.07 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 23:19.45 | VeNoMouS_ | lol |
| 23:19.46 | VeNoMouS_ | <PROTECTED> |
| 23:19.46 | VeNoMouS_ | Segmentation fault |
| 23:19.48 | VeNoMouS_ | gg svn |
| 23:23.55 | J4k3 | always blame hardware |
| 23:23.59 | J4k3 | its more fun to replace. |
| 23:25.43 | VeNoMouS_ | lol |
| 23:30.56 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 23:30.59 | syzygyBSD | hey all |
| 23:31.18 | syzygyBSD | what is the easiest way to get a recording from Allison now? |
| 23:32.01 | JT | going to the correct digium page i assume |
| 23:33.21 | syzygyBSD | well, I know site I used to use isn't used anymore |
| 23:34.06 | JT | yes it changed |
| 23:36.17 | Sputtering | http://www.digium.com/en/products/voice/ |
| 23:37.14 | |ryan| | This is probably a total n00b question, but I'm not finding an answer via google; How does one set up an extension that starts with a *? |
| 23:37.28 | quidpro | TK: Two Polycom's sitting in a 10,000 sq foot warehouse aren't going to have enough juice to get the page out. :) |
| 23:37.33 | Sputtering | exten => *69,1,PlayBack(blah) |
| 23:37.41 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 23:37.55 | Sputtering | if you're using asterisk-gui, it seems to not want to let you use a *. |
| 23:38.05 | |ryan| | ok, then my pap2 is eating the *. damnit. |
| 23:38.51 | Sputtering | |ryan|, test your theory with a soft phone |
| 23:40.43 | Sputtering | allison's rendering of a british accent is bad. |
| 23:42.18 | JT | logic would dictate one gets a british voice person to do british prompts :) |
| 23:42.36 | Sputtering | i guess logic escapes digium |
| 23:42.52 | JT | shrug, if people are silly enough to buy it... |
| 23:43.53 | *** join/#asterisk zorro___ (n=zorrillo@189.128.82.173) |
| 23:44.03 | zorro___ | Hello |
| 23:44.14 | Sputtering | hi |
| 23:44.35 | zorro___ | i need help whit Sphinx |
| 23:44.36 | JT | i'm wondering if there's a market for redneck australian voice prompts :P |
| 23:44.46 | ManxPower | Sputtering: try the asterisk gui channel |
| 23:44.50 | Sputtering | or wesern canadian prompts, eh? |
| 23:44.51 | JT | i was bored the other day and made some redneck sounding prompts |
| 23:45.02 | zorro___ | <PROTECTED> |
| 23:45.33 | Sputtering | i find that allison's prompts are heartless... not as bas as rogers/att voicemail, but close. |
| 23:45.37 | Bobthehunter | by any chance anyone have the x100p clone windows drivers around ? |
| 23:45.47 | JT | heartless is good |
| 23:45.59 | JT | i love neutral tone |
| 23:45.59 | Sputtering | heartless is creepy |
| 23:46.00 | JT | i hate "excited" voice prompts |
| 23:46.09 | Sputtering | jane barbe (rip) was much better |
| 23:46.12 | JT | a phone system should not have emotion |
| 23:46.22 | Sputtering | i didn'tsay excited, just a little less monotonous |
| 23:46.29 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 23:46.34 | JT | yeah depends how it's done i guess |
| 23:46.39 | Sputtering | i find the monotonous voices aggrivating |
| 23:46.50 | JT | true |
| 23:46.58 | Sputtering | telus has re-done their IVR system with a voice that hastually has some inflection, and it is very pleasing |
| 23:47.07 | Sputtering | hasctually = actually |
| 23:47.16 | JT | okay :) |
| 23:47.34 | Sputtering | okay then. |
| 23:47.46 | JT | i am almost never in american IVR systems |
| 23:47.51 | Qwell[] | I hate voice prompts that are plain wrong |
| 23:48.00 | Qwell[] | Don't read me an email address, if it doesn't exist |
| 23:48.06 | Qwell[] | </rant> |
| 23:48.09 | Sputtering | telus = canadian (very offensive to call a canadian an american!!) |
| 23:48.18 | JT | north american then |
| 23:48.23 | Sputtering | damn. |
| 23:48.29 | [hC] | I didnt realize telus had changed theirs recently |
| 23:48.29 | Sputtering | can't hide from that one. |
| 23:48.31 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:48.37 | [hC] | I should call and observe |
| 23:48.46 | [hC] | Allisons are sometimes a bit too... dramatic/stage voice for me |
| 23:48.50 | [hC] | but overall they're good |
| 23:48.51 | Sputtering | well, recent, as in the last few years -- i only call them when I move |
| 23:49.21 | Sputtering | since they switched to their voice recognition system |
| 23:49.29 | [hC] | nod |
| 23:49.49 | [hC] | Sputtering: you're in kelowna? |
| 23:50.13 | Sputtering | hc: yup |
| 23:50.25 | [hC] | Right on.. I'm in vancouver. |
| 23:50.29 | [hC] | I used to live in kamloops though. |
| 23:50.29 | j0 | <-- abbotsford |
| 23:50.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:50.40 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 23:50.52 | [hC] | j0: know anyone looking for work for a voip company? :) |
| 23:51.01 | j0 | i visit kamloops once a year to goto sunpeaks. :) |
| 23:51.02 | Sputtering | i haven't been to vancouver since 1997... I should get out that way more often! |
| 23:51.04 | j0 | [hC]: nope, sorry |
| 23:51.47 | zorro___ | <PROTECTED> |
| 23:52.01 | Sputtering | what is sphinx anyway? |
| 23:52.35 | *** join/#asterisk notoriousrab (n=robert_m@207.47.34.74.static.nextweb.net) |
| 23:52.53 | zorro___ | <PROTECTED> |
| 23:53.04 | Sputtering | oh. |
| 23:53.18 | Qwell[] | speech recognition... |
| 23:53.21 | Qwell[] | big difference :) |
| 23:53.23 | zorro___ | <PROTECTED> |
| 23:53.49 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:54.30 | [hC] | So, is there a quicker implementation of Page() in 1.4? client is complaining that having to add 80+ phones to a meetme when i do a page() takes like 10 seconds, and thats too long |
| 23:54.50 | zorro___ | i need examples, i read about this page |
| 23:54.50 | zorro___ | http://www.voip-info.org/wiki/view/Sphinx |
| 23:55.00 | zorro___ | <PROTECTED> |
| 23:55.01 | zorro___ | <PROTECTED> |
| 23:55.09 | JT | abbotsford must be a real popular place name |
| 23:55.19 | JT | we have a few abbotsfords in australia too |
| 23:55.41 | [hC] | i think its an english name, we have alot of those here |
| 23:55.48 | [hC] | Langley, Surrey, Abbotsford |
| 23:55.50 | JT | yeah |
| 23:55.52 | Sputtering | we have sydneys in canada to |
| 23:55.53 | [hC] | New Westminster |
| 23:56.12 | j0 | jt: i passed through ABbotsford australia a few months ago.. lol |
| 23:56.24 | Sputtering | Slurrey |
| 23:56.30 | j0 | eheh |
| 23:56.49 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 23:56.52 | j0 | if surrey had waterfront, it'd almost be the same in some areas |
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| 23:58.46 | *** part/#asterisk dasenjo (n=dasenjo@190.24.177.229) |